IRC log for #asterisk on 20070624

00:00.15pipwerkI guess actually correctly interpreting what you read is harder for some then for others
00:01.14*** join/#asterisk rnovotny22 (n=ro085181@h460da7b0.area2.spcsdns.net)
00:03.16Strom_Mpipwerk: what we need is drool-proof asterisk
00:03.39Strom_Mso simple, even the totally wasted who can't even order at Denny's can set it up!
00:04.19Strom_Mholy god no
00:04.46Strom_Mthat guy is just way too queer even for me
00:04.46pipwerkI wouldn't know what denny's is, but I get the point
00:04.59Strom_Mpipwerk: 24 hour greasy spoon diner type place
00:05.03pipwerkand yes, * was harder that I expected when I got started
00:05.15Strom_Mthe menu has many pictures so you can order just by pointing if you're so inclined
00:05.21pipwerk:)
00:05.41fileStrom_M: if you are ever in HSV and Robinson invites you out for a midnight Denny's group gathering... go insane
00:05.49pipwerkas I said, I get the point
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00:06.12*** mode/#asterisk [+o anthm] by ChanServ
00:06.19pipwerktime for bed, later
00:06.34Strom_Mlater
00:06.47Strom_Mfile: I will indeed do that
00:06.58Strom_Mseeing as how i'm gonna be in lolabama in a few weeks...
00:08.57[TK]D-FenderStrom_M, I take it he spammed your in PM then gave only you the pastbin?
00:09.34Strom_Myes
00:09.47Strom_Mhow did you know ?!?!?!??!!?!?!?!?!?!?!!!!!!!!!!!!!?!?!!?!??!?!?!?!!?!!!?!?1//11/1/1/1
00:09.56*** join/#asterisk rnovotny22 (n=ro085181@h460da7b0.area2.spcsdns.net)
00:10.39rnovotny22Anyone have any ideas where to start looking for an intermittant no receive audio problem using voip?
00:10.56Strom_Mthe router
00:11.01Strom_Mthe network
00:11.04Strom_Mthe pbx
00:11.07Strom_Mthe phone
00:11.19rnovotny22the provider?
00:11.28Strom_Msure, that too
00:12.06rnovotny22That narrows it down a little.
00:12.43denonthe user :)
00:13.16rnovotny22Well, I think I can rule that one out.
00:13.28denonwell, if you have to ask ...
00:13.31denonI wouldn't rule that one out :)
00:13.48Strom_Mnever rule out the user
00:13.56sevardOMG LOL BOOM
00:14.05Strom_Mhahahahahaha
00:14.19Strom_Msevard: did you just see the biggest explosion evar>?
00:14.58sevardi've regressed into my 12 year old state, i'm currently hiding in my treehouse with crates of ak47 ammo and viagra
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00:29.19[TK]D-Fendersevard, s'ok... you have only 1 year to take back ;)
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00:33.49sevardOOOOOOO there's TK with the really well thought out and very prompt insults.
00:34.05sevardman, they have so much DEPTH, they just cut DEEP
00:42.27[TK]D-FenderThe Wiltshire Wit strikes again!  Never dulls!
00:43.12centrexwow, I thought the asterisk documentation was pretty bad, but freespeech just doesn't have any
00:43.19centrexfreeswitch
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00:49.24anthmwhy don't you go to that channel and complain!
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01:02.08adeelnhi, i'm running asterisk 1.2.17 and i can't seem to get my sip clients to use any zap devices, and vice versa
01:02.34harlequin516adeeln: Why not?
01:02.47harlequin516What happens?
01:02.51adeelnharlequin516: i don't know...it just won't let me dial out on it
01:02.53harlequin516Wheres the log?
01:02.57adeelnlet me pastebin it
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01:03.44guille1983hi
01:03.53Strom_Mbetter yet, pastebin sip.conf and extensions.conf
01:03.59Strom_MI bet you forgot an include :)
01:04.13guille1983i would like to know important companies that are using asterisk... would you please tell me some of them ?
01:04.33denoncisco, avaya, the white house
01:04.46denonpretty sure the FBI uses it
01:04.52denonI think ATT runs it
01:05.07guille1983denon: really? white house and FBI ?
01:05.12MoutaPToh really white house?:P
01:05.13denonwell, no
01:05.15denonbut that'd be nice
01:05.16harlequin516Anyone know how to play a .ulaw file from the linux commandline?
01:06.03guille1983i need to prove who are using *
01:06.12Strom_Mguille1983: why?
01:06.23denonhe's in sales
01:06.26denontrying to convice a manager
01:06.34denonor owner
01:06.38guille1983Strom_M: an assigment in my class
01:06.45denonhaha
01:06.51denonhomework help in #asterisk
01:06.58denonnever thought I'd see the day
01:07.08guille1983:D now you see it
01:08.53adeelnhere's the sip.conf file http://www.pastebin.ca/586790
01:09.16Strom_Moh fucking hell, not trixbox again
01:09.23Strom_M~trixbox
01:09.23jbotsomebody said trixbox was a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org
01:09.55harlequin516adeeln: We want to see your CLI output or logs from your asterisk server.
01:10.09adeelnharlequin516: i'll paste that as well
01:10.15Strom_Mharlequin516: it's trixbox; don't bother trying.  :)
01:11.17Strom_Myou'll only get a headache
01:11.19harlequin516adeeln: HAHAH.  OSI
01:13.26adeelnosi?
01:14.59adeelnharlequin516: http://www.pastebin.ca/586795 is the cli output...when i call from zap to sip, it rings the sip phone, but drops when i pick it up...calling from sip to zap, nothing happens
01:18.33MoutaPTzap show channels
01:18.40MoutaPTwhat it gives u?
01:20.01adeelnzap show channels -- http://www.pastebin.ca/586800
01:21.48adeelnsip to zap call -- http://www.pastebin.ca/586801
01:23.42harlequin516I don't really understand how/why you have 12 Zap channels?
01:23.50harlequin516Whats your hardware?
01:24.14adeelnopvxa1200 .... it's a 12 port card
01:24.19adeeln3 fxo, 9 fxo
01:24.25adeelnerr 9 fxs
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01:25.34adeelneverything used to work great, but i recently migrated machines, and then upgraded asterisk from 1.2.15 to 1.2.17 and then this started happening
01:25.54Strom_Madeeln: that's your punishment for using that horrible gui
01:26.08adeelnStrom_M: any other recommendations for a gui?
01:26.19Strom_Mvim
01:26.33adeelnhaha vim really isn't a gui
01:26.50adeelnalthough gvim is, but that's irrelavent
01:26.55Strom_Mwhat the hell do you need a gui for, anyway? :)
01:27.05adeelnwell, it used to make my life easier
01:27.29adeelnand it's a headless machine
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01:28.03adeelnan http interface was supposed to be the easiest for us
01:28.26Strom_Mssh was too complicated or something?
01:28.46adeelnno, but i don't always have an ssh client on the machines i'm using
01:29.00adeelnssh is setup, and that's how i do debugging if i need to, but i normally don't need to
01:29.21Hymieput a copy of putty on an internal website
01:29.29Hymiethen you can download it to any windows box in 1 second
01:29.33Strom_Mput a copy of putty on a thumbdrive
01:29.39Hymieheh, or that too
01:29.44Strom_Mor hell, just download it from their website every time
01:30.22adeelnbecause i'm rarely at my desk
01:30.34adeelnor in the office for that matter
01:30.48adeelnhttp also allows me to remote admin from my phone
01:30.54Strom_Msounds like you need...a laptop
01:31.03adeelnits in the shop
01:31.15Hymiethat's it's, but anyhow.. look
01:31.16Hymiejust
01:31.17Hymiedownload
01:31.17Hymieputty
01:31.18Hymieand
01:31.19Hymiestop
01:31.20Hymiecomplaining ;)
01:31.26Strom_Mwelcome to the excuses hour, with our special guest adeeln
01:31.28*** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-165-37.columbus.res.rr.com)
01:31.29adeelnbut you guys think it's a config issue?
01:31.32Hymieit takes 10 seconds to download
01:31.37Strom_Madeeln: it's a freepbx issue
01:31.44Strom_Mtoss that crap and build a real asterisk system
01:32.15adeelni also don't hear any prompts being played...like *69 will tell me the digits, but there are some other voice prompts that don't work anymore
01:32.21adeelnthat a freepbx or asterisk issue?
01:32.21Hymieer
01:32.27Hymiethat's broken, they should all work or none
01:32.38adeelnthe logs say they're playing, but i don't hear anything
01:32.49Hymiedo they all not work, or some don't work
01:32.55Hymieon the same call
01:33.07adeelnall the digit related ones work, but anything like 'please enter your pin' don't
01:33.23Hymiewell, that does sound like a messed up dialplan
01:33.35Hymieinternal asterisk apps seem to find them
01:33.39Hymiebut the dialplan doesn't seem to
01:33.57adeelnis there a need to compile the alsa/oss module for asterisk?
01:34.13Hymiesome of the sounds work
01:34.18Hymietherefore, it's not that
01:34.20Hymieand no
01:34.24Hymieno oss needed
01:34.41Hymiewhat phones are you using, just curious
01:34.50Hymienot really related to your prob
01:34.53adeelnwhat's the use for the alsa module? transcoding? playback on the local machine through speakers?
01:35.06Hymieyes, your local box can become an extension
01:35.24Hymiealsa isn't needed for transcoding
01:35.34adeelnwell we have a bunch of analog cordless phones plugged into the zap channels, 2 grandstream bt-120's, & a bunch of different soft phones (x-lite, phonerlite, etc)
01:36.06Hymieinteresting collection
01:36.06Hymiewell
01:36.12Strom_Msounds like you think you can get away with a business-grade PBX for $20
01:36.16HymieI've never used anything buy pure asterisk
01:36.21Hymiebut
01:37.16Hymieanyhow, it sounds like your install is borked, but.. I don't know if you know enough to debug it, if you get what I mean
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01:37.24Hymieit might be better for you long term, to learn asterisk
01:37.32Hymiesave you time over 10 years
01:37.41adeelnStrom_M: i'd hardly say that's the case...this is more of a proof of concept machine; fax doesn't work reliably through asterisk (at least i've had a hell of a time getting it to work) and we're trying to figure out what combination is right for us
01:38.04Strom_Madeeln: i'm surprised that openvox crap works at all, tbh
01:38.06Hymieadeeln: I can't say enough good about the polycoms
01:38.25adeelnthere's no point in spend 10,000 on a demo system
01:38.28Strom_Madeeln: fax is a sensitive one; you need to set things up in specific ways
01:38.40Strom_Mgrandstream is crap
01:38.52adeelnthe grandstreams are crap, and we're aware of that, but if the crap works well, then the expensive stuff will be worth the investment
01:38.53Strom_Mpoor build quality, poor feature set, poor reliability
01:39.42adeelnanyways, thanks for the help
01:39.45__DAWgood or bad, you should test with what you will be using.  Things vary greatly between phones.
01:40.46Strom_M"how to play ostrich" by adeeln
01:41.04__DAW:)
01:41.57Hymiehttp://toronto.ctv.ca/servlet/an/local/CTVNews/20070623/contaminated_chocolate_070623/20070623?hub=TorontoHome
01:42.09Strom_MOH NO NOT THE CHOCOLATE
01:42.13Hymiemy god.. to steal chocolate bars being disposed of an reselling them?
01:42.17Strom_MANYTHING BUT THE CHOCOLATE
01:42.27Hymieso little kids die, I mean, come on, that's freaking beyond low
01:42.37Strom_MHymie: people are greedy, cheap, and stupid.  you should know this by now.
01:42.39HymieI bet those guys go beat up midgets for weekened entertainment
01:43.11HymieStrom_M: the day I become unastonished by such horrid greed, is the day that my morality changes
01:43.16Hymieand, that is not acceptable
01:43.29Hymiethe astonishment, the anger, it remains in the face of such horrid actions
01:43.48Hymiethat does not mean I do not know about it... merely that I am still angered by it
01:43.52Hymieas you should be
01:44.01Strom_Mso you'd rather live in fantasy land and believe that people aren't greedy and selfish and moronic?
01:44.30Hymiethat statement makes no sense, based upon what I've just said to you
01:44.43HymieI already stated I realise that such people exist, this is not news to me
01:44.58HymieI simply refuse to accept that behaviour as something that does not anger me
01:45.02Strom_Mand what has this got to do with phones, anyway
01:45.13denonHymie: I agree.
01:45.19Strom_Mplease check the number and dial again
01:45.21HymieStrom_M: maybe someone spread that chocolate onto a polycom before you received it!
01:45.34Strom_Mor ask your operator for assistance
01:45.39Strom_Mdelicious, poisonous assistance
01:45.45Strom_Mthis is a recording
01:45.49Strom_M213-4
01:49.01*** join/#asterisk Wayra (i=Wayra@201-221-202-37.bk11-dsl.surnet.cl)
01:50.16__DAWAny major problems with 1.4.5?
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02:29.38paragoncdo SMS chargers only effect the end user, or are there origination fees from VoIP providers ?
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03:31.25ez`i need a solar satelite voip phone
03:34.26JTwhy on earth would you want it to be voip?
03:35.50coppicehe doesn't want it to be on earth
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03:41.43plasmidI setup extension 5 on my IVR to be emergency which should ring all phones. I created a ringroup called "emergency" which should dial all my extensions: 1,2,3,4 .. Is the correct syntax in the extensions box: 1# 2# 3# 4# OR 1#,2#,3#,4# ?
03:41.53plasmidextensions 3 and 4 are misc destinations = my cell and my partner's cell.
03:42.18JTplasmid: this channel is for asterisk support
03:42.26plasmidJT: I know.
03:42.43JTyour problem does not sound like an asterisk problem
03:42.51plasmidI was hoping the underlying foundation is similar.
03:43.07JTfreepbx?
03:43.28JTwell asterisk doesn't have an "extensions box" for one
03:43.35plasmidOn asterisk, how would u setup a ring group to ring all all extensions?
03:44.16JTthe dial command can dial multiple extensions at once
03:44.23plasmidI don't think it's a problem per se, but more a syntax issue.
03:44.44plasmidthis dial command you are reffering to... what is the file to modify?
03:44.48JT~freepbx
03:45.59jbotfreepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
03:45.59JTthe dialplan, extensions.conf
03:45.59plasmidso in extensions.conf the dialplan is called say: emergency and the syntax should be 1# followed by a comma or a space or?
03:46.04JT~thebook
03:46.05jbotfrom memory, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
03:46.16plasmidyes...yes. I am aware of that.
03:46.16JTyou need to read the book
03:46.22JTespecially of the dial command
03:46.30JTyou are? then you should have no problems
03:46.43plasmidno i meant the referral.
03:47.11plasmidI dont use the IVR that much so I was hoping to get a quick solution to the matter.
03:47.24plasmidthen again I should peruse through the documentation once more
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04:03.09ech0dishhi
04:05.37ech0dishim a lil curious about asterisk. i've been trying to read a lil about it but its kinda confusing. i was wondering if this could be used to turn my home phone line into a voip that i can use on other computers on the net?
04:06.01ech0dishto make pc to land line calls?
04:06.47*** join/#asterisk syle (n=blah@unaffiliated/syle)
04:09.01ech0dishcause i got a laptop and a desktop... it would be cool if i could use my desktop as a voip server connected through my modem to my home phone and then make calls from my laptop at whatever wifi spot and it would show up as if i was calling from my home phone on the caller id?
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04:14.16fx0not cool
04:14.16ech0dish?
04:15.05ech0dishnot cool what?
04:15.32JTzeros in nicks ;)
04:16.01ech0dishk
04:16.19ech0dishhi, u know about asterisk any?
04:17.04JTdo I know any asterisk? Yes.
04:18.03coppiceI know some asterisk. I know "rm -rf /*"
04:18.26ech0dishsigh
04:18.28*** part/#asterisk ech0dish (n=matt@70-140-179-217.lightspeed.hstntx.sbcglobal.net)
04:18.56JTasking dumb questions doesn't get people far, who knew?
04:19.21coppicePlease, please, I know this one. I know this one!
04:21.29plasmidIf my ring group is "5" (where it will ring all my extensions: 1,2,3,4) do I have to modify the dialplan on those extensions and put callgroup = 5?
04:26.00plasmidack.. nv. Callgroup is for picking up an extesion if the other doesn't pick up by punching for example *8
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04:40.26kusznirHi all:  I'm trying to debug an asterisk problem that claims to be call routing, but I don't understand how it can be such.  I'm running asterisk 1.2.16 on openwrt linux, configured by hand.  I just rebuilt my system and copied the config files that used to work back.
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04:41.04kusznirNow, whenever I attempt an IAX call, I get "unable to create channel of type 'IAX2' (cause 3 - no route to destination).  internal sip-sip calls work fine.
04:42.19kusznirBut it shows it has sucessfully registered to the remote host, and the line before it shows it is trying to pass the call to the correct host.  The only thing I see is an error setting CALLERID a few lines before.  Could that cause the "no route" error?
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04:53.49J4k3B
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05:14.57JTkusznir: no, it couldn't
05:15.56kusznirYea, I've been troubleshooting that one, and that's fixed.  I think there's an issue with IAX peers.  I'm at a loss on how to debug this further, though.
05:16.26kuszniriax2 debug shows registrations (about every 15 sec, I think) on all 3 of my iax peers.  Yet iax2 show peers shows them all as "UNMONITORED".
05:16.48kusznirreload shows "registered to peer..." messages, which all look right.
05:17.13kusznirI have verified I can receive an outside call via SIP, but I cannot receive calls via IAX.
05:19.03b1shopmy whole config is hosed.
05:19.07Strom_Mwhat does 'iax2 show registry' show?
05:19.17JTqualify is not set to yes if they're unmonitored.
05:19.24b1shopi can call ext 12 from ext 10.  but ext 10 cant call 12
05:19.26b1shopetc
05:19.34JTit has no relation to registration
05:20.25kusznirso is that an option in the peer stanza in iax.conf?
05:20.35Strom_Mkusznir: what does 'iax2 show registry' show?
05:20.36kusznirshow registry shows 3 peers registered.
05:20.37JTyes
05:20.53kusznireg, 64.2.142.22:4569      kusznir     66.233.138.60:4569         60  Registered
05:22.24b1shopi bought a sangoma 200 card.  i was pissed when i saw the ports son't accept standard phone line ends.  how do you get around that?
05:22.43coppicethey supply the cables you need
05:22.50b1shopyeah.  but only 4ft
05:23.10b1shopwont reach from the punchdown to my rack
05:23.28coppicethe strange plug is a headset plug, so they are easily available as parts, but not as leads
05:23.49coppiceI understand this was a cockup by the developers, and not part of any master plan
05:24.02b1shopi have a crimper.  just @ a loss as to where to buy the ends
05:24.34coppicesorry, handset plug, not headset plug
05:24.53b1shopyup.  understand the end exactly
05:25.10b1shopaside from cutting the ends and splicing.. i have no idea what to do
05:25.16kusznirI have found small quantities available at Lowes and Home Depot.
05:25.34*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)
05:25.43b1shopkusznir: they cromp ok with a rj-45 crimper?
05:25.47b1shopcrimp*
05:25.58kusznirI've seen them referred to as "4p4c" connectors in bulk from places like digikey.
05:26.39b1shopi'm kinda screwed.  moved offices this weekend.  was hoping to have new system built out by then
05:26.43b1shopnot looking good
05:26.46coppicesomething that rolls of the toungue nicely, like that
05:26.53kusznirIf the crimper has the insert for the smaller connector.  There are only 4 pins, and the body doesn't have any "spare" locations, so the crimper die has to have exactly 4 pins.
05:27.35kusznirA lot of RJ-45 crimpers that also do RJ-11 can do the headset.  I think I got a cheap one form Wal-mart years ago....
05:27.42b1shopif anything can possibly go wrong.  it has with this install
05:28.10coppiceI have a headset crimper in the cupboard next to me, and a few plug. now, if you can arrange crimp over IP.......
05:28.21b1shopdcc?  ;-)
05:28.29*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
05:29.31b1shopi might be forced to try and make my old norstar system work
05:30.03coppicethe reason I have a crimper which can do the headset plugs is because I also have some sangoma A200 cards :-)
05:30.27b1shopengineers screwed up and ran two phone and one data instead of 2 data and one phone.  and they used old cat3.. so it's useless
05:30.57b1shopat&t ran the lines into the basement and the bldg engineer punched them to block in new office.  but i can;t get dialtone off them
05:31.28JTwhat's wrong with cat3?
05:31.36b1shopand of course i bought POE phones... so no one will have PC connections due to lack of ports
05:32.16sylefuckit install wireless router and wireless cards or bitch at at&t
05:32.30JTwireless? evil
05:32.46JTPoE phones usually have a switch, with a PC port
05:32.50syleyeah i hate them to, i use it for laptop only
05:33.11b1shopyup.  had to hack a wireless together
05:33.29JTb1shop: what phones did you buy?
05:33.34b1shopunfortunately.  the polycom 320's i bought don;t have the extra ethernet port
05:33.47JTwell you should've checked that first
05:33.55JTit's like an extra $10 for an ethernet port
05:34.18b1shopdidn;t know iw as goign to be an issue.  the install order was for 2-data 1phone.  not the other way around
05:34.38JTso can't it be changed?
05:35.16b1shopsure.  but they used old cat3 for phone.  and it does't help me on mon morning
05:35.39JTcat3 can carry data
05:36.01b1shopplus.  engineers seemed to have cut power for a 6 cubes in the office
05:36.19b1shopthey have data.  no power
05:36.35kusznirlock the switch port to 10Mbps and you'll be fine...two of my buildings were wired exclusively with Cat-3 due to stupid people working in the contracting office...(we can save $$$ by downgrading CAT-5 to CAT-3...nobody will notice....)
05:36.38b1shopJTcat3 do 100mb or only 10mb?  i was wondering about that
05:36.58JTdepends on length
05:37.19JTi've had success with 100Mbit/s on <50m runs
05:37.21coppicea lot of e-mail tells me that
05:37.25kusznirMy experiences have been at about 10 feet, cat-3 start to get a lot of errors at 100Mbps.
05:37.32b1shopalso have 2ton ac unit plugged into new server room.  that blew a fuse tonight and my 5TB NAS box overheated
05:37.44*** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
05:38.18b1shoplike i said.  NOTHING with this move is working right
05:39.37b1shopmy * install is working like crap also.
05:40.13b1shopprovisioned phones with tftp.  some can dial extensions... others cannot
05:40.17b1shopbut they all ring
05:40.27JTeww tftp
05:40.40b1shopftp better?
05:41.13b1shopthis is my 1st install.  so if you can do it wrong.  i prob am
05:41.17JTftp or http, or the secure versions
05:41.29JTtftp is not recommended by polycom even
05:42.30b1shopi tried ftp.  created user/pass with the polycom defaults.  it seemed to always fail to log in
05:42.47*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
05:43.40russellbprovisioning via ftp works great for me.
05:45.01coppicesucker
05:45.20b1shoplol
05:45.36b1shopanything is better then not having phones on mon morning
05:46.00coppiceah, the sense of peace sounds blissful
05:46.10kusznirSo, any more thoughts on what could be causing my issue? (although its far less important than b1shop's....my * is for personal learning and experimenting)
05:46.35b1shopi'm thinkign about pissed off clients and loss of $
05:46.55b1shopnot like i don;t have 20 other things wrong with the move i could spend time on
05:47.09coppiceif they can't call you, then were is the problem? :-)
05:47.46coppicecue Monty Python.....
05:47.48coppice"Always look on the bright side of life..."
05:47.54b1shopoh.  i dunno.  missed deadlines.  no new business. pissed off clients that send new projects elsewhere, etc..
05:48.19coppicethere you go. focusing on the downside
05:49.00b1shopheh.  i even went out to radio shack today and bought a $10 phone just to punch in so i had SOMETHING that rang.  i still could not get dialtone
05:49.29*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)
05:50.13coppicesee, that's the business you should be in. Those phones leave the factory at around $1, and you pay $10.
05:50.28b1shopodd thing is that my internet punched down fine
05:51.10kclaussenwell you know shipping :)
05:51.41coppiceif I really knew shipping, I'd be buying $1 phones and reselling them :-)
05:52.05sevardyeah, $1 phones, big business, lots and lots of money to be made there.
05:53.20coppicethere's a lot more in that than buying $10 IP phones and reselling them at $50
05:54.42kusznirSo, I think I've traced a part of it down now...I'm pretty sure its coming back to CallerID (I think my peers dont' accept calls without CID info set or something).  Anyway, I'm using: Set(CALLERID(number)=<my DID>), but I get: ERROR[7420]: pbx.c:1437 ast_func_write: Function CALLERID not registered.  How do I fix this?
05:56.20kusznirb1shop: Well, a good chunk of the problems are not your fault:  Insufficient wiring for the AC, data/phone wiring not to spec, insufficient power for cubes, etc.  So even if asterisk did work, you'd still not be able to provide reliable service due to others failing to do their job properly.
05:56.56b1shopyeah.  i know.
05:57.05coppicekusznir: wheta he is experiencing seems pretty normal for a move, so a lack of planning for it might well be his fault
05:58.06b1shopoh.  some if it is def. my fault.  murphy's law.  week before move is always busiest with projects
05:58.33b1shopnot to mention having a crappy admin
05:58.45b1shopher days are numbered....
05:58.55coppicecrappy admin == janitor? :-\
05:59.32b1shopyeah.  i can seem my interview requirements dropping now.
05:59.53b1shopQiestion 1:  can you actually make a decent pot of coffee?
06:00.12coppiceAnswer: No. I work at Starbucks
06:00.30*** join/#asterisk jamon (n=jamon@ppp-70-225-71-139.dsl.covlil.ameritech.net)
06:01.21b1shopshe got the requirements.  it was her job to make sure everything go done
06:01.46b1shopshe spent hrs over there last week.  always came back and said everything looked great.
06:02.09b1shopshe WAS right.  it looks awesome..  nothing works... but ti looks sweet as hell
06:02.15coppiceshe probably referred to the decor :-\
06:02.52*** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar)
06:03.48b1shoplol.  they delivered new cubes thurs evening.  all the surfaces are a nice slate grey.  except for the end pieces.  tyey are a funny blue color
06:04.43jamonI'm trying to write a script to automatically pickup inbound routes and send a DTMF 2 seconds after picking up, then wait 2 seconds, then transfer it to an extension... do I need to do an Answer() or something before my first Wait() and SendDTMF()?
06:04.56*** join/#asterisk GothAlice (n=amcgrego@200.75.246.175)
06:06.07GothAliceI have a Nokia N95, fully updated, which is successfully registering with my Asterisk server.  If I attempt to make a local-extension Echo test call (ext. 613) the connection is made, Asterisk begins playback of a "this is a test" message then switches to the echo test, but I hear nothing on my Nokia.  I've tested every option for the NAT= option, to no avail.
06:07.17GothAliceNeermind!
06:07.28GothAliceHad the wrong IP on the "externip" option!
06:07.33jamon:)
06:07.43jamoni had that happen with all my inbound calls the other day :p
06:07.57jamoni'm going to write a script to auto update it for me :p
06:08.29jamonwow, that's nice :P
06:08.32jamonnow i'm jealous
06:09.33GothAliceTrust me, the N95 -isn't- worth it.  I just happen to have one because a local (cough: chinese) retailer gave me the runaround.  Some things in Panama are good, some things are very, very bad.  Electronics retailers are bad.
06:09.58jamonah :p
06:10.01jamonthanks for the heads up :)
06:10.15jamonwhat are you doing to connect it to freepbx?
06:10.21jamondoes it have a SIP client built in?
06:10.25jamonor are you running a java one or something?
06:10.26GothAliceThe interface is non-intuitive to the max, the OS is slow and doesn't have enough RAM (get out of RAM errors all the time), the GPS doesn't work here (at all - it thinks I'm in Egypt), and the mic picks up all surrounding noise.
06:10.33GothAliceSIP built-in.
06:10.45jamonthat's cool, if it worked, huh? :)
06:10.54jamonimpressive featureset
06:11.09jamonthat's a new one for nokia though... I've always found that their stuff usually doesn't have as many features, but is more stable
06:11.11GothAliceIt's grossly expensive, the feel is of a $20 Linksys VoIP phone, not a $1K gold brick.
06:11.14GothAliceEtc.  :^)
06:11.30GothAliceThe settings to get SIP working are stupid.
06:11.36jamonof course I bought a Motorola Razr v3i the day they were sold in america for $300 + 2 yr contract
06:11.38jamonso I'm an idiot :P
06:11.58GothAliceProxy: your IP, asterisk realm, username, password.  Register server: asterisk is the host, no user, no pass, no realm.  WTF?
06:12.01jamonthat was before I realized it was a piece of garbage with a song limit via itunes and a USB 1.0 interface :)
06:12.01b1shopheh.  i'll sell ya my old one!?
06:12.08GothAliceXD
06:12.39jamonwho the hell creates a device with 512 meg of storage for music and puts it on usb 1?
06:12.46jamonsomeone should be kicked in the head for that
06:13.05GothAliceXD  True.
06:13.09b1shopsame ppl who put 50 song limit
06:13.17jamonit's 100 for v3i, but still, way way too low
06:13.26jamonso I just use the java based mp3 player instead
06:13.28jamonand let it drain my battery
06:13.33coppicejamon: you simply misunderstand the phone market. the goal is to piss off the customer
06:13.46b1shopi have v3i.  mine caps @ 50 songs
06:13.46jamonapparently so
06:13.51jamonbishop: does it?
06:13.55jamonmaybe I'm misremembering
06:13.56b1shopyup
06:13.58coppiceif you make a music oriented phone, you give it a 2.5mm jack socket, or a proprietary row of pins
06:14.00jamonit's been a year since I used itunes
06:14.13jamonrazr 2 looks much better
06:14.17jamonbut i've already been burned once
06:14.24coppiceif you have USB, you use a propritary row of gold pads that get too dirty to ever work
06:14.40b1shopi ditched my v3 for a blackberry
06:14.49coppiceif you have something hot, like the tx PA, you put it right under their ear
06:14.58jamonhehe
06:15.11jamonthough recently my v3i has been rebooting itself randomly
06:15.15jamonwhite screen for a second, then reboots
06:15.28b1shopsounds like i should sell ya mine!  ;-)
06:15.33coppicethe V3 sounds amazingly good, though
06:15.42jamonyeah, it's sound quality is great
06:15.53*** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com)
06:15.55jamonboth music and talking
06:15.57jamoni'll give it that
06:16.04jamonpeopel can't tell i'm on a cellphone usually
06:16.15coppicepeople complain a lot about codecs, but half the time its the phone at fault. The V3 shows what the GSM codecs can really do
06:16.16*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:17.27jamonany of you ever use grandcentral? :)
06:18.08jamonit's a relaying thingy for phones... they give you a local number someplace, and it forwards to your other phones... or gizmo account
06:18.17jamonI've got mine setup to forward to gizmo,a nd gizmo to forward to asterisk :)
06:18.32jamonit actually doesn't sound bad... and it's a free DID anywhere in america pretty much =)
06:18.57jamononly problem is that when they call you... they play a stupid message and ask if you want to pickup the call
06:19.07jamonso i'm trying to get asterisk to answer it right away, and immediately accept the call
06:19.15jamonit works most of the time, but sometimes it doesn't
06:19.34jamonand it ALWAYS plays the first part of the message, even if it did work
06:19.52jamonwhich is very odd to me :p
06:25.31*** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
06:26.50jamonanyone know what the timeout does on senddtmf()?
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07:03.37kusznirwhat causes asterisk to suddenly show all peers as "Host (Unspecified)" after a reload?
07:03.46kusznir(iax2 show peers command)
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07:36.37waKKuevening...
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07:43.12rvhi0how to do multiple registration for one extension?
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07:51.41rvhi0anyone here?
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08:04.56Dovidany one know where I can rent space on a SBC router or any recomendations for a good SBC router/swtich ?
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08:45.41DeadlyDatais any one around that could help me?
08:48.09tzafrir~anyone
08:48.09jbot*** anyone: No such nick/channel - and yes, there probably is someone, somewhere, who knows or runs it; that doesn't mean /I/ do.
08:48.57tzafrirDeadlyData, just ask your question
08:50.13tzafrirDovid, rent space on a SBC? SBC as in Single Board Computer?
08:51.41*** join/#asterisk lorinc (n=ang@pool-2522.adsl.interware.hu)
08:52.25JTtzafrir: i'd say session border controller is much more likely
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09:01.21Dovidtzafrir: a router in a DC
09:01.38JTan sbc is not quite a router
09:01.55Dovidwe are having lots of NAT issues as well as I understand from an SBC we can have it re-route the RTP stream even if behind NAT
09:02.01DovidJT so SBC = ?
09:02.25coppiceSBC == session border controller == VoIP box with VoIP box crossed out, SBC written in crayon, and the price increased at least 5X
09:05.26JTexactly
09:05.30JTsbcs are for suckers
09:07.20coppiceSBCs are for innovators. people at the cutting edge. people who need enterprise scalability.... oh, sorry, you just said that
09:08.17Dovidhehe
09:08.26Dovidso an SBC wont help me at all ?
09:09.00Dovidmy bigest issue is 1) NAT and 2) RTP Stream - I want it to by pass the server even if the calling client is behind NAT
09:12.50JTDovid: no
09:12.55JTDovid: you are dreaming
09:13.11JTif you don't want to deal with rtp traffic, get out of the voip business :)
09:19.58sylenat is not a big issue, but if you don;t want to handle rtp traffic, you gonna loose CDR records, so you can;t bill crap to anyone
09:20.41JTand end users may get nat problems too
09:27.37DovidJT: why can astrisk just handle the set up and tear down of the call ?
09:27.47*** join/#asterisk waptaxi (n=waptaxi@stat-5-160.e-sky.ru)
09:29.26JTbecause nat exists
09:29.37JTand i assume you want accurate billing data
09:34.50DeadlyDatawhat would this error mean if I was attempting to connect to my asterisk system usng SIP, Rejected registration for 'CENSORED' with cause 'bearercapability notauth'
09:43.46*** join/#asterisk DragoraN (n=dragoran@217.67.19.74)
09:43.47DragoraNhi
09:44.08DragoraNdoes this work? if not, how can i do this? exten => 20,1,SetGlobalVar(PARTY_EXT=${EXTEN})
09:44.26DragoraNi want to save originally called extension to variable for using in other context
09:44.37DragoraNnot context, sorry, on other extension
09:44.46DragoraNwhich i jump to with goto
09:46.40DragoraNplease
09:48.55Dovidhuh ?
09:49.04Dovidu want to set the exten of the person that is calling ?
09:49.13Dovidy would u do global ?
09:49.26Dovidu can use just Setvar and use it any where in the system on the same call
09:49.54*** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il)
09:50.58DovidShalom Yonah
09:56.51YonahW-WorkShalom Dovid, How are you
10:03.01Dovid:)
10:03.13Dovidani roeh she atah me-yisrael ?
10:04.23YonahW-WorkI am (I believe this is an English only channel)
10:04.28YonahW-WorkI see you are a local as well
10:04.54Dovidyup
10:05.02Dovidthere is also #asterisk-il
10:05.35YonahW-Workdid not know that! is it active at all?
10:05.57Dovidhere and there
10:06.06Dovidnot many of us on at once
10:06.17YonahW-Workso I see
10:06.49YonahW-WorkI am on the mailing list as well but dont really see any action there
10:07.19YonahW-Workwere you at the IsrAsterisk conference a few months back?
10:07.38Dovidnope. didnt know about it :(
10:07.53Dovidwhere was it ? and i asume it was hebrew only which I am not the best at
10:08.02*** join/#asterisk YonahW-Work (n=YonahW-W@genie03-173-74.inter.net.il)
10:08.14Dovidwhere was it ? and i asume it was hebrew only which I am not the best at
10:08.51YonahW-WorkIt was in Petach Tikva at the IBM building and was mainly Hebrew with parts in English
10:09.15Dovidargh !!!
10:09.22Dovidwhen was it ? and how many people showed up ?
10:09.29DovidI know VON is coming up - i hope to be at it
10:11.06YonahW-Workfunnily enough I was there and don't remember when exactly it was
10:11.23YonahW-WorkI want to say around April and there were probably 50-100 people
10:11.29YonahW-Workwhen is VON?
10:11.33Dovidhttp://www.von.com/israel07/
10:11.46Dovidin oct.
10:12.02Dovid14-15th of Oct.
10:12.22Dovidwanted to go to astricon in the US but it is on Sukkot
10:12.25YonahW-Workcool October is a good time for me
10:12.29Dovidsame here
10:12.30YonahW-Workyeah I noticed that
10:12.46Dovidmaybe in 2 years
10:12.58Dovidor i will go to the one in Europe
10:15.23DragoraNhow can i capture entered digits?
10:16.13*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
10:17.11YonahW-WorkDragoraN: check out the IVR page on the wiki http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu
10:17.31DragoraNthx
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11:03.26DragoraNhow to play music on hold just in background and let ringing the calling party?
11:03.41DragoraNwhen part will answer, the moh will stop..
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11:07.27Dovidanyone know how I can change the packetization time for g729 ?
11:07.59YonahW-WorkDragora: try looking into the Background cmd
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11:36.36DragoraNYonahW-Work: thx
11:50.06YonahW-Workanyone know how I can figure out why 1.4.5 is not building chan_zap.so?
11:50.34DrukenLPYis your zaptel current?
11:51.38YonahW-Workzaptel 1.4.3
11:56.23tzafriris it installed?
11:57.11YonahW-Workyes, I am currently installing again
11:57.46tzafrir./configure
11:57.51tzafrironce again...
12:01.55YonahW-Worklets see how it goes
12:04.20YonahW-Workok compiled no errors and I see chan_zap.so
12:04.34tzafrir_laptopYonahW-Work, look at build_tools/menuselect-deps in the asterisk source tree
12:05.08tzafrir_laptopthis is how menuselect figures out if a certain external dependency (E) is available
12:05.15YonahW-Workcool thanks for the tip, I will look into this further
12:05.34tzafrir_laptopit is updated by ./configure
12:05.58DragoraNhow do you call in english when someone just rings you once or twice then he hangs up?
12:07.25YonahW-WorkDragoraN: I think you mean a prank call
12:08.18DragoraNyeah, thx
12:09.17*** join/#asterisk gardo (n=gardo@121.97.178.163)
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12:36.36sof76Hi all, does anyone know how to install asterisk wihout installing zaptel in a clean way? Is there an option to give during asterisk installation?
12:37.36Strom_Mzaptel and asterisk are completely separate packages
12:37.56Strom_Myou don't even have to download zaptel to build asterisk
12:38.40sof76thank you
12:39.08Strom_Mdon't thank me - thank obviousness
12:41.38DrukenLPYnow Strom_M, be nice... not everyone has the brain power of a common mouse
12:43.03Strom_Mthe hi/cocks protocol
12:44.18*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
12:45.08DrukenLPYStrom_M: you know if anyone has had any luck with a voice activated ivr?
12:47.11Strom_Myeah
12:47.22Strom_Mlumenvox apparently makes it easy
12:47.51DrukenLPYyeah? will have to check that out
12:48.40*** join/#asterisk Splat (n=splat@home.heehawhills.com)
12:56.35DragoraNhow to detect something like this 555444#200
12:56.43DragoraNto get #200 from the number
12:57.03Strom_MDragoraN: during an IVR menu?
12:57.09DragoraNno
12:57.18DragoraNwhen calling for example from my cell phone to my asterisk
12:57.32DragoraNi want to route to other destination when #200 is entered at the end of number
12:57.35Strom_Mare you dialing # before you set the call up?
12:57.50DragoraNno
12:57.55DragoraNor 555444p200
12:58.10Strom_Mgrr, answer my question
12:58.15Strom_Mat which point do you dial #200
12:58.22Strom_Mafter asterisk answers, or before?
13:00.19Strom_Mi guess "at what point in time are you sending DTMF digits?" is too difficult a question
13:01.18*** join/#asterisk CVirus (n=GoD@196.205.193.169)
13:11.23*** join/#asterisk cappiz (n=cappiz@c85-196-102-30.static.sdsl.no)
13:11.42cappizis it possible to make the "on hold music" not restart after a user has been announced his/her queue number?
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13:40.01DrukenLPYuhmm... why?
13:40.10DrukenLPYyou want them to just sit there in silence?
13:41.09cappizwell
13:41.23cappizcant it pick up from where is stoped?
13:41.42cappizkind of bad to only hear 30 seconds of the song
13:41.47cappizover and over again
13:42.02DrukenLPYoh...
13:42.14DrukenLPYyeah, it's in the settings i guess... i've never had that problem
13:42.35DrukenLPYmine stays at it's current position until someone else goes into the moh
13:42.55cappizhum.. settings under what? the music on hold or the queue
13:43.06cappizwel, i tried with 2 in queue
13:43.18DrukenLPYmusic on hold....
13:43.24cappizand each one of the callers heard the song for only 30 seconds,
13:43.25cappizok
13:43.31DrukenLPYmoh has nothing todo with the queue
13:43.40cappizok
13:44.03cappizcould it be that i only have 1 song in the moh category?
13:46.34DrukenLPYwell, that's kinda dumb... hehehe yeah you should have a minimum of 2
13:46.41cappizok
13:46.41cappiz:>
13:46.55DrukenLPYeven if it's the same song
13:48.31*** join/#asterisk zotz (n=zotz@24.244.163.157)
13:52.49DrukenLPYpersonally, i have several moh contexts...
13:53.09cappizhehe
13:53.11DrukenLPYof course,... i also do multipul ivr's on a single system... and seperate pbx's
13:53.23cappizyeah... works after i uploaded more than one song :)
13:53.30cappiznow it picks up
13:53.31cappiz:)
13:53.33DrukenLPYwholly crap :)
13:56.16cappizis it possible to make freepbx/asterisk record calls done from DIAS?
13:58.39UatecWHY AM I IN WORK ON SUNDAY?!
13:58.56cappizhehe... you should know :P
14:01.37DrukenLPYUatec: because your a bitch?
14:03.12Dovidhehe. i work on sunday too. goto bring home the money
14:05.01*** join/#asterisk coppice (n=chatzill@163.201.17.210.dyn.pacific.net.hk)
14:06.53*** join/#asterisk _WildPikachu_ (n=WildPika@about/linux/staff/wildpikachu)
14:07.13_WildPikachu_I'm trying to strip down the number of modules loaded by asterisk, is there a simple list I can explicitly define to load?
14:07.19_WildPikachu_I think i'm missing a few from my list
14:07.45Strom_Mlook in /var/lib/asterisk/modules
14:07.56_WildPikachu_yea ... I'm loading about 10 so far
14:08.20Corydon76-homes/var/usr/
14:08.21_WildPikachu_but when I try dial ... I get engaged without any logs generated, think I'm forgetting something
14:08.24Strom_Mer yes
14:08.34_WildPikachu_one sec, I paste my list on pastebin
14:08.34Strom_M/usr/lib/asterisk/modules
14:08.43Strom_Mit be early here
14:08.58Corydon76-homeYes, what are you doing up at 7am on a Sunday?
14:09.04Strom_Mjet lag
14:09.09_WildPikachu_http://pastebin.ca/587330
14:09.18Corydon76-homeAh...
14:09.20_WildPikachu_anything obvious I'm missing?
14:09.48Strom_Mthe CDR module, i'd imagine
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14:16.11_WildPikachu_ah
14:16.41_WildPikachu_hectic, theres a whole bunch of cdr modules
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14:19.39Corydon76-home_WildPikachu_: func_cdr
14:19.51_WildPikachu_yea ... was trying out one by one
14:22.22_WildPikachu_hrmmmmm
14:22.22_WildPikachu_i'm missing something else
14:22.35_WildPikachu_i dont think my extensions.conf is getting loaded
14:23.13Corydon76-homeWhy aren't you using autoload=yes ?
14:24.23Corydon76-homeI would suggest an alternate approach... Use autoload=yes, then strip out various modules with noload => whatever.so until something breaks
14:24.35Corydon76-homeextensions.conf is loaded with pbx_config.so
14:25.08_WildPikachu_well ... I only want what I need, and would like to understand what each module does  :)
14:25.28Corydon76-homeformats are needed to load sound files from disk
14:25.43_WildPikachu_doubt i'm going to need that
14:25.56_WildPikachu_i basically just want to route voice calls comming into this server to various providers
14:26.02Corydon76-homeYou aren't playing messages (such as pbx-invalid.gsm)?
14:26.08_WildPikachu_nope
14:26.14_WildPikachu_(not yet)
14:26.31Corydon76-homeFormats are also needed for recording
14:26.38Corydon76-homeor monitoring
14:26.40_WildPikachu_don't need that yet  :o)
14:26.54Corydon76-homecodecs are needed to translate
14:27.25_WildPikachu_yea, figured that out and the formats .... just trying to get something to show up in the logs now
14:27.31_WildPikachu_let me try the pbx_config quick
14:27.37Corydon76-homeWell, if you're not playing anything from disk, then you don't need app_playback
14:28.17_WildPikachu_excellent, I have some logs  :)
14:28.20_WildPikachu_let me go play around
14:34.53_WildPikachu_hrmmm, Corydon76-home quick question ... which format_*.so provides ulaw btw?
14:35.04Corydon76-homepcm
14:35.27_WildPikachu_:)
14:41.47_WildPikachu_btw, where are asterisk sounds supposed to be installed?
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15:01.37Crimethinkermine are in /var/lib/asterisk/sounds
15:03.25*** join/#asterisk kusznir (n=kusznir@66-233-138-60.lew.clearwire-dns.net)
15:03.47rob0No they're not. Weasels have eaten those sound files.
15:04.57*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
15:05.21syleferrets are bad ass to
15:05.56Strom_Mweasels have eaten your ferrets
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15:15.55kusznirHi:  I'm having trouble with setting outbound caller ID info.  I'm getting the error:
15:15.58kusznirJun 24 08:15:00 ERROR[2343]: pbx.c:1437 ast_func_write: Function CALLERID not registered
15:16.37Strom_Mwhat line of dialplan code are you executing?
15:16.43kusznirThe line generating that error is: exten => _X.,1,Set(CALLERID(num)=<my num>)
15:16.55Strom_Mwhich version of asterisk?
15:17.04kusznirasterisk 1.2.16
15:17.08Strom_Mthat's odd,
15:17.39Strom_Myou compiled from source, right>
15:17.40Strom_M?
15:18.02kusznirYes (through a build-kit -- OpenWRT embedded linux)
15:18.19Strom_Mhm
15:18.43Strom_Mi don't know if they've slimmed it down so severely so as to not include the callerid function
15:18.55kusznirMost of the app_* stuff was not installed to save space, but I didn't see any that looked related.
15:19.17kusznirI grabbed the app_setcallerid.so, but it appears that's deprecated.
15:19.33kusznirWhere would the callerid function be included?
15:20.14Strom_Mfunc_callerid.so perhaps?
15:20.43*** part/#asterisk lukketto (n=lukketto@host45-89-dynamic.1-87-r.retail.telecomitalia.it)
15:24.22kusznirYep, that was it.  I didn't have any func_* modules on my client, so I didn't even notice that catagory of modules existed.  Thanks!
15:25.23`Seanwow
15:25.27`Seanmust suck using it on wrt tho
15:25.42`Seanseeing how you wont have half the functions and not to mention no way to install a FXS port
15:30.55kusznirI was looking to use it for IP phones only.  I don't have enough CPU power to run the codec's or do inline DTMF, but otherwise things seem to work well.
15:31.07kusznirI'm only running 3-5 phones from it anyway.
15:31.49kusznirI think its really cool how much I am able to get from such a small box!  (I haven't tried musiconhold yet, but that is part of the minimal install, so it might be doable.)
15:31.56kusznirI do have voicemail working.
15:34.15*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
15:34.30kusznirI do have another question for everyone:  I'm having issues with the timestamp on voicemail.  The embedded system doesn't have /usr/share/zoneinfo, but it does have its zone info in /etc/TZ (uclibc-based).  Asterisk console and the rest of the system properly show localtime, but voicemail is always in UTC.
15:36.18Strom_Myou have to set the time zone in voicemail.conf
15:36.34Strom_Msince your voicemail users can be in multiple time zones
15:36.45kusznirI've attempted to...I'm trying some more tests now with other formats.  tz=pacific did NOT work.
15:37.06Strom_Modd, because it works everywhere else
15:37.13Strom_Myou're sure it's in the right spot in the file?
15:37.26kusznirDo I have to reload anything on asterisk to make voicemail.conf changes take effect?
15:37.35Strom_Mreload app_voicemail.so
15:37.40tzafrir_laptop<PROTECTED>
15:37.41[TK]D-Fenderkusznir, You did this on the box definition line?
15:37.49kuszniryes
15:38.44kusznir70 => <PIN>,Jim Kusznir,kusznir@kusznir.net,tz=PST8PDT
15:38.54kusznirI previously had tz=pacific
15:39.04Strom_Mwrong spot
15:39.21Strom_Mpin,name,email,pager_email,options IIRC
15:40.27Uatecwhat extension should i use if i want to catch all other numbers that i haven't already specified extensions for?
15:40.32Strom_M_X.
15:40.41Strom_Mor _X!
15:44.44Uatecapparently it can't match that
15:44.45Uatec?
15:44.58Strom_Mwhy not?
15:45.08Uateci don't know
15:45.13Strom_M_X! will match anything 1 or more digits
15:45.15Uatecoh wait
15:45.17UatecDAMMIT
15:45.19Uatecof course
15:45.23Uatec1 or more digits
15:45.30Uateconly works if your phone company aren't retards
15:45.43Strom_Mlet me guess: GTE
15:45.47Uatecnah
15:45.49UatecNTL
15:45.52Uateci'm in the UK
15:45.58Uatecthey're presenting 0 digits on this line
15:46.04Strom_Mhahahahahaha
15:46.17Strom_Mmatch on s then
15:46.37kusznirOn the voicemail timezone issue:  If I use pacific as my timezone, I get disconnected with a "file vm-received not found".  If I remove the tz or set it to PST8PDT, it finds vm-received just fine.
15:46.44*** join/#asterisk soylentgreen (n=fgast@193.238.89.34)
15:46.48kusznirpacific is defined in zonemessages as: pacific=America/Los_Angeles|'vm-recieved' Q 'digits/at' IMp
15:46.51Uatecstill doesn't want to match s
15:47.45*** join/#asterisk soylentgreen (n=fgast@193.238.89.34)
15:48.30kusznirOk, corrected the typo, and it now plays back :)  Unfortunately, its still using UTC.
15:49.08kusznirMy guess is the difference between glibc and uclibc's handling of timezones.
15:53.15*** join/#asterisk Splat (n=splat@home.heehawhills.com)
16:04.22Uatecwhen dialing out with misdn, what does: mISDN/4-u7
16:04.24Uatec<PROTECTED>
16:04.33Uatecwhat's the 4 and the u7?
16:05.19ai-awho / what told you to use 4-u7 ?
16:05.44*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)
16:06.04Uateci didn't
16:06.06Uatecthat's from my log
16:06.37Uateci used misdn/g:isdn/${EXTEN:1}
16:06.58Uatecthe u number appears to be incrementing every time i dial
16:07.17ai-aline number used ?
16:08.01Uatecthat's what i thought initially
16:08.10Uatecbut i've got 4 lines plugged in
16:08.22Uatec3 of them are flashing red, which i took to mean that they were not working
16:08.35Uatecand i get 4- and 2-
16:08.43Uatec*sigh*
16:10.15*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
16:11.41*** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
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16:17.23UatecFUCK
16:17.29Uateci just rebooted
16:17.32Uatecand it's all gone
16:17.35Uatecall my hardware
16:17.39Uatecraid array failed
16:17.40*** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
16:22.50sylethat'll teach ya to check your system logs
16:23.13syletry loosing 3 years worth of programming, not fun
16:26.19syleso replace the 2 downed drives, reinstall OS, restore backups, reconfigure system, 5 hours maybe, go for a beer
16:27.21Uatecsyle, how can i check my system logs
16:27.27Uatecthe array is fucked
16:27.35Uatec5 hours work
16:27.36Uatecyoure right
16:27.40Uatecon a fracking sunday night
16:27.41Uatecfor no reason
16:30.01syleshit happens dude, its how you deal with it that makes you a person
16:30.52sylecourse if you just needed to vent that works to :)
16:32.54syletell ya one thing though, i;d deal with a downed array anyday of the week instead of going through a divorce again lol
16:33.58*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
16:34.17[TK]D-Fender"Divorce : a word derived from latin meaning 'to rip a man's heart out through his wallet'"
16:35.55Strom_M"Polycom: a word derived from Algonquin for 'telephone which takes five minutes to reboot'"
16:39.07*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
16:39.33syleChild Support: way to do it for next 18 years
16:40.15*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
16:42.50Uateclol
16:42.53Uateci'm fortunate
16:42.57Uatecnoone can divorce me
16:43.42*** join/#asterisk Greek-Boy (n=g@196.45.144.42)
16:43.46*** join/#asterisk waKKu (n=worth@unaffiliated/wakku)
16:44.42Greek-Boywhat is an open-wire infrastructure?
16:44.49Greek-Boyfirst time i hear of that in telecoms...
16:45.27Strom_MGreek-Boy: open-wire carrier as opposed to twisted-pair insulated wiring
16:46.00[TK]D-FenderStrom_M, Exaggerated... takes a hair over 2 to boot :)
16:46.00Greek-Boyhmmm
16:46.05Greek-Boyi've tried to google it
16:46.12UatecStrom_M, you mean cables that are metal open to the air?
16:46.51Strom_Msigh
16:46.53Greek-Boyno results, I'd like to see some technical specs of open wire
16:47.10Strom_Mwelcome to #asterisk, where you apparently don't need to know a thing about telephony to screw up a phone system
16:48.09*** join/#asterisk Greenbox (n=Brett@c-68-59-20-153.hsd1.sc.comcast.net)
16:48.33Uatec....
16:49.25syleto only be a cunt hair of an error to make the whole thing work again
16:51.28Uateclike your expensive new raid mirror which appears to be less stable than the cheapo 10gig hdd you pulled out of your PC you had 7 years ago
16:53.02sylebut we can all agree on at least raid mirrors being more stable than some of the women out there
16:54.44Uatecoh yes
16:56.05*** join/#asterisk javar (n=javar@69.79.134.24)
16:58.34sylethen again you might be lucky and just power supply or raid controller died
16:59.35*** join/#asterisk Metfan2007 (n=metfan@189.146.139.155)
17:00.31Metfan2007Hi, I have installed a Asterisk 1.4.5 and Zaptel 1.4.2 but I cannot use any ZAP command in the Asterisk CLI
17:00.35Metfan2007any clue?
17:01.57russellbyou have to install zaptel before asterisk
17:02.46*** join/#asterisk marv (n=Tim@c-71-228-189-127.hsd1.al.comcast.net)
17:03.27marvwhat was the url to the latest iax2 rfc draft?
17:03.57russellbmarv: http://tools.ietf.org/id/draft-guy-iax-03.txt
17:03.58syleanyone actually running 1.4 branch in production?
17:04.23russellbplenty of people are
17:05.56Uatechey, in SIP.conf is is possible to allocate multiple values to the mailbox?
17:06.24russellbUatec: no
17:06.26sylewhy
17:07.07russellbjust kidding, yes
17:07.19marvrussellb: thanks, for some reason google wasn't finding it, it found the iax enum draft though
17:07.22russellbyou can put multiple mailboxes, separated by commas
17:07.50russellbmarv: yeah, i always search buy "draft-guy-iax"
17:07.53russellbby*
17:08.22russellbUatec: you can put multiple mailboxes separated by commas
17:09.14russellbwhich you would have seen if you had looked in sip.conf.sample ...
17:09.16Metfan2007anyone has compiled succesfully sangoma cards with zaptel 1.4.2?????
17:10.24russellbmaybe a few more question marks will help :-p
17:11.14marvwhy is the word wikipedia in brackets after the first sentence of section 1.1? format conversation error?
17:11.26Metfan2007hehe, sorry, I am frustrated
17:11.35*** join/#asterisk gardo (n=gardo@121.97.208.93)
17:11.41russellbmarv: i have no idea ...
17:12.08Metfan2007someone using sangoma digital cards here?
17:15.53*** join/#asterisk Greek-Boy (n=g@196.45.144.42)
17:17.23sylehaven;t tried 1.4 yet, worth it?
17:17.24Uatec:(
17:17.26UatecP[ 2] GOT IGNORE SETUP
17:17.26UatecP[ 2] CC_RELEASE_COMPLETE|CONFIRM [TE]
17:17.26UatecP[ 2] release_chan: Ch not found!
17:17.39Uateci'm getting that whenever i dial in to my B410P BRI isdn card
17:17.45Uateci have no idea what it means
17:17.49Uatecwhat does it mean?
17:17.54sylesuppose i;d have to recode all my c modules now, hopefully not much has changed
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18:10.30__DAWerrr
18:12.02DeL3e7yea
18:13.53Strom_Mcocks
18:14.01russellb...
18:14.12Strom_Mi mean pones
18:14.14Strom_Mphones
18:14.46DeL3e7this is too mutch for me to handle
18:15.00Strom_M?
18:15.12Strom_Mi mean, I know I'm fun to hug and all, but wow
18:15.33DeL3e7i mean asterrisk
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18:15.35*** mode/#asterisk [+o anthm] by ChanServ
18:15.48Strom_Mah
18:15.51Strom_Mwell that's different then
18:15.57Strom_Mand that's why we're here! :)
18:16.01DeL3e7its installed on my linksys
18:16.09Strom_Mwrt54g?
18:16.12DeL3e7yea
18:16.14DeL3e7GS
18:16.50Strom_Mive got one of those that runs asterisk
18:16.58Strom_Mor that ran asterisk anyway
18:17.05Strom_Mi dont think i've installed it on the latest reflasg
18:17.07Strom_Mreflash
18:17.13DeL3e7what could i use it for a gateway?
18:17.29Strom_Mfor a gateway for what?
18:17.50DeL3e7a main asterisks box with alittle more beef
18:17.58Strom_Msure
18:18.10DeL3e7i dunno ive never used this software before
18:18.12Strom_Myou could use it as an IAX/SIP gateway if you configured it right
18:18.19DeL3e7im reading the huge pdf's
18:18.30Strom_Mbut I'd say start with a basic setup and then go from there
18:18.33Strom_Mdo you have thebook?
18:19.07DeL3e7asterisksTFOT.pdf
18:19.21DeL3e7http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
18:19.25Strom_Mwell, s/asterisks/asterisk/
18:19.29Strom_Mbut yes, that's a good book
18:22.34DeL3e7kewl asterisk -c
18:23.33DeL3e7http://voip-info.org/wiki/
18:24.54Uatechey, does anybody know how to use the distinctive ring feature on the linksys SPA922?
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18:54.51mikpelhow can a legacy phone from ericsson be connected to asterisk, and stil use its functionsbuttons?
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18:59.03Cyber-Doggso... if I wanted to have my system make calls out to someone... what am I looking for?
19:04.05Greek-Boyany gsm network techies around?
19:04.10Uatechaving just finished rolling out a 16 phone asterisk system, but not having a clue how he did it
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19:15.48skoenmanhey all
19:15.57Cyber-Dogghowdy
19:16.02*** part/#asterisk MikeJ (n=MikeJ@d149-67-175-107.try.wideopenwest.com)
19:16.29skoenmancan anybody tell me if its possable to use asterisk to make a pbx with software phones???
19:18.12waKKuexactly what it is
19:18.39skoenmanhe???
19:18.57skoenmank so you can have the pbx and use software phones with it???
19:19.13russellbyes
19:19.26skoenmancool whats the softwares name ???
19:19.35skoenmani have been looking but theres nothing
19:23.14skoenman???
19:24.21Iamnachox-lite, idefisk are 2 that i like
19:24.35skoenmank cool
19:24.41skoenmando they run on windows???
19:24.47Iamnachoyes both of those do
19:24.54skoenmank cool
19:24.55Iamnachoi think idefisk runs cross platform
19:24.58skoenmanthanks
19:25.28Iamnachoyou should look at this page http://www.voip-info.org/wiki-Asterisk+phones
19:25.48Iamnachodown a ways is a soft phone thing
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20:06.34NOT_guruhello  anyone here using a linksys pap2? how well does it work? are you using both ports as 2 different extensions?
20:07.14NOT_guruI am thinking of getting a couple of these and breaking my house into 4 "zones"
20:07.16waKKuNOT_guru: yes
20:08.07NOT_guruwonderfull waKKu  I have played with a grandstream a buddy had and didn't like it much...  are the linksys ones pretty straight forward?
20:08.09waKKueach linksys pap2 haves 2 FXS ports.. u can put 2 differentes numbers and, of course, 2 atas
20:08.57waKKuNOT_guru: yeah.. i believe linksys a best choice for this kind of equip ;)
20:09.14waKKuhowever.. linksys is Cisco SOHO
20:09.17NOT_guruwonderfull  thanks man  I appreciate the information
20:09.33*** part/#asterisk mikpel (i=mikpel@h-195-210.A183.cust.bahnhof.se)
20:09.40NOT_guruI found a place I trust that has some for $25
20:09.54Qwelllocked, no doubt
20:10.22NOT_guruI don't think so  but I guess I will find out
20:10.31waKKuuow.. NOT_guru $25 is cheaper SO MUCH...
20:10.48*** join/#asterisk swift99 (n=dave@ip70-178-9-136.ks.ks.cox.net)
20:10.51NOT_gurueither of you wanna look at the link?
20:11.00waKKui have seem some pap2 for $100 $120 ...
20:11.02NOT_gurusee what you think
20:11.06waKKuok
20:12.00NOT_guruyou get that?  and if so  please let me get mine purchased before you spread that  =)
20:12.26waKKuyeah.. its really a linksys pap2 ... very very cheap..
20:12.28swift99Would someone be willing to walk a newbie (me) through some of the pros and cons of an asterisk based system for a small office (12 phones)?
20:12.48NOT_gurulooks unlocked to me
20:13.28waKKuNOT_guru: if I not live in brazil.. i'll purchase a dozen :P
20:13.35NOT_guruLOL
20:15.29QwellNOT_guru: link?
20:16.14NOT_guruagain  please refrain from spreading that till tomorrow
20:16.21DragoraNwhat softphone on windows do you use?
20:16.24NOT_guruI just have to see if my buddies want some before I buy
20:16.35NOT_guruI personally use x-lie
20:16.41NOT_gurux-lite
20:16.46_VoiceMeUp_COMxlite is ok
20:16.48_VoiceMeUp_COManyone tried bria ?
20:16.52_VoiceMeUp_COMfrom same company
20:16.59waKKuDragoraN: idefisk is a good choice, and free
20:17.06DragoraNyes
20:17.12waKKuand.. one of few that support IAX
20:17.22NOT_guruI am willing to try anything, as I am looking to buy some softphone software
20:18.09NOT_guruunless I find something wonderfull and free
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20:18.27swift99I've just been asked to help choose and configure a small office PBX, from the ground up.  At this point, I don't know what questions to ask.  Asterisk has been suggested as a possible solution.  Would someone with experience be willing to bring me up to speed on what questions to ask?
20:18.36swift99Thanks!
20:18.50_VoiceMeUp_COMfailover/emergency
20:19.03_VoiceMeUp_COMthen backup solution.. example net down
20:19.19*** join/#asterisk dagrim (n=_dagrim_@unaffiliated/dagrim)
20:19.42_VoiceMeUp_COMso for tooll frees you could setup an alternative btn for them in case pri down
20:19.42_VoiceMeUp_COMetc
20:19.51waKKuswift99: use asterisk and be happy :)
20:20.09swift99What kind of equipment would we need to purchase?
20:20.15waKKuasterisk was made to do exactly what u want..
20:20.18_VoiceMeUp_COMthen theres always the budget and , does anyone live byt he phones ? .. example.. would you set up an hospital emgency reposnce unit with asterisk ?
20:20.40waKKuswift99: this depends.. what type of equip. do u want use in the end's ?
20:20.43_VoiceMeUp_COMso how vital is the system in first place, and do they want 80% up time or 99.99999
20:20.56swift99_VoiceMeUp_COM: I'll ask my neighbor - he manages the hospital network   :o)
20:20.58waKKuATA's ? Softphones ? did u have an PABX ?
20:20.59_VoiceMeUp_COMactualy i think 99.99999 is too much
20:21.09_VoiceMeUp_COM5 9's = 1 sec downtime in 1 year
20:21.19swift99Right now, this is a brand new building.  There is nothing.
20:22.05swift99We have limited resources to support the tool, but the resources we do have are world class.
20:22.30swift99Ideally, we want the tool to just run like any turnkey PBX that we would purchase.
20:23.09swift99I was hoping to run it on two midsized ubuntu servers, with a  load balancer to provide automatic failover.
20:23.33swift99What kinds of phones and telco interface hardware would we need?
20:23.58swift99... for an ideal configuration, that is.
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20:26.29swift99By nothing, I mean two walls and a roof have been built.  We need to decide what sort of internal phone system to use between the roof going up andthe ceiling going in.
20:27.33swift99waKKu: One of the things we need to decide are what we want on the ends.  I was thinking cisco VOIP phone.
20:28.17sof76Hello, does someone know why when I launch safe_asterisk, sometimes the stop command is not recognized in the asterisk console?
20:28.32swift99But, I'm an apps programmer, not a telecom tech.  I don't know what the alternatives are for end points.
20:29.30waKKuswift99: well.. if u really choose for IP-Phones and SIP protocol, u wont need any other hardware on asterisk..
20:30.06swift99What about connection to the telco side?
20:30.13waKKubut if u need comunicate asterisk with other PBX, u'll need some PCI card (E1 or ISDN or FXO ports) ..
20:31.33swift99At this point the simpler the system, the better it is.  We have *nix admins and network techs, but little telecom experience in our team.
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20:34.36swift99waKKu: what do you recommend for connecting to the telco side of the pbx?
20:35.59waKKuswift99: i cant say.. i dont know ur structure and i dont live in your country (probably.. i'm from brazil)... Here we have some PBX with R2 or E1 connectors..
20:37.14swift99wakku: I'm USA.  So, I'd need to talk to my telco, see what connector they expect, then get the appropriate card for that connection, I guess.
20:38.39swift99waKKu: At that point, if we use SIP phones, we should be able to handle everything.
20:38.57swift99waKKu: what about using load balancers with SIP Phones?
20:39.46swift99waKKu: Would they work, providing an automatic failover, or would they mess up the assumptions of the system?
20:40.27waKKuswift99: hm.. really dunno.. i never used that..
20:40.46*** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com)
20:41.06waKKubut i thought that its possible .. :)
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20:51.40swift99How long would you expect it to take to configure a basic system with SIP phones, provided linux and asterisk were installed?
20:55.38waKKuthinking out of structure and users - one day, two on max
20:55.58bbryantswift99, there's not too much you have to do except add sip peers in sip.conf and config your phones
20:57.23*** join/#asterisk Dovid (n=Dovid@bzq-82-81-163-218.red.bezeqint.net)
20:58.12swift99Sounds good.  We'll have to set up a demo system.
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21:17.00*** join/#asterisk codey (i=codec@iglu.paranoid-penguin.de)
21:17.05codeyhi all
21:17.16codeyanyone using app-fax?
21:17.20codey(faxreceive)
21:22.56codeyhttp://slexy.org/paste/3235
21:23.00codeymaybe someone is able to help
21:24.09Dovidi donr know the agi. but it is failing after
21:24.09DovidAGI Tx >> 200 result=1 (default)
21:24.20Doviddid u look at the docs for it. maybe u missed a step some where
21:24.42Dovida hunch is this
21:24.56Dovidyou may need to put in exten => s,1,ANswer
21:24.58Dovidand then
21:25.14Dovidexten => s,2,,AGI(receive_fax,receive)
21:25.17codeywell, the docs say you need the part "[recipient foobar]"
21:25.23codeyand then faxrecipient has to be "foobar"
21:25.33Dovidas well as
21:25.33Dovidexten => h,2,Set(MY_CALLERIDNUM=${CALLERIDNUM})
21:25.33Dovidexten => h,3,DeadAGI(receive_fax,deliver)
21:25.46Dovidshould be h,1 and h,2 but that should not be affecting it
21:25.59Dovidgoobar needs to set
21:26.02Dovidits just an example
21:26.07codeyyes, i know
21:26.10codeyI've set it to "default"
21:26.19codey(i've tried a different name before - same result)
21:26.34Doviddo u know y its set to default ?
21:26.53Dovidalso are there any other packages that need to be installed ?
21:27.24codeyi've installed all needed perl-modules
21:27.28codeyand all the postscript stuff the agi uses
21:27.31Dovidmy hunch is that one of the variables sint set
21:28.46Dovidisnt set*
21:29.27codeythe FAXRECIPIENT= stuff doesn't matter as long as you have a section with the same name in the faxreceive.conf
21:29.39Dovidthat may matter
21:29.43sof76Hi everybody, I encountered a bug with asterisk, sometimes asterisk does not recognize the stop commands, it answers : No such command 'stop now' (type 'help' for help), can you help me please?
21:29.47codeyit *is* there
21:30.01codey(read my paste - i've included all the needed config stuff)
21:30.08Dovidu may need to set it differently. like i said i dont know the agi so i cant say for sure
21:30.13Dovidu can try emailing the users list
21:30.35Dovidsof76: what version ? i doubt its a bug. others would of reported it here
21:30.41Dovidcan u replicate it ?
21:31.20sof76I have the latest one, 1.4.5
21:31.33codeysof76: what kind of terminal do you use? maybe there are some strange sequences because you did a backspace or whatever (i experience this on my mac sometimes)
21:31.35Dovidcodey: what codec r u using ? i doubt thats ths issue
21:31.43Dovidsof76: can u replicate it ?
21:31.51sof76I have this problem only on asterisk installed on a virtual server without zaptel and libpri
21:31.54codeyDovid: RxFAX works fine for me
21:32.38Dovidcodey: also try agi debug. see if anything more comes up
21:33.03codeythe output of my paste IS agi debug
21:33.19sof76it does not happen automatically, but when it happens command stop does not exist on the help menu
21:33.41Dovidah ok. sorry i cant help u
21:33.46Dovidlike i said try the users lsit
21:33.53Dovidhmm
21:34.00Dovidif its not in help then it may be a bug
21:34.07Dovidtry asking one of the room ops when the come in
21:34.26sof76what is the room ops?
21:34.41Dovidany of the mods in the room
21:34.48DovidTK: also knows a lot
21:42.22sof76If I launch asterisk with asterisk -c, then sometimes it stops at this level : pbx_dundi.c:437 reset_global_eid: No ethernet interface found for seeding global EID  You will have to set it manually.
21:42.24sof76............................
21:42.41_VoiceMeUp_COMhey guys need a regex expert im a noob regex : sip:78|2@1.2.3.4:5060;blah=udp i need to extract the part between sip: and the | so 78 in this example could be from null to 8 chars
21:42.44_VoiceMeUp_COMif one can help
21:42.58sof76and then the stop commands are not in the help menu and not recognized by asterisk
21:43.09Dovidi am off to bed.
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21:52.58dudesAny suggestions on a good residential DID provider inside the US that is reliable and would run less than $60/mth per did?
21:53.41*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
21:58.29CrimethinkerI'd love to find one that actually bills with dead trees and a stamp at any price at all
21:58.46dudesyea I here you there
21:59.22Strom_Mi'd love to find an IRC user that can tell the difference between homophones
21:59.32dudessadly, I'm thinking about getting raped by Qwest and buying a TDM card /w a couple FXO ports
22:00.38Strom_Mdudes: what's your expected call traffic?
22:01.27dudesnot much
22:01.40dudesprobably like 10 calls a day at most
22:01.41*** join/#asterisk __DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
22:01.51dudesper DID that is
22:02.34*** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net)
22:02.47Strom_M"not much" doesn't tell me how many minutes
22:03.12dudesno more than 500 a month per DID
22:04.19*** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net)
22:04.26[TK]D-FenderStrom_M, homophones!??!?!  I run a *STRAIGHT* PBX thank-you!
22:04.55waKKuLOL
22:04.56waKKuhhaioeha
22:04.59Strom_Mhomophones are words that sound alike but which suck each other off in the men's restroom
22:05.12*** join/#asterisk kombi (n=kombi@213.160.14.18)
22:05.18[TK]D-FenderStrom_M, An apple a day keeps the mixed-metaphor away ;)
22:05.23Strom_Mheheh
22:05.35Crimethinkersnap crackle pop toyota?
22:05.38[TK]D-Fender<- literate bastard
22:06.02Crimethinkerdoes any voip provider bill with dead trees & stamp, and accept payment by same?
22:06.30Strom_Mdudes: teliax is $5 a month for the DID and 2c a minute for the traffic, which would put you at $15 a month per DID
22:06.53CrimethinkerI'm trying to deduct this crap as a business expense, but my accountant is unhappy with printouts of webpages as documentation
22:07.02kombiwhen CLI says: "No application 'MeetMe' for extension ()" when in fact there is one, what might that be?
22:07.07Strom_Mwell tell your accountant that it's the 21st century now
22:07.18__DAWCrimethinker:  PM me
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22:10.18*** join/#asterisk kombi (n=kombi@213.160.14.18)
22:11.33kombiI understand there has to be [rooms] conf=>123 and exten=>123,2,MeetMe(123), correct?
22:13.11*** join/#asterisk saftsack (n=saftsack@pD9E0453C.dip.t-dialin.net)
22:13.50[TK]D-Fenderkombi, You need to have zaptel installed BEFORE compiling *.
22:14.03[TK]D-Fenderkombi, Forget the conf.. your APP didn't build for lack of a timing source
22:14.47kombif* me.., just managed to compile ztdummy against sources, didn't know * needs remaking too.. thank Fender!
22:14.58kombithanks, I meant..
22:15.55[TK]D-Fenderkombi, NP, * needs to see Zaptel being ready becfoe compiling the app.
22:16.03[TK]D-Fenderbefore*
22:16.04[TK]D-Fenderkl;asjdkl;asjd
22:16.33[TK]D-Fender<- just in from a 5 hour bikeride around the West Island.
22:18.13kombithat being a racing bike?
22:19.12kombior street bike, thing your race on streets with with narrow tyres
22:19.44kombiwhat is it really called in your lingo?
22:20.49kombishould do make -j3..
22:21.28kombimaybe make clean wasn't necessary there..
22:29.20[TK]D-FenderHybrid
22:29.20kombiflat bar?
22:29.20[TK]D-Fendernot originally, but replaced custom, yes
22:29.20[TK]D-Fenderwas a "W" like hybrid bar, heavy street (light off-road capable) tires, T bar (loo long for me now, I need a shorter & taller riser post))
22:29.20[TK]D-FenderMore of a light off-road touring bike class now.
22:29.20[TK]D-FenderI'm totally f'n wiped right now...
22:29.21kombiso what is the proper name for racing bike for the street?
22:29.21[TK]D-Fenderkombi, damned if I know!  I just ride!
22:29.21kombilol.. still trying to pick up english better
22:29.21[TK]D-Fenderkombi, used to be the old "10-speed" which looks rather similar still.  But names are so grey these days
22:29.21kombisince there 30 speeds now..;)
22:29.21kombibtw, does it make sense to install progdocs?
22:29.21[TK]D-Fenderkombi, Your's is considerably better that some native english speakers I know :)
22:29.21*** join/#asterisk nephfl (i=nephilim@wsip-70-184-144-158.ga.at.cox.net)
22:29.21kombithankyou! (flattered)
22:29.21[TK]D-FenderI never installed any extra docs personally
22:29.21[TK]D-Fenderkombi, Whats your native tongue?
22:29.21kombigerman
22:29.21[TK]D-Fenderkombi, neato, any other languages you speak?
22:29.21JTvas is das?
22:29.21JTi mean hi
22:29.21kombiJT: was ist das
22:29.21[TK]D-FenderHaagen Dasz? ;)
22:29.21kombi..means What's that?
22:29.21kombilol..
22:29.21JTkombi: it's not vas?
22:29.21kombinope
22:29.21kombicommon misconception..;)
22:29.23JTis it at least pronounced as vas?
22:29.23kombialso hardly ever said as a phrase
22:29.23kombiJT: that is completely right
22:29.35*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)
22:29.47nephflanyone here setup a wrtp54g with asterisk?
22:29.59kombivas is das?
22:30.03*** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)
22:30.37kombiFender: some french, but just enough to get by without being beaten up
22:32.18kombiafter succesfully modprobing ztdummy and re-compiling *, should there be some meetme options in CLI?
22:33.39[TK]D-Fenderkombi, Excellent, trilingual = very cool.
22:34.00[TK]D-Fenderkombi, You should see it via "show application meetme"
22:34.49kombi"your application is not registered" ;(
22:34.51[TK]D-Fenderkombi, Now before touching * I would have suggested doing "modprobe ztdummy" (you DID specifically check that off in make menuselect, right?) and "modprobe zaptel"
22:35.56kombiFender: actually compiled it debian style with apt-get zaptel-sources and m-a a-i zaptel sources
22:36.43kombimodprobe ztdummy went without complains, maybe I try again with modprobe zaptel..
22:37.07kombioh, no, never mind, zaptel's loaded (lsmod)
22:37.33[TK]D-Fenderkombi, I really can't comment on your style of install, I've only done pure-source
22:38.11kombimake menuselect in /usr/src/asterisk you mean? (I've only ever done that to do with kernels so far)
22:38.24[TK]D-Fenderkombi,  in zaptel.
22:38.34kombioh, ok..
22:38.36[TK]D-Fenderkombi, in ther you can choose whic modules to complie
22:39.26kombimodules seem to have come out ok, at least they loaded are nicely listed in lsmod
22:39.30kombiand..
22:40.20saftsackhi, is a via 1ghz cpu capable of handling about 20 calls? (no codec tranlsation)
22:41.49kombidoing make menuselect in asterix shows meetme XXed out, so there is still in issue with zaptel..
22:42.53[TK]D-Fenderkombi, do a ./configure first
22:43.01[TK]D-FenderYou'll need to pick up your changes.
22:43.10kombijust did, still..
22:43.11[TK]D-Fendersaftsack, sure
22:43.15[TK]D-Fender:/
22:43.26kombiweired..
22:43.50saftsackhas the via board a long lifetime?
22:44.00saftsackor will they propably die after about 3 years
22:44.33[TK]D-Fendersaftsack, Oh, I presumes that 1 GHZ was on a POS CPU
22:44.45[TK]D-Fenderwasn't
22:44.48[TK]D-FenderC3 = crap
22:44.54[TK]D-Fendermore like a P550-600
22:45.10[TK]D-Fendersaftsack, It'll probably do the job but.... ICK
22:45.31saftsackoh ok. so it is more recommened to buy a soekris board for long life duration?
22:45.41[TK]D-FenderI can buy a 1 GHZ PC complete for 100$ easy these days with all sorts of extras...
22:45.57[TK]D-Fendersaftsack, Buy a real computer
22:46.16saftsackits for home usage ...
22:46.30saftsackand no ... i have no 20 persons at home :-P
22:46.40[TK]D-Fendersaftsack, So's mine, and a real PC will cost you less and survive more.
22:47.16saftsackwhy will it cost less? a normal pc takes about 100W
22:48.49[TK]D-Fendersaftsack, How much does a Soekris cost?
22:49.55saftsackabout 200$
22:50.04saftsack+ psu i think
22:50.32[TK]D-Fendersaftsack, I can get a real PC for under half of that which includes expansion options, and no extras required like flash, etc.
22:50.54saftsackbut with active cooling i think
22:52.07[TK]D-Fendersaftsack, sure, feel free to guesstimate your energy costs, effort, etc.  I gave you "my" math.  Go see how it applies to you.
22:52.22*** join/#asterisk Corydon76-home (i=two@pdpc/supporter/sustaining/Corydon76-home)
22:52.22*** mode/#asterisk [+o Corydon76-home] by ChanServ
22:52.35saftsackyes your calculation is cool but i dont think that your pc will life longer than my soekris
22:55.09kombisaftsack: Is that thing completely noiseless and easy to run * on?
22:55.22saftsacki have to find out ....
22:56.42JTpeople have run asterisk on soekris before
22:57.09JTin fact, some setups have a full pri with a suitable pci card (no transcoding)
23:00.15saftsackhow to get pci cards in this minipci port? are there adapters available?
23:00.40JTerr
23:01.29*** join/#asterisk obnauticus (n=admin@c-71-59-162-60.hsd1.wa.comcast.net)
23:01.29JTyou buy the soekris net4801
23:01.35JTit has a full pci slot
23:01.35saftsacki know that the soekris has a pci port on the side but what if i want tot use the minipci port? i saw an adapter but it was only available in the usa
23:01.51JTyes you can buy adapters online
23:01.54saftsacki have an ar7 here with a minipci port and i want to test this first
23:02.00saftsackwhere? :)
23:02.07JTwhat's the point?
23:02.11JTgoogle for them
23:02.21JTif you have a pci card... use the pci slot?
23:06.00saftsackno my router just has a minipci slot
23:06.12saftsacki got 1 entry in google but they dont sell :( they just produce
23:06.30JTwell you bought the wrong model
23:06.38JTwhat card do you want to attach?
23:07.37saftsackhfc-s card
23:07.55saftsackthis cards have a one byte fifo so the irqs in the router will be fullfilled but i want to give it a try
23:08.10JTyou can get minipci bri cards, but they cost a lot more than those hfc-s cards ;)
23:08.51saftsackthats the point ;)
23:08.57saftsackthose adapters are at 60 bucks
23:09.13JTso next time what willl you do... buy the correct soekris model? :)
23:09.42kombiFender: got it! debian's zaptel is no good..
23:09.43saftsacki havent no soekris ... i have an asus wl-500gp
23:10.51JTi see
23:11.28saftsackhttp://www.interfacemasters.com/products/pci_tools/im380/index.html
23:13.20saftsackhttp://www.interfacemasters.com/products/pci_tools/mini_pci_to_pci/index.html
23:13.29saftsackthis is what i want .... but where to buy? :(
23:21.36*** join/#asterisk remmo (n=junk@203.62.147.3)
23:23.10*** join/#asterisk Stridernzl (n=neville@125-239-186-131.jetstream.xtra.co.nz)
23:27.31dlynes_laptopsaftsack: how about here?  http://www.google.ca/search?hl=en&q=where+can+I+get+interface+masters+im300%3F&btnG=Search&meta=
23:29.37*** join/#asterisk lesouvage (n=lesouvag@62.140.137.125)
23:30.49lesouvage.
23:31.32lesouvagesorry,just trying irc client on e61
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23:47.15DeL3e7IAX
23:47.27DeL3e7how do i do that ;)
23:49.55lesouvagedel37e: what do you mean or looking for
23:50.35JTdlynes_laptop: i was more interested in performance on the Netra T1 than anything else
23:50.40obnauticushey umm
23:50.48obnauticusI got an inbound trunk on SIP
23:50.51obnauticusbut im answering through IAX2
23:50.52JTi'm sure installation isn't that hard unless you're trying to use a pci card
23:50.55obnauticus-- IAX2/obnautsoft-6 answered SIP/66.54.140.46-081d6690
23:51.04obnauticusI can talk TO the person who's calling me
23:51.08obnauticusbut they cannot talk to me
23:51.11obnauticusis there any reason why that is?
23:52.33JTnat or firewall
23:52.41obnauticusHow would i fix that ?
23:52.43obnauticuslike
23:52.47obnauticusim running a BSD based firewall...
23:55.03JTdepends on where the problem lies i guess
23:55.40lesouvageAbnauticus: udp 5060 and rtp 10000-20000 should be open .
23:56.10JTudp 10000-20000 you mean... rtp runs on top of that
23:56.14obnauticusI can only forward TCP and UDP
23:56.15obnauticuslol
23:56.24obnauticusudp 10000-20000
23:56.25obnauticuswow
23:56.27obnauticuslol.
23:56.51JTyou shouldn't need to forward anything unless people from outside are connecting to your asterisk as a server
23:56.56JTwhy all the lols?
23:57.01obnauticusi lol all the time
23:57.01obnauticusirl too
23:57.13obnauticusI'm 15.
23:57.17lesouvageIf you don t expect lots of connections you can lower the range
23:57.23JTmaking excuses now ;)
23:57.31obnauticusWhere does it start binding sockets?
23:57.31obnauticuslike
23:57.35obnauticus10000 and up
23:57.38obnauticusstarting at 10001
23:57.44JT5060 is bound for sip
23:57.51JTthe rest are chosen per call
23:58.06obnauticusim not running 10000 clients so umm
23:58.07obnauticusis
23:58.09obnauticus300 ok?
23:58.12JTerr
23:58.13obnauticus10000 10300
23:58.25JTwho really cares, unless you have a reason to, leave it at defaults
23:58.34obnauticusya
23:59.10obnauticusit still no workie
23:59.20obnauticus-- Hungup 'IAX2/obnautsoft-7'
23:59.20obnauticus== Spawn extension (inbound, 508, 2) exited non-zero on 'SIP/66.54.140.46-081d6690'
23:59.26JTumm
23:59.39JTi think the problem is probably in the asterisk configuration
23:59.47obnauticuscodec's?
23:59.51JTbut first of all, what is the setup functionally?
23:59.52JTno.
23:59.59obnauticusJust for stupid stuff

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