IRC log for #asterisk on 20080926

00:00.10adr3nalin3I am having a problem with a TE122P, I am unable to make outgoing calls.  All the channels in my PRI say they are on onhook but when I go to make a call I get Service Unavailable: 961 on the phone.
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00:00.28adr3nalin3Also I am getting --> [Sep 25 18:59:39] WARNING[5195]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)  in the CLI
00:00.41andrewyagerwhat is the output of show zap channels
00:00.47andrewyagerand what is your dial string?
00:01.27adr3nalin3show zap channels shows all 23 channels
00:02.17adr3nalin3exten => _9NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}})
00:02.55adr3nalin3I'm using the gui...Don't tell anybody.  jk ;)
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00:03.22andrewyagerwhat is ${GLOBAL(TRUNK)}
00:04.03adr3nalin3TRUNK = Zap/G2
00:04.40andrewyagerdo you have zaptel groups set up in your zapata.conf file?
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00:06.07adr3nalin3andrewyager: its in users.conf but I have group = 1 ...that could be a problem.
00:07.27andrewyagerChange TRUNK = Zap/G1
00:08.23andrewyageryou are asking it to dial group 2, but you have no group 2 defined
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00:09.49Dovidhi
00:10.04Doviddoes anyone know if anyone ever created a fast agi that runs on php ?
00:10.21andrewyagerthere are plenty of good libraries out there to help you do that
00:10.35andrewyagerhttp://www.google.com/search?client=safari&rls=en-us&q=fastagi+php&ie=UTF-8&oe=UTF-8
00:10.47Dovidthanks
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00:31.24Greg25cany idea how to keep the same extension/IAXy from calling the same conf. rm - it sounds like a machine gun when this happens - for everyone in the conf.?
00:34.18Dovid?
00:34.23Dovidit calls its self ?
00:34.35Dovidand u dont want that to happen ?
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00:36.39andrewyagerHi
00:37.10andrewyagerDoes anyone have any suggestions on how to implement the appearance of multiple queues, but with a single queue?
00:37.45andrewyagerAs in, we have multiple calls running into the same set of operators, and we always want the oldest call answered first, and the position and hold time information to be combined, but the ability to have different hold music and messages on answer
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01:33.36jblackThis place is "nerd party" quiet tonight.
01:36.24Dovidoh yea
01:38.54outtoluncpress aux2
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01:55.12zer0x-hey its alex, with old efnet nick
01:55.22mvanbaakhey
01:55.33mvanbaakI'm there as well
01:56.07Qwellefnet is for newbs^1/nick NotQwell
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01:56.14mvanbaak;)
01:56.22zer0x-it was in 1997/98 =)
01:56.25rvhihi, anyone got a polycom 8002 working?
01:56.26mvanbaakhhmm, stuff is happening
01:56.32mvanbaakI think we are going to have steak
01:56.36jeevn00
01:56.37jeevn00b
01:57.36zer0x-yard?
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02:07.09RB2evening
02:07.22DigitalIronyello
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02:08.30drmessanojblack: It's a funeral
02:10.02jayteewhat's a funeral?
02:10.58RB2hey DigitalIrony
02:11.04DigitalIronyhello
02:11.10RB2How's it goin'?
02:11.16DigitalIronyjust working
02:11.36DigitalIronyand you?
02:12.00RB2same here. Just waiting for some downloads to finish.
02:12.50RB2I need to switch ISPs, this connection is entirely too slow. But, don't have much of a choice to pick from.
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02:16.19fakhir~itsp
02:16.19jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
02:16.32fakhir~itsplist-us
02:16.33jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
02:17.30johnakabean~FU
02:18.40johnakabeanrb2, if you don't mind 300 a month, you can find a good T1 Provider
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02:21.51RB2johnakabean, yeah. I'm just biding my time until FiOS gets to my area. But, I think it's gonna' be quite a few years. I have the choice right now between Comcast or Verizon DSL. :-/
02:22.06Dovidboth crap ;)
02:22.27DovidFIOS is supposed to have low latency
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02:22.46RB2Pretty much. Because of distance, Verizon won't provision me at any faster than 1.5Mb/768Kb
02:23.07DovidRB2: That blows.
02:23.47RB2So, for the moment, I'm considering doing what I said I would NEVER do... go with 16Mb service from Comcast.
02:23.50Dovidhere in Israel I have a  meg line (I miss my 30/5 back in Jersey) I get 6-6.5 They push it through so that you get atleast 5
02:24.02DovidRB2: Cable has high latency issues
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02:24.35DovidOptonline boost cable is great for downloads. Now if only I can just live in a Data center...........
02:24.54RB2I've done some testing with Comcast in my local area here and it actually appears not to be too bad compared to other areas I've seen.
02:25.10Dovidcould be u r close to em
02:25.31Dovidin Jersey there were issues. using G729 was bad. G.711U helped out a lot
02:25.34RB2I'm about 2 miles from the local Comcast center.
02:25.41Dovidnot bad
02:25.51RB2You lived in Jersey?
02:26.07Dovidyea. go back and forth between here and there. was there 2 weeks ago
02:27.12RB2My DSL is slow, but I've never had a latency issue.
02:27.17drmessanoJersey FTW
02:27.18DovidBroadvoice should go from
02:27.18Dovid<PROTECTED>
02:27.19jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
02:27.22Dovidto
02:27.25Dovid~ygwypf
02:27.26jbotmethinks ygwypf is You Get What You Pay For.  If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.
02:27.27RB2drmessano, from NJ?
02:27.33drmessanoYes
02:27.47DovidRB2: DSL seems to have very low latency
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02:27.59Dovidalso with DSL its your own circuit. Cable is shared
02:28.36Sinist3rWhy does nmap not show an open port for asterisks web interface?
02:28.38RB2Yeah, I was grilling the local Comcast team to find out how many people were on the loop I would be added to, etc. They couldn't tell me anything hehehe
02:29.02Dovidhehe
02:29.39DovidRB2: My friend was in an area where there was no optimum online. They wanted 20k to pull to hi area. He found some law making them do it ;)
02:29.43Dovidhe moved a year later
02:29.53Dovidhe musta had his own connection there
02:30.30RB2haha Now, OptOnline is, from what I've seen, someone who actually knows how to operate a cable-based ISP.
02:30.53Dovidthey seem to be ok.
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02:31.10Cutlassquestion: to invoke the call park feature in asterisk, does the call between parker and parkee have to be anchored in asterisk?
02:31.14Dovidbut i never used em for TV. DishNetwork was much better
02:31.28Dovidwhat do u mean y by ancored in asterisk ?
02:31.39Dovidanchored*
02:32.06DovidTK: U here ? U seem very quiet tonight
02:32.07CutlassI mean, does asterisk have to be in the signaling path of the original call
02:33.09Cutlass...so what I want to do is...
02:33.15DovidCutglass: Not sure. Why noit try it ? Logic says yes because it needs to pick up on the DTMF (if thats how ur doin it )
02:33.25DovidYou want to test it ;)
02:34.04Cutlassyes...I'll have to try...right now I'm not able to get to my box, but I'm trying to figure out if it's feasible
02:34.32DovidI never tested that way. I always force the rtp through my servers due to NAT issues
02:35.22DovidI don't know the "guts" of call parking to tell u how the DTMF goes through to know if it will work :(
02:35.27Cutlassthe only reason I was not sure is because the parkee is put on hold, then the parker calls the park extension.  that's what triggers the park, correct?  If so, how did asterisk know the extension to be parked unless the call was anchored in asterisk
02:36.19lanningit's a transfer to the park extension.
02:36.37DovidCutglass: Correct. I would assume that the parkee would then get MOH from your asterisk server so the media would go through it
02:37.47lanningYou don't need asterisk to be involved in the original setup, but you need a way to transfer a call to asterisk's park extension.
02:38.14Cutlassoic...I must admit that I have not played with it much so I didn't realize that it was a transfer to that extension.  Is it a blind transfer or consultative?
02:38.21Dovidlannig: I was assuming that the call was feeding throguh asterisk
02:38.39DovidCutglass: attended - cause u need to here where the call was parked
02:38.56Dovidu need to make sure that u do an attended transfer
02:39.03Cutlassyes...ok, that makes sense...
02:39.29lanningCutglass' question was if asterisk was required to be in the original call setup.
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02:39.55Cutlasscorrect...
02:40.23Dovidah ok. 5:40 AM. server issues its been a long night ;)
02:40.35Cutlassit sounds like the answer is "no" due to the fact that a consultative transfer is being made, therefore asterisk becomes aware of both parker and parkee and "park-time"
02:40.44Dovidii agree with lanning: If you can some how send it to asterisk and then grab it from there then ur good
02:40.53Dovidthat would force the call through ur asterisk system
02:40.58RB2Dovid, wow, 5:30am, eh?
02:41.11DovidI work for an ITSP. Sleep does not exist for us
02:41.21CutlassDovid & lanning:  Thanks!
02:41.26lanningnp
02:43.07RB2Dovid, ;)
02:43.42DovidCoffee + Red Bull + Water. I have it down to a science
02:43.54Dovidwhat to drink when it what intervalls
02:45.16RB2hehe
02:45.51Dovidoooh + Smokes
02:45.55Dovidthet go a loooooooong way
02:46.17DovidRB2: Whats gas per gallon in the US now ?
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02:47.49RB2not much has changed in the last two weeks, $3.36-$3.46 in NJ.
02:49.15Dovidhehe
02:49.23Dovidtry $7.25 a gallon
02:49.25Cutlass$4+ in Chicago
02:49.33Cutlasswow!!
02:49.35Dovidnm that the dollar is in the pooper so the Xchange rate sucks
02:49.48Cutlasswhich country?
02:50.07DovidIsrael
02:50.16Dovidwe are on the European system. there its even more
02:50.29DovidI was in Ireland about a year ago. there its ike $8.00 a gallon
02:50.44Cutlasscrazy!
02:51.17Dovidyea. stop cryin !!!!!!!!
02:51.35CutlassI'll think of you next time I'm at the pump :)
02:51.58Dovidmy mom was tellin me that in Jersey it was $4.00 and it was sooooooooo high. When I was there 2 weeks ago I was laughing at the pump
02:52.13Dovidi know the price per barell diped. was wondering if it dipped at the pump as well
02:53.56Cutlassnot sure what it's done in the past week...I've been traveling with no car
02:54.50Dovidhehe
02:54.59Dovidi saw people wit scooters
03:00.25drmessano$6.50 a gallon here right now
03:00.32drmessanoIn the few places that have it
03:00.55RB2drmessano, in NJ?
03:01.29idothat's definitely not in nj
03:01.57drmessanoGA
03:02.00RB2Ahhh
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03:04.30Dovidi am off to bed for a "nap" or atleast I hope a nap. night ev1.
03:04.43Qwellev1...ugh
03:04.44Cutlassgood night
03:05.23drmessanoI think Gas Stations should play the Anthem of the USSR over their loudspeaker and give out a roll of toilet paper with each rationed 5 gallons of gas
03:05.41RB2gnight Dovid
03:05.49drmessanoAT SOVIET GAS STATION, YOU FILL US!
03:05.51RB2lol
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03:06.45drmessanoThe stations that do have gas here have an hour wait
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03:07.09drmessanoI want to get in line and yell "Workers Unite!" every 5 or 10 minutes at random
03:07.12Qwellgrr
03:07.19QwellI can't find gas here...
03:07.25QwellI'm so screwed tomorrow, it's not even funny
03:07.59QwellI'm at < 1/8 tank..
03:08.54drmessanoI have a half tank and 2 onsite calls to run tomorrow
03:09.09drmessanoShould be ok
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03:11.18pcranepossibly a silly question
03:11.43pcranebut
03:11.50pcraneEmbedded Asterisk Business Edition runs 1.4?
03:11.51drmessanoDon't sell yourself short, it's probably a stupid question
03:11.52pcraneor 1.2?
03:11.57drmessanooh..
03:12.02pcranesilly much the same as stupid
03:12.03drmessanoBoth
03:12.14pcraneboth 1.2 and 1.4
03:12.15pcrane?
03:12.20drmessanoDepends
03:12.26Qwellpcrane: it's based on 1.4
03:12.36pcraneok
03:12.59pcraneso, AMD is an application that's available on it?
03:13.21QwellEmbedded?  Highly unlikely.
03:14.30pcranehmm...
03:14.31QwellEABE doesn't exist as a standalone product though.  It's only used on the AA50 appliance
03:14.31pcranethanks Qwell
03:14.42drmessanomissed the embedded part
03:14.52pcraneyeah, hence the quick googling didn't turn much up
03:15.09drmessanoWorkers unite!
03:15.18pcranethanks guys
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03:17.56drmessanoWhat is a 4-letter word for gas rationing?
03:17.58drmessanoCCCP
03:18.27jayteelol
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03:18.32pcrane:D
03:18.54Sinist3rEat less bean burritos
03:19.05Sinist3rELBB
03:19.16pcranenote to self: if you have a g->t typo, don't type Regards at the end of an email
03:20.30drmessanoHAW
03:20.40drmessanoComrades, please..
03:22.10drmessanoThere is no fuel shortage.  Help the motherland.  Have a roll of toilet paper.
03:22.35jayteeMoscow on The Hudson
03:23.06Sinist3rWhy do I get this page not loading when I go to my gui?
03:23.08Sinist3rhttp://209.160.40.98:8088/asterisk/static/config/index.html
03:23.14Sinist3rThis is a fresh install mind you.
03:24.14drmessanoMay I have your attention please.  ignore the man who you have called drmessano.  He has taken quite ill and will no longer be able to join you this evening.  Motherland!
03:25.03jameswf-home~asteriskgui
03:25.29drmessano~gas
03:25.30jboti guess gas is so much cheaper in the US anyway it wont make any difference
03:25.33jameswf-homeSinster try #asterisk-gui
03:25.37drmessanolol
03:26.07drmessanoThe good news is, Saddam Hussein is no longer a threat.
03:26.46jameswf-homeI was thinking about doing a TFTP scan on the interweb and hijack a whole bunch of lines to vote for Americas got tallent
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03:53.24eric2my callerid is showing the number that I'm calling instead of the number I'm calling from
03:53.38eric2and nowhere in the dial plan am I altering it
03:53.41eric2what gives?
03:54.02Cutlasswhat type of channel?
04:14.31jeevjameswf-home, if you watch america's got talent, someone needs to hijack you
04:15.12jameswf-homedude it has the hoff on it
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04:36.27jeevlol
04:41.39jeevasterisk is made with 49% iron
04:43.32drmessano32% skype too
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04:50.13pcraneand 19% blood swear and tears (in varying proportions)
04:51.40jameswf-homeno urine
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04:54.08jeevand some redbull/vodka
04:59.29jplanknothing about skype for asterisk on asterisk.org yet?
05:00.28fakhirhttp://www.digium.com/en/mediacenter/viewpress/Digium-and-Skype-Collaborate-to-Bring-Skype-to-Business-Phone-Systems
05:00.30jameswf-homei would do  the beta but frankly WTF is skype
05:00.32drmessanoNope
05:00.50jplanklol
05:00.52jplanksounds cool
05:00.57drmessanoThis feels like the Metallica black album all over again
05:01.09jplankthey seemed pretty happy about it working at the keynote this morning
05:01.56jameswf-homethere you hav it digiun has sold out so the hard cores go F your selves its main streem time
05:02.06fakhiranyone have a video of the keynote?
05:02.10jplankno
05:02.12jameswf-homedamn korean keyboard
05:02.14jplanknothing really interesting
05:02.20drmessanoor when Kiss took their makeup off
05:03.35jameswf-homewait till they go public... or get bought by AT&T
05:03.42jameswf-homeor both
05:03.44jplankheh
05:03.48jplankbought by AT&T
05:03.59jplankI wouldn't be surprised
05:04.17jplankthe digium guys seemed pretty chummy with the AT&T guys that where there
05:04.24jameswf-homeAsterisk powered by AT&T
05:04.47jplankhell, one of the speeches was inter-op asterisk with AT&T
05:04.48jameswf-homeer SBC err at&t errrr
05:04.54jameswf-homeat&t has mpd
05:05.37jameswf-homewe have at&t formaly cingular now cingular wit now at&t
05:06.05jplankAT&T wireless and AT&T are two different companies
05:06.17jameswf-homestii same name games
05:06.29drmessanoPretty soon you're gonna see the effing asterisk logo on every mobile phone made
05:06.30jameswf-home*still
05:06.36drmessanoThen it will be sell-out city
05:06.43jplankcingular bought AT&T wireless for the name, not for the technology
05:07.02jplankthen we move over to freeswitch
05:07.04jameswf-homeEDGE <__ wtf who would want that
05:07.14drmessanoFreeswitch is a joke
05:07.22jplankhow so?
05:07.32jplankI've heard some good things
05:08.11jameswf-homeI want a blackberry bold its 3G with wifi gps and it will wash you car do your laundry and make you more attractive to women
05:08.12jplankI never used it though
05:08.45drmessanoFailure to innovate.. Trying so hard to be an asterisk killer (which they say they're not, but then they say they're better, but then they say they're not competing), so everything is duplicated functionality "done one better than Asterisk", but lack of anything innovative
05:08.53jplankwell don't get it with verizon, i'm sure all those features will be disabled
05:09.23jameswf-homemy company is contracted through AT&T
05:09.54jplankI thought freeswitch was supposed to be a more efficient asterisk?
05:10.09drmessanoThats the whole problem
05:10.24drmessanoAsterisk sucks, yet it's just like asterisk but better
05:10.31jameswf-homei will get flamed for this but i am seeing some asterisk aspects go the FS way
05:10.44drmessanoI guess Asterisk doesn't suck so much then.. FAILure to "wow" me
05:11.19jameswf-homefiles a bug report: core: failed to WOW! drmessano :)
05:11.25jplanklol
05:11.37jplankyou know what though, what else are you going to use?
05:11.50drmessanoIt would be like if I wrote a clone of MS Outlook, except mail is stored in XML and not JET.
05:12.20jplankwell it just not being MS would make it better
05:12.43drmessanoyeah, thats a semi-intelligent statement
05:13.12jeevjameswf-home i want the bold too
05:13.16jplankI get your point though
05:13.38jameswf-homeon demand needs a playlist
05:13.40jplankclones but different isn't always a bad thing
05:13.45jplanklook at linux
05:13.55drmessanojplank: Its an absolute waste of time
05:14.12drmessanoIf the original sucks so bad, why clone it and then make it better?
05:14.15drmessanoDo something DIFFERENT
05:14.18drmessanoINNOVATE
05:14.26jeevthis skype thing could possibly kill a lot of ITSP business
05:14.26jplankbuts their point isn't it?
05:14.26jeevheh
05:14.36jplankisn't it totally different under the hood
05:14.42mogmaybe
05:14.46jeevsince skype has so much traffic, they can sell cheaper.. boom goes the wiesel
05:14.47jeevor whatever
05:14.49jeevweesel
05:14.58mogskype isnt really competitive with other providers i thought
05:15.01drmessanoI could give a shit under the hood.. I dont look at code.
05:15.03jplankdidn't they do away with the whole channel driver concept?
05:15.07jameswf-homeI think it os pop goes the chicken
05:15.11mogjust give you access to their users
05:15.15jeevit's poop goes to the toilet
05:15.24jeev# Complement existing services with low Skype global rates (as low as 2.1US¢ per minute to more than 35 countries worldwide).
05:15.41jameswf-homeskype is trying not to get burried my gizmo
05:15.43jplankand freeswitch isn't a media gateway
05:15.46drmessanoFS doesn't show me it's any better than asterisk than yate is better than callweaver or callweaver is better than asterisk
05:15.49jplankisn't it more of a sip proxy?
05:16.04drmessanoIt's all the same fish in the same pond with different colored stripes
05:16.37drmessanojplank: Do you even know what Freeswitch is or supposedly does?
05:16.50jplankyea
05:16.52mogi thought callweaver went the way of apherion
05:16.57drmessanoDoesn't sound much like it
05:17.01jeevlook
05:17.06jeevnow that skype is in bed with digium
05:17.06jplankits supposed to be more of a carrier switch
05:17.08jeevthe world is over
05:17.09jplanknot a pbx
05:17.10jameswf-homeFreeswitch will wash you car do your laundry and make you more attractive to women
05:17.12jeeveveryone give me your paypal information
05:17.21jplankmore of a softswitch
05:17.37jplankyour saying freeswitch and asterisk is the same thing
05:18.12jameswf-homedoes it do porn?
05:18.26jplankthey are more concerned about call processing and routing over the fancy features
05:18.26jeevjameswf-home, you know much about asterisk and sms ?
05:18.44jameswf-homeI know nothing about asterisk and sms
05:18.59jameswf-homei know you can make asterisk twitter an sms
05:19.36drmessanoFreeswitch is supposed to be more "carrier grade" because all the cool kids want to work in spaces the other guy supposedly can't compete in..  Read some other blog post and it's a better PBX than Asterisk ever is/was, then "oh, not even on the same playing field.. Asterisk is for building toys... you build empires with FS"
05:19.38jeevahh
05:19.42drmessanoDepends on the day of the week
05:19.45drmessanoAngle of the moon
05:20.09jplankI get you
05:20.16jameswf-home~freeswitch
05:20.17jbotfrom memory, freeswitch is an open source soft switch that is *not* a fork of asterisk http://www.freeswitch.org/interview2.htm
05:20.30jeev~paypal
05:20.30jbotrumour has it, paypal is at http://www.paypalsucks.com and http://www.paypal.com
05:20.52jameswf-home~sex
05:20.53jbotupdatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/
05:21.30jplankdrmessano: why are you such an angry person?
05:21.35drmessanoI am?
05:21.41oilinkihoho
05:21.43jplankyou always seem so
05:21.48drmessanoUh no
05:22.04jeevjameswf-home, running the command wont get you laid
05:22.05jeevf00l
05:22.11jameswf-homehe is competing with n3glv
05:22.17drmessanoha
05:22.38drmessanoI am just opinionated.. BS is BS
05:23.11jplankyaaaay my flight is ready to board
05:23.13jplanknight all
05:23.20jameswf-homeI hear everytime you use skype god kills a puppy.... I'm just saying....
05:23.26jplankLOL
05:23.30drmessanolol
05:24.33drmessanoI guess I am also not getting paid or make money from telling someone that one week Grandstreams pwn, and the next week that anything but Cisco is for loser
05:24.37drmessano+s
05:24.58jeevi wonder if there were a stat for everytime someone acquires a virus or spyware on windows.. what's god do?
05:25.05drmessanoI guess that means I won't be using trixbox anytime soon
05:25.48drmessanoDunno.. what about everytime some loser *nix admin opens up SSH with a weak password and get pwned?
05:25.57drmessanoAhhh... we don't like to talk about those guys
05:33.11jameswf-homewith all due respect most trixbox users are not linux people
05:34.28drmessanotrixbox users don't even come into play here
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05:39.22drmessanoYou're hard pressed to find anyone who really wants to talk about security objectively as a function of us flawed human beings who create and manage these systems, nevermind "MY OS is bigger than yours".  I just find the lack of consideration that while windows boxes up to Vista gave every use god rights on every fresh install, how many *nix boxes have lazy admins running everything as root because it's less hassle, and weak SSH passwor
05:39.59drmessanoI don't care what OS you're on.. People are wreckless.
05:40.11drmessanoPeople are the secuirty threat
05:42.05*** join/#asterisk oilinki (n=oil@ppp-124-120-17-190.revip2.asianet.co.th)
05:44.43drmessanoIt would be interesting to see feature in asterisk to now allow secrets to contain weak passwords
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05:47.56mogasterisk allows weak passwords
05:48.07mogbe more interesting if it complained to you about it
05:48.32EmleyMoorI'm having trouble doing a regular expression match - I came up with something that turned out to be "always true" - and I also want it in conjunction with another condition.
05:48.37drmessanonot* allow
05:48.50drmessanoWhich is what I was saying
05:48.54drmessanoI know it allows them
05:49.08EmleyMoorWhat I want is to detect when the first two characters of something are 0[1-9] AND the same string is longer than 11.
05:49.24mogahh
05:50.24EmleyMoorIt lost me a couple of calls last night (in that they went a different way, not that they should have rung my phones and didn't, fortunately)
05:51.14drmessanoI think maybe a compile-time option to not load a peer definition that doesn't follow the basic definition of a "strong password", being a minimum of 6 characters, to include 1 number and 1 capital letter minimum
05:52.16drmessanoOf course, this is all going back to the same person who would be setting the weak password in the first place and can't be trusted lol
05:52.48drmessanoBut I suppose in trixbox like systems, the integrator could set it for the compiled binaries, which would at least help the issue
05:53.52drmessanoAt the same time, even the best of us have created a test/test account before
05:54.08drmessanoOnly takes a lucky second by a lucky person to exploit it
05:54.39drmessanoWould be a nice barrier to make you check yourself.. "Oh crap, thats right, cant do that"
06:00.17EmleyMoorIn an expression, can you put multiple conditions together with | (OR) or is it limited to two?
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06:02.36*** part/#asterisk SpaethCo (n=SpaethCo@cache.spaethco.com)
06:03.45EmleyMoor(or do you have to do [ 1st | [ 2nd | 3rd ]])
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07:27.22baxtashi all. did someone knows how to solw misdn sound recording problem. when *.wav and *.gsm files become unable to listen. like in example http://drd.chita.ru/test.wav .
07:28.30rednodeHas anyone here integrated genesys with asterisk
07:28.43rcyno, but we just hooked our toilet up to asterisk
07:29.12rednodelol
07:29.25rcywebcam coming in a couple seconds
07:29.36rcyer, minutes maybe
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07:48.10rcyhttp://thehackery.ca:8090/zm
07:48.32rcy604 637 6414 ext 750
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07:48.54creativxlol wtf
07:49.59krdianrotfl
07:53.51creativxis can be called?
07:53.52creativxhehe
07:53.54creativxis.. it
07:54.02creativxi can haz toilet?
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07:57.28rcyyeah, that number flushes the toilet on that webcam
07:57.42torrikfthey all, what do i need to add in my SIP trunk config to point destination to port 5061 instead of 5060
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07:58.39rcycan someone call the number and flush this?
08:01.52creativxwell rcy
08:01.54creativxhow do i dial it from intl
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08:19.46rcycreativx: can you see the toilet?
08:19.50rcywe'll put yer name on it
08:19.58rcyer in it
08:20.01rcy> creativx: its canada, +01 604 637 6414
08:20.42creativxrcy: yeah i can see it
08:20.51rcyok, yer nick is going in
08:21.00rcysee that?
08:21.13creativxhaha yep
08:21.16creativxscary
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08:21.26rcycreativx: where you calling from?
08:21.35creativxnorway
08:21.56rcyboom
08:21.57creativx"thankyou"
08:21.58creativxhehe
08:22.01iamturnipcreativx: thanks man!!!
08:22.06creativxhaha..
08:22.08creativxtechnology
08:22.10iamturnipnew record
08:22.10creativxglad its friday
08:22.16iamturnipcalifornia ws the last one
08:22.43creativxheheh
08:23.47creativxso how much weed went into this idea
08:23.51creativxcanadians are crazy
08:24.00creativxthats what i learned from watching trailer park boys
08:24.12rcyno weed, but a bit of beer
08:24.39rcywe'll pull something more ridiculous next time we're smoking
08:25.24creativxhehe =)
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08:42.18glazwe're no crazy :)
08:42.27glazWe're just different.
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08:57.03rcyhttp://thehackery.ca:8090 updated url
08:57.10rcygot rid of zoneminder crap.
08:57.14rcy*flush*
09:02.55*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
09:03.39ZeeekI have found lightning to be handy for trackin my wife's activities
09:04.45iamturniprcy: but now its not as *live* hmm more work required
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09:09.58mchouholy crap.  That's why the guys use asterisk :)
09:10.21mchouto find out who their wifes are cheating with :)
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09:26.56rednodedoes asterisk work on any netgear router or would i need a specific voip router?
09:30.41Zeeekhi tzafrir_laptop
09:30.58tzafrir_laptophi Zeeek
09:30.59Zeeekany decent router will work with asterisk
09:31.49ZeeekI am watching VP candidate Palin interviews and I feel ill
09:32.01tzafrir_laptopI'm not sure I'm the one to answer that. I don't work much with routers...
09:32.27Zeeekmchou:  lightning is for Thunderbird, not asterisk :)
09:33.10Zeeekthrere are a few diffs in routers such as whether they allow hairpinning (going out and coming back in via SIP
09:34.30tzafrir_laptopThe hebrew name of the scientific species of "moose" is literally called "the calling moose" or "the reading moose"
09:35.01tzafrir_laptopThere is hence a thriving discussions site in Israel called in that name
09:35.33tzafrir_laptopRecently someone posted there a news item for discussion:
09:36.21ZeeekA news item?
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09:38.42ZeeekI feel better: media is showing this in the light of context. She looks bad and it is being seen as looking bad. Thsi is good
09:38.48Zeeekenough if politics
09:39.40tzafrir_laptopa news item for discussion: "Palin lists moose stew and moose burgers among her favorite foods"
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09:40.45tzafrir_laptopbah, my internet conection is bad
09:41.57IllaraneI love some of the things Asterisk does when you mis-configure it.  One of my colleagues just managed to get us a minute's worth of repeated "number not found" messages from various countries before it finally hung up and gave us a proper error tone. :D
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10:12.05teleniekohi ppl. is there any reason why "sip show peers" shows lots of people but "dialplan show sipregistrations" shows only one?
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10:20.26Zeeektelenieko: what are you actually looking for?
10:21.05teleniekoZeeek: As fas as I understand any sip peer that got registered should appear in sipregistrations
10:21.10teleniekoand only a few of them actually appear
10:21.53Zeeekshow peers would show the peers in your sip.conf that aren't registered, too
10:23.08teleniekosure, but they're shown as "OK", I also see the "Saved user agent ..." and "Registered SIP..." messages
10:23.34teleniekoand sometimes "Added extension ..... to sipregistrations priority 1"
10:23.46teleniekobut then they appear/disappear from sipregistrations :o
10:24.21teleniekoI just restarted asterisk, lets see if phones appear :o
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10:46.15jazzmannhi can anyone help me setup softphone behind smoothwall firewall.It is keep saying login time out,firewall blocked.I opened all the relvent port.Using Xlite and centos5
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10:52.14NuitariHi anyone
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11:07.32jazzmannhi can you help me in setting up softphone behind firewall
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11:17.20jblack~sipnat
11:17.22jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
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11:27.44tompawhello
11:30.01tompawI have a question regarding good configuration practices: let's say I have a remote sip server, where my asterisk registers as SIP client
11:30.29*** part/#asterisk telenieko (i=marc@telenieko.com)
11:30.34tompawI define this server as a trunk in users.conf and later call is using Dial(SIP/trunk-name/number)
11:31.07tompawnow, what to do if I also want to ACCEPT calls from that server?
11:31.36tompawnote, that on that server I am visible as some extension, which obviously is different that my local phone's extension at my asterisk
11:32.06tompawthat would suggest I need a dedicated context, which will do exten => remote_extension,1,Dial(SIP/local_extension)
11:32.48tompawand my question is: should I simply add that extension to my trunk's definition in users.conf, along with insecure = very?
11:33.05tompawor should I do a separate definition of that incoming trunk in sip.conf?
11:34.07tompawso: A) users.conf trunk - bound it to a context and add insecure = very or B) sip.conf define an incoming trunk with context and insecure?
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11:55.25Blackvelhi all. setup voicemail and voicemailmain with asterisk 1.2.30.2. its missing sound files (like digits/1F, digits/6-and...). can I safely use sound files from 1.4.21.2?
11:56.10torrikftgive it a try
11:56.21torrikftif filenames are the same
11:56.24torrikftit will work
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11:56.46Blackvellike there is only 1.gsm, 1.ulaw, 1.alaw and 6.alaw. but no 6-and.alaw e.g
11:57.11Blackvelthere are no special sound files on ftp like germans?
11:57.54torrikftlocalization files go into their own folder
11:58.29torrikftinside /var/lib/asterisk/sounds
11:58.43Blackvelfound this: http://downloads.digium.com/pub/asterisk/old-releases/asterisk-sounds-1.2.1.tar.gz.
11:59.41torrikftwell ther eu go then
12:00.03viperdude_ukhi all
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12:01.31Blackveldo you guys use snom 370 wmi retrieve button? I am already using vmext 999 (extensions.conf with _999,VoicemailMain...) and fromdomain. but when the snom 370 tries to dail the '999@192.168.1.2' it gets forbidden. when I manually dial 999 extension its working
12:02.00Blackveldo I have to go into sip debugging?
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15:06.18*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
15:06.18*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc6, 1.4.22-rc5 (2008/09/09), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12.1 (2008/09/09), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon
15:06.20tzangerlike when they fall off a table after trying to perform some acrobatic move, they sit down immediately and wash their face or something and look at you like "what? I meant to do that."
15:06.23Kattythat should have said Chief Taster says your food is ackceptabul for hooman consumption
15:06.36tzangerheh
15:06.36KattyChief Taste Tester
15:06.39c4t3l5w33t lolc@t!!
15:07.01Kattythedonvaughn: [TK]D-Fender might know.
15:07.04*** part/#asterisk TheBonsai (n=bonsai@unaffiliated/thebonsai)
15:07.08Kattythedonvaughn: not sure where he's at tho.
15:07.32c4t3l[TK]D-Fender knows everything
15:07.41Kattynot everything
15:07.54c4t3li have yet to run into an issue he could help on
15:08.01thedonvaughnthanks
15:08.01c4t3lcouldn't**
15:08.01*** part/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za)
15:08.22c4t3l:D
15:08.57Kattyi have.
15:10.01c4t3lwhat kind of issue??
15:10.05Kattykm-: you been to target lately?
15:10.08*** part/#asterisk jtarlton (n=jtarlton@87-194-179-73.bethere.co.uk)
15:10.20Kattyc4t3l: not geekery related kind.
15:10.28km-hmm, target... probably about a month ago, why?
15:10.29c4t3lahh!!
15:10.52Kattykm-: they have a new toy back in the kids dept. a robotic dino
15:10.52km-bought a bunch of school supplies for a fundraiser
15:10.52*** join/#asterisk spike008t (n=Spoukie@ven69-2-82-228-116-153.fbx.proxad.net)
15:10.54spike008thi all :)
15:10.57c4t3lhas anyone here ever used the Nokia N800 series internet tablet?
15:10.59Kattykm-: remote controlled, lights and sounds.
15:11.04km-hahaa, awesome.
15:11.05Kattykm-: about... 36" tall or so
15:11.10km-PET TORMENTOR ENGAGE
15:11.10Kattykm-: i'm thinking new puppy toy
15:11.17km-my cats would FREAK OUT.
15:11.22Kattysadly, i can't find it on their website
15:11.27km-ah nuts
15:11.29c4t3lhehe
15:11.36km-if I see one I'll pick one up for ya
15:11.39spike008tThese someone have already work with the iaxclient lib?
15:11.57Kattykm-: nah it's cool.
15:11.59Blackveldoes DBPut replace the old key or do I have to call DBDel before for replacement?
15:12.09c4t3li run mythtv at home and from time to time I'll shell in and talk to my cat via festival.  I know that freaks him out
15:12.10Kattykm-: at this poitn, i think the puppy is smaller than the dino. i wouldn't want the terrorize the pup ...just yet
15:12.19km-katty: hehe.
15:12.58Kattykm-: actually i'm going to bottle feed riddick for a couple more weeks.
15:14.43tzangerhow young is the pup that you have to bottle feed?
15:15.25Kattythe pup is 8 weeks and already weaned.
15:15.38Kattybottle feeding a pup just makes them sweeter.
15:15.42tzangerso you're unweaning him?
15:15.44Kattyno
15:15.52Kattyit's like giving a cat a saucer of milk
15:15.54Kattyit's yummy
15:15.58Kattyit does not unwean the cat.
15:16.11Kattyand no, i'm not giving the pup cows milk. that'd be awful.
15:16.12tzangerwell no, but the cat has to drink the milk like he would water
15:16.24Kattynot neccesarily
15:16.29Kattyi've bottle fed cats too
15:16.32Kattyeven adults
15:16.37Kattytho, they have a tendancy to chew on them.
15:16.55tzangerinteresting
15:17.02Kattyi've even bottle fed a goat
15:17.11Kattyweaned a pgymy
15:17.19Kattythat was all sorts of crazy mahhing fun
15:18.23*** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr)
15:18.27*** join/#asterisk arpu (n=arpu@chello084114025240.14.vie.surfer.at)
15:18.30ZeeekAt 12 Noon Eastern stop by http://voipUsersConference.org and join #voip-users-conference for the Friday conference
15:18.47Kattyhai Zeeek
15:19.22KattyOHOH!
15:19.25Kattykm-: it's on the website now!!!
15:19.41ZeeekKatty:  what?
15:20.10Kattycheers. someone youtubed it.
15:20.16Kattykm-: http://www.youtube.com/watch?v=7Qw6rLrJX4E
15:20.18Zeeekoh
15:20.20Zeeekya
15:20.36Kattykm-: puppeh toy
15:20.43Zeeekbut I want to hear your opinions of this and more on the conference call
15:20.50Kattyi'm sure you would.
15:21.02ZeeekYou KNOW I would
15:21.56Zeeekwhich is why I hope you will jopin us on the conf
15:21.56Kattykm-: they make lots of different dino models, apparently.
15:21.56Zeeekmy typing is rich
15:21.56Kattyquite rich.
15:22.24Zeeek"My tailor is rich.?"
15:22.33ZeeekAnyway, carob milk
15:22.39Kattymy tailor is broke. always broke.
15:22.57Kattyi'm sure Qwell can relate to the broke tailor.
15:23.08Qwellhuh?
15:23.14Qwelloh
15:23.19Kattygood rebound.
15:23.21Kattyless than a minute.
15:23.22Kattyclaps
15:23.28Zeeek`Qwell
15:23.36Qwell`Zeeek
15:23.45ZeeekIn a few moments, we'll open the bar and buffet
15:24.00ZeeekPlease joinus: http://voipuserconference.org
15:24.13Qwellno phone
15:24.19jeevwhat kind of food
15:24.34Zeeekaw
15:24.42jeevyou misspelled your website name
15:24.43Kattybrain food.
15:24.46*** join/#asterisk knobo (n=bohmersp@148.122.202.168)
15:24.50Zeeekshit
15:24.51Kattyalso, now i'm hungry. thanks
15:24.55Zeeekhahaha
15:25.05Kattykm-: http://www.youtube.com/watch?v=4rquLG8tw2M <- this is a much better video
15:25.06ZeeekI'm on a different conference now
15:25.06jeevhey
15:25.10jeevi heard buffet
15:25.18ZeeekCOME see us on voipusersconference.org
15:25.46jeevZeeek, there isn't a www there so i can't click on it. you're not making it easy!! and what kind of food ?
15:26.40Kattyi think you just need to give me the sip number
15:29.50Blackvelanyone using SMS application? can I use it to send over my sip patton isdn gateway a sms from pstn (landline) to mobile phone?
15:30.00Kattyalso, carob milk?
15:30.02Blackvelthe other solution would be to use the sms gateway provider with the FastSMS application
15:30.40jeevBlackvel, i'm looking for a method too... i've found some services but i want to know how i can not use them and do what they do myself
15:30.57ZeeekI think there is one: http://food4wine.ning.com
15:31.24ManxPowerBlackvel: PSTN SMS uses modem tones, so if a modem works thru that setup running at (I think) 2400 baud then it should work.
15:31.48jeevManxPower, if i wanted to be my own SMS gateway, or aggregator or whatever, what would i need
15:31.57c4t3l<PROTECTED>
15:32.04ManxPowerjeev: where are you located again?
15:32.05*** join/#asterisk CanWood (n=chatzill@24.108.64.80)
15:32.06c4t3ldoes such a thing work?
15:32.22jeevLA
15:32.38ManxPowerjeev: this SMS discussion does not apply to USA/Canada
15:32.54Zeeekfor the VUC, you call in on SIP, PSTN. There should be a click 2 call on http://food4wine.ning.com
15:33.11*** join/#asterisk Nate187 (i=Nate187@64.82.228.228)
15:33.18*** join/#asterisk leif[astricon] (n=Leif@63-255-103-7.ip.mcleodusa.net)
15:33.18*** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-218-95.phlapa.east.verizon.net)
15:33.19*** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34)
15:33.30jeevhow come
15:33.47ManxPowerjeev: because USA carriers do not use FSK SMS
15:33.58QwellQwell Communications does
15:34.04Dr-Linux|homeanybody guide me that what's the "wireshark" command to trace the Asterisk SIP call on local host?
15:34.28*** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa)
15:34.35c4t3ltcpdump :D
15:34.49a1fahas anyone noticed increased bruteforce on their asterisk
15:34.50*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
15:35.01a1fain last 3 days, 10 different IPs hit my box trying to crack users password
15:35.15jeevManxPower, so if i wanted to send sms out on a massive scale... my only option would be to use a company like blackhamsystems or wahtever theyr'e called?
15:35.17c4t3lvia sip?
15:35.26a1fasip
15:35.27ManxPowerjeev: correct
15:35.28moga1fa, i have
15:35.34a1faweerd
15:35.40mogi recently got spammed by three different people
15:35.41a1fai need to put a firewall on my system again
15:35.48a1famog: maybe same IP
15:35.49mogfortunately i just was able to firewall it off
15:35.51Qwella1fa: fail2ban
15:35.52mogas i have one site
15:35.55mogseveral
15:36.02Qwellgood at getting rid of brute force attacks
15:36.04jeevManxPower, i hate the sound of that
15:36.04moghalf in china
15:36.06moghalf here
15:36.13ManxPowerjeev: move to Europe then
15:36.17*** join/#asterisk outtolunc (n=me@c-24-130-75-122.hsd1.ca.comcast.net)
15:36.18Dr-Linux|homec4t3l: and how can i do it with "wireshark"?
15:36.21a1faawesome Qwell
15:36.24ManxPowerHeck, I'd move to Europe anyway.
15:36.38a1fanow If i can only find an IP for a successfull connect in my log files
15:36.41jeevManxPower, so those companies are located in Europe ?
15:36.55a1famy remote phone lost DNS settings, so now I need to connect to it and update, if i knew its IP
15:36.55ManxPowerjeev: European carriers support FSK SMS
15:37.15jeevok
15:37.16jeevbah
15:37.17Nate187I'm migrating a system from 1.2 to 1.4 and ran into a problem. There's no Voicemail application and app_voicemail.so is missing
15:37.34jeevso it's either that or tapping into the SMTP service providers give..
15:37.44ManxPowerNate187: Did you read UPGRADE.txt and UPGRADE-1.2.txt ?
15:38.09Nate187no, i installed via yum with fedora 8
15:38.13Nate187i'll check there
15:38.18ManxPowerNate187: go read them
15:38.45ManxPowerAlso we can't help you with packages here, if you have problems you should contact the packager.
15:39.52a1fado successful authentications get stored in log file?
15:40.15a1faall I am getting is
15:40.18a1fa450 successful
15:40.36a1faerr.. reachable
15:40.47ZeeekThe SIP and PSTN numbers are both apparent at http:/voipusersconference.org - no googler ads, no ads, no bull at all, just the info ma'a''m
15:41.07ZeeekTalkshoe will be open in about 10 min max
15:41.15QwellZeeek: aren't those ads on the top?
15:41.42Blackvelwhat can I do? the asterisk ivr does not work with dtmf on landline/pstn phones or mobilphones (internally it works with snom3 and sip dtmf inband as well as with snom m3 and dtmf info)
15:41.44leif[astricon]Zeeek: are you talking about VoIP this week?
15:41.59Blackvelis this a problem of my patton isdn-to-sip gateway?
15:43.02ZeeekQwell: no, those azre sponsors who help keep it alive. Digium is part of that
15:43.19ZeeekYes! VOIP, vOiP and possible VoIP
15:43.27*** join/#asterisk beek (n=klinebl@65.211.106.242)
15:43.29ZeeekAND Astricon
15:43.43ZeeekTHE BAR IS OPEN
15:43.44c4t3lbut what about "Voice Over IP"
15:43.44Qwellahh, advertising sponsors isn't advertising...  got it. :)
15:43.57Zeeekleaves to get serveed first as usual
15:44.10c4t3lhere comes the flood!
15:44.53*** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net)
15:45.40c4t3lNate187:  If you ask some people here (well most I'd say), you are much better off compiling from source
15:45.45*** join/#asterisk sack (n=sack@106.Red-88-24-156.staticIP.rima-tde.net)
15:46.12jeevNate187, compiling from the source is as easy as taking a shower
15:46.34c4t3lNate197: a warm shower!
15:47.29jeevc4t3l, a golden shower even
15:48.44c4t3llol!
15:49.40c4t3lnew ad campaign... "Asterisk:  It's as easy as taking a warm golden shower."
15:49.55c4t3lok, I'll be good now
15:50.04c4t3l>.<
15:50.13jeevlol
15:50.14jeevit is
15:51.56Nate187yeah, its nothing to do with actually upgrading. somebody didn't bother to make a voicemail package for fedora 8, gotta do a source install
15:52.03*** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa)
15:52.10*** join/#asterisk jtodd (i=cg5b52so@ns2.loligo.com)
15:52.15Qwelljtodd: nice shirt
15:56.23Kattyasterisk supports both G.711u and G.729a right?
15:56.56Kattyah
15:57.01Kattyshow codecs.
15:57.03Kattyhow handy.
16:01.21km-what was jtoddd's cool shirt?
16:01.45Qwellkm-: something only jtodd can get away with - a tie dye polo
16:04.26km-hahaha
16:04.29*** join/#asterisk nny_2 (n=scott@64.203.244.146)
16:04.30km-rock on with your bad self
16:06.03leif[astricon]jtodd: has some pretty funky shirts
16:06.08leif[astricon]I saw a new one this year
16:06.26ajohnsonthey definitely stand out
16:07.29*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
16:07.48*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
16:08.04*** join/#asterisk BobPierce (n=BobPierc@216.36.132.162)
16:13.07Katty8-bit tie would be awesome with a tiedye polo
16:16.57Kattymaybe i can put an 8-bit tie on riddick
16:17.05Kattyhow do you get a clipon tie onto a dog?
16:17.16hardwirewhat a cute name for a dog
16:17.20hardwiredoes he have crazy eyes?
16:17.22Kattythat's not the half of it
16:17.39Kattyhardwire: http://www.new.facebook.com/album.php?aid=2131303&l=de518&id=37617946
16:17.45jayteevery blue eyes
16:18.00jayteealmost the color of a Kashmiri sapphire
16:18.00Kattythey're probably turn brown tho
16:18.05Kattylike most puppeh's whent hey grow up
16:18.07hardwireaww
16:18.20hardwirehe's an aussie mix?
16:18.24Kattyno
16:18.33Kattyhe's a full blodded GSD
16:18.35Kattyblooded, aslo.
16:18.37Katty...
16:18.38Kattyalso.
16:18.39hardwireno kidding
16:18.45Kattyhe's just 8 weeks tho
16:18.50jayteedon't ya wish you could take them to the vet and he'd give them a shot and they'd stay puppies forever?
16:18.51hardwireit's so hard to tell what the heck they are when they are little
16:18.52Kattyhe's all paws, chin fluff, and floppy ears
16:19.03Kattyjaytee: oh hell no
16:19.08hardwireI have an american bull dog / pit bull mix
16:19.13jeevpuppies are the ish
16:19.16Kattyhardwire: bobbed tail?
16:19.18jeevjaytee, you're not helping your case
16:19.22jeevyou rollerblade, love puppies
16:19.25jeevthere was somethin else too
16:19.37jayteethe journal?
16:19.42nny_2jesus
16:19.54jeevyes
16:19.57nny_2can chan_dahdi spam errors just a bit more?
16:19.57hardwireKatty: nope.. long drink spilling heavily wagging tail
16:19.59hardwireno cropped ears
16:20.01hardwirevery pretty dog
16:20.04hardwiregoofy little grin
16:20.06jeevjaytee, you're either the gayest gay guy in the closet or you're everything a woman wants!\
16:20.20hardwireand we have a southern united states hound dog mix too
16:20.49Kattycute (=
16:21.12Kattyjeev: simmer down now.
16:21.20Kattyjeev: jaytee's a sweety.
16:21.23tzafrir_laptopnny_2, what errors? What version of asterisk?
16:21.24jayteejeev, and why should I care what some technically inept, anti-semitic, crazy persian with delusions of grandeur who thinks Amadinejad is a great guy thinks?
16:21.34Kattyjaytee: okay you simmer down too
16:21.38Kattyeveryone just simmer!
16:21.49hardwiredigs up a pic for Katty
16:21.49hardwireoh
16:21.51hardwireduh
16:21.57jeevlol!
16:21.57[TK]D-FenderO>o
16:22.03jeevpolitical attack
16:22.05Kattyoh look, it's fender.
16:22.07jeevjaytee, because i'm kidding around
16:22.08Kattydid you bring me lunch?
16:22.18[TK]D-Fenderjaytee / jeev : Seriously, tone it down.
16:22.34Katty[TK]D-Fender: tone yourself down.
16:22.40Katty[TK]D-Fender: i need to tone too
16:22.48Katty[TK]D-Fender: you're probably toney enough as it is ^_-
16:22.52Katty[TK]D-Fender: how's your thumb?
16:22.56[TK]D-FenderKatty: Yes... my portrait alone can bring up lunch.  Its best viewed at least 2 hourse BEFORE eating ;)
16:22.59hardwirekatty http://mommashila.blogspot.com/
16:23.21Kattyoooh, youtubage
16:23.39*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
16:23.39[TK]D-FenderKatty: I'm close to "toning" myself.  I lost almost 25lbs in 2 months (still time), and then aim for tone.
16:23.41Kattyhardwire: 'wiggle' is the cutest
16:23.56jeevhardwire, that guy needs a vacuum!
16:24.08*** join/#asterisk gr0mit (n=tim@81.187.32.146)
16:24.08hardwireKatty: yer dog needs a blog
16:24.10jaytee[TK]D-Fender, was just reading about the "raw" diet where you eat as much uncooked, unprocessed foods as possible.
16:24.15Kattyhardwire: my dog has a blog.
16:24.20Kattyhardwire: well i have a blog. which my dog is on
16:24.22hardwirejeev: that dog sheds.. constantly.
16:24.25[TK]D-Fenderjaytee: belh
16:24.30*** join/#asterisk gloin (i=me@unaffiliated/gloin)
16:24.31Kattyhardwire: tho i don't blog much.
16:24.31nny_2tzafrir_laptop: 1.4.21.2 one sec getting log start entry
16:24.32jeevhardwire, that was TRASH on the floor!
16:24.41jayteeI'll take that as a bleh
16:24.42hardwirejeev: yeh.. she sheds and out comes trash
16:24.44gloinMorning all
16:24.46Kattyhardwire: nose art!
16:24.47jeevlol
16:24.51tzafrir_laptop1.4.21 has chan_zap, not chan_dahdi
16:25.06[TK]D-Fenderjaytee: add it to the pile-up of Atkins, South Beach, juice, banana, De Montignac, yadda-yadda-yadda.  All raging BS.
16:25.29Kattystop talking about food
16:25.31Kattyi'm starving
16:25.32[TK]D-Fenderjaytee: I subscribe to the massively successful "Stop eating, you fat bastard" diet :)
16:25.33hardwirejeev: we keep a clean house.. but every so often we go crazy w/ the dogs and everything goes everywhere
16:25.37jayteelol
16:25.41jeevheh
16:26.17Kattydisappears for a bit
16:26.27[TK]D-FenderI did 70 lbs in 5 months, and years later now, 25 in 2 (minor rounding+est)
16:26.37gloinSo listen, I've got an Asterisk 1.4 box, runs nice, uses SIP trunks, those work fine, EXCEPT: when I have a pin set in a meetme room, callers from outside our PBX can't enter a pin.  It's as if the PIN gets ignored.
16:26.50[TK]D-Fenderjaytee: I'm currently about 187 @ 6'2"
16:26.53gloinI removed the PIN from a room, dialed in from outside, and can hear keypresses
16:27.08gloinbut for some reason, they're not being heard by asterisk at the right time
16:27.17[TK]D-Fendergloin: Go prove that DTMF works elsewhere.  I highly doubt it.
16:27.25[TK]D-Fendergloin: You probably picked the wrong mode for your peer
16:27.28jaytee[TK]D-Fender, that doesn't sound overweight
16:27.40[TK]D-Fenderjaytee: not now :)
16:27.42gloin[TK]D-Fender: I can use keyed PINS from within the PBX
16:27.53gloinif I dial the extension directly, it works fine
16:28.12[TK]D-Fenderjaytee: I have been as low as 176 & really low fat %
16:28.24jayteedwarves from the Misty Mountain have no business running Asterisk in the first place :-)
16:28.26nny_2tzafrir_laptop: basically i get [Sep 24 15:25:02] NOTICE[3221] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 5 times per second when the T1 has an issue
16:28.28[TK]D-Fendergloin: That peer is set wrong.
16:28.42gloin[TK]D-Fender: can you expand on that a bit?
16:29.17[TK]D-Fendergloin: Go do some IVR testing with the same peer and prove if DTMF is globally clearly off.
16:29.48glointhe peer in this case is a shoretel pbx
16:29.52gloingack
16:30.04nny_2tzafrir_laptop: the cause of original issue we had is still being debated, as the logs just say "Red Alarm" when the issue started and it daisy chained from there
16:30.14*** join/#asterisk simNIX (n=simNIX@82-204-21-111.dsl.bbeyond.nl)
16:31.39*** join/#asterisk CunningPike (n=arodgers@64.251.77.9)
16:31.43*** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
16:31.50*** join/#asterisk fudpucker1 (n=here@h-67-103-23-130.phlapafg.covad.net)
16:32.44*** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org)
16:32.53*** join/#asterisk CunningPike_ (n=arodgers@vpn.dnv.org)
16:34.42gloin[TK]D-Fender: thanks, that was actually it (config option in st was "Enable SIP Infor for G.711 DTMF Signaling")
16:35.26jeev[TK]D-Fender, how is 187 6'2 fat?
16:35.39jeevahh, didn't see your not now thing
16:35.42[TK]D-Fenderjeev: I never said it was.
16:36.13[TK]D-Fenderjeev: I was very "average" at the start and I've dropped a pants size & 1/2
16:36.30Blackvel<PROTECTED>
16:36.32jeevahh, i'm 6'1, 194 now, i used to be a fatty in high school.
16:36.43*** part/#asterisk gloin (i=me@unaffiliated/gloin)
16:36.49Blackvelhow can that be? the snom phone is ringing! shouldn't it be NOANSWER?
16:36.49jeevimmediately after, i lost, i went to 6'1 187.. then after i hurt my ankle, i gained some weight, went up to 205..
16:36.56Blackveltimeout is set
16:37.06jeevi dont like to exercise cause i'm unable to breathe well through my nose (i've hurt it a lot in sports)
16:37.15jeevbut i have a very big build so thats good at least
16:37.54nny_2so last week we had a client lose the T1 connection, and the issue was compounded by the telco bringing the T1 down for further testing AFTER the initial event, or so they say. We found this out when everything started magically working again 3 hours later. I am going over the log and have found what looks like the start of the event
16:38.03fudpucker1is there any way i can tell zaptel to let go of /dev/hwclock.  it makes it difficult to update the hwclock when zaptel isrunning
16:38.43*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
16:39.04*** join/#asterisk xuser (n=JJ@unaffiliated/xuser)
16:40.07nny_2here is the event http://pastebin.com/m554080cc
16:40.17nny_2i noticed a write error in there somewhere as well
16:40.30*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
16:42.01Blackvelohhh. I think I forgot to set an dial timeout both for 1st SIP device and 2nd Local device
16:42.29Blackvelwhat a shame :)
16:42.54*** join/#asterisk jtodd (i=va9kq098@ns2.loligo.com)
16:43.09tzafrir_laptopfudpucker1, hwclock? what kernel do you use? What OS?
16:43.29nny_2i am trying to figure out if the T1 went down or if the pri card/ asterisk etc had an issue, although there isn't enough evidence so far for either
16:43.46*** part/#asterisk viperdude_uk (n=jon@87.127.75.241)
16:44.02nny_2tzafrir_laptop: nah i was just asking about the HDLC Abort error cause it floods the hell out of the logs
16:44.16tzafrir_laptopnny_2, this is just on startup?
16:44.31nny_2tzafrir_laptop: no on a T1 failure
16:44.55nny_2tzafrir_laptop: i am still trying to discern why the failure happened
16:45.01tzafrir_laptopThe span is in red alarm, and thus it sends an alarm on each channel
16:45.06fudpucker1Centos 5 x86_64
16:45.13nny_2yeah that part is understandable
16:45.18nny_2but then i get:
16:45.33fudpucker12.6.18-92.1.10.el5-x86_64
16:45.35nny_2http://pastebin.com/m554080cc
16:45.37nny_2no
16:45.53*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
16:45.53*** mode/#asterisk [+o russellb] by ChanServ
16:45.53nny_2dammit why doesn't control C work for copy in Ubuntu and gterm :]
16:46.03nny_2[Sep 24 15:23:59] WARNING[3221] chan_dahdi.c: No D-channels available!  Using Primary channel 24 as D-channel anyway! [Sep 24 15:25:02] NOTICE[3221] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
16:46.29nny_2NOTICE[3221] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 <-- Gets spammed 5 times per second for 10 minutes or so
16:46.35nny_2literally pages of the same error
16:47.15tzafrir_laptopctrl-c stops a program. You should know that
16:47.25nny_2tzafrir_laptop: ha yeah it
16:47.25fudpucker1ctrl-insert should do it
16:47.29nny_2's shift ctrl c
16:48.37*** join/#asterisk propellerhead (n=yogurt2u@host38.190-137-2.telecom.net.ar)
16:48.52tzafrir_laptopseems you have a bad line if it happens several times over 10 minutes
16:49.47*** join/#asterisk Shotygun (n=thorn@82.166.246.66)
16:50.16nny_2tzafrir_laptop: :D
16:50.26nny_2tzafrir_laptop: yes we did, thought I mentioned that
16:50.35nny_2tzafrir_laptop: my communication skills lack today
16:50.53nny_2ok so fine it spams the error, was just a question, here is the other error i am not clear on
16:51.09*** join/#asterisk growltiger (n=growltig@ip70-179-54-235.sd.sd.cox.net)
16:51.09nny_2[Sep 24 14:42:14] WARNING[3222] chan_dahdi.c: Unable to disable echo cancellation on channel 2: Invalid argument
16:51.30Kattyinvalid argument
16:51.31nny_2this is immediately following a Red alarm on the channel, so I assume this is normal?
16:51.40Kattywhat me and [TK]D-Fender have everyday
16:51.45nny_2hahaha
16:51.59[TK]D-Fenderruns in circles, arms flailing
16:52.18Kattywhatever he's having...i'll take two.
16:52.50nny_2I have asked the telco for the Adtran alarms from the event as well
16:52.56nny_2we'll see how that goes :\
16:53.20*** join/#asterisk oej (n=olle@63-255-103-7.ip.mcleodusa.net)
16:53.31*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
16:55.05*** join/#asterisk pikachu2000 (n=pikachu2@196-209-182-112-rndf-esr-5.dynamic.isadsl.co.za)
16:55.26*** part/#asterisk iamturnip (n=joe@shop.freegeekvancouver.org)
16:56.00Kattyso randomly
16:56.03leif[astricon]wacky waving inflatable arm flailing tube man!
16:56.07Kattyabout baby bottles
16:56.20Kattyyou can get them anywhere right?
16:57.47ZeeekKatty:  not too late to join us!
16:57.58*** join/#asterisk denon (i=denon@synapse.subneural.net)
16:57.58*** mode/#asterisk [+o denon] by ChanServ
16:58.00Zeeek123@ts.x2z.eu
16:58.08*** join/#asterisk denon (i=denon@synapse.subneural.net)
16:58.08*** mode/#asterisk [+o denon] by ChanServ
16:58.23Zeeekor talkshoe@vuc.onsip.com
16:58.24KattyZeeek: very persistant
16:58.26KattyZeeek: i like that
16:58.31Zeeekor whatever that resolves to
16:58.52ZeeekKatty we love ya and we're talkiing with JerJet and a bunch of great folks
16:59.00Kattyhow about baby bottles?
16:59.01Zeeekand I can't type
16:59.10ZeeekGit over here woman!
16:59.23ZeeekI'm talkin' heah
16:59.34ZeeekSIP trunking
16:59.36*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
16:59.40ZeeekSkype/Asterisk
17:00.17Kattyport?
17:00.48Zeeek5060
17:00.57Qwell~trunk
17:01.01jbotmethinks trunk is a word with varying definitions.  In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN.  There is no such thing as a "SIP Trunk" -- Don't use the term.
17:01.12QwellZeeek: ^^ :p
17:01.40*** part/#asterisk fudpucker1 (n=here@h-67-103-23-130.phlapafg.covad.net)
17:02.59*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-5ffc093fa80dcd2b)
17:02.59*** mode/#asterisk [+o putnopvut] by ChanServ
17:03.05tzafrir_laptopQwell, I guess we should not use "branch" either
17:03.06jjshoebut most of all Qwell is my hero
17:03.22Qwelljjshoe: haha, Tyler was talking about a script that Samy had written
17:03.33Qwelldo you have ANY idea how hard it was for me to not shout "Samy is my hero!"
17:04.09Kattythat's just...
17:04.17Kattyone of the ferrets are named Sammy
17:04.59jayteewasn't one of the muskrats in "Muskrat Love" named Sammy too?
17:05.03jjshoeQwell you should have :P
17:05.11jjshoeQwell what did kerry emaiul you?
17:05.40Qwelljjshoe: meh, it's stupid
17:05.42Qwell"If you need help, Tim can help you"
17:05.55jjshoelol, on the asterisk now interface?
17:05.59Qwellyeah..
17:06.01Qwelllike wtf
17:06.02jjshoelol
17:06.05Qwellhow does that benefit him/you?
17:06.09jjshoeno idea
17:06.15jjshoeI think he's trying to schmooze
17:06.16Qwellso, I'm staying far away from that
17:06.18Qwellindeed
17:06.47jjshoeyou guys are just integrating freepbx? not writing your own frontend?
17:06.59Qwellwe have our own already.  it'll have both
17:07.14jjshoeah
17:07.23jjshoewe tried to install asteirks now here once
17:07.31jjshoeit couldn't even detect the proper resolution for our monitor
17:07.34jjshoeand we could never click install
17:07.40jjshoe*shrug*
17:07.44Qwell*shrug* this isn't a custom anaconda
17:07.49Qwellthat probably was.  I don't know
17:08.00jjshoei believe it was
17:08.11Qwellwell, I know it was.  I don't know what was added/removed though
17:08.15*** join/#asterisk lanning (n=lanning@66.151.128.195)
17:08.20Qwellmaybe video drivers - who knows
17:08.39*** join/#asterisk MindTheGap_ (n=MindTheG@189.59.204.61)
17:08.59jjshoeI've been contemplating writing just a front end in my spare time
17:09.06jjshoethat does nothing but post/get json
17:09.09*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
17:09.13jjshoeand leave it up to anyone else to implement a backend
17:09.23Qwelljjshoe: asterisk-gui
17:09.50MindTheGap_hello, im trying to compile asterisk, it compiles just fine but wont compile chan_zap.so. i do have zaptel installed... i ve done it a milion times but now its not compiling it...
17:10.07QwellMindTheGap: 1.4.22-rc?
17:10.42*** join/#asterisk Assimilate (n=Assimila@216.83.78.108)
17:10.47tzafrir_laptopMindTheGap_, fails to build? does not build? What version of asterisk?
17:11.18MindTheGap_it just doesnot build the module, everything else is ok
17:11.27QwellMindTheGap: what version of Asterisk?
17:11.33MindTheGap_its like it cannot see zaptel installed
17:11.40MindTheGap_1.6.0 rc5
17:11.42jjshoeQwell is that whats going on with freepbx?
17:11.46MindTheGap_rc5 sorry
17:11.46QwellMindTheGap: You need dahdi
17:11.50Qwelljjshoe: huh?
17:12.01tzafrir_laptopit's chan_dahdi, and it needs dahdi-tools and dahdi-linux, indeed
17:12.06*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
17:12.06*** mode/#asterisk [+o lmadsen] by ChanServ
17:12.17MindTheGap_damn, it worket w 1.0.6 beta9
17:12.23MindTheGap_1.6.0beta9
17:12.24Qwellbeta9 is very ver different
17:12.31Qwell(it was a beta..)
17:12.46MindTheGap_does the chanel naming conventions stay?
17:13.01MindTheGap_like can i still use dial/zap/000
17:13.02MindTheGap_?
17:13.24*** join/#asterisk drumkilla (n=russell@asterisk/digium-open-source-team-lead/russellb)
17:13.24*** mode/#asterisk [+o drumkilla] by ChanServ
17:13.34jjshoeQwell on asterisk-now
17:13.57Qwelljjshoe: I don't get the question
17:14.30*** join/#asterisk ManxPower (n=manxpowe@125.sub-75-249-4.myvzw.com)
17:14.46*** join/#asterisk russellb_ (n=russell@asterisk/digium-open-source-team-lead/russellb)
17:14.46*** mode/#asterisk [+o russellb_] by ChanServ
17:15.11jjshoeQwell is this being packaged with asterisk-now?
17:15.12ManxPower~book
17:15.13jbotwell, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
17:15.16Qwellyeah
17:15.19ManxPowerYAY!  jbot is alive!
17:15.35ManxPowerjjshoe: bkruse is the one packaged with AsteriskNOW
17:15.46ManxPower8-)
17:15.49Qwelldownload AsteriskNOW, get a free bkruse?
17:15.53ManxPoweryup!
17:16.10jeevhmm
17:16.14MindTheGap_there is no dahdi in downloads.digium.com
17:16.20jeevi'm gonna try to convince bkruse to change asteriskNOW to JEEVnow
17:16.24QwellMindTheGap: Did you read the README file?
17:16.43ManxPower~zeeek
17:16.43jbotsomebody said zeeek was someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff."
17:16.51jeevlol
17:17.14MindTheGap_Qwell this? http://downloads.digium.com/pub/zaptel/README-1.4.12.1
17:17.16MindTheGap_yes
17:17.20Qwellhttp://downloads.digium.com/pub/README.txt
17:17.22MindTheGap_no dahdi anywhere
17:17.32*** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej)
17:17.53MindTheGap_thanks
17:18.16jeevManxPower, want to take over the world with me ?
17:18.41ManxPowerMindTheGap I see it right there!
17:18.46jjshoeManxPower or someone too important with other things :D
17:18.53jjshoebusy with other things
17:18.54jjshoe*sigh*
17:18.56jjshoeit's friday.
17:19.06BlackvelMacro arguments can only be ${ARG1}, ${ARG2}? No way to give the parameter a name?
17:19.08jjshoeI need a job where I do nothing but work with asterisk
17:19.09jjshoe...
17:19.16Qwelljjshoe: yeah, you should quit
17:19.33jjshoeQwell lol
17:19.37Zeeekoh Katty
17:19.43jjshoeQwell are you going to pay me $75k a year and my moving expenses? :D
17:20.16QwellI'll pay you $75/year
17:20.20*** part/#asterisk Damin (n=damin@nucleus.nacs.net)
17:20.28jjshoeQwell hmm. what do I get to do for the $75. ?
17:20.32jjshoeQwell do we get to make out?
17:20.40citywokManxPower, i ended up using meetme instead of zapbarge to listen
17:21.02Qwelljjshoe: you want to do that AND get paid?
17:21.04Qwellas if
17:21.15jjshoeI know, I was stretching
17:22.45[TK]D-FenderBlackvel: Feel free to do "Set" for them at the start of your macro.  Technically the answer is "no"
17:23.00jjshoealthough the $75 might cover the moving expenses
17:24.42jjshoeoh, and don't forget, I'll need my week off in winter for jamaica :)
17:26.17*** join/#asterisk growltiger_ (n=growltig@ip70-179-54-235.sd.sd.cox.net)
17:30.33*** join/#asterisk Cresl1n (n=matt@asterisk/libpri-and-libss7-expert/Cresl1n)
17:30.33*** mode/#asterisk [+o Cresl1n] by ChanServ
17:33.18BlackvelI am playing around with this Set(DB..) stuff like DBPut
17:33.46Blackvelwhat must I do do be able to set the asterisk db parameter when calling 401-405 from snom phone?
17:34.01Blackveljust Set or surrounding Answer,Set,Hangup?
17:41.08Zeeekby golly
17:41.12Zeeekg'bye all
17:41.38Zeeekhttp://voipUsersConference.org for a recorded version
17:41.43*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
17:42.06bpgoldsbI see about 5 Python AGI libraries on Voip-info, anyone used one that they can recommend?
17:46.12*** join/#asterisk mapmagic (n=piyush@chan.cs.fsu.edu)
17:46.17[TK]D-FenderBlackvel: What "parameter"?  AstDB is just a dumb flat non-searchable value list.  Its boring dialplan like any other.
17:46.22mapmagicDoes this work with asterisk: http://www.buy.com/retail/product.asp?sku=202505473
17:48.33jameswfmapmagic: buy it and let us know
17:49.15mapmagiclol, if it doesnt work, i dont have any other use for it.
17:49.37jameswfhow many voip channels do you have
17:50.06mapmagicI just want to connect my computer to my telephone line using asterisk and vmware. I don't know if this can be done using this : http://www.buy.com/retail/product.asp?sku=202505473
17:50.18jameswfg729 has been gold for me
17:50.39*** join/#asterisk propellerhead (n=yogurt2u@host113.190-136-116.telecom.net.ar)
17:51.33StephenFmapmagic: Tthat only does QOS for voip. You need an ATA (Analog Terminal Adapter) if you want to use your existing analog phones with asterisk
17:51.50mapmagicwhat is a recommended ata StephenF?
17:52.03StephenFmapmagic: look for a Linksys PAP2, or PAP2T
17:52.19[TK]D-Fendermapmagic: No.
17:52.48StephenFmapmagic: your talking about setting up your current phone right? not just your computer
17:52.54[TK]D-Fendermapmagic: THIS is what you'll want :: http://www.telephonydepot.com/product_p/105-054-312.htm
17:53.35mapmagicI think there are cheaper places to buy that. $67!
17:53.54mapmagicStephenF: Yes, I want to connect my phone and my computer to a phone line that comes in my house.
17:54.15jameswfg729 $10/chan
17:54.26StephenFmapmagic: oh sorry, yes if you are keeping your existing phone line you want what [TK]D-Fender suggested
17:54.29mapmagicPAP2T = $37. PAP2 = $28. What is g279?
17:54.49StephenFmapmagic: where did you find PAP2T for $37?
17:55.01[TK]D-Fendermapmagic: NONE of those other devices will let you take in your phone **LINE**
17:55.03*** join/#asterisk krokodilerian (n=vasil@63-255-123-238.ip.mcleodusa.net)
17:55.23mapmagicso i need a PAP2T I guess then, right?
17:55.29[TK]D-Fendermapmagic: Those ones will let you plug in regular PHONES and use them as SIP phones.
17:55.41StephenFmapmagic: no, you need http://www.telephonydepot.com/product_p/105-054-312.htm
17:55.44mapmagicI want to keep my phone working as well.
17:55.46StephenFlike [TK]D-Fender said
17:55.58[TK]D-Fendermapmagic: then the device i listed is what you need.
17:56.09[TK]D-Fendermapmagic: it does both.
17:56.37mapmagicso that is not much cheaper i guess...$68 even on newegg!
17:57.00mapmagicStephenF: http://www.compuplus.com/i-Linksys-PAP2T-NA-2-Port-VoiP-Phone-Adapter-Feature-Rich-VoIP-Service-through-your-High-Speed-Internet-Connection-1009628~.html?sid=39u06204h9c1y60
17:57.13[TK]D-Fendermapmagic: Compared to the rest of the telephony world, that IS incredibly cheap.
17:57.16[TK]D-Fender~cheap
17:57.19jbotfrom memory, cheap is a bad idea.  If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.
17:57.19StephenFmapmagic: ohh ok, thanks
17:57.20[TK]D-Fender^^^^^
17:57.52mapmagicSPA3102 is the best i can do i guess, right?
17:58.01[TK]D-Fendermapmagic: Yup
18:00.30*** join/#asterisk feeds (n=feeds@85-135-232-238.adsl.slovanet.sk)
18:01.43infinity1where can i get some free music on hold
18:02.12feedscan anyone help me with running asterisk under Fedora 9??
18:02.23*** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net)
18:02.36[TK]D-Fenderfeeds: Show us a specific probelm and maybe we can help out
18:02.51[TK]D-Fenderinfinity1: there are links on the WIKI page for this.  Go check 'em out.
18:03.00feedswhen i run asterisk -c as root this shows up
18:03.04feeds[root@localhost ~]# asterisk -c
18:03.04feedsAsterisk 1.6.0-beta9, Copyright (C) 1999 - 2008 Digium, Inc. and others.
18:03.04feedsCreated by Mark Spencer <markster@digium.com>
18:03.04feedsAsterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
18:03.04feedsThis is free software, with components licensed under the GNU General Public
18:03.04feedsLicense version 2 and other licenses; you are welcome to redistribute it under
18:03.06feedscertain conditions. Type 'core show license' for details.
18:03.08feeds=========================================================================
18:03.10feeds[ Booting...
18:03.12feeds[ Reading Master Configuration ]
18:03.14feeds[ Initializing Custom Configuration Options ]
18:03.15StephenFfeeds: stop
18:03.16*** kick/#asterisk [feeds!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by lmadsen (flooding)
18:03.20StephenFtoo late
18:03.32*** join/#asterisk feeds (n=feeds@85-135-232-238.adsl.slovanet.sk)
18:03.34feeds[Sep 26 20:01:58] WARNING[2926]: res_config_pgsql.c:888 parse_config: PostgreSQL RealTime: Couldn't establish connection. Check debug.
18:03.37feeds[Sep 26 20:01:58] NOTICE[2926]: config.c:1905 ast_config_engine_register: Registered Config Engine pgsql
18:03.39feeds.[Sep 26 20:01:58] NOTICE[2926]: res_odbc.c:794 load_module: res_odbc loaded.
18:03.39putnopvutlawl
18:03.41feeds.[Sep 26 20:01:58] WARNING[2926]: res_config_ldap.c:1401 parse_config: No directory user found, anonymous binding as default.
18:03.42*** kick/#asterisk [feeds!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by lmadsen (stop flooding)
18:03.42StephenFfeeds: stop
18:03.44StephenFlol
18:03.49lmadsenStephenF: he can't once he's done the paste
18:03.53StephenFohh
18:03.55StephenFright
18:04.36infinity1[TK]D-Fender: voip-info wiki?
18:04.45[TK]D-Fenderinfinity1: Yes
18:04.49*** join/#asterisk feeds (n=feeds@85-135-232-238.adsl.slovanet.sk)
18:04.54tzafrir_laptop~pb
18:04.55jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
18:04.59[TK]D-Fender^^^^^^^^^^
18:05.20feeds~pb help
18:05.33denonwhat more help do you want?
18:05.34feeds~pb can paste it here?
18:05.40denonno, it gives you a url
18:05.43lmadsenfeeds: pastebin gives you a link to post
18:05.45denongo to that website
18:05.54*** join/#asterisk Specialist1 (n=me@119.160.105.172)
18:05.57Specialist1hi everyone
18:06.45Katty[TK]D-Fender: fazoli's!
18:06.49Katty[TK]D-Fender: spaghetti baked pizza!
18:06.52Katty[TK]D-Fender: this is so bad for me.
18:06.57tzafrir_laptopfeeds, alternatively, there's the channel #flood in which you can paste long pastes. But nobody really uses it nowadays
18:06.58*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
18:07.16[TK]D-FenderKatty: 2 minutes in your mouth, 2 hours in your stomach, 2 years on your butt ;)
18:07.23feedswhen i run asterisk -c in fedora 9 this shows up:http://pastebin.com/m731c03e3
18:07.41feedscan anyone help me?
18:07.42Katty[TK]D-Fender: and i'm going to enjoy every bite.
18:07.55[TK]D-FenderKatty: With a side-order of... GUILT!!!
18:08.03Katty[TK]D-Fender: no, not really
18:08.11Kattyki, maybe a little
18:08.22Kattyjust a little tho
18:08.32[TK]D-FenderKatty: You will soon... this is an all-inclusive prepaid guilt-trip I'm sending you on!
18:08.42Specialist1anybody using a2bililng here
18:08.42Kattyoh boy!
18:08.45Kattycan i take my pup with me?
18:08.58feedscan anyone help me?
18:08.59tzafrir_laptopfeeds, built asterisk youself? from apckage?
18:09.03feedsno
18:09.09tzafrir_laptopfeeds, built asterisk youself? from apckage?
18:09.13feedsi installed it from yum
18:09.14[TK]D-FenderKatty: Misery can bring along company so long as it meets carry-on sizing requirements.
18:09.21tzafrir_laptophmmm....
18:09.29Katty[TK]D-Fender: you're just too much sometimes, deary
18:09.32Katty[TK]D-Fender: that's why i love you
18:09.48tzafrir_laptopno previous source installs there?
18:09.53feedsno
18:10.01[TK]D-FenderKatty: If only you could comprehend how much I hold back for you all ;)
18:10.12Katty^_^
18:10.18Kattypepperoni slice?
18:10.31tzafrir_laptopwhat do you see in the logs? /var/log/asterisk/{messages,full} ? which of those exist?
18:10.45feedsill look
18:11.07Kattysomeday, i'm going to backup the ole messages log, then make a new one that says Doom
18:11.15Kattyright before one of the other techs goes to look at it
18:11.28Kattya few hundred lines of Doom
18:12.02*** join/#asterisk albertoandrade (n=alberto@200.195.161.164)
18:13.43Katty[TK]D-Fender: did i show you the spike ulta dinosaur puppy toy?
18:13.54Katty[TK]D-Fender: also suitable for children ages 3 and up
18:14.23feedstzafrir_laptop: http://pastebin.com/m386ceda2
18:14.38[TK]D-FenderKatty: I think so...
18:15.51Kattyit's just swell.
18:16.53*** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer)
18:17.34JerJerugh - how does one install apache in using conary / rpath ?
18:17.50JerJeri think this is a like hacked up BE install - not sure
18:18.24*** join/#asterisk logicwrath (n=no@c-68-42-253-39.hsd1.mi.comcast.net)
18:19.04feedstzafrir_laptop: are you here?
18:19.16tzafrir_laptopyes
18:19.25tzafrir_laptopHow do you start asterisk?
18:19.32Kattydoes a remote controlled todler dino seem like an appropriate gift for a grown up geek?
18:19.32feedsas root
18:19.35feedswith
18:19.36*** join/#asterisk kand (n=kanderso@139.148.188.72.cfl.res.rr.com)
18:19.42feedsaserisk -c
18:19.50kandHello, can somebody explain the difference between P-Asserted-Identity, Remote-Party-ID, and the from headers?
18:19.50Kattyhttp://www.fisher-price.com/img/product_shots/M8126_d_1.jpg <- said dino
18:20.32tzafrir_laptopfeeds, I'd try strace . It's basically a matter of some module failing. So I wonder what was the last module loaded
18:20.48feedsstrace??
18:20.50[TK]D-FenderKatty: Meanie... Pup hasn't even arrived yet and you're already preparing your campaign of terror..
18:21.12Kattyoh yes.
18:21.13Kattynods
18:21.18Kattyi'm getting a very early start.
18:21.37Katty[TK]D-Fender: but! is this suitable for an all-grown-up geek?
18:22.10[TK]D-FenderKatty: Depends on the geek.  You already know the answer, you just don't want to acknowledge it.
18:22.48Kattymaybe a nerf gun would be more appropriate to chase co-workers around with at the office.
18:23.34krokodilerianKatty , why not stick to the classics - an axe ?
18:24.18Kattydon't be silly.
18:24.21Kattyi don't want to kill, i just want to torture
18:24.32feedstzafrir: what is strace?
18:24.37Kattya marshmallow blaster!!!
18:24.40Kattyspock would be proud.
18:25.03tzafrir_laptopyum install strace
18:25.08feedsaha
18:25.22tzafrir_laptopstrace -o trace_file command line to run
18:25.32*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
18:28.37krokodileriantzafrir_laptop , "-s 4096 -f" for the strace
18:28.40krokodilerianit helps a lot
18:29.05Blackvel[TK]D-Fender: is it enough to connect 401 to exten => 401,1,Set(DB(....)? must there be any Answer or Hangup?
18:29.25[TK]D-FenderBlackvel: Doesn't have to be anything.
18:30.43x86oh my god...
18:30.44x86http://www.reuters.com/article/companyNewsAndPR/idUSPEK16693720080925?rpc=64
18:30.50x86this is very, very bad...
18:31.33tzafrir_laptopKatty, be careful it won't backfire on you:
18:31.36tzafrir_laptophttp://thedailywtf.com/Articles/Lyle-Can-Do-Anything-Better-Than-You.aspx
18:32.05feedstzafriri what should i do when ive got it installed
18:32.06feeds?
18:34.01tzafrir_laptopstrace -o trace_file command line to run
18:34.57feedsstrace: asterisk: command not found
18:35.03tzafrir_laptopfeeds, hint: writing the full nick of a name triggers an alarm of some sort with most IRC clients
18:35.29*** join/#asterisk oej_ (n=olle@63-255-123-242.ip.mcleodusa.net)
18:35.30tzafrir_laptopmost IRC cleints complete nick names with the TAB key
18:36.06nny_2so awesome, makes me wish i had a shred of musical talent http://www.youtube.com/watch?v=Bkp9bDGDd1w
18:36.54scooby2where is the proper place to put autofill?
18:37.07feedstzafrir_laptop, http://pastebin.com/m157036e7
18:37.25nny_2I am looking at email notifications from syslogd for specific events (like red alarm on a t1). Is syslogd the best way to approach this or does logger in asterisk have any magic?
18:37.48bpgoldsbCan anyone recommend a book for learning AEL2?
18:38.46[TK]D-Fendernny_2: He's playing on an M-Audio Axiom 61 :)
18:39.33[TK]D-Fendernny_2: And as all the backing is crappy MIDI, I wouldn't feel jealous of that vid ;)
18:39.58nny_2[TK]D-Fender: yeah i think the Moog theremin is pretty cool though.. the way the video was produced is pretty neat
18:40.09tzafrir_laptopfeeds, write(2, "asterisk: invalid option -- \'o\'\n", 32asterisk: invalid option -- 'o'
18:40.09tzafrir_laptop) = 32
18:40.14[TK]D-Fendernny_2: Yes, that is nifty...
18:40.27tzafrir_laptopstrace -o trace_file asterisk -c
18:40.55nny_2the lyrics on his helmet thing are post production video effects.. did a pretty convincing job with it
18:41.06nny_2he has a gnarles barkley cover of crazy that's pretty cool as well
18:41.15nny_2cover of gnarles*
18:41.19nny_2heh parse error
18:41.39*** join/#asterisk tkbeat (n=tk@p54B96213.dip.t-dialin.net)
18:42.03Kattyanyone in here work for isymphony?
18:42.09*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
18:42.38feedstzafrir_laptop, http://pastebin.com/m5a8973a8
18:44.04tzafrir_laptopcan you pase trace_file?
18:44.15feedspase?
18:44.23feedsaha paste
18:44.26tzafrir_laptopeither paste it, or upload it directly through http://paste.debian.net
18:44.44*** join/#asterisk hfb (n=hfb@pool-96-247-117-68.lsanca.dsl-w.verizon.net)
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18:46.20feedstzafrir_laptop,  http://paste.debian.net/18021/
18:47.30BlackvelExecuting PrivacyManager("SIP/patton_ntba-098b1b28", "3|7") in new stack
18:47.30Blackvel<PROTECTED>
18:47.47Blackvelis that the case when CALLERID(number) was set to anonymous?
18:48.52*** join/#asterisk v4mp (n=Gary@82.118.111.250)
18:49.07tzafrir_laptopfeeds, but you ran it with an incorrect asteirsk command-line
18:49.09[TK]D-FenderBlackvel: I don't see you looking at your actual CID anywhere...
18:49.11tzafrir_laptopstrace -o trace_file asterisk -c
18:49.16tzafrir_laptopnot:
18:49.18tzafrir_laptopstrace -o trace_file asterisk -o
18:49.40*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
18:50.36feedstzafrir_laptop,  http://paste.debian.net/18023/
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18:53.50feedstzafrir_laptop,  http://paste.debian.net/18023/
18:54.59tzafrir_laptopthat's it? What about the trace_file?
18:56.49feedsthis is in the trace file: http://paste.debian.net/18025/
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18:57.58v4mphey guys i got problems with asterisk i cnt get it setup so a user can login i've added the user to users.conf nor can i get the http working for it to try edit it that way if it makes it any easier to configure like that
18:59.00*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
19:01.23*** join/#asterisk zionvier (n=pete@c-67-166-22-106.hsd1.co.comcast.net)
19:04.21*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
19:05.47zionviercan anyone tell me what ports I need to route to my * server to connect a sip client to it from outside the network?
19:06.58[TK]D-Fender~sipnat
19:06.59jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
19:07.01*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
19:07.01[TK]D-Fenderzionvier: ^^^
19:07.11feedstzafrir_laptop, this is in the trace file: http://paste.debian.net/18025/
19:07.39zionvierthanks
19:07.49v4mp[Sep 26 20:07:02] WARNING[29531]: chan_zap.c:890 zt_open: Unable to open '/dev/zap/channel': No such file or director << how would i fix
19:08.28tzafrir_laptopfeeds, so this is after loading foramt_mp3.so
19:08.40feeds?wgat?
19:08.44feeds?what?
19:09.25tzafrir_laptopwhat are the outputs of:  rpm -qf /usr/lib/asterisk/modules/format_mp3.so; ldd /usr/lib/asterisk/modules/format_mp3.so
19:09.57*** join/#asterisk Qapf (n=qapf@173.8.140.60)
19:10.13tzafrir_laptopv4mp, lsmod | grep zaptel # and do you have zaptel built?
19:10.21tzafrir_laptopwhat system is that?
19:10.44feedstzafrir_laptop, http://pastebin.com/m72e18b07
19:10.45v4mpsystem as in what ?
19:11.02*** part/#asterisk mapmagic (n=piyush@chan.cs.fsu.edu)
19:11.10v4mpcentos
19:11.17v4mpno commad lsmod
19:12.01Qapfhey guys, i have a customer who is concerned with inbound call reliability and wants to buy a service where someone calls in to their phone system once every X minutes and checks to ensure calls actually terminate and so on. anyone know of some software to do this, or should i tell em to just pick up a phone :)
19:12.11Nuggetv4mp: you probably don't have /sbin and /usr/sbin in your path, but this really isn't the right place to learn how to use your operating system.
19:12.43tzafrir_laptopinteresting. Is there a Fedora resident in the crowd?
19:12.48feedsme
19:13.01v4mpNugget, should be other things work
19:13.17Nuggetwell then it works!  ;)
19:13.31v4mp/dev/zap wasn't created on build either but the install gave no errors
19:13.48Nuggetthat's normal.
19:13.57tzafrir_laptop<PROTECTED>
19:14.11tzafrir_laptop<PROTECTED>
19:14.26tzafrir_laptop<PROTECTED>
19:15.37*** join/#asterisk dwclarkNU (n=dwclarkN@64.8.26.200)
19:15.54dwclarkNUmy phones are becoming "UNREACHABLE" and then "OK" constantly, they constnalty are going Retransmitting #1, #2, #3, #4 (NAT) with the OPTIONS message...any help?
19:15.57v4mp1.4.11
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19:18.24feedstzafrir_laptop, i gotta go, will you be here tomorrow??
19:18.34tzafrir_laptopmaybe
19:18.44feedstzafrir_laptop,  ok
19:18.47tzafrir_laptopthere are a bunch of fedora folks here
19:18.53feedsbye All
19:18.57tzafrir_laptopI suggest you file a bug for that package
19:19.02tzafrir_laptopin their bugzilla
19:19.05feedsok
19:19.24*** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk)
19:19.37maxximhi. anybody have experience in interconnecting h323 and sip ?
19:20.12*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
19:21.43*** join/#asterisk Talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-af76483349ec4065)
19:22.52*** join/#asterisk nooky (n=nooky@customer191-9.iplannetworks.net)
19:23.03Talirk81When  asterisk recieves a call , and then asterisk dial a new number , bridging the calls. Will Monitor record back  the  Recived and the dialed  sound streams , im thinkign it wont since they are seperate channels right?
19:23.46[TK]D-FenderTalirk81: "core show application monitor" , "core show application mixmonitor"
19:23.59Talirk81im reading the explaination
19:24.08Talirk81but i conused on "channel"
19:24.10nookycan i strip the agent number from the channel in queue_log file? is that right?
19:24.29Talirk81does asterisk count  each as seperate channels or the whole call as a channel
19:24.36v4mpso what am i to do ?
19:26.36*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
19:27.00[TK]D-FenderTalirk81: A "call" could be considered as 2 bridged channels.
19:28.06Talirk81ok so each side the  incoming call and the dials out call would each be seperate channels and thus  2 monitor commands would be needed and i would need to then merge the files later.
19:28.39Kattywibbles
19:28.47Talirk81Since Monitor() says is records the current "channel" ?
19:29.06lgjMixMonitor records both sides and mixes them into one file
19:29.09[TK]D-FenderTalirk81: No, monitor records the brdged channels.  Both sides.  read each of their instructions.
19:29.18*** join/#asterisk jasonwoot (n=jasonrot@69.73.89.233)
19:29.21jasonwootwoot
19:29.31[TK]D-Fenderredundant
19:29.33lgjMonitor as he says does see both sides but does sepreate files you have to mix together
19:29.43jasonwootretarded
19:29.46jasonwoot7557Fall*
19:30.00jasonwootany queuemetrics users present?
19:30.43bpgoldsbI'm calling 'Background(main-menu);', it plays the sound correctly, but any digits pressed during the time report 'DTMF begin ignored 'X''
19:30.43Talirk81lgj, i think the  mb option will merge them at the end if i under stand fender and the   Monitor() description in the Oriely book correctly.
19:30.50bpgoldsbAnyone know why that is?
19:31.24lgjTalirk81, prob so I don't have the docs in front of me I use MixMonitor myself so I just have one file :)
19:31.32*** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com)
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19:33.10[TK]D-Fenderbpgoldsb: pastebin your dialplan context.
19:33.35bpgoldsbhttp://pastebin.com/m42f6f0a1
19:33.46lgjbpgoldsb, I bet you not wating for digit/extention
19:34.55[TK]D-Fenderbpgoldsb: Waht version?
19:34.59bpgoldsb1.6-rc6
19:35.47[TK]D-Fenderbpgoldsb: Try setting "rfc2833compensate=yes" for your peer
19:35.47StephenFis that the new syntax for 1.6? or are you using a different language for your diaplan?
19:36.00[TK]D-FenderStephenF: AEL2
19:36.02bpgoldsbAEL2
19:36.05StephenFoh ok
19:36.18bpgoldsb[TK]D-Fender: For that sip peer alone?
19:36.29[TK]D-Fenderbpgoldsb: Yes
19:36.48[TK]D-Fenderbpgoldsb: Test, and upon success scale as needed.
19:37.04bpgoldsb[TK]D-Fender: Thanks, worked great.
19:37.11[TK]D-Fenderbpgoldsb: Glad to hear
19:38.18bpgoldsb[TK]D-Fender: Any downside to enabling this globally?
19:39.26bpgoldsbFrom a quick googling, it says it's caused by using Asterisk 1.2 <-> 1.4 communication
19:39.29bpgoldsbWhich I'm not doing
19:39.49bpgoldsbI am using a custom written Sip Softphone our company made
19:41.11*** join/#asterisk ScarEye (n=scareye@12.27.87.66)
19:43.54Kattykm-: ping?
19:45.36v4mpanyone got a clue how to solve the channel problem then ?
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19:47.13*** join/#asterisk Juxt (n=denis@adsl-233-30-165.mia.bellsouth.net)
19:47.42Juxtwho is a realible did provider out there? i just went thru practially the whole voip-info list and they all look amateurish
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19:54.41maxximhi. anybody have experience in interconnecting h323 and sip ?
19:54.52jeevif i want to build out a network in europe for a massive amount of text/sms.. what equipment do i need?
19:56.31tzangerjeev: the first thing you need is someone technical, because questions like that indicate that you have an idea, and while it may be a good one, you aren't a technical person, or at least not one technical enough.
19:57.15jeevare you trying to apply? :D
19:57.25*** join/#asterisk oej_ (n=olle@63-255-123-231.ip.mcleodusa.net)
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20:00.18v4mpam i just going to be ignored ?
20:00.31c4t3ll8tr guys!!
20:01.25km-katty: pong.
20:01.35km-katty: fighting with nusoap in php.  It's winning.
20:02.28*** join/#asterisk thehar (i=thehar@thehar.xmission.com)
20:02.51*** join/#asterisk [netman] (n=netman@152.Red-88-19-164.staticIP.rima-tde.net)
20:03.02jeevv4mp
20:03.05jeevjust keep asking man
20:04.01Kattykm-: :<
20:04.08Kattykm-: i already forgot what i was going to ask you.
20:04.11Kattykm-: something about babies.
20:04.13Kattykm-: OH!
20:04.18Kattykm-: you have a kid, right?
20:04.37km-yep, 3 months old
20:04.48Kattykm-: so you know those teething rings they put into the freezer
20:04.54Kattykm-: that the babies gnaw on
20:05.05km-yeppers, we have a bunch.
20:05.10Kattykm-: how durable are those?
20:05.30km-you thinking of using them for Riddick?
20:05.36Kattyyeah
20:05.48*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
20:05.48*** mode/#asterisk [+o russellb] by ChanServ
20:05.52km-they're fairly durable for babies, but you're talking about an animal with good jaw strength and sharp canines
20:06.01Kattyyou have a point there
20:06.08Kattyeventually riddick will get his canines
20:06.21Kattymaybe a wet wash cloth in the freezer will be better to gnaw on
20:17.00km-katty: that's much safer, yeah.
20:20.55*** join/#asterisk SwK (n=SwK@freeswitch/developer/swk)
20:22.54*** part/#asterisk leif[astricon] (n=Leif@63-255-123-231.ip.mcleodusa.net)
20:24.15[TK]D-Fenderok, checkout time, later all
20:24.43*** join/#asterisk bijit (n=benji@200.122.158.243)
20:24.48*** join/#asterisk Unlockgod (n=benjamin@88-108-114-59.dynamic.dsl.as9105.com)
20:24.50UnlockgodHey guys
20:25.04Unlockgodwonding if someone could help me isntall askerisk on a small server here
20:25.30Qwellhelp you?
20:25.44*** join/#asterisk sacitec (n=tobi@201.144.211.82)
20:25.58bijitI am am using application Pickup() but sometimes its works and sometimes it doesn't anyone have any comments I should try?
20:26.09Unlockgodim trying to install iv formatted it back to centos 4 times
20:26.22bijitI am using exten=>_**96.,n,Pickup(${EXTEN:4}
20:26.24Unlockgodand i cant seem to get right dependency files
20:26.54*** join/#asterisk wacky_ (n=abourget@mtl.savoirfairelinux.net)
20:27.27wacky_I have some faulty "core show translation" values.. some are "1" and some are "10000"
20:27.38Qwellwacky_: They aren't faulty.
20:28.03wacky_could that cause the horrible voice messages from the lady saying "Comedian mail" ?
20:28.37wacky_Qwell: they change tremendouly each time I run "core show translation recalc 10"
20:28.45wacky_going from 5000 to 2
20:28.51Qwellwacky_: that is usec
20:29.10Qwellit's correct, if you understand how context switches works
20:29.26wacky_it's going crazy :)
20:29.30wacky_how do they work ?
20:30.02QwellI can't explain it without explaining many other things.
20:30.05QwellJust trust me - it's correct.
20:31.13v4mpanyone got a clue to the /dev/zap/channel not existing and how i would fix ?
20:31.23Qwellv4mp: Did you install Zaptel?
20:31.59v4mpyes
20:32.18v4mpi followed a guide that had me install that first incase and problems/errors of which there was none
20:32.23bijitI am using exten=>_**96.,n,Pickup(${EXTEN:4} is this correct since I have more than two context?
20:33.45wacky_Qwell: which contexts are you talking about ?  not the dialplan contexts ?
20:34.01Qwellkernel
20:35.02wacky_Qwell: Can you confirm that this weird behavior is correct: http://www.pastebin.ca/1211802
20:35.23infinity1does polycom have a contact directory other than xml files?
20:35.23QwellI already said it was correct.
20:35.37*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
20:35.53v4mpany ideas Qwell ?
20:35.58*** join/#asterisk krokodilerian (n=vasil@63-255-123-238.ip.mcleodusa.net)
20:36.26*** join/#asterisk growltiger (n=growltig@ip70-179-54-235.sd.sd.cox.net)
20:36.44wacky_the thing is I'm getting the most crappy sound out of system prompts, and I was wondering if it's "costs" for translations didn't find a shorter way to go through 4 codecs that cost at 4 microseconds instead of a direct translation that costs 10000
20:36.52bkrusejeev: Done.
20:37.00jeevwoo hoo!
20:37.04jeevam i getting royalties ?
20:37.09Qwellwacky_: Do you understand how small 10000 is?
20:37.10bkruseof course
20:37.12jeevbut any % of 0 is 0..
20:37.14bkruse$2.00 per download
20:37.17jeev:>
20:37.20jeevdo you have my paypal?
20:37.29wacky_it's 10 mili seconds.. it is small.. but if it transcodes 4 times, it's normal I get some horrible sound..
20:37.31bkrusejeev@jeevnow.com?
20:37.35wacky_my problem is not the delay, it's the sound quality
20:37.37jeev;)
20:42.05v4mpim just wating my time arent i :/
20:42.11v4mp*wasting
20:42.23jeevbkruse, i'om waiting
20:42.57wacky_anyone ever had ugly sound from system prompts ?
20:43.10wacky_and nice sound from Record()'ed prompts ?
20:44.02sacitecHello, i'm working with a SIP trunk + g729 to another PBX(cisco call manager). On sip debug i have the an error of 'Non-codec capabilites', it happends when the other side puts me on Hold and try to pick up the call again. http://pastebin.com/m4f7a3dad  line 111
20:45.32wacky_(could channel lock problem make this happen? :P)
20:48.30wacky_have you ever seen this error: chan_sip.c:18562 handle_request_do: We could NOT get the channel lock  ?
20:48.52wacky_any docs or pointers to some information about that I could find somewhere ?
20:49.12*** join/#asterisk bladerunner05 (n=blade@81-174-56-54.static.ngi.it)
20:50.09bladerunner05hello I configure a voip account, so outgoing calls are ok, but when come a call the caller hear the busy tone and asterisk report this error: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission
20:51.55infinity1anyone have a script to convert users.conf into a polycom contact directory
20:57.30wacky_Qwell: are any decisions based on the "core show translation" table, for stream transcoding ?
21:01.07Kattywibbles
21:01.08Kattywobbles
21:01.22Kattyfidgets
21:01.26russellbswiggles
21:01.45Qwell~lart everybody
21:01.45jbotstrangles everybody with a doohicky mouse cord
21:01.59theharhaha
21:02.21Qwell~lart everybody individually
21:02.21jbotdecapitates everybody individually conan the destroyer style
21:02.22Kattywhy do days go so slowly right before you get a puppy!!!
21:02.31Kattyit is a CONSPIRACY
21:03.12v4mpanyone got a clue to the /dev/zap/channel not existing and how i would fix ? i have installed zaptel i installed that first as followed a guide that said to incase there was problems/errors
21:03.20thedonvaughnmmm puppy burgers
21:04.16thedonvaughnIf I have 2 agents logged into 2 different queues.  Does priority still hold effect between the 2 queues?  Let's say I have queue A and queue B.   If a call comes in on queue A with priority 0 and a call comes in to queue B with priority 10 (assuming they're the only calls) is the agent guarenteed to get the call on queue B first due to priority?
21:04.50thedonvaughnasked this earlier this morning, but Katty said it was too early :)  Anyone know off top of their head how this works?  In about 3 hours i'll just be able to test it myself heh
21:05.03Kattypouts and looks through puppy toys again
21:06.27*** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17)
21:06.48tristanbob_does anyone use hte Polycom 501?
21:06.56tristanbob_do you recommend that over the 430?
21:08.20*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
21:13.21*** join/#asterisk moy (n=moy@63-255-123-231.ip.mcleodusa.net)
21:13.27*** join/#asterisk MrTelephone (n=test@h697179-171.picriverisp.net)
21:13.52*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
21:14.02MrTelephoneJust wondering why the newer wanpipe drivers don't patch zaptel anymore? I'm getting some D-Channel errors and cannot dial out on PRI but RBS channel bank works.
21:14.25MrTelephoneAsterisk also says it cannot turn off echo cancellation on channel 1,2,3 etc. Some warning messages...
21:15.45bladerunner05hello I configure a voip account, so outgoing calls are ok, but when come a call the caller hear the busy tone and asterisk report this error: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission how may resolve this ?
21:17.05krokodilerianbladerunner05 , tcpdump/wireshark and see why a packet is getting lost or something like that, not going to the right place
21:17.38krokodilerianlast time i saw something like that, it was because i had nat=yes and the carrier was using something weird
21:21.25maxximIs it possible to increase the RTP audio volume for incoming SIp connection?
21:21.48maxximIs it possible to increase the RTP audio volume on asterisk (not with gain parameter) for incoming SIp connection?
21:21.55krokodilerianthere's no gain for sip :)
21:22.02theharmaxxim: good rephrasing :)
21:22.14krokodilerianand i don't think so, the only thing asterisk can do to a bridged stream is to transcode it
21:22.58maxximthere is any other products that can do it?
21:23.03maxximsome modules, etc...
21:23.29maxximis it true that via VPN the volume is decreasing? why?
21:23.55maxximrephrasing: is it true that VPN is the cause of volume decreasing?
21:25.52MrTelephonebecause it adds latency and echo cancellation goes into overdrive
21:25.59*** join/#asterisk zer0x- (n=zer0x-@63-255-123-248.ip.mcleodusa.net)
21:26.02MrTelephoneby lowering the volume
21:26.54krokodilerianmaxxim , i have never heard of a VPN that touches your data in the packets
21:27.15maxximkrokodilerian> i've just read on the forum about this...
21:28.59krokodilerianwhere?
21:29.05maxximthe thing is , when i'll calling directly to asterisk (via vpn) the sound is cool (playing music). But when I call to PSTN vi asterisk (via VPN) the incoming volume is low. INcresing the gain makes big echoes... I just don't know how solve this probleam. I've read a lot of forums.. all says to increas the 'gain' on gateway.... but in my case it makes big echoes (echo cansalation on gateway is ON, and on softphone is also ON)
21:29.07MrTelephonepeople are on crack
21:29.34maxximkrokodilerian> on digium foruM: http://lists.digium.com/pipermail/asterisk-users/2008-January/203594.html
21:29.49maxximsomeone wrote: "Are you using VPN between the two sites? If that is
21:29.49maxximthe case, then you wil have a low volume and I faced
21:29.49maxximthis problem. Just try without VPN.
21:29.49MrTelephoneturn off echo cancellation
21:29.51maxxim"
21:30.12maxximMrTelephone> where to turn off echo cancellation: on soft phone, or gateway?
21:30.23krokodilerianwtf
21:30.42MrTelephoneit reduces echo by reducing the volume
21:30.44maxximkrokodilerian> did you read it? i was wondering. this is why i was asking it here
21:30.46MrTelephonenot much you can do
21:31.00krokodilerianmaxxim , this sounds extremely unlikely
21:31.23krokodilerianpretty much impossible
21:32.01maxximMrTelephone> i just can't imagine, at what side this echo is detected, and why the volume is decreased....
21:32.23logicwrathin a NAT scenario are there ports besides 5060 and the rtp.conf range that need to get forwarded to the * server?
21:32.30*** join/#asterisk Strom_C (n=strom@208.127.172.112)
21:32.31maxximkrokodilerian> i think the same like you regarding VPN
21:32.48x86~sipnat
21:32.48jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
21:33.08x86logicwrath: read that
21:33.23logicwrathive read these already
21:33.30maxximMrTelephone> explain me please this think. Why when i'm calling to the SIP client with X-LIte, i hear it well. But when i call to PSTN via gatwway (taht is the same SIP client), the incoming volume is low
21:33.42krokodilerianx86 , is there a list of the ~something stuff from jboot?
21:34.01maxximboths are far from me: SIP client with X-Lite, and SIp client as gateway with exit to PSTN
21:34.10x86krokodilerian: no idea
21:34.15krokodilerianmaxxim , that's a PSTN problem, not a SIP one, e.g. it has to be fixed on the pstn gw
21:34.52maxximkrokodilerian> i dont' have much to do on the gateway... just this parameter : input and output gain...
21:34.58maxximhow can i fix it?
21:35.22krokodilerianmaxxim , just increase the input gain :)
21:35.33maxximkrokodilerian> it makse big echo... by increasing the input gain
21:35.37logicwrathi get no audio when an inbound call gets auto forwarded to an outside phone, it works fine if someone calls and I transfer them manually
21:35.39maxximi can hear myself aster 1 sec
21:36.11maxximI was thinking, if asterisk can decode and incrise the volume of input traffic, and to encode again, and to send it to me
21:36.14[TK]D-Fendermaxxim: then you have traded voluem problems for echo problems.  Thats what you gt with inferior EC
21:36.14maxximit will be great!
21:36.34krokodilerianmaxxim , but there's no such thing yet, it doesn't really touch the audio itself
21:36.43eric2what's the proper order when installing the following:   asterisk, zaptel, libpri and wanpipe (sangoma)?
21:36.44[TK]D-Fendermaxxim: * 1.6 has func_volume.  Go look it up.  Nothing exists before 1.6
21:37.10maxxim[TK]D-Fender> xe-xe-xe...let see what can du this func_volume
21:37.35maxxim[TK]D-Fender> "Thats what you gt with inferior EC" <- what do you mean, can you explain it to me please?
21:37.46x86[TK]D-Fender: if it was an analog channel he could simply adjust the gains... as far back as 1.2 afaik, never used analog with 1.0, so dunno
21:37.58krokodilerianeric2 , libpri, wanpipe,zaptel, asterisk
21:38.08[TK]D-Fendermaxxim: If basic gains screw up your EC, then your EC isn't very good
21:38.21x86ah, yeah
21:38.26maxxim[TK]D-Fender> that is EC?
21:38.28[TK]D-Fender1.0?  WTF
21:38.32x86echo cancellation :)
21:38.52x86man he's talking about a 1 second tail lol
21:38.53x86that's nuts
21:40.06maxxim[TK]D-Fender> may be EC is not good because is a big lancety? hmm 100ms is a big lancety?
21:40.11eric2tx krokodilerian
21:40.47v4mpanyone got a clue to the /dev/zap/channel not bein written to and how i would fix ? i have installed zaptel i installed that first as followed a guide that said to incase there was problems/errors this seems to be the only real problem according to debug
21:41.06[TK]D-Fendermaxxim: Echo is not caused by SIP, its caused by the PSTN interface
21:41.27*** join/#asterisk Tond (n=t@CPE001a70b2449d-CM00194747ae5e.cpe.net.cable.rogers.com)
21:41.52maxxim[TK]D-Fender> is func_volume functionality finished? can i use it?
21:42.04TondHi, is there a way for me to voew active channels on my asterisk from a web browser (possibly iphone) ?
21:42.06[TK]D-Fendermaxxim: Go look.  I haven't touched 1.6 personally yet
21:42.24[TK]D-FenderTond: Natively no.  Easily codable, yes
21:42.28Tondi want to be able to view # of active channels, etc from my iphone
21:42.47TondYa, the coding should be simple, but not sure how simple as I am not a programer..
21:42.49[TK]D-FenderTond: And you just went and said the same thing twice, worded backwards
21:42.59Tondha ha ha
21:43.04TondTrue
21:43.10[TK]D-FenderTond: Then you're going to have to commission one regardless.
21:43.43Tondwell, I am going to try and code it out, any pointers you can give me and I'll go read up on it?  like a head start?
21:45.15[TK]D-FenderTond: Go look at what you can execute from * CLI and thuse through UNIX CLI
21:45.17maxxim[TK]D-Fender> "its caused by the PSTN
21:45.17maxxim<PROTECTED>
21:45.32maxximit means that my gateway device is bad?
21:45.41Tond<[TK]D-Fender> tnx
21:46.07[TK]D-Fendermaxxim: if it causes echo when you increase the gains to "normal" levels, yes
21:47.04maxxim[TK]D-Fender> it's normals (0,-3) make the incoming volume to be low... Lower than an normal call between SIP users (shoft phones), Lower then listening a Playback music on asterisk
21:47.51maxxim[TK]D-Fender> by increasing gains (0,9) or (-18,9). it makes the volume higher, that in this case the ECHO can be heared
21:48.00[TK]D-Fendermaxxim: *'s stock recordings are what is called "normalized".  Conversations should be on about the same level as those.
21:48.15[TK]D-Fendermaxxim: What kind of gateway are you using?
21:48.25maxxim[TK]D-Fender> bought a new AddPac AP200D
21:48.35[TK]D-Fendermaxxim: Never heard of them.
21:48.45maxxim"stock recordings" -> what is this?
21:48.46Tond<[TK]D-Fender>: I know how to use the * cli through SSH, but what i need to know is how to get PHP to connect to ast manager and run the * cli command
21:48.52Tondand view it
21:49.14[TK]D-Fendermaxxim: Just basic * prompts like the voicemail menu, etc
21:49.30maxxim[TK]D-Fender> like Playback. right?
21:49.33[TK]D-FenderTond: Go through UNIX CLI then
21:49.56Tondso u mean ssh from iphone to the ast server, etc etc?
21:50.05[TK]D-Fendermaxxim: not just Playback.  Playback can play ANY sound file.  As long as we are talking about the exact sound files that came supplied with *, then yes
21:50.17[TK]D-FenderTond: No, from your WEB SERVER
21:50.29[TK]D-FenderTond: and no, not SSH.
21:50.42[TK]D-FenderTond: unless you're going to use another box
21:50.45maxxim[TK]D-Fender> thansk for advices!!!
21:51.03[TK]D-FenderTond: At which point I would suggest using AMI instead
21:51.13Tond<[TK]D-Fender> no, the webserver is available on the same server.
21:51.16Katty[TK]D-Fender: :>
21:51.24Katty[TK]D-Fender: puppeh shopping eta 5 minutes!!!!
21:51.46Katty[TK]D-Fender: nothing makes a woman feel better than spending money
21:51.49[TK]D-FenderKatty: I thought you picked it out already...
21:51.53Katty[TK]D-Fender: i'm psyched
21:51.58Katty[TK]D-Fender: oh i did. i'm just buying toys
21:52.01[TK]D-FenderTond: then clearly no need for SSH
21:52.10Katty[TK]D-Fender: and treats, and doggy food, and a pet bed, and the leash, and collar, and pet id
21:52.25[TK]D-FenderKatty: that'd be "shopping for puppy".
21:52.41[TK]D-Fenderconjugates Katty
21:53.07Tond<[TK]D-Fender> Just not sure how to connect with the UNIX CLI, etc.. Kinda confused about it.
21:53.26Katty[TK]D-Fender: take your math terms elsewhere.
21:53.44Katty[TK]D-Fender: i have shopping to do!
21:53.56*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
21:55.09Tond<[TK]D-Fender> nevermind, got it
21:55.17Tond<[TK]D-Fender> asterisk -rx, right?
21:56.43[TK]D-FenderTond: Yes
21:57.48*** join/#asterisk SwK (n=SwK@freeswitch/developer/swk)
21:59.58*** join/#asterisk murdock_ut (n=chatzill@70.102.148.44)
22:00.02murdock_utHowdy all.
22:01.33v4mpanyone got a clue to the /dev/zap/channel not bein written to and how i would fix ? i have installed zaptel i installed that first as followed a guide that said to incase there was problems/errors this seems to be the only real problem according to debug
22:06.07*** join/#asterisk zydoon (n=zydoon@41.225.153.114)
22:06.20*** part/#asterisk zydoon (n=zydoon@41.225.153.114)
22:08.52TondOk, i am running the following line from SSH asterisk -rx 'sip show channels' and it works fine, but when i run it from PHP, i get the following message printed: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?).  ANy idea why?
22:09.10krokodilerianTond , your php is not running as root
22:09.16krokodilerianor is it?
22:09.17Tondasterisk and webserver are on the same computer
22:09.21troubledwhats the highest sampling rate a standard telephone can get?
22:09.35Tondkork> not sure, how do i check?
22:09.41troubledor is everything 8khz mono?
22:09.45krokodilerianTond , if it's a web server, then it's not running the php as root
22:09.51krokodileriantroubled , everything is 8khz mono
22:10.00krokodilerianexcept stuff like skype, which has 16khz stuff
22:10.06krokodilerianbut the pstn is just 8khz mono
22:10.12Tondkrok, ya it is through webserver
22:10.21joattroubled, use one of the better codecs
22:10.21troubledkrokodilerian: no way to get 16khz in asterisk with something like an iaxy instead of the tdm cards?
22:10.26krokodilerianTond , so it's running under a user that can't connect to asterisk
22:10.35troubledor is the telephone equipement itself only sending 8khz?
22:10.47krokodileriantroubled , no, asterisk supports only 8khz
22:11.01Tondthe user will be apache then, how do i make that user access asterisk?
22:11.13joatkroko... not so sure that's true...
22:11.24troubledkrokodilerian: damn. was wondering for use in my console/intercom system if the quality could be better, but im using a telephone
22:11.39joatknow it's 8-bit but think kHz can be higher, depending on codec
22:11.41krokodilerianjoat , well, check :) it says 8000 everywhere in the sources
22:12.01krokodilerianthe slinear stuff which is the uncompressed codec there is 8khz
22:13.34krokodilerianif you do a grep in the sources, you'll see
22:14.58joathmm... ilbc is 13.3 and speex can run up to 24.6 kHz... however, nobody likes those
22:15.12v4mpanyone able to help ?
22:15.13*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
22:15.26troubledthanks. just teh quality is noticable since im using the telephone as a crude mic atm and feeding the console dsp out to my workstation mic in and its terrible
22:15.55troubledwell, standard phone quality, but still very tinny compared to a 44khz headset or something
22:16.15joattroubled, what codec you using?
22:16.31troubledmy input is a TDM400p atm, so I gather its 8khz
22:16.39joatyep
22:17.03joatfeed audio to your mic in could be the problem also... probably overdriving
22:17.09troubledbut since its just a telephone, I figured that its the phone itself anyways. although I do have an iaxy I can use, i dont know if it would help to use a diff codec
22:17.11joatthere's no line in?
22:17.32troubledwell, I have a line in, but then I dont think my apps would see the sound input
22:17.47joatapps?
22:17.56troubledsound apps, anything that uses a mic
22:18.11troubledor is line in = mic for non amplified?
22:18.35joati think mic is amplified
22:18.47joatwait... backwards
22:18.52joatmic needs amplification
22:19.09joatline in doesn't as it comes from a powered source
22:19.21troubledbut from a sound card POV, is line in = mic?
22:19.30joatdon't think so
22:19.53joathow many jacks on that thing?
22:20.02troubled5.1, so I got a few :)
22:20.09troubledsb live! value card
22:20.23joatthe green one should be line in
22:20.27troubledline in, out, mic, coax and a few others
22:20.31troubledI think so ya
22:20.39joattime to read yer docs
22:20.42troubledthe sound works in line in from the line out on the console
22:20.42joat:)
22:21.01troubledmy problem isnt getting audio, its getting higher quality audio :)
22:21.07joatoh..
22:21.35joatthen what krok said... asterisk doesn't do the higher end audio
22:21.35troubledi would use the iaxy if I would get 16khz or something better, but I think the iaxy only supports alaw/ulaw and maybe gsm
22:21.50[TK]D-Fenderno point
22:21.54joatthose three are probably your best bet
22:22.01troubledya, not when the phone im using as a mic only does 8khz :)
22:22.08joatwhy would you push high end audio into a phone system?
22:22.15troubledill just go get a headset for higher quality
22:22.24*** join/#asterisk dlewis (i=4576958c@about/security/staff/dlewis)
22:22.35[TK]D-Fenders/higher quality/not a complete POS/
22:22.38troubledjoat: I already said, its functioning as a crude mic since I cant find my headset
22:22.45joati do listen to spoken podcasts via asterisk
22:22.46joatah
22:23.19*** join/#asterisk grEvenX (n=even@c9A7F8BC3.dhcp.bluecom.no)
22:33.29v4mpguys how would i fix this issue i have looked through configs and im still baffled
22:33.34v4mp[Sep 26 23:32:35] WARNING[29531]: chan_zap.c:898 zt_open: Unable to specify channel 1: No such device or address
22:34.04*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
22:34.10v4mpthe error line is this
22:34.18v4mp[Sep 26 23:32:35] ERROR[29531]: chan_zap.c:7209 mkintf: Unable to open channel 1: No such device or address
22:34.19v4mphere = 0, tmp->channel = 1, channel = 1
22:49.18*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
22:49.34maxxim[TK]D-Fender> fantastic!!!! func_volume works!!!!
22:49.59maxxim[TK]D-Fender> it defenetly increase the incoming volume
22:50.02maxximxe-xe-xe
22:50.57joatmaxxim: i could never get the volume reset function to work... the caller/room  talk/listen functions work nicely though
22:51.44maxximjoat> caller/room  talk/listen -> what are these functions for?
22:51.59TondHow do i run asterisk as user asterisk?  I downloaded and compiled the code as root, and i guess all is running as root now
22:54.59*** join/#asterisk mateo_au (n=chatzill@12.144.159.231)
22:56.48maxximthere is a EC (echo cancalation) in asterisk 1.6 ?
22:56.51joatmaxxim: thought your were talking about the functions to adjust volumes in the conference room(s)
22:57.07joatyou can adjust a single callers volume for talking or listening
22:57.16joator you can do it for the entire conference room
22:57.27maxximjoat> i'm truing to use that function for incoming PSTN call
22:57.30maxximit was to low
22:57.35joatobviously you were talking about a different function
22:57.39joatah
22:57.53maxximexten => s,n,Set(VOLUME(TX)=4)
22:58.02maxximi did like this. and it increased the volume...
22:58.19joatah!  haven't played with that one yet!
22:58.32maxximit defenetly increase the volume...
22:58.49maxximwhen i was using the input/output gain on Gateway, i had a big echo
22:58.50joatscribbles hurriedly
22:58.51joatthanks!
22:59.17maxximbut increasing the volume on asterisk side, it doens't affect the echo
22:59.23*** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com)
22:59.30maxximanyway , I have a little echo.. and i want to cancel it via asterisk too
22:59.33UnixDawgok anyone in the office at digium
22:59.42maxximthis is why i'm asking, if there is any EC function in 1.6
22:59.48UnixDawgI just got a demo login for the switchvox
23:00.00UnixDawgbut there was no passwd in the emil
23:00.04UnixDawgemail
23:00.12murdock_utIsn't a parked call that rings back and is not answered supposed to go to the park-dial context?
23:03.14*** join/#asterisk eric2 (n=nobody@sudbury-cable-69-60-228-64.unitz.ca)
23:03.51v4mpguys how would i fix this issue i have looked through configs and im still baffled
23:03.53v4mpthe error line is this
23:04.00v4mp[Sep 26 23:32:35] ERROR[29531]: chan_zap.c:7209 mkintf: Unable to open channel 1: No such device or address
23:11.34beekv4mp: Is zaptel started?
23:11.40v4mpyes
23:13.08beekpastbin the output of ztscan
23:16.18*** part/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com)
23:22.37maxxim[TK]D-Fender> are you here?
23:29.41maxximThere is echo cancelation for asterisk. I'm using an extern gateway, so i can't use echo cancelaton of Zaptel.
23:29.44maxxim?
23:32.08logicwrathwhat about soft echo
23:32.17logicwrathis anyone using that
23:38.55drmessano<PROTECTED>
23:43.20maxximdrmessano> addpac ap200d
23:43.42maxximdrmessano> it has it own echo cancelation. but anyway i'm hearing myself
23:43.59maxximdrmessano> input , output gain didnt help
23:48.37*** part/#asterisk beek (n=klinebl@65.211.106.242)
23:50.59infinity1i can't get playback to work. it says the file doesn't exist and the file does exist.
23:51.02infinity1any ideas?
23:53.04*** join/#asterisk [gnubie] (n=gnubie@cm141.omega112.maxonline.com.sg)
23:56.38bkw_how are you doing the playback?
23:56.44bkw_the exact line from your dialplan?
23:56.50infinity1ok.
23:57.23infinity1i figured it out. asteirsk-gui saves the file in /var/lib/asterisk/sounds
23:57.41infinity1and playback uses /usr/share/asterisk/sounds
23:58.58bkw_haha
23:59.01bkw_thats ok
23:59.04bkw_:)
23:59.08infinity1took me 30 mins
23:59.09infinity1shit
23:59.13v4mpright ok i've solved my problem but cant find anything of use on google i've tryed and failed at what i've found
23:59.23v4mpits not connectiong to mysql
23:59.32infinity1i'm not sure how i should fix it. maybe link /var/lib/aserisk/sounds to the /usr/share one
23:59.55bkw_or fix the agi

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