IRC log for #asterisk on 20120717

00:00.17sparqyou should have asked some insurance companies to pay for the hardware
00:00.36sparqbetter weather data -> better actuarial tables -> fewer losses
00:00.38mcf3782I wish I'd thought of that.
00:01.32sparqshrugs
00:01.38jpsharpYou can use the Bluetooth channel for Asterisk and peer it up with a mobile phone and get SMS's like that.
00:01.39sparqthey probably wouldn't have gone for it either
00:01.56sparqyou know, because of general corporate derp
00:02.09sparqjpsharp: Yeah, I might have to do that.
00:02.23mcf3782I didn't have a lot of experience dealing with mind numbingly stupid government pencil pushers, so when I got the "deer in headlights" look across the table after the second meeting with the NWS people; I just kind of dropped it.
00:02.29sparqjpsharp: unfortunately, the server is currently in Amazon's EC2
00:03.01sparqmcf3782: sigh.
00:03.05jpsharpOh. Heh.  Yeah, then there's a few ITSPs that will send incoming SMSes via a SIP message packet.
00:03.17sparqjpsharp: ooo?
00:03.23jpsharpYou'll need a 10.something version of Asterisk to support it.
00:03.45sparqjpsharp: do you know of one that doesn't charge per-message fees?
00:04.17WIMPysparq: How much data do you transmit?
00:04.22sparqthe network will generate something like ~10-100 inbound messages per minute, which would accumulate VERY quickly at $0.05 a message
00:04.30jpsharpVitelity says they'll do unlimited for $9.95/mo
00:04.34jpsharpper DID
00:04.54sparqWIMPy: Not much data, but SMSs are stupid expensive
00:05.06sparqlooks at Vitelity
00:05.08WIMPyWho much?
00:05.12WIMPyHow much?
00:05.17jpsharphttp://www.vitelity.com/pricing/#sms\
00:05.21jpsharphttp://www.vitelity.com/pricing/#sms
00:05.23mcf3782Yes, and they are pure profit for the Wireless carriers too.
00:06.05sparqWIMPy: ~one SMS a second
00:06.29WIMPyasked how much, not how often.
00:06.55sparqwell, just take ^(-1)
00:06.56WIMPyWhat do you send in a message?
00:07.41sparqit's going to be operating on a continuous basis (24x365)
00:07.49sparqminus breakage
00:08.41WIMPyOk, I don't know what the contracts look like in your area, but maybe it would be an option to just place calls and encode the date in the setup?
00:08.43jpsharpBetter make sure that the "unlimited" messaging is truly unlimited, then.
00:09.21*** join/#asterisk Iamnacho (~Iamnacho@ip174-70-132-58.ks.ks.cox.net)
00:09.36mcf3782If you could get the cell owner to cooperate, you might be able to piggy-back your traffic into a serial port on the switch, and send it to some other location where it could then be siphoned off and spit out onto a better network.
00:09.50sparqWIMPy: message look like 'pH:8.83,ORP:3.32,DO:53.02,C:32.22;32.32,T:22.03'
00:10.46WIMPyLike 6x16Bit?
00:11.10sparqmcf3782: that would be wonderful... but I doubt it. I've just been plopping these nodes down in weird places in the Sierras
00:11.25sparqWIMPy: sure
00:11.40jpsharpYou'd be hard pressed to get a cell carrier to cooperate like that.
00:11.45WIMPySo it would be possible.
00:12.03sparqjpsharp: Yeah. I'm not going to count on any kind of cooperation
00:12.18sparqjpsharp: it would just limit the number of sites where it will work
00:13.32WIMPyI'd try and place a call and encode the data in to the subaddress.
00:13.40mcf3782Do you have a ham radio license, or can you get one? If so, you could probably push that kind of data onto the amateur radio packet network…..but that's probably more hardware than you'd want to get involved. The Western Inter tie repeater network would cover that area quite well.
00:13.47sparqHmm. $9.95 for unlimited SMS receiving would be awesome.
00:14.12*** join/#asterisk Lepon (Lepon@60-241-109-208.static.tpgi.com.au)
00:14.16sparqmcf3782: That would be nifty
00:14.51sparqmcf3782: that's something I might do once I have SMS receiving working
00:15.04sparqmcf3782: Another thing I like about SMS is that my sattelite modules talk through SMS
00:15.29sparqmcf3782: So I can send messages from the middle of the open ocean, which is actually in the offing
00:15.42jpsharpOhyeah.  APRS is what you want.
00:15.53mcf3782sparq: have a look around on google for APRS.  Might give you enough background to see if it would work for you.  We hams push weather-related data (with GPS even) around via APRS and packet radio all the time.
00:17.00Leponhey all, quick question I need one of my asterisk servers to reject all incoming calls from my sip provider (with an error would be fine) without affecting the calls it is already handling in progress. Any ideas?
00:17.02sparqoh, that's cool
00:17.31sparqmcf3782: Definitely will do.
00:17.31WIMPyLepon: Put a Hangup in your dialplan?
00:17.52Leponcan you do a hangup without an answer?
00:17.53mcf3782My Jeep has a small VHF radio, an embedded GPS/APRS packet node in it. It sends me my position information constantly in the background.
00:18.05WIMPyLepon: Sure.
00:18.08sparqmcf3782: but, it's hard to beat basic GSM for cost. A SimCOM Sim900 module is only about $15 in volume.
00:18.11jpsharpLepon: use Congestion.  Sends a message without answering the line.
00:18.12WIMPyYou can even set a reason.
00:18.35mcf3782wow.
00:18.43mcf3782yea… can't touch those prices.
00:19.05sparqmcf3782: And there are packages that have integrated GPS for maybe $25
00:19.06Leponokay i might try that then thanks. What im doing is rejecting the call so the sip provider sends it on to my next server and I can clear the calls out of the current one without affecting the calls that are currently on it till they finish
00:19.09Leponcheers guys
00:19.39mcf3782That's not bad at all.
00:19.58WIMPyLepon: Why don't you just register the other server?
00:20.01sparqmcf3782: http://wm.sim.com/wirellessen.aspx?id=1090
00:20.10jpsharpA friend of mine used the SimCOM unit in a telemetry system for commercial ice makers.
00:20.34sparqmcf3782: They just talk AT commands over serial, so it's really easy to integrate them
00:20.56mcf3782That's definitely one to bookmark for me.  Thanks! :)
00:21.14Leponboth the servers are registered with the sip provider, but they send all the incoming calls to the first server then the second one if the first doesn't take it. The only way I have of changing that order is logging a ticket with them and waiting for them to response (which could take hours)
00:21.16jpsharpBut he's talking GPRS rather than SMS.
00:21.41sparqmcf3782: Here's what's on my prototype: http://www.seeedstudio.com/depot/gprs-shield-p-779.html?cPath=132_134 (the microcontroller is an Arduino Uno)
00:21.45WIMPyLepon: Or just stop registering the first server?
00:22.00WIMPyLepon: Anyway: A hangupt will do it immediately.
00:22.25Leponhangup sounds like the easiest way to put it in and out at the moment thank you :)
00:22.41sparqjpsharp: yeah, they are really great to work with. Good old AT.
00:23.51WIMPyEven the latest smartphones do it that way.
00:24.21mcf3782The guy who wrote the original "Hayes AT" command set lives not far from me.  He's a really interesting guy.
00:26.09sparqHeh. Tell him I said AT+CGMS="his_name"\r S'up?ctrl-z
00:26.22mcf3782ROFL
00:26.51*** part/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com)
00:27.17mcf3782Dale could probably respond in kind, without having to even think about it. :)
00:28.29mcf3782sparq: http://wa4dsy.net/
00:29.43LeponOnce again you have all been very helpful, have a good day everyone.
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00:32.10*** join/#asterisk sarsaeol_ (~sarsaeol@66-113-78-49.rev.ibsinc.com)
00:32.50sarsaeol_can someone tell me what the (T) means after the port display when issuing a 'iax2 show peers'?
00:33.12sarsaeol_liek so
00:33.13sarsaeol_http://pastebin.com/Q7rqMPpR
00:34.04jpsharpT means trunk enabled.
00:34.16jpsharpI believe.
00:34.53jpsharpYes.  T means IAX2 trunking.
00:35.39sarsaeol_word
00:35.43sarsaeol_thanks jpsharp
00:36.07sparqmcf3782: cool!
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00:49.52mcf3782sparq: you never know what might show up on Dale's web site either.  He cashed out and "retired" from Hayes when he was very young.  Now he just works on whatever interests him.  Like tracking collars for his cats, for instance.   At one point, he had a map that plotted the location in all 3 axis on a map, overlayed with a picture of his farm. Why?  Just because he could. :)
00:50.51jpsharpDamn.  Now I wish I would've/could've met him while I lived in ATL.
00:53.56mcf3782jpsharp - he's a very unassuming person. To pass him in Home Depot or Kroger, you'd never know he was a genius with lots of $$$$
00:55.37jpsharpThat's the kind of people I like.  The ones that call you up at 4am and yell "DUDE, COME OVER HERE.  Check out what I built!"
00:58.00jpsharpOr my friend Dave who came flying down the stairs of his house one afternoon and yelled "DUDE, I JUST BOUGHT A CRAY!"
00:58.10mcf3782You'd be more likely to get an email message, with links to pages detailing what he'd built.  :)
00:58.39mcf3782heh. We had a baby Cray at GaTech for a while.
00:59.03mcf3782The later ones weren't quite as geeky cool as the Cray 1.
00:59.30jpsharpNo, but there aren't too many of them you can run in your living room.
01:00.04mcf3782True. That's either one very warm, or one very cold living room. :)
01:01.46jpsharpAnd noisy
01:08.45*** join/#asterisk TheMan (~garry@nv-208-13-131-232.sta.embarqhsd.net)
01:09.13mcf3782Our machine room was always so loud from the air handlers, it was hard to tell how much noise any of the computers themselves made….with the exception of one of the Sun boxes… When that box powered up or down, it sounded like a jet aircraft was in the room.
01:15.42jpsharpOne of the big E10Ks?
01:15.53mcf3782yep
01:21.03jpsharpMegawatt-hours of SPARC power.
01:27.45mcf3782And it's now less powerful than most desktop computers. :)
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01:42.01nanoha-samaaccess
01:42.07nanoha-samashit wrong window
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03:59.48junminHell all, Given a DID number, may I know where it is "hosted"? Is there something like IP Traceroute that I can know where "calling DID" get through(the path?)?
04:00.47[TK]D-Fendernot really
04:01.52junmin[TK]D-Fender: so... no way?
04:03.42[TK]D-Fenderther is nothing is SIP to tell you this.  Only CLECs+ have any means to try to track down the owner of a #
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04:08.33junmin[TK]D-Fender: thanks. by the way, Given a DID, is there a smart way to get to know how many channel it has?
04:08.51[TK]D-FenderDID's dont have channels
04:09.54junmin[TK]D-Fender: eh? confuse, buying a DID, it always says XXXX has available X channels
04:11.27ChannelZIt says that where?
04:12.01junminEach DID number is provided with two channels, allowing two concurrent incoming calls on the same number. Additional channels may be purchased if required.
04:12.09junminhttp://www.didww.com/virtual_numbers
04:12.17[TK]D-Fender~did
04:12.18infobotdid is, like, Direct Inward Dialing, or just a phone number
04:12.18ChannelZThat's between you and the ITSP..
04:12.35ChannelZThey make up the rules
04:12.37[TK]D-Fenderjunmin: that is a number.  Nothing more.
04:13.04[TK]D-FenderI can have 15 T1's filled with nothing but calls for that one DID
04:13.44[TK]D-FenderWhen you pay for a DID, you get the number.  That is all
04:14.02[TK]D-Fender"Channels" is some other part of the overall product you're paying for.
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04:19.19junminI think I am understanding it ... well. hope you guys can give me some advise. the problem is that I am trying to build a calling card service with local access number, the local number(DID) i want is 2 channels FIXED. i need more people can call in the same time. is there anything i can try?
04:20.23junmin2 channels fixed, maybe it's because  local carrier(or ITSP) maks the rule.
04:20.41junminI have asked them for additional channels, they say no.
04:25.05[TK]D-Fenderjunmin: if you want to limit simultaneous calls fom the same CALLER, then all you have to filter by is whatever ID code you make them enter, or their CALLERID number
04:25.21junmin(looks like there is another company with Virtual PRI with the same local number(both their number and mine star with _159XXXX). That why I would like to know where they get it)
04:26.26junmin[TK]D-Fender: maybe i dont explain it clearly.
04:26.35junmin*I didn't
04:27.37ChannelZVitelity?
04:27.38junmin[TK]D-Fender: I need more simultaneous calls capacity(calling to the same DID number).
04:28.04[TK]D-FenderSo pay for a service that offers you more
04:30.22junminChannelZ: I will ask them. I check didnumbers and didww, no good news.
04:30.53ChannelZI just wondered if that's who you were looking at, they have a service called 'Virtual PRI'
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04:31.58junminChannelZ: yes, but not specifically for the DID i want.
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04:35.50ChannelZWell you'll just have to hunt around.  voip.ms will let you do a bunch of channels but if you pay per minute..
04:36.56junminChannelZ: expensive cost :s if pay per minute
04:37.27junminChannelZ: any other more providers that you know?
04:37.39ChannelZ~itsplist-us
04:37.39infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
04:39.03ChannelZLooks like you're going to be in the $15-20/channel.  Not sure how many you're looking at.  And presumably the calls go somewhere afterward (you said calling card service?) so you've also got outbound to deal with
04:40.00junminChannelZ: yes ...outbound gb's
04:42.57junminChannelZ: do you use voip.ms? how's its service?
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04:44.06ChannelZno I use Vitelity, and just for my home, not a lot of volume
04:44.31[TK]D-Fenderjunmin: You don't seem to have come in understanding the basics of telephony, don't know any common providers and apparently none of the economics of what this business model requires.  This does not look good for your "plan"
04:46.40[TK]D-Fenderjunmin: All of the providers listed so far a consumer-grade and there is no reason your clients shouldn't be able to replace you
04:47.08[TK]D-Fenderjunmin: CC is a volume game, and a very difficult market
04:47.26junmin[TK]D-Fender: you may probably right. really thank you for your observation. I am just starting at this area. newbie.
04:48.33[TK]D-Fenderjunmin: You also have to consider the legal issues given that many of these services don't allow you to resell their use.
04:51.42junmin[TK]D-Fender: my market is the people who dont know how to use computer, very small market, very specific small market.
04:51.48drmessanoAlso, I hope Junmin isn't the name of your service.  It would look terrible on a card hanging from a rack in Walmart.
04:52.57drmessanoYou're probably better off getting into the 419 business than calling cards.  I know this royal prince that could help..
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04:55.27junmindrmessano: thank you ...
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06:25.06schmidtsgood morning
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06:29.55v0lZymorning
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06:53.00stianmornin :)
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07:17.39stianwee
07:17.49stiangot the AT&T prompts
07:17.53stianbetter than allison imo
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07:20.31pavlzhi
07:21.52pavlzi need one information please
07:22.43stianok?
07:23.58pavlzto the modem there is connected a switch on which is connected a phone DECT
07:24.47pavlzTiscali see my traffic but i don't phone with Tiscali, i phne with VoipStunt
07:25.46pavlzi paid 0,53 euro for a phone that i never done with tiscali, i want to masquerade my own traffic (phone)
07:26.46pavlzcan i do it ? is there a way to do it ? awating your reply, happy hacking, paolo
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07:35.10stianuhm
07:35.23stianyou want to mask your own traffic?
07:35.38stianI guess you could run it over a vpn, or encrypt it - but I realy have no idea
07:36.10pavlzi want to mask, i thank you stian for your reply
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07:36.58pavlzi got a regular contract SIP with VOIPSTUNT to phone
07:38.01pavlzand i got a regular contract with TISCALI-IT for the ADSL, i never called with TISCALI-IT, and i paid 0,53 cents of euros for a call that i never done
07:38.45*** join/#asterisk nutxase (~nutxase@unix-02.lair.co.za)
07:38.51stiansorry man
07:38.52nutxaseanyone here got mISDN working on centos 6?
07:38.55stianhoe gaan dit nutxase ;D
07:39.07stiansorry mate, no isdn here :
07:39.12nutxaselol
07:39.26nutxase:( dahdi is killing me
07:39.42stianheh lots of things are killing me as well
07:39.47stiantrying to get the damn IVR working
07:39.53nutxaselol
07:40.35nutxasehave u got dahdi working fine for a b410p?
07:42.09stianweird that it's not working though?
07:42.15stianthat card should be supported?
07:42.47nutxasewell it works but it drops incoming calls :/
07:42.55stiandid you check your logs?
07:43.00stianmaybe something is fubar in the conf
07:43.06nutxaseyea hangup cause 27 all the time
07:44.06stianIs it possible to have announcements in different languages based on caller id?
07:44.28nutxasenot easily
07:44.34nutxasebased on DID yes
07:44.45stianyeah, we have clients all over the world
07:44.47stianso... :/
07:45.38nutxasewhere do u work?
07:45.52stianhttp://swingcatalyst.com
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07:46.42nutxaseu the IT guy?
07:46.48stianHah, yes
07:46.49stianI guess
07:46.51stianand support guy
07:46.59stianand I do photograhy and videography for them
07:47.36nutxaseaah ok are u looking for someone to do ur IT and phones etc?
07:47.42stiannope
07:47.43stian:)
07:47.46nutxaseI get some of the lowest voip rates in SA :)
07:47.56stianI live in Norway
07:47.58stianlol
07:48.04nutxaseoh LOL
07:48.18stianI'm not from SA, just my hostmask is SA :)
07:48.29stianhave lots of friends there, and I speak a bit of Afrikaans
07:49.18nutxaseaah ok
07:49.29kaldemarstian: make a context that handles the language setting and then goes to the actual dialplan. funtions CALLERID and LANGUAGE will be useful.
07:49.55stiankaldemar: I was thinking about incoming
07:50.01*** join/#asterisk eicto (~eicto@2001:5c0:1000:b::672f)
07:50.04kaldemarstian: so was i.
07:50.07stianok
07:50.27kaldemarall calls are incoming from asterisk's point of view.
07:50.35stianhah, that is true
07:50.52nutxasei really wish mISDN would work!!!!!
07:51.08nutxaseid pay someone to fix it lol
07:51.21stianWhere in SA do you live, nutxase
07:51.22stian?
07:51.31nutxaseJHB :)
07:51.41stianI have a friend that lives not too far from Joburg
07:51.43nutxaseI can get misdn working on centos 5 but not 6
07:51.55stianhe could maybe help you lol
07:51.56nutxaseas if it is not compatible
07:52.07stianthen why run centos 6?
07:52.08stian:p
07:52.14nutxaseyea i wish i knew that before lol
07:52.22stianyou sure it's not the asterisk version?
07:52.40nutxaseyea tried 6 diff ones
07:52.43stianok
07:53.05nutxaseyup :S
07:53.19stianbut uhm you could always talk to my friend
07:53.22stianGrog_SA
07:53.24stianjust pm him
07:53.28stianmaybe he can help you
07:53.30stianworth a shot
07:54.31nutxasehe in here?
07:54.36stiannah
07:54.41nutxase:/
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07:57.56nutxasei wish someone from digium was in here today
07:58.22stianask Qwell maybe he knows
07:58.35nutxaseQwell are u there/
07:58.36nutxaseQwell are u there?
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08:08.25nutxasemISDN will be the death of me lol
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08:17.42stianfucking ivr
08:17.43stiangah
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08:27.22irroothi folks long time no see
08:33.01stian:)
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08:38.25nickfennellHi All.
08:38.37nickfennellI get occasional issues with stale calls on SIP channels
08:38.53nickfennellchannel request hangup doesn't clear them and the extension remains as 'in use'
08:40.14nickfennellcore show channels
08:40.15nickfennellSIP/281-000002a2     221@from-internal:1  Up      AppDial((Outgoing Line))
08:40.19nickfennellStuck.
08:40.41irrootnickfennell what ver you usin
08:40.53nickfennell1.8.5.0
08:41.31irrootah makes sense had that a while back
08:41.36nickfennellReally?
08:42.05irrootyeah
08:42.24irrootcant remember when it was fixed but it was
08:46.47irrootnickfennell check if its related to a pickup i recall pickups been the evil that caused this
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08:49.13nickfennellhttps://reviewboard.asterisk.org/r/1400/
08:50.33nickfennellThat's got your name over it ;)
08:50.57irrootit was not me .... :P my evil twin
08:51.04irrootyeah thats the one
08:51.06nickfennell:P
08:51.23nickfennellThat would make sense for here was we do use *8
08:51.36irrootthe ZOMBIE flag ... thats the one
08:53.10nickfennellSo I'm wondering if I can patch this in place rather than haul my entire system up a level
08:54.26irrootif the patch applies clean to the source should be ok
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08:59.27nickfennellyum tells me I'm about to jump to 1.8.14.0
09:00.13nickfennellbut that's cool. At least I've managed to track down what the fault is caused by
09:00.17nickfennellThat's a huge bonus.
09:00.45stian:)
09:00.47stianhah
09:00.52stianmy voice is now our ivr
09:00.53stian:D
09:02.59nutxasehttp://pastebin.com/R0FU9sYm
09:06.38nutxasewhat should the signalling be on a b410p?
09:09.12kaldemardepends on what it is connected to. btw, signalling changes require a restart, a reload is not sufficient.
09:10.07nutxasedahdi restart?
09:12.11kaldemarasterisk restart. that's what you're configuring, right?
09:12.51kaldemarof course if the new asterisk config conflicts with the dahdi config, dahdi needs to be reconfigured too.
09:13.12nutxaseok flip I can not understand this call dropping problem
09:13.42nutxasecheck
09:13.43nutxasehttp://pastebin.com/W7z0xP8Q
09:13.44nutxaseis this right?
09:16.18kaldemarno channel lines there => no configured channels at all.
09:16.33nutxasethats configured in dahdi_channels.conf
09:17.05nutxasehttp://pastebin.com/LDUmnXWh
09:18.02kaldemarlooks good, if it is in the right file.
09:18.37nutxaseyea thats dahdi_channels.conf
09:18.43nutxaseany idea why calls would cut off then?
09:18.55kaldemarapart the context being "default", but that's not likely to be the issue.
09:19.03kaldemaris that file included in chan_dahdi.conf?
09:20.08nutxaseyea it is
09:20.24kaldemarenable debug with "pri set debug 2 span <yourspannumber>" and try to catch what happens when a call drops.
09:20.34nutxaseits giving hangup cause 27
09:20.53kaldemarwhat else?
09:21.12kaldemarwhere did you observe the cause 27?
09:21.23nutxasewhen it cuts thats the error u see
09:23.00kaldemarwhere, and what else is there?
09:23.15nutxaselet me look
09:28.28WIMPynutxase: We found out how to do it some weeks ago here. replace the RedHat kernel with a standard one from kernel.org and it works.
09:30.22nutxasewhich version
09:30.40kaldemarnutxase: is SIP involved in the dropping calls? if so, how? do the drops happen in the middle of a call?
09:30.45WIMPyDoesn't really matter.
09:31.29WIMPy3.5 might be experimental.
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09:31.46nutxaseok so i install the kernel then what
09:32.07WIMPyThen mISDN will work.
09:32.33WIMPyBut it really depends on what you want which way is the right one for you.
09:33.03nutxaseok cool
09:34.44WIMPyOne day I might write some howtos...
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09:56.48nutxasehow do u remove misdn if u have half of it installed?
09:57.24WIMPyHalf of it?
09:57.33WIMPyHalf of the user part or what?
09:57.55WIMPyOr did you try the standalone modules thing?
09:58.13nutxasetried standalone modules thing
09:58.15nutxase:/
09:58.28nutxaseto install the kernel from kernel.org do i just run make?
09:59.12WIMPyThe standalone thing might work again, but I heven't tried, yet.
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09:59.33irrootWIMPy o/ hi there ltns +1 to Qwell
09:59.38WIMPyI only used the standard kernel on production machines.
09:59.57WIMPyHi irroot. Where have you been?
10:00.28irrootbeen a bit of a rough year dude but still here not spent much time on asterisk dev
10:00.52nutxaseneed to remove all traces of the misdn i think :/
10:02.12WIMPynutxase: Just make sure you have only one version, i.e. eithr the one from the kernel or the extra one.
10:03.31WIMPyirroot: Are you also using something made by RedHat?
10:04.31irrootWIMPy depends what you refering too i dont use RE or Fedora distros but there is plenty code from them all over :P
10:05.33WIMPyirroot: Ok, it was just that you said the b410p didn't work too well for you with misdn. So I thought that might have been related.
10:06.25irrootthe b410 works well on mISDN v1 the lcr bits dont work for me ... i use me own distro + src
10:06.44nutxaseiroot do u have any idea how to get it to work with misdn on centso 6?
10:06.55WIMPyWasn't it something about stability?
10:07.22irrootthe USB not work at all
10:07.55irrootthe b410 was unstable back to back NT/TE mode
10:08.17nutxasethe b410p and dahdi are making me pull my hair out
10:08.17irrootseemd to work ok in TE mode but was not happy with it
10:08.19WIMPyThere have been quite some bad git versions, and as we leraned some weeks back, they made their way at least in to RedHat kernels.
10:08.56nutxasewhats the best version of mISDN to use?
10:09.26WIMPyI haven't had any issues with the versions included in the standard kernels so far.
10:09.52nutxasewhat version do u use?
10:10.09WIMPyHere I'm on 3.1.7.
10:10.23nutxaseso thats newer then mISDN-1_1_8.tar.gz
10:10.30nutxasewhere did u get 3.1.7:
10:10.31nutxasewhere did u get 3.1.7?
10:10.38WIMPyThat's for misdn1.
10:10.46irrootim using a self maintained mIDSN 1 + 3.4.4 kernel
10:10.48WIMPyLinux 3.1.7
10:11.03nutxaseaah im using 2.6.32-220.17.1.el6.x86_64
10:11.14WIMPyFor misdn2 you need misdnuser from git.
10:11.42nutxaseso that version i pasted u will i need misdnuser?
10:11.48WIMPy2.6.32 is certainly fine, not sure what difference the addons make.
10:12.09WIMPyYou always need the user part.
10:12.19WIMPybut it has to fit.
10:12.57WIMPyUnfortunatly there are no stable type archives for v2, which is somwhat dull.
10:13.12WIMPyBut the user part hasn't given me issues so far.
10:13.19irrootAPI / ABI compatability is a problem esp when building with header files that are perhaps the wrong ver
10:14.31irrootfolks from Digium are poping in for a visit in a bit any messages for em ?
10:14.57WIMPyWell, I guess they will want dahdi :-)
10:15.35irrootwho is your dahdi ?
10:15.43WIMPyDigium :-)
10:16.24WIMPyMessages? What about questions? Such as when will we get proper support for media after disconnect?
10:16.31irrootnot the best time of the year to be here
10:16.47irrootlol think they sales folk
10:16.50WIMPynever got that bit working with dahdi.
10:17.02WIMPyBut support from Asterisk would be a good thing as well.
10:17.09irrootlol
10:17.17irrootwell seems they here
10:17.24WIMPyWhat's going on at this time?
10:17.40WIMPyAre they getting cooked?
10:21.03nutxaseso what misdn should i use
10:21.35sekilmisdn is in kernel 2.6.32
10:21.38WIMPyWhat features do you want?
10:21.39sekilalready
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10:22.02WIMPyYes, and that's v2.
10:22.38WIMPyAlthough I've used it for quite some time, I wouldn't try v1. Either dahdi or v2.
10:24.19sekilI still have some 1.1.x in production
10:24.23sekilwith b410p
10:24.32sekiland * 1.2 :)
10:24.43WIMPyOuch
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10:39.10eelcobHi all! I've noticed that the asterisk debian packages are a not updated to the current release (latest debian package is 1.8.11.1). Any chance this is happening soon?
10:44.42stianhey does anyone know how to up the audio levels in asterisk?
10:44.50stianthe call audio is a tad too low
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10:51.04kaldemarstian: what kind of calls?
10:51.12WIMPyWell, there's always teh VOLUME function, but sensibly the volume should be set right at the point whete the ADC happens.
10:53.22stianhmm
10:53.35stianboth sip-sip and sip-pstn
10:53.40stiannot via fxo, fxs though
10:53.53stiandon't have cards
10:54.24WIMPyIf it's from a sip phone, it should be done on the phone if any possible.
10:54.31stianwe tried that
10:54.34stianwas still a tad too low
10:54.57WIMPyThen you have to do it in the dialplan. Or get decent phones.
10:55.05stian:)
10:55.20stianJust tried it on our cellphones, and x-lite etc.
10:55.30kaldemarstian: for FXS, you can set rxgain and txgain values in chan_dahdi.conf.
10:55.41stianI know, but as I said - we don't have fxs cards :P
10:55.46kaldemaroh, no cards.
10:55.48stian:)
10:55.49stianI wish though
10:55.53stianbut running without it :)
10:55.56stianit's just easier
10:56.07stianand running it virtually - I know I'll get hate for this
10:56.26stianbut for only 6-10 people, it's really worth it
10:56.32stiansnapshot function <3
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11:07.52stianweird
11:07.58stianhah, all of the sudden I can't register clients
11:07.58stiangah
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12:40.52rolandowi have a question about the nat setting.. is this nat considered between * and the phone?
12:41.19rolandowin other words: if 172.16.0.10 phone connects to 172.16.0.100 * .. it's no nat between them
12:41.24rolandowshould i use nat=no then?
12:41.39rolandowor should i use nat=yes because the phone has no public IP ?
12:42.45[TK]D-Fender"the" nAT setting?
12:42.48[TK]D-FenderThere are multiple.
12:43.05[TK]D-Fender* under [general] and a possibly different value for a device
12:43.39rolandowthe nat value for a device
12:43.41[TK]D-Fenderif the device is local to * then it's setting should be "nat=no"
12:43.50[TK]D-Fenderbut that doesn't say anything about *'s WAN connectivity
12:44.28rolandowyes.. my * has public IP, so under general i have nat=no
12:48.26nickfennellYou may also want to check out localnet
12:48.30nickfennellI found that one fun
12:49.04rolandowlocalnet is configured correctly
12:51.01irrootnickfennell double points trying to bind :: :P and use localnet
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13:10.41stiananyone from norway in here?
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13:17.22WIMPy~polls
13:17.22infobot"Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask>
13:17.27WIMPystian: ^^
13:17.36wdoekesdo half-swedes count?
13:18.09irrootwdoekes only 1/2 ?? greets
13:19.32wdoekeshi irroot
13:19.44WIMPyirroot: That's just a gegographical/political thing :-)
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13:46.40pgraceIs there a way to have an ACD queue where members of the same queue are on different asterisk servers?  I assume with some logic/AGI it is possible, but wondering if there's a simple way to do it with DUNDI or something.
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13:49.59[TK]D-Fenderpgrace, Local channel + dial out to that remote server that will only ring the other device
13:50.21[TK]D-FenderNo need for DUNDI (doesn't change what you have to do on the remote end), and AGI is completely unrelated
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13:52.08Malard_Officehi does anyone have a working config for an ISDN 2 line with BT? for dealing with national and international calls?
13:52.46WIMPySure, but
13:52.48WIMPy~polls
13:52.49infobot"Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask>
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13:53.20[TK]D-FenderThat isn't quite a "poll".  He's looking for something specific....
13:53.40[TK]D-FenderIt is entirely valid.
13:53.53WIMPyHe didn't ask for the config :-)
13:54.18irrootso i take it there is no need for a poll on polls :P
13:54.20Malard_Officeokay
13:54.24Malard_Officedo you have a working config i could use
13:54.28Malard_Officeor advice on how i can make it work
13:54.34Malard_Officebt seem to want me to strip leading zero from calls
13:54.44Malard_Officeso 01733404940 would need to be 1733404940
13:54.48Malard_Officewhich is fine, i do this in the trunk
13:55.05Malard_Officebut then i can't make international calls work, not sure what the format should be
13:55.10WIMPySounds like your're using dahdi and have pridialplan=national.
13:55.17ChB82Hi! I am using Asterisk 1.4 with "rxfax". Everything is working fine. Only the papersize is in "letter" - i need "a4". What can i do?
13:55.20irrootMalard_Office isdn 2 i assume that is BRI as oppsed to PRI [isdn-30] what hardware you using / driver combo
13:55.34WIMPyCommon thing. Change it to unknown and you're fine.
13:56.10Malard_Officeokay its an openvox bri card
13:56.16Malard_Officeand i've just got a basic trixbox install
13:56.23Malard_Officei had a guy come over nickname chainsaw ?
13:56.30Malard_Officewho got the initial system working for me
13:56.49coppiceChB82: don't enable letter size, although if the sender is trying to send letter sized pages prohibiting them may have undesirable consequences
13:57.14Malard_Officei have a single outbound route with lots of X's for Dial Patterns
13:57.22WIMPyAsk his friend BFG9000 ;-)
13:57.34WIMPyMalard_Office: It's in chan_dahdi.conf.
13:57.41Malard_Officeand a single ZAP/g0 trunk which has 0|. in the dial rules
13:58.22WIMPyzap? That was replaced 4 years ago IIRC.
13:58.30ChB82coppice: okay, but how can i get the pdf-files in DIN A4? - i need this because the printer always wants "letter" but there is only A4...
13:58.34irrootWIMPy
13:58.36irrootMalard_Office eish .... that is old ....
13:58.39Malard_Officehttp://pastebin.com/xuLXfgAT
13:58.49Malard_Officewhat is old irroot?
13:59.00WIMPyYour Asterisk.
13:59.06Malard_Officeso far my setup is just a big hack to make local calls working
13:59.14irroot^^^ Asterisk + Driver + ....
13:59.17WIMPyThat is it must be old if it still uses zaptel.
13:59.33Malard_Officeumm, what do you mean my asterisk is old? i just purchased the card from openvox, if the card is no good, any suggestions on what i should use?
13:59.37Malard_Officeand what build of asterisk ?
13:59.53WIMPyNot the hardware. The software.
14:00.27Malard_Officei havent told you what version i'm running yet?
14:00.39WIMPyOk, so it must be in chan_dahdi_additional.conf.
14:00.49WIMPyAnd so it seems to use dahdi, not zaptel.
14:00.57irrootwell if its running Zap it has to be old you did say trixbox puts it @ 1.2 +/-
14:01.14Malard_Officehow can i get asterisk version?
14:01.27WIMPy'core show version'
14:01.32irrootasterisk -V
14:01.37WIMPyAt the *CLI that is.
14:01.45Malard_Officep8-pbx001*CLI> core show version
14:01.46Malard_OfficeAsterisk 1.6.0.26-FONCORE-r78 built by root @ revisor.trixbox.com on a i686 running Linux on 2010-06-08 22:01:27 UTC
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14:02.20irrootdahdi show channels ??
14:02.22WIMPyOk, guess it must be in dahdi-channels.conf, actually.
14:02.46WIMPyBut I still wouldn't use anything older than 1.8 with ISDN.
14:02.50Malard_Officeokay to paste 4 lines?
14:02.50irrootthe Zap bit may be wrong and hangover from previous bits ?
14:03.01pgrace[TK]D-Fender: awesome, I was just thinking that Local would probably do the trick.  Thanks!
14:03.45Malard_Officehttp://pastebin.com/huRk2rfe
14:04.00Malard_Officehttp://pastebin.com/kcLRYzL8
14:05.03irrootok so its definately not Zap
14:05.15irrootit may have a "alias"
14:05.28Malard_Officei dont quite know what your talking about :)
14:05.43Malard_Officedo you understand my problem though? that i am having to strip the leading zero, but then it breaks for international calls?
14:05.52Malard_Officei just have 1 sip phone and then this isdn at the moment
14:05.59WIMPyAdd the pridialplan=unknown to dahdi-channels, somewhere before the channel => 1-2.
14:06.29WIMPyAs I said: It's a common issue.
14:06.39WIMPyOr just a bad default.
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14:09.14Malard_Officei've edited the file, do i need to reload anything?
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14:09.20acidfoowhere can I download asterisk-1.4 ? I need to test something old... thank you
14:09.35wdoekesacidfoo: http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/
14:10.34WIMPyNot sure if 'dahdi restart' reloads the config. Maybe 'module reload chan_dahdi' might be better.
14:11.28acidfoowdoekes, erm, I don't know if there is something wrong... but I only see ChangeLog files :|
14:11.32stianman
14:11.40stianI've set up everything now, and it works so damn well
14:11.49stiancall forwarding, all the extensions, recorded some greetins
14:11.51stian*greetings
14:11.54stianis a happy man
14:11.56acidfoomaybe it just take a while to load ( the repository )
14:11.58acidfooah ok
14:12.08acidfooam seeing more files now
14:12.10acidfoowdoekes, thank you
14:12.19stiannow can anyone tell me how I can have two people call in at the same time (support) without having one not being able to call?
14:13.00WIMPyUnless you do some dialplan magic, you can have as many calls as you want.
14:13.11stianI'd have to enable call waiting?
14:13.21irrootstian or app_queue
14:13.28stianapp_que? :)
14:13.32WIMPyOn your phone maybe.
14:13.36stiangotcha
14:13.54stianOk, so Queue is what I want to set up then
14:14.35irrootstian it manages the waiting callers and rotates calls between available agents
14:14.50stianwell we don't have that many people calling in
14:14.56*** join/#asterisk Faustov (~fst@gentoo/user/faustov)
14:14.56stianmany people calling at all really
14:15.17stianbut I'd like it to at least have the person calling have to wait a few minutes if no one is available
14:15.23stianinstead of just terminating the call
14:15.23stianthanks
14:16.13WIMPyAsterisk would not reject additional calls on its own. Maybe the called phone dos.
14:16.22WIMPydoes
14:16.25stianno?
14:16.36Malard_OfficeWIMPy i made that change and reloaded as you said, then i removed O|. from the trunk and i just get back the all circuits are busy message
14:16.37stianIf I dial our number asterisk dumps me to our IVR
14:16.39stian:)
14:16.51stianI just want the second person calling to have to wait until it's available
14:17.10Malard_Officeand an international call still gives the same message
14:17.22irrootstian a simple way to do this in the dialplan is use a channel group and loop till its free
14:17.44WIMPyMalard_Office: I have no idea what that 0| means, but with that setting, you should be able to dial the way you'd do on any phone.
14:17.46stianirroot: sweet, where do I set up the hannel group?
14:17.55Malard_Officeit removes the leading zero
14:18.05Malard_Officeif i include it, the exchange rejects it
14:18.36irrootfunction GROUP
14:18.44WIMPyDo you use an additional 0 as a trunk prefix?
14:18.50Malard_Officeeh?
14:18.58nickfennellare you dialling 00 ?
14:19.01Malard_Officeno
14:19.03irrootGROUP_COUNT
14:19.04nickfennellfor inter
14:19.07Malard_Officejust 01733404940 for example
14:19.11stianirroot: thanks man
14:19.13nickfennellfor uk?
14:19.15Malard_Officeyes
14:19.25WIMPyMalard_Office: Are you dialling an extra 0 to get access to the line?
14:19.27nickfennellbut you need to send inter as just 00 ?
14:19.28Malard_Officeso if i dial that, without stripping the first zero in the trunk
14:19.28Malard_Officeno
14:19.36Malard_Officei have no idea what the exchange needs
14:19.39nickfennelloh
14:19.49Malard_Officei found that the exchange wants, 1733404940
14:19.54nickfennelleh really?
14:19.55Malard_Officeto dial that national number
14:19.57Malard_Officeyeah
14:19.58nickfennellThat's odd.
14:20.07nickfennellWho runs it?
14:20.10nickfennellThey need fixing :)
14:20.21Malard_Officeits peterborough orton southgate exchange
14:20.25WIMPyMalard_Office: That's only because the type defaults to national. But that makes it impossible to dial local or international calls.
14:20.28Malard_OfficeBT telecom :)
14:20.31nickfennellAh
14:20.33nickfennellexplains a lot
14:20.37nickfennellso you want to strip the 0
14:20.42Malard_Officewell, i already do that
14:20.47nickfennellok
14:20.50Malard_Office0|.
14:20.52nickfennellyeah
14:20.55Malard_Officeis in my trunk
14:20.56WIMPyNo, you don't.
14:21.00irrootnickfennell Malard_Office had a similar problem on a ISDN-30 a while back  here in Johannesburg took a while but the telco eventually fixed it
14:21.08WIMPyNut unless you don;t want international and local to work.
14:21.22WIMPynot...
14:21.22nickfennellSeems odd that you'd send without any prefix
14:21.27Malard_Officeright, but even if i remove the 0|.
14:21.33nickfennelleven in my isdn30 days i had to send as 44
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14:21.36Malard_Officeinternational fails, if i dial 001xxxxxxxxx
14:21.54j4m3shello #asterisk
14:22.14j4m3sis anyone using a Digium PCI card on a xen guest?
14:22.16Malard_Officeso i tried 4477594176xx
14:22.24Malard_Officemy mobile and it still bitched saying all circuits are busy
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14:22.32*** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br)
14:22.32nickfennellwhat about 0044
14:22.34WIMPyMalard_Office: Let's se what's really happening. set 'pri set intense debug span 1' and pastebin the output of a call attempt.
14:22.39nickfennellcanonical dial.
14:23.04irrooto/ Katty
14:23.17Malard_Officeno joy
14:23.20nickfennello/ looks like a little man waving...
14:23.41Malard_Officep8-pbx001*CLI> pri set intense debug span 1
14:23.42Malard_OfficeNo such command 'pri set intense debug span 1' (type 'help pri set' for other possible commands)
14:23.48nickfennellMalard_Office, id back out any dial plan conversions you have and figure out what you need to dial
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14:23.58nickfennellfor nat and inter
14:24.08WIMPyHmm, ist it already 'pri set debug 2 span 1' on that version?
14:24.15Malard_Officehow do i check
14:24.49nickfennellWell if you make sure no conversions are in place then try dialling
14:24.53nickfennellyou should get through.
14:25.01nickfennellas exactly what you dial should be sent
14:25.02Malard_Officei am the only phone
14:25.08Malard_Officei've stripped the 0|.
14:25.12Malard_Officewill pastebin what i see in asterisk
14:25.41WIMPyYou shouldn't modify the number in any way.
14:25.43Malard_Officehttp://pastebin.com/bxX2tncz
14:25.56WIMPyUnless you want an extra 0 to access the line.
14:26.13nickfennellYeah or unless you want to simulate an 80s office environment where you need to dial a 9
14:26.20Malard_Officeno i dont want to do either
14:26.33Malard_Officeand as i say, the only way i've managed to make a call work so far, is to strip the leading zero
14:26.41Malard_Officeso it sends  1733404940 to the exchange
14:26.45Malard_Officeinstead of 01733404940
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14:27.12nickfennellbut then that means you can't sent international as there's no way to identify the destination
14:27.20WIMPyA clear indication of the TON being det.
14:27.23Malard_Officeyeah i am already aware of that
14:27.28nickfennellSo something is so fubar.
14:27.29Malard_Officethe what being what?
14:27.34Malard_Officelook at my pastebin
14:27.47nickfennellI would but that requires a reboot
14:27.50nickfennellwhich I will perform nowe
14:27.51nickfennell*now
14:27.53WIMPyType Of Number
14:28.03WIMPyBut I don't see any pri debug in your PB.
14:28.15Malard_Officei pasted you the output of the command you wanted me to type
14:28.16nickfennellbby
14:28.17Malard_Officeit did'nt work
14:29.12WIMPyNeither of them?
14:29.44WIMPyThen I can't remember the syntax for 1.6. Try to use the tab key to get suggestions.
14:30.27WIMPyBut those Null Frames don't look correct, either.
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14:39.12WIMPyMalard_Office: Try without set: 'pri intense debug span 1'.
14:39.55Malard_Officep8-pbx001*CLI> pri intense debug span 1
14:39.55Malard_OfficeNo such command 'pri intense debug span 1' (type 'help pri intense' for other possible commands)
14:40.17WIMPyHmm.
14:40.29WIMPySo what does it offer after pri <tab>?
14:40.44Malard_Officep8-pbx001*CLI> pri
14:40.44Malard_Officedebug      intensive  no         set        show       unset
14:41.05WIMPyOh, so it's "intensive".
14:41.22WIMPy(or was)
14:41.23Malard_Officep8-pbx001*CLI> pri intensive debug span 1
14:41.23Malard_OfficeEnabled EXTENSIVE debugging on span 1
14:41.37WIMPyFinally
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14:42.34Malard_Officehttp://pastebin.com/Tw8rhbB3
14:43.34WIMPyIt still says national number. No good.
14:44.05WIMPyLooks like it didn't get the pridialplan=unknown.
14:44.13Malard_Officeits in that file
14:44.25Malard_Officejust pm'd it to save time
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15:09.45TomCat2if I wanted to just purchase my g.729 licenses from Digium, but not deal with the license files themselves because my PBX runs in a VM and don't want the hassle every time I change my configuration, are the g.729 binaries available at asterisk.hosting.lv stable enough for production use - or is there another suggestion?
15:12.01WIMPyOh, that's THE question you usually get stoned for.
15:12.27[TK]D-FenderTomCat2, You're suggesting paying for the licenses but using the other codec for the actual job?
15:12.37TomCat2[TK]D-Fender - correct
15:13.08[TK]D-FenderTomCat2, I would say you're legally covered so long as you can ensure you hold yourself within the # of conversions allowed
15:13.20[TK]D-Fendertomthis is "light grey" at worst in my scale
15:13.32TomCat2that's easy for me - very limited # of SIP devices here anyway
15:13.34TomCat2(2)
15:14.11TomCat2my question is more about the stability of the binaries built based on Intel IPP libraries
15:14.11[TK]D-FenderTomCat2, you're already proposing covering yourself.  I don't see this as a problem anyone would pursue
15:14.24[TK]D-FenderTomCat2, they work....
15:15.31TomCat2or if there is a better alternative for use - just anything where I won't be forced to deal with Digium if I change my hardware - which from what I understad, screws with the licenses if you use the digium implementation of g.729
15:16.11WIMPyDo they use more than the mac of eth0?
15:16.28[TK]D-FenderTomCat2, No, what you've proposed seems entirely fair for the circumstances and should work just fine
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15:18.02TomCat2WIMPy - that's a very good question - since I totally control the mac, keeping that consistent wouldn't be a problem and would make my original question moot
15:23.20*** join/#asterisk Katty (~Katty@mail.copi-rite.com)
15:24.08KattyDERP!
15:24.10TomCat2guess I can call digium about that one if they're willing to tell me
15:24.53TomCat2thanks for the info guy - I'll go with the Intel IPP-based implementation then and go get my licenses from Digium soon as I'm done figuring out how it all works
15:25.50Kattywonders if she can knit a digium logo
15:26.53*** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn)
15:27.54pznI saw in a secretary phone, many leds that shows which extensions are in use. it is possible to implement this with asterisk? which phone model can I buy that is compatible with this?
15:28.32WIMPyAlmost any
15:28.44WIMPyAs long as it has keys with LEDs that is.
15:28.47irrootpzn indeed its a BLF [Busy Lamp Field] Snom/Yealink/Polycom/Switchvox all have em
15:29.20Kattyi rather like isymphony for that, personally
15:29.35Kattyof course it'll take up a whole LCD
15:29.54Kattytelnet
15:29.58KattyNugget: TELNET
15:30.00KattyNugget: :<
15:30.10QwellKatty: I think he disabled it, like a nub.
15:30.21[TK]D-FenderKatty, 24H timeout on that script of his
15:30.22[TK]D-Fender^
15:30.30Kattyyes, fender, i am aware of that.
15:30.33_Corey_pzn: Look at the Digium D70 phone, it has what you want
15:30.41Kattyi did talk to Nugget on slashdot before i disocvered this server ;)
15:30.51Qwell_Corey_: I should make them give me commission for your phone sales.
15:30.52Kattyhas known nugget FOREVER.
15:30.56Kattyand by forever i mean...
15:31.06[TK]D-Fenderpzn, How many do you need to indicate?  What kind of use for the phone?
15:31.11Katty13 years? ^_-
15:31.21KattyNugget: holy carptail batman, i've known you for 14 years.
15:31.24[TK]D-Fenderpzn, Lighting & distance to the phone? (visibilty range requirement), etc
15:31.56KattyNugget: no...11 years.
15:31.57_Corey_Qwell: Commission?  I should be getting endorsement money
15:31.58pzn[TK]D-Fender, 10 extensions... secretary phone... it is useful if she sees who is already using the phone before trying to transfer a call
15:32.00Kattysigh, where does the time go.
15:32.19pzn_Corey_, tks, will search for that model. what is "the name of this feature" that I need?
15:32.25QwellTomCat2: Patent licenses are given only for a specific implementation.  You would be violating the G.729 patent (as well as various copyright laws..) by using that module.
15:32.29Kattymister parker
15:32.32Qwellpatents*
15:32.37QwellKatty: ikr?
15:32.45KattyQwell: HIPPO BIRDIE TWO EWE
15:32.56KattyQwell: BIPPO BIRDIE TWO EWE!!! HIPPO BIRDIE DEER EEWE!!!
15:32.59QwellI had a mug that said that when I was a kid. O.o
15:33.01KattyQwell: hippo birdie two ewe :>
15:33.25KattyQwell: so what;'re the barfday plans
15:33.33QwellKatty: drink.
15:33.37Kattyexcellent.
15:33.44Kattyany plans of getting pushed out of a helicopter?
15:33.46_Corey_pzn: Most manufacturers call what you want a "sidecar".  Snom, Aastra and Polycom also make what you want.  Digium's is nice because you don't need multiple sidecars to show more contacts
15:34.03_Corey_pzn: in fact, you don't need any because it's built-on
15:34.22QwellThat, and Digium phones are just awesome.
15:34.41Kattywait, digium phones?
15:35.03Kattyoh look at that, digium makes phones
15:35.17_Corey_lol
15:35.32pzn_Corey_, now, trying to understand a little bit more technically... My phone is extension 200. suppose phone 201 calls phone 202... then 201 and 202 are busy... since my phone is registered in SIP as 800, how does it get the status of the other phones?
15:35.34Kattyare these rebranded astra phones?
15:35.50_Corey_Katty: you've been too busy using Drano to unclog your Asterisk to notice
15:36.11Kattymore like too busy knitting a TARDIS
15:36.24WIMPypzn: Hint entries in your dialplan to map extension to devices, as for dialling.
15:36.29_Corey_pzn: It "subscribes" to the status on the other extensions...  they're not registrations per se
15:36.50Kattythey remind me of a cross between astra and toshiba phones
15:36.52_Corey_Katty: they're not rebranded or anything...  they developed them in-house
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15:36.58Kattyfancy.
15:37.15_Corey_they're priced well too
15:37.22Katty130ish?
15:37.27_Corey_less
15:37.34Kattyooh la la
15:37.52_Corey_msrp on the 2-line is 129, but that's msrp
15:37.53pzn_Corey_, is there any way to have this "busy indicators" as a webservice or app, that secretary will see at her screen?
15:38.05Katty^- isymphony
15:38.07_Corey_pzn: Sure, google FOP or FOP2
15:38.13Kattyisymphony is nicer tho >.<
15:38.22_Corey_I thought they went out of business
15:38.29Kattywhat?!
15:38.35[TK]D-Fenderpzn, Aastra 6739i is pretty nice for this.  Huge colour LCD touch-screen.
15:38.39_Corey_someone that used to work there told me just about everyone is gone
15:38.50_Corey_dunno first-hand though
15:39.01Kattyi emailed mike last week tho
15:39.03[TK]D-Fenderpzn, You get 12 softkeys per screen (can paginate multiple screens worth
15:39.48irrootwill be getting some shiny new digium phones soonest if happy will go into project working on for university voip roll out
15:40.01chuckfKatty: did mike respond?
15:40.03pzn[TK]D-Fender, wow! really big display!
15:40.13chuckf:)
15:40.25Kattyhmm
15:40.29Kattydigs through email
15:40.33[TK]D-Fenderpzn, I just re-equipped my call center with them + Plantronics wireless headsets & EHS cable
15:41.19Kattyyou're right, there was no response.
15:41.26Kattythe last email i got a response from was around 4/17
15:41.28[TK]D-Fenderpzn, live queue stats on idle like I used to do with my Polycoms, but now MUCH more presence indications, greater detail (ringing as well+), immensely more readable screen
15:41.44Katty:< :< :<
15:42.04Kattypesters sean
15:46.56QwellKatty: I'm old.  Fix it.
15:47.44*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
15:47.52Katty:<
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15:49.00KattyQwell: you are not old!
15:49.03Qwelllies
15:54.03Nuggethuggles Katty
15:54.47NuggetKatty: I hired drdink.  He's here in Houston this week working and looking for an apartment.
15:55.50QwellNugget: Is he a real doctor, like drmessano?
15:56.16KattyNugget: wow.
15:56.30KattyNugget: i've not talked to him in years >.<
15:56.50Kattywhich is ok, he was always a little cranky.
15:56.56Kattyand rightly so.
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16:08.09koffelis this correct to get correct caller id ?ext-did xxxxxxxx 2 Set ${CALLERID(num)}=${CALLERID(name)}
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16:08.47leifmadseno.O
16:08.51leifmadsenno
16:08.53WIMPyWhy do you want to set the number to the name?
16:09.01leifmadsenbeyond that, the syntax is wrong
16:09.16koffeli am using mysql realtime
16:09.22WIMPyWhat are you trying to do?
16:09.29koffeli tring to get name and number
16:09.44WIMPyTo where?
16:09.55koffelfrom trunk
16:10.02koffelwhen someone calls in
16:10.17leifmadsenyou don't have to set the callerID if you're recieving it
16:10.20WIMPyWhat do you want to do with them?
16:10.32leifmadsenkoffel: please elaborate and be clear about the problem you're trying to solve
16:10.42koffeli want to show up on my ata
16:11.16koffelwell this what happening when some one calls in it just shows name only i was tring to get the number to show up
16:11.20WIMPyHow could an ATA send a name?
16:11.44WIMPyo.O
16:11.47koffelout going works fine it incoming caller id
16:13.49koffelwhen someone calls  in it was showing there caller id name and my voip number as there number
16:14.05koffelbut i did a debug and number is in the header
16:14.34WIMPyWhere does what show the name?
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16:15.12koffelmy ata rtp300
16:15.22koffelcoreless phone
16:16.41WIMPydidn't even know such a thing existed.
16:17.05WIMPyBut it's been almost 20 years since I cared about the analogue stuff.
16:17.11koffelwell it shows up like this in the header
16:17.14TechSmurfSo I switched from trix to *now for our production phone system... everything seems to be great except CID info is not being properly transmitted to dahdi extensions while ringing
16:17.16koffelFrom: "ALOHN,CHOWS" <sip:ALOHN,CHOWS@192.168.80.1>;tag=as7ae036c8
16:17.36TechSmurfanyone run into something like this?
16:17.58irrootTechSmurf CID on analogue i assume ?
16:18.04TechSmurfnods
16:18.21TechSmurfdigium pri card -> adit 600 -> nortel cics
16:18.28irrootits a bit tricky the default is not helpfull for non americans :P
16:18.43TechSmurfWell, I'm american
16:19.03TechSmurfand I'm pretty sure it's supposed to "just work"
16:19.17Kattyi'm sorry to hear that. you have my condolences.
16:19.23TechSmurfthank you.
16:19.41TechSmurfbut that's a problem for another channel...
16:19.41WIMPyNothig "just works"
16:20.02Kattyno i'm pretty sure that's a problem for another gov't
16:20.48WIMPySo what are you getting? Nothing?
16:21.05Kattywhat are you getting? *picks random card from deck*
16:21.16Katty"dry heaving"
16:21.16TechSmurfthe nortel sets are saying "NO INFO"
16:21.31TechSmurfbut oddly, upon answering they display CID
16:21.34Kattywhat does a no op say
16:21.40[TK]D-FenderTechSmurf, And I'm pretty sure you haven't shown us a complete call with debug to look at yet.
16:21.48[TK]D-Fender~pb
16:21.48infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude.
16:21.49irrootKatty hehe you have not met our gov't :P SITA [State IT Agency] could not join a single pair color coded cable
16:21.50[TK]D-Fender^^^
16:22.01Kattysweet!
16:22.13TechSmurfwanders off to dig out a log snippet to paste
16:22.28irrootTechSmurf add a ring delay for CID possibly ?
16:22.32Kattyi wander...i wa-wa-wa-wa- wander
16:22.33Kattytoo
16:22.39Kattyto to to to to to dig out log snippets
16:23.25WIMPyWhat kind of link is it between Asterisk and the Nortel?
16:23.37koffelhere is debug log http://pastebin.com/fJK1q7sS
16:25.04TechSmurfWIMPy: digium pri -> carrier access adit 600 ata -> nortel analog
16:26.01vastinais there a way to check for the hardware echo cancellation module from software?
16:26.18vastinapossibly in dmesg or a an lspci()
16:26.24WIMPyvastina: dmesg.
16:26.33vastinawhat am i looking for in dmesg?
16:26.41WIMPyThe latest dahdi might heve it available.
16:26.53WIMPygrep for "VPM"
16:26.56sruffelldahdi_scan will also tell you.
16:29.04koffelwhimpy it even show same thing on my softphones
16:29.51WIMPyEasy enough to display stuff on a softphone.
16:30.16koffeldid u see my debug log?
16:30.58WIMPyI'm not the SPI guy.
16:31.06WIMPySIP
16:31.11koffelokay sorry
16:31.16vastinaso VPM450
16:32.35TechSmurfhttp://pastebin.com/tH8UwzZr
16:35.28*** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk)
16:36.02[TK]D-FenderTechSmurf, And you don't have any debug enabled for your DAHDI channels there
16:36.11[TK]D-FenderTechSmurf, "pri debug span X"
16:36.13[TK]D-Fenderfor BOTH ends
16:38.05TechSmurfWhy both ends?
16:38.17*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net)
16:38.30Kattywhy both ends?! *checks deck*
16:38.43Kattyfor a mating display.
16:39.52[TK]D-FenderTechSmurf, because I trust what comes in about as much as I trust what is going out.
16:39.53[TK]D-FenderZERO
16:39.57*** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809)
16:42.00TechSmurfPlease note my challenge is part of my attempt to fully understand, not to infer you're an idiot...
16:42.15*** join/#asterisk shadebob (~shadebob@41.142.254.128)
16:42.40Kattywell he IS an idiot, but that's beside the point ;)
16:42.44Kattypats [TK]D-Fender
16:43.03[TK]D-FenderUnderstand that when things aren't working it's best to throw out all assumptions and look at EVERYTHING
16:43.04TechSmurfWould not the CID info being perfectly accurate in the log files suggest the inbound side is working fine? Or the CID info displaying 100% correctly on SIP handsets?
16:43.08WIMPyWho isn't?
16:43.18Kattytrue story
16:43.23[TK]D-FenderTechSmurf, Show us the call....
16:44.36TechSmurfAnyone who actually enjoys teaching want to try answering the question?
16:44.46TechSmurfI'm not here for a quick fix.. fuck that noise.
16:44.59KattyTechSmurf: someone is trying to help you.
16:45.09carrarheh
16:45.10KattyTechSmurf: if you don't give them the info they need to help, you're not going to get very far here.
16:45.18KattyTechSmurf: same with calling into ANY tech support anywhere.
16:45.24Kattycarrar: ohai
16:45.26TechSmurfI'm not calling tech support.
16:45.28Kattycarrar: how'rechu dear.
16:45.31koffelwaiting in line lol
16:45.35TechSmurfI'm calling my damned peers
16:45.44carrardoing greAT!!
16:45.48Kattycarrar: woot!
16:45.55carrarw00t! is right!
16:46.08TechSmurfI don't want to know the fix, I want to know *why* it broke, *why* the fix, and *why* the questions that don't make sense to me.
16:46.10carrarHow YOU doin?
16:46.16Kattyi'm doin good
16:46.18Katty^_^
16:46.22carrarWoo Woo
16:46.28Kattyworkin on my tardis.
16:46.28[TK]D-FenderAnd we can't see why it broke without looking under the hood
16:46.30TechSmurfIf I wanted a tech support script-reading monkey, I'd call tech support :)
16:46.37[TK]D-FenderYuo are trying to diagnose without looking
16:46.43KattyTechSmurf: fender is one of our best.
16:46.44TechSmurf[TK]D-Fender: slur against tech support, not you.
16:46.55KattyTechSmurf: despite his... shortness and sarcasm.
16:46.59carraragrees with Katty
16:47.08Kattyand down right rudeness, at times.
16:47.09[TK]D-FenderTechSmurf, Time better spent getting what's been requested of you
16:47.21Kattypersonally, i think he needs a shock collar
16:47.22koffeltk fender can u look at my debug log and possible tell me whatz wrong
16:47.38Kattyoh boy! another one for the deck.
16:47.48Kattywhat's wrong with koffel's debug log? *digs through deck*
16:47.53[TK]D-Fender<koffel> is this correct to get correct caller id ?ext-did xxxxxxxx 2 Set ${CALLERID(num)}=${CALLERID(name)} <--- no
16:48.06Katty"the kool-aid man"
16:48.14[TK]D-Fenderkoffel, You are trying to set the number to the name... but you are REFERENCING the function instead of setting it
16:48.22nanoha-samahey Katty
16:48.33Kattyhello
16:48.37carrarsama!
16:48.38[TK]D-Fenderkoffel, You are REFERENCING the function where you shuoldn't be
16:48.38carrarwoah
16:48.49[TK]D-Fenderdangit, screen froze
16:48.49nanoha-samahm... I don't see SwK and bkw_ in here anymore
16:48.51[TK]D-Fenderdouble-sent
16:48.56*** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain)
16:49.06koffelokay so would it be callerid(num)=$callerid(num)?
16:49.19[TK]D-Fenderkoffel, Right on the first, and now wrong on the second
16:49.35[TK]D-Fenderkoffel, You seem to have completely forgotten your variable & function basics
16:49.44koffelso it just callerid(num)?
16:49.46Katty[TK]D-Fender: shock collar.
16:49.58TechSmurfhrm... asterisk seems to indicate only one span
16:50.14[TK]D-Fenderkoffel, Now the case is wrong
16:50.27TechSmurfscratches his head
16:50.36koffelback to looking it up i guess
16:51.06nanoha-samaKatty: how long has SwK and bkw_ not been in here anyways?
16:51.10Kattykoffel: tell him to give you a hint :P
16:51.16Kattynanoha-sama: oh umm. hmm.
16:51.19Kattynanoha-sama: couple years?
16:51.22[TK]D-FenderI just told him exactly what's wrong
16:51.25Kattygive or take
16:51.30[TK]D-Fendertaht isn't a"hint"
16:51.33nanoha-samahehe Katty
16:51.36[TK]D-Fenderit's the answer
16:51.42carrarTK, please add color to it
16:51.46nanoha-samaI heard [intra]lanman got banned here for trolling Katty
16:51.48Kattyyes color would be nice.
16:52.00WIMPynanoha-sama: Those nicks don;t ring a bell and I think I've been here for about 4 years now.
16:52.01Kattynanoha-sama: i line needs grammar.
16:52.06Kattynanoha-sama: i don't quite understand it
16:52.17KattyWIMPy: the original asterisk clue
16:52.24KattyWIMPy: err crew. around 1.2ish
16:52.32Kattyprobably before.
16:52.42nanoha-samahehe, yeah, they're in freeswitch now
16:52.44WIMPystarted on 1.4 beta.
16:52.44nanoha-samaruns
16:52.45carrarnanoha, is this you? http://upload.wikimedia.org/wikipedia/en/6/6d/Nanohadvd2pkg.jpg
16:52.52Kattynanoha-sama: yesh.
16:53.00nanoha-samacarrar: lol
16:53.02nanoha-samayou wish
16:53.05carrarhaha
16:53.08Kattythat's a wonderful picture.
16:53.12Kattyyou should just pretend.
16:53.14nanoha-samaI just use this so people don't ask me who I really am
16:53.21nanoha-samabecause it's an imaginary entity
16:53.23nanoha-samalol
16:53.29carrarWHO ARE YOU REALLY
16:53.29WIMPyYou are?
16:53.30koffeltkdfender i was looking on the function for asterisk 1.8
16:53.35carrarWHO ARE YOU REALLY AM
16:53.36[TK]D-Fenderstarted before they reached the other side of 2 decimal points
16:53.50koffelit says i can do exten => s,1,Set(CALLERID(num)=${CALLERID(num):0:40})
16:53.59nanoha-sama[TK]D-Fender: you'd probably heard of the rebellion then
16:54.05nanoha-samaor saw it happen as a matter of fact
16:55.21Kattyi wouldn't call it a big tramatic ordeal.
16:55.23[TK]D-Fenderkoffel, Technically correct, though I wonder what you have as CID# that you want to chop it to the first FOURTY
16:55.26Kattyerr dramatic.
16:55.32nanoha-samahehe
16:55.41Kattyi'm not sure anyone really cares anymore.
16:55.49Kattyor if they do, they get their panties in a wad quietly
16:56.05Kattynot much of it said these days.
16:56.24Kattyhmm. sweden. hmm
16:56.26carrarI CARE
16:56.30Kattystill trying to figure out who nanoha-sama
16:56.34Kattycarrar: honey badger.
16:56.40nanoha-samaKatty: aka TriJetScud on here
16:56.45Kattyah HA!
16:56.50Kattynow that rings a bell
16:56.54nanoha-samaif you've talked on the FS conf's you'd probably already know who I am
16:57.10Kattyi know everyone who's been around here for awhile
16:57.23*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
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16:57.47carrarwoot
16:57.47Kattywooter.
16:57.49nanoha-samayeah Katty like you knowing SwK owned a spongebob squarepants boxer
16:57.51carrarwooterific
16:57.58Kattyshhh
16:58.05carraroh my
16:58.11Kattythat is not spoken of in here!
16:58.21nanoha-samalol
16:58.39Kattyi swear he is never going to live that down tho.
16:59.11nanoha-samaKatty: some say he owned a pair of boxers with pictures of spongebob sparepants painted all over it but we can never be sure
16:59.12*** join/#asterisk brdude (~brdude@12.155.183.30)
16:59.30Kattymuch better ^_^
16:59.33nanoha-samasome say he used to be in the marines, but all we know... HE'S SWK!
16:59.53nanoha-samapeople who top gear get the joke above lol
17:00.49Nuggetambitious, but rubbish.
17:02.27nanoha-sama[TK]D-Fender: you got a good guide for asterisk for beginners anyways?
17:02.37[TK]D-Fender~book
17:02.37infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
17:02.50Kattyi like the fischer price book better.
17:02.52*** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com)
17:02.54Kattymy first pbx.
17:02.56vastinaon the single span TE122 dmesg will nto tell me whether or not the VPM0CT032 is available
17:03.24vastinahowever on the dual span TE205 the dmesg will tell me if the VPMOCT064 is available or not
17:03.38nanoha-samaheh, I wonder if there's books that deal with users who have lots of FreeSWITCH  background when dealing with asterisk...
17:03.41vastinaany other way I can check from software?>
17:03.46WIMPyvastina: If there;s no information in dmesg, it's not working.
17:04.07vastinaWIMPy: so I have a faulty VPM module?
17:04.12vastinaon three brand new cards?
17:04.22[TK]D-Fendernanoha-sama, Since you can't compare them.. it may as well be "for beginners".  You're in "from scratch" either way
17:04.32vastinas/module/daughterboard/
17:04.34WIMPyvastina: In the same PC?
17:04.36drmessanonanoha-sama:  The tech is the same, but the applications are worlds apart.  The End
17:04.41Kattywhat's a vpm card
17:04.45vastinaWIMPy: distinct PCs
17:04.48[TK]D-FenderkattHWEC
17:04.56Kattyah
17:05.44carrarhere you go katty https://www.osburn.com/fisher-price-network.jpg
17:05.44WIMPyvastina: There might be differences. I once couldn't get it working because I didn't use udev. (off course not. It's a server).
17:05.45carrarhaha
17:06.18vastinaWIMPy: just find it a bit weird VPM wouldn't show... i mean it's all ubuntu (hate ubuntu)
17:06.19Kattycarrar: YES
17:06.24Kattycarrar: that's PERFECT
17:06.24vastinafor the *032
17:06.32vastinabut for the *064 works
17:06.48nanoha-sama[TK]D-Fender: asterisk was designed like a PBX in mind in the first place, right?
17:07.07Kattyno it was designed to be... *digs through deck*
17:07.10WIMPyvastina: Have you tried to swap cards? Or the modules? Or just checked they're seated securely?
17:07.11[TK]D-Fender~asterisk
17:07.12infobotAsterisk is an open source telephony toolkit, or #asterisk on irc.freenode.net, or http://www.asterisk.org/
17:07.14[TK]D-Fender^
17:07.23Katty"kamikaze pilots!"
17:07.24WIMPynanoha-sama: I can't see that at all.
17:07.25vastinaWIMPy: done the basics
17:07.32[TK]D-Fendernanoha-sama, It's whatever you make it into
17:07.51[TK]D-Fendernanoha-sama, For me it's a coffee timer & jukebox.
17:08.00nanoha-samahehe
17:08.03vastinamaybe it's the wcte12xp module?
17:08.09vastinaa flaw
17:08.13TechSmurfscratches his head
17:08.29nanoha-samathen I don't see much of a difference in FS comapred to Asterisk in terms of concepts
17:08.35TechSmurfSo I called to generate log data with pri debugging, and the CID worked.
17:08.51TechSmurfbashes his head against a nearby solid object
17:09.04KattyTechSmurf: maybe you should post that debug now.
17:09.04vastinaWIMPy: i guess what I'm going on about is if there a way beyond dmesg to check? dahdi_scan doesn't check
17:09.07[TK]D-Fender~osmosis
17:09.07infobot[~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ...
17:09.10[TK]D-Fender^^
17:09.11[TK]D-Fender:D
17:09.48WIMPyvastina: So did you swap cards? Is it one card or one PC?
17:10.04nanoha-samalol [TK]D-Fender
17:10.05vastinaWIMPy: i work for a manu... i have about 4 boxes to check on the fly
17:10.19nanoha-samayeah, I don't see much of a difference between the two now
17:10.46nanoha-samashit, this "same shit, different names" disease is getting through me
17:10.52vastinadahdi_scan shows no pertinent info
17:11.00vastinaregarding HWEC
17:11.15TechSmurfsadly, pasting a log of a perfectly working example doesn't seem helpful
17:11.24WIMPyvastina: I do see the information there.
17:11.44WIMPydevicetype=
17:12.05TechSmurfhowever, it gives me a great baseline comparison for the not-working example should I manage to generate one
17:12.53*** join/#asterisk brdude (~brdude@12.155.183.30)
17:12.57vastinaWIMPy: devicetype=Wildcard TE122
17:13.02[TK]D-FenderTechSmurf, That's right.  If life hands you lemons you ask for TEQUILA AND SALT.
17:13.04vastinadirect copied
17:13.22vastinasays nothing of HWEC
17:13.27WIMPyvastina: No module.
17:14.02WIMPygets devicetype=Wildcard TE220 (4th Gen) (VPMOCT064)
17:14.15vastinathe VPMOCT064 is a dual span...
17:14.26vastinathe VPMOCT032 is my target
17:14.32WIMPySo is the base card.
17:14.36vastinai've already tested the dual
17:14.36*** join/#asterisk tris (tristan@camel.ethereal.net)
17:14.44vastinayes I'm aware
17:14.53carrarhi tris
17:14.57vastinaI'm concerned with the TE122, ! the TE20*
17:15.14vastinai have both cards on a few systems
17:15.29vastinamaybe the VPMOCT064 is just more responsive?
17:15.30vastinahaha
17:15.46WIMPyUnlikely
17:16.29vastinaWIMPy: unfortunately you're asking me as though I don't know how to troubleshoot, I have done my troubleshooting and have several machines in my testbed, just not getting the VPMOCT032 to respond is all
17:16.34vastinahave a problem in the field
17:16.51vastinawondering if there's a way outside of dmesg
17:16.55vastinaor dahdi_scan
17:17.04vastinashrugs
17:17.13vastinaI appreciate your help, really
17:17.34WIMPyI'm pretty sure it's not working if you don't see it working in dmesg.
17:17.46WIMPyBut maybe you should contact Digium.
17:17.51vastinaon several machines?
17:18.29vastinayeah maybe I should... but again I appreciate your help, I can repeat your results on the VPMOCT064
17:18.31WIMPyMaybe there was a bad dahdi version?
17:18.36WIMPyI have no idea.
17:18.47newtonrvastina: you should contact Digium support for that issue, the board products include support
17:18.48vastinait's all good, I'm grateful for the help
17:18.57vastinanewtonr: i'm aware :)
17:19.06newtonrnow you are double-aware!
17:19.20vastinalucky me
17:19.43WIMPyAn awareness backup :-)
17:20.17*** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file)
17:20.18*** mode/#asterisk [+o file] by ChanServ
17:21.56mcf3782sfuffell:  Thanks for the info on those parts for my TDM400. My Digi-Key order showed up yesterday. Replaced those two SMD chokes and that channel came right back online. :-)
17:23.47TechSmurf[TK]D-Fender: http://pastebin.com/H6HVLNed
17:24.03TechSmurf[TK]D-Fender: First call, CID worked. Second call, CID failed
17:24.31sruffellmcf3782: woot!
17:25.29koffelpulling my hair can't figure out this damn caller id
17:25.47mcf3782Indeed! :)
17:27.20[TK]D-FenderTechSmurf, You don't have debug on the span that the call is going OUT over so we can't se what & when * sends anything over
17:27.29[TK]D-FenderTechSmurf, Like I said... BO TH SIDES
17:28.07TechSmurfhrm
17:28.17TechSmurfyeah, I was just noticing that in the log
17:28.34TechSmurfsee, problem is as I said earlier asterisk is only showing one span
17:29.36*** join/#asterisk tris (tristan@camel.ethereal.net)
17:30.30vastinatris: name any reference to 5th element? haha
17:30.45koffeltkdfender if i remove callerid string completely would asterisk still work?
17:30.57*** join/#asterisk xaionaro (~xaionaro@nat.mephi.ru)
17:32.27[TK]D-Fenderkoffel, ....that is a very vague question and you ahven't told us the scope of what you're changing.
17:33.01[TK]D-Fenderkoffel, Do you think all of * dies if you change a callerid?  The question is : what cares about it?
17:33.49TechSmurfstares at asterisk like it's crazy
17:33.53koffeljust changed one line in ext to get incoming caller id to work
17:34.14[TK]D-Fenderkoffel, And does it?
17:34.31WIMPyTechSmurf: So where's your number to call you maybe?
17:34.47leifmadsenkoffel: what did you change?
17:35.16koffelths line
17:35.20koffel${CALLERID(num)}=${CALLERID(name)}
17:35.26leifmadsento what....
17:35.39leifmadsenthat line alone is invalid
17:35.46leifmadsenplus the syntax is also wrong
17:35.47koffelto callerid(all)=$callerid(name)
17:35.52leifmadsenthat's wrong
17:35.54leifmadsenas well
17:35.56WIMPyNot on;y alone.
17:36.02koffelcan u tell me correct one
17:36.23leifmadsensame => n,Set(CALLERID(num)=${CALLERID(name)})
17:36.28leifmadsen1) functions must be uppercase
17:36.37leifmadsen2) setting functions does not use ${ ... } surrounds
17:36.47leifmadsen3) reading functions uses ${ ... }
17:37.03WIMPyAnd the fender already said so.
17:37.03leifmadsen4) what you're doing is kind of illogical, but ok....
17:37.10leifmadsenWIMPy: I can't see anything he says
17:37.29p3nguinStop trying to screw up the caller id name and number before it reaches your phone.
17:37.33p3nguinJust stop it.
17:37.41p3nguinThen it will work better.
17:37.56koffelleifmadsen so it (CALLERID(ALL)=$(CALLLERID(NAME)
17:37.56[TK]D-Fenderkoffel, i already told what was wrong in EACH half of that
17:37.59[TK]D-Fenderkoffel, twice
17:38.03leifmadsenno
17:38.14leifmadsennot even close...
17:38.22leifmadsenI literally typed out a working example
17:38.57koffelbut then u say not to use {
17:39.01leifmadsenright
17:39.07leifmadsenI didn't say replace { with (
17:39.08p3nguin(1236.23) <@leifmadsen> same => n,Set(CALLERID(num)=${CALLERID(name)})
17:39.41p3nguinAnd then I said stop trying to mess with the caller id info before it comes to your phone.
17:39.48koffelso then it would be  n,Set(CALLERID(num)=$(CALLERID(name)))
17:39.55[TK]D-FenderNO
17:39.56leifmadsenfacepalms
17:39.57p3nguinJust let it show what it actually is.
17:40.00WIMPywow
17:40.08[TK]D-Fenderkoffel, LOOK AT THE BRACES
17:40.08leifmadsenkoffel: good luck to you
17:40.26p3nguinIf the caller id exists, why are you trying to ruin it before it reaches your phone?
17:40.29koffelok
17:40.30p3nguinJust stop it.`
17:40.56koffeltkdfender like this n,Set(CALLERID(num)=$CALLERID(name))
17:41.01[TK]D-FenderMP
17:41.07leifmadsenand what p3nguin says is true -- you don't actually need to set anything in the dialplan to pass the callerID
17:41.11leifmadsenkoffel: did you look at my example?
17:41.14[TK]D-Fenderkoffel, he gave you the complete answer 3 TIMES
17:41.28leifmadsenSet(CALLERID(num)=some value)
17:41.38p3nguinPutting the name in the number field is pretty ridiculous in itself.
17:41.44leifmadsenNoOp(Output value of CallerID number:  ${CALLERID(num)})
17:41.47leifmadsenp3nguin: +1
17:41.49koffelyeah but what value that darn thing
17:41.53leifmadsengoes back to things that matter
17:41.53WIMPyYes, it's going to be hard to call back a name form a phone connected to an ATA.
17:42.02p3nguinJust leave it be.  Stop messing with it.
17:42.12p3nguinThe caller id will show whatever it actually is.
17:42.22koffelp3nguin i would but don't work
17:43.00[TK]D-FenderThis is just not worth the trouble guys.....
17:43.10WIMPykoffel: Maybe you should take a look at the configuration of that ATA?
17:43.11TechSmurfnods
17:43.12[TK]D-FenderSpare yourselves the aneurism
17:43.13p3nguinInstead of saying it doesn't work, actually prove that it doesn't work.  Check the CALLERID() values with Verbose() or NoOp() in the dial plan.
17:43.30*** join/#asterisk kleszcz (tick@176.111.232.12)
17:43.30p3nguin(1241.43) <@leifmadsen> NoOp(Output value of CallerID number:  ${CALLERID(num)})
17:43.34TechSmurfWHat we have here is a clear-cut case of why I stopped wanting to do chat support.
17:43.56[TK]D-Fenderp3nguin, what doesn't work is the fact he keeps reintroducing syntax errors after a complete and proper sytax version is handed to him over 3 times.
17:43.58TechSmurfIt's even painful to watch...
17:44.03WIMPyTechSmurf: Seriousely, imagine that on the phone...
17:44.20TechSmurfWIMPy: do that all the time, but for money :P
17:44.33p3nguinSo what you're saying is that koffel doesn't work, but CALLERID() probably does.
17:44.39WIMPyI hope it lots of money then.
17:44.43TechSmurfand I reserve the right to tell my users they're fucked
17:44.57pabelangershould include = foo with within a dialplan?
17:45.00[TK]D-Fenderp3nguin, Correct
17:45.00*** join/#asterisk luckman212 (~irc@2001:470:8abb:0:3839:fb97:cb77:8968)
17:45.04pabelangervs include => foo
17:45.30[TK]D-Fenderpabelanger, Both should work
17:45.46koffelp3 i do have noop
17:46.03koffeland all but they all say same darn thing
17:46.24p3nguinIf you have the NoOp() and did not fiddle with the caller id info before it, what does it say?
17:46.48koffelext-did xxxxxxxx 3 Noop Incoming call from ${CALLERID(num)}
17:47.08p3nguinI have no idea what that even means.
17:47.09koffelonly one line i changed that was one after it
17:47.32koffelthat one you guys are yelling at me for
17:47.51p3nguinI don't speak "incomprehensible."  I only speak Asterisk dial plan.
17:48.04koffelp3 it in mysql
17:48.04[TK]D-Fenderp3nguin, DB realtime <-
17:48.59koffeltkdfnder is that last line i pasted wrong too
17:49.06p3nguin${CALLERID(num)} is a valid function which would show the caller id number.  What does that produce when making a call to that extension?
17:49.29[TK]D-Fenderkoffel, assuming your spaces VS colums are right, then yes that last one should work
17:49.41koffelwell okay
17:49.47p3nguinSo what does it show?
17:50.10koffelincoming calls show correct name but number as my voip
17:50.14koffelnumber
17:50.25[TK]D-Fenderhuh?
17:50.28p3nguinI would have used Verbose() instead of NoOp(), but the value can still be retrieved in the case of using NoOp().
17:50.32koffelp3 here is debug call log
17:50.36koffelhttp://pastebin.com/fJK1q7sS
17:50.41[TK]D-Fenderplease reword that into something... legible
17:50.57p3nguinI have a feeling someone hard-coded the callerid value on the SIP peer.
17:51.26[TK]D-Fenderkoffel, I don't see it in there at all
17:52.01koffeltkd want me to screen shot db ext
17:52.22[TK]D-Fenderkoffel, the PB you just provided shows NOTHING
17:52.44[TK]D-Fenderkoffel, You are also not even showing basic VERBOSE at CLI
17:53.28[TK]D-Fenderkoffel, And do not waste our time with another call where you scrub out all of the pertinent information.
17:53.31koffel6kay so u don't want sip debug u want core debug
17:53.42[TK]D-Fenderkoffel, No, I want SIP + VERBOSE
17:54.20koffelokay i just don't want my voip number pasted on the net that al
17:54.32koffelthat only info i scrubbed
17:54.43[TK]D-Fenderkoffel, Last chance on this.
17:57.23koffeltkdfender i have verbise log and sip debug log
17:57.43koffeli will paste them can i xxx out my voip number
17:58.20koffel?
17:58.24p3nguinDon't waste your only remaining chance.
17:58.50*** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com)
17:59.02p3nguinWe don't care what your phone number is, but we do care what numbers are being tossed around in the channel.  They are RELEVANT.
17:59.47koffeli know but last time i pasted my voip number in log in here i had 1 am phone calls
18:00.04p3nguinYou should have expired your pastebin.
18:00.26p3nguinUsing NEVER for an expirating makes it hang around in there for people to find later.
18:00.27WIMPyI never got any calls after pasting my number.
18:00.41WIMPyBut then it isn't hard to find anyway.
18:00.41p3nguins/expirating/expiration/
18:01.09p3nguinYou can also paste it as an unlisted paste so it does not show up in the recent pastes list.
18:01.58p3nguinUnlisted AND a relatively short expire time are good ways to not have your super-secret phone number floating around in the intar webz.
18:02.12koffelhttp://pastebin.com/tM6Wz3Gw
18:03.14[TK]D-Fender[Jul 17 13:56:44] VERBOSE[507] pbx_realtime.c:     -- Executing [2156605225@from-trunk:2] Set("SIP/2156605225-0000001e", "2156605225=ALOHN,CHOWS") <- BORKEN
18:03.17[TK]D-FenderBROKEN even
18:03.30[TK]D-Fender[Jul 17 13:56:44] VERBOSE[507] pbx_realtime.c:     -- Executing [2156605225@from-trunk:3] Noop("SIP/2156605225-0000001e", "Incoming call from 2156605225") <- This one at least did it's job
18:04.26p3nguinits job
18:04.33*** join/#asterisk eva_02 (~jsc@83.170.99.222)
18:05.04[TK]D-Fender[Jul 17 13:56:44] VERBOSE[507] pbx_realtime.c:     -- Executing [2156605225@from-trunk:5] Goto("SIP/2156605225-0000001e", "custom-linuxmce,101,1") <- You must be that one twit using LinuxMCE that left n3glv traumatized and neurotic about everyone in #freepbx as being suspect users of it.
18:05.26drmessanoHAHAHA
18:05.39p3nguinThe problem still seems like you're setting the caller id number to your own number instead of leaving it be.
18:05.55*** join/#asterisk timahvo1 (~rogue@196.200.32.36)
18:06.22drmessanoALOHN CHOWS sounds like a good name for a Chinese restaurant for single folks
18:06.29drmessanoSmall servings
18:06.33[TK]D-Fenderthat broken set shows you still can't read and are referencing a varible as the TARGET of your Set()
18:06.35eva_02how to show peers with Display Name that users type in sip client?
18:06.47eva_02sip show peers - shows only hostnames
18:06.50[TK]D-Fenderdrmessano,  4eva!
18:07.14p3nguinsip show peers shows the peer names.
18:07.18p3nguinthat is all.
18:07.33p3nguinhint:  left column
18:07.39WIMPyAnd the username
18:07.40koffelp3nguin so i do i fix that
18:07.59p3nguinStop trying to Set() the caller id info.
18:08.07p3nguinYou don't need to set it.  It already exists.
18:08.10p3nguinLeave it be.
18:08.12koffelso delete that line
18:08.27p3nguinMaybe.  I didn't see your dial plan.
18:10.30eva_02p3nguin, how I can see peers callerid(name)?
18:11.10p3nguinThe name set in the callerid value of the peer?
18:11.20WIMPyeva_02: sip show peer <peer>
18:11.24p3nguinsip show peer <peer name>
18:11.34p3nguinThat will show the details of the peer entry.
18:11.36eva_02i need list all
18:12.05WIMPyMaybe you should take a look at your configuration then?
18:12.37eva_02list all online peers with names
18:12.43koffelscrew it don't really need caller id
18:14.02eva_02also "sip show peer <name>" not show Display Name that user type in his sip client.
18:14.18p3nguinOf course it doesn't.
18:14.44eva_02it is possible to view it on asterisk side?
18:15.02eva_02or it sended only during the active call?
18:15.02p3nguinThe sip debug will show it per call.
18:15.05WIMPyOnly on an active call from that peer.
18:15.12WIMPyIf you accept it, that is.
18:15.17eva_02thank you
18:16.23eva_02how to show only online peers?
18:16.29leifmadsenyou don't
18:16.32*** join/#asterisk navaismo (~navaismo@189.144.234.127)
18:16.49leifmadsenyou show all configured peers
18:16.50p3nguinYou could enable qualify for all of them, then parse the output.
18:17.05p3nguinThat would show which ones are online.
18:17.23p3nguin(or were perceived to be online at the last qualify)
18:17.27leifmadsenor at least which ones responded within the last X ms :)
18:17.30leifmadsen:)
18:21.51navaismoHi, the DPMA only work on LAN? I tried a phone via VPN but cant contact them i think because the mac trough routers right?
18:22.47leifmadsenI would be surprised if provisioning via the DPMA was at layer 2
18:22.49leifmadsenbut I've never used it
18:22.50_Corey_navaismo: I use DPMA over WAN links...
18:23.13leifmadsenI suspect the MAC is just used as an identifier, not actually layer 2 communications
18:23.50_Corey_navaismo: Make sure port 80 is reachable
18:25.36Kattyugg, want to sell 1 headache.
18:25.38Kattywill deliver
18:26.06navaismoThanks leifmadsen and _Corey_
18:27.46Qwell_Corey_: It only uses 80 for firmware
18:27.48Qwelljust FYI
18:27.55_Corey_Ah, good to know thx
18:27.57Qwell(and cool stuff in 1.2, but I digress)
18:28.30eva_02how sip client's check if user in contact list online?
18:28.57Qwellwonders if _Corey_ knows about 1.2 yet
18:29.05*** join/#asterisk jsman (~jsman@unaffiliated/jsman)
18:29.42leifmadseneva_02: that's done via device state
18:30.27eva_02where i can read about this? because it not work in eyeBeam and my asterisk
18:30.32eva_02all users shows offline
18:30.38Kattyglomps Qwell
18:30.48navaismois there a log like messages where i can see the events of DPMA? Im reading the wiki but cant see something like that. Im receiving this message: Phone 'Gordon's Phone' has not contacted the server.  The server is unaware of the location of this device and can not reconfigure it at this time.
18:31.02[TK]D-Fendereva_02, Go set your HINT's  Dialplan 101
18:31.05[TK]D-Fender~book
18:31.05infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
18:31.06[TK]D-Fender^^
18:32.09*** join/#asterisk fexilal (~felixal@unaffiliated/felixal)
18:32.14fexilalHi
18:32.26_Corey_Qwell: Not so much...!  I'm going to have to hassle malcolmd
18:33.56malcolmdwho, what, where?
18:34.14malcolmddocs aren't ready yet, QA's still doing their work
18:34.15*** join/#asterisk brainiac (~Brainiac@12.52.94.138)
18:34.16fexilalanyone here dealt with 25-pair cables?
18:34.35*** join/#asterisk timahvo1 (~rogue@196.200.32.36)
18:34.47malcolmdQA tells me they're close
18:34.56QwellThen, beer!
18:35.05vastinabeer is the answer
18:35.14QwellMalard_Office: Maybe even *two* beers.
18:35.16Qwellerr, malcolmd
18:35.28Qwell(one per version?)
18:35.29[TK]D-Fenderfexilal, I've owned all sorts of old printers and modems.  Fire away....
18:35.43_Corey_blows a layer of dust off his beta lab
18:35.44malcolmdQwell: ? :)
18:35.58_Corey_that'd be 10 beers, right?
18:36.09malcolmdQwell: oh, right, then beer, yes :)
18:36.12Qwell_Corey_: it should be done real soon now
18:36.41malcolmdQwell: i've been off beer since i took ill a couple of weeks ago, i've nearly forgotten it :(
18:38.37fexilal[TK]D-Fender : Use any memory tricks to remember the color codes of the pairs?
18:38.58[TK]D-Fenderfexilal, Printed sheet <-
18:39.15fexilalOh, yeah..
18:40.32_Corey_malcolmd: I recommend whiskey to kill all germs as a preventive measure
18:40.35Kattyperks up
18:40.37Kattybeer?
18:40.59Kattycould go for a leinenkugel right about now
18:41.10_Corey_mmm, fruity beer
18:41.15Kattymmmm, yes.
18:41.25Kattydo you have a favorite?
18:41.27p3nguinSurely fruit is better than flower.
18:41.53malcolmd_Corey_: i hit this one with a cocktail of pills; didn't seem to phase it
18:46.31_Corey_malcolmd: Ouch...  yeah, I went through that two weeks ago.  (the whiskey as a preventive measure is touchy...  regular dosages keep the germs away, too much "medicine" and the immune system is compromised.)
18:47.07Qwelltoo much medicine, and you don't really care :p
18:47.30_Corey_not at the time of medication anyway
18:47.40_Corey_:)
18:50.58malcolmdi like my medicine served draught
18:52.04*** join/#asterisk timahvo1 (~rogue@196.200.32.36)
18:54.11*** part/#asterisk TheMan (~garry@nv-208-13-131-232.sta.embarqhsd.net)
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18:59.03*** join/#asterisk nutxase (~nutxase@197.87.61.80)
18:59.13gustonaaa
18:59.14nutxaseany b410p gurus here?
19:01.54*** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan)
19:03.43*** join/#asterisk timahvo1 (~rogue@196.200.32.36)
19:03.52leifmadsen~asl
19:03.52infobotOld enough, yes please, my place.
19:03.55leifmadsenbah
19:03.56leifmadsen~ask
19:03.56infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
19:04.11[TK]D-Fenderpwned
19:04.23nutxasedahdi keeps dropping calls and misdn i cant not get working on centos 6
19:04.30[TK]D-Fender~pb
19:04.30infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude.
19:04.31[TK]D-Fender^^^^^^^6
19:04.41leifmadsennutxase: packages or source?
19:04.48nutxasesource
19:04.54leifmadsennutxase: selinux disabled?
19:04.58nutxaseyes
19:05.04leifmadsenis out of ideas
19:05.17nutxasewhere can i get misdn rpms'
19:06.06gustoselinux is sth bad
19:06.13newtonrnutxase: the B410P includes support from Digium, feel free to give them a call
19:06.18p3nguinWho is Seth Bad?
19:07.08nutxasetried they were "reluctant" to help
19:07.43*** join/#asterisk cyborg-one (1000@188-115-140-195.broadband.tenet.odessa.ua)
19:08.34newtonrnewtonr: thats cause mISDN is no longer supporter with that card, the versions that work with it are pretty old
19:08.43newtonr*supported
19:08.49newtonrnutxase: ^
19:09.40nutxasehmm any idea why dahdi would drop calls then
19:09.50newtonrwith recent versions of Asterisk you can use the DAHDI driver for it, wcb4xxp
19:10.01nutxaseim using 1.8.14
19:10.34newtonryou can use wcb4xxp with that version
19:10.42nutxasethats what i do use :/
19:11.17newtonrokay, thought you said you were using mISDN, looks like you were just looking for mISDN
19:11.34nutxaseyea im looking to try get misdn to work on centos 6 :/
19:11.34newtonrDigium support should have no issue helping you on wcb4xxp
19:12.23newtonrI haven't used misdn in years, I won't be of any help :(
19:12.43nutxasewell maybe u can help me get digium working fine ;)
19:14.04nutxasehangup cause 27 all the time
19:14.27newtonrif you post debug in pastebin, someone here will likely look at it.   If you give me the case number for the Digium support issue, I can ping someone to make sure they understand the issue
19:14.54nutxasedo u work there
19:15.20newtonryes, i'll PM you
19:15.23nutxasekk
19:28.13McBoingBotrying to get some different ringtones into our Polycom phones, boot log showing ftp is copying the new wav to the phone but it will not play any sound whether the actual filename shows up on the phone or not (hit or miss), I have been putting the wav files into the tftp directory with appropriate permissions and I am fairly certasin the formats are correct, where do I start checking for problems?
19:29.32jpsharpLook at the logs the Polycom phone tries to upload onto the TFTP server?
19:29.37McBoingBoyes
19:30.05McBoingBosometimes the wav filename makes it into the phone and sometimes now, still have not determined what it is exactly, but always no sound
19:30.12McBoingBonow = not
19:31.24McBoingBoto try to eliminate the format as an issue I am trying to use the LoudRinger.wav that comes with SIP app for phone, and still nothing even though log shows "0717152312|copy |3|00|'ftp://PlcmSpIp:****@192.168.129.254/LoudRing.wav' from '192.168.129.254'"
19:33.38McBoingBorunning file LoudRinger.wav gives me "LoudRing.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz" so its the right format.....meh
19:41.14[TK]D-FenderFunny...
19:41.19[TK]D-Fenderyou say you're using TFTP
19:41.36[TK]D-Fenderand that is blatantly telling you **FTP**
19:43.38*** join/#asterisk adeel|work (~adeel@216.183.80.220)
19:51.20TechSmurfOk, so span 2 is fxo_ks, not pri
19:51.40McBoingBohilarious! LOLOL
19:53.37McBoingBoDont you think thats kind of "trivial", obviously not my problem
19:57.01*** join/#asterisk TSM2 (~the_softw@78-105-6-158.zone3.bethere.co.uk)
19:58.39[TK]D-FenderTechSmurf, Dumb analog \o/
19:59.35*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)
19:59.40carrartftp ftp rtp smtp all the same TK!!
19:59.41TechSmurfbut it works just fine with trixbox, and the issue is intermittent. really points at configuration over hardware
20:00.00*** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
20:00.00*** mode/#asterisk [+o sruffell] by ChanServ
20:00.30McBoingBosigh
20:01.16*** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca)
20:01.34*** join/#asterisk slingr (santas@will.one.day.hack-the-pla.net)
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20:05.27*** join/#asterisk gringo (~gringo@h195202184107.med.cm.kabsi.at)
20:05.27[TK]D-FenderOr the fact the drivers haev all changed
20:05.36[TK]D-FenderTrixbox died YEARS ago.
20:06.43carrarif only there was a way for other trixbox users to talk to each other
20:07.18[TK]D-FenderYou mean "the voices" aren't multi-cast?
20:08.00carrarsure would make listening in alot more easier :)
20:08.16drmessanocarrar: If someone would get access to fonality's DNS, you could redirect all the data sent from the heartbeat application to a portal where we could all share love
20:08.28Kattyprods carrar
20:08.42Kattycarrar: i very much enjoyed reading the lists of things you like and don't like.
20:08.48carrarbends over a little closer to Katty, AGAIN PLS
20:08.48drmessano"Anonymous call data"  <--- Lies
20:09.01Kattycarrar: and mean people are definately at the top of my dislike list!
20:09.01carrarheh
20:09.06carraryeah!
20:09.18Kattyalong with slow download speeds.
20:09.27carrarheh
20:10.23carrarand slow upload speeds!
20:10.30KattyYES!
20:11.13*** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax)
20:11.23carrartrying upload a 10 min 1080 video over DSL, come back next week when it's done!
20:11.32Katty>.<
20:13.18*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
20:13.28Kattyi want a gordiate :<
20:13.39carrarand thanks for reading my web pages!
20:13.49Katty:>
20:14.00carrarI should update those 2007 pics
20:14.34Kattyyes!
20:14.37Kattyand make goofy faces.
20:14.40Kattygoofy faces are the BEST
20:14.42carrarheh
20:14.56Kattymy last one i had a laptop sleeve on my head.
20:15.52carrarhaha
20:15.59carrarthe privacy laptop sock?
20:16.14carrarhttp://www.instructables.com/id/Laptop-Compubody-Sock/
20:16.18carraris that you?
20:16.37carrarI KNEW THAT WAS YOU!!!
20:16.47Kattyteehee, no
20:17.05carraroh
20:17.13Katty-->
20:18.52Kattyno sharing!
20:18.58carrarsharing what
20:19.01Kattyremember, i am a 50yr old male from the basement.
20:19.21Kattylet's not mess up my reputation.
20:19.26carrarand I am a 18 single girl naked in front og my machine
20:19.32Kattyexcellent.
20:20.05Kattyi'm working on a tardis laptop now
20:20.13carraras in time machine?
20:20.21carrarerr time machine laptop
20:20.29Kattyyes. it's bigger on the inside.
20:20.33carrarhaha
20:20.35Kattyinside it has the internets.
20:20.41carrarwoah
20:21.03carrarmust have to use two hard drives
20:21.12Kattyyes. i used ALL the drives.
20:21.36carrarALL DA DRIVES
20:21.40carrarADD technology
20:21.40Kattyi will publish the pattern so you can see it
20:22.35carrar"secret algorithm pattern"
20:22.52carrarADD with SAP
20:24.33Kattyhttp://42ndknitstreet.blogspot.com/2012/04/tardis-laptop-sleeve.html
20:24.56Kattyyoul'l have to imagine that the 'k' at the bottom isn't there. that's how far i've gotten.
20:26.01carraroh I think you should put the redhead on the laptop
20:26.02Kattyyou'll have to pretend to fold it in half too, the winders bit is the front
20:26.17carrarto distracted to see anything else on that web page
20:26.50gustowhat sap?
20:28.24carrarOh there is a tardis also on that web page
20:28.43Katty:>
20:32.13carrarKatty, you should do a ATARI  pattern
20:32.23carrarthe logo
20:32.58carrarhttp://www.salzmafia.com/blog/wp-content/uploads/2010/11/atari_logo.jpg
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20:39.12Kattyhmm
20:39.44Kattythinkers
20:39.49Kattytinkers
20:39.51QwellKatty: come drink beer, kthx.
20:39.59Kattybutbut atari logo
20:40.07Kattybutbut knitting
20:41.38Kattyah HA!
20:42.30Kattycarrar: FRESH KNITS. http://42ndknitstreet.blogspot.com/2012/07/atari-logo.html
20:42.54Qwellicky, use a cleaner source file
20:43.09carrarheh
20:43.11Kattyhttp://i3.squidoocdn.com/resize/squidoo_images/590/draft_lens15310211module132721461photo_128950280351Og-Jj-ebL._SL500_SX300_ <- look a pattern!
20:43.55QwellKatty: http://upload.wikimedia.org/wikipedia/en/7/7a/Atari_old_logo.svg
20:44.27Kattythat I is 5 stitches wide.
20:44.35Kattythat'd be HUGE
20:44.43Qwelldoesn't have to be!
20:44.51Kattyyes i could use that hat pattern
20:44.51Qwell(it's an svg, scale away!)
20:45.01Katty1 stitch = 1 pixel
20:46.53carrarDon't you do retina quality stitches?
20:47.00Katty>.<
20:47.04carrarheh
20:47.06Kattyit would be the size of a blanket!!!
20:47.15Katty..wait.
20:47.17Kattyi could be ok with this.
20:47.21carrarother way around
20:47.35Kattyi could be okay with an atari blanket tho
20:48.16carrar1 pixel would be 264 stitches per inch
20:48.21carrarerr
20:48.34carrar1 stitch would 1/264 of a inch
20:48.53Kattyi think that would be thread, not yarn
20:48.59carrarheh
20:49.06carrarcut by LASERS!
20:50.46Kattydid you see the mario blanket?
20:51.53*** join/#asterisk cyborg-one (1000@79-140-0-96.broadband.tenet.odessa.ua)
20:52.51*** join/#asterisk pigpen (~mark@fw.seamans.cc)
20:54.14*** join/#asterisk Syrex (~syrex@dsl-165-146-17-215.telkomadsl.co.za)
20:56.55*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
21:01.38eva_02Can I dial to few queues at once? Like this Queue(oneq,secondq)?
21:02.05QwellWhy would you?
21:03.36eva_02i have IVR:  1 - call sales, 2 - call support, 3 - call everyone
21:03.43eva_02how better make this?
21:03.58*** join/#asterisk cusco_ (~tralala@a79-168-182-209.cpe.netcabo.pt)
21:03.59*** join/#asterisk uskerine (~Uske@200.Red-2-138-161.dynamicIP.rima-tde.net)
21:04.07uskerinehi, i am trying to playback a .wav file without success
21:04.23uskerinei put the wav file under /var/lib/asterisk/sounds/
21:04.35uskerinethen in dialplan
21:04.37uskerineexten => s,n,Playback(welcome1.wav)
21:04.43uskerinebut i get the following message:
21:04.49junminremove .wav and try
21:04.51Qwelldon't include the .wav
21:04.57uskerine[Jul 17 23:02:12] WARNING[14326]: file.c:663 ast_openstream_full: File welcome1.wav does not exist in any format
21:05.32cusco_uskerine: read what junmin and Qwell said
21:05.33uskerinesame error
21:05.38cusco_permissons
21:05.47uskerinecan be "read" for all
21:05.56cusco_make sure the user running  asterisk can 'execute' it ?
21:05.57uskerine[Jul 17 23:02:12] WARNING[14326]: file.c:663 ast_openstream_full: File welcome1.wav does not exist in any format
21:05.58[TK]D-Fenderremove the extension
21:06.03uskerine-rw-r--r--
21:06.15uskerinefrom the file ?
21:06.21[TK]D-Fenderno, your APP CALL
21:06.22cusco_from the playback cmd
21:06.23Qwelluskerine: That is the same warning as before.  You did not change anything.
21:06.30uskerinei did
21:06.34uskerineand i did a dialplan reload
21:06.44QwellWhat did you change it to?
21:06.55uskerine-rw-r--r--
21:07.00uskerineexten => s,n,Playback(welcome1)
21:07.16QwellIs it in the correct format?
21:07.25uskerinels -l /var/lib/asterisk/sounds/
21:07.30[TK]D-Fenderuskerine: pastebin the complete call attempt, and "ls -la /var/lib/asterisk/sounds/"
21:07.32uskerinewelcome1.wav
21:07.45uskerine-rw-r--r-- 1 root     root     319596 Jul 17 22:59 welcome1.wav
21:07.50[TK]D-Fender~pb
21:07.50infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/; or install pastebinit with yum or aptitude.
21:07.53[TK]D-Fender^^
21:07.55[TK]D-Fenderpastebin them
21:07.57[TK]D-FenderBOTH
21:07.58[TK]D-Fenderin full
21:08.04[TK]D-Fenderincluding the commands issued
21:08.08*** part/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
21:08.10[TK]D-Fender</kirk>
21:08.17uskerinehttp://pastebin.com/raw.php?i=R5H0JSDg
21:09.33uskerinehttp://pastebin.com/raw.php?i=xGWD1KTi
21:10.13uskerinei tried also to do "show file formats" in the CLI
21:10.16uskerinebut the command is not recognized
21:10.29uskerineusing an emmbedded message works
21:10.42[TK]D-Fender-rw-r--r-- 1 root     root     319596 Jul 17 22:59 welcome1.wav
21:10.44[TK]D-FenderROOT
21:10.56uskerinebut it is readable by "all"
21:10.56[TK]D-Fendermaybe .. just MAYBE ... you should make the file owned by asterisk like the rest
21:11.00uskerineok
21:11.11uskerinei will do it, but it should be read by asterisk user anyway
21:11.47*** join/#asterisk nny (~Scott@174.107.223.14)
21:11.52uskerinesame error
21:12.02uskerinemaybe wav files are not loaded in asterisk?
21:12.06uskerinei mean as file format
21:12.16[TK]D-FenderAnd what are the precise specs on the file?
21:16.39nnyi am trying to use Set(CDR(myfield)=something) in the dialplan but the custom fields won't populate (I have created them in the cdr database). What am I missing?
21:17.13uskerine[Jul 17 23:16:37] WARNING[14357]: file.c:958 ast_streamfile: Unable to open welcome2.wav (format 0x4 (ulaw)): No such file or directory
21:17.15uskerinestill same error
21:17.24uskerinei have tried with and without .wav
21:17.29uskerineand i have tried also to convert the wav
21:17.34uskerinehttp://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk
21:18.27uskerinewhy 0x4 ulaw format appears there
21:18.29uskerine?
21:18.32[TK]D-FenderI guess I'm not going to get a straight answer...
21:19.45nnyhmm is custom fields still not implemented in 1.8?
21:20.24uskerine[TK]D-Fender
21:20.29uskerinehow can i get the precise specs of the file?
21:25.16jsmanuskerine: file /var/lib/asterisk/sounds/welcome1.wav
21:26.44*** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210)
21:27.06uskerine> /var/lib/asterisk/sounds/welcome1.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz
21:27.30uskerinei also have
21:27.32uskerine> /var/lib/asterisk/sounds/welcome2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
21:28.42[TK]D-Fender800 khz required
21:28.48[TK]D-Fender8khz
21:28.50[TK]D-Fender8000 hz
21:28.58jsman;)
21:29.13[TK]D-Fenderso welcome1 = dead
21:29.21[TK]D-Fenderthere's an immediate fail
21:29.41uskerinei also tried with welcome2
21:29.53uskerineexten => s,n,Playback(welcome2.wav)
21:30.07uskerine[Jul 17 23:16:37] WARNING[14357]: file.c:663 ast_openstream_full: File welcome2.wav does not exist in any format
21:30.07uskerine[Jul 17 23:16:37] WARNING[14357]: file.c:958 ast_streamfile: Unable to open welcome2.wav (format 0x4 (ulaw)): No such file or directory
21:30.41jsmanuskerine: you're not supposed to add the extension of the file i think
21:30.49[TK]D-Fenderuskerineexten => s,n,Playback(welcome2.wav) <- what part of NO EXTENSIONS were we not clear on?
21:31.00uskerinei also tried with no extension
21:31.09uskerineexten => s,n,Playback(welcome2)
21:31.10uskerine:)
21:31.14uskerinesame result
21:31.26uskerine[Jul 17 23:30:32] WARNING[14369]: file.c:663 ast_openstream_full: File welcome2 does not exist in any format
21:31.26uskerine[Jul 17 23:30:32] WARNING[14369]: file.c:958 ast_streamfile: Unable to open welcome2 (format 0x4 (ulaw)): No such file or directory
21:31.34[TK]D-Fender"module show like format"
21:31.50[TK]D-Fenderand look in asterisk.conf to confirm if your varlib is even where you think it should be
21:32.01[TK]D-FenderBRB
21:32.07junminuskerine: try the directory /usr/share/asterisk/sounds/
21:32.30uskerinehttp://pastebin.com/2hfFbQDm
21:32.52*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
21:33.07uskerine<[TK]D-Fender>
21:33.11uskerinehttp://pastebin.com/2hfFbQDm
21:33.55[TK]D-FenderAt least those are there.
21:33.55uskerineregarding asterisk.conf:
21:33.56uskerinehttp://pastebin.com/cR3bXRfA
21:34.07[TK]D-FenderNew dump as per the last I requested please....
21:34.18uskerinecan't see any sound directory
21:34.21uskerineshould I add one?
21:35.07uskerinejunmin
21:35.15uskerinemoving to /usr worked fine
21:35.32uskerine/usr/share/asterisk/sounds/
21:35.43uskerineshould I add something in asterisk.conf and use /var....
21:35.44uskerine?
21:39.12jsmanuskerine: what user are you running asterisk as?
21:39.27uskerinei guess asterisk
21:39.29uskerinewhy?
21:39.33jsmani would check permissions on the directories
21:40.08uskerine/var/lib/asterisk/ is asterisk:asterisk
21:40.20uskerinesame for sounds
21:40.39uskerinedrwxr-xr-x 4 asterisk asterisk 4096 Jul 17 23:36 /var/lib/asterisk/
21:41.28*** join/#asterisk PipBoy (~PipBoy@206-248-172-38.dsl.teksavvy.com)
21:42.05uskerinei am fine anyway putting audio on /usr/share
21:42.09*** join/#asterisk mcf3782 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net)
21:43.05uskerinewould you point me to the right path on how to implement a time/day of the week check in the dialplan to either pass the call or launch a message and drop the call?
21:44.01jsmanyou set astdatadir to /usr/share/asterisk that's why it's looking for sound files there
21:45.29uskerineunderstood
21:45.30uskerinethanks jsman
21:46.27jsmanhttps://wiki.asterisk.org/wiki/display/AST/Application_GotoIfTime
21:49.36nnyhttp://pastebin.com/4Q2QsLdP <-- this query run manually and logged in as my test user asterisk works fine. From the command MySQL though the fields just don't get updated. Advice?
21:50.30nnyraw dialplan info FYI http://pastebin.com/vTqRtgKa
21:50.43*** join/#asterisk jpsharp (jsharp@ohno.mrbill.net)
21:50.48carraruskerine: you can also use this to find out what you are running Asterisk as
21:50.48carrarps -o user,flags,cmd -p `cat /var/run/asterisk.pid`
21:53.57nnykeep getting conflicting reports on wether or not asterisk 1.8 supports custom cdr fields that can be set using Set(CDR). thoughts?
21:54.55jsmannny: what are you using as cdr backend? odbc or mysql?
21:55.56jsmani think the custom fields were something that adaptive odbc could do, not sure about the mysql cdr backend
21:56.28*** join/#asterisk serafie (~erin@nat/digium/x-uvpxycumvlyjgqtx)
21:56.46nnyjsman: mysql. Yeah that was the original way, conflicting info on 1.6 and beyond
21:57.22nnyjsman: I can't even get a straigh MYSQL query to update it thought, it's freaking odd. Perms are open for my test user, see http://pastebin.com/4Q2QsLdP and http://pastebin.com/vTqRtgKa
21:57.34nnyjsman: er MySQL application in asterisk, the query run manually works fine
22:02.42jsmannny: strange, no errors/messages from the MYSQL app? do you see traffic on the mysql server?
22:03.58*** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net)
22:04.31nnyjsman: no errors or otherwise. the mysql server is localhost, not sure how to diagnose further at this point
22:05.39*** join/#asterisk brdude (~brdude@12.155.183.30)
22:07.23jsmannny: i also don't know ;) try: http://dev.mysql.com/doc/refman/5.6/en/server-logs.html ?
22:13.16*** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com)
22:13.29nnysorry can you repeat that jsman?I lost my connection
22:13.40nnywell, i had to reboot actually -_-
22:18.19nnyarg
22:18.26nnythis is so damn annoying
22:18.34jsmannny: i also don't know ;) try: http://dev.mysql.com/doc/refman/5.6/en/server-logs.html ?
22:19.07nnyjsman: i even tried root as the user and UPDATE to one field, yet nothing. It's almost like asterisk is doing something it's not telling me about
22:19.50jsmansounds more like it's doing nothing
22:20.10jsmani would really check to see if anything is coming in on the mysql server
22:24.42nnyoh...
22:24.49nnydamn it
22:24.49nnyisn
22:24.57nnyisn't the cdr written out after the call is done?!
22:26.34nnyargg wargarble
22:26.58leifmadsenyes
22:27.00*** join/#asterisk shadebob (~shadebob@41.141.25.35)
22:27.17leifmadsenthere is an option in cdr.conf to allow access to the "to be stored" values in the 'h' exten
22:27.27leifmadsenotherwise, by default you don't get access to them
22:27.33nnyleifmadsen: i assume those don't include cutom fields
22:27.43nnyleifmadsen: my goal is to add 3 custom fields to the cdr
22:28.03nnyleifmadsen: but 1/2 of what I read says you can do that post 1.6, the other half says use mysql to do so, but neither seems right
22:29.46nnyleifmadsen: it almost seems like I have to run my query at H and set the endbeforehexten=no to yes
22:30.03nnyseems so ham handed for something that is probably a common need though
22:38.01nnyok figured it out
22:38.25nnyi can use Set(CDR(something) etc if i set endbeforehexten to yes. Sweet
22:42.53nnymy ham hands are now delicious bacon gloves. Thanks
22:48.38*** join/#asterisk Ahmad_Khokhar (~chatzilla@116.71.191.224)
22:48.44Ahmad_KhokharHi all.
22:48.56carrarhi!!
22:48.57Ahmad_KhokharI am having problems with paging
22:49.04carrarwith ageing?
22:49.13Ahmad_KhokharI have Cisco 7911, 7942
22:49.17TomCat2bizarre...  I have my firewall set up to allow outbound TCP/UDP 5060 and UDP 10000-20000...  no audio in either direction when I make a call...   open up the firewall to allow all outbound, no problems...  am I missing some other port that asterisk needs?  I swear I made calls on this earlier today w/o all ports allowed
22:49.45TomCat2using flowroute as the trunk if that makes any difference
22:49.48Ahmad_Khokharconnected and working good, but in the SEP file i have them set to auto answer, so that i can atleast have them answer on an extension if a call comes in (for paging reasons)
22:50.16Ahmad_KhokharIs there any workaround for that ??? I only want them to auto answer on call info not on all calls
22:50.25Ahmad_KhokharI can not seem to find any config for that
22:50.28Ahmad_Khokharanyone\?
22:53.31Ahmad_KhokharI guess no one has been through paging
22:53.42Ahmad_Khokharcmon guys, no one uses this feature?
22:54.37uskerinejsman
22:54.41uskerinei am trying gotoiftime
22:54.53uskerinebut i think i am not fully understanding the basics of asterisk
22:54.58uskerinethis is the workflow i am trying
22:54.58uskerinehttp://pastebin.com/raw.php?i=LLi7dwFf
22:55.23uskerinebut right after executing the gotoiftime to "closed" it also eavluates the next one, so it seems it is not actually jumping into "closed"
22:57.46uskerinehola
23:02.24*** part/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210)
23:02.43*** join/#asterisk sds (~sds@fsf/member/sds)
23:04.01paulcuskerine: I think you're missing a comma on your 25Dec line
23:04.25Ahmad_Khokharanyone have info on the autoanswer on call info for cisco phones?
23:04.36Ahmad_Khokhari cant seem to find a proper info on it
23:04.47Ahmad_Khokhari have autoanswer , but no autoanswer on call info
23:04.59paulcAhmad_Khokhar: How about using a different line button for paging vs regular calls? It's been a while since I played with Cisco phones, but I thought you could do something with alert-info too
23:04.59Ahmad_Khokharso wont work on a single line phone like 7911
23:05.06Ahmad_Khokharhi paulc
23:05.20Ahmad_Khokhar7911 is a single line phone , so cant seem to register a second line on it
23:05.25paulcah.. single line phone.. yeah, I dunno - if you can't get it to work with the right combination of alert-info (if it's supported), then you're probably SOL.
23:05.27Ahmad_Khokhari have tries line 2 in SEP
23:06.00Ahmad_Khokharcould you guide me on the right track plz
23:06.05Ahmad_Khokharwhat about alert-info
23:06.06Ahmad_Khokhar?
23:06.35TomCat2wireshark to the rescue!  good to know that RTP will run wild if you don't specify rtpstart and rtpend...   doesn't default to 10000-20000 range as previously and incorrectly assumed
23:07.19uskerinethanks paulc, i added the comma, but it stills go through both GotoIftime's
23:07.29uskerinewhy does not jump to "closed" ?
23:07.42uskerinesorry i didn't say anything
23:07.48uskerinetoday it is not 25th december
23:07.58carrarChristmas!
23:09.45paulcuskerine: I don't see an extension with the (open) label
23:11.08*** join/#asterisk wudles (~wudles@gateway.secureinstrument.com)
23:11.20sdsany idea where i can find a good international sound package (fr/en) with words and phrases for a payment ivr? things like "amount, money, transaction", etc. something free, or not. just something already done.  i have a rush and cannot wait 24h to 48h delay for a custom ivr voice (which would be the easy way).
23:12.24Ahmad_Khokharsds try http://www.ivona.com/en/
23:13.00paulcsds: Prior planning prevents piss poor performance! ;-)   Go with your own voice for now, and get it done "properly" with a voice talent later in the week?
23:13.54sdspaulc: yes. even though i have a "poor planning on your side doesn't create an emergency on my side" poster on my office's door... seems that it doesn't always work that way! ;)
23:14.01sdsAhmad_Khokhar: i'll check that out. thank you.
23:17.05paulcsds The alternative is Ceptral.. with the Allison voice.. works well enough for TTS, as a stop-gap measure till you can get your whole vocab voiced properly. We use it for dev prompts mainly cos it integrates nicely with Allison's existing prompts/digits/numbers etc
23:17.12paulc$20 or $25 for a license I think
23:17.17paulcI gotta take off.. back in a bit
23:17.39*** join/#asterisk engrxyz (~dgf@host81-150-217-167.in-addr.btopenworld.com)
23:27.53*** join/#asterisk eicto (~eicto@144-71.dsl.aichyna.com)
23:31.39TechSmurfDoes an openvpn tunnel often induce significant lag for iax trunking?
23:32.10*** join/#asterisk egonzalez_ergio (~egonzalez@186.137.1.3)
23:32.47egonzalez_ergioHi
23:33.00egonzalez_ergioI have a problem with calling
23:33.20egonzalez_ergioI can receive calls but when I call an externerl extention
23:33.44egonzalez_ergioever answer the answer machine
23:34.11egonzalez_ergioI'm behind a nat/firewall
23:34.26egonzalez_ergiowith all ports redirect to the asterisk
23:34.32[TK]D-FenderTechSmurf: No
23:35.13egonzalez_ergiosip show peers : shows all extention whith host in internals IPs but are remotes ips!
23:38.00TechSmurfhrm... lag seems to be significantly higher than the average 48ms round trip to the remote pbx...
23:44.34*** join/#asterisk Harzilein (~harzi@harzilein.eu.org)
23:44.38Harzileinhi
23:44.50Harzileindo you know if ovh has an official echotest address?
23:48.41*** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net)
23:48.48TechSmurflol... silly system is dialing out its own phone number rather than routing internally
23:49.12*** join/#asterisk zerohalo (~zerohalo@74.61.196.236)
23:54.09*** join/#asterisk egonzalez_ergio (~egonzalez@186.137.1.3)
23:55.34WIMPyTechSmurf: I'm using openvpn all the time.
23:56.17WIMPyAnd it sounds like someone didn't "include" the external extensions in the phones context(s).
23:58.03*** part/#asterisk D-Boy (~D-Boy@unaffiliated/cain)
23:58.40*** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain)

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