IRC log for #asterisk on 20080820

00:00.10Duddshttp://pastebin.com/m6061f93  <-- That's the current config stuff. (There's an IAX2 trunk from the remote where the TE122P is connected to the T1 from telco).
00:00.25DuddsManxPower - Everything is in Clear status except 24   24 WCT1/0/24 HDLCFCS
00:00.32THX2000rasterix: depending on how much customization you want to do, freepbx/trixbox/elastix provides some pretty enhanced functionality and allows you to fill in the blanks if theres something missing
00:00.37ManxPowergood!  no alarms
00:00.56DuddsYeah, and no action!  :D
00:01.28ManxPower<PROTECTED>
00:01.34DuddsI hit the Line button on a PBX phone, and it goes "Line001 - Line in Use"... Asterisk reports that there is congestion when trying to dial Zap/g1 too.
00:01.38ManxPowerThere is nothing after g1  You need a dialed number
00:01.42rasterixthx200: i think that is true... but for some direct editing of extensions.conf provides the flexibility they require
00:01.47mcab/act
00:02.07ManxPowerlike Dial(Zap/g1/5551212)
00:02.34DuddsChanged it, same result:  Executing [221@bond-remotesite:1] Dial("IAX2/iax-remotesite-12215", "Zap/g1/8452") in new stack
00:02.48Dudds[Aug 19 17:02:20] WARNING[28406]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
00:02.52ManxPowerDudds: BTW, there is no such thing as "TE" in a PRI.  That's a BRI thing.
00:03.07DuddsAhh, I'm referring to the model of the card.
00:03.54rasterixjust bought a sangoma pri card with echo cancellation << in love so far
00:03.55ManxPowerWhat is the other end of the line?
00:04.27THX2000rasterix: True, and if you have some very specific requirements thats a good way to go.
00:04.28ManxPowerI assume it's your PBX, but I want to confirm
00:04.32DuddsThe PBX (Norstar MICS).
00:04.41ManxPowerDudds: It sucks to be you.
00:04.55ManxPowerDudds: As far as I can tell Asterisk is set up correctly
00:05.03*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
00:05.31ManxPowerYou can do a "pri debug span 1" but I suspect all that will confirm is that the PBX is rejecting the call.
00:06.01DuddsWoah, that gives me "Sending Set Asynchronous Balanced Mode Extended" every second.
00:06.08rasterixi dont have enough (almost none) experience of the graphical interfaces... if you load a complex extensions.conf is there any danger it will mess with your file?
00:06.29ManxPowerDudds: Turn off SAMBE on the PBX
00:07.02DuddsI'm not familiar with that term...
00:08.33THX2000rasterix: not really sure I understand your question
00:08.49ManxPowerSAMBE = Set Asynchronous Balanced Mode Extended
00:09.37ManxPowerDudds: http://www.google.com/search?hl=en&q=site%3Alists.digium.com+Set+Asynchronous+Balanced+Mode+Extended&btnG=Search
00:10.44rasterixthx2000: probably didnt make sense its late and im tired...  i have a couple of fairly complex extensions.conf... if i try to edit these files with a graphical interface that did not create them is there a danger the file will be "messed up"
00:11.15Qwellrasterix: there is about a 100% chance that the GUI will hose your config.
00:11.33rasterixlol thank you qwell
00:11.50Qwellunless you do things the way the GUI expects them, it can't possibly handle it
00:12.28THX2000rasterix: Exactly what Qwell said :P
00:12.29rasterixok i just wondered if there were any gui's smart enought to section off code they dont understand and leave it untouched
00:12.44Qwellno, because for them to work, they have to change the way your config is setup
00:13.09ManxPowerDudds: I could be wrong about SAMBE.
00:13.17DuddsHmm, well, I'm looking through the docs.
00:13.19ManxPowerIT might just be indicating the far end is not responding?
00:13.31DuddsI duno if you've ever messed with a Norstar system, it's a PITA.
00:13.37THX2000rasterix: The GUI's pretty much take over your whole system.  You'd have to implement the gui, then figure out how to add in your own customizations
00:13.38ManxPowerDudds: Are you SURE it's a PRI and not just a plain T-1?
00:14.05DuddsWell, the Norstar says it's in T1 mode and has nothing about PRI in it, so I'm not sure if it's mislabelled or I've just missed that.
00:14.15DuddsIf it was in T1 mode, how would that change the config?
00:14.21ManxPowerDudds: If so, well you sure have wasted a lot of my time.
00:15.13ManxPowerYou would change all of your channel => stuff to fxoks or fxsks, depending on how the PBX is set up.
00:15.29rasterixit depresses me that i didnt find this channel 6 months ago... im sure i could have avoided wasting so many days of my life :(
00:15.46Duddsrasterix - That's kind of how I feel.  :)
00:16.47russellbThis channel is great if you've got thick skin :)
00:17.03ManxPowerDudds: next time make sure what you are saying is reality.  A simple mistake in a pbx config could allow anyone to route calls thru your PBX and have you pay for it.  You MUST be very careful.
00:17.05rasterixrussellb: or no pride
00:17.14*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
00:18.03rasterixi dont mind if people call me an idiot on here... because compared to the majority of this room in terms of asterisk i AM an idiot
00:18.19russellbI mind.
00:18.21russellbheh
00:18.36russellbj/k ..
00:19.16ManxPowerHaving your facts correct, asking good and complete questions, not going AFK in the middle of someone trying to help you, all these things help
00:19.37Qwella little common sense helps too
00:20.26rasterixa very general question and if its not the right place to ask tell me... how do you (whoever can be bothered to comment) see asterisk competing with Avaya etc over the next few years?
00:20.49ManxPoweror....
00:20.51ManxPower~ask
00:20.51jbotfrom memory, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
00:20.56ManxPowerthere we go.
00:22.05grandpapadotAsterisk = more flexibility, requires more techical knowlege, open source, open standards, works with a lot of equipment -- Avaya is propriatary and expensive but has the benefit being around for a while
00:23.39DuddsOk, well looking at this line config, it seems the card, on the PBX, is in T1 mode and the channels are in E&M with "Wink Start" on.
00:23.56ManxPowerDudds: then that is what you must configure on the asterisk side.
00:24.13DuddsI've switched the config to match, or so I think, and now it's giving "chan_zap.c:7221 mkintf: Signalling requested on channel 1 is E & M Wink but line is in ISDN PRI signalling"... I restarted the card (rmmod/insmod the modules).
00:24.15ManxPowerBut someone else will have to help you with that.
00:24.32ManxPowerDudds: You do not have a PRI.
00:25.13DuddsI know, I've switched it at this point to "signalling   = em_w".
00:26.11Qapfhey guys, ive been doing my research all day and it seems the general opinion of faxing and voip is don't do it, it doesn't work. does anyone know of a company that offers a fax-email gateway then? i dont have any analog lines in my asterisk box and i kinna wanna keep it that way
00:26.17*** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun)
00:26.38Qapfand not efax, their product is horrible
00:26.41l2trace99rightfax
00:27.04l2trace99it was about $10 a month
00:27.06l2trace99US
00:27.19Qapfl2trace99: thanks, ill take a look
00:27.36grandpapadotOh, sorry, lol
00:28.02rasterixi have forwarded a ddi from isdn30 to an analogue fax number and it worked fine... really quick and nasty fix but worked
00:28.19DuddsManxPower - Thanks for you time.  :)
00:28.20QwellDudds: switch it in zaptel.conf also
00:29.02DuddsQwell - I've set the signalling to em_w but switchtype is still national.
00:29.11DuddsI'm not sure how to get it to T1 from PRI.
00:29.15Qwellin zaptel.conf and zapata.conf?
00:29.23drwelbyManxpower: BLF is sort of working now with limitonpeers=yes in sip.conf
00:29.41QwellI believe e&m is clearchannel..  I have no idea though, I've only ever configured a T1 once
00:29.58l2trace99Qapf: sorry I remembered the name wrong it is rapidfax
00:30.06DuddsYeah, that's definitely what the PBX has for the Lines. (This thing is a pain to configure... MICS system).
00:30.18Qapfl2trace99: i was going to say, rightfax looks like something else
00:31.14Qapfl2trace99: thanks for the pointer, rapidfax looks like what im after
00:31.18Qapfjust need to make a pdf workflow
00:31.46Qwellrapidfax is j2
00:31.53QwellAVOID
00:31.55russellbQwell: E&M is a RB sig type
00:32.05Qwellrussellb: see, I don't even know what RB is :(
00:32.09QapfQwell: seriously? j2 is efax
00:32.09QwellI'm a T1 newb
00:32.09*** join/#asterisk giovani (n=giovani@unaffiliated/giovani)
00:32.10Qapfjesus
00:32.12russellbheh
00:32.18QwellQapf: *fax.com is j2
00:32.18russellbQwell: robbed bit ... analog signaling
00:32.19l2trace99i used it for a year
00:32.20Qwellbasically..
00:32.30l2trace99no problems with it
00:32.32giovaniI'm trying to set up FollowMe with the option to confirm the calls by the external extension it forwards to
00:32.35QapfQwell: is there anyone who doesn't suck?
00:32.45QwellQapf: anybody that j2 hasn't bought, I imagine
00:32.47l2trace99my wife
00:32.54l2trace99oh sorry
00:33.01Qwelll2trace99: really?  huh, that's n..nevermind
00:33.04giovanihowever, it doesn't seem to be recognizing the DTMF tone when I press "1" -- I've used that same trunk for inbound calling in IVR and DISA, so I assume DTMF recognition is working
00:33.05QapfQwell: good luck to me on finding that i guess
00:33.10QwellQapf: basically
00:33.15rasterixif there was an editor with intellisense etc for extensions.conf would anyone be interested in using it?
00:33.26Qwellrasterix: like...vi?
00:33.39rasterixno not like vim
00:33.42grandpapadotvim rocks
00:33.45rasterixyes
00:33.47rasterixi like vim
00:33.55rasterixbut it does not have intellisense
00:33.57Qwellwell, vim does that
00:34.02QapfQwell: this is insane, do you know anyone who isnt just j2 rebranded?
00:34.06rasterixit does?
00:34.07*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-57eacd9b6ee2d3cd)
00:34.07QwellQapf: heh, no
00:34.12Qwellrasterix: it can, sure
00:34.27Qwellrasterix: maybe not exactly the same, but.
00:34.36rasterixit tells you the parameters when you type in a function?  < must have his vim set up wrong?
00:34.57Qwelldunno, I use nano
00:35.04ManxAwayrasterix: sounds like you are more looking for an IDE for extensions.comf
00:35.08russellbnano is for noooobs
00:35.30russellb:-p
00:35.30QwellI'm too cool for vim
00:35.30russellbsuck it up and learn vim, nub
00:35.30ManxAwayI use "joe"
00:35.30QwellManxAway: I'll bet
00:35.30voxteri used joe for yearssss
00:35.31voxterthen i finally sucked it up and learned vim
00:35.38rasterixlol now im confused
00:35.48Qwellrussellb: I don't need no "syntax highlighting"
00:35.52ManxAwayI know enough vi to do simple config file edits and to edit the Makefile for joe.
00:35.54Qwellor "features"
00:36.00rasterixwas the upshot of that that vim or nano do have "intellisense" of sorts?
00:36.43Qwellmy editor?  started life as a mail composer.  yeah.  what now?
00:37.03rasterixlol < gives up
00:37.24QwellYou'll never see a "gnano" like you will with vim or emacs.
00:37.29grandpapadotYou can pry vim from my cold, dead, carpel-tunnel ridden hands ...
00:37.37l2trace99ctl-p  in vim
00:37.38l2trace99http://www.vim.org/tips/tip.php?tip_id=1257
00:37.52Qwellrasterix: ^^
00:37.52l2trace99never looked for it until now
00:37.58*** join/#asterisk iewebguy (n=mark@65.19.81.253)
00:37.59Qwellit's a pretty neat feature
00:38.24grandpapadotMy first text editor was stdin
00:38.31Qwellgrandpapadot: so, ed?
00:38.36voxteredlin.exe
00:38.37grandpapadotlol, ya
00:38.37voxter:P
00:38.46rasterixthat could be better than i have
00:38.54rasterixits still not intellisense tho
00:39.03rasterixi have a bad memory
00:39.12rasterixi need to be reminded of each parameter
00:39.37Qwellrussellb: you know...
00:39.50russellbi know nothing
00:39.53Qwellrussellb: once this xmldoc stuff is done, and there is syntax defined...
00:40.26Qwellhmm, I'll hold that idea for a while
00:40.36rasterix< intrigued
00:40.49rasterixtell me more qwell
00:41.08grandpapadotout, peace
00:41.13iewebguyHello everyone,  I am upgrading my asterisk to 1.4 and I can't seem to get my t-1/pri to come up fully --- does anyone have time to give me a little help?
00:41.26Qwelliewebguy: upgrading to what?
00:41.40*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id)
00:41.41iewebguyasterisk 1.4 from 1.2
00:41.47rasterixwhat xmldoc stuff are you talking about qwell?
00:42.00rasterixa syntax definition for asterisk?
00:42.18Qwellthat's part of the eventual goal
00:42.26Qwellfor applications, anyways
00:42.31rasterix< wants to read more...
00:42.39rasterixwhere please?  :)
00:42.46iewebguymy D-channle is going up and down
00:42.57Qwellrasterix: ask mvanbaak where the branch is
00:43.01iewebguy<PROTECTED>
00:43.05iewebguy<PROTECTED>
00:43.06russellbrasterix: nothing to read, other than the code in the branch right now ...
00:43.07iewebguyetc.
00:43.38russellbiewebguy: sounds like you have your timing settings off ...
00:43.39rasterixiewebguy what card do you use?
00:43.41russellbin /etc/zaptel.conf
00:43.55iewebguya sangoma 104D
00:44.04rasterixi think i know this
00:44.09russellbbah, nevermind :-p
00:44.13iewebguyI pulled the settings from the working system
00:44.32rasterixin your zaptel.conf change the dchan= to say h****=
00:44.37rasterixhold up
00:44.56jayteeanyone here ever use Valcom or Viking equipment for Paging/PA with Asterisk?
00:45.02iewebguyone question is the zap/G1 is that 1 the same as Group=1 or callgroup=1
00:45.11iewebguyin zapata.conf
00:45.25jayteeiewebguy, it's Group
00:45.31rasterixiewebguy what does your zaptel.conf specify dchan as?
00:45.48rasterixif you are using the lastest sangoma drivers its no longer called dchan
00:45.54Qwell...
00:46.02Qwellwhy the crap would they change that?
00:46.06iewebguyI pulled the drivers a few days back
00:46.08rasterixi dont know
00:46.17rasterixi was scratching my head for hours
00:46.29iewebguyI ran the sangoman script and it gives me:
00:46.33rasterixthen i found... change dchan to h*****
00:46.40iewebguyloadzone=us
00:46.40iewebguydefaultzone=us
00:46.40iewebguy#Sangoma A101 port 1 [slot:10 bus:0 span:1] <wanpipe1>
00:46.40iewebguyspan=1,0,0,esf,b8zs
00:46.40iewebguybchan=1-23
00:46.40iewebguydchan=24
00:46.51rasterixi know iewebguy... its nice their generator calls it dchan
00:46.55rasterixbut its wrong!
00:47.34iewebguyso like this:
00:47.35iewebguyspan=1,0,0,esf,b8zs
00:47.35iewebguybchan=1-23
00:47.35iewebguyh*****=24
00:47.45voxteruhhh
00:47.46rasterixlol no wait
00:47.49jayteethat is just so wrong
00:47.51voxterthat is not correct.
00:47.56rasterixi have to find what the *s are supposed to be
00:47.59Qwell~paste
00:48.00jboti guess paste is http://rafb.net/paste/, or see also pb
00:48.01iewebguyoh
00:48.05Qwell~pb
00:48.05jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
00:48.13*** join/#asterisk bad_duck (n=bad_duck@dynamic44-3.MAN-STDOM.cablenet.com.ni)
00:48.14iewebguythanks
00:52.42rasterixsorry iewebguy
00:52.48rasterixi might have to logout
00:52.58rasterixand go into the server
00:53.13rasterixi cant find it on sangoma's site so far...
00:53.21iewebguyok
00:54.31Qwellhardhdlc...  lame
00:55.13*** join/#asterisk codefreeze-lap (n=murf@72.21.67.40)
00:55.50jayteemakes me glad I bought Digium instead of Sangoma
00:56.23rasterixyou found it qwell?
00:57.07rasterixits ridiculous that their script generate an out of date zaptel.conf
00:57.08rasterix><
00:57.52iewebguysomething like: hardhdlc=24 ?
00:57.54rasterixso anyway basicall iewebguy if you swap dchan for hardhdlc in zaptel.conf it should be ok
00:59.02rasterix*crosses fingers* this could be the first time i actually was any help to anyone in this channel
00:59.39iewebguytesting...
00:59.46jaytee1 2 3
01:02.06anonymouz666Qwell: Don't know if you remember, but the static noise on the far end with the TE110P just stopped and I switch the card to another machine. From a QUADCORE to DUALCORE
01:02.09anonymouz666that's...bizarre.
01:02.10iewebguyseems to not work,
01:02.16*** join/#asterisk samy^ (n=samy@72.37.252.206)
01:02.27anonymouz666s/and/when
01:02.53rasterixdid you use the latest sangoma driver ieweb?
01:03.09Qwellsamy^: .
01:03.14iewebguyI think so. I built this box a few days ago.
01:03.20rasterixyah
01:03.30rasterixwell i had exactly that problem
01:03.32samy^hey qwell, how are you?
01:03.32Qwellsamy^: welcome back to the Internet >.<
01:03.43rasterixdchan up dchan down dchan up dchan down
01:03.50anonymouz666and the dual core machine is sharing IRQ!!!!!!!
01:03.56anonymouz666wtf this is totally illogical
01:04.06rasterixand it was because zaptel.conf had dchan =
01:04.27samy^qwell, thanks :)
01:04.37Qwell~samy
01:04.52rasterixieweb: did you restart the machine
01:04.54iewebguywhich logs should I look at ? pri did not even come up
01:04.54Qwelljbot: samy is <reply> Samy is my hero.
01:04.55jbotQwell: okay
01:04.56iewebguyyup
01:05.22rasterixugh i wish i could get on my work pc
01:05.40iewebguyfirewall ?
01:05.54rasterixnah gotta go through vpn and it kicks me off here
01:06.58iewebguyoh.
01:08.20rasterixi cant remember
01:08.23rasterixbut its in /var/log
01:10.17iewebguyI am gonna put it back and see what else might be logged. it was closer b4
01:10.29rasterixiewebguy: you compiled zaptel and libpri before asterisk
01:11.34iewebguyyup
01:12.03rasterixwhat version of the sangoma driver are you compiling?
01:12.27jayteelebenty-leven
01:13.27iewebguywanpipe-3.2.7.tgz
01:13.53iewebguyasterisk-1.4.21.2 libpri-1.4.5  zaptel-1.4.11
01:14.40rasterixhumour me
01:14.49rasterixtry a make clean
01:15.08rasterix./configure
01:15.15rasterixmake
01:15.24rasterixmake install on zaptel
01:15.27iewebguyI'll probably have to do that tomorrow. its about time for me to head home.
01:15.29*** join/#asterisk washburnello (n=johnsy@h36.225.40.162.dynamic.ip.windstream.net)
01:16.15iewebguy8 am - 9pm I am such a sissy.
01:16.31*** part/#asterisk washburnello (n=johnsy@h36.225.40.162.dynamic.ip.windstream.net)
01:17.14rasterix2 minutes to recompile :)
01:18.34jayteefast
01:18.37iewebguyok
01:19.05puzzledmvanbaak: ping
01:19.25jayteeso I'll try again
01:19.48jayteeHas anyone here ever hooked a PA system up to Asterisk? if so what'd you use for equipment?
01:20.41ManxAwayjaytee: yes
01:21.17ManxAwayI set up chan_console, the onsite guy bought some cheap radio shack amp and some speakers.  Plugged it into line out on the server
01:21.18iewebguycompile running... I am getting these now too:  "chan_zap.c: No D-channels available!  Using Primary channel 24 as D-channel anyway!"
01:21.41ManxAwayiewebguy: that means "line not working"
01:21.57rasterixiewebguy: http://sangoma.editme.com/Asterisk-FAQ#hardhdlc%20vs%20dchan
01:22.07iewebguytell me more,  my t1 has a green light.
01:22.30rasterixiewebguy: enger zztcfg -vvv
01:22.31jayteeManxAway, sounds like an inexpensive solution. I'm looking for something I can distribute thoughout a campus environment.
01:22.38rasterixenter*
01:23.12rasterixgeeze... ztcfg -vvv
01:23.13M-I-AManxAway you wouldn't happen to have a link for the cheap ratshack amp would you?
01:23.14jayteecurrently looking over docs for a Valcom and Viking products which are analog but I could use an ATA for each location.
01:23.14rasterixsorry
01:23.53ManxAwayM-I-A: I have never even SEEN the box.
01:24.13M-I-Aoh ok
01:24.15*** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org)
01:24.57iewebguyhttp://rafb.net/p/9sNdWJ26.html
01:25.40iewebguyI am rebooting with the zaptel recompile
01:26.20iewebguyv
01:26.21iewebguy~centos52bug
01:26.22jbot[~centos52bug] There is a bug compiling Zaptel  up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver  in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889
01:26.43ManxAwayM-I-A: I'm not onsite very often
01:26.45rasterixahhh
01:27.05rasterixcd /usr/src/zaptel*******
01:27.11rasterixmake clean
01:27.20rasterix./configure
01:27.26iewebguyoh I fixed that days ago
01:27.34rasterixmake menuselect
01:27.43iewebguyI just saw it in the headline
01:27.46M-I-AManx, the amps that I keep finding are always very expensive and dont seem to be what I'm looking for to run three speakers
01:27.47rasterixdisable everything you dont need
01:27.59rasterixmake
01:28.04rasterixmake install
01:28.13iewebguydisable from a file?
01:28.52ManxAwayM-I-A: where are you located?
01:28.56rasterixno in the menuselect utiltity
01:29.03rasterixdisable everything you dont need
01:29.06iewebguyI see
01:29.18M-I-Atoronto
01:30.48*** join/#asterisk mltlnx (n=mltlnx@65.90.180.42)
01:31.05ManxAwayM-I-A: There are like 10 Radio Shacks in Toronto
01:32.07brodiemM-I-A: what are you looking for in an amp
01:32.18M-I-AManx, Well they are called 'the source' now but, I'm either looking for the wrong thing or the guys working there have no clue
01:32.30M-I-Apowere
01:32.31M-I-Ad
01:32.33ManxAwayM-I-A: Or even Wal-Mart, they have devices for less than US$100 that will take RCA stereo in and send it out speakers with amp.
01:32.47brodiemM-I-A: you won't get your money's worth at RS on an amp that's for sure
01:32.48ManxAwayStandard powered computer speakers will do it.
01:33.15rasterix< didnt realise asterisk needed a power amp to work properly :)
01:33.27ManxAwaybrodiem: I believe the one the customer bought was under $100.
01:33.43brodiemManxAway: bah I joined too late I don't know what's going on, heh
01:33.48ManxAwayrasterix: if you want overhead paging then yes you need an amp or an FXS port
01:35.25M-I-AManx: I have three speakers in the ceiling in different parts of the office. A single pair of wires run from each back to the phone room. I just need something to power them for overhead paging
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01:35.33iewebguyInstall ncurses to use the menu interface  -- but I have ncurses installed with yum install ncurses
01:35.56rasterixyum -y install ncurses-devel?
01:36.07iewebguy3 cell phones with little wires soldered to the internal speakers ;)
01:36.21M-I-AManx, the one I found was like 500 bucks
01:36.29brodiemM-I-A: what impedence are the speakers
01:36.33M-I-Aiewebguy: thx ;)
01:36.34ManxAwayM-I-A: Go find an electronics geek and ask him/her.
01:36.39ManxAwayLOCAL one.
01:36.49ManxAwayThe ham radio club would be a start.
01:37.00ManxAwaygoodwill, pawnshop
01:37.02M-I-ABrodiem i'm thinking 16ohm
01:37.11iewebguyinstalled already
01:37.14iewebguyvery odd
01:37.42brodiemM-I-A: standard is 8oh btw but you'll need to be sure before selecting an amp if you want to run them on a single channel to keep the cost of the amp down
01:38.32ManxAwaysee, ask for an electronics geek and one suddenly appears.
01:38.36brodiemhehe
01:38.38M-I-Alol
01:38.58brodiemI have 6 systems currently between home and cars
01:39.10iewebguyrasterix, I am gonna put the old server back and try again another day.  thanks for your help.
01:39.20rasterixok
01:39.27rasterixdid you do ./configure
01:39.49iewebguyya but I erased it when I was trying for the menu.
01:40.09iewebguyI'll find the menu problem tomorrow.
01:40.27M-I-Abrodiem: I can always rewire for a combo of series/parallel in order to get the right impedence right?
01:41.07rasterixwell i think when you recompile tomorrow with a clear head
01:41.13rasterixit will all just work :)
01:41.17iewebguycooll
01:41.18iewebguy:)
01:41.49russellbiewebguy: make distclean
01:41.51russellbafter you install ncurses-devel ...
01:41.55brodiemM-I-A: yeah but if they're indeed 16oh then I would just run them all in parallel and just make sure to get an amp that's 4 (or even 6 but you probably won't see any with that spec) ohm stable bridged
01:41.57rasterixmy experience... after about 930pm things just stop working
01:41.57russellband then configure, make menuselect... it shoulud work then
01:42.16iewebguyk
01:42.19rasterixand that is why russellb is blue
01:42.26rasterixand the rest of us are black
01:42.29russellblol
01:42.34Qwellwonders why russellb is blue
01:42.43russellbchecks his breathing
01:42.45russellbyep, still working
01:42.47M-I-Abrodiem: ok thanks for your help
01:43.09M-I-AManx: Thanks also
01:43.29iewebguyrussellib: you were right
01:43.44russellbw00t
01:44.01iewebguybut  I don't know which f these 18 choices I need.
01:44.08russellbfor what
01:44.19Qwellto fix his sangoma or something
01:44.27russellb... oh
01:44.30Qwellshould've bought Digium, they would've helped you with the config if it wasn't working..
01:44.34Qwellrussellb: ...same guy
01:44.34russellbimmediately regrets helping
01:44.34russellbha
01:44.38russellboh, darnit
01:44.38iewebguyI have a sangoma 104D card and a PRI/t1 from the telco
01:44.59rasterixsangoma tech support has always been very helpful to me :S
01:45.14russellbuntil it involves asterisk, huh?
01:45.17iewebguyyup but they close when I can take the phone system down.
01:45.18Qwellexactly
01:45.39ManxAwayQwell/ russellb: I love the Digium Zaptel build process, you just make install and you are done.    Samgoma's is very complex in comparison
01:45.46rasterixnah my chap doesnt mind helpin out with asterisk
01:46.04QwellManxAway: well, should start buying Digium again
01:46.58ManxAwayQwell: In any situation where I can use Digium at a new customer, I'll try it out.
01:46.58rasterixread them out iewebguy
01:47.14QwellManxAway: that's fair
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01:47.43ManxAwayQwell: I get new customers very seldom. 8-|
01:48.15iewebguy<PROTECTED>
01:48.20russellbblinks
01:48.25M-I-AIs it possible to have a line appear (flash) but not ring on a IP endpoint (Only on certain incoming calls)... If so what is this called in the * world?
01:48.47ManxAwayQwell: You don't see many HDLC Abort issues anymore, but when you do it seems like it's almost always an older card.
01:48.55QwellManxAway: yeah..
01:49.25ManxAwayI'm still in therapy over the Digium Analog problems we had.
01:49.29iewebguyI figure I just need some of the wct ones and maybe a zt something
01:49.42QwellManxAway: TDM410 is an amazing upgrade to the 400
01:49.48ManxAwaySo much so that I won't even recommend Sangoma analog.
01:50.07rasterixim not qualified to ask answer this... but i think it tells you what cards each relates to in menuselect?
01:50.31iewebguyoh ya at the bottom.
01:50.34russellbsuggests "sudo rm -rf /" to fix it
01:51.09ManxAwaymutters "Never, even in joking say that"
01:51.19russellbheh, sorry
01:51.25russellbretracts his suggestion
01:51.39JerJerrm -rf /boot is better
01:51.45iewebguyfunny, I don't need any of them
01:52.04JerJersince once rm gets to proc and/or dev it will fail  :(
01:52.37russellbheh ... ok, so, rm -rf /boot /
01:52.43russellbget boot first, then go for the kill
01:52.56brodiemJerJer: but easily fixable, how about cat /dev/random > /dev/sda*
01:53.13russellbnice .. /dev/urandom though!
01:53.22JerJerhmm - can one write to it that way if its mounted?
01:53.26Qwellnewbs, all of you
01:53.29QwellJerJer: yes
01:53.32brodiemyep
01:53.33rasterixas i recall they all relate to cards nothing to do with sangoma so u probably dont need any of them... > qwell / russellb
01:53.34russellbdd if=/dev/urandom of=/dev/sda
01:53.38JerJeryeah - dd better
01:53.38Qwellrussellb: exactly
01:53.47JerJercount=500G
01:54.00iewebguy<PROTECTED>
01:54.04brodiembut won't that only overwrite the boot sector
01:54.08brodiemleaving your fs in tact
01:54.22JerJersda3
01:54.22Qwellno
01:54.28russellbno, that will get the whole drive
01:54.37brodiemok I believe you, hehe
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01:55.04rasterixdoes blue mean you are on the payroll of digium?  < genuine question
01:55.16Qwellblue?
01:55.22M-I-Alooks sideways at his 2401E collecting dust
01:55.28russellbmost of the ops are digium people, but not all of them
01:55.40rasterixi see
01:56.34JerJeri am rather fond of:  :(){ :|:& };:
01:56.46QwellJerJer: now explain each char.
01:57.15Qwell(I asked somebody to do that once, and they did...and it sucked for me)
01:57.19JerJerok :() is a function with no arguments
01:57.27rasterixwhat makes me laugh about sangoma is their own compiler creates a zaptel config file that doesnt work... and then they get support calls... presumable one for every fresh install right now...
01:57.55QwellJerJer: will it work in POSIX sh?
01:58.09JerJerhmm - i think its only bash
01:58.13rasterixanyway im out too... its late here in the uk
01:58.17rasterixbye all
01:58.28QwellI wonder what would happen if you ran that in a VM
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01:58.37Qwellsomebody care to try it, and report back?
01:58.41russellbbourne shell doesn't support functions, does it?
01:58.43unpaidbilli just ran it on my asterisk server
01:58.47unpaidbillwhat the hell is happening
01:58.49russellblol
01:58.51brodiemomg
01:58.53JerJerfork bomb
01:58.54Qwellrussellb: it does
01:58.55unpaidbillhaha i kid.
01:59.03Qwellunpaidbill: You wouldn't be able to type.
01:59.08QwellIt's instant resource saturation
01:59.11unpaidbillyeah
01:59.15unpaidbillbut i dont irc from my asterisk box :P
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01:59.20JerJerQwell:  I ran it on xen based guest os' before
01:59.26QwellJerJer: how'd it end up?
01:59.34Qwelldid the host crap itself?
01:59.36JerJernot friendly - unless the config limits processor limitation
01:59.53JerJerbut even then its still very unfrlendly to the whole system
02:00.26JerJera VPS provider i used a while ago decided not to respond to tech support request for 3 days, so i fork bombed it -  oddly enough they responded after about an hour
02:00.45unpaidbillhaha
02:00.54unpaidbillresponded with a termination notice
02:01.00JerJerthey had no clue
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02:01.19JerJerthey just rebooted the system
02:01.35QwellJerJer: my favorite is a bzbomb
02:01.36brodiemI'm kinda surprised it would have had any effect with file descriptor limits inside the VE
02:01.53QwellI haven't seen one in forever though
02:02.05brodiemalthough probably still eats cpu cycles
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02:02.31JerJerit will still bring down a wide open xen config (no limits)
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02:02.56JerJerbut one can limit the effect with a few simple processor / thread controls on the guest config
02:03.08brodiemyeah
02:03.15JerJerstill burns up CPU badly - but not enough to stop the admin from doing something about it
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02:04.25JerJeri do a asston of virtualization -  kinda a pain to remember which host which guest is on   :(
02:04.25brodiemyeah you can limit open files and number of procs easy, but it's still going to go crazy cycling as it tries :)
02:04.33JerJeryuuuup
02:11.20brodiemnice, scored a wip330 from ebay for $50.. going to give one another whirl and see if its any better than the original unit I had (orig firmware sucked)
02:11.35Qwellbrodiem: 330 is Windows
02:11.41brodiemugh yea I know
02:11.55brodiembut it was $50
02:11.56brodiemhehe
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02:51.43heedlywhy would music on hold not play during a Dial()?
02:51.48heedlysometimes it does, sometimes not.
02:52.59JerJerhow you doing MOH?
02:53.02JerJernative or mpg123
02:54.12M-I-AJerJer: Is it possible to have a line appear (flash) but not ring on a IP endpoint (Only on certain incoming calls)
02:54.29JerJersegfaults
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03:01.06heedlyJerJer: native.
03:01.19heedlyI guess formats don't match up?
03:01.28JerJerwhat ver of asterisk?
03:01.57heedly1.4.21.2
03:02.43JerJernot sure - my customers haven't complained about anything like that -  and they love to complain
03:04.16JerJerdo you have codec_g729ab.so  (or whatever) ?
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03:14.29heedlyJerJer: yep
03:15.41brodiemM-I-A: what do you mean by "line appear (flash)"
03:17.26JerJerheedly:   not sure without diving in
03:18.48M-I-Abrodiem: Well in the Nortel world you could set a "Line Appearance" to 'Appr & ring' or 'Appr' or just 'Ring'... looking for something similar
03:20.29brodiemM-I-A: probably the closest thing you could do is use func_devstate paired with a busy lamp on your phone
03:21.07brodiemalthough you won't get an actual "ringing" state, either inuse or not inuse
03:22.38M-I-ASo you wouldn't be able to push the button beside the flashing light to pickup the call?
03:22.54brodiemanything else probably more on the device side though, where for example some phones you can tag on a SIP header that could set a ringtone, which could be a silent ring maybe
03:23.30brodiemM-I-A: I'd say you could rig it up - it'll call the BLF extension, so if it's attached to a pickup group it could work I suppose
03:24.49brodiema little messy though trying to emulate that at the PBX level - doing a silent ring tone may be easiest and most effective if your phone(s) will allow it
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03:24.57M-I-Aok I will investigate more into this... Do you know which phones support a tag on the sip header for ringtone?
03:25.15brodiemhmm sorry I can't remember =/
03:25.38M-I-Ano prob. I'll search it out
03:25.40[TK]D-FenderM-I-A: Polycom, Linksys, Aastra, Cisco, and I'm sure others
03:25.41brodiemmaybe try searching for "alert-info"
03:27.10M-I-Aok thanks guys...
03:28.03brodiem..out
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03:43.59[TK]D-Fender*b00m*
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03:47.43M-I-Anice split
03:49.44JerJermakes me remember the good ole efnet days
03:50.10M-I-ALOL true
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04:01.58obnauticusEfNet never had`good ole' days`
04:02.06obnauticusAlways had skiddies
04:02.07obnauticuslol
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04:17.52drmessanoHmmm
04:19.49jayteethings that make you go Hmmmm
04:23.21drmessanoI got all my linksys provisioning working...
04:23.34drmessanoBut my phones keep resyncing
04:23.39drmessanoMaybe that isn't the work
04:23.56drmessanoThey almost reset, but without the full power cycle
04:24.05drmessanoSurely they're not supposed to do that every x
04:24.12heedlyJerJer: fwiw, stopping and restarting asterisk fixed the problem.
04:30.00SwKanyone want an asterisk dev kit?
04:30.07SwKi have 3 extra Dell 850 1U servers
04:30.35SwK3ghz Dual Core 64Bit CPU (intel) 2G Ram and a tdm400
04:31.17drmessanoOk, you twisted my arm... I'll give you $20USD for all 3
04:31.24SwKhah
04:31.49SwKactually 1U servers, with rails... $650... the tdm cards are unused
04:32.42drmessanoThat's not a bad deal.. If I hadn't accidentally bought $7500 worth of PAP2s, I would be all over that
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04:33.30heedlySwK: were are you located?
04:33.43SwKHuntsville
04:34.00heedlyNC?
04:34.12SwKAL
04:34.43drmessanoNC is always what I think of first when someone says "Huntsville"
04:35.08drmessanoHuntsville, AL is the VoIP epicenter of the planet
04:35.44drmessanoIn Huntsville, SIP does not echo
04:35.57DigitalIronyLOL
04:38.07[TK]D-Fendersicks Trogdor on drmessano for great burnination.
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04:39.37drmessanohopes [TK]D-Fender is fangoriously devoured by a gelatinous monster
04:41.09[TK]D-Fenderfears the Cosby's minions....
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04:50.55SwKhah
04:51.13SwKwhy NC when someone says huntsville
04:52.57SwKi wouldnt call huntsville a voip epicenter... i would call it a blowing shit up epicenter...
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04:59.32M-I-AI thought I was the only one with SIP echo issues ;)
05:04.53drmessanolol
05:04.56drmessanoSIP doesn't echo
05:05.01drmessano_._
05:05.03drmessanoPeriod
05:05.08Qwellyou must be new here
05:05.26drmessanoNo
05:05.32drmessanoif I new
05:05.33drmessanoerr
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05:05.37drmessanoIf I WAS new
05:05.57drmessanoHey guys! :) <_> ^_^
05:07.06drmessanoI just installed astrixbox 2.6 and I am having problems with my grendstram phones connecting to my trunk provider
05:08.05drmessanoI edited my zomg_itsgreen.conf and added the foundthisontheforumsbutitdoesnothing=yes and zomgwtfe=sadface and now i hear echo
05:08.44drmessanocan you tell me how to connect my us robutics winfaxmodem to asterisk too?
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05:09.14M-I-Ayou can't do that?
05:09.24M-I-Alol
05:09.27drmessano<---- DOOR
05:09.50Qwellzomg_itsgreen.conf - that's a keeper.
05:10.42drmessanothats where you set the sendlotsofinfotokerry_g=yes to "sometimes"
05:10.57M-I-AAnyone want to trade something for a 2401E?
05:11.33drmessano2401E?
05:11.54M-I-Adigium TDM2401E
05:12.09drmessanoFully loaded?
05:12.23M-I-Ajust a 4 port fxo
05:13.06drmessanoYou have a 2401E with just 4 ports loaded?
05:13.19M-I-Ayea, don't ask
05:14.13drmessanoI got some quaaludes, an old CB radio, and a box of Garbage Pail Kids Series 1
05:14.22drmessanoTell me what I can do ya for
05:15.01M-I-Athe garbage pail is tempting
05:16.26kd8iktcant wait for iaxtel
05:16.29heedlyI'll take the luds
05:18.57drmessanoiaxtel is a slick idea
05:19.10drmessanoSince FWD frammed it in the dooker
05:22.17kd8iktFWD sucks balls
05:30.03SwKactually ipv6tel is coming
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05:36.59kd8iktsweet
05:39.14JerJerludes man, call up the ludes man, lets get some fucking lueds man
05:39.45JerJeris IAXTel coming back ?
05:40.41QwellJerJer: want to host it?
05:40.59JerJervery possibly
05:41.53JerJeri guess it would depend if I had control over the system that runs it or not
05:42.49QwellI'll see if there's any interest
05:46.09TJNIII think I just had a brute force attack on my sip server today....
05:46.21TJNIIchan_sip.c:14861 handle_request_register: Registration from '"624"<sip:62
05:46.21TJNII4@209.234.89.226>' failed for '217.115.218.68' - No matching peer found
05:46.36TJNIIIn the logs several hundred times, with an incrementing extension.
05:47.04JerJeraddress:        Bucuresti,sect 3, Romania
05:47.12JerJerdescr:          ARtelecom IP Network
05:47.25JerJerabuse-mailbox:  abuse@romtelecom.ro
05:47.49TJNIIYea, I doubt mailing them will do any good....
05:48.54JerJeryeah
05:50.28kd8iktstill
05:50.32TJNIIThey tried 100-999 it looks like.  Probably checking for peers with the peer number as a password.
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05:53.22TJNIIIt's a trixbox machine...
05:54.08JerJeri'm sorry
05:55.59Qwellnot a very bright hacker
05:56.46Qwellespecially considering that IP has an http directory listing...*cough*
05:57.05QwellTJNII: might want to go ahead and fix that
05:57.45TJNIIIf I can guess the password I'll fix it right with a nohup rm -rf /
05:59.54kd8iktsounds like his machine has be hacked already
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06:00.27TJNIIWell, it is trixbox....
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06:35.25M-I-ADo people use trixbox in a professional production systems?
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06:36.08exothermicWhen I have asterisk initiate a call via manager how do I set things like caller ID etc?
06:36.24creativxin the originate event
06:36.28mchou~book
06:36.28jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
06:36.29creativxyou can specify callerid
06:36.41creativxevent.. i mean action
06:36.52exothermiccreativx: can I just specify some system wide default that isn't "asterisk"  ?
06:38.24SwKiirc thats not a setting you just can set as a system wide default... its a compile time constant
06:38.55exothermicSwK: hmm ok
06:39.38SwKexothermic, check out http://www.voip-info.org/wiki-Asterisk+Manager+API+Action+Originate
06:39.49SwKjust set the callerid there
06:40.05exothermicSwK: Well this is for FOP so setting it is a little beyond me.
06:40.30SwKhah
06:40.31SwKyeah
06:40.34SwKthats beyond me
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06:46.57exothermicwell here I go hacking at some perl to see if I'm smart enough to pull this off.
06:47.27SwKif you find where they build the originate command you can probab pull it off easily
06:47.29exothermicsweet well that wasn't so hard I guess.
06:47.46exothermicya a little else statement and I'm all running.
06:47.58exothermicnever written perl before  ;)
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07:30.15fl1pAnyone knows if there are some kind of best practice how to handle ALL those different Release Causes ? when i got a Release Cause 17 User Busy i can use the ${DIALSTATUS} and Jump to an Extension which uses Application Playtones(Busy) and then Busy(), what about the other Release Causes ?
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07:42.18kaldemarfl1p: take a look at ${HANGUPCAUSE}
07:44.01fl1pkaldemar : it´s always 0 when it falls into the failed extension....
07:44.27fl1pkaldemar: ${DIALSTATUS} is a empty String
07:47.09SwKshow app dial
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07:48.09fl1pkaldemar : some Background : I create some call files for the outgoing spool to verify some did numbers if they are available (busy or any other release cause) automatic using a shell script, when the Dialout fails it falls down to the failed extension, ${HANGUPCAUSE} is 0 and ${DIALSTATUS} is empty
07:49.57kaldemarwell, are you maybe directly calling an actual channel in the call file?
07:50.25kaldemarmeaning Channel: Zap/... for example.
07:50.44fl1pSwK: when using Call Files which Dial Parameters are used ? It seems to me there is an internal Dialplan or something Asterisk Dials out, or is it possible to pass Dial Paremeters?
07:51.13fl1pkaldemar: Yes the Call files is Zap/g1/123456789
07:51.19fl1pi mean the Channel of the Call File
07:51.20kaldemarif so, and the extension in the call file is supposed to handle the status codes, no wonder they are empty.
07:52.19fl1pkaldemar : Ah! using a Local Channel would work right ?
07:52.23kaldemarput a Local/... in the Channel part and handle the codes there. the extension in the call file can be anything.
07:52.48kaldemarthose are channel variables, and if you try to read them on a different channel, they are of course empty.
07:53.15fl1pkaldemar : yes you´re right!
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08:13.20SwKflip you you need to see the dial status for the outbound channel leg dont call technology/foo/whatever directly
08:13.27SwKcall it via a local channel
08:13.42SwKthat will loop it thru the dialplan and you'll actually get dialstatus and such that way
08:14.27SwKsomething like this... (replace the \n's with carriage returns)
08:15.31SwKAction: Originate\nChannel: local/18005551212@outbound\nContext: default\nExten: 19009091234\nPriority: 1\n.......
08:16.44SwKthat will send the A leg to the outbound context and find the extension that matches for 18005551212... do your processing as normal on that call... once it is answered, the Extenion line from the originate will fire and it will process thru the default context as specified and go from there
08:17.45SwKfurther help designing and implementing autodialers is 150 euros/hour
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08:37.53fl1pkaldemar: now it works! Thank you very much!
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08:39.38hi365anyone know how i get get the messages key on the polycoms to go straight to vm, instead of that annoying vm status screen?
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08:47.25fl1pSwK : Now everything works fine, so my gained knowledge is  150€/hour worth? good to know, maybe i should talk to my boss :)
08:48.10M-I-Ahe wont listen... already tried
08:48.23M-I-A:)
08:49.15fl1p:) maybe i should get a new job
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08:59.39torrikftif i have 3 sip trunks to the same provider, all showing registered, why do i get congested tone on 2 of them
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09:03.21torrikfti get  is circuit-busy for two of the trunks
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09:40.14phpboyhey all, how can I check against a flat file to see what users may NOT dial out?
09:40.23phpboyor perhaps another way of doing this?
09:43.47kaldemarapp ReadFile and expressions
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09:50.22jblackphpboy: That's determined by your extensions.conf
09:51.59pawproHi all! Can i execute dial to SIP account not specified in sip.conf I.E. "dial sip/username:password@voip.example.com"?
09:52.12jblackpawpro: Yes.
09:52.12tompawhi
09:52.27jblackIn fact, that's almost exactly how you do it.
09:52.35pawpro:)
09:52.44jblackDial(SIP/username:pass@hostname/extension,120)
09:52.55pawproone moment :)
09:52.57jblackof course.. you'll be putting passwords in your extensions.conf
09:54.17jblackwondershaper rocks. Wondershaper rocks. WonderShaper Rocks! Wondershaper rocks. wondershaper rocks
09:54.29pawproits agi script what about no such host IP/extension ??
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09:54.48jblackIt's still the same set of arguments to dial.
09:54.58phpboyjblack: I know, but I need a 'db' to check against
09:55.07jblackperhaps $agi->dial("SIP/username......./extension", 120).
09:55.33jblackphpboy: That's no problem. The argument will still look the same.
09:56.07jblackan arg is an arg, whether it's extensions.conf, perl agi, or talking through stdin/out
09:56.13pawproI do agi->exec("Dial SIP/username:pass@host/0123456789|120")
09:56.18phpboyjblack: do you have a doc that I can real?
09:56.29pawproand i got unknonw host "host/0123456789"
09:56.29jblack~book
09:56.30jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
09:56.33jblackphpboy: There you go.
09:56.34phpboy*read
09:56.53phpboylol, thanks :D
09:57.03tompawjblack: wondershaper?
09:57.12jblacktompaw: Yes. It rocks.
09:58.18jblacktompaw: it's an automagic traffic shaping script. Tell it your max up and down (after subtracing 10k or so), and ssh, voip, everything works, even when bittorrent is running.
09:58.27jblackpawpro: You've got e.
09:58.59jblackIsn't there an $agi->dial() ?
09:59.56pawprowhat is the difference?
09:59.57jblackOh, hmmm. I see the problem.
10:00.18tompawkewl. I prefer pf.conf though ;)
10:00.38jblackSIP/extension@host
10:00.41jblackLet me look a little harder
10:01.09jblackI'm used to iax myself, which is the format I gave you, which is the one you don't need.
10:01.11pawprogot it! Its port
10:01.24pawproI added :whateverport and it gets the host
10:01.38pawpromaybe the next parameter after : or / is port not extension
10:01.56pawproanyway it works. Thanks all! And have a great day
10:02.14jblackOkedoke
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10:02.49phpboytompaw: PF is the shiz :D
10:07.26hi365are there any codec that can playback sln files on windows?
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10:21.39Assidheya
10:22.13Assidhow do i  configure this such as it doesnt get the whole bootrom.ld but the one specific for my device
10:23.16kaldemarconfigure what?
10:23.59Assidpolycom phones, is there any configuration i need to modify so that it chooses the device specific bootrom instead of bootrom.ld ?
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10:25.06qpmorning
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10:55.02Assidkaldemar: also as per the sip documentation of the polycom phones it says i can load these 2 files: 2345-12200-001.sip.ld 2345-12200-004.sip.ld
10:55.08Assidquestion is.. which one do i load?
10:58.01wizzy_If I don't specify an extension *except* in users.conf, can I only override it by putting it in the first context Zap/Dahdi/SIP phones go to (currently "default")? I am contrained this way because I want to use the (great) asterisk-gui AJAX thing
10:58.52wizzy_and I want to have some choice over how voicemail is handled - currently all by macro-stdexten
10:59.30wizzy_It is not clear to me when users.conf is consulted
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11:05.08Assidhrmm anyone here good with polycom phones
11:05.42Assidaccoridng to the sip software release .. the ip330 has 2 files.. how do i know which one to use?
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11:24.39tzafrir_laptopwizzy_, put it in a different context, that deos not inlcude the default one
11:24.53hi365anyone know how i get get the messages key on the polycoms to go straight to vm, instead of that annoying vm status screen?
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11:25.33tzafrir_laptopwizzy_, make a new "dialplan"
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11:35.16barakuda_workpeople help me please: i need to add custom header to SIP (for example - X-Unique-Id: 12345)
11:35.22barakuda_worki wrote in my extensions.conf such strings:
11:35.29barakuda_workexten => 401,1,SIPAddHeader(X-Unique-Id: 12345)
11:35.35barakuda_workexten => 401,2,Dial(SIP/401,20)
11:35.41barakuda_workexten => 401,3,Hangup
11:35.48barakuda_worki saved file, and reloaded asterisk. when i call ext.401 asterisk writes in console, that sipaddheader worked fine, BUT, on the client's side there are no my custom header.
11:35.54barakuda_workWhat's could be wrong?
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11:46.53kaldemarbarakuda_work: check that the header gets added with "sip set debug" in CLI
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11:50.19wizzy_tzafrir_laptop: this is for inbound calls - I cannot change the context to different people when coming from the trunks ?
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11:55.48barakuda_workkaldemar: sip set debug help a lot. i see that header is added. thanx, kaldemar ;-)
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12:34.23mvanbaakpuzzled: pong
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12:34.50Andre101Hey people, anyone able to add a sip account and dial an echo test? Testing some NAT issues
12:34.59puzzledmvanbaak: hey. had a question about the syntax of dialplan.xml for a 7961 but I found it already
12:37.15mvanbaakah
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12:38.35puzzledanyone know what the max number of phones is that you can dial simultaneously with Dial(SIP/10&SIP/11&SIP12....) ?
12:39.43Andre101anyone available to quickly help with my NAT problem?
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12:41.50lmadsenpuzzled: there isn't a limit... the limit is more going to be in the number of characters you can have as a single argument to dial
12:41.57[TK]D-Fenderpuzzled: More than *'s string limitation ;)
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12:42.22[TK]D-Fenderpuzzled: You could of courst nest multiple local channels to break it up.
12:42.33lmadsendone that too :)
12:42.42[TK]D-Fenderpuzzled: For which the limit should be largely that of the load it places.
12:42.59puzzledthat makes total sense
12:43.09lmadsenDial(Local/group1@multicall&group2@multicall)
12:43.23lmadsenexcept... do it right unlike what I just did there :)
12:43.28lmadsenDial(Local/group1@multicall&Local/group2@multicall)
12:43.31lmadsenbetter :)
12:43.40lmadsenhas been up for a whole 10 minutes
12:44.17Andre101Hey people, anyone able to add a sip account and dial an echo test? Testing some NAT issues
12:44.17puzzledlmadsen: right and group1 would be a Dial(SIP10&SIP11...)  ?
12:44.17geniorealHi, i got a question, i got a server with a digium Card and im installing Asterisk on it,  im doing all this steps http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation  i want to know if zaptel is real needed and was does zaptel do? becouse when i make on the zaptel dir i get "You do not appear to have the sources for the 2.6.18-92.1.10.el5PAE" and i got a Linux centralpbx.txp 2.6.18-92.1.10.el5PAE #1 SMP Tue Aug 5 08:14:05 EDT
12:44.18genioreal2008 i686 i686 i386 GNU/Linux
12:44.23lmadsenpuzzled: exactly
12:44.28[TK]D-Fenderlmadsen: Not bad, its one thing to make mistakes, its another to miss them completely.  You're doing jsut fine :)
12:44.33puzzledgreat thanks lmadsen and [TK]D-Fender
12:44.45lmadsen[TK]D-Fender: I'm usually pretty good at finding them :)
12:44.56[TK]D-Fendergenioreal: Yes you need Zaptel, it is the driver interface for your cards <---
12:44.58lmadsencan't wait for his water to boil so he can make green tea
12:45.15lmadsenpuzzled: np!
12:45.43[TK]D-Fendergenioreal: "yum install kernel*"
12:46.01[TK]D-Fendergenioreal: This will no doubt end up upgrading your kernel as well for which you'll need to reboot
12:46.23geniorealOK [TK]D-Fender ill try that, very helpfull thxs
12:46.29lmadsengenioreal: yum install kernel-devel
12:46.37genioreali got kernel-devel installed
12:47.10lmadsenthat will get you the kernel sources, however, if your kernel is not the newest, the newest kernel-devel will be installed, in which case it will not be the right version for your currently (out of date) kernel
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12:49.00genioreallmadsen: i did the first steps of the link (manual) and there is a yum -y update && yum -y upgrade && yum -y install kernel-devel etc... etc.. etc..
12:49.19geniorealim gonna try now what [TK]D-Fender said but thxs lmadsen too :)
12:49.32lmadsenyou need to reboot then because you probably upgraded the kernel
12:49.34[TK]D-Fendergenioreal: I like carpen-bombing :)
12:49.39[TK]D-Fendercarpet*
12:49.56genioreallmadsen: i rebooted the machine  more the once and i updatedb too
12:50.12[TK]D-Fendergoes back to playing "Scorched Earth" with Norad systems...
12:51.52miloux|NotHereI need to troubleshoot interrupts, as i believe there is a device that "locks" to long and then causes "PRI got event: HDLC Bad FCS (8) on PrimaryD-channel of span x"... And i have gotten indications that this might be related to that...So anyone know how i can Troubleshoot this problem?
12:52.31genioreal[TK]D-Fender: now is installing kernel kernel-PAE-devel -debug-level -xen -doc
12:52.36genioreallets hope this solve the problem
12:52.37genioreal:)
12:52.46[TK]D-Fendermiloux|NotHere: "cat /proc/interrupts" + "dmesg" > PASTEBIN
12:52.58miloux|NotHereaight
12:53.04[TK]D-Fendermiloux|NotHere: And "ztcfg -vvvv"
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12:55.28miloux|NotHere[TK]D-Fender: http://pastebin.com/m1da5e2dc <--- interrupts + dmesg
12:56.09miloux|NotHerehttp://pastebin.com/m4eba0c31
12:56.13miloux|NotHerefor ztcfg
12:56.50[TK]D-Fendermiloux|NotHere: Disable APIC for your kernel
12:57.26miloux|NotHereOK
12:57.37BarthezZcan asterisk have different verbosity levels on different "remote" unix connections?
12:57.53miloux|NotHereand just now, 20 secs ago...all calls got hungup and in the cli this: http://pastebin.com/m57898514
12:58.13[TK]D-FenderBarthezZ: Nope.
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12:58.55Blackvelhi. what ssh client do you use for asterisk and what terminal emulation? I use putty and have some weird display problems with ztmonitor
12:59.27[TK]D-FenderBlackvel: Putty, and I lock mine to 80x25
13:00.32tzafrir_laptopBarthezZ, it can, in trunk
13:00.39tzafrir_laptopI'm not sure about 1.6.0
13:01.22[TK]D-Fenderhttp://www.engadget.com/ <--- Lol @ Windows 7
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13:02.05miloux|NotHere[TK]D-Fender: any other suggestions? This problem startde 2 days ago and nothing has changed since...
13:02.10miloux|NotHerehas changed
13:02.12miloux|NotHerehardware
13:02.20miloux|NotHerethe problem just appeared
13:02.33[TK]D-FenderQ: If it takes 1000 Microsoft engineers to code Windows 7, how many are required to change a lightbulb?  A: None, they just define darkness as an industry standard.
13:02.49[TK]D-Fendermiloux|NotHere: You've heard my only suggestion.  Beyond that, contact Digium support.
13:03.11miloux|NotHere[TK]D-Fender: ok, ive already done that tho
13:04.06flushahoy
13:04.28flushi have set some CALLFILENAME in my extensions.conf file to record outgoing calls, but im wondering.. where are the conversations saved ?
13:04.32[TK]D-Fenderchips
13:04.46Blackvel[TK]D-Fender: any special emul options which I have to use? ztmonitor only displays this: 80 columns, 25 rows?
13:04.51[TK]D-Fenderflush: Look in asterisk.conf's spoolfonder\monitor
13:04.53Blackvel80 / 24 standard here
13:05.01yang2[TK]D-Fender: With such a budget that MS has it, they can hire the best coders, but I think they intentionally have their products crippled
13:05.11[TK]D-FenderBlackvel: 80x24 rather, and no, nothing special
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13:05.46Blackvel[TK]D-Fender: ztmonitor only displays this: <----------------(RX <----------------(TX
13:05.55Blackvelthe bar is completely wrong
13:06.15yang2[TK]D-Fender: the people I work with don't use Vista at all - they all use patched XPs, so much about the new systems...
13:07.17[TK]D-Fenderyang2: Ditto.  I have so far evaded it at my office & home.  None I know uses it except 1 coworker who got it on a laptop.
13:08.20yang2[TK]D-Fender: and what is the most annoying you cannot change "locales" in windows vista - if you get a German version only 10% of the menus will be in a different language
13:10.29flush[TK]D-Fender cool ok i got them, theyre in /var/spool/asterisk/recording, asterisk will never flush whats in /var/spool/ ? so its safe to leave it there ?
13:11.02[TK]D-Fenderflush: Yes
13:11.19flushcool thanks
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13:17.15genioreal[TK]D-Fender: the kernel problem is solve, thxs. but now i get this http://pastebin.com/d6d5c337e when i hit the make, any idea why?
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13:17.46[TK]D-Fender~centos52bug
13:17.47jbot[~centos52bug] There is a bug compiling Zaptel  up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver  in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889
13:17.48[TK]D-Fender^^^^
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13:19.19geniorealok, then have to patch it hehe thxs again
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13:25.32genioreal[TK]D-Fender: i wget the patch now i get File to patch:  <-- there i must put this path: /usr/src/asterisk/zaptel-1.4.11/kernel/xpp/xdefs.h <-- right?
13:25.56[TK]D-Fendergenioreal: I would simply have unchecked XPP from menuselect...
13:27.18qpsame question again today, sorry, but is there anyway of flagging in the * log when an operator performs an attended transfer, so we can charge more for operator talk time and less for a transferred calls time
13:28.17genioreal[TK]D-Fender: sorry my english is not the best i did not understand  what u mean with uncheck the XPP from a menuselect, i got this: http://pastebin.com/m74f2016c
13:28.29[TK]D-Fender~centos52bug
13:28.30jbot[~centos52bug] There is a bug compiling Zaptel  up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver  in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889
13:28.31[TK]D-Fender^^^^^
13:28.44[TK]D-Fender"make menuselect" <- UNCHECK "xpp"
13:32.45geniorealthxs again
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13:44.01[TK]D-Fendergenioreal: np
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13:49.05Blackvelwhat results should ztmonitor give me? Milliwat test with magic number for rx/tx 14844?
13:49.20Blackvelthat applies for a BRI card too?
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13:49.25tzangerBlackvel: yes
13:50.05Blackvelthe bar display graph is totally weird in my putty window :)
13:50.29tzangerheh
13:50.35Blackvelare echo cancellation codecs like octasic softecho or digium hpec still depend on the rx/tx gain values?
13:51.03Blackvelso if someting is wrong there might still be problems? read about olsec which seems not caring about rxgain/txgain anymore...
13:51.11Blackvelbut right now I am testing around with octasic
13:51.44Blackvelthe half side of rx and tx is missing..dunno why
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14:01.36bram247anyone use bandwidth.com with switchvox? i need to be able to redial numbers with +1 or chop off the +1 as a call comes into the system, but seems to be a limitation of switchvox
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14:03.45bram247seems switchvox freaks on the + sign
14:04.46bram247i have tried passing all 12 digits, but for some reason it still fails, and i cannot put + in the special chars option box
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14:04.54bram247for outboud ruls
14:04.57bram247rules^
14:09.20[TK]D-Fenderbram247: This is NOT #switchvox
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14:20.12alivercan someone check out my sip.conf and extensions.conf and tell me WTF is wrong? My phones aren't ringing sip debug shows that the calls come in but it never gets to my NoOps(). http://pastebin.com/d7f8731a1
14:20.34aliverBTW, is there any * command to show me errors in my config files?
14:20.46aliver'cause reload goes by WAY too fast and too verbose.
14:20.49[TK]D-Fenderaliver: You get errors on load.
14:21.05[TK]D-Fenderaliver: And what you SHOULD be including is SIP DEBUG from CLI for the attempted call that failes.
14:21.18aliverokay will paste that in. One sec.
14:21.19[TK]D-Fenderaliver: And thats what SCROLLBACK is for.
14:21.34aliverI got that but I still visually have a hard time sifting through it all.
14:24.37aliverhttp://pastebin.com/d304de67c    <--- sip debug
14:25.10[TK]D-Fenderaliver: Found no matching peer or user for '4.79.212.236:5060'
14:25.20[TK]D-Fenderaliver: Looking for +19702033733 in internalphones (domain 216.17.133.231)
14:25.26[TK]D-Fenderaliver: SIP/2.0 404 Not Found
14:25.48[TK]D-Fenderaliver: your itsp peers should be "nat=no"
14:26.13[TK]D-Fenderaliver: and I suggest "insecure=port,invite" as well
14:26.14aliverexten => _+19702037XXX,1,Set(DID=${EXTEN:9})    <--- so this isn't working?
14:26.20aliverokay, I'll change that for sure.
14:27.48aliverOkay, I'll add that insecure=port,invite too.
14:29.03[TK]D-Fenderaliver: problem is it isn't matching your PEER and sending it to the right context.
14:30.25aliverhmm, okay so I need to change the context to "internalphonhes" Doh!
14:30.26[TK]D-Fenderaliver: You probably shouldn't even have those as 2 entrier.
14:30.47aliverwhich two?
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14:32.22[TK]D-Fenderaliver: When you look at it, you don't have a pattern to match +19702033733
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14:32.42aliverOh, that's probably why I'm still getting 404
14:32.49[TK]D-Fenderaliver: It is hitting the "right" context (poor choice of names & contexts, just that the pattern doesn't match
14:32.50aliverLet me scrutinze the pattern.
14:33.01aliveryes, the context now works I fixed that.
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14:33.12aliverit's going to "internalphones" which actually has an extensions.conf entry.
14:33.39[TK]D-Fender+19702037XXX
14:33.41[TK]D-Fender+19702033733
14:33.44[TK]D-FenderNo match
14:33.55alivergah!
14:34.00alivergood catch.
14:35.30geniorealHi i got a little problem when im running asterisk http://pastebin.com/d2ce5e0fd i have done step by step from  here http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation it says if a need zaptel services i must correctly configure Zaptel, but i did allready by the steps on the manual, any idea?
14:35.52aliverit's weird, though. I don't even see my NoOp().
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14:36.01aliverI fixed the pattern... I think I'll double check.
14:36.21[TK]D-Fendergenioreal: Go look at your zaptel config and verify that "ztcfg -vvvv" is OK
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14:38.44genioreal[TK]D-Fender: my /etc/zaptel.conf is all commited with # and i only got one ztcfg that is:  # This file is parsed by the Zaptel Configurator, ztcfg
14:40.24aliverIs there any pattern that will match a "+" and a number?
14:40.30aliverwill "." work?
14:41.10[TK]D-Fenderaliver: Yes.
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14:41.50[TK]D-Fendergenioreal: Go configure your card in zaptel.conf & zapata.conf , and verify that its module is loaded and that ztcfg checks out
14:42.03mort_gibgenioreal: I have a similar issue...
14:42.55geniorealok [TK]D-Fender ill go look for a guide to configure the right way the zaptel.conf and the  zapata.conf
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14:46.45aliverstill getting a 404 Where did I fsck up? http://pastebin.com/m6f7ad93f
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14:48.03genioreal[TK]D-Fender:  if i got a Digium TE412P do you know that fxo/fxs interface cards will i need to set on the zaptel.conf?
14:48.06aliverI'm not even getting to my NoOp()s
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14:48.24aliverit's okay to have multiple "1" priorities in the same context right?
14:48.35aliveras long as they go to different numbers, right?
14:48.39[TK]D-Fenderaliver: You only put "." at the END of the remainder of a pattern, not in the FRONT.  put the "+" back
14:49.42aliverokay, tried that already
14:50.20[TK]D-Fenderaliver: Yah, and the last one you showed me with the + had the pattern wrong
14:50.34aliverbut ill put it back
14:50.54aliverI fixed it, though.
14:50.57aliverI thought...
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14:54.34aliverIs there any way to turn off all the chatter from the phones ?
14:54.48[TK]D-Fenderaliver: Nothing I'd advise for your testing
14:54.54aliverokay.
14:57.01geniorealhttp://pastebin.com/d70a41c5e
14:57.46geniorealZT_SPANCONFIG failed on span 1: Invalid argument (22) <-- where can i look for this invalid argument to fix it?
14:58.09Blackvelguys I don't get it. is the value 14844 really the right one on an ISDN BRI? channel 4 calls into pstn, and channel 5 accepts the call from telco and forward to asterisk Milliwatt/Playtones
14:58.28BlackvelI made sure that there is no echo canceller on
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14:59.24Blackveltx is 4319, rx is 10412
14:59.39[TK]D-Fendergenioreal: What card(s) do you have?
14:59.51genioreal<PROTECTED>
15:00.39[TK]D-Fendergenioreal: IIRC those have a jumper that needs to be set for E1.  Go verify it
15:00.41BlackvelI wonder that tx 4319 can be used with tgain 0.0 and be adjusted to 14844? :)
15:00.48Blackvelthats probably way too much, isn't it?
15:02.26*** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla)
15:02.26*** mode/#asterisk [+o russellb] by ChanServ
15:02.34genioreali set the zaptel conf by this guide [TK]D-Fender http://www.asteriskguru.com/tutorials/wildcard_te407p_wildcard_te412p.html
15:02.39geniorealok ill send
15:02.47geniorealok ill look
15:04.35*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
15:04.42aliverhttp://pastebin.com/m63534b9  <--- can't figure out WTF is wrong with my pattern.
15:05.26genioreal[TK]D-Fender: to look at the jumper for E1 i need to have the server open right? there's no other way? becouse i cant open the server (guaranteed)
15:06.08[TK]D-Fendergenioreal: If your server, your card, your eyes.  Do what you must
15:06.57aliverI guess 1.2 has no "dialplan show" eh?
15:07.06*** join/#asterisk zamba (i=marius@sveigde.hih.no)
15:07.18aliveris there anything like it in 1.2 ?
15:07.30genioreal[TK]D-Fender:  and if the jumper is set for E1, what else might it be?
15:07.35zambai need help shaping my connection for use with voip.. i'm currently using wondershaper, but it's not giving voip traffic ultimate priority.. how can i do this?
15:07.48*** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)
15:07.55aliverhttp://pastebin.com/m63534b9 <--- Anyone see why my pattern won't match and I still get a 404 ?
15:08.05*** join/#asterisk CunningPike (n=arodgers@204.239.8.157)
15:09.08[TK]D-Fendergenioreal: Bad zapte/zapata config.
15:09.27*** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun)
15:10.33genioreal[TK]D-Fender: do you now the right jumper for E1 to move? is my first time with a digium card
15:10.54[TK]D-Fendergenioreal: Go look at the card.
15:11.25[TK]D-Fenderaliver: Yes, there is
15:11.54[TK]D-Fenderaliver: And do a "reload to be sure its in effect
15:12.22*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
15:12.25genioreal[TK]D-Fender: ok, this are my conf anyway http://pastebin.com/d6baca8d7
15:14.21[TK]D-Fendergenioreal: I suggest contacting Digium support
15:15.17geniorealok thxs
15:15.56*** join/#asterisk moy (n=moy@189.169.82.40)
15:16.14anonymouz666[TK]D-Fender: did you ever tried to contact by e-mail?
15:16.30[TK]D-Fenderanonymouz666: once, long ago
15:16.34anonymouz666mine got lost, I think.
15:17.53aliverdamn I can't even get _X. to match.. UGh.
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15:22.26aliverhttp://pastebin.com/d451bc8f9 <-- can someone see why _X. doesn't match?
15:22.55*** join/#asterisk c4t3l (n=root@74.95.210.124)
15:23.16c4t3lwhat up playaz
15:23.41*** join/#asterisk mog (n=mog@nat/digium/x-5364dc7026058077)
15:23.41*** mode/#asterisk [+o mog] by ChanServ
15:24.57alivergrrr. this is weird.
15:25.21aliver_X. should always match right?
15:26.27Qwellaliver: should always match what?
15:26.37[TK]D-Fenderaliver: No, it won't
15:26.53[TK]D-FenderX != +
15:27.41aliverahh, crap.
15:28.15brodiemlol what's with the huge ascii banners in your extensions.conf
15:29.01seanbrightthat's pretty cool, actually.
15:29.02aliverI'm blind.
15:29.06aliverso it helps
15:29.10aliver<PROTECTED>
15:29.12brodiemhehe
15:29.13seanbrightwe could probably add that directly into asterisk
15:29.13aliverWTF.
15:29.28aliverwhy does ${EXTEN} say "t" there?
15:29.36seanbrighttimeout
15:29.39seanbrightt = timeout
15:29.52*** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com)
15:30.01aliverEven if the phone isn't online (which it IS online)
15:30.11aliverit should still not do that should it?
15:30.15aliverwhat does that indicate?
15:31.44aliverahh, I see.
15:31.51aliverit's just at the end when it hangs up.
15:31.57aliverthe one above looks normal.
15:34.05*** join/#asterisk Knightfa| (n=jjj@66.178.134.235)
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15:37.58Blackvelstupid question. how can I tune rxgain/txgain for group 2 with ztmonitor (playing milliwatt application myself) when channel 4 (out) connects into asterisk over channel 5 (milliwatt)?
15:38.56Blackvelwhen I change rxgain from 0.0 to 10.2 its ~14844
15:42.19Blackvelif txgain is 0.0. but when I monitor ztmonitor channel 5 for changes, to raise txgain from 0.0 to e.g 10.0, it changes again the rx ztmonitor values which are now >25k
15:42.22*** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it)
15:43.24Blackvellooks like I have to probe the txgain a little bit different and can not use asterisk to meassure output of Milliwat on ZAP channel 5?
15:44.07Qwellan rxgain of 10 is ridiculously high
15:44.33Blackveltought that too
15:44.42jeeveyes Qwell.
15:44.54Qwellstabs jeev, and wonders where his million dollars is.
15:44.56Blackvelso is the number 14844 not working for ISDN BRI Milliwatt meassurement?
15:45.08jeevQwell, you sure you didn't touch that package?
15:45.12QwellBlackvel: I don't know what number you're referring to
15:45.23jeevred bull stealer.
15:45.36Blackvellikes to throw away junghanns bri card and likes to get patton voip gateway
15:45.53Blackvelqwell: in ztmonitor wiki it says meassurement HAS to be done with Milliwatt testtone
15:46.04*** join/#asterisk AlexTO (n=alex@190.159.213.36)
15:46.06Blackveland values should be raised to stay around 14844
15:46.14Qwellumm, okay
15:46.41Blackvel###################################*                                          Rx: 14588 (14588) Tx:     0 (    0)
15:46.52jeev14844 will make you impotent
15:47.13BlackvelI already have the feeling that the wiki documenation for the HOWTO is missing some important part
15:47.23BlackvelI have no idea :(
15:47.36Blackvelwhats the correct way to tune ISDN BRI with ztmonitor? :(
15:47.37Qwellwhy are you adjusting gains?
15:48.21Blackvelthing is this whole thing is crap
15:49.13Knightfa|Im looking for a little analysis from some one. Can some one look at the link http://pastebin.com/mca3b4d8 and maybe sort thru why my call may not be connecting correctly. I have put an example of a successfull call and a failed call in the link The first example uses a script to dial the call the 8203 are stripped(for CDR purposes) and then its supposed to dial... Thanks in advance
15:49.44Blackveli noticed echos...trying to play around. introduced octasic echo cancel. but then on the opposite side I got told: my voice is too silent. so txgain should be raised
15:49.54Blackvelif you ask me, all does not match anymore
15:50.50*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
15:50.58Blackvelits all about this: To help solve this we have echo cancelers, but echo cancelers can only do so much, the better the signal we can get to begin with, the easier the job the echo canceler has. To get a clean signal we need to adjust our rxgain and txgain values in zapata.conf for Asterisk (http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx)
15:51.17BlackvelI have no feeling if my in/out signals are right
15:51.17kaldemarKnightfa|: RIOTRU
15:51.57[TK]D-FenderBlackvel: Does "SIP/5551212" look like "SIP/RIOTRU/5551212" to you?
15:52.15kaldemarcollateral damage.
15:52.36jeevfender, wrong dood.
15:52.51jeevthat was Knightfa|
15:53.23[TK]D-FenderKnightfa : yeah, you..
15:53.24Qwelljeev: [TK]D-Fender is always right - you just read the answer wrong
15:53.37jeevsees that fender is really struggling since he took out one of his eyes
15:53.43jeevsees that fender is really struggling since he took out one of his eyes with his katana
15:54.27aliverIf I get half-duplex audio from outside I'm probably facing nat or a reinvite=yes/no issue right?
15:54.35[TK]D-Fenderaliver: Quite likely.
15:54.49Blackvelbefore I defenitely throw this equipement thru my window....how can I tune a damn isdn bri line with ztmonitor? What numbers count?
15:54.56Blackveldefinitely
15:54.59[TK]D-Fenderaliver: Considering I've told you messed up your peers and that it wasn't even matching them on inbound anyways
15:55.14QwellBlackvel: what driver are you using?
15:55.25Blackvelzaptel/bristuff + octasic
15:55.39puzzledBlackvel: I never had to tune an isdn line in my life. what's the problem?
15:55.42*** join/#asterisk FabiOne (n=FabiOne@host10-25-static.104-82-b.business.telecomitalia.it)
15:55.44Blackvelshall I just ignore rxgain/txgain and leave to 0.0 default?
15:55.44Qwellbleh, bristuff
15:55.45FabiOnehi all
15:56.08puzzledBlackvel: I never touched tx/rxgain. always leave them at 0.0
15:56.13aliverFender, I got that working now. I think I'm just boneheading my firewall and not letting the RTP ports in.
15:56.33Blackvelpuzzled: but how do you get the optimal volume strength?
15:56.46Blackvellike that is isn't too silent for the called party (txgain)
15:56.54FabiOneexcuse me, there are italian people? I need help for Italian laws
15:57.03*** join/#asterisk nny_2 (n=Scott_My@64.203.244.146)
15:57.07QwellFabiOne: getting legal advice on IRC is a BAD idea.  Consult a lawyer.
15:57.15puzzledBlackvel: never had an issue with volume either. are you sure your isdn line is ok?
15:57.30Blackvelnot sure about nothing ;)
15:57.33puzzledheh
15:57.38Blackvelno idea where to start
15:57.49nny_2looking around to see if there are ways to set options at the end of Dial (,wWk etc) globally. Any advice appreciated
15:58.06FabiOnei need to know what authorization i need to made a pbx with *
15:58.09puzzledBlackvel: start from scratch. rx/tx at 0.0 and no echo can enabled. see what happens then
15:58.15*** join/#asterisk Assid (n=assid@unaffiliated/assid)
15:58.22QwellFabiOne: You need to talk to a lawyer.
15:58.26qpis there an * command to quickly add a row into the cdr?
15:58.27Blackvelpuzzled: i just can say there are echos from time to time on this isdn bri line. with mg2. now i switched to octasic
15:58.28FabiOneokay
15:58.28[TK]D-FenderFabiOne: None.
15:58.46FabiOnetnx anyway
15:58.47Knightfa|[TK]D-Fender: I noticed that But Im having issues with that portion of this .... Let me post up my config ...I thought I had this set correctly
15:58.57Blackvelbut I have the feeling there won't be an 100% echo free solution without optimum values (olsec doesn't care about txgain/rxgain e.g)
15:59.02[TK]D-FenderFabiOne: You don't need any authorization to make a PBX with *
15:59.06puzzledBlackvel: a bit of echo now and then is not uncommon specially when calling a gsm
15:59.09Qwell"Your Honor, the people on IRC said I could murder that guy."
15:59.17nny_2I can? Sweet!
15:59.27nny_2wait just one of them ?>
15:59.32[TK]D-FenderKnightfa|: Those 2 lines are clearly different.  go fix it
15:59.50puzzledBlackvel: did you try oslec yet with rx/txgain at 0.0?
15:59.52Blackvelpuzzled: this diva server >1000 euros is too expensive for me. I am thinking about to get this done with zaptel/bristuff/octasic or to go for 369 euro patton isdn bri voip gateway
16:00.32Blackvelnot yet :) opinions seem to be sooo different of all the guys
16:00.58Blackvelfor one olsec or octasic fixed everything (I dislike the background accoustic noise e.g on octasic)
16:01.18exothermcAnyone know if it is easy to get flash operator panel to use spyexten() instead of moving all legs of the call into a meetme?  It is just so messing since the original legs of the call stay in the meetme.
16:01.26puzzledBlackvel: check ebay.de. I've seen quite a few eicon diva server bri cards on them
16:01.30exothermc*messy*
16:01.54Blackvelone says olsec is nice, but octasic quality is better (its commercial so no compromise)
16:02.28puzzledBlackvel: well since oslec is free it can't hurt to try right?
16:02.28Blackvelother 3-4 ppl describe scenarios, where only voip gateways fixed 100% echos, neither mg2/olsec/octasic
16:02.31exothermc*extenspy*
16:03.32Blackvelpuzzled: true, sure...same problem as with octasic. how can I rely on 1-5 test calls? one try with no echo probably does not mean its gone forever in every situation (e.g voip on the other side, gsm, etc.)
16:03.56*** join/#asterisk brettnem (n=brettnem@user-387oe35.cable.mindspring.com)
16:04.21brettnem~centos52bug
16:04.22jbot[~centos52bug] There is a bug compiling Zaptel  up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver  in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889
16:04.23BlackvelOkay...i forget about this milliseconds 14844 value for isdn bri ;)
16:04.40Qwell14844 milliseconds?
16:04.57Blackveleh nah..wrong
16:05.58Blackvelztmonitor value 14844 for milliwatt 1004 khz test
16:06.14puzzledBlackvel: I built a box with bristuff 0.4.0-RC3b, oslec svn revision 1613, zaptel 1.4.11, libpri 1.4.3 and asterisk 1.4.21.1 and it works quite well. even with gsm calls
16:07.45Blackveloslec ignores rxgain/txgain. dunno about octasic
16:07.50Blackvelits not mentioned in the faq
16:08.34puzzledBlackvel: well I never had to mess with those values both when using an eicon and bristuff/hfc-s
16:08.56*** join/#asterisk airjump (n=chatzill@62.159.95.82)
16:09.02FabiOnei logoff
16:09.05FabiOnesee u
16:09.59airjumphello
16:10.46*** join/#asterisk l2trace99 (n=jr@75.112.133.235)
16:11.32genioreal[TK]D-Fender: it was a conf problem i guess
16:11.43genioreal25 channels to configure.
16:11.46geniorealno error now
16:11.58[TK]D-Fendergenioreal: Good to hear.
16:12.43genioreal:)
16:13.09*** join/#asterisk xacatecas (n=jkroon@196-209-108-119-wblv-esr-3.dynamic.isadsl.co.za)
16:13.19airjumphave someone a good documentation for installing asterisk on a freebsd 7.0 ? thx
16:13.44xacatecashi all, i've got some BT200 phones here.  My problem is that on some phones it sounds like the incoming voice stream is cutting out.
16:13.49neurosysairjump:  Just install from ports?
16:14.16airjumpneurosys: thanks ;-)
16:14.36xacatecashas anybody else experienced this?  managed to locate a fix?
16:14.59kaldemar~gs
16:15.00jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
16:15.11bradleyprice86[TK]D-Fender: Not sure if you remember helping me about a week ago, but I had some problems with recorded messages and voicemails having static when listening to the from the freepbx web interface or through email, but the quality was fine when calling voicemail or downloading the messages straight from the server through winscp. You mentioned it might be an Apache problem. Are there any configuration changes that should be changed in apache for
16:15.49[TK]D-Fenderbradleyprice86: No, Apache would have nothing to do with Aquality issues
16:16.28[TK]D-Fenderbradleyprice86: I asked you to confirm between known decent SIP devices directly, VS whatever interface your PST origin calls come in on.
16:18.07*** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)
16:18.21nny_2meh god sox's syntax is a steaming pile
16:18.41xacatecaslol @jbot
16:18.47nny_2guess i can't do any better
16:19.59xacatecassame problem with the SNOM M3 here that i just managed to locate - something with the PBX setup perhaps?
16:22.06bradleyprice86[TK]D-Fender: I have made several call recordings and voicemails using different devices. We mainly use X-lite softbased phones here, but we have a few polycom hard based phones. The voicemail messages sound fine when calling in from either type of device. The qualit issue is only using the web interface.
16:22.51*** join/#asterisk scampbell (n=scampbel@mail.scampbell.net)
16:22.58[TK]D-Fenderbradleyprice86: So if you download a recording off the web & play it back is sounds bad, but when yuo listen to that same message on a Polycom phone it doesn't?
16:24.11bradleyprice86[TK]D-Fender: Correct. If I listen to it on the softbased phone it sounds fine. It is only when streaming it from freepbx or downloading it through freepbx.
16:24.38[TK]D-Fenderbradleyprice86: Don't know about "streaming".  if you DOWNLAOD the file is it bad?
16:25.43bradleyprice86[TK]D-Fender: Yes. In freepbx once you open the windows in which the audio is streamed, you have the option of downloading it. If I download it from there it has the bad quality.
16:26.22[TK]D-Fenderbradleyprice86: Then clearly the codec used on the PC is bad.
16:26.30*** part/#asterisk airjump (n=chatzill@62.159.95.82)
16:27.14bradleyprice86[TK]D-Fender: I actually checked the md5 between two pc's from which I downloaded the same file. Neither of the two matched each other, or the original from the Asterisk server.
16:28.08[TK]D-Fenderbradleyprice86: If you are looking at the same file on both ends then its the listener who's screwed
16:28.40*** join/#asterisk droops (n=droops@74.193.255.27)
16:29.30bradleyprice86It's not the same file. That's the reason I though it might be a streaming issue. Neither file downloaded from the freepbx ARI matches the original.
16:29.44*** join/#asterisk axisys (n=axisys@155.70.141.45)
16:30.45bradleyprice86Anyone know of a good, reliable, and inexpensive voip provider for residential use?
16:32.20mmlj4teliax
16:32.51mmlj4http://www.teliax.com/?a=64254321
16:33.48*** join/#asterisk fl1p (n=fl1p@ip-88-153-12-37.hsi.ish.de)
16:34.58*** join/#asterisk aliver (n=aliver@ip-216-17-149-97.rev.frii.com)
16:35.10aliverany idea what this means in a half-duplex audio situation:
16:35.15aliverAug 20 10:34:07 WARNING[689]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission DENMGC0320080820163323059802@209.244.63.17 for seqno 1 (Critical Response)
16:35.17Knightfa|[TK]D-Fender: Thx Man after going thru the configs again I saw the issue...appreciate the second set of eyes. :)
16:35.17[TK]D-Fenderok, out for a while, back later
16:35.45aliverDoes that mean I can't get packets outta asterisk to the peer?
16:39.01*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
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16:39.11*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:39.25lmadsenbeside zoiper and x-lite, anyone have recommendations for a decent, free, softphone for OSX? I'm doing testing, and don't want to get out any more hard phones :)
16:39.29lmadsenalready running zoiper and x-lite
16:40.21*** join/#asterisk lejun (n=lejunhu@66.178.134.235)
16:40.32anonymouz666x-lite works fine.
16:41.19lejunexcept you can't transfer unless you buy
16:41.31Qwelllmadsen: Qwell Communications has a solution for you
16:41.36QwellIt's a Layer 1 softphone...
16:41.49lmadsenanonymouz666: I don't think you read what I wrote...
16:42.13lmadsenQwell: I already have the layer 1 firewall... and its not as good as the one offered by telecomjoshleifvoxmartenterprisesinc
16:42.27anonymouz666lejun: can't transfer?
16:42.33QwellThat sir, is defamation.
16:42.35QwellI will sue you.
16:43.13lmadsenQwell: that's fine... you'd just be suing yourself since tcjlvei is a subsidiary of qwell communications
16:43.29Qwellwell...yeah...
16:43.32Qwellbut that's beside the point
16:43.37lmadsenagreed
16:45.07*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-1d50eca9dc05d996)
16:45.24jeevscratches self
16:46.12*** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net)
16:46.44Kobazon a t1/pri is it possible to be the CPE side and provide clocking
16:47.06Kobaz...and have it work
16:47.36*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:47.42Kobazwe're linking up to a nortel and they have it set up to be the user side and provide clock
16:51.00*** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk)
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16:51.28*** join/#asterisk pkinville (n=pkinvill@adsl-76-232-156-10.dsl.sfldmi.sbcglobal.net)
16:51.42pkinvilleHey guys, i have a quick noob query if anyone has the time
16:51.46Qwell~ask
16:51.47jbotask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
16:52.03lejunanonymouz666: yeah, says you can have that feature when you upgrade to eyebeam 1.5
16:52.22pkinvillethanks.  do you guys happen to know if it is possible to use Asterisk to seperate the RX and TX portions of a particular phone call and do as you like with them?
16:52.37Qwellpkinville: can you explain a little more?
16:52.46Kobazlicks Qwell
16:52.52QwellKobaz: O.o
16:53.12seanbrightIRC... it's just like real life... with licking
16:53.16pkinvillesay i have a call come in, right?  We will call it Call A.  And I want to switch the RX portion of Call A with Call B (another call).  is this possible?
16:53.44Qwellpkinville: I'm sure you could hack something up with chanspy
16:53.53pkinvillechanspy?
16:54.00Qwellassuming I'm picturing the scenario properly
16:54.02pkinvillesorry, i'm totally a noob here so forbie me
16:54.07nny_2heh i made a script that is run from our minor web interface to decrease/increase the volume on the moh files exponentially, called by + and - buttons. Hope i am not re-engineering something
16:54.09pkinvilleforgive even
16:54.09seanbright~book
16:54.10jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
16:54.13seanbrightpkinville: ^^^
16:54.20*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
16:54.20brodiempkinville: you could use the original Monitor() and record each leg separately
16:54.33Qwellbrodiem: I think he's wanting to do it realtime
16:54.36brodiemah
16:54.38pkinvilleyes, realtime
16:54.53pkinvilleactually, it would be a total of 4 calls in a loop that I would be dicking around with
16:55.08Qwellwell...
16:55.13QwellI suppose you could use JACK for that
16:55.21Qwellmaybe
16:58.11Kobazso, is it valid to have a pri span be both CPE and the clock source
16:59.21Kobaz[Aug 20 12:55:09] WARNING[14785]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown)
16:59.37Kobazthe span is up and active, but that happens
16:59.49bijitis its possible to transfer a call directly to a voicemail?
17:00.19Kobazyeap
17:00.49bijitwithout waiting for the message to comeup and then send.
17:00.56Kobazyeap
17:01.01Kobazshow application VoiceMail
17:01.12Kobazs    - Skip the playback of instructions for leaving a message to the calling party
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17:07.55Assidoh man anyone here ever played with multivoip from multitech
17:07.57DeeewayneKobaz: if you have green lights then CPE & clock source is not your problem
17:08.15*** join/#asterisk Levonk (n=lk@adsl-76-238-249-245.dsl.lsan03.sbcglobal.net)
17:08.33Assidand linked the outgoing to go through asterisk sip server
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17:14.24bijitmaybe I did not explain that good. If I get a call and I want to transfer to someone else vm. just dial and hungup not wait for an accion so it goes. If I just hung up the xfer doesn't work.
17:18.33KobazDeeewayne: yeah we fixed it
17:21.03DeeewayneKobaz: what was the problem ?
17:24.05KobazDeeewayne: i was using asterisk -rvv and not -rvvv
17:24.11KobazDeeewayne: i thought i wasn't getting any calls
17:24.37Assidi need some help with this multitech device
17:25.06KobazDeeewayne: for some reason, if i provided clock, and the nortel recieved it, the pri went flapping
17:26.46DeeewayneAssid: I know nothing about multivoip from multitech
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17:42.34jayteereferencing the value of ${CALLERID:-7} will give me the first 3 digits of a 10 digit number. Am I correct?
17:42.35jsolaresmeh why do weird problems follow me XD
17:44.45jsolareswhenever i call out one specific number (havent found another that gives the problem) using an e1 i get an isdn release cause 29, facility rejected, and a display message " NUMBERREQUIREDFORAOC-T"
17:45.02GhOnDiEi think you can also do ${CALLERID:{3}} HOWEVER IM NOT 100% ON THAT
17:45.04DuddsDoes anyone know how to get a dialtone from a TE122P card that's in T1 with E&M (with wink) signalling?
17:45.31DuddsI can dial no problem, but there is no initial tone played and it's probably going to confuse people.  :)
17:45.38Dudds(immediate is set to no)
17:46.19pascal_almasterisk just broke without any config changes... using odbc voicemail...  get mkdir '/var/spool/asterisk/voicemail/default/125/' failed: Permission denied (running asterisk as root)
17:46.43jayteeI'm just lookining for a way to strip the 1st 3 digits from 10 digit number inbound call if it matches my local area code.
17:47.13jsolaresi think that's just ${CALLERID:3}
17:47.26GhOnDiECOULD BE
17:47.28GhOnDiEwhoops
17:47.30jsolareslet me check on that one
17:47.44jsolaresthat
17:47.45GhOnDiEhowever that will strip it from every number
17:47.46jsolaresthat
17:47.56jsolaresbah stupid keyboard
17:48.04GhOnDiElol
17:48.07jsolaresthat's why i never leave home without paralles/vmware XD
17:48.25jayteeyeah, I only want to strip from the local area code so I'd have to use GotoIf and Set I think
17:48.36GhOnDiEyeah
17:48.44GhOnDiEsomething like that
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17:48.52jayteeok, back to massaging the code
17:48.59BlackvelNOW I made my decision. throwing away ISDN BRI and getting patton ISDN BRI gateway
17:51.17lowtekIs there some trick to allow multiple SIP registrations from the same phone (aastra/polycom) with 1.4?  Asterisk seems to get the peer/secrets confused.
17:52.09jsolaresit would be ${CALLERID:0:3} to get the area code and ${CALLERID:-7}/${CALLERID:3} to get the remaining 7
17:52.57GhOnDiEok
17:53.32jasonwootBug tracker says that in v1.6 manager.conf only gets parsed on load rather than on connection, can someone verify that for me?
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17:57.52gaetronikhi there
17:58.02gaetroniki've a quuestion about fax detection
17:58.11gaetroniki activate it in zapata.conf
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17:58.56gaetronikI'm using .call in outgoing spool dir to make call and i would detect which number are faxes?
17:59.02gaetronikhoy can i do that?
17:59.13kannanhello, what is the max allowed (if any ceiling) for Set (absolute timeout ) ? i want to do ine for 43 minutes
17:59.38kannan2500 seconds AbsoluteTimeout
18:00.55Qwelljasonwoot: yes
18:01.20Qwelljasonwoot: I'm pretty sure that is the case
18:01.38Qwell(it was always pretty silly for it to check on each connect attempt)
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18:07.21gaetronikno one for fax detect ?
18:09.19heedlyit's difficult.
18:09.41heedlythere is code in SVN to detech the appropriate tones, but I not gotten it to work with any of the existing fax detect apps.
18:09.45Qwellkannan: why 43 minutes? O.o
18:10.03Qwellheedly: that might be something d1mas would be interested in hearing about
18:10.46Qwell(the latter part - he wrote the former part)
18:11.00heedlyWhat is his contact info?
18:11.12Qwelldunno, he's on here occasionally though
18:14.44jasonwootQwell: I had to go back to 1.4 because of zap probs, but this manager.conf thing is just annoying enough to make me want to update again
18:14.55jasonwootI haven't found a level of verbosity yet that supresses it
18:14.59Qwelljasonwoot: 1.6 doesn't use Zaptel anymore
18:15.30Qwellneed to upgrade to DAHDI, as of 1.6.0-rc1 (I don't think it's been released - you'll want to use svn)
18:15.41jasonwootsry, I went back from .18 to .16 but haven't tried 1.6 yet
18:15.51Qwelloh
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18:16.43Pegasus_RPGhi there.
18:17.10Pegasus_RPGJust trying to get faxing working via Asterisk and BroadVoice (VoIP provider) First, is it possible>?
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18:18.21Pegasus_RPGI should also say I'm trying to fax from a PC faxmodem through a GrandStream HandyTone 386, to Asterisk, to BroadVoice, to the destination.
18:19.46Gershwindoes it fall back from 14.4 to 9600 to 1200?
18:20.04Gershwinor does the negotiation not last long enough to get fallback?
18:21.02Pegasus_RPGGershwin: if you're talking to me, negotiation is too short. I get "handle_response_invite: Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled." on the console
18:21.25Pegasus_RPGBut I don't know what it's referring to as the "Peer"
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18:22.32c4t3lwhat up playaz!?
18:22.35Blackvelpuzzled: do I have to understand this? I called my friend (pots analog, analog dect phone + mobile) and I had no echo issues
18:22.56Blackvelpuzzled: when I call myself on my mobile I have problems (same for my three pstn phones)
18:24.27puzzledBlackvel: try muting the phones otherwise you will get feedback/echo
18:25.05Blackvelnow I called I don't have.... when I call with my same setup and configuration at other times I do get that
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18:25.56Blackvelpuzzled: should be no problem because I put this snom 370 on the desktop, leave the room, shut the doors, go to different room, shut the doors and try to speak into the mobile phone
18:26.15BlackvelI could have understood that it is an issue with my isdn line/connection at home
18:26.25puzzledBlackvel: and is the snom muted?
18:26.28Blackvelbut getting problems on my mobile is far away from that
18:27.37Blackvelhandset is not muted, just lies on desk (detached). may THAT be a problem?
18:28.09Blackvelwell yeah
18:28.34Blackvelwhen i am in front of the desk of course there is feedback
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18:29.04puzzledBlackvel: yes cause sound from the speaker in the horn can feed back into the mic which causes echo
18:29.08Blackvelbut is not a problem when i leave the room
18:30.25Blackvelweird...working in 2nd room with my pstn mobile
18:30.49Blackvelso I can say..sometimes its working...sometimes not. octasic only uses about 11% of my 1gig cpu
18:31.39puzzleddunno if that is good or bad :)
18:32.13puzzledalso check /proc/interrupts. hfc-s based cards generate a ton of interrupts/sec and don't like to share an interrupt with something else
18:32.27Blackvelwell its not a duo core 2 with 2,5 mhz  or p4 with 3 gig
18:32.35Blackvelbut hey...its only a single call
18:33.05Blackvelthats fine. qozap on int 10, unique
18:33.23Blackvelmy friend was astonished
18:33.32Blackvelbecause he has that bad analog dect phone
18:35.47tzafrir_laptopBlackvel, to get a feeling of the number of interrupts per second: watch -n1 -d cat /proc/interrupts
18:35.50bradleyprice86Two questions. First, would it be possible to insert a second nic card in my asterisk server and have some of the voip traffic on my network use it instead of the first? Also, would it be worth doing?
18:36.12tzafrir_laptopand just look at the 1000-s digit
18:37.41tzafrir_laptopBlackvel, 11%? for how many active channels? Length of tail?
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18:39.16Blackveltzafrir_laptop: 1, 128ms
18:39.37Blackvelor is top lying?
18:39.52BlackvelI thought it could have been my cpu
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18:40.29Blackvelas octasic recommends for multiple channels an higher cpu (e.g 3 ghz)
18:40.50Blackvelbut there is no prerequisite like: must be 2 ghz
18:41.14Blackvelproblem ist just...its not really testable
18:41.30Blackvelworks and works not depending on unknown factor ;)
18:41.50Blackvelenough for me to return 305 euro junghanns duobri and get a real hardware with hardware echo
18:41.52Blackvelbut....
18:42.10Blackvelonly read good recommendations for patton's isdn bri gateways
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18:42.33Blackvelbut I wonder if octasic 128ms tail would be better than patton G168 20ms hw tail?
18:42.39Blackvelwhy is it only 20ms?
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18:45.57BlackvelI think I read that sangoma HWEC has 128ms
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18:51.58Knightfa|Is there a way to log into a sip phone and assign an ext to an agent. Something like HotSeating  if I sit down at a cube and dial a code and then my ext that phone is assigned that ext?
18:52.52*** join/#asterisk Chris-NB (n=chris@213162066148.public.t-mobile.at)
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18:55.20heedlyman the internet sucks today.
18:56.36n3hxsUFOs are sucking all the bandwidth.
19:00.06heedlymore like breaking routers
19:01.27Blackvelwhat are typical scenarios what echos exceed the tail of 128ms?
19:01.30Blackvelwhere
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19:05.33heedlyBlackvel: maybe when latency is longer than that?
19:07.52Blackveljust trying to find out why it works on the same phones sometimes and sometimes not ...
19:08.49Blackveli called my mobile phone multiple times (with echocancel=no, echocancel=yes) and it was horrible and not working. and now...suddenly it does. this is going to be of interest for the weekend.
19:09.14Blackvelhmm....lost TOO much time yet on this thing
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19:11.41Assidanyone here by chance worked with multivoip from multitech
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19:16.32Blackvelcu the weekend. closing time...
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19:35.04ShaunWingsay... I'm using IAX2 between two asterisk boxes and transfer=mediaonly, asterisk returns Error unable to transfer but there is two way sound... any idea please?
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19:37.54rwaitesimple, you didnt enable the donoterroronmediaonly=yes option
19:37.58aliverwhat's that page with the comman one-way audio issues?
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19:45.24Assidhi
19:45.37nny_2aliver: for nat issues?
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19:46.53nny_2~nat
19:46.54jbotsomebody said nat was Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
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19:48.35ShaunWingHas anyone used transfer=mediaonly iax2?
19:49.20aliverah thanks
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19:53.59JerJerdonthangupcallswhenasterisksegfaults=yes
19:54.15giovaniheh
19:54.50Knightfa|Is there a way to log into a sip phone and assign an ext to an agent. Something like HotSeating  if I sit down at a cube and dial a code and then my ext that phone is assigned that ext?
19:55.35aliverWastn't there a voip.info page about nat? google plays dumb
19:55.37exothermcis there anyway to have asterisk accept registrations without having a sip peer or user?
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19:57.28giovanialiver: google delivers
19:57.40giovani"asterisk hot seating"
19:57.45mvanbaakJerJer: you added that to asterisk ? cool !
19:57.57ShaunWingsay.....Any idea what this means? -- Channel 'IAX2/ECom-iax1-16386' unable to transfer
19:58.20JerJermvanbaak:  no need - a good media gateway can redirect that call   :)
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19:58.53JerJerShaunWing:   most likely you could not talk directly to that endpoint, so a native bridge could not take place
19:59.15JerJernothing serious
19:59.23mvanbaakShaunWing: are you sure all endpoints support G729 ?
20:00.29ShaunWingsay... I'm using IAX2 between two asterisk boxes and transfer=mediaonly, asterisk returns Error unable to transfer but there is two way sound... any idea please?
20:00.48ShaunWingHas anyone used transfer=mediaonly iax2?
20:01.33*** join/#asterisk lanning (n=lanning@66.151.128.195)
20:01.50JerJeris it error or warning ?
20:01.55JerJeror notice
20:02.04*** join/#asterisk Levonk (n=lk@adsl-76-227-117-65.dsl.lsan03.sbcglobal.net)
20:02.28ShaunWingIt appears as this in CLI:-- Channel 'IAX2/ECom-iax1-16386' unable to transfer
20:02.36ShaunWingboth suppor g729
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20:02.54JerJeri think its more informational that the media could not get transfered either due to codec or network sitautions
20:03.08ShaunWingCalling from Linksys PAP2T using G729
20:03.30JerJerwhat on the far end?
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20:04.57ShaunWingAny idea if I need a G729 license loaded in Asterisk as both end points are G729
20:05.10ShaunWingare using G729
20:05.16JerJerif you use dial modifiers
20:05.18ShaunWingThere is no codec translation
20:06.17ShaunWingI dial out with the Linksys which registers to the Asterisk
20:06.29JerJerdo you use T, t or r or M or any other dial modifier ?
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20:07.13ShaunWingThe Asterisk then dials the 2nd Asterisk and it in turn dials a 3rd Asterisk which eventually terminates teh calls
20:07.55ShaunWingI get the message on the first 2 asterisk servers, the 3 rd is that of the service provider
20:08.56*** join/#asterisk fl1p (n=fl1p@ip-88-153-12-37.hsi.ish.de)
20:09.07ShaunWingnone
20:09.13exothermcexit
20:09.18ShaunWingexten => _20.,1,Dial(${OUTTRUNKEComIax1}/27${EXTEN:2},60,)
20:09.18*** part/#asterisk exothermc (n=miles@74.85.89.146)
20:09.20ShaunWingI dial as above
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20:10.58bradleyprice86Is it possible to have to network cards being used by my asterisk server for better performance?
20:12.02aliverIf I can get audio out of my asterisk server to my internet SIP provider, but folks in the "real world" can't be heard responding back to the SIP caller. Where is the most likely issue?
20:12.13*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
20:12.13Qwellnat/firewall
20:12.26JerJerbradleyprice86:  you mean like SSL accelerators ?   TCP engine ?
20:12.43aliverDoes the SIP peer I'm talking to try to connect back to me on RTP UDP ports?
20:12.58*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
20:13.16aliver'cause I'm damn near-certain that I put in those rules.
20:13.28JerJeraliver:  see rtp.conf
20:13.29aliverI guess I'll have to test that somehow.
20:13.36aliverJerJer, hmm, okay.
20:13.50JerJerShaunWing:  then i would look at the network -  is there a firewall and/or nat  ?
20:14.17Arck-FRhi, someone has already test asterisk 1.2 with Sipp on RHLE 5 ?
20:14.26ShaunWingThere are 2 firewalls and nats
20:14.26aliverSo, those ports described in rtp.conf are ports that asterisk GETS data on or does it TRANSMIT data on them. ie.. listen or send?
20:14.57[TK]D-Fenderaliver: 1 port send, 1 port receive
20:15.23aliverI'm assuming that the SIP server tells asterisk when and where to open the UDP recieve port via the 5060 control port once a call is made, correct?
20:15.35[TK]D-Fenderaliver: Yes
20:15.43ShaunWingPerhaps this is a clue. Sometimes I place a call from the Linksys PAP2 there is no sound. But the call right after has even to the same number?
20:16.03aliverhmm.
20:16.04*** part/#asterisk airjump (n=chatzill@p508AD289.dip.t-dialin.net)
20:16.07bradleyprice86JerJer: Well, I am having a problem where my audio recordings have extremely bad quality when streamed over my freepbx web interface. I wasn't sure if it might help if I installed a second network card. Or, if that would even be a bottleneck. I have around 100 users on the system with approximately 15 concurrent connections at anytime.
20:16.40[TK]D-Fenderbradleyprice86: no, like I told you before its the fault of the LISTENER'S playback codec
20:17.29bradleyprice86[TK]D-Fender: I did check the codec's earlier and tried multiple machines with different codecs, but all resulted in the same error.
20:17.50puzzledmaybe it's one of those wav versus wav49 problems
20:18.12nny_2so i am looking to have global Dial options (wW,t,k) etc. Should i just do a Dial(SIP/10,,${doptions}) and set the variable globally or is there a global Dial option setting somewhere i am unaware of?
20:18.27[TK]D-Fendernny_2: Former
20:18.35nny_2ok cool no problems there
20:18.53[TK]D-Fendernny_2: And unless you're working with seriously chump crappy phones, you don't want "tT"
20:19.15bradleyprice86[TK]D-Fender: Also, maybe 1 out of every 10 times the recording actually sounds clear.
20:19.36nny_2[TK]D-Fender: indeed. was just an example
20:19.44[TK]D-Fenderbrad1 out of 10 recordings, or 1 out of 10 attempts to listen to ANY?
20:20.33bradleyprice86[TK]D-Fender: Sorry, 1 out of every 10 attempts.
20:21.10[TK]D-Fenderbradleyprice86: Then its something intermittent like network jitter, congestion, etc
20:26.25ShaunWingI have two asterisk servers and I've created an IAX2 config on both as below. The one server shows host as OK with <20ms and the otreh shows it as unreachable? Please help.
20:26.58*** join/#asterisk edgimar (n=edgimar@dyndsl-080-228-184-063.ewe-ip-backbone.de)
20:26.59ShaunWingWhy does a perfectly fine iax2 host becomes UNREACHABLE?
20:29.20Assiderr. i have  'm' as a parameter in my dial.. but i am sill getting the ringing instead of the music
20:29.24[TK]D-FenderShaunWing: Networkting problem.
20:29.32edgimarCan anyone direct me to a nice howto on setting up asterisk as a personal SIP 'provider', where I can register with a SIP client, and recieve calls on the registered client?
20:29.33[TK]D-FenderAssid: PASTEBIN
20:29.54[TK]D-Fender~book
20:29.54jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
20:29.58[TK]D-Fenderedgimar: ^^^^^^^^^
20:30.37edgimar[TK]D-Fender: thanks.  I'll take a look.
20:31.26Assid[TK]D-Fender: http://assid.pastebin.com/da0fea1f -- i tried with and without 'r' as well
20:32.02[TK]D-FenderAssid: Says it starts...
20:32.50Assidyeah but it doesnt i still hear ringing
20:33.04Assidnot the music
20:33.35[TK]D-FenderAssid: First verify that your MoH class is functional.  Then if that's fine, do an "answer" first
20:33.48gaetronik[Aug 20 16:20:09] NOTICE[6365] chan_zap.c: Fax detected, but no fax extension
20:33.48*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
20:34.00gaetronikthis line means that fax detection is activated, right?
20:34.06[TK]D-Fendergaetronik: Yes
20:34.29Assidyes i had tested it
20:34.34Assidthe moh that is
20:34.41Assidmade an extension to play moh back to me
20:34.45gaetronikwhy this not appear for each call to a fax
20:35.27[TK]D-Fendergaetronik: could be * isn't listening for anything on those calls.
20:35.56gaetronikmiss a answer?
20:36.07ShaunWingHow can I diagnose thenetwork problem... they can ping each other and there is no firewall blocking..
20:36.47[TK]D-Fendergaetronik: Good reason.
20:36.47[TK]D-FenderShaunWing: Sit and watch the traffic
20:36.47*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)
20:36.49ShaunWingwith what?
20:37.12[TK]D-FenderShaunWing: tcpdump, wireshark, etc
20:37.29Assid[TK]D-Fender: if i exclusively call musiconhold() it works
20:37.46[TK]D-FenderAssid: pastebin, and then do as I advised
20:38.08ShaunWingtx. I'm used to wireshark for windwos but what can I use for linux?
20:38.23[TK]D-FenderShaunWing: WIRESHARK
20:38.41gaetronik[TK]D-Fender, don't even work with an answer
20:38.47Assid[TK]D-Fender: http://assid.pastebin.com/d320ed1fb
20:38.54gaetronikwhat could be?
20:38.55aliverWhen I scan my asterisk box from the internet I see the UDP ports open 4520 and 2727 and lsof tells me asterisk is listening on them. nmap shows the ports are open and the adjacent ports are closed. I _really_ think I'm forwarding UDP properly across the firewall. What else could cause one-way audio?
20:39.22ShaunWingG8
20:39.23*** part/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net)
20:39.24aliverWould it help to see a pastebin of my sip debug when a one-way audio call comes in?
20:39.49[TK]D-FenderAssid: And that works?  Now try what I suggested
20:40.13Assidanswer first?
20:40.18[TK]D-Fenderaliver: Do you ask your doctor is examining you might help him find out what's wrong?
20:40.23*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
20:40.23[TK]D-FenderAssid: Yes.
20:42.11ShaunWingsay for IAX2 is this enough in iptables? -A RH-Firewall-1-INPUT -m state --state NEW -m udp -p udp --dport 4569 -j ACCEPT
20:42.29ShaunWingie. only udp 4569
20:42.50alivergood point.
20:42.54gaetronik[TK]D-Fender, why asterisk would not be listening the call?
20:42.54[netman]--state NEW with udp?
20:42.59aliverI'll get together a pastebin
20:43.03[TK]D-FenderShaunWing: Why don't you try loosening things up if you want to test.
20:43.14[TK]D-Fender[netman]: SHHH!!!
20:43.22[TK]D-Fender[netman]: Leave the OSI challenged alone!
20:43.25*** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com)
20:43.35tobiasis it OK to use asterisk 1.4 with zaptel 1.2 ?
20:43.42[TK]D-Fendergaetronik: Not being in an IVR, etc.
20:43.46[TK]D-Fendertobias: NO
20:43.59tobiasdang.  thanks.
20:44.21Assid[TK]D-Fender: yeah apparently i need to answer it first.. like the start of an ivr
20:44.32gaetronik[TK]D-Fender, i don't understand
20:44.45ShaunWingProblem is - its not my server and they firewall controlls are out of my hands..I have to log  a request  and therefore need to know what to allow for 4569
20:44.52[TK]D-Fendergaetronik: Somewhere that * is litening for tones.
20:44.53Assid[TK]D-Fender: and i need to set the rm again for the extension that needs to be called
20:45.58Assidone thing tho.. the audio keeps cutting out
20:46.33gaetronikthe way is answer, dial(Zap/g0/number)
20:47.28gaetronikis this not the way in what asterisk should listen?
20:48.16[TK]D-Fendergaetronik: Does that look like an IVR to you?
20:48.22gaetronikno
20:48.33gaetronikit why i'm wondering
20:48.38*** part/#asterisk nny_2 (n=Scott_My@64.203.244.146)
20:48.39gaetroniki mak the call
20:48.47gaetroniki don't receive the call
20:48.50*** join/#asterisk jameswf-home (n=james@jfinstrom.broker.freenet6.net)
20:48.52[TK]D-Fendergaetronik: Go look at how the calls are different
20:48.52gaetronikto be clear
20:49.58gaetronikwhat happened is that the call which leave log was not originated by me
20:50.15gaetronikso i don't know if there are inbound or outbound
20:52.28*** join/#asterisk edon (n=edon@albalug/edon)
20:56.50*** join/#asterisk asteriskmonkey (n=asterisk@69.77.169.14)
20:57.19asteriskmonkeyanyone got a link or bash script to take care of all the -in and -out.wav files monitor creates using soxmix?
20:57.50asteriskmonkey~centos52bug
20:57.51jbot[~centos52bug] There is a bug compiling Zaptel  up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver  in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889
20:59.28*** join/#asterisk ddunavant (n=David@75.145.240.14)
20:59.56*** join/#asterisk flush (n=SYN_SENT@ip216-239-89-63.vif.net)
21:00.49aliverCan someone look at this pastebin and see if you see any reason why I have half-duplex audio issues still ? I'm pretty sure UDP ports are being let in.
21:00.51aliverhttp://pastebin.com/d2c4877b0
21:02.01gaetronikasteriskmonkey, it's quite difficult since you could not ne sure it's over
21:02.05gaetronikor use inotif
21:02.07gaetroniky
21:02.16gaetronikwith closewrite
21:02.28*** join/#asterisk jameswf-home (n=james@jfinstrom.broker.freenet6.net)
21:03.23*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
21:03.26gaetronikis this possible to detect outgoing fax
21:03.34gaetroniki doubt it with my experiment
21:03.57aliverI suppose there are no codecs that multiplex the audio onto one port eh? I'm trying to find an easy way to weasel out of this one-way audio problem.
21:04.13*** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com)
21:04.16alivermy sip debug is here with my sip.conf and extensions.conf:    http://pastebin.com/d2c4877b0
21:06.21*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
21:06.30*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
21:09.21*** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net)
21:09.22grandpapadotwrote a really fast and simple international number parsing and rate table lookup script if anyone wants it ...
21:09.33grandpapadot.. written in php
21:10.14[netman]grandpapadot: I would like to see your script
21:10.42*** part/#asterisk asteriskmonkey (n=asterisk@69.77.169.14)
21:11.33mchouIf the inbound trunk is SIP or IAX, is it still necessary to do Wait(1) prior to calling the function CALLERID()?
21:12.28*** join/#asterisk fogo (n=Paul@rs-69-169-132-200-0003.broadweave.net)
21:12.31mchouor does Wait() only apply to analog/PRI as far as CallerID is concerned??
21:13.07JerJerthat is  a PSTN thing
21:13.31mchouJerJer: ok.  thx for confirming
21:15.02grandpapadot[netman]: http://pastebin.com/d54215c74
21:15.32grandpapadotThis script is 100% accurate as it 'backs in' to the actual country code instead of some crazy parsing mechanism ...
21:15.41grandpapadotAnd it's surprisingly fast
21:15.53[netman]thx grandpapadot
21:17.02grandpapadotnp
21:17.09*** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net)
21:19.23scooby2any format that is best for greeting messages or is mp3 fine?
21:19.45russellbthe same format that the calls will be using is best
21:19.47[netman]alaw?
21:19.56[TK]D-Fenderscooby2: Should have them in the codec being used
21:20.02russellbmp3 is one of the worst options ...
21:21.24scooby2russellb: I kind of figured as much
21:25.24*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
21:27.51gaetronikno one have a good experience of outgoing fax detection
21:27.59*** join/#asterisk razu (n=razu@razu.data.ee)
21:29.25[TK]D-Fendergaetronik: What are you expecting to do?
21:29.44gaetronikto detect if a fax answer
21:29.56[TK]D-Fendergaetronik: And then?
21:30.06gaetronikand mark the number as fax in a db in order to stop calling him
21:30.17gaetroniks/him/it
21:30.32[TK]D-Fendergaetronik: Don't think that'll work...
21:31.14gaetronik[TK]D-Fender, why?
21:31.40[TK]D-Fendergaetronik: Because I * isn't listening to bridged calls
21:32.18gaetronikNo hay usbekistan patch to do that?
21:32.23[TK]D-Fendergaetronik: If I'm talking to you on the phone and there is a fax machine turned up loud next to me do you think * is going to dump my over it?  no.
21:32.36[TK]D-Fender...
21:33.16gaetronik[TK]D-Fender, there is a difference between your voice and the fax tone and only the fax tone
21:33.16scooby2whats the best format for hold music then for inbound callers via telco?
21:33.49grandpapadotscooby2: If from TDM use ulaw, if from an ITSP use whatever code you're receiving the call with
21:34.11grandpapadotIf you must transcode, sln would likely require the least work by asterisk
21:34.25grandpapadotcode=codec
21:34.35scooby2thank you
21:34.55gaetronik[TK]D-Fender, it looks like i'm screwed
21:35.02scooby2goof ball before me had everything in mp3
21:35.07[TK]D-Fendergaetronik: * isn't listening while 2 people are talking
21:35.16grandpapadotWe use WAV and transcode the hell out of it, works great, of course we have some serious cpu power behind our servers, quade core, 4gb ram, etc
21:35.36gaetronik[TK]D-Fender, and for auto dialup?
21:35.49gaetronikwith .call file
21:35.51[TK]D-Fendergaetronik: in an IVR <-
21:35.52grandpapadotWe use mp3 for moh and let madplay produce 8khz unsigned audio
21:36.39grandpapadotWow, I'd like to back in time 3 years and read what I just wrote, I wouldn't know what the hell I was talking about, lol
21:36.59grandpapadotOf course, I don't know what the hell I'm talking about now  (For TK, since he's thinking it)
21:37.05gaetronikwhy there is option for fax detect in outgoing call?
21:37.22*** join/#asterisk andylockran (n=andylock@genesis.zrmt.com)
21:37.27andylockranhey guys
21:37.58andylockranI'm getting a softly spoken glasweigian to record a new voice for our asterisk PBX.  Do you think many people would be interested in using the same voice?
21:42.20*** join/#asterisk adrianXXX (i=xpl@84.126.207.62.dyn.user.ono.com)
21:42.23Assidhrmm musiconhold keeps disappearing for me
21:42.24Assidstrange
21:42.36adrianXXXhola alguien abla espaniol por aqui /
21:42.46gaetronikadrianXXX, si
21:42.49mvanbaakyou have to pay the commission for moh ;)
21:43.07*** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net)
21:43.22jsolaresgaetronik are you planning on using the fax thing on a kind of dialer? and if it's not fax send it to a person?
21:43.55gaetronikjsolares, kind of
21:44.10razuanyone have experience configuring Cisco 79XX series phone with backupped solution ?
21:44.18jsolaresi haven't done it with fax per se, but what i'm doing on a dialer is calling and if it's an answering machine leave a message and call them again and again and again :)
21:45.21gaetronikjsolares, and how you do that
21:45.22razuactually 79X1 phones ... I configured backup proxy in config file, but when one is dead then outgoing calls don't reach backup asterisk :(
21:45.23Assidokay this is causing me grief.. how do i set this such as the moh shouldnt just disappear mid way
21:45.42heedlyjsolares: get asterisk SVN and hax code.
21:45.52heedlyfax tones are different for incoming and outgoing.
21:46.03heedlythe current 1.4 stuffs only supports outgoing.
21:46.34Assidi just cant hear the music anymore
21:46.43jsolaresi'm using AMD for detecting the voicemail, you'd need something like nvfaxdetect from what i remember
21:47.00heedlymaybe for incoming.
21:47.06jsolaresbut if the tones are different... go hax the code
21:47.20heedlycodes already there, just none of the apps use it.
21:47.43heedlyhow do you find AMD goes?
21:47.56heedlyI have people who use nvbackgrounddetect to get around waiting for voice.
21:48.27gaetronikit works?
21:48.39heedlynot for faxes, but everything else.
21:48.43heedlyfaxes are low on my list.
21:48.48heedlygimme moneys and I'll get it working.
21:49.05mvanbaakfaxes should die
21:49.26gaetronikmvanbaak, yes but there are still alived
21:49.30gaetroniklike minitel
21:49.48mvanbaakI'm ignoring faxes
21:49.57mvanbaakand for now it works out fine for me
21:50.14jsolareshehe
21:50.38gaetronikmvanbaak, but i need to know if number are faxes or no
21:50.40mvanbaak"can you fax that?", "no, I'll mail you a PDF", "ok"
21:51.01jsolareswell what if they are faxes today, but not tomorrow?
21:51.19Assidafter 5 seconds.. moh just stops
21:51.23heedlyyou'll know when you redial :)
21:51.27*** join/#asterisk jameswf-home (n=james@jfinstrom.broker.freenet6.net)
21:51.33heedlyor get bettar data...
21:51.38*** join/#asterisk tkbeat (n=tk@p54B947BF.dip.t-dialin.net)
21:51.47gaetronik~sipbroker
21:51.55gaetronik~sip
21:51.55jbot[sip] http://www.cs.columbia.edu/sip/  X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/  Session Initiation Protocol (see RFC 3261)
21:52.05gaetronik~siptrunk
21:52.05jbotNo such thing, my friend.. Like too much salty plum soda.
21:52.09heedlyAssid: does the provider use silence suppresion?
21:52.12gaetronik~siptrunks
21:52.24mvanbaakjsolares: eh ?
21:52.27heedlyI just had the same thing with MOH and that was one of the causes.
21:52.36jsolaresthat was to gaetronik
21:52.37*** part/#asterisk knarfly (n=vladt@c-75-74-155-198.hsd1.fl.comcast.net)
21:53.15gaetronikjsolares, no change often
21:53.19jsolaresif heedly is correct in that the tones a fax makes when it's in receive mode rather than trying to send you one, you'll need to code it in
21:53.20Assidheedly: nah its between me and asterisk box
21:53.51Assidthe asterisk box tries to call a remote multitech gateway.. but no one answerting the call there yet
21:54.36jsolaresotherwise have the dial send the call on answer to a macro, and do the fax detection on that macro, if you end inside the fax extension then well it was a fax machine
21:55.47jsolaressomething like Dial(Zap/g0/3423423, 120, M(detectfax^param1^param2))
21:55.54gaetronikjsolares, i didn't understand the last
21:56.19jsolareswell if you run nvdetectfax it'll send you to extension fax or talk depending on what it detects
21:56.39heedlylulz
21:56.42heedlythat won't work.
21:56.49gaetronikjsolares, should it works in outgoing
21:57.01*** join/#asterisk kd8ikt (n=kd8ikt@we.trekbeyond.org)
21:57.33adrianXXXalgun espaniol ?
21:57.44jsolaresonly if whatever app you're using to detect faxes can detect a remote fax machine expecting a fax
21:58.03heedlythe spandsp code does the tone detection not the app.
21:58.04*** join/#asterisk km2 (n=x@208.80.69.72)
21:58.59aliverAnything other than NAT/firewall issues that can cause one-way audio?
21:59.23heedlyAssid: maybe a restart?
21:59.26gaetronikaliver, routing issue
21:59.31heedlyor rather full stop then restart.
21:59.35heedlythat was the other problem :)
21:59.48Assidheedly: it randonly works the first time after a restart to asterisk
21:59.51gaetronikwill commit suicide due to fax detection
22:00.24alivergaetronik Okay, but if I can, say ssh into a machine from the outside via it's WAN VIP, that would mean I don't have an asymetric route.
22:00.38jsolaresso spandsp has the code to detect the remote fax machine on receive? but nvfaxdetect is not coded to use it?
22:01.21Assidheedly: i even went out to make ulaw versions of the file so it doesnt need to transcode
22:01.21gaetronikaliver, right
22:01.28heedlyjsolares: depends on the version of asterisks you use.
22:01.36gaetronikaliver, i añlways debug those issue with tcpdump and wireshark
22:01.48heedlyjsolares: SVN has some code, but I have been unsuccefful in getting it to work. But I didn't spend much time with it.
22:01.55jsolaresah
22:01.57aliverswitching codecs wouldn't make a damn bit of difference on one-way audio issues, right?
22:02.01heedlywas fairly easy to hack up 1.4 to get it working.
22:02.17aliverlike, if I use g729 instead of ulaw would that help possibly?
22:05.08*** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com)
22:05.15Assidweird
22:05.24Assidti randomly works. and randomly doesnt
22:05.26mchouI'm a bit confused about how cdr works.  In my dialplan for outbound I call Playback() (which lasts about 3 seconds) before I actually call Dial().  In the cdr billsec field it "charges" for the 3 second recording (which I clearly dont want to count as charged).  How do I go about correcting this?
22:09.16*** join/#asterisk Levonk (n=lk@adsl-75-62-132-77.dsl.lsan03.sbcglobal.net)
22:09.44codefreeze-lapmchou: the cdr record itself is co-created with the channel; If the channel of the call-starter has some playback before the dial, then it might end up delaying the start time... the answer will be delayed 3 sec. the billsec is the end time minus the answer time, the duration time is the end time minus the start time...
22:09.47grandpapadotDoes asterisk understand daylight savings time in voicemail somehow with zonemessages?
22:11.17grandpapadotAhh... rtfm
22:11.20grandpapadotworks every time
22:11.25codefreeze-lapuh, mchou: I meant, "then it might end up delaying the answer time..."
22:12.06mchoucodefreeze-lap: yeah, I understand that.  but I'm not sure where Answer() is started (or implicitly called)
22:12.53*** join/#asterisk korihor (n=korihor@201.211.168.130)
22:12.58codefreeze-lapThe dial app sets the answer time on the CDR when the dial attempt succeeds
22:12.59mchoucodefreeze-lap: like Answer seems to get called b4 playback (for the purposes of CDR)
22:13.54codefreeze-lapmchou: hmmm. I'd check to see if playback calls Answer() or some such. There might be an option to turn it off.
22:14.23mchoucodefreeze-lap: I basically want to set the "answer" breakpoint once the callee has answered.  Not before. :)
22:16.09codefreeze-lapthere it is, see "core show application playback" -- you use the noanswer option, mchou
22:16.20mchoucodefreeze-lap: wow, lol
22:16.35mchoucodefreeze-lap: rtfm ftw
22:16.57codefreeze-lapmchou:  ha, comes under the "read the friendly manual" heading, eh?
22:17.09mchouyeah.
22:18.02outtoluncdid you use the noanswer option?
22:18.12mchououttolunc: nope :)
22:18.12outtoluncoops.. day late
22:18.25*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
22:18.31codefreeze-lapif all else fails....
22:18.43outtoluncresetcdr <G>
22:18.50outtoluncruns
22:19.06codefreeze-lapouttolunc: working on those issues now.... ugh.
22:19.23mchoureset CDR doesnt work if the user hangs up during playback()
22:21.16outtoluncwell do a exten => h,1, and test the playbackstatus
22:22.01codefreeze-lapuh, changing the cdr in the hangup exten is not an option any more
22:22.11outtoluncbummer <G>
22:22.46codefreeze-lapI'm contemplating that entire side of life as I go thru my CDR bug list right now.
22:23.18codefreeze-lapIt's all been an issue of what you want, and what you are willing to pay for it.
22:24.11codefreeze-lapIt's like watching "A Series of Unfortunate Incidents" all over again...
22:24.31codefreeze-lapturns off riddle mode
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22:29.50mgromanHi, When I call my Asterisk box, it seems like there is about a 4 or 5 second delay between hitting connect on the sip phone and when the dialplan starts executing, any ideas what might cause this?
22:30.59tvirus~itsp-usa
22:33.46gaetronik~itsp
22:33.47jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
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23:00.12jayteegod that was frustrating as hell
23:00.17jayteebut I finally cracked it
23:03.55*** join/#asterisk troy|home (n=troy@142.166.111.200)
23:05.06troy|homeI have been asked to setup an asterisk server for my office. I need a tutorial or very basic guide to get asterisk to work with our 1800 number and call forward to cell phones... any suggestions?
23:05.24jaytee~book
23:05.24jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
23:05.45troy|homesweet... thanks. will check it out
23:05.47jayteealso checkout www.voip-info.org or www.asteriskguru.com
23:06.38jayteetroy|home, there's a link there for the book to download the PDF and save it.
23:07.02troy|homedownloading as we speak (err.. type) ;)
23:07.09jeevcoughs
23:08.11troy|homeif I call forward to personal cell phones does that use up one of the lines going to the asterisk box?
23:08.30troy|homeor is it a direct connection between the caller and callee?
23:09.55*** join/#asterisk ManxPower (n=manxpowe@adsl-222-29-13.msy.bellsouth.net)
23:10.41heedlytroy|home: depends.
23:11.01heedlysomething like a zaptel line would tie up two channels I believe.
23:11.15troy|homeI thought that might be the answer... :P
23:11.24heedlywhere as something like SIP could go from caller to callee
23:11.38troy|homethats what I was afraid of
23:12.03troy|homeoh well, I guess it would be that way with any other pbx anyway
23:13.39ManxPowerWith careful planning, the Custom Calling Feature "Conference, Drop, and Transfer" with an analog lines, and clever dialplan stuff you could get calls transfered off the system and off the lines.
23:15.05GhOnDiEthats only with analog lines
23:15.09drmessanoI need some partners for a new business venture
23:15.11GhOnDiEif you had a digital line
23:15.16troy|homewe are just starting out on asterisk, so I think that might be a little in the future. I will just tell them we need 2x the amount of lines planned for.
23:15.17drmessanoHosted PBX: Old and busted
23:15.18troy|homeaha!
23:15.22GhOnDiEyou can hold a couple of calls on the 1 trunk
23:15.22troy|homedigital you say?
23:15.23ManxPowerGhOnDiE: it would also work with a CT-1
23:15.26drmessanoHosted Phones: The New hotness!
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23:15.39drmessanojameswf: I have been looking for you
23:15.46ManxPowerIt would not work with PRI, of course, nor E&M
23:15.57GhOnDiEno
23:16.08*** join/#asterisk drwelby (n=mpfister@mail.enplan.com)
23:16.18ManxPowerBut is you are silly enough to have analog lines and too poor to upgrade to a PRI.......
23:16.24GhOnDiEany ideas how i can get asterisk to dial my page group number and then playback an audio file?
23:16.32GhOnDiEi just cant get it to do what i want
23:16.44drmessanoManxPower: So a 48 lines FXO card is silly?
23:16.47ManxPowerGhOnDiE: .call ciles
23:16.58ManxPowerdrmessano: stop, I'm laughing!
23:17.11drmessanohides his TDM9600P
23:17.41ManxPowerdrmessano: Where "silly" is "one of the stupidest things I've heard of in a long time".  I'm practicing my interview skills.  I've applied to a couple of places
23:18.02drmessanoNice
23:18.11GhOnDiEManxPower, cheers not sure why i did not think of that
23:18.18ManxPowerGhOnDiE: .call files there is a sample and maybe some docs in /path/to/src/asterisk/doc
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23:18.42drmessanoJust remember to learn the bullshit bingo words
23:18.49drmessano"Strategic initiative"
23:18.50drmessanoetc
23:18.58ManxPowerGhOnDiE: Anytime you say "Asterisk make a call" also think ".call file"
23:19.04GhOnDiEyeah
23:19.10GhOnDiEmust be because its late here
23:19.16GhOnDiEmy brain is not functioning
23:19.17GhOnDiElol
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23:22.53drwelbyAny opinions on an small office IP PBX? 20 users, 6 analog lines out. All VOIP traffic is on the office LAN only.
23:23.12drmessanoYeah, sounds like a good idea
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23:23.39GhOnDiEsounds good
23:23.48GhOnDiEhave it on its own lan may be a good idea
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23:23.56ManxPowerdrwelby: ask local providers about a 6 channel PRI, you might be surprised how close in cost it might be to 6 analog lines
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23:25.43ManxPowerI had forgotten how awesome it is to be on a real "unlimited" internet service. 8-)
23:27.18drmessanolol
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23:28.48ManxPowerdrmessano: About a year ago I accidentally left BitTorrent running when I was on my Verizon EVDO internet service.
23:29.34*** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com)
23:29.44grandpapadotWe pay $23/channel for PRIs with level 3...
23:29.56ManxPowerGot a letter that said "You are using too much bandwidth so you must be doing something against our AUP.  Your account is terminated as of (date 2 days in the future).  There is no appeal."
23:30.26scooby2did they charge you the ETF?
23:30.47ManxPowerscooby2:  I don't think so.
23:30.56grandpapadothmm.. good question..
23:31.22deeperrorManxPower, who was that with
23:31.33ManxPowerA few months ago I got a check from Verison for about $250, apparently Verizon lost a class action lawsuit over the exact issue I had.
23:31.46adrianXXXAug 20 19:31:36 WARNING[5623]: file.c:512 ast_openstream_full: File invalid does not exist in any format
23:31.46adrianXXXAug 20 19:31:36 WARNING[5623]: file.c:824 ast_streamfile: Unable to open invalid (format ulaw): No such file or directory
23:31.46adrianXXXAug 20 19:31:36 WARNING[5623]: app_playback.c:133 playback_exec: ast_streamfile failed on SIP/w1-082562b0 for invalid
23:32.08ManxPoweradrianXXX: that means exactly what it says
23:32.26adrianXXXI am now on asterisk
23:32.55adrianXXXwhat means ?
23:33.17ManxPoweradrianXXX: It means the file you specified does not exist in any format that Asterisk can read.
23:33.37deeperrorfile not found
23:33.54ManxPowerdeeperror: don't put too much stock in that message
23:33.57adrianXXXexten => s,n,Background(mens-demo-congrats)         ; "play welcome msg"
23:34.18ManxPoweradrianXXX: and you have some files in formats Asterisk can read in /usr/lib/asterisk/sounds ?
23:34.21deeperroror in some bad format ha
23:34.25adrianXXXexten => s,n,Background(file.wav)         ; "play welcome msg" ??
23:34.26adrianXXX?
23:34.34deeperrorremove the .wav?
23:34.45filethe message says a file named "invalid" did not exist... not mens-demo-congrats
23:34.47ManxPoweradrianXXX: NEVER specify the file extension in Playback or Backgound
23:34.59adrianXXXok
23:35.34adrianXXXsome one speak spanish ?
23:35.47adrianXXXbicause my english it a little bat ..:(
23:35.49ManxPowerfile: maybe quotes could be put around the file name in the message?
23:36.36drwelbyManxPower: Really? I always assumed that we were too small to even think about PRI
23:38.11JTso what is generally the minimum number of channels on a PRI in north america?
23:38.13jblackdrwelby: You ery well might be.  If you have reasonably good bandwidth, you can look at traffic shaping and two voip providers.
23:38.13drwelbyBut anyway, IP PBX hardware. Does the Trixbox evoke vomiting from anybody? Aastralink? Fuzion?
23:38.29*** part/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net)
23:39.02jblackJT: You're usually looking at floor costs of $450 - $600 a month.
23:39.03drwelbyWe're just a small office and doing mostly local calls.
23:39.18ManxPowerdrwelby: It all depends on where you are located.
23:39.26JTjblack: ouch
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23:39.48jblackdrwelby: Ok, well, you've got three choices. Straight voip, a PRI w/ a dedicated card, or two or three FXO ports for analog lines.
23:39.53JTpris are extremely cheap here in australia
23:40.06ManxPowerIt all depends on your market.
23:40.09JTyou can get a 10 channel for AUD$165/mo
23:40.17JTwhich is probably about USD$150
23:40.24ManxPowerIn NYC you could get a PRI far, far cheaper than in Rainbow City, AL
23:40.50jblackThe market definitely matters. Large Cities tend to be cheaper. Medium sized cities, suburbia, and rurual, you can truly get fucked.
23:40.55drwelbyNorthern California, though I'm perfectly happy sticking with an FXO card
23:40.57JTcosts are pretty much flat here regardless of location. you're either in an area with service available, or you're not
23:41.38drwelbyRural northern California, that is. Not Bay Area or Sacramento.
23:41.40jblackJT: Ohh, poor you. You don't have a competitive market (where competitive can be define as "if you don't like us, then move".
23:41.51ManxPowerJT: In the USA each state has it's own telecom regulatory body for stuff that does not go inter-state.  Interstate is regulated by the FCC (feds_
23:41.55adrianXXXalguien abla espaniol por ay ?
23:41.58adrianXXXalguien abla espaniol por ay ?
23:42.02jblackdrwelby: Then a digium card with 2 or 3 fxo ports might be exactly what you want.
23:42.38*** part/#asterisk adrianXXX (i=xpl@84.126.207.62.dyn.user.ono.com)
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23:42.40JTjblack: err it's extemely competitive
23:42.56jblackjt: I was being sarcastic about my country. =)
23:42.57JTjblack: otherwise it'd be much more expensive
23:42.59JTah
23:43.11ManxPowerCost for the EXACT same service from the SAME company, Cost for a BRI in Louisiana: $100/month, in Tennessee: $35/month
23:43.21JTif i chose to go with the incumbent telco, Telstra, it's about AUD$380/mo for 10 channels iirc
23:43.37drwelbyjblack: We have an AA50 but find the little crackerjack processor a little underwhelming
23:43.57JTthe cheapest prices are terminated directly to other telco muxes
23:44.03ManxPowerThe difference is that in Tennessee BellSouth (ILEC) had not found the right people to bribe on the PUC/PSC
23:44.11jblackdrwelby: I'd do initial rollout is voip if you have reasonable bandwdith. Get your initial plant going, make sure things "feel" right. Then, once you know you like where you're doing, consider getting a fxo card (digium sells cards that can do from 1 to 4 fxos based on modules) and replacing the voip.
23:44.37JTbut they can use Telstra's PRI termination gear in areas where Telstra is the only provider, using X.164, and that's around $250/mo for 10ch, iirc
23:44.46JTusing Telstra's hardware, but cheaper than telstra
23:44.56jblackYeah. I didn't think about BRIs. you might want to look at those too welby.
23:45.11ManxPowerjblack: BRIs are not a valid option in USA/Canada
23:45.20jblackI thought BRIs were isdn?
23:45.26JTBRIs aren't a good option for asterisk in USE/Canada
23:45.34JTa BRI is ISDN, just like a PRI is ISDN
23:45.38ManxPowerjblack: They are ISDN
23:46.03JTa lot of people incorrectly assume that BRI is the only type of ISDN
23:46.35jblackturns pink when he realizes that PRIs are isdn
23:46.40*** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320)
23:46.58Yourname`Does anyone know what STATUS 5 is in AMI regarding Queue member statuses?
23:47.03JTjblack: they work in similar ways, just a BRI only has a 16kbit/s D channel instead of 64kbit/s
23:47.27Yourname`Oncall is status 3, idle is status 1.. but status 5?
23:47.46jblackPresumably because the number of channels supported?
23:47.54JTright
23:48.22jblackGot it.
23:48.49JTthe D channel messages are identical pretty much between the two
23:48.55JTthe L1 technology is obviously different
23:50.16Yourname`Somebody?
23:50.23Yourname`Or is there an AMI channel somewhere?
23:50.28adrianXXXwho speak spanish ??? i need help ...:(
23:50.36ManxPowerYourname`: Bet you'll have to look at the source code.
23:50.45*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
23:51.54Yourname`ManxPower: Thanks, any specific pointers where?
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