IRC log for #asterisk on 20080924

00:00.25bird_of_Luckjmardonesk: what does 'show function ODBC_PREGUNTA1' says?
00:01.48jmardonesk<PROTECTED>
00:01.49jmardonesk[Syntax]
00:01.49jmardoneskODBC_PREGUNTA1(<arg1>[...[,<argN>]])
00:01.49jmardonesk[Synopsis]
00:01.49jmardoneskRuns the referenced query with the specified arguments
00:01.49jmardonesk[Description]
00:01.51jmardoneskRuns the following query, as defined in func_odbc.conf, performing
00:01.53jmardonesksubstitution of the arguments into the query as specified by ${ARG1},
00:01.55jmardonesk${ARG2}, ... ${ARGn}.  The values are provided either in whole as
00:01.57jmardonesk${VALUE} or parsed as ${VAL1}, ${VAL2}, ... ${VALn}.
00:01.59jmardoneskThis function may only be set.
00:02.02jmardoneskSQL:
00:02.05jmardoneskINSERT INTO asterisk.encuesta (respuesta1, hora) VALUES ('${SQL_ESC(${ARG1})}',NOW())
00:03.38jmardoneskthe query is fine http://pastebin.com/d26b28664 for more info
00:05.06bird_of_Luckjmardonesk: can you post function definition in func_odbc.conf to pastebin.com ?
00:06.01jmardoneskbird_of_Luck, http://pastebin.com/d26b28664
00:09.52bird_of_Luckjmardonesk: try using Set(XX=${ODBC_PREGUNTA1(0)})
00:14.14jmardoneskbird_of_Luck, when i put that, I see the error [Sep 23 19:53:51] ERROR[5917]: pbx.c:1552 ast_func_read: Function ODBC_PREGUNTA1 cannot be read
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00:16.34bird_of_Luckjmardonesk: dsn 'astrealtime' exists and works ? 'module reload func_odbc.so' shows no warnings/errors?
00:18.19jmardoneskastrealtime works for me in the cdr, and the reload dont show any error
00:21.09bird_of_Luckjmardonesk: something is wrong in func_odbc.conf. See 'This function may only be set' in function desctiption
00:22.05phixjeev: awesome
00:22.29bird_of_Luckjmardonesk: for valid functions with read= 'function show ..' should report that in can be read
00:23.14bird_of_Luckjmardonesk: are you sure you posted real func_odbc config on pastebin ?
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00:29.13bird_of_Luckjmardonesk: you should get something like http://pastebin.com/m22df4df0 for function with read=.. and write=..
00:30.13Qwellyou can't insert on a read..
00:34.47bird_of_LuckQwell: odbc module is not interested in what SQL query do. It simply substitutes variables and execute query. Yes, you cant't get valuable result but you can do INSERT into db.
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01:23.49jmardoneskin read in can only select? and in write i only can insert or update?
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01:34.53drmessanoi do not how open the source
01:34.56drmessanohalp?
01:35.18Qwelldrmessano: Don't you have testing to do or something?
01:36.23jmardoneskbird_of_Luck, I dont understand whats mean "This function may only be set". I have in func_odbc.conf
01:36.25jmardoneskwrite=INSERT INTO asterisk.encuesta (respuesta1, hora) VALUES ('${SQL_ESC(${ARG1})}',NOW())
01:36.31drmessanoSetting back up my DNS for the 3rd time in 5 days
01:37.01jmardoneskand the context say: exten => 102,2,Set(ODBC_PREGUNTA1(0))
01:37.34drmessanoI did check all the VoIP spam tabloid websites and have heard no juicy astricon news
01:37.40drmessanoWTF is up with dat?
01:38.37jmardoneskwhen i put exten => 102,2,Set(${ODBC_PREGUNTA1(0)}) dont work, when i change write for read, dont work... I dont know what i need to do..
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01:42.49bird_of_Luckjmardonesk: Set() wants evaluated expression like X=Y
01:43.45jmardoneskbird_of_Luck, how can only call the query ODBC_XXXX?
01:44.18bird_of_Luckjmardonesk: save function as read, do 'module reload func_odbc.so', verify you get it right (by doing show function ...)
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01:44.53bird_of_Luckjmardonesk: change your set to SET(UNNEEDEDSTUFF=${ODBC_PREGUNTA1(0)}), save, do 'dialplan reload' and retry
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01:48.02Qwellumm, here's a stupid question
01:48.12Qwellif both read and write do the same query...  why bother with fixing it?
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01:55.42pugaanyone knows Russell Bryant?
01:56.19pugabah... have to go.. Oo'
01:57.56jmardoneskbird_of_Luck, muchas gracias dude ;-)
01:58.30jmardoneskbird_of_Luck, it works very fine, now show only a warning
01:59.01jmardoneskbird_of_Luck, but is now storing data in mysql
02:02.27bird_of_Luckjmardonesk: good. another way is to write write=.. with ${ARGX} and ${VALX} and call function like Set(FUNC_NAME(ARG1,ARG2,..)=VAL1\,VAL2\,..)
02:04.41bird_of_Luckjmardonesk: http://pastebin.com/m2464efb0 for example
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02:05.47pcranedoes anyone have experience with Linksys spa8000s and DTMF in asterisk?
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02:20.05jmardoneskhow can detect the answer in a Zap channel? i forgot what are the option in /etc/asterisk/zapata.conf
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03:06.57jameswf-homeMarketing at its best http://www.voipspeak.net/2008/fonality-provides-free-replacement-phone-systems-to-customers-devastated-by-hurricane-ike/
03:08.05eric2hmm, I restarted zaptel and it took the whole machine down
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03:08.17jameswf-homedont do that
03:08.29hardwirewhen did we start naming hurricanes after dead presidents?
03:09.05eric2what's the best way to restart zaptel and wanpipe?
03:09.11jameswf-homereboot
03:09.18eric2hmm... :(
03:09.27eric2is my machine toast?
03:09.39eric2messed up kernel ?
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03:10.20frogonwheelsI'm trying to work a way of having an incoming call with  Ringing() Dial(SIP/etc ,20,rd )  but have it Answer() after (say) 10 seconds
03:12.23glazwait() ?
03:13.02frogonwheelsok.. if I have  Ringing()  Dial(SIP/etc,5,rd) Answer() wait(15)    then it stops ringing as soon as the DIal() finishes
03:13.09drmessanoSo any big annoucements from Astricon today?
03:13.38frogonwheelsglaz: I'm either missing something obvious, or just trying to be too smart :|
03:14.09frogonwheelsglaz: was that what you meant by using wait?
03:15.49frogonwheels.. and iif I do  Ringing() Dial(SIP/etc,5) ANswer() Dial(SIP/etc,15,rd) then, at least for my sip client on my mobile), the second Dial doesn't connect
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03:19.15frogonwheelsThere's an old email that hints on forking a dialplan..  http://tinyurl.com/4snrz5   but I'm not sure I can then direct that incoming channel to Answer()
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03:21.09hardwirehi
03:21.16wscholarhey
03:21.28wscholar<PROTECTED>
03:21.48wscholar[Sep 23 23:09:27] WARNING[16692]: app_channelredirect.c:112 asyncgoto_exec: ChannelRedirect failed for SIP/provider-08237a80
03:22.53hardwirewscholar: read up on this
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03:22.55hardwirehttp://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Transfer
03:23.00hardwirekrokodilerian: fancy seeing you here
03:23.07Qwelldrmessano: it doesn't actually "start" until tomorrow
03:23.18Qwelltoday was developer day, and other random "pre-show" stuff
03:23.19drmessano:(
03:23.45mostyis it possible to get asterisk to refuse sip 302 redirects?
03:23.54drmessanoI just want to know if Skype is buying Paltalk.. then I will shut up...
03:24.11jayteehehe
03:24.29drmessanoTHIS IS IMPORTANT!!!!!ONES!!!!!111!!!!
03:24.42filewscholar: you are sending it to the transfer context, extension "Local/90005119077482879@transfer" with a priority of 1
03:25.10wscholaryes
03:25.14wscholartrying todo a blind transfer
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03:34.04jeevQwell
03:34.07jeevi cancelled my appearance
03:35.00jayteeI bet the fans were shattered.
03:35.02hardwirewscholar: figure it out?
03:35.10hardwirejeev: where are you atm?
03:35.24jeevi'm at the airport
03:35.26jeevi'm coming back home
03:35.30hardwireaww
03:35.31jeevthey only sent me 150k
03:35.35hardwirehah
03:35.36jeevwe aggreed on 250k
03:35.45jeevsomething about russellb complaining that he's not getting his 100k hooker
03:35.45hardwiretwofiddy?
03:36.00hardwirejeev: hah.. I'd hooker to him for that much.
03:36.14jeevlol
03:36.18jeevyou'd hooker to anything for nothing
03:36.18jayteewrites that down in his journal
03:36.31jeevjournal
03:36.33jeevrollerblading
03:36.33jeevwow..
03:36.35jeevyou sure you're not?
03:36.59drmessanojeev: I should apologize
03:37.03jayteeonly in your deepest fantasies
03:37.24drmessanojeev: I know I acted like you are a stark raving lunatic pathological liar
03:37.41drmessanojeev: Now I realize you're just ill and need help :(
03:39.06drmessanojeev: Now please, stop trying to stab yourself with that potato chip and get off the roof of the car.. mommy and daddy love you :`(
03:39.45jayteeI think he's just locked into the emotional state of a 14 year old. That's probably when the "trauma" occurred.
03:40.30drmessanojaytee: Do we need to break out the "Where did he touch you" bear?
03:41.08jeevwhat trauma
03:41.11jayteeI don't want him acting out his inner demons in here particularly
03:41.21drmessanojaytee: or is this more like "tell us where left Uncle Joe "sleeping", son"
03:41.29jeevbreak yoself
03:41.30wscholarwell if i pass in  EXEC "ChannelRedirect" "SIP/provider-081ca788|transfer|SIP/provider/90005119077482879|1"
03:41.33wscholari get failure
03:41.37wscholarif i pass in
03:41.37wscholar<PROTECTED>
03:41.39wscholarsrry
03:41.46wscholar<PROTECTED>
03:41.52wscholari get congestion / fast busy
03:42.59wscholar[transfer]
03:43.00wscholarexten => _X.,1,Noop(Enter transfer context, gr_callid: ${gr_callid})
03:43.00wscholarexten => _X.,n,Agi(agi://domU-12-31-39-00-50-32.compute-1.internal/transfer.agi?gr_callid=${gr_callid})
03:43.20wscholarif that helps
03:43.58jayteewould have been nicer if you'd just have wrapped up all that in a nice pretty package on a pastebin
03:44.02jaytee~pb
03:44.02jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
03:44.29hardwirewscholar: hmm..
03:45.59frogonwheelsok, is there a way I can cause an incoming channel to get 'answered' without stopping the Dial()  (i'll pb what I've tried)
03:46.04hardwireDial(SIP/provider/90005119077482879)
03:46.06hardwireyar
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03:48.31frogonwheelshttp://pastebin.com/d7bb8f70b
03:51.26[TK]D-Fenderfrogonwheels: That will grab it after 5s, hold for 20 and jsut flat out hangup.
03:51.49[TK]D-Fenderfrogonwheels: I see no point to this at all.  Just answer the call then dial like normal.
03:52.11frogonwheels[TK]D-Fender: yep, I know that is wrong 'cause I tried it, and It makes sense.
03:52.33[TK]D-Fenderfrogonwheels: Either way the call will get answered withing 5s regardless unless they hang up nearly instantly, which, really, why do youc are about that so much?
03:52.37frogonwheels[TK]D-Fender: Normally, I prefer not to Answer() straight away - so people on mobiles can hang up and not incur a charge.
03:52.46frogonwheels[TK]D-Fender: ok - so make it 20 seconds
03:53.47[TK]D-Fenderfrogonwheels: there should also be no need for "r" or "ringing".  Your first priority would be the Dial.
03:54.07[TK]D-Fenderfrogonwheels: Net your nested dial will not work to try to leave your original channel ringing.
03:54.17[TK]D-Fenderfrogonwheels: So no need for the split local channel.
03:54.36[TK]D-Fenderfrogonwheels: You'll have to do this as back-to-back dials with an Answer in the middle
03:54.38frogonwheels[TK]D-Fender: Ok - I know that. But you get what I'm trying to do, and I'm wondering if there is a away to do it.
03:54.43frogonwheelsok
03:54.55[TK]D-Fenderfrogonwheels: As I've just described.  That's all there is
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03:55.06frogonwheels.. and that doesn't appear to work with my sip client on the mobile phone.
03:55.25frogonwheels- the second dial doesn't work. (still engaged)
03:56.26frogonwheelsI thought there could be an outside chance, that [cont_Vista] context could send (something like) a ChannelRedirect to cause the connection to Answer
03:57.50[TK]D-Fenderfrogonwheels: All the redirecting in the world won't cause an answer.  An ANSWER will cause an answer.
03:59.23frogonwheels[TK]D-Fender: makes sense. I found this  http://tinyurl.com/4snrz5  - obviously somebody has thought of something similar before me :)
03:59.35frogonwheels[TK]D-Fender: thanks for the help though.
04:00.42krokodilerianhm
04:00.53krokodileriandoes some one feel like looking at some C code
04:01.14krokodilerianasterisk app, "cleaned up" for astricon
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04:02.16geoff2010i have a question on g.729 and asterisk... anyone willing to listen?
04:03.21krokodilerianis not that drunk yet
04:03.33phixkrokodilerian: fail
04:03.42phixyou should drink more
04:03.55krokodilerianshouldn't, i'll try for a dcap tomorrow
04:04.00krokodileriani need at least to get to the hall
04:04.05krokodilerianpreferrably standing
04:04.22mostygeoff2010, ask
04:05.01[TK]D-Fenderfrogonwheels: And that post states failure at every tern
04:05.26frogonwheels[TK]D-Fender: it was just a proof of concept.
04:05.35frogonwheels[TK]D-Fender: a debugging straw-man
04:05.50geoff2010i want my system to be able to handle 100 concurrent calls, but i only want to purchase 50 g729 licenses.  will asterisk automatically start negotiating g711 once all the licenses are used up?
04:06.06frogonwheels[TK]D-Fender: not much point in dotting i's and crossing t's if the concept is flawed
04:06.08[TK]D-Fenderfrogonwheels: If you answer in the multi-dial you cut on the one you ran in "parallel".  Complete waste of time.  You may as well have dialed for the same amount of time and gone on to the next priority.  There is no interrupting dial and changing the rules.
04:06.23[TK]D-Fenderfrogonwheels: Dial for X, Answer, Dial for Y.
04:06.36[TK]D-Fendergeoff2010: No.
04:06.39frogonwheels[TK]D-Fender: gotcha, really I understand.
04:06.46phixkrokodilerian: standing and bladder control is overrated
04:07.04geoff2010Fender: what will the behaivor be once i run out of licenses?
04:07.12[TK]D-Fenderfrogonwheels: On to more productive things then.
04:07.19frogonwheelsexactly
04:07.24[TK]D-Fendergeoff2010: Calls will be rejected
04:07.33geoff2010lame
04:07.37jameswf-homeobscure errors
04:08.04geoff2010so it's all or nothing with asterisk, can't mix/match codecs?
04:08.06krokodilerianwscholar , did you charge your phone? I might need some juice in a while :)
04:08.36[TK]D-Fendergeoff2010: It will match the same compatibility list every time.  Then upon attempt it will fail, and thats the end of the call.
04:09.02[TK]D-Fendergeoff2010: IMO * should "reserve" a license in its negotiation", but it does not.
04:09.18wscholari'm good
04:09.22wscholartake it
04:10.04geoff2010Fender: agreed.  sounds like a somewhat intentional limitation to force people to buy more g729 licenses... perhaps I am wrong.
04:11.23jameswf-homehmmm http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
04:14.23mostygeoff2010, will this box be primarily for incoming or outgoing calls?
04:17.25geoff2010mosty: i actually have 6 asterisk systems which work as one unified platform.  it's the telephony interface for a large virtual call center solution.
04:17.42geoff2010mosty: the servers do everything, inbound, outbound, IVR, etc
04:18.44mostyyou may as well just buy 100 g729 licences
04:19.04mostyinstead of messing around with a solution which might not work 100% of the time
04:19.51geoff2010yeah, 100 was a 'fake' number... we would need a lot more licenses if we switched to g.729.  problem is bandwidth is becomming very expensive.  not cheap to burst to 500Mb/sec
04:19.51oilinkihow much are the g729 licenses?
04:20.12hardwirehow many goats you got?
04:20.21jameswf-homeFree<
04:20.30hardwireoilinki: prices are listed on digiums site for their licenses
04:21.24mostygeoff2010, do your phones support g729?
04:21.33drmessanolol
04:21.45drmessanoyeah, thats legal....
04:22.13geoff2010we don't have any phones, it's all virtual call center stuff, so we are taking calls from SIP gateway and then calling agents out through the SIP gateway going back to the PSTN
04:22.26geoff2010asterisk stays in the middle of all the media
04:22.38geoff2010agents use regular home phones
04:28.57oilinkig729 seems to be usd10/channel
04:29.13hardwireoilinki: using g729 with a provider?
04:31.22oilinkihardwire: same situation as geoff2010 has. man in the middle
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04:36.59hardwireoilinki: mitm happens
04:38.12jameswf-homeis there a way to see the codec in use?
04:38.57krokodilerianjameswf-home , show channel xxxx
04:39.02krokodilerianjameswf-home , and then look at the format part
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04:39.53jameswf-homeno format field first thing i tried
04:40.01mostyjameswf-home, try "sip show channels"
04:40.04krokodilerianhmmm, should I be able to order room service with food and them carrying me from the code zone to the room
04:40.06mostyand "iax2 show channels"
04:40.09krokodilerianwell, not on a tray
04:40.16krokodilerianjameswf-home , what do you mean no format field?
04:40.31krokodileriangoes to look for an asterisk with channels
04:40.57jameswf-homehmm 0x0 (nothing)
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04:42.34krokodilerian<PROTECTED>
04:42.45krokodilerianjameswf-home , what's your channel driver
04:44.18jameswf-homesip/zap.   was checking out an unofficial g729 build
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04:49.08krokodilerianjameswf-home , can't really help you there
04:49.13krokodilerianseems like a bug
04:49.33krokodilerianthe one I tested on has g729
04:50.36krokodilerian<PROTECTED>
04:50.36krokodilerian<PROTECTED>
04:50.36krokodilerian<PROTECTED>
04:50.40krokodilerianthis is from a zap channel
04:52.39krokodilerianso, whatever, i'll sleep and try to dcap tomorrow :)
04:52.53jameswf-homefogot g723 hmm
04:52.57jameswf-homeworks
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05:27.59hardwirehi
05:30.14drumkillahi2u2
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06:23.26jameswf-homeheh http://www.russellbryant.net/blog/2008/09/12/the-voice-of-asterisk-does-monty-python/#more-136
06:24.00Strom_CNOBODY EXPECTS A MONTY PYTHON JOKE !!!
06:25.19jeevi dont even know what monty python is
06:25.49jblackthat's a joke, right?
06:26.23jeevnegative
06:26.38jblackYou should know. I suggest you google it up
06:26.46jeevi'm fuckin <50 years old
06:27.04jblackNo need to be obnoxious.
06:27.05jameswf-homeoy!
06:27.07jeevlol
06:27.12jeevi saw a wikipedia thingy
06:27.13jeev1969
06:27.15jeevjeebus h. christ
06:27.30jameswf-homeMonty Python is timeless
06:27.32Strom_Cmonty python was a group of british guys who wrote brilliant jokes which geeks have, through tireless repetition, made lame and unfunny.
06:27.47jameswf-homeit gave us python
06:27.52jameswf-homeokay i made that up
06:27.52jeevahh cool
06:28.01jeevjameswf-home, are you ever out of your home? :D
06:28.07jeevjameswf No such nick/channel
06:28.11jeevoh BY THE WAY
06:28.13jameswf-homeyes except when im not
06:28.14jeevi'm going to england next month
06:28.19jeevis it cool thereor what
06:28.31jblackYou're home except when you're not home?
06:28.41jameswf-homecorrect
06:29.05jeevhttp://www.africahit.com/news/index.php?mod=article&cat=othersenglish&article=4452
06:29.05jeev!
06:29.07jeevcatch me there
06:29.08jeevfarting
06:29.21jblacki suppose you're also paraplegic, except for when you're not.
06:29.34jblackjeev: Didn't you say you were gonna behave?
06:29.44jameswf-homeLIES!
06:30.06jblackOh.. africa hit. Not africa shit.
06:30.11jblackI'm sorry.
06:30.25jeevheh
06:30.30jeevi am behaving
06:31.01jblackI can apologize again, if you like
06:31.21jeevdood
06:31.23jeevlondon better be fun
06:31.24jeevi'll be pissed
06:31.40jblackPlenty to do in london, if you like getting drunk.
06:31.44jeevi dont drink
06:31.45jeevnever drank
06:31.46jeevnever smoekd
06:33.18jeevhahah, i told my girlfriend i give her too much freedom
06:33.19drmessanoI'm going to jupiter in August
06:33.20jeevshe's the cutest thing ever
06:33.23jameswf-homelondon smells bad
06:33.49drmessanojeev and I are going to go... as long as they don't make us take the orange pills again
06:33.53drmessano:(
06:33.56jblackI never noticed london smelling bad, but I smoke heavily.
06:34.05jblackwell, did at the times I was there.
06:34.24drmessanoWe're taking our Neptune Express Black Cards
06:34.36jameswf-homean odd mix of trash and urine
06:35.09jblack6-13 more days.
06:35.16drmessanoKinda like the smell of a man that just peed in his own vomit, set it on fire, then rolled around in it?
06:35.19drmessanoThats NYC
06:35.22jeevheh
06:51.30hardwireblah
06:51.46hardwirephoenix smells like sulfer to me
06:51.53creativxi thought nyc smelled like food
06:51.59creativxfood food and more food
06:52.02hardwireand even though it's humid.. it's making my nose dry
06:52.18creativxatleast you dont have rocket fuel in your drinking water ;)
06:52.29hardwirecreativx: put down the vodka.
06:54.23fiddurHi.  After a queue-call is finished, is there any way to identify what agent or interface that took the call, still in the dialplan?   I want to make a custom wrap-up-handling....
06:56.34hardwireyup
06:56.48hardwirecheck out the "asterisk variables" section in voip-info
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06:57.07hardwire-> sleep
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07:03.00fiddurI'm sorry, but I can only find QUEUESTATUS being set from application QUEUE, and none of the QUEUE_*-functions say anything about what agent took the call on the channel either... What am I missing?
07:08.31fiddurah, now I found some info about setinterfacevar...
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07:15.12yangIs there a magic key for deleting all messages from the voicemail ?
07:16.02Strom_Cthe closest thing to magic on my telephone is "transfer"
07:16.25Strom_Calthough "8" can be pretty fucking magical if you get it drunk first
07:17.19jeevlol
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09:11.19*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc6, 1.4.22-rc5 (2008/09/09), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12.1 (2008/09/09), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon
09:15.28mort_gibAgain, how to I change the ringing fro a caller to indicate that callee is in another call??
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09:17.44qpmorning all
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10:32.14badcfei have that when i reload
10:32.35badcfeand i have a file specified in the custom config
10:32.48badcfehowever, nothing goes into that file when calls end
10:32.51badcfe8 - (
10:33.04badcfeanyone comes up with a possible reason?
10:40.33badcfecdr-custom is *not* shown by "cdr status"
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10:46.48festr_hi
10:48.09festr_anyone know, why i'm hearing ringing tone immediatly after "SIP/SDP Status: 180 Ringing, with session description" although the device does not send ring for the first five seconds?
10:48.25festr_it is some GSM gateway which sends first RTP packet after five seconds
10:48.44festr_but in this first seconds asterisk assumes something and generates ring which is wrong
10:48.47festr_1.4
10:48.58festr_any idea what to change or if this is normal behaviour?
10:51.49badcfeyour phone will generate rinning on 180
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10:52.28badcfemore info about my problem descirbed above: custom cdr *was* enabled on make menuselect under Call Detail Recording
10:54.14festr_badcfe: dont understand, 180 generates ring although it is "with session description?"
10:54.55festr_aha
10:54.57festr_i got it
10:54.57festr_https://lists.cs.columbia.edu/pipermail/sip-implementors/2006-January/011850.html
10:55.03festr_If a 180 (Ringing) has been received but there are no incoming media packets, generate local ringing.
10:55.12festr_If a 180 (Ringing) has been received and there are incoming media packets, play them and do not generate local ringing.
10:56.16badcfeand sometimes the behaviour is configurable on the phone
10:56.16festr_unfortunately it is not. it seems that i need 183
10:58.26badcfefestr_: you have used custom-cdrs?  i have a little puzzle enabling it right now.
10:59.59badcfefestr_: on reload i see that /etc/asterisk/cdr_custom.conf are being parsed but without any effect ... you know why this could be so?
11:00.38festr_i'm using /etc/asterisk/cdr_mysql_custom.conf
11:01.15badcfehmmm.  when i issue "cdr status" the cdr-custom doesnt show even
11:01.31festr_ast-mezivoda*CLI> cdr status
11:01.31festr_CDR logging: enabled
11:01.31festr_CDR mode: simple
11:01.31festr_CDR output unanswered calls: no
11:01.31festr_CDR registered backend: cdr_manager
11:01.34festr_CDR registered backend: cdr-mysql-custom
11:02.08badcfei have nothing in /etc/asterisk/cdr.conf since its enabled by default, and my cdr_custom.conf now contains the example Master.csv
11:02.41badcfefestr_: i suppose if you uncomment the example in cdr_custom.conf, the file shows up under /var/log/asterisk/cdr-custom ?
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12:22.21yangI am wondering is there a device to integrate asterisk with hamradio ?
12:24.16boolean12This sounds fun.
12:24.23boolean12Well.
12:24.28boolean12Yeah. :-p
12:24.34boolean12You could use the console.
12:24.48yangI would probably need a device
12:24.49boolean12Take the sound output from the soundcard into your base.
12:25.00yangsort of a receiver to trigger it on some frequency
12:25.40boolean12If your base has a serial interface, you could just run a system command to key it, transmit then unkey.
12:26.45boolean12Hmm.
12:27.02yangboolean12: i mean triggering the numbers would be possible via DTMF tones
12:27.06boolean12I wonder if you could use talking detection to key it.
12:27.09boolean12Yeah, it would be.
12:27.32boolean12Hm.
12:27.38boolean12Sounds fun ^^
12:31.18tzafrir_laptopboolean12, pciradio, baiscally
12:31.34tzafrir_laptoperr, the above was for yang
12:31.37boolean12Heh ^^
12:31.46tzafrir_laptopask SteveTotaro
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12:45.44oktayhello.. what does 5 very rapid tones when I take the line off-hook mean?
12:45.58oktayor 8
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12:52.05srthi, i am looking for dids in portugal. can anybody give me a hint?
12:52.46eric2didww
12:53.53srtunfortunately they don't have anything in portugal, same for didx
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13:03.10oktayOutbound Routes/Dial Patterns -> 88
13:03.19oktaygives me a PSTN dial tone when I dial 88
13:03.21oktaywhy is that?
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13:05.47oktayerm. sorry. that was a freepbx question.
13:06.26[TK]D-Fenderoktay: For which they have their own channel ; #freepbx . Go give them a visit I'm sure they can tell you what you need to know
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13:07.38oktayyes. i typed in the wrong window. that's why i said sorry.
13:07.44oktaythanks
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13:12.09oktaybye
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13:15.38traxxoshello
13:17.44Kattywobbles
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13:17.52Kattywho stole my caffeine iv
13:18.08blaylockanyone have 1.6 working with unified messaging?
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13:18.46seanbrightanyone know how to set callerid name on an outbound analog line?
13:18.49seanbrightteehee
13:18.54seanbright<-- s.o.b.
13:19.00Kattybonks seanbright
13:19.22KattySean NotSo Bright
13:19.28seanbrightclever!
13:19.44seanbrightmy middle initial is R
13:19.49hi365is there a limit to the ammont of includes voicemail.conf can have?
13:19.53Kattythere's two new litters of GSDs in the newspaper :>
13:19.54seanbrightso when i say "Sean R. Bright"
13:19.57coppiceseanbright: go to Japan :-)
13:20.18seanbrightjackasses says "shouldn't that be Sean *is* Bright??"
13:20.21Kattyplots phone calls around 10
13:20.43seanbrightcoppice: what is in japan besides really frightening porn?
13:20.52Kattylolita goths.
13:20.57Kattyipod knockoffs
13:21.05coppiceyou actually can set caller ID on some analogue lines there
13:21.36Kattyreally bad food.
13:21.36seanbrightohhh
13:21.36Kattytheir mcdonalds carry soy burgers.
13:21.55Kattywalmart labor camps.
13:21.55seanbrightcoppice: i was just asking that question as a friendly ribbing towards Katty
13:21.55Kattycoppice: yes. he was giving me hell.
13:21.56Kattycoppice: in a nice, sort of way.
13:22.01gr0mitmmmmh soy-burgers
13:22.08Kattyi love you too sean.
13:22.16seanbright<3
13:22.23Kattyboca burgers aren't too bad.
13:22.28Kattyyou can even get those at Denny's!
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13:23.03coppicesome markets kinda make veggie burgers essential for MacD. A big mac would have a limited market in India
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13:23.11coppiceI think they only serve rice in .ph, though
13:23.26Kattysacred-god-burgers
13:23.40Kattyanother example of religion holdin the man down
13:24.02tzafrir_laptopham-burgers?
13:24.10coppicehow exactly does not eating cows keep someone down?
13:24.12Kattycow-burgers
13:24.23Kattycoppice: if they ate the cows, they'd have less famine
13:24.34gr0mitmunches a ratburgeer
13:24.40coppicehuh? it works the other way around
13:24.53Kattycoppice: perhaps we're not on the same page.
13:25.00Kattycoppice: it's still really early in the Angeliverse.
13:25.38coppiceit takes 10kg of veggie to make 1kg of beef. eating animals causes starvation
13:25.52gr0mitnot for me though ;-)
13:26.00coppicedo you eat horse?
13:26.07gr0mitnope.
13:26.12Kattyisn't that illegal?
13:26.18Kattyi know it's illegal to shoot a horse now.
13:26.20gr0mitthe English do not eat horse
13:26.20srtno horse is great
13:26.29Kattyi've heard horse is really good.
13:26.30gr0mitat least not knowingly
13:26.50coppiceno. most western people just don't do it. similarly much of asia treats its traditional farm working animals with the same respect
13:26.51Kattyi also hear cat is really good.
13:27.01viraptoris there any official / proven way to add join/leave sounds to app_conference?
13:27.06coppicedonkey is better than horse
13:27.14Kattyviraptor: show application meetme
13:27.16gr0mitlikes ostrich
13:27.21Kattyviraptor: also perhaps, core show applicaiton meetme
13:27.33Kattyviraptor: show core application meetme?
13:27.36Kattyviraptor: something like that.
13:27.38coppicekiwi is better than oistrich
13:27.39viraptorKatty: I don't want meetme -> is there any way to add it to app_conference?
13:27.54Kattyviraptor: i haven't the froggiest.
13:28.01Kattyviraptor: all the cool kids use meetme.
13:28.25Kattycoppice: you mean... the kiwi fruit?
13:28.38Kattycoppice: i'm having horror flashbacks of kiwi, from youtube.
13:28.48coppiceno. the bird. similar meat to ostrich, but more flavourful and more tender
13:28.52viraptorKatty: doesn't work good enough here... app_conf can handle 2 times more users unfortunately
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13:29.24jameswf-homenever call an aussie a kiwi or a kiwi an aussie
13:29.48gr0mitor a Brit an ozzie
13:29.49Kattyviraptor: best of luck in your search.
13:29.54coppicei think most of the kiwis eaten in asia actually come from .au
13:30.11Kattyi must google this kiwi bird.
13:30.24jameswf-homemost ausies were brits granted long ago
13:30.38Kattyit is the youtube kiwi :<
13:30.45Kattyunacceptable!
13:31.10Kattyjbot: kiwi?
13:31.11jbotkiwi is, like, A framework for Python applications with graphical user interfaces.   URL: http://www.async.com.br/projects/kiwi/
13:31.28Kattyjbot: epic fail.
13:31.44jameswf-homejbot fist jbot
13:31.44jbotACTION uses jbot as a handpuppet
13:32.11blaylockunified messaging and asterisk 1.6 anybody?
13:32.38*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
13:32.44Kattylanman
13:32.59[intra]lanmanKatty
13:33.03Kattyhow's the wifey?
13:33.11[intra]lanmanhugs Katty
13:33.13Kattyand all your critters
13:33.13[intra]lanmanshe's good
13:33.15Kattyhugs intralanman
13:33.26[intra]lanmanooohhhh, the critters aren't doing so good
13:33.30Kattyoh? :<
13:33.34[intra]lanmanwell, at least the chickens
13:33.42Kattywhat happened to the chickens? )_=
13:33.44[intra]lanmanall the furries are still cool
13:33.55[intra]lanmanwell, in short, the dogs happened to the chickens
13:34.05Kattyoh dear :<
13:34.10[intra]lanmanuh huh
13:34.18Kattyi hope it was quick.
13:34.53[intra]lanmanwell, i've seen the dogs play with toys and rip them to shreds in seconds... .so i'm sure it was
13:34.59Kattynods
13:35.05Kattyi'm getting a GSD soon.
13:35.09coppicedon't eat dog with beans
13:35.10Kattyinside pup, tho
13:35.21*** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk)
13:35.30[intra]lanmancoppice: why? give you gas?
13:36.01coppiceyeah. it messes up your stomache really really badly
13:36.07Kattyalso, randomly, (emily elbert)++
13:36.20gr0mitbegins to get very worried at the direction this conversation is heading
13:36.23[intra]lanmancoppice: thanks, i'll keep that in mind
13:36.37Kattygr0mit: it's too early for geekery.
13:36.43[intra]lanmanindeed
13:36.55Kattyi still think seanbright stole my caffeine iv.
13:37.03gr0mitreturns to Electronics for Dogs
13:37.07[intra]lanmanthis might not be the best place to say this... but i finally moved my voicemail server away from asterisk ;-)
13:37.08Kattygr0mit: oh!
13:37.10coppiceit would be hard to outdo someone living in east asia for eating strange things :-)
13:37.18Kattygr0mit: target has a robotic dino
13:37.26Kattygr0mit: $135ish, it'd be a super dog toy
13:37.43Kattygr0mit: it wasn't the common white one from best buy and such, either. it was BIG with lights, and fun noises.
13:37.54adr3nalin3How does one disable moh?
13:38.08Kattymusiconhold.conf?
13:38.40adr3nalin3Katty: thanks for some reason I was looking for moh.conf
13:38.51*** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com)
13:39.00Kattyearly morning score for katty!
13:39.11Katty...oh. that sounded bad.
13:39.15[intra]lanmanyup
13:39.23Kattyseanbright: iv :<
13:40.10seanbrighti admit nothing
13:41.40gr0mitponders an after lunch nap
13:42.02Kattyvolunteers seanbright for after lunch nappery.
13:43.09seanbrighti'm down
13:45.24Kattypoodles are so incredibly ugly.
13:45.38*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
13:46.49*** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr)
13:46.52sehhhey people
13:47.04Kattyhi
13:47.48gr0mithello
13:49.17sehhq: i've got an asterisk server (v1.4.21) which uses a Beronet ISDN card to connect to the ISDN line. It uses mISDN to talk to asterisk. My problem is that some incoming callers can't use the IVR because the DTMF tones aren't recognized.
13:49.35sehhi've enabled debug,verbose,dtmf mode in logger.conf
13:50.10sehhand i've successfuly tested the system by calling from a landline and a mobile phone (dtmf tones are recognized and also logged properly)
13:50.40*** join/#asterisk mog (n=mog@nat/digium/x-2924d08d04cd8846)
13:50.40*** mode/#asterisk [+o mog] by ChanServ
13:51.10sehhunfortunately, the users who complain that the IVR doesn't work are calling from their own ISDN line and when they hit dial keys, my asterisk doesn't even log anything! its as if asterisk can't hear the tones
13:51.53*** join/#asterisk relas (n=hemanuth@port-92-203-120-191.dynamic.qsc.de)
13:52.37gr0mitsehh
13:52.50gr0mitare you sure you are receiving DTMF?
13:53.08gr0mitif you listen via chanspy do you hear the incoming dtmf?
13:53.09relashello! I've got two hfc-s isdn-cards with Asterisk 1.4.21.1-BRIstuffed-0.4.0-RC3b. I'm getting this warning: WARNING[2172]: chan_zap.c:2510 pri_find_dchan: No D-channels available!  Using Primary channel 3 as D-channel anyway!
13:53.36relasWhy do I get this warning?
13:53.57gr0mitone of your isdn2 lines is not working?
13:54.01Katty[TK]D-Fender: where are you this morning?
13:54.22gr0mitRelas, paste your zaptel.conf and zapata.conf pls
13:54.31relasgr0mit: mom
13:54.35[TK]D-FenderKatty: Work, like usual... same sh!t, different day...
13:54.42Katty[TK]D-Fender: ah right.
13:54.50Katty[TK]D-Fender: well i hope your wounded thumb is a bit better.
13:55.12sehhgr0mit, i haven't tried to use chanspy (never used it before) but i'm sure the caller produces tones because i've made a call to my mobile phone for testing and i can hear them fine
13:55.43*** join/#asterisk Specialist1 (n=me@119.160.105.172)
13:55.43Kattydisappears for awhile
13:55.45Specialist1hi all
13:56.03relashttp://rafb.net/p/52LpFT55.html
13:56.15sehhgr0mit, i've also asked the caller to call some local company that uses an IVR on a panasonic PBX and he could navigate their IVR without a problem
13:57.00[TK]D-FenderKatty: Yes, its vissibly better daily.  Also funky when you can FEEL your body repairing itself...
13:57.43gr0mitRelas, looks ok
13:57.47*** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com)
13:58.48sehhgr0mit, interestingly when the tones ARE recognized, i see this in the log (when pressing an 8): DTMF[7993] channel.c: DTMF end accepted without begin '8' on mISDN/1-u1389
13:58.59gr0mithmmm, relas, let me pastebin you my config.
13:59.41gr0mithttp://pastebin.com/d77202254
14:00.21gr0mitspans 1 and 2 are NT mode, span 3 goes to my ISDN line
14:00.36gr0mitwonder if it relates to where you are taking your clock from
14:00.55gr0mityou must take your clock from the public networl
14:01.12hi365is there a limit to the ammont of includes voicemail.conf can have?
14:01.24gr0mitnetwork, but i cant recall what the numbers in span= line do without looking it up
14:01.39*** join/#asterisk _gm (n=gmustafa@202.133.78.60)
14:02.14gr0mitand i am running the same version of bristuff as you, relas
14:02.50gr0mitsehh, never used misdn, i use bristuff
14:03.27*** join/#asterisk zydoon (n=zydoon@41.225.153.114)
14:03.40sehhgr0mit, ok no problem
14:04.15tengulreanybody here make CallCenter with asterisk ?
14:04.18*** part/#asterisk zydoon (n=zydoon@41.225.153.114)
14:05.23*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-9728310ef3a85813)
14:05.23*** mode/#asterisk [+o putnopvut] by ChanServ
14:06.06*** join/#asterisk gambolputty (n=BC43599@cpe-24-175-90-49.tx.res.rr.com)
14:06.13*** part/#asterisk gambolputty (n=BC43599@cpe-24-175-90-49.tx.res.rr.com)
14:07.22*** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal)
14:07.29[TK]D-Fendertengulre: Plenty of us.
14:08.26*** join/#asterisk daveburr (n=dave@208.53.57.91)
14:11.48*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
14:13.46geoff2010g729 licensing question: i am making outbound calls which do AMD, which will require a license reservation to perform the audio analysis, once i get an answer i am going to be doing straight g729 pass-thru to another g729 UA, will asterisk give back the license once it is done decoding for AMD?
14:14.11geoff2010or will that channel hold onto the license for the duration of the call?
14:14.14[TK]D-Fendergeoff2010: It should
14:14.28[TK]D-Fendergeoff2010: Only when actual decode/encode is required
14:21.26*** join/#asterisk BobPierce (n=WESTMAN\@216.36.132.162)
14:21.39geoff2010Fender: Thanks for the help (last night and this AM)
14:23.38gr0mitany luck, relas?
14:24.49*** join/#asterisk buzzyd (n=psp-man@86.54.239.113)
14:24.59buzzydHi all,
14:25.00*** join/#asterisk ToTo (n=ToTo@207.176.6.103)
14:25.17*** join/#asterisk tftech (n=mjolnir1@63.214.236.169)
14:26.00buzzydI'm using the followme app in asterisk 1.4 and it asks push 1 to accept call 2 to reject however I can't do this on my bike headset so anyway to stop it asking that and just connect on answer?
14:28.39tftechHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this?
14:29.53*** join/#asterisk mv2 (n=mv2@83.240.229.38)
14:31.08*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
14:31.30mv2why asterisk opens two udp ports beside 5060 ?
14:31.44km-backdoor.
14:31.52km-it's so digium can hax your gibson.
14:31.53*** join/#asterisk ph0enix (i=disalvo@foster.stonedcoder.org)
14:32.28mv2really ?
14:32.39km-haha, no.  What ports are open?
14:32.58mv24569
14:33.01mv22727
14:33.11km-4569 is IAX2, another voip protocol that asterisk supports
14:33.20*** join/#asterisk jpastore (n=jpastore@adsl-69-106-247-50.dsl.pltn13.pacbell.net)
14:33.29mv2ok
14:33.32tftechHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this?
14:33.43mv2and 2727 manager ?
14:34.02km-nope, 5038 is manager I think
14:34.05BobPierceI'm just in the process of rebuilding our Asterisk server here at the office. We're currently running 1.4.21.2, but I'm tempted to move to 1.6 for some of the new features. Anyone have any opinions or info on how stable it might be for production use?
14:34.08km-not sure what 2727 is.  H323 maybe?
14:34.22km-ah, no, MGCP
14:34.32*** join/#asterisk jpastore (n=jpastore@adsl-69-106-247-50.dsl.pltn13.pacbell.net)
14:34.32mv2km: ok thanks
14:34.34km-MGCP is another voip protocol
14:34.38km-np
14:36.03*** join/#asterisk jpastore (n=jpastore@adsl-69-106-247-50.dsl.pltn13.pacbell.net)
14:36.13JerJer4569 is the segfault port
14:36.56km-guaranteed to segfault your crap when you connect to it?
14:37.32JerJerwell not necessarily a guarantee but pretty damn close
14:37.41km-hahaha
14:38.25JerJerhttp://www.securityscraper.com/pingpoke/iaxControlNew.txt
14:39.57km-interesting
14:41.17tftechHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info
14:41.34JerJerkm-:   try it  :)
14:41.36*** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-570d86b7c4963b77)
14:41.36*** mode/#asterisk [+o Deeewayne] by ChanServ
14:41.49JerJerpick any version of asterisk - even svn head
14:42.46mv2JerJer : not working with me
14:44.23km-hmm
14:44.35km-its spamming iax packets to my server but no crash yet.
14:46.10km-it's up to 40,000 packets
14:46.19km-is there a certain point where it'll fall over?
14:49.03km-hehe, there's definitely a memory leak
14:51.06tftechHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info
14:52.40adr3nalin3Anybody have any tips on tweaking zaptel settings?  I am having some problems with dialing out.  I have the patterns correct but the phone co doesn't seem to be registering all the numbers dialed when they are sent out my tdm400p card.
14:54.00adr3nalin3about 50% a call will go through the rest of the time I will get a recording from the phone co.  That says cannot complete the call as dialed.
14:54.11*** join/#asterisk mahlon (i=mahlon@martini.nu)
14:54.14*** join/#asterisk wampie (n=nancyvb@vanbaak.xs4all.nl)
14:54.37*** join/#asterisk propellerhead (n=yogurt2u@190.210.28.193)
14:54.44wampiemvanbaak, awake yet?
14:54.58glazwhich file is use for echo tuning?
14:55.04tftechadr3nalin3 - What verison are you using?
14:55.26*** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net)
14:55.37wampieHi there btw! Anyone there who is actually on the Astricon?
14:56.13adr3nalin3tftech: current stable
14:56.22seanbrightwampie: bunch of people floating around in #astricon
14:56.36adr3nalin3tftech: Zaptel 1.4.12.1
14:56.38wampieah thanks!
14:56.47tftechadr3nalin3 : I had an issue where I had to update the Zaptel drivers to the very latest. I built a box that worked at one place and would not work at another
14:56.50wampiehadn't checked there yet ;)
14:57.32adr3nalin3tftech: were they using "digital" phones from a cable co?
14:57.53tftechadr : so they are not striaght POTS lines?
14:58.23adr3nalin3tftech: they are terminated as such.
14:58.47adr3nalin3I was using them without issue with a 3COM analog line card adaptor
14:58.55tftechadr : Can you call out on a regular analog phone?
14:58.58adr3nalin3yes
14:59.35adr3nalin3tftech: I will give updating zaptel a shot.
14:59.42relasgr0mit: I changed it to the same in your config. Seems to work
14:59.42tftechI know more about telephony than Asterisk. Sorry. I just know I had the issue and had to update the zaptel drivers
15:00.01adr3nalin3tftech: thanks for your help
15:00.13tftechwish I could help more
15:03.31gr0mitrelas, cool!
15:03.51*** join/#asterisk brodiem (n=brodiem@67.18.114.226)
15:06.05*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
15:06.18*** part/#asterisk BobPierce (n=WESTMAN\@216.36.132.162)
15:06.35relasgr0mit: I've antoher problem. DTFM don't work.
15:07.28*** join/#asterisk hfb (n=hfb@pool-96-247-117-68.lsanca.dsl-w.verizon.net)
15:10.17gr0mitrelas - aaahh not you as well!
15:10.23gr0mitdtmf in or out?!
15:11.29relasin
15:11.33relasout works fine
15:11.46gr0mitok pls paste your zapata.conf file
15:12.08gr0mit1 sec let me look back
15:12.09relashttp://rafb.net/p/52LpFT55.html
15:12.52relastoneduaration=300 did not take any effect
15:13.33gr0mitlet me check my config
15:14.47gr0mitdoes it not work for your isdn line or your phone or both?
15:15.14relasfor isdn-line
15:15.27gr0mitand for your isdn phone?
15:15.37relasit works
15:15.46relasfor my isdn-phone
15:16.02*** join/#asterisk thor (n=mjolnir1@63.214.236.169)
15:16.38thorHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info
15:17.53gr0mitso if someone calls your isdn line, it does not get detected under all conditions, or just some?
15:18.26gr0mitlet me paste my config
15:18.36*** join/#asterisk Dovid (n=Dovid@bzq-79-181-15-215.red.bezeqint.net)
15:19.21*** join/#asterisk ManxPower (n=manxpowe@5.sub-70-222-245.myvzw.com)
15:19.29gr0mithttp://pastebin.com/d180f80c1  <-- Relas, here is my config for a UK ISDN2 line
15:22.00relasgr0mit: I'll try your config later. I have to go right now. I will report you on success.
15:22.10thorHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info
15:22.12gr0mitok. tot ziens
15:22.34gr0mitor tchus
15:22.39gr0mitor wotever it is over there!
15:22.43[TK]D-Fenderadr3nalin3: Most common issue is that * dials the 1st digit too fast and it gets dropped.  When dialing out add "ww" before your actual number to delay it 1s
15:24.14adr3nalin3[TK]D-Fender: Thanks, I called digium and they told me the same and was fixed immediately!
15:26.51thorHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info
15:27.45blaylockis anyone using asterisk 1.6 to connect to unified messaging server without sipx?
15:29.43[TK]D-Fenderadr3nalin3: Excellent
15:32.21tftechHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info
15:33.00*** join/#asterisk oilinki (n=oil@ppp-124-120-17-190.revip2.asianet.co.th)
15:33.05[TK]D-Fendertftech: You should go ahead and pastebin that now.
15:33.07[TK]D-Fender~pb
15:33.08jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
15:33.57tzafrir_laptopjbot's back!
15:34.06creativxwb jbot
15:34.50iCEBrkrumm
15:35.22iCEBrkrhas Cordy's func_mysql stuff been replaced with the app_mysql addon??
15:35.33tftechIt is not really over 3 lines
15:35.37tftechjust will explain
15:36.29tftechgetting this message: ERROR: Could not open H.323 listener port on 1720
15:36.29tftech[Sep 24 04:22:14] ERROR[13127]: chan_h323.c:3170 load_module: Unable to create H323 listener
15:36.48tftechhave installed via SVN today
15:37.00tftechrecompiled 3 times with no installation errors
15:40.15*** join/#asterisk a-s (n=user@89.38.174.194)
15:40.28a-show can I install ilbc in asterisk?
15:40.41a-sI cannot find it in asterisk's sources
15:41.09iCEBrkrAhh I found it.. func_odbc
15:41.58KattyYES!
15:42.04Kattygsd pups have been CONFIRMED!
15:42.09Katty[TK]D-Fender: PUPPY!!!!
15:42.11Katty[TK]D-Fender: THIS WEEKEND!
15:42.17Katty[TK]D-Fender: I"M GOING TO BE A MOM!
15:42.38Katty[TK]D-Fender: i hope. they're sending me pictures :>
15:42.50Kattyboingboingboing
15:43.18*** join/#asterisk riksta (n=rick@92.63.131.41)
15:43.20Katty..i swear i'm not insane.
15:43.26Kattylooks around
15:43.48creativxcraaaaahaazy
15:44.03rikstacan someone tell me how i can set up an extension in the dialplan which when i call into it, it runs a stop now (or shuts down asterisk) ?
15:45.24*** join/#asterisk aaroninnes (n=aaroninn@im.jobdig.com)
15:45.37*** join/#asterisk bijit (n=benji@200.122.158.243)
15:46.03bijithow do i get the colors for the cli?
15:47.47aaroninnesHave you checked google yet? If not I just found this: http://osdir.com/ml/debian.packages.voip.devel/2005-04/msg00151.html
15:48.16bijityeah i checked
15:48.49bijitacually i tried that and it did not wok
15:48.56*** join/#asterisk af_ (n=getsmart@88-149-241-240.dynamic.ngi.it)
15:48.56aaroninnesNice...
15:49.27aaroninnesCan someone tell me if it is normal to have errors on cat/proc/interrupts on an asterisk server?
15:49.45a-snobody can help me about how to install ilbc please?
15:50.38Specialist1any folks have any idea about a good reseller control panel for asterisk
15:51.03a-sI cannot find its sources here... http://ilbcfreeware.org/
15:53.39geoff2010anyone know anything about "frame type 64"?  i am getting the following error after installing some g729 licenses on my server:
15:53.41geoff2010chan_sip.c:3707 sip_write: Asked to transmit frame type 64, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256)
15:54.38*** join/#asterisk ana_micho (n=sniper_v@87.236.144.38)
15:54.45*** join/#asterisk c4t3l (n=root@74.95.210.124)
15:54.51c4t3lhello all
15:55.23*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
15:56.02ana_michoHi all, I need that someone take a look to http://pastebin.com/m285374ec.....When making a call to a DID number the asterisk server in not entering in DID mode...Any comments?
15:56.37iCEBrkrKatty: This is like the 100th time you claim to not be insane..  who you trying to fool?
15:56.41iCEBrkr:D
15:56.51*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
16:00.48MikeJKatty isn't insane..
16:00.55MikeJreally.. i swear
16:02.00*** join/#asterisk grEvenX (n=even@pc107-130.ktv.no)
16:03.36ManxPowerana_micho: You should contact a2billing support
16:03.47*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
16:04.02ManxPowergeoff2010: "show codecs"
16:04.07ManxPoweror core show codecs
16:04.24ManxPoweraaroninnes: what kind or errors?
16:05.01ManxPowerriksta: exten => 666,1,System(/usr/sbin/asterisk -rx "stop now")
16:05.06*** join/#asterisk ant (n=ant@209.11.127.13)
16:06.10*** join/#asterisk CunningPike (n=arodgers@64.251.77.9)
16:06.26*** join/#asterisk ManxPower (n=manxpowe@5.sub-70-222-245.myvzw.com)
16:06.48ManxPoweroops 8-)
16:07.01aaroninnesManxPower: I am getting the info right now, thank you.
16:07.16*** join/#asterisk spokra (n=spokra@gumby.sea0.speakeasy.net)
16:07.16*** join/#asterisk Tomo1657 (n=Tomo1657@63-255-103-7.ip.mcleodusa.net)
16:07.21rikstaManxPower: thanks, ididnt know you could call System()
16:07.45ManxPowerriksta: "core show applications" tells you all the applications available in YOUR Asterisk install
16:07.56ManxPoweraaroninnes: copy the errors to pastebin.ca NOT TO THIS CHANNEL
16:08.10aaroninnesOk
16:08.59*** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org)
16:13.21aaroninnesManxPower: here is the pastebin link: http://pastebin.com/m3eafe8f3
16:13.56ManxPoweraaroninnes: You are either having problems with the PRI line or you are having problems with locked interrupts.  What version of Zaptel are you using?
16:14.38ManxPowerBTW, I believe this problem is something that would be included with the free support for your card.
16:15.54bijitanyone know how I can get the cli colors working?
16:15.56ManxPoweraaroninnes: a Red Alarm almost always means line problems on the telco side.  Sometimes it can be a marginal cable.
16:16.12ManxPowerbijit: not me.  Seems like something stupid to spend lots of time trying to fix
16:16.48*** join/#asterisk lgj (n=leif@76.7.56.12)
16:16.53*** join/#asterisk pyite (n=pyite@63-255-103-7.ip.mcleodusa.net)
16:16.54aaroninnesOk, thank you very much. We believe the problem was with locked interrupts. I'm looking for the zaptel info now.
16:17.15ManxPoweraaroninnes: I have never actually seen locked interrupts causing a red alarm.
16:18.20ManxPowerALL Red alarms that I have had to diagnose and fix were caused by the telco side.
16:19.53aaroninnesManxPower: zaptel version 1.4.10.1 asterisk version 1.4.20.1 and libpri 1.4.4
16:19.56bijitManxPower: thought it would be fun.. anyhow. What I really want to spend time is in blf. I changed call-limit limitonpeers. But its kind of delayed. Dont know if its the phone firmware.
16:21.22*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
16:23.10a-swhy ilbc is not included in tha latest version of * any more?
16:23.24[TK]D-Fendera-s: Licensing
16:24.26coppiceilbc is very messily licenced. its hard to know what you can and can't do with it. this is probably intentional
16:24.27lgjhello channel, have a sip auth/peer question if anyone has time?
16:24.51ManxPower~ask
16:24.52jbotask is probably Questions 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.
16:25.28a-s[TK]D-Fender: waw! however if I want to compile it , can I?
16:25.50[TK]D-Fendera-s: Yes, and there is plenty of documentations telling you how
16:26.01ManxPowera-s: dude, this should be all covered in the Asterisk change log or release notes or upgrade.txt that is included with the Asterisk source code.  If it is not then you should file a bug.
16:26.05lgjanyone know if inbound sip calls which have a peer/user entry matching host and insecure=port, don't match all the time and thus send call to default context even when allow guest calls is no?
16:26.30lgjseems like it only gets the context when I use the domain=<domain>,<context> setting in global
16:27.28*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
16:27.53aaroninnesManxPower: Thanks for the info, I'm a little confused though. I don't see how the telco can cause na IRQ error, plus I feel it is very unlikely that they are having issues with 2 T1 lines. Is it more possible its a PRI issue?
16:28.07lgjwhich seems not the right default function of chan_sip
16:28.26ManxPoweraaroninnes: maybe you are having two UNRELATED problems?
16:28.41ManxPowerlgj: Not all that many people use insecure=
16:29.20aaroninnesManxPower: That is possible...thank you.
16:29.22lgjwhat then is the best way to talk to opensips/openser currently the test is on the same box thus the insecure=port
16:29.48ManxPowerlgj: no idea.
16:29.53lgjnods
16:30.30lgjinsecure is the issue then i would assume?
16:31.06ManxPowerlgj: best of luck.
16:31.20c4t3llgl:  have you been over to the openser channel
16:31.24*** join/#asterisk southtel (n=southtel@68-114-19-101.dhcp.gwnt.ga.charter.com)
16:31.59a-sI understand that ilbc must be compiled as a system library, and asterisk will link something against it... in the past its sources were included in *. is it true?
16:32.06lgjyeah I got the openser side working perfectly. now down to why asterisk only sends the call to the correct context when I use the global value for domain=<domain>,<context>
16:32.27lgjseems insecure to do the context via URI domain instead of a defined peer?
16:32.45c4t3lagreed
16:33.00c4t3lthat's a wierd one
16:33.10*** part/#asterisk southtel (n=southtel@68-114-19-101.dhcp.gwnt.ga.charter.com)
16:33.30lgjthe reason I am asking here is I don't understand why the peer wouldn't match all the time, it matches when an outside call is forwared to asterisk e.g. inbound mainmenu
16:34.02lgjbut when the phone registered with opensips calls the asterisk box it sends it to the default context even when there is a peer with host match ?
16:34.31lgji am thinking I overlooked somthing but i wanted to know if anyone had wrestled with this before
16:34.39c4t3lsounds like some sipheader foo goin on there...
16:35.04lgjprob so, but it was very late last night and I was tired so I figured I would ask fresh here today
16:35.14c4t3li last workd with openser/asterisk over 3 years ago.  Do you hang out on this channel often?
16:35.31c4t3li know a guy who does this sort of thing exclusively
16:35.51c4t3lhe runs a business on it actually
16:35.53lgjnope, unfortunatly don't have time to chat usualy, only try to ask informed questions when I cannot get it myself
16:35.56a-s[TK]D-Fender: can you give me a concretely page where the installation of ilbc is described?
16:36.03a-sI cannot find its sources
16:36.51lgjc4t3l, yeah and they changed a LOT from openser to opensips so almost all the docs out there are slightly wrong
16:37.06c4t3lcrap
16:37.49lgjbut I got all my regex and dialplan code working correctly just this wired bug/feature of how the calls are matched to the peer host records
16:38.06lgjthus it forcing it into default context even when the global context is not default
16:38.12lgjthat is why I came here to chat about it
16:38.26lgjseems like there may be a wired condition within the code
16:38.51ManxPowerBut code is not really discussed on this channel.
16:38.59ManxPowerThat's why they created #asterisk-dev
16:39.17lgjyeah on there but none of the available people want to chat about my issue right now :)
16:39.22c4t3llgl:  sorry man.  I cant raise my buddy on the phone. the only other option i can think of is maybe running in debug logging mode (if not done so already) or perhaps gdb...
16:39.56ManxPowera-s: please report the lack of docs in the code about ilibc on bugs.digium.com
16:39.58*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
16:40.18lgjc4t3l, no need to bother anyone on the phone, yeah I will dig deeper into it if I have to. the work around for now with the gobal domain= settings works just don't feel to comfortable about it
16:40.24lgjer global
16:40.38lgjthanks for the help people! keep up the good work with asterisk!
16:40.52lgjl8r
16:41.08qp[TK]D-Fender: re my connection issue from a couple of days back, we did end up using conference rooms for our inhouse sip softphone, quite handy too as others can "hop" into calls
16:41.23ManxPowera-s: of course if you had read the CHANGES and UPGRADE.txt you would know everything you need to know about why and how to fix your problem.  Shall I also give you line numbers of the information>?
16:41.23qp(managed transfers)
16:41.35*** join/#asterisk Segnale007 (n=Segnale0@host112-254-dynamic.32-79-r.retail.telecomitalia.it)
16:46.23*** join/#asterisk voxter (n=voxter@63-255-103-7.ip.mcleodusa.net)
16:49.56*** join/#asterisk kbdman (n=carlos@unaffiliated/kbdman)
16:51.34*** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130)
16:53.01*** join/#asterisk wscholar (n=wayne@143.sub-75-209-212.myvzw.com)
16:53.08*** join/#asterisk gr0mit (n=tim@81.187.32.146)
16:54.07StephenFHow do you guys handle teleworkers? hardware VPN tunnel back to *?
16:54.16mort_gibHi, is there any way to alert a caller if callee is already in a call. I KNOW i can see if the Channel is in use, but playing a sound file is not really what I want
16:54.32ManxPowermort_gib: what DO you want?
16:55.56mort_gibMe?? -To be left alone! But I need to alert a caller to the fact that the "extension" they are calling is busy, without limiting the ingoing/outgoing calls
16:55.59tftechHello, I am getting an H323 error on the latest verison that is not letting me run asterisk, anybody else getting this? I can provide more specific info
16:56.30*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
16:57.09ManxPowermort_gib: you just said you don't want to play a sound file.  How do you want to notify the caller?
16:57.10[TK]D-Fendermort_gib: Well you're calling this parrty (or in the dialplan about to do so), besides playing a sound, what else could yousexpect?
16:57.32ManxPowerputs on his tin foil hat and glares at [TK]D-Fender
16:58.04mort_gibDunno, options I guess...
16:58.25[TK]D-Fendersues ManxPower for infringing his licensed telepathic band-space.
16:58.38*** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir)
16:58.47mort_gibIn the end I have to record the sexy reception girl saying something :-)
16:58.47ManxPowermort_gib: IIRC ChanIsAvail has an option that will let you know the peer is in use, even if call waiting is enabled on the phone.
16:59.03[TK]D-Fendermort_gib: Guess you'd better be able to come up with what you expect if you want any kind of help
16:59.10*** join/#asterisk mroth_imm (n=chatzill@63.65.26.220)
16:59.46mort_gibI know, I have that working, my client wants, like a busy tone or a different ring tone...
17:00.03ManxPowermort_gib: THEN PLAY A BUSY TONE.
17:00.06mroth_immAnyone know a straightforward way to stop Asterisk from replying to a REGISTER with a '100 TRYING'?
17:00.06ManxPowerthis is not rocket science.
17:00.33mroth_immIt is not playing nice with our SBC.
17:00.53[TK]D-Fendermroth_imm: "vi chan_sip.c"
17:01.24mort_gibManxPower: Not a problem, just wondered...
17:01.31*** join/#asterisk errr_ (n=mike@fedora/errr)
17:01.39*** join/#asterisk lanning (n=lanning@66.151.128.195)
17:01.41ManxPowermort_gib: but you already knew the answer.
17:01.43mroth_immYes, that's quite straightforward.  I'm sure it's right in a define.
17:02.04ManxPowermroth_imm: I imagine not replying would violate the SIP RFCs.
17:02.26[TK]D-Fendermroth_imm: I'm sure it isn't TOO hard to hunt down, but it is that dirty a job...
17:02.37mroth_immFrom the vendor: 'this a known issue in the 3.5.1 release, as the 100 Trying is not normally provided to the REGISTER method.'
17:02.43ManxPowerAsterisk was never designed to give access to low level SIP stuff.
17:02.51errr_when using the manager for QueueStatus it says it takes a parameter of ActionID.. what are the valid actionids?
17:02.55ManxPowerUse OpenSER if you want low level protocol stuff
17:02.58mroth_immI think it expects just an OK or a 40[3-4]
17:03.14ManxPowermroth_imm: 404 means "not found"
17:03.22*** join/#asterisk Defraz (n=T0tal@fw.fuzecore.com)
17:03.29mroth_immYou get the point, but thanks for flexing you e-muscle.
17:03.58mort_gibManxPower: No I did not. I knew ONE way of doing this, but clients ALWAYS come up with another request. That is way we use * because of the flexibility
17:04.50[TK]D-Fendermort_gib: Flexibility is nice, but knowing what you want is more important. what you want is infinitley
17:04.52*** join/#asterisk gbr_ (n=gbr@200.103.96.98)
17:04.58[TK]D-Fenderdang split line.
17:05.05[TK]D-Fendermort_gib: You catch my drift...
17:05.10mroth_immAnyway, no would've been a good enough response, but I guess that's not in your vocabulary.
17:05.30*** part/#asterisk mroth_imm (n=chatzill@63.65.26.220)
17:05.57mort_gib[TK]: Yes, I just wondered how other people delt with this....
17:06.19*** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net)
17:06.42[TK]D-Fendermort_gib: We personally eith just chove them to VM, or just call the phone.
17:06.57[TK]D-Fenderok, I'm going to go caffienate some more...
17:08.00*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
17:08.27KattyMikeJ: ;)
17:08.32KattyiCEBrkr: well of course i'm not insane.
17:08.41mort_gibYeah, which I did initially....
17:09.14anonymouz666app_queue is definitely mad. it reports members as "IN-USE" and that members has a call-limit=99 configured and "show channels" does not show this member in ANY call.
17:09.31Kattyanonymouz666: offer it some chicken salad.
17:10.31anonymouz666Katty: heh, how are you?
17:10.33StephenFSo is using something like a Cisco 800 router with a VPN tunnel back to the * box the cheapest way to have a telework's phone connect to the office network?
17:10.45Kattyanonymouz666: starving. also, eating chicken salad.
17:11.05Kattyanonymouz666: there's also promise of getting a puppy this weekend. i'm quite excited.
17:11.11KattyiCEBrkr: but not insane.
17:11.27iCEBrkrhehe
17:11.46Kattyi went to petco, and i was looking through the puppy sweaters.
17:11.59Kattythat's how excited i am.
17:12.13Kattyi don't even have pictures of the pups yet
17:13.22*** join/#asterisk krokodilerian (n=vasil@63-255-103-7.ip.mcleodusa.net)
17:16.19*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
17:16.20anonymouz666Katty: can I send to the picture of my puppy?
17:16.23puzzledhi
17:16.24anonymouz666you
17:16.41mort_gibStephenF: I have 8 users that connect via Cisco 877/ADSL lines
17:16.58StephenFindividual users? So they each have an 877?
17:17.03StephenFor is that a branch office
17:17.13*** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net)
17:17.16puzzledanyone know which option in Dial(ZAP/<option>/${EXTEN} makes asterisk use the last available line in the group?
17:17.35mort_gibStephenF: No, it's a branch office, they do Outlook/MS Exchange over the  same line :-(
17:17.44mort_gib512 kBit :-(
17:17.45StephenFgotcha
17:17.47StephenFouch
17:18.17mort_gibI would be tempted to use Soekris or similar running OpenBSD
17:18.18krokodilerianpuzzled , yep, use G1 instead of g1
17:18.28puzzledkrokodilerian: thanks
17:18.31mort_gibGives you MUCH better QOS
17:20.19mort_gibYou are going to end up having to deal with explaining to users that limewire and VOIP are not good bed fellows
17:20.52Kattyanonymouz666: SURE!
17:21.10Kattyanonymouz666: i love puppeh pictures
17:24.15mort_gibStephenF: But you have to keep an eye on latency (of course)
17:28.12Kattythinkgeek and jinx need to make doggy shirts.
17:28.49KattyQwell: i could see a dalmation wearing a More Dots! More Dots! shirt.
17:28.57Qwellgroans
17:29.54*** join/#asterisk nny_1 (n=Scott@64.203.237.47)
17:30.08nny_1ManxPower: hey mind if I pm you?
17:30.30QwellKatty: I may have to photoshop that later
17:30.33ManxPowermort_gib: most of the network I manage is 384K
17:30.38KattyQwell: :>
17:30.44ManxPowernny_1: Only if it involves me receiving money.
17:31.09ManxPowerOr something that REALLY is secret, I guess.
17:32.21nny_1ManxPower: the former
17:32.27ManxPowerthen PM away
17:34.41seanbrightholy crap
17:34.44seanbrightclay aiken is gay!?
17:34.47seanbrightwhy wasn't i told!?
17:34.48seanbrightheh
17:35.23*** join/#asterisk tkbeat (n=tk@p54B97062.dip.t-dialin.net)
17:42.03Kattyachmed the undead terrorist told me.
17:42.32creativxi thought he was dead
17:42.37creativx:o
17:42.44Kattyoh right.
17:42.55KattyQwell: Achmed, the undead mage. SILENCE!!! I SHEEP YOU!
17:43.01mort_gibManxPower: Yeah, but I HOPE you only run IAX/SIP over it!
17:43.10QwellKatty: stop breaking my sheep
17:43.22KattyQwell: do you have turtur tome?
17:43.27KattyQwell: or just the piggy
17:43.27Qwellno
17:43.30Katty:<
17:43.34Kattythere was one on the AH. 4500g
17:43.35Qwell5kg >.<
17:43.44Kattyyes, indeed.
17:44.56ManxPowermort_gib: We don't run any voice over it.
17:44.57mort_gibManxPower: I got "lucky" because the two ADSL links are on the same exchange.. Still bandwidth drops like a stone @ 17:00 when kids come home from School
17:45.06mort_gibAh, ok
17:45.13mort_gibSo only ?? MS Exchange
17:45.16ManxPowerStupid to try to run VoIP on a 384K frame relay network
17:45.24Kattysomeone said exchange
17:45.29Kattygoes to get rifle.
17:45.44krokodilerianManxPower , the world is full with such people
17:46.08ManxPowerkrokodilerian: Maybe so, but I don't interact with those people.
17:46.16krokodilerianManxPower , people try to run voip on an unguaranteed bandwidth and then complain to us when it fucks up at the moment some idiot is downloading porn from torrent
17:46.24krokodileriantorrents
17:46.29krokodilerianManxPower , then you must live in a happier place :)
17:46.36mort_gibYeah, I managed to get some 8-9 users to work over a 512 K link, VOIP and all their dear MS stuff
17:46.47ManxPowerkrokodilerian: I just don't currently accept stupid clients.
17:46.58krokodilerianmort_gib , 729 for the voip?
17:47.17mort_gibI happen to manage some 35+ networks, I don't EVER get to decide what technology they implement. Sure I get to suggest, but...
17:47.20mort_gibYes!
17:47.28mort_gibG729 for VOIP
17:47.32krokodilerianManxPower , how do you know if they're stupid? i'll buy you a beer if you tell me how you do that if it doesn't include talking to them :)
17:47.56krokodilerianmort_gib , hm, so then the adsl device was not a shitty one
17:48.09*** join/#asterisk stephank (n=urk@82-197-207-120.dsl.cambrium.nl)
17:48.11krokodilerianwe had to explain to a customer that the router they had couldn't push the voip traffic as it couldn't handle the pps
17:48.23ManxPowerkrokodilerian: if they want to run voice over a 384K network and I cannot change their mind, then they are stupid.
17:48.49mort_gibWell... QoS on the Cisco Soho devices are mostly mentioned in the pamflet that comes with the device, for good reason!
17:49.52mort_gibManxPower: I changed to consultancy BECAUSE I wanted to retain my right to say "Told you so" and not get a "I don't want any negative attitude"
17:50.30mort_gibI was quite open about the risk this particular client was (is) running
17:50.48mort_gibin the end we will setup a WiFi link between offices
17:50.52Katty[TK]D-Fender: I found a very interesting sign. "German Shepherd Security - We Don't Call 911"
17:51.43*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
17:51.43*** mode/#asterisk [+o russellb] by ChanServ
17:53.54Kattyhai russell
17:54.11*** join/#asterisk jewfish (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
17:54.45jewfishHello.  I'm trying to figure out the significance of the underscore in the asterisk dialplan.  i.e., what's different about _9NXXNXXXXXX and just 9NXXNXXXXXX?  Thanks for any help.
17:57.50ManxPowerjewfish: you need to read the Asterisk book.
17:57.57ManxPower_ means "pattern match"
17:57.58ManxPower~book
17:57.59jbotwell, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
17:57.59*** join/#asterisk roe_ (n=roe___@216-164-160-45.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com)
17:58.11jewfishtnx
17:58.25*** part/#asterisk nny_1 (n=Scott@64.203.237.47)
17:59.59roe_I know this is a tough question to ask and I have done my research.  When comparing the polycom 320 with the linksys sp942 - both of them in about the same price bracket (in the US) the features of the spa942 blow the polycom out of the water.  I have to be missing something but I don't know what
18:01.03ph8hi guys, I have a SIP hard phone in my room at uni, I need to tunnel it through my main PC (i'm fine setting up tunnels) to the registration server - what ports will the phone be using? Just the one for SIP?
18:01.08ph8(what is the port for SIP?)
18:02.30Kobaz5060
18:02.40Kobazyou're better off using iax
18:02.41ph8cheers, i'll try just that and hope that's the only blocked thing
18:02.50Kobazmuch more NAT/firewall friendly
18:02.53ph8is IAX a better protocol?
18:02.56ph8i see
18:03.02ph8it's not my firewall unfortunately or else it wouldn't be a problem :p
18:03.06Kobazwell, it's better for certain things
18:03.10Kobazsip has more features
18:04.04Kobazph8: you can set up a little asterisk box so you can use your sip ahrd phone to asterisk. and then send the call over iax to the remote system
18:04.08Kobazor just tunnel the whole entire thing
18:04.18ph8oh sorry i do
18:04.22ph8asterisk takes the phone by SIP
18:04.25ph8then makes outbound calls by IAX
18:04.28ph8although it receives calls by IAX
18:04.30*** join/#asterisk BobPierce (n=WESTMAN\@216.36.132.162)
18:04.31Kobazokay, that should work nicely
18:04.38ph8i'm trying to just get the phone out to my remote * server
18:04.44ph8via my desktop i guess
18:05.04Kobazso what's the real problem
18:05.05[TK]D-Fenderroe_: Polycom can handle the same number fo calls, and better, has a better build & audio quality, microbrowser support, etc, and costs less.
18:05.21ph8Kobaz:  What i said, i have a phone i can't connect to my * server because the phone's behind a university firewall
18:05.33ph8so 5060 is SIP, will outbound be any different?
18:05.35Kobazcan you tunnel through it?
18:05.47ph8yep
18:05.51ph8so i've got one tunnel for 5060
18:05.53Kobazwell for sip you need 5060, and then the client port on the way back in
18:05.54ph8which i'm just testing
18:06.14ph8will the client port be random?
18:06.16Kobazwhich you can force to be within a specific range
18:06.16Maliutaph8: don't forget to pass the rtp ports aswell
18:06.22Kobazyeah the client rtp ports
18:06.25ph8i read RTP, wasn't sure what it is
18:06.27ph8is it a range?
18:06.32Kobazmedia / audio
18:06.37Kobazyou can set a range
18:06.42Maliutayeah a range of ports used to do the media trasmission
18:06.47ph8not in /etc/services
18:06.58Kobaznothing in /etc/services to monkey with
18:07.12ph8i would have used it to find the rtp ports :p
18:07.13Maliutaph8: because it's user defined
18:07.14ph8just googling now
18:07.27Katty[TK]D-Fender: http://shop.cafepress.com/design/12712455
18:07.29ph8ahhh
18:07.33ph8the 'local rtp port' on the phone?
18:08.03[TK]D-FenderKatty: What do you have by way of pets already?
18:08.31ph8hmm
18:08.34roe_[TK]D-Fender, While there is a $32 difference, the linksys handles 4lines instead of 2 and the screen is 50% and is backlit
18:08.35c4t3lis interested to know about this after the dog naming from yesterday
18:08.35ph8can i tunnel *in* to my phone like this?
18:08.39ph8does that make sense?
18:09.13roe_I guess that $32 pays for those extra features
18:09.28c4t3lph8: is firewall stateful.   keep in mind the problems with some firewall and SIP
18:09.40ph8not a clue unfortunately
18:09.42ph8one would hope so
18:10.10ph8right so i've set my phone to use my machine for sip and 'outbound' (what the phone calls it) and my machine forwards 5060 and 5004 to my * server
18:10.16ph8s/forwards/tunnels
18:10.36Katty[TK]D-Fender: just the ferrets. does the boyfriend count?
18:10.56ph8hmm no joy :/
18:11.07MaliutaKatty: that depends on what you do to him
18:11.13KattyMaliuta: erm.
18:11.17KattyMaliuta: it was a joke.
18:11.18[TK]D-Fenderroe_: SPA supports 4 calls, IP320 supports 2x2=4.  Same thing.
18:11.23MaliutaKatty: does Freenode have a #BDSM?
18:11.29KattyMaliuta: i wouldn't know.
18:11.40MaliutaKatty: you say that now
18:12.03Kattyc4t3l: i need a german speaking person to help me finish the name thing :<
18:12.42Maliutadog naming?
18:13.09[TK]D-FenderKatty: Depends... does he have a leash too? ;)
18:13.25KattyNext!!!!
18:13.32*** join/#asterisk stencil (n=stencil@69-196-130-202.dsl.teksavvy.com)
18:13.43Maliuta[TK]D-Fender: no, he just has a ball gag and some restraints :)
18:13.51l2trace99is there a way to get the qualify value for realtime peers ?
18:15.07l2trace99like the from the output from a sip show peer | user ?
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18:18.11nny_2ManxPower: quick q that power supply is a regular 48v, i know those units use negative 48. Still works though right?
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18:27.38keith4where would I find out how much power a Polycom SIP320 draws, over PoE? I don't see it in the PDF manual
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18:31.53ManxPowernny_2: they are what my customer uses
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18:35.46nny_2ManxPower: cool sounds good thanks
18:36.19nny_2ManxPower: the telco is testing themselves for echo :\ tonight. I don't trust them, but have asked for full details of the test.
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18:44.08StephenFkeith4: no more than 15.4 W?
18:45.35StephenFi dunno, you probably need to call them unless someone here has measured it
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18:47.08swampworkI've got a Polycom 501 that I'm trying to upgrade firmware on.  I've never done it before.  I am a near-total n00b.  I have a tftp server on my internal network (same as the phone's on).  I have a .cfg file named the same as the phone's MAC address (0004fd.......cfg) .  I've told the phone the IP address of the TFTP server, and verified from another machine that the cfg file can be retrieved via tftp.  But the phone insists that there's an error retrievin
18:47.29swampworkIs there a step-by-step, so-specific-a-monkey-could-do-it walk-through for doing this?
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18:50.04jayteeso far we haven't been successful in creating a phone provisioning guide that monkeys could use, we have however created a political system so simply screwed that even a chimpanzee can be President.
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19:01.48MindTheGaphello all, asterisk 1.6.0 rc6 getting choppy moh... there is a digium te110p card locaded and correctly configured (although its running w a loopback connector to clear red signal). zttest shows 99.98 all the time, no irq sharing, no . Playback works fine but moh quality is crappy (playing a .wav file). Does the digium board requires a live connection to generate proper timing? I even tried a realtime kernel but it changed nothing... this server is
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19:07.23MindTheGapztdummy also wont change a thing, by the way the kernel is a 2.6.24-19-rt and since ztdummy on 2.6 takes the timer from the rtc i thought it would make the problem dissapear... but it says.
19:07.42MindTheGaps/says/stays
19:07.58swampworkjaytee: So you're saying I should just get myself elected president and have someone ELSE provision the phone for me?
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19:08.50swampworkI was afraid of that.  So what's the best provisioning howto/walk-through/source of useful information you would suggest ?  Even if it's not monkey-usable.
19:11.16[TK]D-Fenderswampwork: There is a decent guide on the WIKI
19:11.18[TK]D-Fender~wikis
19:11.19jbot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
19:11.36jayteeI like following the provisioning example in "the book" along with the Polycom SIP Admin Guide and their whitepaper on provisioning which makes it all seem pretty simple once you've read that. I now have provisioning working and I just got through writing a shell script that I just type the name of with the mac address 1st line key extension number etc and it generates my master config file for that phone and the unique settings <mac>-phone.cfg file
19:12.29swampworkjaytee: Nice!  Care to share such shell script goodness?
19:12.48jayteesure but it has it's limitations
19:13.13jayteeright now I've got two scripts I'll be merging, one is for 550's and the other is for 330's.
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19:14.04jayteeand it "assumes" that the first two line keys are set to the same sip account.
19:15.13swampworkjaytee: That's no problem; I'm not going to use it blindly, I'm just looking for reference stuff that's actually *working* somewhere to read and compare to.
19:17.18jayteefirst off, you need to read the section in the book
19:17.21jaytee~book
19:17.21jbot[book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
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19:19.19Maliutamm book
19:19.21jayteeswampwork, read pages 85-93 of the book and reference a couple of the howtos on the wiki for the FTP setup and make sure you're using FTP and not TFTP.
19:19.33keith4~sip phones
19:19.37geoff2010does MeetMe transcode all audio streams regardless?  i.e. if i have two g729 channels will MeetMe do straight pass-thru, or will it reserve 2 licenses and decode/encode everything?
19:19.39keith4~phones
19:19.39jbotmethinks phones is http://bani.anime.net/phones/.  While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows:  Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else.  Do not consider Grandstream phones.  Ever. places like such as
19:20.43keith4pets jbot
19:21.04jaytee~pb
19:21.04jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
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19:24.11swampworkjaytee: Cool.  I'm off to read.  I was trying to use tftp, but I guess that's not a good idea.  I'll fire up an FTP server and get with it.  I'm trying to work through the voip-info wiki guide, but the tftp failure has been throwing me off.  So I'll see if I can get it working with FTP.
19:24.29jayteeswampwork, hang on a second
19:26.20[TK]D-Fendergeoff2010: yes, it'll kill 2 licenses, and the mix load on CPU will be more significant
19:26.21ManxPowergeoff2010: You cannot mix G729 audio.  All audio must be converted to SLIN for mixing
19:26.55swampworkjaytee: Yeah?
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19:27.31jayteethis is a shell file I use to create the config files for a 330. http://pastebin.com/ddbec48
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19:27.59xaiAnyone know of a webconferencing suite that uses * ?
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19:29.21swampworkjaytee: Coolness!  Thanks much.
19:29.24jayteeswampwork, and this is the file I named 000000000000.330   http://pastebin.com/m3e2174e0
19:29.41swampworkExcellent!
19:29.42gaetronikhi there
19:30.24jayteeswampwork, IMPORTANT. on the second line of the file change the 000000000000.cfg to 000000000000.330
19:30.34gaetroniki sometime have agents which stay conected whereas the softphone is closed
19:30.54jayteeswampwork, the second line of the prepphone.sh script I pastbined first
19:34.48swampworkOK, got it.
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19:37.14jayteeswampwork, so I use a different master config template for each type of phone, the sed just replaced certain preset strings for the extension number passed in the second command line parm.
19:38.26jayteeI'm doing a similar thing with 550's and when I get done I'm merging the two script files into one so I'll just pass it phone type, mac address and lines 1-4 or 1,3-4 in my case since I use the first two line keys for the same number.
19:40.22gaetronikno one have the same issue?
19:40.46swampworkNice.  BTW, shouldn't there be a space between "/overrides" and CONTACTS_DIRECTORY in the 000000000000.330 file?
19:41.10swampworkIs the 550 one significantly different than the 330 one you pasted?
19:42.25gongoputchanyone use Gizmo ?
19:43.57eric2hmm, what's the command to uninstall asterisk?  (ya, probably a dumb question)
19:44.07Qwellmake uninstall
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19:45.03SexyKenAnyone ever setup a Trunk based on IP and no registration for terminating or origination?
19:45.11eric2ah yes... thanks Qwell
19:46.32geoff2010ManxPower: i guess i was wondering if there was only a single speaker in a MeetMe conference, does it do straight packet pass-thru if both users are G.729
19:46.35SexyKenI'm failing at every attempt.
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19:49.00SexyKenAnyone even use iCAll?
19:49.08ManxPowergeoff2010: Meetme always transcodes AFIK
19:49.35ManxPowerSexyKen: you mean like the permit= and deny= in sip.conf/iax.conf?
19:51.14gaetronikis this a known isue the issue of agent which look like conected in asterisk whereas the softphone is down?
19:51.25jayteeswampwork, yes the 550 uses a differnet sip.ld file, Polycom broke them out into smaller files per phone
19:54.05SexyKenManx:  Maybe that's where I'm messing up?  You're supposed to have permit= and deny= ?
19:54.07citywokdoes anybody do Asterisk consulting type stuff?  I wnat to use the ZapScan/ZapBarge feature to monitor channels in a way that would have to link in some weird external shit, that is over my head
19:54.34ManxPowerSexyKen: Try looking in sip.conf.sample, the first place to look for sip.conf questions
19:54.54ManxPowercitywok: I do consulting, but would not touch that project.
19:55.18citywoki'm guessing its a few hours of work for somebody that knows what they are doing
19:55.51ManxPowernot if you are using zapscan/zapbarge
19:56.17citywoki can already use it to camp on a line, but i want to be able to build a "pool" of people to scan
19:57.15citywokI want to bea ble to dial in, and monitor active calls, would it make more sense to use a bridge of some sort, and just bridge in/out calls programmatically from the API?
19:58.17citywok(i am a call center, and i've built a call recording system, but people want to be able to "camp on" and listen to calls)
19:58.52jayteeswampwork, this is a template 000000000000-XXXX.cfg file I use for that script for 330's http://pastebin.com/d2eae38ee
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20:00.41brookshiredoes anyone have pics of astricon
20:01.32citywokManxPower, would it be possible to dial into my asterisk box, and then use a web interface/API to bridge two zap channels together?
20:01.51*** join/#asterisk SamuraiDio (n=diovani@200.180.158.162)
20:01.53SamuraiDiohi
20:02.19citywokwell, not bridge because i dont want it to be a conference, i want to barge
20:02.25gaetronikcitywok, with a script which make ami call
20:03.01citywokwould it have to be an AMI, or an API call?
20:03.49SamuraiDiocan i have an asterisk cluster (two or more asterisk servers) acting simultaneously (with auto balancing) able to be accessed/used (by users, peers, and cli) in a single address?
20:05.42geoff2010does anyone know how asterisk behaves if you have two SIP peers with the same IP address.  this is an unregistered peer, just does a sip_poke to determine if it's alive.  I am wondering how asterisk decides which peer record is used for inbound calls.
20:07.36swampworkWell, my problem remains in that the phone refuses to load the 0004f.......cfg .  I'm using FTP now, and (again) I've verified that I can ftp in as PlcmSpIp and retrieve that file.  But the phone says "Unable to contact boot server, using existing configuration", then a few secs later says "Error loading 0004fblahblah.cfg"
20:08.01swampworkWhich is the same behavior it's given me with the TFTP server as well.  So having config files won't help if I can't coax it to connect to the FTP server and load a new one.
20:09.23SexyKenHrm - Okay, I got outgoing calls to work - but on incoming, it seems to start this undless loop of calling the DID that was actually supposed to be calling us.
20:10.02citywokManxPower, how would i execute ZapBarge(CHANNELNUMBER) from the api/ami, on a specific channel?  Say i'm on channel 17, and i want to Barge in on channel 2
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20:12.29swampworksunuvagun  - it's doing *something* now .  Apparently it was so hosed that it had forgotten the "default" FTP username/passwd combo that the docs said to use.  I put them in and now it's doing things to itself.  I'll hope they're healthy things
20:14.13jeevhttp://rawr.pastebin.com/d39a3318f anyone know what i could do?
20:14.27jeevi guess just add it after
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20:25.00citywokdoes anybody know how i would execute zapbarge on an open zap channel from the api?
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20:26.56AlexTOHi, i'm making and new instalation of asterisk 1.4 on debian and it is fine but on my /etc/asterisk folder the user and group after the installation is root:root and usuarlly is asterisk:asterisk , so i wonder to know if it fine or not? thanks
20:27.22nny_2i have a pri channel in an alarm state and a tech heading over to test the card with a loopback cable. Is there anything else I can do to see if this is the telco having issues or asterisk?
20:27.44nny_2PRI span 1/0: Provisioned, In Alarm, Down, Active
20:27.48nny_2is what it says right now
20:27.49*** join/#asterisk jplank (n=GBove@158.sub-75-208-68.myvzw.com)
20:28.04nny_2wcte12xp: Setting yellow alarm
20:29.24nny_2willing to paypal anyone that wants to help
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20:30.00citywokyou could build a loopback cable yourself, and tell them to try and loop it from there end
20:30.21citywokloopback cable = rj45 connector with a couple wires jumping the pins
20:30.22nny_2yeah someone is heading over there right now to hook the loopback up and see if the alram clears
20:30.30nny_2yeah made one already
20:30.38citywokjust do that, and they dont need to send anybody over
20:30.39citywok:-)
20:30.49citywokthey can test it from their CO
20:30.55*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
20:31.45ManxPowernny_2: RED or YELLOW alarm?
20:31.51nny_2ManxPower: Yellow
20:31.56nny_2ManxPower: according to dmesg
20:32.02ManxPowernny_2: no red alarms?
20:32.02citywokthen its a signalling problem?
20:32.13ManxPoweryellow is not a signalling
20:32.33nny_2Detected alarm on channel 5: Red Alarm
20:32.36ManxPowernny_2: I would have the telco "do a loop test to the smartjack"
20:32.38nny_2from asterisk messages
20:32.51nny_2tech is on site going to test via loopback cable
20:33.00citywokManxPower, is it possible to send an extension to a context from API?
20:33.10ManxPowerRED alarm is line problems that SHOULD be seen by the telco when they do the loopback to the smartjack.  You don't do anything for this.
20:33.18citywoki.e. send zap/13-1 to context asdfasdfasdf
20:33.19nny_2they said the card is bad
20:33.23nny_2the digium card
20:33.26ManxPowerin fact, you don't even have access to the wiring required for a loop test
20:33.41ManxPowernny_2: then they will find no problems when the telco does that
20:33.45krokodileriannny_2 ,yellow means you don't have asterisk running
20:33.50krokodilerianor not configured right
20:33.57nny_2krokodilerian: explains the dmesg, as i restarted both
20:33.58krokodilerianso either zapata.conf or asterisk
20:34.05krokodilerianwell
20:34.19krokodilerianjust run asterisk, ztcfg -vv once, and see what does asterish say on zap show status
20:34.59nny_2http://pastebin.com/m7498440c zapata.conf
20:35.31nny_2Wildcard TE122 Card 0                    RED        1          0          0
20:35.37krokodilerianhm
20:35.41krokodilerianRED is another thing :)
20:35.53nny_2yeah dmesg said yellow, asterisk says red
20:36.04krokodilerianwhat does zttool say
20:36.12krokodilerianor hm, /proc/zaptel/ ... whatever that was
20:36.19jplankwithout starting a flame war (and if I am, please stop me) could someone explain the pros and cons of digium cards vs sangoma cards?
20:36.26nny_2red
20:36.31nny_2has a loop option
20:36.44krokodilerianjplank , sangoma - better hardware, bigger price, a bit weird drivers
20:37.08krokodilerianjplank digium - not so good hardware, i think cheaper, somewhat better support in zaptel/dahdi
20:37.15nny_2IRQ Misses:               1   Total/Conf/Act:      24/ 24/  0
20:37.31nny_2bipolar says 0, rx/tx levels say 0
20:38.04nny_2can i use the loop feature of zttool to test it that way
20:38.25krokodilerianwell, i think you also needed someting on the other side to see the loop,etc.
20:38.40jplankkrokodilerian: the sangoma guy I was talking to said dahdi works with their cards, but the prefer netborder
20:38.45nny_2i have a tech heading over now and on hold for digium support
20:39.10krokodilerianthe checklist for pri problems goes like this - first, is the cable ok, is the zaptel.conf right (signalling parameters) and if those are ok, you either get YEL or OK
20:39.12KattySexyKen: moooooo.
20:39.21krokodilerianor, of course , the card might be fucked up
20:39.40*** join/#asterisk FinboySlick (n=FinboySl@207.134.8.4)
20:39.40krokodilerianjplank netborder ?
20:39.41jplankjust bothers me to install a media gateway ontop of asterisk
20:39.51nny_2if i loop back test and get green does that clear the card?
20:39.58jplanknetborder.com
20:40.17krokodilerianoh, their weird driver?
20:40.29jplankits a software media gateway that works as a driver, and makes Asterisk think its just a SIP trunk
20:40.38jplankor multiple sip trunks whatever the case maybe
20:40.41*** part/#asterisk CrazyTux (n=root@ip68-111-67-4.oc.oc.cox.net)
20:41.00krokodilerianwhy the hell would I want to see my PRI card as a SIP trunk, that's almost retarded
20:41.34jplankI asked the same question
20:41.37jplankfunny enough
20:41.46FinboySlickI need a bit of inspiration here.  An application here would require us to pipe a pcm stereo signal over TCP/IP at relatively low latency.  I'm sure this is something asterisk can do but it's a tad overkill for my need.  IceS2 + Icecast is way too slow (>3sec latency), so, any advices?  I don't need a whole lot of compression.
20:42.07krokodilerianFinboySlick , socat/netcat?:)
20:42.18krokodilerianFinboySlick , should be relatively easy to tune it for latency
20:42.27jplankbasically his answer was it doesn't matter how * sees the trunk, as long as it can communicate with it, and brought up the point * works a lot better as a SIP PBX then a TDM one
20:42.42jplankthen again, he also had a glass of vino in his hand, so who knows
20:42.45FinboySlickkrokodilerian: I've pondered something like that, but what about the endpoints?  I think raw PCM might be a tad too big for me.
20:42.48krokodilerianyes, really, and you do faxing reaaaaally easy with sip and asterisk
20:42.58krokodilerianFinboySlick , well, it's easy to calculate
20:43.16jplankthats what I said, but since it wouldn't really be a SIP trunk, faxing wouldn't be a problem
20:43.25jplankthats what he said at least
20:43.35krokodilerianjplank , i'll go talk to him again today probably, i spoke with him on running a test for me on one of their BRI cards
20:43.51AlexTOHi, i'm making and new instalation of asterisk 1.4 on debian and it is fine but on my /etc/asterisk folder the user and group after the installation is root:root and usuarlly is asterisk:asterisk , so i wonder to know if it fine or not? thanks
20:43.56krokodilerianjplank , also... how the hell do you use a sip trunk for timing :)
20:44.18*** join/#asterisk BiG_NoBoDy (n=BiG_NoBo@88.223.34.50)
20:44.32BiG_NoBoDyhy
20:44.33BiG_NoBoDy(11:26:27 PM) BiG_NoBoDy: does any one knows how to make Asterisk send fax via email when fax is retrieved?
20:44.33BiG_NoBoDy(11:27:59 PM) BiG_NoBoDy: and how to set up fax server that it should send faxes retrieved via email (postfix)
20:44.33BiG_NoBoDy(11:28:28 PM) BiG_NoBoDy: using debian and asterisk 1.6 and hylafax and avantfax
20:44.33BiG_NoBoDy(11:28:36 PM) BiG_NoBoDy: at the moment
20:44.48jplankfunny enough, these are the same questions I asked him
20:45.20krokodilerianand what did he say about timing?
20:45.25jplankhe said the clocking would be taken care of by the card itself and netborder, since * only sees a SIP trunk, * wouldn't care about the clocking
20:45.26*** join/#asterisk devhen (n=devhen@70-58-99-185.slkc.qwest.net)
20:45.59jplankand since it would be all sangoma devices (FXO, PRI ect) the timing issue would be taken care of by netborder
20:46.06krokodilerianso * doesn't need timing at all ?:) wow, i nevr knew that...
20:46.10jplank* wouldn't ever have to know they are actually TDM devices
20:47.04jplankif you don't have TDM devices as far as * is concerned, what do you really need the timing for? (taking meetme out of the equation)
20:47.16krokodilerianjplank , iax trunking
20:47.32jplankI ran a asterisk server with SIP trunking only for two years before ever having to instal zdummy for timing
20:47.32krokodilerianjplank , some t38 fax transmitter that needs good timing (that we wrote)
20:47.37krokodilerianand tons of other stuff
20:47.57jplankwe make it a point not to support t38
20:47.59krokodileriani always tend to have at least ztdummy on my asterisks
20:48.06jplankand I haven't really messed with IAX all that much
20:48.17krokodilerianjplank: ... we make a product for that, not really an option :)
20:48.31jplankyea, I installed ztdummy on ours a couple months ago for meetme
20:48.33jplankwe?
20:48.34*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
20:48.46krokodilerianjplank , attractel (the thing is called attrafax_
20:49.01jplankt38 hardware?
20:49.01krokodileriani'll be talking a bit on timing at my talk today (it's in an hour)
20:49.06krokodileriansoftware
20:49.11jplankwhich conference?
20:49.44jplankerrr
20:49.45krokodilerianastricon, i'm in cira B
20:49.48jplankwhich track should I say
20:50.22jplankmeasuring signal quality in a hybrid system?
20:50.46krokodilerianjplank , yes, although the title is a bit different
20:50.54jplankI'm just going by the book
20:51.08krokodilerianand i'm not joachim vanheverzwijn :) we didn't know if i would be able to travel, so that's why it's his name over there
20:51.23jplankI was going to go to the ss7 one, but I'll come to yours instead
20:51.51krokodilerian:)
20:52.05krokodilerianfor mine I have the whole talk written, so you can read it afterwards, I don't know if it will be the same for ss7
20:52.12krokodilerianand damn, i wanted to attend that one too
20:52.17*** part/#asterisk gbr_ (n=gbr@200.103.96.98)
20:52.24krokodilerianbut not showing for my own talk would be somewhat impolite
20:52.29jplanklol
20:52.31jplanksomewhat
20:52.37*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
20:52.37*** mode/#asterisk [+o russellb] by ChanServ
20:53.04jplankso in your personal experience, you'd go which way, digium or sangoma?
20:53.36krokodilerianjplank , what for, e.g. PRI or TDM, what kind of application?
20:53.43jplankPRI
20:53.48jplankor even TDM
20:54.01krokodileriani have mostly used digium cards as we sell them, but have seen so many of them dead that it's a bit worrying
20:54.04jplankfor a customer installation
20:54.53jplankI guess my best bet would be to get one of each and beat them up
20:54.53krokodileriandigium cards will take less time to setup
20:54.54krokodilerianpretty much sure about that
20:54.54ManxPowernny_2: RED means "line not connected"
20:54.56krokodilerianfor the rest... toss a coin :) sangoma is mre reliable
20:55.07jplankso I've heard
20:55.08krokodilerianat least hardware wise, on dying ports,etc
20:55.21jplankreliability is more important then ease of setup
20:55.39krokodilerianjplank , so then either sangoma or one of the later digium cards
20:56.02jplankwell then, that answers my question :)
20:56.24jplank</sarcasm> just in case
20:56.58krokodilerianif you're not paying it, and you won't need the tdm stuff in asterisk for timing, etc, go with sangoma :)
20:57.06krokodileriantheir guy can probably show you how the installation works
20:57.12*** join/#asterisk VJFROMGT (n=vjfromgt@user-12lcpfg.cable.mindspring.com)
20:57.18krokodilerianwe can in fact go torment him together later on :)
20:57.44jplankplease
20:57.44jplanklol
20:57.44VJFROMGTi need a way to convert SIP to a PRI without a asterisk machine
20:57.44VJFROMGTcan someone tell me of a hardware?
20:58.47*** join/#asterisk implicit- (n=bayan@unaffiliated/implicit)
20:59.09jplankkrokodilerian: I'm going to hold you to that torment comment
20:59.21*** join/#asterisk neurosys (n=neurosys@c-66-229-91-186.hsd1.fl.comcast.net)
20:59.23krokodilerianokay:)
20:59.26jplanklol
20:59.36puzzledVJFROMGT: look at patton.com
20:59.47jplankahhhh
20:59.49jplankNOT PATTON
20:59.51jplankplease
21:00.00jplankbetter off with Adtran
21:00.00nny_2http://pastebin.com/m7498440c <-- can someone peep this and see if anything looks out of place for a standard t1
21:00.12jplankVJFROMGT: Adtran TA904
21:00.25puzzledor cisco
21:00.37nny_2krokodilerian: loopback test and patlooptest show green/ no errors
21:00.45jplanknever got to use any of the Cisco IADs, Adtran did the trick
21:00.52jplankPatton gave us NOTHING but problems
21:01.03jplankand their config is backwards
21:01.04*** join/#asterisk rasterix (n=IceChat7@80.177.176.254)
21:01.36rasterixhi is it possible to retrieve config files including white space and comments through the asterisk manager interface?
21:01.44krokodileriannny_2 , show also the zaptel.com
21:01.46krokodilerianconf
21:02.12krokodilerianzapata looks fine
21:02.55nny_2krokodilerian: http://pastebin.com/m5996a5f5
21:03.06*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
21:03.46nny_2krokodilerian: mind you this all worked until 14:30 today
21:04.01krokodileriannny_2 ahhaaaa
21:04.07krokodileriannny_2 , how many ports do you have on that card
21:04.13nny_2krokodilerian: 1
21:04.22krokodileriannny_2 , any replacements handy?
21:04.26nny_2krokodilerian: no
21:04.28*** join/#asterisk xenonex (n=xenonex@89.218.233.88)
21:04.40nny_2krokodilerian: but all self tests show that it is fine
21:04.42krokodilerianif it just stops working, most of the time it's an issue with your hardware, the one of the carriers doesn't do that often
21:05.04nny_2the carrier has been adjustng things on their ends for other reasons from what we understand
21:05.29*** part/#asterisk BiG_NoBoDy (n=BiG_NoBo@88.223.34.50)
21:05.56krokodileriannny_2 , and so there was a technician where, where the machine with the PRI is, or on the other end
21:06.16krokodilerianthey usually have something they can connect on the PRI and see the signal
21:06.17nny_2both afaik but they haven't been very helpful
21:06.33nny_2krokodilerian: they said the card is transmitting, but the loopback test cleared it
21:08.20*** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net)
21:08.40ManxPowernny_2: YOUR tech or TELCO tech?
21:08.40nny_2krokodilerian: *not transmitting
21:08.46nny_2ManxPower: telco tech
21:08.59ManxPowernny_2: He should have a T-BERD for testing the PRI
21:09.08jplankwhat if you plug a loopback into the *?
21:09.12jplankdoes the card green up?
21:09.15nny_2our tech hooked up a loopback, i restarted ZAP and got green
21:09.17nny_2yes
21:09.30jplankwhen you connect to the smartjack it stays red?
21:09.31nny_2and patlooptest completed 60 second time out no errors
21:09.34nny_2yes
21:09.46nny_2and the T1 is flashing red regardless of the cable connection to it
21:09.52nny_2T1/smartjack
21:09.54jplankdoes the alarm light change on the smartjack when you plug the the * in?
21:10.00jplankgot it
21:10.00nny_2no it stays red regardless
21:10.08jplankit doesn't flash or anything?
21:10.17nny_2flashes red continuiously
21:10.35jplankthe alarm light on the smartjack flashes red? not solid?
21:10.38nny_2yes
21:10.46ManxPowernny_2: My prediction is that the telco will find nothing wrong, but the problem will be fixed.  Either that or the tech will say "Oh!", call the CO and the problem will be fixed.
21:10.48jplankwho's your LEC?
21:11.03nny_2ManxPower: I will buy you a beer if you're right, which i suspect you are
21:11.04nny_2Embarq
21:11.06nny_2:\
21:11.09rasterix< thinks manxpower is right
21:11.09jplanklol
21:11.21jplankwhat happens when you plug the loopback directly into the smartjack?
21:11.23*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
21:11.29nny_2haven't tred that
21:11.29ManxPowerRed alarms are almost always a telco line/smartjack/mux/etc issue or a bad cable between the smartjack and the telco
21:11.38nny_2is that safe to do, it's their equipment
21:11.42nny_2replaced the cable and tested it
21:11.44jplankagrees with ManxPower
21:11.52nny_2beer for everyone!
21:11.56nny_2and double for me :D
21:12.08jplankdid you plug a loopback into the smartjack?
21:12.23MindTheGap_ello all, asterisk 1.6.0 rc6 getting choppy moh... there is a digium te110p card locaded and correctly configured (although its running w a loopback connector to clear red signal). zttest shows 99.98 all the time, no irq sharing, no . Playback works fine but moh quality is crappy (playing a .wav file). Does the digium board requires a live connection to generate proper timing? I even tried a realtime kernel but it changed nothing. ztdummy also wont
21:12.32nny_2i can if thats ok, i don't wanna screw with their equipment and create a liability
21:12.34ManxPowerSmartjacks have a telco controlled loopback feature internal to them
21:12.53jplankManxPower: yea, but i've seen plenty of cases when the back end of the smartjack was bad
21:12.54jdnWESTAnyone had any luck getting a Sangoma A 102 Card into a 1U server?
21:12.55ManxPowerMindTheGap_: why have you not reported this as a bug yet?
21:13.03jplankso when we looped up the smartjack it ran clean
21:13.33jplanknny_2: you just made a quick loopback plug right? rj45 connector with pins 1-4 2-5?
21:13.37nny_2yes
21:13.45jplankthen yea, plug it into the smartjack
21:13.52jplanksee if the alarm light goes out
21:13.52ManxPowerWhen the telco says "line tests fine, we see your equipment" then I unplug the line and ask them if they see any change.  If they don't see a change then they are testing the wrong line (happened to me several times)
21:14.18nny_2ManxPower: hahaha
21:14.26MindTheGap_ManxPower, because i may be missing something. as i stated, the card has a loopback conenctor to clear alarms, dont know if it will generate the timing asterisk needs while in loopback
21:14.44jplankManxPower: we've had the issue where the lec would leave a soft loop up somewhere in their network, so when we try to run to the customers equipment, or loop up their csu, it seems to work, then we send the loop down command and everything starts working magically
21:15.05nny_2ManxPower: and these people somehow think they are better than us heh
21:15.21nny_2I say all playing fields are level until you make it a point to look stupid
21:15.37jplanknny_2: put the loopback into the smartjack and tell me what happens
21:16.11jplankif * greened up when you plugged the loopback into it, I doubt its a problem with the card, at the very least, it should green up
21:16.16jplankits probably a cable issue
21:16.19jplankor a smartjack issue
21:16.34nny_2jplank: loopback made it red
21:16.36jplankor your carrier or lec didn't put their cross-connects in (or properly)
21:16.40jplanksolid red?
21:16.52nny_2jplank: asking
21:17.00jplankwhich lights?
21:18.19nny_2jplank: ds1: cable in = green green red amber amber cable out green/red green red loopback = green/red green/red
21:18.29nny_2jplank: sorry playing monkey n the middle with my tech on site
21:18.45nny_2basically cable out and loopback are the same
21:18.47*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
21:18.57jplankthe lights don't change at all?
21:19.02jplankwith a loopback in
21:19.06jplanklike blink or anything?
21:19.33nny_2says t flashes green red
21:19.38jplankok
21:19.42*** join/#asterisk knobo (n=bohmersp@148.122.202.168)
21:19.42jplankthats something :)
21:20.15jplankwhats the label of the light that flashes green/red?
21:20.20jplankds1?
21:20.20nny_2jplank: stand by my tech is not communicating this well
21:20.30jplankalarm
21:20.31jplank?
21:20.54jplankrealistically, when you plug the lopoback into the smartjack, the alarm should green up
21:21.23jplankif it doesn't, theres def something wrong with either the smartjack, or at the lec (not necessarily your carrier)
21:22.03nny_2jplank: is there a light that specifcally states alarm
21:22.24jplankyea
21:22.30jplankwhat kind of smartjack is it?
21:22.34jplankadtran?
21:22.40jplankhyper_whatever
21:22.44jplank?
21:23.01nny_2jplank: woring on all five light status on 3 different set ups (loopback etc) i will get brand name
21:23.08nhuisman_workI wish there was a better mechanism for having the same # on multiple phones
21:23.16nhuisman_workinstead of having to make up sip users for each phone
21:23.33*** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org)
21:23.35krokodileriannhuisman_work , do a script that generates the config:)
21:24.22nny_2jplank: ds1 light is the only one that changes based on cable
21:24.36nny_2jplank: Westell
21:24.40jplankok
21:24.45nny_2case anyways
21:25.00nny_2jplank: adtran card
21:25.04jplankok
21:25.13jplanktheir should be a alarm light
21:25.32nny_2jplank: alm
21:25.32jplankthere should also be a ds1 light (which you confirmed)
21:25.35jplankyea
21:25.45jplankwhen you plug the loopback in
21:25.51jplankdoes the alarm light do anything?
21:25.56*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
21:25.57jplankit might flash quickly
21:26.19nny_2ok having him retest and confirm light status
21:26.24Katty[TK]D-Fender: they should make a Kong suitable for children.
21:26.50nhuisman_workkrokodilerian, configuring it isn't the issue, it's just needing to have sip users who somehow represent the X phones that the number rings on
21:26.54nhuisman_worklike X_1 X_2
21:27.03jplankif your plugging the loopback into the smartjack and the ds1 light is greening up (which confirms the the smartjack is seeing something being plugged in) but the alarm is staying red
21:27.12nny_2jplank: cable out no loopback alm is orange for a sec than red
21:27.19jplankthat means your carrier doesn't have their crosses in
21:27.20krokodileriannhuisman_work , e.g. an account per did per phone??
21:27.22nny_2jplank: checking now
21:27.27nny_2crosses?
21:27.39jplankyea, its def a carrier issue
21:28.04jplankkeep your loopback in, and tell them you have a "hard" loop directly into the smartjack and ask them to run patterns to it
21:28.05nny_2jplank: with loopback in same thing orange then red
21:28.09nny_2ok
21:28.21nhuisman_workkrokodilerian, I have an extension that I need 4 phones to share
21:28.23jplankassuming your plugged into the right circuit, they wont see their test patterns come back
21:28.31nny_2jplank: ds1 goes solid red when loopback is in
21:28.49nhuisman_workkrokodilerian, I already have it configured using usernames like 300_1 300_2 300_3 300_4 for the sip usernamaes
21:29.05nhuisman_workjust complaining ;P
21:29.11krokodilerian:)))
21:29.34krokodilerianok, i'm going to embarass myself in front of some people :)
21:29.43jplanki'll be there
21:30.28jplanknny_2: you def need to call your carrier, if your not greening up with a loopback, its not your issue
21:30.53jplankunless your loopback is bad
21:30.56nny_2jplank: alarm cleared on loopback after a minute or two
21:31.07jplankbut you confirmed its not since th * greened up
21:31.14jplankok
21:31.27nny_2jplank: on T1
21:31.39jplankbut alm is still red?
21:31.46nny_2jplank: now red is gone
21:31.50jplankok good
21:31.58nny_2jplank: alm is clear and ds1 is green
21:32.12jplanknow make a loopback jack and put it on the end of the cable your plugging into the *
21:32.16jplanksame thing should happen
21:32.58nny_2jplank: SWEET CHRIST
21:33.01nny_2jplank: it works now
21:33.05nny_2just magically started working
21:33.11nny_2on t1 hooked to asterisk
21:33.22jplankeither your carrier realized the crosses werent in
21:33.43jplankor it just took a while for it to sync up (i've seen circuits take 5-10 minutes to green up)
21:33.48jplankeither way good for you
21:33.50jeevhttp://rawr.pastebin.com/d39a3318f anyone know what i could do?
21:34.09jplanknow I'm going to sit in on kroko be back in a bit
21:35.52nny_2jplank: thanks again i owe you need your email addy
21:36.21jplankyou dont owe me anything, but if you need help gbove@nyigc.com
21:36.34nny_2jplank: whats crosses btw?
21:36.38nny_2jplank: thank you again
21:36.41jplankcross connects
21:36.49jplankdigital
21:36.55nny_2roger
21:36.58*** join/#asterisk Carlos_PHX (n=Carlos@63-255-103-7.ip.mcleodusa.net)
21:37.02citywokis it a new t1 nny?
21:37.22citywoki've not once had a t1 install go without a hitch. Qwest, Verizon, PLDT all screw them up, every single time
21:37.39jplankhey our turn ups go pretty good :)
21:37.58jplankmost of the time at least
21:37.58citywoklol our last one we were moving 1 of 4 T1's to another building
21:38.03citywokthey cross patched the wrong one in the middle
21:38.14jayteeit's always easier when the T1 is already in place and you're just swapping out what it connects to.
21:38.17citywokso they broke the other companies T1 cuz they cant read circuit ID's
21:38.31jplankits usually a lec issue not carrier
21:38.37citywokInternational T1's are difficult to deal with
21:38.38jplanklol
21:38.43jplanksounds about right
21:38.51jplankt1/e1 sucks
21:39.01nny_2ManxPower: jplank carrier says somehow it was our fault and tested for echo (another issue) with a tbird and said it cleared
21:39.18jplankof course they did
21:39.23jayteeI once had to deal with Nynex in New Hampshire to straighten out a mess when they put the demarc for the T1 in the laundromat next door.
21:39.27*** join/#asterisk outtolunc (i=outtolun@15-056.143.popsite.net)
21:39.31jplankyou used a magicloopback that fixed it
21:39.36citywokhaha jaytee thats common i think
21:39.58jplankthe lec NEVER installs the circuit in the right place
21:39.59denonjaytee: so how many million did they rack up in 900# terminations before you got it moved?
21:40.10citywokhow the hell do i bridge 2 channels together, or how do i execute a context on an active channel?
21:40.18denonor wasn't this a chinese laundry?
21:40.20jplankwe had verizon once place the circuit on the flooor then refuse to mount it
21:40.37citywokthe only thing i can find is bridge, but thats only in 1.6
21:40.51citywoki want to execute ZapBarge on an active channel, dont care how i have to do it lol
21:40.57ManxPowercitywok: dude, channels are AUTOMATICALLY bridgbed
21:41.03jayteedenon, it had never been turned up so $0.00 but I had them move it. Fixing Nynex's mistakes isn't my job, I just point them out and say, "Fix this or we get another carrier"
21:41.24citywokManxPower, i want to click a button, and have it change what channel i'm barged into
21:41.39ManxPowercitywok: Zapbarge/Zapscan does not bridge channels
21:41.48citywoki can zapbarge no problem in my dialplan, but only locally, cant remove exec it
21:41.57ManxPowercitywok: and I want a 21 yr old blond asian chick.  We can't always get what we want.
21:41.57citywokyea, bridge would be both way so you could talk to it, right?
21:42.21citywoki'll be in the philippines in 2 weeks, i'll get you one if you help me :-)
21:42.32jplankManxPower: come to astricon, theres a strip club not to far from the convention that has one :)
21:42.45ManxPowerThere are incoming calls and outgoing calls to/from Asterisk.  When a call comes into the dial plan and the dialplan executes Dial then the two channels are bridged.
21:43.04citywokis there any way to execute a command like zapbarge from the API?
21:43.18citywoktell channel 13 to barge in on channel 27
21:43.29citywokas far as i can tell in the api there is no way
21:43.39jayteeI want, I want!!! that's all they ever say! I want a button I can click on that makes 1000 people I never met in some countries I've never heard of write and mail me a valid check for $9999.99 everytime I click it.
21:43.40ManxPowercitywok: there are like 5 APIs in Asterisk.  Which do you want to use?
21:43.52ManxPowercitywok: why do you care what channel the barger is on?
21:44.25citywokManxPower, i want to be able to dial into asterisk on my desk phone, and then be able to click a button on a webpage that will figure out what channel i am on, and let me barge in on another channel
21:44.34jeevgives 5 to ManxPower and gives the finger to jaytee
21:44.48ManxPowercitywok: best of luck with that.  The rest of use the dialplan for that
21:44.49citywokright now i can click a button on a webpage, and have asterisk dial my phone, and barge me in on another call, but i dont want it to call me, i want to call it
21:44.53denonweb page figure out what channel your on?
21:44.58denonthat's a pretty loose coupling ..
21:45.28denonyou're going to have to associate location with zap channel *somewhere*
21:45.31denonor name or whatever
21:45.38citywoklol yea, callerid tells me what person called into asterisk
21:45.43citywokand i know who sits at what extension
21:45.46ManxPowerYou've been whining about this on the channel for like the last hour.  There's a reason nobody responds to you.
21:46.13Kattydo you think a bean bag would make a good doggy bed?
21:46.15citywokall i want to know how to do is execute a context, or a command, from the api
21:46.29ManxPowercitywok: STOP USING THE TERM API.
21:46.44denonpresumably you're talking about the manager interface
21:46.45denonso, say that
21:46.58jayteedepends on the size of the beanbag and the size of the dog. Usually, if the beanbag is bigger than the dog it'll work.
21:47.26ManxPowerdenon: I was thinking AGI until a few mins ago.  But maybe he wants to code it in the channel, which would be the C API (come to think of it, that's the only thing that might be called an API"
21:47.31citywokyes, the manager telnet in interface is what i've been using
21:48.02ManxPowercitywok: you pretty much wasted the entire time you've been asking on the channel because you used the wrong term.
21:48.08jeevjatee, msg
21:48.16outtoluncgrins
21:48.18ManxPowercitywok: the manager interface is called either "manager interface" or AMI
21:48.33citywokManxPower, i'm not a god, i dont know everything, and every once in a while we use the wrong term. it doesnt mean you need to treat us like idiots.
21:48.39jayteeflips jeev back the finger. "Right back at ya, buddy!" ;-)
21:48.49ManxPowercitywok: then you need to do more reading
21:48.50citywokthank you for explaining that to me
21:49.16citywokgood ol' RTFM, spent a lot of time donig that trying to find a way to do this
21:49.36citywokand since i couldn't find a way to do it online after a few hours of searching, i figured i would ask those that presumeably might have an answer
21:49.55ManxPowercitywok: manager.txt in the asterisk doc directory and voip-info.org should have many many AMI examples, just remember some of them will be out of date
21:49.56jayteeI carry my ATFOT copy more often than Buchanan carries a bible.
21:50.16citywokthe voip-info wiki has a lot of info, but almost all of it is out of date unfortunately, and much of it is barely explained
21:51.12ManxPowercitywok: Hint: ".call files", also an example in the asterisk doc dir.
21:51.26*** join/#asterisk sucituanbo (n=free@c-24-21-121-148.hsd1.wa.comcast.net)
21:51.28jayteewell, there's always Google! or if you prefer to use the command line there's http://goosh.org
21:51.40nny_2ManxPower: they brought a BCM nortel in to test for echo
21:51.53nny_2ManxPower: mind you this is a low latency system
21:52.00nny_2ManxPower: how freaking evil is that
21:52.04ManxPowernny_2: Um, the BCM has commercial EC built into it./
21:52.13jayteemakes gagging noises and starts dry heaving because someone used the N word with BCM after it.
21:52.14nny_2ManxPower: yeah
21:52.28ManxPowernny_2: you'll NEVER get the telco to admit to an echo problem
21:53.38jayteeRule #1, the problem is always at the customer premise equipment, never at our end of the circuit.
21:54.27jayteeRule #2, if the problem does appear after testing that it is at the CO end of the circuit, fix or stall according to need and then blame it on the CPE side anyways.
21:55.42jayteeRule #3, if standard business hours support contract exists, ensure that circuit testing is done after hours and bill customer for off hours support fees.
21:55.47ManxPowerjaytee: one of my customers went with a couple of BCMs (against my advice) and then went back to installing Asterisk at their offices
21:56.10[TK]D-Fenderrule #3, if we care about the customer('s business) enough and fear losing them simply claim it as a mystery
21:56.49jayteeManxPower, they've dumped them from their product line or are going to. Their new systems run linux with their version of Pingtel's sipXecs.
21:57.01*** join/#asterisk pyite (n=pyite@63-255-103-7.ip.mcleodusa.net)
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22:05.51*** join/#asterisk drumkilla (n=russell@asterisk/digium-open-source-team-lead/russellb)
22:05.51*** mode/#asterisk [+o drumkilla] by ChanServ
22:07.16*** join/#asterisk gaetronik (n=gaetan@186.9.14.229)
22:07.38km-anyone watching dayton's presentation at astricon?
22:07.43km-http://www.ustream.tv/channel/astricon2008
22:09.27denonis
22:09.31denon(just fired it up now)
22:14.36loompekyou farted?
22:14.43loompekoh.. sry...
22:14.46loompekfired .D
22:16.20*** part/#asterisk bradleyprice861 (n=bradleyp@fw.datafax.net)
22:16.49*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
22:22.22*** join/#asterisk SpeedDragon (n=SpeedDra@sm4-84-90-136-254.netvisao.pt)
22:22.57SpeedDragoni can ask questions here ?
22:23.15denonyes
22:23.22nosbigSpeedDragon, depends on the question....  ;-)
22:23.23SpeedDragonok
22:23.34SpeedDragonnosbig , is about asterisk :D
22:23.44nosbigAsterisk stuff, sure...  Meaning of life stuff, we might be less helpful...  ;-)
22:24.05SpeedDragoni have soft phones connection to my private server
22:24.17SpeedDragonit works inside and outside of network
22:24.22SpeedDragonworks realy great
22:24.47SpeedDragoni have account in voipbuster and netcall (netcall is a SIP Provider in Portugal)
22:25.09SpeedDragoni want make call throw both providers
22:25.14SpeedDragonand recive by one
22:25.26SpeedDragonbut i can't find where i need change to do that
22:25.34SpeedDragoni have asterisk-gui instaled
22:25.40*** join/#asterisk CunningPike (n=arodgers@64.251.77.9)
22:25.40SpeedDragonand version 1.4 of asterisk
22:26.24SpeedDragoncan anyone help ? :P
22:27.18[TK]D-FenderSpeedDragon: Look at the channel topic.  GUI's are supported in their own channels, not here
22:27.59SpeedDragonhum ... ok , but is not a relation gui based question, but i post question there
22:29.38AlexTOHi [TK]D-Fender, one quick question, i just install a 1.4 version and the user and group is root root insted of asterisk asterisk, my question is with this version uses root and root? because always after the installation  leave asterisk and asterisk
22:30.09[TK]D-FenderSpeedDragon: then where to change it is extensions.conf
22:30.31[TK]D-FenderAlexTO: no such thing as "version".
22:30.35SpeedDragoni dont have that file :X
22:30.48*** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net)
22:31.02[TK]D-FenderSpeed then your system is either really screwed up or you aren't looking in the right place.
22:31.23SpeedDragon/etc/asterisk ?
22:31.24baliktadsome precompiled distributions of asterisk (from debian and ubuntu, for example) automatically set up asterisk to run as user asterisk
22:31.34[TK]D-FenderSpeedDragon: That is where it normall should be
22:31.36baliktadbut if you compile from source asterisk runs as root by default
22:31.53[TK]D-Fender~asterisk-non-root
22:31.54jbot[~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115
22:31.57[TK]D-Fender^^^
22:31.57SpeedDragoni used debian
22:32.01SpeedDragonbut debian is 1.2
22:32.13SpeedDragoni download from website
22:32.14SpeedDragon1.4
22:32.14[TK]D-FenderSpeedDragon: then I suggest you use "find" to located your configs
22:32.27SpeedDragoni know where the configs are
22:32.29SpeedDragonsip.conf
22:32.34SpeedDragonrtp.conf
22:32.35SpeedDragonetc
22:32.41SpeedDragonbut i dont have extension.conf :X
22:32.44[TK]D-FenderSpeedDragon: If you hve not extensions.conf your system is crippled
22:32.53[TK]D-FenderSpeedDragon: Go rebuild your configs.
22:33.04AlexTOOk  thanks, my OS is debian
22:33.11[TK]D-FenderSpeedDragon: Actually there is the off-chance that EVERYTHING is build in extensions.ael
22:33.18[TK]D-FenderSpeedDragon: Do check in there
22:33.34[TK]D-FenderSpeedDragon: Barring all that go read the book as you're likely going to have to start from scratch
22:34.06jayteersync is your friend
22:34.17jayteeand so is cron
22:34.35[TK]D-Fendercp, tar, take your pick
22:35.30jaytee"gee, I've spent 37 hours getting all this stuff working. Wonder if I should make a copy or leave that till later."
22:36.07*** part/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net)
22:36.31*** join/#asterisk sacitec (n=tobi@201.144.211.82)
22:37.05boolean12What do you guys think about ISPCP?
22:37.54boolean12A little off topic, but I was curious.
22:38.02SpeedDragon[TK]D-Fender sorry, i are dumb ..
22:38.08SpeedDragoni have extension.conf file :P
22:38.42[TK]D-FenderSpeedDragon: then go read the book and look at what you're doing.
22:38.44sacitechello, I'm working with asterisk 1.4.19 and cisco call manager 6.1 with sip trunk. i have these problem with the calls http://bugs.digium.com/view.php?id=9546 . Any idea if 1.4.22 version fix this bug ?
22:38.48*** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net)
22:38.57*** join/#asterisk pyite (n=pyite@63-255-103-7.ip.mcleodusa.net)
22:39.29[TK]D-Fendersacitec: Go look at the dates.  It's more than clear
22:43.27*** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org)
22:45.51*** join/#asterisk l2cache (n=l2cache@adsl-75-21-128-203.dsl.rcfril.sbcglobal.net)
22:46.49l2cacheI have an issue with asterisk 1.2.24,  I go to do 'show application meetme' and it isn't installed, any way to patch it in there?
22:51.39[TK]D-Fenderl2cache: You need to have zaptel compiled prior to Asterisk
22:52.02[TK]D-Fenderl2cache: Go setup Zaptel and ztdummy, and then rebuild * from scratch
22:52.31sacitec[TK]D-Fender: on the version seems to be fixed, but i still have the issue, what could be the problem ?
22:53.05[TK]D-Fendersacitec: You say its so but you haven't shown us anything.  If you wish our help, provide the backup
23:01.24*** join/#asterisk jplank (n=GBove@240.sub-75-209-88.myvzw.com)
23:01.45jplankheh, I just met allison at astricon
23:02.00jplankvery weird talking to her
23:03.42baliktadweird
23:04.12*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
23:05.27EmleyMoorHow can I check for a caller ID beginning with a certain set of four digits and effectively replace them with 1? An unusual call came in today
23:06.08*** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-570d86b7c4963b77)
23:06.39EmleyMoorI want to change any caller ID that comes in as "0044XXXXXXXXXX" to 0XXXXXXXXXX
23:07.46EmleyMoorI think I may know but clarification would help
23:08.18l2cacheI did compile zaptel before asterisk
23:10.20*** join/#asterisk robba (n=robert@203.102.118.155)
23:11.38*** join/#asterisk lyroy (n=lyroy@bas1-montreal02-1096727613.dsl.bell.ca)
23:11.39*** join/#asterisk seanmh (n=seanmh@63-255-103-7.ip.mcleodusa.net)
23:13.21lyroyDoes someone here ever run asterisk inside a VMware Guest... i'm experiencing problems with choppy calls... it seems to be a timing issue with the ztdummy module... my question.. is it possible to run asterisk in a vmware environnement?
23:14.21*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
23:15.10StephenFlyroy: I've heard of it being done. Also I was experiencing choppy calls once ,and disabling ztdummy fixed that for me. Of course without ztdummy you can't use iax trunking or meetme
23:15.27*** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano)
23:15.32l2cacheI've always had horrible delay when using VMWare with *
23:15.48StephenF^ there you go, might just be a side effect of virtualization...
23:15.51drmessanohttp://bringvictory.com
23:17.27StephenFdrmessano: wow...
23:17.36lyroythats strange because I can run asterisk inside a Xen guest but not inside vmware wich is a mature vistualization architecture
23:18.54l2cachegod i love that song, thanks
23:19.55drmessanoyou're welcome
23:20.06drmessanoIt's one of the best. ever.
23:20.22l2cacheI just forwarded that to everyone I know
23:20.30EmleyMoorAm I right in thinking you can use the string slicers (;n:m) on the left hand side in a conditional?
23:21.56*** join/#asterisk CanWood (n=chatzill@24.108.64.80)
23:32.48sacitec[TK]D-Fender: here is my issue, when i make a call to cisco call manager via sip trunk, connects ok, but when cisco part put on hold and try to pick it up, the call is hanged on MOH on asterisk side(cisco side can recover the call). Here is the debug of that part of the call
23:32.49sacitechttp://pastebin.com/m28757a01
23:32.57*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
23:34.37[TK]D-FenderEmleyMoor: "core show application gotoif" <-
23:35.04[TK]D-Fendersacitec: Complete CLI SIP debug only.
23:35.29EmleyMoorDoesn't say I can't, so we'll see
23:37.38lyroyFor those of you that are experiencing some timing issue with asterisk in a vmware guest... here's the answer  http://www.pbxinaflash.com/vm/
23:37.41x86[TK]D-Fender: what's that 24- or 48-port rackmount SIP ATA you were talking about a while back?
23:37.53x86mediatrix?
23:37.59[TK]D-Fenderx86: AudioCodes MP-124 / Mediatrix 1124
23:39.08*** join/#asterisk rednode (n=admin@dehghany.demon.co.uk)
23:39.10rednodeHi
23:39.17rednodeWhat distro do you guys recommend for asterisk?
23:39.24rednodeand can asterisk be used on windows at all?
23:39.26l2cacheCentOS 4.5
23:39.38seanbrightminix
23:39.41rednodeis that what asterisk recommends or what you recommend?
23:39.46EmleyMoorrednode; I use Debian etch but it really is a matter of personal choice
23:39.46[TK]D-FenderReDWhichever you are most comfortable administering and can install *'s requirements on.
23:39.56seanbrightGNU/Hurd
23:39.57[TK]D-FenderReDNAnd FORGET about * under windows.
23:40.13[TK]D-Fenderkicks seanbright in the nads
23:40.16seanbright[TK]D-Fender: ... for now
23:40.34seanbrightonce i make my first million i am porting asterisk to windows
23:40.40rednodewell we are going to be rolling out asterisk to banks and hedge funds in the city
23:40.49rednodeso im not sure which distro to use to be honest
23:40.56rednodeis there a distro which asterisk actually recommend?
23:40.59seanbrightno
23:41.01seanbrightwell
23:41.02rednodeFedora? Suse? Red Hat?
23:41.02seanbrightno
23:41.10rednodelol thats abit weird...
23:41.11[TK]D-Fenderrednode: Even saying that brings into question your qualifications for a project that serious.
23:41.22seanbrightfor banks i would run openbsd
23:41.34seanbrightbut it's not trivial getting asterisk to install there, apparently.
23:41.37seanbright<-- never tried
23:41.42rednodeim not qualified at all for this project but unfortunatley im the most qualified person in the company, so most likeley ill hire a contractor
23:42.00seanbrightrednode: where are you?
23:42.02seanbright:)
23:42.11[TK]D-Fenderrednode: Then since you're doling that task out, ask them what they would use.
23:42.26[TK]D-Fenderrednode: Otherwise you're telling your contractor how to do their job.
23:42.39*** join/#asterisk jksM (i=jks@193.189.93.254)
23:42.45seanbrightoh, the UK
23:42.48seanbrightnevermind that.
23:42.54rednodei c ok, was just curious what asterisk recommends thats all
23:43.17seanbrightrednode: a lot of the asterisk devs run ubuntu
23:43.19seanbrightdebian
23:43.22*** join/#asterisk jeff_smoker (n=jeff_smo@ip70-162-238-155.ph.ph.cox.net)
23:43.25seanbrighti run centos
23:43.25*** join/#asterisk jks (i=jks@193.189.93.254)
23:43.38rednodei c ok thanks
23:44.07jeff_smokerDoes anyone here know a good resource on how to run a gsm phone to asterisk via usb?
23:44.17[TK]D-Fenderrednode: So far the vast majority of all-inclusive installs use CentOS which is RHEL effectively.
23:44.19seanbrightif you want to pay to fly me over and pay me obscene amounts of money to do the work, i would happily come
23:44.42[TK]D-Fenderrednode: that means its quirks are well known and support for those in similar positions is easy to find.
23:44.54seanbrightjeff_smoker: step 1, make sure you have a USB port
23:44.58seanbrightjeff_smoker: step 3, profit!
23:45.05[TK]D-Fenderrednode: That is perhaps the strongest reason to choose it IMO
23:45.29seanbrightwith centos 5 i have run into *0* quirks
23:45.35citywoki'm a debian man myself, never had any problems with it -- getting asterisk installed takes 5 minutes
23:45.37seanbrightit's pretty straight forward
23:45.43seanbrightdon't install from packages though
23:45.49[TK]D-Fenderjeff_smoker: go find some software that cares that you plugged in a phone at all.
23:45.49seanbrightreal men compile from source
23:45.57citywokgo use gentoo then :-D
23:46.05rednodeunfortunatley looksl ike we will be going with fedora or red-hat to be honest
23:46.15rednodeonly distros which I know that you can get commercial support
23:46.19rednodewell suse as well
23:46.35seanbrightcentos is effectively redhat
23:46.37l2cacheWhen running 'make config' in zaptel, I get the error "make: D: Command not found \r\n make: [config] Error 127 (ignored)"
23:46.45seanbrightheh
23:46.49l2cacheany insight
23:46.50lyroyyum install make
23:46.53rednodeCentOS is based on Red Hat?
23:46.55lyroyapt-get install make
23:46.59rednodeWho owns CentOS? Red Hat?
23:47.08seanbrightrednode: correct.  centos is FOSS.
23:47.12*** join/#asterisk krokodilerian (n=vasil@63-255-123-206.ip.mcleodusa.net)
23:47.15seanbrightrednode: owned by no one.
23:47.20l2cachelol, never had 'make' not on a system before
23:47.24rednodeFOSS?
23:47.25seanbrightrednode: check out centos.org
23:47.28l2cachehow did all the other make commands work???
23:47.31seanbrightrednode: google FOSS
23:47.32seanbright:)
23:47.53seanbrightl2cache: `which make`
23:48.10l2cachelibpri, zaptel, asterisk, addons   make, make install
23:48.14drmessanosudo make me a sandwich
23:48.29drmessanoseanbright: He answered you
23:48.33rednodefree and open source softwar :P
23:48.39seanbrightrednode: ding ding ding
23:48.44seanbrightdrmessano: did he?
23:48.55drmessano[19:48] <seanbright> l2cache: `which make` [19:48] <l2cache> libpri, zaptel, asterisk, addons   make, make install
23:49.09drmessanoYou asked which make, right?
23:49.15seanbrightah
23:49.21l2cacheeverything worked but the make config
23:49.32seanbrightyou should include a ;) or :) so i know when things you are saying are funny
23:49.49rednodeHas linux got a similar version to active directory, which is stable as active directory?
23:49.51drmessanoI refuse to use emoticons
23:50.03seanbrightrednode: samba does AD integration
23:50.07drmessanorednode: Not even close
23:50.10rednodeahh ok
23:50.27seanbrightrednode: requires voodoo and goat's blood... but you can make it work
23:50.33l2cacheI still get the 'make config' error I provided before
23:50.34rednodemeh someone needs to write something like AD for linux :P if theres any hope of linux becomning widely used in the commercial road :P
23:50.35rednodelol
23:50.36krokodilerianis the guy we were supposed to go tormenting the sangoma engineer around?:)
23:50.37drmessanoHe didnt ask if it would TALK to AD
23:50.42seanbrightl2cache: where did you get zaptel from?
23:50.47rednodethanks for your help guys, ill be pestering u more with questions later :P
23:50.52drmessanoHe asked if there was a linux drop in replacement
23:50.52l2cachegnudialer.org
23:50.54drmessanoand there is not
23:50.56rednodegoing to try and learn it myself first
23:51.02seanbrightl2cache: pastebin your *entire* zaptel Makefile
23:51.04seanbright~pb
23:51.04jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
23:51.04rednodego through the ebook which is on asterisk site etc
23:51.07rednodecheers again lads
23:51.11rednodeand ladies
23:51.35beekrednode: Linux has Fedora Directory Server
23:52.11rednodesimilar to AD or no where near?
23:52.45beekIt's a great LDAP server.   You can tie it to Kerberos.   AD is basically those two items with Windows proprietary hooks.
23:53.03beekFor a 'doze network it won't provide the full functionality but it may be enough for your purposes.
23:53.09sacitec[TK]D-Fender: sorry for the past debug, here it is the sip debug on cli http://pastebin.com/m70ce25fa
23:53.43beekrednode: http://directory.fedoraproject.org/
23:53.53rednodethanks bek will look @ it
23:54.02seanbrighthis name is beek
23:54.10seanbrighthave some respect for christ's sake
23:54.11seanbrightheh
23:54.37rednodesorry lol :P
23:54.45seanbrightl2cache: where's my pastebin, sucka?
23:54.46rednodecant type for shit its 00:54 here :( tired
23:54.49rednodegot work in 5 hours
23:54.54seanbrightrednode: go to sleep
23:55.03rednodecant rebuilding a server :(
23:55.07rednodefucking arsehole banks
23:55.08rednodelol
23:55.37beekrednode: Working on rebullding a server while tired.... now THERE's a recipe for disaster.
23:55.44seanbrightyeah
23:55.45rednodelool :P yep
23:55.57rednodecan asterisk be implemented with Avya, Nortel or Etrali switches??
23:56.08rednodei mean integrated :P
23:56.21Qwelldo they support sip?
23:56.23Qwellor pri?
23:56.40*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
23:56.40*** mode/#asterisk [+o mog] by ChanServ
23:56.49rednodethey support SIP as far as i knw
23:57.00seanbrightthen yes
23:57.07rednodeand cisco?
23:57.16rednodenot sure if cisco switch supports PRI im assuming it does
23:57.25seanbrightasterisk SIP stack is extensive...
23:57.28seanbrightheh
23:57.36Qwellcisco sip stack is expensive... :p
23:57.43l2cacheseanbright: http://pastebin.com/d5d0c969d
23:57.56seanbrightrednode: lots of the answers you seek are floating upon the interweb
23:58.07seanbrightrednode: voip-info.org for example
23:58.15rednodeyeh i know soz ill stop asking silly questions :P ill come back when iv fucked up the asterisk test install
23:58.16rednodelol
23:58.42seanbrightrednode: thats why [TK]D-Fender is here
23:58.57seanbrightrednode: ANY question at all... feel free to ask [TK]D-Fender.  day or night.
23:59.04seanbrightrednode: i'll PM you his personal e-mail address.
23:59.32rednodethanks!!!
23:59.35rednodeis he a contractor?
23:59.38seanbrighti was kidding
23:59.42seanbrightjust busting balls
23:59.44rednodelol

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