irclog2html for #asterisk on 20061116

`00:00.03`	`JT`	`yeah`
`00:00.11`	`JT`	`you can buy it in every combination`
`00:00.13`	`bulatitoy`	`is that what I need if i will be using 4 phone lines?`
`00:00.17`	`JT`	`and buy modules and add them later`
`00:00.18`	`JT`	`yes`
`00:00.50`	`bulatitoy`	`i can see in ebay, X100P is about $15`
`00:00.58`	`JT`	`fake ones`
`00:01.01`	`bulatitoy`	`can i use this one for testing?`
`00:01.04`	`JT`	`they're no longer produced`
`00:01.06`	`JT`	`you can`
`00:01.12`	`JT`	`but there is no guarantee`
`00:01.15`	`bulatitoy`	`and have several softphones just to simulate`
`00:01.17`	`JT`	`up to you whether it's worth it`
`00:01.21`	`JT`	`sure`
`00:01.28`	`e-horn`	`shido6: any ideas?`
`00:01.36`	`bulatitoy`	`so those cards are old`
`00:01.46`	`bulatitoy`	`might as well invest on a TDM`
`00:01.59`	`JT`	`yes, no longer produced, the main chipset IC is no longer made`
`00:02.12`	`JT`	`so all the ones on ebays are either fakes or factory seconds of the main IC`
`00:02.37`	`bulatitoy`	`yikes! may not work huh`
`00:02.50`	`*** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com)`
`00:04.06`	`bulatitoy`	`thanks a lot JT!`
`00:04.15`	`bulatitoy`	`thanks to the others too :)`
`00:04.21`	`*** join/#asterisk dahunter3 (n=dahunter@122.sub-75-215-171.myvzw.com)`
`00:04.35`	`bulatitoy`	`i will go ahead and give it a try`
`00:04.47`	`bulatitoy`	`again thanks guys!`
`00:05.13`	`JT`	`going to try your luck with an X100P?`
`00:05.28`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`00:05.50`	`bulatitoy`	`nah..i'll invest on a TDM400P :)`
`00:05.59`	`*** join/#asterisk sjobeck (n=sjobeck@66-182-49-26.atgi.net)`
`00:06.20`	`JT`	`ah, yeah, at least you know it should work`
`00:06.25`	`JT`	`as well as analogue lines do ;)`
`00:07.22`	`bulatitoy`	`so 1FXO port = 1 phone line and 1 FXS port = 1 phone connection?`
`00:07.34`	`bulatitoy`	`coz i see a card with this config`
`00:07.53`	`JT`	`yes`
`00:08.36`	`bulatitoy`	`but i can extend my simulation/testing by using softphones, right?`
`00:08.50`	`JT`	`sure`
`00:09.01`	`bulatitoy`	`the card is worth $180 in ebay`
`00:09.12`	`bulatitoy`	`hope this one works :)`
`00:09.12`	`JT`	`the only difference in the asterisk dialplan is unsead of a Zap/ channel it's a SIP/ channel`
`00:09.16`	`JT`	`or IAX`
`00:09.18`	`JT`	`heh`
`00:09.26`	`JT`	`check digium.com and resellers too`
`00:09.37`	`*** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net)`
`00:09.40`	`bulatitoy`	`ok, i will`
`00:09.41`	`JT`	`may be a better bet than going ebay`
`00:09.55`	`JT`	`but if ebay is the cheapest, and you think the seller is ok, then go for it`
`00:10.09`	`bulatitoy`	`yep ill check on it`
`00:10.41`	`*** join/#asterisk hads (n=hads@mail.nice.net.nz)`
`00:11.11`	`*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)`
`00:11.33`	`*** join/#asterisk cloud9 (n=nothing@smtp.a-cti.com)`
`00:11.56`	`Pumas`	`I have a problem when I try to record an Voip incoming call it stops at 5 minutes, I just can record 5 minutes of conversation`
`00:12.25`	`monsted`	`bulatitoy: it may end up being cheaper and easier to buy a network-attached FXO/FXS gateway instead of a card, depending on what you need and if you find a good deal`
`00:12.25`	`Pumas`	`why ? I have no maxduration in command record`
`00:12.29`	`cloud9`	`quick question.. I've been searching voip-info.org to see if my dialplan can detect whether the call came in on zap channel 1 or 2. is there an easy way to do this?`
`00:12.45`	`cloud9`	`the channel is on an fxo card, btw`
`00:14.05`	`JT`	`~dos`
`00:14.08`	`jbot`	`rumour has it, dos is (Disk Operating System) This OS is what got it all started for PCs. Denial of Service...`
`00:14.13`	`JT`	`~docs`
`00:14.14`	`jbot`	`docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html`
`00:14.14`	`JT`	`heh`
`00:15.04`	`bulatitoy`	`thanks monsted`
`00:15.48`	`JT`	`cloud9: set different contexts for the different channels`
`00:15.55`	`JT`	`bulatitoy: oh yeah, an ATA is usually cheaper`
`00:16.03`	`JT`	`although you lose a bit of flexibility then`
`00:16.07`	`JT`	`which you may or may not need`
`00:16.12`	`monsted`	`bulatitoy: somthing like a sipura device is pretty cheap`
`00:16.33`	`monsted`	`JT: how is a zaptel thing more flexible?`
`00:16.39`	`bulatitoy`	`sipura 3000 is fine?`
`00:16.39`	`cloud9`	`JT : but don't I need to detect what channel the call is coming in on, and then point my dialplan to that context?`
`00:16.47`	`Pumas`	`I have a problem when I try to record an Voip incoming call it stops at 5 minutes, I just can record 5 minutes of conversation`
`00:16.50`	`monsted`	`(as long as you have room to stack the sipuras somewhere ;))`
`00:16.51`	`JT`	`monsted: you can configure a lot more options`
`00:16.56`	`JT`	`eg distinctive ring detection`
`00:17.00`	`*** join/#asterisk hads (n=hads@mail.nice.net.nz)`
`00:17.19`	`JT`	`and you also don't have to configure things twice`
`00:17.29`	`monsted`	`i think i have enough options, thankyouverymuch ;)`
`00:17.30`	`JT`	`with a sipura, you have to configure the sipura and asterisk`
`00:17.54`	`JT`	`does it do distinctive ring detection and generation?`
`00:18.18`	`martin5519`	`Anyone know if there's any real difference between Asterisk Open Source and the Asterisk Business Edition from Digium`
`00:18.19`	`JT`	`cloud9: in the zapata.conf file you set the context for a channel or group of channels`
`00:18.34`	`Pumas`	`HELP ME please: I have a problem when I try to record an Voip incoming call it stops at 5 minutes, I just can record 5 minutes of conversation`
`00:18.43`	`JT`	`martin5519: some drivers for proprietary stuff, and greater bugfix priority`
`00:18.50`	`monsted`	`JT: to be quite honest, i haven't dug through all of the options and actually figured out how to get it to register yet, so...`
`00:18.58`	`JT`	`Pumas: we heard you the first 3 times`
`00:19.11`	`JT`	`monsted: don't assume the options are enough for everyone :P`
`00:19.40`	`Pumas`	`ok, I'll wait for your help, thanks`
`00:19.46`	`martin5519`	`Someone told me this though:`
`00:19.47`	`martin5519`	`I'm not sure I would go the ABE route - you'll be versions behind the`
`00:19.47`	`martin5519`	`current stable version of Asterisk (which is what we are running) and`
`00:19.47`	`martin5519`	`it's not cheap.......`
`00:19.49`	`monsted`	`JT: i really wanted my cisco router to do the POTS-to-SIP stuff, but noone felt like buying me the hardware i needed ;)`
`00:20.05`	`CunningPike`	`Hmm - where'd ${FIELDQTY} go in 1.4?`
`00:20.17`	`JT`	`Pumas: to be honest you really haven't provided enough info to debug the problem`
`00:20.31`	`JT`	`people expect you to pastebin errors and configs etc`
`00:21.00`	`JT`	`martin5519: do you have a reason you need ABE? if not, don't`
`00:22.45`	`Pumas`	`sorry, I'm new on this what is a pastebin?`
`00:22.54`	`*** join/#asterisk worldjam (n=worldjam@220-253-80-140.QLD.netspace.net.au)`
`00:22.58`	`*** join/#asterisk dorphalsig (i=dorphals@pcsp168-254.supercabletv.net.co)`
`00:23.02`	`JT`	`~pb`
`00:23.03`	`jbot`	`i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`00:23.15`	`dorphalsig`	`Hello`
`00:23.23`	`bulatitoy`	`thanks guys!`
`00:23.26`	`bulatitoy`	`got to go`
`00:23.29`	`JT`	`cya`
`00:23.32`	`bulatitoy`	`thanks a lot for the help!!!`
`00:23.38`	`*** part/#asterisk bulatitoy (n=rmn@adsl-70-231-130-250.dsl.snfc21.sbcglobal.net)`
`00:23.47`	`dorphalsig`	`I'm trying to build a network of VoIP over Wifi`
`00:24.10`	`dorphalsig`	`however when I called linksys for support they told me I had to connect the wireless routers with cable`
`00:24.21`	`worldjam`	`Hey Is there a to set up a multiple voip chat using asterisk as in a group chat, all in the same convo ?`
`00:24.43`	`JT`	`worldjam: Meetme conferences`
`00:24.44`	`martin5519`	`Cunningpike - cool - I'm around as well... working on stuff... just wondering about the whole ABE thing`
`00:24.47`	`dorphalsig`	`I'm not sure that makes any sencs.`
`00:24.50`	`dorphalsig`	`*sense`
`00:24.59`	`martin5519`	`I guess we can purchase support and books from Digium eh!`
`00:25.05`	`JT`	`martin5519: cunningpike?`
`00:26.25`	`worldjam`	`Meetme conferences. thats pay but. is it possible to make your own voip conference?`
`00:26.45`	`JT`	`no`
`00:26.48`	`JT`	`app_meetme`
`00:26.55`	`JT`	`is an inbuilt asterisk application`
`00:27.02`	`worldjam`	`oh`
`00:27.32`	`worldjam`	`cheers mate`
`00:27.35`	`JT`	`np`
`00:27.39`	`justinu\|laptop`	`no one knows about the "ABE thing"`
`00:27.50`	`dorphalsig`	`Hello`
`00:28.59`	`justinu\|laptop`	`what's new?`
`00:29.10`	`*** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net)`
`00:29.21`	`dorphalsig`	`Me just looking around for somebody who knows wifi :P`
`00:29.23`	`dorphalsig`	`how bout u?`
`00:29.30`	`dorphalsig`	`I finally met Martin ...`
`00:29.38`	`justinu\|laptop`	`no kidding?`
`00:30.06`	`dorphalsig`	`Yeah ... we worked together a couple of times`
`00:30.09`	`justinu\|laptop`	`voip over wifi is sketchy... not sure if I buy into that as a decent technology`
`00:30.12`	`justinu\|laptop`	`too open to interference`
`00:30.21`	`JT`	`dorphalsig: i know about about wifi, but not sure exactly what the secnario is`
`00:30.43`	`JT`	`if by open you mean prone to interference and downtime, then yeah`
`00:30.55`	`justinu\|laptop`	`yeah`
`00:33.58`	`bblack`	`dorphalsig: check WDS (http://en.wikipedia.org/wiki/Wireless_Distribution_System), I don't think linsys supports it in standard firmware but alternative firmwares like OpenWRT do`
`00:34.00`	`*** join/#asterisk Strom_C (i=strom@nat/digium/x-4297617aac6df6a6)`
`00:36.55`	`*** join/#asterisk kschava (n=kschava8@blk-222-18-178.eastlink.ca)`
`00:37.12`	`Pumas`	`I have put my pastebin`
`00:37.22`	`*** part/#asterisk kschava (n=kschava8@blk-222-18-178.eastlink.ca)`
`00:38.27`	`Pumas`	`but it is the first time that I used it, now how you can get it?`
`00:38.50`	`icel`	`pumas: it should give you a url`
`00:38.59`	`JT`	`after you submit the pastebin, it should direct you to the url`
`00:39.05`	`Pumas`	`http://pastebin.ca/index.php`
`00:39.09`	`JT`	`no`
`00:39.12`	`icel`	`pumas: nice try`
`00:39.15`	`JT`	`have you submitted it`
`00:39.54`	`Pumas`	`yes in the page that jbot told me`
`00:40.25`	`JT`	`when you submit it, the new page will have a url`
`00:40.58`	`Pumas`	`http://pastebin.ca/249663`
`00:43.54`	`*** join/#asterisk ma-requin (n=youssef@dyn10.tcom.ou.edu)`
`00:44.01`	`JT`	`maybe it's the asterisk default maxduration`
`00:44.10`	`JT`	`try setting the maxduration argument to 0`
`00:44.39`	`JT`	`or a specific time in seconds`
`00:45.47`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`00:45.50`	`*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`00:47.36`	`Pumas`	`but this problem just happens when the call comes from VOIP to TRrix, when I call from a "normal" line To Trix all goes fine, WHY??`
`00:48.02`	`*** join/#asterisk cbm11211 (n=Administ@66.28.182.170)`
`00:48.26`	`JT`	`oh, trix`
`00:48.36`	`JT`	`what about normal calls not using record from voip?`
`00:52.25`	`dorphalsig`	`bblack --> What hardware would you recommend for WDS?`
`00:53.57`	`Pumas`	`I have a VOIP number and I dial it, and asterisk takes the call, and I start to record the conversation`
`00:54.07`	`dorphalsig`	`bblack --> I mean for VoIP over WiFi`
`00:54.37`	`Pumas`	`and in this case I have the timing problem`
`00:55.05`	`bblack`	`dorphalsig: I'd recommend WRT54G for WDS, if you have a controlled environment (like a big warehouse) without too much interferences you should be fine`
`00:55.48`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`00:56.13`	`dorphalsig`	`bblack ... actually its supposed to be a clinic ...`
`00:56.18`	`bblack`	`dorphalsig: the problems with WiFi phones is that the WiFi chips are quite power-hungry so the batteries don't last too long`
`00:56.54`	`bblack`	`dorphalsig: if you have radiation equipment that's interfering with 11b/g spectrum you're out of luck`
`00:57.17`	`dorphalsig`	`Hrmmmm`
`00:57.53`	`dorphalsig`	`bblack --> here comes my other question ... if I want to use ATAs to go "Physical"`
`00:58.02`	`dorphalsig`	`I would need some bridges, right?`
`00:58.39`	`bblack`	`dorphalsig: you mean having ethernet devices? (ATA/IP phones)`
`01:03.11`	`dorphalsig`	`yes`
`01:04.38`	`*** join/#asterisk cbm11211 (n=Administ@66.28.182.170)`
`01:04.49`	`bblack`	`dorphalsig: I'd recommend an ethernet network only for IP phones, not connected to Internet; the architecture of an ethernet network is kind of off-topic for this channel`
`01:05.43`	`dorphalsig`	`bblack may i priv u`
`01:05.44`	`dorphalsig`	`?`
`01:06.08`	`bblack`	`dorphalsig: sure`
`01:07.55`	`*** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-228-131.dc.res.rr.com)`
`01:08.59`	`*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)`
`01:12.33`	`Sponge_bob`	`<PROTECTED>`
`01:18.08`	`*** part/#asterisk brif8 (n=brif8@rrcs-67-78-24-180.se.biz.rr.com)`
`01:18.26`	`*** join/#asterisk piggie (n=pig@bb219-74-36-37.singnet.com.sg)`
`01:19.45`	`piggie`	`Hi all, I've successfully installed zaptel but when I went back to asterisk to run make, chan_zap.so isn't compiled. Any reasons why that could be so?`
`01:29.04`	`*** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca)`
`01:39.55`	`Pumas`	`somebody spanish?`
`01:46.24`	`dorphalsig`	`ya`
`01:46.25`	`dorphalsig`	`nme`
`01:47.58`	`*** join/#asterisk yl (n=yann@yannl.mtl.istop.com)`
`01:54.21`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`01:54.50`	`martin5519`	`hey, I assume that Asterisk has the built-in drivers for the boards on the Digium website... specifically... TDM2410E`
`01:55.06`	`mog`	`zaptel is what you are looking for`
`01:55.14`	`mog`	`chan_zap is in asterisk and talks to zaptel`
`01:55.36`	`martin5519`	`oh... so that's what zaptel is ...it's like the drivers interface to Asterisk?`
`01:55.47`	`martin5519`	`I always wondered what "zaptel" was`
`01:56.29`	`martin5519`	`so.. if I get that board TDM2410E and Asterisk,... I just have to install Zaptel as well?`
`01:58.24`	`JT`	`yes`
`01:58.37`	`martin5519`	`thanks! YUou know where I can read up on Zaptel?`
`01:58.53`	`martin5519`	`I gotta rtfm on zaptel.. have no idea on that`
`01:59.21`	`JT`	`and your kernel needs the crc-ccitt module`
`01:59.56`	`JT`	`either the book or voip info`
`02:00.03`	`JT`	`and the readmes for zaptel`
`02:00.06`	`JT`	`~thebook`
`02:00.08`	`jbot`	`methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`02:00.17`	`martin5519`	`oh geez... thanks for letting me know... I would be deploying on FreeBSD via the port`
`02:00.24`	`martin5519`	`Ports`
`02:00.32`	`JT`	`err, i have no idea how to do it then :)`
`02:00.44`	`JT`	`i believe there is zaptel for fbsd`
`02:00.56`	`martin5519`	`No - easy on BSD with Ports package it downloads allt eh dependencies for BSD but yes, I have to check all tha tout..`
`02:01.41`	`JT`	`oh, you already know how to do zaptel in fbsd then i assume`
`02:02.03`	`martin5519`	`No - not at all`
`02:02.08`	`martin5519`	`I'm taking notes of what you told me though`
`02:02.44`	`JT`	`you'll have to find documentation on how to do it with fbsd`
`02:02.57`	`martin5519`	`This is what BSD has:`
`02:02.58`	`martin5519`	`zaptel-1.0_1`
`02:02.58`	`martin5519`	`<PROTECTED>`
`02:03.00`	`JT`	`i don't think it's supported by digium, if that's even an issue ;)`
`02:03.04`	`martin5519`	`Is that it?`
`02:03.15`	`Strom_C`	`1.0 is really old`
`02:03.17`	`JT`	`yes except zaptel 1.0 is uber old`
`02:03.20`	`*** join/#asterisk _DAW (n=_DAW@adsl-156-78-145.msy.bellsouth.net)`
`02:03.34`	`martin5519`	`really eh... so we may have to deploy on LInux then - man that sucks`
`02:04.07`	`martin5519`	`It is SOOOO easy to do installs with BSD because of the Ports collections`
`02:04.29`	`Strom_C`	`why, is typing "make clean install" too difficult?`
`02:04.31`	`JT`	`yeah great`
`02:06.20`	`martin5519`	`Because - you have all the dependencies that you might be missing`
`02:06.39`	`Strom_C`	`asterisk and zaptel have something like six dependencies`
`02:06.52`	`Strom_C`	`install those from packages and then install asterisk from source`
`02:06.59`	`Strom_C`	`it's not exactly difficult`
`02:07.22`	`martin5519`	`Yeah... not difficult but a pain ..I used to use LInux all the time but the Ports package on FreeBSD makes it SO easy.`
`02:07.36`	`JT`	`SOO easy`
`02:07.41`	`JT`	`you've said that at least 3 times now`
`02:07.53`	`JT`	`man linux distros are just as easy`
`02:07.56`	`JT`	`if not easier`
`02:07.59`	`JT`	`s/man/many/`
`02:08.06`	`Strom_C`	`so installing the dependencies from packages is a pain, but installing asterisk from packages is SOOO easy?`
`02:08.13`	`martin5519`	`Have you guys used the Ports package on BSD?`
`02:08.14`	`*** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com)`
`02:08.17`	`Strom_C`	`am I missing something here? :)`
`02:08.21`	`*** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn)`
`02:08.53`	`martin5519`	`No.. I can install every individual dependency from a package... arrghh... what's the point..`
`02:09.10`	`Strom_C`	`you are missing the point`
`02:09.17`	`Strom_C`	`I said there were only six to worry about`
`02:09.25`	`JT`	`anyway`
`02:09.30`	`JT`	`in the overall scheme of things`
`02:09.43`	`JT`	`installing asterisk and zaptel will be ones of the least difficult things`
`02:09.53`	`JT`	`compared to the time needed to configure it`
`02:10.15`	`Strom_C`	`ncurses, ncurses-dev, openssl, openssl-dev, zlib, zlib-dev`
`02:10.24`	`martin5519`	`Listen... I've worked extensively with both Linux and FreeBSD ... all I'm saying is that I find administration of software on FreeBSD (installing, removing, modifying) much easier...... JT: Yeah, I ran Asterisk before but not Zaptel.`
`02:10.57`	`martin5519`	`So... because it's more popular for Linux.. I'll probably deploy on Linux - easier to find stuff I need`
`02:11.20`	`JT`	`you can deploy on either, who cares, as long as you can work it out`
`02:11.30`	`martin5519`	`yeah`
`02:12.00`	`*** join/#asterisk WGFreewill (n=chatzill@69-170-244-239.atlsfl.adelphia.net)`
`02:12.11`	`martin5519`	`Well, the version of Zaptel for FreeBSD is old - I just e-mail the author and asked if he has time to build a new Ports Package for it... we'd pay him... contribution to Open Source`
`02:12.35`	`JT`	`are you sure there is no more recent package available?`
`02:12.45`	`martin5519`	`freebsd.org/ports is where I'm going`
`02:12.50`	`martin5519`	`just did a search on zaptel`
`02:13.04`	`WGFreewill`	`anyone using asterisk 1.4 with g729, I see on my system it's not counting the licenses right`
`02:13.13`	`martin5519`	`Is this the most recent version of Asterisk? asterisk-1.2.13`
`02:13.22`	`martin5519`	`Because that's what BSD ports has`
`02:13.22`	`JT`	`yes most recent stable`
`02:13.32`	`martin5519`	`ok.. so it's just Zaptel that's not up to date`
`02:13.45`	`WGFreewill`	`0/2 encoders/decoders of 4 licensed channels are currently in use`
`02:13.56`	`martin5519`	`Maybe author will do something for me ... so... this Kernel module crc-ccitt... where do I get that?`
`02:14.12`	`JT`	`i have no idea what the fbsd equivalent is`
`02:15.43`	`martin5519`	`oh crap`
`02:16.25`	`martin5519`	`So.. I guess what I want to do first is load this special kernel module... then strip my kernel of stuff I don't want.. then intall Zaptel and Asterisk wit the new Board on the motherboard..`
`02:16.50`	`JT`	`you ought to search google for instructions of using zaptel with fbsd`
`02:16.55`	`JT`	`or voip-info.org`
`02:16.59`	`Supaplex`	`can I use EAGI to slice up wav files?`
`02:17.18`	`martin5519`	`cool... will do!!! Thanks JT - great intro!`
`02:17.37`	`JT`	`np`
`02:23.10`	`martin5519`	`yeah.. damn - no support for that board I need to put in - I check on voip-info.org`
`02:27.14`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`02:27.14`	`*** mode/#asterisk [+o Qwell] by ChanServ`
`02:27.18`	`Strom_C`	`it's qwell!`
`02:27.26`	`Qwell`	`where?!`
`02:27.28`	`Qwell`	`oh`
`02:27.31`	`Strom_C`	`right behind you!`
`02:27.49`	`Qwell`	`:D`
`02:27.59`	`Strom_C`	`heheh`
`02:28.08`	`Qwell`	`still in town?`
`02:28.09`	`Strom_C`	`man, this is irritating`
`02:28.10`	`Strom_C`	`yep`
`02:28.14`	`Strom_C`	`i'm at the office`
`02:28.47`	`Strom_C`	`I actually was about to give up on getting an ftp server working and instead go get some dinner. are you back?`
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`02:37.16`	`Qwell`	`yeah`
`02:37.23`	`Strom_C`	`are you hungry? :)`
`02:37.44`	`Qwell`	`nah`
`02:38.22`	`Qwell`	`need to unpack and such`
`02:38.28`	`Strom_C`	`k`
`02:38.38`	`Qwell`	`I'll be in the office tomorrow though`
`02:38.50`	`Qwell`	`barring any...natural disasters :P`
`02:39.12`	`justinu\|laptop`	`lol`
`02:39.37`	`Qwell`	`justinu\|laptop: it makes it funnier, when you realize that we've been on tornado watch all day :p`
`02:40.11`	`hoobastooba`	`I am trying to set up gnudialer and because it uses agents.conf for its user stuff, and I user AddQueueMember... I am trying to fake an agent login. they give an example of how to do this... but it does not work, and their irc channel is asleep. can anyone take a peak at what I am trying to do and see if it should work or not? http://pastebin.ca/249738`
`02:40.15`	`justinu\|laptop`	`well, if you're not around tomorow, we'll know you're in Oz`
`02:40.30`	`hoobastooba`	`i thought once you hung up the call, agentlogin logged you out of the queue.`
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`02:46.25`	`hoobastooba`	`i guess i just have that effect on a channel... i type something in, and the whole room goes silent. sorry.... I will go so you can get back to chatting.`
`02:49.03`	`jtf0518`	`will Asterisk scale to handle 200 phones and what kind of specs should the server have for that, i.e. CPU speed, dual core or not, RAM? We'd throw in a huge SATA RAID anyway for voicemail.`
`02:49.38`	`JT`	`i'd be thinging more clustering there than one huge machine`
`02:49.40`	`JT`	`thinking`
`02:50.00`	`jtf0518`	`how many nodes in the cluster for 200 phones?`
`02:50.13`	`*** join/#asterisk jbt (n=jtac@mnldns10.innodata.com.ph)`
`02:50.30`	`JT`	`no need to go too crazy, even if it's just 2 machines`
`02:50.36`	`JT`	`i guess it depends on connectivity`
`02:50.48`	`JT`	`how do they reach the outside world?`
`02:51.02`	`jtf0518`	`we currently have a Nortel Option 11C with 2 PRI`
`02:51.20`	`JT`	`ah right`
`02:52.38`	`jtf0518`	`we'd use the Asterisk server to handle all the routing and a Digium 4port PRI card for routing any of the calls to legacy phone sets to the PRIs on the Option 11C.`
`02:53.46`	`JT`	`a really nice setup would be with a T1 failover switch or a Redphon fonebridge2`
`02:53.52`	`JT`	`Redphone, even`
`02:54.03`	`JT`	`nope, Redfone`
`02:54.06`	`JT`	`damn strange spelling`
`02:54.15`	`JT`	`depends how important uptime is i guess`
`02:54.31`	`JT`	`http://www.voip-info.org/wiki/view/Redfone`
`02:54.34`	`jtf0518`	`uptime is nice, standing in an unemployment line sucks`
`02:54.48`	`JT`	`http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions`
`02:54.48`	`JT`	`http://www.voip-info.org/wiki/view/Failover+switches`
`02:55.24`	`JT`	`with a redphone unit, it terminates the pri in a hardware box, and sends the channels to asterisk using TDMoE`
`02:55.41`	`JT`	`or you could use a more traditional pri failover box`
`02:55.57`	`JT`	`then you need pri cards in both boxes`
`02:57.10`	`jtf0518`	`thanks JT`
`02:57.54`	`JT`	`http://www.junghanns.net/en/ISDNguard_produkt.html`
`02:58.09`	`JT`	`that's a layer1 bri or pri failover box that does 4 channels`
`02:58.14`	`dorphalsig`	`Cya`
`02:58.19`	`JT`	`designed for asterisk`
`02:59.10`	`JT`	`i'd still recommend redundant power supplies and RAID1 for the servers of course`
`02:59.26`	`JT`	`and it's up to you whether to set it up as a hot spare, or in a load balancing capacity`
`02:59.30`	`JT`	`with the load shared`
`02:59.39`	`JT`	`eg 1 pri in and 1 pri out in each box`
`03:00.58`	`jtf0518`	`yeah, but I'd need to feed two pri lines from the Asterisk to the Option 11C for legacy phones so it's a question of which server takes that role since the hardware can only be in one of them.`
`03:01.24`	`JT`	`and 2 pris from the telco?`
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`03:01.49`	`JT`	`well i was saying you could put 1 pri from the option 11c to 1 asterisk box, and 1 to the other`
`03:01.50`	`jtf0518`	`yes, two from the telco for inbound outbound and 2 to feed to the legacy PBX.`
`03:02.00`	`JT`	`that would be loadbalancing`
`03:02.08`	`JT`	`otherwise you could put them all through one box`
`03:03.21`	`jtf0518`	`Ok, then I have a dilemma. The two pri circuits are programmed in a way that they are both part of the same route and outbound calls try to grab the first open line from the last channel working upwards while inbound come in from the top of the channel and work downward.`
`03:03.46`	`Strom_C`	`so you have NFAS?`
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`03:04.30`	`jtf0518`	`NFAS? I should know that acronym but it's been about 9 years since my last Nortel class.`
`03:04.40`	`Strom_C`	`non facility associated signaling`
`03:04.41`	`JT`	`jtf0518: that could still work, unless it's NFAS`
`03:04.50`	`Strom_C`	`multiple T1 spans comprising a single PRI`
`03:04.52`	`JT`	`as the two boxes would link via ethernet`
`03:05.00`	`Strom_C`	`and sharing one or more d-channels`
`03:05.05`	`JT`	`so they could pass data`
`03:05.15`	`JT`	`but yeah, sharing d chans would be a problem across machines`
`03:05.46`	`jtf0518`	`so if I stat the trunks and each card shows it's own D-channel then it's not NFAS?`
`03:05.56`	`Strom_C`	`not necessarily`
`03:06.05`	`Strom_C`	`NFAS can have more than one d-channel`
`03:06.33`	`jtf0518`	`well, I guess I can dig out the manual and see if it's programmed that way.`
`03:07.00`	`JT`	`you could ring the telco, depending how approachable they are`
`03:07.31`	`jtf0518`	`that would be Time Warner but they don't support our PBX. AT&T does (or pretends to)`
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`03:07.59`	`Strom_C`	`no no`
`03:07.59`	`JT`	`just ask if the lines have NFA enabled, and if so, what is the configuration`
`03:08.05`	`JT`	`NFAS`
`03:08.09`	`Strom_C`	`ask the telco about the lines, not about the pbx :)`
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`03:08.25`	`jtf0518`	`ok, I'll check with them. So NFAS would be a showstopper?`
`03:08.31`	`JT`	`no`
`03:08.41`	`JT`	`it would just mean the lines would come into one box`
`03:08.48`	`JT`	`like you were originally saying`
`03:08.54`	`JT`	`you can still do failover`
`03:08.59`	`JT`	`just not pri load balancing`
`03:09.08`	`jtf0518`	`ah, ok`
`03:09.27`	`JT`	`unless it's something funky like 2 lines in on one box, 2 out on the other, which would be pointless`
`03:09.54`	`*** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net)`
`03:10.02`	`sudhir492`	`Hi All`
`03:10.11`	`*** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net)`
`03:10.17`	`jtf0518`	`thanks for all your help, JT and Strom_C`
`03:10.19`	`sudhir492`	`Has anyone compiled asterisk on FC6?`
`03:10.32`	`jtf0518`	`have a nice nite everyone!`
`03:10.35`	`jtf0518`	`bye`
`03:10.38`	`JT`	`you too`
`03:10.39`	`JT`	`cya`
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`03:20.00`	`puzzled`	`sudhir492: yes`
`03:21.21`	`sudhir492`	`puzzled: do I have to do anything extra for intel 64 bit before compiling ?`
`03:22.05`	`JT`	`anyone know how in the dialplan i could check if all lines in a zap group were in use?`
`03:22.14`	`JT`	`before attempting to dial`
`03:22.33`	`puzzled`	`sudhir492: probably. get the SRPMs from laimbock.com/asterisk/`
`03:22.56`	`sudhir492`	`ok`
`03:23.12`	`Strom_C`	`JT, if the group is full, Dial() will return an error`
`03:23.19`	`Strom_C`	`and execution continues`
`03:23.40`	`Strom_C`	`so you could do a gotoif based on ${DIALSTATUS}`
`03:24.01`	`sudhir492`	`puzzled: did you do the same? does that compile well?`
`03:24.21`	`sudhir492`	`let me try that. thanks`
`03:24.30`	`puzzled`	`sudhir492: yes, works like a charm. check the asterisk.spec file. you can activate a ton of features/patches`
`03:24.40`	`JT`	`Strom_C: i'm just wanting to a bit of emergency service dialling handling`
`03:24.53`	`JT`	`if all the zap lines to the telco are busy`
`03:25.18`	`sudhir492`	`let me try that now.`
`03:25.27`	`JT`	`i want to be able to detect that, and forcefully disconnect a non emergency call`
`03:25.27`	`JT`	`and then put the call through`
`03:25.28`	`puzzled`	`JT: do you use T1/PRI lines?`
`03:25.28`	`*** join/#asterisk doolph (i=doo@200.46.148.42)`
`03:25.28`	`JT`	`yes`
`03:25.29`	`doolph`	`hi`
`03:25.30`	`Strom_C`	`JT, that can be done easily enough`
`03:25.50`	`puzzled`	`JT: then put one of the channels in its own group and dedicate that to 911`
`03:26.11`	`puzzled`	`Strom_C: you can do that? how?`
`03:26.29`	`sudhir492`	`I always download tar files or binary RPMS. This is the first time I will be building from Source RPMs.`
`03:26.42`	`JT`	`puzzled: i'm not wasting a channel on that`
`03:26.47`	`Strom_C`	`it would probably require some AMI or AGI scripting`
`03:26.55`	`JT`	`puzzled: then what if there is more than 1 emergency call?`
`03:27.22`	`puzzled`	`JT: true. prolly a bit risky to make that a statistic :)`
`03:28.07`	`JunK-Y`	`jt, then dial on 2 groups, 1st ur emergency group, and ur regular group?`
`03:28.38`	`sudhir492`	`JT: first try to call on the channel group, assuming it will go through fine. In case you get chan unavailable, soft hangup one of the channels and call on that line`
`03:29.34`	`sudhir492`	`puzzled: I see a tar file and a bunch of patch file. Do I manually apply all the patch files?`
`03:30.29`	`puzzled`	`sudhir492: no, you install the srpm and build it with rpmbuild. you will need to build the zaptel, and libpri SRPMs first though`
`03:30.54`	`JT`	`sudhir492: sounds similar to what i want to do`
`03:31.23`	`JT`	`except it will need to check a variable to see if the channel it is about to soft hangup is an emergency call first`
`03:31.32`	`JT`	`not sure if i need to use the astdb for that`
`03:31.35`	`sudhir492`	`JT: then do it :-)`
`03:32.10`	`JT`	`i know what i want to do`
`03:32.16`	`JT`	`not how to do it simply`
`03:32.44`	`Strom_C`	`JT, it would be fairly simple`
`03:33.58`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`03:34.18`	`symlink`	`[TK]D-Fender: I just want`
`03:34.42`	`[TK]D-Fender`	`symlink : ! ! !`
`03:34.46`	`symlink`	`:D`
`03:34.49`	`[TK]D-Fender`	`symlink : On only just got in from work.....`
`03:34.57`	`[TK]D-Fender`	`asdaskl;djakl;sjda;sd`
`03:34.58`	`symlink`	`[TK]D-Fender: this late?`
`03:35.04`	`JT`	`Strom_C: can one string of execution in a dialplan check a variable out of another seperate existing call?`
`03:35.16`	`Strom_C`	`JT, you'd have to set an astdb value`
`03:35.43`	`JT`	`ah damn`
`03:35.55`	`[TK]D-Fender`	`symlink : Yup. I made a budget sheet in Excel that I was matching up witha re-run. Thing had 4 "key fields" to create a uniqueid. Problem is the MODIFIED them so everything fucked up. Spent hours rebuilding it by hand.`
`03:36.03`	`JT`	`the persistance annoys me, will need to make sure it's reset even if asterisk crashes`
`03:36.05`	`symlink`	`[TK]D-Fender: yikes`
`03:36.23`	`Strom_C`	`JT, you could use global variables instead`
`03:36.32`	`JT`	`ah good idea`
`03:36.43`	`[TK]D-Fender`	`Strom_C : So you can have collision issues instead ;)`
`03:36.56`	`Strom_C`	`i'm too tired and hungry for thinking`
`03:37.07`	`sudhir492`	`puzzled: what if I want to change one of the source files?`
`03:37.14`	`JT`	`collision issues? i can give them unique names :P`
`03:37.19`	`puzzled`	`sudhir492: have fun :)`
`03:37.52`	`sudhir492`	`:-( :-)`
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`03:39.01`	`sudhir492`	`where is asterisk.spec file? I just installed asterisk SRPS but do not see that in SOURCES directory`
`03:40.33`	`orlock`	`sudhir492: SPECS directory`
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`03:44.02`	`nexis`	`ok, stupid newbie question, what are the requirements to get app_meetme to configure and compile?`
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`03:44.43`	`sudhir492`	`thanks.`
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`04:10.50`	`Supaplex`	`how can I find what directory agi's should be kept in?`
`04:10.59`	`Supaplex`	`debian w/asterisk 1.2`
`04:11.05`	`russellb`	`/var/lib/asterisk/agi-bin by default`
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`04:11.20`	`russellb`	`can be changed in /etc/asterisk.conf`
`04:12.16`	`Supaplex`	`ahh, the /etc/asterisk/asterisk.conf says it's in /usr/share/asterisk/agi-bin.`
`04:12.17`	`WGFreewill`	`anyone know how to define NEWJB to make asterisk 1.2 use the new jitterbuffer`
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`04:46.59`	`marbahlarbs`	`is an asterisk server suitable to replace a phone system with 250+ phones?`
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`04:47.33`	`sudhir492`	`marbahlarbs: Yes, Asterisk is very much capable of handling 250 users. Depends on what kind of hardwae you use and what your usage is going to be`
`04:47.33`	`sudhir492`	`puzzled: are you still there?`
`04:49.13`	`puzzled`	`yup`
`04:50.06`	`sudhir492`	`I had to leave. Came back to building asterisk again. I downloaded SRPS for zaptel, libpri, spandsp and asterisk fromwww.laimbock.com/asterisk/.`
`04:50.43`	`sudhir492`	`zaptel built fine. I used the following command for that:`
`04:51.37`	`sudhir492`	`rpmbuild -ba zaptel.spec --target "$(uname -m)"`
`04:52.02`	`*** join/#asterisk hads (n=hads@mail.nice.net.nz) [NETSPLIT VICTIM]`
`04:52.15`	`sudhir492`	`However, when I try to rpmbuild libpri, I get the error message that zaptel-devel`
`04:52.19`	`sudhir492`	`failed dependency`
`04:52.36`	`sudhir492`	`sorry, wrong key got pressed by mistake.`
`04:52.45`	`puzzled`	`sudhir492: then install the zaptel rpms before building libpri. then install the libpri rpms before building asterisk, etc.`
`04:53.37`	`sudhir492`	`hmm, do you suggest, I install RPM binaries for zaptel, libpri and SRPM for Asterisk and SPANDSP`
`04:54.18`	`puzzled`	`sudhir492: you need the zaptel rpms installed before you can build the libpri srpm. you need the libpri rpms installed before you can build the asterisk srpm`
`04:54.48`	`nexis`	can anyone give me some advice as to where to look at fixing this error, svn trunk checkout pbx.o: In function `extension_match_core':
`04:55.53`	`sudhir492`	`I know that and that is what I am doing. My question was if I should better install binary RPMS for zaptel, zaptel-devel and libpri and then install (and rpmbuild) spandsp and asterisk`
`04:56.13`	`sudhir492`	`SRPMs for asterisk I mean`
`04:56.17`	`puzzled`	`sudhir492: yes. if you don't you can not build the asterisk srpm`
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`05:02.42`	`JT`	`hmm, now here's one`
`05:03.04`	`JT`	`is there anyway to set a variable at the same time as the Dial application runs?`
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`05:14.58`	`olarva`	`Alguem que entenda portugues? ( Sorry...)`
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`05:19.56`	`hoobastooba`	`anyone here using gnudialer?`
`05:25.08`	`JT`	`i have a question regarding variable behavious`
`05:25.14`	`JT`	`${CHANNEL}`
`05:25.30`	`JT`	`will return the value of which channel? :)`
`05:25.43`	`JT`	`as many calls have at least 2 channels involved`
`05:28.14`	`ma-requin`	`hello guys I am behind a nat router and trying to make a call to a public ip address ( my router shares the same ip address with the destination ) can anybody help ??`
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`05:28.53`	`*** mode/#asterisk [+o mog] by ChanServ`
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`05:34.44`	`Corydon76-home`	`JT: the name of the channel running in the dialplan`
`05:35.13`	`Corydon76-home`	`JT: bridging pauses the dialplan`
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`05:35.42`	`JT`	`hrm well say there's an extension "incoming channel" and a trunk "outgoing channel"`
`05:35.53`	`JT`	`i'm after the channel name/number of the outgoing channel`
`05:36.41`	`Corydon76-home`	`When would you try to get it?`
`05:37.06`	`Corydon76-home`	`There's no point when you're bridged that you have execution in the dialplan`
`05:37.07`	`JT`	`eg when dialing via "g3", i would need to grab the specific channel number within g3`
`05:37.18`	`Corydon76-home`	`only before bridging or after bridging`
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`05:37.40`	`JT`	`so i can set a global var regarding that channel`
`05:37.51`	`JT`	`you're saying it's impossible to do anything while bridged?`
`05:38.00`	`Corydon76-home`	`Pretty much`
`05:38.06`	`JT`	`:o`
`05:38.25`	`Corydon76-home`	`unless you're writing C code and you start a separate thread`
`05:38.28`	`JT`	`won't the M(x) Dial option execute a macro`
`05:38.34`	`JT`	`upon channel bridging`
`05:38.49`	`Corydon76-home`	`Yes, in the called channel, just before bridging starts`
`05:39.22`	`JT`	`then the voip-info dial entry is a bit off`
`05:39.24`	`JT`	`M(x): Executes the macro (x) upon connect of the call (i.e. when the called party answers)`
`05:39.51`	`JT`	`it actually doesn't really matter if it's just before bridging`
`05:39.51`	`Corydon76-home`	`Correct... the called party answers, the macro executes, then the caller and called are bridged`
`05:40.11`	`JT`	`what matters is that i know the outgoing channel name`
`05:41.04`	`JT`	`(ideally it'd know the channel name as soon as dialling commences, but i can live without that)`
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`05:43.40`	`JT`	`Corydon76-home: so there's no way to grab the outgoing channel name?`
`05:44.08`	`Corydon76-home`	`JT: well, there's the macro like you pointed out`
`05:44.32`	`Corydon76-home`	`but no, you cannot know (from the dialplan) both channel names at the same time`
`05:44.52`	`JT`	`will the macro get the channel name of the ingoing or the outgoing channel?`
`05:45.03`	`Corydon76-home`	`outgoing`
`05:45.13`	`JT`	`sweet`
`05:45.23`	`JT`	`i'll have to give it a go`
`05:45.39`	`Corydon76-home`	`Does it matter that you won't know the corresponding incoming channel name?`
`05:45.40`	`JT`	`writing up emergency calling macros`
`05:45.51`	`JT`	`nope, not in this case`
`05:46.21`	`JT`	`as i am setting a global var that says the outgoing channel's current state is in an emergency phone call`
`05:46.31`	`JT`	`and the incoming and outgoing chans are in different zap groups`
`05:50.57`	`JT`	`Corydon76-home: just another quick question; what's the best way to determine that all outgoing lines in a zap group are currently in use?`
`05:51.08`	`JT`	`atm i have it checking HANGUPCAUSE is 2 or 3 or 4`
`05:51.15`	`Corydon76-home`	`ChanIsAvail`
`05:51.24`	`JT`	`(trying to use the ast hangup cause)`
`05:51.30`	`JT`	`hmm`
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`05:53.34`	`JT`	`does ChanIsAvail check Zap groups?`
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`05:54.17`	`BosHaus`	`rt`
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`05:55.19`	`kph100`	`hello`
`05:55.45`	`kph100`	`anyone knows what the codes in disposition field of cdr mean?`
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`06:01.02`	`ma-requin`	`~seen Dovid`
`06:01.19`	`jbot`	`dovid <n=dovi5988@85.159.160.207> was last seen on IRC in channel #asterisk, 7h 34m 14s ago, saying: 'icel: read the book'.`
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`06:01.44`	`ma-requin`	`~seen benjk`
`06:01.46`	`jbot`	`benjk is currently on #asterisk (1h 4m 42s), last said: 'maybe they are quite good'.`
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`07:48.09`	`Tond`	`Hi, is there a way to retrive info from MySQL DB by typing a command within extensions.conf file?`
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`08:59.18`	`Sargun`	`What different providers that offer unlimited accounts do people in here use?`
`08:59.25`	`Sargun`	`anyone played with arrivaltel?`
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`09:27.36`	`mattfletcher`	`Does anyone here use the UNISTIM driver?`
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`09:35.00`	`mattfletcher`	`Has anyone here got a Nortel Networks phone to work on their system?`
`09:35.22`	`*** join/#asterisk inspired (n=mikael@85.221.7.59)`
`09:35.39`	`*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)`
`09:35.41`	`monsted`	`throw it away and buy one that doesn't suck :)`
`09:36.16`	`*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)`
`09:37.32`	`mattfletcher`	`"Generic asterisk features (since 1.0.0.0) You can use the following entries in unistim.conf - Call Group : callgroup pickupgroup (untested)"`
`09:37.45`	`mattfletcher`	`The rest of the system works fine, just not call pickup`
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`09:45.33`	`normie`	`does anybody know of any sip-service offering free landline calls to sweden? :)`
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`09:54.56`	`mattfletcher`	`Does anyone have call pickup working on a UNISTIM/Nortel phone? If it can be done, I will keep trying. If not, I will forget it`
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`10:09.28`	`SoftIce`	`hi say I setup an inbound route and forward a call to somebodys desk, if he doesn't answer how do I take control of the call again?`
`10:11.40`	`*** join/#asterisk henrique (n=henrique@201-26-76-82.dsl.telesp.net.br)`
`10:12.25`	`*** join/#asterisk SoftIce (n=newbie@vc-196-207-45-253.3g.vodacom.co.za)`
`10:12.30`	`SoftIce`	`sorry got disc`
`10:14.31`	`monsted`	`SoftIce: in extensions.conf, "Dial(SIP/101,20,Ttr)" means it'll ring for 20 seconds.. you can then do whatever you want on the next line`
`10:14.40`	`*** join/#asterisk xnon (i=xnon@200.8.30.3)`
`10:15.49`	`SoftIce`	`well its rang at my extention right, then i've transfered the call to another desk, but he isn't there I want to take control again without just putting down the phone ;)`
`10:16.38`	`monsted`	`ah`
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`10:18.10`	`*** join/#asterisk LarsKalas (n=Lasse@102.Red-88-3-173.dynamicIP.rima-tde.net)`
`10:20.03`	`LarsKalas`	`Is there any way to do a bitwise &-operation in extensions.conf (The regular x & y is a logical and not bitwise afaik)`
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`10:22.30`	`__freedom__lover`	`hi all, my asterisk has returned 'spawn extension' for all calling. what can be?`
`10:23.40`	`monsted`	`LarsKalas: extensions.conf isn't really marketed as a calculator... what are you trying to accomplish? :)`
`10:24.28`	`LarsKalas`	`I am reading a options int from a mysql database and the options are set by bits...`
`10:29.03`	`shellshark`	`why not just make an AGI with Perl?`
`10:29.11`	`shellshark`	`or <insert your language of choice here>?`
`10:30.21`	`LarsKalas`	`I'll use the Math function, but i was just wondering if there was some standard way...thanks anyway!`
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`10:31.09`	`santibiotico`	`hi`
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`10:41.55`	`santibiotico`	`can anyone help me with chanspy??`
`10:42.41`	`santibiotico`	`plz`
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`10:43.43`	`*** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net)`
`10:43.52`	`hohum`	`hello`
`10:44.03`	`hohum`	`why would an Asterisk box ignore an invite request?`
`10:44.04`	`santibiotico`	`hi`
`10:44.20`	`hohum`	`--- (12 headers 16 lines) ---`
`10:44.20`	`hohum`	`Ignoring this INVITE request`
`10:45.20`	`santibiotico`	`i've configured chanspy on asterisk but when i start the spy process through a phone, i can only hear the conversation if i tal through the phone...any idea??`
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`10:59.52`	`hohum`	`why would an Asterisk box ignore an invite request?`
`10:59.55`	`hohum`	`--- (12 headers 16 lines) ---`
`10:59.58`	`hohum`	`Ignoring this INVITE request`
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`11:01.33`	`*** join/#asterisk \|Johny\| (n=gomesper@bacus.corp.fccn.pt)`
`11:01.51`	`\|Johny\|`	`is anyone using cisco 7970 IP phones?`
`11:05.30`	`jaike`	`show channels`
`11:05.45`	`jaike`	`oops sorry`
`11:06.04`	`\|Johny\|`	`wrong window`
`11:06.05`	`\|Johny\|`	`eheheheh`
`11:06.19`	`*** join/#asterisk propagandhi (n=pclaven@ppp201-114.static.internode.on.net)`
`11:06.24`	`propagandhi`	`good day`
`11:06.41`	`propagandhi`	`does anybody here have any experience with the B410p`
`11:08.07`	`propagandhi`	`or could help anyway???`
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`11:20.13`	`ccesario`	`somebody have any documentation about how to configure planet vip-400 and asterisk?!`
`11:21.10`	`*** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se)`
`11:25.41`	`\|Johny\|`	`utried once the planet VIP`
`11:26.03`	`*** join/#asterisk M_at (n=643744f8@82.152.60.82)`
`11:26.07`	`\|Johny\|`	`you just need to put the username and password`
`11:26.11`	`\|Johny\|`	`use the web interface`
`11:29.12`	`*** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com)`
`11:29.17`	`Stephnie`	`hi`
`11:29.54`	`bXi`	`does somebody know a service which allows me to send a few faxes to a certain number for free ?`
`11:30.08`	`Stephnie`	`how to stop simultaneous sip registration for the same SIP PEER but from different IPs`
`11:30.47`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`11:30.56`	`DarKnesS_WolF`	`dose TDM400P works with PCI 5 v ?`
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`11:31.24`	`\|Johny\|`	`<Stephnie> asterisk doesnt allow 2 registrations`
`11:31.47`	`\|Johny\|`	`when you register the second phone, he overwrites the first one`
`11:31.49`	`M_at`	`bXi: try http://tpc.int`
`11:31.50`	`Stephnie`	`it does...u can register 1 PEER from 2 Different IP Addresses`
`11:31.55`	`Stephnie`	`ohhh...`
`11:32.09`	`M_at`	`bXi: You can send emails to fax machines with that.`
`11:32.13`	`Stephnie`	`but I want to stop the second phone if first is already registered`
`11:32.36`	`Stephnie`	`yes I know it over writes....but how to stop the second phone if first is already registered`
`11:32.55`	`\|Johny\|`	`dont give him the password :P`
`11:33.38`	`Stephnie`	`can I use the same technique with username ? ;)`
`11:36.54`	`*** join/#asterisk zotz (n=zotz@208.196.247.175)`
`11:39.01`	`\|Johny\|`	`but usually the username is known, because its his phone number`
`11:39.35`	`Stephnie`	`ok :)`
`11:39.45`	`Stephnie`	`so how to stop the second registration ?`
`11:40.41`	`M_at`	`Why do you need to stop the registration - can you not have a dial plan which routes calls in order?`
`11:43.11`	`Stephnie`	`M_at> one of my client has registered the SIP PEER from Cisco ATA ... and at the same time he is trying to register it from any Softphone as well..`
`11:43.32`	`M_at`	`I see`
`11:43.42`	`\|Johny\|`	`create him a second account`
`11:43.50`	`Stephnie`	`woooops`
`11:43.56`	`*** join/#asterisk viperdude (n=jon@195.74.96.118)`
`11:44.19`	`M_at`	`Yup - put an a on the end of his username, creating a second account and set the dialplan to route calls appropriately.`
`11:44.21`	`Stephnie`	`is there any professional solution just like broadvoice..`
`11:44.55`	`\|Johny\|`	`what do you mean?`
`11:44.57`	`Stephnie`	`they dont allow second registration of first one is up..`
`11:45.23`	`propagandhi`	`any help here for the B410P card??`
`11:45.25`	`M_at`	`Are they using SIP?`
`11:45.43`	`Stephnie`	`yes....`
`11:46.14`	`M_at`	`They may have tailored the SIP stack to be able to reject subsequent registrations`
`11:46.55`	`Stephnie`	`hmmmm`
`11:47.10`	`Stephnie`	`or may not be using asterisk`
`11:47.19`	`santibiotico`	`i've configured chanspy on asterisk but when i start the spy process through a phone, i can only hear the conversation if i tal through the phone...any idea??`
`11:50.45`	`Dr-Linux\|work`	`hi guys`
`11:51.09`	`Dr-Linux\|work`	`how can i set global variable?`
`11:51.58`	`viperdude`	`DrLinux: http://www.voip-info.org/wiki-Asterisk+variables`
`11:52.19`	`\|Johny\|`	`Stephnie - ussually, voip providers allow you to register with more then 1 phone for the same account`
`11:52.25`	`\|Johny\|`	`thats one of the nice things of SIP`
`11:52.26`	`\|Johny\|`	`:)`
`11:52.28`	`wasim`	`Dr-Linux\|work: \|g`
`11:53.04`	`Dr-Linux\|work`	`wasim: janab main kuch aor pooch raha tha`
`11:53.25`	`Dr-Linux\|work`	`viperdude: yeah i know that but.... lemme explain`
`11:53.55`	`Dr-Linux\|work`	`well, i have 2 variables defined in [global] sections`
`11:54.30`	`Dr-Linux\|work`	`english=/temp/english/`
`11:54.39`	`Dr-Linux\|work`	`spanish=/tmp/spanish/`
`11:55.38`	`Dr-Linux\|work`	`so i want if users press 1 then variable should be set ${english}`
`11:55.59`	`Dr-Linux\|work`	`but if users presses 2 then variable should be "spanish"`
`11:56.45`	`viperdude`	`Dr-Linux\|work: so whats the problem?`
`11:57.28`	`Dr-Linux\|work`	`wasim: i have also put here my questions`
`11:57.35`	`Dr-Linux\|work`	`http://networks.pk/forum/viewtopic.php?t=11`
`11:57.56`	`Dr-Linux\|work`	`viperdude: how can i change that variables`
`11:58.01`	`wasim`	`Dr-Linux\|work: yeah, but whats the issue, thats simple using 1,1,Set(english) 2,1,Set(spanish)`
`11:58.21`	`wasim`	`orsomething of that kind`
`11:58.25`	`propagandhi`	`somebody should stab the guy that invented B410P`
`11:58.30`	`viperdude`	`Dr-Linux\|work: just define extensions for 1 and 2 which as part of the definition set the variable`
`11:58.33`	`propagandhi`	`if he's here send him to me`
`11:59.15`	`Dr-Linux\|work`	`hhm...`
`11:59.32`	`Dr-Linux\|work`	`SetVar(LANG=${SPANISH}) << hows it?`
`11:59.45`	`*** join/#asterisk coppice (n=chatzill@213.143.17.210.dyn.pacific.net.hk)`
`11:59.48`	`viperdude`	`yeah thats it`
`12:00.36`	`Dr-Linux\|work`	`hhm..`
`12:00.37`	`Dr-Linux\|work`	`thanks`
`12:02.03`	`wasim`	`setvar is deprecated, use set, or have they changed it again ?`
`12:02.38`	`RoyK`	`wasim: set is still The Right Way`
`12:02.45`	`RoyK`	`set(wasim=0)`
`12:04.11`	`*** part/#asterisk jaike (n=jaike@125.5.144.90)`
`12:06.07`	`*** join/#asterisk sergee (i=opera@195.94.224.197)`
`12:08.23`	`sergee`	`Can anybody explain me how can i enable MWI in linksys phone (spa941)? i switched "Message Waiting -> yes" in phone's web interface - and now lamp is always on even when i don't have VM, should i change additional options? please help`
`12:12.37`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`12:13.54`	`*** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk)`
`12:13.56`	`mattfletcher`	`I use both voipcheap and the PSTN (via a TDM400P) for my outgoing calls. I want to set things up so that if my net access goes down, calls that were destined for voipcheap fall back to using the PSTN. How can I do this. I thought I could just put another dial statement after the voipcheap one but it doesn't seem to work.`
`12:15.52`	`creativx`	`check ${dialstatus}`
`12:18.00`	`*** join/#asterisk jm\|work (n=jamiem@dilbert.jamiem.com)`
`12:23.06`	`tcseke`	`Hello`
`12:23.39`	`tcseke`	`I have a problem with mapi Originate action`
`12:24.44`	`tcseke`	`exactly it's not sure that the problem is in mapi, i start a call witch Originate but i seems the Dial late, quiet much`
`12:26.20`	`tcseke`	`I see that manager.c caled ast_pbx_outgoing_exten function, and it call __ast_request_and_dial, why could the dialing late?`
`12:27.16`	`sergee`	`i can see from sip debug that SPA921 doesn't send SUBSCRIBE for MWI.... althogh i enabled "Message Waiting"...`
`12:30.58`	`*** part/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`12:38.06`	`RoyK`	`does anyone have an example of how to limit the number of concurrent calls for a SIP peer?`
`12:39.38`	`*** join/#asterisk vlt (n=dm@p54B32ADA.dip0.t-ipconnect.de)`
`12:44.27`	`viperdude`	`RoyK: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+incominglimit`
`12:44.53`	`RoyK`	`viperdude: afaik incominglimit is depricated`
`12:45.30`	`viperdude`	`RoyK: there is an example on that page to do it in extensions.conf or use a AGI`
`12:45.53`	`*** join/#asterisk apardo (n=apardo@87.217.145.229)`
`12:45.55`	`*** join/#asterisk kph100 (n=kph100@206-248-157-22.dsl.teksavvy.com)`
`12:50.26`	`*** join/#asterisk ma-requin (n=youssef@dyn10.tcom.ou.edu)`
`12:50.37`	`*** join/#asterisk Kigh (i=kai@unixuni.org)`
`12:50.44`	`vlt`	`Hello. I use asterisk and VoIP/SIP behind a NAT router that's connected to a 3072/512kbit ADSL line. To improve voice quality I setup Traffic Shaping on the router. For outgoing connections it works fine: Even during full upload or file sharing the voice packets are first and voice on the other end is clean. But for incoming VoIP data i don't get a lossless signal even if i keep other traffic a low as 1/3 of max. I pasted the HTB config for inc`
`12:50.44`	`vlt`	`oming connections here: http://pastebin.ca/249954 -- Can anyone reccommend a working setup?`
`12:50.48`	`Kigh`	`hi all`
`12:51.21`	`ma-requin`	`<PROTECTED>`
`12:52.05`	`vlt`	`When I stop all other downloads the incoming voice is clean.`
`12:52.10`	`sergee`	`RoyK: use 'call-limit' instead of 'incominglimit, see also res_limit.c (if you are using trunk)`
`12:52.41`	`sergee`	`so, what about MWI anybody using MWI with Linksuis/Sipura phones?`
`12:53.20`	`Kigh`	`does somebody know a windows TAPI driver that is 64-bit aware?`
`12:53.43`	`RoyK`	`sergee: 1.2`
`12:53.58`	`RoyK`	`htf is using trunk for anything else but playing?`
`12:54.07`	`sergee`	`RoyK: then use incominglimit`
`12:54.34`	`sergee`	`RoyK: i'm using trunk in production`
`12:54.37`	`RoyK`	`afaicr incominglimit was depricated on 1.2`
`12:55.03`	`RoyK`	`well, I'm not. we have a very busy system of five servers, and we don't tolerate downtime`
`12:57.09`	`sergee`	`RoyK: just checked in sources.. 'incominglimit' and 'call-limit' are both ok with 1.2`
`12:57.17`	`*** join/#asterisk ManxPower (n=manxpowe@71-8-11-73.dhcp.leds.al.charter.com)`
`12:58.17`	`*** join/#asterisk oej_ (n=oej@213.115.215.55)`
`12:58.28`	`*** join/#asterisk alerios (n=alerios@190.24.100.110)`
`12:58.41`	`sergee`	`RoyK: 'incominglimit' is a synonym for 'call-limit' in current 1.2 branch`
`12:59.16`	`sergee`	`RoyK: check chan_sip.c - } else if (!strcasecmp(v->name, "call-limit") \|\| !strcasecmp(v->name, "incominglimit")) {`
`12:59.50`	`*** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il)`
`12:59.56`	`drcode`	`hi all`
`13:00.07`	`drcode`	`any one can dcc me chan_zap.so`
`13:00.09`	`drcode`	`thanx`
`13:00.09`	`sergee`	`MWI? Linksys? Anybody, help! :)`
`13:00.13`	`*** join/#asterisk misc-- (n=misc@203.215.121.39)`
`13:00.31`	`drcode`	`I am using ast. 1.2`
`13:00.42`	`misc--`	`hi.. what is the admin username/password on a default asterisk install for the myslql database? I need to create a database`
`13:00.44`	`cpm`	`asterisk warez`
`13:00.45`	`ManxPower`	`drcode: chan_zap won't usually work between Asterisk versions`
`13:00.52`	`misc--`	`asterisk@home btw`
`13:00.55`	`ManxPower`	`drcode: what is your specific problem?`
`13:01.03`	`drcode`	`I use ubuntu`
`13:01.06`	`*** join/#asterisk Darthclue (n=e054502@fw149.northside.isd.tenet.edu)`
`13:01.07`	`drcode`	`6.06`
`13:01.14`	`ManxPower`	`Yes, that would be a problem`
`13:01.18`	`drcode`	`I insall ast. box with apt-get`
`13:01.23`	`drcode`	`but this module is not there`
`13:01.35`	`drcode`	`I seccess compile zaptel`
`13:01.39`	`drcode`	`it seems to work`
`13:01.40`	`ManxPower`	`drcode: then you need to install the package that has it or build from scratch.`
`13:01.52`	`drcode`	`hmm`
`13:01.52`	`ManxPower`	`zaptel needs to be installed before Asterisk`
`13:02.10`	`drcode`	`k`
`13:02.19`	`drcode`	`then apt-get will put chan_zap.so?`
`13:02.42`	`drcode`	`ManxPower: so how can I fix it with apt-get?`
`13:02.45`	`ManxPower`	`drcode: I don't know`
`13:03.10`	`ManxPower`	`You would have to find out from the person that built the Asterisk package`
`13:03.19`	`drcode`	`i don't know`
`13:03.24`	`drcode`	`its debian packge`
`13:03.31`	`drcode`	`I just use apt-get`
`13:03.49`	`RoyK`	`call-limit seems to work nicely`
`13:04.05`	`drcode`	`if I will take from other ver, it will not work?`
`13:04.24`	`ManxPower`	`drcode: correct, it will not work`
`13:04.30`	`drcode`	`hmm`
`13:04.36`	`drcode`	`so I need to recompile ast?`
`13:04.59`	`Kigh`	`its not so difficult :D`
`13:05.03`	`ManxPower`	`that's the way real people do it.`
`13:05.10`	`drcode`	`k`
`13:05.13`	`Kigh`	`if you plan to use ZAP, consider "bristuff"`
`13:05.20`	`drcode`	`I need to use apt-get install ast-devl?`
`13:05.23`	`ManxPower`	`Asterisk changes so fast at this point that installing from packages is silly`
`13:05.29`	`ManxPower`	`Kigh: why?`
`13:05.35`	`Kigh`	`drcode: goto asterisk.org, download tar.gz`
`13:05.46`	`drcode`	`k`
`13:05.51`	`drcode`	`thanx`
`13:06.02`	`Kigh`	`ManxPower: well. you dont have to but it has greater support for HFC cards and some fixes`
`13:06.34`	`Flusher`	`I have some problems with mISDN : i often have to reload the module and asterisk to make it working again. Is there some fixes pls ?`
`13:06.58`	`Kigh`	`we have lots of problems with misdn too`
`13:07.02`	`ManxPower`	`Flusher: upgrade to 1.2.13`
`13:07.10`	`propagandhi`	`SO DO WE`
`13:07.20`	`Kigh`	`thats why we switched to ZAP`
`13:08.03`	`RoyK`	`hm...... call-limit only works for outbound calls, it seems`
`13:08.05`	`propagandhi`	`can you use a BRI card with zaptel??`
`13:08.07`	`Kigh`	`even though misdn may be the future, its a pain in the ass currently`
`13:08.13`	`RoyK`	`propagandhi: yes, several`
`13:08.14`	`Kigh`	`yes you can`
`13:08.21`	`ManxPower`	`propagandhi: with bristuff as long as it is a supported card.`
`13:08.21`	`propagandhi`	`RoyK: can u use the B410P`
`13:08.40`	`RoyK`	`propagandhi: get an el-cheapo hfc-pci card for $20 and use misdn or bristuff`
`13:08.42`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`13:08.45`	`ManxPower`	`B410P doesn't need bristuff`
`13:08.45`	`RoyK`	`:)`
`13:08.59`	`RoyK`	`what does B410P use ?`
`13:09.05`	`propagandhi`	`ManxPower: no matter what we've done so far we cannot get the B410P working`
`13:09.10`	`ManxPower`	`RoyK: it's a card sold by Digium`
`13:09.16`	`RoyK`	`I know that`
`13:09.19`	`RoyK`	`but what protocols?`
`13:09.20`	`ManxPower`	`propagandhi: Digium support can't get it working?`
`13:09.23`	`RoyK`	`someone told me it's not zap`
`13:09.45`	`propagandhi`	`ManxPower: not sure we cannot contact them yet, but i've read they've been unsuccessful in the past`
`13:10.00`	`propagandhi`	`it uses misdn from all reports`
`13:10.22`	`ManxPower`	`I can't imagine Digium releasing a card that is not Zaptel compatable.`
`13:10.24`	`propagandhi`	`and thus far we have had it so close but no cigar`
`13:10.26`	`Kigh`	`i thought zap is supported by digium?`
`13:10.31`	`ManxPower`	`Kigh: it is.`
`13:10.46`	`Kigh`	`so why not use zap with digium cards :)`
`13:10.48`	`propagandhi`	`I also have a Wildcard TE405P and sangoma E1 and analog cards`
`13:10.50`	`propagandhi`	`they are fine`
`13:10.51`	`Kigh`	`misdn is currently crap`
`13:11.16`	`propagandhi`	`i have not seen any way of getting this card working with zap`
`13:11.36`	`propagandhi`	`we've followed the digium setup instructions to the letter`
`13:12.18`	`propagandhi`	`if anyone can point me to a means of using zap for this card I will be forever in your debt`
`13:12.21`	`Kigh`	`propagandhi: do you use this one? http://ftp.digium.com/pub/telephony/zaptel/b410p/`
`13:12.38`	`propagandhi`	`Kigh: thats the one`
`13:12.41`	`Kigh`	`k`
`13:13.55`	`Kigh`	`does somebody know a working TAPI driver for 64 bit windows?`
`13:14.40`	`ManxPower`	`propagandhi: what version of Zaptel are you using?`
`13:14.49`	`Kigh`	`propagandhi: do you use the zaptel beta? in zaptel 1.4.0 misdn is used for B410P`
`13:15.02`	`Kigh`	`but not in the stable tree, i think`
`13:15.30`	`propagandhi`	`Kigh: no we dont, could u give me an idea on how to go about that`
`13:15.46`	`propagandhi`	`we are using 1.2.9`
`13:15.49`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`13:16.38`	`ManxPower`	`Confirmed, even in 1.4 beta the B410P uses mISDN`
`13:16.44`	`Kigh`	`well, give 1.4.0 beta a try then!`
`13:17.07`	`ManxPower`	`see REAMDE.b410p`
`13:17.13`	`Kigh`	`damn i need a 64 bit tapi but dont find a working one`
`13:17.41`	`RoyK`	`any idea what sort of chipsest the B410 is using?`
`13:19.00`	`propagandhi`	`RoyK: its a beronet chipset i believe`
`13:19.26`	`RoyK`	`Kigh: does anyone know a working 64bit windows?`
`13:21.09`	`Kigh`	`lol`
`13:21.09`	`Kigh`	`well, my customer goes well with 64 bit, until i brought them an asterisk PBX :D`
`13:21.36`	`Kigh`	`err i cant talk`
`13:21.59`	`coppice`	`RoyK: I think it uses the silicon sort`
`13:22.42`	`coppice`	`RoyK: unless the B mean means B cup, when it might be the silicone sort`
`13:27.11`	`ManxPower`	`"B" stands for "Bollocks", which is what people scream when they try to use ISDN BRI with Asterisk. 8-)`
`13:27.46`	`propagandhi`	`B stands for blood which will be that of the designer of the card if i ever catch up with him`
`13:28.15`	`*** join/#asterisk zotz (n=zotz@208.196.247.175)`
`13:28.36`	`mitcheloc`	`hola`
`13:28.49`	`coppice`	`don't all these 4 port BRI cards have the same schematic?`
`13:30.21`	`propagandhi`	`i for one don't know`
`13:30.32`	`propagandhi`	`i do know its a major bitch this particular card`
`13:31.55`	`Strom_C`	`propagandhi, wait thirty minutes and call digium support`
`13:31.59`	`ManxPower`	`propagandhi: as I understand it any BRI with asterisk is a major issue.`
`13:32.28`	`ManxPower`	`Yes. If for no other reason than to punish Digium for releasing a card that is not zaptel compatible`
`13:32.44`	`propagandhi`	`ManxPower: i cant believe they even sell it like this`
`13:33.37`	`propagandhi`	`the TE405P and the Sangoma cards I've not had issues with`
`13:35.05`	`mfroes`	`which packages do i need installed so i can have odbc with asterisk ???`
`13:35.32`	`ManxPower`	`mfroes: odbc-devel I would assume`
`13:42.07`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)`
`13:42.26`	`Kigh`	`ODBomfgC`
`13:42.35`	`*** join/#asterisk requin (n=youssef@dyn10.tcom.ou.edu)`
`13:43.13`	`*** join/#asterisk spr1te (n=spr1te@213.227.193.75)`
`13:46.54`	`*** join/#asterisk fholmes (n=fholmes@cpe-68-201-200-58.houston.res.rr.com)`
`13:48.05`	`mfroes`	`when i tri to connect odbc .. i get this error Nov 16 10:47:41 WARNING[25871]: res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified`
`13:50.30`	`mfroes`	`any ideas?`
`13:51.12`	`RoyK`	`yeah, configure odbc`
`13:51.24`	`M_at`	`ON windows boxes it's usually because you're trying to connect with a DSN rather than a string`
`13:51.31`	`*** part/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net)`
`13:51.33`	`M_at`	`Add the DSN and it would work`
`13:51.33`	`mfroes`	`RoyK: it is configured`
`13:51.46`	`mfroes`	`it already has the DSN`
`13:51.55`	`M_at`	`Check your spelling?`
`13:53.03`	`mfroes`	`M_at: already has checked`
`13:53.36`	`M_at`	`Case sensitive?`
`13:53.40`	`mfroes`	`yeap`
`13:53.44`	`hmmhesays`	`nice nick`
`13:53.47`	`M_at`	`Solar flares?`
`13:53.47`	`mfroes`	`i use all lowercase`
`13:54.05`	`M_at`	`hmmhesays: Whose nick?`
`13:54.27`	`mfroes`	`Solar flares?!?`
`13:55.16`	`Kigh`	`mfroes: paste your DSN pls`
`13:55.35`	`mfroes`	`which files exactly do i need to configure to odbc work ??? res_odbc.conf and cdr_odbc.conf`
`13:55.37`	`mfroes`	`?`
`13:55.49`	`Kigh`	`no res_odbc only`
`13:56.09`	`Kigh`	`cdr is for call event logging to database`
`13:57.05`	`mfroes`	`http://www.rafb.net/paste/results/lBn7mc35.html`
`13:57.11`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.136)`
`13:57.28`	`M_at`	`Where have you configured the DSN named asterisk?`
`13:58.47`	`*** join/#asterisk quelo (n=andros@host45-186-dynamic.10-87-r.retail.telecomitalia.it)`
`13:59.57`	`mfroes`	`cdr_odbc.conf`
`14:00.04`	`M_at`	`No - in unixODBC`
`14:00.14`	`M_at`	`Not in Asterisk - that's configured correctly`
`14:00.34`	`M_at`	`You need to configure unixODBC with a DSN to tell it that "asterisk" means connect to this database on this server`
`14:00.41`	`quelo`	`Hi to all I'm trying to compile asterisk-addons 1.4 from branches on my Debian/GNU sarge but it returns this error`
`14:00.47`	`quelo`	`http://paste.debian.net/16751`
`14:01.22`	`quelo`	`How can I solve it?`
`14:01.35`	`*** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch)`
`14:01.41`	`M_at`	`mfroes: You need to setup odbc.ini to contain an asterisk entry`
`14:02.00`	`M_at`	`mfroes: See http://www.unixodbc.org/odbcinst.html for some specifics`
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`14:09.09`	`sergee`	`unbeliveable! i thought that everybody are using MWI...`
`14:09.27`	`martin5519`	`To FreeBSD or not to FreeBSD with Asterisk and Zaptel`
`14:09.40`	`martin5519`	`Linux or FreeBSD... that is the question`
`14:09.46`	`quelo`	`any idea please?!?`
`14:09.55`	`M_at`	`sergee - I use it on a box with FreePBX installed with an SPA942 and it "just works"`
`14:11.25`	`martin5519`	`quelo... saw your paste... couldn't tell you - all I know is on BSD, I never run into any issues...`
`14:11.34`	`martin5519`	`like the one you just ran into`
`14:11.58`	`Katty`	`g'morning!`
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`14:15.04`	`Kigh`	`moin`
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`14:22.42`	`mfroes`	`when i try to connect via asterisk i get a login incorrect. "res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=20014 [unixODBC][FreeTDS][SQL Server]Login incorrect"`
`14:22.51`	`mfroes`	`but when i connect via isql . .it connects.... perfectly`
`14:22.57`	`mfroes`	`anyone knows why ?`
`14:23.28`	`Kigh`	`did you set the login information correctly in res_odbc.comf ?`
`14:25.01`	`mfroes`	`Kigh: yes`
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`14:25.22`	`Kigh`	`cant believe that :D`
`14:25.38`	`Kigh`	`other possible thing is asterisk uses another DSN as iSQL`
`14:26.04`	`Kigh`	`either your DSN isnt correct or you have a typo or what in the login information`
`14:26.04`	`M_at`	`ON MS SQL you sometimes need the user to be entered at the server level before ODBC logins can take place - isql talks directly to the server and bypasses ODBC`
`14:26.24`	`Kigh`	`LOL`
`14:26.41`	`sergee`	`M_at: can you tell me what did you enable in your SPA's config?`
`14:27.38`	`mfroes`	`it has exactly the same password and login that i use on isql`
`14:28.11`	`sergee`	`sergee: i have only one option "Message Waiting: yes/no" - i switched it to yes, but sipura doesn't send "SUBSCRIBE" to server....`
`14:28.44`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`14:28.53`	`M_at`	`sergee: Nothing - factory default apart from the login details`
`14:28.59`	`puzzled`	`hi`
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`14:32.43`	`sergee`	`M_at: kill me, but my linksys doesn't send SUBSCRIBES... :(`
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`14:33.33`	`case_`	`hello`
`14:34.07`	`mfroes`	`it already has a user on server level and odbc.. with but it still wont go`
`14:34.12`	`M_at`	`sergee: To be honest I'm not surprised. I had 2dead out of an order of 7, despite Linksys claiming they have a near zero DOA rate`
`14:34.52`	`case_`	`i try to set up some traffic control on my internet connection, but i wonder if i need to set a TOS of "Minimize-Delay" for anything going out of my Asterisk host...`
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`14:35.45`	`M_at`	`mfroes: Try recreating the user on SQL, start at the Logins section of the Management folder in the SQL MMC, then add one to the database`
`14:37.06`	`quelo`	`are you only BSD users?!?`
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`14:38.24`	`sergee`	`i have some expirience runing asterisk under BSD (4.9, 5.3, 6.0)`
`14:38.33`	`sergee`	`it really sucks`
`14:38.51`	`sergee`	`i have to install one more server for asterisk`
`14:39.02`	`sergee`	`no regrets`
`14:39.22`	`sergee`	`BSD is a very special OS`
`14:39.29`	`adamowitz`	`What's the best way to program special greeting messages for infrequently occurring holidays and other unusual events? Googling and browsing voip-info.org gives me the impression that there are at least two ways to do it: AEL and directly in the dialplan. Maybe others too?`
`14:39.40`	`Nugget`	`I'm down to just one BSD asterisk server now. I hate Linux as much as the next guy, but it's just not worth the grief. I use Slackware for my Asterisk installs where it matters.`
`14:41.01`	`sergee`	`you should use FreeBSD for common tasks (DNS, Routing, Webserver, Mailserver..) - but if you want something more, if you want to be on bleeding edge - you should use linux. Why? Because of community`
`14:41.12`	`Nugget`	`I'm not sure I agree wtih that.`
`14:41.33`	`Nugget`	`the linux "community" is great if you mean "skinnable mp3 player" as "something more".`
`14:41.50`	`M_at`	`I thought it was because BSD is more stable, in that it's changes are slower`
`14:41.54`	`Nugget`	`but if "something more" means bleeding edge of routing or traffic shaping, then the linux community sucks wind a bit.`
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`14:42.30`	`sergee`	`Nugget: linux community also great when you mean "zaptel drivers", it is also great when you mean "asterisk timer" etc etc etc :)`
`14:42.39`	`Nugget`	`yup yup. very true.`
`14:42.58`	`Nugget`	`it's unfortunate, but very true.`
`14:43.03`	`sergee`	`so as i said - only common tasks...`
`14:43.10`	`mfroes`	`M_at: now it says it's connected. how can i test it ?`
`14:43.12`	`Nugget`	`I don't agree with that.`
`14:43.32`	`Nugget`	`there are plenty of common tasks where linux isn't really the best choice.`
`14:43.44`	`M_at`	`mfroes: use it? You have to correct schema there?`
`14:44.14`	`Nugget`	`and there are plenty of "bleeding edge" areas where bsd is the better choice`
`14:45.24`	`mfroes`	`M_at: yes`
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`14:54.19`	`MikeB`	`Hi, Any chance of some help with the B410P we are trying to install? Finally got misdn it compile but not when we asterisk -r reload it segfaults, found something on google but this is fixed in the source we have, Help, Trying to avoid call to Digium`
`14:56.49`	`MikeB`	`[chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri))`
`14:56.50`	`MikeB`	`mISDN_close: fid(20) isize(131072) inbuf(0xb7c10008) irp(0xb7c10008) iend(0xb7c10008)`
`14:56.52`	`MikeB`	`Nov 16 14:43:59 ERROR[4124]: chan_misdn.c:4105 load_module: Unable to initialize mISDN`
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`15:02.20`	`b11d`	`morning lads`
`15:02.42`	`M_at`	`Afternoon`
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`15:05.04`	`codefreeze`	`adamowitz: AEL/dialplan is it. There's no built-in apps for holidays right now.`
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`15:14.32`	`b11d`	`time to musk up`
`15:16.12`	`adamowitz`	`codefreeze: thanks for your reply. I notice that TFOT has no mention of AEL. Do you think that AEL is better than GotoIf constructs in the dialplan for implementing this? If so, do you know of some specific AEL documentation (other than voip-info.org, I mean)?`
`15:16.56`	`codefreeze`	`TFOT?`
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`15:17.04`	`ManxPower`	`~book`
`15:17.05`	`jbot`	`well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`15:17.23`	`codefreeze`	`Ahhhh... of course.`
`15:17.45`	`fourcheeze`	`anyone recommend any asterisk conferences/jollies in europe?`
`15:17.45`	`adamowitz`	`sorry... i should've explained...`
`15:18.24`	`fourcheeze`	`or general voip ones`
`15:18.38`	`fourcheeze`	`preferably ones with lots of mugs/mousemats`
`15:18.45`	`codefreeze`	`adamowitz: Be not deceived... I may *seem* biased about AEL, but I'm not. You should always use AEL for dialplan programming.`
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`15:19.25`	`adamowitz`	`codefreeze: that seems to be the question "why?"`
`15:19.35`	`adamowitz`	`s/be/beg`
`15:19.37`	`codefreeze`	`TFOT probably doesn't have a chapter on AEL (yet) because it is new.`
`15:20.13`	`codefreeze`	`adamowitz: Why? Got some time? (get out your pillows, everyone!)`
`15:20.51`	`ManxPower`	`Many people also did not consider AEL to be "production quality" in 1.2. It is supposed to be stable in the not yet released 1.4`
`15:21.32`	`fourcheeze`	`codefreeze: carry on I'm listening`
`15:22.04`	`codefreeze`	`adamowitz: ManxPower is correct. It's new, and the 1.4 version of AEL is a rewrite (complete) of the 1.2 version.`
`15:23.13`	`codefreeze`	`adamowitz: the 1.2 version was a great start, but has unwritten restrictions, and a lack of warning/error messages that make quite unpleasant to use at times.`
`15:23.50`	`codefreeze`	`Pardon me, I leave unimportant words like "it" out at times. You have to interpolate!`
`15:24.46`	`codefreeze`	`Anyway, for 1.2, AEL2 is available as a sort of backport/patch. AEL2 ended up replacing AEL in 1.4 and up.`
`15:25.22`	`codefreeze`	`So, now that history is done, I progress to the "WHY use it? We already got "extensions.conf"????" part.`
`15:27.23`	`codefreeze`	`1. Clarity. Write a fairly intricate set of conditionals nested in conditionals and tell me with a straight face that its easier to understand in extensions.conf, and I'll nominate you to the head of the poker league.`
`15:28.47`	`adamowitz`	`2.?`
`15:28.56`	`codefreeze`	`2. Quality. The AEL compiler has a ton of checks for simple/subtle problems you might have. extensions.conf (at the moment) has no such checks.`
`15:29.08`	`codefreeze`	`(patience, I'm a slow typer!)`
`15:29.13`	`adamowitz`	`:)`
`15:30.53`	`codefreeze`	`3. Workflow. AEL provides aelparse, a stand-alone compiler that allows you to check your code before you toss it to an asterisk server. It also allows you to dump your AEL in extensions.conf format, so you can feed it to 1.2 (or lower) servers.`
`15:31.16`	`adamowitz`	`nice.`
`15:32.28`	`codefreeze`	`4. Cost. The bottom line, for any programming language, and dialplans ARE programming, is that if you can reduce your error count earlier in the programming cycle, you save $$$$.`
`15:32.58`	`codefreeze`	`(Wasn't it Einstein that proved that time=$$$?)`
`15:33.05`	`adamowitz`	`:)`
`15:33.11`	`adamowitz`	`something like that...`
`15:33.36`	`suma`	`hi i want to use voicemail contexts from mysql database and send my voicemail to my email address`
`15:34.12`	`codefreeze`	`I think I've made a case. Maybe I forgot something. That's the trouble with forgetting.`
`15:34.20`	`adamowitz`	`thanks codefreeze. makes sense.`
`15:34.33`	`suma`	`i have created tables and working fine. i dunno how to take voicemail and send it to my email`
`15:34.39`	`suma`	`can anyone please help me ?`
`15:35.03`	`*** part/#asterisk case_ (n=case@pdpc/supporter/student/case-)`
`15:35.21`	`codefreeze`	`suma: once it's in, it's in. you have to stick an email addr in the voicemail.conf file for your mbox.`
`15:35.29`	`fourcheeze`	`codefreeze: do you know of an AEL tutorial somewhere?`
`15:35.40`	`fourcheeze`	`one with some good examples`
`15:36.07`	`codefreeze`	`fourcheeze: I gave a presentation at Astricon. They said they'd post the slides on their web site.`
`15:36.25`	`adamowitz`	`I'm very slow to upgrade * servers though. I've been bitten when trying to stay current, so now my procedure is something along the lines of: "if it ain't broke (for me), don't upgrade it..."`
`15:36.34`	`adamowitz`	`I'm still running 1.2.9.1, and after doing a cursory browse of the source, it looks like I'd need the AEL2 patch if I wanted to use it. Is that patch stable?`
`15:36.36`	`suma`	`codefreeze: i have configured in mysql database, http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail as mentioned here`
`15:37.13`	`codefreeze`	`suma: I assume you have your email addr in the proper spot in the db?`
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`15:37.33`	`suma`	`codefreeze, i have configured a context and a mailbox number in mysql, but when i call the number it asks for password`
`15:38.08`	`suma`	`codefreeze, VoiceMailMain(su68299930@starhubtrunksvoicemail)`
`15:38.13`	`suma`	`this is how i mentioned there`
`15:38.32`	`oej_`	`RoyK we still have call limits. THey're not going away.`
`15:40.04`	`codefreeze`	`suma: I assume you are not expecting to be asked a password. Therefore, I assume you left the password field blank. Did you try hitting "#" for the password?`
`15:40.24`	`sergee`	`oej_: hi!`
`15:40.24`	`suma`	`codefreeze, the password field is zero`
`15:40.35`	`oej_`	`sergee: Hi`
`15:40.40`	`sergee`	`oej_: Olle, MWI looks broken,`
`15:40.46`	`suma`	`codefreeze, my intention was to record a voicemail and send it to my email, not asking for password`
`15:40.51`	`sergee`	`=))`
`15:40.53`	`codefreeze`	`suma: null, as in empty, or the single digit 0?`
`15:41.04`	`suma`	`codefreeze, it is with digit 0`
`15:41.34`	`sergee`	`oej_: i'll fix it this evening or tomorrow morning`
`15:41.39`	`codefreeze`	`suma: wait a minute, You want to record a voicemail? Then you need to use voicemail(), not voicemailMain().`
`15:41.55`	`sergee`	`oej_: asterisk doesn't send NOTIFY when i delete message`
`15:42.26`	`suma`	`codefreeze, i c`
`15:42.50`	`oej_`	`sergee: delete what message?`
`15:42.56`	`codefreeze`	`suma: voicemailmain() is for listening/deleting/etc. voicemails.`
`15:43.33`	`sergee`	`sergee: MWI... Voicemail message`
`15:43.46`	`sergee`	`oej_: MWI... Voicemail message`
`15:43.55`	`sergee`	`omg, i'm talking to myself now`
`15:43.59`	`ManxPower`	`Message Waiting Indicator`
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`15:44.41`	`ManxPower`	`sergee: VoicemailMain/Voicemail does not change the status of MWI`
`15:44.53`	`ManxPower`	`Since messages could be added/removed outside of those applications`
`15:45.01`	`sergee`	`oej_: when i recieve new voicemail, asterisk sends me NOTIFY, but when i listen it or delete it, asterisk doesn't send me anything`
`15:45.27`	`ManxPower`	`sergee: how long are you waiting?`
`15:45.37`	`suma`	`codefreeze, works now, thanks`
`15:45.50`	`oej_`	`sergoej: you know that the mwi has no direct connection to the voicemail storage?`
`15:45.54`	`sergee`	`ManxPower: i think you are wrong, because when i recieve new message with Voicemail() asterisk generates SIP NOTIFY instantly`
`15:46.04`	`oej_`	`oejserge: So when you delete, it will take time until another MWI comes`
`15:46.11`	`adamowitz`	`codefreeze: was your Astricon presentation on AEL for 2006 or another year? do you know which one at http://www.astricon.net/files/usa06/ it is?`
`15:46.48`	`codefreeze`	`adamowitz: I haven't gone back to check yet. Hang on a minute.`
`15:48.18`	`BurtyB`	`is there something different with IAX2 vs SIP regarding music on hold? I've just moved a phone over to IAX2 and hold music doesnt seem to play anymore - I see a MusicOnHold("IAX2/desk0-6", "") when it doesnt work and a MusicOnHold("SIP/soft0-091bc1d0", "") when it does`
`15:49.03`	`BurtyB`	`(they are different phones but the IAX2 one works ok for in/out calls its just hold music)`
`15:49.05`	`sergee`	`ManxPower: ofcause voicemail() changes mwi state - take a look into source code http://svn.digium.com/view/asterisk/trunk/apps/app_voicemail.c?view=markup`
`15:49.23`	`MikeB`	`Any idea how to stop udev hotpluging misdn modules?`
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`15:51.57`	`codefreeze`	`adamowitz: nope, it's not there! Hmmm! Let me see if I can put it on my server until Steve Sokol can put it on his.`
`15:52.02`	`MikeB`	`on reboot it seems to be loading the hfcmulti module but wrongly, and we then need misdn-init restart to fix it.`
`15:52.50`	`adamowitz`	`Thanks codefreeze.`
`15:52.54`	`adamowitz`	`Does anyone know whether the 1.2.9.1 module pbx_ael.so is AEL2 or the older AEL?`
`15:53.11`	`symlink`	`older AEL`
`15:53.19`	`M_at`	`Not trying to be cheeky but is there a timeframe for release 1.4? Doing some planning here and want to know wether it's worth playing with 1.4 now or waiting a while til the new year.`
`15:53.33`	`adamowitz`	`thanks symlink`
`15:54.03`	`sergee`	`ManxPower: sorry man, looks like you was right`
`15:54.24`	`propagandhi`	`benjk: are u there`
`15:55.01`	`sergee`	`ManxPower: looks like Voicemail() only schedules MWI message, and it is sent independently by channel driver - i will dig this stuff a little bit`
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`15:59.14`	`codefreeze`	`adamowitz: symlink is correct. AEL2 is available via a svn branch; you diff it to produce a patch.`
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`15:59.40`	`adamowitz`	`Thanks codefreeze. I was just reading the wiki page on AEL2.`
`15:59.55`	`codefreeze`	`adamowitz: Try downloading http://www.parsetree.com/Astricon-AEL-presentation.odp`
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`16:00.33`	`adamowitz`	`got it.`
`16:00.35`	`codefreeze`	`adamowitz: And the wiki AEL2 is the most up-to-date reference on AEL. I use it all the time.`
`16:00.37`	`adamowitz`	`thanks codefreeze.`
`16:02.47`	`codefreeze`	`M_at: Another beta coming soon. We need testers! Testers!! Break it, complain, file bug reports!!!`
`16:02.50`	`suma`	`codefreeze, how can i change the greeting for every voicemail inbox?`
`16:03.08`	`kph100`	`k`
`16:03.26`	`M_at`	`codefreeze: I will when I have the chance - for now I have an echoey BRI based system in Sweden and a PRI based system to deploy in Miami`
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`16:04.26`	`codefreeze`	`suma: within the confines of comedia voicemail, which is what the module calls itself, you can have every user record their own little greeting.`
`16:04.55`	`codefreeze`	`suma: actually, set of greetings, like busy, etc.`
`16:05.27`	`suma`	`codefreeze, yes, is there anywhere i can set the voicefiles ?`
`16:06.00`	`adamowitz`	`I've only glanced at it, but it looks like a very comprehensive presentation, codefreeze. Plus, now I understand why you said what you said initially (about always using AEL2). Looks like I've got quite a bit of work here learning AEL2. Thanks very kindly for the comments and other help.`
`16:06.13`	`suma`	`codefreeze, how can i make a customer to record their own greeting ?`
`16:06.15`	`codefreeze`	`Well, I think everything gets stored down under /var/spool/asterisk/voicemail, or somesuch place.`
`16:06.41`	`*** join/#asterisk RoyK (n=roy@ti211310a080-3120.bb.online.no)`
`16:06.52`	`codefreeze`	`suma: get your users/customers to go to voicemailmain() and there's options down in there.`
`16:07.23`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`16:08.05`	`codefreeze`	`adamowitz: It may appear an imposing chunk now, but it is pretty simple. No fancy objects, etc.`
`16:08.34`	`kph100`	`anyone knows how i can convert the codes in the disposition field of cdr odbc?`
`16:08.52`	`adamowitz`	`I'm certain you're right, codefreeze. I look forward to learning how to use it! :)`
`16:12.55`	`codefreeze`	`oej_: murphy here! Did I commit any attrocities in 7433? It was a tough homework assignment!`
`16:13.09`	`Qwell[]`	`codefreeze: You...changed? :p`
`16:13.27`	`oej_`	`codefreeze: Aha, it's you!`
`16:13.43`	`oej_`	`codefreeze: Can you move the branch to the group directory, please`
`16:13.45`	`*** join/#asterisk merbanan (n=Anders@136.240.13.217.in-addr.dgcsystems.net)`
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`16:14.19`	`codefreeze`	`oej_: Uh, .... ok.`
`16:14.45`	`codefreeze`	`oej_: I think I know how to do that.`
`16:15.01`	`Qwell[]`	`codefreeze: should be just an svn mv on the repositories`
`16:15.03`	`oej_`	`codefreeze: Thanks! Good luck! I can't do like russell and just hijack branches`
`16:15.16`	`*** join/#asterisk merbanan (n=Anders@136.240.13.217.in-addr.dgcsystems.net)`
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`16:15.44`	`codefreeze`	`oej_: I will go and do it right now. Qwell[]: txs!`
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`16:17.31`	`oej_`	`codefreeze: It's a messy area where the SIP notify user interfaces (phones, softphones) require on thing to look right, and the queue system have other requirements`
`16:17.53`	`oej_`	`codefreeze: I need to check with BJ to make sure things work right`
`16:18.03`	`af_`	`ok`
`16:18.04`	`sahafeez`	`hey, suck on a console w/out browser. can some remember me the default username and passwd for a polycom 501 phone - web interface.`
`16:18.18`	`Strom_C`	`admin:456 IIRC`
`16:18.29`	`*** join/#asterisk Chris-NB (n=chris@87.247.210.19)`
`16:19.52`	`codefreeze`	`oej_: done in 47741, the bug7433 dir is now in the group dir.`
`16:19.59`	`*** join/#asterisk alamantia (i=alamanti@nat/digium/x-fa9f8ea9c94fdeaf)`
`16:20.53`	`codefreeze`	`oej_: that is, https://origsvn.digium.com/svn/asterisk/team/group/bug7433`
`16:22.28`	`*** join/#asterisk k31th (n=keith@cartman.nzsolutions.net)`
`16:22.47`	`k31th`	`yo. anyone recomend a decent hardware phone for asterisk ?`
`16:23.01`	`Qwell[]`	`k31th: The polycom 430's look good`
`16:23.07`	`k31th`	`Are the Linksys phones any good?`
`16:23.18`	`Qwell[]`	`They're decent, I hear`
`16:23.36`	`k31th`	`Qwell[]: its just a "first phone" i want some thing that is known to work well.`
`16:23.37`	`puzzled`	`k31th: the 942 is good`
`16:23.45`	`k31th`	`942 ?`
`16:23.50`	`Qwell[]`	`spa942`
`16:23.54`	`puzzled`	`k31th: linksys spa942`
`16:24.45`	`k31th`	`are these linksys phones / cisco phones ? or are they totally differnt ?`
`16:24.53`	`k31th`	`iirc they kinda look simular ?`
`16:24.58`	`Qwell[]`	`They aren't cisco, no`
`16:26.56`	`mitcheloc`	`Qwell[]: pokeeee!`
`16:27.02`	`*** join/#asterisk Pumas (n=KAos@207.248.51.107)`
`16:28.38`	`*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)`
`16:28.39`	`k31th`	`Do cisco phones work well with asterisk ?`
`16:28.52`	`Qwell[]`	`k31th: Cisco phones don't work well with Call Manager`
`16:29.02`	`Qwell[]`	`at least, not SIP`
`16:29.13`	`Qwell[]`	`(disclaimer, I dislike SIP on Cisco)`
`16:29.22`	`k31th`	`Qwell[]: the linksys phones they support sip ?`
`16:29.28`	`Zeeek`	`Which phone k31th ?`
`16:29.29`	`Qwell[]`	`sure, but linksys != cisco`
`16:30.25`	`k31th`	`tbh iv heard bad things about cisco and sip before.`
`16:30.45`	`Pumas`	`hi, everybody, I have a little problem with asterisk I posted my problem in http://pastebin.ca/249663`
`16:30.51`	`Pumas`	`please help me`
`16:31.44`	`Pumas`	`I dont what to do`
`16:31.51`	`k31th`	`know*`
`16:31.55`	`Pumas`	`I dont know what to do`
`16:32.18`	`k31th`	`Zeeek: looking at the SPA941 and the SPA942 http://www.voiptalk.org/products/Linksys+SPA+942+Business+IP+Phone`
`16:32.23`	`k31th`	`is that what you use?`
`16:32.40`	`Zeeek`	`I have a 942 (I think it is)`
`16:32.55`	`Zeeek`	`I love that phone`
`16:33.04`	`carrar`	`linksys and cisco 7900 work fine`
`16:33.04`	`Zeeek`	`Use it as my main phone at the office`
`16:33.29`	`Zeeek`	`The 942 boots in about 11 seconds, too`
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`16:34.26`	`Qwell[]`	`carrar: I wouldn't say all 7900's work fine`
`16:34.35`	`Qwell[]`	`s/fine/at all/, in some cases`
`16:34.46`	`carrar`	`I would`
`16:35.02`	`Zeeek`	`yeah that's the one k31th very good phone for the price`
`16:35.08`	`Qwell[]`	`carrar: So, you're saying that a 7985 works?`
`16:35.10`	`carrar`	`well for sure the 7940 & 7960's`
`16:35.15`	`BurtyB`	`k31th we have 3 linksys spa901 phones and they seem to work well`
`16:35.54`	`Qwell[]`	`carrar: That's hardly "all 7900's" :)`
`16:35.57`	`Qwell[]`	`carrar: which was my point`
`16:35.57`	`carrar`	`heh`
`16:36.00`	`carrar`	`in my world it is :)`
`16:36.20`	`carrar`	`ALL = 2`
`16:36.21`	`Qwell[]`	`carrar: might as well say "All SIP phones work"`
`16:36.22`	`carrar`	`heh`
`16:36.32`	`Qwell[]`	`but, that isn't the case, because if the vendor is a complete moron...`
`16:36.38`	`carrar`	`All SIP phones work!!!`
`16:37.11`	`carrar`	`They just might not work with Asterisk`
`16:37.20`	`*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)`
`16:38.27`	`k31th`	`Which phone then this http://www.pcwb.com/catalogue/item/CISIPT35?speedtrapid=kelkoo&lead=kelkoo&tduid=6c39148fd5345b640356c069a790cae9 or http://www.voiptalk.org/products/Linksys+SPA+942+Business+IP+Phone`
`16:39.41`	`carrar`	`the 941 & 942's work fine`
`16:40.17`	`carrar`	`even the SPA-841`
`16:41.02`	`k31th`	`carrar: they have caller ID?`
`16:41.38`	`carrar`	`yes`
`16:42.06`	`carrar`	`i think most sip phones that have a display have that`
`16:42.12`	`k31th`	`Dual RJ-45 10BaseT Ethernet Ports -- ??? surly there 100baset ?`
`16:42.15`	`carrar`	`I haven't worked with any that didn't`
`16:42.32`	`carrar`	`no they are 10`
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`16:43.27`	`k31th`	`is that stanard for phones?`
`16:43.27`	`carrar`	`Cisco 7940/7960 are 100`
`16:43.34`	`carrar`	`not really`
`16:43.44`	`carrar`	`Polycom's are 100 I believe`
`16:43.53`	`carrar`	`I would have to double check`
`16:44.17`	`k31th`	`in reality what codec will max out a 10mbp connection ?`
`16:44.25`	`Qwell[]`	`k31th: none`
`16:44.32`	`Pumas`	`I put the output log file about my problem in http://pastebin.ca/250101`
`16:44.35`	`carrar`	`1 phone is not going to max out a 10mb connection`
`16:44.39`	`Pumas`	`help me please`
`16:44.40`	`k31th`	`ok care about that then.`
`16:44.40`	`Qwell[]`	`but, if you have a PC on the switch port...`
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`16:45.05`	`carrar`	`then the PC can max it out`
`16:45.12`	`carrar`	`if you have some super heavy app`
`16:45.46`	`k31th`	`Oh i see, if you plugin your phone then pc to the same line ?`
`16:45.54`	`k31th`	`so it passes thru the phone.`
`16:46.02`	`k31th`	`?`
`16:46.04`	`carrar`	`942 has a extra rj45 on the back`
`16:46.22`	`carrar`	`you can plug a pc in`
`16:46.27`	`k31th`	`I see...`
`16:46.28`	`carrar`	`save on wall jacks`
`16:47.37`	`carrar`	`I the 942 is 100 I believe`
`16:47.41`	`carrar`	`the 941 is 10`
`16:48.14`	`k31th`	`says 10 on the 942`
`16:48.15`	`RoyK`	`http://karlsbakk.net/fun/andgodsaid.jpg`
`16:48.27`	`k31th`	`might be incorrect info.`
`16:49.02`	`*** part/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`16:49.14`	`Pumas`	`I put the output log file about my problem in http://pastebin.ca/250101`
`16:49.32`	`Pumas`	`somebody help me`
`16:49.57`	`*** join/#asterisk Alric (n=nbowyer@avantacom.com)`
`16:51.03`	`*** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)`
`16:52.09`	`*** join/#asterisk [Airwolf] (n=airwolf@84.241.201.111)`
`16:53.17`	`k31th`	`humm im stuck between a cisco 940g and a linksys 942`
`16:53.22`	`*** join/#asterisk arkonadev (n=julian@65.203.186.131)`
`16:53.54`	`arkonadev`	`hey whatsup...so I am messing around with queue events and I noticed that you can get the number of abandoned calls per queue does anyone know if you can figure out how many abandoned calls per agent?`
`16:54.21`	`*** join/#asterisk SwK[Work] (n=SwK@br0.asteriasgi.com)`
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`16:56.47`	`*** join/#asterisk svenna_ (n=svenna@p548D3C7F.dip0.t-ipconnect.de)`
`16:57.48`	`marbahlarbs`	`I'm adding an asterisk server to my nortel option 11 switch. Do I need FXO and FXS ports on the asterisk?`
`16:58.41`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
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`16:59.49`	`monsted`	`marbahlarbs: FXO, if asterisk is to play a phone towards the nortel box`
`17:00.49`	`marbahlarbs`	`ok. I wasn't sure if each trunk would require an fxo and fxs`
`17:01.38`	`*** join/#asterisk Chris-NB (n=chris@87.247.210.19)`
`17:02.39`	`*** join/#asterisk badcfe_ (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr)`
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`17:11.32`	`*** mode/#asterisk [+o Qwell[]] by ChanServ`
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`17:12.28`	`marbahlarbs`	`another question: is there a low cost/free voip provider that I can connect my asterisk server to on the internet and make some test calls?`
`17:12.53`	`aydiosmio`	`yes, it's call "Don't be a Penny Pincher, Jack."`
`17:12.58`	`aydiosmio`	`uh, Inc.`
`17:13.13`	`marbahlarbs`	`very helpful`
`17:13.17`	`aydiosmio`	`you're welcome.`
`17:14.28`	`marbahlarbs`	`I did include low cost as an option`
`17:14.59`	`BurtyB`	`i think voipjet give you a few $ to make test calls with`
`17:14.59`	`marbahlarbs`	`really its just to show management that voip and an open source pbx isn't voodoo and can indeed route calls`
`17:15.17`	`Strom_C`	`Bjorn's Authentic Tacos and VoIP`
`17:15.17`	`marbahlarbs`	`awesome, thanks`
`17:15.17`	`Strom_C`	`, Inc.`
`17:15.21`	`Qwell[]`	`Strom_C: You here?`
`17:15.24`	`Strom_C`	`yes`
`17:15.28`	`Qwell[]`	`fun`
`17:15.30`	`Strom_C`	`I'll toddle over`
`17:15.30`	`Strom_C`	`brb`
`17:15.37`	`file`	`Strom_C: A-ha!`
`17:16.05`	`*** join/#asterisk hyegeek (n=hakimian@aragorn.aha.com)`
`17:16.44`	`liri`	`I've no idea why my fax2mail is failing, there's a small asterisk log here http://pastebin.ca/250125`
`17:18.04`	`file`	`"fax- process.pl"? looks like there is a space`
`17:19.24`	`Pumas`	`hi, everybody, I have a little problem with asterisk I posted my problem in http://pastebin.ca/249663`
`17:19.29`	`liri`	`file: does it look like the "data conversation" between the fax machine and asterisk has ended successfully though`
`17:19.46`	`liri`	`file: I have 2 other fax extensions like this one which are working just fine.`
`17:20.02`	`file`	`seems fine`
`17:20.25`	`file`	`but it does seem as though there is a a space in your script name`
`17:21.10`	`liri`	`file: I see that, yeah.`
`17:21.13`	`liri`	`file: I also checked on the file system and there's no such file /var/spool/asterisk/fax/1163696220.25731.tif`
`17:21.39`	`file`	`some day we will shine like the moon in the morning...`
`17:21.44`	`file`	`like the sun when it's dawning`
`17:22.28`	`*** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net)`
`17:22.37`	`liri`	`file: doesn't this mean the conversation got disconnected? == Spawn extension (from-pstn, fax, 0) exited non-zero on 'Zap/2-1'`
`17:22.58`	`file`	`it means it was hungup, then the 'h' extension which is where your script executes was executed`
`17:23.47`	`liri`	`file: so thats normal...`
`17:24.29`	`liri`	`file: ok lets check that space thing... where is the filename defined?`
`17:24.41`	`file`	`I don't know how your dialplan is written.`
`17:25.02`	`liri`	`file: ok let me do a quick grep throughout /etc/asterisk`
`17:26.16`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`17:26.59`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`17:27.57`	`b11d`	`Pumas.. your problem is with trixbox.. read the channel topic..`
`17:28.07`	`b11d`	`join #freepbx for trixbox shit`
`17:28.23`	`*** join/#asterisk sjobeck (n=sjobeck@pool-71-111-88-209.ptldor.dsl-w.verizon.net)`
`17:28.35`	`Pumas`	`ok thanks b11d`
`17:29.02`	`b11d`	`np man`
`17:32.49`	`liri`	`file: it's defined just fine in the extensions.conf file (for the file fax-process I mean) so it's probably just asterisk that causes that space to appear there...`
`17:33.04`	`*** part/#asterisk hyegeek (n=hakimian@aragorn.aha.com)`
`17:33.09`	`*** join/#asterisk ian_esper (n=chatzill@c9066412.static.spo.virtua.com.br)`
`17:36.49`	`*** join/#asterisk krondorl (n=krondorl@acid.auricnet.ca)`
`17:37.02`	`krondorl`	`allo allo!!`
`17:37.28`	`b11d`	`bonjour!!`
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`17:40.10`	`krondorl`	`Question: Can asterisk be programmed to do call forwarding?`
`17:40.46`	`Strom_C`	`answer: yes`
`17:41.02`	`b11d`	`I dunno.. asterisk is pretty limited in what it can do`
`17:41.04`	`b11d`	`:)`
`17:41.10`	`viperdude`	`krondorl: i do call forwarding configurable fro, mysql`
`17:41.29`	`krondorl`	`cool, are there docs on how to do this?`
`17:41.51`	`viperdude`	`http://www.voip-info.org/wiki/index.php?page=Asterisk`
`17:42.06`	`krondorl`	`thnaks`
`17:42.15`	`*** part/#asterisk propagandhi (n=pclaven@ppp201-114.static.internode.on.net)`
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`17:42.50`	`mfroes`	`do anyone has zaptel kernel modules?`
`17:42.59`	`jets`	`i'm sure most of us does???`
`17:43.07`	`jets`	`what question are you really asking?`
`17:43.20`	`mfroes`	`can someone tell me where i can find ?`
`17:43.26`	`Qwell[]`	`mfroes: asterisk.org`
`17:43.30`	`Qwell[]`	`front page...`
`17:43.35`	`mfroes`	`there's only zaptel`
`17:43.40`	`b11d`	`lol`
`17:43.50`	`mfroes`	`i want kernel modules for zaptel`
`17:44.11`	`Qwell[]`	`mfroes: So download zaptel and then compile and install it`
`17:44.23`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`17:44.23`	`*** mode/#asterisk [+o mog] by ChanServ`
`17:44.40`	`mfroes`	`i have ....`
`17:44.53`	`mfroes`	`but there is no module named zaptel`
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`17:45.29`	`*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`17:45.31`	`mfroes`	`tried to find an rpm for it.. but could not find`
`17:45.41`	`awannabe`	`does anyone know why the snoms show up in "sip show peers", can make/take calls, but the phone doesnt show registered with the little registered icon...anyone know the setting in the html file to force it to register??`
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`17:46.55`	`hoobastooba`	`anyone know when they are shooting for having version 1.4 out of beta`
`17:47.08`	`Qwell[]`	`hoobastooba: as soon as it's ready, and no sooner`
`17:47.34`	`*** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com)`
`17:47.43`	`hoobastooba`	`good answer.`
`17:47.50`	`hoobastooba`	`just wondering how it is going.`
`17:47.57`	`fourcheeze`	`awannabe: did you read all the info on the snom website on getting them working with * ?`
`17:48.12`	`awannabe`	`yeah, im gonna go thru it again, heh`
`17:48.26`	`fourcheeze`	`getting snoms working can sometimes be a bit of a black art`
`17:48.37`	`fourcheeze`	`what kind of network conditions are there?`
`17:48.52`	`awannabe`	`local LAN, heh`
`17:49.42`	`awannabe`	`im using the mass provisoing, so im guessing im mising a setting, just cant find it out`
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`17:51.08`	`awannabe`	`hrmm, it says you must specify the voicemail in sip.conf as well`
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`18:05.02`	`ian_esper`	`have anyone tested xorcomm 32 fxs/fxo? What about quality on USB?`
`18:05.20`	`Qwell[]`	`ian_esper: USB 2 should be plenty to carry 30+ calls`
`18:05.33`	`Qwell[]`	`There are no technical reasons why it wouldn't have good quality`
`18:06.08`	`ian_esper`	`If I need more channels? Is it Ok?`
`18:06.25`	`ian_esper`	`Like 96`
`18:07.42`	`monsted`	`as long as they're on individual usb ports, it should scale close to endlessly`
`18:07.48`	`krondorl`	`is there something in 1.2.12 that has replaced the StripLSD command?`
`18:08.55`	`ian_esper`	`Thanks.`
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`18:10.42`	`Corydon-w`	`krondorl: wtf were you using StripLSD?`
`18:12.17`	`krondorl`	`Corydon-w: I was attempting to use one of the call forwarding examples I found and that's what they used.`
`18:14.53`	`Qwell[]`	`ian_esper: Your best bet, honestly, would be to contact Xorcom and ask`
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`18:21.13`	`ManxPower`	`krondorl: you need to read README.variables`
`18:21.38`	`clyrrad`	`Do you guys have issues with IAX Lines + Sipura Phones / ATA's not being able to send DTMF's into certan IVR / Auto Attendant Systems? For the most part it works, but some systems simply do not get the DTMF tones....`
`18:21.53`	`ManxPower`	`clyrrad: no.`
`18:22.09`	`clyrrad`	`I have DTMF set to INBAND+INFO`
`18:22.18`	`clyrrad`	`ManxPower: how did you get around this issue?`
`18:22.21`	`ManxPower`	`translating DTMF messages into DTMF tones is the job of the VoIP/PSTN Gateway`
`18:22.47`	`clyrrad`	`Any idea how I can fix this?`
`18:22.48`	`ManxPower`	`clyrrad: pick ONE DTMF mode and set that mode on the phone and on Asterisk.`
`18:23.02`	`ManxPower`	`clyrrad: What version of Asterisk is the PSTN Gateway?`
`18:23.20`	`clyrrad`	`that I am not sure - I get my IAX lines from Unlimitel`
`18:23.31`	`ManxPower`	`Then you will need to contact Unlimitel`
`18:23.40`	`krondorl`	`ManxPower: :) hate to say this but I don't have a README.variables file.`
`18:23.41`	`clyrrad`	`to set the DTMF in Asterisk that is done in extensions.conf or iax.conf?`
`18:23.49`	`ManxPower`	`You should confirm your DTMF is being sent to them via iax2 debug of course.`
`18:23.53`	`luke-jr_work`	`how do I get the return value of an application?`
`18:24.06`	`ManxPower`	`clyrrad: for SIP phones it is set in sip.conf`
`18:24.15`	`ManxPower`	`luke-jr_work: you can't from the Dialplan`
`18:24.27`	`clyrrad`	`ManxPower: the DTMF's are being sent, 99% of systems can get the DTMF's only 1% dont get them`
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`18:24.50`	`ManxPower`	`clyrrad: as long as you are sure the DTMFs are being sent then you MUST contact Unlimitel, there is nothing you can do in Asterisk`
`18:25.28`	`clyrrad`	`ManxPower: ok - so set DTMF to the same in the phones as in sip.conf, and be sure that matches what Unlimitel is using correct?`
`18:25.43`	`ManxPower`	`clyrrad: IAX2 only supports out of band dtmf, there is no config option for it.`
`18:26.06`	`luke-jr_work`	`ManxPower, any other way to determine if a fax failed or succeeded? :\`
`18:26.15`	`ManxPower`	`clyrrad: I always use rfc2833 (aka AVT) DTMF for my SIP stuff.`
`18:26.25`	`clyrrad`	`ManxPower: so that makes it easy then, I just set all the DTMF's to outband`
`18:26.41`	`clyrrad`	`ManxPower: I dont have that option in the Sipura phones and ATA's`
`18:26.47`	`ManxPower`	`luke-jr_work: "show application txfax" does not tell you what variables the application sets.`
`18:26.55`	`luke-jr_work`	`ManxPower, you can't exactly stop in-band DTMF... ;p`
`18:27.08`	`ManxPower`	`clyrrad: All my SIPura boxes have an AVT option`
`18:27.19`	`ManxPower`	`luke-jr_work: um, yes you can.`
`18:27.25`	`luke-jr_work`	`ManxPower, variables are set regardless of success or not`
`18:27.44`	`ManxPower`	`luke-jr_work: then you have to take it up with the maintainer of rxfa/txrax`
`18:27.48`	`luke-jr_work`	`ManxPower, not from a protocol standpoint`
`18:27.55`	`clyrrad`	`ManxPower: ok yes I do see AVT, in sip.conf do I set dtmf=AVT or to rfc2833?`
`18:28.12`	`ManxPower`	`clyrrad: on the phone set AVT, in asterisk set rfc2833`
`18:28.14`	`QbY`	`is there a way to falsify the number of callers in a queue? ie, increase it by x`
`18:28.19`	`ManxPower`	`they are different names for the same thing.`
`18:28.28`	`clyrrad`	`ManxPower: Thanks for the info I will give that a shot :)`
`18:28.36`	`ManxPower`	`luke-jr_work: Asterisk will mute the inband tones if it is converting them to out of ban d`
`18:28.44`	`ManxPower`	`most phones will as well`
`18:29.01`	`luke-jr_work`	`and what if they don't?`
`18:30.12`	`ManxPower`	`luke-jr_work: then you are screwed and may get double DTMFs`
`18:30.27`	`clyrrad`	`ManxPower: can I just confirm the syntax is 'dtmf=rfc2833'`
`18:30.28`	`luke-jr_work`	`ManxPower, no, what if they leave it in-band and don't do any processing of it`
`18:30.28`	`luke-jr_work`	`?`
`18:30.40`	`ManxPower`	`clyrrad: read the damn docs. dtmfmode=rfc2833`
`18:30.55`	`clyrrad`	`Thanks :p`
`18:30.57`	`ManxPower`	`luke-jr_work: then that would be inband dtmf.`
`18:31.23`	`ManxPower`	`if if they leave it inband and also send an oob message then you would be getting two dtmf tones, one as audio and one as an oob message.`
`18:31.23`	`luke-jr_work`	`ManxPower, but you just said IAX2 can't support it...O.o`
`18:31.41`	`ManxPower`	`luke-jr_work: when I talk about phones, I mean SIP phones.`
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`18:47.22`	`CunningPike`	`clyrrad: We had to send DTMF to inband on our SPA-3000s to work with some IVRs`
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`18:50.00`	`phedders`	`can anyone help me get automon to work - debian etch, asterisk 1.2.12`
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`18:53.05`	`k31th`	`evening`
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`18:53.59`	`vader--`	`anyone ever link asterisk with nagios in here?`
`18:54.00`	`vader--`	`im looking for a way my nagios server can send something to asterisk so asterisk can call me if there is a problem`
`18:54.20`	`dlynes_laptop`	`CunningPike: I fixed it on those pos's by forcing them to AVT (instead of autodetect), and forcing asterisk to rfc-2833`
`18:54.25`	`CunningPike`	`vader--: We are using a modified nagios plugin with Intermapper`
`18:54.35`	`CunningPike`	`vader--: It just does an IAX ping`
`18:55.02`	`vader--`	`cunning im looking to be notified through asterisk instead of an sms message or email`
`18:55.03`	`CunningPike`	`vader--: With 1.4, we plan to use the SNMP module to provide more monitoring`
`18:55.05`	`vader--`	`like a voicealert`
`18:55.08`	`dlynes_laptop`	`CunningPike: nmap -sU -p 4569 ip.ip.ip.ip works, too :)`
`18:55.10`	`*** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net)`
`18:55.36`	`CunningPike`	`vader--: But if your asterisk is down...........?`
`18:55.40`	`vader--`	`like if my email system goes down i would like to use asterisk to notify me via my cellphone`
`18:55.49`	`vader--`	`if asterisk is down then i should get an email`
`18:55.51`	`vader--`	`about it`
`18:55.54`	`CunningPike`	`vader--: Ah, I see`
`18:56.34`	`vader--`	`my nagios box and asterisk box is two seperate boxes`
`18:56.48`	`vader--`	`i need to figure out how to get nagios to tell asterisk to call me and tell me what's going on`
`18:58.06`	`*** join/#asterisk xnon_ (i=xnon@200.8.30.3)`
`18:58.21`	`hohum`	`why would an Asterisk box ignore an invite request?`
`18:58.23`	`hohum`	`--- (12 headers 16 lines) ---`
`18:58.24`	`dlynes_laptop`	`CunningPike: btw...iax ping/nmap will only work if the asterisk box isn't behind a firewall`
`18:58.26`	`hohum`	`Ignoring this INVITE request`
`18:58.55`	`CunningPike`	`dlynes_laptop: Uh huh........ your point? :)`
`18:59.17`	`dlynes_laptop`	`CunningPike: that is my point :)`
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`19:07.20`	`saftsack`	`hi`
`19:07.34`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`19:08.13`	`saftsack`	`if i do a redirection of a call in this way, it isnt possible to show the source callerid to the called party, isnt it? Caller -> PSTN -> Asterisk -> PSTN -> CalledParty`
`19:09.01`	`Katty`	`allo.`
`19:09.48`	`*** join/#asterisk macli (n=vincent@mcli.brc.ubc.ca)`
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`19:11.12`	`mercestes`	`Katty! Heyas.`
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`19:13.38`	`mercestes`	`I fingered Katty...but I didn't get a response.`
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`19:14.35`	`luke-jr_work`	`Is 750 GB for $235 a decent deal?`
`19:15.48`	`dlynes_laptop`	`luke-jr_: $cdn or $USD?`
`19:15.56`	`*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`19:16.12`	`hoobastooba`	`is there a way to list up where sip extensions are forwarded to?`
`19:16.23`	`krondorl`	`that better be USD...`
`19:16.39`	`dlynes_laptop`	`luke-jr_work: if it's USD, I guess it's a sorta ok deal`
`19:16.40`	`krondorl`	`they are almost $400 can at my local store`
`19:16.52`	`Nugget`	`it's a fantastic deal if you're talking about RAM. :)`
`19:17.02`	`krondorl`	`lol`
`19:17.35`	`dlynes_laptop`	`krondorl: woah...$400? I've seen a special running on them recently here for around the $200 range`
`19:18.09`	`dlynes_laptop`	`krondorl: the same place was also selling 300GB drives in an external drive enclosure for $100 Cdn`
`19:18.30`	`krondorl`	`wow lucky you!!`
`19:18.40`	`dlynes_laptop`	`krondorl: www.futureshop.ca`
`19:18.59`	`CunningPike`	`saftsack: What is your PSTN connection (on the second leg)?`
`19:19.00`	`dlynes_laptop`	`krondorl: they're owned by Best Buy Canada`
`19:19.06`	`*** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`19:19.08`	`krondorl`	`I got my 400gb for $184.`
`19:19.29`	`krondorl`	`Dynes: yup I am aware for that.. :) lol`
`19:19.42`	`dlynes_laptop`	`krondorl: yeah...didn't realize you were Canadian when I said that :0`
`19:20.05`	`krondorl`	`dlynes: np... :)`
`19:20.08`	`luke-jr_work`	`dlynes_laptop, USD, SATA`
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`19:21.56`	`brodiem`	`Can someone tell me what technically qualify does to measure latency? Is it an ICMP, or does it send a SIP OPTIONS and measure response time, or...?`
`19:22.19`	`saftsack`	`CunningPike, BRI`
`19:22.48`	`CunningPike`	`saftsack: Can you set CID on other types of call through the BRI?`
`19:22.53`	`krondorl`	`Gotta love this: FREE... 2 gig ram, 1-72gig drive, 5-32gig drives, 7-18gig drives, 5-9gig drives to be used in a 2-p4 processor server.. ALL that for FREE>>`
`19:23.07`	`brodiem`	`reason i'm asking is a SIP peer I have keeps flapping over 200ms, but running a constant icmp is < 10ms`
`19:23.09`	`dlynes_laptop`	`damn`
`19:23.14`	`dlynes_laptop`	`krondorl: where's that? :)`
`19:23.24`	`*** join/#asterisk Kyler (n=chatzill@74.132.200.229)`
`19:23.27`	`krondorl`	`It was just droped off to my office..`
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`19:23.36`	`dlynes_laptop`	`krondorl: heh...nice`
`19:23.38`	`saftsack`	`CunningPike, i can only set the cid to one of the numbers which i have ordered for my BRI (30 numbers)`
`19:24.02`	`brodiem`	`nvm found my answer... it sends a SIP OPTIONS`
`19:24.06`	`CunningPike`	`saftsack: OK - so, your question again was...?`
`19:24.07`	`saftsack`	`so the conclusion is, that it isnt possible in my opinion`
`19:24.08`	`ManxPower`	`brodiem: qualify= measures the response time to an OPTIONS packet or IAX2 PING packet. It does NOT measure network latency.`
`19:24.18`	`saftsack`	`if i do a redirection`
`19:24.21`	`brodiem`	`ManxPower ty`
`19:24.27`	`krondorl`	`Me gonna have fun!!!`
`19:24.49`	`CunningPike`	`saftsack: Correct - it's to do with restrictions placed by your BRI provider. We have a PRI that lets us set whatever CID we wanat`
`19:24.56`	`CunningPike`	`s/wanat/want/`
`19:25.49`	`luke-jr_work`	`krondorl, I'll take the RAM`
`19:26.15`	`Kyler`	`I was hoping to whip up a conference system today but I don't have a Zaptel device on this system and it's running under Xen (soon to be OpenVZ) so I'd rather not make a dummy driver. I thought I'd use app_conference but I can't reach cvs.sourceforge.net to get the code.`
`19:26.23`	`saftsack`	`CunningPike, ok. i had the same way of thinking but now i know that it was the right way ... so i will ask at my telco now. but with spoofing cids it is possible to do crimes and so on, or? so a telco wont do this in my opinion`
`19:26.59`	`CunningPike`	`saftsack: Depends on the telco`
`19:27.15`	`krondorl`	`Oh and there are more drives and more ram on the way....`
`19:27.29`	`saftsack`	`CunningPike, ok thank you i will try this`
`19:27.34`	`krondorl`	`lukejr: Ummmm, NO.. :)`
`19:27.44`	`Kyler`	`Oh...I can get to app_conference through Subversion. https://svn.sourceforge.net/svnroot/iaxclient/branches/oldtrunk/app_conference/`
`19:27.50`	`krondorl`	`nice try though..`
`19:28.31`	`luke-jr_work`	`krondorl, ...but!`
`19:29.09`	`krondorl`	`Luke: :) I need this machine quite badly right now...`
`19:29.41`	`bluregard`	`when AgentCallbackLogin asks for a new extension, what does it want? I've tried entering just about everything I can think of.`
`19:29.44`	`*** join/#asterisk dasenjo (n=dasenjo@201.228.128.10)`
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`19:31.22`	`dasenjo`	`hi everybody!`
`19:31.40`	`krondorl`	`hello`
`19:31.44`	`bluregard`	`hi`
`19:31.51`	`mercestes`	`Konichiwa`
`19:31.52`	`aydiosmio`	`HI DOCTOR NICK!`
`19:32.01`	`Katty`	`meesha.`
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`20:00.40`	`GarethTheGreat`	`any way to use the soundcard as a channel on OpenBSD?`
`20:01.35`	`ManxPower`	`GarethTheGreat: does OpenBSD have ALSA or OSS support?`
`20:01.51`	`Qwell[]`	`should have oss support`
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`20:02.17`	`*** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net)`
`20:02.32`	`GarethTheGreat`	`does have OSS, but there's no chan_oss module`
`20:02.48`	*** join/#asterisk Math` (n=privmath@bas4-montreal19-1242358545.dsl.bell.ca)
`20:03.02`	`ManxPower`	`GarethTheGreat: perhaps the oss headers were not installed or are in a location different than on Linux`
`20:03.17`	`ManxPower`	`and there is a chan_oss, parhaps it was not built for some reason.`
`20:03.39`	`GarethTheGreat`	`i installed the binary package from ftp.openbsd.org`
`20:03.53`	`GarethTheGreat`	`should i reinstall from ports or from a tarball?`
`20:04.13`	`adamowitz`	`codefreeze: what's the odp stand for in your presentation file? Open Directory Project ?`
`20:04.20`	`adamowitz`	`I'm finding that OpenOffice.org reads it fine, but it's not one of the file formats that OOo can save in (in my binary anyway), so I'm just curious.`
`20:04.21`	`hoobastooba`	`will fax machines work through atas?`
`20:04.25`	`adamowitz`	`None of these ( http://it.acronymfinder.com/af-query.asp?string=exact&acronym=odp&s=r ) seem to fit...`
`20:04.42`	`ManxPower`	`GarethTheGreat: from the source`
`20:04.45`	`*** part/#asterisk propagandhi (n=pclaven@ppp201-114.static.internode.on.net)`
`20:05.07`	`dlynes_laptop`	`ManxPower: that would be from ports in bsd-speak`
`20:06.07`	`dlynes_laptop`	`GarethTheGreat: there's some parameters you need to pass to make when you build the ports though`
`20:06.48`	`dlynes_laptop`	`GarethTheGreat: you'll have to look through your makefile before you build ports to find out what those parameters will be`
`20:06.48`	`dlynes_laptop`	`GarethTheGreat: i.e. to conditionally make the OSS support into asterisk`
`20:07.19`	`GarethTheGreat`	`ok`
`20:07.38`	`CunningPike`	`hoobastooba: Sometimes :)`
`20:07.48`	`dlynes_laptop`	`GarethTheGreat: i have no idea what those parameters are though, as I've never used openbsd, and it's been a while since I touched freebsd`
`20:08.08`	`GarethTheGreat`	`how would you enable it in a normal tarball build?`
`20:08.14`	`GarethTheGreat`	`a ./configure option?`
`20:08.27`	`adamowitz`	`Nevermind: I think I figured it out. ODP ==? OpenDocument Presentation`
`20:08.56`	`dlynes_laptop`	`GarethTheGreat: 1.2, or 1.4?`
`20:10.00`	`*** join/#asterisk alamantia (i=alamanti@nat/digium/x-028b7974b3bb9651)`
`20:10.11`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`20:10.19`	`GarethTheGreat`	`asterisk-1.2.9.1`
`20:10.20`	`*** join/#asterisk Carp1 (n=none@cpe-69-204-153-25.nycap.res.rr.com)`
`20:10.43`	`Carp1`	`Is NuFone still up?`
`20:10.46`	`hoobastooba`	`I still have not figured out how to correct this issue when someone has their phone forwarded, I get: Dropping incompatible voice frame on Local/2@Progrexion_Inbound-a609,2 of format slin since our native format has changed to gsm`
`20:10.49`	`hoobastooba`	`can anyone help me?`
`20:11.25`	`ManxPower`	`hoobastooba: did you use the no release option to chan local? I don't recall the exact format.`
`20:11.26`	`CunningPike`	`hoobastooba: We get that all the time, too`
`20:11.33`	`GarethTheGreat`	`hang on, there's a file with chan_oss commented out`
`20:11.43`	`GarethTheGreat`	`uncommented it and trying to build now`
`20:11.53`	`hoobastooba`	`ManxPower: where would this be?`
`20:12.38`	`hoobastooba`	`CunningPike: when it happens, the caller calling the phone that is forwarded hears awful noise... I think this is related.`
`20:12.54`	`ManxPower`	`hoobastooba: I think it is a /n at the end of the Dial, but check /path/to/src/asterisk/docs/ and see if any files in there talk about chan_local`
`20:13.12`	`CunningPike`	`hoobastooba: We've had no call quality complaints`
`20:13.19`	`ManxPower`	`hoobastooba: It's been a long time since I've needed to use Local`
`20:13.55`	`*** join/#asterisk mtoups (i=toups@HOWARD.INDYPGH.ORG)`
`20:14.14`	`dlynes_laptop`	`GarethTheGreat: cool`
`20:14.31`	`mtoups`	`anybody else seeing excessive gain/clipping on calls recorded with Monitor() ? similar to this: http://bugs.digium.com/view.php?id=5823&nbn=13`
`20:14.33`	`*** join/#asterisk cbm11211 (n=AB@66.28.182.170)`
`20:14.46`	`Carp1`	`Does anyone know about NuFone? It says my login information is wrong`
`20:15.05`	`dlynes_laptop`	`Carp1: have you tried emailing them?`
`20:16.24`	`knarfly`	`what is the syntax to block a call from a certain number...or zapateller them?`
`20:16.51`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`20:17.23`	`ManxPower`	`knarfly: see "ex-girlfriend" option in the Asterisk docs dir, the sample configs (extensions.conf.sample) and the Wiki`
`20:18.12`	`*** join/#asterisk spr1te (n=spr1te@213.227.193.75)`
`20:19.39`	`ManxPower`	`hoobastooba: yes, add a /n at the end of your Dial`
`20:20.15`	`Carp1`	`dlynes_laptop: No because I just noticed it and thought mnaybe you guys would know something.`
`20:20.15`	`Carp1`	`Thanks`
`20:20.28`	`bluregard`	`if I'm using AgentCallbackLogin(\|\|${CALLERID(num)}@internal) how do I log off?`
`20:21.18`	`h3x0r4t0r`	`AgentCallbackLogin is getting depreciated`
`20:21.26`	`Qwell[]`	`~deprecated`
`20:21.28`	`jbot`	`extra, extra, read all about it, deprecated is a typo of depreciated, see depreciated, or not a typo according to the jargon file... oh well here it is "you can use it, but it's no longer the best way and will probably be unsupported soon" since when become typos new words`
`20:21.28`	`*** join/#asterisk sahafeez (n=sahafeez@dsl092-150-211.wdc2.dsl.speakeasy.net)`
`20:21.29`	`Strom_C`	`and also deprecated`
`20:21.36`	`ManxPower`	`bluregard: just press # instead of the nnnn#`
`20:21.36`	`Qwell[]`	`wtf`
`20:21.44`	`Strom_C`	`the bot is backwards`
`20:21.44`	`Qwell[]`	`jbot: You are an idiot`
`20:21.45`	`jbot`	`I think you lost me on that one, Qwell[]`
`20:21.48`	`h3x0r4t0r`	`its getting decapitated`
`20:21.55`	`Strom_C`	`dCAPitated?`
`20:22.03`	`h3x0r4t0r`	`yes`
`20:22.04`	`h3x0r4t0r`	`!`
`20:22.20`	`bluregard`	`ManxPower: I tried that. As soon as the agent # and password is entered it goes right to "Agent Logged In"`
`20:22.34`	`ManxPower`	`bluregard: check the wiki then`
`20:22.37`	`bluregard`	`if I just hit # it gives me "Invalid login"`
`20:23.04`	`ManxPower`	`bluregard: Ah! press # when it prompts for the callback number`
`20:23.11`	`sahafeez`	`hello. i have a strange issue. adding 2 new polycom phones to a working setup. phones boot and all good. i can dail, etc. however i am not getting a sound between them. they are setup 100% the same as the other 10 phones. and configured for allow = all`
`20:23.12`	`sahafeez`	`192.168.120.5 4222 5ed7cafe550 00102/00000 ulaw No Tx: ACK`
`20:23.12`	`sahafeez`	`192.168.120.4 4223 426da4f2-d6 00101/00001 ulaw No Rx: ACK`
`20:23.20`	`sahafeez`	`and they seem to be talking that same?`
`20:23.57`	`bluregard`	`ManxPower: Yeah, that's what I thought, but using CALLERID(num) inserts the callback extension automagically.`
`20:24.16`	`ManxPower`	`bluregard: then use a different extension to run it without the CALLERID stuff`
`20:24.20`	`codefreeze`	`adamowitz: odp == the openoffice equiv to powerpoint.`
`20:25.04`	`scurb`	`Which presentation did you present codefreeze?`
`20:25.23`	`codefreeze`	`scurb: the one on AEL.`
`20:25.25`	`bluregard`	`I'd rather not use the variable but I can't get it to log in without it. When it asks for the extension, I put in my phone's extension but it says it's invalid`
`20:25.28`	`*** join/#asterisk saftsack (n=oliver@p54A7DDD0.dip.t-dialin.net)`
`20:25.58`	`scurb`	`ah, ok :) The one trying to impress with your wife :)?`
`20:26.44`	`codefreeze`	`scurb: guilty as charged!`
`20:26.47`	`Carp1`	`Wow, I'm on the phone with NuFone right now......AWFUL!!!`
`20:26.59`	`Carp1`	`Cant hear the man becuase their service keeps cuttubg out`
`20:27.06`	`scurb`	`:), Good presentation however.`
`20:27.13`	`ManxPower`	`Carp1: are you using PSTN to call or VoIP to call?`
`20:27.25`	`ManxPower`	`Carp1: I never had significant audio issues with NuFone when I used them.`
`20:27.30`	`*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`20:28.12`	`Carp1`	`I am using VoIP.....But I'm sure its not me beucase I've been on the phone all day with no problems.`
`20:28.54`	`*** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg)`
`20:29.01`	`codefreeze`	`scurb: many txs!`
`20:31.28`	`bluregard`	`ManxPower: I misunderstood what you said about the other ext. I got it, thanks`
`20:32.42`	`bluregard`	`can I use the set application in sip.conf or iax.conf?`
`20:33.00`	`*** join/#asterisk Dater (n=dater@adsl-75-38-109-118.dsl.irvnca.sbcglobal.net)`
`20:33.28`	`scurb`	`Anyone know if that sun-guy has released conman?`
`20:34.01`	`ManxPower`	`bluregard: no, but sip.conf at least (maybe iax.conf) suport a Set or SerVar config option that does the same thing.`
`20:34.09`	`ManxPower`	`I think it was first available in 1.2.x`
`20:34.54`	`bluregard`	`That's what I need. I'd like to be able to use names in sip.conf but set their callerID to a number`
`20:35.43`	`ManxPower`	`bluregard: see sip.conf.sample for the format.`
`20:35.48`	`bluregard`	`thanks`
`20:35.58`	`ManxPower`	`I think it is something like SetVar=BOB=Happy`
`20:36.07`	`ManxPower`	`but you need to check the sample config to be sure.`
`20:37.43`	`bluregard`	`ManxPower: yep, setvar=VAR=value thanks`
`20:42.23`	`*** join/#asterisk _DAW (n=_DAW@adsl-156-78-145.msy.bellsouth.net)`
`20:43.08`	`*** join/#asterisk designdream (n=felipe@rrcs-71-40-49-30.sw.biz.rr.com)`
`20:43.18`	`_DAW`	`Hello, can asterisk make good use of dual core and quad core processors?`
`20:44.15`	`bluregard`	`_DAW: I would think so, as long as the kernel you use does. I have mine running on a dual CPU machine and it runs great.`
`20:46.08`	`wasim`	`_DAW: 124 channels of g729 on a dual xeon 3.0 ghz with no issues`
`20:47.02`	`_DAW`	`thanks for the feedback.`
`20:47.36`	`QbY`	`_DAW.. I have Dual Opterons, a huge dialplan, apps, and never counted the channels but a lot.`
`20:55.22`	`*** join/#asterisk macli (n=vincent@mcli.brc.ubc.ca)`
`21:00.57`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`21:06.47`	`bluregard`	`wasim: is that 124 simultaneous channels or total?`
`21:07.49`	`wasim`	`bluregard: simultaneous`
`21:08.29`	`wasim`	`bluregard: err ... whats the difference between simultaneous and total?`
`21:08.56`	`bluregard`	`wow, that's good. I have mine on a dual MP 1800+ that I'm wondering how much it can handle.`
`21:09.10`	`bluregard`	`124 calls at the same time?`
`21:09.19`	`monsted`	`I do 1700 channels on a 300 Mhz`
`21:09.21`	`bluregard`	`or you have 124 available channels`
`21:10.44`	`wasim`	`bluregard: at the same time`
`21:11.33`	`wasim`	`do remember call setup rate also affects load handling capacity`
`21:11.46`	`bluregard`	`wasim: that's good to know. I'm trying to guage how many my server can handle.`
`21:11.49`	`bluregard`	`call setup rate?`
`21:12.05`	`wasim`	`number of new calls being setup per second`
`21:12.10`	`bluregard`	`ok`
`21:12.43`	`wasim`	`bluregard: well, compile something on your box, and run show translation recalc 200`
`21:12.47`	`bluregard`	`yeah, I can see where call setup could be more stressfull than the call itself`
`21:13.00`	`wasim`	`bluregard: that should give you a more realistic figure`
`21:14.12`	`bluregard`	`wasim: which codec should I be looking at?`
`21:14.26`	`*** join/#asterisk jaike (i=jaike@58.69.25.176)`
`21:14.29`	`wasim`	`bluregard: whichever ones you intend to use`
`21:14.36`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`21:14.53`	`bluregard`	`actually right now all I can use is ulaw or alaw, DID provider restriction`
`21:15.13`	`*** join/#asterisk zotz (n=zotz@208.196.247.175)`
`21:20.39`	`*** part/#asterisk jaike (i=jaike@58.69.25.176)`
`21:24.19`	`*** part/#asterisk yassine (n=yassine@xdsl-87-78-115-211.netcologne.de)`
`21:26.34`	`*** join/#asterisk klictel (n=klictel@207.107.208.137)`
`21:27.11`	`CunningPike`	`Does anyone else find the sounds in 1.4 are`
`21:27.13`	`CunningPike`	`really`
`21:27.15`	`CunningPike`	`spaced`
`21:27.17`	`CunningPike`	`far`
`21:27.21`	`CunningPike`	`apart?`
`21:27.33`	`CunningPike`	`Much more thanin 1.2?`
`21:27.41`	`CunningPike`	`s/thanin/than in/`
`21:28.04`	Math`	`they are`
`21:28.18`	Math`	`just voicemail gives you a nice example`
`21:28.31`	`CunningPike`	`Right - I think I'll go back to our old sounds`
`21:28.59`	`CunningPike`	`We got the ULAW ones that were posted to the list a while back and we're happy with them`
`21:37.05`	`*** join/#asterisk Pumas (n=KAos@207.248.51.107)`
`21:37.25`	`ManxPower`	`the 1.4 sounds are supposed to be fixed in SVN`
`21:37.49`	`ManxPower`	`I don't know the value of "fixed", however.`
`21:37.59`	`CunningPike`	`ManxPower: OK - good news. That'll make it into 1.4, then?`
`21:39.25`	`Pumas`	`I have a VOIP number with SIP protocol How can I solve the NAT problem, I'm new in Asterisk`
`21:40.30`	`hads`	`CunningPike: The sounds aren't included in 1.4 anymore AFAIK so they can be updated seperately.`
`21:41.35`	`CunningPike`	`hads: Aye - we already were running third party sounds - I was just wondering if the bundled sounds in the 1.4 install would include the 'fixed' sounds`
`21:42.10`	`hads`	`I know but I was saying that AFAIK 1.4 doesn't bundle sounds anymore.`
`21:42.34`	`hads`	`(although I never download the tarball so I don't know about that)`
`21:45.15`	`sahafeez`	`i am having the strangest issue and i really could use some help`
`21:45.54`	`*** part/#asterisk krondorl (n=krondorl@acid.auricnet.ca)`
`21:46.30`	`klictel`	`hi all... anyone knows if the new web interface (gui framework) is available in beta3?`
`21:46.31`	`*** join/#asterisk ikey (i=ikey@220.226.36.47)`
`21:47.41`	`ManxPower`	`klictel: see the /topic`
`21:47.49`	`sahafeez`	`i have some polycom sip phones. everything has worked forever. i have 2 remote sites via vpn.`
`21:48.07`	`sahafeez`	`the one site has been up for a year and has one phone. no isseus.`
`21:48.32`	`sahafeez`	`just added a new site to the vpn and i am trying to setup the phones. no issues there. they boot, etc.`
`21:48.51`	`klictel`	`Manx: where do I see /topic?`
`21:48.55`	`sahafeez`	`hum...i think i know..`
`21:49.12`	`ManxPower`	`klictel: at the top of this window or try typing "/topic"`
`21:49.16`	`*** join/#asterisk ShadwDrgn (i=arcane@adsl-217-6-103.asm.bellsouth.net)`
`21:49.39`	`ShadwDrgn`	`is it possible to make it so when i dial in with a certain number... asterisk dials OUT to another number and plays a sound file when they pick up?`
`21:50.17`	`ManxPower`	`ShadwDrgn: yes`
`21:51.12`	`ShadwDrgn`	`i can get it to dial the number`
`21:51.15`	`ShadwDrgn`	`but it won't play the audio`
`21:51.21`	`ManxPower`	`"show application dial"`
`21:51.23`	`ShadwDrgn`	`yeah`
`21:51.24`	`klictel`	`well /topic needs a parameter... and i see no mention of the web interface`
`21:51.28`	`ShadwDrgn`	`i think i just found that as well, thanks.`
`21:51.33`	`mercestes`	`Ok, I have an IAX channel going from one * box to another, via a LAN connection. HOwever, they seem to be going "Unreachable" Time: 3 and then "Reachable" Time: 3 pretty consistently.`
`21:51.37`	`mercestes`	`How can I fix this??`
`21:51.44`	`ManxPower`	`(15:49:04) current topic is: Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Zaptel 1.2.11 released (Nov. 9, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework`
`21:53.34`	`klictel`	`Manx thanks... /topic did not work and the topic is incomplete on my window`
`21:54.17`	`*** join/#asterisk test34 (n=test34@unaffiliated/test34)`
`21:56.51`	`*** join/#asterisk jusse (n=jusse@201.240.83.160)`
`21:57.07`	`jusse`	`hi`
`21:57.09`	`jusse`	`deny of service in asterisk?`
`21:57.11`	`jusse`	`:(`
`21:57.12`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`21:57.36`	`ManxPower`	`jusse: yes.`
`21:57.43`	`jusse`	`waaa`
`21:57.58`	`ManxPower`	`jusse: just like denial of service for any other internet service`
`21:58.10`	`jusse`	`aps`
`21:58.11`	`jusse`	`yes`
`21:59.16`	`jusse`	`how many companies use asterisk`
`21:59.26`	`*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)`
`21:59.34`	`ManxPower`	`jusse: when you are done with your survey let us know.`
`22:00.09`	`jusse`	`^^`
`22:00.36`	`Supaplex`	`jusse: a) 1-5, b) 20-30, c) 421, d) lots`
`22:01.36`	`jusse`	`d)`
`22:01.59`	`jusse`	`^^^`
`22:02.12`	`jusse`	`mmm... i need a hamburger o.O`
`22:02.31`	`Pumas`	`I have a VOIP number with SIP protocol How can I solve the NAT problem, I'm new in Asterisk`
`22:03.30`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`22:06.16`	`ShadwDrgn`	`damn`
`22:06.24`	`ShadwDrgn`	`i can't get DISA to work all i hear is silence when i dial in :(`
`22:06.40`	`Corydon-w`	`ShadwDrgn: did you Answer the line first?`
`22:06.55`	`Ryushin`	`I'm chomping at the bit for the openmoko phone to be released so I can run asterisk on it.`
`22:07.46`	`ShadwDrgn`	`yes`
`22:08.01`	`ShadwDrgn`	`i answer and then do DISA(1111)`
`22:08.15`	`ShadwDrgn`	`i also tried DISA(no-password)`
`22:08.28`	`ShadwDrgn`	`i see on the CLI it's executing the DISA application, but i never hear anything. :(`
`22:08.43`	`bluregard`	`does anyone know of any retail stores that sell plain old IP phones not locked to Vonage or some other damn provider.`
`22:08.58`	`ManxPower`	`ShadwDrgn: does it work even if you don'thear dialtone`
`22:09.08`	`ManxPower`	`try making sure you have /etc/asterisk/indications.conf`
`22:09.20`	`ShadwDrgn`	`i'll look`
`22:09.31`	`ShadwDrgn`	`i have indications.conf`
`22:09.35`	`ShadwDrgn`	`but i haven't done anything with it`
`22:10.00`	`sahafeez`	`can you list more then one localnet in sip.conf?`
`22:10.16`	`ManxPower`	`sahafeez: I doubt it`
`22:10.22`	`sahafeez`	`shit`
`22:11.31`	`ManxPower`	`but you should be able to lie about the localner`
`22:11.34`	`sahafeez`	`i have some vpn networks that are getting tagged as being sip ext@external addess when i need it ot be the internal`
`22:11.55`	`sahafeez`	`the vpn nets are all 192.168.x.x`
`22:12.18`	`ManxPower`	`i.e. if your networks are 172.16.5.0 172.16.10.0 and 172.16.44.0 you could use localnet=172.16.0.0/255.255.0.0`
`22:13.51`	`ShadwDrgn`	`currently the lines for this are only`
`22:13.57`	`ShadwDrgn`	`exten => 6782613529/6784512668,1,Answer()`
`22:13.57`	`ShadwDrgn`	`exten => 6782613529/6784512668,2,DISA(no-password)`
`22:14.02`	`ShadwDrgn`	`awesome`
`22:14.02`	`sahafeez`	`yah.`
`22:14.06`	`ShadwDrgn`	`i advertised myp honenumbers`
`22:14.11`	`ShadwDrgn`	`oh well`
`22:14.51`	`*** join/#asterisk Un1x (i=Un1x@CPE001731208485-CM00080d850684.cpe.net.cable.rogers.com)`
`22:14.52`	`sahafeez`	`that fixed it. shit spent 3 hrs when it was that simple - did not figure out the issue until then`
`22:15.09`	`sahafeez`	`new version of polycom code showed the sip ext@ip`
`22:15.23`	`Un1x`	`Hey, how do i put a certain extension so when i put in extensions lets say 33 it dails a specific number?`
`22:16.39`	`ManxPower`	`exten => 33,1,Dial(Zap/g1/5551515)`
`22:16.44`	`ManxPower`	`noob`
`22:16.48`	`Un1x`	`heh thanks :)`
`22:16.58`	`Un1x`	`whats the g1 for tho?`
`22:17.48`	`ManxPower`	`I assume you have a zaptel group 1`
`22:17.59`	`ShadwDrgn`	`ManxPower i don't get it what could i be doing wrong?`
`22:18.20`	`[TK]D-Fender`	`ShadwDrgn : You may want to consider calling DISA properly and actually pass it the CONTEXT its supposed to dial as...`
`22:18.30`	`ShadwDrgn`	`i tried that too`
`22:18.36`	`ManxPower`	`ShadwDrgn: do other things like playback work`
`22:18.44`	`ShadwDrgn`	`ManxPower: yes. everything else works great.`
`22:18.51`	`[TK]D-Fender`	`ShadwDrgn : Pastebin a full sample including this for us to see and the CLI output of the failed attempt`
`22:19.00`	`ShadwDrgn`	`[TK]D-Fender: I had DISA(1111\|sipphone) at one point`
`22:19.05`	`ShadwDrgn`	`sure`
`22:19.53`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)`
`22:20.30`	`Kyler`	`O.k., after years of using Asterisk I've never made calls from the console. Is there a way to call a SIP host by IP address (instead of extension@context)?`
`22:20.39`	`*** join/#asterisk jm\|work (n=jamiem@zen.jamiem.com)`
`22:22.02`	`CunningPike`	`Is there a way in the dialplan to look up the callerid in sip.conf for a given extension`
`22:22.08`	`*** join/#asterisk jets (n=jets@72.22.230.207)`
`22:22.35`	`[TK]D-Fender`	`Kyler : Dial(SIP/exte@ipaddress)`
`22:22.55`	`Kyler`	`[TK]D-Fender: That doesn't seem to work from the console.`
`22:23.55`	`[TK]D-Fender`	`Kyler : the format is similar for the console, though I can't confirm it exactly.`
`22:24.07`	`ShadwDrgn`	`http://pastebin.ca/250384`
`22:24.09`	`ShadwDrgn`	`there you go ManxPower`
`22:25.09`	`Kyler`	`[TK]*CLI> dial SIP/1234@12.34.56.78`
`22:25.10`	`Kyler`	`No such extension 'SIP/1234' in context '12.34.56.78'`
`22:25.47`	`JT`	`ShadwDrgn: it certainly doesn't look like you are Answer()ing calls before DISA`
`22:25.53`	`ShadwDrgn`	`oh`
`22:25.57`	`ShadwDrgn`	`yeah i took answer out by mistake`
`22:25.59`	`ShadwDrgn`	`it was in there.`
`22:26.13`	`[TK]D-Fender`	`ShadwDrgn : And what are you trying to dial? Also, whats in [sipphone2] ?`
`22:26.31`	`ShadwDrgn`	`sipphone2 is jsut my catchall so when i dial out it goes through my sunrocket`
`22:26.47`	`*** join/#asterisk FLP- (n=Flo@86.121.3.170)`
`22:26.48`	`ShadwDrgn`	`hold on i've just added answer`
`22:26.55`	`[TK]D-Fender`	`ShadwDrgn : And what # are your attempting to dial in it?`
`22:27.10`	`ShadwDrgn`	`6781111111`
`22:27.16`	`[TK]D-Fender`	`ShadwDrgn : And yes, you should definately ANSWER the call first.`
`22:27.22`	`ShadwDrgn`	`okay i added answer`
`22:27.31`	`ShadwDrgn`	`and it does the SAME thing but the debug shows it answering first`
`22:28.18`	`[TK]D-Fender`	`ShadwDrgn : You're using DISA to dial the same number that LANDS you on it? Thats not swift...`
`22:28.36`	`ShadwDrgn`	`uhm...`
`22:28.37`	`ShadwDrgn`	`no`
`22:28.43`	`ShadwDrgn`	`:(`
`22:28.51`	`ShadwDrgn`	`DISA is supposed to give me a dialtone right?`
`22:28.56`	`ShadwDrgn`	`not dial anything.`
`22:31.58`	`ShadwDrgn`	`my asterisk box is at 6781111111 and the phone i dialing from is 6782222222`
`22:32.09`	`ShadwDrgn`	`it answers and calls DISA but i never hear anything`
`22:33.29`	`*** join/#asterisk saftsack (n=saftsack@pD9E04C82.dip.t-dialin.net)`
`22:34.11`	`*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`22:34.53`	`*** join/#asterisk juanmanuel (n=jmacz@190.24.100.110)`
`22:35.27`	`hoobastooba`	`i am trying to do something that is not working... well i am trying to do a bunch of things that are not working.... but to be specific...`
`22:36.08`	`hoobastooba`	`i am trying to call one extension, that rings one phone and a queue that has cell numbers logged into it. and then if neither of those get answered I am trying to dial a specific cell phone. here is how I am trying to do it: http://pastebin.ca/250395`
`22:37.06`	`*** join/#asterisk bhima (n=gf2e@server1.mdsamerica.com)`
`22:37.40`	`*** part/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net)`
`22:37.50`	`hoobastooba`	`its not working... it does not call the cell phone after the timeouts of the queue and the dial command.`
`22:37.57`	`hoobastooba`	`can anyone offer some suggestions?`
`22:39.37`	`[TK]D-Fender`	`ShadwDrgn : Actually I don't see an answer in there at all. you should put that before your call to DISA`
`22:40.04`	`ShadwDrgn`	`i added answer`
`22:40.37`	`ShadwDrgn`	`it does the same thing except now it answers first`
`22:41.34`	`JT`	`ShadwDrgn: does indications.conf exist?`
`22:41.43`	`ShadwDrgn`	`yes.`
`22:42.51`	`*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`22:43.05`	`*** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net)`
`22:44.53`	`bhima`	`Is there a way to make a particular DID on a PRI redirect to another phone number? ie: person dials a DID, but the call gets forwarded to a cellphone, without using a pair of our T1 channels.`
`22:45.53`	`JT`	`as in a pri redirect message?`
`22:46.01`	`JT`	`as opposed to just dialling it out somewhere`
`22:46.26`	`bluregard`	`does anyone have any luck good or bad using Dlink IP phones?`
`22:46.27`	`*** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)`
`22:46.46`	`JT`	`dlink make ip phones? :o`
`22:46.47`	`*** join/#asterisk tbic (n=tbic@protious.fciautomation.com)`
`22:47.00`	`bluregard`	`yeah, a few`
`22:47.01`	`EmleyMoor`	`Whenever I connect my FXO port to my line, it jams my line off-hook - and I can't see why. What should I check?`
`22:47.03`	`[TK]D-Fender`	`bhima : http://www.voip-info.org/wiki-Asterisk+bounty+PRI+2B+channel+transfer`
`22:48.06`	`JT`	`bristuff can do pri ECT`
`22:48.12`	`bhima`	`TK: Interesting. Now, this sounds like a fancier feature than I want, somewhat.`
`22:48.39`	`JT`	`bhima: not sure if standard asterisk can do it, by the sounds of it, BRIstuff can`
`22:48.41`	`*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)`
`22:48.58`	`bhima`	`I don't actually mind if I have to set it up "in advance" - basically, I want SIP telephony when you're in the office, cellular when you're not.`
`22:49.36`	`bhima`	`So, if I can tell phone company to forward a DID the way you usually can with one of your POTS phones, that would actually be sufficient.`
`22:49.52`	`JT`	`junghanns.net`
`22:50.17`	`EmleyMoor`	`What makes an FXO channel go "off hook"? Mine is stuck, it seems`
`22:52.14`	`CunningPike`	`EmleyMoor: Usually, it's an FXO channel that fails to go back on hook due to a lack of call progress supervision on the line`
`22:52.40`	`EmleyMoor`	`Mine is stuck (and I mean stuck - even after a reboot) off hook`
`22:53.42`	`bhima`	`Do you think that I could use a standard POTS *72 call forward on a DID on a PRI?`
`22:53.43`	`EmleyMoor`	`Is there a guide to "correct settings with BT"?`
`22:54.22`	`CunningPike`	`bhima: I highly doubt it...... but why not forward it in your dialplan?`
`22:54.42`	`JT`	`bhima: btw i think you want CD not ECT`
`22:54.43`	`*** join/#asterisk Lord_Pall (i=Flerb@cpe-66-69-213-131.austin.res.rr.com)`
`22:54.50`	`JT`	`call deflection`
`22:54.53`	`bhima`	`CunningPike: because then I'll use two channels...`
`22:55.15`	`ShadwDrgn`	`well my idiot friend turned the asterisk box off`
`22:55.21`	`ShadwDrgn`	`so i guess i'll have to figure this out tommorow`
`22:55.38`	`Lord_Pall`	`anyone know why I'm getting a chanunavail message from voxee? right after the "call accepted" line in the debug log.`
`22:55.40`	`CunningPike`	`bhima: Ah - well, your telco can forward your DID wherever you want..... but I don't think *72 will work - ymmv`
`22:55.56`	`EmleyMoor`	`What command do I issue to make an FXO channel hang up?`
`22:56.15`	`ShadwDrgn`	`oh well i think i know what's wrong`
`22:56.20`	`bhima`	`CunningPike: Is there a way to ask the telco to forward it programatically, or must I call their customer support?`
`22:56.46`	`CunningPike`	`bhima: I think you need to talk to their customer service.....`
`22:57.16`	`bhima`	`(We have a data T1 right now, and we are thinking of converting a few B channels to a partial PRI. If, however, every person who's out of the office costs us two active Bs per call, that would suck)`
`22:58.10`	`JT`	`bhima: i think you want CD`
`22:58.14`	`pigpen`	`Can anyone tell me how to set * to pick up the last channel of the pri on outbound calls, where incoming calls are on the first channel? (ie: glare)`
`22:58.15`	`EmleyMoor`	`Is there no way to force an FXO channel "on-hook"?`
`22:58.18`	`Lord_Pall`	`interesting. Voxee iax2 is horked. 3 works fine.`
`22:59.01`	`*** join/#asterisk macli (n=vincent@mcli.brc.ubc.ca)`
`22:59.06`	`Un1x`	`Hey, can someone tell me how to unblock private numbers with Asterisk?`
`22:59.19`	`EmleyMoor`	`Un1x: "unblock private numbers"/`
`22:59.20`	`EmleyMoor`	`?`
`22:59.37`	`bhima`	`JT: Yeah, that's probably what I want. Is that normally supported on a PRI? Does that depend on my telco...?`
`22:59.41`	`Un1x`	`Yes, i read about in a article before that it was possible and it had been done`
`22:59.56`	`puddingpimp_work`	`Un1x, you need PRI IIRC`
`22:59.57`	`JT`	`bhima: it probably would`
`23:00.14`	`EmleyMoor`	`What do you mean by "unblock", and what do you mean by "private numbers"/`
`23:00.15`	`EmleyMoor`	`?`
`23:00.31`	`puddingpimp_work`	`I presume he means getting CID info for an unlisted number`
`23:00.42`	`JT`	`and more importantly, which directions are the calls doing`
`23:00.57`	`Un1x`	`puddingpimp_work no i ment, you know before you call someone you can dial *67 to block youre number from showing on the receivers screen`
`23:01.07`	`macli`	`Hi, when I type command 'show agi' at the asterisk CLI, it gave me "No such command 'show agi' (type 'help' for help)"`
`23:01.22`	`Un1x`	`i read on a white paper before that it had been done and possible i rember them demonstrating it at Blackhat conv but i forgot how to :S`
`23:01.29`	`puddingpimp_work`	`Un1x, that's what I mean, it is the same as having an unlisted number, and you need PRI`
`23:01.32`	`Un1x`	`JT incoming`
`23:01.33`	`EmleyMoor`	`Un1x: Do you mean to make that ineffective?`
`23:01.43`	`Un1x`	`puddingpimp_work how would a PRI help me?`
`23:01.46`	`EmleyMoor`	`(on calls to you)`
`23:01.53`	`JT`	`Un1x: helps if you explain things better`
`23:02.01`	`JT`	`Un1x: ugh, the signalling is digital`
`23:02.01`	`pigpen`	`Can anyone tell me how to set * to pick up the last channel of the pri on outbound calls, where incoming calls are on the first channel? (ie: glare)`
`23:02.06`	`JT`	`there's heaps more info sent`
`23:02.09`	`JT`	`than an analogue line`
`23:02.10`	`pigpen`	`Sorry to repeat...I am on the phone with ATT....`
`23:02.38`	`suma`	`pigpeb: zap/g(span)-(channel)/ is the format`
`23:02.47`	`suma`	`pigpen: zap/g(span)-(channel)/ is the format`
`23:02.49`	`EmleyMoor`	`My FXO is stuck "off-hook". If I plug it in, it takes my line off hook. How do I force it down?`
`23:02.52`	`puddingpimp_work`	`Un1x, it just does, without PRI you don't have access to that information, pri tells you IIRC the originating ANI and reverse number, which is different to the CID number that *67 blocks`
`23:03.12`	`puddingpimp_work`	`s/reverse/originating/`
`23:03.25`	`JT`	`puddingpimp_work: depends on telco, most won't send ANI to end users if blocked i'd think`
`23:03.32`	`dlynes_laptop`	`EmleyMoor: sipura 3000, sangoma, digium, ..., ?`
`23:03.38`	`EmleyMoor`	`Digium`
`23:03.58`	`pigpen`	`tks`
`23:04.07`	`puddingpimp_work`	`JT, well I heard different, and have no experience with it, but it's definitely not going to happen without PRI or some other trunk connection`
`23:04.12`	`dlynes_laptop`	`EmleyMoor: have you tried unloading asterisk, then unloading your zaptel drivers, and then reloading everything?`
`23:04.32`	`EmleyMoor`	`I've tried rebooting - surely that should have been effective?`
`23:04.37`	`dlynes_laptop`	`EmleyMoor: it sounds like the hardware's probably stuck in a certain mode`
`23:04.44`	`JT`	`puddingpimp_work: well i've been doing testing here, and unless the ANI doesn't come up in PRI debug messages, when i call from private numbers, i can't see the number anywhere`
`23:04.45`	`dlynes_laptop`	`EmleyMoor: or rebooting should've solved it, too`
`23:04.47`	`puddingpimp_work`	`anyone know of a guide/example dialplan for Peoples Republic of China?`
`23:05.01`	`dlynes_laptop`	`EmleyMoor: I would think the hardware is defective, then`
`23:05.28`	`puddingpimp_work`	`JT, ok, I stand corrected, though I did read somewhere that this is possible, doesn't mean all carriers will allow you this`
`23:05.34`	`EmleyMoor`	`Oh hell! My supplier have made a right mess of things as it is`
`23:06.02`	`puddingpimp_work`	`PRI also allows you to fake CID, but not ANI IIRC, again shout me down if I'm wrong`
`23:06.04`	`dlynes_laptop`	`EmleyMoor: Digium has a 5 year(?) warranty on all their products`
`23:06.12`	`EmleyMoor`	`Is there a command I can use to see if it is on or off hook?`
`23:06.13`	`dlynes_laptop`	`puddingpimp_work: correct`
`23:06.14`	`*** part/#asterisk bhima (n=gf2e@server1.mdsamerica.com)`
`23:06.26`	`dlynes_laptop`	`EmleyMoor: zap show channels`
`23:06.34`	`JT`	`puddingpimp_work: certainly if the carrier sends the info, asterisk can use it`
`23:06.47`	`dlynes_laptop`	`EmleyMoor: and then zap show channel xx where xx is the channel you're interested in`
`23:06.54`	`EmleyMoor`	`Digium's main UK agent have already let me down by only shipping part of the order`
`23:07.01`	`JT`	`puddingpimp_work: i've tried faking ANI, unfortunately my telco has sanity checking and checks whether you own the number first, it will only send a number you own`
`23:07.01`	`EmleyMoor`	`What do I look for?`
`23:07.09`	`JT`	`faking CLI i mean`
`23:07.13`	`dlynes_laptop`	`EmleyMoor: 'offhook'`
`23:07.22`	`dlynes_laptop`	`EmleyMoor: in zap show channel xx`
`23:09.10`	`Lord_Pall`	`hmm. How do i speed up hangups? I think I'm getting orphaned calls or something and it's causing voxee to kick out when I try to make another call before waiting`
`23:10.33`	`EmleyMoor`	`Hmmm - it goes offhook when I plug it in and returns to onhook when I unplig`
`23:10.35`	`EmleyMoor`	`unplug`
`23:11.07`	`dlynes_laptop`	`EmleyMoor: Do you have another jack wired in parallel with that one?`
`23:11.22`	`dlynes_laptop`	`EmleyMoor: Or maybe something tapping the phone?`
`23:11.29`	`dlynes_laptop`	`EmleyMoor: I'd measure the line voltage`
`23:11.35`	`dlynes_laptop`	`EmleyMoor: see if it's 50 vdc`
`23:11.53`	`JT`	`nominal -48VDC ;)`
`23:11.59`	`EmleyMoor`	`I have an extension socket with a caller display, a modem and a satellite TV box connected`
`23:12.08`	`dlynes_laptop`	`JT: 48, 50... :)`
`23:12.20`	`EmleyMoor`	`Should I try without that lot or does it only make a difference if they go offhook?`
`23:12.44`	`dlynes_laptop`	`EmleyMoor: If it's much less than 48vdc as JT stated, you might have to make some adjustments in the driver parameters`
`23:12.53`	`dlynes_laptop`	`EmleyMoor: if the driver supports that`
`23:12.57`	`dlynes_laptop`	`EmleyMoor: or unplug some stuff`
`23:13.21`	`dlynes_laptop`	`EmleyMoor: but like i said...measure the voltage first before you get all crazy`
`23:13.34`	`dlynes_laptop`	`EmleyMoor: the card might be defective, too`
`23:13.44`	`riddlebox`	`can someone help me get asterisk to start when the pc boots?`
`23:14.01`	`*** join/#asterisk Waverly360 (n=mirc@adsl-070-148-122-203.sip.bna.bellsouth.net)`
`23:14.01`	`puddingpimp_work`	`riddlebox, what platform?`
`23:14.19`	`riddlebox`	`puddingpimp_work, ubuntu`
`23:14.37`	`puddingpimp_work`	`riddlebox, /etc/rc.local or /etc/conf.d/localstart <- on linux one of these should exist, put asterisk in here`
`23:15.12`	`EmleyMoor`	`If the card is defective, I want someone from myphonecall to actually come here - I have been let down badly by them already`
`23:15.29`	`EmleyMoor`	`Measuring the voltage is not an easy proposition TBH`
`23:15.38`	`dlynes_laptop`	`EmleyMoor: tbh?`
`23:15.44`	`JT`	`to be honest`
`23:15.47`	`dlynes_laptop`	`ah`
`23:15.51`	`dlynes_laptop`	`it's not?`
`23:15.58`	`JT`	`tbh, it's a good diagnostic step to take`
`23:16.00`	`dlynes_laptop`	`Just stick a dmm on the leads`
`23:16.08`	`puddingpimp_work`	`EmleyMoor, why? DMM + beige box, can't be that hard`
`23:16.16`	`*** join/#asterisk adamowitz (n=adamowit@ip68-14-27-224.ri.ri.cox.net)`
`23:16.22`	`JT`	`what on earth do you need the beige box for`
`23:16.33`	`EmleyMoor`	`What is a beige box?`
`23:16.41`	`dlynes_laptop`	`puddingpimp_work: even just opening up the wall plate, and sticking your dmm leads on the back of the plate`
`23:16.44`	`puddingpimp_work`	`EmleyMoor, the wall socket that the phone line comes into`
`23:16.54`	`mercestes`	`WHY does a Lan IAX2 channel go "unreachable" at every registration sync and then reachable again a second after? how do I correct that??`
`23:16.59`	`puddingpimp_work`	`well the wall plate is a beige box`
`23:17.07`	`JT`	`EmleyMoor: something you don't need`
`23:17.11`	`dlynes_laptop`	`puddingpimp_work: why not call it a wall plate then?`
`23:17.15`	`JT`	`puddingpimp_work: that is NOT the definition of beige box`
`23:17.19`	`puddingpimp_work`	`because I'm stupid`
`23:17.27`	`dlynes_laptop`	`puddingpimp_work: most of the time they're white, or almond...never seen a beige one`
`23:17.40`	`puddingpimp_work`	`JT, well it worked for me when I was a naughty teenager`
`23:17.41`	`JT`	`beige box is phreaking terminology for a linesman's handset`
`23:17.59`	`dlynes_laptop`	`JT: never seen a beige one of those, either :)`
`23:18.05`	`JT`	`dlynes_laptop: i have one.`
`23:18.11`	`JT`	`it's beigeish`
`23:18.17`	`dlynes_laptop`	`JT: I've got several...yellow, blue, red...never beige`
`23:18.17`	`JT`	`browny-beige`
`23:18.27`	`JT`	`i know they make them all safety solours these days`
`23:18.28`	`dlynes_laptop`	`JT: you're talking about a butt set, right?`
`23:18.30`	`JT`	`yes`
`23:18.32`	`JT`	`buttinski`
`23:19.01`	`JT`	`old standard issue Telecom Australia DTMF buttinskis`
`23:19.03`	`EmleyMoor`	`I haven't got anything to hand which will let me get at the wires enough to put a meter across...`
`23:19.16`	`puddingpimp_work`	`my bad, anyway just pull the wall socket off the wall and jam the probes into the back of the crimps`
`23:19.26`	`mercestes`	`So plug a wire into a wall plate and cut the other end, EmleyMoor.`
`23:19.56`	`dlynes_laptop`	`JT: ah...these ones are all standard Harris butt sets...same ones Telus, Bell, ... all use`
`23:20.21`	`mercestes`	`Harris. Now wholly owned by Fluke.`
`23:20.23`	`dlynes_laptop`	`EmleyMoor: your meter doesn't have probes?`
`23:20.32`	`_DAW`	`Can someone tell me if it is possible to replicate the paging port from a conventional pbx with *? I thought the port was just fxs but obviously not.`
`23:20.52`	`EmleyMoor`	`My probes aren't sharp enough`
`23:21.03`	`JT`	`dlynes_laptop: yeah they don't make their own branded ones anymore i don't think, and they're Telstra now`
`23:21.04`	`mercestes`	`??`
`23:21.04`	`EmleyMoor`	`I don't have any leads I can sacrificer`
`23:21.11`	`JT`	`most would use harris or whatever`
`23:21.37`	`dlynes_laptop`	`JT: yeah...never been down under, so I wouldn't know about aust`
`23:22.03`	`dlynes_laptop`	`EmleyMoor: how about some pieces of wire you can screw onto the back with the existing connections?`
`23:22.31`	`dlynes_laptop`	`EmleyMoor: or do you have a mod end crimper?`
`23:22.55`	`dlynes_laptop`	`EmleyMoor: you could make a new rj11 mod end with some leads running out of it, plug it into the plate, and hook it up to your dmm`
`23:23.35`	`puddingpimp_work`	`dlynes_laptop, I haven't seen any phone sockets with screws, all the ones in NZ have punchdown termination, and I can see how it would be difficult to jam probes into those, though you can take the pcb off the plate and access the tracks on the other side`
`23:23.37`	`*** join/#asterisk [hC] (n=hardcore@S010600404521309c.vc.shawcable.net)`
`23:24.06`	`dlynes_laptop`	`puddingpimp_work: wall plates are punchdown termination in nz?`
`23:24.15`	`dlynes_laptop`	`puddingpimp_work: you mean like 110 punch, or 66 block?`
`23:24.23`	`mercestes`	`that explains alot.`
`23:24.30`	`EmleyMoor`	`-49V`
`23:24.38`	`mercestes`	`dlynes_laptop: I think he means like a Rj-45 port punchdown.`
`23:24.40`	`dlynes_laptop`	`EmleyMoor: there ya go`
`23:24.48`	`dlynes_laptop`	`mercestes: ah..yeah...usually 110 punch`
`23:25.18`	`dlynes_laptop`	`EmleyMoor: looks normal then`
`23:26.14`	`puddingpimp_work`	`dlynes_laptop, BT telecom punch, I think they are different to Krone, they're not very good and the sockets come with cheap plastic terminators`
`23:26.14`	`dlynes_laptop`	`EmleyMoor: it's probably an issue with your driver settings, or the hardware is deffective`
`23:26.14`	`puddingpimp_work`	`*termination tools`
`23:26.14`	`JT`	`puddingpimp_work: do new installs in nz still use the nz proprietary connector or is RJ-12 or something?`
`23:26.14`	`JT`	`still the BT connector?`
`23:26.14`	`dlynes_laptop`	`puddingpimp_work: yeah...the krone's require a special tool from krone...thankfully we don't have any of those here`
`23:26.14`	`puddingpimp_work`	`JT, all nz plugs use BT, before that they were hard wired (back in pulse dial days)`
`23:26.17`	`EmleyMoor`	`I need to find out which is most probable really`
`23:26.25`	`JT`	`dlynes_laptop: what's special about the tool?`
`23:26.32`	`dlynes_laptop`	`JT: It's not 110 punch`
`23:26.36`	`tsurk0`	`hello, i have to access an asterisk behind nat router via IAX. Could you tell me which ports I am supposed to redirect? I'm trying with 5060 and 4569 (udp), but no good.`
`23:26.38`	`EmleyMoor`	`If the lead I'm using is four wires, would that make a difference?`
`23:26.47`	`JT`	`i have no idea what the difference is`
`23:26.54`	`JT`	`we only have krone stuff around here`
`23:26.57`	`dlynes_laptop`	`JT: 110 punch is an american standard`
`23:27.05`	`EmleyMoor`	`puddingpimp_work: Weren't NZ rotary phones differently ordered than those in ROTW?`
`23:27.06`	`dlynes_laptop`	`JT: krone is a proprietary standard`
`23:27.28`	`JT`	`uhuh`
`23:27.29`	`dlynes_laptop`	`JT: there's also 66 block and bix`
`23:27.41`	`dlynes_laptop`	`JT: bix is another proprietary standard`
`23:28.06`	`puddingpimp_work`	`EmleyMoor, yes, the digits except 0 were back to front, you dial (10 - digit) pulses instead of digit pulses`
`23:28.09`	`JT`	`american standards are proprietary too :P`
`23:28.14`	`EmleyMoor`	`I am still awaiting the cables I ordered from myphonecall`
`23:28.37`	`dlynes_laptop`	`JT: no idea...all i know is that there's plenty of third party manufacturers that manufacture the same thing`
`23:28.44`	`dlynes_laptop`	`JT: so to me, that's not proprietary`
`23:28.46`	`JT`	`same with krone`
`23:29.12`	`JT`	`how does the actualy punchdown differ?`
`23:29.28`	`puddingpimp_work`	`oh well I've got to finish this dialplan for China, I know the emergency numbers and toll access prefix, but I'm stumped as to anything more complicated`
`23:29.34`	`dlynes_laptop`	`JT: I don't know what the krone punchdown is like..I just know it's different, so I couldn't tell you`
`23:29.42`	`JT`	`right`
`23:29.49`	`dlynes_laptop`	`JT: one sec, and i'll try to find you some images of 110 blocks`
`23:30.44`	`puddingpimp_work`	`the 110 is what I use for Cat6 termination IIRC, I definitely have the right tool, and I think it's a 110 but not sure`
`23:32.06`	`puddingpimp_work`	`It's so much fun punching UTP, not like trying to put RJ-45 mod's on UTP, that's a pretty tricky one ;)`
`23:32.43`	`JT`	`the worst is putting RJ-45 jacks on stranded cable`
`23:32.52`	`JT`	`also depends on the insulation of the strands`
`23:33.33`	`dlynes_laptop`	`JT: the tool is at: http://www.pccables.com/21001.htm; the panel is at: http://cableorganizer.com/phone-data-connection-blocks/`
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`23:34.39`	`puddingpimp_work`	`dlynes_laptop, yep, that's the exact tool I have for punching Cat6`
`23:34.54`	`JT`	`visually, i cannot determine why that tool would not work on krone`
`23:35.04`	`dlynes_laptop`	`puddingpimp_work: yeah...all the cat6 we do uses 110`
`23:35.09`	`dlynes_laptop`	`puddingpimp_work: even nordx`
`23:35.41`	`dlynes_laptop`	`JT: Here is the tool and the block for 66 block`
`23:35.46`	`dlynes_laptop`	`JT: oops http://www.tech-faq.com/66-block.shtml`
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`23:36.55`	`dlynes_laptop`	`JT: bix block: http://www.electronicproductonline.com/catalog/product_info.php?products_id=819; bix tool: http://indonetwork.net/cheerwe/300628`
`23:37.35`	`dlynes_laptop`	`JT: as you'll notice, 66 block tool and 110 punch tool are similar, but not the same`
`23:37.41`	`dlynes_laptop`	`JT: the bix tool is considerably different`
`23:37.42`	`puddingpimp_work`	`mmm, is it only North America that uses 1 for tolls, here and China they use 0ZXX for national, and 00Z for international`
`23:38.19`	`dlynes_laptop`	`puddingpimp_work: in NA, we use '1' for domestic (domestic == Canada, US and the Carribbean), and 011 for international (everything else)`
`23:38.34`	`JT`	`puddingpimp_work: everything is different everywhere in the world`
`23:38.42`	`JT`	`there is no standard international dialling prefix`
`23:38.53`	`JT`	`0011 in Australia`
`23:39.11`	`EmleyMoor`	`UK is 0ZX and 00Z too`
`23:39.18`	`JT`	`dlynes_laptop: yes bix looks very different`
`23:39.19`	`puddingpimp_work`	`dlynes_laptop, so how do you tell if a call is international, if dialing 1XXX. you could get Carribbean?`
`23:39.30`	`EmleyMoor`	`That's in the NANP`
`23:39.53`	`dlynes_laptop`	`puddingpimp_work: because we don't care if it's in the carribbean or not...domestic long distance is all different rates`
`23:39.55`	`JT`	`puddingpimp_work: part of the annoyance of being american :P, although the 3 digits after the 1 says where it's going`
`23:40.32`	`dlynes_laptop`	`puddingpimp_work: if i'm calling the US, it'll be different rates depending on which part of the US I'm calling`
`23:40.37`	`EmleyMoor`	`On a TDM400P, are the connectors in the RJ11 sockets NOT used for the A and B wires connected anywhere?`
`23:40.42`	`dlynes_laptop`	`puddingpimp_work: if i'm calling from a traditional phone company, that is`
`23:41.03`	`puddingpimp_work`	`ok, anyone know where I would get info on the CT dialplan in PRC?`
`23:41.22`	`dlynes_laptop`	`puddingpimp_work: what is it that you want to know about the prc dialplan?`
`23:41.27`	`puddingpimp_work`	`dlynes_laptop, is this because the US doesn't have a single national carrier, or just because of distance?`
`23:41.39`	`dlynes_laptop`	`puddingpimp_work: just because of long distance`
`23:41.53`	`dlynes_laptop`	`puddingpimp_work: Canada and the US both cover a huge area`
`23:42.23`	`JT`	`does the 110 tool automatically cut excess wire?`
`23:42.45`	`dlynes_laptop`	`puddingpimp_work: i'm sure even in Australia, it's probably a different rate for long distance from sydney to hobart than it is from Sydney to Brisbane`
`23:42.51`	`dlynes_laptop`	`JT: yes...so does the bix`
`23:43.22`	`puddingpimp_work`	`dlynes_laptop, I'm trying to setup outgoing calls for a PBX in china, I want to know what numbers are valid, and what numbers incur tolls etc.`
`23:43.25`	`dlynes_laptop`	`JT: the 110 does it by pressure with a sharp edge; the bix does it by actually cutting the wire using a pair of cutters that's part of the tool`
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`23:43.42`	`dlynes_laptop`	`puddingpimp_work: if i remember correctly, the whole country is one toll`
`23:43.56`	`dlynes_laptop`	`puddingpimp_work: but coppice would be able to give you a more accurate answer`
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`23:44.03`	`dlynes_laptop`	`puddingpimp_work: he's usually on late at night`
`23:44.04`	`puddingpimp_work`	`dlynes_laptop, come to think of it, it's different rates for shorter hops even in NZ and we're tiny`
`23:44.13`	`dlynes_laptop`	`puddingpimp_work: exactly`
`23:44.18`	`JT`	`dlynes_laptop: the krone tool cuts the cable with cutters`
`23:44.30`	`puddingpimp_work`	`JT, mine doesn't I`
`23:44.31`	`dlynes_laptop`	`JT: ah...so it's similar to the bix tool for that, then`
`23:44.44`	`puddingpimp_work`	`'ve seen one which does, but it felt light weight and flimsy`
`23:45.02`	`JT`	`dlynes_laptop: unless there's something i'm missing, i can't see why it wouldn't do 110 too`
`23:45.06`	`GarethTheGreat`	`chan_iax2.c:1387 reload_firmware: Error opening firmware directory '/usr/local/share/asterisk/firmware/iax': No such file or directory`
`23:45.13`	`GarethTheGreat`	`any idea what that's all about?`
`23:45.24`	`JT`	`puddingpimp_work: the krone tools with cutters are sturdier than the crappy little disposable punchdown tools`
`23:45.25`	`puddingpimp_work`	`dlynes_laptop, late at night where?`
`23:45.30`	`GarethTheGreat`	`i've got 2 asterisk boxes here, both on OpenBSD - trying to link them together with IAX2`
`23:45.32`	`dlynes_laptop`	`GarethTheGreat: it's not an error, even though it's telling you it is; it's just a warning`
`23:45.47`	`dlynes_laptop`	`GarethTheGreat: but you should have an iaxy.bin file somewhere`
`23:45.52`	`puddingpimp_work`	`JT, mine isn't flimsy, it has a solid plastic handle, and a cast steel head`
`23:45.54`	`GarethTheGreat`	`iax2-provision.c:515 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.`
`23:46.02`	`GarethTheGreat`	`i get that immediately afterwards`
`23:46.03`	`puddingpimp_work`	`JT, but it is a chinese clone`
`23:46.06`	`dlynes_laptop`	`puddingpimp_work: sorry....late at night, pacific standard time`
`23:46.11`	`JT`	`dlynes_laptop: impressed you got australian capital city names right :P`
`23:46.18`	`GarethTheGreat`	`and registration refused when the other box tries to register`
`23:46.18`	`dlynes_laptop`	`JT: and how's that?`
`23:46.24`	`dlynes_laptop`	`JT: I'm Canadian, not American :)`
`23:46.30`	`JT`	`heh`
`23:46.45`	`JT`	`i think even with traditional telcos, inter-capital is the same price`
`23:46.52`	`JT`	`it's non-capitals that are different`
`23:46.56`	`JT`	`iirc`
`23:46.56`	`GarethTheGreat`	`dlynes_laptop: any idea what the iax2-provision error is all about?`
`23:47.00`	`dlynes_laptop`	`JT: must be an asian thing`
`23:47.04`	`dlynes_laptop`	`JT: that's not true, here`
`23:47.11`	`JT`	`no, it's australian`
`23:47.23`	`JT`	`cheaper to call a capital city than a country town i think`
`23:47.36`	`dlynes_laptop`	`JT: Yeah...it's the same thing for China for calling cards`
`23:47.43`	`JT`	`although i've never paid too much attention to traditional telco pricing to regional areas`
`23:47.51`	`dlynes_laptop`	`JT: however, if you're actually in China, i think the prices are the same, no matter where you're claling`
`23:48.03`	`JT`	`heh`
`23:48.10`	`dlynes_laptop`	`GarethTheGreat: dump the iaxy.bin file in that directory, and the error will go away`
`23:48.24`	`JT`	`we have untimed local calls here for residential lines`
`23:48.41`	`dlynes_laptop`	`JT: When did that change?`
`23:48.48`	`EmleyMoor`	`How do I make sure the permissions on /dev/zap/* stay appropriate?`
`23:48.50`	`dlynes_laptop`	`JT: I could've sworn telstra used to have timed local calls`
`23:49.00`	`JT`	`dlynes_laptop: on some business plans`
`23:49.06`	`JT`	`never ever been timed by default for home`
`23:49.18`	`dlynes_laptop`	`JT: or maybe it was they charge you for local calls...just not per minute; per call`
`23:49.28`	`JT`	`yes`
`23:49.37`	`EmleyMoor`	`They charge per call in Hull`
`23:49.41`	`dlynes_laptop`	`JT: ah...yeah...here, we get charged by the month`
`23:49.48`	`JT`	`heh`
`23:49.51`	`dlynes_laptop`	`JT: not by the call or by the minute`
`23:50.13`	`JT`	`and we never pay to receive calls, unless we use call forwarding or reverse charge`
`23:50.14`	`dlynes_laptop`	`JT: so Telus typically charges $30/mo for residential service and up to $70/mo for business service`
`23:50.30`	`dlynes_laptop`	`JT: but unlimited incoming and outgoing (with the exception of long distance)`
`23:50.32`	`JT`	`does that include line rental?`
`23:50.40`	`dlynes_laptop`	`JT: yeah...that includes line rental`
`23:50.51`	`JT`	`that's still pretty pricey`
`23:51.01`	`JT`	`line rental is 30 or $40 AUD here`
`23:51.13`	`dlynes_laptop`	`JT: so why do you say that's pricey?`
`23:51.30`	`dlynes_laptop`	`JT: $30 is pretty comparable to your 30 or 40, and we don't have pay per call`
`23:51.44`	`dlynes_laptop`	`JT: $1Cdn ~= $1AUD`
`23:51.45`	`JT`	`so USD, well that's getting closer to $45AUD, and USD$70 for business is heaps more`
`23:51.48`	`JT`	`ah`
`23:52.04`	`JT`	`business lines are pricey then :)`
`23:52.05`	`GarethTheGreat`	`still not working`
`23:52.05`	`GarethTheGreat`	`== Loaded firmware 'iaxy.bin'`
`23:52.06`	`GarethTheGreat`	`Nov 16 23:51:36 NOTICE[3024]: iax2-provision.c:515 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.`
`23:52.11`	`mercestes`	`Gah, how do I get into the web admin of a cisco phone again?? It' sbeen so long since I dinked with these things.`
`23:52.12`	`*** join/#asterisk [pyro] (n=pyro@tor/regular/bracketed-pyro)`
`23:52.15`	`dlynes_laptop`	`GarethTheGreat: that's working`
`23:52.24`	`JT`	`GarethTheGreat: it's just a NOTICE, ignore it`
`23:52.39`	`JT`	`it's not an ERROR, or even a WARNING`
`23:52.41`	`dlynes_laptop`	`mercestes: do an nmap scan of it to find out what port it's listening on`
`23:53.06`	`mercestes`	`dlynes_laptop: lol..guess that's one way`
`23:53.17`	`dlynes_laptop`	`mercestes: what's wrong with that way? :)`
`23:53.30`	`mercestes`	`dlynes_laptop: Nothing...always looking for a good reason to nmap something.`
`23:54.00`	`dlynes_laptop`	`mercestes: that's how i foudn out what port my voip phones listen on...the stupid distributor gave me some crappy manual on them that didn't even list the port number`
`23:54.02`	`GarethTheGreat`	`"IAX provisioning disabled" and i can't get my 2 boxes to register to each other`
`23:54.27`	`dlynes_laptop`	`GarethTheGreat: iax provisioning is for the iaxy device (Digium's ATA that speaks IAX)`
`23:54.30`	`JT`	`do you need to do IAX provisioning`
`23:54.35`	`JT`	`GarethTheGreat: do you even have an IAXy?`
`23:54.42`	`dlynes_laptop`	`GarethTheGreat: it has nothing to do with having two asterisk boxes talking to each other`
`23:55.12`	`GarethTheGreat`	`ah`
`23:56.35`	`EmleyMoor`	`Can unexpectedly bringing the ring capacitor into circuit on an FXO port cause it to go offhook?`
`23:57.00`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`23:57.38`	`dlynes_laptop`	`EmleyMoor: huh?`
`23:58.39`	`EmleyMoor`	`On British lines, we have a ring capacitor in the NTE. If I'm using a 4-wire cable between the NTE and FXO, I could be doing something unexpected with it, I guess... could that be causing my "goes offhook" condition?`
`23:58.52`	`GarethTheGreat`	`what can cause the registration refused messages then?`
`23:59.24`	`GarethTheGreat`	`i've double-triple checked the username/password and IPs`

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