irclog2html for #asterisk on 20060416

00:05.29*** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com)
00:05.48Shotta30fiveSt, Petersburg.... Used to be broward county until I started USF
00:06.26Shotta30fiverene: What you think about shootwall firewall for the asterisk box
00:06.31*** join/#asterisk `Kevin (n=Kevin@64.243.236.10)
00:09.33rene-Shotta30five: i have never used it, people talk good things about it
00:10.02rene-but first try connecting with no firewall
00:11.27Shotta30fiveWill let you know my progress..
00:11.41nainHi
00:13.31*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
00:14.14*** join/#asterisk redcap1 (n=phez@redcap.xs4all.nl)
00:14.36*** join/#asterisk suma (n=suma@222.165.112.215)
00:15.19sumaWhere can I get the list of asterisk actions to use in asterisk IVR C Program
00:18.35DoktorGregok, pri question
00:18.48rene-hi nain
00:18.52DoktorGregi figured how to dig into MICS system, the proper menu system
00:19.05nainI am getting -- Transmitting RFC2833 on payload 101
00:19.06nainOuch ... error while writing audio data: : Broken pipe
00:19.06nainSegmentation fault
00:19.08DoktorGregmy PRI line..??? doesnt support d channel
00:19.16QwellDoktorGreg: Then it isn't PRI
00:19.30nainCan any one know why this call failed and asterisk crashed on h323 call, while sip is working fine
00:19.31DoktorGregno, in the MICS, it says its PRI
00:19.41DoktorGregbut for D channel it says... None
00:19.43QwellPRI has D channel
00:19.46QwellSo set one
00:20.56fileif anyone in here has bugs on mantis they want me to look at, or I replied to... well... say your number now or forever hold your peace and quiet!
00:22.03*** part/#asterisk rene- (n=rene-@dsl-201-128-115-107.prod-infinitum.com.mx)
00:22.37drrayis your PRI going to a channel bank first and then into your asterisk box?
00:22.41Qwellfile: close em all!
00:23.25fileQwell: marvelous idea
00:24.00DoktorGregno
00:24.13nainAny one can let me know what's wrong with my setup
00:24.21DoktorGregdirct from the csu/dsu unit from phone company
00:24.27nainwhile dialing h323 * crashed with this log
00:24.28nain-- Transmitting RFC2833 on payload 101
00:24.28nainOuch ... error while writing audio data: : Broken pipe
00:24.28nainSegmentation fault
00:24.28nainPK_Server:/etc/asterisk# Warning, flexibel rate not heavily tested!
00:24.46drraydoes it work without the d-channel?
00:24.52*** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
00:25.35wwalkerIf I downloaded a bad config file into a polycom 501, and I now want it to not use the contents of the config, how do I reset the phone
00:27.14wwalkerI'm stuck in a loop where it
00:27.40wwalkercomes back with "config file error"  "error is 0x4020"
00:28.24[TK]D-Fenderwwalker : fix the file and reboot
00:28.35filenooooo don't fix me
00:28.43wwalker:)
00:28.52Qwellfile: snip, snip
00:29.19wwalkerI want the phone to NOT download an app.  I want it to use the APP it has in flash.
00:29.20filego after Qwell! he's smaller, it won't matter!
00:29.26Qwellpfft
00:31.04Qwellblitzrage: where art thou?
00:31.33QwellIt's April 15th!  Why isn't Astricon registration open? :(
00:31.36wwalkerI don't have a sip.ld :(  so I gave it a config file without a app_file_path entry
00:32.16fileQwell: who knows!
00:35.34[TK]D-Fenderwwalker : You think you've screwed up your sip.ld load?
00:35.36Shotta30fiveRenee: Having it on the internet work
00:35.39Shotta30fivethanx you
00:36.14Shotta30fiveJust got to figure out what is going on with the router
00:37.21wwalker[TK]D-Fender: no, not yet.  I went back and added app_file_path and booted once (with no sip.ld to load) so now it seems to do Ok.
00:37.46wwalkerIt gets thru loading application, then loading sip.ld with no errors, but it then reboots
00:38.28wwalkerlooks like it made it this time... maybe...
00:39.48[TK]D-Fenderwwalker : Sounds like you should reset your <mac>.cfg file to something normal and get the firmware in the folder where it belongs
00:40.53wwalkerback to don't have the firmware :(
00:41.07[TK]D-Fenderwwalker : Which version were you on?
00:41.17wwalker1.6.2.0041
00:41.28wwalkerI get all the way to the Welcome screen.
00:41.38wwalkerIt shows the version.
00:41.39[TK]D-Fenderwwalker : Ok, which version would you LIKE?
00:41.46wwalker1.6.2.0041
00:42.07wwalkerI think sip.ld is fine.  it doesn't like something in the config file it pulled down
00:42.32*** join/#asterisk Anexs_and_Mai (n=Dartagna@pc-58-176-104-200.cm.vtr.net)
00:42.41wwalkerHow do I get the polycom to push it's config up to the server?
00:42.47*** join/#asterisk elg (n=fugalh@falcon.fugal.net)
00:43.05DoktorGregok now im getting somewhere
00:43.20DoktorGregi just made the phones print from * to mics
00:43.28DoktorGreger ring
00:43.40DoktorGregbut
00:43.50DoktorGregi cant do it a second time
00:44.07DoktorGregcli gives me a all channels busy error
00:44.42[TK]D-Fenderwwalker : typilcally it only updates the locally amde changes in <mac>-phone.cfg
00:44.54[TK]D-Fenderwwalker : As an override to base.
00:45.32wwalker<PROTECTED>
00:46.10[TK]D-Fenderwwalker : Though you really DON'T want the phone thinking it knows better... thats the POINT of provisioning them.
00:46.29luke-jr_Is there a website that explains what "SNFC CNTRL" and similar 'city names' are?
00:46.30wwalkerI agree, can't convince the client
00:46.45[TK]D-Fenderwwalker : Have you tried blunt trauma? ;)
00:47.10wwalker:)
00:48.48wwalkerback in an hour, dinner calls
00:57.13luke-jr_BTW, SellVoIP lets you use both IAX2 and SIP for calls ;)
00:57.30kamileonwhoa i found a FXS module sitting in a drawer!
00:57.34luke-jr_which is potentially useful for supporting reinvites between different outgoing services
00:58.27luke-jr_Is it possible to set 'use reinvites' in the dialplan?
01:01.21*** join/#asterisk jofre (n=jofre@201.2.192.43)
01:02.04luke-jr_in particular, I'd like to disable all reinviting for local destination calls (which are recorded) and reinvite anything goes back out over the net
01:02.41[TK]D-Fenderluke-jr_ : Nope... once a calls path is decided you can't jsut go and grab it back...
01:04.01SplasPoodQuestion, when using Agents /w AgentCallbackLogin and ackcall=yes, where do you define the announcement played *before* the callee hits '#'
01:06.20[TK]D-FenderSplasPood : I believe ackcall only applies to "AgentLogin" users
01:07.37[TK]D-Fendersince they're already "on the call" already as a warning.
01:07.58*** join/#asterisk angom_h (n=angom@red-corp-200.76.229.86.telnor.net)
01:08.01*** join/#asterisk hfb (n=hfb@adsl-69-231-83-94.dsl.irvnca.pacbell.net)
01:08.10Qwellagents are silly
01:08.20*** join/#asterisk k-man (n=jason@unaffiliated/k-man)
01:08.21QwellWhy would they be used over queue members?
01:08.31*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
01:08.47[TK]D-FenderQwell : Useful in cases where you want to trigger events what the agent is called I guess.
01:09.00*** join/#asterisk brockj49464 (n=brockj49@63.87.56.236)
01:09.05[TK]D-FenderQwell : that style was how I was going to implement "screen-pops" for mine
01:09.23*** join/#asterisk websae_ (n=icechat5@CPE-24-167-204-30.wi.res.rr.com)
01:09.41Qwellmanager can send for queue members too...
01:10.54[TK]D-FenderQwell : yeah, but thats an approach that'd require more programming skills.
01:11.24SplasPood[TK]D-Fender: nope, its the opposite.. it def works.. waits for me to hit '#', its just all slient before that...
01:11.50[TK]D-FenderSplasPood : that blows... hope you rigged the callerID before passing off the calls o you know what to do :)
01:12.17SplasPood[TK]D-Fender: After '#' it plays an announcement defining the name of the queue so people know how to react
01:12.24*** join/#asterisk phrog123 (n=francois@ns.menards.ca)
01:12.29SplasPoodI'd like something before the '#' that says "Please press '#' to accept this call"
01:14.15phrog123folks, with Asterisk at Home, can you tell me which SIP header on an inbound SIP connection from another asterisk box (not trunked) binds to the exten command in extensions.conf ?
01:14.33QwellTo: ?
01:14.55phrog123I've spent more than 4 hours today trying every possible permutation of DID
01:15.20mitchelocphrog123: what for?
01:15.24phrog123what I have in my log is that the other server sends the call as s@192.168.0.1
01:15.26SplasPoodQwell: Agents allow a given person to sit down at any phone and then take calls.. without just having all phones ring
01:15.40QwellSplasPood: like AddQueueMember?
01:16.07phrog123I want a rule for trapping that DID so that I can have a different IVR on my SIP account dialing into my * box than from my digium board
01:16.31SplasPoodQwell: Well I suppose you could put something together with that, but why use that OVER agents?
01:16.44QwellSplasPood: Because, Agents are lame :p
01:16.50Qwellnot as flexible, IMO
01:17.21phrog123It is as if the DID is not recognized by asterisk, yet the calls comes in and rings my IP phones just fines, but throught the default rule which is the same than a call coming over POTS and ringing my IP phones
01:17.25SplasPoodhrm...  what type of flexibility?
01:17.28luke-jr_[TK]D-Fender: this is before the path is decided
01:17.34luke-jr_eg, before I execute Dial
01:17.53*** join/#asterisk angom_h (n=angom@red-corp-200.76.229.86.telnor.net)
01:17.54phrog123Of course, I had to do the normal extensions.conf tweak (wonder why this is not yet enableable through the Web UI
01:18.02[TK]D-Fenderluke-jr_ : perhaps you could set up 2 peer entries, each with the same connection details, only different rules...
01:18.08luke-jr_:/
01:18.25luke-jr_hackish
01:18.33phrog123anyways, the DID does not work, so I want to trace the SIP flow and figure out what's missing in the SIP header so that I can check with the ISP who's providing me that DID
01:19.06luke-jr_phrog123: Ethereal?
01:19.29phrog123why?
01:19.45phrog123sip debug peer myispaccount spits out what I want
01:20.26mitchelocdoes ethereal cook and clean?
01:20.47Qwellluke-jr_: Wanna record me a pcap file? :p
01:20.51phrog123I just do not know what * binds on exten=>5555551212 = what in the SIP message ??? from: 5555551212@sip.sipprovider.com?
01:20.56phrog123Is this the from field?
01:20.58[TK]D-Fendermitcheloc : You've got to keep current with your plug-ins!
01:21.04Qwellphrog123: To:
01:21.06macTijnmitcheloc: yes, and it vacuums too!
01:21.36luke-jr_Qwell: no, why?
01:21.39phrog123Qwell: to doesn't make sense, the to is the telephone number being dialled, not the DID
01:21.42Qwellluke-jr_: because I need one, heh
01:21.46luke-jr_Qwell: of what?
01:21.46Qwellphrog123: duh?
01:21.58QwellYou don't match the number dialed to the From:
01:22.08Qwellluke-jr_: about 10 seconds of a gsm (or g729) rtp stream
01:22.20phrog123ok, assuming this is the case, is it what's at the left of the @
01:22.29luke-jr_Qwell: why can't you do that yourself? O.o
01:22.40phrog123does it matter what's right of the @?
01:22.43Qwellphrog123: exten => _NXXNXXXXXX/5555551212,1,Blah()
01:22.49Qwellluke-jr_: don't know how :(
01:22.59luke-jr_Qwell: ...
01:23.07QwellI'm an ethereal newb :p
01:23.21*** join/#asterisk TTT_Travis (n=Travis@bal-broadband2-ws-14.dsl.airstreamcomm.net)
01:23.32luke-jr_Qwell: the problem is, I don't know enough about RTP to know that keys/passwords won't be crackable from it =p
01:23.56luke-jr_and also that I use ulaw...
01:24.00Qwellmeh :p
01:24.04phrog123if the * box on my ISP's side sends me a to: s@x.y.z.w (public IP address), can I assume that this is where the problem is?
01:24.13phrog123wtf is s@ ?
01:24.17Qwellexten s
01:24.25luke-jr_phrog123: that means you didn't give them an extension to call
01:24.27QwellThen thing you're adding at the end of your register => line
01:24.31TTT_Travishi guys, I'm interested in learning about asterisk so I though I'd take the beginners path and install Asterisk@home, I have a Rockwell Voice Modem do you think it will be possible to make this work with Asterisk without too much effort?
01:24.44QwellTTT_Travis: As what?
01:24.50Shotta30fiveI think i found out why my Asterisk Box was not working with external client
01:24.51Qwella modem?  sure
01:25.03luke-jr_when I register, how can I specify the username the remote side should authenticate with when placing calls?
01:25.18Shotta30fiveThe router I was using was a Linksys ATA that I wasn't using
01:25.33TTT_TravisQwell well I want to connect it to my phoneline in my house and then for example beable to pickup the phone and dial an extension and stuff
01:25.38luke-jr_TTT_Travis: a modem will only work for a phone line, not for a phone itself
01:25.40QwellTTT_Travis: no
01:25.44Shotta30fiveSo make a quick Smoothwall Firewall and now it works like a charm
01:25.48QwellTTT_Travis: It'll be a modem, thats about it
01:25.55Qwellunless you can write zaptel fxo drivers for it
01:26.15TTT_Travishow come with like windows I can dial numbers on it and it will dial it on my phoneline?
01:26.17luke-jr_Qwell: don't those exist for Rockwell?
01:26.32QwellTTT_Travis: because there are drivers
01:26.35tainted-anyone know how to speed up boot process with polycom 301s
01:26.35*** join/#asterisk Flauto (n=zhao@adsl-75-3-189-92.dsl.chcgil.sbcglobal.net)
01:26.47luke-jr_TTT_Travis: will it transmit voice on windows too?
01:26.47phrog123what does it mean if I'm getting an invite from another * box which is s@IP.IP.IP.IP ?
01:26.59TTT_Travisluke-jr_ not really sure
01:27.02TTT_TravisI think it would
01:27.05TTT_Travissince its a voice modem
01:27.17luke-jr_what's that mean?
01:27.24luke-jr_modem != voice
01:27.25Qwellit means they can charge $45 for it
01:27.28luke-jr_they're kinda exclusive
01:27.51TTT_Travisit says on the modem that its a Voice/Fax Modem
01:28.01QwellIt's a winmodem.  Nothing more, nothing less
01:28.06phrog123if my register line is not a telephone number, but rather a username:passsword@sip.sipprovider.com? Does this mess up asterisk insofar as parsing a DID?
01:28.08TTT_Travisso its impossible?
01:28.10QwellWhatever extra "stuff" it can do, is done in software
01:28.16QwellTTT_Travis: no, you just need to write a driver
01:28.23TTT_Travisso impossible ;)
01:28.30luke-jr_TTT_Travis: not impossible if its a winmodem
01:28.36phrog123the problem is that this provider, the username is not the 10 digit number, but rather an account name
01:28.58TTT_Travisluke-jr_ what do you mean by winmodem?
01:28.58luke-jr_TTT_Travis: if nobody's written a driver, you can either write one yourself or hire someone to
01:29.08Qwellphrog123: It will send the DID in the To: header
01:29.16luke-jr_TTT_Travis: software modem
01:29.19TTT_Travisk
01:29.28TTT_TravisI don't know if it is
01:29.39QwellTTT_Travis: Did you buy it after 1999?
01:29.42TTT_TravisI am new to think kind of stuff
01:29.50TTT_TravisQwell I am guessing right around 2000
01:30.02TTT_TravisI got this computer from someone that didn't want it
01:30.07TTT_Travisand that was the card that was in there
01:30.08QwellThen there is a fairly high chance that it is
01:30.18*** join/#asterisk somegeek (i=levin@unaffiliated/somegeek)
01:30.23Qwelljbot: tell phrog123 pastebin
01:30.26luke-jr_channels/chan_modem_aopen.c: * A/Open ITU-56/2 Voice Modem Driver (Rockwell, IS-101, and others)
01:30.58TTT_Travisfair chance that it is what?
01:31.03Qwella winmodem
01:31.15TTT_Travisluke-jr_ whats that about rockwell?
01:31.19TTT_Travisit has a rockwell chip on it
01:31.25luke-jr_TTT_Travis: if you want to buy a non-winmodem, you'd need to spend extra money and time looking for it
01:31.41luke-jr_TTT_Travis: apparently, the chan_modem_aopen module supports Rockwell modems
01:31.50Qwellchan_modem is dead
01:32.00luke-jr_...
01:32.20luke-jr_dead how?
01:32.20QwellIt won't be there in 1.4
01:32.41TTT_Travisso there is a chance it might work?
01:32.45luke-jr_what replaces it and provides the same features?
01:32.57Qwellluke-jr_: chan_zap?
01:33.08luke-jr_Qwell: chan_zap supports regular winmodems?
01:33.12Qwellno
01:33.29luke-jr_so chan_modem is being removed because it cuts into Digium's sales?
01:34.03QwellI'm pretty sure zap supports other hardware
01:34.12luke-jr_what version will TTT_Travis be stuck with if he wants to use chan_modem?
01:34.19Qwell1.2
01:34.28TTT_Traviswill the latest Asterisk@Home work?
01:34.34marcus2zap supports intel winmodems
01:34.37luke-jr_TTT_Travis: @Home gets no support here
01:34.45TTT_Travisis there a channel for it?
01:34.51luke-jr_see topic
01:34.59luke-jr_tho it's a waste of time, IMO
01:35.07TTT_Traviswhy?
01:35.17marcus2but there has definitely been some questionable behavior in the past with regards to digium natively supporting cards that they dont get profits from
01:35.18luke-jr_it's user friendly
01:35.25TTT_Traviswhats so bad about that?
01:35.37luke-jr_it's not smart-people friendly
01:35.46marcus2one big reason that it sucks is that it is based on centos
01:36.13QwellThe reason *@~ is crap, is because AMP is crap, as is phpmyadmin
01:36.49Qwellphrog123: 1) See channel topic.
01:36.56Qwellphrog123: 2) Unless you're paying me, don't msg me
01:37.05TTT_Travisthe rockwell chip says this:       RCV336ACF/SP     R6749-21    ? Rockwell 96 9742 B31697-4 MEXICO
01:37.14TTT_Travisso its a pretty old card
01:37.17TTT_Travisits not PCI
01:37.28QwellTTT_Travis: Should've said it wasn't PCI before
01:37.30*** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz)
01:37.31TTT_Travisits the long black slots
01:37.36QwellIt's almost definitely not a winmodem
01:37.39TTT_Travisk
01:37.42Nuggetthe "long black slots".
01:37.46Nuggetoof.
01:37.46TTT_Travislol
01:37.49iceyphey guys, been some time sicne i setup voicemail... how does one add a new mailbox to the system? i justed added it to voicemail.conf
01:37.52TTT_Travisthey were before my time
01:38.01luke-jr_ISA
01:38.06TTT_Travisyeah
01:38.06Qwelliceyp: That's all
01:38.07TTT_Travisthats it
01:38.16iceypQwell mmm, doesnt appear to be working
01:38.22QwellDid you reload?
01:38.27iceypyea
01:38.35TTT_Travisso what would I need to use this Chanzap driver?
01:38.36luke-jr_but
01:38.44luke-jr_if it's not a winmodem, how does it claim 'voice'?
01:38.46QwellTTT_Travis: Get real telephony hardware
01:38.47iceyp<PROTECTED>
01:38.52iceyp<PROTECTED>
01:39.10TTT_TravisQwell I'm working with what I got for now
01:39.22TTT_Travisbut it says Fax and Voice modem card
01:39.22QwellTTT_Travis: well, it isn't going to work
01:39.29TTT_Travisok
01:39.37TTT_Travisso what is the cheapest card I can get?
01:39.38iceypcan i start a mailbox with a 0 on the beginning?
01:39.45phrog123if My ISP is giving me a To: <sip:pro01473@sip2.isptel.ca> in the SIP header rather than 5555551212@sip2.isptel.ca, does this mean that the extensions.conf entry should be more like exten => pro01773@,1,SetVar(FROM_DID=pro01473 rather than exten => 8196018096,1,SetVar(FROM_DID=8196018096)
01:39.47Qwelliceyp: sure
01:40.00luke-jr_TTT_Travis: you could get a Linksys PAP2-NA for about $60
01:40.06TTT_Travisummm
01:40.10iceyp099742910 => 1111,Barry SIP Phone,barry@unix.co.nz
01:40.12TTT_Travisyeah thats way more then I want to spend
01:40.21QwellTTT_Travis: telephony isn't cheap
01:40.25TTT_Travishttp://voip-forum.tmcnet.com/voip-forum/forum/forum_posts.asp?TID=4300
01:40.29QwellJust because the software is free, doesn't mean hardware is
01:40.33TTT_Travisthat says
01:40.34TTT_Travis"Laptops4me.com sells the same cards for US $7.70 each, which is hard to beat. "
01:40.43TTT_Travisbut when I click on the link its dead
01:41.03iceypahh i think its the context i using :)
01:41.23luke-jr_Qwell: also, chan_zap supports Intel because they're the same as Digium hardware, IIRC
01:41.28TTT_Travishow about X100P card from Digium
01:41.35luke-jr_and it only supports intel when you hack it
01:41.39QwellTTT_Travis: Digium hasn't sold those in a long time
01:41.57iceypok, all fixed
01:42.07TTT_TravisQwell but they do work with Asterisk
01:42.09TTT_Traviscorrect?
01:42.21TTT_Travisyep
01:42.40TTT_Travisand they're like $15
01:44.50phrog123so folks, if account name is not a telephone number, will exten => accountname,1,... work?
01:44.54iceypI have an issue from my cisco 7912 that when i make a call to my remote asterisk box for time, audio, voicemail or anything, it gets through and has no sound till i press something
01:45.56Qwelliceyp: skinny?
01:45.58luke-jr_TTT_Travis: http://www.laptops4me.com/product_info.php/modem/all-56k-modems/p/i-bis-v-92-pci-intel-chip-voice-modem/cPath/176_239/products_id/5232
01:46.03iceypumm sip client
01:46.09iceypi beleive
01:46.26iceypif i make a call to an external number, like pdsn shes sweet
01:46.31luke-jr_TTT_Travis: though I'd give the Rockwell chan_modem a try first anyway
01:46.32TTT_Travisluke-jr_ that doesn't work
01:46.35TTT_Travisk
01:46.36TTT_TravisI will
01:46.40luke-jr_TTT_Travis: what doesn't work?
01:47.06TTT_Travisthat link
01:47.09luke-jr_no?
01:47.12luke-jr_does for me
01:47.27TTT_Travisnow it does
01:47.28iceypQwell yeah its sip based
01:47.34TTT_Travisso that card will work?
01:47.44luke-jr_TTT_Travis: looks like it
01:47.49luke-jr_but no guarantees
01:48.07TTT_Travisyeah
01:48.13TTT_Travisfrom what I see the intel based ones do
01:48.25luke-jr_it has an intel chipset, note
01:48.33*** join/#asterisk op3r (i=op3r@210.4.31.234)
01:48.34TTT_Travisthey show up as generic clones to the X100P
01:48.36luke-jr_looks like it's merely rebranded
01:48.42TTT_Traviswell if my rockwell one fails then I will get that one
01:48.43op3rdoes anyone know the pricing of aheeva?
01:48.53TTT_Travisand hey if that one doesn't work I'm only out $10
01:49.24Qwelland if it does work, you'll have a poorly performing PBX
01:49.33Qwelland WE'LL have to support you
01:49.57iceypQwell it would appear i get no voice until the pbx hears something, if i wisper or blow into the handset / speakerphone then voice comes through
01:50.05TTT_Traviswhat do you mean poorly performing?
01:50.12TTT_Traviswill it just be bad quality?
01:50.15QwellTTT_Travis: cheap, as in $10
01:50.19QwellIt's a crap card
01:50.32TTT_Travisjust so I can hear whats coming
01:50.37luke-jr_X100P is a crap card
01:50.37TTT_Travisthis is just for learning purposes
01:50.38luke-jr_?
01:50.49Qwellluke-jr_: umm...yes
01:50.55op3rQwell: do you know the pricing for Aheeva?
01:51.04*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
01:51.04luke-jr_what makes it crap?
01:51.05QwellTTT_Travis: So, you're using *@~ to "learn" also?
01:51.16TTT_TravisI just want to play around
01:51.34TTT_Travisand according to what I've read Asterisk@home is the easiest way to start
01:51.38Qwellluke-jr_: the hardware?
01:51.41luke-jr_TTT_Travis: learn how to use real Asterisk, not some dumb UI ;)
01:52.17Qwelliceyp: You must have the silence threshold thing
01:52.27*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
01:52.56iceypQwell on the phone it's self?
01:52.56TTT_Travismeh
01:52.58iceyplet me look
01:53.01TTT_TravisI'll start with the ui
01:53.05TTT_Travisthen move onto the commandline
01:53.27Qwelliceyp: in voicemail.conf
01:53.28[TK]D-FenderTTT_Travis : It will NOT teach you * at all, just how to fill in a couple of stupid blank lines on a web-form.
01:53.54iceypmmmm
01:53.55TTT_Travisif it works thats enough
01:53.56De_Mon[TK]D-Fender well, to be fair. You can create the dialplan and then SEE how to do it manually.
01:54.06iceypsilencethreshold=128
01:54.08mitcheloc[TK]D-Fender: be nice now ;)
01:54.16QwellDe_Mon: You've obviously never seen an AMP config
01:54.19TTT_Travisthis way I can still digg around the config files if I need too
01:54.20QwellIt's complete shit
01:54.30op3rI learned * by installing it and messing it up and editing confs on a production servers
01:54.32op3r:(
01:54.32[TK]D-FenderTTT_Travis : Yeah, in all likelyhood it'll work, just forget about learning anything of value or being able to truely control your PBX.
01:54.35Qwelltakes about 50 lines to do anything at all
01:54.40op3rand I am still 0 clue about it thougj
01:55.00iceypQwell  how can i turn it off
01:55.02De_MonQwell wow.. ok nevermind
01:55.15QwellDe_Mon: It's like 5 layers deep in pointless macros, etc
01:55.17[TK]D-FenderDe_Mon : Yeah, and spend HOW long picking apart the AGI's, etc?  And how is it that you'd even understand what it all means without oing a lot of it yourself OUTSIDE of AMP anyways?
01:55.45Qwellyeah, dumb ass AGIs too
01:55.57mitchelochey now, i'm working on a gui, they aren't *all* bad!
01:56.09Qwellmitcheloc: Yes they are.  Even the one I'm writing :p
01:56.14TTT_Travisso how hard is chan_zap to setup?
01:56.18mitchelocnoo! mine isn't!
01:56.19TTT_TravisI just install zaptel and ?
01:56.22[TK]D-FenderQwell : yup.... KNOWELDGABLE * users get lost in there... its like spaghetti code with meatballs and way too much parmesan.
01:56.26QwellTTT_Travis: Eithout hardware?  impossible
01:56.34QwellWithout*
01:56.39[TK]D-FenderTTT_Travis : Get compatible hardware first
01:56.47iceypQwell  how can i turn the silence thing off?
01:56.49iceypset it to 0?
01:56.53[TK]D-FenderTTT_Travis : They Rockwell wonder of yours is USELESS to *.
01:56.55Qwelliceyp: no, that's fine as it is
01:56.59[TK]D-Fenderthat*
01:57.13Qwelliceyp: minimum length?
01:57.17iceypQwell  why do i have to blow or make a sound into the mic before it talks to me
01:57.19iceypQwell  10
01:57.24QwellThat's why. :)
01:57.34QwellIt removes the first part, if it's 100% silence
01:57.45iceypmaxsilence=10
01:57.46iceyp?
01:57.52Qwellno
01:58.01QwellSo, your 2 seconds of blowing, after 8 seconds of silence, is only counted as 2 seconds
01:58.14QwellI think that was how it worked
01:58.27Qwellor maxsilence
01:58.27iceypi can wait 20 seconds with nothing till i make any sound into the handset then it talks to me
01:58.46Qwellminmessage
01:58.54Qwellminmessage and/or maxsilence can affect it
01:59.15iceypmmm
01:59.42iceypguess i just need to tell people that when they call the pbx for anything, just to make some sort of sound into the handset?
01:59.50Qwellor turn off those options
02:00.01Qwelland, no, those are only for voicemail
02:00.33op3rdoes anyone have any idea of the pricing of Aheeva?
02:00.39iceyp;maxgreet=60got them all ;commented
02:03.04TTT_Travis[TK]D-Fender why is it useless?
02:05.04luke-jr_TTT_Travis: he's assuming chan_modem won't work with it, which is probably true
02:05.12luke-jr_but like I said, I'd give it a chance just in case
02:05.18TTT_Travisyeah
02:05.24TTT_Travisif not I'll get the $10 one
02:05.29TTT_TravisI just need something that works
02:05.36TTT_Travisdon't care how crappy the quality is
02:05.39*** join/#asterisk rowter (n=Silver@201.138.157.112)
02:06.04rowteranyone has heard of dr nolazco corpus files for sphinx for spanish recognition?
02:06.29iceypthanks for your help guys
02:06.30iceypcya
02:07.28QwellTTT_Travis: I never said the quality was crappy
02:07.36*** join/#asterisk esculapio_ (n=ESCulapi@142stb68.codetel.net.do)
02:07.39QwellI said the card is.  Meaning, it'll BARELY work, if at all
02:07.57QwellDon't count on having any sort of hangup detection.
02:08.09QwellSo, anticipate 24 hour long voicemail messages
02:12.28*** join/#asterisk synaptic (i=synaptic@68.62.176.196)
02:13.14hinckcTTT_Travis: if you want to make a "science project", just do sip only, and get a sip service provider for 20 bucks a month.  then when your frankenstein is ready, just forward your regular # to the SIP #.
02:20.05luke-jr_hinckc: if I understand correctly, he wants other phones to be on the line too
02:20.57[TK]D-FenderTTT_Travis : What do you want out of *?  Then we'll see about what best help you get there.
02:21.43*** join/#asterisk coppice (n=chatzill@37.162.17.210.dyn.pacific.net.hk)
02:24.51TTT_Travis[TK]D-Fender just something to fill my spring break boredom
02:24.57TTT_Travislike maybe making a voice prompt menu
02:25.41[TK]D-FenderTTT_Travis : For your existing line?
02:25.56TTT_Travispossibly?
02:26.03[TK]D-FenderTTT_Travis : What would use as phones on your system?
02:26.03TTT_TravisI really haven't looked into it much
02:26.17TTT_Traviscan you just pick up a phone and dial an extension?
02:26.52[TK]D-FenderTTT_Travis : With what is the question.  You can't jsut plug a phone into that modem of yours and it'd be almost miraculous if it was uable for even your line.
02:27.23[TK]D-FenderTTT_Travis : You'd need a special interface in order to use your analog phone with *.  Either ana ATA or special PCI card.
02:27.23TTT_Travisk
02:27.24[TK]D-Fenders/ana/ATA
02:27.28[TK]D-Fenders/ana/ATA/
02:27.33[TK]D-Fendergrrr
02:28.02[TK]D-Fenderclose enough.  either way an ATA will run you about $70 (the best way to go), and the PCI solution $130 +/- (I think)
02:29.52TTT_Travismaybe I will just use it as sip phone over my network
02:29.57TTT_Travisor something
02:30.39hinckcyou can get a cheap (but functional) sip phone for ~50 bucks.
02:30.56TTT_Travisdid I say sip?
02:30.58TTT_TravisI mean't soft
02:31.12hinckcoh, sure... even cheaper
02:32.05*** join/#asterisk suma (n=suma@222.165.112.215)
02:32.46hinckcit just doesn't really feel like a PBX until a _phone_ rings... :)
02:33.35TTT_TravisI wonder what my school uses for a PBX
02:33.45TTT_Travisits not asterisk since they've had it for a long time
02:34.11mitchelocreplace it!
02:37.25*** join/#asterisk inv_Arp (i=junya@adsl-11-225-195.mia.bellsouth.net)
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02:39.28wwalkerOK, I can call my Polycom501 from Asterisk (Dial 510) and it rings and gets sidetone.  But it won't dial out.  It doesn't send anything to asterisk (sip debug show nothing)
02:39.52wwalkerIt's registered (sip show peers show address and 68 ms).  What am I missing?
02:41.19*** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com)
02:42.48wwalkerOK, I see that I'm getting sip stuff now. "Failed to authenticate user "SoundPoint IP""  I set the user for the line to 510...???
02:53.13*** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36)
02:54.36MoutaPTAny one could explain me or guess, why using Xlite if i'm in a call and choose music on hold ant then just click to hangup, Xlite presents the call like finish and it stills runing on asterisk?? The only way to finish this was soft hangup ??
02:55.37DoktorGregoh i think i found it...
02:55.56DoktorGregI was mapping pri channels onto a pseudo channel
02:56.10DoktorGreg???
02:56.19DoktorGreghow do i tell if i have a pseudo channel?
02:57.04MoutaPTAny one expert with PRI? any way to hang up a call automatically if the called party hangs the call?
02:57.54DoktorGregI am working with pri....  But it sounds like you have more connectivity than i do right now...
02:58.05MoutaPTi'm using E1
02:58.20DoktorGregim on t1
02:58.28DoktorGregno fair you get more channels!
02:58.35MoutaPTeverything is fine, only problem is some of my users seem to not hangup correctly the calls!
02:58.37[TK]D-Fenderwwalker : bad user:pass
02:58.49MoutaPTi need to finnish the calls if the called party hangs
02:59.12[TK]D-FenderMoutaPT : typically the call IS terminated immediately.
02:59.31[TK]D-FenderMoutaPT : Thats the point of digital signalling, so you don't have to "guess"
02:59.39MoutaPTi have * behind a legacy PBX
02:59.46MoutaPTand i've been tracing
02:59.49wwalker[TK]D-Fender: thx
02:59.51MoutaPTthe operator sends
02:59.58MoutaPTPROGRESS message
03:00.03MoutaPTCause Code =16
03:00.13MoutaPTand asterisk answers with cause code =98
03:01.02MoutaPTISDN says that cause code= 98 is This cause is sent when the equipment sending this cause has received a message which includes the information elements not recognized because the information element identifier is not define or it is defined but not implemented by the equipment sending the cause. However, the information element is not required for the equipment sending the cause to process the message.
03:01.56MoutaPTi got a technician with equipment to watch the ISDN packets between asterisk and old PBX...
03:02.04MoutaPTany one could help me?
03:02.51*** part/#asterisk apple (i=appleboy@about/cooking/nakedchef/apple/tarts)
03:05.19MoutaPTDoktorGreg any troubles with your PRI?
03:06.15MoutaPTmy mistake cause code=98 is :
03:06.18MoutaPTThis cause indicates that the message received is not compatible with the call state or the message type is non-existent or not implemented.
03:06.58DoktorGregi have some sort of pseudo channel driver loaded
03:07.08MoutaPTthat one is Ztdummy
03:07.12DoktorGregand its messing with how the channels are mapped by zaptel
03:07.15MoutaPTare you with kernel 2.6?
03:07.23DoktorGreg2.4
03:07.28MoutaPTedit your zaptel.conf
03:07.37MoutaPThave you done it ?
03:07.48DoktorGregits not being loaded in zaptel
03:08.07MoutaPTwhich card you have?
03:08.14DoktorGreg205
03:08.23MoutaPTdigium?
03:08.24DoktorGregxorcom rapid distro
03:08.26DoktorGregyah
03:08.35MoutaPTdigium?
03:08.38*** join/#asterisk websae2k (n=icechat5@CPE-24-167-204-30.wi.res.rr.com)
03:08.47DoktorGregdigium card, the 205
03:08.59DoktorGreg2 pri [ports at 5 volts
03:09.01MoutaPTok, i've only experience with TE110P
03:09.27MoutaPTbut i can tell you that for me worked fine editing zaptel.conf
03:09.37MoutaPTztcfgv -vvvv
03:09.38DoktorGregediting editing...
03:09.56MoutaPTmodprobe
03:10.05MoutaPTgenzaptelconf
03:10.16MoutaPThave you done all this stuff?
03:11.33MoutaPTAny one here has tried Mitel Hardphone with Asterisk?
03:11.49DoktorGregive done all sorts of stuff
03:11.52QwellMoutaPT: Is it SIP?
03:11.57DoktorGregthats the thing with asterisk
03:12.02DoktorGreger linux
03:12.09DoktorGregthere is always more stuff to try
03:12.13MoutaPTyes
03:12.13QwellDoktorGreg: You have to know what you're doing, to config stuff? :P
03:12.19QwellMoutaPT: Then it should work fine
03:12.38MoutaPTi will try it on monday:)
03:13.04MoutaPTcan u tell me why with xlite i seem to loose control of call if i put it on hold?
03:13.22MoutaPTthen i won't get control of the call any more...
03:13.23Qwellbecause they want you to use the pro version
03:13.47QwellI think they introduce random bugs, and stupid config options to the free version
03:13.49*** join/#asterisk rickb|server (n=none@cpe-71-66-110-248.neo.res.rr.com)
03:13.53rickb|serverHello. :)
03:13.54MoutaPTr u sure? i need to explain my boss...
03:14.03Qwellwell, I'm paranoid
03:14.13MoutaPThe though we could get it working with softphone
03:14.28rickb|serverI am really new to PBX and asterisk, is there a way that someone could help me out with just a few things?
03:14.29MoutaPTi've 70 users... and the problems are arriving...
03:14.37Qwellsoftphone for 70 users?
03:14.38Qwellugh
03:14.43MoutaPTi think hardphone would help me
03:14.45MoutaPTa lot
03:14.50Qwellrickb|server: Only if we know what those things are - ie; ask a question
03:14.51MoutaPTi get all the emails....
03:14.57MoutaPTcomplaining...
03:14.59MoutaPTand so on...
03:15.01rickb|serverWell.. Ok
03:15.06MoutaPThard to manage... 70 users
03:15.11QwellMoutaPT: not really
03:15.25MoutaPTsjphone seems to get critical error
03:15.45MoutaPTand xlite seems to loose control of the call if u click to music on hold...
03:16.00rickb|serverI have a POTS line in my house.. So I run the PSTN line from my house into the server running Asterisk, then that Asterisk Box, (if setup properly) will pickup those calls? And direct them to whereever I set?
03:16.02MoutaPTQwell not really what?
03:16.07Qwellnot really hard
03:16.27Qwell70 users should be a snap
03:16.37MoutaPThow do u explain this calls...
03:16.44MoutaPTlike zombie
03:16.45MoutaPTcalls
03:16.58Qwellxlite sucks :p
03:17.03MoutaPTfor a mistake users click music on hold then click to hangup
03:17.10MoutaPTxlite presents no call running
03:17.17MoutaPTand they are running
03:17.22MoutaPTi see it on CLI
03:17.37MoutaPTimagine you have 70 outbound calls like this
03:17.44MoutaPTyou are paying!!!!
03:17.57MoutaPTwhich SIPsoftphone do u recommend me?
03:19.36rickb|serverI am using the Rapid Xorcom version of the Asterisk. :) It is nice from what I can tell..
03:20.10Qwell70 calls?  I wouldn't USE a softphone
03:20.18QwellIf the company can't afford hardware, they can't afford my services...
03:20.27MoutaPTOk now i understand you!
03:20.40Qwelland if they're just too cheap...screw them
03:20.48Qwellsoftphones do not a production PBX make
03:21.01MoutaPTyou r right , i though u were saying to me 70 users with softphone were peanuts
03:21.10Qwellit is
03:21.16rickb|serverI am running the newest version of the Rapid, I tried to use the Asterisk Flash Operator Panel, It prompts for a password, I don't know it and consulted google for 2 hours to find the default. Any ideas?
03:21.33MoutaPTpassw0rd
03:21.40Qwellrickb|server: maybe "asterisk"?
03:21.52Qwellor "password"
03:21.55Qwellor a blank string
03:22.04MoutaPTQwell is peanuts with sofphone???
03:22.12MoutaPTwhich softphone?
03:22.23mitchelocrickb|server: why not ask xorcom?
03:22.26QwellMoutaPT: doesn't really matter, because it's stupid to do
03:24.02MoutaPTit depends... if you an IT company every one with its own workstation...
03:24.13MoutaPTthey thing it is good idea...
03:24.17MoutaPTthink
03:24.20Qwellwell, it isn't
03:24.38MoutaPTyeah i got it now :( unfortunately...
03:25.05MoutaPTIAX trunks working perfectly, fax to email too
03:25.15MoutaPTonly problem are the softphones.
03:25.15rickb|serveryeah.. I tried.
03:25.30rickb|serverI am just setting up a third party program for admining
03:25.46Qwellrickb|server: passw0rd, like MoutaPT said
03:25.55rickb|serverI tried that to. :)
03:25.59rickb|serverIt didn't work
03:26.10rickb|serverI tried all the configs to see if there were entries to..
03:26.12rickb|server:(
03:26.23MoutaPTcheck your .conf files
03:26.35MoutaPTsomewhere probably u will find it
03:26.40rickb|serverYeah
03:27.21MoutaPTany one could say me if i can by BRI connection to my operator and use  a PRI card?
03:27.38QwellMoutaPT: no
03:27.38MoutaPTit's ISDN so...?
03:27.43rickb|serverThe default password for asterisk its self, how do you set it? or is it allready set?
03:27.52Qwellrickb|server: There is no password for asterisk
03:27.57rickb|serverok. :)
03:28.18MoutaPTQwell why not do u know? did u try it?
03:28.34QwellMoutaPT: It won't work
03:29.03MoutaPTok
03:29.13Qwellbesides, a BRI card would be cheaper
03:29.22MoutaPTit is not so much..
03:29.31MoutaPTTE110P is around 500Euros
03:29.38MoutaPTand BRI could be 400
03:29.39drraygovarion
03:29.41QwellThat's sad
03:29.43drrayhas a cheaper tor
03:30.19MoutaPTwhich hardphones u ser Qwell?
03:30.24Qwellser?
03:31.45MoutaPT?
03:31.50MoutaPTsorry
03:31.51Qwellexactly
03:32.04Qwellplz2be using proper english
03:32.15MoutaPTwhich hardphones do you use?
03:32.21Qwellmuch better
03:32.23QwellCisco
03:33.14MoutaPTok, most of the people i talk they advice me cisco
03:33.23websae2kcisco for speaker phone :)
03:33.31drrayonce it's working the 7960 is a damn fine phone
03:33.31SplasPoodpolycom for speaker phone
03:33.38brookshirepolycom designed cisco's speaker phone ;)
03:33.39websae2kgrandstream works quite nicely, polycom even better, and then cisco
03:33.39QwellIt's the same speaker phone :p
03:33.42[TK]D-FenderPolycom is typically a cheaper and equal choice.
03:33.53websae2ki think 7960 has great speaker phone
03:33.57websae2kyes exactly
03:34.09SplasPoodI just bought a few Grandstream BT-101s...  great value for $48/ea
03:34.20drrayif the polycom lets you dowload firmware without paying, tehn I'd go that way
03:34.30SplasPooddrray: previous revision, yea
03:34.35SplasPoodnot the latest, but one behind
03:34.49SplasPoodyour vendor should be able to hook you up /w the latest firmwarez tho
03:35.03MoutaPTeasy one question, could you tell me what for is the dialparties.agi ?
03:35.11rickb|serverWhat is a good free gui web based administration program to use?
03:35.18QwellMoutaPT: To dial...parties
03:35.23Qwellrickb|server: no such thing
03:35.25drrayrick - winvi
03:35.40rickb|serveri'l check it out
03:35.57MoutaPTQwell, but wouldn't be better to just use a simple macro to dial ?
03:36.04QwellMoutaPT: I don't know
03:36.17[TK]D-FenderMoutaPT : Thats an AMP/FreePBX script.  Not the sort of thing to speak of here...
03:36.26Qwellugh
03:36.28Qwellfigures
03:37.33MoutaPTok, Does any one could answer me how to make asterisk to hang up a call when the called party hangs, i'm using PRI.
03:38.08VoIPMastaMoutaPT: it should hang up by itself
03:38.36MoutaPTI have PSTN---OLDPBX----Asterisk ,  and it doesn't happens...
03:38.52VoIPMastamaybe something is wrong in your oldpbx
03:38.56MoutaPTif i dial a local extension on OLDPBX, it happens
03:39.02VoIPMastawhy don't you plug your pri directly into asterisk?
03:39.05MoutaPTbut if i dial to outside world...
03:39.27VoIPMastamy best guess would it be that your old pbx isn't handling the call signaling correctly
03:39.49MoutaPTmy users just start listenning busytones...
03:40.06VoIPMastayup, there's a signaling problem in your oldpbx
03:40.07MoutaPTbut the call stills there...
03:40.30MoutaPTi got there a technician watching this...
03:40.54MoutaPTand i found the asterisk receives PROGRESS , CAUSE CODE=16
03:41.00MoutaPTwhen the called party hangs
03:41.14MoutaPTthen asterisk answers with cAUSe CODE=98
03:41.18MoutaPTThis cause indicates that the message received is not compatible with the call state or the message type is non-existent or not implemented.
03:41.41MoutaPTany test i can make with the local extensions
03:41.44MoutaPTon old pbx
03:41.59MoutaPTto show my boss that oldpbx is the problem?
03:42.30Qwellremove it from the loop
03:43.02brookshireso why does asterisk support the jpeg codec?
03:43.04brookshireheh
03:43.15Qwellbrookshire: pr0n for the 7970, duh
03:43.22mitchelocit's good a compressing audio?
03:43.37Qwellpr0n IVR!
03:43.39Qwellomg, brb
03:43.51brookshirepush one for pr0n
03:43.54coppiceright. a picture paints a thousand words, so JPEG compression is excellent
03:43.54mitcheloclol wash your hands before you come back
03:43.54brookshirepush two for pr0n
03:44.01brookshirepush three for pr0n
03:44.04Qwellbrookshire: no, no, no
03:44.09Qwellbrookshire: Seen the jukebox AGI?
03:44.13brookshireno
03:44.18Qwellit's in trunk
03:44.28Qwelljukebox menu thing
03:44.37brookshireahh.. for mp3?
03:44.40Qwellyeah
03:44.49brookshirethat's so pre-stable 1.0
03:44.51mitcheloc*wonders how the jukebox agi got in*
03:44.52Qwellthat could be so easily modified for chan_skinny, pr0n.agi :D
03:45.07Qwellmitcheloc: because people tested it, and it worked well
03:45.39brookshireso like you can change the photos on a cisco phone?
03:45.47mitchelocwhen you get your pr0n make sure to use ulaw to get the best experience
03:45.59filepr0n? WHERE
03:45.59Qwellbrookshire: I don't know if the protocol can
03:46.00brookshireg722!
03:46.09MoutaPTQwell do you know which message i should receive in Asterisk when the called party hangs, with zap intense debug?
03:46.11mitcheloclol file
03:46.20Qwellfile: qwell.com/pr0n/
03:46.29fileQwell: is that... Qwell pr0n?
03:46.31Qwellwget WILL crash if you try to mirror it
03:46.35Qwellumm...mebbe
03:46.36coppiceare the women in ulaw pr0n fatter than in other pr0n?
03:46.46Qwellcoppice: less compression, is all
03:46.53brookshire:(
03:46.53QwellThey actually look better...no lossage
03:46.55brookshireyou lie
03:47.05Qwellg729 pr0n is ...ugh
03:47.06filebrookshire: Damn Best!
03:47.14Qwelllpc10 is like...yeah
03:47.19brookshirefile: where did you get Damn Best?
03:47.25filebrookshire: somewhere.
03:47.29brookshirehehe
03:47.31brookshirerar!
03:47.38fileit MAY have appeared on my account
03:47.55Qwell"Damn Best"?
03:48.07brookshireyes.. Damn Best!
03:48.13fileit's the damn best!
03:48.14Qwellwhich are?
03:48.50fileexactly!
03:51.18fileQwell: nub!
03:51.22Qwell:(
03:51.30*** join/#asterisk bmg505 (n=leon@dsl-146-14-214.telkomadsl.co.za)
03:53.35*** join/#asterisk isamar (n=isamar@202.95.220.92)
03:54.11isamarhi folks
03:59.04*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
03:59.07wwalkerI've got two IP501's.  one will authenticate, the other fails.  I've set and reset them over and over.  the sip.conf paragraph for one is a copy of the other with the [sip_name] changed.  So I can't see how one works and the other doesn't
03:59.12wwalkerAny ideas?
04:00.09alephcomI have a favor to ask...  Would somebody be willing to try calling 14035387846 and see if you can get through?
04:00.36alephcomIt should hit a recording.
04:01.13wwalker"connected to the server"
04:01.22QwellservICE
04:01.37Qwell"hoi, you are being connected to the service"
04:01.41drrayyou are being connected to the service"
04:01.46drrayis that festival?
04:01.52wwalkeralephcom: I don't think it works :(
04:02.02alephcomThanks a bunch everyone...  This is weird.
04:02.10alephcomwwalker, you couldn't get through?
04:02.28alephcomdrray:  Gotta love my voice. :-P   That's me :-(
04:02.30wwalkerjust joking.  I heard server instead of service as many corrected me
04:02.41drraythat's you>?
04:02.51drraywhat codec are you using?
04:02.58Qwelllpc10?
04:03.05wwalkerAllison makes nice recordings....
04:03.39alephcomtrashy microphone on a bad day.  I know I have bought all the recording from her.  For some reason this one hasn't been replaced yet.
04:03.54fileQWELL
04:05.04fileawwww
04:05.22Qwell<3
04:06.04Qwelldrray: perv
04:10.40isamartoday my 1.2.6 astersik stopped responding SIP connections... suddenly..
04:12.18VoIPMastaa firewall maybe?
04:12.23drraydid you do a yum updated?
04:12.26drray-d
04:12.33Qwellyum -d updated?
04:13.08Qwelloh, yum upated
04:13.15VoIPMastayum?
04:13.21VoIPMastaIs that another "automated" installer?
04:13.22drraydid you upgrade your kernel
04:13.29drrayor something on your box
04:14.30*** join/#asterisk synaptic (i=synaptic@68.62.176.196)
04:14.32isamaryum ?
04:14.47isamarnot firewall..
04:14.48QwellI prefer apt and synaptic
04:14.49isamarpublic IP
04:14.59synapticlol
04:15.03VoIPMastaisamar: maybe your ISP added a new firewall?
04:15.15isamarnot first time with 1.2.x ... :-(
04:15.17VoIPMastaisamar: or something else is opening port 5060
04:15.26isamareverything ws ok.. except sip :-(
04:15.29wwalkerI've got two IP501's.  one will authenticate, the other fails.  I've set and reset them over and over.  the sip.conf paragraph for one is a copy of the other with the [sip_name] changed.  So I can't see how one works and the other doesn't
04:15.33wwalkerAny ideas?
04:15.47VoIPMastaQwell: apt? as in debian?
04:15.57QwellVoIPMasta: sure
04:16.02VoIPMastawwalker: what's in your logs?
04:16.17VoIPMastaQwell: I prefer make, gmake, configure
04:16.32Qwellmake, then gmake, THEN configure?
04:16.39VoIPMastaQwell: not in the right order
04:16.50VoIPMastaQwell: as a matter of fact you can't use make and gmake in the same compile
04:16.52fileQwell: come on baby I want to party, come on right to the dance floor
04:16.54Qwellindeed you can't
04:16.58Qwellfile: umm
04:17.03Qwellyou scare me
04:17.10fileexcellent
04:19.44*** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com)
04:20.40wwalkerVoIPMasta: Apr 16 00:13:45 NOTICE[3423]: chan_sip.c:10879 handle_request_register: Registration from '<sip:511@192.168.50.1>' failed for '192.168.50.200' - Username/auth name mismatch
04:20.49SplasPoodHrm.. is it possible to have the grandstream bt-101 hang up the call when the remote party hangs up when on speakerphone?   Mine starts giving a busy tone
04:21.38VoIPMastawwalker: Username mismatch
04:21.49VoIPMastawwalker: double check your username and password settings
04:21.53QwellSplasPood: what is on the other end?
04:22.31VoIPMastaSplasPood: I don't think it's possible
04:22.49wwalkerthe "username" is the part between the []'s in sip.conf, right?
04:23.00VoIPMastawwalker: remove any type of encryption, use cleartext usernames and passwords stored in sip.conf (don't use mysql to set usernames)
04:23.30wwalkerplaintext in sip.conf.
04:23.37VoIPMastawwalker: you can also set it using username=yourusername
04:24.48filebeeeeep no you can't....
04:25.02VoIPMastafile: you can't what?
04:25.02filewhatever is in the context name, ie between [ and ] is what is the username...
04:25.30VoIPMastafile: AFAIK it can be overrided using username= in sip.conf
04:25.38wwalkerhttp://rafb.net/paste/results/A2nlZz59.html 510 works, 512 works, 511 doesn't...
04:25.40filethat's used for something different
04:28.11VoIPMastawwalker: your sip.conf looks ok\
04:28.38VoIPMastawwalker: check your settings in your IP501
04:30.27wwalkerWhich IP501 setting is the password for SIP?  Which setting is the username?  I think that is Auth under Line 1
04:34.10SplasPoodQwell: Asterisk...
04:34.18SplasPoodVoIPMasta: So this is a known...  bug/feature?
04:34.18VoIPMastawwalker: don't know, I don't have an IP501
04:34.27QwellSplasPood: and on the other end of that?
04:34.28wwalkerthx
04:34.37SplasPoodQwell: nothing..  local dialplan app stuff..
04:34.40VoIPMastaSplasPood: most speakerphone phones won't hang up until you press the hangup button
04:34.53SplasPoodpolycom will
04:35.02SplasPoodas will cisco
04:35.44VoIPMastayup, but Grandstream's chipsets are usually less feature-packed
04:35.55SplasPoodhaha... "feature"
04:36.07SplasPoodI don't consider this a feature, but for $48, i can deal :)
04:36.34VoIPMastaI said "less feature-packed" what I mean is that there are less features in a Grandstream chipset
04:36.54SplasPoodYes there are, but hanging up the phone when it gets a hangup indication isn't much to ask.
04:38.54drraymy budgetone was worth the $70 i paid for it 2 years ago
04:39.10drrayI'd not deploy them in an office though
04:39.27SplasPoodyea we bought a few to throw in people's homes
04:39.39SplasPoodprolly end up going for IP301s in the long run
04:39.48drrayjust don't power cycle it during a reboot
04:40.00SplasPoodhrm?
04:40.20drrayit'll lose its mind
04:42.03SplasPoodso like.. i plug it in... then while it's booting, unplug/replug it?
04:42.25drraywell, I was rebooting it from the webpage
04:43.53SplasPoodwhen you say it'll lose it's mind do you mean.. it'll forget it's config.. or it'll brick itself..
04:45.05drraymine bricked itself, I had to set a subnet up to mimic the one it was looking for to load a new firmware on it
04:45.12drrayit kept clicking every 10 seconds
04:45.29VoIPMastawhat you mean is to not unplug it while upgrading, not rebooting
04:45.43drraymine locked up during rebooting
04:45.47VoIPMastamost devices will have problems if you cycle the power while they're having the firmware updated
04:45.51drraynot an upgrade
04:45.56VoIPMastammm odd
04:46.11drrayI was able to revive it
04:47.23drrayI hit reboot in the budgetones web gui, and while it was clicking to restart I unplugged it
04:47.29drrayI don't know what possed me to do it
04:48.57*** join/#asterisk achandra (n=achandra@static-71-103-255-118.lsanca.dsl-w.verizon.net)
04:49.15VoIPMastaI wish there was a local grandstream dealer here, but there aren't any
04:49.30Qwellgrandstream is pretty bad too
04:49.38VoIPMastaQwell: but pretty cheap
04:49.50VoIPMastaQwell: it can be the perfect solution for lame residential users
04:49.54drrayI'd buy an iaxy before I bought another grandstream
04:49.57QwellI was actually supposed to email one of the grandstream guys...
04:54.00*** join/#asterisk op3r (i=op3r@210.4.31.234)
04:54.09op3ranyone using aheeva here?
04:55.16achandraHello, had a few questions about * in a LB setup. Can multiple * boxes, using say SRV records function with a central PGSQL boxes ( assume active, active setup for LB).
04:56.34*** part/#asterisk TTT_Travis (n=Travis@bal-broadband2-ws-14.dsl.airstreamcomm.net)
04:56.38Qwellachandra: LB?
04:56.51achandraload balanced
04:57.22achandra; )
05:17.01*** join/#asterisk Plecebo (n=x@216-160-101-90.tukw.qwest.net)
05:18.42*** join/#asterisk suma (n=suma@222.165.112.215)
05:18.54sumai'm havng problem with iax registration
05:19.01sumacan anybody please help
05:20.25suma<PROTECTED>
05:22.18sumathis is in my iax.conf file
05:22.20suma[kans]
05:22.20sumatype=user
05:22.20sumacontext=kanscontext
05:22.53brookshiretype=user is for outgoing lines.. is that what you want there?
05:23.32sumathanks a million
05:24.07fileer
05:24.17brookshireheh
05:24.19brookshirefile!
05:24.26fileuser is incoming, as they use your system
05:24.35filepeer is outgoing, as you peer with another system
05:24.44brookshireoh now you've got me all confused
05:24.54filegood.
05:25.00brookshireA user makes calls. The following would be needed in iax.conf on the user machine to identify (authenticate) itself to the peer before the peer will take the call.
05:25.09tainted-why not type=incoming, type=outgoing, type=bidirectional
05:25.26brookshireoooo! lets just make everything 'friend'
05:25.34QwellLet's kill the user
05:25.49Qwelltype=whateveritdoesntmatteranymore
05:25.50tainted-type = friend, type = aquaintance, type = onenighter, type = voipwanker
05:26.04Qwelltype=bootycall
05:26.15tainted-i need more of those users
05:26.25Qwellmore, as in 1?
05:26.40tainted-one is the new more
05:28.36Qwelltype=user ~= fxs?
05:28.48Qwelltype=peer ~= fxo?
05:30.30brookshireoh yeah.. don't get me started about the fxs fxo thing
05:31.00tainted-yea talk about confusicating
05:31.51tainted-get started brookshire get started
05:36.24Shotta30fiveAnyone know abou unlocked a VOIP device
05:36.38Shotta30fiveunlocking
05:36.46QwellShotta30five: You don't
05:38.55achandraAre you reffering to say unlocking a Fleebay special sipura that has been "locked" and you want to ulock it?
05:39.31QwellNo, he got a free vonage ATA, and wants to unlock it
05:39.51drraycan't you pay vonage $5 to unlock it?
05:39.52file[laptop]Qwell: are you locked?
05:40.02Qwellfile[laptop]: depends where
05:40.16file[laptop]eep
05:40.24Qwellsicko, not like that
05:41.01file[laptop]Qwell: you're sooooooo cool
05:41.07QwellI so know
05:41.09dlynesbtw...why not just have one context that defines outgoing and incoming, so you don't need two separate contexts?
05:41.21dlynesMost of the info is duplicated, anyways
05:41.22Qwelldlynes: because that would be silly
05:41.37Qwelltwo seperate contexts is a VERY good thing
05:42.09dlynesyes, it makes sense, conceptually
05:42.37dlynesbut if you don't need the type= line, what's to differentiate the two contexts?
05:42.50achandrathis cleary explained...because I myself re-read it here -  page 78 of Asterisk: The future of Telephony...about contexts and why they are seperate...
05:43.38dlynesthat book isn't terribly useful for the most part, either
05:43.43achandraI suppose you could argue and be convincing that seperate contexts enforce security.
05:43.47brookshirefriend is bad!
05:43.51brookshirei learned this today
05:44.04dlynesYeah...friends are bad; enemies are good!
05:44.14achandrawtf??
05:44.34achandrai thought we were discussing use of contexts..anyhow.
05:44.37file[laptop]brookshire: am I your friend?!?
05:44.47brookshirefile: no..
05:44.51file[laptop]good
05:44.52file[laptop]I don't want to be
05:44.53achandralol
05:44.58dlyneslmao
05:45.45dlynesachandra: I think the different contexts are just for conceptualizing direction of traffic
05:45.52dlynesachandra: i.e. for humans
05:46.08tainted-omg freshly baked poundcake + ice cream = crazy delicious
05:46.10Qwellcontexts help IMMENSELY with security
05:46.17Qwelltainted-: poundcake linux?!
05:46.19dlyneshow so?
05:47.02tainted-apt-get consume poundcake
05:47.06dlynesAnd how's that gonna help you get another free pstn proxy?
05:47.18file[laptop]I'm watching William Shatner... rap...
05:47.22file[laptop]I feel violated
05:47.23Qwelldlynes: Because you obviously  haven't read README.security
05:47.29achandraon contexts again...quoting... " One of the most important use of contexts is to enforce security. By using contexts xorrectly, you give certain callers access to features...If you dont design the dialplan carefully you may indaventrly allow others to fradulently use your system.....se SECURITY"
05:47.37tainted-dlynes by forcing intruders to be so innundated with extra text that they run away with despair
05:48.24achandraanyhow thats on page 79 ;)
05:48.26tainted-dlynes he hax0rs your skinny session
05:48.26dlynesAnd?  You think I send all iax and sip traffic into the same incoming context?
05:48.47Qwell100 things going into one incoming context != insecure
05:48.59Qwellincoming AND outgoing in the same context == stupid
05:49.06brookshireyes
05:49.09brookshirei agree
05:49.16Qwelland covered in README.security
05:49.17tainted-i like to use [incoming] + _. matching..
05:49.23tainted-it's really cool
05:49.27dlynesI thought we were talking about sip/iax contexts, not dialplan contexts?
05:49.27brookshireqwell: and don't forget about the headaches
05:49.31file[laptop]let's put our outbound PSTN calling into our incoming context... yes
05:50.01achandraplus imagine having a complex dialplan with multiple macros, and definitions, what a friking nightamare keeping it all straight...and imagine changing say one or two things and having crap blow up...yuck.
05:50.13Qwellpublic IPs on a DS3, with a simple password, is also insecure :D
05:50.19Qwellfile[laptop]: eh, eh?! ^^
05:50.22dlynesI wasn't even talking about the context= line in iax/sip.conf
05:50.50file[laptop]The Running Man is on, like omg
05:50.52achandragive us an example...maybe a good place to start??
05:50.53file[laptop]what an old movie!
05:50.59Qwellfile[laptop]: does he really run?
05:51.07dlynesAn outgoing iax/sip context afaik doesn't even use the context= line, does it?
05:51.15file[laptop]Qwell: yes
05:51.21achandrafile[laptop]: The asternator? lol
05:51.35file[laptop]dlynes: you can send a context over IAX2 actually
05:51.40QwellDoes he also run...for governer?!
05:51.46Qwellwow, I botched that word
05:51.49Qwellgovernor?
05:52.01dlynesfile[laptop]: but that's only if you're including the remote iax switch into your dialplan context, right?
05:52.12achandraits okay you can account the mis-spelling to his lack of funding for education.
05:52.23dlynesfile[laptop]: which, in theory is not a sound concept from a security standpoint, to begin with
05:52.35file[laptop]no, you can send it when you dial as well
05:53.16dlynesfile[laptop]: Yeah...I never did understand that concept...why you want the outside world to know what one of your contexts is
05:54.04dlynesIt should route you into the context you've defined in your iax.conf file for that user; they shouldn't be able to choose which context to go into
05:54.29file[laptop]the music on this movie is so... yeah
05:54.38achandradlynes: assuming the complexity, a simple context might be figured out by simply calling the number, and running through scenarios of key presses. No?
05:55.30dlynesachandra: Ok....where are you going with this?
05:55.55sumawhen i make a call to asterisk it answers and playback a sound file and i could not hear that
05:56.02sumathe call is iax2
05:56.05achandrawell...in essence you program the contexts based on what you want the user to say experience...so in essence the user DOES know the context.
05:56.18achandraif he/she goes through the presses..
05:56.23dlynesThey don't know the name of that context, though
05:56.40achandraright..but the function yes...
05:56.48dlynescorrect
05:57.03dlynesso why should they be allowed to specify which context to use?
05:57.20*** join/#asterisk cced (n=dev2003@222.33.36.205)
05:57.21dlynesThe context should be chosen, based upon which username and password they have chosen
05:58.00achandrawhich that function exists in * today.. a little confused...go on though..
05:58.42sumawhen i make an iax2 call, i could not hear the audio, asterisk says it is playing ? ! can anyone please help me
05:58.44dlynesWell, if you're able to specify a dialplan context outside of what's defined in your iax context, you can theoretically invoke someone else's context
05:58.57dlynesand thus get access to a dialplan you wouldn't normally be able to use
05:59.22cced:)
05:59.22dlynesi.e. possibly dialing long distance if say normally you're restricted to local pstn calls
05:59.35achandraby seperating them though..you could have them access international calls or not etc. but that is based on  seperate contexts to keep some users from doing one thing or another..
05:59.44achandrawoops you beat me to it.
05:59.47dlynesBut if you don't allow the remote user to specify which context to use, they don't have that ability
06:00.21dlynesWhich brings us back to my original question, why does iax allow you to specify the context in the dial command?
06:01.03achandrai just see it in reverse...the contexts are rules by which a user can do one thing or another...and the user abides by those set rules... sure as a usere youd be able to understand those rules...(contexts in this case), but your bound by them.
06:01.05dlynesOr in the switch command for that matter?  I realize that's the whole concept of the switch command is to share dialplans, but it seems to me that is insecure in and of itself
06:02.20dlynesIs there a way to lock down asterisk so that the remote caller is not allowed to specify which context to use?  i.e. it'll just use the context that's specified in the 'context=' line of iax.conf?
06:02.46dlynesOtherwise, what's the point of even having the context= line?
06:04.12achandradesign issue...i guess...define the dial plan differently?...im not sure..unless we start talking about "what" you are intending on designing.
06:04.35kamileonwhat voip provider is suggested for use with * with local DIDs
06:04.39dlynesI'm not intending on designing anything
06:04.40achandrawhat do you want the system to do and how do you want the user to interact with the system.
06:05.10dlynesQwell was just saying i'm running a sip bot, but I don't see any way to lock down the context issue, either
06:05.53achandradlynes: I will say this...your making me think :)
06:06.06achandraon a friking saturday too.
06:06.13dlynesWell, wouldn't that seem like a pretty basic thing?
06:06.28dlynesAllowing the system admin to lock out any requests for an alternate context?
06:06.51dlynesI've never understood why iax even allows that
06:06.54cced:
06:07.55dlynesSure, it makes it more flexible, but at the risk of making it less secure
06:08.32dlynesIt's like Microsoft Windows...Windows allows you to do a lot of things as a regular user that UNIX doesn't, but it also makes it less secure
06:09.28dlynesI guess you must be on PDT, too
06:09.43dlynesEverywhere else on the continent, it's Sunday :)
06:13.01sumayes
06:13.07achandrayep
06:13.27sumai could not hear audio in an iax2 call
06:13.37sumaasterisk says it is playing the gsm file
06:13.44sumait is the demo which comes with asterisk
06:14.09achandrabehind nat or.. ??
06:14.50sumaI forwarded my port 4569 to my PC
06:15.00sumai mean the router port
06:15.06sumawhere the client is
06:15.19sumashall i show you the ethereal output ?
06:16.00achandranot sure if im entirely qulaified to solve your issue but someone may know whats up
06:16.47sumaserver is through DMZ
06:16.55sumaand my client is through port forwarding
06:17.01sumawill there be a NAT problem
06:18.28suma46.534929 222.165.112.215 -> 192.168.1.13 IAX2 IAX HANGUP, source call# 16216, timestamp 990ms
06:18.29suma<PROTECTED>
06:18.44achandraso you have the * in the DMZ and then you are connecting to external ip of firewall or router which routes to internally natted * box?
06:19.34achandrawhat is in your iax.conf file?
06:19.48sumait is the default conf file
06:19.55sumai added new user with that
06:20.13suma[kans]
06:20.13sumatype=friend
06:20.13sumacontext=default
06:20.13sumahost=dynamic
06:20.32sumadefault is the default demo that comes with asterisk
06:20.53achandrathe sample file....okay
06:21.05sumayes
06:21.51sumawhen i say ifconfig, it says 192.168.0.12 and 192.168.0.13 respectively in linux
06:23.12*** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz)
06:23.19iceypwhats the variable for digit input?
06:23.27iceypi.e. if i want to do SayDigit($input)
06:25.23sumai could not find the application saydigit
06:25.30sumadoes it comes with asterisk ?
06:25.55*** join/#asterisk cced2 (n=dev2003@222.33.36.205)
06:25.58X-Robsuma, try 'show application say' and then push TAB
06:26.02X-Rob(in the asterisk console)
06:26.10X-Robor just 'show applications'
06:26.12*** join/#asterisk MaddieBoi (n=MaddieBo@210-84-15-248.dyn.iinet.net.au)
06:27.21achandraim not sure exaclty but do you need to define qualify=no, and port=4569 in the definition which you posted explicitly?
06:27.39sumayes, there is SayDigits but not SayDigit
06:27.56achandraas well as context=from-internal ( at least thats in my case).
06:28.55sumaport=4569 is in global definition of iax.conf
06:29.13achandraokay
06:29.34achandraalso defined the externip ?
06:30.13sumaexternip in the general or in the user context ?
06:30.20achandrain general
06:31.15sumano, just now mentioning it
06:31.29achandraahhhh
06:33.55achandraso.. externip = aaa.bbb.ccc.ddd ; localnet = 192.168.0.0./255.255.255.0; bind addr: 0.0.0.0,  etc.
06:34.18achandraor whatever the internal network is..
06:34.51achandrathen you must recompile the kernel....just kidding... :)
06:35.03cced2who is familar with H323?
06:35.48achandracced2: your going to use for media streaming of video ? cool.
06:36.47*** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com)
06:36.53sumano luck
06:38.33sumaI also get these messages in the meantime
06:38.34sumaupdate_registry: Restricting registration for peer 'kans'
06:38.54sumato '60' seconds
06:40.59achandrado you have a delayreject =yes in general as well?
06:42.19sumajust now i did
06:42.24sumaand restarted asterisk
06:42.30achandraokay
06:43.24iceypachandra whats the cariable to remember my input digits?
06:43.34iceypor actually it's just my dialed number
06:43.36iceypi should be ok
06:44.04sumaApr 16 07:42:57 NOTICE[8311]: chan_iax2.c:5692 update_registry: Restricting registration for peer 'kans' to 60 seconds (requested 1200)
06:44.11sumastill this message comes
06:44.47achandrathis is what my general section has -
06:45.44*** join/#asterisk ptblank (n=MURDER1@68.233.145.253)
06:47.03achandrabindport = 4569 , externip = aaa.bbb.ccc.ddd , local net = 192.168.0.0 / 255.255.255.0 , bindaddr = 0.0.0.0 delayreject = yes disallow=all allow=g729,allow=ilbc, allow=ulaw, allow=alaw. allow=gsm jitterbuffer = yes dropcount =1
06:50.09achandrasuma: did that help or??
06:50.58Shaun2222is their a uniq id assigned to every call that comes in on asterisk, hopfully one that can be read using ${ID}
07:02.54dlynessuma: This might seem like a stupid question, but are 192.168.0.12 and 192.168.0.13 both on the same machine?
07:04.14websaehrm
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07:16.20esculapio_hola quien habla espanol
07:17.03cced2:) .
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07:27.45dlynesShaun2222: ${UNIQUEID}
07:28.32dlynesShaun2222: Try checking out http://www.voip-info.org/wiki/view/Asterisk+variables
07:34.29esculapio_hola quien habla espanol
07:34.34esculapio_?
07:35.49cced2why irc often offline?
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07:56.48thx2000Does anyone know of a way to get sendmail to use the /etc/hosts file instead of looking up the host record from the dns server?
07:59.14stoffell_hthx2000, uhm, hostfile is used 'before' using dns server, but i don't see the *-relation here ;)
07:59.27thx2000sendmail seems to bypass it
07:59.51thx2000hang on, i might have a way to fix it, and it'll save a lot of explaining :)
08:00.46tecnicochecked /etc/nsswitch.conf ?? make sure files is before dns
08:05.36thx2000doesn't look like that file exists in osx
08:06.14thx2000Is anyone still here to listen to me ramble about my problem? :P
08:07.23tecniconsswitch is part of glibc , weird that you don't have it...
08:07.38dlynesthx2000: /etc/host.conf:  line 1:  order hosts,bind   line 2:  multi on
08:07.54*** part/#asterisk angom_h (n=angom@red-corp-200.76.229.86.telnor.net)
08:08.08dlyneshost.conf afaik, is part of bind
08:08.24tecnicois sendmail bound to 127.0.0.1 as well ? or just your external IP ?
08:08.28jj1to grab syslog messages from a pap2 to a special file anybody know the syslog.conf line? eg pap2.*  /var/log/pap2.log?
08:08.53thx2000im not sure
08:09.14thx2000dlynes, that file doesn't exist, so just create a new one w/ those 2 lines?
08:09.15tecnicojj1: in syslog-ng you can create a reg. expr. filter.. if that's an option for you.
08:09.37dlynesjj1: there's no service defined for pap2
08:09.48dlynesthx2000: Yes, you can create the file; ymmv
08:10.00jj1k, what do most people do? throw syslog messages into mysql somehow?
08:10.11dlynesthx2000: it all depends on whether or not your bsdsockets actually makes use of that file
08:10.13*** join/#asterisk NirS (n=NirS@62.90.49.98)
08:10.22NirShello everybody
08:10.29thx2000does sendmail cache mx records?
08:10.39dlynesthx2000: no..your dns cache does
08:10.44jj1whats better SER or OpenSer?
08:11.29dlynesthx2000: your dns server will have that cache
08:11.53NirSSay, anyone ever tried working with /n with Manager originate method to Local channel ?
08:12.01thx2000what if i created a new zone on my local dns server that should completely bypass the old record?
08:12.12thx2000local meaning the dns server i use for my local network
08:13.05dlynesthx2000: why would you?
08:13.35thx2000because it didn't look like sendmail was paying any attention to my hosts file
08:13.36dlynesthx2000: Why not just override your dns server with /etc/hosts?
08:13.53thx2000i tried that and it was still tryin to send to the external ip
08:13.57dlynesthx2000: try commenting out the entries in your /etc/resolv.conf file
08:14.19dlynesthx2000: or renaming your /etc/resolv.conf file so sendmail can't find it
08:14.24dlynesthx2000: and then restart sendmail
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08:15.05dlynesthx2000: What operating system are you running on?  OSX?
08:15.21thx2000yea, 10.4.4 i think
08:15.48dlynesyeah...i'm not totally familiar with it, but I do remember it's lacking a few files
08:15.53thx2000Basically, im just tryin to get sendmail to use my mail server's local IP instead of the external one since my firewall wont accept loopback connections
08:16.25dlynesOk, and what's that got to do with your nameserver or hosts, or anything else for that matter?
08:16.45thx2000well the dns for the the domain im trying to send mail to is hosted on netsol
08:16.49dlynesSendmail is your local mail server, right?
08:17.16thx2000so when sendmail goes out to look for the ip for my mailserver it resolves to my external ip
08:17.23dlynesOr are you trying to use a smart host that's on your local network?
08:17.53thx2000I'm trying to send the message to my kerio mailserver, on another box
08:18.02dlynesoh...nvm...it sounds like you haven't even configured sendmail yet
08:18.10dlynesIt sounds like you're using the default configuration
08:18.19thx2000probably
08:18.22thx2000:P
08:18.35dlynesYou need to define a smarthost in your /etc/mail/sendmail.cf or /etc/sendmail.cf file
08:18.46dlynesgo to www.sendmail.org to learn how to set up a smart host
08:19.00dlynesAnd configure your smart host by ip address, not by host address
08:19.07dlynesthat way it forces it to go to the lan address
08:19.29thx2000neither of those files exist :/
08:19.52dlynesfind / -type f -name sendmail.cf
08:21.40thx2000now that i've created a record in my local dns server, to point the mx record to the local ip of my mailserver, shouldn't that bypass it from finding the old record at all though?
08:22.51dlynesWell, that's not the proper way of doing it, for one
08:22.59dlynesAnd two, did you remove the old entry?
08:23.24dlynesIn order to solve a problem properly, it usually helps if you understand why you're having a problem
08:24.02dlynesThe smarthost option in sendmail was designed specifically for your problem in mind, where all mail gets sent to one specific mail server
08:24.06thx2000I'm just frusterated and all the help i find has absolutely nothing to do with osx
08:24.16thx2000so at this point anything that'll make this thing work im completely fine with
08:24.21dlynesYour problem is a sendmail problem, not an osx problem
08:24.40dlynesSendmail doesn't look at /etc/hosts
08:24.56thx2000correct
08:24.57dlynesReason being is that it needs to do a reverse lookup
08:25.03dlynesAnd it can't do that with a hosts file
08:25.42dlynesIt does the reverse lookup to make sure nobody's trying to use your mailserver as a spambot
08:26.02dlynesIf you choose to configure sendmail to do that
08:26.18thx2000ok, i understand that now
08:26.23dlynesbut regardless, it still behaves that way whether you want that functionality or not
08:26.34thx2000the search for sendmail.cf came up w/ nothing
08:26.39dlynessendmail has a lot of power
08:26.49dlynesbut with that power comes a lot of bulk
08:27.06thx2000understood
08:27.17dlynesDo you have an /etc/mail directory?
08:27.25thx2000no
08:27.36dlynesOk, next question...is sendmail in your process list?
08:27.46dlynesI get the feeling your mailserver might not be sendmail
08:28.02dlynesIt might be some other mailserver masquerading as sendmail
08:28.18thx2000thats very possible
08:28.21thx2000postfix exists
08:28.31dlynesPostfix is in your process list?
08:29.15dlynesThe reason I suspect you're not using sendmail is because of the lack of existence of a sendmail.cf file
08:29.28dlynesThat file is necessary for sendmail to start up
08:29.57thx2000makes sense...i just figured it was there because it would respond through the terminal
08:30.04thx2000i can do postfix start
08:30.06dlynesHow so?
08:30.13dlynesI don't want you to do postfix start
08:30.22dlynesI want to find out what mail server you've got running, currently
08:30.29dlynestry this:
08:30.48dlynesps auxffww | grep -E "postfix|sendmail|qmail|smail"
08:31.09dlynesDid you get any results?
08:32.14thx2000well postfix, but i had started it before u told me not to :P
08:32.43dlynesOk, so postfix is the only thing running in taht list?
08:33.03thx2000root       238   0.0  0.1    27356    744  ??  Ss    1:28AM   0:00.07 /usr/libexec/postfix/master
08:33.03thx2000postfix    239   0.0  0.1    27380    756  ??  S     1:28AM   0:00.03 pickup -l -t fifo -u
08:33.03thx2000root       251   0.0  0.0    27820      4  p0  R+    1:31AM   0:00.00 grep -E postfix|sendmail|qmail|smail
08:33.08thx2000thats the full output
08:33.11dlynesThere ya go
08:33.19dlynespostfix is your culprit then, not sendmail
08:33.29dlynesI know...you started it after I asked you not to
08:33.40dlynesbut i suspect that's the only mail server you have installed, too
08:33.41thx2000*before*
08:33.47dlynesor before :)
08:33.49thx2000hehe
08:34.00dlynesAnyways...I can't help you with postfix, though
08:34.04thx2000doh
08:34.05dlynesI know sendmail quite well
08:34.10dlynesBut I don't know postfix at all
08:34.33dlynesBut, I suspect it probably has an option for a smarthost as well
08:34.38thx2000a heck of alot more than i know im sure :P
08:34.41thx2000thanx for the help though
08:34.50thx2000a shove in any direction is welcome at this point
08:35.25dlynestry http://freshmeat.net/projects/postfix for more info
08:35.46dlynesIt should have a link to the homepage for postfix there
08:35.59dlynesOn the postfix home page, you can probably find more info about setting up a smarthost
08:39.20glm2kthx2000: you might find this guide quite straightforward to follow. it is however a mysql/postfix install
08:39.22glm2khttp://flurdy.com/docs/postfix/
08:39.41*** join/#asterisk heka (n=heka@82.114.68.124)
08:39.47glm2kjust skip the section you don't need
08:39.52glm2ker, sections
08:40.04thx2000cool, thanx
08:41.47*** join/#asterisk franck (n=franck@tikiwiki/franck)
08:42.03franckHi all
08:42.37franckWhen you compile zapata drivers it gives you several option for echo canceller, which one to use?
08:42.46franckI have a wctdm24xxp card
08:42.51dlynesfranck: Are you using trunk?
08:43.12franckdlynes, all are FXO interfaces
08:43.23dlynesfranck: If you're using trunk, you can use the MG2 echo canceller
08:43.40dlynesIt's the best one, but I don't think it's available in zaptel 1.2.5
08:43.41franckto call outside, inside is all sip phones
08:44.46dlyneshave you tried MG2?
08:45.04franckthis is the one I have compiled
08:45.06franckin
08:45.20hekaCan I apply the ast_jb-1.2.0.patch3 to asterisk version 1.2.7.1 ?
08:45.33franckbut I get in my asterisk log, cannot set echo cancelelr on channel ...
08:45.52dlynesyou mean zaptel trunk, franck?
08:46.30franck<PROTECTED>
08:46.48franckthis is the error message I get each time an outside line is used
08:47.31franckeither to call out or receive a call
08:48.00franckI have Zap/g0 with Zap/1 to Zap/8
08:48.01*** join/#asterisk CMike (i=daemon@c-544171d5.116-1-64736c10.cust.bredbandsbolaget.se)
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08:49.05dlynesYeah...sorry, dude...no idea what's wrong
08:49.25franckdlynes: ok thx for trying
08:49.43dlynesBut then again, I'm using a PRI on one asterisk box, and x100p's on everything else
08:52.10franckI think the wctdm24xxp is a little bit new...
08:52.41dlynesit's the guy with 4 modules; each module has 6 fxs ports, or 6 fxo ports
08:52.48dlyneserm
08:52.58dlynes6 modules, each with 4 fxs ports, or 4 fxo ports i mean
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08:53.24dlynesright?
08:54.24franckyes
08:57.38franckyou have 24 ports in group of 4 fxo or fxs
08:57.47franckI have 8fxo total on this card
09:06.19dlynesyeah...there's a couple people i've seen on here using that card
09:21.11franckdlynes: yes it puzzles me... this error message
09:21.34*** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com)
09:21.43franckand especially where I am, I neve r know if it is the card or the other telcos who have a lot of echo
09:27.50thx2000relayhost= ... thats all i needed
09:31.01hekaanybody can help me about jitter buffer for sip?
09:34.15RoyKheka: using the patch from mantis?
09:34.27RoyKslav's stuff?
09:35.43hekaRoyK: yes!
09:36.36hekaI have try to use it with cdr_mysql but because of memory leak in earlier versions I couldn`t use it
09:37.08hekabut in later versions of asterisk memory leak has been fixed so my question is: Can I apply that patch to version 1.2.7.1?
09:38.27RoyKI have a newer patch
09:38.41hekacan you share it with me please?
09:38.45RoyKsure
09:38.50RoyKbut the leak is still there
09:39.31hekaRoyK: that`s bad bacause it dosent give me a chance to use it with any cdr gennerator
09:39.41RoyKwhy?
09:40.12RoyKafter four days uptime on 1.2.6 asterisk eats something like 400 megs, so it's kinda ok. just need to restart it every now and then
09:40.20hekabecause while trying to put the data to database it get segfaulted
09:40.23RoyKhave there been any leak fixes in 1.2.7?
09:40.30RoyKthat's not the leak
09:40.52RoyKheka: what db?
09:40.52RoyKsegfault != leak
09:41.03hekaRoyK: I think that some fixes has bean done in 1.2.7
09:41.08RoyKmake a backtrace and report the crash in mantis
09:41.24hekaRoyK: but it looks to me from the backtrace that it is the result of the leak
09:41.37RoyKobviously lots of fixes has been done in 1.2.7, but the leak is in the jitterbuffer code
09:42.09hekaRoyK: let me get the newer patch and try with that please!
09:42.53hekaare you thinking about oej`s svn version?
09:43.13RoyKno
09:43.30RoyKi had slav logged on to a test server to setup 1.2.6 with the jb
09:43.53RoyKwe paid for this, and we're experiencing problems
09:44.03RoyKheka: email address?
09:44.15hekamay I pm?
09:45.19RoyKsent
09:45.40hekathanks!
09:45.50hekahave you patch the 1.2.6?
09:46.01RoyKwot_
09:46.02RoyK?
09:46.04CMike*yawn*   morning all...
09:46.09RoyKCMike: morning
09:46.16CMikeHiyas RoyK ... allt väl ? :)
09:46.21hekahave you apply this patch to the 1.2.6 version of asterisk or what version?
09:46.39hekaCMike: morning!
09:46.51CMikemorning..
09:47.02RoyKCMike: alt vel :)
09:47.07hekaRoyK: got it! let me give a try.
09:47.26RoyKheka: please do report that segfault anyway
09:48.05hekaRoyK: I`ll do as soon as Im done with this patch. what do you think about trying to apply this patch against 1.2.7.1?
09:48.05*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
09:48.14RoyKshould work
09:48.19hekaok
09:48.35RoyKheka: btw, I use cdr_mysql in production and have been for almost two years
09:49.08CMikeme too :)
09:49.34CMikemaybe I should upgrade.. my servers..
09:49.56hekaRoyK: both with jitter buffer patch?
09:51.07CMikewtf.. hm ..  a big pigeon just sat down om my sat.dish .. my picture disapeard..   *mumble*
09:51.08CMikeBRB
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09:52.46robin_szmeep?
09:53.22robin_szand .. as expected, its worse than the previous version :(
09:53.40robin_szgoing backwards and fast :(
09:56.36RoyKheka: with or without - doesn't matter
09:57.57robin_szI wonder if there would be any interest in making Grnadstream GXP2000s into something useful?
09:58.03robin_szlike ... plant pots?
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10:34.19hekaHi
10:34.22hekaRoyK: ?
10:35.13RoyKka-ding
10:35.43hekathe same hapens with the new patch
10:35.49RoyKwhat happens?
10:35.51hekado you have time to look at the backtrace?
10:36.02RoyKwhat happens?
10:36.09hekahttp://pastebin.com/662839
10:36.09RoyKcoredump? where?
10:36.51hekathis is with another cdr application
10:37.03hekanot cdr_mysql
10:37.14RoyKapp_prepaid_call.c
10:37.17RoyKpastebin the dialplan as well
10:38.17hekadial plan is simple! Im calling the app_prepaid_call wich handles the dial plan. puts the initial data in cdr table of database
10:38.24hekalike call id, start time etc etc
10:38.35hekaand the completes it after the call is finished
10:38.37*** join/#asterisk ToTo (n=ToTo@host110-142.pool874.interbusiness.it)
10:40.08RoyKperhaps the same bug as this
10:40.08RoyKhttp://bugs.digium.com/view.php?id=6846
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10:40.22RoyKhm
10:40.30RoyKno
10:40.30RoyKnot really
10:40.44RoyKheka: but pb the config and let me see if i can reproduce it
10:41.08*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)
10:41.30FuriousGeorgeexten => s,n(repeat),setvar(COUNT = $[${COUNT} + 1])
10:41.38FuriousGeorgethis is constantly evaluating to the same thing for me
10:42.38RoyKs/setvar/set/
10:42.45FuriousGeorgei tried both ways
10:42.47RoyKbut that shoulnd't really matter
10:42.54FuriousGeorgeand what i meant to say was that it keeps evaluating to two
10:43.01hekaRoyK: wich configs should I pb?
10:43.03RoyKwhat asterisk version?
10:43.11RoyKheka: all relevant
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10:45.02RoyKalso
10:45.02RoyKheka: i need a url to the prepaid app you're using
10:45.25FuriousGeorgehttp://pastebin.ca/49496
10:45.33FuriousGeorgecan anyone tell me why im not getting out of that loop
10:45.46FuriousGeorgei know its b/c my counter is not incrememnting but i dont see why it shouldnt
10:47.15RoyKFuriousGeorge: what version of asterisk?
10:47.19FuriousGeorge1.2.6
10:47.54RoyKk
10:48.15FuriousGeorgeits showing the counter being increased from 1 to 2 but it doesnt get any farther, and i dont see any cli output for the background and noop there
10:48.19FuriousGeorgeso im real confused
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10:50.48snittMw3: respect
10:51.13RoyKFuriousGeorge: the while statement is wrong
10:51.35FuriousGeorgewhich statement?  the counter increment?
10:51.49FuriousGeorgeoh sorry
10:51.57FuriousGeorgei read the "whole statement"
10:52.24*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
10:52.32RoyKit should be
10:52.32RoyKexten => s,n,while($[ ${COUNT} < 7 ])
10:52.41puzzledmorning
10:52.50FuriousGeorgeoh yeah
10:53.06RoyKpuzzled: morning
10:54.41RoyKFuriousGeorge: and...
10:54.48RoyKFuriousGeorge: try this
10:54.50RoyKFuriousGeorge: exten => s,n,set(COUNT=$[ ${COUNT} + 1 ])
10:54.56RoyKthat works for me
10:55.17RoyKseems like asterisk doesn't handle spacing too well around the '='
10:57.13puzzledand wasn't there the opposite too, where you have to use spacing or it won't work?
10:57.24wasimoui
10:58.00RoyKpuzzled: i beleive there was, in the $[ asdf ] thing, but iirc that's fixed
10:58.23puzzledah ok
10:58.28FuriousGeorgeRoyK: thanks the counter is working now
10:58.45FuriousGeorgei would have never figured out that spacing thing
10:59.15RoyKdon't use spaces around =
10:59.17RoyKuse spaces after [ and before ]
10:59.46RoyKand between elements between [ and ]
11:02.36wasimofcourse this may all change without notice
11:02.42puzzledimho that should be sooo fixes to take spaces or not
11:02.48puzzledwasim: indeed
11:04.19Rawplayervrolijk paasfeest allemaal
11:04.26FuriousGeorgecall me crazy but my counter is getting up way past counter < 7 and the loop is STILL not exiting
11:04.33RoyKwasim: what? asterisk is _stable_ and _productional_ and doesn't _change_ unless there's a _major_ upgrade
11:04.49RoyK_and_it_is_the_best_pbx_in_the_word_and_so_on_
11:04.56Rawplayeren dikke eieren
11:05.16RoyKRawplayer: alt vel? trenger du hjelp? medisiner?
11:06.18Rawplayerlol
11:06.23FuriousGeorgemy dialplan says this:  exten => s,n,while($[${COUNT}<7])
11:06.25Rawplayervalium
11:06.26FuriousGeorgeand the CLI says
11:06.35RoyKFuriousGeorge: http://pastebin.com/662870 this one works for me
11:06.35FuriousGeorge-- Executing While("Zap/2-1", "1") in new stack
11:07.54RoyKwasim: and STABLE doesn't mean it doesn't crash, but that new features aren't added....
11:09.04FuriousGeorgeRoyK: shouldnt the CLI say something else besides - Executing While("Zap/2-1", "1") in new stack
11:09.09RoyKno
11:09.16FuriousGeorgeconsidering my while has the > operator and a #7 in there
11:09.28RoyKthe $[ bladdi ] is evaluated and the CLI returns the evaluated result
11:09.56FuriousGeorgeok, but we just got the counter working, so shouldnt that 1 at least change to a 2
11:09.58FuriousGeorgeand so on
11:10.09RoyKno
11:10.18FuriousGeorgeok
11:10.34FuriousGeorgeso i'm gonna assume this part is working and go on to fix the next part :)
11:10.38FuriousGeorgetomorrow
11:10.38RoyKtry NoOp( $[ 1 > 2 ] ) and NoOp( $[ 1 > 1 ] )
11:10.48RoyKthe result returned from $[ asdf ] is a bool
11:11.01FuriousGeorgei see what you mean
11:13.17puzzledRawplayer: same to you :)
11:13.36Rawplayer:D
11:13.44isamaranybody using SER?
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11:16.10RoyKno!
11:16.55puzzledthat harry guy is :)
11:21.40FuriousGeorgejesus, im confused by this new n(label) priority.  apparently if im still in extension s then goto(label) works, but if im not in that extension goto(s,label) doesnt
11:21.54FuriousGeorgesame context though
11:22.15RoyKdon't
11:22.15RoyKjust use goto(label)
11:22.37FuriousGeorgeim told it doesnt exist
11:22.46RoyKpb dialplan again :)
11:23.05RoyKhm
11:23.15FuriousGeorgehold on, this is another part
11:24.57RoyKi think labels are local only
11:24.57RoyKlocal to the context
11:27.00FuriousGeorgehttp://pastebin.ca/49497  im always in the same context, but now its a different extension
11:27.17FuriousGeorgewhat a mess this has become :)
11:28.05FuriousGeorgeim sure there are other mistakes but im just confused by the lables right now
11:28.47wasimFuriousGeorge: well said
11:29.07*** part/#asterisk franck (n=franck@tikiwiki/franck)
11:29.11FuriousGeorgewasim: which part
11:29.24RoyKer
11:29.25RoyKexten => _${ROOMATE[${LASTCOUNT}]},1,random(${MATEWEIGHT[${LASTCOUNT}]},${ROOMATE[${LASTCOUNT}]},3)
11:29.59RoyKcan you use a variable like in the search part of an extension?
11:30.09wasim16:27 < FuriousGeorge> what a mess this has become :)
11:30.11FuriousGeorgei was wondering the same thing
11:30.25FuriousGeorgei guess you can because its complaining about the goto
11:30.49FuriousGeorgewasim: if you can think of a better way to give 5 variables a relative weight and pick one based on that, im all ears
11:31.02wasimno, no, i meant you were talking about *
11:31.04RoyKwhat is a roo-mate? :)
11:31.36RoyKkangaroo-mate
11:31.38RoyK:)
11:31.42FuriousGeorgewasim: well tbh, the spaces thing with my = before did piss me off a bit :)
11:32.01RoyKFuriousGeorge: welcome, to the real world...
11:33.12FuriousGeorgeany idea about that label though, royk?  i really would like to undsertand wtf is happening with that
11:36.35isamaranyone using SER?
11:36.53isamarwhat is the most common use of ser+asterisk ?
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11:37.15FuriousGeorgeisamar: as a sip gateway
11:37.27FuriousGeorgefor handling many many sip user registrations
11:37.33FuriousGeorgeroutiong, etc
11:37.35FuriousGeorgeor so i hear
11:38.02X-Roboooh
11:38.08X-Robnew series of dr who has started
11:38.14X-Robmininova is your friend
11:38.33FuriousGeorgeon hbo
11:39.10FuriousGeorgeRoyK: actually forget that label, im not gonna eork on this anymore till i get some sleep.  if i didnt know any better i'd say its almost working
11:39.46FuriousGeorgei jsut wanna know if there is some glaring error that is gonna force me to start over :)
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11:42.03BladeRunner05Happy easter to all
11:42.35FuriousGeorgesame 2 u
11:43.20BladeRunner05I need to use my eicon diva server bri to asterisk 1.2.7.1, I have installed chan_capi-com.0.6.5 but when I start asterisk it say:  chan_capi.c:4577 cc_init_capi: CAPI not installed, CAPI disabled!
12:03.01puzzledBladeRunner05: what does capiinfo say?
12:09.12*** join/#asterisk Garaan (n=jfleisch@user-142h64a.cable.mindspring.com)
12:09.21GaraanGood morning
12:09.59GaraanAnyone not AFK?
12:16.09GaraanIs anyone on?
12:17.51tzafrir~ask
12:17.52jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a quesiton first.  Don't ask if a person is there, just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily.  See also http://catb.org/~esr/faqs/smart-questions.html
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12:19.52snitt:)
12:21.13GaraanI am currently trying to configure zaptel-1.0.10 with an X100P clone and a TDM400P with 1 FXS module installed. The reason for the old version is that is what is currently in place at my work location.  I am having issues making the cards initialize at boot, as the /etc/init.d/zaptel script sees the error from wcfxo loading and not seeing the fxs port and quits.  Any suggestions?
12:23.40GaraanLink for zaptel.conf and error from /etc/init.d/zaptel http://pastebin.ca/49500
12:26.06Mw3hi. how can i use an auth username with @ in it in sip.conf (register =>)? is there any way to escape the @ ?
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13:03.27cced2:_)
13:04.03cced2?
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13:26.34PoWeRKiLLhi
13:27.22PoWeRKiLLI have a strange thing with my sipura when I'm in a call and call a call wiating I can see the caller id but when there is no call I can't see the caller id any idea ?
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13:43.09cced2<cced2> IN chan_zap.c start_pri() pri->fds[i] = open("/dev/zap/channel", O_RDWR, 0600);
13:43.10cced2<cced2> use /dev/zap/channel as dchannel?
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13:46.48cced2:)
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14:00.20WeeZyyyGood Morning Everyone
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14:15.09n0cturnal_is it possible to have any calls from exten x go out one route, but any call from exten z go out another ?
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14:38.12MoutaPThi, is there any simple tip to make CDR of all calls that go to my voicemail if i'm busy or unavailable?
14:38.48tzangerwhat does that have to do with CDR?
14:39.23tzangerDial() with option j will jump to n+101 if you can't be contacted, and you can use that to go to VoiceMail()if you're busy
14:39.37tzangerand add a timeout to Dial and n+1 to go to voicemail if you're unavailable (i..e the Dial() times out)
14:40.22MoutaPTyes but i want to make record that this call went to voicemail, and i didn't pickup that call
14:41.34MoutaPTthen i would be able to measure how many calls i receive per month , and how many of them go to the voicemail, because i didn't pick it up
14:41.52coppicewhy do some people believe 64kbps is some magic number that creates perfect phone calls? :-)
14:44.45tzangercoppice: :-)
14:45.20tzangerMoutaPT: you can alter the user field with Set(CDR(...)) before hitting the VoiceMail() app
14:45.27tzangershow function CDR will give you details
14:45.36tzanger64kbps is teh winz
14:45.39MoutaPTthanks tzanger!
14:46.31coppicetzanger: I think the same clowns listen to MP3s at 64kbps and never notice how much better than a phone call they sound :-)
14:48.23tzangermeh.  that's just because we can't convince the telcos to do CBR MP3s or OGGs in the trunks
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14:57.09websaeHappy Easter everyone!
15:00.48DaminHappe Easter...
15:02.11robin_szeaster would be better if my GXP2000 was working :(
15:02.53robin_szwheres helix and/or thetatag when you want someone to whinge at? huh?
15:06.34MoutaPTexten => s-BUSY,2,Set(CDR(dst)=Voicemail_busy)
15:06.34MoutaPTexten => s-BUSY,3,Voicemail(b${ARG1}@${VMCONTEXT})
15:06.47MoutaPTis this enough to get CDR of a call that went to voicemail?
15:07.02MoutaPTi couldn't get it yet..
15:11.05BladeRunner05puzzled: r u there ?
15:15.03BladeRunner05I need to use my eicon diva server bri to asterisk 1.2.7.1, I have installed chan_capi-com.0.6.5 but when I start asterisk it say:  chan_capi.c:4577 cc_init_capi: CAPI not installed, CAPI disabled!
15:15.21BladeRunner05mits on debian kernel 2.6.15
15:16.30puzzledBladeRunner05: check the output of capiinfo
15:18.00BladeRunner05capiinfo say: capi not installed - No such device or address (6)
15:19.11puzzledBladeRunner05: you have to load the capi modules first, then start asterisk with chan_capi-cm
15:20.34puzzledBladeRunner05: here's what I have in /etc/rc.local  http://pastebin.com/663188
15:21.51BladeRunner05ok, take a look puzzled
15:24.24BladeRunner05puzzled: ok now the driver are loaded correctly, I also copy the firmware in /usr/share/eicon ...
15:24.59BladeRunner05can I see a copy of your capi.conf file ?
15:25.10puzzledBladeRunner05: no :)
15:25.15puzzledit's mine!
15:26.30puzzledBladeRunner05: there is nothing special to see in my capi.conf. The author has it commented very well. just read it thoroughly
15:27.19puzzledBladeRunner05: I only added the msn number and changed the callgroup and pickupgroup
15:27.32puzzledlike that it works fine for me
15:27.57BladeRunner05kk
15:31.35tecnicoanyone knows of a AGI or any type of script or way to let iax2 client users to know who else is "online" (registered) , without using Jabber/aim/etc. ? Maybe a webpage ?
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15:37.06x86tecnico: dont know of one existing, but i could write you some web software to do it if you wanted to hire me
15:37.48tecnicolooks like there was someone already with an app (http://www.sineapps.com/news.php?rssid=342), but the link is dead...
15:38.11tecnicotnx. x86 , but this is just personal.. not for profit..  wouldn't make sense for me to pay anyone
15:38.25x86tecnico: maybe you could trade something then?
15:39.37tecnicoI could tell you where to maybe get a free asterisk enabled VPS account. (ztdummy/etc enabled)
15:40.21x86CPS?
15:40.22x86err
15:40.23x86VPS?
15:40.36tecnicovirtual private server... you get your own root... arround 5GB
15:40.45x86free?
15:41.17tecnicomaybe.... can't guarantee
15:41.35BladeRunner05puzzled: r u using debian ?
15:41.54puzzledBladeRunner05: nope, FC4/5 and CentOS43
15:42.01BladeRunner05kk
15:44.07RoyKhm. nice sensors output http://pastebin.ca/49514 i wonder what they were smoking.
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15:47.45RoyKhehe
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16:02.07Corydon76-homeLucas_Fernando: please turn off your auto-away announce.
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16:09.56dlynestecnico: try flash operator panel
16:10.20tecnicotnx dlynes
16:10.42dlynestecnico: it's overkill for what you want, but it does what you want, too
16:11.09BladeRunner05I'm getting error compiling mpg123 on debian kernel 2.6.15 i486, I try make linux, make linux-486, make linux-pentium, make generic but always get errors
16:11.18tecnicodlynes: tnx...I remember it from Astricon2004 but never used it..
16:11.35dlynestecnico: Yeah...it's pretty straightforward to set up
16:11.47robin_szbladerunner: which mpg123 .. the standard one or the one that comes with *?
16:12.22ManxPowerrobin_sz, they are the same
16:12.23BladeRunner05I try both .r and .s get it from http://www.mpg123.de/mpg123/mpg123-pre0.59s.tar.gz
16:12.39ManxPowerWell s prolly isn't going to work with Asteirsk.
16:12.47ManxPowerWhere did you paste the error message?
16:12.59robin_szManxPower: thought the * one was tweked a bit ... no?
16:12.59dlynesBladeRunner05: I wouldn't even suggest using mpg123...it's got a really bad bug in it, where the process loses contact with asterisk
16:13.10BladeRunner05but .r version won't compile
16:13.13dlynesBladeRunner05: madplay plays nicer with asterisk
16:13.30dlynesBladeRunner05: and asterisk 1.2 also supports mpg natively for moh; you don't need an external player any more
16:13.41robin_szkewl
16:13.56BladeRunner05really ? I have compiled 1.2.7.1 and I don't need mpg123 ?
16:14.08robin_szseems not
16:14.10dlynesBladeRunner05: nope...use mode=files
16:14.27RoyKnative moh starts the music for every new call
16:14.31RoyKi dislike that
16:14.36robin_sznow ... fix my GXP2000 for me ;)
16:14.40RoyKso i use a sox wrapper instead....
16:14.42dlynesRoyK: so use madplay then :)
16:14.45BladeRunner05dylnes: where I have to use it ?
16:15.04dlynesRoyK: I just hate mpg123 because when it loses contact with asterisk, it hogs up 20% of the cpu
16:15.33dlynesLucas_Fernando: lose your autoaway
16:16.13dlynesBladeRunner05: if you take a look at your sample musiconhold.conf file, it gives you an example for mode=files
16:16.30dlynesBladeRunner05: So just copy that info into your [default] musiconhold class
16:16.40BladeRunner05ok
16:16.44RoyKdlynes: they all do that
16:16.56dlynesRoyK: all what do what?
16:17.32dlynesRoyK: all external players hog up 20% of your cpu?
16:17.58dlynesRoyK: I've never had madplay do that; only mpg123
16:18.04RoyKsox does it
16:18.23RoyKmy soxwrapper spawns a sox process and that process hangs
16:18.28dlyneswhat does sox have to do with mpg123?
16:18.29RoyKsame with mpg123
16:18.52ManxPowerribnope
16:18.54dlynesor madplay for that matter?
16:19.00ManxPowerrobin_sz, nope
16:19.02RoyKit doesn't matter what software. the problem is that children of the called process aren't killed
16:19.16dlynesRoyK: Whether they're killed or not
16:19.18RoyKbut if madplay doesn't spawn new children, it should work well
16:19.30dlynesRoyK: I've never had an issue with madplay using up 20% of the cpu
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16:19.42RoyKmodplay doesn't spawn a child, then, does it?
16:19.44dlynesRoyK: I've never had an issue with more than one process of madplay running, either
16:19.59RoyKor perhaps it kills the child if it receives a sigterm
16:20.14dlynesRoyK: Wouldn't that be the way it's supposed to run?
16:20.22dlynesRoyK: i.e. play nice with the OS?
16:21.35dlynesRoyK: to me, any process that hogs up 20% of the cpu for no good reason is a security risk
16:21.54RoyKit is
16:22.02dlynesRoyK: Especially when it causes my main process not to respond in a timely fashion (asterisk)
16:22.29dlynesI haven't used mpg123 for quite some time now
16:22.49dlynesWhether it starts from the beginning or not is of little concern to me, when it's doing stuff like that
16:22.49RoyKasterisk calls soxwrapper, soxwrapper calls sox, asterisk stops and soxwrapper is killed, sox goes on. same story with mpg123
16:23.18dlynesThen it's a bug in mpg123 and soxwrapper
16:24.03dlynesI never used it for long enough to find out why it was doing it
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16:24.19dlynesAs soon as I found out it was doing it, I started looking for alternatives
16:24.33dlynesThat's when I found out about madplay on the musiconhold part of the asterisk wiki
16:25.06dlynesWell..it was actually the musiconhold/faxing part of the asterisk wiki
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16:26.24dlyneshttp://www.bartroos.com/asterisk/ <-- Asterisk + ISDN HFC_PCI + Music-on-hold + Soft fax HOWTO
16:27.22dlynesYou can also get libmadplay and write your own music-on-hold module
16:28.27dlynesBut, if you like sox, soxwrapper probably wouldn't be that hard to fix
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16:31.38wwalkerWhat about a patch to asterisk that starts the external moh player at nice 20?  then even if it goes into runaway it should always relinquish CPU to asterisk when asterisk wants it?
16:36.52wasimany body use astpp?
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16:47.33dlyneswwalker: that's not a fix though...that's an ignore the problem and hope it goes away philosophy
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16:49.42wwalkerI miscommunicated.  The soxwrapper problem should be fixed, but many things can make the child go wild, therefore, I'd always want any MoH child to be niced.
16:50.07Hmmhesayshey hey, you you, get off of my cloud
16:51.27wwalkerI first saw vi runaways over 15 years ago.  I thought they had them fixed for years.  I've recently been seeing runaway vi/vim processes again.  someone introduced a new bug.  Somewhere someone is reading from the tty and not catching that it's gone...ina tight loop, 99%cpu....
16:51.59wwalkerSo, I expect that sox*, mpg123, and libmad derived works all have that same probable failure.
16:54.02Hmmhesaysthis is rock n roll radio, come on lets rock n roll with the ramones
16:55.50MoutaPThow do i enable call waiting for every SIP user i create? any tip?
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17:02.54dlyneswwalker: ah....but libmad doesn't have that failure, afaik...it doesn't have any child processes...only the main process
17:03.05dlyness/libmad/madplay
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17:13.17dlynesbbl\
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17:18.49SkramXI have a very nice pipe for our company's servers but how do you all do QoS?
17:18.58BladeRunner05what does it means:  Music class default requested but no musiconhold loaded.
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17:35.39QwellLucas_Fernando: stop that
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17:37.12wasimoui, non hable espanol, capice?
17:37.58BladeRunner05what does it means:  Music class default requested but no musiconhold loaded.
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17:48.41*** mode/#asterisk [+b %Lucas_Fernando!*@*] by file
17:49.16talljon84Hi all-- I have a VM box that has a blank message in it that the user can not delete. It always says they have one new message in the mailbox but there is no recorded message. How do I clear that?
17:49.17Hmmhesaysyou gots no music
17:49.27Hmmhesaysdelete it out of the users folder
17:50.29talljon84will Asterisk recreate it on it's own?
17:50.40Hmmhesaysdelete the voicemail file itself
17:50.44Hmmhesaysnot the whole folder
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17:50.52talljon84ok.
17:50.56rene-~seen Qwell[]
17:51.07jbotqwell[] <i=north@unaffiliated/qwell> was last seen on IRC in channel #asterisk, 1d 20h 25m 15s ago, saying: 'VoIPMasta: anything that gets added to the bug, will get emailed to me'.
17:51.07Qwell?
17:51.09rene-heh
17:51.25rene-Qwell: what is the name of the realtime family for agents
17:51.43QwellI don't know.  Agent?
17:52.07HmmhesaysQwell, make me some coffee
17:52.18talljon84Hmmmhesays: awesome. that worked. thanks a ton
17:52.25Hmmhesaysnp
17:53.04rene-I downloaded trunk 1.2 from svn, i am going to try that
17:53.46talljon84Ok, now next problem: I have a lawyer who has a legal research program he uses. It's only method of update is through a dialup connection that IT must dial. Faxing works fine for me over SIP; however, using the same SIP ATA, the computer won't dialup. Is there any extra configuration anyone knows of to make a dialup connection work across SIP? (using a Sipura ATA)
17:54.29BladeRunner05Problem: with asterisk 1.2.7.1 in modules.conf I have load res_musiconhold.so, in musiconhold.conf I have enabled the default, but when I make a call and press hold button asterisk say Music class default requested but no musiconhold loaded, what this mean ? what I have to do ?
17:54.36Hmmhesaysgotta look at the error messages the program spits back
17:54.46*** join/#asterisk jofre (n=jofre@200.215.42.23)
17:55.56nettieHey guys anyone know what's the latest firmware for the grandstream handytone 286 please? Mine is Program-- 1.0.7.19    Bootloader-- 1.0.8.9    HTML-- 1.0.7.18    VOC-- 1.0.0.10 and I'm having stability issues during heavy T.38 faxing. Any idea please?
17:57.58*** join/#asterisk MstlyHrmls (n=mh@66.193.14.132)
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18:03.53Qwellrene-: realtime agents won't be in 1.2
18:04.07*** join/#asterisk IceManRISK (n=kart@201.10.99.247)
18:05.16rene-i tried to configure it the same way as rt queues and that failed, i am looking at the patch, is dated 11/05 i wonder if i can apply it to SVN trunk 1.2...
18:05.44IceManRISKanyone here uses asterisk with a2billing ?
18:13.31*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-92-37.red.bezeqint.net)
18:16.01*** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt)
18:16.42wiseguy_sorry, is it possible to do something (extensions.conf) after call ends?
18:17.24rene-wiseguy_: just add a priority after your dial or hangup statement
18:17.40*** join/#asterisk heka (n=heka@82.114.68.124)
18:17.44rene-are you receving or making this call?
18:18.28wiseguy_i have something like that
18:18.30*** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net)
18:18.44*** join/#asterisk lyroy (n=toor@modemcable146.87-83-70.mc.videotron.ca)
18:19.03*** join/#asterisk Assid (n=assid@203.115.64.8)
18:19.07Assidheylo
18:19.11wiseguy_http://pastebin.ca/49525
18:19.33Assidanyone got a changelog available from 1.2.7 all the way down to 1.2.4
18:19.35wiseguy_but system command is not executed after call ends..
18:20.14lyroyIf I have more than 1 sip device, and when I receive a call I want all my sip device to ring, after someone answer the call, the other device should not ring... how it's possible with asterisk?
18:20.47Assidlyroy: just use the dial cmd
18:22.44rene-wiseguy: in your example system gets executed both when the call finished and also when the call cant be connected
18:22.59rene-e.g. no answer
18:23.00AssidDial(SIP/blah&SIP/user2&SIP/user3)
18:23.38Assiderr.. anyone jhave a changelog ?
18:23.49rene-wiseguy_: if this is not what you want you can take a look at defining an h extension for your incoming context
18:24.03rene-wiseguy_: http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
18:24.11lyroyalrigth thx assid
18:25.57wiseguy_rene-: ghem, the problem is the command is not executed in this case
18:26.38wiseguy_:/
18:26.40rene-well that is because when the other party cant connect, your priority will jump to +101
18:27.16rene-http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities
18:27.36rene-so http://pastebin.ca/49527
18:30.15wiseguy_it does not work
18:30.16wiseguy_:/
18:31.06rene-is the dial command connecting the call?
18:31.26wiseguy_yes, the dial command
18:31.29wiseguy_goes okaj
18:32.36wiseguy_rene-: any other ideas?
18:32.52rene-well if the dial goes ok, then priority 103 doesnt get executed but 3 should
18:33.11rene-re add the priority 3 above priority 103
18:33.53cybergypsywiseguy - have you tried the g option ?
18:34.01cybergypsywiseguy - i have a similar problem
18:34.18wiseguy_g option?
18:34.33cybergypsyg: When the called party hangs up, exit to execute more commands in the current context.
18:34.54wiseguy_cybergypsy: can you paste an example?
18:35.00wiseguy_to pastebin.ca
18:35.44cybergypsyexten => _0044808XXXXXXX,n,Dial(IAX2/iaxfwd/*${EXTEN:2},60,hHg)         ; UK 0808 Numbers - Free
18:35.52*** join/#asterisk _42 (n=r2d2@nvader.sh.nu)
18:35.53PakiPenguindid anyone try configuring a  TE110P  with mitel?
18:36.20_42Anybody have any experience in getting asterisk working with Sunrocket's voip? www.sunrocket.com
18:36.22rene-PakiPanguin, they work great both in R2 and ISDN modes
18:37.27wiseguy_cybergypsy: it doesn't work for me
18:37.55_42Nevermind, Sunrocket blocks access to their SIP servers
18:38.38PakiPenguinrene-, can you share your zaptel configuration please?
18:38.57PakiPenguini have a mitel 3300 right now , and the the e1 card is in red alarm constantly
18:39.05*** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com)
18:39.25rene-well if its in red then the problem is with your cable i think
18:39.48rene-i no longer have access to configs, are you using ISDN?
18:40.17*** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net)
18:40.31rene-does zttool shows your card as being up?
18:40.58PakiPenguinyes
18:41.03PakiPenguinfor the ISDN
18:41.18PakiPenguinAlarms          Span                                               â
18:41.18PakiPenguin<PROTECTED>
18:41.27PakiPenguinand this is the zttool output
18:41.28rene-what type of cable are you using to link the mitel and asterisk?
18:41.44*** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net)
18:43.34PakiPenguina sec
18:45.03wiseguy_rene-: have you got an working example? I need to execute that command when call ends not depending on the reason of end
18:49.20talljon84I'm trying to setup outbound SIP calling on * but even if I DMZ the server, it always fails when the remote end trys to answer the phone. It will ring, but when they answer, it fails. Any ideas?
18:49.46rene-talljon84: codecs?
18:50.16talljon84the trunk is alowing g729 or ulaw
18:50.29rene-use of stun in the remote end?
18:50.47rene-better
18:50.51rene-use qualify
18:50.59rene-qualify=yes
18:51.02rene-nat=yes
18:51.31talljon84on the SIP entry or the general sectioN?
18:51.38rene-on the sip entry
18:52.11talljon84ok. lemme try
18:52.11rene-of the remote peer
18:53.02talljon84ok, now it will connect (yay!) and I can transmit, but the remote phone can't send audio back (like it's on mute)
18:53.24rene-they cant talk back to you
18:53.30talljon84right
18:53.52rene-what device?
18:54.12Qwellis the * server behind NAT also?
18:54.14talljon84the remote end is a cellphone on the PSTN.. the local device is a Cisco 7960 runing SIP firmware
18:54.20Qwellof course it is...
18:54.26QwellYou need externip and localnet
18:55.01talljon84Qwell: my IP here is dynamic so I can't set externip
18:55.10Qwellthen use externhost
18:55.13rene-get a dyndns account
18:58.16*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
18:59.31talljon84Qwell: I set externip to see if it would resolve the problem and it does. Thanks a ton for that. I know this is outside of the scope of this group but, are you aware of anyway to configure a remote BIND server to accept an update for it? Example: host.domain.com wants to update it's static IP address as itgets it to ns1.domain.com which is running BIND9. Is this even possible?
18:59.54talljon84*it's new dynamic IP address
19:01.35QwellI think that'd part of the point of rndc
19:02.21rene-talljon84: use externrefresh and externhost with a dyndns acct, supported in asterisk 1.2+
19:03.12talljon84rene-  Ok, thanks.
19:03.42talljon84One last Q: Any preference for a compressed codec? g723, g726 or g729?
19:04.42rene-g729 is your friend if bandwidth is limited but you have to purchase licenses for your *
19:05.48talljon84rene-: I'm not as concerned about bandwidth usage in this deployment. thoughts on g723 or g726?
19:06.00rene-havent really used them
19:06.26rene-if you are on lan use g711, best audio quality
19:06.39rene-g729 is not as crisp but well is 8:1 compresion
19:06.44Qwellg723 isn't supported in asterisk
19:07.18*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
19:07.34talljon84I'm using g711 for the LAN, but I'd like to compress it a little for traversal across the net.
19:07.43*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
19:07.51Qwellso use g729
19:08.22CMikehm.. anybody if I easily can save the useragent for a client in realtime ?
19:09.09*** join/#asterisk apardo (n=apardo@87.218.45.206)
19:10.51*** join/#asterisk Dovid (n=Dovid@85-250-190-83.bb.netvision.net.il)
19:15.03rene-Qwell: the patch applied almost cleanly but it still isnt working, i have already loading agents from database via include files  so i guess i will wait for 1.4 for that
19:15.24*** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net)
19:16.04DovidHappy easter everyon e
19:24.33*** join/#asterisk austinnichols102 (n=austinni@dsl-10-169.cofs.net)
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19:42.41mitchelocditto Dovid
19:42.48Dovidhehe
19:42.52Dovidthis room is busy
19:43.02Dovidi am jewish so i got nothin to celebrate
19:43.08mitcheloceveryone is getting stuffed or preparing to ;)
19:43.44mitcheloci think even non-religious people take the opportunity to spend time with family, so it's not necessarily only for religious people
19:44.35Dovidguess so
19:44.41Dovidi am here. tryin to get some work done
19:44.51*** join/#asterisk DeV-rAd (n=jesse@fl-69-69-130-197.sta.sprint-hsd.net)
19:45.16Dovidlol
19:45.23Dovidyes tzafir it is Pesach
19:45.44mitchelocenglish please =P
19:45.57DovidPesach is hebrew for Passover
19:46.14mitchelocahh
19:46.20file[laptop]it's rabbit day
19:46.26file[laptop]KILL HIM!
19:46.28*** join/#asterisk delacko (n=delacko@lns01-1076.dsl.iskon.hr)
19:46.37Dovidlol
19:46.54DeV-rAdNeed Some help with asterisk@home get the error 404 on out dial
19:47.29mtghAll asterisk people, I need some C help with a non asterisk thing, but it should be quick, is there anyone who can help me?
19:47.58mitchelocDeV-rAd: from the channel topic - "asterisk@home users should join #freepbx for support"
19:48.13DeV-rAdok thank you
19:48.21mitchelocnp
19:49.24tzafrir_laptopmtgh, tried #c or something similar?
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19:49.51*** part/#asterisk DeV-rAd (n=jesse@fl-69-69-130-197.sta.sprint-hsd.net)
19:56.48talljon84Does anyone know of a way to get MOH (using madplay) to get streaming audio as a source (such as shoutcast)?
20:04.01*** join/#asterisk hodrige (n=Hodrige@ip68-98-172-123.dc.dc.cox.net)
20:04.23*** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
20:04.37*** join/#asterisk angom_h (n=angom@200.76.229.86)
20:06.17*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
20:08.33dlynestalljon84: I'm using g726 on some installs...it's pretty good
20:08.47dlynestalljon84: afaik, asterisk still only supports g726-32, however
20:22.30*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
20:32.28blitzrageQwell: I am here!
20:32.36file[laptop]blitzrage: OH NOES
20:32.44blitzrage:-O
20:33.30file[laptop]blitzrage: LJAM
20:34.01*** join/#asterisk redcap1 (n=phez@redcap.xs4all.nl)
20:35.06blitzrageoh no you didn't
20:35.24*** join/#asterisk Seggy (i=rbutler@tsss.org)
20:39.48*** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net)
20:47.54*** join/#asterisk Sedorox (n=penbra67@smartserv/cna/Sedorox)
21:23.42Dovidso am i
21:25.28*** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net)
21:26.41robin_szpoxy new software ... worse than before
21:26.45Sedorox8
21:26.47Sedoroxer
21:27.08Sedoroxwell I'll bbl
21:28.47robin_szhttp://www.atcom.cn/En_products_At320EE.html <=== are these any good?
21:30.24tecnicoanyone knows what module provides "ast_park_call" ?? when trying to load chan_sip, I get "undefined symbol: ast_park_call"
21:30.30file[laptop]res_features
21:30.35tecnicotnx. file
21:30.45Dovidit looks cheap
21:30.55robin_sztrue .. it does
21:31.00robin_szbut it is cheap ...
21:31.09robin_szthe question is: does it work OK :)
21:31.27robin_szvoip phones are WAY overpriced
21:31.40robin_szthey cost no more than POTS phones to make
21:31.45*** join/#asterisk mwright1night (n=mwright1@203-214-48-213.dyn.iinet.net.au)
21:31.53robin_szand sell for 30 times the rate ... things have to change
21:32.54robin_szwell, 10 times the rate then
21:33.47Dovideh
21:33.53Dovidi like the spa-841
21:34.16robin_szprice?
21:34.42*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
21:34.53Dovidi think 85
21:35.20Dovidvoipsupply.com
21:35.25Dovidor ebay
21:35.45DoktorGregactually, if you compair voip phones to key system phones, their prices are quite favoriable
21:35.52*** join/#asterisk apardo (n=apardo@87.218.45.213)
21:36.07file[laptop]VoIP phones have to be smarter then you think, and have more then a regular ol' POTS phone
21:36.12DoktorGregor even a nice analog wireless phone...
21:36.30DoktorGregcan easily pay 150 for a nice analog wireless phone
21:36.41tecnicoI have the spa-841... I wouldn't recommend it..
21:37.05Dovidi like it for a basic phone
21:37.11Dovidpolycoms r the best
21:37.11tecnicospeakerphone is terrible..
21:37.24Dovidyes, but for a basic phone its good
21:39.21DoktorGregman last night i got stuck using a 4x cd burner
21:39.22*** join/#asterisk CukX (n=cuk@nu.cuk.nu)
21:39.40DoktorGregthis 32x cd burner i have at home is much nicer
21:39.43CukXcan anyone enlighten me, please ?
21:39.50CukXI have ISDN HFC-S card and wich drivers to use with it ?
21:40.56CukXno, really, I don't have a clue on all that... mISDN, isdn4linux, zaptel, ....
21:42.44tecnicoany hints on what's causing this? : "chan_sip.c:9633 handle_response_register: Got 200 OK on REGISTER that isn't a register"
21:44.22robin_szfile[laptop]: no, you misunderstand .. the phoen might have to be "smarter" ... BUT  that doesnt mean it costs more in bulk ... infact, the ethernet if is probably cheaper to if than a POTS interface
21:44.42blitzrage<PROTECTED>
21:44.43robin_szthe smartness is in the code ... code it once and you;re done
21:46.01mwright1nightrobin_sz: I think at the cisco buying millions of them level, they both cost a few cents
21:46.11robin_szright
21:46.35robin_szbut a cisco 7940g still sells for like $350
21:46.44Dovidget a polycom
21:47.01mwright1nightchinese (near) slave labour gives us lots of cheap plastic things... uh oh, we need oil for plastic stuff, and china needs oil for it.. but but but
21:47.01robin_szthey should come down to the <$100level in the not too far off
21:47.14Dovidhehe
21:47.14mwright1nightwe want the oil for us, but we want to buy the plastic chinese things
21:47.16Dovidu wish
21:47.17mwright1nightwhat do we do
21:47.18Dovidu get what u pay for
21:47.41mwright1nightThe cisco's will come down to that pricing point you think?
21:47.50robin_szsure ... well
21:47.54DoktorGregthe phones with the big color displays???
21:47.55Dovidnope
21:47.56robin_sznot as cisco
21:48.11DoktorGregthe discount cicsos will be the sipuras
21:48.15robin_szbut when Walmart have them rolling off the shelves
21:48.31Dovidi wish
21:48.35Dovidthat will be the day
21:48.37robin_szand they are in 50% of broadband-equipped homes ...
21:48.47robin_szI say, thats .. what .. 18 months away
21:49.00DoktorGregin 5 years the POTS service providers will be closing up ship
21:49.03DoktorGregshop
21:49.07robin_szagreed
21:49.26DoktorGregand you will be able to buy a no frills voip phone for $30
21:49.27robin_szalready BT who had been fighting voip takeup by businesses
21:49.39robin_szare now offereign it as an add on on business broadband
21:49.52robin_szthats a sure sign its gone mainstream
21:50.06mwright1nightsomeone has to do complicated switching
21:50.11Dovidi wish
21:50.18DoktorGregcomcast is about ready to go zero $ install on their voip service
21:50.19mwright1nightdirectory for all the number allocations
21:50.22robin_szand now it will happen very fast ... because the under pinnin technolgy (adsl) is already widely available and installed
21:50.36mwright1nightsomeone has to maintain the last mile copper
21:50.41robin_szyeap
21:50.45Dovidsome people will still not get the internet
21:50.49mwright1nightin Australia where I'm from this is very expensive
21:50.49robin_szthats broadband providers ...
21:51.18mwright1nightour 49% state owned Telecommunications provider owns it
21:51.29robin_szit will just "happen" .. you'll pay your $30 for the line and broadband
21:51.31DoktorGregI think the cable IP backbone makes more sense than the pots telco backbone
21:51.38robin_szand another $10 for your mothly calls
21:51.45Dovidonly time will tell
21:51.57DoktorGregwith phone you need  pair of wires between you and co
21:51.59robin_szi think it wont be long now ... its already happening
21:52.00*** join/#asterisk Lino` (n=Lino@i577BDF8D.versanet.de)
21:52.14robin_szanyway ...
21:52.18mwright1nightI pay 89.95 + 33 to my broadband provider atm and get 1x POTS line 1x VOIP line and 1x 40GB 8AM --> 12Midnight, 1x 40GB 12midnight --> 8AM
21:52.24DoktorGregwith cable you need a piece of coax between you and drop...
21:52.48*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
21:52.59mwright1nightexchange rate is .70
21:53.02robin_szin the UK, it generally about £24/month for 512K adsl, no limit
21:53.08mwright1nightso that's a hell of a lot more expensive
21:53.25mitchelocwhat is a "1x 50GB 8AM"??
21:53.27mwright1nightwell this is 19000/1024
21:53.30mitcheloc*40
21:53.42mwright1night40GB quota per month
21:53.45DoktorGregIm on comcast business about $160 to get 1.5Mb up and 8Mb down
21:53.48mwright1nightfrom Midnight to 8AM
21:53.51mitchelocoh, ouch, where are you from?
21:53.53mwright1nightand 40 GB for peak
21:54.10mitcheloci use that in a week =P
21:54.22mwright1nightAustralia,  we have the worlds most sparsely populated contenent
21:54.25robin_szsigh . if nly I hadnt "upgraded" this GXP200 I could seel the fscker on ebay :(
21:54.29mwright1nightso do I
21:54.43robin_szAustralia?
21:54.43mwright1nightthat's why I'm about to be throttled in 6mins to 64k
21:54.52mwright1nightyep
21:54.57robin_szIve always wanted to go to Australia ...
21:55.14robin_szbut I don't have a criminal record :(
21:55.15mwright1nightexpensive broadband.. don't come (hehe)
21:55.32mwright1nightwell our immigration dept won't let you in if you don't have a criminal record
21:55.35mwright1nightI'm a kiwi
21:55.36Dovidlol
21:55.39Dovidu need one to get in ?
21:55.43mwright1nightI live in aus
21:55.53mwright1night5 mins to Shaping (throttling)
21:56.00robin_szDovid: traditionally, from the UK yes ...
21:56.00DoktorGregdo you guys have on demand cable down under yet?
21:56.05Dovidlol
21:56.08Dovidgood to know
21:56.15robin_szDovid: stealing a sheep is the most popular route I think
21:56.20Dovidlol
21:56.29Dovidso if i am caught wit something i jus tmove there ?
21:56.44mitchelocstop giving me more reasons to stay in california!
21:56.46mwright1nightI think stealing a loaf of bread
21:56.52mwright1nightor not having a home to go to
21:56.57DovidLA ROCKS
21:57.00Dovidi am thinkin of movin there
21:57.06mwright1nightdoktorgreg: what do you mean by ondemand cable?
21:57.10robin_szmwright1night: heh, indeed :)
21:57.30mitcheloclol, i'll trade Dovid ;)
21:57.43Dovidhehe
21:57.46Dovidto where i am ?
21:57.57mitchelocmmm...good point, maybe not
21:58.02Dovidlol
21:58.05Dovidits nice here
21:58.36Dovidnot too hard to get a gun
21:58.37Dovidlol
21:58.41Dovidand maybe kill some one
21:58.43robin_szmwright1night: at one point, they offered £400 cash and a free boat ticket ... that was in the 30s I think
21:58.44Dovid:)
21:58.51Dovidlol
21:58.58Doviddamn. to get rid of the criminals ?>
21:59.13mitcheloc*creepy*
21:59.14robin_szno, criminals just got sent anyway
21:59.21mwright1nightrobin_sz
21:59.24mitchelocheh i just remembered i need to do my taxes!
21:59.28Dovidso do i
21:59.32mwright1nightyep that is how all the Italians came
21:59.36robin_szheh
21:59.36mwright1nightand Greeks, and slavs
21:59.37Dovidthanks for reminding me
21:59.41mwright1nightthey did really well
21:59.43mitchelocyour welcome
22:00.04Dovidcalling accountant now
22:00.06Dovid...
22:00.08DoktorGregwow i havent looked at a proper linux distro in a while debial is really really nice
22:00.09mwright1night1 minutes until capping
22:00.10mwright1nightdoh
22:00.27mitchelocquick send mwright pr0n in non-compressed format!
22:00.29mwright1nightdoktorgreg: check out ubuntu dapper drake, due to be released june 1
22:00.50robin_szDoktorGreg: ignore them ... stick with debian. the one true way
22:00.59mwright1nightI have paused my torrents waiting for the last 20mb of my asterisk@home 2.8 to come in
22:01.16DoktorGregwhats not to like about debian?
22:01.17Dovidnah
22:01.18mwright1night2:53 to go @ 246KB/sec
22:01.20DovidCENT OS RULES
22:01.21Dovidlol
22:01.32mwright1night30MB to cover
22:01.34robin_szDoktorGreg: nothing .. its great.
22:01.36mwright1nightin 0 secs
22:01.46Dovidi get 7.0 KB where i colo
22:01.52DoktorGregit found all my hardware?!!
22:01.58robin_szexactly
22:02.02DoktorGreghow can I complain about that?
22:02.03mwright1nightand you're in a big broadband country
22:02.05mitchelocI get 600+KB/sec where I colo ;)
22:02.13DoktorGregomg i just converted distros
22:02.14Dovidoops
22:02.15robin_szand new stuff is just an "apt-get install" away
22:02.17Dovidnot KB
22:02.18mitcheloci can get a full linux distro in less then 12 minutes
22:02.18DovidMB
22:02.20DoktorGregI was slackware forever type
22:02.31mwright1nightit musn't be 8AM yet, my isp hasn't throttled me, 1min 18 to go
22:02.49Dovidi recently converted
22:02.50mwright1nightI am a Fedora / RHEL person, however ubuntu rocks
22:02.54Dovidi am now CentOS
22:03.02robin_szIused to be RH,
22:03.05DoktorGregbut slackware is too old school now days
22:03.08robin_szbut debian is MUCH better
22:03.15mitchelocthe fedora people are awesome for including mono in fc5 ;)
22:03.18Dovidwhat do u like with debian ?
22:03.25mwright1nightI think after I Dapper Drake comes out, I will switch my www.ltsp.org / FreeNX based terminal server to Ubuntu Dapper Drake
22:03.27Dovidi dont know y anyone uses fc
22:03.33Dovidif u have cent os
22:03.34mitcheloci do
22:03.36robin_szfedora tries hard, but has WAY too short a lifecycle for servers
22:03.43mwright1nightwhy is mono good
22:03.46Dovidthats y i use cent os
22:03.46mwright1nightit is a slow pig
22:03.47DoktorGregso far ive looked at asterisk on debian in rapid asterisk distro
22:04.05mitcheloci prefer to program on it, and there is nothing wrong with that!
22:04.08DoktorGregused apt-get to get some thigns
22:04.20robin_szapt-get roxxors
22:04.20mwright1nightI like ubuntu for desktop
22:04.21DoktorGregwas like, "OMG PONIES!!!"
22:04.31robin_szyeah
22:04.39robin_szpony drop!
22:04.40mwright1nightok m ISO came through
22:04.44mwright1nightand I still don't feel shaped yet
22:04.55mitchelocpink slashdot hahahaa
22:05.00DoktorGregwas having some problems getting the asterisk 1.2 branch compile
22:05.16DoktorGregso decided to start from a clean debian install and dump the rapid thing
22:05.50DoktorGregam experiencing apt-get based install right now
22:05.59DoktorGregand so far I am like
22:06.04DoktorGregOMG PONIES!
22:06.47DoktorGregOMG Ponies, fastest meme ever
22:07.21Dovidponies ?
22:07.32DoktorGregapril first edition of slashdot this year
22:07.49DoktorGreggo look at cuitest website ever article
22:07.53mwright1nightI have been a computer junkie this weekend and got a bad back
22:08.01mwright1nightdo you wall have really nice workstations with high quality chair?
22:08.32mitcheloci had a decent chair but i broke it =/
22:10.35robin_szponies was something a long long time ago on #london.pm
22:12.01Dovidcan u elaborate ?
22:13.15DoktorGregwhen you buy an office chair now days spend the extra money and make sure you get one with a steel truck
22:13.36DoktorGregthe chairs at the office super store almost never have steel trucks
22:13.51DoktorGregyou have to go to office furniture store
22:14.01*** join/#asterisk Greek-B0y (n=fusion@193.220.93.162)
22:15.27*** join/#asterisk lacym (n=lacym@h-68-164-20-5.hstqtx02.covad.net)
22:18.50*** join/#asterisk AsteriskAlbania (i=Asterisk@80.91.113.253)
22:19.21*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
22:19.26QwellAlbania?
22:19.34AsteriskAlbaniayes
22:20.44robin_szhttp://mightymake.en.alibaba.com/product/50108544/50491128/VoIP_Phones/VoIP_SIP_IAX2_H_323_MGCP_Phone.html
22:20.51robin_sz^^ interesing price ...
22:21.13robin_szless than £30 each on ebay, new
22:21.53Greek-B0yanyone played with wi-fi phones?
22:22.02QwellI see no price on that site
22:22.21*** join/#asterisk heka (n=heka@82.114.68.124)
22:24.01talljon84All of a sudden, my * is considering every extension to be busy. Extensions with VM are sent directly there and those without are shown as busy. Help! Any ideas why this would be happening?
22:24.26robin_szGreek-B0y: yeah, I have a few Zyxel wifi phones
22:24.40robin_szQwell: search on ebay.co.uk
22:24.45filetalljon84: look on the asterisk console and see what it says...
22:24.54robin_szQwell: sorry less than £40 /// 39.99
22:25.17Dovidtall is ur internet down ?
22:25.22Dovidcan u make calls on the lan ?
22:25.35[av]banirobin_sz: how are they?
22:25.45robin_sz[av]bani: crap!
22:25.46talljon84i can access thing such as VM; however, any call to an extension gets busy.
22:25.47Greek-B0yrobin_sz, what do u think of the zyxel ones? any good? i've read negative reviews about them
22:25.57robin_szGreek-B0y: they are crap
22:26.52robin_szshort battery life, crap networking (more like not-working!), crap menus/gui, only 10% chance of it sitting in the charger and charging etc etc etc etc
22:27.17robin_szinshort, its a bit like a portable GXP2000 :)
22:27.46DoktorGregwhat POE switch do yall recommend?
22:28.11QwellDoktorGreg: anything, as long as it also has poe on the wifi
22:28.17[TK]D-FenderDoktorGreg : D-Link DES-1526
22:28.22mitchelocoooh, that would be cool
22:28.24robin_szPOE-knee!
22:28.53talljon84I'm using aoh which is using an agi script to make dialing. It seems to be exiting the script with "no extensions to dial" and sends it to vm..  Nothing has been changed with the AGI script though so I don't understand why it suddenly stopped working.
22:29.15[TK]D-Fendertalljon84 : Please read the channel topic....
22:29.16Qwelltalljon84: type /topic
22:29.49Greek-B0yrobin_sz: i was hoping to get my hands on something workable. I guess I'll have to go for the more expensive linksys wi-fi phone
22:30.03QwellGreek-B0y: those seem real nice
22:30.17Greek-B0yyeah
22:30.21Qwellphysically anyhow
22:30.26robin_szGreek-B0y: or a DECT phone plugged into a iaxy adapter!
22:30.39[TK]D-FenderGreek-B0y : Check your reviews first.  If its for a fixed SITE, then I'd suggest an ATA+Cordless phone still.
22:30.52Greek-B0ymaybe i should just wait until they release a cellphone with built-in wifi and sip client
22:31.04[TK]D-FenderGreek-B0y : They're out there already...
22:31.08robin_sztheres a dutch thing, that does 10 dect phones to SIP or H323
22:31.18robin_sza sorta DECt base for sip
22:31.24MoutaPTAsterisk in a DMZ, Xlite registers is just fine and the calls are ok, but when i click to hangup the call it takes about 3 or 5 seconds to get the call hanged in the Xlite.. I've checked ASterisk CLI and the call is hanged correctly! any tip?how should i debug this?
22:31.45mitcheloc*hung
22:31.53robin_szI have one somewhere .... I failed to get my DECT phoens to register with it and lost interest  .. its under the desk or behind the bin or something
22:32.34robin_szother people have had good success with them .. if only I could remeber what it was called
22:32.53Greek-B0yso i take it iax2 is still the recommended protocol even for hard phones
22:33.00*** join/#asterisk fugitivo (n=fugitivo@201.255.177.88)
22:33.08QwellGreek-B0y: Only if the hardphone runs asterisk
22:33.13*** join/#asterisk oej (n=oej@tcn003124.tcn-catv.ne.jp)
22:33.14Qwell~iax
22:33.15jbothmm... iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for  Inter-Asterisk Exchange
22:33.26*** join/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com)
22:33.31mitchelocMoutaPT: i suggest buying eyebeam or get another softphone, lots of problems are fixed in their paid version, counterpath secretly leaves us in the dark on that
22:34.11Qwellmitcheloc: They're quite clear that bugs aren't often fixed in xlite :p
22:34.16MoutaPTi've this at home working fine, but at home * is running in my lanno
22:34.21MoutaPTLan no dmz
22:34.32Greek-B0yand what codec is best for lan? ulaw?
22:34.42MoutaPTtomorrow i'm trying a MITEL hardphone
22:34.53MoutaPTcould this be better with hardphone? i hope...
22:34.57mitchelocyea, well it sucks, i was trying out xlite and the audio quality sucked, switched in eyebeam and it worked like a charm....i think it's a conspiracy!
22:35.29Qwellmitcheloc: glad to see I'm not the only paranoid one
22:37.22MoutaPTi think i will try one eyebeam to be sure that is not my mistake... i'm getting paranoid too
22:38.43*** join/#asterisk CrummyGummy (n=wayne@dsl-145-69-235.telkomadsl.co.za)
22:46.10CukXwich driver to use for HFC-S cards, please...
22:46.48puzzledCukX: bristuff patch, mISDN or vISDN
22:47.48puzzledCukX: and if you decide to use bristuff I read on the mailing list that adding the florz patch improves it
22:49.45*** part/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net)
22:49.51*** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net)
22:50.31CukXpuzzled are there any manuals on that ?
22:50.55puzzledCukX: no idea, don't use them
22:51.08CukXi am from freebsd world, but anyway... i have installed debian 3.1 with 2.4.27 kernel...
22:51.18CukXpuzzled should i have 2.6 for mISDN ?
22:51.56puzzledCukX: I would prefer 2.6 over 2.4 in general. no idea if you need 2.6 for either bristuff, mISDN or vISDN
22:52.50CukXi didn't found any relative cheap hw GW's for ISDN
22:53.10dlynesI take it bri is still quite cheap in Europe?
22:53.21dlynesI don't know of anyone using bri in na
22:53.30puzzleddlynes: depends on what you call cheap. I think it's still a ripoff
22:53.42CukXoh, and... one of our ISP offer VoIP, SIP based... can I "make" asterisk, that send traffic to their SIP server and GW ?
22:53.56dlynesCukX: yes, and h323
22:54.21puzzledCukX: search eBay for an Eicon Diva *Server* card and get chan_capi. I have had the best experience with that so far
22:54.26dlynespuzzled: ah...so why do people use it instead of dsl?
22:54.34CukXand have fancy SCCP cisco phone, hihi :)
22:55.12puzzleddlynes: I don't know anyone who uses ISDN instead of DSL. I know people that have downgrade their ISDN line to analog and get DSL on it
22:55.42dlynespuzzled: so why is isdn still used then?
22:55.57dlynespuzzled: is it cheaper than dsl in europe?
22:56.02puzzleddlynes: it's big in businesses
22:56.23dlynespuzzled: yeah...not here...everyone uses dsl or cable
22:56.25CukX
22:56.25CukX
22:56.25CukXEicon DIVA Server BRI-2M PCI aktive ISDN Karte
22:56.46CukXand better than hack with passive HFC-S cards ? and had troubles with echo cancelation, etc, etc ?
22:57.05puzzledCukX: yup that's the one. You can also search for AVM Fritz! card to use with chan_capi but I no experience with that
22:57.42dlynesbtw...anyone know how the hdlc code in the oej branch will help with pri's?
22:57.53puzzleddlynes: it's the smart thing to do money wise. prolly cable even more now they added phone services to it over here and you no longer need a pots line with the added cost
22:57.59CukXdlynes you don't wanna know... our Telco has bought ISDN, 10 years ago, and they offered DSL only over ISDN... because they wanted to cover expenses in ISDN
22:58.36puzzledCukX: typical traditional old telco monopoly
22:58.51CukXpuzzled yep
22:59.07CukXand now... they offer "change" to VoIP phones..
22:59.32CukXand they take all ISDN stuff away and put one Sagem with VoIP and analog POTS
22:59.36puzzleddlynes: I would think that any hdlc stuff lives in libpri but am not sure
23:00.00puzzledCukX: so then you are stuck with your ISDN phones and have to buy new analog ones :)
23:00.02dlynespuzzled: but how is hdlc related to pri?  i'm not sure what it is
23:00.10CukXpuzzled sort of...
23:00.11puzzleddlynes: low level data framing
23:00.28dlynespuzzled: I just know with libpri-trunk and zaptel-trunk that i'm getting messages on my screen about hdlc frames getting dropped now
23:00.48CukXpuzzled or clean dust from old Panasonic phones, wich we didn't throw away
23:01.06puzzleddlynes: testing of trunk is good but if you need it for production I would have a look at the latest 1.2 branch
23:01.16puzzledCukX: good for you :)
23:01.37CukXpuzzled so you recomend Diva over HFC-S cards ?
23:01.40dlynespuzzled: that's just it...latest 1.2 branch wasn't stable for me...trunk is
23:02.03dlynespuzzled: last release that was stable was libpri for zaptel 1.0.9.2
23:02.05CukXpuzzled what about a FXO interface ? Sipura ?
23:02.21puzzledCukX: yeah cause they have echo cancellation on board. If you can find the latest/newer 2.0 revision on eBay than it has even more goodies on board
23:02.52CukXpuzzled how do I know, that it's ver 2 ?
23:03.07puzzleddlynes: well if you had issues/bugs I can only recommend to file them on bugs.digium.com
23:03.08dlynespuzzled: libpri 1.2.2 and zaptel 1.2.5 randomly hang my pri on me...sometimes as often as once a day, other times as often as once a week
23:03.26*** part/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-28-100.houston.res.rr.com)
23:03.40puzzledCukX: the sipura spa-xxxx series are quite popluar. I have a couple here but there are still sitting in their box
23:03.45dlynesthe hanging issue isn't happening with zapte-trunk and libpri-trunk
23:04.16puzzledCukX: version 2 has a different product id. Search the eicon website for it. Iirc it is noted on the datasheet
23:04.36puzzledCukX: if they don't mention it on eBay than it's prolly a version 1.0 which works fine for me
23:04.54puzzleddlynes: hanging pri sucks. did you get core dumps or anything else?
23:05.48dlynespuzzled: nope...asterisk was still running...just giving everyone a busy signal that tried calling into the pri
23:06.08*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
23:06.09CukXpuzzled can I do all that Asterisk on dual 333Mhz PII ? max 4 concurent connections  ?
23:06.09*** join/#asterisk hodrige (n=Hodrige@ip68-98-172-123.dc.dc.cox.net)
23:06.21puzzleddlynes: even if you get zaptel/libpri from trunk going reliably be aware that asterisk trunk is in serious flux at the moment
23:06.40dlynespuzzled: i'm not updating it every day
23:06.42hodrigeHi
23:06.54dlynespuzzled: i just downloaded it once, it's stable, and i'm not updating it again
23:07.00puzzleddlynes: did you try upping the filedescriptor count etc. the ulimit stuff?
23:07.10dlynespuzzled: my policy is, if it ain't broke, don't fix it
23:07.13hodrigeanyone was able to setup DISA on AAH 2.8
23:07.36dlynespuzzled: you mean ulimit -c?
23:07.46puzzledCukX: I have a PII-350 that can do 2 ISDN channels simultaneously without a problem. Just don't do any transcoding
23:07.52dlynespuzzled: I always have it set at unlimited
23:08.01puzzleddlynes: ah ok, that's good
23:08.20dlynesand filedescriptor count is an msdos/windows thing
23:08.28dlynesand even then, it's a throwback to cp/m-80
23:08.42puzzleddlynes: ulimit -n is fd count iirc
23:09.04dlynes1024
23:09.20puzzledlemme check waht I usually throw it to
23:09.27CukXpuzzled do you remember, your Diva card is low-profile, or full sire ?
23:09.29CukXsize
23:09.38puzzledfull height
23:09.47CukXso probably 2.0 is low...
23:09.48puzzledthey all are iirc
23:09.54puzzledthe 1.0 ones
23:10.13dlynesnice to see ulimit doesn't have a manpage :)
23:10.24AsteriskAlbaniaany one has experience TE110P with QUINTUM ?
23:10.50*** join/#asterisk hinckc (n=hinckc@c-68-45-24-192.hsd1.nj.comcast.net)
23:11.34puzzleddlynes: I use ulimit -n 8192
23:12.13dlynespuzzled: what does that do, and why do i need it?
23:12.19*** join/#asterisk mwright1nigh1 (n=mwright1@203-214-48-213.dyn.iinet.net.au)
23:12.31mwright1nigh1what's the default username password for centos asterisk@home
23:12.37mwright1nigh1I am at a login prompt now
23:12.40puzzleddlynes: if you open a lot of files e.g. on an IVR platform
23:12.48bkw_mwright1nigh1, RUDE.. don't bother saying Hi
23:12.53bkw_just bust right in and ask questions
23:12.58dlynesmwright1nigh1: check the topic
23:13.01puzzledmwright1nigh1: browse the manual on the A@H website
23:13.04bkw_mwright1nigh1, I'm sure its like in the docs
23:13.05dlynesmwright1nigh1: try #freepbx
23:13.18SplasPoodMWRIGHT STANDING BY!
23:13.27SplasPoodHe's poised at the prompt
23:13.32SplasPoodhanging on your every word!
23:13.35timscotthaha.
23:13.57dlynespuzzled: the pri machine is just acting as a softswitch, that plays the odd music on hold file and records the odd voicemail, but that's it
23:13.58mwright1nigh1what is the diff between freepbx and @home
23:14.02mitcheloclol mwright1nigh1 got jumped
23:14.03dlynespuzzled: no big usage of files on there
23:14.27dlynesmwright1nigh1: freepbx is the site that produces AMP...@home uses AMP
23:14.35puzzleddlynes: did you check for weird stuff like NMI error messages if it's a Dell or HP. Is the card on it's own interrupt
23:14.52dlynespuzzled: nah...nothing weird like that
23:15.15dlynespuzzled: The biggest problem with that damned machine
23:15.25mwright1nigh1bummer root password password isn't working
23:15.28mwright1nigh1handbook isn't much good
23:15.31dlynespuzzled: is that it's the only rackmount machine we've got that has that particular type of pci slot
23:15.38*** join/#asterisk Araluccl0 (n=ciccio@adsl-ull-1-4.46-151.net24.it)
23:15.44puzzleddlynes: all I can say is shoot an email to the list but have it well documented, e.g. compile zaptel and libpri with all debug options and log the stuff
23:15.49dlynespuzzled: so i can't even take the pri card out of there and throw it into another machine
23:16.14puzzleddlynes: that doesn't make it any easier :/
23:16.39dlynespuzzled: well, if i run into that problem with libpri-trunk and zaptel-trunk, i'll try that...but like i said...i'm not running into the problem with the version of trunk i'm using now
23:17.00puzzleddlynes: ok, hope that the asterisk version you got works too
23:17.15dlynespuzzled: using asterisk 1.2.5 on that machine, I think
23:17.35dlynesasterisk 1.2.6
23:17.47puzzleddlynes: afaik you can't mix zaptel/libpri from trunk with asterisk from 1.2 branch
23:17.57dlynesI did, and it works
23:17.58dlynes:)
23:18.00puzzledmaybe that's been "fixed" but it used to be like that a while back
23:18.07puzzledlucky you :)
23:18.21filedepends how much we change things
23:18.30*** join/#asterisk mog_home (n=mogorman@68.62.237.103)
23:18.35dlyneswhat i did was compile libpri-trunk, install that
23:18.37fileor rather
23:18.41filemog_home!
23:18.46dlynesthen compiled zaptel-trunk, installed that
23:18.47mog_homefile!
23:18.57dlynesand hten compiled asterisk-1.2.6, compiled that, and installed it
23:19.07dlyneserm...fix the second compile :)
23:19.11AsteriskAlbaniadlynes: do you have any idea if TE110P works with quintum CMS960 ?
23:19.16puzzleddlynes: wrong order. you need to remove all old header/libs. than compile/install zaptel then compile/install libpri
23:19.19dlynesAsteriskAlbania: no idea
23:19.41dlynespuzzled: i compiled into a slackware binary package
23:19.49dlynespuzzled: and installed the slackware binary package
23:19.54dlynespuzzled: i.e. using upgradepkg
23:20.01dlynespuzzled: it autoremoves everything
23:20.17puzzleddlynes: sure as long as you compile/install in the right order zaptel -> libpri -> asterisk
23:20.49dlynespuzzled: oops...when i said the order i used...that wasn't the order i used
23:21.03dlynesi actually did do zaptel->libpri->asterisk
23:21.20dlynesbecause obviously libpri depends on zaptel
23:21.37puzzledyup
23:21.46mitcheloci thought libpri was first =X
23:21.47filelibpri doesn't depend on zaptel...
23:21.53dlynesusing 2.6.15.5 on there, too, with low latency optimizations
23:22.33dlynesfile: well, i wasn't sure if it did or not, but compiling zaptel first and then libpri should work regardless of whether libpri depends on zaptel, or not :)
23:22.49filedepends
23:23.08puzzledfile: so the CFLAGS+=-I../zaptel are in the libpri Makefile for fun? :)
23:23.15dlynesdidn't you say it wasn't dependant? :)
23:24.37filesome of the utility programs use zaptel
23:24.43filebut libpri itself, does not
23:24.54puzzledok
23:25.19*** join/#asterisk hinckc (n=hinckc@ool-4354c627.dyn.optonline.net)
23:25.23filechan_zap is what uses libpri
23:25.45dlynescorrect
23:26.10*** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36)
23:26.22*** join/#asterisk AsteriskAlbania (i=Asterisk@80.91.113.253)
23:29.45DoktorGregdammit!
23:29.56DoktorGregi just found the problem i was having
23:30.04DoktorGregsave the trouble
23:30.11DoktorGregif you want to do PRI
23:30.27DoktorGregDO NOT USE BRIStuff branch
23:30.40dlynesi don't
23:31.00DoktorGregI was sharing the solution to my problem
23:31.15puzzledDoktorGreg: if you had taken a look at what the patch touches you would have stayed well away from it :)
23:31.36dlynesi haven't used bristuff since 1.0.9
23:31.36DoktorGregI was using the rapid asterisk distro
23:31.54dlynesdidn't even think there was a patch for it for 1.2
23:32.22DoktorGregwas not patch, I assumed that i was using a stable build
23:32.55DoktorGregIm gonna post this on every forum i can find
23:33.02filelol
23:34.34Rawplayerdont use it then
23:34.45*** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) [NETSPLIT VICTIM]
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23:35.30key2someone knows a lil bit about capi ?
23:35.51puzzledkey2: a bit but only in relation to Eicon Diva Server cards
23:36.22key2puzzled: it's with avmc4
23:36.44puzzledkey2: no idea but there is a page on voip-info.org dedicated to making it work with asterisk/capi
23:36.55key2I can't take more than 2 lines at the time, if the 2 lines of the first port are busy, then it doesnt take the line from the second line
23:37.27*** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) [NETSPLIT VICTIM]
23:38.18key2puzzled: something looks wrong here : http://pastebin.com/664015
23:40.57puzzledkey2: looks ok. maybe try to change [AVMC4.2] to [AVMC42] so loose the dot. alternatively have a look on www.chan-capi.org
23:41.15key2puzzled: tryed that
23:42.03puzzledkey2: yeah I see there not a lot there yet. Tried the chan-capi mailing list yet?
23:43.22key2nop
23:43.34*** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) [NETSPLIT VICTIM]
23:43.35key2it's just weird
23:43.49*** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler)
23:44.08key2puzzled: here is what it says: http://pastebin.com/664025
23:44.29key2== Everyone is busy/congested at this time (1:0/1/0)]
23:45.28*** join/#asterisk IceManRISK (n=kart@200-181-208-180.mganm7001.dsl.brasiltelecom.net.br)
23:46.12puzzledkey2: if you already have 2 incoming calls on that port than that does not surprise me
23:46.46*** join/#asterisk MrCraig (n=Craig@bb-87-82-12-210.ukonline.co.uk)
23:46.49MrCraighi
23:46.53*** join/#asterisk IceManRISK (n=kart@200-181-208-180.mganm7001.dsl.brasiltelecom.net.br)
23:46.59dlyneshihi
23:47.00key2puzzle: AVCM4 is 2 lines and AVCM42 is two other line
23:47.03key2in the same group
23:47.25key2puzzled: so basically, If I even use one line of AVCM4 and one of AVCM42, i can't dial from any of those anymore
23:47.44MrCraigI wants to be able to make or recieve calls using my fax modem - I've been told to try asterisk, so my first question is do I have the right software?
23:47.48puzzledkey2: sorry, no idea. I would try the chan-capi mailing list
23:49.27CukXstupid starting question... i'm getting messages, registration from "11" <sip:11@ASTERISK_IP> failed for PHONE_IP
23:49.49CukXwhat to enter ? into extensions.conf something, right ?
23:50.18puzzledsip.conf
23:52.58Lino`hmmm
23:53.35Lino`key2: german isdn or what?
23:55.28Lino`ok
23:55.31Lino`most probably french ISDN

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