irclog2html for #asterisk on 20060825

`00:00.01`	`ringhals`	`I am running 1.2.11 on centos 4.3`
`00:00.21`	`ringhals`	`anyone have any suggestions?`
`00:02.05`	`*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-25-53.cybersurf.com)`
`00:02.37`	`ringhals`	`I am also getting a warning about`
`00:02.47`	`ringhals`	`WARNING[12237] chan_zap.c: getdtmf on channel 1: Operation now in progress`
`00:02.47`	`ringhals`	`WARNING[12237] chan_zap.c: getdtmf on channel 1: Operation now in progressWARNING[12237] chan_zap.c: getdtmf on channel 1: Operation now in progress`
`00:04.55`	`*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)`
`00:05.01`	`*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)`
`00:12.44`	`UForgotten`	`anyone know wtf is going on with vitelity?`
`00:18.00`	`*** part/#asterisk mountainm2k (n=mountain@216.87.64.218)`
`00:19.48`	`*** join/#asterisk wglenncamp (n=wcampbel@208.203.43.125)`
`00:19.57`	`*** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net)`
`00:30.52`	`*** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net)`
`00:42.25`	`bkw_`	`er music`
`00:42.28`	`bkw_`	`fark`
`00:42.29`	`bkw_`	`I can't type today`
`00:42.40`	`UForgotten`	`lol`
`00:43.02`	`*** join/#asterisk EtienneG (n=etienne@montreal.canonical.com)`
`00:47.06`	`Bullseye_Network`	`I've been all of a sudden getting bad echo on incoming audio. Were doing SIP to a voip provider. any ideas?`
`00:49.03`	`russellb`	`blame your provider`
`00:49.13`	`wunderkin`	`blame canada... or...file..`
`00:49.20`	`russellb`	`wunderkin: even better`
`00:49.55`	`Qwell`	`I fail`
`00:52.23`	`*** join/#asterisk test34 (n=test34@unaffiliated/test34)`
`00:52.55`	`*** join/#asterisk Gregabyte (n=greg@gateway.digium.com)`
`00:54.54`	`dieno`	`encounter webmin`
`00:54.59`	`dieno`	`how can i solve this thin`
`00:55.26`	`Bullseye_Network`	`If I bypass the asterisk system and dial their system directly I dont get the echo.`
`00:55.59`	`russellb`	`what type of phones?`
`00:56.32`	`Bullseye_Network`	`sjphones. We have been using them for several years. And the problem just started up tuesday.`
`00:56.48`	`Bullseye_Network`	`BUT. We also have problems with the Cisco phone.`
`00:56.52`	`Bullseye_Network`	`Its across the board`
`00:56.55`	`russellb`	`ok, well if you have IP phones to asterisk, and it is IP to your provider, how do you think Asterisk has anything to do with your echo?`
`00:57.01`	`russellb`	`it's not ... possible`
`00:57.07`	`Bullseye_Network`	`lol`
`00:57.08`	`Bullseye_Network`	`ok`
`00:57.21`	`russellb`	`:)`
`00:57.44`	`Bullseye_Network`	`I quarantee it is.`
`00:57.48`	`Bullseye_Network`	`Possible`
`00:58.04`	`Bullseye_Network`	`Echo is worse witht he soft phones`
`00:58.18`	`Bullseye_Network`	`Choppy audio with the cisco, MOSTLY.`
`00:58.40`	`russellb`	`echo happens at analog interfaces ... if there is no analog interface in your asterisk system, it is not possible`
`00:59.16`	`Bullseye_Network`	`The weird part is the problem is ONLY with the incoming audio.`
`00:59.29`	`Strom_C`	`Bullseye_Network, incoming audio on outbound calls?`
`00:59.34`	`Strom_C`	`or incoming calls only?`
`00:59.51`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:59.56`	`Bullseye_Network`	`all calls. But we mostly make out going calls`
`01:00.30`	`Bullseye_Network`	`The echo is only on outgoing calls the choppy is on inand out`
`01:01.02`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`01:01.03`	`*** mode/#asterisk [+o mog] by ChanServ`
`01:01.05`	`*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)`
`01:01.06`	`Bullseye_Network`	`And Bandwidth is not a problem.`
`01:01.46`	`russellb`	`once again, blame your provider :-p`
`01:02.49`	`russellb`	`it's more fun that way`
`01:03.30`	`Bullseye_Network`	`OK, I would but our other locations are connecting to the same provider and are not having the problem`
`01:07.24`	`[TK]D-Fender`	`Bullseye_Network : Do you get echo from 2 phones directly connected to *?`
`01:07.45`	`[TK]D-Fender`	`Bullseye_Network : As in SIP-SIP direct.`
`01:09.24`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`01:10.09`	`russellb`	`Bullseye_Network: really, the problem is in your phones`
`01:10.14`	`russellb`	`they have to do the echo cancellation`
`01:10.35`	`russellb`	`and ... they must suck at it`
`01:11.42`	`Bullseye_Network`	`ok CHecking`
`01:14.35`	`*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-54.intrastar.net)`
`01:21.19`	`*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id)`
`01:32.13`	`*** join/#asterisk mcreedjr (n=mcreedjr@adsl-75-13-62-230.dsl.toldoh.sbcglobal.net)`
`01:32.57`	`mcreedjr`	`Can anyone help me out here? Whenever I enable the externip option in my sip.conf, my SIP UAs fail to register with an Auth Mismatch error.`
`01:33.01`	`mcreedjr`	`What am I doing wrong?`
`01:33.48`	`*** join/#asterisk Strom_C (n=strom@fl-65-41-146-225.sta.embarqhsd.net)`
`01:35.22`	`*** part/#asterisk Bullseye_Network (n=info@72-166-37-114.dia.static.qwest.net)`
`01:43.44`	`EtienneG`	`does somebody know which CPU the SNOM phone use ? Just curious, wonder about hackability`
`01:44.36`	`*** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`01:45.24`	`*** join/#asterisk tengulre (n=tengulre@221.11.5.180)`
`01:47.07`	`[TK]D-Fender`	`EtienneG : Well it runs Linux, and the source is available......`
`01:48.36`	`shmaltz`	`mcreedjr, that SIP UA are they local or NATed?`
`01:53.41`	`*** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`01:56.45`	`EtienneG`	`[TK]D-Fender, i know that, just wondering which CPU`
`01:57.02`	`EtienneG`	`I a somewhat interested in gadget running on ARM`
`01:58.01`	`*** join/#asterisk crochat (i=crochat@84-74-146-28.dclient.hispeed.ch)`
`01:58.03`	`*** join/#asterisk Bullseye_Network (n=info@72-166-37-114.dia.static.qwest.net)`
`01:58.45`	`crochat`	`Hello !`
`01:59.12`	`crochat`	`Is there a way to do a simple automatic callback with Asterisk ?`
`01:59.29`	`docelmo`	`You mean like *69`
`02:00.17`	`crochat`	`I mean, I wanna call my Asterisk server on a specific number, and if Asterisk recognises my CALLERID, it makes a hangup, and it calls me back`
`02:00.47`	`JT`	`crochat: rep`
`02:00.49`	`JT`	`yep`
`02:00.56`	`docelmo`	`haha`
`02:01.01`	`docelmo`	`You want ANI callback`
`02:01.08`	`docelmo`	`Asterisk by itself is too dumb`
`02:01.18`	`docelmo`	`You would need a custom module or AGI`
`02:01.25`	`Strom_C`	`or the GotoIf command :)`
`02:01.28`	`JT`	`http://www.junghanns.net/en/callback.html - it's pretty basic, so you could reimplement it in something better than PHP if you wanted`
`02:01.54`	`russellb`	`or a simple system command to create a call file ...`
`02:02.17`	`hads`	`Oh the possibilities`
`02:02.20`	`crochat`	`Strom_C: No, with the GotoIf command, I couldn't make a callback`
`02:02.35`	`Strom_C`	`crochat, as one component of the dialplan script`
`02:03.01`	`crochat`	`Strom_C: I can recognize my CALLERID, but if I make a hangup, Asterisk couldn't go further in the context`
`02:03.14`	`Strom_C`	`crochat, thats why you do the following:`
`02:03.21`	`Strom_C`	`(1) send a progress message (do not answer)`
`02:03.31`	`hads`	`exten => h`
`02:03.33`	`Strom_C`	`(2) process the data`
`02:03.38`	`Strom_C`	`(3) send a hangup`
`02:03.42`	`Strom_C`	`(4) use the h extension`
`02:04.11`	`mog`	`what?`
`02:04.16`	`mog`	`you can do that`
`02:04.19`	`mog`	`without anything`
`02:04.30`	`crochat`	`Strom_C: You mean that Asterisk should Answer my call ?`
`02:04.39`	`Strom_C`	`crochat, no, you're not listening`
`02:04.47`	`Strom_C`	`i said do NOT answer the call`
`02:05.00`	`hads`	`I believe that was (1)`
`02:05.00`	`mog`	`its easy`
`02:05.31`	`Strom_C`	`mog, I'm always amused when people think you need to do great feats of AGI scripting to do something you can accomplish in a few lines of extensions.conf scripting :)`
`02:05.47`	`JT`	`crochat: have you looked at the url i pasted? it does EXACTLY what you are asking`
`02:05.53`	`mog`	`exactly`
`02:06.05`	`mog`	`you can do just about anything with dial plan`
`02:06.07`	`mog`	`and with ael`
`02:06.10`	`mog`	`its even easier`
`02:06.11`	`crochat`	`Strom_C: Sorry, I'm very tired... 04:06 AM here :-(`
`02:06.20`	`Strom_C`	`crochat, go to sleep`
`02:06.35`	`crochat`	`Strom_C: That's a good idea ;-)`
`02:06.37`	`JT`	`crochat: http://www.junghanns.net/en/callback.html`
`02:07.08`	`hads`	`"Use the dialplan: It's the way of the future."`
`02:07.30`	`Qwell`	`mog: you now...maybe I should learn ael soon, heh`
`02:07.31`	`[TK]D-Fender`	`docelmo : no need for AGI... just use a shell script to copy over a .call file`
`02:07.44`	`Qwell`	`know too`
`02:08.02`	`*** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)`
`02:10.11`	`Strom_C`	`mog, my favorite was someone who wrote a giant AGI to do caller ID spoofing`
`02:10.35`	`crochat`	`Strom_C: You said "send a progress message"... like Ringing() ?`
`02:10.45`	`Strom_C`	`crochat, like Progress()`
`02:10.59`	`Strom_C`	`if I wanted you to send a Ringing message, I would have said so.`
`02:11.14`	`hads`	`callerid spoofing is just so difficult with asterisk though.`
`02:11.48`	`Strom_C`	`yeah, all that horrid Set(${CALLERID(num)}=5552368) stuff is soooo insanely difficult`
`02:11.57`	`Strom_C`	`er`
`02:12.07`	`Strom_C`	`Set(CALLERID(num)=5552368)`
`02:12.11`	`hads`	`:)`
`02:12.13`	`Strom_C`	`god`
`02:12.16`	`Strom_C`	`what's wrong with me tonight :)`
`02:12.47`	`shmaltz`	`interrsting:http://www.ft.com/cms/s/c2eab782-3394-11db-981f-0000779e2340,_i_rssPage=6700d4e4-6714-11da-a650-0000779e2340.html`
`02:12.49`	`shmaltz`	`anybody here running Asterisk on one of those babys?`
`02:13.04`	`hads`	`I don't know but my problem is that it's a Friday afternoon and I'm working.`
`02:13.13`	`crochat`	`CallerID spoofing depends of what your gateway provider allows... it's his problem if it allows those things, no ?`
`02:14.04`	`Strom_C`	`crochat, i was making a joke`
`02:14.15`	`hads`	`shmaltz: Running Asterisk on a Sony battery?`
`02:14.48`	`shmaltz`	`hads, could be, I have to check who made those batteries`
`02:14.49`	`crochat`	`Strom_C: I'm not sure, but I think with my provider, I can make those things (discussed on a forum)...`
`02:15.21`	`hads`	`I must say it wouldn't be particually comfortable having your laptop burst into flames on your lap.`
`02:15.32`	`crochat`	`Strom_C: My provider is not very... professional :-(`
`02:16.13`	`UForgotten`	`is anyone having issues getting to vitelity?`
`02:16.15`	`crochat`	`Strom_C: But their costs are the cheapest`
`02:18.20`	`file`	`Strom_C: NUB`
`02:19.35`	`Strom_C`	`NUB NUB NUB`
`02:24.56`	`Chuji`	`Hmm, is this not right?`
`02:24.59`	`Chuji`	`exten => _5.,n,Set(${EXTEN}=${CUT(EXTEN,,4-)})`
`02:25.09`	`Chuji`	`wiki is misleading here`
`02:25.38`	`*** part/#asterisk yosyp (n=yosyp@pool-70-108-138-234.res.east.verizon.net)`
`02:26.02`	`crochat`	`Strom_C: With you callback recipe, the caller can't see that Asterisk has Hangup, right ?`
`02:29.07`	`Strom_C`	`crochat, you can do a hangup with cause without having answered the channel`
`02:30.13`	`Chuji`	`wait, maybe I can't alter ${EXTEN} here? Is that my problem?`
`02:30.22`	`hads`	`Yes`
`02:30.29`	`Strom_C`	`Chuji, your syntax is wrong`
`02:30.52`	`russellb`	`that Set command is very wrong`
`02:31.12`	`Chuji`	`apparently`
`02:31.17`	`russellb`	`what do you want to do?`
`02:31.36`	`Chuji`	`chop first 3 digits off of ${EXTEN}`
`02:31.40`	`*** join/#asterisk pbx1 (i=pbx1@netblock-66-245-193-230.dslextreme.com)`
`02:31.42`	`Chuji`	`and leave it that way`
`02:32.00`	`Strom_C`	`Chuji, why not just use substrings`
`02:32.05`	`Strom_C`	`${EXTEN:3}`
`02:32.06`	`Chuji`	`Not like ${EXTEN}:3`
`02:32.13`	`russellb`	`exten => _5.,1,Goto(${EXTEN:3}\|1)`
`02:32.27`	`Chuji`	`It doesn't alter it for good though does it?`
`02:32.38`	`hads`	`No`
`02:32.41`	`Strom_C`	`why in god's name do you want to alter it for good?`
`02:32.48`	`russellb`	`Strom_C: when you're doing a goto`
`02:32.51`	`russellb`	`that's what a goto is ...`
`02:33.11`	`Strom_C`	`russellb, i was talking to Chuji`
`02:33.13`	`Strom_C`	`:)`
`02:33.19`	`russellb`	`right`
`02:33.19`	`Strom_C`	`<3 russellb`
`02:33.21`	`russellb`	`well ...`
`02:33.26`	`russellb`	`um .. yeah`
`02:33.44`	`Chuji`	`I'm passing it from a media gateway to an internal box`
`02:33.55`	`Chuji`	`once it hits internal, I don't want the office code in it`
`02:34.11`	`russellb`	`Chuji: use what i said ..`
`02:34.36`	`Strom_C`	`chop it off in the dial statement you use to pass the call`
`02:35.08`	`Chuji`	`russellb : ${EXTEN} Never changes from the original value though right?`
`02:35.28`	`Chuji`	`So if I evaluate it later in the dialplan, I'll have to use the same substring?`
`02:35.38`	`russellb`	`no, it will be changed.`
`02:35.49`	`Chuji`	`Ok, groovy... I'll do that`
`02:35.56`	`Strom_C`	`${EXTEN} is the name of the current extension`
`02:35.56`	`russellb`	`to keep it, you'll have to do a Set(ORIGEXTEN=${EXTEN}) before the goto`
`02:36.06`	`Chuji`	`Was misunderstanding substrings`
`02:37.11`	`Chuji`	`russellb : Any of you Alabammers making the trip up here to Phreaknic this year?`
`02:37.17`	`*** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)`
`02:37.33`	`russellb`	`i'm not in Alabama right now, so not me`
`02:37.37`	`Strom_C`	`when is phreaknic again?`
`02:37.46`	`Chuji`	`Oct 20th`
`02:37.52`	`Strom_C`	`so many conventions`
`02:37.59`	`Chuji`	`so little time?`
`02:38.01`	`Chuji`	`;)`
`02:38.02`	`Strom_C`	`yes`
`02:38.34`	`Chuji`	`I've been out of the look for a year or so. Think I might make a couple this year`
`02:38.45`	`Chuji`	`Phreaknic is easy cuz I'm in Nashvegas`
`02:38.58`	`Chuji`	`look=loop btw`
`02:39.36`	`Strom_C`	`to compare nashville to las vegas would be an insult to nashville :)`
`02:40.48`	`Chuji`	`hah`
`02:42.37`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`02:44.14`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`02:44.14`	`*** mode/#asterisk [+o mog] by ChanServ`
`02:46.07`	`*** join/#asterisk ANTILOCAS (n=uoiuyiu@200.87.51.226)`
`02:46.56`	`Strom_C`	`all I want to do is find out whether AT&T California can deliver CNAM over BRI. For this I have to wait on hold twenty minutes.`
`02:47.15`	`file`	`I doubt it ... just to spite you`
`02:47.17`	`crochat`	`Strom_C: I'm trying to make the callback with the same provider, but Asterisk console says circuit-busy, and it works exactly as if I call from my mobile phone like usually... so I can't have a working callback like that :-(`
`02:47.21`	`*** part/#asterisk EtienneG (n=etienne@montreal.canonical.com)`
`02:47.36`	`Strom_C`	`crochat, you have to hang up the call before you place a new call`
`02:48.20`	`crochat`	`Strom_C: I make a hangup on Asterisk, but it's like I've never made any hangup...`
`02:48.44`	`Strom_C`	`crochat, pastebin me some extensions.conf and some console output`
`02:49.11`	`ANTILOCAS`	`which linux is good enough to install asterisk on it? could you give me the 2 more used options?`
`02:49.41`	`intralanman`	`centos and debian?`
`02:49.42`	`teleweb`	`Debian and CentOS are good choices`
`02:49.46`	`teleweb`	`lol`
`02:49.48`	`intralanman`	`rofl`
`02:49.49`	`Strom_C`	`ANTILOCAS, I like debian. Some people like centos or redhat or gentoo or....`
`02:50.07`	`intralanman`	`i personally use gentoo, but it's not used as much`
`02:50.11`	`ANTILOCAS`	`what about arch linux or lsf?`
`02:50.22`	`ANTILOCAS`	`or freeBSD`
`02:51.00`	`Strom_C`	`are you running the box with any hardware?`
`02:51.11`	`Strom_C`	`or are you running a pure-voip setup?`
`02:51.32`	`teleweb`	`astlinux :)`
`02:51.48`	`ANTILOCAS`	`puro voip`
`02:51.58`	`teleweb`	`well, I've wanted to take a look at that astlinux for some time now, anyone here using it?`
`02:52.14`	`Strom_C`	`ANTILOCAS, use whatever OS you like best`
`02:52.24`	`hads`	`Arch Linux is cool except I wouldn't really call it suited to a server because of it's rolling release cycle.`
`02:52.54`	`ANTILOCAS`	`and Asterisk is only for linux? cant be used with winxp?`
`02:53.04`	`JT`	`lol xp`
`02:53.10`	`teleweb`	`it can in a VM or with colinux`
`02:53.15`	`Strom_C`	`hhahaha`
`02:53.16`	`hads`	`But ya, what Strom_C said.`
`02:53.26`	`teleweb`	`there was some win32 build too`
`02:53.27`	`Strom_C`	`who in their right mind would run a telephony application on windows?`
`02:53.31`	`teleweb`	`but hasn't been updated`
`02:53.45`	`ANTILOCAS`	`haha`
`02:53.45`	`teleweb`	`I wouldn't run asterisk on linux`
`02:53.54`	`teleweb`	`oops`
`02:54.00`	`teleweb`	`that should be *windows of course ;)`
`02:54.04`	`JT`	`yes but you're insane`
`02:54.04`	`JT`	`well`
`02:54.07`	`ANTILOCAS`	`and with unix?`
`02:54.17`	`ANTILOCAS`	`does it work with free versions of unix?`
`02:54.18`	`JT`	`define unix`
`02:54.25`	`JT`	`name an os`
`02:54.28`	`ANTILOCAS`	`unix the operating system`
`02:54.38`	`Qwell`	`which unix?`
`02:54.38`	`hads`	`heh`
`02:54.39`	`Strom_C`	`ANTILOCAS, no, it will not run on your toaster`
`02:54.41`	`JT`	`SysV UNIX hasn't been released in a long time`
`02:54.49`	`JT`	`doesn't sound like you know what you're talking about`
`02:54.49`	`teleweb`	`I mean: I wouldn't run asterisk on windows, but there are other telephony server apps that can run perfectly fine on a Windows 2003 server`
`02:55.00`	`Strom_C`	`ANTILOCAS, nor will it run on your microwave oven`
`02:55.03`	`Qwell`	`depending on your definition of "perfectly fine"`
`02:55.13`	`ANTILOCAS`	`hah`
`02:55.22`	`Strom_C`	`i wouldn't run any telephony server app on windows ever ever ever ever ever`
`02:55.32`	`teleweb`	`let's not start that discussion`
`02:55.43`	`JT`	`if you can't even name a UNIX, i don't know why you'd be interested in running it on that`
`02:56.07`	`ANTILOCAS`	`SCO unix`
`02:56.11`	`Qwell`	`no`
`02:56.13`	`intralanman`	`haven't you guys heard windows outperform linux in every way.... or so say the M$ biased tests that they pay for all the time`
`02:56.16`	`ANTILOCAS`	`i dont remember the new free unix`
`02:56.21`	`ANTILOCAS`	`there is one`
`02:56.40`	`ANTILOCAS`	`well nevermind`
`02:56.54`	`teleweb`	`GUYS, please, don't start the flamewar of windows vs linux`
`02:56.59`	`JT`	`linux is unix-like, ANTILOCAS`
`02:57.06`	`Qwell`	`ANTILOCAS: Solaris maybe?`
`02:57.10`	`JT`	`it may not BE unix`
`02:57.11`	`Qwell`	`one of the many BSDs?`
`02:57.11`	`JunK-Y`	`opensolaris`
`02:57.17`	`Nivex`	`slowlaris?`
`02:57.26`	`Qwell`	`slow, pfft`
`02:57.36`	`ANTILOCAS`	`yeah the BSD ones`
`02:57.37`	`JT`	`on x86 stuff mostly`
`02:57.39`	`teleweb`	`ANTILOCAS: you could install Freeswitch on windows, they have an MSI installer`
`02:57.45`	`JunK-Y`	`Nivex: cause u dont have machines like Qwell :P`
`02:57.48`	`JT`	`solaris works better on sun arch stuff`
`02:57.48`	`Qwell`	`:D`
`02:57.50`	`teleweb`	`runs natively on windows`
`02:57.54`	`Qwell`	`JunK-Y: most people don't, heh`
`02:58.01`	`Qwell`	`most big businesses don't :D`
`02:58.09`	`JunK-Y`	`Qwell: maybe, at least, ya can share with me ? :P`
`02:58.10`	`JT`	`what does qwell have?`
`02:58.13`	`JT`	`an e10k`
`02:58.19`	`JunK-Y`	`a 386 running solaris`
`02:58.21`	`JunK-Y`	`!`
`02:58.22`	`Qwell`	`Qwell has some pretty badass servers at sunlabs`
`02:58.36`	`Nivex`	`yeah, I think the "slowlaris" remark was made to me in reference to x86 gear`
`02:58.38`	`JT`	`e10k?`
`02:58.44`	`*** part/#asterisk redondos (n=redondos@190.48.23.163)`
`02:58.46`	`ANTILOCAS`	`how do u tie an ATA to a voip service provider?`
`02:58.46`	`Qwell`	`lets see. quad dualcore opteron, 8 core sunfire, quad opteron, and some quad sparc`
`02:59.05`	`JT`	`not bad`
`02:59.16`	`JT`	`i have a sun enterprise 4000 at home`
`02:59.19`	`JT`	`not using it though`
`02:59.30`	`teleweb`	`ANTILOCAS: just register it with a sip account, or do you mean LOCK it for a certain ITSP?`
`02:59.33`	`Qwell`	`I have asterisk running on all of these :P`
`03:00.17`	`teleweb`	`so, has anyone here any experience with astlinux?`
`03:00.55`	`teleweb`	`0.4.3 release is out`
`03:01.17`	`teleweb`	`I wonder... how does it work for updates or packages that you want to add?`
`03:01.40`	`crochat`	`Strom_C: Did you see any wrong way in my configuration ?`
`03:01.48`	`teleweb`	`does it have a tool like apt or yum?`
`03:01.51`	`Strom_C`	`crochat, let me look`
`03:02.04`	`*** join/#asterisk fyrfaktry (i=fyrfaktr@adsl-149-219-201.pns.bellsouth.net)`
`03:02.10`	`teleweb`	`does it have an operator panel?`
`03:02.14`	`teleweb`	`does it support FOP?`
`03:02.54`	`teleweb`	`I mean, it looks like an interesting distro for a small pbx, but nearly all our clients require some sort of operator panel...`
`03:03.25`	`Chuji`	`The op_panel shouldn't run right on that * box neway IMHO`
`03:03.50`	`Chuji`	`But, if you must, it'll run in astlinux`
`03:03.57`	`ANTILOCAS`	`thanks`
`03:05.06`	`teleweb`	`Chuji: if you need to run it on another box, there is no point in using astlinux`
`03:05.20`	`*** join/#asterisk hax (n=hax@httpcraft/hax)`
`03:05.25`	`teleweb`	`I'm talking of a small setup with 5 to 10 phones...`
`03:05.30`	`teleweb`	`or a home setup`
`03:06.07`	`teleweb`	`you don't want TWO servers for such a setup`
`03:06.40`	`Chuji`	`I won't disagree. Just mearly stating I'm not a fan of running the op_panel on the same box`
`03:07.22`	`Chuji`	`can always use trixbox in that scenario`
`03:07.31`	`Chuji`	`it's all packaged up with a panel`
`03:07.43`	`Chuji`	`HUD iirc`
`03:08.16`	`JT`	`but who wants to use trixbox :P`
`03:08.43`	`Chuji`	`hah, again, not my recommendation`
`03:09.00`	`UForgotten`	`what's better than trixbox? (runs and hides)`
`03:09.05`	`teleweb`	`hm, yes, i've never wanted to use trixbox because of its custom config structure etc...`
`03:09.08`	`Chuji`	`I am using PoundKey right now at the office`
`03:09.12`	`*** part/#asterisk fyrfaktry (i=fyrfaktr@adsl-149-219-201.pns.bellsouth.net)`
`03:09.25`	`Chuji`	`The new ABE linux distro`
`03:09.30`	`teleweb`	`btw, what do you think is the best OP? still FOP? (it hasn't been updated in quite a while)`
`03:09.51`	`UForgotten`	`actualy I'd love to evaluate any asterisk gui's before we roll out our enterprise pbx next week`
`03:10.19`	`teleweb`	`escaux seems to have a nice one`
`03:10.23`	`Strom_C`	`UForgotten, dont waste your time`
`03:10.23`	`Chuji`	`teleweb : I like where HUD is going, but it's a thicker client`
`03:10.30`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`03:10.47`	`Chuji`	`Can't beat the lightweight flash app for distribution`
`03:10.48`	`teleweb`	`has HUD been updated recently? I tried the original release but it was still too buggy back then`
`03:10.53`	`UForgotten`	`what's your recommendation then, stromc?`
`03:11.23`	`Chuji`	`UForgotten : I'm going to guess he will say.... "If you want a gui, try gvim"`
`03:12.10`	`Strom_C`	`haha, exactly`
`03:12.23`	`teleweb`	`" <Chuji> teleweb : I like where HUD is going " >> my colleague here says that he has received an e-mail from fonality that they are abandoning the HUDlite project!!!`
`03:12.40`	`UForgotten`	`heh. I'm all for CLI, I don't even run x anywhere... I can handle the config editing... but my managers can't. They want something they can walk their happy asses to a web browser and click to change things and they wont pay me to design my own interface`
`03:12.42`	`Chuji`	`teleweb : that's too bad. I like the UI`
`03:12.58`	`teleweb`	`yes, I liked it too`
`03:13.02`	`teleweb`	`that's really a shame`
`03:14.08`	`teleweb`	`that's true UForgotten, we get the same questions from our clients...`
`03:14.36`	`teleweb`	`the first thing they ask: where is the web admin interface? :p`
`03:14.39`	`teleweb`	`tsss...`
`03:15.10`	`Strom_C`	`UForgotten, your managers have no more business administering the phone system than the janitor has auditing your legal documents`
`03:15.52`	`teleweb`	`I installed a nice looking FOP setup at a client, and had to show it to the boss... his first question: how can I put photos of our people on the buttons? :)`
`03:16.02`	`UForgotten`	`that reminds me, I need to ask the janitor to review my tps reports again`
`03:17.21`	`teleweb`	`btw, still have to check that one out, is it configurable in fop to put pictures on the extension buttons? :-s`
`03:20.45`	`hads`	`Exactly, a phone system is an appliance, it's not something you let anyone with a browser go and mess around with.`
`03:21.46`	`JT`	`i guess it's a question of whther the browser allows operation or administration`
`03:21.50`	`teleweb`	`hads: not with the core setup, no, but people want things like changing the name of an extension or adding a new internal extension...`
`03:22.21`	`teleweb`	`a good designed interface can let people do just that and not mess with any other part of the system or setup`
`03:22.27`	`JT`	`adding a new extension, with a traditional pabx, the installers do that for you`
`03:22.33`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`03:23.07`	`hads`	`Tell them to call their friendly phone system tech :)`
`03:23.16`	`teleweb`	`JT: not always, the installers often setup a block of internal extensions, and the client can use them when needed...`
`03:23.29`	`JT`	`then they're already set up`
`03:23.37`	`JT`	`and don't need configuration`
`03:23.49`	`teleweb`	`they still need a name configured ;)`
`03:23.50`	`JT`	`or cabling`
`03:24.04`	`JT`	`yeah, normal pabx, you can set names with a phone`
`03:24.14`	`JT`	`but it doesn't get much more complex than that`
`03:25.40`	`teleweb`	`indeed, but now for a different challenge: starting an ITSP with a partner network of VoIP providers :)`
`03:26.53`	`teleweb`	`these partners have some level of tech knowledge of course, but they still need some sort of interface to manage their "part" of a big asterisk setup in your datacenter`
`03:27.16`	`teleweb`	`you don't want them to edit .conf files`
`03:27.30`	`JT`	`yeah, they'll need something pretty specific setup`
`03:27.38`	`JT`	`with well defined limitations`
`03:28.26`	`teleweb`	`true, and that's a challenge I'm working on right now :)`
`03:28.44`	`teleweb`	`first thing was the design of a good dial plan`
`03:28.45`	`*** part/#asterisk ANTILOCAS (n=uoiuyiu@200.87.51.226)`
`03:29.17`	`teleweb`	`with a prefix for every partner, and their own numbering space after that`
`03:31.29`	`teleweb`	`well, i'm off to do some more programming, cu :)`
`03:36.03`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`03:45.25`	`hax`	`kind of an odd question... but say i needed someone to dial in letters and numbers, like a short code... 'a7zx' or something... and have asterisk parse that somehow... is that possible? is there an approach for doing that?`
`03:51.21`	`*** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir)`
`03:51.31`	`*** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au)`
`03:53.04`	`*** join/#asterisk }btorch{ (n=btorch@c-66-176-87-59.hsd1.fl.comcast.net)`
`03:53.11`	`JT`	`almost everything is _possible_`
`03:53.34`	`JT`	`and i'm sure that is`
`03:53.40`	`JT`	`it wouldn't be super simple though`
`03:56.35`	`hax`	`well, i meant "would it be super simple" :)`
`03:56.37`	`hax`	`i didn't think so`
`03:56.49`	`hax`	`but i figured i'd ask anyway, you know, in case there was something i hadn't thought of`
`04:00.18`	`fafnir`	`oh oh`
`04:00.22`	`fafnir`	`this sounds interesting`
`04:00.27`	`fafnir`	`what are you doing?`
`04:04.41`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`04:12.21`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`04:12.34`	`*** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`04:13.29`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`04:13.30`	`*** mode/#asterisk [+o mog] by ChanServ`
`04:17.30`	`}btorch{`	`does anyone here know what's the difference between yate and asteisk and if yateclient phone works with asterisk ?`
`04:20.32`	`*** join/#asterisk adorah (n=Administ@87.68.146.95.cable.012.net.il)`
`04:24.12`	`hax`	`fafnir: its just for a customer support code thing`
`04:24.20`	`hax`	`fafnir: well, this is very preliminary`
`04:24.43`	`hax`	`fafnir: the idea was to use the system already in place, say a login or some short code, and have them type it in on the phone`
`04:25.05`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`04:26.48`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`04:32.56`	`*** join/#asterisk mivck (i=1000@200.114.70.228)`
`04:34.38`	`*** join/#asterisk dieno (n=Dieno@124.29.194.150)`
`04:35.52`	`*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221)`
`04:39.19`	`*** part/#asterisk [Airwolf] (n=airwolf@83.98.235.221)`
`04:39.53`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`04:40.18`	`dieno`	`please help me to setup IAX2 Account by using DeStar`
`04:51.46`	`*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)`
`04:51.47`	`*** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au)`
`04:53.27`	`*** join/#asterisk hatamen (i=hatamen@222.183.32.11)`
`04:54.58`	`hatamen`	`hi,can asterisk support srtp???`
`04:55.34`	`*** join/#asterisk MGSsancho (n=sancho@adsl-67-126-128-134.dsl.irvnca.pacbell.net)`
`04:55.52`	`hatamen`	`hello?`
`04:56.18`	`MGSsancho`	`hi`
`04:56.41`	`*** part/#asterisk }btorch{ (n=btorch@c-66-176-87-59.hsd1.fl.comcast.net)`
`04:57.07`	`hatamen`	`can asterisk support srtp???`
`05:00.45`	`hatamen`	`can asterisk support srtp??? somebody know??`
`05:03.04`	`MGSsancho`	`i dont know`
`05:03.39`	`wunderkin`	`try asking 5 more times`
`05:04.31`	`hatamen`	`can asterisk support srtp???`
`05:04.35`	`hads\|home`	`~google srtp asterisk`
`05:04.41`	`hatamen`	`can asterisk support srtp???`
`05:05.09`	`Nugget`	`stop asking. if we didn't know four minutes ago we don't magically know now.`
`05:05.17`	`hads\|home`	`Telnet!`
`05:05.40`	`hads\|home`	`Nugget: You given up on telnet?`
`05:05.52`	`Nugget`	`telnet is evil.`
`05:05.58`	`hads\|home`	`Heh.`
`05:08.40`	`hatamen`	`sorry`
`05:09.19`	`benjk`	`you have to ask 42 times over a period of 42^42 seconds`
`05:09.34`	`benjk`	`then the asnwer will be calculated by Deep Thought`
`05:10.29`	`benjk`	`that will take another 42^42^42 seconds though`
`05:10.34`	`Juggie`	`the answer to srtp is yes`
`05:10.37`	`Juggie`	`and if you LOOKED`
`05:10.37`	`benjk`	`so be patient`
`05:10.38`	`Juggie`	`you could find it`
`05:11.42`	`benjk`	`somebody told me about this unbelievable thing`
`05:11.50`	`Juggie`	`google`
`05:11.51`	`Juggie`	`OR`
`05:11.53`	`benjk`	`I think it was called something like er`
`05:11.55`	`Juggie`	`log onto the bug tracker`
`05:11.56`	`Juggie`	`and search`
`05:11.57`	`benjk`	`yes, google`
`05:11.58`	`Juggie`	`crazy ideas`
`05:12.00`	`benjk`	`that was the thing`
`05:12.31`	`benjk`	`do you think this google thing really exists?`
`05:13.11`	`benjk`	`its probably a government disinformation conspiracy, so better not use it`
`05:15.28`	`benjk`	`does anybody here ever use apps, context or variable names that go over 6 lines (80 chars * 6 lines = 512 chars) ?`
`05:16.00`	`benjk`	`perhaps more interesting, what's the maximum length you ever used for a context or variable name?`
`05:17.15`	`*** join/#asterisk techie (n=gus@antibala.com)`
`05:18.53`	`mog`	`ive used 200 or so before benjk`
`05:20.46`	`*** join/#asterisk _mh (n=largo@202.5.145.13)`
`05:27.17`	`*** join/#asterisk sumasuma (n=sumkji@61.14.86.23)`
`05:28.13`	`sumasuma`	`Is there is any linux iax client? open source to understand the iax2 code`
`05:28.41`	`JT`	`there are`
`05:28.47`	`JT`	`check out voip-info.org`
`05:29.11`	`sumasuma`	`JT: before i came here, i visited this page http://www.voip-info.org/wiki/view/Asterisk+IAX+clients`
`05:29.23`	`sumasuma`	`JT: is this not the page ?`
`05:30.12`	`JT`	`looks right`
`05:30.25`	`JT`	`what's wrong, is not a single one of all those softphones open source?`
`05:30.46`	`sumasuma`	`i guess so, on linux`
`05:31.02`	`sumasuma`	`i have seen kiax, it is all for windows based`
`05:31.11`	`sumasuma`	`Page looks right ?`
`05:31.24`	`sumasuma`	`you been to any of those websites and downloaded the code ?`
`05:31.47`	`JT`	`if you're just interested in the code, why not?: http://www.voip-info.org/wiki/view/IAXClient`
`05:35.28`	`sumasuma`	`JT: yes, seen this page ? http://sourceforge.net/project/showfiles.php?group_id=72851`
`05:35.42`	`dieno`	`sumsuma :`
`05:35.49`	`sumasuma`	`it says iaxclient in sourceforgenet !`
`05:36.32`	`*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`05:38.27`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`05:39.27`	`JT`	`sumasuma: http://iaxclient.sourceforge.net/devinfo.html`
`05:39.28`	`JT`	`use svn`
`05:39.32`	`*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`05:41.34`	`*** join/#asterisk twisted[work] (n=twisted@pdpc/supporter/active/twisted)`
`05:41.34`	`*** mode/#asterisk [+o twisted[work]] by ChanServ`
`05:42.13`	`Strom_C`	`you're up late, twisted`
`05:42.27`	`mog`	`Strom_C, what you up to`
`05:42.51`	`Strom_C`	`proofread and printed up tomorrow's test`
`05:43.40`	`Strom_C`	`listening to neil snore like a boeing`
`05:43.48`	`mog`	`lewis?`
`05:43.51`	`Strom_C`	`yeah`
`05:43.54`	`mog`	`lol`
`05:57.57`	`*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)`
`06:01.09`	`harryvv`	`Does anyone know of any auto dial phones on the market`
`06:03.15`	`*** join/#asterisk ANTILOCAS (n=uoiuyiu@200.87.51.226)`
`06:03.49`	`benjk`	`define auto=dial phone`
`06:04.55`	`harryvv`	`phone does not have a keypad and it will dial a pre set number`
`06:05.04`	`harryvv`	`when the reciver is picked up`
`06:06.00`	`benjk`	`the Barbietones have such a feature, but they do have a keypad`
`06:06.25`	`harryvv`	`barbietones?`
`06:07.57`	`wunderkin`	`heh heh :)`
`06:08.02`	`benjk`	`yeah, Barbietones from Grandstream`
`06:08.55`	`harryvv`	`okay`
`06:09.37`	`harryvv`	`how about a outside phone that does this.`
`06:17.12`	`Strom_C`	`harryvv, either you provision the line as a ringdown or you get an adjunct module`
`06:19.18`	`Snake-Eyes`	`Is there a way to play a sound file when some tries to place a call with Asterisk but isn't a user on the system ?`
`06:20.02`	`Snake-Eyes`	`eg normally a sip message 404 would be sent but instead a sound found is played and then hangup`
`06:20.21`	`Snake-Eyes`	`found = file`
`06:21.20`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.254)`
`06:22.46`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`06:27.18`	`*** join/#asterisk vistac (n=vistac@211.22.77.218)`
`06:31.30`	`*** join/#asterisk daysmen3 (n=primus@host86-143-6-179.range86-143.btcentralplus.com)`
`06:33.16`	`*** join/#asterisk BugKham (n=bugkham@ppp-58.8.7.125.revip2.asianet.co.th)`
`06:35.33`	`*** join/#asterisk anthonyu (i=user5211@netblock-68-183-123-247.dslextreme.com)`
`06:35.54`	`anthonyu`	`is there an extensions.conf command for writing a line to the logfile?`
`06:37.34`	`*** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com)`
`06:37.36`	`*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)`
`06:38.25`	`FarrisG`	`I've just added 6 channels to my PRI. It seems to be configured fine, but is there any way to really test it, short of attempting to make a bunch of outgoing/incoming calls at once?`
`06:38.43`	`*** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com)`
`06:39.11`	`*** join/#asterisk tabmeister1 (n=troy@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com)`
`06:39.16`	`tabmeister1`	`does asterisk require a soundcard?`
`06:39.27`	`FarrisG`	`tabmeister1: No`
`06:39.39`	`wunderkin`	`FarrisG: call using the specific channel..?`
`06:40.04`	`tabmeister1`	`FarrisG, i get a broken pipe error when trying to start asterisk... any ideas?`
`06:40.06`	`FarrisG`	`wunderkin: I didn't think of that. How would I go about specifying which zap chan to use?`
`06:40.31`	`tabmeister1`	`Ouch ... error while writing audio data: : Broken pipe`
`06:40.47`	`FarrisG`	`tabmeister1: Sounds to me like some piece didn't build properly`
`06:41.19`	`tabmeister1`	`specifically: - codec_lpc10: using generic PLC`
`06:41.39`	`tabmeister1`	`i have never encountered this error before :)`
`06:42.42`	`tabmeister1`	`should i try the later build?`
`06:43.14`	`wunderkin`	`Zap/chan/..`
`06:43.42`	`vistac`	`hello sir`
`06:44.09`	`vistac`	`does anyone know where is the default system voice file locate :)`
`06:44.47`	`wunderkin`	`/var/lib/asterisk/sounds`
`06:45.13`	`vistac`	`thnaks`
`06:45.14`	`hads\|home`	`setup in asterisk.conf`
`06:45.19`	`FarrisG`	`wunderkin: Should I create a test context in extensions.conf to do that?`
`06:46.07`	`wunderkin`	`you dont need a whole new context, you can just add an exten`
`06:48.13`	`vistac`	`Which program can I play the .gsm voice file ?`
`06:48.57`	`tabmeister1`	`damn it still same error`
`06:49.11`	`wunderkin`	`tabmeister1, you need to look further up`
`06:49.31`	`tabmeister1`	`wunderkin, it what respect?`
`06:49.58`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`06:50.23`	`wunderkin`	`on the scroll when you start asterisk`
`06:50.35`	`tabmeister1`	`i must be out of it`
`06:50.49`	`tabmeister1`	`when i start asterisk, i get that error?`
`06:51.02`	`wunderkin`	`...`
`06:51.06`	`wunderkin`	`do you? what error?`
`06:51.17`	`tabmeister1`	`[codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder)`
`06:51.17`	`tabmeister1`	`<PROTECTED>`
`06:51.17`	`tabmeister1`	`<PROTECTED>`
`06:51.17`	`tabmeister1`	`Ouch ... error while writing audio data: : Broken pipe`
`06:51.17`	`tabmeister1`	`Segmentation fault`
`06:51.24`	`wunderkin`	`thats nothing`
`06:52.14`	`tabmeister1`	`really?`
`06:53.15`	`tabmeister1`	`i mean how do i start asterisk if i get that error`
`06:54.38`	`*** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it)`
`06:54.51`	`wunderkin`	`you must be doing something wrong if you are getting a seg fault, are you using trunk, freebsd, ... i dunno`
`06:55.50`	`FarrisG`	`wunderkin: Thanks! It worked. That was a hell of a lot easier than I was expecting. I'm a little depressed that something so expensive (adding 6 more channels to our PRI) was so damn easy to implement :)`
`06:56.10`	`wunderkin`	`wrong optimizations or something`
`06:56.38`	`tabmeister1`	`wunderkin, could it be the kernel i am using?`
`06:58.56`	`wunderkin`	`dunno, does it always stop at the same place`
`06:59.11`	`*** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it)`
`06:59.51`	`tabmeister1`	`wunderkin, well it compiles fine.. it's just starting asterisk is the problem`
`07:01.07`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`07:02.41`	`tabmeister1`	`actuall wunderkin`
`07:02.47`	`tabmeister1`	`it doesnt exit it just hangs..`
`07:02.54`	`*** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp)`
`07:03.04`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-96-211.red.bezeqint.net)`
`07:03.06`	`tabmeister1`	`after broken pipe it doesnt give me a command prompt again it just hangs`
`07:03.38`	`tzafrir_laptop`	`wiki Q.: can anybody take a look at http://voip-info.org/wiki/view/Asterisk+config+sip.conf and tell me why no TOC generated?`
`07:03.48`	`tzafrir_laptop`	`Who do I aks this?`
`07:03.56`	`BugKham`	`anyone knows if we need to answer a call before "RxFax" using spandsp?`
`07:04.04`	`tzafrir_laptop`	`(apart from "everybody" in the shoutbox, that is)`
`07:04.50`	`BugKham`	`I actuallty did "answer" it but the fax thing still doesn't work`
`07:05.44`	`tzafrir_laptop`	`I tend to believe it will work both ways.`
`07:06.33`	`BugKham`	`tzafrir_laptop, r u using spandsp?`
`07:07.36`	`*** join/#asterisk xnon (i=xnon@200.82.222.64)`
`07:11.30`	`tzafrir_laptop`	`BugKham, no. However, do you have fax detection enabled in zapata.conf?`
`07:11.49`	`tzafrir_laptop`	`If so: asterisk must answer the call to hear its beginning`
`07:13.25`	`BugKham`	`tzafrir_laptop, let me see`
`07:14.06`	`BugKham`	`tzafrir_laptop, yes, I have faxdetect=both`
`07:14.22`	`*** join/#asterisk DGnome (i=dgnome@slashbox.net)`
`07:14.27`	`tzafrir_laptop`	`Why do you need fax detection for outgoing calls?`
`07:15.11`	`DGnome`	`Heyho! Whats the status on video conferance features?`
`07:15.33`	`Qwell`	`DGnome: What specifically would you like to know?`
`07:16.08`	`BugKham`	`tzafrir_laptop, don't really know if I need`
`07:16.41`	`BugKham`	`tzafrir_laptop, it's my first day testing it`
`07:18.52`	`DGnome`	`Qwell: Can I do videoconference with Ekiga against asterisk using SIP protocol?`
`07:19.01`	`Qwell`	`DGnome: sure`
`07:19.41`	`DGnome`	`Qwell: straight out of the box? or does one need extra software patches or something?`
`07:20.02`	`Qwell`	`I don't know about 1.2, but trunk and 1.4 (when it's released) will work just fine`
`07:20.43`	`DGnome`	`Great :)`
`07:23.56`	`DGnome`	`Qwell: is there a roadmap for 1.4? Would like to see what's still pending and todo so to say`
`07:24.17`	`DGnome`	`To get an idea of how far the release is`
`07:24.26`	`Qwell`	`it's...close`
`07:24.35`	`tabmeister1`	`Qwell, i have a segfault error when i attempt to start asterisk any ideas?`
`07:24.57`	`hads`	`mpg123?`
`07:25.01`	`DGnome`	`Qwell: okay, thanks for the info :)`
`07:25.16`	`tabmeister1`	`hads, i already installed it from scratch :)`
`07:25.19`	`Qwell`	`DGnome: If you feel like testing bugs...`
`07:25.26`	`Qwell`	`DGnome: there are quite a few that could use it`
`07:26.02`	`hads`	`tabmeister1: Yes, I believe that error is from mpg123`
`07:26.08`	`Qwell`	`some stuff that would be nice to have in 1.4, that may not necessarily make it - unless they get some testing`
`07:26.21`	`tabmeister1`	`hads, alrighty any thoughts on what i can do to correct it?`
`07:26.50`	`hads`	`tabmeister1: Sorry, I don't use mpg123`
`07:26.58`	`tabmeister1`	`thanks`
`07:27.03`	`tabmeister1`	`Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp.`
`07:34.07`	`*** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it)`
`07:35.07`	`vistac`	`have any one know how to integrate Asterisk with LDAP or RADIUS?`
`07:35.56`	`vistac`	`let Asterisk load SIP user configuration and extention from LDAP atuomaticlly realtime?`
`07:37.33`	`vistac`	`hello`
`07:46.54`	`corruptor`	`<PROTECTED>`
`07:57.59`	`[Airwolf]`	`vistac, I believe there is no realtime implementation for LDAP right now.`
`08:00.01`	`[Airwolf]`	`A wait`
`08:00.09`	`[Airwolf]`	`LDAP is availible`
`08:00.10`	`[Airwolf]`	`http://www.voip-info.org/wiki/view/Asterisk+RealTime`
`08:00.14`	`[Airwolf]`	`have fun :)`
`08:02.39`	`*** join/#asterisk lukketto (n=lukketto@host156-134.pool8257.interbusiness.it)`
`08:03.10`	`*** part/#asterisk hatamen (i=hatamen@222.183.32.11)`
`08:04.33`	`*** join/#asterisk patient_zero (n=tjzewez@209.107.188.100)`
`08:05.23`	`patient_zero`	`can anyone tell me how to disable acpi if I cant in my bios?`
`08:05.47`	`intralanman`	`dude, i think you're on the wrong channel`
`08:06.01`	`patient_zero`	`i read somewhere that it may cause dropped calls`
`08:06.23`	`zoa`	`hello hello`
`08:08.11`	`BugKham`	`can we process other things while playing a sound file?`
`08:09.33`	`*** join/#asterisk tiab (i=214819@bud.cc.swin.edu.au)`
`08:09.36`	`BugKham`	`and stop playing whenever we want it to`
`08:10.00`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`08:15.38`	`tzafrir_laptop`	`patient_zero, why do you need to disable acpi?`
`08:16.03`	`*** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp)`
`08:16.23`	`tzafrir_laptop`	`anyway, try to look into kernel options such as 'pci=noacpi'`
`08:22.05`	`patient_zero`	`thanks`
`08:23.03`	`*** join/#asterisk Vec (n=Vector@dsl-146-99-57.telkomadsl.co.za)`
`08:33.54`	`*** join/#asterisk Ahrimanes (n=michael@81.7.159.2)`
`08:34.46`	`Ahrimanes`	`morning`
`08:46.20`	`*** join/#asterisk svenna_ (n=svenna@p548D1FBD.dip0.t-ipconnect.de)`
`08:54.31`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`08:56.41`	`*** join/#asterisk nailbags (i=someone@c220-237-123-137.randw1.nsw.optusnet.com.au)`
`08:56.54`	`*** join/#asterisk Anotsu (n=kvirc@a81-14-187-199.net-htp.de)`
`08:58.33`	`anthonyu`	`is there an extensions.conf command for writing a line to the logfile?`
`08:59.37`	`Ahrimanes`	`Noop`
`09:00.01`	`anthonyu`	`do you have to increase the verbosity somewhere? mine aren't going through to the messages log`
`09:00.34`	`Ahrimanes`	`hm dont quite remember`
`09:01.17`	`anthonyu`	`k, thanks anyway`
`09:06.18`	`JT`	`you can get asterisk to log`
`09:07.20`	`anthonyu`	`that's what i'm looking for`
`09:07.31`	`*** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue)`
`09:12.05`	`Anotsu`	`verbosity must be at least set to 5 i think to see noop messages`
`09:12.46`	`*** join/#asterisk pbx1 (i=pbx1@netblock-66-245-193-230.dslextreme.com)`
`09:13.11`	`anthonyu`	`ty. can i set that in a config file, or just with -vvvvv?`
`09:17.13`	`hads\|home`	`Either that or 'set verbose 5' from the CLI`
`09:19.44`	`anthonyu`	`ty. so it's either -vvvvv or the cli? no config file entry anywhere?`
`09:20.00`	`Anotsu`	`correct`
`09:20.01`	`*** join/#asterisk inspired (n=mikael@85.221.0.46)`
`09:20.07`	`anthonyu`	`thanks again`
`09:27.04`	`*** join/#asterisk phearless (n=phearles@host81-138-68-106.in-addr.btopenworld.com)`
`09:27.23`	`tzafrir_laptop`	`anthonyu, asterisk.conf , yes`
`09:27.26`	`phearless`	`asterisk is great !`
`09:27.29`	`phearless`	`(when it works)`
`09:27.45`	`tzafrir_laptop`	`however it will also trigger a -f`
`09:27.57`	`tzafrir_laptop`	`unless you use trunk and also set -F`
`09:28.24`	`tzafrir_laptop`	`(The current 1.2 debian debs have a trivial patch to apply an equivalent switch, -D)`
`09:28.33`	`shodan`	`hmm I have one weirdo problem , whenever * transcode audio , it introduce crackles and pop in the sound , at first I thought they were just gsm artifact , so what I did is I set my sip config to disallow=all allow=ulaw (my original files are ulaw) and the crackles and pops are gone , then I changed to allow=alaw and even for alaw I get the exact same crackle and pops then I get from gsm ?! how weird is that !`
`09:29.03`	`*** join/#asterisk evilmnky (i=evilmnky@69-163-156-13.atlsfl.adelphia.net)`
`09:29.38`	`hads\|home`	`tzafrir_laptop: Huh?`
`09:30.11`	`tzafrir_laptop`	`hads, an option to start asterisk with verbose/debug but still daemonize`
`09:30.11`	`evilmnky`	`#asterisk on freenode`
`09:31.08`	`anthonyu`	`that'd be nice`
`09:31.17`	`hads\|home`	`What does that have to do with asterisk.conf though?`
`09:31.47`	`*** join/#asterisk Nebukadneza (n=daddel9@i3ED6F2F7.versanet.de)`
`09:33.13`	`*** join/#asterisk End3r (n=end3r@voodeedoo.org)`
`09:33.22`	`tzafrir_laptop`	`because you can set the command-line option of -v in asterisk.conf as well, IIRC`
`09:33.31`	`anthonyu`	`all i see are directories in asterisk.conf docs`
`09:33.34`	`phearless`	`member => SIP/200,1`
`09:33.34`	`phearless`	`member => SIP/201,2`
`09:33.34`	`phearless`	`member => SIP/202,3`
`09:33.40`	`phearless`	`I have put that in queues.conf`
`09:33.43`	`phearless`	`it's fine ?`
`09:33.47`	`tzafrir_laptop`	`But maybe this is just trunk`
`09:33.53`	`anthonyu`	`ah`
`09:34.06`	`tzafrir_laptop`	`asterisk.conf is not "documented" by asamples file`
`09:34.25`	`phearless`	`exten => s,1,Dial(SIP/200,30)`
`09:34.31`	`phearless`	`and this in extensions.conf`
`09:34.48`	`phearless`	`but if somebody call, and 200 is busy, 201 do NOT receive the call :(`
`09:35.30`	`hads\|home`	`tzafrir_laptop: Ah, quite right. I hadn't noticed doc/asterisk-conf.txt`
`09:36.49`	`phearless`	`?`
`09:38.26`	`anthonyu`	`i found it here in the asterisk book, [options] verbose=<value>`
`09:39.43`	`phearless`	`?`
`09:40.23`	`hads\|home`	`If it's in the book then it's quite possibly in 1.2`
`09:42.29`	`Anotsu`	`[options] verbose=<value> put where ? in asterisk.conf?`
`09:44.52`	`anthonyu`	`yeah`
`09:45.08`	`anthonyu`	`you have to make the [options] header if it's not there`
`09:45.17`	`anthonyu`	`then verbose=5`
`09:56.39`	`*** part/#asterisk lukketto (n=lukketto@host156-134.pool8257.interbusiness.it)`
`10:10.32`	`RoyK`	`.`
`10:12.48`	`*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)`
`10:13.51`	`*** join/#asterisk xnon (i=xnon@200.82.222.64)`
`10:16.54`	`*** join/#asterisk L-info (n=Adam@62.69.102.99)`
`10:18.24`	`*** part/#asterisk Ahrimanes (n=michael@81.7.159.2)`
`10:28.34`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`10:31.20`	`phearless`	`hello !`
`10:31.32`	`phearless`	`anybody know how to make a BASIC queue ?`
`10:31.37`	`phearless`	`without agents`
`10:31.51`	`inspired`	`member => SIP/username`
`10:31.57`	`phearless`	`exten => s,1,Dial(SIP/200,30)`
`10:32.06`	`phearless`	`this is for the ext'`
`10:32.11`	`phearless`	`and in queues :`
`10:32.25`	`phearless`	`member => SIP/200,1`
`10:32.25`	`phearless`	`member => SIP/201,2`
`10:32.25`	`phearless`	`member => SIP/202,3`
`10:32.38`	`phearless`	`but if 200 is busy, 201 do NOT ring :(`
`10:32.54`	`inspired`	`don't put 1,2,3`
`10:33.04`	`phearless`	`ok`
`10:33.09`	`inspired`	`try if that works petter`
`10:33.44`	`*** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk)`
`10:34.07`	`phearless`	`does not work`
`10:35.01`	`inspired`	`hmm, what kind of phones do you use?`
`10:35.08`	`phearless`	`http://paste-bin.com/181`
`10:35.11`	`phearless`	`check my logs`
`10:35.29`	`inspired`	`when I did that with x-lite I had a problem since the extra lines on the x-lite would be ringing, which was not what I was after`
`10:35.32`	`phearless`	`I got 2 cisco 7960`
`10:35.44`	`inspired`	`what version of asterisk are you using?`
`10:35.47`	`phearless`	`200 is one and only one phone`
`10:35.51`	`phearless`	`201 is one and only one phone`
`10:36.03`	`phearless`	`asterisk 1.2.10`
`10:36.08`	`phearless`	`please check http://paste-bin.com/181`
`10:36.20`	`inspired`	`do show queues`
`10:36.23`	`inspired`	`and paste-bn`
`10:36.39`	`phearless`	`during the call?`
`10:37.35`	`phearless`	`No queues.`
`10:37.45`	`phearless`	`this is weird`
`10:38.02`	`*** join/#asterisk xorotude (n=labsoard@mail.xorotude.com)`
`10:38.37`	`inspired`	`not during the call. just from the asterisk CLI`
`10:39.06`	`phearless`	`No queues.`
`10:39.06`	`phearless`	`same`
`10:39.38`	`inspired`	`paste queues.conf to pastebin`
`10:40.07`	`phearless`	`[general]`
`10:40.07`	`phearless`	`persistentmembers = yes`
`10:40.17`	`phearless`	`+ the 3 lines that I pasted before i the chan`
`10:40.22`	`phearless`	`all the rest is comments`
`10:40.42`	`inspired`	`well, you haven't defined any queues`
`10:40.59`	`inspired`	`members belong to queues, and queues can have many members`
`10:41.11`	`inspired`	`google config queues.conf`
`10:41.36`	`hads\|home`	`There might be something in the book.`
`10:41.51`	`nettie`	`Hello guys, I have a couple of ip phones behind NAT connecting a remote asterisk server. When I make a call and the called party picks there's a 2 seconds gap before the called party can hear the caller. I think it might be realted to some RTP and NAT, some of the packets get lost before the router actually open the ports and makes the transaltion. Does anyone know if this makes sense and if there's a way to fix it? Thanx in advance.`
`10:42.27`	`phearless`	`http://www.asteriskguru.com/tutorials/queues_conf.html`
`10:42.36`	`phearless`	`2) If you do not use the agents.conf file. The so called static way. The disadvantage of this way is that the user is bind to one work station(the one on which is registered) and it cannot login into the queue from another work station. The form is: member => Agent/interface/user>,<penalty>`
`10:42.38`	`phearless`	`I used that`
`10:43.22`	`phearless`	`maybe I miss a "[default]"`
`10:43.26`	`phearless`	`or something like this`
`10:43.30`	`inspired`	`nettie, are you using port triggering or port range forwarding in your router?`
`10:43.41`	`inspired`	`you are probably missing [my-queue-name]`
`10:43.49`	`inspired`	`and then add your members and settings for this queue`
`10:44.24`	`phearless`	`but how do I "call" [my-queue-name] ?`
`10:44.38`	`inspired`	`Queue(my-queue-name)`
`10:44.39`	`phearless`	`I have to put my-queue-name somewhere else, no ?`
`10:44.42`	`phearless`	`ahh!`
`10:44.43`	`phearless`	`thank you`
`10:44.45`	`inspired`	`;-)`
`10:44.45`	`inspired`	`np`
`10:44.49`	`nettie`	`inspired port forwarding`
`10:45.05`	`inspired`	`nettie, hmm, then the port should be open all the time. strange`
`10:45.11`	`nettie`	`uhmm`
`10:45.24`	`nettie`	`but wait`
`10:45.24`	`nettie`	`IO`
`10:45.25`	`nettie`	`I`
`10:45.36`	`nettie`	`where I have the problem I'm not actually forwarding`
`10:46.11`	`nettie`	`I have multiple phones`
`10:46.39`	`nettie`	`they're on dhcp and they use random ports fro rtp`
`10:47.43`	`phearless`	`thanks inspired it works`
`10:47.56`	`inspired`	`:)`
`11:08.23`	`xorotude`	`Hi all..`
`11:08.35`	`xorotude`	`I have a realy simple question`
`11:08.47`	`xorotude`	`I have a Duron 1100MHz with 256MB ram`
`11:08.58`	`xorotude`	`it's running a 2.6 linux kernel`
`11:09.02`	`xorotude`	`postfix`
`11:09.05`	`xorotude`	`apache and mysql`
`11:09.26`	`xorotude`	`and iptables firewall as an internet gateway`
`11:09.50`	`xorotude`	`should it also be able to handle running asterisk ok for a single line?`
`11:10.26`	`xorotude`	`ooh, and I have a 512/128k ADSL link`
`11:10.46`	`RaYmAn-Bx`	`easily. depending on the number of users and whether it has to transcode`
`11:11.08`	`RaYmAn-Bx`	`I have a via epia 1ghz doing roughly the same as yours..has 512mb ram though`
`11:11.36`	`xorotude`	`maybe transcode... I am going to get an SPA-3000 ATA`
`11:12.05`	`xorotude`	`I have no web sites to speak of..`
`11:12.07`	`RaYmAn-Bx`	`should still be able to handle it for a single line i'd say..might want to get more ram though..can't cost a whole lot`
`11:12.10`	`xorotude`	`but I'm working on it`
`11:12.30`	`xorotude`	`unfortunately it's SDRAM... but I shoul dbe able to find something`
`11:12.41`	`RaYmAn-Bx`	`you can prolly get some used fairly easily`
`11:12.48`	`xorotude`	`indeed..`
`11:13.04`	`RaYmAn-Bx`	`the inet line might a tad on a the low side for voip though (upload in particular)`
`11:13.35`	`RoyK`	`http://www.ryanair.com/site/EN/notices.php?notice=060822-ASP-EN`
`11:13.37`	`xorotude`	`until I get the spa3000 I plan to use a software sip phone.. .hoping to be able to at least be able to connect to another user's asterisk box`
`11:13.58`	`*** join/#asterisk morex (i=morex@host86-133-26-102.range86-133.btcentralplus.com)`
`11:14.00`	`morex`	`Hey all`
`11:14.06`	`xorotude`	`yeah, well ADSL in australia sucks.. I wish we could get 512/256 like you can in the USA`
`11:14.11`	`morex`	`Anybody got any experience with using Yate with Asterisk?`
`11:14.21`	`morex`	`I'm trying to get the h323 <-> sip gateway working...`
`11:14.48`	`RaYmAn-Bx`	`xorotude: I have 8mbit/1mbit adsl =P (well adsl2+)..just to brag a bit ;)`
`11:14.59`	`*** join/#asterisk erwinism (n=erwinpog@203.115.172.242)`
`11:15.06`	`xorotude`	`yeah.. have to get me one of those..`
`11:15.30`	`erwinism`	`hello, how can i set the ring count to zero before the IVR starts?`
`11:15.30`	`xorotude`	`I believe adsl (1) modems can do the 8/1 on an adsl 2+ line.. is that right?`
`11:17.15`	`ltd`	`<PROTECTED>`
`11:18.05`	`erwinism`	`anyone can help me for my IVR starts automatically when theres a call?`
`11:19.36`	`xorotude`	`cool.. might just have to upgrade to adsl2+ then.. and keep the current modem..`
`11:19.48`	`xorotude`	`but I also hear the new modems have better line diagnostics`
`11:20.44`	`breakdisk`	`hi guys,anyone experience noisy lines using tdm400`
`11:22.05`	`erwinism`	`breakdisk: i used that card but have no noise`
`11:23.35`	`xorotude`	`ahh.. and here's an important question for when I get my spa-3000`
`11:23.51`	`xorotude`	`I'm looking on ebay at the moment, and there's a co-branded box going cheap`
`11:24.04`	`xorotude`	`I'm guessing it's locked to the provider that sold the box though`
`11:24.20`	`xorotude`	`does anyone have a link on how to 'unlock' an SPA-3000?`
`11:24.27`	`xorotude`	`I've seen plenty for the 2000`
`11:24.33`	`xorotude`	`but none specifically for the 3000`
`11:26.55`	`breakdisk`	`erwinism:yes,correct i used it for 5months.but now im experiencing noise.i formatted my server already.`
`11:27.43`	`breakdisk`	`erwinism:i upgrade also my asterisk and zaptel software.`
`11:28.01`	`breakdisk`	`is there a troubleshooting comman to know if the card has a problem.`
`11:28.28`	`erwinism`	`what is the noise?`
`11:28.39`	`erwinism`	`like hizz? or white noice?`
`11:28.52`	`breakdisk`	`is there any command for diagnosing tdm card?`
`11:28.54`	`*** join/#asterisk backblue (n=igor@82.102.1.42)`
`11:29.04`	`backblue`	`hi`
`11:29.41`	`breakdisk`	`combination of static and hizz noise`
`11:35.58`	`*** join/#asterisk heffernan (n=hwt@195.139.204.157)`
`11:36.48`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`11:38.54`	`*** part/#asterisk xorotude (n=labsoard@mail.xorotude.com)`
`11:41.36`	`backblue`	`hello party people!`
`11:43.18`	`puzzled`	`morning`
`11:48.22`	`erwinism`	`breakdisk: i think there is no tool for that`
`11:48.44`	`erwinism`	`breakdisk: what module you have? do you have fxs?`
`11:51.59`	`*** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net)`
`11:54.31`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`11:59.22`	`breakdisk`	`erwinism:no i dont have fxs only fxo`
`12:00.14`	`*** join/#asterisk potsboy (n=chrisg@196.211.16.202)`
`12:07.15`	`*** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se)`
`12:09.51`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`12:11.49`	`*** join/#asterisk jpeeler (n=jbpeele@130-127-44-95.chouse.resnet.clemson.edu)`
`12:14.16`	`Sonderblade`	`is there an asterisk command to get the unix time?`
`12:14.25`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`12:15.07`	`*** join/#asterisk daysmen3 (n=primus@host86-143-6-179.range86-143.btcentralplus.com)`
`12:17.05`	`FlatFoot`	`Sonderblade: SayUnixTime()`
`12:17.10`	`*** join/#asterisk oej (n=oej@63.117.53.60)`
`12:17.35`	`Sonderblade`	`FlatFoot: i want to get it as an int or string, saying it is pointless`
`12:18.38`	`FlatFoot`	`fair enough thats the only time i have used it`
`12:18.40`	`FlatFoot`	`exten => 1234,1,SayUnixTime()`
`12:18.50`	`*** join/#asterisk oej (n=oej@63.117.53.60)`
`12:20.54`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`12:24.20`	`*** join/#asterisk negativecreep (n=xaeem@210.2.151.110)`
`12:24.37`	`*** join/#asterisk cocomp (n=jeremy@80-193-182-106.cable.ubr03.pres.blueyonder.co.uk)`
`12:25.20`	`negativecreep`	`I am running asterisk on my server with freepbx..basically a trixbox installation..i would like to run SER/openSER on the same box on multiple ports including 5060 and hand off all incoming sip calls to asterisk`
`12:25.23`	`negativecreep`	`is that possible?`
`12:26.25`	`*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)`
`12:30.33`	`cocomp`	`hi - asterisk newbie with a basic question (tried google / docs / book :( ) when run ztcfg -vv get ZT_CHANCONFIG failed on channel 1: No such device or address (6) at end - I've tried it in 2 different machines same issue - what's it mean? PCI issues or config?`
`12:31.28`	`cocomp`	`card is TDM412P on CentOS 4.3 x86_64 using atrpms versions`
`12:32.58`	`*** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net)`
`12:33.59`	`*** join/#asterisk cbrake (n=cbrake@oh-69-34-21-229.sta.embarqhsd.net)`
`12:40.25`	`*** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it)`
`12:41.42`	`MrChimpy`	`cocomp: what version are these atrpms?`
`12:41.52`	`MrChimpy`	`TDM412 is new AFAIK`
`12:42.24`	`cocomp`	`zaptel-1.2.7-16.el4.at zaptel-kmdl-2.6.9-42.0.2.ELsmp-1.2.7-16.el4.at`
`12:42.40`	`MrChimpy`	`and the standard answer would be to download the newest version and build it yourself`
`12:43.14`	`cocomp`	`it's a TDM411P with an extra red module`
`12:43.29`	`cocomp`	`e.g. Green, none, Red, REd`
`12:43.55`	`MrChimpy`	`sorry, I'm confusing it with TE range which I use`
`12:44.01`	`MrChimpy`	`analog stuff, I dunno`
`12:44.14`	`cocomp`	`np - I used the svn stuff yesterday - same prob`
`12:44.34`	`cocomp`	`hence I thought I'd try the rpms`
`12:52.31`	`[TK]D-Fender`	`That'd be a TDM12B, not TDM412....`
`12:52.36`	`Chuji`	`cocomp : Did you load the modules?`
`12:52.49`	`[TK]D-Fender`	`which is just a TDM400B fram w 2 FXO / 1 FXO`
`12:52.53`	`[TK]D-Fender`	`fram*`
`12:53.35`	`cocomp`	`So it's wrong on digium's website then?!?`
`12:54.37`	`cocomp`	`ok TDM400P is the generic card name TDM12B is what I've got then`
`12:54.51`	`cocomp`	`yes loaded zaptel and wctdm`
`12:55.30`	`*** join/#asterisk Bambr (n=Bambr@213-35-236-25-dsl.end.estpak.ee)`
`12:55.52`	`Chuji`	`cocomp : after loading the modules look ad dmesg?`
`12:56.02`	`[TK]D-Fender`	`cocomp: Pastebin your zaptel.conf and "cat /proc/interrupts"`
`12:56.52`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`12:58.24`	`cocomp`	`http://pastebin.com/775617`
`12:58.38`	`*** join/#asterisk roving_prole (n=Harper@72-254-127-253.client.stsn.net)`
`12:59.00`	`cocomp`	`can't see card in proc interrupts`
`12:59.11`	`Chuji`	`did dmesg say anything interesting after loading the modules?`
`12:59.15`	`cocomp`	`nope`
`12:59.32`	`*** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)`
`12:59.49`	`*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)`
`13:00.41`	`tzafrir_home`	`anybody used TestCleint/TestServer (app_test) with zaptel?`
`13:01.09`	`cocomp`	`I think I'm gonna move it's position in the pci slots - see if that turns up anything different :S`
`13:01.24`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`13:01.24`	`*** mode/#asterisk [+o anthm] by ChanServ`
`13:02.12`	`*** join/#asterisk Dr-Linux\|work (n=Linux@202.125.139.198)`
`13:02.26`	`tzafrir_home`	`I'm trying to use it with a an analog loopback (a wire from an FXS port to an FXO port). It appears that the version string is sent fine, but the test ID sending seems to fail`
`13:02.34`	`Dr-Linux\|work`	`anybody is using realtime?`
`13:02.51`	`tzafrir_home`	`Though a quick look at the code there makes me believe that the error message is a bit misleading`
`13:03.33`	`tzafrir_home`	`Dr-Linux\|work, why not just ask your question?`
`13:07.35`	`Dr-Linux\|work`	`Aug 25 18:21:05 ERROR[3214]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server on . Check debug for more info.`
`13:08.27`	`Dr-Linux\|work`	`what debug can i use to check mysql connectivity`
`13:09.06`	`Dr-Linux\|work`	`tzafrir_home, while mysql is connected for:`
`13:09.07`	`Dr-Linux\|work`	`LHR-PBX*CLI> cdr mysql status`
`13:09.08`	`Dr-Linux\|work`	`Connected to cdrasterisk on socket file /var/lib/mysql/mysql.sock using table cdr for 8 minutes, 23 seconds.`
`13:09.27`	`Dr-Linux\|work`	`but it's not for realtime :S`
`13:10.23`	`tzafrir_home`	`Dr-Linux\|work, that's the CDR config. What about the realtime config? extconfig.conf or whatever?`
`13:10.53`	`*** join/#asterisk hypa7ia (i=hypatia@judecca.aculei.net)`
`13:11.53`	`Dr-Linux\|work`	`tzafrir_home, yes it's extconfig.conf`
`13:12.33`	`Dr-Linux\|work`	`voicemail => mysql,rasterisk,voicemail_users`
`13:16.13`	`phearless`	`MP3Player()`
`13:16.25`	`phearless`	`can you play shoutcast stream with it ?`
`13:16.34`	`phearless`	`streams contain ID3 tags`
`13:16.50`	`phearless`	`so I hear only the first second of the stream and then <blank>`
`13:17.00`	`phearless`	`how can I fix that ?`
`13:18.05`	`Nivex`	`get an external program to convert the stream to the proper format`
`13:19.35`	`*** join/#asterisk marv[work] (n=timr@64.89.118.139)`
`13:20.15`	`phearless`	`some apps can remove the tags but for static files`
`13:20.17`	`phearless`	`not streams`
`13:20.21`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`13:21.42`	`*** join/#asterisk rbd (n=rbd@adsl-221-119-125.rmo.bellsouth.net)`
`13:22.40`	`rbd`	`hey guys, I have an app that prints messages at LOG_DEBUG level...how do I get asterisk to display those messages in the console? (starting up with -vvv doesn't seem to do it...is it -d?)`
`13:23.04`	`phearless`	`debut level 666`
`13:23.09`	`phearless`	`debug level 666`
`13:23.16`	`phearless`	`try that in the console`
`13:23.20`	`phearless`	`(I AM NOT SURE)`
`13:30.33`	`*** join/#asterisk hatamen (n=hatamen@222.183.33.234)`
`13:32.39`	`tzafrir_home`	`rbd, edit logger.conf and add "debug" to the "console" line`
`13:32.52`	`tzafrir_home`	`It's documented in the default/sameple logger.conf`
`13:33.30`	`tzafrir_home`	`rbd, alternatively, tail -f /var/log/asterisk/full.log # in another terminal`
`13:33.41`	`inspired`	`rbd, set verbose 8 ?`
`13:33.54`	`inspired`	`oh, follow tzafrir_home's advice instead`
`13:34.24`	`*** join/#asterisk ssokol (n=ssokol@dsl017-122-217.mci1.dsl.speakeasy.net)`
`13:34.58`	`*** part/#asterisk negativecreep (n=xaeem@210.2.151.110)`
`13:40.51`	`*** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.201)`
`13:42.44`	`*** join/#asterisk negativecreep (n=xaeem@210.2.151.110)`
`13:46.15`	`*** join/#asterisk ToyMan (n=stuq@74-32-51-182.dsl1.mdl.ny.frontiernet.net)`
`13:49.10`	`*** join/#asterisk _deg_ (n=deg@200.250.222.8)`
`13:50.21`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`13:50.34`	`*** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.51)`
`13:51.36`	`tdonahue-laptop`	`hi all`
`13:51.56`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`13:52.07`	`tdonahue-laptop`	`has anyone had any isses with Polycom phones not having the MWI turn on with new VM messages?`
`13:52.54`	`tdonahue-laptop`	`or at least any thoughts as to why this would be happening?`
`13:52.57`	`*** part/#asterisk hatamen (n=hatamen@222.183.33.234)`
`13:57.34`	`phearless`	`<phearless> modprobe wctdm opermode=UK`
`13:57.34`	`phearless`	`<phearless> I started a module like this`
`13:57.35`	`phearless`	`<phearless> how can I know, after that it has been started, that the opermode is right ?`
`14:04.07`	`*** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com)`
`14:04.09`	`tdonahue-laptop`	`no one has ever experienced issues with the MWI not coming on before?`
`14:04.20`	`clyrrad`	`I have`
`14:04.22`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`14:04.23`	`*** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)`
`14:05.39`	`tdonahue-laptop`	`oh, oops... helps if the mailbox is defined...`
`14:05.50`	`tdonahue-laptop`	`this is what I get for not checking other people's work.... public embarassment`
`14:05.57`	`tdonahue-laptop`	`:-/`
`14:06.48`	`clyrrad`	`Hi Guys - this is the first time I have had to patch asterisk - I got the security bulitin and downloaded the lastest patch file - my question is how do I properly use it? I am using CentOS`
`14:07.25`	`tdonahue-laptop`	`clyrrad, did you install asterisk from the source code or using a RPM?`
`14:07.31`	`clyrrad`	`from source`
`14:08.39`	`tdonahue-laptop`	i usually just use the 1.2 branch on SVN personally but the basic format is going to be `patch < /path/to/patch.file`
`14:09.10`	`clyrrad`	`i do that from the command line in the sources directory?`
`14:09.43`	`tdonahue-laptop`	`yes most likely in the asterisk directory`
`14:10.11`	`tdonahue-laptop`	`you can check the patch to see if it uses a relative path with the subdirectories`
`14:10.17`	`clyrrad`	`ok then I would need to re-compile astierks?`
`14:11.28`	`tdonahue-laptop`	`yep, i would do a make clean, then compile normally`
`14:11.51`	`clyrrad`	`ok - and you 100% sure thats the proper way to do it?`
`14:12.38`	`tdonahue-laptop`	`make clean` will clear out any old binaries/.o files from the source tree and ensure that you compile everything from scratch with the updated sources
`14:12.47`	`*** join/#asterisk devooo (n=al@124.6.183.1)`
`14:12.51`	`tdonahue-laptop`	`you could always move the old source tree to asterisk.old`
`14:13.21`	`*** join/#asterisk op3r (n=op3r@202.71.179.140)`
`14:13.22`	`devooo`	`can you create a text-based chat application on top of asterisk?`
`14:13.26`	`tdonahue-laptop`	`then just download the asterisk-1.2.11.tar.gz`
`14:13.36`	`clyrrad`	`yea - thats one way to do it`
`14:13.48`	`clyrrad`	`so why dont people just do that instead of use the patch?`
`14:13.56`	`op3r`	`hiya`
`14:13.58`	`tdonahue-laptop`	`you get to avoice all the possible nastiness if you accidently skip a patch`
`14:14.04`	`clyrrad`	`is there an advantage to using the patch instead of downling?`
`14:14.12`	`tdonahue-laptop`	`errr... avoid even`
`14:14.15`	`clyrrad`	`downloading new version*`
`14:14.27`	`clyrrad`	`yea makes sense`
`14:14.34`	`op3r`	`can anyone tell whats this error? Aug 25 06:33:59 WARNING[32233]: channel.c:2719 ast_channel_make_compatible: No path to translate from SIP/acl-08417430(256) to IAX2/37337-3(4)`
`14:14.34`	`op3r`	`<PROTECTED>`
`14:15.07`	`clyrrad`	`prolly Im better to do that instead of using the patch`
`14:15.19`	`clyrrad`	`and since I need to recompile anyway - patch file does not make alot of sense`
`14:15.20`	`tdonahue-laptop`	`it is a tradeoff... you need have a larger download if you use the whole source tree but in the end you usually get to the same place`
`14:15.41`	`tdonahue-laptop`	`make clean isn't a big deal`
`14:15.54`	`clyrrad`	`yea - its the make install that takes the time`
`14:16.01`	`clyrrad`	`after make clean have to make install`
`14:16.23`	`tdonahue-laptop`	`i would hightly recommend that whichever way you do it you make a backup copy of your working source tree`
`14:16.41`	`tzafrir_home`	`phearless, you want to know if the string "UK" is right? or if it got through?`
`14:16.44`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`14:16.51`	`tdonahue-laptop`	`that way if you need to roll back in a hurry, you don't need to run around looking for the source that your originaly had working`
`14:17.06`	`tzafrir_home`	`to see if it got through: cat /sys/module/wctdm/parameters/opermode`
`14:17.06`	`clyrrad`	`todanhug- yea that sounds like a good plan`
`14:17.28`	`*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`14:17.51`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`14:17.54`	`tzafrir_home`	`phearless, to see if it is right, look for "opermode" in the source of wctdm. Theer's a big "table" with all the possible values`
`14:18.02`	`tdonahue-laptop`	`well the saying does go "Once bitten, twice shy"... or otherwise "Been there, done that. didn't get the T-shirt"`
`14:18.28`	`clyrrad`	`lol`
`14:18.41`	`clyrrad`	`I have a backup of the sources as a tarball`
`14:19.04`	`tdonahue-laptop`	`hmm... can't type today... should have been "didn't even get the T-shirt"`
`14:19.14`	`tdonahue-laptop`	`close enough`
`14:19.22`	`blitzrage`	`hail coppice`
`14:19.23`	`tzafrir_home`	`phearless, it has two relevant options: "UK" and "TBR21"`
`14:19.33`	`coppice`	`hi`
`14:20.02`	`tzafrir_home`	`"TBR21" is listed as relevant to many european countries. Perhaps coppice will know if it is relevant`
`14:20.42`	`tzafrir_home`	`(an opermode for the wfxo ports of wctdm)`
`14:20.46`	`coppice`	`shouldn't it be CTR21?`
`14:21.28`	`phearless`	`sorry for the late answer, tzafrir_home`
`14:21.29`	`phearless`	`1s`
`14:21.30`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`14:21.56`	`phearless`	`<tzafrir_home> phearless, you want to know if the string "UK" is right? or if it got through?`
`14:21.57`	`phearless`	`no`
`14:22.18`	`phearless`	`I wanted to set the module options in a config file`
`14:22.31`	`phearless`	`but it seems to be in modprobe.conf`
`14:24.08`	`*** join/#asterisk iso (n=allen@host-24-225-209-177.patmedia.net)`
`14:24.08`	`tzafrir_home`	`phearless, those are module options, so modprobe.conf is the right place for them`
`14:24.44`	`tzafrir_home`	`coppice, both "BTR21" and "CTR21" are chineese to me. IIRC that table of values is taken from the specs of the chip`
`14:25.17`	`coppice`	`CTR21 is the ETSI spec for pan-european analogue lines`
`14:25.50`	`phearless`	`ok thanks tzafrir_home`
`14:26.18`	`tzafrir_home`	`phearless, in case of doubt, try both and choose that one that gets you less echo...`
`14:26.34`	`tzafrir_home`	`I don't have that daa specs with me right now`
`14:26.34`	`phearless`	`ok`
`14:26.59`	`op3r`	`<op3r> can anyone tell me about this error? Aug 25 06:33:59 WARNING[32233]: channel.c:2719 ast_channel_make_compatible: No path to translate from SIP/acl-08417430(256) to IAX2/37337-3(4)`
`14:28.06`	`*** join/#asterisk devooo (n=al@124.6.183.1)`
`14:28.24`	`devooo`	`Hello. Can you create a chat application on top of Asterisk?`
`14:29.10`	`phearless`	`like irc ?`
`14:30.01`	`devooo`	`yes`
`14:31.47`	`devooo`	`given that there is an asterisk server, could this act as a server so that computers who are registered on the asterisk server could chat with one another and each computer oculd be identified by their extension numbers?`
`14:32.11`	`bkw_`	`SIP Simple`
`14:32.18`	`bkw_`	`now if only Asterisk supported that`
`14:32.26`	`op3r`	`anyone?`
`14:33.37`	`blitzrage`	`you could run an irc server on the same box as asterisk, yes ;)`
`14:34.56`	`*** part/#asterisk negativecreep (n=xaeem@210.2.151.110)`
`14:35.08`	`devooo`	`blitzrage: but can asterisk act as the server? say in a LAN, could each user chat with each other using asterisk?`
`14:35.15`	`mut`	`maaaan`
`14:35.29`	`mut`	`we're switching our carrier`
`14:35.35`	`caio1982`	`with the jabber support in asterisk that could be possible to do not that hard, right?`
`14:35.35`	`mut`	`going to be a task`
`14:36.02`	`mut`	`gotta move a ton of crap to a co in southfield`
`14:37.09`	`*** join/#asterisk sivana (n=sivana@mixdown.ca)`
`14:38.04`	`sivana`	`which module is GotoIfTime in?`
`14:38.06`	`devooo`	`caio1982: where could i find information on that?`
`14:38.56`	`*** join/#asterisk kore (i=kore@mindwipe.org)`
`14:39.44`	`tzafrir_home`	`I repeast: anybody tried app_test (TestClinet/TestServer) over a zap channel?`
`14:42.19`	`phearless`	`how can I STOP the SIP debug mode ?`
`14:42.26`	`*** part/#asterisk phearless (n=phearles@host81-138-68-106.in-addr.btopenworld.com)`
`14:42.34`	`*** join/#asterisk phearless (n=phearles@host81-138-68-106.in-addr.btopenworld.com)`
`14:42.36`	`phearless`	`oops`
`14:43.52`	`*** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35)`
`14:44.53`	`caio1982`	`devooo: here http://svn.digium.com/svn/asterisk/trunk/doc/jingle.txt, here http://svn.digium.com/svn/asterisk/trunk/doc/jabber.txt, here http://svn.digium.com/svn/asterisk/trunk/configs/jingle.conf.sample, here http://svn.digium.com/svn/asterisk/trunk/configs/jabber.conf.sample and herehttp://www.voip-info.org/wiki-Asterisk+Jabber (but i was just thinking out loud, I neved touched in these things)`
`14:45.48`	`devooo`	`caio1982: thanks a lot! :)`
`14:46.36`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`14:48.15`	`*** join/#asterisk dacleric (n=dacleric@p5482274E.dip0.t-ipconnect.de)`
`14:52.01`	`*** part/#asterisk hmmhesays (n=ohyeah@66.173.103.110)`
`14:53.52`	`*** join/#asterisk bmg505 (n=leon@dsl-146-19-97.telkomadsl.co.za)`
`14:54.27`	`phearless`	`how can I STOP the SIP DEBUG MODE ?`
`14:55.08`	`Chuji`	`ugh, you need qt for mysql connectors?`
`14:55.20`	`caio1982`	`phearless: guess what? sip no <TAB>`
`14:56.39`	`phearless`	`thank you caio1982`
`14:56.41`	`phearless`	`!`
`14:57.54`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`14:58.50`	`*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)`
`15:02.33`	`*** join/#asterisk kpettit (n=keith@69.15.174.114)`
`15:05.05`	`*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)`
`15:05.05`	`*** mode/#asterisk [+o Cresl1n] by ChanServ`
`15:08.04`	`*** join/#asterisk doolph (i=doolph@200.46.148.43)`
`15:08.07`	`doolph`	`hi`
`15:08.15`	`doolph`	`how do I record something in g729 format?`
`15:09.01`	`Corydon-w`	`Record(foo.g729)`
`15:09.29`	`doolph`	`thnx`
`15:11.28`	`*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)`
`15:17.21`	`*** join/#asterisk anthonyl (n=anthony@c-71-207-228-190.hsd1.al.comcast.net)`
`15:17.37`	`*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)`
`15:26.27`	`*** join/#asterisk nDuff (n=ccd@64.128.31.220)`
`15:29.19`	`*** join/#asterisk ik_5 (n=ik@85.64.203.170.dynamic.barak-online.net)`
`15:30.12`	`nDuff`	`What's comparable to Background(), but simply providing a dialtone rather than playing an audio file?`
`15:30.29`	`Corydon-w`	`DISA`
`15:30.39`	`Corydon-w`	`or Playtones(dialtone)`
`15:32.03`	`*** join/#asterisk \|dennis\| (n=dennis@etcdsl1.btl.net)`
`15:32.22`	`*** join/#asterisk bobnormal (n=bob@218.63.238.3)`
`15:33.18`	`*** join/#asterisk ellisgl2 (i=keefejoh@seraph.techwareit.com)`
`15:33.43`	`ellisgl2`	`Any one try setting up Frame Relay on a t1 on a * box`
`15:34.47`	`bobnormal`	`hey, can someone help me test my new german sipgate.co.uk number? my phones here in china are international-locked and voipbuster (my preferred dialout system) has that range blocked. # is +49 1801 993322 3568. i think you might need to be in germany and dial it locally for it to work. sipgate says its a local call from anywhere in germany. you should hear a chinese pbx (getting german pbx translation if this works :)`
`15:35.22`	`*** join/#asterisk Assid (i=assid@203.115.83.215)`
`15:37.39`	`*** join/#asterisk alexmontoanelli (n=alexmont@200.193.10.156)`
`15:41.40`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.127)`
`15:44.46`	`*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)`
`15:46.01`	`ellisgl2`	`Or even PPP over t1`
`15:47.18`	`nDuff`	`Corydon-w: Thank you; DISA is exactly what I was looking for.`
`15:48.40`	`*** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)`
`15:48.56`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`15:49.16`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`15:49.51`	`*** join/#asterisk c4t3l (n=c4t3l@69.15.174.114)`
`15:49.59`	`rollergrrl`	`Anyone have opinions on G723 vs. G729?`
`15:50.47`	`ellisgl2`	`G729 is 6 greater thatn 723`
`15:50.49`	`ellisgl2`	`=P`
`15:50.54`	`c4t3l`	`:D`
`15:51.00`	`*** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw)`
`15:51.45`	`rollergrrl`	`nice opinion`
`15:52.54`	`twisted[work]`	`omg`
`15:52.55`	`twisted[work]`	`it's rollergrrl`
`15:53.42`	`twisted[work]`	`g723.1 is nicer and IMO less cpu intensive, but good luck getting license and/or modules for it`
`15:53.54`	`tzafrir_home`	`rollergrrl, both are pattently absurd to me`
`15:54.03`	`rollergrrl`	`I know! She lives!`
`15:54.08`	`[TK]D-Fender`	`G723 is a massively patent encumbered and old code that is best forgotten about and only used by old and/or stupid equipment.`
`15:54.22`	`rollergrrl`	`eek`
`15:54.24`	`c4t3l`	`i agree`
`15:54.36`	`c4t3l`	`fax technology also sucks!`
`15:54.43`	`tzafrir_home`	`[TK]D-Fender, g723 or g723.1?`
`15:54.44`	`twisted[work]`	`[TK]D-Fender, yeah, but you gotta admit, it's a nice codec outside of that`
`15:54.57`	`rollergrrl`	`I'm looking for something easier on the CPU than G729`
`15:54.58`	`Qwell`	`g723 is more compressed, right?`
`15:55.03`	`twisted[work]`	`g723 is old and dead... g723.1 is the new black`
`15:55.04`	`[TK]D-Fender`	`tzafrir_home: YES`
`15:55.05`	`tzafrir_home`	`rollergrrl, gsm`
`15:55.08`	`Qwell`	`rollergrrl: g729 isn't that bad`
`15:55.10`	`twisted[work]`	`Qwell, yes`
`15:55.13`	`rollergrrl`	`with similar compression ratio`
`15:55.25`	`twisted[work]`	`rollergrrl, ilbc`
`15:55.29`	`c4t3l`	`he :)`
`15:55.36`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`15:55.44`	`tzafrir_home`	`ilbc is no lighter on the CPU.`
`15:55.49`	`TripleFFFF`	`ANY one experience echo on SIP -> SiP`
`15:55.52`	`twisted[work]`	`tzafrir, sure it is`
`15:56.02`	`twisted[work]`	`unless you're running archaic hardware`
`15:56.02`	`TripleFFFF`	`i hear echo on .11`
`15:56.06`	`rollergrrl`	`since cpu is limited on this Soekris box`
`15:56.17`	`c4t3l`	`eew no`
`15:56.28`	`*** join/#asterisk DarKnesS_WolF (n=wolf@62.135.2.227)`
`15:56.37`	`twisted[work]`	`then what about lpc10`
`15:56.42`	`twisted[work]`	`you may sound like a robot`
`15:56.47`	`twisted[work]`	`but at least it's streamlined :P`
`15:56.52`	`rollergrrl`	`hehe`
`15:56.56`	`*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`15:56.59`	`tzafrir_home`	`And it beats g723.1`
`15:57.05`	`Qwell`	`Why not just use gsm?`
`15:57.16`	`rollergrrl`	`so iLBC over GSM?`
`15:57.28`	`twisted[work]`	`personally, i would go ilbc over gsm`
`15:57.29`	`*** part/#asterisk ik_5 (n=ik@85.64.203.170.dynamic.barak-online.net)`
`15:57.31`	`twisted[work]`	`but YMMV`
`15:58.03`	`Corydon-w`	`ilbc over gsm if you have the CPU to spare`
`15:58.14`	`twisted[work]`	`Corydon-w, it didn't ever seem that intensive to me`
`15:58.18`	`Corydon-w`	`It's a rather costly codec, comparatively`
`15:58.19`	`twisted[work]`	`on a 1ghz box`
`15:58.54`	`twisted[work]`	`guess the real question is`
`15:59.00`	`twisted[work]`	`what is the application and estimated volume?`
`15:59.01`	`rollergrrl`	`It's a 266MHz 586 Geode`
`15:59.22`	`twisted[work]`	`ouch.`
`15:59.37`	`blitzrage`	`yah, she said soekris`
`15:59.37`	`rollergrrl`	`"ideally" i'd like 24 channels with the smallest bandwidth consumption possible`
`15:59.53`	`Corydon-w`	`ulaw->gsm=4 ulaw->ilbc=16`
`15:59.54`	`caio1982`	`24 channels in a soekris board?`
`15:59.56`	`twisted[work]`	`ooooooh...`
`16:00.01`	`rollergrrl`	`but I highly doubt that's possible`
`16:00.21`	`twisted[work]`	`just do slin`
`16:00.28`	`file`	`twisted[work]: :D`
`16:00.34`	`rollergrrl`	`hehe`
`16:00.36`	`twisted[work]`	`you'll get 24 channels easy, but the bandwith will suffer`
`16:00.46`	`rollergrrl`	`I've got about 600kps to work with though`
`16:00.50`	`rollergrrl`	`kpbs`
`16:00.56`	`blitzrage`	`ouch`
`16:01.01`	`Corydon-w`	`GSM or ADPCM are probably going to be your choices`
`16:01.10`	`blitzrage`	`I'd agree`
`16:01.20`	`blitzrage`	`not sure if you'll get 24 GSM though`
`16:01.45`	`rollergrrl`	`GSM will be a good starting point for testing then`
`16:01.57`	`Corydon-w`	`You might could give the first 12 channels GSM and the last 12 channels ADPCM`
`16:02.11`	`Corydon-w`	`so if you stayed below 12 channels, it'd all be GSM`
`16:02.18`	`rollergrrl`	`interesting`
`16:02.33`	`rollergrrl`	`that just might do it`
`16:02.43`	`Corydon-w`	`but you could fall back to a lesser codec if you exceeded that`
`16:03.14`	`rollergrrl`	`very neat idea`
`16:03.54`	`twisted[work]`	`Corydon-w, that's actually a good idea in that scenario... why can't I think of stuff like that when I have a hangover?`
`16:03.55`	`Corydon-w`	`They will give roughly the same quality`
`16:03.56`	`*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)`
`16:04.40`	`Corydon-w`	`twisted[work]: heh`
`16:05.07`	`Corydon-w`	`twisted[work]: I haven't had a hangover in nearly 10 years`
`16:05.10`	`rollergrrl`	`twisted[work]: you don't have enough hangovers`
`16:05.25`	`rollergrrl`	`twisted[work]: if you have one every morning... maybe you'll get used to it`
`16:05.31`	`*** join/#asterisk anthonyl (n=anthony@gateway.digium.com)`
`16:05.39`	`Corydon-w`	`One was enough for me not to do that again`
`16:05.57`	`twisted[work]`	`rollergrrl, haha, i'll also wind up gaining enough weight to make me not want to move`
`16:06.12`	`twisted[work]`	`Corydon-w, i never had them until a couple years back`
`16:06.25`	`twisted[work]`	`then all of a sudden WHAM! you get hangovers now!`
`16:06.29`	`TripleFFFF`	`hhe`
`16:06.41`	`rollergrrl`	`just buy an extra large seqway`
`16:06.43`	`Corydon-w`	`Might also be the cheap vodka`
`16:06.49`	`rollergrrl`	`segway`
`16:06.52`	`TripleFFFF`	`question.. if u use CLI> DIAL .. how does it assign to a sip device ?`
`16:06.58`	`twisted[work]`	`segway? hahaha`
`16:07.13`	`twisted[work]`	`i wouldn't be caught on a segway`
`16:07.47`	`rollergrrl`	`that's only if you gain too much weight`
`16:07.52`	`*** join/#asterisk DarKnesS_WolF (n=wolf@62.135.2.227)`
`16:07.59`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:07.59`	`*** mode/#asterisk [+o mog] by ChanServ`
`16:07.59`	`twisted[work]`	`well, I don't plan on that happening`
`16:08.00`	`rollergrrl`	`you may have to get it strapped on to your body`
`16:08.02`	`twisted[work]`	`i'm down to like 213 now`
`16:08.51`	`Corydon-w`	`I wonder if they called it that only because the Average Joe couldn't pronounce "segue"`
`16:08.52`	`rollergrrl`	`If you drink enough after every meal... you can go on a purge diet`
`16:09.05`	`twisted[work]`	`rollergrrl, that's just nasty`
`16:09.16`	`TripleFFFF`	`SEG YOU WAY`
`16:09.46`	`*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)`
`16:09.55`	`rollergrrl`	`just keep mouthwash in your purse`
`16:10.10`	`Corydon-w`	`twisted doesn't carry a purse, last I checked`
`16:10.30`	`twisted[work]`	`nope`
`16:10.33`	`Corydon-w`	`but who knows, HSV may have changed him`
`16:10.59`	`twisted[work]`	`HSV has changed me in several ways, none of which include carrying a purse`
`16:12.49`	`blitzrage`	`lol`
`16:14.06`	`blitzrage`	`damnit... I gotta go down to the locker room to find a console cable for this soekris...`
`16:17.10`	`eKo1`	`I'm getting "Unable to create channel of type 'zap' (cause 34 - Circuit/channel congestion)" but I have no channels in use. Could it be my PRI?`
`16:19.32`	`*** join/#asterisk _deg_ (n=deg@200.250.222.8)`
`16:25.16`	`intralanman`	`eKo1: it's possible, although i personally would expect that to come back as another cause code.... (44 maybe)`
`16:26.34`	`tzafrir_home`	`eKo1, what's the relevant Dial command?`
`16:26.36`	`TripleFFFF`	`YO i need a hanf`
`16:26.57`	`TripleFFFF`	`i get frigin echo now .. was 1.2.9.1 to 1.2.11 .. sip to sip ulaw via asterisk`
`16:27.09`	`TripleFFFF`	`as in exten dial sip /user`
`16:27.10`	`intralanman`	`~ecfo`
`16:27.13`	`jbot`	`Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as "screeching", "feedback", "static", or other useless terms. If users report "static" on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. what happens when the echo canceller suddenly ...`
`16:27.35`	`TripleFFFF`	`im sip to sip`
`16:27.46`	`bkw_`	`you shouldn't have to monkey with the RX or TX gains`
`16:27.52`	`TripleFFFF`	`yeah`
`16:27.55`	`TripleFFFF`	`hey brian you know about this ?`
`16:27.58`	`TripleFFFF`	`man this is weird`
`16:28.02`	`intralanman`	`yeah, i was looking for something else, my bad`
`16:28.07`	`TripleFFFF`	`BUT.. sip/#@provieder is perfect`
`16:28.18`	`TripleFFFF`	`and echotest is perfect..`
`16:28.28`	`TripleFFFF`	`so sip to spstn , pstn to sip is ok`
`16:28.34`	`TripleFFFF`	`but sip to sip SUCKS balls`
`16:28.47`	`TripleFFFF`	`i hate to revert pre 11 dues to iax DDOS`
`16:29.09`	`*** join/#asterisk lilalinux (i=e-trolle@deepthroat.deswahnsinns.de)`
`16:29.12`	`TripleFFFF`	`brian.. can i give you a test accoutn on our box and i dial you ?`
`16:29.14`	`lilalinux`	`hey guys`
`16:29.50`	`*** join/#asterisk gJon (n=jellis@dev.integrumtech.com)`
`16:30.14`	`lilalinux`	`is it possible to have festival support with unix sockets instead of port 1314?`
`16:30.27`	`TripleFFFF`	`ghosting in the background`
`16:31.40`	`*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)`
`16:31.47`	`lilalinux`	`TripleFFFF: pardon?`
`16:33.08`	`gJon`	`what codec do you guys recommend for connecting 2 asterisk machines using IAX2?`
`16:33.40`	`[TK]D-Fender`	`gJon: Depends on your bandwidth. If you can, use G.711 at all times`
`16:33.44`	`mog`	`ulaw, unless you have bandwith constraints`
`16:34.30`	`gJon`	`yeah, i do have limited bandwidth and i am having problems with g.711... what is the next best?`
`16:34.51`	`*** join/#asterisk mercestes (n=merceste@216.54.143.2)`
`16:35.24`	`mog`	`in order from largest to smallest, g726, gsm, ilbc, g729, g723`
`16:35.45`	`TripleFFFF`	`hey`
`16:36.02`	`gJon`	`thanks... i will play around with it some more`
`16:36.07`	`TripleFFFF`	`still getting ghosting on .11`
`16:36.09`	`TripleFFFF`	`sip to sip`
`16:36.12`	`TripleFFFF`	`this is REALLY weird`
`16:36.14`	`Cresl1n`	`TripleFFFF: if you're calling SIP to SIP and you get echo, your phones are brox0red`
`16:36.24`	`TripleFFFF`	`cisco 7960 ;ol`
`16:36.27`	`TripleFFFF`	`doubt it`
`16:36.32`	`TripleFFFF`	`i call pstn fine`
`16:36.48`	`[TK]D-Fender`	`TripleFFFF: I heard another guy yesterday making the same comments..... try a different firmware release on them`
`16:36.52`	`TripleFFFF`	`and receive.. so its prolly other exten im dialing.. ill check in 4 with party #2`
`16:36.58`	`Cresl1n`	`well, no echo cancelation is done on SIP to SIP calls, so it's the phones' fault`
`16:37.12`	`Cresl1n`	`echo cancelation is supposed to be on the SIP phones`
`16:37.23`	`Cresl1n`	`local acoustic echo cancelation`
`16:37.39`	`[TK]D-Fender`	`Cresl1n: Only AEC, but thats minimal`
`16:37.55`	`TripleFFFF`	`hmmm so .. if phone does EC.. and asterisk does.. it would be bad`
`16:38.02`	`TripleFFFF`	`so all phone should turn off no ?`
`16:38.02`	`Cresl1n`	`you only do echo cancelation on zap interfaces`
`16:38.06`	`TripleFFFF`	`yeah`
`16:38.08`	`TripleFFFF`	`stilll`
`16:38.14`	`Cresl1n`	`TripleFFFF: no, you don't get it`
`16:38.18`	`Cresl1n`	`it's local echo cancelation`
`16:38.28`	`TripleFFFF`	`oh`
`16:38.29`	`Cresl1n`	`so the phone is supposed to remove any echo that it generates`
`16:38.31`	`TripleFFFF`	`on his phone ?`
`16:38.38`	`TripleFFFF`	`hmm`
`16:38.54`	`*** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw)`
`16:38.54`	`Cresl1n`	`if the remote end generates echo, then the remove phone is broken and should be doing local echo cancelation`
`16:38.56`	`TripleFFFF`	`so i need to target if its me or him.. then figure it out..`
`16:39.22`	`Cresl1n`	`if you hear it, it's him`
`16:39.26`	`Cresl1n`	`if he hears it, it's you`
`16:40.03`	`TripleFFFF`	`oh then that easy .. its him .spa 942`
`16:40.15`	`TripleFFFF`	`so we need anyone with spa942 to test .11`
`16:41.45`	`TripleFFFF`	`Linksys/SPA942-4.1.15 actually`
`16:42.03`	`Cresl1n`	`I would think so`
`16:42.13`	`Cresl1n`	`the cisco phones are usually really good phones`
`16:42.41`	`TripleFFFF`	`the problem is .. its worked 100% till last night.. when i made .11 from 1.2.9.1`
`16:42.45`	`Qwell`	`as long as they run skinny firmware`
`16:42.48`	`TripleFFFF`	`so thats the problem..`
`16:45.07`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`16:45.37`	`*** join/#asterisk Strom_C (n=strom@gateway.digium.com)`
`16:46.38`	`[TK]D-Fender`	`Qwell: I heard some freakish sounding rumour that Cisco may offload all their VoIP line to their Linksys division. Anything you can corroborate?`
`16:46.58`	`Qwell`	`[TK]D-Fender: no.. not like cisco knows or even likes me :p`
`16:47.34`	`[TK]D-Fender`	`Qwell: Just figure you are more "in the know" than most as far as they go (as I am with Polycom).`
`16:47.41`	`Qwell`	`nah`
`16:47.56`	`Qwell`	`I've only given them a couple dozen sales. I mean nothing to them.`
`16:48.01`	`blitzrage`	`Qwell: !!!`
`16:48.46`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`16:49.43`	`Strom_C`	`[TK]D-Fender: offloading everything voip to the linksys division doesn't make a lot of sense to me...they're trying to position their linksys line as the consumer / prosumer stuff and the cisco stuff as the business equipment`
`16:49.47`	`*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)`
`16:50.03`	`Qwell`	`I'd agree with that`
`16:50.21`	`Qwell`	`oh, ha, have you guys seen the movie "The Island"`
`16:50.30`	`blitzrage`	`yes`
`16:50.38`	`Qwell`	`there is so much product placement in that movie, it isn't even funny...EXCEPT!`
`16:50.39`	`blitzrage`	`has a 1984 flavour to it`
`16:50.49`	`Qwell`	`there is a cisco ad in there, and that is hilarious`
`16:50.54`	`[TK]D-Fender`	`blitzrage: You two timing bastard ! ! !`
`16:50.57`	`Strom_C`	`what is cisco advertising in the movie?`
`16:51.05`	`Qwell`	`Strom_C: Have you seen the movie?`
`16:51.11`	`blitzrage`	`[TK]D-Fender: ?`
`16:51.11`	`Strom_C`	`no`
`16:51.19`	`Qwell`	`then you wouldn't get it if I told you what part :P`
`16:51.20`	`Strom_C`	`i remember seeing the billboard, though`
`16:51.30`	`[TK]D-Fender`	`blitzrage: Our use of "! ! !" (c)`
`16:51.36`	`Qwell`	`there is a part where they do a video chat...and when it starts up, it's a full on cisco ad`
`16:51.48`	`[TK]D-Fender`	`:D`
`16:51.49`	`blitzrage`	`[TK]D-Fender: I didn't use ! ! !, I used !!! which is different`
`16:51.57`	`*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)`
`16:52.05`	`blitzrage`	`!!! is generally reserved for greetings :)`
`16:52.20`	`Strom_C`	`hah`
`16:52.29`	`[TK]D-Fender`	`blitzrage: well in that case ... ! ! !`
`16:52.35`	`blitzrage`	`hehehe`
`16:53.12`	`blitzrage`	`oh, and going to the locker room to attempt to find a console cable...`
`16:53.16`	`TripleFFFF`	`its seem to disapear when io sepak softley... butbut if i talk loud while he talks its horrible`
`16:53.42`	`Strom_C`	`huh??`
`16:54.41`	`*** join/#asterisk adorah (n=Administ@87.68.146.95.cable.012.net.il)`
`16:55.20`	`TripleFFFF`	`sip -> sip echo same ulaw codec.. on .11`
`16:55.29`	`TripleFFFF`	`BTW whats linksys one line of product ? not sip i think`
`16:57.12`	`[TK]D-Fender`	`TripleFFFF: SIP + non-RFC extra's I believe.`
`16:57.28`	`TripleFFFF`	`not it seem proprytary`
`16:58.54`	`TripleFFFF`	`ok so my sip does it my cisco 7960 does the echo crap`
`16:59.01`	`TripleFFFF`	`ill reboot iand see`
`16:59.30`	`TripleFFFF`	`i thnk i saw something about that anyho`
`17:00.18`	`eKo1`	`How can I find out what port chan_iax2 is listening on?`
`17:00.26`	`Qwell`	`eKo1: netstat -pl`
`17:01.11`	`Cresl1n`	`eKo1: 4569`
`17:01.55`	`eKo1`	`I meant, within the Asterisk CLI.`
`17:01.58`	`*** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`17:02.01`	`rene1`	`hello`
`17:02.11`	`eKo1`	`It is not possible is it.`
`17:02.27`	`bobnormal`	`is it normal to get ACK then BYE on SIP call setup?`
`17:02.27`	`bobnormal`	`my calls, coming in via gizmo project, generate traffic but dont start and instead fall back to voicemail (which i turned off). what gives?`
`17:02.27`	`bobnormal`	`voicemail@gizmo, that is.`
`17:02.54`	`eKo1`	`Cresl1n: that is the default`
`17:03.17`	`*** join/#asterisk test34 (n=test34@unaffiliated/test34)`
`17:04.28`	`*** join/#asterisk Mandrak3 (n=io@81.27.211.30)`
`17:05.04`	`*** join/#asterisk rbordeaux (i=hidden-u@80.169.196.234)`
`17:05.11`	`rene1`	`what could cause a situation where a E1 line gets an all or some channels congestion, after several hundred calls? i am using PRI ISDN zaptel 1.2.7 and asterisk 1.2.10. restarting asterisk and issuiing a ztcfg often resolves the issue but sometimes a full driver reload is required and some times a full reboot was required.. Debian 3.1 r2 platform 2.6.8-3-686 kernel. 2 port E1 digium card being used. 1 E1 port active`
`17:06.34`	`rene1`	`Clock via telco. i.e. 1/1/0. not using CRC4.`
`17:07.06`	`blitzrage`	`[TK]D-Fender: is it possible to make a # from the configuration on the phone (not in the GUI)`
`17:07.14`	`blitzrage`	`let me quality -- IP501 :)`
`17:07.17`	`blitzrage`	`qualify*`
`17:07.19`	`blitzrage`	`damnit! :)`
`17:07.30`	`eKo1`	`rene1: I had that condition just a while ago. I had no calls going through at all and everytime I made a call trhough Zap, I would get congestion.`
`17:07.44`	`eKo1`	`I just did a reboot and it fixed the problem.`
`17:07.56`	`eKo1`	`I suspect the driver is to blame.`
`17:08.13`	`Cresl1n`	`rene1: if you do a "show channels" what comes up?`
`17:08.40`	`rene1`	`eko1: i rebooted a couple days ago and that fixed it until it happened 2-5 hours later`
`17:08.51`	`*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)`
`17:09.23`	`rene1`	`Cresl1n: just a sip user/agent connected to app: agent/login listening to MOH`
`17:09.35`	`Cresl1n`	`rene1: no, when it gets in that state`
`17:09.40`	`rene1`	`MOH is in native format i.e. gsm`
`17:09.40`	`rene1`	`ok`
`17:09.45`	`TripleFFFF`	`cisco 7960 so tis a falw by design....if you volume... up ..it up sensitivty of mic + volume of ear`
`17:10.01`	`rene1`	`Cresl1n: right now is in that state: Aug 25 12:10:07 NOTICE[10877]: channel.c:2445 __ast_request_and_dial: Unable to request channel Zap/g1/2504718`
`17:10.20`	`rene1`	`so local calls do work but calls that go tru the zap interfase dont`
`17:10.30`	`Cresl1n`	`that's weird`
`17:10.42`	`rene1`	`zttool says it is OKk, and sometimes shows REC or RED`
`17:10.47`	`rene1`	`right now it is OK`
`17:11.06`	`rene1`	`but it is not ;)`
`17:11.49`	`bobnormal`	`does anyone have incoming working with a gizmo project / area775 / callwave number? i get sip traffic but the call falls back to the voicemail box, despite me turning voicemail off. i guess that means sip is failing, but us+them codecs agree and i cant see anything obviously wrong.`
`17:12.39`	`rene1`	`Cresl1n: if CRC4 was required by telco and i dont have it configured, would i get calls at all?`
`17:12.43`	`bobnormal`	`could it be a latency issue? i am <300ms away.`
`17:12.44`	`*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`17:12.50`	`Nivex`	`bobnormal: works fine for me`
`17:14.14`	`rene1`	`OTOH i have heard that having a system not properly connected to ground can create synchronicity issues with the line, comment on this 1 please?`
`17:14.18`	`Mandrak3`	`Hi everybody`
`17:14.22`	`rene1`	`Hi`
`17:14.30`	`bobnormal`	`nivex: mind msging me your config minus the secret bits?`
`17:14.47`	`Mandrak3`	`someone can help me with asterisk addons`
`17:14.56`	`rene1`	`what is giving you trouble?`
`17:15.09`	`clyrrad`	`im getting an error on the CLI saying an agi does not exsit at /var/lib/asterisk/agi-bin/check-queue.agi yet if I VI /var/lib/asterisk/agi-bin/check-queue.agi the script is there.... what would cause this?`
`17:15.10`	`Mandrak3`	`I want to enable cdr_mysql module`
`17:15.11`	`*** join/#asterisk EmleyMoor (n=phil@firthpark.tinsleyviaduct.com)`
`17:15.19`	`*** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net)`
`17:15.30`	`rene1`	`what OS`
`17:15.38`	`clyrrad`	`CentOS`
`17:15.41`	`rene1`	`ok`
`17:15.44`	`clyrrad`	`and permissions is 777`
`17:15.59`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`17:16.05`	`clyrrad`	`but the CLI sais the script is not there (no such file or direcotry)`
`17:16.47`	`rene1`	`Mandrake3 you need to install client and development libraries for mysql`
`17:17.02`	`Nivex`	`bobnormal: my sip.conf http://pastebin.ca/149251`
`17:17.31`	`Mandrak3`	`I have them`
`17:17.41`	`*** join/#asterisk crCernier (n=rochatc@01-mars-ext.he-arc.ch)`
`17:17.51`	`bobnormal`	`nivex: thanks a lot.`
`17:18.00`	`rene1`	`ok you might need to tweak the makefile`
`17:18.15`	`Mandrak3`	`http://www.voip-info.org/wiki-Asterisk+cdr+mysql`
`17:18.27`	`Mandrak3`	`I read from here`
`17:18.51`	`clyrrad`	`can anyone help me?`
`17:19.03`	`rene1`	`Mandrak3: make sure you ve got mysql`
`17:19.03`	`rene1`	`mysq-server`
`17:19.03`	`rene1`	`mysql-devel`
`17:19.21`	`docelmo`	`clyrrad well is it?`
`17:19.27`	`clyrrad`	`yes its there`
`17:19.28`	`docelmo`	`is it really there or did it run away?`
`17:19.34`	`clyrrad`	`if i VI that path it opens and i can see the code`
`17:19.37`	`docelmo`	`What did you name it?`
`17:19.39`	`rene1`	`Mandrake3: and also kernel-source bison openssl-devel`
`17:19.42`	`docelmo`	`What kinda script is it..`
`17:19.48`	`clyrrad`	`check-queue.agi`
`17:19.52`	`clyrrad`	`its a PHP AGI`
`17:19.55`	`docelmo`	`You see when you ask a VERY LAME question you will get alot of lame answers..`
`17:19.59`	`docelmo`	`I wrote that.. :)`
`17:20.11`	`Mandrak3`	`I have Mysql 5 installed`
`17:20.26`	`docelmo`	`Anywho.. do you have check-queue.agi as your filename in the script?`
`17:20.30`	`docelmo`	`err dialplan?`
`17:20.31`	`rene1`	`you need the development libraries so asterisk can bind to mysql`
`17:20.35`	`bkw__`	`what no more markocam?`
`17:20.36`	`clyrrad`	`yes`
`17:20.47`	`docelmo`	`is it executable?`
`17:20.55`	`*** join/#asterisk NeonLevel (i=HydraIRC@200.52.142.184)`
`17:20.55`	`*** join/#asterisk EmleyMoor (n=phil@firthpark.tinsleyviaduct.com)`
`17:20.55`	`bkw__`	`"Where on earth is Sparky Menser!" thats the new game`
`17:21.09`	`Mandrak3`	`mysql-devel.i386 4.1.20-1.RHEL4.1 installed`
`17:21.29`	`docelmo`	`is it setup in ld.conf???`
`17:21.32`	`EmleyMoor`	`I have just installed asterisk and want to connect linphone to it - how do I match up username and password?`
`17:21.32`	`clyrrad`	`yes its 777 and +x`
`17:21.46`	`docelmo`	`What are the first 3 lines of your script?`
`17:21.50`	`clyrrad`	`-rwxrwxrwx 1 root root 13326 Aug 25 11:54 check-queue.agi`
`17:21.51`	`Mandrak3`	`rene1: mysql-devel.i386 4.1.20-1.RHEL4.1 installed`
`17:22.03`	`clyrrad`	`#!/usr/bin/php -q`
`17:22.03`	`clyrrad`	`<?php`
`17:22.03`	`clyrrad`	`<PROTECTED>`
`17:22.09`	`rene1`	`Mandrak3: that version you have for mysql and mysql-server`
`17:22.23`	`NeonLevel`	`good day everyone, anyone has noticed that the process of mpg123 "get stuck" even asterisk stopped, and this produces a 100% usage of the cpus power////. is there some wayto fix this?`
`17:22.25`	`docelmo`	`You have something wrong in you rdialplan`
`17:22.27`	`crCernier`	`I'm registered to an outbound SIP named "switzernet", and in my dialplan, I can successfully make a call with "Dial(SIP/123456789@switzernet)". But now, I want to make the same call with a .call file, and it doesn't work :-( I tried with "Channel: SIP/switzernet/123456789" (and other directives like CallerID, Context, Extension and Priority), but the Asterisk console say "Call failed to go through, reason 3"... what should I do ?`
`17:22.41`	`rene1`	`NeonLevel: dont use mp3 for your music`
`17:22.58`	`rene1`	`if you take the time to convert to your native format you will have less CPU usage and better stability`
`17:23.04`	`clyrrad`	`exten => 4,n,Agi(check-queue.agi)`
`17:23.07`	`NeonLevel`	`i see`
`17:23.50`	`NeonLevel`	`rene1, native music format, which format should be that?`
`17:23.52`	`rene1`	`it is worth it,`
`17:23.53`	`rene1`	`well`
`17:23.55`	`docelmo`	`and your file is located in /var/lib/asterisk/agi right ?`
`17:23.55`	`rene1`	`it depends`
`17:24.03`	`rene1`	`what codec are your phones using?`
`17:24.07`	`*** join/#asterisk Dr^Mouse (n=noneof@89-145-198-64.xdsl.murphx.net)`
`17:24.16`	`Dr^Mouse`	`HELLO EVERYBODY!`
`17:24.19`	`rene1`	`hey`
`17:24.32`	`clyrrad`	`yes its in there`
`17:24.34`	`rene1`	`if you are using GSM then convert your audio to GSM`
`17:24.37`	`NeonLevel`	`oh i see`
`17:24.47`	`[TK]D-Fender`	`NeonLevel: Not a format, but rather *'s built in MoH player....`
`17:24.53`	`NeonLevel`	`rene1 ill do taht`
`17:25.05`	`docelmo`	`I dont know what to tell you except you screwed something up somewhere..`
`17:25.13`	`EmleyMoor`	`I get this:`
`17:25.16`	`[TK]D-Fender`	`NeonLevel: Use MP3's, they're fine, just us the * "Natve" MoH mode.`
`17:25.19`	`docelmo`	`Cause everything I saw was right`
`17:25.23`	`EmleyMoor`	`Aug 25 18:24:57 NOTICE[5349]: chan_sip.c:11045 handle_request_register: Registration from '<sip:phil>' failed for '62.49.246.87' - Username/auth name mismatch`
`17:25.30`	`docelmo`	`~pb`
`17:25.34`	`jbot`	`i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/`
`17:25.35`	`NeonLevel`	`ok but how do i disable asterisk to use mpg123 for MoH?`
`17:25.35`	`clyrrad`	`yea the script works on my other server`
`17:25.35`	`EmleyMoor`	`... when I try to connect a linphone to asterisk`
`17:25.59`	`Dr^Mouse`	`ok, i cant get the register utility for digiums g729 codec to work. how do i tell which glibc version i have? or does anyone know - i'm running latest debian stable on em64t`
`17:26.01`	`NeonLevel`	`[TK]D-Fender: how do i do that?`
`17:26.26`	`EmleyMoor`	`How do I resolve this? I can't tie up in my mind what the Username and auth name are and where they are defined`
`17:26.36`	`rene1`	`NeonLevel, you have to take the mp3 out of your moh directory then issue a moh files reload`
`17:26.43`	`rene1`	`or restart asterisk]`
`17:27.04`	`*** join/#asterisk DarKnesS_WolF (n=wolf@62.135.2.227)`
`17:27.05`	`[TK]D-Fender`	`NeonLevel: Set your type to "mode=files" in musiconhold.conf and make sure you have compiled asterisk-addons`
`17:27.32`	`clyrrad`	`anyone else know what could be wrong with asterisk saying that agi script does not exist?`
`17:27.49`	`NeonLevel`	`thanks guys ill do that`
`17:28.20`	`crCernier`	`NeonLevel: You can convert a mp3 with one command line, with mplayer : "mplayer -really-quiet -quiet -shuffle -ao pcm -af format=alaw,pan=1:0.5:0.5,resample=8000 file.mp3"`
`17:28.40`	`NeonLevel`	`ok got that`
`17:28.46`	`NeonLevel`	`but which way to go`
`17:28.53`	`NeonLevel`	`convert my mp3 files to alaw`
`17:28.56`	`bobnormal`	`nivex: still no luck :( http://pastebin.ca/149251 <-- log`
`17:29.05`	`bobnormal`	`whats going on? incoming call from gizmo causes weirdness: http://pastebin.ca/149251`
`17:29.05`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`17:29.21`	`NeonLevel`	`or use the files directive in musiconhold.conf`
`17:29.21`	`crCernier`	`NeonLevel: Now remove some useless parameters: "mplayer -ao pcm -af format=alaw,pan=1:0.5:0.5,resample=8000 file.mp3"`
`17:29.24`	`rene1`	`D-Fender: any clue on why my PRI E1 card is getting congestion signals after some hundred calls or some hours ??`
`17:29.25`	`clyrrad`	`here is my error can someone please help: Launched AGI Script /var/lib/asterisk/agi-bin/check-queue.agi`
`17:29.25`	`clyrrad`	`Failed to execute '/var/lib/asterisk/agi-bin/check-queue.agi': No such file or directory`
`17:29.27`	`rene1`	`NeonLevel: both`
`17:29.33`	`anthonyl`	`i usually just use mpg123 -w name.wav name.mpg , then use sox to convert the auto file to whatever codec is neeeded`
`17:29.36`	`clyrrad`	`that AGI script is there at that path`
`17:30.25`	`*** join/#asterisk redondos (n=redondos@190.48.23.163)`
`17:30.40`	`redondos`	`SIP using broadvoice -> http://pastebin.ca/149270`
`17:30.41`	`crCernier`	`anthonyl: Why use 2 commands when mplayer can do everything in one command ?`
`17:30.47`	`redondos`	`Can you please tell me what's wrong?`
`17:30.55`	`caio1982`	`is it really needed to patch both asterisk and zaptel sources to get sangoma's wanpipe built ok? or is it only for hdlc support for real?`
`17:31.10`	`anthonyl`	`crCernier, welk the one using mpg123 and sox is just easier for me to recall of the top of my head there are much less flags..etc`
`17:32.34`	`crCernier`	`anthonyl: I'm an mplayer fan ;-) It's much more powerfull than sox`
`17:32.48`	`bobnormal`	`also posted my sip.conf there too now. (@ http://pastebin.ca/149251 - gizmo incoming calls broken.)`
`17:32.51`	`anthonyl`	`:)`
`17:32.58`	`Strom_C`	`redondos: perhaps you've dialed an invalid number?`
`17:33.46`	`crCernier`	`I'm registered to an outbound SIP named "switzernet", and in my dialplan, I can successfully make a call with "Dial(SIP/123456789@switzernet)". But now, I want to make the same call with a .call file, and it doesn't work :-( I tried with "Channel: SIP/switzernet/123456789" (and other directives like CallerID, Context, Extension and Priority), but the Asterisk console say "Call failed to go through, reason 3"... what should I do ?`
`17:33.48`	`*** join/#asterisk Givemelove2k (n=bob@208.57.229.162)`
`17:33.57`	`redondos`	`Strom_C: Oh, maybe.`
`17:34.08`	`[TK]D-Fender`	`NeonLevel: Leave them as MP# and use Native MoH. You can change them to anotehr format as well if you really want to, but the key is getting rid of MPG123`
`17:34.22`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`17:34.30`	`Dr^Mouse`	`ok, just checked by instruictions i found elsewhere - im on glibc 2.3.2, so i got the glibc2.3 version on the register utility and now it says...`
`17:34.37`	`*** join/#asterisk QbY (n=Kelvin@cm-64-221-172-66.dhcp.southerncoastalcable.net)`
`17:34.41`	`Dr^Mouse`	`-- /root/register: relocation error: /root/register: symbol __guard, version GLIBC_2.3.2 not defined in file libc.so.6 with link time reference`
`17:34.54`	`*** join/#asterisk a1fa (n=a1fa@207.210.210.202)`
`17:35.03`	`QbY`	`Does anyone know a way to get the value of MONITOR_FILENAME into queue_log?`
`17:35.39`	`a1fa`	`http://svn.digium.com/svn/asterisk/tags/1.2.11.0 is the right tag?`
`17:35.52`	`bobnormal`	`http://pastebin.ca/149277 <- improved bug report for incoming call failure, includes: SIP dump & sip.conf. please help!`
`17:36.08`	`*** join/#asterisk svenna_ (n=svenna@p548D373F.dip0.t-ipconnect.de)`
`17:36.11`	`a1fa`	`http://svn.digium.com/svn/asterisk/tags/1.2.11/`
`17:36.13`	`a1fa`	`:P`
`17:36.30`	`*** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net)`
`17:36.36`	`FlyboySR22`	`on AR02`
`17:36.53`	`c4t3l`	`anyone ever hear of the pcom 601 expansion module crapping out after a reboot. Specifically the red leds lighting up when extensions are in the middle of calls?`
`17:37.37`	`*** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`17:37.47`	`*** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`17:38.20`	`Dr^Mouse`	`anyone?`
`17:38.24`	`redondos`	`Strom_C: Does it have anything to do with the extension you put in "register => <phonenumber>@sip.broadvoice.com:<password>:<phonenumber>@sip.broadvoice.com/<extension>" in sip.conf ?`
`17:39.24`	`charles___`	`hey guys`
`17:39.29`	`charles___`	`anyone using MCC or ASTPP ?`
`17:39.44`	`c4t3l`	`negative`
`17:39.58`	`eKo1`	`I don't even know what those are.`
`17:40.05`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`17:40.34`	`rene1`	`nada`
`17:40.55`	`rene1`	`never heard either`
`17:41.04`	`c4t3l`	`fax technology also sucks!`
`17:41.28`	`rene1`	`this congestion problem.. what could it be causing, i am reading pri intense debugs but cant really find out anything about it....`
`17:41.38`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`17:41.51`	`TripleFFFF`	`is there a way to pass a php file to application of musiconhold`
`17:42.16`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`17:42.16`	`*** mode/#asterisk [+o mog] by ChanServ`
`17:42.23`	`rene1`	`TripleFFFF; that didnt make much sense`
`17:42.35`	`rene1`	`what do you want to do`
`17:43.02`	`TripleFFFF`	`like run a php applicationt will stream the files`
`17:43.21`	`rene1`	`well you can change the MOH class via a CLI application`
`17:43.27`	`TripleFFFF`	`i mean php application or agi that iwll decide WAHT to stream`
`17:43.30`	`TripleFFFF`	`instead of contexts`
`17:43.40`	`rene1`	`you could`
`17:43.43`	`TripleFFFF`	`so nothing in files`
`17:44.42`	`Dr^Mouse`	`has anyone got any idea about this error?`
`17:44.57`	`rene1`	`TripleFFFF: show application SetMusicOnHold`
`17:45.46`	`rene1`	`on the other hand if you try to stream the music from a remote source like shoutcast or what not well you could change the music in the streaming server but thats upt o you to find out how`
`17:46.10`	`*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)`
`17:46.19`	`rene1`	`if you dont control the server and want to change the streaming source i guess you need to define your streaming sources in files and show application SetMusicOnHold again`
`17:46.19`	`TripleFFFF`	`yes but i mean`
`17:46.25`	`[TK]D-Fender`	`bobnormal: I see a "+" at the front of that # you are dialing... get rid of it...`
`17:46.33`	`TripleFFFF`	`no`
`17:47.30`	`rene1`	`what are you trying to achieve`
`17:49.11`	`TripleFFFF`	`trying to use res_musiconhold.c:852 moh_register: Unable to open pseudo channel for timing... Sound may be choppy.`
`17:49.11`	`TripleFFFF`	`*CLI>`
`17:49.15`	`TripleFFFF`	`its that normal ?`
`17:49.17`	`*** part/#asterisk pointer (i=pointer@aj.catt.com)`
`17:49.27`	`rene1`	`mm i am not sure`
`17:49.29`	`TripleFFFF`	`BTW im trying to run a php script instead of madplay.. but IT will run madplay etc`
`17:50.10`	`rene1`	`well do you have any zaptel interfaces compiled or ztdummy maybe?`
`17:50.45`	`*** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net)`
`17:50.52`	`*** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`17:51.01`	`Dr^Mouse`	`ok then, im ringing digium tech support about it`
`17:51.14`	`*** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`17:52.14`	`rene1`	`D-fender, should i ask in DEV? about this problem i am having? fill a bug report? capture backtraces?`
`17:53.00`	`TripleFFFF`	`MusicOnHold cant mak it like 60 sec ?`
`17:53.02`	`TripleFFFF`	`as befre`
`17:53.06`	`Dr^Mouse`	`another one, if asterisk has an incoming sip call using one codec, and then transfers to a sip device which doesnt support that codec, what is the default behaviour? does it find a codec which works with both, or does it transcode?`
`17:53.12`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`17:53.21`	`Strom_C`	`Dr^Mouse: it transcodes`
`17:53.37`	`TripleFFFF`	`zaptel worked`
`17:54.02`	`blitzrage`	`Dr^Mouse: asterisk will try to find a codec that both end points understand, if not, then it transcodes and stays in the media path`
`17:55.01`	`rene1`	`Dr^Mouse, each endpoint advertises to asterisk what codecs it understands`
`17:55.33`	`TripleFFFF`	`make izaptel was done`
`17:55.37`	`TripleFFFF`	`so why is taht ?\`
`17:55.43`	`TripleFFFF`	`theres no soundcard on box`
`17:55.47`	`blitzrage`	`rene1: yes, done in the SDP`
`17:55.58`	`Strom_C`	`TripleFFFF: did you modprobe ztdummy?`
`17:56.04`	`blitzrage`	`soundcards have nothing to do with zaptel`
`17:56.16`	`Dr^Mouse`	`Strom_C rene1 and blitzrage: ok, but what if it changes endpoints, eg transfer?`
`17:56.28`	`blitzrage`	`Dr^Mouse: then asterisk continues to stay in the media path if it needs to`
`17:56.29`	`rene1`	`blitzrage: is there something in asterisk that negotiates towards the lowest cpu intensive transcoding`
`17:56.32`	`blitzrage`	`both end points just talk to Asterisk`
`17:56.51`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`17:56.51`	`*** mode/#asterisk [+o mog] by ChanServ`
`17:56.52`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`17:56.55`	`*** part/#asterisk QbY (n=Kelvin@cm-64-221-172-66.dhcp.southerncoastalcable.net)`
`17:57.00`	`blitzrage`	`rene1: it does if you order you disallow=all; allow=<my_codec> in such a way that it negotiates it that way`
`17:57.08`	`blitzrage`	`the order you list the codecs matters`
`17:57.10`	`rene1`	`Dr^Mouse: unless told not to via canreinvite`
`17:57.14`	`rene1`	`blitzrage: cool`
`17:57.16`	`*** join/#asterisk angom (n=angom@red-corp-200.79.148.126.telnor.net)`
`17:57.24`	`EmleyMoor`	`Can I sign up for IAXtel from outside the USA#`
`17:57.24`	`EmleyMoor`	`?`
`17:57.32`	`Qwell`	`sure`
`17:57.32`	`jbalcomb`	`WTF mate, I have 'wrapuptime = 30' on my queues but reps are reporting that calls are coming in as soon as they hang up or even while they are on calls. what gives?`
`17:57.35`	`blitzrage`	`canreinvite=yes will be ignored if codecs cannot be negotiated for both end points`
`17:57.55`	`rene1`	`EmleyMoor: i was able to why you arent?`
`17:57.58`	`TripleFFFF`	`whats soundcad name in dmesg normaly`
`17:58.12`	`EmleyMoor`	`renel: Did you have to lie to it?`
`17:58.18`	`rene1`	`no`
`17:58.24`	`rene1`	`IIRC`
`17:58.38`	`Dr^Mouse`	`ok i cant get thru to digium support. i have paid(!) for 2 g729 licenses to test, and now they wont register. the registration program wont work.`
`17:58.39`	`TripleFFFF`	`modprobe ztdummy`
`17:58.39`	`TripleFFFF`	`Notice: Configuration file is /etc/zaptel.conf`
`17:58.39`	`TripleFFFF`	`line 0: Unable to open master device '/dev/zap/ctl'`
`17:58.48`	`EmleyMoor`	`I can't select other than USA`
`17:58.51`	`Strom_C`	`Dr^Mouse: call 877-linux-me`
`17:59.00`	`rene1`	`TripleFFFF modprobe zaptel first?`
`17:59.10`	`TripleFFFF`	`that the modprobe result`
`17:59.10`	`blitzrage`	`redhat based systems can use rc scripts if you do "make config" which will load ztdummy if no hardware is found`
`17:59.29`	`*** join/#asterisk zoa (n=d@pirus.securax.be)`
`17:59.30`	`zoa`	`hey ho`
`17:59.36`	`blitzrage`	`zoa: !!!`
`17:59.36`	`rene1`	`hey`
`17:59.39`	`zoa`	`aha!`
`17:59.45`	`Strom_C`	`a-ha?`
`17:59.47`	`blitzrage`	`zoa: I still need to report some bugs :)`
`17:59.52`	`zoa`	`hehe`
`17:59.55`	`blitzrage`	`stupid time not slowing down or stopping :)`
`17:59.57`	`zoa`	`go go go`
`18:00.14`	`blitzrage`	`they are all minor though... except when the jitterbuffer has to resync ;)`
`18:00.18`	`TripleFFFF`	`FATAL: Error running install command for ztdummy`
`18:00.28`	`Dr^Mouse`	`Strom_C: i'm in the UK (i know this makes me sad and a geek coz its 7pm friday night and im still sat here working)`
`18:00.43`	`intralanman`	`TripleFFFF: check dmesg`
`18:00.49`	`Strom_C`	`Dr^Mouse: either use IAX2 to call in or dial +1 256 428 6000`
`18:00.49`	`*** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net)`
`18:01.27`	`*** join/#asterisk ToyMan (n=stuq@74-32-7-1.dsl1.mdl.ny.frontiernet.net)`
`18:01.51`	`rene1`	`sorry to keep asking about this: what can i do about a pri E1 line that gets congestion after either some time or some hundred calls... i can restart asterisk and issue a ztcfg, sometimes it requires a driver reload, and sometimes only a full reboot would inlock it... Ast 1.2.10, Zaptel 1.2.7, Debian 3.1r2, Kernel 2.6`
`18:02.28`	`rene1`	`s/inlock/unlock/`
`18:02.46`	`Strom_C`	`rene1: which card?`
`18:02.54`	`rene1`	`2 port E1 card, digium`
`18:02.58`	`intralanman`	`damn.... i just realized... jbot knows sed`
`18:03.01`	`Strom_C`	`rene1: contact digium support`
`18:03.02`	`Dr^Mouse`	`Strom_C: what is that number? if its for digium, i've tried via IAX2/guest@misery.digium.com/s but ill try again`
`18:03.19`	`TripleFFFF`	`so`
`18:03.20`	`Strom_C`	`Dr^Mouse:thats the iax2 connection for digium`
`18:03.29`	`TripleFFFF`	`i did`
`18:03.33`	`Strom_C`	`Dr^Mouse: the 256 number is the direct non tollfree number`
`18:03.50`	`rene1`	`Strom_C thanks, do you think my telco could be throttling (sp?) down my calls?`
`18:04.16`	`Strom_C`	`rene1: i have no idea`
`18:04.42`	`rene1`	`it wouldnt hurt to ask them first`
`18:04.49`	`TripleFFFF`	`ok weird`
`18:04.52`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`18:05.24`	`*** join/#asterisk Assid (i=assid@203.115.83.215)`
`18:08.35`	`TripleFFFF`	`hmm ifconfig -a doesnt show interface 10 or 100`
`18:09.08`	`backblue`	`TripleFFFF: that coulde be very good, could mean you have a 1000 or 10000 interface.`
`18:09.12`	`zoa`	`there is another tool for that`
`18:09.41`	`zoa`	`afaik ifconfig -a never shows the interface type`
`18:09.44`	`TripleFFFF`	`no`
`18:09.45`	`TripleFFFF`	`lol`
`18:10.00`	`zoa`	`just a sec`
`18:10.06`	`TripleFFFF`	`i saw fuckers at adatacenter where running /sbin/mii-tool -F 10BaseT-FD`
`18:10.08`	`TripleFFFF`	`k`
`18:10.11`	`zoa`	`iidiag or so`
`18:10.21`	`TripleFFFF`	`so i commented lol`
`18:10.22`	`zoa`	`mii-diag it is`
`18:10.31`	`zoa`	`and ethtool or something`
`18:10.38`	`TripleFFFF`	`mii-diag`
`18:10.38`	`TripleFFFF`	`-bash: mii-diag: command not found`
`18:10.38`	`zoa`	`or mii-tool`
`18:10.43`	`zoa`	`google :p`
`18:11.05`	`zoa`	`http://freshmeat.net/projects/mii-diag/`
`18:11.10`	`TripleFFFF`	`mii-tool -v`
`18:11.10`	`TripleFFFF`	`eth0: 10 Mbit, full duplex, link ok`
`18:11.10`	`TripleFFFF`	`<PROTECTED>`
`18:11.11`	`TripleFFFF`	`<PROTECTED>`
`18:11.11`	`TripleFFFF`	`<PROTECTED>`
`18:11.11`	`TripleFFFF`	`<PROTECTED>`
`18:11.12`	`TripleFFFF`	`<PROTECTED>`
`18:11.18`	`zoa`	`��`
`18:11.21`	`zoa`	`�� `
`18:11.28`	`zoa`	`oops`
`18:11.28`	`TripleFFFF`	`hmm does say what its running .. sorry.. tough i pasted last line`
`18:11.40`	`Qwell`	`Just because it's advertised, doesn't mean it'll work`
`18:11.51`	`TripleFFFF`	`h,,,`
`18:12.20`	`TripleFFFF`	`divert: allocating divert_blk for eth0`
`18:12.21`	`TripleFFFF`	`e1000: eth0: e1000_probe: Intel(R) PRO/1000 Network Connection`
`18:12.24`	`rene1`	`Qwell: like the ab shaper or ma -evans?`
`18:12.26`	`TripleFFFF`	`e1000: eth0: e1000_watchdog: NIC Link is Up 10 Mbps Full Duplex`
`18:12.26`	`TripleFFFF`	`d`
`18:12.43`	`TripleFFFF`	`ok ... divert_blk is ipv6 crap i thnkg`
`18:12.53`	`TripleFFFF`	`they boycutting my eth0 fuckers`
`18:13.35`	`*** join/#asterisk goozbach (n=goozbach@brooks.netradius.com)`
`18:14.10`	`Qwell`	`btw...`
`18:14.12`	`Qwell`	`~pb`
`18:14.19`	`jbot`	`rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/`
`18:14.21`	`TripleFFFF`	`sory`
`18:14.21`	`TripleFFFF`	`qwesl`
`18:14.49`	`EmleyMoor`	`Is there a "no fog to knit" way to have a go at setting up asterisk with a softphone?`
`18:14.49`	`intralanman`	`s/qwesl/qwell/`
`18:15.34`	`TripleFFFF`	`yeah`
`18:15.38`	`TrixVox`	`is "show registry" an accurate representation of whether or not you're registered? i changed my iax.conf to have the wrong password and it still says Registered`
`18:15.45`	`TripleFFFF`	`yes`
`18:16.35`	`cocomp`	`hi - basic question - how can you tell if mobo supports pci 2.2 as required by digium analogue cards (TDM12B)?`
`18:16.56`	`Qwell`	`cocomp: If it's fairly even, it should be 2.2`
`18:17.08`	`Dr^Mouse`	`another one: if you specify canreinvite=yes, but have dtmfmode=info, will asterisk still remain in the media path with the t or T dial options?`
`18:17.28`	`cocomp`	`Qwell: not sure I understand that ;)`
`18:17.41`	`Qwell`	`cocomp: The machine/mb.`
`18:18.10`	`cocomp`	`Mobo is tyan S2882 1x PCI 32bit 4 x PCI-X 64`
`18:18.20`	`DrkShdw`	`TrixVox: did you try iax2 reload before the registry? or is it still registered from before when the passwrod was correct`
`18:19.11`	`cocomp`	`card is in 32bit slot - shows up in lspci but no interrupt assigned (not sure if it should have) and zaptel doesn't find hardware so loads ztdummy`
`18:19.20`	`redondos`	`I am configuring a broadvoice peer, and when calling it says that "the device you're using is not registered to place calls on the network". What does it mean?`
`18:19.25`	`bobnormal`	`still no luck receiving calls from gizmo project :(`
`18:21.00`	`*** join/#asterisk asteriskbie2 (n=at@59.93.70.212)`
`18:21.29`	`*** join/#asterisk yosyp (n=yosyp@pool-70-108-99-62.res.east.verizon.net)`
`18:22.03`	`yosyp`	`What's the hardware I need to just connect my (1) home phone line to my computer to use with asterisk? I'm not looking to spend a lot of money, just trying it out.`
`18:22.23`	`EmleyMoor`	`A suitable FXO card`
`18:22.37`	`cocomp`	`Dev kit from digium.com - $241US`
`18:22.46`	`yosyp`	`$241?!!`
`18:22.57`	`intralanman`	`an x100p for about $30`
`18:23.01`	`cocomp`	`k`
`18:23.05`	`asteriskbie2`	`salutations! I am new to AGI and looking for help related to DIAL command invoked with macro-screen which sets MACRO_RESULT`
`18:23.07`	`yosyp`	`intralanman: anything cheaper?`
`18:23.38`	`intralanman`	`heh, not that i know of.... the x100p is about as "cheap" as they get`
`18:23.39`	`asteriskbie2`	`how do I get that value (MACRO_RESULT) after EXEC DIAL SIP/NUM@sip\|\|M(Screen)?`
`18:25.00`	`redox-`	`you can probably get it used even cheaper on ebay`
`18:25.06`	`asteriskbie2`	`can I do that at all?`
`18:25.30`	`tzafrir_home`	`asteriskbie2, what's wrong with $PMACRO_RESULT} ?`
`18:25.34`	`intralanman`	`not much cheaper i think.....`
`18:25.44`	`tzafrir_home`	`asteriskbie2, ${MACRO_RESULT}, that is`
`18:26.14`	`tzafrir_home`	`intralanman, I got a couple for ~20$ . Which included shipping overseas`
`18:26.29`	`tzafrir_home`	`20$ per card`
`18:26.41`	`asteriskbie2`	`tzafrir: does it work in AGI?`
`18:26.48`	`iso`	`what does the TDM400P have over the x100p besides expandability`
`18:26.51`	`intralanman`	`shipping included, that's not too bad`
`18:26.57`	`asteriskbie2`	`tzafir: using GET VAR`
`18:27.15`	`intralanman`	`iso: FXO and FXS`
`18:27.19`	`myiagy`	`asteriskbie2 http://www.voip-info.org/wiki/view/get+variable`
`18:27.35`	`intralanman`	`or at least the options`
`18:28.10`	`iso`	`intralanman: the x100p has a FS0 as well as FSX doesn't it?`
`18:28.32`	`intralanman`	`i don't think so`
`18:28.35`	`iso`	`you can all or receive calls with the x100P`
`18:28.41`	`asteriskbie2`	`myiagy: I already took a look at it. However my experience with AGI is very limited. I have issued DIAL and it immediately returns 1.`
`18:28.44`	`Dr^Mouse`	`iso: many things. it has 4 ports rather than one, each can be fxo or fxs, it can be tuned to the country you are in's line types, it has onboard hardware echo cancellation (i think), and is overall better quality.`
`18:29.47`	`myiagy`	`asteriskbie2 well, what language are you using? php?`
`18:29.54`	`asteriskbie2`	`yes php`
`18:30.01`	`iso`	`Mouse,Intralan: thanks`
`18:30.59`	`*** join/#asterisk patient_zero (n=tjzewez@209.107.188.100)`
`18:31.11`	`myiagy`	`write("GET VARIABLE MACRO_RESULT"); $m = read(); $macro_result = substr($m,14); $macro_result = substr($macro_result,0,-1);`
`18:31.16`	`TripleFFFF`	`intralanman i want one please`
`18:31.16`	`myiagy`	`that should do it`
`18:31.22`	`myiagy`	`there is an easier way though`
`18:31.22`	`tzafrir_home`	`iso, the X100P has a single FXO port`
`18:31.36`	`myiagy`	`if you use some already made class.. like phpagi`
`18:31.41`	`redondos`	`If a user dials "0012" ${EXTEN} expands to just that, but I want to trim the first zero. How can I do that?`
`18:31.43`	`intralanman`	`TripleFFFF: ????`
`18:31.44`	`myiagy`	`there are links to classes on the wiki`
`18:33.17`	`intralanman`	`phpagi rocks`
`18:33.27`	`asteriskbie2`	`myiagy: Thanks. I will try that. A related question. Do you have Macro-Screen which works perfectly - I mean accepts 1 and then connects. I have seen few versions but none of them work if the callee hangs up the call`
`18:34.06`	`myiagy`	`i don't even know what macro-screen does..`
`18:34.33`	`TripleFFFF`	`wat ?`
`18:34.37`	`asteriskbie2`	`hmm - why were you using MACRO_RESULT for?`
`18:34.38`	`TripleFFFF`	`i watn a 100xp card`
`18:35.09`	`myiagy`	`i'm not, i was just telling you how to do it`
`18:35.32`	`Dr^Mouse`	`tripleFFFF: no you dont, the sound quality (on the 3 i've tried) is appaling`
`18:35.33`	`asteriskbie2`	`but that was fast - very fast. I thought you copied those lines from your code and hence the question`
`18:35.48`	`redondos`	`Done.`
`18:35.48`	`TripleFFFF`	`oh`
`18:35.55`	`TripleFFFF`	`lol well anyothr way to get a zap line ?`
`18:36.00`	`TripleFFFF`	`pci based`
`18:36.02`	`TripleFFFF`	`and cheap`
`18:36.15`	`myiagy`	`asteriskbie2 i used to do that a lot before i learned about phpagi :)`
`18:36.41`	`EmleyMoor`	`I will try FWD with a straightforward softphone first then do asterisk another day I think`
`18:36.52`	`tzafrir_home`	`TripleFFFF, no cheaper way.`
`18:37.35`	`tzafrir_home`	`TripleFFFF, if you have ISDN, you can get a HFC-S-based card and use bristuff`
`18:38.50`	`Dr^Mouse`	`tripleffff: the way we are trying atm is main lines over sip (shameless plug for UK VoIP Line Provider: LVDX.com is fantastic) and an x100p as an emergency backup line in case the internet dies.`
`18:38.56`	`tzafrir_home`	`TripleFFFF, but a X100P card is very cheap. Quite a nice toy if you need to play with connecting asterisk to a telco`
`18:40.10`	`EmleyMoor`	`Once I have all of that working I will be after a TDM31B or TDM2411B`
`18:41.10`	`*** join/#asterisk copantl (n=galel@190.4.23.242)`
`18:41.14`	`copantl`	`hello`
`18:42.15`	`*** join/#asterisk ToyMan (n=stuq@74-32-7-1.dsl1.mdl.ny.frontiernet.net)`
`18:43.32`	`EmleyMoor`	`Is there a non-encumbered softphone I can use on Debian?`
`18:44.12`	`copantl`	`any body know what this mean: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 31`
`18:44.12`	`copantl`	`?`
`18:45.19`	`rene1`	`EmleyMoor: moziax?`
`18:46.31`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`18:46.32`	`cocomp`	`Anyone know if a tdm12B analogue card should show up in /proc/interrupts (and what it would be called if it should)?`
`18:47.58`	`[TK]D-Fender`	`EmleyMoor: Ekiga`
`18:48.11`	`copantl`	`any body from digium ?`
`18:48.11`	`cocomp`	`EmleyMoor: linphone`
`18:48.15`	`[TK]D-Fender`	`cocomp: "WCTDM"`
`18:48.19`	`cocomp`	`k`
`18:48.29`	`copantl`	`i got this: chan_zap.c:2289 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway!`
`18:48.41`	`EmleyMoor`	`linphone is complicated :-(`
`18:48.43`	`copantl`	`my te110p`
`18:49.19`	`*** join/#asterisk ncrucial (i=[XNrZp5E@bolt.sonic.net)`
`18:49.24`	`cocomp`	`[TK]D-Fender: and that should show up before modules etc are loaded right?`
`18:49.29`	`*** join/#asterisk Givemelove2k (n=bob@208.57.229.162)`
`18:49.37`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.69)`
`18:50.34`	`*** join/#asterisk Givemelove2k (n=bob@208.57.229.162)`
`18:51.16`	`Mandrak3`	`Problem Compiling Asterisk-Addons "undefined symbol: _intel_fast_memcpy"`
`18:51.26`	`Mandrak3`	`what can I do?`
`18:51.28`	`ncrucial`	`Anyone good at tuning FXO? I have been reading up on it but it seems like a nightmare to try and patch ztmonitor for numbers, etc. etc. Is this old information or is it still the way easiest way to tune them?`
`18:51.53`	`ncrucial`	`and when i run fxotune -s, or fxotune -i i never seem to get any numbers in my /etc/fxotune.conf`
`18:51.54`	`crCernier`	`I'm registered to an outbound SIP named "switzernet", and in my dialplan, I can successfully make a call with "Dial(SIP/123456789@switzernet)". But now, I want to make the same call with a .call file, and it doesn't work :-( I tried with "Channel: SIP/switzernet/123456789" (and other directives like CallerID, Context, Extension and Priority), but the Asterisk console say "Call failed to go through, reason 3"... what should I do ?`
`18:52.07`	`*** join/#asterisk fgifgi123 (n=Hobbit@cpe-24-30-185-100.socal.res.rr.com)`
`18:53.18`	`fgifgi123`	`can anyone recommend or not recommend a wifi sip phone? I can't seem to find any in-depth reviews or hands on experience. Looking at d-link, zyxel, ustarcomm, linksys`
`18:53.55`	`Mandrak3`	`!!! Problem Compiling Asterisk-Addons "undefined symbol: _intel_fast_memcpy" !!! Help please`
`18:54.51`	`Dr^Mouse`	`ncrucial: are the zaptel and wcfxo modules loaded? issue "cat /proc/zaptel/1" to check`
`18:55.13`	`[TK]D-Fender`	`cocomp: Make sure you've modprobed it and ZAptel first, adn it should show up`
`18:55.48`	`cocomp`	`[TK]D-Fender: tried - still comes up with error no hardware found`
`18:56.11`	`[TK]D-Fender`	`fgifgi123: All WiFi phones suck, some just a little less than others.`
`18:56.29`	`[TK]D-Fender`	`cocomp: Recompile Zaptel...`
`18:56.32`	`fgifgi123`	`that's what I figured`
`18:56.38`	`fgifgi123`	`which one sucks the least? :)`
`18:57.03`	`[TK]D-Fender`	`fgifgi123: I'f you're talking about wireless use in a fixed area, get an ATA + analog cordless phone`
`18:57.16`	`Dr^Mouse`	`fgifgi123: Dont use the UTStarcom F1000`
`18:57.18`	`cocomp`	`[TK]D-Fender: tried with 2 machines using rpms and svn source versions -same issue - I think it's pci / irq trouble`
`18:57.20`	`ncrucial`	`Dr^Mouse the zaptel and the wctdm are loaded (using a TDM400P)`
`18:57.32`	`[TK]D-Fender`	`fgifgi123: The Hitachi one is pretty decent as far as they go, and apparently Linksys isn't TOO bad either...`
`18:57.44`	`Dr^Mouse`	`ncrucial: are you running it as root? dont know if that makes a difference`
`18:58.13`	`ncrucial`	`Dr^Mouse Yeah, as root.`
`18:58.15`	`cocomp`	`[TK]D-Fender: should show up in bios as well - eg. multimedia controller or similar - not seeing that - so got to be hardware issue.`
`18:59.16`	`cocomp`	`however bizarre that is shows up in lspci output :(`
`18:59.24`	`EmleyMoor`	`OK - can anyone recommend a softphone that works behind nat, is easy to connect directly to FWD and will also connect easily to asterisk when I come to try that out, that doesn't come with the need for loads of otherwise needless stuff?`
`19:00.24`	`fgifgi123`	`thanks Dr^Mouse and [TK]D-Fender, btw the hitachi ip3000 or ip5000?`
`19:00.32`	`Dr^Mouse`	`cocomp: try what d-fender suggested - recompile zaptel. i would also suggest getting the latest fxotune from trunk, it does a much better job. but just try recompiling and reinstalling zaptel first.`
`19:00.50`	`[TK]D-Fender`	`fgifgi123: I only knew of the 3000 at the time, unsure of the other`
`19:00.53`	`EmleyMoor`	`(could use linphone if someone could translate it for me)`
`19:00.53`	`cocomp`	`k`
`19:01.27`	`Dr^Mouse`	`fgifgi123: I am currently trialing the UTStarcom F3000, its going well, and its a cool toy :)`
`19:02.16`	`[TK]D-Fender`	`Dr^Mouse: One of my clients is using a bunch of UTSC's and they time out constantly with ping times > 1s and general flakeyness.... YMMV`
`19:03.27`	`Mandrak3`	`<PROTECTED>`
`19:03.58`	`rene1`	`hitachis are not that bad but not flawless either`
`19:04.00`	`Dr^Mouse`	`[TK]D-Fender: i found the ping was constantly 10-160ms on a lan, but I had to increase the qualify time as that was often over 2s. However speach quality was still fantastic.`
`19:04.18`	`crCernier`	`EmleyMoor: Did you try KIAX ?`
`19:04.46`	`crCernier`	`EmleyMoor: Or iaxcom ?`
`19:04.55`	`EmleyMoor`	`Not heard of those`
`19:05.18`	`rene1`	`EnleyMoor: IAX softphones are easier to connecto to asterisk or asterisk compatible ITSPs like FWD or voicepulse connet`
`19:05.20`	`rene1`	`connect`
`19:05.31`	`crCernier`	`EmleyMoor: Search for IAX softphones... and forget SIP !`
`19:05.53`	`rene1`	`there are zero nat issues it just plain works, think msn or think skype`
`19:06.15`	`*** join/#asterisk ToyMan (n=stuq@74-32-7-1.dsl1.mdl.ny.frontiernet.net)`
`19:06.18`	`[TK]D-Fender`	`Dr^Mouse: I never heard about the call aspect, just noticed the network aspect as I worked on their system.`
`19:07.02`	`yosyp`	`i'm calling digium`
`19:07.05`	`yosyp`	`over asterisk :P`
`19:07.50`	`EmleyMoor`	`crCerenier: Do you know a good iax softphone that is in Debian stable?`
`19:07.51`	`clyrrad`	`I am having a strange issue - I have some call forwarding logic in my dial plan which is working - but as soon as the dial plan reaches the Dial statment - it dials the call forward number - the problem is - Asterisk keeps going to the next step in the dial plan - once it hits a Dial should the channel not be sent out and stop executing the dial plan?`
`19:08.41`	`TripleFFFF`	`hey.. enum would be a great way to control a did.. right ? .. like to choose what provider it goes to.. but how does it work internaly ? on asterisk`
`19:08.57`	`clyrrad`	`it dials out the IAX and the IAX (the forwarded number) accepts the call - but asterisk keeps going down to the next step in the dial plan.... its really got me baffled`
`19:09.11`	`rene1`	`EmleyMoor: you can easily use moziax in your mozilla/firefox based browser`
`19:09.35`	`*** join/#asterisk GlobbeTotssz (n=eric@200.122.149.57)`
`19:09.53`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`19:10.04`	`[TK]D-Fender`	`clyrrad: Pastbin the CLI output and the whole exten from your dialplan`
`19:10.09`	`EmleyMoor`	`About to try it`
`19:10.22`	`*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)`
`19:10.43`	`*** join/#asterisk [hC] (i=root@donkey.voxter.ca)`
`19:10.55`	`clyrrad`	`ok`
`19:11.16`	`[hC]`	`This is kinda messed.. I'm trying to get rid of the MWI splash on my polycom ip430.. Ive done this a million times before on 501's and 601's, but on this one, as soon as the MWI indicator shows up, the phone locks up`
`19:11.19`	`[hC]`	`Anyone seen this happen?`
`19:11.25`	`EmleyMoor`	`renel: OK - what do I fill in to try it out on FWD?`
`19:12.03`	`rene1`	`basically FWD iax address, username and password`
`19:12.14`	`rene1`	`that would get you ready`
`19:12.44`	`EmleyMoor`	`But what is the FWD iax address?`
`19:12.59`	`*** join/#asterisk novafirst (n=kosta@wrt1.niclab.com)`
`19:13.03`	`Dr^Mouse`	`[TK]D-Fender: The F1000 is terrible. The speach was crackly, and the volume was too quiet even on max. Also, using it remotely our receptionist (who works from home) found that it killed her wifi router in the middle of a call. Often. The F3000 is miles better, although it still suffers from a painfully slow Web interface, which often times out.`
`19:13.14`	`novafirst`	`hi guys`
`19:13.38`	`[TK]D-Fender`	`Dr^Mouse: So I'll stick to my story that all WiFi = SUCK for now :)`
`19:14.02`	`novafirst`	`I see this error in log file: db.c: Unable to find key 'xxxx' in family SIP/Registry`
`19:14.29`	`novafirst`	`I have only one entry in sip.conf file`
`19:15.44`	`EmleyMoor`	`This is even less productive than sip was!`
`19:16.43`	`*** join/#asterisk lordbaron (n=redbaron@host55-226.rancor.birch.net)`
`19:16.56`	`[TK]D-Fender`	`novafirst: Pastebin CLI output and something else to back it up...`
`19:16.58`	`[TK]D-Fender`	`~pb`
`19:17.01`	`jbot`	`somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/`
`19:17.17`	`GlobbeTotssz`	`Hi what are the procedures that i need to follow to got some questions answered about Asterisk and NAT ?`
`19:18.58`	`[TK]D-Fender`	`GlobbeTotssz: For SIP : Forward 5060-5070, 10000-20000 UDP to *. in sip.conf under [general] put "externip=[yourip]", "localnet=[yourlocalsubnet]/[netmask]", and "nat=yes".`
`19:19.52`	`[TK]D-Fender`	`GlobbeTotssz: If you're referring to running * behind NAT. If * is public and a remote PHONE is behind NAT then you should only need set "nat=yes", and "qualify=yes" for that phone's entry`
`19:20.11`	`EmleyMoor`	`How do I connect moziax direct to FWD? I will of course be intending to connect it to asterisk at a later date but at the moment I am almost convinced this stuff is mythical`
`19:20.35`	`novafirst`	`[TK]D-Fender: I posted here http://pastebin.com/775877`
`19:21.32`	`lordbaron`	`Anyone have experiences (good or bad) with Centos x86_64 (v4.3) and compiling zaptel/asterisk 64bit?`
`19:22.04`	`[TK]D-Fender`	`novafirst: So far that has nothing to do with AstDB....`
`19:22.48`	`*** join/#asterisk kotique[male] (n=burumba@host-86-106-210-146.moldtelecom.md)`
`19:23.49`	`fgifgi123`	`rene1: which wifi phone do you prefer?`
`19:24.16`	`novafirst`	`[TK]D-Fender: the log file: db.c: Unable to find key 'myxlite' in family SIP/Registry`
`19:25.23`	`[TK]D-Fender`	`novafirst: Who said that had anything to do with sip.conf?`
`19:25.46`	`*** join/#asterisk zotz (n=zotz@24.244.133.115)`
`19:26.03`	`[TK]D-Fender`	`novafirst: When does that warning get generated? We need to see this as it happens in CLI to debug it...`
`19:26.26`	`novafirst`	`[TK]D-Fender: I assumed, since it mention SIP/Registry`
`19:26.34`	`[TK]D-Fender`	`EmleyMoor: Go to FWD and getthe connection instructions for your client.`
`19:26.37`	`novafirst`	`ok`
`19:27.03`	`Mandrak3`	`[app_saycountpl.so] => (Say polish counting words)`
`19:27.03`	`Mandrak3`	`<PROTECTED>`
`19:27.03`	`Mandrak3`	`<PROTECTED>`
`19:27.03`	`Mandrak3`	`Aug 25 21:08:05 WARNING[10235]: loader.c:554 load_modules: Loading module cdr_addon_mysql.so failed!`
`19:27.41`	`[TK]D-Fender`	`EmleyMoor: And you really DO want * to reg to FWD for you and you then need only connect your softphone to *.`
`19:28.12`	`rene1`	`fgifgl123: i like my macbook running jackeniax ;) no i havent played much with them, wifi phones, i have only played with hitachi models.. and well sjphone on pocket pc is also real nice`
`19:28.12`	`EmleyMoor`	`[TK]D-Fender: I've so far got precisely nowhere with any of it`
`19:28.21`	`rene1`	`i tried on symbol pocket pc machines`
`19:28.22`	`EmleyMoor`	`FWD do not have instructions`
`19:28.47`	`EmleyMoor`	`I need only to prove that this stuff is for real, for myself, first`
`19:29.00`	`novafirst`	`[TK]D-Fender: full log: http://pastebin.com/775889`
`19:29.04`	`[TK]D-Fender`	`EmleyMoor: http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+FWD`
`19:29.08`	`rene1`	`EmleyMoor: try searching in goggle first`
`19:29.20`	`fgifgi123`	`thanks`
`19:29.27`	`rene1`	`EmleyMor: http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76`
`19:29.42`	`*** join/#asterisk backblue (n=moo@87-196-105-168.net.novis.pt)`
`19:30.19`	`GlobbeTotssz`	`0k thanks for your help`
`19:30.27`	`GlobbeTotssz`	`i am going to try it now`
`19:31.18`	`EmleyMoor`	`renel: Can you translate that directly into moziax for me first? If I can get that much to work, I will believe that this is for real`
`19:31.36`	`file`	`for real? interesting`
`19:32.10`	`yosyp`	`so who can i call online? :P`
`19:33.05`	`file`	`yosyp: yourself!`
`19:34.00`	`yosyp`	`file: can't i call online to other PBXs?`
`19:34.01`	`*** join/#asterisk adorah (n=Administ@87.68.146.95.cable.012.net.il)`
`19:34.06`	`file`	`sure`
`19:34.20`	`yosyp`	`so don't you guys have one setup for this channel for people to call in?`
`19:34.22`	`EmleyMoor`	`The advice I am getting here seems to be "do it through asterisk" - which is fair enough`
`19:34.37`	`file`	`yosyp: not particularly, no`
`19:34.50`	`yosyp`	`file: do you know any PBXs that I can call in??`
`19:35.09`	`rene1`	`EmleyMoor: we are all real people here and many of us make a living out of this, you have the information, if you are serious about learning VoIP or anything computer related you have to read there is no other way`
`19:35.15`	`kotique[male]`	`<PROTECTED>`
`19:35.21`	`file`	`off the top of my head? not really`
`19:35.32`	`yosyp`	`think file think :)`
`19:35.38`	`yosyp`	`and it's...`
`19:35.39`	`file`	`SIP/333@conf.vonality.net`
`19:35.40`	`file`	`there!`
`19:35.43`	`yosyp`	`/quit`
`19:36.11`	`rene1`	`but anyway`
`19:36.16`	`rene1`	`host is iax2.fwdnet.net`
`19:36.24`	`yosyp`	`No such extension 'SIP/333' in context 'conf.vonality.net'`
`19:36.51`	`file`	`that was a dial line.`
`19:37.03`	`rene1`	`user is your FWD user and pass your FWD pass. apparently FWD supports ulaw only so select that codec`
`19:37.21`	`*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)`
`19:37.50`	`EmleyMoor`	`renel: Responds "not registered"`
`19:38.09`	`*** join/#asterisk x3me (n=x3me@201.11.199.111)`
`19:38.10`	`x3me`	`hi`
`19:38.18`	`*** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk)`
`19:38.31`	`x3me`	`anyone has using an MD3200 modem to FXO port ?`
`19:38.41`	`runa`	`hey :) I have a grandstream 488 and I want to make calls thru the FXO port. I have "exten => 113,1,Dial(SIP/pstnFXO/133,60)" but when I dial 113 and I hear a tone where I can dial (using the pstn line). but this wasn't what I wanted to do (I wanted asterisk to dial 113 on the line). Any hints?`
`19:38.42`	`x3me`	`my zttool only reports that my decide is with a RED ALERT`
`19:38.44`	`x3me`	`why?`
`19:39.06`	`EmleyMoor`	`I am on the verge of calling it a day and trying again tomorrow`
`19:39.22`	`robin_sz`	`seems best`
`19:39.33`	`EmleyMoor`	`I really don't understand why this doesn't work - it certainly should!`
`19:39.51`	`robin_sz`	`then, life would be too easy`
`19:39.57`	`file`	`when it all amounts to nothing in the end... I won't worry my life away`
`19:40.01`	`EmleyMoor`	`Part of the problem seems to be that when I read the "getting started", there's a lot of "noise" in my conf files`
`19:40.25`	`rene1`	`conf files?`
`19:40.47`	`robin_sz`	`yeah ... theres a soltion t that ...`
`19:41.25`	`robin_sz`	`mainly involving dumping all the lines begining with #`
`19:41.28`	`ghenry`	`going to be trying astlinux on net4801 soon ;-)`
`19:41.34`	`*** join/#asterisk crlshn (i=kvirc@operaciones3.globalnet.hn)`
`19:41.59`	`EmleyMoor`	`robin_sz: There's still a lot of "noise" in them even then`
`19:42.13`	`rene1`	`if you are connecting moziax to FWD you dont need no asterisk and no conf files`
`19:42.21`	`robin_sz`	`not so much, dump the demo stuff and its a lot clearer`
`19:42.49`	`EmleyMoor`	`renel: moziax will not connect to FWD for me - leading me into even greater annoyance`
`19:42.58`	`rene1`	`EmleyMoor: i dont use make config and then just create the files i need`
`19:43.12`	`rene1`	`that will require you some time learning what file does what`
`19:43.13`	`EmleyMoor`	`I don't use make anything with asterisk`
`19:43.37`	`blitzrage`	`lies`
`19:43.55`	`file`	`blitzrage: hush you! I know where you live!!!`
`19:44.10`	`EmleyMoor`	`I'm not trying to do this quickly - I'm just trying to get any progress at all - I think I started too late in the day`
`19:44.19`	`blitzrage`	`file: another lie`
`19:44.31`	`EmleyMoor`	`(mind you, if I'd been able to get my headset when I first tried, I'd have had more time`
`19:44.56`	`file`	`blitzrage: you live... in Toronto!`
`19:45.12`	`blitzrage`	`that's not very granular :)`
`19:46.42`	`novafirst`	`[TK]D-Fender: please take a lookt full log: http://pastebin.com/775889`
`19:51.53`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`19:52.43`	`*** join/#asterisk Slackuser_ (n=aaaa@200.194.102.234)`
`19:53.00`	`*** join/#asterisk Makenshi (n=makenshi@2001:630:1c0:2006:20c:29ff:fe7f:ba41)`
`19:57.28`	`ncrucial`	`quit`
`20:01.33`	`*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)`
`20:01.57`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)`
`20:12.07`	`*** join/#asterisk designdream (n=designdr@rrcs-71-40-48-202.sw.biz.rr.com)`
`20:13.02`	`designdream`	`alright.. so i bought the poly com 501's as recommended and i have yet to get ahold of the files needed to remotely provision the phones. i have tried manually making the xml .config files and have received errors. anyone have experience with polycom?`
`20:15.38`	`designdream`	`<PROTECTED>`
`20:15.47`	`Strom_C`	`.`
`20:15.48`	`Strom_C`	`..`
`20:15.54`	`Strom_C`	`nub.so`
`20:15.54`	`designdream`	`hahah`
`20:16.03`	`Strom_C`	`silly.conf`
`20:16.04`	`*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)`
`20:16.17`	`designdream`	`i was in wrong gnu screen split..`
`20:16.23`	`Strom_C`	`hehe i know`
`20:16.23`	`[hC]`	`Hey, any of you guys use the plantronics wireless headset (cs50/hl10 combo) with a polycom phone?`
`20:16.35`	`Dr-Linux\|work`	`chmod 777 silly.conf`
`20:17.17`	`designdream`	`find / -name \silly\ \| xargs rm -f`
`20:18.03`	`tzafrir_laptop`	`chmod 666 silly.conf # maybe`
`20:18.05`	`designdream`	`who was it in here who told me to get the polycom 501's from telephonydepot`
`20:18.37`	`tzafrir_laptop`	`chmod 644 silly.conf # probably better. or maybe a chown`
`20:19.33`	`[TK]D-Fender`	`designdream: You're expected to get them with the SIP firmware from your reseller`
`20:19.46`	`[TK]D-Fender`	`designdream: And that'd be me I'm sure ;)`
`20:20.37`	`designdream`	`[TK]D-Fender: it didnt come with anything... and i read a couple tutorials, next thing i knew i had 4+6+8 reset one.. and now it just loops in booting`
`20:20.40`	`tzafrir_laptop`	`designdream, 'locate' is faster.`
`20:21.30`	`designdream`	`[TK]D-Fender: did i remove the sip firware off when i did that?`
`20:22.00`	`fgifgi123`	`does skinnt protocal work over a wan?`
`20:22.04`	`[TK]D-Fender`	`designdream: If you misprovisioned it you need to rebuild a proper config for it to override whats stored local to the phone or get to a point to format off that bad config.`
`20:22.08`	`fgifgi123`	`skinny even`
`20:22.53`	`designdream`	`[TK]D-Fender: well thats what i did... i have 9 other good phones`
`20:23.54`	`[hC]`	`[TK]D-Fender: you dont have any plantronics handset lifters on your polycom phones, do you?`
`20:23.59`	`[TK]D-Fender`	`designdream: Well then its still picking up its config AGAIN from the server... you need to stop it for the local settings to stick after a factory reset`
`20:24.24`	`Strom_C`	`[hC]: why do you need a handset lifter when you can just press the "headset" button on the phone?`
`20:24.28`	`[TK]D-Fender`	`[hC]: Used to. I switched them to M10 straight amps a while back though...`
`20:24.28`	`designdream`	`[TK]D-Fender: i reset the flash.. and shutdown the ftp server`
`20:24.44`	`[TK]D-Fender`	`Strom_C: For use with wireless headsets, its required`
`20:24.50`	`Strom_C`	`what?!`
`20:24.56`	`Strom_C`	`that seems......kludgy.`
`20:25.05`	`[TK]D-Fender`	`designdream: Not sure... I'd try to rebuild the config and reprovision properly.`
`20:25.15`	`[hC]`	`[TK]D-Fender: how on earth did you get the headset to stay on the phone? mine ALWAYS screws up and drops down a bit, making it impossible to hang up the receiver.`
`20:25.25`	`coppice`	`call centre phones always seem kludgy`
`20:25.29`	`[hC]`	`Strom_C: Its because you answer the phone by pushing a button on your ear. no interaction with the phone.`
`20:25.34`	`[TK]D-Fender`	`Strom_C: Extremely, and none to fucntional. no remote xfer or anything without sitting in front of it. All you can do is answer and hangup.`
`20:26.09`	`GlobbeTotssz`	`DFender,,-- i did this (For SIP : Forward 5060-5070, 10000-20000 UDP to *. in sip.conf under [general] put "externip=[yourip]", "localnet=[yourlocalsubnet]/[netmask]", and "nat=yes".) I cannot register my phone.. But it registers if a set it to have no password`
`20:28.26`	`GlobbeTotssz`	`I used my local info for the extrernal ip and localnet settings`
`20:28.26`	`designdream`	`[TK]D-Fender: have any sample configs laying around?`
`20:28.26`	`[hC]`	`[TK]D-Fender: it seems that because the handset has nothing on the bottom to 'hold it' there, like a cisco does, that i need to add something to the phone to get this to work. It lifts up, then slides down a bit, so it wont hang up properly.`
`20:28.26`	`designdream`	`[TK]D-Fender: for some reason i feel that the instructions i followed were wrong`
`20:28.33`	`robin_sz`	`on my windows box, Firefox regularly checks for updates and updates itself ... how do I persuade firfox on my debian box to do the same thing?`
`20:29.44`	`robin_sz`	`oops, ww`
`20:37.26`	`Mandrak3`	`bye`
`20:37.45`	`*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)`
`20:39.01`	`designdream`	`i plugged in a brand new, unconfigured polycom 501... if i change the server setting to my ftp's ip address.. and there are no files in the folder involved in provisioning and configurating.. what will happen?`
`20:39.27`	`designdream`	`it boots up fine if i dont touch any settings..`
`20:41.52`	`syzygyBSD`	`it will burst into flames`
`20:43.12`	`designdream`	`omg! like a dell or mac battery!`
`20:43.44`	`syzygyBSD`	`blame the battery manufacturer, Sony`
`20:44.31`	`syzygyBSD`	`so on a more serious note, does SS7 require HDLC?`
`20:44.57`	`eKo1`	`no`
`20:45.17`	`Strom_C`	`on a less serious note, WHEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEE`
`20:45.33`	`*** join/#asterisk netstatic (i=netstati@neptune.negativeblue.com)`
`20:45.39`	`designdream`	`i am asking question`
`20:45.46`	`Cresl1n`	`syzygyBSD: yep`
`20:45.47`	`netstatic`	`quick question - where do i send RMA's to?`
`20:45.47`	`Cresl1n`	`:-D`
`20:45.54`	`Cresl1n`	`why?`
`20:45.55`	`Strom_C`	`netstatic: for digium?`
`20:46.02`	`netstatic`	`Strom_C: yes`
`20:46.12`	`Strom_C`	`netstatic: support didnt give you the address?`
`20:46.16`	`eKo1`	`ss7 does its own link management`
`20:46.34`	`netstatic`	`Strom_C: i took over someone else's project - they are no longer with the company`
`20:46.38`	`syzygyBSD`	`oh, just setting up a server and going to be doing some ss7 work. It frezzes randomly on reboot with hdlc compiled in (but not on powerup)`
`20:46.52`	`syzygyBSD`	`just making sure it is required or if i could ditch it`
`20:47.00`	`Qwell`	`netstatic: http://www.digium.com/en/company/contact.php`
`20:47.00`	`Strom_C`	`netstatic: call digium support, confirm the rma, and they will give you all necessary info`
`20:47.14`	`Cresl1n`	`syzygyBSD: what do you mean by "hdlc compiled in"?`
`20:47.16`	`eKo1`	`but then again, i'm no ss7 expert`
`20:47.17`	`*** join/#asterisk arkonadev (n=arkonaj@65.203.186.131)`
`20:47.22`	`netstatic`	`Strom_C/qwell :thanks`
`20:48.02`	`arkonadev`	`how is it possible that my originate action throws a timeout exception but yet still completes the call`
`20:48.28`	`Cresl1n`	`syzygyBSD: you don't have to do any special compiling for ss7`
`20:48.36`	`Cresl1n`	`stock zaptel should work just fine`
`20:49.43`	`Un1x`	`anyone know whats this mean`
`20:49.43`	`Un1x`	`<PROTECTED>`
`20:49.44`	`Un1x`	`Aug 25 16:51:05 WARNING[3281]: chan_zap.c:1586 zt_set_hook: zt hook failed: Device or resource busy`
`20:49.44`	`Un1x`	`<PROTECTED>`
`20:51.00`	`arkonadev`	`rooms not to active today`
`20:51.16`	`*** part/#asterisk netstatic (i=netstati@neptune.negativeblue.com)`
`20:51.23`	`syzygyBSD`	`lol.. ok, I mean the HDLC data that you can compile into the kernel, it allows you to split off data/voice lines from the same T1`
`20:51.37`	`Cresl1n`	`ahh.... ok`
`20:51.45`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`20:51.46`	`Cresl1n`	`well, if you're just doing SS7 you don't have to do that`
`20:51.48`	`Cresl1n`	`fwiw`
`20:51.48`	`syzygyBSD`	`I have it working on another server`
`20:51.50`	`Cresl1n`	`:-)`
`20:52.02`	`syzygyBSD`	`thanks, what about for a CAMA trunk`
`20:52.20`	`Cresl1n`	`nah, you only need if you're running data lines`
`20:52.57`	`syzygyBSD`	`thanks, that is what I was thinking, just wanted to doublecheck`
`20:53.13`	`arkonadev`	`how is it possible that my originate action throws a timeout exception but yet still completes the call?`
`20:55.44`	`*** join/#asterisk mountainm2k (n=mountain@216.87.64.218)`
`20:57.00`	`syzygyBSD`	`arkonadev: because, originate waits for an answer before returning`
`20:57.20`	`mountainm2k`	`What exactly is the correct syntax for a multi-word Caller-ID name? SetCIDName(Some Company,a) didn't seem to work right...`
`20:57.34`	`syzygyBSD`	`mountainm2k: try with ""'s`
`20:57.38`	`Qwell`	`mountainm2k: Set(CALLERID(name)=Some Company)`
`20:57.49`	`syzygyBSD`	`or better yet what he said`
`20:58.03`	`mountainm2k`	`Qwell: Unfortunately ABE-A... :-(`
`20:58.09`	`mountainm2k`	`Qwell: Upgrading this weekend tho`
`20:59.42`	`mountainm2k`	`With "";s, still only getting the number when I call myself (out through the PRI and back)... Don't have a line with analog caller-ID handy at the moment`
`20:59.47`	`mountainm2k`	`perhapps my carrier is blocking it?`
`21:00.01`	`Qwell`	`mountainm2k: most do. cidname is looked up on the far end`
`21:00.12`	`Qwell`	`it only (if even that) travels through the same carrier`
`21:00.42`	`mountainm2k`	`They're sending "Blocked call" when I call my house... Perhapps if I send them anything, they're blocking it...`
`21:01.30`	`syzygyBSD`	`what does it say when you dont' send anything`
`21:01.42`	`syzygyBSD`	`I know our carrier only allows the number to be set`
`21:02.09`	`syzygyBSD`	`only for the next couple weeks though till we get on SS7 hopefully`
`21:02.22`	`mountainm2k`	`dunno, haven't tried it -- it's coming through with the number to my cell (but I think cell reads the ANI), and nothing ("Blocked call") on home analog`
`21:02.35`	`eKo1`	`syzygyBSD: are you using chan_ss7 by any chance?`
`21:02.39`	`mountainm2k`	`Hahah, no way we'll be on SS7 -- PRI is where it's at for now...`
`21:02.59`	`syzygyBSD`	`not yet, still setting up our development box, but I will be`
`21:03.06`	`Cresl1n`	`syzygyBSD: you should run libss7`
`21:03.09`	`Cresl1n`	`that's the new coolness`
`21:03.11`	`Cresl1n`	`:-)`
`21:03.13`	`syzygyBSD`	`lol`
`21:03.28`	`syzygyBSD`	`well, i haven't started to get into the programming so we will see`
`21:03.36`	`syzygyBSD`	`either of you 2 work for digium?`
`21:03.57`	`syzygyBSD`	`my boss was talking with someone over there about it`
`21:04.09`	`eKo1`	`I'm connected to my telco. through ss7. It bites.`
`21:04.09`	`syzygyBSD`	`ahhh... you might be the one`
`21:04.22`	`Cresl1n`	`eKo1: what's wrong?`
`21:04.32`	`Cresl1n`	`eKo1: just remember, libss7 is where it's at :-D`
`21:04.36`	`eKo1`	`Getting no audio on channels 48-62`
`21:04.38`	`syzygyBSD`	`lol`
`21:04.56`	`syzygyBSD`	`hmm.. isn't 48 normally a data channel?`
`21:05.35`	`eKo1`	`I'm using a signaling gateway that does PRI<->SS7 though.`
`21:05.58`	`eKo1`	`It bites.`
`21:06.32`	`eKo1`	`I will eventually dump it though in favor of chan_ss7.`
`21:06.47`	`eKo1`	`I will try libss7 if that doesn't work.`
`21:08.34`	`redondos`	`When using a VoIP provider through asterisk using a softphone, I see that packets are not routed through the asterisk server. Rather, the client establishes a direct connection with the VoIP provider.`
`21:08.56`	`redondos`	`In that case, is Asterisk the one who actually handles the control connection?`
`21:09.18`	`redondos`	`Then it tells the client how to directly contact the remote host, or what?`
`21:09.46`	`redondos`	`I don't think * will give every client the credentials to use the remote service, so I guess what I say makes sense. Now, where can I read more about this?`
`21:09.54`	`eKo1`	`ReD-MaN: Are you using SIP?`
`21:10.00`	`redondos`	`yeah`
`21:10.09`	`eKo1`	`That is called REINVITEing`
`21:10.19`	`blitzrage`	`does the 1.6.6 version of the Polycom firmware allow TFTP anymore?`
`21:10.24`	`redondos`	`Nice, you gave me the golden clue, eKo1.`
`21:10.25`	`redondos`	`Thanks.`
`21:10.26`	`*** join/#asterisk gmfm (i=gmfm@rtr.enterprisemtg.net)`
`21:10.45`	`GlobbeTotssz`	`[TK]D-Fender: i did this (For SIP : Forward 5060-5070, 10000-20000 UDP to *. in sip.conf under [general] i put externip=216.194.164.124, "localnet=192.168.3.0/255.255.255.0", and "nat=yes".) I cannot register my phone.. But it registers if a set it to have no password`
`21:12.01`	`*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)`
`21:12.16`	`*** join/#asterisk noahvvvv (n=kydtiglo@wireless-141.media.mit.edu)`
`21:12.18`	`charles___`	`does anyone here uses a opensource billing ?`
`21:13.18`	`*** join/#asterisk Defraz (n=t0tal@fw.centrisys.com)`
`21:14.05`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`21:14.54`	`syzygyBSD`	`that is a lot of _ after your name, let me guess charles__ was taken?`
`21:14.55`	`[TK]D-Fender`	`GlobbeTotssz: Oh, and "canreinvite=no" in [general] as well`
`21:14.58`	`eKo1`	`charles___: not I. Had to dev. my own.`
`21:15.34`	`syzygyBSD`	`ya, it is hard when you want it to integrate into your whole CRM system`
`21:15.43`	`noahvvvv`	`has anyone ever used NOQUELCH mode in order to record DTMF?`
`21:16.22`	`noahvvvv`	`I've found I need to set that variable in the dsp structure, but I can see that the called to ast_dsp_digitmode will block NOQUELCH.`
`21:16.56`	`noahvvvv`	`so I'm wondering if that's a bug, or a rarely-used feature....or if there is some way to config NOQUELCH digitmode in a .conf file.`
`21:18.50`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`21:18.51`	`arkonadev`	`anyone had issues with ehcoing when you originate a call to a zap channel?`
`21:19.38`	`*** join/#asterisk hads\|home (n=hads@mail.nice.net.nz)`
`21:19.59`	`arkonadev`	`also i get a timeout exception as well but the call still goes through`
`21:20.35`	`[TK]D-Fender`	`blitzrage: 1.6.6 is the SIP application, not the BootROM. You can use any BR you want (within minimum spec), and yeahits supports only more and moer as tie goes on.`
`21:20.38`	`redox-`	`GRiDMAN101`
`21:20.46`	`*** part/#asterisk redox- (n=redox-@CPE0080c8242420-CM000a73664ccc.cpe.net.cable.rogers.com)`
`21:20.59`	`eKo1`	`arkonadev: I can relate to your echo problem, but not your timeout function.`
`21:21.01`	`noahvvvv`	`arkona: echoing is covered in the FAQ on asterisk.com.`
`21:21.13`	`*** join/#asterisk AJaymn (n=boiwonde@71-36-133-35.mpls.qwest.net)`
`21:22.10`	`arkonadev`	`hmm any way to fix it`
`21:22.46`	`arkonadev`	`asterisk.com is some sports website`
`21:23.04`	`blitzrage`	`[TK]D-Fender: nevermind, I had to use the right arrow instead of the up/down`
`21:23.42`	`[TK]D-Fender`	`blitzrage: :p`
`21:24.00`	`blitzrage`	`how annoying`
`21:24.01`	`noahvvvv`	`bleh .org sorry`
`21:24.22`	`GlobbeTotssz`	`SIP/2.0 403 Forbidden (Bad auth)`
`21:24.22`	`GlobbeTotssz`	`Via: SIP/2.0/UDP 75.30.124.150:5060;branch=z9hG4bK663F261F59944B5C9A3D341DF1E561A3;received=75.30.124.150;rport=5060`
`21:24.45`	`GlobbeTotssz`	`<PROTECTED>`
`21:24.45`	`GlobbeTotssz`	`<GlobbeTotssz> Via: SIP/2.0/UDP 75.30.124.150:5060;branch=z9hG4bK663F261F59944B5C9A3D341DF1E561A3;received=75.30.124.150;rport=5060`
`21:25.10`	`GlobbeTotssz`	`but it registers if i setup the phone with out a password`
`21:25.11`	`[TK]D-Fender`	`GlobbeTotssz: Congrates... SIP works... your user & pass DON'T :)`
`21:25.30`	`robin_sz`	`no, its forbidden, bad paste ;)`
`21:25.42`	`*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)`
`21:27.08`	`robin_sz`	`what about a Perl plug-in for Xchat, that delivers the full output of the console straight to the IRC channel ... now that would be so handy for newbies wanting some assistance ...`
`21:28.03`	`arkonadev`	`how could a originate action still work if there was a response timeout exception?`
`21:30.26`	`eKo1`	`robin_sz: err, you can do that already`
`21:30.31`	`*** join/#asterisk intralanman (n=lanman@pool-71-253-195-19.nrflva.east.verizon.net)`
`21:30.51`	`tzafrir_laptop`	`robin_sz, "the full output of the console" can be emulated using tail -f /var/log/full.log`
`21:31.04`	`tzafrir_laptop`	`<PROTECTED>`
`21:31.13`	`eKo1`	`you can do that with /exec`
`21:35.51`	`syzygyBSD`	`so if i recompile the kernel I should probably recompile all my programs too right?`
`21:37.48`	`*** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net)`
`21:38.05`	`redondos`	`syzygyBSD: Nah.`
`21:38.29`	`syzygyBSD`	`what if I am removing the HDLC network driver?`
`21:38.31`	`eKo1`	`syzygyBSD: that made no sense`
`21:39.05`	`eKo1`	`removing a driver will not affect much of anything`
`21:39.15`	`eKo1`	`except those dependent on it`
`21:39.45`	`syzygyBSD`	`eKo1: how so... when asterisk is compiled it uses some elements from the kernel source, if I recompile it, it is possible that will affect asterisk`
`21:39.47`	`*** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`21:40.03`	`eKo1`	`syzygyBSD: try it and find out`
`21:40.10`	`rene1`	`does remote_form_for has an oncomplete handler`
`21:40.18`	`syzygyBSD`	`lol fine, maybe I will, on monday when this memtest is complete`
`21:40.29`	`syzygyBSD`	`16% in 1 hour`
`21:40.46`	`copantl`	`any body know what it this?: No D-channels available! Using Primary channel 16 as D-channel anyway!`
`21:40.52`	`shodan`	`why doesn't this work ?`
`21:40.56`	`shodan`	`exten => 107,1,Dial(Zap/1/555-0001)`
`21:41.00`	`shodan`	`exten => 107,2,Record(/tmp/myfile.wav:wav)`
`21:41.02`	`shodan`	`I use a sip client , I type 107 , it calls the phone number but it doesn't record ?`
`21:41.19`	`copantl`	`I got a te110p and i have this warning : No D-channels available! Using Primary channel 16 as D-channel anyway!`
`21:41.21`	`rene1`	`oops, sorry wrong channel`
`21:41.43`	`copantl`	`any clue?`
`21:41.54`	`rene1`	`copantl i am seeing the same issue in my box`
`21:42.05`	`copantl`	`ohh`
`21:42.08`	`rene1`	`reloading asterisk fixes it but thats the only clue i have`
`21:42.27`	`copantl`	`reloading?`
`21:42.50`	`eKo1`	`restarting is what rene1 meant`
`21:42.50`	`copantl`	`i did it , i still have it`
`21:42.53`	`eKo1`	`I think`
`21:42.56`	`copantl`	`yes`
`21:43.06`	`GlobbeTotssz`	`[TK]D-Fender: Is there anything that i should check jn myu setting to get this to work?`
`21:43.16`	`eKo1`	`copantl: have any alarms?`
`21:43.20`	`copantl`	`yes`
`21:43.22`	`copantl`	`a lot`
`21:43.25`	`copantl`	`yellow`
`21:43.27`	`copantl`	`in zttool`
`21:43.32`	`eKo1`	`well, that is your problem.`
`21:43.38`	`eKo1`	`eliminate the alarms`
`21:43.41`	`syzygyBSD`	`copantl: in your zaptel.conf, do you have a d channel specified`
`21:43.46`	`copantl`	`yes`
`21:43.59`	`copantl`	`dchan=16`
`21:44.17`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`21:44.41`	`syzygyBSD`	`well, I am gueesing like what they said, fix your alarms`
`21:45.05`	`copantl`	`how i can clean my alarms?`
`21:45.15`	`*** join/#asterisk si_alchemist (n=sialchem@jingle-fw0.cust.e-xpedient.com)`
`21:45.16`	`syzygyBSD`	`shodan: that doesnt' work because Dial doesn't return`
`21:45.43`	`syzygyBSD`	`at least, not like you think it will`
`21:46.13`	`eKo1`	`copantl: talk to your PRI provider`
`21:47.06`	`copantl`	`eKo1: what i need to tell them?`
`21:47.46`	`copantl`	`eKo1: they need to clean all alarms?`
`21:48.06`	`shodan`	`ah ok , the problem I have is I want to call and old answering machine and record it's message in a file, is there a way I can do that with my setup ?`
`21:48.42`	`eKo1`	`copantl: you need to work with them to try to figure out what the problem is.`
`21:49.22`	`copantl`	`eKo1: i got this warning: WARNING[2652]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 2`
`21:49.35`	`copantl`	`a lot on my screen`
`21:49.39`	`rene1`	`copantl: you could try reloading the zapata driver, shutdown asterisk`
`21:50.13`	`copantl`	`rene1: i did it, nothing happens`
`21:50.19`	`copantl`	`the same`
`21:50.22`	`rene1`	`ask the telco`
`21:50.33`	`rene1`	`did you reloaded the driver and then ran ztcfg ?`
`21:50.40`	`copantl`	`yes`
`21:50.43`	`copantl`	`i did it`
`21:50.53`	`rene1`	`then do as everyone is telling you`
`21:51.02`	`copantl`	`ok`
`21:51.11`	`copantl`	`thanx`
`21:51.20`	`redondos`	`How can I make asterisk avoid concurrent connections with a remote voip provider?`
`21:51.44`	`redondos`	`I want to make it so if someone is using the voip account, it gives a busy tone when another person tries dialing through it.`
`21:52.55`	`rene1`	`if SIP, then tru the call-limit parameter which takes an integer for the number of concurrent connections`
`21:53.28`	`rene1`	`a.k.a incominglimit`
`21:54.31`	`tzafrir_laptop`	`shodan, is zapata.conf configured to use echocan?`
`21:56.24`	`*** join/#asterisk vader-- (n=johndoe@204.183.88.101)`
`21:57.12`	`redondos`	`rene1: thanks.`
`21:58.09`	`shodan`	`tzafrir_home, yes`
`21:58.11`	`shodan`	`oops`
`21:59.23`	`rene1`	`np`
`22:00.01`	`shodan`	`I tried to replace the dial with Dial(Zap/1/555-0001,,GOTO:107^2) but that didn't work :\`
`22:03.37`	`De_Mon`	`http://www.asterisk.org/developers/releasecycle`
`22:03.41`	`bobnormal`	`do any popular voip providers let you make more than one call simultaneously?`
`22:03.54`	`De_Mon`	`hehehe 1.4 is due in July, the NEXT major release is due in January!`
`22:04.05`	`De_Mon`	`oh, thats .org nevermind`
`22:04.10`	`*** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net)`
`22:05.24`	`redondos`	`bobnormal: Broadvoice and Voxee do. But since I have an account that lets me call for free to a number of countries with broadvoice, I'mlimiting it in case they charge me for the duplicate connections.`
`22:06.43`	`*** join/#asterisk test34 (n=test34@unaffiliated/test34)`
`22:07.31`	`charles___`	`eKo1: around ?`
`22:07.46`	`shodan`	`hmm , I used Dial(Zap/1/555-0001,,G(maison-interne^107^2)) I get transfered , and it created a /tmp/myfile.wav but I can't hear anything (except a beep from the record() function) so it's probably just recording silence ?`
`22:08.24`	`*** join/#asterisk redox- (n=redox-@CPE0080c8242420-CM000a73664ccc.cpe.net.cable.rogers.com)`
`22:09.34`	`*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)`
`22:09.51`	`eKo1`	`charles___: ?`
`22:11.41`	`shodan`	`oh never mind it did work my message is in the .wav , this G() argument is strange , the documentation says it sends both parties to the destination , how can it send both my sip client and the zap/1 to a record() to the same file ! and why does it cuts communication between the sip client and zap/1 also I had to create a second priority also with a record() because the first record exited immediately`
`22:14.09`	`redondos`	`How can I select what codec each SIP client uses? Or a global default.`
`22:18.05`	`eKo1`	`redondos: read sip.conf`
`22:20.13`	`Qwell`	`shodan: 1) It's sending you to two different places, so yes, it "cuts communication". 2) It doesn't send them both to record the same file. If you use a non-unique filename, only one of the channels will be recorded. 3) You had to "create a second priority", because it sends both channels to different priorities, on purpose.`
`22:22.08`	`shodan`	`thanks ! that was very clear`
`22:29.13`	`*** join/#asterisk Bullseye_Network (n=info@72-166-37-114.dia.static.qwest.net)`
`22:29.24`	`redondos`	`eKo1: Ok, got it, thanks. What codec do you recommend using if wanting only free software?`
`22:31.09`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.97)`
`22:31.11`	`hads\|home`	`ulaw/alaw if you have the bandwidth`
`22:32.05`	`*** join/#asterisk warmbowski (n=warmbows@dsl231-055-233.sea1.dsl.speakeasy.net)`
`22:32.23`	`redondos`	`hads\|home: And if I don't?`
`22:32.25`	`redondos`	`Heh.`
`22:32.38`	`*** join/#asterisk santiago (i=santiago@debian/developer/santiago)`
`22:33.15`	`eKo1`	`I have it worse. I'm in CA.`
`22:35.12`	`redondos`	`Heh. :)`
`22:36.05`	`hads\|home`	`redondos: I don't know, try GSM, some people don't mind it, some people hate it.`
`22:37.30`	`redondos`	`if I have "allow=gsm" as the first entry, I get a 603-Declined when picking up the phone.`
`22:38.38`	`eKo1`	`Does the phone/ATA support GSM?`
`22:38.46`	`redondos`	`I don't know, but it seems not.`
`22:38.55`	`redondos`	`It's twinkle, a linux softphone written in QT.`
`22:39.05`	`eKo1`	`Never heard of it.`
`22:39.19`	`eKo1`	`Does it do IAX?`
`22:39.31`	`[hC]`	`any of you guys played with this voice changer thing on voip-info?`
`22:40.00`	`hads\|home`	`Twinkle appears to support G711 and GSM`
`22:40.13`	`redondos`	`eKo1: It does.`
`22:40.29`	`redondos`	`hads\|home: Oh, really? Then why would the 603?`
`22:41.00`	`hads\|home`	`I don't know, check the settings.`
`22:41.16`	`hads\|home`	`And it's a SIP client not an IAX one`
`22:42.03`	`eKo1`	`I knew it.`
`22:42.12`	`eKo1`	`redondos: you liar`
`22:42.46`	`redondos`	`Ouch.`
`22:42.52`	`redondos`	`My bad.`
`22:43.50`	`charles___`	`eKo1: wanna share your billing software :P ?`
`22:44.20`	`*** join/#asterisk areq (n=areq@pld-linux/areq)`
`22:44.20`	`eKo1`	`charles___: at a price`
`22:44.54`	`areq`	`hi`
`22:44.58`	`charles___`	`eKo1: pleasure`
`22:45.10`	`charles___`	`eKo1: and I can introduce you my sister, she is cool.`
`22:46.42`	`eKo1`	`charles___: err, no`
`22:47.02`	`Givemelove2k`	`guys`
`22:47.15`	`charles___`	`eKo1: I promess she is cute`
`22:47.15`	`Givemelove2k`	`anyone knows how to use LookupBlacklist?`
`22:47.50`	`Bullseye_Network`	`I dont remember which line Im supposed to edit before I compile in the Makefile does anybody know? So I can get backtrace info.`
`22:48.08`	`hads\|home`	`Bullseye_Network: 'show application LookupBlacklist' ?`
`22:48.36`	`Bullseye_Network`	`hads\|home, no mixmonitor is crashing system, bug 7700`
`22:48.50`	`eKo1`	`charles___: I said NO`
`22:48.58`	`hads\|home`	`erm, sorry, that was for Givemelove2k`
`22:49.44`	`Givemelove2k`	`Hads: thanks, but I've seen that we have to put (j) to the end of the function in v1.2`
`22:49.51`	`Givemelove2k`	`just wanted a confirmation`
`22:50.31`	`hads\|home`	`Why, that's only if you want priority jumping?`
`22:50.58`	`areq`	`I have strange problem with asterisk + Sangoma A101 + E1, when someone call me from ISDN phone with display`
`22:50.59`	`Givemelove2k`	`I do`
`22:51.01`	`areq`	`after asterisk answer form their display disappear two last digits (I have 100 DDI)`
`22:51.09`	`Givemelove2k`	`but for some reason, it doesn't works`
`22:51.38`	`hads\|home`	`Bullseye_Network: I don't know for 1.2, maybe make dont-optimize`
`22:52.31`	`areq`	`I use wanpipe-2.3.3-3`
`22:52.46`	`Bullseye_Network`	`hads\|home, ohh thanks`
`22:54.27`	`areq`	`Can anybody help me ?`
`22:56.12`	`*** join/#asterisk dieno (n=Dieno@124.29.194.150)`
`22:56.23`	`*** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`22:56.52`	`runa`	`Im logged into FWD (using iax), I can call 613, 612, etc but I cannot receive calls. I click on "call me" and nothing happens. If I "iax2 debug" I see some messages but they don't seem to be related to the call`
`22:56.54`	`*** join/#asterisk Waverly360 (n=mirc@209.12.249.243)`
`22:56.58`	`runa`	`I can paste those messages in pastebin, if you want`
`22:57.03`	`dieno`	`for FC 4 Kernel 4.6.11 which version will gona wrk properlyon it`
`22:57.42`	`Waverly360`	`I know this is a bit quick..but I have a scenario with a problem..was wondering if you guys could help me with it. I'm kinda stumped.`
`22:57.59`	`runa`	`Waverly360: ask your question, please`
`22:58.13`	`shodan`	`is there a way I can see what codec a particular channel is using ? I tried show channel mychannel but it didn't contain that information`
`22:58.46`	`Waverly360`	`I dial an extension and wait until it hits the voicemail. Once it starts recording my message, I hit 0 to cancel out and dial the operator.`
`22:59.11`	`Waverly360`	`well...it's creating a message0000.txt file in my voicemail directory, but with no wav file`
`22:59.36`	`Waverly360`	`so now my phone says I have a voicemail..but I can't delete it through the phone.`
`22:59.42`	`hads\|home`	`Waverly360: What version?`
`22:59.51`	`Waverly360`	`1.2.9 I think?`
`22:59.54`	`Waverly360`	`lemme check`
`23:00.09`	`hads\|home`	`I believe there was a bug like that fixed at some stage.`
`23:00.23`	`Waverly360`	`oh?`
`23:00.27`	`Waverly360`	`hmm`
`23:00.32`	`Waverly360`	`I've thought about upgrading to see`
`23:00.37`	`Waverly360`	`but I haven't found any info on it.`
`23:01.51`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`23:02.03`	`TripleFFFF`	`whats issue again on WMI not beign passed from asterisk to phones via SIP`
`23:02.03`	`*** join/#asterisk atapi2 (n=virgill4@c-69-180-119-156.hsd1.fl.comcast.net)`
`23:02.22`	`areq`	`how I may send by PRI which DDI receive connection ?`
`23:02.58`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.127)`
`23:03.05`	`TripleFFFF`	`or not`
`23:04.03`	`*** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no)`
`23:05.18`	`*** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`23:06.26`	`TripleFFFF`	`a`
`23:06.46`	`*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`23:06.52`	`*** join/#asterisk justinu\|laptop (n=Justin@c-24-130-145-153.hsd1.ca.comcast.net)`
`23:07.19`	`Nivex`	`a`
`23:07.21`	`Nivex`	`oops`
`23:09.18`	`*** join/#asterisk dhanes (i=dhanes@hubble.switched-it.com)`
`23:09.23`	`dhanes`	`allo`
`23:10.31`	`*** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net)`
`23:17.26`	`dieno`	`Asterisk 1.2.11 is diz a Stable Version`
`23:17.53`	`Aurs`	`yes`
`23:19.45`	`warmbowski`	`Has anyone here had issues on outgoing zap (in my case PRI) with a delay between the call being answered and the incoming audio connecting. The delay is about 1sec in my case. Just enought to wipe out a quick 'Hello'.`
`23:20.49`	`JunK-Y`	`just do an answer(1000) ?`
`23:20.55`	`*** join/#asterisk vlt (n=daniel@dslb-088-073-211-136.pools.arcor-ip.net)`
`23:22.17`	`GlobbeTotssz`	`can anyone help me out.. my ip phone will not autenticate when using * behind NAT.. but it registers if i use no password`
`23:23.15`	`dhanes`	`Anyone elses have a problem with a TDM400P where the first two FXSs won't light up, but the 3rd and 4th FXO are lit up?`
`23:23.30`	`dhanes`	`Glob, using STUN?`
`23:23.32`	`Juggie`	`dhanes, did you apply power :)`
`23:23.50`	`dhanes`	`heh, pretty sure, would the FXOs not light up too if i didn't?`
`23:24.10`	`hads\|home`	`FXO's don't usually need power`
`23:24.14`	`dhanes`	`ahh`
`23:24.15`	`GlobbeTotssz`	`No no stun`
`23:24.37`	`dhanes`	`Glob, are your phones behind a NAT from *?`
`23:24.46`	`Juggie`	`isnt there a power adapter on the TFM400`
`23:24.51`	`dhanes`	`yes`
`23:24.52`	`Juggie`	`to power the dialtone on the FXS`
`23:24.52`	`dhanes`	`molex`
`23:24.56`	`Juggie`	`uhhuh`
`23:25.00`	`Juggie`	`and did you put something in it :)`
`23:25.07`	`dhanes`	`sigh, i guess i'll stop being lazy and go check it :)`
`23:25.56`	`areq`	`anybody use Sangoma A101?`
`23:26.16`	`vlt`	`Hello all. For sending and receiving faxes I used an analog fax connected to an ATA for several weeks now and think it's not stable enough ... Can asterisk receive faxes sent to it via SIP/G711 and save it as a file (or forward it to a hylafax machine or similar)?`
`23:27.13`	`GlobbeTotssz`	`Yes,, some phones are behind NAT.. some are not,, this is what i have done so far:: (For SIP : Forward 5060-5070, 10000-20000 UDP to *. in sip.conf under [general] i put externip=216.194.164.124, "localnet=192.168.3.0/255.255.255.0", and "nat=yes".)`
`23:28.23`	`GlobbeTotssz`	`canreinvite=no`
`23:28.59`	`dhanes`	`also foward 69`
`23:29.08`	`dhanes`	`forward`
`23:29.31`	`GlobbeTotssz`	`port 69 ?`
`23:29.42`	`dhanes`	`i think you need to use STUN on the phones that are nat'd before the * box`
`23:29.59`	`dhanes`	`OMFG`
`23:30.08`	`dhanes`	`guess what wasn't plugged in?`
`23:30.22`	`dhanes`	`yes, port 69`
`23:30.23`	`Dr-Linux`	`any new feature in app_queue application after 1.2.0 ?`
`23:30.57`	`dhanes`	`Glob, your ports plus 69 are what i have and everything works`
`23:31.35`	`Un1x`	`anyone ever use Asterisk on a laptop?`
`23:31.46`	`GlobbeTotssz`	`ok gonna try now`
`23:31.48`	`dhanes`	`jeebus...i've even gone so far as to reinstall everything to get those FXSs working ;)`
`23:31.49`	`GlobbeTotssz`	`thanks`
`23:31.51`	`dhanes`	`np`
`23:31.57`	`GlobbeTotssz`	`is that UDP ?`
`23:32.00`	`dhanes`	`yes`
`23:32.11`	`GlobbeTotssz`	`ok thanks`
`23:32.13`	`dhanes`	`don't forget STUN`
`23:32.52`	`dhanes`	`and on to the other box`
`23:32.58`	`Dr-Linux`	`Qwell:`
`23:33.16`	`dhanes`	`anyone have the recurring problem of 'Recordings' not being able to write/read the custom dir?`
`23:33.33`	`dhanes`	`i've uninstalled and deleted, rebooted and reinstalled module Recordings`
`23:33.37`	`dhanes`	`checked perms`
`23:34.10`	`Dr-Linux`	`dhanes: i have no problems, my recordings are going to different dir's`
`23:34.43`	`dhanes`	`of course you don't...i'm the idiot that forgets to plug in molex power on a TDM400p :)`
`23:35.07`	`Un1x`	`LOL!`
`23:35.10`	`Un1x`	`read the manual`
`23:35.13`	`Dr-Linux`	`dhanes: TDM400P is FXS?`
`23:35.15`	`Un1x`	`that comes with the card :P`
`23:35.23`	`dhanes`	`2 fxs 2 fxo`
`23:35.29`	`Un1x`	`TDM22B`
`23:35.33`	`Dr-Linux`	`dhanes: you don't need really`
`23:35.36`	`dhanes`	`forget what digium calls that particular config`
`23:36.00`	`Dr-Linux`	`dhanes: i have 4 TDM400P cards at different servers, i never pluged molex connector`
`23:36.09`	`Dr-Linux`	`and they work fine for me`
`23:36.13`	`dhanes`	`really?`
`23:36.19`	`Dr-Linux`	`dhanes: yes`
`23:36.23`	`dhanes`	`both my FXSs wouldn't light up`
`23:36.28`	`hads\|home`	`So you don't have FXS modules?`
`23:36.31`	`Dr-Linux`	`card gets power from bus`
`23:36.38`	`dhanes`	`until Juggie mentioned it`
`23:36.46`	`hads\|home`	`Depends on the board.`
`23:36.48`	`dhanes`	`what voltage are you runing?`
`23:36.58`	`Dr-Linux`	`i see`
`23:36.58`	`Un1x`	`hads\|home: he's more of a fool then we thought... unless you plan on not using the FXS' then you want the cable plugged in..`
`23:37.06`	`areq`	`where I can found source of my problem ?`
`23:37.08`	`Dr-Linux`	`dhanes: i'm sorry, i have all FXO's`
`23:37.12`	`dhanes`	`:)`
`23:37.13`	`dhanes`	`np`
`23:37.18`	`hads\|home`	`Un1x: Nice attitude.`
`23:37.23`	`Dr-Linux`	`dhanes: i never use FXS`
`23:37.35`	`Un1x`	`theres youre answer.`
`23:37.54`	`hads\|home`	`Un1x: There's no need to be so blatently rude.`
`23:37.56`	`dhanes`	`I think *n1x was just being sarcastic`
`23:38.03`	`Un1x`	`dhanes: you dont plan on using FXS portS?`
`23:38.10`	`dhanes`	`and since i flubbed so badly i deserve it :)`
`23:38.16`	`dhanes`	`yes, we are`
`23:38.19`	`Un1x`	`dhanes it wasn't to you`
`23:38.26`	`Un1x`	`ok then you need to plug in the molex cable`
`23:38.42`	`*** part/#asterisk warmbowski (n=warmbows@dsl231-055-233.sea1.dsl.speakeasy.net)`
`23:38.46`	`dhanes`	`yup, figured that out after Juggie mentioned it, did it, and now all are lit up`
`23:38.57`	`dhanes`	`swore I did it in the beginning`
`23:39.02`	`dhanes`	`meh`
`23:39.02`	`vlt`	`~thebook > vlt`
`23:39.19`	`vlt`	`~thebook`
`23:39.21`	`jbot`	`well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`23:39.33`	`Un1x`	`heh`
`23:39.39`	`Un1x`	`vlt: it's a good book :)`
`23:40.54`	`Dr-Linux`	`any new feature in app_queue application after 1.2.0 ?`
`23:40.56`	`vlt`	`Un1x: I know. Just needed the url again ;-)`
`23:41.09`	`Un1x`	`ahh i saved my copy :)`
`23:41.59`	`Dr-Linux`	`my boss sent me a hard copy of TFOT , but i never touch it :)`
`23:42.09`	`alrs`	`My PDF of TFOT is totally scrambled in chapter 7`
`23:42.20`	`*** join/#asterisk Givemelove2k (n=bob@208.57.229.162)`
`23:42.28`	`Givemelove2k`	`re`
`23:43.01`	`alrs`	`at least, it was under gpdf`
`23:43.08`	`alrs`	`seems fine now under evince`
`23:43.47`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`23:47.58`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.