IRC log for #asterisk on 20080629

00:00.30afg_chasdx: I guess your Linux workstation will recognize the embedded USB soundcard as plug and play device and you can choose it as input/output sounddevice from the dropdown list in your preferred application.
00:00.33asdxlol
00:00.45asdxyeah
00:01.37asdx<PROTECTED>
00:01.37asdx<PROTECTED>
00:01.40asdxi will fix it
00:02.05afg_chasdx: any nowadays the sound wizard will configure it for you at a comfortable soundlevel.
00:02.07asdxi just have to chage my ~/.asoundrc and put "card 0" on t
00:02.08asdxit*
00:07.11tzafrir_laptopasdx, full-speed? (USB1)?
00:08.57asdxinput: USB HID v1.00 Device [Logitech Logitech USB Headset] on usb-0000:00:02.0-1
00:09.00asdxyeah
00:09.28asdxguys what does this means: http://pastie.org/224203
00:09.39asdxthe numbers
00:09.43asdx1001
00:09.46asdx2000 etc
00:10.00asdxah the seconds
00:10.12afg_chthe cost of translation
00:10.57asdxcool
00:11.08afg_chunlikely to be seconds. rather milliseconds but I am not sure on that.
00:11.16asdxyeah
00:12.33afg_chbe told that this value seems a bit high. I only see 11 for ilbc. The rest of the heard goes at about 1-2
00:13.00asdxyeah this is in asterisk 1.6
00:13.40asdxin asterisk 1.4 the value is lower
00:14.08asdxbut asterisk 1.4 speaks about "milliseconds" and 1.6 in "microseconds"
00:17.51asdxfeels like watching a good movie and eating pizza
00:21.19afg_chmy eyes are also heavy. I move my operations over to the sofa.
00:21.38afg_chgood night everybody!
00:21.40asdxheh
00:21.44asdxnight dude
00:31.03mvanbaakmy eyes start to be heavy as well
00:31.08mvanbaakSun Jun 29 02:31:07 CEST 2008
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01:40.23Blacrafthello can anyone help me, i can't keep a dialtone on my tdm410 fxs modules
01:44.11Strom_Cwhat do you mean "keep a dialtone"?
01:50.40Blacraftif i reboot the server i get a dialtone for about 1 minute, then it goes away.
01:56.04Strom_Cok...
01:56.06Strom_Chmm
01:56.45Strom_CBlacraft: does simply unloading and re-loading the modules cause the dialtone to come back?
01:58.54Blacraftyes
01:59.03Blacraftony for a bit though
02:00.40Strom_CBlacraft: I'd suggest you give Digium support a call
02:02.00Blacraftyeah ok thz
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03:11.32asdxhow do i make sip work with tcp, transport=tcp?
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03:14.43JayTee52asdx, you have to use * 1.6 for starters
03:15.39asdxJayTee52: i use 1.6
03:16.04JayTee52oh, well it's documented in the sip.sample.conf file and the docs
03:17.05asdxk
03:17.37asdxso it's tcpenable=yes
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06:49.46^^Johnny5can anyone help with asterisk and digium tdm04P boards...
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07:13.45iuytohi
07:14.09iuytoi got sipura 3000 and asterisk installed on my system
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07:15.33iuytomy server asterisk is at home
07:15.48iuytomy sipura 3000 adaptor is on my work
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07:16.40iuytosipura3000 connected with my work pstn line
07:17.39iuytohow to remote my pstn work line to work with my asterisk asterisk extension ,??
07:21.05TJNII~trixbox
07:21.06jbot[~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org
07:21.19TJNIIiuyto: ^^^^^
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07:27.42iuytothx TJNII
07:28.15iuytothe problem is not the asterisk installation
07:28.29iuytoi have allready install asterisk
07:28.56iuytojust wanna configure my sipura3000 work line to my asterisk box
07:29.19TJNIINo, you installed trixbox
07:29.25iuytoeventough i try to install trixbox
07:29.26TJNIIWhich configures asterisk for you
07:29.33TJNIISo we can't support it here
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07:29.40TJNIIBecause we don't know what trixbox did
07:29.49TJNIIAnd since it did it for you, niether do you.
07:30.16iuytook
07:30.24iuytothx TJNII
07:30.32TJNIITry #trixbox
07:30.39iuytook thx
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07:33.51iuytonoone there
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07:47.16TJNIIiuyto: So you have the fxo line on the Sipura connected to a POTS line, and you want * to answer it?
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07:55.20iuytoye
07:55.29iuytoyes TJNII
07:55.29TJNIIAnd then what
07:55.38TJNIIWhat do you want * to do with the call
07:56.11iuytomy sipura3000 is connected to my work pstn line
07:56.22iuytomy asterisk server is at home
07:56.25TJNIIThrough the fxo port, yes
07:56.30iuytoyes
07:56.51TJNIIThe server is physically at home or you are using asterisk@home (aka trixbox)
07:57.02iuytoi create an sip extension from asterisk server
07:57.16TJNIIAre you using a gui or did you install from source?
07:57.18iuytoat home
07:57.25iuytopsycally
07:58.03iuytoyes from source
07:58.11iuytoand i compile the program
07:58.23TJNIIOkay, than that is different.
07:58.28TJNIIThen we can help you here.
07:58.37iuytothx
07:58.58TJNIISo you have the sipura at your work and asterisk at your home.  You want work calls to go to the asterisk box
07:59.18iuytono
07:59.30iuytoi just want to use my pstn work line
07:59.36iuytofrom my home
08:00.09TJNIIWhat hardware do you have at your home
08:00.17TJNIIAny telephone adapters?
08:00.26TJNIIOr do you want to answer the call on a softphone
08:00.38iuytofrom softphone
08:00.42iuytoyes
08:00.43iuytoat home
08:00.58TJNIISo you have the sipura at your work and asterisk at your home.  You want work calls to go to the asterisk box which will then send it to an softphone.
08:01.19iuytono
08:01.47iuytoi just want to use
08:02.06iuytomy work line to call outside the world
08:02.20iuytofrom my softphone
08:02.29TJNIISo what I said but the other way around
08:02.43TJNIISoftphone -> asterisk -> sipura
08:02.49iuytoyes
08:02.53TJNIIAnd the sipura will not answer the line
08:03.03iuytothat 's why i put sipura 3000 at my work
08:03.40iuytoyes it can answer the line
08:03.46iuytousing my sip extension
08:03.55TJNIINow, before we go too far, do you have NAT at home?  Do you have a static IP?
08:04.10iuytono NAT
08:04.17iuytoStatic IP yes
08:04.19iuytoi have
08:04.38TJNIICool.  Then you won't have to deal with the one-directional auido headaches NAT causes.
08:05.00TJNIIHow far have you gotten with your configuration?
08:05.43iuytosorry ??
08:05.52TJNIIWhat have you done so far?
08:06.49iuytoi just my asterisk
08:06.51iuytoserver
08:07.02iuytowith my sip extension
08:07.25TJNIISo you have the softphone connected already?
08:07.30iuytowhich means i can call to my work using my asterisk server sip extension
08:07.35iuytoyes
08:07.55TJNIISo you can make outgoing calls already?
08:07.57iuytono
08:08.24iuytofrom home softphone  i can only join my work sipura3000
08:08.31iuytousing my sip extension
08:08.53TJNIIWhat do you mean ny "join my work sipura3000"
08:08.57TJNIIYou can call it?
08:09.36iuytoas sipura3000 got 1 FXS port and 1 FX0 port
08:09.57TJNIIRight....
08:10.01iuytoi configure my work sipura3000 to use my sip extension
08:10.35iuytonow i just want to connect my work pstn line on FXO port
08:11.04iuytoand remote using my work pstn line from home
08:11.04TJNIISo you have the FXS port set up now?  Or the FXO port?
08:11.24TJNIIYes, I'm clear on what you want to do, but I'm unclear on how far you've gotten
08:13.40iuytoi got i pc wich is running
08:13.48iuytoon asterisk server at home
08:14.02iuyto1 softphone at home
08:14.12iuytoand 1 siprua3000 at work
08:14.27iuytothat's all
08:14.58TJNIIAnd the softphone is setup successfully to *
08:16.11iuytoyes
08:16.41TJNIIAnd you have the FXO line set up to connect to?
08:16.42iuytoi can make call from my home softphone to my work sipura3000 which is connected on fxs port
08:16.52TJNIIs/connect to/connecto to */
08:17.44TJNIIOkay, so you can call a phone on the fxs port from the softphone, and now you want to set up the fxo port?
08:17.56iuytoyes
08:17.57TJNIIDo I understand that correctly?
08:18.13iuytoyes it's correct
08:18.23TJNIIDo you uunderstand dialplan contexts?
08:18.38iuytoyes
08:18.56iuytonot really
08:19.03iuytono not really
08:19.25TJNIIOkay, I need to go to bed, but I'll outline what you need to do.
08:19.32TJNII1) Read chapter 5 of the book
08:19.34TJNII!book
08:19.38TJNII~book
08:19.38jbotAsterisk: 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
08:19.50TJNIIThat will give you an understanding of dialplans.
08:20.02iuytoyes
08:20.05iuytoi will
08:20.09TJNII2) add a entry in sip.conf for the fxo port
08:20.30TJNIIShouldn't be too hard since you were successful with the fxs
08:20.31iuytoyes i have done
08:20.51iuytowait i show u an example on what i have done
08:21.12TJNIImmmkay
08:21.25TJNII~pastebin
08:21.25jbot[~pastebin] 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
08:24.20iuytoplz here http://pastebin.com/m77a5cc10
08:25.14TJNIIpastebin your dialplan.
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08:26.43TJNIIAnd your entire sip.conf
08:27.02iuytohere
08:27.34iuytohttp://pastebin.com/m2faba158
08:29.54TJNIII only see one phone in that sip.conf
08:29.58TJNIIThere should be 3
08:30.35iuytowhy 3 ?
08:31.00iuytomy asterisk server LAN ip is 192.168.0.1
08:31.00TJNIIOne for the FXO line, One for the FXS line, one for the softphone.
08:31.08TJNIIEach device is independent.
08:32.06iuytoi create 6 sip extensions all work
08:32.14iuytowithout any problem
08:32.18iuytousing my softphoen
08:32.23iuytosoftphoen
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08:33.02TJNIIThere is no such thing as a sip extension.  You have SIP peers which devices register as.
08:33.14TJNIIYou define extensions in your dialplan.
08:33.48TJNIINow in the sip.conf you posted, I only see one sip peer.  So One device.
08:33.59TJNIIYou need a minimum of two for what you want to do.
08:34.40iuytook
08:35.07iuytogive the right way on what i have to do ??
08:35.15iuytoon my asterisk server side
08:35.24iuytoand on sipura 3000 side
08:35.29iuyto?
08:36.16TJNIIYou need a SIP.conf entry for the softphone, and a second SIP.conf entry for the sipura.
08:36.21iuytohave i have to have a static or dynamic ip  type of connection where sipura3000 is connected ?
08:36.40iuytook
08:36.47TJNIIYou're going to set the sipura to look to your server's static IP.
08:37.20iuytoyes i configure one of my sip extension on sipura3000
08:37.23TJNIII thought you said you could call the sipura from the softphone?  Based on what you posted, that isn't possible.
08:37.26iuytois it's enough ?
08:38.20iuytono
08:38.54TJNIISo you can register with the softphone _OR_ you can register with the sipura right now
08:38.55iuytoi can make call from softphone to join my sipura3000
08:39.03TJNIIThat makes no sense.
08:39.16TJNIIGo to the cli and type sip show peers
08:39.23TJNIICopy the output.
08:40.03TJNIIMake sure to have your verbosity set to >= 5
08:40.20iuytowait
08:42.36iuytoplz see here
08:42.38iuytohttp://pastebin.com/m7c3edc40
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08:43.00iuytowhere 1234 is my softphone
08:43.21iuytoand 0987654321 is my sipura box
08:44.09TJNIIPlease pastebin your dialplan.
08:44.27iuytoi  post u all my sip.conf
08:45.04iuytowhere i can find it ?
08:45.13TJNIIOkay.
08:45.24TJNIIYou really need to read this:
08:45.26TJNII~boot
08:45.26jbotfrom memory, boot is what you get when you act like a UnderNet user, or #debian-boot
08:45.28TJNII~book
08:45.29jbotAsterisk: 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
08:45.35TJNII^^^^^^^
08:46.02iuytoyes
08:46.03TJNIIYou can safely skip the beginning, since you already have the server up
08:46.10TJNIIRead chapter 5 CAREFULLY
08:46.16iuytowhat i
08:46.18TJNIIIt is key
08:46.33iuytohave allready buy this book from 75 dollars
08:46.45iuytobut it's in french
08:47.30TJNIIWell, read the chapter on dialplans
08:47.54TJNIIAnd things should start to make more sense for you.
08:48.32iuytook thx a lot TJNII
08:48.36TJNIIMe, I need to go to bed.  It is 10 till 4 in the morning, and if I sleep into the afternoon my GF will be pissed
08:48.44TJNIIGood luck
08:48.49iuytow
08:48.51iuytook
08:48.53iuytothx a lot
08:48.55iuytobye
08:48.57iuytogoood night
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10:40.35unpaidbillfuck nanpa in their dirty microsoft loving asses
10:40.50unpaidbillwtf are these assholes doing distributing the area code list in .mdb format
10:41.04unpaidbillfucking idiots
10:42.55unpaidbillalso, i want to punch them all in the goddamn faces
10:47.33tzafrir_laptopdirects unpaidbill to a near-by glass of water
11:04.22unpaidbillhaha tza
11:04.24unpaidbillyeah.
11:04.35unpaidbilli had to install openoffice
11:04.37unpaidbillirritating
11:05.12unpaidbilland the info i got didnt have what i wanted
11:05.18unpaidbillso i am still digging through their damn site
11:09.01tzafrir_laptopunpaidbill, I have to agree. openoffice is irritating
11:09.05tzafrir_laptop;-)
11:09.07unpaidbillyeah
11:09.13unpaidbillbeats paying though
11:09.47unpaidbilldo you have any idea where on the nanpa site they list the local numbers ranges for an areacode and exchange?
11:10.02unpaidbilli know i downloaded that info in CSV format .... 7 years ago
11:10.06unpaidbillbut i cant seem to find it now
11:10.30unpaidbilland hawaii has one area code with exchanges broken into local/long distance, so sometimes you have to dial 1808 and sometimes you dial only 7 digit
11:10.33unpaidbillannoying as hell.
11:11.13unpaidbilli guess i can have my clec send it
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11:16.48Fiapo-CEwhat the best linux-distro to install asterisk?
11:16.57unpaidbillwhichever one you prefer
11:17.44Fiapo-CEis debian a good choice?
11:17.50unpaidbillif you like it
11:18.25Fiapo-CEyes, i like it... but i need recompile kernel etc?
11:18.35unpaidbillnope
11:18.51Fiapo-CEgood, thanks!
11:19.01unpaidbillthe most you will need to do is install the kernel headers
11:19.11unpaidbillwhich is just apt-get install blah
11:19.22unpaidbilland of course all your compiling utilities
11:19.27unpaidbill(if you want to use zaptel, that is)
11:19.53unpaidbillyou'll need zaptel for MeetMe (conferencing), unless you're installing 1.6 i believe
11:20.00unpaidbillfor the ztdummy module
11:20.47Fiapo-CEsure
11:21.05Fiapo-CEthanks for help
11:23.02[netman]Fiapo-CE: I love Debian, but I think CentOS based Asterisk system r better supported
11:23.25[netman]i.e. there r more information on the web related to that distro
11:24.06[netman]so I have installed my Asterisk over CentOS 5.2
11:27.21Fiapo-CE[netman] i will test it
11:27.25Fiapo-CEI'm testing asterisk as active pbx..
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11:27.37Fiapo-CEwith php i'm creating a .call and moving for outgoing directory
11:27.54Fiapo-CEI´m using php to integrate my actual system w/ asterisk to make the call
11:28.07Fiapo-CEsample: http://myserver/call.php?numer=XXX&session=xxxx&extensionline=XXXX
11:28.18Fiapo-CEI have 10 persons doing active calls, well asterisks make the call and give to extension..
11:28.35Fiapo-CEthis is the best way to do it? through .call file?
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11:30.49unpaidbilluse the manager interface
11:31.02unpaidbillinfo here http://www.voip-info.org/wiki/view/Asterisk+manager+API
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11:39.52rolershello. I've got 2 hfc-s cards (one in NT and one TE mode). The NT-card works fine. But with the TE-card I'm just getting the error: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
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11:50.38yangvncrolers: have you configured zapata.conf correctly ?
11:50.49yangvncrolers: defined channels etc.?
11:51.21yangvncrolers: maybe you could paste your configuration
11:51.25yangvnc~pb
11:51.26jbot[~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
11:52.04rolersyangvnc: I think so: http://rafb.net/p/zw89dT86.html
11:53.12yangvncrolers: I had problems with signalling=bri_cpe_ptmp try simply signalling=bri_cpe
11:53.59yangvncrolers: and there were problems with SPANS, if you unplug the ISDN cable you should restart asterisk or reboot machine
11:56.05rolersyangvnc: My extenstion: exten => 35,1,Dial(Zap/g2/85695). I changed the value to bri_cpe, but there is still the same error.
11:56.47rolersincomming calls also don't work
11:57.22yangvnctry ZAP/g2/${EXTEN})
11:58.07rolersyangvnc: nothing changed.
11:58.22yangvncfor incoming call you should specify extensions, if your number is 1234 you should make exten => 1234,1,Dial(SIP/<your phone>
11:58.36rolersyes I know
11:59.28rolersI'm running asterisk in verbosity level 3. On incomming calls just the capi-card reacts
12:07.02rolersyangvnc: pri_find_dchan: No D-channels available!  Using Primary channel 6 as D-channel anyway!
12:13.42tzafrir_laptoprolers, It may mean that the span is down
12:14.53tzafrir_laptoptry 'head -n 1 /proc/zaptel/*' . Is layer 1 up in those cards?
12:16.20rolerstzafrir_laptop: http://rafb.net/p/2LYqPy38.html and /proc/zaptel*:http://rafb.net/p/0W4c1m27.html
12:17.54tzafrir_laptoprolers, the second parameter in the span= lines should not be the same for all cards, IIRC . Though I don't think it is the issue here
12:19.37tzafrir_laptopThe first card is actually connected as NT . But in zaptel.conf you set it to get timing from the remote side
12:20.10tzafrir_laptopDo you want to use it as NT (e.g.: to connect an ISDN phone) or as a TE (to connect to the telco)?
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12:22.08rolerstzafrir_laptop: I want to use the first card as NT and the first card works fine.
12:22.23rolersI want to use the second card as TE.
12:22.55tzafrir_laptopIf you use that card as NT, better write in the span line span=1,0,<whatever>
12:23.27tzafrir_laptopSo that card will provide timing to the remote side
12:24.42rolerstzafrir_laptop: ok I changed it.
12:24.55rolersWhy doesn't the second card in TE-mode work?
12:25.41tzafrir_laptopcan you try a loopback?
12:25.50tzafrir_laptopbetween the NT and TE cards?
12:26.47rolersHow to do that?
12:27.42tzafrir_laptopConnect them
12:28.33rolersokay I'll do. But I have to restart asterisk?
12:32.10tzafrir_laptopactually, no
12:32.31tzafrir_laptopJust pretend that the NT is your telco :-)
12:33.03tzafrir_laptopBut if you changed zaptel.conf, you do need to re-run ztcfg
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12:37.24rolerstzafrir_laptop: okay I've got a loopback. But TE-mode don't seems to work
12:37.32rolers*doesn't
12:38.08tzafrir_laptopdoesn't seem to work == ?
12:38.15rolersUnable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)
12:39.07rolersI've got two extenstions in the local-context:
12:39.07rolersexten => 35,1,Dial(ZAP/g2/85)
12:39.07rolersexten => 36,1,Dial(ZAP/g1/86)
12:39.24tzafrir_laptopwhat is the output of: asterisk -rx 'zap show channels' and what is in your zapata.conf ?
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12:41.58phlackmorning
12:42.10phlacki was just wondering could someone give me a quick help with my asterisk install
12:42.18phlackits hanging every time i do a reload
12:42.51[netman]I bet it's a bad zaptel configuration
12:42.57phlacki think you could be right
12:43.18phlackbut unfortunately i don't know enough about it, I have no zaptel hardware installed but i installed the drivers for completeness
12:43.43[netman]don't install drivers for hardware u haven't
12:43.47rolerstzafrir_laptop: http://rafb.net/p/KiTFcg80.html
12:43.59phlackit was just for future proofing because i intend on getting the hardware
12:44.14phlacki just noticed the banner at the top about Contos 5.2 issues
12:44.24phlack~centos52bug
12:44.25jbot[~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2.  Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889
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12:44.47phlackwould that cause it?
12:44.53*** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21)
12:45.40Mike8861hello everyone
12:45.54tzafrir_laptoprolers, but what is in your zapata.conf? (for the groups)
12:46.10phlackjust a second ill check
12:47.05tzafrir_laptopphlack, that specific bug is a build issue. If you successfully built zaptel it won't byte you at runtime
12:47.21phlackthere is nothing enabled in the groups of zapata.conf
12:47.30rolerstzafrir_laptop: zapata.conf: http://rafb.net/p/WuYOA271.html
12:47.40tzafrir_laptopAnd it has now been fixed in SVN (the above bug report includes a patch)
12:47.50phlackah ok
12:47.57phlacki compiled from source though
12:51.26phlack== Using TOS bits 0
12:51.34phlackits hanging here during reload
12:52.44phlack== Using TOS bits 0
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12:52.51phlackoh sorry didnt mean to paste that twice
12:55.39phlackthat seems to be part of the MGCP
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13:11.52icenicolahello. how can i do auto provisioning in asterisk? what do i need?
13:12.55Mike8861icenicola: do u mean auto provisioning a hard phone ?
13:13.40icenicolai want to plug a sip device and it will take configurtions from my server that runs asterisk
13:15.08Mike8861well, i have not done it from asterisk. if i remember correctly
13:15.09icenicolacan i auto provision a sip device?
13:15.27Mike8861u can only provision a sip device that support auto provision
13:15.32Mike8861which is usually a hardphone
13:16.15Mike8861polycom, aastra, snom supports auto provision
13:16.32icenicolawhat about linksys and batm?
13:16.51icenicolai could search them
13:16.53Mike8861check with the manufacturer if it support
13:16.59icenicolacan u tell me please how to do it in asterisk?
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13:18.02vk4akpAnyone interseted in helping me with a compile error?
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13:22.11trafimHi guys. Can anyone help me with my prob? I have dialplan like: incoming -> voicemenu (here customer chooses one of the languages) -> queue-decision (day and night queue for each language) -> queue. And now my task is to make member in night queue for every language to change every night (every night different guy will answer).
13:22.16trafimthe only way i can think of is to create a bunch of queues for every phone numbe
13:22.19trafimr, and huge pack of gotofitime's, but i feel it's not efficient way. i'm not an
13:22.22trafimasterisk expert though, so i maybe missing the point.
13:22.29trafimaww, linebreaks. sorry.
13:24.12trafimanyway, can't figure out how to implement it without such a headache.. can anyone give an advice?
13:24.26tzafrir_laptopflorz, ping
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13:45.57vk4akpmake[3]: asnparser: Command not found
13:46.02vk4akpAny idea's? :(
13:46.38riddleboxvk4akp, what distro
13:46.47vk4akpSabayon
13:46.50lmadsentrafim: I'd use a Local channel which then checks in your local AstDB (or you could hard code it into the dialplan too) which does the check to determine who the call should go to.
13:47.35lmadsentrafim: so your queue would have a single member, and that member would be a Local channel, then inside the Local channel configuration (which is just a context inside the dialplan), you could add your GotoIfTime()'s, or whatever other method you want.
13:47.51lmadsentrafim: just like you do your night and day menu's for the IVR, you would do the same kind of idea
13:48.07riddleboxvk4akp, http://www.google.com/search?hl=en&q=make%5B3%5D%3A+asnparser%3A+Command+not+found+asterisk&btnG=vkSearch
13:48.40phlackis it possible to run asterisk, im only using voip without zaptel and libpri or do i still need the ztdummy module for timing?
13:49.00riddleboxphlack, I did that for a long time
13:49.00vk4akpno
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13:49.14vk4akpyou only need the ztdummy for the meetme conference.
13:49.19riddleboxphlack, you need ztdummy for conference
13:49.27phlackyeah i want conference so im gonna need it
13:49.43riddleboxphlack, yup
13:51.33vk4akpLooks like my problem is the new OS.
13:51.42riddleboxok which codec uses the smallest bandwith, ulaw,alaw,or g.726?
13:51.45phlackso during the make menuselect, i can disable everything apart from the ztdummy
13:51.52trafimlmadsen: tnx, i thought i'll get no response at all. i'll go read about these local chan's now.
13:51.56phlackand still allow conferencing
13:52.00vk4akpG.729
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13:52.22vk4akpG,729, followed by G.726
13:52.26riddleboxvk4akp, dont you have to pay for that codec?
13:52.56vk4akpI believe it's free now for non commercial use.
13:54.02coppiceits free for nothing
13:57.15riddleboxhrmm well I have been using u-law here at home for almost 2 years and havent had any issue, so I guess I will keep it
13:59.56coppiceif u-law gives you no trouble don't change. it will most certainly give you better quality than G.726 or G.729
14:00.10trafimlmadsen: and, if it won't disturb you, what would example of calling this local chan from queue look like? like, in queues.conf, there must be something like "member => Local/s@night/n" and somewhere in dialplan context [night] full of gotoiftime+dial? i got it correctly?
14:11.05riddleboxcoppice, yeah thats kind of what i am thinking
14:12.22coppicewideband voice is far better, but the only broad support for it is in Skype
14:14.50lmadsentrafim: you got it pretty much, but abstract the context name a bit more:  [queue_members], then member => Local/s@queue_members/n, then in [queue_members] you just determine what time it is, such as with GotoIfTime(), then you can have a bunch of GotoIfTime()'s that go to an extension in queue_members such as 'bob', 'nancy', 'george' depending on when you want to distribute calls to those queue members
14:15.34lmadsenso the 's' extension is just the check to determine which extension to actually send the call to based on your time requirements, then an extension like 'george' might be:  exten => george,1,Dial(SIP/george)
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14:16.14lmadsenrussellb: morning!
14:16.31lmadsenriddlebox: don't fix what ain't broke
14:17.19trafimlmadsen: thanks alot.
14:17.38lmadsentrafim: no problem, that should give you enough information to get going
14:17.59lmadsentrafim: thanks for asking a well phrased question :)
14:18.26trafimlmadsen: :)
14:22.07riddleboxlmadsen, thats how i feel abot it too
14:22.12lmadsen:)
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14:22.22lmadsenall happy admins live by that mantra :)
14:29.30coppiceI thought happy admins achieved that state by being fault tolerant
14:30.24ManxPoweri thought admins chieved that state by killing the problem users?
14:30.40ManxPowerand achieved too!
14:31.12coppicethat sounds fault intolerant. not good for the karma
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14:37.12ManxPowerIt's always the user's fault!  But I'll tolerate it. 8-)
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14:38.59riddleboxlol
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14:40.24riddleboxis the fax extension built in now, or do you have to add some 3rd party software
14:42.25outtoluncwonders if i drink some coffee this channel will appear to move faster <G>
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14:46.36coppicedrink syrup of figs and everything will move faster
15:03.46outtoluncmmm coffee
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15:06.15riddleboxspeaking of coffee, I need breakfast
15:06.34coppiceat 11PM? :-\
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15:07.29mwallingnot everyone is on your side of the world...
15:07.41mwallingSun Jun 29 11:07:41 EDT 2008
15:08.50coppicewell 11AM is also a little off for breakfast
15:08.59outtolunconly 8am hee
15:09.03outtoluncer here
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15:14.41mwallingcoppice: never heard of brunch?
15:16.12vk4akpis ther ea newer svn release other then 1.4 ?
15:19.18russellbasterisk/trunk is the most bleeding edge
15:19.23russellbthere is also asterisk/branches/1.6.0
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15:31.04vk4akpCan I get 1.6 using SVN ?
15:31.23vk4akpI'm having trouble getting 1.4 to compile using a newer distro.
15:31.45vk4akpseems to be a common problem across lnewer linux distro's.
15:31.56vk4akpasnparser: doesn't exist.
15:32.00[TK]D-Fendervk4akp: just download off digium.com
15:32.17[TK]D-Fendervk4akp: And which distor would that be?
15:32.24[TK]D-Fenderdistro*
15:32.24vk4akpSabayon.
15:32.32vk4akpBut google shows others havign the same problem.
15:32.38[TK]D-Fendervk4akp: And whats the problem?
15:33.02vk4akpmake[3]: asnparser: Command not found
15:33.37vk4akpasnparser -m CISCO_H225 -c cisco-h225.asn
15:33.37vk4akpmake[3]: asnparser: Command not found
15:33.52[TK]D-Fender"If your version of OpenH323 requires ASNParser, then it is over a year out of date." <---
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15:34.11vk4akpthats 1.4
15:34.22vk4akpAsterisk 1.4 needs it
15:35.00[TK]D-Fendervk4akp: Only for chan_h323 apparently
15:35.00mwallingwow.. a gentoo distribution using old software?
15:35.10mwalling/o\
15:35.12[TK]D-Fendervk4akp: So unselect it in menuconfig
15:35.20vk4akphugh?
15:35.29[TK]D-Fender"make menuconfig"
15:35.40vk4akpYou will have to dumb this down for me. I am no linux guru unfortunatly.
15:35.40[TK]D-Fenderunselect h323.
15:36.09vk4akpI haven't had to do any of this b4 on older releases. Why has it become  aproblem now?
15:36.32[TK]D-Fendervk4akp: What versions does this problem appear on?
15:36.42vk4akpHow do I get this asnparser into the OS instead?
15:37.06vk4akpSabayon 3.5 loop 3, with Asterisk SVN install of 1.4
15:37.28[TK]D-Fendervk4akp: What other versions of *?
15:37.45[TK]D-Fendervk4akp: And don't jsut say SVN.  EVERY versions is available through SVN
15:37.46vk4akpI don't know.
15:37.52vk4akpI am trying to install 1.4
15:37.59vk4akpIts what I was using with Sabayon 3.4f
15:38.07[TK]D-Fendervk4akp: Stop using SVN and work of digium's download site
15:38.15vk4akpwhy
15:38.22vk4akpsvn was simple last time.
15:38.34mwalling[TK]D-Fender: hes a gentoo user... have patience
15:38.39[TK]D-Fendervk4akp: Because you can't track down where the problem is when you can't even tell what version of * you're running.
15:39.05[TK]D-Fendergenpoo
15:39.11[TK]D-Fenderlights up the torches...
15:39.27Mike8861HI [TK]D-Fender
15:40.02mwallingfwiw, * builds *perfectly* on slackware ;)
15:40.06[TK]D-Fendervk4akp: If you aren't a linux guru you should stick to a popularly supported distro.
15:40.21vk4akphttp://www.asterisk.org/developers/get-source    <<< This is what I've always used b4.
15:40.21[TK]D-Fendermwalling: Yup, Slack "just works"
15:40.34mwallingvk4akp: are you a developer?
15:40.38[TK]D-Fendervk4akp: Good, now stop.
15:40.45vk4akpwhat?
15:40.51vk4akpWhat the hell R U talking about.
15:41.09[TK]D-Fendervk4akp: http://www.asterisk.org/downloads
15:41.20mwallingwelcome to irc, which is not aim. we spell words out here
15:41.25[TK]D-Fendervk4akp: SVN is the latest "to the minute" including the latest BUGS <-
15:41.47[TK]D-Fendervk4akp: FUCK SVN.  Get over it, get off you ass and create a CONSISTENT working environment.
15:41.49vk4akpThey are working on 1.6
15:41.56vk4akpwhy would 1.4 now be a problem?
15:42.04[TK]D-Fendervk4akp: 1.4 is a SERIES!
15:42.09vk4akpI've installed it plenty of time sthis way b4.
15:42.13[TK]D-Fendervk4akp: not a specific versioN!
15:42.23[TK]D-Fendervk4akp: And its not working, so stop whining.
15:42.36mwallingb-e-f-o-r-e
15:42.38vk4akpWhining. Amazing.
15:42.43Mike8861[TK]D-Fender , how do i get video ivr working on asterisk ?
15:42.50[TK]D-FenderMike8861: no idea.
15:42.50vk4akpTrying to get around a problem is whining. OK right.
15:43.03[TK]D-Fendervk4akp: You don't even know what release you are working on.
15:43.03mwallingvk4akp: you're creating a problem for yourself
15:43.08Mike8861[TK]D-Fender are u serious ?
15:43.17mwallingy-o-u
15:43.19vk4akphow can I know.
15:43.20[TK]D-FenderMike8861: Yes.  Who uses video for ivr?
15:43.29russellbvideo IVR is the same as audio IVR, actually
15:43.36[TK]D-Fendervk4akp: If you don't know you shouldn't be using SVN.
15:43.38russellbas long as the files are formats that asteirsk supports ...
15:43.46russellband the caller has video support
15:43.57russellbworks well ..
15:44.06Mike8861i saw video on youtube, some taiwan people got nokia E61 with video IVR working
15:44.12[TK]D-Fenderrussellb: will * play multiple streams on mathing the video codec as well?
15:44.18russellbyes
15:44.33russellbit's ... magic
15:44.47[TK]D-Fenderrussellb: neato.  How would you record both audio & video simultaneously for playback like that?
15:44.58russellbsame, with magic
15:45.09russellbjust using the Record app ... or any other recording stuff in asterisk
15:45.10[TK]D-Fenderrussellb: Sorry, all out of goats to sacrifice :)
15:45.18russellbcalling into voicemail from a video phone will record audio and video
15:45.24russellbcalling back in will play them both back
15:45.26[TK]D-Fenderrussellb: thought record only recorded a single format
15:45.26russellbvideo voicemail ftw
15:45.41russellbok, the Record app may not work ...
15:45.46russellbbut i know voicemail does
15:46.01Mike8861can i post the youtube link over here ?
15:46.02[TK]D-Fenderrussellb: So use VM as a recording mechanism & move them out for IVR usage?
15:46.07Mike8861it demo about vid ivr
15:46.08russellbha
15:46.14russellbif record doesn't work, we should make it work
15:46.17russellbi haven't tried in forever
15:46.18russellbi don't remember
15:46.38russellbjust checked, Record() supports video
15:46.48Mike8861well, i have magic passion, i will make this magic to happen!
15:46.52[TK]D-Fenderrussellb: just the I recall Record using the file extension to specify the format which to me looks like it implies only a single format can be recorded at a time.
15:47.09russellbwell, that's because we support audio transcoding
15:47.19russellbwe don't support video transcoding, if there is a video stream, it will just record it in the native format
15:47.21Mike8861@russellb: which AGI script do i need to use to push video ??
15:47.33russellbum
15:47.44[TK]D-Fenderrussellb: and it will record it accordingly, transcoding audio if thats the case?
15:47.50Mike8861@russellb i planning to push it in h.263
15:47.54russellbnods
15:48.06russellbMike8861: whatever AGI script you write, i guess?
15:48.10[TK]D-Fenderrussellb: I should try it sometime then.  Also looking to go 1.6 shortly.
15:48.17russellb[TK]D-Fender: brave man :)
15:48.27[TK]D-Fenderrussellb: jsut at home.  Little to lose
15:48.29seanbright[TK]D-Fender: i'd recommend sticking with 1.2
15:48.32seanbrightducks
15:48.40[TK]D-Fenderseanbright: that'd require DOWNGRADING ;)
15:48.46russellb~fist seanbright
15:48.46jbotACTION uses seanbright as a handpuppet
15:48.49seanbrighthaha
15:48.52seanbrightnooooooooooooooo!
15:48.55[TK]D-Fenderrussellb: that looked... wrong...
15:49.04russellb~punch seanbright
15:49.05jbotACTION lets fly with a wild haymaker which catches seanbright right on the nose
15:49.08russellbthat's what i meant!
15:49.09russellb:-p
15:49.13seanbright[TK]D-Fender: i made that unfortunate discovery last week
15:49.13[TK]D-Fender...
15:49.23seanbright[TK]D-Fender: tried ~fist fight Qwell
15:49.28seanbrightand all hell broke loose
15:49.39Mike8861example :
15:49.39Mike8861<PROTECTED>
15:49.40Mike8861<PROTECTED>
15:49.49Mike8861i need to have a script to play back video
15:50.04Mike8861whichh command does the job ?
15:50.05russellbif you also have tt-weasels.h263, then it will work
15:50.08seanbrightnone of the apps support video playback i don't believe
15:50.14russellbstares at seanbright
15:50.15seanbrightor i am lying
15:50.21seanbrighti am lying.
15:50.23seanbrightheh
15:50.32[TK]D-FenderMike8861: apparently "playback" alone will do if you have a "tt-weasels.h263" from what russellb just said
15:50.40russellbyes
15:50.51[TK]D-Fenderseanbright: ignorance != lying
15:51.00russellbtouche.
15:52.39seanbrightgoes back to something else
15:53.37Mike8861@russellb thanks
15:53.47russellbsure
15:53.51russellbseanbright: no!
15:53.57russellbjbot: make seanbright work on asterisk
15:53.58jbotmake: *** No rule to make target `seanbright work on asterisk'.  Stop.
15:54.07russellbheh
15:54.09seanbrightis working on hoard
15:54.15russellbseanbright: w00t
15:54.21russellbjbot: make love
15:54.22jbotmake: *** No rule to make target `love'.  Stop.
15:54.29russellbjbot: make
15:54.40seanbrightjbot: make war
15:54.41jbotmake: *** No rule to make target `war'.  Stop.
15:54.44seanbrightah well
15:54.48russellbi see.
15:55.00seanbrightjbot: make something for god's sake
15:55.01jbotmake: *** No rule to make target `something for god's sake'.  Stop.
15:55.10russellbjbot: i hate you
15:55.10jbotYou hate you?
15:55.20russellbjbot: die
15:55.21jbotACTION takes two shots to the head and crumples to the ground, lifeless.
15:55.21seanbrightjbot: you hate you
15:55.27Mike8861maybe video ivr is good for sex hotline
15:55.36Mike8861it can increase sales ???
15:55.58seanbrightMike8861: have you *seen* phone sex operators?
15:56.30Mike8861seanbright: nope.
15:56.47Corydon76-digThere's a good reason why they're on the phone and not in a strip club
15:56.50Mike8861seanbright: are they IVR or nerv network ??
15:57.00seanbrighthas no idea
15:57.12seanbrighthas yet to resort to such measures
15:57.14seanbright:)
15:57.17JayTee52picture a 400lb woman wearing spandex and a pound and a half of makeup with green eye shadow
15:57.22coppiceignorance is truly bliss in this case
15:57.26Mike8861seanbright: i see
15:57.29seanbrightholding a baby
15:57.31seanbrightand ironing
15:57.41seanbrightand smoking
15:57.45JayTee52while smoking a cheroot
16:01.48*** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net)
16:03.30*** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola)
16:06.32Mike8861hello Ebola
16:06.53*** join/#asterisk smach (n=smach@bas15-ottawa23-1279684007.dsl.bell.ca)
16:06.58EbolaHi
16:07.18Mike8861its so quiet over here
16:07.50EbolaIt being Sunday will have something to do with it.
16:09.21Mike8861yeah, its monday midnight over here
16:09.31Mike8861are u living in the US ?
16:10.09hi365(how) can i use [zonemessages] (in voicemail) to set the default zone settings w/o having to add the settings to every mailbox?
16:11.06Corydon76-dighi365: set tz=foo in the [general] section
16:11.14hi365thanks
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16:25.39Mike8861[TK]D-Fender: i wanna ask about Playback()
16:25.56Mike8861exten => 123,2,Playback(tt-weasels)  <-- the parameter means path to file ??
16:29.22gr0mitMike8861, yes
16:29.22JayTee52yep, that's the path to the file allright and it's even documented in the book!
16:36.03Mike8861it doesnt works. i got some error message from console
16:36.10Mike8861<PROTECTED>
16:36.26Mike8861can anyone help me to debug ?
16:37.28[TK]D-FenderMike8861: maybe you should prove the files are THERE
16:38.17QwellMike8861: how is that an error?
16:38.49Mike8861please hold on, i am pasting extension.conf to pastebin.
16:38.59[TK]D-FenderMike8861: NO
16:38.59Mike8861this is my first attempt to modify extension.conf
16:39.15Mike8861[TK]D-Fender: No ??
16:39.23[TK]D-FenderMike8861: maybe you should prove the files are THERE <--
16:40.06Mike8861[TK]D-Fender: i am using a build-in blue claim river file, and the file is there, i proved it by "moh show files"
16:40.26[TK]D-FenderMike8861: those aren't in the same place.
16:40.35Mike8861[TK]D-Fender: oh....
16:40.39Qwellalso...
16:40.44Qwell~freepbx
16:40.45jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
16:40.58[TK]D-FenderMike8861: go look in your lib folder under sounds and verify that the ones you are playing back are there
16:41.10QwellSurely you aren't using freepbx...are you?
16:41.15[TK]D-FenderQwell: He may have gone independant, so I was going to leave that alone.
16:41.51Mike8861@Qwell, i got freepbx, but i am trying to do this without the help of gui
16:42.11Qwellthe next time you use the gui, it's going to overwrite everything you've just done
16:42.13[TK]D-FenderMike8861: You're frightfully close to "burn in Hell" territory.
16:43.12Mike8861i will set it to permission to 666, so freepbx wont write this ?
16:43.33[TK]D-FenderMike8861: You have no clue.  Good luck with all that...
16:49.43Mike8861[TK]D-Fender: I have changed the file, but its still not working to "exten => 123,2,Playback(you-sound-cute) "
16:50.07[TK]D-FenderMike8861: and you haven't proved the file is there.
16:50.30Mike8861[TK]D-Fender: you-sound-cute is under /var/lib/asterisk/sounds
16:50.50[TK]D-FenderMike8861: PASTEBIN <---- and show us the ERRO (full CLI output)
16:51.06[TK]D-FenderMike8861: And show us the file itself with mermissions, etc
16:51.44Mike8861[TK]D-Fender: thanks, please hold, i takes some time
16:57.41Mike8861[TK]D-Fender: the files are ready
16:57.48Mike8861http://www.pastebin.sk/en/7225/ http://www.pastebin.sk/en/7224/ http://www.pastebin.sk/en/7223/
16:59.48[TK]D-FenderMike8861: BLIND
17:00.00[TK]D-FenderMike8861: that isn't even executing the same code as your show from your dialplan.
17:00.09[TK]D-FenderMike8861:  -- Executing [123@from-internal:4] Playback("SIP/1025-b77c6b70", "silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer") in new stack
17:00.24[TK]D-FenderMike8861: Clearly whatever you are looking at in that other pastebin is NOT getting used at all
17:00.53Mike8861what  does that mean excatly, i am dialing from a extension 1025
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17:01.17[TK]D-FenderMike8861: http://www.pastebin.sk/en/7224/ <-- this code is not being exewcuted at all
17:01.42[TK]D-FenderMike8861: and the fact you think "but I diead it from extension 1025" matters just showes you have no idea what's going on.
17:02.02[TK]D-FenderMike8861: http://www.pastebin.sk/en/7225/ <-- look at whats being CALLED <-
17:02.17[TK]D-FenderMike8861: Nothing at all alike.
17:02.22mwalling!book
17:02.24mwalling~book
17:02.25jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
17:02.31mwallingMike8861: reaqd that, start fresh
17:02.39[TK]D-Fendermwalling: He's on FreePBX.  There is no hope
17:02.49Mike8861[TK]D-Fender: it has been called to s@macro-hangupcall:1] ???!!!
17:03.13mwalling[TK]D-Fender: if i can figure out how to build a svn-1.5.0 package for ubuntu, he can figure out how to write a dialplan ;)
17:03.17Mike8861[TK]D-Fender: to solved this, i need to modify extension_addition.conf ?
17:03.27[TK]D-FenderMike8861: Listen up.  You made an exten (123) to try and play back thqat recording you mentioned.  its not getting USED
17:03.45[TK]D-FenderMike8861: FreePBX is not supported here and you have no Clue.  this is no the channel for you.
17:04.26Mike8861[TK]D-Fender: i am not using FreePBX to made this code, i swear i code it manually
17:04.56[TK]D-FenderMike8861: yes and its NOT GETTING CALLED.  You have no idea where to put it or how to set up FreePBX to get it EXECUTED.
17:05.13[TK]D-FenderMike8861: And we are not here to figure out how to use that garbage to teach you.
17:09.23JayTee52changes channel topic of #freepbx to "Abandon hope, all ye who enter here!"
17:11.05mwallingisnt it "dial here"?
17:11.29ManxPowerhow about just "Abandon all hope!"?
17:17.50gr0mitmaybe we should have bot that looks for freepbx
17:18.17gr0mitany mention of the word should initiate a tyrade of messages
17:18.33gr0mit'not supported - abandon hope, You Are On Your Own, etc!
17:18.34[TK]D-Fendergr0mit: Or maybe we should just KB users who don't get the picture and continue to press on in here..
17:18.39gr0mithehe
17:21.02*** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com)
17:21.55gr0mithas had Bad Experiences with customers who insist on using Freepbx
17:22.39gr0mitto the extent that i will not touch any box that runs anything other than native Asterisk
17:24.50[TK]D-Fendergr0mit: I'm very close to that point myself.  I make some exceptions for certain paid clients.
17:25.09[TK]D-Fenderpaying*
17:33.37gr0mithas had a client who has not paid
17:34.02gr0miti took the project on before i realised it was freepbx
17:34.13gr0mittold him to abandon it if he wanted my help
17:34.50gr0mitpersuaded me to help anyway - project was a disaster - he never got paid, and thus I never got paid.
17:34.54gr0mitgrrr.
17:35.23gr0mittwo lessons i learned .
17:35.38gr0mit1) Avoid freepbx like the plague.
17:35.49gr0mit2) Demand payment in advance.
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17:38.09gr0mitwonders if the mention of the plague rattled Ebola's cage!
17:38.10[TK]D-Fendergr0mit: Words to live by.  I do a quick free log-in inspection first, then quote.  Payment upon initiation of work.
17:38.31[TK]D-Fendergr0mit: Never a dissatisfied customer.
17:39.00gr0mitdo you stop work when the money runs out?
17:39.13gr0mitquoted by the hour
17:39.55gr0mitnext time i will quote for the job
17:40.23gr0mitthe project in question involved freepbx installs in 3 countries
17:40.30gr0mitshudders
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17:41.44[TK]D-Fendergr0mit: I usually job quote, and it turns our rather accurate.  Its always stated that if things go quicker that benifit gets passed on as follow-up support.
17:42.24[TK]D-Fendergr0mit: I discount certain "exploratory" work if it will expand my knowledge even if its likely to exceed my typical hourly rate.
17:42.33[TK]D-Fender(within reason)
17:42.50gr0mitwell,  the issue involved a lot of fiddling trying to bypass voip-blocking in certain ahem, Arab countries.
17:46.24[TK]D-Fendergr0mit: Yeah we've got a few people here who are trying they best to get themselves arrested :)
18:00.16styelzis there any benefit in using the -s command line option for asterisk ?
18:00.39styelzif you have zap hardware?
18:01.34[TK]D-Fenderstyelz: I don't even see that option listed.
18:01.43styelzsorry i mean -I
18:02.03styelzhad s on the brain
18:02.20styelz<PROTECTED>
18:02.43styelzshould i be using this if i have zap hardware ?
18:02.49[TK]D-Fenderstyelz: Never needed it personally... it "just works"
18:02.56[TK]D-Fender(without)
18:03.00styelzok cool.. i pretend i didnt see it
18:08.12[TK]D-Fenderok, out for a while...
18:26.14*** join/#asterisk shtoom (n=shtoom@121.246.167.147)
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18:39.27*** join/#asterisk James|TCC (i=James@87-194-161-247.bethere.co.uk)
18:39.56James|TCCdoes anyone know ho i could go about a prank call system in asterisk - sounds strange i know lol
18:40.20James|TCCbasically we want to be able to from an IRC bot cause a call to go out to my mobile, ring for 20 secs, then hang up
18:40.53James|TCCits easy to do if i have an extension, but without makingn our irc bot a SIP compliant extension, im not sure how to proceed
18:40.57plikshell scripts and call files by the sound of things
18:41.20James|TCCoh and another thing, the irc bot is not on the same machine as the asterisk
18:41.45plikslightly more complexshells scripts and ssh keys
18:42.03James|TCCthe bot is php, so SOAP is my method of choice,
18:42.16James|TCCdo you know of any tutorials on making calls from a webpage, as its not far different
18:42.42plikgoogle does, the book will help with call file syntax
18:43.29James|TCCah cool
18:48.37styelzheh
18:49.27styelzer wrong square
18:49.31plikJames|TCC: thinking about it, if you're using php you could prolly do something in AGI - again see the book for details
18:49.50James|TCCAGI?
18:51.05plikAsterisk Gateway Interface
18:51.15plik~book
18:51.15jbotAsterisk: 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
18:51.24plikor voip-info.org
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19:00.40*** part/#asterisk VoipMasta (n=fabio@201.139.138.246.cable.dyn.cableonline.com.mx)
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19:12.38adr3nalin3hi guys, I am trying to verify that I have correctly installed a digium TDM800p.  When I run ztcfg -vvvv I get 8 channels to configure at the bottom.  Is there something else I need to do?  Also when I go into the asterisk CLI and type 'zap show channels' all I see is a psuedo channel.
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19:21.28styelzadr3nalin3: try running genzaptelconf
19:22.03styelzthen restart zaptel
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19:38.28adr3nalin3styelz: I have done so, but I am getting the same output
19:39.47styelzdo you see anything in /proc/zaptel ?
19:40.00styelzand have /dev/zap ?
19:40.23styelzmaybe cat /proc/zaptel/1
19:41.16adr3nalin3styelz: Yes when I cat /proc/zaptel/1 it shows 8 channels "1 WCTDM/0/0 FXSKS
19:41.41adr3nalin3and I yes I do have /dev/zap
19:42.21styelzand they are set in /etc/zaptel.conf ?
19:42.35styelzk
19:42.57adr3nalin3such as fxsks=1  fxsks=2 and so on?
19:43.00adr3nalin3yes
19:43.07styelzwhat about /etc/asterisk/zapata-auto.conf
19:43.16styelzor is it zapata-channels.conf
19:43.24styelzare yo including the correct file
19:43.33styelzand are they set in there
19:44.28styelzyou may have #include zapata-channels.conf  in zapata.conf
19:44.42tzafrir_laptopls -l /etc/asterisk/zapata*.conf
19:45.14adr3nalin3the zapata-channels.conf looks like it is all setup
19:45.18styelzok
19:45.29styelzdoes zapata.conf have #include zapata-channels.conf ?
19:45.32styelzit needs it
19:45.37adr3nalin3checking
19:45.41styelzat the end somewhere
19:45.48styelzpossibly
19:46.34styelzthat should do it
19:49.12adr3nalin3I have added that but I still get the same output.  I have to go for now.  thank you for all your help
19:49.22styelzok good luck
20:00.04mchouanyone here ever subscribe to voipjet?
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20:08.33Strom_Cmchou: years ago
20:09.29mchouStrom_C
20:09.45mchouStrom_C: Do they give you a SIP address?
20:10.17mchouStrom_C: I couldn't find this on their pages.....
20:10.37Strom_Ca SIP address for what purpose?
20:11.12mchouStrom_C: so ppl can call in to my SIP address (as an example)
20:11.22Strom_C...
20:11.29mchouas opposed to a bona fide DID #
20:11.29Strom_Cyou don't need an ITSP for that
20:12.29mchouStrom_C: That's like saying I dont need the telco to get a phone #
20:12.34Strom_Cno, wrong
20:12.40Strom_Cyou have an IP address
20:12.59Strom_Cyou can very easily have people call your box DIRECTLY
20:13.28Strom_Cthe whole point of the ITSP is for PSTN interconnect
20:13.40Strom_Cif you just want people to call you directly via SIP, you don't need the ITSP
20:14.03mchouthis is not enterprise level stuff....IPaddr is dynamic (via ISP)
20:14.23Strom_Cdid I ever claim this was enterprise-level stuff? :)
20:14.29mchouand dont really want to be messing with dyndns and crap
20:15.06Strom_Cno ITSP that I know of is going to provide you with an address like that
20:15.44Strom_Cif you want to allow unauthenticated SIP calls to your PBX, you must do that yourself, either via dyndns (or similar) or some box with a static IP address
20:25.57*** join/#asterisk nirz (n=nir@bzq-79-177-133-58.red.bezeqint.net)
20:27.28nirzhello, i'm trying to set an iax trunk between asterisk appliance and other asterisk 1.2, from some reason i having problem to send the caller id ... do i need to make any modification because of the appliance ?
20:51.47*** join/#asterisk Pctech37|Mac (n=Pctech37@unaffiliated/pctech37)
20:52.13Pctech37|MacCan I join the project?
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20:55.50*** join/#asterisk RoyK (n=roy@ip-194-54-149-91.dialup.ice.no)
20:56.17variable_officeI am trying to get chan_mobile to work with lg vx5800(chocolate) but it is not working; one thing I noticed is that it is coming up as a "handset"
20:56.22variable_officeerr headset
20:56.44variable_officeand usable=no
21:09.21variable_officehas anyone here even gotten chan_mobile to work?
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21:18.20florztzafrir_laptop: pong
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21:28.09qwertyfcukerwhats up folks, what's the good word on asterisk 1.6?
21:28.15qwertyfcukerneed SIP over TCP!
21:28.37implicituse openSER u can use it as a proxy to 1.4
21:28.51qwertyfcukeryeah yeah i know the workarounds :)
21:28.55implicittls, tcp, etc
21:29.00impliciti only use asterisk as a media server
21:29.18implicitdo all routing, accounting, sdp mangling, normalization in openser
21:29.20qwertyfcukeryou use openSER for telephony?
21:29.26implicitabsolutel
21:29.40impliciti can do 10,000 calls per second on that thing
21:30.02qwertyfcukerwow
21:30.12implicitprobably the most powerful sip proxy ever, its more like a SIP scripting language
21:30.13qwertyfcukeryou done microsoft UM integration?
21:30.16implicityeah
21:30.18impliciti have
21:30.30implicitbut you will need another component too if you want to do REFERs properly
21:30.33implicitb2bua
21:30.39qwertyfcukerk
21:30.45qwertyfcukerhow is the MS UM voicemail btw?
21:30.57impliciti don't know ive just done integration on a consulting basis
21:30.58qwertyfcukerasterisk's VM is absolutely awful
21:31.08impliciti wrote my own VM in python
21:31.14qwertyfcukernice
21:31.16implicitand tied it into SEMS
21:31.19qwertyfcukerusing tts?
21:31.31implicitu seems sems?
21:31.35implicitiptel.org/sems
21:31.40qwertyfcukerlemme have a look
21:31.42implicitbut it's not really good if your not a programmer
21:31.52implicitasterisk is a much easier to use media server and for general purpose asterisk is good
21:32.16impliciti jsut don't use it for SIP routing cause it's not a sip proxy. it doesn't make sense to make new dialogs and deal w/ media etc when doing sip routing if you don't have to
21:32.17qwertyfcukeri'm a decent programmer - just getting into the telephony/voip side of things now
21:32.42qwertyfcukerthere was some Mono AGI thingie that caught my eye the other day and I started playing w/ asterisk
21:32.48implicitah cool
21:32.53implicitwhat are you trying to develop
21:33.09qwertyfcukerbasically a call in timeclock solution for home health aides
21:33.17qwertyfcukerand any other type of industry really
21:33.22qwertyfcukertemp workers, etc
21:33.28implicitoh ok
21:33.41qwertyfcukercall from a location from a range of numbers, verify the # and clock the punches in and out
21:33.51qwertyfcukerthen pump out a file for payroll
21:33.57qwertyfcukerpretty basic really
21:33.58implicithow are you integrating to PSTN?
21:34.22qwertyfcukeri've been playing with a sangoma a200 card w/ 4 lines in a hunt group
21:34.34impliciteasier to just get it over SIP, isn't it?
21:34.36qwertyfcukerthinking about maybe using a SIP trunk provider instead
21:34.39implicityeah
21:34.40qwertyfcukerdefinately cheaper
21:34.47implicityep
21:34.51qwertyfcukerthe POTS call quality seems much better though
21:34.54implicitdo you ened a lot of phone #s?
21:34.59qwertyfcukerdon't think so really
21:35.19qwertyfcukerSIP would suit my needs just fine
21:35.35implicitfor 100 contiguous or so i can give you $50/mo
21:36.04qwertyfcukerwow that's probably more than i'd need for the foreseeable future
21:36.05qwertyfcukerlol
21:36.09implicitlol
21:36.10qwertyfcukerbut a great deal
21:36.15Strom_Cqwerty: your SIP calls will sound like PSTN quality if you use the right codecs
21:36.38implicitStrom_C: absolutely, if not better cause of impedence and grounding isseus of some hardware
21:37.04qwertyfcukerg729?
21:37.12Strom_Cqwertyfcuker: ugh, no
21:37.13implicitg729 is toll quality
21:37.15Strom_Cg711u
21:37.15implicitfor sure
21:37.20Strom_Cno, g729 is ass quality
21:37.25qwertyfcukerlol
21:37.29implicitStrom_C, absolutely not
21:37.41Strom_Cimplicit: it sounds like shit to me
21:37.41qwertyfcukerulaw is probably fine
21:37.46Strom_CI even hate talking on my mobile phone
21:37.51qwertyfcukerthis is true....
21:37.52implicitwhat is your setup with it though
21:38.03qwertyfcukerbasically employees would call a number to clock in
21:38.04implicitcause it soudns WAY better than my gsm phone
21:38.06*** join/#asterisk dFence (n=chatzill@ings-d93226b8.pool.mediaWays.net)
21:38.13qwertyfcukerit's not like they have to have amazing call quality anyway
21:38.22qwertyfcukerthey're not calling 1900spankalot
21:38.39implicitif you have other issues, like resizing rtp frames, messed up jitter buffers etc
21:38.42implicityeah it will sound like ass
21:38.58qwertyfcukerwell what would matter would be if IVR didn't work
21:39.02Strom_Cimplicit: heh, any lossy codec sounds like crap over the phone to me
21:39.08implicitotherwise g729 has a mos score of 4.1
21:39.12implicitversus 4.4 for g711
21:39.23implicitalso disable VAD
21:39.26Strom_Cthat's because most people are ignorant :)
21:39.28implicitthat's the biggest issue w/ bad quality of g729
21:39.41implicitbad VAD implementations
21:40.02qwertyfcukeris sorry for starting a codec pissing contest
21:40.02Strom_CI'm just particularly picky about the subtleties of what it does to the audio
21:40.02implicitthen don't use g711
21:40.02implicituse g722
21:40.02implicithaha
21:40.02implicitand call ip-ip only
21:40.06implicitso that you don't get cut off at 4khz
21:40.09Strom_Csigh...missing the point
21:40.13*** join/#asterisk magic_hat (n=geoffdou@h-68-164-10-43.chcgilgm.dynamic.covad.net)
21:40.35implicitperhaps, but g729 itself especially if you do IP-IP is extremely good
21:40.37*** join/#asterisk bmg505 (n=leon@196-209-8-2-ndn-esr-2.dynamic.isadsl.co.za)
21:41.09implicitif your going to PSTN it's really hard to isolate the isseus to say what it really is
21:41.15magic_hathowdy folks. I'm seeing two entries in the logs for an autodialing app that I have. One is SIP response 503, service unavailable. The other is CONGESTION... everyone is busy. What's going on with these?
21:41.20implicitif you have the bandwidth then use g711
21:41.20Strom_Cheh, no, even just running it over my local LAN from my polycom phone to my asterisk box, I don't like the sound of g729
21:41.34implicittry it without your * box
21:41.39implicit* sometimes resizes frames etc
21:41.40Strom_Ctried that too.
21:41.41implicitmakes it sound ass
21:41.56implicithave rtp go point-to-point
21:42.20Strom_Clook, i know you think g729 is the holy grail of voice codecs, but I've tried it dozens of different ways and I don't like the vocoding.
21:42.28impliciti never relay RTP when unnecessary, i let it go point to point as much as possible
21:42.52implicitit's not that, it's just that people blame audio issues more on codecs than other problems that are generally teh real sources
21:43.03qwertyfcukerheh, i spoke to the folks at bandwidth.com and they told me the same thing implicit
21:43.07Strom_CI understand what you're arguing.
21:43.12Strom_CI don't like the vocoders.
21:43.14qwertyfcukeri was looking at a solution that involved MPLS and lots of NATing
21:43.36qwertyfcukerand they said it was better just to use a level 3 ISP w/ a T1 and let it go over the public internet to them
21:44.10*** join/#asterisk beek (n=klinebl@65.211.106.242)
21:44.35magic_hatEveryone is busy/congested at this time (1:0/1/0)... does this mean my VOIP is busy? I've exceeded my allotted number of channels? Or the phone is off the hook o the other end?
21:44.47Strom_Cmagic_hat: congested circuit
21:45.15magic_hatStrom_C: Anything I can do about that? Or it's a problem at Teliax?
21:45.32Strom_Cincrease your timeout
21:45.44Strom_Cit could be that teliax is just slow to respond
21:45.52magic_hatah, okay.
21:46.27beekFWIW, I just placed a call through Teliax with no problem.
21:46.51magic_hatbeek: Yeah, it's generally working. Just wondering what the occasional error messages mean.
21:47.43qwertyfcukerhah I just got my father in law to take my wife to sex and the city......PWNED
21:48.17qwertyfcukernn fellas
21:48.26*** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net)
21:48.42magic_hatStrom_C where do I set that timeout.... not seeing it in sip.conf
21:49.22seanbrightis it just me, or does reading some of the long threads on the -users list make anyone else want to stab themselves in the eyes?
21:56.23jblackI'm not inclined to blame teliax. I've never seen a problem with them.
21:58.26magic_hatso... if not them, where's the congestion?
21:59.22jblackbad number, the number is busy?
21:59.35jblackMisconfiguration of your system?
22:00.03magic_hatjblack: system's working fine on most calls. But you're saying I can get an 'everyone is busy' if it's a bad number?
22:00.05jblackBump up your verbose and debug settings on the console, and watch the call try to place.
22:00.39jblackpastebin it if you like
22:04.17magic_hatI'm doing random-digit dialing, so I know a bunch of #'s are going to come back bad. I just wasn't sure if the 'congestion' notice was that, or something else.
22:05.38Strom_Crandom-digit dialing?
22:06.19magic_hatyeah, for public opinion polling. take a list of valid exchanges, add a random 4 digits on the end, and you have a random sample.
22:06.52Strom_CI hope to God you're not calling mobile phones and people on the do not call list
22:07.09magic_hatthere's an exemption for public opinion research.
22:07.13implicitit doesn't count as telemarketing
22:07.17magic_hatand we're a nonprofit, and exempt because of that, too. lol
22:07.38Strom_Cdie in a fire :)
22:07.48jblackI hope you get syphillis.
22:08.29magic_hatlol. we're doing some really important work that's in the public interest. and I already have syphilis :)
22:08.41jblackThen I hope you get herpes too.
22:08.58magic_hatwe'll make sure you're on the call list!
22:10.51*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)
22:10.52Strom_CI always answer the opinion polls out of curiosity, and I've NEVER answered one that wasn't full of ambiguous, badly-written, and loaded questions.
22:11.17jblackstrom: Don't answer opinion polls. They establish a business relationship that can be used to call back and telemarket with.
22:12.06Strom_Cjblack: the last one I received was three telephone numbers ago :)
22:13.24Strom_Cbut yeah
22:13.35Strom_Cthe opinion polls are always badly-written and terribly biased
22:13.40*** join/#asterisk Fuzix (n=fuzix@250-118.citynet.ftth.internl.net)
22:13.41Strom_Cso fuck that shit
22:14.48jblackHeh, you think opinion polls are used to survey what people think?
22:14.54Strom_Cof course not
22:15.01Strom_Ci'm not that naive :)
22:16.29jblacksome day I'm going to make a survey, and call every mcdonald's in the land with one question "Are you comfortable with your participation in feeding america unhealthy food"
22:17.49Strom_Cheh
22:18.44jblackThen, I'll follow up by calling every red cross with "Are you happy to personally profit from the practice of selling for $600 a pint blood that people gave you for free?"
22:21.15*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
22:21.15*** mode/#asterisk [+o russellb] by ChanServ
22:21.20*** join/#asterisk ^^Johnny5 (n=JeanCote@dsl-67-55-16-142.acanac.net)
22:21.49JayTee52"wouldn't you like to be a Pepper too?"
22:22.07magic_hatStrom_C: we're actually pretty good at writing them to find out how people think about policy issues without biasing them. But yeah, there's a lot o'crap out there.
22:22.48Strom_Cmagic_hat: I'll reserve judgment until I actually read one of your surveys
22:23.19magic_hatlol if I hadn't busted the code that runs the dial-in number, I'd give it to you.
22:23.39^^Johnny5anyone know how to debug a digium tdm400P board with 4 pots (green) interface daughter boards?
22:24.04^^Johnny5was working and does not anymore...rrr
22:24.07JayTee52the green modules are called FXS modules
22:24.11Strom_C^^Johnny5: what's the actual problem you're having?
22:24.12^^Johnny5yes
22:24.49Strom_Cmagic_hat: i said "read," not "call" :)
22:25.28^^Johnny5the same boards (2x) were purchased in jan 08 and all of a sudden no dial tones on my pots anymore... I go in asterisk there is no zap* commands there... zapta.conf registers 1 board only rrr
22:26.52Strom_C^^Johnny5: pastebin your zapata.conf and zaptel,conf files
22:26.54^^Johnny5I have setup a ssh route to the box from jcdirect.no-ip.org (my home)
22:26.56russellbif there are no zap commands, then chan_zap didn't get loaded, or failed to initialize ...
22:27.03Strom_C~pb
22:27.04jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
22:27.04russellbzaptel itself probably isn't properly configured
22:27.24implicitor even better, dpaste
22:27.33implicitdpaste.com is faster than all of them
22:27.40Strom_Ccongratulations
22:27.41implicitit takes only 3.3ms to process
22:27.53^^Johnny5I am currently on a Win XP Pro using opera
22:28.02implicitcongratulations
22:28.05Strom_C^^Johnny5: just pastebin please
22:28.58russellbmy pastebin is faster than your pastebin
22:29.20Strom_Ci've got the fastest pastebin in six counties
22:31.29*** join/#asterisk minime (n=afg_ch@84-73-144-128.dclient.hispeed.ch)
22:31.36^^Johnny5# Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
22:31.36^^Johnny5# Zaptel Configuration File
22:31.36^^Johnny5#
22:31.36^^Johnny5# This file is parsed by the Zaptel Configurator, ztcfg
22:31.36^^Johnny5#
22:31.36^^Johnny5# It must be in the module loading order
22:31.38^^Johnny5# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
22:31.40^^Johnny5fxoks=1
22:31.42^^Johnny5fxoks=2
22:31.43Strom_Coh christ
22:31.44^^Johnny5fxoks=3
22:31.44Strom_CPASTEBIN
22:31.46Strom_Cnot paste
22:31.46^^Johnny5fxoks=4
22:31.48^^Johnny5# Global data
22:31.50^^Johnny5loadzone= us
22:31.52Strom_Crussellb: kick plz
22:31.52^^Johnny5defaultzone= us
22:32.06implicitlol jut let him do it
22:32.08implicitit's alright
22:32.19Strom_C^^Johnny5: PASTEBIN PASTEBIN PASTEBIN
22:32.21Strom_C~pb
22:32.22jbot[~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
22:32.27mchoulol
22:32.40implicithttp://www.dpaste.com
22:32.44Strom_Cdraws a circle on the whiteboard and bangs his head into the centre of it
22:32.47^^Johnny5never used pastebin I am there I must create an account ?
22:32.50implicitno
22:32.53implicitplease just go to the site
22:32.54implicitand SEEEEEEEEEEEEEE
22:33.09implicitSEE
22:34.04^^Johnny5juste pasted in dpaste
22:34.11implicitok now give the link to us
22:34.25^^Johnny5http://dpaste.com/hold/59834/
22:34.31implicitperfect
22:34.36implicitbut we already saw this in the channels
22:34.42implicitso we didn't need to see it agan
22:34.53implicit*channel
22:35.32Strom_C^^Johnny5: ok -- now what happens when you run "ztcfg -vv" at the bash prompt?
22:35.34^^Johnny5I don't understand....
22:35.44implicit^^Johnny5: you are a nice guy, don't be scared
22:35.58^^Johnny5do you see the zaptel.conf contents in dpaste?
22:36.07VeggenJohnny: yes, we do.
22:36.15^^Johnny5how can I check/debug my board...
22:36.27Strom_C^^Johnny5: follow my instructions
22:36.34Strom_C^^Johnny5: what happens when you run "ztcfg -vv" at the bash prompt?
22:37.34^^Johnny5http://dpaste.com/59835/
22:39.07Strom_Crestart asterisk and see if your "zap" CLI command shows up
22:39.07^^Johnny5Besides this problem the PBX is working fine with XLite...
22:39.20^^Johnny5ok I will try
22:39.26Strom_Cjust asterisk
22:39.28Strom_Cnot the whole box
22:39.35^^Johnny5how is that done
22:39.47Strom_Cconnect to the asterisk CLI and type "restart now"
22:39.50^^Johnny5asterisk -r stop now
22:41.51^^Johnny5http://dpaste.com/59836/
22:41.58^^Johnny5no zap*
22:42.27Strom_Cpastebin your zapata.conf
22:42.44adr3nalin3this problem above is the same problem I have with a tdm800p
22:42.52adr3nalin3jumps in
22:43.06Strom_Cadr3nalin3: wait till I fix this one
22:43.24^^Johnny5http://dpaste.com/59837/
22:43.42adr3nalin3im just gonna watch for now
22:43.56Strom_C^^Johnny5: uhm...are you using trixbox?
22:45.27russellbi could have guessed that a long time ago :-p
22:45.32^^Johnny5yes stable ver 2.4
22:45.38russellb~trixbox
22:45.39jbot[~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org
22:46.14^^Johnny5do youi suggest I use asterisk....
22:47.42russellbalmost anything but trixbox
22:47.44russellbbut yes
22:47.57russellbusing asterisk directly is just about all you'll get help with in this channel
22:48.08^^Johnny5can you tell me why?
22:48.20russellb~trixbox
22:48.21jbot[~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org
22:48.21*** join/#asterisk d-k-t (n=dt@125.120.132.230)
22:49.17^^Johnny5yes I understand that part...
22:49.20beek^^Johnny5: My first * box was Trixbox.   It took me longer to figure out how to get it to do what I wanted (or to customize it) than it did just to load a fresh install of CentOS, then Asterisk, and configure it.
22:49.47^^Johnny5oh yeah well that hits me!!
22:49.54^^Johnny5thanks...
22:51.15^^Johnny5if I want to go asterisk then what about web interface like freePBX... is it ok?
22:51.35Strom_Cno
22:51.37Strom_C~freepbx
22:51.38jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
22:51.44beekThat's where things get complicated.
22:51.55^^Johnny5ok, thanks
22:51.58Strom_Clearn to write your own dialplan
22:52.00Strom_C~book
22:52.01jbotAsterisk: 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
22:52.29^^Johnny5thanks for the pdf... and links
22:52.30beekfreepbx is attractive because it seems to be a quick way to a working system  but the code that it writes is spaghetti code.
22:52.42beekIt's a bitch to work on.
22:52.49Strom_Cspaghetti is a nice way of describing that mess
22:53.56^^Johnny5lol
22:54.38beekThe other thing Johnny is that by learning to configure Asterisk by hand gives you the opportunity to load it on other platforms not supported by Trix/FreePBX, such
22:54.50beekas the little Asus router I have running * at home.
22:55.58adr3nalin3could someone give me a hand with a problem similar to ^^Johnny5's?  I am using a TDM800P.  It appears that my channels are not configured.
22:56.20Strom_Cadr3nalin3: pastebin your zaptel.conf and your zapata.conf
22:59.26*** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net)
22:59.55*** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de)
23:00.00*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
23:00.45adr3nalin3Strom_C: zapata.conf,  http://pastebin.com/m5fcf9562  The only thing I have changed in there is I added #include zapata-channels.conf
23:02.11Strom_Cok, and where are the contents of zapata-channels.conf?
23:03.08*** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com)
23:03.35adr3nalin3here --> http://pastebin.com/m31661c55
23:04.25Strom_Cok, and what about zaptel.conf?
23:05.02*** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de)
23:06.44*** join/#asterisk craigk (n=craigk@58.174.150.119)
23:06.49adr3nalin3zaptel.conf -> http://pastebin.com/m56f0907f
23:07.19Strom_Cum, no
23:07.30Strom_Cwell, i suppose that will work
23:07.50adr3nalin3gathered from reading http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf
23:07.53Strom_Cbut, just for simplicity's sake, change that to "fxsks=1-8"
23:08.00adr3nalin3ok
23:08.21Strom_Cjust to clarify, you've got FXO (red) modules, right?
23:08.28adr3nalin3correct
23:09.12Strom_Cwhat happens when you run "ztcfg -vv"?
23:09.55adr3nalin3http://pastebin.com/m6f812208
23:10.11Strom_Cok
23:10.14Strom_Cnow restart asterisk
23:10.47adr3nalin3ok
23:10.58adr3nalin3done
23:11.12Strom_Cdoes it appear to work now?
23:11.21adr3nalin3same output
23:11.27adr3nalin3^^ from ztcfg -vv
23:11.28Strom_C?
23:11.38Strom_Cthat response makes no sense
23:11.46adr3nalin3I know...
23:11.59Strom_Cnow that you've restarted ASTERISK, do your channels appear to work now?
23:12.14*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:12.53^^Johnny5do you type in help in the asterisk prompt? asterisk -r , help
23:13.07adr3nalin3they are all showing just as they did in the previous ztcfg -vv pastebin but it still says 8 channels to configure.  Is that the expected output?  I was expecting 8 channels configured
23:13.18Strom_Cadr3nalin3: so wait
23:13.24Strom_Cyou didnt actually test them?
23:13.33Strom_Cand just assumed from the output that it was broken?
23:13.39Strom_CTEST THEM
23:13.41*** part/#asterisk minime (n=afg_ch@84-73-144-128.dclient.hispeed.ch)
23:14.26^^Johnny5how?
23:14.45adr3nalin3the box is 150 miles away.  I am trying to make sure it is ready to go when I arrive next week
23:15.07Strom_Cadr3nalin3: um, ok?
23:15.21Strom_Clog into asterisk and type "zap show channels"
23:15.45^^Johnny5do you have zap* commands when you type help?
23:15.54adr3nalin3Strom_C: they DO show up now!
23:15.59adr3nalin3thank you!
23:16.06adr3nalin3they did not previously
23:16.18^^Johnny5thanks guys!!
23:16.56^^Johnny5it seems I have faulty boards since they were working for 6months at least... thunder!
23:17.20Strom_C^^Johnny5: or freepbx fucked up.
23:17.29Strom_CI would blame trixbox before I blame the hardware
23:17.34^^Johnny5is there a way to check the boards...
23:18.45^^Johnny5I have used another hard disk and redone the config as previously... in jan 08 ... on one of the lines I here a tone very low.... with white noise
23:19.26Strom_C^^Johnny5: you're using trixbox
23:19.45Strom_Cthat in and of itself is probably at least 97% of the problem
23:20.48^^Johnny5well it was working for a while... no auto updates either... this happened after they told me that thunder striked about 500 meters (3x=ft) away
23:21.11Strom_C...you honestly think I'm so ignorant that I don't know what a meter is?
23:21.23^^Johnny5hum sorry
23:21.53^^Johnny5no offence in mind just respect...
23:22.03Strom_C...
23:22.12Strom_Cyeah, I believe that one </sarcasm>
23:22.20Strom_Ccall digium support, although they're not going to be much help if you're running trixbox
23:22.21*** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net)
23:22.42*** join/#asterisk xlogik (n=xlogik@c-71-232-176-24.hsd1.ma.comcast.net)
23:23.08^^Johnny5ok, then I will install what they require me to install right as per their web site... all dev stuff ?
23:23.23Strom_Cno, you don't need to install the dev stuff
23:23.40Strom_Cjust go with whatever linux distro you like best and then a release version of asterisk/zaptel
23:23.58^^Johnny5which one do you sugggest...
23:24.38^^Johnny5linux distro
23:24.47Strom_CI like debian
23:24.57Strom_Cbut personal preference will dictate which one you choose
23:26.16^^Johnny5I have fedora 8 working on another machine here
23:26.25^^Johnny5is that ok?
23:26.32Strom_Care you listening to me?
23:26.34Strom_CIT DOESN'T MATTER
23:26.39Strom_Cuse whatever you like best
23:26.47^^Johnny5thanks...
23:27.56adr3nalin3~snom
23:27.57jbotrumour has it, snom is like all German products. High quality, but wacky engineering. :)
23:29.35*** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de)
23:31.54*** join/#asterisk moos3_ (n=richard@cpe-204-210-72-206.maine.res.rr.com)
23:32.04moos3_is this a good card http://store.pbxeq.com/store/product.php?productid=17104&cat=283&page=1
23:32.38*** join/#asterisk Strom_C (n=strom@208.127.172.112)
23:34.24Strom_Cmoos3_: i'd advise you not buy that one
23:34.38JayTee52it's a clone knockoff of a Digium card.
23:35.00moos3_Strom_C: what do you recommend then
23:36.06Strom_Cmoos3_: a Digium TDM410
23:36.36moos3_Strom_C: I would have i had that kind of cash
23:36.58adr3nalin3moos3_: e4 technologies has digium cheap
23:37.05Strom_Clol
23:37.13Strom_Cancient crappy digium shit for cheap, maybe
23:37.19*** join/#asterisk levi (n=user@166.70.39.219)
23:37.20JayTee52buying cheap hardware is the easiest way to mess up your system
23:37.41adr3nalin3Strom_C: all the stuff I have gotten from them has been up to date and worked great
23:37.46moos3_yeah but I have to voip four offices for less then 1k
23:37.53*** part/#asterisk levi (n=user@166.70.39.219)
23:37.57adr3nalin3moos3_: good luck with that
23:37.58moos3_thats for all the hardware
23:38.00JayTee52what moron decided on 1K as a budget?
23:38.04Strom_Cmoos3_: lololololololololololololololololololol
23:38.30adr3nalin3moos3_: I fucking hope that doesn't include phones...
23:38.43moos3_Strom_C: i have read stuff tht zapmicro cards work great
23:38.47*** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de)
23:39.01moos3_no phones dont come out of the IT budget
23:39.02moos3_:)
23:39.06JayTee52I've read Lord of the Rings but I don't believe in hobbits
23:39.13adr3nalin3I wouldn't go with anything except digium, I bought sangoma one time and returned it the next day.
23:39.33adr3nalin3the drivers wouldn't even install
23:41.01jblackmoos3_: On a budget of 1K, you'll have to go with voip.
23:41.40Strom_Con a budget of 1k, I'll give you some cans and string
23:41.43adr3nalin3moo3_: sip trunking?
23:42.28moos3_yeah
23:42.30*** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan)
23:42.43adr3nalin3moos3_: Do you already have the machines to use?  4 boxes will be > $1k
23:42.54JayTee52I thought there wasn't any such thing as a sip trunk
23:43.18adr3nalin3qwest has it, they call it IPLD.
23:44.00moos3_yeah I have the boxes in place because they are old poweredges that we use to for staging
23:44.21jblackmoos3_: Then blow the $1k on 4 dedicated dsl lines, and a voip provider.
23:46.15moos3_i would do that i'm required by law to have dedicated hardlines
23:46.46adr3nalin3moos3_: for 911>
23:46.48adr3nalin3?
23:47.47Strom_Cmoos3_: then ask for a budget increase, because you don't want to run your business on some shit clone of a discontinued card
23:48.11adr3nalin3moos3_: or get one pstn and a red analog phone per office
23:48.22moos3_lol
23:48.43moos3_so the only aterisk cards that will work are nothing but digium
23:49.05adr3nalin3digium wrote asterisk.
23:49.22moos3_if there isn't any other way but digium then they will have to give me the increase
23:49.33adr3nalin3I would be willing to bet their cards work the best
23:49.53moos3_but if there is other cards that will work and my boss finds out my ass is gonna get crewed
23:49.54Strom_Cmoos3_: trust me -- being unreasonably cheap now will cause you no end of headache and wasted money in the future
23:50.02jblackI've heard sangoma cards are the best.
23:50.28jblackThe rhini I "have" works also fine, now that it's set up.
23:50.32jblackrhino, that is.
23:50.33adr3nalin3^^ except there drivers don't always work.  Kind of a big problem
23:51.01jblackafaik, All of the cards are a disaster if you care about free software.
23:56.31*** join/#asterisk mwalling_ (i=mwalling@you.dontlike.us)
23:58.12LiNeTuX|HomeRhino cards are a pain in the butt.  Even their own tech support has issues getting them working.

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