IRC log for #asterisk on 20070926

`00:00.01`	`bintut`	`ManxPower: there's no content inside the cdr-csv`
`00:00.07`	`ManxPower`	`bintut: then you disabled it.`
`00:00.36`	`bintut`	`ManxPower: but i have contents and keeps on updating my /var/log/asterisk/cdr.db`
`00:00.46`	`ManxPower`	`bintut: then you enabled that`
`00:00.57`	`*** part/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`00:01.02`	`ManxPower`	`unless they did something stupid in 1.4 that turned off text CDRs`
`00:02.23`	`*** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es)`
`00:02.32`	`ManxPower`	`bintut: your asterisk install is not a trixbox or asterisk-gui or something like that, is it?`
`00:03.07`	`bintut`	`ManxPower: it's the asterisk-1.4.11 from the debian unstable i386`
`00:03.25`	`ManxPower`	`bintut: now you know why we don't install from packages`
`00:03.57`	`bintut`	`:(`
`00:03.59`	`BockBilbo`	`hello again. Is it possible to run the Saydigits command inside a background command?`
`00:04.10`	`ManxPower`	`BockBilbo: no`
`00:04.18`	`BockBilbo`	`:/`
`00:04.48`	`BockBilbo`	`so the only way to reproduce digits in background is by adding them one by one?`
`00:05.25`	`ManxPower`	`you could write a macro to parse out each number, then background the soundfile for that digit`
`00:05.48`	`*** join/#asterisk luke-jr (n=luke-jr@2002:48ce:72ec:0:20e:a6ff:fec4:4e5d)`
`00:05.54`	`ManxPower`	`seems like a lot of work`
`00:06.07`	`luke-jr`	`So anyone know why Asterisk insists on playing hold music at 10x the volume of the call? ☺`
`00:06.08`	`ManxPower`	`you could also do it in an agi`
`00:06.27`	`ManxPower`	`luke-jr: maybe YOUR asterisk does, but mine does not.`
`00:06.31`	`BockBilbo`	`what do you mean by "doing it as an agi"?`
`00:07.18`	`ManxPower`	`BockBilbo: I mean write an AGI to take the number, parse out each digit and Background the actual sound file for that digit. I don't know how well that would work (I don't know if you can run background from an AGI)`
`00:07.29`	`ManxPower`	`I macro would be better, I think.`
`00:08.12`	`BockBilbo`	`ok, so i'll try the macro`
`00:08.14`	`BockBilbo`	`thanks!`
`00:09.32`	`Ritzerisk`	`anyone ever get to config the vicidial with *`
`00:09.32`	`s34n`	`RazaMetaL: so you've got your answer, right?`
`00:09.42`	`rob0`	`luke-jr: Eh sonny? Whazzat you say?`
`00:09.45`	`*** join/#asterisk fujin_ (n=aj@unaffiliated/fujin)`
`00:09.46`	`RazaMetaL`	`s34n, no :(`
`00:10.08`	`RazaMetaL`	`s34n, I'm looking for a /etc/default/zaptel file that I do not have`
`00:10.34`	`bintut`	`ManxPower: thanks for the help..`
`00:10.42`	`*** join/#asterisk jm\|home (n=jm\|home@dilbert.jamiem.com)`
`00:10.44`	`bintut`	`i have to go now.. see you later..`
`00:11.04`	`s34n`	`RazaMetaL: you said you have an /etc/init.d/ something or other linked into runlevel 2`
`00:11.27`	`s34n`	`RazaMetaL: but you computer is running in runlevel 3`
`00:11.37`	`RazaMetaL`	`s34n, yes.. I've one /etc/init.d/zaptel file running at runlevel 3`
`00:12.00`	`*** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net)`
`00:12.36`	`*** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9e056f92bccce2e4)`
`00:12.51`	`RazaMetaL`	`s34n, S20zaptel -> ../init.d/zaptel`
`00:12.53`	`*** join/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca)`
`00:12.59`	`Voicemeup`	`http://bugs.digium.com/view.php?id=9044&nbn=19`
`00:13.01`	`RazaMetaL`	`s34n, my computer is running at runlevel 3`
`00:13.14`	`Voicemeup`	`check_auth: username mismatch, have <trunk_179>, digest has <s>`
`00:13.15`	`Voicemeup`	`this bug..`
`00:13.16`	`*** join/#asterisk Rants (i=Rants@ool-182f4d05.dyn.optonline.net)`
`00:13.18`	`Voicemeup`	`any way around it ?`
`00:13.23`	`Voicemeup`	`registering 1.4 to a 1.2`
`00:13.24`	`s34n`	`RazaMetaL: do you have an /etc/sysconfig/zaptel?`
`00:13.29`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`00:14.07`	`RazaMetaL`	`s34n, humm....`
`00:14.17`	`RazaMetaL`	`danm.. ubuntu use /etc/sysconfig ?`
`00:14.21`	`Voicemeup`	`foudn it insecure=very`
`00:14.30`	`RazaMetaL`	`my debian is using /etc/default/zaptel`
`00:14.53`	`Rants`	`Im looking for a good sip provider who allows multiple channels in and out, hopefully with a flat fee.`
`00:15.04`	`Rants`	`any one know of a good one?`
`00:16.06`	`*** join/#asterisk tuxd00d (n=tuxinato@128.187.154.118)`
`00:16.35`	`RazaMetaL`	`s34n, I see the error now :)`
`00:16.58`	`RazaMetaL`	`i need to move the /etc/sysconfig/zaptel to /etc/default/zaptel`
`00:17.04`	`RazaMetaL`	`now is working`
`00:17.05`	`RazaMetaL`	`thank you`
`00:17.47`	`Rants`	`anyone have any opinions on providers?`
`00:18.21`	`RazaMetaL`	`Rants, for international or long distance?`
`00:18.38`	`Rants`	`US long distance, multiple channel in/out flat rate.`
`00:19.12`	`Rants`	`I want to do something like a conf. center.`
`00:19.22`	`Rants`	`so more in then out.`
`00:20.15`	`s34n`	`is anyone around that is willing to wade through a sip invite exchange to see why I 404 Not Found on calls from a peer?`
`00:20.57`	`RazaMetaL`	`Rants, check http://www.google.com/url?sa=t&ct=res&cd=5&url=http%3A%2F%2Fwww.voip-info.org%2Fwiki%2Fview%2FVOIP%2BService%2BProviders%2BBusiness%2BNorth%2BAmerica&ei=yKX5RoOtMpKwgATPhZXyDA&usg=AFQjCNEWsEhl02LtDiSISGSPD3NS8TwNHw&sig2=Njqsar_-HYXfQEgPk7ui7w`
`00:21.34`	`RazaMetaL`	`sorry . . http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business+North+America`
`00:21.36`	`rudholm`	`does the SIP control channel (5060) use UDP or TCP normally?`
`00:21.47`	`luke-jr`	`ManxPower: every * system I've worked with does`
`00:24.22`	`*** join/#asterisk saftsack (n=saftsack@pD9E05A91.dip.t-dialin.net)`
`00:24.40`	`*** join/#asterisk adr3nalin3 (n=andrewf@ip68-13-89-102.om.om.cox.net)`
`00:24.50`	`s34n`	`rudholm: udp`
`00:25.10`	`rudholm`	`s34n: I know the data is udp, but wasn't sure about the control channel`
`00:25.13`	`rudholm`	`thanks`
`00:25.38`	`s34n`	`rudholm: dbl check that`
`00:26.34`	`s34n`	`rudholm: but I've just spent the day sniffing 5060 control traffic that was all udp`
`00:27.04`	`*** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju)`
`00:27.21`	`s34n`	`and now my head is tired and wants to go home`
`00:33.55`	`*** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es)`
`00:34.44`	`*** join/#asterisk FireMac (n=mccasian@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)`
`00:34.55`	`FireMac`	`hi`
`00:35.00`	`FireMac`	`i need your guidance`
`00:35.14`	`FireMac`	`very new to asterisk,`
`00:35.37`	`FireMac`	`i have voip service`
`00:36.03`	`FireMac`	`i want asterisk to manage calls to my voip number`
`00:36.15`	`*** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net)`
`00:36.26`	`knarfly`	`howdy world`
`00:36.31`	`knarfly`	`8-)`
`00:36.40`	`FireMac`	`one task is to forward calls for example to my cell phone if the caller choose to talk to me in my cell phone`
`00:36.55`	`FireMac`	`hi knarfly`
`00:36.55`	`knarfly`	`can someone tell me why my incoming extension is hosing up: http://www.pastebin.ca/715082`
`00:36.59`	`*** part/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca)`
`00:37.26`	`knarfly`	`I've been toying around with this for awhile now and it appears I've gotten something stupid in it.`
`00:39.35`	`*** join/#asterisk brian (n=brian@unaffiliated/brian)`
`00:40.58`	`*** join/#asterisk brian (n=brian@unaffiliated/brian)`
`00:41.36`	`CCFL_Man2`	`rudholm: i picked up a green imperial WE202`
`00:42.11`	`RazaMetaL`	`arghh.. does not work :(`
`00:42.33`	`CCFL_Man2`	`rudholm: $103 :P`
`00:43.00`	`RazaMetaL`	`I've two cards, one e1 and one tdm with 4 fxs ports`
`00:43.44`	`CCFL_Man2`	`phone is beautiful, uses a 4H dial and a clear plastic fingerwheel with metal bracket for mounting the plastic fingerwheel`
`00:44.38`	`CCFL_Man2`	`5H i mean`
`00:44.59`	`CCFL_Man2`	`rudholm: you have experience with 5H dials?`
`00:46.38`	`rudholm`	`CCFL_Man2: my 202 has a 4H`
`00:46.56`	`rudholm`	`CCFL_Man2: with the metal fingerwheel`
`00:47.47`	`rudholm`	`CCFL_Man2: and a type E handset`
`00:48.00`	`CCFL_Man2`	`rudholm: but mine is an imperial model`
`00:48.14`	`rudholm`	`CCFL_Man2: the colored ones?`
`00:48.17`	`rob0`	`ot: anybody here on LUNA?`
`00:48.20`	`*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)`
`00:48.24`	`rudholm`	`CCFL_Man2: it's just painted, isn't it?`
`00:48.25`	`CCFL_Man2`	`the rare 50s remake of the 202 or WE's 75th anniversary`
`00:48.30`	`CCFL_Man2`	`right`
`00:48.36`	`rudholm`	`CCFL_Man2: yeah, I've heard of those`
`00:48.37`	`CCFL_Man2`	`mine is green :P`
`00:48.57`	`rudholm`	`CCFL_Man2: well, mine is an original :-p`
`00:49.14`	`rudholm`	`CCFL_Man2: but you did get a good price on that`
`00:49.35`	`CCFL_Man2`	`oh i know, but i was just boasting that i got a rare green imperial 202 :P`
`00:49.37`	`CCFL_Man2`	`i know`
`00:50.03`	`CCFL_Man2`	`i've never seen any color of imperials other than chrome or brass with ivory handset`
`00:50.49`	`CCFL_Man2`	`i took some rubbing alcohol and cleaned off the dirt, it shines so nicely now`
`00:51.20`	`CCFL_Man2`	`F1 handset of course, dated in the 50s, i forget which year`
`00:51.27`	`rudholm`	`yeah`
`00:51.41`	`rudholm`	`the F handsets sound a lot better than the E handsets`
`00:51.50`	`CCFL_Man2`	`i heard that`
`00:52.16`	`*** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net)`
`00:52.17`	`rudholm`	`the Fs sound as good as a G IMO. but the E is very tinny and too quiet`
`00:53.00`	`CCFL_Man2`	`hmm.. my F1 on my 302 sounds kind of muffled`
`00:53.15`	`*** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net)`
`00:53.50`	`CCFL_Man2`	`that normal?`
`00:53.54`	`rudholm`	`no`
`00:53.56`	`*** join/#asterisk ManxPower (n=manxpowe@209.16.72.135)`
`00:54.04`	`*** join/#asterisk mihinomenest (i=HB9Q@66.255.220.17)`
`00:54.26`	`CCFL_Man2`	`think the speaker needs changing?`
`00:54.52`	`rudholm`	`hard to say, could be a number of things, but that's definitely a possibility`
`00:54.59`	`jayk-`	`i just upgraded to asterisk 1.4.11 from 1.2 and i have lost my zap channel..it is Zap/g2 and doesn't work with 1.4.11. i get: No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66 - Channel not implemented)`
`00:55.03`	`jayk-`	`does anybody have any ideas?`
`00:55.26`	`CCFL_Man2`	`i'd hate to replace it and not have the speaker match in date`
`00:55.34`	`rudholm`	`does it match?`
`00:55.39`	`*** join/#asterisk PepOSX (n=pepOSX@190.72.149.199)`
`00:55.43`	`CCFL_Man2`	`yeah`
`00:55.57`	`CCFL_Man2`	`everything in my 302 was all dated the same`
`00:56.04`	`ManxPower`	`jayk-: you did not install zaptel 1.4.x before installing asterisk 1.4.x I assume you read upgrade.txt as well`
`00:56.11`	`CCFL_Man2`	`same with my green 202`
`00:56.20`	`jayk-`	`manxpower: i installed librpi, zaptel, and then asterisk.`
`00:56.27`	`ManxPower`	`jayk-: all 1.4.x?`
`00:56.29`	`jayk-`	`yues`
`00:56.33`	`rudholm`	`CCFL_Man2: does the F handset on your 302 sound muffled?`
`00:56.49`	`CCFL_Man2`	`rudholm: yeah`
`00:57.01`	`rudholm`	`CCFL_Man2: what about the F on your 202?`
`00:57.06`	`ManxPower`	`jayk-: then for some reason zaptel either did not actually install (a known bug in the latest zaptel 1.4.x on Debian) or asterisk did not correctly detect the zapte`
`00:57.08`	`CCFL_Man2`	`not sure with my 202 yet, i just heard dial tone so far`
`00:57.23`	`rudholm`	`you might try swapping them to see if it's the receiver`
`00:57.41`	`jayk-`	`ManxPower: hmm.. the driver was loaded into the kernel. how do i check to see if asterisk loaded the mod?`
`00:57.59`	`ManxPower`	`jayk-: how do you know what is loaded in memory is 1.2.x zaptel or 1.4.x zaptel?`
`00:58.17`	`ManxPower`	`jayk-: asterisk will not build chan_zap.so if it does not see zaptel installed.`
`00:58.19`	`CCFL_Man2`	`rudholm: i cleaned out the ear holes`
`00:58.57`	`CCFL_Man2`	`because i thought the F1 should sound as good as the G3`
`00:59.03`	`rudholm`	`they really should`
`00:59.08`	`ManxPower`	`anyway, I'm outta here.`
`01:01.19`	`CCFL_Man2`	`rudholm: my 302 has a 5H dial, as my imperial 202, but they both eith pulse with too long a pulse width or too slowly`
`01:01.32`	`CCFL_Man2`	`either`
`01:02.11`	`rudholm`	`CCFL_Man2: probably just need to clean and/or lubricate the speed governors`
`01:02.45`	`^JimmyRidge^`	`heh holy crap some of those asterisk gsm sounds are funny`
`01:03.03`	`^JimmyRidge^`	`blue eyed polar bear? wtf lol`
`01:03.08`	`*** join/#asterisk fujin (n=aj@unaffiliated/fujin)`
`01:03.54`	`rudholm`	`^JimmyRidge^: the monkeys, the monkeys!`
`01:04.20`	`CCFL_Man2`	`my my channel bank doesn't detect anything but 1 dialed, and on my rotary trimline and my G1 spacesaver it's detected most of the time, some digits are missed`
`01:04.33`	`^JimmyRidge^`	`though how can a put a lil pause before a ,playback()?`
`01:04.57`	`rudholm`	`wait(seconds)`
`01:05.25`	`rudholm`	`you can use decimals`
`01:05.28`	`CCFL_Man2`	`rudholm: my pots line all phones work great and never miss a digit, but i wonder, are my dials a little off?`
`01:05.39`	`rudholm`	`CCFL_Man2: could be`
`01:05.43`	`^JimmyRidge^`	`thanx cause during connect im loosing some of the playback`
`01:06.07`	`rudholm`	`CCFL_Man2: my IAXy doesn't like the dial on one of my ericofons, but that ericofon works fine on pots lines and Zap lines`
`01:06.29`	`CCFL_Man2`	`rudholm: yeah, thats the way it is with me`
`01:06.31`	`rudholm`	`^JimmyRidge^: yeah, I find that it cuts off a bit, so a Wait(0.5) or even less should solve it`
`01:07.22`	`rudholm`	`CCFL_Man2: and my trustly WECO 500 works perfectly with that IAXy, so I suspect the ericofon is a bit off. POTS lines are really tolerant to pulse timing. hookswitch dialing usually works pretty easily on a POTS line.`
`01:07.37`	`*** join/#asterisk brian_ (n=brian@unaffiliated/brian)`
`01:07.58`	`CCFL_Man2`	`rudholm: yeah, i had issues hook switch dialing on my channel bank too`
`01:08.43`	`CCFL_Man2`	`rudholm: i thought it was the dial needs to either be adjusted or something, but i hear to never ever ever adjust the weights on the flywheel`
`01:09.06`	`rudholm`	`you shouldn't need to, but they do get dusty sometimes`
`01:09.26`	`rudholm`	`and the works sometimes benefit from some light oil`
`01:10.15`	`CCFL_Man2`	`rudholm: so oil the shafts and the givenor wheel?`
`01:10.21`	`CCFL_Man2`	`govener`
`01:10.34`	`rudholm`	`yeah, something real light`
`01:10.39`	`rudholm`	`maybe even wd-40`
`01:10.49`	`rudholm`	`but use some dust-off first`
`01:10.50`	`CCFL_Man2`	`ahh, ok`
`01:10.54`	`CCFL_Man2`	`yeah`
`01:11.17`	`CCFL_Man2`	`here i thought the dials needed adjusting`
`01:11.20`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`01:11.55`	`*** join/#asterisk FireMac (n=mccasian@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)`
`01:11.58`	`CCFL_Man2`	`this channel bank is "carrier class", i thought it would work without issue`
`01:13.58`	`CCFL_Man2`	`well, i did order an upgraded fxs card for it, so maybe that'll make it more tolerant`
`01:14.26`	`CCFL_Man2`	`nothing beats a good old WECO 5ESS`
`01:14.34`	`rudholm`	`I don't have one`
`01:14.47`	`rudholm`	`I'd settle for a 355A`
`01:14.55`	`CCFL_Man2`	`a channel bank or a switch?`
`01:14.59`	`rudholm`	`switch`
`01:15.20`	`CCFL_Man2`	`even the nerdiest nerds don't have class 5 switches :P`
`01:15.40`	`rudholm`	`a friend of mine has a 355A`
`01:15.52`	`CCFL_Man2`	`those are hard to find`
`01:15.55`	`rudholm`	`it's connected to my Asterisk`
`01:16.03`	`rudholm`	`sometimes I just call it to listen to progress tones.`
`01:16.03`	`CCFL_Man2`	`they use inband signalling?`
`01:16.11`	`CCFL_Man2`	`farkin sweet!`
`01:16.41`	`CCFL_Man2`	`it's in his basement?`
`01:16.51`	`rudholm`	`he has a C.O.`
`01:17.10`	`CCFL_Man2`	`hook me up!!! :P`
`01:17.30`	`rudholm`	`he has an X-Y and a No. 3 x-bar`
`01:17.47`	`*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)`
`01:18.18`	`CCFL_Man2`	`nice`
`01:19.45`	`CCFL_Man2`	`oh, i forgot thats an old ckick-n-bang`
`01:20.03`	`rudholm`	`yes, yes it is`
`01:20.23`	`Strom_M`	`CCFL_Man2: hello`
`01:20.29`	`Strom_M`	`CCFL_Man2: i just read the scrollback`
`01:20.46`	`Strom_M`	`CCFL_Man2: and i've told you countless times that no, it's NOT me who put the iaxy into a 500 set`
`01:20.57`	`Strom_M`	`nor was it rudholm`
`01:21.08`	`rudholm`	`hehe`
`01:21.16`	`rudholm`	`it sounds like fun, though.`
`01:21.19`	`CCFL_Man2`	`Strom_M: you said it was Mark`
`01:21.31`	`rudholm`	`oh, maybe some other Mark.`
`01:21.34`	`rudholm`	`I can think of one.`
`01:21.46`	`Strom_M`	`CCFL_Man2: yeah, Mark Spencer`
`01:21.55`	`CCFL_Man2`	`there lies the source of the cornfusion!`
`01:22.08`	`CCFL_Man2`	`sorry :P`
`01:22.48`	`Ritzerisk`	`anyone ever get to config the vicidial with *`
`01:23.24`	`CCFL_Man2`	`rudholm: that ol' click-n-bang still in service for regular customers?`
`01:23.40`	`rudholm`	`CCFL_Man2: no`
`01:23.52`	`rudholm`	`CCFL_Man2: it's his personal project`
`01:24.54`	`CCFL_Man2`	`rudholm: you don't know how much i'd like such a personal project`
`01:24.59`	`*** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ced695885f057890)`
`01:26.26`	`*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)`
`01:27.14`	`CCFL_Man2`	`rudholm: he ever come on irc?`
`01:27.18`	`rudholm`	`CCFL_Man2: no`
`01:27.23`	`*** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)`
`01:27.37`	`CCFL_Man2`	`smart guy`
`01:27.42`	`rudholm`	`heh`
`01:28.08`	`rudholm`	`he doesn't really have a great internet connection either. some kind of fixed wireless.`
`01:28.21`	`rudholm`	`cant' get DSL and there's no cable plant`
`01:29.32`	`CCFL_Man2`	`ahh, unfortunate`
`01:30.02`	`CCFL_Man2`	`who needs that when you have a click-n-band switch as a personal project`
`01:31.47`	`CCFL_Man2`	`how is he able to connect digital trunks to his switch?`
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`01:32.25`	`rudholm`	`last time I was there, he didn't have that set up yet, but I believe it's now set up via T1 and channel bank.`
`01:33.10`	`rob0`	`Mark used to be on here a lot ... "kram".`
`01:36.02`	`CCFL_Man2`	`rob0: ahh, ok, thanks`
`01:36.22`	`hmmhesays`	`bah app_skel.c sucks`
`01:36.54`	`CCFL_Man2`	`rudholm: so thats the only way to interface a T1 trunk to it then? via a channel bank and fxo ports?`
`01:37.23`	`rudholm`	`I suppose so`
`01:37.24`	`metfan2007`	`hi all, is there a way to sotre in the CDR data what agent answer a call in a queue in a inbound campaign???`
`01:38.13`	`CCFL_Man2`	`ahh`
`01:38.39`	`CCFL_Man2`	`rudholm: he ever give you tours of his CO?`
`01:38.59`	`rudholm`	`CCFL_Man2: yes`
`01:39.26`	`rudholm`	`CCFL_Man2: Strom_M and I helped him a bit with Asterisk integration`
`01:41.35`	`CCFL_Man2`	`lucky bastards`
`01:46.17`	`CCFL_Man2`	`he a small CLEC?`
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`01:46.28`	`rudholm`	`CCFL_Man2: no`
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`01:46.59`	`CCFL_Man2`	`AN ILEC?`
`01:47.04`	`CCFL_Man2`	`whoops`
`01:47.05`	`rudholm`	`haha, no`
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`01:47.21`	`rudholm`	`just a guy with a devotion to his hobby`
`01:47.48`	`*** part/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl)`
`01:48.31`	`CCFL_Man2`	`how did he get his hands on a CO then?`
`01:50.16`	`Rants`	`does anyone here use ipkall?`
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`02:00.06`	`Rants`	`anyone use packet8?`
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`02:05.36`	`BockBilbo`	`is it possible to add more than one notification email addresses on the voicemail.conf file?`
`02:05.37`	`*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`02:05.37`	`*** mode/#asterisk [+o blitzrage] by ChanServ`
`02:07.55`	`metfan2007`	`hi all, is there a way to sotre in the CDR data what agent answer a call in a queue in a inbound campaign???`
`02:08.24`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`02:08.31`	`metfan2007`	`BockBilbo: you have to create email alias in you smtp system, like sendmail`
`02:09.00`	`BockBilbo`	`thanks metfan2007`
`02:10.01`	`BockBilbo`	`and do you know where can i change the default message?`
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`02:13.28`	`CCFL_Man2`	`rudholm: oh, i need a 684A subset and 4 conductor green rubber cord, you know any sources?`
`02:13.32`	`metfan2007`	`BockBilbo: in voicemail.conf, check the emailbody variable`
`02:13.52`	`BockBilbo`	`:)`
`02:13.55`	`BockBilbo`	`thank you`
`02:22.41`	`holiday42`	`rants: i use ipkall, why?`
`02:29.28`	`Rants`	`holliday, Im looking to do a conference calling system, and need incomming sip lines with alot of incomming channels for my college. I am willing to pay to make it legit, but I was wondering if ipkall could handel multiple incomming lines (like 20 or so) or if anyone knows of a did -> sip provider that I can use that would be great.`
`02:32.50`	`CCFL_Man2`	`rudholm: i found this site with the replacement cords i need`
`02:33.21`	`CCFL_Man2`	`coiled cloth moth green cord is $34, but bad for a rare cord for the F1 handset`
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`02:48.55`	`CCFL_Man2`	`Strom_C: i found a supplier for some reproduction colored cloth coiled cords for F1 handsets`
`02:49.24`	`Strom_C`	`cool`
`02:50.02`	`CCFL_Man2`	`Strom_C: they sell an original blue 302, $2500 :P`
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`02:50.53`	`CCFL_Man2`	`oldphoneworks`
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`03:31.34`	`Ritzerisk`	`yello`
`03:31.36`	`Ritzerisk`	`anyone there`
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`04:04.24`	`*** join/#asterisk nighty^ (n=nighty@72.0.72.123)`
`04:07.57`	`Ritzerisk`	`hey is there a unzip command besides unzip`
`04:11.19`	`*** join/#asterisk TrevMeister (n=trevor@166.129.103.55)`
`04:12.28`	`[TK]D-Fender`	`Ritzerisk, MC can browse through them`
`04:12.41`	`Nugget`	`tar on freebsd can unzip.`
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`04:26.24`	`Ritzerisk`	`it wont let me tar it`
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`04:32.55`	`tzafrir_home`	`Ritzerisk, why do you need a different command? unzip -l ?`
`04:33.26`	`flenders`	`~seen JT`
`04:33.37`	`jbot`	`jt is currently on #asterisk-dev #asterisk #slug. Has said a total of 68 messages. Is idling for 5h 35m, last said: 'nny: you dont have to buy cards fully populated'.`
`04:33.43`	`tzafrir_home`	`you tar with tar`
`04:34.04`	`tzafrir_home`	`zip is a different archive format`
`04:34.37`	`tzafrir_home`	`you refer to tar.gz ?`
`04:35.57`	`Ritzerisk`	`it wont let me unzip it`
`04:37.57`	`Ritzerisk`	`how would i extract a zip file if i dont have unzip command`
`04:38.11`	`tzafrir_home`	`Ritzerisk, what do you mean by "unzip"? extract a tar.gz?`
`04:38.13`	`[TK]D-Fender`	`Ritzerisk, MC <-`
`04:38.34`	`tzafrir_home`	`on which linux distribution?`
`04:38.40`	`tzafrir_home`	`MC uses unzip`
`04:38.55`	`[TK]D-Fender`	`tzafrir_home, does it?`
`04:39.07`	`[TK]D-Fender`	`tzafrir_home, Seems kinda internal....`
`04:39.08`	`tzafrir_home`	`Last time I checked`
`04:39.37`	`tzafrir_home`	`On debian: apt-get install unzip`
`04:42.26`	`Ritzerisk`	`i think it says asterisknow on ayer`
`04:42.55`	`Ritzerisk`	`haha whats mc ..`
`04:43.09`	`Ritzerisk`	`- unzip astguiclient_2.0.3.zip it tells me on the install guide for vicidial`
`04:43.31`	`*** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au)`
`04:50.33`	`luke-jr`	`So anyone know why Asterisk insists on playing hold music at 10x the volume of the call? ☺`
`04:50.59`	`luke-jr`	`(maybe not literally 10x, but it sounds like at least 2x, and ears are logarithmic…)`
`04:54.10`	`coppice`	`for music, ears are logarythmic`
`04:57.19`	`luke-jr`	`coppice: ☺`
`04:57.23`	`luke-jr`	`wow`
`04:57.24`	`luke-jr`	`OUCH`
`04:57.32`	`luke-jr`	`just finally listened to the recording of the call with MOH`
`04:57.40`	`luke-jr`	`the music is so loud it's totally distorted`
`04:57.46`	`*** join/#asterisk tsurko (n=tsurko@213.91.216.130)`
`04:58.31`	`coppice`	`d B doo, d B doo, d B doo....`
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`05:03.57`	`litage`	`"echocancel=yes" is specified in zapata.conf for each FXO port. why would "zap show channel 1" say that echo cancellation is off?: Echo Cancellation: 128 taps unless TDM bridged, currently OFF`
`05:07.55`	`*** join/#asterisk hijacked (i=N9XY@66.255.220.17)`
`05:09.06`	`*** join/#asterisk israr (n=israr@210-56-12-34.DSL.isb.comsats.net.pk)`
`05:16.23`	`litage`	`n/m! it says "currently OFF" when the FXO port is not engaged in a call, and changes to "currently ON" when there's a call`
`05:20.48`	`Ritzerisk`	`does anyone know what os is asterisknow on`
`05:22.19`	`luke-jr`	`…`
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`05:36.29`	`*** join/#asterisk brian_ (n=brian@unaffiliated/brian)`
`05:36.35`	`Ritzerisk`	`<PROTECTED>`
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`05:37.40`	`luke-jr`	`Ritzerisk: the purpose of that would be?`
`05:38.07`	`Ritzerisk`	`just to unzip a regular zip file to a directory`
`05:38.23`	`luke-jr`	`"unzip bla.zip"`
`05:38.37`	`luke-jr`	`ZIP is not tar-based`
`05:38.38`	`Ritzerisk`	`but i dont have the command unzip nor do i know where to download it`
`05:38.50`	`Ritzerisk`	`unzip doesnt wanna work ..`
`05:38.51`	`luke-jr`	`your OS's package repository`
`05:38.58`	`Ritzerisk`	`and i cant apt-get`
`05:39.05`	`Ritzerisk`	`or sudo apt`
`05:39.12`	`Ritzerisk`	`its the asterisknow ayer`
`05:39.17`	`luke-jr`	`…`
`05:39.25`	`luke-jr`	`then download it to a normal systme and unzip it there`
`05:39.36`	`luke-jr`	`or mount the LiveCD via sshfs`
`05:39.40`	`Ritzerisk`	`how do i do that`
`05:39.47`	`Ritzerisk`	`its just a wget file for vicidial`
`05:39.48`	`luke-jr`	`…`
`05:39.52`	`Ritzerisk`	`predictive dialer`
`05:40.03`	`luke-jr`	`there is no such thing as a 'wget file'`
`05:40.06`	`luke-jr`	`get a clue -.-`
`05:40.12`	`Ritzerisk`	`or like a wget http://....`
`05:40.19`	`luke-jr`	`it's called a zip file`
`05:40.24`	`luke-jr`	`no matter how you got it`
`05:40.24`	`coppice`	`the prediction being that those called will be pissed off`
`05:40.42`	`luke-jr`	`coppice: wtf?`
`05:40.57`	`coppice`	`predictive dialer`
`05:40.57`	`Ritzerisk`	`hmmm how can i unpack it then .. ...`
`05:41.11`	`luke-jr`	`oooh-- that's what that is?`
`05:41.16`	`luke-jr`	`Ritzerisk: I'm not helping you`
`05:41.21`	`Ritzerisk`	`haha`
`05:42.00`	`Ritzerisk`	`its not any of the other os im familar with`
`05:45.00`	`*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)`
`05:45.16`	`Ritzerisk`	`<PROTECTED>`
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`05:48.22`	`styelz`	`Ritzerisk: conrary update unzip ;)`
`05:48.27`	`Strom_C`	`Ritzerisk: if it's debian, "apt-get install unzip" always works`
`05:49.26`	`Ritzerisk`	`says command not found`
`05:49.41`	`lala45`	`can someone help a newb`
`05:49.57`	`styelz`	`sorrt`
`05:50.01`	`styelz`	`Ritzerisk: conary update unzip ;)`
`05:50.04`	`Strom_C`	`Ritzerisk: which distro is it?`
`05:50.06`	`styelz`	`typo`
`05:50.13`	`Strom_C`	`lala45: just ask your question`
`05:50.41`	`Ritzerisk`	`asterisknow ayer`
`05:50.48`	`styelz`	`Ritzerisk: conary update unzip ;)`
`05:51.20`	`Strom_C`	`Ritzerisk: i don't think you're supposed to be able to install vicidial on asterisknow`
`05:51.25`	`lala45`	`im total newb---im unhappy with my voip and i was looking into asterisk as a replacement---how easy or difficult is it to implement?`
`05:51.41`	`Strom_C`	`lala45: what kind of "voip" are you trying to replace?`
`05:52.15`	`lala45`	`i have voip from my internet provider, wich is local`
`05:52.18`	`Ritzerisk`	`ohh snazzy thanks`
`05:52.23`	`Ritzerisk`	`wait you cant ?`
`05:52.35`	`styelz`	`welcome`
`05:52.44`	`Strom_C`	`Ritzerisk: asterisknow is a software appliance`
`05:52.56`	`lala45`	`id like to have at least a couple of lines coming in`
`05:53.17`	`Strom_C`	`if you want to do heavy customization like installing vicidial, you'll have to go with a more traditional linux + asterisk + etc setup`
`05:53.22`	`lala45`	`i understand that ill still have to provide the did's or pots`
`05:53.24`	`Ritzerisk`	`so i tag a regular 1.41 with the vicidial`
`05:53.42`	`luke-jr`	`1.41 what?`
`05:53.42`	`*** join/#asterisk apardo (n=apardo@139.65.220.87.dynamic.jazztel.es)`
`05:53.52`	`Ritzerisk`	`like whats a good distro`
`05:54.04`	`Strom_C`	`lala45: how well do you understand linux, IP networking, and telephony?`
`05:54.10`	`Ritzerisk`	`is ubuntu any good`
`05:54.18`	`Strom_C`	`Ritzerisk: whichever one you're most comfortable using`
`05:54.19`	`luke-jr`	`Strom_C: fyi, he's a telemarketer type guy or something`
`05:54.22`	`styelz`	`ubuntu works ok for me`
`05:54.31`	`Ritzerisk`	`hmmm`
`05:55.01`	`Strom_C`	`Ritzerisk: and if you are a telemarketer, then please do us all a favor and go die in a fire`
`05:55.03`	`lala45`	`small understanding of linux, good understanding of networking and small understanding of telephony`
`05:55.06`	`Ritzerisk`	`so does asterisk have a web interface like the asterisknow or is it the same thing`
`05:55.44`	`Strom_C`	`lala45: you'll probably have a moderate but not insurmountable learning curve with asterisk`
`05:56.30`	`lala45`	`is there another open source system that would be more appropriate?`
`05:57.06`	`Strom_C`	`probably not`
`05:58.20`	`*** join/#asterisk yang (i=yang@static-ip-62-75-255-124.inaddr.intergenia.de)`
`05:58.22`	`Ritzerisk`	`im a mitel dealer trying to learn on asterisk haha`
`05:59.33`	`luke-jr`	`Strom_C: lol you can say that here?`
`05:59.41`	`lala45`	`then my question is can mutiple voip lines come into the *server, and how does that work?`
`05:59.52`	`Ritzerisk`	`setting up acd agents and eventually try to get the mitel to asterisk via sip trunks and use the predictive dialer with calls in the que to send a msg to all the users mailboxes`
`05:59.54`	`Strom_C`	`lala45: yes, though they're not "lines"`
`06:00.10`	`Strom_C`	`Ritzerisk: you don't need a predictive dialer to do that`
`06:00.45`	`lala45`	`? what do you mean? there not lines`
`06:00.56`	`Strom_C`	`TWILIGHT ZONE MUSIC`
`06:01.01`	`JT`	`lala45: voip doesn't have "lines"`
`06:01.04`	`lala45`	`can i dial in and out from them?`
`06:01.11`	`Strom_C`	`lala45: they are telephone connections`
`06:01.16`	`Strom_C`	`but "lines" is the wrong term`
`06:01.28`	`lala45`	`just a figure of speech, my bad`
`06:01.39`	`Strom_C`	`your bad what`
`06:01.49`	`Strom_C`	`incomplete sentence !!!!!`
`06:02.18`	`rudholm`	`Strom_C: I think lala45 meant "mea culpa"`
`06:02.20`	`*** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl)`
`06:02.24`	`Strom_C`	`ah`
`06:02.28`	`Strom_C`	`that makes more sense then :)`
`06:02.32`	`rudholm`	`yes, yes it does.`
`06:02.35`	`lala45`	`lol, i realize that these "mutiple lines" will be coming in over my network connection`
`06:04.13`	`rudholm`	`I often wonder why common usage favors nonsense like "my bad" and "mash up" when there are perfectly good terms already available like "mea culpa" and "pastiche" that mean precisely the same things.`
`06:04.20`	`*** join/#asterisk ta^3 (n=tacvbo@65.116.224.30)`
`06:04.56`	`lala45`	`some people are just more street`
`06:04.58`	`Strom_C`	`rudholm: because people are idiots?`
`06:05.20`	`lala45`	`thanks`
`06:05.42`	`rudholm`	`yes, that's why I favor rusting cars on my front lawn on jack stands and no wheels. it makes me more street. :)`
`06:06.08`	`styelz`	`get wid da program`
`06:06.12`	`Strom_C`	`and they'd probably think "pastiche" is in the same category as "vermicelli" and "fusilli"`
`06:06.17`	`rudholm`	`hahaha`
`06:06.18`	`lala45`	`but how does the hardware layout? can asterisk take these incoming connections and rerout?`
`06:06.31`	`Strom_C`	`lala45: essentially, yes`
`06:06.54`	`lala45`	`i thought "pastiche" is that nut they dye red.`
`06:07.07`	`Strom_C`	`that's pistachio`
`06:07.12`	`rudholm`	`lala45: what kind of incoming connections are they?`
`06:07.12`	`lala45`	`lol`
`06:07.33`	`lala45`	`voip connections mixed with one pots`
`06:07.49`	`rudholm`	`yeah, you can do whatever you like with inbound calls`
`06:08.27`	`*** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no)`
`06:08.52`	`JT`	`< lala45> some people are just more street <--- that is not english`
`06:08.59`	`lala45`	`does the network cable carry all connections to the *server or`
`06:09.25`	`rudholm`	`it won't carry the calls that come in via the pots line`
`06:09.30`	`lala45`	`thats not proper english, but it is accepted english`
`06:09.48`	`JT`	`accepted by no-one`
`06:09.58`	`JT`	`i've never seen anyone talk exactly like that`
`06:10.12`	`Ritzerisk`	`welp my boss was like this is what i need to look up and configure and we also needed to put everything in the mysql db`
`06:10.36`	`lala45`	`so i will have one network cable for multiple voip lines and 1 phone line for each pots line?`
`06:10.48`	`rudholm`	`yes`
`06:11.21`	`lala45`	`cool, any recommendations on did/voip resellers?`
`06:11.32`	`rudholm`	`I like teliax.com`
`06:11.45`	`rudholm`	`not the cheapest, but very stable`
`06:11.48`	`lala45`	`accepted by anyone`
`06:11.54`	`rudholm`	`for cheap, go with voipjet`
`06:12.07`	`lala45`	`hows the quality`
`06:12.21`	`rudholm`	`I haven't used voipjet in a while, but they were spotty`
`06:12.25`	`rudholm`	`and had outages`
`06:14.15`	`Ritzerisk`	`moss bros`
`06:18.21`	`*** join/#asterisk ^JimmyRidge^ (n=JimmyRid@EV-ESR-74-215-26-69.fuse.net)`
`06:23.49`	`*** join/#asterisk af_ (n=getsmart@81-174-8-20.dynamic.ngi.it)`
`06:24.36`	`*** join/#asterisk Rtx (n=xtr@bas1-quebec09-1178041595.dsl.bell.ca)`
`06:24.39`	`Rtx`	`hi`
`06:24.49`	`Rtx`	`someone could help me with a little thing ?`
`06:25.41`	`Strom_C`	`Rtx: you need to add more orange oil`
`06:25.54`	`Rtx`	`heh ok`
`06:26.10`	`Rtx`	`it is normal for zaptel to detect a fxo card as a fxs`
`06:26.13`	`Rtx`	`? :\`
`06:26.26`	`rudholm`	`no`
`06:26.42`	`Strom_C`	`the signalling type for FXO should be FXS`
`06:27.04`	`Rtx`	`ok it is because i have an openvox a400p`
`06:27.09`	`Rtx`	`with a fxo module on it`
`06:27.25`	`Rtx`	`and zaptel say this Channel 04: FXS Kewlstart (Default) (Slaves: 04)`
`06:27.27`	`Strom_C`	`ah, openvox`
`06:27.40`	`Strom_C`	`the yugo of ip pbx hardware`
`06:27.51`	`Rtx`	`huh ?`
`06:27.55`	`rob0`	`haha`
`06:28.16`	`^JimmyRidge^`	`yeah thats got me stumped too, i think stroms making words up agian`
`06:28.33`	`Strom_C`	`^JimmyRidge^: you don't remember Yugos?`
`06:28.36`	`Rtx`	`sorry could you explain yourself ?`
`06:28.39`	`rob0`	`No, the reference is to the Yugo automobile.`
`06:28.48`	`Strom_C`	`Rtx: openvox is junk. crap. garbage.`
`06:28.53`	`rob0`	`it's a joke rtx, means your card is cheap and lousy.`
`06:29.03`	`Rtx`	`err ...`
`06:29.04`	`^JimmyRidge^`	`the car?`
`06:29.09`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`06:29.09`	`rob0`	`yes`
`06:29.13`	`Rtx`	`this card have good feedback`
`06:29.30`	`Rtx`	`i readed about 30 feedback before buying it :\`
`06:29.52`	`Rtx`	`but is is normal to be detected as an fxs ?`
`06:29.57`	`Rtx`	`ist my only question ;\`
`06:30.07`	`Strom_C`	`Rtx: FXO ports use FXS signaling`
`06:30.10`	`Strom_C`	`try and see if it works`
`06:30.16`	`Rtx`	`ok`
`06:30.18`	`Rtx`	`will try`
`06:30.37`	`Strom_C`	`also: "readed" is not the past tense of "read"`
`06:31.13`	`Rtx`	`Strom_C: but think this card beter than x100p card on the net yes ?`
`06:31.29`	`Rtx`	`because i have buy in past x100p based card and .... poor`
`06:31.55`	`coppice`	`Rtx: its exactly the same as a TDM400P from digium. its as good or bad as that`
`06:32.02`	`*** join/#asterisk Lawbringer (n=Lawbring@212.183.134.130)`
`06:32.17`	`Rtx`	`but zaptel detect fxo module as fxs one heh`
`06:32.20`	`Rtx`	`crap heh`
`06:33.31`	`arekm`	`configuring FXS as groundstart doen't look sane. my phone didn't even had the signal then heh`
`06:34.11`	`CCFL_Man2`	`when i look at a western electric phone i see a beautiful piece of engineering`
`06:35.01`	`coppice`	`the good old bake-no-so-lite phones`
`06:35.09`	`Strom_C`	`http://www.stromcarlson.com/misc/P5300076.JPG`
`06:35.15`	`Rtx`	`does the default zaptel route should work with my card or i need to create another zaptel trunk ?`
`06:35.37`	`Strom_C`	`"zaptel route"?`
`06:35.42`	`Rtx`	`trunk`
`06:35.45`	`Rtx`	`sorry`
`06:35.46`	`Rtx`	`heh`
`06:35.59`	`Strom_C`	`what software are you using there?`
`06:35.59`	`Rtx`	`sleepy here ist 2am here :\`
`06:36.03`	`rob0`	`San Francisco ... area code 415`
`06:36.08`	`Rtx`	`freepbx`
`06:36.18`	`Strom_C`	`~freepbx`
`06:36.19`	`jbot`	`hmm... freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there`
`06:36.20`	`Rtx`	`and asterisk`
`06:36.25`	`*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)`
`06:37.47`	`Strom_C`	`http://www.stromcarlson.com/misc/agcs/P5120082.JPG`
`06:38.05`	`arekm`	`what's "wink"? there is rxwink var in zapata.conf`
`06:38.11`	`Strom_C`	`http://www.stromcarlson.com/misc/agcs/P5120084.JPG`
`06:38.24`	`Strom_C`	`arekm: T1 winkstart signaling`
`06:38.41`	`*** join/#asterisk IvanV3835 (n=Miranda@styx.mcn.ru)`
`06:39.10`	`coppice`	`not necessarily t1. wink was used on analogue pairs too`
`06:39.38`	`coppice`	`A wink is just a short pulse of the signaling`
`06:42.19`	`arekm`	`aha. I have FXSDN setting here, too - what's that?`
`06:45.17`	`arekm`	`found, some stripped down form of FXS`
`06:46.15`	`Rtx`	`ha !`
`06:46.23`	`Rtx`	`im able to make outbound call`
`06:46.25`	`Rtx`	`:)`
`06:46.32`	`Rtx`	`good news ?`
`06:54.52`	`*** join/#asterisk justdave (n=dave@unaffiliated/justdave)`
`07:02.45`	`*** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no)`
`07:02.53`	`*** join/#asterisk zeeesh (i=zeeesh@202.166.161.45)`
`07:04.39`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`07:04.50`	`*** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-b2350f8598f838e8)`
`07:07.54`	`*** join/#asterisk dominic1 (n=dob@213.221.82.242)`
`07:08.06`	`*** part/#asterisk dominic1 (n=dob@213.221.82.242)`
`07:13.27`	`*** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl)`
`07:16.23`	`Ritzerisk`	`does asterisk have a web interface or is it mostly conf files`
`07:16.34`	`*** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org)`
`07:17.00`	`J4zen`	`config files`
`07:17.08`	`J4zen`	`with third-party web interfaces`
`07:17.11`	`J4zen`	`afaik`
`07:18.06`	`J4zen`	`Does anyone happen to know a work-around/place to get the sata_piix driver? i cannot seem to find it and google agrees.`
`07:18.37`	`tzafrir_home`	`J4zen, that's a linux question . ask in the channel for your distro`
`07:18.51`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`07:19.05`	`J4zen`	`Will do.`
`07:20.27`	`tzafrir_home`	`BTW: the "default zaptel route" in FreePBX is Zap/g0 . Hence you just need to put your trunks in group 0 (as e.g. done by genzaptelconf)`
`07:22.56`	`mvanbaak`	`Ritzerisk: there are a couple of projects that provide a webinterface to manage asterisk configuration. The stock asterisk comes without one.`
`07:23.10`	`Ritzerisk`	`ohhh`
`07:23.47`	`Ritzerisk`	`hmm is there like a way to put vicidial on the asterisk with a web interface addon or the web interface is only through the asterisknow`
`07:24.08`	`Ritzerisk`	`sorry to bug but whats the differecnce betweeen asterisk and asterisknow`
`07:24.34`	`mvanbaak`	`asterisk is the application, asterisknow is a complete linux distribution that comes with asterisk preinstalled`
`07:24.56`	`Ritzerisk`	`ohhh yea it said like asterisknow Ayer`
`07:25.16`	`mvanbaak`	`vicidial is a thirdparty addon for asterisk. You can put it on your existing asterisk setup`
`07:25.23`	`Ritzerisk`	`with a web interface`
`07:25.36`	`Ritzerisk`	`can i put vicidial on a asterisknow though`
`07:25.51`	`mvanbaak`	`yup, but I dont know the product so dont know what you can do with the webinterface`
`07:26.00`	`Ritzerisk`	`ahh`
`07:26.07`	`mvanbaak`	`Ritzerisk: I have no idea, ask in #asterisknow`
`07:26.28`	`Ritzerisk`	`someone was telling me eariler i couldnt here because the liux distro is limited`
`07:26.41`	`Ritzerisk`	`asterisknow is noone really in there ehehe`
`07:27.26`	`Ritzerisk`	`im trying to figure to do either sip trunking or pri between my mitel pbx and the asterisk`
`07:28.45`	`*** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com)`
`07:28.50`	`Ritzerisk`	`but the sip trunking licence on the mitel cost a buttload of money`
`07:28.51`	`Ritzerisk`	`haha`
`07:30.56`	`Strom_C`	`well, ISDN PRI would be easy`
`07:31.07`	`*** join/#asterisk justdave_ (n=dave@unaffiliated/justdave)`
`07:31.56`	`Ritzerisk`	`meaning i would have to get the digium card`
`07:32.23`	`Ritzerisk`	`would the sip trunking pass caller id through`
`07:32.24`	`Strom_C`	`well, obviously`
`07:32.29`	`Strom_C`	`yes`
`07:32.43`	`Ritzerisk`	`reallly hmm`
`07:32.50`	`mvanbaak`	`Ritzerisk: have a look here for vicidial on asterisknow: http://forums.digium.com/viewtopic.php?p=56349&sid=f31903bca6ceaa3e7435d68dac46bb6d`
`07:33.06`	`Strom_C`	`how do you think caller ID gets passed to SIP phones?`
`07:33.09`	`Strom_C`	`magic?`
`07:33.14`	`Strom_C`	`bicycle messenger?`
`07:33.19`	`Ritzerisk`	`i didnt know if there was a source to emulate the pri`
`07:33.34`	`Strom_C`	`perhaps a good cabernet sauvignon?`
`07:33.44`	`J4zen`	`lol`
`07:33.45`	`Ritzerisk`	`wellp i dont know if it would because say i have a setup of a mitel system going to the *`
`07:33.45`	`Strom_C`	`what do you mean "source to emulate the pri"?`
`07:33.47`	`mvanbaak`	`18 year old single malt scotch`
`07:34.55`	`Ritzerisk`	`so the originating call comes from the main pbx .. mitel.. - the * so i was just checking if it passes CID info ..`
`07:35.42`	`Ritzerisk`	`or like an ext off of the mitel calling the huntgroup going tot the *`
`07:36.20`	`Strom_M`	`.....huh?`
`07:36.32`	`Strom_M`	`sense making little`
`07:41.24`	`*** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se)`
`07:41.30`	`*** join/#asterisk bintut (n=bintut@203.125.63.150)`
`07:41.54`	`*** join/#asterisk blackhole (n=pkakkar@unaffiliated/blackhole)`
`07:42.16`	`blackhole`	`Do i need to take any SIP addrerss or route for placing calls in a country using Asterisk..?`
`07:42.55`	`Strom_M`	`blackhole: that question makes no sense`
`07:43.28`	`blackhole`	`Strom_C, I need to configure asterisk to be able to place calls to any number in my country, else than confg asterisk what would i need?`
`07:44.10`	`Strom_M`	`you need an ITSP`
`07:44.17`	`blackhole`	`Strom_C, what is that?`
`07:44.25`	`blackhole`	`Strom_C, I am being told i need A_Z route account`
`07:44.32`	`Strom_M`	`~itsp`
`07:44.32`	`jbot`	`An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower`
`07:44.55`	`blackhole`	`Strom_C, Someone is telling me i need A_Z route account, Is that what i need..?`
`07:45.32`	`Strom_M`	`i have no clue what "a_z route" is supposed to be`
`07:46.23`	`bintut`	`tzafrir: still there? :)`
`07:47.40`	`blackhole`	`Strom_C, Some Service Provider says A_Z route is account which would help to send calls from your asterisk box to their gateway`
`07:48.58`	`blackhole`	`Strom_C, Any idea?`
`07:50.10`	`Ritzerisk`	`is iax the protocol just for asterisk`
`07:50.22`	`denon`	`other devices run iax`
`07:50.32`	`denon`	`iax is open source, anyone can implement it in their hardware or pbx`
`07:50.37`	`Ritzerisk`	`like other pbx`
`07:50.50`	`denon`	`yeah, others use it`
`07:50.55`	`denon`	`not real common yet`
`07:51.04`	`Ritzerisk`	`is it good`
`07:51.22`	`denon`	`depends on your needs, but yes, it tends to be very simple for client access, and very nice for trunking`
`07:51.24`	`blackhole`	`I need to configure asterisk to be able to place calls to any number in my country, else than confg asterisk what else would i need and can you point to any website which provide that service`
`07:51.35`	`Ritzerisk`	`doesnt sip use like random ports all the time and opens alot just for one call really`
`07:51.43`	`denon`	`kind of`
`07:51.55`	`denon`	`that's a good share of the reason iax was developed`
`07:52.00`	`denon`	`single port, simple communication`
`07:52.29`	`Ritzerisk`	`sip is all over the place in a ethereal log i was like whoaa`
`07:53.16`	`*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)`
`07:54.28`	`Ritzerisk`	`can asterisknow do everything that asterisk can do or is it limited`
`07:54.36`	`denon`	`~asterisknow`
`07:54.49`	`denon`	`hmm .. well ..`
`07:54.55`	`denon`	`you should google it :)`
`07:54.58`	`denon`	`Im headin to bed`
`07:54.59`	`mvanbaak`	`~gui`
`07:55.00`	`jbot`	`it has been said that gui is (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli`
`07:55.11`	`Ritzerisk`	`i know that asterisknow has web interface`
`07:56.13`	`Ritzerisk`	`laters denon thanks ;)`
`07:56.17`	`*** join/#asterisk appelza (n=d@dsl-240-129-197.telkomadsl.co.za)`
`07:56.25`	`mvanbaak`	`it cannot do what you can do while editing configfiles`
`07:56.34`	`appelza`	`hi, how do I pick up a call from anywhere by default?`
`07:57.11`	`Ritzerisk`	`is there like a call pickup feature access code you can dial`
`07:57.32`	`*** join/#asterisk fenlander (n=fenlande@82.152.81.57)`
`07:58.36`	`mvanbaak`	`you have to look in /etc/asterisk/features.conf`
`07:58.59`	`mvanbaak`	`if the phone and the incoming call are in the same call/pickup group the default code is *8`
`08:00.13`	`blackhole`	`I need to configure asterisk to be able to place calls to any number in my country, else than confg asterisk what else would i need and can you point to any website which provide that service`
`08:00.32`	`*** join/#asterisk qdk (n=qdk@213.150.62.32)`
`08:01.03`	`mvanbaak`	`you either need a phoneline or an ITSP`
`08:01.07`	`mvanbaak`	`~itsp`
`08:01.08`	`jbot`	`An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower`
`08:01.24`	`appelza`	`thanks mvanbaak`
`08:01.32`	`mvanbaak`	`appelza: welcome :)`
`08:02.08`	`mvanbaak`	`blackhole: if you want to connect a physical phoneline you will need a pci card for that. http://www.digium.com has some for almost all types of phonelines`
`08:03.39`	`blackhole`	`mvanbaak, I have one VOIP provider he is telling me they will provide me A_Z route which will help my asterisk box to connect to their gateway and hence place call..`
`08:03.44`	`blackhole`	`mvanbaak, Is that true?`
`08:04.26`	`mvanbaak`	`I have no idea what they mean with A_Z route`
`08:05.25`	`blackhole`	`A_Z route would be something in their gateway like they will provide me some gateway addr which would allow my calls to be sent to anywhere in world, my calls would connect to gateway.. mvanbaak`
`08:06.23`	`mvanbaak`	`well, that's what you need right ?`
`08:06.54`	`blackhole`	`mvanbaak, yes kind of .. but would it be called would it be called SIP details?`
`08:08.07`	`*** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net)`
`08:08.47`	`blackhole`	`mvanbaak, Someone earlier told me i would require internet service and SIP provider for it..`
`08:08.55`	`blackhole`	`mvanbaak, Is it same as SIP provider`
`08:10.07`	`blackhole`	`mvanbaak, Any idea?`
`08:12.28`	`mvanbaak`	`I have no idea blackhole`
`08:13.49`	`*** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net)`
`08:13.50`	`l0verb0y`	`hey does anyone know what the best voice recogniition software for asterisk would be`
`08:14.07`	`mvanbaak`	`If you want voip to route calls to outside numbers you need a good and stable internet connection and an account with a voip provider`
`08:14.16`	`mvanbaak`	`that voip provider link can be sip or IAX2`
`08:17.50`	`*** join/#asterisk d3wayne (n=deeewayn@c-76-29-245-9.hsd1.al.comcast.net)`
`08:17.50`	`*** mode/#asterisk [+o d3wayne] by ChanServ`
`08:27.56`	`*** join/#asterisk VictorP (n=fusma@143.Red-83-61-17.staticIP.rima-tde.net)`
`08:31.44`	`*** join/#asterisk hank (n=hank@netwichtig.de)`
`08:31.58`	`hank`	`hi`
`08:32.11`	`VictorP`	`hi there`
`08:34.33`	`hank`	`i am not sure where to start, so i hope i can explain my question clearly enough. we have asterisk 1.2 with snom phones. before we switched to that it was possible to say 'call me back' when one person calls another person who is currently on the phone. if person2 hangs up, person1's phone rings and when answered calls person2. is there a function like that in asterisk?`
`08:34.51`	`hank`	`if not: any recommendations how to achieve it using agi scripts or something like that?`
`08:38.48`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`08:39.44`	`yang`	`mvanbaak: you just enter SIP details to your asterisk, and it will connect over that to the uplink VOIP provider`
`08:39.52`	`yang`	`mvanbaak: -> blackhole`
`08:49.46`	`blackhole`	`yang, Thanks.. Which file in asterisk have SIP details?`
`08:51.36`	`*** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro)`
`09:02.21`	`*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)`
`09:04.57`	`yang`	`blackhole: sip.conf`
`09:09.06`	`luke-jr`	`blackhole: do note that all ITSPs are unreliable ☺`
`09:09.17`	`luke-jr`	`to an extent`
`09:09.54`	`blackhole`	`yang, Most of asterisk conf file are in /etc/asterisk am i correct?`
`09:10.05`	`luke-jr`	`all should be`
`09:10.07`	`J4zen`	`ya`
`09:11.05`	`blackhole`	`Okay, Thanks.`
`09:30.49`	`*** join/#asterisk orn (n=orn@85.197.193.24)`
`09:31.37`	`orn`	`I have a problem in asterisk-1.4.11; it seems I can't change callerID name without changing callerid num too`
`09:32.34`	`orn`	`I use Set(CALLERID(name)=Whatever), but the original callerid name displays`
`09:33.02`	`orn`	`if I do Set(CALLERID(name)=Whatever) and Set(CALLERID(num)=123), it changes both name and number`
`09:33.23`	`orn`	`but I want to change only the name and keep the number unchanged`
`09:33.32`	`orn`	`is there a bug or am i doing something wrong?`
`09:36.00`	`luke-jr`	`orn: a directory service might be overriding whatever you send for name, if it knows the num`
`09:44.56`	`orn`	`hmmm`
`09:45.05`	`orn`	`interesting`
`09:50.26`	`*** join/#asterisk stkn\|work (n=stkn\|wor@gentoo/developer/pdpc.active.stkn)`
`09:50.57`	`*** join/#asterisk kay2 (n=two@gob75-7-82-247-113-230.fbx.proxad.net)`
`09:51.19`	`blackhole`	`How can i check which version of asterisk i am running?`
`09:51.57`	`hank`	`asterisk -rx 'show version'`
`09:52.16`	`blackhole`	`okay`
`09:54.35`	`*** join/#asterisk pkunkra (n=chris@cpe-74-73-28-89.nyc.res.rr.com)`
`09:55.10`	`knarfly`	`blackhole, show version is being deprecated...they want you to type 'core show version'...more key strokes for the same thing`
`09:57.23`	`knarfly`	`why do these two lines whack up my extension?`
`09:57.26`	`knarfly`	`exten => s/7605320417,n,Goto(blocking,s,1) ; Block This Pesky Telemarkete`
`09:57.26`	`knarfly`	`exten => s/9012792345,n,Goto(wife,s,1) ; Take wife's call`
`09:57.46`	`knarfly`	`without these, the same dial plan works fine`
`09:58.06`	`knarfly`	`but when I uncomment them it goes into auto failthrough`
`09:59.47`	`hank`	`what does 's/' do?`
`10:00.31`	`knarfly`	`it's supposed to let * determine if the call is coming from the phone number that proceeds it.`
`10:00.50`	`knarfly`	`sometimes, when I only use one of these it works, with two it always fails`
`10:01.45`	`knarfly`	`I want to add a list of about 5-6 numbers that should get trapped and sent either to the blocking (zapateller) or go to another extension for handling`
`10:02.01`	`knarfly`	`but since I cannot get this working, I'm stuck`
`10:07.50`	`*** join/#asterisk tsurko (n=tsurko@78.90.100.214)`
`10:15.43`	`*** join/#asterisk saftsack (n=saftsack@pD9E0685A.dip.t-dialin.net)`
`10:15.58`	`orn`	`luke-jr: That was it! Thanks man :)`
`10:16.25`	`thomas`	`have a error:`
`10:16.25`	`thomas`	`[Sep 26 12:15:57] NOTICE[1923]: chan_sip.c:5331 process_sdp: No compatible codecs, not accepting this offer!`
`10:16.34`	`thomas`	`my codec: G.729A`
`10:16.38`	`thomas`	`what is wrong?`
`10:16.42`	`orn`	`thomas: Do you have the G.729 license?`
`10:16.48`	`thomas`	`orn: hmm. no?`
`10:17.03`	`thomas`	`orn: what is the best free codec?`
`10:17.04`	`orn`	`thomas: Try sending a codec freely available (711 alaw/ulaw, 723, gsm etc)`
`10:17.17`	`luke-jr`	`orn: np`
`10:17.35`	`thomas`	`orn: 723.1 ?`
`10:17.38`	`luke-jr`	`thomas: alaw/ulaw are uncompressed, thus lossless`
`10:17.52`	`luke-jr`	`eg, better quality than anything else`
`10:19.36`	`*** join/#asterisk Martin79 (n=martin@cpc4-leed12-0-0-cust360.leed.cable.ntl.com)`
`10:20.22`	`Martin79`	`Hi, could somone help me work out why chan_misdn isn't compiling on one of my machines please?`
`10:23.29`	`*** join/#asterisk nfi\|ermes (n=ErMeS@217.220.121.62)`
`10:30.14`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)`
`10:30.37`	`*** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-b2350f8598f838e8)`
`10:32.39`	`awk`	`hmm, is this normal. when parking a call it takes 2/3 seconds to actually do the park, on your phone you can see it stating parking but takes as I said 2/3 seconds? is there a way to make that kind of instant?`
`10:47.35`	`l0verb0y`	`does anyone know where I can get liblv_lvbts.so ?`
`10:49.40`	`thomas`	`is asterisk 1.4.11 stable?`
`10:49.46`	`mvanbaak`	`yes`
`10:50.23`	`thomas`	`i have a snom 370 and asterisk 1.4.11 but the led is not blinking if i recieve a call`
`10:50.29`	`thomas`	`and no pickup...`
`10:50.39`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`10:50.59`	`mvanbaak`	`did you setup hints ?`
`10:51.28`	`thomas`	`mvanbaak: i think yes: 84@hints : SIP/84 State:Idle Watchers 0`
`10:51.33`	`thomas`	`but watcher = 0`
`10:52.09`	`thomas`	`mvanbaak: http://tm.muc.de/up/archiv/Unbenannt.PNG`
`10:52.17`	`thomas`	`(P6)`
`10:52.23`	`thomas`	`this is also correct or?`
`10:54.12`	`*** join/#asterisk gardo (n=gardo@121.97.249.68)`
`10:54.31`	`*** join/#asterisk qdk (n=qdk@213.150.62.32)`
`10:54.48`	`mvanbaak`	`my german is terrible`
`10:55.07`	`thomas`	`MrChimpy: hehe :-)`
`10:59.18`	`*** join/#asterisk Renacor (n=kvirc@62.157.211.194)`
`10:59.26`	`Renacor`	`is there a way to join 2 gsm files with sox?`
`11:04.48`	`mvanbaak`	`sox file1 file2 output`
`11:05.08`	`mvanbaak`	`it's on the first page of the command 'man sox'`
`11:11.44`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`11:14.05`	`*** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg)`
`11:22.39`	`*** join/#asterisk vgster (n=vgster@h14862.navonline.net)`
`11:23.26`	`*** join/#asterisk xipi (n=xipi@212.17.244.236)`
`11:24.05`	`*** join/#asterisk vgster (n=vgster@217.78.148.62)`
`11:25.12`	`xipi`	`how can i send an inbound call to a voice menu?`
`11:25.33`	`xipi`	`with the stdext macro?`
`11:25.51`	`mvanbaak`	`exten => <my_inbound_nr>,1,Goto(ivr,s,1)`
`11:26.09`	`xipi`	`mvanbaak, thanks. i'll try`
`11:26.18`	`*** join/#asterisk IgorG (n=FeedomPa@195.162.32.126)`
`11:27.43`	`mvanbaak`	`I'm off to a customer`
`11:27.44`	`mvanbaak`	`latero`
`11:28.53`	`xipi`	`ok. thanks`
`11:29.44`	`*** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net)`
`11:31.47`	`*** join/#asterisk zdrulio (n=krlozano@82.119.72.130)`
`11:31.50`	`zdrulio`	`hello`
`11:31.54`	`*** join/#asterisk CVirus (n=GoD@196.205.191.113)`
`11:32.01`	`zdrulio`	`i have a question`
`11:32.11`	`CVirus`	`I'm getting the following errors when trying to compile asterisk-addons http://rafb.net/p/eMFOqP47.html`
`11:35.27`	`zdrulio`	`i have 1.4.11 asterisk. When i talk with other side i want to put them "on hold" and dial to another phone and when i hangup a second line to pickup a first call. how can i make this ?`
`11:36.53`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`11:37.17`	`sevard`	`that's called an attended transfer. While on a call, hit flash, you'll get a dialtone and caller1 will be on hold listening to MoH. Dial another number, tell caller2 that caller1 wishes to speak to caller2. Hangup. The two calls are now bridged on your Asterisk system.`
`11:38.55`	`sevard`	`and with that very congested-head influenced answer, brought to you by the letters F and U and the number 1-SinusInfection in association with ColdSeasonSucks, good-night.`
`11:39.58`	`*** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk)`
`11:41.50`	`styelz`	`CVirus: maybe you need libncurses and libmxml before menuselect will work`
`11:42.52`	`CVirus`	`styelz: dev-cpp/libmxmlplus ?`
`11:43.11`	`*** join/#asterisk dominic1 (n=dob@213.221.82.242)`
`11:43.18`	`styelz`	`sorry i dont know`
`11:43.24`	`CVirus`	`styelz: Thanks alot`
`11:43.33`	`CVirus`	`will try with it anyways`
`11:43.42`	`styelz`	`but it says it needs them in menuselect/README`
`11:45.21`	`xipi`	`what does this messge mean: "Failed to authenticate on INVITE"`
`11:45.40`	`xipi`	`it was from an outbound call`
`11:46.15`	`CVirus`	`styelz: Thanks again`
`11:47.34`	`*** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net)`
`11:47.37`	`zdrulio`	`hm`
`11:47.48`	`zdrulio`	`with *7 i transfer other side but`
`11:47.58`	`zdrulio`	when i`m with PAP2t and hardphone
`11:48.17`	`zdrulio`	`i must push * 7 very fast to work`
`11:48.40`	`zdrulio`	`where can i raise seconds for *7 feature ?`
`11:48.41`	`CVirus`	`styelz: I installed libmxml and asterisk-addons still doesn't want to compile`
`11:52.05`	`*** join/#asterisk erwinpogz (n=chatzill@124.106.223.17)`
`11:54.35`	`*** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`11:54.43`	`erwinpogz`	`hello, i would like to ask how can I get a voip lines to accept 1-800 calls?`
`11:55.08`	`Daviey`	`erwinpogz: googld it?`
`11:55.12`	`Daviey`	`*googled`
`11:55.47`	`*** join/#asterisk Lawbringer (n=Lawbring@212.183.134.211)`
`11:56.36`	`erwinpogz`	`Daviey: i am using asterisk for my PBX, just want a telco provider for this`
`11:56.45`	`erwinpogz`	`i will google too :)`
`11:58.17`	`kaldemar`	`erwinpogz: what is it that you expect to be a problem with asterisk?`
`11:59.39`	`erwinpogz`	`kaldemar: i just want want to know what type of voip cards (if theres any) i should use to comply with the telco that i should connect to`
`12:00.29`	`Daviey`	`erwinpogz: Often a freephone numper points to a regular number`
`12:00.38`	`Daviey`	`s/numper/number`
`12:02.02`	`erwinpogz`	`the scenario is this... I am from philippines, I am going to get an IPL connection to US (Los Angeles)`
`12:03.13`	`erwinpogz`	`two E1 connections`
`12:04.22`	`*** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com)`
`12:04.22`	`erwinpogz`	`i really dont have any idea which voip provider should I look into to provide me a toll free lines`
`12:04.59`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`12:06.59`	`*** join/#asterisk lirakis (n=lirakis@65.200.191.253)`
`12:07.02`	`lirakis`	`morning`
`12:07.12`	`erwinpogz`	`good morning lirakis`
`12:08.04`	`*** join/#asterisk coppice (n=chatzill@79.193.17.210.dyn.pacific.net.hk)`
`12:08.26`	`*** join/#asterisk nighty^ (n=nighty@72.0.72.123)`
`12:09.44`	`kaldemar`	`erwinpogz: a regular NIC is enough to connect with VoIP. you only need special hardware for non VoIP calls.`
`12:09.45`	`*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)`
`12:11.00`	`Daviey`	`NIC = Network Card`
`12:12.28`	`erwinpogz`	`okay`
`12:14.52`	`*** join/#asterisk PepOSX (n=pepOSX@190.72.149.199)`
`12:15.38`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`12:26.16`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`12:27.32`	`*** join/#asterisk guillote_GNU (n=bancaria@host35.201-253-17.telecom.net.ar)`
`12:33.39`	`xipi`	`what does this messge mean: "Failed to authenticate on INVITE"`
`12:33.47`	`xipi`	`?`
`12:35.18`	`*** part/#asterisk dominic1 (n=dob@213.221.82.242)`
`12:36.19`	`*** part/#asterisk zdrulio (n=krlozano@82.119.72.130)`
`12:37.21`	`*** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net)`
`12:37.34`	`*** part/#asterisk fenlander (n=fenlande@82.152.81.57)`
`12:40.38`	`*** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl)`
`12:46.08`	`[TK]D-Fender`	`xipi: means one side asked the other to verify its identity and the auth doesn't match`
`12:49.54`	`*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)`
`12:50.32`	`Dr-Linux`	`i can dial a tollfree from my extension directly, but i can't dial through agi`
`12:50.39`	`Dr-Linux`	`please someone have a look here: http://phpfi.com/265212`
`12:51.27`	`xipi`	`[TK]D-Fender, oh! interesting. i'll have to check`
`12:52.17`	`[TK]D-Fender`	`Dr-Linux: -- Got SIP response 482 "Loop Detected" back from 127.0.0.1 <----`
`12:52.45`	`Dr-Linux`	`[TK]D-Fender: yes, but i didn't understand it`
`12:53.03`	`*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)`
`12:53.03`	`[TK]D-Fender`	`Dr-Linux: Why are you dialing SIP INTO your same server?`
`12:54.14`	`_x86_`	`hahahhaha`
`12:54.19`	`_x86_`	`that's awesome :P`
`12:54.33`	`[TK]D-Fender`	`GI Script Executing Application: (Dial) Options: (sip/4565566@localhost\|50000\|L(10:80000:20000)) <--------`
`12:54.47`	`[TK]D-Fender`	`I prefer to think of it as... RETARDED`
`12:54.52`	`kaldemar`	`maybe he's generating load on purpose.`
`12:55.21`	`[TK]D-Fender`	`-- Now forwarding SIP/4092-08517e88 to 'Local/4565566@users' (thanks to SIP/localhost-085072b0)`
`12:55.25`	`[TK]D-Fender`	`And FORWARDEING!`
`12:55.54`	`[TK]D-Fender`	`-- Executing Answer("Local/4565566@users-c73d,2", "") in new stack`
`12:56.08`	`[TK]D-Fender`	`And some more vrtualization w/ Chan_local! wheeee!!!`
`12:56.09`	`_x86_`	`[TK]D-Fender: spelling? :P`
`12:56.18`	`[TK]D-Fender`	`_x86_: No, typing :p`
`12:56.23`	`_x86_`	`hehe`
`13:01.23`	`[TK]D-Fender`	`_x86_: Haven't been sleeping much and my caffeine/blood ratio is too low...`
`13:03.21`	`*** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`13:05.32`	`*** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`13:07.21`	`cpm`	`[TK]D-Fender, seems like a lot of that going around these days, are you waking up in strange places?`
`13:08.50`	`_x86_`	`[TK]D-Fender: I know how that is... of course, I have the same problem well after I'm caffinated properly, but i still blame it on low caffiene levels ;)`
`13:09.33`	`*** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr)`
`13:13.50`	`*** join/#asterisk d3wayne (n=deeewayn@c-76-29-245-9.hsd1.al.comcast.net)`
`13:13.50`	`*** mode/#asterisk [+o d3wayne] by ChanServ`
`13:15.29`	`*** join/#asterisk masus (n=tet@88.248.73.2)`
`13:15.57`	`masus`	`hi all`
`13:16.10`	tru_`z24	`[TK]D-Fender: how do you put asterisk into debug mode?`
`13:16.10`	`masus`	`have a little problems -> "res_odbc.c: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Can't open lib '/usr/lib/libtdsodbc.so"`
`13:16.48`	`masus`	tru_`z24: do u mean "set debug n"
`13:16.51`	`hank`	`what is esel.conf for?`
`13:16.52`	tru_`z24	`[TK]D-Fender: enable zap debug, debug 10, verbose 10`
`13:16.55`	tru_`z24	`how do i do those things`
`13:17.16`	`masus`	`set debug 3; set verbosee 99`
`13:17.21`	`masus`	`set verbose`
`13:17.30`	tru_`z24	`is this at the asterisk CLI?`
`13:17.31`	`masus`	`yes`
`13:17.40`	tru_`z24	`what about zap debug?`
`13:17.42`	`*** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com)`
`13:17.46`	tru_`z24	`set zap debug i assume?`
`13:17.51`	`masus`	`i do not use zap maybe the same`
`13:18.12`	tru_`z24	`rgr`
`13:21.46`	`*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)`
`13:23.16`	`lirakis`	`hmm... when a caller comes out of a queue and is talking to an agent... if the agent conferences them to another number, and then tries to enter dtmf tones .. asterisk interpets it as a transfer..`
`13:23.38`	`lirakis`	`but i dont have any "feature codes" set up`
`13:23.46`	`lirakis`	`.. im not sure why its trying to transfer`
`13:23.58`	`lirakis`	`after the 3 way call has already been established`
`13:24.35`	`*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)`
`13:24.36`	`Katty`	`herro`
`13:29.15`	`masus`	`"Can't open lib '/usr/lib/libtdsodbc.so" but the file is there ... can anybody help`
`13:29.50`	`creativx`	`Katty: herrrrrrrrr hoooo`
`13:30.35`	`_x86_`	`masus: it's a link... see where the destination is, verify the destination exists, and use the 'file' utility to verify the destination is actually what it needs to be (in this case, a shared object library)`
`13:31.04`	`*** join/#asterisk ManxPower (n=manxpowe@ip68-107-250-148.no.no.cox.net)`
`13:31.50`	`masus`	`it's still exists`
`13:32.35`	`masus`	`"lrwxrwxrwx 1 root root 19 2007-09-26 12:26 libtdsodbc.so -> libtdsodbc.so.0.0.0"`
`13:32.57`	`[TK]D-Fender`	`lirakis: PASTEBIN time....`
`13:33.01`	`[TK]D-Fender`	`Katty: Mew.`
`13:33.27`	`thomas`	`spricht hier wer deutsch?`
`13:33.37`	`orn`	`ein bisschen`
`13:33.52`	`thomas`	`orn: hallo :-)`
`13:34.11`	`orn`	`hallo :)`
`13:34.21`	`tzafrir_laptop`	`masus, test -r /usr/lib/libtdsodbc.so && echo readable`
`13:34.23`	`thomas`	`orn: kennst du dich mit hints aus? snom?`
`13:34.39`	`tzafrir_laptop`	`thomas, English, please`
`13:35.08`	`masus`	`tzafrir_laptop: readable`
`13:35.12`	`*** join/#asterisk thansen\|laptop (n=thansen@19.243.sfcn.org)`
`13:35.33`	`tzafrir_laptop`	`ls -l /usr/lib/libtdsodbc.so.0.0.0`
`13:36.02`	`masus`	`tzafrir_laptop "-rwxr-xr-x 1 root root 789694 2007-09-26 12:26 /usr/lib/libtdsodbc.so.0.0.0"`
`13:36.44`	`tzafrir_laptop`	`Any more specific error that "can't open"?`
`13:36.54`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`13:36.54`	`*** mode/#asterisk [+o anthm] by ChanServ`
`13:37.39`	`masus`	`tzafrir_laptop: http://pastebin.ca/715770`
`13:38.02`	`Katty`	`[TK]D-Fender: mew (=`
`13:38.20`	`Katty`	`creativx: what did you call me?! ^_-`
`13:39.27`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`13:39.33`	`tzafrir_laptop`	`masus, hmmm, there's only an opening quote in that error message. That looks strange to me`
`13:40.11`	`Katty`	`[TK]D-Fender: i need undepressing.`
`13:40.16`	`tzafrir_laptop`	`Does the name /usr/lib/libtdsodbc.so appear in some config file?`
`13:40.28`	`Katty`	`[TK]D-Fender: do you have a couch and chocolate? )=`
`13:40.35`	`masus`	`hmmm`
`13:41.42`	`masus`	`i dont know`
`13:42.01`	`*** join/#asterisk sangee (i=ravi@206.191.114.82)`
`13:42.15`	`masus`	`it's for func_odbc`
`13:42.57`	`sangee`	`Hi my outbound working with pri card but inbound does not work and i got "Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]"`
`13:43.12`	`sangee`	`someone help me to solve this issue?`
`13:44.02`	`s34n`	`[TK]D-Fender: Although my peer was working off and on yesterday, it finally settled into not working.`
`13:44.20`	`hank`	`anyone know what res_esel does?`
`13:44.42`	`s34n`	`[TK]D-Fender: It registers consistently now. I think that part is finally straightened out`
`13:45.02`	`s34n`	`[TK]D-Fender: but * doesn't accept incoming calls`
`13:45.20`	`s34n`	`[TK]D-Fender: It returns a "404 Not Found"`
`13:45.44`	`s34n`	`[TK]D-Fender: which leads me to believe that my peer is misconfigured in sip.conf`
`13:46.10`	`s34n`	`or my extentodial is coordinated properly with my dialplan`
`13:46.20`	`s34n`	`s/is/isn't/`
`13:47.30`	`s34n`	`[TK]D-Fender: but I'm sniffing the incoming invites, and they are using the extension I expect`
`13:48.17`	`s34n`	`[TK]D-Fender: So perhaps they are not coming to the context I expect?`
`13:48.19`	`*** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com)`
`13:48.29`	`Katty`	`jbot: :<`
`13:48.30`	`jbot`	`< is, like, redirection of stdin to a program`
`13:48.36`	`Dr-Linux`	`[TK]D-Fender: back, sorry i was away`
`13:48.37`	`Katty`	`jbot: today is not a good day.`
`13:48.38`	`jbot`	`...but today is already something else...`
`13:48.45`	`Katty`	`jbot: oh? what?`
`13:48.53`	`Katty`	`...didn't think so.`
`13:49.11`	`Katty`	`jbot: i still love you.`
`13:49.11`	`jbot`	`You still love you.?`
`13:49.20`	`Katty`	`jbot: no, i love jbot.`
`13:49.31`	`Katty`	`speechless!`
`13:49.43`	`Sauron	`Mew`
`13:49.48`	`Katty`	herro `Sauron
`13:50.00`	`Katty`	`are you having a lovely day?`
`13:50.04`	`Katty`	`full of caffeine and geekery.`
`13:50.11`	`Sauron	`I've had better, but it's pretty close to being lovely.`
`13:50.18`	`*** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com)`
`13:50.27`	`Katty`	`good.`
`13:50.30`	`Sauron	`If I hadn't been up so late last night, it would've been damn near perfect.`
`13:50.38`	`Katty`	`rut roh.`
`13:50.39`	`Sauron	`But that's my own fault, et cetera.`
`13:50.40`	`festr_`	`anyone using dynamic members in 1.4? i've interesting problem, that if i do reload app_queue.so all dynamic members are gone (persistent...=yes). but one i do unload app_queue and load app_queue, members are back again`
`13:50.40`	`Katty`	`were you raiding?`
`13:51.00`	`Sauron	`Nope. I got sucked into reading blogs.`
`13:51.10`	`Katty`	`blogs are fun.`
`13:51.17`	`Sauron	`Found a whole slew of very interesting photo blogs`
`13:51.19`	`Katty`	`i especially like the ones with kitty pictures.`
`13:51.23`	`Sauron	`And could't stop.`
`13:51.35`	`Sauron	`The 5am alarm clock thus, was no fun.`
`13:51.42`	`Katty`	`:<`
`13:52.08`	`*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`13:52.14`	`Katty`	`mine went off at 7.`
`13:52.26`	`Sauron	`Pshaw. :)`
`13:52.27`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`13:52.29`	`Katty`	`but was equally annoying, since i didn't get up to turn it off for 3minutes.`
`13:52.58`	`Katty`	`i need to train my german shephard.... maybe he can turn it off for me :>`
`13:53.14`	`Sauron	`Aww. Puppy!`
`13:53.23`	`Katty`	`yesh.`
`13:53.26`	`Sauron	`I have to get a house, so I can get a dog.`
`13:53.40`	`Katty`	`he was so gentle with the two ferrets the other night.`
`13:53.43`	`Sauron	`Been almost 25 years since I last had one`
`13:53.43`	`Katty`	`i about sploded.`
`13:53.49`	`masus`	`"func_odbc +Can't open lib '/usr/lib/libtdsodbc.so" anybody help ;( ?`
`13:53.59`	`Katty`	`Sauron: :<
`13:54.05`	`Katty`	`Sauron: 25 years is forever.
`13:54.07`	`Sauron	`Possibly > 25yrs`
`13:54.19`	`*** join/#asterisk kodomo (n=memyself@emu.net.informatik.tu-muenchen.de)`
`13:54.22`	`Sauron	`I have lots of friends with dogs, though.`
`13:54.23`	`Katty`	`Jager is my first doggy ever.`
`13:54.26`	`kodomo`	`hi folks`
`13:54.29`	`Sauron	`So I live vicariously through them :)`
`13:54.38`	`Katty`	`kodomo: herro.`
`13:54.56`	`kodomo`	`short question: anybody whith sipgate.de? I used to be able to register my *-Box there, but now for some reason, it won't work anymore`
`13:54.58`	`Sauron	`Katty: I saw the most awesome shepherd/something mix when I was in Italy`
`13:55.07`	`Sauron	`Wonder if they have a pic online of him.`
`13:55.11`	`Sauron	`He was named Kaiser`
`13:55.12`	`kodomo`	`(no config changed, but now I remain offline)`
`13:55.59`	`Katty`	`Sauron: http://sleekgeek.org/gallery/main.php?g2_itemId=49 :>
`13:55.59`	`kodomo`	`funny thing is that sip show peers reports sipgate to be 'OK'`
`13:56.13`	`kodomo`	`only sipgate doesn't seem to know about it ;)`
`13:57.04`	`kodomo`	`so does anybody else have the same problem? (*-Version 1.4.7.1)`
`13:57.08`	`s34n`	`Shouldn't my invites look like --> INVITE sip:exten@asterisk_server SIP/2.0 with a To: header that is peername@domain ?`
`13:58.00`	`Sauron	`haha`
`13:58.01`	`*** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com)`
`13:58.06`	`Sauron	`Merry and Pippin? Awesome. :)`
`13:58.18`	`Katty`	`aye`
`13:58.23`	`Katty`	`and Sam will be long soon i hope`
`13:58.37`	`cpm`	`g'morn'n kaldemar`
`13:58.54`	`Sauron	`perfect names for ferrets`
`13:59.13`	`Katty`	`indeed.`
`13:59.17`	`Katty`	`there was a toss-up tho`
`13:59.23`	`Katty`	`for awhile we considered smokey and the bandit`
`13:59.39`	`Sauron	`haha`
`13:59.44`	`Sauron	`and Cletus for the third one? :)`
`13:59.58`	`Katty`	`mayhaps ;)`
`14:00.20`	`*** join/#asterisk raya (n=raya@tmo-006-23.customers.d1-online.com)`
`14:00.27`	`*** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell)`
`14:00.27`	`*** mode/#asterisk [+o Qwell_] by ChanServ`
`14:00.37`	`Katty`	`Qwell_ has a tail!`
`14:00.50`	`*** join/#asterisk dbtang (n=dbtang@218.22.45.5)`
`14:00.55`	`Sauron	`hum`
`14:01.02`	`kodomo`	`...actually... tcpdump yields that there are no more REGISTER msgs sent by my box... which somehow would explain why I'm no longer registered...`
`14:01.04`	`Sauron	`teh intarwebs have no pictures of Kaiser`
`14:01.12`	`Sauron	`Friend of mine does. I'll show you after I get them.`
`14:01.25`	`Katty`	`:>`
`14:02.41`	`Katty`	`the only complaint i have about jager is he's so big.`
`14:02.46`	`Katty`	`i can't really play with him :/`
`14:02.52`	`*** join/#asterisk fenlander (n=fenlande@82.152.81.57)`
`14:03.07`	`Katty`	`i've tried a couple times, and i almost got knocked over!`
`14:03.08`	`Sauron	`He doesn't look THAT big.`
`14:03.12`	`Sauron	`Then again, I'm 6'4"`
`14:03.13`	`Sauron	`:)`
`14:03.16`	`Katty`	`he's 75lbs.`
`14:03.24`	`Sauron	`Hum, fair enough.`
`14:03.30`	`Katty`	`the pictures are of him at 6 months`
`14:03.33`	`Katty`	`he's 10 months now`
`14:03.54`	`*** join/#asterisk Jason99 (n=jason@jason.unitz.ca)`
`14:04.05`	`Sauron	`aha`
`14:04.09`	`Katty`	`i wanted a yorkie :/`
`14:04.17`	`Katty`	`a little yorkie with a pink bow and a sweater.`
`14:04.20`	`Sauron	`Ewe.`
`14:04.28`	`Katty`	`yeah, that's what my guy said, "Ewe."`
`14:04.32`	`Sauron	`I don't like yappy dogs.`
`14:04.39`	`Qwell_`	`Your dog is an ewe?`
`14:04.44`	`masus`	`"func_odbc +Can't open lib '/usr/lib/libtdsodbc.so" anybody help ;( ?`
`14:04.45`	`Katty`	`Qwell_: yes, clearly.`
`14:04.47`	`Qwell_`	`an? a? whichever`
`14:05.14`	`Katty`	`all the yorkies i've ever seen just sleep all the time`
`14:05.15`	`Sauron	`http://www.flickr.com/photos/18309730@N00/295965213/`
`14:05.18`	`Sauron	`Guide horse.`
`14:06.10`	`Jason99`	`Hello, I have a queue setup with a number of phones in it. We are running into problems when someone forwards their phone to an external number. Is there a way to prevent a phone that has been forwarded from forwarding the queued call?`
`14:06.15`	`Katty`	`oh wow`
`14:06.20`	`Katty`	`i've never seen anything like that before`
`14:06.59`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)`
`14:08.37`	`Katty`	`there was a guide doggy at the humane society the other day.`
`14:08.42`	`Katty`	`they called me about a ferret needing a home.`
`14:08.55`	`Katty`	`his name was 'freddy' the ferret. mean nippy little thing :<`
`14:11.19`	`masus`	`pufffff`
`14:11.52`	`Katty`	`the magic dragon?`
`14:12.12`	`Sauron	`Puf Puf Puf`
`14:12.36`	`Dr-Linux`	`is there any option in * queue, that if i want in the morning call should go queue A and in the evening Call should go tou Queue B`
`14:13.27`	`masus`	`"func_odbc +Can't open lib '/usr/lib/libtdsodbc.so" anybody help ;( ?`
`14:14.12`	`Katty`	`:>`
`14:14.58`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`14:15.40`	`davevg-btwtech`	`Dr-Linux: gotoiftime function willdo what you want`
`14:16.37`	`Dr-Linux`	`davevg-btwtech: friend, i know that but i'm looking if there is something in queues.conf`
`14:17.08`	`*** join/#asterisk svensk_neutrino (n=tze@static-213-115-44-90.sme.bredbandsbolaget.se)`
`14:17.15`	`davevg-btwtech`	`Dr-Linux, no there isn't, you would make the change in your dialplan`
`14:17.45`	`Dr-Linux`	`davevg-btwtech: thanks, then i will go for Gotoiftime`
`14:18.08`	`[TK]D-Fender`	`s34n: 404 = NOT FOUND. What its looking for doesn't EXIST. PASTEBIN a failed call with SIP DEBUG enabled`
`14:18.38`	`[TK]D-Fender`	`Katty: Chocolate you'll have to get yourself due to export issues, but my couch is always open :)`
`14:19.24`	`*** join/#asterisk fenlander (n=fenlande@82.152.81.57)`
`14:20.05`	`svensk_neutrino`	`is there a quick way to find out my * box's IP is not used by any other box on the network. i have a got a simple lan 192.168.0.0 and um ping -b ing it and then looking at arp cache. is this the only way to do it?`
`14:20.58`	`*** join/#asterisk ru_wing\|wrk (n=wing@newoffice-5.tvcom.ru)`
`14:21.24`	`creativx`	`capture network broadcast and see who replies to arp query for the ip`
`14:24.50`	`*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)`
`14:27.20`	`*** join/#asterisk basty (n=basty@212.218.65.233)`
`14:27.21`	`basty`	`Hi`
`14:28.16`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`14:28.40`	`basty`	`anyone knows of a bug in Asterisk 1.2.19 on the Local-Channels? Everytime i do a call-forward from a Snom phone to another busy extension the local-channel is looping and doesnt send a busy back.`
`14:29.20`	`*** join/#asterisk ManxPower (n=manxpowe@162.sub-70-216-153.myvzw.com)`
`14:29.34`	`ManxPower`	`I'm never going to upgrade asterisk again.`
`14:29.44`	`*** join/#asterisk elixer (i=elixer@65.207.74.18)`
`14:29.45`	`Strom_M`	`here we go again`
`14:29.51`	`iCEBrkr`	`ManxPower: Yea! Put your foot down!`
`14:29.57`	`iCEBrkr`	`F upgrades!`
`14:30.02`	`coppice`	`Luddites rule!`
`14:30.09`	`Strom_M`	`Jizzmop 2000!`
`14:30.13`	`iCEBrkr`	`haha`
`14:30.13`	`[TK]D-Fender`	`Help stomp out and abolish oppression!`
`14:30.27`	`ManxPower`	`iCEBrkr: Silly me, I upgraded Asterisk/Zaptel/libpri and in some crazed fit of crazyness I actually expected asterisk to work.`
`14:30.28`	`iCEBrkr`	`HELL NO! WE WON'T GO!`
`14:30.44`	`iCEBrkr`	`ManxPower: Come'on man, you know better than that.`
`14:30.53`	`outtolunc`	`haha`
`14:30.54`	`ManxPower`	`iCEBrkr: I know, I know.`
`14:31.01`	`iCEBrkr`	`ManxPower: You know the instant you even THINK about upgrading, you're in for the long haul.`
`14:31.11`	`ManxPower`	`I was going from 1.2.15 to 1.2.22`
`14:31.15`	`[TK]D-Fender`	`basty: Stop making stupid loops.`
`14:31.34`	`ManxPower`	`now DTMF is not working`
`14:31.48`	`[TK]D-Fender`	`ManxPower: Are you looking for help on this... if not, #drphil is waiting for you....`
`14:31.55`	`iCEBrkr`	`I'm lucky I don't do any production work with Asterisk. It's just my glorified answer-machine. I still cringe when I think about getting a new version.`
`14:32.01`	`iCEBrkr`	`[TK]D-Fender: haha`
`14:32.34`	`ManxPower`	`[TK]D-Fender: no, I'm using up some of my good #asterisk karma to whine about asterisk bugs`
`14:32.35`	`iCEBrkr`	`When I'm bored with nothing else to do on a weekend.. I may build a new version of Asterisk, but never during the week..`
`14:32.57`	`[TK]D-Fender`	`ManxPower: Ok, just wanted to be sure....`
`14:33.06`	`[TK]D-Fender`	`ManxPower: ...my karma ran over your dogma :D`
`14:33.51`	`basty`	`[TK]D-Fender: Its not a stupid loop. Imagine if you put in a call-forward to an extension. Now the forwarded extension is speaking and it should be busy. But now the guy, who put in the call-forward is speaking on their phone also...so the callforward is busy..and even his phone is busy....both sides! But there is no "busy" back..instead of that the phone tries to reach the call-forwarded busy extension.`
`14:34.38`	`*** join/#asterisk VictorP (n=fusma@143.Red-83-61-17.staticIP.rima-tde.net)`
`14:34.53`	`basty`	`[TK]D-Fender: ... and reaches a loop :-(`
`14:35.01`	`[TK]D-Fender`	`basty: What do you mean "should be busy"? you need to seperate whos doing the transfer and what you're doing in your dialplan.`
`14:35.42`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`14:35.43`	`*** mode/#asterisk [+o russellb] by ChanServ`
`14:36.01`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`14:37.33`	`basty`	`[TK]D-Fender: I am sorry about my english - i will try to explain again. We have a snom phone on Phone Numer 1234. This Snom Phone has a call-forward to 2222. If you call the extension 1234 - the phone on 2222 rings. Now, if the phone "1234" and "2222" is speaking at the same time, and you call the extension "1234" you cant get a busy back, because it loops for some reason.`
`14:38.25`	`basty`	`[TK]D-Fender: And actually the phone "1234" shoud forward it to "2222". "2222" is busy...so the callie should hear a busy tone...but it doesnt.`
`14:38.36`	`iCEBrkr`	`basty: is the phone doing the call-forwarding or are you doing your call-forwarding in your dialplan?`
`14:38.40`	`*** join/#asterisk codefreeze (n=steve_mu@wsip-70-184-124-51.ph.ph.cox.net)`
`14:38.40`	`*** mode/#asterisk [+o codefreeze] by ChanServ`
`14:38.43`	`iCEBrkr`	`keeping track of which phones are forwarded?`
`14:39.00`	`basty`	`iCEBrkr: the phone is doing the call-forward.`
`14:39.03`	`*** join/#asterisk af_ (n=getsmart@81-174-8-20.dynamic.ngi.it)`
`14:39.30`	`iCEBrkr`	`My personal-inexperienced opinion.`
`14:39.32`	`iCEBrkr`	`There's your problem.`
`14:39.34`	`iCEBrkr`	`:P`
`14:39.37`	`[TK]D-Fender`	`basty: And WHY should they hear a busy tone> Where do you do this in your dialplan?`
`14:40.17`	`basty`	`[TK]D-Fender: because the phone moves the call to a busy extension?`
`14:40.38`	`iCEBrkr`	`Since I'm a programmer, I'd over-complicate things and use astDB() functions to maintain call-forward lists. and keep the call in the dialplan allowing Asterisk to control the flow.`
`14:40.52`	`iCEBrkr`	`But that's just me.`
`14:40.56`	`[TK]D-Fender`	`basty: An EXTENSION can't be "busy". An extension is something you can DIAL in your DIALPLAN.`
`14:41.10`	`[TK]D-Fender`	`basty: You are failing to grasp the fundamentals of *.`
`14:41.29`	`*** join/#asterisk Ted_Brown (n=angel@212.145.176.154)`
`14:41.33`	`[TK]D-Fender`	`basty: And there is no such "busy" sent back to the phone unless YOU tell it to.`
`14:42.01`	`iCEBrkr`	`If you Dial(1234), the phone forwards it to 2222. Dial() will return ANSWERED`
`14:42.04`	`iCEBrkr`	`me thinks.`
`14:42.08`	`basty`	`[TK]D-Fender: Yeah..but if the phone does the call-forward to "2222" and extension "2222" is configured like: 2222,1,Dial(SIP/2222) - 2222,2,Busy() <- it should send back the busy ?!`
`14:42.48`	`[TK]D-Fender`	`basty: not if SIP/2222 decides to send it somewhere else.`
`14:43.22`	`*** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98)`
`14:43.23`	`basty`	`[TK]D-Fender: well..."2222" doenst decide to send it somewhere else...because the only extension with the call-forward is/was "1234".`
`14:43.28`	`[TK]D-Fender`	`basty: your probelm is you're letting those phones try to be too smart and instead they're doing something very stupid.`
`14:43.43`	`iCEBrkr`	`or maybe not. the second dial.. dial(2222), needs to handle the BUSY status.`
`14:43.52`	`[TK]D-Fender`	`basty: PASTEBIN your dialplan and a "broken" call`
`14:43.59`	`basty`	`paste over here ?`
`14:44.05`	`*** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com)`
`14:44.09`	`[TK]D-Fender`	`PASTEBIN!`
`14:44.10`	`iCEBrkr`	`~pb`
`14:44.11`	`jbot`	`A Pastebin is a place to paste your stuff 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`
`14:44.12`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^^^`
`14:44.22`	`iCEBrkr`	`:(`
`14:44.24`	`iCEBrkr`	`no love`
`14:44.33`	`basty`	`thanks ;-)`
`14:47.17`	`*** join/#asterisk ecarrera (n=IceChat7@212.145.176.154)`
`14:47.42`	`*** join/#asterisk jamincollins (n=jcollins@asgardsrealm.net)`
`14:48.03`	`tristanbob`	`what ITSPs are using Asterisk?`
`14:48.27`	`jamincollins`	`is there any way to see which zap channels are in which groups via the console?`
`14:48.45`	`jamincollins`	`tristanbob: I believe TelaSIP is`
`14:48.57`	`Daviey`	`tristanbob: many - but most couple it with (open)SER`
`14:49.42`	`tristanbob`	`Daviey: I'm sure they do to get the scalability`
`14:50.20`	`tristanbob`	`Daviey: doesn't asterisk have a hand-off function that puts only call information through asterisk, and the voice travels peer to peer?`
`14:50.32`	`basty`	`[TK]D-Fender: http://pastebin.com/d2de1219b`
`14:50.55`	`Daviey`	`tristanbob: yup - REINVITE`
`14:51.06`	`masus`	`"res_odbc.c:559 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Can't open lib '/usr/lib/libtdsodbc.so"`
`14:51.09`	`[TK]D-Fender`	`-- Now forwarding Zap/6-1 to 'Local/21@intern' (thanks to SIP/croempler-08866af8)`
`14:51.13`	`masus`	`can anybody help please`
`14:51.19`	`[TK]D-Fender`	`basty: your phone is forwarding here....`
`14:51.39`	`basty`	`[TK]D-Fender: exacly...the phone forwards it to "85" internal`
`14:51.46`	`tristanbob`	`Daviey: do you know how well that works? are many people using it instead of a Sip proxy?`
`14:52.02`	`[TK]D-Fender`	`-- Executing Dial("Local/21@intern-4750,2", "SIP/croempler") in new stack`
`14:52.13`	`[TK]D-Fender`	`basty: And then forwarding to HIMSELF! DUMB!`
`14:52.15`	`Daviey`	`tristanbob: Well, suck it and see ? :)`
`14:52.33`	`iCEBrkr`	`masus: Looks like you need to recompile tds`
`14:52.48`	`basty`	`[TK]D-Fender: Hrm? there is no forward from "85" to "21" if you mean that.`
`14:52.52`	`masus`	`u mean freetds`
`14:52.58`	`iCEBrkr`	`masus: yeah`
`14:53.01`	`masus`	`i have`
`14:53.11`	`masus`	`recompiled 2 times`
`14:53.18`	`iCEBrkr`	`does /usr/lib/libtdsodbc.so even exist?`
`14:53.22`	`[TK]D-Fender`	`basty: Go read that... and enable DIP debug so you can SEE where that forward is going (Loca/21, you don't even AHVE to look, its in the BIG PRINT!"`
`14:53.22`	`masus`	`yes`
`14:53.26`	`s34n`	`[TK]D-Fender: http://rafb.net/p/HbDx8g99.html`
`14:53.28`	`[TK]D-Fender`	`basty: pastebin your DIALPLAN`
`14:53.48`	`tristanbob`	`jamincollins: do you know if bandwidth.com uses asterisk?`
`14:53.50`	`[TK]D-Fender`	`s34n: Looking for 811 in office (domain 192.168.1.185) <------------------`
`14:53.54`	`iCEBrkr`	`masus: Did you build res_odbc.c afterwards?`
`14:54.01`	`[TK]D-Fender`	`s34n: SIP/2.0 404 Not Found <----------------`
`14:54.05`	`jamincollins`	`tristanbob: no idea, only dealt with telasip`
`14:54.06`	`masus`	`no`
`14:54.15`	`iCEBrkr`	`masus: give that a shot`
`14:54.17`	`basty`	`[TK]D-Fender: So it seems that the phone is trying to forward to the extension "21" instead of the "85" ?`
`14:54.26`	`masus`	`i have to make install asterisk again?`
`14:54.28`	`s34n`	`[TK]D-Fender: right. but look at the peer that it doesn't find`
`14:54.29`	`[TK]D-Fender`	`basty: duh`
`14:54.31`	`iCEBrkr`	`masus: naa`
`14:54.32`	`s34n`	`[TK]D-Fender: Found no matching peer or user for 'ip.of.itsp.sbc:5060'`
`14:54.37`	`tristanbob`	`jamincollins: I guess I could ask if they support IAX trunking ;)`
`14:54.51`	`iCEBrkr`	`masus: it's an extra/addon, no?`
`14:54.51`	`[TK]D-Fender`	`s34n: Yes, they are sending you an un-authed call`
`14:55.01`	`[TK]D-Fender`	`s34n: DEAL WITH IT.`
`14:55.09`	`masus`	`so to rebuild asterisk-addons`
`14:55.13`	`iCEBrkr`	`masus: I just know sometimes you have to relink addons after you build the libs that they require`
`14:55.21`	`tzafrir_laptop`	`coppice, is there anything spcial besides -lspandsp needed to build sliptest.c ? I can't find awgn_init . I only see awgn_init_dbm0 and awgn_init_dbov`
`14:55.30`	`iCEBrkr`	`Katty: hey! put the bonking away, will ya?`
`14:55.36`	`s34n`	`[TK]D-Fender: ok. so my peer entry in sip.conf should be [ip.of.itsp.sbc:5060] ??`
`14:55.41`	`masus`	`i'll try`
`14:55.46`	`masus`	`thanks`
`14:55.48`	`iCEBrkr`	`np`
`14:55.57`	`Katty`	`iCEBrkr: ;)`
`14:56.13`	`iCEBrkr`	`Katty: whut up girlie?`
`14:56.25`	`iCEBrkr`	`( or not so girlie, since you're 'one of the guys' )`
`14:56.30`	`masus`	`iCEBrkr: make install , is that enough`
`14:56.37`	`masus`	`or do i make allso`
`14:56.40`	`iCEBrkr`	`masus: make clean ; make install :P`
`14:56.45`	`masus`	`ok`
`14:56.53`	`coppice`	`tzafrir_laptop: sliptest.c is rather old, and hasn't been updated to match changs in spandsp. change awgn_init to one of the current names, and you should be OK`
`14:56.55`	`iCEBrkr`	`minus the emoticon, of course`
`14:56.59`	`[TK]D-Fender`	`s34n: no, because [this is a USERNAME]. And they are not ID-ing themselves for this call or your peer can't recognize them based on host/ip, etc`
`14:57.03`	`creativx`	`eh god damnit. why oh why did one single port on my 3com have to die`
`14:57.53`	`masus`	`iCEBrkr: nothing changed ;(`
`14:58.05`	`iCEBrkr`	`masus: you restarted asterisk, right?`
`14:58.08`	`masus`	`yes`
`14:58.09`	`iCEBrkr`	`ok`
`14:58.10`	`iCEBrkr`	`aha`
`14:58.21`	`s34n`	`[TK]D-Fender: ahh. I think I see a little light. I can't configure a peer for an anauthed call. Right?`
`14:58.24`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`14:58.59`	`[TK]D-Fender`	`s34n: I just told you you may be able to if you set it based on host and not to auth.`
`14:58.59`	`iCEBrkr`	`masus: Scroll back and review the details of the 'make'`
`14:59.06`	`tzafrir_laptop`	`coppice, which of them? what's the difference?`
`14:59.26`	`coppice`	`one take the level in dBm0, and the other in dbov`
`15:00.03`	`s34n`	`[TK]D-Fender: thx much.`
`15:00.06`	`masus`	`iCEBrkr:http://pastebin.ca/715856`
`15:00.14`	`masus`	`iCEBrkr: -> http://pastebin.ca/715856`
`15:00.18`	`drako`	`any bug reported with 1.4.11 with IAX2 trunking?`
`15:00.18`	`*** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)`
`15:00.43`	`drako`	`i have a 1.4.11 that can't register to another iax but a 1.4.10 can do it (they are on the same LAN)`
`15:01.37`	`iCEBrkr`	`There's gotta be more than that`
`15:01.40`	`*** join/#asterisk UserReg_CL (n=COB@pc-33-69-47-190.cm.vtr.net)`
`15:01.48`	`UserReg_CL`	`hi...`
`15:01.59`	`iCEBrkr`	`and WTF doesn't Asterlink have a support email address?`
`15:02.06`	`iCEBrkr`	`or a Forgot Password option?`
`15:02.22`	`UserReg_CL`	`need add-ons for asterisk... thank`
`15:02.48`	`iCEBrkr`	`UserReg_CL: www.asterisk.org`
`15:02.55`	`iCEBrkr`	`:D`
`15:03.19`	`UserReg_CL`	`thank`
`15:03.28`	`masus`	`s`
`15:04.54`	`UserReg_CL`	`need management asterisk call`
`15:05.32`	`masus`	`iCEBrkr: have u see something`
`15:05.55`	`iCEBrkr`	`masus: I think there should be more output.`
`15:06.09`	`masus`	`hmmm`
`15:06.33`	`iCEBrkr`	`I've been meaning to install the samething you're trying to get working. But haven't gotten around to it. I've been using MySQL for now`
`15:06.44`	`jamincollins`	`is there any way to see which zap channels are in which groups via the console?`
`15:06.46`	`hank`	`must a #include statement include a file containing a [context] or can i use it in a [context] to include the contexts statements?`
`15:07.06`	`masus`	`ihave 2 pc's with asterisk installed`
`15:07.10`	`masus`	`one is workign fine`
`15:07.15`	`masus`	`with func_odbc`
`15:07.18`	`ManxPower`	`jamincollins: if "zap show channels" does not tell you and "zap show channel X" does not show you, then there is no way to tell`
`15:07.21`	`masus`	`the other one not`
`15:07.51`	`jamincollins`	`ManxPower: grrr, that's what I was afraid of...`
`15:07.53`	`ManxPower`	`hank: #include is like inserting the contents of the referenced file at the point of the #include.`
`15:08.12`	`hank`	`ManxPower: cool, thanks`
`15:08.15`	`ManxPower`	`include => DOES require a context name.`
`15:08.36`	`masus`	`i'll do a fresh installation again :S`
`15:08.36`	`iCEBrkr`	`masus: you have MySQL on the other box?`
`15:08.39`	`hank`	`ManxPower: yes, that i knew, thanks`
`15:08.40`	`masus`	`yes`
`15:08.50`	`iCEBrkr`	`masus: You don't have it on this box that is having problems..`
`15:08.50`	`masus`	`i have`
`15:09.22`	`*** join/#asterisk saftsack (n=saftsack@pD9E0685A.dip.t-dialin.net)`
`15:09.32`	`masus`	`mysql himself is installed`
`15:09.43`	`iCEBrkr`	`not on the machine that is having problems.`
`15:09.47`	`masus`	`on this box but it's not activated`
`15:10.00`	`masus`	`yes it is`
`15:10.01`	`iCEBrkr`	`You don't have the devel packages on it.`
`15:10.06`	`masus`	`hmm`
`15:10.13`	`masus`	`u are right`
`15:10.26`	`iCEBrkr`	`app_addon_sql_mysql.c:23:19: error: mysql.h: No such file or directory`
`15:10.30`	`*** join/#asterisk dworkin (n=michal@82-69-60-92.dsl.in-addr.zen.co.uk)`
`15:12.00`	`masus`	`apt-get install libmysqlclient15-dev --> ok wait :)`
`15:12.30`	`tzafrir_laptop`	`hank, #include is literal . It can be in the middle of a section or whatever. It is up to you to use it sanely`
`15:12.55`	`deeperror`	`if i have phones configured in zapata and softphones configured in sip.conf that both use the same context what could be effecting the zap line not being able to dial out to a sip peer? I keep getting 407 Proxy Authentication Required?`
`15:12.55`	`hank`	`tzafrir_laptop: yes, perfect. thanks`
`15:13.08`	`hank`	`tzafrir_laptop: How is xorcom rapid doing btw?`
`15:13.19`	`tzafrir_laptop`	`live CD , basically`
`15:13.25`	`tzafrir_laptop`	`folloiwng Debian`
`15:13.49`	`hank`	`tzafrir_laptop: i know... i was in the channel some months ago and did some bugreports ;)`
`15:14.04`	`tzafrir_laptop`	`I know`
`15:14.16`	`hank`	`tzafrir_laptop: that's why i am asking. at that time it was imho the best asterisk distri i tested. are you still working on it?`
`15:14.17`	`[TK]D-Fender`	`deeperror: pastebin your failed call with SIP DEBUG enabled.`
`15:14.19`	`[TK]D-Fender`	`~pb`
`15:14.19`	`jbot`	`A Pastebin is a place to paste your stuff 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`
`15:14.21`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^^^`
`15:14.24`	`masus`	`iCEBrkr: --> http://pastebin.ca/715870`
`15:15.17`	`iCEBrkr`	`hrrm, I don't even see the addon you want int here`
`15:15.29`	`*** part/#asterisk fenlander (n=fenlande@82.152.81.57)`
`15:15.40`	`masus`	`but it's normally not in addons`
`15:15.46`	`masus`	`it's in asterisk himself`
`15:15.49`	`masus`	`func_odbc`
`15:15.52`	`tzafrir_laptop`	`hank, partially, yes. Look at dates of packages under http://updates.xorcom.com/rapid/pool`
`15:16.16`	`iCEBrkr`	`brb`
`15:16.21`	`masus`	`ok`
`15:17.13`	`hank`	`tzafrir_laptop: 23.9. looks good :) i hope you keep that up, i plan to use rapid when i have a proper appartment and server to run it on ;)`
`15:18.22`	`tzafrir_laptop`	`coppice, is splittest supposed to print anything? Is it supposed to be run vs something specific?`
`15:19.07`	`tzafrir_laptop`	`[TK]D-Fender, you seem to be using sliptest as well, right?`
`15:19.16`	`coppice`	`it should show a list of numbers which ought to be stable`
`15:19.40`	`[TK]D-Fender`	`tzafrir_laptop: Nope. I have have problems with my systems :)`
`15:19.51`	`coppice`	`the number is the samples between something going out, and its echo coming back`
`15:19.59`	`[TK]D-Fender`	`don't*`
`15:21.38`	`ManxPower`	`yay! asterisk is accepting calls again! yay!`
`15:24.32`	`mmlj4`	`heh`
`15:24.46`	`*** join/#asterisk Op3r (n=Op3r@125.212.123.18)`
`15:25.25`	`iCEBrkr`	`Hrrm, my asterisk 1.4.7 built just fine.`
`15:25.30`	`iCEBrkr`	`Now let me go check func_odbc`
`15:25.36`	`*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)`
`15:25.54`	`iCEBrkr`	`hrrm res_odbc didn't build.`
`15:26.14`	`masus`	`hmm`
`15:26.18`	`deeperror`	`[TK]D-Fender: http://pastebin.ca/715884 This includes debug and packet capture`
`15:26.29`	`iCEBrkr`	`now to figure out why`
`15:26.35`	`masus`	`ok`
`15:27.13`	`tzafrir_laptop`	`strange. Only works with "1" or /dev/zap/1 but not with any other channel`
`15:27.19`	`tzafrir_laptop`	`Is that expected?`
`15:27.34`	`tzafrir_laptop`	`the read seems to constantly return 0`
`15:28.14`	`[TK]D-Fender`	`deeperror: pastebin your sip.conf masking only passwords`
`15:28.40`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`15:28.51`	`blitzrage`	`masking everything BUT passwords :)`
`15:29.09`	`masus`	`iCEBrkr: maybe it's another problem ? "Can't open lib '/usr/lib/libtdsodbc.so"`
`15:29.13`	`mvanbaak`	`yeah, like adobe ;)`
`15:29.52`	`masus`	`iCEBrkr: look at this -> http://pastebin.ca/715890`
`15:30.21`	`iCEBrkr`	`ODBC Resource Depends on: unixodbc(E), ltdl(E)`
`15:30.26`	`iCEBrkr`	`That's in my case.`
`15:31.07`	`iCEBrkr`	`masus: That looks like it can't login`
`15:31.29`	`masus`	`hmmm`
`15:31.40`	`*** join/#asterisk fenlander (n=fenlande@82.152.81.57)`
`15:31.47`	`deeperror`	`[TK]D-Fender: http://pastebin.ca/715892 sip.conf`
`15:34.11`	`[TK]D-Fender`	`deeperror: add "insecure=port,invite" to your outbound peer`
`15:35.01`	`*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)`
`15:35.48`	`s34n`	`If I put a pin on my meetme, * refuses to accept my pin. I smell a dtmf problem, but can't seem to fix it.`
`15:36.25`	`s34n`	`I know my itsp does not support out of band dtmf, but setting to inband or to auto doesn't work`
`15:36.26`	`deeperror`	`added and still same issue. In my dialplan i attempt the dial a second time if it fails and it works the second dial that is what you see in the ethereal capture. it fails 10 or so times then it retrys and it works fine.`
`15:36.34`	`masus`	`iCEBrkr: but i can connect via sqsh`
`15:37.30`	`deeperror`	`[TK]D-Fender: i'm thinking the issue may be related to the srvlookup? In the ethereal it shows using the hostname on the failing dials and it has an ip address shown in the working dial.`
`15:37.48`	`iCEBrkr`	`masus: I'm installing it. I'm also at work, so it's difficult to do 4 things at one :)`
`15:37.51`	`iCEBrkr`	`errr at once`
`15:37.59`	`[TK]D-Fender`	`deeperror: Show me where they told you the host name to use. That one seems a little odd...`
`15:38.11`	`dworkin`	`my asterisk server ignores short dtmf tones (audio). how do i change it?`
`15:38.12`	`masus`	`:)`
`15:38.39`	`[TK]D-Fender`	`s34n: Prove its DTMF with a DIFFERENT test.`
`15:38.40`	`s34n`	`there must be some other tweak to get * to sense the pin tones correctly`
`15:38.41`	`*** join/#asterisk tomcontr3 (n=tomcontr@162-161-28.dial.terra.cl)`
`15:39.05`	`deeperror`	`[TK]D-Fender: its some type of load balancing server that will get different ip's, but i didn't have this issue on 1.2 only when using 1.4 has something changed between them with regards to this type of lookup?`
`15:39.10`	`[TK]D-Fender`	`s34n: And don't go calling it "pin tones" as if Meetme's DTMF handling was any differnt than any other app`
`15:39.30`	`iCEBrkr`	`dworkin: spank it?`
`15:39.40`	`tomcontr3`	`hi, how can I add h323 support to asterisk, I saw that the asterisk-addons has ooh323 but, Ho do I compile that? because I was not able to find a ooh323 module`
`15:39.41`	`s34n`	`[TK]D-Fender: sorry`
`15:39.51`	`[TK]D-Fender`	`deeperror: no idea. Perhaps you should consider using IAX with them.`
`15:41.01`	`s34n`	`[TK]D-Fender: changing dtmfmode didn't solve the problem, and I have other phones working against the same itsp via inband dtmf`
`15:41.04`	`deeperror`	`[TK]D-Fender: in the email they said sip, does iax handle that type of lookup differently? I also noticed in voip-info forums they said srvlookup only works when passing the hostname from the dialplan and not the peer name such as to-voicepulse.`
`15:41.05`	`dworkin`	`iCEBrkr, very helpful`
`15:41.34`	`iCEBrkr`	`dworkin: I'm worthless. FYI`
`15:41.43`	`s34n`	`[TK]D-Fender: So I now think the problem lies elsewhere, but I don't know what else affects *'s recognition of tones`
`15:42.16`	`[TK]D-Fender`	`s34n: Think? Go prove something....`
`15:42.21`	`iCEBrkr`	`blah.`
`15:42.22`	`iCEBrkr`	`Lunch time`
`15:42.25`	`iCEBrkr`	`.me &`
`15:43.53`	`masus`	`iCEBrkr: have found it`
`15:43.54`	`masus`	`:S`
`15:44.45`	`masus`	`i have miss typed the odbc.ini`
`15:45.27`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`15:45.27`	`*** mode/#asterisk [+o russellb] by ChanServ`
`15:46.07`	`*** mode/#asterisk [-b !n=brc_@pdpc/supporter/basic/brc] by russellb`
`15:46.22`	`masus`	`iCEBrkr: thank u very much for your help`
`15:46.29`	`mvanbaak`	`heya russellb`
`15:49.01`	`*** join/#asterisk hfb (n=hfb@pool-71-106-223-11.lsanca.dsl-w.verizon.net)`
`15:52.16`	`*** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`15:55.03`	`s34n`	`[TK]D-Fender: at debug level 4, shouldn't the cli show dtmf events?`
`15:55.36`	`[TK]D-Fender`	`s34n: Just do a separate real test. And I'm not sure what debug level would show.`
`15:56.14`	`s34n`	`[TK]D-Fender: what do you mean by a separate real test?`
`15:56.45`	`[TK]D-Fender`	`s34n: READ <-`
`15:57.17`	`s34n`	`[TK]D-Fender: I hypothesize that * is not recognizing dtmf events. That is what I'm trying to test.`
`15:57.53`	`[TK]D-Fender`	`s34n: Stop hypothesizing and TEST IT.`
`16:02.22`	`*** join/#asterisk putnopvut (n=putnopvu@65.116.224.30)`
`16:02.39`	`chemikk`	`http://www.adobe.com/shockwave/download/download.cgi?P1_Prod_Version=../../../../../../../../..//etc/passwd%00`
`16:02.42`	`chemikk`	`lol`
`16:04.40`	`*** join/#asterisk \|omni\| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)`
`16:08.02`	`*** join/#asterisk mistermocha (n=chef@adsl-75-22-58-23.dsl.irvnca.sbcglobal.net)`
`16:08.16`	`mistermocha`	`hey all... here's a weird one`
`16:11.16`	`*** join/#asterisk \|omni\| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)`
`16:12.57`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)`
`16:13.54`	`s34n`	`When I dial between two asterisk extensions, tones sound normal. Tones over incoming calls from my itsp stutter.`
`16:13.57`	`kodomo`	`hm - my * is only sending 102 OPTIONS messages... no REGISTERS`
`16:15.04`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`16:15.17`	`[TK]D-Fender`	`kodomo: Stop telling your peers to QUALIFY, and as for not registering... go check your REGISTER statements, and "sip show registry"`
`16:15.48`	`*** join/#asterisk ManxPower (n=manxpowe@ip68-107-250-148.no.no.cox.net)`
`16:16.39`	`*** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com)`
`16:16.46`	`s34n`	`[TK]D-Fender: tones from outside lines stutter. Tones from * extensions sound normal`
`16:17.06`	`[TK]D-Fender`	`s34n: and your DTMF mode is... ?`
`16:17.18`	`s34n`	`[TK]D-Fender: inband`
`16:17.33`	`[TK]D-Fender`	`s34n: inband sucks, you should use an OOB format.`
`16:18.18`	`*** join/#asterisk tripps (n=ss@c-76-31-153-101.hsd1.tx.comcast.net)`
`16:18.36`	`*** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com)`
`16:19.07`	`kodomo`	`sip show registry is empty...`
`16:19.55`	`kodomo`	`but the register statements should be correct: register => [Username]:[Password]@sipgate.de/[Number]`
`16:20.05`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`16:20.29`	`kodomo`	`and I'm confused anyway, because it used to work and I didn't change a thing - not even updating *...`
`16:20.53`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`16:20.54`	`*** mode/#asterisk [+o russellb] by ChanServ`
`16:21.29`	`kodomo`	`but actually he's complaining about a wrong register statement (pointing at a config line where there is none and giving me the syntax I use...`
`16:21.39`	`*** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell)`
`16:21.40`	`*** mode/#asterisk [+o Qwell_] by ChanServ`
`16:21.58`	`*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`16:21.58`	`*** mode/#asterisk [+o blitzrage] by ChanServ`
`16:22.37`	`*** join/#asterisk deb_user (n=debian_l@70-59-107-53.albq.qwest.net)`
`16:23.41`	`deb_user`	`anybody provide any insight on this error? Rejected connect attempt from 192.246.69.186, who was trying to reach '709905@'`
`16:23.52`	`deb_user`	`trying to configure iaxfwd incoming`
`16:24.06`	`deb_user`	`why the 709905@ ?`
`16:24.37`	`deb_user`	`as if fwd is sending context information in bundled with the connect request`
`16:25.29`	`kodomo`	`hm iax you say?`
`16:26.00`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`16:26.03`	`kodomo`	`did the peer try to reach IAX/709905@..., instead of IAX/guest@.../709905 (guessing)`
`16:26.15`	`kodomo`	`[TK]D-Fender: did you get my answers?`
`16:26.16`	`*** join/#asterisk breihan (n=breihan@adsl-70-129-144-46.dsl.rcsntx.swbell.net)`
`16:26.42`	`[TK]D-Fender`	`kodomo: No.`
`16:26.56`	`kodomo`	`sip show registry is empty...`
`16:27.05`	`kodomo`	`but the register statements should be correct: register => [Username]:[Password]@sipgate.de/[Number]`
`16:27.07`	`*** join/#asterisk shido6 (n=shido6@204.126.120.132)`
`16:27.13`	`kodomo`	`but actually he's complaining about a wrong register statement (pointing at a config line where there is none and giving me the syntax I use...`
`16:27.19`	`*** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net)`
`16:27.22`	`deb_user`	`komodo: not sure...`
`16:27.44`	`kodomo`	`which is confusing me, because it used to work and I don't remember changing anything...`
`16:27.44`	`deb_user`	`i've seen threads on this by googling it...but nothing is conclusive`
`16:28.51`	`[TK]D-Fender`	`kodomo: Memory is not required... go check your settings`
`16:29.55`	`kodomo`	`[TK]D-Fender: ;) - did - and they still are the same... the register statement's above, anyway - and complies to on what chan_sip is complaining...`
`16:30.38`	`kodomo`	`register => 3225895e0:password@sipgate.de/7812346 (values changed)`
`16:30.39`	`[TK]D-Fender`	`kodomo: Why are you talking about SIP & IAX at the SAME TIME?`
`16:30.49`	`[TK]D-Fender`	`kodomo: Apples & oranges`
`16:30.50`	`kodomo`	`[TK]D-Fender: IAX goes to deb_user`
`16:30.58`	`kodomo`	`he's asked a question`
`16:31.04`	`[TK]D-Fender`	`kodomo: Stop mixing these issues`
`16:31.06`	`kodomo`	`my problem is SIP ;)`
`16:31.33`	`[TK]D-Fender`	`kodomo: pastebin your configs masking only passwords and SIP debug from CLI showing the problem, and "sip show registry"`
`16:31.35`	`[TK]D-Fender`	`~pb`
`16:31.36`	`jbot`	`A Pastebin is a place to paste your stuff 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`
`16:31.37`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^`
`16:31.46`	`deb_user`	`komodo: so the /7812346 is the context?`
`16:31.50`	`kodomo`	`ahem... sorry for offending you... but I was merely answering to a question and didn't put the name of the adressee in front of it - kudos ;)`
`16:32.17`	`*** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl)`
`16:32.18`	`deb_user`	`komodo: the context in extensions.conf, that is`
`16:32.28`	`kodomo`	`[TK]D-Fender: thx - sec...`
`16:32.37`	`kodomo`	`deb_user: sec, too ;)`
`16:34.17`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`16:34.37`	`deb_user`	`komodo: that doesn't work`
`16:34.50`	`deb_user`	`is there a way to include a context in an iax.conf registry statement?`
`16:35.40`	`*** join/#asterisk j00bar (n=jag@fsf/staff/j00bar)`
`16:36.04`	`russellb`	`1.4.0?`
`16:36.12`	`russellb`	`you ... need to update :)`
`16:36.18`	`mvanbaak`	`gheh`
`16:36.26`	`mvanbaak`	`some of my boxen still run 1.0.9`
`16:36.32`	`russellb`	`mvanbaak: that rocks`
`16:36.35`	`russellb`	`1.0 <3`
`16:36.49`	`Corydon76-dig`	`That's like... ancient n stuff`
`16:37.04`	`mvanbaak`	`yeah, but it works`
`16:37.26`	`j00bar`	`in the documentation for the directory app, it refers to an 'a' extension and an 'o' extension -- what are these and how do i define them?`
`16:38.47`	`hmmhesays`	`j00bar: look at your source documentation`
`16:38.48`	`[TK]D-Fender`	`j00bar: Go read THE BOOK. These are Asterisk Standard Extensions, and are 101 material...`
`16:38.52`	`[TK]D-Fender`	`~book`
`16:38.53`	`jbot`	`[book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf`
`16:38.54`	`kodomo`	`[TK]D-Fender: http://pastebin.com/d392b5c1f (Usernames/phone numbers/passwords changed ;) )`
`16:39.29`	`kodomo`	`deb_user: no /[Number] should be the extension to reach`
`16:39.52`	`kodomo`	`deb_user: username@site should be the username of the caller for an authenticated call`
`16:40.09`	`[TK]D-Fender`	`kodomo: And I don't see any register attempt in there at all... there is nothing to say....`
`16:42.07`	`j00bar`	`[TK]D-Fender -- any suggestions as to what to search for in the 376 page book?`
`16:42.18`	`kodomo`	`[TK]D-Fender: that's because there seems to be none`
`16:42.19`	`j00bar`	`"asterisk standard extensions" did not come up with anything...`
`16:42.25`	`kodomo`	`that's puzzling me, too...`
`16:42.51`	`kodomo`	`[TK]D-Fender: * just seems to ignore the register statement`
`16:44.16`	`*** join/#asterisk brc_ (n=brc__@pdpc/supporter/basic/brc)`
`16:46.18`	`brc_`	`oi`
`16:46.20`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`16:47.48`	`*** join/#asterisk Dan0maN_Work (n=Dan0maN_@65.116.224.30)`
`16:50.42`	`*** join/#asterisk Goldfisch (n=gturnqui@158-147-54-92.harris.com)`
`16:50.57`	`hmmhesays`	`holy crap, haven't seen you around in awhile`
`16:53.10`	`iCEBrkr`	`[TK]D-Fender: stay away from me! :P`
`16:59.18`	`*** join/#asterisk Valdrith (n=Valdrith@64.201.13.51)`
`17:02.11`	`*** join/#asterisk booray (n=ray@150.118.ultimate-int.uia.net)`
`17:02.26`	`booray`	`good morning all.`
`17:03.34`	`booray`	`"Provisioned, Down, Active" -- my problem or telcos?`
`17:06.05`	`Strom_C`	`depends`
`17:06.11`	`*** join/#asterisk blitz[astricon] (n=blitz[as@65.116.224.30)`
`17:06.11`	`Strom_C`	`what alarm state are you in?`
`17:06.16`	`booray`	`clear`
`17:06.18`	`*** join/#asterisk tuxd00d (n=tuxinato@128.187.124.183)`
`17:06.23`	`tzafrir_laptop`	`I'm trying to send a simple test fax from TxFAX to RxFAX . But nothing gets sent. I used: 230,1,TxFax(/tmp/test.tiff,caller)`
`17:06.32`	`tzafrir_laptop`	`Is that OK?`
`17:07.32`	`tzafrir_laptop`	`I can hear a short initial negotiation, and then the line goes quiet`
`17:07.46`	`tzafrir_laptop`	`until a time out`
`17:08.04`	`booray`	`Strom_C: light's green, etc, niu unit seems to see the card`
`17:08.12`	`tzafrir_laptop`	`on the debug log I see "HDLC up" and "HDLC down" constantly`
`17:08.25`	`Strom_C`	`booray: it's an ISDN PRI I'd assume?`
`17:08.30`	`booray`	`yes`
`17:08.45`	`booray`	`Strom_C: at least I think so, first t1 i've set up`
`17:09.09`	`Strom_C`	`well, according to the telco, what do you have?`
`17:09.11`	`booray`	`esf/b8zs from verizon`
`17:09.26`	`Strom_C`	`right, that's the framing and line coding`
`17:09.36`	`Strom_C`	`what else did they tell you?`
`17:10.16`	`booray`	`I have specific numbers on specific channels`
`17:10.24`	`booray`	`but that's pretty close to all the info they gave us`
`17:10.27`	`[TK]D-Fender`	`kodomo: Then don't "snip" your pastebin of sip.conf, you probably have things in the wrong places.`
`17:10.33`	`Strom_C`	`so channel 1 is...let's say (311) 555-2368?`
`17:10.42`	`Qwell_`	`311?`
`17:10.46`	`booray`	`sure`
`17:10.55`	`Strom_C`	`booray: ok, it's not an ISDN PRI circuit`
`17:10.56`	`Qwell_`	`Strom_C: you should know better than 311 :p`
`17:11.05`	`Strom_C`	`Qwell_: standard fake area code`
`17:11.09`	`Qwell_`	`O.o`
`17:11.13`	`booray`	`okay.`
`17:11.15`	`booray`	`awesome.`
`17:11.20`	`Qwell_`	`apparently I should know better than 311`
`17:11.20`	`matt_`	`lol`
`17:11.23`	`matt_`	`i'm gonna try to dial it :)`
`17:11.32`	`booray`	`so.. fxo loop start or something?`
`17:11.41`	`booray`	`the paperwork does say loop start`
`17:11.49`	`Qwell_`	`matt_: if it's used for that, it's gonna be reserved in NANP`
`17:11.51`	`Strom_C`	`booray: ugh, why why why did you order a circuit like that`
`17:11.55`	`Qwell_`	`it won't go anywhere`
`17:11.57`	`matt_`	`:O some fat guy answered`
`17:11.57`	`Strom_C`	`you should have ordered a PRI :)`
`17:12.12`	`*** join/#asterisk saftsack (n=saftsack@pD9E0685A.dip.t-dialin.net)`
`17:12.58`	`Strom_C`	`matt_: try configuring the individual channels in zaptel and zapata as FXS loopstart signaling`
`17:13.12`	`Strom_C`	`s/matt_/booray/`
`17:13.16`	`matt_`	`Strom_C, 0.o`
`17:13.28`	`booray`	`so, singalling = fxo_ls`
`17:13.32`	`Strom_C`	`no`
`17:13.33`	`booray`	`then channel => 1-23`
`17:13.35`	`Strom_C`	`FXS signaling`
`17:13.39`	`Strom_C`	`not FXO signaling`
`17:13.39`	`booray`	`oh`
`17:16.28`	`booray`	`"Signalling requested on channel 1 is FXS Loopstart but line is in PRI Signalling signalling"`
`17:16.49`	`Strom_C`	`booray: you also need to reconfigure zaptel.conf`
`17:17.22`	`booray`	`wow, so how is it different than the normal spam, bchan, and dchan lines?`
`17:17.30`	`booray`	`I haven't seen any other possibilities in the docs I've read`
`17:17.36`	`booray`	`Strom_C: thanks for the help btw`
`17:18.01`	`Strom_C`	`brb, phone call`
`17:18.32`	`booray`	`cool`
`17:19.33`	`*** join/#asterisk svenna_ (n=svenna@p548D0477.dip0.t-ipconnect.de)`
`17:20.13`	`*** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org)`
`17:20.18`	`booray`	`I think I got it anyway`
`17:20.20`	`booray`	`rebooting`
`17:20.28`	`Strom_C`	`you dont need to reboot`
`17:21.11`	`booray`	`Strom_C: yeah, but I just want to make sure, b/c this card was acting funny with module unloading and reloading`
`17:21.35`	`Strom_C`	`ok`
`17:21.38`	`Strom_C`	`back from phone call :)`
`17:21.50`	`Strom_C`	`which card are you using, by the way?`
`17:21.54`	`booray`	`everything I see online indicates that I should be using fxo`
`17:22.06`	`booray`	`wildcard te120p i think? I can check to be sure`
`17:22.09`	`booray`	`I know its a 120`
`17:22.12`	`Strom_C`	`no, you want FXS signaling if the telco is giving you circuits that act as analog lines`
`17:22.15`	`Strom_C`	`yeah, TE120`
`17:22.19`	`Strom_C`	`good choice :)`
`17:22.36`	`Strom_C`	`you want FXS signaling because the asterisk box has to pretend to be station equipment`
`17:22.47`	`Strom_C`	`go off-hook, outpulse numbers, etc`
`17:23.25`	`booray`	`oaky`
`17:23.27`	`booray`	`okay*`
`17:25.05`	`booray`	`alright.. I have it all changed;`
`17:25.09`	`booray`	`so would my channel be zap/1 ?`
`17:25.45`	`Strom_M`	`ZAP/1 through ZAP/24, assuming they're giving you all 24 channels as individual trunks`
`17:25.49`	`booray`	`yes`
`17:25.53`	`booray`	`well, 1-6`
`17:26.10`	`Strom_M`	`then your definitions in zaptel and zapata.conf should only be for channels 1-6`
`17:26.22`	`*** part/#asterisk putnopvut (n=putnopvu@65.116.224.30)`
`17:27.27`	`booray`	`okay`
`17:27.38`	`booray`	`I'm getting a unable to create channel of type 'zap' cause 0, unknown`
`17:28.06`	`Strom_M`	`pastebin the output of "zap show channels"`
`17:28.25`	`booray`	`pastebin.ca ? what was the site again?`
`17:28.28`	`Strom_M`	`~pb`
`17:28.29`	`jbot`	`A Pastebin is a place to paste your stuff 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`
`17:28.42`	`Strom_M`	`pick one :)`
`17:29.12`	`*** join/#asterisk Ciber311 (n=Ciber311@user-1087eds.cable.mindspring.com)`
`17:29.16`	`booray`	`http://pastebin.ca/716245`
`17:29.36`	`booray`	`i included some lines before`
`17:29.37`	`*** join/#asterisk asdx (n=foo@adsl-153-45.click.com.py)`
`17:30.45`	`booray`	`http://pastebin.ca/716247 for 'zap show channel 1'`
`17:30.49`	`*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)`
`17:31.14`	`Strom_M`	`ah`
`17:31.17`	`Strom_M`	`your dial string is incorrect`
`17:31.28`	`Strom_M`	`you need to specify ZAP/1/13115552368`
`17:31.46`	`Strom_M`	`channel type / channel address / number to dial`
`17:31.46`	`booray`	`that simple huh`
`17:31.50`	`booray`	`dialplan reloading`
`17:32.18`	`booray`	`same error, just says zap/1/number now`
`17:32.51`	`Strom_M`	`booray: are you in los angeles? :)`
`17:32.54`	`booray`	`yes`
`17:33.02`	`Strom_M`	`ah cool`
`17:33.05`	`booray`	`853-1212 was cut off on sept 19 however`
`17:33.06`	`Strom_M`	`so am I`
`17:33.09`	`Strom_M`	`yes, i know`
`17:33.11`	`booray`	`verizon is no longer serving time`
`17:33.12`	`booray`	`bastards`
`17:33.16`	`Strom_M`	`uh, that's AT&T`
`17:33.21`	`booray`	`well whoever`
`17:33.27`	`Strom_M`	`where in los angeles are you?`
`17:33.31`	`booray`	`san dimas`
`17:33.33`	`Strom_M`	`west LA or covina somewhere`
`17:33.34`	`Strom_M`	`ah ok`
`17:33.51`	`booray`	`it's verizon flexgrow`
`17:33.59`	`*** join/#asterisk gorauskas (n=lgorausk@66-224-20-131.atgi.net)`
`17:34.47`	`booray`	`hmm.. any other ideas of stuff I can look at?`
`17:34.47`	`Strom_M`	`what happens when you try to call the DN associated with channel 1?`
`17:34.55`	`booray`	`trying now`
`17:35.23`	`booray`	`calling from my cell phone, just ringing`
`17:35.28`	`booray`	`nothing at the CLI`
`17:35.31`	`Strom_M`	`any output on the console`
`17:35.32`	`Strom_M`	`?`
`17:35.33`	`Strom_M`	`hmm`
`17:35.39`	`Strom_M`	`what's your verbosity set at?`
`17:35.41`	`*** part/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`17:35.48`	`booray`	`I think 3`
`17:36.01`	`booray`	`yes it is`
`17:36.49`	`Strom_M`	`ive not worked with this type of signaling on a T1 before...though if I were sitting in front of the console, i could probably bang out a solution relatively quickly`
`17:37.18`	`kodomo`	`[TK]D-Fender: http://pastebin.com/deb87765`
`17:37.27`	`kodomo`	`(sorry - interrupt by work ;) )`
`17:38.09`	`[TK]D-Fender`	`kodomo: permanently remove all commented out lines and repastebin.`
`17:39.25`	`outtolunc`	`strom/booray, what is the issue (sorry wasn't watching)`
`17:39.52`	`booray`	`verizon t1, loop start apparently, but I had it configured as pri`
`17:40.22`	`booray`	`but even though I've got that fixed, I'm getting a unable to create channel of type zap error when attempting to dial out`
`17:40.51`	`booray`	`and the psuedo numbers they gave me don't ring through yet`
`17:41.14`	`*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)`
`17:41.37`	`outtolunc`	`grr`
`17:41.37`	`booray`	`outtolunc: wb`
`17:41.39`	`booray`	`hehe`
`17:41.50`	`outtolunc`	`ty`
`17:42.11`	`[TK]D-Fender`	`booray : pastebin your zaptel.conf , zapata.conf , and extensions.conf`
`17:42.54`	`booray`	`just had a thought`
`17:42.55`	`outtolunc`	`and what type of line it is`
`17:43.12`	`Jason99`	`Hello, I have a queue setup with a number of phones in it. We are running into problems when someone forwards their phone to an external number. Is there a way to prevent a phone that has been forwarded from forwarding the queued call?`
`17:43.25`	`booray`	`since I thought i was in PRI mode, I commented out the wcfxo/wcfxs modules out of /etc/default -- I imagine those would need to be enabled now?`
`17:43.46`	`booray`	`[TK]D-Fender: I'll do that in just a second here`
`17:43.51`	`[TK]D-Fender`	`Jason99: No.`
`17:44.00`	`[TK]D-Fender`	`booray: No.`
`17:44.17`	`[TK]D-Fender`	`booray: Don't think so much, and just pastebin your configs.`
`17:44.36`	`Jason99`	`[TK]D-Fender: Ok, thanks`
`17:45.03`	`booray`	`okie. going to copy/paste now`
`17:45.15`	`[TK]D-Fender`	`~pb`
`17:45.15`	`jbot`	`A Pastebin is a place to paste your stuff 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`
`17:45.16`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^`
`17:45.18`	`[TK]D-Fender`	`PASTEBIN`
`17:46.53`	`*** part/#asterisk gorauskas (n=lgorausk@66-224-20-131.atgi.net)`
`17:47.08`	`*** join/#asterisk Strom_M (n=strom@208.127.172.112)`
`17:47.15`	`*** join/#asterisk Strom_C (n=strom@208.127.172.112)`
`17:47.20`	`kodomo`	`[TK]D-Fender: http://pastebin.com/d5aa30154`
`17:47.24`	`Norm`	`anyone have a contact at U Penn regarding their SER/Asterisk deployment?`
`17:47.25`	`Strom_C`	`booray: still there?`
`17:47.35`	`booray`	`yes`
`17:47.40`	`booray`	`copying and pasting configs`
`17:47.47`	`Strom_C`	`alright - sorry, something on my DSL provider's end took a shit`
`17:47.58`	`Strom_C`	`don't mung things up too badly; you're close`
`17:48.07`	`[TK]D-Fender`	`kodomo: So your * server has a public IP ?`
`17:48.32`	`kodomo`	`[TK]D-Fender: yep`
`17:48.53`	`booray`	`[TK]D-Fender: http://pastebin.ca/716260 minus extensions.conf. I'd rather only paste sections as necessary`
`17:49.23`	`[TK]D-Fender`	`kodomo: remove [authentication] (the line), and then enable SIP debug and then do "sip reload" and try and trap the reg attempt.`
`17:50.25`	`[TK]D-Fender`	`booray: span=1,1,0,esf,b8zs <- should be`
`17:50.39`	`[TK]D-Fender`	`booray: And please pastebin the CLI output of a failed call attempt at verbose 10`
`17:50.39`	`booray`	`okay, so use their timing`
`17:50.44`	`booray`	`ok`
`17:50.45`	`[TK]D-Fender`	`booray: yes`
`17:50.46`	`booray`	`brb`
`17:51.24`	`*** part/#asterisk j00bar (n=jag@fsf/staff/j00bar)`
`17:52.33`	`*** join/#asterisk jcanfield (n=jcanfiel@68.109.242.162)`
`17:55.08`	`booray`	`Now, when I dial, it seems to call successfully and says 'Zap/1-1 answered SIP-3051-081fc870'`
`17:55.17`	`Strom_C`	`booray: awesome`
`17:55.21`	`booray`	`so that means I'm connecting to the telco properly?`
`17:55.25`	`Strom_C`	`yeah`
`17:55.33`	`booray`	`the number I called isn't ringing yet. :)`
`17:55.37`	`Strom_C`	`try and call 310-454-9999 and see if you get a recording`
`17:56.35`	`kodomo`	`[TK]D-Fender: hooray :D (I'm suddenly confused why this ever worked... but it did...`
`17:56.57`	`[TK]D-Fender`	`kodomo: You're welcome`
`17:56.58`	`kodomo`	`[TK]D-Fender: probably, I've commented something out in that section :-/`
`17:57.06`	`kodomo`	`[TK]D-Fender: thanks a lot :)`
`17:57.12`	`booray`	`Strom_C: zap/1-1 answered but no audio is passing atm. I'm on the phone w/verizon right now anyway so I think I have my end up and working correctly`
`17:57.27`	`*** join/#asterisk denon (n=denon@208.122.43.201)`
`17:57.27`	`*** mode/#asterisk [+o denon] by ChanServ`
`17:57.33`	`Strom_C`	`booray: maybe, maybe not`
`17:57.34`	`kodomo`	`lesson: don't let comments garble your configuration file :P`
`17:59.34`	`mvanbaak`	`that's why I let a script generate my configfile`
`17:59.57`	`*** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`18:02.39`	`lirakis`	`oy .. sigh`
`18:08.17`	`*** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net)`
`18:08.37`	`*** join/#asterisk [hC] (n=hardcore@65.116.224.30)`
`18:08.48`	`Qwell_`	`[hC]: You're here?`
`18:09.07`	`mvanbaak`	`Qwell_: try /whois [hC]`
`18:09.08`	`mvanbaak`	`;)`
`18:09.14`	`Qwell_`	`I saw the 65.`
`18:09.22`	`Qwell_`	`that's the crappy T1 we're on :D`
`18:09.34`	`mvanbaak`	`ah`
`18:09.42`	`mvanbaak`	`Qwell_: you @astricon as well ?`
`18:09.45`	`Qwell_`	`the ENTRE resort shares a T1`
`18:09.46`	`Qwell_`	`yeah`
`18:09.53`	`[hC]`	`I am indeed`
`18:09.58`	`[hC]`	`a little hung over this morning`
`18:09.59`	`Qwell_`	`including the hotel rooms, and the villas, and the actual hotel...`
`18:09.59`	`[hC]`	`:)`
`18:10.07`	`mvanbaak`	`hhmm, T1 == 24x64kbps right ?`
`18:10.10`	`Qwell_`	`yeah`
`18:10.13`	`Qwell_`	`1.5mbit`
`18:10.18`	`mvanbaak`	`damn`
`18:10.21`	`Qwell_`	`yeah`
`18:10.27`	`[hC]`	`know how they get wireless out to the villas? ive been looking for the aps and havent found em yet`
`18:10.27`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`18:10.33`	`mvanbaak`	`even a public toilet has more bandwidth here in .nl`
`18:10.37`	`[hC]`	`er, not how they get it out there, but where the ap's are`
`18:10.45`	`Qwell_`	`[hC]: yeah, same as here, it's just CAREFREE_RESORT`
`18:11.26`	`[hC]`	`Qwell: yup.. ive been trying to find where the phsical ap is sitting.. it must be outside cause my signal is full out there`
`18:11.32`	`Qwell_`	`[hC]: there are a ton`
`18:11.55`	`Qwell_`	`I can see about 5 from my villa`
`18:12.01`	`mvanbaak`	`when devcon in .nl is a fact I will have at least 100mbit`
`18:12.09`	`Qwell_`	`mvanbaak: yay`
`18:12.22`	`Qwell_`	`astricon rarely has good bandwidth`
`18:12.40`	`*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`18:12.43`	`mvanbaak`	`I'm trying to get the university in amsterdam to give us access to one of their locations`
`18:12.48`	`mvanbaak`	`they have 1.5gbit`
`18:13.08`	`mvanbaak`	`and it's close to the airport schiphol`
`18:14.24`	`GreggB`	`I'm setting up a blacklist for tele-marketers and other harrasing callers - does anyone know of a good source for annoying recordings which could be played back at these folks?`
`18:14.56`	`mvanbaak`	`record a barfing person`
`18:15.46`	`[TK]D-Fender`	`GreggB: tt-weasels`
`18:15.46`	`mvanbaak`	`but I think there's an application in asterisk that produces a very annoying tone`
`18:16.07`	`GreggB`	`lol - the barfing is just the route I'm looking for`
`18:16.10`	`davevg-btwtech`	`echo() :)`
`18:16.25`	`Corydon76-dig`	`[TK]D-Fender: I had hospital staff walking around for days talking about the weasels`
`18:16.33`	`mvanbaak`	`lol Corydon76-dig`
`18:17.23`	`Corydon76-dig`	`It wasn't working beforehand, and I couldn't figure it out, so I left it in debugging mode... and it started working while I was gone...`
`18:17.50`	`mvanbaak`	`hahahahaha`
`18:18.05`	`mvanbaak`	`dont you love it when that happens ?`
`18:18.21`	`mvanbaak`	`I had that with my MoH`
`18:18.23`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`18:18.36`	`mvanbaak`	`as a test I uploaded Cradle of Filth`
`18:18.38`	`*** join/#asterisk drutlandxpt (n=drutland@px1.xfoneusa.com)`
`18:18.42`	`*** join/#asterisk Ciber311 (n=Ciber311@user-1087eds.cable.mindspring.com)`
`18:18.54`	`mvanbaak`	`nothing happened, so I told the customer I would get back to them a week later to fix it`
`18:19.05`	`mvanbaak`	`during the weekend the server rebooted and stuff started to work`
`18:19.16`	`mvanbaak`	`now they had a very nice tune as moh`
`18:19.31`	`mvanbaak`	`they were not happy, can tell you that`
`18:20.57`	`Corydon76-dig`	`I had lawyers who didn't like the default moh... so I uploaded Weird Al... they were not amused`
`18:21.09`	`mvanbaak`	`whehehehe`
`18:22.40`	`mvanbaak`	`weird al ++`
`18:23.02`	`Corydon76-dig`	`What's worse is that they replaced it with <gag> soft jazz`
`18:23.05`	`sevard`	`Corydon76-dig: the default * MoH?`
`18:23.12`	`Corydon76-dig`	`sevard: correct`
`18:23.23`	`mvanbaak`	`food is here, latero`
`18:26.02`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`18:26.14`	`sevard`	`Corydon76-dig: how can you not like that`
`18:26.41`	`sevard`	`Corydon76-dig: there's a whole directory somewhere listed on voip-info.org with royality free classical music. It's something like 40 MP3s`
`18:27.18`	`_x86_`	`i found some california-style music free by googling for "free MoH"`
`18:28.49`	`drutlandxpt`	`this may be a sore subject, but can anyone help me with mgcp dialing out? I can dial into it just fine, but it comes back with errors dialing out`
`18:30.00`	`luke-jr`	`Corydon76-dig: breaking copyright law right in front of lawyers in attempt to annoy them? bad idea imho`
`18:30.02`	`Corydon76-dig`	`sevard: I never said I had a problem with it`
`18:30.18`	`Strom_C`	`_x86_: what is "california style music"`
`18:30.21`	`Corydon76-dig`	`luke-jr: it gets worse. I'm in Music City USA`
`18:31.11`	`Corydon76-dig`	`luke-jr: you think these lawyers licensed their current MoH?`
`18:35.07`	`*** join/#asterisk drwelby (n=mpfister@mail.enplan.com)`
`18:35.08`	`jameswf`	`does asterisk have a place to set the Time Zone Offset`
`18:35.45`	`Corydon76-dig`	`We haven't used timezone offsets for years`
`18:35.57`	`Corydon76-dig`	`Timezones are considerably more complex`
`18:36.15`	`GreggB`	`sevard: Looks like the stuff is here now: http://www.signate.com/products/?page=moh (linked from: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold) Thanks for pointing that out, now I've got good hold music too :-)`
`18:36.50`	`Deeewayne`	`Corydon76-dig: are we going to add Hugo Chavez's new timezone ?`
`18:37.06`	`Corydon76-dig`	`Deeewayne: that would be up to the US govt`
`18:37.13`	`Deeewayne`	`chavez-time-zone`
`18:37.20`	`jameswf`	`I think asterisk is using utc but cdr (mysql) is using Unix time`
`18:37.52`	`Corydon76-dig`	`jameswf: that's the same thing`
`18:38.03`	`jameswf`	`*with the offset`
`18:38.05`	`Corydon76-dig`	`unixtime IS utc`
`18:38.33`	`Corydon76-dig`	`neither unixtime nor utc have an offset`
`18:38.35`	`*** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net)`
`18:38.44`	`Qwell_`	`GMT-0`
`18:38.58`	`*** join/#asterisk ta^3 (n=tacvbo@65.116.224.30)`
`18:39.12`	`drutlandxpt`	`does anyone have any infromation on mgcp?`
`18:39.17`	`jameswf`	`explained: CDR Time stamp now asterisk recording timestmp = now+5 hours`
`18:40.35`	`jameswf`	`utc gmt ... da** achronyms`
`18:42.39`	`Deeewayne`	`drutlandxpt: I've had enough information to successfully avoid mgcp so far....`
`18:44.28`	`drutlandxpt`	`Deeewayne: that is better than nothing. but I pretty much don't have a choice`
`18:45.03`	`*** join/#asterisk anonymouz666 (n=anonymou@201.19.162.243)`
`18:45.07`	`*** join/#asterisk cnet2 (n=nada@196.40.38.130)`
`18:45.12`	`cnet2`	`hi guys..`
`18:45.16`	`Deeewayne`	`sorry I can't help`
`18:45.39`	`cnet2`	`Ihave a tdm card but here (CostaRica) on pstn lines when you hangup, there's no busy tone`
`18:45.48`	`drutlandxpt`	`thanks`
`18:46.03`	`cnet2`	`so, asterisk doesn't do the hangup detection`
`18:46.28`	`cnet2`	`I found some CostaRica specific vars in the dsp.c, but don't know if that has to manually be set,.. anyone know?`
`18:48.30`	`cnet2`	`could it be the fxsks?`
`18:48.32`	`cnet2`	`that?`
`18:48.55`	`*** join/#asterisk [hC] (n=hardcore@65.116.224.30)`
`18:51.41`	`cnet2`	`...anyone?.. any clue`
`18:55.43`	`*** join/#asterisk tsurko (n=tsurko@78.90.100.214)`
`18:56.44`	`*** join/#asterisk YakoS (n=source@130.89.171.8)`
`18:56.47`	`YakoS`	`hello all`
`18:57.23`	`YakoS`	`I'm having a small problem`
`18:57.49`	`scooby2`	`Anyone know which field the auto directory gets lastnames from?`
`18:57.56`	`Strom_C`	`is it smaller than a breadbox?`
`18:58.02`	`Strom_C`	`is it larger than an amoeba?`
`18:58.22`	`YakoS`	`I set the context of a phone that is connected to my asterisk server to 1370, then I add an 's'-extension to the 1370-context, but somehow it doesn't work, I jus get the dial tone like it is waiting for me to dial an extension`
`18:58.26`	`YakoS`	`anyone know why?`
`18:58.27`	`Strom_C`	`is it roughly, give or take an inch or two, the size of a tennis ball?`
`18:59.00`	`Corydon76-dig`	`Are you talking about tennis balls and garden hoses again?`
`18:59.03`	`YakoS`	`a tennis ball would do`
`18:59.15`	`YakoS`	`but, anyone know the solution to my problem?`
`18:59.25`	`Strom_C`	`YakoS: are you setting up your phone in ringdown mode?`
`18:59.39`	`scooby2`	`I always suggest B. if youhave to guess`
`18:59.50`	`YakoS`	`I feel stupid saying that I have no clue what that is.`
`18:59.59`	`sevard`	`Strom_C: do you know of any free reverse cellphone lookups, or... is all this $30 dollars a month! only 9.95! crap online freaking scams?`
`19:00.19`	`sevard`	`Strom_C: and what the heck happened to bellsmind? Wasn't there a very similar website to it?`
`19:00.24`	`Strom_C`	`YakoS: is the phone supposed to automatically connect somewhere (think the red phone to Moscow on Reagan's desk), or is the person picking up the phone supposed to dial a number?`
`19:00.42`	`YakoS`	`well, its a gsm to voip gateway`
`19:01.00`	`*** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com)`
`19:01.04`	`YakoS`	`so if somebody calls the gsm number, its supposed to forward the call to asterisk using SIP, and asterisk should start an extension`
`19:01.15`	`YakoS`	`like a voice menu`
`19:01.24`	`YakoS`	`but all I'm getting is a dial tone from asterisk`
`19:01.46`	`Strom_C`	`YakoS: may I gently suggest that perhaps you should read a book on Asterisk before going any further`
`19:01.50`	`*** part/#asterisk [hC]-Dayton (n=hardcore@65.116.224.30)`
`19:01.56`	`YakoS`	`well I have done that`
`19:02.26`	`YakoS`	`there's this guide that says if I add an 's' exten entry to a context, that extension should be started automatically`
`19:02.34`	`Strom_C`	`not exactly`
`19:02.53`	`Strom_C`	`'s' is only used in situations where the inbound call has no dialed number information associated with it`
`19:03.15`	`[TK]D-Fender`	`scooby2: the second.`
`19:03.22`	`[TK]D-Fender`	`scooby2: Where you fill in the NAME`
`19:04.02`	`YakoS`	`Strom_C, then is there a relatively simple way to automatically start a voice menu once a SIP call comes in?`
`19:04.20`	`mistermocha`	`hi`
`19:04.40`	`[TK]D-Fender`	`YakoS: and it assumes your sip.conf is setup right to accept the call, and that it passes it to the proper context in which you HAVE a matching exten for the number the call is looking for...`
`19:04.58`	`mistermocha`	`asterisk 1.2 - what variable assumes the incoming caller ID?`
`19:05.02`	`Strom_C`	`YakoS: yes there is. Does your inbound call have dialed number information associated with it?`
`19:05.11`	`Strom_C`	`mistermocha: ${CALLERID(num)}`
`19:05.17`	`Strom_C`	`mistermocha: it's a function, not a variable`
`19:05.21`	`[TK]D-Fender`	`mistermocha: CALLERID is a FUNCTION in 1.2+`
`19:05.34`	`Strom_C`	`there's also name, all, ANI, DNID, and so on in place of 'num'`
`19:05.50`	`*** join/#asterisk jmikeharvey (n=jmikehar@px1.xfoneusa.com)`
`19:05.52`	`mistermocha`	`ok, talking specifically about incoming caller id`
`19:06.24`	`hmmhesays`	`catfish noodlers on dirty jobs`
`19:06.35`	`hmmhesays`	`wow, red neck`
`19:06.46`	`jmikeharvey`	`I know you said that mgcp.conf was almost a dead topic, just a quick questions. I have DT and can receive calls but when I try and dial out I get a fast busy`
`19:07.05`	`Strom_C`	`jmikeharvey: it's called a reorder, not a "fast busy"`
`19:08.50`	`YakoS`	`Strom_C, I think not`
`19:09.09`	`[TK]D-Fender`	`jmikeharvey: enable MGCP DEBUG and verify your dialplan.`
`19:09.16`	`Strom_C`	`YakoS: try this - send the inbound call to a context containing only the following extension:`
`19:09.19`	`jmikeharvey`	`Thank you`
`19:09.31`	`[TK]D-Fender`	`jmikeharvey: And keep in mind the phone may have its own dialplan getting in the way...`
`19:09.36`	`Strom_C`	`exten => _.,1,NoOp(The DNIS is ${EXTEN})`
`19:09.48`	`Strom_C`	`YakoS: then pastebin the CLI output from that call`
`19:09.54`	`jmikeharvey`	`It is an analog phone connected to an Adtran TA624`
`19:09.57`	`*** part/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`19:10.02`	`jmikeharvey`	`So no other dial-plans`
`19:10.34`	`[TK]D-Fender`	`jmikeharvey: Oh yeah... ok, just pastebin your * dialplan and a failed call with MGCP debug enabled`
`19:10.35`	`YakoS`	`Strom_C, thanks, it will be a minute since my asterisk machine has no gui`
`19:10.36`	`[TK]D-Fender`	`~pb`
`19:10.37`	`jbot`	`A Pastebin is a place to paste your stuff 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`
`19:10.38`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^^`
`19:10.53`	`Strom_C`	`YakoS: the lack of gui should speed things up actually :)`
`19:11.05`	`Strom_C`	`YakoS: you should be ssh'd in from a workstation that has a gui :)`
`19:14.06`	`YakoS`	`Strom_C, http://pastebin.com/m5bb77315`
`19:14.34`	`Strom_C`	`ok, so you do have DNIS information`
`19:14.39`	`Strom_C`	`you're being sent "1370"`
`19:14.45`	`YakoS`	`ok`
`19:14.46`	`Strom_C`	`which means 's' isn't going to work.`
`19:14.49`	`YakoS`	`well`
`19:15.00`	`Strom_C`	`now, what you could do is something like this`
`19:15.04`	`YakoS`	`when i called the gateway, i pressed 1370 on my mobile phone`
`19:15.12`	`YakoS`	`otherwise it just gives me a dial tone`
`19:15.26`	`Strom_C`	`well then that's a gateway configuration problem, not an asterisk problem`
`19:15.37`	`Strom_C`	`unless you didn't give me full CLI output`
`19:15.42`	`*** join/#asterisk tuxd00d (n=tuxinato@128.187.124.183)`
`19:15.56`	`YakoS`	`well i did`
`19:16.03`	`YakoS`	`hmm then i have to look into that again`
`19:16.16`	`Strom_C`	`the entire call only generated those three lines of output?`
`19:16.27`	`Strom_C`	`i don't see anything about "accepting call" and so on`
`19:18.35`	`tristanbob`	`anyone here work for a university that deployed asterisk?`
`19:19.50`	`YakoS`	`Strom_C, here's some more`
`19:19.51`	`YakoS`	`http://pastebin.com/m77edb25f`
`19:20.43`	`Strom_C`	`YakoS: ok, so you see nothing on the console until after you've dialed "1370" at the dial tone?`
`19:21.40`	`Strom_C`	`my laptop has narcolepsy`
`19:23.34`	`Strom_C`	`YakoS: ping`
`19:24.00`	`Strom_C`	`YakoS: i'm asking you questions which should not, realistically, be taking 3+ minutes to answer`
`19:24.16`	`wishes`	`good mornin`
`19:24.34`	`De_Mon`	`what does this mean?`
`19:24.34`	`De_Mon`	`<PROTECTED>`
`19:24.47`	`De_Mon`	`It looks like the number was a fax line, does that sound right?`
`19:24.54`	`De_Mon`	`(t38 = fax?)`
`19:25.30`	`wishes`	`Strom_C: not everyone sits at the computer all day every day :)`
`19:25.30`	`YakoS`	`sorry Strom_C, i was figuring out some stuff`
`19:25.45`	`YakoS`	`my gateway was dialing the asterisk server directly, instead of using it as proxy`
`19:25.47`	`wishes`	`or rather, not everyone looks at the one console all day every day :D`
`19:26.03`	`YakoS`	`and no, nothing else`
`19:26.17`	`Strom_C`	`YakoS: ok, but if i'm actively helping you debug something, then a simple "hang on, let me try something" will suffice`
`19:27.00`	`YakoS`	`yeah, I'm sorry`
`19:27.10`	`YakoS`	`well i think it is working now`
`19:27.24`	`Strom_C`	`YakoS: cool`
`19:27.39`	`YakoS`	`i get a -- Executing [1370@1370:1] NoOp("SIP/1370-084256d0", "The DNIS is 1370") in new stack`
`19:27.43`	`YakoS`	`without dialing anything`
`19:27.59`	`*** join/#asterisk codefreeze (n=steve_mu@65.116.224.30)`
`19:27.59`	`*** mode/#asterisk [+o codefreeze] by ChanServ`
`19:28.42`	`[TK]D-Fender`	`YakoS: how do you "not dial anything" on a GSM phone?`
`19:29.00`	`dlynes`	`When you post a message to the asterisk-users mailing list, you're supposed to get your message echo'd back on the list, correct?`
`19:29.06`	`[TK]D-Fender`	`YakoS: AFAICT You have to dial a NUMBER on your phone.`
`19:29.17`	`*** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu)`
`19:29.41`	`YakoS`	`yes I dial the number of the GSM gateway`
`19:30.03`	`YakoS`	`but I used to have to dial an extension number after calling, but now i don't have to do that anymore`
`19:30.08`	`YakoS`	`which brings me a step further`
`19:31.16`	`YakoS`	`hey, it works, but only with "_.", not with "s"`
`19:31.34`	`YakoS`	`when I put "s" as exten entry, my call gets refused`
`19:32.08`	`Strom_C`	`what is _. indicating the DNIS is set to?`
`19:32.19`	`YakoS`	`first 1370 then h`
`19:32.37`	`Strom_C`	`the gateway is still sending "1370" then`
`19:32.45`	`YakoS`	`yes, but it should do that`
`19:32.48`	`Strom_C`	`booray: por que?`
`19:32.55`	`*** join/#asterisk guillote_GNU (n=bancaria@host35.201-253-17.telecom.net.ar)`
`19:33.06`	`YakoS`	`since i want it to trigger extension 1370 when i call it with my GSM`
`19:33.22`	`Strom_C`	`YakoS: then you need extension 1370, not extension s`
`19:33.25`	`Strom_C`	`do you follow?`
`19:33.25`	`YakoS`	`before it worked i used to just dial 1370 on my phone`
`19:33.54`	`YakoS`	`wait, I see`
`19:34.05`	`YakoS`	`so, _. is like a wildcard?`
`19:34.43`	`booray`	`Strom_C: I can't seem to get a hold of anyone who can watch the t1 switch to see if any of my call attempts are actually working. I believe I have it configured correctly now, with the right signalling, and calls "connect" (i.e. "Zap/5-1 answered SIP/3051-etcetc") but nothing rings... incoming calls aren't identified by the system, etc. And in all of this, Verizon can't even look directly at their switch to watch while I make calls`
`19:34.49`	`YakoS`	`that obnoxious asteriskNOW interface confuses me`
`19:34.59`	`Strom_C`	`YakoS: yes, sort of -- didn't you read that chapter of the book?`
`19:35.47`	`YakoS`	`Strom_C, I guess I didn't`
`19:36.30`	`Strom_C`	`booray: well let's see if we can troubleshoot this before you fly off the deep end with verizon`
`19:36.46`	`YakoS`	`I just ended up editing the config files trough the asteriskNOW interface`
`19:36.52`	`YakoS`	`because the interface itself sucks ass`
`19:37.16`	`Strom_C`	`YakoS: you should probably just go with asterisk and a linux distro of your choice rather than asterisknow`
`19:37.35`	`Strom_C`	`asterisknow is a decent software appliance, but for custom things like GSM gateways, plain ol' asterisk is better`
`19:38.51`	`*** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com)`
`19:38.57`	`YakoS`	`okay thanks for your time`
`19:40.38`	`[TK]D-Fender`	`YakoS: Well * seems fine. You just need to setup your dialplan to handle the extensions you want to let your phone call.`
`19:41.06`	`Strom_C`	`booray: are you still there?`
`19:41.17`	`mistermocha`	`back for more...`
`19:42.30`	`mistermocha`	`just to make sure I'm not crazy... if the ${NUMBER} field is empty, then does that mean the caller ID is not being sent by the carrier>`
`19:42.33`	`mistermocha`	`?`
`19:42.46`	`Strom_C`	`what's ${NUMBER}?`
`19:43.05`	`Strom_C`	`didn't I explicitly say ${CALLERID(num)}?`
`19:43.07`	`booray`	`Strom_C: sorry, took a quick phone call. I'm here`
`19:43.30`	`mistermocha`	`ok, good...`
`19:43.46`	`mistermocha`	`so if ${CALLERID(num)} is not set, then it's not being set by the carrier?`
`19:43.50`	`booray`	`Strom_C: I appreciate the help. using the fxs signalling as you suggested before has gotten me the furthest, as it actually connects my call with the zap channel, but it doesn't seem to get any further than that`
`19:44.01`	`Strom_C`	`booray: ok. the first thing I need you to do is to compile everything about the T1 you can find`
`19:44.10`	`mistermocha`	`sorry for the push-back`
`19:44.18`	`booray`	`Strom_C: I have all the e-mails here, some printed, some in a window.`
`19:44.35`	`Strom_C`	`I need to get a clear picture of what the T1 is set to before I can go stabbing in the dark at it with asterisk`
`19:44.42`	`Strom_C`	`mistermocha: what kind of carrier?`
`19:45.16`	`mistermocha`	`analog`
`19:45.50`	`Strom_C`	`mistermocha: on a zaptel card, i'd assume, right?`
`19:46.20`	`*** join/#asterisk shido6 (n=shido6@204.126.120.132)`
`19:46.44`	`Strom_C`	`mistermocha: do you have usecallerid=yes in your zapata.conf /before/ the channel=> line?`
`19:46.53`	`*** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com)`
`19:47.06`	`mistermocha`	`Strom_C: Sangoma`
`19:47.11`	`mistermocha`	`lemme check`
`19:47.47`	`mistermocha`	`Strom_C: yep`
`19:48.10`	`Strom_C`	`ok - is asterisk waiting till the second ring cycle to answer the channel?`
`19:49.55`	`Seldon75`	`hi, we have just switched out our Digium TDM2400 for a Sangoma`
`19:50.24`	`mistermocha`	`aha.... that may be it`
`19:50.41`	`mistermocha`	`I forget... how do I set that?`
`19:50.54`	`Seldon75`	`calls made from our SIP phones work fine, however the analog phones that go into the Sangoma arent able to make calls`
`19:51.15`	`Strom_C`	`mistermocha: if you're using a digium card, i know that the usecallerid=yes setting forces the card to wait a full ring cycle before the call goes into asterisk`
`19:51.51`	`Strom_C`	`mistermocha: but i'm not familiar with the particular method sangoma uses to completely fuck zaptel limp, so i don't know whether the same applies with sangoma cards.`
`19:52.15`	`mistermocha`	`ok`
`19:52.23`	`Seldon75`	`is there more configuration I need to do other than specifying those channels as fxo_ks in zapata.conf?`
`19:52.30`	`mistermocha`	`I'll dig thru the sangoma documentation and see if I can find out more`
`19:52.41`	`dlynes`	`Seldon75: pastebin the following: wanrouter hwprobe, wanrouter status`
`19:52.51`	`Seldon75`	`ok`
`19:52.53`	`Seldon75`	`1 sec`
`19:53.11`	`dlynes`	`Seldon75: and 'wanrouter messages \| grep wanpipe'`
`19:56.03`	`*** join/#asterisk hyperactivecrond (n=chris@unaffiliated/hyperactivecrond)`
`19:56.34`	`hyperactivecrond`	`i have asterisk 1.4. i had music on hold through a shoutcast server working last night, then i edited the config file for musiconhold.`
`19:56.47`	`hyperactivecrond`	`now it doesn't work, even if the syntax appears to still be valid.`
`19:56.54`	`hyperactivecrond`	`i'm pastebinning my relevant config files`
`19:57.16`	`dlynes`	`hyperactivecrond: did you try putting back your old config file?`
`19:57.32`	`hyperactivecrond`	`dlynes: dummy me didn't make a backup one.`
`19:57.46`	`hyperactivecrond`	`what happens is that asterisk spews out this error while putting music on hold:`
`19:58.24`	`Katty`	`meesha.`
`19:58.34`	`hyperactivecrond`	`started music on hold, class 'default', on channel'SIP/200-<snip>'`
`19:58.35`	`dlynes`	`musha.`
`19:58.37`	`Katty`	`is there anyone here based out of las vegas?`
`19:58.59`	`hyperactivecrond`	`--stopped music on hold on SIP/200<snip>`
`19:59.03`	`hyperactivecrond`	`right afterwards.`
`19:59.11`	`dlynes`	`heh`
`19:59.16`	`Sauron	`Katty: No.`
`19:59.20`	`Katty`	`Sauron: :<
`19:59.25`	`rob0`	`I am based WAY out of Las Vegas. Several thousand kilometers outside, in fact.`
`19:59.28`	`Katty`	`sad panda.`
`19:59.30`	`Sauron	`Is Austin close enough?`
`19:59.35`	`Katty`	`no, no it's not.`
`19:59.36`	`Sauron	`Bring Jaeger :p`
`19:59.39`	`dlynes`	`How about Canada?`
`19:59.39`	`Sauron	`:(`
`19:59.48`	`Katty`	`no, no i'm sure i need las vegas.`
`19:59.58`	`rob0`	`How about HuntsVegas?`
`19:59.59`	`Katty`	`i skeered to move to a city with no geeky friends :<`
`19:59.59`	`Sauron	`For what?`
`20:00.16`	`Katty`	`yep, there's the inevitable why.`
`20:00.23`	`dlynes`	`hahaha`
`20:00.28`	`hyperactivecrond`	`http://pastebin.com/d44c816e0`
`20:00.32`	`Katty`	`foiled again >.<`
`20:00.40`	`rob0`	`Katty should move to HSV and work for Digium.`
`20:00.42`	`Corydon76-dig`	`Huntsvegas is much nicer`
`20:00.43`	`hyperactivecrond`	`Why?`
`20:01.01`	`Katty`	`i don't want to work for digium.`
`20:01.04`	`Seldon75`	`dlynes: http://www.pastebin.ca/716406`
`20:01.22`	`rob0`	`Okay, we'll have them put you in charge.`
`20:01.34`	`Katty`	`i don't want to be in charge of digium either.`
`20:01.39`	`rob0`	`In Soviet Russia, Digium work for YOU`
`20:01.44`	`Katty`	`hee`
`20:01.46`	`hyperactivecrond`	`well obviously i need a -s in my mpg123 line`
`20:01.51`	`dlynes`	`Seldon75: you don't have it configured, by the looks of things`
`20:01.56`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`20:02.02`	`hyperactivecrond`	`but still i don't even get it playing on the machine`
`20:02.03`	`Katty`	`i'm skeered of getting a new job.`
`20:02.05`	`Katty`	`i really REALLY am.`
`20:02.08`	`dlynes`	`Seldon75: did you run wancfg?`
`20:02.14`	`Katty`	`i know my stuffs tho >.<`
`20:02.42`	`Seldon75`	`dlynes: we ran wancfg_zaptel`
`20:02.44`	`Sauron	`You can have mine.`
`20:02.56`	`hyperactivecrond`	`D'OH i think i've got it`
`20:03.00`	`Sauron	`*nix admin and head asshole`
`20:03.32`	`Katty`	`i don't want to be head admin.`
`20:03.37`	`hyperactivecrond`	`got it!`
`20:03.45`	`hyperactivecrond`	`i added an extra dash in the stream 'directory'`
`20:04.12`	`Seldon75`	`dlynes: we got the instructions from the Sangoma wiki`
`20:04.14`	`hyperactivecrond`	`so we're good.`
`20:04.25`	`hyperactivecrond`	`now, to find a setting that actually makes it sound decent :\`
`20:04.36`	`Katty`	`hyperactivecrond: two tin cans with string!`
`20:04.54`	`hyperactivecrond`	`it sounds terrible and slow (PIV 2.0GHZ proc, 512MB of ram, PCLinuxOS2007, kde running)`
`20:04.56`	`YakoS`	`Strom_C, does regular asterisk offer queues and agents?`
`20:05.15`	`Katty`	`hyperactivecrond: it might be KDE`
`20:05.18`	`Strom_C`	`YakoS: yes`
`20:05.23`	`YakoS`	`ok thanks`
`20:05.30`	`Seldon75`	`dlynes: we ran with http://wiki.sangoma.com/wanpipe-linux-asterisk-install`
`20:05.40`	`Katty`	`hyperactivecrond: kde is purrty, just kinda big and bulky...`
`20:05.41`	`hyperactivecrond`	`lemme see if i can change the bitrate and if that'd help`
`20:05.48`	`hyperactivecrond`	`yeah i know katty :)`
`20:05.54`	`Katty`	`i'm sure you do (=`
`20:06.09`	`Seldon75`	`dlynes: what indicates to you that it's not configured`
`20:06.20`	`deeperror`	`I've got $50 for anyone that can find a solution to a 407 proxy authentication failure`
`20:06.30`	`shido6`	`sure`
`20:06.54`	`Katty`	`for 50 bucks i might get out of bed in the morning.`
`20:07.05`	`Katty`	`no offense ;)`
`20:07.18`	`Katty`	`i'm sure someone will jump at that to help you out.`
`20:07.21`	`Katty`	`just not me`
`20:07.25`	`hyperactivecrond`	`deeperror: the username / password are wrong?`
`20:07.30`	`hyperactivecrond`	`that's A solution`
`20:07.33`	`hyperactivecrond`	`so pay me.`
`20:07.35`	`hyperactivecrond`	`:p`
`20:07.37`	`Katty`	`hyperactivecrond: be nice :P`
`20:07.55`	`deeperror`	`solution = fix`
`20:08.06`	`hyperactivecrond`	`although, correct me if i'm wrong, that would bring up a 407 error, right?`
`20:08.38`	`Katty`	`deeperror: any way you can look at SIP debug?`
`20:08.46`	`Katty`	`not sure if that would help or not`
`20:08.53`	`Katty`	`but maybe you can see what the phone is actually /trying/ to do.`
`20:09.35`	`deeperror`	`sip client x-lite can make an outbound call 100%....my analog phones on zap channels don't send auth info along on the first dial attempt....but on the second dial attempt it works fine`
`20:09.56`	`Seldon75`	`hello`
`20:10.10`	`Seldon75`	`oh no, dlynes seems to have gone a.w.o.l.`
`20:10.15`	`Katty`	`deeperror: can you manually do a wait?`
`20:10.47`	`Katty`	`deeperror: 1,Dial(ww12093810293)`
`20:10.57`	`Katty`	`or whatever.`
`20:11.41`	`Sauron	`j/k`
`20:11.59`	`Sauron	`:(`
`20:12.01`	`Katty`	`:>`
`20:12.03`	`Katty`	`sorry.`
`20:12.05`	`Katty`	`hehehehe.`
`20:12.08`	`Sauron	`:`
`20:12.10`	`Sauron	`)`
`20:12.23`	`Katty`	`i dunno if i have iax forwarded through the firewall anymore or not`
`20:12.25`	`*** join/#asterisk flenders (n=fserto@unaffiliated/flenders)`
`20:12.37`	`Katty`	`file: can you still call me?`
`20:13.26`	`dlynes`	`Seldon75: nope...was just on a call`
`20:14.44`	`dlynes`	`Seldon75: normally wanrouter messages \| grep wanpipe will show you a number of log entries stating what modules are in the sangoma card, whether you've got hardware echo can installed, ...`
`20:14.58`	`hyperactivecrond`	`omgwtfbbq google's not working for once!`
`20:15.01`	`dlynes`	`Seldon75: gimme a few secs, and i'll pastebin you a sample`
`20:15.05`	`hyperactivecrond`	`oops never mind`
`20:15.36`	`dlynes`	`hyperactivecrond: omgwtfbbq's coming up just fine on google...wtfayta?`
`20:15.58`	`hyperactivecrond`	`heh`
`20:16.01`	`Katty`	`WTBomgwtfbbqplz.`
`20:16.17`	`Seldon75`	`dlynes: ok cool.. thx`
`20:16.17`	`hyperactivecrond`	`it was giving me 500 errors`
`20:16.20`	`Katty`	`will pay 50s.`
`20:16.29`	`Seldon75`	`500 errors! thats a lot of errors`
`20:16.33`	`Katty`	`haha`
`20:16.51`	`Sauron	`Time to go home.`
`20:16.53`	`Sauron	`Tarra!`
`20:17.00`	`Katty`	`buhbye`
`20:17.01`	`Seldon75`	`ciao`
`20:17.24`	`Sauron	`Yassas`
`20:18.35`	`hyperactivecrond`	`later`
`20:18.37`	`dlynes`	`Seldon75: http://pastebin.ca/716445`
`20:18.47`	`Katty`	`anyone worked with Rhino stuff?`
`20:19.06`	`dlynes`	`Seldon75: the last two lines mentioning 'open' are where asterisk has started using the zaptel channel through wanpipe`
`20:20.17`	`*** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)`
`20:20.18`	`Seldon75`	`thanks`
`20:20.41`	`dlynes`	`Seldon75: is wanrouter started?`
`20:21.03`	`Seldon75`	`it's status is 'connected'`
`20:21.06`	`Seldon75`	`does that mean yes?`
`20:21.13`	`dlynes`	`probably`
`20:21.32`	`Seldon75`	`so this output in the pastebin; that all came from running the 'wanpipe' command?`
`20:21.37`	`hyperactivecrond`	`from within a dialplan, how do you dial another extensions`
`20:21.41`	`hyperactivecrond`	`extension*`
`20:21.43`	`hyperactivecrond`	`for example:`
`20:21.51`	`dlynes`	`Seldon75: that all came from 'wanrouter start' and ztcfg -vvvvvvvvvvvvvvvvv`
`20:22.07`	`dlynes`	`Seldon75: erm actually....ztcfg isn't part of that`
`20:22.12`	`hyperactivecrond`	`dial an extension, play a message, than dial another extension`
`20:22.19`	`Katty`	`noob!`
`20:22.25`	`Seldon75`	`so I just issue 'wanrouter start'`
`20:22.27`	`Katty`	`i can has walkthrough, yes?`
`20:22.32`	`dlynes`	`Seldon75: correct`
`20:22.36`	`*** join/#asterisk \|omni\| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)`
`20:22.38`	`Seldon75`	`ok thx trying now`
`20:22.41`	`Katty`	`i must be the only gamer in this channel :/`
`20:22.42`	`dlynes`	`Seldon75: but make sure you issue that command when asterisk isn't running`
`20:22.47`	`dlynes`	`Katty: must be`
`20:22.47`	`Katty`	`<PROTECTED>`
`20:22.57`	`dlynes`	`Katty: only game i play is mud`
`20:23.01`	`hyperactivecrond`	`Katty: i'm in your PBX, eating your channelz`
`20:23.03`	`Katty`	`oooh mud is fun :>`
`20:23.08`	`Katty`	`hyperactivecrond: >.<`
`20:23.13`	`Katty`	`hyperactivecrond: /kill`
`20:24.14`	`dlynes`	`Mervyn's doesn't exist on the west coast anymore?`
`20:24.41`	`Seldon75`	`aha`
`20:24.48`	`dlynes`	`Seldon75: ?`
`20:25.12`	`dlynes`	`Seldon75: keep in mind you have to run ztcfg -vvvvvvvvvvvvvvvv after running wanrouter start, and then run asterisk`
`20:25.21`	`Katty`	`oh.`
`20:25.23`	`Katty`	`that reminds me.`
`20:25.29`	`dlynes`	`Seldon75: if you don't do the ztcfg -vvvvvvvvvvvvvvv, asterisk won't see the sangoma card`
`20:25.33`	`Katty`	`i can't get asterisk to launch on boot up anymore.`
`20:25.38`	`dlynes`	`cool`
`20:25.41`	`Katty`	`and it makes me all sad inside.`
`20:25.44`	`Katty`	`i should go look at it ;<`
`20:25.55`	`dlynes`	`Katty: that means you get to charge service calls every time their power goes out`
`20:26.06`	`Katty`	`no`
`20:26.09`	`dlynes`	`hehehehe`
`20:26.15`	`Katty`	`that means i have to go into the freeze cold SERVER ROOM everytime the power goes out`
`20:26.27`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`20:26.39`	`Seldon75`	`ok`
`20:26.58`	`dlynes`	`Seldon75: so have you figured out what the issue was, then?`
`20:27.51`	`Katty`	`will someone look at my rc.local for me?`
`20:27.53`	`*** part/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net)`
`20:28.04`	`Katty`	`maybe i'm staring right at my problem.`
`20:28.15`	`dlynes`	`Katty: pastebin it`
`20:28.24`	`*** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-230-158.dsl.irvnca.pacbell.net)`
`20:28.27`	`UnixDog`	`hey guys`
`20:28.44`	`UnixDog`	`whats the line for sip.conf to allow anon sip`
`20:29.30`	`Katty`	`http://pastebin.ca/716459 <- rc.local, asterisk doesn't come up on it's own anymore but everything else before it seems to do just fine`
`20:29.44`	`dlynes`	`Katty: you're using slackware?`
`20:29.51`	`Katty`	`debian.`
`20:30.01`	`dlynes`	`Katty: asterisk 1.4?`
`20:30.03`	`Katty`	`Linux aphrodite 2.6.18-4-686 #1 SMP Wed Apr 18 09:55:10 UTC 2007 i686 GNU/Linux`
`20:30.19`	`Katty`	`Asterisk 1.4.4 built by root @ aphrodite on a i686 running Linux on 2007-05-10 13:18:36 UTC`
`20:30.21`	`Seldon75`	`dlynes: my initial problem was that the analog phones had dialtone, but coulnt dial out. after running those sommands, they no longer have dialtone`
`20:31.52`	`*** join/#asterisk saftsack (n=saftsack@pD9E0685A.dip.t-dialin.net)`
`20:33.01`	`Seldon75`	`thing is, if the Sangoma card wasn't configured, could my SIP phones still have made outgoing calls?`
`20:35.58`	`Seldon75`	`anyone?`
`20:36.05`	`Katty`	`oh`
`20:36.05`	`Katty`	`hi`
`20:36.10`	`Katty`	`the answer is 47`
`20:36.12`	`Katty`	`next!`
`20:36.13`	`Seldon75`	`hi ;)`
`20:36.14`	`Katty`	`;)`
`20:36.22`	`Katty`	`i don't htink you oculd`
`20:36.34`	`*** join/#asterisk [hC] (n=hardcore@65.116.224.30)`
`20:36.43`	`dlynes`	`Seldon75: they could, just not through analog...i.e. through voip`
`20:36.45`	`Katty`	`unless it was via something else...`
`20:36.48`	`Katty`	`like iax`
`20:37.11`	`Katty`	`or Magic(tm)`
`20:37.12`	`dlynes`	`Katty: if you type 'make config' on asterisk, it'll install the startup files for you`
`20:37.23`	`Katty`	`^_-`
`20:37.25`	`Katty`	`i don't mean that`
`20:37.25`	`dlynes`	`Katty: and it does not install anything into /etc/asterisk for you`
`20:37.32`	`Katty`	`i mean the server doing asterisk -gvvvvvvvc for me`
`20:37.42`	`Katty`	`i have all the files silly`
`20:37.46`	`dlynes`	`Katty: yes, of course...it does a debian-style startup for you`
`20:37.47`	`*** part/#asterisk [hC] (n=hardcore@65.116.224.30)`
`20:37.56`	`Katty`	`whatt?`
`20:37.57`	`dlynes`	`Katty: and then you don't have to mess with rc.local`
`20:38.00`	`Katty`	`:<`
`20:38.05`	`shido6`	`you mean make config ?`
`20:38.11`	`Katty`	`make menu config?`
`20:38.15`	`dlynes`	`Unless of course you like messing with rc.local`
`20:38.20`	`Katty`	`let me look at my blog real quick`
`20:38.21`	`dlynes`	`Katty: no ... make config`
`20:38.25`	`Katty`	`i can't remember what i did`
`20:38.43`	`Katty`	`i think i did a make menu select`
`20:38.45`	`Katty`	`and then make samples.`
`20:38.49`	`dlynes`	`Katty: after the install is done, you can do a make config, and it'll install the startup files for you`
`20:38.50`	`Seldon75`	`dlynes: the SIP phones are able to make calls out onto analog trunks`
`20:38.57`	`dlynes`	`Seldon75: right now?`
`20:39.01`	`Seldon75`	`yes`
`20:39.07`	`Katty`	`dlynes: oh.`
`20:39.14`	`Seldon75`	`it's just the analog lines going into the Sangoma that dont seem to work`
`20:39.20`	`dlynes`	`Seldon75: can you do a pastebin of wanrouter messages \| grep wanpipe?`
`20:39.27`	`Katty`	`dlynes: so, ./configure, make, make install, make samples.. and make config is something different from the sample files for asterisk`
`20:39.34`	`Seldon75`	`even though they are specified as fxo_ks in zaptel.conf`
`20:39.43`	`dlynes`	`Seldon75: analog lines going into the sangoma? you mean analog extensions going into the sangoma?`
`20:39.51`	`Seldon75`	`dlynes: we got no messages from that command`
`20:39.58`	`Seldon75`	`yes`
`20:40.29`	`Qwell_`	`make config installs the init scripts`
`20:40.42`	`Katty`	`ah ha!`
`20:40.48`	`Katty`	`yes, yes i could use some of that.`
`20:42.21`	`*** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro)`
`20:44.44`	`Zefk`	`After any call that ends with disposition 4 or 2 (no answer or busy) an additional call record is added in the log with dst field=s. Is it possible to have only one record in the log?`
`20:45.20`	`Katty`	`Zefk: Hi!`
`20:45.31`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
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`20:47.09`	`hmmhesays`	`run to the hills! run for your life!`
`20:47.12`	`Zefk`	`Katty: Hi!`
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`20:49.45`	`Seldon75`	`so, if I followed the install procedude from http://wiki.sangoma.com/wanpipe-linux-asterisk-install and zaptel.conf has my analog extensions properly specified as fx0_ks - what else could be preventing them from making calls?`
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`20:51.57`	`dlynes`	`Seldon75: hrm`
`20:53.04`	`Katty`	`russellb: (=`
`20:53.37`	`dlynes`	`Seldon75: pastebin your zaptel.conf and zapata.conf files`
`20:53.51`	`dlynes`	`Seldon75: also, do you get anything if you type 'ztcfg -vvvvvvvvvvvvvvvvvvv'?`
`20:54.46`	`dlynes`	`Seldon75: also, did you install it as follows: ./Setup install --linux-vanilla?`
`20:55.39`	`dlynes`	`Seldon75: and did you run make and make install on the zaptel drivers before the wanpipe drivers? did you get any errors during the makes?`
`20:56.04`	`dlynes`	`Katty: so did you run make config, then?`
`20:56.22`	`Katty`	`oh`
`20:56.23`	`Katty`	`no`
`20:56.26`	`Katty`	`i'm fixing a laptop right now`
`20:58.20`	`UnixDog`	`ok what is the line I add to asterisk to allow anonymous sip calls`
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`20:58.29`	`UnixDog`	`I am not finding it in the wiki`
`20:58.30`	`Katty`	`:<`
`21:00.14`	`UnixDog`	`?`
`21:00.19`	`UnixDog`	`anyone?`
`21:01.04`	`russellb`	`UnixDog: allowguest`
`21:01.19`	`UnixDog`	`ok`
`21:03.14`	`Seldon75`	`if I run ztcfg -vvvvvvvvvvvvvvvv I get the same output as wanpipe start`
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`21:09.52`	`dlynes`	`Seldon75: you mean absolutely nothing?`
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`21:10.39`	`Seldon75`	`no, the confirmation about '12 channels configured'. the command I got nothing from was 'wanrouter messages'`
`21:11.02`	`Seldon75`	`http://www.pastebin.ca/716481 is my zaptel`
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`21:11.23`	`Seldon75`	`http://www.pastebin.ca/716482 is my zapata`
`21:11.40`	`Seldon75`	`should they have the FXO/FXS signalling the same, or reversed?`
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`21:13.03`	`dlynes`	`Seldon75: can you pastebin the ztcfg -vvvvvvvvvv output as well?`
`21:13.39`	`JoeDeveloper`	`I am using Lumenvox speech recognition, and I just discovered that if I do a SpeechBackground and then hangup, if the script then executes SpeechDeactivateGrammar, it gets an error and exits. Is there something different I should be doing on a hangup to deactivate a grammar?`
`21:14.12`	`Seldon75`	`ok, but while I do that - is it correct that the FXO/S channel speciification should be the same in both files`
`21:16.04`	`dlynes`	`Seldon75: correct`
`21:16.08`	`Seldon75`	`http://www.pastebin.ca/716488 is the ztcfg -vvvvvv output`
`21:17.17`	`deeperror`	`Seldon75: may have something to do with using the wrong configs in zapata try switching fxsks to fxoks and fxoks to fxsks`
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`21:18.48`	`Seldon75`	`really? all the way through both files?`
`21:19.04`	`dlynes`	`Seldon75: ummm...something's severely screwed up there`
`21:19.07`	`deeperror`	`what is channel 1 connected to?`
`21:19.17`	`dlynes`	`Seldon75: you're using an a200, right?`
`21:19.21`	`Seldon75`	`this is what happens when I use an analog phone, i pick up, it gives me dialtone, I type any number, I get busy signal immediately`
`21:19.24`	`Seldon75`	`a400`
`21:19.44`	`Seldon75`	`channel 1 is connected to an analog trunk line`
`21:19.57`	`dlynes`	`Seldon75: oh yeah...forgot a400 shows up as an a200 in hwprobe`
`21:20.31`	`Seldon75`	`so if channel 1 is hooked up to a copper trunk, it should be FXS, no?`
`21:21.14`	`Seldon75`	`using this configuration we are able to get i/c and o/g calls if we use the SIP phones`
`21:22.41`	`Seldon75`	`so should I really go reversing the signalling?`
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`21:29.35`	`Lucky7`	`how do i send Asterisk a command from a shell prompt?`
`21:29.39`	`Lucky7`	`without actually opening the CLI?`
`21:29.40`	`dlynes`	`Seldon75: signalling would be fxs, yes, and the module should be fxo`
`21:29.50`	`dlynes`	`Lucky7: asterisk -rx 'insert command here'`
`21:30.03`	`Lucky7`	`thanks dlynes`
`21:30.24`	`dlynes`	`Seldon75: do you have any fxs modules on that card?`
`21:30.51`	`dlynes`	`Seldon75: 4 fxo modules, and 2 fxs modules, right?`
`21:30.55`	`Lucky7`	`Annnd when you've do asterisk -rx 'zap show channels'`
`21:30.56`	`Lucky7`	`and you get`
`21:30.57`	`Lucky7`	`http://rafb.net/p/1qvz6216.html`
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`21:31.10`	`Lucky7`	`that, How do i hangup from-pstn-c9c2?`
`21:31.44`	`*** part/#asterisk Cresl1n (n=matt@65.116.224.30)`
`21:31.47`	`Lucky7`	`i dont know how people get into that`
`21:32.07`	`Lucky7`	`but it spams the system so fast, that it makes the full log files, generate about a gig of data in about 15 minutes`
`21:32.25`	`dlynes`	`Lucky7: asterisk -rx 'soft hangup Local/s@from-pstn-c9c2,2'`
`21:32.29`	`cnet2`	`hi guys.. is there a asterisk developer here?`
`21:32.33`	`Lucky7`	`thanks dlynes`
`21:32.36`	`dlynes`	`Lucky7: or killall safe_asterisk asterisk rasterisk`
`21:32.47`	`dlynes`	`cnet2: LOTS`
`21:32.52`	`cnet2`	`jeje ok..`
`21:33.04`	`Lucky7`	`dlynes > i have to do it witout killing the system :p`
`21:33.07`	`dlynes`	`cnet2: ask a question that'll get a usable response :)`
`21:33.16`	`dlynes`	`Lucky7: your spam is already killing the system`
`21:33.16`	`Lucky7`	`15 sales people who would get royally pissed if I dropped their calls`
`21:33.23`	`Lucky7`	`yea..`
`21:33.27`	`Lucky7`	`thanks thouh, that fixed it`
`21:33.32`	`dlynes`	`Lucky7: because you've got a buggy dialplan`
`21:33.54`	`cnet2`	`i have an understanding.. if you make a call to pstn, asterisk knows the called person hangs-up, if asterisk hears the busy signal (tone that we all know).. is this right?`
`21:34.38`	`dlynes`	`cnet2: no...asterisk knows they hung up, because the polarity on the tip/ring pins got reversed momentarily`
`21:34.55`	`*** join/#asterisk tsurko (n=tsurko@78.90.100.214)`
`21:35.16`	`dlynes`	`cnet2: assuming you have disconnect supervision on the line...otherwise, there's no way to tell that the other end hung up`
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`21:35.50`	`BockBilbo`	`hello`
`21:35.55`	`anders047`	`hello. is there an accepted method for setting up a hardphone with multiple lines via SIP in asterisk? Ie: does each line get its own extension or is there a built-in way of handling this situation?`
`21:36.08`	`cnet2`	`so .. if asterisk is gateway between pstns.., and there is no supervision.. it will hang.`
`21:36.10`	`Strom_C`	`depends on the phone`
`21:36.15`	`anders047`	`polycom 320`
`21:36.28`	`UnixDog`	`ok Ineed help on this`
`21:36.28`	`Strom_C`	`do you want the line appearances individually addressable?`
`21:36.33`	`UnixDog`	`http://pastebin.ca/716499`
`21:36.38`	`Strom_C`	`if so, give them separate sip accounts`
`21:36.39`	`cnet2`	`i guess that is what happens here.., there must not be any supervision.`
`21:36.48`	`struct`	`if I have a net2phone account and an ATA can I use asterisk to login to my net2phone account instead of using the ATA?`
`21:36.49`	`dlynes`	`cnet2: correct`
`21:36.51`	`UnixDog`	`the sip debug is there and the full extenUnixDog: sions.conf`
`21:36.52`	`cnet2`	`thanks for the explanation..`
`21:36.55`	`Strom_C`	`if not, just give them one account and set "number of line keys" to two`
`21:37.13`	`*** join/#asterisk PSU_Boss (n=Eric@unaffiliated/psuboss/x-309451)`
`21:37.16`	`dlynes`	`cnet2: also, another way where you might not see the other end hangup, is when you call numbers offered by certain cellphone service providers`
`21:37.16`	`anders047`	`Strom_C: is there an issue with multiple sip connections from same network IP?`
`21:37.26`	`Strom_C`	`anders047: no`
`21:37.28`	`dlynes`	`cnet2: because they don't hang up the call properly`
`21:37.55`	`anders047`	`currently I have them set up as 2 separate extensions, but "sip show peers" shows that only the first line on each phone actually has an IP`
`21:38.12`	`cnet2`	`ok.., i was reading on alcatel public pbxs, and they talk about the resistance, and loop on open n close`
`21:38.18`	`pepse`	`hi, this may be a strange question..`
`21:38.18`	`dlynes`	`cnet2: one such cellphone service provider I can think of, off the top of my head is Fido/Microcell`
`21:38.20`	`anders047`	`the other is listed as Unspecified with port 0`
`21:38.22`	`UnixDog`	`it a very basic dial plan`
`21:38.27`	`pepse`	`can i make asterisk dial a number for me when I pick up the phone?`
`21:38.37`	`UnixDog`	`to come in on a did and go back out a trunk`
`21:38.41`	`pepse`	`i mean, generate a dtmf '1' when i pick up, rather than having me do it manually`
`21:38.53`	`dlynes`	`pepse: you mean like a taxi hotline?`
`21:38.54`	`Strom_C`	`pepse: why`
`21:39.03`	`cnet2`	`i'll research a little more on these old alcatel pbx. thanks again.`
`21:39.06`	`booray`	`pepse: you just want to 10 digit dial instead of 11?`
`21:39.18`	`pepse`	`well no, i have to dial a 1 to accept a call`
`21:39.29`	`pepse`	`so instead of doing it from the handset, i want * to just do it when i pick up`
`21:39.32`	`UnixDog`	`pepse go install freepbx`
`21:39.45`	`pepse`	`UnixDog: uhm.. heh, ok.`
`21:39.56`	`dlynes`	`UnixDog: how does that help him?`
`21:40.37`	`PSU_Boss`	`Hi, Does anyone know if asterisk is able to call multiple phones at once?`
`21:40.39`	`dlynes`	`pepse: what's forcing you to dial '1' to accept a call?`
`21:40.43`	`pepse`	`anyway, I have exten => yadda,1,Set(CALLERID(all)=etc), and 2,Dial(${PHONE})..`
`21:40.58`	`dlynes`	`PSU_Boss: yes...Dial(Tech/exten&Tech/exten&Tech/exten&...)`
`21:41.02`	`pepse`	`dlynes: Grandcentral`
`21:41.31`	`dlynes`	`pepse: is that a brand of pbx?`
`21:41.53`	`pepse`	`dlynes: no, grandcentral.com, gives you a local number that forwards to multiple numbers, so one number for cell/home/office/etc`
`21:41.59`	`UnixDog`	`qwell you alive`
`21:42.12`	`Seldon75`	`im a bit confused about the fxo/fxs terminology but we have 8 ports that go to trunks and 4 that connect to analog phones :S`
`21:42.14`	`UnixDog`	`I need help with my a fast easy dial plan is not working`
`21:42.15`	`PSU_Boss`	`dlynes, what i want to do is when someone dials the number, asterisk will forward it to two cellphones, will that situation work?`
`21:42.19`	`UnixDog`	`its small`
`21:42.25`	`UnixDog`	`http://pastebin.ca/716499`
`21:42.35`	`anders047`	`PSU_Boss: yes. http://www.voip-info.org/wiki-Asterisk+cmd+Dial`
`21:42.36`	`dlynes`	`PSU_Boss: yes`
`21:42.37`	`Seldon75`	`dlynes: i guess thats 8 fxs and 4 fxo?`
`21:42.51`	`*** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)`
`21:43.12`	`dlynes`	`Seldon75: how many red modules do you have? how many green ones do you have?`
`21:43.35`	`dlynes`	`Seldon75: it seems something in wanrouter isn't working the way it's supposed to`
`21:43.45`	`dlynes`	`Seldon75: actually...just thought of something`
`21:44.14`	`pepse`	`Strom_C: no ideas?`
`21:44.15`	`dlynes`	`Seldon75: try lsmod \| grep wan`
`21:44.18`	`WilliamK`	`do you tend to find that nothing in life ever works the way you want?`
`21:44.20`	`WilliamK`	`:)`
`21:44.22`	`dlynes`	`Seldon75: and pastebin that`
`21:44.25`	`PSU_Boss`	`dlynes & anders047 : ok, the problem with that is that if one of the cellphones is turned off, and it goes straight to voicemail, asterisk sees that as an answer, and the other simultaneous call will be dropped, right?`
`21:44.37`	`dlynes`	`PSU_Boss: correct`
`21:44.40`	`Strom_C`	`pepse: my idea is "you are too lazy"`
`21:44.45`	`PSU_Boss`	`is there anyway to detect a voicemail or get around that?`
`21:44.50`	`pepse`	`Strom_C: Agreed. But is it possible :)`
`21:45.05`	`Seldon75`	`4 fxo and 2 fxs modules came with our card`
`21:45.06`	`WilliamK`	`Strom, you're being sooo nice today :)`
`21:45.30`	`pepse`	`can asterisk send dtmf on it's own after that 2,Dial(${PHONE})?`
`21:45.34`	`Seldon75`	`ok`
`21:45.38`	`dlynes`	`PSU_Boss: host the voicemail yourself?`
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`21:45.49`	`pepse`	`like, does me picking up make it go to 3, or does it go to 3 only after I hang up`
`21:46.03`	`*** part/#asterisk Cresl1n (n=matt@65.116.224.30)`
`21:46.13`	`dlynes`	`pepse: go to the wiki and read up on the dial command...it can do a whole bunch of stuff you're probably not aware of`
`21:46.15`	`PSU_Boss`	`dlynes, but it's a cellphone.. can the voicemail of the cellphone be turned off?`
`21:46.41`	`dlynes`	`PSU_Boss: or you can have the cellphone call forward on busy/no answer to your asterisk pbx's voicemail`
`21:46.59`	`dlynes`	`PSU_Boss: instead of to the cellphone's voicemail`
`21:47.10`	`Seldon75`	`dlynes: http://www.pastebin.ca/716506`
`21:47.44`	`PSU_Boss`	`dlynes, hmm, ok. one more question. do you have any suggestions on service providers, or a good website for information? i'm very new to this whole thing`
`21:47.50`	`Seldon75`	`i have the packaging for the modules for our card; it's definitely 4 x fxo and 2 x fxs`
`21:48.01`	`Seldon75`	`does that mean my zaptel is wrong`
`21:48.13`	`pepse`	`dlynes: hm, I'm not sure dial will do what I am looking for tho.`
`21:48.16`	`dlynes`	`Seldon75: do you have the molex adapter connected to a power cord?`
`21:48.59`	`UnixDog`	`anyone have input on my pastebin`
`21:49.05`	`UnixDog`	`http://pastebin.ca/716499`
`21:49.05`	`pepse`	`oh, maybe M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers)`
`21:49.26`	`pepse`	`but how would I make it send just a dtmf '1' to the call?`
`21:49.41`	`pepse`	`haha oops`
`21:49.42`	`pepse`	`sorry`
`21:49.48`	`pepse`	`D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel.`
`21:49.59`	`UnixDog`	`in your dialplan 1${EXTEN}`
`21:50.00`	`dlynes`	`pepse: wrong`
`21:50.00`	`pepse`	`they already thought of that!`
`21:50.12`	`dlynes`	`pepse: that'll send a '1' to the person picking up the phone`
`21:50.13`	`UnixDog`	`but that would do it on every call`
`21:50.29`	`Seldon75`	`dlynes: yes`
`21:50.31`	`UnixDog`	`this is for your outbound dialing only`
`21:51.09`	`dlynes`	`Seldon75: what i would do then, is get sangoma to ssh into your box to see if your fxs modules are working properly`
`21:51.22`	`dlynes`	`Seldon75: but in the meantime, I would check all analog phones to see fi they have the issue`
`21:51.34`	`dlynes`	`Seldon75: maybe it's just 1 or 2 on that specific module`
`21:52.00`	`UnixDog`	`exten => NXXNXXNXXX,s,1,DIal(protocal/provider,1${EXTEN})`
`21:52.09`	`Seldon75`	`aha ok`
`21:52.12`	`UnixDog`	`would add a 1`
`21:52.17`	`Seldon75`	`thanks dlynes`
`21:54.05`	`*** join/#asterisk Strom_M (n=strom@208.127.172.112)`
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`21:56.51`	`Strom_C`	`UnixDog: except the NANP format is NXX-NXX-XXXX`
`21:56.56`	`Strom_C`	`and you misspelled "protocol"`
`21:57.03`	`JoeDeveloper`	`I am using Lumenvox speech recognition, and I just discovered that if I do a SpeechBackground and then hangup, if the script then executes SpeechDeactivateGrammar, it gets an error and exits. Is there something different I should be doing on a hangup to deactivate a grammar?`
`21:57.24`	`UnixDog`	`exten => X.,s,1,DIal(protocal/provider,1${EXTEN})`
`21:57.30`	`UnixDog`	`exten => X.,s,1,DIal(protocol/provider,1${EXTEN})`
`22:01.08`	`Strom_C`	`UnixDog: still wrong`
`22:02.18`	`Seldon75`	`dlynes: hmm, strange thing; the fxs-es are channels 9,10,11 and 12. If i plug the phone into 9 i get nothing, 10 i get dialtone, 11 i get dialtone and 12 I get nothing`
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`22:11.27`	`UnixDog`	`http://pastebin.ca/716515`
`22:13.05`	`UnixDog`	`any input on this`
`22:13.55`	`UnixDog`	`its very basic`
`22:14.21`	`UnixDog`	`yet they get fast busy every time you dial`
`22:15.15`	`UnixDog`	`you/they`
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`22:17.03`	`UnixDog`	`http://pastebin.ca/716519`
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`22:18.12`	`UnixDog`	`thats the full extensions.conf`
`22:18.45`	`UnixDog`	`we are making a pass threw / hub system`
`22:19.06`	`UnixDog`	`that takes in a call from a ip and shoves it back out a trunk`
`22:19.44`	`UnixDog`	`strom can you look at it`
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`22:29.49`	`UnixDog`	`Strom_C: you here still`
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`22:33.39`	`*** join/#asterisk shtoom (n=shtoom@59.93.115.126)`
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`22:40.08`	`pepse`	`i can't tell if i'm getting flooded/ddosed, or if my kernel is just lame.`
`22:40.43`	*** join/#asterisk Yourname` (i=Miranda@unaffiliated/yourname/x-837320)
`22:41.10`	Yourname`	`Hi. Who thinks using manager to parse CLI output is a good idea? (Don't want to use CDRs because of CPU load)`
`22:41.43`	`Mavvie`	Yourname`: I use it to monitor the status of SIP/IAX and PRI neighbours.
`22:42.01`	`Strom_C`	`UnixDog: yes, I'm here`
`22:42.05`	`Strom_C`	`what do you want me to look at`
`22:42.14`	`UnixDog`	`http://pastebin.ca/716519`
`22:42.20`	Yourname`	`Mavvie: I'll be needing it to monitor things like a few NoOps, etc.`
`22:42.21`	`pepse`	`UnixDog: were you trying to answer my question? Because I'm not adding a 1 to numbers I'm dialing. I want asterisk to dial a 1 when I answer a call`
`22:42.36`	`Mavvie`	Yourname`: use /var/log/asterisk/something
`22:42.42`	`UnixDog`	`yes`
`22:42.50`	`UnixDog`	`exten => X.,s,1,DIal(protocol/provider,1${EXTEN})`
`22:43.04`	`Strom_C`	`UnixDog: wrong wrong and wrong`
`22:43.11`	`UnixDog`	`why`
`22:43.18`	`*** join/#asterisk anthm (n=anthm@mb30736d0.tmodns.net)`
`22:43.18`	`*** mode/#asterisk [+o anthm] by ChanServ`
`22:43.29`	`Strom_C`	`because that's like answering a yes or no question with "cheesecake"`
`22:43.29`	`pepse`	`UnixDog: you're still not getting it`
`22:43.32`	`Strom_C`	`it's entirely the wrong answer`
`22:43.33`	`UnixDog`	`?`
`22:43.56`	`Strom_C`	`UnixDog: also, what you sent me is completely not what you need in order to accomplish the task of creating a tandem type switch`
`22:44.00`	`hmmhesays`	`i love how firefox eats your system resources`
`22:44.00`	`pepse`	`UnixDog: i'm not adding a 1 to numbers i'm dialing`
`22:44.17`	`Strom_C`	`pepse: forget it - UnixDog is clearly not understanding the basics`
`22:44.31`	`UnixDog`	`I am going by what the books say`
`22:44.49`	`pepse`	`then you're reading the books wrong maybe :)`
`22:44.54`	`pepse`	`err understanding`
`22:44.58`	Yourname`	`Mavvie: Yeah, /var/log/asterisk/messages, but that only has SIP regitrations, queue activity, etc. Not NoOP info..`
`22:45.08`	`UnixDog`	`but whats wron with my switch setup`
`22:45.16`	`Mavvie`	Yourname`: add a debug logger instance and it will show up there.
`22:45.19`	`UnixDog`	`it comes in and dials rigt back out`
`22:45.37`	`Strom_C`	`UnixDog: it's overcomplicated and you're using that stupid macro`
`22:45.38`	Yourname`	`Mavvie: What do you mean?`
`22:45.41`	`UnixDog`	`Strom_C: its bulling the data from everything I set`
`22:45.55`	`Strom_C`	`UnixDog: which gui are you using?`
`22:45.58`	`Mavvie`	`[Sep 27 08:45:46] VERBOSE[21773] logger.c: -- Executing [try@macro-call-ext-sjh-try:1] NoOp("SIP/10.252.11.130-0944ca00", "Try") in new stack`
`22:46.01`	`UnixDog`	`it needs to know the dial status`
`22:46.02`	`Mavvie`	`That's a NoOp`
`22:46.16`	`UnixDog`	`thats whyI wrote the macro`
`22:46.26`	Yourname`	`Ok, and how did you get it into /var/log/asterisk/messages?`
`22:46.36`	`UnixDog`	`then explain what should be done`
`22:46.37`	`Mavvie`	`Add an extra instance in the logger configuration.`
`22:46.44`	Yourname`	`ah`
`22:46.48`	`Strom_C`	`UnixDog: you wrote that macro?`
`22:46.54`	`*** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell)`
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`22:46.58`	`UnixDog`	`es`
`22:47.02`	`Strom_C`	`from scratch?`
`22:47.02`	`UnixDog`	`yes`
`22:47.04`	`UnixDog`	`yes`
`22:47.33`	`Strom_C`	`call me skeptical, but I find that hard to believe`
`22:47.48`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`22:47.48`	`*** mode/#asterisk [+o russellb] by ChanServ`
`22:48.49`	`UnixDog`	`I pulled it from all my other working dial plan`
`22:49.04`	`UnixDog`	`and basicly scripted it out`
`22:49.39`	`*** join/#asterisk THX2000 (n=the@netblock-208-127-150-56.dslextreme.com)`
`22:49.43`	Yourname`	`Mavvie: Thanks, reading logger.conf and logging on the wiki.`
`22:49.44`	`Strom_C`	`UnixDog: you stole that macro from freepbx`
`22:49.48`	`UnixDog`	`no`
`22:49.51`	`Strom_C`	`you did not write it from scratch`
`22:49.57`	`UnixDog`	`I dont touce freepbx`
`22:50.03`	`Strom_C`	`i'm willing to help, but please don't make stuff up`
`22:50.03`	`Strom_C`	`no?`
`22:50.12`	`UnixDog`	`I use thirdlane and write dial plan for thirdlane all day`
`22:50.23`	*** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq)
`22:50.24`	`UnixDog`	`they stole 3/4 of the dieal plan we use`
`22:50.30`	`Strom_C`	`explain again why you were showing me the pastebin?`
`22:50.36`	`*** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net)`
`22:51.01`	`pepse`	`damn. D(#) does send it to the called party :)`
`22:51.04`	`UnixDog`	`because its not going threw`
`22:51.10`	`UnixDog`	`they get a fast busy`
`22:51.20`	`pepse`	`so I have to execute a macro, obviously, but what command will send a dtmf to the calling party?`
`22:51.28`	`UnixDog`	`I have never used aster as a redirect`
`22:51.31`	`UnixDog`	`like this`
`22:51.43`	`Strom_C`	`UnixDog: so pastebin some console output`
`22:51.47`	`Strom_C`	`also, when did you write this macro?`
`22:51.52`	`UnixDog`	`and it should be working so I need 2nd eyes to go point out`
`22:52.29`	`UnixDog`	`the orignal macro was bigger and was writen a year ago`
`22:52.41`	`UnixDog`	`I yanked parts out of it for tis`
`22:52.52`	`UnixDog`	`andon thirdlane gui I use and dev`
`22:53.02`	`pepse`	`ok. SendDTMF. thanks :)`
`22:53.13`	`UnixDog`	`I use thirdlane gui wich is older. and cleaner`
`22:53.46`	`Strom_C`	`UnixDog: pastebin some CLI output already`
`22:54.12`	`UnixDog`	`hold a min`
`22:54.44`	`*** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net)`
`22:55.38`	`pepse`	`so, I will have 2,Dial(${PHONE},,M(macro-send-one))..`
`22:55.59`	`UnixDog`	`having a ip call shoved right now`
`22:56.06`	Yourname`	`Mavvie: What about for realtime monitoring?`
`22:56.12`	`pepse`	`but then, what's the exten => etc,1,SendDTMF(1)?`
`22:56.19`	`pepse`	`like, what goes in place of etc`
`22:57.25`	`*** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-59-195.pskn.east.verizon.net)`
`22:57.38`	`Mavvie`	Yourname`: what do you want to monitor realtime?
`22:57.46`	`hmmhesays`	`so who wants to write me chan_yahoo?`
`22:57.53`	`Qwell_`	`why?`
`22:58.36`	Yourname`	`Mavvie: I was thinking parsing CLI through manager will give me more realtime information, like a NoOp that says DTMF pressed 1. And then that'll autoupdate a website that shows "5 people did DTMF." "6 people did DTMF.", etc.`
`22:58.40`	`pepse`	`ah, it's s`
`22:59.00`	`Mavvie`	Yourname`: just tail it with "tail -F"
`22:59.22`	*** part/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq)
`23:00.05`	`pepse`	`damnit. SendDTMF only sends to the called party also`
`23:00.09`	`UnixDog`	`the person making the test call is slower then (*^`
`23:00.19`	`pepse`	`how do I send dtmf to the calling party :/`
`23:02.26`	`UnixDog`	`waint`
`23:02.30`	`UnixDog`	`waiting`
`23:04.37`	`Strom_C`	`UnixDog: make the test call yourself`
`23:04.57`	`*** join/#asterisk linagee_ (n=linagee@about/linux/staff/linagee)`
`23:05.52`	`pepse`	`any idea how the G option works?`
`23:06.00`	`pepse`	`wiki says G(context^exten^pri), does ^ mean or or and?`
`23:06.15`	`pepse`	`or does it actually mean nameofcontext^extennumber^pri?`
`23:06.36`	`Qwell_`	`it's actual syntax`
`23:06.49`	`pepse`	`what is the pri?`
`23:06.53`	`Qwell_`	`priority`
`23:07.19`	`UnixDog`	`brb trying to set exten now`
`23:07.21`	`pepse`	`hm, but it says it assigns the calling party a priority, then called party priority x+1.. does that x mean the pri i set?`
`23:07.26`	`UnixDog`	`so I can dial`
`23:07.27`	Yourname`	`Mavvie : Hmm, I got that working. But to run a parser, I'd need to keep that file stream open in the background. :S`
`23:08.33`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`23:08.36`	`sandorp`	`where do I configure which ports are supposed to pick a call? only port 1 of the 8 ports I have answers; the other ports appear to answer`
`23:09.54`	`*** join/#asterisk saftsack (n=saftsack@pD9E0685A.dip.t-dialin.net)`
`23:09.57`	`Mavvie`	Yourname`: yes.
`23:09.58`	`sandorp`	`I updated users.conf already`
`23:10.25`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`23:10.25`	`*** mode/#asterisk [+o russellb] by ChanServ`
`23:10.35`	`UnixDog`	`I have to wit they just rebooted the server no notice`
`23:11.05`	Yourname`	`Mavvie: And I can't figure out how to.`
`23:11.20`	`Mavvie`	`tail -F foo \| parser &`
`23:11.31`	`UnixDog`	`brb`
`23:11.37`	Yourname`	`Tried that out, but the stream closes.`
`23:14.39`	`pepse`	`What do you think, could I use G() to send the caller and callee into a bridge and then send the dtmf into there?`
`23:14.45`	`*** join/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca)`
`23:15.00`	`Voicemeup`	`hi all, if i have spa942 - >asterisk -> asterisk 2 -> pri`
`23:15.13`	`Voicemeup`	`how can i have hte moh play from asterish and not asterisk 2 when spa puts one on hold`
`23:18.55`	`Voicemeup`	`silence here is hmm unabitual`
`23:21.03`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
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`23:21.04`	`sandorp`	`nm, my phone company is to blame`
`23:21.23`	`*** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)`
`23:24.39`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`23:25.59`	`*** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
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`23:26.46`	`Voicemeup`	`<PROTECTED>`
`23:27.05`	`pepse`	`hey, is there a .conf where all the dtmf frequencies are specified?`
`23:27.06`	`JT`	`i swear i saw the same question 10 minutes ago`
`23:27.15`	`Voicemeup`	`yeap its me`
`23:27.32`	`Qwell_`	`pepse: no, it's a static list`
`23:27.42`	`Voicemeup`	`since then 10 people joined, with the non response i assumed all where awol, and tought these could answer it`
`23:27.59`	`JT`	`weird thought`
`23:28.06`	`pepse`	`Qwell: can Playtones do dtmf?`
`23:28.24`	`struct`	`if I have a net2phone account and an ATA can I use asterisk to login to my net2phone account instead of using the ATA?`
`23:28.58`	`UnixDog`	`ok it works`
`23:29.15`	`pepse`	`struct: http://www.voip-info.org/wiki/view/Asterisk+settings+Net2phone`
`23:29.42`	`pepse`	`struct: I guess if net2phone allows standard sip registration, i would think it'd work fine`
`23:29.47`	`UnixDog`	`if I register but if I try to send a straight sip anon call with out registering it faisl`
`23:29.52`	`pepse`	`struct: maybe look at the settings in the ATA`
`23:29.53`	`UnixDog`	`fails even`
`23:29.54`	`Voicemeup`	`yeah JT i know.. i kep having weird tought about some number.. 818.. not sure why either but its in my mind anyhow`
`23:30.03`	`struct`	`pepse: ok thanks`
`23:30.21`	`struct`	`pepse: I was thinking that since its a hardware device maybe software would not be able to use it`
`23:30.42`	`UnixDog`	`this is a pisser`
`23:33.42`	`vader--`	`any of you guys know what the price of a full T3 runs on average?`
`23:34.24`	`*** join/#asterisk [hC] (n=hardcore@wsip-70-184-124-51.ph.ph.cox.net)`
`23:35.42`	`JT`	`vader--: a voice t3 or something else?`
`23:38.27`	`pepse`	`damn. Playtones doesn't seem to work either`
`23:39.34`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
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`23:41.46`	`pepse`	`woah.`
`23:42.02`	`pepse`	`This is extra lame. When I use DIAL(${PHONE},,M(macroname))`
`23:42.20`	`pepse`	`it doesn't actually pick up (connect to the originating caller) until after the macro is done`
`23:42.48`	`*** part/#asterisk [hC] (n=hardcore@wsip-70-184-124-51.ph.ph.cox.net)`
`23:43.38`	`pepse`	`bah, i say!`
`23:44.00`	`pepse`	`there should be an option to pick up right away`
`23:46.41`	`dlynes`	`Seldon75: sounds like modules 5 and 6 are defective to me, but i wouild get sangoma to log in to check to make sure`
`23:47.09`	`*** join/#asterisk ta^3 (n=tacvbo@65.116.224.30)`
`23:47.23`	`dlynes`	`Seldon75: they can run a few commands against wanrouter to check to see what the issue is for sure, and then send a ticket into rma's for you to let rma's know that two defective modules are coming back`
`23:48.38`	`*** part/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net)`
`23:49.18`	`russellb`	`pepse: well you have to understand how much more extremely complex that would be`
`23:49.18`	`russellb`	`because then you have the connection to another channel as well as a connection to applications at the same time`
`23:49.18`	`russellb`	`and having to do audio mixing between the two ... and not to mention dealing with signaling being passed from two different connections ...`
`23:49.19`	`russellb`	`it's just ... not trivial :)`
`23:51.13`	`dlynes`	`pepse: on Aastra phones...not sure about others, you can have it send an xml message to an ip address when the phone goes off hook`
`23:51.41`	`dlynes`	`pepse: perhaps you could use that to dial a '1' on a line, and then call the aastra phone after it's finished`
`23:51.52`	`dlynes`	`pepse: using an agi script, or something similar`
`23:52.13`	`dlynes`	`pepse: but that's far from a trivial solution`
`23:53.29`	`*** join/#asterisk kiscokid (n=ron@208.106.35.66)`
`23:54.35`	`*** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`23:56.59`	`*** join/#asterisk Toerkeium (i=oo@201.216.206.221)`
`23:58.22`	`*** join/#asterisk brian (n=brian@unaffiliated/brian)`
`23:58.27`	`kiscokid`	`anybody know if you can have an annoucement occur immediately when a caller enters a queue?`
`23:58.45`	`Voicemeup`	`kisco`
`23:59.14`	`Voicemeup`	`ried that.. only way i found.. (cause it waits for first agent timeout.) is to use playbacl( this can be monitored) then exten -=> queue`
`23:59.30`	`Voicemeup`	`if htat waht your trying to acchieve`
`23:59.47`	`kiscokid`	`yeah, I just thouight of that as I finished typing in my question`
`23:59.57`	`Voicemeup`	`yeah , been there , done that too`

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