IRC log for #asterisk on 20071001

`00:04.17`	`*** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`00:08.09`	`hrmphh`	`yeah zttest`
`00:08.11`	`hrmphh`	`sorry`
`00:08.18`	`hrmphh`	`call support?`
`00:08.27`	`hrmphh`	`they'll get back to you in about 2 weeks`
`00:08.37`	`hrmphh`	`and fwiw, there are some issues w/multicore systems`
`00:08.40`	`hrmphh`	`id go single if possible`
`00:08.43`	`hrmphh`	`race conditions`
`00:09.30`	`JunK-Y`	`go with dual, since this is future.`
`00:10.20`	`hrmphh`	`future?`
`00:10.29`	`asdx`	`im planning on getting a vps/dedicated server for running asterisk there, i will be able to use ip phones and connect to it right`
`00:11.10`	`[TK]D-Fender`	`asdx, vps sure, just don't bet on zaptel`
`00:11.45`	`JunK-Y`	`hrmphh: single core is certainly not the future`
`00:12.09`	`asdx`	`[TK]D-Fender: why?`
`00:13.44`	`[TK]D-Fender`	`asdx, Zaptel requires serious timing you can't usualy get in a virtualized envirnment.`
`00:24.43`	`*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)`
`00:24.53`	`asdx`	`[TK]D-Fender: i see`
`00:24.59`	`*** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose)`
`00:27.56`	`NightMonkey`	`Howdy. I'm checking out the Xorcom Astribank to use for a small office with Asterisk. Has anyone here used it with an OpenWRT-enhanced device w/ Asterisk?`
`00:30.05`	`NightMonkey`	`Since the Asus WL-500G Premium has two USB 2.0 ports, it seems like it might be a good candidate, but does it have enough CPU "oomph" to handle the Astribank?`
`00:36.27`	`*** join/#asterisk mltlnx (n=mltlnx@cpe-72-225-244-190.nyc.res.rr.com)`
`00:37.32`	`*** join/#asterisk gmustafa (n=gmustafa@221.132.115.3)`
`00:38.11`	`[TK]D-Fender`	`NightMonkey, And the reason for suck a wimpy solution is...?`
`00:40.42`	`NightMonkey`	`[TK]D-Fender: I was just curious. I like using OpenWRT for other purposes, but I suspect that the codec processing and such might be too much for typical OpenWRT devices.`
`00:41.32`	`[TK]D-Fender`	`NightMonkey, WRT is ok for a HANDFUL of non-codec translated calls with next to no services.`
`00:42.46`	`hmmhesays`	`this song is kicking my ass`
`00:43.23`	`*** join/#asterisk ljd (n=ljd@nelug/coreteam/luisjose)`
`00:44.32`	`NightMonkey`	`[TK]D-Fender: Thanks. Your statement confirms my suspicions. :)`
`00:45.38`	`[TK]D-Fender`	`hmmhesays, which?`
`00:46.24`	`*** join/#asterisk ManxPower (n=manxpowe@stirprop-s4-0-0-21.ndcr2.datasync.net)`
`00:56.37`	`hmmhesays`	`[TK]D-Fender: crazy train`
`00:57.31`	`[TK]D-Fender`	`hmmhesays, Not so hard....`
`00:58.00`	`JunK-Y`	`here's some pics of astricon 2007: http://picasaweb.google.com/cpatry/Astricon2007`
`00:58.11`	`[TK]D-Fender`	`hmmhesays, I dare say less than challenging`
`00:58.15`	`hmmhesays`	`timing doesn't come naturally to me, so I have to work with a metronome for the rhythm parts`
`00:58.26`	`*** join/#asterisk remmo (n=junk@203.32.47.250)`
`01:00.17`	`[TK]D-Fender`	`hmmhesays, tiem to work on a short picking depth & alternated palm muting :)`
`01:00.30`	`hmmhesays`	`of course`
`01:00.39`	`hmmhesays`	`I actually have some stylus picks that help with that`
`01:02.05`	`*** join/#asterisk Strom_M (n=strom@adsl-66-127-181-92.dsl.lsan03.pacbell.net)`
`01:03.04`	`*** join/#asterisk ^JimmyRidge^ (i=JimiRidg@EV-ESR-74-215-26-69.fuse.net)`
`01:06.02`	`[TK]D-Fender`	`hmmhesays, http://www.aocomputing.net/ct.mp3`
`01:09.52`	`[TK]D-Fender`	`hmmhesays, simple non-gimmicky way it to pull the pick in towards your knuckle so only the smallest amount of the tip is exposed past your thumb`
`01:10.19`	`[TK]D-Fender`	`hmmhesays, Also marke artifical harmonics fast & easy`
`01:10.32`	`*** join/#asterisk Entr4nced (n=IMG001@cpe-76-190-132-60.neo.res.rr.com)`
`01:11.22`	`*** join/#asterisk iamamoron (n=kerlnel2@202.137.121.84)`
`01:11.27`	`iamamoron`	`hi there`
`01:11.46`	`iamamoron`	`anybody have make dfm-5621s of dlink running on asterisk?`
`01:11.56`	`iamamoron`	`is there a driver for this v92 modem?`
`01:13.29`	`ManxPower`	`iamamoron: there are no modem drivers for Asterisk except for one specific chipset`
`01:13.38`	`ManxPower`	`and that chipset has not been made in several years`
`01:13.59`	`iamamoron`	`hmm`
`01:14.16`	`iamamoron`	`any ideas how this dfm-5621s be my fxo?`
`01:14.30`	`iamamoron`	`according to some blogs`
`01:14.40`	`iamamoron`	`v92 modem can be used as fxo port`
`01:16.16`	`*** join/#asterisk minkus (n=minkus@pool-71-182-38-220.clrkwv.east.verizon.net)`
`01:16.51`	`*** join/#asterisk Ard0gx (n=adiaz0@190.128.167.127)`
`01:17.41`	`ManxPower`	`I see you ignored what I said.`
`01:17.52`	`ectospasm`	`iamamoron: I'd like to see a link to that`
`01:18.26`	`[TK]D-Fender`	`iamamoron, funny I can't find any reference to that....`
`01:18.50`	`[TK]D-Fender`	`iamamoron, you should be ready to accept that it is not supported by Zaptel and thus worthless to *.`
`01:18.58`	`hmmhesays`	`I have problems with my thumb hitting the strings when I don't want it to`
`01:19.06`	`hmmhesays`	`the stylus picks helps that`
`01:19.15`	`hmmhesays`	`and they help with uniform strumming`
`01:20.20`	`hrmphh`	`can someone recommend fax software?`
`01:20.35`	`[TK]D-Fender`	`hrmphh, www.hylafax.org`
`01:20.36`	`hrmphh`	`we'd like to use one of our DIDs as a fax # and have thea bility to route out to diff. people with some sort of gui`
`01:20.43`	`hrmphh`	`and have faxes delivered via email`
`01:21.19`	`iamamoron`	`http://voip.weblogsinc.com/2005/07/14/use-a-v92-modem-as-an-fxo-card-on-asterisk/`
`01:21.21`	`Ard0gx`	`Hi guys, I'm working on creating a channel driver for asterisk 1.4.11 and found there is no more the "usecount" to update when requesting to allocate a channel, have somebody know I have to update the number of channels to allocated for my channel module in this asterisk version ?`
`01:21.23`	`hrmphh`	`does it have good email integration?`
`01:21.31`	`ManxPower`	`hrmphh: We support inbound faxing on all of our DIDs. The extension has a Set to specify the e-mail address.`
`01:21.32`	`hrmphh`	`http://www.hylafax.org/content/Email_to_Fax_Gateway`
`01:21.33`	`hrmphh`	`nm`
`01:21.48`	`hrmphh`	`manx; oh interesting, asterisk can detect if fax ?`
`01:21.55`	`hrmphh`	`you mean your voice/fax # is the same for each user?`
`01:22.09`	`iamamoron`	`that is v92 modem`
`01:22.11`	`ManxPower`	`hrmphh: fax and voice number the same for each user/extension`
`01:22.21`	`hrmphh`	`wow`
`01:22.23`	`hrmphh`	`thats pretty cool`
`01:22.34`	`hrmphh`	`so the user's phone obviously rings tho?`
`01:22.38`	`ManxPower`	`hrmphh: heck no,.`
`01:22.44`	`hrmphh`	`you have some sort of AA?`
`01:23.06`	`hrmphh`	`or does asterisk answer and play ring`
`01:23.10`	`hrmphh`	`while it determines if voice/fax`
`01:23.14`	`hrmphh`	`just trying to figure out how this works`
`01:23.47`	`ManxPower`	`A call comes in, first the script checks to see if there is an FAX_DEST set. if it is set, the script answers the call, starts to play a ringing tone and runs NVFaxDetect for 3 seconds to listen for a fax tone. if it hears a fax tone it runs RxFax, if it does not, it sends the call to the phone`
`01:23.52`	`[TK]D-Fender`	`iamamoron, And that refers to the one that IS. Yours is NOT.`
`01:24.10`	`iamamoron`	`hmmm`
`01:24.30`	`ManxPower`	`doing it this way DOES introduce an extra 3 second delay.`
`01:24.52`	`ManxPower`	`NVFaxDetect also detects voice, so if it hears voice it will immediatly send the call to the phone device.`
`01:24.56`	`hrmphh`	`what sets FAX_DEST? the fax machine calling?`
`01:25.06`	`ManxPower`	`The admin sets it.`
`01:25.23`	`hrmphh`	`sorry you set FAX_DEST on each extension? trying to figure out what that is used for`
`01:25.34`	`hrmphh`	`it indicates that the extension is dual use?`
`01:25.49`	`ManxPower`	`http://pastebin.ca/721205`
`01:25.58`	`ManxPower`	`FAX_DEST is used by my script.`
`01:26.09`	`hrmphh`	`ah`
`01:26.09`	`ManxPower`	`the pastebin is a copy of an actual extensions.conf extension`
`01:26.27`	`ManxPower`	`the macro-std-exten-v2 then does everything else, including fax tetection and call routing.`
`01:26.34`	`*** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il)`
`01:26.42`	`hrmphh`	`what is CFBL?`
`01:26.49`	`ManxPower`	`Here is a slightly out of date version of the macro`
`01:26.50`	`ManxPower`	`http://www.fnords.org/~eric/macro-std-exten-v2.inc`
`01:27.06`	`ManxPower`	`CFBL - Call Forward Busy Line. CFNA - Call Forward No Answer`
`01:27.14`	`Ard0gx`	`Hi guys, I'm working on creating a channel driver for asterisk 1.4.11 and found there is no more the "usecount" to update when requesting to allocate a channel, have somebody know I have to update the number of channels to allocated for my channel module in this asterisk version ?`
`01:27.16`	`hrmphh`	`got it`
`01:27.22`	`ManxPower`	`Ard0gx: try #asterisk-dev`
`01:27.35`	`hrmphh`	`do you need to assign separate extensions for a user to have "two lines"?`
`01:27.36`	`*** join/#asterisk exvito (n=exvito@89-180-33-79.net.novis.pt)`
`01:27.42`	`hrmphh`	`or is it possible to ring a second call on a busy extension`
`01:27.42`	`iamamoron`	`any ideas on how to use dfm-6521 as fxo?`
`01:28.02`	`[TK]D-Fender`	`iamamoron, you can't. GET OVER IT.`
`01:28.39`	`ManxPower`	`hrmphh: An extension is not a device.`
`01:28.42`	`[TK]D-Fender`	`iamamoron, Unless you're willing to code a zaptel driver for it from scratch yourself. and to that end...`
`01:28.44`	`[TK]D-Fender`	`~wglwat`
`01:28.45`	`jbot`	`well, wglwat is well, good luck with all that`
`01:28.45`	`iamamoron`	`what should i buy to have a call to pstn device?`
`01:28.48`	`ManxPower`	`Each phone has a sip.conf entry for each line`
`01:29.07`	`*** join/#asterisk \|omni\| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)`
`01:29.25`	`ManxPower`	`that way we can totally control call routing for what lines calls come in on. We use the MAC address of the device with -a, -b, -c, etc appended to it depending on the line appearance`
`01:29.36`	`[TK]D-Fender`	`iamamoron, pick up an X100P used on ebay, or a Linksys SPA-3000 or SPA-3102, Digium TMD400P, or Sangoma A200`
`01:29.58`	`ManxPower`	`[TK]D-Fender: at least he has Truth In Nick NAme`
`01:30.21`	`[TK]D-Fender`	`hrmphh, ManxPower's scripts will cause your brain cells to spontaneously explode....`
`01:30.33`	`ManxPower`	`[TK]D-Fender: feel free to rewrite it 8-)`
`01:30.36`	`hrmphh`	`im liking his stuff`
`01:30.39`	`hrmphh`	`cool ideas`
`01:30.53`	`ManxPower`	`hrmphh: That script started out 5 years ago.`
`01:31.01`	`hrmphh`	`ok so you have a phone logged in to two diff sip confs`
`01:31.05`	`hrmphh`	`so it has multiple appearances`
`01:31.06`	`Ard0gx`	`Hi guys, any idea of how to update the usecount counter in asterisk 1.4.11 ?`
`01:31.16`	`hrmphh`	`dialing same extension can ring either`
`01:31.19`	`hrmphh`	`depending on status?`
`01:31.38`	`ManxPower`	`hrmphh: depending on the value of DIALSTATUS`
`01:31.43`	`hrmphh`	`got it`
`01:31.45`	`hrmphh`	`CONGESTION, etc.`
`01:31.54`	`ManxPower`	`you also have to make sure call waiting is disabled on the device, of course.`
`01:31.54`	`hrmphh`	`thats pretty neat`
`01:32.02`	`hrmphh`	`oh hmm`
`01:32.08`	`[TK]D-Fender`	`hrmphh, Basically he's coding something that the Polycom's do THEMSELVES if he'd only LET THEM :)`
`01:32.10`	`hrmphh`	`wonder if these polycom 330s have call waiting on by default`
`01:32.24`	`[TK]D-Fender`	`hrmphh, Depends how you configured them.`
`01:32.31`	`hrmphh`	`msotly default`
`01:32.38`	`hrmphh`	`didnt change anything pertaining to call waiting`
`01:32.44`	`[TK]D-Fender`	`hrmphh, mostly USELESS... description that is :p`
`01:33.03`	`ManxPower`	`If you do a reload in your browser you'll see the current macro-std-exten-v2.inc production file`
`01:33.21`	`ManxPower`	`polycom has call waiting on by default.`
`01:33.32`	`*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)`
`01:33.35`	`ManxPower`	`calls.per.linekey or something like that is the setting`
`01:33.52`	`hrmphh`	`sweet`
`01:33.54`	`hrmphh`	`its already working`
`01:33.59`	`hrmphh`	`multiple calls`
`01:34.24`	`hrmphh`	`wondering if thatd jive with the fax routing stuff`
`01:34.31`	`hrmphh`	`guess could answer call anyway`
`01:34.40`	`hrmphh`	`before letting the polycom figure out which line to send to`
`01:34.55`	`hrmphh`	`i really really really like these 330s`
`01:35.02`	`hrmphh`	`so much nicer than the pos sipura's we had before`
`01:35.18`	`ManxPower`	`hrmphh: You would have to install spandsp and rxfax and my postprocessing fax-to-email script as well`
`01:36.30`	`ManxPower`	`The biggest issue I've found with fax to email is that my users (real estate agents) frequently get 20 page faxes. The message size of those faxes exceeded the max allowed message size on our e-mail server.`
`01:37.19`	`*** part/#asterisk Ard0gx (n=adiaz0@190.128.167.127)`
`01:37.39`	`hrmphh`	`hmm`
`01:37.42`	`*** join/#asterisk Hoondie (n=andre@ppp121-44-55-116.lns10.syd7.internode.on.net)`
`01:37.43`	`hrmphh`	`guess you could tweak the pdf creation`
`01:37.46`	`hrmphh`	`so its smaller`
`01:37.55`	`Hoondie`	`hey people..`
`01:37.57`	`hrmphh`	`another option would be to write to an smb share`
`01:38.02`	`hrmphh`	`and just send the user a UNC path`
`01:38.03`	`hrmphh`	`thatd be neat`
`01:38.10`	`ManxPower`	`that would not work in our enviroment`
`01:38.20`	`hrmphh`	`Hello Dumb User: You have a fax waiting in "H:\Faxes" named Blah.pdf`
`01:38.21`	`ManxPower`	`as users need to access these faxes while not in the office`
`01:38.26`	`hrmphh`	`yeah`
`01:38.28`	`Hoondie`	`quick question, i get two way audio if an external sip phone calls my phone on my local network.. if my phone calls the external sip phone, i get no audio`
`01:38.33`	`hrmphh`	`could use public folders`
`01:38.37`	`hrmphh`	`if they use outlook`
`01:38.38`	`Hoondie`	`anyone know what the problem could be?`
`01:38.40`	`hrmphh`	`w/rpc over https`
`01:38.55`	`ManxPower`	`seems like a lot more work than just imcreasing the max message size`
`01:38.57`	`hrmphh`	`haha`
`01:38.58`	`hrmphh`	`true`
`01:39.01`	`hrmphh`	`just getting creative`
`01:39.30`	`ManxPower`	`we have over 400 users, 28 offices in 2 states with an IT department of 3 people plus 2 consultants. creative is not something we do.`
`01:39.38`	`ManxPower`	`Save time is what we care about.`
`01:39.41`	`hrmphh`	`yeah`
`01:39.47`	`hrmphh`	`so im really interested in getting this going`
`01:39.57`	`hrmphh`	`i need spandsp, rxfax, and the postproc script?`
`01:40.03`	`hrmphh`	`any chance youd share the script?`
`01:40.10`	`ManxPower`	`and NVFaxdetect (check the Wiki)`
`01:40.25`	`hrmphh`	`where is the wiki?`
`01:40.45`	`[TK]D-Fender`	`~wikis`
`01:40.46`	`jbot`	`methinks wikis is http://www.voip-info.org`
`01:41.06`	`Hoondie`	`is anyone able to help with my one way audio problem?`
`01:41.24`	`hrmphh`	`oh that wiki`
`01:41.39`	`ManxPower`	`only of you promise to not EXPECT support and promise to report bugs.`
`01:42.20`	`hrmphh`	`http://voip-info.org/tiki-index.php?page=Asterisk%20fax`
`01:42.22`	`[TK]D-Fender`	`Hoondie, read this, NOW :`
`01:42.25`	`[TK]D-Fender`	`~sipnat`
`01:42.25`	`jbot`	`[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`01:42.25`	`hrmphh`	`of course not`
`01:42.27`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^`
`01:42.33`	`hrmphh`	`if i wanted support, id ask you how much it cost me`
`01:42.40`	`hrmphh`	`if cost was too high, id say no :)`
`01:42.55`	`hrmphh`	`do you do any consulting work on the side?`
`01:45.00`	`ManxPower`	`If I can find out why I'm getting a 403 error when trying to view fax2email.pl.txt you will be able to see it.`
`01:45.18`	`ManxPower`	`hrmphh: I am a consultant. I refer to "us" and "we" when referring to my client.`
`01:45.22`	`hrmphh`	`ah ok`
`01:45.35`	`ManxPower`	`I don't usually accept new customers except under specific situations`
`01:45.47`	`hrmphh`	`hmm business must be good :)`
`01:45.59`	`*** part/#asterisk exvito (n=exvito@89-180-33-79.net.novis.pt)`
`01:46.38`	`ManxPower`	`hrmphh: My needs are modest, my customers are long term.`
`01:46.54`	`Hoondie`	`[TK]D-Fender: is there a hostname i can use in place of externip=222.222.222.222?`
`01:47.00`	`ManxPower`	`My two largest (and only two frequent customers) have been customers for 12 and 7 years respectivly`
`01:47.56`	`[TK]D-Fender`	`Hoondie, you'd need "externhost=mydomain.com", and "externrefresh=120" (seconds, or something else reasonable)`
`01:48.45`	`Hoondie`	`Thanks.. i'll check it out`
`02:08.44`	`*** join/#asterisk Hoondie (n=andre@ppp121-44-5-124.lns10.syd7.internode.on.net)`
`02:09.16`	`Hoondie`	`[TK]D-Fender: I have followed everything in that guide you gave me, i still get one way audio when calling from the handset on the local network`
`02:11.17`	`[TK]D-Fender`	`Hoondie, pastebin your sip.conf masking only passwords`
`02:11.19`	`[TK]D-Fender`	`~pb`
`02:11.20`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`02:11.21`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^`
`02:11.42`	`Hoondie`	`only the relivant parts? global and the two sip peers?`
`02:13.13`	`[TK]D-Fender`	`Hoondie, sure, make sure its complete`
`02:14.40`	`Hoondie`	`here's almost everything..`
`02:14.41`	`Hoondie`	`http://pastebin.ca/721227`
`02:16.08`	`[TK]D-Fender`	`Hoondie, register => 029043xxxx:xxxx@sip.internode.on.net/s <- this has to be AFTER everything else under [general]`
`02:17.02`	`Hoondie`	`would that be why it was working sometimes and not others when trying to call inbound?`
`02:17.08`	`Hoondie`	`it'd work sometimes and others it wouldn't`
`02:17.30`	`[TK]D-Fender`	`Hoondie, what hoave you forwarded to * from your router?`
`02:18.02`	`Hoondie`	`what you mean? what ports have i forwarded?`
`02:18.19`	`Hoondie`	`5060 - 5070 TCP and UDP.. and 10000-20000 UDP`
`02:18.54`	`[TK]D-Fender`	`Should be fine then. apply your changes abnd retest`
`02:20.11`	`Hoondie`	`still not working..`
`02:20.13`	`*** part/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`02:20.25`	`Hoondie`	`i called the external phone, they cann hear me, i can't hear them..`
`02:20.31`	`Hoondie`	`if they call me, it works fine`
`02:20.32`	`[TK]D-Fender`	`Hoondie, which is your outside phone?`
`02:20.36`	`Hoondie`	`104`
`02:20.59`	`Hoondie`	`andre is the phone here`
`02:21.04`	`[TK]D-Fender`	`Hoondie, did they forward anything?`
`02:21.13`	`Hoondie`	`i have set their phone as DMZ`
`02:21.39`	`[TK]D-Fender`	`Hoondie, undo that. they should not have any forwarding`
`02:21.47`	`[TK]D-Fender`	`Hoondie, And repastebin please.`
`02:23.16`	`ReDNeQ`	`can anyone here help with zaptel now loading proper module`
`02:27.05`	`Hoondie`	`new pastebin`
`02:27.06`	`Hoondie`	`http://pastebin.ca/721233`
`02:27.57`	`Hoondie`	`and have removed the DMZ and rebooted the external phone`
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`02:30.47`	`Hoondie`	`[TK]D-Fender: after removing the DMZ i don't get any audio either way no matter who calls`
`02:32.28`	`[TK]D-Fender`	`Hoondie, general] <- missing brace a paste typo?`
`02:33.32`	`Hoondie`	`yea, didn't grab the first character`
`02:34.17`	`[TK]D-Fender`	`Hoondie, Ok, so * should have ports forwarded, the remote side should not. Have them reset the remote phone`
`02:34.32`	`[TK]D-Fender`	`Hoondie, Also, what router models are used at each site, and what phone remote?`
`02:34.41`	`Hoondie`	`yea, that has been done..`
`02:34.53`	`Hoondie`	`using DD-WRT WRT-54G's`
`02:35.05`	`Hoondie`	`I got some bugetone at the remote end..`
`02:35.11`	`Hoondie`	`and a linksys phone here`
`02:35.44`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`02:36.49`	`[TK]D-Fender`	`Hoondie, Also typically neither phone should be told that it's behind NAT`
`02:36.55`	`*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)`
`02:36.58`	`[TK]D-Fender`	`Hoondie, (on the phone itself)`
`02:37.37`	`Hoondie`	`ok, i'll configure that..`
`02:37.41`	`[TK]D-Fender`	`Hoondie, Then have the remote phone do an echo test to isolate that its audio to/from the server is OK before involving your "inside" phone`
`02:37.56`	`riddlebox`	`what would cause a zap fxo channel to ring an extra time after the called party answers?`
`02:41.07`	`ReDNeQ`	`my tdm800p isnt loading with the normal modules`
`02:41.25`	`[TK]D-Fender`	`riddlebox, describe the call flow`
`02:41.41`	`[TK]D-Fender`	`riddlebox, better yet.. PASTEBIN it`
`02:42.11`	`riddlebox`	`I am`
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`02:42.56`	`riddlebox`	`[TK]D-Fender, http://pastebin.ca/721248`
`02:43.29`	`riddlebox`	`ohh sorry, wrong way, you want the outbound`
`02:43.56`	`[TK]D-Fender`	`riddlebox, I'd like CLI output actually....`
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`02:46.35`	`riddlebox`	`[TK]D-Fender, weird, now its working fine`
`02:47.03`	`Hoondie`	`[TK]D-Fender: done that, removed NAT from the external phone.. no audio either way.. tried to get them to do an echo test, they were not hearing anything at all`
`02:47.16`	`[TK]D-Fender`	`Hoondie, hrmmm..`
`02:47.30`	`[TK]D-Fender`	`Hoondie, Somethings whacked... go check your servers firewall config`
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`02:47.49`	`Hoondie`	`i turned the firewall off on the server`
`02:48.38`	`[TK]D-Fender`	`Hoondie, SOMETHING is off here....`
`02:49.17`	`Hoondie`	`yea, it's just tryying to find that something..`
`02:49.26`	`Hoondie`	`I'll try to DMZ my * box at this end..`
`02:50.15`	`Hoondie`	`brb`
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`03:09.04`	`ZX81`	`hi all, anyone know how to change the NPI in mISDN? The Digium b410p won't work with the New Zealand telephone system without it`
`03:10.51`	`[TK]D-Fender`	`ZX81, http://www.asteriskguru.com/tutorials/pri_zaptel.html`
`03:11.32`	`[TK]D-Fender`	`ZX81, I think this sample might have what you're looking for...`
`03:12.01`	`Strom_M`	`ZX81: you know, also, if you wait for roughly ten hours or so, you can call digium and get support`
`03:12.39`	`ZX81`	`heh yeah will try - at 2am :)`
`03:12.52`	`Strom_M`	`or do it in the morning your time`
`03:13.02`	`Strom_M`	`8 AM your time, they'll still be open`
`03:13.03`	`ZX81`	`[TK]D-Fender: mISDN though`
`03:13.09`	`ZX81`	`know how to do it in zap`
`03:13.21`	`ZX81`	`Strom_M: yeah`
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`03:35.19`	`nickzxcv`	`what does this error mean: chan_zap.c: Signalling requested on channel 1 is FXS Kewlstart but line is in FXO Kewlstart signalling`
`03:35.22`	`nickzxcv`	`?`
`03:35.49`	`nickzxcv`	`my channel 1 is an FXO`
`03:36.24`	`JT`	`so you use fxs kewlstart signalling`
`03:36.33`	`JT`	`don't you read the documentation?`
`03:37.24`	`jbroome`	`bah`
`03:38.03`	`[TK]D-Fender`	`humbug`
`03:38.32`	`JT`	`ZX81: yeah i certainly would not ever buy a B410P`
`03:38.37`	`ReDNeQ`	`is there a better way of making the zaptel load better without adding extra lines in the rc.local`
`03:39.21`	`nickzxcv`	`thats what I have configured http://www.schmalenberger.us/files/zapata.conf`
`03:40.13`	`JT`	`nickzxcv: that web site does not work`
`03:40.22`	`JT`	`try pastebin.ca`
`03:41.37`	`nickzxcv`	`try it now`
`03:42.21`	`[TK]D-Fender`	`nickzxcv, LINK please.....`
`03:42.40`	`nickzxcv`	`http://www.schmalenberger.us/files/zapata.conf`
`03:43.07`	`JT`	`nickzxcv: and zaptel.conf?`
`03:43.59`	`nickzxcv`	`http://www.schmalenberger.us/files/zaptel.conf`
`03:44.57`	`JT`	`nickzxcv: compared the two`
`03:45.05`	`JT`	`it's pretty clear what the problem is`
`03:45.08`	`JT`	`compare, even`
`03:45.17`	`JT`	`the error even states what the issue is`
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`03:48.08`	`nickzxcv`	`um, yes... I guess I thought ztcfg says what it is, not the signalling. sorry!`
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`03:52.25`	`DirtyD`	`Hello`
`03:52.51`	`*** join/#asterisk bmg505 (n=leon@196.209.181.201)`
`03:53.04`	`DirtyD`	`Is it possible to bridge two sip callers together who are both on hold?`
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`03:56.27`	`[TK]D-Fender`	`DirtyD, through AMI, yes`
`03:59.31`	`*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`04:02.24`	`DirtyD`	`can only be done through AMI?`
`04:03.31`	`[TK]D-Fender`	`DirtyD, Correct`
`04:04.06`	`[TK]D-Fender`	`DirtyD, Now the question becomes : What prompts the system to bridge these calls?`
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`04:05.07`	`DirtyD`	`a user decides to invite another user to "live chat", invited user accepts invitation.`
`04:05.27`	`carrar`	`Are they in a bar?`
`04:05.32`	`DirtyD`	`sorta.`
`04:05.36`	`carrar`	`Thats hot`
`04:05.41`	`[TK]D-Fender`	`DirtyD, What does this have to do with "on hold"?`
`04:05.48`	`hmmhesays`	`why can't you bridge a call with channel redirect?`
`04:06.06`	`[TK]D-Fender`	`hmmhesays, that is what I alluded to`
`04:06.18`	`DirtyD`	`I guess because channel redirect can only be done via AMI?`
`04:06.47`	`[TK]D-Fender`	`DirtyD, can you answer my last question.... I fail to see the implication of HOLD in this case.`
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`04:07.56`	`DirtyD`	`scratch the hold stuff. The user is at a menu prompt.`
`04:08.25`	`hmmhesays`	`DirtyD: no`
`04:08.33`	`hmmhesays`	`if you are using 1.4 you can use app_channelredirect`
`04:10.31`	`[TK]D-Fender`	`DirtyD, How does MeetMe NOT solve this need for you?`
`04:10.33`	`hmmhesays`	`ahh just watched first episode of prison break season 3`
`04:10.40`	`hmmhesays`	`good episode`
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`07:16.13`	`aiksa[LV]`	`hi everyone.`
`07:16.34`	`aiksa[LV]`	`have anyone seen this in asterisk? chan_zap.c:4950 zt_write: Unable to set linear mode on channel 3`
`07:17.46`	`aiksa[LV]`	`google only returns a list of webpages having code for chan_zap`
`07:22.36`	`sheppard`	`check the zap source?`
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`07:26.14`	`aiksa[LV]`	`sheppard - the only thing i could imagine that asterisk is trying to push slinear into the PRI`
`07:26.50`	`aiksa[LV]`	`but as far as i understand - it should be ulaw or alaw, right?`
`07:39.27`	`*** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no)`
`07:41.33`	`aiksa[LV]`	`so no any other suggestions?`
`07:42.06`	`aiksa[LV]`	`changed to only allow ulaw and alaw for end voip phones. hope this helps`
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`08:17.23`	`yidiyuehan`	`hi, can any one assist me on the B410p card configuration? I saw == Primary D-Channel on span 1 down WITH ASTERISK -VVVVVVVVVVVVVVR`
`08:18.14`	`yidiyuehan`	`I am using bristuff driver for B410P card.`
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`08:26.26`	`tzafrir`	`yidiyuehan, does it work?`
`08:26.45`	`tzafrir`	`what version?`
`08:30.10`	`yidiyuehan`	`hi tzafrir`
`08:30.14`	`yidiyuehan`	`no it doesn't work`
`08:30.21`	`yidiyuehan`	`I tried to call out but failed,`
`08:30.47`	`yidiyuehan`	`Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g built by root @ PT8 on a i686 running Linux on 2007-09-24 11:16:08 UTC`
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`08:33.36`	`yidiyuehan`	`I tried call in, there is no response in CLI`
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`08:36.32`	`aiksa[LV]`	`nope didnt help either`
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`08:36.46`	`aiksa[LV]`	`I really wouldnt like to report this to digium if this is know issue`
`08:37.48`	`aiksa[LV]`	`what could cause : chan_zap.c:4950 zt_write: Unable to set linear mode on channel 3`
`08:37.51`	`aiksa[LV]`	`?`
`08:38.34`	`aiksa[LV]`	`@ Asterisk 1.2.20`
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`08:40.25`	`citats`	`aiksa[LV]: what type of zap devices do you have?`
`08:42.22`	`yidiyuehan`	`can any one here help me on the ISDN 4 port card? I have checked everything is fine but just the light is red color`
`08:44.31`	`aiksa[LV]`	`1 port E1 card`
`08:44.43`	`aiksa[LV]`	`digium`
`08:45.26`	`aiksa[LV]`	`the one which uses wcte11xp , te110p i suppose`
`08:45.39`	`citats`	`aiksa[LV]: hmmm i was thinking you had 2 different cards and the channel numbers werent being assigned properly. are you having problems, or just getting the warning?`
`08:46.01`	`aiksa[LV]`	`citats: when these messages appear it cant dial out`
`08:46.29`	`aiksa[LV]`	`the end device (SIP phone, which initiated the call) has a terrible noise, and after a while`
`08:46.42`	`aiksa[LV]`	`i get these messages`
`08:46.43`	`aiksa[LV]`	`Oct 1 10:11:44 WARNING[30745]: chan_zap.c:7963 pri_fixup_principle: Call specified, but not found?`
`08:46.46`	`aiksa[LV]`	`Oct 1 10:11:44 WARNING[30745]: chan_zap.c:9047 pri_dchannel: Hangup on bad channel 0/3 on span 1`
`08:47.33`	`citats`	`aiksa[LV]: is this happening with lots of calls at once or just one?`
`08:47.36`	`aiksa[LV]`	`incomming calls work fine. this is only problem for outgoing`
`08:47.54`	`aiksa[LV]`	`citats: it happens for aprox 10% of all of the calls`
`08:48.41`	`aiksa[LV]`	`and it pattern looks much like a wave - it starts out of the blue, lasts for a period of time and then disapears for time being`
`08:48.55`	`citats`	`aiksa[LV]: i've never run into that specific issue, so you might want to contact digium support.`
`08:49.30`	`aiksa[LV]`	`ok. i already thought that this was a rather rare issue as google couldnt give any decent locationsd`
`08:49.44`	`citats`	`aiksa[LV]: i'd suspect it might be more related to activity on your system though. are you running an underpowered system?`
`08:50.09`	`aiksa[LV]`	`underpowered meaning what? PSU supplying less than system is using?`
`08:50.51`	`citats`	`aiksa[LV]: i mean underpowered processor wise. is your system busy handling other tasks when this happens`
`08:51.41`	`aiksa[LV]`	`i normally have 99% idle processor when this happens. and tons of free ram`
`08:51.44`	`*** part/#asterisk atomicd (n=atomicd@69.107.248.157)`
`08:51.45`	`aiksa[LV]`	`:)`
`08:52.20`	`citats`	`aiksa[LV]: but you have a decent cpu right? 99% idle on a 386 still isnt much :)`
`08:52.21`	`aiksa[LV]`	`the server is not doing nothing more than being a pbx. and this happens even if there is only one call going through`
`08:52.47`	`aiksa[LV]`	`citats: yes i have a decend proc. Xeon, 2.4 i suppose`
`08:52.55`	`citats`	`aiksa[LV]: i'd also check to see if your zaptel card is sharing interrupts with another device. look at /proc/interrupts`
`08:53.15`	`citats`	`aiksa[LV]: ok just making sure, your not drastically underpowered`
`08:53.28`	`aiksa[LV]`	`11: 453212109 XT-PIC-XT wcte11xp`
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`08:53.41`	`aiksa[LV]`	`means zaptel has its own irq right?`
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`08:54.21`	`citats`	`aiksa[LV]: yeah thats how it looks. i'd contact support, this might be an issue they've dealt with before`
`08:54.42`	`aiksa[LV]`	`ok.`
`08:54.58`	`citats`	`sorry i cant help more, just wanted to cover the basics`
`08:54.59`	`aiksa[LV]`	`better to submit a bug. or to write to a mailing list/forum?`
`08:55.25`	`aiksa[LV]`	`citats: i have pretty much of * experince. but i have stumbled upon this one for first time.`
`08:55.46`	`citats`	`aiksa[LV]: i'd contact digium support directly, either by phone or email`
`08:55.52`	`aiksa[LV]`	`citats: nevertheless many thanks for the help`
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`09:03.03`	`*** mode/#asterisk [+o Corydon76-dig] by ChanServ`
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`09:05.31`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`09:08.10`	`*** join/#asterisk Provo (n=at@85.233.236.59)`
`09:08.15`	`*** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net)`
`09:08.38`	`[T]ank`	`anyone here savy with hylafax?`
`09:08.54`	`[T]ank`	`i have it working with a brooktrout fax card for fax to email`
`09:09.00`	`[T]ank`	`but i want to send email to fax.`
`09:09.19`	`[T]ank`	`would I just use standard "AT" commands with that? or would iaxmodem be required?`
`09:12.02`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`09:13.58`	`ghatak`	`Hi, i have a small setup for asterisks for testing {PSTN<---->Asterisks<---->SIP<---->End User}. My question is that, can i simulate the PSTN part? I dont have a spare line to test with asterisks PSTN.....`
`09:14.35`	`[T]ank`	`what do you mean by simulate?`
`09:14.49`	`[T]ank`	`you just want calls to reach your pstn server?`
`09:14.51`	`Strom_M`	`ghatak: first off, "asterisk" is the name of the program, not "asterisks"`
`09:14.53`	`*** join/#asterisk stkn\|work (n=stkn\|wor@gentoo/developer/pdpc.active.stkn)`
`09:15.10`	`ghatak`	`Sorry, typooo :)`
`09:15.27`	`Strom_M`	`ghatak: secondly, to asterisk, a channel is a channel is a channel. it doesn't matter if it's a POTS line or a SIP phone or an ISDN PRI channel or what`
`09:16.17`	`ghatak`	`Strom_M: Ahh... i c, i can put another SIP server on other side..... to achieve same thing ?`
`09:16.29`	`Strom_M`	`yeah, or IAX, or whatever`
`09:17.05`	`ghatak`	`Ok, makes me feel much better now :)`
`09:17.07`	`ghatak`	`thanks`
`09:18.17`	`Strom_M`	`some of the details will be different, of course, but you'll still get the apparent result of "calls tandem through the asterisk box"`
`09:19.52`	`ghatak`	`basically i want to do some testing as my calls currently cut off after 20 seconds if not answered, that does not happen when call is not being routed via asterisks. just trying to figure out what is causing that.`
`09:20.14`	`Strom_M`	`gah`
`09:20.16`	`Strom_M`	`ASTERISK`
`09:20.22`	`Strom_M`	`there is no "s" on the end`
`09:20.41`	`ghatak`	`Asterisk, sorry again`
`09:20.55`	`Strom_M`	`now the calls are being cut off because your Dial() line is probably specifying a 20 second timeout`
`09:21.14`	`Strom_M`	`pastebin the relevant section of extensions.conf`
`09:21.19`	`ghatak`	`Strom_M: that seems to commented out in the config, extensions.conf`
`09:21.19`	`Strom_M`	`~pb`
`09:21.20`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`09:23.53`	`ghatak`	`Strom_M: http://pastebin.com/mce7a418 here`
`09:25.04`	`Strom_M`	`ok....why is it commented out?`
`09:25.17`	`Strom_M`	`and what is "SIP/ghatak"?`
`09:26.19`	`ThoMe`	`hello`
`09:27.38`	`*** part/#asterisk ghatak (n=ghatak@84-93-217-81.plus.net)`
`09:27.57`	`Strom_M`	`clearly, those were simple questions with inarticulably complex answers`
`09:40.45`	`*** join/#asterisk MicW (n=michael@dslb-088-074-136-089.pools.arcor-ip.net)`
`09:40.49`	`MicW`	`hi`
`09:41.04`	`MicW`	`how can i lookup callerid names using an external database or application?`
`09:42.10`	`keulin`	`MicW, you can use for example`
`09:42.17`	`keulin`	`HUD`
`09:42.25`	`keulin`	`it's running under windows`
`09:42.57`	`keulin`	`and is linked to asterisk in a way and outlook in an other way`
`09:43.02`	`Strom_M`	`keulin: HUD has a connection to the telco's CNAM database?`
`09:43.09`	`keulin`	`and work perfectly with x-lite`
`09:43.24`	`keulin`	`Strom_C, I think not`
`09:43.25`	`Strom_M`	`MicW: I assume you're asking about CNAM, right?`
`09:43.55`	`keulin`	`sorry, I've not seen that is talking about cnam`
`09:44.15`	`Strom_M`	`"callerid name" strongly suggests CNAM`
`09:44.46`	`keulin`	`right`
`09:45.21`	`keulin`	`how telco's cnam db is working ?`
`09:45.54`	`MicW`	`i run it under linux and i have different caller id sources. i can use 2 ways to implement it: a script that asks all sources or a script that writes all known ids in a database`
`09:45.59`	`keulin`	`in my country the CID is displayed by any soft or hardphone without specific sofwtare`
`09:46.02`	`MicW`	`(i'd prefer the first way)`
`09:47.56`	`*** join/#asterisk michael-i (n=michael-@Ldf84.l.pppool.de)`
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`09:53.08`	`*** join/#asterisk ussrback (n=MAX@81.95.160.147)`
`09:53.11`	`ussrback`	`Hi all`
`09:53.56`	`ussrback`	`How can i implement private chat in meetme? possibility to chat in private`
`09:56.04`	`*** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu)`
`09:56.16`	`ussrback`	`?`
`09:57.03`	`tcseke`	`Hello, could you please tell me how to compile asterisk without chan_misdn?`
`10:00.24`	`tzafrir`	`./configure --without-misdn ?`
`10:02.54`	`*** join/#asterisk duckz (n=duckz@81.180.83.75)`
`10:07.43`	`*** join/#asterisk shido6 (n=shido6@74-130-59-184.dhcp.insightbb.com)`
`10:08.01`	`tcseke`	`i don't have configure...`
`10:08.56`	`tcseke`	`in asterisk 1.2, i think there is no configure`
`10:09.32`	`tzafrir`	`first off, if misdn libs are not present on the system, it won't build`
`10:09.58`	`tzafrir`	`And then, a little bit of editing channels/Makefile will get rid of anything you don't need`
`10:11.01`	`tcseke`	`i get these errors: http://pastebin.com/d1e341dc3, how can i solve this?`
`10:13.26`	`*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)`
`10:14.51`	`*** join/#asterisk MrMister2 (n=mrmister@195-23-105-193.net.novis.pt)`
`10:15.58`	`MrMister2`	`Hi. I've just seen a Thomson SpeedTouch 2020 for sale at an great price. Anyone had any experience with them? good or bad?`
`10:17.23`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`10:17.41`	`MrMister2`	`I would need about 4 of them and would be used for normal use. i.e: make, receive and transfer calls. nothing too exotic :)`
`10:22.00`	`MrMister2`	`mmm... Guess everyone's still asleep`
`10:22.42`	`awk`	`hrm, flash operator panel, when the light is orange or yellow not sure exactly what`
`10:22.48`	`awk`	`what does that mean?`
`10:26.06`	`MicW`	`Strom_M: yes, i meant CNAM`
`10:26.07`	`MrMister2`	`awk: light orange where? on the extension or trunk?`
`10:26.23`	`MicW`	`seems that's easy to script with the build in scriptiong language`
`10:26.27`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`10:27.59`	`awk`	`MrMister2 exten`
`10:35.11`	`MrMister2`	`awk: have you checked if you have any agents configured? I don't use agents but I know that the led color changes based on agent state (says so on the CHANGES file)`
`10:36.21`	`awk`	`where would I find a description stating this means this that mean sthat, etc`
`10:36.40`	`MrMister2`	`If you set this option to one, the LED of the button corresponding to a logged on agent will change its color to led_color_agent (specified in op_style.cfg)`
`10:36.56`	`MrMister2`	`check the op_style.cfg and see what you have set there`
`10:37.18`	`*** join/#asterisk enex (n=roman@p3EE269D5.dip.t-dialin.net)`
`10:37.32`	`awk`	`thanks`
`10:37.35`	`MrMister2`	`the option = "change_led_agent"`
`10:42.48`	`MrMister2`	`Anyone had any experience with a Thomson SpeedTouch 2020 or 2030? good or bad?`
`11:12.08`	`*** join/#asterisk caio1982 (n=usuario@CAcert-br/caio1982)`
`11:18.48`	`*** join/#asterisk shinao1 (n=chatzill@217.20.242.52)`
`11:23.16`	`*** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net)`
`11:24.41`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`11:26.31`	`*** join/#asterisk zumbush (n=ztriver@81.25.80.131)`
`11:29.26`	`creativx`	`oh this is fun`
`11:32.05`	`*** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro)`
`11:33.26`	`zumbush`	`When i call *97 to check voicemail the VoiceMailMain app fails when there is only one msg. I get this ->`
`11:33.26`	`zumbush`	`-- <SIP/202-00b053f0> Playing 'vm-youhave' (language 'se')`
`11:33.26`	`zumbush`	`<PROTECTED>`
`11:33.26`	`zumbush`	`When i run a test with SayDigits(1) the swedish soundfile 1.gsm is played w/o problem.`
`11:33.26`	`zumbush`	`Any idea whats wrong?`
`11:34.15`	`zumbush`	`Using Asterisk SVN-branch-1.4-r84135`
`11:35.47`	`Zefk`	`hi! I dial in gsm network via SIP and Atheus VoiceBlue Enterprise (4 SIM cards). I have a call-limit=4 setting in sip.conf. Is it possible to inform somehow the agents when all SIMs are bussy by other calls? In console I receive: call to peer 'voiceblue' rejected due to usage limit of 4. Thx`
`11:35.54`	`*** join/#asterisk javb (n=javb@190.80.224.136)`
`11:35.59`	`javb`	`hi guys.`
`11:36.58`	`javb`	`im using ubuntu server, asterisk 1.2..., i wonder, how can make asterisk console (using ssh) to show in colors, as in red had it used to be ? and how can i make "verbose mode" by default higher than cero ?`
`11:37.27`	`gremzoid`	`lots of v's on the cli?`
`11:37.39`	`gremzoid`	`asterisk -rcvvvvvvvvv ?`
`11:39.31`	`*** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net)`
`11:42.54`	`[T]ank`	`anyone here done anything outbound with hylafax? I know this is not the appropriate place for this question, but figured some people here are using it.`
`11:43.10`	`[T]ank`	`having trouble on outbound dialing with hylafax. it is only sending 1 digit.`
`11:43.39`	`[T]ank`	`log files say that the full number is received from the sendfax command, but my asterisk gateway is not getting anything more than the first digit.`
`11:44.17`	`*** join/#asterisk orn (n=orn@85.197.193.24)`
`11:44.44`	`[T]ank`	`gremzoid: dont use all the v's that just increases the verbose by each v you add. use the command in the cli of "set verbose", ie "set verbose 3"`
`11:45.36`	`orn`	`Hi. I'm having odd 1way RTP problems; two SIP servers, asterisk is one of them. If asterisk receives a call from a SIP client on remote SIP server there is one way RTP. If the call originates from PSTN and comes through the same SIP server the RTP is fine.`
`11:45.46`	`orn`	`asterisk rtp debugging even shows that asterisk doesn't send any rtp`
`11:45.53`	`orn`	`but there's no indication of codec incompatability`
`11:49.06`	`*** join/#asterisk CaRb0n^ (n=Omer@202.133.69.117)`
`11:49.17`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`11:53.31`	`orn`	`http://www.arnarson.net/~orn/calldebug.txt`
`12:00.32`	`tzafrir`	`gremzoid, both -r and -c is kind of useless`
`12:08.33`	`*** join/#asterisk tc3driver-nii (n=huh@rrcs-67-52-113-254.west.biz.rr.com)`
`12:09.58`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`12:12.41`	`javb`	`im using ubuntu server, asterisk 1.2..., i wonder, how can make asterisk console (using ssh) to show in colors, as in red had it used to be ? and how can i make "verbose mode" by default higher than cero ?`
`12:15.54`	`rob0`	`1. Colors in the console are a factor of YOUR TERMINAL, not the server. Use a smarter terminal emulator. Any xterm should work fine. 2. Change the command which starts asterisk to include the desired number of -v's.`
`12:16.22`	`rob0`	`It's possible that the ubuntu /etc/termcap is at fault, too.`
`12:17.04`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)`
`12:20.35`	`*** join/#asterisk mltlnx (n=mltlnx@pool-96-224-1-190.nycmny.east.verizon.net)`
`12:25.19`	`tzafrir`	`javb, use recent debs buildserver.net , as they have applied the patch in http://bugs.digium.com/view.php?id=9048`
`12:33.05`	`*** join/#asterisk Aeudian (n=Aeudian@74.92.134.190)`
`12:37.19`	`*** join/#asterisk airjump (n=MZ@p508ADE02.dip.t-dialin.net)`
`12:37.33`	`*** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net)`
`12:38.16`	*** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.242)
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`12:44.59`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`12:45.03`	`Aeudian`	`Is there a way to diagnose CDR with MYSQL? i have "what i believe is setup" correctly but i see no data in my sql database from asterisk, is there a way to see if asterisk is sending cdr information like a debug to the mysql server`
`12:49.14`	`*** join/#asterisk lirakis (n=lirakis@65.200.191.253)`
`12:49.22`	`lirakis`	`morning`
`12:49.56`	`Aeudian`	`morning`
`12:55.52`	`codefreeze`	`Aeudian: First, look in the console messages when asterisk starts, and verify the mysql module loaded, and no error messages.`
`12:56.13`	`*** join/#asterisk blq (n=Bl@dslb-088-066-226-229.pools.arcor-ip.net)`
`12:57.33`	`*** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net)`
`13:03.53`	`*** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net)`
`13:04.21`	`marc7`	`is there any way to have the asterisk console output to /var/log/asterisk? the 'messages' file in that directory just shows me the bare minimum`
`13:04.35`	`*** join/#asterisk ussrback (n=MAX@81.95.160.147)`
`13:04.38`	`*** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`13:05.07`	`ussrback`	`Please, help me! app_meetme.so module is missing in my asterisk 1.4.11`
`13:05.17`	`ussrback`	`how can i fix this problem?`
`13:07.54`	`[TK]D-Fender`	`ussrback: If its not there it means you didn't have a zaptel timing source installed when compiling Asterisk`
`13:08.52`	`ussrback`	`ok. let me clearrafy this issue. first should i need to install zaptel and then asterisk?`
`13:09.48`	`*** join/#asterisk theHub (n=karl@firewall.cierant.com)`
`13:10.06`	`[TK]D-Fender`	`ussrback: Yes`
`13:10.22`	`ussrback`	`Thanks Ill try ;)`
`13:10.50`	`ussrback`	`anyway is it possible to make some sound streaming`
`13:10.54`	`ussrback`	`in background`
`13:11.00`	`ussrback`	`during conference ?`
`13:11.22`	`ussrback`	`for example some songs in background`
`13:11.35`	`[TK]D-Fender`	`ussrback: It does that when there is only 1 person in the conference`
`13:11.59`	`[TK]D-Fender`	`ussrback: I think you can set it so it also does that until the "host" arrives (admin)`
`13:12.07`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`13:12.35`	`ussrback`	`but if there is some users in conference?`
`13:12.50`	`[TK]D-Fender`	`ussrback: Worry about that once you get it WORKING...`
`13:13.02`	`ussrback`	`ok :)`
`13:13.10`	`ussrback`	`one more question`
`13:13.12`	`ussrback`	`may i ?`
`13:13.16`	`ussrback`	`ask u?`
`13:13.22`	`rob0`	`tzafrir: so that's an asterisk issue? Interesting. Shouldn't a daemonized process not make any assumptions about the terminal type, and just take the $TERM of the attaching client?`
`13:13.47`	`creativx`	`morning [TK]D-Fender`
`13:13.53`	`ussrback`	`is it possible to make private chat?`
`13:14.03`	`ussrback`	`for example in conference there is 10 users`
`13:14.19`	`ussrback`	`and user 1 wants to make private chat with user 2`
`13:14.32`	`rob0`	`I usually try to post either drivel or correct information here, sorry, that was neither.`
`13:14.34`	`lirakis`	`man...`
`13:14.43`	`lirakis`	`safari bookshelf is sick`
`13:15.23`	`lirakis`	`.. its kinda expensive.. but ... its like .. a dreamy wonderland of books ... so much information at your fingertips ... aughhh.... (drooling)`
`13:17.41`	`Zefk`	`hi! I dial in gsm network via SIP and Atheus VoiceBlue Enterprise (4 SIM cards). I have a call-limit=4 setting in sip.conf. Is it possible to inform somehow the agents when all SIMs are bussy by other calls? In console I receive: call to peer 'voiceblue' rejected due to usage limit of 4. Thx`
`13:19.04`	`[TK]D-Fender`	`ussrback: Yes, you can make dynamic confreences, and you would ahve to make some dialplan to allow this.`
`13:20.14`	`ussrback`	`do i need any modules to make dynamic conference or dileplan is enough?`
`13:20.43`	`[TK]D-Fender`	`ussrback: once you have meetme, the rest of your work is in the dialplan`
`13:22.06`	`ussrback`	`[TK]D-Fender: I have installed meetme, according to your instruction. Thanks a lot!`
`13:22.37`	`[TK]D-Fender`	`ussrback: You're welcome`
`13:22.40`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`13:26.00`	`ussrback`	`[TK]D-Fender: I do not want to bother you with my questions, but .... I made some dialplan for IVR + Meetme but it doesnt work, seems i've made some mistakes. Can u check it, plzz? http://sial.org/pbot/27818.`
`13:26.27`	`[TK]D-Fender`	`ussrback: exten => s,5.backgroung (/home/asterisk/chatprogram/option1) <- spelling`
`13:26.46`	`[TK]D-Fender`	`exten => conf,1,Meetme(0001) <- how do you dail an ALPHA name?`
`13:26.49`	`[TK]D-Fender`	`ial*`
`13:26.59`	`[TK]D-Fender`	`exten => conf,3,Hangup() <- where's step 2?`
`13:27.12`	`*** join/#asterisk iBuMp (i=ibump@cpe-66-68-37-190.austin.res.rr.com)`
`13:27.14`	`[TK]D-Fender`	`exten => 1,1,MeetMe (0004) <- pretty sure extra spacing = BAD`
`13:27.42`	`[TK]D-Fender`	`ussrback: And next time pastebin the CLI output of your failed call, not jsut your dialplan code. Let me see what its actaully EXECUTING`
`13:28.56`	`ussrback`	`I have read several documents and there are different methods how to create meetme conference, cause i messed`
`13:30.46`	`[TK]D-Fender`	`ussrback: "show application meetme", and read the meetme.conf sample config`
`13:31.12`	`De_Mon`	`[TK]D-Fender do you have a trunk install available? I've got a dialout problem using SIP that I wanna know if it happens in trunk too`
`13:31.26`	`[TK]D-Fender`	`De_Mon: nope.`
`13:31.36`	`[TK]D-Fender`	`De_Mon: pastebin it and lets take a look.`
`13:32.04`	`De_Mon`	`http://bugs.digium.com/view.php?id=8078`
`13:32.08`	`ussrback`	`[TK]D-Fender: I have made changes with config, but it still not working. I incuded debug message http://sial.org/pbot/27819.`
`13:33.01`	`De_Mon`	`thats from 1.4.11 btw`
`13:33.22`	`[TK]D-Fender`	`ussrback: you have no priority #2, and you have autofallthrough enabled. Your dialplan is very broken`
`13:33.37`	`ussrback`	`:(`
`13:33.40`	`[TK]D-Fender`	`De_Mon: Ok, this is clearly out of my league....`
`13:34.03`	`De_Mon`	`Yeah, mine too. heh`
`13:34.10`	`[TK]D-Fender`	`ussrback: I also hope this isn't * 1.4 because you are using deprecated timeout apps....`
`13:34.39`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`13:34.39`	`*** mode/#asterisk [+o anthm] by ChanServ`
`13:34.46`	`ussrback`	`yes this is 1.4 version`
`13:34.57`	`*** join/#asterisk deeperror (n=deeperro@adsl-75-46-148-11.dsl.sfldmi.sbcglobal.net)`
`13:35.11`	`De_Mon`	`[TK]D-Fender the short story is that I can't call certain numbers with fax detection on them, I get that unsupported SDP media error and the call fails to connect. Pretty serious problem for us`
`13:36.03`	`tzafrir`	`rob0, asterisk will only like a very short list of terminal names: "xterm", "vt100", "linux", and maybe one or two others`
`13:36.09`	`*** join/#asterisk bintut (n=bintut@cm44.gamma185.maxonline.com.sg)`
`13:36.20`	`[TK]D-Fender`	`ussrback: Go read upgrade.txt , the readme both included in your source download. In fact read ALL of the DOCS included, because 1.2's functions have taken over. Your code is no longer valid.`
`13:36.22`	`tzafrir`	`At the time of daemonizing`
`13:36.29`	`tzafrir`	`Quite sad, indeed`
`13:39.13`	`*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)`
`13:43.15`	`*** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-2204ec29780e4615)`
`13:45.44`	`*** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)`
`13:49.21`	`*** join/#asterisk ming_zym (n=ming_zym@124.254.56.33)`
`13:50.59`	`orn`	`Hi. I'm having odd 1way RTP problems; two SIP servers, asterisk is one of them. If asterisk receives a call from a SIP client on remote SIP server there is one way RTP. If the call originates from PSTN and comes through the same SIP server the RTP is fine. SIP and RTP debugging shows no RTP being sent from Asterisk. I don't see any indication of a codec incompatability. Output of Asterisk debug at: http://www.arnarson.net/~orn/calldebug.txt`
`13:51.21`	`*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)`
`13:51.26`	`dandre`	`hello,`
`13:51.52`	`dandre`	`is there any documentation about users.conf file anywhere?`
`13:53.41`	`*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)`
`13:53.55`	`zumbush`	`anyone know why Asterisk 1.4 writes the voicemail files with theese permissions set`
`13:53.56`	`zumbush`	`-------rw- 1 asterisk asterisk 272 30 sep 19.14 msg0000.txt`
`13:53.57`	`zumbush`	`-rwxr-xr-x 1 asterisk asterisk 92844 30 sep 19.14 msg0000.wav`
`13:53.58`	`zumbush`	`?`
`13:54.01`	`*** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)`
`13:54.42`	`zumbush`	`the txt gets all fu permission as far as i can tel`
`13:54.44`	`zumbush`	`tell`
`13:55.06`	`[TK]D-Fender`	`dandre: Yes... in the sample files.`
`13:57.46`	`*** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136)`
`13:58.28`	`*** join/#asterisk ManOfMilk (n=root@c-71-193-242-0.hsd1.or.comcast.net)`
`13:58.30`	`dandre`	`ok I have read the sample file for users.conf.`
`14:00.38`	`dandre`	`For instance, there is no clue on hasexten keyword`
`14:00.46`	`*** join/#asterisk waltj (n=walt@216.179.31.170)`
`14:00.54`	`[TK]D-Fender`	`dandre: If you can't find your answers in there and can't reverse engineer a GUI created config, your last bet is hoping its in the NEW BOOK.`
`14:01.28`	`[TK]D-Fender`	`dandre: hasexten I think determines whether or not its dial-able as an exten.`
`14:01.49`	`[TK]D-Fender`	`dandre: AKA creating an entry for a phone that can dial out, etc, but not one that can perhaps be CALLED.`
`14:01.59`	`[TK]D-Fender`	`dandre: users.conf = flaming pile of shit.`
`14:02.18`	`[TK]D-Fender`	`Oh... I forgot the opening tag....`
`14:02.34`	`creativx`	`gaash`
`14:02.35`	`[TK]D-Fender`	`<bile>`
`14:02.46`	`[TK]D-Fender`	`there`
`14:03.03`	`[TK]D-Fender`	`so yeah.... users.conf = horrific mistake....`
`14:03.09`	`[TK]D-Fender`	`</bile>`
`14:03.11`	`*** join/#asterisk ecto (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net)`
`14:03.37`	`*** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au)`
`14:04.22`	`*** join/#asterisk tsurko (n=tsurko@213.91.216.130)`
`14:04.48`	`Norm`	`anyone have any experience with a SIP->PSTN termination service?`
`14:05.00`	`[TK]D-Fender`	`Norm: plenty.`
`14:05.02`	`[TK]D-Fender`	`~ask`
`14:05.03`	`jbot`	`Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will.`
`14:05.05`	`*** join/#asterisk asdx (n=foo@adsl-136-52.click.com.py)`
`14:05.22`	`*** join/#asterisk jmesquita (n=jmesquit@189.31.65.15)`
`14:05.43`	`Norm`	`can you recommend one? i see plenty of references all over the place, but would like to try one (preferably with some sort of free trial) before plunking down the money`
`14:06.01`	`[TK]D-Fender`	`~itsp`
`14:06.01`	`jbot`	`An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower`
`14:08.11`	`*** join/#asterisk allforthelulz (i=lolsofco@nat/digium/x-9ecef2894305a46a)`
`14:08.22`	`Norm`	`thanks`
`14:10.16`	`*** join/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca)`
`14:12.04`	`*** join/#asterisk cpina (n=carles@77.226.21.170)`
`14:12.18`	`*** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net)`
`14:13.02`	`cpina`	`hi,`
`14:13.57`	`cpina`	`i need to call to some phone numbers (just to check) in parallel way. Using Dial in CLI or AMI doesn't allow me to call to more than one number, because tries to use the same channel (OSS/dsp)`
`14:14.09`	`cpina`	`how I can avoid it? (create new channels, redirect to some place, etc.)`
`14:14.48`	`*** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com)`
`14:15.38`	`*** join/#asterisk BadPacket (n=John@unaffiliated/badpacket)`
`14:16.31`	`dandre`	`ok [TK]D-Fender`
`14:17.17`	`*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)`
`14:17.18`	`*** mode/#asterisk [+o russellb] by ChanServ`
`14:20.24`	`dandre`	`what is the use of meetme.conf?`
`14:20.43`	`Aeudian`	`dandre: conference rooms`
`14:22.09`	`deeperror`	`anyone know of some good resources or case studies on asterisk hardware specs for high volume setups?`
`14:24.41`	`*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)`
`14:25.41`	`dandre`	`yes but meetme room are accessed from the dialplan with app_meetme and there is not much configuration available in the meetme.conf file. So I questionned about its real usability`
`14:29.23`	`*** join/#asterisk SplasPood (n=jwb@schizophrenia.paravolve.net)`
`14:29.37`	`*** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net)`
`14:29.50`	`*** join/#asterisk tsurko (n=tsurko@213.91.216.130)`
`14:31.10`	`[TK]D-Fender`	`dandre: Indeed due to dynamic conferences, meetme.conf is nearly pointless....`
`14:31.39`	`russellb`	`pretty much ...`
`14:31.53`	`russellb`	`just for backwards compatability, and for people that want to put meetme.conf in a database`
`14:32.22`	`russellb`	`but i believe it supports dynamic realtime now ...`
`14:32.27`	`russellb`	`so pretty much pointless, yes.`
`14:32.37`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`14:32.58`	`*** join/#asterisk coppice (n=chatzill@30.168.17.210.dyn.pacific.net.hk)`
`14:33.59`	`*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)`
`14:34.24`	`Dr-Linux`	`hi file`
`14:34.54`	`file`	`hello`
`14:35.17`	`Dr-Linux`	`lol`
`14:36.04`	`Dr-Linux`	`sorry to call you directly, actually if you remember you helped to setup iax2 trunk`
`14:36.33`	`Dr-Linux`	`actually, i wanna discuss something about jitter buffer, that how can i get some voice quality benifit out of it`
`14:36.52`	`*** join/#asterisk shinao1 (n=shinao1@217.20.242.52)`
`14:36.55`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`14:37.31`	`Dr-Linux`	`file: since i've low bandwidth at Pakistan side, so what should the option set to in iax.conf :`
`14:38.05`	`Dr-Linux`	`jitterbuffer= yes/no`
`14:38.06`	`Dr-Linux`	`maxjitterbuffer=n`
`14:38.07`	`Dr-Linux`	`maxexccessbuffer=100`
`14:38.40`	`Dr-Linux`	`file: everything is well described over here: http://bugs.digium.com/file_download.php?file_id=5250&type=bug`
`14:38.51`	`Dr-Linux`	`but still want to know your sugestions`
`14:39.24`	`cpina`	`i need to call to some phone numbers (just to check) in parallel way. Using Dial in CLI or AMI doesn't allow me to call to more than one number, because tries to use the same channel (OSS/dsp)`
`14:39.29`	`cpina`	`how i can create "dummy" channels?`
`14:39.31`	`cpina`	`is it possible?`
`14:40.23`	`[TK]D-Fender`	`cpina: dump a pile of call-files that dump into a mettme.`
`14:40.28`	`[TK]D-Fender`	`meetme*`
`14:40.51`	`volker__`	`bye guys :)`
`14:40.56`	`*** part/#asterisk volker__ (n=volker@proton.uhrig.eu.org)`
`14:41.15`	`cpina`	`ahá... i will try [TK]D-Fender...`
`14:44.09`	`cpina`	`[TK]D-Fender: Now I'm trying using Originate AMI command, what we should put in channel? (I will look further now...)`
`14:44.22`	`*** join/#asterisk jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)`
`14:44.51`	`[TK]D-Fender`	`cpina: depends how you want to dial. I tend to like using Chan_local for the outbound channel.`
`14:45.00`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`14:45.01`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`14:45.16`	`puzzled`	`hi`
`14:45.58`	`cpina`	`[TK]D-Fender: do you mean local/9999 , no?`
`14:46.02`	`cpina`	`Channel: local/9999`
`14:46.09`	`cpina`	`(I was using SIP/ext_number)`
`14:46.18`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`14:46.31`	`[TK]D-Fender`	`cpina: "Local/[exten]@[context]/n"`
`14:47.03`	`*** join/#asterisk akasa47 (i=0ca5bc82@gateway/web/cgi-irc/ircatwork.com/x-fe9f0c71027d4fc8)`
`14:47.04`	`[TK]D-Fender`	`cpina: And let your dialplan do the dialing out your PSTN connection (however it is that you get there)`
`14:47.05`	`cpina`	`[TK]D-Fender: ops thanks, I was only using SIP/... :-)`
`14:47.11`	`cpina`	`yeap...`
`14:47.59`	`akasa47`	`Can somebody help me with some audio problems?`
`14:49.42`	`akasa47`	`?`
`14:49.55`	`jbroome`	`can't hear you`
`14:50.04`	`[TK]D-Fender`	`:D`
`14:50.13`	`[TK]D-Fender`	`akasa47: ...`
`14:50.15`	`[TK]D-Fender`	`~ask`
`14:50.16`	`jbot`	`Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will.`
`14:50.17`	`[TK]D-Fender`	`^^^^^^^^^^^^^`
`14:50.38`	`akasa47`	`ok. I am newbie here`
`14:51.09`	`akasa47`	`I have meetme wiht 15 users and I have robotic voice. What I can set for improve the audio quality?`
`14:51.17`	`asdx`	`is there something i can show to normal people about asterisk, so they can understand it easily how it works`
`14:51.30`	`[TK]D-Fender`	`akasa47: What are you using for a timing source? And what kind of channels are these?`
`14:51.39`	`akasa47`	`ztdummy`
`14:51.47`	`akasa47`	`all sip channels`
`14:51.50`	`[TK]D-Fender`	`akasa47: Thats 1 "X".....`
`14:52.01`	`akasa47`	`connected with avaya pbx`
`14:52.02`	`[TK]D-Fender`	`akasa47: Any possibility of bandwidth issues?`
`14:52.08`	`[TK]D-Fender`	`akasa47: What codecs?`
`14:52.22`	`[TK]D-Fender`	`asdx: What is there to show?`
`14:52.34`	`cpina`	`[TK]D-Fender: thank you very much, using a "pile" to Meetme has worked :-)`
`14:52.34`	`akasa47`	`Not really. I have many Ip phone with no problems`
`14:52.39`	`akasa47`	`only ulaw`
`14:52.40`	`[TK]D-Fender`	`asdx: perhaps you could clarify what it is about * you want them to understand.`
`14:52.52`	`asdx`	`[TK]D-Fender: yeah`
`14:52.54`	`[TK]D-Fender`	`cpina: Glad to help.`
`14:53.02`	`Jameno123`	`For 100ms RTT whats the best jitter buffer settings?`
`14:53.33`	`[TK]D-Fender`	`akasa47: Where are they located relative to the sever, and what kind of bandwidth per site?`
`14:54.02`	`akasa47`	`I set meetme jitterbuffer to 32 and in zapata jitterbuffer to 32 and I have delays in audio`
`14:55.03`	`[TK]D-Fender`	`akasa47: Guess that could have issues as you would be spanning jitter buffer delay across all members of the meetme. I think you'd best leave JB to the phones themselves for this case...`
`14:55.47`	`[TK]D-Fender`	`akasa47: I can imagine that you are getting a "worst case" JB averaging effect because of the number of people involved.`
`14:56.05`	`[TK]D-Fender`	`akasa47: that is assuming I don't just blame ZTDUMMY outright.`
`14:56.20`	`[TK]D-Fender`	`akasa47: It really isn't a great timer...`
`14:59.16`	`akasa47`	`if I leave dafult buffer then it is worse`
`15:00.00`	`akasa47`	`I have asterisk and avaya in same network connected through same switch 100mb`
`15:00.30`	`akasa47`	`if I leave deafult buffer size then it is worse`
`15:00.40`	`[TK]D-Fender`	`akasa47: What is the default?`
`15:02.53`	`akasa47`	`I think was 8 at least was configured before with 8`
`15:03.09`	`*** join/#asterisk cadsii (n=mario@206-248-190-20.dsl.teksavvy.com)`
`15:03.15`	`[TK]D-Fender`	`akasa47: And if you remove it entirely?`
`15:03.36`	`akasa47`	`I don't know I never tried`
`15:03.42`	`[TK]D-Fender`	`akasa47: Now's a good time...`
`15:03.48`	`akasa47`	`:D`
`15:03.48`	`cadsii`	`anyone need help upgrading their cisco 7940 7960 phones to sip ? if so i can get it dont for you in 5 minutes`
`15:04.46`	`akasa47`	`is there any other configuration for conferences or for audio?`
`15:05.35`	`[TK]D-Fender`	`akasa47: Thats pretty much it... its either your timing source, your jitter buffering, or your network....`
`15:05.54`	`akasa47`	`Can I troubleshoot or get statistics in asterisk?`
`15:06.06`	`cadsii`	`asterisk -rvvvvvvvvv`
`15:06.10`	`cadsii`	`in consol`
`15:06.12`	`cadsii`	`in console`
`15:06.16`	`[TK]D-Fender`	`akasa47: I'm pretty sure there are options for this, but I have no experience with them.`
`15:06.27`	`[TK]D-Fender`	`cadsii: Doesn't help him any.....`
`15:06.34`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)`
`15:06.41`	`cadsii`	`what is the situation`
`15:06.44`	`*** join/#asterisk _matt (n=root@82-33-68-129.cable.ubr01.trow.blueyonder.co.uk)`
`15:06.44`	`akasa47`	`I use -rvvvvvvdddd`
`15:06.45`	`cadsii`	`give me a brief out`
`15:07.05`	`akasa47`	`I would need some statistics for rtp`
`15:07.14`	`*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)`
`15:07.32`	`akasa47`	`The problems is that in a meetme with about 15 users I get robotic voice`
`15:07.40`	`_matt`	`hello, i'm looking for some confrencing software that supports a whiteboard, voip and video, does anybody know of such software ?`
`15:08.01`	`[TK]D-Fender`	`_matt: Netmeeting <-----`
`15:08.16`	`_matt`	`[TK]D-Fender: lol yea that would be good, but its not continued anymore :(`
`15:08.28`	`[TK]D-Fender`	`_matt: Doesn't matter... still works...`
`15:08.28`	`cadsii`	`akasa47 what do you mean ?`
`15:08.32`	`cadsii`	`everyones voice`
`15:08.43`	`cadsii`	`or u get a msg after cetraint amount of users`
`15:09.03`	`*** join/#asterisk dominic1 (n=dob@213.221.82.242)`
`15:09.03`	`akasa47`	`sometimes, yes`
`15:09.11`	`_matt`	`[TK]D-Fender: it has to be supported tho :( its for use in production`
`15:09.14`	`*** part/#asterisk dominic1 (n=dob@213.221.82.242)`
`15:09.40`	`akasa47`	`when is starting then everybody has problem. Sometimes is cutting the audio too.`
`15:09.59`	`[TK]D-Fender`	`_matt: www.polycom.com`
`15:10.21`	`_matt`	`[TK]D-Fender: cheers, i will look at that`
`15:10.23`	`akasa47`	`Sometimes I get echo but I think this is because user use speaker phones`
`15:10.30`	`[TK]D-Fender`	`akasa47: It only really shows up in meetme right? Standard JB seems fine?`
`15:11.02`	`cadsii`	`yea conference should have no speaker`
`15:11.04`	`akasa47`	`[TK]D-Fender: Yes only in meeting. And no cpu usage on the server.`
`15:11.05`	`cadsii`	`unless its on mute`
`15:11.30`	`[TK]D-Fender`	`akasa47: Ok, I think we can scratch off everything... see if you can gt your hands on a TDM400P for use as a timing source...`
`15:11.39`	`cadsii`	`what compression ur clients using ?`
`15:12.08`	`akasa47`	`cadsii:no compresion only ulaw`
`15:12.17`	`*** join/#asterisk Ard0gx (n=adiaz0@190.128.167.127)`
`15:12.21`	`akasa47`	`I don't have TDM400P :(`
`15:12.51`	`cadsii`	`not a bandwidth issue ?`
`15:12.59`	`cadsii`	`u doing lan or nat?`
`15:13.12`	`cadsii`	`users local or remote ?`
`15:13.51`	`*** join/#asterisk agile (n=mike@68-187-47-224.dhcp.ftwo.tx.charter.com)`
`15:13.52`	`akasa47`	`I set in asterisk.conf internal_timing = yes. Does help this? I have Asterisk BE`
`15:13.59`	`Ard0gx`	`Hi guys, does anybody know if there is problem on installing asterisk on intel dual core servers ?`
`15:14.16`	`cadsii`	`i heard only problems with amd`
`15:14.23`	`akasa47`	`Users are registered to avaya pbx and I have sip trunks to avaya`
`15:14.29`	`cadsii`	`ive been only doing intel pcs and no prolbems`
`15:14.56`	`Ard0gx`	`I'm getting no audio using TE405P`
`15:15.15`	`Ard0gx`	`I have changed the board and still no audio`
`15:15.20`	`cadsii`	`why not use freepbx?`
`15:15.23`	`akasa47`	`I hava analog phones, cell phones, digital phones,`
`15:15.56`	`cadsii`	`how are the cells registering or is it just an inbound trunk ?`
`15:15.57`	`akasa47`	`but for asterisk is only SIP everything`
`15:16.45`	`akasa47`	`I call out from asterisk for conferenc so I call cellphones too through avaya`
`15:17.21`	`akasa47`	`Avaya and Asterisk are connected in the same switch so should not be network issue`
`15:18.17`	`cadsii`	`have you tried solo clients`
`15:18.28`	`cadsii`	`like 20 cells or 20 analog or 20 digital`
`15:18.43`	`cadsii`	`at least you can knock the phones out of the question if any of those work`
`15:18.54`	`cadsii`	`i mean if they all dont work`
`15:19.30`	`Ard0gx`	`I'm using asterisk-1.4.11 + zaptel-1.4.5.1 + TE405P + Intel server SR1530CLNA Barebone - Dual Core processor`
`15:20.08`	`akasa47`	`Yes, good point. I need to test with different phones.`
`15:20.51`	`akasa47`	`cadsii: with rtp debug can I get some statistics from asterisk?`
`15:21.28`	`Ard0gx`	`cadsii: what kind of intel server are you using ?`
`15:22.15`	`*** join/#asterisk Quintana (n=sylvain@office.proformatique.com)`
`15:22.23`	`cadsii`	`wish i could ttell ya aka`
`15:22.26`	`cadsii`	`havent used it`
`15:22.36`	`cadsii`	`u name it... i got a few`
`15:22.56`	`[TK]D-Fender`	`akasa47: Since individual channels seem fine I think you really need to test a hardware timing source.`
`15:23.32`	`cadsii`	`1 dual core p4 3.0/ 1 fujitsu laptop running vmware with trixbox / 1 p 3 600 running & a dl380 hp server (xeons)`
`15:25.00`	`akasa47`	`[TK]D-Fender: Maybe you'll right. I just wondering if somebody used meetme for large groups and had the same problem`
`15:25.53`	`[TK]D-Fender`	`akasa47: Pretty sure I've heard this around here a few times... The circumstances definitely start to single out your timer.... see if you can get a "loaner" to test.`
`15:26.03`	`akasa47`	`I am not yet asterisk guru so I try to figure out what settings are available`
`15:26.38`	`cadsii`	`how about using aussie's meetme teleconfrerence faq for trixbox ?`
`15:26.43`	`cadsii`	`not an option ?`
`15:26.51`	`cadsii`	`its for freepbx`
`15:27.23`	`akasa47`	`cadsii: I don't know about that`
`15:27.39`	`[TK]D-Fender`	`cadsii: This is NOT a Trixbox channel, nor do we suggest its use around here unless to person coming in is incompetant as an admin, and he is obviously far more aware of * setup than that.`
`15:27.43`	`*** join/#asterisk Zipper_32 (n=None@s206-116-61-252.bc.hsia.telus.net)`
`15:27.56`	`[TK]D-Fender`	`cadsii: You are leading him around in circles`
`15:28.17`	`agile`	`in my moh directories I want to ensure the files get played in a certain sequence, but I don't seem to be able to consistently do that. Tried naming the files a.sln, b.sln, c.sln etc also tried touching to set the mtime, anyone know how I might do that?`
`15:29.05`	`[TK]D-Fender`	`agile: Make 1 super-recording and use that instead of seperate recordings....`
`15:29.21`	`akasa47`	`agile: I think is played in name order`
`15:30.10`	`agile`	`[TK]D-Fender, is there an easy way to merge sln files?`
`15:31.24`	`tzafrir`	`concatenate them?`
`15:31.36`	`akasa47`	`agile:I use mp3 files`
`15:32.08`	`tzafrir`	`cat file1 file2 file3 >merged`
`15:32.17`	`[TK]D-Fender`	`agile: I think you'd best transcode them to something managable, merge and then leave as-is`
`15:32.36`	`Zipper_32`	`I can't seem to get MOH working from a fresh install. Am I missing something: Put mp3's in the /var/lib/asterisk/mohmp3/ as noted by musiconhold.conf, and that should be it?`
`15:32.41`	`*** join/#asterisk Nivex (n=kjotte@user-0c8hvoj.cable.mindspring.com)`
`15:32.45`	`agile`	`tzafrir, gonna try that`
`15:33.06`	`tzafrir`	`Zipper_32, why do you use mp3s for moh?`
`15:33.19`	`tzafrir`	`Unless you enjoy wasting CPU`
`15:33.25`	`akasa47`	`Zipper_32: yes you need files in the that directory`
`15:33.31`	`Zipper_32`	`Because that's what I have so far. I'm just trying to get it to work. Waves aren't working, and neither are the mp3's`
`15:33.53`	`*** join/#asterisk tsurko (n=tsurko@213.91.216.130)`
`15:34.00`	`tzafrir`	`Do you have format_mp3 ?`
`15:34.04`	`*** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com)`
`15:34.05`	`[TK]D-Fender`	`Zipper_32: You also need to either install a 3rd party player like MPG123 0.59r, or use Native MoH which requires Asterisk-addons to be installled.`
`15:34.13`	`Zipper_32`	`I have 8bit wav's, and the 64kbps mp3's, but none seem to play.`
`15:34.24`	`Zipper_32`	`I've installed the addons,`
`15:34.34`	`tzafrir`	`What wav files, exactly? file those_those_files.wav`
`15:35.01`	`akasa47`	`Zipper_32: I had to install manually mpg123 on my box`
`15:35.03`	`Zipper_32`	`It's called 'hot.wav'`
`15:35.12`	`tzafrir`	`They should be 16bit, 8kHz, mono`
`15:35.16`	`agile`	`nice`
`15:35.38`	`tzafrir`	`akasa47, that is a nice waste of CPU cycles in transcoding`
`15:35.39`	`Zipper_32`	`Is there a mpg123 for x64?`
`15:35.44`	`agile`	`concatenating the sln files worked, thanks!`
`15:36.03`	`tzafrir`	`on Debian:apt-get install mpg123 . But again: a constant waste of CPU time`
`15:36.13`	`Zipper_32`	`I'm not caring about wasting cpu cycles if it works. It's better than nothing right now.`
`15:36.22`	`*** join/#asterisk loompek (n=NoName@noname.rula.net)`
`15:36.26`	`loompek`	`hi ppl`
`15:36.26`	`akasa47`	`tzafir: should I use only native for moh?`
`15:36.31`	`Zipper_32`	`Don't I need 0.59r?`
`15:38.42`	`tzafrir`	`akasa47, if you use uncompressed wavs rather than high quality mp3s you will not waste CPU time for transcoding at play time`
`15:38.53`	`[TK]D-Fender`	`Zipper_32: You lso need to make sure your MP3's are NOT VBR and don't have ID3 taqgs`
`15:39.39`	`*** join/#asterisk hfb (n=hfb@pool-71-106-223-11.lsanca.dsl-w.verizon.net)`
`15:39.45`	`Zipper_32`	`Thanks [TK]D-Fender. The ones included, fpm-calm-river.mp3 fpm-world-mix.mp3, should work though, shouldn't they?`
`15:41.03`	`akasa47`	`Zipper_32: for me works.`
`15:41.18`	`Zipper_32`	`sigh. Okay.`
`15:41.48`	`Zipper_32`	`I have format_mp3 in my /usr/lib/asterisk/modules, so I'm trying to figure out why Asterisk immediately terminates the MOH session as soon as it starts.`
`15:42.14`	`akasa47`	`Zipper_32: beacuse something wrong.`
`15:42.33`	`akasa47`	`Zipper_32: it means doesn't play the audio`
`15:42.46`	`*** join/#asterisk ted_brown (n=angel@212.145.176.154)`
`15:43.00`	`Zipper_32`	`Alright, I'll keep testing, Thanks.`
`15:43.59`	`akasa47`	`Zipper_32: See what says the log or console and start from there`
`15:44.08`	`*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)`
`15:44.33`	`Zipper_32`	`oh! I've got it!`
`15:44.50`	`Zipper_32`	`Too high of a frequency on the .wavs`
`15:44.58`	`Zipper_32`	`WARNING[11472]: format_wav.c:169 check_header: Unexpected freqency 44100`
`15:45.01`	`tzafrir`	`And you'll also see that there's no big difference in file sizes`
`15:45.20`	`Zipper_32`	`Man, I've got to get some sleep... Thanks for your help [TK]D-Fender, tzafrir and akasa47.`
`15:45.34`	`*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)`
`15:45.36`	`tzafrir`	`Zipper_32, use sox or whatever to convert them to 8000 Hz`
`15:46.13`	`[TK]D-Fender`	`Zipper_32: Verify you have "format_mp3" and then check your permissions`
`15:46.51`	`akasa47`	`I need to get back to test my audio problem. :(`
`15:47.39`	`loompek`	`i have a problem...`
`15:47.44`	`akasa47`	`<PROTECTED>`
`15:47.47`	`loompek`	`http://rula.net/131`
`15:48.15`	`loompek`	`two asterisk servers... one as a server, second one as a client (register and type=friend)`
`15:48.28`	`loompek`	`calling doesn't work as i would like`
`15:48.28`	`loompek`	`:S`
`15:49.09`	`*** join/#asterisk ussrback (n=MAX@80.92.183.84)`
`15:49.20`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net)`
`15:49.53`	`Ard0gx`	`does somebody know if TOR3e works on 64 bits servers ?`
`15:53.06`	`ussrback`	`I have EL4, I have errors when installing zapatel. Please, help me`
`15:53.24`	`[TK]D-Fender`	`ussrback: PASTEBIN it....`
`15:53.58`	`keith4_`	`wtf is EL4?`
`15:54.11`	`[TK]D-Fender`	`keith4_: Red Hat Enterprise Linux 4`
`15:54.18`	`keith4_`	`oh, RHEL4`
`15:54.19`	`coppice`	`three steps before EL7`
`15:54.20`	`[TK]D-Fender`	`keith4_: And welcome to planet earth....`
`15:54.20`	`ussrback`	`Enterprise linux`
`15:54.20`	`keith4_`	`eww`
`15:54.36`	`[TK]D-Fender`	`nothing wrong with EL4...`
`15:54.52`	`coppice`	`nothing wrong with EL7`
`15:54.53`	`[TK]D-Fender`	`ussrback: So please pastebin your attempt so we can see the failure`
`15:54.53`	`keith4_`	`it's pretentions to suppose that "Enterprise Linux" could only refer to RHEL`
`15:55.08`	`keith4_`	`doesn't SUSE have an EL product?`
`15:55.09`	`[TK]D-Fender`	`coppice: unload chan_fluxcapacitor.so`
`15:55.52`	`coppice`	`chan_unicall.so can do EL7, though I have yet to release the protocol module for that`
`15:56.44`	`tzafrir`	`well, there was never a SUSE EL 4. They started at 7`
`15:56.59`	`[TK]D-Fender`	`coppice: So as a point for context... care to elaborate on what "EL7" you are referring to? :)`
`15:57.16`	`coppice`	`its a telephony signalling protocol`
`15:57.37`	`tzafrir`	`ah, so it isn't SLES7. Fine`
`15:59.18`	`ussrback`	`[TK]D-Fender: Here is my log http://sial.org/pbot/27823.`
`15:59.54`	`[TK]D-Fender`	`ussrback: I've never seen that make option before...`
`16:00.17`	`*** join/#asterisk gardo (n=gardo@121.97.211.248)`
`16:00.41`	`*** join/#asterisk viking78 (n=ViKing78@cerberus.franklinamerican.com)`
`16:00.50`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`16:01.50`	`ussrback`	`[TK]D-Fender: Look here, its a asterisl log http://sial.org/pbot/27825.`
`16:02.05`	`ussrback`	`when i looked for in Google why this occures`
`16:02.30`	`ussrback`	`there was some answer need to install ztdummy`
`16:02.33`	`viking78`	`Anyone have a suggestion of whether to use an IAXy or Grandstream ATA? Which is better?`
`16:02.41`	`ussrback`	`cause im trying to install it`
`16:02.48`	`cadsii`	`cisco`
`16:02.52`	`[TK]D-Fender`	`viking78: Both junk`
`16:03.11`	`cadsii`	`ata186`
`16:03.15`	`coppice`	`the evil of 2 lessers`
`16:03.21`	`viking78`	`[TK]D-Fender: What's good for a single line ATA then?`
`16:03.27`	`[TK]D-Fender`	`viking78: Linksys SPA's are the best entry-level option.`
`16:03.44`	`viking78`	`[TK]D-Fender: K, thanks`
`16:03.54`	`cadsii`	`cisco ata186 works great`
`16:03.58`	`cadsii`	`got 3 hookup up`
`16:04.05`	`cadsii`	`with cordless phones`
`16:04.46`	`drako`	`cadsii, how much does it cost?`
`16:05.01`	`cadsii`	`ebay around 100`
`16:05.18`	`Zipper_32`	`Just a side note, I got the music working. Had something to do with the bitrate of the music I had on there.`
`16:05.21`	`Zipper_32`	`Thank you guys,`
`16:05.31`	`*** join/#asterisk Deep_Dish (n=DaFresh@office.proformatique.com)`
`16:05.34`	`Deep_Dish`	`hi all,`
`16:06.00`	`[TK]D-Fender`	`cadsii: You are suggesting overpriced licensed hardware and pimping Trixbox.... you may want to reevaluate your approach and consider where you are...`
`16:06.19`	`cadsii`	`wtf is your problem loser`
`16:06.25`	`cadsii`	`cisco is bad hardware now`
`16:06.28`	`cadsii`	`?`
`16:06.39`	`Deep_Dish`	`i am trying to setup an Asterisk in Thailand, and it seams that there is some very specific configurations needed by the locals telco with the PRI link,`
`16:06.40`	`cadsii`	`and 100 is not too much for a quality ata`
`16:06.59`	`Deep_Dish`	`locals calls works, but internationnal calls always fails,`
`16:07.07`	`[TK]D-Fender`	`cadsii: First, tahts e-bay, and you that doesn't give you legal access to firmware.`
`16:07.19`	`Ard0gx`	`cad`
`16:07.21`	`viking78`	`cadsii: How much did you pay for the wireless phones connected to the ATAs? Just curious.`
`16:07.23`	`Deep_Dish`	`anyone have an idea ?`
`16:07.26`	`JT`	`cadsii: cisco is overpriced crap`
`16:07.33`	`JT`	`cadsii: and you were promoting freepbx`
`16:07.37`	`JT`	`which is just crap`
`16:07.37`	`cadsii`	`just regualr cordless phones`
`16:07.42`	`[TK]D-Fender`	`cadsii: Second, you can get a Linksys SPA-2102 for $67 NEW and not have to worry about it later.`
`16:08.14`	`cadsii`	`cause i use cisco i dont know crap`
`16:08.17`	`cadsii`	`nice assumption`
`16:08.18`	`cadsii`	`lol`
`16:08.20`	`Ard0gx`	`cadsii: are you using E1 boards connected to you 1 dual core p4 server ?`
`16:08.23`	`JT`	`where did we say that?`
`16:08.36`	`JT`	`cadsii: stop being defensive and don't put words in our mouthes`
`16:08.38`	`cadsii`	`you know cisco is the world leader in networking hardware rigbht`
`16:08.38`	`[TK]D-Fender`	`cadsii: And you may want to be a little careful about name-calling.`
`16:08.39`	`cadsii`	`?`
`16:08.46`	`viking78`	`So you've got a $50 phone hooked up to a $100 ATA?`
`16:08.50`	`JT`	`if you have a proper response, expand upon the reasoning`
`16:09.04`	`JT`	`not using stupid rhetoric like "they make the best hardware in the world"`
`16:09.07`	`JT`	`which they don't`
`16:09.16`	`coppice`	`who pays $50 for a phone?`
`16:09.19`	`cadsii`	`yea`
`16:09.24`	`cadsii`	`and who doesnt have a cordless at home`
`16:09.25`	`[TK]D-Fender`	`viking78: http://www.telephonydepot.com/Linksys_ATA_s/33.htm`
`16:09.34`	`[TK]D-Fender`	`cadsii: That'd be me...`
`16:09.44`	`cadsii`	`well maybe you should hit a garage sale`
`16:09.46`	`viking78`	`[TK]D-Fender: Thanks, That's where I'm shopping`
`16:09.51`	`cadsii`	`and u will find one for 2$`
`16:10.12`	`cadsii`	`so many haters on here`
`16:10.13`	`cadsii`	`jesus`
`16:10.14`	`JT`	`maybe you should hit an english lesson`
`16:10.15`	`ussrback`	`http://sial.org/pbot/27825.`
`16:10.17`	`[TK]D-Fender`	`viking78: I have bought fromt hem personally (internationally), and have several happy clients of mine I've referred there as well.`
`16:10.22`	`JT`	`and you will find more than $2 in your future`
`16:10.26`	`Ard0gx`	`cadsii: as you said you have asterisk on a 1 dual core p4 3.0, just wondering if you have any kind of E1 or T1 board connected to your server`
`16:10.42`	`*** join/#asterisk \|omni\| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)`
`16:10.43`	`viking78`	`[TK]D-Fender: I have about a $10k order ready with Polycom phones and Sangoma A101D cards. Figured I'd upgrade my ATA at home while I was at it.`
`16:10.46`	`Ard0gx`	`cadsii: or just using Voip ?`
`16:10.52`	`[TK]D-Fender`	`cadsii: You seem to be the one yelling and putting words in other peoples mouths. Go grab a coffee...`
`16:10.55`	`cadsii`	`i have 2 x100s 3 ata186 and sip providers`
`16:10.57`	`cadsii`	`home setup`
`16:10.58`	`cadsii`	`not office`
`16:11.01`	`[TK]D-Fender`	`viking78: hold that thought...`
`16:11.03`	`*** join/#asterisk \|omni\| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)`
`16:11.16`	`cadsii`	`2 x100s for backup`
`16:11.22`	`cadsii`	`no access to t1 at home house`
`16:11.36`	`[TK]D-Fender`	`viking78: Figured I'd go through a Q&A on your hardware just so see if I might have a different suggestion for you.`
`16:11.49`	`ussrback`	`app_meetme.c:769 build_conf: Unable to open pseudo device <--- what does it mean?`
`16:11.50`	`cadsii`	`how can you tell me cisco hardware is crap`
`16:11.54`	`viking78`	`[TK]D-Fender: pm me`
`16:11.54`	`Ard0gx`	`cadsii: ok, thanks`
`16:11.58`	`cadsii`	`is it cause you dont know howto confugre it`
`16:12.01`	`[TK]D-Fender`	`ussrback: Means ztdummy isn't loaded`
`16:12.05`	`JT`	`cisco hardware is overpriced and overhyped`
`16:12.17`	`JT`	`has no relationship to configuration`
`16:12.23`	`JT`	`cisco hardware is piss easy to configure`
`16:12.25`	`coppice`	`the voip stuff is not even cisco's`
`16:12.34`	`Ard0gx`	`Someone using E1 or T1 on a 64 bits server ?`
`16:12.54`	`coppice`	`Ard0gx: lots of people are`
`16:13.37`	`Ard0gx`	`and without audio problems ?`
`16:14.38`	`coppice`	`oh, no. people only use them to add a little trouble to their lives. if everything works properly the support people are out on their ears`
`16:14.50`	`loompek`	`and please.. not all at once!`
`16:15.00`	`*** join/#asterisk JasonK1909 (n=JasonK19@12.18.111.2)`
`16:15.24`	`JasonK1909`	`Anybody there? I'm a newbie...`
`16:16.00`	`Ard0gx`	`well, I have 3 E1 boards and tested on 2 intel 64 bits processor server and got audio problems`
`16:17.57`	`*** part/#asterisk agile (n=mike@68-187-47-224.dhcp.ftwo.tx.charter.com)`
`16:18.24`	`Ard0gx`	`I have tested those 3 E1 boards and the problem is there, but testing on a 32 bits processor server it works fine`
`16:19.00`	`*** join/#asterisk hfb (n=hfb@pool-71-106-223-11.lsanca.dsl-w.verizon.net)`
`16:19.42`	`*** join/#asterisk clandmeter (n=Carlo@81.175.82.2)`
`16:19.53`	`JasonK1909`	`I have a question re: failiure to register with an asterisk server, not sure if this is the right place. My sip phone (Leadtek BVP 8882) will reg with one server, but not another. Both servers are comparably configured but have different builds. In the "good" case the phone will intercept the WWW-authorize & include it in the 2nd REGISTER request; but in the "bad" case every it doesn't seem to see the WWW-authorize and never includes it in subsequ`
`16:19.54`	`JasonK1909`	`ent registrations. Anyone seen this problem before?`
`16:21.36`	`coppice`	`Ard0gx: which E1 boards? I have used tor2, and TE410P OK. I have used sangoma A104D, but with some recent versions of the drivers there is audio trouble on 64 bit machines. I think the last version has fixed that`
`16:22.29`	`*** join/#asterisk bfrance (n=brian@adsl-68-72-34-207.dsl.ipltin.ameritech.net)`
`16:22.57`	`*** join/#asterisk ajohnson (n=ajohnson@corpex.pivotal.televerde.com)`
`16:23.04`	`ussrback`	`[TK]D-Fender: Do i need Chanspy, to make private chats with conference members?`
`16:23.18`	`*** join/#asterisk aninoSAdilim (n=a@58.69.243.203)`
`16:23.39`	`[TK]D-Fender`	`ussrback: not sure how that'd work....`
`16:23.57`	`*** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose)`
`16:24.07`	`ussrback`	`yes finally, i configured meetme`
`16:24.24`	`ussrback`	`but now i want something to do with private calls`
`16:24.32`	`ussrback`	`and dont know hot to do it`
`16:25.06`	`[TK]D-Fender`	`ussrback: from who to who?`
`16:25.25`	`[TK]D-Fender`	`ussrback: and you want this to occur DURING the conference?`
`16:25.39`	`*** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue)`
`16:25.54`	`*** part/#asterisk bfrance (n=brian@adsl-68-72-34-207.dsl.ipltin.ameritech.net)`
`16:26.03`	`*** join/#asterisk tsearle (n=tsearle@cable-213-132-158-189.upc.chello.be)`
`16:26.24`	`ussrback`	`let say, in some conference room i have 5 participants. they chat about politics :) and some person decided to talk tet-a-tet with someone from that conference room`
`16:26.42`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`16:26.51`	`elriah`	`tet-a-tet?`
`16:27.01`	`[TK]D-Fender`	`tete-a-tete*`
`16:27.07`	`ussrback`	`is it possible to make such dialplan, that any user can ask to someone else from conference to go in private`
`16:27.14`	`JasonK1909`	`Is this the right place to ask for help w Asterisk REGISTER problems?`
`16:27.18`	`elriah`	`Ahh...`
`16:27.35`	`cadsii`	`its call msn`
`16:27.36`	`cadsii`	`lol`
`16:27.46`	`[TK]D-Fender`	`ussrback: Only way is to leave the conference or just place a completely seperate call.`
`16:28.19`	`[TK]D-Fender`	`cadsii: You're right... MSn has voice chat.... we should clearly all give up on * and use that instead! :)`
`16:28.50`	`ussrback`	`yes but i mean something like phone chat`
`16:28.55`	`ussrback`	`on asterisk`
`16:29.43`	`ussrback`	`i have read thet there is something like whisper`
`16:29.50`	`ussrback`	`in chanspy`
`16:29.58`	`[TK]D-Fender`	`ussrback: Sorry, there is no "direct chat" option while in a conference. You'll either have to place a new call or leave the conference and join a private on...`
`16:30.02`	`ussrback`	`which should be used for this`
`16:30.25`	`[TK]D-Fender`	`ussrback: You can't even execute chanspy while IN the conference like that... only a NEW channel can call that.`
`16:30.26`	`ussrback`	`yes sure, leave conference`
`16:30.39`	`ussrback`	`if the other part accepts private chat`
`16:31.01`	`[TK]D-Fender`	`ussrback: Like if you knew someone in the conference and wanted to whisper to them that they have a meeting in 10 mins... that you could do... but not between 2 members like that`
`16:31.25`	`ussrback`	`ahhh okkkk`
`16:31.26`	`[TK]D-Fender`	`ussrback: So sorry, it just isn't made for that. You're welcome to go try and code it yourself though...`
`16:31.36`	`[TK]D-Fender`	`ussrback: Since thats what it'll come down to.`
`16:32.22`	`ussrback`	`[TK]D-Fender: do u have any info or doc how to make a such action with asterisk?`
`16:33.01`	`[TK]D-Fender`	`ussrback: For what you want? Nothing exists for that... you'll have to learn how *'s code works and use a lot of your own imagination.`
`16:33.08`	`JT`	`K&R C`
`16:33.28`	`ussrback`	`:) OK thanks`
`16:33.56`	`elriah`	`... and it would be 10-feet thick`
`16:33.56`	`*** part/#asterisk gabbernaster (n=jshanks@69.10.147.2)`
`16:38.51`	`*** join/#asterisk GreggB (n=GreggB@66.206.86.107)`
`16:40.50`	`GreggB`	`I'm looking for a reputable SIP or IAX provider - anyone heard of voicetrading.com?`
`16:41.18`	`elriah`	`vitelity, les.net, heavylogic`
`16:41.24`	`*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)`
`16:41.35`	`[TK]D-Fender`	`~itsp`
`16:41.35`	`jbot`	`An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower`
`16:41.36`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^`
`16:41.47`	`*** part/#asterisk ming_zym (n=ming_zym@124.254.56.33)`
`16:42.20`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`16:42.42`	`JasonK1909`	`looking for help with asterisk REGISTER authorization problem`
`16:42.53`	`tzafrir`	`~thebook`
`16:42.54`	`jbot`	`rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`16:42.54`	`elriah`	`JasonK1909: shoot`
`16:43.38`	`*** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net)`
`16:43.54`	`tzafrir`	`~thebook`
`16:43.54`	`jbot`	`somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://openbooks.oreilly.com/`
`16:44.07`	`[TK]D-Fender`	`JasonK1909: PASTEBIN it...`
`16:44.08`	`[TK]D-Fender`	`~pb`
`16:44.09`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`16:45.00`	`*** join/#asterisk ming_zym (n=ming_zym@124.254.56.33)`
`16:48.13`	`elriah`	`JasonK1909: Dude, don't appreciate the flood. Just pastebin your sip.conf and ask your question in the main channel.`
`16:50.01`	`*** join/#asterisk squall (n=squall@ns2.squallnetwork.net)`
`16:51.42`	`JasonK1909`	`My sip.conf is...`
`16:51.47`	`JasonK1909`	`[general]`
`16:51.47`	`JasonK1909`	`context=default`
`16:51.47`	`JasonK1909`	`bindport=5060`
`16:51.47`	`JasonK1909`	`bindaddr=0.0.0.0`
`16:51.47`	`JasonK1909`	`srvlookup=yes`
`16:51.48`	`JasonK1909`	`[authentication]`
`16:52.16`	`*** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)`
`16:52.56`	`elriah`	`JasonK1909: Somebody will ultimatly just kick you out of the channel if you just paste sh*t ... Use pastebin (google it)`
`16:54.47`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-94-123-110.dsl.hstntx.swbell.net)`
`16:55.13`	`hmmhesays`	`indeed`
`16:55.52`	`*** join/#asterisk ussrback (n=MAX@80.92.183.84)`
`16:56.14`	`elriah`	`JasonK1909: This IRC channel is easy, just be polite, don't ask to ask, ask questions thouroughly and pastebin any config files or multi-line text and use the URL you generate from pastebin in your question.`
`16:56.45`	`tzafrir`	`elriah, next time just write '~pb' and let jbot do the rest :-)`
`16:56.46`	`elriah`	`i.e., I can't get my Polycom 301 to register with Asterisk 1.2.24, here's my sip.conf: http://pastebin.com/12345`
`16:57.37`	`Nivex`	`heh, nice pastebin number :)`
`16:58.27`	`ussrback`	`How can i limit the number of participants in conference call?`
`16:58.28`	`*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)`
`16:59.19`	`JasonK1909`	`OK sorry for spam, my sip.conf is http://pastebin.com/m502435ee & asterisk log is http://pastebin.com/m1f80c6fe`
`16:59.56`	`elriah`	`JasonK1909: You don't have any peers defined in your sip.conf`
`17:00.06`	`tzafrir`	`elriah, I believe that you pasted an incorrect pastebin number, though`
`17:00.24`	`elriah`	`tzafrir: I was showing JasonK1909 how to use the channel, sorry, it was an example`
`17:00.25`	`tzafrir`	`ah, oops, I wasn't reading :-)`
`17:00.43`	`elriah`	`JasonK1909: http://www.voip-info.org/wiki-Asterisk+config+sip.conf`
`17:01.22`	`JasonK1909`	`hm I have a Digium vendor box, the user shows in the webpage so I guess they save it somewhere besides /etc/sip.conf`
`17:01.25`	`*** part/#asterisk ming_zym (n=ming_zym@124.254.56.33)`
`17:01.38`	`JasonK1909`	`looking for that elsewhere`
`17:01.44`	`elriah`	`JasonK1909: If you do a "sip show peers" from your cli, what do you get? (pastebin if it's big)`
`17:01.56`	`elriah`	`er.., in 1.4, it may be "core sip show peers"`
`17:02.00`	`tzafrir`	`JasonK1909, what version of asterisk?`
`17:02.18`	`tzafrir`	`is that "something" users.conf?`
`17:03.45`	`JasonK1909`	`http://pastebin.com/m1c81b7eb shows peer 6001 is known`
`17:05.19`	`elriah`	`JasonK1909: What's a Digium Vendor Box? You mean Asterisk Business Edition?`
`17:05.38`	`elriah`	`Also, pastebin your /etc/asterisk/extconfig.conf`
`17:05.48`	`JasonK1909`	`asterisk appliance aa50`
`17:05.51`	`*** join/#asterisk mltlnx (n=mltlnx@209.10.153.194)`
`17:06.09`	`elriah`	`Ahh.. so you're configuring everything via a GUI?`
`17:06.41`	`[TK]D-Fender`	`JasonK1909: SIP/2.0 401 Unauthorized <- bad user/pass. End of story`
`17:07.14`	`[TK]D-Fender`	`JasonK1909: and you'd be referring to USERS.CONF.`
`17:07.24`	`[TK]D-Fender`	`JasonK1909: * GUI is not supported here.`
`17:07.32`	`JasonK1909`	`extconfig.conf http://pastebin.com/m5eb84e4d`
`17:07.43`	`elriah`	`[TK]D-Fender: Yea, I'm trying to figure out his setup, he's using the new Asterisk appliance apparently with 1.4 but his "sip show peers" shows a single peer but his "sip.conf" shows none so I'm thinking static realtime or something.`
`17:07.46`	`*** join/#asterisk saftsack (n=saftsack@pD9E05A1B.dip.t-dialin.net)`
`17:07.57`	`[TK]D-Fender`	`elriah: No.... USERS.CONF`
`17:08.04`	`elriah`	`[TK]D-Fender: Ahh..`
`17:08.11`	`JasonK1909`	`ah sec`
`17:08.14`	`[TK]D-Fender`	`elriah: Realtime on an APPLIANCE? get off that crack :p`
`17:08.15`	`elriah`	`JasonK1909: Pastebin your users.conf`
`17:08.35`	`elriah`	`[TK]D-Fender: No clue, never seen the appliance other than the nice fancy picture on the digum website ...`
`17:09.28`	`[TK]D-Fender`	`elriah: Don't dream of it using anything fancy.. it doesn't have the CPU to spare...`
`17:09.50`	`JasonK1909`	`users.conf http://pastebin.com/m3dc4201c`
`17:10.20`	`elriah`	`Ok, now in your phone, what SIP username/password are you using?`
`17:10.33`	`*** join/#asterisk KnowWhat (n=jjk@host210-2-174-38.isb.dancom.net.pk)`
`17:10.37`	`JasonK1909`	`for both phone & server i set name 6001, pwd 6001`
`17:10.55`	`[TK]D-Fender`	`JasonK1909: What kind of phone?`
`17:10.56`	`JasonK1909`	`... and it works fine for a different aa50`
`17:11.09`	`JasonK1909`	`phone is Leadtek BVP8882`
`17:11.23`	`[TK]D-Fender`	`registersip = yes <--- BAD.`
`17:11.38`	`[TK]D-Fender`	`Your phone is NOT an ITSP, * should not be trying to register to IT.`
`17:11.59`	`JasonK1909`	`i'll try changing that to no`
`17:12.26`	`[TK]D-Fender`	`JasonK1909: And other than that it domes down to not trusting your device setup.`
`17:12.33`	`elriah`	`Take it down to this: http://pastebin.com/m1ed27e12`
`17:13.06`	`[TK]D-Fender`	`elriah: ...... missing important stuff....`
`17:13.15`	`elriah`	`And add type=friend`
`17:13.37`	`JasonK1909`	`ok i'll make that change & get back to you ... ugh boss is bugging me, g2g`
`17:13.46`	`JasonK1909`	`thx for the help, sorry for the spam earlier`
`17:14.05`	`[TK]D-Fender`	`elriah: .....`
`17:14.39`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`17:16.15`	`GreggB`	`I've got a PRI (Zaptel TE120P) which keeps failing (asterisk gives fast-busy), yet no alarms are present via "zap show status" and the status of "pri show span 1" shows`
`17:16.18`	`GreggB`	`"Status: Provisioned, Up, Active"`
`17:17.22`	`*** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net)`
`17:17.47`	`GreggB`	`It's been working fine for ~9 months, with no changes but library/asterisk upgrades (running latest) - the telco is now blaming the card for the failures, but both bpviol and CRC4 show 0 for this card. Is there any way to further test this card to rule out possible hardware/software issues?`
`17:20.38`	`[TK]D-Fender`	`GreggB: Well you COULD enable PRI debug and pastebin an incoming call at verbose 10.....`
`17:20.52`	`[TK]D-Fender`	`GreggB: Since you haven't shown us anything of use yet...`
`17:22.18`	`*** join/#asterisk vutamhoan (n=hoavq@58.187.93.124)`
`17:24.02`	`vutamhoan`	`I have an IVR server using E1/R2 - Incomming call got the noise at silent of playback. Can anyone help?`
`17:25.42`	`vutamhoan`	`Using Record, audio received is perfect :(`
`17:25.43`	`GreggB`	`Oops, sorry... :-) - http://www.pastebin.ca/721934`
`17:26.38`	`*** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net)`
`17:33.56`	`vutamhoan`	`plz take a look at http://www.pastebin.ca/721947 - Thanks`
`17:35.11`	`*** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187)`
`17:36.49`	`*** join/#asterisk farish (n=chatzill@124.105.193.68)`
`17:40.33`	`*** part/#asterisk deeperror (n=deeperro@adsl-75-46-148-11.dsl.sfldmi.sbcglobal.net)`
`17:42.14`	`[TK]D-Fender`	`GreggB: [Oct 1 10:22:20] NOTICE[13578]: pbx.c:1613 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') <--- dialplan errors`
`17:43.01`	`GreggB`	`Thanks - I'll correct that, but could that be causing the random fast-busy and dropped T1?`
`17:43.28`	`[TK]D-Fender`	`GreggB: Sure, since * is never ANSWERING the call. Its a failure`
`17:43.59`	`GreggB`	`If I reset the card with /sbin/ztcfg -vvvv, we can usually make calls for a while, though sometimes we have to call from an outside (ie: mobile) line - once we do that, we can make outbound calls again`
`17:44.02`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-94-123-110.dsl.hstntx.swbell.net)`
`17:45.27`	`[TK]D-Fender`	`GreggB: Fix your dialplan.....`
`17:47.53`	`*** join/#asterisk DirtyD (i=DigiD@ool-18bddac2.dyn.optonline.net)`
`17:48.28`	`GreggB`	`[TK]D-Fender: done`
`17:48.45`	`DirtyD`	`so.. two users are dialed into my asterisk PBX both one is checking his voice mail.. Is there a way that the other user can invite him to "chat"?`
`17:49.23`	`GreggB`	`[TK]D-Fender: would another pastebin of an inbound call now help?`
`17:49.41`	`[TK]D-Fender`	`DirtyD: You could always use chanspy to whisper page him...`
`17:49.52`	`[TK]D-Fender`	`GreggB: Yes, at verbose 10 please...`
`17:50.28`	`DirtyD`	`Fender: After chanspy, how does one connect the two channels?`
`17:51.27`	`iBuMp`	`how to setup my licensed g729 on only my iax2 trunk`
`17:52.06`	`GreggB`	`[TK]D-Fender: http://www.pastebin.ca/721970`
`17:52.36`	`*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)`
`17:52.58`	`[TK]D-Fender`	`DirtyD: use a private Meetme room I guess.`
`17:54.24`	`[TK]D-Fender`	`GreggB: Why aren't I seeing any normal CLI output?`
`17:54.32`	`*** join/#asterisk freezey (n=freezey@maher.mercy.edu)`
`17:54.46`	`[TK]D-Fender`	`GreggB: show me you're setting the verbose level and try again, and set debug 10 as well`
`17:55.00`	`freezey`	`in the conf file manager.con at the bottom you permit certain ips... what exactly is that permitting them to do?`
`17:55.04`	`*** join/#asterisk bfrance (n=brian@adsl-68-72-34-207.dsl.ipltin.ameritech.net)`
`17:55.33`	`GreggB`	`[TK]D-Fender: I copy pasted just after the command to enable debugging`
`17:55.37`	`freezey`	`asterisk giving me a few problems so i am trying to debug the situation`
`17:55.59`	`_x86_`	`hey guys, I'm trying to figure out how to make the "Messages" button on a Polycom IP501 phone to dial my voicemail extension correctly`
`17:56.03`	`_x86_`	`any ideas?`
`17:56.15`	`[TK]D-Fender`	`_x86_: try reading the admin guide :)`
`17:56.55`	`freezey`	`i have 5 different locations... and in 2 of them one persons extension is the same as the other one and when i dial from inside the building it goes to the wrong person but if i dial form the outside it goes trhoguh no problem... i know i have to set +x for the other campus i am just not sure where thats being inputed...`
`17:56.56`	`_x86_`	`[TK]D-Fender: i'm looking all over to find the value in the provisioning files i need to change`
`17:57.01`	`freezey`	`if your confused by my statement believe me so am i`
`17:57.15`	`_x86_`	`[TK]D-Fender: last i checked, the admin guide didn't cover the provisioning files`
`17:57.26`	`GreggB`	`[TK]D-Fender: http://www.pastebin.ca/721980`
`17:58.01`	`[TK]D-Fender`	`_x86_: Umm... LOL!`
`17:58.08`	`bfrance`	`Anybody know the difference between TDM400P version E/F, H and I? Are there any known issues with any of them? One version better than the others?`
`17:58.14`	`[TK]D-Fender`	`_x86_: Get off that crack... thats ENTIRELY what they are for!`
`17:58.15`	`_x86_`	`[TK]D-Fender: url me?`
`17:58.22`	`[TK]D-Fender`	`_x86_: www.polycom.com`
`17:58.24`	`[TK]D-Fender`	`:p`
`17:58.34`	`_x86_`	`[TK]D-Fender: i'm looking at the admin guide for the IP501 now, no privisioning stuffs in it`
`17:58.42`	`_x86_`	`[TK]D-Fender: which admin guide should I be looking at?`
`17:59.05`	`[TK]D-Fender`	`_x86_: thats not the ADMIND guide then`
`17:59.19`	`[TK]D-Fender`	`_x86_: Go download your SIP app's admin guide.`
`18:00.18`	`freezey`	`[TK]D-Fender: in the manager.conf file at theres a section called [asterisk] under that deny=blahblah permit=blah whats the permit section for exactly?`
`18:01.12`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`18:01.58`	`[TK]D-Fender`	`freezey: says what IP's can connect to that account`
`18:02.26`	`freezey`	`[TK]D-Fender: and when you say connect what exactly are they connecting to?`
`18:02.59`	`freezey`	`[TK]D-Fender: and if i add another do i have to restart anything or will it just update it`
`18:05.25`	`[TK]D-Fender`	`freezey: they are connecting to the * AMI`
`18:05.47`	`freezey`	`[TK]D-Fender: and what is what exactly?`
`18:06.09`	`freezey`	`ok ok`
`18:06.10`	`freezey`	`i got that`
`18:06.15`	`freezey`	`so if i add another IP into that`
`18:06.20`	`freezey`	`then i should beable to it hit via http`
`18:06.28`	`freezey`	`cuase when i try to hit it via http now it says i do not have access`
`18:07.37`	`freezey`	`ok so i did add myself but still cant hit it via http`
`18:08.27`	`adeel`	`i keep seeing SIP/2.0 404 Not Found sip messages, even though the user/secret is correct...is there a way to get rid of those 404 messages?`
`18:08.27`	`freezey`	`[TK]D-Fender: can you explain how exactly i login?`
`18:10.38`	`[TK]D-Fender`	`adeel: 404 can be a DIALPLAN error as well if you are not dialing a valid #.`
`18:10.46`	`[TK]D-Fender`	`freezey: Do you even know what AMI is FOR?`
`18:11.05`	`adeel`	`[TK]D-Fender, i see those error's without any activity really going on`
`18:11.36`	`[TK]D-Fender`	`adeel: enable SIP debug & pastebin a failed call @ verbose 10`
`18:11.54`	`freezey`	`[TK]D-Fender: i have an idea... i am just wondering howto access it... my problem is i am gettin crossed extensions`
`18:12.00`	`adeel`	`well that's the thing, the calls don't fail...it goes through, but i keep seeing those random 404 error's`
`18:12.03`	`freezey`	`[TK]D-Fender: so i am wondering if that could help me`
`18:13.09`	`adeel`	`[TK]D-Fender, here's the output...http://pastebin.ca/722004`
`18:13.29`	`*** join/#asterisk famicon (i=scenesta@c51447ddc.cable.wanadoo.nl)`
`18:13.44`	`*** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl)`
`18:14.37`	`*** join/#asterisk ectospasm (n=ectospas@c-68-62-214-83.hsd1.al.comcast.net)`
`18:16.00`	`[TK]D-Fender`	`freezey: "crossed extensions"?!`
`18:16.31`	`[TK]D-Fender`	`adeel: [Oct 1 11:10:41] Looking for s in incoming (domain 64.71.180.2) SIP/2.0 404 Not Found <------ yup, DIALPLAN ERROR`
`18:17.08`	`adeel`	`[TK]D-Fender, oh okay...any ideas where to look?`
`18:17.30`	`[TK]D-Fender`	`adeel: Geee.... I dunno.. how about in your [incoming] context! :p`
`18:17.50`	`[TK]D-Fender`	`adeel: Its telling you to your face exactly what it is looking for... what more do you need?`
`18:18.01`	`adeel`	`[TK]D-Fender, haha...apparently someone to hold my hand =p`
`18:18.19`	`[TK]D-Fender`	`adeel: My rates are very accessible :)`
`18:18.31`	`adeel`	`[TK]D-Fender, hahah, i recall`
`18:18.59`	`*** join/#asterisk juliux (n=juliux@ubuntu/member/juliux)`
`18:19.02`	`juliux`	`hi all`
`18:19.13`	`karleeto`	`for a 10 phone asterisk system do i need a gigabit switch? or is 10/100 fine?`
`18:19.37`	`jbroome`	`10/100 is fine`
`18:19.45`	`_x86_`	`[TK]D-Fender: is it possible to provision the admin password to the phones as well?`
`18:19.59`	`karleeto`	`how much bandwidth does a single phone have?`
`18:19.59`	`_x86_`	`or am i going to have to put that in each time i setup a new phone manually`
`18:20.14`	`[TK]D-Fender`	`_x86_: I'm not sure actually....`
`18:20.42`	`juliux`	`i have a problem with my asterisk installation, i have two peers configure peer 1 is connected via the localnetwork on ip 172.23.42.x peer 2 is connected via openvpn on ip 172.23.50.x but if i try to connected the second peer i only get a time out but now log entry in asterisk, what did i wrong?`
`18:20.49`	`[TK]D-Fender`	`karleeto: 10 k BYTE / s :p .....`
`18:21.15`	`freezey`	`[TK]D-Fender: one person calling somebody with say extension 2255 trying to reach them at one building but gettin exteion 2255 at another building when`
`18:21.19`	`*** join/#asterisk mltlnx (n=mltlnx@m0a5f36d0.tmodns.net)`
`18:21.23`	`[TK]D-Fender`	`karleeto: bandwidth on a LAN is IRRELEVANT`
`18:21.48`	`[TK]D-Fender`	`freezey: Sounds like a dialplan error or forwarding issue..`
`18:22.29`	`*** join/#asterisk FlatFoot (n=simon@80.88.192.83)`
`18:22.30`	`_x86_`	`[TK]D-Fender: cool... well i found the parameter i needed, i think`
`18:22.35`	`freezey`	`[TK]D-Fender: what is the dialplan conf file?`
`18:22.46`	`[TK]D-Fender`	`freezey: .......`
`18:22.48`	`[TK]D-Fender`	`....`
`18:22.49`	`[TK]D-Fender`	`.`
`18:22.52`	`[TK]D-Fender`	`UGH.`
`18:22.59`	`[TK]D-Fender`	`freezey: extensions.conf <-----`
`18:23.12`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`18:23.37`	`freezey`	`[TK]D-Fender: i apologize first day in asterisk and i am trying to figure out somebodys old setup and trying to fix it ha when i have no idea about aterisk so i am just playin as i go`
`18:23.39`	`*** join/#asterisk juanjoc (n=juanjoc@host127.190-31-247.telecom.net.ar)`
`18:23.56`	`[TK]D-Fender`	`freezey: Ah.... then :`
`18:24.01`	`[TK]D-Fender`	`~hafc`
`18:24.02`	`jbot`	`extra, extra, read all about it, hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out.`
`18:24.03`	`[TK]D-Fender`	`:p`
`18:24.45`	`[TK]D-Fender`	`freezey: And..... my rates are very accessible ;)`
`18:24.56`	`freezey`	`[TK]D-Fender: when there is an _ infront of an extension what does that mena exactly`
`18:24.57`	`freezey`	`lol`
`18:25.18`	`_x86_`	`[TK]D-Fender: i set msg.mwi.1.subscribe="8500", msg.mwi.1.callBackMode="contact", msg.mwi.1.callBack="8500", where 8500 is my voicemail extension`
`18:26.00`	`[TK]D-Fender`	`_x86_: Leave the subscribe blank, its not needed...`
`18:26.06`	`[TK]D-Fender`	`_x86_: but congrats on the rest`
`18:26.15`	`SplasPood`	`hrm, are there any instances where ForkCDR would not in fact end up writing out two CDR records?`
`18:26.19`	`[TK]D-Fender`	`freezey: Here :`
`18:26.21`	`[TK]D-Fender`	`~book`
`18:26.22`	`jbot`	`[book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf`
`18:26.23`	`*** join/#asterisk akaast47 (i=0ca5bc82@gateway/web/cgi-irc/ircatwork.com/x-f63a896508f0fd2b)`
`18:26.28`	`adeel`	`_x86_, I set the callBack to *97 so it takes me directly to the users voicemail, instead of ringing the ext`
`18:26.57`	`adeel`	`brb`
`18:27.07`	`SplasPood`	`freezey: it means what follows is a pattern match not an explicit extension, but as others suggested the book is helpful`
`18:27.43`	`_x86_`	`[TK]D-Fender: but it doesn't work!`
`18:27.50`	`SplasPood`	`[TK]D-Fender: btw my polycom reg issue seems to be config related.. I haven't had time to narrow it down to a specific setting, but once I changed some of my configs the problem went away`
`18:28.13`	`_x86_`	`[TK]D-Fender: when my phone booted up, it didn't try to pull phone1.cfg, just sip.cfg, macaddress.cfg, and x7099.cfg`
`18:28.16`	`freezey`	`[TK]D-Fender: yeah i have that`
`18:28.29`	`_x86_`	`[TK]D-Fender: i think that's the problem`
`18:28.33`	`freezey`	`SplasPood: thanks man`
`18:28.36`	`SplasPood`	`_x86_: if macaddress.cfg doesn't reference phone1.cfg it'll never be pulled`
`18:28.51`	`*** join/#asterisk DemonJester (n=DemonJes@65.249.165.118)`
`18:29.16`	`*** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca)`
`18:29.24`	`freezey`	`Also exten => _6XXX,1,Macro(dialout-callmanager,${EXTEN}) ; MANVOIP`
`18:29.27`	`freezey`	`when something looks like that`
`18:29.30`	`freezey`	`the 1 means what?`
`18:29.36`	`_x86_`	`ah... i thought it was a default thing... dunno how it got taken out of macaddress.cfg`
`18:29.36`	`[TK]D-Fender`	`CCM!!!! WHEE!!!!`
`18:29.50`	`DeeJayTwo`	`is there something unsupported with ptime control in sip.conf? (allow=ulaw:10)`
`18:30.00`	`[TK]D-Fender`	`freezey: Ok, you are BEYOND newb right now. If we're answering this you've come to the wrong place...`
`18:30.04`	`DeeJayTwo`	`We're using it in 1.4.3 and it always sends a ptime of 20..`
`18:30.17`	`[TK]D-Fender`	`freezey: You do not appear to understand the slightest thing about the dialplan.`
`18:30.22`	`freezey`	`nope`
`18:30.23`	`freezey`	`lol`
`18:30.27`	`freezey`	`like i said`
`18:30.32`	`freezey`	`today is my first day going into this`
`18:30.44`	`[TK]D-Fender`	`freezey: Either hire a consultant or sit down with the book for a few hours working on a separate * install.`
`18:31.00`	`freezey`	`alright i will read the book`
`18:31.04`	`freezey`	`but if i have questions i am comin here!`
`18:31.05`	`freezey`	`ha`
`18:31.07`	`[TK]D-Fender`	`freezey: This is not a place to teach you every single character of syntax fromt he ground up.`
`18:31.24`	`[TK]D-Fender`	`freezey: Sorry, but you are in the "tragic overkill" category.`
`18:31.25`	`freezey`	`ok thats cool`
`18:31.28`	`freezey`	`hahah`
`18:31.34`	`freezey`	`well i have never used this so i am just asking`
`18:31.37`	`freezey`	`but`
`18:31.46`	`freezey`	`i think i understand the basic of that`
`18:31.52`	`freezey`	`the 1 is just the first entry in that department`
`18:31.54`	`_x86_`	`freezey: get The Book`
`18:31.56`	`_x86_`	`~thebook`
`18:31.57`	`jbot`	`from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://openbooks.oreilly.com/`
`18:31.58`	`freezey`	`i got i got it`
`18:32.05`	`[TK]D-Fender`	`Thats like joining #ubuntu and asking "whats a CD-ROM"? There is sucha thing as "minimum knowledge"`
`18:32.12`	`freezey`	`ok ok`
`18:32.14`	`freezey`	`i will read`
`18:32.16`	`freezey`	`then i am comin back`
`18:32.16`	`freezey`	`ha`
`18:32.17`	`[TK]D-Fender`	`GOOD`
`18:32.24`	`_x86_`	`freezey: they even give The Book away in digital form so it doesn't even cost you anything`
`18:32.33`	`freezey`	`i know i have the book already`
`18:32.34`	`freezey`	`i just got it`
`18:33.07`	`_x86_`	`then open it newb`
`18:33.12`	`_x86_`	`;)`
`18:33.14`	`freezey`	`haha`
`18:33.20`	`freezey`	`gonna do that now`
`18:33.32`	`_x86_`	`read the characters that formulate words, which in turn combine to make coherent sentances`
`18:33.51`	`_x86_`	`multiple sentances combine to convey an idea onto the reader`
`18:34.12`	`_x86_`	`it's the coolest thing since sliced bread OMGWTFBBQKTHX`
`18:34.33`	`[TK]D-Fender`	`mmmmmm BBQ`
`18:35.14`	`*** join/#asterisk eldon (i=eldon@nat/digium/x-6ce711c2380d33fe)`
`18:35.35`	`_x86_`	`[TK]D-Fender: ok man, this is pissing me off.. it's still calling itself...`
`18:36.07`	`_x86_`	`[TK]D-Fender: i hit messages, and it DOESNT bypass the instant messaging crap, then i hit connect, and the damn thing tries calling itself`
`18:36.15`	`_x86_`	`i verified that phone1.cfg is being requested now`
`18:36.35`	`[TK]D-Fender`	`_x86_: Make sure its not getting overridden int he other config....`
`18:37.01`	`_x86_`	`aha!`
`18:37.11`	`_x86_`	`damn ur smart lolz kthx`
`18:37.16`	`_x86_`	`</aol>`
`18:37.54`	`karleeto`	`anyone know a secret spot where i could grab polycom's bootrom 4.0.0?? my retailer's download site is broken, and i really need to upgrade these phones today, as im installing them tomorrow`
`18:38.32`	`*** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net)`
`18:39.22`	`[TK]D-Fender`	`karleeto: Get another vendor to help you`
`18:40.16`	`_x86_`	`karleeto: read this carefully: http://www.voip-info.org/tiki-index.php?page=Polycom+SoundPoint+IP+501`
`18:40.39`	`_x86_`	`karleeto: it has a link to a polycom download site`
`18:41.06`	`asdx`	`how much does a tdm400p cost right now?`
`18:41.38`	`[TK]D-Fender`	`asdx: Depends who you buy it from.`
`18:41.40`	`*** join/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net)`
`18:41.55`	`[TK]D-Fender`	`asdx: I can get you one for 500$ USD fully loaded :)`
`18:41.56`	`*** part/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net)`
`18:42.17`	`asdx`	`i want one for getting started with asterisk`
`18:42.26`	`asdx`	`testing purposes, etc`
`18:42.52`	`*** join/#asterisk thekidrio (n=thekidri@66.107.42.13)`
`18:42.53`	`styelz`	`you can get cheaper yeastar ones`
`18:43.14`	`[TK]D-Fender`	`asdx: Why get one for TESTING? What do you want to use when you're finished?`
`18:43.29`	`*** join/#asterisk moprilo (n=jjohn@201.192.35.138)`
`18:43.49`	`asdx`	`[TK]D-Fender: well, i want to learn how to set up things like pstn <-> voip, and things like that`
`18:43.59`	`[TK]D-Fender`	`asdx: Waste of money then....`
`18:44.16`	`asdx`	`hm, why?`
`18:44.19`	`[TK]D-Fender`	`asdx: A call is a call is a call... setting up a TDM card is not big deal.`
`18:44.39`	`moprilo`	`hi guys.. i just installed a new network nat/gw (pfsense,freebsd), and now my sjphone is not working. (worked before, and works from other networks.) i get "process_sdp: Insufficient information for SDP (m = '', c = '')"`
`18:45.05`	`[TK]D-Fender`	`asdx: The difference between setting up an ITSP vs a local SIP gateway, vs an analog card, vs a PRI is pretty irrelevant.`
`18:45.07`	`karleeto`	`_x86_: polycom wont let you download the new versions.. i already got sip 2.2.0, my vendor's site did let me download that, its just hanging on the bootrom download`
`18:45.45`	`asdx`	`well, i have a lot to learn then`
`18:45.47`	`asdx`	`i guess`
`18:46.12`	`moprilo`	`karleeto, i've done lots of polycom installs, what's the problem? wasn't here before.`
`18:46.33`	`*** join/#asterisk hrmphh (i=patrick@notchill.com)`
`18:46.56`	`karleeto`	`moprilo: no problem, just my vendors download site is crappy, and i need the 4.0.0 bootrom file`
`18:47.20`	`moprilo`	`i don't have the site now,.. but i'm sure you can find download sites on google`
`18:47.29`	`[TK]D-Fender`	`karleeto: TECHNICALLLY... you don't.... SIP 2.2.0 doesn't require it :)`
`18:47.36`	`thekidrio`	`anyone here mess with aastra phones?`
`18:47.45`	`hrmphh`	`what is the preferred way to integrate hylafax with asterisk on T1 PRI?`
`18:47.49`	`karleeto`	`[TK]D-Fender: yeah, i'm using 3.2.3 with 2.2.0 right now`
`18:47.55`	`[TK]D-Fender`	`thekidrio: Yeah... they're a mixed bag and have only 1 use.`
`18:48.01`	`asdx`	`[TK]D-Fender: what do you suggest for starting`
`18:48.01`	`thekidrio`	`I am having an issue setting the time with my 9112i, http://pastebin.ca/722059`
`18:48.15`	`[TK]D-Fender`	`asdx: What do you actually want to do?`
`18:48.16`	`*** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)`
`18:48.28`	`_x86_`	`karleeto: ah, that site only goes up to 3.2.x`
`18:48.32`	`_x86_`	`karleeto: sorry`
`18:49.03`	`asdx`	`[TK]D-Fender: i would like to play with pure voip first and then start to attach things like ip-phones and play with the pstn too`
`18:49.45`	`_x86_`	`any idea why when I call any extension that tries to play a file, the playback executes but no audio is heard? polycom IP501 <--> Asterisk 1.2.21.1 with no NAT`
`18:49.57`	`[TK]D-Fender`	`asdx: so "play" basically? I mean if buying hardware you arent seriously planning on using is your idea of fun, then by all means...`
`18:50.01`	`_x86_`	`asdx: get an X100P`
`18:50.26`	`[TK]D-Fender`	`_x86_: Actaully.... not a bad idea for him. At least he'll be wasting LESS money...`
`18:50.33`	`hrmphh`	`should i be using IAXmodem?`
`18:50.42`	`[TK]D-Fender`	`_x86_: And it should be the same approximate learning experience.`
`18:50.44`	`asdx`	`[TK]D-Fender: i will use it`
`18:50.52`	`asdx`	`[TK]D-Fender: but i want to start small (playing) first`
`18:50.54`	`[TK]D-Fender`	`hrmphh: That's typically considered the best way`
`18:51.02`	`_x86_`	`[TK]D-Fender: any idea why i can't hear when i call voicemail? (i dont hear the login prompts / etc)`
`18:51.21`	`[TK]D-Fender`	`_x86_: firewall / NAT`
`18:51.48`	`asdx`	`[TK]D-Fender: i would like to go work with asterisk in the future`
`18:53.38`	`_x86_`	`[TK]D-Fender: no and no`
`18:53.58`	`_x86_`	`[TK]D-Fender: i'm going to the same switch, no firewall on the asterisk box, and definitely no NAT`
`18:55.00`	`[TK]D-Fender`	`_x86_: guess you'd have to show some CLI output....`
`18:55.17`	`_x86_`	`[TK]D-Fender: Polycom is using the G.711u codec, sound files are encoded as GSM, but that shouldn't matter`
`18:55.43`	`_x86_`	`[TK]D-Fender: the CLI output is rather useless`
`18:55.50`	`_x86_`	`[2007-10-01 13:55:27] -- Executing VoiceMailMain("SIP/7099-082855c0", "7099") in new stack`
`18:55.53`	`_x86_`	`[2007-10-01 13:55:27] -- Playing 'vm-login' (language 'en')`
`18:56.12`	`[TK]D-Fender`	`_x86_: have you tried ANSWER first?`
`18:58.20`	`*** join/#asterisk Strom_M (n=strom@208.127.172.112)`
`18:58.25`	`_x86_`	`hmm, i'll try that`
`18:59.10`	`_x86_`	`[TK]D-Fender: no go`
`18:59.26`	`[TK]D-Fender`	`_x86_: Would have to see SIP debugs, etc...`
`19:00.12`	`_x86_`	`[TK]D-Fender: wierd thing is, I know it's getting my DTMF, because i put in a bunch of crap, and it tries to play vm-incorrect.gsm`
`19:00.33`	`hrmphh`	`hrm`
`19:00.34`	`[TK]D-Fender`	`_x86_: and that can be OOB....`
`19:00.34`	`hrmphh`	`which is better`
`19:00.39`	`hrmphh`	`hylafax or asterfax?`
`19:00.45`	`[TK]D-Fender`	`_x86_: which does not say much for RTP...`
`19:00.46`	`_x86_`	`and i can call other extensions on the same asterisk box and get two-way audio no problem`
`19:00.54`	`_x86_`	`even outside that asterisk box...`
`19:01.05`	`_x86_`	`[TK]D-Fender: i'm doing G.711u with inband dtmf`
`19:01.09`	`[TK]D-Fender`	`_x86_: And none of that commentary means anything....`
`19:01.24`	`[TK]D-Fender`	`_x86_: INBAND means it MUST be getting audio.`
`19:01.30`	`[TK]D-Fender`	`_x86_: And you'd KNOW IT.`
`19:01.34`	`_x86_`	`[TK]D-Fender: exactly`
`19:01.40`	`_x86_`	`[TK]D-Fender: i think the audio path is fine`
`19:01.47`	`[TK]D-Fender`	`_x86_: Which means if it isn;t making it to the other side thats a problem on the * server side and screams networking.`
`19:02.05`	`_x86_`	`[TK]D-Fender: i think there is something fubar in the codec translation stuff`
`19:02.32`	`_x86_`	`[TK]D-Fender: DTMF makes it both ways`
`19:02.46`	`hrmphh`	`i was thinking hyla`
`19:02.48`	`_x86_`	`which is inband, so the audio path is good both ways`
`19:02.52`	`hrmphh`	`but people on sangoma wiki are saying go w/aster`
`19:03.28`	`*** join/#asterisk afrosheen (n=cj@207.71.49.137)`
`19:03.43`	`_x86_`	`[TK]D-Fender: just for grins, i'm going to use sox to transcode one a random file, and make a phantom extension that just plays back the new ulaw file`
`19:03.44`	`afrosheen`	`what's up gang`
`19:04.52`	`afrosheen`	`I'm using hyperic to monitor our asterisk server and the asterisk process itself. However, when someone remotes in and runs asterisk -r, hyperic says the asterisk process has died. Any reason for this? Does the asterisk PID change when you asterisk -r?`
`19:06.05`	`[TK]D-Fender`	`hrmphh: asterfax uses SpanDSP +rxfax/txfax which is considered less reliable than IAXmodem + Hylafax`
`19:06.50`	`_x86_`	`[TK]D-Fender: interesting... i can't hear the ulaw file either`
`19:10.31`	`_x86_`	`[TK]D-Fender: what else would you recommend?`
`19:10.49`	`[TK]D-Fender`	`_x86_: I'd have to pick apart every bit of your setup myself to say...`
`19:11.00`	`[TK]D-Fender`	`_x86_: and I just odn't have the time for that right now.`
`19:13.02`	`hrmphh`	`cool`
`19:14.09`	`hrmphh`	`iaxmodem is annoying me as its complaining about not being able to find libiax.so.0 yet its blatantly in /usr/local/lib/libiax.so.0`
`19:14.12`	`*** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144)`
`19:14.13`	`hrmphh`	`prob need to check env settings mutter`
`19:15.17`	`jcanfield`	`I just got an IAX2 test trunk from a local provider, but incoming connection is being rejected. The channel appears to be registered fine. What should I be looking for?`
`19:15.56`	`[TK]D-Fender`	`hrmphh: ldconfig <-`
`19:16.16`	`hrmphh`	`woot thanks`
`19:19.00`	`hrmphh`	`erm thats annoying, iax is binding to port 4569`
`19:19.15`	`hrmphh`	`even tho i tell it port=4671, just seems to ignore`
`19:20.01`	`file`	`it's bindport, not port`
`19:20.04`	`hrmphh`	`ah`
`19:20.06`	`hrmphh`	`sangoma example is wrong`
`19:22.12`	`hrmphh`	`anyone have a sample iax.conf i can look at?`
`19:23.53`	`hrmphh`	`[Oct 1 12:23:48] NOTICE[30582]: iax2-provision.c:518 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.`
`19:24.00`	`hrmphh`	`im getting that with this iax.conf file:`
`19:24.25`	`hrmphh`	`http://pastebin.com/mf7e06eb`
`19:25.00`	`*** join/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu)`
`19:25.07`	`[TK]D-Fender`	`hrmphh: that SHOULD be "port=4671", and "bindaddr" is just wrong...`
`19:25.24`	`[TK]D-Fender`	`hrmphh: should say "host="`
`19:25.25`	`hrmphh`	`12:00 <@file> it's bindport, not port`
`19:25.26`	`*** part/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu)`
`19:25.35`	`[TK]D-Fender`	`hrmphh: "thats nice"......`
`19:25.39`	`hrmphh`	`lol`
`19:25.46`	`hrmphh`	`ok`
`19:25.47`	`[TK]D-Fender`	`hrmphh: But in the context you have jsut shown us... NO.`
`19:25.53`	`hrmphh`	`still getting the error`
`19:25.59`	`hrmphh`	`after removed bindaddr and changed to port`
`19:26.22`	`[TK]D-Fender`	`hrmphh: You have no [general] section either... NOT good.`
`19:26.30`	`[TK]D-Fender`	`hrmphh: And clearly a bad thing`
`19:26.34`	`hrmphh`	`heh this is what i get for following incomplete sangoma docs`
`19:26.38`	`hrmphh`	`can you suggest a good doc to follow?`
`19:26.49`	`[TK]D-Fender`	`hrmphh: they show you the PIECE you need to add.`
`19:26.58`	`hrmphh`	`yeah`
`19:27.03`	`hrmphh`	`i had no prior IAX config`
`19:27.06`	`hrmphh`	`not using at all heretofore`
`19:27.14`	`*** join/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu)`
`19:27.19`	`[TK]D-Fender`	`hrmphh: You don't get a car stereo system with a COMPLETE car manual including instructions on how to change your batter as well do you?`
`19:27.30`	`[TK]D-Fender`	`battery*`
`19:27.40`	*** join/#asterisk Yourname` (n=Miranda@unaffiliated/yourname/x-837320)
`19:27.43`	`*** part/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu)`
`19:27.44`	`[TK]D-Fender`	`hrmphh: Don't become a cut&paste coder!`
`19:27.53`	`hrmphh`	`heh`
`19:28.38`	Yourname`	`Hi. Sometimes when I make a call from Sipura SPA941.. the called party's phone starts ringing, but then, all of a sudden, the called party picks up.. AND there is also some ringing going.`
`19:28.46`	Yourname`	`How can I rectify whatever is happening?`
`19:29.04`	`[TK]D-Fender`	Yourname`: Without watching the call in progress...... SUFFER.
`19:29.33`	`hrmphh`	`and ditch the 941s`
`19:29.35`	`hrmphh`	`best move we made`
`19:29.36`	Yourname`	`haha oops`
`19:29.39`	Yourname`	`Forgot to do that`
`19:29.40`	`hrmphh`	`go w/polycom 330s`
`19:31.10`	Yourname`	`Amendo-polis.`
`19:31.15`	Yourname`	`Onenna these days.`
`19:31.32`	`*** join/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu)`
`19:32.10`	`jcanfield`	`shouldn't I see something after the @ on an incoming IAX call? http://www.pastebin.org/3874`
`19:33.02`	`[TK]D-Fender`	`jcanfield: only if the DIAL had a context... which it clearly DOESN'T`
`19:33.45`	`jcanfield`	`[TK]D-Fender: doh! :)`
`19:35.06`	`*** join/#asterisk mltlnx (n=mltlnx@cpe-68-174-159-251.nyc.res.rr.com)`
`19:35.56`	`*** join/#asterisk jsmith (n=jsmith@000-222-951.area3.spcsdns.net)`
`19:35.56`	`*** mode/#asterisk [+o jsmith] by ChanServ`
`19:36.31`	`*** join/#asterisk Kurin- (n=Kurin@lithium.delete.org)`
`19:36.47`	`*** join/#asterisk FinboySlick (n=Miranda@207.134.8.4)`
`19:37.15`	`Kurin-`	`What could cause echo between a VoIP phone and a remote phone if it only happens when the remote phone calls the VoIP phone?`
`19:37.26`	`Kurin-`	`this would be echo at the VoIP end`
`19:37.56`	`akaast47`	`[TK]D-Fender: Can I use a TDM11B card for meetme timing?`
`19:38.07`	`[TK]D-Fender`	`akaast47: Sure.`
`19:38.15`	`[TK]D-Fender`	`akaast47: Don't actually need any modules though`
`19:38.27`	`afrosheen`	`[TK]D-Fender, I'm in a bad mood too, big diesel on the highway this morning slung pieces of a blown tire into the bumper on my 3 week old car, cut up the paint :(`
`19:38.38`	`akaast47`	`[TK]D-Fender: What I would need then?`
`19:38.56`	`[TK]D-Fender`	`Kurin-: that would be acoustic echo. What phones?`
`19:39.10`	`[TK]D-Fender`	`akaast47: You could use a TDM400P without any modules...`
`19:39.17`	`hrmphh`	`yuck`
`19:39.23`	`Kurin-`	`I don't know what the remote end has. We're using polycom 550s here`
`19:39.29`	`hrmphh`	`tdm400p=poop`
`19:39.36`	`akaast47`	`[TK]D-Fender: Digium sells with modules. Where I can get without?`
`19:39.40`	`[TK]D-Fender`	`hrmphh: You missed his REASON for them.`
`19:39.40`	`*** join/#asterisk blq (n=Bl@dslb-088-064-134-161.pools.arcor-ip.net)`
`19:39.40`	`*** join/#asterisk zippytech (n=ron@244.zippytech.com)`
`19:39.57`	`[TK]D-Fender`	`akaast47: another store. telephonydepot.com , voipsupply.com , etc`
`19:40.07`	`afrosheen`	`voipsupply has been good for us`
`19:40.19`	`zippytech`	`any one know is the zaptel-echo-octasic modules has to be the same as zaptel drivers`
`19:40.19`	`akaast47`	`[TK]D-Fender: That means I don't need to connect another analog line to use the timer?`
`19:40.20`	`zippytech`	`veriosn`
`19:40.28`	`[TK]D-Fender`	`akaast47: Correct`
`19:41.01`	`FinboySlick`	`Yo gang. Quickie question about an update to 1.4: How do I tell VoiceMailMain not to supply its first param? Here's what the console gives me: VoiceMailMain("SIP/504-00812630", "504@intenral\|s") However, I only supply the part that comes after the coma.`
`19:41.07`	`afrosheen`	`zippytech, what are you using that needs the octasic module?`
`19:41.18`	`zippytech`	`for softecho`
`19:41.37`	`afrosheen`	`zippytech, oh, I was under the impression that octasic was hardware-based`
`19:41.41`	`hrmphh`	`tk;`
`19:41.43`	`hrmphh`	`theres no good reason for one :)`
`19:41.53`	`hrmphh`	`i dont care if its to wedge the door open`
`19:42.01`	`afrosheen`	`hrmphh, oh, you've been bitten by the tdm400 also huh`
`19:42.06`	`hrmphh`	`yeah`
`19:42.14`	`zippytech`	`i have a box with really bad echo tried the basics and then had to purchase an softech licence`
`19:42.20`	`afrosheen`	`we've had 2 and both of them sucked, modules regularly died`
`19:42.24`	`FinboySlick`	`[TK]D-Fender: Oh, and by the way: Yo.`
`19:42.42`	`zippytech`	`fixed the problem but did an update to the box and lost the softecho`
`19:42.43`	`lirakis`	`YES! .. my new ATFOT 2nd ed is here!`
`19:42.46`	`[TK]D-Fender`	`FinboySlick: y0`
`19:43.02`	`[TK]D-Fender`	`lirakis: Great.... now you can :`
`19:43.07`	`[TK]D-Fender`	`~osmosis`
`19:43.08`	`jbot`	`from memory, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ...`
`19:43.09`	`[TK]D-Fender`	`:p`
`19:43.16`	`afrosheen`	`so, do any experts know if the asterisk PID changes when you open a remote asterisk cli?`
`19:44.14`	`[TK]D-Fender`	`FinboySlick: ummmmm reword that please I'm thinking you might be silly...`
`19:44.37`	`lirakis`	`[TK]D-Fender: i learn through practice.. not osmosis... ill work on setting up a new 1.4 system and learn more as i play.... thats not to say i wont keep the book under my pillow at night.. who knows .. maybe osmosis will occur then`
`19:44.49`	`lirakis`	`:P`
`19:45.05`	`hrmphh`	`<PROTECTED>`
`19:45.07`	`hrmphh`	`is that normal?`
`19:45.17`	`hrmphh`	`for asterisk to cycle through all the b chans and restart them?`
`19:46.06`	`afrosheen`	`yeah`
`19:46.13`	`afrosheen`	`don't worry about that, it won't restart channels that are active`
`19:46.17`	`FinboySlick`	`[TK]D-Fender: You're too polite. What I'm being is ignorant ;) Anyway, in my dialplan, I call: VoiceMailMain("504@internal\|s") and I get VoiceMailMain("SIP/504-00812630", "504@intenral\|s") in the log. Where does the part before the coma comes from?`
`19:47.44`	`[TK]D-Fender`	`FinboySlick: "SIP/504-00812630"? that is your current CHANNEL. this is not YOU passing anything...`
`19:49.16`	`FinboySlick`	`[TK]D-Fender: You saying it'll only take into account the part after the coma? 'cause things aren't working (note that there's a couple other spots where I could have an error, investigating those right now.`
`19:49.52`	`[TK]D-Fender`	`FinboySlick: Yes, you are barking up the wrong tree... pastebin your actual complete call CLI and tell us where the problem occurs in it.`
`19:50.26`	`FinboySlick`	`[TK]D-Fender: Always so polite. Well, if that isn't an issue, I'm pretty sure I know how to fix it already ;)`
`19:51.06`	`hrmphh`	`ok im officially retarded`
`19:51.19`	`[TK]D-Fender`	`FinboySlick: ..... you're welcome... I think.... O_O`
`19:51.20`	`hrmphh`	`can someone please pastebin a working iax.conf file`
`19:51.35`	`[TK]D-Fender`	`hrmphh: look at the sample * came with.`
`19:51.57`	`hrmphh`	`any where to find locally without installing samples?`
`19:52.02`	`hrmphh`	`in /usr/share or something?`
`19:52.18`	`hrmphh`	`woot /usr/local/src/asterisk-1.4.4/configs/iax.conf.sample`
`19:53.14`	`afrosheen`	`find? lol, do you not have locate`
`19:53.23`	`hrmphh`	`find is more brute force :P`
`19:53.29`	`hrmphh`	`i cant be bothered to update indices`
`19:53.38`	`afrosheen`	`neither can a cron job evidently :)`
`19:53.42`	`hrmphh`	`nope`
`19:53.59`	`afrosheen`	`find and grind, locate = no wait`
`19:54.11`	`FinboySlick`	`[TK]D-Fender: See, I fixedated it :)`
`19:54.22`	`[TK]D-Fender`	`FinboySlick: Okerator.`
`19:55.07`	`FinboySlick`	`[TK]D-Fender: Now if I could get the %!"$ thing to hear my fax tones.`
`19:57.39`	`*** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net)`
`19:57.46`	`FinboySlick`	`[TK]D-Fender: I'm begining to think it's the fax itself that isn't working right.`
`19:58.03`	`moprilo`	`I have this weird thing happening, that between 2 peers (that work fine), if i put the (receiving peer) host=dynamic, asterisk[receiving peer] returns (No compatible codecs, not accepting this offer) when calling. why could this be?`
`19:58.11`	`[TK]D-Fender`	`FinboySlick: hope you're not expecting an opinion on that right now....`
`19:58.34`	`[TK]D-Fender`	`moprilo: PASTEBIN is your friend...`
`19:59.22`	`syzygyBSD`	`no he isn't, Pastebin slept with my woman, a real friend wouldn't do that`
`20:00.04`	`[TK]D-Fender`	`syzygyBSD: I said PASTEBIN what HIS friend, you have nothing to do with this :)`
`20:00.25`	`syzygyBSD`	`touche`
`20:01.04`	`akaast47`	`[TK]D-Fender: why is needed the timming in the meetme conference?`
`20:01.06`	`FinboySlick`	`[TK]D-Fender: No need for an opinion ;) I was right apparently. Just had another office call us and it works`
`20:02.01`	`[TK]D-Fender`	`akaast47: Because * gets voice data from multiple channels and has to time-synch and MIX them.`
`20:02.13`	`[TK]D-Fender`	`FinboySlick: You're welcome.`
`20:02.46`	`moprilo`	`i'll do the pastebin`
`20:02.47`	`akaast47`	`[TK]D-Fender: and the hardware is doing better timing than ztdummy?`
`20:03.17`	`afrosheen`	`akaast47, it's supposed to`
`20:03.18`	`hrmphh`	`this is absurd`
`20:03.23`	`hrmphh`	`ive followed the sample iax.conf`
`20:03.26`	`*** part/#asterisk jbroome (n=jbroome@unaffiliated/jbroome)`
`20:03.30`	`hrmphh`	`and still getting this f'in provisioning message`
`20:04.07`	`hrmphh`	`hmm didnt see iaxprov.conf :)`
`20:04.09`	`akaast47`	`afrosheen: :)`
`20:04.11`	`afrosheen`	`akaast47, which do you trust more, a Rolex wristwatch or an autistic guy counting in his head?`
`20:04.21`	`hrmphh`	`lol`
`20:04.31`	`*** part/#asterisk juliux (n=juliux@ubuntu/member/juliux)`
`20:04.35`	`akaast47`	`afrosheen: :D`
`20:04.35`	`afrosheen`	`that's the difference between software and hardware timing for meetme and other stuff`
`20:04.42`	`Nugget`	`neither, really. a rolex is jewelry, not an accurate timepiece. :)`
`20:04.59`	`afrosheen`	`Nugget, I see your sharp razor, and raise you a split hair :)`
`20:05.10`	`Nugget`	`close shave.`
`20:05.22`	`akaast47`	`afrosheen: Then definitly I need hardware otherwise I will be limited to 10 user in one meetme conference`
`20:05.35`	`afrosheen`	`akaast47, yeah, just get an unpopulated TDM card`
`20:05.42`	`hrmphh`	`udp 0 0 0.0.0.0:4571 0.0.0.0:*`
`20:05.45`	`hrmphh`	`wee at least its listening`
`20:05.57`	`afrosheen`	`hrmphh, I respect your determination, iax is always fun`
`20:06.19`	`akaast47`	`everywhere I see at least 1 module fxs or fxo. Maybe one fxo doesn't hurt to have`
`20:07.42`	`hrmphh`	`hmm wont even let me turn on debugging`
`20:07.44`	`hrmphh`	`for iax`
`20:08.19`	`hrmphh`	`guess i could tcpdump the udp port`
`20:08.20`	`hrmphh`	`mutter`
`20:09.27`	`moprilo`	`guys..: http://pastebin.com/m4478855d`
`20:10.15`	`moprilo`	`the thing is a get an error if a change the type=peer to type=user, or if i put my host=dynamic instead of the ip`
`20:10.57`	`moprilo`	`a=> I`
`20:11.06`	`hrmphh`	`tiax2 set debug`
`20:11.08`	`hrmphh`	`is useless`
`20:11.11`	`hrmphh`	`err -t`
`20:12.03`	`*** join/#asterisk uwe (n=uwe@athedsl-306148.home.otenet.gr)`
`20:13.23`	`akaast47`	`Where I can get a gsm card or gateway for asterisk?`
`20:15.35`	`[TK]D-Fender`	`moprilo: pastebin the ENTIRE call with sip debug enabled please...`
`20:16.32`	`hrmphh`	`$`
`20:16.38`	`hrmphh`	`just needed a full blown restart of asterisk`
`20:16.43`	`hrmphh`	`iax2 reload wouldnt cut it`
`20:16.48`	`hrmphh`	`or reload chan_iax2.so`
`20:19.12`	`*** join/#asterisk mltlnx (n=mltlnx@static-70-107-237-66.ny325.east.verizon.net)`
`20:21.10`	`*** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl)`
`20:21.52`	`stimpie`	`does someone know how to remove the subscribe option from the allow list in an invite message coming from asterisk`
`20:22.12`	`*** join/#asterisk Mavvie (n=edwin@ppp121-44-20-238.lns10.syd7.internode.on.net)`
`20:22.28`	`hrmphh`	`hrm you guys recommend hylafax from src or from debian apt package?`
`20:23.09`	`[TK]D-Fender`	`hrmphh: if you did * from src, then from src it is...`
`20:23.13`	`hrmphh`	`k`
`20:24.16`	`*** join/#asterisk tsurko (n=tsurko@78.90.100.214)`
`20:27.04`	`*** join/#asterisk \|omni\| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)`
`20:27.19`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`20:27.28`	`moprilo`	`can i have a peer (for testing reasons) with no username, and no specific ip?`
`20:27.40`	`moprilo`	`i mean a user.. (me receiving the calls)`
`20:27.57`	`moprilo`	`or at least an ip range.. (if no dynamic)`
`20:30.23`	`[TK]D-Fender`	`BBIAB`
`20:30.59`	`*** join/#asterisk tsurko (n=tsurko@78.90.100.214)`
`20:31.21`	`*** join/#asterisk [hC] (n=hardcore@66.119.167.163)`
`20:31.38`	`stimpie`	`moprilo: you should be able to use the permit,deny,mask rules but not sure if that works without a password/host`
`20:32.21`	`moprilo`	`true.. thanks for the tip, but i think that-s my problem.. (why asterisk doesn't receive calls from an undentified user)`
`20:32.46`	`stimpie`	`strange ;-)`
`20:33.14`	`*** join/#asterisk tsurko (n=tsurko@78.90.100.214)`
`20:33.26`	`stimpie`	`an option would be to have an sip proxy on a specific ip to forward all messages to asterisk`
`20:36.07`	`akaast47`	`moprilo:domain=<domain>[,<context>]`
`20:37.38`	`DeeJayTwo`	`strangely...even when setting allow=ulaw:10 in sip.conf, we get a ptime of 20 ...`
`20:37.45`	`DeeJayTwo`	`what could lead to this?`
`20:37.56`	`DeeJayTwo`	`This same server offers a ptime of 20 when sending an INVITE..`
`20:38.09`	`DeeJayTwo`	`on 1.4.3`
`20:38.10`	`DeeJayTwo`	`err..`
`20:38.12`	`DeeJayTwo`	`1.4.2`
`20:41.37`	`*** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net)`
`20:42.11`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`20:43.08`	`*** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com)`
`20:47.09`	`lirakis`	`later all`
`20:47.12`	`*** part/#asterisk lirakis (n=lirakis@65.200.191.253)`
`20:47.32`	`*** join/#asterisk blq (n=Bl@dslb-088-067-028-153.pools.arcor-ip.net)`
`20:51.04`	`*** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)`
`20:52.02`	`DeeJayTwo`	`is ptime something esoteric nobody is playing with?!`
`20:52.05`	`DeeJayTwo`	`;)`
`20:52.08`	`*** join/#asterisk shinao1 (n=shinao1@217.20.242.52)`
`20:52.18`	`DeeJayTwo`	`Each time I put "ptime" in my question, I get no answer ;)`
`20:52.49`	`*** join/#asterisk irq (n=dan@ip72-197-39-50.sd.sd.cox.net)`
`20:53.06`	`FireMac`	`hi can someone help me pls`
`20:53.16`	`irq`	`how can i make um, my outbound trunk, when a call goes to it, to start at the last line of the trunk and not the first?`
`20:53.22`	`FireMac`	`i couldnt get my voip service and asterisk work`
`20:53.28`	`FireMac`	`what configurations should i configure in order asterisk control my incoming and outgoing calls to my voip service?`
`20:55.50`	`Sci_05`	`FireMac: your voip provider should be able to help with that. SHould just be the iax/sip.conf and your extensions.conf files`
`20:56.30`	`FireMac`	`(Sci_05): what are the information should i ask and get from my provider?`
`20:56.53`	`Sci_05`	`ask them what needs to be setup on your box`
`20:57.37`	`irq`	`please`
`20:57.38`	`irq`	`seriously`
`20:57.39`	`irq`	`anyone`
`20:57.42`	`irq`	`hello?`
`20:57.44`	`FireMac`	`is it ok to tell them that i am running asterisk`
`20:57.53`	`FireMac`	`and that i want to set it up?`
`20:57.54`	`irq`	`i'm sending phone calls out a 5 line port group`
`20:57.56`	`irq`	`1,2,3,4,5`
`20:58.06`	`irq`	`i want outgoing calls to be sent to 5 first, then 4, etc`
`20:58.08`	`akaast47`	`Can tell somebody what is the max phone calls in one asterisk box?`
`20:58.11`	`irq`	`because new calls come in on 1,2,3 in that order`
`20:58.13`	`Sci_05`	`I don't see why not, they really shouldn't care`
`20:58.28`	`hrmphh`	`grr hylafax is being annoying on debian`
`20:58.40`	`irq`	`k`
`20:58.41`	`irq`	`g1`
`20:58.44`	`irq`	`instead of G1`
`20:58.44`	`hrmphh`	`wonder what i need to symlink to /usr/share/gs-esp/fonts`
`20:59.28`	`FireMac`	`(Sci_05): thanks, ill try to contact them and see`
`21:04.01`	`akaast47`	`FireMac: you need phonenumber,auth user (usually phonenumber), password, host (sip server, realm), eventually port and outbound proxy ,`
`21:06.07`	`*** part/#asterisk akaast47 (i=0ca5bc82@gateway/web/cgi-irc/ircatwork.com/x-f63a896508f0fd2b)`
`21:06.17`	`*** join/#asterisk akaast47 (i=0ca5bc82@gateway/web/cgi-irc/ircatwork.com/x-f63a896508f0fd2b)`
`21:07.13`	`akaast47`	`Can tell somebody what is the max phone calls in one asterisk box?`
`21:07.50`	`_x86_`	`that's a limitation of your specific hardware`
`21:07.57`	`thekidrio`	`akaast47, there is no max, you could theoretically cluster a bunch of boxes for unlimited scalablilty`
`21:08.07`	`thekidrio`	`not just hardware eh x86, but your pipe too`
`21:08.23`	`FireMac`	`(akaast47): thanks, from the host side is there anything else aside from sip server and realm that i should ask from my provider?`
`21:08.26`	`_x86_`	`thekidrio: i include that in hardware, as it's not software ;)`
`21:08.30`	`errr`	`I have a small task I am trying to make more generic.. http://fluxbox.pastebin.ca/722222 everwhere I have 231 in that paste I would like to make it generic so I dont have to add the same lines for every user.. is that possible??`
`21:08.35`	`keulin`	`is somebody able to explain me what is the DID ?`
`21:08.40`	`thekidrio`	`x86, hehe makes sense :)`
`21:09.04`	`thekidrio`	`DiD is a direct inward dial number`
`21:09.13`	`thekidrio`	`like 18182557405`
`21:09.28`	`thekidrio`	`or 16029318686`
`21:09.32`	`akaast47`	`<FireMac> : probably you should ask about outbound proxy too for outgoing calls`
`21:10.29`	`errr`	`never mind, I figured it out`
`21:11.11`	`keulin`	`thekidrio, is it the number to type to directly phoning something in a company ?`
`21:11.28`	`FireMac`	`(Sci_05): well, my provider told me that they dont support asterisk.`
`21:11.52`	`FireMac`	`(Sci_05): i have access on my ata, is it possible to get this informations from the ata?`
`21:12.42`	`thekidrio`	`keulin, yes as in not an extension, but something that directly dials into an extenstion/IVR/group`
`21:13.39`	`keulin`	`thekidrio, I think in france we tell a SDA (direct sequence on arrival)`
`21:15.21`	`keulin`	`direct inward dialling, DID, direct dialling-in`
`21:15.39`	`keulin`	`I understand`
`21:15.44`	`keulin`	`all is right`
`21:15.49`	`keulin`	`thanks the_lalelu`
`21:15.52`	`keulin`	`oupsss`
`21:15.56`	`thekidrio`	`hehe`
`21:16.01`	`keulin`	`thanks you thekidrio`
`21:16.04`	`keulin`	`sorry`
`21:16.18`	`*** join/#asterisk remedex (n=none@192.217.255.35)`
`21:20.50`	`akaast47`	`thekidrio: I have dualcore hyperthreaded Intel(R) Xeon(TM) CPU 3.20GHz with 2 GB memory`
`21:22.34`	`akaast47`	`So what would be the max calls for this pc?`
`21:23.06`	`akaast47`	`the lan has 100mb`
`21:24.13`	`_x86_`	`heh`
`21:24.55`	`_x86_`	`akaast47: it has a lot more to do with motherboard, south bridge, and north bridge`
`21:25.14`	`_x86_`	`akaast47: motherboard's ability to bang out IRQ's like crazy`
`21:25.37`	`_x86_`	`akaast47: also, SIP calls? IAX calls? H323 calls? Zaptel calls? the technology makes a huge difference`
`21:26.03`	`akaast47`	`only SIP calls and meetme conference`
`21:26.09`	`remedex`	`transcoding?`
`21:26.22`	`akaast47`	`ulaw only`
`21:26.52`	`remedex`	`bandwith?`
`21:27.09`	`thekidrio`	`akaast47, http://www.fonality.com/pop.cgi?page=which_server_is_right_for_me.tt that will give you a VERY rough guideline`
`21:27.22`	`moprilo`	`[TK]D-Fende: could you take a look at this.. http://pastebin.com/m3e2199a2 i get that on my receving side. (i have my receiving side with no host ip, nor username.. but i need it to work like that.).. ) the sending side just gets congestion`
`21:27.51`	`remedex`	`tata * peeps`
`21:28.02`	`*** part/#asterisk remedex (n=none@192.217.255.35)`
`21:29.52`	`akaast47`	`thekidrio: thanks`
`21:33.34`	`akaast47`	`moprilo: no rule configured in you receiving dialplan for 5062331122 that's why you get 404`
`21:34.15`	`*** part/#asterisk akaast47 (i=0ca5bc82@gateway/web/cgi-irc/ircatwork.com/x-f63a896508f0fd2b)`
`21:40.29`	`*** join/#asterisk JasonK1909 (n=JasonK19@12.18.111.2)`
`21:41.41`	`hrmphh`	`do you need vgetty+egetty installed for hylafax to work correct?`
`21:44.02`	`*** join/#asterisk jtodd (n=jtodd@gw0.office1.talkplus.com)`
`21:45.40`	`JasonK1909`	`those users.conf changes earlier didn't fix the problem`
`21:48.02`	`JasonK1909`	`anyone else got tips for fixing registration problems w authentication?`
`21:50.25`	`*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)`
`21:51.03`	`syzygyBSD`	`JasonK1909: use the right username/password?`
`21:51.40`	`moprilo`	`disallow=all`
`21:51.41`	`moprilo`	`allow=g729`
`21:51.41`	`moprilo`	`allow=g723.1`
`21:51.47`	`moprilo`	`<PROTECTED>`
`21:51.52`	`moprilo`	`disallow=all`
`21:51.52`	`moprilo`	`allow=g729`
`21:51.52`	`moprilo`	`allow=g723.1`
`21:51.56`	`moprilo`	`sorry sorry`
`21:52.11`	`moprilo`	`akaast47:.. thanks`
`21:52.15`	`[TK]D-Fender`	`moprilo, feel free to pastebin that failed call I asked for.`
`21:52.27`	`[TK]D-Fender`	`moprilo, And the configs to match`
`21:52.32`	`_x86_`	`hmm`
`21:52.42`	`JasonK1909`	`ya, just using user 6001 / secret 6001 to get it up & running ... couple guys here earlier gave tips for simplifying my users.conf but to no avail`
`21:52.48`	`moprilo`	`i've realized since if the incomming peer didn't have an IP nor username, it would go to the default context.. that was the problem`
`21:53.18`	`moprilo`	`thanks for the help.. n pacience on this stupid issue :S`
`21:53.28`	`JasonK1909`	`just bizarre... identically configured Digium aa50 devices, phone connects to one and not the other...`
`21:53.42`	`syzygyBSD`	`JasonK1909: is it just me, or what the hell is users.conf for?`
`21:55.30`	`Toerkeium`	`guys, does trixbox works for a small business with 30 pure voip lines?`
`21:55.59`	`syzygyBSD`	`Toerkeium: depends on how you define "works"`
`21:56.20`	`Toerkeium`	`syzygyBSD: will I have to reboot asterisk every 5 minutes? :)`
`21:56.30`	`[TK]D-Fender`	`Toerkeium, Yes it works. The questio is, do you WANT IT.`
`21:56.38`	`russellb`	`and then answer is, "no"`
`21:56.39`	`russellb`	`:-p`
`21:56.42`	`[TK]D-Fender`	`:D`
`21:56.43`	`syzygyBSD`	`I don't know about trixbox, I know my asterisk install has been up for many months`
`21:56.44`	`Toerkeium`	`heh`
`21:57.00`	`Toerkeium`	`need to do a quick install..`
`21:57.45`	`_Sam--`	`[TK]D-Fender : 1.4.9 just crashed`
`21:57.50`	`syzygyBSD`	`heh, quick installs take three times longer than if you just did it right in the first place`
`21:57.51`	`_Sam--`	`System uptime: 2 seconds`
`21:57.55`	`_Sam--`	`:/`
`21:58.21`	`Toerkeium`	`better go with asterisk and do manual setup?`
`21:58.38`	`_Sam--`	`my 1.2.5 ran for 7 months before :/`
`21:58.53`	`_Sam--`	`1.4.9 got about 9 days`
`21:59.35`	`*** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net)`
`22:03.11`	`JasonK1909`	`newb question, what is CLI command to get asterisk version #?`
`22:03.27`	`syzygyBSD`	`asterisk -v`
`22:03.28`	`Qwell`	`core show version`
`22:03.45`	`syzygyBSD`	`sorry, V`
`22:03.59`	`*** join/#asterisk ectospasm (n=ectospas@c-68-62-219-252.hsd1.al.comcast.net)`
`22:04.00`	`[TK]D-Fender`	`_Sam--, :/`
`22:04.44`	`_Sam--`	`im trying to remember why i switched to 1.4 in the first place`
`22:04.56`	`_Sam--`	`other than boredom, im drawing a blank.`
`22:05.20`	`_Sam--`	`i may try to go backward`
`22:05.43`	`JasonK1909`	`hmm that's what I thought ... the stock build on this server tells me sx00i-1.0.3, whatever that means ... was hoping there was another command I'd missed`
`22:05.50`	`[TK]D-Fender`	`_Sam--, well 1.2 support is over.... and 1.4 handles several things better...`
`22:06.07`	`_Sam--`	`i dont need support -- other than what you provide :)`
`22:06.16`	`_x86_`	`[TK]D-Fender: 1.2 support is "over" ?`
`22:06.16`	`_Sam--`	`and 1.2 did everything I do perfectly`
`22:06.17`	`thekidrio`	`[TK]D-Fender, how is the fax support in 1.4?`
`22:06.20`	`[TK]D-Fender`	`_Sam--, That may well be the deciding factor.`
`22:06.20`	`_x86_`	`[TK]D-Fender: it's EOL?`
`22:06.39`	`_x86_`	`_Sam--: i'm a devout 1.2 user also`
`22:06.40`	`[TK]D-Fender`	`_x86_, for many weeks now`
`22:07.42`	`_Sam--`	`[TK]D-Fender : there's no way to trace what happened when my asterisk died / dies?`
`22:08.05`	`_x86_`	`http://www.freepbx.org/2007/05/31/version-23-plans-and-asterisk-14/`
`22:08.26`	`_x86_`	`[TK]D-Fender: that says that Asterisk 1.2 will be supported for "years" after it's been EOL'd`
`22:09.19`	`[TK]D-Fender`	`_x86_, Really? Where?`
`22:09.27`	`_x86_`	`heh, and I just replaced an ancient 1.0 box with 1.2 last Friday ;)`
`22:09.42`	`_x86_`	`[TK]D-Fender: Asterisk 1.2 will be strong and running for many months (and years) to come.`
`22:09.48`	`[TK]D-Fender`	`_x86_, "Procrastination : The art of keeping up... with YESTERDAY"`
`22:10.02`	`_x86_`	`[TK]D-Fender: 1.2 is rock solid for what I use it for`
`22:10.13`	`[TK]D-Fender`	`_x86_, Nowhere does that say SUPPORTED anywhere. Thats a statement of FAITH.`
`22:10.25`	`_x86_`	`ah, true ;)`
`22:10.30`	`_x86_`	`i mis-interpreted :P`
`22:10.30`	`[TK]D-Fender`	`_x86_, Oh yeah.. they BELIEVED there were WMDs.....`
`22:10.36`	`_x86_`	`hahahaha`
`22:10.36`	`_Sam--`	`_x86_: i dont think anyone is forcing or asking you to upgrade...do so at your own peril!`
`22:10.53`	`_Sam--`	`if your stuff is fine, then its fine.`
`22:10.54`	`_x86_`	`_Sam--: [TK]D-Fender is trying to twist our arms ;)`
`22:11.09`	`_Sam--`	`my stuff WAS fine, and i upgraded. now it isnt.`
`22:11.09`	`[TK]D-Fender`	`_x86_, No, I'm just reporting the news, not MAKING it :)`
`22:11.12`	`JasonK1909`	`if this aa50 is using 1.4, which I guess it is, it's flaky as hell`
`22:11.13`	`_x86_`	`hehe`
`22:11.14`	`_Sam--`	`my own choice, my own fault.`
`22:11.20`	`*** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com)`
`22:11.45`	`_x86_`	`heh, it's misleading that asterisk.org still has 1.2.24 on the front page ;)`
`22:11.48`	`ajohnson`	`I would upgrade to 1.4 when it reaches EOL ;-)`
`22:12.14`	`ajohnson`	`1.4 hasn't passed our internal QA yet, though it is still getting much much better`
`22:12.44`	`_Sam--`	`[TK]D-Fender : if i go back to 1.2.X, can i use the zap modules that you made and installed or i just have to make install the 1.2 zaptel stuff again?`
`22:13.00`	`[TK]D-Fender`	`_Sam--, All from scratch`
`22:13.07`	`[TK]D-Fender`	`_Sam--, Zaptel to follow`
`22:13.26`	`_Sam--`	`i already have the old binary, and old modules directory still from 1.2`
`22:13.31`	`_Sam--`	`i just need old zap`
`22:13.52`	`[TK]D-Fender`	`_Sam--, make sure the versions are compatible`
`22:14.03`	`*** part/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue)`
`22:14.22`	`syzygyBSD`	`where is there a list of features that may break when upgrading from 1.2 to 1.4`
`22:15.06`	`[TK]D-Fender`	`syzygyBSD, Take a look at the what's new with 1.4 list :)`
`22:15.20`	`_Sam--`	`[TK]D-Fender : 100 bucks and you roll me back to 1.2 i was using?`
`22:15.54`	`[TK]D-Fender`	`_Sam--, I'll get you to 1.2.current inclusive for our previous work.`
`22:16.03`	`[TK]D-Fender`	`_Sam--, just prep the account again`
`22:16.10`	`_Sam--`	`i dont want current...i want what i had :)`
`22:16.16`	`_Sam--`	`cause i know how it worked.`
`22:16.20`	`[TK]D-Fender`	`Ok, will research for that.`
`22:16.28`	`_Sam--`	`i have all the tarballs`
`22:16.36`	`[TK]D-Fender`	`_Sam--, Oh, ok, no issue then`
`22:17.05`	`_Sam--`	`goin home now, will contact you in a few, after i first break everything :)`
`22:17.09`	`[TK]D-Fender`	`:D`
`22:18.03`	`_Sam--`	`thank you.`
`22:22.22`	`hrmphh`	`do i need more than one iax2 device to process simultaneous faxes?`
`22:25.09`	`[TK]D-Fender`	`hrmphh, Yes.`
`22:25.12`	`wishes`	`what softphones do video and are semi decent`
`22:25.15`	`hrmphh`	`k`
`22:25.25`	`hrmphh`	`stupid -- Registered IAX2 'iaxmodem1' (AUTHENTICATED) at 127.0.0.1:4571 -- Registered IAX2 'iaxmodem1' (AUTHENTICATED) at 127.0.0.1:4572`
`22:26.01`	`hrmphh`	`its hould be iaxmodem0 and iaxmodem1`
`22:31.02`	`*** join/#asterisk fujin (n=aj@unaffiliated/fujin)`
`22:33.28`	`*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)`
`22:35.22`	`wishes`	`hmm my GoToIf are working wrong`
`22:36.50`	`fujin`	`use AEL ;]`
`22:37.03`	`fujin`	`what are they doing wrong?`
`22:38.27`	`wishes`	`if it finds a match, surely it jumps to where its supposed to go`
`22:38.34`	`wishes`	`but its reading the next few gotoif's`
`22:38.55`	`fujin`	`not quite sure I follow you`
`22:38.58`	`fujin`	`console output=>?`
`22:40.14`	`hrmphh`	`ael should be the default =]`
`22:40.19`	`*** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net)`
`22:40.20`	`wishes`	`http://pastebin.ca/722329`
`22:40.27`	`wishes`	`thats the gist of it`
`22:40.38`	`wishes`	`i did snip a chink of sql code out`
`22:41.55`	`wishes`	`is there something wrong there with my syntax or soemthing?`
`22:42.41`	`codefreeze`	`wishes: mortal sin: you use n for priority nums, and goto fixed nums? tisk tisk! put labels on your n's and use those labels in the gotoif's....`
`22:42.49`	`[TK]D-Fender`	`wishes, Sure LOOKS like it matches and GOTO's`
`22:42.56`	`wishes`	`im fixing somebody elses codefreeze :)`
`22:42.57`	`fujin`	`^5 hrmphh`
`22:42.58`	`wishes`	`but thanks anyway`
`22:43.04`	`wishes`	`previous admin was a big weird`
`22:43.14`	`wishes`	`he quit because of mental health illness`
`22:43.19`	`[TK]D-Fender`	`and yes, using hard numbers on moving targets = dumb :p`
`22:43.42`	`fujin`	`using 'n' is generally bad`
`22:43.43`	`fujin`	`heh`
`22:43.52`	`fujin`	`I'm going to go with using extensions.conf standard language is generally bad`
`22:44.00`	`fujin`	`migrate to ael, solve all your problems!`
`22:44.10`	`codefreeze`	`wishes: try those fixes and see if it works...`
`22:44.35`	`[TK]D-Fender`	`fujin, and joint he dozen of users out there to support you! ;)`
`22:44.42`	`wishes`	`yeah good call codefreeze`
`22:44.44`	`wishes`	`cheers`
`22:44.50`	`wishes`	`he uses n EVERYWHERE`
`22:44.59`	`wishes`	`i should have thought of that - its been the cause of several other bugs too`
`22:44.59`	`fujin`	`[TK]D-Fender: hardly requires support, it's self explanatory`
`22:45.09`	`fujin`	`mind you, I'm from a code background, probly why it appeals to me`
`22:45.18`	`fujin`	`not going to turn this into a .conf vs. .ael debate though`
`22:45.22`	`Wonka`	`ael rocks.`
`22:45.25`	`codefreeze`	`fujin: you mean the other way round... you should always use the n's and (labels)... and stay away from fixed numbers wherever possible.... or, better yet.... as you said, use AEL.`
`22:45.31`	`[TK]D-Fender`	`fujin, it only explains itself... AFTER explaining itself! Documentation is sparse, users moreso`
`22:45.54`	`fujin`	`^^`
`22:45.56`	`fujin`	`codefreeze > you`
`22:46.11`	`[TK]D-Fender`	`No, sticking with hard numbering = GOOD. You KNOW whats going on and you will not get screwed by silly assumptions.`
`22:46.21`	`wishes`	`yeah`
`22:46.25`	`wishes`	`im with Fender here`
`22:46.48`	`fujin`	`gah, I'd hate to have to rewrite all of my super cool AEL magic in the standard language`
`22:46.55`	`[TK]D-Fender`	`<- doesn't play games or take "chances". I bet when the odds are rigged :p`
`22:47.15`	`codefreeze`	`Sorry guys, I disagree. There's no REN command (like in basic of days gone by!) Stay away from fixed numbers, use labels, and you are better off. But who am I to crimp your style?`
`22:47.16`	`wishes`	`bingo, fixed the problem`
`22:47.34`	`[TK]D-Fender`	`fujin, You already had to rewrite it inot there, of course you don't want to go back. but most dialplans have little need of it offers.`
`22:49.39`	`fujin`	`I'm inclined to agree with n(label)`
`22:49.44`	`fujin`	`at least then you can goto label`
`22:49.48`	`fujin`	`like a true label! ohmygosh`
`23:02.46`	`[TK]D-Fender`	`FinboySlick has quit ("Miranda IM! Smaller, Faster, Easier. http://miranda-im.org")`
`23:02.48`	`[TK]D-Fender`	`hrm`
`23:03.16`	`fujin`	`I use Miranda for MSN, it's nice.`
`23:03.27`	`[TK]D-Fender`	`an IM client named "Miranda". Like as in "Miranda Rights" as in "You have the right to remain SILENT"`
`23:03.47`	`[TK]D-Fender`	`A chat program named after "shut up if you know whats good for you"`
`23:03.57`	`[TK]D-Fender`	`irony++`
`23:04.12`	`fujin`	`dunno about any of that`
`23:04.14`	`fujin`	`ignorance++`
`23:04.57`	`[TK]D-Fender`	`FTW!`
`23:05.25`	`fujin`	`bliss, indeed.`
`23:05.39`	`fujin`	`Miranda IM was first started by Roland Rabien (figbug) on February 2, 2000. At that time, it was a minimalist ICQ clone and officially called Miranda ICQ`
`23:05.48`	`fujin`	`iirc, miranda was the protocol ICQ used to run, right?`
`23:06.20`	`*** join/#asterisk shinao1 (n=shinao1@217.20.242.52)`
`23:06.22`	`[TK]D-Fender`	`fujin, No, ICQ was the protocol ICQ STILL runs :)`
`23:06.35`	`fujin`	`meh`
`23:06.37`	`fujin`	`icq<*`
`23:12.16`	`ThoMe`	`hello`
`23:12.36`	`ThoMe`	`i search a good web-interface for cdr (i have my cdr (calllist) in the mysqldb) .. in php?`
`23:12.43`	`ThoMe`	`any ideas?`
`23:13.30`	`fujin`	`have you tried asterisk_stat?`
`23:13.49`	`fujin`	`http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54`
`23:13.49`	`*** join/#asterisk charasky (i=jgflk44i@201.250.49.173)`
`23:13.51`	`ThoMe`	`fujin: no, gracias, i search demos for "asterisk_stat".. thx :-)`
`23:14.00`	`charasky`	`buenas`
`23:14.03`	`charasky`	`una pregunta`
`23:14.05`	`fujin`	`err`
`23:14.05`	`fujin`	`http://www.areski.net/asterisk-stat-v2/about.php`
`23:14.17`	`ThoMe`	`pregunta = question`
`23:14.26`	`fujin`	`English please.`
`23:14.38`	`charasky`	`ok`
`23:14.45`	`hrmphh`	`charasky; callate la boca, mmk?`
`23:15.32`	`charasky`	`from outside they estan making called international redirecting the calls, as blockade this?`
`23:15.42`	`ThoMe`	`fujin: oh, i have asterisk csr (a_stat) installied hihi`
`23:16.00`	`charasky`	`hrmphh mmk?`
`23:16.08`	`ThoMe`	`already yesterday, fujin :-)`
`23:16.12`	`fujin`	`that's great`
`23:16.13`	`ThoMe`	`fujin: its ok...`
`23:16.20`	`ThoMe`	`fujin: but better? no ?;)`
`23:16.27`	`fujin`	`make one yourself`
`23:16.35`	`ThoMe`	`fujin: hihi .-)`
`23:16.36`	`ThoMe`	`jep.`
`23:16.37`	`charasky`	`from outside they making called international redirecting the calls, as blockade this?`
`23:17.02`	`charasky`
`23:17.03`	`charasky`	`as I stop that`
`23:17.04`	`charasky`	`?`
`23:17.12`	`fujin`	`wow, fail`
`23:17.15`	`fujin`	`I have no idea what you're talking about`
`23:17.23`	`syzygyBSD`	`something was lost in the translation`
`23:17.38`	`charasky`	`yes syzygyBSD`
`23:17.47`	`fujin`	`it'd appear so`
`23:18.12`	`charasky`
`23:18.12`	`charasky`	`the internal ones have the capacity to redirect calls`
`23:18.44`	`charasky`	`somebody from outside this redirecting itself to itself, to make calls free`
`23:18.58`	`charasky`	`internationals`
`23:19.44`	`charasky`
`23:19.45`	`charasky`	`he is bug or a bad configuration?`
`23:19.55`	`charasky`	`*it`
`23:20.37`	`fujin`	`I don't really understand what you're asking, so I can't say`
`23:20.42`	`fujin`	`can you give us some examples`
`23:20.44`	`fujin`	`console output`
`23:20.45`	`fujin`	`configuration etc`
`23:21.07`	`syzygyBSD`	`it is bad configuration`
`23:21.22`	`charasky`	`thanx syzygyBSD`
`23:22.19`	`charasky`	`this i want know`
`23:22.25`	`charasky`	`wanted...`
`23:22.59`	`syzygyBSD`	`charasky: what is your native language?`
`23:23.08`	`*** join/#asterisk tsurko (n=tsurko@78.90.100.214)`
`23:23.14`	`charasky`	`spanish`
`23:23.22`	`syzygyBSD`	`k, thanks`
`23:23.24`	`charasky`	`my english is not very welll`
`23:23.31`	`charasky`	`is horrible..`
`23:23.31`	`fujin`	`We can tell.`
`23:23.59`	`charasky`	`but understand`
`23:23.59`	`syzygyBSD`	`ya, knowing what language you are comming from helps me identify which words you would leave out`
`23:24.39`	`charasky`	`also thanks`
`23:24.53`	`charasky`
`23:24.53`	`charasky`	`single it needed to know if he were bug`
`23:25.36`	`hrmphh`	`http://pastebin.com/m41871f1a`
`23:25.43`	`hrmphh`	`any hydrafax gurus?`
`23:26.05`	`jsmith`	`charasky: Por favor preguntar en castillano... tratate' responder`
`23:26.23`	`jsmith`	`hrmphh: Hylafax, you mean?`
`23:26.42`	`hrmphh`	`err yeah :)`
`23:29.49`	`*** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr)`
`23:32.44`	`ThoMe`	`hello`
`23:32.46`	`ThoMe`	`jsmith: good morning`
`23:32.51`	`PSU_Boss`	`hello, when i run /var/lib/asterisk/bin/module_admin listonline i get this error: Fatal error: Unsupported operand types in /var/lib/asterisk/bin/module_admin on line 140`
`23:32.58`	`PSU_Boss`	`does anyone know how i can fix this?`
`23:32.58`	`ThoMe`	`fujin: emm, how i can record a call, on demand ?`
`23:33.36`	`hrmphh`	`any idea how to debug incoming faxes?`
`23:34.31`	`*** join/#asterisk ChrisN (n=ChrisN@tz.tztalk.com)`
`23:34.38`	`jsmith`	`ThoMe: You need to Google for "automon asterisk features.conf"`
`23:35.02`	`jsmith`	`PSU_Boss: To be honest, I've never heard of module_admin before`
`23:35.29`	`PSU_Boss`	`jsmith, hmm.. it must be part of freepbx then`
`23:36.06`	`jsmith`	`Maybe so...`
`23:41.20`	`ChrisN`	I'm compiling asterisk from svn trunk. Things seem to be fine until linking when I get errors like this: "/usr/bin/ld: Dwarf Error: Abbrev offset (17576) greater than or equal to .debug_abbrev size (1120)." "../apps/app_dial.eo: In function `dial_exec_full': app_dial.c:(.text+0x368a): undefined reference to `ast_bridge_call'" What's up?
`23:42.52`	`fujin`	`looks like fun`
`23:44.02`	`Nugget`	`it might simply be that asterisk from svn trunk is broken right now. Is there a particular reason you want to run the development branch?`
`23:44.30`	`hrmphh`	`i take it you can use the same ttyIAX bad boys for both incoming/outgoing?`
`23:46.10`	`ChrisN`	`Nugget: I've been having problems with meetme conferences and wanted to see if they exist in trunk before filing a bug`
`23:46.37`	`ThoMe`	`fujin: is "monitor" a modul?`
`23:46.50`	`ChrisN`	`lots of "unable to write frame" errors`
`23:49.56`	`*** join/#asterisk blq (n=Bl@dslb-088-064-159-177.pools.arcor-ip.net)`
`23:50.54`	`*** join/#asterisk iBuMp (i=iBuMp@cpe-66-68-37-190.austin.res.rr.com)`
`23:52.06`	`*** join/#asterisk craigk (n=ckowald@58.174.122.198)`
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`23:54.46`	`*** mode/#asterisk [+o anthm] by ChanServ`
`23:55.19`	`hrmphh`	`any way to put some logic in FaxDispatch so that HylaFax knows where to send to based on the number dialed?`
`23:55.35`	`hrmphh`	`i.e. if num#1 is called it goes to email1 and if num#2 is called it goes to email#2`
`23:55.50`	`wishes`	`should be do-able`
`23:56.34`	`wishes`	`but not using FaxDispatch at a guess`
`23:56.40`	`wishes`	`bbiab`
`23:57.37`	`hrmphh`	`hmm`
`23:57.47`	`hrmphh`	`so can you have FaxDispatch call some external script?`
`23:58.22`	`hrmphh`	`http://www.hylafax.org/content/Handbook:Advanced_Server_Configuration:Per-call_Dynamic_Configuration\`
`23:58.25`	`hrmphh`	`-\`
`23:58.28`	`hrmphh`	`hmm`
`23:59.30`	`ChrisN`	`I get the same errors when using the 1.4 branch`

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