IRC log for #asterisk on 20071025

`00:00.09`	`Qwell`	`if only it were that easy, that somebody could just pick up somebody else's dialplan and use it`
`00:00.40`	`Alowishus`	`Alan_Hicks: ignore that this is from AAH, the writeup walks you through it a bit - http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+7#7224HowtouseProvisioningCentralBootServe`
`00:00.56`	`Alan_Hicks`	`Who is AAH?`
`00:01.04`	`*** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com)`
`00:01.05`	`Alan_Hicks`	`Nevermind.`
`00:01.56`	`Alan_Hicks`	`In other words, I only need to list things that I change from the phone's default dial-plan.`
`00:02.09`	`Alowishus`	`phone's default config yes`
`00:02.15`	`Alowishus`	`Polycom has a guide on the settings`
`00:02.19`	`Alowishus`	`but it's some icky XML editing`
`00:02.31`	`Alan_Hicks`	`Yeah, I read a bit of it but it was really icky like you said.`
`00:02.51`	`Alan_Hicks`	`I was just trying to figure out why I could dial say 511, but not 100.`
`00:03.05`	`JT`	`irule: no, i don't see how the help in here could possibly be better quality with some generic jack of all trades dialplan`
`00:03.06`	`Alan_Hicks`	`And ManxPower figured it was something to do with the dialplan in the phone.`
`00:03.34`	`irule`	`JT no offense? duhhh`
`00:03.58`	`JT`	`irule: what?`
`00:05.24`	`Ritzerisk`	`so in the Sip config do i have to use the providers config if im using the 3102`
`00:05.30`	`irule`	`JT not just you, irc users in general want everyone to make smart questions and RTFM all the time, a link toa step by step instruction on that speciffic part is a lot better than whining about stupid questions`
`00:05.44`	`*** join/#asterisk Defraz (n=t0tal@69-20-180-238.static.ida.net)`
`00:05.58`	`JT`	`irule: there is a step by step insruction..`
`00:06.00`	`JT`	`~thebook`
`00:06.01`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf`
`00:06.06`	`Defraz`	`1`
`00:06.31`	`Alan_Hicks`	`Alowishus: I figure the phone's config probably has its own rules set up for most common two-digit "beginnings" (is there a better for the first couple of digits in a phone number?).`
`00:06.56`	`Alan_Hicks`	`Example: When I tried to dial "100", the phone gave me a second dial-tone after puching in "10".`
`00:06.59`	`irule`	`but none of you will EVER give step by step instructions, that is documenting, there is one method in voip-info, the book is another, a better basic dialplan is just one more way of documenting`
`00:07.17`	`Alan_Hicks`	`irule: Step-by-step instructions != documentation!`
`00:07.27`	`irule`	`ok, name it`
`00:07.44`	`Alan_Hicks`	`This coming from the guy that authored/edited the Slackware handbook.`
`00:07.50`	`JT`	`irule: there are already sample dialplans/configs shipped with asterisk`
`00:08.09`	`Alan_Hicks`	`Step-by-step isn't documentation, it's brain-dead hand-holding.`
`00:08.09`	`JT`	`irule: that IS a "basic dialplan"`
`00:08.12`	`irule`	`imagine, cp http/base-dialplan, and add those 3 other partds, and ready :P`
`00:08.24`	`Alan_Hicks`	`You should just pay some one to do it for you if you don't want to learn about it.`
`00:08.41`	`JT`	`Alan_Hicks: high five`
`00:08.55`	`Alan_Hicks`	`After all, do you expect there to be step-by-step instructions to fix your car when you don't even know what the hell the problem is?`
`00:09.23`	`JT`	`yeah, i should sue my car manufacturer, it does not come with step by step instructions on rebuilding the engine!`
`00:09.24`	`irule`	`its not for me dont you get it? many people dont touch * because it is stilll a geek only thing`
`00:09.31`	`Alan_Hicks`	`"I need an engine for my car. Is there an open-source engine I can just plug in?" No there isn't, because everyone has different needs.`
`00:09.43`	`JT`	`irule: building PBXes is NOT a task for nubs`
`00:09.57`	`irule`	`well seee :P`
`00:09.58`	`*** part/#asterisk jordanb (n=jordanb@adsl-68-20-22-211.dsl.chcgil.ameritech.net)`
`00:10.10`	`Alan_Hicks`	`The 2.2 4-banger in your four-door sedan won't cut it in my truck, but the 444 turbo diesel in my truck won't cut it in your car.`
`00:10.15`	`JT`	`it shouldn't be`
`00:10.22`	`Alan_Hicks`	`JT: Um, I'm a noob. :^)`
`00:10.27`	`JT`	`i've seen the results of unleashing idiots on TBXes`
`00:10.37`	`JT`	`Alan_Hicks: as in stupid people who don't want to learn`
`00:10.42`	`Alan_Hicks`	`Touche.`
`00:10.42`	`JT`	`s/TBXes/PBXes/`
`00:11.07`	`Alan_Hicks`	`</rant>`
`00:11.35`	`Alan_Hicks`	`Now I'll go back to occasionally requesting assistance from people who know much more about this stuff than I do.`
`00:14.39`	`*** join/#asterisk asdx (n=diego@adsl-155-217.click.com.py)`
`00:16.23`	`*** part/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`00:18.13`	`syle`	`open source engine interesting concept`
`00:18.20`	`*** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net)`
`00:18.23`	`syle`	`well i guess there is fuel cells and solar power`
`00:19.16`	`syle`	`that wouldn;t be great for long trips though`
`00:20.40`	`*** join/#asterisk dlynes_ (n=dlynes@216.251.149.68)`
`00:21.55`	`Alowishus`	`Alan_Hicks: that's the dial mask`
`00:22.11`	`Alan_Hicks`	`dial mask. Thanks.`
`00:22.21`	`Alowishus`	`or digit mask or something with the term "mask"`
`00:22.22`	`Alan_Hicks`	`Gives me something else to google.`
`00:22.37`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`00:25.13`	`javb`	`I have two * PBX, one with 1.2, the other, 1.4 ... 15 Polycom 330, and 5 Polycom 501, same config FILES. (copy and paste) ; SITUATION: 1.2 = PERFECT ; 1.4: Polycom 330 cant tranfer and cant handle call waiting, polycom 501 works ok. Any one here has had something similar?`
`00:26.42`	`*** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)`
`00:34.30`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`00:35.07`	`irule`	`http://swik.net/AsteriskDialplan there, my first draft! heuaheuaheuaheauehuae`
`00:36.03`	`irule`	`hmm, I should have posted this sooner! lol`
`00:36.28`	`*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)`
`00:39.38`	`*** join/#asterisk dlynes_laptop (n=dlynes@216.251.149.68)`
`00:44.32`	`Ritzerisk`	`what ever i try to do i cant get this line to ring through to the asterisk ... 3102 sipura to asterisk`
`00:46.35`	`Alan_Hicks`	`Got another rookie problem here. Gotta be something I'm doing wrong.`
`00:46.37`	`Alan_Hicks`	`http://pastebin.com/m8436fa2`
`00:47.38`	`Alan_Hicks`	`When I dial the line, Asterisk answers and plays me vm-enter-num-to-call.gsm fine, but when I hit "1" to proceed to Playback(digits/1) it never plays and the Asterisk console shows it hanging up.`
`00:48.10`	`Alan_Hicks`	`Actually, it hangs up whether I enter the number or not.`
`00:49.15`	`*** join/#asterisk mcab (n=mb@mostly-harmless.ca)`
`00:49.44`	`Ritzerisk`	`i am going sip set debug peer asterisk ... and it says no such peer and i set the peer in the sip conf`
`00:49.54`	`Alan_Hicks`	`Nevermind, it's just me being a dumb-ass.`
`00:50.15`	`Alan_Hicks`	`Everything works if I enter "1" or whatever before Background() completes.`
`00:52.51`	`*** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net)`
`00:58.27`	`*** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com)`
`00:58.33`	*** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq)
`01:02.26`	`[hC]`	`This is amusing, the asterisk appliance insists that my compact flash card be ext3, yet it only has mke2fs on it.`
`01:02.58`	`fujin`	`mke2fs -j`
`01:03.16`	`fujin`	`I assume you're trying to do somethign silly like mkfs.ext3 or something?`
`01:04.35`	`[hC]`	`no, i just wanted to format the CF card w/ ext3 so that this device will agree to use it. i didnt know about mke2fs -j, i thought i had to use tune2fs`
`01:05.20`	`[hC]`	`im trying that now.`
`01:06.02`	`[hC]`	`definitely have to make my own image for this thing, its missing a ton`
`01:06.18`	`[hC]`	`no less, tcpdump, scp, ssh, so far that ive found`
`01:06.51`	`[hC]`	`yay, that worked.`
`01:06.55`	`[hC]`	`Thanks fujin ;)`
`01:06.59`	`JT`	`a journalling fs may reduce the life of a flash card`
`01:08.39`	`[hC]`	`i agree, but for now this thing refuses to moount the flash card ext2, based on its init scripts`
`01:08.45`	`[hC]`	`another thing i will change.`
`01:09.07`	`[hC]`	`however, the reduced life may be better than a reboot one day and having no fs.`
`01:12.59`	`fujin`	`That's probably likely to happen regardless of a journal`
`01:13.01`	`fujin`	`with a cf disk`
`01:13.05`	`fujin`	`(eventually, anyway)`
`01:13.57`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net)`
`01:15.32`	`*** join/#asterisk bkw_ (n=brian@ppp-70-128-126-184.dsl.tulsok.swbell.net)`
`01:17.53`	`[hC]`	`fujin: eventually, yes.. but in our testing its quite a long time. a lot longer than originally thought`
`01:25.29`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-70-240-164-157.dsl.hstntx.swbell.net)`
`01:30.31`	`*** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)`
`01:31.00`	`ReDNeQ`	`sup`
`01:33.02`	`*** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)`
`01:36.13`	`*** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch)`
`01:47.05`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`01:54.05`	`*** join/#asterisk CrazyTux[m] (n=CrazyTux@70.240.164.157)`
`01:55.36`	`*** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com)`
`01:57.33`	`*** join/#asterisk kn0x (n=pinochle@75.127.83.151)`
`01:58.19`	`xai`	`how do i restart asterisk from the commandline or console? there is no service in this install.`
`01:59.22`	`HarryR`	`uh, don't you have init scripts setup?`
`01:59.57`	`xai`	`no, they didnt install it.. its a hack job. I didnt do it.`
`02:00.10`	`HarryR`	`stop now from the commandline`
`02:00.17`	`HarryR`	`then "asterisk" and whatever options it needs`
`02:04.49`	`xai`	`They use AMP too, so maybe that app is starting/stopping it.`
`02:06.07`	`HarryR`	`amp?`
`02:06.43`	`xai`	`its some gui they use..`
`02:07.02`	`HarryR`	`never heard of it`
`02:07.19`	`xai`	`it really is ugly, and has a bunch of alternate config files in /etc/asterisk..`
`02:07.28`	`xai`	`makes its a bit harder to track things.`
`02:07.34`	`HarryR`	`don't they all :)`
`02:08.16`	`*** join/#asterisk ming_zym (n=ming_zym@121.0.31.121)`
`02:10.40`	`*** join/#asterisk AsTeRiSk_2 (n=A@84sdl30m51.codetel.net.do)`
`02:14.48`	`*** part/#asterisk asdx (n=diego@adsl-155-217.click.com.py)`
`02:17.44`	`*** join/#asterisk ctooley (n=ctooley@209.33.109.249)`
`02:18.16`	`ctooley`	`is there a way to have a failover host in sip.conf in the event that the primary hostname fails?`
`02:18.33`	`JT`	`primary hostname of what?`
`02:18.51`	`ctooley`	`sorry, the primary address of the peer`
`02:19.49`	`JT`	`ip or dns`
`02:20.03`	`ctooley`	`ip`
`02:20.16`	`JT`	`dns srv`
`02:21.24`	`ctooley`	`can't use dns srv`
`02:22.55`	`*** join/#asterisk bintut (n=bintut@203.125.63.150)`
`02:24.35`	`AsTeRiSk_2`	`<PROTECTED>`
`02:24.44`	`AsTeRiSk_2`	`please see error in this page http://dpaste.com/23308/`
`02:25.14`	`JT`	`ctooley: then what do you want to use?`
`02:26.54`	`ctooley`	`JT, I'm asking if chan_sip supports a failover IP address`
`02:27.06`	`ctooley`	`that's what I would prefer to use.`
`02:27.26`	`JT`	`i don't think so`
`02:27.36`	`JT`	`sounds like a dialplan issue`
`02:28.51`	`AsTeRiSk_2`	`please run this on cli tail -f /var/log/asterisk/full`
`02:29.11`	`AsTeRiSk_2`	`you can see all proceses in real time and see what is the problem on the system`
`02:29.29`	`AsTeRiSk_2`	`or go to cli asteris -vvvr`
`02:29.35`	`AsTeRiSk_2`	`or go to cli asterisk -vvvr`
`02:29.56`	`AsTeRiSk_2`	`but i like more the tail -f /var/log/asterisk/full`
`02:30.05`	`AsTeRiSk_2`	`show all step by step`
`02:30.19`	`JT`	`AsTeRiSk_2: what are you talking about?`
`02:30.58`	`AsTeRiSk_2`	`sorry jt bad window`
`02:30.59`	`AsTeRiSk_2`	`lol`
`02:31.54`	`Mw3`	`does somebody around know what the time is in the PDT time zone currently?`
`02:32.06`	`*** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)`
`02:32.41`	`dlynes_laptop`	`Mw3: 7:30PM, PDT`
`02:33.04`	`Mw3`	`thanks, paypal is down till 3:00AM :(`
`02:33.14`	`AsTeRiSk_2`	`:x`
`02:35.45`	`*** join/#asterisk abritton (n=abritton@207-255-210-182-dhcp.cbe.md.atlanticbb.net)`
`02:36.18`	`abritton`	`hello`
`02:36.43`	`abritton`	`anybody available to answer a couple of questions`
`02:43.52`	`*** part/#asterisk abritton (n=abritton@207-255-210-182-dhcp.cbe.md.atlanticbb.net)`
`02:46.27`	`*** join/#asterisk Op3r (n=edwin@121.97.179.227)`
`02:54.52`	`phix`	`[hC]: I use hints :)`
`02:55.37`	`phix`	`hmmm voice mail recording has a very low volume`
`02:56.05`	`phix`	`There is a volume variable somewhere?`
`02:56.42`	`*** part/#asterisk ctooley (n=ctooley@209.33.109.249)`
`02:58.35`	`*** join/#asterisk modulus_ (n=modulus@shell.blacksun.net)`
`02:58.38`	`modulus_`	`hi`
`02:59.32`	`modulus_`	`anyone here ever tried bayhamsystems.com?`
`02:59.43`	`modulus_`	`for FastSMS() ?`
`03:01.01`	`modulus_`	`i tested via their web interface, it just says "Queued" forever`
`03:01.08`	`modulus_`	`bleh`
`03:18.53`	`phix`	`no`
`03:35.18`	`*** join/#asterisk PepOSX (n=pepOSX@190.72.148.244)`
`03:37.47`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`03:40.12`	`*** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey)`
`03:40.59`	`*** join/#asterisk dmitry-ita (n=dmitry@89-97-253-222.ip2.fastwebnet.it)`
`03:41.11`	`dmitry-ita`	`hi all`
`03:41.49`	`dmitry-ita`	`i have a strnage problem when i upgrade to asterisk 1.4 from 1.2`
`03:43.22`	`dmitry-ita`	`the call international not work, i get this error : app_dial.c:1111 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown)`
`03:43.33`	`dmitry-ita`	`only for international call`
`03:44.09`	`dmitry-ita`	`i think some zap parameters`
`03:44.37`	`dmitry-ita`	`hello!!! Is there anybody????`
`03:44.43`	`dmitry-ita`	`:)`
`03:47.01`	`*** join/#asterisk blq (n=Bl@dslb-088-067-019-040.pools.arcor-ip.net)`
`03:49.32`	`TrentCreek`	`no`
`03:50.21`	`phix`	`I am currently unavailable, please leave a message and I will get back to you, Please leave your message after the tone, when done hang up or press the pound key BEEP!`
`03:57.25`	`*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)`
`04:01.12`	`*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`04:01.12`	`*** mode/#asterisk [+o blitzrage] by ChanServ`
`04:17.16`	`*** join/#asterisk mihinomenest (n=argh@66.255.220.22)`
`04:18.18`	`*** join/#asterisk Op3r (n=Op3r@121.97.179.227)`
`04:24.16`	`*** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-243.pskn.east.verizon.net)`
`04:32.19`	`*** join/#asterisk asdx (n=diego@adsl-155-217.click.com.py)`
`04:32.22`	`*** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus)`
`04:33.08`	`*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)`
`04:43.30`	`*** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net)`
`04:43.46`	`*** part/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com)`
`04:51.00`	`*** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net)`
`05:16.17`	`Snake-eyes`	`Any one have any idea's how to get Asterisk to write the correct dst to cdr's when the callee hangs up first. I keep getting 's'/Default instead of the true dst number.`
`05:17.30`	`*** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl)`
`05:28.08`	`*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)`
`05:30.27`	`*** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net)`
`05:35.23`	`*** join/#asterisk ming_zym (n=ming_zym@121.0.31.121)`
`05:43.59`	`l0verb0y`	`hey hows it going`
`05:47.52`	`*** join/#asterisk bkw_ (n=brian@ppp-70-128-126-184.dsl.tulsok.swbell.net)`
`05:49.42`	`*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)`
`05:53.23`	`phix`	`hmmmm, is it normal to be able to hear other channels while on a call?`
`05:55.32`	`enalert`	`phix, generally no, Zaptel hardware?`
`05:55.59`	`phix`	`ok nm, I figured it out :) my bro had his softphone going through his stero speakers :P lol`
`05:56.16`	`enalert`	`hrm`
`05:56.36`	`enalert`	`sounds more like a genetic issue than a bug`
`05:56.40`	`enalert`	`Ducks`
`06:07.23`	`*** join/#asterisk dan__t (i=dan@neener.neener.org)`
`06:07.23`	`TrentCreek`	`I see a lot of people having problem with that Zaptel stuff. I say use SIP goodies and you cann;t go wrong`
`06:11.51`	`*** join/#asterisk dominic1 (n=dob@213.221.82.242)`
`06:14.34`	`*** part/#asterisk dominic1 (n=dob@213.221.82.242)`
`06:20.05`	`*** join/#asterisk iamamoron (n=kerlnel2@202.137.121.84)`
`06:20.08`	`iamamoron`	`hi there`
`06:20.20`	`iamamoron`	`${EXTEN:4} what is this supposed to mean?`
`06:20.40`	`iamamoron`	`according to doc ${EXTEN} is for current extension`
`06:21.12`	`iamamoron`	`but i have seen some dial plans that like this on ${EXTEN:1} or ${EXTEN:-4}`
`06:21.15`	`iamamoron`	`any ideas?`
`06:21.19`	`iamamoron`	`please shed some light`
`06:21.51`	`JT`	`~thebook`
`06:21.52`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf`
`06:25.56`	`iamamoron`	`is there no need to have DIAL() in call forwarding?`
`06:27.40`	`*** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com)`
`06:35.19`	`iamamoron`	`?`
`06:35.19`	`iamamoron`	`is there no need to have DIAL() in call forwarding?`
`06:37.55`	`*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)`
`06:39.49`	`iamamoron`	`?`
`06:39.59`	`iamamoron`	`any howto for call forwarding?`
`06:42.41`	`creativx`	`iamamoron: read up on the wiki on how to manipulate variables`
`06:52.40`	`*** join/#asterisk dlynes_laptop (n=dlynes@d154-20-34-39.bchsia.telus.net)`
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`06:56.06`	`iamamoron`	`any links?`
`06:57.44`	`*** join/#asterisk iamamoron (n=kerlnel2@202.137.121.84)`
`06:57.47`	`iamamoron`	`any ideas?`
`06:59.02`	`iamamoron`	`call forwarding`
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`07:18.34`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`07:22.50`	`Strom_M`	`CALL FORWARDING?!!?!?!?!?!!??!!?!?!?!?!!???!?!!!???!?!? HYPERVENTILATING`
`07:24.52`	`asdx`	`lol`
`07:31.00`	`jql`	`breathe`
`07:34.13`	`*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)`
`07:35.30`	`agx`	`In -1.4 is still active the freeze of all the SIP phones registered with the when ADSL freeze and chan_sip stop at syncronoush DNS querys?`
`07:36.20`	`*** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com)`
`07:36.27`	`TrentCreek`	`I see a lot of people having problem with that Zaptel stuff. I say use SIP goodies and you cann;t go wrong`
`07:45.46`	`*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)`
`07:45.48`	`Chris-NB`	`hi`
`07:46.01`	`Chris-NB`	`is it possible to store user defined values for a sip peer?`
`07:46.24`	`Chris-NB`	`in sip.conf, in addition to the normal sip parameters`
`07:48.09`	`jql`	`store?`
`07:48.18`	`jql`	`not that I know if`
`07:48.22`	`jql`	`of`
`07:48.22`	`linagee`	`has anyone here heard of something called E&M?`
`07:48.51`	`jql`	`linagee: yes, but I expect most would wish they hadn't`
`07:49.15`	`linagee`	`jql: is it like a pair of wires for mic and a pair for speaker, or is E&M some sort of digital signaling?`
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`07:49.29`	`jql`	`it's analog signalling`
`07:49.39`	`jql`	`as in tones`
`07:49.54`	`linagee`	`jql: so i could make a E&M test set with a RJ45 crimper and a speaker?`
`07:50.06`	`jql`	`like back in the old blue box days of the 70s`
`07:50.29`	`linagee`	`jql: we have a channel bank with E&M cards that goes to some sort of radio transmitter thingy`
`07:50.29`	`Chris-NB`	`jql, in sip.conf add a customiced variable/value for a sip user, so I can do special filtering in dialplan, depending on that parameter`
`07:51.01`	`linagee`	`jql: in order to take the E&M channels and combine them into a T1 cable to go into other equipment. (or so I understand it)`
`07:51.02`	`jql`	`Chris-NB: I'm a Perl programmer -- don't ask me how I'd implement that. it's frightening`
`07:51.14`	`Chris-NB`	`jql, *hehe : )`
`07:52.28`	`linagee`	`jql: the E&M card has an LED that says 600 ohm impedance. do i need a 600 ohm speaker? (does such a thing exist?)`
`07:52.59`	`JT`	`600ohm is the standard line imedence`
`07:53.02`	`JT`	`impedence`
`07:54.01`	`jql`	`and, a speaker? don't you just need a dumb $5 handset?`
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`07:54.25`	`jql`	`oh, I suppose it could come in digital. never mind`
`07:54.47`	`linagee`	`jql: there is 2 wire and 4 wire for E&M. that makes it even more confusing`
`07:55.01`	`linagee`	`jql: another provider delivers us a signal in E&M, we didn't chose it. heh`
`07:55.08`	`linagee`	`:(`
`07:55.14`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
`07:55.38`	`kaldemar`	`Chris-NB: take a look at parameter setvar in sip.conf`
`07:55.38`	`linagee`	`jql: and i'm not really the designer of said setup, just a confused maintainer. (i guess they like to trip us up with a lack of documentation)`
`07:55.56`	`Chris-NB`	`kaldemar, ok, thanks! I'll look for that`
`07:56.18`	`linagee`	`jql: geez this is confusing and annoying. "Wink StartWink is the most commonly used protocol. In this technique, the originating switch goes off-hook, waits for a temporary off-hook pulse from the other end (which is interpreted as an indication to proceed), then sends the dial digits."`
`07:58.16`	`linagee`	`jql: it was very stupid of them to use E&M instead of just T1 to T1.`
`07:58.43`	`linagee`	`jql: "yes, let's let the signal degrade over a long distance by using analog!"`
`07:58.47`	`linagee`	`:(`
`07:58.55`	`Shaun2222`	`anybody kno why i might get this error when trying to add a call to a queue...`
`07:58.56`	`Shaun2222`	`[Oct 25 00:53:10] WARNING[13262]: app_queue.c:3663 queue_exec: Unable to join queue 'sales'`
`07:59.10`	`Chris-NB`	`kaldemar, jup, that's exactly what I was looking for. thanks!`
`07:59.29`	`*** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se)`
`07:59.46`	`linagee`	`interesting. "The 2 wire E&M is technically same thing with a normal CO trunk."`
`08:00.02`	`linagee`	`2 wire E&M is like a telephone line i guess`
`08:00.39`	`TrentCreek`	`true true`
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`08:06.08`	`linagee`	`does anyone here use channel banks?`
`08:06.32`	`linagee`	`i guess if you had some bizarre reason to have a lot of FXS cards and wanted to terminate them to a T1`
`08:07.55`	`Strom_M`	`linagee: ive used channel banks`
`08:08.14`	`*** join/#asterisk qdk (n=qdk@85.235.253.139)`
`08:08.21`	`*** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)`
`08:08.58`	`linagee`	`Strom_M: interesting how a 50 pin amphenol connector has 25 pairs and there are 23 pairs available on a T1. it almost goes together perfectly. ;)`
`08:09.12`	`linagee`	`i'm sure this is no coincidence. ;)`
`08:09.18`	`Strom_M`	`linagee: you fail at T1`
`08:09.19`	`cy3o3`	`Strom_M: rumor has it you scored my loldong`
`08:09.23`	`linagee`	`Strom_M: 24`
`08:09.26`	`Strom_M`	`T1 has 24 channels`
`08:09.29`	`*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)`
`08:09.34`	`Strom_M`	`cy3o3: no, i merely auctioned it off`
`08:09.37`	`linagee`	`Strom_M: with one being used for data signaling`
`08:09.43`	`Strom_M`	`linagee: no, that's PRI`
`08:09.50`	`Strom_M`	`linagee: you need to go learn about how phones work`
`08:09.54`	`Strom_M`	`~101`
`08:09.54`	`jbot`	`hmm... 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf`
`08:09.58`	`Strom_M`	`^^^^^^^^^^^^^^`
`08:10.02`	`Strom_M`	`go read that`
`08:10.10`	`cy3o3`	`Strom_M: err, yeah :) Anyways great con`
`08:10.40`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`08:14.07`	`linagee`	`Strom_M: am back`
`08:14.18`	`linagee`	`Strom_M: PRI/T1/etc`
`08:14.32`	`linagee`	`Strom_M: still though. no small coincidence it's about the same amount? heh`
`08:14.38`	`Strom_M`	`uh`
`08:14.44`	`Strom_M`	`let me explain`
`08:15.00`	`Strom_M`	`if your ADD doesn't send you bounding towards cocks three lines in, that is`
`08:15.06`	`linagee`	`why is a T1 24 channels and not 25 though?`
`08:15.20`	`Strom_M`	`linagee: because 24 is an even multiple of eight`
`08:15.33`	`linagee`	`Strom_M: but there are 25 pairs on an ampenol connector`
`08:15.40`	`linagee`	`+h`
`08:15.41`	`Strom_M`	`linagee: and that connector predates T1`
`08:15.46`	`linagee`	`Strom_M: :->`
`08:16.28`	`*** join/#asterisk yang (i=yang@static-ip-62-75-255-124.inaddr.intergenia.de)`
`08:16.38`	`linagee`	`Strom_M: we have a 100 pair coming into the building`
`08:16.45`	`linagee`	`big fat cable`
`08:17.01`	`yang`	`I am trying to establish a callback to my asterisk gateway which would then call my VOIP phone....but these are the errors that I get http://openpaste.org/en/3594/`
`08:17.27`	`linagee`	`Strom_M: oh yes. but anyhow about the channel bank. i don't see why anyone would use a T1 24 FXS channel bank. heh`
`08:17.41`	`linagee`	`Strom_M: it would be like, "yes! let's set up 24 old time phones! wee!" :(`
`08:17.52`	`Strom_M`	`linagee: i've done just that`
`08:18.02`	`linagee`	`Strom_M: did the customer hate voip or something?`
`08:18.07`	`Strom_M`	`the company had about 60 analog sets that they were in no hurry to replace`
`08:18.11`	`linagee`	`lol`
`08:18.13`	`*** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru)`
`08:18.17`	`linagee`	`that sucks`
`08:18.24`	`Strom_M`	`linagee: it worked out quite nicely actually`
`08:18.38`	`Strom_M`	`there was no training to do, because the phones worked the same way`
`08:18.51`	`linagee`	`Strom_M: where they PBX-type phones, or just single line ones?`
`08:19.03`	`Strom_M`	`linagee: WECo 2500 sets`
`08:19.57`	`linagee`	`Strom_M: blech`
`08:20.07`	`linagee`	`Strom_M: beige phones you could run over with a truck`
`08:20.17`	`Strom_M`	`actually, these were all green :)`
`08:20.21`	`linagee`	`lol!`
`08:20.28`	`Strom_M`	`now stop being a ditkopf and go read that document`
`08:20.45`	`linagee`	`Strom_M: i've still got it on my desktop. it's a little dry. i'll have to add some salt`
`08:21.16`	`linagee`	`Strom_M: is it possible to have a channel bank that talks to multiline phone (or PBX phones, whichever term is "better")`
`08:21.21`	`linagee`	`+s`
`08:21.40`	`Strom_M`	`sure. run four analog lines to the phone...`
`08:21.51`	`*** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net)`
`08:22.07`	`linagee`	`Strom_M: we have an inter-tel axxess and i wish i could interject asterisk into it somehow`
`08:22.33`	`Strom_M`	`throw it into the dumpster`
`08:22.38`	`Strom_M`	`install asterisk`
`08:22.40`	`Strom_M`	`done and done`
`08:22.43`	`*** join/#asterisk shinao1 (n=shinao1@217.20.242.50)`
`08:22.50`	`linagee`	`LOL!`
`08:22.51`	`linagee`	`nice`
`08:23.10`	`*** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com)`
`08:23.18`	`linagee`	`Strom_M: they are 2-wire phones that use some "magic" to do digital signaling + power`
`08:23.30`	`Strom_M`	`linagee: yeah, they're digital.`
`08:23.35`	`Strom_M`	`not analog.`
`08:23.52`	`Strom_M`	`therefore, you can't use them with a channel bank no matter how many times you say "lol"`
`08:24.09`	`agx`	`is there any way to avoid SIP phones to freeze when chan_sip.c freeze on DNS queries? at least this happen to me with 1.2`
`08:24.12`	`linagee`	`aww`
`08:24.41`	`linagee`	`Strom_M: i could get a T1 option card and then lead that into an asterisk system somehow (whatever that would do)`
`08:25.14`	`Strom_M`	`linagee: you have to define your goal before you start determining how to get there`
`08:25.47`	`linagee`	`Strom_M: yikes. $700 for the card!`
`08:25.53`	`linagee`	`Strom_M: i'd like to record all calls`
`08:26.06`	`linagee`	`Strom_M: coming from an extension`
`08:26.15`	`linagee`	`Strom_M: which may or may not already be a feature of the system. :)`
`08:26.55`	`Strom_M`	`if you're trying to kludge that onto the inter-tel, i guarantee that won't work very well or be easy to implement`
`08:27.27`	`agx`	`is there any external repository for extra apps_* not included in default source tree?`
`08:27.52`	`linagee`	`Strom_M: in addition, it will have to have a warning announcement that will be a real kludge to add on`
`08:28.11`	`linagee`	`Strom_M: maybe i'll get a teddy roxbin tape cassette player to read the announcement. ;)`
`08:28.29`	`*** join/#asterisk litage\|w (n=nick@70.55.220.203.static.comindico.com.au)`
`08:28.31`	`linagee`	`ultra-kludge`
`08:28.35`	`Strom_M`	`linagee: yeah. so read telephony 101, read the asterisk book, replace the inter-tel with asterisk, and learn how to fucking spell "teddy ruxpin"`
`08:29.26`	`defswork`	`anyone know how to clear the directory on an aastra 9133i before loading a new one ? It seems to append`
`08:29.28`	`linagee`	`Strom_M: this thing is so creepy. http://images.businessweek.com/ss/06/09/popular_toys/image/teddy_ruxpin.jpg`
`08:29.33`	`Chris-NB`	`is it possible to distinguish if a call comes form a queue or if it's a 'direct' call?`
`08:29.50`	`Strom_M`	`linagee: i know what teddy ruxpin is`
`08:30.50`	`linagee`	`the company that made him declared bankruptcy 3 years after manufacturing him. heh`
`08:31.27`	`Strom_M`	`linagee: keep going like this and you'll win the off-topic moron of the month award`
`08:31.34`	`*** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com)`
`08:37.27`	`*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)`
`08:38.45`	`*** join/#asterisk sircrown (i=freetibe@206.71.151.43)`
`08:39.44`	`sircrown`	`ZT_SPANCONFIG failed on span 1: Invalid argument (22)`
`08:40.09`	`sircrown`	`any clues what this might refer to?`
`08:40.15`	`sircrown`	`moved a working config to a new box`
`08:40.20`	`Strom_M`	`sircrown: check your zaptel.conf`
`08:40.26`	`Strom_M`	`check your drivers`
`08:40.31`	`Strom_M`	`check layer one`
`08:40.32`	`sircrown`	`had a fonality box die on us`
`08:40.44`	`sircrown`	`installed trixbox at their suggestion`
`08:41.29`	`sircrown`	`layer1 is ok`
`08:41.38`	`sircrown`	`drivers i'm not 100% sure about`
`08:42.10`	`Strom_M`	`let me guess`
`08:42.18`	`Strom_M`	`sangagaglhaglahglahglaghlahglagoma`
`08:42.22`	`sircrown`	`yar.`
`08:42.25`	`sircrown`	`A104`
`08:42.55`	`Strom_M`	`fragile sangoma drivers + fragile trixbox == failure`
`08:42.58`	`Strom_M`	`try #trixbox`
`08:43.03`	`Strom_M`	`or call fonality for support`
`08:43.06`	`Chris-NB`	`is it not possible to check if a call comes from a queue (call an agent) or if it's a 'normal' call ?`
`08:43.36`	`jql`	`Chris-NB: I wouldn't know without checking the source. :(`
`08:45.50`	`Shaun2222`	`is there anyway to detect if a call is being sent to a phone from a queue?`
`08:46.31`	`Shaun2222`	`problem i have right now is that if a phone is on DND asterisk see's the phone as busy and proceeds to the next exten => which is voicemail..`
`08:46.47`	`Strom_M`	`uh`
`08:46.53`	`Strom_M`	`presumably if the phone is in DND...`
`08:47.01`	`Strom_M`	`...the person doesnt want to receive calls`
`08:47.14`	`Shaun2222`	`this is a problem because all calls in a queue are being routed to voicemail as if the person was answering the call`
`08:47.22`	`*** join/#asterisk cypherdelic (n=cypher@p5B27D0F0.dip.t-dialin.net)`
`08:47.57`	`yang`	`what does it mean -- SIP/pritrunk-091df940 is circuit-busy ?`
`08:48.03`	`Strom_M`	`tell your user to log out of the queue before pressing the DND button`
`08:48.06`	`*** join/#asterisk bantu (n=Miranda@p54A32A0F.dip0.t-ipconnect.de)`
`08:48.16`	`Shaun2222`	`Strom_C: there is no login/logout..`
`08:48.20`	`Shaun2222`	`it's a ring all type setup`
`08:48.30`	`Strom_M`	`Shaun2222: add a login/logout then`
`08:48.42`	`Shaun2222`	`but if one phone has dnd set, all calls are "answered" by voicemail`
`08:49.13`	`Shaun2222`	`Strom_M: dont want to... they dont have a choice of wether or not they are logged into the queue or not.`
`08:49.31`	`Strom_M`	`queue members are defined as....what?`
`08:50.29`	`Shaun2222`	`local/extension`
`08:51.01`	`Strom_M`	`don't use local if you don't want voicemail`
`08:52.05`	`Shaun2222`	`well i could do sip/ but that posses a problem with the screening process..`
`08:52.28`	`Strom_M`	`what screening process`
`08:52.30`	`Shaun2222`	`when the member answeres i have a macro run that tells them who it is and what they can do with them...`
`08:53.03`	`*** join/#asterisk harpal (n=Harpal@124.125.254.227)`
`08:53.08`	`Shaun2222`	`you have a call from ... press 1 to answer the call, press 2 to tell the caller to fuck off, press 3 to send caller to voicemail....`
`08:53.28`	`Strom_M`	`so...ok...then have a separate dialplan context for queued calls that doesnt include auto voicemail`
`08:53.38`	`Strom_M`	`and then use local to send queued calls into that context`
`08:54.15`	`Shaun2222`	`well ya, i could do that, but if i could detect if a call was comming from a queue i could just write a few extra lines to skip..`
`08:54.39`	`Shaun2222`	`i suppose i could set a a __VAR before they are added to the queue..`
`08:54.59`	`Shaun2222`	`thats what i do with the screen audio file..`
`08:56.22`	`*** join/#asterisk ming_zym (n=ming_zym@121.0.31.121)`
`08:59.48`	`agx`	`in sip.conf bindport= should be used before bindaddr= like in iax.conf or not?`
`09:00.32`	`Chris-NB`	`is it possible to translate the time/date in voicemail emails sent from asterisk?`
`09:00.40`	`Chris-NB`	`... into other languages`
`09:01.14`	`Strom_M`	`el deux de esperanto, nineteen seventy swahili`
`09:06.02`	`sircrown`	`any fonality users in here? :p`
`09:07.56`	`*** join/#asterisk billybongo (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk)`
`09:08.15`	`Shaun2222`	`global var works perfect... exten => _2XX,6,GotoIf($["${FROM_QUEUE}" != ""]?8)`
`09:08.49`	`Strom_M`	`Shaun2222: yeah, now try that under heavy load and see how wonderfully it totally fucks everything up`
`09:09.16`	`Shaun2222`	`why would it fuck everything up?`
`09:13.06`	`*** join/#asterisk ozus (n=ozus@ip72-205-206-86.sb.sd.cox.net)`
`09:18.53`	`*** join/#asterisk Op3r (n=edwin@121.97.179.227)`
`09:28.24`	`yang`	`what does it mean -- SIP/...trunk is circuit-busy ?`
`09:32.31`	`*** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net)`
`09:32.59`	`yang`	`i have these 2 lines in extension`
`09:33.03`	`yang`	`exten => _X.,1,Dial(SIP/pritrunk/${EXTEN:})`
`09:33.04`	`yang`	`exten => _X.,2,Hangup`
`09:33.14`	`yang`	`it used to work before`
`09:33.59`	`*** join/#asterisk pjezek (n=pj@193.85.164.154)`
`09:45.43`	`*** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)`
`10:03.03`	`*** join/#asterisk ccesario (n=ccesario@189.44.237.98)`
`10:15.38`	`*** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83)`
`10:17.52`	`dan__t`	`'morning.`
`10:19.13`	`sircrown`	`anyone good with sangoma cards?`
`10:19.31`	`sircrown`	`need some urgent help`
`10:23.08`	`*** join/#asterisk shtoom (n=shtoom@59.93.124.108)`
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`10:37.00`	`*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)`
`10:38.07`	`dan__t`	`Is there a way to tell Asterisk which ports to use for SIP on a device behind NAT?`
`10:38.09`	`*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)`
`10:38.13`	`dan__t`	`So I can just port forward those ports to the device`
`10:40.08`	`dan__t`	`Does that make sense? heh`
`10:41.55`	`dan__t`	`RTP ports.`
`10:42.11`	`yang`	`I have this error (circuit busy) http://openpaste.org/en/3595/`
`10:48.18`	`*** join/#asterisk gardo (n=gardo@125.212.12.23)`
`10:52.10`	`*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)`
`10:53.48`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`11:06.00`	`*** join/#asterisk myiagy (n=myiagy@189.4.73.139)`
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`11:29.44`	`*** part/#asterisk lukketto (n=lukketto@host192-27-dynamic.7-87-r.retail.telecomitalia.it)`
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`11:38.28`	`phix`	`Hey, what are some good VoIP providers in the UK?`
`11:41.45`	`k31th`	`phix: me :D`
`11:42.03`	HarryR`Work	`phix, voiptalk.org :)`
`11:42.53`	`k31th`	`we resell gradwell`
`11:42.58`	`k31th`	`which have been great`
`11:43.01`	HarryR`Work	`gradwell :\`
`11:43.09`	HarryR`Work	`their support is supposed to be pretty thin`
`11:44.30`	`k31th`	`well we do the support as we resell them.`
`11:44.38`	`k31th`	`we have found there support to be good.`
`11:45.10`	`k31th`	`less prompt at email replys lately`
`11:47.07`	`*** join/#asterisk stkn\|work (n=stkn\|wor@gentoo/developer/pdpc.active.stkn)`
`11:49.45`	`*** join/#asterisk michael-i (n=michael-@Lc2ac.l.pppool.de)`
`11:50.06`	`geminidomino`	`argh.. again with the no functioning hardware found...`
`11:52.04`	`michael-i`	`when Asterisk is looking for a sound file, does it use the file extension to determine what format each file is in, or does it look at structure? I have a prompt package that uses .g711u instead of .ulaw and was curious`
`11:58.44`	`*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)`
`12:02.30`	`*** join/#asterisk coppice (n=chatzill@232.196.17.210.dyn.pacific.net.hk)`
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`12:11.13`	`*** mode/#asterisk [+o blitzrage] by ChanServ`
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`12:36.21`	`yang`	`where could I ask questons regarding TELES?`
`12:39.25`	`geminidomino`	`Anyone have any idea why zaptel_hardware would detect my cards, but starting the zaptel drivers tells me "no functioning zap hardware found in /proc/zaptel" ?`
`12:48.13`	`*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)`
`12:48.13`	`*** mode/#asterisk [+o anthm] by ChanServ`
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`12:59.15`	`*** join/#asterisk walhala_ (n=walhala@resonix.fr)`
`12:59.35`	`walhala_`	`hi does anyone know how can i test my asterisk with sipsack and some bash scripts ?`
`13:00.30`	HarryR`Work	`walhala_, you use sipsack and some bash scripts`
`13:00.46`	`[TK]D-Fender`	`load chan_regurgitate.so`
`13:01.11`	`blitzrage`	`walhala_: I don't think anyone in here is going to build the scripts for you, if that's what you're asking...`
`13:02.23`	`walhala_`	`blitzrage: that's not what i'm asking but i want to know if there is a way to test asterisk with this tools :)`
`13:02.41`	HarryR`Work	`walhala_, what sort of testing?`
`13:02.45`	`*** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com)`
`13:02.55`	HarryR`Work	`if sipsack can do what you want, then together with bash you can do what you want`
`13:03.27`	`coppice`	`try #hollywood. there are lots of people there with scripts :-)`
`13:03.28`	`walhala_`	HarryR`Work: just if it's up
`13:03.34`	`blitzrage`	`coppice: lol`
`13:03.43`	`*** join/#asterisk nettie (n=nettie@ns.coolgadgets.it)`
`13:03.56`	`walhala_`	`coppice: :-D`
`13:05.50`	`nettie`	`Hi guys, I'm running a bristuffed version of asterisk 1.2 -- asterisk has problems detecting DMTF codes coming from the voip carrier. I tried all possible configurations, rfc, info, inband using a g711 codec, noway. I opened a ticket with them without receiving any answer yet so I decided to try here. Anyone know what could be wrong? we definitely need DTMF to navigate through our IVR. Thanx in advance`
`13:06.32`	`*** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net)`
`13:06.38`	`waKKu`	`folks... have many differences between a dialplan for 1.2 and 1.4 ? I wondering migrate my 1.2 site :)`
`13:08.30`	`[TK]D-Fender`	`waKKu: Go rad UPGRADE.TXT and all the other docs in the 1.4 tarball.`
`13:10.02`	`*** part/#asterisk \|Rain\| (i=rain@2001:440:eeee:fffb:42:0:0:2)`
`13:11.05`	`*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)`
`13:11.57`	`*** join/#asterisk Bl0w_M0nk (n=ak@66-168-56-207.dhcp.mdsn.wi.charter.com)`
`13:12.51`	`*** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)`
`13:12.53`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`13:13.07`	`*** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se)`
`13:13.08`	`jeremy_g`	`hi`
`13:13.37`	`*** join/#asterisk sasch (n=sasch@host117-234-static.4-79-b.business.telecomitalia.it)`
`13:13.39`	`jeremy_g`	`does asterisk 1.2 contain any alarms`
`13:14.06`	`*** join/#asterisk sasch (n=sasch@host117-234-static.4-79-b.business.telecomitalia.it)`
`13:15.11`	`sasch`	`hi all ... I'm italian and I try to install chan_mISDN for firt time ... do you have a link for how to ??`
`13:15.12`	`sasch`	`thanks`
`13:15.50`	`[TK]D-Fender`	`jeremy_g: load chan adt.so / chan_chubb.so`
`13:15.58`	`*** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)`
`13:18.52`	`Katty`	`mew.`
`13:19.15`	`blitzrage`	`mew urns`
`13:20.57`	`*** join/#asterisk Boones (i=Boones@port-212-202-42-6.dynamic.qsc.de)`
`13:21.46`	`Qwell`	`boo urns`
`13:22.09`	`Nugget`	`hey mister, that's a candle. you don't want to drink that!`
`13:25.57`	`Katty`	`mister burns...`
`13:26.17`	`*** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net)`
`13:30.42`	`Siya`	`anyone here that can point me to a cdr web frontend (billing/accounting only, no * admin required)`
`13:31.07`	`Siya`	`I'm finding loads of discussions via google but solutions seem far to be found`
`13:34.53`	`Siya`	`hmmm`
`13:35.19`	`phix`	`hey`
`13:35.34`	`phix`	`k31th: you provide VoIP ay`
`13:36.15`	`phix`	`k31th: nice, what are the plans like? I have a family friend in the UK that requires cheap overseas calls (to AU)`
`13:36.27`	`phix`	`Good local calls would be great too`
`13:36.54`	`phix`	`michael-i: depends`
`13:37.50`	`michael-i`	`phix: re: file extensions? i've found that asterisk does not like the g711u extensions`
`13:37.59`	`michael-i`	`thanks for the response :)`
`13:41.29`	`*** join/#asterisk duckz (n=duckz@81.180.83.75)`
`13:42.38`	`phix`	`yep, re file extensions, for use on moh or Playback() ?`
`13:43.14`	`phix`	`hmmm, is echo cancelling only applicable to zap?`
`13:43.34`	`phix`	`because I am getting an echo on my SIP line (probably from headset)`
`13:45.11`	`[TK]D-Fender`	`michael-i>phix: re: file extensions? i've found that asterisk does not like the g711u extensions <--- ???`
`13:45.39`	`[TK]D-Fender`	`phix: That would be ACCOUSTIC echo where you mic is picking up signal from your headphones.`
`13:45.55`	`michael-i`	`[TK]D-Fender: you missed this:`
`13:45.56`	`michael-i`	`when Asterisk is looking for a sound file, does it use the file extension to determine what format each file is in, or does it look at structure? I have a prompt package that uses .g711u instead of .ulaw and was curious`
`13:46.14`	`[TK]D-Fender`	`michael-i: it is indeed .ulaw`
`13:46.33`	`[TK]D-Fender`	`michael-i: mv .g711u .ulaw`
`13:46.35`	`[TK]D-Fender`	`<PROTECTED>`
`13:46.53`	`michael-i`	`already done, but first I got to hear the default prompts one last time :)`
`13:46.55`	`phix`	`[TK]D-Fender: yeah`
`13:47.38`	`phix`	`[TK]D-Fender: Asterisk doesn't have filter it can run it through?`
`13:47.48`	`phix`	`or do I need one for my headset?`
`13:47.52`	`[TK]D-Fender`	`phix: no.`
`13:47.56`	`phix`	`<PROTECTED>`
`13:48.04`	`[TK]D-Fender`	`phix: Fix your gains. Thats up to your soundcard`
`13:48.06`	`phix`	`hmmmm`
`13:48.20`	`[TK]D-Fender`	`phix: Or your mic is psycho.`
`13:48.24`	`phix`	`yeah I tried that, but then the person can't hear me well or I can't hear them :(`
`13:48.47`	`BBHoss`	`get new shit`
`13:48.49`	`*** join/#asterisk BadPacket (n=John@unaffiliated/badpacket)`
`13:48.56`	`phix`	`arh my ears! St Anger was in my playlist!`
`13:48.56`	`coppice`	`i thought it was norman that was psycho :-\`
`13:49.42`	`*** join/#asterisk jtexter3 (n=jamest@ppp-70-128-111-195.dsl.tulsok.swbell.net)`
`13:49.45`	`phix`	`therefore normal == my microphone?~`
`13:49.53`	`phix`	`norman even`
`13:50.11`	`*** join/#asterisk lukketto (n=lukketto@host192-27-dynamic.7-87-r.retail.telecomitalia.it)`
`13:50.25`	`*** part/#asterisk lukketto (n=lukketto@host192-27-dynamic.7-87-r.retail.telecomitalia.it)`
`13:50.29`	`[TK]D-Fender`	`coppice: Yeah... he's a "master" all right :)`
`13:51.09`	`[TK]D-Fender`	`phix: Soft phones suck, largely because of the soundcars & mics....`
`13:51.40`	`phix`	`[TK]D-Fender: ok, how can I resolve this? what are some good soundcards and mics?`
`13:51.55`	`[TK]D-Fender`	`phix: Buy a DECENT hard-phone.`
`13:52.06`	`[TK]D-Fender`	`~cheap`
`13:52.06`	`jbot`	`cheap is probably a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE.`
`13:53.16`	`*** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl)`
`13:53.38`	`phix`	`[TK]D-Fender: hmmmm, so a grandstream then :P :P`
`13:53.49`	`phix`	`heh`
`13:54.19`	`*** join/#asterisk bkw_ (n=brian@70.128.126.184)`
`13:55.11`	`Siya`	`as new, with love ebay :D`
`13:55.31`	`*** join/#asterisk tengulre (n=tengulre@219.144.192.2)`
`13:56.13`	`*** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net)`
`13:57.01`	`tengulre`	`hi,all`
`13:57.07`	`tengulre`	`where is Mark?`
`13:58.04`	`*** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au)`
`13:58.17`	`jtexter3`	`grr...10659 sucks`
`13:59.26`	`*** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au)`
`14:01.26`	`Katty`	`>.<`
`14:02.58`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`14:04.20`	`*** join/#asterisk DaveCanoe (n=Dave@adsl-065-007-135-002.sip.asm.bellsouth.net)`
`14:07.27`	`*** join/#asterisk ming_zym (n=ming_zym@218.108.10.151)`
`14:08.13`	`Katty`	`[TK]D-Fender: i has no caffeine :<`
`14:08.20`	`*** join/#asterisk jbondc (n=jbondc@modemcable158.97-203-24.mc.videotron.ca)`
`14:08.35`	`[TK]D-Fender`	`Katty: U acn has sum uv myne`
`14:08.36`	`jbondc`	`anyone have a mediatrix 1104?`
`14:09.12`	`coppice`	`mediatrix sucks`
`14:09.26`	`jbondc`	`i'm trying to find out if it's normal the fan's so noisy! :/`
`14:09.31`	`coppice`	`they have the most screwed up protocol implementation in the universe`
`14:10.37`	`jbondc`	`what?`
`14:11.08`	`jbondc`	`what's a better device.. i was told to get a mediatrix for my small home office`
`14:11.55`	`[TK]D-Fender`	`jbondc: FXS ro FXSO?`
`14:12.03`	`[TK]D-Fender`	`FXO*`
`14:12.08`	`[TK]D-Fender`	`asdfsfdgsfdgfagfdgg`
`14:13.00`	`jbondc`	`FXS`
`14:13.26`	`[TK]D-Fender`	`jbondc: That IS a waste...`
`14:13.39`	`[TK]D-Fender`	`jbondc: SPA-8000 <------`
`14:15.33`	`jbondc`	`that does look better :/`
`14:16.49`	`*** join/#asterisk jm\|laptop (n=jm\|home@dilbert.jamiem.com)`
`14:19.10`	`jbondc`	`it's working ok now, i'll see how it goes... had to put some tape to silence the fan! it's amazingly annoying, wish I could turn down the spin rate`
`14:20.00`	`agx`	`you can also try to use CheapStream gateway but you need to set them with static IP and obtain secret firmware to able to work :-P we had 2x8 port doing segmentation fault every`
`14:20.12`	`coppice`	`I have several workarounds in spandsp specifically to tolerate mediatrix bugs`
`14:20.18`	`phix`	`hey, are there any plugins / modules that can be used to connect to skype?`
`14:20.21`	`phix`	`hides`
`14:20.37`	`phix`	`also, how is asterisk with SCCP?`
`14:21.28`	`jbondc`	`phix, i wish (though there's some hacky solutions) then again don't use skype until they support open protocols`
`14:22.03`	`phix`	`yeah, it is just a client would like to contact skype users, I should tell them to tell their clients to setup a real PBX solution using asterisk :)`
`14:22.10`	`[TK]D-Fender`	`~skype`
`14:22.10`	`jbot`	`Skype is the bastard child of telephony. It uses a proprietary protocol for which only commercial channel drivers exist, all of which are DISGUSTING hacks at best. Forget about using Skype with Asterisk...`
`14:22.34`	`jbondc`	`lol that sums it up`
`14:22.35`	`[TK]D-Fender`	`phix: And you just keep looking for trouble... you're sure to find it...`
`14:22.43`	`agx`	`does Skype uses LPC10 as codec?`
`14:23.12`	`phix`	`[TK]D-Fender: :)`
`14:23.33`	`coppice`	`now now. LPC10 is much maligned, because there are so many buggy implementations around. its really not that bad for its bit rate`
`14:24.30`	`[TK]D-Fender`	`agx: ILBC I believe`
`14:24.39`	`*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)`
`14:24.55`	`[TK]D-Fender`	`coppice: Its the waiver at the end that doesn't spare it a bad review :)`
`14:25.05`	`coppice`	`it uses iLBC and the wideband code from the same people (forgot its name)`
`14:25.19`	`agx`	`[TK]D-Fender, i just wonder why nobody already reversed engineered skype protocol...`
`14:25.23`	`Dr-Linux`	`1.4 doesn't support ${TIMESTAMP} and same some other variables`
`14:25.34`	`Dr-Linux`	`what i can use instead?`
`14:25.41`	`[TK]D-Fender`	`Dr-Linux: Go read the docs.`
`14:25.57`	`jm\|laptop`	`GRTD aka RTFM?`
`14:26.08`	`coppice`	`[TK]D-fender: most people are surpised when I show them LPC10 working properly, as they have only ever heard broken versions`
`14:26.25`	`Dr-Linux`	`[TK]D-Fender: i read a doc which says "use function" but i didn't undersatand,`
`14:26.32`	`Dr-Linux`	`[TK]D-Fender: can you suggest any doc's link?`
`14:26.44`	`[TK]D-Fender`	`Dr-Linux: yeah.. the DOCS folder in your source tarball.`
`14:26.58`	`[TK]D-Fender`	`Dr-Linux: and "show functions" <-----`
`14:27.05`	`Dr-Linux`	`cool`
`14:27.16`	`geminidomino`	`was 'zap show *' among the deprecated commands in 1.4.13? chan_zap is loaded but it tells me there's no such command as 'zap show channels'`
`14:27.19`	`coppice`	`you can do a lot better than LPC10 at the same bit rate with most modern techniques, though`
`14:27.32`	`*** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net)`
`14:31.11`	`Katty`	`[TK]D-Fender: ^_^`
`14:31.12`	`Katty`	`[TK]D-Fender: i has discovered teh caffeines.`
`14:31.15`	`Katty`	`[TK]D-Fender: they are MINE`
`14:31.19`	`Katty`	`[TK]D-Fender: all of them ^_^`
`14:32.43`	`tzafrir`	`geminidomino, no. if you don't have 'zap <whatever>, chan_zap.so is not currently loaded for whatever reason`
`14:33.58`	`*** join/#asterisk Yoe (n=wouter@samba.grep.be)`
`14:34.24`	`Yoe`	`Is anyone here aware of a hardware SIP phone vendor that provides extended warranty contracts without a time limitation?`
`14:34.37`	`geminidomino`	`thats wierd...`
`14:34.42`	`Yoe`	`geminidomino: not so much`
`14:34.58`	`Yoe`	`geminidomino: I have a customer who has something like that currently for their POTS PBX`
`14:35.05`	`BBHoss`	`heh`
`14:35.12`	`BBHoss`	`two different worlds :)`
`14:35.19`	`geminidomino`	`Yoe, I wasn't talking to you about that being wierd. :)`
`14:35.28`	`Yoe`	`geminidomino: oh, heh. Sorry :)`
`14:35.41`	`geminidomino`	`tzafrir: module show like zap shows chan_zap.so. Does that mean its present but not loaded?`
`14:35.48`	`BBHoss`	`technology moves so much faster with VoIP phones, but im sure there is SOMEONE out there that does it`
`14:36.18`	`phix`	`hmmm, when someone rings my DID number the callerid is picked up as my DID Number, is that normal?`
`14:36.31`	`*** join/#asterisk Op3r (n=edwin@121.97.179.227)`
`14:36.37`	`BBHoss`	`not usually`
`14:36.51`	`phix`	`hmmm, I need to do some experimentations`
`14:37.14`	`Splat`	`phix: what provider are you using?`
`14:37.46`	`phix`	`Pennytel`
`14:37.54`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`14:38.04`	`*** join/#asterisk Netyciukas (n=root@87-239-84-128.kis.user.lt)`
`14:38.05`	`phix`	`I think I had a callerid= line in my VoIP provider context, I should investigate`
`14:38.31`	`Netyciukas`	`hey`
`14:38.59`	`Netyciukas`	`is it able asterisk to bind sip address on more then one ip address`
`14:39.05`	`Netyciukas`	`but not on all addresses`
`14:39.25`	`Splat`	`phix: I haven't heard of any caller if issues with them.. but it's probably worth checking whirlpool and see if anyone else has been complaining in there about it..`
`14:39.28`	`Netyciukas`	`in ex. i have 5 ips, 1.1.1.1 2.2.2.2 3.3.3.3 4.4.4.4 and 6.6.6.6`
`14:39.39`	`Netyciukas`	`i want asterisk to bind on 1.1.1.1 ant 6.6.6.6`
`14:39.55`	`phix`	`Netyciukas: do it then`
`14:39.55`	`Netyciukas`	`how could i be done without doing port forwarding?`
`14:40.25`	`phix`	`Splat: you know of any shared address books?`
`14:40.39`	`phix`	`that work with ip phones and softphones?`
`14:40.47`	`phix`	`need a centralised address bool`
`14:40.48`	`phix`	`book`
`14:40.53`	`*** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl)`
`14:41.20`	`phix`	`would also like a shared address book in thunderbird too, although apparantly it doesn't have write support`
`14:41.31`	`Netyciukas`	`ldap?`
`14:41.51`	`*** join/#asterisk supjigator (n=sysgod@152.53.16.10)`
`14:42.41`	`Splat`	`phix: nope.. only a very crappy web based one that I maybe half finished.. less then half finished I would think.. heh I just wanted to play with setting up my own click to dial interface`
`14:43.04`	`BBHoss`	`Netyciukas: i think using multiple bindaddr lines in sip.conf might work, but i havent tried it`
`14:43.50`	`phix`	`Splat: hmmm, I was thinking about getting an ip phone that supports XML/HTML`
`14:44.01`	`BBHoss`	`Polycoms are nice`
`14:44.07`	`BBHoss`	`Aastra's are good too`
`14:44.15`	`blitzrage`	`Netyciukas: asterisk will either bind to all, or 1 IP`
`14:44.17`	`*** join/#asterisk zeroHalo-BB (n=zeroHalo@64.78.21.135)`
`14:44.32`	`phix`	`BBHoss: yeah`
`14:44.52`	`phix`	`Netyciukas: yeah LDAP would be nice`
`14:45.11`	`phix`	`especially if I could integrate it with my current LDAP setup`
`14:45.25`	`BBHoss`	`phix: you could write a php script that interfaces with LDAP, then goes over XML-Microbrowser`
`14:45.25`	`Splat`	`only Aastra I've ever used wasn't fit to be a door stop or for decorating a fish tank.. mind you it's a rather old Aastra which is why it's such a piece of crap.. heh`
`14:45.29`	`geminidomino`	`Yay! I got green!!!`
`14:45.41`	`Maliuta`	`f00f`
`14:45.53`	`BBHoss`	`Splat: the aastra i use now rivals the polycom IMO`
`14:45.53`	`phix`	`BBHoss: yeah, that would be nice, although I would use perl or j2ee instead of PHP :)`
`14:45.56`	`*** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl)`
`14:46.06`	`*** part/#asterisk geminidomino (n=ciro@65.41.157.192)`
`14:46.15`	`BBHoss`	`yeah perl would work, theres probably already something in cpan to connect to LDAP sources`
`14:46.23`	`Netyciukas`	`blitzrage: it's not possibe to bind just few of ips, not all?`
`14:46.32`	`blitzrage`	`Netyciukas: I wasn't lieing`
`14:46.33`	`phix`	`BBHoss: everything is in CPAN`
`14:46.39`	`BBHoss`	`:)`
`14:46.42`	`Maliuta`	`cpan is evil'`
`14:46.51`	`Maliuta`	`it's like pear`
`14:46.52`	`phix`	`shits all over PEAR :)`
`14:47.15`	`phix`	`is it like marvin too? :)`
`14:47.20`	`Maliuta`	`it's unstable untrustable shite :P`
`14:47.26`	`[TK]D-Fender`	`BBHoss: Only thing on Aastra's side is their sidecars & soft-key handling. Everything else is socond rate :/`
`14:47.37`	`Splat`	`pear isn't evil.. it's just crap..`
`14:47.40`	`phix`	`(javas equiv, although it is also a builder as well like ant, makefile, etc..`
`14:47.43`	`phix`	`)`
`14:48.25`	`Maliuta`	`Splat: my point about cpan being?`
`14:49.27`	`phix`	`any way, lets asterisk :)`
`14:49.32`	`bkw_`	`lets not`
`14:50.26`	`phix`	`yeah ditto`
`14:50.33`	`phix`	`getting on 1:00AM`
`14:50.39`	`Splat`	`heh`
`14:50.49`	`Splat`	`that's only because you start DST later then us..`
`14:50.56`	`Splat`	`it's almost 2am here`
`14:50.58`	`*** join/#asterisk irule (n=irule@200.53.61.4)`
`14:51.03`	`phix`	`your a tasy?`
`14:51.04`	`irule`	`gmornin!`
`14:51.10`	`Maliuta`	`phix: yeah, and I have had too much to drink`
`14:51.11`	`Splat`	`yeah`
`14:51.15`	`phix`	`Maliuta: heh`
`14:51.19`	`supjigator`	`Anyone aware of any class 5 type sip routing platforms that are opensource?`
`14:51.25`	`phix`	`Splat: nice :) never knew the Internet worked there :)`
`14:51.25`	`Maliuta`	`damn work functions with OS execs`
`14:51.26`	`phix`	`:P`
`14:51.48`	`*** join/#asterisk Darthclue (n=e054502@fw149.nisd.net)`
`14:52.08`	`Splat`	`phix, I'm currently going through a number of relays run off tin cans and string.. the messages get garbled quite often..`
`14:52.18`	`Maliuta`	`if it was next week it would be 2am, daylight savings doesn't start until saturday night`
`14:52.19`	`phix`	`hehe :P`
`14:52.36`	`phix`	`yeah`
`14:52.37`	`Maliuta`	`the internet "works"`
`14:52.43`	`*** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-f7bedd9d060b20e1)`
`14:52.55`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`14:53.17`	`Splat`	`think Chinese whispers where everyone speaks a different language.. that's my internet connection.. :P`
`14:53.29`	`phix`	`:)`
`14:53.49`	`phix`	`what ISP?`
`14:54.00`	`phix`	`I know the ISP I am with doesn't touch TAS :)`
`14:54.43`	`Splat`	`I'm on internode.. it's actually pretty good but then I'm on a business connection and I'm in collinsvale so there's only like 12 other people using adsl off the exchange.. :P`
`14:54.44`	`coppice`	`whispering doesn't work too well on IRC, but if they type Chinese it will be OK`
`14:55.10`	`phix`	`ay`
`14:55.11`	`phix`	`ah`
`14:55.30`	`phix`	`Splat: always good`
`14:55.53`	`Splat`	`none of this new fangled adsl2 stuff..`
`14:56.54`	`Splat`	`the only upgrade telstra will do up here is sticking up a next g tower and saying if you want faster internet (or internet at all if you can't get it yet) then it's on to next g with you..`
`14:57.49`	`irule`	`hi guys, I am atarting a small effor to create a basic PBX dialplan to spped up initial * deployment http://swik.net/AsteriskDialplan have any of you suggestions? thanks!`
`14:57.56`	`phix`	`next g ay, that is after 3g right?`
`14:58.11`	`phix`	`brb`
`14:58.31`	`*** join/#asterisk litage_ (n=nick@70.55.220.203.static.comindico.com.au)`
`14:58.45`	`*** join/#asterisk l2trace99 (n=asd@75.112.133.235)`
`14:59.52`	`*** join/#asterisk Op3r (n=edwin@121.97.179.227)`
`15:01.14`	`[TK]D-Fender`	`irule: Priority jumping is DEAD, and you really need a spell-checker.`
`15:01.40`	`irule`	`[TK]D-Fender its a start! lol thanks!`
`15:01.42`	`Maliuta`	`[TK]D-Fender: a speel checker?`
`15:01.44`	`*** join/#asterisk ussrback (n=MAX@81.95.160.147)`
`15:01.53`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`15:02.12`	`[TK]D-Fender`	`irule: And the "dialplan" is really so empty as to have nothing worth commenting on. Its got 1 exten.`
`15:02.19`	`Maliuta`	`Splat: helstra don't "do" upgrades`
`15:02.53`	`Maliuta`	`Splat: their copper network in my area is horrible`
`15:03.27`	`*** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net)`
`15:03.32`	`Splat`	`heh`
`15:04.09`	`Splat`	`well I'm gonna hit the sack now I think`
`15:04.24`	`Maliuta`	`helstra: sucking the life out of .au telecoms`
`15:05.19`	`ussrback`	`does mysql no more supported in * ?`
`15:05.54`	`adeel`	`i'm having the most acinine problem with my * box...i replaced my router (and copied the config over) and now * won't register with my service provider...it just says 'unreachable' ...i don't see any register attempts in the sip debug output, is there a way to force * to send a register request?`
`15:06.21`	`l2trace99`	`asterisk -r -x "sip reload"`
`15:06.32`	`Katty`	`anyone attending the wow zombie raid on stormwind?`
`15:07.15`	`*** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net)`
`15:07.26`	`*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)`
`15:08.05`	`l2trace99`	`adeel : also "sip debug peer <peername>" from the console may give you more info to what is going on`
`15:08.51`	`Maliuta`	`logging the TCP requests on the SIP port may also help`
`15:09.06`	`Maliuta`	`see what is being dropped by the "router"`
`15:09.31`	`l2trace99`	`anyone seeing high cpu loads when adding members to queue ?`
`15:09.52`	`adeel`	`l2trace99, i've done both, the sip reload still doesn't send out the registration requests...let me pastebin the sip debug output`
`15:10.16`	`[TK]D-Fender`	`Maliuta...... cough UDP cough`
`15:10.36`	`l2trace99`	`before you do that`
`15:10.57`	`sasch`	`i try to compile the mISDN support for asterisk ... http://www.misdn.org/index.php/MISDN_with_Asterisk`
`15:11.03`	`l2trace99`	`are you sure asterisk can ping the peer`
`15:11.04`	`Maliuta`	`[TK]D-Fender: bite me, the same thing can be logged :P`
`15:11.17`	`Maliuta`	`so I have had "a few"`
`15:11.19`	`Maliuta`	`:P`
`15:11.20`	`sasch`	`but i recevie this http://pastebin.ca/749227`
`15:11.25`	`[TK]D-Fender`	`Maliuta : I would.... if not for fear of FOOD POISONING :)`
`15:11.44`	`Maliuta`	`[TK]D-Fender: oh, I do more than poison :P`
`15:12.21`	`irule`	`pretty good fot a 000000000000.0 version ;)`
`15:12.23`	`coppice`	`be constructive. poison some telstra managers`
`15:12.25`	`Maliuta`	`l2trace99: I think your conf is "queer"`
`15:13.01`	`Maliuta`	`coppice: I would do away with helstra if I could`
`15:13.05`	`adeel`	`l2trace99, http://nopaste.com/p/a9nhYqKMjb`
`15:13.25`	`coppice`	`Maliuta: well you gotta start somewhere`
`15:14.02`	`Maliuta`	`coppice: sending 9.99% of my stuff over VoIP is a start`
`15:14.23`	`Maliuta`	`coppice: the next stage is assasinating Sol`
`15:14.41`	`Maliuta`	`I'll work on that when I have time`
`15:15.28`	`*** join/#asterisk naitram (n=chatzill@216.77.58.40)`
`15:17.17`	`l2trace99`	`adeel are you sure your router is natting the 192.168.1.10 to the 209.157.131.178 ?`
`15:17.36`	`naitram`	`on a sip channel, I originate a call via AMI. I don't want an audible ring to be played on the sip ua. How do I stop asterisk from sending a ring? The sip ua i am using does not have the capability to play the ring itself so I know ast* is doing it.`
`15:19.21`	`*** join/#asterisk punkgode (n=punkgde@rev-200-40-119-222.netgate.com.uy)`
`15:19.30`	`adeel`	`l2trace99, that's the thing, i've added the entries to port forward any requests on 5060, the udp range, etc, to 192.168.1.10....but i'm thinking it's a router problem as well`
`15:19.46`	`adeel`	`s/udp range/rtp range`
`15:20.14`	`Maliuta`	`adeel: but are you dealing with the incoming SIP and RTP?`
`15:20.19`	`*** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com)`
`15:20.27`	`*** join/#asterisk voipnet-tech (n=voipnet-@216.195.128.62)`
`15:20.37`	`voipnet-tech`	`morning all`
`15:20.49`	`voipnet-tech`	`who's going to the VON conference in Boston next week ?`
`15:20.49`	`Maliuta`	`adeel: the outgoing is thing the incoming is another`
`15:22.25`	`l2trace99`	`Maliuta: why would you say my config is queer ?`
`15:22.42`	`cygar`	`Hey guys I have converted an old trixbox to Realtime and everything works so far BUT when I try to make an incoming call to an extension that was created through an INSERT in the sip_buddies table of the realtime db some AGI [looks something trixbox/freepbx related] tells me that there's no AMPUSER with that extension [ even if it's registered and it can make outgoing calls ]. After a few research I could "patch" it making in the CLI> database put AMPUSER ext`
`15:22.49`	`Maliuta`	`l2trace99: because they are ghay?`
`15:22.50`	`cygar`	`BUT, it's very ugly to do this.`
`15:23.31`	`*** join/#asterisk pjezek (n=pj@193.85.164.154)`
`15:23.33`	`Maliuta`	`l2trace99: it _may_ have something to do with and excess of single malt`
`15:23.51`	`[TK]D-Fender`	`cygar: Dialplan BS.`
`15:23.53`	`cygar`	`Does anyone knows how to skip this "trixbox" [ some AGI's in there ] stuff to not search for this AMPUSER ? FreePBX is not used anymore right now with Realtime in that box.`
`15:23.57`	`[TK]D-Fender`	`~freepbx`
`15:23.58`	`jbot`	`[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there`
`15:24.07`	`cygar`	`[TK]D-Fender: What do you mean by Dialplan BS ?`
`15:24.29`	`[TK]D-Fender`	`cygar: Means we don't support FreePBX and its silly scripts and dialplan.`
`15:24.30`	`adeel`	`Maliuta, i don't know what you're trying to get at....the outgoing packets have nothing to do with the incoming ones...the incoming packets must be routed correctly through the firewall, which according to my config should be correct; however, things aren't working and i can't find any meaningful output`
`15:24.48`	`[TK]D-Fender`	`adeel: If its checking AstDB, go rip that out.`
`15:25.07`	`[TK]D-Fender`	`adeel: or do "whatever", but we don't support that stuff here.`
`15:25.19`	`cygar`	`[TK]D-Fender: hehhe I agree with that mate, but I had no option since this was an already installed box. I just convert it to realtime.`
`15:25.47`	`[TK]D-Fender`	`cygar: The fact you feel you have no option makes little difference.`
`15:26.12`	`Maliuta`	`adeel: if you are having problems getting a register, and the outgoing packets are being SNAT'd properly it has to do with the DNAT of the incoming packets`
`15:26.40`	`Maliuta`	`adeel: anyhow it's not an asterisk issue`
`15:26.54`	`Maliuta`	`it's got to do with your "router" config`
`15:27.26`	`adeel`	`Maliuta, well the thing is, i don't ever see a register packet leaving my box..`
`15:27.50`	`ussrback`	`hi all. I get errors when im trying to install asterisk-addons ---> http://pastebin.ca/749244`
`15:27.52`	`*** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au)`
`15:27.54`	`Maliuta`	`you're logging the outgoing?`
`15:28.07`	`Maliuta`	`not dropping the packets with an earlier rule?`
`15:28.41`	`adeel`	`Maliuta, the outgoing is going through a linksys router, so i doubt it's doing anything as advanced as packet shaping`
`15:28.52`	`*** join/#asterisk FlatFoot (n=bigflatf@80.88.192.83)`
`15:29.19`	`adeel`	`yeah sip and nat is pretty straight foward...until you hit an obscure problem like i'm having`
`15:29.27`	`Maliuta`	`adeel: so I doubt the NAT is being done properly too`
`15:29.28`	`adeel`	`i can post a longer sip debug output`
`15:29.40`	`*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)`
`15:30.27`	`naitram`	`is there a way to specify a different ring tone per channel`
`15:31.26`	`cygar`	`adeel: can't you sniff at application level to see if , when negotiation where the voice is going to be sent, if it's sending the right source IP address and not the local one?`
`15:32.48`	`adeel`	`cygar, well so far, i haven't even gotten to that point where i'm able to make a call...i first need to get the box to register again...`
`15:33.01`	`adeel`	`Maliuta, http://www.nopaste.com/p/ad6aOQYNw`
`15:33.33`	`adeel`	`the only thing that had been changed was the router`
`15:34.00`	`adeel`	`so it's leading me to believe that the router is where the problem lies, but i just wanted another opinion to confirm my suspicions`
`15:34.54`	`naitram`	`i am trying to orginate a call between two sip clients. I have the clients set up to autoanswer. They do, but I always get 1 audible ring. From asterisk? Don't want the ring tone. how to remove`
`15:35.44`	`agx`	`adeel, some router has buggy SIP handling, check if the router has an option for it or change bindport in sip.conf to 5062 ... in Italy FastWeb provider filter 5060 traffic`
`15:36.38`	`adeel`	`agx, i can try changing the port, but i don't think that's where the problem is (the box is in the USA and never had a problem on 5060 before)`
`15:37.05`	`*** join/#asterisk sriramnrn (n=chatzill@220.226.20.113)`
`15:40.01`	`ussrback`	`hi all. I get errors when im trying to install asterisk-addons ---> http://pastebin.ca/749244`
`15:40.07`	`ussrback`	`How can i fix it?`
`15:40.25`	`sriramnrn`	`~ asterisk book`
`15:40.40`	`sriramnrn`	`help download asterisk book`
`15:40.43`	`Corydon76-vcch`	`~thebook`
`15:40.43`	`jbot`	`Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf`
`15:41.00`	`*** join/#asterisk shinao1 (n=shinao1@217.20.242.50)`
`15:41.03`	`sriramnrn`	`thanks :)`
`15:41.07`	`*** join/#asterisk bkw_ (n=brian@ppp-70-128-126-184.dsl.tulsok.swbell.net)`
`15:41.37`	`k31th`	`humm i am following that book but i cant get my phones to register using secret=password`
`15:42.20`	`Maliuta`	`gah! can't resist ...`
`15:42.31`	`Maliuta`	`k31th: you lose! :P`
`15:42.54`	`k31th`	`:( yeah`
`15:43.32`	`k31th`	`Maliuta: the phone apears in sip show peers`
`15:43.36`	`k31th`	`i dont get it.`
`15:44.04`	`*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)`
`15:44.10`	`Maliuta`	`hey, my stuff works (IAX and SIP) with the secret entries`
`15:44.35`	`Maliuta`	`sounds like you screwed a client conf`
`15:45.32`	`k31th`	`Maliuta: well im filling it in by hand on a handset (both a snom and a linksys).`
`15:45.49`	`*** part/#asterisk sriramnrn (n=chatzill@220.226.20.113)`
`15:46.48`	`*** join/#asterisk blq (n=Bl@dslb-088-066-229-151.pools.arcor-ip.net)`
`15:47.36`	`naitram`	`k31th: if you hardcode the ip addresses in your sip.conf, they will show up when you do the sip show peers even though they have not really registered. So, set it to host=dynamic. I think that is right and then only after the credentials on both are correct will they show up in sip show peers`
`15:48.09`	`*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)`
`15:49.11`	`*** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru)`
`15:50.46`	`k31th`	`naitram: that is what i have done, let me show you my sip.conf`
`15:50.51`	`k31th`	`http://pastebin.ca/749265`
`15:51.02`	`k31th`	`very basic, (i am following the book)`
`15:51.37`	`*** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net)`
`15:51.58`	`*** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210)`
`15:54.11`	`naitram`	`k31th: you dont have any username or callerid. Take another look at the sip.conf. The username and callerid, (both i think) are used to authenticate.`
`15:54.37`	`blitzrage`	`callerid would not be used to authenticate`
`15:54.59`	`punkgode`	`k31th, u didn't setup a username`
`15:55.07`	`naitram`	`blitzrage: ok, i dont know which is used but....`
`15:55.15`	`blitzrage`	`but?`
`15:55.40`	`*** join/#asterisk Xenon3DN (n=Xenon@mail.3dnature.com)`
`15:55.48`	`k31th`	`naitram: punkgode thanks... was just following this book tbh by the looks of it to blindly`
`15:55.57`	`*** join/#asterisk sakic (n=sakic@adsl-154-55-232.clt.bellsouth.net)`
`15:57.05`	`naitram`	`k31th: anyway, once you get that set up, you might have to experiment with you sip phone setup to figure out where you actually have to place the username. Sometimes they use username, sometimes auth name....`
`15:58.13`	`*** join/#asterisk s0lid (n=_freq@210.213.243.193)`
`16:00.09`	`k31th`	`naitram: yeah i use SPA942's and Snom's`
`16:01.10`	`Xenon3DN`	`Heya. I have been trying unsuccessfully to post a question to the asterisk-users mailing list. I've given up on that -- would anyone mind looking at the text of my question here and telling me if you have any suggestions? It's kinda verbose (written for the ML) so I'll send it to pastebin to not flood the channel.`
`16:01.59`	`*** part/#asterisk Yoe (n=wouter@samba.grep.be)`
`16:02.17`	`*** join/#asterisk DataCompBoy (n=datacomp@213.187.250.34)`
`16:02.50`	`*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)`
`16:02.56`	`DataCompBoy`	`Hi all! I have some problem with zaptel and outgoing CID presentation`
`16:04.27`	`Strom_M`	`DataCompBoy: what kind of probem`
`16:04.29`	`Strom_M`	`problem`
`16:05.37`	`Xenon3DN`	`http://pastebin.com/d51c9e2`
`16:05.46`	`Xenon3DN`	`Do those symptoms ring a bell with anyone?`
`16:06.24`	`DataCompBoy`	`http://pastebin.ru/227267 -- here is mine problem...`
`16:07.40`	`Strom_M`	`DataCompBoy: that's not a problem so much as it is just a block of code`
`16:08.53`	`Strom_M`	`which country are you in....who is your telco...is this ISDN PRI...what is the numbering plan for your country...etc`
`16:09.38`	`k31th`	`naitram: username still doesnt work`
`16:10.42`	`DataCompBoy`	`Strom_M: that not code :) there at top -- cut from config, at bottom - list from debug`
`16:11.03`	`irule`	`guys Ive made a few mods, how do you like it? http://swik.net/AsteriskDialplan`
`16:11.10`	`DataCompBoy`	`Strom_M: I have set up callerid, but on outgoing Calling number "Number not available (67)".`
`16:11.34`	`Strom_M`	`DataCompBoy: ok...will you answer my other questions above, please?`
`16:11.58`	`*** part/#asterisk billybongo (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk)`
`16:12.16`	`DataCompBoy`	`Strom_M: server in germany, 2*E1, provider Deutsche Telekom (he know nothing about asterisk's config :( )`
`16:12.55`	`*** join/#asterisk krondorl (n=chatzill@207.245.216.9)`
`16:12.58`	`Strom_M`	`DataCompBoy: share your sapata.conf`
`16:12.59`	`krondorl`	`hello!!`
`16:13.26`	`Strom_M`	`er`
`16:13.29`	`Strom_M`	`zapata.conf`
`16:14.03`	`DataCompBoy`	`Strom_M: full?`
`16:14.11`	`Strom_M`	`DataCompBoy: yes`
`16:14.20`	`agx`	`is there any external addon package with apps for * 1.4 ?`
`16:14.52`	`krondorl`	`I have CLOCK="UTC" and TIMEZONE="EST5EDT" in my clock.conf file but my time is 6 hours in the future and I get the following: Thu Oct 25 16:11:30 Local time zone must be set--see zic manual page 2007 Anyone know what I might have missed?`
`16:16.28`	`Corydon76-vcch`	`krondorl: you missed the symlink to /etc/localtime from /usr/share/zoneinfo/America/New_York`
`16:16.29`	`DataCompBoy`	`Strom_M: http://pastebin.com/d4b4aea26`
`16:17.11`	`*** join/#asterisk ManxPower (n=manxpowe@122.sub-75-200-141.myvzw.com)`
`16:17.24`	`Strom_M`	`DataCompBoy: try changing pridialplan to "national"`
`16:17.38`	`krondorl`	`Corydon76-vcch: I don't seem to remember any of the docs I read saying anything about a symlink..`
`16:17.39`	`Strom_M`	`or "international"`
`16:17.52`	`ManxPower`	`Revenge is sweet! http://www.networkworld.com/community/node/20981`
`16:18.02`	`Corydon76-vcch`	`krondorl: that would be your distro at fault, then`
`16:18.29`	`Strom_M`	`DataCompBoy: if that doesnt work, try the same with prilocaldialplan`
`16:18.58`	`DataCompBoy`	`Strom_M: have tried, that change only Called Number, not Calling Number.`
`16:19.35`	`DataCompBoy`	`Strom_M: Called Number are fine. but I need to set up Calling Number, so user see mine CID...`
`16:20.10`	`Strom_M`	`DataCompBoy: you tried both pridialplan and prilocaldialplan?`
`16:20.11`	`ManxPower`	`DataCompBoy: what country are you in?`
`16:20.21`	`DataCompBoy`	`Strom_M: yes.`
`16:20.38`	`DataCompBoy`	`ManxPower: server in germany, 2*E1, provider Deutsche Telekom`
`16:20.52`	`*** join/#asterisk Arc^^ (n=Arc_@a82-95-179-89.adsl.xs4all.nl)`
`16:20.53`	`ManxPower`	`DataCompBoy: Ah!`
`16:21.09`	`Arc^^`	`Hi guys, anybody know how i can have asterisk ignore L2 Down on group-out call mISDN?`
`16:21.18`	`Strom_M`	`DataCompBoy: i assume these are two ISDN PRI circuits, correct?`
`16:21.43`	`Arc^^`	`I'm using mysql based config so i can't easily add the misdn_check_l2l1 command`
`16:22.08`	`DataCompBoy`	`Strom_M: yes, two ISDN PRI E1 lines, into one TE card`
`16:22.34`	`ManxPower`	`DataCompBoy: What is the problem?`
`16:22.58`	`DataCompBoy`	`ManxPower: http://pastebin.ru/227267 -- here is mine problem... that not code :) there at top -- cut from config, at bottom - list from debug`
`16:23.10`	`DataCompBoy`	`ManxPower: asterisk doesn't send Calling Number.`
`16:23.18`	`*** join/#asterisk jm\|laptop (n=jm\|home@dilbert.jamiem.com)`
`16:23.21`	`DataCompBoy`	`Strom_M: TE210P, from digium.`
`16:23.59`	`Strom_M`	`DataCompBoy: have you tried setting the callerid in extensions.conf before executing Dial()?`
`16:24.42`	`DataCompBoy`	`Strom_M: yes. and have tried before Dial, have drived different way of writting callerid in zapata.conf, etc. no result :(`
`16:25.09`	`bkruse`	`wow`
`16:25.12`	`bkruse`	`with a digital card?`
`16:25.14`	`bkruse`	`thats hard to believe.`
`16:25.22`	`Strom_M`	`show me what you're trying in extensions.conf`
`16:25.30`	`bkruse`	`Strom_M: agreed`
`16:25.37`	`Strom_M`	`bkruse: it's Deutsche Telekom`
`16:25.47`	`Strom_M`	`which I have 0 experience with`
`16:25.53`	`bkruse`	`same same`
`16:25.54`	`bkruse`	`Strom_M: I bet he is sending it, but its just getting ignored`
`16:25.55`	`*** join/#asterisk AsTeRiSk_1 (n=A@190.80.139.29)`
`16:26.04`	`AsTeRiSk_1`	`hello, some body can help ?`
`16:26.12`	`bkruse`	`DataCompBoy: you have your loadzone and tonezone setup for your land?`
`16:26.17`	`AsTeRiSk_1`	`i nee make some question about T1 Framing.`
`16:26.19`	`bkruse`	`AsTeRiSk_1: do not ask to ask, ask`
`16:26.37`	`DataCompBoy`	`bkruse: http://pastebin.com/d4b4aea26 -- full config`
`16:26.50`	`Strom_M`	`AsTeRiSk_1: what's your question?`
`16:26.58`	`AsTeRiSk_1`	`ok`
`16:27.03`	`AsTeRiSk_1`	`tnks for the help`
`16:27.08`	`AsTeRiSk_1`	`this is the question`
`16:27.08`	`sakic`	`anyone know an asterisk guy in the Charlotte area?`
`16:27.09`	`DataCompBoy`	`Strom_M: for now -- only have Dial(), can't touch alive system to re-test.`
`16:27.27`	`DataCompBoy`	`Strom_M: asterisk 1.4, was 1.2 -- same trouble`
`16:27.31`	`Strom_M`	`DataCompBoy: you can reload the extensions.conf without restarting asterisk, you know`
`16:28.33`	`*** join/#asterisk saftsack (n=saftsack@pD9E05FEC.dip.t-dialin.net)`
`16:28.37`	`AsTeRiSk_1`	`I have a T1 E&_W/Ami/SF in the side of my telco company i can use in the asterisk D4? because Asterisk no support SF only ESF`
`16:29.07`	`Strom_M`	`AsTeRiSk_1: asterisk supports ESF and D4/SF`
`16:29.20`	`Strom_M`	`SF and D4 are one and the same`
`16:29.59`	`DataCompBoy`	`Strom_M: exten => 1,1,Dial(ZAP/r1/${number},${tio},j). ok, moment, will try to add Set(CALLERID(num)="02212601906"), moment...`
`16:30.35`	`agx`	`wich is the best way to check a null/empty variable with 1.4? ISNULL ?`
`16:30.41`	`Strom_M`	`DataCompBoy: try without the quotes`
`16:31.34`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`16:31.47`	`AsTeRiSk_1`	`ok, i have this configuration. but the asterisk no answer the incoming calls Strom_M: you can see the full erro log in this page http://dpaste.com/23341/`
`16:32.03`	`AsTeRiSk_1`	`again tnks for your help and time`
`16:32.48`	`DataCompBoy`	`Strom_M: at least, it doesn't broke anything...`
`16:33.29`	`ManxPower`	`DataCompBoy: have you confirmed this via pri debug, or are you assuming asterisk is not sending the calling number based on the fact it does not show up on the destination phone?`
`16:33.50`	`*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)`
`16:33.56`	`ManxPower`	`DataCompBoy: normally the leading 0 is not part of the callerid info`
`16:34.02`	`DataCompBoy`	`ManxPower: http://pastebin.ru/227267 -- see here.`
`16:34.34`	`*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)`
`16:34.54`	`DataCompBoy`	`ManxPower: i see in pri debug that no CID sent :(`
`16:37.02`	HarryR`Work	`Any londoners here?`
`16:37.09`	`DataCompBoy`	`ManxPower, Strom_M: ask to dial to known number... difficult to see pri debug now (18 active channels)`
`16:37.32`	`*** join/#asterisk tripps (n=ss@ASA-ParksLuttrell.phonoscope.com)`
`16:37.34`	HarryR`Work	`That can do PHP and need contract/permanent work?`
`16:37.59`	`DataCompBoy`	`ManxPower, Strom_M: Set(CALLERID(num)=02212601906), on phone seen "AVS 03", no Calling Number in pri debug.`
`16:38.01`	`coppice`	`londoner, or someone in london?`
`16:38.08`	HarryR`Work	`someone in London`
`16:38.34`	HarryR`Work	`coppice, unless you're willing to commute :D`
`16:39.06`	`*** join/#asterisk Snake-eyes (n=hgjg@70.55.220.203.static.comindico.com.au)`
`16:39.08`	`ManxPower`	`DataCompBoy: try setting your prodialplan=unknown and remove that silly leading 0 from the CLID number.`
`16:39.09`	`DataCompBoy`	HarryR`Work: I'm working from russia with ppl from Hastings. Can I help you? :)
`16:39.23`	`*** join/#asterisk dijungal (n=kdaniel@63.175.159.171)`
`16:39.26`	`*** part/#asterisk dijungal (n=kdaniel@63.175.159.171)`
`16:39.29`	`coppice`	`its more than 12 hours to London, so true commuting would seem impossible.... unless the EU succeeds with that new plane`
`16:39.34`	`awk`	`good day`
`16:39.35`	`*** join/#asterisk dijungal (n=kdaniel@63.175.159.171)`
`16:39.36`	HarryR`Work	`:D`
`16:39.37`	`awk`	`what am I missing /usr/bin/ld: cannot find -lidn`
`16:39.44`	`Qwell`	`coppice: 12 hours there, 12 back.. just work from the car`
`16:39.46`	`awk`	`I have openssh and openssh devel?`
`16:39.49`	`DataCompBoy`	`ManxPower: for now both, pridialplan and prilocaldialplan are unknown.`
`16:40.00`	`Qwell`	`erm, plane`
`16:40.02`	`dijungal`	`hello, i am trying to understand the local/ channel construct.. can someone explain it to me?`
`16:40.09`	`dan__t`	`This sucks, I still feel hungover from this past weekend.`
`16:40.13`	`DataCompBoy`	`ManxPower: removed 0. everything same. (AVS 04, no Calling Number).`
`16:40.21`	`dijungal`	`chan_local`
`16:40.22`	`DataCompBoy`	`I'm in panic :(`
`16:40.24`	`coppice`	`Qwell: I said its more than 12 hours. >12 hours == non-commute distance`
`16:40.33`	`Qwell`	`excuses`
`16:40.37`	`ManxPower`	`dijungal: Local/ is similar to a goto`
`16:40.46`	`dijungal`	`oooh`
`16:40.57`	`coppice`	`unless they build hypersonic planes`
`16:40.59`	`dijungal`	`goto allows u to jump to a extension/context`
`16:41.36`	`dijungal`	`i'm reading up on create .call files for autodial out, and it's suggested that the local channel be used for dialing... but i'm wondering why`
`16:41.45`	`AsTeRiSk_1`	`i need help, asterisk is having problem in the incoming calls :x`
`16:41.49`	`dijungal`	`why can't i just say... zap/g2`
`16:42.04`	`dijungal`	`Asterisk_1 .. what's the problem?`
`16:42.26`	`coppice`	`a radio station in london looking for the longest commute found someone in the Midlands passing through London to Paris, and a lawyer from London commuting to Tel Aviv. Anything up to 12 hours seems fair game for a commute`
`16:42.29`	`AsTeRiSk_1`	`please check the log in this page http://dpaste.com/23341/`
`16:43.20`	`ManxPower`	`DataCompBoy: You understand that callerid=123456 does NOT set the NUMBER, right? That sets the NAME.`
`16:43.34`	`ManxPower`	`callerid=<123456> sets the NUMBER`
`16:44.06`	`DataCompBoy`	`ManxPower: I have tried ;callerid="Expert-Call.com"<00492601906> -- same thing`
`16:44.28`	`ManxPower`	`DataCompBoy: I assume without the leading ;`
`16:44.40`	`tzafrir`	`dijungal, who says you can't ? Could you be more specific?`
`16:44.42`	`dijungal`	`manxpower why the need for the opening and closing "<>"`
`16:44.43`	`ManxPower`	`also you still have leading 00 on that.`
`16:44.57`	`DataCompBoy`	`ManxPower: i have commented now, as one of testing variat... forget about zapata.conf (that i can't reload now), why Set(CALLERID(num)) doesn't work?`
`16:45.04`	`ManxPower`	`dijungal: because without the <> asterisk thinks you are setting the name.`
`16:45.07`	`dijungal`	`i'm just trying to figure out what's the real point of the LOCAL channel`
`16:45.27`	`dijungal`	`manxpower ohooo..`
`16:45.32`	`*** join/#asterisk axscode (n=axscode@103.211.111.218.klj04-home.tm.net.my)`
`16:45.33`	`DataCompBoy`	`ManxPower: or i MUST set BOTH ? num and name?`
`16:45.48`	`axscode`	`whats the best codec for VIDEO? small but good quality that can be used on ASTERISK? thank you!`
`16:45.50`	`ManxPower`	`DataCompBoy: I'm sure it is working. put in a Noop(CALLERID(num) is ${CALLERID(num)}) right before and after the Dial line, that will show you on the console.`
`16:46.02`	`ManxPower`	`DataCompBoy: setting the name will do nothing for PSTN calls.`
`16:46.06`	`*** join/#asterisk PepOSX (n=pepOSX@190.72.148.244)`
`16:47.12`	`AsTeRiSk_1`	`dijungal you check the log ?`
`16:47.34`	`ManxPower`	`so, set the CLID number, do the Noop, do the Dial, do the Noop.`
`16:47.39`	`DataCompBoy`	`ManxPower: -- Executing [2@test:1] NoOp("Zap/58-1", "CALLERID(num) is 2132136960") in new stack`
`16:47.46`	`DataCompBoy`	`ManxPower: -- Executing [2@test:2] Set("Zap/58-1", "CALLERID(num)=2212601906") in new stack`
`16:47.53`	`DataCompBoy`	`ManxPower: -- Executing [2@test:3] NoOp("Zap/58-1", "CALLERID(num) is now 2212601906") in new stack`
`16:47.59`	`DataCompBoy`	`ManxPower: and same result :(`
`16:48.08`	`ManxPower`	`I do NOT see the DIAL in there.`
`16:48.21`	`DataCompBoy`	`ManxPower: -- Executing [2@test:4] Dial("Zap/58-1", "ZAP/r1/021321369622\|100\|j") in new stack -- of course:)`
`16:48.42`	`*** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)`
`16:49.33`	`ManxPower`	`DataCompBoy: is 2212601906 a number assigned to your PRI?`
`16:50.27`	`ManxPower`	`DataCompBoy: now do you get the same "presentation: Number not available (67)" message?`
`16:51.04`	`*** part/#asterisk supjigator (n=sysgod@152.53.16.10)`
`16:51.46`	`ManxPower`	`DataCompBoy: what is the switchtype= set to?`
`16:52.09`	`DataCompBoy`	`ManxPower: euroisdn, yes, Number not available (67).`
`16:52.21`	`dijungal`	`AsTeRiSk_1: no`
`16:52.39`	`DataCompBoy`	`ManxPower: span 1 have 2212601906, span 2 have 221260122`
`16:53.36`	`AsTeRiSk_1`	`dijungal ok, if you look i have the error in this page http://dpaste.com/23341/ of th error`
`16:53.54`	`AsTeRiSk_1`	`the only problem i have in the asterisk "incoming Calls"`
`16:54.24`	`keith4`	`if I start with a single T1, how hard is it to add another and bond them at a later time?`
`16:54.42`	`ManxPower`	`DataCompBoy: try putting a SetCallerPres(allowed) before the Dial line`
`16:55.05`	`coppice`	`keith4: depends where you live`
`16:55.09`	`dijungal`	`AsTeRiSk_1: thats allot of different errors`
`16:55.10`	`DataCompBoy`	`ManxPower: usecallingpres=yes in conf not same?`
`16:55.45`	`keith4`	`coppice: why is that the answer to so many questions? ;-)`
`16:56.18`	`ManxPower`	`DataCompBoy: actually usecallingpres=yes might be CAUSING your problem.`
`16:56.40`	`ManxPower`	`In fact I do NOT see that option in the pastebin you gave.`
`16:56.56`	`DataCompBoy`	`ManxPower: http://pastebin.com/d4b4aea26 -- here full config.`
`16:57.21`	`DataCompBoy`	`ManxPower: added SetCallerPres(allowed) -- no effect.`
`16:57.24`	`ManxPower`	`try my suggestions`
`16:57.25`	`AsTeRiSk_1`	`dijungal this error are from incoming calls :s`
`16:59.32`	`dijungal`	`hmm`
`17:00.14`	`dijungal`	`AsTeRiSk_1: what channel is the call coming in on?`
`17:00.48`	`dijungal`	`AsTeRiSk_1: u may want to paste the context in your extensions.conf where the call is being places into`
`17:01.09`	`AsTeRiSk_1`	`dijungal ok i change some config and i have this new error chan_zap.c: zt hook failed: Device or resource busy`
`17:01.20`	`AsTeRiSk_1`	`chan_zap.c: Exception on 21, channel 4`
`17:01.38`	`DataCompBoy`	`ManxPower: i can't edit zapata.conf right now -- that needs asterisk restart, but i have 38 active calls. SetCallerPres(allowed) doesn't helps.`
`17:01.43`	`dijungal`	`are you using a Te1XXP card?`
`17:01.51`	`AsTeRiSk_1`	`TE210P`
`17:02.00`	`*** join/#asterisk jmls (n=jmls@62.49.235.130)`
`17:02.07`	`AsTeRiSk_1`	`Card TE210P`
`17:02.07`	`*** join/#asterisk PDani (n=pdani@mail.istennotemplom.com)`
`17:02.08`	`PDani`	`hi`
`17:02.12`	`Xenon3DN`	`Anybody have any insight on my silence-when-coming-off-hold problem? http://pastebin.com/d51c9e2`
`17:02.20`	`dijungal`	`the phone is off hook?`
`17:02.21`	`jmls`	`what is the format for STRFTIME to show milliseconds ?`
`17:02.27`	`jmls`	`I can get seconds ..`
`17:02.29`	`PDani`	`how can i enable sending text messages through an asterisk with SIP?`
`17:03.07`	`PDani`	`is this require some extra configuration, or is it enaabled by default?`
`17:03.17`	`AsTeRiSk_1`	`dijungal i have T1 in the port one of the T201P`
`17:03.21`	`AsTeRiSk_1`	`Sorry T210P`
`17:03.25`	`ManxPower`	`DataCompBoy: try this one http://pastebin.com/m76b24c28 then stop then start Asterisk`
`17:03.47`	`AsTeRiSk_1`	`24 Channels E&M_W T1 Signalling /AMI/D4`
`17:03.54`	`[TK]D-Fender`	`PDani: * doesn't support SIP IM`
`17:03.57`	`ManxPower`	`DataCompBoy: it is best to work on problems when the system is not busy with calls`
`17:04.24`	`DataCompBoy`	`ManxPower: heh... 36 active calls. :( i just can't do anything with zapata conf right now.`
`17:04.54`	`DataCompBoy`	`ManxPower: you mean put config with such, minimalistic way, leaveing everything other on defaults?`
`17:06.08`	`agx`	`is there anybody outside that has some extra apps for * 1.4 (apart from the package asterisk-addons) ??`
`17:06.30`	`dijungal`	`hmmm... it's not seeing the hooks, on or off`
`17:06.37`	`*** part/#asterisk dijungal (n=kdaniel@63.175.159.171)`
`17:07.00`	`k31th`	`anyone got a working sip.conf example for a linksys spa94*`
`17:07.19`	`Xenon3DN`	`I've been trying to Google for an answer to my question, but "asterisk hold silence" just comes up with a bunch of people having trouble getting music-on-hold right. Any suggestions for a better search query string?`
`17:07.32`	`AsTeRiSk_1`	`dijungal look i have this new error : zt hook failed: Device or resource busy`
`17:07.55`	`AsTeRiSk_1`	`Took Zap/6-1 off hook`
`17:08.10`	`AsTeRiSk_1`	`<PROTECTED>`
`17:08.23`	`AsTeRiSk_1`	`Got wink in weird state 6 on channel 6`
`17:08.36`	`AsTeRiSk_1`	`chan_zap.c: Ring/Off-hook in strange state 6 on channel 6`
`17:09.58`	`keith4`	`coppice: coppice as in "thicket"?`
`17:10.34`	`coppice`	`or possibly as in Underwood`
`17:10.53`	`Arc^^`	`anyone know where i can hook misdn_check_l2l1 in a dial rule when i use a mysql based asterisk config`
`17:11.11`	`Alan_Hicks`	`According to the book I'm reading, if I want to dial multiple lines I can call them like this "Dial(Zap/1&Zap/2&SIP/alan)". Is there any shorter way of specifying this if say I want to ring 10 phones and bridge the call to the first one to answer? I'm thinking something like a variable? What's something like this called so I'll know what to read up on?`
`17:11.41`	`keith4`	`it's called a variable`
`17:11.57`	`PDani`	`[TK]D-Fender: i read about this presence patch + sip_message_support patch solution. is there somewhere a precompiled stuff from this, which is capable to forward text messages? (maybe, in a .deb? :)))`
`17:12.10`	`*** join/#asterisk trippsss (n=ss@ASA-ParksLuttrell.phonoscope.com)`
`17:12.18`	`Alan_Hicks`	`chuckles Figures. I just haven't seen it anywhere in the book yet.`
`17:12.19`	`Xenon3DN`	`Hey, something unrelated just occurred to me -- anyone here have any use for some free AT&T 8510T ISDN sets? I think I have 4 of them in the storage room as well as 1 or more ISDN sets by some other mfr, and I hate to through out well-made telco gear like that.`
`17:12.19`	`Arc^^`	`where do i p ut this: exten => _X.,1,misdn_check_l2l1(g:out\|2)`
`17:12.34`	`Xenon3DN`	`s/through/throw`
`17:13.30`	`keith4`	`Alan_Hicks: ALL_PHONES=Zap/1&Zap/2&SIP/alan and then you can say "Dial(${ALL_PHONES})"`
`17:14.11`	`Alan_Hicks`	`keith4: Thanks. I'm sure it's somewhere in the book but I probably glossed over it.`
`17:18.03`	`*** join/#asterisk tristezo2k (n=seba@200.117.247.43)`
`17:18.14`	`*** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus)`
`17:19.07`	`tristezo2k`	`Hi *.`
`17:19.08`	`tristezo2k`	`I am using asterisk for an itsp with some providers...`
`17:19.26`	`tristezo2k`	`I need that * can place outbound calls by a second provider`
`17:19.34`	`*** part/#asterisk Xenon3DN (n=Xenon@mail.3dnature.com)`
`17:19.40`	`tristezo2k`	`one that the first has completed their pool...`
`17:19.58`	`tristezo2k`	`Prov1 has 50 voip channels available`
`17:19.59`	`tristezo2k`	`Prov2 has many more`
`17:20.05`	`*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)`
`17:20.11`	`tristezo2k`	`but prov1 has to be used first...`
`17:20.22`	`tristezo2k`	`how can this be done in a elegant way? :-S`
`17:20.57`	`tristezo2k`	`Sounds like a group concept in behind of this..`
`17:21.35`	`*** join/#asterisk dwclarkNU (n=Administ@c-76-19-196-67.hsd1.ma.comcast.net)`
`17:22.11`	`dwclarkNU`	`i am getting an echo over my sip channel on inbound calls from certain people, anyway to diagnose why or enable something to stop it?`
`17:23.00`	`tristezo2k`	`dwclarkNU: is there pstn hardware in the middle of the call? or is it IP only?`
`17:23.51`	`dwclarkNU`	`tristezo2k: landline -> sip provider -> sip trunk -> asterisk -> ip phone`
`17:24.10`	`jmls`	`please, can someone show me what the format is for STRFTIME to show milliseconds ?`
`17:24.50`	`tristezo2k`	`chances are thata the echo is generated in the landline > sip provider part`
`17:24.56`	`*** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl)`
`17:25.07`	`dwclarkNU`	`any way to counteract it in asterisk?`
`17:25.12`	`tristezo2k`	`Did you tried to record at the asterisk to see if it already has the echo?`
`17:25.51`	`tristezo2k`	`the ip phone hears the echo or the landline?`
`17:25.57`	`dwclarkNU`	`the ip phones`
`17:26.08`	`dwclarkNU`	`the landline hears teh call perfectly clear`
`17:26.56`	`k31th`	`\o/ my phones registered`
`17:27.07`	`tristezo2k`	`try recording at the asterisk to see if it has echo.`
`17:27.31`	`tristezo2k`	`if it does not the echo is generated at the landline < sip provider part.`
`17:28.12`	`tristezo2k`	`when talking from the ip phone to landline the landline is not acopling`
`17:28.17`	`tristezo2k`	`(I guess)`
`17:28.20`	`dwclarkNU`	`and if that's the case, is there something i can do to reduce the echo?`
`17:28.28`	`*** join/#asterisk one47 (n=one47@dyn-168.ipcortex.bp.ipcortex.net)`
`17:28.42`	`tristezo2k`	`lower your sending volume in steps of 3db...`
`17:28.46`	`tristezo2k`	`or so.`
`17:29.08`	`tristezo2k`	`Does it allways have echo?`
`17:29.13`	`tristezo2k`	`or with some special numbers?`
`17:29.14`	`dwclarkNU`	`no`
`17:29.17`	`dwclarkNU`	`special numbers`
`17:29.17`	`one47`	`hi, anybody here who can reopen a mantis bug for me so I can add a proper resolution to it?`
`17:29.32`	`*** join/#asterisk Falkon_MX-5 (n=Falkon@c-76-29-226-148.hsd1.al.comcast.net)`
`17:29.44`	`tristezo2k`	`you see.. the problem is at the sip provider, they usually have many different providers for themselves...`
`17:29.46`	`Falkon_MX-5`	`Anyone here use a Polycom 500/501?`
`17:29.47`	`jmls`	`one47: which one ?`
`17:29.48`	`*** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net)`
`17:30.08`	`one47`	`bug#10052 - It also needs * version changing to 1.4.12`
`17:30.13`	`tristezo2k`	`one of them is the problem`
`17:30.47`	`jmls`	`so you've reproduced it in 1.4 ?`
`17:31.03`	`dwclarkNU`	`i will contact my provider and find out`
`17:31.14`	`jmls`	`one47: ping ..`
`17:31.33`	`dwclarkNU`	`should i still reduce my sendng volume?`
`17:31.47`	`one47`	`yes - reproduced on a 1.4.x box that no-longer exists, fixed on a production 1.2.24 box, but the logic is faulty in 1.4.12 also`
`17:32.02`	`one47`	`one line logic fault fixes it :)`
`17:33.01`	`jmls`	`one47: done`
`17:33.17`	`one47`	`thanks. I'll upload patch(es) and an explanation ASAP.`
`17:34.12`	`stimpie`	`I have openser in front of asterisk and defined a peer in sip.conf, issue is that I have one-way audio`
`17:34.24`	`dwclarkNU`	`how would one go about reducing the sending volume?`
`17:34.27`	`stimpie`	`running a trace shows no rtp packets being send from asterisk`
`17:34.42`	`stimpie`	`any pointers on what could be wrong?`
`17:35.03`	`*** join/#asterisk wunderkin (n=wunderki@ip72-208-16-104.ph.ph.cox.net)`
`17:36.13`	`tristezo2k`	`dwclarkNU: that can help, but if it is ok with the others, keep it that way, and call your provider..`
`17:36.48`	`*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)`
`17:37.33`	`tristezo2k`	`Hi *.`
`17:37.33`	`tristezo2k`	`I am using asterisk for an itsp with some providers...`
`17:37.33`	`tristezo2k`	`I need that * can place outbound calls by a second provider`
`17:37.38`	`tristezo2k`	`one that the first has completed their pool...`
`17:37.38`	`tristezo2k`	`Prov1 has 50 voip channels available`
`17:37.38`	`tristezo2k`	`Prov2 has many more`
`17:37.48`	`tristezo2k`	`but prov1 has to be used first...`
`17:37.49`	`tristezo2k`	`how can this be done in a elegant way? :-S`
`17:37.49`	`tristezo2k`	`Sounds like a group concept in behind of this..`
`17:38.56`	`*** join/#asterisk shinao1 (n=shinao1@217.20.242.50)`
`17:39.45`	`dwclarkNU`	`tristezo2k: i use freepbx to manage my routes and trunks, on my outbound route i can set the order of the trunks I want to use and then on the trunk config, i can specify the number of channels it has, when they are all full, it will rollover to the next trunk`
`17:41.04`	`one47`	`jmls: Thanks, notes and patches attached to bug#10052`
`17:42.08`	`*** join/#asterisk eBody (n=icechat5@207.71.51.162)`
`17:42.16`	`tristezo2k`	`dwclarkNU: thanks!, I will look at freepbx and how it does that :D`
`17:42.33`	`eBody`	`<PROTECTED>`
`17:42.44`	`eBody`	`what is that about?? we have a PRI`
`17:43.46`	`*** part/#asterisk one47 (n=one47@dyn-168.ipcortex.bp.ipcortex.net)`
`17:44.24`	`dwclarkNU`	`tristezo2k: suppose my provider tells me i'm SOL, is it possible to fight the echo inside asterisk?`
`17:47.03`	`*** join/#asterisk arekm (i=arekm@pld-linux/arekm)`
`17:48.26`	`arekm`	`now I wonder how to setup such sip`
`17:48.37`	`blitzrage`	`dwclarkNU: there is HPEC to do echo canceling in software`
`17:49.50`	`[TK]D-Fender`	`dwclarkNU: What phones are you using?`
`17:50.03`	`dwclarkNU`	`polycom ip330`
`17:50.35`	`*** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net)`
`17:50.36`	`arekm`	`type=peer? user?`
`17:50.51`	`docelmo`	`no =friend !!!!!!!!!!!!!1`
`17:50.52`	`dwclarkNU`	`friend`
`17:50.52`	`docelmo`	`:)`
`17:51.15`	`arekm`	`ok, friend for anonymous sip`
`17:51.31`	`[TK]D-Fender`	`dwclarkNU: Change providers. You'[re screwed with them.`
`17:51.56`	`docelmo`	`You would setup anonymous with the directive autocreatepeer or setup in general section of sip.conf to handle anonymous sip connections`
`17:52.05`	`dwclarkNU`	`b/c of the echo?`
`17:52.15`	`docelmo`	`dwclarkNU who do you use?`
`17:52.17`	`[TK]D-Fender`	`dwclarkNU: Yes. This is on THEIR side.`
`17:52.22`	`dwclarkNU`	`iCall`
`17:53.27`	`docelmo`	`eww they are using dundi or enum and dont control their end points.. Well I guess you get what you dont pay for..`
`17:54.00`	`dwclarkNU`	`how do you know that?`
`17:54.21`	`docelmo`	`Ive been around the block a few times when it comes to ITSP's..`
`17:54.42`	`dwclarkNU`	`do you have a suggestion?`
`17:54.42`	`docelmo`	`Im not a novice in anyway shape or form`
`17:54.43`	`*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`17:54.43`	`*** mode/#asterisk [+o blitzrage] by ChanServ`
`17:55.00`	`*** join/#asterisk Xenon3DN (n=Xenon@mail.3dnature.com)`
`17:55.02`	`dwclarkNU`	`i wasn't saying you were, i just didnt know if i had overlooked it on their site`
`17:55.15`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`17:55.20`	`*** join/#asterisk jsaunders (n=nevermin@70.70.0.33)`
`17:55.37`	`Katty`	`what's IVR?`
`17:55.49`	`dwclarkNU`	`interactive voice response`
`17:55.54`	`Xenon3DN`	`Would someone here mind posting a question to the asterisk-users mailing list for me? I'm subscribed and can read the messages, but every time I post the messages goes silently into a black hole.`
`17:56.04`	`docelmo`	`Its not on their site.. But since they are 100% free.. You have to asume a couple of things.. They are either charing for an ass load of ads.. Which I dont see any.. Have someone giving them money for shits and giggles.. I doubt.. or they use Enum or Dundi which would allow them to use a free network to send and recieve calls`
`17:56.19`	`dwclarkNU`	`http://en.wikipedia.org/wiki/Interactive_voice_response`
`17:56.27`	`docelmo`	`Katty come on.. You know better than that..`
`17:56.32`	`docelmo`	`MEW MEW MEW MEW!`
`17:56.42`	`Katty`	`docelmo: you're right. i asked my question wrong.`
`17:57.22`	`docelmo`	`Katty == Senior Overworked IT Chic..`
`17:57.29`	`docelmo`	`:P`
`17:58.41`	`dwclarkNU`	`docelmo: they're not free... http://carriers.icall.com`
`17:59.19`	`arekm`	`oh, something starts working. How can I make anonymous sip calls (to my pbx) go into some sip.conf section and not generic?`
`18:00.10`	`arekm`	`I have http://pastebin.com/m2bb8a02e but anon sip calls still go as [generic]`
`18:00.11`	`[TK]D-Fender`	`arekm: How is anonymous NOT generic?`
`18:00.13`	`docelmo`	`For an everyday use they appear to be.. their commercial or wholesale side might be where they are making their money to cover the free costs elsewhere`
`18:00.43`	`[TK]D-Fender`	`arekm: Pastebin the CALL with SIP debug and verbose 10 and maybe we'll see.`
`18:01.05`	`docelmo`	`arekm the only thing you can do in the general section is set your context=anonincoming or something to direct all of those calls to that conect in extensions.conf.`
`18:01.11`	`docelmo`	`Else they would need to register`
`18:01.42`	`DataCompBoy`	`URAAAAAAAAAAAAAAAAAAAAAAA`
`18:01.59`	`docelmo`	`[TK]D-Fender what do you do for work? Every time I see you in here its like you never work.. Does someone pay you to sit in IRC and talk all day long?`
`18:02.04`	`docelmo`	`:) If so where do I sign up?`
`18:02.05`	`DataCompBoy`	`ManxPower, Strom_M: Have fixed! SetCallerPres(allowed_not_screened) -- helps me`
`18:02.08`	`arekm`	`[TK]D-Fender: http://pastebin.com/d2043cf36`
`18:02.23`	`*** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)`
`18:02.37`	`[TK]D-Fender`	`docelmo: Ask not what I do for work ask what work does for me!`
`18:03.00`	`arekm`	`actually generic could be fine for me if it is never used for other types of SIP traffic than anonymous one`
`18:03.16`	`docelmo`	`arekm what is defined as context under general in sip.conf?`
`18:03.18`	`arekm`	`I'm trying to make sure that anonymous is handled with different way than other sip traffic`
`18:03.35`	`docelmo`	`[TK]D-Fender funny.. I just landed a new job I am excited about.`
`18:03.42`	`arekm`	`docelmo: "from-inotel"`
`18:03.54`	`[TK]D-Fender`	`arekm: Ah! "allowguest=yes" belongs under [general] and you had BETTER not dream of trying to call an ENTRY [generic] or anything silly like that.`
`18:04.26`	`docelmo`	`then in your extensions.conf [from-inotel] needs to have some sort of pattern matching for what you are trying to accomplish`
`18:04.30`	`[TK]D-Fender`	`<PROTECTED>`
`18:05.10`	`[TK]D-Fender`	`arekm: So ditch that "section" go add your context under [general] and move allowguest=yes under there.`
`18:05.25`	`arekm`	`[TK]D-Fender: so to just make sure - [generic] is used _only_ for anonymous sip right?`
`18:05.48`	`docelmo`	`um, [general] handles anonymous calls`
`18:05.57`	`docelmo`	`[generic] would be a sip account in sip.conf`
`18:06.19`	`arekm`	`right, sorry, I was thinking about [general]`
`18:06.51`	`DataCompBoy`	`Many thanks to all! going out.`
`18:06.59`	`dwclarkNU`	`now i am having an echo for internal calls as well`
`18:06.59`	`docelmo`	`And general also sets default permissions and configuration settings for the sip stack as well as the other peers that dont set directives differently than [general]`
`18:07.04`	`stimpie`	`I posted a trace of my one-way audio problem: http://www.pastebin.ca/749419 if someone could have a look that would be great`
`18:07.08`	`[TK]D-Fender`	`arekm: NO. the IS NO GENIC NAMED SECTION.`
`18:07.22`	`[TK]D-Fender`	`asdasdaskl;dkl;jdakl;sdhhksfd`
`18:07.47`	`docelmo`	`dwclarkNU if you are experiencing echo on a local network you either are using shitty sip hardware or shitty networking gear`
`18:07.49`	`[TK]D-Fender`	`arekm: And yes if a call comes in without auth, it hits [general]`
`18:08.08`	`[TK]D-Fender`	`docelmo: How does shittiy networking gear creat echo? :)`
`18:08.17`	`docelmo`	`[TK]D-Fender haha..`
`18:08.27`	`[TK]D-Fender`	`docelmo: Delayed sound perhaps, but REPEAT? :)`
`18:08.39`	`docelmo`	`[TK]D-Fender gee I dont know.. Packet loss, colissions, etc..`
`18:08.47`	`arekm`	`[TK]D-Fender: but only such calls (with no auth) or such cals also (among some others)?`
`18:08.49`	`[TK]D-Fender`	`docelmo: I'd offer you some crack, but clearly your stuff if far superior`
`18:08.51`	`docelmo`	`[TK]D-Fender I had a bad cisco 1924 do it to me`
`18:09.15`	`[TK]D-Fender`	`arekm: All calls that don't match a user (some ITSP's send you calls that way too.`
`18:09.38`	`[TK]D-Fender`	`docelmo: COLLISIONS?! Stop using hubs silly!`
`18:09.53`	`docelmo`	`I havent used a hub is a couple days thank you.. :)`
`18:10.06`	`[TK]D-Fender`	`docelmo: And how can you have echo on packet loss... they never arrived!`
`18:10.21`	`[TK]D-Fender`	`docelmo: you = funneh`
`18:10.32`	`docelmo`	`ok true.. I just threw some ideas out there..`
`18:10.52`	`[TK]D-Fender`	`docelmo: I threw them out too :p`
`18:11.03`	`docelmo`	`haha.. anywho.. back to cleaning.. cya`
`18:11.04`	`*** join/#asterisk Quintana (n=sylvain@213.215.63.10)`
`18:11.34`	`peanut-`	`what all has to be set for xten client to work over NAT?`
`18:13.03`	`*** join/#asterisk unstable (n=unstable@tor/regular/sid)`
`18:13.43`	`arekm`	`[TK]D-Fender: so it's not possible to distinguish anonymous from wrong-user sip calls, right? (at sip.conf level)`
`18:14.37`	`*** join/#asterisk pkwong (n=chatzill@68.195.200.20)`
`18:16.36`	`[TK]D-Fender`	`peanut-: ...`
`18:16.36`	`*** join/#asterisk PepOSX (n=pepOSX@190.72.148.244)`
`18:16.38`	`[TK]D-Fender`	`~sipnat`
`18:16.38`	`jbot`	`[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions`
`18:16.49`	`arekm`	`hm, almost works`
`18:17.03`	`arekm`	`calling 77@my.asterisk.ip`
`18:17.19`	`arekm`	`but http://pastebin.com/mfa948f`
`18:18.42`	`arekm`	`daamn, typo, nevermind`
`18:21.05`	`docelmo`	`hmm.. I wouldnt roast anything above a gasoline fueled fire.. taste nasty..`
`18:21.08`	`Wonka`	`bad Katty!`
`18:21.08`	`*** join/#asterisk anonymouz666 (n=anonymou@201.19.178.16)`
`18:22.24`	`Katty`	`i wanna go work for a zoo :<`
`18:23.27`	`anonymouz666`	`lol`
`18:23.52`	`waKKu`	`i have some monkeys heere.. can i send its to u ? :P`
`18:24.01`	`outtolunc`	`someone stop her before she pours gasoline on the zoo animals next <G>`
`18:24.38`	`Xenon3DN`	`Flaming meerkats.`
`18:25.09`	`peanut-`	`orly.`
`18:25.28`	`*** join/#asterisk Chris-NB (n=chris@smtp137.tech.t-mobile.at)`
`18:25.38`	`Xenon3DN`	`Flaming owls?`
`18:28.10`	`Katty`	`whhhyyyyyyy me`
`18:28.20`	`Katty`	`[TK]D-Fender: make it work :<`
`18:34.20`	`[TK]D-Fender`	`Katty: Show me a problem and we'll see...`
`18:34.43`	`Katty`	`no i don't think i'll bother`
`18:34.50`	`Katty`	`cause you'll just give me some sarcastic comment`
`18:34.56`	`Katty`	`and presume i should already know the answer`
`18:35.05`	`Katty`	`then try to make me feel even more dumb after you talk down to me`
`18:35.26`	`Katty`	`also, did i mention i'm grumpy?`
`18:37.21`	`*** join/#asterisk J4k3 (n=jsuter@pimpin.aint.easy.in.grapeland.us)`
`18:37.52`	`Xenon3DN`	`http://en.wiktionary.org/wiki/catty`
`18:38.02`	`k31th`	`Hows it going katty`
`18:38.04`	`Xenon3DN`	`Not that I'm sayin' anything...`
`18:40.50`	`Katty`	`k31th: very frustrated.`
`18:40.53`	`Katty`	`[TK]D-Fender: sorry, i still love you!`
`18:42.40`	`Sweeper`	`http://ablk.com/c8.html <-- el oh el`
`18:43.25`	`[TK]D-Fender`	`Sweeper: JURASSIC`
`18:43.26`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`18:46.46`	`outtolunc`	`i like this one <G> http://ablk.com/cd.html`
`18:47.56`	`*** part/#asterisk dwclarkNU (n=Administ@c-76-19-196-67.hsd1.ma.comcast.net)`
`18:49.28`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`18:49.31`	`*** join/#asterisk PhiberOptik (n=PhiberOp@75.148.50.141)`
`18:52.07`	`*** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl)`
`18:53.29`	`PhiberOptik`	`I'm having a little trouble getting voice mail broadcast lists configured. I've been through the forums, with no luck. I'm 99 percent sure that I have the correct syntax for the extension in extensions.conf, and the correct definitions for the broadcast mailbox set. Can I get a few ideas?`
`18:53.33`	`*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)`
`18:59.54`	`Strom_M`	`If you specify a hostname rather than an IP address for the asterisk server in a polycom config file, does the polycom phone do one DNS lookup when the phone boots up, followed by periodic refreshes thereafter, or does the phone check DNS every time a call is placed?`
`19:02.51`	`*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)`
`19:03.54`	`[TK]D-Fender`	`PhiberOptik: Yeah I highly recommend you show us the CLI output of the failed attempt at verbose 10.`
`19:03.56`	`[TK]D-Fender`	`~pb`
`19:03.56`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`19:03.58`	`[TK]D-Fender`	`^^^^^^^^^^^^^^^^^^^^^^^^^^^^^`
`19:04.46`	`*** part/#asterisk Xenon3DN (n=Xenon@mail.3dnature.com)`
`19:05.39`	`*** join/#asterisk gandhijee (n=user@mail.win-ent.com)`
`19:06.02`	`gandhijee`	`hey, my asterisk box has 2 nics, with phones on both sides`
`19:06.38`	`gandhijee`	`and there is a wifi segment as well, when i try to dial from the wifi phones (192.168.2.x) to the ip phones (192.168.1.x) i get a pbx_extension_helper: Cannot find extension context '10.0.0.1'`
`19:06.40`	`gandhijee`	`any idea?`
`19:06.56`	`gandhijee`	`the 2 interfaces in my machine are 10.0.0.x and 192.168.1.97`
`19:07.02`	`Strom_M`	`do the wifi phones work for calls out to the universe?`
`19:07.30`	`gandhijee`	`Strom_M: i just switched over to asterisk from Yate`
`19:07.46`	`Strom_M`	`ok`
`19:07.48`	`Strom_M`	`do the wifi phones work for calls out to the universe?`
`19:07.51`	`Alan_Hicks`	`Hey guys. I'm trying to setup a provisioning server for my Polycom Soundpoint IP 320 phone, and I need the configuration files and bootrom to do this. I've tried to contact my reseller to obtain them, but I'm having trouble getting in touch with their support department right now. Does anyone know some place I could download the latest files?`
`19:07.52`	`gandhijee`	`they were able to call the ip phones with that software no problems`
`19:08.19`	`gandhijee`	`i haven't setup outbound calling yet`
`19:08.27`	`gandhijee`	`i can have my ip phones call the wifi phones`
`19:08.36`	`Strom_M`	`gandhijee: pastebin your sip.conf`
`19:08.42`	`Strom_M`	`~pb`
`19:08.42`	`jbot`	`A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste`
`19:09.07`	`Katty`	`file: save me.`
`19:09.14`	`Katty`	`also!`
`19:09.21`	`Strom_M`	`Save, Save As...`
`19:09.22`	`Katty`	`is anyone going to zombie raid 2007 on maulstrom?`
`19:09.32`	`Strom_M`	`wait, who's mauling strom?`
`19:09.34`	`arekm`	`<PROTECTED>`
`19:09.43`	`Katty`	`Strom_M: nono, silly, it's a warcraft server.`
`19:09.48`	`arekm`	`should I double-escape \ for bind?`
`19:09.52`	`hmmhesays`	`warcraft eh?`
`19:09.55`	`Katty`	`Strom_M: 1000 people have RSVPed.. we're making lvl 1 alts..`
`19:09.58`	`Katty`	`Strom_M: and inviading a city`
`19:10.10`	`gandhijee`	`http://pastebin.ca/749520`
`19:10.15`	`Strom_M`	`Katty: I have less than zero clue what you're talking about`
`19:11.19`	`Strom_M`	`gandhijee: ok, and now extensions.conf`
`19:11.26`	`gandhijee`	`k`
`19:12.15`	`gandhijee`	`http://pastebin.ca/749525`
`19:12.26`	`BBHoss`	`could be a bug`
`19:12.32`	`BBHoss`	`what version * are you suing`
`19:12.34`	`BBHoss`	`using`
`19:13.04`	`gandhijee`	`BBHoss: me??`
`19:13.10`	`BBHoss`	`yes`
`19:13.33`	`gandhijee`	`1.2.24`
`19:13.56`	`BBHoss`	`ahh`
`19:14.04`	`gandhijee`	`?`
`19:14.15`	`BBHoss`	`it very well could be a bug then`
`19:14.27`	`stimpie`	`Iam still having an issue on calls which are received trough my openser proxy have one-way audio (from ua to asterisk)`
`19:14.35`	`BBHoss`	`there are a number of bugs related to bindaddr=`
`19:14.42`	`BBHoss`	`google: asterisk bindaddr`
`19:15.06`	`BBHoss`	`Alan_Hicks: did you ever find the bootrom and applciation files?`
`19:16.08`	`*** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net)`
`19:16.10`	`*** join/#asterisk eBody (n=icechat5@207.71.51.162)`
`19:16.17`	`Alan_Hicks`	`BBHoss: Not yet. I'm gonna call telephonydepot back in 15 minutes or so. Maybe they won't be as busy then?`
`19:16.28`	`BBHoss`	`http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip601.html#download`
`19:16.36`	`eBody`	`anyone know how to remedy this error from a PRI: Dial failed due to CONGESTION`
`19:16.37`	`BBHoss`	`those are the release before last`
`19:16.42`	`*** join/#asterisk bantu (n=Miranda@p54A32A0F.dip0.t-ipconnect.de)`
`19:16.59`	`gandhijee`	`you guys have the new 601 boot roms?`
`19:17.01`	`Alan_Hicks`	`Oh thanks. I read that Polycom doesn't host those files for download. Maybe they changed practice?`
`19:17.08`	`BBHoss`	`unless there is something you require of the newest`
`19:17.36`	`Alan_Hicks`	`No, I'm just still learning, and I need to figure out how to edit the phones dial mask (or is it digit mask?) and stuff like that.`
`19:18.25`	`Alan_Hicks`	`Example: after entering ten digits the phone attempt to call the number entered, but that won't match _91NXXNXXXXXX.`
`19:18.33`	`BBHoss`	`heh`
`19:18.56`	`*** join/#asterisk CrashSys (n=kumba@216-199-37-76.tpa.fdn.com)`
`19:18.58`	`*** part/#asterisk tristezo2k (n=seba@200.117.247.43)`
`19:18.59`	`peanut-`	`I have a neat issue. I have a SIP client behind NAT, using STUN and nat=yes in sip.conf, after 20 seconds of talk time, every time, it disconnects with the message - [Oct 25 14:16:31] WARNING[2474]: chan_sip.c:1938 retrans_pkt: Maximum retries exceeded on transmission Zjk1ZjNlMDBlZWJkOTM3OTA1MjcwMGY5ODBjOThiMTI. for seqno 2 (Critical Response)`
`19:19.05`	`BBHoss`	`also, looks likes the bootroms aren't on the site`
`19:19.12`	`Alan_Hicks`	`I do need to setup a provisioning server to fix that problem, right?`
`19:19.33`	`BBHoss`	`what the dial string?`
`19:19.45`	`Alan_Hicks`	`Yeah.`
`19:20.04`	`BBHoss`	`im not sure, the web interface is horrid though`
`19:20.27`	`Alan_Hicks`	`Say I want to dial 14787849500 (my office number), and I need to dial 9 to get out. After entering 9147878495 the phone dials.`
`19:20.35`	`Alan_Hicks`	`Yeah, the web interface sucks.`
`19:20.41`	`J4k3`	`a phone for three times the cost is ten times as goofy.`
`19:20.47`	`*** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com)`
`19:20.48`	`CrashSys`	`Does anyone know if Flash Operator Panel has a forum/irc-channel somewhere?`
`19:20.52`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`19:21.20`	`BBHoss`	`heres another mirror`
`19:21.21`	`BBHoss`	`http://www.freedomphones.net/polycom/files/`
`19:21.50`	`ai-a`	`We got asterisk installed and we're getting echos on our side (inside the the building) when dialing landline (BT) phones.. mobiles seem ok.`
`19:22.04`	`BBHoss`	`ai-a what card are you using`
`19:22.10`	`ai-a`	`Sangoma.`
`19:22.36`	`BBHoss`	`try sangoma support`
`19:23.01`	`ai-a`	`;) could do.`
`19:23.13`	`BBHoss`	`alan_Hicks - this site has the latest bootrom/SIP images`
`19:23.14`	`BBHoss`	`http://www.polycomnz.co.nz/software.html`
`19:23.33`	`Alan_Hicks`	`Thanks.`
`19:23.43`	`BBHoss`	`good old new zealand`
`19:23.45`	`Alan_Hicks`	`Now to figure out exactly what I need, and get started. Obliged.`
`19:23.48`	`hmmhesays`	`I just put in a a200 sangoma card`
`19:23.50`	`hmmhesays`	`this thing rocks`
`19:23.52`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`19:24.03`	`BBHoss`	`bbiab`
`19:24.05`	`Alan_Hicks`	`They must have a law that says you gotta make that stuff publicly available or something.`
`19:24.15`	`BBHoss`	`heh NO`
`19:24.30`	`J4k3`	`the trick is`
`19:24.38`	`J4k3`	`folks from .nz won't dick with shithead manufacturers`
`19:24.46`	`Alan_Hicks`	`hehe`
`19:24.49`	`J4k3`	`but north americans and europeans enjoy being shafted and fucked with`
`19:24.54`	`J4k3`	`its a central part of the marketing plan`
`19:25.50`	`Alan_Hicks`	`Yeah, my phone's got bootrom 3.2.3. I reckon it won't hurt to go to 4.0.0.`
`19:26.39`	`CrashSys`	`4.0.0 likes to hang at initializing network`
`19:26.54`	`*** join/#asterisk Micc (n=dotirc@c-67-168-123-42.hsd1.wa.comcast.net)`
`19:27.05`	`CrashSys`	`well if you wait 15 minutes it eventually gets past it`
`19:27.29`	`Micc`	`Is there anyone here that can answer some questions about the asterisk manager api and originate?`
`19:28.01`	`Shaun2222`	`CrashSys: you need to set the boot server option from whatever it's on to static..`
`19:28.16`	`Shaun2222`	`i had the same problem, cuz my dhcp server wasnt giving a boot server`
`19:28.22`	`trippsss`	`what would be the best way to match 5555795800-5819 and then forward to the ${EXTEN} that would be the last 4 digits? Something like exten _55557958XX and ${EXTEN:6} but how to 00-19`
`19:28.28`	`Shaun2222`	`once you do that the phone will boot fast`
`19:28.43`	`Alan_Hicks`	`Shaun2222: What do you mean by "to static"?`
`19:29.11`	`Shaun2222`	`Alan_Hicks: under setup->dhcp there's a option that says boot server`
`19:29.12`	`trippsss`	`_555579580X and _555579581Xwould be easy I suppose in two entries . . .`
`19:29.16`	`CrashSys`	`Alan: If you dont set it to static it will wait for DHCP to give it the boot server option`
`19:29.18`	`Shaun2222`	`it's normaly set to custom+opt66 or somthing`
`19:29.29`	`Shaun2222`	`you change it to static`
`19:29.42`	`Alan_Hicks`	`Oh ok.`
`19:29.55`	`trippsss`	`oh I guess I could use _55557958[01]X`
`19:29.57`	`CrashSys`	`And in 4.0.0 if you dont change it to static it sits there for 15 minutes or so at "initializing network"`
`19:30.01`	`Alan_Hicks`	`But if my dhcp server does farm out a boot-server option, does it still boot slowly?`
`19:30.02`	`Shaun2222`	`ya if you dont the phone takes like 10 minutes to boot, it's rediculis.`
`19:30.08`	`trippsss`	`that should work . . .`
`19:30.09`	`CrashSys`	`Where as every OTHER bootrom fired up right away`
`19:30.10`	`Shaun2222`	`Alan_Hicks: no.`
`19:30.17`	`Alan_Hicks`	`Thanks.`
`19:30.39`	`Alan_Hicks`	`That should be a non-issue for me then.`
`19:30.54`	`CrashSys`	`It just means you now have more hands-on provisioning for non-local phones is all...`
`19:30.57`	`Alan_Hicks`	`I've already got dhcpd farming that out.`
`19:31.04`	`Alan_Hicks`	`Makes sense.`
`19:31.10`	`k31th`	`if i have this in my extensions.conf`
`19:31.11`	`k31th`	`exten => 100,1,Answer()`
`19:31.11`	`k31th`	`exten => 100,n,Background(main-menu)`
`19:31.23`	`k31th`	`I should be able to dial 100 right?`
`19:31.35`	`Alan_Hicks`	`Depends on your context, but yes.`
`19:31.39`	`Shaun2222`	`Alan_Hicks: ya i didnt, but i use ftp to pull/push configs`
`19:31.40`	`Corydon76-vcch`	`Assuming you're starting in the same context, yes`
`19:32.00`	`*** join/#asterisk gandhijee (n=user@mail.win-ent.com)`
`19:32.21`	`k31th`	`well it does not work.`
`19:32.23`	`k31th`	`Call from '1005' to extension '100' rejected because extension not found.`
`19:32.26`	`k31th`	`from console.`
`19:33.00`	`Alan_Hicks`	`keith4: What context are you placing the call from?`
`19:33.10`	`Alan_Hicks`	`SIP phone?`
`19:33.24`	`keith4`	`sure`
`19:33.29`	`keith4`	`why not`
`19:33.35`	`k31th`	`lol`
`19:33.46`	`k31th`	`Alan_Hicks: yeah`
`19:33.52`	`Alan_Hicks`	`keith4: Sorry. :^)`
`19:33.59`	`k31th`	`getting a bit keith'd out :D`
`19:34.01`	`[TK]D-Fender`	`k31th: fROM CONSOLE YOU HAVE TO TELL IT WHAT context <-`
`19:34.03`	`Alan_Hicks`	`k31th: Then pastebin extensions.conf and sip.conf please.`
`19:34.09`	`keith4`	`no worries, I do it all the time`
`19:34.16`	`keith4`	`holy caps lock, [TK]D-Fender`
`19:35.45`	`ai-a`	`sip debug turns it on.. what turns it off ?`
`19:35.58`	`ai-a`	`sip no debug ;)`
`19:36.04`	`outtolunc`	`'help sip'`
`19:36.05`	`*** join/#asterisk PhiberOptik (n=PhiberOp@75.148.50.141)`
`19:36.35`	`ai-a`	`that doesnt stop it Heh.`
`19:36.39`	`k31th`	`Alan_Hicks: http://pastebin.ca/749552`
`19:36.49`	`ai-a`	`just comes up with loads of words i cant read... Help files are for girls ;)`
`19:37.21`	`Alan_Hicks`	`k31th: And sip.conf?`
`19:37.58`	`k31th`	`http://pastebin.ca/749553`
`19:38.11`	`Alan_Hicks`	`I bet the problem is your 100 extension is in the [incoming] context and not the [internal] context.`
`19:38.12`	`k31th`	`sorry im slacking...`
`19:38.41`	`Alan_Hicks`	`context=threshold-out`
`19:38.59`	`k31th`	`ah shit`
`19:39.05`	`Alan_Hicks`	`You con't have a threshold-out context defined in extensions.conf.`
`19:39.08`	`*** join/#asterisk s0lid (n=_freq@210.213.243.193)`
`19:39.11`	`PhiberOptik`	`D-Fender, verbosity is at least 10. 39 actually. When leaving the message in the broadcast extension mailbox, 8810, asterisk cli shows that it's recording the wav file, but the file is only saved in the broadcast extensions mailbox, /var/spool/asterisk/voicemail/default/8810, the voicemail application is not saving the channel data in the other mailboxes specified. I.E. exten => 5000,1,VoiceMail(u5000&2203&2202)`
`19:39.18`	`Alan_Hicks`	`If that was context=incoming, it would work.`
`19:39.31`	`k31th`	`thats my bad i copied ad pasted that from some where.`
`19:40.04`	`k31th`	`tbh i have not learned contexts yet, im getting to that part in the book now.`
`19:40.14`	`Alan_Hicks`	`#asterisk 47 v. Alan_Hicks 1`
`19:40.24`	`k31th`	`thanks :D`
`19:40.32`	`*** part/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)`
`19:40.46`	`*** join/#asterisk Ciber311 (n=Ciber311@user-1087eds.cable.mindspring.com)`
`19:40.52`	`Alan_Hicks`	`k31th: Yeah, I'm a total rookie too. contexts are fairly easy to understand though. It's simply a way of segmenting portions of your dial-plan so they don't interact with one another.`
`19:40.52`	`*** join/#asterisk callguy (n=callguy@pool-72-70-81-172.bstnma.east.verizon.net)`
`19:40.58`	`[TK]D-Fender`	`PhiberOptik: Wheres the pastebin?`
`19:41.34`	`k31th`	`Alan_Hicks: I understand the concept,just not how to put it in place (yet)`
`19:41.36`	`Alan_Hicks`	`Example: You may have an FXO module configured for incoming calls and it's context might be [incoming] while your sip phones inside your network use context [internal]`
`19:42.07`	`Alan_Hicks`	`Calls coming in Zap/? would hit [incoming] and calls from sip phones would hit [internal].`
`19:43.21`	`k31th`	`Alan_Hicks: I see, and i presume i can do the same for out going`
`19:43.31`	`*** join/#asterisk PepOSX (n=pepOSX@190.72.148.244)`
`19:43.44`	`*** join/#asterisk mohsen (n=chatzill@213.217.51.84)`
`19:43.48`	`k31th`	`I am going to have to learn this as I need to specify handsets to different outbound trunks`
`19:44.59`	`[TK]D-Fender`	`k31th: No. Outgoing has nothing to do with this.`
`19:45.45`	`Alan_Hicks`	`Contexts don't care WHERE the call is going to IIUC.`
`19:45.47`	`[TK]D-Fender`	`k31th: Contexts control what a given device can dial. All calls are "incoming" in your dialplan. What they do in terms of dialout OUT a resource is another matter`
`19:46.00`	`karleeto`	`any people with aastra experience (especially cordless), PLEASE join us in #aastra`
`19:46.08`	`karleeto`	`thanks in advance'`
`19:46.13`	`k31th`	`http://pastebin.ca/749565 this means there is no sound file there? does asterisk look in default sounds dir for files? or should I have specified where the sound file should be?`
`19:46.28`	`*** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com)`
`19:46.44`	`k31th`	`[TK]D-Fender: thanks`
`19:47.07`	`Alan_Hicks`	`k31th: It'll look in the default sound files directory unless you specify a full PATH.`
`19:47.19`	`Alan_Hicks`	`Usually that means /var/lib/asterisk/sounds`
`19:47.20`	`k31th`	`humm i wonder why i have no sounds.`
`19:47.43`	`k31th`	`Ahh i do have sounds...`
`19:47.49`	`k31th`	`dir is full of .gsm files.`
`19:48.16`	`[TK]D-Fender`	`k31th: show us THAT FILE is there..`
`19:48.59`	`k31th`	`It is not, which would explain this error.`
`19:49.23`	`k31th`	`I wonder why my box does not have this file. when i have followed the book... and that install does.`
`19:49.30`	`Alan_Hicks`	`Yeah, usually when you see "No such file or directory" it means it's not there. :-)`
`19:49.38`	`k31th`	`lol`
`19:49.43`	`PhiberOptik`	`D-Fender. http://pastebin.com/dec71315`
`19:49.48`	`Alan_Hicks`	`k31th: I don't have that file either.`
`19:50.05`	`mohsen`	`How can I get the list of registered "realtime" sip peers/users?`
`19:50.16`	`Alan_Hicks`	`Are you reading the first edition of the O'Reilly book?`
`19:50.19`	`[TK]D-Fender`	`k31th: the book is giving yuo a SAMPLE. Stop taking everything it shows so literally.`
`19:50.32`	`[TK]D-Fender`	`k31th: Make your own damn recordings! :p`
`19:50.37`	`k31th`	`I am readon the second edition.`
`19:50.42`	`k31th`	`reading*`
`19:50.57`	`k31th`	`:D`
`19:51.08`	`k31th`	`I dont kow how to do that part yet`
`19:51.37`	`k31th`	`Is there a way of dialing in and recording ?`
`19:51.59`	`[TK]D-Fender`	`PhiberOptik: And where is your broadcast attempt in there?`
`19:52.20`	`[TK]D-Fender`	`k31th: "show applications" <- get reading.`
`19:52.54`	`ai-a`	`doing a wanpipe hwprobe returns sangoma 101c. is this reliable? it seems 101d has Echo Cancellation?`
`19:53.09`	`PhiberOptik`	`D-Fender, the procedure is, *98 on phone, Log in to a regular extension. (Not 8810) 3 for adv opts, 5 to leave msg, 1 to specify ext 8810, then record message, and 1 to accept`
`19:53.15`	`ai-a`	`can it be misrepresenting itself due to config settings ?`
`19:53.55`	`k31th`	`[TK]D-Fender: wow...`
`19:54.07`	`k31th`	`hahahah`
`19:54.18`	`k31th`	`@ tt-monkeysintro`
`19:54.44`	`PhiberOptik`	`From what I understand, setting, exten => 5000,1,VoiceMail(u5000&2203&2202) should tell the voice mail application to save the channel data in the other two mailboxes. Thus saving the wave file in the other two boxes.`
`19:55.20`	`PhiberOptik`	`That line is set in /etc/asterisk/extensions.conf`
`19:56.14`	`PhiberOptik`	`D-Fender, That pastebin output is all I get during this attempt.`
`19:56.15`	`[TK]D-Fender`	`PhiberOptik: 8810 has nothign to do with "broadcast.`
`19:56.32`	`PhiberOptik`	`Please explain.`
`19:56.50`	`[TK]D-Fender`	`PhiberOptik: You CANNOT broadcast to multiple boxes from withing Voiemailmain.`
`19:56.56`	`PhiberOptik`	`OOPS, I meant ext 5000`
`19:57.12`	`[TK]D-Fender`	`PhiberOptik: Show me something USEFUL`
`19:57.36`	`[TK]D-Fender`	`PhiberOptik: I see no sign of failure because I don't see anything successfully deposited in the first place.`
`19:57.47`	`PhiberOptik`	`ext 5000 is the broadcast extension.`
`19:58.00`	`PhiberOptik`	`Yeah, that's the problem, it's like it's just not doing it.`
`19:58.02`	`[TK]D-Fender`	`PhiberOptik: Show me it FAILING.`
`19:58.28`	`*** join/#asterisk coolhp (n=coolhp@74.15.123.88)`
`19:58.31`	`PhiberOptik`	`What are the dependancies of voicemailmain? Perhaps I'm missing a library, or something.`
`19:59.03`	`unstable`	`I'm looking for a good analog phone, one of those nice office speaker phones, and a cordless headset to go with it. Anyone know a good product? something with nice reception, I got this crappy phone from home depot, and I can't hear much when I'm on the second floor because of this sheet metal ceiling.`
`19:59.33`	`J4k3`	`wow`
`19:59.34`	`[TK]D-Fender`	`PhiberOptik: No, you aren't. We aren't talking about voicemailmain here.`
`19:59.44`	`J4k3`	`you want a lot of garbage there, unstable.`
`19:59.46`	`Alan_Hicks`	`Whoa.... this has got me confused. From the Release notes for Bootrom 4.0.0 for Polycom phones: "Removed Features: BootROM will not provision configuration files anymore."`
`19:59.47`	`[TK]D-Fender`	`PhiberOptik: Show me your BROADCAST ATTEMPT FAILING.`
`19:59.51`	`J4k3`	`go buy a BT adapter, stick it in your USB port, ahve a nice day.`
`20:00.04`	`Alan_Hicks`	`If I upgrade to 4.0.0, how am I supposed to fix configuration problems on the phone?`
`20:00.25`	`BBHoss`	`u dont need 4.0 anymore`
`20:00.26`	`coolhp`	`Hi all... I have a bit of an mistery happening here. I'm installing asterisk 1.4 on a Debian Etch (Kernel 2.6.18). Everything looks fine except for the following : When I load ztdummy, I get no audio. If I remove ztdummy and restart asterisk, I get audio again... any ideas ?`
`20:00.40`	`BBHoss`	`w/e is on the phone is fine unless you are having a specific problem`
`20:01.02`	`PhiberOptik`	`Is it not this line? Oct 25 12:44:27 VERBOSE[20718] logger.c: -- x=0, open writing: /var/spool/asterisk/voicemail/default/5000/tmp/jNNEdE format: wav, 0x6f6860. Should it not say saving the file to box 5000 as well as 2202 and 2203?`
`20:01.05`	`Alan_Hicks`	`I understand that, but I'm just curious as to how to answer that question.`
`20:01.30`	`[TK]D-Fender`	`PhiberOptik: That is 1 solity meaningless line by itself. It says NOTHING to me.`
`20:01.58`	`mohsen`	`How can I list sip registered users from CLI?`
`20:02.02`	`[TK]D-Fender`	`PhiberOptik: And does it SAY to the other boxes? No.`
`20:02.07`	`*** join/#asterisk rpm (n=russell@75.155.167.90)`
`20:02.50`	`unstable`	`J4k3: What do you mean a lot of garbage?`
`20:02.58`	`unstable`	`J4k3: I just want a good speaker phone with a head set.`
`20:03.07`	`*** part/#asterisk krondorl (n=chatzill@207.245.216.9)`
`20:03.18`	`*** join/#asterisk wishes (n=wishes@60.234.20.178)`
`20:03.37`	`wishes`	`whats a windows sip phone software that works over a VPN ?`
`20:03.48`	`wishes`	`i was using wengophone but it doesnt bind to a specific interface`
`20:04.15`	`unstable`	`wishes: #openwengo`
`20:04.28`	`*** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`20:04.34`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`20:05.04`	`wishes`	`i think you must be mistaken, i dont want to fix wengo, i want to find alternative software that works`
`20:05.36`	`wishes`	`nm http://www.voip-info.org/wiki-VOIP+Phones#SoftPhones will do`
`20:05.37`	`mcab`	`Alan_Hicks: in newer applictions (well, since 1.5+), the Application has managed the downloading of the config files. the notice just means that 4.0 won't work with really old apps...`
`20:06.03`	`Alan_Hicks`	`Ah.`
`20:06.31`	`PhiberOptik`	`D-Fender, What is the correct syntax to use to create a broadcast extension? I used this and other pages from voip-info as an example. http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf See the section "voicemail broadcasts"`
`20:07.21`	`*** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it)`
`20:07.54`	`*** part/#asterisk arekm (i=arekm@pld-linux/arekm)`
`20:08.49`	`[TK]D-Fender`	`PhiberOptik: Ok, you are wasteing time. Show me a COMPLETE attempt to call it in the first place <-`
`20:10.40`	`coolhp`	`Would anyone happen to have any issues with ztdummy on a 2.6.18 kernel ?`
`20:10.57`	`coolhp`	`When I load ztdummy up, I can no longer playback any audio.`
`20:11.15`	`coolhp`	`Yet if I remove it, everything works fine (except MOH obviously).`
`20:11.32`	`tzafrir`	`coolhp, try running zttest`
`20:11.36`	`coolhp`	`Running a zttest returns Best: 0.000000 -- Worst: 100.000000 -- Average: 100.000000`
`20:11.42`	`hmmhesays`	`anyone have the problem where voicemail hangs up if there is a blank message?`
`20:12.24`	`PhiberOptik`	`D-Fender, if I dial extension 5000 directly and leave a message, then it works perfectly. However if I dial into some other extension and follow the procedure I listed above, then it does not work. Aparently asterisk does not have the ability to ad-hoc record a message in the voicemail system and then send it to a broadcast extension?`
`20:12.39`	`BBHoss`	`no`
`20:12.53`	`BBHoss`	`the extension you call broadcasts to the other extensions`
`20:13.05`	`BBHoss`	`they dont all broadcast to each other`
`20:13.20`	`*** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com)`
`20:13.43`	`[TK]D-Fender`	`PhiberOptik: You do not send a vm to an EXTENSION.`
`20:15.44`	`[TK]D-Fender`	`PhiberOptik: And as I said, you cannot brodcast from within VOICEMAILMAIN <-----------`
`20:15.57`	`*** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com)`
`20:16.33`	`PhiberOptik`	`D-Fender, is there another way to accomplish this? Perhaps leave voicemailmain()`
`20:17.16`	`[TK]D-Fender`	`PhiberOptik: CLEARLY. You have a way already. its to call VOICEMAIL like aready do.`
`20:18.18`	`*** join/#asterisk bkw_ (n=brian@70.128.126.184)`
`20:18.56`	`coolhp`	`Could my ztdummy problem be a kernel issue... should try recompiling my kernel ?`
`20:20.11`	`*** join/#asterisk seipak (i=kapies@cm96100.red.mundo-r.com)`
`20:20.28`	`*** join/#asterisk saftsack (n=saftsack@pD9E06D09.dip.t-dialin.net)`
`20:20.41`	`*** join/#asterisk Darthclue (n=e054502@fw149.nisd.net)`
`20:20.42`	`PhiberOptik`	`D-Fender OK Thank you for your help.`
`20:21.46`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`20:24.14`	`Alan_Hicks`	`Eureka! dialplan.digitmap! That's what I need to edit on these phones!`
`20:25.42`	`*** join/#asterisk ManxPower (n=manxpowe@254.sub-70-221-93.myvzw.com)`
`20:26.42`	`*** join/#asterisk b_lay (n=blay@c-66-41-250-224.hsd1.mn.comcast.net)`
`20:27.56`	`TrentCreek`	`okay..go get to it`
`20:28.04`	`docelmo`	`hehe`
`20:28.05`	`Alan_Hicks`	`So... I should be able to set it to something like XXX0T\|9,XXXXXX1T\|9,1XXXXXXXXXX`
`20:28.11`	`docelmo`	`Alan_Hicks yes`
`20:28.13`	`docelmo`	`on a polycom`
`20:28.47`	`Alan_Hicks`	`That means if any three digits are dialed (that don't start with a 9) the phone will immediately dial that extension.`
`20:29.17`	`docelmo`	`should be XXX\|`
`20:29.21`	`Strom_C`	`Alan_Hicks: no, that's not what you specified`
`20:29.26`	`Alan_Hicks`	`If a 9 is dialed, it gives a second dialtone, then if 7 digits are entered, 1 second later it'll dial that extension.`
`20:29.39`	`Strom_C`	`any three digits not starting with 9: [0-8]XX`
`20:29.51`	`Alan_Hicks`	`And a 9 followed by 11 digits would dial that 11 digit extension.`
`20:29.57`	`Alan_Hicks`	`Strom_C: Gotcha.`
`20:30.25`	`Alan_Hicks`	`docelmo: I understand that you can also use [0-9]T to specify a timeout.`
`20:30.42`	`Alan_Hicks`	`Though that might be better put in dialplan.digitmap.timeout`
`20:30.56`	`Alan_Hicks`	`s/out/Out/`
`20:31.18`	`coolhp`	`Ok... I have a little more information now. It seems that it is a ZtDummy problem. When ZTDUMMY is loaded, I get no RTP from the server to the phone what so ever. If I remove ztdummy and place the same call, I now get RTP... any ideas ?`
`20:32.07`	`Alan_Hicks`	`This is fsckin' awesome. I can't believe there's a company out there that publishes freely downloadable, complete documentation for their devices.`
`20:36.39`	`ManxPower`	`Alan_Hicks: IMPOSSIBLE! You must be on drugs. tease`
`20:36.54`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)`
`20:37.09`	`Alan_Hicks`	`Um, does nicotene count? :^)`
`20:37.25`	`ManxPower`	`Alan_Hicks: usually not.`
`20:38.02`	`ManxPower`	`alan, paste a copy of your dialplan for the polycom and I'll optimize it for you. I assume you are in the USA.`
`20:40.04`	`ManxPower`	`Alan_Hicks: with a well designed dialplan users should almost never have to wait for the timeout to kick in.`
`20:40.18`	`mohsen`	`Anybody can give me a hint about listing my SIP users from CLI/AMI?`
`20:40.30`	`ManxPower`	`mohsen: "sip show peers"`
`20:40.47`	`Alan_Hicks`	`I am in the USA. I'll take you up on that offer, but it may be a little while yet. I've still got reading and ciphering to do.`
`20:41.20`	`mohsen`	`ManxPower: That does not list my realtime users, even though there is 'realtime' column in the table it throws`
`20:41.34`	`ManxPower`	`mohsen: you did not say you are using realtime`
`20:41.50`	`ManxPower`	`Alan_Hicks: perhaps tomorrow then`
`20:42.53`	`mohsen`	`ManxPower: I do. I wonder what's the use of 'realtime' column in the table. Could be a bug?`
`20:42.55`	`Alan_Hicks`	`Perhaps tomorrow, perhaps later today. I like to try to figure things out as much as possible for myself first, then let some one who knows more about it critique it rather than show you a sow's ear and ask for a silk purse.`
`20:43.00`	`BBHoss`	`users are not peers i thought`
`20:43.27`	`ManxPower`	`mohsen: I have never, ever in my entire life used Realtime`
`20:43.49`	`BBHoss`	`and to show users its sip show users`
`20:43.54`	`mohsen`	`ManxPower: I see. You store all the users in plain files?`
`20:44.01`	`ManxPower`	`mohsen: correct`
`20:44.12`	`BBHoss`	`:)`
`20:44.18`	`mohsen`	`ManxPower: How many of them do you have?`
`20:44.25`	`*** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com)`
`20:44.26`	`mohsen`	`Users, I mean`
`20:44.28`	`ManxPower`	`mohsen: 80 or so on the largest system`
`20:44.38`	`ManxPower`	`80 or so PHONES on the largest system`
`20:44.49`	`mohsen`	`aha`
`20:44.52`	`Alan_Hicks`	`Hmm.... just had a thought (and I know it's in the book somewhere, but I'd like an answer before I forget the question)... let's assume all 4 FXO channels are full at the moment, and a user tries to dial-out....`
`20:44.54`	`ManxPower`	`close to 300 total phones across all systems`
`20:45.11`	`BBHoss`	`all circuits busy is default i think`
`20:45.14`	`ManxPower`	`Alan_Hicks: Dial will exit, set DIALSTATUS and HANGUPCAUSE and expect you to handle it from there.`
`20:45.17`	`mohsen`	`BBHoss: what you said (sip show users) does not work for realtime users.`
`20:45.18`	`Alan_Hicks`	`Is there an application that will inform the user that all circuits are busy and hold for one to become available?`
`20:45.37`	`ManxPower`	`Alan_Hicks: no, you would write that in your dialplan`
`20:45.46`	`Alan_Hicks`	`Ok, thanks.`
`20:45.49`	`BBHoss`	`righto`
`20:46.24`	`Alan_Hicks`	`When Dial() does that, will it default to +101 like it does if the remote line is busy?`
`20:46.44`	`Alan_Hicks`	`Or should I just go RTFM? :^)`
`20:47.55`	`ManxPower`	`Alan_Hicks: +101 is not used anymore. don't use it.`
`20:47.59`	`ManxPower`	`I just said what it does.`
`20:48.04`	`ManxPower`	`ManxPower: Alan_Hicks: Dial will exit, set DIALSTATUS and HANGUPCAUSE and expect you to handle it from there.`
`20:48.17`	`ManxPower`	`actually it will set those variables, THEN exit.`
`20:48.25`	`Alan_Hicks`	`Ok. Guess that's my fault for having the first edition of the book.`
`20:48.26`	`*** join/#asterisk mog (n=mog@c-71-207-220-220.hsd1.al.comcast.net)`
`20:48.26`	`*** mode/#asterisk [+o mog] by ChanServ`
`20:48.57`	`ManxPower`	`Alan_Hicks: look at extensions.conf.sample for the macro-stdexten stuff for a simple example`
`20:48.59`	`Alan_Hicks`	`I assume it has different error codes in DIALSTATUS for busy, congestion, et al.`
`20:49.03`	`Alan_Hicks`	`ok`
`20:49.11`	`ManxPower`	`Alan_Hicks: they are all listed in "show application dial"`
`20:50.46`	`*** part/#asterisk unstable (n=unstable@tor/regular/sid)`
`20:51.02`	`ManxPower`	`in fact the most current documentation for ALL asterisk applications is "show application X" where X is the application. You can also do a "show applications" for a list of applications, as well as "show functions" for a list of functions. Functions are case sensitive.`
`20:53.47`	`ManxPower`	`I finally found a feature that might compel me to upgrade to 1.4`
`20:54.35`	`*** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-58-243.pskn.east.verizon.net)`
`20:55.04`	`Alan_Hicks`	`the console needs a "\| less" feature.`
`20:55.28`	`Deeewayne`	`O.O`
`20:56.01`	`Alan_Hicks`	`Did I just say something stupid?`
`20:56.12`	`*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)`
`20:57.42`	`Deeewayne`	`Alan_Hicks: never mind me, I'm just looking around`
`20:58.58`	`b_lay`	`Hi, we're thinking about setting up an Asterisk system and we have one outside line and want three inside extensions to anaolog phones. Currently we're looking at the TDM400P. Does anyone know if this card would be good for what we want to do and what modules we need?`
`21:00.31`	`[TK]D-Fender`	`b_lay, I'd suggest Linksys SPA-3102 + SPA-2102 +/- $150 total`
`21:01.03`	`Alan_Hicks`	`b_lay: Good card, but pricey. You'd need 1 FXO module and three FXS modules.`
`21:01.26`	`*** join/#asterisk cyclone (n=cyclone@antitech.xmission.com)`
`21:01.48`	`b_lay`	`Thanks, what are the differences between FXO and FXS?`
`21:02.05`	`Alan_Hicks`	`Superficially, it's what they connect to.`
`21:02.16`	`cyclone`	`Anyone know of a Good Free SIP SoftPhone with Push to Talk capabilities and is Free?`
`21:02.35`	`Alan_Hicks`	`FXO modules connect to your telco and use FXS signalling. FXS modules connect to a station (analogue phone) and use FXO signalling.`
`21:02.44`	`[TK]D-Fender`	`~fxsfxo`
`21:02.44`	`jbot`	`hmm... fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it.`
`21:04.02`	`b_lay`	`oh, okay. Thanks a lot`
`21:04.04`	`cyclone`	`Trying to replace a Ventrilo setup for a big gaming group with ASterisk and a couple conference rooms. Everything works cept getting people a really easy to use Push to Talk SIP Client.`
`21:04.59`	`b_lay`	`And also, what's involved with using IP phones? Do we need a special card? What all would we need?`
`21:06.29`	`cyclone`	`b_lay: IP Phones usually are just ethernet and suppose TCP`
`21:06.35`	`cyclone`	`So you'd just need IP Phones and a switch.`
`21:06.36`	`Darthclue`	`IP Phones run over the computer network. Nothing special unless you want power over ethernet.`
`21:07.59`	`irule`	`http://swik.net/AsteriskDialplan step! right in and be part of history while enjoying the new open source PBX dialplan for Asterisk! ;)`
`21:08.24`	`roxlu`	`hi`
`21:08.34`	`roxlu`	`does someone knows a good speech recognition engine for asterisk?`
`21:08.47`	`Alan_Hicks`	`ManxPower: Still around?`
`21:08.56`	`BBHoss`	`roxlu: lumenvox is very nice, its not free though`
`21:09.14`	`cyclone`	`hrmm`
`21:09.20`	`roxlu`	`is there a free recognition engine?`
`21:09.23`	`cyclone`	`must not be a good free softphone with PTT`
`21:10.16`	`J4k3`	`I have a CB radio`
`21:10.22`	`J4k3`	`you might want to get one, too.`
`21:11.04`	`Alan_Hicks`	`From what I understand, PTT is pretty proprietary.`
`21:12.10`	`cyclone`	`hrmmm doesn't make sense. I figure just something that unmutes you when you hold an assigned key down.`
`21:13.02`	`Alan_Hicks`	`PTT doesn't work that way.`
`21:13.20`	`cyclone`	`I know.. but even if there was something that simple.. I'd be really happy. hehe ;)`
`21:14.10`	`Alan_Hicks`	`Is there anyway to look at the current settings inside a Polycom phone?`
`21:14.19`	`Alan_Hicks`	`Other than the horrid web interface of course.`
`21:15.27`	`b_lay`	`Thanks for all the help guys`
`21:17.30`	`cyclone`	`So to refine my question.. sorry. Is there a free soft phone that has an option to assign a hotkey to quickly mute and unmute yourself?`
`21:19.42`	`wishes`	`x-lite is cool`
`21:19.56`	`wishes`	`sorry, dunno if it does what you want, but i just installed it on another pc`
`21:20.07`	`wishes`	`and it worked and was nice and purdy and easy to setup :)`
`21:20.14`	`*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)`
`21:20.26`	`grandpapadot`	`Is there a t.38 patch for asterisk 1.2.24?`
`21:20.29`	`wishes`	`actually- it does have mute`
`21:20.32`	`cyclone`	`yeah.. xten makes a good soft phone.. but it just doesn't do what I need it to do.`
`21:20.37`	`wishes`	`because i accidentally muted myself`
`21:20.55`	`cyclone`	`but can you assign a key to mute and unmute within the application?`
`21:21.00`	`wishes`	`not to sure about hotkeys though`
`21:21.09`	`wishes`	`i would have thought it wouldnt be too hard`
`21:21.11`	`wishes`	`but try it?`
`21:21.22`	`cyclone`	`been using x-lite for 3 years now`
`21:21.58`	`cyclone`	`no worries.. I'll keep hunting.. or figure out how hard it is to write a simple one. :(`
`21:22.05`	`cyclone`	`Thanks all for your help. take care.`
`21:22.07`	`*** part/#asterisk cyclone (n=cyclone@antitech.xmission.com)`
`21:29.04`	`[TK]D-Fender`	`Alan_Hicks, Whats to look at?`
`21:29.40`	`Alan_Hicks`	`[TK]D-Fender: You mean my "\| less" comment? That was in reference to how many asterisk console commands scroll past my buffer.`
`21:29.52`	`Alan_Hicks`	`Examples are "help" and "show application dial".`
`21:29.57`	`[TK]D-Fender`	`Alan_Hicks, No about Polycom configs`
`21:30.26`	`Alan_Hicks`	`Oh! Nothing in particular; I was just curious is all.`
`21:31.49`	`grandpapadot`	`Is there a t.38 patch for asterisk 1.2.24?`
`21:32.12`	`[TK]D-Fender`	`grandpapadot, Do onto Mantis and take a look`
`21:41.35`	`hmmhesays`	`bah I need the latest sip.ld and bootroom for a 501`
`21:41.43`	`hmmhesays`	`I can only find 2.01`
`21:42.10`	`hmmhesays`	`anyone have it handy?`
`21:42.50`	`*** part/#asterisk jmls (n=jmls@62.49.235.130)`
`21:42.53`	`hmmhesays`	`I could really use it`
`21:47.19`	`alrs`	`hmmhesays: It looks like 2.1.2 is up on Polycom's site. Am I confused?`
`21:47.25`	`alrs`	`hmmhesays: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip501.html`
`21:48.20`	`alrs`	`hmmhesays: 2.2 is in the trixbox repo as an rpm: http://yum.trixbox.org/centos/4/RPMS/repodata/repoview/firmware-polycom-0-2.2.0-1.html`
`21:48.43`	`Qwell`	`they're redistributing polycom firmware?`
`21:49.15`	`*** join/#asterisk Midmania (n=no@cpc2-midd5-0-0-cust770.midd.cable.ntl.com)`
`21:50.18`	`Strom_M`	`Qwell: isn't that against the polycom license agreement?`
`21:50.28`	`Qwell`	`Strom_M: sounds like it`
`21:50.41`	`Qwell`	`I don't know the details though - I just think it's interesting.`
`21:51.13`	`*** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai)`
`21:51.30`	`Midmania`	`HI,`
`21:51.46`	`Midmania`	`Is there anyone who can help me with the digium card?`
`21:52.21`	`Corydon76-vcch`	`Why don't you call Digium support?`
`21:52.35`	`Strom_M`	`Why don't you ask us a specific question?`
`21:53.24`	`Midmania`	`I could do .... I am trying to get the disgium te120p card running but when I connect it via CAT5 to the BT ISDN30, the Customer Alarm does not go out?`
`21:53.28`	`Midmania`	`any ideas?`
`21:53.52`	`Strom_M`	`Midmania: is it a straight-through cable?`
`21:54.08`	`Midmania`	`yes ... I have tried a crossover cable too`
`21:54.15`	`Midmania`	`both dont work`
`21:54.19`	`Strom_M`	`is your card provisioned?`
`21:55.13`	`Midmania`	`well this is where is get slightly complicated. I am using Sipcat which is based on the asterisk platform, but they just want remote access all the time ..... not that easy on a corporate lan`
`21:55.22`	`hmmhesays`	`yeah I found some`
`21:55.37`	`Midmania`	`i have it set in the web interface`
`21:56.26`	`Strom_M`	`Midmania: why not use asterisk?`
`21:57.15`	`*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)`
`21:57.18`	`Midmania`	`probably because in depth linux scares me!?`
`21:57.48`	`*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)`
`21:57.50`	`Strom_M`	`this is not in-depth linux`
`21:58.06`	`Strom_M`	`you have to know how to edit files and load modules and that's about it`
`21:58.26`	`Midmania`	`with sipcat, it is all based in an ISO image for sipcat`
`21:58.39`	`Midmania`	`i can edit files`
`21:58.41`	`Strom_M`	`and we can't support sipcat here`
`21:58.46`	`outtolunc`	`they can't even get the online demo working <G>`
`21:58.53`	`Midmania`	`true`
`21:59.17`	`Midmania`	`how does the asterisk configuration differ from sipcat?`
`21:59.55`	`BBHoss`	`text based`
`21:59.59`	`BBHoss`	`its faster in the end`
`22:00.10`	`Strom_M`	`ive never used sipcat, so i cant speak to differences...but it's all text files you edit`
`22:00.50`	`BBHoss`	`i dont see how they can sell the product, since it is based on asterisk which is GPL`
`22:01.00`	`BBHoss`	`hmm`
`22:01.00`	`Strom_M`	`you can sell GPL'd software`
`22:01.12`	`Strom_M`	`please go read the GPL again :)`
`22:01.17`	`*** join/#asterisk seanmh (n=John@208.50.194.18)`
`22:01.28`	`Midmania`	`what is the best operating system to install for asterisk?`
`22:01.35`	`Strom_M`	`Midmania: whichever linux you like best`
`22:01.38`	`Strom_M`	`personally I like debian`
`22:01.44`	`seanmh`	`Trying to install a polycom 330, I get and error after it boots up: Error: 1.`
`22:01.49`	`seanmh`	`Does anyone know how to fix this error?`
`22:02.03`	`seanmh`	`The phone can receive and make calls but th error is on the screen`
`22:03.24`	`Alowishus`	`never seen that... try a reformat and reflash?`
`22:04.50`	`seanmh`	`already did :S`
`22:04.53`	`seanmh`	`still shows`
`22:04.58`	`seanmh`	`and I just got more detail on it`
`22:05.07`	`seanmh`	`XML Error: 13,8 Junk aftr document`
`22:05.24`	`Midmania`	`Looking at the Debian website, do I need to download the full DVD set?`
`22:05.28`	`Strom_M`	`seanmh: try starting over from a fresh set of config files`
`22:05.34`	`Strom_M`	`Midmania: no, start with the net install CD`
`22:05.44`	`hmmhesays`	`setting busy-limit=1 does not seem to be working`
`22:06.03`	`Strom_M`	`hmmhesays: don't you mean....call-limit?`
`22:06.18`	`BBHoss`	`Midmania: just download the basic install cd`
`22:06.19`	`alrs`	`seanmh: are you sure that your sip.cfg is written for the version of the firmware you are using? A lot changed between 2.1 and 2.2`
`22:06.22`	`hmmhesays`	`I want the phone to indicate busy at 1 call, but still be able to call out for transfers and what not`
`22:06.30`	`seanmh`	`Strom_M: do you know where I can get the config files for the 330?`
`22:06.39`	`Strom_M`	`seanmh: should be included with the firmware`
`22:07.02`	`hmmhesays`	`asterisk is still sending the call to the phone, even though there is one call present`
`22:07.12`	`hmmhesays`	`busy-limit=1 should do that right?`
`22:07.17`	`BBHoss`	`Midmania: http://cdimage.debian.org/debian-cd/4.0_r1/i386/iso-cd/debian-40r1-i386-netinst.iso should work, the stuff you need to install later can be installed with apt-get`
`22:07.32`	`BBHoss`	`it even has asterisk in apt, but i think it is very old`
`22:07.50`	`Strom_M`	`yeah, don't install asterisk from apt`
`22:08.36`	`Midmania`	`ok ..... once I have run the ISO image for debian, do I need to run anything else before installing the askerisk modules?`
`22:08.44`	`hmmhesays`	`or am I wrong about that?`
`22:09.07`	`BBHoss`	`they actually have 1.12.13 on debian 4`
`22:09.07`	`Strom_M`	`Midmania: there are some dependencies you'll need to install`
`22:09.20`	`Midmania`	`ok`
`22:09.27`	`Strom_M`	`BBHoss: the dependency tree for the debian package is bonkers`
`22:10.10`	`hmmhesays`	`Strom_M: I thought there was a busy-limit, am I wrong?`
`22:10.17`	`Strom_M`	`hmmhesays: i've never heard of busy-limit`
`22:10.45`	`Midmania`	`Thanks. I will try installing this in the morning .... if I get stuck, am I alright coming back?`
`22:10.53`	`hmmhesays`	`hrm ok, thats odd`
`22:10.54`	`Strom_M`	`Midmania: yeah`
`22:10.58`	`hmmhesays`	`I guess I'll disable call waiting`
`22:12.36`	`*** join/#asterisk jm\|laptop (n=jm\|home@dilbert.jamiem.com)`
`22:12.49`	`hmmhesays`	`now how do I disable call waiting in the sip.cfg file on a polycom`
`22:13.35`	`Midmania`	`thanks chaps`
`22:13.38`	`Midmania`	`take care`
`22:16.22`	`*** join/#asterisk CrossRoad (n=CrossRoa@209.172.67.146)`
`22:16.52`	`hmmhesays`	`call-limit=1 is not working either`
`22:17.08`	`BBHoss`	`does * 1.2~latest send a DURATION in RFC2833 packets?`
`22:17.38`	`*** join/#asterisk rpyne (n=rpyne@69.77.169.14)`
`22:18.07`	`Corydon76-vcch`	`No, it does not`
`22:18.13`	`BBHoss`	`ok`
`22:18.25`	`BBHoss`	`thanks`
`22:18.27`	`Corydon76-vcch`	`1.2 sends a start and a stop`
`22:18.32`	`hmmhesays`	`any reason call-limit wouldn't be working?`
`22:18.42`	`*** join/#asterisk mohsen (n=chatzill@82.99.234.189)`
`22:18.56`	`BBHoss`	`my ITSP says they are upgrading and this fact will cause issues with DTMF RFC2833`
`22:19.01`	`*** join/#asterisk CVirus (n=GoD@196.205.193.87)`
`22:19.10`	`BBHoss`	`so i guess i'll be moving to 1.4`
`22:20.00`	`*** join/#asterisk s34n (n=smcmurra@ip-208-76-93-125.mvdsl.com)`
`22:22.08`	`s34n`	`Calls coming in from my sip trunk are not succeeding. When Dial the call to one of my SIP phones, the phone answers, then disconnects`
`22:22.26`	`s34n`	`and * tells me Failed to authenticate on INVITE`
`22:23.31`	`*** join/#asterisk remmo (n=junk@202.1.119.80)`
`22:24.19`	`rpyne`	`Hello everyone. I am currently setting up asterisk RealTime Sip. I have edited my extconfig.conf sipusers => mysql,asterisk,sip_buddies`
`22:24.20`	`rpyne`	`sippeers => mysql,asterisk,sip_buddies and my res_mysql.conf has been edited. My Realtime Voicemail works fine. I have created my SIP table in MySQL and have inserted test entries, but my SIP entries are not registering. Any ideas are welcomed. Thanks.`
`22:24.23`	`s34n`	`What does * do once the extension picks up the call that would cause an INVIT failure?`
`22:26.34`	`*** join/#asterisk rva (n=rafael@189-68-99-65.dsl.telesp.net.br)`
`22:27.53`	`*** join/#asterisk tripps (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net)`
`22:27.55`	`*** join/#asterisk craigk (n=ckowald@58.174.122.198)`
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`22:41.36`	`seanmh`	`where can i get the sip.cfg for a polycom 330?`
`22:41.46`	`Qwell`	`seanmh: write one`
`22:43.14`	`seanmh`	`is there a manual or tutorial on how to write one?`
`22:45.19`	`outtolunc`	`http://www.voip-info.org/wiki-Polycom+Phones has bunch of info`
`22:45.52`	`*** join/#asterisk terracon (n=greisky@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com)`
`22:46.01`	`outtolunc`	`no idea if any of it is current <G>`
`22:46.46`	`seanmh`	`outtolunc: I've looked through that already. To this point I've failed to see information that could aide in writing a sip.cfg file from scratch.`
`22:46.59`	`seanmh`	`Is anyone willing to give me a sip.cfg for a polycom 330?`
`22:47.25`	`JT`	`seanmh: www.polycom.com is much more useful`
`22:47.46`	`seanmh`	`polycom.com doesn't allow download of those files without a proper login`
`22:48.05`	`*** join/#asterisk thx2000 (n=the@netblock-208-127-150-56.dslextreme.com)`
`22:48.06`	`JT`	`...`
`22:48.15`	`JT`	`all of the documentation is freely downloadable`
`22:48.38`	`seanmh`	`thanks, you're a big help.. I'm sure you'll sleep better at night knowing how smart you are.. Smile!`
`22:49.26`	`Alowishus`	`seanmh: Polycom does allow download of the older files without login... you can't get 2.2, but you can get 2.1.2`
`22:49.28`	`JT`	`what's up your arse?`
`22:50.06`	`seanmh`	`alowishus: correct me if I'm wrong but are those two version compatible with each other?`
`22:50.41`	`Alowishus`	`seanmh: well probably not... but if you d/l the 2.1.2 firmware for the 330 then load that on it and use the matching sip.cfg... just ignore 2.2 altogether`
`22:50.52`	`JT`	`seanmh: all the info on writing sip.cfg is FREELY DOWNLOADABLE`
`22:51.07`	`Alowishus`	`unless you already bumped up to 2.2, at which point... you shouldn't have done that or you'll be writing sip.cfg by hand :)`
`22:51.14`	`JT`	`http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip330_320.html#document`
`22:51.24`	`Alowishus`	`JT: well yeah but having the 100+k default sip.cfg is nice, and then he only has to write overrides`
`22:51.45`	`JT`	`www.polycom.com > Support > Voice/IP320/330`
`22:52.39`	`Shaun2222`	`ummm.... the sip software comes with a config to match.. just plug in your info.`
`22:53.35`	`outtolunc`	`and the admin guide has a 'configuration' section`
`22:54.01`	`*** join/#asterisk beeew (n=chatzill@c-24-7-43-146.hsd1.ca.comcast.net)`
`22:54.32`	`*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)`
`22:55.13`	`beeew`	`hi..i'm a newbie...when my log files show SIP/some_number-some_other_number...what is 'some number'..and what is 'some_other_number'`
`22:55.37`	`Qwell`	`beeew: in what context?`
`22:55.48`	`outtolunc`	`tech/dev-occurance`
`22:55.49`	`*** join/#asterisk abritton (n=abritton@207-255-210-182-dhcp.cbe.md.atlanticbb.net)`
`22:56.28`	`beeew`	`Qwell, i'm looking at my log files in Master.cvs`
`22:56.50`	`beeew`	`the tech number is my own id number?`
`22:56.57`	`Shaun2222`	`beeew: SIP/channel-channel_id i beleive is what they stand for`
`22:57.19`	`outtolunc`	`tech = ZAP/SIP/IAX/IAX2/MGCP/etc etc etc`
`22:57.37`	`Shaun2222`	`bahh.. sorry SIP/channel-session_id i ment`
`22:58.54`	`beeew`	`ok..`
`23:01.53`	`*** join/#asterisk implicit (n=implicit@ip72-197-20-157.sd.sd.cox.net)`
`23:04.05`	`*** join/#asterisk knarfly (n=vladimir@c-75-74-155-198.hsd1.fl.comcast.net)`
`23:04.12`	`*** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com)`
`23:05.18`	`seanmh`	`JT: Thank you for your answer. Much more helpful than "Write one". Thank you.`
`23:05.23`	`rva`	`what is wrong with this dialplan:`
`23:05.26`	`rva`	`exten => 777,4,Read(digito\|\|2)`
`23:05.26`	`rva`	`exten => 777,5,GotoIf($["$digito" == "10"]?6:8)`
`23:07.15`	`rva`	`forget it...i did not past it. sorry guys`
`23:09.23`	`JT`	`seanmh: my answer was always to check polycom.com ;)`
`23:10.30`	`*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)`
`23:11.44`	`*** join/#asterisk dijungal (n=kdaniel@209.59.110.38)`
`23:11.48`	`*** join/#asterisk jm\|laptop (n=jm\|home@dilbert.jamiem.com)`
`23:12.06`	`dijungal`	`what text-to-speech addon would you recommend for asterisk`
`23:12.08`	`dijungal`	`?`
`23:15.40`	`thx2000`	`dijungal: I like cepstral. They did a pretty good job matching up Allison w/ the real thing as well.`
`23:16.04`	`dijungal`	`oh yea...`
`23:16.12`	`dijungal`	`is cepstral free tho?`
`23:17.22`	`thx2000`	`No...pricing isn't terrible though`
`23:17.31`	`dijungal`	`k`
`23:23.26`	`wwalker`	`in escalations, I want an escalation to apply to 50 hosts. nagios requires that I enter a host_name directive. I want to use hostgroup. Is there some way to not enter 50 separate host_name directives?`
`23:24.03`	`*** part/#asterisk mog (n=mog@c-71-207-220-220.hsd1.al.comcast.net)`
`23:24.21`	`Shaun2222`	`looking for a alternative to voicepulse connect, anybody know of a good solid provider with decient rates, offer callerid(name) in/out, and support more than 4 concurrent channels?`
`23:33.46`	`knarfly`	`why would *8 not let me pickup a call...I only get a message on the CLI that says there's nothing to pickup?`
`23:34.11`	`Alan_Hicks`	`Before I spend a bunch more time rebooting my phone over and over again to load a new dialmap, will some one kindly check what I've got so ar for sanity?`
`23:34.15`	`dijungal`	`ahhh.. i can use flite for text-to-speech`
`23:34.17`	`knarfly`	`Shaun2222: try splitinfinity.com`
`23:34.21`	`Alan_Hicks`	`dialplan.1.digitmap="[2-9]11\|9,[2-9]xxxxxx\|9,1xxxxxxxxxx\|3xx" dialplan.1.digitmap.ti`
`23:34.25`	`Alan_Hicks`	`meOut="0\|0\|0\|0"`
`23:34.44`	`thx2000`	`knarfly: huh?`
`23:34.53`	`Alan_Hicks`	`Sorry, line wrap screwed that up, but you should be able to make it out. :^) This is a Polycom Soundpoint IP 320 phone.`
`23:35.12`	`Alan_Hicks`	`Intra-office extensions will all be of the form 3xx.`
`23:35.24`	`Alan_Hicks`	`And I'm un the USA.`
`23:36.04`	`k31th`	`morning`
`23:37.25`	`knarfly`	`thx2000: what did you want me to reference?`
`23:37.37`	`*** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch)`
`23:38.06`	`thx2000`	`knarfly: Just didn't understand your question`
`23:39.15`	`knarfly`	`thx2000: I have a POTS line into my * box. when a POTs call comes in I want to pick it up on one of my SIP phones but I'm getting the message above and a busy signal...no call pickup happens???`
`23:39.45`	`thx2000`	`ahh, didn't realize *8 was a typo :P`
`23:40.16`	`thx2000`	`zaptel?`
`23:40.26`	`knarfly`	`thx2000: I have configured the SIP phone to be in GROUP 1 and PICKUP GROUP 1, the zapata.conf also configures to PICKUP GROUP1`
`23:41.37`	`knarfly`	`still I cannot pickup the incoming calls....only the analog phones connected to the POTS extensions will answer. The caller can dial one of the SIP extensions and reach me but that's not what I'm wanting`
`23:51.42`	`*** join/#asterisk felix_da_catz (i=felix_da@65.57.248.187)`
`23:55.48`	`enalert`	`wwalker, lol`
`23:56.53`	`*** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com)`
`23:58.21`	`*** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au)`

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