IRC log for #asterisk on 20070619

`00:01.58`	`ManxPower`	`THERE IS ONLY ONE TRUE BOOK!`
`00:02.04`	`ManxPower`	`ATFOT!`
`00:03.12`	`Teccy`	`i've got a digium 400p with 2 FXO and i'm having issues with them answering calls`
`00:03.33`	`ManxPower`	`and that issue is.....?`
`00:03.42`	`Teccy`	`asterisk says it's answered the call (with an Answer() statement) and follows through the dial plan, but it never actually answers the line`
`00:04.19`	`Teccy`	`i get a message from chan_zap saying something along the lines of Ring/Off-hook in strange state 6`
`00:04.25`	`Teccy`	`(i dont have the message handy atm)`
`00:04.29`	`Teccy`	`any thoughts?`
`00:05.13`	`Teccy`	`googlgin only reveals some messages from 2004, to do with now-fixed issues`
`00:05.26`	`ManxPower`	`Teccy: that happens when a call comes in too soon after another call hangs up on the same line (FXO) or when a user goes onhook/offhook too fast (FXS)`
`00:05.32`	`Teccy`	`i'm using zaptel 1.4.4 drivers (bsd), but i've tried the asterisknow cd, and i get exactly the same issue`
`00:05.48`	`Teccy`	`ManxPower: no other calls on the line for several minutes`
`00:05.57`	`ManxPower`	`Teccy: it has never caused a major problem on the systems I've seen that on.`
`00:06.14`	`JT`	`err`
`00:06.15`	`JT`	`bsd`
`00:06.18`	`ManxPower`	`sounds like you might have noisy lines`
`00:06.22`	`Teccy`	`i've tried both on pstn lines and on a pbx`
`00:06.25`	`ManxPower`	`JT: Errr, AsteriskNOW`
`00:06.30`	`JT`	`is not a paranthesis issue`
`00:06.46`	`JT`	`Teccy: the zap drivers for bsd are unofficial and may or may not work`
`00:06.55`	`Teccy`	`ive opened a ticket with both the card provider and digium last week, but i've yet to get a response`
`00:07.08`	`ManxPower`	`JT: test with AsteriskNOW or most people will tell you BSD is not supported.`
`00:07.17`	`ManxPower`	`..not JT, but Teccy`
`00:07.27`	`Teccy`	`JT: hence why i tried asterisknow. but i get exactly the same issue, hence i dont think it's driver relayed`
`00:07.30`	`Teccy`	`related*`
`00:07.36`	`mrdigital-work`	`what im trying to do isnt covered`
`00:07.47`	`Teccy`	`i have a feeling it could be dodgy hardware`
`00:07.53`	`ManxPower`	`Teccy: I don't think it is either, but people will still waste time.`
`00:07.59`	`Teccy`	`indeed`
`00:08.45`	`Teccy`	`as i say, i think i've covered most bases. linux/bsd. different phone cables, different motherboards (different chipsets), modules in different places on the card, 2 different pstn lines, and a pbx line`
`00:09.38`	`JT`	`and the software is all setup correctly?`
`00:09.48`	`Teccy`	`i did notice that if i run ztmonitor and try to dial out (havent tried in), the line is really noisy before the dialout begins, then it goes silent, then i hear the dtmf tones crystal clear`
`00:09.54`	`Teccy`	`yeh, ive checked everything is set for UK`
`00:10.11`	`Teccy`	`but i never hear the dialtone from the line`
`00:10.49`	`JT`	`pastebin.ca zaptel.conf and zapata.conf`
`00:10.53`	`Teccy`	`however, before it picks up to dial out, you can hear over the noise, if someone speaks into another phone connected to the pstn line, so there is a connection there`
`00:11.04`	`Teccy`	`JT: it's not the issue`
`00:11.46`	`Teccy`	`but if you insist, 2 secs`
`00:11.48`	`*** join/#asterisk perf3kt (n=perf3kt@adsl-68-73-150-167.dsl.ipltin.ameritech.net)`
`00:12.10`	`perf3kt`	`~trixbox`
`00:12.11`	`jbot`	`Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org`
`00:14.36`	`Teccy`	`http://www.pastebin.ca/575369`
`00:15.03`	`Teccy`	`and yes, ive tried without the callerid bits, and i've tried the busydetect=no and callprogress=no that several people have suggested before`
`00:15.10`	`Teccy`	`any thoughts ManxPower ?`
`00:17.07`	`bkruse`	`hey, when using manager, you can do applicaion: blah with call originates, but how do you pass args to it?`
`00:17.30`	`*** join/#asterisk denon (n=denon@tooth.decay.org)`
`00:17.30`	`*** mode/#asterisk [+o denon] by ChanServ`
`00:19.37`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
`00:28.25`	`*** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net)`
`00:31.27`	`*** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au)`
`00:39.35`	`*** join/#asterisk emiquelito (n=emiqueli@c93456d6.virtua.com.br)`
`00:44.20`	`*** part/#asterisk emiquelito (n=emiqueli@c93456d6.virtua.com.br)`
`00:44.53`	`*** join/#asterisk b00gz (n=b00gz@d233-124-245.col.wideopenwest.com)`
`01:01.25`	`JerJer`	`is my email phreakin out or was there a very old asterisk-announce message re-set today?`
`01:01.58`	`Strom_M`	`JerJer: no, version 0.7 really is almost ready`
`01:03.19`	`Corydon76-home`	`JerJer: -addons, not the main release`
`01:03.39`	`JerJer`	`The Asterisk development team has announced the releases of`
`01:03.39`	`JerJer`	`Asterisk-addons 1.2.7 and 1.4.2`
`01:03.51`	`JerJer`	`Date: Today, 6:56pm`
`01:03.58`	`JerJer`	`ohhhhhhhhhhh`
`01:04.00`	`JerJer`	`addons -`
`01:04.02`	`JerJer`	`don't mind me`
`01:04.04`	`JerJer`	`long day`
`01:04.09`	`JerJer`	`ugh`
`01:04.28`	`JerJer`	`i'm like 1.2.7 - H.323 driver? what ?!`
`01:04.40`	`Corydon76-home`	`ooh323`
`01:04.47`	`*** join/#asterisk mitcheloc (n=mitchelo@rrcs-64-183-110-250.west.biz.rr.com)`
`01:04.50`	`JerJer`	`that smell is my brain melting`
`01:11.18`	`*** join/#asterisk AndrewKT (n=Miranda@user-0c8h5ql.cable.mindspring.com)`
`01:19.43`	`*** join/#asterisk [hC] (n=hardcore@66.119.167.162)`
`01:20.34`	`[hC]`	`Not sure whats going on here with this manager enabled app im using. I'm tcpdumping and watching the responses, (its trying to originate a call) asterisk comes back and says that the origination was successfully queued, then it never does anything.`
`01:20.34`	`[hC]`	`how can i check further, like to see the origination queue, or something?`
`01:31.35`	`*** join/#asterisk logicwrath (n=some@c-68-60-121-112.hsd1.mi.comcast.net)`
`01:34.03`	`logicwrath`	`I can make internal calls from my new 7940 but when I try to make internal calls to the cisco I get app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) can someone give me a hint?`
`01:34.59`	`logicwrath`	`I am also seeing chan_sip.c: Error in codec string '=audio 53398 RTP/SAVP 107 119 100 106 6 0 97 105 98 8 18 3 5 101'`
`01:35.07`	`*** join/#asterisk fujin (n=aj@unaffiliated/fujin)`
`01:35.17`	`logicwrath`	`do I need to set a special RTP port range for the cisco?`
`01:36.01`	`*** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net)`
`01:36.03`	`fujin`	`wow, I'm sure I was asking the same question yesterday`
`01:36.06`	`fujin`	`just came to ask it again`
`01:36.08`	`*** join/#asterisk wotcha (n=jim@cust4716.qld01.aanet.com.au)`
`01:36.29`	`logicwrath`	`same as mine?`
`01:37.07`	`fujin`	`possibly, I was wondering how to configure an as5400's RTP port range to see if it resolved an issue I'm seeing`
`01:37.13`	`fujin`	`just sounded like half of my question :P`
`01:37.52`	`logicwrath`	`are you referring to an ASA firewall?`
`01:38.08`	`fujin`	`no, a cisco as5400 universal gateway`
`01:38.13`	`fujin`	`It's what I'm using to terminate our E1 on.`
`01:38.20`	`fujin`	`and then sip->asterisk`
`01:38.43`	`logicwrath`	`Are you using cisco phones?`
`01:38.50`	`fujin`	`nope, unfortunately not`
`01:38.54`	`fujin`	`Mitels`
`01:39.36`	`logicwrath`	`i just got a free 7940. they seem to behave differently than normal SIP phones as far as configuration`
`01:39.57`	`fujin`	`I have a 7912 sitting on my desk`
`01:40.03`	`fujin`	`can't configure it cause I don't have the cisco tools`
`01:40.24`	`logicwrath`	`what cisco tools do you need`
`01:40.38`	`fujin`	`no idea`
`01:40.45`	`fujin`	`no matter, we wouldn't be rolling the Ciscos anyway.`
`01:43.13`	`logicwrath`	`my device registers I just cant route to it from my softphone`
`01:43.33`	`logicwrath`	`i can however call the softphone from my cisco`
`01:44.37`	`logicwrath`	`ive tried so many diffrent .cnf and .xml configurations im sick of rebooting it`
`01:49.51`	`fujin`	`Is it possible to enable silence supression in asterisk?`
`01:52.13`	`*** join/#asterisk rsd99 (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net)`
`01:53.41`	`*** join/#asterisk rsd99 (n=rsd095@c-71-224-187-182.hsd1.pa.comcast.net)`
`01:54.32`	`rsd99`	`quick question. how do i go about creating a mailbox? from the docs i have read there is a command addmailbox. i can't seem to find it.`
`01:55.07`	`*** join/#asterisk pigpen (n=mark@fw.seamans.cc)`
`01:55.57`	`pigpen`	`Ok...please bear with me. I normally setup * on a custom box, custom gentoo load using custom scripts and my custom front end.`
`01:56.04`	`pigpen`	`however, this takes a bit of time I don't have right now.`
`01:56.21`	`pigpen`	`I -really- don't want to use trixbox.`
`01:56.29`	`pigpen`	`So what is the opinion about AsteriskNow.`
`01:56.30`	`pigpen`	`?`
`01:56.49`	`logicwrath`	`i like the front end and it has a file editor`
`01:57.08`	`logicwrath`	`it also doesnt overwrite hand written changes like ive seen other UI's do in the past`
`01:57.17`	`pigpen`	`k. thanks.`
`01:57.35`	`logicwrath`	`im not sure I like how it stuffs everything in the users.conf file though`
`01:57.38`	`pigpen`	`Just need to deploy a quick system for a customer who got their Strata hit by lightening last night.`
`01:58.08`	`pigpen`	`on a crap system, to buy enough time to buy a nice server, switches, phones, etc...`
`01:58.24`	`pigpen`	`Just the shear fact I haven't gotten flamed yet is a good sign.`
`01:58.27`	`pigpen`	`:)`
`01:59.18`	`logicwrath`	`well i think most are occupied elsewhere`
`01:59.37`	`pigpen`	`If they are smart.`
`01:59.59`	`logicwrath`	`or lurking for a good discussion`
`02:00.16`	`pigpen`	`.... and figure this one is stupid.`
`02:00.30`	`logicwrath`	`along with mine :) and a few others`
`02:00.45`	`rsd99`	`how do i go about creating a voice mailbox. is there a command that does it, or is it done within the CLI?`
`02:02.58`	`logicwrath`	`�rsd99� /usr/src/asterisk/addmailbox`
`02:03.03`	`logicwrath`	`have you tried that?`
`02:03.53`	`*** join/#asterisk GothAlice (n=amcgrego@209.161.123.42)`
`02:05.41`	`GothAlice`	`I have an IAX provider (IAX/provider) on the default context. How do I get it to ring my extension (6000) when anyone calls the DID provided by said provider?`
`02:06.07`	`logicwrath`	`�GothAlice� an inbound route?`
`02:06.13`	`GothAlice`	`Correct.`
`02:06.17`	`GothAlice`	`Right now I've got exten => s,1,Answer and exten => s,x,Dial(6000)`
`02:06.31`	`GothAlice`	`DIal out works perfectly, BTW.`
`02:06.42`	`GothAlice`	`I just need the simplest method to get a DID working.`
`02:06.53`	`*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id)`
`02:07.09`	`GothAlice`	`With my current setup (s,1 & s,x) when the DID calls the server I get a "Rejected connect attempt from 204.11.194.34, who was trying to reach 's@'" message.`
`02:08.01`	`logicwrath`	`im really not that good im sorry`
`02:08.08`	`logicwrath`	`i typically use a UI`
`02:08.12`	`GothAlice`	`;_;`
`02:09.19`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`02:09.19`	`*** mode/#asterisk [+o anthm] by ChanServ`
`02:09.35`	`logicwrath`	`are you using any cisco phones goth maybe you can help me`
`02:09.48`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
`02:09.54`	`*** join/#asterisk infernix (i=nix@unaffiliated/infernix)`
`02:10.01`	`GothAlice`	`Linksys, which is cicco, but I'm on a call right now.`
`02:10.21`	`JT`	`they're not the same phones though`
`02:10.21`	`flenders`	`linksys is not a cisco phone`
`02:10.31`	`flenders`	`it's actually far from it`
`02:11.28`	`*** join/#asterisk santiago (i=santiago@debian/developer/santiago)`
`02:12.11`	`logicwrath`	`do cisco phones use non standard rtp ports?`
`02:12.28`	`logicwrath`	`why is my SIP image trying to use 5061? Isnt that SCCP?`
`02:13.26`	`aptura`	`logic look at ciscos web site?`
`02:13.47`	`logicwrath`	`not as much as the voip-info site or some other sites linked from there`
`02:14.02`	`logicwrath`	`i cant route to the cisco`
`02:14.16`	`logicwrath`	`but I can call internally from the cisco`
`02:14.39`	`logicwrath`	`[Jun 18 21:30:43] WARNING[2291] chan_sip.c: Error in codec string '=audio 53398 RTP/SAVP 107 119 100 106 6 0 97 105 98 8 18 3 5 101'`
`02:14.39`	`logicwrath`	`[Jun 18 21:30:44] WARNING[3264] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)`
`02:15.30`	`*** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-212-192.dsl.irvnca.pacbell.net)`
`02:22.25`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`02:23.06`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
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`02:30.03`	`[TK]D-Fender`	`GothAlice, problem is that you did not set an inbound context.`
`02:30.16`	`GothAlice`	`Nope. Not the problem at all. I just got it working.`
`02:30.47`	`GothAlice`	`DIDWW's mapping was crazy: IAX2/context:pw@domain/extension@default`
`02:31.35`	`GothAlice`	`The two at symbols threw me for a loop.`
`02:35.19`	`*** join/#asterisk jarnaud (n=jarnaud@64.235.205.43)`
`02:38.03`	`*** join/#asterisk enjay5150 (i=PJirc@wsip-24-249-169-168.ph.ph.cox.net)`
`02:44.43`	`enjay5150`	`has anyone experienced CPU util problems when using Queues?`
`02:44.53`	`*** join/#asterisk fbffff (n=fbffff@c-24-12-176-127.hsd1.il.comcast.net)`
`02:47.04`	`*** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com)`
`02:53.06`	`*** join/#asterisk davidcsi (n=davidcsi@82.158.35.53.dyn.user.ono.com)`
`02:53.21`	`logicwrath`	`yea, i got it working`
`02:53.33`	`logicwrath`	`finally`
`02:53.34`	`Cyber-Dogg`	`when I use the playback in my extension... do I need to specify a path?`
`02:53.40`	`Cyber-Dogg`	`cause it doesn't seem to be working`
`02:54.01`	`davidcsi`	`hello all: I know you can add custom fields to cdr via the cdr_custom. BUt that only works for Master.csv... but how do I add custom fields and set them via the dialplan??`
`04:23.18`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`04:23.18`	`*** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.19 and 1.4.5 (June 15, 2007) *-addons 1.2.7 and 1.4.2 (June 18, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support.`
`04:23.29`	`JT`	`flenders: i guess, i think you'll need to provide your own 12v adpater though`
`04:23.32`	`neoalex`	`Nuitari: really... what's wrong with grandstreams?`
`04:23.37`	`JT`	`~gs`
`04:23.38`	`jbot`	`GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.`
`04:23.44`	`JT`	`just in the nick of time there, jbot`
`04:23.52`	`enjay5150`	`hahaaha`
`04:23.53`	`Nuitari`	`noalex I just don't trust sales pitches`
`04:23.55`	`enjay5150`	`thats awesome`
`04:24.30`	`flenders`	`JT: would it work with one of those multi-voltage dicksmith's adapters?`
`04:24.35`	`JT`	`probably`
`04:24.36`	`Nuitari`	`neoalex: There is one actually coming in for evalutation purpose, I should get it on thursday`
`04:24.45`	`neoalex`	`yes but why are they bad?`
`04:24.52`	`neoalex`	`sound quality, ease of use`
`04:24.54`	`JT`	`Nuitari: they're not even heavy enough to hold doors open`
`04:25.01`	`neoalex`	`they break for no reason`
`04:25.02`	`Nuitari`	`wow, I haven't heard from dse from a long time`
`04:25.08`	`flenders`	`no, sound quality is 'award winning'`
`04:25.29`	`enjay5150`	`hah`
`04:25.37`	`Nuitari`	`JT: the clients wants me to try them`
`04:26.05`	`JT`	`client will be clients...`
`04:26.10`	`Nuitari`	`yeah`
`04:26.56`	`*** part/#asterisk enjay5150 (i=PJirc@wsip-24-249-169-168.ph.ph.cox.net)`
`04:30.02`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
`04:30.12`	`neoalex`	`blah... if I don't like ti I'll sell it back on ebay`
`04:30.45`	`JT`	`why are you so convinced on buying it in the face of unanimous opposition?`
`04:31.27`	`neoalex`	`I already bid on it`
`04:31.30`	`neoalex`	`before talking to you`
`04:31.44`	`JT`	`i see`
`04:31.53`	`neoalex`	`anyway... found that award... some Internet Telephony excellence 2006 award`
`04:32.00`	`JT`	`research is useful :)`
`04:32.09`	`JT`	`they must've been the only competitor`
`04:32.22`	`neoalex`	`haha`
`04:32.56`	`neoalex`	`anyway... this model is they're enterprise solutions phone, one of them at least`
`04:33.10`	`aptura`	`neo you also get telephony mag?`
`04:33.10`	`neoalex`	`so maybe it's better then the budgetones`
`04:33.17`	`neoalex`	`no`
`04:33.21`	`neoalex`	`should I?`
`04:33.29`	`JT`	`the problem is that it's still a granstream`
`04:33.43`	`JT`	`grandstream`
`04:33.46`	`aptura`	`I mabey one of the few here that does recieve the magazine.`
`04:34.32`	`aptura`	`I wonder what would be a ideal ip phone for motels and inns. I have seen only propriatory phones so far.`
`04:34.37`	`neoalex`	`so what else do you think I should get?`
`04:34.40`	`JT`	`i doubt many get such magazines`
`04:34.52`	`neoalex`	`say I don't like the GS when I get it`
`04:35.10`	`JT`	`polycom`
`04:35.15`	`JT`	`~phones`
`04:35.16`	`jbot`	`i guess phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever.`
`04:36.21`	`aptura`	`something low cost for non chain hotels.`
`04:37.21`	`Nuitari`	`aptura: define low cost as a price range, please`
`04:37.23`	`aptura`	`motels have some of the more interesting networks setup.`
`04:37.34`	`aptura`	`100 dollars`
`04:37.38`	`aptura`	`per unit.`
`04:37.47`	`*** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk)`
`04:38.01`	`JT`	`you can get polycoms for that`
`04:38.02`	`Nuitari`	`aptura: you can get some Polycom`
`04:38.24`	`Nuitari`	`like the 320 or 330`
`04:38.44`	`*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)`
`04:39.32`	`aptura`	`looking at it now`
`04:40.04`	`aptura`	`109 cdn`
`04:40.25`	`aptura`	`thats a nice looking phone`
`04:40.26`	`JT`	`for which`
`04:40.26`	`aptura`	`:)`
`04:40.30`	`aptura`	`320`
`04:40.46`	`Nuitari`	`aptura: plus you could easily get bulk pricing deals`
`04:42.17`	`aptura`	`possible`
`04:42.26`	`neoalex`	`greaaaat... paypal is down`
`04:47.22`	`aptura`	`Jt whats your experiance with them?`
`04:47.52`	`JT`	`haven't used the model in particular, but in general, polycoms are reliable and have quality audio`
`04:47.56`	`aptura`	`I get a little speaker audio feedback on my 501 but other then that it works fine`
`04:48.36`	`*** join/#asterisk rad07 (i=raca@64-126-95-37.static.everestkc.net)`
`04:49.26`	`coppice`	`well, they only put a little speaker in there :-)`
`04:49.41`	`aptura`	`Also the case may need more cooling since doing a eval at some of these hotels the IDFs room temps can be in the mid to high 70s. One had a way undersized portable ac unit inside it. Bad design. IT was one room away from the corner of the building which I think should have been made into the wiring room.`
`04:50.18`	`coppice`	`boxes that cook is now the industry best practice`
`04:50.24`	`aptura`	`hehe`
`04:50.58`	`aptura`	`hotel owners dont seem to care or are very uneducated in the way a small data center room needs to be built.`
`04:51.31`	`Nuitari`	`aptura: most owners / managers don't unless you can prove it with hard bottom line amounts`
`04:52.03`	`aptura`	`I know. One room had a 3 inch copper water line mounted over all the equipment.`
`04:52.27`	`JT`	`most of your phones won't be in the idf will they?`
`04:52.30`	`aptura`	`in fact i have a picture of it ;)`
`04:53.22`	`aptura`	`well actually IDF is probebly not the proper naming for just one building. make it simple call it the mdf :)`
`04:55.39`	`JT`	`yes, but your phones won't need to endure the conditions of the idf/mdf, right?`
`04:56.29`	`*** join/#asterisk Avalone (n=Avalone_@mail.kawkazrg.ru)`
`04:57.10`	`aptura`	`no not at all`
`04:57.31`	`aptura`	`each room has a/c as I expect all hotels do.`
`04:57.50`	`JT`	`right`
`04:57.55`	`*** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal)`
`04:58.00`	`coppice`	`that is usually off during the hotest part of the day`
`04:58.15`	`aptura`	`check this out. build a one of a kind case that looks more pbx then a pc case. http://www.protocase.com/`
`05:01.38`	`JT`	`aptura: at what cost though?`
`05:01.57`	`*** join/#asterisk unspin (n=unspin@24.82.161.85)`
`05:02.01`	`aptura`	`wow some big company names use there service`
`05:02.25`	`coppice`	`probably high, but if you know who to go to getting custom cases made can be quite cheap`
`05:02.37`	`aptura`	`well I did not read the article by tomshardware if it was a one up protype of production cost but there version was around a grand.`
`05:03.22`	`aptura`	`I guess if you want to impress a big company it better not look cheap or off the shelf.`
`05:04.19`	`aptura`	`coppice what do you use`
`05:04.28`	`JT`	`that company only likes to do small runs`
`05:04.41`	`JT`	`what if you like the prototype and what to do large quantities, have to go elsewhere?`
`05:05.25`	`coppice`	`aptura: I haven't done it for some years, but when we wanted modest numbers of cases for specialist things - say 50 off - the price spread was huge, and the one that was easily the cheapest was actually the best`
`05:05.44`	`aptura`	`sure`
`05:06.17`	`coppice`	`JT: I doubt anyone only does small quantities`
`05:07.05`	`aptura`	`On top of my commute to washington stopped by the owner of www.turbinefun.com to look at his jet boat :)`
`05:07.30`	`aptura`	`Entire boat is made out of kevlar.`
`05:08.06`	`JT`	`then you went to namedrop.com? :)`
`05:08.07`	`aptura`	`And honycomb paper and balsa. Very strong weighs in at 2100 bls.`
`05:08.48`	`aptura`	`JT I have experaince working in the same jet engines you see in that site from 20 years ago.`
`05:09.25`	`JT`	`ok, and this was relevant how? :)`
`05:10.29`	`aptura`	`Well the boat will be powered by the T58-GE-100 engine at only 1,500 SHP`
`05:10.39`	`coppice`	`the stuff we used to make radar dishes from should make a very light strong boat`
`05:11.13`	`aptura`	`He has sanded it down and is going to paint it a bright yellow. Personally I would put some orange in it.`
`05:11.49`	`coppice`	`paint adds weight :-)`
`05:12.08`	`JT`	`absorbing water adds more`
`05:13.35`	`aptura`	`I know`
`05:15.47`	`nowork`	`hi I want to setup 67 block caller ID function, how can I take the called num out from 67xxxx?`
`05:21.13`	`[TK]D-Fender`	`nowork, depends on what "*67xxxx" is in.`
`05:21.39`	`nowork`	`*67countrycode+num`
`05:21.57`	`nowork`	`thanks TK:)`
`05:22.33`	`nowork`	`TK,anotherquestion here:: my sip device call to my asterisk , with almost 0 PDD..i think this is because asterisk provided ringback tone instead remote end sip longdistance provider`
`05:22.39`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`05:22.58`	`[TK]D-Fender`	`nowork, ${[var-or-function-call]:3}`
`05:23.30`	`[TK]D-Fender`	`?`
`05:24.05`	`nowork`	`TK: will try that for *67xx; .. the second question , is that confusting?`
`05:24.35`	`rad07`	`Hi, Anybody knows how to access Asterisk-GUI from a Windows machine? I can access it locally. I checked default Apache test page on port 80 and it works so I know that Apache http is responding. I don't know how to diagnose "mini built-in" Asterisk HTTP Server.`
`05:27.21`	`*** join/#asterisk steliosk (n=Stelios@62.169.217.209)`
`05:31.41`	`tzafrir`	`http://asteriskmachine/`
`05:32.01`	`tzafrir`	`http://ip.of.aste.risk/`
`05:32.23`	`tzafrir`	`maybe you need something of the sort of :8088/ there`
`05:33.09`	`tzafrir`	`ah, you have a wrong path. It probably needs to be: /asterisk/config/basic.html`
`05:33.28`	`tzafrir`	`(so long for .html being "static contents")`
`05:34.19`	`rad07`	`I setup port 8080. I checked on Linux CLI "http show status". it says " Server Enabled and bound to 192.168.1.70:8080"`
`05:35.17`	`rad07`	`I am using this: http://192.168.1.70:8080/asterisk/static/config/cfgbasic.html Same line works on local Asterisk machine`
`05:36.31`	`rad07`	`It says Enabled URI's: /asterisk/static/... => Asterisk HTTP Static Delivery`
`05:36.46`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`05:37.37`	`*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)`
`05:38.13`	`*** join/#asterisk jmls (n=jmls@62.49.235.130)`
`05:38.37`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`05:40.18`	`rad07`	`tzafrir: what are these 2 urls. They don't work`
`05:41.25`	`JT`	`you mean the examples he snet?`
`05:42.11`	`rad07`	`Can I setup SIP channel with Linksys SPA-3102 through Asterisk-GUI? Does it use the same files: sip.conf, extensions.conf or it uses DB?`
`05:42.40`	`tzafrir`	`it uses extensions.conf and users.conf`
`05:43.44`	`nowork`	`tzafrir: 1793159183185677`
`05:43.48`	`nowork`	`sorry,`
`05:43.51`	`nowork`	`mistake`
`05:44.32`	`*** join/#asterisk hijacked (i=Jao5@cerebus.clandestineresearch.com)`
`05:45.49`	`*** part/#asterisk Avalone (n=Avalone_@mail.kawkazrg.ru)`
`05:47.31`	`rad07`	`What about sip.conf. I am an asterisk novice and I am reading a book Asterisk TFOT and I am trying to setup a basic dial plan. The books talks about using extensions.conf, sip.conf, voicemail.conf. What is users.conf?`
`05:48.52`	`rad07`	`What is underlying storage for Asterisk-GUI? Can I achieve the basic setup only via GUI?`
`05:52.21`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)`
`05:52.54`	`tzafrir`	`the underlying storage is the asterisk config files`
`05:53.29`	`tzafrir`	`Each and every time you refresh the gui screen , it actually parses them.`
`05:54.32`	`tzafrir`	`that's easy to check: add an entry in users.conf and save it. Don't reload anything in asterisk`
`05:54.43`	`tzafrir`	`then refresh the GUI`
`05:54.49`	`*** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au)`
`05:55.57`	`tzafrir`	`TFOT is (still) in the past of Asterisk 1.2 and still lacks any documentation of users.conf or the manager-over-http features.`
`05:55.58`	`rad07`	`So there is no extra storage that is exclusive to GUI. Basically GUI is just a front end to *.conf files. Not DB involved? Right?`
`05:56.39`	`tzafrir`	`it's not exclusive to the GUI. You can edit users.conf with $EDITOR as well`
`05:57.03`	`rad07`	`Do you want to say that users.conf was used in version 1.2. I seached for that file name in the book and didn't find any mention of it.`
`05:57.04`	`tzafrir`	`essentially it is a front to the config file, yes.`
`05:57.24`	`tzafrir`	`I said that it's a new feature of 1.4`
`05:58.15`	`rad07`	`When I say exclusive to GUI I mean only GUI can edit certain files and Asterisk don't use those in any way. I believe that GUI is just nice organised interface`
`05:58.53`	`rad07`	`Any book on Asterisk 1.4 with GUI included`
`06:01.07`	`mosty`	`rad07, no`
`06:01.57`	`rad07`	`mosty: I am just looking AsteriskNOW user Guide?`
`06:02.05`	`mosty`	`rad07, 1.4 is too new, and there is no standard gui`
`06:02.19`	`mosty`	`you might find some online docs, i doubt there are books`
`06:02.30`	`rad07`	`If I installed Asterisk 1.4 on Centos 5, Added Asterisk GUI Did I make my installation ASteriskNOW?`
`06:02.38`	`mosty`	`also, you should start in #asterisknow for that`
`06:02.50`	`*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)`
`06:02.52`	`mosty`	`asterisk itself does not include any gui's`
`06:04.04`	`rad07`	`If I installed Asterisk 1.4 on Centos 5, Added Asterisk GUI Did I make my installation ASteriskNOW? I mean can I follow AsteriskNOW user Guide?`
`06:04.42`	`Nuitari`	`rad07: you should ask in #asterisknow`
`06:05.08`	`rad07`	`That guide says that user information is kept in users.conf adn the trunk info in providers.conf. What about sip.conf that I read about int the book TFOT?`
`06:05.30`	`Nuitari`	`that's for sip devices`
`06:05.47`	`rad07`	`Nuitari: That channel seems to be dead. Nobody answers.`
`06:05.59`	`JT`	`1.4/ast gui stuffed around using users.conf`
`06:06.06`	`JT`	`you can choose not to use it`
`06:06.27`	`Nuitari`	`rad07: I don't use either the gui or asterisk now`
`06:06.45`	`rad07`	`I have Linksys SPA-3102 ATA. How can I setup this with Asterisk 1.4? Can I do it via GUI?`
`06:07.28`	`rad07`	`Nuitari: I have a clean installation Asterisk 1.4 and I just added Asterisk-GUI.`
`06:07.31`	`mosty`	`rad07, what gui did you install?`
`06:07.54`	`rad07`	`I can certainly use gedit or similar`
`06:08.00`	`JT`	`asterisk-gui it seems`
`06:08.03`	`rad07`	`yes`
`06:08.19`	`rad07`	`did it change anything in my ASterisk 1.4 installation?`
`06:09.19`	`*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)`
`06:09.23`	`Nuitari`	`rad07: http://forum.voxilla.com/linksys-sipura-voip-support-forum/`
`06:11.12`	`rad07`	`Guys I just want to learn and setup my system at the same time. I don't have any examples to follow. I cannot seem to rely on ASterisk 1.2 TFOT book? Can you recommend some good documentation for my case. Centos 5, Asterisk 1.4, Asterisk-GUI added, Linksys SPA-3102 ATA, No voip service provider for now, I wish to connect to Free World Dialup network and do all cool stuff later on. It is hard in the beginning as you know`
`06:11.13`	`*** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk)`
`06:11.20`	`mosty`	`rad07, there are several asterisk gui's. where did you download this software?`
`06:11.52`	`Nuitari`	`rad07: You should spend some time on voip-info.org, true some of the info will be outdated for 1.4, but it is how I started`
`06:12.07`	`Nuitari`	`rad07: it has plenty of device specific examples that are easy to follow`
`06:13.37`	`rad07`	`tx Nitari. I followed some Guide for my ATA and I was only able (with the help of one of you guys) to get calls on my analog phone attached to FXS port, but not to call out or receive calls via PSTN line`
`06:13.56`	`Nuitari`	`try more`
`06:14.19`	`*** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com)`
`06:14.26`	`rad07`	`I got calls from a software phone`
`06:14.37`	`Nuitari`	`you basically need the config for the device in sip.conf, then the entries in extensions.conf to connect to it`
`06:14.40`	`adelas`	`hey does anyone know a software, that will take sip, and take that line as fax?`
`06:15.04`	`mosty`	`adelas, fax over voip sucks ass`
`06:15.05`	`JT`	`no,`
`06:15.06`	`adelas`	`cause i'm tired of this buggy fax server software(winfax) and converting to anaglog line`
`06:15.15`	`adelas`	`well`
`06:15.17`	`JT`	`you don't do fax with voip`
`06:15.18`	`adelas`	`i have a T1 line`
`06:15.26`	`mosty`	`adelas, stick with the analoge line and use hylafax instead of winfax`
`06:15.29`	`JT`	`well that's slightly different`
`06:15.31`	`adelas`	`go to asterisk server`
`06:15.32`	`rad07`	`Nuitari: Can I do it via GUI`
`06:15.42`	`adelas`	`then splits out to phones already`
`06:16.00`	`adelas`	`i have a stinkin linksys converter to analog and using it :\|`
`06:16.00`	`mosty`	`adelas, i would recommend hylafax for fax, and asterisk for phones`
`06:16.03`	`JT`	`mosty: umm, asterisk-gui is put out by digium`
`06:16.14`	`JT`	`it's the name of the gui`
`06:16.16`	`adelas`	`um we don't have any analog lines`
`06:16.16`	`mosty`	`jt: is that asterisknow ?`
`06:16.18`	`adelas`	`just the t1`
`06:16.28`	`adelas`	`and linksys pata2 converter`
`06:16.29`	`JT`	`no, asterisknow is the distro containing centos + asterisk-gui`
`06:16.56`	`mosty`	`JT, ahh ok. they could have picked a more distinctive name :)`
`06:17.08`	`rad07`	`I have clean install of Asterisk 1.4. I first installed Centos 5 on my own and then installed ASterisk`
`06:17.12`	`mosty`	`jt: btw i just got calls working on one of my E1 lines!`
`06:17.28`	`JT`	`cool`
`06:17.31`	`JT`	`what was the issue?`
`06:17.40`	`mosty`	`rad07, it looks like the asterisknow docs should be a good starting point`
`06:17.55`	`adelas`	`mostly, hylafax for linux app?`
`06:18.07`	`mosty`	`jt: yesterday i got it to the point where it would dial but get disconnected by the remote end, they said i was sending too many digits`
`06:18.42`	`mosty`	`adelas, if you just want to use one channel from a E1 for fax and the rest for voip, you can safely use asterisk i think`
`06:19.11`	`rad07`	`mosty: Did I make my machine ASteriskNOW from the moment I installed Asterisk-GUI? Did that installation change anything that the regular Asterisk 1.4 installation did?`
`06:19.35`	`JT`	`mosty: that's what i thought`
`06:19.43`	`JT`	`mosty: i thought it was a digit sending issue`
`06:19.58`	`mosty`	`rad07, i don't know enough about asterisknow to answer. try #asterisknow`
`06:20.23`	`mosty`	`jt: compiling everything from scratch got me that far, then i was just dialling wrong :)`
`06:20.35`	`rad07`	`mosty: they don't answer. I am trying to get started you know.`
`06:20.45`	`adelas`	`mosty, hylafax, a linux based software?`
`06:20.52`	`adelas`	`cause i need something windows :\|`
`06:20.54`	`mosty`	`adelas, yes`
`06:21.01`	`adelas`	`was gonna try microsoft faxing`
`06:21.04`	`*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)`
`06:21.05`	`adelas`	`but.. its ms`
`06:21.14`	`adelas`	`with the sip converter`
`06:21.21`	`rad07`	`Do you guys use editing programs to edit *.conf files.`
`06:21.28`	`mosty`	`adelas, no, keep voip out of the picture when dealing with fax`
`06:21.30`	`JT`	`it needs to support T.38 to use an ATA`
`06:21.37`	`rad07`	`If you say so I am fine to use gedit`
`06:22.00`	`adelas`	`then how can i do this?`
`06:22.08`	`adelas`	`cause i only have a pri T1 card`
`06:22.18`	`adelas`	`to work off of`
`06:22.35`	`JT`	`pri is not voip`
`06:22.39`	`adelas`	`yea`
`06:22.44`	`adelas`	`but only way i can use it`
`06:22.51`	`adelas`	`is with asterisk server right now :\|`
`06:25.30`	`JT`	`look into spandsp`
`06:26.35`	`mosty`	`adelas, why do you need to use windows for fax?`
`06:27.13`	`rad07`	`JT, mosty, adelas: I would like to create basic dial plan for my Centos 5, Asterisk 1.4, Linksys SPA-3102 ATA and later I wish to connect to Free World Dialup network and VOIP providers. Which *.config files I should edit?`
`06:27.47`	`JT`	`extensions.conf sip.conf`
`06:28.16`	`tzafrir`	`rad07, I use vim`
`06:28.21`	`mosty`	`rad07, if you installed the gui use that`
`06:28.26`	`tzafrir`	`it has nice syntax hilighting`
`06:28.30`	`rad07`	`What about users.conf, providers.com`
`06:28.49`	`tzafrir`	`and it is a great editor altogether`
`06:29.10`	`rad07`	`mosty and tzafrir. Everybody discurages me from doing it?`
`06:29.28`	`tzafrir`	`learn to use vi. It pays`
`06:29.29`	`rad07`	`I cannot find a way to edit sip.conf via GUI`
`06:29.34`	`mosty`	`rad07, it's just that we don't usually support the gui's here`
`06:29.58`	`mosty`	`rad07, we deal with lower level details in this channel, mostly`
`06:30.22`	andrew`	`i think you should learn how to use a text editor before trying to learn asterisk :)`
`06:30.25`	`tzafrir`	`rad07, it depends for what`
`06:30.44`	`tzafrir`	`The GUI doesn't support anything`
`06:30.56`	`tzafrir`	`It is still a nice way of starting`
`06:31.09`	`rad07`	`I know these channels are dead. I don't insist on GUI, but since you mention them as being nice. So how to get to sip.conf? I know that there is an option to edit row files, but what about nice GUI form fields stuff`
`06:31.14`	`tzafrir`	`Actually at least the asterisk-gui is not as complex as some others`
`06:31.22`	`sergee`	`~seen puzzled`
`06:31.37`	`jbot`	`puzzled <n=patrick@puzzled.xs4all.nl> was last seen on IRC in channel #asterisk, 20h 9m 6s ago, saying: 'hey tzafrir. thanks for the patch. haven't yet tried it but will soon'.`
`06:31.37`	`sergee`	`!seen puzzled`
`06:32.30`	`rad07`	`tzafrir: Can you edit sip.conf on you ASterisk-Gui`
`06:33.43`	`rad07`	`I have menu options: User, Conferencing, Voicemail, Call Queues, Service Providers, Calling Rules, Voice Menues, Activer Channels? What options will edit sip.conf and extensions`
`06:33.56`	`tzafrir`	`rad07, the GUI edits it for you. Not you directly`
`06:34.48`	`rad07`	`tzafrir: but if I know that something needs to go in sip.conf how can I translate this into GUI option`
`06:35.15`	`tzafrir`	`maybe the gui will show it, maybe it won't. I'm not sure`
`06:35.24`	`tzafrir`	`I don't know it well enough`
`06:35.51`	`rad07`	`For example for my case: Asterisk 1.4, Linksys SPA-3102 ATA and later I wish to connect to Free World Dialup network and VOIP providers. What menu options I will need to deal with?`
`06:39.17`	`*** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net)`
`06:40.38`	`snuffy22`	`anyone good at 'sipp' xml writing?`
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`07:26.53`	`kova`	`anyone here has some experience with channel gtalk and jabber?`
`07:27.24`	`*** join/#asterisk svenna_ (n=svenna@p548D1017.dip0.t-ipconnect.de)`
`07:32.57`	`*** join/#asterisk Quintin (n=quintin@74-133-79-92.dhcp.insightbb.com)`
`07:33.11`	`Quintin`	`Quick, who should I sign up with to prototype asterisk?`
`07:33.21`	`Quintin`	`I was going to get with stanaphone, but they aren't open for new clients now`
`07:34.25`	`jql`	`freeworlddialup?`
`07:36.22`	`Quintin`	`ok, I was thinking about them`
`07:36.29`	`Quintin`	`Do you know the rates?`
`07:36.55`	`mvanbaak`	`it's for free`
`07:37.08`	`mvanbaak`	`read their webpage`
`07:37.16`	`Quintin`	`nooo`
`07:37.22`	`Quintin`	`I mean for connecting to the PSTN`
`07:37.31`	`*** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru)`
`07:37.32`	`mvanbaak`	`no idea`
`07:38.57`	`jql`	`prototype != pstn connectivity, imho`
`07:38.57`	`kova`	`Quintin: use Voipbuster`
`07:39.27`	`Quintin`	`kova: that will work with asterisk?`
`07:39.27`	`*** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it)`
`07:39.32`	`kova`	`Quintin: you can dial out to landlines for free in a lot of countries`
`07:39.49`	`Quintin`	`I don't have any regular telephone.`
`07:39.51`	`Quintin`	`only internet :)`
`07:39.59`	`kova`	`Quintin: is supposed to work as it is SIP`
`07:40.27`	`Quintin`	`hm`
`07:40.45`	`Quintin`	`I wonder if voipbuster client runs in wine`
`07:40.53`	`*** join/#asterisk Strom_M (n=strom@60-241-200-26.static.tpgi.com.au)`
`07:40.54`	`kova`	`Quintin: so now you can use internet to call regular phones .. isn't that nice`
`07:41.42`	`kova`	`why not just add SIP account in asterisk`
`07:42.10`	`kova`	`still no one here with gtalk experience in asterisk ?`
`07:42.33`	`Quintin`	`kova: what do you mean add sip account in asterisk? does digium provide sip?`
`07:50.23`	`*** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com)`
`07:52.37`	`creativx`	`oh my fuck i hate the moron who invented wrapmail`
`07:53.23`	`JT`	`voipbuster, eww`
`07:53.35`	`JT`	`Quintin: asterisk can talk sip`
`07:54.03`	`Nuitari`	`wrapmail ?`
`07:54.57`	`*** join/#asterisk neoalex (n=chatzill@user-0ccengj.cable.mindspring.com)`
`07:55.08`	`neoalex`	`does anyone know what port asterisk -r works on`
`07:55.16`	`JT`	`port?`
`07:55.20`	`neoalex`	`because I'm getting some weird msgs in the cli`
`07:55.25`	`JT`	`it's a cli application`
`07:55.45`	`neoalex`	`<PROTECTED>`
`07:55.48`	`neoalex`	`<PROTECTED>`
`07:55.49`	`neoalex`	`<PROTECTED>`
`07:55.51`	`neoalex`	`<PROTECTED>`
`07:55.58`	`Nuitari`	`it's a unix socket`
`07:56.00`	`neoalex`	`I get like 10 of those every once in a while`
`07:56.13`	`JT`	`you must have something connecting`
`07:56.19`	`JT`	`maybe a manager interface or something`
`07:56.31`	`neoalex`	`well... I don't... not using the manager (manager.conf)`
`07:56.37`	`neoalex`	`don't have freepbx`
`07:56.44`	`*** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com)`
`07:56.44`	`neoalex`	`just pure asterisk`
`07:56.47`	`JT`	`or a program that uses the cli to login`
`07:56.59`	`neoalex`	`nope... nothing like that either`
`07:57.04`	`JT`	`or even running commands via asterisk -rx will probably do that`
`07:57.06`	`JT`	`check cron`
`07:57.13`	`*** join/#asterisk Strom_M (n=strom@dsl-202-173-183-69.vic.westnet.com.au)`
`07:57.17`	`*** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk)`
`07:57.36`	`neoalex`	`by the way... It started when I upgraded from 1.4.4 to 1.4.5`
`07:58.01`	`JT`	`check cron`
`07:58.03`	`neoalex`	`nothing in cron either`
`07:58.07`	`neoalex`	`just looked`
`07:58.17`	`JT`	`there might be some new script somewhere`
`07:58.23`	`JT`	`well`
`07:58.29`	`JT`	`just activate the full log`
`07:58.36`	`JT`	`and see if anything happens when the logon occurs`
`07:59.06`	`neoalex`	`no... it happens every couple of minutes`
`07:59.22`	`JT`	`so have you checked full log?`
`07:59.41`	`neoalex`	`no... where is it saved... because it is enabled`
`07:59.52`	`JT`	`/var/log/asterisk`
`08:01.33`	`tzafrir_laptop`	`neoalex, use netstat or fuser to try to catch the other side of the unix socket in action?`
`08:01.43`	`neoalex`	`ok... there's nothing in messages log... how do I increase verbosity for the logs`
`08:01.50`	`JT`	`not messages`
`08:01.51`	`JT`	`full`
`08:01.58`	`JT`	`make sure full is enabled in logger.conf`
`08:02.00`	`KpoH`	`:)`
`08:02.02`	`*** join/#asterisk HarryR (n=Administ@host-83-146-53-46.bulldogdsl.com)`
`08:02.16`	`*** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com)`
`08:02.25`	`JT`	`and debug messages should be logged`
`08:02.36`	`*** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl)`
`08:02.49`	`*** join/#asterisk Daejeo1 (n=chatzill@124.62.150.49)`
`08:03.45`	`neoalex`	`so I should also enable debug?`
`08:04.05`	`neoalex`	`anyway full was off, turned it on now`
`08:04.18`	`neoalex`	`let me open a netstat like tzafrir said`
`08:04.30`	`Daejeo1`	`I am trying to call in from other box. I do not see anything on cli`
`08:04.45`	`KpoH`	`neoalex: tail -f /var/log/asterisk/full`
`08:05.09`	`neoalex`	`I'm looking in the cli now... no messages yet :D`
`08:05.22`	`neoalex`	`ok... there they go once`
`08:05.46`	`Daejeo1`	`any educated guess?`
`08:06.15`	`neoalex`	`increase verbosity Daejeo1`
`08:06.22`	`neoalex`	`either asterisk -vvvvvvvvr`
`08:06.26`	`*** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk)`
`08:06.37`	`Daejeo1`	`neoalex: 15`
`08:06.39`	`neoalex`	`or core set verbose 10 or whatever you like`
`08:07.11`	`neoalex`	`that also happens when the extension you're trying to call is not in extensions.conf`
`08:07.28`	`neoalex`	`does the call go through?`
`08:07.34`	`Daejeo1`	`it is in the extensions`
`08:07.44`	`Daejeo1`	`I can call out`
`08:07.54`	`Daejeo1`	`but unable to call in`
`08:09.02`	`walhala`	`io`
`08:09.07`	`neoalex`	`ok... the context set in the general section of sip.conf is the context that will contain the incoming extensions... is the extension you are trying to call there?`
`08:09.50`	`neoalex`	`tzafrir what should I look for in netstat?`
`08:10.38`	*** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com)
`08:10.41`	`Daejeo1`	`let me paste whatever I have in sip and extension`
`08:10.50`	`neoalex`	`pastebin`
`08:13.44`	`neoalex`	`JT it doesn't happen when I connect with asterisk -r`
`08:13.59`	`neoalex`	`I mean I see one message as I should`
`08:15.24`	`Uatec`	`hey, does anybody know where i can get sales type paperwork for asterisk and for asterisk based phone systems?`
`08:16.06`	`Daejeo1`	`neoalex: http://www.pastebin.ca/575922`
`08:17.46`	`*** join/#asterisk saftsack (n=oliver@p54A7E41F.dip.t-dialin.net)`
`08:18.57`	`neoalex`	`ok... in the general section of sip.conf you should have context=internal so you can accept the call 300@yourasterisk`
`08:19.27`	`Daejeo1`	`yes, but I am unable to receive`
`08:19.59`	`neoalex`	`I know... that's what I'm saying accept=receive... same thing :D`
`08:20.00`	`Daejeo1`	`i am trying to call sip/3000@mybox from other box`
`08:20.29`	`Daejeo1`	`sorry 300`
`08:21.05`	`*** join/#asterisk matsk (n=mk@194.68.102.173)`
`08:21.47`	`neoalex`	`I know... that's what you need... put context=internal in general in sip.conf`
`08:22.18`	`Daejeo1`	`i have context=incoming in general (sip.conf)`
`08:22.57`	`neoalex`	`I don't see it in the stuff you pasted me`
`08:23.08`	`neoalex`	`but if so do you also have context called incoming in extensions`
`08:23.16`	`neoalex`	`cause I only see one called internal`
`08:25.59`	`neoalex`	`JT... ok those messages are exactly 5 minutes apart`
`08:26.17`	`JT`	`well it's something accessing it`
`08:26.28`	`JT`	`maybe a daemon`
`08:26.39`	`neoalex`	`there's gotta be something in 1.4.5`
`08:26.55`	`neoalex`	`but I don't know what the heck it could be`
`08:27.48`	`Daejeo1`	`neoalex`
`08:28.00`	`Daejeo1`	`it is working now`
`08:28.03`	`Daejeo1`	`thank you`
`08:28.08`	`neoalex`	`no prob.`
`08:28.18`	`neoalex`	`glad to help`
`08:28.44`	`Daejeo1`	`:)`
`08:30.15`	`*** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk)`
`08:31.23`	`neoalex`	`JT this is weird`
`08:36.47`	`*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)`
`08:37.12`	`neoalex`	`ok... fixed it... I think, asterisk was being started by a script in /etc/init.d/asterisk`
`08:37.23`	`neoalex`	`but something else also starts it`
`08:37.31`	`JT`	`like safe_asterisk?`
`08:37.42`	`neoalex`	`because I disabled that script and it still started at boot time`
`08:37.57`	`neoalex`	`I don't know... trying to find it now`
`08:38.06`	`*** join/#asterisk saftsack (n=oliver@p54A7DE15.dip.t-dialin.net)`
`08:38.35`	`neoalex`	`I remember I put something to start asterisk and disabled the script now it was back from the upgrade`
`08:38.49`	`neoalex`	`but I don't remember how I started it in the first place`
`08:39.19`	`Daejeo1`	`NOTICE[4528]: rtp.c:783 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 145.998.999.00`
`08:39.52`	`neoalex`	`look in the settings of your sofphone, hardphone ATA, whatever bill is using to connect to asterisk`
`08:40.07`	`Daejeo1`	`eyebeam`
`08:40.23`	`neoalex`	`ah ok... it is in device settings I believe`
`08:40.29`	`JT`	`wtf`
`08:40.33`	`JT`	`Client IP: 145.998.999.00`
`08:40.35`	`JT`	`...`
`08:40.36`	`neoalex`	`preserve bandwidth during silence periods`
`08:40.38`	`JT`	`impossible ip`
`08:40.49`	`neoalex`	`ha... didn't notice`
`08:41.11`	`neoalex`	`yeah... you only see those in crappy movies`
`08:41.14`	`Daejeo1`	`JT: i just typed`
`08:41.26`	`cy303`	`yo`
`08:41.39`	`neoalex`	`ah... ok`
`08:42.22`	`neoalex`	`Daejeo1: it is in options->advanced->network->preserve bandwidth during silence periods`
`08:42.26`	`neoalex`	`uncheck that`
`08:42.30`	`*** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk)`
`08:43.10`	`Daejeo1`	`it is already unchecked`
`08:44.05`	`creativx`	`comfort noise`
`08:44.10`	`neoalex`	`ok... now... what about the other client?`
`08:44.12`	`creativx`	`keeps annoying me too`
`08:44.35`	`Daejeo1`	`tivi`
`08:44.58`	`neoalex`	`never worked with it but look for the same setting some where`
`08:45.02`	`Sean	`~tfot`
`08:45.05`	`jbot`	`i guess tfot is "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details`
`08:45.11`	`neoalex`	`something similar`
`08:45.17`	`Sean	`~thebook`
`08:45.18`	`jbot`	`i guess thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`08:47.23`	`Polis_ttt`	`that's a very good book, i got it both in pdf and paper, a lot of useful information in it`
`08:48.34`	`neoalex`	`ok... found it... I was starting asterisk with initab`
`08:48.42`	`neoalex`	`forgot about that`
`08:48.48`	`neoalex`	`anyway works fine now`
`08:50.02`	`neoalex`	`ok... that's it for me`
`08:50.10`	`neoalex`	`ttyl`
`08:55.09`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`08:55.50`	`*** join/#asterisk MAGNET\|C (n=asd@213.132.241.78)`
`08:56.41`	`MAGNET\|C`	`need help with connecting * with Meridian SL1 PBX ...`
`08:58.40`	`*** join/#asterisk saftsack (n=oliver@p54A7CED3.dip.t-dialin.net)`
`09:01.52`	`*** join/#asterisk casix (n=casix@edifici-pub.adam.es)`
`09:01.55`	`casix`	`hello`
`09:02.10`	`casix`	`I've a problem with asterisk 1.4.5 and cdr`
`09:02.23`	`casix`	`when I make a cdr submit asterisk crash`
`09:02.59`	`casix`	`but there are no error logs`
`09:14.01`	`casix`	`any idea?`
`09:17.11`	`sergee`	`any MeetMe users around?`
`09:18.13`	`*** join/#asterisk saftsack (n=oliver@p54A7C1AC.dip.t-dialin.net)`
`09:18.41`	`*** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com)`
`09:35.50`	`*** join/#asterisk jacq (n=jal@203.187.143.130)`
`09:38.11`	`*** join/#asterisk saftsack (n=oliver@p54A7E43A.dip.t-dialin.net)`
`09:39.37`	`tzafrir_laptop`	`sergee, what type of "meetme users"?`
`09:39.39`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`09:40.36`	`sergee`	`tzafrir: anybody who is using 'I' option for MeetMe, and who's willing to test patch for it (#9430)`
`09:40.40`	`sergee`	`:)`
`09:40.47`	`sergee`	`tzafrir_laptop: anybody who is using 'I' option for MeetMe, and who's willing to test patch for it (#9430)`
`09:44.18`	`tzafrir_laptop`	`I don't use that option normally...`
`09:51.32`	`A-data`	`<PROTECTED>`
`09:52.19`	`*** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au)`
`09:57.55`	`*** join/#asterisk saftsack (n=oliver@p54A7F85E.dip.t-dialin.net)`
`09:58.33`	`mosty`	`is there a way to give the caller a different ringing sound if the destination is already on a call?`
`09:58.46`	`*** join/#asterisk Kenoby (n=hmng@195-23-23-14.net.novis.pt)`
`10:02.27`	`creativx`	`ya`
`10:02.41`	`creativx`	`check ${dialstatus} after dial()`
`10:03.08`	`mosty`	`that won't work, the ringing sound happens before dial() exits`
`10:03.20`	`creativx`	`1, ringing`
`10:03.22`	`creativx`	`2, dial()`
`10:03.32`	`creativx`	`3,goto s-dialstatus`
`10:03.43`	`creativx`	`pseudo`
`10:04.17`	`mosty`	`i don't see how that will change what happens during the dial command. these phones support multiple lines`
`10:04.59`	`creativx`	`ah`
`10:05.19`	`cy303`	`can macro names have variables in them?`
`10:05.20`	`creativx`	`then im not sure, i would assume the phone will always be available as long as it has atleast 1 line free`
`10:05.27`	`cy303`	`[macro-name${VAR}] ?`
`10:06.35`	`mosty`	`creativx, yes but asterisk knows the destination is already on a call and asterisk generates the ringing tone. i think it's a setting in indications.conf`
`10:07.58`	`creativx`	`mosty: im sorry, dont have any good suggestions at hand`
`10:08.54`	`*** join/#asterisk yassaccan (n=yassacca@admin192.hgo.se)`
`10:08.58`	`mosty`	`thanks anyway`
`10:13.49`	`*** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose)`
`10:15.02`	`*** join/#asterisk saftsack (n=oliver@p54A7F71F.dip.t-dialin.net)`
`10:21.49`	`casix`	`I've a problem with asterisk 1.4.5 and cdr`
`10:21.53`	`casix`	`when I make a cdr submit asterisk crash`
`10:21.56`	`casix`	`but there are no error logs`
`10:21.59`	`casix`	`any idea?`
`10:23.06`	`cy303`	`damn 1.4.5 is out? :P`
`10:24.18`	`casix`	`yes`
`10:24.20`	`casix`	`it is`
`10:24.30`	`casix`	`from yesterday I think`
`10:31.47`	`*** join/#asterisk basty (n=basty@212.218.65.199)`
`10:31.48`	`basty`	`Hi`
`10:32.04`	`basty`	`I am having a Problem with "exten => _XXX,3,Gotoif(${BLINDTRANSFER}=""?4:5)" Anyone knows why ?`
`10:33.22`	`casix`	`basty: I think it have to be like "GotoIf($[${BLINDTRANSFER}=""]?4:5)"`
`10:35.19`	`*** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr)`
`10:35.36`	`basty`	`casix: Hmm: Jun 19 12:35:13 WARNING[9657]: ast_expr2.y:815 op_div: non-numeric argument -> Executing GotoIf("SIP/22-a96f", "0?4:7") in new stack`
`10:36.50`	`creativx`	`GotoIf($["${BLINDTRANSFER}"=""]?4:5)`
`10:36.52`	`creativx`	`try that`
`10:37.05`	`*** join/#asterisk friedrich\| (n=friedric@e177246208.adsl.alicedsl.de)`
`10:37.19`	`creativx`	`encapsulate it in quotes to do a textual comparison`
`10:37.47`	`basty`	`creativx: Still the same "Executing GotoIf("SIP/22-5e2a", "0?4:7") in new stack"`
`10:39.50`	`casix`	`basty this is ok`
`10:40.00`	`basty`	`but why is it "0" ?`
`10:40.19`	`casix`	`if the evaluation of the condition`
`10:40.31`	`casix`	`0 false`
`10:40.33`	`casix`	`1 true`
`10:40.39`	`casix`	`I thing`
`10:41.14`	`basty`	`Mhh..okay..but it should fail...because I am trying to do a callback when doing a blindtransfer.`
`10:41.23`	`basty`	`s/should/shouldnt`
`10:41.49`	`basty`	`I am trying the example on: http://voip-info.moltentelecom.com/wiki/index853004e33a6db40ecf5469e0344eddf6.html?comment_page=1&page_id=1483&maxComments=10&comments_maxComments=10&comments_sort_mode=commentDate_desc&comments_style=flat`
`10:42.19`	`creativx`	`gotoif($[ "${BLINDTRANSFER}" = "" ]?truelabel:falselabel)`
`10:42.28`	`casix`	`basty: debug using NoOp(${BLINDTRANSFER}) to see if it is blank or not`
`10:42.32`	`creativx`	`that should work`
`10:44.15`	`basty`	`creativx: Thanks..but it still doesnt.. ;-( Jun 19 12:43:19 WARNING[9875]: pbx.c:6451 ast_parseable_goto: Priority 'falselabel' must be a number > 0, or valid label`
`10:44.39`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`10:45.04`	`creativx`	`hehe my bad`
`10:45.12`	`creativx`	`change falselabel to 7 and truelabel to 4 accordingly`
`10:45.22`	`creativx`	`i use named extensions and not number when doing gotos`
`10:45.24`	`creativx`	`easier to read`
`10:45.31`	`basty`	`ceasix: Executing NoOp("SIP/22-6798", "SIP/21-0949") in new stack`
`10:45.52`	`basty`	`creati: oh sorry..my fault.. :) doh`
`10:47.24`	`cy303`	`:w`
`10:48.34`	`*** join/#asterisk saftsack (n=oliver@p54A7E02F.dip.t-dialin.net)`
`10:49.22`	`creativx`	`well did it execute basty=`
`10:50.08`	`basty`	`yeah well..but now it seems that I have another, different problem ;)`
`10:50.37`	`creativx`	`welcome to asterisk`
`10:50.40`	`*** join/#asterisk ghenry (n=ghenry@212.159.59.85)`
`10:50.41`	`creativx`	`you fix one problem, get atleast two nwe`
`10:50.41`	`casix`	`hehehe`
`10:50.42`	`creativx`	`new`
`10:50.42`	`creativx`	`:)`
`10:51.08`	`basty`	`yeah...`
`10:51.14`	`basty`	`now it looks like that:`
`10:51.15`	`basty`	`-- Executing GotoIf("SIP/22-a4c8", "0?4:7") in new stack -- Goto (transfercontext,18,7) -- Executing Dial("SIP/22-a4c8", "\|\|mTt") in new stack -> Jun 19 12:50:32 WARNING[10045]: app_dial.c:794 dial_exec_full: Dial requires an argument (technology/number)`
`10:52.34`	`basty`	`this example is kinda strange...nothing really works... ;-)`
`10:52.41`	`basty`	`the example on the url i posted before`
`10:58.36`	`creativx`	`dial(\|\|mTt) wouldnt help you much`
`10:59.13`	`creativx`	`dial(tech/ext\|\|options) would`
`11:00.01`	`basty`	`Mh..yeah I know..but why does it not send the extension into the variable ? I mean...i thought it was a working example on the page ;)`
`11:03.30`	`creativx`	`i havent seen the example`
`11:03.32`	`creativx`	`so i dunno`
`11:08.26`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`11:08.47`	`*** join/#asterisk saftsack (n=oliver@p54A7E346.dip.t-dialin.net)`
`11:10.03`	`*** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net)`
`11:15.08`	`*** join/#asterisk gardo (n=gardo@121.97.194.235)`
`11:17.01`	`A-data`	`every time i update somthing in SIP.conf do i have to reload or it auto work`
`11:19.06`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`11:20.01`	`casix`	`A-data: you have to reload or sip reload`
`11:25.47`	`*** join/#asterisk Nobbie (n=anony@fwb003.fw.is.co.za)`
`11:26.36`	`the_5th_wheel`	`hi. say i have a isdn premicell, would i be able to connect that to an NT1, with no line going to the telecoms, and use it as aisdn to pots converter?`
`11:27.13`	`*** join/#asterisk saftsack (n=oliver@p54A7FDBB.dip.t-dialin.net)`
`11:34.45`	`tzafrir_laptop`	`why not connect the ISDN directly?`
`11:34.59`	`tzafrir_laptop`	`what would you do with the POTS?`
`11:38.23`	`casix`	`there is any problem with asterisk 1.4 and cdr??`
`11:46.57`	`Teccy`	`the_5th_wheel: the answer is no`
`11:47.42`	`*** join/#asterisk saftsack (n=oliver@p54A7EF25.dip.t-dialin.net)`
`11:48.42`	`*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)`
`11:50.28`	`the_5th_wheel`	`blast`
`11:52.19`	`the_5th_wheel`	`mmm, ive heard of some isdn' modems' that have an analouge port on them, would i be able to use that as an fxs line?`
`11:54.24`	`JT`	`the_5th_wheel: so let's try and answer the question this time, why don't you connect isdn directly to the server?`
`11:55.43`	`the_5th_wheel`	`sorry, didnt see that last time. Well, i would like one plain analouge line for my faxmachine.`
`11:56.15`	`the_5th_wheel`	`and im running on a seemingly nonexisten budget, so i need to uyse the bare minimum`
`11:57.42`	`*** join/#asterisk zapp-branigan (n=zapp-bra@141.Red-83-44-133.dynamicIP.rima-tde.net)`
`12:00.35`	`*** join/#asterisk javar (n=javar@69.79.134.24)`
`12:03.30`	`A-data`	`exten => t,1,Playback(vm-goodbye); <<< can any one please tell me what t here represent?`
`12:05.54`	`*** join/#asterisk saftsack (n=oliver@p54A7F39E.dip.t-dialin.net)`
`12:06.09`	`b00gz`	`is 250ms of lag going to cause issues with VoIP phone calls?`
`12:06.40`	`sergee`	`A-data: 't' - special extension for timeout`
`12:07.36`	`HarryR`	`b00gz: it should be just about ok`
`12:07.54`	`sergee`	`A-data: http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf`
`12:07.57`	`HarryR`	`it takes a bit of getting used to the lag`
`12:08.08`	`sergee`	`A-data: http://www.voip-info.org/wiki/index.php?page=Asterisk+t+extension`
`12:09.02`	`A-data`	`ok sergee i am reading TOFT but it was not clear the t so thanks every one i have one more question i will paste it in site and say it`
`12:10.14`	`A-data`	http://paste-it.net/2602 <<< when i dial 5055 it auto hang up after finish it don`t wait for time out also if invalid response it don`t say invalid
`12:10.41`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`12:12.03`	`JT`	`A-data: is english your first language?`
`12:12.30`	`Polis_ttt`	`I got a nice little problem: "You do not appear to have the sources for the 2.6.17-10-server kernel installed." on my ubuntu6.10-lamp-server. I do got that kernel installed, but get this error anyway when i use 'make' for zaptel1.2.12 :( What can i do?`
`12:14.22`	`[TK]D-Fender`	`Polis_ttt: Its not the kernel SOURCE you need, its the HEADERS.`
`12:14.26`	`*** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar)`
`12:14.39`	`*** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net)`
`12:15.12`	`Chris-NB`	`anyone uses astlinux?`
`12:19.13`	`purplet`	`hi, does anyone know the meaning/cause of these messages: "Internal RTCP NTP clock skew detected" ? I get them on the CLI on SIP -> PSTN (PRI) calls... Google isnt very helpful to me...`
`12:20.46`	`mosty`	`purplet, are you running an ntp server?`
`12:21.10`	`*** part/#asterisk jmls (n=jmls@62.49.235.130)`
`12:21.30`	`purplet`	`mosty: not on the asterisk server... But the asterisk server is daily synced with ntpdate ...`
`12:21.56`	`*** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com)`
`12:22.18`	`mosty`	`purplet, you could try running an ntpd instead, it will sync much more often and skew less`
`12:22.39`	`*** join/#asterisk indend7 (n=indileos@220.227.46.41)`
`12:22.41`	`mosty`	`or ntpdate more often`
`12:22.59`	`purplet`	`aha, ok i will try that`
`12:23.09`	`purplet`	`but what does it mean exactly?`
`12:23.14`	`A-data`	`jt no`
`12:23.24`	`purplet`	`time difference between server and clients or something?`
`12:23.44`	`mosty`	`purplet, skew is when the clock drifts away from the correct time`
`12:24.02`	`A-data`	`JT no but is my english not clear`
`12:25.00`	`purplet`	`mosty: ok, how often do you recommend a ntp sync?`
`12:25.42`	`mosty`	`purplet, how often do you get the warnings?`
`12:26.02`	`HarryR`	`Is anybody using the jabber/jingle extension for Asterisk? I have a few questions about how I could pass parameters to scripts`
`12:26.18`	`*** join/#asterisk saftsack (n=oliver@p54A7E5ED.dip.t-dialin.net)`
`12:26.22`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`12:26.27`	`HarryR`	`e.g. jingle call comes in to asterisk, is there any way I can pass a unique ID to asterisk ?`
`12:27.19`	`kippi`	`whats the best verison of asterisk to get 1.45 or 1.2.19 ?`
`12:27.39`	`mosty`	`harryr: if it's the same as every other channel type, just make those calls start in a particular context, and use dialplan magic`
`12:27.40`	`HarryR`	`If you need compatibility with 1.2x series - 1.2.19 - otherwise go with 1.4.x`
`12:27.56`	`mosty`	`kippi: 1.2 is more stable, has less bugs. 1.4 has newer features`
`12:27.58`	`HarryR`	`mosty: yah but I need to pass a unique ID from the jabber server to asterisk`
`12:28.04`	`b00gz`	`HarryR, if I have 250ms of lag should I use gsm instead of ulaw?`
`12:28.18`	`kippi`	`1.4.x connect back to a 1.2.x version using IAX etc?`
`12:28.20`	`mosty`	`harryr: unique id for what? for the account? see what i said before`
`12:28.23`	`*** join/#asterisk mmagik (n=mmagic@host81-149-128-104.in-addr.btopenworld.com)`
`12:28.29`	`HarryR`	`mosty: no, for the call`
`12:28.55`	`mosty`	`HarryR, that is simple, do it where the call enters the dialplan`
`12:29.09`	`mmagik`	`hi everyone... if i get "Got I-frame while link state 2" using a loopback cable into my te110p.... is this normal?`
`12:29.13`	`HarryR`	`basically, jingle user calls 4412312341234@whatever.com, jabber server redirects that to the asterisk user`
`12:29.19`	`HarryR`	`asterisk routes it to pstn..`
`12:29.42`	`HarryR`	`bah i'll experement instead of asking questions`
`12:31.02`	`purplet`	`mosty: during a call i get them like every 5 seconds....`
`12:31.27`	`mosty`	`HarryR, if you ignore the fact that it's jabber/jingle instead of something like sip or iax, there's nothing special or unusual about that`
`12:31.43`	`HarryR`	`yah I figured`
`12:31.52`	`mosty`	`purplet, how much does ntpdate update your clock by approximately when you run it once a day?`
`12:31.54`	`HarryR`	`hopefully I can get jingle calls routing through out existing billing stuff`
`12:32.20`	`purplet`	`let me check`
`12:32.34`	`mosty`	`HarryR, the cdr record should have a unique number already, just use that?`
`12:32.38`	`*** join/#asterisk kombi (n=kombi@195.158.185.196)`
`12:33.06`	`purplet`	`mosty: about 3 seconds`
`12:33.39`	`A-data`	http://paste-it.net/2602 <<< when i dial 5055 it auto hang up after finish playing the first voice file it don`t wait for time out also if invalid response it don`t say invalid
`12:33.52`	`mosty`	`purplet, that's not very much over the course of an entire day. i say ignore the warnings until it causes problems with calls`
`12:34.13`	`kombi`	`for Music on hold, do you just stick mp3 files into mohmp3 and tweak extensions.conf?`
`12:35.02`	`kombi`	`because it says "Started music on hold.." but nothing can be heard..`
`12:35.34`	`creativx`	`convert them first`
`12:35.44`	`kombi`	`into what?`
`12:35.45`	`creativx`	`let asterisk play them in the native format`
`12:36.09`	`purplet`	`mosty: ok, thanks. What kind of problems can it cause? When I know I might recognize them if they occure ;)`
`12:36.24`	`*** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru)`
`12:36.30`	`kombi`	`creativx: 8khz 8bit ulaw?`
`12:36.41`	`mosty`	`purplet, i'd imagine dropped voice packets`
`12:37.01`	`[TK]D-Fender`	`kombi: You need to have aserisk-addons installed adn your MP3's should be 128kbps non VBR`
`12:37.02`	`creativx`	`kombi: theres some great info about that`
`12:37.05`	`creativx`	`on.... eh.. what its called`
`12:37.31`	`kombi`	`thanks people, some tweaking to do there..`
`12:37.40`	`[TK]D-Fender`	`A-data: You should not be running IVR's of of anything but the "s" exten.`
`12:38.11`	`mosty`	`[TK]D-Fender, do you know if it's possible to give a caller a different ringing tone if the destination already has an active call? ie so they know the person they're calling is on the phone, while allowing call waiting`
`12:38.21`	`[TK]D-Fender`	`A-data: And for that even you need to set "autofallthrough=no" in [general]`
`12:38.29`	`purplet`	`mosty: thanks! I'll keep an eye on it :)`
`12:38.47`	`creativx`	`kombi: asteriskguru.com i think it was`
`12:38.53`	`[TK]D-Fender`	`mosty: depends on the phone and I don't know of ANY offhand that would support such a thing.`
`12:39.01`	`*** join/#asterisk mkl1525 (n=qwertz@i59F7136B.versanet.de)`
`12:39.14`	`[TK]D-Fender`	`mosty: Oh wait... the CALLER.`
`12:39.22`	`creativx`	`mosty: http://astrecipes.net/?n=152`
`12:39.32`	`mosty`	`[TK]D-Fender, yes. the callee is done in indications.conf`
`12:39.43`	`[TK]D-Fender`	`mosty : dirty trick : use "m" and set a different MoH class with the sound you want, looped`
`12:39.48`	`mosty`	`but i want the caller to hear a different ring if the callee is already on the line`
`12:40.14`	`mkl1525`	`Hi, does anybody know how to enable the auto answer function on snom phones? found the "Auto Answer:" but there's no enable/disable only config stuff - any hint?`
`12:40.19`	`mosty`	`[TK]D-Fender, to do that i suppose i'd need to use agi to find out if the destination is in use already?`
`12:40.32`	`[TK]D-Fender`	`mosty: "show application chanisavail"`
`12:41.05`	`creativx`	`heh [TK]D-Fender, beat me by 2 seconds there`
`12:41.07`	`mosty`	`ahh, interesting`
`12:41.10`	`mosty`	`thanks`
`12:41.16`	`[TK]D-Fender`	`mkl1525: tahts not how it works. the serve sets a header when sending a call to the phone to TELL it to auto-answer.`
`12:41.38`	`*** join/#asterisk dijungal (n=kdaniel@64.86.52.254)`
`12:42.30`	`creativx`	`mkl1525: some use "answer-after" header, some "alert-info"`
`12:42.42`	`creativx`	`mkl1525: my ip10s obeys the answer-after header, but x-lite doesnt`
`12:43.04`	`creativx`	`[TK]D-Fender: would chanisavail(sip/587) work or would it have to be a qualified channel name?`
`12:43.05`	`dijungal`	`hello i was following the instructions here "http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/doc-svn6.txt" to install the intel based g729 codec on asterisk.. but the files it's asking me to download are not there.. any idea where i can find them?`
`12:43.15`	`mosty`	`mkl1525, there's a page on the wiki about it`
`12:43.18`	`[TK]D-Fender`	`creativx: X-Lite doesn't OFFER that feature and a ton of other stuff, thats why its teh FREE version.`
`12:43.31`	`creativx`	`[TK]D-Fender: x-lite does not offer it in any versions, since its not part of the RFC`
`12:43.38`	`[TK]D-Fender`	`creativx: And few phones use a common header style for that anyways.`
`12:43.38`	`creativx`	`i told you this last time nagged about it ;)`
`12:43.59`	`creativx`	`those headers are mostly hardware phone manufacturers imaginative ways of solving the autoanswer problem`
`12:44.15`	`mkl1525`	`creativx, [TK]D-Fender thanks for the info will have a look at the wiki`
`12:44.32`	`*** join/#asterisk jmacz (n=jmacz@190.24.103.191)`
`12:44.49`	`mosty`	`mkl1525, SIPAddHeader(Call-Info: <sip:something>\;answer-after=0)`
`12:45.12`	`dijungal`	`how do i ontain and install the g729 codec?`
`12:46.01`	`[TK]D-Fender`	`dijungal: Do NTO expect any help with this "free" version. It is questionably legal, and considered very innapropriate.`
`12:46.02`	`mosty`	`dijungal, see the digium website`
`12:46.11`	`[TK]D-Fender`	`NOT*`
`12:46.12`	`[TK]D-Fender`	`a;asklfd;ljdkl;f`
`12:46.34`	`dijungal`	`no prob.s`
`12:46.34`	`creativx`	`np`
`12:46.45`	`kombi`	`[TK]D-Fender: what is non VBR in mp3 files?`
`12:46.51`	`creativx`	`kombi: variable bitrate`
`12:46.54`	`[TK]D-Fender`	`creativx: My spelling is fine, my ability to type this morning is in serious doubt however`
`12:46.57`	`kombi`	`thanks!`
`12:47.08`	`tzanger`	`morning [TK]D-Fender`
`12:47.09`	`creativx`	`[TK]D-Fender: you could rename it to typecheck.exe, it will do the trick :)`
`12:47.13`	`tzanger`	`still having trouble with the typing, eh? :-)`
`12:47.18`	`*** join/#asterisk saftsack (n=oliver@p54A7F4FA.dip.t-dialin.net)`
`12:47.37`	`[TK]D-Fender`	`tzanger: Not doubt aboot it! ;)`
`12:47.41`	`tzanger`	`heh`
`12:47.55`	`mkl1525`	`mosty, thanks will try it`
`12:47.59`	`dijungal`	`awww.. it's only$10 on digum... if i buy that one can it work on asterisk?`
`12:48.11`	`tzanger`	`[TK]D-Fender: any recommendations on a SIP ATA to do paging? Anything with an audio out and relay I guess`
`12:48.19`	`mosty`	`dijungal, yes. $10 USD for one call`
`12:48.28`	`dijungal`	`huh..??? 1 call`
`12:48.37`	`mosty`	`one call at a time`
`12:48.42`	`dijungal`	`damn`
`12:48.51`	`tzanger`	`we've got the amp, and I imagine Bogen makes the POTS version... maybe I don't even need audio out in that case, just a cheap SIP FXS ATA`
`12:48.55`	`creativx`	`man i dont understand why i didnt use a softphone before.. now i can listen to mp3s inbetween`
`12:49.18`	`dijungal`	`where can i get one that allows for more than 1 call`
`12:49.41`	`mosty`	`dijungal, buy two licences`
`12:50.00`	`mosty`	`then you can have two g729 calls at a time`
`12:50.49`	`creativx`	`but 15, you get 15 concurrent calls!`
`12:51.02`	`creativx`	`damnit [TK]D-Fender, your typing has negative impact on mine`
`12:52.55`	`*** part/#asterisk javar (n=javar@69.79.134.24)`
`12:53.24`	`flenders`	`mosty, sorted out those damn PRIs yet?`
`12:54.04`	`mosty`	`flenders, i have one working :) the other keeps losing the D channel`
`12:54.05`	`*** join/#asterisk javar (n=javar@69.79.134.24)`
`12:54.39`	`flenders`	`and have they tested that one too?`
`12:54.44`	`mosty`	`flenders, the one i had mostly working yesterday afternoon was just a prefix issue, it didn't want the leading 0`
`12:55.09`	`mosty`	`i haven't had the other one tested yet, i need to double check my settings between the one that works and the one that doesn't`
`12:55.25`	`flenders`	`it wouldn't dial out because you had a 0?`
`12:55.38`	`zapp-branigan`	`someone has used g729 in xscale ?`
`12:55.52`	`A-data`	`i am stucked in variables i made this in Global`
`12:55.52`	`A-data`	`Mohamed=SIP/6062;`
`12:55.52`	`A-data`	`and in context`
`12:55.52`	`A-data`	`exten => 3,1,Dial(${Mohamed},10);`
`12:55.52`	`A-data`	`in console i get`
`12:55.53`	`A-data`	`== CDR updated on SIP/6060-09b10f40`
`12:55.55`	`A-data`	`<PROTECTED>`
`12:55.57`	`A-data`	`[Jun 19 15:49:57] WARNING[23242]: app_dial.c:838 dial_exec_full: Dial requires an argument (technology/number)`
`12:56.00`	`A-data`	`<PROTECTED>`
`12:56.02`	`A-data`	`and then it hangup`
`12:56.06`	`flenders`	`A-data: heard of pastebin?`
`12:56.13`	`mosty`	`flenders, correct`
`12:56.16`	`zapp-branigan`	`<PROTECTED>`
`12:56.28`	`flenders`	`mosty: that's a weird problem mate`
`12:56.32`	`mosty`	`zapp-branigan, does digium even provide it for that?`
`12:56.42`	`mosty`	`flenders, easy enough to fix though`
`12:57.13`	`flenders`	`so, dialing a 02XXXXXXXX number wouldn't work?`
`12:57.41`	`flenders`	`or you mean, a zero like 'dial zero to dial out'?`
`12:57.41`	`zapp-branigan`	`mosty intel offer g729 library for xscale`
`12:58.02`	`zapp-branigan`	`the ipp libraries`
`12:58.35`	`[TK]D-Fender`	`A-data: pastebin your dialplan.`
`12:58.36`	`mosty`	`zapp-branigan, but that does not include the g279 patent licence does it?`
`12:58.55`	`JT`	`mosty: so what was the prefix problem exactly?`
`12:58.55`	`zapp-branigan`	`you pay a license for intel`
`12:59.03`	`zapp-branigan`	`or 30 days trial`
`12:59.18`	`JT`	`mosty: you can't call numbers normally, like 03XXXXXXXX`
`12:59.30`	`JT`	`or 04XXXXXXXX`
`12:59.33`	`mosty`	`zapp-branigan, that is a licence for intel's ipp library, it does not include a licence for the g729 patent`
`12:59.47`	`zapp-branigan`	`i do n't know`
`12:59.59`	`Strom_M`	`JT: where in australia are you again?`
`13:00.01`	`flenders`	`JT: I was gonna txt you, but thought it was too late`
`13:00.10`	`mosty`	`jt: correct, it wants 3XXXXXXXX or 4XXXXXXXX`
`13:00.13`	`JT`	`Strom_M: sydney`
`13:00.17`	`Strom_M`	`ah ok`
`13:00.20`	`JT`	`mosty: dodgy second rate telco :P`
`13:00.21`	`flenders`	`mosty: that's odd`
`13:00.24`	`Strom_M`	`i'm in melbs this week :)`
`13:00.29`	`JT`	`nice`
`13:00.33`	`JT`	`not coming to syd?`
`13:00.42`	`Strom_M`	`no; just connecting through on the way home`
`13:00.43`	`mosty`	`Strom_M, cold enough for you?`
`13:00.45`	`JT`	`ah`
`13:00.47`	`Strom_M`	`bleh`
`13:00.51`	`Strom_M`	`grey, wet, cold`
`13:00.55`	`Strom_M`	`I hate this weather :)`
`13:00.59`	`JT`	`flenders: too late, you're funny`
`13:01.00`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`13:01.00`	`A-data`	`[TK]D-Fender http://paste-it.net/2603`
`13:01.40`	`flenders`	`:D`
`13:02.09`	`Chris-NB`	`anyone played around with astlinux?`
`13:02.21`	`flenders`	`what is astlinux again?`
`13:02.39`	`flenders`	`is it like a distro with asterisk?`
`13:02.55`	`Chris-NB`	`flenders, a embedded linux running on a CF card`
`13:02.59`	`Chris-NB`	`flenders, jep`
`13:03.03`	`[TK]D-Fender`	`A-data: remove ALL of those ";", this isn't PASCAL. then its [globals] , not [global]`
`13:03.32`	`A-data`	ok [TK]D-Fender it`s my C sytle :> i will do that now
`13:03.51`	`JT`	`A-data: and you like backwards quotqation marks too? :)`
`13:03.59`	`JT`	`quotation`
`13:04.06`	`flenders`	`Chris-NB: and what's the problem?`
`13:04.12`	`A-data`	`JT where?`
`13:04.24`	`*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)`
`13:04.35`	`JT`	it`s
`13:04.39`	`[TK]D-Fender`	`A-data: "Style" huh? ... http://www.despair.com/los24x30prin.html`
`13:04.40`	`Chris-NB`	`flenders, I've built it from de development environment. anything went fine, until it comes to grub install on the disk image`
`13:04.43`	`JT`	`that's the wrong quotation mark`
`13:05.14`	`flenders`	`Chris-NB: so, you haven't even got it running?`
`13:05.29`	`Chris-NB`	`flenders, not from the dev-environment, noop`
`13:05.39`	`A-data`	`lol [TK]D-Fender i am not loser :p`
`13:06.04`	`Chris-NB`	`flenders, the image from their website runs fine, but this image doesn't contain all packages what I need. So I used the dev-env`
`13:06.10`	`*** join/#asterisk saftsack (n=oliver@p54A7C2A6.dip.t-dialin.net)`
`13:06.21`	`JT`	`you better get this 501 to work flenders ;)`
`13:06.24`	`Chris-NB`	`flenders, have you built it from dev-env?`
`13:06.25`	`flenders`	`sorry mate, but I have no idea what the development environment is... my question really is, you need help to get ast linux installed?`
`13:06.38`	`flenders`	`Jt: I'll do my best mate!`
`13:06.49`	`flenders`	`JT: thanks a lot!`
`13:07.04`	`flenders`	`Chris-NB: I've never built it`
`13:07.16`	`Chris-NB`	`flenders, dev-env is a environment to crosscompile the astlinux distro on another box`
`13:07.17`	`flenders`	`I dont even know how it works... :o)`
`13:07.21`	`Chris-NB`	`flenders, ok`
`13:07.22`	`JT`	`you need 12v, positive centre pin`
`13:07.26`	`*** join/#asterisk ghenry (n=ghenry@212.159.59.85)`
`13:07.34`	`flenders`	`JT: that I can find a DS`
`13:07.45`	`Chris-NB`	`no one ever used the dev-env for astlinux?`
`13:07.48`	`flenders`	`there's one on your building...`
`13:08.08`	`flenders`	`Chris-NB: I guess you should be looking for support on their website, not here`
`13:08.09`	`JT`	`you could even make an adapter to connect to a molex ;)`
`13:08.16`	`Chris-NB`	`flenders, jup`
`13:08.31`	`flenders`	`Chris-NB: sorry.`
`13:08.42`	`[TK]D-Fender`	`Chris-NB: Let clear this up : EXTREMELY few people here will have used it. Fewer still are on at this hour. GET GOOGLING :)`
`13:09.11`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`13:09.14`	`tzanger`	`CAS signalling on span 1 conflicts with HDLC with FCS check on channel 16.`
`13:09.15`	`tzanger`	`wtf`
`13:09.27`	`tzanger`	`it's ztdynamic, I am not specifying CAS anywhere`
`13:09.29`	`Chris-NB`	`[TK]D-Fender, thanks`
`13:09.30`	`flenders`	`JT: I'd be better off just buying an adapter for 10 bucks.`
`13:09.34`	`tzanger`	`zaptel.conf is only 3 lines`
`13:09.41`	`JT`	`flenders: 10bucks, dreaming? ;)`
`13:10.06`	`[TK]D-Fender`	`tzanger: WTF are you doing messing with E1? :)`
`13:10.12`	`flenders`	`JT: or borrow your solding iron and all that stuff you have lying around at your place`
`13:10.17`	`tzanger`	`[TK]D-Fender: top secret :-)`
`13:10.24`	`tzanger`	`it's E1 over TDMoE even`
`13:10.30`	`JT`	`[TK]D-Fender: E1 is the bomb :)`
`13:11.31`	`flenders`	`well, off to bed...`
`13:11.42`	`*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)`
`13:11.43`	`JT`	`night`
`13:11.49`	`Teccy`	`JT: btw, the zaptel drivers were initially written for freebsd (wrt your comment last night)`
`13:11.58`	`flenders`	`I'll let you know tomorrow what time I'm coming over`
`13:12.08`	`JT`	`Teccy: err, proof?`
`13:12.35`	`JT`	`maybe the old original tormeda cards, but digium develops for linux`
`13:13.09`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`13:13.39`	`Teccy`	`true, but still`
`13:14.20`	`JT`	`and how many people come in here asking for tormeda help?`
`13:15.41`	`Teccy`	`i was just stating a fact, is all`
`13:15.44`	`*** join/#asterisk ELBunce (n=erik@kde/developer/bunce)`
`13:16.43`	`[TK]D-Fender`	`TORMENTA`
`13:16.43`	`JT`	`but my point the other day was that if you choose to go zap bsd, you are choosing an unsupported and possibly more flakey route`
`13:17.26`	`coppice`	`lots of people want tormenta help, because lots of people use the tormenta 2 cards from various suppliers`
`13:18.36`	`JT`	`silent majority eh??`
`13:19.41`	`coppice`	`I wonder how many of those have been made by various people`
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`13:53.27`	`*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`13:54.13`	`*** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)`
`13:54.13`	`frood`	`my boss has given me a new cisco 7960 he wants added to the network but he wants it to have different permissions to the rest. ie. it can't dial out.`
`13:54.39`	`frood`	`is there an easy way of doing this without adding if statements to all the existing extensions?`
`13:54.45`	`*** part/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)`
`13:54.57`	`Mercestes`	`frood: Smart use of contexts and includes will solve your problem.`
`13:55.09`	`[TK]D-Fender`	`frood: Put him in a context that doesn't allow him to dial out.`
`13:55.26`	`frood`	`thanks guys`
`13:56.13`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`13:56.21`	`frood`	`i'm only using one context at the moment`
`13:56.26`	`frood`	`[default]`
`13:56.48`	`*** join/#asterisk oej_ (n=olle@apollo.webway.se)`
`13:56.56`	`casix`	`hello`
`13:57.11`	`casix`	`I've problems with asterisk 1.4 and cdr`
`13:57.21`	`[TK]D-Fender`	`frood: well ad more and break it up`
`13:57.39`	`casix`	`asterisk crash when I make a cdr submit. The cdr is in a mysql db`
`13:58.16`	`mosty`	`casix: submit a bug report`
`13:59.02`	`casix`	`ok`
`14:00.32`	`mosty`	`check that one hasn't already been submitted for that bug yet, of course`
`14:03.18`	`*** join/#asterisk kc-lamda (n=Elive_us@proxy.hostopia.com)`
`14:05.47`	`kc-lamda`	`Hi, Does anyone know how to handle an error with asterisk. I have connected one server to another through an IAX trunk. Outgoing calls work from one server however for that same server incoming calls don't work. The contexts are set correctly. The error message is No Authority Found.`
`14:05.52`	`*** join/#asterisk oej (n=olle@apollo.webway.se)`
`14:06.35`	`kc-lamda`	`When I try to debug the only thing I see different between the incoming calls from another server connected through IAX and this one that doesn't work is the format being 256`
`14:06.36`	`mosty`	`woohoo, both E1 lines working`
`14:07.03`	`creativx`	`kc-lamda: checked the case sensitivity?`
`14:07.09`	`[TK]D-Fender`	`kc-lamda: 256 means G.729. Do you have that licensed on your servers?`
`14:07.38`	`kc-lamda`	`Not on the one that is recieving.`
`14:07.50`	`[TK]D-Fender`	`kc-lamda: Well that would be a problem then, wouldn't it....`
`14:07.57`	`[TK]D-Fender`	`kc-lamda: Go fix the codes`
`14:08.01`	`[TK]D-Fender`	`codecs*`
`14:08.06`	`kc-lamda`	`[TK]D-Fender, yes, thanks let me try that out.`
`14:09.03`	`*** join/#asterisk fenril (n=Fenril@LLamentin-151-1-77.w81-248.abo.wanadoo.fr)`
`14:10.02`	`*** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk)`
`14:10.05`	`zeeesh`	`hi all`
`14:10.36`	`*** join/#asterisk FreezeS (n=bla@82.208.157.125)`
`14:10.41`	`FreezeS`	`hey guys`
`14:10.49`	`FreezeS`	`I've got a lot of problems with a new installation`
`14:10.55`	`FreezeS`	`crashes very often`
`14:11.00`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`14:11.14`	`FreezeS`	`can anyone guide me to find the cause of this ?`
`14:11.22`	`mosty`	`what's the last thing in the logs when asterisk crashes, and what version are you running?`
`14:12.04`	`anonymouz666`	`anyone in here use pap2 to pass-through fax?`
`14:12.32`	`FreezeS`	`I'm running 1.4.5 now with zaptel 1.4.3 and asterisk-addons 1.4.2`
`14:12.47`	`FreezeS`	`I'm getting disconnected from the console`
`14:13.27`	`FreezeS`	`the message is something like "Asterisk exited cleanly"`
`14:13.37`	`mosty`	`freezes: enable full logging, set debug and verbose to 10`
`14:13.47`	`mosty`	`look at the full log, not the concole`
`14:13.50`	`mosty`	`console, even`
`14:13.57`	`FreezeS`	`I was looking at the debug file`
`14:14.03`	`FreezeS`	`and saw nothing weird`
`14:14.04`	`mosty`	`anonymouz666, no because fax over voip does not work`
`14:14.19`	`zeeesh`	`trial configuring voicemail for sip users .. due to my configuration if user's online through xlite and not receiving the call then after 10 rings u can leave voice msg ... i need if user is not online caller can leave msg. how should i perform?`
`14:14.23`	`mosty`	`freezes: how are you starting asterisk?`
`14:14.28`	`*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)`
`14:14.30`	`FreezeS`	`mosty: how do I set the log verbosity ?`
`14:14.39`	`FreezeS`	`using the debian startup script`
`14:14.44`	`mosty`	`freezes: core set verbose 10, and core set debug 10`
`14:15.01`	`FreezeS`	`will they remain after I close the console ?`
`14:15.04`	`[TK]D-Fender`	`zeeesh: go read THE BOOK and "show application dial"`
`14:15.06`	`[TK]D-Fender`	`~book`
`14:15.07`	`jbot`	`well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`14:15.15`	`mosty`	`FreezeS, yes but not after you restart asterisk`
`14:15.20`	`anonymouz666`	`mosty: the fax will be placed through mfc/r2 link`
`14:15.34`	`anonymouz666`	`voip is using g711 on lan`
`14:15.36`	`anonymouz666`	`it works`
`14:15.49`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`14:15.49`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:15.49`	`*** join/#asterisk EricL (n=eric@74.9.83.194)`
`14:15.53`	`mosty`	`anonymouz666, i've heard reports of fax over sip using g711 being flakey too`
`14:16.13`	`*** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk)`
`14:16.13`	`EricL`	`Can someone point me to a plugin (if it exists) to use Astiersk with Ximian Evolution?`
`14:16.18`	`FreezeS`	`mosty: I've set the debug and verbose levels. Now let's see what happends...`
`14:16.26`	`[TK]D-Fender`	`mosty: Can be, but can be OK if you're lucky. I for one don't like betting my business on LUCK.`
`14:16.29`	`EricL`	`Google just points me to an Asterisk based PBX called Evolution.`
`14:16.42`	`mosty`	`[TK]D-Fender, i agree`
`14:16.59`	`*** join/#asterisk saftsack (n=oliver@p54A7EDB7.dip.t-dialin.net)`
`14:17.00`	`anonymouz666`	`it works here`
`14:17.08`	`*** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net)`
`14:17.11`	`anonymouz666`	`with grandstream doing pass-through`
`14:17.22`	`mosty`	`EricL, what do you want evolution to do?`
`14:17.24`	`anonymouz666`	`all the time using fxo`
`14:17.32`	`FreezeS`	`I've got a lot of theese: [Jun 19 17:17:13] WARNING[4620]: chan_sip.c:12434 handle_response: Remote host can't match request NOTIFY to call '17c85f483182e3106edc147802a95586@192.168.105.150'. Giving up.`
`14:17.46`	`FreezeS`	`about one every couple of seconds`
`14:18.25`	`FreezeS`	`what are they ?`
`14:19.05`	`EricL`	`mosty: I want to be able to call the contact by pressing the phone number or having a menu that says call.`
`14:19.11`	`kc-lamda`	`[TK]D-Fender, unfortunately it didn`
`14:19.19`	`kc-lamda`	`[TK]D-Fender, '`
`14:19.55`	`mosty`	`they're just warnings, i would ignore them in this case`
`14:20.05`	`EricL`	`mosty: Similar to the way in which you can press "Compose Message to Contact". I want it to dial my extension and then dial the number of the contact.`
`14:20.10`	`[TK]D-Fender`	`kc-lamda: pastebin ALL of the associated bits of your configs on BOTH sides, along with the CLI output of the failed attempt.`
`14:20.11`	`kc-lamda`	`[TK]D-Fender, unfortunately it didn't help. I am getting the following error: Call rejected by IP-Of-SERVER: No authority found`
`14:20.12`	`[TK]D-Fender`	`~pb`
`14:20.12`	`jbot`	`pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org`
`14:20.14`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^`
`14:20.58`	`*** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil)`
`14:21.07`	`mosty`	`EricL, look at click to call scripts, there are some for firefox out there. a plugin for evolution would work in a similar way`
`14:22.07`	`*** join/#asterisk floppp (n=flop@nat-staff.b3g-telecom.com) [NETSPLIT VICTIM]`
`14:22.07`	`*** join/#asterisk smurf (n=smurf@debian/developer/smurf) [NETSPLIT VICTIM]`
`14:22.07`	`*** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net) [NETSPLIT VICTIM]`
`14:22.07`	`*** join/#asterisk Pagautas (n=bigman@83.171.14.250) [NETSPLIT VICTIM]`
`14:22.29`	`EricL`	`mosty: I have found ones for firefox. I am just looking for one for evolution.`
`14:23.41`	`*** join/#asterisk tsurko (n=tsurko@77.70.24.142)`
`14:24.24`	`[TK]D-Fender`	`EricL: Could very well be that it doesn't exist. Go ask in #sasquatch and see if anyone there has seen it....`
`14:25.08`	`EricL`	`[TK]D-Fender: It may not exist, but I figured if anyone would know, it would probably be the folks in here.`
`14:25.27`	`[TK]D-Fender`	`EricL: checked the WIKI?`
`14:26.56`	`[TK]D-Fender`	`EricL: Doesn't seem to be anything...`
`14:27.02`	`[TK]D-Fender`	`EricL: get coding!`
`14:27.11`	`EricL`	`I did and I didn't find anything.`
`14:27.22`	`*** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)`
`14:27.22`	`*** mode/#asterisk [+o mog] by ChanServ`
`14:27.33`	`EricL`	`I actually may just code one. Depends on how much time I can free up at work.`
`14:28.33`	`SirThomas`	`EricL: it would be super cool!`
`14:29.10`	`frood`	`i've put the new phone in a different context... but how do i get it to dial an extension in the default context?`
`14:29.33`	`EricL`	`Alright...thanks everyone. If I get around to writing it, I will be sure to let everyone know.`
`14:29.39`	`*** part/#asterisk EricL (n=eric@74.9.83.194)`
`14:31.29`	`mosty`	`frood, use Goto or include`
`14:31.38`	`[TK]D-Fender`	`frood: you have that context "include => [contextnamewithoutbraces]" the ones that contain the extens you want to allow it access to.`
`14:32.30`	`frood`	`so if i wanted to dial extension 8000 in [default] if i dial 100 on the phone, i'd just do "exten => 100,1,Dial(default,8000) ?`
`14:32.33`	`mrdigital-work`	`exten => s,6,GotoIf($["{$orderstatus}" = "P"]?10:20) how come this is always reporting the varaible as false?`
`14:32.38`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`14:33.32`	`[TK]D-Fender`	`frood: No, you use INCLUDE statements to tell that context that it can do whatever is in that other context.`
`14:33.47`	`frood`	`hmm`
`14:33.48`	`frood`	`ok`
`14:34.40`	`[TK]D-Fender`	`mrdigital-work: NoOp your variable before you call GototIf to prove what its filled with.`
`14:36.01`	`frood`	`[TK]D-Fender: but all the extensions are in the default context. Do i really have to split them up into other contexts, or can i just do a Goto jump to the correct place?`
`14:36.16`	`*** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl)`
`14:36.20`	`mosty`	`frood: it's your choice`
`14:36.34`	`[TK]D-Fender`	`frood: "include => default" gives you new context access to EVERYTHING in [default]. its ONE LINE.`
`14:36.56`	`frood`	`[TK]D-Fender: but i dont want this one phone to have access to everything in default`
`14:37.05`	`robl^`	`moooo! hey boys and girls. ;-)`
`14:37.21`	`frood`	`mosty: how would i format the Goto statement?`
`14:37.38`	`frood`	`mosty: Goto(default,8000)?`
`14:37.41`	`FreezeS`	`I'm curious, will they allow includes in realtime in 1.6 ?`
`14:37.47`	`*** join/#asterisk saftsack (n=oliver@p54A7FF40.dip.t-dialin.net)`
`14:38.02`	`mosty`	`frood: i recommend you break the default context up into smaller contexts. the default context would then include them all, and this other context would include just the ones you want there`
`14:38.32`	`frood`	`mosty: i will soon enough. but i've got the boss breathing down my neck`
`14:38.53`	`frood`	`mosty: and if 1 goto line will get the phone to phone the switchboard on "100" then i'll do that`
`14:39.11`	`frood`	`mosty: but i'm getting a busy signal`
`14:39.21`	`robl^`	`[TK]D-Fender: hey mate! Have you tried the new Polycom 320/330s?? Any comments about them? Considering those instead of 430s`
`14:39.52`	`mrdigital-work`	`NoOp??`
`14:40.10`	`mosty`	`frood, the one goto line would make it do everything the default context does`
`14:40.16`	`robl^`	`NoOp does No Operation. It does NOTHING`
`14:40.38`	`frood`	`ah i see`
`14:40.46`	`[TK]D-Fender`	`frood: this where you should have come up with the idea to BREAK DEFAULT UP.`
`14:40.47`	`mrdigital-work`	`ok`
`14:41.03`	`[TK]D-Fender`	`frood: you don't need Goto. that is the wrong approach.`
`14:41.05`	`mosty`	`robl^, it does slightly more than nothing, it adds output to the verbose log`
`14:41.24`	`*** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com)`
`14:41.43`	`frood`	`can a SIP user belong to more than one context?`
`14:42.09`	`[TK]D-Fender`	`robl^: Not personally, but i intend to. 430 has a place if you need to handle more than 2 calls per line-key, and plan on PoE later, but not immediately.`
`14:42.12`	`mosty`	`frood, sip users don't "belong" to contexts. their calls start in a context`
`14:42.19`	`[TK]D-Fender`	`frood: No.`
`14:42.23`	`frood`	`i see`
`14:42.27`	`*** join/#asterisk plla (n=nekomimi@corporacionlely.com.pe)`
`14:42.39`	`robl^`	`mosty: true.. but it really exists for legacy reasons. it was there in the old days when you just to jump to like priority + 101 for some applications based on result. You could put in a NoOp as a place holder to keep the dial plan from doing anything..`
`14:42.43`	`plla`	`Hello`
`14:43.03`	`mosty`	`robl^, it's also used for debugging`
`14:43.15`	`mrdigital-work`	`NoOp?? didnt do anything`
`14:43.32`	`JT`	`isn't that what people just said?`
`14:43.46`	`mrdigital-work`	`yeah [TK]D-Fender: said to use it before the gotoif statement i did`
`14:43.56`	`[TK]D-Fender`	`frood: you put your "dial out" extens in say CONTEXTA , you then put your other internal stuff in CONTEXTB. You then make a CONTEXTC that INCLUDES A & B. You give NORMAL users CONTEXTC as their context, and your restricted phones CONTEXTB <---------`
`14:44.03`	`plla`	`I have problems with one of my asterisk installations. Can anyone give me a hand?`
`14:44.13`	`[TK]D-Fender`	`NO STUPID GOTO'S WHERE YOU SHOULD BE USING INCLUDES!`
`14:44.27`	`s0ck`	`is it normal to for the asterisk make script to whinge about /usr/lib/asterisk/modules being incompatible?`
`14:44.27`	`frood`	`ok :)`
`14:44.28`	`[TK]D-Fender`	`plla: ....`
`14:44.30`	`mosty`	`plla, ask a specific question`
`14:44.31`	`s0ck`	`like, 20 of them`
`14:44.31`	`[TK]D-Fender`	`~ask`
`14:44.32`	`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.`
`14:44.33`	`file`	`[TK]D-Fender: easy now!`
`14:44.38`	`robl^`	`[TK]D-Fender: ahh!! yeah. I am looking for something very basic -- caller ID and 2 calls would likley be the max. I am tempted to try the 320s for that reason`
`14:44.38`	`file`	`[TK]D-Fender: no heart attacks allowed`
`14:45.06`	`*** join/#asterisk FoxTrot- (n=lfc@200.204.142.181)`
`14:45.24`	`FoxTrot-`	`can anybody help me, i keep getting this msg "outgoing spool failed"`
`14:45.29`	`FoxTrot-`	`how do i fix that`
`14:45.42`	`robl^`	`the Polycom 320s look like they would be good for low volume "telecommuter" phones.`
`14:45.43`	`[TK]D-Fender`	`robl^: You'll be more than happy with them I'm sure. I wish I was using my bed-side IP 301 rather than the Aastra 57i CT I have at my office :)`
`14:45.44`	`plla`	`I have two asterisk installed, each with a digium card TE110p and TE210P.`
`14:45.54`	`plla`	`I am having problems with my PRI line`
`14:46.14`	`plla`	`Asterisk randomly restarts the channels every 20 minutes or so`
`14:46.14`	`JT`	`[TK]D-Fender: surely you need speakerphone for the bedside ;)`
`14:46.29`	`JT`	`plla: are the channels in use when restarded?`
`14:46.30`	`robl^`	`[TK]D-Fender: hah! How are the new Aastras?? Junk? I have a 480i I use at home..on my dev box`
`14:46.30`	`[TK]D-Fender`	`FoxTrot-: How are you making them, putting them into the spool folder?`
`14:46.34`	`JT`	`restarted`
`14:46.47`	`plla`	`In my testing server while the same PRI line in the other server works fine.`
`14:46.56`	`JT`	`plla: are they in use or not?`
`14:47.03`	`[TK]D-Fender`	`JT : No, if I had speakerphone I'd be tempted to whack the phone to answer, and in missing likely hang up the caller :)`
`14:47.03`	`plla`	`the channels can be in use or not`
`14:47.14`	`JT`	`[TK]D-Fender: heh`
`14:47.21`	`plla`	`I read the logs and it has happened all the day with calls or not.`
`14:47.31`	`mosty`	`jt: btw after adding some dialplan options to zapata.conf i can dial numbers as normal now :) both E1 channels working perfectly`
`14:47.42`	`plla`	`I am thinking it may be a timing problem with the digium card of the testing server.`
`14:47.44`	`JT`	`plla: asterisk normally restarts pri channels which aren't used at intervals specified in the config files`
`14:47.49`	`JT`	`plla: no, it's normal.`
`14:48.01`	`[TK]D-Fender`	`robl^: the 480i is a better phone than the 5i series IMO. 5i's have rubbery shit buttons, make shit use of their pixel based screen, and the handset has NO weight`
`14:48.03`	`JT`	`mosty: heh, set the dialplan to unknown?`
`14:48.04`	`plla`	`it hangs every call`
`14:48.16`	`plla`	`that's not right.`
`14:48.36`	`JT`	`plla: is the D channel flapping?`
`14:48.40`	`mosty`	`jt: yes. at one point somebody asked me to try removing the dialplan options, but i think that was long before i was even close to getting this thing working`
`14:48.59`	`JT`	`mosty: yeah, it should always be set to unknown in australia afaik`
`14:49.16`	`plla`	`the log is mostly:`
`14:49.17`	`JT`	`i was going to suggest it the other day, but you failed to paste all your configs...`
`14:49.22`	`plla`	`WARNING[2116] chan_zap.c: Detected alarm on channel XX: Yellow Alarm`
`14:49.35`	`plla`	`NOTICE[2115] chan_zap.c: PRI got event: Alarm (4) on Primary D-channel of span 1`
`14:49.38`	`mosty`	`JT, anyway thanks for the help over the various times when i was in here`
`14:49.43`	`plla`	`WARNING[2115] chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway!`
`14:49.43`	`frood`	`ahh working sweet now`
`14:49.46`	`frood`	`thanks`
`14:49.53`	`plla`	`NOTICE[2116] chan_zap.c: Alarm cleared on channel X`
`14:49.58`	`JT`	`plla: maybe you should look into zap timing`
`14:50.10`	`JT`	`plla: pastebin.ca zaptel.conf and zapata.conf`
`14:50.34`	`plla`	`ok`
`14:50.55`	`*** join/#asterisk mrichmanM (n=richmanm@70.89.184.1)`
`14:51.00`	`purplet`	`Hello, can someone determine the cause of a dropped call based on this output? http://pastebin.ca/576491 (calls always drop after 11 minutes, suspecting NAT, but still think the output is weird)..`
`14:52.17`	`JT`	`plla: and btw what you are describing is the D channel flapping, not B channel resets`
`14:52.54`	`plla`	`I see.`
`14:53.08`	`plla`	`http://pastebin.ca/576501 zaptel.conf`
`14:53.10`	`mosty`	`purplet, you probably need to do an iax debug`
`14:53.17`	`plla`	`http://pastebin.ca/576503 zapata.conf`
`14:53.37`	`JT`	`plla: are both spans connected to the telco??`
`14:53.38`	`plla`	`I have the exact same configuration in my production server and it doesn't happen there.`
`14:53.47`	`JT`	`well the configs are wrong anyway`
`14:54.06`	`plla`	`it's only one span`
`14:54.21`	`JT`	`span 1?`
`14:54.31`	`JT`	`#`
`14:54.32`	`JT`	`span=1,0,0,ccs,hdb3`
`14:54.42`	`JT`	`span=1,1,0,ccs,hdb3`
`14:54.51`	`JT`	`change it to that ^`
`14:55.02`	`plla`	`ok`
`14:55.06`	`JT`	`it must receive timing from the telco`
`14:55.13`	`JT`	`not try and act as sync master`
`14:55.49`	`JT`	`you'll need to fully restart zaptel and ast after making the change`
`14:56.05`	`plla`	`ok`
`14:56.43`	`plla`	`I get this message, should I get worried? Loading zaptel hardware modules: wcte11xpNo Zaptel timing source sound. loading ztdummy`
`14:57.23`	`JT`	`sounds like it shouldn't be coming up`
`14:57.40`	`tzafrir_laptop`	`plla, hmm... head -c 1 /dev/zap/pseudo`
`14:57.53`	`*** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com)`
`14:58.22`	`*** join/#asterisk svenna_ (n=svenna@p548D0E55.dip0.t-ipconnect.de)`
`14:58.24`	`plla`	`what does it do?`
`14:58.32`	`tzafrir_laptop`	`plla, sorry`
`14:58.43`	`tzafrir_laptop`	`plla, ignore that`
`14:58.54`	`ai-a`	`how easy is it to set up recording on N internal lines on asterisk ? to record inbound / outbound call from external numbers. And also place these into a folder we can reference easy ?`
`14:59.07`	`mrdigital-work`	`why is asterisk failing to goto the script for no reason`
`14:59.26`	`plla`	`so do I ignore the "No Zaptel timing source sound. loading ztdummy"`
`14:59.28`	`JT`	`ai-a: pretty easy as long as asterisk has access to the audio`
`14:59.29`	`mosty`	`ai-a, 42 easy. see the Monitor command`
`14:59.33`	`[TK]D-Fender`	`ai-a: Relatively easy`
`14:59.44`	`*** join/#asterisk GlobeTrotter (i=erivvnni@190.10.0.188)`
`14:59.51`	`*** join/#asterisk bintut (n=bintut@203.125.63.150)`
`14:59.59`	`[TK]D-Fender`	`JT : * always has access to the audio ;)`
`15:00.06`	`JT`	`[TK]D-Fender: not true`
`15:00.15`	`JT`	`sip endpoints reinviting`
`15:00.46`	`[TK]D-Fender`	`JT : thats * ALLOWING it to reinvite.. is HAS access to it ALL THEM time, if you want to WAIVE the traffic, then by all means ;)`
`15:01.22`	`JT`	`[TK]D-Fender: it's still a valid point anyway and plenty of people get caught out by it`
`15:01.30`	`[TK]D-Fender`	`JT : And mere use of the Monitor app will PREVENT it from re-inviting, jsut like Dial opts ;)`
`15:01.35`	`*** join/#asterisk HockeyInJune (n=HockeyIn@pool-70-18-14-219.nycmny.east.verizon.net)`
`15:01.41`	`JT`	`also not always true`
`15:02.05`	`JT`	`i have fixed a few people's monitor problems by getting them to change their sip.conf to have canreinvite=no`
`15:02.08`	`JT`	`maybe a bug`
`15:02.10`	`[TK]D-Fender`	`JT : Monitor most assuredly.`
`15:02.11`	`*** join/#asterisk perf3kt (i=perf3kt@149.166.34.141)`
`15:02.14`	`JT`	`but it's there nevertheless`
`15:02.21`	`[TK]D-Fender`	`Jt : I'd have to be :)`
`15:02.30`	`[TK]D-Fender`	`it'd`
`15:03.33`	`*** join/#asterisk shay\|work (i=shay@unaffiliated/shay)`
`15:04.44`	`indend7`	`hi there....having prob with fwd calling....says "All circuits are busy now, try again later"....on 39355555`
`15:05.05`	`indend7`	`393612 and 393613 gives busy tone!!....`
`15:05.19`	`*** join/#asterisk eald (n=eald@189.157.254.176)`
`15:05.28`	`indend7`	`tried forums and goole search....not much helpfull....`
`15:05.59`	`eald`	`hi, how do you use Monitor application in order to get both legs syncronized in the recorded file?`
`15:06.00`	`indend7`	`can anybody help me regarding this??`
`15:06.57`	`[TK]D-Fender`	`indend7: Pastebin the complete CLI output of your failed call attempt at verbose 10.`
`15:07.09`	`plla`	`hmm`
`15:07.15`	`[TK]D-Fender`	`~pb`
`15:07.16`	`jbot`	`pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org`
`15:07.17`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^`
`15:07.31`	`purplet`	`mosty: overhere I have the debug output... server1: pastebin.ca/576529 server2: pastebin.ca/576535 ... I don't see anything strange... BTW only audio is dropping, connections stays...`
`15:07.40`	`plla`	`how do I explain that the same configuration works in the other server?`
`15:07.51`	`plla`	`with the same pri line`
`15:08.02`	`indend7`	`I have it's output in file....I can send you here....`
`15:08.30`	`JT`	`plla: there's probably something set wrong`
`15:08.42`	`*** join/#asterisk mrichmanM (n=richmanm@70.89.184.1)`
`15:08.45`	`plla`	`It's an exact copy.`
`15:08.53`	`JT`	`your timing was wrong`
`15:08.54`	`plla`	`Just the span 2 commented out.`
`15:08.58`	`JT`	`it's pure luck it worked`
`15:09.06`	`plla`	`I see.`
`15:09.14`	`JT`	`on the old server anyway`
`15:09.18`	`indend7`	`yea...read alot about that doesn't work easily!!...`
`15:09.23`	`JT`	`is all the versions of everything identical?`
`15:09.38`	`plla`	`nope, the new server is using asterisk 1.4.5`
`15:09.49`	`indend7`	`I've installed 1.2.18`
`15:09.55`	`plla`	`the older 1.4.0`
`15:10.01`	`JT`	`1.2.19 is out`
`15:10.09`	`indend7`	`it's a latest trixbox 2.2.* I installed....`
`15:10.13`	`JT`	`..`
`15:10.15`	`JT`	`~trixbox`
`15:10.16`	`jbot`	`Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org`
`15:10.21`	`plla`	`I tested with the svn but got the same problems.`
`15:10.34`	`JT`	`plla: correct version of zaptel i assume`
`15:10.45`	`plla`	`yes.`
`15:11.07`	`JT`	`full kernel source and headers installed, with zap drivers compiled against them?`
`15:11.40`	`[TK]D-Fender`	`indend7: PASTEBIN, not DCC`
`15:11.41`	`plla`	`yes, kernel 2.6.18-8.1.4.el5`
`15:11.43`	`[TK]D-Fender`	`~pb`
`15:11.43`	`jbot`	`well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org`
`15:11.45`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^`
`15:12.15`	`plla`	`the older server is using kernel 2.6.9-42.0.10.ELsmp`
`15:12.23`	`JT`	`plla: what card?`
`15:12.51`	`kombi`	`Internal RTCP NTP clock skew detected <<- how to fix that?`
`15:13.03`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`15:13.03`	`*** mode/#asterisk [+o anthm] by ChanServ`
`15:13.18`	`*** join/#asterisk codey (i=codec@iglu.paranoid-penguin.de)`
`15:13.20`	`codey`	`hi there`
`15:13.26`	`kombi`	`does asterisk keep its own time?`
`15:13.36`	`codey`	`anyone running asterisk-bristuff on etch with a junghanns duoGSM pci-card?`
`15:13.36`	`plla`	`Digium TE110P the new server and the older TE210P`
`15:13.49`	`*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)`
`15:14.22`	`tzafrir_laptop`	`codey, jost note that the Debian packages of asterisk do not include libsmat`
`15:14.38`	`*** join/#asterisk zdrulio (n=krlozano@82.119.72.130)`
`15:14.46`	`tzafrir_laptop`	`libgsmat, that is`
`15:14.49`	`zdrulio`	`where can i find change log for 1.4.5`
`15:14.50`	`zdrulio`	`?`
`15:14.56`	`Qwell[]`	`zdrulio: in the source`
`15:15.05`	`zdrulio`	`ok thx`
`15:15.08`	`codey`	`tzafrir_laptop: i've installed libgsmat-0.0.2 by hand`
`15:15.13`	`JT`	`plla: does the card come up on lspci -vv ?`
`15:16.01`	`plla`	`explain, I don't understand the question.`
`15:16.10`	`tzafrir_laptop`	`modprobe ztgsm ?`
`15:16.14`	`JT`	`type in lspci -vv`
`15:16.22`	`codey`	`Jun 19 17:16:08 ERROR[2842]: chan_zap.c:12491 setup_zap: Unknown signalling method 'gsm'`
`15:16.25`	`codey`	`:(`
`15:16.50`	`tzafrir_laptop`	`codey, asterisk was built without support for gsm. Is this from debs?`
`15:16.54`	`kombi`	`where do you set asterisk's "internal clock", if any?`
`15:16.55`	`JT`	`codey: just compile from source`
`15:16.58`	`codey`	`yes, it's from debs`
`15:17.04`	`tzafrir_laptop`	`hmm...`
`15:17.10`	`JT`	`wonder why it doesn't work ;)`
`15:17.29`	`codey`	`actually i didn't want to compile it myself`
`15:17.33`	`JT`	`too bad`
`15:17.37`	`JT`	`it's your best bet`
`15:17.55`	`JT`	`otherwise sent me the duo-gsm if compiling is too much effort :P`
`15:17.59`	`JT`	`send`
`15:18.00`	`tzafrir_laptop`	`or rebuild the package with libgsmat installed on your system. Should work as well`
`15:18.20`	`tzafrir_laptop`	`(compile: == bristuffed asterisk and libpri)`
`15:18.25`	`indend7`	`[TK]D-Fender: this is the output of call I made on 393612 http://paste.debian.net/30888`
`15:18.47`	`*** join/#asterisk Ciber311 (n=Ciber311@user-12ld42j.cable.mindspring.com)`
`15:18.55`	`plla`	`JT: http://pastebin.ca/576585 lspci -vv`
`15:19.12`	`indend7`	`and Nobody is answering in #trixbox :(`
`15:19.18`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`15:19.18`	`*** mode/#asterisk [+o anthm] by ChanServ`
`15:19.27`	`codey`	`JT: okay, so what do I need to compile the whole stuff?`
`15:19.46`	`JT`	`plla: well it comes up, i'm still puzzled about the timing error message`
`15:19.57`	`FoxTrot-`	`where do i add channels in my asterisk`
`15:19.57`	`FoxTrot-`	`?`
`15:20.17`	`JT`	`codey: bristuff tarball, it has a script to download patch and compile zaptel, libpri and asterisk inside it`
`15:20.19`	`[TK]D-Fender`	`indend7: the other side SAYS its "busy".`
`15:20.31`	`[TK]D-Fender`	`indend7: And you are running FreePBX which is not supported here`
`15:20.39`	`plla`	`the same server was used before with the same card with the same pri with Asterisk 1.2.X and it worked fine.`
`15:20.44`	`[TK]D-Fender`	`FoxTrot-: What KIND of channels?`
`15:20.51`	`JT`	`plla: then use 1.2`
`15:21.04`	`plla`	`I have a production server already working.`
`15:21.05`	`indend7`	`[TK]D-Fender: Other side? means FWD side or wat??`
`15:21.18`	`JT`	`most people still use 1.2 in production`
`15:21.19`	`plla`	`I was testing the new Asterisk.`
`15:21.23`	`*** join/#asterisk Jingles (n=dfbarth@74-61-126-58.anc.clearwire-dns.net)`
`15:21.24`	`FoxTrot-`	`[TK]D-Fender active channels`
`15:21.29`	`[TK]D-Fender`	`indend7: Correct`
`15:21.44`	`[TK]D-Fender`	`FoxTrot-: "active" is not a KIND, its a STATE`
`15:21.54`	`FoxTrot-`	`whats KIND then`
`15:22.01`	`JT`	`FoxTrot-: think about it`
`15:22.03`	`FoxTrot-`	`�:<`
`15:22.06`	`JT`	`you want a new channel`
`15:22.12`	`JT`	`but can't specify what it is`
`15:22.14`	`indend7`	`[TK]D-Fender: but TrixBox installes FreePBX by default :-/`
`15:22.16`	`[TK]D-Fender`	`FoxTrot-: If you want to add channels, I'd suggest getting a Satelite receiver dish.`
`15:22.16`	`JT`	`do you know what you want?`
`15:22.23`	`JT`	`indend7: trixbox is also not supported`
`15:22.33`	`indend7`	`ahum??`
`15:22.34`	`[TK]D-Fender`	`indend7: around here thats sorta TFB. Nobody wants to deal with it.`
`15:22.42`	`FoxTrot-`	`[TK]D-Fender i mean, in which configuration file i do that? sip.conf?`
`15:22.43`	`JT`	`indend7: we support asterisk here`
`15:22.48`	`indend7`	`[TK]D-Fender: so what should I try for that to work??`
`15:23.04`	`JT`	`trixbox and freepbx are modifications to asterisk that change it greatly`
`15:23.05`	`plla`	`could it be that the card is having isues?`
`15:23.15`	`plla`	`a malfunctioning card perhaps?`
`15:23.24`	`indend7`	`[TK]D-Fender: I have downloaded AsteriskNow too!... Do you think it's fine to work ??`
`15:23.27`	`indend7`	`Okay..`
`15:23.31`	`JT`	`plla: always a possibility, work trying 1.2 first`
`15:23.36`	`[TK]D-Fender`	`FoxTrot-: You can have SIP channels, IAX2 channels, MGCP channels, Zap channels, H.323 Channels, LOCAL channels, and more.... so what the heck are you talking about>?!`
`15:23.39`	`JT`	`~thebook`
`15:23.39`	`jbot`	`extra, extra, read all about 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`
`15:23.43`	`JT`	`and real asterisk is best`
`15:24.05`	`[TK]D-Fender`	`indend7: I already told you that FWD said the # you dialed is BUSY as well.`
`15:24.22`	`[TK]D-Fender`	`indend7: So FreePBX is doing its just just fine I'd guess`
`15:24.27`	`FoxTrot-`	`[TK]D-Fender GOT IT hehe, zap channels`
`15:24.40`	`codey`	`JT: so which bristuff version do you recommend?`
`15:24.44`	`plla`	`I will redirect the traffic for 20 minutes to test this change, perhaps it was the wrong configuration in zaptel.conf`
`15:24.52`	`[TK]D-Fender`	`FoxTrot-: Now clarify what you mean by ADDING Zap channels.`
`15:24.57`	`indend7`	`okay....`
`15:25.08`	`JT`	`codey: i don't have experience with the last couple of releases, but usually the most recent version`
`15:25.09`	`*** join/#asterisk saftsack (n=oliver@p54A7D681.dip.t-dialin.net)`
`15:25.13`	`*** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net)`
`15:25.19`	`[TK]D-Fender`	`indend7: So FreePBX is doing its job just fine I'd guess`
`15:25.32`	`indend7`	`[TK]D-Fender: Okay....`
`15:25.42`	`[TK]D-Fender`	`indend7: "Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: CONTROL Subclass: BUSY" <------------`
`15:26.26`	`errr`	`anyone have any idea why when I get an incoming call the caller id info only shows the number of the caller and never shows the name`
`15:26.49`	`JT`	`name is not supported in a lot of places`
`15:26.53`	`JT`	`or costs money`
`15:27.10`	`JT`	`or you failed to put a Wait in your incoming dialplan`
`15:27.10`	`errr`	`JT: its supported here, it worked with our old pbx`
`15:27.17`	`JT`	`you must wait for it`
`15:27.19`	`errr`	`maybe I failed on that`
`15:27.23`	`JT`	`it is sent after the name`
`15:27.24`	`codey`	`JT: where will it place it's binaries? i don't want my compile-version to overwrite the debian version...`
`15:27.31`	`JT`	`as the CO does a db lookup and it's slow`
`15:27.34`	`indend7`	`But whenever I try to call on another extension like 393859890 or 393859890201 says that Busy msg again!...`
`15:27.37`	`JT`	`or something along those lines`
`15:27.50`	`indend7`	`[TK]D-Fender: But whenever I try to call on another extension like 393859890 or 393859890201 says that Busy msg again!...`
`15:28.03`	`errr`	`JT: Ill look into the wait, thanks`
`15:28.03`	`indend7`	`[TK]D-Fender: Okay...what that means exactly?`
`15:28.10`	`JT`	`sent after the number i meant to say`
`15:30.36`	`[TK]D-Fender`	`indend7: it means.. IT BUSY. How much bigger do you need to warning to be? Perhaps in flashing NEON?`
`15:30.55`	`plla`	`codey: compile with ./configure --prefix=/some/directory/asterisk they can share whatever is in /var/*/asterisk in the /etc/asterisk/asterisk.conf change the "astmoddir" to point to your asterisk lib/modules directory.`
`15:31.05`	`[TK]D-Fender`	`JT : Don't need wait for CID on analog..... ;)`
`15:31.15`	`JT`	`[TK]D-Fender: ?`
`15:31.18`	`JT`	`what?`
`15:31.58`	`[TK]D-Fender`	`JT : If Zaptel is told to "usecallerid=yes" dialplan will not start executing until after the appropriate wait has already taken place.`
`15:32.13`	`JT`	`[TK]D-Fender: but i'm not talking about analogue`
`15:32.23`	`[TK]D-Fender`	`JT : your callerID won't change in the middle of your dialplan you know, the channel has to START as "complete"`
`15:32.27`	`codey`	`plla: which configure script? there is no configure script :)`
`15:32.34`	`[TK]D-Fender`	`JT : And what else would you have to WAIT for CID on?`
`15:32.39`	`JT`	`[TK]D-Fender: callerid NAME`
`15:32.40`	`plla`	`from the source.`
`15:32.46`	`JT`	`it is sent in a message after setup`
`15:33.00`	`tzafrir_laptop`	`in 1.2 the makefiles do their own directory lookups`
`15:33.03`	`plla`	`asterisk source files, you require it to compile anything.`
`15:33.04`	`[TK]D-Fender`	`JT : All comes in at the same time...`
`15:33.14`	`JT`	`[TK]D-Fender: not from what i've seen`
`15:33.15`	`[TK]D-Fender`	`JT : part of the same FSK.`
`15:33.17`	`tzafrir_laptop`	`codey, -^`
`15:33.19`	`JT`	`on digital`
`15:33.22`	`JT`	`not fsk`
`15:33.23`	`codey`	`plla: there's no configure script in the bristuff directory. neither in its subdirectories (like asterisk)`
`15:33.24`	`plla`	`ah yes, I remember. :\ well, edit MakeFile`
`15:33.26`	`*** join/#asterisk sulan (n=ksjoberg@1-1-4-23a.lio.sth.bostream.se)`
`15:33.28`	`JT`	`there's a SETUP Q.931 message`
`15:33.37`	`FreezeS`	`I'm getting a lot of theese: channel.c: Internal timing is disabled (option_internal_timing=0 chan->timingfd=63)`
`15:33.39`	`codey`	`tzafrir_laptop: ?`
`15:33.40`	`[TK]D-Fender`	`JT : I assembled CID boxes before for a Bell tech's home PBX (ANCIENT Vantage 25 series :))`
`15:33.43`	`JT`	`the name comes in on a supplementary Q.931 message`
`15:33.46`	`FreezeS`	`is that a bad thing ?`
`15:34.41`	`[TK]D-Fender`	`JT : and * wiats for the complete call to be ready. You can't have shit change behind *'s back once the dialplan is executing, that'd be like pulling the rug out from under it! How silly!`
`15:34.51`	`plla`	`JT: I have found this: http://bugs.digium.com/view.php?id=6259 it describes the same problem I am having.`
`15:35.08`	`JT`	`[TK]D-Fender: how does it know if the telco will send a name or not?`
`15:35.22`	`[TK]D-Fender`	`JT : it either gets it, or it doesn't.`
`15:35.37`	`[TK]D-Fender`	`JT : just like SIP headers. I get it, if I get it.`
`15:35.44`	`JT`	`meh, i've seen this behaviour with us telcos before, but feel free to pointlessly debate it`
`15:35.49`	`JT`	`i know what i've seen :)`
`15:36.16`	`[TK]D-Fender`	`JT : you should join #sasquatch to and join the lookout! ;)`
`15:36.36`	`sulan`	`hi guys. I want to originate a new outgoing call from an AGI. I use AstMan for this, and send it to an extension that launches another AGI. However, for the outgoing call to be placed, the AGI need to answer first. How do I know if the remote party has answered in the AGI so I can play an automated message?`
`15:36.39`	`JT`	`i am talking about PRIs with callerid numbers coming in over SETUP and the name coming in later messages after call proceeding`
`15:36.45`	`*** join/#asterisk CunningPike_ (n=CunningP@204.239.12.183)`
`15:37.38`	`sulan`	`(I later need to bridge the original incoming call with the outgoing call, if the callee chooses to accept the call)`
`15:37.49`	`*** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com)`
`15:37.50`	`sulan`	`*called`
`15:40.06`	`JT`	`q.931 messages can be sent at any stage of a call, hence the whole out of band signalling thing :)`
`15:42.26`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`15:44.24`	`indend7`	`[TK]D-Fender: Thanks...`
`15:45.24`	`*** join/#asterisk saftsack (n=oliver@p54A7FDA3.dip.t-dialin.net)`
`15:46.04`	`sulan`	`maybe I should just reverse channel/extension and be happy. :)`
`15:48.58`	`*** join/#asterisk mcf3782 (n=mfreeman@ip67-90-136-181.z136-90-67.customer.algx.net)`
`15:50.16`	`[TK]D-Fender`	`sulan: What you are looking for is AMI, and if the other end answers, THEN it will Bridge to the requested part of your dialplan to do whatever you want.`
`15:51.33`	`mcf3782`	`Anyone in here familiar with app_cepstral?`
`15:51.54`	`*** join/#asterisk dcm_ (n=dcm@207.59.3.77)`
`15:52.26`	`denon`	`hmm .. someone needs to invent a good quality, cheap sip wifi phone`
`15:53.15`	`[TK]D-Fender`	`denon: Like the sign at my local bar/pool-hall says "Good. Fast. Cheap. Pick TWO."`
`15:53.25`	`denon`	`nod`
`15:53.37`	`denon`	`I'll take good and cheap`
`15:54.52`	`[TK]D-Fender`	`denon: Ok, I'll like you to it.... in about 5 years time ;)`
`15:55.08`	`[TK]D-Fender`	`denon: But don't complain that its not 6G compatible!`
`15:55.12`	`denon`	`nah, I meant I dont mind if it takes a couple minutes to boot`
`15:55.31`	`[TK]D-Fender`	`denon: How's 5 years? ;)`
`15:55.39`	`denon`	`hah, yeah that'll be ok`
`15:55.41`	`denon`	`give me the phone now`
`15:55.44`	`denon`	`I'll wait for the 5y bootup`
`15:55.54`	`[TK]D-Fender`	`denon: only 2628000 minutes!`
`15:56.01`	`denon`	`hehe`
`15:56.46`	`denon`	`tk: you ever played with the 7920s?`
`15:57.41`	`mcf3782`	`I found it out on the net when I was looking for a way to use the Cepstral voice synthesis package with my Asterisk box. Works well with my 1.2.9.1 install; but won't compile under 1.4.4, and I havent' been able to find its author to see if he's got a new version.`
`15:57.52`	`denon`	`oh, it doesnt do any SIP`
`15:59.10`	`*** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com)`
`16:00.38`	`*** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net)`
`16:00.54`	`magic_hat`	`hey everyone. wondering if there's an easy way to set up * to receive faxes.`
`16:01.17`	`denon`	`criminy, linksys's new IP over Power stuff is 200Mbps`
`16:01.19`	`denon`	`that's insane`
`16:01.26`	`coppice`	`nope/ nobody has ever set up * to receive faxes`
`16:01.37`	`denon`	`haha - ironic, coming from coppice`
`16:02.28`	`magic_hat`	`coppice, uh.... lol`
`16:03.47`	`*** join/#asterisk saftsack (n=oliver@p54A7E5FD.dip.t-dialin.net)`
`16:04.36`	`*** join/#asterisk DaveCanoe (n=Dave@H6.C30.B96.tor.eicat.ca)`
`16:05.36`	`magic_hat`	`no, seriously. faxes?`
`16:06.25`	`perf3kt`	`magic: everythign workign good for you?`
`16:06.55`	`magic_hat`	`perf3kt: pretty much, yeah. I had to up the bandwidth on our ISP yesterday, but other than that the system is totally functional.`
`16:06.56`	`magic_hat`	`you?`
`16:07.47`	`*** part/#asterisk oej (n=olle@apollo.webway.se)`
`16:11.43`	`eald`	`hi, just a small question, how do you use Monitor application in order to get both legs syncronized in the recorded file?`
`16:12.05`	`rob0`	`I think that's MixMonitor.`
`16:12.09`	`*** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ma.comcast.net)`
`16:13.25`	`*** join/#asterisk srd2 (n=srd@207.80.63.129)`
`16:13.37`	`eald`	`I guess that too, just that MixMonitor stop recording calls at random`
`16:13.41`	`srd2`	`when I try to make a phone call with my 7920 I get:`
`16:13.42`	`srd2`	`[Jun 19 17:13:07] WARNING[35536]: chan_skinny.c:1255 find_subchannel_by_instance_reference: Could not find subchannel with reference '0' on '2005'`
`16:14.03`	`srd2`	`would anyone know what I am to do to fix that? I've looked through google without success.`
`16:20.17`	`*** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu)`
`16:22.59`	`*** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)`
`16:26.59`	`*** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse)`
`16:33.34`	`codey`	make[1]: Entering directory `/usr/local/src/bristuff-0.3.0-PRE-1y/asterisk-1.2.14/res'
`16:33.37`	`codey`	make[1]: *** No rule to make target `res_watchdog.so', needed by `all'. Stop.
`16:33.39`	`codey`	`:/`
`16:33.41`	`*** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca)`
`16:34.21`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`16:40.07`	`*** join/#asterisk trevarthan (n=jesse@c-71-59-54-137.hsd1.ga.comcast.net)`
`16:42.19`	`trevarthan`	`Hello. I have an odd request. Is there a way to get a zap channel to hold the line open until the other end hangs up? Unfortunately I don't need to record and playback anything. I just need to hold the line open and sit on it until the calling party hangs up. Any way to do that?`
`16:42.20`	`tzafrir`	`codey, get latest bristuff`
`16:42.34`	`tzafrir`	`it has a more recent asterisk`
`16:43.04`	`tzafrir`	`codey, also have a look at: http://updates.xorcom.com/astribank/bristuff/`
`16:43.32`	`tzafrir`	`latest bristuff is 1y-g , not 1y`
`16:44.12`	`tzafrir`	`anyway, there's a news entry on the wiki for: 2007-06-18 - Free Music on Hold for Asterisk in MP3 format and licenced as NonCommercial`
`16:44.33`	`tzafrir`	`Is there any value to MoH files licensed as non-commercial?`
`16:45.19`	`Corydon76-work`	`Sure, it provides example functionality`
`16:45.29`	`tzafrir`	`(not to mention it is mp3, and hence requires extra format conversion)`
`16:45.59`	`tzafrir`	`yeah, but not-totally-free moh files are already provided with asterisk...`
`16:46.10`	`tzafrir`	`and with a more liberal license`
`16:46.17`	`tzafrir`	`license`
`16:46.26`	`Corydon76-work`	`Mmmm, interesting`
`16:48.26`	`*** join/#asterisk _DAW (n=_DAW@72-12-58-58.wan.networktel.net)`
`16:48.30`	`[TK]D-Fender`	`trevarthan: when would * seize the line?`
`16:48.40`	`trevarthan`	`immediately`
`16:48.52`	`[TK]D-Fender`	`trevarthan: what would initiate this?`
`16:49.00`	`*** join/#asterisk tylerhunt (n=thunt@6-5-111-208-in-addr-arpa.omnispring.net)`
`16:49.21`	`trevarthan`	`a third party system. It makes a call to asterisk and asterisk needs to answer and wait.`
`16:49.54`	`trevarthan`	`This is all very rediculous and hackish. I understand that. I just need to know how to make it happen. :)`
`16:50.47`	`[TK]D-Fender`	`trevarthan: oh, asterisdk just needs to wait? easy. answer. Wait`
`16:51.04`	`trevarthan`	`will a wait hang up when the user hangs up?`
`16:51.06`	`[TK]D-Fender`	`then check if it should stop, otherwise keep waiting`
`16:51.12`	`trevarthan`	`ah.`
`16:51.15`	`trevarthan`	`a loop. nice`
`16:51.19`	`[TK]D-Fender`	`trevarthan: clearly`
`16:51.23`	`trevarthan`	`thanks`
`16:51.25`	`trevarthan`	`that'll work`
`16:51.35`	`Jingles`	`doesn't 'answer()' just sit there and hold the line open until someone hangs up?`
`16:52.05`	`trevarthan`	`Jingles: no, answer() immediately returns`
`16:52.28`	`trevarthan`	`it waits until the channel rings....`
`16:53.50`	`trevarthan`	`wait... how do I detect a hangup on the other end of the channel?`
`16:54.16`	`trevarthan`	`nm... the 'h' extension, right?`
`16:57.08`	*** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com)
`16:57.38`	`*** join/#asterisk harlequin516 (n=sham@styk.net)`
`16:57.56`	`harlequin516`	`My call never connects because of : -- Attempting native bridge of ... ? What is this?`
`16:59.35`	`*** join/#asterisk Splat (n=splat@home.heehawhills.com)`
`17:00.58`	`*** join/#asterisk holiday_42 (n=no@spike.wcta.net)`
`17:00.59`	`Corydon76-work`	`A native bridge happens when both protocols match up and there is no need for Asterisk to remain in the middle`
`17:01.30`	`Corydon76-work`	`i.e. if you run a Monitor on the channel, then Asterisk has to remain in the middle`
`17:01.57`	`[TK]D-Fender`	`trevarthan: If you want to do something on disconnect, sure`
`17:02.20`	`Corydon76-work`	`harlequin516: you might want to set 'canreinvite=no' on one or the other peer`
`17:02.23`	`_VoiceMeUp_COM`	`harlequin516 i think attempting message is when it fails`
`17:02.28`	`*** part/#asterisk trevarthan (n=jesse@c-71-59-54-137.hsd1.ga.comcast.net)`
`17:02.34`	`_VoiceMeUp_COM`	`there should be an addition message coded but they never did`
`17:02.46`	`_VoiceMeUp_COM`	`aka " SUCCESS" or "OVERIDE per app" or "FAIL"`
`17:03.02`	`_VoiceMeUp_COM`	`coz liek you are saying we never know waht /where and if it succesed`
`17:03.07`	`_VoiceMeUp_COM`	`succedded`
`17:03.08`	`_VoiceMeUp_COM`	`ah`
`17:03.09`	`Corydon76-work`	`_VoiceMeUp_COM: more likely the reinvite succeeded, but due to NAT, the media path wasn't set up correctly`
`17:03.19`	`_VoiceMeUp_COM`	`yeah`
`17:04.04`	`*** join/#asterisk viperdudeuk (n=viperdud@195.74.96.113)`
`17:06.58`	`holiday_42`	`help with asterisk manager please: manual Telnet to asterisk manager from the console works fine. But when I run a bash script to do it, I always get "connect attempt from x.x.x.x unable to authenticate" asterisk CLI verbosity is set to 999999999`
`17:06.58`	`Nugget`	`telnet is eeeeeeevil!`
`17:07.35`	`codey`	`-- Executing Dial("SIP/65", "ZAP/g1/1234\|60") in new stack`
`17:07.45`	`codey`	`Jun 19 19:06:53 NOTICE[2544]: app_dial.c:1089 dial_exec_full: Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)`
`17:08.24`	`Corydon76-work`	`holiday_42: uh, that would be because you aren't actually sending through telnet when using a script like that`
`17:08.46`	`Corydon76-work`	`holiday_42: if you want to use telnet, please learn Expect scripting`
`17:09.33`	`holiday_42`	`meh, a simple bash script should be fine, i would think`
`17:09.46`	`*** join/#asterisk KryoStoffer (n=kri@helium.kri.dk)`
`17:10.02`	`Corydon76-work`	`Good. You can ignore my advice and fail miserably, or you can take my advice. Your choice.`
`17:10.56`	`[TK]D-Fender`	`codey: Error says it all. BUSY`
`17:11.04`	`mvanbaak`	`lol Corydon76-work`
`17:11.17`	`mvanbaak`	`there you are. giving advice and all and _NOONE_ listens`
`17:11.28`	`mvanbaak`	`the joy of being in a -users channel`
`17:11.46`	`[TK]D-Fender`	`Corydon76-work: Like they say about marriage "You can right.... or you can be HAPPY. 'Yes, dear.'".`
`17:11.59`	`Corydon76-work`	`I'm not going to get my panties in a wad about someone who won't listen...`
`17:12.00`	`[TK]D-Fender`	`mvanbaak: thats the point of advice.`
`17:12.15`	`[TK]D-Fender`	`mvanbaak: People take what they want and discard the rest.`
`17:12.31`	`*** part/#asterisk holiday_42 (n=no@spike.wcta.net)`
`17:12.32`	`codey`	`[TK]D-Fender: but thats like ... not possible.`
`17:12.36`	`[TK]D-Fender`	`mvanbaak: Those that are dead set on the path of stupidity get....`
`17:12.41`	`[TK]D-Fender`	`~ygwypf`
`17:12.42`	`jbot`	`extra, extra, read all about it, ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.`
`17:12.44`	`[TK]D-Fender`	`and...`
`17:12.46`	`[TK]D-Fender`	`~wglwat`
`17:12.47`	`jbot`	`[wglwat] well, good luck with all that`
`17:12.49`	`[TK]D-Fender`	`^^^^^^^^^^^^^^`
`17:13.44`	`[TK]D-Fender`	`codey: Apparently NOT. Check your configs if you must. Turn up the debug on your PRI (as thats an ISDN 34 - YES it IS busy) and see`
`17:13.45`	`mvanbaak`	`:)`
`17:14.03`	`*** join/#asterisk holiday_42 (n=no@spike.wcta.net)`
`17:14.07`	`[TK]D-Fender`	`codey: I've encountered that on my own NI1 PRI`
`17:14.26`	`codey`	`it's a gsm card..`
`17:14.40`	`[TK]D-Fender`	`codey: Connected to * HOW?`
`17:14.46`	`codey`	`?`
`17:15.02`	`codey`	`err.. it's a junghanns duoGSM.`
`17:15.03`	`[TK]D-Fender`	`codey: Link me to a page about this card of yours`
`17:15.13`	`*** join/#asterisk badcfe (n=cso@d83-177-231-219.cust.tele2.fr)`
`17:15.20`	`codey`	`http://www.junghanns.net/en/GSM-PCI_produkt.html`
`17:15.21`	`*** join/#asterisk eliter (n=eliter@66.179.79.69)`
`17:15.24`	`eliter`	`hey`
`17:15.33`	`[TK]D-Fender`	`codey: Perhaps they are using their ISDN interface internally to fake things out.`
`17:15.42`	`badcfe`	`i have an expert question about * re-INVITE mechanisme`
`17:16.27`	`eliter`	`I followed the gentoo setup for hylafax with an iaxmodem and the modem registers against the asterisk server but won't dial out, I keep getting : SEND FAILED: JOB 13 DEST 14125199225 ERR No answer from remote {E003}`
`17:16.29`	`eliter`	`any ideas?`
`17:17.07`	`[TK]D-Fender`	`codey: Well I can confirm the nature of that error code, how it is that the call came back as such is another matter I guess.... depends on how the rest of that unit operates`
`17:18.39`	`*** join/#asterisk mitcheloc (n=mitchelo@h46077954.area7.spcsdns.net)`
`17:18.41`	`[TK]D-Fender`	`codey: Thats a damn cool looking card though.`
`17:19.01`	`codey`	`maybe i just got the wrong slot`
`17:19.35`	`*** join/#asterisk Corydon76-home (i=white@pdpc/supporter/sustaining/Corydon76-home)`
`17:19.35`	`*** mode/#asterisk [+o Corydon76-home] by ChanServ`
`17:19.44`	`badcfe`	`if it takes a thousand characters to formulate my question, should i pastebin it? its about * re-INVITE`
`17:20.37`	`tzafrir_laptop`	`codey, again, are you using latest bristuff?`
`17:22.28`	`codey`	`-- Zap/1-1 is ringing`
`17:22.30`	`codey`	`yeah`
`17:23.18`	`*** join/#asterisk teh_recon (n=Recon@75.bartizanadsl.adsl.golden.net)`
`17:23.45`	`tzafrir_laptop`	`libgsmat is basically a fork of libpri`
`17:23.57`	`codey`	`tzafrir_laptop: its bristuff-0.3.0-PRE1-y`
`17:24.34`	`codey`	`err`
`17:24.36`	`codey`	`y-g`
`17:24.43`	`tzafrir_laptop`	`codey, latest is bristuff-0.3.0-PRE1y-g`
`17:25.00`	`codey`	`yes, its y-g`
`17:25.05`	`tzafrir_laptop`	`(ok, I have a dash missing there)`
`17:25.32`	`*** join/#asterisk Marshall-Laptop (n=eman0n@cpe-76-181-166-16.columbus.res.rr.com)`
`17:25.48`	`harlequin516`	`http://answers.yahoo.com/question/index?qid=20070504164406AA`
`17:26.57`	`harlequin516`	`yInBH/leave #`
`17:27.07`	`*** join/#asterisk saftsack (n=oliver@p54A7FC9E.dip.t-dialin.net)`
`17:27.18`	`harlequin516`	`Hmm, canreinvite=no for all my peers.`
`17:28.01`	`holiday_42`	`I tried expect (instead of a bash script) to connect via telnet to asterisk management... i STILL see "connect attempt from x.x.x.x unable to authenticate" whereas a manual telnet session works fine. both from the console`
`17:28.30`	`[TK]D-Fender`	`holiday_42: Since you're not showing us what you're doing, don't expect much help....`
`17:28.50`	`Jingles`	`besides. telnet is the devil.`
`17:28.53`	`Jingles`	`ssh ftw.`
`17:28.54`	`holiday_42`	`:)`
`17:29.16`	`Qwell[]`	`I telnet over an ssh tunnel`
`17:29.19`	`Qwell[]`	`so there`
`17:29.56`	`tzafrir_laptop`	`holiday_42, \r\n issue?`
`17:30.22`	`*** join/#asterisk Marshall- (n=Marshall@cpe-76-181-166-16.columbus.res.rr.com)`
`17:30.26`	`[TK]D-Fender`	`tzafrir_laptop: He never showed. :) go load chan_psychic.so ;)`
`17:31.25`	`tzafrir_laptop`	`filter what you send through sed -e 's/\n$/\r\n/' ? (untested)`
`17:31.30`	`holiday_42`	`yes, very possible! hmm. i would expect expect to get it right though`
`17:32.20`	`*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)`
`17:32.26`	`holiday_42`	`i will show you tkd, just a min please.`
`17:32.43`	`harlequin516`	`Any other ideas about -- Attempting native bridge of from sip?`
`17:33.46`	`harlequin516`	`I have a sipura box behind a firewall. I am sure this is the problem but, canreinvite=no is supposed to make it work as far as I know.`
`17:36.33`	`Jingles`	`ack! I lost the token!`
`17:37.13`	`[TK]D-Fender`	`harlequin516: You need a whole bunch of settings for NAT`
`17:38.16`	`*** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com)`
`17:44.39`	`*** join/#asterisk lucho81 (n=lgarcia@67.151.114.205)`
`17:44.47`	`lucho81`	`hi ..`
`17:45.00`	`*** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com)`
`17:49.22`	`*** join/#asterisk Hmmhesays (n=Neg@24-119-176-74.cpe.cableone.net)`
`17:50.57`	`*** join/#asterisk irule (n=irule@189.164.43.19)`
`17:54.00`	`*** join/#asterisk myiagy (i=myiagy@201.31.20.47)`
`17:54.44`	`*** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se)`
`17:55.37`	`*** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se)`
`17:55.58`	`*** join/#asterisk pfn (n=pfnguyen@hanhuy.com)`
`17:56.05`	`harlequin516`	`Ick..`
`17:56.11`	`harlequin516`	`whole bunch?`
`17:56.36`	`*** join/#asterisk rpm (n=rpm@S010600111155e117.vc.shawcable.net)`
`17:56.55`	`*** join/#asterisk pfn (n=pfnguyen@64.235.249.50)`
`17:59.15`	`[TK]D-Fender`	`harlequin516: For your remote phone (behind its own nat : "nat=yes", "canreinvite=no", "qualify=yes". If * is behind NAT as well, thats another pile of settings`
`18:00.10`	`blitzrage`	`externip and localnet`
`18:00.15`	`*** join/#asterisk saftsack (n=oliver@p54A7C1FB.dip.t-dialin.net)`
`18:00.46`	`[TK]D-Fender`	`harlequin516: Or externhost +externrefresh`
`18:00.49`	`harlequin516`	`Hmm qualify`
`18:00.52`	`harlequin516`	`Lemme see`
`18:01.15`	`russellb`	`[TK]D-Fender: have you ever set up a polycom to use TCP?`
`18:01.17`	`*** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)`
`18:01.19`	`[TK]D-Fender`	`blitzrage: I've seriously gotta install a CMS on my server so I can properly host articles.`
`18:01.26`	`mrdigital-work`	`where can i get a female american voice for festival?`
`18:01.44`	`[TK]D-Fender`	`russellb: Sure, provisioning through FTP is great! :)`
`18:02.14`	`blitzrage`	`[TK]D-Fender: ya.... getting a CMS setup is a bit of a pain in the ass if you don't want it to look stock`
`18:02.19`	`blitzrage`	`I really like Wordpress -- EASY to setup`
`18:02.25`	`russellb`	`[TK]D-Fender: well cool! We have our phones provisioned via FTP already ... I was just hoping you knew the option to make it use TCP instaed of UDP off of the top of your head ...`
`18:02.28`	`[TK]D-Fender`	`blitzrage: That'd do I'm sure...`
`18:02.42`	`russellb`	`[TK]D-Fender: someone here is hacking on making TCP/TLS work in chan_sip`
`18:02.42`	`blitzrage`	`russellb: isn't FTP using TCP by definition?`
`18:02.47`	`blitzrage`	`oooooh`
`18:02.47`	`russellb`	`yes, of course`
`18:02.49`	`russellb`	`i mean for SIP`
`18:02.50`	`blitzrage`	`you mean in the signalling`
`18:02.51`	`blitzrage`	`:)`
`18:02.55`	`blitzrage`	`ok, I'm all caught up :)`
`18:03.11`	`[TK]D-Fender`	`russellb: Oh you wanted to know about SIP via TCP ... Well you should have been more specific ;)`
`18:03.13`	`russellb`	`excuse me for not being clear`
`18:03.18`	`russellb`	`yeah yeah :-p`
`18:03.24`	`*** join/#asterisk pfn (n=pfnguyen@hanhuy.com)`
`18:03.42`	`russellb`	`or whatever that option is :-p`
`18:03.54`	`[TK]D-Fender`	`russellb: and I was 100% sure that's what you were referring to.... and I simply have nothing for one to TALK to using TCP for SIP.`
`18:04.20`	`russellb`	`well we're trying to fix that, heh`
`18:04.42`	`russellb`	`back to google then :)`
`18:04.46`	`[TK]D-Fender`	`russellb: that be -f"pdeantic,sarcastic,delusional,etc..."`
`18:05.05`	`[TK]D-Fender`	`russellb: Check out .... M$'s POS`
`18:05.33`	`[TK]D-Fender`	`russellb: And if you're looking for something to do... SIP-B! For ENDPOINTS ;)`
`18:05.40`	`rob0`	`uckhead? :) j/k btw :)`
`18:05.48`	`[TK]D-Fender`	`russellb: CPID! Merge!`
`18:06.00`	`russellb`	`no, i have plenty to do`
`18:06.10`	`russellb`	`i'm just trying to help bbryant get a SIP phone using TCP for testing`
`18:06.18`	`russellb`	`and obviously, i have never had this need before :)`
`18:07.32`	`justdave`	`anyone know how I enable res_snmp in make menuselect (1.4.5) ?`
`18:07.41`	`russellb`	`bbryant: i ... have no idea ...`
`18:08.05`	`russellb`	`bbryant: the sip/tcp bug in the tracker mentioned something`
`18:08.22`	`justdave`	`it's acting like I don't have the prereqs, but I do according to the docs... (net-snmp is installed, and so is net-snmp-devel)`
`18:08.34`	`russellb`	`justdave: did you re-run the configure script?`
`18:08.35`	`[TK]D-Fender`	`russellb: I seem to recall passing it in the admin guide. Was a single flag on the <reg`
`18:08.44`	`russellb`	`[TK]D-Fender: yeah, probably ...`
`18:09.01`	`bbryant`	`[TK]D-Fender, do you know where you found that guide?`
`18:09.09`	`russellb`	`bbryant: google :-p ... i got it`
`18:09.13`	`blitzrage`	`justdave: don't forget about libtool-ltdl-devel`
`18:09.15`	`[TK]D-Fender`	`bbryant: www.polycom.com <---------------`
`18:09.23`	`mrdigital-work`	`anyone know how to change the voice in festival to American Female?`
`18:09.31`	`justdave`	`russellb: yeah, didn't help.`
`18:09.35`	`justdave`	`blitzrage: ok, looking for that now`
`18:09.43`	`blitzrage`	`justdave: ya, it's a tricky one to know`
`18:10.34`	`*** join/#asterisk Lawbringer (n=Lawbring@84-45-215-247.no-dns-yet.enta.net)`
`18:12.16`	`justdave`	`blitzrage: installed, configure re-ran, still shows up as missing dependencies in make menuselect`
`18:12.42`	`blitzrage`	`check the log that the configure script generates`
`18:12.46`	`blitzrage`	`that'll tell you what you're missing`
`18:13.01`	`justdave`	`ok, I'll try that next if this didn't help`
`18:13.51`	`irule`	`hi there, I created a call queue, call in and get the default MOH, but there are no agents registered, how can I send the caller to a recording or something when there are no agents?`
`18:15.03`	`blitzrage`	`irule: leavewhenempty=yes; joinonempty=no`
`18:15.06`	`justdave`	`configure:20941: checking for snmp_register_callback in -lnetsnmp`
`18:15.12`	`justdave`	`/usr/bin/ld: cannot find -lz`
`18:15.19`	`justdave`	`nice :)`
`18:15.34`	`russellb`	`[TK]D-Fender: indeed ... reg.2.server.1.transport="TCPpreferred"`
`18:16.06`	`[TK]D-Fender`	`russellb: they make one damn fine manual, no? :)`
`18:16.22`	`irule`	`blitzrage thanks`
`18:16.34`	`blitzrage`	`irule: np`
`18:16.47`	`russellb`	`[TK]D-Fender: it's TERRIBLE`
`18:16.49`	`holiday_42`	`justdave: install libz? (or was it zlib?)`
`18:16.52`	`[TK]D-Fender`	`russellb: LIES!`
`18:17.00`	`blitzrage`	`holiday_42: ahhh yes, that is true`
`18:17.04`	`justdave`	`holiday_42: yeah, that was my thought. yum says it's already installed`
`18:17.10`	`blitzrage`	`zlib-devel?`
`18:17.17`	`blitzrage`	`(is there such a thing... I always forget :))`
`18:17.22`	`holiday_42`	`doh`
`18:17.23`	`justdave`	`yeah, just did that, that fixed it. now it can't find -lcrypto :)`
`18:17.31`	`blitzrage`	`yum install openssl-devel`
`18:17.53`	`irule`	`so is it possible to define where to leave to, when empty with the leavewhenempty=yes, and what to do when joinonempty=no?`
`18:18.05`	`blitzrage`	`irule: it'll just continue on in the dialplan`
`18:18.13`	`irule`	`ok thanks`
`18:18.15`	`blitzrage`	`so catch it after the Queue()`
`18:18.30`	`irule`	`oh I see thanks`
`18:18.31`	`blitzrage`	`if a call is answered in the Queue(), it won't follow that logic`
`18:18.50`	`justdave`	`now it can't find -lsensors`
`18:18.59`	`justdave`	`love dependency hell ;)`
`18:19.51`	`blitzrage`	`justdave: http://www.leifmadsen.com/blog/?p=11`
`18:20.25`	`blitzrage`	`~build_snmp`
`18:20.37`	`blitzrage`	`jbot: build_snmp is http://www.leifmadsen.com/blog/?p=11`
`18:20.37`	`jbot`	`okay, blitzrage`
`18:20.43`	`*** join/#asterisk rantsh (n=chatzill@201.210.16.238)`
`18:20.53`	`rantsh`	`hello everyone`
`18:21.07`	`*** join/#asterisk saftsack (n=oliver@p54A7C43F.dip.t-dialin.net)`
`18:21.29`	`rantsh`	`I've been having a horrible time with recording calls, can anyone PLEASE help me?`
`18:21.31`	`justdave`	`ok, lm_sensors-devel is the one for that`
`18:22.00`	`justdave`	`configure:20941: checking for snmp_register_callback in -lnetsnmp`
`18:22.01`	`justdave`	`configure:21000: result: yes`
`18:22.03`	`justdave`	`whee :)`
`18:22.12`	`rantsh`	`btw, am an * newbie so please be patient with me`
`18:23.08`	`blitzrage`	`justdave: :)`
`18:23.31`	`rantsh`	`I keep trying to record calls using either Monitor() or MixMonitor and all my recordings stop when the receiver picks up the phone... I'm using asterisk 1.4.0`
`18:23.31`	`*** join/#asterisk tekor (n=will@adsl-155-158-78.bhm.bellsouth.net)`
`18:24.21`	`blitzrage`	`1) you should probably be using 1.4.5...`
`18:24.36`	`justdave`	`my existing asterisk 1.2.x seems to be using a mysql-based CDR module... I don't see something like that in 1.4, is that available somewhere else, or not available for 1.4 or?`
`18:24.37`	`blitzrage`	`2) .... that seems strange... what does your dialplan look like? (use pastebin)`
`18:24.39`	`blitzrage`	`~pb`
`18:24.39`	`jbot`	`pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org`
`18:24.58`	`eliter`	`I followed the gentoo setup for hylafax with an iaxmodem and the modem registers against the asterisk server but won't dial out, I keep getting : SEND FAILED: JOB 13 DEST 1xxxyyyzzzz ERR No answer from remote {E003}`
`18:24.58`	`blitzrage`	`justdave: mysql stuff is installed via asterisk-addons`
`18:25.17`	`rantsh`	`oh well I've posted this in the * forums I can paste the link`
`18:25.21`	`rantsh`	`give me a sec, please`
`18:25.46`	`*** join/#asterisk QbY_ (n=Kelvin@66.236.241.67.ptr.us.xo.net)`
`18:26.16`	`rantsh`	`http://forums.digium.com/viewtopic.php?t=16283`
`18:26.21`	`QbY_`	`Anyone see a problem with this? exten => 2001,1,Dial(SIP/12345,L(120000:60000:30000)) -- I keep getting Jun 19 14:18:22 WARNING[30976]: app_dial.c:1214 dial_exec_full: Invalid timeout specified: '`
`18:27.09`	`file`	`QbY_: yes, you are specifying options in the place where a timeout is supposed to be`
`18:27.56`	`QbY_`	`its a one comma world.`
`18:28.07`	`Mercestes`	`eliter: Iaxmodem cannot dial out`
`18:28.21`	`walhala`	`HI all someine may explain tome what is the difference between chan_sccp and chan_skinny ?`
`18:28.50`	`eliter`	`Mercestes, its only so hylafax with iaxmodem is only used for receiving?`
`18:28.54`	`Qwell[]`	`walhala: chan_sccp is entirely unsupported and dead. chan_skinny rocks, is supported, actually works, and is maintained`
`18:28.58`	`Mercestes`	`eliter: Yes.`
`18:29.14`	`robl^`	`there is also chan_sccp2 and chan_sccp3 ;-)`
`18:29.20`	`Qwell[]`	`both of which are also dead`
`18:29.21`	`eliter`	`Mercestes, thanks a lot, how would you suggest sending, we are trying to send multipage pdfs`
`18:29.38`	`Mercestes`	`eliter: I guess you could somehow create a context for Iaxmodem to dial out on and send a fax to it, and it can automatically dial a string but....`
`18:29.38`	`harlequin516`	`Damn why can't the stupid sipura boxes just implement iax instead of sip`
`18:29.57`	`walhala`	`Qwell[]: but can I pickup calls, play with my 7914 and hint ?`
`18:30.01`	`Qwell[]`	`pretty much all of the people who were working on chan_sccp that have the ability to add features or fix major bugs, are all working on chan_skinny now`
`18:30.05`	`Qwell[]`	`walhala: in trunk, you can, yes`
`18:30.19`	`Mercestes`	`eliter: When i did sending I had PRIs available and defaulted to a PRI`
`18:30.29`	`robl^`	`7914 support is in chan_skinny now?? wow`
`18:30.34`	`walhala`	`Qwell[]: so not with 1.2.x ?`
`18:30.42`	`Qwell[]`	`walhala: no, I wouldn't recommend using chan_skinny in 1.2`
`18:30.51`	`eliter`	`Mercestes, we were trying the direct approach, but the tiff from the pdf was timing out and the fax machine was receiving only 1 page out of 10`
`18:31.05`	`Qwell[]`	`it was very majorly redesigned and fixed in 1.4. a couple of new features have been added to trunk, such as adding hints`
`18:31.18`	`walhala`	`Qwell[]: ok thanks for this information :) but just an other question ... If use 1.2.x that's because i have snom`
`18:31.24`	`Qwell[]`	`snom works on 1.4...`
`18:31.33`	`Qwell[]`	`anybody who told you otherwise is very much wrong :)`
`18:31.38`	`walhala`	`pick up too ?`
`18:31.40`	`*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)`
`18:31.52`	`Qwell[]`	`if it was in 1.2, I don't see why it wouldn't work in 1.4...`
`18:32.06`	`walhala`	`because we should patch chan_sip.c to pick up a call with the keys`
`18:32.28`	`walhala`	`that's why i'm still on 1.2.x`
`18:32.32`	`rantsh`	`can anyone tell me why was * divided into 2 branches (1.2 and 1.4)?`
`18:32.58`	`Qwell[]`	`rantsh: features don't go into release branches.. 1.2 was an older release branch, a bunch of stuff was added to trunk, which was then branched as 1.4`
`18:33.01`	`walhala`	`if it's working on 1.4.x that's will be so great !`
`18:33.02`	`Mercestes`	`eliter: Looks like it migh tbe possible but I'm not certain`
`18:33.12`	`rantsh`	`blitzrage: did you get to see the link a pasted?`
`18:33.19`	`Qwell[]`	`and eventually all the new features that are currently in trunk will be branched as 1,6`
`18:33.33`	`eliter`	`Mercestes, it seems really sketchy, and there is no real valid test, it registers but can't dial out....its strange...`
`18:33.44`	`Mercestes`	`eliter: Does it have a default context?`
`18:33.58`	`walhala`	`Qwell[]: no information about pickup keys and snom in 1.4.x ?`
`18:34.09`	`Qwell[]`	`walhala: no idea, sorry`
`18:34.22`	`rantsh`	`so 1.2 is eventually going to be deprecated, presumably earlier than later... no?`
`18:34.31`	`eliter`	`Mercestes: ya`
`18:34.34`	`Mercestes`	`eliter: and in that context, do you have the exten => xxxxxxxxx,1,Dial(Tech/User) that allows it to dial out?`
`18:34.46`	`walhala`	`Qwell[]: ok thanks very munch for your prescious advertissement :)`
`18:34.47`	`Qwell[]`	`1.2 is already "deprecated". There will be no more bug fixes in 1.2 starting August 1st - only security fixes.`
`18:34.58`	`rantsh`	`:p`
`18:35.08`	`Qwell[]`	`For anybody not already looking at 1.4, now is definitely the time to start. You have just over a month left.`
`18:35.09`	`eliter`	`Mercestes, I'll have to check with the guy who set it up, but he did previously say it did`
`18:35.24`	`Qwell[]`	`bug reports against 1.2 will be pretty much closed without resolution...`
`18:35.25`	`Mercestes`	`K, let us know`
`18:35.35`	`Mercestes`	`Qwell[]: Isn't that how it works now?`
`18:35.37`	`Mercestes`	`>.>`
`18:35.42`	`Qwell[]`	`(unless the same bug is in 1.4, in which case it'll be fixed in 1.4, and not 1.2)`
`18:35.51`	`rantsh`	`is it too traumatical to upgrade from 1.4.0 to 1.4.5`
`18:35.53`	`rantsh`	`?`
`18:35.56`	`Qwell[]`	`rantsh: no`
`18:36.08`	`Qwell[]`	`minor versions shouldn't have anything changed in the way of configs, etc`
`18:36.33`	`rantsh`	`then again i come to my original reason i logged in today`
`18:37.45`	`rantsh`	`we need to record some outgoing calls, this worked perfectly in our 1.2.3 * server, we updgraded (testing) to 1.4.0 and the same dialplan that used to record perfectly doesn't work`
`18:37.48`	`*** join/#asterisk peanutb (n=paulb@c-24-16-243-186.hsd1.mn.comcast.net)`
`18:37.55`	`rantsh`	`any ideas on what the problem could be?`
`18:37.57`	`eliter`	`Mercestes, this is the context:;exten => 2201,1,Answer()`
`18:37.57`	`eliter`	`;exten => 2201,n,Wait(4)`
`18:37.58`	`eliter`	`;exten => 2201,n,NoOp("NO FAX PICKUP")`
`18:37.58`	`eliter`	`;exten => 2201,n,Macro(faxtransmit)`
`18:37.58`	`eliter`	`;exten => t,1,GoTo(fax,1)`
`18:37.58`	`eliter`	`exten => _.,1,Answer()`
`18:37.59`	`walhala`	`Qwell[]: for information : http://bugs.digium.com/view.php?id=5014`
`18:38.02`	`eliter`	`exten => _.,n,Set(CALLERID(num)=xxxyyyzzzz)`
`18:38.04`	`eliter`	`exten => _.,n,Dial(Zap/g4/${EXTEN})`
`18:38.05`	`Mercestes`	`...`
`18:38.10`	`Mercestes`	`~pb`
`18:38.11`	`jbot`	`rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org`
`18:38.26`	`Qwell[]`	`rantsh: it'll tell you if any of the applications are deprecated`
`18:38.30`	`walhala`	`Qwell[]: simple pick up may work with snom 320 and 1.4.x :)`
`18:38.36`	`eliter`	`sorry bout that, my bad`
`18:39.30`	`Mercestes`	`eliter: first, never use _. I hope you didn't pay the guy that set that up`
`18:39.34`	`rantsh`	`Qwell[]: I'm sorry, what will? I think I didn't get that line :)`
`18:39.46`	`Qwell[]`	`and applications that "don't work"`
`18:40.06`	`Qwell[]`	`You'd have to show somebody your dialplan and any errors you get, for somebody to debug it`
`18:40.55`	`rantsh`	`I did post it in the * forums, don't know how to use this pastbin apps.... :S`
`18:40.56`	`Mercestes`	`eliter: two, that answer is in a retarded place.`
`18:41.29`	`rantsh`	`either way, the link to my post is this, I'd very much appreciate if any one can help me a little here: http://forums.digium.com/viewtopic.php?t=16283`
`18:41.42`	`*** join/#asterisk imperial- (n=nick@weld.imperial.org)`
`18:42.01`	`*** join/#asterisk saftsack (n=oliver@p54A7EEA7.dip.t-dialin.net)`
`18:42.28`	`imperial-`	`anyone use the Asterisk::Manager perl module?`
`18:42.43`	`*** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca)`
`18:42.49`	`_VoiceMeUp_COM`	`need your wize knoweledge`
`18:43.04`	`Mercestes`	`rantsh: Use record, not monitor.`
`18:43.21`	`_VoiceMeUp_COM`	`why ?`
`18:43.25`	`_VoiceMeUp_COM`	`monitor bad ?`
`18:43.34`	`Mercestes`	`wait, I lied`
`18:43.39`	`rantsh`	`is it any better? i tried monitor and Mixmonitor`
`18:43.54`	`rantsh`	`someone told me record is a one way recording or something`
`18:44.01`	`Mercestes`	`Lemme look`
`18:44.25`	`*** join/#asterisk kombi_ (n=kombi@213.160.14.18)`
`18:44.45`	`eliter`	`Mercestes, is there any reason that wouldn't pick up the line though?`
`18:45.20`	`Mercestes`	`rantsh: What is not working about that?`
`18:45.38`	`Mercestes`	`I think monitor_exec broke on me at some point too (and you have to use \|m to make it run monitor_exec anyways)`
`18:45.43`	`kombi_`	`when there is one way audio just on incoming calls, is there a hint on where to look for the cause?`
`18:45.44`	`rantsh`	`it records the ringing, stops recording when receiver picks up`
`18:45.51`	`Mercestes`	`oh`
`18:46.41`	`rantsh`	`I'm thinking it may be a version bug, but can't risk to upgrade my server now, and don't have anymore pc's to test it out`
`18:46.45`	`Mercestes`	`rantsh: welll, I would try upgrading, and barring that, file a bug report, I guess, your code looks right to me.`
`18:47.12`	`rantsh`	`that's what i tought`
`18:47.19`	`kombi_`	`..or is it the sip provider that is responsible?`
`18:47.33`	`rantsh`	`(sorry, my english is getting a little rusty :D )`
`18:47.55`	`rob0`	`Host access issue was fixed by RTFWiki. "SIP/2.0 407 Proxy Authentication Required" ... Google points to a lot of questions, but no answers. I'm trying to receive inbound calls from Asterlink.`
`18:48.50`	`rantsh`	`either way thanks qwell[] and Mercestes... you've sort of confirmed what i suspected`
`18:49.06`	`Mercestes`	`np, good luck`
`18:49.16`	`rantsh`	`so I'll just try to get a testing server with a newer version and see what comes out from there`
`18:49.24`	`kombi_`	`which is better, ethereal or wireshark?`
`18:49.46`	`*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)`
`18:49.46`	`kombi_`	`..rolling his sleeves up to dig into packet sniffing..`
`18:49.59`	`rantsh`	`I wish I could help others, but I'm still a n00b in *, if anyone needs php help I can help there...`
`18:50.12`	`Mercestes`	`Cool. I use agiphp :)`
`18:50.23`	`rantsh`	`kombi_ isn't it the same, but they changed the name or something`
`18:50.49`	`rantsh`	`feel free to contact me whenever you need`
`18:50.51`	`Hmmhesays`	`wireshark is what ethereal turned into I believe`
`18:50.59`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`18:51.07`	`Mercestes`	`you are correct, Hmmhesays`
`18:51.17`	`kombi_`	`Hmmhesays: does it run from command line?`
`18:51.41`	`kombi_`	`rantsh: you're right, I just checked..`
`18:52.17`	`kombi_`	`a hefty 64 Meg install but I guess there is no way around it`
`18:52.31`	`imperial-`	`so any of you guys spend any time w/ Asterisk::Manager?`
`18:53.36`	`rantsh`	`Alright guys, thank you all for your help`
`18:53.48`	`*** join/#asterisk kushal06062007 (n=kushal06@202.70.69.64)`
`18:53.55`	`rantsh`	`I'll come back later, have a nice one`
`18:54.58`	`*** part/#asterisk kushal06062007 (n=kushal06@202.70.69.64)`
`18:55.31`	`kombi_`	`oh jeez, 2400 line man page..`
`18:55.53`	`Hmmhesays`	`Atlantis undocks from space station. Claims to still respect space station. Promises to call`
`18:57.32`	`rob0`	`Outta this world!`
`18:58.15`	`blitzrage`	`Hmmhesays: lol`
`18:59.29`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`18:59.35`	`*** join/#asterisk oej (n=olle@apollo.webway.se)`
`19:01.09`	`holiday_42`	`[TK]D-Fender, tzafrir_laptop: issues solved. the whole "request" must end with /r/n/r/n. a "request" may be comprised of one or more lines. nothing to do with using bash or expect technically.`
`19:01.33`	`holiday_42`	`but i knew that`
`19:02.10`	`*** join/#asterisk robin_sz (n=robin@rapid2.gotadsl.co.uk)`
`19:02.36`	`*** join/#asterisk saftsack (n=oliver@p54A7DE67.dip.t-dialin.net)`
`19:02.41`	`robin_sz`	`Hi,`
`19:03.11`	`robin_sz`	`I am trying to dial out over IAX and keep getting a "format unknown' probelm after the call is trying to establish`
`19:03.28`	`robin_sz`	`Call accepted by 193.111.201.75 (format unknown)`
`19:03.34`	`robin_sz`	`clooes?`
`19:04.02`	`[TK]D-Fender`	`robin_sz: Set your codecs`
`19:04.18`	`holiday_42`	`i would guess the codec... try disable=all enable=<whatever>`
`19:04.19`	`robin_sz`	`where?`
`19:04.27`	`robin_sz`	`in iax.conf?`
`19:05.20`	`Jingles`	`there's a ${CALLERIDNUM} - is there a similar ${CALLERSERVER} or something?`
`19:06.15`	`robin_sz`	`ok, it seems my provider supports G711u and G729a ... probably somehting related to that`
`19:06.17`	`*** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com)`
`19:06.22`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`19:07.02`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net)`
`19:08.19`	`_VoiceMeUp_COM`	`so`
`19:08.25`	`_VoiceMeUp_COM`	`can we jump out a macro ?`
`19:08.39`	`_VoiceMeUp_COM`	`like s,1,blah s,2, jump out.. or exit..`
`19:08.45`	`_VoiceMeUp_COM`	`s,3, other stuff`
`19:09.04`	`_VoiceMeUp_COM`	`or i jump a prio ?`
`19:09.25`	`*** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga)`
`19:09.27`	`blitzrage`	`MacroExt()`
`19:09.31`	`blitzrage`	`MacroExit()`
`19:09.38`	`*** join/#asterisk Zion800 (n=None@cpe-76-167-156-224.socal.res.rr.com)`
`19:10.32`	`_VoiceMeUp_COM`	`GotoIf( $[${LEN(${FOOBAR})} > 0]?2:5)`
`19:10.33`	`_VoiceMeUp_COM`	`ok`
`19:10.34`	`_VoiceMeUp_COM`	`thanks`
`19:10.38`	`robin_sz`	`ahh, wait. G729 and 711 ... are they both uncompressed codecs, ie 64K?`
`19:10.39`	`_VoiceMeUp_COM`	`this doesnt match up`
`19:10.44`	`blitzrage`	`use priority labels`
`19:10.48`	`_VoiceMeUp_COM`	`but a noop on 2 shows found 10`
`19:10.53`	`blitzrage`	`using numbered priorities is just wrong`
`19:11.12`	`_VoiceMeUp_COM`	`so foobar = 10 chars`
`19:11.17`	`_VoiceMeUp_COM`	`but not evaliation`
`19:11.20`	`Zion800`	`Hey, can anyone help me edit the page.agi script found on Voip-Info.org? It worked in Asterisk 1.2, but now that the output for "show hints" has slightly changed in Asterisk 1.4, the script doesnt work.`
`19:11.57`	`harlequin516`	`Okay I can't figure it out... I even put a W option in my Dial command and it still attempts to do a native bridge... Any ideas?`
`19:12.54`	`irule`	`is there a way to see a queue caller's priority while waiting? I would like to save that priority to database for callback instead of always giving them 10`
`19:13.15`	`_VoiceMeUp_COM`	`yeah labels suck`
`19:13.24`	`_VoiceMeUp_COM`	`hmm no`
`19:13.43`	`_VoiceMeUp_COM`	`label is a prio alias right ? or exten`
`19:13.49`	`_VoiceMeUp_COM`	`coz exten wont work`
`19:13.56`	`_VoiceMeUp_COM`	`as in exten => label,1,noop`
`19:14.23`	`_VoiceMeUp_COM`	`ah`
`19:14.25`	`_VoiceMeUp_COM`	`nm`
`19:14.39`	`Zion800`	`Asterisk Priority Lables: http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities`
`19:15.14`	`_VoiceMeUp_COM`	`nah i jusut needed to gotoif ? label\|1:label2\|1`
`19:15.19`	`Zion800`	`ah..ok`
`19:15.27`	`_VoiceMeUp_COM`	`but when using just prios' i didnt have that`
`19:15.39`	`_VoiceMeUp_COM`	`guess its as much as goo dpractce as escaping table names in sql`
`19:15.58`	`harlequin516`	`Any ideas on how to completely disable native bridging?`
`19:17.17`	`Zion800`	`Anyone wanna help me out with a small perl script?`
`19:18.18`	`*** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net)`
`19:19.14`	`*** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29)`
`19:20.55`	`*** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il)`
`19:21.11`	`srd2`	`when I try to make a phone call from my 7920, all I get is: -- Starting simple switch on '2005@wifi'`
`19:21.24`	`srd2`	`what do I need to do to have it be able to make phone calls?`
`19:21.30`	`Qwell[]`	`dial`
`19:21.52`	`srd2`	`i do, it does nothing but give me a dialtone`
`19:23.22`	`*** join/#asterisk saftsack (n=oliver@p54A7CEEE.dip.t-dialin.net)`
`19:23.57`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`19:25.23`	`mrdigital-work`	`is exten => x,x,MYSQL( feature in all asterisk installs?`
`19:25.25`	`mrdigital-work`	`by default?`
`19:26.23`	`FoxTrot-`	`yep`
`19:26.32`	`Juggie`	`no`
`19:26.42`	`Juggie`	`you need asterisk-addons`
`19:26.47`	`*** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)`
`19:26.53`	`Juggie`	`but you should use func_odbc`
`19:27.52`	`Nuitari`	`Juggie: why not mysql?`
`19:28.07`	`Juggie`	`because its not really maintained, func_odbc is much better`
`19:28.10`	`Juggie`	`you can still use mysql`
`19:28.17`	`Juggie`	`just mysql through odbc`
`19:28.19`	`srd2`	`now I get:`
`19:28.19`	`srd2`	`[Jun 19 20:27:54] WARNING[36042]: chan_skinny.c:1255 find_subchannel_by_instance_reference: Could not find subchannel with reference '0' on 'wifi'`
`19:28.27`	`*** join/#asterisk seele_ (n=seele@200.30.85.186)`
`19:28.33`	`codefreeze`	`Hey, everybody!!!! I forgot to ask!!! Did y'all have a wonderful Father's Day out there?`
`19:28.39`	`seele_`	`how can I change the CID format ??`
`19:29.00`	`rob0`	`codefreeze: best one for me in at least 10 years!`
`19:29.19`	`*** join/#asterisk Toerkeium (i=oo@201.216.206.221)`
`19:29.42`	`codefreeze`	`rob0: really? Me too! I had a ham dinner, treats, didn't have to do dishes. Wow!!!`
`19:30.27`	`codefreeze`	`rob0: what made it so good for you?`
`19:31.38`	`mrdigital-work`	`juggie: how do i use func_odbc?`
`19:33.41`	`[TK]D-Fender`	`mrdigital-work: "show function FUNC_ODBC"`
`19:34.37`	`Corydon76-work`	`[TK]D-Fender: uh, not quite`
`19:35.15`	`[TK]D-Fender`	`Corydon76-work: Ok, fine, sure, you can take over now :)`
`19:35.50`	`Corydon76-work`	`mrdigital-work: see configs/func_odbc.conf.sample`
`19:35.56`	`rob0`	`First Fathers' Day in 12 years that I heard from ALL my kids.`
`19:36.55`	`codefreeze`	`rob0: Are they all out of the house now?`
`19:37.17`	`_VoiceMeUp_COM`	`: chan_sip.c:9427 func_header_read: This function can only be used on SIP channels.`
`19:37.20`	`_VoiceMeUp_COM`	`so.. hmm`
`19:37.34`	`rob0`	`no, only one is, but I went a long time without being allowed to see him.`
`19:39.29`	`*** join/#asterisk stkn\|work (n=stkn\|wor@gentoo/developer/pdpc.active.stkn)`
`19:40.13`	`mrdigital-work`	`i got it`
`19:40.44`	`srd2`	`don't suppose anyone could help with:`
`19:40.45`	`srd2`	`[Jun 19 20:40:23] WARNING[36092]: chan_skinny.c:1255 find_subchannel_by_instance_reference: Could not find subchannel with reference '0' on 'wifi'`
`19:40.45`	`srd2`	`?`
`19:41.15`	`srd2`	`It happens when I try to make an outgoing call from a 7920`
`19:42.13`	`codefreeze`	`rob0: Cool!`
`19:42.59`	`robin_sz`	`bah, this IAX thing still istn working ... I put disallow=all allow=g729 inthe iax.conf, I still get "format unknown" when diallign out :(`
`19:44.03`	`*** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar)`
`19:44.20`	`*** join/#asterisk saftsack (n=oliver@p54A7D99C.dip.t-dialin.net)`
`19:45.03`	`[TK]D-Fender`	`robin_sz: PASTEBIN`
`19:45.17`	`*** join/#asterisk lucho81 (n=lgarcia@67.151.114.194)`
`19:45.19`	`lucho81`	`hi ..`
`19:45.48`	`lucho81`	`I'm looking for information about how to dimension an Asterisk server..`
`19:45.53`	`seele_`	`hello`
`19:45.58`	`lucho81`	`anybody may help me ..?`
`19:46.02`	`seele_`	`how can I change the caller ID format ??/`
`19:46.20`	`Corydon76-work`	`dimension?`
`19:46.57`	`seele_`	`actually my CID is LOC(PREFIX) NUMBER`
`19:47.31`	`seele_`	`I need to remove the LOC and leave only the NUMBER`
`19:47.34`	`lucho81`	`yes, .. how to determine how much memory in RAM, hard disk space, processor ... you ll need to implement Asterisk for 20 SIP users ....`
`19:47.52`	`srd2`	`anyone?`
`19:48.13`	`Corydon76-work`	`~thebook`
`19:48.14`	`jbot`	`i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`19:48.15`	`lucho81`	`and 20 analog endpoints ..`
`19:48.17`	`Corydon76-work`	`Read that`
`19:48.28`	`[TK]D-Fender`	`lucho81: lets say a basic 1GHZ PC w/ 256meg would probably be fine.`
`19:49.05`	`[TK]D-Fender`	`lucho81: HD is nearly irrelevent unless you're recrding calls all over the place.`
`19:49.27`	`[TK]D-Fender`	`lucho81: Even then with HD's these days it falls under the realm of "whocares?"`
`19:49.38`	`lucho81`	`ok, let me see ..`
`19:49.46`	`[TK]D-Fender`	`< $120 for 500 gigs... shees.`
`19:51.20`	`[TK]D-Fender`	`lucho81: Your servre could very easily cost a small fraction of the price of the PSTN connectivity card you'll likely buy for it :)`
`19:51.22`	`bkruse`	`nice!`
`19:51.26`	`bkruse`	`[TK]D-Fender: ide? sata?`
`19:51.42`	`[TK]D-Fender`	`bkruse: Either. Prices are par between them still`
`19:51.46`	`bkruse`	`wow`
`19:51.48`	`lucho81`	`hmm ..`
`19:51.49`	`bkruse`	`true`
`19:51.57`	`*** part/#asterisk mcf3782 (n=mfreeman@ip67-90-136-181.z136-90-67.customer.algx.net)`
`19:52.10`	`robin_sz`	`[TK]D-Fender, http://www.pastebin.ca/577118`
`19:53.16`	`[TK]D-Fender`	`robin_sz: ...... and the rest?! BTW, one of my clients used Gradwell for a London DID :)`
`19:53.30`	`robin_sz`	`[TK]D-Fender, rest?`
`19:53.39`	`robin_sz`	`[TK]D-Fender, liek my iax.conf?`
`19:53.54`	`[TK]D-Fender`	`robin_sz: Like I should even have to ASK :)`
`19:54.05`	`robin_sz`	`mmm, k`
`19:54.22`	`[TK]D-Fender`	`robin_sz: Of COURSE I don't trust your configs! If they were right.... they'd WORK!`
`19:55.40`	`robin_sz`	`well, its the virgin one from samples, just with disallow=all allow=g729 ..`
`19:55.51`	`[TK]D-Fender`	`~[TK]D-Fender:`
`19:55.56`	`[TK]D-Fender`	`~[TK]D-Fender`
`19:55.56`	`jbot`	`[TK]D-Fender is the Zen Master of the blatantly obvious.`
`19:56.01`	`[TK]D-Fender`	`:D`
`19:56.29`	`[TK]D-Fender`	`robin_sz: "Show me the money!" - Jerry Maguire`
`19:56.32`	`robin_sz`	`whats the sed thing for removign all ines begining with ;`
`19:56.49`	`[TK]D-Fender`	`robin_sz: "sed don't gimme no comments!"`
`19:57.47`	`*** join/#asterisk \|dennis\| (n=dennis@200.32.233.82)`
`19:57.48`	`*** join/#asterisk daveburr (n=Miranda@66.7.124.15)`
`19:58.07`	`_VoiceMeUp_COM`	`so`
`19:58.08`	`_VoiceMeUp_COM`	`sorry`
`19:58.17`	`_VoiceMeUp_COM`	`is there a way to see the channel type ?`
`19:58.22`	`_VoiceMeUp_COM`	`tough there was is local`
`19:58.23`	`*** join/#asterisk asterisknerds (n=asterisk@66.7.122.93)`
`19:58.24`	`asterisknerds`	`<PROTECTED>`
`19:58.25`	`_VoiceMeUp_COM`	`is chanislocal()`
`19:59.26`	`robin_sz`	`I'll paste as soon as ive stripped all the comments`
`19:59.56`	`kombi_`	`why does music on hold stop immediately after being started?`
`20:00.38`	`*** join/#asterisk _DAW (n=chatzill@adsl-222-30-84.msy.bellsouth.net)`
`20:01.05`	`kombi_`	`there is a notice before about schedule in the past but that shouldn't harm, right?`
`20:01.06`	`_VoiceMeUp_COM`	`hmm`
`20:01.15`	`_VoiceMeUp_COM`	`DumpChan(Type) maybe`
`20:01.32`	`[TK]D-Fender`	`_VoiceMeUp_COM: ${CHANNEL} <- use your imagination`
`20:01.58`	`_VoiceMeUp_COM`	`yeah im cuting and regexing like crazy`
`20:02.03`	`[TK]D-Fender`	`kombi : means you are using MPG123 for MoH, and that warning is mostly harmless.`
`20:02.46`	`[TK]D-Fender`	`_VoiceMeUp_COM: ${CHANNEL:3} should be more than enough to branch however you like.`
`20:02.56`	`kombi_`	`but why does it stop just after it starts?`
`20:03.17`	`kombi_`	`moh I mean.. does mpg123 log to somewhere?`
`20:03.45`	`robin_sz`	`[TK]D-Fender, http://www.pastebin.ca/577151`
`20:03.57`	`robin_sz`	`[TK]D-Fender, anything else?`
`20:04.25`	`[TK]D-Fender`	`robin_sz: Make a peer for gradwell.... put everything in there.`
`20:04.34`	`robin_sz`	`a peer?`
`20:04.35`	`[TK]D-Fender`	`robin_sz: and I believe they like ALAW`
`20:04.39`	`[TK]D-Fender`	`robin_sz: Yes`
`20:04.44`	`robin_sz`	`G729 and 711`
`20:04.52`	`robin_sz`	`a peer, for ougoing only?`
`20:05.05`	`[TK]D-Fender`	`robin_sz: Correct`
`20:05.14`	`robin_sz`	`hey ho ...`
`20:06.43`	`*** part/#asterisk daveburr (n=Miranda@66.7.124.15)`
`20:06.44`	`kombi_`	`does it make sense to switch to rawplayer like they say on asteriskguru?`
`20:07.24`	`*** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net)`
`20:07.38`	`robin_sz`	`[TK]D-Fender, doen that, iax2 reload, no change`
`20:07.42`	`[TK]D-Fender`	`kombi : Read the differences and YOU decide`
`20:07.42`	`raidenz`	`hi guys`
`20:07.59`	`[TK]D-Fender`	`robin_sz: and your EXTENSIONS.CONF?`
`20:08.09`	`robin_sz`	`what about it?`
`20:08.10`	`_VoiceMeUp_COM`	`5thanks`
`20:08.15`	`raidenz`	`Is it possible to send/set variables to pass in a .call file?`
`20:08.29`	`[TK]D-Fender`	`robin_sz: Your dial was DIRECT before!`
`20:08.30`	`robin_sz`	`the whole thing, or just the [gradwell] bit?`
`20:08.42`	`robin_sz`	`?`
`20:08.46`	`[TK]D-Fender`	`robin_sz: It's not going to MIRACULOUSLY use taht new peer entry! ;)`
`20:08.56`	`robin_sz`	`err`
`20:09.14`	`[TK]D-Fender`	`robin_sz: fix everything up. Upon failure, make a consolidated pastebin of all the new material`
`20:09.56`	`tzafrir_laptop`	`anybody built the spandsp app dtmftotext with asterisk 1.4? anybody actually needed it?`
`20:10.24`	`robin_sz`	`[TK]D-Fender, im using the lines gradwell supplied`
`20:10.33`	`robin_sz`	`for extension.conf`
`20:10.33`	`kombi_`	`how do I debug the damn mpg123? can't even find it..`
`20:10.39`	`bkruse`	`tzafrir_laptop: ask coppice :]`
`20:11.27`	`[TK]D-Fender`	`robin_sz: Dial(iax2/whateveryourgradwellpeerishere/1234567)`
`20:11.45`	`robin_sz`	`umm, k`
`20:11.47`	`tzafrir_laptop`	`kombi_, why do you need mpg123?`
`20:12.06`	`tzafrir_laptop`	`for a remote stream or a local file?`
`20:12.07`	`kombi_`	`tzafrir_laptop: trying to make moh work..`
`20:12.19`	`kombi_`	`local file so far`
`20:12.35`	`tzafrir_laptop`	`for a local file use native moh`
`20:13.00`	`*** join/#asterisk logyati (n=suporte@201.29.73.49)`
`20:13.02`	`tzafrir_laptop`	`transcode the file once from mp3 to wav offline`
`20:13.26`	`logyati`	`hello :D thanks to your tips, now i can operate asterisk by my self :D`
`20:13.35`	`kombi_`	`got wav files in /var/lib/asterisk/moh but they don't f** play..`
`20:13.36`	`logyati`	`now i want to make another step`
`20:13.50`	`logyati`	`how does asterisk pass calls to SER?`
`20:13.58`	`tzafrir_laptop`	`file /var/lib/asterisk/moh/*.wav`
`20:14.05`	`tzafrir_laptop`	`or even just the first one`
`20:14.18`	`kombi_`	`how do you mean?`
`20:14.22`	`_VoiceMeUp_COM`	`exten => s,4,GotoIf( $["${NCHAN}" = "Local"]?notfound\|2:5`
`20:14.26`	`_VoiceMeUp_COM`	`not really working`
`20:14.57`	`kombi_`	`oh, ok, wait..`
`20:15.03`	`raidenz`	`2,5`
`20:15.10`	`_VoiceMeUp_COM`	`and ncvhan = Local`
`20:15.17`	`*** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com)`
`20:15.47`	`tzafrir_laptop`	`file /var/lib/asterisk/moh/*.wav \| head -n 2 # what is the output of that`
`20:16.09`	`raidenz`	`Does anyone know if it is possible to send/set variables to pass in a .call file?`
`20:16.23`	`kombi_`	`RIFF little-endian data, Wave audio, Microsoft PCM, 16 bit, mono 8000 ht`
`20:16.26`	`Hmmhesays`	`yes`
`20:16.26`	`kombi_`	`Hz`
`20:16.28`	`Hmmhesays`	`it is`
`20:16.35`	`*** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com)`
`20:16.36`	`Hmmhesays`	`documented on the wiki`
`20:16.40`	`tzafrir_laptop`	`looks OK`
`20:17.38`	`raidenz`	`I am looking at the wiki and I can make calls and go to an extension but I cam't see anything about how to set a variable in a call file.`
`20:17.44`	`kombi_`	`first in the alphabet should be played first I assume, just how does one test on a low level..`
`20:18.07`	`raidenz`	`nevermind :_p`
`20:18.15`	`logyati`	`any tips?`
`20:18.19`	`logyati`	`:(`
`20:18.39`	`[TK]D-Fender`	`logyati: Get off your ass, place a call and see what happens. Wireshark it.`
`20:18.40`	`_VoiceMeUp_COM`	`how about SET:`
`20:18.48`	`_VoiceMeUp_COM`	`like SET: blah=234`
`20:18.54`	`robin_sz`	`[TK]D-Fender, http://www.pastebin.ca/577184`
`20:19.11`	`*** join/#asterisk THX2000 (n=bob@netblock-208-127-94-59.dslextreme.com)`
`20:19.33`	`raidenz`	`I already found it... thanks though VoiceMeUp`
`20:19.44`	`[TK]D-Fender`	`robin_sz: Lets see your phone's entry`
`20:19.53`	`kombi_`	`I set up an extension with just two lines to Answer and MusicOnHold(), CLI says moh starting but then stops again immediately..`
`20:20.12`	`_VoiceMeUp_COM`	`anyway to friggin disable the stop now from being the first damn thing on cli ?`
`20:20.23`	`[TK]D-Fender`	`kombi : Go install asterisk-addons <-------`
`20:20.26`	`_VoiceMeUp_COM`	`or disable that history pos`
`20:20.38`	`logyati`	`[tk]d-fender, man i didnt sit down since i started playing astarisk ^^`
`20:20.42`	`[TK]D-Fender`	`_VoiceMeUp_COM: You have the source, so stop whining ;)`
`20:20.55`	`logyati`	`and what is wireshark?`
`20:21.10`	`brea`	`uh oh`
`20:21.17`	`_VoiceMeUp_COM`	`yeah`
`20:21.27`	`_VoiceMeUp_COM`	`wiereshark ? is a card like paypal not to trust in germany`
`20:21.39`	`_VoiceMeUp_COM`	`will go belly up for fraud as they authen acocutns via callerid`
`20:21.42`	`_VoiceMeUp_COM`	`and sms`
`20:21.51`	`tzafrir_laptop`	`Why asterisk addons? to waste CPU cycles on converting the mp3 to slinear?`
`20:21.53`	`_VoiceMeUp_COM`	`now aint it easy to get an asterisk sms and a did`
`20:21.59`	`robin_sz`	`[TK]D-Fender, my phoens entry in?`
`20:22.18`	`_VoiceMeUp_COM`	`heehehe and also a sniffer ;)`
`20:22.26`	`logyati`	`lol :D`
`20:22.27`	`_VoiceMeUp_COM`	`and i mean wirecard`
`20:22.39`	`_VoiceMeUp_COM`	`sorry for confusion .. crawling abck to my nest`
`20:22.47`	`logyati`	`ok ok, but i think i asked wrong, you missunderstood me`
`20:22.57`	`[TK]D-Fender`	`robin_sz: ****SIP.CONF ***** Omg, you really aren't awake today are you?`
`20:23.24`	`sulan`	`hmm, if I have two channels (one incoming call, the other is an outgoing call), both in an AGI - how can I bridge them?`
`20:23.36`	`robin_sz`	`err, I am very awake, but since I get exactly the same response from the dial cmd on the console ....`
`20:23.50`	`kombi_`	`do you install addons with configure make install or that strange install-sh script?`
`20:24.08`	`tzafrir_laptop`	`kombi_, please pastebin your musiconhold.conf`
`20:24.09`	`tzafrir_laptop`	`also: do you see any error messages when someone tries to be on-hold? if so: paste them there as well`
`20:24.18`	`[TK]D-Fender`	`robin_sz: You should know that when were trying to debug something that you should provide everything related to the call. On BOTH ends. And SIP/IAX2 debug, etc....`
`20:24.26`	`logyati`	`i ment: how do i configure asterisk to pass SIP calls to openser? i wanna call to asterisk from pstn, asterisk answers (this is already configured), then i type a number like 1234 and asterisk call to user 1234 that exists in my SER`
`20:24.28`	`robin_sz`	`[TK]D-Fender, ok, coming up ...`
`20:24.45`	`[TK]D-Fender`	`robin_sz: Asking for it piece-meal is like pulling teeth, and I'm seriously getting out of dentistry in here..`
`20:25.14`	`kombi_`	`tzafrir_laptop: it has two lines only: mode=quietmp3, directory=/var/lib/asterisk/moh`
`20:26.05`	`_VoiceMeUp_COM`	`the firs tdigit after a gotoif from cli is the result 0/1 ?`
`20:26.06`	`_VoiceMeUp_COM`	`0?notfound\|1:s\|5`
`20:26.35`	`[TK]D-Fender`	`heading home, BBIAB`
`20:26.42`	`*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)`
`20:26.45`	`robin_sz`	`[TK]D-Fender, apologies, i did try and make that last post as inclusive as possible, OK, its updated now with a bit of sip.conf`
`20:26.51`	`logyati`	`_voicemeup_com understood?`
`20:27.22`	`tzafrir_laptop`	`kombi, is there [default] or something before that?`
`20:27.29`	`tzafrir_laptop`	`kombi_, what do you see in the asterisk CLI as output of: moh show classes`
`20:27.35`	`kombi_`	`sorry, yes of course`
`20:27.39`	`_VoiceMeUp_COM`	`?`
`20:27.47`	`robin_sz`	`http://www.pastebin.ca/577203`
`20:27.53`	`logyati`	`_voicemeup_please read above`
`20:27.53`	`_VoiceMeUp_COM`	`oh i dont do ser sosrry`
`20:27.54`	`robin_sz`	`oh, tk has gone :(`
`20:28.05`	`logyati`	`_voicemeup_ :(`
`20:28.17`	`logyati`	`_voicemeup_ do you know someone that could help me?`
`20:28.26`	`kombi_`	`tzafrir_laptop: class default, mode quietmp3, directory /var/lib/asterisk/moh, format slin`
`20:29.12`	`tzafrir_laptop`	`do you see anything in 'moh show files' ?`
`20:29.32`	`kombi_`	`NOTHING at all! that is it..!`
`20:29.36`	`kombi_`	`good one!`
`20:30.01`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
`20:30.02`	`tzafrir_laptop`	`maybe the directory is not readable to asterisk or something?`
`20:30.17`	`kombi_`	`very good indeed, let me check that..`
`20:30.34`	`tzafrir_laptop`	`hmmm... mode=files`
`20:30.34`	`QbY_`	`does anyone know of a way to make a cisco 79xx phone re-register..`
`20:30.46`	`QbY_`	`does anyone know of a way to make a cisco 79xx phone re-register more frequently?`
`20:31.26`	`kombi_`	`tzafrir_laptop: would you reckon? I first look for file permissions now`
`20:31.56`	`*** join/#asterisk zeeesh (n=aadilism@202.125.143.70)`
`20:31.59`	`zeeesh`	`hi`
`20:32.18`	`tzafrir_laptop`	`for starters, edit musiconhold.conf`
`20:32.27`	`tzafrir_laptop`	`set: mode=files`
`20:32.43`	`kombi_`	`tzafrir_laptop: all world-readable, vi'ing into the .conf now..`
`20:33.49`	`kombi_`	`trafrir_laptop: ;)`
`20:33.55`	`kombi_`	`you are disco!`
`20:34.01`	`waKKu`	`folks.. which is diffs between linksys pap2 and pap2t ??`
`20:34.26`	`kombi_`	`that only took me freaking day to figure out, oh my god..`
`20:34.59`	`kombi_`	`now on to the next hurdle: stream that to icecast..`
`20:36.03`	`_VoiceMeUp_COM`	`exten => s,5,GotoIf( $[${LEN(${SIP_HEADER(X-BLAH)})} > 0]?found\|1:notfound\|1)`
`20:36.07`	`_VoiceMeUp_COM`	`ok`
`20:36.10`	`_VoiceMeUp_COM`	`LEN is 10`
`20:36.12`	`_VoiceMeUp_COM`	`BUT`
`20:36.20`	`_VoiceMeUp_COM`	`the > 0 doesnt evaluate.. any idea ?`
`20:36.31`	`*** join/#asterisk Maan (n=maan@c-24-218-24-255.hsd1.ma.comcast.net)`
`20:37.46`	`*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)`
`20:38.37`	`waKKu`	`thanks : http://forums.linksys.com/linksys/board/message?board.id=VoIP_Adapters&message.id=455`
`20:39.26`	`zeeesh`	`through asterisk server without using any trunk (like voxbone or verizone) is it possible to make call conferencing ?`
`20:40.21`	`sulan`	`Am I required to have a special dial-plan application to bridge two channels active in AGIs?`
`20:40.34`	`*** join/#asterisk Cresl1n (i=matt@nat/digium/x-d81ac2f4c0d3b52b)`
`20:40.35`	`*** mode/#asterisk [+o Cresl1n] by ChanServ`
`20:40.39`	`Mercestes`	`On astrerisk 1.2.13, I have a group of SIP members in a queue, and one agent. When my agent logs in, and I get a call in that queue, her Agent immediately answers the call, and then it looks like it calls her phone. Is this normal behavior and can I disable that?`
`20:41.46`	`*** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk)`
`20:42.12`	`*** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk)`
`20:42.23`	`*** join/#asterisk kiscokid (n=ron@208.106.33.66)`
`20:42.53`	`*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)`
`20:43.24`	`robin_sz`	`right ... it seems gradwell works with ulaw`
`20:43.30`	`_VoiceMeUp_COM`	`except LEN(${VAR} ) anyother way to test for a var ?`
`20:43.34`	`_VoiceMeUp_COM`	`just want it there`
`20:43.46`	`_VoiceMeUp_COM`	`and : exten => s,5,GotoIf( $[${LEN(${SIP_HEADER(X-BLAH)})} > 0]?found\|1:notfound\|1)`
`20:43.50`	`_VoiceMeUp_COM`	`doestn work`
`20:44.06`	`Qwell[]`	`because " 0" and " 1" are both true`
`20:44.07`	`_VoiceMeUp_COM`	`Noop("${SIP_HEADER(X-BLAH)}") is ok`
`20:44.16`	`_VoiceMeUp_COM`	`tried 1`
`20:44.18`	`_VoiceMeUp_COM`	`also >1`
`20:44.27`	`Corydon76-work`	`$["${VAR}" != ""]`
`20:44.33`	`_VoiceMeUp_COM`	`doh`
`20:44.33`	`_VoiceMeUp_COM`	`thanks`
`20:44.36`	`_VoiceMeUp_COM`	`tired`
`20:44.38`	`[TK]D-Fender`	`robin_sz, And now your dialplan and everything looks saner too :)`
`20:44.41`	`_VoiceMeUp_COM`	`i have htat line like 2 lines down`
`20:44.41`	`Qwell[]`	`all of those would still return " 0" or " 1"`
`20:44.57`	`Qwell[]`	`note the spaces`
`20:44.59`	`robin_sz`	`[TK]D-Fender, yeah, thanks`
`20:45.17`	`*** join/#asterisk Toerkeium (i=oo@201.216.206.221)`
`20:45.21`	`robin_sz`	`[TK]D-Fender, ulaw is not the final answer though, they say they support G729a .. umm`
`20:45.35`	`_VoiceMeUp_COM`	`still 0`
`20:45.44`	`_VoiceMeUp_COM`	`ah`
`20:45.47`	`Qwell[]`	`no, still " 0"`
`20:45.57`	`_VoiceMeUp_COM`	`Thanks`
`20:45.58`	`_VoiceMeUp_COM`	`lol`
`20:46.02`	`robin_sz`	`[TK]D-Fender, is g729a different from g729 in * ? thye use asterisk @ gradwell ...`
`20:46.07`	`_VoiceMeUp_COM`	`must be an obscure reason`
`20:46.21`	`robin_sz`	`the digium paidn thng is G723 isnt it?`
`20:46.46`	`Qwell[]`	`robin_sz: our hardware does G723, but no, there is no G723 codec module`
`20:46.50`	`Qwell[]`	`paid or otherwise`
`20:46.53`	`srd2`	`When I try to make a call from my 7920, I get the following (and just dialtone on phone for like 10 seconds):`
`20:46.54`	`srd2`	`[Jun 19 21:45:48] WARNING[36324]: chan_skinny.c:1255 find_subchannel_by_instance_reference: Could not find subchannel with reference '0' on 'wifi'`
`20:46.57`	`*** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net)`
`20:47.01`	`srd2`	`is there something I'm doing wrong?`
`20:47.15`	`*** join/#asterisk denon (n=denon@tooth.decay.org)`
`20:47.16`	`*** mode/#asterisk [+o denon] by ChanServ`
`20:47.22`	`robin_sz`	`Qwell, ok, so G729a is ?`
`20:48.44`	`_VoiceMeUp_COM`	`dang..`
`20:49.10`	`[TK]D-Fender`	`robin_sz, not sure of its implication`
`20:49.14`	`_VoiceMeUp_COM`	`exten => s,5,GotoIf( $["${LEN(${SIP_HEADER(X-BLAH)})}" != " 0"]?found\|1:notfound\|1)`
`20:49.27`	`_VoiceMeUp_COM`	`so which function returns " 0"`
`20:49.49`	`Strom_M`	`why is there a space before 0?`
`20:49.59`	`_VoiceMeUp_COM`	`coz qwell said so`
`20:50.08`	`Strom_M`	`k`
`20:50.14`	`robin_sz`	`[TK]D-Fender, it just says "g711u and g729a codecs" on their faq`
`20:50.46`	`robin_sz`	`[TK]D-Fender, g711u is a bandwidht hog comapred to G729a right?`
`20:50.58`	`Mercestes`	`http://pastebin.ca/577250 Can someone look at this please? My queue is giving a half-ring then dropping the call`
`20:52.23`	`[TK]D-Fender`	`robin_sz, Yup`
`20:52.33`	`_VoiceMeUp_COM`	`HEADER ? (10)`
`20:52.35`	`*** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket)`
`20:52.36`	`_VoiceMeUp_COM`	`that the len`
`20:52.48`	`_VoiceMeUp_COM`	`so wth does it not work`
`20:54.38`	`Jingles`	`I didn't have any luck with trying to find if anything (headers, callerid, or whatever) had a length of 0.`
`20:54.43`	`[TK]D-Fender`	`Mercestes, First the dialplan you are using for that has an ANSWER in it which is typically a SHOORT-ON-SIGHT-OFFENSE.`
`20:55.04`	`[TK]D-Fender`	`Mercestes, But barring that, please pastebin the actual dialplan being used by that mess`
`20:55.14`	`*** part/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket)`
`20:55.21`	`Jingles`	`so, I did GotoIf($["foo+${CALLERIDNUM}"="foo"]?5:10)`
`20:56.58`	`aptura`	`can asterisk redirect a fax from a fxo to a fxs based on a ivr request?`
`20:56.59`	`[TK]D-Fender`	`Jingles, Get rid of that "+" and that variable is DEPRECATED`
`20:57.20`	`aptura`	`Want to install a fax without buying another line.`
`20:57.21`	`_VoiceMeUp_COM`	`this thing needs is_present(HEADER)`
`20:57.23`	`_VoiceMeUp_COM`	`like ser`
`20:57.26`	`_VoiceMeUp_COM`	`coz its broken`
`20:57.32`	`[TK]D-Fender`	`aptura, It can have the call DIAL that FXS, sure, but the term "redirect" DOES NOT EXIST`
`20:58.02`	`aptura`	`k`
`20:58.37`	`[TK]D-Fender`	`aptura, Dial(Zap/4,30) ; Yippy-kai-yay ^@%#^% a fax!`
`20:59.07`	`Strom_M`	`_VoiceMeUp_COM: if you'd stop complaining, you'd realize you're supposed to take the space out between the opening parenthesis and the dollar sign prefixing the expression...`
`20:59.11`	`aptura`	`Just need to buy another fxs to do this then. I am not sure if asterisk does fax detection and if so what would be the command that detects it.`
`20:59.30`	`_VoiceMeUp_COM`	`ok foun the bug`
`21:00.08`	`[TK]D-Fender`	`aptura, time to head back to * 101 and learn your asterisk Standard Extensions.`
`21:00.21`	`[TK]D-Fender`	`aptura, And thats not even what you asked the first time...`
`21:00.38`	`robin_sz`	`[TK]D-Fender, ok, I got gradwell to come up in G729 mode, but it seems to be trying to go from G729 to slin and then slin to GSM .. and fialing .. I shall paste :)`
`21:00.55`	`[TK]D-Fender`	`robin_sz, if it works one way, to hell with it!`
`21:01.07`	`robin_sz`	`but ulaw is so ... so ... bloaty`
`21:01.59`	`_VoiceMeUp_COM`	`pbx.c: Expression result is '0'`
`21:02.01`	`Strom_M`	`robin_sz: it sounds great though. stop kvetching.`
`21:02.03`	`robin_sz`	`Jun 19 21:59:07 WARNING[26910]: channel.c:2415 set_format: Unable to find a codec translation path from g729 to slin`
`21:02.04`	`Mercestes`	`[TK]D-Fender, That's the rub.....it does nto have an answer in it. :(`
`21:02.26`	`robin_sz`	`Jun 19 21:59:10 WARNING[27858]: app_dial.c:1638 dial_exec_full: Had to drop call because I couldn't make SIP/home-081b7fd8 compatible with IAX2/gradwell-1`
`21:02.30`	`robin_sz`	`sigh ...`
`21:02.47`	`Mercestes`	`[TK]D-Fender, http://pastebin.ca/577282`
`21:03.20`	`Mercestes`	`[TK]D-Fender, the IVR that points to it has an answer to get to that extension, but, that extension does not have an Answer()`
`21:03.57`	`Mercestes`	`that's what has me confused`
`21:05.38`	`robin_sz`	`Strom_M, yeah, but we only have a single adsl ... getting three ulaw channels outgoin on it?`
`21:06.07`	`robin_sz`	`it will be fine until someone browses the pesky internet`
`21:07.22`	`[TK]D-Fender`	`#`
`21:07.23`	`[TK]D-Fender`	`Called Agent/4913`
`21:07.23`	`[TK]D-Fender`	`#`
`21:07.23`	`[TK]D-Fender`	`<PROTECTED>`
`21:07.29`	`[TK]D-Fender`	`^^^^ excuse me?`
`21:07.34`	`Mercestes`	`[TK]D-Fender, Ditto`
`21:07.40`	`Mercestes`	`That is the dialplan you are looking at for 4050`
`21:07.48`	`[TK]D-Fender`	`Mercestes, Sure looks like it does`
`21:07.56`	`Mercestes`	`I agree, it does look like that.`
`21:08.02`	`Mercestes`	`But I checked twice and she said she did not pick up the phone.`
`21:08.18`	`[TK]D-Fender`	`Mercestes, ASTERISK andswered the call!`
`21:08.26`	`Mercestes`	`I agree.`
`21:08.35`	`Mercestes`	`and then asterisk calls 4913`
`21:08.35`	`[TK]D-Fender`	`Mercestes, Welcome to Local Channel Sillyness, population YOU!`
`21:08.41`	`*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)`
`21:08.47`	`[TK]D-Fender`	`Mercestes, that answers the Queue for all it cares`
`21:08.52`	`Mercestes`	`Right.`
`21:08.56`	`Mercestes`	`but ...I didn't do that I don't think.`
`21:09.07`	`[TK]D-Fender`	`Mercestes, do NOT dial extens with an explicit "Answer" of any kind!`
`21:09.16`	`Mercestes`	`I didn't.`
`21:09.24`	`Mercestes`	`"4050" does not exist as a peer, it's jsut an extension`
`21:09.29`	`[TK]D-Fender`	`Mercestes, Blame whomever you will, but thats why.`
`21:09.34`	`_VoiceMeUp_COM`	`hmm`
`21:09.35`	`_VoiceMeUp_COM`	`wow`
`21:09.36`	`_VoiceMeUp_COM`	`lame`
`21:09.42`	`*** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com)`
`21:09.44`	`Mercestes`	`....there is not an answer there...you HAVE THE CODE`
`21:09.52`	`[TK]D-Fender`	`-- Executing Answer("Local/4913@houston-616b,2", "") in new stack <- this means YOU = FUBAR'd`
`21:09.58`	`Mercestes`	`WTF`
`21:09.59`	`_VoiceMeUp_COM`	`so its adding the space in GotoIf( $[`
`21:10.00`	`_VoiceMeUp_COM`	`lol`
`21:10.15`	`Mercestes`	`can you READ?`
`21:10.27`	`[TK]D-Fender`	`Mercestes, sigh`
`21:10.33`	`Mercestes`	`what part of QUEUE(SUPPORT) are you not getting?`
`21:10.34`	`Mercestes`	`I gave you the code`
`21:10.40`	`Mercestes`	`I'm not dialing any extensions with an explicit answer`
`21:10.57`	`Mercestes`	`and I sure as hell don't do anything with local/`
`21:10.59`	`[TK]D-Fender`	`Mercestes, your queue is initiating a Local channel to call that agent. the Local channel ANSWERED. YOU are not getting it.`
`21:11.11`	`Mercestes`	`WTF is it doing that then??`
`21:11.27`	`[TK]D-Fender`	`Mercestes, The fact that said channel would EVERNTUALLY try and dial a SIP device or pick its nose or whatever is IRRELEVENT`
`21:11.53`	`Mercestes`	`wait..wait...`
`21:11.59`	`[TK]D-Fender`	`Mercestes, Chan_local doesn't just ACCEPT the call, it ANSWERS IT.`
`21:12.05`	`Mercestes`	`....does it dial the "4913" extension?`
`21:12.18`	`Mercestes`	`because I do have a exten => 4913,1,Answer`
`21:12.20`	`[TK]D-Fender`	`Mercestes, YES!`
`21:12.22`	`Mercestes`	`but, that's off in another context.`
`21:12.23`	`Mercestes`	`oh...`
`21:12.28`	`Mercestes`	`that explains where it gets answre.`
`21:12.41`	`Mercestes`	`...`
`21:12.52`	`Mercestes`	`so...the queue tries to dial local/4913....`
`21:13.09`	`_VoiceMeUp_COM`	`yeah`
`21:13.17`	`[TK]D-Fender`	`Mercestes, Yank that stupid ANSWER out and you'll be fine! (Assuming you don't something else retarded like... letting it fall to *VM`
`21:13.18`	`_VoiceMeUp_COM`	`had to find away around local and queues`
`21:13.18`	`_VoiceMeUp_COM`	`also`
`21:13.48`	`[TK]D-Fender`	`_VoiceMeUp_COM, Way around? No, that is a blatantly and easily avoidable dialplan snafu.`
`21:14.05`	`Mercestes`	`[TK]D-Fender, I don't see how my answer under exten => 4913 is causing this though.`
`21:14.17`	`Mercestes`	`asterisk shouldn't be dialing the extension 4913, it should be contacting the peer 4913`
`21:15.03`	`Mercestes`	`exten => 4913 could dial my ass for all asterisk knows, it shouldn't even be looking at that.`
`21:15.04`	`[TK]D-Fender`	`Mercestes, that is an AGENT. It dials through the dialplan. that is a channel. for all * knows you're gonn have it pick up and play some stupid recording and hang up. a channel is a channel is a channel`
`21:15.35`	`Mercestes`	`oh...`
`21:15.52`	`[TK]D-Fender`	`Mercestes, If you do "Answer", then the channel has answered. If You just dial that sip device FIRST THEN IF THEY DON'T ANSWER THE CALL WILL FALL THROUGH AS IT SHOULD`
`21:16.14`	`Mercestes`	`indeed.`
`21:16.24`	`[TK]D-Fender`	`Mercestes, OOPS, gratuitous caps-lock ;)`
`21:16.30`	`Mercestes`	`;) It's because you love me.`
`21:16.45`	`*** join/#asterisk Jabroni (n=Jabroni@red-corp-200.76.249.142.telnor.net)`
`21:16.48`	`Mercestes`	`..and I thought you just held down shift for a bit.. >.>`
`21:17.30`	`kiscokid`	`I have a question about what kind of channel cards I need to buy to replace my old Norstar Nortel PBX. According to my Vcom phone bill I have 5 "measured business lines" with "hunting" and one "analog DID trunk" plus a separate fax line.`
`21:17.31`	`[TK]D-Fender`	`No small irony there ;)`
`21:17.48`	`Mercestes`	`hehe`
`21:18.10`	`[TK]D-Fender`	`kiscokid, Sangoma A200d w/ 3 FXO Modules (6 ports)`
`21:18.47`	`[TK]D-Fender`	`kiscokid, then drop-kick that old POS into a dumpster`
`21:18.52`	`[TK]D-Fender`	`:D`
`21:18.57`	`kiscokid`	`Fender: does the analog DID trunk appear as a separate port ?`
`21:19.50`	`[TK]D-Fender`	`kiscokid, forget every tech term you just used. You have 5 effectively boring analog lines going into your PB now. the card I mentioned will let you take them into * and you can process your calls any which way you please.`
`21:20.24`	`kiscokid`	`so * will know the extension number being called?`
`21:20.52`	`[TK]D-Fender`	`kiscokid, Does your telco send DTMF signalling to fake-out DID's over analog?`
`21:21.14`	`kiscokid`	`don't know, how can I find out?`
`21:21.22`	`[TK]D-Fender`	`kiscokid, I have heard of this before, but only once.`
`21:21.39`	`[TK]D-Fender`	`kiscokid, Ask them or plug an analog phone in parallel to a line and test`
`21:22.03`	`*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)`
`21:22.57`	`*** join/#asterisk nexilus (i=nexilus@c-d87d71d5.011-143-6c756c1.cust.bredbandsbolaget.se)`
`21:23.14`	`nexilus`	`where do i set the "global EID" ?`
`21:23.29`	`kiscokid`	`fender what is the other option to DTMF signalling to fake out DIDs?`
`21:27.58`	`[TK]D-Fender`	`kiscokid, nothing I'm aware of on analog...`
`21:32.03`	`*** join/#asterisk THX2000 (n=bob@netblock-208-127-94-59.dslextreme.com)`
`21:32.54`	`THX2000`	`Anyone know how i might get the hudlite client to initiate calls to my aastra phones w/ an auto-answer header?`
`21:33.22`	`sulan`	`Is AgentLogin removed in Asterisk 1.4?`
`21:34.59`	`*** join/#asterisk alex1234 (n=lolz@adsl-71-156-37-180.dsl.irvnca.sbcglobal.net)`
`21:35.52`	`*** part/#asterisk alex1234 (n=lolz@adsl-71-156-37-180.dsl.irvnca.sbcglobal.net)`
`21:40.54`	`zeeesh`	`using 1 DID .. exten => _X.,1,answer exten => _X.,2,WaitExten(15) exten => 5557,3,Meetme,54321 ... dialing from 2 users ... failed ... anyother way ..?`
`21:42.02`	`Strom_M`	`because _X. isn't the same extension as 5557`
`21:42.35`	`Qwell[]`	`that actually should work`
`21:42.39`	`Strom_M`	`or are you dialing after the waitexten`
`21:42.52`	`Qwell[]`	`I mean, the exten switching like that. Not the content of the extensions :p`
`21:42.56`	`Strom_M`	`well, no, 5557 has no priority 1`
`21:42.59`	`Strom_M`	`so it's not going to work`
`21:43.02`	`Qwell[]`	`shouldn't matter`
`21:43.13`	`Qwell[]`	`it evaluates exten on each priority`
`21:43.36`	`Strom_M`	`....? AFAIK you can't have an extension start with a priority other than 1`
`21:43.42`	`Strom_M`	`but I could be wrong`
`21:43.48`	`Qwell[]`	`You can, if there's something else letting it get past 1`
`21:44.02`	`Strom_M`	`doesn't work with "n" anyway`
`21:44.03`	`xkev`	`the regex will match`
`21:44.05`	`xkev`	`I do that a lot`
`21:44.08`	`Qwell[]`	`it'd be kinda like a goto to an exten without a priority 1, as long as you specify a priority with the goto`
`21:44.17`	`Qwell[]`	`yeah, it wouldn't work with n, but if it was 3, then n, it might`
`21:44.30`	`xkev`	`exten => _89XX,1,Answer;`
`21:44.30`	`xkev`	`exten => _89XX,2,Wait(1);`
`21:44.30`	`xkev`	`exten => 8900,3,Read(recnum,xm/beeps/amfmbeep,3);`
`21:44.35`	`xkev`	`exten => 8969,3,Goto(conferences,create,1);`
`21:44.38`	`Qwell[]`	`yeah, same deal`
`21:44.40`	`xkev`	`exten => 8990,3,Goto(findme_init,139,1);`
`21:44.42`	`xkev`	`..etc`
`21:44.44`	`zeeesh`	`so i m dialing my DID from my cell phone ... so i think 1st should be the answer .. so thats y 2nd is waitexten .`
`21:45.13`	`xkev`	`all my feature-code contexts are built that way, each context is like 89xx or 88xx etc`
`21:45.42`	`xkev`	`strom is right about the 'n' though`
`21:45.53`	`*** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net)`
`21:46.02`	`xkev`	`exten => 8900,3,Read(recnum,xm/beeps/amfmbeep,3);`
`21:46.03`	`xkev`	`exten => 8900,n,Record(xm/recorded/${recnum}.wav);`
`21:46.03`	`xkev`	`exten => 8900,n,Playback(xm/recorded/${recnum});`
`21:46.07`	`Qwell[]`	`that should work`
`21:46.08`	`xkev`	`^works`
`21:46.20`	`sulan`	`AgentLogin, I can't find it in my list of applications - I'm running Asterisk 1.4. Any ideas?`
`21:46.25`	`xkev`	`but _89XX,2,Blah then 8900,n,Blah would not`
`21:46.28`	`Qwell[]`	`right`
`21:46.46`	`Qwell[]`	`sulan: is chan_agent.so loaded?`
`21:46.53`	`Toerkeium`	`[TK]D-Fender: are you over there?`
`21:46.58`	`xkev`	`(also you should burn chan_agent and run far)`
`21:47.09`	`xkev`	`stick with app_queue for maximum happiness`
`21:47.16`	`[TK]D-Fender`	`Toerkeium, no, YOU'RE over "there", I'm here!`
`21:47.51`	`xkev`	`login with AddQueueMember(queuename\|${CHANNEL}) :)`
`21:47.52`	`sulan`	`Qwell[]: thanks, I removed noload => chan_agent and reloaded, it seems I need to restart for it to work.`
`21:48.06`	`Toerkeium`	`[TK]D-Fender: can I message you privately?`
`21:48.13`	`Qwell[]`	`sulan: reload doesn't load modules that weren't already loaded. you could've just done "module load chan_agent.so"`
`21:48.19`	`[TK]D-Fender`	`Toerkeium, if need be, sure`
`21:48.25`	`Qwell[]`	`erm, that weren't previously loaded`
`21:48.40`	`sulan`	`xkev: ah, that might work! It doesn't work just to AddQueueMember(queuename) because it adds the interface implicitly.`
`21:49.04`	`Qwell[]`	`There is even an alias option to AddQueueMember`
`21:49.16`	`Qwell[]`	`so you can have several devices that are "called" the same thing in the logs`
`21:51.22`	`sulan`	`interesting, trying my system now ;)`
`21:51.30`	`zeeesh`	`<Qwell[]>: my users dial access number from mobile, after getting beep they dial their destination number .. now i want they dial the same access number and then dial any specifice extension like 5557 for call conferencing so how to possible?`
`21:53.26`	`sulan`	`[Jun 19 23:50:16] NOTICE[790]: app_queue.c:3220 aqm_exec: Added interface 'Local/00701234567@private-c612,1' to queue 'test'`
`21:53.29`	`sulan`	`later i get:`
`21:53.42`	`Toerkeium`	`does anyone is able to do a asterisk+vicidial+sugarcrm job professionaly?`
`21:53.43`	`sulan`	`[Jun 19 23:51:37] NOTICE[1319]: chan_local.c:566 local_alloc: No such extension/context 00701234567@private-c612,1 creating local channel`
`21:54.30`	`*** join/#asterisk bkruse (i=bkruse@nat/digium/x-e8c300aa520b2fce)`
`21:55.55`	`[TK]D-Fender`	`sulan, go check your dialplan. It's not lying`
`21:56.28`	`*** part/#asterisk QbY_ (n=Kelvin@66.236.241.67.ptr.us.xo.net)`
`21:56.51`	`sulan`	`well, my goal is to create a system like this:`
`21:57.15`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`21:58.59`	`sulan`	`An incoming calls gets to an AGI that places it into a queue. The AGI also Originates a new call via Asterisk Manager, where the remote call, when answered, gets connected to another AGI that makes the agent accept/decline the call. On accept, the two calls get bridged - somehow.`
`21:59.08`	`harlequin516`	`Okay I have now tested all cases that I know. Still I cannot get the native bridging to stop being called. Where can I find exactly the criteria for native bridging? Do I have to go to source?`
`21:59.32`	`[TK]D-Fender`	`sulan, this has nothing to do with AGI. Just use the local channel and the macro efature of Dial.`
`21:59.50`	`*** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`22:00.11`	`*** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com)`
`22:00.21`	`[TK]D-Fender`	`harlequin516, pastebin the CLI output of your call in its entirety, and what exactly is it you want to have happen instead?`
`22:00.36`	`harlequin516`	`sulan: Yeah if you are new to asterisk, it is frustrating to want to do everything in agi, later to find that most things can be done without it.`
`22:01.01`	`harlequin516`	`[TK]D-Fender: Alright be back in 2`
`22:01.45`	`*** part/#asterisk guigouz (n=guigouz@unaffiliated/guigouz)`
`22:02.14`	`sulan`	`[TK]D-Fender: I'm not following you...`
`22:02.58`	`[TK]D-Fender`	`sulan, go read "show application dial" for how to have the other side accept before bridging.`
`22:03.27`	`*** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net)`
`22:04.12`	`irule`	`what is fullname user for in queues.conf?`
`22:04.30`	`irule`	`what is fullname used for in queues.conf?`
`22:04.49`	`Qwell[]`	`irule: That's the alias I talked about above, I believe`
`22:05.06`	`harlequin516`	`Okay how do I disbale native bridging? Here's my CLI: http://pastebin.ca/577414`
`22:05.14`	`Qwell[]`	`It lets you call multiple devices the same thing in logs`
`22:05.29`	`Juggie`	`Qwell, dropping an index on a table w/ 220million rows, and re-adding it, wish me luck :)`
`22:05.56`	`[TK]D-Fender`	`harlequin516, See this? Executing Dial("SIP/sham-081a8be8", "SIP/16232294754@sip.broadvoice.com\|60\|W\|")`
`22:06.08`	`harlequin516`	`yeah?`
`22:06.18`	`[TK]D-Fender`	`harlequin516, Your in-line recording ability FORCES * to keep itself in the audio path incase it needs to record`
`22:06.30`	`sulan`	`[TK]D-Fender: aha!`
`22:06.50`	`harlequin516`	`I only added the W to try and force ASterisk to not use native bridging`
`22:06.56`	`[TK]D-Fender`	`sulan, Good, now beat yourself over the head with it for a few hours till it works perfect then come back.`
`22:07.20`	`harlequin516`	`I didn't really want the W, but I don't think it would interfere.`
`22:07.38`	`harlequin516`	`It didn't work without it either`
`22:07.45`	`[TK]D-Fender`	`harlequin516, That is backwards. One of the 2 ends it undoubtably behind NAT and wouldn't survive a re-invite ANYWAYS`
`22:08.10`	`[TK]D-Fender`	`harlequin516, but alas I have to go for a few hours.`
`22:08.13`	`[TK]D-Fender`	`BBIAB all`
`22:09.46`	`harlequin516`	`ASterisk should see both channels plainly. One is a NAT address but in the same ethernet and address space as asterisk, the other is public IP.`
`22:10.29`	`harlequin516`	`The funny thing is that it works when I call from one device to the other, but not in reverse.`
`22:12.03`	`*** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net)`
`22:24.00`	`*** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net)`
`22:27.04`	`*** join/#asterisk _DAW (n=chatzill@adsl-222-41-108.msy.bellsouth.net)`
`22:29.09`	`*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)`
`22:35.18`	*** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com)
`22:35.29`	`robin_sz`	`meep?`
`22:38.39`	`*** join/#asterisk Strom_M (n=strom@60-241-200-26.static.tpgi.com.au)`
`22:52.20`	`harlequin516`	`Okay I have looked into the asterisk source to find out what is going on. I have decided that I am not qualified as a C programmer.`
`22:52.33`	`sulan`	`[TK]D-Fender: thanks, it's coming together now!`
`22:52.54`	`harlequin516`	`What gobbledeygook! Java debugging is sooo much easier.`
`22:53.13`	`harlequin516`	`Let's port the whole mess to java.`
`22:53.21`	`JT`	`it's a pity java can't do much useful`
`22:53.26`	`JT`	`java is a horrible language`
`22:53.47`	`JT`	`a perfect example of a language designed by committie instead of by real programmers`
`22:57.50`	`*** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net)`
`23:01.54`	`harlequin516`	`Java is awesome and easy`
`23:03.14`	`JT`	`someone's been smoking too much uni crackpipe`
`23:03.44`	`JT`	`java has its uses, they're just fairly minimal`
`23:03.51`	`JT`	`especially if performance is a requirement`
`23:04.01`	`Jingles`	`java is a great way to teach oop.`
`23:04.14`	`JT`	`yes, it's used for teaching a lot`
`23:04.18`	`bkruse`	`EWW`
`23:04.25`	`bkruse`	`javascript ;]`
`23:04.26`	`JT`	`it's easy for educational institutions to teach`
`23:04.27`	`bkruse`	`:P`
`23:04.32`	`bkruse`	`this is true.`
`23:04.33`	`JT`	`that doesn't make it a good language to use`
`23:07.20`	`*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)`
`23:10.18`	`harlequin516`	`I'm no expert about performance or anything like that, but Java syntax is clear and easy to follow.`
`23:10.39`	`Aces1Up`	`what you guys think of ruby?`
`23:11.01`	`harlequin516`	`If something needed to be built for high performance I would still use Java where possible and use native methods when prudent.`
`23:11.33`	`Qwell[]`	`java for high performance? O.o`
`23:11.43`	`*** join/#asterisk mightnare (n=mike@s230165.ppp.asahi-net.or.jp)`
`23:12.20`	`harlequin516`	`Most of code is glue anyways`
`23:12.30`	`harlequin516`	`The hard meat is liek 15%`
`23:14.01`	`*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)`
`23:16.37`	`mihinomenest`	`if I needed something for high performance, I'd let JITC decide for me.`
`23:16.51`	`JT`	`harlequin516: i still think you're talking crack`
`23:17.01`	`Qwell[]`	`talking or smoking`
`23:17.36`	`*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)`
`23:17.43`	`neverblue`	`how can I reset the pass on my Grandstream Budget Tone-100 phone?`
`23:18.56`	`*** join/#asterisk lmoreira (n=xxx@201009076233.user.veloxzone.com.br)`
`23:19.18`	`mihinomenest`	`supposedly, sun's got an OS on one of their sparcservers that's as close to a java environment without the encumberment of a virtual machine.`
`23:20.21`	`*** join/#asterisk axisys (n=axisys@ip68-98-146-161.dc.dc.cox.net)`
`23:20.51`	`lmoreira`	`Hi, getting chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1`
`23:21.14`	`lmoreira`	`Random call dropping`
`23:21.28`	`JT`	`lmoreira: run zttest`
`23:21.35`	`lmoreira`	`Any clue how fix it?`
`23:21.45`	`JT`	`find out what the lowest score is, and what it normally scores`
`23:23.14`	`lmoreira`	`Running zttest now.`
`23:24.15`	`lmoreira`	`max 100% min 99.987793%`
`23:24.27`	`*** join/#asterisk Mavvie (n=edwin@ppp121-44-63-246.lns2.syd6.internode.on.net)`
`23:24.42`	`lmoreira`	`results: Best: 100.000000 -- Worst: 98.034668 -- Average: 99.955219`
`23:24.54`	`bkruse`	`thats not bad`
`23:25.00`	`JT`	`that's awful`
`23:25.03`	`JT`	`98%`
`23:25.16`	`bkruse`	`oh, thought he said 99.98`
`23:25.39`	`JT`	`he did, no idea why he contradicted himself`
`23:25.40`	`lmoreira`	`Setup TE110p + P4 + RAM2GB`
`23:25.43`	`bkruse`	`didnt read the second....what else you have on the pci bus?`
`23:26.20`	`lmoreira`	`Interrupts>> 11: 199630826 XT-PIC libata, eth0, wcte11xp, wcfxo`
`23:26.35`	`JT`	`lmoreira: you have a zap timing issue, try disabling unneeded hardware, not loading drivers for uneeded hardware, ensuring there is no interrupt sharing..`
`23:26.38`	`JT`	`wtf`
`23:26.46`	`JT`	`irq 11 shared with 4 devices`
`23:26.47`	`bkruse`	`haha`
`23:26.52`	`JT`	`that is completely unnacceptable`
`23:26.56`	`bkruse`	`hahaha`
`23:27.02`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`23:27.38`	`lmoreira`	`So, how can I change the IRQ?`
`23:28.10`	`JT`	`try changing pci slots`
`23:28.11`	`*** join/#asterisk mmartinn (n=martin@adsl-065-005-200-225.sip.gnv.bellsouth.net)`
`23:28.19`	`JT`	`how many digium cards do you have?`
`23:28.35`	`mihinomenest`	`your motherboard manufacture should have some info on which slots share IRQs.`
`23:28.42`	`lmoreira`	`2 casds: 1 TE110p and 1 X100p`
`23:28.52`	`*** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335)`
`23:29.19`	`Jon335`	`Is there a place to get a VoIP DECT phone in the US?`
`23:29.19`	`bkruse`	`micro-atx mobo?`
`23:29.37`	`lmoreira`	`yes`
`23:29.58`	`bkruse`	`with onlu like 2 pci slots right?`
`23:30.15`	`lmoreira`	`yes`
`23:30.16`	`*** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net)`
`23:30.18`	`bkruse`	`eww`
`23:30.29`	`*** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net)`
`23:30.33`	`lmoreira`	`well, actually, I have more PCI slots`
`23:30.40`	`JT`	`you're not really meant to run ast on pos desktop boards`
`23:30.50`	`JT`	`well, go right ahead and swap te cards to other sltos`
`23:30.52`	`bkruse`	`^^^true`
`23:30.53`	`bkruse`	`try em`
`23:31.38`	`lmoreira`	`I will, thanks.`
`23:32.22`	`harlequin516`	`Hmm can't asterisk connect a gsm channel to a ulaw?`
`23:32.26`	`lmoreira`	`So, one IRQ for each board, right?`
`23:32.50`	`lmoreira`	`<PROTECTED>`
`23:32.51`	`lmoreira`	`<PROTECTED>`
`23:32.51`	`lmoreira`	`<PROTECTED>`
`23:32.51`	`lmoreira`	`<PROTECTED>`
`23:32.51`	`lmoreira`	`<PROTECTED>`
`23:32.51`	`lmoreira`	`<PROTECTED>`
`23:32.53`	`lmoreira`	`<PROTECTED>`
`23:32.55`	`lmoreira`	`<PROTECTED>`
`23:33.10`	`JT`	`lmoreira: stop`
`23:33.21`	`JT`	`lmoreira: don't every paste that much to channel`
`23:33.26`	`JT`	`s/every/ever/`
`23:33.28`	`lmoreira`	`ok, sorry`
`23:33.52`	`JT`	`~pb`
`23:33.52`	`jbot`	`methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org`
`23:34.16`	`JT`	`lmoreira: digium cards must not share irqs with anything`
`23:34.32`	`JT`	`let along hard drive and ethernet, and each other, that's about the worst possible`
`23:35.46`	`harlequin516`	`I thought asterisk could bridge any two supported protocols`
`23:35.56`	`JT`	`harlequin516: ?`
`23:36.22`	`lmoreira`	`Ok, thank you. The paste is on http://pastebin.ca/577569`
`23:36.45`	`mrdigital-work`	`http://paste.debian.net/30925 can someone take a look at that?`
`23:36.55`	`harlequin516`	`How do I get asterisk to convert gsm to ulaw?`
`23:37.01`	`JT`	`lmoreira: yes i've already seen the irq 11 line twice before... it still isn't fixed`
`23:37.07`	`JT`	`harlequin516: it's automatic`
`23:38.02`	`lmoreira`	`JT, I cannot work on the server now. But I'll tomorrow. See you tomorrow. bye`
`23:38.23`	`JT`	`ok`
`23:38.46`	`harlequin516`	`I get : *CLI> Jun 19 16:38:25 NOTICE[22383]: chan_sip.c:3770 process_sdp: No compatible codecs!`
`23:39.19`	`JT`	`what's trying to negotiate for what codec?`
`23:39.30`	`mmartinn`	`hey folks... what's up with zttool and TE405p's and the lack of information when using dms100 signalling? zttool basically shows no changing information about each channel like it did when I tried with e&m signalling?`
`23:39.58`	`russellb`	`because e&m is an analog signalling protocol`
`23:40.01`	`russellb`	`and zaptel itself knows about it`
`23:40.11`	`russellb`	`dms100 is a higher layer protocol, and is handled up inside of asterisk.`
`23:40.25`	`mmartinn`	`russellb: hey Russell, so there's no way to know about channels being used that isn't massively-fast-scrolling debug?`
`23:41.12`	`harlequin516`	`JT: Oh okay I am frazzled... Too much going on i figured it out. I allow=gsm, but my sipura don't do gsm.`
`23:41.19`	`russellb`	`it may be available from CLI commands in asterisk, i don't remember off of the top of my head`
`23:42.03`	`russellb`	`mmartinn: pri show span X maybe?`
`23:42.18`	`mmartinn`	`russellb: there's a few, like zap show channels or "show channels" but they aren't really focused on channel status so much as what is currently bridged... I'm interested in a channel-oriented view, not a call-oriented one`
`23:42.36`	`russellb`	`is there a zap show channel x?`
`23:42.44`	`mrdigital-work`	`anyone?`
`23:42.55`	`russellb`	`looks like there is ..`
`23:43.04`	`mmartinn`	`russellb: There is, but you can't sit and watch all of your channels with it`
`23:43.09`	`mmartinn`	`russellb: Like zttool`
`23:43.22`	`russellb`	`ah, right. then in that case, I do not know of such a tool.`
`23:43.26`	`mmartinn`	`russellb: I suppose you could poll a couple hundred channels`
`23:43.33`	`russellb`	`heh, yeah, you could ...`
`23:43.46`	`mmartinn`	`russellb: hmm... I meant to look in chan_zap for manager events I could use`
`23:43.52`	`russellb`	`you could write an app using the manager interface, yeah`
`23:43.56`	`mmartinn`	`russellb: I'm thinking that might be those efficient way`
`23:44.01`	`russellb`	`yeah`
`23:44.11`	`mmartinn`	`russellb: We're trying to gauge pri usage during peak hours`
`23:44.16`	`russellb`	`gotcha.`
`23:44.24`	`russellb`	`CDRs?`
`23:44.44`	`russellb`	`for that, you don't need to know which channel it is specifically`
`23:44.46`	`mmartinn`	`russellb: I suppose given enough of them, I could look at what Zap channels were used when`
`23:44.51`	`russellb`	`just how many channels are used on the PRI as a group`
`23:45.03`	`russellb`	`right`
`23:45.16`	`mmartinn`	`russellb: We do use a TAPI driver that generates its own CDRs though, at least twice the Zap/ related ones`
`23:45.19`	`russellb`	`that would probably be an easier thing to analyze`
`23:45.39`	`mmartinn`	`russellb: That's a very good idea... harder for realtime, but easily enough information to look back`
`23:45.40`	`russellb`	`well, you can filter for the ones you care about :)`
`23:45.51`	`russellb`	`yeah, agreed`
`23:46.03`	`russellb`	`i mean, if you had the CDRs go into a database, you could still poll the db for new entires`
`23:46.07`	`russellb`	`or have some kind of trigger, i don't know`
`23:46.25`	`russellb`	`and with CDRs, you will only get them at the end`
`23:46.32`	`russellb`	`so you still wouldn't have in progress information ...`
`23:46.35`	`mmartinn`	`I wanna say chan_zap has some events, but I just walked in the door`
`23:46.38`	`russellb`	`depends what you're looking for`
`23:46.46`	`mmartinn`	`manager events, that is`
`23:46.58`	`russellb`	`i just looked, not for what you're looking for`
`23:47.07`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`23:47.08`	`russellb`	`not in chan_zap directly, anyway.`
`23:47.09`	`mmartinn`	`d'oh`
`23:47.21`	`russellb`	`however, the asterisk core will generate channel state events`
`23:47.39`	`mmartinn`	`Maybe I'm thinking of eoj's moremanager`
`23:47.40`	`russellb`	`which will probably get you what you want`
`23:47.43`	`mmartinn`	`err oej`
`23:47.44`	`russellb`	`perhaps`
`23:48.04`	`mmartinn`	`russellb: well, I know i've thanked you before on IRC, but thanks again for more useful ideas and tips ;)`
`23:48.13`	`russellb`	`mmartinn: you are very welcome :)`
`23:48.27`	`mmartinn`	`hmm... mostly alars`
`23:48.29`	`mmartinn`	`err alarms`
`23:48.32`	`russellb`	`yeah ..`
`23:48.39`	`russellb`	`and some other DNDstate thing ...`
`23:49.36`	`mmartinn`	`I swear there's a bridge event somewhere...`
`23:49.42`	`russellb`	`that's in the core`
`23:49.59`	`russellb`	`main/channel.c`
`23:50.13`	`mmartinn`	`Ah there's some state events`
`23:50.21`	`mmartinn`	`That might work, if I only pay attention to zap ones`
`23:50.28`	`russellb`	`yeah`
`23:51.04`	`mmartinn`	`hmmm... that should be more than enough to have a realtime status of Zap channels`
`23:52.55`	`russellb`	`i would think so`
`23:52.59`	`russellb`	`anyway, i'm off, good luck`
`23:53.13`	`mmartinn`	`thanks again!`
`23:57.01`	`*** part/#asterisk kiscokid (n=ron@208.106.33.66)`
`23:57.38`	`*** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com)`

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