irclog2html for #asterisk on 20060131

00:00.03*** part/#asterisk [dc] (n=dc@24-205-223-175.dhcp.slto.ca.charter.com)
00:00.58mzo_iwhat's the going rate for a paypal thank you to imlement a call block list and some fancy stuff? :P
00:01.13Qwell[]mzo_: "fancy stuff"?
00:01.14Qwell[]:p
00:01.47mzo_well, like, a custom set of voice mail prompts when people on the block list so it'll say 'the caller you hae called does not accept calls from facist companies like citibank.  please try your call again later when you aren't an ass' and have it hang up
00:02.26Qwell[]that could be done in dialplan
00:02.37mzo_i'm sure it can be done a bunch of awyas but i'm wary of fucking up my system again ;)
00:02.38SkumlingQwell: are you one of the *-devs?
00:02.52Qwell[]Skumling: I'm a bug marshall
00:03.14Qwell[]I do write code though, sure
00:03.29*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)
00:03.39SkumlingQwell: okay... must make friends with you ;)
00:03.43*** join/#asterisk dominix (n=dominix@CA03FD17.adsl.mana.pf)
00:04.16*** join/#asterisk elg (n=fugalh@dhcp25.cs.nmsu.edu)
00:04.40lucasjbWhen a SIP call connects to my Asterisk system, the process events/0 jumps to 99% CPU usage. This has only just started happening, I can't figure what configuration change I've made that's caused it...
00:05.59*** join/#asterisk johnnyb (n=jonathan@207.155.33.225)
00:06.18AJMn<PROTECTED>
00:06.47Qwell[]AJMn: What sip phone?
00:07.00Qwell[]There was an echo issue with a certain firmware version of...umm...polycom?
00:07.01knight_; switch => IAX2/user:password@bigserver/local     <--- will this let me pass traffic between asterisk servers?
00:07.05*** join/#asterisk family (n=family@c-67-163-169-5.hsd1.ct.comcast.net)
00:07.14Qwell[]knight_: Should, but it's better to use a peer
00:07.17familyanyone in here have astcc working
00:07.28Qwell[]family: I've gotten it working...I don't currently though
00:08.13familyi have it running it all, but when i try to use it, , using the sampels, dialign 1234, its says 20 then hangs up
00:08.34Qwell[]ugh, I saw that before...what was it?
00:08.43Qwell[]it just tells you the length of the card
00:08.53Qwell[]or, rather, the length it expects a card to be
00:09.27Qwell[]family: You'd have to pastebin your CLI output.  I think you missed a step
00:09.35familyso isnt there suppsoed to be a prompt askign for the #
00:09.43familysure brb
00:09.44Qwell[]I think that's what that is, is the prompt
00:09.45knight_Qwell, I have an asterisk box here at home, but I want my asterisk on a colo box to handle all the calls and pass them to my home... I have 8+ trunks, and various dialplans, so I'd hate to have to duplicate my dialplans
00:09.59Qwell[]knight_: read up on switch =>
00:10.06knight_Qwell, yeah that's what I thought
00:10.35*** join/#asterisk ardor (n=vircuser@las-cust-66.18.135.148.mpowercom.net)
00:10.43AJMnQwell[]  Zyxel P2000W, but both have updated firmware, it only happens when i call each other, but if i call an outside line from a sip phone or outside to sip theres no echo, just calls that stay inside the asterisk system
00:11.02Qwell[]AJMn: Are they reinviting?
00:11.14ardorhow do shutdown asterisk via the command line... asterisk -xstopnow
00:11.17ardordoesnt work..
00:11.21familyQwell[] http://pastebin.ca/39182
00:11.29Qwell[]ardor: asterisk -rx "stop now"
00:11.31Qwell[]should do the trick
00:11.34Math[laptop]ardor: asterisk -rx "stop now"
00:11.37ardorthanks
00:11.41Qwell[]Math[laptop] slow :p
00:11.45Math[laptop]lol
00:11.55ardorsweet!!
00:11.59Qwell[]family: Yeah, I'd say you missed a step
00:12.04AJMn<Qwell[]> set to no
00:12.11familywhat in particular
00:12.12*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)
00:12.14Math[laptop]ardor: actually its asterisk -rx "cli command here"
00:12.25Qwell[]family: Dunno...I'd have to install it to try to figure it out
00:12.31Qwell[]or see yours
00:12.36familyyou still ahve your old config files from it
00:12.41familyhave
00:12.42Qwell[]no, it was for somebody else
00:12.52familydamn
00:13.12AJMn<Qwell[]> reinviting set to NO on both
00:13.35*** join/#asterisk troy (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)
00:14.12Zodiacalanyone know if my line assignments can be both speed dial -AND- show the status of ext.'s? i.e. on/off hook using hint. if i use sccp with my cisco 7960 phones?
00:14.45*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
00:15.02Qwell[]Zodiacal: chan_sccp?
00:15.08Zodiacalya
00:15.10Qwell[]yep
00:15.15Zodiacalcoolness
00:15.19Zodiacalqwell thank you!
00:15.22Qwell[]1234,name,hintexten
00:15.51*** part/#asterisk eieiyo (n=eieiyo@68-119-68-89.dhcp.mtgm.al.charter.com)
00:15.51Zodiacali havn't tried sccp yet ,but i have heard good things about it..
00:15.57Zodiacallike that feature for instance.
00:16.01Qwell[]I love chan_sccp
00:16.25Qwell[]especially with my realtime patch...mmm
00:16.46Zodiacalcan i have the latest firmware :P j/k im still waiting for my retailer to give me my cisco account so i can d/l it.. i can only play with sip right now :P and its ok..
00:17.33*** join/#asterisk johnnyb (n=jonathan@207.155.33.225)
00:17.33Zodiacalqwelll realtime patch?
00:17.33Qwell[]mmhmm
00:17.33Qwell[]let's me put the sccp configs in a database
00:17.33Zodiacaloic
00:17.33Qwell[]Sergio is going to add it "any time now"...heh
00:17.33Qwell[]I need to get on him about that
00:17.48*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
00:18.12Zodiacalhints and programable soft keys is what sounds nice
00:18.23Qwell[]They aren't exactly programmable
00:18.27Zodiacalqwell have you tried changing the softkeys? how easy is it?
00:18.33Qwell[]I mean, form the services menu...sure
00:18.41Qwell[]I looked a few times...I wasn't feeling too ambitious
00:18.42Zodiacallike i wanta add a page button
00:18.56Zodiacalto just dial an ext. to get loud speaker paging..
00:18.56Qwell[]it's not impossible
00:19.14Qwell[]certainly isn't easy though
00:20.45dmzhey y'all, i enabled caller announcing on meetme and whenever i enter the conference, it says to say name & press #, however right after it does that, it goes directly into conference. i don't hit # and it doesn't capture my name...also once in a conference how can i get into administrative mode? or do i have to setup another extension and join as an admin?
00:21.21Qwell[]dmz: You have to join as admin
00:21.51*** part/#asterisk bertd (n=admin@adsl-220-179-181.mob.bellsouth.net)
00:22.35Qwell[]brb
00:22.58*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)
00:23.47*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
00:24.36*** join/#asterisk elg (n=fugalh@hfugal.NMSU.Edu)
00:25.34*** part/#asterisk jaike (n=a@203.131.137.76)
00:28.43*** join/#asterisk Math[laptop]_ (n=Math_@modemcable148.4-81-70.mc.videotron.ca)
00:30.26*** join/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda)
00:31.41*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)
00:32.45*** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net)
00:38.38dmzQwell, thanks. so i need to setup a specific extension for each conference for admin's to join as? hmm i might look into the code, should be able to use admin code instead of pin and have it recognize admin vs regular pin-auth'd user
00:39.41[av]baniwow, the asterisk phpbb forums got thoroughly trashed
00:40.56drumkillaum ... which ones?
00:41.07drumkillaforums.digium.com looks fine.
00:41.19[av]banii cant login anymore, says wrong password
00:41.42drumkillawell, I wouldn't classify that as thoroughly trashed
00:41.51[av]baniof course they got really screwed on friday... someone redirected them to the sales forum for allison
00:42.09drumkillai can log in just fine
00:42.21Qwell[]people actually use forums?
00:42.46drumkillaalmost 13,000 posts ...
00:42.54Qwell[]how many are answered? ;/
00:42.55[av]baniwell, somehow my account got corrupted since friday
00:43.12[av]baniafter the big mangle
00:43.15*** join/#asterisk cyburdine (n=cyburdin@208.2.145.2)
00:43.47drumkillawhat is your username
00:44.00[av]banibani
00:44.33[av]baniwell, guess i get to wait another 30 minutes because it says i've exceded login attempts
00:45.52drumkillawell, I can reset your password if you need it .....
00:46.01[av]banii already tried.
00:46.05[av]banithen it locked me out.
00:46.39*** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
00:48.39Frawghey
00:48.47Frawgg729 uses 8kbit/sec correct?
00:48.52Frawgis there anyway to change it's timeslicing?
00:48.56Math[laptop]plus overhead
00:49.19FrawgMath[laptop]: what's the overhead?
00:49.31Math[laptop]it uses around 20kbps total
00:49.44Math[laptop]http://www.packetizer.com/voip/diagnostics/bandcalc.html
00:53.56neon_kli want to setup sms server using gsm modem can suggest a gsm modem 4 sim card support
00:54.41litageis it possible to limit a particular extension to a certain number of concurrent/simultaneous calls?
00:54.57*** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo)
00:55.10FrawgMath[laptop]: can you change the frames/packet easily?
00:56.30dlyneslitage:  sip.conf/iax.conf/...: incominglimit=n
00:57.04FrawgMath[laptop]: of g729
00:57.14Math[laptop]uhm let me check
00:57.16*** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com)
00:57.42Math[laptop]Frawg: no idea
00:59.44litagedlynes: thanks for that. i looke dup incominglimit on voip-info.org and it turns out that it's been deprecated in favour of setgroup and checkgroup
00:59.56litages/looke dup/looked up/
01:01.05Qwell[]Frawg: There is/was a patch on the tracker to allow changing packet sizes
01:01.33Math[laptop]litage/dlynes: which have then been deprecate for the GROUP() and CHECKGROUP() dialplan functions
01:01.42Math[laptop]er, not CHECKGROUP() but GROUP_COUNT()
01:06.05litagethanks Math[laptop]
01:06.58FrawgMath[laptop]: cool, you don't happen to know the overhead that an ssh tunnel adds(perpacket) ?
01:07.55Math[laptop]er, you're tunnelling a voice conversation over ssh?
01:08.09trixterI dont know about the IP layer, presumably it has some (I fail to see how it cant) but there is also the cipher chosen, most are block ciphers which will padd to make an even block, that will be at most a couple bytes per packet..
01:08.26trixterit shouldnt be that hard to figure out what you are sending and what ssh packet sizes are and do the math
01:09.13trixterMath[laptop]: if its a local network the tcp difference wont matter that much, by local I mean direct connection could be a wan could be physically local, either way the tcp part of it shouldnt intefere if the network is managed
01:09.29trixterhowever if, and I suspect this is what you were getting at, you tunnel via the inet then there may be problems
01:09.53trixterI didnt know his configuration so I didnt comment on that
01:10.25*** part/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda)
01:12.16Erryou don't want to tunnel VoIP over ssh
01:12.31*** join/#asterisk p0g0__ (n=pogo@mrtc-dsl-610045.mis.net)
01:12.32trixterthat depends
01:13.01Errnot really - it's pretty much a universal constant
01:13.02*** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0)
01:13.14Errif you're on a network with virtually NO loss, then you might be able to get away with it
01:13.20trixterif its a local network as described above on a managed network it wont be a problem
01:13.31trixterok thanks for confirming what I said
01:13.47NetgeeksI'll provide a third confirmation if you need it
01:13.51Errotherwise, if TCP's congestion control ever drops below the required bitrate, you'll experience strangeness
01:14.10trixterthat doesnt sound like a managed network to me
01:14.16trixterwhy I specifically added that qualifier
01:14.19Errthat sounds like packet switching to me
01:14.46trixtersome people maintain networks that are more than just internet links
01:14.53trixterwhy I specifically commented on those
01:14.58trixterI wasnt talking about home dsl
01:15.28ErrTCP's congestion window won't ever grow much above the committed bit rate, which means that ANY loss (including corruption, collision, ARP timeout, whatever) will cause your congestion window to possibly drop below the CBR
01:15.33ErrI'm not either
01:15.39ErrI'm just talking about TCP dynamics
01:16.47trixteron a non home network there are ways to deal with that
01:16.53*** join/#asterisk ctooley (n=ctooley@24-155-179-239.dyn.grandenetworks.net)
01:16.56Errto deal with TCP's congestion window?
01:16.58trixterhowever if you want to talk about home dsl then you are 100% right you shouldnt use ssh
01:17.12Erryour implication that I've never seen a non-home network is misguided
01:17.12trixterI however see little point in arguing the difference between a managed network and home dsl with you
01:17.21ctooleyThere a good client for the Nokia 770 tablet available yet?
01:17.27trixterI never said you hadnt seen one
01:17.45trixterI am saying that you, like many in the asterisk community, think in terms of a home install and try to make assumptions and comments that may not apply
01:17.50Errno, I don't
01:17.56trixterits a common asterisk thing, dont worry about it
01:18.12ErrI happen to do TCP research, and I'm well aware of what can fail with TCP and CBR streams
01:18.16trixtermost people do what you do, think solely in terms of what runs fine on their box at home on their home network applies to everyone else
01:18.29trixterI dont think less of you for doing that
01:18.44mzo_my work asterisk is flakey :P
01:18.47Errthat's big of you
01:19.11trixterwell I didnt want you to start boasting about some mythical work you do to justify what you said, after all that isnt proof that you are even right
01:19.11ErrI don't think less of you for assuming that I'm an idiot who's never worked on a "real" network
01:19.14trixterits just ego boosting
01:19.20*** join/#asterisk MGSsancho (n=user@adsl-67-127-173-128.dsl.irvnca.pacbell.net)
01:19.27Netgeekswhoah
01:19.29trixterso I felt that if I boosted your ego by saying I didnt look down of you for thinking opnly in terms of home networks you wouldnt do that
01:19.47trixterafter all only idiots will say some mythical work they do as proof they are right
01:19.49Erryes, you are indeed God's gift to networking - it's clear to me now
01:19.59mzo_no wait, i'm gods gift to networking!
01:20.02Netgeeksthere is some massive testosterone flying about
01:20.17trixterI never claimed that I did mythical research work in tcp and thus was gods gift to networking
01:20.20[av]banipenis wars
01:20.20trixterI left that for the ego impaired
01:20.21Qwell[]Netgeeks: NO, you're thinking the other one...
01:20.25Qwell[]what was it called?
01:20.32Qwell[]ahh yes...bullshit
01:20.43Netgeekshehe, it's hard to tell the difference, Qwell
01:20.48Qwell[]Netgeeks: indeed
01:20.50trixterafter all what validity does a comment about some mythical made up research job have in proving anything?
01:21.08[av]banithis reminds me of the rednecks who were redlining their engines in an attempt to prove who had the larger penis
01:21.29[av]baniwhoever had the louder engine had a bigger penis
01:21.31trixterthere *is* a difference between a managed network and home dsl or just inet links, that is all I said, and I was quite clear about it
01:21.39Errtrixter: http://www.ietf.org/internet-drafts/draft-allman-rto-backoff-02.txt <-- I'm the first author
01:21.54Errit's not "mythical"
01:22.00Qwell[]best comeback EVER
01:22.03Qwell[]:P
01:22.08trixterI dont know that you are true, I could pull up a link to the constitution and claim I am the first signer
01:22.19trixterlemme see if 'err' is listed
01:22.34trixternope its not
01:22.37[av]baniyay penis
01:22.41Errwhatever
01:22.42trixteryou sure proved me wrong
01:22.54Qwell[]hostname matches
01:22.57trixterthat still doesnt address the core issue which you are now trying to avoid
01:22.59Errof course my hostname matches
01:23.10Errwhat core issue might that be?
01:23.11trixterI really dont think less of you for thinking only in terms of home dsl and not properly managed networks
01:23.15*** part/#asterisk Utah_Dave (n=boucha@0-1pool139-4.nas28.salt-lake-city1.ut.us.da.qwest.net)
01:23.54fiber0ptiWhat is the best way to include include or goto a context depending on the hour of the day, day of the week or date?
01:24.01trixteroh that you were trying to base decisions and comments on home networks despite my continued comments about properly managed networks
01:24.11trixterfiber0pti: gotoiftime ?
01:24.15Erroh, right, because packet loss never happens in "properly managed" networks
01:24.26Netgeeksfiber: are you asking if there is an option in addition to GotoIfTime?
01:24.34fiber0ptiah..thank you.. didn't know gotoiftime existed
01:24.36Qwell[]can't include do stuff based on time?
01:24.44Qwell[]or something like that
01:25.04trixterErr: depends on the network manager I guess
01:25.07NetgeeksI lost a packet the other day at the store, I had to go back and get it, but by the time I got there they had discarded it.
01:25.14NetgeeksThey gave me a new one tho
01:25.26trixteryou can build a network that doesnt have such issues, you just have to think about the problem a little differently
01:26.00Qwell[]mythical no packetloss network?
01:26.11dudesI've never heard of one
01:26.13Erryeah - Shannon talks about it a lot in his information theory books :-)
01:26.16Netgeeksdon't use packets, and you can't lose any
01:26.34*** join/#asterisk TedC (n=ted@gray.impulse.net)
01:27.28Math[laptop]Qwell[]: LANs usually don't suffer from packet loss
01:27.35Qwell[]usually != never
01:27.37trixterif you send packets in the same old way and deal with them in the same old way you have the same old problems
01:27.59trixterthere are solutions but they dont work well over unmanaged networks like the internet (unmanaged from the endpoints perspective)
01:28.11Math[laptop]well as long as you don't do rate-limiting and that your links aren't maxed-out, there's none
01:28.16Math[laptop]except in a case of hardware failure
01:29.04wilymage...or crap hardware
01:29.36Error bit corruption - which MUST happen
01:30.12Math[laptop]bit corruptions?
01:30.35Math[laptop]transmission errors on an ethernet network is handled by the card itself, the faulty packet is discarded and retransmitted automaticly
01:30.41*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)
01:30.43mzo_hehe, i just made a 75 minute call on asterisk, using my fxo.  yay. ;)  it's so nice when it just works.
01:30.51Math[laptop]hehe :)
01:30.52Netgeekscosmic ray beans a bit in the wire, changing it
01:30.58rayvdAnyone here using adelphia.net?
01:30.59Qwell[]Math[laptop]: I think that's the issue
01:31.04Qwell[]rayvd: god no
01:31.09Qwell[]rayvd: Need help?
01:31.11ErrMath[laptop]: not necessarily - COLLISIONS are detected by CSMA/CD networks, but bit corruption can be unrelated to collisions
01:31.21NetgeeksI used to work for adelphia, and I stayed at a holiday inn express last night
01:31.24mzo_what's the other kind of network besides CSMA/CD? there's some other kind
01:31.25rayvdjust want to prove to my boss that email from adelphia.net is not being rejected by our servers :)
01:31.26Qwell[]I'm on a first name basis with their CEO and a bunch of execs :p
01:31.39mzo_haha, and i'm calling another asterisk server to leave voice mail now. ;)
01:31.41rayvdso wanted to find someone taht can use their mx server
01:31.46*** join/#asterisk dan_ (n=dan@202.147.139.26)
01:31.50Errwell, wireless is CSMA/CA
01:31.57Qwell[]rayvd: call support, tell them to try
01:32.08Errthere are TDM networks that don't do any CSMA whatsoever
01:32.15rayvdooh, that's a swell plan
01:32.27Err(there are a million types of networks)
01:32.36dan__howdi peoples... just wondering if anyone knew if a cisco AS5200 loaded with MICA modems can be used in conjunction with an asterisk to do SIP ?
01:32.40mzo_yeah that's it CA.  I forgot.  It's something i read years and years and years ago. ;)
01:33.27mzo_is there a reccomendation for an OSX Softphone application?
01:33.35Qwell[]mzo_: idefisk works
01:33.35litagewhat does it mean when someone says "this channel has only one leg"?
01:33.48mzo_ill try that. ;)
01:34.04mzo_is that spelled right?
01:34.07Qwell[]litage: means it only goes between a phone and the pbx
01:34.12trixterI am still waiting for the great research god of all things networking to explain this comment: Err or bit corruption - which MUST happen
01:34.23trixterspecifically his emphasis on the use of the word 'must'
01:34.37mzo_oh it's with a c, not a k
01:34.46Qwell[]no, it's idefisk
01:34.56Errtrixter: see, this dude named Shannon wrote a bunch of papers on information theory, which proves that bit corruption must exist in any network - you can use ECC to lower the rate, but you can never totally eliminate it
01:34.57mzo_heh google had it the other way :P
01:35.03Qwell[]well, google is wrong :p
01:35.21trixterso when using ECC a retransmission is required?
01:35.26litageQwell[]: where/what else could the channel go between?
01:35.33rtwell, that's sort of what it says, yes.
01:35.34Qwell[]litage: another phone
01:35.40Qwell[]two phones, no pbx between
01:35.42trixteryour whole point was about retransmissions, so taken in context according to what you have said parity checking wouldnt be able to compensate
01:35.46Errno - link-layer retransmission is completely different - and also doesn't play particularly nicely with TCP, because it jacks with the RTT calculations
01:36.00rtwhat it really means is that for any given communication channel, there is an associated capacity that depends on the bandwidth and the signal to noise ratio of the channel.
01:36.01*** join/#asterisk santiago (n=santiago@63.245.86.155)
01:36.18litageQwell[]: how can 2 phones communicate without some sort of pbx/softswitch in the middle?
01:36.19Errno, *TCP* retransmissions are unrelated to link-level retransmissions, and care caused by packet loss
01:36.22rtany attempt to transmit beyond this capacity cannot succeed with greater chance than just guessing the bits.
01:36.25Qwell[]litage: easily
01:36.38trixterahh so this stuff about bit corruption doesnt really apply
01:36.39trixterI see
01:36.43litageQwell[]: oh, using something like enum?
01:36.49Qwell[]litage: no
01:37.03trixterI think I finally understand you now, when you couldnt back up your comments you went on a tangent to try to divert attention away
01:37.04Qwell[]just dial the user@hostname of another phone
01:37.04litageQwell[]: woud you care to shed some more light on this for me please?
01:37.08trixteryou sure showed me up
01:37.09Errtrixter: yes, you're right - I don't know anything about computer networks; I'm sorry for having tried to help, when I'm clearly a fool
01:37.18litageah  =P
01:37.30trixterwell trying to compare apples to oranges after you couldnt back up your claim doesnt help your image any
01:37.33Qwell[]litage: works fine with sip and iax, most phones
01:37.37litageErr: don't insult yourself dude. it doesn't do anyone any good
01:37.41trixterit gave me a laugh though
01:37.45trixteroh and btw now I do think less of you
01:37.59NetgeeksI'm laughing too, but not for the same reason
01:38.25trixternot becuase you only think in terms of home dsl links going over the internet but instead becuase you tried to cover up when you were wrong and hide the fact later finally admitting that bit corruption (your holy grail to prove your point) was not quite accurate for that point
01:39.51*** join/#asterisk Jun_ (n=wj1918@pool-138-89-62-149.nwrk.east.verizon.net)
01:40.12familyyea so hmmm looks liek my problem with astcc is it cant connect to the database
01:40.22Qwell[]family: That'd do it
01:42.02familybut on the astcc wiki it doesnt say whats needed for the db connected... odbc and what not
01:42.10*** join/#asterisk [Atlas] (n=whois@216.190.144.90)
01:42.12inv_Arpdamn i need some good caribbean rates
01:42.23*** join/#asterisk prufrock (n=orange80@252.84.cm.sunflower.com)
01:42.25mzo_ooh, ty for the softphone thing it works
01:42.36*** join/#asterisk Hali_303 (n=surfk@dsl51B6ACC5.pool.t-online.hu)
01:42.42Hali_303hi!
01:43.33Netgeekswhat a quotable quote!  "any attempt to transmit beyond this capacity cannot succeed with greater chance than just guessing the bits."  not sure what it meant in context, but out of context, it's pretty funny
01:43.49Errheh, it's actually true, too
01:43.53ErrShannon ruled
01:44.30[av]baniguh, can the cross conversation leakage stop? i put err and trixster on /ignore so i wouldnt get this stupid babble
01:44.56NetgeeksBack in 'the day' I worked with some sun hardware running SunOS 4.1.3 or so, I don't remember the exact version, and they had a great quote for the option to shutdown that would stop the computer without syncing disks
01:45.11mzo_suicide? :P
01:45.19trixterNetgeeks: yeah it also goes to prove my point about 'properly managed networks' which by definition dont exceed their allowable bandwidth
01:45.28Hali_303does anyone here know SIP on a protocol level? my question: how to setup a SIP request like "send the reply back to the same IP and same port from where you have received this" and also for the RTP: "initiate an RTP connection to the ip and port where you receive and rtp connection".. this way I could get around if only one of the hosts is behind a symmetric NAT. any ideas?
01:45.32*** join/#asterisk dsasda (n=OIGEORGE@ool-43540102.dyn.optonline.net)
01:45.36dsasdahttp://www.valvehacks.zaccum.com/ - great cs dod hl2 (valve) hacks
01:45.39Netgeeksit said something like this:  "this option is used to shut down the computer without syncing the disks.  You should only use this when the CPU is on fire"
01:45.43Qwell[]Netgeeks: like shutdown -n?
01:45.49mzo_hahahah, that's a real man(tm) option
01:45.56trixterNetgeeks: are you refering to going into prom mode by pressing stop-a (or l1-a depending on keyboard)
01:46.16trixterthat sounds like something that was in the prom
01:46.20Netgeeksnah, it was in the man page for the shutdown command
01:46.33trixterahh
01:46.39trixterthere were a few things that sun did back in those days
01:47.09trixterthe screensaver had a backdoor of 'hasta la vista' to bypass anyones password, the prom actually was interesting in itself becuase you could do forth (it still should support that at least)
01:47.10Netgeeksnow I also worked with some Apollo computers, and they had a error message in the error.txt file that gave you text error descriptions for error codes
01:47.13trixterand a few other things
01:47.24trixtervax had the best WAY back in the day
01:47.26NetgeeksI don't remember the code number, but the description was "Won't fit through an 18 inch hatch"
01:47.38trixterif 'love' was not a valid make target and you typed 'make love' it would reply 'not war?'
01:47.48trixterlikewise if you did make war it would reply not love?
01:48.06Netgeeksapparently they had lost a navy contract after putting a ton of cash into the development because they failed to meet the spec requirement that the computer fit through a standard 18 inch hatch
01:48.18*** part/#asterisk dsasda (n=OIGEORGE@ool-43540102.dyn.optonline.net)
01:49.13trixterall I have to say is I am glad the 70s are over and I no longer have to 'attach' to a drive before using anything off it
01:49.42Qwell[]What, you mean like mounting it?
01:49.53trixterno in a prime you have to attach its like doing   c:
01:50.40Qwell[]oh, so a command you have to type before you can access files on it
01:50.47NetgeeksAt my first real post college job, we had a computer that you had to boot by entering binary based commands using 10 flip-switched and 10 lamps on the front panel
01:50.51Qwell[]not like mount at all
01:50.56trixterthe seperate drives like windows does is actually rooted WAY back when, its not anything remotely current or even that handy, but I guess its less confusing to some
01:50.58justinuprime?
01:51.11trixterprime computer made mainframes back in the day they went under in the 80s
01:51.18trixterright after a buyout the new owner drove it into the ground
01:51.24justinuyeah, i've read about them
01:51.43trixtertheir pixel was used in 1 star trek TNG episode though...  they were doing a trade show in LA and a producer saw some stupid ass grfx and wanted one
01:51.58trixteryeah I was using em in the 70s...  your 'read about em' comment is um ...  :P
01:52.14trixterprime also had their own private world wide network which was also fun
01:52.28trixternot much to do then the intarweb wasnt around..  but it was fun to talk to people
01:52.36Qwell[]bbl, hom
01:52.37Qwell[]e
01:52.52justinui was born in 76
01:52.57trixterI was born in 72
01:53.16trixterit just so happens in the late 70s my father got a job doing education for prime so I got to play
01:53.37trixterthat is also when I started programming, I must say dartmouth basic on a prime isnt as much fun as languages today
01:53.49justinuso you were playing with a corporate mainframe before age 10?
01:53.54trixteryup
01:53.58justinuheh, nice
01:54.25Math[laptop]heh I was coding under the age of 10
01:54.31trixtermy first computer was built by my dad (mom soldered the keyboard together)
01:54.34Math[laptop]well I started some qbasic at 8
01:54.43justinui was keying in basic programs on my ti/994a from "99er magazine"
01:54.51trixterMS made a lot of money rewriting the io routines in dartmouth basic
01:55.00justinui dunno if you would call that programming
01:55.04trixterqbasic wasnt that bad compared to what I used..  infact it was a lot better in many ways
01:55.16Math[laptop]heh
01:55.36Math[laptop]the first thing I did is make some sort of siren sound using a for loop and the SOUND instruction
01:55.54Netgeeksheh
01:55.54trixterI took a programming class in um 7th grade and we had to type in code from a magazine that was soooo boring
01:56.03justinui also had one of those radio shack 3000-n-1 electronics kits
01:56.42trixterthe teacher and I didnt get along because she was trying to teach the difference between print 4+2 and print "4+2" and I was making the result blink and fly across the screen, that lasted 1 week before I switched to theaqter (it was that or AG and I didnt want to raise livestock - yes it was a rural farming area school)
01:56.46SibRphrekhow does asterisk handle 911 calls?
01:56.54trixterjustinu: lucky I only had a 50:1
01:57.06justinusomething like that
01:57.07trixterSibRphrek: it technically doesnt, it hands it off to someone who does
01:57.09justinuit was really cool tho
01:57.10ErrSibRphrek: it doesn't do anything out-of-the-box - you have to put in rules for it
01:57.12*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
01:57.19SibRphrekErr: which conf is that?
01:57.34trixterSibRphrek: what exactly do you want to do?
01:57.35Netgeekssib: Asterisk doesn't know the difference between 911 and 123 or xxx until you tell it to
01:57.41dlynesSibRphrek:  Your upstream has to be able to handle 911, or you have to have a contract with a call center that does it
01:57.57trixtermap it so when someone dials 911 it calls?  or set up the E911 information data?  it doesnt do the E911 information stuff out of the box, you have to get a service provider for that
01:57.58SibRphreki just wanna make sure if i dial 911, it gets somewhere
01:58.08dlynesSibRphrek: and of course you have to submit your 411 data to them so that they have something to send to the 911 call center
01:58.12trixterextensions.conf typicaly
01:58.31NetgeeksThen you will need to create a dialplan that captures your 911 calls and routes them where you want
01:58.39dlynesSibR: exten => _911,1,...
01:59.05trixterI prefer to route them to the local pizza place, figured since its a take and bake if they call about a fire it would all work out
01:59.12Math[laptop]dlynes: no _
01:59.14Netgeeksthe very basic for a home office install with an analog card would be something as simple as 'exten => 911,1,Dial(Zap/1/911)
01:59.31trixterMath[laptop]: that shouldnt hurt anything since there isnt any wildcard stuff ...  but yeah
02:00.01Netgeeksbut then you need to make sure all of the inbound context for any handset you might dial 911 from has the above extension available to it
02:00.20trixterNetgeeks: what if someone is on line 1?  like the burglar outside your business trying to break in just so you cant call out :P
02:00.20Netgeeksand in order, so if you are allowing _9X. you better check 911 before that
02:00.48*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
02:00.51Netgeekstrixter: thus my qualifying 'very basic' prefix to the rest of the comment
02:01.04*** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com)
02:01.22Netgeeksi didn't want to lead into a possible discussion of what G1 meant
02:01.28dlynesWe generally won't sell a client a system unless they have at least one hard line
02:01.39trixterextensions handling doesnt do real good matching where multiple targets exist that both wildcard match, it may match correctly one time it may not the other, version seems to be the biggest determiner of that
02:01.45dlynesThat way they can still dial 911, when there's no network or power or whatever else
02:02.06trixterso its better to extend your wildcard _9X. to _9XX. (. means 1 or more) so you dont have a dup match
02:02.17*** join/#asterisk Flyboy (i=rsears@gateway.adnc.com)
02:02.40FlyboyHey Everyone
02:02.52Netgeeksdid any of that help at all SibRphrek?
02:02.53justinuhi flyboy
02:03.00Flyboygot a Digium TDM40B quiestion is anyone has some time
02:03.07Math[laptop]just ask it
02:03.12trixterI have one of those in a box waiting to be given away
02:03.17clyrradI am having a really strange problem, I have 6 DID's all configured the exact same way in iax.conf, and extension.conf. Only 3 of the DID's are able to accept incomming, and all can make outgoing.  I get the error Rejected while trying to reach 'DID@' instead of it saying 'DID@incoming'  Can anyone suggest what can be the problem?
02:03.25NetgeeksI'd use it as a paper weight
02:03.29trixterit looks quite secksi but ..
02:04.03dlynesclyrr: make sure in your channel config that you have a context
02:04.21clyrraddlynes, i have the context in iax.conf
02:04.25clyrradis that what you mean?
02:04.46dlynesyeah...make sure all of your iax peers/friends/users have a context specified
02:04.46FlyboyInstalled the 40B card, got all modules loaded, ztcfg -vv checks good, running 1.2.3, all the latest drivers from cvs installed no problem and I get no dialtone on my phone attached to the ports (which are greeg) and when I try to dial the extension attached I get a "Unable to create a channel type of 'Zap"
02:04.52dlynesand that's it's not overridden
02:04.59clyrraddlynes, I have checked they all have context's
02:05.04Netgeeksdo you see any CLI message that looks something like 'unable to find match for <insert something here>, using default context.?
02:05.19clyrradNetgeeks, nope nothing like that
02:05.26dlynesyou might have it coming into a non-existent extension, too
02:05.31trixterFlyboy: zap show channels   does that show anything?  I am almost thinking that chan_zap.so isnt loaded in modules.conf
02:05.44dlyneswhich is probably what netgeeks was insinuating
02:05.52clyrraddlynes, sorry didnt quite follow you on the last part
02:06.03SibRphrekWTF
02:06.05clyrradthey all have context=incoming in iax.conf
02:06.06Flyboytrixter, when I run zap show channels from asterisk, it tells me No such caooman 'zap'
02:06.17mzo_caooman?
02:06.19trixteryou dont have chan_zap.so loaded
02:06.23Netgeekssounds like the peer/user matching algorithm for determining which iax entry to use when getting a call from an remote iax speeking device may not be able to determine then entry you want it to use
02:06.24trixteredit modules.conf to make that happen
02:06.40dlynesclyrrad what is the extension they're trying to reach though?  Is it a valid extension in your incoming context?
02:06.42clyrradNetgeeks, what do you suggest?
02:06.44SibRphrekhttp://pastebin.com/531497  someone please help me understand this?
02:06.51*** join/#asterisk santiago (n=santiago@63.245.86.155)
02:06.57dlynesclyrrad:  make sure you don't have a #include statement that includes another context inside of your [incoming] context, too
02:07.01clyrraddlynes, yes its the DID in the form of [DID]
02:07.28Netgeeksclyrrad capture cli output, with iax debug on, verbosity set 3 or higher, and include the iax.conf section and the context in extensions.conf in a pastebin
02:07.45trixter~pb
02:07.47jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
02:07.52justinuhttp://www.sky.com/skynews/picture_gallery/picture_gallery/0,,70141-1210748-1,00.html
02:08.09Netgeeksthat should give enough info to get a handle on the issue
02:08.15clyrradNetgeeks, I have done that, the error i get is CAUSE: No authority found
02:09.06clyrradand all the DID's can make outgoing calls with out issue, its just when they come in.  So I think the problem is in iax.conf somehow? But all DID contexts are identical just username and secret are different
02:09.52Flyboytrixter, I added   load => chan_zap.so  to /etc/asterisk/modules.conf and restarted Asterisk - it crashed
02:09.55*** join/#asterisk austinnichols10 (n=austinni@dsl-10-169.cofs.net)
02:10.40litageMath[laptop]: you said earlier that SETGROUP() and CHECKGROUP() have been deprecated by GROUP() and CHECKGROUP(). however, voip-info.org says that CHECKGROUP() is only in v1.0.x
02:11.00Netgeeksclyrrad: that leads me close to the belief that the incoming iax request is not finding a match (peer, user, or friend) from all the entries in your iax.conf
02:11.14austinnichols10do start/end media ports on the server need to match the phone (cisco 7960)?  Server is 10000/20000, phone is 16384/32766.
02:11.15Flyboytrixter - also there is no such file on my system (chzn_zap.so)
02:11.33NetgeeksI'd have to see the above requested output to be any more help, clyrrad
02:11.43*** join/#asterisk Seedy (n=Seedy@cpe-24-90-35-96.nyc.res.rr.com)
02:12.04clyrradNetgeeks doing it now
02:12.09SeedyI am a total newb so please excuse my ignorance...
02:12.19Netgeeksthe wiki at www.voip-info.org has some valid text on how the selection is made when looking for a user/peer/friend match, that may help...
02:12.27ErrSibRphrek: it looks to me like your SIP control packets aren't getting through to the other end, or the other end's responses aren't getting back, and so the channel is being torn down
02:13.02SeedyCan an sip.conf file be used for incoming calls too? Or does iax.conf handle that?
02:13.04NetgeeksSeedy, you are in good company, we are all newbs at something here
02:13.13trixterFlyboy: I hpe that you see the typo :P
02:13.17trixterer hope
02:13.21Netgeekssip and iax are two different signalling protocols
02:13.26clyrradNetgeeks, here it is http://pastebin.ca/39196
02:13.53trixterif you dont have a chan_zap.so you need to build asterisk with that
02:14.12Netgeeksand I'm half wrong there, iax actually does signalling as well as media
02:14.20Netgeekslooking at it now cly
02:14.25Flyboytrixter, :-) WHat type..
02:14.33Flyboytrixter, :-) WHat typo
02:14.47trixterwas it a typo or is it really not there?
02:14.49Netgeeksso yes, sip handles both 'incomming' and 'outgoing' calls as well as iax
02:15.17Flyboytrixter, I did an updatedb and locate chan_zap and just the the chan_zap.c file
02:15.25Flyboytrixter, nothing else
02:15.40trixterwhen you built asterisk did you already have the zaptel stuff installed?
02:15.46trixterand most likely libpri
02:15.49Flyboynope, but I can build again if necessary
02:15.58trixteryou may have to
02:16.04Flyboytrixter, OK - I will try that now
02:16.09trixterbecuase it may not build that if the dependancies arent t here, I know I wouldnt if they arent there :P
02:16.12Flyboytrixter, thanks and I will elt you know how it goes
02:16.17*** join/#asterisk dijit0 (n=eric@adsl-69-106-42-147.dsl.pltn13.pacbell.net)
02:16.38Netgeeksclyrrad, and you have a iax.conf entry that starts with [4168484163] and has host=64.26.157.230 ?
02:17.06dijit0is nufone or iax.cc any good? or can anyone recommend something better/cheaper?
02:17.11clyrradyes, I have it matched exactly as you just said
02:17.26Netgeeksclyrrad and what is type= set to?
02:17.29Flyboydijit0, I am using NuFOne
02:17.37clyrradpeer
02:17.39Flyboydijit0, seems ok
02:17.48FlyboyI have about a dozen DIDs
02:17.58clyrradjust like all the other DID's 3 of which can get incomming calls
02:18.04Netgeeksclyrrad do you have a matching iax.conf entry with type=user?
02:18.11Flyboydijit0, would be nice if they were closer to my servers in CA
02:18.30clyrradNetgeeks, no they are all peer, the working ones and non working ones, thats how its been working all along
02:18.41Netgeeksokay, here is the issue
02:18.42dijit0cali or canada?
02:18.55Flyboydijit0, Calif
02:18.59*** join/#asterisk svenl_ (n=sven@AStrasbourg-251-1-35-113.w82-126.abo.wanadoo.fr)
02:19.03dijit0ahh alright, lol, thats where im at
02:19.14Netgeekswhen an iax request comes in, it's going to look for a iax.conf type=user or type=friend entry with a matching username, if it can't find that, it looks for a matching host= entry
02:19.20dijit0i notice there are cheaper rates than nufone around... but i dont know how good any of them really are
02:19.40Flyboydijit0, same here, I was going to try another carrier or two to see how well they worked
02:19.54Netgeeksif it finds neither, it is *supposed* to check the peer entries based on secret....  don't ask, I don't know who was on what drugs when they did that
02:20.08clyrradlol
02:20.14Flyboydijit0, where are you at in Calif..?
02:20.24Netgeeksanyway, the fact that you have no peer or interpeted peer via friend statement, is bad
02:20.26clyrradAny idea why 3 are working setup like this but not the other 3?
02:20.30Netgeekssorry, I mean user
02:20.42NetgeeksI would try first changing the non-working ones to type=friend
02:20.44austinnichols10saw a good review of providers on mundy.org
02:20.52dijit0bay area
02:21.00Netgeeksrestart (I don't like reloads) your asterisk if possible and test
02:21.03FlyboyI am in San Diego
02:21.06austinnichols10currently using voxee.com
02:21.18Flyboythanks austinnichols10
02:21.20Netgeeksclyrrad: bit corruption is as likely as anything
02:21.21clyrradNetgeeks, ok going to give that a try, becase the contexts are used for incomming and ougtoing, what i mean is its all IAX for calls incomming or outgoing
02:21.39Flyboyhow do you like voxee.com and do they hit you for multiple calls at once..?
02:21.51FlyboyI run a Friends and Family system with about 50 users
02:21.54Netgeeksthe context statement is only valid for 'incomming' calls, it's ignored when you make an outbound call referencing a iax.conf entry that has one
02:21.55Flyboynon-profit
02:22.13FlyboyI like NuFOne becuase they could care less how many inbound and outbound calls I have at once
02:22.31dijit0flyboy, and those are toll free numbers?
02:22.32clyrradwell i'll be damned! :p
02:22.51clyrradset as type friend it works
02:22.59austinnichols10flyboy: no - voxee is outgoing only
02:23.04NetgeeksGood to hear, Cly
02:23.12austinnichols10flyboy: but they don't care how many concurrent
02:23.20Flyboydo they care how many outgoing connections you ahve at once..?
02:23.26clyrradany security or related issues with using friend instead of peer?
02:23.31Flyboydijit0, My inbound calls are 800
02:23.39NetgeeksI read 'I like NuFOne becuase they could care less.....' and I stopped reading and agreed with the later haf of that phrase
02:23.52*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
02:23.57dijit0id REALLY like a service that has inbound caller id, WITH NAME! lol
02:24.31Flyboydijit0, sometimes we get caller ID from NuFone and sometimes we don't :-(
02:25.06Netgeeksclyrrad: type=friend is less perfect that having a type=peer and type=user, but the issue isn't that big, if you run into a problem with authentication then I would look into breaking the friend entries into peer and user, but for now... I wouldn't worry
02:25.22FlyboyNetgeeks, HA
02:25.26dijit0i c...
02:25.41clyrradNetgeeks, so type user is used for incoming and peer for outgoing?
02:26.02*** join/#asterisk unixgeek_ (n=unixgeek@216-220-234-197.exploremaine.com)
02:26.11Netgeekstype=user is scanned when a call request is recieved from a remote party (read iax box calling you in this case)
02:26.53Netgeekstype=peer is used when you reference it via a dial command, such as Dial(IAX2/acme/${EXTEN})
02:27.05Netgeekstype=friend means I'm both a peer and user entry
02:27.17wilymagehmm, any good papers on stopping denial of service attacks using asterisk? (i.e. someone spawning multiple IAX calls from multiple hosts simultaneously)
02:27.20clyrradgotcha, thanks so much for your help much appreciated )
02:27.48Netgeeksin some cases, you need a username and secret for calling through a provider, but they expect you not to challenge them when they send a call back to you
02:27.55Netgeeksin that case a friend entry wouldn't work
02:28.03Netgeeksyou'd need to split into user and peer
02:28.16familyargh
02:28.29clyrradGot it, thanks for clarifying :)
02:28.35familyi upgraded mysql connectors perl dbi etc etc and astcc stillc ant connect to the db
02:28.38clyrradMakes sense now
02:28.45Netgeeksno worries, good luck!
02:28.54clyrradthanks again :)
02:29.36Netgeeksglad I could help, the problem you had can be pesky if you don't know what you are looking for
02:29.39*** part/#asterisk unixgeek_ (n=unixgeek@216-220-234-197.exploremaine.com)
02:30.15austinnichols10Anyone know if start/end media ports on the server need to match the phone (cisco 7960)?  Server is currently 10000/20000, phone is 16384/32766
02:30.26clyrradyea tell me about it, i was looking at this for a couple hours before i decided to ask here
02:30.56Flyboyaustinnichols10, I have a 7960 - How owould I tell
02:31.00*** join/#asterisk Math[laptop] (n=Math_@modemcable148.4-81-70.mc.videotron.ca)
02:31.27Netgeeksthe phone and the server should be able to negotiate for a valid media port
02:31.35austinnichols10settings / sip / scroll down to 16 or 17 where the ports are shown
02:31.42Netgeekscan't say I've ever tried testing the fact
02:31.49austinnichols10netgeeks: tks - that's kind of what I thought
02:32.13NetgeeksI wouldn't be suprised if either asterisk or the cisco or both even ignored the setting altogether
02:32.15Errit would be broken if it required the ports to match (since you're not guaranteed that the server has any ports free in whatever range is specified)
02:32.15SeedyI'm having trouble getting asterisk to recognize DTMF. I have a simple exten => _x,1,Playback(beep) in my extensions.conf. But it never gets triggered. Is this a common problem?
02:32.40austinnichols10I'm just trying to debug a problem with SER on my Linksys (running dd-wrt).  The start/end media ports are about the last thing on my list to check
02:33.10NetgeeksSeedy, there have been many many pubs who have been saved by bankruptcy by people trying to get asterisk to work with DTMF in a simple manner
02:33.23Netgeekssave from bankruptcy that is
02:33.42*** join/#asterisk CoiL (n=bah@68.62.165.236)
02:34.32SeedyNetgeeks: So it should be easy?
02:34.41wilymageor perhaps given their woe saved *by* bankruptcy would be more appropriate ;)
02:34.54Errhm, can you do that without using some command to prompt for an extension to be input first?
02:35.08Netgeeksone could only wish it was easy, but IMHO echo and DTMF are the two most difficult asterisk issues to resolve
02:35.17Err(i.e. shouldn't use use Background() or some other mechanism to listen for tones before trying to key off of them...?)
02:35.32NetgeeksWaitExten would work as well
02:36.02NetgeeksI just assumed you had one....
02:36.02wilymagemusiconhold.c is the most buggy component, in my experience; it's the only thing that crashes our boxes on a regular basis
02:36.23justinuwow
02:36.26Netgeeksyeah, but that issue is easily fixed
02:36.34*** part/#asterisk santiago (n=santiago@63.245.86.155)
02:36.42Netgeeksnoload => res_musiconhold.so
02:37.01NetgeeksI think it's a resource
02:37.01wilymageand if one wishes to utilise musiconhold?
02:37.11Flyboytrixter, rebuild asterisk - still fails - messages now say - undefinded symboy: ast_pickup_call
02:37.49wilymagewe wrote another module relying moreso on sox, but it merely stays up for longer before leaking then dying.
02:38.02Netgeeksmake sure you are using the perfect mpg123 version (make mpg123 in asterisk source directory) and make sure you convert all the mp3's to the right format, mono 8k I believe
02:38.11Math[laptop]wilymage: musiconhold is crashing? or mpg123 is?
02:38.17Netgeeksthe native stuff in 1.2 seems to be nice
02:38.36*** join/#asterisk Triffid_Hunter (n=Splat@funkmunch.net)
02:39.06wilymagebut yeah, with the native moh worked very poorly
02:39.24*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
02:39.38wilymageall of the standard distortion, etc., that timers and silence suppression are purported to fix.
02:39.40SeedyNetgeeks: So how does one debug dtmf. I don't even think asterisk is getting my phones signals.
02:40.19ErrSeedy: does your dialplan have a default entry that waits for DTMF entries?
02:40.25Flyboytrixter, HA
02:40.26wilymageTriffid_Hunter ended up writing a piece of software that replaced using mpg123, which works amazingly for a few days, then eats the CPU and kills asterisk.
02:40.37Flyboytrixter, interesting - it fails to load the module, but now it works
02:40.45Triffid_Hunterlol asterisk eats the cpu, not sox
02:41.09wilymageNetgeeks: version of mpg123 was the correct clean version, as per the gentoo ebuild.
02:41.20Triffid_Hunterthe one that baffles me is why asterisk insists on starting seven or more moh processes, even when sending only one stream to one place
02:42.12SeedyErr: I have this exten => _x,n,SayDigits(${EXTEN}), which I thought would echo my button being pressed. But it does nothing
02:42.41[TK]D-FenderSeedy : Pastebin the entire context for us please....
02:42.42[TK]D-Fender~pb
02:42.44jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
02:42.44*** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com)
02:42.52ErrSeedy: is this coming from an analog line?  you'll need something like "exten => s,1,Background(file-to-play)" or use WaitExten() (I don't know anything about it...)
02:43.22SeedyErr: Thanks... I'll try that
02:43.30Err[TK]D-Fender
02:43.32*** join/#asterisk hellop (n=hellop@cpe-70-95-165-136.hawaii.res.rr.com)
02:43.35Err's suggestion is probably better :-)
02:43.39NetgeekswaitExten(x) waits for a keypress for the period of <x> seconds
02:43.52Errif you show us your dialplan, someone here can almost certainly help out
02:44.02[TK]D-FenderErr : yeah... I don't trust single lines pasted like that.. I have no idea what other BS is in the breaking it :)
02:44.04SeedyOk... One second
02:44.26Err[TK]D-Fender: I completely understand :-)
02:44.49NetgeeksWily: how busy is your asterisk system that has this MOH / CPU eating problem?
02:45.02familyexit
02:45.35SeedyHere is the conf file http://pastebin.com/531533
02:46.16Flyboylithi, I can't send private messages
02:46.32Flyboyzap show channels
02:46.39lithiFlyboy, ah 1 sec
02:46.40[TK]D-FenderSeedy : Got Autofalltrouhg=off I hope......
02:46.56litageafter a call has finished, how would you get asterisk to send the call's CDR to another box, or write to another file?
02:47.10*** join/#asterisk kuj (n=kuj@c-67-174-106-30.hsd1.co.comcast.net)
02:47.26Netgeekslitage, cdr_mysql pgsql odbc, etc
02:47.49Seedy[TK]d-fener: Autofalltrouhg In my sip.conf file?
02:48.54Seedythe _NXXNXXXX numbers are replaced with my reail number in that  file
02:49.06Seedylike 15551212
02:49.18Seedyor 12125551212 i mean
02:49.39Netgeeksyou are dialing in using a sip phone?
02:49.58[TK]D-FenderSeedy : in extensions.conf
02:50.30SeedyI am using a regular analog phone
02:51.10Seedy[TK]d-Fender: what does Autofalltrough do?
02:51.53litageNetgeeks: to be more specific, my CDRs are currently being stored in a mysql db using cdr_mysql, but i also need each a program to parse each cdr['s data]
02:52.50ErrSeedy: I suspect that he typo'd Autofallthrough
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02:53.19SeedySo, will my conf not work with an analog phone?
02:53.41Netgeekslitage, you could use mysqldump to remotely dump the contents of the cdr table, or you could even use a perl/php/python/ruby/c program to go and grab the specific entries you want... once in the db they should be easy to get at
02:54.05Netgeeksyou could use scp to copy the text file cdr's from the log section
02:54.05Errwell, you won't get an extension when dialed from an analog phone adapter
02:54.38Err(at least, that's my understanding - I don't have one)
02:55.27Netgeekssee my addition onto the end of your pastebin, seedy, swap out what you have for what I added, and you should get farther
02:55.54Netgeeksif you aren't sure, in incoming you could add this:
02:56.01Netgeeksexten => _X.,1,Goto(s,1)
02:56.38Flyboytrixter, THANKS MUCH - system works wonderful
02:56.52Netgeeksyou could actually add _.,1,....  but you would suffer the ire of the asterisk warning gods in doing so
02:56.52Flyboytrixter, now I can hook it into my NEX IPS2000 :-)
02:57.01SibRphrekwhere do voicemail passwords get kept?
02:57.07Netgeeksvoicemail.conf
02:57.14Flyboyor a MySQL db
02:57.19Flyboy:-)
02:57.21Netgeeksor a database if you are using such a.....
02:57.30ptiggerdine<PROTECTED>
02:57.30trixterFlyboy: np
02:57.43wilymageptiggerdine: err?
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02:57.53Erryour /etc/passwd has extensions in it?  :-)
02:58.00*** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com)
02:58.05wilymagewhoa freaky
02:58.17ptiggerdinedon't worry I just realised it was a bad idea as I pressed enter :)
02:58.45Netgeekssounds like my first marriage
02:58.52trixterI wrote a pam module to tie the unix logins into the database system we had at a unified messaging company I worked at in 1999
02:58.55ptiggerdineROFL!
02:58.59wilymageNetgeeks: on-line marriage?
02:59.10SibRphreki found my error
02:59.11SibRphrekthanks
02:59.11ptiggerdinehe marrige a PBX
02:59.14SibRphrekstill having problems calling out
02:59.15Netgeeksnope, not online....
02:59.21trixterpeople werent fond of that, they wanted them to be seperate even if they were the same ...  mostly I think its becuase it was 'strange code they didnt undertand'
02:59.30ptiggerdineROFL!
02:59.30Netgeeksstill, the similarities are amazing....
02:59.56wilymageNetgeeks: just curious as to how you sealed your fate with the press of a key
03:00.11ptiggerdinepr0n
03:00.20trixterthey freaked when I wrote javascript client to directly interact with  the database and basically bypassed some of the stupid things the tool the DB guys gave tech support (which meant he couldnt do his job) iut was sanctioned code they just didnt know that you could do such a thing
03:00.24trixterlive connecti s wonderful
03:00.57NetgeeksI'm just extending the thought... I would have made the same comment to 'I knew it was a bad idea as the front wheels passed over the edge of the cliff'
03:01.11trixterI think that scared them the most because it also showed them a huge gaping hole in their design
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03:01.20wilymagetrixter: I'd freak out if I saw javascript anywhere near my important systesm . . .
03:01.25trixterand by gaping hole I mean like you might find on a 50 cent whore on the docks on payday
03:01.55trixterwell this was for a webbrowser to parse, it just allowed network connectivity to the oracle webserver (which is art of the DB)
03:01.59trixterer part
03:02.50wilymageso you didn't use perl because ... ?
03:03.05Errprobably because web browsers don't do client-side perl
03:03.11Errjust a guess, here :-)
03:03.17trixterperl doesnt execute within a browser nearly as well
03:03.18litageNetgeeks: that requires checking the db every second though. i was thinking more along the lines of asterisk sending the CDR data to a program rather than a program periodically grabbing the data
03:04.02wilymageperl doesn't execute within a browser at all, that's the wonder of it.
03:04.14Netgeekslitage: I think you can dump cdr to the manager api (don't quote me on this). you could then have your remote app connected and listening on the manager api
03:04.15trixterit might, wouldnt suprise me if someone by now has written a plugin of some sort
03:04.25trixterbut it certainly didnt in 1999, at least none that I was aware of
03:04.52trixterNetgeeks: do you mean concurrent?  if it werent I would be afraid of losing CDR
03:04.56Netgeekslitage: my knowledge in cdr to manager api comes from either reading it somewhere or having dreamt that i read it somewhere
03:05.20Netgeekstrixter: I would hope it could be done in parallel with permanent storage methods
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03:06.03SeedySo... An analog phone won't register extensions with asterisk from what I have heard. So how can I get it to intercept what numbers I am pressing!?!
03:07.11Netgeeksthe only time I worked with analog phones, I used DISA and immediate=yes to capture the dtmf entries
03:07.28Netgeekssee immediate= directive information in your sample zapata.conf file
03:07.40Netgeeksand DISA is an app you can read about at www.voip-info.org
03:07.46trixterat the very least you can do a 'userevent'
03:07.59trixterin your dial plan, but it might be better to watch call creation and termination
03:08.07Netgeeksagain, I warn you that I am no where near an expert in analog phone and asterisk... more like a newbie
03:08.13Math[laptop]I just make up a digitmap for the ATA
03:08.18Netgeeksso take anything I say with a grain of salt and do your homework
03:08.58trixterwell what you proposed can be  done
03:08.58*** join/#asterisk whoknows (n=nav_swt@cpe-70-117-5-47.satx.res.rr.com)
03:09.08trixterhowever it may require a userevent if the functionality isnt already there
03:09.42whoknowsneed some help in configuring pri for two separate phone lines using separate context
03:09.45trixteralthough for the most part you can watch the call setup/teardown with  the manager api and do it that way, you may not have *all* the cdr info but you would have a general overview
03:09.52trixterI personally like tossing CDR into a DB and accessing it that way
03:10.03SeedyOh, just to make things clear too. My asterisk only accepts incoming calls. And I am making these calls from an analog phone
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03:10.17Netgeekshe wants push methodology versus a pull, though, Trixter
03:10.37trixterahh there is an app for that
03:10.42trixterforget its name (broadcast?)
03:11.08trixterbasically its a message thing where the server sends arbitrary messages as defined in your dial plan, its not even in add-ons but it does exist, I recall the author saying something um ...  a year ago on asterisk-users
03:11.11trixtermaybe only 6 months ago
03:11.16whoknowscan anyone help this newbie with pri card
03:12.23whoknowsconfiguring pri for two different companies phone number
03:12.28whoknowspri is 16 channel
03:12.41whoknowsneed help in zapata conf file
03:13.08trixterby send I mean it has a bunch of programs that listen and you select which get the message ...  thought my explanation wasnt exactly clear earlier
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03:34.32JunK-Yhey hey angler__ .
03:38.56*** join/#asterisk mattwj2006 (n=Matt@dialup-4.254.83.80.Dial1.Chicago1.Level3.net)
03:39.40mattwj2006so can you make a call on an actually asterisk box (not a sip phone or iax softphone) using a soundcard
03:39.49mattwj2006if so how do you do it?
03:40.41Math[laptop]dial extension@context
03:40.42Math[laptop]on the cli
03:41.24mattwj2006do you have to load any special kernel mods?
03:41.50mattwj2006ex zaptel
03:42.20Erryou need soundcard drivers that provide the oss or alsa API, and load chan_oss.so or chan_alsa.so (whichever is appropriate)
03:43.33mattwj2006awesome :)
03:44.31litageNetgeeks: hahah thanks for that. i'll look into the manager api
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03:57.16whoknowsasterisk cannot detect if the phone is busy and go directly into voicemail
03:57.28whoknowsis there any way to fix this guys
03:57.33Qwellwhoknows: define busy
03:57.55whoknowsthks qwell, but how do i do that
03:58.00whoknowsin extensions.conf file
03:58.11Qwellby typing words, that explain to me what you think "busy" means
03:58.37Math[laptop]whoknows: what kind of phone
03:59.44whoknowsif the extensions is busy then the call should directly go into voicemail rather than ringing and then going to voicemail
03:59.53whoknowshope i make myself clear
04:00.19Math[laptop]what do you mean by busy
04:00.25Math[laptop]that's what he's asking
04:01.20whoknowswell the person at 103 is on phone and talking and if you are calling 103 then it should go to voicemail
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04:21.31*** topic/#asterisk by drumkilla -> Asterisk 1.2.4 and Zaptel 1.2.3 Have been released! => Includes a significant memory leak fix for Asterisk
04:22.08*** join/#asterisk coppice (n=chatzill@46.155.17.210.dyn.pacific.net.hk)
04:22.15opsyswhat had the mem leak asterisk or Zaptel?
04:22.18*** join/#asterisk burton (i=mimx@w201.ljudmila.org)
04:22.27drumkillaopsys: Asterisk
04:22.31drumkillait was the expression parser
04:22.39Qwelllike $[]?
04:22.42drumkillayes
04:22.44Qwellnice
04:23.03drumkillathank murf for fixing it and Corydon for getting it merged ...
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04:33.25FuriousGeorgewadup kids
04:34.25FuriousGeorgemogorman: que pasa
04:36.47clyrradI have call forwarding app enabled on my * box and a sipura ata, the ata does not let me do the *21*forward_number# like other voip phones do, anyone know how to use call forard with this ATA?
04:38.11Math[laptop]*21*? isnt that on cells
04:38.35clyrradno i have that enabled on my asterisk box
04:38.40clyrradas teh call-forward app
04:38.48Corydon76-homeand blame Corydon for not getting it into 1.2.3
04:38.58Flyboygood night everyone !!
04:41.41wasimwoo hoo ... 1.2.4
04:42.57SwK1.2.4?
04:42.59SwKalready?
04:43.05fugitivowell, i knew this was going to happen
04:43.17fugitivothat's why i stick with 1.2.1
04:43.17fugitivoi'll wait for 1.2.5
04:43.28Corydon76-home1.2.1 also has the memory leak
04:43.33fugitivoi know
04:43.39fugitivobut who knows what has 1.2.4
04:43.40SwKasterisk 1.X.X has a memory leak
04:43.57Corydon76-homeNo, it's just 1.2.x prior to 1.2.4
04:44.16Corydon76-home1.0.x had a significantly different expression parser
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04:47.34__Krush__Hi all, anyone know a source for Grandstream ringtones?
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04:48.47__Krush__Hi all, anyone know of a source for good Grandstream rintones?
04:48.56__Krush__ringtones that is
04:49.48robbytkrush: a moog and an ftp server
04:50.07__Krush__moog?
04:50.08*** part/#asterisk mattwj2006 (n=Matt@dialup-4.254.83.80.Dial1.Chicago1.Level3.net)
04:50.18robbytgoogle it ;)
04:52.10__Krush__google "grandstream ringtones" yields only 2 entries in a forum with a similar request...
04:52.35*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
04:52.48knight_hey SwK!
04:53.00robbytkrush: my point is, i haven't been able to find much for ring tones outside of making them my self
04:53.19__Krush__ah...I'm not the only one then...
04:53.29robbytkrush: http://freesound.iua.upf.edu/
04:53.32Corydon76-homerobbyt: Wendy Carlos would be proud
04:53.43robbytthere might be something there, but be ready to dig!
04:55.22robbytkrush: grab some samples off of free sound and convert them in audacity
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04:56.59__Krush__thanks robby...which specific format as seems quite nitpicky?
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04:57.29robbytkrush: i'm not sure what the GS uses, i'm sure the format is listed on voip-info though
04:57.52__Krush__Ok 10x will have a browse
04:57.56robbythaha here you go: http://freesound.iua.upf.edu/samplesViewSingle.php?id=14252
04:59.28__Krush__:)
05:00.30__Krush__Is there some utility to convert mobile phone ringtones to GS format?
05:00.42robbytmidi files?
05:00.58__Krush__MIDI or polyphonic..
05:01.11robbytaudacity is a free audio editor
05:01.17robbytbut you can't import midi files into it
05:01.29robbytmidi is basicly sheet music
05:01.52robbyti think winamp might let you dump midi files to disk as wavs
05:02.08robbytthen, for example on the cisco phones, i beleive they need to be compressed in GSM
05:02.15robbytas 22khz 8bit files
05:02.21robbytso that's where audacity would come in
05:02.46*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
05:02.47__Krush__good...will consider this route than...have tons of mobile phone ringtones....
05:03.19robbythaha here's another great one:
05:03.20robbythttp://freesound.iua.upf.edu/samplesViewSingle.php?id=11729
05:03.30robbytwow, know how i'm going to spend my morning at work
05:03.35{zombie}you can use timidity to convert midi to wav
05:03.46robbytmaking up new amazing annoying rings for my cisco deskphone
05:03.50{zombie}and gs provide a hacked version of sox to convert to their ringtone format
05:04.51*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
05:06.51__Krush__thanks zombie will have a look
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05:10.15__Krush__What are you opinions about the different SIP phones money can buy?
05:10.55robbyti like the polycom 501s
05:11.30CANO-1982__Krush__, you could try timidity
05:11.33robbyti've worked with the newer sipura, cisco 7940s, and polycoms
05:11.39CANO-1982is the best choice
05:11.47__Krush__Bought GXP2000s as a low cost intro to Asterisk
05:12.06__Krush__robby new sipura is 941?
05:12.21robbytumm, the one with the cisco up/down button
05:12.25robbythold on, i'll check
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05:12.46__Krush__cisco/linksys....?
05:12.53austinnichols10wooHoo!  double-nat finally working!
05:13.08robbyt841
05:13.15robbytit's a toy
05:13.18__Krush__price-wise same as polycom 301
05:13.45robbytno wait
05:13.54robbytit is the 941
05:13.55robbytsorry
05:14.01robbytit's a toy-
05:14.08robbytthe poly301 is nice for the money
05:14.16robbytbut the kicker is that there's no speaker phone Mic
05:14.34robbytthe 501 is the best i've used though
05:14.40robbytamazing speaker phone
05:14.59__Krush__ok...
05:15.08__Krush__501 not yet ce certified in Europe here
05:15.14__Krush__500 ok but not 501
05:15.22robbytahh
05:15.34robbytnot sure what the difference is-
05:15.49__Krush__seems more mem for bigger better firmware
05:16.24robbyt<shrug> the newest firmware has https support for the services buttons
05:16.30robbytguess that takes up some room
05:16.44__Krush__yeps
05:17.27__Krush__Would you know of some AAH resource to upgrade to latest asterisk without breaking config?
05:17.52*** part/#asterisk CANO-1982 (i=alejandr@201.255.53.122)
05:18.17robbytAAH?
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05:18.29__Krush__Asterisk@Home
05:18.35wasim* @ ~
05:19.21robbytahh
05:19.28robbyt*@~ hehe
05:19.47robbytnot sure krush- never used AAH
05:19.54austinnichols10krush: try tom vile at baldwintech.com
05:20.15austinnichols10wait - wrong url
05:20.30austinnichols10baldwintechsolutions.com
05:20.39austinnichols10He helped me get started with my AAH
05:21.01austinnichols10tell him michael dyer reffered you
05:21.39austinnichols10there are a few more names in the AAH forums - do a search on baldwintechsolution.com and you'll find the others
05:22.02__Krush__Ok...will give it a browse
05:22.11austinnichols10which version you running?
05:22.38__Krush__2.2 but read that 2.4 has addressed major mem leaks
05:22.54austinnichols10I just upgraded my 2.2 to 2.4
05:23.04__Krush__how...?
05:23.10austinnichols10let me get the link
05:23.15__Krush__any reference docs?
05:23.19austinnichols10yes
05:23.45austinnichols10do you know how to do the yum updates?
05:23.50__Krush__yes
05:24.50__Krush__anything really tangible from the change 2.2 > 2.4?
05:25.13austinnichols10nothing that I've needed
05:25.57austinnichols10you need to be careful because AAH uses AMP.  AMP has a lot of deprecated commands and if those commands go away between 2.2 and 2.4 your SOL
05:28.11__Krush__What is the better alternative to AMP...besides hand changing the configs?
05:28.20wasimvi
05:28.36Qwellhire a monkey to do it
05:28.57rob0tt-monkeys.gsm
05:29.24__Krush__I see there are some monkeys dangling on the tree...
05:29.47austinnichols10Here's the link to what I used: http://sourceforge.net/forum/message.php?msg_id=3542026
05:29.49rob0I resemble that remark!
05:30.02austinnichols10you should readh the whole thread so you can see what's really happening
05:30.31austinnichols10afterwards I did a yum update which included some kernel stuff so I had to rebuild zaptel afterwards
05:31.03FuriousGeorgewhere are those modules i gotta delete before upgrading.  i thought they were in /var/lib/modules/asterisk
05:31.06austinnichols10I read that doing this manual update may break the fax over IP receiving stuff
05:31.56austinnichols10in /usr/src
05:32.24FuriousGeorge?
05:32.28FuriousGeorgeno its not
05:32.29__Krush__OK...thanks
05:33.31austinnichols10sorry george - was on mine
05:34.20FuriousGeorgeyou were close though, theyre in usr lib
05:36.13__Krush__Bye alll...
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05:55.10mattwj2006I just made my first call from cli
05:55.20tzafrir_laptopusing what?
05:55.25mattwj2006pretty sweet....I have always been just doing sip or iax
05:55.35tzafrir_laptopdial?
05:55.37mattwj2006alsa
05:55.40mattwj2006yup
05:56.22mattwj2006I am still quite the noob with asterisk
05:56.27inv_Arpmattwj2006: how?
05:56.28mattwj2006:)
05:57.18mattwj2006well you have to load => chan_alsa.so
05:57.24mattwj2006reboot it
05:57.32mattwj2006dial extension@context
05:58.21mattwj2006does that help inv_arp?
05:58.21inv_Arpahh
05:58.26inv_Arpthx
05:58.39mattwj2006yup
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05:59.55libilaI'm setting up my dgium card for the first time. I have four FX0 ports on it. When I do ztcfg -vv I get 'Channel map: Channel 01: FXO Kewlstart (default) (Slaves: 01) etc etc etc 4 channels configure. Notice: Configuartion file is /etc/zaptel.conf line 26: Unable to open master device `/dev/zap/ctl`' line 26 is 'defaultzone=us' I did ls /dev/zap and there isn't a directory by that name. Could someone explain why I'm getting that error?
06:00.29*** join/#asterisk coppice (n=chatzill@30.196.17.210.dyn.pacific.net.hk)
06:01.08libilaor why it's looking for /dev/zap/ctl when it doesn't exist? lspci detects my card: Communication controller: Tiger Jet Network Inc. Tiger3xx Modem/ISDN interface
06:01.12Qwelllibila: README.udev
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06:03.55mattwj2006anyone know how to give my sound card an extension?
06:03.59mattwj2006I am using ALSA
06:04.06QwellDial the alsa channel
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06:15.14mattwj2006Qwell you do know what the dial syntax is?
06:15.47mattwj2006a brief look at voip-info.org didn't turn up anything
06:15.48*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
06:15.55QwellDial(Console/dsp) ?
06:16.07*** join/#asterisk yoyoma (n=S@mbl-99-58-31.dsl.net.pk)
06:16.50yoyomahello
06:18.21yoyomawhen i make an external call, there is no audible ring for the called party at the other end. can anyone help here?
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06:19.58tzafrir_laptopmattwj2006, how application dial
06:20.03yoyomahello
06:20.38tzafrir_laptopyoyoma, what type of call? from what phone to what phone?
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06:22.23yoyomatzafrir: I have a TDM04B with asterisk 1.2. Internal calls, ie. extension to extension calls, work fine. When calls come in from the outside, all internal phones ring. When I place an external call using the zap channel to a regual pstn number, the phone at the other end does not ring
06:22.59yoyomaif they happen to pick up the phone at the moment i'm trying to call them, the call is connected...
06:23.07yoyomaat my end i hear the ring back tone as well...
06:23.17yoyomait's just that the phone does not ring for them
06:23.18Qwellyoyoma: Did you plug in the power connector?
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06:23.33Qwellit won't be able to generate ring voltage otherwise
06:24.07austinnichols10or could be fxs/fxo swapped
06:24.30yoyomaQwell: excellent point.... recently the digium board was reinstalled... it may have been left unplugged. let me check
06:25.06mattwj2006it worked.....but I got a notice
06:25.24mattwj2006NOTICE[5509]: rtp.c:510 ast_rtp_read: Unknown RTP codec 72 received
06:25.56yoyomaQwell: the power connector is plugged and the lights at the ports are on
06:25.59austinnichols10anyone know if PRI circuits need to be rx/tx gain tuned (for echo)?
06:27.12OloBolaI would like a caller to be able to "press 2" to speak with whoever, which is then forwarded to their cell phone. How can I do this from my php script?
06:28.01opsysaustinnichols10: PRIs shoudl NOT need to be turned as they are a balanced cirquit
06:28.14austinnichols10that's what I thought
06:28.18yoyomai had an older version of asterisk using the same hardware and this problem was not present then
06:28.30opsysaustinnichols10: PRIs should NOT need to be tuned as they are a balanced circuit (sorry late)
06:29.27austinnichols10opsys: I'm having echo issues with cisco 7960s going to PSTN endpoints via the PRI.  Any idea where else to look since the PRI should be clean?
06:30.03opsysAre you in FLorida, using a Bell PRI??
06:30.17austinnichols10florida using an FDN PRI
06:31.31robbytpri isn't always echo free! :)
06:31.55opsysFDNs are trick, they don;t put good echo cans on them. YOu can try calling from a landline a DID that runs echo. DO NOT use a Cell. If you have echo there its on your Telco side, Yuo can try using a dirrect echo can in Asteisk. What kind of hardware do you have?
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06:32.41austinnichols10TE110P (was thinking to switch to an EC card)
06:33.04opsyswhat are you getting as far as interupts in /proc, have you run zttest?
06:33.08austinnichols10FDN = Florida Digital Network
06:34.27austinnichols10zttest runs at either 99.987793 or 100
06:34.33austinnichols10jumps back and forth
06:34.48austinnichols10avg 99.991708
06:35.31opsysanything above 99.975 is good. Congrats your PRI to PCI interface is solid, now on to the the next problem.
06:36.10austinnichols10what am I looking for on interrupts?
06:37.49litageare there any other options/settings you can specify in cdr_manager.conf besides "enabled=" ?
06:38.16opsysaustinnichols10: did you get my reply?
06:41.05austinnichols10not to the x600 question
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06:42.31austinnichols10opsys: you still there?
06:42.56opsysaustinnichols10: I'm still here
06:43.16austinnichols10k - interrupts seem to be > 1000/sec (dual proc box)
06:43.43opsysYour OK on interupts too.
06:44.09austinnichols10is there a better way to test other than just cat | interrups?
06:44.59opsysyou can do.. while: do; cat /proc/interrupts >> /tmp/out; sleep1 ; done
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06:45.42opsysafter about 10sec ctrl-c and the grep for the module (wcte1xxp) and add up the differences.
06:46.15yoyomawhat reasons could there be for the no audible ring at an external number? i have a pstn line coming in to one port of my tdm04b. when i place an outside call using that pstn line, the phone at the other end does not ring but i get a ring back tone.
06:46.35austinnichols10I'm running off of an aah build so there's no 600, but I do have dids.  What does 600 normally do?
06:47.17opsysadd this into your menu context.  exten => 600,1,Echo()
06:48.09opsysecho does just that it echos everything it gets back to you.  If you are on a landline there is a chance you will get an echo if it on the Teclo side.
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06:50.09austinnichols10got it
06:51.29opsysYou could also try this on the VoIP side and see if you get an echo also.
06:51.37opsysHow BAD is the echo?
06:53.39austinnichols10testing now
06:53.57opsysyoyoma: once you place the call on the PSTN the ringing comes from a ring gen at YOUR CO. Unless of course you have Asterisk set to generate rings.
06:54.17austinnichols10opsys: check your PM.  I don't think it's bad at all
06:54.48opsysWhat PM
06:55.06austinnichols10I opened up a separate window - hmmm
06:55.26trixterI would open a window but its cold outside
06:55.53opsysI know it about 67F hear on the coast of Florida, BRRRRRRR!!
06:56.24opsystrixter: Are you in SFO?
06:56.32austinnichols10going down to 56 tonight
06:56.37trixterno about 150 miles or so away
06:57.21opsystrixter: Thanks for some of the info on your site, really helpfull, got me out of a crunch the other day
06:57.30austinnichols10trixter: have you tried hooking up a 'real' phone and making a test call?
06:57.43trixterum which site and what info?
06:57.44austinnichols10sorry - not trixter, yoyo
06:57.56trixteraustinnichols10: define 'real'
06:57.58yoyomayes..
06:58.11austinnichols10yoyoma: princess phone
06:58.30opsysthe only way its going down to 56 here is if my A/C mal-functions. I am on the Beach. GulfStream BABY
06:58.47yoyomaaustinnichols10: i have a second line here that is not connected to my asterisk server...
06:58.59Qwellmeh, you people and your "weather"
06:59.11Qwell"snow" and "rain" and the like
06:59.21opsysyoyo: are both lines off of the same CO?
06:59.39opsysQwell: (err North)
06:59.52Qwellopsys: hrm?
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07:01.27opsysQwell: I figured cause your name is North that your up there Lat wise.
07:01.41Qwellnot so much, no..
07:01.51Qwellpretty far south actually
07:01.59opsysSorry, How far??
07:02.03Qwellsouthern CA
07:02.48opsysAhh SoCal the area where the weather is great but you may find your house at the bottom of the Vally!!
07:03.23austinnichols10opsys: what do you do for work?
07:03.49opsysaustinnichols10: Asterisk Consulting and integration help.
07:04.04austinnichols10cool - send me info
07:04.13austinnichols10ALWAYS helps to have someone local
07:04.30opsysgo to www.opsys.com. Site outdated but contacts are still good. And yourself.
07:04.36yoyomaopsys: yes both lines are from the same co
07:04.49austinnichols10www.tieronehosting.net
07:05.03*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
07:05.25[TK]D-FenderWow... another quick release....
07:05.27austinnichols10ha - I tried to call you guys when I was first thinking about setting up a server
07:05.44*** join/#asterisk kimosabe (n=kimosabe@201.133.216.51)
07:05.52opsysaustinnichols10: How long ago??
07:06.04Supaplexaustinnichols10: but the line was busy eh?
07:06.06Supaplex;)
07:06.38austinnichols10a couple of months ago
07:06.43austinnichols10right before Christmas
07:06.49Math[laptop]thats 1 month ago
07:06.58austinnichols10it's 2:05
07:06.59opsysyoyo: If asterisk is ringing and you pick up the other phone are the calls bridged.
07:07.56opsysopsys: sorry we missed ya.
07:07.57yoyomaopsys: yes the calls are bridged. i get ring back tone when i call the other number. the other phone does not ring, but if i pick it up i'm connected
07:08.01austinnichols10np
07:08.30*** join/#asterisk Abbas (i=Abbas@203.81.200.29)
07:08.48opsysyoyo: try swapping lines and see if problem continues, alt. can you call second line from other phone (ie cell)?
07:09.17opsysAustinnichols10: Call me at 305-503-3000 ext 122. Lets see if we can kill the echo prob.
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07:11.00yoyomaopsys: i'll give that a shot. thanks a lot for your help!
07:11.13opsysyoyoma: no probblem
07:11.48austinnichols10opsys: when is a good time to call?
07:12.06opsysI'm up now.
07:12.21austinnichols10cool
07:13.10clive-found it
07:14.13*** part/#asterisk dataworm (n=dataworm@modemcable192.46-130-66.mc.videotron.ca)
07:15.04lucasjbHiyas, just a quick one, how do I tell WaitExten() I want more than one digit? I've set my Digit timeout to four seconds, but the system always continues when my user presses their first key.
07:17.10*** join/#asterisk Psykick (n=anon@203-167-215-33.dsl.clear.net.nz)
07:17.12Psykickhi guys
07:17.23PsykickI got a weird yet kind of interesting question for ya'll
07:17.44Psykickjust wondering if its possible to change the firmware in an avaya phone to talk either SIP or IAX2
07:18.03PsykickI know avaya makes SIP phones but just wondering if it's possible using other avaya models
07:18.30PsykickI've got about 20+ avaya 5420 phones from our previous phone system
07:18.45[TK]D-FenderGoogle it...
07:19.04Psykickbeen googling it
07:19.18Psykickthought I'd come to the one place I might get an answer
07:24.16coppicecoming here sounds really desperate :-)
07:24.49Psykickkinda am :)
07:25.21dpryoPsykick: avaya.com probably knows ;)
07:26.46Psykicknot like they'd tell anyone
07:26.52*** join/#asterisk burtonez (i=mimx@w201.ljudmila.org)
07:27.12Psykickthey'd have people changing firmware left right and center then be expected to fix the problems of those people
07:27.18dpryoI've found sip-images for 4620 on their site.
07:27.26Psykickso I don't think they'd mention it
07:27.35Psykickyeah I found those as well
07:27.42dpryoAll my 40 phones are running sip now :)
07:27.56Psykickdo you have original 4620's?
07:28.03Psykickor another model?
07:28.17dpryoI have 4620 phones.
07:28.28Psykickwell there ya go ... I have 5420 phones
07:28.36Psykickand around 20+ of them
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07:28.47dpryoI'm sure you find what you look for if you put your mind to it :)
07:29.40Psykickwell considering I have 5420 phones that are not advertised as SIP phones I don't think I'd be able to do what I want with them which is change the firmware so that it supports SIP
07:30.13dpryoMy 4620 were not sip-phones, until i downloaded another firmware from support.avaya.com.
07:31.18lucasjbCan anyone explain to me why WaitExten() is returning after only one digit is entered?
07:31.28PsykickI suppose I could at least screw up 1 phone ... an expensive test though
07:31.36[TK]D-Fenderlucasjb : Pastebin you extensions.conn
07:31.37[TK]D-Fender~pb
07:31.39jbotpb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
07:31.41Psykickdamned phones cost $550
07:31.46*** join/#asterisk pobre (n=seymore@203.215.73.192)
07:31.59Psykickthat's each
07:32.40dpryoHeh
07:33.18dpryoSo why buy them in the first place?
07:33.27dpryoSounds like a really bad deal :)
07:33.31Psykickwe originally had an entire avaya phone system
07:33.56Psykickbut ... for basic call reports the suppliers wanted us to pay $30K
07:34.05Psykickon top of what we had already paid
07:34.13lucasjb[TK]D-Fender, http://pastebin.com/531775
07:34.15dpryoHeard that one before :)
07:34.25Psykickeven though we were supposed to get CCC
07:35.48Psykickconsidering the 5400 isn't exactly advertised as an IP phone I wonder if it would be possible to change the firmware to another firmware that supports SIP
07:36.17[TK]D-Fenderlucasjb : Don't just show me a little clip of what YOU think is relevent.  Pastebin it ALL
07:39.04lucasjb[TK]D-Fender, hmm... I'll try...
07:39.21*** join/#asterisk EriSan (n=erisan@151.8.109.88)
07:40.35coppicemost PBX phones only work as RIP phones, especially when they are OFF :-)
07:41.48bigjbpsykick, you can get call reports for next to nothing
07:41.55bigjbon a an avaya system
07:41.57Psykicknot on avaya you can't
07:42.06bigjbyes you can
07:42.07Psykickcbc ... yeah fine
07:42.08lucasjb[TK]D-Fender, http://pastebin.com/531780
07:42.20Psykickwhat we were supposed to get was CCC
07:42.26Psykickcbc is just a waste of time
07:42.31lucasjb[TK]D-Fender, the interesting context is [atp-incoming]
07:42.46bigjbnope, when i get to work i will find the name of the company that makes the software
07:43.09bigjbin fact i might even be able to find you the site now
07:43.12Psykickbigjb: doesn't matter now .... using asterisk and we have all the reports we need
07:43.30Psykickonly thing is just trying to re-use the equipment
07:43.38bigjbhttp://www.ctidata.co.uk/office_product.htm
07:44.15*** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
07:45.19Psykickbigjb: errr ... don't see any kind of reference to integrates with avaya IP office .....
07:45.30bigjbit does
07:45.33*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
07:45.38[TK]D-Fenderlucasjb : You are trying to use "n" as the first priority on several extens in there which is something you can not do.  You therefor have NO valid extens in taht menu and everything leads to "i"nvalid
07:45.41bigjbive installed it to use several systems
07:46.46Psykickbigjb: doesn't matter anymore
07:47.17bigjbheh definitely not if your using asterisk
07:47.32lucasjb[TK]D-Fender, Ah, ok let me fix that...
07:47.59Psykickbe good if I could change the firmware in these 5400 phones to use Avaya's SIP firmware
07:48.21Psykickassuming that the phones still work after changing the firmware
07:48.22dpryoWhat is the difference on 4600 and 5400?
07:49.04lucasjb[TK]D-Fender, Ah, that's fixed it - thank you!
07:49.06Psykickother than the 4600 series being IP phones .... can't really differentiate between the two
07:49.17*** join/#asterisk corruptor (n=andrew55@www.tae.ru)
07:49.37[av]banibleh
07:49.54bigjbthat is the difference
07:49.57knight_I am having problems getting ast to ast via iax nat to auth
07:50.03Psykickwouldn't it just really rip your shorts if there is no difference between the different series other than firmware
07:50.09dpryoPsykick: But they do have an ethernet interface?
07:50.19Psykickdpyro: yip
07:50.22[TK]D-Fenderlucasjb : ywc
07:50.23*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
07:50.33Psykickpower is supplied over ethernet as well as firmware updates
07:50.43dpryoJust as with 4600.
07:51.00dpryoI'd try to put in a 4600-sip-image :)
07:51.03bigjbthats the same with all ipoffice digital phones
07:51.13Psykickit's tempting ... but $550 is a fair bit o cash to waste if it screws the phone
07:51.27dpryoIt probably won't. You can always put back the old image.
07:51.52dpryoThe netbootloader is probably never changed.
07:52.18bigjbgive me 3 or 4 houts and once im at work i can get a request in with my supplier
07:52.21dpryoAnd I bet they have some kind of a system to check if the image is compatible
07:52.39Psykickhmm ...
07:52.48Psykickcan you email me?
07:52.54Psykickor leave me a memo
07:52.54bigjbyup
07:52.55*** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu)
07:53.09bigjbnot a problem
07:53.12Psykickkewl
07:53.19bigjbi would be interested to see myself
07:53.35PsykickTHANKS! bigjb
07:54.22bigjbwe have a ipoffice 406 v3 with 5402 and 5420 handsets
07:54.24*** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13)
07:54.40bigjbi also have a asterisk box running on the same network
07:54.45*** join/#asterisk tomas_ (n=tomas@78.121.broadband3.iol.cz)
07:55.04bigjbbeen meaning to speak to supplier about best way to get them to talk to each other anyway
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08:01.10Psykickbigjb: I found a doc somewhere on the net for getting an asterisk box to act as a gateway for avaya ip office
08:02.47*** join/#asterisk S3RG3US|W (n=SERGEUS@ippe-245.ippe.ru)
08:04.23knight_k
08:05.25dpryoPsykick: care to share?
08:06.19*** join/#asterisk gevious (n=chatzill@dsl-146-112-82.telkomadsl.co.za)
08:06.41geviousHi All
08:07.49*** join/#asterisk acehunky (n=chat_jok@221-128-138-148.exatt.net)
08:08.10Psykickdpyro: http://www.inventigo.co.uk/home/forum/index.php/topic,20.new.html
08:08.22Psykickthey just talk about their setup ...
08:08.45*** join/#asterisk razu (n=razu@tln-kontor.norby.ee)
08:08.57PsykickI believe it works quite well though ... have heard of someone else doing the same thing
08:09.16Psykickok well ... goin home now
08:09.21Psykicktalk to ya'll tomorrow
08:11.14*** join/#asterisk webmind (n=webmind@feather.perl6.nl)
08:12.16knight_an asterisk over iax over a nat does not seem to authenticate to a remote ast server.... ideas?
08:13.43OloBolaI need to use call files to forward calls from an AGI script?
08:14.26*** join/#asterisk diLLec (n=dillec@a15182648.alturo-server.de)
08:14.32OloBolaso person calls, recording: "press 1 to call cell phone", user presses 2, AGI script drops a call file?
08:15.48OloBolapress 2,
08:17.07lucasjb[TK]D-Fender, thanks again for your help.
08:17.08*** join/#asterisk Bambr (n=Bambr@213-35-235-26-dsl.end.estpak.ee)
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08:19.36mmmToophi...anyone got an idea what this WARNING is:  "Received mini frame before first full voice frame"
08:20.16clive-mmmtttooop, ignore that,,,just bad connectivity normally
08:22.05mmmToopwhat do you mean by bad connectivity...? LAN issues?
08:23.16clive-I see that on international linmks, but I don't realy have expereince over a LAN
08:23.28clive-you can usually ignore those warning messages
08:24.47*** join/#asterisk nurfe (n=rgff@h24-207-70-68.dlt.dccnet.com)
08:25.38*** join/#asterisk OnuR (n=HaLuK_Le@62.220.216.163)
08:25.46mmmToopmaybe the soft phones that we are using are doing it...?
08:27.52OloBolacan someone suggest a way to forward a call from an AGI script?
08:28.22*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
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08:29.08*** join/#asterisk Password (n=oFF@62.220.216.163)
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08:32.27pobreim using TDM400P im know on asterisk and dont know how to make a call via this zap card?
08:32.39pobreim using TDM400P im new on asterisk and dont know how to make a call via this zap card?
08:33.01pobreit is configure no alarms
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08:35.35robbie2helllo
08:35.45*** join/#asterisk O-Zone (n=O-Zone@moloch.asb.unisi.it)
08:35.47O-Zonehi all
08:36.39OloBolacan someone suggest a way to forward a call from an AGI script?
08:37.40O-Zonethere's a way to do ringing on two extn ?
08:38.13robbie2anyone here configured an isdn 10 ?
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08:40.01OloBolaI think you need to use & symbol or something to ring two extensions
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08:44.05KeX_WorXhi
08:44.28KeX_WorXi'v a problem getting a variable from an agi script back into the dialplan
08:44.35KeX_WorXhere are my scripts: http://phpfi.com/99391
08:44.54KeX_WorXcan someone pls look at them and give me a hint where the problem is/could be ?
08:45.49KeX_WorXwhen i call the sh script from the cmd line, i get the expected result, but if i call it from within the dialplan i just get nothin : /
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08:51.56robbie2are rtp ports udp or tcp ?
08:52.00robbie2for forwarding ?
08:54.14Krilludp i'd imagine
08:54.53robbie2loos so
08:55.47*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
08:57.12KeX_WorXcan someone pls look at that and probably find an error? http://phpfi.com/99391
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09:12.30jhiverhi all
09:12.33dogtanian<PROTECTED>
09:13.07*** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13)
09:13.22enemy^xI`ve been trying to find out why my MOH is causing so much misbehaviours with asterisk. When running mpg123-0.59r, it seems more stable than any other mpg123 release. But still, it seems to crash once in a while... Anyone have any good suggestions to keep this stable? I`m running 1.2.3
09:13.30jhiveris there a way to have the current time (in seconds or so) in a variable so that I can get asterisk to record each of the calls I place through it?
09:18.53*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
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09:23.31jhiverseriously I can't I do something like Servar(TIME,Now()) ?
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09:30.48justnullingwhat is this error Auto-congesting call due to slow response?
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09:34.49rezafilberts, eh?
09:35.08*** join/#asterisk gvag11 (n=gvag11@ipa201.5.tellas.gr)
09:35.11gvag11Hi all
09:35.52gvag11Any suggestions of how can i monitor an Asterisk box ? Monitor the process and the ZAP channels....
09:35.55*** join/#asterisk P0L0 (n=n0n3@140.Red-83-58-255.dynamicIP.rima-tde.net)
09:38.18diLLecconnect via manager account. there are zap status commands available
09:39.02gvag11dillec : yes i know but i would like to find an application which can do this 24h and report any alarms, any idea ?
09:40.04diLLeci think that alarms are sent by events to the manager
09:41.28gvag11dillec: yes they are comming like events, but instead of make something from scratch (catch the event and then report) i am looking to find if there is something ready to do that.
09:43.06diLLecah ok - i thought you need a way to implement something :-)
09:43.49gvag11no ... ;-)
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09:46.02*** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it)
09:50.34areskigood morning
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09:52.14*** join/#asterisk phpboy (n=shane@196.26.21.106)
09:52.57phpboyHey all, I'm trying to make a call with my Swissvoice hardphone... I can call local extensions but I can't call out to my PSTN... these are the errors I get on asterisk
09:52.58phpboyJan 31 11:51:36 WARNING[7101]: chan_sip.c:703 retrans_pkt: Maximum retries exceeded on call 106b3030-aa10a8c0-13c4-17d-5b5b6-72cc@192.168.16.170 for seqno 1 (Critical Response)
09:55.15phpboy:T
10:00.06phpboyI really love you guys
10:00.10phpboybut I need your help :<
10:01.36*** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
10:01.36*** mode/#asterisk [+o drumkilla] by ChanServ
10:01.41tronixI imagine * doesn't support 7960G's extension mobility
10:01.51tronixit's probably a CCM thing... and relies on SCCP specific features?
10:02.02areskiphpboy, peoples have never been so quite
10:02.41phpboyI know
10:02.46phpboythat's why I'm so heart sore :<
10:03.19tronixslackers in north america sleeping. how dare they! :-)
10:03.23*** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13)
10:03.30*** join/#asterisk secure75 (n=mic@host-62-245-230-34.customer.m-online.net)
10:03.47tronixfor some reason, I keep writing down 'twinkie' instead of 'twinkle' in my extensions.conf. freudian slip? :P
10:03.51phpboyIf I call to my pstn(via ISDN) I cometimes get an echo... how do I avoid this?
10:04.24tronixhmm the usual means is to adjust rx/tx gain, I understand. beyond that, not too familiar with echo issues.
10:06.20tronix(unrelated), looks like I found something on extension mobility at voip-info.org's wiki. cool.
10:07.16*** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk)
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10:20.56tronixhmm. where does one find NVFaxDetect?
10:21.37tronixn/m
10:21.39chapeaurougehmm.. why would i have no sound, if i have * setup in my internal network, with no firewall whatsoever?
10:21.43tronixI see the info on how to find it via wiki.
10:23.49*** join/#asterisk grey (n=grey@193.220.84.198)
10:23.55*** join/#asterisk M-I-A- (n=mai@wsp05975102wss.cr.net.cable.rogers.com)
10:23.55greyhi all
10:24.11greycan anyone give me some help with amp ?
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10:25.23*** part/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com)
10:25.31tronix~amp
10:25.37jbothmm... amp is NOT supported here! people using it should join #amportal
10:25.45tronix:)
10:25.53tronixbest place for it.
10:26.35tronixchapeauro: not sure. call establish ok? no messages in logs that looks unusual? debug messages enabled?
10:27.07tronixsic tcpdump/ethereal/etc and watch traffic to see if RTP established ok
10:27.37tronixgrey: no offense intended. I'd help if I knew anything about amp. sorry.
10:27.42chapeaurougetronix, calls establish ok. sip debug gives no real good info
10:28.01tronixhmm.
10:28.10chapeaurougemessages are being played, but i hear nothing.
10:28.44tronixsip softphone or hardphone used?
10:29.02chapeaurougespftphone
10:29.08*** join/#asterisk BSDaemon (i=hbf@CPE00032f0d286f-CM014380004179.cpe.net.cable.rogers.com)
10:29.10tronixhmm.
10:29.24chapeaurougei had specified an RTC port..
10:29.27*** join/#asterisk fulgas (n=fulgas@209.8.233.254)
10:29.27chapeaurougeim gonna try without it
10:29.47chapeaurougebleh. i can't :)
10:31.50chapeaurougeit works at home (differnt * install, but very similar)... odd.
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10:33.38chapeaurougebtw, for a 1.2.3 to 1.2.4 upgrade, nothing in particular to worry about?
10:33.51fourcheezeif I have N asterisks all looking at the same database (realtime) what else is required for me to get dynamic routing between them?
10:35.26fourcheezeis there some way to make * save the name of the host that each client is registered with to the database?
10:36.05Dabbaif $fred contains 12345 how do i make it contain null ?
10:36.15fourcheezeit seems a waste of a RDBMS otherwise
10:37.32X-Rob_Dabba,  - 'Set(fred=)'
10:38.09X-Rob_ANyone here got a GXP-2000?
10:39.12susinsome with astgui experience?
10:39.27fourcheezeany realtime experts around?
10:39.32fourcheezeany realtime hackers around?
10:39.38fourcheezeI'd like to know where it's going
10:39.54fourcheezethere doesn't seem to be much more work to have automatic clustering
10:40.10fourcheezeor is that available already with some module I don't know about?
10:40.54oejfourcheeze: It's not automatic, but if you have the proper fields in the database, Asterisk will save information for you
10:41.14fourcheezeoej: which fields do I need?
10:41.35oejchapeaurouge: No, there should not be anything to worry about more than upgrading your binaries
10:41.47oejfourcheeze: Can't remember of the top of my head. Wait.
10:42.09fourcheezeoej: because then I'm 90% of the way towards my goal
10:42.31fourcheezeoej: BTW my snoms still are not *subscribing* for less than 3600 seconds
10:42.38oejRead doc/README.extconfig to find the fields
10:42.44fourcheezeoej: ok thanks
10:43.13oejIn Addition, a field named "fullcontact" is used
10:43.27oejFor saving the Contact: header of the registered peer
10:43.41oejMost people use dundi in combination with regcontext/regexten
10:43.53*** join/#asterisk sevendeathlyvirt (n=shinux@196.207.6.13)
10:46.11fourcheezeoej: but fullcontact doesn't tell me where a client registered
10:46.13*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
10:46.23fourcheezeor don't I need to know that?
10:46.43*** join/#asterisk Vinsik (i=vinsik@84-240-73-181.dsl.maxinetti.fi)
10:46.50chapeaurougehmmm.... i can now hear, the 'bye' part of good-bye, but nothing else before... :\
10:47.25VinsikNeed a bit of help. How to dial out with Realtime peer added to MySQL?
10:48.53chapeaurougevoicemail isn't playing mp3 file, is it?
10:52.00chapeaurougeno. hmm.
10:52.26DabbaX-Rob thanks
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10:55.48mutilator:O
10:55.49mutilatorno +r?
10:57.13X-Rob_Oooh.
10:57.19X-Rob_This new GXP-2000 firmware is interesting
10:57.27X-Rob_OOh! Pretty!
10:57.29*** join/#asterisk RoyK (n=roy@80.239.107.70)
10:58.45X-Rob_Fuck me. I think they've fixed it.
10:59.10*** join/#asterisk cfh (n=luca@82.193.23.6)
10:59.10gaupeX-Rob_: what has been fixed?
10:59.13VinsikNeed a bit of help. How to dial out with Realtime peer added to MySQL?
10:59.36X-Rob_gaupe, well, the AGC that was causing handset feedback seems to be fixed.
10:59.39X-Rob_BLF's work
10:59.43X-Rob_haven't tested call pickup yet.
10:59.43RoyKVinsik: it's all in the docs :P
10:59.55VinsikRoyk: cant find it..
11:00.04gaupeRoyK: takk
11:00.07RoyK~realtime
11:00.09jbotrumour has it, realtime is http://www.voip-info.org/wiki-Asterisk+RealTime
11:00.41gaupeX-Rob_: I will try to load in on the only one I have, but it's still not a good phone :)
11:00.55X-Rob_For the money, they seem pretty reasonable.
11:01.08cfhwhat can i do to configure a sip trunk with a server asterisk and a cisco router?
11:01.33RoyKmethinks grandstream is quite reasonable, only perhaps a little lightweight, for dorstoppers
11:02.00gaupeX-Rob_: I like the thomson 2030 for that money
11:02.48VinsikRoyK: the problem is .. when i put Dial(SIP/${EXTEN}@user-out,20,r) <= this says no route to host. isnt it suppose to look for the user in database?
11:03.33RoyKhm...
11:03.45RoyKthe syntax is Dial(SIP/user)
11:03.53RoyKor Dial(SIP/user/${EXTEN})
11:04.02Vinsikbah
11:04.07RoyK:)
11:04.18VinsikRoyK: im calling through a SIP provider..
11:05.24*** join/#asterisk P0L0 (n=n0n3@140.Red-83-58-255.dynamicIP.rima-tde.net)
11:05.34*** join/#asterisk backblue (n=igor@82.102.1.42)
11:05.45backbluehi.
11:06.52RoyKVinsik: ithen use the latter
11:06.58RoyK<
11:06.58RoyK>
11:07.03*** join/#asterisk drumkill1 (n=russell@host-12-179-65-65.nctv.com)
11:07.15VinsikRoyK: nope
11:07.23Vinsiknot workin
11:07.25RoyKthen pastebin the output
11:07.34Vinsikw8
11:07.36phpboyI need to get rid of the echo on my line
11:07.37phpboypomple :<
11:08.04X-Rob_phpboy, you using -trunk?
11:08.43X-Rob_-trunk has _shithot_ echo cancellation now. Well. Shithot in comparison to what it used to be. Still woefully inadequate in comparison to a hardware EC.
11:08.45RoyKVinsik: why are you using realtime if you're just dialling through a provider? that's just one peer....
11:10.27VinsikRoyK: got.. thanx!!!
11:10.30VinsikRoyK: got it
11:10.46RoyKX-Rob_: what's so cool about it?
11:10.52RoyKX-Rob_: how's cpu load?
11:10.58X-Rob_RoyK, it works, is what's cool about it 8)
11:11.08RoyKdoesn't the one in 1.2 work?
11:11.28X-Rob_cpu load's not that bad. I don't have any machines that are loaded up enough for me to notice difference in cpu load.
11:11.51X-Rob_10 line PRI into a Via Eden 1Ghz machine with software ec == bugger all cpu utilisation
11:14.18*** join/#asterisk simondotsi (n=simon@mindtrip.entered.net)
11:14.58robin_szOK girls
11:15.21X-Rob_heh
11:15.25X-Rob_damn that tab expansion
11:15.28robin_szarse biscuits
11:15.43simondotsiHello, I'm having troubles with echo while doing MixMonitor on a call (between two Sirrix ports), anyone aware of this ?
11:15.44robin_szso where was I .. oh yes *
11:15.58robin_szdamn thing keeps quitting on me :(
11:16.14robin_szim running monit now on a 1 minute cycle to keep it alive
11:16.28robin_szbut tis quit twice in two weeks now ...
11:16.32robin_szthat aitn good :(
11:16.35X-Rob_robin_sz, there's a thing called 'safe_asterisk' that does that for you.
11:16.42robin_szahh ....
11:16.57robin_szno other changes required?
11:17.06robin_szjust run safe_asterisk??
11:21.25X-Rob_that restarts asterisk when it crashes, and saves the core in /tmp
11:21.27X-Rob_then read README.backtrace
11:22.02*** join/#asterisk Johan (n=kvirc@194.151.113.2)
11:22.04JohanHi all
11:22.14greywhat is the channel for AMP help ?
11:24.09X-Rob_grey, #amportal
11:24.14X-Rob_but you can ask me
11:25.13greythanks Rob
11:25.16susinsome installed astgui?
11:25.42greywhat is the best GUI for asterisk ?
11:26.10DarkFlibblegrey, depends what you are trying to do...
11:26.44JohanIs here someone with experience with mISDN? I am trying to dial-out (or recieve a call) but it doesn't work. It seems asterisk failes to get an index or so, but the output is pretty verbose.
11:27.01*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
11:27.01greysetup an asterisk gateway to provide cheap international calls
11:27.26DarkFlibblegrey, to the public?
11:27.54*** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au)
11:28.06X-Rob_Holy shit.
11:28.12X-Rob_The EC actually works!
11:28.16*** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net)
11:28.23DarkFlibbleX-Rob_?!?
11:28.26X-Rob_I have two phones, next to eachother, on hands free, and they're not feeding back!
11:28.34DarkFlibbleahhh
11:28.35X-Rob_(well, only a little bit, for 1/2 a second or so before it chops it)
11:28.46DarkFlibbleEC as in Echo Cancellation..
11:28.50greyyes
11:29.25mutilatorwish it worked that well for me
11:29.32mutilatori had to put my cisco box back into play
11:29.34DarkFlibblegrey, then I would write a custom one or hack someone elses code... since every business I've seen works differently...
11:29.43mutilatorbecause the echo on my new te405p i put in was so bad
11:36.35Money5ackhey ho
11:36.42X-Rob_mutilator, were you using -trunk?
11:37.26mutilator1,2,1
11:37.28X-Rob_(and, hint, echo problems -> http://bugs.digium.com/view.php?id=5520 )
11:37.30mutilatorand trunk
11:37.57*** join/#asterisk zotz (n=zotz@24.231.47.175)
11:39.27mutilatori hope nothing in trunk is broken right now
11:39.42X-Rob_trunk seems pretty good.
11:40.49mutilatornah
11:41.03mutilatorthat was already posted to trunk
11:41.09mutilatorbefore i had my echo problems
11:41.26*** join/#asterisk sevendeathlyvirt (n=shinux@196.207.4.220)
11:41.27mutilatorthis was like last...
11:41.37mutilatorwed morning
11:41.44mutilator25th in the early AM
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11:44.02*** join/#asterisk leopardus (n=leopardu@217.22.179.69)
11:44.33moreecehelp!?@! - does anyone have an idea how to disable my Call rating engine and routing -> rate_engine.conf?
11:44.43moreecemy asterisk is crashing cause their is no database
11:45.18clive-moreece howzit
11:45.33moreecehowzit clive
11:45.33moreeceany ideas???
11:46.22clive-I am not familiar with that software working with * but why dont you try changing your dialplan so it doesnt use this cal rating stuff
11:47.08moreecehmmm, not sure I've configured my sip.conf and extensions.conf and I dont see it within there however this was preconfigured ---> let me check quick
11:47.28DarkFlibblemoreece, what rating engine are you using?
11:47.35DarkFlibbleI know of two...
11:47.42moreece*checking*
11:47.46*** join/#asterisk burton (i=mimx@w201.ljudmila.org)
11:49.04moreece; Call Rating Engine Configuration File
11:49.04moreece;
11:49.04moreece; Copyright (C) 2003 by Troll Phone Networks AS
11:49.04moreece;
11:49.04moreece; This program is distributed under the terms of the GNU General Public License
11:49.05moreece; as published by the Free Software Foundation; either version 2, or (at your
11:49.08moreece; option) any later version.
11:49.18moreece;
11:49.18moreece;       $Id: rate_engine.conf.sample,v 1.5 2003/11/28 16:20:46 tholo Exp $
11:49.21moreecesorry for the paste, should have used pastebin
11:49.24DarkFlibbleahh... the troll phone one..
11:49.30DarkFlibblenope, never used it..
11:49.58DarkFlibblehttp://www.voip-info.org/wiki/view/Asterisk+addon+rate-engine <-- might help tho
11:51.13moreeceta
11:51.18moreece*checking*
11:51.46DarkFlibblehttp://www.trollphone.org/files/ <-- might be useful..
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11:57.40buzzydDoes anyone know how I can play a message when I get  Got SIP response 404 "Not Found" message ?
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12:02.51moreeceall righty, simply remote the entire rate_engine package from the systems starts up without it
12:02.58moreece*removed*
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12:16.15Johanwhat is the difference between ptp and ptm? When I have an ISDN-card and want to dial to a phone number, then I am talking about ptp, is that right?
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12:20.22Tribastianhello all
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12:28.52snoopshey - I've just purchased a linksys SPA-3000 Sipura, and was wondering if you guys could point me in the right direction (I grabbed the O'Reilly Asterisk - The Future of Telephony book, since it was published under creative commons license, which I am currently reading through) of some resource links/information I should know, etc
12:29.29*** join/#asterisk beside (n=beside@tigbis.lt)
12:29.38DarkFlibblea linksys sipura?
12:29.53besidei would like to ask if someone here worked on passive monitoring using hfc-e1 board?
12:29.56DarkFlibblevoip-info.org
12:30.21bigjbdoes anyone know of a guide to intergrating asterisk with a h.232 gatekeeper?
12:30.36DarkFlibblevoip-info.org
12:30.44bigjb=oP
12:30.50DarkFlibbleits h.323
12:30.58Skumlingwhat are people doing this "monitoring" things for? for logging purposes or for debugging?
12:31.11besideSkumling: logging
12:31.12DarkFlibbleSkumling, recording all calls normally...
12:31.17gaupeDarkFlibble: well it probably says cisco linksys on the box and sipura on the software
12:31.22DarkFlibbleso they can laugh at them later...
12:31.35besidein the case I do, I have E1 stream copy and give only RX to hfc-e1 board
12:31.36snoopsThat's one of the things I'll be doing - recording calls when I call power company etc(after telling them of course)
12:31.44Skumlingbeside, DarkFlibble: hum okay... not that nice :-/
12:31.52besideI need a way to read D channel HDLC packets and open and read B channel
12:32.18Tribastianhello! i do have a slight problem. i do have firehol running and asterisk. i can phone out, i can phone internal, but nobody can tel in. my sip.conf is to find at http://www.codepaste.net/301, my extensions.conf is at http://www.codepaste.net/302 and my firehol.conf is at http://www.codepaste.net/300. please help or my boss will kill me!!!
12:33.08besideas I load multihfc module with debugging
12:33.15besideI can see hdlc packets in kern.log
12:33.34besidebut how can I take them using for example C
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12:38.27Wiizhi
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12:47.53Dr-Linuxhi
12:48.00Dr-Linuxexten => _81NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
12:48.33Dr-Linuxi don't want my users to dial _81 << this "1"
12:48.43Dr-Linuxwhat do i need to change in this line?
12:48.45Dr-Linuxexten => _81NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
12:51.15Tribastianhello! i do have a slight problem. i do have firehol running and asterisk. i can phone out, i can phone internal, but nobody can tel in. my sip.conf is to find at http://www.codepaste.net/301, my extensions.conf is at http://www.codepaste.net/302 and my firehol.conf is at http://www.codepaste.net/300. please help or my boss will kill me!!!
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12:52.27WiizDr-Linux
12:52.30Wiizwhat do u awnt them to dial
12:52.38Dr-LinuxWiiz: yes sir
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12:53.09Dr-Linuxi want them to dial _8NXXXXXX  the other 1 is needed by telco
12:53.35Wiizso delete the 1
12:53.42Wiizadd an X
12:54.33Dr-LinuxWiiz: no, but user will still need to dial 816504500
12:54.44Dr-Linuxi don't want to dial 1
12:56.08Errso *asterisk* has to dial the 1, right, and not the user?
12:56.22Dr-LinuxErr: yessssssssss you are right
12:56.25Dr-Linuxthats what i want
12:56.35Dr-Linuxbut sorry i can't explain with my bad english
12:56.40Errso you have asterisk dial 81${EXTEN:1}
12:56.59Err(which dials 8, then 1, then everything that the user dialed except the first number (the 8)
12:56.59Dr-LinuxErr: and what would be the petren ?
12:57.26ErrDr-Linux: hm?  what's a petren?
12:57.39Dr-Linuxexten => _81NXXXXXX,1,Dial(Zap/g1/${EXTEN:1})
12:58.00tbsSkumling: *You* think recording calls is mean? ;-)
12:58.01Errexten => _8Nxxxxxx,1,Dial(Zap/g1/81${EXTEN:1})
12:58.51DarkFlibbleSkumling, in the UK a private individual can record a call as long as one party is aware of it...
12:59.04DarkFlibble.. at least according to my last emplyer...
12:59.06Dr-LinuxErr: so now user will not dial 8 and 1 ?
12:59.09tbsDarkFlibble: It's the same in .dk
12:59.29ErrDr-Linux: yes, with that line, users would dial 8<number>, and asterisk would dial 81<number>
12:59.32tbsIt's a weird law...
12:59.56Dr-LinuxErr: okey great . let me try this
13:00.14Errheh, in the US the *federal* law requires that one of the parties know - but many states have laws that require both parties to know
13:00.15dpryohehe, I hate that my cellphone sends *BEEP* every minute when recording calls
13:00.22viperdudeDarklibble: thats how I understand it but have you noticed how many calls centers now say calls may be recorded for training purposes.... i am guessing they are fearing litigation
13:00.26DarkFlibbleit sometimes comes in useful, since most big companies record all calls... and you can prove that one of their representitives said x but did y
13:00.49Errcommon courtesy suggests that you should tell people that you're recording their calls
13:00.52DarkFlibblecall centres aren't individuals...
13:01.05Err(not that common courtesy has anything to do with telemarketing ;-)
13:01.21fugitivoviperdude: call centers use recordings for training, that's true
13:01.32cron"your message might be recorded for training purposes"
13:01.52cronooh .4
13:01.54dpryo«all you say will be held against you»
13:02.04DarkFlibble"your call maybe recorded because we are paranoid"
13:02.09cronwell
13:02.28crondifferent laws require user information to be recorded and held for ammounts of time
13:02.34viperdudei think they say training purposes as it sounds less big brother
13:02.45dpryoof course :)
13:02.47Tribastianor 1982
13:02.53Err84 even ;-)
13:02.59viperdudelol
13:03.00cronexample, I wonder if PIPEDA would go into play for social engineerign
13:03.00besideDarkFlibble: in my case I'm not talking about individuals or companies, but about Law Enforcement Agency
13:03.30DarkFlibbleLaw enforcement hold all calls in the uk
13:03.47DarkFlibblesince they may need them to "safeguard the life of the caller"
13:03.54cronman thats sad :(
13:04.03viperdudethe BBC was talking about the NSA and the wire tap scandel in the US, last night. Does anyone reckon VoIP taps are in common use by the law enforcement yet?
13:04.37DarkFlibbleI recon voip taps are used...but nowhere near to the extent normal landline taps are
13:04.40croni wouldn't doubt it
13:04.41ErrI'm certain that VoIP calls are monitored in and out of the US; internally, maybe so and maybe not
13:04.44warmcatUse a VPN if you're worried
13:04.46Dr-LinuxErr: 8 is also going out, i wanna strip it
13:04.54Dr-LinuxExecuting Dial("SIP/4092-26e9", "Zap/g1/815935400") in new stack
13:04.57besidein my country no, but in some yes, VoIP are monitored
13:04.57DarkFlibblesince a lot of voip is untappable without major problems
13:04.57viperdudedoes a VPN worry the NSA?
13:04.57tbsDarkFlibble: A few months ago a Danish consumer tv-programme described a case with a person, who had been called by a tele marketing company, who wanted to subscribe him to a new ADSL-line. He agreed to have some more info sent to his home. A few days later, he received an order confirmation. When he called them, they claimed to hold a recordring of him agreeing to subscribe.
13:05.05ErrDr-Linux: oh, so you want outbound to dial only 1<number>?  that's not what I thought you said
13:05.14warmcatThe question is how much budget you attract
13:05.16Errexten => _8Nxxxxxx,1,Dial(Zap/g1/1${EXTEN:1})
13:05.33tbsDarkFlibble: however, they wouldn't let him hear the recorded conversation, because it "contained sensitive information"
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13:05.53Dr-LinuxErr: 8 is my patren
13:05.56warmcatThere's only so much cracking capability around, it will be focussed on the Really Bad Guys
13:05.59fugitivotbs: lol
13:06.06DarkFlibbletbs, surely that would be covered in the data protection act in the uk, and across the EU
13:06.11tbsDarkFlibble: They couldn't quite explain what the point of the recording was then :D
13:06.21warmcatif you are not a bad guy at all but interested in privacy there's no reason to burn that limited resource on your VPN
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13:07.01ErrDr-Linux: so, give me an example of what you want to dial internally, and what needs to be dialed externally; the last line I wrote will take an internally-dialed 8<number> and dial, outbound, 1<number> - if that's not what you want, what *do* you want?
13:07.08Tribastianhello! i do have a slight problem. i do have firehol running and asterisk. i can phone out, i can phone internal, but nobody can tel in. my sip.conf is to find at http://www.codepaste.net/301, my extensions.conf is at http://www.codepaste.net/302 and my firehol.conf is at http://www.codepaste.net/300. please help or my boss will kill me!!!
13:07.14Dr-LinuxErr: this is the way 1(number) i wanna strip 8
13:07.43Dr-Linuxi don't want 8 to send out, user will dial 8 but asterisk will not send it out
13:07.57Errthat last line I wrote should work, then
13:08.08Err(a concrete example with numbers would help out a bunch, here)
13:08.31Erryou're using too many pronouns for what you're trying to describe :-)
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13:08.51Dr-LinuxErr: sorry for my bad english
13:09.16ErrDr-Linux: it's not a problem - just give me an example of a number dialed internally, and what you want asterisk to actually dial on the outbound line
13:10.23Dr-Linuxwell, i want like this 85935400   << 8 is a patren and 5935400 is a number
13:10.52Dr-LinuxErr: but the damn telco says dial 1 before the number
13:11.26Errok, so "exten => _8Nxxxxxx,1,Dial(Zap/g1/1${EXTEN:1})" should work, I think
13:11.28Dr-Linuxso it will be like this 8+1+5935400
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13:13.21Dr-LinuxErr: yeah, thanks it works, i tried same before, but i was missing an X in patren
13:13.51Dr-LinuxErr: i wanna ask you a another question
13:14.32Dr-LinuxErr: one of my x-lite client has low bandwidth, so when he calls me on my x-lite extension, he hears me good, but i can't hear him fine
13:14.50Dr-Linuxbut when i call him on his xlite extension, we hear fine each other
13:14.59Dr-Linuxwhat could be happen? :S
13:15.23DarkFlibbledifferent negotiation depending on the call setup?
13:15.46Erryeah, that'd be my guess - you probably need to force a codec for his extension
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13:16.51DarkFlibblejust dropped my lappy...
13:17.10Dr-Linuxhhm..
13:17.17Dr-LinuxErr: which one?
13:17.43DarkFlibblelanded open on the screen and base like an A... seems alright tho..
13:18.00DarkFlibbleDr-Linux, what codecs does x-lite support?
13:19.04Dr-Linuxulaw, alaw, iLBC and GSM
13:19.16DarkFlibblehmmm... supposedly speex and ilbc need a reg hack to work with asterisk in xlite
13:19.35DarkFlibblegsm would be the codec I would force for low bandwidth
13:19.47Dr-LinuxDarkFlibble: i don't think ilbc needs registration
13:19.55Dr-Linuxyeah, i know
13:19.58DarkFlibbleworks out about 20kbits/sec per channel
13:20.12DarkFlibblereg hack == registry hack
13:20.18DarkFlibblein windows...
13:20.21Dr-Linuxbut my codecs sequence is something like this >> iLBC >> GSM >> ulah >> alaw
13:20.27DarkFlibblehttp://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite
13:21.02Dr-Linuxi also have g729, but that doesn't work with x-lite
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13:34.18jaikewow..new release?
13:34.20bigjbwhats the easiest way to add a few hundredths of a second pause into the dialplan before it goes into playing an automated messasge?
13:35.52mutilatormake a perl script and exec it via agi to pause for a few hundredths
13:36.02mutilatoror just pause for 1 second with Wait()
13:36.45tronixthe mere act of exec'ing might very well provide the delay. :-)
13:37.10bigjbheh
13:37.24mutilatoryea
13:37.41mutilatordunno about a few hundredths tho
13:37.43tronixcan probably count on at least 10ms (1/100th sec) delay alone
13:37.43mutilatorit's not that slow
13:37.49tronixdue to disk i/o
13:38.10cpmUmm, can you stack Wait()Wait() ?
13:38.30tronixit's just that I don't think you can get that granular with Wait()
13:38.50mutilatoronly deals in seconds as far as i know
13:39.10mutilatorbigjb: why ya need the pause?
13:39.20tronixcould be for callerid or other stuff
13:39.59Money5ackanybody here who gets t.38 running with latest svn ?
13:40.09mutilatorbut for what.. usually theres always something that executes before you'de need to use callerid
13:41.27bigjbwhen it goes into auto attendant and asks "please dial the extension of the person" it just seems to imidieate
13:41.30bigjbimediate
13:41.35bigjbimmediate
13:41.40bigjbbollocks
13:41.41*** join/#asterisk fugitivo (n=ajf@201.255.176.83)
13:41.44bigjbyou know what i mean
13:41.50Tribastianhello! i do have a slight problem. i do have firehol running and asterisk. i can phone out, i can phone internal, but nobody can tel in. my sip.conf is to find at http://www.codepaste.net/301, my extensions.conf is at http://www.codepaste.net/302 and my firehol.conf is at http://www.codepaste.net/300. please help or my boss will kill me!!!
13:41.57bigjbits probably a lot to do with the fact that im using the pc without a headset
13:44.35bigjbWait doesnt seem to affect it anywho =oS
13:44.44bigjbwoman is just there straight away
13:44.45ErrTribastian: your firewall is blocking inbound calls
13:45.15ErrTribastian: your "firehol" rules are far too simple - you'll need to allow the UDP block that asterisk uses for incoming SIP data streams
13:45.34fugitivowhat is "firehol" ?
13:45.41bigjbill have a gander at sounds directory when back from customers, pretty sure that there is a silence in there somewhere
13:45.48Errit's some firewalling frontend, according to google
13:46.01fugitivoso, it's just iptables?
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13:46.13saftsack_Sam--, hi
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13:46.58Errit's a front-end to iptables, I guess
13:47.08ErrYet Antoher Program to make firewalls "easy"
13:47.58besideit's sounds like Firewall Hole :)
13:48.24*** join/#asterisk Paulo (n=paulos@200-168-112-132.dsl.telesp.net.br)
13:48.38Drew___not only do you need to allow the udp block - but if its a NAT you need portforwarding
13:48.52TribastianErr how would the line in the firehol would look like? btw thanks for answering
13:49.02ErrTribastian: I have no idea - I don't know anything about firehol
13:49.02iCEBrkryay 1.2.4
13:49.12ErrDrew___: it doesn't look like a NAT
13:49.18Errof course, I don't know anything about firehol
13:49.32TribastianErr: thanks anyway, at least somebody did talk to me...
13:49.44iCEBrkrlol
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13:49.54Drew___1.2.4???? holy shit - more updates... ;)
13:49.56iCEBrkrWTF is Firehol?
13:50.04Stephniehi
13:50.05iCEBrkrDrew___: Don't blink man!
13:50.11ErrTribastian: I've seen your question three or four times, but I don't really know what the answer is - which I'm sure is why nobody else answered either
13:50.17Err"I don't know" isn't a very useful answer
13:50.24Drew___hi Stephnie
13:50.33StephnieIf outbound call is connected/Answered ...Can I play a FILE (.wav) ?????
13:50.33_Paulo_<PROTECTED>
13:50.43coppiceFirehol == a firewall that only protects you for 15 minutes
13:50.44iCEBrkrStephnie: Sure
13:50.54iCEBrkrcoppice: 15mins?! Woah! Crazy stuff!
13:50.55Stephniewait wait
13:50.56Tribastianthanks again, so i see why nobody did answer...
13:51.20iCEBrkrTribastian: What's wrong with iptables? Why not use it instead?
13:51.46Flyboy-SR22Tribastian, have you tried dropping your firehol to make sure the system is operating properly without the firewall in place..?
13:52.02StephnieiCEBrkr:  actually I need to first check that I need to play a file or not....I mean not every time I want to play a file
13:52.07Flyboy-SR22I use iptables on all of my * servers, works great
13:52.09Errwell, it looks like it's also operating as the NAT box itself, so I'm not sure that he can
13:52.15Flyboy-SR22ah
13:52.17TribastianICEbrKr: well i first suggested that but my team said that it is stupid what i am doing and i shall use the firehol because it should be much easier
13:52.21iCEBrkrStephnie: Ok? You can check for conditions
13:52.31*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197)
13:52.31Stephniewhat should I check ?
13:52.36Flyboy-SR22no way to move the * box to a public IP protectec by iptables..?
13:52.36StephnieDial application?
13:52.42iCEBrkrTribastian: You'd get more help/support using iptables.. It's pretty standard.
13:52.42Errtell firehol not to block any inbound connections (forward them all to the machine), and see if it works
13:52.51iCEBrkrStephnie: I dunno, what are you checking before you play the file?
13:53.49iCEBrkrTribastian: First of all, if your team is telling you what you're doing is stupid-- Don't take advice from them :)
13:53.58Tribastiani will try, but even as i said in my conf that everything should not be blocked it still did not work, sadly we do need a masquarating wich is in the firehol...
13:54.01Stephnieif there is a voice mail...then I want to play a file...otherwise NOT...
13:54.18iCEBrkrStephnie: Simple
13:54.29iCEBrkrStephnie: Start here.. http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands
13:54.52iCEBrkrStephnie: HasNewVoicemail()
13:55.39Flyboy-SR22Tribastian: Have you verified that the traffic is actually making it through firehol..? Do you have logging enabled on the firewall..?
13:56.21StephnieiCEBrkr : it's not an incoming call.....It's an outbound call
13:56.32StephnieiCEBrkr : for example
13:56.41iCEBrkrStephnie: OK, I think you need to describe what your end result is going to be..
13:56.59Tribastianit is enabled, i will have to check again, lot of changes have been taking place over the last days, so did not have time to check...
13:57.01iCEBrkrStephnie: A lot of people have obfuscated ideas how things work and go about doing it the wrong way.
13:57.01StephnieiCEBrkr : I dial your number.....if you pick it up then I'll talk to you...otherwise I will leave a message through a .WAV file...
13:57.22Stephnie:)
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13:57.54iCEBrkrStephnie: So you're trying to make it call someone when they have a voicemail?
13:58.16Flyboy-SR22Tribastian: Here is what I would do..check your logs on the firewall and fire off asterisk -rvvvvvvv and watch to see if you see anything coming in at all while you are attempting the connection. If the logs on the firewall are no help, maybe you will see the traffic hit the * box..
13:58.30StephnieI want to play .WAV manually when there is a voice mail.....
13:58.33fugitivois xlite free for a company?
13:58.42fugitivofor internal use
13:58.55StephnieI want to play .WAV manually when there is a voice mail at dialed number
13:59.06Flyboy-SR22Tribastian: You could also throw iptables on the * box and allow all but log it to get a better picture of what you were seeing. TCPDUMP or Ethereal may also help you pinpoint where the traffic is blocked, but my first suggestion would be the firewall logging
13:59.11iCEBrkrStephnie: What's the wav file gonna say?
13:59.18Tribastianbut we could one time as we turned of the firehol in a very early status (before everything was connected with our callcenter) call in, so it must be the firewall, but i will do as you said an come back here again, thanks...
13:59.41StephnieiCEBrkr : "LALALALAL ....Laaaaa...Lets make things better"
13:59.42Flyboy-SR22Tribastian: NP
14:00.16StephnieiCEBrkr : :-)
14:01.06iCEBrkrStephnie: You're still not being descriptive enough for me to help you do exactly what you want to do
14:01.31StephnieiCEBrkr : sorry...ok now let me explain...
14:01.34*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
14:02.53StephnieiCEBrkr : for example: I dial your number ...if you are available then ofcourse We'll talk....but if you are unavailable then I want to PLAY a WAV file at your voice mail...
14:03.29StephnieiCEBrkr : that's the as simple as I can explain.....
14:03.41iCEBrkrStephnie: So why not use VoiceMail()?
14:05.15Stephnieis that what I need?
14:05.26iCEBrkrStephnie: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail
14:06.28Stephnieyep reading ..
14:08.14Stephniethat is for incoming voice mails
14:08.28iCEBrkrThat's for when someone is on the phone or doesn't answer
14:08.36*** join/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p224.telkom-ipnet.co.za)
14:08.49Stephniewhen someone calls me ....or at my number...right?
14:09.12iCEBrkrStephnie: If someone calls you and your line is busy or you don't answer, you have Asterisk jump to VoiceMail()
14:09.28iCEBrkrIt'll prompt the caller to leave a message depending on your extensions status
14:10.02Stephnieok ..what about if I call friend and want to play MP3 at his/her voice mail???
14:10.06iCEBrkrDamnit!!!! How do you tell Meetup.com to stop sending you daily updates?!?!
14:10.13Stephniemy friend*
14:10.16iCEBrkrStephnie: Huh?
14:10.28Supercrosshello everyone
14:10.37Stephniewhat about if I call my friend and want to play MP3/WAV at her voice mail???
14:10.41Stephniegot it now???
14:10.54iCEBrkrSo you call someone, and then what?
14:11.03iCEBrkrYou want to stream in mp3's while you're on the call?
14:11.14StephnieYES!!!! MP3 or WAV...
14:11.36I-MODwav, not that hard....mp3, harder
14:11.44Stephnieok I got for WAV
14:11.45iCEBrkrThere's a way to do it, but I'm not sure how.. You have to do it from outside of Asterisk I believe.
14:11.47Stephniego*
14:11.57*** mode/#asterisk [+o drumkilla] by ChanServ
14:12.34StephnieI-MOD have something to say ...
14:12.37Stephnie:)
14:13.28*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:13.30StephnieI-MOD ??
14:13.52I-MOD1 sec, reading backlog
14:14.32Stephnieok..
14:15.11mutilatorhas anyone ever run into problems with a
14:15.15mutilatorZoom X5 GS Ver 2.0.1-00 ADSL modem
14:15.19mutilatorand sip?
14:15.25mutilatori keep getting sipura 2002 ata's
14:15.36I-MODiCEBrkr: what Stephnie wants is for asterisk to detect whether or not she has been sent to voicemail on someone else's box and if she has hit voicemail, play a pre-recorded message into it
14:15.38iCEBrkrI wish I had that problem.
14:15.39mutilatorand when i plug them into the zoom modem it locks up the modem
14:15.52mutilatorsoon as i unplug the ata it works perfect
14:16.08mutilatorthis has happened with 6 different atas and different modems
14:16.18iCEBrkrI-MOD: Sounds like Telemarketing spam to me.
14:16.31I-MOD:)
14:16.46Flyboy-SR22mutilator: what is the problem..?
14:16.48*** join/#asterisk RevK (n=RevK@flawless.1ec.aaisp.net.uk)
14:16.58mutilatorthe dsl modems lockup with i plug in ata's
14:16.59iCEBrkrmutilator: umm, does the modem have RJ45 ports for extra network devices or something???
14:17.08mutilatoryea it's has 4 ports
14:17.09StephnieI-MOD: no
14:17.10fugitivoI-MOD: app_amd (answering machine detection)
14:17.12RevKI have a dumn question on asterisk's handling of jitter...
14:17.14Flyboy-SR22mutilator: I had to replace my sisters Zoom modem several time due to problems with the hardware, but it only effected * when it ws down !!
14:17.33iCEBrkrfugitivo: You realize there's no traces of app_amd on the wiki or anywhere else, right?
14:17.34fugitivoI-MOD: it's not 100% accurate
14:17.40fugitivoiCEBrkr: right
14:17.44StephnieI-MOD: example:   I call my friend and want to play MP3/WAV at her voice mail???
14:17.53fugitivoiCEBrkr: do you want the module? ;)
14:18.01*** join/#asterisk Utah_Dave (n=boucha@0-1pool138-169.nas28.salt-lake-city1.ut.us.da.qwest.net)
14:18.01iCEBrkrfugitivo: :P Duh!
14:18.06Flyboy-SR22mutilator: Wow - haven't seen that problem...its the only device you are plugging into the modem..? have you tried a switch instead..?
14:18.26mutilatorthere have been 1 & 2 pc's also plugged in
14:18.33mutilatorbut the problem was resolved down to plugging in the ata
14:18.45mutilator6 different times now
14:18.46*** join/#asterisk maggit (n=maggit@customer-200-36-59-130.uninet.net.mx)
14:18.47mutilatordifferent locations
14:18.56mutilatordifferent ata's and different modems
14:18.58iCEBrkrmutilator: You got a hub?  Plug the ATA's into a hub instead of into the modem
14:19.03fugitivoiCEBrkr: where do i send it?
14:19.10iCEBrkrfugitivo: icebrkr@cyberdyne.org
14:19.12Flyboy-SR22mutilator - so basically you plus the ata into a switch on your network and it kills the modem..?
14:19.12mutilatori'll try it at the next customer with that problem
14:19.13iCEBrkrthat'll work
14:19.56RevKWHilst asterisk may cope with jitter, does it cope well with packet reordering at all?
14:20.40mutilatori was thinking maybe a firmware problem with the modem
14:20.48mutilatorbecause i ran into it on one customer
14:20.56mutilatorwehre they tried the new msn voice chat, and that uses SIP
14:21.01mutilatorand it locked their dsl modem
14:21.11StephnieI-MOD: still reading log?
14:21.11*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
14:21.12*** join/#asterisk Seedy (n=Seedy@65.200.153.2)
14:21.37*** join/#asterisk gr0mit (n=w10277@206.41.25.138)
14:21.51RevKWe are seeing signs that iax does not cope with packet reordering... I am wondering if this is expected...
14:22.49*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
14:24.47*** join/#asterisk danzig (n=chatzill@ruc-kj-013.ruc.dk)
14:25.09danzigEHLO all!
14:25.37danzigAnyone know about any progress with a skype channel for *?
14:26.40*** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net)
14:26.50cucodanzig, this will not happen. skype does not like that.
14:27.16*** join/#asterisk zotz (n=zotz@24.231.47.175)
14:27.28fourcheezedanzig: how could that work?
14:27.33iCEBrkrcuco: You'd almost think they'd want some sort of inter-connect...
14:27.42danzigYes, they just published their specifications! (for people to make hardware phones, but if one can make a HW, one can also make a SW)
14:27.48fourcheezemore likely skype have to join the sip world eventually
14:27.51Dr-Linuxany open source voice recognation program work with asterisk?
14:28.05danzigdr>>Festival
14:28.06bweschkeDr-Linux: sphinx would be your best bet there
14:28.06iCEBrkrDr-Linux: Sphinx  Supposedly..
14:28.06fourcheezedanzig: url?
14:28.17iCEBrkrdanzig: Festival isn't voice recognition
14:28.21*** join/#asterisk svenna_ (n=svenna@p548D23B9.dip0.t-ipconnect.de)
14:28.23ErrRevK: define "does not cope" - does it drop connections, or simply ignore the late packets?
14:28.26danzigsorry
14:28.29synthetiqany asterisk agi people here who use perl?
14:28.34iCEBrkrFestival is text to speech
14:28.37cucodanzig, you cannot connect skype to another vpio network, as far as i understood, this is part of the eula
14:29.01RevKErr, it asks as if the packets were dropped, i.e. audio break up.
14:29.10ErrI would expect that
14:29.19danzigyup, I got mixed up with text to speech. Will just go find skype URL
14:29.34Err...unless the late packets are still in time to be played, which is doubtful if they're behind later packets :-)
14:29.39RevKWe have 40ms packets, and around 60ms random jitter independantly on two channels, so it is possible for packets to arrive one out of order
14:29.49Dr-LinuxSphinx is an open source? and good to use with asterisk?
14:30.02Errdo you have a jitter buffer sufficient to allow delayed packets to still be in time to be played?
14:30.18RevKWhat I wanted to confirm is that an out of order packet would be held to be played if the jitter buffer is at a sensible level, and the later out of order packet correct slotted in if in time.
14:30.26clive-use the jitterbuffer
14:30.32RevKAlso, if the late arrival of an out of order packet caused the jitter buffer time to push up to allow for more.
14:30.34bweschkeDr-Linux: yes - it's open source - is it good to use with Asterisk, well it's probably the best you can find w/OpenSource tools.
14:30.50RevKThe jitter buffer settings are set up in iax.conf, anything else I need to know about setting up jitterbuffers?
14:30.58ErrI wouldn't expect the jitter buffer to auto-tune by default
14:31.18*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
14:31.38RevKYes, but tuning for variable jitter without reordering is not quite the same as allowing for packet reordering, and I am wondering if the code copes with that. I guess I can go and read the code.
14:32.12iCEBrkrI really wish I could build a 'TellMe' type system wit Sphinx. :P
14:32.28Errno, packet reordering should be the same, really
14:32.31Dr-Linuxbweschke: Thanks, could you please suggest Sphinx link or any user manual for this?
14:32.37RevKI would hope so...
14:32.41Errjitter is jitter
14:32.56RevKPractical testing suggests not
14:33.16Errsufficient jitter must cause reordering, unless your delay monotonically increases forever :-)
14:33.17iCEBrkrDr-Linux: Getting Sphinx to integrate into Asterisk == PITA
14:33.19SibRw0rkdoes the topic mean that the new version of asterisk is a memory leak, or fixes a memory leak?
14:33.34RevKErr, jitter is jitter. I can see how one could code for jitter without reordering quite easily and break when there is reordering.
14:33.35Err(jitter or drops, one, of course)
14:33.45Dr-LinuxiCEBrkr: PITA ? :S
14:33.59RevKBut reorder looks like a drop until the out of order packet arrives...
14:34.00iCEBrkrDr-Linux: Pain in the ass
14:34.09RevKI'll have to read the code won't I... oh well.
14:34.11Dr-LinuxOpsss
14:34.17RevKffff
14:34.24iCEBrkrSibRw0rk: I read that as having fixed a memory leak
14:34.25bweschkeDr-Linux: ya - it's not a real user friendly process - but there are folks that have gotten it working
14:34.52iCEBrkrGetting Sphinx to work alone was a chore
14:34.55Dr-Linuxhhm..
14:35.19Dr-Linuxyou mean, using voice recognition system is not a good approach with asterisk yet?
14:35.36*** join/#asterisk saftsack (n=lottc@p54A7EC24.dip.t-dialin.net)
14:35.47iCEBrkrDr-Linux: There's just no real good voice recognition software for linux
14:35.48bweschkeDr-Linux: you can do it - it's just going to take alot more work than, say, configuring asterisk by itself does now.
14:35.52iCEBrkrDr-Linux: that's free, of course
14:35.56saftsack_Sam--: hi
14:36.04bweschkethere is a company working on a low cost solution for it
14:36.19bweschkewithin a couple months I think they plan to have something available
14:36.59DarkFlibbleits not really that complex...
14:37.15saftsackis it importan to upgrade to asterisk 1.2.4?
14:37.26DarkFlibbleyou just need to use Fourier Tranforms and patten matching...
14:37.32fugitivoiCEBrkr: yes, loquendo is good
14:37.37RevKLooks like code tries to handle out of order - I'll read more
14:37.45fugitivocommercial software, obviously
14:37.46DarkFlibblewhich a limited set of patterns to match it should be pretty acurate...
14:38.09DarkFlibblefreeform speech recognition tho... thats a different matter
14:38.18iCEBrkrfugitivo: Is that what TellMe uses?
14:38.39fugitivoiCEBrkr: i don't know, www.loquendo.com, they have asr, tts, and a lot of voice software
14:38.51iCEBrkrOh, I think I've been there
14:39.07fugitivoi saw it working, it's wonderful
14:39.22*** join/#asterisk ivanfm (n=ivanfm@201-1-164-43.dsl.telesp.net.br)
14:39.59Money5ackguys ?
14:40.21Money5acki've got some curios compiling failures in chan_sip when i try to compile t38 support in
14:41.15Money5ackis here somebody who can help me a little bit ?
14:41.28clive-money I never knew t38 worked with asterisk yet
14:41.35SibRw0rkwhen i call out - i get a shit load of time out warnings then get a 603 declied
14:41.57danzigI dunno... Maybe I got hopefiull too fast - slashdot har "MSNBC has a look at some of the interesting gadgets that will be available for purchase now that Skype has published instructions on how to build the service into phones", but I cannnot offhand find the actual published specifications...
14:42.07SibRw0rkhttp://pastebin.com/532134
14:43.16Dr-Linuxhhm..
14:43.45Dr-Linuxhttp://turnkey-solution.com/asterisk-sphinx.html << here is a guide for Sphinx
14:43.57*** join/#asterisk _-_ (n=nabudoco@206.135.48.98)
14:44.01Money5ackclive: on the bugtracker list are some people who got t38 support into asterisk and its running, i had running it to for a while without know it and without any changes in sourcecode...
14:44.19iCEBrkrDr-Linux: Yea, good luck with that :P
14:44.33Dr-LinuxiCEBrkr: could you tell me please what makes it bad? PITA ?
14:44.35SibRw0rkiCEBrkr: http://pastebin.com/532134   - help?
14:44.58sivanais there a list of conf files that can safely be deleted?
14:45.03Money5acknow i had to reinstall that machine and there is no t38 support compiled in so i downloaded the t38_bits work from steveu and patched all files..
14:45.14Money5acknow i have some big failures in chan_sip
14:45.20*** join/#asterisk junbug (n=junya@c-66-176-211-109.hsd1.fl.comcast.net)
14:45.20iCEBrkrDr-Linux: Getting Asterisk to talk to Sphinx in general.. It uses AGI() and file descriptor 3?  Which is proprietary to Asterisk
14:45.54Dr-Linuxyeah, it uses AGI()
14:46.04iCEBrkrDr-Linux: I've gotten Sphine to recognize simple words.. UP, DOWN, LEFT, 1, 2, 3, YES, NO type stuff.
14:46.17iCEBrkrDr-Linux: But I'll be damned if I could get it to control Asterisk
14:46.48iCEBrkrSibRw0rk: What's the issue?
14:46.57elgcan I log in agents from the CLI?
14:47.17elgas in callback login
14:47.21SibRw0rkiCEBrkr: http://pastebin.com/532134  when i make a phone call i get these warnings, and then a 603 Declined
14:47.37iCEBrkrSibRw0rk: I'm not much of a network guy.
14:48.03iCEBrkrAFK
14:48.08fugitivois ok a dual xeon 2gb ram for 100 g711 simultaneous calls?
14:48.08SibRw0rki don't understand why i get a everyone is busy/congested @ this time - when i'm the only one on the network
14:48.10SibRw0rkargh
14:48.14SibRw0rki have to go onsite anyway
14:48.15SibRw0rkbbl
14:48.18junbughmm, any majors cost barries FCC etc.. if I wanna provide outbound voip access to a couple of clients on my colo box
14:48.34RevKWe were not setting minjitterbuffer, I wonder if that is a factor - I'll test
14:48.40brettnemjunbug: not besides the normal 911 requirements
14:49.25junbugbrettnem: oh i have to provide that? even tho they  have a land line
14:50.16brettnemmy understanding is that if you are providing telephone service, you better provide 911 as well.
14:51.04saftsackfugitivo: 100 simaltaneous calls. wow :>
14:51.09brettnembah
14:51.33junbugbrettnem: *sigh*  alrighty then ...  oh well i can do like inphonex.com   they charge $25 per 911 dial
14:51.34danzigfug> if you are not transcoding (translating from one protocol to another) calls, that box is much more than necesarry. If you are translating all the calls to another codec, it may not be enough.
14:51.46fugitivono transcoding
14:51.58brettnemjunbug: they charge $25 per 911 call? What a scam
14:52.41wunderkinfugitivo: a single xeon could easily handle that, actually im sure something smaller could also.. im doing a lot more than that on a single xeon
14:53.00fugitivowunderkin: what do you do with your box?
14:53.16*** join/#asterisk unixgeek (n=unixgeek@12.45.238.189)
14:53.17saftsackyou guys a crazy. howto connect 100 telehones on asterisk? with a pri card?
14:53.21danzigI run 170 users  on a pIII 750 Mhz 256 Mb ram - 10 simulatnious calls = 0.00 load. But it depends a lot on what u are doing - playing music, speech, conferences, transcoding, all take a lot more
14:53.48fugitivosaftsack: 4 E1, ip phones or softphones in my case
14:53.52danzigsaft>> ethernet, if they are IP phones :-)
14:54.02saftsackfugitivo: :)
14:54.16wunderkinfugitivo: i tested 4 t1 doing backgrounddetect and connecting to a remote pgsql server
14:54.26brettnemwhen I get > 100 registrations.. asterisk starts acting funny.. leaving open file decriptors, etc.. no transcoding..
14:54.27saftsacka e1 one port card is cheaper than a bri 8 port card. why? ^^
14:54.29fugitivowunderkin: with a single xeon?
14:54.33wunderkinya
14:54.47fugitivobrettnem: sip registrations?
14:54.56brettnemfugitivo: yes
14:55.08fugitivobrettnem: 1.2.x?
14:55.11coppicexeons are like breasts. nature intends for them to come in pairs
14:55.25brettnemfugitivo: peers.. actually on both 1.0 and 1.2.x
14:55.44fugitivobrettnem: hmmm
14:55.45brettnemand who said coppice couldn't tell jokes?
14:55.56fugitivocoppice: for 100 simultaneous g711 calls, dual or single?
14:56.38coppicefor just G.711 a single current xeon should do
14:56.39wunderkinfugitivo: im doing all ulaw, on a single xeon 3.0 2mb cache
14:56.46wunderkinyeah
14:57.04fugitivogreat, the problem is if they want to start using voip providers
14:57.13fugitivoi'll tell them to buy a dual motherboard
14:58.56*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)
14:59.07*** join/#asterisk Skumling (n=skumling@fw.sg12.dk)
14:59.34*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197)
15:00.28danzigbrettne>> I have 350 registrations all the time, and * i rock solid - never reboot. But I have no hardware in the box - all connected over ethernet. My impression is that zaptel and other hw drivers have more memory leack etc than * itself
15:01.03brettnemdanzig: I'm not using any zaptel or any other hardware
15:01.21fugitivodanzig: what hardware for the server?
15:01.37brettnemsingle P4, 1 Gb mem
15:01.53danzigbrett>> hmmm... well,all I can say is  I am on * 1.0.9 on Debian, works great fo me...
15:02.13brettnemit'll be a long time before I ever say that asterisk works well for me.. heh
15:02.30*** part/#asterisk Supercross (n=superX@thbh-ip-vsat-2-p224.telkom-ipnet.co.za)
15:02.32danzigfug>> trash bamboo pIII 733 Mhz. Not to be recommended, but works fine. I have a backup.
15:03.01fugitivodanzig: g711??
15:03.16fugitivodon't tell me you use g729 for 300 users with that box
15:03.36danzigum.. ALAW - that is g711A, is it not? I can't remember...
15:04.08gaupedanzig: that's right
15:04.20*** join/#asterisk santiago (n=santiago@63.245.86.155)
15:04.35danzigno, I use ALAW (occansionally ULAW), there are 170 users, but that have 170 phones and 150 SIP trunks to various providers
15:04.49fugitivodanzig: are you using channel banks or gateways?
15:04.50*** join/#asterisk ComPuTeR (n=NAZAN___@85.107.169.248)
15:04.54*** part/#asterisk santiago (n=santiago@63.245.86.155)
15:05.34oej~seen cresl1n
15:05.36jbotcresl1n <n=matt@gateway.digium.com> was last seen on IRC in channel #asterisk, 3d 13h 7m 13s ago, saying: 'nmsclera: DS0s on the PRI'.
15:05.37danzigneither - all phones are ethernet attatched (Grandstream GXP 2000), all trunks are SIP all the way out to the provider.
15:06.04fugitivodanzig: you use g711 for the trunks?
15:06.22fugitivohow many simultaneous calls?
15:06.22*** join/#asterisk neon_kl (i=neon_kl@218.208.240.171)
15:06.25danzigfug>> 711a, yes
15:07.13danzigfug>> Very seldom see over 10 - i.e. "20" from asterisks point of view
15:09.29danzigfug>> we bought an Athlon 64 server for it, but saw that it was overkill, så haven't been bothered to move from the test box yet - but these people are students in a dorm - we don't lose 100000$/hour if the phones don't work
15:12.34*** join/#asterisk CaViCcHi (n=matteo@81.208.84.216)
15:12.44CaViCcHiHI
15:13.24Nuggetgros glandeur!
15:13.38CaViCcHican someone help me with a function?
15:13.39JunK-Ynugget: hjeheh
15:13.48CaViCcHiIFTIME function
15:14.26CaViCcHiin ael
15:14.47CaViCcHiextensions.ael I have an extension... i use...
15:14.50CaViCcHiIFTIME(9:00-13:00|mon-fri|*|*?goto s|work);
15:15.26CaViCcHiit answers back... WARNING[38235]: pbx.c:1690 pbx_extension_helper: No application 'IFTIME' for extension ...
15:15.42CaViCcHiand hangs up
15:16.36CaViCcHiAny Help?
15:16.43JunK-Ywhat it does if u do show function IFTIME ?
15:16.53CaViCcHiit shows me how to use
15:16.56CaViCcHiso its compiled in
15:17.15JunK-Yshow me 1 line before and after.
15:17.19CaViCcHi-= Info about function 'IFTIME' =-...
15:17.37CaViCcHiTemporal Conditional: Returns the data following '?' if true else the data following ':'
15:17.46CaViCcHi[Description]
15:17.46CaViCcHiNot available
15:17.51*** join/#asterisk sevard (n=kynan@198.174.233.25)
15:17.56JunK-Yi mean in ur AEL
15:18.01CaViCcHioh ok
15:18.21sevardCan some one please point me to a document explaining modules, how to load them, where to download them, what each module does.  I can't find any documents and would like to learn this.
15:18.27*** join/#asterisk P0L0 (n=n0n3@140.Red-83-58-255.dynamicIP.rima-tde.net)
15:18.30danzigCa>> What * version? Can it be that that function is only available in a newer version?
15:18.37CaViCcHiinizio:
15:18.37CaViCcHi<PROTECTED>
15:18.42CaViCcHi<PROTECTED>
15:18.42CaViCcHinolavoro:
15:19.19CaViCcHiAsterisk 1.2.1
15:19.31danzigthats not the problem then
15:20.06DannyFdamn, internet is broken today :/
15:20.41CaViCcHiyes... but I'm following instructions... i cant understand why
15:21.17*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
15:21.43*** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com)
15:22.21RevKIFTIME is not a conditional command, it is a function. What did you think goto s|lavora evaluates too exactly?
15:22.38saftsackhi i have asterisk + hylafax. are some hylafax experts here?
15:22.49CaViCcHiit just evaluates?
15:22.49saftsackbecause i want to know, howto print with hylafax
15:22.55RevKYou want GotoIfTime
15:23.03saftsackalso that all outgoing faxes are printed with hylafax
15:23.14RevKfunctions do evaluate to a value... What did you think Temporal Conditional: Returns the data following '?' if true else the data following ':' means?
15:23.20RevKIt "returns a value", not "does a command"
15:23.32RevKHope that helps
15:23.38CaViCcHiIt helps a lot...
15:23.46CaViCcHii work too much :P i need vacancy
15:28.33*** join/#asterisk SplasPood (i=jwb@206.252.198.100)
15:29.32*** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0i7.dialup.mindspring.com)
15:29.41*** join/#asterisk apardo (n=apardo@87.218.45.104)
15:29.57wunderkingone fishing
15:30.05*** join/#asterisk b_52FREE (n=b_52FREE@adsl-212-18-192-81.adsl.iam.net.ma)
15:30.06danzigI think he meant holiday. Vacance in french. Or something.
15:31.09*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
15:31.16[TK]D-Fenderwell.. that was neat...
15:31.25b_52FREEhi
15:32.42CaViCcHiyea
15:32.51CaViCcHijust a Zlang :P
15:33.15CaViCcHianother nice question could be... how can i retrieve # from a call?
15:33.33*** join/#asterisk klictel (n=klictel@207.107.208.137)
15:34.42wunderkinwhat number?
15:34.54CaViCcHiPhone number
15:34.59CaViCcHicaller
15:35.18CaViCcHiof Mr. I'm calling your asterisk
15:35.54wunderkin${CALLERID(num)}
15:36.47CaViCcHiHere I am in italy... and sometimes u need that ur ISP allows you to see the #
15:37.15CaViCcHii'll try anyway
15:37.32*** join/#asterisk mhnoyes_ (n=mhnoyes@user-38lc0f9.dialup.mindspring.com)
15:40.52Nuggethttp://colo.slacker.com/stuff/italy.swf
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15:46.45*** part/#asterisk saftsack (n=lottc@p54A7EC24.dip.t-dialin.net)
15:47.12[Atlas]Nice!
15:50.56ast_freakDoes 1.2.3 have a memory leak?  I thought I saw some conversation about that yesterday.
15:51.03ast_freakDoes 1.2.4 have a memory leak?  I thought I saw some conversation about that yesterday.
15:51.08ast_freak:^)
15:51.29[Atlas]i thought i saw a conversation yesterday about that too but i think it was but .3
15:51.38[Atlas]if i remember correctly
15:52.00ast_freakWierd, I could have sworn it was 1.2.4.
15:52.08sevard<PROTECTED>
15:52.14mzo_argh, a 1.2.4 out? already?
15:52.14[Atlas]can i give you my word as a spaniard?
15:52.23[Atlas];p
15:53.13[Atlas]ast realtime via odbc is in 1.2.x stable right? no need to cvs co?
15:53.56ast_freaklol
15:54.01ast_freakno good
15:54.03ast_freak:^)
15:54.07[Atlas]LOL
15:55.30mdavesevard, exten => 3 refers to the '3' being dialed
15:55.44*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
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15:56.00mdaveexten => s,3 refers to nothing being dialed, and the '3' refers to the order in which the exten => specifications are referred to
15:56.06mdaveaka 'priority'
15:56.26sevardmdave: thank you very much.  is there a 'current time' module?
15:56.42mdaveno idea
15:56.56mdavebut see http://voip-info.org/wiki/view/Asterisk+config+extensions.conf
15:57.08mdavefor more details on the syntax for extensions.conf
15:57.26sevardawesome, is there a module respository?
15:57.44*** join/#asterisk stack_ (n=stack@63.239.190.202)
15:57.52mdaveim not even sure what a module is
15:57.54mdave:P
15:58.09sevardlike the wake-up call module
15:58.15stack_Is it possible to have a queue with members that are external phone numbers or voicemail boxes?
15:58.28sevardi believe it works after  placing the module in the extensions.conf file but I think the time on my server is screwed up
15:59.22rob0<== just got back from watching http://colo.slacker.com/stuff/italy.swf and LOL :)
15:59.37*** join/#asterisk mhnoyes__ (n=mhnoyes@user-2ivfmv1.dialup.mindspring.com)
16:00.29RoyKka-ding
16:00.32RoyKanyone here from uk?
16:00.37RoyKgr0mit: ping
16:00.56*** join/#asterisk ComPuTeR (n=TrgirL_@85.107.169.248)
16:02.34fourcheezeRoyK: me me
16:02.58*** join/#asterisk rmorris (n=rmorris@d221-85-117.commercial.cgocable.net)
16:02.59RoyKfourcheeze: do you know a good place to place a colo?
16:03.02*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
16:03.09fourcheezeaha yes
16:03.11fourcheezewell
16:03.15fourcheezewhat kind of place are you after?
16:03.32RoyKwe prolly need a PRI and 1-4U worth rackspace
16:03.49Dr-Linuxanybody is using sphinx voice recognition with asterisk?
16:04.03fourcheezeif RevK is awake I've a feeling that's his line, if he's the RevK that I think he is
16:04.30*** join/#asterisk diclophis (n=diclophi@Sac-12-201.cisdata.net)
16:04.36diclophishowdy all
16:04.41diclophis... so what is up with SMS?
16:04.41*** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81)
16:04.53diclophisis that the same thing as sending an email to 1231231234@vmail.net ?
16:05.11fourcheezeRoyK: sent you a PM
16:06.45sevardwhy doesn't this work?
16:06.46sevardexten => 556,1,Answer
16:06.47sevardexten => 556,2,Playback,current-time
16:07.05diclophisIIRC the syntax for playback is Playback(filename)
16:07.15rmorrisI have * 1.2.x my x-lite phone works fine for logging into voice mail, but it seems * does not hear the tones from my hardware phone
16:07.38RoyKfourcheeze: ping?
16:07.40sevarddiclophis: so current-time would not spit out the current time?
16:07.45[Atlas]so , if im reading the doxygen for asterisk realtime correctly, you still have to configure some things by hand in the config files?
16:07.47*** join/#asterisk b_52FREE (n=b_52FREE@adsl-212-18-192-81.adsl.iam.net.ma)
16:07.48fourcheezeRoyK: not getting anything from you
16:07.50rmorrisAT-320 phone sip
16:08.00fourcheezeRoyK: you on jabber?
16:08.08diclophisthere might be a variable in the channel for the current time
16:08.16diclophisthen you would need to do something like SayTime
16:08.17diclophisperhaps?
16:08.25diclophisbut IIRC playback is only for files
16:08.40RoyKfourcheeze: wtf? hm...
16:08.50RoyKfourcheeze: only msn and ichat
16:08.56fourcheezeok, I can do msn
16:09.12RoyKroy@karlsbakk.net
16:09.13sevarddiclophis: Do you know where I could pull up a document that would have the correct syntax for that?
16:09.32*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-227-29.claranet.co.uk)
16:10.05sevarddiclophis: I'm looking at http://www.voip-info.org/wiki/view/say+time  but to tell you the truth I don't fully understand it
16:11.25*** join/#asterisk leopardus (n=leopardu@217.22.179.69)
16:11.40rmorrisanyone? ideas why asterisk would not understand/hear tones correctly?
16:11.43diclophisdo youi know anything about unixtime?
16:11.51sevarddiclophis: I do not.
16:12.08diclophisthe "time" variable needs to be an integer... that represents the number of seconds that have elapsed from the epoch
16:12.10sevardexec date ;)
16:12.19sevardI knew that. Yes.
16:12.32leopardushello : which is the best, and simple to install, softphone for linux?
16:12.48sevardleopardus: if you're using X then the xlite phone from xten.com is simple.
16:12.54diclophisso now you need to find the current time somehow
16:13.06diclophisi would imagine there has to be a channel var
16:13.23sevardSAY DATETIME <time> <escape digits> [format] [timezone]
16:13.31leopardussevard : xlite acts funny on linux
16:13.46sevardleopardus: I haven't experienced 'funnyness'
16:14.42leopardussevard : I'm running xlite on the same server as asterisk
16:14.42leopardussevard : should that be a problem?
16:14.50sevardI have yet to try but i imagine not.
16:14.51diclophistry SAY TIME ${EPOCH}
16:15.46sevarddiclophis: I get the same thing.  It connects but gives me dead air.  I did 'reload'.
16:16.03leopardussevard : I'm downloading lipz4 ,,,,??
16:16.06sevard#time extension
16:16.06sevardexten => 556,1,Answer
16:16.06sevardexten => 556,2,SAY TIME ${EPOCH}
16:16.12rmorrisjust got off the phone with the phone support people. They say (surprise) this is an asterisk problem not a phone problem
16:16.13sevardleopardus: what?!
16:16.18*** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com)
16:16.37leopardussevard : I'm going to try lipz4
16:16.39JunK-Ysevard: SAY TIME is an agi command.
16:16.47diclophisah...
16:17.00diclophisis there a way of doing that in the dialplan?
16:17.10JunK-Yshow application SayUnixTime
16:17.13leopardussevard : going for tea, maybe I catch you again, bye
16:17.40sevardSayUnixTime([unixtime][|[timezone][|format]])
16:18.19*** join/#asterisk crich1999 (n=crich@p54BFC4D9.dip0.t-ipconnect.de)
16:18.36diclophisthat should doit
16:18.37ast_freakJan 31 09:18:18 DEBUG[20337] pbx_spool.c: Delaying retry since we're currently running '`,
16:18.37ast_freak<PROTECTED>
16:18.41sevardAwesome!
16:18.45ast_freakWhat's up with that?
16:18.47stack_Is it possible to have a queue with members that are external phone numbers or voicemail boxes?
16:18.58sevardexten => 556,1,Answer
16:18.58sevardexten => 556,2,SayUnixTime
16:19.03sevardworks perfectly
16:19.04JunK-Ysevard: * is awesome :)
16:19.14diclophisha, i knew it was something easy
16:19.21sevardvery much so, each day i get my hands dirtier with this and it surprises me even more
16:19.36sevardthere is a weather module also, no?
16:19.37diclophisso now back to my question.. what is the most reliable way to sen hpone mail?
16:20.01*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
16:20.11rob0sevard: that's my extension 8463 :)
16:20.11NDTAnyone have a copy of the file that was on the wiki with this description? --> PHP ZAP GUI: A single php page to view calls taking place on ZAP channel (copper line). Has the option to hang up call. Very Basic.
16:20.36NDTdead link now
16:20.49sevardrob0: can you copy and paste to me (perhaps in /msg) your 8463 part in your extensions.conf ?
16:20.55rob0sure
16:21.07sevardthanks man
16:22.08_Paulo_Steve Underwood is the man!
16:22.31*** join/#asterisk beraldi (n=beraldi@gw-telecorp.telecorp.com.br)
16:23.09sevardrob0: oh, i thougt you meant weather :P
16:23.23*** part/#asterisk CaViCcHi (n=matteo@81.208.84.216)
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16:26.40badboyzis there a way to make asterisks playback a message to the caller, while the dialplan is still executing?
16:26.53Nuggetbackground()
16:26.56badboyzbackground() doesnt do it .. background waits till its done playing before moving on
16:27.32wunderkinwhat else do you want to do badboyz
16:27.52badboyzwunderkin: i want the caller to hear a sound file, while the dialplan is still executing
16:28.01wunderkinwhat else is it executing
16:28.10iCEBrkrbadboyz: you can't spawn 'threads'
16:28.20iCEBrkrbadboyz: Everything executes top-down
16:28.25badboyzlets say they are hearing a greeting, while the dialplan is trying to find someone to answer the call
16:28.33*** join/#asterisk j4m3s_ (n=j4m3s@user-24-214-119-188.knology.net)
16:28.49wunderkinbadboyz so you are doing a findme
16:28.53iCEBrkrbadboyz: You'll have to dump them in a queue or park their call while the dialplan is running around
16:29.20mdaveanyone have any non-US test numbers? I just got fwdOUT setup and working, would like to give it a spin
16:29.46badboyzwunderkin: yes, but w/ audio playing during the findme process
16:29.51iCEBrkrmdave: FWD completed their PSTN inter-connect??
16:30.14wunderkini know there is a findme app in mantis
16:30.17mdaveer, eh? no this is the community line-sharing thing
16:30.25mdavefwdout.net
16:30.32iCEBrkroh so it works like DUNDi?
16:30.40mdaveive seen that, but dont know what it is
16:30.40*** join/#asterisk Assid (n=assid@203.115.64.12)
16:30.43*** join/#asterisk Krill (n=majestic@210-84-11-13.dyn.iinet.net.au)
16:30.57mdavealthough something in fwdout mentions it will 'use dundi' if it doesnt have a route
16:31.23mdavewhere to find inf on DUNDi?
16:31.34mdavenvm
16:31.41Dr-Linuxanybody is using sphinx voice recognition with asterisk?
16:31.48iCEBrkrDr-Linux: LOL
16:31.58iCEBrkrmdave: Yea, it appears to be a lot like DUNDi
16:32.36mdavedundi.com appears to have a low of theory and 'why its great' type info, but seems to be lacking any detailed info on how it works or how to use it
16:33.11mdaveugh. and their press release is a MSword file
16:33.22iCEBrkryeah and their whitepaper is a PDF
16:33.25iCEBrkrmy machine is about to die
16:33.32iCEBrkrjunk workstation here at the office :(
16:33.33Dr-LinuxiCEBrkr: heh ;)
16:33.33mdavepdf isnt as reprehensible as word
16:33.38Assidsup iCEBrkr
16:33.38mdaveannoying at times
16:33.45iCEBrkrIt's sad when you have better hardware at home than at work.
16:34.12iCEBrkrmdave: Yea, well, I'm already in .NET and spawning a copy of PDFreader is thrashing my system
16:34.12mdavebut regardless, there seems to be no tech info there.. any idea where to find info that gets to the point?
16:34.37Dr-LinuxiCEBrkr: i insatalled everything for sphinx, but have a question
16:34.38Dr-Linux[root@I2C-PBX root]# ./sphinx-netclient.pl /var/lib/asterisk/sounds/thanks-for-using.gsm
16:34.39Dr-LinuxResult: YES
16:35.04serg_bhow can i move both legs of bridged call into meetme ? for example by assigning dynamic feature ?
16:35.07iCEBrkrmdave: The whitepaper describes it a bit better
16:35.22iCEBrkrDr-Linux: I'm headed to lunch.. and it's been almost a year since I've tinkered with Sphinx
16:35.22infoboxhi
16:35.50rmorrisAnyone know why mpg123 would be running in the background all the time?
16:36.08Dr-LinuxiCEBrkr: oky :S
16:36.50mdavermorris, it sits waiting, paused, to play music on hold
16:37.01rmorristhx !
16:37.03wunderkinDr-Linux, im still fighting to get sphinx3 working properly
16:37.05mdaveif you dont need or want any music on hold
16:37.06sevardCan anyone help me with this wake-up module?  It seems to be working but not making making the correct call
16:37.09wunderkinin batch mode
16:37.09mdaveedit musiconhold.conf
16:37.22sevardwhen somebody requests a wakeup it's put into a file in /var/spool/asterisk/wakeups/
16:37.34sevardin the asterisk manager it says:
16:37.39sevard<PROTECTED>
16:37.45sevardbut the phone doesn't recieve a call
16:38.39Dr-Linuxwunderkin: did you try sphinux2 ?
16:38.52*** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it)
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16:42.15wunderkinyeah, my problem is just getting it configured and working with the dictionary i want
16:43.30rmorrisdo I need to place the voicemail setting in both internal and external contexts?
16:44.14rmorrisie. voicemail works if I dial in zaptel, but not in the internal context
16:45.06matteomm
16:45.06*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
16:49.02sevardwhiskey tango foxtrot!
16:49.03sevardawesome
16:50.11moverany help needed for a notify problem
16:50.39movermwi on realtime works with rtcache right?
16:50.56rmorrisso back to my #1 question ... are there any settings that tell asterisk how to listen for tones?
16:52.37*** join/#asterisk pointer (i=pointer@aj.catt.com)
16:53.38pointeris there a way to check to see if a file exists from within the dialplan?  STAT() doesn't exist in the code but is in the wiki and EXISTS() appears to look in the DB or something along those lines
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16:56.26znoGanyone know how to view active Zap calls?
16:57.25synthetiqzap show channels
16:57.38pointerno, that shows all zap channels
16:57.40synthetiqerr
16:58.06pointershow channels
16:58.11pointerbut that shows all of them
16:58.55wasimasterisk -rx 'show channels' | grep Zap
17:00.15*** join/#asterisk bigjb (n=nbigjb@195.60.10.114)
17:00.30znoGno, i want to view active calls to the Zap chans
17:00.50znoGah show channels is good
17:00.54pointerany ideas on the file exists problem?
17:01.39moveranyone here expert of sip protocol and asterisk?
17:01.57jontowhmm, whats the consensus on FreeBSD 6.0-REL and asterisk 1.2+?
17:02.06jontowusable for VoIP only, in a very reliable fashion?
17:04.24*** join/#asterisk masonf_ (n=masonf@dungle.vineyard.net)
17:05.36znoGhrm, is it normal to see this:
17:05.37znoGZap/8-1              s@macro-llamar:100   Busy    Congestion()
17:05.42znoGin show channels
17:05.48znoGthey don't seem to disappear either
17:07.08wasimznoG: that means it hasn't detected calling channel hangup, put a congestion(60) or something
17:07.22znoGok
17:07.24masonf_can I make the hold button on my polycomm soundpoint play asterisk music on hold?
17:07.24*** join/#asterisk matteo (n=matteo@81.208.84.216)
17:08.49*** join/#asterisk EriSan (n=erisan@81-174-35-6.f5.ngi.it)
17:09.11znoGwasim: its actually showing channels that are not active anymore, such a call that took place earlier, shows "state" as Up, but i've confirmed the call has already taken place.
17:09.15*** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc)
17:11.52jaikemasonf: it does for me...i didnt have to make any special configuration
17:12.05copantlany body know how to convert a v400p from t1 to e1?
17:13.11wasimcopantl: jumper it
17:13.15mdavehow does one find out the available formats for the record and/or monitor file? one would think either of the doc pages on those would mention, but they dont seem to
17:13.19wasimcopantl: or hardwire it in the driver
17:13.53wasimUse 'show file formats' to see the available formats on your system
17:14.06mdaveaha
17:14.07mdavethank you
17:14.18jaikemdave: wav, wav49, gsm
17:14.45copantlwasim: dont have any jumper
17:14.50diclophisso... yea, whats up with phone mail?
17:14.53diclophisand Sms?
17:15.03wasimcopantl: v400?
17:15.15mdavethere a page anywhere comparing size/quality for those?
17:15.20copantlv400 t/e1
17:15.37copantlvarion right
17:15.38copantl'
17:15.43wasimright, haven't used it
17:16.29*** join/#asterisk _zebras (n=chris@62.69.89.38)
17:16.48copantland www.zapatatelephony.org is down
17:17.23wasimits on the tor2, so those were manufactured either t1 or e1
17:17.28wasimnot switchable ...
17:17.44jaikemdave: i suggest wav49, small size. can be played on windows media player
17:17.55_zebrasI'm running asterisk at home, when I simulate an external incoming call it works great, rings the right group etc - but when I ring from outside the line is permanently engaged (fxo connected to pstn) - any ideas how I can start to troubleshoot this..
17:18.52copantlany body have a V400p card?
17:19.51mdavewindows media player is irrlevent
17:19.54mdavethis is for archival
17:20.16mdavei dont run windows anywhere for any reason
17:20.57*** join/#asterisk QbY (n=Kelvin@adsl-068-209-210-253.sip.cha.bellsouth.net)
17:22.29copantlwasim: do you think is not switchable?
17:22.47mizticis there a way to make asterisk wait for a dialtone before sending digits with Dial() ? I'm having an odd problem where local calls don't work 80% of the time, its like the exchange isn't getting the first digit
17:23.05znoGthis is weird, some calls that are made don't seem to close the channel when they've finished. I presume it's because Asterisk is not detecting a hangup, which is strange.
17:23.21mdavei dont need full wav/pcm quality, but id like slightly better than wav49/gsm
17:24.00miztici'd even settle for inserting a delay between opening the line and dialing the number, inserting "," into the dialed number doesn't appear to help any
17:24.07mdaveid like to find a comparison/overview of all the formats
17:24.36mdaveanyway, i dont have time now
17:24.45jaikei think youll have to do some tests
17:25.06mdavei suppose
17:25.10mdavethanks
17:25.18mdave:)
17:25.20mdaveim off
17:25.46SibRw0rkAH WTF!
17:25.50SibRw0rki can't figure out what this error means
17:26.00diclophisso does anyone know anything about phone mail?
17:26.26SibRw0rkhttp://pastebin.com/532370
17:26.28SibRw0rkanyone pleas3
17:26.29wasimmiztic: w is wait
17:26.34mizticaha
17:26.37mizticlet me try that
17:26.38mizticthanks
17:26.40SibRw0rkwhat does Warning[315] entail?
17:27.08wasimdial(zap/1/wwwww9484858)
17:27.17mizticgotcha, doing that now
17:27.20mizticappreciate it :)
17:27.39mizticbingo
17:27.39mizticgreat
17:27.44mizticlets see if that helps
17:28.47*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
17:31.20*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197)
17:32.50miztici think that's done it, thanks a lot
17:32.59*** join/#asterisk roulduke_ (i=5sbgb7qg@p508D21A8.dip0.t-ipconnect.de)
17:33.12*** join/#asterisk Lurr (n=pr0ph3t@adsl-11-11-32.mia.bellsouth.net)
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17:34.44*** part/#asterisk Lurr (n=pr0ph3t@adsl-11-11-32.mia.bellsouth.net)
17:37.19pauldyI haatee commminggg back ttto aaaa blueeee channeeeel window   without thhhat    faintest idddea of whooo said    my nnname or wwwhat the connntext wasss
17:38.07znoGthats why you use irssi and do /lastlog pauldy
17:38.20rob0It was me who said that pauldy, but I was just joking.
17:39.17pauldyxchat isfor me cause I'm   guid like  that
17:39.47*** join/#asterisk kio (n=kio@195-11.customer.cloud9.net)
17:39.53*** join/#asterisk basty (n=basty@212.218.65.235)
17:40.00bastyHi
17:40.54pauldyyour one  letter off
17:41.00*** join/#asterisk SERGEUS (n=s@195.112.98.13)
17:41.03iCEBrkrpauldy: Yea, but with irssi, you can use screen which allows you to attach and then reattach console sessions from different terminals. :P
17:41.28*** join/#asterisk samueltc (n=samuel@levinux.UQAR.UQUEBEC.CA)
17:41.35samueltchi
17:41.48samueltchow I can track an Originate [call] in the manager api?
17:41.58wasimuniquecallid
17:42.04samueltcactually, I set a random callerid
17:42.13samueltcyes but how I known the uniquecallid
17:42.19wasimits a var
17:42.24Assidumm.. is it possible to set the timezone that will be used in gotoiftime ?
17:42.35iCEBrkrsamueltc: Track it by the extension
17:42.38pauldyiCEBrkr, I know but then I  couldn't bitch and how much   fun would that be
17:42.46samueltcI known, but when I originate the call, I don;t known the uniquecallid
17:43.35samueltciCEBrkr: the extensions is not present in all events
17:43.59iCEBrkrsamueltc: The ones you care about it is..
17:44.42iCEBrkrsamueltc: I've already started work on a soon to be professional grade call manager. I've worked extenstively with the manage port.  The stuff you care about is trackable by extension
17:44.43samueltcI care about Newchannel and the extension is not present
17:44.47iCEBrkror well.. Channel
17:44.59*** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
17:45.02iCEBrkrsamueltc: I think you're going about it the wrong way then.
17:45.09bastyI have a little Problem with setting up dialplans on Asterisk. We have a main Asterisk-Server, that is connected to a SIP-Provider. The SIP-Provider requests calls in a string like: Dial (SIP/11<callingnumber>*20*0049{EXTEN:1}@sip.provider... If I insert a Number into the field "Dialingno" it sends this number for external calls. Now I am trying to figure out a way on how to make this more flexible. Means a customer connects with an other Asterisk to a SIP
17:45.29*** join/#asterisk FastJack (i=fastjack@p5091F6F3.dip.t-dialin.net)
17:45.54*** part/#asterisk gr0mit (n=w10277@206.41.25.138)
17:46.01*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)
17:46.19Dr-Linuxwunderkin: did you try sphinux2 ?
17:46.28samueltciCEBrkr: Newchannel give me the state of the channel, but there is no extension in that event
17:46.42iCEBrkrsamueltc: Why are you looking for Newchannel?
17:46.49samueltc'ringing'
17:47.00Dr-LinuxiCEBrkr: welcome back :P
17:47.12iCEBrkrsamueltc: I suspect you're trying to do the same deal with like desktop manager/callerID type app?
17:47.31samueltciCEBrkr: no
17:47.54Nuggethttp://www.echostore.com/skype-usb-to-rj11-adaptor.html  <-- has anyone here used one of these (or equivalent) as a fairly lame way to link asterisk and skype?
17:48.11NuggetI'm thinking that box, plus an FXO might let me route to skype
17:48.25iCEBrkrNugget: haha, maybe
17:48.43*** join/#asterisk RoyK (n=roy@ti211310a080-2037.bb.online.no)
17:48.47Nuggetit'd be ugly, for sure, but it might just work.
17:49.03_zebraswhere do I set the initial delay before asterisk answers the call, whether I direct incoming pstn direct to extensions or groups it still seems to ring 2-3 times before performing the incoming rules...
17:49.09samueltcIf I could get the uniqueid of my call, that would work..., actualy I'm Originate the call with a uniqueid in the callerid field, then with that field I known the uniqueid
17:49.10iCEBrkrNugget: No uglier than those cellphone cradles
17:49.15Nuggetyeah
17:49.31austinnichols101samueltc: but you still have to have your PC on
17:49.39austinnichols101running skype
17:49.42samueltcaustinnichols101: my pc?
17:49.53iCEBrkrNugget: Shit, $60??
17:49.58samueltci'm not talkinga about skype...
17:50.09NuggetI think I'm going to order one, just to play with.
17:50.17Nuggetcheap enough that if it doesn't work I won't mind
17:50.22austinnichols101sorry: meant nugget
17:50.22iCEBrkrNugget: Go for it, lemme know how it works out :P  Then I'll have to get one too
17:50.43iCEBrkrsamueltc: I'm lost as to what the end result of you're trying to do is...
17:50.51Kattyso. i got the xrays.
17:50.58Kattyand the dentist even said i had gorgeous teeth.
17:51.08*** join/#asterisk malcolmd (n=malcolmd@gateway.digium.com)
17:51.24Kattybad news is, i'm going in for surgery :<
17:51.57_Sam--he liked your teeth so much he wants to pull a few out and keep them as his own!
17:52.07samueltciCEBrkr: it's pretty simple, I originate call on the manager API, and then I track that call. But the problem is, once the call originated, I have no way to track it because I don;t known the uniqueid.
17:52.08Kattyhe can't do it.
17:52.13Kattyit will require an oral surgeon
17:52.20Kattyall 4 of them must go, he says.
17:52.23Kattycurse of the little people.
17:52.24iCEBrkrsamueltc: Even if you set ActionID or anything like that??
17:52.39iCEBrkrsamueltc: I guess ActionID is more for when you're waiting for a response back from a manage command.. hrrrm.
17:52.42samueltciCEBrkr: yes, actionID don't give me the uniqueid
17:53.16NetgeeksKatty: Wisdom teeth?
17:53.29SwK[Work]anyone get a situation where  chan_zap.c:1583 zt_set_hook: zt hook failed: Device or resource busy is filling the screen and the message log
17:53.37*** join/#asterisk jpablo (n=jpablo@200.94.130.194)
17:53.48samueltciCEBrkr: actualy i'm generating a random number in the callerid field of my Originate command
17:54.04samueltcthen the callerid is present in all event I need to track
17:54.09KattyNetgeeks: yes.
17:54.17JunK-YNugget: let me know how it works, that will be a nice gift to brother-in-law.
17:55.09iCEBrkrsamueltc: If you know the channel or extension/phone number you're doing the originate on----hrmm shit
17:55.16iCEBrkrOk, I'm kinda stumped
17:55.25NuggetI just ordered one.  I'll let everyone know how it works out
17:55.25NetgeeksKatty: Well, the good news is that you will most likely get the fun gas... I did
17:55.29iCEBrkrCuz with my call manager, I know the originating extension number and I track that..
17:55.34NuggetHolding off on buying the FXO, though.  :)
17:55.59iCEBrkrNugget: Well, if it works the way the device claims, then it's quite obvious you could use it to interface with Asterisk
17:56.05iCEBrkrerrr, wait.
17:56.05samueltciCEBrkr: if 2 people call the same extensions?
17:56.05NetgeeksI had all 4 removed at the same time, and it took me out of commission for about 3 days
17:56.06iCEBrkrMaybe not
17:56.08*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)
17:56.25iCEBrkrsamueltc: Wouldn't matter, cuz you have the Originating extension/number
17:56.25*** join/#asterisk fugitivo (n=ajf@201.255.176.83)
17:56.36samueltcok
17:56.44iCEBrkrNugget: That USB thing is gonna try to literally dial, which means it'll want a dialtone
17:57.03NetgeeksKatty, I don't remember any significant pain.
17:57.15Netgeekssome soreness, dull ache kind
17:57.15Nuggetyeah, let me clarify.  I've got one FXO I use for POTS right now.  I'll use it for testing and if it works out I'll add a second FXO.
17:58.04iCEBrkrNugget: I might be confused.. But those cheap-o Intel voice-modems, they're considered FXO or FXS?  I continually get those confused/backwards
17:58.14NuggetFXO
17:58.17justinuFXO
17:58.19iCEBrkrok
17:58.22NuggetFXO plugs into a dial tone.  FXS makes a dial tone.
17:58.28KattyNetgeeks: :>
17:58.32fugitivo~fxofxs
17:58.33jbotrumour has it, fxofxs is 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.
17:58.33iCEBrkrNugget: ok, ok ok.. gotcha
17:58.34junbugany good external fxo's these days
17:58.37Kattynow /thats/ what i wanna hear!
17:58.46justinujunbug: spa-3000
17:58.54iCEBrkrNugget: So, how's that USB device going to work with an FXO card if the FXO card doesn't provide dialtone?
17:59.09justinukatty: i had 4 teeth pulled when I was a teen to make room for the wisdom teeth
17:59.10Nuggetit only needs a dialtone if you want it to be able to route calls out over PSTN.
17:59.10junbugthe 3000 is fxo ... lemme check it out
17:59.19NuggetI just plan to use it as a gateway to skype
17:59.21Kattyjustinu: they're going in after mine.
17:59.27KattyNetgeeks: how do they put you out?
17:59.30iCEBrkrNugget: Oh.. outbound from Asterisk to Skype
17:59.32Nuggetso it's acting as an FXS, so to speek.
17:59.36Nuggetor inbound.
17:59.39justinuKatty: when's d-day?
17:59.45Kattyjustinu: feb 17th
17:59.47Nuggetbut no PSTN connection at all.  I already have asterisk for that
17:59.47NetgeeksKatty: I got Gas to start and Sodium Pent after the gas had me goofy
17:59.48iCEBrkrNugget: You wouldn't be able to take inbound Skype calls
17:59.54Nuggetit says it can.
17:59.57KattyNetgeeks: was it an iv?
18:00.15justinuKatty: boo, plenty of time to dwell on it too :(
18:00.18iCEBrkrNugget: But how? The USB device is going to dial a phone number to connect the two legs of the call.
18:00.22Nuggetno.
18:00.35iCEBrkrNugget: It'll ring a phone?
18:00.36Nuggetunless your definition of "phone number" is a lot looser than mine
18:00.39Nuggetyes, it will
18:00.42Kattyjustinu: i'm not all that worried about the surgery, to be honest.
18:00.43iCEBrkroh well, cool
18:00.45NetgeeksKatty I don't know.  the gas had me soo goofy, I can't remember ANY shots, IV's etc.  I came to in a dentist's chair with no IV in me
18:00.59Kattyjustinu: i was more worried about the financial issue...
18:01.09KattyNetgeeks: oh, goodly.
18:01.12NuggetIt's exactly like an SPA-3000, except for skype instead of PSTN.
18:01.14iCEBrkrNugget: Hrrm.  Damnit, now I want one of these, just for fun
18:01.15*** join/#asterisk MatsK (n=mk@cC30123C5.inet.catch.no)
18:01.19jpablohey people, I'm having a problem with my pri, when i dial a number trougth it and the number doesn't exits it gives a busy tone imediatly, instead of passing the providers error message.
18:01.23samueltciCEBrkr: http://pastebin.ca/39276 I can match the uniqueID with the callerid on line 14
18:01.24NetgeeksHeya Drumkilla!
18:01.29rob0<== had oral surgery in about 1968
18:01.42jpabloany idea how can i hear the prividers error message?  the people in the call center really need it
18:01.55KattyNetgeeks: will i remember much?
18:02.01justinuKatty: there is that too, isn't there...
18:02.05samueltcthen I knonw that the phone is ringing on line 31
18:02.24Kattyjustinu: yeah, but my dad volutneered to put it all on a credit card for me to pay back.
18:02.37Kattyjustinu: so i'm not all freaked out about billing anymore (=
18:02.38NetgeeksKatty: I had dreams that I was talking to my Teeth (they were like 5 feet tall and walked) and we were reminiscing the good times before they left... thats all I remember
18:02.45NetgeeksKatty: Like I said, gas is good
18:02.46justinuw00t, what would we do with out parents
18:02.49iCEBrkrsamueltc: See, I match all my stuff on Channel
18:02.55KattyNetgeeks: so you dreamed...during the operation?!
18:03.02NetgeeksKatty: yes
18:03.02KattyNetgeeks: you're not like completely /out/?
18:03.05Kattyoh dear.
18:03.08Kattythis isn't good
18:03.14iCEBrkrsamueltc: Cuz I'm only looking for events that are directed towards me, or my desktop call manager
18:03.17iCEBrkrhrrrm
18:03.20NetgeeksKatty: I had no recollection of the surgury itself
18:03.31ZeeekKatty it's less good than rice milk
18:03.34justinuNO2 will put you on the rings of saturn
18:03.34NetgeeksKatty: Like a real dream that you wake up and wonder where you are
18:03.48justinua completely different plane
18:04.10samueltciCEBrkr: but i'll probably originate call to the same channel (IAX2/out-gw00/14185551212)
18:04.14KattyNetgeeks: hmmmmmmmmmmmmm...
18:04.15Zeeekmovies are lousy on planes
18:04.21KattyNetgeeks: were you not scared of the effects of the drugs?
18:04.24samueltcinitiate more than one call on the same channel
18:04.27*** join/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net)
18:04.27*** join/#asterisk [ToTo] (n=ToTo@host72-146.pool872.interbusiness.it)
18:04.30iCEBrkrsamueltc: That's external. All your extensions will be SIP
18:04.38justinuthe drugs are the best part!
18:04.39iCEBrkrOhhh
18:04.40iCEBrkrok
18:04.42iCEBrkrhrrm.
18:04.54ZeeekKatty get a general anasthetic
18:04.57ErrKatty: it's called "twilight" anesthesia around here - you won't remember it at all (I've had it done several times)
18:05.05NetgeeksKatty: No, I was too scared of the whole surgery event to worry about the drigs
18:05.05Zeeekthen even if you die, it won't matter
18:05.08Netgeeksdrugs
18:05.12KattyNetgeeks: oh, ok
18:05.17KattyZeeek: :<<
18:05.27*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
18:05.37ZeeekIs it more than like wisdom teeth pulling?
18:05.39KattyNetgeeks: describe procedure please
18:05.43samueltcIf I could just update the callerid (after the match on the newcallerid event) in the manager API that would probably work...
18:05.51brettnemErr: Do they IV you for twilight?
18:06.01justinunah, it's just gas
18:06.06justinunawwws
18:06.07Errbrettnem: it's just gas
18:06.15ErrIVs are if they put your completely under, AFAIK
18:06.18NetgeeksAll I can say is that I don't recall any un-pleasantness associated with the event... I remember going to the oral surgeon, getting the gas and having a finger pulse/oxy monitor I kept trying to push off as the gas took hold
18:06.26_Sam--Katty:  you need someone to hold you and tell you its going to be fine....because it will, its a minor procedure
18:06.26brettnemhmm.. last time I had NO2, I remembered everything.. I suppose I didn't have enough
18:06.54NetgeeksKatty: I remember the dream as clear as it was 2 minutes ago, and waking up in a pretty empty room in a reclining dentist chair with a dental assistant
18:07.32NetgeeksKatty: I went home and lived on jello/milkshakes, yogurt, etc. for the weekend and was back to work like tuesday (surgury was on friday)
18:07.39ZeeekI woke up in a waiting room with three other people holding bloody kleenexes just like the one I was holding!
18:07.40brettnemI remember my dentist trying to calm me down by telling me how much better it is being at the dentist than the orthodontist
18:07.41justinuNetgeeks: one question... was she hot?
18:07.44iCEBrkrLOL @ rGd's signoff
18:08.02iCEBrkrjustinu: ^5
18:08.15SwK[Work]http://pastebin.ca/39277
18:08.17SwK[Work]h0 h0 h0
18:09.03Netgeeksjustinu: I honestly don't remember.... if the drugs do anything like a few beers, then I don't see how she couldn't have been hot
18:09.10justinulol
18:09.13Zeeekheh
18:09.50Dr-Linuxhi justinu ;)
18:10.23justinuhey there
18:10.36_Sam--whats up Dr-Linux...you get your project finished?
18:11.27*** join/#asterisk Lurr (n=pr0ph3t@adsl-11-11-32.mia.bellsouth.net)
18:11.37*** part/#asterisk Lurr (n=pr0ph3t@adsl-11-11-32.mia.bellsouth.net)
18:11.40Dr-Linux_Sam--: we didn't try to start it yet. they java guy was on leave :)
18:11.41*** part/#asterisk retentiveboy (n=retentiv@h73.90.40.69.ip.alltel.net)
18:11.46NetgeeksKatty: When are you getting the procedure done?  Any idea yet?
18:12.07_Sam--did you ever figure out why the one read() never worked right?  and gotoif
18:12.37_Sam--that was still bothering me
18:12.51_Sam--where gotoif was going to the wrong priority..remember?
18:13.17Dr-Linux_Sam--: i just figured out that the problem is with the script. not with dialplan
18:13.50_Sam--did you find out what values the script/agi is wanting you to give it from the dialplan?
18:14.07Dr-Linux_Sam--: dialplan is fine,
18:14.12justinuNetgeeks: she said feb 17th ;)
18:14.20_Sam--but the dialplan has to talk to the agi and give it values
18:14.34Dr-Linuxyes
18:14.47_Sam--you know what values the agi needs to get?
18:14.50Dr-Linuxbut what if the agi script was wrong ?
18:15.22_Sam--if the agi script was wrong then it wouldnt connect to the informix database
18:15.23Assidhrmm.. is there a way to set the owner of voicemails to something else besides root?
18:15.30Dr-Linux_Sam--: do you think dialplan was wrong?
18:15.57_Sam--Dr-Linux:  i feel really stupid, but i couldnt figure out why the gotoif was going to the wrong priority
18:16.05_Sam--so im not sure exactly what is/was wrong
18:16.10_Sam--because i looked at it like 50 times
18:16.17*** join/#asterisk CoderCR (n=creyna@ip21.farheap.net)
18:16.38*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
18:17.02*** part/#asterisk CoderCR (n=creyna@ip21.farheap.net)
18:17.03_Sam--but you tihnk it is because the agi was returning bad values
18:17.05_Sam--?
18:17.07Dr-Linux_Sam--: no thats not Read() or gotoif problem, thats script problem
18:17.13Dr-Linuxyess
18:17.24Assidi set the owners to daemon:daemon in the asterisk.conf
18:17.29Dr-Linux_Sam--: agi always return 0
18:17.29Assidbut it still doesnt do that
18:17.30_Sam--ok, that does make sense, but we SAW what the agi was returning...it was returning 0
18:17.37_Sam--and we said if 0, goto 20
18:17.40_Sam--but it went to 30
18:18.03tekatiI have the ability to switch providers to get me a little better upload speed from 412 to a meg.  The new link is wireless and doing tests it will be a difference to my SIP provider from 85ms to 125ms.  How will that effect performance?  Anyone know?
18:18.06Dr-Linux_Sam--: but always 0, if i hit wrong i never saw anything else except 0
18:18.53_Sam--but even when it was 0, the gotoif didnt go to the right place i thought
18:18.58_Sam--forget that it never returns 1
18:19.05_Sam--even when it returned 0, it didnt go to the right place
18:19.08badboyzanyone messed w/ the d option for the Dial commmand? its where a person can DTMF during the call -- i cant find more documentation but i would like it while its ringing, they could punch 1 and it goes to VM
18:19.22Assidhey iCEBrkr you around?
18:19.48Assidany clue on the file owner ship?
18:19.56iCEBrkrAssid: For?
18:20.11Dr-Linux_Sam--: i hate AGI's one thing, it doesn't show error output, it always show returning 0 with any thing
18:20.31Assidvoicemails
18:20.37Assidthey keep doing as root:root
18:20.46Assidi tried setting the ownership in asterisk.conf
18:20.48Dr-Linux_Sam--: as i'm playing with another thing since 4 hours, AGI returns 0
18:20.49Assidbut it doesnt help
18:20.49_Sam--what about /var/log/asterisk/my_agi.log
18:21.00Dr-Linuxooo ic
18:21.47iCEBrkrAssid: Apparently you're running Asterisk as root....
18:22.16*** join/#asterisk A-jay (n=quirc@62.217.245.194)
18:23.04_Sam--Dr-Linux:  do you have any experience with embedded linuces on  SBC computers or MINI ITX?
18:23.23*** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)
18:23.47*** join/#asterisk muzzz (n=chatzill@218.111.66.117)
18:24.07fugitivoi do with mini itx
18:24.14*** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)
18:24.17_Sam--epia?
18:24.20fugitivoyes
18:24.30_Sam--how many concurrent calls can a newer 1.3ghz epia handle?
18:24.37_Sam--w/ say 512ram
18:24.38justinuembedded meaning what? no hard drive?
18:24.47_Sam--embedded into a diskonchips or something
18:25.05fugitivo_Sam--: what kind of calls, codec, etc?
18:25.06_Sam--embedded into the actual sbc or something as well
18:25.24*** join/#asterisk rpm (n=russell@S010600111155e117.cg.shawcable.net)
18:25.24_Sam--fugitivo:  no PSTN, just SIP --> IAX Provider or something
18:25.33_Sam--using U
18:25.57fugitivog711?
18:26.01_Sam--yep
18:26.28`lymeDial failed due to CHANUNAVAIL
18:26.33`lymewhat all could that mean?
18:27.18*** join/#asterisk dijit0_ (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net)
18:27.46dijit0_would anyone happen to have any idea that when i call someone the caller id shows up as 19999991234 ?? thats nothing close to what i have set, lol
18:28.18*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
18:29.27junbugdijit0_: some providers dont support call id forwarding ...
18:29.30*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
18:30.06dijit0_it USED to work though
18:30.17dijit0_just recently, me and my friend havce the same problem all of a sudden
18:30.24*** join/#asterisk razu (n=razu@ip59.cab62.mus.starman.ee)
18:31.48dijit0_grr! when i call through idefisk and connect to nufone or wahtever, it works fine
18:32.04dijit0_bypassing asterisk, that is...
18:36.02dijit0_ugh! now its working... but why did it stop in the first place...
18:36.12mzo_it's called a 'feature'
18:36.21*** part/#asterisk mhnoyes__ (n=mhnoyes@user-2ivfmv1.dialup.mindspring.com)
18:37.28dijit0_meaning it works half the time? lol
18:37.40mzo_yes!
18:37.42mzo_a feature! )
18:38.45*** join/#asterisk MattH (n=MattH@63.174.244.174)
18:38.54dijit0_heh, know of any voip providers that allow you to receive callerid WITH NAME? that asterisk can work with
18:39.07MattHif someone is registering their asterisk server to mine with sip (yes I know maybe we should be using iax) what is causing this error when I try to send a call to them?
18:39.07MattHJan 31 12:36:28 NOTICE[3716] chan_sip.c: Failed to authenticate user "+15703232166" <sip:+15703232166@63.174.244.172>;ta g=as7facadd4
18:39.12_Sam--dijit0_:  teliax is doing caller id with name, works well
18:39.25DarkFlibbledijit0_ its generally an additional service... about 0.2c/ call
18:39.59rajiv|work_Sam--: is teliax's network fixed ?
18:39.59_Sam--im sure there other places too, but teliax does all our origination so i dont know any others
18:40.04_Sam--rajiv:  is it ever?
18:40.20dijit0_i c... ok thx
18:40.33rajiv|work_Sam--: can you recommend them?
18:40.39_Sam--i dont have any problems with teliax's routing...but i have noticed for days 1-3% packet loss at their router.
18:41.04_Sam--rajiv:  i think it depends on your expectations...i could recommend them, but if you're not happy with them, then my recommendation doesnt matter :)
18:41.08FuriousGeorgeis it asterisk or eyebeam that is playing the double "boop" callwaiting sound?  if its the former, can i stop it?
18:41.10_Sam--they work reasonably well for me.
18:41.16zamslerhmm
18:41.16_Sam--i do have backup accounts, just in case.
18:41.26zamslerwhat kind of prices are you guys looking for ?
18:41.54FuriousGeorgewell, i want to stop the sound, not the call from being "answerable" by that extension
18:42.10FuriousGeorgethe booping drowns out the conversation for local party
18:42.26_Sam--darwin_35:  say something
18:42.41_Sam--when are you going to fix the 1-3% packet loss at your router
18:43.11justinuheh, 1-3% loss isn't all that bad
18:43.21_Sam--i notice it
18:43.23_Sam--on calls
18:43.24FuriousGeorgehow do we test packet loss at the router it
18:43.34justinuulaw?
18:43.37_Sam--yeah
18:43.46[av]baniyay its sam
18:43.48justinumust be the gxp not doing any PLC
18:43.50_Sam--FuriousGeorge:  something like mtr works
18:44.33_Sam--hey their bani...how goes it?
18:45.13[av]baniyour gxp2000 comment on the wiki got nuked somehow
18:45.35_Sam--i tried
18:45.35zamslerhooooooah
18:45.36justinucongressional staffers screwing with the voip-info.org wiki too?? :P
18:45.42_Sam--lol
18:46.18[av]banii think it got nuked in an edit
18:46.29[av]banisomeone else must have been editing at the same time
18:46.58jhiver~seen p0lar
18:47.00jbotp0lar <~p0lar@64.254.225.62> was last seen on IRC in channel #asterisk, 318d 22h 10m 6s ago, saying: 'time to order..hehe'.
18:47.22[av]bani_Sam--: the only phones you have are gxp-2000 ?
18:47.47zamslerlol
18:47.54_Sam--yep, and a few of those utstarcomm wifi phones
18:48.09[av]banihow are the utstarcomm?
18:48.18*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
18:48.18[av]banii'm assuming they are equally cheap
18:48.25_Sam--i guess since i dont have anything to compare them to....i would say they work ok
18:48.30malverian[work]Anyone know of some good festival voices (eg.. better sounding than the mbrola ones)
18:48.46_Sam--i just dont know what else i would compare em to...i got my wife a plantronics USB wireless headset...
18:48.57_Sam--i think the plantronics headset through a softphone works better , sound quality wise
18:49.03[av]banicompare them to the gxp2000's
18:49.41_Sam--the gxp2000s sounds much better
18:50.03_Sam--the utstarcomm gets some 'clipping' type sounds, like there is too much gain on someplace
18:50.17_Sam--i have to turn the volume down pretty low for it to go away
18:50.32_Sam--but all in all, i have 3 guys in a warehouse with the f1000s , and they never complain
18:50.46_Sam--sometimes they have to reboot them because calls will stop working
18:51.01justinumy lusers would complain non-stop about that.
18:51.15justinunon-fucking-stop
18:51.20_Sam--we are talking about once a week
18:51.22_Sam--not once a day
18:51.25justinudoesn't matter
18:51.26_Sam--maybe even less than once a week
18:51.36rajiv|work_Sam--: i want origination that works. i want my phone to ring when calls come in, and the quality to be acceptable. i'm using gizmo now and surprisingly it works.
18:51.44*** part/#asterisk _Paulo_ (n=paulos@200-168-112-132.dsl.telesp.net.br)
18:51.52_Sam--see, it helps too that i am the owner of the place, because when people complain, i can always tell them , dont like it, get your own phone, or theres the door
18:51.57justinucool
18:52.04hypnoxany dundi experts here? I am running a local dundi 'cloud'. Numbers are advertised and looked up okay, but the IP of the destination is always set to 127.0.0.1 - even though i use ${IPADDR} in dundi.conf as recommended
18:52.30_Sam--rajiv:  i think you could find alot of companies that would be able to provide reliably what you're looking for
18:52.35*** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
18:52.37_Sam--i am anxious to give asterlink a shot
18:52.57_Sam--their server(s) are the closest ones that ive seen route-wise to me
18:53.39*** join/#asterisk INOT|Bewildered (n=thehouse@80-195-138-8.cable.ubr07.uddi.blueyonder.co.uk)
18:54.35trixterthey are a good bunch of guys
18:54.48dijit0_this caller id problem seems to be related to whether i set it in idefisk or not... i dont want the user to be able to set it, but if not specified, it doesnt work for some reason
18:55.03[av]bani_Sam--: who do you use for origination?
18:55.09_Sam--teliax currently
18:55.13badboyzso if you set a timeout to a s,1,Dial(SIP/200,30) -- where does it go when it times out? +101 ?
18:55.28[av]banido your callers complain about choppiness?
18:55.45[av]banii call out and people say its choppy, but i hear them fine
18:56.48_Sam--no, i dont get many complaints at all.  i am like 55ms away from the server i connect to.  when i listen in on sales calls they always sound good, and i never hear the customer complain...
18:56.50justinuthat's upstream problems
18:57.03[av]banii dont get problems with junction networks, and theyre farther away
18:57.13[av]baniwe have a ds3
18:57.17[av]baniand its not loaded
18:57.43_Sam--i am on a good ISP who is homed on three or 4 different backbones, i think that helps me out often
18:57.47[av]baniteliax is 30ms away and jn is 90ms
18:57.52_Sam--because my path to teliax goes all the way over cogent
18:57.54[av]baniwe're multihomed too
18:57.58_Sam--no interconnects or anything
18:58.11[av]baniteliax has issues, jn doesnt even though jn is 3x as far
18:58.22[av]baniso i'm inclined to assume teliax has inbound issues
18:58.28jpablohey people, I'm having a problem with my pri, when i dial a number trougth it and the number doesn't exists it gives a busy tone imediatly, instead of passing the providers error message, any idea how can i pass the error message audio back to my extensions ?
18:58.36[av]banisince our outbound goes over the same pipes
18:59.04_Sam--brad_mssw was complaining about the same thing with teliax, but i think the problem is related to the route you take to teliax
18:59.07*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:59.16[av]bani_Sam--: i tried different routes, same results
18:59.53_Sam--do you connect to voip-co3.teliax or voip-co2?
18:59.58[av]banico3
19:00.06_Sam--for some reason co2 is like 1 hop closer
19:00.10[av]banii hear them fine, callers say i sound choppy
19:00.17_Sam--i connect to co2
19:00.29*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
19:01.13MRH2anyone use polycom phones and sync to a ntp server?
19:01.20_Sam--bani if you want the guys IM info from teliax let me know, he is usually pretty helpful, which is one of the reasons ive stayed around.
19:01.46ZeeekMRH2 ya
19:01.52_Sam--if he realizes that you know possibly as much as he does, he does tend to listen and try to help
19:01.59_Sam--otherwise he will tell you to reboot your computer and send you on your way
19:02.27MRH2mine are consistently 10 secs slower - do you get the same?
19:02.38[av]bani_Sam--: cool, icq?
19:02.41Zeeekno I kept getting on ehour diff!
19:02.47Zeeekone hour
19:02.48_Sam--aolim, will message name
19:03.20ZeeekMRH2 I'd say you're not syncing with the server at all
19:03.47_Sam--[av]bani:  what are you doing with a DS3 over there?
19:03.52brettnemHey, anyone know why I'd get a bunch of stuck SIP channels with method REFER?
19:03.56_Sam--maybe you could talk teliax into putting a server on it :)
19:03.59MRH2definately am ;)
19:04.08[av]bani_Sam--: isp
19:04.16[av]baniwe resell dsl
19:04.16Zeeekthe phone is running free though?
19:04.27*** part/#asterisk pointer (i=pointer@aj.catt.com)
19:04.28MRH2running free?
19:04.32_Sam--nice, i owned an ISP from 94-2002, sold it off
19:04.36[av]bani:)
19:04.40_Sam--who babysits your servers? :)
19:04.41[av]bani94-02, wow
19:04.43Zeeekyou can't have sync and be 10 secs off!
19:04.46[av]banii do :()
19:05.08[av]baniwe're replacing our inhouse POS phone system with * for PBX
19:05.10_Sam--we have the new verizon fiber down here that is killing off the dsl resellers
19:05.14_Sam--verizon FIOS
19:05.18[av]banitrying to figure out a good solution for PSTN
19:05.21_Sam--i dont think they can resell it
19:05.37[av]baniteliax would work if they werent choppy
19:05.48*** join/#asterisk stack_ (n=stack@63.239.190.202)
19:05.58rajiv|work[av]bani: maybe you need QoS on your upload?
19:06.04_Sam--i use teliax to run my current biz...we do over 7 million a year in telephone sales...and teliax does work ok
19:06.09[av]banii also tried iax, same
19:06.16_Sam--telephone sales = mail orders sales, not sales of telephones :)
19:06.23[av]bani:)
19:06.27[av]banimail orders of what?
19:06.39_Sam--we sell motorcycle stuff for sportbikes
19:06.42brettnemhere's a pastebin of those STUCK SIP channels in REFER method: http://pastebin.ca/39286
19:06.46brettnemAny ideas anyone?
19:06.47iCEBrkr_Sam--: SELL ME STUFF
19:06.53[av]bani:)
19:06.53_Sam--anything you want, at cost
19:06.56_Sam--www.kneedraggers.com
19:07.02iCEBrkr_Sam--: Oh you're one of THOSE guys?
19:07.06*** join/#asterisk DarkFlibb (n=darkflib@cpc4-nfds9-6-0-cust148.leic.cable.ntl.com)
19:07.07iCEBrkr:-/
19:07.11_Sam--which guys?
19:07.17[av]baninice website
19:07.21[av]baniimpressive
19:07.28iCEBrkr_Sam--: A few of the kneedragger.com guys used to hang out in #Motorcycles on EFNet
19:07.30MRH2if the ntp is down i get flashing date/time of 1st jan  - if ntp gets a successful response time is 10secs out
19:07.35_Sam--that was me, ice.
19:07.36iCEBrkrThis was years ago tho.. I'm not sure now
19:07.39iCEBrkrhahaha
19:07.40[av]baniheh
19:07.41_Sam--i still have friends on efnet #motorcycles
19:07.44_Sam--ryanb
19:07.48brettnemI have around 83 stuck sip channels.. anyone have any clues?
19:07.52iCEBrkr_Sam--: The internet is too small :P
19:07.53[av]banithe owner gets to loaf around on irc all day  <3
19:07.57DaminiCEBrkr: What's up bitch? :)
19:07.59ZeeekMRH2 never heard anything like that (not that that helps you, sorry)
19:08.00stack_how does the "include" statement work with regards to the time options?  If I have a set of holiday rules and then a daytime rule and the holiday was on a weekday, wouldn't it include both the holiday rule and the daytime rule?
19:08.02iCEBrkrDamin: Nap time!!!!
19:08.09_Sam--iCEBrkr:  that is funny shit...do you still hang out there?
19:08.14DaminiCEBrkr: Is that a sanctioned work event? :)
19:08.17MRH2ok  r urs in sych exactly?
19:08.20denon_Sam--: your "You may be interested in" feature is broke
19:08.27brettnemargh
19:08.28iCEBrkr_Sam--: Seriously tho, I got a '89 gixxer.. I need plastics!! not fiberglass shit :P
19:08.30[av]bani'hacker safe' eg safe for hackers!
19:08.36_Sam--i had a falling out with someone on that channel because i wouldnt give them a deal....the guy was 'desmo' i think
19:08.39iCEBrkrDamin: If I were President, it would be!
19:08.48_Sam--denon:  give me a page
19:08.57denon_Sam--: http://www.kneedraggers.com/list/1.2 - click "Goodridge speed bleaders"
19:09.01iCEBrkr_Sam--: LOL I remember Desmo..
19:09.04denonI assume its an old product or something
19:09.11justinudesmodromic?
19:09.12iCEBrkr_Sam--: I was supposed to meet up with him at Daytona Bikeweek
19:09.14brettnemargh
19:09.14denonbut you need to take it out of both tables ;)
19:09.20brettnemanyone want to talk about asterisk? :)
19:09.26ZeeekMRH2 like I said, it was off for an hour no matter what I did on the web interface. Rebooting without an ftp server didn't help, only with ftp and XML did it straighten up
19:09.26MRH2what firmware are you on Zeeek?
19:09.35_Sam--iCEBrkr:  if you make it down there...look me up :)
19:09.38_Sam--we race there every year
19:09.39jhivernite' all
19:09.41Zeeek1.4 something, it's a ip500
19:10.06iCEBrkr_Sam--: I'm in Tampa/St. Pete.. if I get my bike together, I'll be at Bikeweek.
19:10.08_Sam--denon:  thanks...i will have the web guy fix it
19:10.08brettnemAnyone know anything about STUCK SIP CHANNELS? :) http://pastebin.ca/39286
19:10.10*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-204-92.red.bezeqint.net)
19:10.10denon_Sam--: also http://www.kneedraggers.com/details/Brembo_Radial_Brake_Master_Cylinders--12-BREMBO-1.html - alt tag in the wrong place
19:10.16iCEBrkr_Sam--: hopefully this year, it won't be a 3hr shower....
19:10.39denonalso .. heh .. well, just tell your web guy to get his act together ;)
19:10.45iCEBrkrdenon: haha
19:10.47_Sam--iCEBrkr:  nice, when im at the don ceaser next time i'll look you up :)
19:10.57iCEBrkr_Sam--: Rich bastard.
19:10.58_Sam--but as far as plastics for your 89...that is a tougher proposition
19:11.04iCEBrkr_Sam--: LOL
19:11.19iCEBrkrI know.. I know.
19:11.32iCEBrkrI only need 3 more.. and a lower is the toughest to find
19:11.45MRH2wll my config files are sorted fine
19:11.49_Sam--maybe you should step up to the 21st century with a 6 year old bike :)
19:11.54iCEBrkrlol
19:11.55_Sam--an 00 R6 would be fine :)
19:12.01iCEBrkr<-- Gixxer dude
19:12.11iCEBrkrtho, I find the R6 & R1 pretty sexy
19:12.14_Sam--i bleed blue, we run a yamaha race team
19:12.17_Sam--they give us bikes and money
19:12.21MRH2i mean they are fine even adjust for daylight savings
19:12.22*** join/#asterisk dijit0 (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net)
19:12.30_Sam--but ive always been an R6 guy
19:12.36iCEBrkr_Sam--: I have a handful of friend who race CCS
19:12.42_Sam--i probably know em
19:12.50_Sam--in the florida region?
19:12.52iCEBrkrYeah
19:12.52fugitivook
19:12.55iCEBrkrTampa Track Junkies
19:13.00fugitivoenough of bikes
19:13.03_Sam--sorry
19:13.03iCEBrkrfugitivo: lol
19:13.04fugitivonow let's talk about trains
19:13.07ZeeekMRH2 actually I probably never looked to see if there was 10 sec diff
19:13.11_Sam--its like asterisk...addicting.
19:13.12iCEBrkrCHOOCHOOO
19:13.12fugitivotrains and monkeys
19:13.28iCEBrkrfugitivo: Whatever man, I got VoIP on my bike!
19:13.29_Sam--if you really ever need anything i will be glad to hook ya up, ice.
19:13.37fugitivoiCEBrkr: 1337!
19:13.38_Sam--and tell those losers on #motorcycles that i really miss them :)
19:13.51iCEBrkr_Sam--: Cool stuff!! I'll be in the market for a new helmet
19:13.56iCEBrkrOh, I don't hang out there anymore
19:13.58iCEBrkr:)
19:14.06_Sam--i am friends too with D3scart3s
19:14.12_Sam--from #motorcycles, you remember him?
19:14.14iCEBrkrI remember him too
19:14.21_Sam--we raced together at ccs a few times
19:14.24_Sam--him and mrcrash too
19:14.24justinuwhat do you guys ride?
19:14.51_Sam--i have 2 yamaha R6s (600cc) and 1 yamaha R1 (1000cc), and a yamaha 450 dirt bike
19:14.51MRH2ok :)
19:15.00justinucool
19:15.02synthetiqa huffy.
19:15.04iCEBrkrhttp://www.cyberdyne.org/~icebrkr/cpg142/thumbnails.php?album=50
19:15.06_Sam--do you ride at all?
19:15.06iCEBrkrsynthetiq: LOL
19:15.14justinuyeah
19:15.16iCEBrkr^^^^^^^^^^ URL to my bike pics!
19:15.20justinui used to have a gsxr600
19:15.24justinuand a gsxr1000
19:15.26iCEBrkrIt's old skewl, but it's clean!
19:15.34fugitivo_Sam--: i like supercross bikes
19:15.56_Sam--fugitivo:  hell yeah, they are great if you are jumping 3000 feet in the air
19:16.12fugitivo:)
19:17.31justinui had this for a while also: http://justinu.smugmug.com/photos/31062641-O.jpg
19:18.01iCEBrkrjustinu: Oh fuck a duc :P
19:18.08justinuheh
19:18.11iCEBrkr:P
19:18.56Kattylet's leave the poor ducks alone.
19:19.01[av]bani<_Sam--> to show my appreciation for #asterisk, free bikes for everyone in the channel
19:19.10iCEBrkr_Sam--: Man, you rock!
19:19.11_Sam--lol!!
19:19.15[av]baniyay!
19:19.22_Sam--justinu:  too bad bennie b. sux anymore
19:19.25[TK]D-Fender... oh the joy of the donorcycle....
19:19.29iCEBrkr[av]bani: The only deal is, they're orange with Asterisk stars on them.
19:19.32[av]banihaha
19:19.35mzo_i'd still ride it
19:19.36Katty[TK]D-Fender: i'll donor your cycle in a minute.
19:19.37[av]baniiCEBrkr: no problem!
19:19.39iCEBrkrlol
19:19.45_Sam--lol
19:19.51[TK]D-FenderKatty: Mew.
19:19.52_Sam--this is one my current bikes
19:19.54_Sam--http://www.kneedraggers.com/racer6/images/DSC00086.jpg
19:19.57Katty[TK]D-Fender: mew :<
19:20.18[TK]D-FenderKatty: Actually saying what said backwards sounds like some sort of proposition ;)
19:20.20[av]banii want a hello kitty cycle
19:20.35iCEBrkr_Sam--: Yea, I have to admit, it looks nice
19:20.42fugitivojustinu: you have a ducati???
19:21.05justinui did
19:21.12dougmmm
19:21.12justinui sold my bikes
19:21.14*** part/#asterisk doug (i=doug@zaxxon.telerama.com)
19:21.21iCEBrkrFine, fine. I'll link directly to my bike too ... http://www.cyberdyne.org/~icebrkr/cpg142/albums/pics/Motorcycles/Mine/dsc00137.jpg
19:21.48_Sam--this is some race team pictures from 05:
19:21.49_Sam--http://www.kneedraggers.com/gallery/
19:22.14justinumy 600: http://justinu.smugmug.com/photos/3171203-O.jpg
19:22.27_Sam--iCEBrkr:  damn, you got the 20th anniversary edition? :)  jk
19:22.31iCEBrkrLOL
19:22.35iCEBrkr_Sam--: The original!
19:22.42_Sam--thats a cool bike.
19:22.59iCEBrkr_Sam--: I actually would love to have the 20th anniversary edition one so I could park them next to each other :P
19:23.07_Sam--iCEBrkr:  my brother has one
19:23.16_Sam--justinu:  that is a nice picture...where is that from?
19:23.32iCEBrkrI'd park'm like this...
19:23.32iCEBrkrhttp://www.cyberdyne.org/~icebrkr/cpg142/albums/pics/Motorcycles/Mine/DSC01963.JPG
19:23.39iCEBrkrwarning, HUGE pic
19:23.41_Sam--<looks like you need a rear tire, justin>
19:24.06justinuon the 600?
19:24.06fugitivohow much is a bike like that?
19:24.26_Sam--8000-11,000 USD
19:24.36_Sam--the racebikes are like 20 with the work we do
19:24.41*** join/#asterisk pbd (n=plancomm@12.144.118.36)
19:24.44fugitivothat's cheap
19:24.52_Sam--the bikes are the cheap part
19:24.53pbdGreetings, all.
19:25.03fugitivoi'm sure i can't get a bike like that for less than 15k here
19:25.20_Sam--fugitivo:  then someone is ripping you off.
19:25.29_Sam--the retail price of the units is like 8000-11000 USD
19:25.35_Sam--except for the ducati that justin showed
19:25.40fugitivo_Sam--: yes, the goverment with taxes :)
19:25.57iCEBrkrI just want one of each...
19:26.08_Sam--iCEBrkr:  where was the last picture from?
19:26.09iCEBrkr1000RR, Ducati 999, etc.
19:26.14pbdQuick question, and yeah, I'll take lumps for not knowing this off the top of my head.. but to those of you who have installed the 729 codec- how did you decide between the i386, i586, and i686 versions?  'arch', or should I know something I'm blind to at the moment.
19:26.27fugitivopbd: wrong channel, this is #bikes
19:26.37iCEBrkr_Sam--: We have Quaker Steak and Lube ( it's a bar/resturant ) which holds a weekly bikenight
19:26.42iCEBrkrfugitivo: lol
19:26.55fugitivo#monkeys-and-trains soon
19:27.06justinui think i still like the duc 998 better
19:27.09pbdWoa.  Interesting.  Client tells me I'm in asterisk users, shows the right person list.. but isn't.  Sorry for the intrusion.
19:27.27fugitivopbd: do you like bikes?
19:27.41*** join/#asterisk saftsack (n=lottc@p54A7EC24.dip.t-dialin.net)
19:27.45_Sam--lol
19:27.52saftsack_Sam--: hi
19:27.52iCEBrkr_Sam--: The bike scene is HUGE down here in Tampa.. I'm sure I could send a lot of business your way
19:27.57_Sam--its funny that we (many of us) have the same interests with the bikes
19:27.58_Sam--and phones
19:28.04saftsacki have a little question with hylafax. it runs now :)
19:28.06_Sam--hey there sacky
19:28.09fugitivoand trains
19:28.15iCEBrkrand monkeys
19:28.17saftsackbut howto print every fax which is sent?
19:28.21iCEBrkrMOOSE PENIS
19:28.24saftsacki thought on notify?
19:28.31_Sam--saftsack:  i use cypheus for that
19:28.38_Sam--cypheus is a client that runs on a windows pc
19:28.39fugitivosaftsack: what bike model/brand?
19:28.50saftsackcypheus is a client for faxing
19:28.54saftsackor?
19:29.03_Sam--yes, for faxing, and for retrieving and printing faxes
19:29.14_Sam--it submits faxes from the windows pc to the hylafax server
19:29.17saftsacki  mean the server as internal which should print all sent faxes for archiving them
19:29.27_Sam--it does archive them on its own
19:29.28*** join/#asterisk sthw45ywyw5 (n=sthw45yw@38.136.42.2)
19:29.28iCEBrkrI really should start looking into faxing with Asterisk
19:29.36saftsack_Sam--: yes i know in doneq
19:29.37iCEBrkrSo I can have a 'fax machine'
19:29.39fugitivoi use spandsp
19:29.46_Sam--<PROTECTED>
19:29.50iCEBrkrTho, I don't see that happening since I don't have a land-line
19:29.53iCEBrkr:(
19:29.59saftsack_Sam--: yes but i want to have them in paperform
19:30.03saftsackalso printed
19:30.15fugitivosaftsack: using lpd?
19:30.22_Sam--like i said, i use cypheus for that....so that is all i know.
19:30.24saftsacki use cups
19:30.28_Sam--cypheus prints every incoming fax for me
19:30.31*** part/#asterisk pbd (n=plancomm@12.144.118.36)
19:30.48saftsack_Sam--: is cypheus running on your server?
19:30.52fugitivosaftsack: you could use some kind of script with lp, i don't know how hylafax works
19:31.06_Sam--cypheus runs on the windows PC and connects to the hylafax server , like POP3 email
19:31.08sthw45ywyw5someone help. I am trying to call from Zap to Polycom 301 phone.  The polycom phone rings , I pick up, but neiter side can hear the other speak.
19:31.09_Sam--same type thing.
19:31.10saftsackfugitivo: i want to do exact this one ;)
19:31.16jpablohey people my asterisk installationg is missing vm-youhaveno.gsm, but asterisk needs it, any idea where can i get it ?
19:31.31fugitivojpablo: www.asterisk.org
19:31.43saftsack_Sam--: do you know the notify script?
19:31.52_Sam--saftsack:  im sorry, no
19:32.11jpablofugitivo, where exactly ? it isn't in the asterisk tarball nor asterisk-sound tarball.
19:32.16dijit0does anyone know how to force the caller id in asterisk whether idefisk has set it or not?
19:32.26*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
19:32.33saftsackdo you know another method to execute a command directly after sending a fax?
19:33.01*** join/#asterisk mutil (i=WebChat@i.think.napoleon.dynamiteblows.com)
19:33.05fugitivojpablo: what asterisk version?
19:33.15mutilhey all how goes it
19:33.26jaikejpablo: try downloading old asterisk-sounds
19:33.35mutquestion.. whats the point in qualify?
19:33.41_Sam--saftsack:  did you check out all the info on hylafax.org?
19:33.48saftsackyes
19:33.49mutwhat does it do other than give me incorrect 'pings'
19:33.53_Sam--i think it tells you there how to do it
19:33.55jpablofugitivo, i'm running svn 1.2
19:33.57saftsacki found a faq for this point but just for incoming
19:34.04_Sam--http://www.hylafax.org/content/Automatically_print_incoming_faxes
19:34.06saftsackbut theres no outgoing help
19:34.14_Sam--oh, i thought you meant just incoming.
19:34.16*** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net)
19:34.20saftsackno
19:34.25saftsacki mean just outgoing ;)
19:34.42saftsackso i wanted to use the notify script for this
19:35.05_Sam--why do you need hard copy of every outgoing fax?
19:35.13_Sam--it it all stored and accessible if you need it later
19:35.20_Sam--then you can print manually the ones you need
19:35.34saftsackok
19:35.40saftsackbut my archive folder is empty
19:35.45saftsacki can find them in docq
19:35.52fugitivothat's true
19:35.52_Sam--you need to setup a cron script to archive
19:36.02_Sam--#Fax Stuff
19:36.02_Sam--00 07 * * *             /usr/local/sbin/faxcron -rcv 45 > /dev/null
19:36.02_Sam--15 07 * * *             /usr/local/sbin/faxqclean -aA > /dev/null
19:36.08fugitivowhy you use a faxserver if you need to print all faxes? buy a fax machine for that :)
19:36.30saftsackwhat is a fax machine? ^^ do you mean the hardware faxes?
19:37.16*** join/#asterisk Seedy (n=Seedy@65.200.153.2)
19:37.38SeedyHello fellow phone nerds!
19:37.47fugitivosaftsack: yes
19:37.53NetgeeksHi Seedy, did you get everything figured out last night?
19:38.21SeedyNetgeeks: Nope! But I'm understanding things more and more.
19:38.37SeedyAnd I have a few more basic questions I can't find answers too
19:39.08Netgeekswell, ask away, someone here is likely to answer in some format
19:39.19mutso does anyone know what sip qualify does?
19:40.14justinuit "pings" the UA
19:40.17justinuwith a sip options
19:41.10mutyea but for what purpose
19:41.21mutbecause i can ping it normally and get like.. 30ms replies
19:41.23saftsack_Sam--: faxcron is just for incoming, right?
19:41.46mutbut the sip 'ping' bounces from 40ms to 300ms to timing out (3000ms)
19:41.48MattHon asterisk box A I get "Got SIP response 481 "Call Leg Does Not Exist" back from " and on asterisk box B I get " Failed to authenticate user "+15703232166" <sip:+15703232166@63.174.244.172>;tag=as03ed0edb
19:41.48MattH" even though I have it setup as a peer and user any thoughts?
19:42.22Netgeeksmut: it isn't an ICMP ping.. it's a SIP ping, meaning it just doesn't test to see if the network stack on your system in running
19:42.26SeedyMy setup is an Asterisk Box that is accepting incoming calls from an external analog phone. I'm trying to figure out if IAX or SIP would work better with this (From what I've read IAX is better, but I don't think it will work with external phone lines)
19:42.49fourcheezeMattH: I bet there is a user of that name somewhere in your config
19:42.49mutNetgeeks: yea.. and?
19:42.53[TK]D-FenderSeedy : either can work just fine
19:42.59Netgeeksit checks to see that the far end is accepting SIP messages and waits for a response, so the round trip time includes the time it takes for the pinged system to process the sip message and respond
19:43.16mutso even though a normal ping on the ata gives 30ms
19:43.18Netgeeksso not only do you check that the system is there and alive, you also check that it's talking SIP and understanding it
19:43.22mutthis sip ping gives 200ms
19:43.27mutwhats that telling me?
19:43.46MattHfourcheeze, there isn't though that "user" is my callerid
19:43.47mutsometimes its unreachable alltogether
19:43.54mutbut i can still log into the ata and configure it
19:44.03mutand it says it's registered, it has dialtone, it can make calls
19:44.17Netgeeksright, if you can ping an ATA at the exact same time you do a sip options (qualify) message, and the ping time is 30 ms, and the sip round trip is 230ms, then 200 ms was spent by the ata processing the sip message and responding
19:44.34*** join/#asterisk outtolunc (n=me@adsl-66-218-53-170.dslextreme.com)
19:44.49*** join/#asterisk brock05 (n=admin@c220-239-93-31.rochd1.qld.optusnet.com.au)
19:44.51SeedySo I'm having trouble with receiving DTMF digits. Asterisk doesn't seem to recognize them at all. I am using SIP right now and was wondering if using IAX might fix this problem. Or is that a dumb idea.
19:44.57jaikemut: try replacing patch cables...happened once with me
19:45.08fourcheezeMattH: that's strange. That error normally means that either something shold have authenticated which didn't, or shouldn't have which did
19:45.09mutthis happens to a lot of people though..
19:46.19jpabloSeedy, there are three ways to send dtmf, both ends must be configured to use the same method
19:47.36Seedyjpablo: What are these three methods? Inband, rfcxxxx
19:47.42*** join/#asterisk los415 (n=los415@64.201.109.62)
19:48.04[av]baniSeedy: try changing your dtmf mode, try rfc2833
19:48.33[av]banimy guess is you're using inband
19:48.41MattHfourcheeze, right ... it's odd.. I can't figure it out :) hehe still trying... thinking maybe peer/user statements are messed up on the client end
19:48.45[av]baniwith sip, that's almost always wrong :)
19:49.22*** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com)
19:49.33fourcheezeMattH: yeah, do you have something as a peer in the sending box?
19:49.38jpabloSeedy, inbad, rfc?? and sip info
19:50.00SeedyThanks guys, I'll try using rfc2833
19:50.12`lymewhy would a FXS be reporting as unavilable, when i know its not in use?
19:50.24MattHfourcheeze-away, it's setup as a friend in the setuping box.
19:50.39jpablo`lyme, they somethings get "stuck", yeah that sucks.
19:51.22`lymewell, im still in the initial setup phases.. and they ahve never reported themselves as being available.  Im baffeled.
19:52.13`lymei have a 4 port digium card, only 1 line is plugged in, and the outbound route checks through all those ports, and they all come back unavail, even the one that i know is good...
19:52.28`lymeis there a manual way to set them as avail?
19:52.54austinnichols101what's a good pc for asterisk for a home system?  Price isn't as much of a concern as low power, footprint, etc.
19:53.17_Sam--austinnichols101:  build a mini-itx solid state machine
19:53.22fugitivo`lyme: /etc/zaptel.conf and /etc/asterisk/zapata.conf ?
19:53.30SeedyIf I am using a router, does that mean I am behind a NAT?
19:53.44*** join/#asterisk flashn253 (i=flashnet@Darkstar.AceShells.com)
19:53.44austinnichols101seedy: not necessarily
19:54.04austinnichols101seedy: what's the ip address on your local machine?
19:54.42fugitivohow does a person learn asterisk without knowledge of networking basics?
19:54.59`lymefugitivo: they are there, and configured, and will take incomming calls just fine.... its just the outbound :(
19:55.25`lymei set the channels with 1 avail connection each, and even tried bumping that to 50, and it still didnt work :(
19:55.28fugitivo`lyme: pastebin your outbound part of extensions.conf
19:55.29_Sam--fugitivo:  thats like saying how do you learn physics if you dont know basic math
19:55.34Seedyaustinnichols101: How do you mean? Other computers can access me with 192.168.0.x but my external ip is something else
19:55.43mzo_you learn physics by dropping heavy things off tall things.
19:55.44I-MODnat
19:55.52fugitivolol mzo_
19:56.13_Sam--i guess if you didnt know basic networking, you would learn it really fast when you are trying to setup your asterisk box.
19:56.18austinnichols101seedy: go to http://www.whatismyip.com and check your addy.  That shows what you look like when you're accessing the internet
19:56.25*** join/#asterisk SocialD (n=SocialD@CPE0040f45b3a28-CM00407b85d7bb.cpe.net.cable.rogers.com)
19:56.27_Sam--or at least, you would learn just enough to configure your ip and default routes :)
19:56.28mzo_nat is like being a super secret agent who has to carry messages to the free people outside without divulging what you're carrying. :P
19:56.37*** join/#asterisk trek (n=rsi@lns-bzn-35-82-250-207-194.adsl.proxad.net)
19:56.40SocialDHey
19:56.48*** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net)
19:56.50austinnichols101seedy: so the router is doing the translation between your local (private) address and the public address
19:57.33SocialDI just installed NetBSD 3:
19:57.43SocialDAnd I want to run Asterisk on it
19:57.45fugitivoSocialD: does zaptel work on that?
19:57.49Seedyaustinnichols101: Yeah... I'm trying to figure out if the NAT is causing my DTMF problems
19:57.53SocialDHavn't tryed yet
19:57.57brock05Hi I have one SIP account that keeps coming up with Bad Auth. All the other SIP accounts are working fine and everything in the account that comes up with bad auth looks ok. Does anyone know what might be doing it ?
19:58.06austinnichols101seedy: you're definitely behind nat
19:58.15ErrSeedy: if you're getting a connection, NAT is not the issue :-)
19:58.29SeedyErr: Thanks, that is what I needed to know
19:58.38fugitivoErr: you can have a connection, but not audio, and nat could be the problem
19:58.40Errif the audio and control channels are both working, NAT isn't in the way
19:58.45SocialDCan you guy's give me a sec, im gona run downstairs to my pentium 1 I wana run Asterisk on.. when I try to do a 'make install' I get a nasty GCC compilation error, 4 minutes be right back
19:58.47*** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com)
19:59.05SocialD<NetBSD 3>
19:59.28SeedySo I'm sure that incoming audio is working fine. But outgoing audio I am unsure of. Could this be a nat issue?
19:59.38fugitivoSeedy: yes
19:59.42Erryes
19:59.42jaikeone way audio?
19:59.43I-MODit could be you need to update your asterisk
19:59.58fugitivoSeedy: what asterisk version?
20:00.15jaikelol
20:00.16SeedyI-MOD: I am using 1.2.3
20:00.26I-MODawww....so much for an easy fix
20:00.30fugitivoI-MOD: i think the bug was 2-way no audio
20:00.34SeedyI am not sure if it is one way audio
20:01.08SeedyThe problem is I am not interfacing with another phone, just the server. So I don't know how to test if the server is getting my audio
20:01.21SeedyI guess I could try and record some audio to a file...
20:01.29I-MODMonitor()
20:01.43Error Echo()
20:01.46fugitivo`lyme: are you using asterisk2home?
20:01.47I-MODthat too
20:01.52fugitivoasterisk@home
20:01.53`lymesadly, yes.
20:01.57fugitivowell
20:02.06*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
20:02.07fugitivo~amp
20:02.09jbotamp is, like, NOT supported here! people using it should join #amportal
20:02.09SeedyErr: Echo doesn't work for sure
20:02.21fugitivo`lyme: trying to debug it is a pain in the ass
20:02.30`lymeah...
20:02.33`lymethaNKS ANYWAYS
20:02.51Errheh, if Echo doesn't work, you don't have two-way audio
20:03.08*** join/#asterisk miketaht (n=mtaht@67-127-179-114.ded.pacbell.net)
20:03.33austinnichols101aah has echo function: *43
20:04.31SeedyHmmmm.... Ok, I've got enough info to try out some more tests. Thanks again!
20:06.21`lyme#amportal is a quiet place. LOL
20:06.38*** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it)
20:08.15*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
20:08.43*** part/#asterisk brock05 (n=admin@c220-239-93-31.rochd1.qld.optusnet.com.au)
20:11.51*** join/#asterisk dijit0_ (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net)
20:13.45SocialDOk im back !
20:13.46SocialDmake: stopped in /AsteriskHAX/asterisk-1.2.0-beta1
20:14.09SocialDThats what I get when I do a make install in my NetBSD 3, box
20:14.23iCEBrkrSocialD: That's what you get for running NetBSD
20:14.25iCEBrkr:P
20:14.33SocialDI tried DSL before
20:15.12SocialDI spent 4 hours trying to get GCC work.. that failed, I got help from people in #damnsmalllinux .. they told me to install g++
20:15.12iCEBrkrSocialD: and CentOS, Fedora, or Debian isn't an option?
20:15.41SocialDMy harddrive can only hold 2 gigs
20:15.54iCEBrkrMinimal install.
20:15.55*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@62.240.252.22)
20:15.59SocialDIt's a pentium 150MHz
20:15.59[TK]D-FenderSocialD : Slackware is for yoU!!!!
20:16.01*** join/#asterisk dlynes (n=dlynes@216.251.149.66)
20:16.08[TK]D-Fender;)
20:16.09stack_in extensions.conf, "include" lets you specify a time for the include.  If I have an include for a holiday and an include for normal operations, and the holiday falls on a normal operation day, do both includes get used?
20:16.15SocialDOh yeah I got that distro round here somewhere
20:16.33iCEBrkrSocialD: I'm not even sure if that's enough horse power to do all the codec/transcoding in Asterisk :)
20:16.36fugitivoSocialD: did you try lfs? (linuxfromscratch)
20:16.49SocialDI've gotten asterisk on there before
20:17.01SocialDI just don't know if its gona do anything else then CLI>
20:17.45[TK]D-FenderYeah, if you're hard-core Linux, LFS would be perfectly fine choice...
20:17.46stack_if you have the time
20:17.46SocialDIm not that hard core sorry
20:17.46[TK]D-FenderSlackware is jsut sort of an easy way out.
20:17.46fugitivoyou don't need much time
20:17.46fugitivoless than gentoo
20:17.46SocialDAlright i'll go install it, be back in an hour if im still alive
20:17.49fugitivobecause you don't install crap :)
20:18.42[av]banihttp://www.voip-info.org/tiki-index.php?page=AstLinux  \o/ ?
20:19.32justnullingwhat is this error Auto-congesting call due to slow response?
20:19.33*** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com)
20:19.50*** join/#asterisk pb__ (n=pb@cpc3-cmbg6-5-1-cust33.cmbg.cable.ntl.com)
20:21.32*** join/#asterisk fdgfd (n=fdgfd@adsl-ull-16-220.42-151.net24.it)
20:21.37fdgfdhello!
20:22.12fdgfdSomeone tried to put asterisk on embedded device such an Access Point?
20:22.29[av]baniyes
20:22.55iCEBrkrI'm not sure why you'd want to do that, other than geek factor
20:22.59*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
20:23.06[av]baniiCEBrkr: solid state pbx
20:23.08_Sam--reliability
20:23.16[av]banialso very cheap
20:23.16_Sam--if you are just using a home setup, its fine
20:23.20[av]baniyep
20:23.24_Sam--people put it on the linksys wrt54g
20:23.28fdgfdyes
20:23.33[av]baniits good for 4-6 lines, especially if you use reinvite
20:23.34fdgfdI mean wrt54g
20:23.50iCEBrkrYeah cuz 100mhz is OOO so much horse power to run an office on
20:23.56_Sam--who said office?
20:23.59fdgfdI've heard it today
20:23.59_Sam--nobody but you
20:24.02[av]baniiCEBrkr: you dont need ghz to run signaling
20:24.18iCEBrkrLOL
20:24.19[av]banifdgfd: www.openwrt.org
20:24.20dijit0_does anyone else hear use idefisk and knows why the caller id will not send right if one isn't specified in idefisk itself?
20:24.24iCEBrkrYou guys crack me up
20:24.35_Sam--dijit0_:  ask ZOA
20:24.40_Sam--er he's not here, he wrote it
20:25.05dijit0_argh@! ok, thx
20:25.12_Sam--fgd:  another fun project would be to build your own solid state asterisk machine
20:25.15fugitivowhat are the "watchers" when doing show hints?
20:25.15fdgfdavbani: yes, I know it. I would like to know if It can be usefull (for you) to have a network of AP (WRT) with asterisk
20:25.25iCEBrkrSo explain to me something...
20:25.26harryvvI wonder how much faster a hardware based asterisk solution would be that over software. Say by the time a extention is pressed to the time the phone rings.
20:25.27[av]baniiCEBrkr: as long as you know the limitations, its fine. no transcoding, use reinvite where possible. works fine
20:25.27_Sam--using something like mini-itx or a something
20:25.28*** join/#asterisk batphone (n=will@69.15.174.114)
20:25.28stack_in extensions.conf, "include" lets you specify a time for the include.  If I have an include for a holiday and an include for normal operations, and the holiday falls on a normal operation day, do both includes get used?
20:25.39[av]bani_Sam--: i wanna try a gumstix :D
20:25.44fdgfdwait, I draw :-)
20:25.47*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
20:25.47batphoneanyone in here have luck using IAXYs over a VSAT link for FAX?
20:25.49_Sam--ive been looking into some sbc's
20:25.50iCEBrkrIf a 3Ghz machine has issues with timing and such.. How well is a shitty router CPU going to handle that??
20:26.04[av]baniiCEBrkr: oh yeah, no meetme :)
20:26.08iCEBrkrlol
20:26.09harryvvThere is a asterisk firmware based solution on A ZORCOM BOX.
20:26.10iCEBrkr:(
20:26.14dijit0_i am not sure whether my issue is asterisk or idefisk... but when i set a custom callerid in my iax.conf, it has no effect unless a callerid is set in idefisk, which can be any callerid, THEN the one i have in my iax.conf will work
20:26.21[av]baniiCEBrkr: home answering machine usually doesnt need meetme...
20:26.23_Sam--that xorcom box is nothing
20:26.30_Sam--my solid state boxes are nicer than that thing
20:26.31harryvvsam, have you used it?
20:26.36_Sam--no i saw the specs last night on it
20:26.38iCEBrkr[av]bani: oh, you silly guys thing so small
20:26.42[av]baniiCEBrkr:  you might as well ask "well how does cisco do it with a 200mhz mips"
20:26.49[av]banithe answer: very carefully
20:26.52trixterisnt the4 xorcom basically a rip of thevoipconnection.coms vs1, including the name
20:26.58iCEBrkr[av]bani: um, a cisco router doesn't do half the shit asterisk does.
20:27.01_Sam--http://www.xorcom.com/ts-1/features.html
20:27.08harryvvsam, have done performance test on yours?
20:27.13[av]baniiCEBrkr: for routing sip, it does
20:27.14iCEBrkr[av]bani: Which is why they got away with 386 CPUs for such a long time
20:27.34fdgfdplease look here : http://www.frascati1.org/asterisk.jpg
20:27.38[av]baniiCEBrkr: the only idfference is ciscos dont do pbx functions,b ut they route sip the same basically -- they're a gateway
20:27.42_Sam--harryvv:  no i havent...but for the same price as the xorcom running a via 1gh, i run a celeron 2.8ghz w/ 1g ram
20:27.49[av]baniiCEBrkr:  so they do signaling, transcoding, etc.
20:27.59harryvvI see
20:28.07iCEBrkr[av]bani: All I'm saying is I hear about all these time sensitive and quality issues even on a beefy machine, and now you're going to tell me that some PoS home networking router is a solution??
20:28.08[av]baniso again, the answer is: 'very carefully'. know your limitations
20:28.14harryvvsam, and for the storage you use a flash card?
20:28.19[av]baniiCEBrkr: yep
20:28.20fugitivowhere is the documentation about "watchers"?
20:28.27iCEBrkr[av]bani: Limitations.. 1 phone call...
20:28.29[av]baniiCEBrkr:  it's _a_ solution, and works very well
20:28.32[av]baniiCEBrkr: hardly
20:28.37iCEBrkr[av]bani: Duct tape is a solution...
20:28.40[av]baniiCEBrkr: read up on "reinvite"
20:28.45iCEBrkrDoesn't mean it's the RIGHT solution
20:28.56_Sam--yes, a 1G cf flsah card runs the main os and asterisk with a GUI and many utilities.../var runs from ram so hardly any writes to the CF card, and i backup with rsync all the critical data to a USB thumb drive
20:29.01[av]baniand get back to me when you're done reading
20:29.24iCEBrkruh
20:29.25_Sam--if i used mini itx it wouldnt be much bigger than 4 packs of cigarettes or so
20:29.26iCEBrkrsure dude.
20:29.29s34nThe Dial command should allow me to dial more than one number/device separated by &, true?
20:29.30harryvvsam, thats interesting :) how fast does it boot up?
20:29.42znoGdoes asterisk support distinctive ring in the SIP protocol? (apparently a certain tag can be set to let ATAs know its a diff ring)
20:29.42_Sam--harryvv:  its takes about three minutes because it has to load /var into ram
20:29.49harryvvI see
20:29.50[av]bani_Sam--: www.gumstix.org  :D~~~~~~
20:29.56*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
20:29.58znoGnever mind
20:30.00fdgfdIt's usefull for you to have WRT with asterisk in this topology?? http://www.frascati1.org/asterisk.jpg
20:30.01harryvvAbout the same time as a standars asterisk box
20:30.07[av]baniznoG: distincitve ring is dependent on the endpoints
20:30.11_Sam--its a bit slower for sure
20:30.11Drew___mv /me /dev/bed
20:30.17treki'am lost between hisax, midsn and zaphfc for hfc cards. What do you advise ?
20:30.25_Sam--it takes about 1 minute to load the 120 megs of /var into ram
20:30.28harryvvsam, and power consumption?
20:30.31_Sam--maybe a bit less
20:30.44[av]banifdgfd: should be fine, just remember you can't do any transcoding
20:31.01*** join/#asterisk Dusty_ (n=Dusty@64.89.118.139)
20:31.02_Sam--it uses maybe 50 watts max, i dont have a way to test it.
20:31.10_Sam--but the only moving parts are the chassis fans
20:31.21_Sam--so i cant imagine it draws very much
20:31.23iCEBrkr_Sam--: Sounds like this system called ShoreTel :P
20:31.27harryvvsam, borrow a Digiital volt meter with amp clamp.
20:31.45_Sam--i want to build the next one on a mini itx
20:31.51_Sam--ive built three using micro atx
20:31.51batphoneanyone in here have luck using IAXYs over a VSAT link for FAX?
20:31.53[av]bani_Sam--: gumstix uses like <1W  :)
20:32.01[av]baniyou can run it off AA batteries
20:32.05_Sam--power consumption wasnt THAT high on my list
20:32.09[av]bani:D
20:32.12[av]baniits cool though
20:32.21*** part/#asterisk Dusty_ (n=Dusty@64.89.118.139)
20:32.24fdgfdavbani: I don't need transcoding but I need "nat" function of asterisk (as an SBC) is UP. So how many call do you think a WRT and a true server can hold (to balance the network)
20:32.30Errheh, I'm not sure that most people know someone with a multimeter and amp clamp well enough to borrow one
20:32.42_Sam--i have a multimeter but know amp clamp
20:32.42[av]banifdgfd: probably maximum of 3-4
20:32.47[av]banifor wrt
20:32.49_Sam--know = no
20:32.51fdgfdok
20:32.52[av]banifor 'true server' probably 100's
20:32.53Erryes, me too
20:32.58fdgfdand a 3ghz server?
20:33.05*** join/#asterisk _dusty (n=Dusty@64.89.118.139)
20:33.05[av]bani3ghz? 100's probably
20:33.19fdgfdok and, sorry, the last question
20:33.21[av]banii think someone tried the other day and hit 500
20:33.24iCEBrkr_Sam--: Have you looked into Soekris
20:33.31fdgfdif I want "101"
20:33.32_Sam--ice;  yeah, the CPU power is too limited
20:33.35[av]baniiCEBrkr: soekris is rather limited cpu
20:33.41fdgfd(more than one server :P )
20:33.43iCEBrkrCompared to?
20:33.43[av]baniiCEBrkr:  a gumstix is fater!
20:33.49_Sam--compared to other small computers
20:33.49iCEBrkrReally?
20:33.50[av]banifaster
20:33.53[av]baniyep
20:33.56fdgfdI must do something like DNS load balancing?
20:33.57[av]baniits pretty slow actually
20:34.01iCEBrkr[av]bani: You called the gumstick fat! LOL
20:34.07fugitivogod, asterisk even runs on a nokia 770
20:34.10[av]banisoekris is mainly intended for APs
20:34.14_Sam--there are some cool SBCs
20:34.18fugitivoand it's a crappy cpu
20:34.21_Sam--that could do asterisk in about the size of a pack of smokes
20:34.26_Sam--and be at least 1/2 decent
20:34.31fugitivo(www.nokia.com/770)
20:34.34*** join/#asterisk Simon-_ (i=byte@2001:4bd0:1000:1:2e0:4cff:feed:1cfb)
20:34.55harryvvwho here has had the most reliable * system running a year strait?
20:35.06[av]banihttp://www.gumstix.com/spexboards.html
20:35.17_Sam--los415:34:36 up 329 days,  3:08,  3 users,  load average: 0.77, 0.77, 0.83
20:35.20iCEBrkr_Sam--: I wonder if you could run something like that and rsync changes from a 'master' asterisk server.
20:35.22_Sam--not quite a year yet
20:35.35fugitivo_Sam--: show uptime inside the CLI
20:35.37_Sam--iCEBrkr:  you wouldnt need rsync if you use sql replication and realtime
20:35.56*** join/#asterisk benjk (n=benjamin@66.89.140.136.ptr.us.xo.net)
20:36.09iCEBrkr_Sam--: Wouldn't that be a bit of traffic?
20:36.10*** join/#asterisk saftsack (n=saftsack@p54A7CB78.dip.t-dialin.net)
20:36.15[av]banifdgfd: its unlikely given 1.2.3 and such :)
20:36.16malverian[work]Wow.. the Festival OGI voice models are really nice
20:36.23malverian[work]Like.. suprisingly good.
20:36.29iCEBrkrmalverian[work]: You found some good voices?
20:36.30_Sam--my solid state boxes run sql / realtime fine
20:36.39malverian[work]iCEBrkr, Yeah.
20:36.49malverian[work]Listen to the demos here (non-commercial only .. like mbrola)
20:36.59[av]bani_Sam--: you're running from ram, mainly :) of course they run fine
20:37.00iCEBrkr_Sam--: What kinda call volume can it handle tho?
20:37.09malverian[work]iCEBrkr, http://cslu.cse.ogi.edu/demos/ttsdemos.htm
20:37.43fdgfd<[av]bani>: sorry I don't understand ! :( I mean: If I need more power, asterisk give me something to make 2 server asterisk work as it was one? (scalability)
20:37.48_Sam--iCEBrkr:  i wouldnt want to route a quad pri card over it, but the ones ive built (all custom) could easily do 2 PRI
20:38.00_Sam--it depends on the customer needs
20:38.16[av]banifdgfd: not really, asterisk doesnt do clustering yet :))
20:38.22_Sam--but my lowest grade hardware so far has been celeron 2.6ghz / 512ram
20:38.42iCEBrkr_Sam--: There's a system my friend sells called ShoreTel. They have 'switches' that route calls apparently.. They're all solid state like your setup
20:38.54*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
20:39.02iCEBrkr_Sam--: I'm just trying to find a way to duplicate that
20:39.06[av]banias long as you dont transcode, you shouldnt need much cpu at all. its just routing packets
20:39.07fdgfdavbani: thank you for your precious help!
20:39.09*** join/#asterisk Insanity5 (n=feaw@ip68-111-5-23.sv.om.cox.net)
20:39.19[av]baniuse reinvite as much as possible and youre not even routing anymore, just signaling
20:39.53iCEBrkr[av]bani: I thought you coudln't reinvite behind NAT?
20:40.05*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
20:40.24_Sam--iCEBrkr:  i dont know, but it seems like shoretel is just a pbx type system
20:40.32_Sam--im not sure what they do that yo couldnt do with *
20:40.38iCEBrkr_Sam--: It is, but it's using SIP phones now and it's all VoIP
20:40.44_Sam--same here
20:40.48iCEBrkrExactly.
20:41.05_Sam--i could sell a unit with fxo though
20:41.08_Sam--if you wanted one :)
20:41.13iCEBrkrFrom what I saw, Asterisk is pretty damn close to what ShoreTel can do..
20:41.35iCEBrkrBut ShoreTel has a 'master' machine which tells the 'switches' how to work.
20:42.03*** join/#asterisk areski (n=areski@36.Red-83-55-99.dynamicIP.rima-tde.net)
20:42.09_Sam--how is that any different than , say, registering your remote asterisks to your main asterisk server?
20:42.13iCEBrkrI don't know much more about it than that, so it could be doing something really simple like [av]bani keeps saying about just using reinvite and routing
20:42.18justinuasterisk seems to hit a bandwidth wall of about 15mbit/s when "routing" rtp
20:42.20justinuno transcoding
20:42.36iCEBrkr_Sam--: Apparently the main ShoreTel box could go down and the switches still work.
20:42.36harryvvWho has used a voip whosale provider that has been nearly flawless in performance over the last year?
20:42.52*** join/#asterisk [ToTo] (n=ToTo@host72-146.pool872.interbusiness.it)
20:43.15_Sam--in that way, it does seem like using mysql replication would work
20:43.20Insanity5How would I record phone calls only to-from an extension?  I know how to record all incoming/outoging calls.
20:43.25_Sam--so that if the main * went down, the remotes would still have configs
20:43.34iCEBrkr_Sam--: Yea, that's what I was getting at with the whole rsync idea..
20:43.40iCEBrkrhrrrm
20:43.51_Sam--i use replication for a bunch of other things...never thought it about it for that
20:44.04iCEBrkrThere ya go
20:44.25_Sam--when i get up to 3 million minutes and get on level 3 with servers in 2 places, you will have solved one of my problems :)
20:44.29harryvvI guess the idea of a ALMOST reliabile whosale voip provider is a myth
20:44.36QbYanyone know of a good example of asterisk and ser?
20:44.52iCEBrkr_Sam--: haha cool
20:45.12iCEBrkr_Sam--: My friend just likes to diss opensource/linux projects.. So I need ammo once in awhile.
20:45.24harryvvSam, are you a installer, end user provider or both?
20:45.31_Sam--iCEBrkr:  what do you do down there in tampa?
20:45.36saftsack_Sam--, if i want to write a post to the hylafax mailing list
20:45.37justinuiCEBrkr: what the hell? is he some kind of windoze luser?
20:45.42iCEBrkr_Sam--: I told him what I'm doing with Asterisk and he's trying to sell me a ShoreTel + SDK solution.. :-/
20:45.49saftsacki have just to write a mail to hylafax-users@hylafax.org or?
20:45.51iCEBrkrjustinu: Exactly.
20:46.01iCEBrkr_Sam--: I'm a code monkey.
20:46.03_Sam--harryvv:  at this point i am nothing more than a wannabe.....i do some small office installs and resell some service.  (voip isnt my main business endeavoer - yet)
20:46.04Insanity5iCEBrkr - IT works fine.  You pay the $$$, I won't :)
20:46.05justinuand you call him a friend? yikes
20:46.33iCEBrkrjustinu: He's actually more of a jock-Microsoft loser.
20:46.40justinulol, wrong
20:46.50iCEBrkrjustinu: He jumped on the bandwagon way back when.. When he heard computers will make you money
20:46.50Insanity5iCEBrkr - Some people say the comemrcial products are bad; I don't pretend to think so.  I just say they're an very bad value :)
20:47.01harryvviCEBrkr down there boy... I worked at micrsoft
20:47.03justinuheh
20:47.12iCEBrkrjustinu: So, he's not even a computer geek, he just knows the stuff he learns without any of the technical knowledge of how/why it works that way
20:47.23[TK]D-FenderGET HIM!!!!!!
20:47.25iCEBrkrharryvv: sucks to be you
20:47.52harryvviCEBrkr yea? It was a good income stream for me then :) but its all a permatemp enviroment.
20:47.55iCEBrkrjustinu: Of course I call him my friend. We drink and throw darts and ride motorcycles!!
20:47.57justinuyeah - i've known people like that
20:47.59Insanity5Ok guys, how do I "Monitor" calls originating from a single extension only?
20:48.08iCEBrkrharryvv: Hey, whatever pays the bills!!
20:48.15saftsack_Sam--, i dunno how2 write to the list, sry :(
20:48.17_Sam--iCEBrkr:  what does he ride? :)
20:48.19*** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net)
20:48.20harryvviCEBrkr Thats ALL that I care.
20:48.23iCEBrkr_Sam--: One guess :P
20:48.35[TK]D-FenderInsanity5 : Just shove the monitor only on secions of your dialplan limited to that phone.
20:48.42harryvvTo many arogant IT people find out thay will get layed off then feel like a ass.
20:48.42_Sam--hmm...if he is a money guy, goes with corporate, non-opensource...
20:48.49_Sam--i have to say a honda cbr1000rr
20:48.52iCEBrkr_Sam--: A Harley...  at the moment.. But he's been eye'n a 'Busa
20:48.52iCEBrkrLOL
20:48.58NetgeeksI don't know alot about shoretel, but look at http://www.netgeeks.net  It is an asterisk based system that provides N+1 node clustering and expansion by adding a box or boxes
20:49.27Insanity5[TK]D-Fender - I guess I don't know how to do that.  All I have is a [default] with setcallerid and dial.  I can monitor there, but it catches everyone.
20:49.56[TK]D-FenderInsanity5 : You need to make a seperate context or something then, or seperate the functionality with CALLERID checks, etc.
20:50.00iCEBrkrNetgeeks: Shameless plug
20:50.01harryvvHeard of a IT manager that had this arogance by saying to the employees at the company"Im a GOD, I am the network administrator" in this one company..he got layed off 3 weeks after starting
20:50.26Insanity5[TK]D-Fender - Yikes.  I understand how to seperate with incoming (simply put under the incoming extension), but outgoing, where would I start?
20:50.43Insanity5harryvv - Arrogant network admins are the worst.  Laid off or fired?
20:50.47iCEBrkrharryvv: Those kind of people typically don't know their ass from a hole in the ground...
20:50.48NetgeeksiCEBrkr: shameless yes, plug maybe, answer to the guy who was looking for options, definately
20:50.53[TK]D-FenderInsanity5 : Make a whole new context jsut for that phone and shove the monitor stuff in there.
20:50.55iCEBrkrNetgeeks: lol
20:51.20[av]banihttp://lestblood.imagodirt.net/archives/106-Asterisk-on-OpenWRT-part-2.html
20:51.23[TK]D-Fenderkillall -9 iCEBrkr
20:51.30[TK]D-Fender:O
20:51.36synthetiqanyone know how many call set ups per second asterisk can handle?
20:51.40harryvvInsanity5 in this one case fired. I knew a few years ago I would spend time in the #windows channel of efnet. There were a few of them there.
20:51.43NetgeeksI'm not ready to just shamelessly plug the system, as it's still in beta testing and the integrated billing system is still a week or two away from beta testing quality
20:52.06Insanity5harryvv - I've actually tried to get help in there once wih complex windows problems.  What a waste of time.
20:52.18harryvvInsanity5 did thay put you down?
20:52.27Insanity5harryvv - No community support, whatsoever.  The best you can hope for it newsgroups
20:52.34harryvvI know
20:52.38Insanity5harryvv - Smart-ass one-off we don't want to help remarks.
20:52.46harryvvyup
20:52.48harryvvI know
20:52.51[av]banisynthetiq: i heard someone the other day saying they had 50/sec or so
20:52.55Insanity5harryvv - And it was a very complex issue.  filters with file replication services cross forest over a one way trust.
20:53.01Insanity5harryvv - Yuck, any way you look at it :)
20:53.07harryvvI even said I worked at microsoft at that time and was kicked off the #windows channel.
20:53.12synthetiqbut SER can only do 20
20:53.29synthetiqand SER has less overhead
20:53.36harryvvthen asterisk?
20:53.37Insanity5harryvv - The MVP's in the newgroups are usually pretty helpful though.
20:53.45harryvvAhh
20:54.25Insanity5[TK]D-Fender - where would I start for outbound filtering?  i made the new context, but how to I direct my outgoing calls to go through that, without saying, implementing a long-distance access code like system.
20:54.35harryvvInsanity5 so you do alot of asterisk support on the side?
20:54.44[TK]D-Fenderwhat are you filtering for?
20:54.48Insanity5harryvv - Nope.  No production asterisk on my company :(
20:54.54harryvvahh
20:54.55Insanity5[TK]D-Fender - origination from a certain SIP device.
20:54.57*** join/#asterisk arcy (n=arcanum@ppp45-adsl-90.ath.forthnet.gr)
20:54.58harryvvwhy not!
20:54.59harryvv:)
20:55.02Insanity5harryvv - My home asterisk box, lol
20:55.08harryvvk
20:55.13Insanity5hardwire - they pay too much money for a commercial solutioon.
20:55.17Insanity5err, that was for harry
20:55.23[TK]D-FenderInsanity5 : Well rigth before your dial lines maybe shove in some GOTOIF's based on the callerid of the phone originating the call...
20:56.15*** part/#asterisk austinnichols101 (n=austinni@70.46.69.130)
20:56.15harryvvInsanity5 how much does your company pay?
20:56.25Insanity5harryvv - I don't even know.
20:56.33*** join/#asterisk infobox (n=dpizarro@libra.infostar.com.pe)
20:56.42Insanity5harryvv - I do love the software, and it is something asterisk is missing, big time.
20:56.45Insanity5harry - http://www.sipcenter.com/sip.nsf/html/Sponsors+Interactive+Intelligence
20:56.46justinuone billion dollars
20:56.55*** join/#asterisk MatsK (n=mk@cC30123C5.inet.catch.no)
20:56.58Insanity5The outlook integration, transfer, record, conference, etc, from the PC works great.
20:57.03infoboxhello
20:57.06infoboxPlease
20:57.27Insanity5infobox - hi :)
20:57.28hypnoxhmm, dundi looked-up calls seem to be using GSM codec only, i am not sure where to override this. dundi.conf ?
20:57.28infoboxI have a big problem making a ISDN/FXS gateway
20:57.42Insanity5[TK]D-Fender - I guess if then's would work.  I can see that getting real messy though :).
20:57.47infoboxhi Insanity
20:57.56harryvvInsanity5  would have any kind of voip domain name registered by now but 99.9% of any domain i type in...even some strange names are used.
20:57.57Insanity5hello :)
20:58.12_Sam--bkw_ you around?
20:58.14Insanity5hardwire - What do you mean?
20:58.16*** join/#asterisk kpettit (n=keith@69.15.174.114)
20:58.18*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
20:58.19kpettitlo
20:58.27kpettitAnother day, another version of asterisk :)
20:58.27infoboxI am using TDM2406B and TE110P boards
20:58.38hardwirequit it
20:58.41hardwiretab tab
20:58.45hardwirethats all I gotta say
20:58.47*** join/#asterisk sevard (n=kynan@198.174.233.25)
20:58.54harryvvinfobox how has those cards been working out for you
20:59.08fugitivoinfobox: how does the tdm2400 perform?
20:59.38sevardgrr I can't figure this out at all. asteriskcdrdb is freaking empty.
21:00.06infoboxI can not configure them successfully
21:00.10kpettitAnybody have any experice transfering calls with AMI?
21:00.24infoboxI am using a PIV 2.66 Ghz with 1GB rAM
21:00.32fugitivoinfobox: why not?
21:01.14Insanity5harryvv - So, are there any good win32 front ends for asterisk?  It should be able to conference, transfer, record, outlook integration with voicemail, etc.  And it shouldn't be some clunky Java thing that is a pain to set up :).
21:01.32Insanity5harryvv - I kind of doubt that it's possible without a proprietary hook-in to asterisk though :(.
21:01.37znoGif only there was a "remote" way to reboot an ATA
21:01.38infoboxfugitivo: the zaptel cannot load
21:01.46Netgeeksthere was that .NET app someone wrote....  I foget the name
21:02.16sevardwhy the heck is my asteriskcdrdb empty graaaa
21:02.18NetgeeksIPSwitchBoard
21:02.24Netgeekssearch for it
21:02.46infoboxI put the TDM240X boards on PCI's number 2 and three
21:03.11infoboxI connected them to power connectors
21:03.18*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
21:03.53infoboxmayeb someone has exprience using both models of boards (TDM2406 and TE110P) ain the same pc ???
21:05.07libilaanyone know what the cause of this is? ZT_CHANCONFIG failed on channel 1: No such device or address (6) I googled and someone moved their PCI card from the top slot (next to agp) to a different one and that error stopped, I tried that but it didn't work.
21:05.19Insanity5Netgeeks - That's pretty far on the clunky side last time I poked around with it :P
21:05.47Insanity5Netgeeks - Required lots of custom seutp anyways.
21:06.15infoboxhas anyone  intent to built a ISDN E1/30FXS gateway?
21:06.16NetgeeksInsanity: I never played with it myself, so I can't say for sure, just remembered it was a windows app that supposably did alot of nice stuff with an asterisk install
21:07.07Insanity5Netgeeks - It will, but it's got a little ways to go.  It's probably help if asterisk had a method of hooking into it other than SIP for stuff like that.
21:11.53*** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com)
21:12.11QbYwow..  Asterisk is EVERYWHERE..
21:12.28NetgeeksOh, how I love cascading style sheets..... *puke*
21:12.40QbYI just talked to a wholesale origination and termination provider..  and they are using Asterisk too..  interesting..
21:13.00*** join/#asterisk Defraz_ (i=t0tal@tim.mychoice.cc)
21:13.14*** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
21:13.24*** join/#asterisk wunderkin- (i=kev@ip68-226-113-228.ph.ph.cox.net)
21:13.52_Sam--QbY:  who did you talk to
21:13.55*** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-3-0-cust141.bagu.broadband.ntl.com)
21:14.01*** join/#asterisk Igbothom (n=HiltonT@office.quarkit.com.au)
21:14.07QbYi3 Networks
21:14.19_Sam--havent heard of em yet
21:14.22*** join/#asterisk Little-L_ (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk)
21:14.46*** join/#asterisk dijit0 (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net)
21:15.02QbYthey are a Master Reseller and VAR for Level 3
21:15.18_Sam--there are only a few real level 3 partners
21:15.22_Sam--like 8x8 and delathree
21:15.42_Sam--deltathree rather
21:15.51*** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it)
21:15.56iCEBrkr_Sam--: Damn you.. Now I wanna build a solidstate asterisk box like you.
21:16.05_Sam--iCEBrkr:  do it, i dont have any patents...yet :)
21:16.09iCEBrkrhehe
21:16.15_Sam--QbY:  they are not listed as an master reseller
21:16.19_Sam--http://www.level3.com/userimages/dotcom/microsites/MasterResellers/
21:16.29iCEBrkr_Sam--: I wanna build it in a 1U
21:16.46_Sam--i found some nice 1u mini itx cases
21:16.55QbYi3 Networks is listed as a Master
21:17.07_Sam--give me a url
21:17.15QbYthe one you just sent me
21:17.22_Sam--oh hah!
21:17.25QbYi loaded it and see i3networks, listed just below Delta Three
21:17.28_Sam--guess it must be close to quitting time
21:17.33_Sam--you are indeed correct!
21:17.35QbYhehehe
21:17.39[av]bani_Sam--: i'm still looking for nano-itx ...
21:17.53_Sam--ravenpi:  i thought there is one or two boards out
21:18.27_Sam--QbY:  i called lvel3 yesterday to find out what you need in order to get to resell their service
21:18.34justinumoney
21:18.35_Sam--i am only about 2.9 million minutes short
21:18.37justinulike everything
21:18.43[av]banii have yet to find anyone selling any nano-itx boards, only cases(!)
21:18.51*** join/#asterisk JohnJacob (n=m00p@pool-71-127-74-138.aubnin.fios.verizon.net)
21:19.06NetgeeksSam: 3 million minutes required?
21:19.08_Sam--justinu:  who else besides l3 provides that type of reseller stuff?
21:19.10*** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com)
21:19.17_Sam--netgeeks 3mil for termination, 2.5 mil for orig
21:19.25justinui think global crossing
21:19.25_Sam--roughly 50K a month a commitment
21:19.50NetgeeksXO as well
21:19.57QbYXO doesn't do origination and termination
21:20.23_Sam--i think with my current volume i am stuck reselling other 2nd tier services like deltathree and 8x8
21:20.27justinuwe're pretty stoked with level3's quality so far
21:20.37justinusam: try tmccom.com
21:20.39NetgeeksXO doesn't do orig and term, or XO doesn't do both?
21:20.48QbYneither
21:20.54QbYper my last agent meeting
21:20.57*** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at)
21:21.12_Sam--ty for the advice
21:21.45Netgeekshrm, wonder if they have a hidden product.  One of the companies I do consulting work for has a XO contract for origination
21:22.04Martincit0Advice pls: Is there any way to run a system call (i.e. perl script) and catch the returning value (from the dialplan)? Thanks
21:22.05znoGahh would be sooo neat to send some command to a Sipura/Linksys unit to force reboot
21:22.26Netgeekshrm, maybe res_perl
21:22.37Martincit0thanks
21:22.59Netgeeksis that even in stable?  res_perl?  or just head?
21:23.02*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
21:23.25sevardWhat wakeup agi does everyone use?
21:23.39_Sam--i use the agi on my cell phone to make some noise and wake me up
21:23.46_Sam--its called a clock
21:23.58libilaI'm getting an error when doing ztcfg, could someone take a look at it? http://tinyurl.com/csh3v
21:24.06Assidis there a way to set a timezone manually for a call?
21:24.18*** join/#asterisk insomni_ (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk)
21:24.18_Sam--<sorry couldnt help it> you make it seem like everyone is doing wake up calls (maybe they are and i dont know it)
21:24.27*** join/#asterisk PSC (n=chatzill@gateway.imtco.com)
21:24.29sevard_Sam--: nice.
21:24.48_Sam--you saw the wake up scripts under the "Tips and Tricks" section of the wiki?
21:24.50sevardI do have a working wakeup php script but currently it plays "Music on hold" for the wakeup call, which.. isn't so great.
21:24.57sevard_Sam--: yes, there are a few.  I'm trying them out.
21:25.46znoGsevard: my wakeup script swears at me in the morning :)
21:25.53_Sam--iCEBrkr:  when you build your solid state box and want a CF image to run on it, let me know.
21:26.04_Sam--i am a bit slacking with it, i still have to get it to kpettit
21:26.21*** part/#asterisk diclophis (n=diclophi@Sac-12-201.cisdata.net)
21:26.23sevardi really like another wakeup AIG that the tips and tricks suggests but unfortunatly it doeesn't like to call the user, i watch the call being placed but it doesn't go through
21:26.39_Sam--my image has apache, asterisk, mysql 5, x windows w/ fluxbox, mozilla, and some other stuff that i forget right now
21:26.40iCEBrkr_Sam--: It's not going to be any time soon.  I'm in the middle of restoring my bike and well, having too hobbies is a bit expensive :P
21:26.57*** join/#asterisk PSC (n=chatzill@gateway.imtco.com)
21:27.02_Sam--but its fully featured, even for development asterisk work
21:27.08iCEBrkr_Sam--: I'm thinking I won't have a GUI/Desktop.  It's going to be more of an appliance
21:27.37_Sam--for the extra space required to have a real webbrowser, and that enviroment,...and with how cheap CF is...its well worth it!
21:27.59_Sam--my stuff is still an appliance...but if it had a problem..and i needed to go onsite, i would be self contained
21:28.03iCEBrkr_Sam--: But why? My 'solution' is to have a 'switch' much like a hub or smart switch
21:28.07iCEBrkrNo need for an interface.
21:28.08_Sam--with a web browser, and all the other tools i need to fix it
21:28.18iCEBrkrmaybe a web server on there to serve up web pages for configuration stuff
21:28.21[av]bani_Sam--: x windows? O NOES DONT YOU KNOW X KILLS ASTERISK
21:28.26_Sam--no need for an interface, until you need one
21:28.42iCEBrkr_Sam--: Who needs prety when you have bash? :P
21:28.43_Sam--it doesnt startx unless you log in :)
21:28.47iCEBrkrs/prety/pretty
21:29.05_Sam--sometimes you need a web browser or GAIM
21:29.10iCEBrkrlol
21:29.18kpettit_Sam--, that's alri8ght man.  I'm keeping busy with all these Asterisk updates
21:29.24*** join/#asterisk signaleleven (n=evan@lion.ragga-jungle.com)
21:29.32iCEBrkr_Sam--: You shouldn't be surfing the web from your Asterisk box!! :)
21:29.35_Sam--there is no downside to having those tools available
21:29.46iCEBrkr_Sam--: lynx works just fine.
21:29.48_Sam--i surf the web when im doing development asterisk work all the time
21:29.51dlynespeople actually use X on an asterisk box? :)
21:29.59_Sam--lynx doesnt work jsut fine.
21:30.01SocialDHow do I set up asterisk, so it detects what port my modem is on, so I can dial into my house
21:30.03iCEBrkrlynx + wget.  What more do you need?
21:30.09_Sam--not anymore, with the complexity of webpages
21:30.24dlynessam: links...does frames and javascript
21:30.30kpettitJust got my gentoo build read for 1.2.4 and zaptel 1.2.3
21:30.34iCEBrkr_Sam--: Well, the end result wouldn't have the overhead of a GUI on it..
21:30.39kpettitman I love ebuild's
21:30.45_Sam--i have no overhead of a gui except disk space
21:30.47synthetiqsam needs flash and graphics to give you a siezure
21:30.49_Sam--which is CF card space
21:30.53[TK]D-FenderI run X, KDE, Samba, Apache, FTP, and plenty more on my server and it seems to work just fine....
21:30.54_Sam--which is cheap.
21:30.57iCEBrkr_Sam--: even then, I surf from my work station, and cut-n-paste URLs into wget
21:31.02Netgeeksah, i hate gentoo  ;0
21:31.06[av]banii usually just take fedora and cut it down to barebones
21:31.07_Sam--iCEBrkr:  these are devices that deployed at customer locations
21:31.12_Sam--when its breaks, and when i show...
21:31.17_Sam--i can be fully self contai9nted on my appliance
21:31.21_Sam--without needing to use their pcs
21:31.32iCEBrkr_Sam--: If it breaks at a client location, you take a new one out there and bring the old one back.
21:31.34[av]banithinking of trying a cut down ubuntu though
21:31.35fugitivo[av]bani: isn't easier to lfs insted of cutting down fedora?
21:31.45_Sam--if it breaks, i bring a new CF card out there,r eally.
21:31.50_Sam--but i can still have tools at my disposal
21:31.51_Sam--to fix it
21:31.54iCEBrkr_Sam--: There ya go :P
21:31.56[av]banifugitivo: not really, it takes more work to 'build up' lfs than it does to cut down fedora
21:32.15_Sam--there is no downside to having the tools available, if you have the space
21:32.15fugitivo[av]bani: hmmm
21:32.16iCEBrkr_Sam--: Thoughts of network devices can't be wrong.
21:32.16_Sam--that is my personal opinion
21:32.20[TK]D-FenderAnd then we tried Slackware and it was JJUUUUSSSSTTT right!
21:32.25fugitivo[av]bani: i think it's impossible to cut down fedora :)
21:32.25_Sam--i have a 1G card, might as well use the space
21:32.32iCEBrkrerrr Thousands of network devices...
21:32.37iCEBrkrIf I could type, I'd be dangerous
21:32.48_Sam--running asterisk is a bit different then running say an 8 port hub.
21:32.48[av]banifugitivo: no, i got it trimmed down to <512m
21:32.55[av]banistill a bit large though
21:33.07[av]banii'm thinking ubuntu shold be able to trim down more
21:33.11iCEBrkr_Sam--: Why tho? Why's it gotta be different?  The idea is to plug the shit in and it just works..
21:33.15tzanger512M?
21:33.18tzangerI have slackware in 52M
21:33.30[av]baniyeah but ... slackware ... :(
21:33.30_Sam--the idea behind a solid state asterisk box is not plug and play
21:33.30signalelevenwhen I call Dial from an AGI application if the called party hangs up then I get ANSWER for dialstatus but if the caller hangs up then I get noresponse... how can I get a ANSWER (and more importantly DIALEDTIME) when the caller hangs up?
21:33.32tzanger[av]bani: slackware is teh win
21:33.34_Sam--the idea behind it is plug and run for a while.
21:33.37fugitivo[av]bani: i have a custom lfs + asterisk + a lot of stuff with <120mb
21:33.37signalelevenanyone know?
21:33.39[av]banitzanger: bring on those floppes!
21:33.43[av]banifloppies
21:33.48fugitivo[av]bani: took only one day of work
21:33.51tzanger[av]bani: oh please
21:34.02[av]banitzanger: my first linux was slackware...
21:34.11tzanger[av]bani: my linux is still slackware
21:34.11[av]banidownloaded over 14.4k modem
21:34.16fugitivoi think everybody's first linux was a slackware
21:34.16[av]banii moved on :)
21:34.17tzangerI've tried all the others, slackware just works for me
21:34.37iCEBrkrI really need to try LFS
21:34.47fugitivo[av]bani: same here, no packages at all, that was good :)
21:34.55tzangerlfs was good if you have a specific need in mind (I was making CF-based firewalls) but for a general distro it's got no advantage
21:35.05fugitivoagreed
21:35.11fugitivoi use it for my asterisk servers only
21:35.11iCEBrkrtzanger: Yea, for sure
21:35.14fugitivoCF
21:35.23[av]banitheres still use for auto-updating and package management, even on tiny embedded systems
21:35.27[av]bani(eg, openwrt, ipkg)
21:35.30tzangerI just use slackware ofr 'em.  I can pare it down for what I need, and it just goes
21:35.30*** join/#asterisk clive- (n=pirch@dsl-165-149-246.telkomadsl.co.za)
21:35.38tzangerfor the embedded stuff I'll be rolling my own but that's hardly a distro then :-)
21:35.38[av]baniactually a mangled openwrt for x86 would be almost idea
21:35.51[av]banisince you can get openwrt into 4mb :)
21:35.56*** join/#asterisk miguel3239 (n=myoung@h-68-167-124-170.cmbrmaor.covad.net)
21:36.15[av]baniand it has a package manger, and nice build system
21:36.21fugitivoi see (and talking with my experience) harder to cut down a distro than building a linux from scratch
21:36.24*** part/#asterisk miguel3239 (n=myoung@h-68-167-124-170.cmbrmaor.covad.net)
21:36.33tzangerI agree with that
21:36.45*** join/#asterisk miguel3239 (n=myoung@h-68-167-124-170.cmbrmaor.covad.net)
21:36.47_Sam--im not sure about that
21:36.57_Sam--it was easier for me to start with debian base than from scratch
21:37.07_Sam--adn add/remove packages that were needed using synaptic
21:37.28_Sam--i started from scratch
21:37.30fugitivo_Sam--: how many mb did you reach?
21:37.43_Sam--how many COULD i have done it in?
21:37.47_Sam--or how many am i using?
21:37.47fugitivothe problem is that a base distro still have a lot of crap
21:37.57iCEBrkr_Sam--: So you use debian as your model until you figured out exactly what was needed?
21:38.00_Sam--CF cards are cheap...i dont care if i have to buy 512 vs. 256
21:38.29_Sam--no, debian is the base.
21:38.33_Sam--you can still apt-get stuff
21:38.38iCEBrkrAhh
21:38.51fugitivobut you don't really know what you have in your system
21:39.06_Sam--in terms of what?
21:39.12_Sam--i know which packages and libraries are installed
21:41.11libilaanyone know what ZT_ChANCONFIG failed on channel 1 means? http://tinyurl.com/csh3v is the complete error.
21:41.16*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)
21:42.03signalelevenanyone know why dial returns noreponse from an AGI app when the caller hangs up but not when the callee does?
21:42.26*** join/#asterisk zotz (n=zotz@24.231.47.175)
21:43.52clive-signaleleven, is that like for astcc ?
21:44.03iCEBrkr<VOICE TYPE="Cartman"> Screw you guys, I'm going home </VOICE>
21:44.12signalelevenclive-: no, it's a custom AGI
21:44.42signalelevenclive-: but similar purpose, if the callee hangs up then everything's cool, if the caller does then I get noresponse for all the Dial vars
21:44.43mzo_ii wish that would actually work
21:44.50clive-signaleleven are you writing it in perl or php ?
21:44.57signalelevenclive-: perl
21:45.02sevardi'm editing the crap out of wakeup.php
21:45.08sevardi just changed the menus so they're more helpful
21:45.09*** join/#asterisk PupenoL (n=pupeno@200.123.183.89)
21:45.27sevardwhat would be better than 'music on hold' which is generally soft and not very much like WAKE UP, DUDE for a wake up call?
21:45.38fugitivosevard: publicity
21:45.45*** join/#asterisk PSC (n=chatzill@gateway.imtco.com)
21:45.51sevardfugitivo: ..what?
21:45.54signalelevenclive- and the strange thing is that on the console when the caller hangs up you get a message saying that it exited non-zero
21:46.20clive-signaleleven, im no expert, but did you try "deadagi"
21:46.35fugitivosevard: promotions, advertising, etc
21:46.48signalelevenclive-: no, I saw reference to it but haven't played with it yet... I'll give it a try.
21:46.55sevardfugitivo: would you be very happy if you got an advertisement on your wakeup call?
21:47.02clive-I think that may solve it for you
21:47.13fugitivosevard: no, but if I run the hotel i will
21:47.28AgilixSupportOk ... newbie question:  what is VoicePet?  Is it just an old .tar.gz of asterisk with some install scripts?  Anything unique?
21:47.41signalelevenclive-: indeed you're right... thanks :)
21:48.02*** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM)
21:48.13[av]bani[TK]D-Fender: around?
21:48.20sevardfugitivo: I guess a custom mp3 would be the best.  This is your schedualed wake up call.  After you've gotten ready for the day why not try our breakfast menu in the lobby.  The weekly specials are only $4.00 and are better than your sister.
21:48.35justinu4 dollars? that can't be a hotel
21:48.38justinuit'd hae to be 40
21:48.39sevardheh
21:48.44fugitivosevard: that's great
21:49.17fugitivosevard: with feng shui music the hotel will increment sales
21:49.20sevardI wouldn't be exactly sure on how to play a custom mp3, what would I change this to
21:49.21sevard$parm_application = 'MusicOnHold';
21:49.42sevard$parm_application = 'mpg123 /path/to/file.mp3'; ?
21:50.17*** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com)
21:52.20[TK]D-Fender[av]bani : yup
21:52.27[av]bani[TK]D-Fender: you have a sangoma 104d right?
21:52.31[TK]D-Fenderyup
21:52.42[av]baniwith the hardware EC?
21:52.45[TK]D-Fenderyup
21:52.50[av]banihow is the EC
21:52.57[TK]D-Fender<- man of many words... or was it 1 word many times...?
21:53.12[TK]D-Fender[av]bani : *0* echo.  period.  Ever.
21:53.17[av]baniyay?
21:53.20*** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com)
21:53.20[TK]D-Fendernigh-Godly.
21:53.33[av]baniso the a200 should be good since it uses the same EC
21:53.49[av]baniof course, just because it works good on digital doesnt mean it will work well (or at all) on analogue :)
21:54.05[TK]D-FenderYUP!  A200 = TMD400/2400 killer in vast majority of scenarios IMO.
21:54.29*** join/#asterisk KrIS83 (n=kris@p549B2226.dip0.t-ipconnect.de)
21:54.38[TK]D-Fender[av]bani : The A200 is build on their AFT card and uses the same DSP.  Basically its like an A104d + Channel bank all in one
21:54.42[av]bania104d: G.168-2002 EC with 128ms tail
21:54.48[av]banispa-3000: G.165 EC with 8ms tail
21:54.50[av]baniheh...
21:55.00[TK]D-FenderThats G.168-2002 at 100% on all densities...
21:55.07*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
21:55.09[av]baniyes... 'godly ec'
21:55.18[TK]D-FenderLook at the degrade rate on the TDM2400....
21:55.21[av]banihow much did the 104d cost?
21:56.01[TK]D-FenderMe?  *0*.  My reseller burned some of his rep on our troubles with 2 x TE405P's which we ditched for the A104d very soon after its release.
21:56.04_Sam--sorry to chime in...is the 104d a 4 port FXO?
21:56.13[av]bani_Sam--: 4 port PRI
21:56.15[TK]D-Fender_Sam-- : No, a 4 Port T1
21:56.19_Sam--damn i see
21:56.30[TK]D-Fender_Sam-- : You'd be thinking of their new A200 series
21:56.38[TK]D-Fender2-24 port
21:56.43[av]bani[TK]D-Fender: not his fault though! i bet he doesnt sell te405p's anymore though :)
21:56.58[TK]D-Fender[av]bani : Actually yes so far....
21:57.06[av]banibig mistake :)
21:57.10[TK]D-FenderWe just had so many problems with them that I don't care where they end up :)
21:57.43[TK]D-FenderBest part is I'm only really using 1 port on it :)
21:58.04[av]bani:)
21:58.16[av]banisangoma doesnt seem to have any EC cards < 4 ports
21:58.28[av]bania 2 port EC T1 would be nice
21:59.12[TK]D-Fender[av]bani : coming out within 3 months from what I'd been told on the inside...
21:59.19[TK]D-Fender:)
21:59.28libilais it bad to have wctdm as number 22 on the list of interrupts? (cat /proc/interrupts)
21:59.35[av]banithey should just sell a generic EC card so people with digiums can use them
21:59.54_Sam--i never realize echo was a big problem on PRIs?
22:00.08[av]bani_Sam--: usually isnt, because PRI cards usually have EC on them
22:00.23[TK]D-Fender[av]bani : I has to be tightly integrated to be effective.  Extra latency would be problematic.  Their design is fairly elegent and its YGWYPF
22:00.26_Sam--i had a PRI here with digium no EC card...never had echo problems
22:00.41[av]bani_Sam--: you're lucky then
22:00.46_Sam--but most PRIs EC is recommended?
22:00.51[av]baniyes
22:00.59*** join/#asterisk copantl (n=galel@63.245.93.138)
22:01.01clive-did everyone forget that no EC cards for asterisk existed until like a few months ago
22:01.03tzanger_Sam--: no, EC is recommeneded for any VOIP application.
22:01.04[TK]D-Fender_Sam-- : Always.  Anything that his the PSTN should have EC
22:01.24_Sam--what does the EC within the phones accomplish?
22:01.25tzangerpersonally I really love the MG2 canceller but on large systems you need a lot of horsepower to run EC on all the channels
22:01.36*** join/#asterisk RoyK (n=roy@193.80-202-93.nextgentel.com)
22:01.47tzanger_Sam--: VOIP is inherently (much) more latent than circuit-switched networks
22:01.52tzangerand echo is caused by latency
22:02.06h3xnah
22:02.13h3xecho is caused by the analog crap on the other end
22:02.13h3xheh
22:02.15_Sam--i am 100% voip, and i dont have many echo problems
22:02.39h3x_Sam-- i had a PRI here with digium no EC card...never had echo problems
22:02.40[av]baniyou should have _zero_
22:02.42h3x_Sam-- i am 100% voip, and i dont have many echo problems
22:02.43_Sam--i wouldnt go so far as to say I NEVER have echo problems, but they are really rare
22:02.46h3xthat aint 100% voip?
22:02.48rajiv|workwhat can you do if you do not have enough side tone on a sip phone?
22:03.04[av]bani_Sam--: a proper setup should have zero
22:03.15[av]banirajiv|work: yell at the vendor
22:03.30_Sam--h3x:  100% voip would be (in my own opinion) if i didnt use the digium card / PRI and routed all my traffic to a remote gateway
22:03.36_Sam--that is what i do now, and what i call 100% voip
22:03.52h3xok well your remote gateway turns it into a pstn line
22:04.01h3x100% voip would be a voip phone to a voip phone
22:04.34rajiv|worki need new phones
22:05.04_Sam--bani, so what does the EC in the phones do?
22:05.43mzo_it stops errors between the user and the phone.
22:05.58tzangerh3x: well yes, but analog will exist on any system since your mouth and ears aren't modular.
22:05.58[av]bani_Sam--: kills echo from local (handset, speakerphone)
22:06.22h3xi have digital ears!!!
22:06.31tzanger[av]bani: a "proper" setup can still have hideous echo
22:06.37tzangerhence the insidiousness of the situation
22:07.10h3xit aint the analog speaker/mic that causes echo (usually)
22:07.16tzangerPSTN hopoff requires echo cancellation to be 100% effective.  You can get lucky, and most people do, but without echo cancellation you're screwed
22:07.28*** join/#asterisk sergey (n=Sergey@sergey.iks.ru)
22:07.36_Sam--how would you do echo canc if you hand off to a remote gatway?
22:07.40h3xits the... what do they call it
22:07.42h3xtalk bridge?
22:07.51tzangerh3x: well that's exactly what echo is created from (the mic picking up the speaker, or the hybrid reflecting too much energy back) but what makes it NOTICEABLE is the latency
22:08.02h3xthe analog shit in a analog phone that mixes your own voice into the speaker
22:08.08h3xyes!!!
22:08.10h3xHYBRID!
22:08.10tzangerthat's why a telephone call can sound perfectly fine but use an X100P or other FXO module and suddenly it's hideously echoey
22:08.35tzangerPRI and digital (VOIP) circuitry does not CAUSE echo.  but it doesn't mean you won't have it
22:08.36bkw__fadsf
22:08.50*** join/#asterisk Seldon1975 (n=someone@199.243.101.131)
22:08.52tzangerbkw__: I agree
22:09.07tzangerbkw__: but how are we going to get congress to approve?
22:09.46s34ndo changes in voicemail.conf require some action to become effective?
22:09.46_Sam--hey bkw...asterlink doesnt do local numbers for origination?
22:09.58_Sam--the only things i saw were 8**
22:10.18[av]bani_Sam--: EC needs to be done at the point where PSTN handoff occurs, so with a remote gateway the remote gateway would do the EC
22:10.32[av]bani_Sam--: EC is done at endpoints. origination and termination
22:10.55[av]bani_Sam--: so, your voip phone would have EC and the PSTN gw would have EC
22:10.56_Sam--i see...thank you for explaining...maybe that is why my calls are ok, maybe teliax actually has EC (doubt it)
22:11.01[av]baniteliax has EC
22:11.08[av]bania reasonable one too
22:11.16[av]banijunction networks has a less good one
22:11.32tzangerit's good to echo cancel on the PSTN side too instead of in software
22:11.44tzangerbecause a lot of the delay occurs at the PCI level
22:11.54mutis MARK2 & aggresive better than kb1?
22:12.00tzangerso if you can kill the echo before it hits the PCI bus, that means the PCI latency isn't going to make what is left all that bad
22:12.06tzangermut: I hate agressive
22:12.16_Sam--echo canc all works on like sine waves and stuff?
22:12.21tzangermut: I have found that MG2 is better than KB1 which it enhances, which in turn is an enhanced MARK2
22:12.35_Sam--im just curious what it does / how it works in a really small nutshell
22:12.45_Sam--it like does phase shifts or something?
22:12.46muti tried MG2
22:12.51mutbut it just didn't get rid of the echo
22:12.54muti dunno what to do about it
22:13.14[av]bani_Sam--: EC is an art :)
22:13.47[av]bani_Sam--: it keeps a buffer of outgoing audio, and looks for the returning echo, and applies an out-of-phase signal to cancel it out.
22:13.49[Atlas]Cheap PoE switches/hubs?
22:13.53*** part/#asterisk jaike (n=a@203.131.137.76)
22:14.03[av]bani_Sam--: but it's hard because the delay can vary, and the level of the returning signal can vary
22:14.06_Sam--hmmm so it is basically a phase shift?
22:14.10h3x!PoE
22:14.11h3xer
22:14.13h3x~PoE
22:14.14jbot[poe] Perl Object Environment, an event driven daemon architecture, http://www.perl.com/pub/2001/01/poe.html?wwwrrr_20010117.txt
22:14.17[av]banisorta
22:14.18h3xdammit
22:14.20h3xwhat the hell
22:14.32h3xjbot is from #perl !? hahahah
22:14.35_Sam--i see the concept
22:14.39[Atlas]LOL
22:14.51h3xi remember jbot
22:15.12_Sam--what frequency does the EC sample at?
22:15.30h3xits a time shift not a frequency
22:15.46[Atlas]~sub
22:15.56h3x~CPAN
22:15.57jbotmethinks cpan is the Comprehensive Perl Archive Network; the store of all Perl modules, or 'The World', or reasonably evil, or 'Reasonably Evil', or "perl -MCPAN -e shell"
22:15.57_Sam--but to know if its out of line (echoing) it seems it would need to sample to know how far skewed it is
22:16.19jlewisI've got something weird going on with 2 customer cvs-stable boxes
22:16.20h3xthis is gonna be funny
22:16.22h3x~python
22:16.23jbotit has been said that python is Available at http://www.python.org    Python is an interpreted, interactive, object-oriented programming language. It is often compared to Tcl, Perl, Scheme or Java.
22:16.27[av]bani_Sam--: it keeps a buffer of the audio it sends out... so it already has it sampled
22:16.33_Sam--i see isee
22:16.36jlewisabout the same time this afternoon, both started doing  chan_zap.c:8015 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too.
22:16.55[Atlas]ahh lame perl bot didnt bash python!
22:17.05h3xyeah
22:17.13h3xmaybe coz one of the CREATORS of perl writes python! hahaha
22:17.17jlewisinteresting note...this error says nothing about which PRI its talking about...one of the servers has 2
22:17.20[Atlas]~wheresthegooldoldcoderflamewars
22:17.34h3xjlewis: Uhm, do you have the asterisk boxes connected back to back?
22:17.50sevardalright, if anyone knows php are they able to toss me some quick help?
22:18.13sevardin wakeup.php i changed $parm_application = 'MusicOnHold'; to $parm_application = 'MP3Player(/var/lib/asterisk/Norah_Jones-Come_Away_With_Me-Dont_Know_Why.mp3)';
22:18.16sevardand it just doesn't work.
22:18.20[Atlas]sevard: depending on the value of the know variable i might be able to
22:18.20jlewisno...one connects to some kind of call center PBX, the other to a cisco voip router
22:18.29jlewisboth have worked for months
22:18.29RoyKhej
22:18.42*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
22:18.57sevardi know the file will play because if i add this to extensions.conf it works exten => 557,1,MP3Player(/var/lib/asterisk/Norah_Jones-Come_Away_With_Me-Dont_Know_Why.mp3)
22:19.13_Sam--im telling NOrah Jones
22:19.20sevardit's legal!
22:19.26_Sam--lol ok !
22:19.38sevardand just for testing :\ it's the only mp3 i had available :P
22:19.46_Sam--you know i am just kidding.
22:19.49sevardi do know
22:19.58*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
22:19.59sevardbut damnit, i can't figure out why it's not playing
22:20.30_Sam--did you check /var/log/asterisk/my_agi.log
22:20.31[Atlas]Asterisk does not care for norah,, try some nofx
22:20.38[Atlas]:D
22:21.05sevardi don't have a /var/log/asterisk/my_agi.log
22:21.09_Sam--i guess that could be specific to me only...sinc ei think i tell mine to log
22:21.34_Sam--$stdlog = fopen("/var/log/asterisk/my_agi.log", "a");
22:21.44sevardyou're putting that where?
22:21.51_Sam--$in = fopen("php://stdin","r");
22:21.51_Sam--$stdout = fopen('php://stdout', 'w');
22:21.51_Sam--$stdlog = fopen("/var/log/asterisk/my_agi.log", "a");
22:21.57RoyK~pb
22:21.58jbotwell, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
22:22.04_Sam--bah 3 lines, no pb
22:22.07sevardheh
22:22.18sevard_Sam--: still confused on where that's going
22:22.24harryvvsam perl?
22:22.29_Sam--that is the top of my php
22:22.33sevardputting.
22:22.54harryvvsam, do you know what that same command is in c?
22:23.00_Sam--sevard:  sending PM
22:23.12harryvvfunction i mean :)
22:23.22_Sam--to log your agi?
22:24.10[av]bani_Sam--: how many gxp2k's you got?
22:24.19_Sam--between my place and the offices i setup, maybe 50
22:24.23[av]bani:O
22:25.31_Sam--its the only hard phone ive ever seen or worked with :/
22:25.35[av]bani:/
22:25.42_Sam--makes selling it easy
22:25.48*** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be)
22:26.19_Sam--the last place i brought brochures for a bunch of phones, but mainly the sales im getting are all really price sensitive
22:26.26_Sam--that is why they are calling me to begin with...
22:26.40_Sam--so when i show them a $150 phone or an $100 phone...they are always picking the cheapest
22:26.50_Sam--i brought brochures and tried to show them nicer phones
22:26.52_Sam--but they all want cheap
22:27.07[av]banigxp is certainly cheap
22:27.29mutsoo
22:27.35mutshow them really expensive
22:27.37mutthen expensive
22:27.42_Sam--hah that is not a bad idea
22:27.44trixterthe gxp2000 is supposed to get updatable lcd scrollys soon so you can have arbitrary messages scroll on them
22:27.56trixterthe beta firmware isnt that bad either, added some features that should have been there all along
22:28.05mutlike "Get back to work you slacker"
22:28.08mutscrolling all day?
22:28.11_Sam--but my problem is if i price the whole setup (including phones) out of their reach or come in with a package that is too much dollars, then i get no sale
22:28.18[av]banitrixter: lcd scrollys?
22:28.33RoyKoh well... Iran is making WMDs. Bush is out there again :)
22:28.37trixterarbitrary messages that you can send to the hpnoes to display on the lcd
22:28.43[av]baniwheres that from?
22:28.53_Sam--i hope bush gets some military action in iran so my oil options go through the roof
22:28.58_Sam--just what i was banking on
22:29.03trixterI spoke to a contact at grandstream today
22:29.08_Sam--and knowing bush, he will, because he has oil options too :)
22:29.29trixterand knowing the democrats they will vote for it becuase they too have oil options
22:29.34trixterremember john kerry has far more money than bush
22:29.44trixtersomething about heinz ketchup
22:29.48trixter$700M right there
22:29.54rob0oh yeah That's his wife
22:30.03trixteryeah as in they are married thus its johns too
22:30.07[av]banithe bush family is a mega conglomerate, not just gwb
22:30.25trixterodd that the beureau of prisons uses heinz isnt it?
22:30.25cpmIt doesn't matter, The fix is in. Mark Warner, 2008. Done deal. enjoy it.
22:30.28trixtercaptive audience
22:30.32[av]banithe whole family has huge oil corporations
22:30.41justinuhey, i like heinz
22:30.45trixtermany democrats do too
22:30.48trixterthat is my point
22:30.50trixterits not one sided
22:31.03_Sam--i am the furthest thing from a bush supporter...but ive found the way so if you cant beat em join em...and oil has paid off handsomely
22:31.04rob0I don't think there is anyone at the national political level in *any* major country who has any honor left.
22:31.12*** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
22:31.26cpmwhat rob0 says
22:31.51cpmthrow all the bums out. All of 'em (except Sen Byrd, he's my hero)
22:32.02_Sam--until people have a means to initiate real change themselves, then politics will always be a sham, like organized religion
22:32.04mzo_hah, the cia office of clandestine services was recrutiing on campus today.
22:32.15justinubyrd is great!
22:32.18cpmcool, did you get an interview?
22:32.20_Sam--<sorry to all organized religion zealots>
22:32.30cpmNo offense taken :)
22:32.32[av]bani_Sam--: heretic!
22:32.48[Atlas]<-- scarred for life ^_^
22:33.11_Sam--but because the common person's voice is still never heard in politics, how will that system ever work?  people instead of getting more into politics and trying to effect change now just bury their heads in the sand because they know its useless
22:33.25Seldon1975Bush's atrocity of a government has made Americans hated all over the world
22:33.28*** join/#asterisk Zodiacal (n=hehehe@bdsl.66.14.242.199.gte.net)
22:33.44Psykickyou guys know what I believe ......
22:33.50Psykickwe'd all be better off without money
22:33.50*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com)
22:34.06[Atlas]wow im quite the outcast here :D
22:34.07_Sam--Psykick:  come work for me? :)
22:34.08[av]baniPsykick: give me yours. problem solved.
22:34.10rob0Psykick: I've tried that. In fact I am trying that as we speak. :)
22:34.29mzo_I vote for whatever party gets me lots of sex and gives me a free macintosh, and working software. :P
22:34.47Psykickit helps having it ..... but if we really want to effect a positive change for all ..... money must not be the sole point of living
22:35.24[Atlas]money and mans ego are the fountainhead of human progres
22:35.28_Sam--i dont see money as the issue...some people work hard for it and get it ...and some people are lazy bums and drink 40s of milwaukees best all day and dont get any.
22:35.49[av]bani_Sam-- sits and IRCs all day
22:35.50[Atlas]progress*
22:35.51Seldon1975democracy is flawed
22:35.53_Sam--i really do see people who work hard getting financially rewarded though
22:35.54Psykickpoverty is an induced form for lack of money
22:36.15[Atlas]laziness and lack of education induces laziness
22:36.18_Sam--and i dont think that is wrong, it gives people motivation to work harder
22:36.19Psykickdemocracy isn't flawed .... it's the people that are supposed to cause effect from that democracy
22:36.52Psykickand what usually causes no effect in the areas where change is needed most is money
22:36.56Seldon1975Psykick: that's just it; democracy is inextricable from the flaws of everyday people
22:37.03_Sam--i think instead of having elected officials that the public in this day and age should able to vote outright on any issues instead of having a "representative" of the public vote on our behalf
22:37.14[av]banicorporation = person, thats the major flaw
22:37.15[Atlas]money doesnt do that -- lack of moral fiber does that
22:37.22Psykickto a degree Sam ....
22:37.25Seldon1975bani: true dat
22:37.40Psykickthe biggest problem is that not all people share the same opinion or views
22:37.55[av]baniPsykick: yeah, it would be far simpler if everyone just STFU and obeyed
22:37.55sevardBRASS MONKEY
22:37.59sevardYEAHG!!!
22:38.02_Sam--lol
22:38.04sevardthat makes me happy when things work
22:38.23Psykickand in an extreme case ..... the wrong changes can be invoked
22:38.29[Atlas]http://www.atlasshrugged.tv/speech.htm     ---- an interesting not often thought of Point of view for money
22:38.40Seldon1975Psykick: the problem is that the average man is too short-sighted to see whats good for him
22:38.51Psykicktoo true seldon
22:39.05[Atlas]Seldon1975: that is a very dangerous path of logic
22:39.06[av]baniwhatever happened to #asterisk ?
22:39.19_Sam--Seldon1975:  what makes you think you know whats better for the average man than he does?
22:39.24_Sam--that is what politicians think
22:39.25Seldon1975[Atlas]: perhaps, but it can't be denied
22:39.29_Sam--"i know whats good for america"
22:39.37[av]bani#asterisk renamed to #politics  :<
22:39.37[Atlas]that tends to lead to lack of individual freedom because some appointed person "knows better"
22:39.38PsykickI believe alot of what is wrong with this world is that we allow ourselves to be influenced .... many of us say that we aren't but .... if you look at it from all viewpoints .... we are
22:40.03Seldon1975_Sam--: I am not proposing myself as a solution!
22:40.06[av]banioh well someone wake me up when the pseudophilosophy is over
22:40.12_Sam--lol
22:40.23_Sam--only so much SIP you can speak before it turns to something else :)
22:40.47Psykickprejudice ... hate crimes ... all because of influence
22:40.48[Atlas]money is the only means man can deal with another for trade without including force
22:40.50_Sam--we've had motorcycles, SIP, and politics today...a broad array of topics.
22:40.58Seldon1975we're too enamoured with Western science/philosophy
22:41.04*** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82)
22:41.09Zodiacalanyone know of a way to speed up dialing with my digum tdm400p? it seems like it takes about 5 seconds from the time Dial() is issued to the time my trunk acctualy gets opened and starts ringing..
22:41.24cpmSeldon1975, or not enamoured enough. You pick.
22:41.39rpmwhat variable is the current date stored in? ${DATE} ?
22:41.39Seldon1975cpm: I pick 'too mcuh'
22:41.44[Atlas]Seldon1975: there is no honor in being the poor of ethiopia or romania
22:41.47Seldon1975much*
22:41.48*** part/#asterisk Navman_Lap (n=icechat5@62.108.206.82)
22:41.52*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-204-92.red.bezeqint.net)
22:41.54[av]banihmm someone should make RJ11 EC's
22:42.10Seldon1975[Atlas]: irrelevent comment
22:42.12PsykickI believe there are groups or organizations of people who influence the world for the worse
22:42.17Psykickin order to get what they want
22:42.25_Sam--they are called terrorist
22:42.26_Sam--s
22:42.39[Atlas]Seldon1975: no it wasnt lol it is western culture that make our "poor"  the richest people in the world
22:43.08Psykicksam: what if those terrorists were paid to do what they're doing in order to distract us from a much more hideous problem
22:43.11[Atlas]i would so much rather live here and be poor with 2 microwaves a tv and 3 dvd players than not be able to feed my family somewhere else
22:43.15*** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au)
22:43.23[Atlas]oh god i have to stop
22:43.25[Atlas]sorry guys
22:43.29Seldon1975[Atlas]: still irrelevent - is being 'rich' by material standards what's best for us?  look at society, look at whats happening to the planet
22:43.51*** join/#asterisk dijit0 (n=dijit0@adsl-68-127-10-103.dsl.pltn13.pacbell.net)
22:43.56PsykickI agree seldon
22:43.59cpmSeldon1975, good essay for you to read someday: http://www.michaelmoser.org/books/book_engine.htm
22:44.09Seldon1975this is the hottest year in the last hundred
22:44.19_Sam--im not sure how us being any less 'wealthy' would make us a 'better' society in the USA
22:44.27_Sam--would we be more compassionate?
22:44.36Seldon1975_Sam--: geez I hope so
22:44.52Psykicksam: you look at those that don't
22:45.00cpmcompassionate hasn't hurt a lot, more would be a nice thing to try for a change.
22:45.03*** part/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co)
22:45.11Psykicksam: do you see them killing each other ... for money ... for a better way of living
22:45.16*** join/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co)
22:45.29Seldon1975_Sam--: you could hardly be less compassionate
22:45.43cpm<PROTECTED>
22:45.44[Atlas]Seldon1975: watching my family starve becuase there is no way for me to get ahead,, watching my children murdered and my wife raped at the hands of a brutal govnmt is alot less attractive for me
22:45.49PsykickI believe that if they had money they would only use as much is necessary to survive
22:46.06_Sam--im the least compassionate person you could ever meet...but i would argue money has nothing to do with it...its because ive worked hard and im not very compassionate for people who havent worked as hard.
22:46.14[Atlas]and as far as killing eachother for money... capitalist societies are by far the least violent,, that is a fact
22:46.30Psykicksam: compassion is not something that is influenced by money
22:46.33Seldon1975[Atlas] erm... the alternative to squandering the worlds resources isnt starving!
22:46.36Psykickits a matter of the heart
22:46.56Seldon1975[Atlas] you dont think we can all subside without material wealth?
22:47.10*** join/#asterisk girllinux_21 (n=ircap8@126.Red-83-56-107.dynamicIP.rima-tde.net)
22:47.13_Sam--i see the poor people in american society, and i have no compassion, or heart...because im like "that f'er could have studied  as hard as i did and got a good job too"
22:47.21[Atlas]Seldon1975: the fact is no country in history has found your enlightened way -- the closest we have ever came as the human race to peacable comfortable life is through capitalism
22:47.24cpmI think we would subside without material weath.
22:47.29Psykicksam: what's not to say that they didn't
22:47.35Seldon1975[Atlas] bullshit
22:47.50Seldon1975[Atlas]: throughout history there are heaps of examples
22:48.02Psykicksam: and that their job was made redundant .... partner left them ... world ... family all turned their backs on them
22:48.03_Sam--Psykick:  most of the poor are poor because they are lazy!  <sorry that is a HUGE generalization and not exactly my point>...but that is how i often feel
22:48.03[Atlas]go for it
22:48.14[Atlas]agree with sam
22:48.18Seldon1975[Atlas]: for a time, the romans, the greeks, the chinese
22:49.04[Atlas]ill leave you to stufy some history to find out how wildly erroneous of a statement that ws
22:49.04[Atlas]was
22:49.08[Atlas]study*
22:49.09*** join/#asterisk rene- (i=rene@201.144.60.114)
22:49.13*** part/#asterisk Martincit0 (n=martin@litigaractivos1.att.net.co)
22:49.29Seldon1975[Atlas]: ill leave you to study it and learn
22:49.29Seldon1975[Atlas]: the truth
22:50.10[Atlas]Seldon1975: i am not much of a programmer or asterisk hacker,, but one thing i do know is world history -- especially in classical times
22:50.31Seldon1975[Atlas]: I'm not saying they were like that throught all of history; but they enjoyed times of peaceful prosperity
22:50.32Psykicksam: I agree to a certain degree
22:50.52rene-hello
22:50.53Psykicksam: most have not been exposed to encouraging parents .... or loving parents for that matter
22:51.04[Atlas]and by the way so there is no confusion rome was a republic built on money much like america today the only difference was the lack of democratic influence
22:51.18shmaltzsilly:
22:51.19[Atlas]when it turned into a dictatorship for all intents and purposes the people were abused
22:51.20shmaltzhttp://ask.yahoo.com/20060131.html
22:51.30Seldon1975[Atlas]: ok I'm talking about before that
22:52.07cpmwhat we have now is a corportacracy, a capitalist fuedalism. Not exactly a republic.
22:52.45Seldon1975[Atlas]: do you really think that society is at it's peak?
22:52.53Seldon1975[Atlas]: no, infact it's falling apart
22:52.58_Sam--good conversation, i would love to stay and continue, but ive already stayed at work an extra to debate...its quittin time ...i will think of some good arguments on the way home :)
22:53.08cpmg'night
22:53.10Seldon1975hehe
22:53.11Seldon1975ciao
22:54.04*** join/#asterisk lesouvage (n=lesouvag@82.74.11.143)
22:54.04[Atlas]Seldon1975: at its peak -- technologically yes,, as far as peace goes we have alot of things to get over in the world i dont believe money is one of them
22:54.05cpmSeldon1975, it is certainly at a cusp, that's pretty hard to argue. Either we outgrow this feudalism AGAIN, or we go back through it all AGAIN. No argument.
22:54.53*** join/#asterisk RoyK (n=roy@193.80-202-93.nextgentel.com)
22:55.14[Atlas]all of the major bloodbathes of the world were not influenced by money ,, but by the forcing of power, through religion, ideals and such
22:55.26_Sam--they (the govt) needs to setup asterisk servers for voting...then when they have big issues, we just vote through asterisk :)
22:55.33[Atlas]the fact is there will always be someone in power
22:55.40_Sam--then we could vote on the issues instead of leaving it to our representatives
22:55.56[Atlas]money is the only means to get there -- for the most part -- without force
22:56.09Psykick[Atlas]: quite a few that we don't know about ..... and I don't believe that they are apart of any government
22:56.10lesouvageDoes asterisk support/make use of dual core processor power on the motherboard.
22:56.19[Atlas]what we lack is moral fiber and honesty -- that mean is easily corrupted with or without money
22:56.32[Atlas]there always needs to be a medium of exchange if not money then what?
22:56.34Psykicklesouvage: that's a linux thing not asterisk .... someone correct me if I'm wrong
22:56.38[Atlas]religion, virtue, faith?
22:56.39mzo_i thought this was #history?
22:57.18austinnichols101set ramblingOffTopicConversation = on
22:57.23[Atlas]lesouvage: Asterisk runs quite well on my dualcore bro :)
22:57.31austinnichols101nice
22:57.36lesouvagePsykick: as far as I now it depends on the application but I may be wrong.
22:57.38Psykick[Atlas]: where are you based
22:57.49[Atlas]idaho
22:57.58[Atlas]for now :D
22:58.17Seldon1975Atlas: do you contend that the Iraq war is not about money?
22:58.25justinuyou're THE ho?
22:58.31justinuquite an honor
22:58.32mzo_i thought they were fighting Nod?
22:58.43lesouvageAtlas: and both cores are used by Asterisk.
22:58.46Seldon1975Atlas: because if you do we might as well stop talking
22:58.50[Atlas]Seldon1975: im no bush lover
22:58.51mzo_asterisk does smp fine?
22:59.03[Atlas]no asterisk is not using both cores
22:59.08mzo_it doesn't?
22:59.09cpmSeldon1975, Not directly, it's about empire.
22:59.13[Atlas]but..... os and other processes are
22:59.20mzo_i have an smp box, but i never bothered to check if it did or not
22:59.24[Atlas]there was a *drastic* improvement
23:00.02bigjbdid you recieve my message Psykick ?
23:00.03*** join/#asterisk AgilixSupport (n=AgilixSu@i.agilix.com)
23:00.03Seldon1975[Atlas]: you said "all of the major bloodbathes of the world were not influenced by money"  were you ignoring the Iraq conflict or have you not heard of it?
23:00.32*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
23:00.33Seldon1975cpm: my ass
23:00.37Seldon1975cpm: its about Oil
23:00.41Seldon1975cpm: = money
23:00.43[Atlas]Seldon1975: you ahve asked me a question i take an unusual stance on,, i believe that may be a big motivating factor yes,, but i do believe we should be there --- not for money or WMDs or anything like that but saddam did need dethroned as it were
23:00.48mzo_is there some way to tell if asterisk is doing somthing with smp? some show (something) command?
23:01.03cpmNaw, it's about empire. Oil is also a big part, as is money, but principally, it's about control.
23:01.14[Atlas]Seldon1975: i also believe the war goes much deeper than that
23:01.20Seldon1975forget saddam, forget WMDs; the biggest motivation for Bush to go to Iraq was Oil == money
23:01.33Seldon1975Im trying not to rant
23:01.38Seldon1975but it's so undeniable
23:02.05Psykickbigjb: yes thank you ....
23:02.08lesouvageAtlas: thanks for the info, buying a mb with a Intel 775 800 serie Dual Core seems to be a good idea.
23:02.25cpmSeldon1975,http://www.newamericancentury.org/
23:02.34[Atlas]lesouvage: no prob bro :)
23:02.43cpmCome back and argue it again once you have read the manifesto.
23:03.07mzo_that website is scary
23:03.09[Atlas]seldon1975: that is not a bloodbath on the scale of the major one of which i was talking
23:03.14mzo_i always thought it was fictional for a movie plot
23:03.15cpmIt's about empire. Yes, oil is a part, money is a part, but in the end, it's control and empire building.
23:03.24[Atlas]indeed their hatred for us is as great as we want power
23:03.28bigjbam waiting for word from avaya but ive had a play today and am pretty sure he was right
23:03.30mzo_...I find your lack of faith disturbing...
23:03.37[Atlas]if they showed us love we would not be over there
23:03.59Seldon1975Atlas: that is such an ignorant statement I just cannot bring myself to argue any more
23:04.01[Atlas]agree with cpm for the most part
23:04.15justinuPNAC called "a new pearl harbor" an "the opportunity of ages"
23:04.17[Atlas]Seldon that was as great a copout as i have ever heard :)
23:04.19justinuthat was in the 90s
23:04.29*** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au)
23:04.30Seldon1975theres just no point
23:04.45cpmBig purges of the last 100 years, Stalin, Hitler, Pol Pot, nothing money related there. All about empire, control, power.
23:04.58justinu"the process of transformation, even if it brings revolutionary change, is likely to be a long one, absent some catastrophic and catalyzing event -- like a new Pearl Harbor."
23:05.00Seldon1975"if they showed us love we would not be over there" thats going in my book of all time American ignorances
23:05.00*** part/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net)
23:05.12[Atlas]let us not forget the crusades and dark ages
23:05.19sevardbbl
23:05.22*** part/#asterisk sevard (n=kynan@198.174.233.25)
23:05.29cpm[Atlas], yeah, again, pretty much the same thing.
23:05.32[Atlas]Seldon1975: you misunderstood
23:05.36[Atlas]my statement
23:05.51[Atlas]would the people stand for it if they were not in fear?
23:05.59[Atlas]would anyone stand for it?
23:06.04[Atlas]no.
23:06.17Seldon1975Atlas: do you have any idea why terrorists (not Iraquis mind you) attacked America?
23:06.31cpmjustinu, yeah, that's a pretty chilling statement, esp in retrospect, yeah?
23:06.37[Atlas]but throughout the conflict -- we have gone everywhere form as a country being 50/50 divided on us being there give or take 20 on each side
23:06.42Seldon1975Atlas: was it because they just hate capitalism that much?
23:06.43Psykickeverything in this world is about control
23:06.54justinucpm - agreed.
23:06.55[Atlas]Seldon1975: i believe i understand that topic better than you
23:07.01[Atlas]no
23:07.04[Atlas]but that is part
23:07.07Psykickstart losing it ... start a hate crime ..... fly a plane into a building killing thousands ...
23:07.13Seldon1975Atlas: ok then; please explain why terrorists attacked merica
23:07.18Seldon1975Atlas: I'm all ears
23:07.18[Atlas]it has alot to do with our support for israel
23:07.22mzo_weird, so asterisk isn't smp.
23:07.31justinumerica.... merica.... fuck ya!!
23:07.39cpmIt has a huge amount to do with our support for Israel,
23:07.46[Atlas]but it also has much to do with their hatred for our wasy of life
23:07.52*** join/#asterisk dfroe (n=chatzill@dslb-084-056-227-185.pools.arcor-ip.net)
23:07.56justinuwhat I want to know is why we've never seen any surveillence footage of the so-called hijackers
23:07.56bigjbcan anyone tell me why is it that im able to connect via sip to asterisk from home via nat port forwarding, but once it connects the call it fails to make a outgoing channel?
23:07.58Seldon1975Atlas: oh my god
23:08.03Seldon1975Atlas: pls stop
23:08.17[Atlas]Seldon1975: you can not believe ther might be a few reasons?
23:08.23Seldon1975"it also has much to do with their hatred for our wasy of life"
23:08.37cpmfolks who don't exactly hate us outright have been warning us since before the clerical revolution in Iran, toppling our puppet tyrant, that our support of Israel really pisses them off to no end.
23:09.09PsykickI don't agree either seldon
23:09.18Seldon1975don't agree with what?
23:09.26Psykick<Seldon1975> "it also has much to do with their hatred for our wasy of life"
23:09.33rob0Sure ... America's way of life is to exterminate or assimilate anyone who's different
23:09.36Psykickit has absolutely nothing to do with it
23:09.45Seldon1975what has nothing to do with what?
23:09.47maskedi just had mormans come to my door who will probably end up suicide bombing themselves in the name of whats good, thats enough relgion and polotics for me for one day
23:09.55*** join/#asterisk brockj49464 (n=brockj49@63.87.56.252)
23:09.58ReD-MaNterrorism is nothing but a bunch of cowards thinking they are in control and can bully people around anyways
23:09.59mzo_mormons :P
23:10.01maskedso now can you pipe it and take it to a voip conference?
23:10.04Psykickthe terrorists and the choices they made we're not over lifestyle
23:10.09[Atlas]ok i have to be done -- i have to do some work
23:10.13Seldon1975ok
23:10.18Seldon1975nice chattin' :}
23:10.23maskedyeh it wont be long before born again christians start terrorism for their beliefs
23:10.25maskedwell...
23:10.38[Atlas]Seldon1975: even though i disagree with you ,, no hard feelings :)
23:10.40maskedits a bit late for that really isn't it, bush already kills people in the name of god
23:10.44[Atlas]take care
23:10.53Psykicksee ya atlas
23:10.54Seldon1975[Atlas]: take care mate
23:11.01cpmI have to go also, Seldon1975, [Atlas], all, thanks for the chat!
23:11.09Seldon1975cpm: you too
23:11.13cpmg'night.
23:11.15[Atlas]yeah thanks guys :) made my day fun hehe
23:11.19Seldon1975heh
23:11.20Seldon1975ciao
23:12.27Psykickmasked: I don't believe that for a sec
23:12.41Psykickmasked: I believe someone will get too greedy
23:12.54Psykickthen all hell will break loose
23:14.25PsykickI better go as well
23:14.33Psykicklater guys ... thanks for the chit chat
23:14.54*** join/#asterisk oatis (n=oatis@c-67-172-176-64.hsd1.ca.comcast.net)
23:15.35oatisHi, my asterisk server is behind my router at the office and I would like to use it from home too... what ports do I need to forward in the router at the office to the asterisk server so I can do this?
23:15.40oatisIm using SIP
23:15.53*** part/#asterisk clive- (n=pirch@dsl-165-149-246.telkomadsl.co.za)
23:16.11dfroeHi, I read http://www.voip-info.org/wiki/view/GXP-2000 and found it very interesting. The new features like an adressbook are great for my phone! Does anyone of you know the (main) author of this site so I can contact him for further details?
23:18.54austinnichols101oatis: nat at the office?  nat at home?
23:19.04rene-how can one queue be spanned across multiple asterisk boxen?
23:19.19*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com)
23:19.38RoyKanyone here that knows why digium hasn't included PLC in codec_g729?
23:19.54rob0I think the wiki pages all have an edit history available.
23:19.54austinnichols101oatis: and what do you want to use at home - 1 phone, multiple phones?
23:20.17RoyKrene-: it can't
23:20.48maskedi have a problem installing zaptel it doesn't put the module entries into /etc/modules.conf so it fails anyone know how to fix this?
23:20.49*** join/#asterisk santiago (n=santiago@63.245.86.182)
23:20.56RoyKrene-: i beleive zoa is working on a commercial project on that subject, but standard asterisk doesn't support it
23:21.21shmaltzidiot:
23:21.22shmaltzhttp://www.breitbart.com/news/2006/01/31/D8FFUHC01.html
23:21.48rene-i saw something about it on *-biz i believ
23:22.13rene-im on a 6E1 + 300 user setup i dont think one box will suffice
23:22.29shmaltzrene, why not?
23:22.31tronixmasked: don't need to put zaptel in modules.conf
23:22.33mzo_one machine can handle 300 users?
23:22.40shmaltzlol
23:22.57rene-i wont use compression for calls but i will be doing gsm recording of calls
23:23.16*** join/#asterisk sindy_84_ (n=ircap8@126.Red-83-56-107.dynamicIP.rima-tde.net)
23:23.17ptiggerdineSCSI disks u must.
23:23.32rene-ptiggerdine
23:23.49mzo_so that grandstream disk is the best bargain these days?
23:23.50rene-i was thinking of having a realtime data in a separate mysql box
23:24.04rene-i could rig that with scsi and offload recording via nfs to that
23:24.05mzo_hah, er, phone, not disk, that scsi thing got to me. :P
23:24.11tronixmasked: start asterisk with -vvvvc option and see if it mentions chan_zap.so loading
23:24.14ptiggerdinenfs is slow.
23:24.33rene-how can i offload the complexities of recording to a third machine
23:24.34*** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.107.102.Dial1.Chicago1.Level3.net)
23:24.44ptiggerdineone storge box connected to the two systems would work better.
23:24.45*** join/#asterisk lucasjb (n=lucas@mail.stabat.com)
23:24.49litageto install h323 support, do you just run 'make' from within the asterisk-addons directory, or do you need to specifically run 'make' within asterisk-addons/asterisk-ooh323c/ ?
23:25.11tronixlitage: not sure but you could do 'make -n install' from both dirs and see which looks sane
23:25.12ptiggerdinegood luck compiling it.
23:25.22brockj49464is there an IRC channel for AAH?
23:25.25tronixlitage: -n pretends to install but doesn't actually do it, so you find out what it'd have installed
23:25.27maskedtronix: ok, make install failed, but asterisk still loads that module
23:25.38*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.92)
23:25.52maskedtronix: so that should be fine?
23:26.05litagethanks tronix
23:26.07rene-tigger: do yo mean something like the apple xserve raid? something fiber based?
23:26.14tronixmasked: on the asterisk side, think so, yes. on kernel side, will want to check with 'lsmod | grep zaptel'
23:26.28tronixditto for whatever wcxxx module you use too
23:26.32ptiggerdinerene-, not I'm talking SCSI here.
23:26.32lucasjbHiyas, I have a problem with my Asterisk system. When I've got one SIP user connected testing extensions.conf, everything seems to work fine, but if I have two users, as soon as they system wants to play an audio file, my CPU usage goes from 99% idle to 99% system. The process events/0 is taking all system resources. Where should I start looking to fix this problem?
23:26.52ptiggerdineF/C it going to be over kill
23:27.01ptiggerdinebut NFS is going to be very slow.
23:27.18hardwirewtf is libzap?
23:27.28maskedtronix well its not loaded
23:27.57rene-lucas you must certainly have an mp3 player issue
23:27.59tronixmasked: okay. what happens if you do 'modprobe zaptel'
23:28.10rene-try to upgrade your mpg123 imple,entation
23:28.11maskedtronix module not found
23:28.26tronixmasked: you've got two parts of zaptel stuff: one is kernel side, one is asterisk side. you've got asterisk side ok, but need kernel side
23:28.31tronixkernel side talks with the actual hardware
23:28.34maskedtronix ill show u the install error
23:28.49lucasjbrene-, Thanks I'll try that.
23:28.50tronixmasked: could you use www.pastebin.com?
23:28.54maskedyep
23:28.58tronixsweet
23:29.32maskedhttp://pastebin.com/532995
23:29.42maskedok the error is at the end of line 3
23:29.50*** join/#asterisk sevarrd (n=kynan@198.174.233.25)
23:30.03sevarrdis there such a thing as a Clean Room / Dirty Room module for *?
23:30.09dlynesdoes anyone know of a way to get a timing device working under freebsd for musiconhold, if you're using a pure software solution for Asterisk?
23:30.16maskedwhere it says echo "alias etc etc" >> (modules.conf should be here) ;
23:30.59maskedsevarrd: asterisk will clean my room?
23:31.10maskedyou mean to say all this time i've been cleaning my room without the need to?
23:31.19maskedomgbbq
23:31.21tronixmasked: i'm thinking a null variable is causing your install failure. should be easy to work around. what distro you using?
23:31.31sevarrdno, hotels use clean room / dirty room systems where a maid calls in and reports a room clean and ready for sale
23:31.37rene-tigger: is this possible? asteriskbox-scsiadapter-scsicable-scsiadapter-scsidrives-mysqlbox
23:31.42sevarrdand it's omgwtfbbq :P
23:31.45tronix:-)
23:31.46*** join/#asterisk zotz (n=zotz@24.231.47.175)
23:31.55*** join/#asterisk troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)
23:32.20znoGsevarrd: you could do one in Perl (asterisk->AGI)
23:32.22sevarrdI was searching and didn't find one already written for *, has anyone heard of one?
23:32.43*** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com)
23:32.54*** join/#asterisk dfgas (n=dfgas@adsl-69-210-72-31.dsl.milwwi.ameritech.net)
23:33.06dfgasis there a howto on installing from source not iso
23:33.40*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-169.nas28.salt-lake-city1.ut.us.da.qwest.net)
23:33.44rene-so do you people think one top of the line dual xeon box or dual Opteron can handle 6xE1 with 300 sip users?
23:33.47maskedsevarrd you could make something simple like that with a dialplan
23:33.57sevarrdmasked: ?
23:34.00maskedtronix im using LFS 6.0
23:34.11infoboxhello has anybody had problems with INtel 915 chipset and TE110P?
23:34.15tronixmasked: ahh-ha. hm. what hardware card do you use?
23:34.25*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.92)
23:34.30hardwireanybody have a good process for retrieving recorded calls (via snom 3x0 record)
23:34.31maskedtronix i just installed a digium x100p, yet to use it
23:34.36[av]banirene-: 6 x E1 is only 180 channels...
23:35.03tronixmasked: ok. you're doing zaptel installation by doing a source build, right? what version?
23:35.09maskedsevarrd, i assume they pick up the room phone, dial an extension then indicate whether it's clean or not?
23:35.12_Sam--sevarrd:  to do what you want should be really easy
23:35.24maskedtronix 1.2.3
23:35.26_Sam--but will need some custom programming most likely
23:35.26sevarrd_Sam--: how would one go about that
23:35.33rene-[av]bani: agreed but doesnt converting from zap to sip (g711) count?
23:35.33_Sam--i personally would keep an sql table
23:35.48_Sam--and when they (the maids) call in, i would update that table
23:35.51[av]banirene-: probably not much
23:36.23sevarrd_Sam--: then maybe paste that table to a html document
23:36.30maskedsevarrd have something like, they dial an extension, get prompted for the room number, then idicate 1 for clean 2 for dirty, and that gets logged to the database
23:36.35_Sam--or have your hotel be able to query against that table
23:36.55tronixmasked: ok. let me look at something .. brb
23:37.07maskedthe table doesn't have to be pasted sevarrd, you will query it like _Sam-- said
23:37.12_Sam--like you could have a webpage using php that would show that status of any rooms
23:37.13maskedtronix thanks mate
23:37.14sevarrdwow, i'm not looking forward to trying to program that.  I know _nothing_ about sql
23:37.15rene-[av]bani: i would need call recording for 75% of calls would that impact performance very badly?
23:37.38_Sam--it will need some custom programming from what it sounds
23:37.53maskedyeah shouldn't be too hard tho
23:38.16*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-98.rockynet.com)
23:38.20maskedjust paypal donate the right ppl and you'll have it done in no time :)
23:38.23_Sam--there are usually at least a dozen ways to do the same thing
23:38.32_Sam--but that is my first thought on how i would do it
23:38.38*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
23:39.24_Sam--if the phones are voip and connect to the same asterisk server they wouldnt even need to do anything else except call the server when its clean
23:39.25_Sam--and hang up
23:39.34_Sam--the server would know what room it is
23:39.36masked_Sam--, sevarrd, considering each room has an extension/callerid, it could be as simple and dialing an extension and hanging up
23:39.47maskedexactly.
23:40.06rene-i should buy opteron just to stay safe
23:40.21maskedtronix it did it in 1.2.2 aswell, i was trying to install it for the ztdummy module
23:40.31ptiggerdinerene-, buy something that from the major players.
23:40.37ptiggerdineIBM, HP, DELL
23:40.44maskedtronix just fyi
23:41.05sevarrdmasked: most def
23:41.11tronixmasked: ah, good info. still checking, btw.
23:41.54sevarrdit looks like i'm going to have to learn mySQL and php
23:41.56_Sam--sevarrd:  how do you handle voice mails after the customer checks out?
23:41.58sevarrdbecause I know neither.
23:42.08_Sam--like if the customer had voice mail, then they check out...do you have something to get rid of it?
23:42.15sevarrd_Sam--: currently I don't, this is just a test server sitting in my room
23:42.20[av]baniwhats a good WinCE softphone?
23:42.38_Sam--im always interested to hear how specific industries use asterisk...
23:42.44_Sam--hearing about the hotel is good
23:42.49[av]bani_Sam--: ILECs use it for... phones!
23:42.54[av]bani!!!
23:42.54tronixmasked: what kernel do you use? 2.4 or 2.6? also, think I see your installation problem
23:43.05maskedtronix 2.6
23:43.07sevarrd_Sam--: I suppose i'd have to look into that
23:43.17_Sam--but now after just listening to sevarrd i would feel pretty comfortable talking to hotels about their phones
23:43.25tronixmasked: Makefile in zaptel-1.2.3 has a part that tries to figure out MODCONF variable -- which file to dump the alias entries in
23:43.41tronixmasked: and from your pb output, looks like it couldn't figure out MODCONF location and left it empty
23:43.45[av]bani_Sam--: most hotel phone systems are unbelievably shit
23:43.49tronixmasked: which causes odd errors that you saw
23:44.04maskedso, export that variable?
23:44.11sevarrd_Sam--: you sound excited, want to teach me? :)
23:44.22tronixmasked: could hack an entry like: MODCONF=/path/to/somewhere after that big MODCONF section
23:44.28tronixmasked: just to get stuff going.
23:44.45tronixmasked: but you're going to need to figure out which file/dir your distro uses for modules definitions
23:45.00maskedtronix yeah thats no drams
23:45.04maskedthanks mate.
23:45.07litagecompiling and installing ooh323c that comes in asterisk-addons produces chan_ooh323.so , but if i compile a version of ooh323c that i downloaded, chan_ooh323.so isn't created. how are you supposed to do this?
23:45.08tronixsplendid
23:45.12_Sam--sevarrd what are some other things specific to hotels that you have to worry about?
23:45.27_Sam--wake up call, room clean / not clean, voicemail, calling cards
23:45.50*** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au)
23:45.59sevarrdthat sounds pretty much it besides maybe a head maid going around and overriding room clean entries
23:46.33_Sam--not that it matters one bit, im just curious how many rooms are you talking about?
23:47.17sevarrdnot sure yet, it's hypothetical, i wanted to help mom and pop places with ~50 rooms
23:47.18maskedtronix ok that worked.
23:47.28tronixgreat
23:47.34*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
23:47.43_Sam--what are you going to use for phones / fxs?  existing phones with ata?
23:47.47maskedtronix one thing tho
23:47.52_Sam--im just wondering what your strategy / plan is
23:48.01sevarrd_Sam--: I don't have one yet
23:48.03tronixmasked: what's that?
23:48.13maskedtronix it added module entries in like install wct4xxp /sbin/modprobe --ignore-install wct4xxp && /sbin/ztcfg
23:48.18maskedas well as aliases
23:48.23sevarrdwell, i've looked at serveral ATAs
23:48.29[av]bani_Sam--: sounds like a job for powerline ethernet/wifi and ATAs
23:48.29_Sam--so its your goal to sell these type setups to hotels?  but you dont have a hotel that wants it right now?
23:48.44[av]banior a DSLAM and CPEs
23:48.45*** join/#asterisk j4m3s_ (n=j4m3s@gateway.digium.com)
23:49.07tronixmasked: aye. problematic?
23:49.14*** join/#asterisk Aldo (n=aleyva@200.62.180.209)
23:49.21_Sam--if all the phones run back to a phone closest, couldnt you use some type channel bank there?
23:49.22tronixmasked: don't think it'll load at next boot
23:49.27tronixmasked: for the stuff you don't use
23:49.29justinuhow much do DSLAMs run?
23:49.36[av]banidepends on the dslam
23:49.42[av]bani$1500 - $150,000
23:49.43Qwell[]$12.50, black market
23:49.55tronix:-)
23:49.56[av]banihow fast/far you wanna go?
23:49.57mzo_i don't think you get warranty on the black marketz
23:49.59justinuQwell[]: i'll meet you in the parking lot tonight
23:50.05justinu[av]bani: no idea, just kinda curious
23:50.10maskedtronix my concern is are those entries supposed to be there?
23:50.14[av]banijustinu: ebay + dslam
23:50.18maskedtronix or were they placed there by mistake?
23:50.24austinnichols101good, fast, cheap - pick any two you want.
23:50.24tronixQwell[]: you forgot the key part -- make sure it isn't Vonage-locked. :-)
23:50.28Aldohi
23:50.36Qwell[]vonage-locked dslam?
23:50.37[av]baniof course if you dont want nifty phones, a channel bank in the office would suffice
23:50.41[av]bani50 FXS...
23:50.41AldoI have an server asterisk 1.12
23:50.47_Sam--yeah thats what im thinking
23:50.50_Sam--easy sell
23:50.51Qwell[]Aldo 1.12?
23:50.52_Sam--existing phones
23:50.54tronixmasked: it just dumps a pile of stuff in by default
23:50.57tronixmasked: so that's normal
23:51.01maskedoh ok cool
23:51.01[av]banijust no nifty new phone features
23:51.05Aldov1.1.2
23:51.07tronixmasked: makes it easier to throw in different cards in the future
23:51.13Qwell[]Aldo: No such thing
23:51.16maskedrighto
23:51.18[av]banihmm one thing, hotels usually use phones locked down to legacy PBX
23:51.23_Sam--when was the last time you stayed at a hotel with a phone that had an LCD display?
23:51.29[av]baniwith the custom MWI and stuff
23:51.34rpmis it illegal to record phone conversations for personal use? im not a business..?
23:51.37*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
23:51.49Qwell[]rpm: Do you have permission?  Are you in a state that requires it?
23:51.50[av]banirpm: depends on your jurisdiction and who is calling
23:51.52AldoIt was configuraded but I don't Hear nothing
23:51.54Qwell[]Go see a lawyer...
23:51.54tronixrpm: depends on your local/state|provincial|regional laws
23:51.58rpmi am in canada, im calling dell.
23:52.07[av]baniah canada, go hog wild then
23:52.10[av]banino law there
23:52.11_Sam--lol
23:52.19austinnichols101you can listen in for QA purposes and then you're allowed to record if you hear something illegal
23:52.21AldoI think the problem are the codecs
23:52.23maskedtronix maybe i should reboot.... but when i modprobe zaptel i get this zaptel: Unknown symbol crc_ccitt_table
23:52.31Qwell[]Aldo: What version of *?
23:52.35Qwell[]1.1.2 is not valid
23:52.35tronixmasked: ah think that's easy fix. no reboot needed btw. hang on
23:52.37*** join/#asterisk bjohnson (n=bjohnson@i216-58-62-9.cybersurf.com)
23:52.38[av]baniif they threaten you tell them they can go shove their head up a mooses butt
23:52.48Qwell[]Do you mean 1.2.2?  If so, upgrade
23:52.56Aldo1.2.2
23:52.57hardwirecrap
23:53.00Aldov 1.2.2
23:53.01Qwell[]go upgrade
23:53.02maskedtronix k
23:53.03_Sam--[av]bani:  if you needed 50 fxs what would you buy?
23:53.05hardwirewhat are the upsides to 1.2.3 vs 1.2.1 :(
23:53.05Qwell[]come back when you're done
23:53.06QbYanyone know how to reset a Sipura SPA-2000 when you don't have the password??
23:53.08*** join/#asterisk sevardz (n=kynan@198.174.233.25)
23:53.09hardwirewell 1.2.4
23:53.11sevardzconnection. :P
23:53.11tronixmasked: look at README. has solution
23:53.14Qwell[]hardwire: less memory leakage
23:53.21hardwireIhate how this many stable releases just like.. happened
23:53.27[av]baniQbY: pick up fxo, pres **** and then 73738
23:53.32hardwireand their changelogs leave something to be desired
23:53.35Qwell[]hardwire: "release" versions
23:53.38tronixmasked: it's in the 'Brief F.A.Q.' section at the end. e-z fix tho
23:53.49[av]baniQbY: err FXS
23:53.52QbY[av]bani -- Its asking for a password
23:53.52hardwireQwell[]: you would think release meant stable :)
23:54.15[av]baniQbY: nice, its locked then. you might be able to provision it via tftp
23:54.18[av]banior dhcp
23:54.24brockj49464Question about SPA2100:  How do you find out how long the flash button on a phone is?  Is there a way to log hook switch?
23:54.30QbYyou got any instructions for that?
23:54.39[av]baniQbY: have you got a dhcp server?
23:54.42_Sam--who else makes a 24port fxs channel bank besides rhino?
23:54.44QbYyeah
23:54.55Qwell[]_Sam--: can get an adit
23:54.58znoGis there any difference between doing a quick tap on the hangup switch and pressing FLASH?
23:55.04[av]baniqby just a mo
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23:55.07justinuadtran, mainstreet
23:55.08Qwell[]znoG: not really
23:55.11_Sam--ty.
23:55.11maskedtronix breif faq is where?
23:55.14justinua bunch of people
23:55.34tronixmasked: end of README file in zaptel-1.2.3 directory
23:55.36justinufxs channel banks have been around for 30+ years
23:55.43justinumaybe 40
23:56.25_Sam--most play fine with asterisk?
23:56.43maskedtronix oh sorry missed u saying that
23:56.51tronixmasked: ha, no worries, mate
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