irclog2html for #asterisk on 20051108

00:00.30Dr-Linuxi'm using soft clients "sip" i wanna avail transfer call feature, is this feature comes with softphone or i need to do something in asterisk configuration?
00:04.16FuriousGeorgechanisavail else priority n+101
00:04.33FuriousGeorgethat what you mean?
00:08.42*** join/#asterisk nagl (n=nagl@213.235.241.6)
00:10.05*** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-11-9.w81-248.abo.wanadoo.fr)
00:12.17justinuDr-Linux: eyebeam supports transfer
00:12.28justinuDr-Linux: xlite doesn't, iirc
00:13.05file[laptop]marketing tactic cause it's free
00:13.33justinuthat's what I figured
00:13.46fugitivouse #
00:14.18*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
00:16.26harryvvanyone ever test asterisk in single board computers before?
00:16.44fugitivohm?
00:17.20file[laptop]just Google it.
00:17.53*** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com)
00:18.36*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
00:19.01harryvvno not just google it.
00:19.21file[laptop]then I'll give you a name, Kristian Kielhofner
00:19.22*** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-80.z142-153-67.customer.algx.net)
00:19.24harryvvasking for personal experaince is a bit different. and i really dont think its been done.
00:19.58file[laptop]it has been done.
00:20.21file[laptop]in fact, Google "asterisk single board computers" and look at the first thing that comes up :)
00:20.31file[laptop]it's an article on linuxdevices.com with Kristian aboot it
00:21.23hardwirecrazy crazy
00:21.25hardwirelittle file
00:21.32*** part/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com)
00:21.56hardwirethe gumstix scares me
00:22.01hardwirebut I think it would be cool for a phone platform
00:22.17file[laptop]hehe
00:22.22Vconexcom has ok stuff
00:22.50harryvvif your talking about the gumstix that was not what i was refering to :)
00:23.02hardwiredoing some filtering.. and creating a better percieved quality
00:23.10hardwiresample at 32khz
00:23.20file[laptop]harryvv: soekris boards are popular too
00:23.24harryvvthin client that fits in a 10baset wall box.
00:23.27harryvvhttp://www.windowsfordevices.com/news/NS3139003780.html
00:23.29hardwireharryvv: I use WRAP's
00:23.30harryvvod
00:23.32harryvvodd
00:23.33harryvv:)
00:23.36harryvvwraps?
00:23.59hardwireok
00:24.04Funaranyone want to see what happens to a TNT when it meets a 12-gauge up close and personal?
00:24.04hardwireI am in love with that device now
00:24.10hardwireharryvv: you suck
00:24.13hardwirenow I have to get one
00:24.15harryvvhehehe
00:24.19file[laptop]Funar: ...SURE!
00:24.24Funarlol
00:24.57harryvvkinda interesting..imagine the look on somones face if thay see peripheral connections in a wall outlet box.
00:24.59MRH2drumkilla patch on bug 5630 works great thanks
00:25.12Funari'm about ready to give up on this..
00:25.20drumkillaMRH2: cool, just note in the bug :)
00:26.27marcus2anyone else running asterisk on a wrt54gs-class device?
00:26.33mog_worki do
00:26.49file[laptop]I run Asterisk on my toaster.
00:27.08*** join/#asterisk cia-dabar (i=bosas@82.135.166.46)
00:27.13file[laptop]which just happens to be a highly fast undercooled machine, naturally
00:27.19mog_worki still havent gotten that crappy linksys sip device to work
00:27.23tzafrir_laptopfile[laptop], is your laptop your toaster?
00:27.33drumkillamog_work: nub.
00:27.39file[laptop]tzafrir_laptop: nope
00:27.50mog_workits not my fault drumkilla
00:27.57drumkilla:-p
00:28.08*** part/#asterisk cia-dabar (i=bosas@82.135.166.46)
00:28.25mog_worklinksys admit it didnt work as they say it did
00:28.42Vcosuch as "One of our Customer Service employees has already tryed to telephonically reach you. As our employee did not manage to reach you, this email has been sent to your notice.
00:28.43Vco"
00:28.47Vcoi mean....really now...
00:28.58file[laptop]telephonically?
00:29.01file[laptop]telepathically!
00:29.01marcus2i have a linksys pap2 working with asterisk
00:29.03marcus2it works great
00:29.17marcus2took me like 5 minutes to unlock the vonage crap, and another 2 minutes to set it up
00:29.29Funargood to know.. i ordered a PAP2-NA to test with.
00:29.37mog_workbut my pap2-na doesnt talk on the lan
00:29.38Vcoi just bought a PAP2 that wasnt' locked...
00:29.39mog_workonly on the wan
00:29.45mog_workthus it is pretty useless to me
00:29.48file[laptop]mog_work: that makes me sad
00:29.55marcus2huh?
00:29.56mog_worki know
00:30.03marcus2"doesn't talk on the lan" ?
00:30.08mog_workyou cant make it talk to a local address marcus2
00:30.14mog_workonly to a far end sip provider
00:30.22mog_workso i cant have it talk to my asterisk box at home
00:30.25Funarhmm.. wrong netmask?
00:30.27mog_workwhich makes me sad
00:30.34file[laptop]he's using the router...
00:30.35marcus2well, i will experiment with that tonight
00:30.38drumkillaso don't use it as a router
00:30.38mog_worki think its just hard locked to the wan port
00:30.39Funaroh, that one
00:30.42file[laptop]although you could make it loop back, but that would be interesting
00:30.42*** join/#asterisk JunK-Y (n=junky@69.156.171.57)
00:30.45mog_workwell then whats the point
00:30.47mog_worki have an iaxy
00:30.49marcus2since i'm setting up asterisk at home, finally
00:30.57marcus2uhm wait, you're using a pap2, or something else?
00:30.57drumkillamog_work: to have another device?
00:31.00file[laptop]JunK-Y: you two left me all alone in here! 'tsk 'tsk
00:31.29file[laptop]Star Academie!
00:31.49harryvvIs there a way for a phone to dial my extention if its picked up automaticly? Was thinking of putting a phone in the outside of the house where if its picked up it will ring the extentions inside.
00:31.50mog_workwhy do i need three phones for my little apt. drumkilla
00:32.02drumkillamog_work: fun!
00:32.10Vcosheer laziness?
00:32.16Vcolazyness
00:32.17test34asterisk need root by default ?
00:32.18Vcowhatever
00:32.25drumkillar0000t!
00:32.39marcus2no it doesn't
00:32.41Funarharryvv- i have a callbox at my office that does that.  you push a button on it and it dials a number.
00:32.47harryvvi have two seperate extentions so my wife does not pick up my calls
00:32.57harryvvfunar, where can i get one?
00:33.16file[laptop]Junky ScArEs me
00:34.02*** join/#asterisk danzig (n=nicolas@130.226.169.135)
00:34.03*** join/#asterisk bwzb (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net)
00:34.28fugitivohttp://www.nslu2-linux.org/wiki/Unslung/Asterisk
00:34.36bwzbhi, any one had issue on installing TDM400P card?
00:34.44bwzbI got following error:
00:34.46bwzbzttool
00:34.46bwzbUnable to open /dev/zap/ctl: No such file or directory
00:34.52Funari don't seem to have the bookmark here at home.  i found it on google tho.  try "outdoor call box" or something similar.
00:34.57fugitivobwzb: ls -la /dev/zap/ctl
00:35.09drumkillabwzb: /usr/src/zaptel/README.udev
00:35.23bwzbfugitivo: ls: /dev/zap/ctl: No such file or directory
00:35.29Funarvery simple device.. just a speakerphone with a set of dip switches to assign a phone number to dial.. single button on the font grill face.
00:35.31fugitivobwzb: that's your problem
00:36.12bwzbfugitivo: I have the same 2 cards installed in another machine which is working fine
00:36.31fugitivobwzb: did you create udev rules correctly?
00:36.33drumkillabwzb: README.udev
00:37.02bwzbdrumkilla: k, I read it now and see what's the issue.
00:37.10drumkillacool
00:37.54fugitivoanyone tried this? http://www.nslu2-linux.org/wiki/Unslung/Asterisk
00:40.49marc324how do you update the aliases table in SER  in realtime?
00:41.22Vcosee if i had a buncha money...it would be fun to get a bunch of these..  http://www.vikingelectronics.com/products/view_product.php?pid=79   and put them around town
00:41.36Vcojust put "DO NOT PUSH THIS BUTTON" on them...
00:41.38*** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com)
00:41.46Vcobut have it connect to a meetme if they do...
00:44.14MRH2off topic - anyone know a good noncisco, cisco router / IOS site  - (other than  http://www.routergod.com/agentsmith/ of course)
00:44.43maskedcould someone explain to me why my 'iax2 friend's registration is restricted to 60 seconds?
00:48.55maskedhow can i change this? Nov  8 11:49:44 NOTICE[3894]: chan_iax2.c:5650 update_registry: Restricting registration for peer 'masked' to 60 seconds (requested 300)   ???
00:52.22enemyanyone here run e1`s or t1`s even in conjunction with asterisk?
00:52.37mog_worklots of people enemy
00:52.39mog_workwhats up
00:53.28marc324ne1 knows ser here?
00:53.38enemymog_work: still having some issues managing to native bridge a call from one of the e1`s over to the other. Im thinking it could be some setup within the zaptel.conf or the zapata.conf. I`m running a dual e1 card with TE/NT.
00:54.26Peggeranyone have acess to cisco firmware?
00:55.15mog_workk what seems to be the problem
00:57.09enemyfirst of all, I get some errors, No D-channels available!  Using Primary on channel anyway 28! ... I guess this is a bad error?
00:57.39enemythe zttool shows no errors :(
00:58.53pooh_enemy: afaik those errors should not be a *real* problem
00:59.23pooh_enemy: find out what gear the connected telco is using
00:59.39enemymog_work: Im thinking about the clock/timing on the NT (which is provided by the dual E1 card) clock source should be taken from the TE. But Im not sure how to configure that.. maybe that will solve my issues
01:01.19bwzbdrumkilla: after reading README.udev, and rebuilding zaptel, I got this message after modprobe wctdm:
01:01.20bwzbFATAL: Error inserting wctdm (/lib/modules/2.6.9-5.0.3.EL/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg)
01:01.20bwzbFATAL: Error running install command for wctdm
01:01.37enemymog_work: also thinking about the framing, I`m getting some messages: wait_for_answer: Unable to forward frame and...
01:01.37enemyNov  8 02:00:40 WARNING[6501]: chan_zap.c:7570 zt_pri_error: PRI: !! Got reject for frame 23, but we have nothing -- resetting!
01:01.37enemyNov  8 02:00:40 WARNING[6501]: chan_zap.c:7570 zt_pri_error: PRI: !! Got S-frame while link down
01:03.16bwzbanyone know why I can not load "wctdm"?
01:04.26FuriousGeorgebwzb: you didnt make the modules?
01:04.36FuriousGeorgeor
01:04.39*** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com)
01:04.39drumkillabwzb: yeah, and that's in the regular README, i think :)
01:04.43FuriousGeorgeyou didnt read readmen.UDEV
01:04.46drumkillabut it's something that is not enabled in your kernel
01:04.50FuriousGeorgereadme*
01:05.02enemymog_work: also got some of theese: Got a UA, but i'm in state 1
01:05.11bwzbI thought I did after reading "README.udev", I rebuild the module
01:05.21drumkillabwzb: this is something else
01:05.21*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
01:05.41FuriousGeorgespeaking of wctdm:  does it take the boostringer option when loading?
01:06.06FuriousGeorgeive heard the wxfxs does but i use ectdm and it pulls in everything i need
01:06.29drumkillabwzb: you need CONFIG_CRC_CCITT enabled in your kernel
01:06.49nesyshi folks ... CCM -->* (multiple sip UA) --> ISP ... I need g729, that's pass-thru topology, or not?
01:06.49drumkillabwzb: under 'Library Routines -> CRC-CCIT functions' in make menuconfig
01:07.06bwzbdrumkilla: What do I do to enable CONFIG_CRC_CCITT in the kernel?
01:07.16drumkillayou'll have to rebuild a new kernel ...
01:07.19*** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au)
01:07.42bwzbdrumkilla: Godsh, do I need to rebuild a new kernel?
01:08.29drumkillayes.
01:10.03deezedi have a couple of toll free DIDs from nufone. how do I assign the number of the DID called to a variable
01:10.35*** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com)
01:11.49drumkilladeezed: it's already available in the variable ${EXTEN}
01:12.25deezedok.. thats what i thought. for some reason asterisk types out
01:12.31deezedtimes*
01:12.54deezedi guess i need to check spelling and syntax
01:13.04drumkillawell yeah, that's a different problem :)
01:14.24Qwelldrumkilla: any idea who I would talk to, to get a broken link on bugs.digium.com fixed?
01:14.30srtdrumkilla: would u mind to have a look a look at the patch submitted as bug 5571 (set global variables via Manager API) - its the equivalent to the GetVar patch you commited some time ago.
01:14.46drumkillaQwell: webmaster@digium.com
01:15.07Qwelldrumkilla: I emailed bugs@ - think it'll get to the right place?
01:15.16drumkillasrt: I'll put it on my list, but it might not happen before the release :(
01:15.40drumkillaQwell: bugs@digium.com?  I doubt that goes anywhere ...
01:15.49drumkillawebmaster will create an RT ticket for them
01:15.53drumkillaso email it there
01:15.53deezedodd.. syntax looks right, but asterisk times out -> exten => 8662942000,2,NOOP(EXTEN: ${EXTEN})
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01:16.13*** join/#asterisk file[laptop] (n=jcolp@69.156.171.57)
01:16.16srtdrumkilla: its very short and provides some consistency :)
01:16.19Qwelldrumkilla: k
01:16.21drumkilladeezed: do you have a priority 1 above that?
01:16.26drumkillasrt: I agree
01:16.28deezedyes Answer()
01:16.34Qwelleww, file[laptop] !
01:16.35file[laptop]french television is great
01:16.43deezedworks fine.. when I skip the noop
01:16.45file[laptop]it's so corny watching soap operas in another language
01:16.49drumkillasrt: I just have to write a speech for school first ...
01:17.08srtdrumkilla: hehe ok - that has priority :)
01:17.18file[laptop]no matter language, it's all the same
01:17.19bweschkedrumkilla: your not done yet?!? :)
01:17.20file[laptop]er no matter the language
01:17.23file[laptop]drumkilla: Junky says hi
01:17.26file[laptop]fine - hello
01:17.42drumkillabweschke: NO!  I can't find a good article!!!
01:17.48drumkillafile[laptop]: Russell says hi
01:17.51bweschkewhat's the topic?
01:18.15file[laptop]he said hi to drumkilla, not Russell :P
01:18.32file[laptop]uh oh it's back on!
01:18.47drumkillabweschke: it has to be an article from a technical magazine or journal about transistors or diodes, basically.  Oh, and published in the last few months ...
01:19.02drumkillabweschke: I'm finding plenty of press releases :)
01:19.10deezedthere isn't anything wrong with the syntax is there?
01:19.17file[laptop]drumkilla: how was your day?
01:19.23drumkillafile[laptop]: just dandy
01:19.38drumkillaI'm leaving my apartment now, so I can get this done, heh
01:19.40drumkillacya guys ...
01:19.44file[laptop]ttyl
01:20.31bweschkelater
01:21.09Dr-Linuxwhats diffenece between Dial(SIP/222,20,t) and Dial(SIP/222,20|t) ?
01:21.25Dr-Linuxi mean difference between , and | ?
01:21.33QwellDr-Linux: not a alot
01:21.35Qwella lot*
01:21.55Qwellmost (all?) options parsing stuff can use either
01:22.42Dr-LinuxQwell: it always confuse me when i read guides
01:22.48*** part/#asterisk docelm0 (n=docelmo@static-71-251-95-2.tampfl.fios.verizon.net)
01:23.21*** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com)
01:23.46_Sam--hi, how would i force asterisk to re-register with my IAX provider without resarting it?
01:24.54file[laptop]how creepy is it that despite not being able to speak french, I can apply past knowledge and common stuff to be able to get what's going on in this soap opera?
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01:33.42enemyafter some researching, I think I got my E1`s up running. But still, I can't dial out from my sip client using the e1 trunk. Using debugging, I now get >   M3: 0   P/F: 0 M2: 0 11: 3  [ UI (unnumbered information) ] ....
01:34.54delmari have a little question.... lets say you have contexts.. for example [local] and [local-restricted], they both have access to a context [local-free] but [local-restricted] doesn't have access to other contextx that cost money.... what happens if the dialplan of the [local-free] context has a timeout that jumps to say.. your chargable VoIP gateway next because the 1st path wasnt available... will the [local-restricted] context be denied access to that pa
01:35.15delmaror will ie be allowd because the timeout jumped over and allowed it to happen?
01:35.34delmaranyone know what im on about? lol
01:36.01file[laptop]since you used a goto, it doesn't matter if it's reachable in the context or not
01:36.11file[laptop]you're literally going to the context specified
01:36.31delmarfile[laptop] so the answer is.. GoTo overrides context includes.
01:36.43file[laptop]correct
01:36.49delmarok cheers
01:37.13file[laptop]if you use Goto without a context, it'll use the current context...
01:37.20file[laptop]but that's aboot it
01:37.24delmaryep
01:37.53delmari think i have a plan.
01:37.54delmar:P
01:39.42*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
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01:47.37mikemeenewbie q: my pots line into a TDM400P rings once (almost twice) before asterisk starts ringing the extensions. Can I make this more immediate?
01:48.22|Vulutre|mikemee: thats a phone company deal
01:49.52mikemeethx, I have a phone on the pots line as well as asterisk, and the phone rings. I turned callerid off in the zaptel config which dropped it from 2 rings to 1, but still one. Surely if the phone at my end rings, Asterisk can see that sooner?
01:49.59*** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net)
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02:11.20justinugoogletalk should allow you to dial sip uris
02:11.23harryvvmike, you cannot...because that first ring sends the callerid info then it will show up on the phone when it rings once.
02:11.45Math`justinu: googletalk doesnt use sip
02:12.34Math`(tho support is planned)
02:12.35mikemeeharryvv, thx!: I don't have/need callerid, so can I reduce the delay?
02:13.00justinumath: supposedly they do on the backend or something
02:13.12justinuif you look at the googletalk packets, you can see SDP
02:15.51harryvvmikemee, why?
02:15.54harryvvanyway
02:16.16harryvvanyone recomend a ups that will signal linux to shutdown if there is a power outage?
02:17.00mikemeeharryv: I don't pay SBC for callerid on my POTS line, so I'd rather avoid the extra rings when people call me
02:19.27Nivexharryvv: I swear by the APC line.
02:21.37*** join/#asterisk sumonish (n=God@203.12.249.168)
02:21.41sumonishhi all
02:21.48harryvvnivex okay what small unit do you recomend for me? might include it inside a protective cover along site the pbx
02:21.52*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
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02:22.50sumonishthe company i work for has just installed an Asterisk SVR for Voip work i wish to set it up so as a the phone rings just for reciption but can be remotly answered by anyone how can i do this?
02:22.59Nivexharryvv: that depends on how much power draw the PBX has, how long you want it to run in the event of a power outage, what your physical constraints are, etc.  Gonna have to do some research.
02:23.21shmaltzsumonish, look at app_pickup
02:23.30shmaltzsumonish, or you could try features.conf
02:23.33harryvvenough to allow it to power off the pbx fafely
02:23.36harryvvsafely
02:23.53javowhen i call one SIP phone from another, it says the phone i am calling is busy.  It is the same if I call the other way around.  I can call out just fine, but my phone will not ring.  it goes straight to voice mail.  I have restarted asterisk and the phone, with no luck.  does anyone know what a common cause of this would be?
02:23.59harryvvnivex im not concerned about keeping it up for x amount of time.
02:24.08*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
02:24.12sumonishthanks for the help Shmaltz
02:24.30Math`what I wanted to do is get a via embedded board (7W total power @ 1ghz) and run * on it with a ups.... it can stay up hours on battery
02:25.09harryvvmath nice idea
02:25.11Math`javo: check if the phone is registered
02:25.36Math`javo: sip show users
02:26.23javoit is
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02:28.53masked~mailinglist
02:28.54jboti guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
02:28.57javowhen someone calls, it tells them it is busy and sends them to voicemail
02:32.06harryvvhow can asterisk be setup to dial 1800 numbers into the states even though the number does not service this area of canada? My asterisk is seeing 1800 and dialing out zap. But want to go into my provider into the states
02:32.24harryvvhalf the 1800 calls are in the states and some dont work.
02:32.42Qwellharryvv: call dial, then check the DIALSTATUS variable?
02:32.50justinuset up a dial string for _1800
02:33.02harryvvI dont want to make a custom dial plan for each and every 1800 I stumble across and configure it. to bad there was a way to force asterik to dial iax
02:33.16harryvvjust...1800 calls here in canada!
02:33.19justinuyou can make it dial whatever service you want
02:33.34harryvvjustinu you dont understand.
02:33.38Qwellharryvv: What happens when you dial a US 800 over zap?
02:33.53harryvvit dails local 1800
02:33.56harryvvbut
02:33.56justinuyou want it to use a SIP provider for certain 800 numbers, right?
02:34.06harryvvsome us companies dont have 1800 service up here.
02:34.11harryvvyes
02:34.11Qwellhmm
02:34.14harryvvinto canada.
02:34.15harryvverr
02:34.19harryvvinto the states
02:34.25Qwellharryvv: I bet you could write a little agi to hit like infousa or something
02:34.32harryvvi dont know agi
02:34.37justinuso you want to route all toll frees that terminate into the USA out via a specific SIP provider?
02:34.38Qwellshouldn't be hard
02:34.46harryvvnooooo
02:34.48justinuand the others out to your local LEC
02:35.30harryvvsorry yes, All toll free 1800 numbers those companies that dont have there service up here.
02:35.51Qwellso, wait...
02:35.56justinulol
02:36.01Qwellcan the same number exist in both canada and us?
02:36.06justinuyeah
02:36.08Qwellif so, that'd be fucking retarded
02:36.13justinuwe have toll frees that work in canada and the USA
02:36.30Qwellokay, let me rephrase
02:36.32justinubut they always terminate into the USA
02:36.38Qwellcan two DIFFERENT companies have the same 800 number?
02:36.42Qwellone in canada, one in the US?
02:36.44justinuoh
02:36.47harryvvqwell, that depends on what the company wants. I often want to call 1800 to a company in the states but it dials out zap. If i found a way to dial out sip then it would work.
02:36.48justinuno, it's still NANP
02:36.56justinupart of the same number plan
02:37.18jake1932just prompt for a code to dial into SIP
02:37.42harryvvhehe
02:37.49Qwellnah, it should be automatic
02:37.56harryvvyou know, i forgot about my calling card feature i programed into this.
02:37.56jake1932or call both using &
02:38.03harryvvthat would work then.
02:38.08Qwelljake1932: SIP will probably always answer
02:38.12justinuthe zyxel p2000 wifi phone is stupid
02:38.16justinuno dns srv support
02:38.20harryvvi dial my extention put in pass then it should work.
02:38.59jake1932and get a SIP provider that won't work in Canada :)
02:39.12justinuyou need a password to dial toll frees? wow :)
02:39.24justinuoh, nm
02:39.29justinudidn't see that about calling cards
02:39.36harryvvsuck..it did not work
02:40.18harryvvjust, well i created this if I am at a goverment office and want to call into the states.
02:40.41harryvvi call my number then enter in extention and pass and its charged to my account. Like that of a calling card
02:40.56justinucool
02:41.23harryvvso call home on cellular, call into states at 2 cents per min plus local cell rates.
02:41.35Qwellthere is probably some simple site that can tell you if a tollfree number is in canada...  call that from an agi, if it's in canada, dial out zap, otherwise, dial out sip
02:41.46justinuget a flat rate US sip account :)
02:42.08*** join/#asterisk docelm0 (n=docelmo@static-71-251-95-2.tampfl.fios.verizon.net)
02:42.17justinuqwell: i think that would be an SMS/800 db query
02:42.35harryvvbrb
02:42.39justinuif there's a site that'll tell you that, i'd love to see it
02:42.41Qwellspeaking of which...
02:42.45Qwellwhere did file go?
02:42.54jake1932great question!
02:43.34jake1932it's too early for bed
02:43.37Qwelldid somebody accidently rm him again?
02:43.43Qwells/accidently//
02:43.47jake1932hehe
02:44.40jake1932qwell - do you work for asterlink?
02:44.40*** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net)
02:44.44Qwellno
02:44.55jake1932but you use them, right?
02:45.00QwellI do
02:45.17jake1932you seeig weirdness with toll-free outbound?
02:45.20jake1932seeing
02:45.43Qwelldunno, lets see
02:46.25Qwellodd, seems they don't like the codec I'm using...or something
02:46.52Qwellwoah, heh
02:46.58Qwellyeah, weirdness indeed
02:47.01jake1932:)
02:47.09Qwell"If you remain on the line, you will be charged for the international portion of this call."
02:47.13jake1932right
02:50.06justinunice!
02:51.41justinuis there any way to do old fashioned call screening with a sip phone?
02:51.54Qwelllook at the callerid, and don't answer if you don't want to talk?
02:52.00justinuhave someone leave a message, but allow you to answer the phone(s) while you hear the ICM
02:52.12justinulike an analog answering machine
02:52.32Qwellmight be able to something creative with monitor or something
02:52.41justinuhmm
02:53.02Qwellhave it fork the call to an autoanswer line on the sip phone, and mute the channel or something?  dunno
02:53.14justinumaybe drop both phones into a meetme
02:53.20Qwellyeah
02:53.47Qwellor you could write app_callscreening, heh
02:53.58justinubut is there any phone that doesn't actually answer the call with a 200 ok, but still will play the audio on the speaker? like early media works with 183 progress
02:54.09Qwellgot me
02:54.19justinucuz i'd like an answer on ANY of the phones ringing to interrupt the message and bridge the call
02:54.33justinubut I guess all the phones would still be ringing
02:54.51justinucan't replicate the old analog scenario very well with sip
02:56.11*** part/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
03:02.35Peggeranyone have acess to cisco firmware?
03:02.50QwellPegger: people who pay for the access do
03:03.12Qwell(that's my subtle way of saying to pay cisco what you owe them, if you want the firmware)
03:03.31PeggerQwell well it only makes sense that if I buy a 200 dollar phone that I should have acess to the phones firmware
03:03.42QwellPegger: you'd think so, but no
03:03.51PeggerQwell how do i owe them?
03:04.09*** join/#asterisk copantl (n=galel@63.245.93.138)
03:04.17copantlhello
03:04.26QwellPegger: because you want the firmware
03:04.26Peggeryo
03:04.50copantli have a problem with a sip peer conexion
03:04.58copantlcan some one help me?P
03:05.24*** join/#asterisk SplasPood (n=sp@paravolve.net)
03:05.35Peggerwhat ever just keep on thinking cisco twisted tatics are cool
03:05.46QwellPegger: Who said I liked what they do?
03:05.53copantli need to route a calls from a peer to my asterisk
03:06.03Qwellfact of the matter is, stealing the firmware would be copyright infringement, plain and simple.
03:06.35copantlbut when i place a call , goes to a extension, not to a DID... any Idea?
03:06.37justinui dunno if he thinks it's cool
03:06.41justinuhe's just saying it's the law
03:06.43Peggerbut it is included withthe phone
03:06.49Peggeram i stealing that
03:06.54justinuthe version that was on it when you bought it was included
03:06.56justinunot the latest
03:07.11copantlsome one from digium ?
03:08.00Peggerso has anyone set up a phone with sccp before?
03:08.09QwellPegger: yep, sccp is fun
03:08.58justinuseems kinda cool
03:09.08justinui like that event driven model
03:09.16harryvvqwell get that link?
03:09.18[TK]D-Fendercopantl : pastebin you extensions.conf and I'll take a look
03:09.20Qwellyep, kinda cool
03:09.35harryvvqwell, that would make a good wall mount pbx style of case.
03:09.41Qwellyeah...
03:09.43Peggerso on the topic of sccp how does it compare to sip when connecting to *
03:09.52QwellPegger: personally, I'm really digging sccp
03:10.04Qwellif not only because of system messages, heh
03:10.10justinuwhat's that?
03:10.25Qwellsccp system message "The boss is gone.  Time to party!" 0
03:10.26PeggerQwell so what kind of feature diffrences are there
03:10.29copantli just wanna configure  a media gateway
03:10.32Qwellsends the shit to all the sccp phones on the system
03:10.35justinuheh
03:10.42justinupolycoms have IM
03:10.46justinuso i guess you could do that
03:10.54Qwellbut...that 0...was a timeout
03:11.03Qwellso, I can make a message display for only 30 seconds
03:11.07justinuah
03:11.14justinuthat's kinda neat
03:11.18QwellI'm in the process of hacking it up to be per device
03:11.26Qwellthat'll be slick...
03:11.27[TK]D-FenderCisco = overpriced Polycom class :)
03:11.43[TK]D-FenderPoltcom <3
03:11.52[TK]D-Fenderpolycom even ;)
03:11.55PeggerQwell   what other neat/usefull features can it do? how well does it handle nat
03:12.01QwellPegger: got me...
03:12.12copantlJoe?
03:12.12Qwellit's rtp, so I imagine it isn't difficult
03:12.19rikstanah cisco are the rolls royce man
03:12.28PeggerQwell so message display is the only benifit of sccp
03:12.34justinulol
03:12.34QwellPegger: no, certainly not
03:12.46justinuno, sccp sends events for onhook offhook, dtmf presses, etc.
03:12.53justinuit's a way less complex protocol
03:12.55Peggerthen what else might there be
03:12.56Qwellyeah, everything is an event
03:13.07justinusip is all crazy and tries to do way too much
03:13.15copantl[TK]D-Fender: i just wanna setup my asterisk as a media gateway
03:13.36Peggerwell i might have to leave sccp on the cisco phone that I oredred last week, hopefully it arrives soon
03:13.59*** join/#asterisk file[laptop] (n=jcolp@69.156.171.57)
03:14.09QwellPegger: or you could get a cco license (which still wouldn't give you a legal right to use it...but, you'll be able to download it)
03:14.15Qwellfile[laptop]: that really you?
03:14.37file[laptop]yup
03:14.37Peggerqwell how much are cco licenses?
03:14.41[TK]D-Fendercopantl : what do you want on both ends?
03:14.49justinulike 7 bucks per phone
03:15.05Peggeryou have to pay per phone?
03:15.05copantl[TK]D-Fender: asterisk
03:15.09Qwellfile[laptop]: remember that script you asked me for a week or so ago?  cidname lookup
03:15.11file[laptop]oh noes it's [TK]D-Fender
03:15.21[TK]D-Fendercopantl : as in Asterisk <-> Asterisk?
03:15.27Peggerwell i oonly have one phone so whatever
03:15.30justinui dunno, i just heard that in passing
03:15.33justinui don't own ciscos
03:15.39justinuor work with them
03:15.45linageejustinu: what's this now? AT&T owns cisco?
03:15.57justinui have polycom, snom, sipura, grandstream, zyxel
03:16.02copantlis all the calls from asterisk1 routes to a did from asterisk2
03:16.03justinulinagee: what are you on about now!?
03:16.15linageejustinu: oh, i thought you were starting conspiracy theories. :-)
03:16.18justinulol
03:17.10[TK]D-Fendercopantl : oh.. sorry, never did * bridging before.
03:17.24copantlok
03:17.28copantlsome one?
03:18.09FuriousGeorgecopantl: didnt ake sense to me
03:18.11Peggeranyone here do asterisk failover, like one asterisk dies for some reason so a second asterisk box takes over
03:18.18FuriousGeorge\ake=make
03:18.24blitz[laptop]not yet
03:18.25copantlmake sanse?
03:18.32QwellPegger: there is something on the wiki about that
03:18.34FuriousGeorgesense
03:18.40blitz[laptop]Pegger: there was a thread about it recently on asterisk-users
03:18.44copantli just wanna create a media gateway
03:18.51justinublitz: link?
03:18.57linageePegger: why would you want something like that? nobody likes it better than when your asterisk box barfs up and nobody can make a phone call!
03:18.58blitz[laptop]justinu: no idea :)
03:19.04justinubastage :P
03:19.12FuriousGeorgecopantl: i dunno
03:19.21blitz[laptop]justinu: basically I think people used a "heartbeat" type of method...
03:19.23linageehigh availability systems save you from hardware problems, not software problems. ;-)
03:19.48copantlasterisk1-----------sip peer--asterisk2-did------>
03:19.50blitz[laptop]if a server dies... all calls are going to die on it either way
03:19.55Peggerwhy cant they save you from software problems
03:20.02justinuibm was doing work to solve that
03:20.18justinuthey did a talk at astricon
03:20.19linageePegger: well i suppose you could fail over to an older version, but i've never heard of that being done.
03:20.25Peggerblitz[laptop] no  I do not see why it is not possible to set up a heat beat for the two servers
03:20.34*** join/#asterisk jontow (i=jontow@bsd.adminforrent.com)
03:20.38blitz[laptop]Pegger: I never said you couldn't
03:20.42linageePegger: it would be a MAJOR annoyance to have two versions of the config
03:20.53justinusync them with rsync
03:20.57Qwellactive calls will be distrupted though
03:20.59linageejustinu: naw.
03:21.05linageejustinu: just use a SAN. ;-)
03:21.10Qwellregardless of whether you use a heartbeat type deal or not
03:21.12justinui'm too poor for a san
03:21.13blitz[laptop]Qwell: yep -- thats what I said, but no one ever pays attention to me :)
03:21.13justinu:(
03:21.15linageejustinu: put an EMC box there, go HA the full 9 yards. ;-)
03:21.22Qwellblitz[laptop]: was helping reiterate the point, heh
03:21.30blitz[laptop]box dies, all active channels on that box dies too -- regardless
03:21.32blitz[laptop]Qwell: hehehe :)
03:21.51blitz[laptop]Qwell: I reiterated your iterated point
03:21.51linageeblitz[laptop]: not necessarily.... hrm
03:22.02blitz[laptop]linagee: let me know when you figure it out
03:22.09Qwellblitz[laptop]: and by doing that, it helped clear everything up...
03:22.13blitz[laptop]Qwell: :)
03:22.14justinuthey'll only die if the box was proxying rtp
03:22.15linageeblitz[laptop]: if you were able to switch over fast enough, and you had the TCP/IP stack intergrated into the HA system...
03:22.16blitz[laptop]Qwell: apparently not :)
03:22.22justinuif you reinvite to the pstn gateway, that's not a problem
03:22.25Qwellreiterating the reiterated point of the previous iteration
03:22.30linageeblitz[laptop]: switch over faster than 2 minutes (TCP timeout) or xx minutes (application timeout)
03:22.31justinuprovided the pstn gateway was not the box that just went down
03:22.47blitz[laptop]linagee: I think the caller would hang up before 2 minutes
03:22.52linageethat's it. fail over to a PSTN line. lo.
03:22.56Qwelllinagee: and your dialplan would crap itself
03:22.59blitz[laptop]linagee: and we don't use TCP in Asterisk
03:23.01linageeblitz[laptop]: lol. like i said, application timeout
03:23.10Qwellthere would be literally no way of knowing where each call was in the dialplan.
03:23.15linageeblitz[laptop]: er, UDP... so it's all application timeout. hehehe
03:23.16blitz[laptop]linagee: tcp is bad for realtime applications
03:23.19QwellThey went over some of this stuff at astricon...
03:23.23linageeblitz[laptop]: exactly.
03:23.27justinutcp is anceint
03:23.34blitz[laptop]justinu: uhhh...
03:23.48linageejustinu: will TCP die with IPv6?
03:23.54Qwellno...
03:24.04blitz[laptop]TCP is layer 4 -- IP is layer 3
03:24.14blitz[laptop]IP encapsulates TCP
03:24.19Math`tho there is a tcp6 afaik
03:24.24justinuyes
03:24.24linageeblitz[laptop]: yes, but maybe they have solved it with IPv6? i hear there is encryption. that is not layer 3....
03:24.36Qwellencryption?
03:24.42linageeQwell: or VPN or something
03:24.50linageeQwell: some new routing method in ipv6
03:25.00linageeQwell: unless i was speaking to a complete liar! (possible)
03:25.10linageeor was that on the internet...
03:25.22linageeipsec? i think that was it.
03:25.26justinulol
03:25.28Qwellthat'd probably be a layer 4+ thing, heh
03:25.28linageei think ipv6 has ipsec built in?
03:25.31Math`ipsec exists on v4 too
03:25.45blitz[laptop]we should just start using SCTP
03:25.49linageeMath`: yes, but only if you have it.
03:26.05linageeMath`: i think ipsec might be REQUIRED for ipv6....
03:26.14justinuSTCP is a good idea
03:26.17Math`blitz[laptop]: it is not
03:26.21linagee(required as in, you're breaking the standard by not having it)
03:26.23*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
03:26.34blitz[laptop]Math`: what is not?
03:26.44Peggerlinagee not required
03:27.09linageePegger: moment...
03:27.20Math`ipsec aint required for v6
03:27.27rikstait certainly is not
03:27.50blitz[laptop]Math`: I never said it was
03:27.57blitz[laptop]Math`: notice I laughed when linagee said that
03:28.10linageeblitz[laptop]: it says so on an ieee page... argh
03:28.11Math`oh sorry it was linagee
03:28.15linagee"buy this article now"
03:28.20justinulol
03:28.29blitz[laptop]I love the IEEE
03:28.39blitz[laptop]</sarcasm>
03:29.04Math`I like people who establish standards... make people pay for the spec, then complain their standard isnt repected
03:29.06rikstai love PIE
03:29.07linagee"It is also expected that extended features of IPv6 will be used when IPv6 gets commonly used. Therefore, Phase-2 will test extended features of IPv6 such as IPsec and MIPv6, in addition to IPv6 core protocol.
03:29.07linagee"
03:29.08Math`respected, that is
03:29.14linageeok, everyone line up for punches
03:29.20linageeor you can pay the fee
03:29.38rikstalinagee: that doesn't mean it;s required
03:29.41linageeyou're all fools. see above
03:29.44copantli just need a asterisk like a pstn gateway for other asterisk? ..... can be posible?
03:29.46blitz[laptop]screw IPv6, I'm starting my own layer 3 protocol
03:29.55linageeriksta: required as in you're breaking the standard if not. see above where i said that right after.
03:29.59rikstawe're hardly even in phase 1 never mind pahse 2
03:30.03justinulinagee: bahahaaha
03:30.06linageeriksta: it might even be compiled into the stack. not sure.
03:30.49blitz[laptop]riksta: I have a feeling even if all of asia adopts IPv6, and even Europe, North America will continue to use IPv4 for a while, simply using IPv4<-->IPv6 gateways
03:31.02rikstayep
03:31.09linageeit will be nice though once it's required.
03:31.09linageethis means an encrypted connection to anyone using IPv6
03:31.11linagee(soon everyone? lol)
03:31.16rikstathere's gonna be mega bux in those gateways
03:31.23justinulol
03:31.49linageeof course, do you think required ipsec in internet communications will mean less big brother? ROFL
03:31.55linagee"sense of security" ;-)
03:32.16rikstaare you blabbering nonsense, or am I just tired
03:32.23justinuno, he's always like that
03:32.31justinubut he eventually makes sense
03:32.45linageejustinu: exactly. as soon as i found that article sniplet
03:32.57*** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com)
03:33.04justinudinner time
03:33.07linageeyes!
03:33.11linageeindeed! taco bell!
03:33.14linageewho's buying? :-)
03:33.41rikstaive never had a taco bell and i dont think i really ever want one
03:34.01linageeriksta: lol. probably not.
03:34.10rikstawhat is in them
03:34.22linageeriksta: fast food mexican.
03:34.22rikstai know it's a chain...but what stuff do they do
03:34.29rikstafajitas n shit?
03:34.39linageenot really. tacos n shit. (TACO bell. ;-) )
03:34.45rikstayeah i gathered that
03:34.53rikstasounds unhealthy :P
03:34.57linageeand burritos too. although if you want a good burrito there are other places.
03:35.03linageeriksta: very! :o)
03:35.29linageeriksta: or i should say, isn't all fast food? heheh
03:35.33rikstatrue
03:35.43rikstayou yanks :P
03:36.02*** join/#asterisk wolfson (n=ggggg@usr-kdh-208-6-58-26.beachlink.com)
03:40.21*** join/#asterisk Brijn (n=bas@S0106004063c0fa1f.vn.shawcable.net)
03:40.37BrijnGood evening!
03:41.41BrijnAfter three weeks of telling my SIP provider they have a problem, they finally sent me their part of the config. Took me a minute to discover a type in their config :-(((
03:42.25BrijnNow I at least see the call coming my way :) But unfortunately the problem is now on my end... i think at least
03:43.41*** join/#asterisk bsdz0r (n=bsd@24-247-92-53.dhcp.bycy.mi.charter.com)
03:45.06BrijnI see some stuff with sip debug enabled, but I don't understand enough of it to pinpoint the problem.. If someone could have a look.... http://pastebin.ca/27948
03:45.10iCEBrkrmd
03:45.17iCEBrkrgrr
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03:45.50linageemagic
03:45.56linageeasterisk is magic
03:46.22*** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net)
03:46.24implicithi
03:50.45harryvvI wonder how tough it would be to make a embeded interface work with asterisk
03:50.56BrijnSo close but so far, wtf is happening :(
03:51.04*** join/#asterisk unabonger (i=Bhongwan@lunacity.themoon.org)
03:51.13unabongerhello
03:51.47unabongeranybody in here able to tell me why my music on hold works when I call an extension from one source, but not another?
03:51.53*** join/#asterisk bmg505 (n=leon@rndf-146-59-144.telkomadsl.co.za)
03:54.04harryvvunabonger, do a cli on it.
03:54.08Brijnunabonger: not to many of the wizards around i'm afraid :) You get audio if you accept the call
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03:57.10unabongerjachin*CLI>
03:57.10unabongerVerbosity is at least 5
03:57.10unabongerNov  7 20:05:39 NOTICE[426]: chan_sip.c:7787 handle_request: Registration from 'Zhonka <sip:zhonka@kush.inwa.net>' failed for '66.228.198.186'
03:57.10unabonger<PROTECTED>
03:57.10unabonger<PROTECTED>
03:57.12unabonger<PROTECTED>
03:57.14unabonger<PROTECTED>
03:57.16unabonger<PROTECTED>
03:57.18unabonger<PROTECTED>
03:57.20unabonger<PROTECTED>
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03:57.24unabonger<PROTECTED>
03:57.25harryvvdork
03:57.26unabonger<PROTECTED>
03:57.27Math`*pastebin*
03:57.28unabongerNov  7 20:06:11 WARNING[426]: chan_sip.c:697 retrans_pkt: Maximum retries exceeded on call 473ab2205e8477684abb7d525b20b8d2@x.228.19x.77 for seqno 12360 (Non-critical Response)
03:57.29*** kick/#asterisk [unabonger!n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted] by twisted (flood detected)
03:57.35harryvvuse pastbin.ca
03:57.46Math`lets tell him when he joins back
03:57.47kr5i am having some issues with dtmf
03:57.52*** join/#asterisk unabonger (i=Bhongwan@lunacity.themoon.org)
03:57.56Math`unabonger: www.pastebin.ca
03:58.00unabongerdumb ass bots
03:58.08unabonger8 lines is NOT a flood, jeebus help us.
03:58.20harryvvmmm i would say ask about doing that before flooding the channel
03:58.31Math`thats 14 lines, not 8
03:58.33harryvvgoto pastbin.ca and past it there.
03:58.37unabongerrepeat "8 lines is NOT a flood"
03:58.38harryvvthats the rule here
03:58.43unabongeranal retentive bots
03:58.44rikstayou pasted about 14 lines, and 8 is considered a flood to me
03:58.58harryvveveryone here past to pastbin.ca
03:59.01unabongerperhaps you have a form of digital ADD
03:59.08unabongeryes, I know about pastebin,
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03:59.12unabongerhold on.
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04:00.36BrijnAnyone willing to have a look at a sip debug trace.. It must be something simple if you are used to these traces :( http://pastebin.ca/27948
04:00.41kr5what else would i need to set other than the dtmf mode in sip.conf and in the config for sjphone?  Both are inband, but dtmf still refuses to work.
04:00.56justinuBrijn: what's the problem
04:03.18justinuyour auth username looks wrong to me
04:03.19unabongerhttp://pastebin.ca/27950
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04:03.35Qwellunabonger: be glad it didn't include a +b
04:03.53unabongerhumbug
04:04.29Brijnjustinu: I got the config from the provider.. And their secret matches mine.. But it seems that my server goes back to their instead of sending the call to the extention (300)
04:04.31unabongeranyway, it just seems weird to me that the MOH doesn't work when calling from one source.
04:05.00Math`unabonger: check for voice activity detection on the phone
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04:05.55file[laptop]Brijn: want an easy way out?
04:06.35BrijnWell, as long as I understand it :)
04:07.03file[laptop]Brijn: put insecure=very in your lightspeed-out entry... do a sip reload, then try it...
04:07.06unabongerMath`   It's the same phone?  Do you mean "suppress silence"?
04:07.47unabongeri'm using X-Ten Lite for my test, is the softphone my problem.  I can try with hardware VOIP phone too. . . .
04:07.48Math`yeah, should be set to NO
04:07.52*** part/#asterisk [bsd] (n=bsd@203.134.194.11)
04:08.00Math`transmit silence should be set to YES in x-lite
04:08.16unabongeralready done
04:08.21unabongerhmmm
04:09.34unabongerlike I said, I'm calling the same phone extensions (one through a SIP provider, and one through an IAX provider), and when the SIP provider calls come in, I can't hear the MOH when I put myself on hold.
04:09.40justinuBrijn: i'm not sure what's going on... been looking at sip too long today, just can't concentrate
04:09.47Brijnfile[laptop]: Unfortunately that didn't fix it.. Let me compare the SIP output
04:09.53file[laptop]their way for calling you is weird
04:10.08Math`unabonger: that is kinda weird
04:10.30file[laptop]cause you won't get which phone number was called... but meh
04:10.45unabongermaybe the SIP provider who forwards my calls is blocking MOH ?
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04:10.56Brijnjustinu: line 60-62, does that tell my server to talk to the mobile phone (the nr in there) thru the lightspeed server
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04:11.07Qwelloh file[laptop]...
04:11.12file[laptop]hi Qwell
04:11.29Qwellremember that cidname lookup script we talked about a week ago or so?
04:11.35file[laptop]sure
04:11.39Qwellstill need it?
04:11.47file[laptop]meh it's not overly urgent, so no
04:11.52Qwellheh
04:12.19Peggeron cisco phones can you load the firmware so that it is saved to flash memory or do you have to reload the firmware every time the phone boots???
04:12.35QwellPegger: it doesn't need to load it every time
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04:12.46PeggerQwell oha good
04:12.46unabongeryou only have to reload the firmware once
04:13.21Peggerit is just most of the docs talk about setting upa tftp server in order to get sccp working with asterisk
04:13.37QwellYou do need a tftp server, to pull the config from
04:13.37unabongerer, once for every version you have to ugrade, which may be a chain of them, to get to the latest version. . . .
04:13.40justinuyou need tftp
04:13.49Brijnfile[laptop]: I tried with and without the ext added to the register statement. This is without any ext added http://pastebin.ca/27951
04:14.32PeggerQwell so you can just enter the server info from he touch pad?
04:14.36BrijnIt seems so close to actually making the phone ring :)
04:14.47Qwellyou probably could, but...
04:14.59Qwellit'd be extremely freaking tedious
04:15.14PeggerQwell  why you would have to do it every time?
04:15.29Qwellno, but you'd have to do it for every phone
04:15.52rajivwhere can i get the patch to fix http://www.assurance.com.au/advisories/200511-asterisk.txt ?
04:16.06PeggerQwell oha well I will only have one phone so not a big deal, so i guess you really only need the tftp for bigger instilaitons
04:16.39Qwellrajiv: just don't use the vmail.cgi script
04:17.09rajivQwell: i dont. but i want to put the patch into the gentoo ebuild for *
04:17.12Peggerso asterisk wil work with out any additional software right ,  what about chan_sccp doe sthat need to be loaded ?
04:17.29Qwellrajiv: check cvs..
04:17.36QwellI haven't seen any bugs about that fly by yet.
04:17.36Dr_Raychan sccp is part of asterisk
04:17.47Qwelloh, guess they have
04:17.49QwellDr_Ray: no it isn't
04:17.50Math`Pegger: its chan_skinny
04:17.58PeggerDr_Ray   awsome
04:18.01Qwellchan_sccp is a seperate module...
04:18.05Dr_RayI was wrong
04:18.10Math`Qwell: isnt skinny == sccp?
04:18.24Dr_Raythere are two sccp modules for asterisk
04:18.27QwellMath`: it's the same protocol, sure
04:18.30QwellDr_Ray: more than that
04:18.34Qwellat least 3
04:18.36Dr_Rayone built in, one not
04:18.51Math`Qwell: so sccp == skinny && skinny [is part of] *
04:18.55Qwellrajiv: yeah, cvs
04:18.55Math`so sccp is part of *
04:18.58Peggeryaha ther are chan_skinny and chan_sccp2
04:19.00Qwellno
04:19.05Math`ah there are 2 versions
04:19.10Qwell* can use the sccp protocol...
04:19.15Qwellbut, chan_sccp isn't in *
04:19.17Peggerchan_sccp is suposadly newer
04:19.24rajivQwell: http://cvsweb.digium.com/index.cgi/asterisk/contrib/scripts/vmail.cgi.diff?r1=1.15;r2=1.16 looks like
04:19.28Math`ok there's another implementation
04:19.30QwellPegger: chan_sccp is based off modifications to chan_skinny
04:19.39QwellMath`: several more
04:20.05PeggerQwell do you have an ipinion on which would would be better to use???
04:20.12QwellPegger: I use chan_sccp
04:20.20Qwellbut, I've never used chan_skinny, so...ymmv
04:20.39PeggerQwell ok thanks
04:20.52Brijnjustinu: http://pastebin.ca/27951 line 130... It's looking for my own IP in the [default] context...... Why :((
04:22.47Dr_Raygastman is ghastly
04:23.08Dr_Raybut I'm not happy with any of the management stuff
04:24.53file[laptop]Brijn: you need an extension 's' in the context 'lightspeed-in'
04:25.06file[laptop]Brijn: once that's done, that should be working
04:26.19Qwellman...that name always bothers me...lightspeed
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04:27.13rajivstkn: hi
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04:30.00Brijnfile[laptop]: there is one, let me pastebin it http://pastebin.ca/27953
04:30.19Brijnfile[laptop]: so close!
04:30.23[TK]D-FenderQwell : yeah lightspeed bothers me too : Since light is both a particle and an EM band to itself this lends us to ponder several cute ideas like : If light is a wave, that means that light meanders, but if it isn't going straight, doesn't that mean its actually going faster than the speed of light?
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04:30.38Qwellumm
04:30.39BrijnQwell: There support isn't at the speed of light anyway
04:30.39[TK]D-Fenderthat would mean we compare everything to the speed of light taking its time :D
04:30.55Qwellmy reference to lightspeed is the first hit on google images
04:31.05Qwell(NSFW!)
04:31.22Brijnhahaha
04:31.42BrijnTheir support staff??
04:31.45Qwellso yeah...needless to say, I wouldn't use a voip provider of the same name. :)
04:31.51Qwellwell...unless that were the case
04:31.53file[laptop]Looking for 300 in lightspeed-in
04:31.59file[laptop]make sure there is an extension 300 in lightspeed-in
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04:33.20BrijnWOEHOE!!!!!
04:33.22*** join/#asterisk spootnick (n=irc@50.118.233.220.exetel.com.au)
04:33.32BrijnThanx man!
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04:34.16kuku5Anyone here ever seen a c program that write to the serial port?
04:34.26BrijnNow I at least have something work, and can start from there.. See if understand what was wrong all the time..
04:34.26justinuon linux?
04:34.46spootnickhi all. how do i display the realtime SIP peers currently registered in my * box via CLI ?
04:34.52mog_homeyou can dump to /dev/tty
04:34.55mog_homeright
04:35.01javoi am having trouble with my sip phone.  i can call out anywhere, but when people call me, it says that i am on the phone and goes straight to voicemail.  does anyone know a likely cause of this?
04:35.01Math`kuku5: just open /dev/ttySX
04:35.08kuku5how do i do that
04:35.15Math`open() or fopen()
04:35.19kuku5ergh
04:35.25Qwellkuku5: it's going to be a long night for you, if you have to ask that...
04:35.46kuku5great :)
04:36.33[TK]D-Fenderjavo : pastebin your extensions.conf
04:37.07kuku5Qwell: do i compile with gcc or cc
04:37.41kuku5:)
04:37.47Qwellkuku5: This is not the forum for that.
04:37.53kuku5i know
04:38.03kuku5but i get and give a lot of help here :)
04:38.30justinugo to #c
04:38.34justinutalk to mixe
04:38.39justinuon efnet
04:38.53justinutell him j2 said to help you
04:39.23delmarin HEAD, usage of SetVar() has been changed to Set(), does anyone know if this is similar for SetGlobalVar? is it not SetGlobal ?
04:39.30kuku5they need a key :(
04:39.30delmarnot=now rather
04:39.44marc32344what does fifo_db do in ser?
04:40.15kuku5mixe isnt online :(
04:40.25justinudrat...
04:40.30justinusvetmixe?
04:41.08Qwelldelmar: show applications like set
04:41.23kuku5justinu: nope
04:42.52delmarcheers, looks like SetGlobalVar is still the thing
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04:44.44gambolputtyIs kpfleming here?
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04:46.45znoGcould someone fill me in on the technical reason why an Ambient MD5XXX card doesn't work with Asterisk?
04:47.01QwellznoG: without knowing, I'd guess lack of drivers
04:47.02znoGis there some hardware restriction or simply support for it hasn't been written?
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04:48.40kuku5Qwell: http://pastebin.com/421397   < invalid argument
04:49.41Math`thats EINVAL?
04:51.10mog_homei much prefer EAGAIN
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04:52.31Math`kuku5: what do u want to do
04:53.38*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
04:54.05kuku5read the serial port - thats it
04:54.13kuku5( for starters )
04:54.32Math`but why do u want to do that, whats your final goal
04:55.52pcmis there anywhere the log from the channel ?
04:56.26Math`0x378 is the parallel port, fyi
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04:59.02denonunless of course your parallel port is 0x278
04:59.14Math`lol
04:59.17denon</being_difficult)
04:59.19denon>
04:59.52blitzragehypa7ia: j00 'round?
05:00.22kuku5i have an embedded device that has proximity card readers and relays connected to it that control door access
05:00.49kuku5what is the com port
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05:04.22gambolputtyIs anyone here a bug marshall?
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05:05.26mog_homegambolputty are you cyberdjheffer
05:05.31mog_homecause i am a marshall
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05:23.05delmarwhats the best way to do like... ok national dialing can be 0 then 8-10 digits... and international is 00 then could be 8-12 digits... ie. exten => _0(what goes here)... and exten => _00(what goes here) ?
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05:23.16file[laptop].
05:23.20delmarjust never got the hang of the X and N and stuff :P
05:23.22file[laptop]. will match any length
05:23.43Math`N matches > 1
05:23.48Math`X matches anything
05:23.53Math`(except *)
05:23.59QwellMath`: not A, B, C, or D, or #
05:24.07Math`matches any digit**
05:24.08Math`:P
05:24.59delmarok but i cant do _0., because they need to be seperate
05:25.18Qwelldelmar: what country?
05:25.22delmarNZ
05:25.25Qwellno clue
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05:25.34delmarok let me explain further...
05:25.58Qwellthere are rules you need to follow...  You can either do 3 seperate lines for NZ national, or 1 line with 8 digits followed by .
05:26.21Qwellis there a "match zero or one character" match?
05:26.41QwellI think it'd be similar to ? elsewhere
05:27.15delmarok numbers would be like this...
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05:28.24delmarnational can be for example..   063271234(lanline)  or 021409123 but can also be as long as 0211123456....
05:29.13Qwelldelmar: is there a minimum each digit can be?
05:29.20delmarinternational is for example... 00 678 29333 (example of a short number in Vanuatu) or 0013038281234 for a number in the usa etc
05:29.27Qwellie; in the US, the lowest possible number would be 2002000000
05:29.33blitzrageyou match as much as you can, then use a pattern or . after... most specific will match
05:29.41blitzrage_NXXNXXXXXX
05:29.46blitzrageN == 2-9
05:29.48Qwellblitzrage: I'm thinking more in the middle.
05:29.57blitzrageZ == 1-9
05:29.58Qwelllike NXX?XXXXXXX
05:30.02delmarQwell, yeah i guess so... the minimum would be... 0XXXXXXXX so 0+8
05:30.05blitzrageX == 0-9
05:30.26Qwelldelmar: so, any part of the number can be any digit?  even 0 or 1?
05:30.43Qwellif so, _0XXXXXXXX. would do the trick
05:30.49delmarlet see...
05:31.08Qwelland if there were ? (there isn't...I just checked), you could do like 0XXXXXXXX??
05:31.13Qwell_0XX...etc
05:31.22delmarok
05:31.29blitzragethere is no ? afaik
05:31.36blitzrageand what would ? be?
05:31.47Qwellblitzrage: 0 or 1 of any digit
05:31.49Qwelllike...
05:31.55Qwell_X? would match 1, or 11
05:32.00Qwellbut not 111
05:32.09Qwellwhereas _X. would match 1, 11, 111, 11111111111
05:32.24delmarso the answer is... exten => _0XXXXXXXX. etc for NZ National and that should cover it... then I can have _00. for International?
05:32.38implicit??
05:32.48blitzragehrmm...
05:32.51Qwelldelmar: no, see...that won't work.  Something is either broken with the NZ implementation, or you're wrong
05:33.00blitzrageprobably need _X and _XX lines
05:33.03implicit??????
05:33.07QwellYou need to dial 00 to exit the country?
05:33.22Qwellthat couldn't possibly work if an NZ phone number started with 0
05:33.29blitzragethen don't match on the 00 -- just dial the number and add 00 to the front of the ${EXTEN} lines
05:34.41delmarQwell, ok what im "trying" to do is simulate NZ dialing on *, but it would probably go via VoIP provider most of the time... so .. one line for NZ National where it will Dial(IAX2/user@provider/64${EXTEN:1} sorta thing.. and another where the 64 code is ommited.. for all other dialing.... you get me?
05:35.01Qwellnope
05:35.49delmarok 0 then area code then number.. national dialing... 00 then country code.. then area code .. international dialing
05:35.51Qwellbbl
05:36.28Math`delmar: can an area code start with 1?
05:36.33delmarthe national dialing rule... needs to add a 64 when sending to the provider.
05:36.41delmarhrm
05:36.42delmarno
05:36.52delmarMath` what u thinking?
05:37.06Math`so exten => 0N. ; in-country
05:37.16Math`exten => 00. ; international
05:37.25delmaronly numbes starting with 1 are service numbers and are like.. 123 126  u dont dial 0123 .. just 123
05:37.38Math`and they are 3 digits?
05:37.42blitzrageMath`: don't forget the _ :0
05:37.45blitzrageerr :)
05:37.50Math`blitzrage: ah thanks for that one
05:38.07delmarMath`, yeah but i already have a prior context for service numbers to override them and do Congestion :P
05:38.17Math`lol
05:38.22delmarMath`, ok cheers, ill impliment that and see if it works :P
05:40.08delmarOh hey so just to clarify.. the _ is needed when doing like _00., etc but not needed when doing stuff like 1234, etc... when else is a _ required and wahts the deal?
05:40.21Math`yeah
05:40.30Math`_ is when you need to match
05:40.40Math`aka: when its not only digits (or special extension)
05:40.43delmari take it it doesnt matter if u have _1234,etc either?
05:40.51Math`u dont do _1234
05:41.01Math`but you do _NXXNXXXXXX etc...
05:41.29delmarah i get it. _ for pattern matching .. without when u are not.
05:43.54Math`:)
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06:03.02SkramXhello all
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06:21.58delmarusage of goto is like.. exten => 1234,1,Goto(context,X,N) sorta thing right?
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06:23.54delmarhmm
06:23.54Math`delmar: goto(context,extension,priority)
06:23.54delmaryeah
06:23.54Math`goto(outgoing,1234,1)
06:23.54Math`what was your question
06:23.54delmarcould also be exten => _1234.,1,goto(context,extension,priority) too right?
06:23.54Math`of course
06:23.54delmarhrm.
06:24.32delmardidnt think i had it messed up.. not playing ball tho.
06:24.33*** join/#asterisk ComputerWarm (n=dan@rddrpx29-port-20.dial.telus.net)
06:25.14ComputerWarmHello all, anyone here use astbill?
06:25.14Math`delmar: whats the error
06:25.45delmarLabel missing trailing ')' at line 223
06:25.45Math`whats line 223?
06:26.10delmarsec.
06:26.30delmarnm. i see.
06:26.38Math`ur missing a ) :P
06:26.52delmarno that would say missing parenth.
06:27.01Math`ah
06:27.02*** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de)
06:27.04delmarits quite good at tell u whats wrong.
06:27.09*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
06:27.19Math`missing trailing ) sounds like missing ) for me
06:27.41FuriousGeorgeso how come the freenode op is called lilo
06:28.45Math`FuriousGeorge: because lilo  is a staff member of freenode?
06:29.00FuriousGeorgeMath`: sounds good
06:29.13Math`indeed
06:29.18mog_homei get those messages from dmwaters
06:29.24mog_homeand always think he is someone i know
06:29.30mog_homeand am dissapointed everytime..
06:29.40FuriousGeorgelol
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06:30.05FuriousGeorgewell i guess you know him now, but regardless of the name hes almost always disappointing
06:31.20Math`if u say so...
06:31.39ComputerWarmhello. anyone here used astbill?
06:33.28Math`uhm I've a deja-vu impression
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06:40.53SarahEmmhihi
06:42.43moraleirc'ing from your workstation?
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06:43.45SarahEmmmorale: me?
06:43.47SarahEmmoh lol
06:43.49SarahEmmheya morale.
06:43.55SarahEmmtook me a second to realize who you were.
06:43.59Vcohttp://rmeek141.home.comcast.net/TrollBooth.JPG
06:44.04SarahEmmyep! ircing at work, as are you :)
06:44.27Math`what time is it for you?
06:44.34SarahEmmit's 0144 for me
06:44.38moraleSarahEmm: i go through my home system so no one dos's me
06:44.40SarahEmmi work rotating shifts. nights this week.
06:44.55SarahEmmmorale: heh. i usually do, but having some VPN issues right now so i'm not.
06:45.04moraleah.
06:45.42delmarhmmm
06:45.53SarahEmmmorale: you ssh home and use a console client, or vpn home?
06:46.44sylecat cost me 150 bucks at vet tonight because she had a fever!
06:46.53moralei just ssh home..
06:47.11SarahEmmmorale: ahh.
06:47.12delmarI'm trying to setup a variation of the emergency call handling found at http://www.voip-info.org/wiki-Asterisk+tips+911... the last one.. and it's just looping out and not doing what it should.
06:47.13moralesyle: it costed me 400$ two weeks ago on a dead cat.
06:47.41delmarcan anyone see anything major wrong with the last variation at http://www.voip-info.org/wiki-Asterisk+tips+911 ?
06:47.51sylegod i think it cost me that much to get both my cats first shots and spaded when they were kittens
06:48.40syleyou paid 400 bucks to put a cat to sleep?
06:48.51delmarsyle, yeah one of our cats had some kinda spasm which we now think was a pulled mussel but she was lookin all wierd walkin lopsided so off to the Vet.. $80 call out plus another $70 for the consult
06:49.33delmarbah. im off to mow the lawns. bbl
06:49.46sylemow the lawn
06:49.51sylewhat part of the world do you live in
06:50.00sylelike 1 am here hehe
06:50.02Vcolawns?
06:50.21Qwelloh shit...  http://www.voip-news-net.com/2005/11/voip_e911_cutof.html
06:50.30QwellFCC folded?
06:50.42implicithehehe
06:50.50implicityou ddin't know?
06:52.47sylesays they are only folding on members before nov 28
06:52.52sylenot any new ones after that
06:53.09Qwellso?  heh
06:53.51*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
06:54.13sylethey don;t really have a choice but to fold i think, a million people pissed off at the FCC
06:54.25Vco"more" pissed at teh FCC
06:54.38sylebudget cuts would come to that dept easily
06:55.42implicitVco, yeah :)
06:57.18Vcooooh
06:58.04QwellVco: they have pricing?
06:58.25Vcohttp://www.voipsupply.com/index.php?cPath=0_99_300
06:59.09Qwellstill nothing at the digium store. :(
06:59.15Vcoheh..heh..
06:59.23mog_homeyeah
06:59.27mog_homeits comming
06:59.29mog_homeits comming
06:59.30Vcosurprise: Please Note: We are currently taking pre-orders for the new Digium TDM2400 Series, Full-Lenth PCI cards. Product is anticipated to begin shipping 11/18/05.
06:59.34Qwellahh...
06:59.36Vcoas per ususal
06:59.41Vcousual even
06:59.53Vcoat they have some numbers tho
07:00.11Qwellwow, thats pricey
07:00.22syleprice?
07:01.06syleholy crap
07:01.10implicit2495 for 24fxs w/ on board echo cancellation
07:01.34*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
07:01.38Vcoi mean...fxs i can kinda see
07:01.46sylewell i guess i won;t be returning my rhino channel bank lol
07:01.49implicitVco, why, lol
07:01.50Vcohaha
07:01.53clive-a channel bank is way cheaper than that
07:01.56RoyK24 pri?
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07:02.02RoyK:)
07:02.09SarahEmmmorale: thanks for reminding me to fix my VPN :) all better now.
07:02.32RoyKclive-: a channel bank seems to me a far cleaner solution as well
07:02.38implicitRoyK, yeah
07:03.10Qwellhmm
07:03.17Qwellfreaking idiots at voip-supply, I swear
07:03.32implicitdigium developed it?
07:03.37mog_homeyes
07:03.40Qwell40fxs / 0fxo - $789.95
07:03.46mog_homedigium develops all digium hw
07:03.46QwellTHAT is why I don't like them
07:03.49implicitlol
07:03.54mog_home40fxs! awesome
07:03.57QwellThey constantly fuck up like that, heh
07:03.58mog_homei need that card
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07:04.08mog_homeolle!
07:04.32SarahEmmwowie, there's a new TDM card.. niftys :)
07:04.43implicitSarahEmm, lol
07:04.49SarahEmmimplicit: mew?
07:04.55impliciti wouldn't use them
07:05.13syle40 fxs for 789 ?
07:05.14sylewhere
07:05.16SarahEmmimplicit: because?
07:05.20Vcoalthough, for the 4 ports, compared to buting a tdm400....
07:05.26Vcocompared to buying..
07:05.27Vcoeven
07:05.58implicitSarahEmm, expensive for what it is, uses softwaredsp, and probably shitty quality
07:06.28mog_homenah implicit
07:06.28mog_homeit sounds clean
07:06.28mog_homelike any card
07:06.36SarahEmmimplicit: ahh. mew.
07:06.46SarahEmmimplicit: i'm running an X100p for part of my home system, works well for me :)
07:06.52*** join/#asterisk Teeli (i=Tili@219.136.106.152)
07:07.01mog_homeall zap cards sound like any dsped card
07:07.02sylei;m authorized to resell rhino channel banks if anyone wants one
07:07.10Qwellsyle: sure, I'll take one
07:07.14implicitmog_home, not exacty
07:07.20implicitand they end up being more expensive
07:07.28implicitbecause of the high requirements on DSP
07:07.28mog_homeumm id love for you to show me the difference
07:07.33mog_homeor if you could pick one out of a line up
07:07.40Vcoif you're authorized to take bits of string as payment i'll take a few
07:07.43mog_homeon normal t1 lines or analog etc
07:08.03mog_homei think people just dont realize that pcs have gobs of proccessing power
07:08.09mog_homethat you can throw at silly things like echo
07:08.37implicitPRI coming in through a digium card on * i could completely tell apart from a one coming into a cisco AS5400
07:08.38Vcodo you have the one with echo cancel on the card?
07:08.45mog_homesure implicit
07:08.54mog_homei find that pretty hard to believe
07:09.00mog_homeunless your blind
07:09.05mog_homeand have super human hearing
07:09.06implicitmog_home, i am
07:09.09implicitfucker
07:09.10Qwellimplicit: I hear the Navy is hiring human sonar.  You might want to apply
07:09.13implicit;)
07:09.13mog_home8khz mono sucks going through anything
07:09.41mog_homewhatever
07:09.55mog_homeif you want to be an audio head thats fine
07:10.01mog_homebut the 99% of the rest of us
07:10.05mog_homecant tell jack squat
07:10.39impliciti admit i'm pretty anal about audio quality, that's why i buy things like http://www.sensaphonics.com/soft2x.html
07:10.53implicitbut the quality difference is pretty large
07:11.02implicitsomeone untrained could easily pick it out if they were listening for it
07:12.03Vcowow..it's a good thing voip supply marks this Digium hardware as Asterisk tested, or i don't know what i might have ended up buying....
07:12.14SarahEmmlol
07:12.18implicitVco, lol
07:17.07*** join/#asterisk lehel (n=lehel@82.79.20.17)
07:17.20lehel'mornin
07:17.31mog_homemornin
07:17.31Qwellwow...umm...who said T1 card + rhino channel bank was cheaper?
07:17.37mog_homeyeah
07:17.45mog_homeused channel bank
07:17.46mog_homeand t1
07:17.47Qwell$1800 for one of these suckers
07:17.49mog_homemight be cheaper
07:17.56mog_homebut getting fully populated channel bank
07:17.58mog_homeis pricy....
07:18.12mog_homeused zhones though do rock if you can find em
07:18.23Qwellzhones?
07:18.32mog_homezhone is a crappy channel bank
07:18.36mog_homeyou can find em on ebay
07:18.37mog_homesometimes
07:18.41mog_homebut they go for like 500
07:18.57*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
07:18.57Qwellcrappy but they rock?
07:19.11mog_homewell they dont have all the niceties
07:19.14mog_homelike for example
07:19.18mog_homeno fxo caller id
07:19.29Qwellthats dirty
07:19.30mog_homeand they have been known to drop all the channels
07:19.41mog_homebut other than that
07:19.45mog_homethey rule
07:19.49Qwellwe have a nortel pbx at work...we don't get cid...
07:20.02mog_homeew
07:20.08Qwellbiggest pile of shit phones too...
07:20.13Qwellthey don't have a god damned mute button
07:20.17Qwellno, they do...
07:20.25Qwellbut if you press it...BAM, speakerphone
07:20.26Vco>crotch<
07:20.31mog_homelol
07:20.35*** join/#asterisk shido6 (i=shido6@d221-68-216.commercial.cgocable.net)
07:20.36mog_homethat rules
07:21.11Qwelllike...when you think about something like that...
07:21.30Qwellimagine the biggest cluebie coming in here, asking about buying a hundred budgetones...
07:21.33mog_homewould make a great gag to screw the new guy in  office
07:21.41Qwellthats the type of person who buys this pbx, I swear
07:21.53Vcothat ranks up there with the wallmount budgetone that doesn't have a hook to keep the reciever on .....
07:22.00mog_homelol
07:22.02QwellVco: you're kidding?
07:22.09mog_homeput it on a slanted wall
07:22.10Vcothe 100's
07:22.11mog_homeor the roof
07:22.13mog_homeit works great
07:22.14Qwellthats awesome
07:22.37Vcoi have a frankentone phone...
07:22.42Vcoadd BT100 phone,
07:23.03Vcoand old USR wifi bridge into the case
07:23.28Vcovoila! The single shittiest wifi phone you could even order pizza with
07:24.18Vcoit was kinda annoying, since it was a white phone, and the bridge had those super bright blue leds..
07:24.28Vcofrickin lightshow at night...
07:24.53SarahEmmlol
07:24.58mog_homedamn someones car alarm is going off
07:25.04SarahEmmand a web interface tty app
07:25.06mog_homei wish someone would just steal the damn thing
07:25.16mog_homebeen going off for 30 minutes
07:25.28mog_homeits like the guy just wants us to blow his car up
07:25.40Vcohmm..
07:25.41Qwellhere, they'll tow it if the alarm goes off for longer than x minutes
07:25.57mog_homeno one is doing anything
07:26.02mog_homesome people came outside angry
07:26.04mog_homethen left
07:26.18Vcowhere you live?
07:26.26mog_homehuntsville
07:27.06Vcoohh..speaking of car alarms andsuch i should get remote start installed like....soon..
07:27.13implicitmog_home, explains why you are so into that digium card
07:27.14implicitlol
07:27.26implicitmog_home, i see where your bias comes from
07:27.35syleoww vco you should book that right now
07:27.37mog_homei thought it was pretty well known
07:27.39implicit:)
07:27.41syletook me 3 weeks to get in
07:27.52implicitmog_home, always used mog_home as your handle?
07:28.05mog_homemog_home, mog_work, or mogorman proper
07:28.14implicithow long have you been working for digium?
07:28.22mog_homejust over a year
07:28.30sylewhat do you do?
07:28.33implicitand how long around */
07:28.35mog_homesupport and dev
07:28.37RoyKmog_home: what do you think about the openpbx project?
07:28.39impliciti'm surprised we don't know eachother better
07:28.40mog_homejust a bit before that
07:28.44syle#DEFINE dev
07:28.48clive-mog_home when are digium going to get G729 with PLC or G729b ?
07:28.50mog_homeroyk you know what i think about that ^_^
07:29.00RoyKthree small characters
07:29.04mog_homeg729 b should probably work as is
07:29.09mog_homebut with plc no idea
07:29.11ComputerWarmquestion all does Asterisk support G723.1?
07:29.12potsboyhi all.. is it possible to register a analogue phone though a rhino bank into a queue?
07:29.14implicithavn't been around much lately but i was around a lot more before
07:29.19RoyKwhat's the diff between g729a and -b?
07:29.19mog_homei do little stuff for digium, i do bigger things in spair time
07:29.24clive-computerworm tyes illegally it does
07:29.28Qwellpotsboy: sure, why not?  a channel is just a channel
07:29.36RoyKComputerWarm: no it does not. g723.1 license is way to expensive
07:29.42clive-RoyK b has PLC, plus a few extras in it
07:29.56RoyKclive-: what is plc?
07:30.01mog_homeg723 is difficult legally as well
07:30.04clive-packet loss concealemnt
07:30.09RoyKok
07:30.10ComputerWarmRoyK oh. will asterisk do a pass thru  for that ?
07:30.12RoyKthat's nice
07:30.13potsboyaah k so it will reference it by channel not ip
07:30.17RoyKComputerWarm: it will
07:30.27RoyKComputerWarm: at least in theory. haven't tried it, though
07:30.31potsboytx
07:30.53ComputerWarmRoyK ok thanks. i guess I will have to try it or talk the carrier in to switching to G729.
07:31.13clive-didnt know any carrioers used 723.1
07:31.23ComputerWarmclive- these guys do
07:31.27Qwellis 723.1 even any good?
07:31.45RoyKQwell: i beleive so, for slow links
07:31.54mog_homeg723 is just really small
07:31.56clive-723.1 is ok , but sucks when you have packet loss
07:32.02mog_homeits like 729 doesnt sound better
07:32.02ComputerWarmQwell its crappy. most the time
07:32.04mog_homebut is tiny
07:32.30mog_homedamn im just gonna call the police on this guy
07:32.35mog_homeit wont stop
07:32.47clive-what guy mog?
07:32.54Qwellclive-: you
07:32.55mog_homethis guy left his car alarm
07:33.00mog_homeits been going off
07:33.04mog_homeFOREVER
07:33.26clive-lol, in south africa you can cal teh police for an armed robbery and they dont arrive
07:33.29sylecall it in to towing company, they will be more than happy to take it
07:33.43mog_homecan you really do that
07:33.50Qwellcall and ask
07:33.58mog_homethat would be awesome
07:34.00syleof course, they make money this way
07:34.00Qwellif so, they'll be there in a second
07:34.04mog_homelike parking lot is full
07:34.08mog_hometow away some guy
07:34.09sylecosts them to pick up their vehicle after
07:34.11mog_hometake his spot
07:34.16Qwellmog_home: no :p
07:34.19Vcocall in a noise complaint
07:34.21mog_homewhy not.....
07:34.26Qwelljust like you can't call a locksmith to break into a ferrari
07:34.31mog_homeyeah i did , the management where i live sucks
07:34.50mog_homei picked the place closest to campus with the cheapest rent
07:34.57mog_homethere are drawbacks to this though....
07:35.08Vco"car alarm has been going off for a while, angry mob gathering downstairs.."
07:35.41Vcopitchforks and clubs and such
07:35.49Qwellyou could probably get away with throwing a rock through the windshield at this point
07:35.54mog_homelol
07:36.03Qwell"dude, the alarm was totally going off when I got home...I've got like...200 witnesses"
07:36.03mog_homesomeone is under the car now
07:36.15mog_homei think they are gonna disco his battery
07:36.18Qwellheh
07:36.24mog_homeor cut his breaklines
07:36.28mog_homewhatever it is
07:36.31Qwellproblem solved
07:36.32mog_homei dont care at this point
07:36.49Vcoy'know...depending on what floor you are on...a 2L pop bottle full of water does some nasty ass damage (at least off the 25th floor it did anyway)
07:37.11mog_homeheh im just on second
07:37.12QwellVco: I imagine 5th and above would start doing it
07:37.21Vcowell..
07:37.21mog_homewell whatever that guy did he killed the alarm
07:37.38Qwellwell...2liter...
07:37.43Qwellmaybe 4th
07:37.54Vcohey ..he killed the alarm, unlocked the door AND popped the ignition...fancy that..
07:38.18Qwell*10 minutes later* <mog_home> so, turns out it was my car
07:38.20Vcopissed off people tossing him gas money to get it out of the area
07:38.23mog_homelol
07:38.24mog_homehehe
07:38.34SarahEmmrofl Qwell
07:38.46mog_homeman i could leave my car with the windows down and the key in the ignition no one would steal it
07:39.11mog_homei drive the out on my own broke as hell college student mobile, thats right an 89 olds mobile calias
07:39.30Qwellbrown?
07:39.38mog_homeits started red
07:39.42mog_homebut its getting there
07:40.31*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
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07:47.45Qwellbed time
07:48.17RoyKhm... does anyone know how I can allow a server to listen to two ip addresses, one on each a network, and allow each of them to be accessed from the internet? currently i have a default gateway on one of the networks, but that means if I try to access it on it's other address, it'll try shipping back the data out the other nic, which is rather a bad thing....
07:48.25RoyKa little OT, perhaps, but then :P
07:48.36Qwellby server, you mean?
07:48.52Qwelljust any program?
07:49.04RoyKlinux system
07:49.31RoyKmeaning having two default gateways, one for each nic
07:49.33RoyKsomehow
07:50.01QwellRoyK: think there is something on tldp.org - perhaps the advanced routing howto
07:51.36Qwellprobably just a little bit of iptables magic
07:52.43*** join/#asterisk JohnJacob (n=JohnJaco@pcp0011543623pcs.mainf01.in.comcast.net)
07:54.00Qwellhttp://www.tldp.org/HOWTO/Adv-Routing-HOWTO/lartc.rpdb.multiple-links.html  part of it
07:54.50QwellRoyK: I accept paypal. :P
07:54.57mog_homelol
07:55.06mog_homedont we all qwell
07:55.10Qwell;]
07:55.13mog_homei have gotten 20 bucks thus far
07:55.17mog_homeit rocked
07:55.30RoyKQwell: lol
07:56.57QwellThat link should actually do most of what you need...as long as your programs can listen on multiple interfaces (not difficult)
07:58.28*** join/#asterisk phara0h (n=yah@229.80-202-56.nextgentel.com)
07:58.37phara0hHi!, anyone working on documentation for v1.2 yet? (I am a couple of other are, and I was wondering if there was some place we should put our documentation/read others documentation so we don't create the same thing :))
07:58.44mog_homethere is
07:58.46mog_home~thebook
07:58.49jbotfrom memory, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
07:59.00mog_homeand asteriskdocs
07:59.04mog_homeand wiki
07:59.07mog_homeetc etc
07:59.14Qwelland bugtracker, if you can do doxygen docs
07:59.24mog_homemake progdocs did rock
07:59.29mog_homebut needs work
07:59.39phara0hmine crashed :(
07:59.54phara0h(on SuSE OSS 10.0)
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08:00.43*** join/#asterisk AgentRed (n=bharatsa@210.211.246.47)
08:00.48AgentRedHello there
08:01.10SarahEmmhihi AgentRed
08:01.23phara0hSo what you are saying there is really no central place where everyone is putting most of the new docs.. actually I first went to voip-info, but there was hardly anything on the page called "Asterisk v1.2" ... just a listing of dialplan functions ..
08:01.50AgentRedIam unable to load Asterisk cos the of the failure in loading the app_voicemail.so
08:02.16mog_homeasteriskdocs
08:02.19mog_homeis good place
08:02.25AgentRedcan anybody please guide me as to how do I come up with a solutino for this problem
08:02.32phara0hok, i'll check it out :)
08:02.35mog_homewhat is the error
08:03.03AgentRedhmm
08:03.22AgentRedwait let me paste the error in the pastebin
08:04.20mog_homehurry as im about to sleep
08:04.25AgentRedok
08:04.30AgentRedwait a sec [please
08:05.59AgentRedhttp://pastebin.com/421501
08:06.03AgentRedthis is the link
08:06.13AgentRedplease have a look at it
08:07.29mog_homeit looks like you had some issue with adsi not being fully installed
08:07.43mog_homeis this latest head?
08:07.50AgentRedya
08:08.25mog_homelet me do update
08:08.39AgentRedshould I go ahead installing the latest ADSI
08:08.39AgentRed?
08:08.46mog_homeadsi is part of asterisk
08:08.52mog_homedo you have zaptel hardware?
08:08.57AgentRedno
08:09.10Qwellheh, res_config_odbc is failing right now on head
08:09.14mog_homehmm i think i know whats wrong
08:09.25AgentRedone more thing is Iam getting errors compling the zaptel
08:10.11mog_homewell if you remove zaptel.h from your libraries
08:10.15mog_homeand rebuild it should work
08:10.21mog_homeand build without adsi stuff
08:10.40AgentRedis it because of that I am getting this error
08:10.42AgentRed?
08:10.55Qwellhmm, what is res/ on the bug tracker?
08:11.00mog_home<PROTECTED>
08:11.02mog_homeget rid of that
08:11.05mog_homeand rebuild asterisk
08:11.11mog_homemake clean make asterisk
08:11.18AgentRedalright
08:11.27mog_homegoodnight all good people
08:11.37AgentRedGood night mog _home
08:11.41AgentRedthanks a lot
08:13.39*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
08:14.05*** join/#asterisk sjobeck99 (n=sjobeck9@london.sjobeck.com)
08:17.41Qwellok, now bed
08:21.19ComputerWarmanyone here use A2Billing?
08:21.41ComputerWarmor maybe better known as ASterisk CallingCard Platform
08:21.48*** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com)
08:21.57delox99hi all
08:22.05phara0hanyone know how to get MOH Native going on v1.2 ?
08:22.32delox99can someone explain what is Dundi exactly?
08:22.49phara0hwww.dundi.com
08:24.09delox99i m looking for a way to connect to pstn lines
08:24.27delox99anyone know of a pstn lines provider?
08:24.51phara0hwhere do you live?
08:25.02delox99canada ottawa
08:25.18tclark_3rd world country
08:25.52shido6i used to visit ottawa
08:26.09delox99i want to have my asterisk to conenct to some pstn switch that can give me good prices for long distances
08:26.17phara0hhttp://www.voipbuster.com/en/index.html <--- not free to canada but ...
08:26.25delox99where you from shido?
08:27.51delox99i mean i want to provide clients with good prices for landline calls
08:27.52tclark_shido6: is hding out near winsor, kitchener corridor some where
08:28.03shido6yep
08:28.38shido6I used to have a place near vanier parkway
08:28.49ComputerWarmHey anyone know of any test numbers for canada. i am trying to test out this new provider find out where all they do not cover
08:28.50emdubanyone have a cisco 7960g?  man what a pain in the neck configuring this thing is
08:29.02shido6its not that bad :) emdub
08:29.14emdubugh can i poke your brain some?
08:29.22shido6shoot
08:29.24emdubmaybe its just the factory firmware sucks
08:29.34shido6i run 7.4 here
08:30.02emdubwell, i plugged it in, it has an ip, etc... i hit **# to unlock it and go under configuration and try to edit the tftp server so i can upgrade it and then it says "this key is not active here" or something
08:30.09emdubevery time i try to change the contrast it never saves, etc
08:30.10emdubwtf!
08:30.25shido6you sure its unlocked?
08:30.32shido6if it is - then build a conf file
08:30.39emdubthe little lock ucon under network config shows unlocked
08:30.48shido6and force it to look for a tftp server in you dhcp server settings
08:30.51emdubproblem is i cant configure my dhcp server to allocate a tftp server ]
08:30.53emdubya, cant
08:31.04shido6then build a little network of your own
08:31.07emdubugh
08:31.07shido6grab a switch
08:31.13emdubyou have got to be kidding me
08:31.27shido6dood its not that hard :)
08:31.28emdubman how useless
08:31.37emdubthats not the point
08:31.43clive-lol, in south africa you can cal teh police for an armed robbery and they dont arrive
08:31.44emdubi should be able to change the tftp server ip from the phone!
08:31.45shido6it will take 6 minutes
08:31.46clive-oops
08:31.47clive-:)
08:32.22h3xget an office next door to krispy kreme, and you will get them to show up for suspcious activity
08:32.55SarahEmm.ckear
08:33.23emdubi should go wake up my cisco sales guy ;)
08:33.26emdubhehe
08:33.28emdubor call tac
08:33.30emdubfix my phone!
08:33.47delox99will dundi help me connect to pstn around the world?
08:33.59delox99sry for m newbieness
08:35.34ComputerWarmdelox99 have you tried www.didx.org ? i have seen it on the biz list alot
08:35.35delox99i need to find a way to redirec sip calls to pstn lines in other cities
08:35.54ComputerWarmoh wait sorry you are looking to buy termination minutes?
08:36.48delox99termination minute?
08:37.19delox99probably yes
08:37.40ComputerWarmsorry i can`t recommend a cheap provider. i have heard nufone is good
08:57.35*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
09:15.29*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
09:19.08*** join/#asterisk bzbw (n=wlwzhang@24-205-27-223.dhcp.nrwl.ca.charter.com)
09:19.22bzbwanyone know why I got following error?
09:19.23bzbw<PROTECTED>
09:19.23bzbwNov  8 17:05:25 WARNING[4489]: chan_zap.c:890 zt_open: Unable to specify channel 1: No such device or address
09:19.23bzbwNov  8 17:05:25 ERROR[4489]: chan_zap.c:6650 mkintf: Unable to open channel 1: No such device or address
09:19.23bzbwhere = 0, tmp->channel = 1, channel = 1
09:19.24bzbwNov  8 17:05:25 ERROR[4489]: chan_zap.c:10030 setup_zap: Unable to register channel '1-3'
09:19.26bzbwNov  8 17:05:25 WARNING[4489]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1
09:19.28bzbwNov  8 17:05:25 WARNING[4489]: loader.c:543 load_modules: Loading module chan_zap.so failed!
09:19.45bzbwI use 2 TDM400
09:20.55*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
09:22.08*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
09:22.21bzbwanyone here that can help me on my issue?
09:22.24bzbw<PROTECTED>
09:22.24bzbwNov  8 17:05:25 WARNING[4489]: chan_zap.c:890 zt_open: Unable to specify channel 1: No such device or address
09:22.24bzbwNov  8 17:05:25 ERROR[4489]: chan_zap.c:6650 mkintf: Unable to open channel 1: No such device or address
09:22.24bzbwhere = 0, tmp->channel = 1, channel = 1
09:22.24bzbwNov  8 17:05:25 ERROR[4489]: chan_zap.c:10030 setup_zap: Unable to register channel '1-3'
09:22.26bzbwNov  8 17:05:25 WARNING[4489]: loader.c:403 __load_resource: chan_zap.so: load_module failed, returning -1
09:22.28bzbwNov  8 17:05:25 WARNING[4489]: loader.c:543 load_modules: Loading module chan_zap.so failed!
09:22.28SarahEmmgah
09:22.30SarahEmmplease don't paste that much
09:22.32SarahEmmuse a pastebin1
09:22.40SarahEmmand don't repeat....
09:22.44bzbwk
09:22.49SarahEmmit sounds like it can't find your card
09:22.53SarahEmmdoes lspci show the card?
09:22.58bzbwk
09:23.32bzbwgot a bunch of output, not sure what to look at.
09:23.41SarahEmmpaste it into a pastebin.ca pastebin pls
09:23.44SarahEmmand paste the URL in here
09:24.16*** join/#asterisk bon (n=bon@voip.in.radiolan.sk)
09:24.22bzbwwhat do u mean by paste the URL? It's from my linux command line.
09:24.39lehel~pastebin
09:24.41jbothmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
09:24.57*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
09:25.01moralecalgary is getting very sleepy...
09:25.04SarahEmmbzbw: copy the output of lspci, and go to pastebin.ca and paste it in
09:25.05lehelbzbw: learn using it
09:25.07bzbwthx
09:25.09SarahEmmmorale: heh :)
09:25.15SarahEmmmorale: Toronto is mildly tired.
09:25.16*** join/#asterisk voipjoy (n=root@1.fix.netvision.net.il)
09:25.29voipjoyanybody can advice what is this? Nov  8 11:25:15 voice kernel: Got pulse digit 13 on TE4/0/1/15???
09:25.34moralei have my emergency red bull in the car
09:26.15*** join/#asterisk zoa (n=kkk@pirus.securax.be)
09:26.16zoayo
09:26.19zoaHa Ho
09:26.39lehel'llo
09:26.48bonhello
09:26.54boni'm having problems with extensions
09:26.58bonhttp://pastebin.com/421534
09:27.01*** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com)
09:27.06bonthat is a user registering with mysql db no problem
09:27.15bonbut when i call that number,it doesn't work :(
09:27.57bzbwSarahEmm: here is the link for lspci: http://pastebin.ca/27967
09:28.41SarahEmmbzbw: is the only telephony card in the box the TDM400p?
09:28.53bzbwyes, there are 2 of them
09:28.54boneven tough it says Dial(SIP/${EXTEN})
09:28.55bon:(
09:30.10SarahEmmbzbw: hrm, yeah i do see them there.. pastebin your /etc/zaptel.conf
09:33.19bzbwthx SarahEmm, here it is http://pastebin.ca/27968, it should be the default one
09:33.42*** join/#asterisk rking (n=rking@ip68-1-234-152.dl.dl.cox.net)
09:33.46bzbwSarahEmm, I thought we don't use /etc/zaptel.conf any more, right?
09:34.45SarahEmmerr
09:34.54SarahEmmwhat has it been replaced with?
09:34.57SarahEmmlast i heard it was still used
09:35.28SarahEmmyeah, you need to configure that file bzbw
09:35.32moraleyou can delete your /vmlinuz and/or /boot/vmlinuz - i hear those aren't used anymore either.
09:35.34SarahEmmthen run ztcfg -vvvvvv to configure the card
09:35.36SarahEmmlol morale
09:36.07bzbwemm, I thought it stated in the zaptel REAME
09:36.09bonmorale :)
09:36.41newlSarahEmm: *cracks whip* got that ringer done yet?
09:37.09voipjoyi have partly E1 channel only 14 instead of 30! how i disable the not used chanel that zapata will not handle it and not manage it?
09:37.17SarahEmmnewl: no :/ not much time lately for that
09:37.21bzbwSarahEmm: Channel Map: 0 channels configured.
09:37.35SarahEmmbzbw: that's because you didn't configure the zaptel.conf file
09:37.46SarahEmmbzbw: you need to edit it and put in the right config before you run ztcfg
09:37.49newlSarahEmm: Eh :)
09:38.00SarahEmmnewl: are you actually interested in it?
09:38.18bzbwSarahEmm: thx.
09:38.39SarahEmmbzbw: there's a page on the wiki that might help if you're having trouble with the zaptel.conf config :)
09:39.25newlSarahEmm: no, but someone has to prod you along hehe
09:39.38newlbesides, I thought it was a good idea.
09:40.12voipjoySarahEmm: could you tell me please - how i can disable chanell in zaptel.conf ?
09:40.45SarahEmmvoipjoy: disable channel? what do you mean?
09:40.47SarahEmmnewl: lol
09:40.59SarahEmmnewl: well right now there's little use for it because * doesn't have any off the shelf TTY apps :)
09:41.04SarahEmmwhich is another thing i need to finish
09:43.03gsthmm... since the upgrade to beta2 i get "Avoided initial deadlock" msgs and then asterisk kills the (sip) channel. i have seen this messages before but the calls worked fine nevertheless before.
09:43.13gstare there any known current problems about this?
09:46.00voipjoySarahEmm: i getting E1 line (30 lines) i use only 14, and receive some "noise" on other 16 lines...I like to disable them...that zapata driver will ignore them
09:46.25fourcheezedoes anyone know if a sipura spa-3000 can forward an incoming call on its fxo port to a sip url?
09:46.51fourcheezein other words operate in reverse
09:47.15fourcheezeor is the fxo port purely a backup in case the net goes down?
09:47.47SarahEmmvoipjoy: oh... no idea.
09:49.30*** join/#asterisk remibreval (i=Remek@pro75-3-82-234-175-208.fbx.proxad.net)
09:49.51remibrevalHello Everyone !
09:50.28remibrevalI'm still getting into troubles to answer a ringing PSTN call. THe communication goes to the IAW service provider when I answer. This is the post : http://pastebin.ca/27885
09:51.59bzbwSarahEmm: I just add one line in zaptel.conf: fxols=1-8, and got error: ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
09:52.18bzbwSarahEmm: WiKi won't help.
09:52.22SarahEmmbzbw: one sec.
09:52.30SarahEmmwhat pastebin was yours again bzbw
09:52.31SarahEmm?
09:52.45SarahEmmssh tonkinese
09:53.21SarahEmmbzbw: do you have FXS or FXO modules on the card?
09:53.21bzbwSarahEmm: you mean my previous zaptel.conf?
09:53.24SarahEmmyeah
09:53.43bzbwSarahEmm: only FXO
09:53.52bzbwSarahEmm: I believe
09:54.08SarahEmmerr.. okay, then you need to say fxsks=1-8
09:54.10SarahEmmnot fxols
09:54.31SarahEmmFXO interfaces use FXS signalling, FXS interfaces use FXO signalling.
09:54.36SarahEmmwelcome to the strange world of telco ;)
09:54.48*** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com)
09:54.53bzbwSarahEmm: thx, will try now.
09:59.50remibrevalnobody has a basic conf to accept an incomming call ??
10:00.15SarahEmmremibreval: err, you're just trying to accept a call? you've looked at example dialplans in the wiki?
10:02.23delox99i want to use a US satelite to send and receive large amount of data. Where should i look?
10:02.48SarahEmmerr
10:02.53SarahEmmlarge amount of data to the internet?
10:03.18delox99yes
10:03.36*** join/#asterisk Alystair (i=Alystair@CPE000d88a7a3b5-CM00407b8794db.cpe.net.cable.rogers.com)
10:03.42AlystairOh hey awesome
10:04.04SarahEmmdelox99: www.starband.com
10:04.15SarahEmmdelox99: latency on sat is awful, but it works if you're out in the middle of nowhere
10:04.32AlystairI need some help here :(
10:04.37SarahEmmAlystair: with?
10:04.45Alystairis there a generic site you can reffer me to?
10:04.51SarahEmmwww.voip-info.org
10:04.53SarahEmm~wiki
10:04.57SarahEmmhrm
10:05.00SarahEmm~rtfw
10:05.01jbotsomebody said rtfw was http://www.voip-info.org, the only place to get any real answers about * and it's many, many uses.
10:05.26Alystairis this mainly for personal or business info?
10:05.32SarahEmmAlystair: either
10:05.34delox99what do you mean in the middle of nowhere?
10:05.39Alystairawesomepossum, thanks Sarah!
10:05.50SarahEmmdelox99: err, why are you looking at sat as opposed to land-based ways of data transfer?
10:05.58SarahEmmAlystair: np :)
10:06.00bzbwSarahEmm: it works, thx!!
10:06.24delox99because it covers a larger teritory
10:06.30SarahEmmdelox99: okay...
10:06.42SarahEmmdelox99: i meant as in 'it works anywhere' because it's kind of a last-resort technology
10:06.50SarahEmmyou wouldn't use it if you had some kind of land-based internet available
10:06.50delox99we need to get send and receive data in the middle of mexico
10:06.54SarahEmmokay
10:06.57SarahEmmlook at starband
10:07.04SarahEmmbzbw: np! :)
10:07.08SarahEmmwoo, i'm useful. :)
10:07.17delox99ok ill have a look
10:07.29delox99is it better to use some kind of wifi?
10:07.40SarahEmmerr, wifi is short-distance tho
10:07.46SarahEmmwhat would you connect to with wifi?
10:07.55*** join/#asterisk Abbas (n=Abbas@203.81.216.47)
10:08.15delox99it s to terminate voip lines
10:08.24SarahEmmdelox99: you don't want sat for voip lines.
10:08.29SarahEmmyour latency will be awfl
10:08.34SarahEmmmuch like standard telco over sat
10:08.40delox99sent from US to Mexico
10:08.56SarahEmmyou're looking at 1s or so RTTs
10:09.02*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
10:09.03delox99what?
10:09.15delox99sry im pretty newbie...
10:09.22SarahEmmdelox99: sat internet has about 1 second or so round trip time on packets.. high latency.
10:09.32SarahEmmspeech takes too long to get to the other end over sat, it makes calls really hard to do
10:09.33delox99ok
10:09.40SarahEmmyou likely don't want to run VoIP over it.
10:09.49*** join/#asterisk duckz (n=duckz@193.192.46.26)
10:10.02delox99ok so what would you suggest
10:10.48delox99?
10:11.00SarahEmmdelox99: you can't get any kind of PRIs or anything?
10:11.03fourcheezedelox99: two tin cans and a long piece of string?
10:11.40delox99we cant aford that length of string!
10:11.50emdubman, 7960's with sccp firmware on them is enough to make you commit suicide
10:11.55fourcheezedelox99: exactly how long are you talking?
10:11.57emdubatleast when trying to upgrade them
10:11.58emdubheh
10:11.59SarahEmmdelox99: then you don't want to look at what sat costs :)
10:12.07fourcheezedelox99: ok, not "exactly"
10:12.08delox99ah ok
10:12.17SarahEmmdelox99: you can't get PRIs there?
10:12.18fourcheeze1 mile, 10 miles, 100 miles, 1000 miles?
10:12.55*** join/#asterisk Alystair (i=Alystair@CPE000d88a7a3b5-CM00407b8794db.cpe.net.cable.rogers.com)
10:13.13delox99i would have to check first
10:13.30delox99but im afraid i could be expensive
10:13.36SarahEmmerr
10:13.36delox99it*
10:13.38SarahEmmsat will be expensive too
10:13.43SarahEmmand low quality
10:13.48*** join/#asterisk MatsK (n=mk@55.80-203-80.nextgentel.com)
10:14.02delox99because Telmex is too much of a monopol
10:14.28*** join/#asterisk cjk (n=cjk@80.92.64.103)
10:15.00cjkhi, i know it does not really belong here. but did anyone here use asterisk together with mysql together with federated tables over a long distance ip link?
10:15.57h3xfederated tables?
10:16.11fourcheezedelox99: point to point microwave link with line of sight can go a fair way, but it depends what you mean by "middle of nowhere"
10:16.24cjkh3x: yes, tables stored on a remote mysql server
10:17.07delox99sarahemm was saying that to use a sat i would need to be in the middle of nowhere
10:17.29delox99microwave link can travel how far?
10:17.57SarahEmmerr, i didn't say you *had* to be in the middle of nowhere
10:18.05SarahEmmi said you really want to AVOID using sat.
10:18.13SarahEmmso the only reason you WOULD use it is if you were in the middle of nowhere.
10:18.14h3xoh
10:18.21SarahEmmadding 1+ seconds of latency onto your voice conversation makes it hard to tlka
10:18.53h3xLots of "private line" services provided by telcos goes over microwave anyway without you knowing it
10:19.08h3xmicrowave dosent slow it down any more than wire or fiber
10:19.17delox99ah i thought ou said "hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka
10:19.17delox99<h3x>
10:19.25delox99hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka
10:19.25delox99<h3x>
10:19.32h3xmicrowave dosent add no damn 1000ms
10:19.34h3xsatellite does
10:19.37delox99oops sry im newbie with mirc
10:20.01delox99trying to paste sarahemm answer
10:20.13delox99but pressing ctrl-v does:
10:20.14delox99hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka
10:20.14delox99<h3x>
10:20.15delox99hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka
10:20.15delox99<h3x>
10:20.15delox99hEmm> adding 1+ seconds of latency onto your voice conversation makes it hard to tlka
10:20.16delox99<h3x>
10:20.59delox99ah i see as if i would be in the middle of nowhere
10:21.35SarahEmmdelox99: umm...... if you're having trouble with mirc, are you sure you're ready to tackle a VoIP deployment away from the city?
10:21.37h3xwhat difference does that make
10:21.42h3xhaha
10:21.47h3xhes gonna do TCoIP
10:21.50h3xTin Can over IP
10:21.53SarahEmm:)
10:21.59delox99hehe
10:23.03delox99i never really learned all the little mirc tricks like real chat machines =)
10:23.16h3xchat machines?
10:23.23h3xare you in afghanistan or what
10:23.30delox99no in Canda
10:23.33delox99Quebec
10:23.38h3xI figured you would say romania
10:24.08delox99i just dont speak english as good as i speak french
10:24.24h3xi worked in quebec for a little while
10:24.28h3xlike, 3 weeks
10:24.42delox99where you from?
10:24.48h3xi live in las vegas
10:24.58delox99hehe
10:25.09delox99where were you working?
10:25.14h3xwhen i was up there, i was fixing some peoples porn sites and gambling sites
10:25.22h3xat teleglobe colocation
10:25.30delox99nice
10:26.02delox99fixing server issues or coding issues?
10:26.25h3xwell all of their stuff was screwed up
10:26.33h3xthey had a bunch of russian programmers up there
10:26.37h3xi was just fixing sysadmin problems
10:26.49delox99ok
10:27.13h3xi got a new russian girl every week
10:27.13h3xhaha
10:27.33delox99on pictures?
10:27.38h3xno i mean in person
10:27.44delox99hehe
10:27.51delox99live from russia
10:28.05h3xthey all move to canada coz its hard to get a visa here
10:28.05*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
10:28.10delox99they were shipping em by fedex or what?
10:28.22h3xand then eventually they move to the US after they have canadian citizenship
10:28.34delox99ah yeah i see
10:28.46h3xwell, everybody except my friends that were running that business
10:29.04h3xthe guy's wife was a smart ass bitch to the immigration folks
10:29.08h3xand got them booted out to russia
10:29.13h3xi havent heard from him since
10:29.29delox99he was russian too?
10:29.37h3xyeah
10:29.59h3xbefore i knew him, he got rich running zooporn.com
10:30.01h3xhahahaha
10:30.27delox99and you met him on mirc? or while surfing one of his sites?
10:30.36h3xbut he screwed up... he registered the domain in his old college professor's name coz he didnt want people tracking him down
10:31.02h3xso when he moved from uunet to teleglobe, he simultaenously lost his email account the domain was registered under, so he couldn't move it over
10:31.32h3xno, somehow he was a partner of some guys i was working with here in vegas on some other projects
10:32.08voipjoyi have partly E1 channel only 14 instead of 30! how i disable the not used chanel that zapata will not handle it and not manage it?
10:32.40delox99bad for him
10:33.04h3xheh
10:33.14delox99yeah so microwave would do it
10:34.16*** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net)
10:34.27moverhi
10:34.41moveranymone here experienced wit t38 stuff in HEAD?
10:34.46delox99we want to bring voip termination in mexico without using telmex infrastructure
10:35.42delox99we are just looking into some solutions so we can hire more professional people to do the job better than me
10:35.54h3xhahahahha
10:36.11h3xthat is a good way to get killed
10:36.19delox99hehe
10:36.25delox99i know
10:36.31h3xtelmex gets really pissed off at grey market minutes
10:36.44delox99hehehe
10:36.58delox99you didnt see me coming?
10:37.08SarahEmmdelox99: wait... termination?
10:37.08delox99the market there is just incredible
10:37.11*** join/#asterisk Marcel-AS16215 (i=Marcel-A@gic-msg-exc-01.genotec.ch)
10:37.19SarahEmmdelox99: where are you dropping it onto the PSTN then, if you're not using telmex?
10:37.31h3xTCoIP
10:37.33h3xim tellin ya
10:37.38h3xthem bean counters love it
10:37.52Marcel-AS16215Hi to all, anyone a idee to fix this -> Nov  8 11:08:24 NOTICE[8180]: chan_sip.c:10270 handle_request_invite: Failed to authenticate user Marcel<sip:993@192.168.254.230>;tag=AIC99B37F7-ECB1BE2E066724AC ?
10:38.29SarahEmmMarcel-AS16215: Genotech Internet Consulting? :)
10:38.33*** join/#asterisk KriS83 (n=KriS@212.202.141.92)
10:38.36KriS83Hi
10:38.38SarahEmmMarcel-AS16215: sounds like your authentication info is wrong, password or such
10:38.51delox99we will use pstn for delivery but we ll see once there
10:39.07delox99first lets bring the pipe =)
10:39.29SarahEmmdelox99: err... okay. you'll still have to use telmex for the PSTN connection :)
10:39.32Marcel-AS16215SarahEmm yes :-)
10:39.39delox99we could distribute using wifi or something once there
10:39.39SarahEmmand i'm not sure where you're going to get a pipe :)
10:39.45h3xtheres plenty of competition in mexico already
10:39.49h3xyou just need to go shopping
10:39.52SarahEmmdelox99: err.. but you still have to connect to the PSTN, wifi won't help you there
10:40.04SarahEmmMarcel-AS16215: :)
10:40.23delox99true i need to make more research
10:40.29h3xyou havent been on #cisco apparently
10:40.30h3xheh
10:40.31delox99i was starting here =)
10:40.49SarahEmmh3x: heh, i used to hang out there all the time :)
10:40.53h3xdammit irc is always a last resort! :)
10:41.07h3xSarahEmm: Yeah i ran across your nick in an irc log when i was going a google search
10:41.09Marcel-AS16215SarahEmm but wehre is my Password wrong ? i think all my settings sut be okay, this comes only if a add tow connection my my Company PBX
10:41.12h3xbut i dont remember if it was that or some other program
10:41.14SarahEmmh3x: heh, i show up all the time :)
10:41.14h3xlike
10:41.21h3xh-sphere maybe
10:41.23h3xummmmm
10:41.38h3xwhat the hell was it was looking for and you replied to somebody and i was gonna ask you about it...
10:41.48h3xer I was looking for
10:41.58SarahEmmh3x: oh?
10:42.04*** join/#asterisk ComputerWarm (n=dan@rddrpx29-port-20.dial.telus.net)
10:42.37delox99i ve been looking into asterisk stuff for only like 4 months but i see he big potential
10:42.50h3xHmmm Maybe it was mythtv?
10:42.53Marcel-AS16215SarahEmm can i can come quick privat to discus this maybe you can give me the right hint that i dont see if i post you my Settings without username/password.
10:43.02ComputerWarmQuestion anyone here using A2Billing?
10:43.52delox99i m running an Asterisk server and already have a couple clients using sip phones
10:44.00lehelComputerWarm, i'm trying to..
10:44.22delox99now i m looking for a way to connect my server to pstn lines
10:44.23ComputerWarmlehel have you been able to get logged-in?
10:44.31lehelyap
10:44.32delox99i found that today in here
10:44.45ComputerWarmlehel oh... i wonder what i did wrong then
10:44.47SarahEmmh3x: could have been, i love mythtv :)
10:45.00SarahEmmMarcel-AS16215: umm, can't really help one-on-one right now sorry. have other stuff going on
10:45.02SarahEmm(am at work)
10:45.13lehelComputerWarm, you r reading the pdf document?
10:45.15SarahEmmdelox99: get a PRI :)
10:45.25ComputerWarmlehel no i have been reading the html doc
10:45.39h3xi cant get my damn pvr-350 to work
10:45.47delox99i would like to but the price scares me :/
10:45.56h3xwhen the kernel module is loaded it bombs dmesg with tons of APIC errors
10:46.00h3xive got HT on
10:46.15lehelComputerWarm, it is the same huh;) which dbtype u r using? [pgsql or mysql] ?
10:46.27remibrevalsarahEmm, yes I look at dialplan exemples in Wiki and I made http://pastebin.ca/27885, but still when I answer, It hang off instead of answering :-(
10:46.27delox99i ll use a provider instead
10:46.29ComputerWarmpsql
10:46.32ComputerWarmpgsql
10:46.55Marcel-AS16215SarahEmm okay
10:47.19delox99what prices should i look for when shopping for voip termination resellers for US/Canada?
10:47.20lehelComputerWarm, i succeded with mysql, never tried with pgsql
10:47.34delox99i saw 0.02US / min
10:47.38delox99is that good
10:47.43delox99or expensive
10:48.04*** join/#asterisk TK9 (n=Administ@p54B29019.dip0.t-ipconnect.de)
10:48.22ComputerWarmlehel maybe i should be trying mysql.
10:48.26*** part/#asterisk TK9 (n=Administ@p54B29019.dip0.t-ipconnect.de)
10:48.27*** join/#asterisk mmmToop (n=chatzill@196.31.11.194)
10:48.30h3xSarahEmm: Well you win, you have a bigger mess on your desk than me
10:49.05ComputerWarmdelox99 that would depend on how much volume u have
10:49.08lehelComputerWarm, ' should work as well with pgsql too
10:49.32ComputerWarmlehel i think once i get this error figured out in there script. i should be on my way.
10:50.19AlystairWait so are all these systems linux only? :O
10:50.28mmmToophi...not sure if you guys can help...we are using swissvoice ip10s's & it sounds like we are talking under water
10:50.34mmmToopthey were working fine & then stop working?
10:50.37mmmToopany thoughts?
10:50.52SarahEmmh3x: lol. old picture, it's better now :)
10:51.03mmmToopit has happened to two phones now...firmware?
10:51.12h3xthat sucks
10:51.21h3xgod damn it, there was a specific issue you solved on mythtv
10:51.22h3xand i cant find it
10:52.30AlystairOk this is rediculous, 90% of the offsite links from the informational wiki are half ass pages with adverts all over them and the content isn't what it is supposed to be
10:53.16SarahEmmh3x: ahh, i use a bttv card and software-encode
10:53.26cryzeck<
10:53.28ComputerWarmi don`t know Alystair the wiki works for me... for the most part
10:53.38h3xyeah my bttv card works fine
10:53.40SarahEmmmy current plan is to move to a PVR-500 as soon as i can get an external box to decode captions before it hits the PVR-500, as the card won't do it for me
10:54.11h3xknoppmyth is garbage
10:54.22h3xi ended up using fedora 4
10:54.32ptiggerdineremibreval, that's one huge dial plan
10:54.35ptiggerdinenice work though
10:54.41AlystairComputerWarm, no I mean it links to all these offsite places for general information
10:54.53h3xi think im gonna switch to via epia with a couple bttv cards :P
10:54.59ComputerWarmoh ok. is there a problem you are having?
10:55.55mmmToopso nobody has any ideas about the swissvoice phones why I sound like I am 10 000m under the sea?
10:55.57SarahEmmh3x: heh, you'll never get enough CPU for it :)
10:56.02SarahEmmh3x: that's why i only hsave one tuner now
10:56.14h3xthe via has mpeg4 acceleration built in
10:56.19h3xtheres a dual via cpu mobo
10:56.21SarahEmmerr
10:56.24SarahEmmmpeg4 decoding
10:56.28SarahEmmnot encoding, right?
10:56.31h3xi thought it did both?
10:56.36AlystairYeah, where can I find a link to a no-crap website which describes setting up a VOIP system for small companies (everything from outbound calling to a PBX like system)? :\
10:56.39SarahEmmmyth doesn't support it, even if it does :)
10:56.43SarahEmmand you'll max out your PCI bus
10:56.44h3xyeah it does
10:56.47SarahEmmerr
10:56.48SarahEmmit does?
10:56.51h3xyes
10:56.51SarahEmmfor encoding?
10:56.54h3xunichrome
10:57.08h3xthey have special setup instructions for it
10:57.13SarahEmmurl?
10:57.27h3xwhich distro? heh
10:57.34SarahEmmn/m
10:57.39rkingAlystair: well, that's a very good question
10:58.20h3xim trying to figure out some way to make these things "cheaper" so i can sell them
10:58.42Alystairand even then, when I try clicking on a website link like packet8's, when it should go to packet8.com (duh) they go to some reseller page or something (not direct from wiki)
10:58.44rkingAlystair: if such a thing doesn't exist, it would be a valuable contribution to the voip-info.org wiki (I want such a document myself)
10:58.59SarahEmmh3x: ahh :)
10:59.01h3xI guess PVR-500 is a good option
10:59.04SarahEmmyeah
10:59.07SarahEmmi have one, works great
10:59.10SarahEmmexcept no CC support
10:59.10h3xbecuase most mini-itx cases only have one pci slot thats usuable
10:59.15Alystairok, here's a question that should work here then
10:59.15SarahEmmyeah, mine does
10:59.20h3xwith some place to fudge that extra bracket in
10:59.39AlystairIf I setup asterix at the office, can I plug it in to our own backend system (php/mysql) to give people information over the phone?
10:59.51SarahEmmure
10:59.52h3xits times like this
10:59.55SarahEmmerr sure even Alystair
11:00.00h3xi wish mark didnt release asterisk to the world
11:00.00h3xhah
11:00.17AlystairI would need someone to code some sort of bridge for me right :\
11:01.30SarahEmmAlystair: yep.
11:01.40SarahEmmobviously, as asterisk can't include support for some custom backend you have ;)
11:02.17Alystairduh, even a PHB like me knows that :O
11:02.23h3xit does with some magic_witches_brew.so
11:02.36mutilatoranyone noticed problems with 2.6.14 kernel
11:02.57h3xAlystair: asterisk dialplan has a MYSQL() command to do database queries
11:03.05h3xor you can write AGIs using whatever language you want
11:03.09Alystaircool
11:03.33h3xmythtvs interface needs some work
11:03.35SarahEmm<-- NOCstrich
11:03.38emdubhmm so i finally got my 7960 up and working and its registering properly etc but when i dial an extension (one that just goes to voicemail) i see asterisk playing sounds but hear nothing on the phone
11:03.43*** join/#asterisk ful|work (n=fulgas@209.8.233.205)
11:03.50emdubsame extension works on my soft phone when i call it... what gives?
11:03.56ful|workhey
11:04.10h3xi dont understand why they made it so when you exit "watch tv" it quits recording
11:04.42h3xi guess maybe because its concept of multiple frontend, multiple backend dosent go with the tivo monolithic concept
11:05.17h3xfast forward is broken
11:05.22h3xwhen it hits the end
11:06.17Alystairguh
11:06.24Alystairwhere are the voip providers in Canada :\
11:06.32SarahEmmAlystair: whereabouts?
11:06.44ComputerWarmAlystair are you looking for termination? or dids
11:06.48AlystairOntario
11:07.16SarahEmmAlystair: what area code(s)?
11:07.21ComputerWarmare looking to make long distance calls or have people call you via voip?
11:07.42AlystairEverything.
11:07.50Alystairincluding 1-800# potentially
11:08.04SarahEmmAlystair: i use voctel.. there's lots of providers tho.
11:08.04Alystair416 area code
11:08.06SarahEmm~rtfw
11:08.07jbotit has been said that rtfw is http://www.voip-info.org, the only place to get any real answers about * and it's many, many uses.
11:08.07SarahEmmheh, me too
11:08.12SarahEmm<-- currently in downtown toronto
11:08.18AlystairHaha
11:08.19h3xhahahahhaah rtfw
11:08.21*** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net)
11:08.27SarahEmmthe wiki has a provider list
11:08.28AlystairSarahEmm, 151 Front St.?
11:08.28SarahEmmin canada
11:08.31zgorhi people!
11:08.39SarahEmmAlystair: not quite. one street north and slightly east :)
11:08.49AlystairHaha alright, I'm in Toronto.
11:08.50SarahEmmi might be able to see 151 out the window tho :)
11:09.28SarahEmm:)
11:09.35AlystairThe jist is, my dad's starting a business and I really want to save him cash on the phone stuff (lots of sales folk will be going all over Canada/USA) and make it easy for us to connect to the back end
11:09.56SarahEmmalright
11:09.59AlystairI'm already taking care of the office network and I'm interested in putting more hardware on the rack
11:10.00h3xhaha
11:10.54SarahEmmAlystair: okay.
11:11.02SarahEmmAlystair: http://www.voip-info.org/wiki/view/VOIP+Service+Providers has a list of providers
11:11.27h3xman
11:11.30h3xi am on there
11:11.33h3xand some guy calls me today
11:11.36h3xasking for 1 DID
11:11.44h3xand im like
11:11.45h3x^#!^%#!%!#%#!
11:12.09h3xoh
11:12.15SarahEmmh3x: if you're on there and don't have a website with info or anything, how are poeple to know?
11:12.16h3xim just on voip providers b2b and business
11:12.23h3xmy website is in there
11:12.29h3xand every single page has a contact form
11:12.40SarahEmmahh okay h3x
11:13.16h3xim suprised they dont charge for em
11:13.27h3xive gotten about 120 leads off voip-info in the past few months
11:13.40h3xabout 20% of them were major accounts
11:13.53emdubok so... more details... calls to a DID number in * which then dials the 7960 via SIP works fine, audio in both directions, but if i dial an outside number from the cisco i hear nothing and the other side can't hear me... same thing if i dial an extension on * which just goes to VM... no sound... i think im using ulaw or something, could that be the problem?
11:14.02h3xwhats really damn funny though is that comm partners runs that site and they didnt list themselves
11:14.06*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
11:14.08puzzledmorning
11:14.14h3x*derrrr*
11:14.42h3xim gonna start reselling their did footprint since they dont seem to care to
11:14.47*** join/#asterisk gun84 (n=gun84@60.48.169.13)
11:15.05h3xcomm partners is across the street from me
11:15.15h3xand im working on getting a gigE connection installed between us
11:15.58h3xtheir web site looks like crap too
11:15.59h3xheh
11:16.44*** part/#asterisk gun84 (n=gun84@60.48.169.13)
11:18.37*** join/#asterisk BerndR (n=broessl@193.83.150.54)
11:19.02moveranymone here experienced wit t38 stuff in HEAD?
11:21.23*** part/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com)
11:21.37Alystairh3x, I may not know much about voip but I'm a pretty spiffy graphic/web designer :D
11:22.06Alystair*... proproses
11:23.08*** join/#asterisk [MUPPETS]Gonzo (i=gonzo@80.69.47.16)
11:26.33ptiggerdineAlystair, do you deal with coperate identity stuff?
11:26.46ptiggerdinesorry bout the spelling
11:31.31AlystairSure thing
11:31.52Alystairlogos, business cards, letterheads, pamphlet material, etc.
11:33.14AlystairThough I've been pretty busy lately, gotta make a sign for some guys office, as well as some of those sticky-advertisments-on-glass type of things
11:33.38*** join/#asterisk Cherebrum (i=fucker@fucker.fuckthefuckingfucks.com)
11:33.47Alystair2 websites and editing some written material *yawn*
11:35.27*** part/#asterisk Cherebrum (i=fucker@fucker.fuckthefuckingfucks.com)
11:42.11remibrevalArff, why could Aswering machine and picking up a call macros work from local call and not from PSTN call ? http://pastebin.ca/27885
11:42.50remibrevalI'm going crazy on what's happening !
11:49.00*** join/#asterisk coppice (n=chatzill@105.201.17.210.dyn.pacific.net.hk)
11:50.12KriS83Anyone here that uses Junghanns.Net quadBRI cards with current * Beta2?
11:53.50*** part/#asterisk lubomier (n=lubomier@217.118.109.179)
11:55.23syleanyone have a PRI line?
11:55.31syleor voip site
11:55.44*** join/#asterisk mazza[W] (i=mazzanet@unaffiliated/mazzanet)
11:57.21Alystairwow Voxter looks good
11:58.05mazza[W]has anyone managed to successfully compile the intel ipp g729 codec?
11:58.51*** join/#asterisk danzig (n=chatzill@ruc-kj-013.ruc.dk)
12:00.53KriS83Anybody got BriStuff compiled for * 1.2.0-beta2?
12:01.20*** join/#asterisk thewiizard (n=nick@host217-34-132-179.in-addr.btopenworld.com)
12:04.00puzzledKriS83: I haven't heard that the patch has been updated for 1.2b2
12:04.29puzzledmazza[W]: yes people have done it before. search voip-info.org for g729 codec
12:04.31KriS83:/
12:04.48KriS83puzzled, major differences between beta2 and CVS Version?
12:05.13KriS83I mean like is the CVS the next beta3 or maybe even final 1.2.0?
12:05.13puzzledI would say about 90 commits so I guess yes
12:05.40mazza[W]puzzled: i was hoping for someone awake in here
12:05.48mazza[W]i'm getting compile errors :(
12:06.06puzzledmazza[W]: do the search. there is a site that offers precompiled codecs
12:06.24puzzledKriS83: afaik cvs is leading up to 1.2
12:06.46*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
12:07.27*** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net)
12:07.43danzigEHLO * Gurus :-)
12:08.40danzigUsing * as a SIP client, when it gets 'Failed to authenticate on REGISTER', it tries again after 20 seconds. Is there any way to increase this timeout without recompiling?
12:10.37*** join/#asterisk robbie2 (n=rob@CPE-60-231-141-131.qld.bigpond.net.au)
12:10.42KriS83puzzled, thank you, I will try the CVS then.
12:12.43KriS83Carefull the floor is hard ;) might knock your head ;)
12:12.52puzzledKriS83: while you are at it, some useful patches for CVS are #5374, #3599, #4252
12:13.20KriS83ok, if you told me where to get those, I will apply :)
12:13.26KriS83Sorry, but I'm new to this
12:14.16puzzledbusg.digium.com and #3599 does not work right now but (hopefully) should be updated soon
12:14.22puzzledugh, bugs.digium.com
12:15.05robbie2my voicemail messages that come through in email have the ends clipped
12:15.29darkskiezrobbie2: probably be your audio player
12:15.34robbie2mplayer
12:15.39darkskiezexactly
12:15.42robbie2oh
12:16.56*** join/#asterisk c0w (n=c0w@staff-ns50-3.as25178.net)
12:16.59c0whello all.
12:17.06robbie2hmm
12:18.00c0w?/
12:19.47robbie2yep
12:19.58*** join/#asterisk Inkubot (n=inkubot@200.75.4.7)
12:20.05robbie2darkskiez: well thats a bit of a bugger
12:20.17robbie2wav file generated on linux box cant play in linux mplayer
12:20.46KriS83puzzled, thx
12:20.48darkskiezi can play in linux, but mplayer is a bit shit
12:20.52darkskiezi=it
12:22.40c0wdoes anyone know about the digium wildcard te4100P cards, and if the new firmware that digium has made is for this card?
12:22.47robbie2hmm
12:23.00robbie2i see gcc3.0 works a bit better than 2.95
12:23.09darkskiezhow do you see that?
12:23.43robbie2cause its building now
12:25.06thewiizardgot a circuits are busy error
12:25.18puzzledc0w: ask digium?
12:25.21thewiizarddoes that mean that it cant stick the call out thru the provider
12:26.03c0wi am just about to send an email just thought i might try here first see if any engineers were about.
12:35.47robbie2umm
12:38.30*** join/#asterisk RoyK (n=roy@55.80-202-161.nextgentel.com)
12:41.34*** join/#asterisk Thumann (n=brianm@80.163.152.30)
12:41.50*** join/#asterisk manelideo (n=mdsilva@194.117.41.7)
12:42.29manelideohi ppl! is anyone using AMP ??? I have this error when I type /usr/sbin/amportal start
12:42.47manelideoSETTING FILE PERMISSIONS
12:42.47manelideoPermissions OK
12:42.47manelideoSTARTING ASTERISK
12:42.47manelideoAsterisk is already running
12:42.47manelideoSTARTING FOP SERVER
12:42.48manelideo-bash: line 1: 17528 Killed                  /var/www/html/panel/safe_opserver
12:42.50manelideo-bash: line 1: 17591 Killed                  /var/www/html/panel/safe_opserver
12:42.52manelideo-----------------------------------------------------
12:42.54manelideoThe FOP's server (op_server.pl) could not start!
12:42.56manelideoPlease correct this problem
12:42.58manelideo-----------------------------------------------------
12:43.24manelideoI tried to google but it didnt solved my problem
12:43.57Thumannhi guys, i'm wondering if asterisk is able to 'lift' the task i present.. can it (if we use more than one server naturally) handle +8.000 users? And if so, do any of you have some reference stories ?
12:44.04Inv_arpmanelideo: pastebin.ca for multple line paste
12:44.40c0wthumann, what are you wanting to run it as just SIP or what.
12:45.05Thumannc0w: afaik, just sip
12:45.12pooh_join #smeserver
12:45.12c0wthen i wouldn't recommend it.
12:45.19pooh_:-) wrong windows
12:45.25Thumannc0w: SER instead?
12:45.27c0wuse a sip proxy (SER) or something
12:45.38c0wand then have asterisk as your gateway / sevices box
12:45.43wasimThumann: ser + *
12:45.45c0wyep
12:45.57c0wworks very well together. =0
12:46.49Thumann:>
12:47.00Thumannjust one * ?
12:47.04thewiizardsounds interesting
12:47.30wasimThumann: nyet ... multiple ser, multiple * is always a better proposition
12:47.39Thumannnaturally
12:47.49Thumannbut are there any limitations to *, and ser?
12:48.06Thumannas in.. max, 600 users pr. box.. etc.. or is it a question of hw sizing?
12:49.47puzzledwasn't there an issue with MWI when using ser+*?
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12:52.22ful|workpuzzled: i got my ser+* with MWI
12:53.50manelideohmmm.... and with the browser AMP, when I want to add a new extension: "This module requires access to the Asterisk Manager. Please ensure Asterisk is running and access to the manager is available."
12:54.04manelideocan anybody help me?
12:54.14thewiizardmanager running?
12:54.24thewiizardamportal start
12:54.49manelideoyes...I did that, but I got the error above (the FOP server)
12:55.07robbie2did i see mention of a web based voicemail interface ?
12:55.11manelideoor it doesnt matter if the FOP server does not start
12:57.34danzigUsing * as a SIP client, when it gets 'Failed to authenticate on REGISTER', it tries again every 20 seconds. Is there any way to increase this default without recompiling?
12:58.03thewiizardmanelideo not really
12:58.16thewiizardrobbie2 yeah new amp release has it
12:58.34thewiizarddanzig its all about the sip timing
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13:02.03robbie2hmm
13:02.18robbie2what plugin this vmail.cgi require
13:03.43danzig>thewiizard>Yes... But it is up til * as the client what it does... Since the problem is a wrong password, it seems to be a waste to try every 20 sec.
13:03.45Thumannbut are there any limitations to *, and ser?
13:03.46Thumannas in.. max, 600 users pr. box.. etc.. or is it a question of hw sizing?
13:04.03JamesDotComsip is extremely scalable
13:05.43coppicescalable like a fortress under attack? :-)
13:06.01JamesDotCommeaning?
13:06.33coppicescalable is a much abused and seldom meaningful buzzword
13:07.30JamesDotComwell we're talking about numbers of users a network can handle
13:08.00thewiizardnetwork can handle = * / x + y
13:08.00JamesDotComand i see that being a scale, for example 1 - 10000 users
13:08.26JamesDotComi know for a fact that a properly designed sip network will be the least of your scalability worries when dealing with 10000 users
13:08.27coppicewhich has almost nothing to do with SIP. SIP is a signalling protocol - i.e. it hardly does anything, and should definitely be able to handle a lot of user on a small box
13:08.50JamesDotComthat's what i'm saying
13:08.58thewiizardits all about hardware
13:09.03JamesDotComit's too open a question
13:09.08JamesDotComtotally depending upon the environment
13:10.21JamesDotComdoes that answer your question Thumann ;)
13:11.48Thumannit does :)
13:11.49JamesDotComthere's a billion factors... codec selection (quality:bandwidth), concurrent calls, features, basically, SIP and almost any SIP proxy will have no issue with that amount of users, but the rest will depend on other things
13:14.57coppiceThumann: saying "only SIP" says absolutely nothing. the real work is in handling the audio. if you want a box to act as a soft-switch, never seeing any of the audio, a small box might support 100K calls, unless the average call is rather short. If the audio goes through your box don't expect too many concurrent calls. If you boxes transcodes, using heavyweight codecs like G.729, except even...
13:14.59coppice...less concurrent calls.
13:17.15coppiceI shouldn't type while nearly asleep :-\
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13:21.42Inv_arpopps wrong cable
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13:27.32thewiizardgoddamn
13:28.49tzangerhahaha
13:29.30Alystairman, where are the free peanuts and beverage
13:29.30tzangerVideotron (Quebec cable company) is suing Bell ExpressVu (Satellite TV company) for not doing enough to protect themseves from piracy
13:29.30tzangergoddamn ducks?
13:29.30AlystairVideotron's still alive?
13:29.30tzangerapparently
13:30.12tzangerhttp://www.cbc.ca/story/business/national/2005/11/07/videotron-051107.html
13:30.55AlystairI gotta call up Bell and Rogers and nag them for quotes on a good business line
13:31.15tzangerheh
13:31.15tzangerwe just use resold powerdsl here
13:31.17tzanger(Ikano)
13:31.29tzangerit works well and we have excellent connectivity to our networks at 151 front
13:31.35Alystairwhat an f'n rip off
13:31.47tzanger$80/mo ain't that bad
13:31.48tzangerhonestly
13:31.51Alystair$199.95 a month for 5mbit
13:31.55tzangerfor what?
13:31.59Alystairand 640kb up
13:32.01Alystairon rogers HAH
13:32.02tzangercable?
13:32.05Alystairyeah
13:32.05tzangerjesus
13:32.14tzanger$80/mo for up to 8meg down and 800k up
13:32.19tzangerno caps
13:32.21Alystairthat's sexy
13:32.32tzangerthat's standard ADSL.  static IP too even
13:32.35tzangerno port 25 block
13:32.38Alystairok, where's the info
13:33.20jake1932i get 3/768 now for $30
13:33.44Alystairwhat's the damn deal with upload speeds anyway
13:33.48af_I am getting this stuffs randomically: PRI: !! Got a UA, but i'm in state 1 (quadbri + bristuff). any hints?
13:33.59Alystairwhatever happened to the SDSL :(
13:35.21jake1932you can prob still get SDSL - but it'll cost a little more :o)
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13:36.22thewiizardyo i cant get this damn amp wired to my proxy
13:36.26thewiizardany special settings i need/
13:36.49thewiizardtrying to get away with just using authuser=, username=, host=, secret= and type=
13:36.54tzangeryeah I hate 800k up it's just not enough
13:37.08tzangerI'd prefer 1.5meg symmetrical
13:37.21tzangeralthough 8meg down does have its advantages
13:37.58puzzledanyone seen a compile error with res_config_odbc with cvs HEAD from an hour ago?
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13:39.04demetriohi
13:40.50demetrioso I have this account with a VoIP provider connected to a "real" phone number. I set up asterisk to register as a client to accept incoming calls, but when I try to call from a phone with number X asterisk will say "Failed to authenticate user X". Why is this?
13:43.24thewiizardgot the right password?
13:43.47demetriosure, but that's not the point
13:44.05demetrioshould I set up SIP accounts for every phone number in the world?
13:44.08file[laptop]registration and accepting calls are two different things
13:44.15bonhm
13:44.23bonwhy do all of my calls fall into one context?
13:44.24mutilatoranyone have a mrtg => cacti config converter?
13:44.55thewiizardbon have u told them to?
13:44.56demetriowell, I thought that the register directive in sip.conf is for incoming calls
13:45.18thewiizarddemetrio Failed to authenticate means you failed to authenticate
13:45.27thewiizardcheck ur account settings
13:45.42demetriobut that doesn't make sense
13:45.58KriS83When does Digium have their working hours?
13:46.01file[laptop]Registrations only tell the other side, "here I am if you need to send calls to me"
13:46.05demetriothe user whose authentications fails is the pone number from wich I try to call
13:46.10demetrioright
13:46.13file[laptop]it's up to you to still configure your side to accept the calls
13:46.18enemyI have a dual E1 card which I`ve connected to the telco and to other equipment. My issue is that when I transfer calls from one E1 to the other, it seems to work nicely when it's isdn originating traffic. If it's an analogue connection (modem) which is going to be bridged, then I get req transfer capability 3K1AUDIO instead of DIGITAL, and then it just gets  span 1 got hangup
13:46.28file[laptop]what you'll probably want to do is make a peer entry, with the IP address where calls will come from, and set insecure=very and context
13:46.35file[laptop]it'll match based on the IP and throw it into the right context
13:46.56thewiizardi allways find outgoing calls more fun to get going
13:46.57demetrioOk I'll try the insecure=very
13:47.05bonthewiizard: i don't know, outgoing calls do go to right context but how do i tell incoming ones to use other context?
13:47.22thewiizardbon: basically for your provider youll have a context defined
13:47.36thewiizardwhen calls are recevied by said provider the calls are placed into said context
13:47.40bonah
13:47.42bonlemme see..
13:47.47thewiizard[voip_provider]
13:47.49thewiizardhost=blabla
13:47.52thewiizardsecret=blabla
13:47.55thewiizardusername=blabla
13:48.06thewiizardcontext=[whatyouwant]
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13:48.11thewiizardi think its context
13:48.19thewiizardmaybe something slightly different
13:48.20demetriofile[laptop], thanks a lot, that worked
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13:48.53demetrioI was missing the insecure=very, now I'll try to understand what that directive does
13:49.21file[laptop]causes asterisk to not challenge for user/password
13:49.31file[laptop]just does IP auth
13:50.26thewiizardtype=friend might be good with that also ;0
13:50.29*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:51.01file[laptop]not really
13:51.09file[laptop]type=friend just generates two entries in memory, a user and a peer
13:51.18file[laptop]and most providers don't do user authentication, which is what a user entry is for
13:51.27file[laptop]they want you to do IP auth, which you use a peer for
13:51.35thewiizardIP auth with sip
13:51.47thewiizardwhy not use h323
13:52.10file[laptop]I'm just going to... ignore... that
13:52.32Dr-Linuxi want to avail Conference call feautre, i define a room 1111 in meetme.conf and i put an extension in extention.conf exten => 5557,1,Meetme,1111
13:52.52*** part/#asterisk jake1932 (n=jake1932@pool-68-163-57-114.phil.east.verizon.net)
13:53.00bonthewiizard: :(
13:53.02bondidn't help
13:53.07boni thought it would
13:53.16bonjust looked at my providers entry in mysql
13:53.21bonhas context=whatiwant
13:53.27Dr-Linuxso when i dial 5557 at softphone, i doesn't work and i see on CLI > Nov  8 18:45:18 WARNING[8432]: pbx.c:1293 pbx_extension_helper: No application 'Meetme' for extension (default, 5557, 1)
13:53.28Dr-Linux<PROTECTED>
13:53.31bonbut it always gets written into cdr as what i don't want it to :)
13:53.47file[laptop]Dr-Linux: do you have zaptel installed, with a timing source?
13:53.47Dr-Linuxwhat could be wrong ?
13:54.02Dr-Linuxfile[laptop]: no sir
13:54.08file[laptop]then that's the problem
13:54.10file[laptop]Meetme requires it
13:54.22Dr-Linuxi'm only working on softphones
13:54.41file[laptop]Meetme itself requires it, you don't have a zaptel timing source - it won't work, if you don't have zaptel installed - it won't compile
13:54.49file[laptop]what you can do is try ztdummy
13:55.05file[laptop]instructions for making it and stuff are on the net, Google is your friend
13:55.17emdubgoogle <3
13:56.06Dr-Linuxfile[laptop]: whats easy way to install work for Meetme? ztdummy or zaptal ?
13:56.11emdubanyone familiar w/ cisco 7960 phones?
13:56.14*** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk)
13:56.14Dr-Linuxi don't have any hardware
13:56.53file[laptop]Dr-Linux: ztdummy is part of zaptel
13:57.37file[laptop]it's a dummy driver, uses the UHCI USB controller on a 2.4 kernel, and the RTC on a 2.6 kernel
13:58.09Dr-Linuxfile[laptop]: i have installed asterisk 1.0.9 , so should i need to install zaptal seperately ? or it should already exists ?
13:58.38file[laptop]it's separate
13:58.41Dr-Linuxas i don't have any hardware, but i need to avail Meetme feature
13:59.48tzangerDr-Linux: so use ztdummy or zaprtc
14:00.29file[laptop]tzanger: so what's your tagline today...
14:00.37file[laptop]bah
14:00.43tzangerhang on I didn't change it
14:00.58iCEBrkrhey hey
14:00.59puzzledanyone yet tried to compile cvs HEAD from a couple of hours ago?
14:01.04tzangerbetter?
14:02.16mutilatorcocacola is teh suck
14:03.08*** join/#asterisk psk (n=psk@golia.caltanet.it)
14:03.19Dr-Linuxi have cisco 79xx ip phone in US, i'm in pakistan .. i wanna access them to configure with SIP , how can i do ?
14:04.40*** join/#asterisk Einon (n=einon@ka.wa.hu)
14:06.12*** join/#asterisk frix (i=frix@p54A876EB.dip.t-dialin.net)
14:06.17*** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca)
14:06.58Dr-Linuxtzanger ?
14:07.06tzangerI don't know anything about cisco sorry
14:07.09frixdont know if you could read it before... i need some help with compiling chan_capi, which blows me up with 1mb error log
14:08.51*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:08.59Einonhoi! I sat up asterisk to put outgoing calls through an ISDN modem. From linphone I dial a number asterisk dials it and the called party receives the call. But from any other client (twinkle, kphone, hardphone) I tried to dial the same number, asterisk dials but the called party do not receive the call.
14:09.53thewiizardDr-Linux u gotta unlock it
14:10.04thewiizardupload new firmware SIP7.3 or 7.5
14:10.08thewiizardvia tftp
14:10.17thewiizardconfig and boom
14:10.23*** join/#asterisk PBXtech (i=nik@55.sub-70-218-24.myvzw.com)
14:10.23thewiizardsorted
14:10.33*** part/#asterisk PBXtech (i=nik@55.sub-70-218-24.myvzw.com)
14:11.15clive-frix use the sourceforge chan_capi
14:11.59enemyhow can I change from ulaw to alaw on my zap channels?
14:11.59frixclive: thank you, i'll give it a try
14:13.44queuetueI am setting up VoicePulse over IAX2  - outgoing seems to work fine, and I have made a few calls with it, but incoming, I get tri-tones, and "your call cannot be completed.  075T"  Can anyone shed any light into this error (does it come from asterisk, my outgoing provider, or from voicepulse?)) and how I might fix it?
14:14.01Dr-Linuxthewiizard: yeah but i'm far away for phone i'm in pakistan and the phones located in US , they are new i am the one who will do all ? what should i do ?
14:14.37tzangerqueuetue: have you registered to voicepulse?
14:14.44tzangerthey have pretty good documentation on their site last I looked
14:15.16frixclive: pretty much the same
14:15.23queuetuetzanger: Yes.  I am registered with voicepulse (I just mentioned I could make outgoing calls with them) and I have followed their instructions.
14:15.28frixbut a little less than 1mb :)
14:15.35tzangerno
14:15.40tzangerregister => user:pass@voicepulse
14:15.43tzangerso they know where to find you
14:15.51queuetuetzanger: Yes.
14:16.07*** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net)
14:16.14tzangerqueuetue: ok, and with iax2 debug turned on what do you see on the console (use pastebin) whey you try to call your DID from a cellphone?
14:16.20SpaceBasswhat is the standard vertical code to enable call recording?
14:16.22queuetueXXX:YYY@gwiax-in-01.voicepulse.com
14:16.28*** join/#asterisk asteriskgeeks (n=SIPdawg@pbxtech.com)
14:16.28asteriskgeeks<PROTECTED>
14:16.41queuetuetzanger: I don't believe I see anything happen at all - will try again, though.
14:16.54*** join/#asterisk PBXtech (i=nik@55.sub-70-218-24.myvzw.com)
14:17.05*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
14:17.37brad_msswtzanger: are you starting the asterisk console using the  -cvvvd  or similar ?
14:17.55tzangernah
14:18.09tzangerasterisk -vvvgc is what I do for testing and just -g for normal (using -vvvrc for connecting to it then)
14:18.49brad_msswthrow a d in there
14:18.58brad_msswfor some extra debugging statements
14:18.58tzangernah
14:19.02SpaceBassanyone...quickly... on the phone with sprint customer service and really need to record it... is there a way to do so mid-call?
14:19.02tzangerI never find the use for it
14:19.31iCEBrkrManage port.
14:19.44tzangerSpaceBass: park them and do up a dialplan real quick.  :_0
14:19.46tzangerI don't know how else
14:19.52tzangerif you don't already have it done
14:19.56brad_msswtzanger: uh, it could help you solve your issue ... most likely you'd see a message on the console stating that it couldn't find a certain context, or the context doesn't contain the extension it's looking for
14:20.01iCEBrkrtzanger: Can't you issue the zapbarge or whtever via the manageport?
14:20.07SpaceBasstzafrir im using AMP and think its in the dial plan, just cannot find the code
14:20.10tzangeriCEBrkr: no idea
14:20.14iCEBrkrI've done it before.
14:20.19iCEBrkrUse the manage port to turn on recording
14:20.21brad_msswhaha, tzanger should have been directed towards queuetue
14:20.32brad_msswthat's what I get for jumping in late
14:20.52iCEBrkrhttp://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Monitor
14:21.01iCEBrkrACTION: Monitor
14:21.01iCEBrkrChannel: SIP/x7062618529-643d
14:21.02iCEBrkrFile: channelsavefile
14:21.02iCEBrkrMix: 1
14:25.41morale<PROTECTED>
14:25.43moralenifty.
14:25.49queuetuetzanger:  Ok, I did get a message:  http://rafb.net/paste/results/VlYKTO29.html  And the [problem appears to be "No such context/extension" ... what context or extension is it looking for?
14:26.17tzangerqueuetue: in your type=user (or type=friend) for voicepulse, what conext are you specifying?
14:26.38queuetuetzanger: from-pstn
14:26.54tzangerok, and what does your [from-pstn] context look like (use pastebin)
14:27.10moralehow do i set the cost on a codec in asterisk? i want all my outbound calls to use the g.729 codec by default now
14:28.14*** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net)
14:28.44queuetuetzanger: I'm using AAH, so it's a bit spaghetti-like.  Essentially, I get the impression  need to set up a DID... Let me try that and get back to you all if i can't take it from here.
14:29.20tzangerqueuetue: well set up this did
14:29.33tzangerexten => _X.,1,NoOp(EXTEN is ${EXTEN})
14:29.41queuetuetzanger: I'm pretty new - didn't realize I had to (the zap lines "just worked". :) )
14:29.47tzangerqueuetue: :-)
14:31.07tzangerwe were all newbies at one time
14:34.45mutilatoranyone delt with call forward billing and call accounting?
14:34.51mutilatorhow do ya handle your cdr?
14:36.38Dr-Linuxtzanger: i installed the zaptal, and now nothing works, softphone can't dial any extension, "sip show users"  doesn't show any registerd user?
14:36.55Dr-Linuxwhat could be happend? i just zaptal and nothing else i face this happend :S
14:38.10Dr-Linuxis the sip users disabled when i installed zaptal or what ? :S
14:38.29tzangerDr-Linux: you really need to read up... you've been an asterisk user now for several months, you should know ho to do basic diagnostics by now
14:39.20*** join/#asterisk grimse (n=grimse@p5481E912.dip.t-dialin.net)
14:39.27Dr-Linuxi left long ago , i just started few days back ..
14:39.36moralehmm.. theoretically on my adsl connection at home.. i can run 8 inbound lines with this g729 codec.. hmm.. ideas start flowing
14:39.54Dr-Linuxeverything was ready and working for demo .. but everything is gone :( as i installed zaptal
14:40.36moralei need to find some chicks with no self-esteem and setup a 1-900 number.
14:42.05Rowterwhen connecting asterisk to a panasonic, you will have a cable from fxs of panasonic to fx0 of a tdm asterisk card?
14:42.20Rowterthats correct?
14:42.25|Vulutre|no
14:42.34|Vulutre|it would be FXS to FXO
14:42.43|Vulutre|you cant connect two FXOs
14:42.45mutilator..
14:42.48Rowterthats what I said
14:42.52|Vulutre|omg
14:43.00RowterFXS to FXO
14:43.02|Vulutre|yea
14:43.08RowterFXS panasonic  FXO asterisk
14:43.09|Vulutre|I literally climbed out of bed
14:43.12|Vulutre|yea you got it
14:43.16Rowterhehe, thanks
14:44.17*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivflm3.dialup.mindspring.com)
14:47.45ManxPowerRowter, you want TDM400P FXS into the Panasonic CO (aka FXO) port.
14:48.07ManxPowerMost PBX FSX ports do NOT support any way to tell Asterisk to hangup the line.
14:48.46ManxPower~fsxfxo
14:48.52ManxPower~fxsfxo
14:48.53jbotmethinks fxsfxo is An FXO port expects to receive dialtone and receive ring voltage.  An FXS port expects to provide dialtone and provide ring voltage.
14:50.11*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
14:50.16*** join/#asterisk BerndR (n=broessl@193.83.150.54)
14:51.11EinonHow is it possible that the very same asterisk with the very same SIP peer but with two different softwares (linphone and twinkle) works different. If I place a call from SIP to my cellphone via Asterisk's ISDN modem, from linphone my cellphone receives the call, from twinkle it does not.
14:51.33Einonany ideas
14:51.43moraleI JUST SWITCHED TO GEICO AND SAVED A BUNDLE ON CAR INSURANCE... argh! i hate those commericials
14:52.51BerndRpart
14:52.55*** part/#asterisk BerndR (n=broessl@193.83.150.54)
14:53.01*** join/#asterisk copantl (n=galel@63.245.93.138)
14:54.20mutilatorfor cdr_mysql
14:54.30mutilatorsetCIDNum changes the src field
14:54.35mutilatorbut how do i change the dst field?
14:55.17*** join/#asterisk tmccrary (n=tmccrary@68.78.185.254)
14:56.30*** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-206.hsd1.ut.comcast.net)
14:56.40ManxPowermorale, I hate them because they would not insure me.
14:57.14ManxPowermutilator, do a Goto
14:57.31gambolputtyThe speed racer geico commercial is funny
14:57.43tmccraryhey, I experience choppyness during phone calls with voip over a high latency connection (250ms). Is there some easy ways to tweak this? Jitter?
14:57.44Kattymishehu: come find me later.
14:59.47mutilatorManx: goto another extension?
14:59.49iCEBrkrtmccrary: Turn off your Gnutella and eMule clients! :D
15:00.12tmccraryhehe, bandwidth shouldn't be a problem. I was thinking maybe doing QoS
15:01.43gambolputtyanyone use Gentoo?
15:02.02iCEBrkrgenpooh!
15:03.42mutilatorah sweet
15:04.38iCEBrkrWhacked.
15:05.01iCEBrkrWe have an analog card in our PBX and my X100P keeps looping-- thinking the line is ringing.
15:06.33*** join/#asterisk alephcom (n=Miranda@207.34.97.130)
15:08.02Dr-Linuxany can tell me, how can i access Cisco 79xx ip phone remotely to configure with SIP ?
15:08.04*** join/#asterisk ian_k (n=ian@gateway.digium.com)
15:08.19iCEBrkrHow would you normally configure it?
15:09.19mutilatorthx manx
15:09.28wasimiCEBrkr: check your wire
15:09.29ManxPoweriCEBrkr, Are you SURE it's an analog port on the PBX?  i.e. can you plug a standard home analog phone into the port and have it work?
15:09.44iCEBrkrManxPower: Yeah.
15:10.00iCEBrkrManxPower: It's meant for that.  So you can plug fax machines and such into the PBX
15:10.23ManxPoweriCEBrkr, I don't know what to suggest.
15:10.31*** join/#asterisk Geert (i=geert@irssi/staff/geert)
15:10.36iCEBrkrI put a standard phone on it.  I got a dialtone.  Had to dial 9 to get out, but that's cool.
15:10.40iCEBrkrReally weird shit
15:11.01iCEBrkrThing is.  I'm not sure if it's the 1.2 zaptel stuff or what.
15:11.25iCEBrkrI had all this working with 1.0.3-- long ago when I first started tinkering with Asterisk.
15:11.51ManxPoweriCEBrkr, I've had X100P cards just top working before.  Make sure the card is seated in the socket.
15:11.57ManxPowertop == stop
15:12.03GeertDoes somebody have information about how to get the call duration through an IVR? Example: Customer calls IVR -> IVR pickups (starts the timer?) -> Users does his thing -> IVR forwards call to VoIP phone -> IVR closes call (stops timer) -> IVR inserts INSERT query
15:12.16thewiizardeeeeeeeeeeeee
15:12.45iCEBrkrManxPower: I'll do that, the machine has been transported a bunch of times.
15:13.11iCEBrkrI just wish I had access to PSTN.  I don't trust the shit pulled into this building.
15:13.24ManxPoweriCEBrkr, I know what you mean.
15:13.30olivier_<Geert> Have a look to the ANSWERED variable
15:13.48Geertolivier_: thanks
15:13.54iCEBrkrGeert: That's all part of the CDR
15:13.58*** join/#asterisk _T3_ (n=rposada@200.63.231.210)
15:14.16olivier_sorry ANSWEREDTIME variable
15:20.03*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
15:20.41lunkhow do i check if a call that's been placed is busy or not in a dial plan?
15:21.14RowterManxPower, whats the best combination to have some extensions to ASterisk from a panasonic?
15:21.19iCEBrkr${DIALSTATUS}
15:21.44lunkiCEBrkr: thanks
15:22.11Einonbye
15:22.12*** part/#asterisk Einon (n=einon@ka.wa.hu)
15:22.45ManxPowerRowter, there really isn't any good solution for that.
15:23.13ManxPowerRowter, But I can tell you that if you go Panasonic FXS -> Asterisk FXO you are going to have lines that don't hang up.
15:23.30RowterManxPower, they want to make voip calls, its better to add a simple ata then..
15:24.15RowterManxPower, I see.. thanks for telling me about that hangup issue..
15:24.30ManxPowerRowter, I speak from personal experience 8-)
15:24.55tzangerManxPower: panasonics don't signal disconnect well?
15:25.05*** join/#asterisk paryl (n=paryl@209.236.78.59)
15:26.09_T3_anybody knows who wants a job to modify dsp.c for callprogress ??
15:30.01parylwill one of the little analog adapters work for a fax machine?
15:30.37queuetuetzanger: BTW< I doubt you were thinking about it at all, but adding the DID fixed everything.  Thanks for your help.
15:31.00*** join/#asterisk shout (n=rcsw@host213-123-195-3.in-addr.btopenworld.com)
15:31.20tzangerqueuetue: good
15:31.23ManxPowertzanger, Very few PBX FXS ports signal disconnect with anything other than a tone.
15:31.31tzangerthis is true
15:31.40tzangerAbsoluteTimeout() may be your only hope
15:31.55ManxPowerAmd we all know how well Asterisk handles tone disconnects
15:32.06tzangervery well... it ignores them perfectly every time
15:33.52Dr-Linuxwhts difference between ztdummy and ztdynamic ?
15:34.10fourcheezeManxPower: is it possible to use something like a sipura SPA-3000 to connect to the FXS and get stuff into the voip world?
15:34.40*** join/#asterisk RoyK (n=roy@80.239.107.70)
15:36.12fourcheezeI mean can the spa-3000 forward stuff from its FXO port to a sip url
15:37.14mutilatoranyone know what kinda warranty wildcards have?
15:37.46parylagain, sorry, but can an analog adapter handle fax/modem calls?
15:38.35azzieparyl, when configured correctly - yes
15:38.37*** join/#asterisk Craziman2 (n=Craziman@boromir.apid.com)
15:39.19parylazzie: thank you.  any specific brand?  and what do you mean by properly configured... a specific codec, or what?
15:39.39thewiizardyoright ive got this tedious issue
15:39.50thewiizardall calls apart from internal are getting "All circuits are busy now"
15:39.57azzieparyl, what do you want exactly from fax/modem ?
15:40.03mutilatoranyone know?
15:40.13mutilatori don't see any warranty info on digiums site..
15:40.28thewiizardyears manufacturer i reckon
15:40.40thewiizardusually the detault for most hardware these days
15:40.44thewiizard*default
15:41.04parylazzie: i need to provide at least 11 analog lines... 8 fax machines, 3 modem lines
15:42.31azzieparyl, any bandwidth limitations? can you have all calls G.711 ?
15:42.46*** join/#asterisk santiago (n=santiago@208.195.215.124)
15:43.01*** join/#asterisk mkrufky (n=mk@68.160.103.77)
15:43.16parylazzie: this is just for the connection to asterisk... i'll be routing these calls directly to one of the T1 channels
15:44.13azzieso what's the answer to my question
15:44.18parylso... no bandwidth limitations, aside from the normal network
15:44.31parylthey'll be on a 100mbit switch
15:44.40Dr-LinuxManxPower: how can i access Cisco 79xx ip phone remotely to configure with SIP ?
15:44.45*** join/#asterisk miksi (i=monte@a80-186-17-71.elisa-laajakaista.fi)
15:44.54miksihello room
15:45.19miksii need help on asterisk anybody can help
15:45.26azzieparyl, then just use G.711 and it won't matter - fax, modem, voice...
15:46.18parylazzie: awesome.  and there aren't any brands to stay away from, as long a they handle G.711?
15:46.26ManxPowerDr-Linux, I have no idea, I don't use Cisco phones.
15:46.27mutilatordoes anyone know tho?
15:46.31*** part/#asterisk Craziman2 (n=Craziman@boromir.apid.com)
15:46.33azzieparyl, yes...
15:46.48parylazzie: thanks so much.
15:46.56emdubanyone familiar with cisco 7960 phones?
15:47.04azzieparyl, problems begin when you need to maximize compression of voice/fax calls etc. If you don't care about compresing them, there's no problem
15:47.13thewiizardU need the sip firmware from Cisco
15:47.18*** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com)
15:47.23thewiizardphat luck unless u have a support contract
15:47.27manelideothewiizard the problem with /usr/sbin/amportal was solved with a reboot lolol
15:47.35thewiizardhehehe nice manelideo
15:47.50parylgotcha... i've been spooked by the fax pages on all of the asterisk sites.  i guess that's what they were referring to
15:48.19ManxPowerDr-Linux, One might start by reading the documentation for your phone and firmware
15:49.21enemyis there some way I can automaticly add agents to queues without having to get them to dial an extension executing AddQueueMember?
15:49.54*** join/#asterisk _santiago_ (n=santiago@208.195.215.124)
15:50.09iCEBrkrenemy: They'd have to login no?  So the system knows they're at their desk?
15:50.49ManxPowerenemy, set a member= line in queues.conf for "agents" that are always "logged in"
15:50.50enemyiCEBrkr: I don't require that, they are always at their desk
15:51.04ManxPoweri.e. member=SIP/happyagent
15:53.33*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
15:54.12*** join/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
15:54.12*** mode/#asterisk [+o twisted[asteria]] by ChanServ
15:54.43enemymanxpower, like group=1 member = SIP/236 ackcall=no ?
15:54.57ManxPowerender, I don't use group, but yes.
15:55.10ManxPowerHere, I'll post my queues.conf
15:55.37*** join/#asterisk viLeR (i=1000@66.128.47.232)
15:56.00*** part/#asterisk miksi (i=monte@a80-186-17-71.elisa-laajakaista.fi)
15:56.08ManxPowerhttp://pastebin.ca/27998
15:59.07thewiizardDr-Linux
15:59.17thewiizardgoogle for Cisco 7960 SIP
15:59.27ManxPower~mailinglist
15:59.28jbotmailinglist is, like, Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
15:59.29ManxPower~docs
15:59.31jboti heard docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
16:03.06*** join/#asterisk oej (n=Olle@apollo.webway.se)
16:03.09*** join/#asterisk thdei (n=DD@nat1.cri74.org)
16:03.58Pazzohi! I'm successfully running SER / Asterisk / Mediaproxy / Stund toghether, BUT: incoming calls on Asterisk are going to be routed to SER - and if nobody answers the call should go to Voicemail (on Asterisk). Voicemail works fine and so do incoming calls - but redirecting them from asterisk to ser and back once again fails.
16:04.07Pazzois there a solution for such a scenario?
16:04.31iCEBrkrAsterisk should handle the transfer to Voicemail.. No?
16:04.37ManxPowerIsn't that a SER question?
16:04.37Pazzoyep
16:04.55ManxPowerPazzo, you would need a timeout on your Dial line on Asterisk
16:05.18enemymanxpower, could I see your extensions.conf also? doesnt look like member had any effect
16:05.39mutilatorhmm
16:05.44mutilatorclec fubared
16:05.50mutilatori can set my ani and cidname
16:05.52file[laptop]Pazzo: you silly person, you're trying to route the same INVITE Asterisk sent out, back to itself aren't you?
16:06.07mutilatorcidnum anyway
16:06.13ManxPowerenemy, http://pastebin.ca/27999
16:08.53*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
16:09.36Pazzofile[laptop]: yep, I'm such a silly person :-)
16:10.08file[laptop]Pazzo: Asterisk will freak out and go, "no no no - LOOP DETECTED you twat!"
16:10.55PazzoSER is handling timeouts (based on sql-based user preferences), will detect the users language (italian, german etc) and route the call to the appropriate Asterisk extension
16:11.16PazzoIf the call comes from a SER client naturally everything is fine
16:11.19file[laptop]'tis not a SER thing, it's an Asterisk thing
16:11.32file[laptop]it's because you're trying to route the call back to itself, and it freaks out
16:11.33Pazzobut as you are saying, it's a kind of loop
16:11.52Pazzono workaround?
16:11.58file[laptop]not off the top of my head
16:12.16blitzrageAsterisk says, "send this call to this address", then SER says, "this call is for you", then Asterisk is like, "no way Jose, I just sent you that call"
16:13.22Pazzomaybe I could put a flag on calls from asterisk to recognize them later and don't do voicemail routing on SER if this flag is set?! and in parallel also try to handle timeouts the same way in asterisks extension.conf...
16:14.06file[laptop]mmm what about rejecting the call once it hits the failover route, if it's going back to asterisk and it came from asterisk
16:14.19file[laptop]and then in asterisk deal with the rejection... might be able to add a header or something
16:14.33file[laptop]that was just randomly off the top of my head
16:14.36blitzragecustom SIP header maybe?
16:14.46file[laptop]blitzrage: that's what I was thinking
16:14.55file[laptop]anyway, I'm leaving in like 5 minutes
16:14.56blitzragefile[laptop]: since we can set those now :)
16:15.00blitzrageand parse them too
16:15.05Pazzofile[laptop]: extactly, that's what I'm trying to explain with my poor English words :)
16:15.15blitzrageMy HD died on my PBX last night :(
16:15.39blitzragewonder if I have an extra one lying around
16:15.45Pazzoif (call from asterisk) { let call fail } else { do all the preference-based stuff including voicemail etc }
16:15.47blitzrageanyways, I'm out -- lates
16:16.22Pazzoand on asterisk I have to implement my preference-based voicemail system - but that shouldn't be hard to realize
16:16.52Pazzothanks a lot for helping to clean up my mind!
16:17.16*** join/#asterisk santiago (n=santiago@208.195.215.124)
16:17.51*** join/#asterisk theNOTO (n=biggs@69-165-25-59.clvdoh.adelphia.net)
16:18.29thewiizardyo is there a default incoming context for asterisk
16:18.40thewiizardand is it 'incoming'
16:19.15theNOTOI am going to be running asterisk with nufone service at home.  What are the advantages of IAX vs SIP?
16:19.20Pazzoblitzrage: asterisk is now able to set custom SIP headers?
16:19.35ManxPowertheNOTO, IAX2 is easier to deal with when NAT is involved.
16:19.55Pazzoblitzrage: (I'm running 1.2.0b1 as I had some trouble with 1.2.0b2)
16:20.22blitzragePazzo: yes -- do a show applications
16:20.59theNOTOManxPower: Ok, what about bandwidth requirements?
16:21.51Pazzoblitzrage: hey, that's great! thanks a lot!
16:22.37Pazzoblitzrage: has this also been there in 1.0.x and been overseen by me or is this a new feature in 1.2.x?
16:22.49blitzragenew feature
16:22.52blitzrage1.0.x is OLD
16:24.27*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
16:24.40ManxPowertheNOTO, they are about the same for small numbers of calls.
16:24.57ManxPowerIAX2 might use a little bit less bandwidth, assuming the same codec.
16:25.29tzangeronly if trunking
16:25.46tzangerthe difference in RTP and IAX2 overhead is negligable
16:25.54Pazzoblitzrage: that's true :-) I have had to start a VoIP project for a local ISP two or three months ago - it was the first time I have been in contact with Asterisk & SER and started with their stable versions - but I realized very soon that Asterisk 1.0.x won't make me happy ;-)
16:26.01ManxPowertzanger, I did say :little bit:
16:26.07tzanger:-)
16:26.16*** join/#asterisk tzhs (n=thomas@p5497DBD5.dip.t-dialin.net)
16:29.13sahafeezdoes asterisk keep call records by default
16:29.43ooglesahafeez: check /var/log/asterisk/cdr-csv
16:30.15sahafeezoogle: thanks
16:31.39KriS83Short question. When using CDR, can it be used for logging _all_ incoming and outgoing calls?
16:32.08KriS83Yes/no answer prefered ;)
16:32.23*** join/#asterisk toddf (n=toddf@adsl-65-66-18-204.dsl.okcyok.swbell.net)
16:32.34QwellKriS83: thats what it does
16:32.41*** join/#asterisk jlewis (i=jlewis@solo.atlantic.net)
16:33.02KriS83Qwell, ok. Perfect! Thats all I wanted to know ;)
16:33.06KriS83thx
16:33.35tmccraryHas anyone here done VOIP over a VPN?
16:33.48tmccraryor more specifically with OpenVPN (SSL VPN)
16:34.06tmccraryIf so, should I do anything special to optimize performance?
16:34.14azziequality should suck big time ;)
16:34.22tzhsTrying to wait longer then 10s on an incoming call from capi causes Spawn extension (isdn-in, XXXXXXXXXX, 1) exited non-zero on 'CAPI/ISDN1XXXXXX-3'. is there a solution?
16:34.24*** part/#asterisk pif (n=ldm@zenon.apartia.fr)
16:34.41Inv_arptmccrary: i know a friend that does it using 3com/ulaw on dsl  works ok for him
16:36.06Inv_arpi just know that certain codecs being ecapsulated in VPN is kinda screwy (for lack of a better term)
16:36.26Lostfrogtmccrary: I use IPsec and Openvpn.. no problem with either.
16:36.44ManxPowertzhs, that means you don't have a priority 2
16:37.37jlewisI'
16:37.38[TK]D-FenderI use an SPA-2000 over an IPSec connection here to link to my other plant.
16:37.40LostfrogThe thought of not using a VPN scares me.
16:37.52jlewisI've done iax over pptp with encryption
16:38.03jlewisit seemed to work
16:38.14tzhsManxPower: there is one, but capi gets an disconnect to early (still ringing)
16:38.47ManxPowertzhs, PASTE the section of extensions.conf to pastebin.ca
16:38.49thewiizardyo well quick question
16:38.58thewiizardi have context called [from-sip]
16:39.13thewiizardhow cna i forward all calls that come into that context to my callingcard app
16:39.15thewiizardon 850
16:39.51ManxPowerthewiizard, you don't "forward" the calls, you just match the call in extensions.conf that runs your app
16:40.10jlewisexten => _. will match all calls
16:40.20thewiizard_.
16:40.25thewiizardso _.,1,Answer
16:40.33thewiizardthat what we're talkin
16:41.44|Vulutre|Anyone know how to set it so that a Polycom IP500 can only accept 1 call per line at a time? so that the second call with go into busy voicemail?
16:41.44tzhsManxPower: http://pastebin.ca/28002
16:42.10ManxPowertzhs, What version of Asterisk are you using?
16:42.34ManxPower|Vulutre|, it's documented in the release notes for 1.5.x I think.  Maybe the 1.6.x
16:42.43tzhsManxPower: cvs head, but the same with 1.0.9
16:43.09ManxPowertzhs, priority n is not supported by 1.0.x or CVS -r v1-0
16:43.24|Vulutre|ManxPower: is it incominglimit=1 ?
16:43.32tzhsManxPower: thats true. but now im using head
16:43.46ManxPowertzhs, I don't have any suggestions
16:44.01[TK]D-Fender|Vulutre| its all in your sip.cfg
16:44.08[TK]D-FenderOr phone.cfg
16:44.11ManxPower|Vulutre|, only if you want to use Asterisk's crappy call limiting stuff.  Why not use the polycom call limiting stuff -- that actually works.
16:44.22tzhsManxPower: anyway tnx
16:44.30[TK]D-FenderUse the Polycom offer feature, trust us...
16:44.32ManxPowertzhs, talk to someone that uses BRI
16:44.46|Vulutre|[TK]D-Fender: do you know what that command is, so I can research it?
16:45.24ManxPowertzhs, can you do an Answer then Ringing, then the Wait?
16:45.35ManxPowerYour telco might be upset about nothing happening for 10 seconds
16:45.40thewiizardanyone ever used areskicc?
16:46.12ManxPower|Vulutre|,  call.callsPerLineKey="1"
16:46.13tzhsManxPower: no, because there is an other phone connected which i want to use
16:46.24|Vulutre|ManxPower: thank you
16:46.35ManxPowertzhs, Ask on the mailing list.
16:46.46tzhsManxPower: i like to use it as an extended answering machine
16:47.00ManxPowertzhs, *nod*  Asterisk is not designed for that.
16:47.01tzhsManxPower: ok. i'll do it
16:47.32ManxPowerAsterisk is designed to be a PBX, not an answering machine and so doesn't not really support stuff like "use a phone on the same line, but not controlled by asterisk"
16:51.43bjohnsonhehe .. my son broke my plastic "H" key for my Compaq notebook and I have to add a whole new keyboard
16:52.01ManxPowerI suggest selling him on the back market.
16:52.42mog_worki have an h key
16:52.50mog_workfor a compaq presario notebook
16:53.34bjohnsonthis is a M700 Armada .. no idea if they are compatible
16:53.44bjohnsonI've been trolling service shops
16:54.01mog_workblack and thick?
16:54.34mog_workcerts are lame
16:54.44javoya
16:54.46thewiizardim self certified
16:54.50thewiizardprobably just as good :)
16:54.56javowhoa!
16:54.58mog_workbut so was i when i was 12
16:55.02thewiizardim 11
16:55.04thewiizard:P
16:55.12mog_workwe can tell
16:55.16thewiizardhah
16:55.16thewiizardur wrong
16:55.20thewiizardcuz im not really ;)
16:55.26thewiizardjust yanking ur chain
16:55.28mog_workbut im right on the inside
16:55.39thewiizardbut are you Allright or just partiality
16:55.43thewiizard*partially
16:55.50mog_workwell thanks
16:56.03thewiizardno probs
16:56.24Marcel-AS16215quick question how i can filter in the extension.conf circuit-busy that my Asterisk say this and not forward me to the Demo-thanks box ?
16:57.12mog_work?
16:57.13thewiizardright got another interesting one for ya
16:57.26bjohnsonmog_work: light grey and thin
16:57.52mog_worki dont think it will work then
16:57.57mog_workmy keys are real hard
16:58.02thewiizardGot calling cards app installed with asterisk
16:58.12mog_workwhy not move the} key or something you dont use a lot over
16:58.22mog_workI never use the number 9
16:58.24thewiizardi have a Siptone2 registered as an extension i dial 850 which the access number for the CC App.
16:59.11thewiizardwhen i dial in the App listens for DTMF for the Calling Card Number and then logs me in with that number
16:59.33thewiizardwhen i dial from this local extension the app goes skitz and believes ive entered some incorrect digits
16:59.52thewiizardthen its skitzs again and ends the call
17:00.45thewiizardfine from a ddi tho
17:00.54mog_workdo dtmf debug
17:00.57ManxPowerthewiizard, You prolly have a problem with SIP DTMF
17:00.59mog_worktell me what is interpreted
17:01.02bjohnsonmog_work: the H key connectors on back are upside down compared to the rest and the corner is notched out for the pointer thing
17:01.03mog_workand what you are sending
17:01.10ManxPoweri.e. phone set for inband, asterisk set for rfc2833 or something like that
17:01.15bjohnsonlooks like I can buy a keyboard on ebay for $20
17:01.22mog_workwell there you go
17:01.35thewiizarddtmf debug = noon
17:01.42thewiizardno cmd
17:01.49mog_worklol
17:01.54mog_workgo to logger.conf
17:02.01mog_workenable dtmf in console
17:02.04mog_workthen logger reload
17:02.43thewiizardive got full loggin
17:02.51thewiizardnotice warning error debug verbose
17:03.09mog_workand add dtmf
17:03.11mog_workits new
17:03.12thewiizardah
17:03.14thewiizardsweet
17:03.18mog_workyou are running current asterisk right?
17:03.29sjobeck99hi, all, how's it goes the good fight today?
17:03.36sjobeck99i wonder why i'm seeing :
17:03.36sjobeck99<PROTECTED>
17:03.42thewiizard1.0.9
17:04.17sjobeck99that comes after:
17:04.17sjobeck99Executing Dial("Zap/4-1", "IAX2/NuFone/5088888888|30") in new stack
17:04.19mog_workoh its not in 1.0.9
17:04.24mog_workitd be in the betas
17:04.33mog_workseeing dtmf is harder in older versions
17:04.45mog_workis that account active sjobeck99
17:04.59sjobeck99yes, thx
17:05.16sjobeck99i edited the # in my paste, but, yes
17:05.50*** join/#asterisk digime (n=drooth@ip68-111-235-172.sd.sd.cox.net)
17:05.55sjobeck99is my iax.conf goobered?
17:06.07*** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com)
17:06.09synthetiqwhen putting in signaling of e&m wink in zaptel.conf is it em_w or emw
17:06.15mog_workmaybe
17:06.39mog_workin zaptel its just e&m
17:06.48Inv_arpsjobeck99: paste it in pastebin.ca
17:06.59Inv_arp~seen ariel
17:07.10jbotariel <n=ariel@200-70-76-10.mrse.com.ar> was last seen on IRC in channel #debian-br, 79d 12h 26m 1s ago, saying: 'mas o multiplicador sta en 18x'.
17:07.10synthetiqi dont inticate the wink?
17:07.10mog_worknope
17:07.10mog_workyou do that in zapata
17:07.22Inv_arp~seen ariel_
17:07.24jbotariel_ <n=9999@dsl-20-177.cofs.net> was last seen on IRC in channel #asterisk, 18h 7m 55s ago, saying: 'harryvv, OK.  If you call saying at home due to me falling off the roof just after the hurricane ok.  Oh well I guess I am ok just can't do much getting out yet.'.
17:07.58Inv_arpisnt it possible on freenode to leave someone a message
17:08.11*** part/#asterisk santiago (n=santiago@208.195.215.124)
17:09.18demetriowhat does "failed to authenticate on INVITE" mean?
17:09.44Math`it means that the authentication failed on an SIP invite
17:09.50Math`which means: bad login
17:10.19JunK-YInv_arp: if the user is offline? memoserv
17:11.03demetriowell the problem I get is this one: with sipgate I'm able to receive calls but when I try to make an outgoing call i get the error above
17:11.09Inv_arpJunK-Y: ahh k thx
17:11.15demetrioso can't be bad login
17:11.46manywhats your extension.conf partial for dialing to sipgate?
17:12.05*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
17:12.39demetrioexten => _6.,1,Dial(SIP/${EXTEN:1}@sipgate.co.uk,100,"T")
17:12.53*** join/#asterisk Assid (n=assid@203.115.64.59)
17:13.07manydo you have a peer section in sip.conf?
17:13.09Qwellwell, no wonder it's failing...
17:13.09demetriosure
17:13.23demetrioI can paste if it doesn' count as flood :)
17:13.23QwellWhat is the name of the peer?  Replace "sipgate.co.uk" with that
17:13.31demetriothe name of the peer is sipgate.co.uk
17:13.40demetrio[sipgate.co.uk]
17:14.09manydoes it contain u/p?
17:14.25demetriosure
17:14.28QwellI bet it's trying to use dns instead of the peername
17:14.34QwellI don't know what order * does
17:14.42manyme neither.
17:14.50Qwelldemetrio: try making it just sipgate
17:14.59Qwellsee if that changes the error at least
17:15.05demetriowell, it works with other providers (same structure, peer name equals host name) and the only one giving trouble is sipgate
17:15.10demetrioanyway I'll try
17:15.26manymaybe the others need reg only, no auth for invite
17:15.30Qwelloff to work
17:16.48*** join/#asterisk vlrk (n=vlrk@59.93.67.217)
17:17.02demetrionope, the error remains the same
17:17.11Assidhey.. what could be the issue if people are complaining of call dropping?
17:17.16Math`can you make a sip debug trace to pastebin?
17:17.34manysip debug peer <peername>
17:17.50manyand dont forget to obfuscate passwords, incase theyre in.
17:17.50mutilatoris there any way to silence registration failures in console yet?
17:17.55bweschkedemetrio: post your sip debug trace to pastebin as suggested here, and also your sip.conf definition of sipgate (XXX'ing our user/pw)
17:18.03mutilatori have a guy that tries to register every second
17:18.44demetriohmm I think I found the problem
17:18.56demetrioWe're at 192.168.0.2 port 11246
17:19.11bweschkehuh?
17:19.18justinuthat's just for RTP
17:19.39demetriowhat does that mean?
17:19.55demetrioasterisk is telling sipgate to contact him on a local address
17:19.58Math`many: ah I didnt know about the "peer" option to sip debug, will be useful as hell
17:20.02bweschkethat's the local resource/socket selected for RTP transmissions
17:20.04justinuit means it has nothing to do with your auth failure
17:20.18ManxPowerRTP == AUDIO
17:20.20manyMath`: what? limiting sip debug to only the peer you actually want to debug? :)
17:20.25Math`many: yeah :)
17:20.29justinuheh
17:20.34manyMath`: yeah. :)
17:20.39Math`especially because my DID provider sends alot of MWI messages
17:20.46Math`(even if I have 0 new)
17:21.04demetrioI don't think so, the same debug block says INVITE sip:10000@sipgate.co.uk SIP/2.0
17:21.40bweschkeright... but if you're getting an auth failure, you're never getting to the point where the RTP is setting up
17:21.43manydemetrio: Authentification is IN SIP, not in RTP.  it just preallocates the RTP Socket.
17:21.52bweschkeso - as justinu said: that's not the problem
17:22.27bweschkedemetrio are you setting type=peer for both inbound/outbound in your sip.conf or type=friend ?
17:22.31*** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com)
17:22.50*** join/#asterisk paryl (n=paryl@209.236.78.59)
17:23.26bweschkethere could be a cpl things happening here... 1) you're not matching your profile on the outbound connected, so * doesn't know to authenticate with the peer and is getting turned away
17:23.56tmccraryDo you know what would cause a buzzing sound with Asterisk / GSM and 250 ms latency?
17:23.59bweschke2) sipgate may be particular about what's coming in the SIP headers, and you may need to look at fromuser= or something like that in your profile once you've verified that * is finding that profile on the dialout
17:24.10Math`demetrio: you should see "SIP/2.0 407 Proxy Authentication Required" followed by another packet with the line "Proxy-Authorization: Digest username="5144487726", realm="spectravoice.com", alg
17:24.10Math`orithm=MD5, uri="sip:4506861000@66.59.150.36", nonce="12345abc", response="abcdef12345678901234567890abcdef0", opaque=""
17:25.04justinuare there any sip DID providers that would sell me a block of 20 DIDs for a reasonable price?
17:25.14Math`justinu: for what area
17:25.20justinulos angeles
17:25.21*** join/#asterisk bartpbx (n=bartpbx@proxy.prodyna.com)
17:25.56bartpbxI'm searching for markster. Whats his nick`?
17:26.15parylwhat can i do about an echo on an iax connection?
17:26.17Marcel-AS16215is this the right way to fix the circuit-busy ? Or is ther a better way to filter all the Messages out from my SIP Provider ?
17:26.17Marcel-AS16215exten => _00049[23456789].,1,Dial(SIP/${EXTEN:1}@voipbuster-out,90)
17:26.17Marcel-AS16215exten => _00049[23456789].,2,Busy(pls-try-call-later)
17:26.17Marcel-AS16215exten => _00049[23456789].,3,Hangup
17:28.30parylto clarify: i have an echo on an IAX connection between SIP phones on both servers
17:28.44justinutry turning down the volume on the sip phones
17:28.50justinuprobably acoustic echo
17:29.01paryljustinu, it helps, but it doesn't fix it.  the echo is still there, just quieter
17:29.10justinuwhat kind of phones?
17:29.16parylpolycom 501's
17:29.20justinuwow
17:29.46justinuanyways, if it's purely digital, the echo can only be acoustic
17:30.01justinuthe reason you hear it over IAX, is probably the latency introduced by your WAN link
17:30.23*** join/#asterisk flynux (i=bod6vc1@pingou.in)
17:30.32parylok, that makes sense.  so is there any way to get rid of it?
17:30.57justinureduce the latency, or figure out why the phones are hearing themselves
17:31.04justinuwhat polycom firmware?
17:31.45*** join/#asterisk jtodd (n=jtodd@ti.fox-den.com)
17:31.47*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
17:32.12parylit's the latest as of 2 weeks ago, don't remember the version number right off hand
17:32.20justinu1.6.2 is the latest, iirc
17:32.27*** join/#asterisk Gourou_fou (n=x@ACaen-151-1-20-125.w86-195.abo.wanadoo.fr)
17:32.32Gourou_fouolé
17:32.37parylsounds right.  the bootrom says 2.6.2
17:32.59parylright.. 1.6.2.0041
17:33.05justinuok, so firmware is good
17:33.36justinuwhen party A hears echo, have party B cover his mic
17:33.41justinusee if that makes it go away
17:34.02Math`nice accoustic echo test
17:34.02justinuif it does, you know for sure it's poor acoustic coupling
17:34.56*** join/#asterisk Lurr (n=pr0ph3t@63.69.20.3)
17:35.58parylit doesn't go away
17:36.19justinubizzare
17:36.39justinuhow about having party be hit MUTE?
17:36.41justinuparty B
17:37.38Math`those phones are purely digital?
17:37.45justinu501s? yeah
17:37.46*** join/#asterisk jonarildh (n=vircuser@195.28.172.4)
17:38.26*** join/#asterisk Utah_Dave (n=boucha@0-2pool130-248.nas28.salt-lake-city1.ut.us.da.qwest.net)
17:38.32parylmuting on the other side makes the echo go away
17:38.41justinuok, so it's gotta be acoustic
17:38.49justinuparyl: i'd talk to polycom about that... seems wrong
17:39.09jonarildhsomeone that can help me with compile errors on Suse 10.0 and asterisk with bristuff-0.2.0-RC8f-CVS
17:39.11tzangerjustinu: you can't just cover the mic though
17:39.11jonarildh?
17:39.19tzangerit could be transmitted through the handset mechanically
17:39.27justinutzanger: good point.
17:39.28tzangerbest to disconnect the headset from the base
17:39.36tzangermost of them use the tiny 4-pin jacks
17:39.55justinuparyl: just know that the person that hears echo isn't the cause of it
17:39.58justinuit's the other sie
17:39.59justinuside
17:41.25*** join/#asterisk JASON-0 (n=jason@jason.unitz.ca)
17:41.39tmccraryis there a way to have asterisk boost volume on calls/
17:41.53iCEBrkrtmccrary: what type of calls?
17:41.56tmccrarySIPO
17:41.57tmccrarySIP
17:42.07JASON-0Hello,  I just setup a queue and no matter what settings I do, it sends the call to fail over when no one answers their phone.  What I need is for the call to stay in the queue.
17:42.07justinuno
17:43.05puzzledis that normal: show translations ulaw -> gsm 105 & gsm -> ulaw 2 ?
17:43.15tzafrir_laptopjonarildh, bristuff "CVS" is quite old (from 29-may-2005)
17:43.18justinuheh
17:43.22parylshe has a headset on the other end, so i'm having her disconnect it to see it that's the problem
17:43.45justinuoh, headset problems
17:43.55justinu:P
17:44.05*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net)
17:44.11justinudon't buy cheap headsets :)
17:44.20justinupolycom 501 audio is almost a work of art, it deserves better
17:45.10parylwell.. they're nice plantronics headsets
17:45.17justinuhmm
17:45.21parylbut i'm wondering if it's a setting with the amp
17:45.22justinumaybe it's defective
17:45.31justinubut yeah, could be lots of stuff
17:47.28Assidcan someone help me thing up what could be the issue for this.. calls just drop on their own..
17:47.40Assidi am thinking is maybe cause i have qualify=yes
17:47.52Assidand i have seen the phone disappear for 2 seconds
17:49.04Assidcoulod it be because of that
17:49.35*** join/#asterisk SplasPood (n=sp@paravolve.net)
17:50.18parylok, that settles it... it's the headset
17:50.37paryli think it's the gain on the amp... but after unplugging it the echo went away
17:50.40justinucool
17:51.42perda quad span t1 card can run on a dual 3ghz system with 2gig ram right?
17:51.49perdi'd have the channels used most of the time
17:52.26perdin the book it says 15 hcannels requires a 3ghz 1gb ram.. yet a quad span is more than 4x the amount of calls they say it can handle
17:54.26zoayes it can
17:54.36perdok
17:54.53perdi dunno wtf the book is talking about saying that for more than 15 channels you need to spread it across multiple servers
17:55.04perdunless by channel they mean t1
17:55.17Math`so a T3 would require 2 servers probably
17:55.58Math`or some kinda hypercube-architecture-based server
17:56.30*** join/#asterisk Abbas (n=Abbas@203.81.216.47)
17:56.38perdmy pbx is powered by the ball sweat of NFL athletes
17:56.45perddon't ask me how i harvest it.
17:58.05Math`lol
18:03.52*** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg)
18:04.09*** join/#asterisk leibniz11 (n=leibniz@163.1.203.62.cust.bluewin.ch)
18:04.24JASON-0I need help setting up a queue in asterisk.. can someone help me ?
18:04.48yxacan someone help me test if my isp blocks my sip port @ 202.156.228.121?
18:04.53leibniz11is it possible to have asterisk transcode between SIP and MGCP? I have a sipura and want to have it talk to my CCM 3.3.X
18:05.21*** join/#asterisk psk (n=psk@golia.caltanet.it)
18:06.02justinuasterisk MGCP can only talk to phones
18:06.12justinuit can't talk to other gatekeepers or media gateways, iirc
18:07.35demetriowell, looks like I'm not smart enough to solve the problem
18:07.38leibniz11how can you communicate asterisk with a non-sip CCM?
18:07.43demetrioso here you are: http://pastebin.com/421947
18:08.12justinuleibniz11: don't think you can
18:09.15yxacan someone call me @ 202.156.228.121? i'm not sure i setup sip correctly...
18:10.39Math`which country is that
18:10.45yxasingapore
18:10.58Math`+202.156.228.121?
18:11.08yxaer SIP
18:11.24Math`ah thats an ip
18:11.29justinuheh
18:11.31Math`what extension
18:11.34yxa2222
18:11.37Math`k
18:12.40Math`<PROTECTED>
18:12.40Math`<PROTECTED>
18:13.10yxai'm not seeing anything on my console.
18:14.22justinudemetrio: try setting  fromdomain=sipgate.co.uk
18:14.27Math`yxa: http://pastebin.ca/28014
18:14.45Math`extension 2222 doesnt exist on your side
18:15.13Math`(well doesnt exist in the guest context you've defined)
18:16.15demetriojustinu: thanks, it works :)
18:16.26yxaMath` so it did connect?
18:16.26demetriowhy is that?
18:16.52Math`SIP/2.0 404 Not Found
18:17.22yxai have exten => 2222,1,Dial(SIP/grandstream) under [test] in extensions. is that rite?
18:17.46Math`thats ok, in your sip.conf file, check for context=test
18:18.05justinudemetrio: wrong authentication realm if you don't put that
18:18.14yxaMath` under [general]?
18:18.29Math`yxa: well under the sip.conf user you have for accepting guest sip calls
18:18.58jonarildhtzafrir_laptop: I have downloaded the latest zaptel from cvs, but still errors on compile
18:19.07*** join/#asterisk darwin_35 (n=darwin35@208.139.193.178)
18:19.07yxaMath` sorry i'm not sure how to do that. been trying to look at wikis
18:19.21Math`under general then
18:19.24Math`whats the context defined
18:19.37yxadefault
18:19.49Math`you probably want that changed to: test
18:20.14Math`now all sip calls will jump to context "test" unless specified otherwise in the sip user entry
18:20.15yxaMath` ok. pls try
18:20.24Math`rings :)
18:20.37yxaMath` great thanks
18:20.41Math`did u hear me?
18:20.53tzafrir_laptopjonarildh, right, because it is not intended for latest CVS
18:20.54yxano i didn't. i said hello
18:21.02Math`is your asterisk box nat'ed
18:21.11Math`I heard you but I guess you had no sound on your end
18:21.17*** part/#asterisk darwin_35 (n=darwin35@208.139.193.178)
18:21.37yxaMath` well 202.156.228.121/5060 is port forwarded to my box inside 192.168.2.10
18:21.45Math`you need to forward RTP ports too
18:21.58yxawhich are?
18:22.16*** join/#asterisk Blackthorn (i=blacktho@72.236.88.10)
18:22.23Math`by default
18:22.24Math`rtpstart=10000
18:22.24Math`rtpend=20000
18:22.28Math`you can change them in rtp.conf
18:22.39*** join/#asterisk miguel3239 (n=myoung@h-68-167-124-170.cmbrmaor.covad.net)
18:22.56yxaudp?
18:22.59Math`yes
18:23.24tmccraryi thought rtp was its own protocol?
18:23.35Math`no rdp is over udp
18:23.40yxadont think my home router has that function of forwarding a whole range :(
18:23.55tmccraryahh, I see. Thank you for the correction
18:24.00Math`tmccrary: it adds sequence numbers etc over udp to ensure that stuff is delivered in the right order
18:24.16tmccrarykind of like weak tcp
18:24.19tmccrarywithout the overhead
18:24.43sjobeck99hi, all, it tells me no class "default" for MOH, but I commented that out since I want 'native-random'. what am I misunderstanding? how does one use native-random?
18:24.47BlackthornHello. I have a SPA-2000 ip 192.168.1.2 behind a nat router 192.168.1.1. On the spa menu nat = off. The Asterisk server has nat = yes and qualify = yes. The problem is that one of the two lines keep droping registration. REbooting the router, the * server, or the sipura dosn't fix. But turning both lines off on the sipura, reboot, then both lines back on does. Thoughts?
18:25.00Math`tmccrary: yeah, and not connection-oriented I think.... see ftp://ftp.rfc-editor.org/in-notes/rfc1889.txt for more info
18:26.04yxaMath` can you try again?
18:26.51Rowteranyone knows if micronet SP5054 could connect to a sip voip reseller?
18:27.11Math`yxa: u getting any sound?
18:27.12sjobeck99what is micronet sp5054 ?
18:27.18yxaMath` no dude
18:27.37justinuqualify = yes means how often?
18:28.06Dr-Linuxi have install zaptal, and loaded ztdummy module as well, but meetme is not working .. any clue ?
18:28.16Dr-Linuxi have not hardware
18:28.18Rowtersjobeck99, a voip gateway
18:28.20Dr-Linuxi'm using soft client
18:29.11*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
18:29.13Math`Dr-Linux: any log emssages?
18:29.17Math`messages, that is
18:29.24justinudr-linux: did you create the udev stuff?
18:30.33Dr-Linuxi just created a 1111 room in meetme.conf and define an extension 1112 in extension.conf, is it enough to work confierence call ?
18:30.41sjobeck99http://www.micronet.info/Products/voip/SP5054.asp says that no it wont, if we're reading it correctly
18:30.42tmccraryWhat is the best free codec available? GSM has bad quality and PCMU/ulaw takes up too much bandwidth
18:30.50Kattyi just /love/ how polycom won't talk to me
18:30.54justinudr-linux: you need to have the proper /dev entries for zaptel
18:30.55tmccraryI guess I could go with pass-thru
18:30.56yxaMath` so this is the famous sip nat problem
18:31.04tmccrarysip + nat = hell
18:31.06tzangertmccrary: I have no issue with gsm
18:31.09Math`yxa: actually its an RTP nat problem
18:31.14tzangeryou could try speex
18:31.30justinugsm makes me realize how shitty cellphones sound
18:31.32sjobeck99arent we all quite used to gsm from using our mobile phones every day?
18:31.33Rowtersjobeck99, it wont mmh
18:31.38Math`justinu: HEHEHE
18:31.41tmccraryGSM is okay, I mean its understandable. But not even as good as regular telephone
18:31.50justinutmccrary: agreed
18:31.52sjobeck99ilbc, perhaps
18:31.58Math`yxa: you can set the externip= parameter in sip.conf
18:31.59tmccraryYeah, we use it for mobile phones... but they are mobile
18:32.12*** join/#asterisk folsson (n=filip@lund-meje-sr0-vl101-249.perspektivbredband.net)
18:32.25yxaMath` i idid
18:32.30justinudr-linux: check README.udev in the zaptel directory
18:32.30Blackthornjustinu: i'm not sure how often. it's what ever the default is for qualify=yes. I belive I coudl put qualify=#
18:32.47*** join/#asterisk vlrk (n=vlrk@59.93.67.38)
18:32.48sjobeck99looks like that micronet wants to connect to another same box on other end
18:33.00Math`yxa: you are sure all rtp ports are forwarded?
18:33.14justinublackthorn: it sounds like your NAT might be closing the nat bindings
18:33.14tmccraryhas anyone here used a Sipura SPA-841. Is it buggy and weird for you too?
18:33.25Dr-Linuxjustinu: let me check
18:33.26justinusipura 841 is ok, except for the shitty buttons
18:33.32Math`Dr-Linux: sounds like your missing app_meetme.so
18:33.33BlackthornOne strange thing is that the spa says line one uses 5060 and line two uses 5061. but on the * server they show using 1030 and 5060
18:33.41vlrkmy voicemail takes /var/lib/asterisk/sounds/voicemail where as it  has to take /var/spool/asteirsk/voicemail
18:33.44Math`Dr-Linux: which * version are u running
18:33.59justinumath, i think his problem is that he didn't create the zaptel entries in /dev
18:33.59Dr-LinuxMath`: 1.0.9
18:34.13Dr-Linuxyes
18:34.20justinuapp_meetme wont load unless they're there
18:34.25yxaMath` i'm trying out this special application portion of my home router
18:34.25Dr-Linuxi didn't create any entries in /dev
18:34.28vlrkis the issue with addmailbox script
18:34.31Blackthornjustinu: What does that mean and how do I correct it.. is it settings ont he spa? the nat router, or * server?
18:34.40perdis there some kind of null channel i can specify in my .call files?
18:34.53sjobeck99any one familiar with 'native-random' music on hold ?
18:34.56perdso i can just have it traverse my extensions without having a connection?
18:35.02Math`justinu: [13:33] <Dr-Linux> Nov  8 23:30:26 WARNING[2561]: pbx.c:1293 pbx_extension_helper: No application 'Meetme' for extension (default, 1112, 1)
18:35.07Math`justinu: he pasted that to me in privmsg
18:35.14yxaMath` fire away
18:35.22justinublackthorn: maybe on the SPA, set "nat keep alive enable"
18:35.26justinumath: ah
18:35.34Math`sound over yours?
18:35.41yxaMath` nope.
18:35.52yxaMath` ok, can i jus forward 2 rtp ports?
18:35.56Math`no
18:36.09Math`why don't u set a dmz on your * box?
18:36.10Dr-Linuxjustinu: i have no hardware, i wanna use ztdummy, still i need work with /dev  right ?
18:36.20justinuDr-Linux: correct
18:36.37Math`why do people stick with 1.0.9 :(
18:36.50Blackthornjustinu: there you go. sure enophe. line two keep alive enable was turned off. Thanks!
18:36.57azzieMath`, because 1.2 is far from 1.0.9 in stability
18:36.59justinuBlackthorn: no prob
18:37.05yxaMath` dmz? i only have a home dynamic ip connection ;)
18:37.11Math`you've a router?
18:37.24yxaMath` yeah, a very low end one
18:37.42Math`you probably have a DMZ option, this option is going to forward all traffic to the ip you specify in there
18:37.47Math`even very low end ones have that
18:38.18yxaMath` oh i saw it
18:38.20Kattydoes anyone have to reboot their polycom500s on a regular basis?
18:38.28Kattyif by regular i mean every 10-12 calls?
18:38.32yxaMath` fire :)
18:39.12Math`i got no sound
18:39.14Math`u got any?
18:39.16yxame neither
18:39.23Kattyyes? no?
18:39.25Kattymaybe so?
18:39.27justinukatty: no
18:39.32Kattyjustinu: thanks for answering
18:39.39Kattydoes anyone /else/ have problems with their polycoms?
18:39.41justinubut i have 501s
18:39.49justinunot much diff tho
18:39.52Kattyspecifically they can hear you, but you can't hear them!
18:40.14yxaMath` i guess it doesnt work
18:40.26Math`you've rebooer the router after setting the dmz?
18:40.35Dr-Linuxjustinu
18:40.42justinukatty: if you pastebin a sip debug from one of the bad calls, we might be able to diagnose it
18:40.52Kattyjustinu: k
18:41.16yxaMath` do i expose * to the dmz or the phone?
18:41.27Math`*
18:41.44Math`* is gonna do proxying for the rtp stream
18:41.49*** join/#asterisk luke-jr_ (n=luke-jr@user-0c938qu.cable.mindspring.com)
18:42.02Dr-Linuxjustinu: the path in README.xxx file  is not in my machine :S or i didn't understand
18:42.10yxaMath` yeah. didn't ask for a reboot
18:42.19Math`reboot it anyways
18:42.22yxaMath` but i'll do it anyway. hang on
18:42.28IronHelixhmmm
18:42.33IronHelixjustinu were you looking for me?
18:42.59justinuyeah, but i forgot why :P
18:43.04Math`haha
18:43.10IronHelixlol
18:43.11IronHelixGJ
18:43.18justinuthat was yesterady
18:43.20Rowtersjobeck99, know a 4 fxo gateway to connect a panasonic that could connect to a voip reseller?
18:43.22justinuwhole different day and stuff
18:43.26mutilatoranyone here have callerid with name in michigan?
18:43.27IronHelixway too long ago
18:43.29Dr-Linuxjustinu:  /etc/udev/rules.d/50-udev.rules  << this file is not located at the given path
18:43.34mutilatoranyone here have callerid with name, in michigan
18:43.41justinuDr-Linux: what linux distro?
18:43.53Math`Rowter: if the gateway is sip, I got no reason why it wont connect to a voip provider
18:43.55Dr-Linuxjustinu: RHEL
18:44.01Math`plus... are you sure you don't mean 4x fxs?
18:44.05justinuDr-Linux: RHE4?
18:44.21Dr-Linuxjustinu: RHEL 3  update 6
18:44.25NuggetLinux is poo.
18:44.27justinuhmm
18:44.35justinumaybe RHEL3 doesn't use udev?
18:44.45Dr-Linuxhhm..
18:44.45sjobeck99rowter: not really. huh? must be one. I presume you cant use * to do that?
18:44.52justinuDr-Linux: just for reference, I got ztdummy working on FC4
18:45.00justinuso I only know specifics for that
18:45.04Dr-Linuxjustinu: how can i verify ..
18:45.14Math`justinu: ztdummy can work on any kernel
18:45.15RowterMath`, thats what I tought but some of them are expecting another of their kind to make a network.. mmh Mediatrix 1204 might do the trick
18:45.23justinuDr-Linux: ps -ef | grep udev
18:45.29Dr-Linuxokey
18:45.29justinuis udevd running?
18:45.42*** join/#asterisk axscode (n=axscode@203.213.217.122)
18:45.42Math`Rowter: for mediatrixes.... even if they are profiled for a specific provider you can manage to flash it
18:45.52Math`Rowter: I found firmwares on a russian site for my 2102
18:45.57Dr-Linux[root@RHEL-TAC-TEST zaptel-1.0.9.2]# ps -ef | grep udev
18:45.57Dr-Linuxroot      4402  2481  0 23:45 pts/1    00:00:00 grep udev
18:45.57Dr-Linux[root@RHEL-TAC-TEST zaptel-1.0.9.2]#
18:46.03justinuok, so no
18:46.13Dr-Linuxyep .. :S
18:46.24justinuDr-Linux: that means you need to create the /dev/sappseudo entry manually I think
18:46.33justinuer /dev/zappseudo
18:46.51RowterMath`, ahh, the idea is to connect it to a panasonic pbx to get some extensions with voip rates with a reseller ..
18:47.27sjobeck99rowter: sounds like a perfect place to install * to intercept 'those' calls
18:47.39Math`Rowter: so you want to connect 4 outgoing lines to the 1204
18:47.45Dr-Linuxjustinu: should i create zappseudo  file or dir ?
18:48.01sjobeck99going, going, gone, any one familiar with native MOH ?
18:48.07Math`Rowter: where did u buy the unit?
18:48.12justinuDr-Linux: somehow, it needs to get created if it doesn't exist.
18:48.15Rowtersjobeck99, no, because asterisk does not work well with the panasonic hangup singal.
18:48.32justinuDr-Linux: as far as how to do that, i'm unsure
18:48.33Dr-Linuxjustinu: yeah but is it a file? or folder ?
18:48.37justinufile
18:48.40RowterMath`, I think I'll buy it from voipsupply
18:48.42justinuit's a dev link
18:48.46Dr-Linuxok
18:48.48justinuit has a major and minor number
18:48.52Math`Rowter: ok... mediatrix are configurable only via snmp
18:49.03Math`Rowter: so you gotta have a win32 box nearby to install UMN (Unit Manager Network)
18:49.10Math`Rowter: the free version allows you to manage 3 units
18:49.14tzafrir_laptop/dev/zap/pseudo
18:49.37Math`Rowter: usually, they provide you the cd with the installer and the firmware for the unit, if they don't, email me
18:50.02*** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg)
18:50.08justinuon my system, /dev/zap/pseduo is mj 196, mn 255
18:50.27justinualso, /dev/zap/timer is mj 196, mn 253
18:50.27Dr-Linuxooo
18:50.36Dr-Linuxjustinu: this file already exist
18:50.37justinunot sure if that's the same for every box
18:50.38Dr-Linuxcd /dev/zap/pseudo
18:51.03justinuDr-Linux: ok, so then that's not it... check your /var/log/asterisk/messages
18:51.14justinuDr-Linux: there should be some clue why app_meetme won't load
18:51.14Dr-Linuxbut its empty
18:51.21Dr-Linuxok
18:51.28justinuDr-Linux: to a reload from the CLI
18:51.33justinuit forces it to reload everything
18:51.36*** join/#asterisk Frosted (n=Procrast@buzzbud.plus.com)
18:51.43Kattyjustinu: not everything everything
18:52.09justinuwell, for his purposes
18:52.20Kattyporpoises.
18:52.32jonarildhHave anyone got asterisk 1.2-b2 to run with hfc ISDN cards?
18:53.11Dr-Linuxjustinu: this is logs >> Nov  8 23:52:15 WARNING[2561]: No application 'Meetme' for extension (default, 1112, 1)
18:53.30bweschkeahh.. well that helps.. :)
18:53.36justinuDr-Linux: modify logger.conf, uncomment the full line
18:53.40pooh_Dr-Linux: what does show modules show you?
18:53.51axscodeNov 10 14:42:55 WARNING[11601] loader.c: /usr/lib/asterisk/modules/res_config_mysql.so: cannot open shared object file: No such file or directory
18:53.52axscodeNov 10 14:42:55 WARNING[11601] loader.c: Loading module res_config_mysql.so failed!
18:53.53tzafrir_laptop<PROTECTED>
18:53.55justinuDr-Linux: then restart asterisk and look in /var/log/asterisk/full
18:53.55bweschkeDr-Linux: check to see if app_meetme.so shows up in /usr/lib/asterisk/modules
18:53.59axscodeis there anything I have to do during installation?
18:54.10justinubweschke: oh, that's a good point
18:54.14justinuheh
18:54.23pooh_axscode: you need to compile,install and copy over some files from asterisk-addons
18:54.37hardwirehmmhmpmhm
18:54.51*** join/#asterisk TheCops (n=xdz@206-248-136-187.dsl.teksavvy.com)
18:54.53TheCopsHi
18:54.53pooh_axscode: the res_mysql conf file has to be copied over manually
18:54.57hardwiredoes anybody know if asterisk has the ability to set its source address?
18:55.03bweschkedr-linux: if you compile and make install asterisk before compiling and make installing zaptel, app_meetme will not build because it doesn't see zaptel around for a timing source
18:55.07hardwirebased on which address a sip client accessed?
18:55.13justinuDr-Linux: that's true, i had that problem as well
18:55.14pooh_hardwire: SIP.conf and IAX2.conf
18:55.21hardwirepooh_: dynamically
18:55.26Dr-Linuxooo
18:55.28axscodewhat do you mean pooh_?
18:55.29justinuDr-Linux: recompile your source, and do a make install
18:55.40TheCopsSomeone know if there's a way to put 200 PSTN line to an asterisk server ?
18:55.52bweschkeTheCops: what kind of PSTN line?
18:55.53Dr-Linuxbweschke: thats not there app_meetme.so
18:55.56TheCopsbweschke, analog
18:55.58Kattydoesn't asterisk start to combust after 120 calls?
18:56.01Dr-Linuxall modules are there , but thats not :S
18:56.10justinuwow, 200 analog lines
18:56.11pooh_axscode: you need to copy over the mysql conf file manually to /etc/asterisk from asterisk-addons/scrips
18:56.21TheCopsjustinu, yeah, that's what I said when I saw that shit.
18:56.22justinu200 FXO lines?
18:56.22TheCopslol
18:56.30justinuor FXS?
18:56.35pooh_hardwire: nope, sorry. Maybe with mysql backend
18:56.40TheCopsFXO
18:56.43justinui'd think SIP channel banks might be your best option
18:56.51Kattyi love how no one answers me. heh
18:56.57bweschkewell... you can put a couple of the new TDM2400 cards in a box, but I don't think you're going to get 10 of them in a box.. better off to do multiple boxes and use dundi or something like that
18:57.06Dr-Linuxjustinu: how can i get that module ?
18:57.08Dr-Linuxbweschke
18:57.20justinuDr-Linux: does the module exist in /var/lib/asterisk/modules?
18:57.28mog_workyeah 10 of anything is crazy
18:57.37bweschkeDr-Linux: if you've compiled and "make install"'d zaptel, now go ahead and "make clean" and then "make" and "make install" on asterisk again
18:57.39Dr-Linuxjustinu: no,
18:57.41TheCopsho my god, justinu
18:57.42pooh_justinu: /usr/lib/asterisk/modules
18:57.43TheCopsyour nice
18:57.49TheCopsjustinu, that's what I searched.
18:57.50justinu?
18:57.57justinuoh
18:58.02bweschkethe build process should then see zaptel and built app_meetme.so because it has a timing source
18:58.21Dr-Linuxbweschke: ok let me try
18:58.23bweschkeKatty: no. asterisk doesn't start to combust after 120 calls
18:58.24TheCopsjustinu, a hannel banks is very nice
18:58.35Kattybweschke: that's not what i've been told.
18:58.37Math`bweschke: MeetMe() need zaptel for mixing functions
18:58.42Math`not for timing
18:58.43justinuthecops: i think it's a better solution than putting cards into a PC
18:58.50TheCopslol
18:59.02pooh_simply test with ztdummy guys
18:59.04TheCopsjustinu, dont worry, I dont had this idea, this is a crazy idea heh
18:59.11justinueasier to maintain, you won't have to shut off the server if one breaks
18:59.31pooh_but meetme *will* load without any *zap*isch stuff arund
18:59.37bweschkeMath`: MeetMe needs zaptel to create a zap psuedo channel. the psuedo channel requires a timing source. you can't create it w/o one
19:00.04Math`bweschke: true, but still, MeetMe uses mixing functions of zap
19:00.17bweschkeMath`: yes. it does
19:00.18TheCopsjustinu, this is converting all analog line plugged into the channels banks equipement to a PRI for Asterisk ? right ?
19:00.25pooh_MeetMe *will* load without zaptel-thingy around, the error says it is NOT loaded, so it is something else
19:00.35Math`pooh_ got a point
19:00.41justinuthecops: no, more like converting analog lines to VoIP (sip signalling)
19:00.43*** part/#asterisk frenzy (n=frenzy@193.220.82.108)
19:00.53justinuSIP channel bank plugs into ethernet
19:01.10TheCopsok
19:01.16justinuwhy waste your money on a PRI channel bank? TDM is out
19:01.18FrostedAnyone know where to get an up to date list of dial parameters?
19:01.27justinushow application dial
19:01.31pooh_Frosted: voip-info.org
19:01.44*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:01.48pooh_justinu: even better :-)
19:01.48Dr-Linuxbweschke: i just did, cd zaptal dir, make clean,  make,  make install
19:01.49bweschkepooh_: meetme will not load w/o zaptel. it needs to create a zap psuedo channel to function
19:01.49TheCopsjustinu, very nice, this is a good idea, I can use Hints function to see the status of the channel
19:02.00axscode~PRI
19:02.01jboti guess pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc.
19:02.11Dr-Linuxand i have checked, this module is still not there :S
19:02.12pooh_bweschke: yes to *function* but not to *load*
19:02.28justinuDr-Linux: no no no, rebuild your asterisk source, no zaptel!
19:02.31pooh_bweschke: the error says it is NOT loaded
19:03.19bweschkepooh_: yes, I know. Dr-Linux's problem is different than zaptel being there. he built asterisk before zaptel was around so the build process didn't build the module
19:03.22axscode~h323
19:03.23jbotsomebody said h323 was An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on ...
19:03.55Dr-Linuxoooo ic
19:04.07Dr-Linuxyeah i install asteris 1st
19:04.20justinujust recompile, man :)
19:04.21Dr-Linuxjustinu: but what about my current asterisk configuration ?
19:04.24justinuthen make install
19:04.30justinuit won't disturb your config
19:04.38bweschkedr-linux: unless you do make samples which you should NOT do, it will leave /etc/asterisk alone
19:04.40axscodenot untill make sample?
19:04.44justinuyou should probably tar it up, just to be safe anyways
19:04.44pooh_bweschke: he needs to start over
19:04.53Dr-Linuxi don't wanna loose sip.conf extension... etc
19:04.59Math`ah ok he didnt follow the order on the site
19:05.01bweschkeu won't
19:05.06justinudr-linux: backup time
19:05.06bweschkejust don't do "make samples"
19:05.07Math`Dr-Linux: make install , u wont loose it
19:05.14axscodecp -vrf /etc/asterisk* ~/
19:05.20Math`Dr-Linux: I do that all the time when cvs update -dP'ing
19:05.22bweschke"make install" will only take the asterisk bin and compiled modules and move them to the build dir
19:05.25justinuyou won't lose it, but if you're afraid of losing your configs, you should back it up
19:05.40justinuwhat if your hard drive dies in 30 seconds?
19:05.50blitzragemine died last night
19:05.57justinuthat sucks
19:05.58Dr-Linuxoo okey cool let me do it
19:06.04*** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net)
19:06.05justinulose all your porn? :P
19:06.42*** join/#asterisk santiago (n=santiago@208.195.215.124)
19:06.53Math`justinu: yeah that sucks
19:08.05justinulifetime of the component
19:08.08justinu2 seconds :)
19:08.14justinuin the case of IBM Deathstars
19:08.20Dr-Linuxi'm comiling sterisk again now
19:08.22*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
19:08.36axscodehow to install addons?
19:08.47pooh_axscode: download it and complie ;-)
19:08.55axscodemake;make install ?
19:08.59pooh_yup
19:09.09pooh_but configs have to be copied over manually
19:09.18axscodeohh. got errors.. where should it resides..?
19:09.33axscodeinside /usr/src/asterisk/asterisk-addons <--should be fine?
19:09.35pooh_axscode: make sure you have apache-devel installed if you want to use mysql backend
19:09.43axscodehmmm ok..
19:09.55pooh_'/usr/src/asterisk-addons
19:10.23Dr-Linuxoo yes i got now that
19:10.24Dr-Linux[root@RHEL-TAC-TEST modules]# cd app_m
19:10.24Dr-Linuxapp_macro.so      app_meetme.so     app_milliwatt.so  app_mp3.so
19:10.48justinudr-linux: *clap*
19:11.04axscodelol
19:11.20Dr-Linuxjustinu: but sir everything is if call conference works :P
19:11.28justinuit'll work
19:11.38axscodepooh.. can I just instlal res_mysql..?
19:11.53pooh_Dr-Linux: post a conference room number and we will test :P
19:12.14pooh_axscode: make install will do that for you
19:12.15axscodewhat's the difference between res_mysql and res_config_mysql ???
19:12.36pooh_axscode: res_mysql = telling asterisk how to use mysql
19:12.50pooh_res_config_mysql = telling * mysql is available
19:12.53oogleis anyone else having problems compiling res_config_odbc in head?
19:12.57Dr-Linuxpooh_: its locally
19:13.01Dr-Linuxjustinu: same problem
19:13.07justinulol
19:13.11axscodeso I really need both? or res_mysql is just fine..
19:13.16pooh_Dr-Linux: wel....INVITE us over!!!!!!!!!!!!! and party!
19:13.25pooh_axscode: both
19:13.30axscodeo sad
19:13.31justinuDr-Linux: well, now we need to figure out why app_meetme won't load
19:13.45justinuDr-Linux: is the ztdummy module loaded in the kernel? lsmod to find out
19:15.18*** join/#asterisk zeedo (n=zeedo@80.68.92.188)
19:15.18axscodewhat should be in my /etc/zaptel.con for ztdummy?
19:15.22ManxPowerZaptel needs to be installed BEFORE Asterisk is built/installed or chan_zap, app_meetme, etc won't be built
19:15.27pooh_axscode: nothing
19:15.33Dr-Linuxjustinu: i load the before doing this >> modprobe ztdummy
19:15.37justinumanx: we already resolved that problem
19:15.43pooh_axscode: just modprobe zaptel and modprobe ztdummy
19:16.04axscodegut problem with modeprobe ztdummy though..
19:16.04Dr-Linuxjustinu: its shows
19:16.05Dr-Linux[root@RHEL-TAC-TEST modules]# lsmod
19:16.05Dr-LinuxModule                  Size  Used by    Not tainted
19:16.05Dr-Linuxztdummy                 2464   0  (unused)
19:16.05Dr-Linuxztdynamic               8544   0  (unused)
19:16.05Dr-Linuxzaptel                179872   0  [ztdummy
19:16.09pooh_axscode: comment out *ALL* lines in zaptel.conf
19:16.17justinuDr-Linux: ok, then modify logger.conf
19:16.23justinuDr-Linux: uncomming the line for "full"
19:16.24pooh_Dr-Linux: is on his way
19:16.24axscodeall?
19:16.33justinuDr-Linux: restart asterisk, and check /var/log/asterisk/full
19:16.38pooh_axscode: yes, unless you have a real zap device
19:17.16axscodedon't have a device
19:17.25pooh_axscode: uncomment all stuff then
19:18.24axscodenice.. twas loaded..
19:18.29axscodegreat help
19:19.12axscodebut I have to search for rpm of apache-devel
19:19.14axscodefor SuSE 10
19:19.34Dr-Linuxjustinu: there is no full  file in the /var/log/asterisk/  dir ?
19:19.37*** join/#asterisk SplasPood (n=sp@paravolve.net)
19:19.41Dr-Linuxthis is my CLI> error
19:19.42Dr-LinuxNov  9 00:18:20 WARNING[2561]: pbx.c:1293 pbx_extension_helper: No application 'Meetme' for extension (default, 1112, 1)
19:19.42Dr-Linux<PROTECTED>
19:19.50justinuDr-Linux: yeah, that's not enough
19:19.59justinuDr-Linux: did you restart?
19:20.04justinuDr-Linux: after moding logger.conf
19:20.12Dr-Linuxjust i reload ?
19:20.20justinuDr-Linux: no, stop now, and restart
19:21.05Dr-Linuxoo i did
19:21.12Dr-Linuxi restart
19:21.18Dr-Linuxand i can see new things on CLI
19:21.21Dr-Linuxcheck your pvt
19:21.38justinuok, so it looks good
19:21.41justinuapp loaded
19:23.06Assidis there anything we can do for echo?
19:23.13Dr-Linuxjustinu: when i diald , it says "currently you are only person in the conffernece :P ;)
19:23.28justinuthen it's working
19:23.33justinuenjoy
19:23.57pooh_justinu: job well done, cheers
19:23.58Dr-Linuxgreat justinu, i won't forget , if i face same problem next time
19:24.03justinulol
19:24.11Dr-Linuxjustinu: i think problems are good to learn
19:24.13justinunow, which one of you is going to help me get SER working with mysql? :P
19:25.03justinumaybe i should try openser
19:25.08Dr-Linuxjustinu: i can help you will apache, php, hosting, dns MySQL 4.1.xx etc  but can't with asterisk :P
19:25.14Dr-Linuxi'm newbie
19:25.42Dr-Linuxjustinu: one thing still remaining,  thats just info nothing else
19:26.51justinuwhat?
19:26.56justinugo ahead
19:27.00Dr-LinuxThanks
19:27.04justinui stayed home from work today to help you guys :P
19:27.12pooh_rofl
19:27.20axscodebeb
19:27.29pooh_BOB
19:27.32axscodehehe.. I got a probs installing addons
19:27.57axscodeapp_saycountpl.c: In function ‘sayword_exec’:
19:27.58axscodeapp_saycountpl.c:101: warning: incompatible implicit declaration of built-in function ‘sscanf’
19:27.58axscodemake: *** [app_saycountpl.o] Error 1
19:28.02delmar(yawn) morning everyone
19:28.30delmarpoooo stinky :P
19:28.41pooh_axscode: do you have your compile environment all together?
19:29.04bweschkepooh_ : no this is a bug. axscode: one moment, I'll give you the URL for the patch
19:29.20axscodeo.. nice..
19:29.22bweschkeaxscode: http://bugs.digium.com/view.php?id=5651
19:29.27axscodethanks
19:29.30Dr-Linuxjustinu: my boss is in US, i'm in pakistan, he got 2 Cisco 79xx _new_  i wanna configure them
19:29.33Dr-Linuxhow can i do that
19:29.42justinuum, they need the SIP software
19:29.50Dr-Linuxeven i never seen these fone in real
19:29.55delmarhey so im trying ti impliment the last variation found here http://www.voip-info.org/wiki-Asterisk+tips+911 to handle emergency dialing.... but it doesnt work... it just loops out... i think it thinks the Zap is busy when its not... anyway... what i have is the same as the 3rd variation found here http://www.voip-info.org/wiki-Asterisk+tips+911 can anyone see anything wrong with that?
19:30.08justinuDr-Linux: do you know if they have the SIP software loaded on them?
19:30.10pooh_bweschke: good catch
19:30.15Dr-Linuxyeah, justinu, but whats appropriate way to do that remotely?
19:30.21TheCopsjustinu, the VoiP gateway with FXO to SIP is very great, but very expensive vs the FXO to PRI interface and TDM...
19:30.38Dr-Linuxjustinu: they are just new, i don't thing the software is there
19:30.38justinuTheCops: really? hmmm how much more?
19:30.43pooh_delmar: /me is not living in 911 area
19:30.45*** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com)
19:30.46TheCopsduh
19:30.48TheCopsjust closed the document
19:30.49TheCopslol
19:30.50axscodebweschke: thank you.. how to apply this patch? what's the command?
19:30.53Dr-Linuxhttp://cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip1
19:30.53justinuDr-Linux: well, your boss needs a TFTP server.
19:30.54TheCopssorry
19:30.57Dr-Linuxhave a look please
19:30.58heroine°O°  <-- mickey
19:31.00delmarpooh_, that is irrelevant
19:31.07pooh_axscode: take a look at the patch, it is 1 adjustment only
19:31.10delmarpooh_, the actual number to dial doesn't matter
19:31.20delmaru can change it to 111 or whatever
19:31.22pooh_delmar: pastebin the trouble
19:31.25delmarwhere are u pooh_ ?
19:31.29pooh_delmar: NL
19:31.31axscodeok have to find the file
19:31.52justinuDr-Linux: well, first of all, you need to have the cisco SIP software image, then you load it onto a TFTP server, then you tell the phone to upgrade
19:32.10justinuDr-Linux: there are instructions on cisco's site, but you need the image, which costs money.
19:32.22axscodecan't find it.. the <stdio.h> is added?
19:32.49pooh_axscode: yes
19:33.05axscodestill same
19:33.31pooh_axscode: output pls?
19:33.41axscodenstall: cannot stat `app_saycountpl.so': No such file or directory
19:33.42axscodeinstall: cannot stat `cdr_addon_mysql.so': No such file or directory
19:33.42axscodeinstall: cannot stat `app_addon_sql_mysql.so': No such file or directory
19:33.45axscodebad..
19:33.59justinubring me some modules, stat!
19:34.02pooh_it can't find the files?
19:34.09axscodeguess..
19:34.11axscodeso
19:34.14delmarOK so the example I am using and the output are at http://pastebin.ca/28025
19:34.43Dr-Linuxjustinu: i'm already using many Cisco products, like PIX, Swithces and Routers etc, i never bought any image with money, i have logins, but i'm not sure about SIP images .. so tell me what should i tell to me boss, bcoz he said configure it
19:35.15axscodepooh the file. with .o exist..
19:35.29axscode<PROTECTED>
19:35.30pooh_delmar: it fails at line 9 already
19:35.34Dr-Linuxshit
19:35.48Dr-Linuxhow i can left the conference room, i'm using x-lite?
19:36.15pooh_delmar: dunno how your trunk is set up
19:36.33pooh_delmar: ah, zap/17
19:36.33delmarpooh_, so u reckon it's trunk setup?
19:36.37pooh_yup
19:36.42axscodepooh: http://pastebin.ca/28028
19:36.47pooh_delmar: chan is NOT available
19:37.07delmarpooh_ thats the example... from the web page.. the emergency trunk = Zap/g2
19:37.49pooh_axscode: you need mysql-devel
19:37.56axscodehmm ok
19:38.13delmarpooh_ should be Zap/g1 perhaps?
19:38.15pooh_delmar: you defined a specific channel zap/17 so you have 17 channels at least ?
19:38.16Assiddamn
19:38.21Assidi cant find anything regarding this echo
19:38.25pooh_delmar: try zap/gx
19:38.30hardwirewow
19:38.32Assidanyone know what can be done regarding echoes?
19:38.32*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
19:38.36hardwirechanisavail kills my asterisk server
19:38.48pooh_Assid: cover your ears ;-)
19:38.50axscode./mysql  Ver 14.12 Distrib 5.0.15, for pc-linux-gnu (i686) using readline 5.0
19:38.55Assidhaha
19:38.57axscodeis there a version for mysql-devel ?
19:39.01Assidseriously tho
19:39.04Assidwhat can we do abt it
19:39.22pooh_axscode: depends on your distro, it is a distro thing not *
19:39.37pooh_Assid: too little info atm from you
19:39.49delmarpooh_ just noticed most other stuff is Zap/g1 so ill change it to Zap/g1
19:40.14pooh_delmar: depends on your zap cards and configs, but give it a try
19:40.26Assidgot sipura 841 ip phones.. it has a horrible echo issue which is pretty constant
19:40.45delmarpooh_, yeah that was the bugger
19:40.49pooh_Assid: turned on echo cancellation on the phone itself ?
19:40.56pooh_delmar: good
19:40.57Assiddont see it
19:41.07pooh_Assid: webinterface to the 841
19:41.11*** part/#asterisk oej (n=Olle@apollo.webway.se)
19:41.14Assidyeah.. im searching there
19:41.18synthetiqis it possible to execute shell commands in the dialplan?
19:41.24pooh_Assid: advanced
19:41.35pooh_synthetiq: yes, use the system command
19:42.27delmarOk so another problem, I have a TDM400 with 1FXO for PSTN... and for some reason in the last few days.. possibly since I updated to HEAD, I get noticable echo in an incoming call, usually much more at the start of the call but lessening.  I used to get this with an X100 card but much worse and gave up for the TDM400 and was happy with it until just recently...what's up with this anyone know?
19:43.10pooh_delmar: check your zaptel.conf and check settings, set traing to on, echocanellation=yes
19:43.49delmarpooh_, when i had the X100, i basically had a crash course in everything to do with echo settings.... all that stuff is setup. here ill tell ya the settings...
19:44.10pooh_delmar: pastebin
19:44.18delmarpooh_, oh and those settings are in zapata.conf not zaptel
19:44.21axscodeif I did modprobe zaptel and modprobe ztdummy earlier.. do I have to do it in my next reboot?
19:44.40pooh_axscode: yes
19:44.50pooh_axscode: unless you make it automagically
19:45.03axscodeok.. ill add it somewhere..
19:45.04pooh_axscode: play with rc.d or rc.local
19:45.20axscodeI don't know if SuSE has a rc.local
19:45.31pooh_axscode: or even with asterisk start/stop script
19:45.31axscodethough I can add it already to my init.d
19:45.51axscodeyupz I made one from skeleton
19:46.13Assidpooh_: silence supression takes care of that?
19:46.36pooh_Assid: please explain
19:46.44*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
19:46.53generalhanwhats up everyone ?
19:46.53Assidwould silence supression take  care of echo cancellation?
19:47.01pooh_Assid: nope
19:47.08pooh_Assid: let me check
19:47.11justinuassid: who hears echo?
19:47.15delmarpooh_ http://pastebin.ca/28030
19:47.17Assidthe other end
19:47.19generalhanwhats the command to reboot asterisk ?
19:47.27enemywhen trying to call a group Dial(ZAP/g3) , then I get app_dial.c:805 dial_exec: Unable to create channel of type 'ZAP' ..... could anyone point me in the right direction?
19:47.28generalhani need to shut it all down and bring it back up
19:47.36delmarpooh_, zapata is an edited example and i leave all the stuff in there to help me know whats what :P
19:48.32justinuassid: try turning down the volume on the spa841
19:48.58bweschkeenemy: what zap channels have you defined in group3?
19:49.39delmarpooh_, so as u can see echocancel=yes, echocancelwhenbridged=yes, wchotraining=800 to give it more echocan time at the start of the call, rxgain/txgain have been tweaked according to the local Telco recommendations until their service no longer said things like.. increase dB by X.X etc...
19:50.14pooh_delmar: what the heck are you doing with all those [context] entries?
19:50.20delmarpooh_, the anoying thing is.. ok we have a service here that allows us to send a Fax.. and it will be sent back to us with dB level adjustment readings.... so the testing is done while you send a Fax.
19:50.59delmarpooh_, there is only 2...
19:51.06Assid- decreases it right ?
19:51.06*** join/#asterisk gman55s1f (n=joe@128.100.33.138)
19:51.43delmarpooh_, dring1 makes sure that is recognized at inbound-FXO, dring2 is obviously fax, and the default for the channel...
19:51.44gman55s1fquestion:  I've upgraded to the latest amportal, which seperates users and devices..  not all voicemail extensions say user is "on the phone" instead of "unavailable".. any ideas?
19:51.57delmarpooh_, so 3 contextx... 1 default, and 1 for the 2 distinctive rings
19:51.58asteriskmonkeyanyone know the cause of echo when there is 2 iaxys talking ot each other?
19:52.00pooh_delmar: Let me take another loo
19:52.10gman55s1fcorrection, *now* all voicemail extensions say the above..
19:52.25Igbothom_IIII wondered who stole all of those toilets!
19:52.28delmardring1context=inbound-FXO
19:52.28delmardring2context=fax
19:52.28delmarcontext=inbound-FXO
19:52.38delmarpooh_ those are the only contexts available
19:52.55*** part/#asterisk santiago (n=santiago@208.195.215.124)
19:54.07delmarpooh_ anyway as I was saying, we can send a fax for testing of the call dB levels, and a fax comes back telling us what to adjust... I have the correct levels set and the Telco no longer says they need adjusting, yet I cannot receive a fax, it gets killed pretty quick, which would suggest the incoming fax call may be experiencing issues... perhaps the same echo issues I get when I receive an audio call?
19:54.19pooh_delmar: pls clear out zapata.conf with *only* relevant lines
19:54.37delmarpooh_ lol ok :P
19:55.18*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)
19:57.15*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
19:59.57delmarpooh_, doh.
19:59.59justinulol
20:00.03delmarpooh_, almost done tidying this up
20:00.12pooh_delmar: about time!!!!! ;-)
20:00.33bweschkeasteriskmonkey: what does that environment look like?    iaxy <-------> * <----------> iaxy ?
20:00.55pooh_bweschke: nice?
20:01.37delmarpooh_ http://pastebin.ca/28033
20:01.42asteriskmonkeybwecheke: looks like this iaxy ---> internet ---> asterisk <--- internet ----- iaxy
20:01.50bweschkek
20:01.52asteriskmonkeyso yes pretty much
20:01.52axscodeinstall: cannot stat `app_addon_sql_mysql.so': No such file or directory
20:01.52axscodeinstall: cannot stat `res_config_mysql.so': No such file or directory
20:01.52axscodemake: *** [install] Error 1
20:01.54pooh_lloks=looks
20:02.08bweschkewhat are the iaxy's hooked up to? just regular analog phones?
20:02.13asteriskmonkeybewschke: yes
20:02.33delmarpooh_ i dont think rxwink=300 and switchtype= even need to be in there .. for an analog PSTN FXO
20:02.40asteriskmonkeythe connections have really low latencys and no jitter either so its hard to figure out
20:02.49delmarpooh_ but they are there...
20:03.04bweschkei don't think it's latency or jitter related at all...
20:03.14pooh_delmar: what is the *exact* problem again ?
20:03.16axscodemake: shared: Command not found <-- what error is this???
20:03.19asteriskmonkeynor do i as its a mint network
20:03.29asteriskmonkeybewschke: whats the next thing to look at
20:03.39bweschkeany chance it's acoustic? - eg. the speaker voume is finding it's way back around to the mic?
20:03.43pooh_axscode: the programm has not been found
20:03.48asteriskmonkeybweschke: no
20:03.53axscodesahred? what program is shared
20:03.57delmarpooh_, I can hear echo of myself, more noticable on an incoming call.. outgoing seems fine.
20:04.02asteriskmonkeyits been used on many different handsets and even digium got an echo
20:04.13delmarpooh_, and i think this is also why incoming fax's are failing to receive but seem to send ok
20:04.28bweschkeu called digium support on it?
20:04.32delmarpooh_, if it was to do with rxgain or txgain... it would be the txgain that is the cause correct?
20:04.40pooh_yup
20:04.42*** join/#asterisk shido6 (i=shido6@d221-68-216.commercial.cgocable.net)
20:04.52pooh_delmar: lower training value
20:04.53asteriskmonkeybweschke: they said they had no idea tried to blame it on network and sell me support
20:04.55delmarpooh_ trouble is, if i lower this, the exchange tells me that the dB this end is too low
20:05.14pooh_uncomment echotraining
20:05.34delmarpooh_ and i thought this TDM400 thing had a built in DSP echo can, i shouldnt even have this problem
20:05.34pooh_delmar: tell != result
20:05.43pooh_delmar: practice == reult
20:05.43delmarpooh_ it is... echotraining=800
20:05.57bweschkedelmar: no, tdm400's have no built in hardware echo can
20:05.57mog_workdsp is done in software
20:06.00pooh_delmar make it 400 and uncomment the echotraining, try again
20:06.01delmarpooh_ it is... echotraining=yes is the same as echotraining=400
20:06.09shido6txgain and rxgain incrememnts of 2 will help but if ur faxing , good luck :)
20:06.27mog_workno rx and tx gain
20:06.28delmarbweschke, so the echocan on a TFM400 with 1FXO is the same as an X100?
20:06.32mog_workuse the kb1 or mg2 echocan
20:06.36delmarbweschke, whats the difference then?
20:06.42mog_workno card is differnet
20:06.45asteriskmonkeybweshcke: no idea? what to do next?
20:06.47bweschkedelmar: no. the card is not the same
20:06.53mog_workfxo module is better than the x100p
20:06.53bweschkeasteriskmoney: one sec.
20:07.03delmarpooh_, that will lower the echotraining time.... should make it worse if anything.
20:07.05bweschkemog_work: any thoughts on the iaxy?
20:07.16pooh_delmar: only the *new* digium cards have this 'feature'
20:07.26delmarbweschke, obviously .. they are physically different....
20:07.54shido6rx and tx deal with the db
20:08.10shido6gain
20:08.14delmarbweschke, other than that.. what is the difference? I mean.. i HAD an X100 and it sucked.. i made no software changes.. ripped it out.. stuck a TDM400 in there.. loaded wctdm instead of wcfxo, and it worked.. mint
20:08.33bweschkedelmar: it's a different chip completely
20:08.44gman55s1fall my voicemail extensions now say user "is on the phone" instead of "is unavailable" .. any ideas?
20:08.47delmarbweschke, right, so there are different features....
20:09.07delmargman55s1f, change the recording?
20:09.22blitzragesomeone was complaining about echo on an iaxy to iaxy connection...
20:09.33bweschkeblitz: ya, asteriskmonkey
20:10.08delmarpooh_ ok so change to echotraining=400 anything else?
20:10.27*** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it)
20:10.33pooh_delmar: yeah, save and RESTART
20:10.46shido6Voicemail(u${EXTEN}) and b${EXTEN} are different :) u is unavailable and b is busy / on the phone
20:10.52blitzragebweschke: him too? Someone was saying he was having problems on the on-asterisk mailing list
20:11.08blitzrageunless its the same person :)
20:11.09delmarpooh_ what do u mean only the *new* cards have a built in DSP etc? I was told this one would have... and this was one of the benifits of the TDM400 etc?
20:11.35pooh_delmar: check www.digium.com and look at the specs
20:11.38mog_workthe 2400p has echo can module you can get
20:11.40delmarpooh_ it certainly seemed pretty damn fine to start with... but something has changed... i suspect in software....
20:11.47bweschkedelmar: tdm2400 has an optional echo can, the 400's do not
20:12.06shido6http://pastebin.ca/28035
20:12.22bweschkedelmar: does the echo stop if you roll back?
20:12.25*** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net)
20:12.34delmarbweschke, what does that say for their products then... that they need to make an additional echocan hardware option.
20:12.43ronaldl79Hello
20:13.00pooh_ronaldl79: everybody say HI to ronaldl79 ;-)
20:13.08delmarbweschke, in future I will copy my original CVS out of the way before upgrading so I can easily.. roll back :P
20:13.09bweschkedelmar: what does it say for their products? I think it says that they're making products to accomodate for difficult echo situations where the software dsp doesn't suffice.
20:13.28ronaldl79Regarding number porting: Are your clients able to maintain their phone listing in the telco's White pages? I've been trying for two weeks to get an answer about this from Qwest, to no avail. And they still don't know as of today.
20:13.32delmarbweschke, its not like its a huge corporate business here... just my little home-buz box :P so no big deal
20:13.34mog_workbut very rarely does the software dsp not suffice
20:13.35pooh_delmar: do not forget your handset, echo is many things related
20:13.44mog_workthe software echo can rocks delmar
20:13.47delmarbweschke, yeah i agree I guess.
20:14.17delmarpooh_, SIP to SIP calls and so on have no echo at all.
20:14.28bweschkeecho can come from a number of places and be caused by a number of different things.. if you've just got one or two fxo lines, the software echo can and the advances it's made are a cool deal
20:14.31mog_workunless the phone sucks
20:14.54delmarpooh_, and outbound SIP to PSTN is fine. i mean.. if u try hard enuf to echo and scream down the phone u can hear something but.. yeah.. outgoing calls are great.
20:15.04mog_workanywhere there is analog echo can happen
20:15.07bweschkebut when you start look at quantities of lines (eg. DS1's / E1's / 24 port FXO/FXS cards) you're starting to ask the processor to work hardware on the CPU to accomodate echo canceling for all those ports
20:15.14bweschkethat's why you get hardware echo can options there
20:15.27delmarpooh_, ok ill test this a sec. brb
20:15.45blitzrageronaldl79: ours will be able to via Verizon within a couple of weeks.
20:15.51axscodeanyone please login to my asterisk
20:16.05axscodesip:10001:10001@203.213.217.122
20:16.07asteriskmonkeybwesche: no idea yet?
20:16.20ManxPowerHmm... for $560/month I can become a mini-ISP and mini-CLEC.
20:16.22axscodesip:10002:10002@203.213.217.122
20:16.30ronaldl79blitzrage: Is the VoIP provider you're converting a client to handling the white page listing, or Verizon is simply maintaining it because the number hasn't changed?
20:16.34bweschkeasteriskmonkey: I really don't know.. I'm sorry. the echo can functionality on the iaxy is pretty closed.. there's not much you can tune once that gets to * as an IAX stream.
20:16.39axscodesip:10003:10003@203.213.217.122
20:16.52axscodeanyone please... I just want to know
20:17.11bweschkewhat would be interesting to see asteriskmonkey would be to take iax soft phones at those same locations where the iaxy's are at and see if the the echo is still there
20:18.01asteriskmonkeycould you reccommend a good one to try?
20:18.39delmarpooh_ ok outbound calls, there is slight echo at the start of the call but is quickly dealt with within about 5seconds or so.....and.....
20:19.01delmarpooh_, ok i was messing with my dialplans alot yesterday and didnt finish so incoming is br0ked.. let me fix it. lol
20:19.15asteriskmonkeybeeschke: whats a good soft iax i could use? ie site link or name :)
20:19.28blitzrageronaldl79: we're creating a portal where the manager of the DID (client) can update their E911 and 411 listing -- then we parse it, update our gateways, and submit the updates to Verizon -- and they do whatever they do with that
20:19.43bweschkeasteriskmonkey: one moment. trying to find a URL for u
20:19.54asteriskmonkeythanks
20:19.55*** join/#asterisk logicalonline (n=Ken@atlantis.clearshout.com)
20:20.00sjobeck99hey blitzrage
20:20.08zoahey blitzrage
20:20.14blitzragesjobeck99: I don't know who that is
20:20.19blitzrage:D
20:20.24blitzragesjobeck99 / zoa:  zup guys!
20:20.26sjobeck99jason@sjobeck
20:20.28zoablitzrage: what provider do you work for ?
20:20.33blitzragesjobeck99: I know who you are :)
20:20.47zoaah the ones from astricon ?
20:20.47bweschkeasteriskmoney: linux or windoze?
20:20.50zoatrue
20:20.52zoayou told me that
20:20.53zoalol
20:20.54zoa:)
20:20.56asteriskmonkeywindozr atm
20:21.01blitzragezoa: yep, those guys
20:21.26*** part/#asterisk logicalonline (n=Ken@atlantis.clearshout.com)
20:21.30bweschkeasteriskmonkey: this this one http://asteriskguru.com/tools/idefisk_beta.php
20:21.50zoabweschke: that one also exists for linux
20:21.56zoabut its not online yet
20:22.08bweschkezoa: good 2 know
20:22.26asteriskmonkeybweschke: is there a good windows sip client also that you would recommend?
20:22.38bweschkeX-Lite?
20:22.40zoaxlite is probably the best one now
20:22.40bweschkeSJPhone?
20:22.51denonanyone know if Nathan Pralle is ever on irc?
20:23.26blitzrageX-Lite!
20:23.27ManxPowerAll softphones suck
20:23.32blitzrageManxPower: true :)
20:23.36justinusoftphones do suck
20:23.38ManxPowerX-Lite seems to suck less, according to some people.
20:23.41demetriois it possible to retrieve from within asterisk the value of a SIP header?
20:23.51justinuprobably because of the audio latency in windows
20:24.02bweschkeasteriskmonkey: try that. if your echo still exists from the softphones through the same connections and * server, then we've got to look elsewhere than the iaxy's
20:24.07ManxPowerWell, I finally called some CLECs for quotes on a location 11 miles from the nearest CO.
20:24.16Augheymy vote is SJPhone
20:24.54zoasjphone gives too much shit to install
20:24.58zoademetrio: it is yes
20:24.59justinuT1 quotes?
20:25.02*** join/#asterisk shmaltz (n=chatzill@69.28.255.210)
20:25.15ManxPowerjustinu, Yup.
20:25.20justinuand?
20:25.29demetriozoa: how is it done? any link?
20:25.32ManxPowerjustinu, seems to be around $500/month or so.
20:25.38bweschkezoa: my version of x-lite didn't have the transfer button working. dunno if that's because it's an eyebeam/x-pro thing only... that's why I use SJPhone too
20:25.42justinuthat's actually not as bad as I thought
20:25.49ManxPowerWhich is more than I really want to pay.  I'll find ways to reduce the cost.
20:26.12ManxPowerjustinu, $546/month for 4 voice lines and 512K interent.
20:26.16zoathats only working for the pro version
20:26.17bweschkeManxPower: ya, that's actually pretty competitive. that's PRI including the monthly loop, no?
20:26.17ManxPowerOf course, I don't need 6 voice lines.
20:26.28justinumanx: ack
20:26.30zoacompetitive ?
20:26.39ManxPowerAbout $400/month for just 512K internet OR 384K NNI FrameRelay
20:26.44zoai pay 250$/month for 30 lines
20:26.47ManxPowerbweschke, Yeah.
20:26.47zoaalthough i dont get internet
20:26.50justinuhow much of that is loop cost?
20:26.59ManxPowerIt's still expensive for just me working out of my home.
20:27.05zoahmm yeah it sounds good anyway
20:27.25ManxPowerThey don't break out loop cost
20:27.44bweschkemanxpower: is there not the normal compliment of broadband options (cable, dsl) where you're at?
20:27.51ManxPowerbweschke, no.
20:28.03bweschkemanxpower: that stinks.
20:28.07asteriskmonkeybweschke: cant get any audio out of this ide program
20:28.08ManxPower11 miles from CO means no DSL, and there is no local cable service.  This location is RURAL.
20:28.34ManxPowerThe nearst town has a population of 6,000 people and is 11 miles away.
20:28.44justinuwhat do they want to have 1.5meg internet?
20:28.48bweschkemaxpower: yep - i hear ya... and honestly, 11 miles from the CO for only $500/mo on the DS1, you're getting a REALLY good price. that's not a short loop the clec has to pay for access
20:28.52ManxPowerBut that's part of the reason I like the location
20:28.57shmaltzManxPower, you shlould look into 512k satellite
20:28.58ManxPowerjustinu, "they"?  No, me.
20:29.13ManxPowershmaltz, SSH via satellite is banned under the Geneva Convention.
20:29.15justinuno, what's the lec charging for a full DS1 IP link
20:29.18ManxPower(or it should be)
20:29.24bweschkeasteriskmonkey: which program? idefisk?
20:29.32shmaltzlol
20:29.33asteriskmonkeynever mind :
20:29.34ManxPowerjustinu, I don't know, I'm not getting quotes for a full T
20:29.38asteriskmonkeyjust takes a second to wake up
20:29.57shmaltzManxPower, I heard from others that it is not so bad anymore
20:30.19ManxPowerMy other option is, of course, sat for web and dialup with tuned MTU/MRU/MSS for SSH
20:30.32zoaasteriskmonkey: if you have a problem with it, find me and we will help you or fix it for you if needed
20:30.39zoabut it should be stable
20:30.50ManxPowerThere is someone near the location with DirectWay internet via Sat, so I'll see how SSH runs over that.
20:31.06justinulatency is gonna be a bit annoying
20:31.42ManxPowerohad, the 512K internet/ 6 voice includes PRI and 600 voice LD mins, and 5 cents/min after that.
20:31.50*** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com)
20:32.02bweschkei've gotta imagine you're going to be hating life doing ssh over a sat link
20:32.15ManxPowerbweschke, I don't consider it a valid option.
20:32.31_Sam--hi i dont mean to sound like an ass (it comes naturally) but doesnt anyone actually pay royalties for their music on hold?
20:32.52_Sam--im just wondering what the costs to ASCAP, CSAC and BMI would be to legally play MOH
20:32.54ManxPower_Sam--, Yes.
20:32.57bweschke_Sam: the moh provided with * is provided free so no commercial royalties are required
20:33.13ManxPowerThere is royalty free music too.  You just pay a one time fee.
20:33.17_Sam--there is music that comes with it?  i must have overwrote it
20:33.18bweschke_Sam: but if you introduced your own, sure royalties would apply
20:33.29_Sam--im just wondering how much it would cost to paly good music, legally
20:33.48bweschke_Sam: yessir. from Freeplay Music
20:34.08bweschke_Sam: http://www.freeplaymusic.com/
20:34.14_Sam--thank you, there now
20:34.22ManxPowerI'm still waiting on quotes from a few other CLECs
20:34.39_Sam--ssh over sat?  with what, 500ms latency?
20:34.47SwK[Work]anyone running the AudioCodes MP10X's with the latest firmwares?
20:35.11sjobeck99any one know native MOH ?
20:35.41*** join/#asterisk jazor (n=jazor@u25-8.dsl.vianetworks.de)
20:35.53_Sam--what do you mean native MOH?
20:36.32ManxPower_Sam--, It's a new 1.2 feature
20:37.00sjobeck99i think it means that * doesnt use mpg123 or other players to play files, it plays the files fright from the drive itself, using whatever codecs the OS has for those files (ie: mp3, ogg, etc).
20:37.02enemyI have agents in alot of queues, how can I build a prioritized queue for each agent so that direct calls get prioritized infront of all the other queues.
20:37.13delmarpooh_ u still here?
20:37.19sjobeck99it's option in musiconhold.conf
20:37.24enemydirect calls (go into an agents personal queue)
20:37.25_Sam--enemy:  set it when they press the key for the queue in your IVR
20:37.47_Sam--exten => 1,1,SetVar(QUEUE_PRIO=10)
20:37.47_Sam--exten => 1,2,Queue(salesq)
20:38.10delmarok outbound calls, there is slight echo at the start of the call but is quickly dealt with within about 5seconds or so...incoming calls are somewhat worse.. louder echo, goes eventually but takes a while. start of call is quite nasty.
20:38.33ManxPowerdelmar, Welcome to the world of VoIP.
20:38.36enemythanks
20:38.48delmarManxPower, yeah but .. with a TDM400?
20:38.55_Sam--higher priority number = call gets through faster
20:39.12_Sam--in theory anyway, i have my queues setup with priorities
20:40.12delmarWhat echocan algs should I be using with the TDM400? ie in zconfig.h etc??
20:40.26ManxPower*grumble*  There are still no hotel rooms north of New Orleans, and no rooms in New Orleans for under $200/night
20:40.38bweschkedelmar: yessir. the software echo can is doing it's job because you're telling us the echo is eventually dealt with... but that's just hiding what's really causing the echo... have you messed with txgain and rxgain yet on these lines?
20:40.51delmarManxPower, nasty.
20:41.07delmarManxPower, that because of of storms/flooding or ?
20:41.09ManxPowerlooks like my next trip to the area will mean staying with my boss again.
20:41.29ManxPowerdelmar, It's because of 1) cleanup workers and 2) so many people without homes anymore.
20:41.59delmarbweschke, cheers for reminding me. im going to set them to 0.0 on both which is where they used to be
20:42.05delmarbweschke, then start again from there
20:42.17bweschkeoh mog_work!! what's the echo can most preferred by "those who know echo cans" these days?? :)
20:42.35mog_workmg1 and kb1
20:42.43mog_workkb1 for 90% of cases
20:42.52mog_workmg1 for the 10% kb1 doesnt fix
20:43.10delmarManxPower, yep. I can imagine it's pretty rough over there right now.
20:43.11mog_workmg1 is a little better i think
20:43.17mog_workbut it can cause issues on some lines
20:43.21mog_workso i reccomend kb1 first
20:43.37bweschkedelmar: try those for zconfig.h - u gotta be sure you have the latest CVS-HEAD
20:44.27ManxPowerdelmar, north of the lake it's fine.
20:44.30mog_workkb1 is the default as of recently
20:44.47*** join/#asterisk kb1_kanobe (n=jsmith@h24-207-96-50.cst.dccnet.com)
20:45.14zoahehe
20:45.18zoa<mog_work> kb1 is the default as of recently
20:45.19zoa* kb1_kanobe has joined #asterisk
20:45.22kb1_kanobed'gay all.
20:45.35kb1_kanobeg'day, even...!
20:46.54mog_workwho is kb1_kanobe that rocks
20:47.04kb1_kanobequick question: I have a problem with a harrasing caller on a PRI circuit. They're blocking their CallerID (ie. it's coming up as null in CDR). I would like to capture the originating number from the ISDN control messages without running w/pri debugging on all the time... any suggestions?
20:47.26asteriskmonkeythere is a function in asterisk
20:47.28zoai dont think you can
20:47.34delmarbweschke mog_work cheers, ok tx_gainand rx_gain set to 0.0 and its still there just not as loud, but you can tell it's happening.. and with rxgain/txgain set to 0.0, the dB levels this end and equalization will be all out of whack and fax send/receive will most likely no longer work.. not that fax receive worked right anyway which is probably due to the echo on incoming calls effecting the incoming fax as well...
20:47.35asteriskmonkeythat lets you ask callers for there id
20:47.36zoais it passed as a debug message ?
20:47.46delmarbweschke, so ill try some echo-can's
20:47.48zoai mean it shows the callerid in the debug messages ?
20:47.55kb1_kanobeasteriskmonkey: not an option - this is an 11 channel pri and very busy at that.
20:48.03asteriskmonkeyjust run an rule on incomming, if its null have them enter it or just send it to hangup
20:48.26asteriskmonkeyah ok..
20:48.29delmarmog_work> kb1 for 90% of cases, this seems to be the default #define already...
20:48.30kb1_kanobezoa: I presume it is being passed in as part of the IE during call setup, however I don't actually know.. let me check.
20:49.04zoait should not be passed unless you are 911 or police or so
20:49.11*** join/#asterisk freespace-in (n=special@ppp-70-227-27-183.dsl.ipltin.ameritech.net)
20:49.22zoaunless you are a carrier
20:49.40kb1_kanobezoa: end user, however we are gov't.
20:49.57delmarok so since echocan KB1 is already in use and not working... MG2 perhap?
20:49.58zoai dont think it should be there
20:50.08bweschkedelmar: try mg1
20:50.10tzangerMG2 has made quite a idfference for us
20:50.19bweschkeor MG2?
20:50.23tzangerthere is no MG1
20:50.28delmartzanger, explain.. what hardware u using?
20:50.31mog_worki mean mg2
20:50.32mog_workim sorry
20:50.44tzangerTE405
20:50.49mog_worknever trust me for exact facts
20:50.59mog_workerr specifics
20:51.10mog_workim generally right, just cant be trusted on names
20:51.17bweschkedelmar: MG2, KB1, etc - these are all software algorithms that are slightly different approaches taken to cancel out your echo
20:51.23delmartzanger, sorry.. not familiar with the model numbers.. what is it.. PSTN, Bri/Pri?
20:51.38delmarbweschke, yeah i figured that
20:51.39tzangeryup
20:51.52delmartzanger, which lol?
20:51.53*** join/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz)
20:52.06tzanger?
20:52.06delmarbweschke, i still dont see why the echocan isn't done on the board.
20:52.14patpatnzhi guys
20:52.16tzangerTE405 = quadspan T1/E1
20:52.19tzangerusing it on PRI
20:52.27delmartzanger, what is the card... pstn/fxo ? or isdn Bir/Pri?
20:52.30patpatnzdoes anyone know about ooh323c and dtmf tones?
20:52.31bweschkedelmar: because it's considerably more expensive, and for most instances, not necessary in small numbers of ports
20:52.31delmarright
20:52.45delmartzanger, i cant imagine u would have echo anyway with a good channelbank
20:52.49kb1_kanobedelmar: the earlier cards are simply protocol adapters/physical interfaces. The later cards include a 3rd party echo cancellation ASIC.
20:53.22delmarkb1_kanobe, hrm. well thats contrary to what I was told before buying the TDM400. i was told they had hardware echocan.
20:53.29tzangerkb1_kanobe: well
20:53.42tzangerT100P/TE110/TE410/TE405 all have no sw echocan
20:53.43bweschkedelmar: no. whoever told you that was mistaken
20:53.51tzangeronly the TE406/TE411 have hardware echocan module
20:53.51delmarbweschke, indeed they were
20:53.59tzangerand the TDM2400 has a variant of the module
20:54.48delmartzanger, yeah but a good channel bank has echocan built in the firmware.
20:54.48tzangerdelmar: not that I'm aware of
20:54.48tzangerI don't know of a signle "channel bank" that has hardware echocan
20:54.49tzangera lot of the good ones have adaptive hybrids to *minimize* reflected energy
20:54.55delmartzanger, i have an adit600 here and I read that it has echocan and also impedance balancing stuff which would probably take care of most problems anyway.
20:54.58tzangerbut I don't know of any that have echocan
20:55.00bweschkedelmar: there's also a whole process to go through with fxotune to work on tuning to eliminate echo.. I'm not 100% familiar with it
20:55.04tzangermy adit600 most certainly does not have echo can
20:55.24delmarbweschke, i have read that fxotune is a waste of time and doesnt work right
20:55.27bweschkemog_work: has anyone put together a howto process for fxotune and the tdm400 cards?
20:55.31tzangerthere's nothing in an adit600 that could do echocan
20:55.40tzangerexcept for, as I said, having a very well balanced hybrid
20:55.43mog_workits easy as 1 2 3 tune
20:55.55kb1_kanobeI beleive there was an ATM-based channel bank that included echo cancellation firmware because of the ATM transport delay issue. But it only works on the ATM bridged calls.
20:56.01denonI've actually seen a few channel banks that do echo cancellation, but its usually an optional module .. so not really part of the CB itself
20:56.02mog_workfxotune is generally not needed though
20:56.05delmartzanger, well since i don't have a T1 card, and cant see myself getting one until they come down in price, I will never know :P
20:56.15mog_workas 90 % of world uses 600 ohms of impedence
20:56.24tzangerdelmar: wellyes, that's an external echo can and they work with any T1/E1
20:56.33Beirdo90% of the world?  ummm
20:56.38BeirdoI think you mean North America
20:56.40bweschkeright.. I have a nack for finding that 10%
20:56.42bweschkeit seems
20:56.47mog_worklol
20:56.52mog_workthe world beirdo
20:56.56*** join/#asterisk Toadyus (n=Im@S010600121746f9fe.mh.shawcable.net)
20:56.57delmarok.. so.. .testing echo-can algs.
20:56.58kb1_kanobeI recommend mg2 and fiddling w/the echotraining parameter if you're having issues. If you _still_ have issues with all calls, fall back to kb1, though I don't think that will help.
20:57.03mog_worka lot of places use 600 ohms
20:57.04denonthat 10% is .. what .. europe? :)
20:57.16tzangerechotraining is teh suck... MG2 is teh winz though
20:57.22mog_worknot just europe just random places
20:57.24*** part/#asterisk patpatnz (n=pjs@unaffiliated/patpatnz)
20:57.30mog_worki mean we didnt have fxotune till this year
20:57.30denonnod
20:57.41bweschkethere's a LEC down in Hilton Head Island, SC who uses 600 ohms, and then when you get lines in a hunt group from them, all of a sudden the lines are 900 ohms
20:57.41delmardoes anyone in here actually know what the numbers in fxotune.conf really do ?
20:57.43mog_workand didnt have a huge demand for this ablity
20:57.46mog_workyes
20:57.50kb1_kanobe'echotraining' is a hack. Plain and simple - a variable convergence adaptation rate needs to be implemented.
20:57.50tzangerkb1_kanobe: I wanted to ask you something about norstar but I've forgotten now
20:57.51perdtzanger. you dirty bastard.
20:58.01mog_workdelmar it set s the impedance for the fxo moduls
20:58.02delmari mean.. one time I tested it a few months back it was... 1=11,0,0,0,0,0,0,0,0   and the other day I tested it .. it was 1=5,0,0,0,0,0,0,0,0
20:58.04tzangerperd: ?
20:58.05mog_workmodules
20:58.09mog_workwow
20:58.14perdlwz hoe
20:58.15mog_workwell it has gotten updated
20:58.17perdhah
20:58.19mog_workand is better now delmar
20:58.30tzangerperd: ;-)
20:58.33mog_workmust people get all 0s or the first 1 is 1 but i cant remeber to be sure
20:58.35*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
20:58.46bweschkedelmar: it's worth a shot.... improper impedance settings can certainly be a cause for bad echo
20:58.51delmarmog_work, perhaps it's the problem?  it has been updated... and fxotune might be the cause of the echo? i might try setting back to 11,0,0 etc
20:58.54perdtzanger you familiar with asterisk? i have a question that i cant find an answer for anywhere
20:59.07mog_workyou could
20:59.14tzangerperd: yeah you could say I am
20:59.22delmarok. things for me to try.. .lots of things .. for the next few mins I guess.
20:59.24delmarbrb
20:59.25mog_workare you using fxotune -s /etc/fxotune.conf after you load module?
20:59.30delmaryes
20:59.37mog_workgood good
20:59.44delmari even unload module and reload...
20:59.47delmarjust be be sure
20:59.48delmarok brb
21:00.24*** join/#asterisk flynux (i=n85yb89@pingou.in)
21:01.07perdtzanger, ok.. i have a box set up for PSTN termination but i'm only allowed one connection out.  i connect to this termination service with IAX2.  i am using nagios to generate error messages that dial out and play a message.  the problem is that if i have like 5 or 10 or 20 errors there will be 20 .call files in there and it will attempt to make 20 outbound calls at the same time
21:01.45tzangerperd: yeah it doesn't serialize that at all
21:01.48perdsomehow i need to limit the number of outbound calls over that iax2 channel.. i tried putting some SetGroup stuff in my extensions.conf, then realized that the .call file doesnt use the extensions file to dial out
21:02.00*** join/#asterisk hhoffman (n=hhoffman@tor/session/x-395008eb87af861e)
21:02.04tzangerbetter to use an agi that reads the outbound call list and tries to concatenate it into one call
21:02.17tzangeractually the callfile does indeed use extensions.conf to call out
21:02.18hhoffman<PROTECTED>
21:02.22tzangerit can't do it any other way
21:02.29perdhmmmi  can use an agi to trigger outbound calls?
21:02.36perdi thought i could only use agi for inbound crap
21:02.42tzangernope
21:02.48perdoh thats sexy
21:02.54tzangerchan_local and agi go together like sex and a condom
21:03.01tzangerwell maybe a little better than that
21:03.03perdmy penis gets hard just thinking about it
21:03.06_Sam--perd you can just drop the call info in /var/spool/asterisk i think
21:03.14_Sam--we do it here at work but i forget what we use
21:03.25axscodecan someone gimme a exten sample for meetme?
21:03.32perdyeah i can generate outbound calls without a problem, its the limiting the amount of current outbound calls that i cant figure out sam
21:03.49_Sam--cant you have your IAX provider limit your trunks?
21:04.02perdi think with agi i should be able to though, assuming tzanger is correct in saying i can generate calls from agi scripts without human interaction
21:04.11tzangerperd: yep you can
21:04.16_Sam--he's pretty bright, id believe him
21:04.17delmarbweschke, ok first test was to re-confirm set at 5,0,0 etc.. in/out .. then set to 11,0,0 etc .. not really any difference between the two... none that I could detect.
21:04.43perdi'll just have nagios dump the warnings into a mysql database and have the agi script poll it until it sees a warning then generate a call
21:04.46delmarbweschke, so I will leave it at 5,0,0,0,0 etc which is what is currently detected, .. .now onto echocan algs.
21:05.07perdthat way i can keep track of calls in the agi scripta nd be happy
21:05.13perdtzanger thank you!
21:05.26tzangerperd: np
21:05.27kb1_kanobezoa: you're correct. It's coming back as 'Presentation: Presentation prohibited of network provided number (35) ', which I hadn't noticed before... I will have to prod the telco and see how they wish to proceed because it's not as if we can grab the line the offending call came in on and dial the 'this was an offensive call' vsc.
21:05.52perdknow a good place for agi docs?
21:05.53perd~docs
21:05.56jbotrumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
21:05.58*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
21:05.58bweschkedelmar: fair enough. like mog_work said, most places use common impedence but it was worth a shot. :)
21:06.00perdhmm
21:06.18_Sam--http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI
21:06.25delmarbweschke, yeah i mean... NZ is supposed to have the same as the USA
21:06.54delmarbweschke, but even then, you just don't know whats going on.. and I dont have a sophometer to test it all :P
21:07.06_Sam--you could do it all with something like php and sql as well, perd, just setup a table that would be checked for current connections...if connections > 1 , then dont call
21:07.18_Sam--and just run the agi that checks the table before each call
21:07.47delmaru would think that the wctdm module load could check /etc/fxotune.conf and set stuff at load time.. :P
21:07.58_Sam--could probably even do it with the off the rack mysql CDR table
21:09.13_Sam--it would be probably easiest to just ask your IAX provider to limit you to 1 outgoing ( i think for me that would be easiest)
21:09.34delmarbweschke, ok so im lookin at zconfig and there are a few other optimizations which im not sure i should touch... CONFIG_ZAPTEL_MMX ? (its intel cpu). ?
21:10.05bweschkesure - if it's an intel cpu u can turn it on
21:10.54*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net)
21:11.45delmarbweschke, now this 2100 Hz echocan disable thing i always wondered about... does that turn off echocan when say.. a fax comes in? detects that kinda tone .. or what?
21:12.06bweschkeyes
21:12.18bweschkeu don't want the echo can interfering with faxes
21:12.33delmarbweschke, ok so i will try this out :P
21:12.52bweschkebut that probably won't fix your echo problem if you didn't have it on originally. :/
21:13.08delmarbweschke, no i was reading it the wrong way.. thats to turn OFF the thing that turns off the echo can LOL
21:13.12bweschkea # of folks have reported good results with MG2 in forums
21:13.25bweschkeso - that's probably one a bastion of hope
21:13.41delmarbweschke, ok then. compile time
21:13.56mog_workyes mg2 works well for people that need it
21:14.48delmarshould I re-do fxotune with the new compiled module and stuff?
21:14.56delmarthink i might anyway...
21:17.26delmarsage: fxotune
21:17.27delmar<PROTECTED>
21:17.37delmarJust looking at that....
21:17.50delmari dont know what the number is.. if any.. to "clear" dialtone off the line....
21:17.51*** join/#asterisk asteriskmonkey (n=phil@HSE-Toronto-ppp300757.sympatico.ca)
21:17.58asteriskmonkeydamn it my asterisk is going to crap..
21:18.22asteriskmonkeyi just called a client not only do i hear myself but they here them selves but neither of us can hear each other echo
21:18.39Beirdodelmar: that means "hit a key so the dialtone stops"
21:18.46Beirdojust tell it what to dial
21:18.46asteriskmonkeywas sipsoftphone > asterisk > vonage client
21:19.00delmarBeirdo, that i realise
21:19.10delmarBeirdo, but there IS no key in NZ to do that ... that i know of
21:19.19asteriskmonkeydoes anyone here have a pri?
21:19.22*** join/#asterisk _T3_ (n=rposada@200.63.231.210)
21:19.28Beirdowhat happens when you dial the first digit of a number?
21:19.34Beirdodoes the dialtone stop?
21:20.06delmarBeirdo, there is silence... then after a few seconds it will time out and give a sort of .. fast busy tone for ages.
21:20.07_T3_hello
21:20.18Beirdowell, if the test is fast...
21:20.38Beirdojust don't dial the operator
21:20.55*** join/#asterisk SERGEUS (i=sergey@195.112.98.13)
21:20.58_Sam--asteriskmonkey:  it could be firewall related
21:21.01delmarBeirdo, i will plug another phone into the line, wait for silence... then run the test.. and as soon as i head the card pickup the line ill hang up and let it have the line.. with silence.
21:21.02_Sam--search one-way audio
21:21.14_T3_anybody knows how much is tdm2400p
21:21.25dougargh.
21:21.27SERGEUSHi everybody!
21:21.30*** join/#asterisk _dave01_ (n=Miranda@21.208.65.212.contactel.net)
21:21.40delmarBeirdo, like.. u can dial 1, wait for 5-6secs.. it will do fast busy for maybe.. a minute.. then go silent. so ill do it like that sorta thing
21:21.42bweschket3: pricing is up on voipsupply.com and I believe Digium's site too
21:22.05Beirdodepends on how that test works
21:22.07SERGEUSi have a lot of crashes since NOV 02, caused by channel.c - anybody has the same problem?
21:22.12_dave01_how do I know that my Digium TE110P has new v2 firmware?
21:22.13_Sam--asteriskmonkey :  are you behind NAT on either side?
21:22.17dougNov  8 15:32:34 NOTICE[57351]: chan_sip.c:7532 handle_request: Registration from 'boob<sip:boob@asterisk.aaronsen.com>' failed for '140.221.241.239'
21:22.20dougwtf
21:22.29_T3_no in digium no, i will check in the other site thanks
21:22.32BeirdoI did an echo-cancellation algorithm that worked in that 5-6s
21:22.33dougwhy would it fail with that?
21:22.39dougand no more details?
21:22.57douglots of
21:22.58dougNov  8 15:33:17 WARNING[57351]: chan_sip.c:598 __sip_xmit: sip_xmit of 0x8120704 (len 430) to 140.221.241.250 returned -1: Invalid argument
21:23.02dougas well
21:23.07dougwith eyebeam sip client
21:23.14delmarBeirdo, oh?
21:23.25Beirdothat wasn't for asterisk though
21:23.29delmarBeirdo, added to * or not?
21:23.36Beirdoa long time back
21:23.46delmarBeirdo, did it work?
21:23.53Beirdoso I don't know how THAT test is working
21:24.02Beirdoit did for our application yess
21:24.08delmarcool
21:24.33Beirdoit was for ship-to-shore impedance matching for a shipboard PBX system
21:24.49delmarnice
21:26.41harryvvohh Beirdo, how did it work out?
21:26.55Beirdopretty well, actually
21:26.58*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
21:26.58harryvvthere is one other person who made a asterisk system for ships linked to a sat.
21:27.45Ariel_ManxPower, I may have a new customer in your area for you. Do you have some time to talk about this?
21:28.04Beirdodunno if they still use it or not, but the idea is... the first time they hook up the lines when porting at a new place, they echo-train and impedance match
21:28.21BeirdoI never got to test it in the field as I quit before then
21:28.21Beirdoheh
21:28.26ManxPowerAriel_, No time right now.  What area?
21:28.27enemyHow to set unlimited timelimit when adding to a queue in extensions.conf?
21:28.35ManxPowerI'm dealing with Cisco access lists at the moment
21:28.36Beirdobut I think there are some frigates with it live
21:28.51Beirdoand maybe an aircraft carrier still
21:29.10Ariel_ManxPower, it's not for today. But it's in the Monroe?? area
21:29.15Ariel_it's a hospital
21:29.27ManxPowerAriel_, that's about 2 - 3 hours east of me
21:29.38Ariel_Bummer
21:29.49harryvvaccess control list?
21:30.43Beirdoharryvv: http://www.drs.com/products/index.cfm?gID=18&productID=203
21:31.05*** part/#asterisk hhoffman (n=hhoffman@tor/session/x-395008eb87af861e)
21:31.15tmccraryis it possible to have two Asterisk PBX communicate via speex codec, but have the phones that register on the pbx be using ulaw?
21:31.27Beirdougly fucker, but there it is.
21:33.16delmarBeirdo, hey thats interesting.. what it does is.. picks up the line.. dials ie. 1, then starts test tones ascending in pitch only for a few seconds, then hangs up, goes off hook again, dials 1 to clear tone, then continues until finished the pitch cycle.. then picks up and dials 1 and repeats the process.. musta done it about 20 times.. anyway the test are short enuf that they would fit within the silence period
21:33.29Beirdoyup
21:33.33Beirdothat sounds about right :)
21:34.02SwK[Work]pile of shit
21:34.07SwK[Work]thats all it is
21:34.11SwK[Work]a big stinking PoS
21:34.16delmarBeirdo, however this echo test blows... the echo while off hook vs a call in progress will be vastly different.... its only testing this side of the exchange.. i dont think thats an accurate test.
21:34.29SwK[Work]any AudioCodes users around?
21:34.32Beirdothat will fix impedance mismatch
21:34.43Beirdothere's no easy way to fix remote echo
21:35.23delmarBeirdo, umm no.. at what point did it measure the line with a sophometer and work out the correct impedance settings? that was purely echo/audio testing.. nothing to do with resistance
21:35.45Beirdolocal echo is almost 100% caused by impedance mismatch
21:35.52delmari agree
21:36.07Beirdoso what it was actually testing was impedance mismatch
21:36.08delmarbut this card does NOT have a sophometer built in.. i can assure you :P
21:36.14*** join/#asterisk delox99 (n=delox99@206-248-149-59.dsl.teksavvy.com)
21:36.15delmarno
21:36.18delmarits testing audio
21:36.20delmarnot resistance
21:36.22Beirdogah
21:36.35delmarresistance = impedance.
21:36.45Beirdoyou don't need to measure the impedance to test for mismatch
21:36.55delmarheh :P
21:36.59Beirdothe test is by minimizing the echo by fiddling the impedance
21:37.23Beirdoat minimum echo, your complex impedance is matched as best you can
21:37.31delmarsee.. even u agree its not testing impedance.. its testing for mismatch by way of echo detection using an audio test.. nothing to do with resistance/impedance.
21:37.47Beirdoit is NOT nothing to do with impedance
21:37.57delmarBeirdo, ok i get what u are saying
21:38.05Beirdothat's what is causing the echo it's measuring :)
21:38.40delmarBeirdo, it cant test the resistance, but can test echo and fiddle it's own impedance matching and re-test the echo/audio to see how that works out... hence the number of tests
21:38.47Beirdobingo
21:38.49Beirdo:)
21:39.06delmarnow that makes sense :P
21:39.13Beirdothe impedance of a line is partially resistive, partially inductive or capacitive
21:39.27*** join/#asterisk citats (n=james@bgp925576bgs.brghtn01.mi.comcast.net)
21:39.43Beirdomost SLIC chips will let you tweak the impedance it presents to allow you to try to match the line
21:40.06delmarso its not testing and adjusting some software echncan or nuthin it's purely trying to illiminate echo by adjusting it's own matching to the line
21:40.16delmarwell thats better than nothing.. but still doesn't work :P
21:40.20Beirdothat would be my guess
21:40.32synthetiqis there a softphone operators console for asterisk?
21:40.38delmarim gonna run another test with the phone on-hook, see what it says, then try the new mg2 alg etc.
21:40.43Beirdothat depends on if that's how they implemented it.  that's precisely how I had implemented it
21:40.45delmarsynthetiq, yep
21:40.56synthetiqwhere can i find it
21:41.01delmarsynthetiq, CVS
21:41.20delmarsynthetiq, gastman
21:41.25Beirdoand we had the fun of using a programmable DSP to do the hard work
21:41.27delmarsynthetiq, at least I think thats it.
21:41.33*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
21:41.49delmarBeirdo, cool
21:42.10Beirdothat was... oh GOD.  almost 10 years ago
21:42.11Beirdodamn
21:42.14*** join/#asterisk it0 (n=it0@zwanebloem.xs4all.nl)
21:42.17_dave01_anyone has te110p with v2?
21:42.31*** join/#asterisk kn0x (n=nunya@tor/session/x-bc08ecc4ffc96481)
21:42.42it0wow, this channel is full!
21:42.48kn0xwhy isnt nickserv working!?!?!?
21:42.57*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
21:42.59it0kn0x: it just idéd me?
21:43.21kn0xwhoops
21:43.22synthetiqwow thats pretty nice
21:43.23kn0xnevermind
21:45.11it0i'm quite an * noob but, it's working for me, pretty well actually, still I would like to know how I can dial someone from the console and play a sound? i guess it's trivial but don't know how...
21:46.09dudesit0 - have an extension that plays the sound? dial 1234@context (if I recall correctly; haven't used console dial for awhile.)
21:46.38shido6chan_oss?
21:46.50it0dudes: so I set up an external number as an extention, then issue the playback command?
21:47.01sylechan_alsa
21:47.05dudesI'd think that'd work
21:47.17*** join/#asterisk Flauto (n=zhao@71.194.38.168)
21:47.22dudesYou could do it via a callfile in the event dial from console doesn't work
21:47.38it0what is a callfile?
21:48.05dudesvoip-info.org (you make a call file and drop it to /var/spool/asterisk/outgoing
21:48.15dudesand it places the call and drops it to an extension if it's answered
21:49.27it0thanks I'll look it up
21:53.48_dave01_how do I know that I have new v2 firmware?
21:53.54mog_workdmesg
21:54.24_dave01_TE110P: Setting up global serial parameters for T1 FALC V1.2
21:54.43_dave01_nothing about version
21:54.51mog_workte110p has only 1 version
21:54.58mog_workv2 refers to the quad cards
21:56.01_dave01_Ops. I didn't know that.
21:56.03_dave01_Thanks.
21:56.09mog_workno problemo
21:56.12_dave01_looks like v2 is for dual and quad only
21:56.20mog_workyes
21:56.37myke420247the * console isn't a full phone, is it?  you can't stick a headset in the sound card on your * box and use that as a phone?
21:57.04mog_workyes you can
21:59.18shido6yep
22:05.47*** join/#asterisk kn0x (n=root@ppp-69-222-164-89.dsl.emhril.ameritech.net)
22:06.15kn0xcan anyone assist me with creating asterisk  init startup scripts for gentoo?
22:06.26mog_workthere is one !
22:06.31mog_workin contrib
22:06.32kn0xi used the cvs build and im stuck..
22:06.32mog_worki believe
22:06.36kn0xcontrib?
22:06.40mog_workasterisk/contrib
22:06.48mog_workasterisk/contrib/init maybe
22:07.03kn0x/var/lib/asterisk/contrib?
22:07.10kn0xwat is the full location
22:07.36mog_workno its in the source
22:07.40mog_workits not installed by default
22:08.11delmarok, tried lots of stuff. this echo is just crap
22:08.23kn0xim looking through the contrib/scripts
22:08.33delmarim going to go back to a regular PBX phone system and attach Asterisk as a VoIP gateway. fuck this.
22:08.43mog_worksorry delmar
22:08.50kn0xahhaa!
22:08.53kn0xinit.d
22:08.56kn0xthanks mog
22:09.15harryvvdelmar, whats the problem?
22:09.35mog_workbe back in a bit
22:10.31harryvvdelamr having echo problems?
22:13.46delmarharryvv, sorry was afk, wife etc.
22:13.50delmarharryvv, yeah
22:13.54delmarharryvv, echo pain in the ass
22:14.08delmarharryvv, it WAS working ok
22:14.20delmarI think HEAD broke it
22:14.57ardDid you also use new zaptel drivers?
22:15.07delmarso calls outbound are fine.. still some faint echo, and holding phone up to mouth and giving it shit causes more...
22:15.14ardpart of the echo can is in the kernel...
22:15.31delmarcalls inboudn tho.. are crappy especialyl at the start but only a little improvement after 10secs.. still too much echo
22:15.36delmarard, yes zaptel as well
22:15.47ardHmmmm
22:15.55harryvvzaptel is involved with the echo?
22:16.03delmari have been using * with TDM400 for months and its been running great...
22:16.04ardmaybe you have another echo canceller selected...
22:16.21delmarthe echo is SIP to/from * via PSTN
22:16.33delmarso yes.. Zaptel FXO port on the TDM400
22:16.47ardharryvv : there are 2 echo's to cancel: the local echo which is not only done by right impedance matching....
22:17.21arddelmar : I can only tell you that I am a very happy isdn user ;-)...
22:17.48demetriowhen I call freeworlddialup test numbers like echo test or time, they will ring then answer but say nothing. Is it possible that this is becaus I'm behind NAT?
22:17.50ardI've tried writing a softphone with a headset and a modem, and I now know how painful analog lines are :-(
22:18.21arddemetrio : if you are using SIP, then it probably is...
22:18.36harryvvard i know
22:18.50demetrioard: using IAX instead will help?
22:18.51arddelmar : the kernel drivers of zaptel have special echo cancelling stuff...
22:19.07delmarrxgain/txgain is 0.0 and has been this way for months until I was playing with it the otehr day trying to get incoming/outgoing fax to work right, but its back to 0.0 again now
22:19.11arddemetrio : yes. But I heard rumours that the iax server of fwd has some problems lately
22:19.18delmarard, yeah i know.. tried kb1 and mg2
22:19.19harryvvecho is also known as impedence mismatch, swr , and who knows how many terms I have learned in the last 15 years.
22:19.22demetrioard: ok, thanks
22:19.28*** join/#asterisk bjohnson (n=bjohnson@i216-58-13-119.cybersurf.com)
22:19.44ardharryvv : but a mere impedance mismatch can be corrected by software...
22:19.53delmarim going to go back to an older Asterisk for testing... just because I can
22:19.59harryvvand hardware
22:20.00ardI was able to cancel out the local echo completely
22:20.09ardin software :-)....
22:20.26ardbut far-end echo and timing problems made it really hard.
22:20.41ard(I used the echo-canceller from speex)
22:20.49*** join/#asterisk dustyservers (n=dustyser@d198-53-254-205.abhsia.telus.net)
22:20.52ardwell...
22:21.00ardI am off to bed now
22:21.11dustyserverscan some one tell me what hard will I will need outher then freebsd box?
22:21.21dustyserversto setup asterisk
22:22.49*** join/#asterisk luite (i=luite@belphegor.deadlysins.nl)
22:22.51delmardustyservers, do u want to attach to a line?
22:22.59dustyserversyes I do
22:23.04dustyserversto my bussine phone line
22:23.06delmardustyservers, doesnt need to be freebsd, can be RH, Debian etc.
22:23.14delmardustyservers, PSTN? ISDN?
22:23.21dustyserversyuppers
22:23.23luitewhat does this error mean, when starting asterisk: asterisk: relocation error: /usr/lib/asterisk/modules/res_watchdog.so: undefined symbol: ast_load
22:23.24delmarwhich?
22:23.29dustyserversumm pstn
22:23.52delmardustyservers, then u will need an FXO card... good luck with fuckin echo
22:24.11dustyserversdose it echo bad?
22:24.26delmardustyservers, for most no, for some its a cunt
22:24.35dustyserversreally
22:24.45delmarno....
22:24.46dustyserversthat sucks
22:24.47freatdelmar: what are you like 16 years old or something?
22:24.54*** join/#asterisk _Madar (n=tophe@219-84-129-163-adsl-tpe.static.so-net.net.tw)
22:25.05delmarfreat, no im just pissed off.
22:25.29luiteah, found it, there were still some asterisk 1.0.9 files left in /var/lib/asterisk/
22:25.38freatdelmar: this is a bit more professional of a forum than some you may be used to
22:25.49freatdelmar: please try to watch the language
22:25.49delmarfreat, i wend thru all this nonsense early this year.. with an X100 card... bought a TDM400.. problems solve.d.. now they are back again because Digium are cowboyz and borke the echocan stuff.
22:26.06freatdelmar: I hear that
22:26.30dustyserversI know a few people using asterisk and they say it good
22:26.39dustyserversso is it a good chose for pbx phone systme?
22:26.41dustyserverssystem
22:26.42delmardustyservers, yep. its fantastic.. very powerful
22:26.51delmardustyservers, i mean.. the echo asside... i love it
22:26.53dustyserversok cool I will try it out then :D
22:26.57freatdustyservers: plenty of people us it in production environments with lots of phones
22:27.06dustyserversas long as it works
22:27.08dustyserversthat all I care
22:27.28dustyserversas 3com charges way too much money for there system
22:27.32dustyserversso I rather build my own
22:27.43delmardustyservers, getting it to work might be a fight.... i mean..depending on what you want to get out of it.. there is so much stuff you can do.. hence so much to learn.. lots of stuff to debug..
22:27.59dustyserversoh ic
22:28.18dustyserversdose the auto atence works good too?
22:28.19freatdustyservers: try asterisk@home to start with is my recommendation
22:28.35dustyserversthat what was mhy plan
22:28.37dustyservers:D
22:28.38delmardustyservers, just takes time.. and plenty peeps in here are guru's so.... u will get there
22:28.45dustyserversto  make sure it suit my needs
22:29.15freatif you want basic IVR and autoattendant you can do all that w/ @home until you get more comfortable at the command line
22:29.18dustyserversand do any of you use this at more then on office?
22:29.19delmari didn't go the way of asterisk @ home or any other kinda user-friendly thing.. just ripped into it manually.
22:29.22*** join/#asterisk gman55s1f (n=joe@128.100.33.138)
22:29.24oogleso what's the difference between an opteron 280 and 880 besides 1000 dollars?
22:29.41delmardustyservers, actually I have it at more than one location.. and tie calls between here and there over IAX2....
22:29.51luiteoogle: the number of hypertransport links
22:29.53freatdustyservers: I've got 3 offices each w/ about 15-20 phones
22:29.54dustyserversthen that wokrs good
22:29.58delmardustyservers, but i am having other issues with that which I haven't got to spend time on yet.
22:29.59oogleluite: thank you
22:30.01*** join/#asterisk rculp (n=rculp@66.173.240.20)
22:30.02freatnot that big but enough
22:30.10gman55s1fquestion -- after a Dial command to one of my SIP phones, the console shows "Called 510" (the extn), and then continues with a busy script..  The phone blips as if it almost started ringing...
22:30.16luiteoogle: the 880 has 3 ht links, and you can use it in 8-way smp
22:30.23delmardustyservers, it works mint apart from some echo im getting but .. there shouldnt be any echo at all so .. its a fault with my situation
22:30.29dustyserverslike I know how to setup freebsd all that
22:30.37luiteoogle: the 280 only in dual or single cpu servers
22:30.38dustyserversas that what I have for a server at the office right now
22:30.51dustyserversbut would just like to try asterisk to see how good it would work thought..
22:30.58delmardustyservers, most people want to use something that runs on linux.. and the first big problem is that they don't know anything about linux.. so u are 90% of the way there
22:31.00dustyserversI will have to definely play iwth it
22:31.03*** join/#asterisk rculp (n=rculp@66.173.240.20)
22:31.21dustyserversI played with linux some too
22:31.26delmardustyservers, just point your cvs at digium and grab it and off u go
22:31.34dustyserversoh that easy
22:31.52dustyserversI will definely play with it I will need a card to plust in my phone line into it right?
22:32.13dustyserversor can I test it withput a phone line?
22:32.18dustyserverswithout
22:32.22rculpquick question: I'm getting the following error when I try to test from xlite client to xlite client
22:32.23delmardustyservers, u can use it without a phone line
22:32.27rculpI think I missed something
22:32.29rculphere's the error
22:32.30rculpNov  8 17:16:39 WARNING[10458]: chan_oss.c:581 setformat: Unable to re-open DSP device /dev/dsp: No such device
22:32.36sylemore feature requests for billing system?
22:32.38delmardustyservers, but obviously need one if u wanna interface to your PSTN
22:32.40rculpforgive my noobness :)
22:32.50delmarrculp, nah
22:32.57dustyserversok thanks for the help
22:32.59delmarrculp, i dont have that setup either. its not needed :P
22:33.07dustyserversthat all I need to know for now..
22:33.28rculpdelmar: hrm, xlite to xlite tests are broken then. I must need to do something else. first asterix setup
22:33.46delmarrculp, br0ken how?
22:33.49gman55s1fwhen calling one of my SIP phones, the console reads Dial(... , 'SIP/510'), and then says 'Called 510', and then launches into a busy script
22:34.05gman55s1fthe phone registers a missed call but never really rings
22:34.11rculpdelmar: calls are not going through. I just think I need to do some more testing
22:34.18delmargman55s1f, then the phone was unreachable
22:34.19rculpdelmar: I edited the sip.conf file
22:34.23*** join/#asterisk brownjor (n=Jordan@c-67-185-209-222.hsd1.mi.comcast.net)
22:34.29rculpdelmar: just don't think I setup the dialplan right
22:34.39gman55s1fdelmar: it seems as if it rings for a split second..
22:34.43delmarrculp, check extensions.conf must be a dialplan issue
22:34.48rculpdelmar: testing to my test extension doesn't ring through either
22:34.52rculpdelmar: will do
22:35.28gman55s1fdelmar: plus, I'm using a dialparties script to check the line is free before calling
22:36.02delmargman55s1f, do ... show sip peers, sip debug peer 510, place a call, copy all the output from those into pastebin and lets have a look
22:36.53delmarerr.. sip show peers *
22:36.57delmar:P
22:37.10rculpdelmar: I'll read up more on extensions
22:37.18rculpand ask again for help tomorrow if I don't have it figured out :)
22:37.22rculpthx
22:37.40*** part/#asterisk rculp (n=rculp@66.173.240.20)
22:37.54*** join/#asterisk swebb (n=swebb@216.183.121.53)
22:38.05delmargman55s1f, to work out if its your script, remove it from the picture.. setup some simple dialplan stuff and test calling that way
22:38.13delmargman55s1f, then debug the script.
22:38.23delmargman55s1f, or whatever :P
22:40.35*** part/#asterisk mazza[W] (i=mazzanet@unaffiliated/mazzanet)
22:40.51gman55s1fdelmar: http://pastebin.ca/28048
22:40.57gman55s1fdelmar: I don'
22:41.05gman55s1fdelmar: I don't think it's the script's fault..
22:41.18*** join/#asterisk shimi (n=shimi@unaffiliated/shimi)
22:42.24shimiHi all. I was wondering: Is it possible to make some asterisk magic, such that someone can call the PBX, and "dial" while "talking" with asterisk (through touchtones, of course), to a different number, outside asterisk, and get connected? If this can be password protected, even better! Any ideas?
22:44.02justinushimi: look up DISA
22:44.27gman55s1fdelmar: it would appear that it's deciding to send a 'cancel' message..
22:44.41delmargman55s1f, indeed
22:45.46delmaranyone else see whats going on there?
22:46.07delmarthis happens from any client to any other client?
22:46.21gman55s1fdelmar:  yes... the identical phone sitting right next to it (extn 511)
22:46.46gman55s1fdelmar:  I don't think the phones are faulty.. I had everything up and running a few weeks ago.. but I had to reinstall asterisk today..
22:46.58delmarremove sensitive bits and pastebin your sip.conf and the relevant contexts from extensions.conf
22:47.00shimijustinu, thanks. I looked up in google, I found numerous leads. is there something official from digium?
22:47.17justinunot really, but voip-info.org is a good place to read up on asterisk
22:47.20justinu~docs
22:47.22jbothmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
22:47.28shimiok, thanks again :)
22:47.30*** join/#asterisk paryl (n=paryl@209.236.78.59)
22:47.42shimihttp://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DISA  I assume :)
22:48.19justinuright-o
22:48.30delmarWhats the story with zaptel and asterisk in terms of versions..is for example.. zaptel v1-0-9 compatible with asterisk HEAD?
22:48.54gman55s1fdelmar: sip.conf: http://pastebin.ca/28050
22:49.07*** join/#asterisk liran_ (n=liran@80.178.120.123.adsl.012.net.il)
22:49.22enemyNov  8 23:46:31 WARNING[31134]: chan_zap.c:7570 zt_pri_error: PRI: !! Not good - head of queue has not been transmitted yet ..... (anyone seen this one?) Running BRIstuff (latest).
22:49.31liran_heys
22:49.52delmargman55s1f, ok firstly.. lets look at some network info....
22:50.01gman55s1fdelmar: something, with extensions appended: http://pastebin.ca/28052
22:50.04delmargman55s1f, the phones, and the * box are all talkingon the same network ranges?
22:50.09gman55s1fdelmar: yep
22:50.21delmargman55s1f, ok set nat=no
22:50.35gman55s1fdelmar: I've had issues with that before.. nat=no doesn't work
22:50.42gman55s1fdelmar: has to do with our switches or something
22:50.48delmargman55s1f, you are not behind nat.. you are not using nat.
22:51.02delmargman55s1f, ok this sounds dodgy already....
22:51.10gman55s1fdelmar: I've give it another shot...
22:52.22gman55s1fdelmar: ok --- I set nat=no,... no visible change
22:52.27delmargman55s1f, ok also there is no codec definition for any of them..at the end of those sip entrys.. do... disallow=all  then allow=(codec)  .. which ever codec u prefer to use with the phone for now... and as long as the phone supports it.
22:52.27gman55s1fdelmar: same result.
22:53.19delmarwe are not done
22:53.33delmarand dont forget when making thos echanges to do either.. reload, or sip reload
22:53.42gman55s1fdelmar: yep
22:53.44delmarok lookin at your extensions.conf...
22:53.56justinug729a sounds like shit
22:54.03justinuwhy would anyone pay 10 bucks a channel for that?
22:54.21gman55s1fdelmar: thanks -- gtg-- be back in a bit
22:54.28liran_I haven't worked with Asterisk before but I've been asked by my firm to create a perl program to analyze the Master.csv (which is supposed to be the log of all calls/duration/etc) file and create a sort-of a billing system out of it. This seems too far-fetched, isn't there already some open-source billing system I can use to integrate into Asterisk?
22:54.39delmargman55s1f, ?
22:55.44delmargman55s1f, ok the info u pasted on http://pastebin.ca/28052 is not complete...whats in your from-internal context? those 4 lines? .. pastebin the exten-vm Macro ...
22:56.24demetrioanyone knows how to retrieve the freeworlddialup's IAX number? should they send it to you via email or something?
22:56.36delmardemetrio, i think so
22:56.43shimidemetrio, you need to request one in their site
22:57.01*** join/#asterisk zotz (n=zotz@24.231.47.168)
22:57.27demetrioshimi: that's what I've done, but I can't see where it is, and have received no mail
22:57.46demetriowell, I'll wait
22:59.25shimithey said it should take about 10 minutes, IIRC
22:59.33shimior was it one hour? I don't remember.
22:59.37*** part/#asterisk mkrufky-gone (n=mk@68.160.103.77)
23:00.14demetrio10 min
23:00.25demetrioI bet I won't see it till tomorrow though
23:01.29shimiwhat does Authenticate(XXXX) do ?
23:04.20*** join/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net)
23:05.39*** join/#asterisk santiago (n=santiago@63.245.87.62)
23:09.18*** join/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net)
23:11.19brownjorHey everyone, I was wondering what distro/OS do you all use for asterisk, and why?
23:12.09luitedebian, because that's what I always use, and know best
23:12.18sylefedora, case developpers use it
23:13.25delmarbrownjor, Debian... just because :P
23:13.33Limbeauxis anyone using Quantum Voice?
23:14.27brownjorluite & delmar, would you recommend learning Debian to run asterisk?  How long does it usually take you to install asterisk and debian together?
23:14.50shimithe fastest way to install asterisk is Asterisk@Home ;)
23:14.59delmarbrownjor, this is the trouble... people find that there is something Linux is doing that they want... getting to know Linux can be 90% of the effort
23:15.12*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
23:15.17luitebrownjor: if you just want to run asterisk, there are other/faster options
23:15.22*** join/#asterisk nvrs (i=RUR@toronto-HSE-ppp4256323.sympatico.ca)
23:15.26luitelike shimi said
23:15.27luite:)
23:15.30tzafrir_laptopbrownjor, I would naturally recommend Rapid
23:15.43tzafrir_laptophttp://xorcom.com/
23:15.58delmarYeah, u are better to get an Astrisk distro which is based on .. say.. Debian or RedHat, but is setup to make life easy for you setting up Asterisk
23:16.11shimiand asterisk@home comes with cool stuff, like call monitoring, call logs, conference management, CRM, etc... all that, out of the box.
23:16.28delmarah i have heard of Xorcom but not tried it out for myself yet.. its based on Debian too right?
23:16.37brownjorwell I'm interested in doing it right... I'm trying to design a solution to work with a cisco call manager system... and I kinda want to get started in the right direction first...  I know things about linux... just not debian...
23:16.38tzafrir_laptopyup.
23:16.57tzafrir_laptopand remains very close to it, so you can easily install any other Sarge package
23:17.24delmaris there a way to do it the other way around?
23:17.41delmari dont wanna re-install an existing box say.. but wanna add the bits from xorcom or whatever?
23:17.44luitebrownjor: debian is not that difficult to learn, fairly easy actually, will probably just take some time to get used to it, if you're comfortable with editing config files
23:17.53tzafrir_laptopdelmar, Yes. http://rapid.dotsrc.org/ , basically
23:18.25tzafrir_laptopWe also have there packages for amportal and destar
23:18.42delmarmight be worth a look im thinkin
23:19.05lesouvageIs there a special reason for a lot of flat enoyning noise when using an ATA. My ATA is working but on the receivng site the noise is to loud.
23:19.42*** join/#asterisk mrec (n=revenger@p54B00A80.dip0.t-ipconnect.de)
23:19.54mrechmm
23:20.00mreccan anyone help me with following error:
23:20.01mrecNov  9 00:18:53 NOTICE[6620]: chan_iax2.c:5876 socket_read: Registration of '719698' rejected: Registration Refused
23:22.15*** part/#asterisk brownjor (n=Jordan@c-67-185-209-222.hsd1.mi.comcast.net)
23:22.17delmarmrec, could be any number of things.
23:22.40delmarmrec, sip client?
23:23.27Math`delmar: its chan_iax2
23:23.49mrecI'm trying to get it work withfreeworlddialup.com
23:24.04Math`iax is currently broken at freeworlddialup
23:24.09Math`you're better off with sip
23:24.10*** join/#asterisk nagl (n=nagl@213.235.241.6)
23:24.43harryvvwell well
23:24.45mrecI'm behind a router will that work without any problems?
23:26.07Math`if you forward your ports properly, it shouldn't, but the best is... get a second nic for your linux box and use linux as router
23:26.23Math`this was asterisk isn't on a nat
23:27.50Limbeauxany suggestion for providers?  The only one i looked at was quantumvoice.com
23:29.08myke420247the grandstream gxp2000 is really a quality phone
23:29.32myke420247if you put in too long of a string for the account name the firmware breaks in weird and entertaining ways
23:29.34FuriousGeorgei figured it out how to make call parking less annoying:  users park calls on $USEREXTEN$PARKINGSPOT where userexten is obvious and parking spot is a number 1-X
23:29.52delmarhrm
23:29.55FuriousGeorgehey all
23:30.09delmaris the zaptel available via CVS.. in MAIN the same as HEAD?
23:30.19*** part/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
23:30.20*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
23:30.31FuriousGeorgehey all
23:30.40sjobeck99hi, all, hope all is well, any one know native MOH?
23:33.19FuriousGeorgeso ive been thinking about call parking.  why dont we do this:  calls get parked on ${PARKEREXTENSION}${PARKINGSPOT} where parking spot is a range of numbers starting >=1.  so if im extension 101, and i get 10 spots, i can just xfer a call to 1010 to have the system park it for me, or just manually transfer to a parking spot, 1011 to 1019
23:33.29FuriousGeorgewhat do you guys think
23:33.42swebbI've got a "wildcard" X100P card - what can I do with it and asterisk on a pots line in my home?  Is it worth playing with?
23:34.10FuriousGeorgeswebb:  isnt that an fxo?  you can recieve calls from ma bell
23:34.23FuriousGeorgebut ive heard their quality is iffy
23:34.27swebbGot me if it's an fxo -.
23:34.28ManxPower~fxofxs
23:34.29jbot[fxofxs] An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
23:34.29FuriousGeorgesound wise
23:34.54FuriousGeorge~furiousgeorge
23:34.55jbot[furiousgeorge] a knife-fighting monkey last seen with The Man with the Yellow Bat
23:35.01myke420247i thought the idea of parking was that the spot wasn't tied to any extension
23:35.02FuriousGeorge~jbot
23:35.03jbotjbot is, like, a nub
23:35.17swebbOk, so I've probably got one of those.  Can I do anything cool with it and asterisk, or do I need a T1 or something to get any mileage out of it?
23:35.18myke420247otherwise why not just do an attended transfer
23:35.57FuriousGeorgemyke420247: so have a global spot, im thinking about it this way:  reverse transfer
23:36.20FuriousGeorgeyou cant "pull" a call on any channel ive used
23:36.26delmarok guys.. i FIXED my echo problems.
23:36.32myke420247yeah ok
23:37.39delmarcan anyone tell me what the difference is in doing "cvs checkout zaptel" vs doing "cvs checkout -r HEAD zaptel" ?in terms of the code? IS there a difference? what version is HEAD and what version is the other one... ?
23:38.09FuriousGeorgei used some prop system at a law firm i worked for (way before i even knew what * was) and i remembered when we had someone on hold, one party could "pull it" or the other could xfer it
23:38.15delmarbeacuse HEAD is utterly br0ken
23:38.20delmarin terms of echo
23:38.50FuriousGeorge:)
23:38.52delmar:P
23:39.21swebbHey.  Do I need a special card and special line from Ma-bell to do cool stuff, or can I do it with my normal POTS line?
23:39.21FuriousGeorgei guess they came out with a beavis and butthead movie, i think its on my mind
23:39.30swebb... and my W100P card?
23:39.47*** part/#asterisk Limbeaux (n=icechat5@ip-209-124-211-234.static.eatel.net)
23:39.52FuriousGeorgeswebb: any analog line will do
23:39.53justinusettle down, beavis
23:40.08FuriousGeorgeswebb: it will be as 'cool' as you pay mabell to make it, i suppose
23:40.25FuriousGeorgeswebb: the /real/ cool part is what you make * do with the mabell line
23:41.26swebbSo, I can do anything with that card and a normal POTS line?  What can I *not* do?
23:41.31delmarso what I have is.. fxotune.conf with 1=11,0,0,0,0,0,0,0,0 in it... I have echotraining=800 and other things, currently txgain/rxgain set 0.0/0.0 .. i leave them as they are .. and using zaptel from HEAD.. it blows.. just the plain zaptel from doing a "cvs checkout zaptel" .. no customizations .. just make install (hence it uses KB1) and it works without echo
23:41.41FuriousGeorgefor instance, why pay for voicemail from ma bell, when it would be much cooler to have * send you an email with an attachment
23:41.52FuriousGeorgeswebb: good question...
23:42.07justinuyou can't make caller ID work without paying ma bell for it
23:42.10justinuwhich sucks
23:42.29linageejustinu: of course. ma bell 0wn3z your pocketbook
23:42.29*** join/#asterisk bumblefsck (n=bumblefs@69-160-145-156.ontrca.adelphia.net)
23:42.37*** join/#asterisk TheCops (n=xdz@got.securebinary.com)
23:42.38FuriousGeorgejustinu: technically yes, but thats not *'s fault, ma bell just isnt sending the info right
23:42.39justinunot mine
23:42.48justinui have no connections into my home from them anymore
23:42.50delmarjustinu, what sux is.. callerID info used to be passed anyway.. years ago... i remember hooking up a modem and setting it to bell ccitt or whatever it was.. and i could see incoming callerID
23:42.51TheCopssomeone is using a Rhino T1 channels banks ?
23:43.01delmarjustinu, then the phone company realised they coudl turn it off.. and charge to turn it back on
23:43.05justinuheh
23:43.06linageedelmar: really?!?
23:43.06FuriousGeorgejustinu: what happens when there is a nuclear fallout and the web is down?
23:43.09linageedelmar: wtf
23:43.19justinuFuriousGeorge: i'll have bigger problems than the web :)
23:43.24delmarlinagee, yep here in NZ back in like.. 1990, i did this.
23:43.46swebbCool, thanks guys.  I guess I'll hook it up and play with it tonight.  Thanks again!
23:43.47FuriousGeorgejustinu: seriously though, one thing that is cool about ma bell, the only thing i suppose, is that their lines almost always work
23:43.57FuriousGeorgeand they provide the power
23:44.09justinuyeah, RBOCs do have it down cold
23:44.12linageedelmar: maybe the electric company will decide it can turn your electricity off every morning and charge you to turn it back on. ROFL
23:44.13FuriousGeorgeso even when you have none, they work
23:44.36justinubut the problem is, if the power goes out, every idiot picks up the phone and tries to call people
23:44.36justinuso all you get are reorders
23:44.44*** join/#asterisk voipjoy (n=root@1.fix.netvision.net.il)
23:45.06linageejustinu: wait, ma bell's switch isn't big enough if everyone picked up the phone at once...
23:45.07delmarlinagee, they already charge enuf to keep it turned on and make food money im sure they wont turn it off :P
23:45.17FuriousGeorgeswebb: thats why i say, get one pots line, and a few voip dids, and if your pots line is the number every1 knows and you cant get it ported (or dont want to) just have it rollover on busy to the voip did
23:45.20delmarlinagee, power here has trippled in the last 3-5yrs
23:45.22FuriousGeorgejustinu: true
23:45.51delmarSo can anyone please tell me... Zaptel on CVS... MAIN and HEAD must be aliases for versions I assume.. what versions ?
23:45.54linageedelmar: maybe they'll change the voltages and charge you to put it back. ROFL. screwy still to mess with people's heads
23:45.59justinuthey have switches that service 60-120k phones
23:46.00*** join/#asterisk test34 (n=test34@unaffiliated/test34)
23:46.06voipjoyanybody can advice? kernel 2.6.14  problem: ztdummy.c:103:2: error: #error ztdummy requires 1000 hz jiffies
23:46.17justinubut they only have 10,000 DTMF receivers!
23:46.18FuriousGeorgedelmar: the only thing i need to do in 1.0.9 to avoid echo (besides have clean wiring and less than crappy handsets) is echocancel=yes, and st jitterbuffer
23:46.31SkramXvoipjoy: Israeli?
23:46.32linageejustinu: yes, but if everyone tried to call from switch A to switch B, do you think the interconnect would hold up? LOL
23:46.41delmarFuriousGeorge, you actually have a jitterbuffer set on your incoming line?
23:46.42voipjoySkramX:yes
23:46.46SkramXCool, Cool
23:46.47justinuno, they have even less trunks, i think
23:47.04test34how would I check with GOTOIF if the incoming call has a specific area code ?
23:47.10FuriousGeorgedelmar: incoming voip did
23:47.22linageejustinu: exaclty. which means you have a slightly higher change of reaching someone on the same switch in an emergency, and a lesser change of reaching someone even just a couple of miles away. ;-)
23:47.23delmarFuriousGeorge, oh ok thats diff.
23:48.05linageejustinu: which makes sense now why local calls are free and zone 3 calls aren't.
23:48.20delmarlinagee, but seriously.. there used to be caller information on the lines here many years ago, and as time went by it was turned off, then a chargable service came along to turn it on again
23:48.22FuriousGeorgedelmar: i mentioned jitterbuffer b/c the only time you should get echo is when voip goes to pots
23:48.26justinulinagee: never thought of that
23:48.34FuriousGeorgeor viceversa
23:48.52justinupure ip calls get echo all the time
23:49.04justinushitty phones/headsets
23:49.09justinupoor acoustic coupling
23:49.17linageepoor user
23:49.38delmarFuriousGeorge, well in zapata.conf i have jitterbuffer commented out, and I seem to have fixed the echo. Clearly the zaptel from HEAD has an issue
23:49.39voipjoySkramX: you?
23:49.42*** part/#asterisk tmccrary (n=tmccrary@68.78.185.254)
23:49.52FuriousGeorgedelmar: well, technically there is echo on every pots call, but as long as the other party is not overseas, your voice comes back to you fast enough that you dont notice.  you just kinda expect to hear yourself, and you do
23:50.01SkramXvoipjoy: Jewish, spoke hebrew somewhat fluently for a while.
23:50.17delmarFuriousGeorge, yeah i know a little about side-tone
23:51.09FuriousGeorgedelmar: i didnt know zapata had a jitter buffer, i dont see how analog in could be jittering.  for an incoming call from pots the only you shouldnt get jitter on your local lan
23:51.12TheCopssomeone is using a Rhino T1 channels banks ?
23:51.15justinupots phones put sidetone on the speaker deliberatly
23:51.28delmarFuriousGeorge, now what gets me is why another situation i have.. does have echo.. almost like delayed side tone.. ... sip-hardphone(ulaw) to * ===IAX2/ulaw==== to * to sip-hardphone(ulaw)
23:51.41FuriousGeorge*for an incoming call from pots you shouldnt get jitter on your local lan
23:51.52voipjoySkramX: good!
23:52.12delmarFuriousGeorge, no.. i mean.. calls from PSTN to * to SIP on the LAN is great
23:52.18tzafrir_laptopvoipjoy, you should USE_RTC there, I believe
23:52.21delmarFuriousGeorge, heh it wasnt this morning but is again now
23:52.42FuriousGeorgedelmar: wait im confused
23:53.03test34what does: Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN' mean ?? my card isnt ready ?
23:53.11FuriousGeorgedelmar: but im pretty sure if its all voip from origination to termination you shouldnt get an echo
23:53.23delmarFuriousGeorge, exactly
23:53.26voipjoytzafrir_laptop: i should declare it by my self?
23:53.58delmarFuriousGeorge, but I do, altho i havent tested it again in a couple of days.. been busy with this pots/pstn echo problem here .. but now that this is resolved i need to look at it again
23:53.59voipjoytzafrir_laptop: x64, kernel 2.6.14..asterisk 1.2beta2
23:54.08tzafrir_laptopvoipjoy, not sure
23:54.16FuriousGeorgedelmar: is your phone trying to do some stuff that * is already doing, or doesnt need to do?  for instance i find "voice activity detection" options on some voip clients isnt necessary
23:54.22delmarFuriousGeorge, what gives me the shits is that clearly.. the zaptel from HEAD, is br0ke
23:54.23tzafrir_laptopin zconfig.h?
23:54.48delmarFuriousGeorge, exactly.. i turn all that crap off.
23:55.04jtoddHrm.  Anyone getting this error on CVS checkout (cvs co asterisk) of HEAD after successful login?  "cvs [checkout aborted]: Cannot check out files into the repository itself "
23:55.12*** join/#asterisk MajestiK (n=MajestiK@S0106000024c058cc.ed.shawcable.net)
23:55.14FuriousGeorgedelmar: i dont use head so i cant tell you if have found the same issue (thats why i wouldnt want to mess with head)
23:55.15delmarFuriousGeorge, none of that is turned on at either end.. yet both ends of the SIP to SIP call can hear themselves with delay....
23:55.41delmarFuriousGeorge, oh i got an idea. sec
23:55.47FuriousGeorgedelmar: just thinking aloud, for there to be echo there must be delay
23:55.58delmarFuriousGeorge, sure
23:56.02delmarFuriousGeorge, brb sec
23:56.29*** join/#asterisk paryl (n=paryl@209.236.78.59)
23:56.41voipjoytzafrir_laptop: i declared it ztdummy.c before 1st line
23:56.57paryli just don't get it... i've tried EVERYTHING i can find on echo cancellation, but it still have it really bad at the beginning of incoming calls
23:59.00FuriousGeorgeparyl: how long is the "beginning"
23:59.04FuriousGeorgefirst five seconds?
23:59.20justinuthat's when the EC is "converging"
23:59.41delmarFuriousGeorge, ok get this... if a SIP client calls the * VM on the remote * box, not locally, and if I swear and curse at it..there is not one bit of echo

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