irclog2html for #asterisk on 20070118

00:00.52rpmi hear this is a bug in asterisk... [Jan 17 16:59:44] WARNING[15243]: chan_sip.c:8023 check_auth: username mismatch, have <lightspeed>, digest has <s>, its matching one of my [lightspeed] entry in my sip.conf and then dieing..
00:01.51*** join/#asterisk arcanine (n=arcanine@203.82.44.179)
00:03.12nick125_lappyIs there a max length of phone numbers on the PAP2?
00:03.49*** part/#asterisk amdtech (i=adaniel@nat/digium/x-404855a2947fc2c8)
00:04.19*** join/#asterisk atlantia (n=scott@cpe-024-088-091-121.sc.res.rr.com)
00:05.09arcanineis there any box similar to redfone?
00:05.29atlantiahi.. just set up my first asterisk system.. with the asterisknow cd.. very very impressive.  using the system to route calls to our cell phones in the field..  I understand i either need two analog lines to handle the routing or i can use an off-hook transfer function of the telco providing the lines, does this sound right?
00:05.36JTarcanine: not that i've seen
00:05.51JTarcanine: i've seen plenty of L1 isdn failover switches
00:05.59JTbut they don't convert to TDMoE
00:07.53nick125_lappyIn the PAP2 dial plan, what's a character that means *ANY* kind of character, * and # included?
00:08.19nick125_lappyx usually only means 0-9
00:08.34JT.
00:08.40JToh
00:08.41JTpap2
00:08.44JTdunno then
00:08.58nick125_lappy. means to repeat the last character multiple times
00:09.23Supaplexthen what's + or *?
00:09.42JTdoesn't the pap2 have any documentation?
00:09.45nick125_lappy* is just a normal character, not sure what + means
00:10.24arcanineJT: so the ratio for a redfone and asterisk would be 1:1
00:10.57arcanineor i can expand another redfone for my existing asterisk server
00:12.43atlantiais my question more built for the forums?
00:13.01atlantiareading those.. seems some topics.. but none really follow through... i'll keep searching
00:13.37JTatlantia: i'm not sure what the off hook transfer function of your telco is
00:13.45JTbut yes, usually you'd need 2 lines
00:14.02atlantiaJT yeah i thinkmy buddy at the telco made it up to sound cool
00:14.05JToutbound lines can be via voip to a voip provider if you have a suitable Internet link
00:14.18atlantiaJT we have a vonage accoutn as well
00:14.22atlantiaaccount*
00:14.33JTonly digital isdn circuits, and then only some of them, can transfer calls using the telco network
00:14.39atlantiacan you use a second line without an analog card? we have 1 card right now, with 1 port
00:14.52JTsounds like an X100P
00:14.54atlantiaJt understood
00:14.58JTnot very good for business
00:15.04atlantiaJT lol, yeah
00:15.19atlantiaJT we are going to buy a better digium one once the initial testing etc is done
00:15.32JTok
00:15.32atlantiawe have a small IT shop and want a better way to manage incoming calls
00:15.40JThmm
00:15.57atlantiayeah the x100p was very unimpressive, lloks like someone slapped a chip on a modem card
00:16.18JTactually, it was a sticker and heatsink
00:16.21JTnot a chip
00:16.27atlantialol no kidding!
00:16.30perdheatsinks rock, though
00:16.45Supaplexheh yea. :-p the sticker just makes you feel special.
00:17.05JTatlantia: the main problem is the chipset is discontinued, so all the supposedly new x100p clones coming out now are likely to be rubbish
00:17.17JTmade with inferior chips like factory seconds
00:17.23JTyou may get a good one, you may not
00:17.24perdi got a 'powered by tux' sticker on my workstation, i'm pretty sure it gained a few megahertz
00:17.28JTand they weren't that good to begin with
00:17.45atlantiayeah i had a windows xp pro sticker on my laptop i took that off and it hasn't broken since
00:17.53perdcrazy
00:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
00:18.02*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
00:18.05atlantiaJT... i am looking at new cards right now for production
00:18.07SupaplexI have a box full of clones :-)
00:18.24atlantiaJT is there a way to use the vonage account without the analog port?
00:18.43JTif you do not use your analogue line with asterisk, sure
00:18.52JTi'm not in the us, but i hear vonage is not that good
00:18.59atlantiaI noticed the ability for voip in the providers setup phase, but not vonage specifically
00:19.27atlantiasounds like best bet is to get two analog lines, buy a multi port card, and have at it
00:19.27JTno experience with asterisknow either, isn't it still beta?
00:19.37perdyeah it is jt
00:19.46atlantiait appears so, but the experience was pleasant
00:20.01atlantiaat leats for someone who has never really dealt with pbx systems
00:20.02JTi prefer configuring the files by hand :)
00:20.04atlantialeast*
00:20.10*** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)
00:20.30atlantiaeventually i would too..
00:20.38JTcool
00:22.41atlantiawhats a good recommendation for a multiple port card that inst x100p?
00:23.03atlantiaactually two ports would be about right
00:24.06Supaplexby queue 'member' (queues.conf) do they mean the caller, the agent, or both?
00:24.59Supaplexeg, announce =
00:25.19JTatlantia: TDM400P if you need only 4 ports
00:25.30flendersatlantia: I recently bought 2 TDM04B with 4 FXO modules each
00:25.33JTyou can buy the TDM400P with just 2 FXO modules
00:25.52Supaplexnm, asterisktoft.pdf answers it yet again. yay. pdf > wiki > config comments
00:26.33flendersSupaplex: I bought it over here for 70 AUD, worth every dollar
00:27.23Supaplexflenders: my bad, is that the softmodem or the T1 card?
00:27.30Supaplexthe latter I hope :)
00:27.41*** join/#asterisk DrCron (n=rszasz@2001:470:1f01:ffff:0:0:0:c49)
00:28.30atlantiaJT thanks just ordered one for 75.00 from a reseller
00:28.36atlantiaflenders, thanks as well
00:28.58flendersSupaplex: I meant the book
00:29.01JTatlantia: ordered what?
00:29.13flendersasterisktoft
00:29.19atlantiai am hoping that inb the long haul, our company can learn enough about the asterisk setup to be able to offer it to my customers, currently, i do admin and network support for acompany with a mitel setup, and no offense to anyone, but it's horrible
00:29.26atlantiaJT the wildcard TDM400P
00:29.31flendersI bought the cards with 4FXO modules for 570AUD
00:29.40JTatlantia: it won't come with any modules for $75
00:29.50JTyou need to buy modules for it to do anything
00:30.20atlantiaJT yeah just noticed that.. 140 rather
00:30.25JTflenders: $70 for the book?? sound a bit pricey
00:30.44flendersJT: angus&robertson
00:30.52JTatlantia: you'll need 2 FXO modules going by what you've been saying
00:30.58JTflenders: explains it
00:30.58Supaplexflenders: oh :) I paid $0 to download it.  I helped author bits and pieces of it (*documentation project), but that was long time ago (no idea where my tidbits are at)
00:31.04JTit's like USD$25
00:31.16JTfrom amazon
00:31.22atlantiaJT ok thank you
00:31.30flendersJT: i got it delivered in less than 24 hours
00:31.31flenders:D
00:32.12flendersSupaplex: because of blokes like me and you that we have such good books out there
00:32.12JTflenders: sure, but i'd rather read the pdf while i wait (or while i don't bother to order it)
00:33.20*** join/#asterisk rickead (n=richard@88-96-99-66.dsl.zen.co.uk)
00:33.35flendersJT: lately I can't read much on the screen
00:33.42Supaplex24 hours for snail mail, or 24 seconds to leech it
00:34.15flendersI get home, and start reading stuff on the laptop, I end up falling asleep in minutes
00:34.33JTheh
00:34.43JTi'm more likely to fall asleep reading a book
00:34.53JTprobably something to do with lying on a bed
00:36.13flendersI'd rather drop a book when I fall asleep than dropping my laptop, as I also use the laptop in bed
00:36.14JThah
00:36.14JTlaptops make good fires
00:36.22flenders:D
00:36.47SupaplexI'd rather leave the book where it belongs, and just fall asleep when that time comes.
00:40.34*** join/#asterisk xnon (n=xnon@200.8.31.93)
00:41.26*** join/#asterisk SECGOD (i=SECGOD@c-71-57-36-106.hsd1.il.comcast.net)
00:41.35*** join/#asterisk nahirean (n=nahirean@unaffiliated/nahirean)
00:43.35atlantiacool.. got the whole FXO/FXS thing figured out, in theory.. got a card with two fx0 modules for 235.. sound about right?
00:45.37atlantiai assume a larger system, more lines would require a t1 for the multiple out going calls, etc.  I need to find a good book on the subject
00:45.51atlantiabook*
00:46.22JT~thebook
00:46.28jbotthebook is, like, a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
00:47.17ManxPower~fxofxs
00:47.18jbotfrom memory, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
00:47.23Supaplexgood call (pun intended)
00:47.26ManxPowerit's Eff Ex Oh.  Not Eff Ex ZERO
00:47.40atlantiafxo, ok thanks
00:47.53JTatlantia: a t1 would be nice for lots more lines, not mandatory though
00:47.58JTforeign exchange office
00:48.04Supaplexor parital T1
00:48.55atlantiai need a laser printer.. i don't know howmany more of my future kids i can name HP, but i am gonna print this up anyways
00:49.28Supaplexwith registration marks and all? :-)
00:49.35atlantiamaybe there is a print version, that'd be smart, let me look
00:49.43JTSupaplex: yeah i know, but it's a big step from 2 lines to whatever the minimum is for fractional T1
00:50.07Supaplexyup
00:50.08JTatlantia: it's been mentioned already, yes, there is a print book, it's published by o'riely
00:50.15atlantiaok thank you
00:50.41JTthe pdf is good in that you can search it
00:50.49Supaplexpdf++
00:51.29atlantiaindeed.. i'll get both..
00:51.32JTttp://www.amazon.com/Asterisk-Telephony-Jim-Van-Meggelen/dp/0596009623/sr=8-1/qid=1169081455/ref=pd_bbs_sr_1/103-1821274-3488656?ie=UTF8&s=books
00:52.35flendersatlantia: that's what I did
00:52.36atlantiayeah amazon has a great price
00:53.45*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)
00:53.59atlantiathanks for the advice all.. i'll return someday soon a man wealthier in his knowledge of asterisk.. or a noob asking my next question.. either way,... thanks!
00:56.44Supaplexif queues.conf has persistentmembers = yes, how can I access/enumerate/lookup values that would be stored in astdb for these queues?
01:01.18*** join/#asterisk Nukemizer (n=Nuke@160.7.249.15)
01:02.25nick125_lappyHmm...
01:02.35nick125_lappyThere's got to be a way to get this to work on my PAP2
01:02.48JThave you read the documentation for it?
01:03.10nick125_lappyYeah, I have..
01:03.28JTand it doesn't tell you what all the patterns are
01:03.59*** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au)
01:04.46nick125_lappyI've look in the manual
01:04.52nick125_lappydidn't really help too much
01:05.14ManxPowernick125_lappy: What specifically are you trying to do?
01:06.01nick125_lappytry to be able to dial this number on my PAP2 and pass it to my asterisk box: *585*1NXXNXXXXXX*1NXXNXXXXXX*
01:06.07*** join/#asterisk JoeLlama (n=snork@66-192-6-6.static.twtelecom.net)
01:06.09JoeLlamaHi :)
01:06.29JoeLlamaIs there a really cheap system I can set up now and call and receive calls cheap?
01:06.58ManxPowernick125_lappy: First three links:  http://www.google.com/search?hl=en&q=sipura+dialplan&btnG=Google+Search
01:07.28ManxPowerChances are the * at the beginning is conflicting with some of the built in SIPura featuers.  Disable those features
01:07.39*** join/#asterisk fiber0pti (n=John@207.114.199.107)
01:07.46fiber0ptiIs there a timeout on manager connections?
01:07.52ManxPowerJoeLlama: Yes, it is called "Sykpe" and it does not work with Asterisk
01:08.24perdskype isnt just providing sip?
01:08.29perdhow dirty.
01:08.37ManxPowerperd: no.
01:08.38nick125_lappyManxPower: I'm not sure which features to disable.
01:08.48ManxPowerSkype uses it's own protocol and it's own codec.
01:08.48perdwhat the hell do they provide? some bastardized sip?
01:08.52ManxPowernick125_lappy: AQLL OF THEM
01:08.53perdno kidding
01:08.53perdwow
01:09.08perdway to go skype... .... </sarcasm>
01:09.31ManxPowerperd: Their market is "I want to make and receive cheap calls with my PC"
01:09.41ManxPowerThat is not the market Asterisk is after.
01:10.01ManxPowernick125_lappy: look in the SIPura, disable any feature that starts with *
01:10.07perdseems like it would be more cost effective to use already existing voice protocols/codecs
01:10.16ManxPowerspecifically any feature that starts with *58
01:10.21perdbut im not a multi million dollar company so what the hell do  i know:)
01:11.00ManxPowerperd: No.  By keeping everything closed, they control all clients and servers for the system and do not (in theory) have interop issues.
01:11.54perdyeah so in other words they suck
01:12.20perdi get it!
01:12.25perdneeds more chan_skype
01:12.28ManxPowerI'm sure they are great for the nitche market they want
01:12.31perdno doubt
01:13.26ManxPowerUsing open protocols can drastically lower the cost of the hardware and software, but it ALSO can cause significant interop issues.
01:13.55Supaplexthe great thing about standards ...
01:14.06perdis taht everyone has their own take on them?
01:14.07perdhehe
01:14.15*** part/#asterisk xnon (n=xnon@200.8.31.93)
01:14.18Supaplexpretty much :P
01:14.47perdsuch as microsoft and IE have shown us for many, many years
01:14.52perdahh the good times i've had.
01:14.58ManxPowerYou can reduce interop issues by keeping the number of vendor's equipment involved in a project as low as you can.
01:14.59JoeLlamaso use skype not asterisk?
01:15.18perdi want to cockpunch cisco for being such nazis about their friggen software upgrades
01:15.21ManxPowerFor example we use Polycom Phones, Adtran channel banks, Digium or Sangoma T-1 cards.
01:15.25ManxPowerCisco switches.
01:15.40perdpay 10 grand for service, never use it once, then get cold shouldered when i ask for one damn softawre update
01:15.50ManxPowerSpecificallly Cisco Catalysy 550x switches and Cisco 2621 routers
01:16.01perdi just swapped over to foundry networks switches
01:16.04perdthey are nice.
01:17.28*** part/#asterisk JoeLlama (n=snork@66-192-6-6.static.twtelecom.net)
01:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
01:18.03*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
01:18.51*** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com)
01:20.20nick125_lappyI finally got my little trick to work..
01:20.30nick125_lappyI guess it doesn't like it with multiple asterisks
01:20.40nick125_lappy(x doesn't include * or #, sadly)
01:21.35Strom_Cwhy should it?
01:21.40Strom_CX is 0-9
01:21.58nick125_lappyWell, I need something that includes * and #
01:22.11AtomicStackuse .
01:22.38*** join/#asterisk re-pete (n=chatzill@24.96.201.72)
01:22.43Strom_C. is an indefinite number of characters though
01:22.54JTStrom_C: on a PAP?
01:22.59Strom_Coh
01:23.05Strom_Ci thought we were talking about extensions.conf
01:23.07Strom_Ci lose
01:23.22*** join/#asterisk xnon (n=xnon@200.8.31.93)
01:24.13JTperd: skype has a slight advantage in terms of audio quality in that it uses a wideband codec
01:24.19*** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C)
01:24.54DocHollidayanyone know what results would be obtained with 'Novell Failover' and asterisk?
01:25.22Supaplexcan you simulate it and find out? what's novell failover do anyhow?
01:25.26*** join/#asterisk RoyK (n=roy@217-175-222.100710.adsl.tele2.no)
01:25.55Strom_Cwell, except when you use skype to call the pstn, and then it sounds like hell
01:26.17JTheh
01:26.29JTnever been willing to spend money with them
01:30.01[TK]D-Fendernick125_lappy : [*#0-9]
01:30.02`SeanJT; how are you man
01:30.19JTnot bad
01:31.17*** join/#asterisk Techie-Micheal_ (n=Techie-M@phpbb/support/techie-micheal)
01:31.57nick125_lappy[TK]D-Fender: I hate you.
01:31.58nick125_lappy:P
01:32.36`SeanJt, have you ever used a Cisco IP phone?
01:32.39nick125_lappy[TK]D-Fender: (That means that it did work, thanks)
01:32.49*** join/#asterisk |dennis| (n=dennis@200.32.233.82)
01:32.53JT`Sean: nup
01:35.03*** part/#asterisk variable_office (n=variable@208.73.60.2)
01:39.24*** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net)
01:40.20*** join/#asterisk coppice (n=chatzill@129.168.17.210.dyn.pacific.net.hk)
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01:49.59nick125_lappylol
01:51.07DocHollidayso anyone know if novell failover can be implemented with asterisk?
01:51.15Strom_Chey, 800-free-411 is asterisk :)
01:51.17QwellDocHolliday: novell failrover?
01:51.20Qwellfailover too
01:51.22QwellStrom_C: neat
01:51.29Strom_Callison told me my number
01:51.31Qwellheh
01:51.37*** join/#asterisk Natham (n=mrDak@Dynamic-IP-cr20011877179.cable.net.co)
01:51.46JTallison likes to talk
01:51.46ManxPowerStrom_C: Don't count on it, Allison does voices for many IVR systems.
01:51.57nick125_lappyStrom_C: They should offer IAX and SIP to 800-free-411 :P
01:52.11Strom_CManxPower: perhaps, but those are the asterisk 1.2 samples of 0-9
01:52.24ManxPowerStrom_C: Ah, then you are prolly right.
01:53.48Qwelland free411.com is static HTML running on astman
01:56.25ManxPowerhow does 800-FREE-411 make any money
01:56.47rudholmJT: isn't that the point of toll-free service?
01:56.50ManxPowerJT: And we americans thing it's funny that you have to pay much, much more to call a mobile .vs. landline
01:57.13nick125_lappyManxPower: advertisements
01:57.22Strom_CManxPower: you have to listen to an irritating advertisement
01:57.26JTManxPower: sure, but a mobile network is much more expensive to provide
01:57.36rudholmyep, and we get to call them for "free"
01:57.39rudholm:-p
01:57.40ManxPowerJT: the mobile customer should pay that
01:57.45JTrudholm: ah yeah, i wasn't talking about that though, the call receivers pay for 1800 here too
01:57.47rpmin a SDP header i see a=rtpma\000\000!\000\000\000/lib/tls/libnss_files.so.2\000\000!\000\000\000/lib/tls\000libnss_files.so.2\000\377!\000\000\000/usr/lib/libdb3.so.3\000\000\000\000l-\031\010!\000\000\000/usr/lib\000libdb3.so.3\000\000\000\000\000\000\000\000\021\000\000\000\3704\031\010\370.. Does this mean something is broken :P
01:57.48JTManxPower: lol
01:58.05rudholmJT: also, a line charge would constitute paying for inbound calls
01:58.12rudholmJT: technically
01:58.15JTgenerally the person making the call is the one responsible for the cost
01:58.33JTrudholm: well that's paying for the line, but if you want :)
01:59.39ManxPowerWe Americans also think it's funny that most of the rest of the world has to PAY PER MIN for a local call.
01:59.57rudholmI have a POTS line that costs me 5$/month and I get free inbound calls.
02:00.02JTweird
02:00.12JTfor the most part we don't
02:01.02Nivexrpm: that definitely doesn't look good.  1.2 or 1.4 ?
02:01.14*** join/#asterisk pat_lehem (i=lehem@bzq-88-152-186-83.red.bezeqint.net)
02:02.40ManxPowerWhy can't VoIP providers offer 20 DIDs for $5/month like the telcos do?
02:02.43*** join/#asterisk RoyKa (n=roy@217-175-39.100710.adsl.tele2.no)
02:02.54JTdunno
02:02.59JTnoticed that problem here too
02:03.05Strom_Cbecause the telcos make ludicrous amounts of money from the ISDN circuits?
02:03.08JTDIDs are cheaper over BRI/PRI than voip
02:03.46nick125_lappyHmm...this is weird
02:03.52[TK]D-FenderJT : lol..... what do you thing VoiP TERMINATES on?
02:04.07JT[TK]D-Fender: i know what it terminates on
02:04.09nick125_lappyI can transfer calls (though #) and do assisted transfers (though *2), but, I can't park a call (though *9)
02:04.14JTvoip providers are often telcos though
02:04.15[TK]D-FenderPRI!  whee
02:04.21JTso shouldn't pay the same as a consumer
02:04.38ManxPowerI got a quote today for 60 DIDs at $15/month for all 60
02:04.53JTi think voip provders charge a premium because it provides convenience to people who get DIDs from areas outside their local area
02:05.01*** part/#asterisk pat_lehem (i=lehem@bzq-88-152-186-83.red.bezeqint.net)
02:05.17ManxPowerThat excluded the local loop, of course.
02:05.37Strom_CJT: exactly, ITSPs see numbers as a high-margin item
02:05.57Strom_Cwhereas telcos see numbers as a way to encourage more traffic on the circuit
02:06.37ManxPowerStrom_C: We never pay for incoming DID calls on our telco circuits
02:06.51ManxPowerOr do you mean "get customers to order more circuits"?
02:07.17Strom_CManxPower: the telco receives termination fees when people call in from elsewhere
02:07.25Strom_Cthat too
02:08.45nick125_lappyYay, 3 way calling actually works
02:09.00ManxPowerOne would assume that any ITSP worth anything would also get termination fees from the calling telco
02:11.05nick125_lappyHmm..anyone here have any ideas on how to repark a call that has already been parked (and picked up though 701)?
02:12.28nick125_lappyIt just doesn't seem to work (TM)
02:12.53Techie-Micheal_Hrm. I really should take TM off my highlight list ... :P
02:13.02nick125_lappylol
02:13.17Techie-Micheal_Speaking of not working, I still can't seem to register my SIP phone to the server. :(
02:13.31JTThat'd be a plan. (TM)
02:13.35ManxPowernick125_lappy: what version of Asterisk?
02:13.40nick125_lappyManxPower: 1.4.0
02:13.45ManxPowerSorry, Asterisk(R)
02:13.50nick125_lappyHaha
02:13.51ManxPowernick125_lappy: several things are broken in 1.4.
02:14.02Techie-Micheal_You did that on purpose. :P
02:14.42nick125_lappyManxPower: I've noticed that
02:14.52ManxPowernick125_lappy: Why are you using a .0 release?
02:15.20JTLies. (TM)
02:15.30nick125_lappyManxPower: No idea really.....
02:15.51perdyeah, i upgraded to 1.4 thinking 'whooaaaaa AWESOME!' then two days later i downgraded
02:16.00perdworst day of my life.
02:16.10nick125_lappyI don't want to downgrade my asterisk system :/
02:16.11rpmNivex: its 1.4.0
02:16.13ManxPowerI'll use 1.4 when Digium uses 1.4 on their corporate production PBX.
02:16.25NivexManxPower++
02:17.26rpmhow do i generate a md5secret for a SIP 407 for proxy authentication, when i dial i have to authenticate to my proxy, but i dont know what realm to use.. username:realm:secret
02:17.26[TK]D-FenderManxPower : "Do ask I say, not as I do" (Boy-scout leader proceeds to use lighter fluid to start HIS fire)
02:17.33JTthe .0 is a little (read big) clue that it's probably not that great for something important
02:17.35DocHollidayTechie-Micheal_, might want to drop the Techie :D
02:17.39DocHollidayoh woops
02:17.41DocHollidaywas scrolled up, sorry
02:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
02:18.02*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
02:18.46Techie-Micheal_DocHolliday: Why's that? Just because I'm having issues with asterisk doesn't mean I don't know what I'm doing when it comes to other things. ;)
02:19.03Techie-Micheal_Besides, I did that years ago when I got my A+ cert. *shrugs*
02:19.11DocHollidayTechie-Micheal_, oh of course.. i wouldnt doubt your intelligence like that.
02:19.23DocHollidayi have complete faith in your abilities :)
02:19.51coppiceHe's been certified. :-)
02:20.40DocHollidayand he got an A
02:20.46*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
02:21.25[TK]D-FenderDocHolliday : A PLUS
02:21.43JTi've got two pluses
02:21.50JTi'm certified JT++
02:22.04Supaplexoh great, another C++ clone gone bad
02:22.07DocHolliday[TK]D-Fender, LOL i feel bad for making fun of him like this.
02:22.24DocHolliday[TK]D-Fender, have you ever used Novell eDirectory?
02:22.51[TK]D-FenderDocHolliday : Nope, and I will celebrate the exorcism of Novell 5.0 from my work file server this year...
02:23.16nick125_lappyDoes anyone know if this issue I'm having is specific to 1.4.0 or if its a general asterisk issue?
02:23.46[TK]D-Fendernick125_lappy : what issue?
02:23.49DocHolliday[TK]D-Fender, im actually  looking forward to Novell eDir, ZEN and groupwise
02:24.25ManxPowernick125_lappy: We don't park calls using * codes (I think it's sloppy, silly, and not needed ever), but if there was a problem with reparking a call using the device native transfer feature, I'm sure I would have heard about it by now.
02:24.51ManxPower[TK]D-Fender: he can't park a call a 2nd time using * transfers
02:25.08filegrab 1.4 from SVN, that was fixed
02:25.49ManxPowerfile: Your IT people need to be spanked for not installing 1.4.0 on the main Digium production PBX
02:26.05ManxPowerbefore it was released.
02:26.08fileI've counted how many times you have said that since you started
02:26.10Sweeperthis channel needs a bot that suggest names for telephony providers
02:26.13fileand the number is getting rather huge
02:26.35[TK]D-Fendernick125_lappy : What kind of phone?
02:26.59SupaplexSweeper: I'd suggest localhost, but their support can't teach me anything new. ;)
02:27.22SweeperSupaplex: no, I mean, suggest names for people looking to start their own provider service :P
02:27.33ManxPowerfile: and yet Digium is not confident enough to do so.
02:28.23SweeperManxPower: maybe it's because they're working on implementing 1.5, to install it before release :o
02:28.34QwellMisery already runs 1.8
02:28.43Qwelloops, forget I said that
02:28.49perdcan i get the pre pre alpha sir
02:29.00perdmaybe chan_skinny works
02:29.07FuriousGeorgeso i was just reading up on debugging my deadlocks.  my test system never deadlocks, and when a production system does I have to get it working again right away.  i can tell people to hold on while i attempt a backtrace
02:29.10Qwellperd: flawlessly
02:29.14perdi knew it.
02:29.20coppiceSweeper: You mean things like Crap-o-tel, or Drop-yr-call?
02:29.32perdfriggen digium, they made asterisk just to waste months of geeks time
02:29.35perdi bet ti doesnt even work.
02:29.45Sweepercoppice: tempting, but I know from experience that customers have no sense of humor
02:29.48ManxPowerFuriousGeorge: Almost all issues with Asterisk I have these days (and there are not many) happen on production systems as well.
02:29.54perdonce you figure it out you're suddenly blessed with the realization that asterisk in fact does nothing.
02:30.07FuriousGeorgeManxPower: are you using 1.4 or 1.2.x?
02:30.16perdthat's why this # has such a high turn over, right? because people find otu and kill themselves
02:30.28JT#?
02:30.45perdpound, channel, hash, whatever you like
02:30.50JTyeah you can never test asterisk like a real production environment can
02:30.56JTah
02:30.59ManxPowerFuriousGeorge: 1.2 of course.
02:31.04fileyou can never anticipate what a user will do
02:31.07JTi thought you meant *
02:31.13SupaplexI call it the tic-tac-toe marker.
02:31.13perdnah, jt :)
02:31.30perdsupaplex that's a definition with a bit too many syllables for my taste
02:31.43perdi like simple things, like curse wods.
02:31.52[TK]D-FenderWaffle :)
02:32.01ManxPoweroctothorpe is the correct term, I believe
02:32.12Nathamhi, does an intel 536ep modem based works with asterix?
02:32.25perdhttp://en.wikipedia.org/wiki/Octothorpe
02:32.28Strom_Cbeats me; i've never used asterix
02:32.31perdi did not know that, but it's a wicked cool name.
02:32.41ManxPower#*#*#*
02:32.50Nathamasterisk
02:32.51perdthat's one hell of a tounge twister
02:32.52ManxPower^^^ swatting flys
02:33.06perdoctothorpeasteriskoctothorpeasteriskoctothorpeasterisk
02:33.08perdahhh
02:33.10SweeperJellyTel!
02:33.16Supaplexvoice prompt "Please press octothorpeasteriskoctothorpeasteriskoctothorpeasterisk to continue"
02:33.24perdhaha
02:33.43perdthat would be awesome, someone needs to pay the voice of asterisk to say octothorpe
02:34.56Supaplexto make it more interesting, randomly state any of "octothorpe, hash, pound, number sign, tic-tak-toe key" throughout the call when navigating menus.
02:35.21perdhaha
02:35.43perdwe should write a 'how to piss people right the fuck off with your new Asterisk PBX!'
02:36.03ManxPowerperd: That is under the Telemarketer Torture page on the Wiki
02:36.07perdalternating delete and next for the voicemail menu would be good too! set it up to randomly decide
02:36.08nick125_lappyHahaha
02:36.13JTit wouldn't be a big book, it'd just say "install ultra beta version in big corporate environment"
02:36.16nick125_lappyManxPower: I read that, it is so funny.
02:36.25perdhaha jt
02:36.40perdhmm i need to check out this torture page
02:36.42SupaplexI wonder where that wavfile of two netzero support agents bridged to each other is at.
02:37.09ManxPowerI would love to hear that .WAV file.
02:37.27perddude i love those prank calls where they call up dominos and papa johns and conference them in
02:37.27FuriousGeorgeManxPower: ive tried two different computers, the only thing that has helped is rebooting asterisk nightly.  i just got another one today after maybe 4 months.  i've read that asterisk 1.4 has "many many bugfixes" according to digium.
02:37.28FuriousGeorgeManxPower: even once every 4 months is too much
02:37.34Supaplexeach assumed a call was from the outside world, and both attempted to assert control of the call asking for an account or creditcard  number.  Lasted far longer then it should have, and I've busted my gut over and over on it.
02:37.43ManxPowerFuriousGeorge: The only systems we have to reboot regularly are ones with TDM400Ps in them.
02:38.19JTFuriousGeorge: this problem that came up after 4 months, was it on a system that rebooted daily?
02:38.28ManxPowerBut since they are production systems we can't bring them down for testing to try to reproduce the problem (only happens after a fairly large number of calls)
02:39.05DocHollidayhmm my new phone number used to be owned by a 'spa' AKA Prostitute order-line.. i get people asking what i charge / what kind of services i can do
02:39.19perdwow doc
02:39.28perdi dont think they have phone #s here
02:39.31perdthere are so many of them
02:39.41rpmwell, what kind of services can you provide :P
02:39.46DocHollidayhaha
02:39.55perddoc really knows how to work the balls
02:39.57DocHollidayrpm, i do reformats and grave digging
02:40.09perdi gave him an A++++++ whore'er on shebay
02:40.26JTDocHolliday: you need an IVR
02:40.28DocHollidayheh
02:40.34DocHollidayJT, yeah your right :P
02:40.51JTDocHolliday: "if you are calling for special services, dial 1" 1 "diverting to pizza hut"
02:41.01DocHolliday'Callers please note the sex hotline has changed.. the new number is ..., for all other callers.. stay on the line
02:41.12rpmtheres no way to mask passwords in sip.conf if i gotta reply to a invite (sip 407 proxy auth.) or in my register => to the proxy?
02:42.11perdrpm, yeah, chmod 600
02:42.12SupaplexDocHolliday: setup an extension to record their number, and always direct them into the ivr from hell after they've paid or something :p
02:42.35SupaplexDocHolliday: I'm sure you can get creative. "Oh yea, hold on... " :-d
02:42.40perdalternatively you could use 'md5password'
02:42.41DocHollidaySupaplex, problem is i actually want to use the line :)
02:42.47perdwhich allows you to specify a prehashed password (i believe)
02:43.01perdnot sure how that works with register, though
02:43.12perdgurus will know.
02:45.18ManxPowerrpm: if your sip.conf is readable by others you are already screwed.
02:45.47rpmit is 0600, but other people do have access to this system. i'd prefer to keep my passwords protected.
02:46.35perdif they have root you cant, really
02:46.39nick125_lappyHmm, this is weird
02:46.53perdsecure your box
02:46.57perdget the rifraff out
02:47.14rpmi'll just use selinux extensions
02:47.36DocHollidaywhere oh where could nem be
02:47.51nick125_lappyfor some reason, when a call comes in, the screen lights up, but, it doesn't actually ring outloud
02:47.58JTrpm: why do they need read access to sip.conf?
02:48.08perdturn the speaker on, nick!
02:48.16nick125_lappyperd: speakerphone and such works
02:49.18rpm6JT, they don't.
02:51.05JTrpm: hrm, what's the issue then?
02:51.48perdsounds like 'bad people' have root on his server
02:52.17nick125_lappythe base rings...this is odd
02:52.27Strom_Cwould those be the evil hackers the local news is always going on about?
02:52.45perdstrom, yeah, they're getting out of hand
02:52.54perdi saw one the other day hiding behind my refrigerator
02:53.23ManxPowerperd: Prolly snorting freon
02:53.25perdhad to swat at him with a broom till he left, bastard stole some packets from my condiment drawer
02:55.54*** join/#asterisk Guest^DJ (n=me@espeed24-92.brunet.bn)
03:03.58*** join/#asterisk mog (i=ejabberd@71.207.215.93)
03:03.58*** mode/#asterisk [+o mog] by ChanServ
03:05.05perdhahah oh man that telemarketer torture is awesome
03:05.10perdim totally using that.
03:05.36nick125_lappyYay, call waiting works as well
03:06.42perdhigh five
03:07.23coppicethe only telemarketer torture I'm interested in involves racks and red hot pokers under the fingernails
03:07.49*** part/#asterisk Guest^DJ (n=me@espeed24-92.brunet.bn)
03:08.03JTdon't hate the player, hate the game
03:08.45coppicethe game only exists because the players play
03:09.30JTbut the players are the companies and bosses, the telemarketers themselves are just there to try and earn a dollar or two
03:09.36JTbefore they get burnt out and quit
03:09.55coppiceyeah, and their bosses will claim the same
03:10.30JTsure, but most telemarketers who man the phones actually hate their job
03:10.40Supaplexfor good reason :)
03:10.42Strom_Cyou know, if no one in this country bought anything from a telemarketer for a year, the whole industry would die out
03:10.45coppiceIts a matter of basic fairness. If someone wakes me up at 3AM in a hotel room on the far side of the planet, I should be able to hear them squirming in agony
03:10.56Supaplexsame thing for spam.
03:10.57JTheh
03:13.39flendersfuck, it's been sooo hard to filter spam here
03:13.55flenderslately we're getting a few thousands a day
03:14.56flendersI think spammers are even worse than telemarketers... as telemarketers are trying to sell you something... most spam, is just rubish, not real
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03:16.31coppicespam doesn't wake me up at 3AM
03:16.44FuriousGeorgeJT: sorry for the delay in responding, asterisk is restarted daily, but the system is persistently on
03:17.43FuriousGeorgeManxPower: and yes the system has a tdm400p in it
03:17.44robl^coppice: spam wakes me up at 3:00AM!!  damn, you RIM and your Blackberry!
03:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
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03:19.06FuriousGeorgeJT: maybe i'll try rebooting weekly and see if i get it down to once a year.  i'm always a little scared to cron a reboot of the system for fear it wont come back
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03:26.22JTheh
03:26.22coppiceIf you have a blackberry, being woken at 3AM soon becomes a minor issue :-)
03:26.57JTflenders: oh, the rubbish from spam is real alright
03:27.06dongcJT: Hi, i managed to make the TE405p LEDs light up. with loopback connector the LED turn green.
03:27.30dongcJT: zap show status, alarm shows "REC", is this normal?
03:27.45DocHollidaycoppice, blackberry + lead-lined container
03:27.49DocHolliday'I swear it was on'
03:28.14JTdongc: yes, asterisk is not designed to work with loopback
03:28.22JTdongc: how did you make the lights light up?
03:28.55dongcJT: reinstall centos 4.4, yum update and install some of the packages. recompile zaptel1.2, libpri and asterisk.
03:28.57JTFuriousGeorge: what was the issue again?
03:29.06JTdongc: ah neat
03:29.08dongcJT: suddenly the lights turn RED.
03:29.12coppice"How long have you been blackberry free?"
03:29.38JTdongc: that sounds much better
03:30.26coppiceDoesn't the Blackberry tend to reduce most e-mails to something almost meaningless?
03:30.28coppice> Doesn't the Blackberry tend to reduce most e-mails to something almost meaningless?
03:30.29coppiceyes
03:30.31dongcJT: yeah. what is this "REC" means? normally when u plugged into an ISDN, what is the alrm shows?
03:30.41JTgreen
03:31.01JTsure it's not RED
03:31.21dongcJT: Ok. i will bring to data center and retest with our Excel switch.
03:31.30JTcool
03:31.47JThave you tried a crossover t1 cable between span 1 and 2?
03:31.52nick125_lappyHmm...
03:32.11dongcJT: yes. Both spans LED turn green. But status still "REC"
03:32.23JThrm
03:32.27JTwhat about in asterisk
03:32.29JTdo they come up?
03:32.42dongcnope. zap show status still shows "REC"
03:33.48nick125_lappyMeh, might as well try 1.2.x and see if it has less issues then 1.2.x
03:34.00JT1.4.x you mean?
03:34.01*** part/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com)
03:34.13nick125_lappyyeah
03:35.49^sandro^good afternoon
03:35.58^sandro^anyone here very familiar with trunk groups?
03:36.09^sandro^and spanmap of course
03:36.16JTyou only need them for NFAS
03:36.24^sandro^yup.. i have to use NFAS
03:36.33JTok
03:36.41^sandro^i have a PRI and today they added another 24 B channels
03:36.48^sandro^thing is i dont know how to join them
03:37.01^sandro^i mean i looked at my config but calls can't be made. i know im doing something wrong... but im not sure what
03:37.48^sandro^so i need a bit of help that's all
03:38.06JTsorry, i can't help other than by reading documentation
03:38.26^sandro^ic
03:38.26^sandro^np
03:38.46Strom_C^sandro^: what happened, man?  I asked you for a pastebin and you disappeared on me
03:38.49^sandro^ya i read docs.. followed them .. but i guess i followed wrong :P
03:38.58^sandro^strom sorry.. i had an emergency come up
03:39.13Strom_Cand you couldn't tell me "I have to go - back later"?
03:39.18^sandro^had to leave my terminal .. i still have the screen but looked and didn't see you speak anymore here so i thought maybe you were sleeping or something
03:39.31^sandro^actually i walkd away for a sec and didn't get back to the computer
03:39.45^sandro^sorry dude.. my fault.. but i could not help this one
03:39.58nick125_lappyDoes anyone here know of call reparking works in 1.2.x?
03:40.01^sandro^but im back and i have all night if i have to .. to figure this out
03:40.11^sandro^im now completely free of distractions
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03:43.39FuriousGeorgeJT: the issue is that i get deadlocks, but only every 4 months or so
03:43.48FuriousGeorgei do restart asterisk daily, but i dont restart the server
03:43.53JTFuriousGeorge: well that sucks
03:44.18FuriousGeorgeyeah, sorry for the extended delay in responses
03:44.52FuriousGeorgeim thinking of upgrading to asterisk 1.4 because according to changes file it has "many many bugfixes"
03:45.10FuriousGeorgeso maybe many bugfixes = less deadlocks
03:46.12JTi'm guessing you can't move the TDM400P
03:46.27FuriousGeorgeto another bay?
03:46.34FuriousGeorgeor to the hardware recycling bin
03:46.44FuriousGeorgei use apic so its not an irq thing
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03:46.50JTcomputer or recycling bin
03:46.57FuriousGeorgei wish
03:47.01EyeCuemornin
03:47.09JTgoing off what others have said, it can be a weak link reliability wise
03:47.51FuriousGeorgei sometimes wonder if i should have gotten the sangoma, but i feel like that would probably be more of the same
03:48.15JTheh
03:48.38FuriousGeorgewhy would using hardware not put out by digium lead to less deadlocks
03:49.07JTshrug
03:49.13JThow does a deadlock materialise?
03:49.17FuriousGeorgei guess ill try 1.4.3 or something, that should be out before the next deadlock
03:49.22FuriousGeorgeclassic symptoms
03:49.31JTdon't think i've had one
03:49.40FuriousGeorgecertain cli commands make cli stop responding (like stop now) but others work (like show channels)
03:49.56FuriousGeorgeincomming calls cant be bridges to channel that tries to answer
03:50.01FuriousGeorgeoutgoing calls dont go out
03:50.01JTah hmm
03:50.33FuriousGeorgeget a lot of "Avoided initial deadlock at (hex value)"
03:50.48FuriousGeorgei guess it couldnt avoid the "subsequent deadlocks"
03:51.04nick125_lappyugh, downgrading to 1.2.13 is becoing a hassle
03:51.26JTwhen you restart daily, what do you do, just restart asterisk?
03:52.18Qwellhttp://www.youtube.com/v/Bj1Mtv9cD0I
03:52.35FuriousGeorgeJT: my script is _very basic_.  i just try a asterisk -xr "stop when convenient" then a "stop now" then a killall asterisk / mpg123 / safe asterisk then i safe_asterisk
03:52.49JTok
03:52.49FuriousGeorgebut when it deadlocks i need to kill -9 the asterisk pid
03:53.12JTmay i suggest that, before you start safe asterisk
03:53.24JTyou rmmod whatever specific zap modules you are using
03:53.27JTmodprobe them
03:53.32JTztcfg -vv
03:53.36JTthen do safe asterisk
03:53.39FuriousGeorgeyou may and i will
03:54.14FuriousGeorgei could even see if i could use my basic shell programming to try a kill -9 and find the pid (not in that order)
03:54.35JTit could be a problem in the kernel modules that builds up over time
03:54.44nick125_lappylets see if asterisk 1.2.x will fix my issues
03:55.03FuriousGeorgeJT: any word on 1.4 stability vs. 1.2?  does anyone do metrics on that sort of thing?  someone should
03:55.08JTi had problems with kernel modules for a multiport BRI card, although i don't think they were deadlocked
03:55.15JTbut they surfaced in now time at all :P
03:55.35JTFuriousGeorge: personally, i think it would be less stable, especially given the .0 version number
03:55.50nick125_lappyYay, it works in 1.2.x
03:55.55JTs/now time/no time/
03:56.03JTnick125_lappy: sweet, what did you fix?
03:56.03FuriousGeorgethats what i think too
03:56.08FuriousGeorgethanks for the input
03:56.14FuriousGeorgeill let you know in 4 months if it helped
03:56.15*** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com)
03:56.16FuriousGeorge:)
03:56.21JTlol, np :)
03:56.26nick125_lappyJT: reparking calls after they've been parked
03:56.30JTah
03:56.36JTmaybe it's a bug/
03:56.43nick125_lappypossibly
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03:56.49JTmight be worth checking if a report has been filed
03:57.56nick125_lappy:/
03:57.56nick125_lappyJan 17 15:02:54 NOTICE[5578]: res_features.c:2053 load_config: Unknown feature 'parkcall'
03:58.29nick125_lappyI guess that's a 1.4.x only feature
03:58.31nick125_lappy*removes
04:01.55JTi thought you said it was working
04:05.25nick125_lappyit is, if I transfer the call to 700
04:06.00nick125_lappyparkcall (#72) didn't work anyways ;)
04:06.31JTah, so what does parkcall do differently?
04:09.29nick125_lappyI guess it skips a step
04:10.29nick125_lappyinstead of hitting # then hitting 700, you just hit #72, I guess
04:10.35nick125_lappyit never worked for me, so, not a big loss
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04:22.30rpmin asterisk, all host= statements have to be unique in sip.conf or you have to use type=friend if you want to recieve/send calls right?
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04:24.28Strom_Chost= is only for outbound calls
04:24.35Strom_Cinbound calls match on the username
04:24.47Strom_Cyou can also restrict by IP range
04:25.04ManxPowerIf you want to match by address use permit/deny
04:25.46ManxPowerIn fact you can have multiple sip.conf sections with exactly the same host= line.  You can do this to allow/disallow different codecs
04:26.45ManxPower(for OUTBOUND calls, of course)
04:29.36rpmi can't seem to figure out why im getting failure to authenticate on invites when dialing out. even though im registered to the proxy
04:30.31[TK]D-Fenderrpm : Registering has nothing to do with your peer being set up right or being able to place calls period.
04:37.14*** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1096579263.dsl.bell.ca)
04:37.16JunK-Ywe're having a discussion, is the CLi needs filtering for specific patterns, channel and session?
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04:40.46[TK]D-FenderJunK-Y : The CLI could use a loggin function to be initiated on-demand
04:40.50[TK]D-FenderJunK-Y : like
04:41.01[TK]D-Fender"log [filename]"
04:41.19JunK-Yjoin #asterisk tk
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04:47.32DocHolliday[TK]D-Fender, how are the napping sessions?
04:48.38polinuxHi to all
04:48.58*** join/#asterisk coppice (n=chatzill@198.199.17.210.dyn.pacific.net.hk)
04:49.58[TK]D-Fenderdoc?
04:50.29*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
04:54.37DocHolliday[TK]D-Fender, i did 3 pm to 6 PM today
04:54.40ealdhi all
04:55.04ealdis asterisk multi threaded?
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04:55.45*** part/#asterisk nbits (n=chris@unaffiliated/nbits)
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04:56.35JunK-Ysome part has threads depends of your definition of multi-thread
04:57.03mike052279hey could i get some really noob help?
04:57.03[TK]D-FenderDocHolliday : Don't follow you at all...
04:57.47nick125_lappyanyone here know if the FWD Call Me function is working?
04:58.26polinuxcan some one tell me how and from where does the *97 for voicemail works
04:58.30JunK-Ynick125_lappy: in what this is related to * ? :)
04:58.44JunK-Ypolinux:
04:58.48JunK-Y~book
04:58.50jbotextra, extra, read all about it, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
04:58.51nick125_lappyJunK-Y: Trying to test my asterisk box ;)
04:59.04nick125_lappyAnd i'm trying to figure out if its FWD not working or asterisk not working
04:59.41polinuxJunK-Y: the Oreally?
05:00.08ealdI see, then how good is for asterisk to run in a multiprocessor system?
05:00.49JunK-Yoreilly ya
05:03.13mike052279ok would someone mind helping me out? :)
05:05.00Supaplexhow can I conditionally switch between two extensions if queue(...) returns 0 or not?
05:05.13[TK]D-Fendermike052279 : Try asking a specific questio.... you might jsut get a specific answer...
05:05.37[TK]D-FenderSupaplex : "show application gotoif"
05:05.42rpmwho would have through dialing out could be so difficult.
05:06.35SupaplexI'm using that, but apparently my syntax is foobar. humm..
05:07.17[TK]D-FenderSupaplex : keep at it..
05:10.48mike052279ok my question is - does anyone know the basic script that answers, plays one of the default prompts and hangs up using a did/iax?
05:11.07mike052279i looked everywhere on the web and everything i found was way more advanced than basic stuff :P
05:12.48Supaplexyou want to do something based on the did over iax? eg, play a greeting based on the number they called.
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05:13.39mike052279yes very basic stuff
05:13.49mike052279i just learned this stuff like 3 hrs ago
05:13.50mike052279lol
05:14.01mike052279i am setup with junction for the did's
05:14.11mike052279and i followed their sample script and it actually worked
05:14.12Supaplexthey have a sample already
05:14.14mike052279yeah
05:14.31mike052279but when i tried to do my own thing, all i get is a fast busy now
05:14.59*** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
05:14.59Supaplexyou have to answer the call :)
05:15.01*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
05:15.38mike052279wellllllllll how it was setup before is it was all ivr based, so i never used anything internal and my "buddy" did it, so now im trying to do the same thing
05:15.42Supaplexeg, exten => _18001234567,1,Goto(company1,1,1);
05:15.50mike052279ah
05:15.56mike052279i was trying this stuff
05:16.14mike052279exten => 12126600009,1,Answer()
05:16.14mike052279exten => 12126600009,2,Playback(welcome)
05:16.14mike052279exten => 12126600009,3,HangUp()
05:16.34[TK]D-Fendermike052279 : Start with THE BOOK....
05:16.35[TK]D-Fender~book
05:16.45jbotit has been said that book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
05:16.45mike052279um basic stuff isnt there
05:16.45mike052279i already read it
05:16.46mike052279:)
05:17.10Supaplexit's all there :P it's just takes time to figure out how the pieces relate to each other, and to your situation
05:17.27[TK]D-Fendermike052279 : all you get is a fast busy... from where?
05:17.32mike052279from my house phone
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05:18.33mike052279supa: i just want to get a successful connection doing one simple thing, then i will worry about piecing it together :P
05:18.48mike052279cuz my buddy had it very advanced so that will be shortly :P
05:20.07Supaplexit's not like I know anything, I'd be lost w/o the book.
05:20.07[TK]D-Fendermike052279 : clarify "from my hose phone" please...
05:20.30mike052279lol
05:20.31mike052279ok
05:20.42mike052279i have different voip service through my cable company
05:20.52mike052279so i am using that phone to test to dial to the trixbox
05:21.17mike052279and it worked with the sample script from junction but now it just does a fast busy when i put that basic script in
05:21.45[TK]D-Fendermike052279 : ...
05:21.48[TK]D-Fender~trixbox
05:21.57jbottrixbox is, like, unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/
05:22.40mike052279yeah
05:23.20mike052279im testing right now to make sure i can get this to work cuz then im hoping to just copy the scripts over to a real asterisk box
05:23.22mike052279:)
05:23.27mike052279i asked in there too
05:23.30mike052279no help!
05:23.40[TK]D-Fender~wglwat
05:23.43jbotit has been said that wglwat is well, good luck with all that
05:23.43mike052279but im not setting up any analog lines
05:23.51mike052279lol
05:23.57mike052279thx!
05:24.05[TK]D-Fendermike052279 : The idea of salvaging anything useful from FreePBX is a pipe dream
05:24.16mike052279hmm so i should start over
05:24.17mike052279?
05:24.24[TK]D-Fendermike052279 : You are better off starting from scratch then trying to come back from there.
05:24.28mike052279ok
05:24.55[TK]D-Fendermike052279 : What provider?
05:25.08Supaplexhe said junction
05:25.15Supaplexwhich works fine for me, fwiw
05:25.27*** part/#asterisk simplton (n=root@c-24-99-119-243.hsd1.ga.comcast.net)
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05:26.35Supaplexgrrr why is this call in queue if joinempty = strict and leavewhenempty = yes
05:26.53flendersmike052279: personal experience, I tried trixbox, and ended up installing asterisk from source
05:27.00[TK]D-Fendermike052279 : http://www.junctionnetworks.com/asterisk.php
05:27.11flendersnot long ago, read the book, and now all the pieces are coming together.
05:27.19nick125_lappyI gave up on asterisk control panels about...2 years ago
05:28.08Supaplexnothing is prepared to handle configuration like a good ole' text editor.
05:28.22nick125_lappyAMP (that's what FreePBX or whatever it is used to be called) is junk
05:28.25Supaplexnot to be confused with O.L.E.
05:29.33mike052279hmmmmmmm
05:29.39mike052279does it take awhile for a # to register
05:29.50mike052279like if i just registered a new did like a couple hrs ago
05:30.31polinuxPlease I cant get the *97 to work any ideas?
05:31.10Supaplexmike052279: does "iax2 show registry" show any "Registered" entries?
05:31.11[TK]D-Fenderpolinux : what is "THE" *97?
05:31.32mike052279yea
05:31.53[TK]D-Fenderpolinux : Extensions don't exist out of thin air.  its your config, how about SHOWING us?  PASTEBIN please ( www.pastebin.ca )
05:32.04mike052279does IAX need firewall port open?
05:32.10mike052279er supa
05:32.12mike052279:P
05:32.21[TK]D-Fendermike052279 : Yes, and forwarded to your * box.
05:32.24[TK]D-Fender4569
05:32.24mike052279lmao
05:32.28mike052279duh@!
05:32.29mike052279ok
05:32.33mike052279thats probably the problem
05:32.48[TK]D-Fenderindeed brilliant.
05:32.53mike052279lol
05:32.57mike052279thx fender :D
05:33.05[TK]D-FenderIts not like * has to actually COMMUNICATE with the outside world or anything...
05:33.08Supaplexlog before deny :)
05:33.09mike052279lmao
05:33.48polinuxwhen you dial *97 it should take you to voicemail
05:34.20CunningPikepolinux: Only if you program it that way. It's *98 on ours
05:34.32bkruse_homefile: <3
05:34.38filebkruse_home: !
05:34.51[TK]D-Fenderpolinux  : says who?
05:35.05polinuxCunningPike: how do I setup that, in extensions.conf? isnt that part of asterisks funcitons?
05:35.14Supaplex[TK]D-Fender: well, there is app_orthogonal_persistency
05:35.28[TK]D-Fenderpolinux : you very clearly don't have the slightest clue about *.  Go read the BOOK.
05:35.30[TK]D-Fender~book
05:35.43jbotit has been said that book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
05:36.43Supaplexpolinux: big hint #1, it's what voicemailmain() is for.
05:36.44bkruse_homefile: i should go to sleep :[
05:36.49filebkruse_home: whyfor?
05:36.55Supaplexhow you configure it, is your bidding.
05:36.59bkruse_homefile: school! :[
05:37.04bkruse_homewhen you coming to visit meh!?
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05:37.29mike052279okkkkk
05:37.39mike052279so
05:37.42file2 weeks!
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05:37.51mike052279that fixed one of the numbers
05:37.52mike052279:P
05:37.58CunningPikepolinux: You'll need something like exten => *97,1,VoicemailMain(${CALLERID(num)}@default)
05:37.59[TK]D-FenderSupaplex : Ok, youg ot me on that app... I cannot clearly understand the joke :)
05:38.01mike052279i looked at cid log in junction
05:38.17mike052279it says failed
05:38.23mike052279so is that a junction issue?
05:38.33bkruse_homeone day im just going to look around the corner and youll be like WEEE IM FILE!
05:38.40bkruse_homeand ill be like, ZOMGZ LETS GO GET CAKE
05:38.51bkruse_home:X
05:38.57perdcake?
05:38.58polinuxCunningPike, thaks alot, thats all I wanted to know, I was not sure if *97 was a function within Asterisk or if I had to put it in the dialplan
05:39.00perdthat sounds good
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05:39.04Supaplex[TK]D-Fender: hehe. define orthogonal persistency first :)
05:39.05bkruse_homeperd: indeed
05:39.11[TK]D-Fendermike052279 : I'd lay bets that your ITSP is doing their job just fine, and its YOUR config and routing that needs to be corrected.
05:39.27bkruse_homelol [TK]D-Fender agreed.
05:39.28mike052279lol
05:39.32CunningPikepolinux: There are no built in extensions in asterisk - you have to program them all yourself
05:39.33mike052279k
05:39.46[TK]D-FenderSupaplex : I looked up "orthogonal", and couldn't come up with a meaningful usage in that context :)
05:39.58coppiceA telco doing their job fine is highly implausible
05:40.19[TK]D-Fendercoppice : But they are well stocked on brownian substances...
05:41.21SupaplexI guess this would fall under 'software engineering': http://en.wiktionary.org/wiki/orthogonal
05:41.36Supaplexwhich makes little sense still.
05:41.49Supaplex22:34 <dpkg> "start a fs war" is "<reply>((ufs|ffs|xfs|jfs|reiserfs|ext3|FAT|UDF|NTFS) (uses too much overhead!|is too slow!|will eat all your data!!)|Who needs a file system? Use orthogonal persistency!)"
05:41.53Supaplex:P
05:42.01Supaplexthat's where I stole the idea from.
05:42.08bkruse_home~zomgz
05:42.11jbotzomgz is probably a word that brandon said that is omgx2=zomg zomgx2=zomgz omgx4=zomgz. It is the equivalent to the LOL of laughter, and the YAY of excitement
05:43.27*** part/#asterisk bkruse_home (n=kruz@69.73.127.92)
05:43.46rpmwhich type=peer variable do i use to set an authuser? is it username=
05:45.27[TK]D-Fenderpl, I'm fried.... later all...
05:45.27Supaplex[TK]D-Fender: my vague memory thought it meant ~ 'omniscience of the topic w/o any special requirement/work/etc'. :-} oh well hehe.
05:45.42*** part/#asterisk polinux (n=gimmesom@ip70-190-159-144.ph.ph.cox.net)
05:46.01coppiceif two related things are orthogonal, they don't affect each other (see husband and wife) :-)
05:48.33*** part/#asterisk simplton (n=simplton@c-24-99-119-243.hsd1.ga.comcast.net)
05:48.40joeso I have a new polycom 301 that has a much newer version of sip and the bootroom than the other phones I have, any reason why it wouldn't work as it or do I need to upgrade them?
05:51.27perdoh noooo
05:51.29perdnot that word again
05:52.23flenders# = octothorpe??
05:52.50coppiceI wonder who came up with that name? and why?
05:52.52^sandro^hey can anyone answer something for me
05:53.01^sandro^i have pri's right.. and incoming calls when i do zap show channels show up
05:53.08^sandro^but i can't see the ones that are going outbound
05:53.36^sandro^any way to say zap show channels and display all incoming / outgoing ?
05:57.01*** part/#asterisk mog (i=ejabberd@71.207.215.93)
05:58.08^sandro^anyone?
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06:22.47x86AJaymn: hey man
06:23.44rpmhttp://pastebin.ca/320266, anyone see a major difference between those two invites besides the user-agent and caller id and call-sequence numbers?
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06:35.39AJaymnx86: Hi! long time no talk
06:36.03AJaymnx86 i cant belive your really here ;)
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06:42.08x86AJaymn: hehehe
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06:59.48Fileflyi'm having trouble with my inbound DTMF
07:00.16Fileflyi had to set dtmf=inband and dtmfmode=rfc2833 in my sip.conf
07:00.27Fileflyand that works fine with my main DID
07:00.44Fileflybut on my alternate DID the settings don't seem to take effect
07:01.09Fileflyi put those lines in the [global] section, but that doesn't seem to be the solution
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07:05.33Supaplexsome features only work in the global section. afiak, atleast one is a wishlist/bounty items for multiple contexts
07:08.00Fileflyi'm not entirely clear on what a context is
07:08.08Fileflyi'm a newbie :)
07:08.14Supaplex[foo]
07:08.17Fileflyi understand the basic concept
07:08.22perdit's one of the most basic parts of the config files
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07:08.53nixboxhi all
07:08.56Supaplexit's an autofail item for asteirsk certification (I'm kidding, but if there's such a thing, so be it)
07:09.05Fileflygrin
07:09.08nixboxis there any other windows client except X-lite that I can use with asterisk?
07:09.23perdsjp phone works
07:09.25Supaplexsomething that runs on glass?
07:09.29perdor something like that
07:09.46perdand there's an iaxphone that works, forgot the name.  voip-info.org will have links
07:09.47Supaplexnetmeeting! *cringe*
07:09.49JTidefisk for iax works
07:10.02Fileflythanks Supaplex :D
07:10.27Supaplexanytime hehe
07:10.41perdoh sjphone i was thinking
07:11.56nixboxone that uses SIP?
07:12.22Fileflyi'm not sure how to go about telling * to apply those settings to my alternate DID as well as the context i've defined for the main DID
07:12.41JTnixbox: there's a big softphones list on voip-info.org
07:13.35Supaplexnetcat ;)
07:14.27perdi use netcat to irc
07:15.00nixboxok thanks
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07:33.47data23*yawns* Morning all
07:34.05Supaplex*GASP* already?
07:34.25Supaplexit's not morning until there's sunlight!
07:34.39mitchelocit's only 11:34 here! hah!
07:34.58mitchelocstill prime work hours :)
07:35.24nixboxhow do i know which client will provide me the option to use any service provider? :S
07:35.35AJaymnask it ;)
07:35.42Supaplexwhich what client?
07:35.44nixboxthere is a long list on voip-info but most of them are specific it to the service
07:35.56nixboxSIP client/softphone for windows
07:36.03Supaplexoh, softphones. man good luck.
07:36.07data2307:35 here :)
07:36.07nixboxi want it to be configurable for any service
07:36.15nixboxlike X-lite is
07:36.26Supaplexuse * inbetween.
07:36.33nixboxis there some other like X-lite which can be configured?
07:37.21JTnixbox: X-pro/eyebeam? :P
07:38.59data23guess that's a no ;(
07:39.08perdthat's a hell no
07:39.19Charles[NS]Good Morning
07:41.44JTnot realy clear what you want to do
07:41.46perdseems you'd have to have some kind of callback to the meridian server
07:41.55*** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
07:42.26data23JT: [userA] ----> [meridian] ----> [Asterisk] ----> [meridian] ---> [userB]
07:42.48perdcanreinvite=yes
07:42.50perdheh.
07:42.57perdi kid.
07:42.57JTare these users on local extensions or on the pstn?
07:43.29data23JT: the users are local extensions on the meridian, but it gets to the meridian via a q.931 pri
07:43.54JTweird that local users would be using the ivr i would've thought
07:44.10data23well, take the example of a store shortcode system
07:44.13JTi doubt you can do it, to be honest
07:44.24data23I have 420 stores in the UK, each with 01xxx yyyzzz numbers
07:44.37data23if i map 3xxx to my asterisk box, i can get it to redial the proper store number
07:44.45Strom_Cdata23: if the meridian doesnt support the call transfer part of q.931, you'll just have to be content with looping back
07:44.54data23i.e. 3001 = store 1 = 01255 523431
07:45.05JThow well does asterisk support call transfering?
07:45.12Strom_Cthat i dont know :)
07:45.33data23i've been told by the BT engineers that the meridian should be able to do it and all the q.931 trunk pruning options are set
07:45.42JTi know bristuff includes some ECT (explicit call transfer) code for libpri
07:45.48JTyou might need bristuff
07:46.04data23Flash() doesn't work on PRI's and Transfer() just does nowt, and if you dial() it uses 2 channels :)
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07:47.27JThttp://72.14.253.104/search?q=cache:ti7WmLItCoMJ:lists.digium.com/pipermail/asterisk-users/2005-May/100662.html+bristuff+ect&hl=en&gl=au&ct=clnk&cd=9
07:48.55JTalso http://lists.digium.com/pipermail/asterisk-users/2005-May/098340.html
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07:54.08data23ooh thanks, i'll have a proper read of those when i get to work
07:54.29data23as for now, i have an hour and 6 minutes to get there :|
07:54.32data23bbl :)
07:54.51JTdata23: also, you may need to read source code comments for the bristuff patches to work out how to do it
07:54.58JTdocumentation seems fairly poor
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08:38.52Charles[NS]hello
08:39.23Charles[NS]who use misdn with BN8S0 and asterisk 1.2.14
08:40.15Charles[NS]phone works in local but i can't use to external call
08:40.39Charles[NS]Jan 17 19:44:53 WARNING[2832] chan_misdn.c: Could not create channel on port:-1 with extensions:0141163490
08:40.43Charles[NS]Jan 17 19:44:53 NOTICE[2832] app_dial.c: Unable to create channel of type 'mISDN' (cause 0 - Unknown)
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08:45.30yxaCharles[NS] what kernel?
08:46.27yansolo90hi, does it exist a version of spandsp, rx_fax ans tx_fax for Asterisk 1.4 ?
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09:07.39linageei have an IAX2 connection to my ITSP on my asterisk box. i have a remote site with a SIP phone (and can set up any hardware needed). is it possible to have calls come into my main site through the ITSP, but if someone dials the extension of the remote site, there will be some sort of SIP reinvite so i can have better latency and save bandwidth?
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09:30.16inspiredyansolo90, no, but it works with openpbx
09:30.57inspirednatively, as it is developed for it. however, I think there is a test version of rxfax and txfax for 1.4. not sure if it works
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09:34.23lupino3hello everybody
09:34.32lupino3small question:
09:34.39lupino3which extension does Asterisk call
09:34.54lupino3when the dialled extension is not available in the current context?
09:35.05lupino3I thought it would call "I", but it doesn't!
09:39.44Gido-Ei
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09:55.15tzafrirlunaphyte, s
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10:13.51lupino3is it possible, via a normal IAX call, to choose the context of the extension that I'm dialing?
10:14.08lupino3something like "dial someext@somecontext"
10:15.58jqlis somecontext a context on the remote server?
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10:51.17Aces1Upanyone here have experience with call shops?
10:54.51bkw__thats like asking "are you human?"
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10:58.33coppiceI'm sure some bushmen of the Kalahari or head hunters from Borneo have not suffered the negative effects of call shops.
10:59.03coppiceon the other hand, if the head hunters started seeing the negative effects, they might take some positive action :-)
11:07.56Aces1Upbkw i take it you have some experience in it?
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11:41.24BrokenNozeasterisk is unable to send voicemail notification emails, anyone help me out where why?
11:41.44x86can you guys try hitting https://voip.shellshark.net please
11:41.45x86i want to verify the SSL cert is not causing any issues for anyone
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11:43.10Makenshiok in ff 1.0.4
11:43.14jeremy_gdont do that guys, x86 is a damn fingerprinter
11:43.15zeeeshhi
11:43.17jeremy_g:P
11:44.03ping2921is asterisk compatible with mysql 5.0?
11:45.39x86jeremy_g: hehe, no
11:46.08x86ping2921: asterisk can't directly communicate with any database, but the mysql realtime driver in asterisk-addons will work with mysql 5.0, sure
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11:46.42jeremy_gor he wants to impress his boss by the hit rate and making similar requests on other channels as well
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11:47.25jeremy_gx86:mysql 5.0 for 1.2 or 1.4?
11:47.31ping2921x86 -- have you actually tested it?  I upgraded from 4.12 to 5.0 and now I get segmentation erros.
11:48.01x86ping2921: yeah I run 5.1 actually
11:48.08x865.1.26, iirc
11:48.13x86perhaps .24
11:48.17ping2921and ast 1.4?
11:48.19x86jeremy_g: 1.2
11:48.27x86ping2921: ah, i have not tested 1.4 yet
11:50.13jeremy_gok
11:50.36jeremy_gwhich * ver did you test it with
11:50.41jeremy_g1.2.?
11:51.13x861.2.13 is what i'm running in production with it right now
11:51.32jeremy_gok
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11:57.12buzzydexHi guys anyone know what this is all about
11:57.15buzzydexchan_sip.c:2542 sip_write: Asked to transmit frame type 8, while native formats is 256
11:57.33buzzydexgot a load of them coming up in cli
11:57.49buzzydexI think it is form some snom 320 phones
11:59.40phearlesshi guys
12:00.06phearlesshow does work power-over-ethernet for a phone ? (in my case I will buy 15 Linksys SPA942)
12:00.29phearlessI do need only one "thing" that push the electricity on the ethernet cables ?
12:01.44phearlessI have seen "The Linksys 942 can be used with a POE Converter (if your network is not POE-enabled)."
12:04.33monstedyou either pop in a brick ("injector") between the switch and the phone or buy a switch with POE built-in
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12:05.39monstedthe injector or switch then probes the cable looking for special characteristics that indicate that the other end wants power and if so, turns it on
12:06.28monstedinjectors are fine if you only need a few or if you have to spread them around, but if it's all in one place you'll be better off with a POE-enabled switch
12:09.16jeremy_gphearless:will poe work on regular ethernet networks (typical switches with cat5 or 7 calbe)
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12:18.17McGeeHi
12:19.04phearlessokay thank you jeremy_g and monsted !
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12:23.17McGeeHow can i dial multible extensions? like if someone dials 200, internal extensions 101 102 and 103 should be triggered...
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12:31.12x86Dial(SIP/101&SIP/102&SIP/103)
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12:46.28in-pthi all
12:48.01in-ptcan anyone tell me about grandstream ip phone GXP-2000, if it supports power over ethernet
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12:58.59stoffellin-pt; no it doesn't..
12:59.47DrukenLPYwtf is with all the poe questions this morning....
13:00.24stoffellprobably PoE day in europe or so.... lol
13:00.34DrukenLPYmust be...
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13:01.05stoffelllol
13:01.26DrukenLPY:)
13:01.51DrukenLPYgoogle always anounces the useless and obscure hollidays that no one knows or remembers
13:02.09e-ddiei guess you should put 'PoE support on #poe' in the topic
13:02.24DrukenLPY~poe
13:02.35jbotpoe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt
13:02.48DrukenLPYhmmm.....
13:03.11e-ddiethere you go
13:03.13e-ddie:D
13:03.30McGeex86, I came this far but actually i want to throw it into the verry fist stage of the dialplan. The problem is that Pickup does not work if i dial using the SIP/ techn.
13:03.33DrukenLPYjbot your so dumb
13:05.38tzafrir~lart DrukenLPY
13:06.53tzafrirjbot, poe is also Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices
13:06.55jbottzafrir: okay
13:07.08tzafriranybody with a better definion?
13:07.17DrukenLPYoh sexy, it's even a canadian slap :)
13:07.21tzafrirA highly relevant link with all the answer?
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13:08.28DrukenLPYhttp://en.wikipedia.org/wiki/Power_over_Ethernet
13:08.30DrukenLPY??
13:09.46coppicejbot, poe is also a control freak's atttude to networking
13:09.48jbotokay, coppice
13:10.02tzafrirhttp://www.poe.org  , http://www.realpoe.org   (not)
13:10.14tzafrir~poe
13:10.15jbotpoe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt.  Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices, or a control freak's atttude to networking
13:11.01tzafrirjbot, no poe is poe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt.  Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet
13:11.04jbottzafrir: I think you lost me on that one
13:11.14tzafrirjbot, no, poe is poe is, like, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt.  Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet
13:11.15jbotI think you lost me on that one, tzafrir
13:11.44tzafrirjbot, no, poe is, Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt.  Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet
13:12.28coppice~poe
13:12.30jboti guess poe is Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt.  Power Over Ethernet, a method to fed power through a RJ45 connector from the ethernet switch to devices: http://en.wikipedia.org/wiki/Power_over_Ethernet
13:13.46tzafrirthe typo there is now fixed
13:13.55x86POE++
13:14.20coppicethat would be edgar alan poe
13:15.12DrukenLPYwhich typo? fed to feed?
13:15.12tzafrirPOF
13:15.13tzafriryes
13:15.32DrukenLPYexcrement
13:16.49tzafrircoppice, sadly, Edgar Allan Poe is only the second and third hit for "poe" on google. The first one is some model
13:17.43coppicethe first hit I get is bau2.uibk.ac.at/sg/poe/
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13:18.26DrukenLPYcoppice: i get that too.. the work of edgar allen poe
13:20.50DrukenLPYhmm... i wonder why all my clocks seem to be 9 mins out from the rest of the world....
13:21.46coppicetime dilation on the fibres
13:22.11`Seanhey someone quick question
13:22.14`Seanwhat timne is itr in NY
13:22.42x868:22am
13:22.47`Seanthanks
13:23.03x86np
13:23.07DrukenLPYcoppice: could the time drift be wrong in my ntpd.conf?
13:24.07jeremy_gwhy are all you guys gang raping poe
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13:32.54dezent_hello, trying to compile zaptel-1.4.0 on debian etch.. got this error " WARNING: Symbol version dump /usr/src/linux-source-2.6.17/Module.symvers \is missing; modules will have no dependencies and modversions." found this bugreport http://bugs.digium.com/view.php?id=8732 ... does anyone have a solution ?
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13:40.51radcliffanyone running asterisk 1.4? if so, on what linux-distro... I'm having problems getting zaptel to work on debian etch due to bug: http://bugs.digium.com/view.php?id=8732
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13:52.33littleballhello, what is the best way to get the channel status of asterisk? i need to put these info on web.
13:53.05littleballchannel include SIP channels and Zap channels
13:53.59Gido-Ethe best, is not!
13:54.32littleballGido-E, ok. change to "good way"
13:54.53littleballsome people just like web console.
13:55.21radcliffmake all extensions members of a queue, and do a show queue queuename
13:55.27radcliffugly...but it works =)
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13:55.39radcliffupdate each second via ajax
13:55.42radcliff=)
13:56.59littleballradcliff, what is the relation btw queue member and channels status?
13:57.06radclifflittleball: you can do this via asterisk manager from a php script and regexp the data you want
13:57.26radclifflittleball: the command shows channel status of each member
13:57.31littleballi want to show the status like in using or idle on web
13:58.05radclifflittleball: try the command: show queue queuename in asterisk console and you will understand
13:58.27DrukenLPYlittleball: why not use FOP ?
14:00.14littleballFOP? what is that? i already implement lots of service based on asterisk. Now want to make the channel available throught web also...
14:00.32littleballor. flash one
14:00.46littleballno, i am using j2ee stuff.
14:01.56littleballradcliff, i never use queue before. I though queue should be used for "call center", right? like if all staff are busy, then the incoming calls are queued. right?
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14:03.37lstephi all
14:04.04SheriF_SpacEwhy the auto dial with .call files dosn't work with zaptel channel :-s?
14:05.00lstepIf I use 'Asterisk NOW', can I install "easily" a Digium BP410P (for Euro-ISDN), and have it detected by the GUI?
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14:26.27drakoOk so again, for a SOHO with 12 extenions should I go for Sipuras ATA or VoIP phones?
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14:27.25Op3riphones
14:27.30drakowhat's the cheapest and usable IP phone out there in the market?
14:27.50cpmwhen talking cheap, usable is quite relative
14:28.01cpmcheap is cheap, period.
14:28.04Dr-Linuxi forgot cisco's 7960 default password, what should i do to modify phone's setting???
14:38.12x86drako: i was in your shoes once
14:38.19x86drako: i wanted the cheapest IP phone available
14:38.28x86drako: so i got the Grandstream BT-101
14:38.39x86drako: i've been kicking myself in the nads ever since
14:38.51x86drako: now i have 4 Polycom IP601's sitting here
14:38.57x86and i'm in love with them ;)
14:39.47drakox86, how much they go for?
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14:40.46robl^601s are like $250
14:41.33robl^Polycom 430s (2 lines vs 6 lines on the 601) are $130
14:41.41drakotoo much
14:41.46drakoill go for atas.
14:43.01x86drako: a decent ATA is like $70 anyway
14:43.10x86drako: for 1 line with no features at all
14:43.13drakoyah for 2 ports
14:43.26x86look at the IAXy ;)
14:43.29drakoa sipura 2002 are pretty nice
14:43.34x86single port for like $85 heh
14:43.39drakoand they can hook to phones
14:43.45x86but there's no features in that at all
14:43.53drakotwo*
14:44.06drakox86, like what?
14:44.46x86call reject, micro browser, line keys with directory, the list goes on for miles ;)
14:44.54robl^drake:  seriously.  once you play with Asterisk a bit, you will decide that the decent phones are the way to go.  ATAs will cost you from $65 to $100, usually a single line.  Add an analog phone for another $20.  So For about $90-$120 youget an analog single line phone with few features.  a polycom 430 has a great speakerphone, easy to use LCD, 2 lines, etc.
14:45.01Aursis it possible to set up logger.conf so that verbose level 0 is saved to a separate file?
14:45.23Aursfilename => verbose won't do the trick
14:45.32x86robl^: or there is the 650 now with the backlit display :)
14:45.58robl^x86: yeah.. but if he's complainign about the price, the 650 is out of his budget ;-)
14:51.03robl^the 430 is a closer comparison to the BT101.. just the 430 is 100x better. ;-)
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14:55.12nick125_lappyBy the way, grandstream's are cheap both in quality and price :p
14:56.01robl^nick125_lappy: we are training you well, I see .  ;-)
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14:58.23nick125_lappyI think it was only $15 more for the PAP2T-NA, so, I just spent the extra $$ and got the PAP2T-NA
14:58.50asteriskdude2Hello, any someone point me to database queries. What it means and how is it useful in retrieving the area and city of the call
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15:04.26sumawhy i get no such host
15:04.27sumachan_sip.c:1989 create_addr: No such host: 213.166.5.135
15:04.43asteriskdude2I think I am sounding stupid.. How do I get the area code ad city of the caller?
15:04.46sumait exists and i can ping from the same system where asterisk is installed
15:04.56expat_Need some opinions on the type of server people would advise for a 40 extension SIP based server, PRI card, voicemail, fax2email, conferencing about 15 PSTN calls at any point in time, aLaw and G.729 codecs...perhaps GSM also.
15:05.17expat_What do you think for CPUs and RAM?
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15:36.15nicklinnHello all, I am having a rather weird problem with 1.4 on a 2.4 kernel.  When I run asterisk without ztdummy things work fine with the execption of stuff that requires it.  But the second I load ztdummy, calls work fine but the second I hang-up using a softphone the whole system hangs.  Anyone seen this problem before?
15:40.53mercestesnicklinn:  =/  Weird.
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15:41.58nicklinnI am not even getting a kernel panic anywhere.  The machine just stops
15:42.08nicklinnIt's very odd
15:42.34nicklinnlitterally this:
15:42.35nicklinnJan 18 10:58:29 ideals kernel: Zapata Telephony Interface Registered on major 196
15:42.35nicklinnJan 18 10:58:29 ideals kernel: Zaptel Version: 1.4.0 Echo Canceller: MG2
15:42.35nicklinnJan 18 10:58:30 ideals kernel: Registered tone zone 0 (United States / North America)
15:42.35nicklinnJan 18 11:09:40 ideals syslogd 1.4.1#17: restart.
15:43.21mercestesThe *entire* machine freezes?
15:43.31nicklinnYup not even local keyboard
15:43.36mercestes...only when you hang up a softphone?  Not on a conference/meetme call?
15:44.22mercestesThis reminds me of a house call I got back when I was doing my own computer consulting business...
15:44.26nicklinnI tested a meetme call a few times and confirmed that a hangup will cause it.  But I only tried an extention to extention call ones with a freeze
15:44.35mercestesGot a call, a young girl, said she replaced her soundcard...and her monitor went out.
15:44.37nicklinnonce rather
15:45.11mercestesSwore up and down that she touched *nothing* on the motherboard, didn't move any wires...just put in a new soundcard and *poof*  monitor went out.
15:45.30mercestesTurned out, her system wasn't posting.  Spent a few hours testing..told her she needed a new mobo.  Must be "static electricity."
15:45.52nicklinnIt happens
15:45.54mercestesShe asked me to wait a "month or so" until she got paid, and I said ok.
15:46.26mercestesSix months later, she calls back (much to my surprise) ready to order a new Mobo.  Just to "refresh my memory" I went through her system again.
15:46.45mercestesshe had inverted the Power LED plug on her Mobo.  Powered up just fine, took five minutes to fix.
15:47.45nicklinnMy lord, whoever designed a motherboard that depends on a Power LED should be shot
15:47.53[TK]D-Fenderyup.....
15:47.57mercestesSo my question is....what else is different on your system that could be causing the entire system to freeze when you hang up a softphone with ztdummy installed?  Which distro?  which Softphone?  What are you calling?  etc.
15:48.23jeremy_gmercestes:you got style baby!! ;)
15:48.34mercestesjeremy_g:  hehe
15:48.40mercestes....
15:48.42mercestes=/
15:48.44jeremy_gmouth off course
15:48.48mercestesahhh.
15:49.35nicklinnLets put it this way.  If I do not load ztdummy I can call softphone to softphone without a problem.  I cannot use meetme etc for the obvious reasons.  I am using Debian sarge.  Kernel 2.4.18bf.  Eyebeam and another softphone X-ten lite
15:50.04mercestesEyebeam and X-ten are extremely similar.  I have used both.
15:50.25mercestesBy extremely similar that means I am 98% certain they are identical source.
15:50.27nicklinnYup pretty much
15:50.53mercestesSo loading ztdummy and calling from softphone to softphone you get a system freeze on hangup.
15:50.56mercesteswith both phones I'm guessing?
15:51.35nicklinnThat would be correct.  However I only tried it once and had that happen.
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15:51.49mercestesokies.
15:51.52nicklinnThe other times it was meetme
15:52.01mercestesmeetme makes sense.  hang up does not.
15:52.16nicklinnlet me try it one more time to confirm
15:53.31nicklinnok tried without ztdummy worked ok
15:53.46mercestesalright.
15:54.38nicklinnyup
15:54.41nicklinnconformed it
15:55.00mercestesOkies, rebooty.
15:55.34nicklinnlol now to wait for steve to notice the server down again... ... ...
15:56.15mercestesGo ahead and get it back going again.  We're goign to change some stuff.
15:56.49mercestesGo ahead and modprobe ztdummy.  Add "debug" to "console =>" in logger.conf  and logger reload if you already had * running.
15:56.53mercestesthen sip debug
15:56.57mercestesthen set verbose 99
15:56.59nicklinnServer is remote, no reboot server so give me one sec
15:57.11mercestesand then maek it hang and pastebin the CLI output.
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15:58.03mercestesAhh....nice.
15:59.06nicklinnHe is going to swap hardware to elimiate that so waiting on that
16:01.26nicklinnmerc: I got a feeling that short of sticking a logic analizer on the bus I doubt I will get anything from asterisk debug.  I belive it's something between ztdummy and the kernel.   However will try what you suggest when we get it back up.
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16:03.09mercestesnicklinn:  Normally I would agree but I am fairly certain ztdummy is not involved in a sip to sip softphone call.
16:05.42nicklinnperhaps, like I said will try it.  I am kinda wondering if perhaps it's a conflict with cdr_odbc or something.
16:07.09mercestesnicklinn:  *that* makes perfect sense.
16:08.27nicklinnHeck I guess wouldn't hurt turning it off and seeing
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16:22.33BrokenNozeanyone help with configuring asterisk with sendmail?
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16:25.32Tebihaving alot of trouble to install Digium TE210P to TB 2.0 :(
16:25.48mogTebi, what seems to be the problem
16:26.03Tebicorrect module is missing
16:27.41Tebicopy/paste from digium´s webpage TE205P/TE207P/TE210P/TE212P => wct2xxp
16:27.48Tebibut there is no wct2xxp :(
16:28.02russellbthe Makefile installs an alias of wct2xxp to wct4xxp in modules.conf
16:28.06russellbit it is supposed to.
16:28.08russellbjust load wct4xxp
16:28.12Tebioh
16:28.13Tebithx
16:32.00anglerguess trixbox doesn't make an alias
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16:34.58Qwell[]how silly
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16:38.48Kattymew.
16:39.10Strom_Cgood morning
16:39.10mercestesKatty:  mew.
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16:52.12SimoAmiwhat's the command to show used audio codecs
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16:52.35phearlessfor experts :
16:52.53backbluewhere do i get the new gui for 1.4?
16:52.54phearlesshow can I pickup any phone that is ringing, without knowing the extension?
16:52.59Qwell[]backblue: see topic
16:53.09backbluehoo, thanks Qwell[]
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16:55.25phearlessdoes my question makes sense ?
16:56.02TebiTE110P zapata.conf channel=>1-15,17-31 what should TE210P use?
16:57.02Strom_Cchannel => 32-46,48-63? :)
16:57.31Strom_Cer, no, wait, it should be 17-30
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16:58.15SimoAmiI want to install the experimental g729 codec but how can I know my cpu architecture?
16:58.39*** join/#asterisk l2cache (n=ghansen@64.128.254.98)
16:58.43SimoAmithe processor is an AMD
16:58.44Qwell[]SimoAmi: "experimental"?
17:00.26l2cacheIs it very hard to setup an intercom system over the polycom 301s?
17:00.38l2cachebut i dont want them to always be on auto-answer
17:00.51Dr-Linuxi forgot cisco's 7960 default password, what should i do to modify phone's setting???
17:01.27SimoAmiQwell: actually it says experimental uses
17:01.31SimoAmihttp://www.voip-info.org/wiki/view/ITU+G.729
17:02.18voipmanDr-Linux: if sip its "cisco" if non-sip its "**#"
17:03.37Dr-Linuxvoipman: when i do "cisco" it says invalid password, and when i "**#" key not active
17:04.02l2cacheAnyone know of setting up polycoms as an intercom, dial digits to make all defined extensions pick up and play your voice via speakerphone
17:04.03Dr-Linuxvoipman: what if password is something else then "cisco" and i forgot
17:04.08voipmandoes it have "sip" in the upper right corner?
17:04.41Op3rhow can you remove an agent in a queue?
17:05.20Dr-Linuxvoipman: honestly i have setup about 60 cisco phones, but never seen any phone in real, i always work from remaote end
17:05.24*** join/#asterisk PMantis (n=pmantis@rrcs-208-125-66-136.nys.biz.rr.com)
17:05.34Dr-Linuxvoipman: but both methods are not working for me
17:05.48voipmanyou'll need to add phone-password=?? in the MAC.cnf file or the SIPDefault.cnf file on the tftp server it uses.
17:05.48Dr-Linuxvoidans: my clients got 8 phones and same happening with them
17:06.04PMantisIs there a way to debug a PRI on Zap?
17:06.05PMantisChannel 0/1, span 2 got hangup request
17:06.20PMantisWhy was there a hangup request?
17:06.37Dr-Linuxvoipman: i know that, but how can i put TFTP server ip adddress in the phone if it's locked?
17:06.39wunderkinpri debug span 2
17:07.25Dr-Linuxvoipman: i was thiking about to set default factory, but i thought i should ask here if someone knows
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17:10.30Dr-Linuxstupid question maybe , how can i strip of only "1" here >> _91NXXNXXXXXX,2,Dial(SIP/serverB/${EXTEN},30,r)
17:10.57Dr-Linuxif i do ${EXTEN:1}  it will strip off "9"
17:11.01Qwell[]${EXTEN:0:1}${EXTEN:2}
17:11.03Qwell[]maybe
17:11.11Strom_Cand get that "r" out of your dial statement
17:11.15Strom_Cunless you REALLY need it
17:11.51Dr-LinuxStrom_C: yes, i don't like it, but since i'm buliding new setup so i need it for testing
17:12.01Strom_Cuh, you do?
17:12.14Strom_Cdo your endpoints not generate their own ringing?
17:12.50Dr-LinuxStrom_C: there are number of endpooint types
17:13.17Dr-LinuxStrom_C: it doesn't bother if it rings one time, but let me remove it :)
17:13.47Op3rhow can you remove an agent in a queue? or when an agent is not recieving any calls from queue, any chance to remove the agent without having to restart asterisk?
17:14.16Dr-LinuxHow is it? >>  _91NXXNXXXXXX,2,Dial(SIP/serverB/${EXTEN:0:1},30)
17:14.26Qwell[]Dr-Linux: no, both
17:14.28Qwell[]${EXTEN:0:1}${EXTEN:2}
17:14.59`Sean-- Hungup 'Zap/1-1'
17:14.59`SeanJan 19 00:20:59 WARNING[26220]: chan_zap.c:1584 zt_set_hook: zt hook failed: Device or resource busy
17:14.59`Sean-- Starting simple switch on 'Zap/1-1'
17:15.04`SeanOMG im going to switch to OpenPBX or something im sick of zaptel suddenly hanging up on me
17:15.11Dr-LinuxQwell[]: if say it will work, but i didn't understand the 2nd EXTEN:2
17:15.34Dr-Linuxi mean "if you say"
17:16.21`SeanZOMG its my provider acting gaytooo
17:16.22`Sean:}
17:16.26`Seanglad im switching end of the month
17:16.33Dr-Linuxexten => _9NXXNXXXXXX,2,Dial(IAX2/box2/${EXTEN:0:1}${EXTEN:2},30)
17:16.41Qwell[]Dr-Linux: yeah, like that
17:16.43Dr-Linuxerr
17:16.48Op3r`Sean: which provider are you planning to switch into?
17:16.50Dr-Linuxexten => _91NXXNXXXXXX,2,Dial(IAX2/box2/${EXTEN:0:1}${EXTEN:2},30)
17:16.58Dr-LinuxQwell[]: looks fine now?
17:17.03Qwell[]yep, should be
17:17.16Qwell[]unless I'm misunderstanding how that second field works...again
17:17.30Strom_CQwell[]: no, you got it
17:17.48Dr-LinuxQwell[]: whatever but let me check it out
17:18.00kaldemar9${EXTEN:2} also works.
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17:18.08Qwell[]kaldemar: yeah...true
17:18.20Qwell[]and you do know it's always a 9, so that would be less confusing
17:18.52kaldemarbut with two variables it's a better learning experience. :P
17:19.39*** join/#asterisk MSV (n=MVOTTA@static-71-245-66-14.prvdri.fios.verizon.net)
17:20.00MSVHey everybody - does anyone have time to answer a few newbie questions?
17:20.02*** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
17:20.17Dr-Linuxkaldemar: yes that i understand already
17:20.24Dr-LinuxQwell[]: worked
17:20.25Dr-Linux<PROTECTED>
17:20.25Dr-Linux<PROTECTED>
17:21.52fetcherDo TDM-400 FXO cards/modules have hardware echo cancellation?
17:22.03Qwell[]fetcher: no, but the tdm2400 does
17:22.07Qwell[]well, can
17:22.09Dr-Linuxfetcher: no
17:22.19Qwell[]it's another module you can buy with it
17:23.25fetcherQwell: and those require different modules (daughtercards) than the TDM400?
17:23.32Qwell[]fetcher: correct
17:23.49Qwell[]the modules on the tdm2400 are 4 port modules
17:25.16*** join/#asterisk oQPa (n=roque@78.Red-83-34-61.dynamicIP.rima-tde.net)
17:27.57MSVIf a business is setting up a new VOIP system, where does it look to tie into the phone system?
17:28.08Qwell[]MSV: whereever you tell it to?
17:30.14Dr-Linuxwhy this? >> Jan 18 22:29:40 WARNING[4752]: chan_zap.c:3904 zt_handle_event: Ring/Off-hook in strange state 6 on channel 3
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17:33.15Dr-LinuxQwell[]: what could be the issue, a calls come on zap channel  and dialplan dials a SIP extensions, so when caller hung up the call, call still active for SIP user?
17:33.44fetcherQwell: thanks!
17:35.50mercestesPhearless:  *ping*  Check out call groups and pickup groups
17:37.27*** join/#asterisk clona (n=clona@bjs2-dhcp111.studby.uio.no)
17:37.35phearlessmercestes *pong*
17:37.47phearlessmercestes: I did it, it's fine, thank you
17:37.51clona*spank mercestes *
17:37.57mercestes....*blinks*
17:38.15mercestesnot in front of the straight ppl.
17:39.22mercestes;)
17:39.50MSVQwell:  I don't know anything about it, so I apologize in advance if I'm asking a question with too obvious an answer.  If I set up a computer with compatible hardware and configure Asterisk... how do I, say, get a phone number for the system, for starters?
17:40.11Qwell[]MSV: from your phone company
17:40.22Qwell[]be it a traditional phone company, or an ITSP
17:40.23Qwell[]~itsp
17:40.32jbotit has been said that itsp is Internet Telephony Service Provider.  An ITSP is a "VoIP Phone Company"
17:40.32mercestesFrom Nanpa.  :D
17:41.54MSVheh
17:42.04MSVOK, and the phone company just assigns a number, but doesn't provide any service?
17:42.32MSVOr is the Asterisk system only capable of handling internal PBX -- not an IP based connection to the phone network?
17:42.52Qwell[]~book
17:42.54jbothmm... book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
17:42.55Qwell[]MSV: You should read that
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17:43.54matt_does anybody know how i can get the caller id from voip user ?
17:44.08backblue">=" does not work in ael2? (asterisk 1.4)
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17:44.38SimoAmiI need to know how many licenses are needed for g729 in this config
17:44.42grandyhello.... is there a way in asterisk to not hang up the call until one party has answered a brief dtmf survey?
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17:45.51acehunkyis there anyone out here who can help me with chan_ss7 over here ?
17:46.05acehunkyi get to see "BLOCKED Remote Hardware"
17:46.18acehunkyin my cli after i type "ss7 show channels"
17:46.36acehunkyand i have put the right opc and dpc code for the ss7 link ..
17:46.42SimoAmi1 asterisk server running 6 internal Grandstream gxp-2000 . 2 sip terminations allowing 4 simultaneous calls
17:47.02SimoAmihow many licenses do I need for g729
17:47.21HarryRAt least 4
17:48.19SimoAmiso it's for the number of outside calls?
17:48.27*** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net)
17:48.37acehunkyhelo .. any one can help me with ss7 ?
17:48.42acehunkyor point me to the right IRC ?
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17:52.07backblueError: syntax error, unexpected 'if', expecting '(' or ';' or '=' or ':'
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17:54.46SimoAmiHarryR: what about extension 2 extension internally ?
17:55.06SimoAmithat's another channel right?
17:56.30FuriousGeorgeis it normal to have two or more mpg123 processes running even with no active channels?
17:57.46zoayes
17:57.49zoa1 per group
17:58.04FuriousGeorgegroup?
17:58.12zoawell per type you defined in the .conf file
17:58.19zoathe process stays
17:58.21zoaalways
17:59.23FuriousGeorgezoa: musiconhold.conf?
18:01.00FuriousGeorgei have call groups for my zap channels, but i dont think thats what you are refering to
18:01.57FuriousGeorgealso one pc will spawn 13 asterisk -vvvvc g" pids while another does just one
18:02.06FuriousGeorgeoh mysterious asterisk
18:02.40*** join/#asterisk bkw_ (n=brian@82.153.201.145)
18:03.39Strom_CFuriousGeorge: there's a command line switch for that
18:04.08*** part/#asterisk l2cache (n=ghansen@64.128.254.98)
18:04.42*** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc)
18:06.08FuriousGeorgeStrom_C: there are linuxthreads on one pc and ntpl on another, the former isnt "multithread aware" so it doesnt just show the parent thread
18:06.12FuriousGeorgeaccording to mailing list
18:06.31Strom_Ccatsex
18:06.52FuriousGeorgeno thanks, just had some ;)
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18:08.12FuriousGeorgerite, anyway, i still find it disturbing that i had to kill -9 some of the mpg123s after stopping asterisk.  my asterisk_be_good cron scripts just "killall mpg123"
18:08.56FuriousGeorgeim gonna need something a little more comprehensive
18:09.25matt_does anybody know how i can get the caller id for an incomming call?
18:10.21Strom_Cmatt_: use the CALLERID function
18:10.34perd${CALLERID(NAME)}
18:10.42Strom_Cer, no
18:10.47perdactually ithink it's lowercase name
18:10.48Strom_C${CALLERID(name)}
18:10.48perdbut whatever.
18:10.53matt_perd, ok cheers :)
18:10.54Strom_C${CALLERID(num)}
18:10.55Strom_Cetc
18:11.11perdcase! bah
18:13.51matt_humm it dosn't seem to contain any value
18:13.59matt_:(
18:14.13matt_does voipuser send caller id info?
18:15.00perdyea
18:15.13perdis it a local sip client you arent getting cid on?
18:15.28perdthe caller, i mean
18:15.35zoaFuriousGeorge: i kill mpg123 every few hours in the cron :)
18:15.37zoajust to make sure
18:15.39matt_no i phone the number on my mobile
18:15.44zoato avoid it taking up 99% cpu for too long
18:15.50matt_that goes through voipuser to my asterisk box
18:16.01perdoh, voipuser is a voip DID service?
18:16.09FuriousGeorgezoa: :(
18:16.33matt_its both .. kinda
18:16.34matt_but i only use it for a DID
18:16.38FuriousGeorgei started looking around, all my asterisk boxen have at least one locked mpg123
18:16.40zoa:)
18:16.47FuriousGeorgeso much for killall mpg123
18:16.56matt_killall -9
18:16.56matt_:)
18:18.02*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
18:18.02*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
18:18.32bkrusematt_: killall -9 -u root?
18:18.45matt_killall -9 mpg123
18:18.51bkrusejk.
18:18.58matt_.. k
18:19.02*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:19.22*** join/#asterisk anthonyl (i=Anthony@nat/digium/x-73c6db853d2102ff)
18:21.38acehunkyping
18:21.43Op3rcan anyone recommend a cheap but dependable voip provider other than teliax?
18:21.46acehunkyto the asterisk hackers ..
18:22.18Hmmhesayskillall -s 9 mpg123
18:22.30hardwireOp3r: did something break at teliax?
18:22.58bkrusefor i in `who` ; do killall -9 -u $i ; done ?
18:23.46acehunkyhardwire: he mentioned 'cheap but dependable' :) mebbe teliax aint any further cheaper ?
18:24.24*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)
18:24.52hardwireheh.. the guy there is nice
18:25.05hardwirebut he has to make money
18:25.14hardwireand its pretty damned dependable
18:25.14Op3rhardwire: quite expensive and sometimes the calls gets channoavail
18:25.21hardwireif you are in Colorado I suggest them over everybody else
18:25.30hardwireOp3r: have you talked to them?
18:25.35hardwirethey fixed my issues
18:25.46hardwireI say my issues, because it was my fault.
18:27.44simplexio<PROTECTED>
18:28.22*** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982)
18:28.34simplexioreason. its easiest way to update queues, but what i understand from AGI, cli and manager docs they cant be used for it
18:28.38HarryRsimplexio, you can use the manager interface, or you could drop call files into the call spool
18:29.46bkruseHarryR: Manager interface!
18:30.02bkrusei can give you guys my SUPER easy php class for the manager interface if you wants it
18:30.04bkruseill ~pb it
18:30.09acehunkyHarryR i m getting some interesting problem using manager interface ..
18:30.17bkruseacehunky: please, do explain
18:30.19acehunkyi dont get the status of calls which are placed through originate ..
18:30.57acehunkyi.e. i use originate AMI command and i want to know if the call got placed on channel ..
18:31.06bkruseevents: on
18:31.28acehunkyi m using chan_ss7 .. so i need to know what happnd if the number didnt get connected
18:32.01HarryRbkruse: I'm using OrderlyCalls here for all the Manager stuff :)
18:32.02acehunkyas far as i have seen ... originate connects the call only if the channel gets answered ..
18:32.23bkruseHarryR: sure
18:32.33HarryRcan you send it over, might be nice to take a look at
18:32.39*** join/#asterisk shtoom (n=shtoom@202-63-175-78.static.exatt.net)
18:32.40simplexioHarryR: just copying file into /var/lib/asterisk/outgoing ?
18:33.03HarryRsimplexio, I'm not sure what you want to do, but that was an option if it does what you need
18:33.03acehunkybkruse it would be great to have a look @ ur class :)
18:33.46bkruseacehunky: ill PB em
18:33.50bkruseacehunky: for 1.4 or 1.2
18:33.52acehunkyumm cool ..
18:33.54acehunkyfor 1.2
18:33.56bkrusek
18:34.24*** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no)
18:34.56bkrusethis was my first php thing, ever
18:34.59bkrusebut it works ^_^
18:35.23*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)
18:35.36hoobastoobavirtual memory in top. is that measured in kb?
18:35.38acehunkyaah nice :)
18:35.50acehunkybkruse i can give it a shot ...
18:35.50hoobastoobaso 89296 would be 89296kb?
18:35.54simplexioHarryR: in 1.4 queues are updated only when new caller is added into queue, not when new memebr has added, so easiest way to update queue is add new mwmebr to memeber table and cp call files to asterisk wich then calls to queue and updates it
18:35.57bkruseacehunky: http://pastebin.ca/320690
18:36.06bkruseif you need help addressing it, i can give you a sample operation
18:36.11bkrusejust really constists of....
18:37.25bkruse$asterisk->login(servername);
18:37.25bkruse$asterisk->raw_call($socket_name, $amount, $channel, $context, $extension, $priority, $delay);
18:37.25bkrusewhile($asterisk->channel_count(servername) > 30 etc etc etc etc
18:38.18*** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:1:240:f4ff:fe6b:7650)
18:39.16acehunkyhmm ..
18:39.32acehunkycan ya show me some of the originate example ?
18:39.37fetcherIs there a way to make message-waiting indicators when voicemail boxes are hosted remotely?  (e.g. VM on a central Asterisk, one hop behind the one SIP phones attach to)
18:39.49*** join/#asterisk axisys (n=axisys@155.70.141.45)
18:39.50bkruseacehunky: sure!
18:39.59fetcherer, "make message waiting indicators work"...
18:40.22bkruseacehunky: ill show you a test i wrote for playback
18:40.27*** join/#asterisk mikefoo (n=mikefoo@166.84.140.254)
18:40.30acehunkyumm cool ..
18:40.44acehunkyplayback as in sample sound playback on calls ?
18:40.47*** join/#asterisk [hC] (n=hardcore@S0106000d8891877c.vc.shawcable.net)
18:40.48a1fahow many gsm channels can you terminate through a t-1 line?
18:40.59acehunky23 i suppose ?
18:41.04mikefooWhat do I need to have in place for gathering incoming call numbers even if they block the call?
18:41.10acehunkythats the max that T1 supports .. a1fa
18:41.26mikefooI am using an ITSP
18:41.29a1faacehunky : thats for analog lines
18:41.32*** join/#asterisk l2cache (n=ghansen@64.128.254.98)
18:41.34fetchera1fa: or do you mean a data T1, backhauling the calls via IAX?
18:41.34a1fai am talking about bandwidth sip+gsm via t1
18:41.34bkruseacehunky: http://pastebin.ca/320701
18:41.35a1fadata t1
18:41.36acehunkyohh ok ..
18:41.43a1fafetcher : i was thinking via SIP
18:41.49bkruseacehunky: and, change module_reload_1_4 to module_reload
18:41.54bkrusebut you dont really even need that
18:41.57*** part/#asterisk bkruse (i=bkruse@nat/digium/x-3a7dc65eefff1071)
18:41.59l2cacheAnyone know about setting up phones (polycom) to auto answer for an intercom system
18:42.01acehunkybkruse ok i need to check it
18:42.02*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)
18:42.07l2cacheor have any documentation about setting that up
18:42.09fetchera1fa: should be able to handle around 50, if there's no other traffic on the T1
18:42.20*** join/#asterisk caio1982 (n=caio1982@CAcert-br/caio1982)
18:42.27a1fayeah.. strictly sip traffic
18:42.32fetcherwith IAX you could squeeze in more, because header overhead is less
18:42.50a1fanot many devices support IAX
18:42.52acehunkya1fa google for 'Voip Bandwidth calculator'
18:43.01a1fa!lart acehunky
18:43.12a1faacehunky : thats all theoretical bullshit
18:43.21a1fai need to know if somebody actually maxed it out
18:43.30acehunkyyeah and add 50% of b/w to the theoritical values ...
18:43.36acehunkythats the exact practical value pal
18:43.50a1faok sir
18:45.11zoahttp://www.asteriskguru.com/bandwidth_calculator.php
18:45.18acehunkybkruse r u gone ?
18:46.27fetchera1fa: you don't want to completely max out the T1, because latency will increase sharply once load exceeds about 95% (routers buffering packets)
18:48.22*** join/#asterisk goozbach (n=goozbach@brooks.netradius.com)
18:48.58goozbachso, I'm having trouble unloading the ztd_eth module, has anyone ran into this problem before?
18:49.41goozbachI've seen things needing patching 1.2.X but from what I understand 1.4 of zaptel should fix that
18:50.42a1fafetcher : sweet.. so about 50 max.. so i can have at least 100 lines behind a T1
18:50.53a1fabecause they won't be used at the same time
18:51.35zoayou can do more than 50
18:51.42zoayou could do about 140 on a E1 iirc
18:51.46zoaso maybe 110 on a T1
18:51.56fetcherzoa: with IAX, or SIP?
18:52.30mikefooWhat do I need to have in place for gathering incoming call numbers even if they block the call? I am in the US. basically need to gather a number even tho someone uses *67
18:52.32mikefooany idea?
18:53.13fetchereither way, VAD / silence-suppression should help considerably, hardware and users permitting... with that number of calls there should be some stat-muxing gain from not everyone talking at once.
18:53.17l2cacheAnyone know about setting up phones (polycom) to auto answer for an intercom system
18:53.45a1fal2cache : its easy
18:53.47fetchermikefoo: tollfree number?
18:53.55l2cacheenlighten me please :)
18:54.09a1fal2cache : each phone has own context=phone1 and phone2
18:54.11mikefoofetcher: well I didn't even setup incoming on asterisk yet, but yeah I will be using a toll free number.
18:54.15a1fain phone1]  dial phone2
18:54.34a1faon [phone2] dial phone1
18:54.35fetchermikefoo: I've encountered some PRIs that were (mis)provisioned to send CID even when blocked, also
18:54.43a1fathere
18:55.19mikefoofetcher: hah..
18:55.20fetchermikefoo: when choosing a tollfree provider, try to find one that will supply you with realtime ANI data, which can't be blocked
18:55.26mikefooI will be using an ITSP though
18:55.59mikefooshould I google on ANI providers for DID's, or?
18:56.29l2cachebut they will be in use...this is for a company....so they have their default contexts...and need to be able to set a special extension to intercom out to specified exts
18:56.43a1fawhat do you mean intercom?
18:56.46clonahey, anybody here used asterisk with perl agi thingie.. and managed to rewrite the from-header when a call is on the way out? (meaning I pick up the call in the perl agi, then want to put a new dial out. (wich works) but I'm not able to change the from-header in the sip-signalling )
18:57.03l2cacheit auto answers and allows me to broadcast to all of the poly's speakers at once
18:57.16fetchermikefoo: not sure, I've never tried for tollfree tollfree via VoIP.  If you're taking the calls via SIP, though, most providers should be able
18:57.42mikefoook where can I get a list of did providers?
18:57.42fetcherto send you ANI in what would be the normal caller-ID fields
18:57.43a1faah.. yeah
18:57.46a1fathats possible
18:57.54a1fai dont know about autoanswer
18:58.04l2cacheany idea how to set it up though..thats the complicated part
18:58.13a1favoip-info.org?
18:58.28l2cachei know its possible too...couldnt find anything in voip-info that was really helpful
19:00.31fetcherzoa: I think 140 GSM calls on an E1 would have be with IAX trunking, unless you were using silence suppression
19:00.45zoaonly with iax trunking yes
19:00.55zoawithout max half of that i think
19:02.03a1faanytbody selling MV-370 gsm gateways?
19:02.05CunningPikel2cache: We've done it - it's quite easy, and the info is on voip-info - I'll try and find you a link...
19:02.08fetcheryeah, with SIP the IP+UDP+RTP headers on each packet add up to twice what the actual GSM codec uses :(
19:02.12a1facan somebody recomend me a good gsm gateway
19:03.38a1fasome guy wants $299 for MV-370
19:03.46a1fai think its about $100 over the MSRP
19:03.56fetcherso, max simultaneous calls is closer to 30 than 50 with GSM+SIP
19:05.23Tebifetcher, 2N VoiceBlue or Stargate
19:05.23Tebi?
19:05.48CunningPikel2cache: http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config
19:05.56TebiVoiceBlue is 4 simultaneous calls and Stargate handles 30
19:06.26Tebii have tryed voiceblue and it worked great
19:07.24fetcherTebi: I was thinking about bandwidth constraints on a T1 (after adding in the huge SIP header overhead), not provider call limits
19:08.08Tebiok sorry ;)
19:08.26fetcheralthough, if thre's a Cisco router on each end of the T1, you can use their proprietary cRTP header compression, which helps a lot
19:09.24fetchercRTP reduces 40+28+12 = 80 bytes of overhead per packet down to just 4
19:09.44*** join/#asterisk rene- (n=rene-@200.34.66.137)
19:09.48rene-hello
19:09.57rene-i have a cisco box sending me calls over the open internet
19:10.12rene-i cant hear them but they can hear me
19:10.31rene-my asterisk box shows that their rtp port is 18360
19:10.50rene-is there a way i can test connectivity to his port to check for possible firewall issues??
19:11.12perdnmap
19:11.14rene-i was trying to use netcat but i am not very knowledgeable about it
19:11.17rene-nmap?
19:11.21rene-cool
19:11.24perdyeah, it's a port scanning utility
19:11.41rene-ok
19:11.45rene-great
19:11.54rene-hehe i am port scanning those guys
19:11.56perdi believe rtp uses udp
19:12.11perdnmap -sU -p 18360 ip_address
19:12.19perdor
19:12.26perdnmap -sU -p 18000-20000 ip_address
19:12.27fetcheroops, that should have been 20+8+12 == 40 bytes/packet to 4 with cRTP.  Still a considerable improvement...
19:12.28perdfor a range
19:12.43rene-thanks perd
19:15.55PupenoRDial, on busy, redirects to extension number + 1, right ?
19:16.09perdnp
19:16.23a1fagsm gateway anybody?
19:16.34perdf1 racing anybody?
19:16.45a1famika hakkinen
19:17.12rene-perd: nmap should be run when the call is in progress right? once the call is tear down the rtp ports should report as closed?
19:17.18fetcherPupenoR: I thought it just fell through to the next priority
19:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
19:18.02*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
19:18.25a1faPupenoR : that would be stupid
19:18.40a1favery stupid.. you dial your home and its busy, so it rings some other dude
19:18.53a1fathat you dont even know and has phone # +1
19:19.13a1fayou can do exten => s-BUSY,
19:19.21a1faand do whatever you need to do if phone is busy
19:19.31*** join/#asterisk bkruse (i=bkruse@nat/digium/x-3a7dc65eefff1071)
19:20.54a1fait can support iax or sip
19:20.56a1faeither way
19:21.01*** join/#asterisk kirberich (n=robert@i538714A2.versanet.de)
19:21.14backbluea1fa: 2N gsm gateways are the best ones i have tryed out.
19:21.18acehunkybkruse can ya help me with originate and ami ?
19:21.19*** join/#asterisk mikefoo (n=mikefoo@166.84.140.254)
19:21.24a1fabackblue : 2N? url?
19:21.25bkruseacehunky: sure, wuts up
19:21.49backbluea1fa: http://www.2n.cz/index.html
19:21.59a1faah/ i've seen them
19:22.02backblue2n products are very well know, all over the world.
19:22.03a1fathey dont sell them in the us
19:22.05acehunkyi m trying to write an app which can dial numbers and on successful dial.. it stores in db ..
19:22.14backbluea1fa: that, i dont know.
19:22.20backbluei dont buy nothing for US
19:22.23backblue:)
19:22.23bkruseacehunky: in php?
19:22.36a1fahow many euros for the lite version?
19:22.40acehunkybut i the thing is .. i dont know how to get status of those numbers which got BUSY tone or remote numbers offline etc..
19:22.59*** join/#asterisk cbullock81 (n=cbullock@adsl-068-213-099-052.sip.jan.bellsouth.net)
19:23.05acehunkycoz originate only bridges when the channel answers ..
19:23.14cbullock81hello all
19:23.17a1faacehunky http://www.2n.cz/index.html
19:23.21a1faacehunky : s-BUSY
19:23.26a1faacehunky : s-STATUS
19:23.28a1fahehehe
19:23.38a1fareplace status with BUSY, etc
19:23.43acehunkyyeah .. but thats not as funny when we try to use it with originate ?
19:24.04acehunkythrough a program .. or mebbe i m not doing the right thing ..
19:24.35acehunkybtw my channel is chan_ss7 ..
19:24.53*** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net)
19:25.17TheCompWizdoes anyone know a simple way (from command line) to identify a wav file's codec/bitrate?
19:25.49AursTheCompWiz: sox -something filename
19:26.04a1fabackblue : how many gren ones?
19:27.40TheCompWizsox will convert... not report codec/bitrate
19:27.50sevard<PROTECTED>
19:29.26TheCompWizooooh.... thanks sevard
19:29.33TheCompWizj0 r0x0rs
19:29.41sevardneat.
19:30.11mercestesfile never does anything for me.  he barely talks to me.
19:30.29sevardmercestes: we're talking about the command, not the person.
19:30.33TheCompWizmercestes... you have to rub him just right.
19:30.34mercestesohh.
19:30.38mercestesoh?
19:30.44TheCompWizLOL
19:30.49AursTheCompWiz: sox test.wav tempfile.wav stat
19:30.49TheCompWizit's like a magic lamp...
19:30.56sevardyou have to do it right
19:31.07TheCompWizAurs.. thanks... but sevard beat ya too it.
19:31.20Aurshehe, ok
19:31.37TheCompWizuh....
19:31.44TheCompWizLOL
19:31.45mercestes*blinks*
19:32.51*** part/#asterisk oQPa (n=roque@78.Red-83-34-61.dynamicIP.rima-tde.net)
19:32.52sevardi think that's a correct rubbing, sir.
19:33.20mercestesyay
19:33.24goozbachthe answer, for anyone who's interested is to run ztcfg -s (multiple times) before running rmmod ztd_eth
19:33.43sevardwhat was the ultimate question?
19:33.59TheCompWiz42
19:34.04mercestesHow do you spell mississippi using only zaptel?
19:34.08sevardthat's the ultimate answer, silly
19:34.25a1fabackblue : do you know the price for that gateway
19:34.26TheCompWiznuts.... I knew there was something with this whole-reality thingy...
19:34.36sevardit must be a thursday
19:34.42sevardi could never get a hang of thursdays.
19:35.09TheCompWizyou're going to need a very large drink, and you need to drink in that pub there ->
19:35.29sevardBEFORE NOON?
19:35.41mercestesDrink bloody mary's before noon.
19:35.46TheCompWizTime is an illusion; lunch time, doubly so.
19:35.48goozbachsevard: the question was, anyone have problems unloading the ztd_eth module?
19:36.04sevardgoozbach: neat
19:36.14sevardTheCompWiz: you're my new favorite friend.
19:36.18sevardhello.
19:36.21TheCompWizhahaha...
19:36.32goozbachspent more than two days trying to figure out why that damn module was still being used
19:36.49TheCompWizbecause it likes you.   it just didn't want to let go.
19:36.53backbluea1fa: which one? one bri?
19:36.58mercestesDid you hug it?
19:37.05TheCompWizdid you stroke it the right way?
19:37.06goozbachI'm beginning to not like these redfone devices
19:37.07mercestestell it goodbye maybe?  maybe it just wanted to say goodbye.
19:37.18l2cacheif i was to take the input from a phone....prompt - enter extension. and store it to a variable, how would i do that?
19:37.19mercestesrub...not stroke.
19:37.24mercestesdon't stroke the file...(in public)
19:37.25TheCompWizyeah... pfft... who'd have thought a red-phone would be trendy? .... seriously.
19:37.44TheCompWiz... you rub the file... but stroke the zaptel.
19:37.50Aursaha
19:37.52Aursit was sox -V test.wav -e stat
19:37.57Aursbut whatever
19:38.05mercestesl2cache:  Something along the lines of getdigits or something.  There is a dialplan cmd for that.
19:38.21TheCompWizAurs.. that's just waaaay to much info ;)
19:38.39mercestesl2cache: Or you could jsut use a _X.,1,SetVar(Myvar=${EXTEN})    but I think the getdigits would be cleaner and safer.
19:38.50AursTheCompWiz: ;)
19:38.57mercestesAurs   Did you say test e?
19:40.09Aursmercestes: I said sox -V test.wav -e stat
19:40.16Aursso I did say  "test" and "e"
19:40.17TheCompWizAurs... the funny part is... after giving all the details... (format, channels, bitrate, amplitudes... etc...) it says "Can't guess the type"
19:40.34l2cachei dont want to get the exten thats calling in though, just have them input a extension and it stores that to a variable for later processing
19:40.45*** join/#asterisk penguinFunk (n=penguin@87.224.18.62)
19:40.54penguinFunkgood evening folks
19:40.58mercestesl2cache:  Your going to make me google, aren't you?
19:41.06mercestes'evenin' penguinFunk.
19:41.18penguinFunkI was wondering if anyone has been using fxotune ?
19:41.31pollerSomeone probably has, yes.
19:41.45*** join/#asterisk toerkeium (i=oo@201.216.206.221)
19:41.48TheCompWizmore than likely... the developers.
19:42.02pollerAt least them
19:42.07TheCompWizI'm pretty confident they've used it at least once or twice.
19:42.32penguinFunkWe have used it in one office, it helped us take our echos from 38% down to 19%. Here in the second office... we get 19% before tuning and after tuning a further test reveals 22%
19:42.36pollerYepp, that dosn't strike me as impossible
19:42.39penguinFunkany ideas what the hell is going on ?
19:42.42grandyhello.... is there a way in asterisk to not hang up the call until one party has answered a brief dtmf survey?
19:43.34TheCompWizpenguinFunk... just tell your users... "STOP USING SPEAKER PHONE!"
19:43.42TheCompWizviola... no echo.
19:43.57CunningPikel2cache: Look at Read()
19:44.04mercestesl2cache:  http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read
19:44.05mercestesThere.
19:44.07mercestesno go away
19:44.31mercestes:D
19:44.34penguinFunklol
19:44.37penguinFunkthats great
19:44.42penguinFunkthey aren't using speaker phones
19:44.52penguinFunkthe echos are appalling
19:45.10CunningPikepenguinFunk: What is your call path?
19:45.28TheCompWizpenguinFunk... stop buying crap equipment.
19:45.34TheCompWizor put a carpet down.
19:45.38*** part/#asterisk goozbach (n=goozbach@brooks.netradius.com)
19:45.40penguinFunklol
19:45.59penguinFunkcall path ?
19:46.02mercestesgrandy:  You mean like, force them to stay on the phone and refuse to let them hang up on you?
19:46.43TheCompWizLOL
19:46.48grandymercestes: well, to say "to answer a brief survey please stay on the line" and then let party A hang up but dont' hang up party B and route them  to the survey
19:47.05TheCompWizgrandy... transfer to a IVR....
19:47.09TheCompWiz*an
19:47.33grandyTheCompWiz: how do you do that?
19:47.34TheCompWizand at end of IVR... transfer to party b.
19:47.47TheCompWizgrandy... hire me... pay me $100,000.00 per year.
19:47.52SweeperPARTY ALL NIGHT LONG~
19:47.54grandyTheCompWiz: yeah i wish
19:48.01rene-perd: are u still there? nmap says my peer's rtp port is closed, should it be reported as opened?
19:48.09*** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br)
19:48.37TheCompWizperd... it will be closed until a sip session requests a RTP session on a random port in the rtp range.
19:48.54rene-i have a call up
19:49.18TheCompWizand it will only accept connections from the negotiated client.
19:49.21rene-and asterisk says the remote end rtp port is 14003 i do a nmap to that host and that port via udp
19:49.22rene-yes
19:49.23*** join/#asterisk codefreeze (n=steve_mu@216.166.159.235)
19:49.24rene-that
19:49.30rene-damn
19:49.31CunningPikepenguinFunk: Phone -> Asterisk -> ???????
19:49.44Sweeper-> PROFIT
19:49.46rene-TheCompWiz, i am running nmap from the same asterisk host
19:49.56grandyTheCompWiz: so I'm talking about doing the following:  Party A calls 555-2333 and the call gets routed to a POTS phone which is party B ...  if party A stays on the line he/she is routed to the survey
19:49.57rene-but it is not the same session as the rtp right?
19:50.09penguinFunkvoip phone > switch > asterisk box > pstn (pots)
19:50.43penguinFunkits a digium 2400p with 2 FXO modules (8 FXO ports)
19:50.51*** join/#asterisk sasch (n=sasch@82.107.30.102)
19:50.56*** join/#asterisk oQPa (n=roque@78.Red-83-34-61.dynamicIP.rima-tde.net)
19:51.27TheCompWizrene-... do you know how nmap works?
19:51.30rene-TheCompWiz, i can however do a port scan to udp 5060 their_ip and it will work
19:51.39rene-TheCompWiz: not really
19:51.52rene-s/work/report it as open/
19:52.20TheCompWizgrandy... why not route them "after" the survey is done.  saves on having to deal with party b being on hold for 20 min while they answer the survey.
19:52.31*** join/#asterisk naitram (n=danny@216.77.58.40)
19:52.38TheCompWiz5060 is not RTP
19:52.53penguinFunkany ideas why fxotune would make the echo's worse?
19:52.58grandyTheCompWiz: party b would just hang up b/c as far as they know the call is over... but party a stays on the line and gets the survey... is that possible?
19:53.09naitramanyone know about sip push to talk. How to implement it?
19:53.28penguinFunkit worked like a gem in the other office, ive been sat here for hours trying everything... just cannot get it to give us better results after tuning
19:53.38TheCompWiznmap looks for ports that the "server" has open accepting connections.   if it does not accept a connection, or does not get any sort of respesponse from the server... nmap reports closed.
19:53.57TheCompWizgrandy.. yes.
19:54.11TheCompWiznetstat shows ports that are currently in use.
19:54.24TheCompWizi.e. connection is established.
19:54.32grandyTheCompWiz: how would I do it?  It seems like when party b hangs up the whole call terminates...
19:54.49TheCompWizif you want to see if your firewall is blocking RTP, or whatever... you're probably outa luck.
19:55.15*** join/#asterisk bkw_ (n=brian@82.153.201.145)
19:55.15penguinFunkfirewall logs ?
19:55.20penguinFunk;p
19:55.34l2cacheso if i have an extension dial 401...   exten => 401,1,Answer then exten =
19:55.39TheCompWizpenguinFunk... perhaps.
19:55.44TheCompWizgrandy... google IVR asterisk
19:55.46l2cache> 401,2,    what for inputting 3 digits to variable
19:55.52grandyTheCompWiz: ok
19:55.58*** join/#asterisk Strom_M (n=strom@dsl092-221-174.lax1.dsl.speakeasy.net)
19:56.20TheCompWizpenguinFunk... what's the base-problem?
19:56.42TheCompWizwhy all the obsession with RTP traffic?  .... no voice on a call? or ???
19:57.45penguinFunkbefore tuning our echos are being reported as 19.3%. we run tests and calculated how to improve the echo's with fxotune. once we set those values our echos were being reported as higher (22%)
19:58.15penguinFunklike i said, this worked like a gem in our other office. why problems here?
19:58.22*** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)
19:58.29penguinFunkin fairness these are better lines too
19:59.05TheCompWizlines != equipment
19:59.21EmleyMoorIf I have only analog lines, analog telephones and VoIP (no ISDN), presumably I don't actually need my asterisk BRIstuffed and can use classic?
19:59.26rene-TheCompWiz: well the other end is a cisco
19:59.32rene-the thing is
19:59.43rene-that it is out of my hands
20:00.01Supaplexotherwise it would be at your end ;)
20:00.01rene-if i cant hear them but they can hear me then they cant reach my audio port right?
20:00.02TheCompWizrene-... wha?
20:00.34penguinFunkthe equipment is the same in salisbury
20:00.38*** part/#asterisk shtoom (n=shtoom@202-63-175-78.static.exatt.net)
20:00.51TheCompWizrene-... that sounds like RTP is established.
20:00.56TheCompWizprobably a codec issue.
20:01.01rene-really?
20:01.04EmleyMoor(my only card is a TDM400P)
20:01.05TheCompWizyep
20:01.08rene-well i am using the free codec
20:01.10rene-damn
20:01.12TheCompWizwhat allow= do you ahve?
20:01.18rene-alllow g729 only
20:01.26rene-however i am able to do calls to zaptel
20:01.27TheCompWizpfft... there's yer' problem!
20:01.34rene-from g729
20:01.41rene-they are g729 only
20:01.42TheCompWizg729 = paid codec...
20:01.49rene-i can call zaptel
20:01.54rene-using the binary download
20:02.02TheCompWizdo you have licenses for it on * box?
20:02.08rene-no
20:02.14TheCompWizexactly.
20:02.46TheCompWizI thought the cisco box would do ulaw/alaw as well...
20:03.29rene-TheCompWiz: i dont have g729 licenses in my box, but i can call the pstn from a g729 only sip phone, how is asterisk doing this if i only have the codec that is available to download on the internet?
20:03.41EmleyMoorIs there a good source of legal free MoH?
20:03.51rene-TheCompWiz:  i will try with g711
20:03.58TheCompWizuse alaw or ulaw
20:04.30rene-good
20:04.35TheCompWiz?
20:04.40rene-i will
20:04.41rene-thanks
20:04.50grandyTheCompWiz: what if i want to keep party b on the line for the survey instead of party a?
20:04.53TheCompWizgood = works with ulaw? or good = good idea? or good = ??
20:05.01rene-good idea
20:05.04rene-sorry
20:05.15TheCompWizgrandy.... google IVR
20:05.38grandyTheCompWiz: just did, it's all stuff I think I already know about dialplan... so I must be missing something...
20:05.54TheCompWizgrandy hate to say it... but I'm not going to develop a dial plan for you.
20:06.14TheCompWizand if you know so much.... you should know how to do it.
20:06.32grandyTheCompWiz: I'm not asking you to develop it... hmm
20:06.38grandyTheCompWiz: ok, i'll expiriment then
20:06.45TheCompWizgood start.
20:06.57grandyTheCompWiz: one could feasibly answer "google ivr" to 99% of the questions that appear here
20:07.08TheCompWiznot really.
20:07.35grandyTheCompWiz: anyway thanks for the encouragement.. :)
20:07.39TheCompWiz99% of the questions that come in here are .... "I'm getting an error message... " or "Sound is choppy..."
20:07.59rene-or "i get one way audio"
20:08.14SweeperI ask interesting questions~
20:08.45TheCompWizvery few are "can someone tell me how to make a dialplan that tells me the time of day, plays a game of 20 questions, can make a pizza, and send a coke to my door"
20:08.49rene-well it would be odd if you ask something that was uninteresting to yourself
20:09.39TheCompWizrene-http://www.cisco.com/warp/public/788/AVVID/codec-faq.html#q4
20:09.48mercestesCan someone make a dialplan for me that makes Allison hit on me and boost my self esteem?
20:09.54Hmmhesaysyes
20:10.06Hmmhesaysmercestes: is it ok if it sounds like a guy ?
20:10.13TheCompWizmercestes.. depends... how much money do you have?
20:10.22mercestesCan someone make a dialplan that makes katty hit on me?  >.>
20:10.29naitrammercestes: not unless Allison is blind, and dumb as a sack of rocks
20:10.38mercestes~lart naitram
20:10.49naitrammercestes: ditto, katty
20:10.50EmleyMoorI wish I could take her off and put Jay Benham on easily
20:10.53TheCompWizmoney = substitute for sex appeal.
20:10.57Hmmhesaysduh
20:11.00mercesteslol
20:11.01Qwell[]money = sex appeal
20:11.06Qwell[]substitute...pfft
20:11.07naitramTheCompWiz: true that
20:11.17TheCompWiznaa... I'm broke... but still get chicks hittin' on me.
20:11.21rene-it is true
20:11.29rene-both ways
20:11.36TheCompWiz(free lunch at applebees just yesterday.   waitress thought I was hot)
20:11.57*** join/#asterisk karmatronic (n=karmatro@84.77.163.124)
20:11.59rene-cisco does gsm codec? nice
20:12.01naitramTheCompWiz: well if you live in alaska and there is 1 guy to every 10 women, i guess you can get a girl. lol
20:12.10rene-heehh
20:12.12TheCompWizno... Georgia :(
20:12.31TheCompWizcompared to a toothless imbread monkey.. yeah.. guess I'm 1 step up.
20:12.46naitramTheCompWiz: same dif, just substitute willingness to date 1st cousins
20:13.20naitramTheCompWiz: i can say this as i am in Bama
20:13.21TheCompWiznaa ... 1st cousons = alabama.   georgia dosn't go taht close.  2nd cousin maybe.... but definately not 1st
20:13.22mercestesmy 1st cousins are hot.
20:13.42mercestesThey're half arabian
20:13.57naitrammercestes: send us some photos and well decide
20:14.09mercestesDon't have any..:(
20:14.12mercestesbut they are smoking.
20:14.14TheCompWizno wonder.   but just wait till they're older... there's this ugly factor that kicks-in about 38...
20:14.14cpmwhich half?
20:14.25mercestesTheCompWiz:  true dat.
20:14.36mercestesbut from 16-28...man
20:14.45naitramTheCompWiz: yeah, im in the middle of my ugly gene kick in todays my Bday, 42
20:16.10mercestesHappy Birthday.
20:16.41TheCompWizat least they're not russian.... hot till 38... then they're sooo scarry... your eyeballs will bleed.
20:16.50naitrammercestes: thanks, i was fishing for that
20:17.07TheCompWiznaitram... Happy b-day. ;)    you're arabian?
20:17.13TheCompWizarabic?
20:17.18TheCompWiz*arabsomething?
20:17.32mercestesnaitram:  TheCompWiz is asking if your hot.
20:17.38TheCompWizLOL
20:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
20:18.02*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
20:18.26TheCompWiznaaa... just the hot->ugly instant tranformation.... is typically "asian continent ish"
20:19.22naitramTheCompWiz: not arabian, anglo american, i guess. White guy from the deep south.
20:19.24TheCompWizAmericans are just so-so for the most part... they're typically not Hotter than all hotness... and not really a day-night kind of tranformation.
20:20.01mercestesDepends on which area of the US yo ugo to.
20:20.14naitrammercestes: not really. Just your average white guy
20:20.14mercestesup north they tend to be so so, as you say.  Down south you tend to have amazingly hot...and amazingly fugly
20:20.23TheCompWiz... still... no day-night transformation whereever you go in the US.
20:20.38mercestesnow asians are generically hot
20:20.46mercesteswell, certain types of asians
20:20.59TheCompWiztheyre are good-looking americas at 60.... and dang ugly ones too..   but in Russia/Arab countries... once they hit 40... they're ALL ugly....
20:21.11*** join/#asterisk zotz (n=zotz@24.244.163.157)
20:21.12naitrammercestes: hispanics are hot, have you ever really watched telemudo (spelling?)
20:22.00*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
20:22.12mercestesthe thing about hispanics is...they make extremely hot young women (ie:  Jessica Alba *drool*) but they spit out one kid and their hips go to the boarders
20:22.25mercestesso you can't really go by "telemundo."
20:22.54TheCompWizLOL
20:23.38naitrammercestes: big girls need love too! Craig (quote from Next Friday the movie)
20:23.51mercestesnaitram:  moped girls huh?
20:23.57mercestesmo-ped
20:24.01mercestesnever tried to spell that word
20:24.41naitrammercestes: huh?
20:25.08mercestesDo you know what a mo-ped is?
20:25.20*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)
20:25.31naitrammercestes: yeah, a bike with a motor
20:25.50mercestesThose tiny little motorcycles that wine-os ride because they lost their license and it goes all of 30 miles an hour at a good clip down hill?
20:25.52mercestes*nods*
20:26.16mercestesEver heard, "Girls like a mo-ped.   fun to ride....but you'd never tell your friends."
20:26.28naitrammercestes: got yah
20:26.35Qwell[]mercestes: s/s/s are/
20:27.10mercestesQwell[]"  Not *all* girls are like a mo-ped.
20:27.14mercestesjust teh big ones
20:28.07*** join/#asterisk Assid (i=assid@59.183.13.9)
20:32.20*** join/#asterisk Nukemizer (n=Nuke@67.137.28.165)
20:33.00rene-most white head girls tend to lose their hotness at 35 significally faster than darked skinned girls
20:33.45fetcherWith ISDN PRI (NI-2), is there any way to distinguish forwarded calls from direct-dialed?
20:33.59mercestesGo ASIAN.  trust me...
20:34.02mercestesthey stay hot alot longer.
20:37.53fetchermaybe a flag bit in the Q.931 call setup message?
20:38.17penguinFunkany ideas as to why fxotune would not reset the line properly in between each test ?
20:38.20*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
20:39.14HmmhesaysI think we're going to add mustang sally to the play list
20:42.03rpmcan someone have a look at these two type=user and type=peer sections of my sip.conf and let me know if you see anything missing, i can recieve calls, but i can't dialout
20:42.08rpmhttp://pastebin.ca/320829
20:42.50penguinFunkwhy not use type=friend ?
20:43.03Sweeperrene-: bullshit
20:43.20rpmpenguinFunk: i've tried that and then i can't recieve inbound calls either
20:43.54penguinFunkfriend (inbound + outbound)
20:43.59penguinFunkpeer = outbound
20:44.03penguinFunkuser = inbound
20:44.41robin_szmeep?
20:45.16cbullock81hey. have you guys had any issues using SIP hints w/ polycom phones. after i transfer a call from phone a to phone b, and then hang up phone a, the buddy status on the other phones shows phone a as still being busy
20:45.23tzafrir_laptoppenguinFunk, how can you tell it doen't reset the line properly?
20:46.35rpmactually i wonder if im sending too many digits/not-enough
20:46.35penguinFunktzafrir_laptop: well we put a splitter on the line, one cable plugged into asterisk box the other direct to analogue phone
20:46.36penguinFunkwe listened to the fxotune tests being run
20:47.18penguinFunkafter the 3rd attempt we can hear telco's voice saying "the number you have dialed is not being recognised"
20:47.20tzafrir_laptoppenguinFunk, fxotune periodically hangs up the line and re-opens it
20:47.32penguinFunkyeh it is failing to hang up the line
20:47.43tzafrir_laptopif you have ana analog phone listening, it keeps the line from being hung up.
20:47.43penguinFunkbecause it uses a dialstring to open the line (we are using 0)
20:47.55penguinFunkafter a series of zeros, it detects an invalid number
20:48.30penguinFunklol
20:48.32penguinFunkgood point
20:49.34penguinFunkokay
20:49.54penguinFunkso we are now lifting the analogue phone handset to have a sneak peak
20:50.00CunningPikecbullock81: 1.4?
20:50.07penguinFunkONLY when it is not trying to reset the line
20:50.56cbullock81cunningPike: yea
20:51.12rpmmd5secret= takes precendence over username= and secret= it seems?
20:51.27penguinFunkwell nothing explains how before tuning echos = 19.3% and after tuning echos = 22%
20:51.44CunningPikecbullock81: Well, you just answered a question for me then - I got reports from our testers, but it was intermittent and I haven't had a chance to figure out what was making it happen
20:51.46SplasPoodanyone have the lddefault.cfg for a Cisco 7905?
20:51.52CunningPikecbullock81: Let me verify
20:51.57CunningPikecbullock81: brb
20:52.24cbullock81k
20:53.30CunningPikecbullock81: That's it
20:53.38CunningPikecbullock81: Mantis time
20:55.52*** join/#asterisk coil (i=coil@isafailure.com)
20:56.45cbullock81CunningPike: Mantis time?
20:57.29CunningPikecbullock81: Sounds like a bug in Transfer() - I'll play around a little more and see
20:57.31*** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com)
20:57.50cbullock81CunningPike: cool. could you email me if you figure anything out?  cbullock@columbiacomputers.com
20:58.12mercestesCan I email you too?
20:58.26cbullock81heh. sure. send me all your goat porn :P
20:59.04mercestescan I send you adds for viagra and cialis too?
20:59.10mercestesand stock tips?  I have lots of stock tips.
20:59.20cbullock81haha. PLEASE DO! i click on every stock tip i get
20:59.26mercestes:D  yay.
20:59.30cbullock81takes 2hrs out of my day, but dang i am rich now
20:59.40*** join/#asterisk ManxPower (n=manxpowe@68.113.119.198)
20:59.57mercestesoh....and I have this bank account in Zimbabwe that has 1 million in uncollected money from thsi dead man, and I need a US bank account to deposit it in so I cam embezzle it.  I'll give yo ua cut if you just give me your bank account info
21:00.26rpmmercestes: you are hooked up with that deal too?
21:00.36CunningPikemercestes: Tell you what, I'll do that for you if you can help me get my Bulgarian Lottery winnings
21:00.45mercestesCunningPike:  Sweet!
21:00.46mercestes:D
21:00.48zoahmm
21:00.49CunningPike:D
21:00.51zoai am in bulgaira
21:00.52ManxPowerJust send me the account number of the uncollected account and I'll pull it out for you.
21:01.07mercestescbullock81:  Probably shouldn't post email addys to IRC in archived (and google searchable) channels.  spambots and all.
21:01.12SweeperI want to buy your car for twice your asking price, only can I pay with cashier check, and I will include the shipping and the check, and please forward the shipping to the shipping company I specify
21:01.35*** join/#asterisk IronMan2000 (n=kent@midsouth.com)
21:01.47cbullock81IRC is google searchable??
21:02.06ManxPowercbullock81: Various bots put the logs on the web
21:02.18mercestes:D
21:02.25mercestesyou weren't....*fond* of that email address were you?
21:02.33cbullock81wow! thats cool (kinda). I havent used IRC in years (until the last few days)
21:02.43rpmhttp://pastebin.ca/320846, md5secret= seems to take precendence over username and secret, i can receive incoming calls when my md5 secret is set but not dial out, when it is unset i can dial out but not recieve calls..
21:02.55cbullock81lol... I think that address has a direct tube to all spammers in the world
21:02.57rpmthe md5secret is a hash of username:realm:password
21:03.05IronMan2000Does anyone know if you can use a dual digium card, and split your PRI. Use 1/2 for VoIP and pass the other 12 channels over to a dial-up server like a Ascend Max or something.
21:03.10ManxPowerrpm: then break up the entry into type=user and type=peer
21:03.37ManxPowerIronMan2000: In theory yes.  See "show application zapras"
21:03.37cbullock81anyone ever get messages like this on the console: Got SIP response 500 "Internal Server Error"
21:03.46ManxPowerBut very few people do that so there won't be much help.
21:03.47mercestescbullock81:  On a polycom phone?
21:03.56ManxPowercbullock81: that is common with polycom phones and is harmless
21:04.05mercestesis it harmless?
21:04.07IronMan2000yea, I was told it will work, but I am not fidning anything on it... Figures..
21:04.08cbullock81ah. thanks
21:04.09IronMan2000Thanks.
21:04.12mercestesit doesn't...stop them from ringing?
21:04.30cbullock81they work fine... except that buddy status problem i mentioned earlier
21:04.33ManxPowerIronMan2000: What we do is put a CSU/DSU between the T-1 and Asterisk and send the PRI channels to Asterisk and the data channels to our Cisco router.
21:04.53ManxPowermercestes: It doesn't stop any of our 80+ phones from ringing.
21:04.59SweeperI weep for someone who ahs to share a t1 between internet and voice ;_;
21:05.00rene-i have the bank account info of some russian scammer posing as a beautiful girl that wants to move with me to america
21:05.23mercestesrene-:  got pics?
21:05.27rene-yes
21:05.31rene-tons of it
21:05.31mercestesI wanna see.
21:05.41mercestessend them to cbullock@columbiacomputers.com please.
21:05.54ManxPowerNo!  Send them to postmaster@localhost
21:06.44rene-http://www.anti-scam.org/scammer/1777-10.html
21:07.34cbullock81hahah! you're wrong
21:07.52mercestesnice
21:07.55tzafrir_laptoppenguinFunk, try "listening" with ztmonitor insead of that separate phone
21:08.25mercestescbullock81:  ;)  S'what private messages are for.
21:08.36ManxPower10thdedabdbi@ncifcrf.gov 1stdedabdbi@ncifcrf.gov 2nddedabdbi@ncifcrf.gov
21:08.41mercestesOMG...yes..I can reregister *ALL* my "free" software yet again.  Good..I had a new PC I'd been trying to get that stuff on.
21:08.49ManxPowerthose are s p a m  t r a p boxes
21:09.12*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)
21:09.42*** join/#asterisk Gankhuu (n=gankhuu@72-166-51-162.dia.static.qwest.net)
21:10.09rene-it would be cool to give the nigerians the bank account of the russians
21:10.30cbullock81heh... it did cross my mind, but i didnt realize there were bots posting this crap on the net... learn something new daily
21:11.03mercestesbots?  hells...I copy this stuff down and use it for registrations.
21:11.05mercestesI give it to telemarketers
21:11.18ManxPowermercestes: you're a bot and you know it!
21:11.28ManxPowerYou are jbot's alter ego!
21:11.34mercestes...
21:11.37mercestesmore..."evil twin."
21:11.47rene-talk like a bot, walk like a bot
21:11.51mercestesyou go ~question and he gives you useful information and I insult your mom.
21:12.13cbullock81on a serious note :)... where can I get some tips on setting up auto attendant. anyone know any good resources. the wiki hasn't really come through for me
21:12.26*** join/#asterisk klictel (n=klictel@207.107.208.137)
21:12.27mercestescbullock81:  google asterisk IVR
21:12.48cbullock81k. thx
21:12.51IronMan2000Manx, I had someone tell me I could plug my T1 into the first port on my digium card, and then use the second port to plug into out Max 6000 dial-up switch. We had our T1 carrier break our T1 into two sperate banks, using channels 1-12 for voip, and then a new set of DID #'s on channels 13-24. Ever heard of this?
21:12.53mercestescbullock81:  If that doesn't work google asterisk consultants
21:12.59mercestescbullock81:  and your mom wears combat boots
21:13.32mercestesI've never heard of a 24 line pri.
21:13.47Hmmhesaysheh what?
21:13.53cbullock81looks like i've stirred up the insult-bot
21:13.54mercestesis this a data T1 for SIP?
21:13.56mercestesor a PRI?
21:14.33GankhuuI am using asterisk 1.4, I reloaded asterisk to see if it was the culprit for a downed iax trunk and now get "Asterisk died with code "Asterisk died with code 1 Automatically restarting Asterisk" over and over again.
21:14.38IronMan2000Channelized T1, same as a PRI except 24 instead of 23 channels. A pri used one of the B channels for sync.
21:14.43ManxPower~mercestes
21:14.45jboti heard mercestes is is the almighty dark overlord.  Worship him!  Worship or lament and suffer!  All hail Mercestes!  Dark lord of existance.  Mercestes is also my Evil Twin!
21:15.00GankhuuI made no changes to configs and was working properly before. Any ideas?
21:15.15mercestesIronMan2000:  Technically...yes, you could set a "g2" to go to the ..thing your plugging into as long as it can accept incoming "dial out" connections in whatever data mode yoru connecting to it with.
21:15.34HmmhesaysGankhuu: how are you starting asterisk?
21:15.45Gankhuuservice asterisk start
21:15.55IronMan2000hmmm, just can't seem to get it to work..
21:15.59Hmmhesaystry starting it manually
21:16.10GankhuuI also tried just asterisk -vvvvvvvvvvvgc
21:16.16Gankhuubut it ends
21:16.16mercestesIronMan2000:  Yea there are about a 1000 variables to worry about there.
21:16.33mercestesGankhuu:  *First*   do a killall -9 safe_asterisk && killall -9 asterisk
21:16.45Gankhuudid that
21:16.49mercestesGankhuu:  that's what's spamming you with "asterisk died, restarting.." blah crap.
21:16.56Gankhuuno asterisk processes right now
21:16.58mercestesthen to an asterisk -ccccccccccccccccccccccccccccccccccccccccccccccccccccccccv.
21:17.06mercesteswith exactly that number of c's.
21:17.14ManxPowermercestes: be n ice.
21:17.26mercestesit should pop out an error as to why * can't start.
21:17.33ManxPoweryou would usually want "asterisk -cvvv"
21:17.45mercestes>.>  ok, what Manxpower said..;)
21:17.48*** part/#asterisk klictel (n=klictel@207.107.208.137)
21:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
21:18.02*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
21:18.22*** join/#asterisk klictel (n=klictel@207.107.208.137)
21:18.30Gankhuulooking for errors...
21:18.34mercestesdid it die?
21:19.05mercestesif it ran successfully it's a different issue...if it died....then .....we're on the right track
21:20.46Gankhuudied
21:20.54[hC]So, is the internet telephony expo any good?
21:20.59*** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net)
21:21.21Gankhuuoh, new error.... chan_oss.c:1867 load_module: Unable to register channel class MGCP
21:21.31GankhuuDon't remember ever doing anything with MGCP
21:21.44ManxPowerGankhuu: that should be a harmless error
21:21.58Gankhuuthat is only error
21:22.03ManxPowerjust put a noload => chan_mgcp.so  /etc/asterisk/modules.conf
21:22.13Gankhuucouple warnings and a few notices, but those are normal for my system
21:22.15ManxPowerthen do asterisk -cvvv again and see what the real error is
21:22.20*** join/#asterisk manolo01 (n=josemanu@ppp-70-129-133-213.dsl.rcsntx.swbell.net)
21:23.36manolo01is there a good guide out there to start learning about the CLI commands?
21:23.48[hC]manolo01: http://www.voip-info.org
21:23.51mercestesmanolo01:  show applications.
21:23.54ManxPowermanolo01: "help"
21:23.58mercesteslol
21:23.59ManxPower"show applications"
21:24.55manolo01thanks!
21:25.14mercestes:)  NP.  Go well asterisk warrior
21:25.53Gankhuuok, so only warning I don't recognize now is channel.c:435 ast_channel_register: Already have a handler for type 'Console'
21:26.03Gankhuumaybe pid still in system?
21:26.15ManxPowerGankhuu: so, does asterisk exit?
21:26.22Gankhuuyes
21:26.39mercestesps -aux |grep -i asterisk     ?
21:26.42ManxPowerGankhuu: do you have any load => lines in /etc/asterisk/modules.conf?
21:26.54*** part/#asterisk l2cache (n=ghansen@64.128.254.98)
21:27.00Gankhuuyes, let me look them up...
21:27.01ManxPowerdo you use the Console channel?
21:27.10ManxPowerIf not remove oss.conf and alsa.conf
21:27.43*** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br)
21:27.54*** join/#asterisk inspired (n=mikael@62.141.128.222)
21:28.00Gankhuuthat did it...
21:28.12ManxPower<-- smarter than he looks
21:28.15Gankhuuwhy do those channels kill asterisk
21:28.26ManxPower<-- poorer than he looks.  send your Paypal thank you to eric@fnords.org
21:28.32mercestesWhy is oss and alsa referred to as "console."  =/
21:28.40GankhuuTrying to configure paging and need one of them
21:28.46rpmomg, this is so frustrating.. i have never had this hard of a time getting a sip trunk working, it makes broadworks broadsoft look nice.
21:28.48Gankhuuat least that is what I read
21:28.51mercestesnevermind....I'll just commit it to memory for future use.
21:28.57ManxPowerThe more zeros you have at the end of your thankyou, the better your chances are of avoiding going to hell.
21:29.07GankhuuI will keep that in mind
21:31.02ManxPowerrpm: there is no such thing as "sip trunking"  You might as well be saying "I never has this much trouble riding a unicorn."
21:31.26mercestesriding unicorns is hard.
21:31.46ManxPowerrpm: It is generally a good idea to only use type=friend for PHONES.  For gateways and providers you should split the configs into type=peer and type=user
21:32.15rpmi think its my provider who has something messed up or gave me the wrong information for authentication
21:32.18ManxPower53 mins until I start my drive by bidding war.
21:32.33bkruseManxPower: actually.......in the GUI, its going to make a user and a peer for each phone
21:32.37ManxPowerrpm: very, very few providers will work with type=friend
21:32.39bkrusewe are fading from type=friend
21:32.42mercestesrpm:  By "generally a good idea" he really means, "This is how it usually works."
21:32.52mercestesrpm:  Damn, he beat me to it..;)
21:33.06rpmim using type=user and type=peer again now, i tried type=friend this morning and it failed also
21:33.37ManxPowerbkruse: does anyone actually thing Asterisk-GUI is not totally worthless?  I can't actually see ANY usefulness of it.
21:33.50Gankhuuso what is the most verbose mode in reality? I like to watch my systems and really learn about them from the messages
21:33.59bkruseManxPower: uh.........its awesome
21:34.12*** part/#asterisk clona (n=clona@bjs2-dhcp111.studby.uio.no)
21:34.18mercestesManxPower:  except by diluting any real talent our community has by a writhing hoarde of retards with a GUI.
21:34.30mercestesit's useful for that.
21:34.36GankhuuI like CLI for most everything...
21:34.49bkruseGankhuu: thats because thats what you grew up with, its what you know
21:34.56Gankhuubetter understanding needed, better execution, more powerful
21:35.10Gankhuunot limited some programmer's imagination
21:35.45Gankhuureally gets more intimate with system
21:35.51Gankhuuand I LIKE that. LOL
21:36.01Gankhuucan you tell I don't get out much?
21:36.06mercestesGankhuu:  Cli makes me look smart.
21:36.27ManxPowerbkruse: What exactly does the GUI do as it currently exists?
21:36.38nays85Gankhuu : have you heard of the new version of the cli?
21:36.45mercestesreally, I fail to see why IT developers continually make the technology easier for the average joe...and then complain when the demand for IT professionals plummets.
21:36.50data23jeez, i can't make head nor tail of these bristuff patches ;{
21:36.57GankhuuI know that the 'service asterisk start' has verbose level 3 by default but what is the max level?
21:37.09bkruseManxPower: if you havent tried, i seriously feel sorry for you.
21:37.19Gankhuuor makes it easier for some dummy to screw up you mean...
21:37.21bkrusei am not trying to flame, but for people just learning to use asterisk, its very appealing
21:37.30mercestesGankhuu:  I keep my verbosity set to around 99.....I know another guy who had 3,047.  but, honestly, the source code doesn't check for anythign above 3.
21:37.41nays85Gankhuu : you should install CLIt, it's much more intimate than the regular CLI
21:37.43ManxPowerbkruse: that does not answer my question.
21:38.08bkruseManxPower: if youve never used it, try for yourself, I am not your slave
21:38.13bkruseit has MANY different features.
21:38.20Gankhuuanyone know a really good windows IRC client. this one demo is about to  expire on me
21:38.21ManxPowermercestes: But the demand for people that can figure out the gui goes up.
21:38.31bkruseGankhuu: MIRC?
21:38.36bkruseyou can still use it when it expires.
21:39.14data23does anyone use the misdn or capi drivers?
21:39.57mercestesManxPower:  Yea,..like the demand for ppl who can figure out windows.  =/  like my grandma.  My six year old, however, teaches me daily.
21:40.06GankhuuMIRC?
21:40.28ManxPowerbkruse: perhaps you could point me to a page that describes it.
21:40.32GankhuuI am using X-chat on windows now... how do i keep using after demo expires?
21:40.53ManxPowerGankhuu: you pay the few dollars to register it, ya cheapskate
21:41.14GankhuuLOL
21:41.17Gankhuuyou are right
21:41.34bkruseManxPower: The topic for #asterisk-gui is: Asterisk finally has a (very alpha) GUI! || http://svn.digium.com/view/asterisk-gui/trunk/ || Screenshots: http://asterisknow.org/images/gui || AsteriskNOW is at #asterisknow and http://asteriskNOW.org
21:42.04GankhuuI don't like it as it is now written
21:42.16Gankhuuprefer CLI
21:42.21Gankhuuwhat is CLIt anyway?
21:42.35mercestes....
21:42.44mercesteswell....it's that small piece...right above......
21:42.58mercestesCan I msg you Gankhuu?  This might get a little graphic.
21:43.05Gankhuubring it on...
21:43.56ManxPowerbkruse: no offence, but a couple of screen shots and a "quickstart guide" that tells you how to make a CD from the ISO is not exactly a features list.
21:44.18Gankhuu<-- agrees with ManxPower
21:44.41ManxPowerBut THANK YOU for creating a separate IRC channel for it.
21:44.49Gankhuu!
21:45.00*** join/#asterisk groogs[h] (n=chatzill@cbl-66-102-80-229.wtccommunications.ca)
21:46.30*** join/#asterisk X-Rob_ (n=Rob@dsl-202-173-151-24.qld.westnet.com.au)
21:47.15kirberichdoes anyone here have some basic experience with festival and asterisk? when i use it though the festival function the speech sounds really weird and the first word is omitted
21:47.33kirberichand for some reason does not seem to have text2wav in the portage tree
21:50.22*** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com)
21:50.26zapp-braniganhi,
21:51.13zapp-branigani have compiled the speex codec in 1.4 but when i load the module loader.c:362 load_dynamic_module: Error loading module 'codec_speex.so': /usr/lib/asterisk/modules/codec_speex.so: undefined symbol: speex_nb_mode
21:51.33rpmkirberich: cepstral works much better than festival.
21:51.33zapp-branigan:?
21:51.34zapp-braniganwhat is the problem ?
21:51.36rpmit is expensive though.
21:52.03kirberichexpensive is bad ;)
21:53.10zapp-branigansomebody know what is the problem in the speex ?
21:56.10*** join/#asterisk diclophis-work (n=jbardin@adsl-69-237-115-101.dsl.scrm01.pacbell.net)
21:56.13mikefooWhat do I need to have in place for gathering incoming call numbers even if they block the call? I am in the US. basically need to gather a number even tho someone uses *67
21:56.14diclophis-workwhat does this mean ? "Unable to open format wav"
21:56.49ManxPowerdiclophis-work: usually means "can't open file, no permission or file does not exist"
21:56.58diclophis-workhmm
21:57.28ManxPowerMaybe you did a Playback(/path/to/file.wav)
21:57.37ManxPoweryou NEVER specify an extension for playback.
21:57.43sevardin which you shouldn't have included the extension
21:58.01rene-hi, can somebody send me a sip call to 123456@200.34.66.132 codec g729?
21:58.07diclophis-work"RIFF (little-endian) data, WAVE audio, mono 8000 Hz" is an OK file type right?
21:58.20sevardyasureyabetcha
21:58.30diclophis-workoh
21:58.32diclophis-workmaybe...
21:59.35diclophis-worktheres also this: "Not a wav file 3"
21:59.43diclophis-worki am not specifiying the extension...
22:01.15ManxPowerpaste the cli output of the line
22:03.08diclophis-workits through an AGI script
22:03.09diclophis-workSTREAM FILE /comp/lib/sounds/greetagent2 "1234567890*#" 0
22:03.28CunningPikecbullock81: http://bugs.digium.com/view.php?id=8848
22:03.29*** join/#asterisk anthonyl (i=Anthony@nat/digium/x-9dc1585c5ece0b30)
22:03.43diclophis-workand this is the output from "file" /comp/lib/sounds/greetagent2.wav: RIFF (little-endian) data, WAVE audio, mono 8000 Hz
22:04.50diclophis-workoh.. maybe i dont have the wav codec loaded...
22:06.13diclophis-worknope, format_wav is loaded
22:07.22*** join/#asterisk anthonyl (i=Anthony@nat/digium/x-16ff1789fd42a967)
22:09.16diclophis-workhow do i convert wav to gsm ?
22:09.32perdsox
22:09.33rene-diclophis: there are several recipes in the wiki
22:10.14rene-can somebody send me a test call to 123456@200.34.66.132 using sip/g729 ?
22:11.56rene-please?
22:12.03*** join/#asterisk crich1999 (n=crich@port-212-202-210-130.dynamic.qsc.de)
22:13.28*** join/#asterisk saftsack (n=saftsack@pD9E04F07.dip.t-dialin.net)
22:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
22:18.03*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
22:18.29*** join/#asterisk crshman (n=asdf@FTS-27-146.resnet.ucsb.edu)
22:18.44crshmanhello all
22:19.40b11d|bblhey
22:20.35crshmani have a question about asterisk....i have 2 boxes box 1 has an extension set up 10000 and box two has a trunk that connects to the extension, calls can be made and recieved and all is good
22:21.04crshmanhowever, when i call out from box 2, the caller id isn't set the outbound caller id shows up as 10000 and not the number i set up on the trunk
22:21.14*** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir)
22:21.45crshmanis there a better way to connect the two boxes, am i going about that wrong? or where do i start to fix this?
22:22.21mercestes=/
22:22.31mercestesHow are you connecting box 2 to box 1?
22:22.46mercestesSip?  Iax?  Zap?  two cans and taut string?
22:23.04crshmansip
22:23.14Assidi want trxtel!
22:23.16mercestesThere is no such thing as a sip trunk.
22:23.23Assidi wonder when they will go back live
22:23.23crshmani set up a trunk on box 2 that connects to extension 10000 on box 1
22:23.26JTdata23: ?
22:23.28mercestesyou have a connection.
22:23.32mercestesvia sip
22:23.36perdi love two big cans with only a string between them
22:24.12JTdata23: doh, was scrolled up, but the bristuff patches are easy
22:24.12mercestesSo....how do you have this connection "connected" to extension 10000?
22:24.12crshmanlol ok technically yes...sorry i'm using freepbx to configure so i'm tainted by the wording but technically yes it's just a connection
22:24.13mercestes....
22:24.13perdhaha
22:24.13perdoh no, now you've done it
22:24.13crshmani'm talking in both
22:24.13mercestes~trixbox
22:24.15jbotmethinks trixbox is unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/
22:24.16mercestes~freepbx
22:24.18jbotfreepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
22:24.29crshmanbut i understand the underlying concepts of the asterisk config files....
22:24.40mercestescrshman:  that's a great start to using Freepbx.
22:24.41mercestes:)
22:24.43crshmani have done it with .conf and with freepbx...so bear with me lol
22:24.46mercestesmaybe someday...you'll use asterisk.
22:25.02perdtwist it!
22:25.07mercestesyou shouldn't cross post either, btw.
22:25.26kirberichperd, is that string-phone protocoll implemented in asterisk yet?
22:25.36*** join/#asterisk _DAW (n=chatzill@adsl-222-55-112.msy.bellsouth.net)
22:25.50perdit's in development, much like chan_skinny. yell at qwell, he's lazy
22:25.50*** join/#asterisk topping (n=topping@207.47.6.185.static.nextweb.net)
22:25.52crshmanthey are different channels lol....how is that considered cross posting? =P well anyways.....the sip "connection" from box2 to box1 doesn't pass the callerid info....
22:26.06Assiderr. any one here using voicepulse
22:26.09mercestescrshman:  ....how do you crosspost in *one* channel?
22:26.11crshmanbox2 registers just fine with box1, but the callerid doesn't get passed
22:26.14Assidrecently having shitty connection
22:26.20_DAWDoes someone here know, how many records can the asterisk database hold before problems?
22:26.50CunningPike_DAW: Fewer than yours, I'm guessing ;)
22:26.55mercesteslmao
22:27.22*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
22:27.25mercestes_DAW:  Depends on which database you use, how you setup the configs, your HDD space, your file system, the alignment of the planets and the moon, your karma, your ancestors karma.
22:27.36_DAWHavent tried yet, I was thinking less than 2000.
22:27.41diclophis-workman this is hopeless
22:27.41mercestes_DAW:  so, exacty 4,987,345,207 records.
22:27.52_DAWIn the asterisk database
22:27.56mercestesoh
22:28.02mercestesin the *asterisk* database
22:28.05perdstrange, that's the exact send i was born, mercestes.  epoc time
22:28.15mercestessame list, minus configs and database type
22:28.35CunningPikediclophis-work: What is?
22:28.42diclophis-workwav files
22:28.48diclophis-worki got them in 32bit float
22:28.57diclophis-workformat, sox can convert from that
22:29.03diclophis-workand audacity is crapping out on me
22:29.05CunningPikediclophis-work: What are you trying to do?
22:29.11diclophis-workwhen i export from that it loses the audio
22:31.18data23JT: you got 2 secs?
22:32.22*** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net)
22:33.05perdyou used up your two seconds by asking that question
22:33.06crshmandoes extension callerid override outbound connection cid or vice-versa?
22:33.07perdNEXT!
22:33.18*** join/#asterisk Vec (n=Vector@dsl-244-211-61.telkomadsl.co.za)
22:34.04data23I guess i'm just a little confused on the whole, bristuff, misdn, capi vs Native Zaptel setup
22:34.12diclophis-workcan asterisk spport 44kz ?
22:34.20diclophis-worker well sox resample should fix that no?
22:34.50VecAnyone know, or can u point me in the direction of some documentation on configuring asterisk, to only allow people access to certain phones by entering a pin, and adding that pin to the call records for biling? I have a few ideas on how to do it, just would be nice to read something on it.
22:35.11data23JT said ineed the bristuff patches to get Call Deflection working, but as far as i read, it's only for BRI's using the CAPI interface, a E100P card won't be supported? or am i getting the wrong end of the stick
22:36.25crshmancan i connect to a running asterisk process and view the debug in color or do i have to start the asterisk process to have color debugging?
22:36.46bkrusecrshman: that is such a good question, i still cannot do it
22:36.53perdvec would be pretty easy.. exten => _*555*NXXNXXX,1,Set(CODE=${EXTEN:X:X}) or something along those lines
22:36.55*** join/#asterisk flenders (n=fserto@unaffiliated/flenders)
22:36.59flendersmorning
22:37.20perderr_*XXX*NXXNXXX
22:37.32perdthen check your C${CODE} against a db
22:37.36crshmanbkruse: no sarcasm intended? or are you just prodding fun because i "don't use asterisk" ?
22:37.44*** join/#asterisk N9URK_lappy (n=icechat5@cpe-075-178-088-168.ec.res.rr.com)
22:37.45perdyour CDR should show the *XXX* which you can use for billing
22:38.08perdor you can just do an insert to mysql ro something
22:38.14*** join/#asterisk A500mg (n=x@86.205.139.254)
22:38.20A500mghello :)
22:38.26Vecperd : thanks for the advice
22:38.51N9URK_lappyHi,  I am having a problem with *.  It was working fine.  Now when we rebooted the server I went to start with "asterisk" then issued "asterisk -r" then got this error "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)"
22:38.53Vecperd : I gues i'll work on it, and if I have issues, I'll come ask. Thanks
22:38.54N9URK_lappyCan anyone help?
22:39.00perdnp
22:39.21A500mglappy: ps -aux
22:39.27A500mgasterisk is present in the list ?
22:39.34perddf -h, did var mount? :P
22:39.36Vecn9urk : I make sure asterisk is not running, if it is delete that file
22:39.45A500mg(sorry for my english, it's not my own language)
22:39.58*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)
22:39.59*** mode/#asterisk [+o mog] by ChanServ
22:40.06N9URK_lappyA500mg N9URK_lappy It's not in the list
22:40.15N9URK_lappyVec its not running
22:40.27data23N9URK_lappy: try asterisk -vvvgc and see if it loads
22:40.57N9URK_lappyI got some errors at least
22:41.10VecA500mg : Do u speak french ?
22:41.16A500mgben oui :)
22:41.23b11d|bblAlors!
22:41.26N9URK_lappyno sprechen Francois
22:41.33N9URK_lappy;)
22:41.36A500mgsi si on va causer francais
22:41.41A500mgcomme ca personne va rien capter
22:41.43b11d|bblwass est los?
22:41.44A500mg:D
22:41.49flendersWTF?
22:42.01A500mgnothing :)
22:42.02mercestesbien parfait?
22:42.06Vecn9urk : check if that file exists in the error msg, if it does cat it
22:42.13VecI don't speak french ahhh :)
22:42.18*** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com)
22:42.22N9URK_lappyYo hablo ingles solamente
22:42.39data23do you speak *? :)
22:42.43JTdata23: that's wrong, you can use any zap interface with bristuff
22:42.47rene-123456@200.34.66.132 for hot sex (sip/g729) just kidding
22:42.51JTit adds more than bri support
22:42.52rene-but please call
22:43.00N9URK_lappyHere are the errors I get:
22:43.00N9URK_lappyJan 18 17:42:33 ERROR[4262]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server  on  (err 2002). Check debug for more info.
22:43.00N9URK_lappyJan 18 17:42:33 ERROR[4262]: chan_zap.c:10323 setup_zap: Unable to load config zapata.conf
22:43.00N9URK_lappyJan 18 17:42:33 WARNING[4262]: loader.c:414 __load_resource: chan_zap.so: load_module failed, returning -1
22:43.00N9URK_lappyJan 18 17:42:33 WARNING[4262]: loader.c:554 load_modules: Loading module chan_zap.so failed!
22:43.04N9URK_lappyJan 18 17:42:33 ERROR[4262]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server  on  (err 2002). Check debug for more info.
22:43.25bkruseN9URK_lappy: ~pb
22:43.26A500mgwhy realtime is active ??
22:43.42bkruse~pb
22:43.52jboti heard pb is a place to paste your stuff without flooding the channel - try  http://pastebin.ca, or http://channels.debian.net/paste
22:43.52N9URK_lappysorry, I only meant to past the first one
22:44.08data23JT: do i need the experimental .3?
22:44.48A500mgres_mysql.conf ?
22:44.51A500mgextconfig.conf ?
22:44.51CunningPikecrshman: Colored CLI output depends on two things - passing '-c' when asterisk is started, and specify a CONSOLE=
22:44.52JTprobably
22:45.02JTthe latest experimental i have found to be good
22:45.10A500mgi don't know how to disable realtime
22:45.19A500mgvoip-info.org will help you
22:45.20JTthe old "stable" versions use asterisk 1.0.x
22:45.29data23k
22:45.34AursCunningPike: any way to get colored cli if asterisk starts from a shell script? (when asterisk crashes)
22:46.02N9URK_lappyI think I know my problem.
22:46.08CunningPikeAurs: Same as above, afaik
22:46.21N9URK_lappyThanks for the help!
22:46.36VecDoes asterisk inherintly support accessing a mysql database ?
22:46.44*** join/#asterisk terrapen_ (n=cjs@208.64.89.90.utahbroadband.com)
22:47.01A500mg:)
22:47.01CunningPikeAurs: We use RHEL /sbin/service to start asterisk with safe_asterisk, but pass in -c, and have CONSOLE=tty9 in the appropriate file
22:47.07mercestesVec:  Pretty much but the planets have to align just right.
22:47.15*** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net)
22:47.22Vecmercestes : how do I align the planets :)
22:47.27AursCunningPike: I use the /etc/init.d/asterisk script
22:47.29mercestesVec:  magic.
22:47.31perdi jsut sacrifice a goat before i fire up asterisk. seems to work well
22:47.44CunningPikeAurs: Same deal then - make sure to use the -c option
22:47.47Vecmercestes : Merlin is unavailible at present please try again later
22:47.47JTdata23: you've found all the scripts in the bristuff tarball yeah?
22:47.55Aursbut if that one is called from cron, we get no colors in cli.
22:47.59mercestesVec:  or by having mysql installed ,and * installed, and the appropriate config files (res_mysql.odbc or res_mysql.conf, extconfig.conf, etc.) and the proper username and passwords ,and permissions.
22:48.12Aursso it's probably the CONSOLE= variable I need?
22:48.13CunningPikeAurs: From cron??????
22:48.16Vecmercestes : that does not sound too bad
22:48.26AursCunningPike: yes, if/when asterisk crashes
22:48.38mercestesVec:  Yea, it's mostly just setting the config files correctly.  googled asterisk realtime and look for voip-info matches
22:48.41Aurswe run a php script in cron that checks if asterisk is running
22:48.52Vecmercestes : thanks!
22:48.56mercestesVec:  I use the res_mysql.conf frmo asterisk-addons to bypass the odbc crap
22:49.07CunningPikeAurs: I see - it really shouldn't be crashing that much...........
22:49.17VecDaemontools might work nicely to keep asterisk running ?
22:49.21A500mgi've a little problem: i use a tdm01b, it works fine, but.. at the beginning of the call, i hear my voice, and after 5-6s it's good. I've try echotraining=yes, it solve the problem partially (only for one direction), any idea ?
22:49.26CunningPikeAurs: Maybe fix that, and then worry about getting colors in your CLI :)
22:49.48AursCunningPike: have fixed it actually. so we dont ahve as many crashes anymor
22:49.48Vecmercestes : thanks
22:49.48perdfunroll-loops.
22:49.53Aursbut still. it might crash
22:50.00CunningPikeAurs: Nah :)
22:50.16Aurson the other hand... it is a good way to see: "aha! no colors.. must have had a crash"
22:50.17Aurs:P
22:50.33perdyeah or you could monitor the system
22:50.37perdlike normal people
22:50.54perdnagios is your friend.
22:51.25Aursour little php script checks service, restarts if down, sends email if down
22:51.59Aursbut nevermind
22:52.01data23JT: i've patched my libpri, recompiled and installed, done the same to zaptel, just recompiling * atm :)
22:52.20JTerr
22:52.28JTthere are scripts that do it all
22:52.43JTno need to do each manually
22:52.59data23ah well, i like patching :)
22:53.27perdno doubt, my favorite part is the -p0 <
22:53.28crshmani have asterisk set up like this: extension on box2 --> outbound sip connection on box2 --> extension on box1 --> out to pstn
22:53.28crshmandoes the callerid from the outbound sip connection
22:53.32crshmanoops...
22:53.49crshmandoes the callerid from the outbound sip connection get overwrited by the extension on box1?
22:54.02data23wheres old jerjer these days anyway?
22:54.32N9URK_lappyhow can I unload an addon without * starting up?  I think I found my problem
22:54.43ManxPowerdata23: You got married?  I'm sorry to hear that.
22:54.45N9URK_lappyand I need to unload the mysql cdr addon
22:54.50mercestesN9URK_lappy:  I think unload would work.
22:55.00N9URK_lappyumm, how do I do that?
22:55.02data23ManxPower: ta
22:55.05mercestes....
22:55.06mercestesumm.
22:55.09ManxPowern9urk: in the cli: unload themodulefilename
22:55.12mercestesunload app_cdrmysql
22:55.17ManxPower"Show modules" will list that.
22:55.22ManxPowermer. you forgot the .so
22:55.27mercestesyea, I know.
22:55.31data23yay, asterisk reloaded, pri is back active :)
22:55.32mercestesthanks, Manx.. :)
22:55.33N9URK_lappyhow do I do that sine * isn't starting up?
22:55.38N9URK_lappysince not sine
22:55.40mercestes...
22:55.47mercestesohhh.
22:55.51ManxPowern9urk: in /etc/asterisk/modules.conf put noload => thefilename
22:55.55mercestesput noload => modulename  in modules.conf
22:56.01N9URK_lappycool thanks
22:56.03mercestesdamnit ManxPower..:P  you type too fast
22:57.13*** join/#asterisk rc-1 (n=rc-1@ip68-229-102-1.hr.hr.cox.net)
22:58.11justdavehmm, are there any utilities for Asterisk to have an extension number be a Skype presence or something so that people could request a chat with that Skype ID and connect to your system?
22:58.27*** part/#asterisk A500mg (n=x@86.205.139.254)
22:58.46justdavemaybe not for Skype, but anything else similar like that (short of having to set up softphone accounts and using a softphone)
22:58.53ManxPowerjustdave: not really.  There are a couple of hacks
22:58.54JTdata23: ./install :)
22:59.04JTor ./compile
22:59.12ManxPowerjustdave: Asterisk is a PBX, not a chat server 8-)
22:59.28justdavewell, meetme has its uses. :)
22:59.29Qwell[]ManxPower: res_ircd
22:59.41mikefooWhat do I need to have in place for gathering incoming call numbers even if they block the call? I am in the US. basically need to gather a number even tho someone uses *67
22:59.43data23JT: uhuh, asterisk is loaded, do i need the capiCD module? that failed to install due to no capi drivers?
22:59.46justdaveI am talking audio chat by the way.
22:59.53monstedManxPower: any hacks that don't require a windows pc and usb FXS box?
22:59.53JTno
22:59.57rene-can some one please call me at 123456@200.34.66.132 sip/g729?
23:00.02Qwell[]monsted: not really, no
23:00.07monstedpoo
23:00.08JTyou can use zap to access channels
23:00.16data23uhuh
23:00.21JTyou have bristuff-0.3.0-PRE-1w
23:00.25JT?
23:00.29data23yes
23:00.33JTit does support capi for legacy use
23:00.47rene-recommending an echo test server that supports g729 is cool too
23:00.57data23hmm actually, mines bristuff-0.3.0-PRE-1x
23:01.02ManxPowermonsted: I think all the Skype hacks require a windowspc
23:01.07data23:}
23:01.16cbullock81hey. with the directory program is there a way to make it playback the recording of the users name instead of spelling out the name?
23:01.32justdaveOnly thing I can think of that would be easy to do is to publish the account/pass info for a SIP account, and lock that account down so the only thing it can do is use the conference system, then let people use SJPhone and the like to connect to it.
23:01.49data23JT: how do i deflect a call back out a channel tho? I still don't see how doing all this has helped just yet :)
23:02.00*** join/#asterisk backblue (n=moo@87-196-32-185.net.novis.pt)
23:02.00ManxPowermikefoo: The caller must be calling a toll free number in order for you to override thier callerid blocking
23:02.00Qwell[]cbullock81: it tries to playback the "greet" file.
23:02.02Qwell[]cbullock81: so, set your greeting in voicemail
23:02.09JTdata23: damn, moving fast, didn't know x was out already
23:02.29JTdata23: it's not well documented, you may have to read some source notes and CHANGES
23:02.35ManxPowermonsted: Skype is a closed protocol.
23:02.36mercestescbullock81:  "Press 3 to record your name."   That's what your users have to do.  Or what you have to do.
23:02.52cbullock81ah! ok. thanks
23:03.07mercestesnp
23:03.12cbullock81you guys rock :)
23:03.27mercestesIt's true.
23:03.27data23hmm, they're talking about what i'm after in asterisk-dev aren't they?
23:04.10*** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com)
23:07.25*** join/#asterisk sjobeck (n=sjobeck@windsorsolutions.biz)
23:07.49JTi'm not sure it 2BCT is the same as ECT or CD at all
23:08.33JTi think 2BCT may be specific to one or two american switchtypes, too
23:09.18*** join/#asterisk sjobeck (n=sjobeck@windsorsolutions.biz)
23:09.33data23aye i think ya maybe right, seems its for 5ESS switchtypes
23:10.19JTwhat switchtype are you connecting to?
23:10.30data23euroisdn
23:10.31*** join/#asterisk dasenjo (n=dasenjo@190.24.176.69)
23:10.54JTah
23:11.00JTyou will need ECT or CD then
23:12.16dasenjoHi! I'm having a problem with the background command, there are some extensions on the context, all of them have three digits and can be dialed without problem, one hasone digit and can't be dialed, ¿what can I do?
23:12.26JTi can't remember the exact difference
23:12.36JTit'd be nice if was easy to talk to junghanns and ask him
23:12.50JThow to use it, most importantly
23:13.31data23heh, i've found a post from John Todd, dated 2003, saying how bad an idea it was to think about implementing 2B Transfers :)
23:14.34JTdoesn't make sense, if the telco offers it, it's a good idea
23:14.54JTi assume you've checked if the telco offers it
23:14.55data23I think it was meant from a billing point of view :)
23:15.00JToh wait, it was a pabx right
23:15.03data23yep
23:15.04JTah
23:15.14data23Meridian (Read: Nortel) PBX
23:15.22JTyes it would be screwed if you needed to bill
23:15.41data23reminds me of the old IAX Transfer debates :)
23:17.22data23ahha
23:17.36data23finally some definitions
23:17.48data23CD (Call Deflection) is for deflecting calls, whilst they're still ringing and not answered
23:18.01*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
23:18.02*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
23:18.27data23whereas, ECT (Explicit Call Transfer) [otherwise known as TBCT, 2BCT] i guess allows you just to handoff calls
23:18.31*** join/#asterisk nicklinn (n=maximag@mrtc-mm-630052.mis.net)
23:18.32JTyeah this was all in the mail list posts i sent you the URL of
23:18.42nicklinnhey guys
23:18.57*** join/#asterisk bkw_ (n=brian@88-111-165-165.dynamic.dsl.as9105.com)
23:19.05bkrusehey guys, in the manager interface, if i want the status of a sip peer
23:19.13bkrusei know they have the Status: command, but?
23:19.24dasenjoI do not understand the m option for the background command. Does it only work for one digit extensions?
23:19.37*** join/#asterisk droemel (n=droemel@p548E84B0.dip0.t-ipconnect.de)
23:19.57mercestesdasenjo:  Coul dyou maybe pastebin yoru dialplan?
23:20.10dasenjomercestes, of course
23:20.25JTdata23: http://www.voip-info.org/wiki/view/Asterisk+CAPI+readme shows how to use CD and ECT in CAPI... i hope that's not the only place they're implemented :/
23:21.06data23JT: yea i saw that earlier and tbh i'm beginning to think it is :)
23:21.20N9URK_lappyThanks for all of your help.  I got * back going.  You guys are the greatest!  I really appreciate all of the assistance I get from everyone
23:21.30mercestesyour welcome.
23:21.35mercesteswe prefer paypal tho..:D
23:21.44N9URK_lappy(that I understand)
23:22.03N9URK_lappyI may be willing to hire someone to get ztdummy going sometime soon
23:22.35N9URK_lappypm me your email addr
23:22.56data23N9URK_lappy: you using a 2.6 kernel?
23:23.02*** part/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net)
23:23.03data23tis much easier with that
23:23.19N9URK_lappydumb question, how do I tell which kernel I have?
23:23.40data23uname -a
23:23.47rene-bkruse: Action: SIPShowPeer
23:23.47rene-Peer: <peer>
23:23.53N9URK_lappy2.6.15-26-server
23:24.01N9URK_lappyso yeah data23
23:24.05rene-can somebody please ring me at 123456@200.34.66.132 sip/g729
23:24.05dasenjomercestes, the important contexts of the dialplan are at http://pastebin.ca/320967
23:24.38mercestesyea that took two seconds.
23:24.41N9URK_lappyrene, give me a minute
23:24.45rene-cool
23:24.46rene-!
23:24.46mercestes_X. matches 2 or more characters.
23:24.48dasenjoin-8756321 is a zap trunk incoming context
23:25.14mercestesadd a exten => _X,1,   and whatever _X. says to match a single digit and your fixed, dasenjo
23:25.16nicklinnAnyone know if it's normal for my 's' Playback/Wait tree to block incoming keypresses?
23:25.58dasenjomercestes, in the in-* context?
23:26.19mercestesdasenjo:  right below _X.   or right above.  Just for reference.
23:27.16N9URK_lappyhang 1 rene
23:27.23rene-sure
23:27.26N9URK_lappyit didn't let me dial you like I wanted to, one more second please
23:27.36rpmwhats the difference between Dial(SIP/provider/NPA-NXX) than Dial(SIP/NPA-NXX@provider) ?
23:29.25data23right, giving up for the night now (11:30pm), nn folks
23:29.58JTdata23: alright, night
23:34.01N9URK_lappyrene it won't let me ring your number
23:34.30N9URK_lappyrene do I have some screwed up here? "exten => 999, n, Dial(SIP/123456@200.34.66.132, 20)"
23:37.30*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)
23:37.57rene-no
23:38.00rene-it looks well
23:38.02ManxPowerActually _X. means match TWO or more digits
23:38.14ManxPowern9urk: don't put in extra spaces
23:38.52ManxPowern9urk: Also, if you dial by IP address or host name, then only the stuff in sip.conf [general] will be used.
23:39.35dasenjoManxPower, now my dialplan is at http://pastebin.ca/320975, but I still can't dial the 9 extension, ¿can you help me?
23:39.38N9URK_lappythanks, ManxPower,  How is the best way to get it to dial that ip addr?
23:39.44N9URK_lappythanks ManxPower
23:39.53ManxPowerdasenjo: what kind of phone are using?
23:40.16ManxPowern9urk: put an entry in sip.conf with host=200.34.66.132
23:40.26N9URK_lappyok thanks Manx.
23:40.38*** join/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia)
23:40.53N9URK_lappyI get this error in idefisk "18:40:03 Line 1 : ended (bearercapability notavail)"  What does it mean?
23:41.02Burgworkwith 1.2, is there a way to redirect unused extensions to a generic voicemail?
23:41.29ManxPowerBurgwork: exten => _XXX,1,Voicemail(u666)
23:41.35dasenjoManxPower, my analog house phone and an IP phone dialing trough asterisk and a zap trunk
23:41.43ManxPowermake sure it is in the same context as the extensions, not an included context
23:41.45BurgworkManxPower: thanks
23:42.00rene-n9urk_lappy:  do you have g729 installed in your asterisk system?
23:42.07ManxPowerdasenjo: IP phones have their own dial plan, so you must fix it in the phone to allow dialing 9
23:42.13dasenjoManxPower, you can see http://pastebin.ca/320967 too
23:42.19ManxPowerthe call will not be sent to asterisk from an ip phone if the dialplan is messed up
23:43.11N9URK_lappyrene- does g729 have to be installed?  It is not bundled with *?  If not then I don't have it
23:43.39rene-n9urk_lappy: yes it should be installed, it is not bundled with asterisk, but thanks buddy
23:43.53N9URK_lappyrene-, ok thanks, I just read about it on the wiki
23:43.54dasenjoManxPower, but i'm testing with a "plain POTS line" and can't dial 9 either
23:44.09N9URK_lappyrene-I gotcha, sorry I didn't catch that in the beginning.
23:44.16rene-np
23:44.18rene-thx
23:44.24ManxPowerdasenjo: I do not understand.  plain pots line would not allow you to dial 9 from the dialtone
23:44.51N9URK_lappyrene- the wiki says that one can test it.  Is that the case? or is that a "liberal" interpretation?
23:45.49dasenjoI can dial the 9, I got the tone, but the asterisk that answers me, do not dial the 9 extension when I press the key on my phone, asterisk does nothing
23:46.12*** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com)
23:47.26N9URK_lappygood night
23:47.37dasenjoManxPower, ¿do I make me understan?
23:55.41*** join/#asterisk droemel (n=droemel@p548E84B0.dip0.t-ipconnect.de)
23:58.19rene-n9urk_lappy: thereis a free to download and use g729 version but it might be ilegal inyour countru
23:58.20rene-country
23:58.34rene-as g729 is covered by patents
23:58.54arcaninehi

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