irclog2html for #asterisk on 20060829

`00:00.21`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`00:01.15`	`*** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com)`
`00:01.35`	`MRH2`	`hi anyone know what zttranscode does?`
`00:05.52`	`tessier__`	`Well bah`
`00:06.10`	`tessier__`	`The d channel wouldn't come up and it's 5 oclock and everyone has to go home. We will reconvene tomorrow.`
`00:08.10`	`file`	`MRH2: ...yes`
`00:08.43`	`MRH2`	`lol cool just checking`
`00:15.26`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`00:16.09`	`*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)`
`00:25.49`	`*** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com)`
`00:27.34`	`kronos003`	`does * have a web interface - is there a demo of this anywherew`
`00:27.46`	`evilrabbi`	`freepbx`
`00:27.54`	`evilrabbi`	`is probably the closest thing`
`00:27.55`	`kronos003`	`link?`
`00:28.02`	`evilrabbi`	`www.freepbx.org i belive`
`00:28.29`	`kronos003`	`i see`
`00:28.38`	`*** join/#asterisk grexk (n=grexk@124.107.72.42)`
`00:28.47`	`kronos003`	`do they keep up with the asterisk code base pretty well`
`00:29.26`	`grexk`	`hello everyone`
`00:30.35`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`00:32.04`	`*** join/#asterisk epoch (i=epoch@octane.breakbeats.org)`
`00:32.42`	`*** part/#asterisk epoch (i=epoch@octane.breakbeats.org)`
`00:33.13`	`evilrabbi`	`kronos003 yeas`
`00:33.15`	`evilrabbi`	`yes*`
`00:34.14`	`kronos003`	`i see - they tell me that freepbx is a frontend more or less for asterisk and asterisk needs to be installed for it to work`
`00:34.31`	`evilrabbi`	`freepbx is just a web based front end for *`
`00:34.46`	`evilrabbi`	`i miss understood your question`
`00:34.57`	`kronos003`	`cool - that works - i have webmin - has anybody written anything for it`
`00:34.58`	`*** join/#asterisk wundaboy (n=uo@c-71-193-206-88.hsd1.or.comcast.net)`
`00:35.14`	`evilrabbi`	`not that's any good that I know`
`00:36.05`	`kronos003`	`darn - that would be nice -i use webmin to administer a lot of things on my server already - wish there was a cool snapin`
`00:36.32`	`*** join/#asterisk dieno2 (n=dienno2@124.29.194.128)`
`00:37.23`	`dieno2`	`does cchanging of ops can disturb calls`
`00:40.36`	`*** join/#asterisk bjohnson (n=bjohnson@i216-58-91-93.cybersurf.com)`
`00:42.05`	`kronos003`	`just found * web manager webmin plugin - http://webadminmodules.sourceforge.net/?page=New+Modules I cant beleive they are charging 295 USD for it disgusting`
`00:43.05`	`evilrabbi`	`wtf`
`00:43.21`	`evilrabbi`	`just use freepbx it's good`
`00:43.23`	`evilrabbi`	`i use it`
`00:44.11`	`kronos003`	`ya maybe i will - realy wanted something in a nice neat little snapin`
`00:45.31`	`*** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au)`
`00:45.33`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`00:45.49`	`wundaboy`	`im just setting up my voicemail, I was curious if i can have it go to someones mailbox and have the caller not have to enter the voicemail box number.`
`00:45.52`	`wundaboy`	`is that possible?`
`00:46.37`	`shodan`	`anyone got a voip store in canada that sells polycom & sipura ? I was gonna order from canadianvoipstore.com but it's just a front for voipsupply.com they ship from the us so that means 40$ shipping and customs on top of that :\`
`00:49.15`	`evilrabbi`	`that sucks`
`00:49.29`	`evilrabbi`	`the name is kinda missleading`
`00:50.59`	`blitzrage`	`shodan: www.voipdepot.com`
`00:51.15`	`blitzrage`	`shodan: I've ordered from there before (twice) -- fantastic shipping speed and located IN Canada`
`00:51.41`	`blitzrage`	`shodan: sorry, www.voipdepot.ca`
`00:52.01`	`shodan`	`whanks`
`00:52.04`	`Druken`	`shodan: what do you need?`
`00:52.18`	`shodan`	`a polycom 301 and a sipura spa-2002`
`00:53.20`	`Druken`	`let me check the prices, i'll let you know in a few mins...`
`00:53.28`	`shodan`	`k`
`00:55.38`	`Druken`	`hmmm... i'm not seeing a spa-2002.. i see a spa2102...`
`00:56.39`	`*** join/#asterisk r0ckst4r (n=Snake@20151231095.user.veloxzone.com.br)`
`00:57.14`	`shodan`	`I guess as long as it's sipura, has dual fxs and isn't much more expensive than a spa-2002 it'll do`
`00:57.50`	`Druken`	`oh, your basically lookin for a pap2?`
`00:58.09`	`bkw__`	`Enterprise Asterisk?`
`00:59.13`	`*** join/#asterisk babyju (n=babyju@ool-4351dd68.dyn.optonline.net)`
`00:59.24`	`kronos003`	`what is ZAP`
`00:59.50`	`shodan`	`Druken, that would probably do as well`
`01:00.03`	`bkw__`	`Enterprise Asterisk? What taking a segfault to enterpise scale?`
`01:00.09`	`Druken`	`when you drag your feet on the carpet, and touch the door knob, you get ZAPPED...`
`01:00.37`	`kronos003`	`LOL`
`01:00.52`	`kronos003`	`ok WHat is ZAP vs SIP`
`01:00.55`	`babyju`	`who is better than broadvoice at international rates--can anyone give me a short list for consideration?`
`01:01.24`	`blitzrage`	`bkw__: I like you as a person... but I haven't seen a productive comment from you in months`
`01:02.26`	`SwK`	`babyju home many minutes you need?`
`01:03.05`	`bkw__`	`blitzrage, its ok. :P`
`01:03.13`	`bkw__`	`but I wasn't aware that anything I say has to be productive?`
`01:03.13`	`babyju`	`300-400 minutes`
`01:03.41`	`blitzrage`	`bkw__: suppose not...`
`01:03.53`	`bkw__`	`I just don't like the direction Asterisk and Digium as a company is headed`
`01:03.55`	`blitzrage`	`seems kind of a waste of time to hang out in here then`
`01:04.00`	`blitzrage`	`bkw__: fair enough`
`01:04.09`	`bkw__`	`I still have to support Asterisk for a bit longer`
`01:04.14`	`JunK-Y`	`blitzrage: i see a lot of productive comments in months, but i dont like ya as a person :P`
`01:04.20`	`blitzrage`	`JunK-Y: lol`
`01:04.27`	`blitzrage`	`JunK-Y: you rock :)`
`01:04.46`	`bkw__`	`blitzrage, and btw I was joking but apparently it was hard to tell via IRC :P`
`01:04.51`	`SwK`	`yeah well I make a lotta productive comments and I like me`
`01:04.59`	`blitzrage`	`bkw__: you need to write things in <sarcasm></sarcasm> :)`
`01:05.08`	`blitzrage`	`SwK: every night...`
`01:05.08`	`*** join/#asterisk MikeJ (n=mikej@d14-69-8-30.try.wideopenwest.com)`
`01:05.17`	`SwK`	`yep`
`01:05.20`	`SwK`	`whats up blitz`
`01:05.30`	`blitzrage`	`nada much man... just about to go and play with minisplat`
`01:05.38`	`bkw__`	`blitzrage, Well really Asterisk in the Enterprise?? Are they kidding or do they have a code base that we know nothing about`
`01:05.43`	`blitzrage`	`was working on a new website, and thinking I might install gallery later`
`01:05.46`	`JunK-Y`	`blitzrage: julie wants to know how much u liked her tylenol, extra-kodeine?`
`01:05.53`	`SwK`	`oh btw looks like telic may h ave actually fixed that reinvite problem`
`01:06.15`	`blitzrage`	`JunK-Y: was ok I guess -- doesn't mess me up like other people... I have a fairly high drug tolerance`
`01:06.26`	`blitzrage`	`SwK: no way!`
`01:06.29`	`bkw__`	`blitzrage, you a pot head? NEVER :P`
`01:06.36`	`blitzrage`	`bkw__: we weren't talking about pot`
`01:06.41`	`SwK`	`blitzrage atleast they arent dropping calls now`
`01:06.44`	`bkw__`	`oh IC`
`01:06.45`	`bkw__`	`:P`
`01:06.47`	`JunK-Y`	`blitzrage: too much experience huh!`
`01:06.52`	`blitzrage`	`I was talking about pharmaceutical drugs`
`01:07.04`	`blitzrage`	`SwK: HBF went down today though, lol`
`01:07.13`	`bkw__`	`OH`
`01:07.18`	`bkw__`	`then I do too`
`01:07.25`	`bkw__`	`blitzrage, you going to VON ?`
`01:07.35`	`blitzrage`	`bkw__: nah, I need a break from flying :)`
`01:08.31`	`AvoidingDeadlock`	`AVOID AVOID`
`01:09.38`	`*** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net)`
`01:09.58`	`Druken`	`thank god for anti lock brakes`
`01:11.13`	`SwK`	`HBF?`
`01:12.15`	`blitzrage`	`Happy Bun Fest`
`01:14.48`	`EyeCue`	`thank god for driver 'assist' technologies holding drivers hands.`
`01:14.49`	`EyeCue`	`:\|`
`01:17.22`	`blitzrage`	`I love all that technology. There is simply stuff the computer can evaluate and react to faster than any professional driver`
`01:17.36`	`blitzrage`	`and most people on the road are far from professional :)`
`01:17.41`	`EyeCue`	`i find its based on a flawed logic though`
`01:17.51`	`EyeCue`	`they should be taught to be professional and courteous`
`01:17.57`	`blitzrage`	`hell, I'm in favour of retesting every 5 years, and more advanced graduated licensing with harder tests`
`01:18.05`	`EyeCue`	`amen`
`01:18.17`	`blitzrage`	`EyeCue: but emergencies don't usually care about that`
`01:18.30`	`EyeCue`	`im all about prevention, not cures`
`01:18.31`	`blitzrage`	`EyeCue: stopping distance and control is significatly improved with ABS`
`01:18.32`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`01:18.32`	`*** mode/#asterisk [+o mog] by ChanServ`
`01:18.44`	`EyeCue`	`most people can learn in a shot period of time to not slam their breaks and lock up`
`01:18.51`	`blitzrage`	`mog, mog, he's our man, if he can't do it...`
`01:19.03`	`mog`	`?`
`01:19.13`	`evilrabbi`	`azns can`
`01:19.34`	`blitzrage`	`EyeCue: it's not even that... ABS gives you control over the car with the shortest stopping distance -- you won't get that from feathering or even not locking the brakes`
`01:20.08`	`EyeCue`	`fruitless discussion, abs fair enough, its the proliferation of other systems im talkin about`
`01:20.22`	`EyeCue`	`they had their place with soccer moms and weekend 4wd'ers, but meh.`
`01:20.36`	`blitzrage`	`I saw a test with two BMW 700 series cars. One had crazy advanced traction control, and another had it turned off (exact car). Two professional drivers were on a course, and the one with the technology on was able to drive significantly faster with a great deal more control than the one without`
`01:20.57`	`blitzrage`	`EyeCue: I'm not sure what you mean?`
`01:23.06`	`*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)`
`01:26.39`	`*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`01:26.45`	`*** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)`
`01:26.48`	`Qwell`	`brimstone: random`
`01:31.45`	`Strom_C`	`catsex`
`01:32.14`	`*** join/#asterisk Cherebrum (n=jgarland@ares.jasongarland.com)`
`01:37.57`	`brimstone`	`had a caller today say that no one would help him in here with his *.ima not found problem`
`01:38.52`	`brimstone`	`thanks file!`
`01:39.03`	`file`	`...yesterday in fact!`
`01:39.05`	`brimstone`	`apparently file can do it if mog can't...`
`01:39.10`	`brimstone`	`o'rly!?`
`01:39.21`	`mog`	`?`
`01:39.22`	`mog`	`ima?`
`01:39.23`	`brimstone`	`wait, yesterday File Standard Time?`
`01:39.41`	`Strom_C`	`the Makefile was looking for the *.ima files in the wrong place`
`01:39.50`	`Strom_C`	`so I bugged file until he fixed it`
`01:39.54`	`file`	`the octasic firmware`
`01:40.13`	`mog`	`i dont ever remember being asked`
`01:40.19`	`Strom_C`	`however, I now owe file two point four million muffins`
`01:40.29`	`brimstone`	`mhmmm muffins`
`01:40.30`	`Strom_C`	`mog: it was sunday and you werent around`
`01:40.50`	`brimstone`	`file, (psst, go with lump sum)`
`01:41.58`	`Qwell`	`Strom_C: two point four million?`
`01:42.04`	`Qwell`	`Is that like 2.4000000`
`01:42.23`	`Strom_C`	`two million four hundred thousand`
`01:42.30`	`brimstone`	`sweet, we get muffins!`
`01:42.46`	`Qwell`	`two million four hundred thousand fourty-two`
`01:43.20`	`file`	`yay gigamuffins!`
`01:43.25`	`Qwell`	`jiga`
`01:44.57`	`brimstone`	`examuffins`
`01:45.03`	`Strom_C`	`http://xkcd.com/c124.html`
`01:45.31`	`brimstone`	`not eating dinner`
`01:45.39`	`brimstone`	`not watching Voyager with the roommates`
`01:45.44`	`blitzrage`	`:)`
`01:45.48`	`blitzrage`	`Voyager? ewwwww`
`01:45.52`	`brimstone`	`not working on homework`
`01:45.52`	`file`	`oic`
`01:45.59`	`blitzrage`	`hell... Star Trek/Wars... ewwww`
`01:46.07`	`brimstone`	`not getting up to get more food and sitdown to watch his own movie`
`01:46.27`	`brimstone`	`blitzrage, stargate?`
`01:46.46`	`blitzrage`	`lets just say sci-fi is a thumbs down`
`01:46.51`	`brimstone`	`aww`
`01:47.08`	`blitzrage`	`way too nerdy for me :)`
`01:47.13`	`*** join/#asterisk FullService (n=jiggaman@CPE0002724fc55e-CM000f211fd29c.cpe.net.cable.rogers.com)`
`01:47.15`	`Dimik_`	`pft.`
`01:47.29`	`FullService`	`Anyone know anything about this company: "Mino Wireless"`
`01:47.38`	`FullService`	`I dont understand how their service works`
`01:47.44`	`JT`	`blitzrage will have a riot on his hands saying that in here`
`01:48.01`	`blitzrage`	`JT: I'm not the typical nerd :)`
`01:48.09`	`brimstone`	`you wouldn't like the conversation at work then blitzrage`
`01:48.18`	`blitzrage`	`brimstone: I imagine not`
`01:48.23`	`brimstone`	`great, now i have to explain it`
`01:49.03`	`blitzrage`	`why is that?`
`01:49.05`	`JT`	`one of these "anti-sci-fi nerds" right... i get it`
`01:49.06`	`blitzrage`	`I don't want to know :)`
`01:49.15`	`blitzrage`	`JT: I'm not a nerd :)`
`01:49.21`	`blitzrage`	`I don't even have glasses :)`
`01:49.25`	`Qwell`	`blitzrage: contacts? :p`
`01:49.27`	`JT`	`amazing`
`01:49.32`	`JT`	`contacts are available`
`01:49.33`	`JT`	`zing`
`01:49.34`	`blitzrage`	`Qwell: nope, 20/20`
`01:49.43`	`Qwell`	`...corrective eye surgery?`
`01:49.45`	`JT`	`geek in denial :P`
`01:50.13`	`blitzrage`	`Qwell: nope, all natural! :)`
`01:50.22`	`blitzrage`	`JT: I said I wasn't a nerd -- not a geek`
`01:50.32`	`blitzrage`	`totally different`
`01:50.49`	`JT`	`fine`
`01:50.54`	`JT`	`"anti-sci-fi geek"`
`01:51.23`	`blitzrage`	`heh`
`01:51.23`	`JT`	`i thought you implyed you were some type of nerd with blitzrage> JT: I'm not the typical nerd :)`
`01:51.23`	`blitzrage`	`ah... yes, error on my part`
`02:08.05`	`*** join/#asterisk spr1te (i=spr1te@194.187.130.227)`
`02:14.20`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`02:17.00`	`*** join/#asterisk Horvath (i=grabowsk@i.use.efnut.com)`
`02:17.16`	`*** join/#asterisk b1ch0 (n=aa@host-206-107-150-177.acelerate.net)`
`02:18.23`	`Horvath`	`How can I set my asterisk console so it's always in verbose mode? I tryed using logger.conf and console => console => notice,warning,error,verbose but its not working or not the same as set verbose 10`
`02:18.59`	`Horvath`	`With only one console => of course`
`02:19.27`	`*** join/#asterisk Skyelar (n=planet@222-152-83-181.jetstream.xtra.co.nz)`
`02:20.16`	`Horvath`	`I guess I can just do asterisk -rv but I thought logger.conf should work`
`02:21.15`	`JunK-Y`	`if i understand, how to remove notice,warning?`
`02:22.29`	`*** join/#asterisk grexk (n=grexk@124.107.72.42)`
`02:22.35`	`Horvath`	`No, I don't want to remove notice and warning`
`02:23.04`	`brimstone`	`Horvath, once asterisk has verbose set, it should stay set`
`02:23.25`	`JunK-Y`	`or connect with asterisk -v , then do set verbose 10`
`02:23.59`	`*** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net)`
`02:24.18`	`Horvath`	`brimstone: true`
`02:24.41`	`Horvath`	`anyways just thought it was weird that logger.conf didin't seem to work at all for me`
`02:25.02`	`Horvath`	`no affect on the verbosity of console messages`
`02:25.04`	`brimstone`	heh, put `asterisk -vvvrx "set verbose 3"` in a startup script then :P
`02:25.17`	`rpm`	`is there a g729a codec for asterisk-trunk, it seems the module does not export all of the correct symbols.`
`02:25.36`	`Qwell`	`rpm: not yet`
`02:25.38`	`Qwell`	`I don't think`
`02:25.39`	`brimstone`	`yeah, why is g729 for trunk broken?`
`02:25.40`	`*** join/#asterisk amdtech (n=amd011@ip70-179-175-131.dl.dl.cox.net)`
`02:26.21`	`brimstone`	`rpm, just curious, why are you running trunk and not branch-1.2?`
`02:26.44`	`rpm`	`brimstone: i wanted to play with some of the new features, smdi, manager interface (ajam)`
`02:27.00`	`brimstone`	`ah, that's cool`
`02:32.43`	`rpm`	`i wish unixodbc didn't hate me so much, it seems enabling tracing/debugging never works.`
`02:33.08`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`02:34.40`	`*** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`02:35.12`	`*** join/#asterisk lordbaron (n=redbaron@host55-226.rancor.birch.net)`
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`02:36.05`	`*** part/#asterisk Cherebrum (n=jgarland@ares.jasongarland.com)`
`02:37.08`	`lordbaron`	`my tdm400p won't disconnect when far-end hangsup. Digium Support tells me to get the telco to provide KewlStart. SBC doesn't have that option (nor COD, or the like)..what are my options?`
`02:37.25`	`*** join/#asterisk jayb (n=jaybinks@59.167.212.49)`
`02:37.40`	`brimstone`	`lordbaron, callprogress? busydetect?`
`02:37.45`	`jayb`	`can anyone provide a little help with "`
`02:37.48`	`jayb`	`"iax2 show netstats"`
`02:38.02`	`jayb`	`cant find much documentation, on how to understand the data`
`02:38.14`	`jayb`	`EG... what does -1 jitter mean`
`02:38.27`	`lordbaron`	`brimstone: busydetect ..tried multiple options, but there is no busy signal. I can listen to what happens. CallProgress doesn't work either`
`02:38.36`	`lordbaron`	`I can hear the audible 'click' when far-end hangsup`
`02:38.39`	`jayb`	`and I have a few 1000 RTT Values... whats that mean (surely its not 1sec ping time)`
`02:39.18`	`lordbaron`	`brimstone: for busydetect ---should I hear the busy signal?`
`02:39.25`	`brimstone`	`yeah`
`02:39.28`	`brimstone`	`i think...`
`02:39.48`	`lordbaron`	`I get another dialtone, then a message 'hangup or make a call', then a 'fast busy'`
`02:40.03`	`lordbaron`	`tried the timing on the fast busy, but it would never happen`
`02:40.14`	`lordbaron`	`it was 100ms on, 70ms off`
`02:40.31`	`lordbaron`	`used audacity to measure`
`02:40.40`	`brimstone`	`is there anything that that the telco does at the end of the call?`
`02:40.46`	`brimstone`	`polarity change? anything?`
`02:41.21`	`lordbaron`	`no polarity. They put on disconnect supervision. This now produces a click noise, and causes the buttset light to flick`
`02:41.31`	`Strom_C`	`that would be a battery drop`
`02:41.35`	`lordbaron`	`yes`
`02:41.38`	`lordbaron`	`I see that in the debug`
`02:41.52`	`lordbaron`	`I also get that on the first ring`
`02:41.55`	`lordbaron`	`when dialing`
`02:41.57`	`hads`	`Kewlstart will work with that`
`02:42.21`	`lordbaron`	`when I set my signalling to KewlStart, I cannot make outbound calls`
`02:42.28`	`lordbaron`	`it disconnects after the first ring`
`02:42.36`	`lordbaron`	`I think this is because of the battery change on the first ring`
`02:42.57`	`lordbaron`	`DIALSTATUS=answered`
`02:43.11`	`Skyelar`	`lordbaron: do you have answeronpolarityswitch=yes?`
`02:43.18`	`hads`	`DIALSTATUS is always answered with analog`
`02:43.26`	`lordbaron`	`skelar: no`
`02:43.43`	`lordbaron`	`Skyelar: Should I?`
`02:44.09`	`jayb`	`can somone offer me a suggestion (re above)...`
`02:44.18`	`Skyelar`	`lordbaron: it'd be worth a try`
`02:44.24`	`jayb`	`I just want to try and measure network stats, of each of my channels..`
`02:44.29`	`jayb`	`jitter, RTT etc...`
`02:44.36`	`lordbaron`	`with kewlstart signalling?`
`02:44.41`	`Skyelar`	`lordbaron: yeah`
`02:45.03`	`Skyelar`	`lordbaron: incoming calls were working, and getting hung up correctly, but outbound ones would hang up as soon as the remote answered?`
`02:45.24`	`lordbaron`	`yes`
`02:45.34`	`lordbaron`	`actually before remote answered`
`02:45.40`	`lordbaron`	`when remote first ring occurred`
`02:45.59`	`Skyelar`	`lordbaron: hmm... might still be worth a try`
`02:46.09`	`lordbaron`	`Skyelar: trying now`
`02:48.50`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`02:49.19`	`*** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net)`
`02:49.23`	`lordbaron`	`Skyelar: same results`
`02:50.33`	`brimstone`	`well, i guess more the "solution"`
`02:51.15`	`lordbaron`	`I set signalling to ks->and changed answeronpolarityswitch=yes`
`02:51.27`	`lordbaron`	`inbound, caller hangs up in voice mail..works`
`02:51.40`	`lordbaron`	`outbout..caller hangs up, asterisk does not detect this`
`02:51.45`	`lordbaron`	`will keep the line open`
`02:52.04`	`lordbaron`	`when set to ks - dialing outbound causes a hangup immedidate`
`02:52.06`	`brimstone`	`asterisk needs a silence detect or something`
`02:52.06`	`lordbaron`	`on first ring`
`02:52.32`	`lordbaron`	`so with kewlstart, I get hangups working on inbound proper, but I can't make outbound calls`
`02:54.38`	`*** join/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net)`
`02:54.59`	`rene-`	`hello, how can i use setmusiconhold to disable music on hold for a particular call?`
`02:55.44`	`rene-`	`basically i have an inbound queue served with the agentlogin application. i want agents to listen to music while waiting for calls but i dont want callers to listen to any music`
`02:56.05`	`lordbaron`	`I just noticed: During dialing is when the hangup occurs. I have a dialplan that gives me a zap direct. Then I dial when I get the dialtone, and midway during dial, it hangsup`
`02:57.25`	`brimstone`	`lordbaron, try with a call file or something simple`
`02:58.13`	`brimstone`	`rene-, you can make a queue have ringing instead of moh`
`03:00.04`	`rene-`	`brimstone good idea`
`03:01.07`	`Skyelar`	`lordbaron: you did a full restart of Asterisk after changing the config?`
`03:01.11`	`lordbaron`	`brimstone: same result`
`03:01.26`	`lordbaron`	`Skyelar: yes...and a ztcfg -vvv`
`03:01.48`	`Skyelar`	`lordbaron: hmm... sounds like your telco is doing something odd with polarity reversal/battery drops`
`03:02.16`	`JT`	`anyone know if the debian testing distribution of asterisk has any hardware support?`
`03:02.23`	`brimstone`	`man, telco's are so... picky sometimes`
`03:02.40`	`Skyelar`	`lordbaron: I don't have an immediate answer in this case - you'd have to spend some time figuring out exactly when they do it, and possibly hack Asterisk to support the scheme they're using (or find out if they support other options)`
`03:02.41`	`*** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net)`
`03:02.42`	`lordbaron`	`Aug 28 23:02:25 asterisk1 kernel: NO BATTERY on 2/2!`
`03:02.46`	`brimstone`	`JT, probably, but i wouldn't use it`
`03:02.54`	`JT`	`brimstone: yeah?`
`03:03.03`	`brimstone`	`nope`
`03:03.17`	`JT`	`i've installed it, doesn't look like there's any zap hardware support as far as i can see`
`03:03.20`	`lordbaron`	`<PROTECTED>`
`03:03.21`	`JT`	`any reason?`
`03:03.37`	`lordbaron`	`Skyelar: That is where I am headed next I believe`
`03:03.46`	`brimstone`	`package managers wouldn't let one of digium's own maintain packages :P`
`03:04.03`	`lordbaron`	`any ideas what this pattern should be ?`
`03:04.46`	`brimstone`	`lordbaron, if you do have to hack something, it'd be totally awesome if you submitted it to bugs.digium.com, provided it's somewhat elegant`
`03:05.32`	`lordbaron`	`youmeanputinspacesandtabsandthelike?`
`03:05.58`	`brimstone`	`heh, yeah, and it's primatives are settings in zapata.conf or something`
`03:06.07`	`brimstone`	`not like "usingsbc=yes"`
`03:06.12`	`JT`	`brimstone: is that an issue?`
`03:06.23`	`brimstone`	`but like "batterydroponhangup=yes"`
`03:06.36`	`rpm`	`bleh, no wonder my voicemail is being weird, the default build arguments are to -DODBC_STORAGE`
`03:07.01`	`lordbaron`	`do you know anyone on #dev that has experience in the wcfx? modules?`
`03:07.12`	`brimstone`	`JT, eh, i didn't like it, it could have other problems too maybe`
`03:07.19`	`brimstone`	`hardware support for one`
`03:07.37`	`lowlevel`	`think i'm gonna put on some foreigner :\|`
`03:07.53`	`*** join/#asterisk tengulre (n=tengulre@61.185.224.66)`
`03:08.42`	`rpm`	`lowlevel: that sounds cold as ice.`
`03:08.57`	`JT`	`brimstone: debian packages are often maintained seperately from the corporates who do software packages`
`03:09.05`	`lowlevel`	`rpm: urgent... urgent... emergency :\|`
`03:09.16`	`JT`	`but yeah, i thought you had more conrete reasons for not using the packaged version`
`03:09.32`	`brimstone`	`eh, i don't like the other 3rd party stuff built into it?`
`03:11.10`	`JT`	`such as?`
`03:11.15`	`brimstone`	`bristuff`
`03:11.21`	`lowlevel`	`jukebox hero....`
`03:12.52`	`*** join/#asterisk atapi2 (n=virgill4@c-69-180-119-156.hsd1.fl.comcast.net)`
`03:14.43`	`JT`	`sure`
`03:14.50`	`JT`	`i don't need bristuff at home`
`03:15.03`	`JT`	`and debian has asterisk-bristuff instead of asterisk-classic available`
`03:24.24`	`jayb`	`soooo... can anyone offer me some help understanding`
`03:24.33`	`jayb`	`"iax2 show netstats"`
`03:24.56`	`jayb`	`(or suggest a similar way, to monitor jitter & latency on calls)`
`03:29.25`	`*** join/#asterisk Jenocin (i=jenocin@99.3.118.70.cfl.res.rr.com)`
`03:40.22`	`Jenocin`	`anyone an op in #freepbx?`
`03:40.22`	`*** join/#asterisk Corydon76-home (i=indigo@pdpc/supporter/sustaining/Corydon76-home)`
`03:40.23`	`*** mode/#asterisk [+o Corydon76-home] by ChanServ`
`03:40.23`	`docelmo`	`No respected asterisk user would be caught dead in there..`
`03:40.23`	`jenocin`	`hehe why all the turmoil between the two groups?`
`03:40.23`	`lowlevel`	`no ieda`
`03:40.23`	`CunningPike`	`jenocin: FreePBX is the Microsoft Bob of PBXs`
`03:40.23`	`lowlevel`	`hAHAHAAHAH`
`03:40.31`	`lowlevel`	`yeah ok, that did it, I'm definatly not checking it out now`
`03:40.32`	`lowlevel`	`;)`
`03:41.11`	`jenocin`	`hehe so what pbx systems you all using?`
`03:41.18`	`CunningPike`	`jenocin: And a lot of us spend a lot time time helping victims with seemingly nonsensical symptoms only to discover that they're running the damn thing`
`03:41.24`	`lowlevel`	`< Panasonic TDA-100`
`03:41.25`	`CunningPike`	`s/lot time time/lot of time/`
`03:41.49`	`CunningPike`	`jenocin: Asterisk, of course`
`03:42.30`	`jenocin`	`k, trying to grasp all the differences here, isn't freepbx just a gui built on top of asterisk?`
`03:43.00`	`brimstone`	`compare it to a frontpage for html`
`03:43.32`	`CunningPike`	`jenocin: It is, but as a GUI, it requires some structure to the underlying files that constrain what you can do with them freehand. That can be OK, but they have implemented all sorts of weirdness that's impossible to debug by hand`
`03:43.42`	`CunningPike`	`brimstone: That is an excellent analogy`
`03:50.49`	`*** join/#asterisk AJaymn (i=AJaymn@70.59.126.204)`
`03:50.49`	`*** part/#asterisk AJaymn (i=AJaymn@70.59.126.204)`
`03:50.49`	`*** join/#asterisk AJaymn (i=AJaymn@70.59.126.204)`
`03:50.49`	`jenocin`	`hehe I hear ya there, sounds like more of a frontpage of the voip world`
`03:50.49`	`Strom_C`	`brimstone: that is BRILLIANT`
`03:50.49`	`Strom_C`	`<3 <3 <3 <3 <3`
`03:50.49`	`jenocin`	`hehe u beat me too it`
`03:50.49`	`CunningPike`	`~freepbx`
`03:50.53`	`jbot`	`well, freepbx is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP)`
`03:50.53`	`lowlevel`	`if you think frontpage is bad, try iWeb.`
`03:50.53`	`jenocin`	`so can you reccomend a decent gui :-P`
`03:50.53`	`CunningPike`	`jenocin: vi`
`03:50.53`	`Strom_C`	`jenocin: vim`
`03:50.53`	`jenocin`	`netobjects fusion`
`03:50.53`	`brimstone`	`gvim :D`
`03:50.53`	`jenocin`	`thats good for editing, but i'm afraid my managers wouldn't wanna use it`
`03:50.53`	`CunningPike`	`jbot: where are you?\`
`03:50.56`	`jbot`	`CunningPike: I think you lost me on that one`
`03:50.56`	`CunningPike`	`jenocin: Your managers shouldn't be messing around with your PBX - do they do that now?`
`03:50.56`	`Strom_C`	`jenocin: your managers have no more business administering the PBX than your secretary has managing space shuttle flights`
`03:50.56`	`CunningPike`	`jenocin: If you are incapable of editing a text file with vi, you probably shouldn't be within an ass' roar of it`
`03:50.56`	`jenocin`	`not administering, but there is the operator panel in freepbx and the reporting module`
`03:50.57`	`jenocin`	`and just for the record, I usually use vim, I'm a freebsd guys`
`03:50.57`	`blitzrage`	`vim!`
`03:50.57`	`CunningPike`	`telnet`
`03:50.57`	`brimstone`	`awk & sed`
`03:50.57`	`CunningPike`	`Nugget must be asleep`
`03:50.57`	`Strom_C`	`balls`
`03:50.57`	`CunningPike`	`Strom_C: balls? Does that require XY-Windows?`
`03:50.57`	`Strom_C`	`exactly`
`03:50.57`	`*** join/#asterisk AJaymn (i=AJaymn@70.59.126.204)`
`03:50.57`	`CunningPike`	`:D`
`03:51.11`	`brimstone`	`~lag`
`03:51.13`	`jbot`	`it has been said that lag is very bad here :(`
`03:51.13`	`CunningPike`	`Welcome back, jbot`
`03:52.56`	`jenocin`	`well jbot, if u followed the stream, I was just trying to find what is being used for reporting and operator type functions`
`03:54.54`	`JT`	`jbot only does as it is instructed`
`03:54.56`	`jbot`	`JT: I think you lost me on that one`
`03:59.17`	`CunningPike`	`file and brimstone?`
`04:00.28`	`file`	`moo?`
`04:00.49`	`*** join/#asterisk Sedorox (i=sedorox@smartserv/cna/Sedorox)`
`04:00.51`	`brimstone`	`baa`
`04:00.57`	`blitzrage`	`honk`
`04:01.03`	`cornet`	`moo`
`04:01.32`	`*** part/#asterisk jenocin (i=jenocin@99.3.118.70.cfl.res.rr.com)`
`04:02.54`	`*** part/#asterisk Here_And_There (n=Here_And@pool-71-246-6-164.phlapa.fios.verizon.net)`
`04:12.56`	`rene-`	`????\`
`04:13.20`	`rene-`	`jbot asterisk as a body part`
`04:13.37`	`rene-`	`jbot: asterisk is also a body part`
`04:13.39`	`jbot`	`okay, rene-`
`04:13.58`	`rene-`	`asterisk~`
`04:14.02`	`rene-`	`~asterisj`
`04:14.05`	`rene-`	`~asterisk`
`04:14.07`	`jbot`	`asterisk is the best free PBX in the world. a body part`
`04:14.25`	`brimstone`	`yeah yeah, stupid bot, now put it back :)`
`04:14.28`	`rene-`	`jbot: asterisk is not a body part`
`04:14.29`	`jbot`	`...but asterisk is already something else...`
`04:14.35`	`CunningPike`	`no, jbot, asterisk is the best free PBX in the world (www.asteriskorg)`
`04:14.42`	`rene-`	`thx`
`04:14.50`	`rene-`	`~asterisk`
`04:14.51`	`jbot`	`asterisk is the best free PBX in the world. a body part`
`04:15.04`	`CunningPike`	`no, jbot, asterisk is the best free PBX in the world (www.asterisk.org)`
`04:15.12`	`rene-`	`~asterisk`
`04:15.14`	`jbot`	`asterisk is the best free PBX in the world. a body part`
`04:15.31`	`brimstone`	`jbot, asterisk is the best free PBX in the world (www.asterisk.org)`
`04:15.32`	`jbot`	`...but asterisk is already something else...`
`04:15.39`	`rene-`	`jbot: no, asterisk is the best free PBX in the world`
`04:15.40`	`jbot`	`rene-: okay`
`04:15.42`	`brimstone`	`jbot, so cchange it back!`
`04:15.45`	`rene-`	`~asterisk`
`04:15.46`	`jbot`	`methinks asterisk is the best free PBX in the world`
`04:16.00`	`rene-`	`vandalize =off`
`04:27.04`	`*** join/#asterisk ma_dzen (n=ma_dzen@217.66.17.141)`
`04:33.53`	`*** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com)`
`04:34.04`	`*** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net)`
`04:35.01`	`rene-`	`~jbot`
`04:35.03`	`jbot`	`well, jbot is only marginally useful at best, He got a C- on his Turing Test, or a complete idiot, or a dolt`
`04:35.24`	`rene-`	`~irq`
`04:35.26`	`jbot`	`rumour has it, irq is (Interrupt request) This is a gateway for devices to request time from the processor to do their jobs. For instance, every time you hit a key on your keyboard an interrupt is generated on the keyboard IRQ. This is mainly only a concern for PC users.`
`04:35.41`	`rene-`	`~deadlock`
`04:35.55`	`rene-`	`~BOFH`
`04:35.56`	`jbot`	`from memory, bofh is Bastard Operator From Hell - see http://www.theregister.co.uk/odds/bofh/`
`04:36.05`	`rene-`	`~digg`
`04:36.20`	`*** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com)`
`04:37.07`	`rene-`	`jbot: FreePBX is the Microsoft Bob of PBXs`
`04:37.09`	`jbot`	`...but freepbx is already something else...`
`04:37.26`	`rene-`	`~pra`
`04:39.55`	`CunningPike`	`rene-: You liked that? :)`
`04:40.27`	`CunningPike`	`jbot, FreePBX is also the Microsoft Bob of PBXs`
`04:40.29`	`jbot`	`CunningPike: okay`
`04:40.48`	`CunningPike`	`~freepbx`
`04:40.49`	`jbot`	`freepbx is, like, NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP), or the Microsoft Bob of PBXs`
`04:41.44`	`r0d3nt`	`hi`
`04:41.45`	`rene-`	`hehe i liked it`
`04:41.46`	`r0d3nt`	chan_zap.c: In function `pri_dchannel':
`04:41.46`	`r0d3nt`	chan_zap.c:9025: error: structure has no member named `call'
`04:41.46`	`r0d3nt`	`make[1]: *** [chan_zap.o] Error 1`
`04:41.49`	`r0d3nt`	`any ideas ??`
`04:41.50`	`CunningPike`	`jbot, no, freepbx is the Microsoft BOB of PBXs and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP)`
`04:41.51`	`jbot`	`CunningPike: okay`
`04:42.03`	`rene-`	`:)`
`04:42.07`	`CunningPike`	`;)`
`04:42.10`	`Strom_C`	`<PROTECTED>`
`04:42.11`	`jbot`	`Strom_C: okay`
`04:42.22`	`Strom_C`	`sorry, had to correct a pluralization error :)`
`04:42.29`	`Strom_C`	`jbot, no, freepbx is the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP)`
`04:42.31`	`jbot`	`okay, Strom_C`
`04:42.36`	`Strom_C`	`and a punctuation error :)`
`04:42.59`	`Strom_C`	`MUFFINY DISGUISE!`
`04:44.24`	`rene-`	`~537`
`04:46.33`	`rene-`	`this is new`
`04:46.35`	`rene-`	`~modems`
`04:46.37`	`jbot`	`modems is, like, something you can not use as an fxo interface under asterisk`
`04:46.53`	`rpm`	`this isn't right, it takes about 3-4 seconds for my grandstream bt-100 (test phone) to recieve rtp traffic g711.u after a call has been established`
`04:50.57`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`04:55.04`	`*** part/#asterisk amdtech (n=amd011@ip70-179-175-131.dl.dl.cox.net)`
`04:55.58`	`*** part/#asterisk shidan (n=shidan@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com)`
`04:56.19`	`*** join/#asterisk sxpert (n=sxpert@navsys.org)`
`05:04.09`	`r0d3nt`	`dwgvoip init.d # asterisk -r`
`05:04.09`	`r0d3nt`	`Asterisk 1.2.11, Copyright 1999 - 2006 Digium, Inc. and others.`
`05:04.09`	`r0d3nt`	`Andrew says (10:03 PM):`
`05:04.09`	`r0d3nt`	`it's ALIVE !!!!!!!!!!!!!!!!!!!!!!!!!!!`
`05:06.57`	`fafnir`	`Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.`
`05:07.14`	`fafnir`	`does it go "ABSOLUTELY NO INFORMATION HERE"`
`05:07.59`	`JT`	`fafnir: what?`
`05:08.28`	`fafnir`	`when you start asterisk... it says Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.`
`05:08.38`	`fafnir`	`i havnt typed 'show warranty' yet`
`05:08.45`	`Strom_C`	`fafnir: it prints out a giant disclaimer of liability`
`05:08.49`	`fafnir`	`ah`
`05:08.55`	`JT`	`ok, so what were you trying to get at?`
`05:09.02`	`fafnir`	`it was a joke.`
`05:09.46`	`fafnir`	`should be a good read`
`05:09.47`	`JT`	`seems to have missed the mark :]`
`05:10.01`	`fafnir`	`youre either too nerdy, or not nerdy enough`
`05:10.11`	`fafnir`	`in either case, you were not just the right amount of nerdy`
`05:11.04`	`JT`	`fafnir's jokes require a very specific level or nerdyness`
`05:11.11`	`JT`	`to suit their nerd niches`
`05:11.25`	`fafnir`	`Actually, i just amuse myself`
`05:11.28`	`fafnir`	`skrew you other guys`
`05:11.58`	`grexk`	`How does actually VPN affects asterisk?`
`05:12.00`	`*** join/#asterisk KaiHanari (n=Kai@stjhnf01-23-142163031217.nf.sympatico.ca)`
`05:12.12`	`KaiHanari`	`is it possible to have a live audio stream as the music on hold?`
`05:12.26`	`grexk`	`I think so`
`05:13.03`	`Skyelar`	`KaiHanari: yes, look at the "custom" MOH mode`
`05:13.10`	`*** join/#asterisk enjay- (n=yea@wsip-24-249-169-168.ph.ph.cox.net)`
`05:13.29`	`*** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net)`
`05:13.36`	`grexk`	`http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ICES`
`05:13.39`	`KaiHanari`	`Skyelar, got a link handy? or a specific site to search on?`
`05:13.54`	`KaiHanari`	`grexk, thanks`
`05:14.00`	`grexk`	`np`
`05:14.09`	`JT`	`is there any simple way of making asterisk forward a call so that when it receives say a pstn call, it does not pick up the call, until the other end that it dials, to forward the call, picks up?`
`05:14.33`	`KaiHanari`	`grexk, wait, thats backwards`
`05:14.48`	`grexk`	`what do you mean?`
`05:14.51`	`KaiHanari`	`thats for sending a stream of a confrence`
`05:15.02`	`KaiHanari`	`i mean have the music thats playing on hold, come from a alive stream`
`05:15.18`	`grexk`	`well just search on that site`
`05:16.36`	`Skyelar`	`KaiHanari: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf`
`05:16.47`	`Skyelar`	`see the`
`05:16.56`	`Skyelar`	`"Asterisk 1.2 Custom Applications" bit (damn enter key)`
`05:19.33`	`enjay-`	`Im experiencing and issue with a zap t1 not dialing anyone mind checking this out: http://pastebin.ca/152806`
`05:19.45`	`KaiHanari`	`Skyelar, Thanks! after a bit of scrolling i found exactly what i wanted`
`05:20.57`	`Skyelar`	`KaiHanari: works nicely, as long as you get your audio formats correct`
`05:24.35`	`*** join/#asterisk doolph (n=doolph@200.124.28.155)`
`05:29.59`	`*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`05:30.23`	`Qwell`	`Strom_C: ping`
`05:30.31`	`enjay-`	`Im experiencing and issue with a zap t1 not dialing anyone mind checking this out: http://pastebin.ca/152806`
`05:31.00`	`*** join/#asterisk daysmen3 (n=primus@host86-140-41-79.range86-140.btcentralplus.com)`
`05:31.17`	`*** join/#asterisk phalacee (n=Sunforge@202.3.110.33)`
`05:33.59`	`CunningPike`	`enjay-: I don't think your zaptel.conf is correct - you need a dchan and bchan line after each span line`
`05:34.15`	`fafnir`	`anyone seen shido around recently?`
`05:34.26`	`CunningPike`	`~seen shido`
`05:34.41`	`jbot`	`shido <n=shido@d221-68-216.commercial.cgocable.net> was last seen on IRC in channel #asterisk, 213d 10h 28m 59s ago, saying: 'someone is getting a nice phat check, jbalcomb'.`
`05:34.41`	`Strom_C`	`Qwell: ack`
`05:34.41`	`Qwell`	`Strom_C: know anything about this cellphone "grace period" bill?`
`05:34.51`	`Qwell`	`~seen shido6`
`05:34.54`	`jbot`	`shido6 <n=shido6@d221-68-200.commercial.cgocable.net> was last seen on IRC in channel #asterisk, 5d 6h 44s ago, saying: 'any max tnt & sip users?'.`
`05:35.07`	`CunningPike`	`Why do people do that?`
`05:35.27`	`Strom_C`	`Qwell: I know that in california you have thirty days`
`05:35.34`	`Qwell`	`Strom_C: nah, not for over a year`
`05:35.38`	`Qwell`	`it was repealed`
`05:36.06`	`Strom_C`	`oh? because when I got this phone a few months ago, tmobile told me I had thirty days`
`05:36.12`	`Qwell`	`They lied. ;/`
`05:36.15`	`Qwell`	`well, no`
`05:36.23`	`Qwell`	`They told me the same, but it isn't to cancel, afaik`
`05:36.31`	`Qwell`	`It's if you don't like the phone, you can trade it in`
`05:36.47`	`Qwell`	`You can get a refund on it, and get another one or whatever`
`05:37.04`	`*** join/#asterisk ltd (n=z@202-161-16-50.dyn.iinet.net.au)`
`05:37.52`	`enjay-`	`CunningPike: I have a shared D channel in that trunkgroup.. 24B's 23B's 1 shared D.`
`05:38.01`	`Qwell`	`I was gonna try to find the bill, and email Sprints CEO. Tell him I strongly believe he should vote for it :D`
`05:38.18`	`JT`	`24Bs, 23Bs, contradictory`
`05:38.35`	`Strom_M`	`JT: ?`
`05:38.40`	`CunningPike`	`enjay-: Ewww. So, 1D, 47 B ?`
`05:38.44`	`JT`	`< enjay-> CunningPike: I have a shared D channel in that trunkgroup..`
`05:38.45`	`JT`	`<PROTECTED>`
`05:38.46`	`enjay-`	`right..`
`05:38.54`	`Strom_M`	`JT, two T1s`
`05:38.59`	`JT`	`ah right`
`05:38.59`	`enjay-`	`JT - breaking them out by T1's bud`
`05:39.00`	`JT`	`nevermind`
`05:39.05`	`Strom_M`	`one has 24 B channels, one has 23 Bs and a D`
`05:39.12`	`Strom_M`	`yay, reasoning`
`05:39.44`	`enjay-`	`changing to 23/1 23/1.. dealing with inheretance at this time..`
`05:40.27`	`enjay-`	`but those arent my issue.. its a different circuit..`
`05:45.25`	`Strom_M`	`also, towels`
`05:50.55`	`mover`	`g morning`
`05:55.11`	`*** part/#asterisk grexk (n=grexk@124.107.72.42)`
`05:59.25`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`05:59.45`	`*** join/#asterisk novafirst (n=kosta@wrt1.niclab.com)`
`06:01.04`	`novafirst`	`anyone?`
`06:02.09`	`Strom_M`	`cheesecake`
`06:03.08`	`Snake-Eyes`	`yes please`
`06:04.00`	`novafirst`	`having problem with moh, asterisk 1.2.11`
`06:04.15`	`*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)`
`06:04.32`	`Snake-Eyes`	`i had to make a cheesie remark :P`
`06:04.39`	`novafirst`	`lol`
`06:05.18`	`novafirst`	`I am pist of on poor doc... for the vers. 1.2+`
`06:05.41`	`novafirst`	`most examples are or older versions`
`06:05.53`	`Strom_M`	`novafirst, so quit complaining and ask a question`
`06:06.15`	`novafirst`	`it wouldn't play mp3 moh.`
`06:06.46`	`novafirst`	`I am using x-lite for testing and vm is working fine but moh is not`
`06:06.59`	`Strom_M`	`what kind of moh are you using?`
`06:07.03`	`Strom_M`	`native?`
`06:07.06`	`novafirst`	`yes`
`06:07.09`	`novafirst`	`files`
`06:08.10`	`novafirst`	`I can postbin conf file`
`06:09.51`	`*** join/#asterisk marta (n=a@host184-5.pool8711.interbusiness.it)`
`06:09.53`	`marta`	`hello`
`06:09.54`	`novafirst`	`the log file show that it is playing and stoping moh properly but there is no sound. Maybe x-lite is not compatible with mp3 moh`
`06:10.10`	`Strom_M`	`what files are you using`
`06:11.52`	`novafirst`	`I downloaded mp3 from the web - signate.com`
`06:12.15`	`novafirst`	`I was told they are Asterisk ready`
`06:12.32`	`Strom_M`	`where did you put them`
`06:13.25`	`*** join/#asterisk Rahail (n=dd@ip66-104-51-162.z51-104-66.customer.algx.net)`
`06:13.48`	`novafirst`	`http://pastebin.ca/152855`
`06:15.32`	`Strom_M`	`ok then`
`06:15.53`	`*** join/#asterisk novafirst (n=kosta@wrt1.niclab.com)`
`06:16.01`	`novafirst`	`I am back.`
`06:16.31`	`Strom_M`	`mode should be quietmp3`
`06:16.38`	`novafirst`	`so I put all mp3 for moh in var-lib-asterisk-mohmp3`
`06:17.24`	`novafirst`	`I tried changing mode to quietmp3 and still no sound`
`06:17.36`	`Strom_M`	`did you reload?`
`06:18.14`	`novafirst`	`sure, yes I did`
`06:18.53`	`Strom_M`	`try a full restart?`
`06:19.04`	`novafirst`	`I did that as well`
`06:20.24`	`Strom_M`	`maybe the files are borked`
`06:22.22`	`novafirst`	`I played the mp3 files localy and it works fine.`
`06:25.08`	`novafirst`	`did anyone tested x-lite with moh ?`
`06:33.37`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`06:35.39`	`*** join/#asterisk fdepp (n=fdepp@c2cpc3.camptocamp.com)`
`06:37.35`	`*** join/#asterisk Assid (i=assid@203.115.83.214)`
`06:41.57`	`*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)`
`06:46.31`	`*** join/#asterisk Frogdude (i=Frogdude@c-24-16-72-159.hsd1.mn.comcast.net)`
`06:46.43`	`Frogdude`	`hola`
`06:47.50`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`06:47.53`	`Frogdude`	`is there a way to get asterisk to log information about a call to a file or to open a database and directly write info to it?`
`06:49.37`	`Frogdude`	`how can I save information about a call so that I can access it outside of asterisk?`
`06:51.47`	`fdepp`	`Frogdude: Hi, by default asterisk logs each call in the file /var/log/asterisk/cdr-csv/Master.csv`
`06:52.29`	`fdepp`	`and there are plugins such as cdr_odbc which can log call details in a database`
`06:56.09`	`*** join/#asterisk Frogdude (i=Frogdude@c-24-16-72-159.hsd1.wa.comcast.net)`
`07:01.37`	`Frogdude`	`fdepp: is there a way to write only some information to a database with the stuff that you mentioned? (as opposed to all general call data)`
`07:01.55`	`Frogdude`	`or can that be done with the cdr_odbc stuff that you mentioned`
`07:02.19`	`Frogdude`	`I just looked at a page about it and got the impression that it logs a bunch of stuff by default`
`07:02.51`	`Frogdude`	`I apologize in advance if I lose my connection. It's been a little flakey here lately.`
`07:03.29`	`*** join/#asterisk bjohnson (n=bjohnson@i216-58-9-75.cybersurf.com)`
`07:04.38`	`fdepp`	`Frogdude: yes each call is logged but you can disabled CDR generation in your dialplan with the NoCDR() application`
`07:06.06`	`Frogdude`	`then can I just log some data (only stuff specified by me)?`
`07:06.58`	`kshyvy`	`i had similiar problem lately`
`07:07.37`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`07:07.48`	`Frogdude`	`is there not a good way to only log certain things?`
`07:08.09`	`kshyvy`	`talked to our db staff, and they told me that it's better to create view containing only data you need, and leave other table with all data intact`
`07:08.38`	`Frogdude`	`ahhh`
`07:08.38`	`kshyvy`	`it's easier`
`07:08.44`	`Frogdude`	`ok`
`07:09.35`	`*** join/#asterisk e-ddie (n=oal@62.61.133.90.generic-hostname.arrownet.dk)`
`07:10.52`	`Frogdude`	`kshyvy: so what did you use to create your database stuff?`
`07:12.48`	`kshyvy`	`i have postgres installed on remote server - all data goes there, and i have view created which returns only connecttime, disconnecttime, dn, calledpartynumber, connectiontime, port`
`07:13.16`	`kshyvy`	`based on those data billing is generated`
`07:14.12`	`Frogdude`	`I'm just curious, what kind of project were you working on?`
`07:14.41`	`Frogdude`	`I mean what was it all for?`
`07:15.07`	`Frogdude`	`I'm pretty new to asterisk and it's pretty interesting hearing about all of the uses for it :)`
`07:16.09`	`x86`	`any freemasons around?`
`07:16.16`	`kshyvy`	`heh`
`07:16.23`	`kshyvy`	`i work for a university`
`07:16.39`	`Frogdude`	`ahh`
`07:17.19`	`kshyvy`	`we implemented callmanager for main location, and 3 other locations, and now - we connected asterisk to our callmanager allowing workers to connect from homes`
`07:17.56`	`kshyvy`	`it's mainly because one have to pay to use cisco ip communicator, and doesn't have to pay for x-lite ;)`
`07:18.06`	`Frogdude`	`nice`
`07:18.27`	`kshyvy`	`so all cdr's from callmanager, from callmanager express and from asterisk go to one database`
`07:21.49`	`kshyvy`	`now i'm testing Linksys SPA942 - quite nice phone`
`07:22.31`	`Frogdude`	`that's cool`
`07:22.43`	`*** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it)`
`07:23.41`	`sx-wks`	`kshyvy: is that the wifi thingie ?`
`07:24.02`	`*** join/#asterisk znoG_ (n=gs@162-148-235-201.fibertel.com.ar)`
`07:24.09`	`kshyvy`	`sx-wks: nah, desktop phone`
`07:24.22`	`kshyvy`	`i'll have this wifi next week probably`
`07:24.24`	`kshyvy`	`:)`
`07:24.27`	`sx-wks`	`kshyvy: hah :D`
`07:24.46`	`kshyvy`	`from what i've seen it's sweet ;)`
`07:24.51`	`sx-wks`	`lucky you... my personal cash flow doesn't allow for having any of those toys :(`
`07:25.32`	`kshyvy`	`so do mine - but i have bosses who want's something like that, so i'll have to test it for... let's say 2 weeks? 3 weeks? :)`
`07:25.47`	`sx-wks`	`kshyvy: heh.`
`07:26.01`	`sx-wks`	`the ITU website is slow as hell`
`07:26.25`	`sx-wks`	`anyhow, I made a nice filter with callerid`
`07:26.26`	`kshyvy`	`working for university is quite cool - good access to all those toys - money are poor - but hey - you can't have everything`
`07:26.29`	`kshyvy`	`;)`
`07:26.50`	`sx-wks`	`kshyvy: I work for university too, but not in the telecom department :(`
`07:27.19`	`*** join/#asterisk stephane_ (n=stephane@merlin.cabale.net)`
`07:27.27`	`kshyvy`	`i work in "department of it implementation and innovation"`
`07:27.31`	`stephane_`	`jour`
`07:27.34`	`sx-wks`	`and I'm not sure I'd like to be transfered there... they don't seem to want to hear about anything besides alcatel and big bucks crap`
`07:27.38`	`kshyvy`	`subsection of networking ;)`
`07:28.02`	`sx-wks`	`kshyvy: the telecom department is wary of anything IT that could eat their lunch.`
`07:28.38`	`sx-wks`	`(which, somehow, means they already know they are irrelevant and are fighting tooth & nails to stay there - can't blame them)`
`07:28.38`	`kshyvy`	`yeah i know how it looks ;) - i have few of them here too`
`07:29.28`	`kshyvy`	`when they heard that voip will be our playground they almost killed us`
`07:29.39`	`kshyvy`	`well, at least they tried ;)`
`07:29.47`	`sx-wks`	`lol`
`07:30.05`	`kshyvy`	`i told them to go play with wires and don't bother me`
`07:31.03`	`sx-wks`	`well. here, it's not that they don't want to hear the words VoIP, it's just that if it doesn't say 'alcatel' all over the equipment, and the price tag is not in the 1+M€ ballpark, it's not worth shit`
`07:31.11`	`*** join/#asterisk inspired (n=mikael@85.221.7.59)`
`07:31.29`	`kshyvy`	`and lately their boss came and asked if we could install him one of those soft phones.... i almost rotfl'ed`
`07:31.43`	`sx-wks`	`kshyvy: hahahah`
`07:31.54`	`sx-wks`	`kshyvy: sweet justice, I'd say :D`
`07:32.15`	`kshyvy`	`yep, exactly my thoughts ;)`
`07:32.17`	`sx-wks`	`talking about work, it's time for me to head there ;D`
`07:32.28`	`kshyvy`	`already there`
`07:32.30`	`kshyvy`	`;)`
`07:32.35`	`kshyvy`	`it's 2 hours now`
`07:32.46`	`*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`07:32.46`	`kshyvy`	`another 6 and i'm free`
`07:33.45`	`sx-wks`	`heh`
`07:34.06`	`sx-wks`	`I start at 10 am or so`
`07:34.35`	`kshyvy`	`8 here in poland`
`07:34.45`	`kshyvy`	`we work from 8 till 16`
`07:35.01`	`sx-wks`	`we work from 10 to 17`
`07:35.28`	`kshyvy`	`nice`
`07:36.17`	`*** join/#asterisk novafirst (n=kosta@wrt1.niclab.com)`
`07:36.49`	`novafirst`	`what are agents used for? are they like switch-operators?`
`07:37.55`	`j0`	`is it possible to buy an "unlimited outbound" trunk for multiple phone numbers?`
`07:37.58`	`mover`	`exist any backport from format_h264.c (trunk) to 1.2.x ?`
`07:37.58`	`*** join/#asterisk TeKaBloK (n=k-misol@80.227.118.6)`
`07:38.04`	`TeKaBloK`	`hello`
`07:38.15`	`TeKaBloK`	`i need some help :)`
`07:38.32`	`TeKaBloK`	`Does anyone know howto change the tone after a WaitExten ?`
`07:38.44`	`TeKaBloK`	`I hate the default one :/`
`07:38.54`	`TeKaBloK`	`Please`
`07:39.40`	`TeKaBloK`	`I tried to change it in the indications.conf but it cant find wich ne it is`
`07:39.45`	`TeKaBloK`	`+o`
`07:40.35`	`Rahail`	`seen Bk`
`07:40.53`	`x86`	`any freemasons around?`
`07:41.39`	`*** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za)`
`07:41.56`	`*** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au)`
`07:45.35`	`*** join/#asterisk ltd (n=z@202-161-16-50.dyn.iinet.net.au)`
`07:46.34`	`TeKaBloK`	`pllllzzzz help me`
`07:50.51`	`Rahail`	`TeKaBloK sorry man .... If i know the answer I would tell you`
`07:50.53`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`07:52.29`	`TeKaBloK`	`ok thank you`
`07:52.32`	`TeKaBloK`	`because this tone is just annoying`
`07:52.46`	`TeKaBloK`	`and sure that there's a way`
`07:53.09`	`Rahail`	`I am just a new comer on * so I am still learning`
`07:53.46`	`TeKaBloK`	`ok`
`07:53.48`	`TeKaBloK`	`:)`
`07:54.55`	`Rahail`	`I am trying to find info If I have two Asterisk box how do I make call each other free`
`07:54.57`	`Rahail`	`any one know`
`07:56.07`	`Skyelar`	`Rahail: search voip-info,org for "iax trunk" and/or "sip trunk"; will hopefully get you started`
`07:56.25`	`Rahail`	`thanx for the hint`
`07:57.37`	`*** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp)`
`07:57.53`	`TeKaBloK`	`IAX`
`07:58.03`	`TeKaBloK`	`it s better, it s very easy`
`07:58.20`	`Rahail`	`if you can explain little if you have time if not I understand`
`07:58.22`	`TeKaBloK`	`u just need to create to friends and u cn call each other`
`07:58.47`	`TeKaBloK`	`[asterisk] ------ IAX Protocol ------- [asterisk]`
`07:59.13`	`TeKaBloK`	`try to look for IAX on google`
`07:59.14`	`TeKaBloK`	`!google iax`
`07:59.51`	`benjk_`	`dont use friends for linking two asterisk boxes`
`07:59.59`	`TeKaBloK`	`why ?`
`08:00.02`	`benjk_`	`friends are for phones`
`08:00.09`	`TeKaBloK`	`?!`
`08:00.17`	`TeKaBloK`	`friends can make call and receive`
`08:00.19`	`benjk_`	`use one user and one peer each`
`08:00.30`	`TeKaBloK`	`it s the same`
`08:00.36`	`benjk_`	`its not`
`08:00.38`	`TeKaBloK`	`user and peer is friend`
`08:00.40`	`TeKaBloK`	`why ?`
`08:00.45`	`benjk_`	`friends have side effects`
`08:01.00`	`TeKaBloK`	`what do u mean by side effects ?`
`08:01.04`	`Rahail`	`ok if i understand coorect`
`08:01.12`	`benjk_`	`trust me on this`
`08:01.33`	`benjk_`	`everybody who's been doing asterisk for a number of years will tell you the same`
`08:01.50`	`TeKaBloK`	`ok, i didn't now this`
`08:01.56`	`TeKaBloK`	`is it security problems ?`
`08:01.57`	`Rahail`	`I have Extension on box 1 201 and did 313-445-xxxx on Box2 I have another Extension 303 and did 313-478-xxx`
`08:02.27`	`Rahail`	`how do i make them call free each other insted of using my us route`
`08:03.41`	`TeKaBloK`	`try to create an IAX between them`
`08:03.47`	`TeKaBloK`	`with a user / peer ;)`
`08:03.58`	`TeKaBloK`	`and add a prefix for it`
`08:05.15`	`TeKaBloK`	`If anyone know how to change the ringone on the WaitExten() application after somebody dials, please tell me`
`08:05.59`	`Skyelar`	`benjk_: one thing that's never been adequately explained to me is why use user/peer rather than friend - can you elaborate?`
`08:06.13`	`*** join/#asterisk dpryo (n=hn@batman.ws.interweb.no)`
`08:08.00`	`TeKaBloK`	`yeah :)`
`08:08.08`	`TeKaBloK`	`need to explain more`
`08:08.20`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`08:08.38`	`*** join/#asterisk meshuga (i=meshuga@c-71-231-141-145.hsd1.or.comcast.net)`
`08:08.57`	`benjk_`	`when you use friends, asterisk will populate some parameters implicitly and you have no control over it`
`08:09.29`	`benjk_`	`side effects basically`
`08:09.38`	`benjk_`	`what you want is to be able to set those independently`
`08:10.17`	`Rahail`	`lol`
`08:10.30`	`Rahail`	`I dont even know what is friend and what is peer`
`08:10.47`	`Rahail`	`dont lough please some how I got * installed`
`08:10.49`	`RoyK`	`friend is user+peer`
`08:11.03`	`*** join/#asterisk casterman (n=Administ@kasadsyed.net8.nerim.net)`
`08:12.07`	`Skyelar`	`benjk_: makes sense - is there any docs on just what the side effects are, or am I off into channels/chan_iax2.c to look? :)`
`08:12.29`	`RoyK`	`a SIP friend is exactly the same as a SIP user and a SIP peer with the same name and the same parameters`
`08:12.43`	`benjk_`	`you can see a table inside iax.conf.sample which explaines which settings are independent and which are joined`
`08:13.04`	`benjk_`	`RoyK, we are talking about IAX trunks`
`08:13.14`	`Rahail`	`ok when you create exetnsion is it friend or user`
`08:13.41`	`TeKaBloK`	`lol rahail`
`08:13.58`	`Rahail`	`:)`
`08:14.06`	`TeKaBloK`	`we talk about iax.conf / sip.conf`
`08:14.10`	`TeKaBloK`	`not extensions.conf`
`08:14.24`	`RoyK`	`benjk_: ah`
`08:14.34`	`casterman`	`i have trouble with sip and a NAT is there arecomandable howto somewhere ? can i just port forward to fix the trouble`
`08:14.45`	`*** join/#asterisk Anotsu (n=kvirc@a81-14-180-246.net-htp.de)`
`08:14.49`	`Rahail`	`as I said I am just start out but sim like lots fun at *`
`08:15.02`	`TeKaBloK`	`yeah`
`08:15.22`	`*** join/#asterisk LoneShadow (n=duh@59.92.131.151)`
`08:15.52`	`LoneShadow`	`anyone using a callback agi script without freepbx ?`
`08:19.12`	`TeKaBloK`	`nop sry`
`08:20.16`	`Rahail`	`as I said I am just start out but sim like lots fun at *`
`08:20.20`	`Rahail`	`ops sorry`
`08:21.28`	`j0`	`anyone here on shaw cable in canada?`
`08:21.55`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`08:23.36`	`RoyK`	`key2: ping`
`08:24.23`	`TeKaBloK`	`hey, did someone change his indications.conf ? For me, it sounds like nothing changed, it's still the US tones :/`
`08:26.21`	`TeKaBloK`	`even if i change the country had I can see that the sttings are taken, it doesnt change`
`08:28.19`	`j0`	`whats the traceroute program that continually updates all hops?`
`08:28.43`	`Rahail`	`pingpotters`
`08:28.45`	`Rahail`	`try that one`
`08:28.51`	`Rahail`	`:)`
`08:29.00`	`j0`	`spelt right?`
`08:29.05`	`j0`	`thats not the one i was thinking of.. but hey`
`08:29.18`	`Rahail`	`hold on let me get the link`
`08:29.18`	`Rahail`	`duh`
`08:29.26`	`Rahail`	`I am halfsleep its 4.30am`
`08:29.47`	`Rahail`	`http://www.pingplotter.com/`
`08:29.59`	`Rahail`	`i missed l`
`08:30.04`	`Rahail`	`and add s`
`08:30.17`	`j0`	`wow overkill.. thanks tho`
`08:30.29`	`Rahail`	`lol`
`08:30.48`	`Rahail`	`what do you mean by overkill`
`08:30.49`	`Rahail`	`?`
`08:30.58`	`j0`	`does far more than i need`
`08:31.11`	`Rahail`	`oh`
`08:32.27`	`*** join/#asterisk heison (n=heison@ip-89-168-23-157.cust.homechoice.net)`
`08:33.30`	`*** join/#asterisk darkskiez (n=mbryars@194.247.78.146)`
`08:33.51`	`*** join/#asterisk darkskiez (n=mbryars@194.247.78.146)`
`08:33.52`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`08:34.05`	`heison`	`hi there... does anyone here uses IaxComm or loudHush on OS X?`
`08:34.44`	`*** join/#asterisk firstknight (n=firstkni@202.155.150.244)`
`08:35.38`	`Rahail`	`IaxComm Softphone`
`08:35.41`	`Rahail`	`?`
`08:37.54`	`firstknight`	`Hi, I try to use Playback function in asterisk configuration ( exten => 5,3,Playback(/usr/share/asterisk/sounds/hello-world) ). I test it with yate softphone, but I hear a very unclear sound. In asterisk console, I get this message: Playing '/usr/share/asterisk/sounds/hello-world' (language 'en'). How do I make it clear?`
`08:39.37`	`heison`	`Rahail: yes, iaxcomm softphone`
`08:40.59`	`inspired`	`firstknight, is that a sound you recorded yourself? if yes, then adjust the volume with sox or something, or re-record it. if it is not a sound you recorded yourself, adjust the volume of your speakers. if that doesn't help, change softphone`
`08:41.00`	`Rahail`	`I did used it`
`08:41.05`	`heison`	`Rahail: it's been working fine until recently.... it stopped working all of a sudden, no packets leaving my machine when a call is place, although IAX registration seems to be working - I can see it on my asterisk box`
`08:41.11`	`heison`	`Rahail: on OS X?`
`08:42.15`	`firstknight`	`inspired, I get that from asterisk-sounds-main package..... ok, then I try with another softphone`
`08:42.17`	`Rahail`	`xp yeap`
`08:44.24`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`08:46.24`	`*** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35)`
`08:48.04`	`phearless`	`anybody got a UK CallerID experience ?`
`08:48.19`	`*** join/#asterisk vlt (n=dm@p54B33914.dip0.t-ipconnect.de)`
`08:54.08`	`*** join/#asterisk heison (n=heison@ip-89-168-23-157.cust.homechoice.net)`
`08:55.16`	`*** join/#asterisk s1gny\|wrk (n=s1gny@p54916A09.dip.t-dialin.net)`
`08:57.00`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`08:58.38`	`*** join/#asterisk daysmen3 (n=primus@host86-140-41-79.range86-140.btcentralplus.com)`
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`09:07.48`	`*** join/#asterisk potsboy (n=chrisg@196.211.16.202)`
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`09:16.52`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
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`09:22.15`	`*** join/#asterisk sxpert (n=sxpert@raph.imag.fr)`
`09:22.23`	`vlt`	`Hello *.`
`09:22.31`	`vlt`	`Question1: Can asterisk communicate with a SIP phone's busy lamp field?`
`09:26.18`	`vlt`	`Question2: I get an error message "app_queue.c:3099 queue_exec: Unable to join queue 'myqueue'" when trying to enqueue a call incoming on a special extension (CAPI in). Enqueuing SIP calls works fine. What could be the problem here?`
`09:28.47`	`Chris-NB`	`hi`
`09:28.59`	`Chris-NB`	`anyone owns a linksys spa942 ?`
`09:31.42`	`*** join/#asterisk geoffl (n=geoff@gjctech.plus.com)`
`09:36.41`	`*** join/#asterisk Anotsu (n=kvirc@a81-14-202-246.net-htp.de)`
`09:38.17`	`*** part/#asterisk geoffl (n=geoff@gjctech.plus.com)`
`09:38.59`	`mover`	`bounty: is anybody interested to backport the format_h264.c (trunk) to 1.2.x ?`
`09:44.28`	`*** join/#asterisk Bert- (n=bert@bas33-1-82-66-4-198.fbx.proxad.net)`
`09:44.36`	`Bert-`	`hi here :)`
`09:45.21`	`Bert-`	`I've a strange issue : system is working fine for weeks, and now suddently it is impossible for some (not all) remote users to register to asterisk`
`09:46.29`	`Bert-`	`exemple : http://pastebin.ca/152992`
`09:46.58`	`*** join/#asterisk ribbo (n=riaan@196.1.57.88)`
`09:51.48`	`L-info`	`Bert-, I cant tell from just the SIP 401. could you pb the complete SIP dialogue (starting with the REGISTER)?`
`09:54.20`	`Bert-`	`yep`
`09:59.03`	`Bert-`	`http://pastebin.ca/152998`
`09:59.22`	`Bert-`	`A REGISTER, A TRYING, then 401 :(`
`09:59.32`	`Bert-`	`really don't understand why ... :(`
`10:00.14`	`*** join/#asterisk [shodan] (n=shodan@ip157.99-113-216.pppoe4.joliette.intermonde.net)`
`10:02.37`	`L-info`	`Bert-, thats odd, and your UA doesnt try sending anything else?`
`10:02.42`	`L-info`	`after the 401`
`10:03.17`	`Bert-`	`there is nothing else provided by SIP debug ip $myip`
`10:03.30`	`Bert-`	`I added the client config in Asterisk`
`10:03.33`	`Bert-`	`http://pastebin.ca/152999`
`10:04.05`	`L-info`	`the presence of the www-Authenticate field is there to give the UA the chance to respond with the correct credentials`
`10:04.42`	`L-info`	`as far as this dialogue goes, it is correct. Your UA needs to send a REGISTER in response to this 401, with the authentication credentials`
`10:05.15`	`L-info`	`check your SIP UA has the correct user/pass as defined in its peer/user context in sip.conf`
`10:05.36`	`*** join/#asterisk LarsSundqvist (n=Lasse@81-232-79-126-no46.business.telia.com)`
`10:09.46`	`LarsSundqvist`	`Hello all, Anyone know of a way to change the timer in the Dial application set with L(X:Y:Z) or S(X) in the macro called by M()? I am calling multiple channels in Dial and want to remove the limit if a specific channel gets answered.`
`10:20.56`	`*** part/#asterisk casterman (n=Administ@kasadsyed.net8.nerim.net)`
`10:23.26`	`*** join/#asterisk hads\|home (n=hads@mail.nice.net.nz)`
`10:24.39`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`10:31.57`	`*** join/#asterisk Ahrimanes (n=michael@81.7.159.2)`
`10:33.27`	`kmilitzer`	`Does anyone of you know what can trigger a message like that: pbx.c:1742 pbx_extension_helper: Cannot find extension context 'default', even if I nowhere have a context default in the dialplan?`
`10:33.52`	`*** join/#asterisk razu (n=rasmus@tln-kontor.norby.ee)`
`10:34.17`	`benjk`	`kmilitzer, that's your problem`
`10:34.35`	`benjk`	`some modules and apps make use of the default context`
`10:34.40`	`benjk`	`they assume it is there`
`10:35.00`	`benjk`	`they use it under certain circumstances, mostly as a fallback, if anything else fails (doesn't match)`
`10:35.54`	`benjk`	`so if you don't have a default context, and you run into such a fallback scenario, then pbx.c won't be able to find it and that's the error you are getting`
`10:36.14`	`kmilitzer`	`benjk: So to sinlence it, I need an (emtpy) default context?`
`10:36.23`	`kmilitzer`	`s/sinlence/silence/`
`10:36.33`	`benjk`	`empty may not be enough`
`10:36.59`	`benjk`	`because it is llikely that some app or some channel module has tried to transfer control to s@default`
`10:37.25`	`benjk`	`you should have a [default] context in your dialplan with an s extension in it`
`10:37.43`	`benjk`	`for security reasons you may want to just print something to the console and hangup`
`10:37.47`	`benjk`	`[default]`
`10:38.02`	`kmilitzer`	`But how do I find out, what triggered this? Defining an default context only cures the symptomes, but not the cause`
`10:38.02`	`benjk`	`exten => s,1,NoOp(s at default was called)`
`10:38.09`	`benjk`	`exten => s,2,Hangup`
`10:38.30`	`benjk`	`that's what you will use the NoOp() in the s extension in default for`
`10:38.48`	`benjk`	`you can add more lines with NoOp()s to print out channel variables and stuff`
`10:39.02`	`benjk`	`this will allow you to find out why you got there`
`10:39.47`	`kmilitzer`	`Hmm ... I wish, I could say, that my dialplan is waterproof, so that this woulnd't be able to happen, but somehow it isn't`
`10:40.00`	`benjk`	`its not the dialplan`
`10:40.14`	`benjk`	`its triggered in some app or some channel`
`10:40.21`	`kmilitzer`	`But I am not using any application besides dial or voicemail ...`
`10:40.33`	`kmilitzer`	`And I only use SIP-Channels ...`
`10:40.35`	`benjk`	`as a result of not having the information how to route an incoming call`
`10:40.44`	`kmilitzer`	`So my enviroment is pretty controlled ...`
`10:40.56`	`kmilitzer`	`And I _never_ have calls that I don't know what to do with ...`
`10:41.03`	`benjk`	`in that case put NoOp()s into your s@default context to print out the SIP related channel variables`
`10:41.23`	`kmilitzer`	`benjk: Which would be what exactley?`
`10:41.27`	`benjk`	`also SIP debug will give you more info on what is happening`
`10:41.43`	`benjk`	`goto voip-info.org and seartch for channel variables`
`10:42.45`	`vlt`	`Hello *`
`10:43.01`	`vlt`	`Does anyone know if asterisk can communicate with a SIP phone's busy lamp field?`
`10:43.15`	`*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)`
`10:50.48`	`*** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net)`
`10:53.21`	`Ahrimanes`	`vlt: it can with a few phones, which phone do you have ?`
`10:58.21`	`vlt`	`Ahrimanes: I have a Grandstream GXP-2000 and consider extending it with its new EXT module.`
`10:59.01`	`vlt`	`Ahrimanes: But I need to buy some more SIP phones. Can you recommend a goodone for Small Office?`
`10:59.41`	`Ahrimanes`	`vlt: i have really good experiences with snom phones`
`11:00.08`	`Ahrimanes`	`vlt: the gxp2000 has asterisk blf support in newer firmwares, but dont know if the ext module is supported`
`11:06.59`	`kmilitzer`	`Just adding an empty default context seemed to have solved my problem ...`
`11:08.51`	`benjk`	`good`
`11:12.31`	`kumamoto`	`Ahrimanes:I would recommend the sipura SPA942 for small office`
`11:20.26`	`Dr-Linux\|work`	`guys`
`11:20.31`	`Ahrimanes`	`kumamoto: ok, havent tried sipura phones, only ata's.. but snom/asterisk BLF integration is really nice :)`
`11:20.51`	`Dr-Linux\|work`	`exten => _9XXXX,1,VoiceMailMain(${EXTEN:1}@default) << i wanna use here "s" option to skip the password, where can i use it?`
`11:21.02`	`vlt`	`kumamoto: Thank you. I'll have a look.`
`11:21.24`	`*** part/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`11:22.24`	`Ahrimanes`	`Dr-Linux\|work: exten => _9XXXX,1,VoiceMailMain(s${EXTEN:1}@default)`
`11:22.28`	`vlt`	`Ahrimanes: I found "Asterisk BLF" in the GXP's web config (multip. keys) and activated it. But it doensn't show me anything ...`
`11:22.40`	`Ahrimanes`	`vlt: you need to add hints to your dialplan`
`11:22.50`	`Dr-Linux\|work`	`Ahrimanes, thanks, lemme try`
`11:23.31`	`Ahrimanes`	`vlt: http://www.jackenhack.com/blog/archives/2005/11/22/setting-up-subscribenotify-blf-in-asteriskhome-for-grandstream-gxp-2000-phones/`
`11:23.54`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`11:24.10`	`vlt`	`Ahrimanes: THANK YOU!!! Never tried hints ... I'll read it ... and then ask again ;-)`
`11:25.02`	`*** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com)`
`11:25.41`	`Stephnie`	`is there a way to check the QUEUE STATUS in Asterisk RealTime ?????? other than Asterisk Manager API`
`11:26.00`	`Ahrimanes`	`vlt: you're welcome :) i have a gxp2000 here if all goes wrong.. :)`
`11:26.15`	`RoyK`	`Stephnie: no`
`11:26.40`	`Stephnie`	`I am interested to develop a Realtime queue monitoring front end ..`
`11:27.03`	`Stephnie`	`so what is the best way to do so ??? Manager or AGI`
`11:27.12`	`*** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk)`
`11:27.42`	`RoyK`	`http://www.youtube.com/watch?v=1XuEGHHoZGM`
`11:27.57`	`Stephnie`	`Royk: is that for me?`
`11:28.45`	`*** join/#asterisk sgeigerbot (n=user@h460d3136.area2.spcsdns.net)`
`11:29.04`	`Stephnie`	`lol..thats nice`
`11:31.40`	`Stephnie`	`RoyK: any suggestions to develop a queue monitoring interface?`
`11:31.59`	`Stephnie`	`whats the professional and best way to do that`
`11:32.20`	`RoyK`	`use the manaager, perhaps?`
`11:32.43`	`RoyK`	`or setup queue logging to a db and then use that for 'realtime' monitoring`
`11:32.49`	`RoyK`	`it's event based anyway, so it should work`
`11:32.57`	`sxpert`	`I am looking into using a postgresql db for blacklist lookup`
`11:32.58`	`RoyK`	`anyway, i beleive there are one or two out there already`
`11:33.39`	`Stephnie`	`if any event is missed due to Internet problem....then...`
`11:34.26`	`Stephnie`	`I think I should check wiki :)`
`11:37.45`	`sxpert`	`haha... anonymous calls to home get tt-monkeys ;D`
`11:42.50`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`11:43.48`	`Ahrimanes`	`has anyone here succesfully setup icecom switchboard?`
`11:44.30`	`sxpert`	`what's icecom switchboard ?`
`11:46.11`	`Ahrimanes`	`piece of software so that eg. a receptionist can monitor, queue, transfer etc calls from a pc`
`11:46.42`	`*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)`
`11:47.50`	`benjk`	`RoyK, this one is much funnier ... http://www.youtube.com/watch?v=VJwAE5Mtrxc&mode=related&search=`
`11:49.51`	`sxpert`	`benjk: wtf is the dude saying ?`
`11:50.13`	`benjk`	`I have no idea`
`11:50.18`	`sxpert`	`hah. the referee is getting pissed ;D`
`11:50.26`	`sxpert`	`starts distributing colored cards`
`11:50.30`	`benjk`	`but I can imagine what they players are saying to the referee`
`11:50.53`	`benjk`	`"come on dude, you went to school with my cousin"`
`11:51.16`	`benjk`	`7 times`
`11:51.56`	`benjk`	`hilarious`
`11:53.10`	`*** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com)`
`11:53.13`	`sxpert`	`yeah`
`11:53.45`	`sxpert`	`damn stupid PBX here not setting CallerID correctly...`
`11:54.08`	`sxpert`	`the damn thing sends "anonymous" for all phones here`
`11:54.10`	`benjk`	`some arabic speaking guy commented on the page ...`
`11:54.14`	`benjk`	`"muahahaha the one who's talking arabic said that's a wrong way to leave his name on the football history, and its a good training for the one who's shooting the penalty."`
`11:55.10`	`Dr-Linux\|work`	`what's this warning mean? >> Aug 29 04:34:40 WARNING[8760]: chan_sip.c:9525 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Saud Shah 4092" <sip:16505935400@70.89.66.128>;tag=as487fa6c0'`
`11:55.37`	`sxpert`	`Dr-Linux\|work: the peer refuses the invite`
`11:55.46`	`sxpert`	`you need to use canreinvite=no`
`11:56.22`	`Dr-Linux\|work`	`sxpert, for what peer? canreinvite=no ?`
`11:56.43`	`Dr-Linux\|work`	`for sip trunk peer? or for my extenstion?`
`11:56.55`	`sxpert`	`for the sip peer, whoeever 70.89.66.128 is`
`11:57.06`	`benjk`	`sxpert, another commenter translated this ... "For God's Sake , its a goal!! , can u just not waste more time so maybe we can score back "`
`11:57.17`	`sxpert`	`benjk: yeah. read the page ;D`
`11:57.28`	`benjk`	`that;s hilarious :D`
`12:01.23`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`12:02.13`	`Dr-Linux\|work`	`sxpert, well, i'm using canreinvite=no for both peers, but same warning`
`12:02.14`	`Dr-Linux\|work`	`:S`
`12:02.25`	`Dr-Linux\|work`	`even i am sending dummy CID`
`12:02.45`	`Dr-Linux\|work`	`Aug 29 04:44:25 WARNING[8760]: chan_sip.c:9525 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"Saud Shah 4092" <sip:16505935400@70.89.66.123>;tag=as1390d338'`
`12:02.45`	`Dr-Linux\|work`	`<PROTECTED>`
`12:03.40`	`Dr-Linux\|work`	`i see`
`12:04.07`	`sxpert`	`you can't dial out then I guess ?`
`12:04.14`	`Dr-Linux\|work`	`sxpert, i can dial via serverA , but i can't dial IN serverA`
`12:04.21`	`sxpert`	`hah.`
`12:04.50`	`sxpert`	`your contract offers a proper DID ?`
`12:05.07`	`sxpert`	`or are you trying to call <yournumber>@sip-provider ?`
`12:05.52`	`Dr-Linux\|work`	`sxpert, i'm calling thorought my extension 4092 which is registered with serverB`
`12:06.00`	`Dr-Linux\|work`	`look here:`
`12:06.16`	`Dr-Linux\|work`	`i am calling through serverA, no problem:`
`12:06.16`	`Dr-Linux\|work`	`Executing Dial("SIP/4092-6ab3", "SIP/server1/5105805032\|30\|r") in new stack`
`12:06.17`	`Dr-Linux\|work`	`<PROTECTED>`
`12:06.17`	`Dr-Linux\|work`	`<PROTECTED>`
`12:06.20`	`sxpert`	`then server A refuses to be called from server B`
`12:06.34`	`Dr-Linux\|work`	`nope`
`12:06.47`	`sxpert`	`no idea then`
`12:06.47`	`Dr-Linux\|work`	`i can diat outside number`
`12:07.06`	`Dr-Linux\|work`	`but i can't dial serverA DID`
`12:07.19`	`sxpert`	`can you pick up an analog phone and call the DID on server A ?`
`12:07.37`	`Dr-Linux\|work`	`sxpert, that works`
`12:07.49`	`sxpert`	`ok. then you have the same kind of setup as I have.`
`12:08.13`	`Dr-Linux\|work`	`sxpert, actually i know, what is the problem, but i'm not sure how can i fix it`
`12:08.27`	`E-bola`	`How can i make it in my dialplan so if a number is busy it dials another number?`
`12:08.31`	`sxpert`	`there may be no way to fix it if this is the provider's policy`
`12:08.55`	`sxpert`	`E-bola: just add a dial command to the subsequent number`
`12:08.56`	`Dr-Linux\|work`	`sxpert, there is nothing related provider`
`12:09.20`	`E-bola`	`sxpert: then it will ring both?`
`12:09.22`	`E-bola`	`i dont want that`
`12:09.31`	`sxpert`	`one after the other`
`12:09.34`	`E-bola`	`should only rign the second number if the first is busy`
`12:09.39`	`Dr-Linux\|work`	`sxpert, maybe it's due to my extension 4092 exists in both serverA and ServerB`
`12:09.40`	`E-bola`	`hmm can u gimmi an example line?`
`12:10.11`	`sxpert`	`E-bola: exten => blah,1,dial(number-1)`
`12:10.22`	`sxpert`	`exten => blah,n,dial(number-2)`
`12:10.27`	`E-bola`	`ohh hehe`
`12:10.30`	`sxpert`	`exten => blah,n,voicemail()`
`12:10.32`	`E-bola`	`it just goes 1 down if its busy?`
`12:10.35`	`sxpert`	`yeha`
`12:10.40`	`E-bola`	`im so stoned today`
`12:10.44`	`E-bola`	`should hve figured that out hehe`
`12:10.44`	`sxpert`	`lol`
`12:10.48`	`E-bola`	`i already use the concept for voicemail :)`
`12:10.50`	`E-bola`	`sory :)`
`12:10.54`	`sxpert`	`nm`
`12:11.23`	`sxpert`	`Dr-Linux\|work: don't think so`
`12:11.24`	`Dr-Linux\|work`	`sxpert, anyidea? ?`
`12:11.53`	`sxpert`	`I have this sip setup here, where I can't actually call the sip number via sip, but only via the pstn`
`12:12.00`	`sxpert`	`(provider policy)`
`12:12.20`	`sxpert`	`your problem seem pretty similar to what I have`
`12:13.08`	`Dr-Linux\|work`	`sxpert, aww dude, i have 5 servers and all are connected with each other with sip trunk and always i use, but this time it's occuring problem`
`12:13.19`	`sxpert`	`ah. ok`
`12:13.39`	`sxpert`	`then if they are all your servers, I am totally clueless`
`12:14.38`	`LarsSundqvist`	`Hello all, Anyone know of a way to change the timer in the Dial application set with L(X:Y:Z) or S(X) in the macro called by M()? I am calling multiple channels in Dial and want to remove the limit if a specific channel gets answered.`
`12:15.52`	`sxpert`	`Dr-Linux\|work: instead of defining your trunk in sip.conf, you define it in iax.conf ;D`
`12:16.58`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`12:19.39`	`Dr-Linux\|work`	`i don't knwo the way`
`12:19.50`	`*** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it)`
`12:20.01`	`sxpert`	`Dr-Linux\|work: the best way to learn is to read the manual ;D`
`12:20.05`	`Dr-Linux\|work`	`sxpert, however now i'm trying to fix the problem, i can make a call to serverA`
`12:20.11`	`RoyK`	`erm. anyone seen this? setting CALLERID(number) works fine, but that change isn't reflected in the cdr logs`
`12:20.46`	`Dr-Linux\|work`	`english is hard to read`
`12:20.54`	`*** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com)`
`12:20.54`	`sxpert`	`Dr-Linux\|work: where are you from /`
`12:21.10`	`Stephnie`	`how to remove SOX from my linux box ???`
`12:21.13`	`Stephnie`	`uninstall*`
`12:21.24`	`sxpert`	`Stephnie: depends on your distribution I guess`
`12:21.28`	`Stephnie`	`RH9`
`12:21.35`	`*** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se)`
`12:21.57`	`sxpert`	`Stephnie: some option to rpm with sox as a parameter`
`12:22.06`	`Stephnie`	`the latest version of SOX doesnt mixup IN and OUT files when I run asterisk as deamon...`
`12:23.19`	`*** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com)`
`12:26.44`	`[TK]D-Fender`	`Stephnie: "man rpm"`
`12:28.35`	`*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)`
`12:28.36`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`12:29.07`	`*** join/#asterisk budairc (n=chatzill@urano.mhnet.com.br)`
`12:29.19`	`*** join/#asterisk kratzers (n=kratzers@martha.pa.net)`
`12:29.34`	`puzzled`	`hi`
`12:30.56`	`*** join/#asterisk stoffell_h (n=stoffell@pot.catsanddogs.com)`
`12:33.29`	`Dr-Linux\|work`	`sxpert, i'm from pakistan`
`12:33.40`	`Dr-Linux\|work`	`sxpert, hey my problem is fixed`
`12:33.53`	`Stephnie`	`what about if it was not RPM ....it was source doce`
`12:33.58`	`Stephnie`	`code*`
`12:34.19`	`Dr-Linux\|work`	`sxpert, shame on me, i was dialing pakistan server instead of US datacenter server :P`
`12:34.20`	`sxpert`	`Dr-Linux\|work: can you explain ?`
`12:34.26`	`Dr-Linux\|work`	`sxpert, sure`
`12:34.47`	`*** join/#asterisk Anotsu (n=kvirc@a81-14-202-246.net-htp.de)`
`12:35.10`	`Dr-Linux\|work`	`sxpert, actually i had to dial server1 with 91...prefix which goes to Datacenter IVR server`
`12:35.35`	`Dr-Linux\|work`	`but i was dialing serverA 92.. prefix which goes to Pakistan termination :D`
`12:36.00`	`sxpert`	`Dr-Linux\|work: hmmm sure that's gonna fail somehow ;D`
`12:36.12`	`Dr-Linux\|work`	`sxpert, actually i'm going to setup a seperate server for only voicemail`
`12:37.10`	`*** join/#asterisk Mugatu (n=mugatu@unaffiliated/Mugatu)`
`12:37.12`	`mover`	`bounty: is anybody interested to backport the format_h264.c (trunk) to 1.2.x msg me`
`12:37.17`	`kratzers`	`murf: I left work yesterday before I could respond. Everything's working fine now.`
`12:37.42`	`sxpert`	`Dr-Linux\|work: you can do that too ;D`
`12:38.13`	`sxpert`	`Dr-Linux\|work: btw, you should contact wasim on the #asterisk-drinkers channel, he's also from .pk`
`12:38.18`	`kratzers`	`it had been that way for a few days`
`12:38.40`	`Mugatu`	`may be a touch off-topic, but can anyone happen to point me to some starting points for using NIST JAIN-SIP to place a PSTN call through an asterisk server? This info seems to be surprisingly hard to find :)`
`12:38.48`	`Dr-Linux\|work`	`sxpert, contact him/her for what?`
`12:39.11`	`sxpert`	`Dr-Linux\|work: dunno, have a chat, drink something at a local bar, whatever ;D`
`12:39.51`	`Dr-Linux\|work`	`sxpert, well, i'm fine here :)`
`12:40.34`	`Dr-Linux\|work`	`sxpert, don't expect from a Pakistani .....`
`12:40.59`	`sxpert`	`Dr-Linux\|work: it would also allow you to talk whatever language you talk there instead of english, which would probably make the understanding easier`
`12:43.38`	`*** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr)`
`12:43.40`	`jhiver`	`hi all`
`12:43.53`	`jhiver`	`i have a question about codecs and transcoding`
`12:44.05`	`sxpert`	`ask, don't ask to ask`
`12:44.24`	`jhiver`	`currently i am connecting two sip legs, both can send or accept g711 g729 and g723`
`12:44.35`	`jhiver`	`the caller sends g729`
`12:44.54`	`jhiver`	`and asterisk /decides/ to transcode despite the fact that the callee could very well take g729`
`12:45.00`	`Ahrimanes`	`hm`
`12:45.00`	`Ahrimanes`	`anyone else experiencing lag? (51+)`
`12:45.18`	`jhiver`	`any ideas how to fix this other than the obvious disallow=all allow=g729?`
`12:47.13`	`Ahrimanes`	`jhiver: hm there's the issue with asterisk and g729 licensing ?`
`12:47.20`	`*** join/#asterisk mitcheloc_ (n=mitchelo@69-167-145-62.lmdaca.adelphia.net)`
`12:48.20`	`Ahrimanes`	`jhiver: setting canreinvite=yes if there's no nat, and let the endpoints decide for themselves ?`
`12:48.26`	`sxpert`	`Ahrimanes: when does this stupid patent dies ?`
`12:48.59`	`Ahrimanes`	`sxpert: i'm not sure it will`
`12:49.18`	`sxpert`	`heh`
`12:49.39`	`Ahrimanes`	`sxpert: and really, people smart enough to create a codec with that little bandwidth and and such quality would deserve some cash.. maybe not as much as they're getting now.. but some`
`12:51.42`	`sxpert`	`Ahrimanes: I don't think standards institute should accept patented things, after all, this is contrary to the concept of a standard...`
`12:52.16`	`Ahrimanes`	`sxpert: hm ok.. well these things dont interest me a lot, i'm more into making things work.. :)`
`12:53.12`	`sxpert`	`Ahrimanes: heh`
`12:55.27`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`12:55.27`	`*** mode/#asterisk [+o anthm] by ChanServ`
`12:57.57`	`murf`	`katzers: excellent. Thanks!`
`12:58.20`	`murf`	`oops, kratzers: excellent, Thanks!`
`13:00.55`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`13:01.05`	`*** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.f5.ngi.it)`
`13:01.19`	`*** join/#asterisk pif (n=ldm@zenon.apartia.fr)`
`13:09.02`	`*** join/#asterisk evilmnky (i=evilmnky@69-163-156-13.atlsfl.adelphia.net)`
`13:09.44`	`*** join/#asterisk tparcina (n=tparcina@lns01-0698.dsl.iskon.hr)`
`13:11.03`	`*** join/#asterisk jhiver_ (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr)`
`13:11.04`	`tparcina`	`i have tried to download SVN but this is what i get - http://pastebin.ca/153159 How to install Asterisk SVN?`
`13:11.31`	`jhiver_`	`sorry, got disconnected for some reason`
`13:11.43`	`jhiver`	`so... i was wondering if maybe there is a variable in the dialplan i could use to see what codecs are being accepted by the sender, then route it to the appropriate peer? i could have one peer per codec, even if the ip address is the same...`
`13:11.55`	`tparcina`	`I need SVN to try asterisk with h264 codec and to test Grandstream GXV-3000 Videophone`
`13:12.31`	`vlt`	`Ahrimanes: [blf, * and GXP] I read the blog page you pointed me to. I created a new context [blf] and placed the hints there. Then I added "subscribecontext=blf" to sip.conf. But when the GXP registered there was an error message like "cannot find 555 in context outgoing ...". So I added the hint extens directly to outgoing and now it works. Is this the right way? Where is the sip.conf mod for?`
`13:12.49`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`13:13.21`	`tparcina`	`so, how can i install asterisk svn?`
`13:13.26`	`brimstone`	`tparcina, use your package manager to install "subversion"`
`13:13.27`	`vlt`	`Ahrimanes: *directly to context [outgoing] ...`
`13:13.33`	`brimstone`	`then try again`
`13:14.03`	`brimstone`	`and i'd use branches/1.2 not trunk`
`13:14.17`	`Ahrimanes`	`vlt: yeah, i basically do the same.. i have a [blf] context and then include it in the context the phones use`
`13:15.02`	`vlt`	`Ahrimanes: And wtf does that sip.conf line do?`
`13:15.34`	`Ahrimanes`	`vlt: uhm, notifyringing=yes ?`
`13:15.45`	`*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)`
`13:16.15`	`*** join/#asterisk Luke-Jr (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d)`
`13:16.23`	`Luke-Jr`	`Is it possible to debug authentication?`
`13:16.25`	`tparcina`	`brimstone: what package manager? you mean yum (I use Fedora Core 4)? I need CSV version, because 1.2.11 doesn't have h264 codec...`
`13:16.47`	`brimstone`	`run yum install subversion`
`13:16.48`	`*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)`
`13:17.01`	`brimstone`	`then try the svn checkout line again`
`13:17.18`	`vlt`	`Ahrimanes: ... not in my sip.conf ...`
`13:17.25`	`tparcina`	`brimstone: than you, i'll try that`
`13:19.01`	`Ahrimanes`	`vlt: what line then ?`
`13:19.28`	`vlt`	`Ahrimanes: subscribecontext=blf`
`13:19.38`	`vlt`	`Ahrimanes: I read that in the blog page ...`
`13:20.05`	`sxpert`	`gah, stupid madplay doesn't do http`
`13:20.50`	`Ahrimanes`	`vlt: ah, think it's supposed to make it work with blf stuff in [blf], but if it works without it.. :)`
`13:21.18`	`vlt`	`Ahrimanes: Aah, ok :)`
`13:21.41`	`vlt`	`Question: Is there a way asterisk can dial an extension when an incoming CAP line rings without answering (means without charging the caller for listening to ringtone)?`
`13:22.18`	`*** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com)`
`13:22.25`	`vlt`	`Now I have "exten => 555,1,Dial(SIP/10,20)"`
`13:22.43`	`Dr-Linux\|work`	`how can i avoid this problem, a call hangs`
`13:23.11`	`Dr-Linux\|work`	`when i do "show channels" i can see`
`13:23.12`	`Dr-Linux\|work`	`Channel Location State Application(Data)`
`13:23.13`	`Dr-Linux\|work`	`Zap/pseudo-130059768 s@ivr:1 Rsrvd (None)`
`13:24.01`	`Dr-Linux\|work`	`i can't even soft hangup "Zap/pseudo-130059768"`
`13:24.10`	`Dr-Linux\|work`	`any clue?`
`13:25.42`	`Luke-Jr`	`How do I seperate incoming/outgoing contexts in Asterisk 1.2 now that 'peer' does both? :\|`
`13:26.28`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`13:26.48`	`Dr-Linux\|work`	`Luke-Jr, define both context in dialplan, and do n't include which you don't want`
`13:27.41`	`Luke-Jr`	`Dr-Linux\|work: I want both...`
`13:27.52`	`Luke-Jr`	`oh`
`13:28.03`	`Luke-Jr`	`well, the problem is that 'secret' should be used for outgoing, but not incoming`
`13:28.36`	`*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)`
`13:31.04`	`Dr-Linux\|work`	`Luke-Jr, what secret?`
`13:31.15`	`Luke-Jr`	`in sip.conf`
`13:32.49`	*** join/#asterisk `Kevin (n=Kevin@64.243.236.20)
`13:34.26`	`Dr-Linux\|work`	`Luke-Jr, secret is for registering the sip users, not for context`
`13:34.42`	`ssokol`	`Corydon76-home: Question for you regarding func_odbc. You around?`
`13:35.30`	`Luke-Jr`	`Dr-Linux\|work: no? then how do I authenticate my outgoing calls?`
`13:35.45`	`Luke-Jr`	`as long as the outgoing context is defined, Asterisk insists on my incoming calls using it :(`
`13:36.23`	`*** join/#asterisk tparcina_ (n=tparcina@lns01-1513.dsl.iskon.hr)`
`13:36.53`	`tparcina_`	`i get this error when i try to install svn asterisk - http://pastebin.ca/153173`
`13:37.04`	`tparcina_`	`why this happends?`
`13:39.06`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
`13:40.34`	`*** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net)`
`13:40.52`	`jtexter3`	`Is there a way to play a file to conference?`
`13:41.01`	`brimstone`	`tparcina, run "./configure && make && make install"`
`13:41.33`	`brimstone`	`just like it says, "./configure" must be run first`
`13:42.10`	`*** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com)`
`13:42.22`	`tparcina_`	`brimstone: thank you, now it works. the thing is that they don't say anything about that on asterisk web pages...`
`13:42.43`	`brimstone`	`most likely because trunk is the unstable development version`
`13:46.40`	`Luke-Jr`	`ok... so order of contexts in sip.conf is important`
`13:46.53`	`RoyK`	`~seen file`
`13:47.07`	`jbot`	`file is currently on #asterisk #openezx #elinux. Has said a total of 14 messages. Is idling for 9h 46m 39s, last said: 'moo?'.`
`13:47.07`	`*** join/#asterisk caloi (n=caloi@nat-66-218-1-140.usadatanet.com)`
`13:47.57`	`caio1982`	`coppice: i got a not-so-useful core file and the log just before asterisk segfaults here, using unicall0.0.3pre8 (pre9 does not work for me), MFC/R2 and the experimental t.38 code`
`13:48.12`	`caio1982`	`coppice: would you like to take a look at it? seems like a unicall problem (i'm guessing)`
`13:48.49`	`Dr-Linux\|work`	`Luke-Jr, don't inclucde outgoing context in incoming context`
`13:49.17`	`*** join/#asterisk mitcheloc_ (n=mitchelo@69-167-145-62.lmdaca.adelphia.net)`
`13:49.49`	`file`	`moo?`
`13:50.01`	`blitzrage`	`moo urns good buddy`
`13:50.08`	`coppice`	`caio1982: pre9 should work. its pre8 that has the bugs :-)`
`13:50.17`	`file`	`blitzrage: !!!`
`13:50.26`	`blitzrage`	`file: how goes?`
`13:50.34`	`coppice`	`caio1982: I'll take a look at the core file if you post it somewhere`
`13:50.43`	`file`	`not too bad, flipping through movie channels... you?`
`13:50.53`	`caio1982`	`coppice: did you read my email about pre9? i could not get it working because those internal functions missing in unicall or something`
`13:51.00`	`mitcheloc_`	`file, anything on?`
`13:51.07`	`caio1982`	`coppice: sure :) http://pastebin.ca/153175 and http://caio.ueberalles.net/debug/`
`13:51.24`	`file`	`some movie about two teachers competing for a teacher's award (that's the movie)... and the rest isn't that good`
`13:51.32`	`mitcheloc_`	`lame!`
`13:51.38`	`file`	`so I was suppose the answer is no`
`13:51.45`	`mitcheloc_`	`, do you watch prison break?`
`13:52.22`	`file`	`I do not`
`13:52.32`	`file`	`but I do watch House, so I am happy that a new episode will be on next week I believe`
`13:52.42`	`dasenjo`	`Hi, was an equivalent function to ast_copy_string in 1.0.10?`
`13:52.47`	`coppice`	`caio1982: You don't have permission to access /debug/core.6073 on this server.`
`13:52.50`	`mitcheloc_`	`awesome! I'm looking forward to finding out why he was shot, and so on`
`13:52.57`	`mitcheloc_`	`But check out PB, it's kick ass`
`13:53.37`	`caio1982`	`coppice: ops`
`13:54.26`	`caio1982`	`coppice: sorry, please retry`
`13:54.44`	`Luke-Jr`	`Dr-Linux\|work: I don't`
`13:55.20`	`*** join/#asterisk stdio (n=stdio@c-69-248-217-8.hsd1.pa.comcast.net)`
`13:55.38`	`*** join/#asterisk spr1te (i=spr1te@194.187.130.227)`
`13:55.57`	`coppice`	`caio1982: it looks OK now`
`13:56.09`	`mover`	`file: you read my msg`
`13:56.10`	`mover`	`?`
`13:56.45`	`file`	`I don't accept messages from random people, if I did I would be flooded`
`13:56.59`	`*** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no)`
`13:57.07`	`caio1982`	`coppice: i believe the core wont help much without all debug symbols, i believe i really should try pre9 again, but those errors in sent to you seems very weird`
`13:57.28`	`sxpert`	`(mpg123)`
`13:57.29`	`Dr-Linux\|work`	`file, :)`
`13:57.41`	`sxpert`	`dunno why it says it doesn't work or something`
`13:58.14`	`file`	`brimstone: !!!`
`13:58.44`	`*** join/#asterisk bpiper (n=bpiper@70.159.49.40)`
`13:59.47`	`sxpert`	`lol`
`14:01.15`	`Luke-Jr`	`brimstone: is flooding file fun?`
`14:01.28`	`brimstone`	`sometimes`
`14:01.39`	`coppice`	`caio1982: what was wrong when you used pre9?`
`14:01.55`	`brimstone`	`i like to fill him up with descripters that don't go anywhere :D`
`14:02.17`	`caio1982`	`coppice: i mailed you with the log but it looks like some internal functions or something missing in the chan_unicall module (it cant get loaded because it)`
`14:02.28`	`caio1982`	`coppice: i'll pastebin it`
`14:02.41`	`mover`	`file: so can you accpt. it from me?`
`14:02.41`	`dasenjo`	`I want to "backport" an 1.2 module to 1.0, and I found the function ast_copy_string that does not exist in 1.0, is there an equivalent function or I have to use the str* C functions instead?`
`14:02.43`	`mover`	`:-)`
`14:02.51`	`E-bola`	`Anybody got an example on how i can record a welcome msg for my telephone system?`
`14:03.06`	`mitcheloc_`	`brimstone, you could always try the clasic: mv file /dev/null -f`
`14:03.44`	`*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`14:04.07`	`Seyr`	`E-bola: http://www.voip-info.org/wiki-Asterisk+cmd+Record`
`14:04.12`	`Dr-Linux\|work`	`awww :(`
`14:04.17`	`Dr-Linux\|work`	`fuck mpg123 :(`
`14:04.19`	`Dr-Linux\|work`	`<PROTECTED>`
`14:04.19`	`Dr-Linux\|work`	`14829 root 25 0 2600 2600 2456 R 49.8 0.1 63529m 0 mpg123`
`14:04.19`	`Dr-Linux\|work`	`16531 root 25 0 2596 2596 2456 R 49.8 0.1 1384m 1 mpg123`
`14:04.21`	`caio1982`	`coppice: here steve http://pastebin.ca/153210`
`14:04.33`	`mover`	`urgs`
`14:05.38`	`caio1982`	`coppice: although that all pre9 stuff compiled fine (it's "just" chan_unicall.so giving these errors)`
`14:06.36`	`mover`	`file: i need a little job done. please msg me`
`14:07.00`	`E-bola`	`Seyr: super. thanks`
`14:07.03`	`file`	`I don't do jobs for people, sorry`
`14:07.18`	`*** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu)`
`14:07.27`	`mover`	`ah ok i got wron infos about that`
`14:07.33`	`mover`	`sorry`
`14:08.17`	`mitcheloc_`	`mover: google "asterisk consultants" and ignore the fact that i'm on the list somewhere :)`
`14:08.34`	`coppice`	`caio1982: that makes no sense. the chan_unicall.c for pre8 is exactly the same as for pre9. I just checked on the download server`
`14:08.43`	`sxpert`	`Dr-Linux\|work: version s works very nicely here, despite whatever the doc says ;D`
`14:08.57`	`mover`	`mitcheloc_: i dont need an consultant`
`14:09.03`	`caio1982`	`coppice: iirc, it's really is, you changed the mfcr2 library and nothing more`
`14:09.25`	`*** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net)`
`14:09.33`	`caio1982`	`coppice: do you have any idea about what more i could try before bothering you again? ;)`
`14:09.58`	`coppice`	`dunno. are you sure you are using the downloaded versions of chan_unicall.c?`
`14:10.29`	`caio1982`	`coppice: yep, i did "wget -m -np" in your pre9 directory`
`14:10.44`	`mover`	`i need a little help`
`14:12.02`	`caio1982`	`coppice: well, as i said, the asterisk got patched with your t38 code, so maybe that's his fault?`
`14:13.53`	`[TK]D-Fender`	`mover: Well just ask your question, don't ask to ask..`
`14:15.22`	`coppice`	`caio1982: OK, I will ignore this until you kick me again :-)`
`14:16.42`	`caio1982`	`coppice: haha ok, i'll try to debug things a little bit more`
`14:19.04`	`*** join/#asterisk seebs (n=seebs@216-243-131-210.static.iphouse.net)`
`14:19.47`	`Luke-Jr`	`Any way to test calling from a pay phone?`
`14:20.03`	`Dr-Linux\|work`	`what we can do to get rid of this fucking mpg123`
`14:20.16`	`Dr-Linux\|work`	`not sure how native moh works`
`14:20.19`	`Luke-Jr`	`or better yet-- any way to make calls as if from a payphone? ;)`
`14:20.27`	`Ahrimanes`	`Dr-Linux\|work: use format_mp3 from asterisk-addons`
`14:20.28`	`sxpert`	`Dr-Linux\|work: create a gstreamer resource ;D`
`14:20.32`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`14:20.52`	`sxpert`	`Ahrimanes: doesn't work for network streams`
`14:21.10`	`caio1982`	`coppice: by the way, are you working in unicall for asterisk trunk/1.4 or for openpbx only?`
`14:21.42`	`Ahrimanes`	`Dr-Linux\|work: or convert your mp3's to another format such as gsm, wav or g729 that * understands natively`
`14:21.52`	`Ahrimanes`	`sxpert: no, but he didnt mention streaming..?`
`14:22.02`	`sxpert`	`Ahrimanes: true ;D`
`14:22.34`	`sxpert`	`wav is better though, if you have lots of available space... it's much faster`
`14:23.10`	`Ahrimanes`	`well if you're running purely alaw, you should encode it in alaw, and so on, to avoid transcoding`
`14:23.24`	`sxpert`	`right`
`14:23.50`	`Dr-Linux\|work`	`i see`
`14:24.02`	`*** part/#asterisk bpiper (n=bpiper@70.159.49.40)`
`14:24.03`	`Dr-Linux\|work`	`Ahrimanes, your statement makes sense`
`14:24.55`	`Ahrimanes`	`Dr-Linux\|work: i know ;)`
`14:26.19`	`*** join/#asterisk sevard (n=sev@c-67-182-119-152.hsd1.ca.comcast.net)`
`14:26.22`	`Luke-Jr`	`Does anyone offer payphone termination?`
`14:26.45`	`Luke-Jr`	`where toll-free calls charge the toll-free # for the payphone usage`
`14:26.48`	`coppice`	`vandals terminate huge numbers of payphones`
`14:26.54`	`sxpert`	`lol`
`14:27.01`	`sxpert`	`terminator style ;D`
`14:27.48`	`*** join/#asterisk anthonyl (n=rmerz@c-24-12-179-217.hsd1.il.comcast.net)`
`14:27.58`	`Dr-Linux\|work`	`Ahrimanes, sometime i'll grab your mind to setup at one of my server`
`14:28.28`	`Dr-Linux\|work`	`Ahrimanes, actually i have currently 5 production server, and going to setup a few more in coming week`
`14:28.58`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`14:29.01`	`Dr-Linux\|work`	`currently i'm facing problem with main PBX server with mpg123, it gets high load, it eats all CPU`
`14:29.33`	`blitzrage`	`don't use mpg123, use native MoH`
`14:30.53`	`creativx`	`mpg123 is designed to use all your cpu`
`14:31.26`	`[TK]D-Fender`	`Dr-Linux\|work: The instructions are all in the the sample musiconhold.conf file. Read it.`
`14:31.29`	`Ahrimanes`	`Dr-Linux\|work: that's a common problem, we're using format_mp3.so at my workplace, it works nicely`
`14:31.31`	`sxpert`	`creativx: supposedly not. at least if you use a sensible version`
`14:31.35`	`Seyr`	`I stopped using mpg123 a long time ago because of that as well :-)`
`14:32.00`	`vlt`	`Hello. I try to pick up a ringing phone. In features.conf I enabled "8". When a phone rings I dial "8" on another but I get 503 and CLI says: "Nothing to pick up". What did I miss?`
`14:32.28`	`sxpert`	`the mpg123 for my stream music on hold takes 0.3% cpu. load is 0.00`
`14:32.39`	`Dr-Linux\|work`	`Ahrimanes, do i need only load format_mp3.so module from adones, or what else?`
`14:33.28`	`Dr-Linux\|work`	`sxpert, what's your server uptime?`
`14:34.01`	`sxpert`	`elephant*CLI> show uptime`
`14:34.01`	`sxpert`	`System uptime: 19 hours, 16 minutes, 30 seconds`
`14:34.01`	`sxpert`	`Last reload: 8 hours, 9 minutes, 36 seconds`
`14:34.22`	`sxpert`	`I'm using mpg123 version s`
`14:34.30`	`sxpert`	`successfully I might add`
`14:34.48`	`Ahrimanes`	`Dr-Linux\|work: download asterisk-addons, build format_mp3, install it, load it into asterisk, change your musiconhold.conf file to use native moh instead of mpg123`
`14:36.20`	`[TK]D-Fender`	`Dr-Linux\|work: Its all on the WIKI page for musiconhold.conf and in the same file itself......`
`14:37.09`	`sxpert`	`the one thing I found wierd on the wiki page is that they have config lines in the`
`14:37.21`	`sxpert`	`blah => something,something,something`
`14:37.22`	`sxpert`	`format`
`14:37.47`	`sxpert`	`and the 1.2.11 I have here requires the other format`
`14:37.50`	`E-bola`	`sxpert: if i want music palying instaid of a ringing tone, can asterisk do that?`
`14:37.59`	`E-bola`	`(when ppla re waiting for somebody to pick up the phone int he office)`
`14:38.29`	`sxpert`	`E-bola: yeah. in the dial app you have an M (I think, it may be m) option`
`14:38.40`	`E-bola`	`alright ile look into the dial syntax`
`14:39.32`	`*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)`
`14:40.29`	`blitzrage`	`anyone in here ever build an image from Astlinux's trunk?`
`14:40.49`	`blitzrage`	`I'm at the mkimage.sh script, but not sure if it's trying to write to the file... or to my local partition :)`
`14:41.56`	`benjk`	`yes, I did`
`14:42.10`	`benjk`	`and it booted alright`
`14:42.18`	`benjk`	`just couldn't figure out how to go from there`
`14:42.42`	`benjk`	`and their forum, mailing list or whatever resources there are seem pretty dead`
`14:43.03`	`blitzrage`	`astlinux mailing list? Certainly isn't dead`
`14:43.20`	`Dr-Linux\|work`	`sxpert, what's output of your "uptime" at shelll prompt?`
`14:43.23`	`blitzrage`	`but I'm looking around to figure out how to really use this mkimage.sh script, and can't find anything really in my archives :)`
`14:43.29`	`benjk`	`the most useful information on astlinux was a page on john todd's site`
`14:43.43`	`Dr-Linux\|work`	`mine is :`
`14:43.45`	`Dr-Linux\|work`	`[root@i2c-pbx root]# uptime`
`14:43.45`	`Dr-Linux\|work`	`<PROTECTED>`
`14:43.49`	`Dr-Linux\|work`	`240 days`
`14:43.50`	`tzanger`	`that's not bad`
`14:43.52`	`benjk`	`somewhere on loligo.com`
`14:44.01`	`coppice`	`astlinux is definitely in development - based on questions the developers asked this week`
`14:44.14`	`tzanger`	`$ uptime`
`14:44.14`	`tzanger`	`<PROTECTED>`
`14:44.28`	`mitcheloc_`	`anyone know if Popups is right, or Pop-ups?`
`14:44.41`	`benjk`	`coppice, do you mean to say they are showing interest in integrating Unicall?`
`14:45.07`	`coppice`	`unicall and faxing`
`14:45.36`	`Dr-Linux\|work`	`tzanger, what MOH player you are using?`
`14:45.46`	`Dr-Linux\|work`	`mine is damn mpg123`
`14:45.52`	`tzanger`	`I don't use MOH`
`14:45.54`	`benjk`	`cool`
`14:47.15`	`E-bola`	`Do anybody know how u can pickup another persons call?`
`14:47.17`	`RoyK`	`s/to/at`
`14:47.22`	`RoyK`	`s/to/at/`
`14:47.23`	`ghenry`	`HI all, we normal use Aastra 9133I IP Telephones, but was looking at a cheaper one for a standard SIP handset for non-advanced users. Can anyone recommend the Aastra 9112I phone?`
`14:47.23`	`creativx`	`pickup()`
`14:47.25`	`E-bola`	`like transfer it to your own phone without walking over to his phone`
`14:47.29`	`*** join/#asterisk marv[work] (n=timr@64.89.118.139)`
`14:48.07`	`phearless`	`[QUESTION]`
`14:48.14`	`*** join/#asterisk Slackuser_ (n=aaaa@200.194.102.234)`
`14:48.23`	`phearless`	`how can I pick up the (ringing) phone of somebody else ?`
`14:48.51`	`phearless`	`I give my liver to the person who will help me on this problem`
`14:48.53`	`creativx`	`<PROTECTED>`
`14:48.53`	`creativx`	`that is calling the specified extension. If no context is specified, the current`
`14:48.53`	`creativx`	`context will be used.`
`14:49.03`	`E-bola`	`ya pickup must be the one`
`14:49.07`	`[TK]D-Fender`	`Dr-Linux\|work: : its a 2 minute change! Just READ THE MANUAL!`
`14:49.07`	`phearless`	`thks I will have a look`
`14:49.09`	`creativx`	`hand over that liver.`
`14:49.09`	`E-bola`	`But cant u make a generic pickup?`
`14:49.15`	`E-bola`	`so it just picksup any ringing call?`
`14:49.47`	`[TK]D-Fender`	`ghenry: 9112 should be the same quality as the 9133 just with few features. Though you must be really cheap to be considering them.`
`14:50.21`	`ghenry`	`[TK]D-Fender: Yeah, it's one of our clients. I think the 9133i is perfect`
`14:50.25`	`[TK]D-Fender`	`ghenry: Linksys is valid in their price range where you are if I'm not mistaken`
`14:50.37`	`[TK]D-Fender`	`ghenry: Should try a Ploycom or Cisco then :)`
`14:50.49`	`ghenry`	`[TK]D-Fender: Prefer to stick one make`
`14:51.13`	`ghenry`	`[TK]D-Fender: Yeah, wondering if we should move our standard model to Polycom and Cisco for the guys with lots of money ;-)`
`14:51.19`	`[TK]D-Fender`	`ghenry: Yeah, Aastra is pretty decent overall... not a great finish or usability, but they get the job done.`
`14:51.38`	`*** join/#asterisk }btorch{ (n=kvirc@adelphi.geofocus.com)`
`14:51.47`	`ghenry`	`[TK]D-Fender: They look ok, and have a good quality/per pound, etc.`
`14:51.57`	`[TK]D-Fender`	`ghenry: Polycom is a fair bit more expensive in the UK unfortunately. In North America it'd be a clear choice however.`
`14:51.58`	`mover`	`<PROTECTED>`
`14:52.11`	`}btorch{`	`hey guys sometimes , and its not that often, but I get this -> Got reject for frame 6, retransmitting frame 6 now, updating n_r!`
`14:52.18`	`Dr-Linux\|work`	`fuckkkkkk,`
`14:52.25`	`ghenry`	`[TK]D-Fender: Yeah, that was one of our main reasons. Don't want to price clients out of a move to * and VoIP`
`14:52.28`	`[TK]D-Fender`	`ghenry: But addmitingly Polycom's cost effectiveness is much lower where you are.`
`14:52.28`	`Dr-Linux\|work`	`a bomb blast here :(`
`14:52.37`	`}btorch{`	`asterisk pretty much crashes when that happens, I have to ztcfg -vvvv and then restart asterisk`
`14:52.45`	`[TK]D-Fender`	`ghenry: But for the quality of the product itself it wins hands down.`
`14:52.48`	`}btorch{`	`any ideas what that might be ?`
`14:52.54`	`ghenry`	`[TK]D-Fender: I think we'll get some in to have a play with soon`
`14:53.10`	`E-bola`	`Hmm`
`14:53.17`	`ghenry`	`[TK]D-Fender: Which model would you recommend as a midrange, like the 9133I?`
`14:53.20`	`[TK]D-Fender`	`ghenry: If you want the low-down on them plenty of people use them here.`
`14:53.21`	`E-bola`	`This application also allows you to specify partial extensions. For example, you have block 200 to 209 and you want to pick up any calls that my be ringing on those 10 phones, you just pickup 20 and it'll pick up the most recent ringing one`
`14:53.25`	`E-bola`	`this doesnt apear to work?`
`14:53.29`	`awannabe`	`on the polycom's any there any reason why i cant get it to dial the right voicemailbox #, even though its set right??`
`14:53.30`	`E-bola`	`(its for pickup()`
`14:53.33`	`ghenry`	`[TK]D-Fender: Polycom that is`
`14:53.38`	`[TK]D-Fender`	`ghenry: Depends. If you don't need speakerphone the IP301 is great.`
`14:54.03`	`ghenry`	`[TK]D-Fender: Looking here: http://www.voipon.co.uk/index.php?category=Polycom_IP_Telephones&cPath=1_49`
`14:54.12`	`[TK]D-Fender`	`ghenry: If you do : If you plan on using PoE, get the IP 430, otherwise aim for the IP 501 (w/o PoE adapter)`
`14:54.14`	`ghenry`	`[TK]D-Fender: Seems like the same price as the 9133I`
`14:54.24`	`awannabe`	`ghenry, if you need speakerphone the 501s are great`
`14:54.28`	`DrkShdw`	`hey, quick question.. is a TDM400p able to tell if a line is in use, if it's not the TDM that is using it? (for example, if I have an analog fax machine plugged into the same line, and it's using the line) would the TDM be able to realize the line isn't available as a trunk for a bit?`
`14:54.32`	`sxpert`	`does asterisk caches the music packets or something ?`
`14:54.34`	`ghenry`	`[TK]D-Fender: We have to have the PoE adapters :-(`
`14:54.36`	`phearless`	`<PROTECTED>`
`14:54.36`	`phearless`	`<PROTECTED>`
`14:54.36`	`phearless`	`<PROTECTED>`
`14:54.36`	`phearless`	`<PROTECTED>`
`14:54.39`	`phearless`	`is it normal?`
`14:54.43`	`E-bola`	`hmmm`
`14:54.45`	`phearless`	`I can't pickup`
`14:54.46`	`E-bola`	`apears to be a bug`
`14:54.56`	`E-bola`	`partial pickup doesnt work acording to voip-info.org`
`14:55.32`	`ghenry`	`awannabe, [TK]D-Fender: They're not that cheap ;-) But you do get what you pay for, and I think here (Scotland), the £100~ mark seems to be the sweet spot I'm afraid`
`14:55.50`	`phearless`	`http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup`
`14:55.57`	`phearless`	`there are some weird commands`
`14:55.58`	`[TK]D-Fender`	`phearless: Prove your groups and everything are set up right. Pastebin your configs for EVERYTHING related to those 2 calls.`
`14:56.10`	`*** join/#asterisk vaewyn (i=ejabberd@jabber.andrews.edu)`
`14:56.16`	`phearless`	`which "groups"?`
`14:56.18`	`awannabe`	`i paid $170USD for my 501, and i think they are worth every penny, ive been messing with all the advanced features this week, they rock`
`14:56.23`	`ghenry`	`[TK]D-Fender: I think the price versus feature list, wins it for us. But I haven't played with them, so I might change my mind very quickly ;-)`
`14:56.29`	`[TK]D-Fender`	`ghenry: Aasta is a great value where you are. It would be harder to sell anything else....`
`14:56.34`	`awannabe`	`except, i cant get it to dial the right voicemail box, uggg he`
`14:57.00`	`ghenry`	`[TK]D-Fender: Yeah, plus our reseller discount, it's a good choice. I wish the UK and the US discount rates :-(`
`14:57.01`	`[TK]D-Fender`	`ghenry: But you'll WANT Polycom before you're though Maybe not PAY for them though ;)`
`14:57.05`	`ghenry`	`had the`
`14:57.36`	`ghenry`	`[TK]D-Fender: I agree completely. It's a enterprise name here, so it will depend on the end customer/client for us.`
`14:57.47`	`vaewyn`	`IMO polycoms have the best speakerphone... hands down... I'll use them for conferences they are so good`
`14:57.58`	`ghenry`	`[TK]D-Fender: New to VoIP == Aastra, been in the game before == Polycomm/Cisco`
`14:57.59`	`[TK]D-Fender`	`ghenry: Some would definately pay for Polycom. They are renowned.`
`14:58.15`	`Dr-Linux\|work`	`a huge bomb blast here . :(`
`14:58.18`	`ghenry`	`[TK]D-Fender: Hence previous comment ;-)`
`14:58.25`	`*** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org)`
`14:58.29`	`awannabe`	`and far far better then those evil ciscos!`
`14:58.31`	`coppice`	`i think polcoms are quite poor as conferencing phones. much better is possible`
`14:58.37`	`[TK]D-Fender`	`ghenry: Aastra is old school though. Very classic and a know maker of more "normal" looking phones. Not so great to shuffle a lot of business calls on though.`
`14:58.47`	`*** join/#asterisk Filar (n=none@sw.m5net.com)`
`14:58.59`	`[TK]D-Fender`	`coppice: YMMV. True for most makes & models.`
`14:59.20`	`*** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`14:59.28`	`vaewyn`	`coppice: really? Man... the voice quality and full duplex on those just rocks... zero echo on both ends... and ... well... I just love them :P`
`14:59.48`	`Filar`	`rofl`
`14:59.53`	`coppice`	`yeah, but polycom have a name for conferencing phones, but don't like up to it. their IP phone are crude, with no EC in the handset`
`15:00.26`	`L-info`	`ghenry, if you're in UK, i can point you in the direction of some of the cheapest prices for polycom.. we go through quite a few at my company and searched through a fair few suppliers`
`15:00.38`	`vaewyn`	`coppice: I use the 501 and 601 for conferencing all the time... their true conference phones are WAY overpriced`
`15:00.42`	`brif8`	`I have a remote IP phone that is connected / registered using SIP. shows up in sip show peers. I can make outbound calls from the remote phone. I can dial the "extension" number and it rings. But when the phone is picked up. It keeps ringing on the callED party side then Console shows a "481 "Call Leg/Transaction Does Not Exist" back from Remote_IP_Public_Address" what is wrong ?`
`15:00.42`	`awannabe`	`polycom are crude? this most be a different polycom were talking about`
`15:01.20`	`vaewyn`	`polycom's conference "blobs" are crude... they work fairly well... but I think their desk set phones actually work better`
`15:01.51`	`vaewyn`	`and ... they should always get better... they did buy the best DSPs in the business after all :P`
`15:02.08`	`ghenry`	`[TK]D-Fender: Yeah, the 9133i comes with > 3 lines, so should be ok`
`15:02.23`	`[TK]D-Fender`	`coppice: It does list AEC support, though it doesn't imply that its for the handset as well, though I would suspect it would apply...`
`15:02.26`	`ghenry`	`L-info: We are just about to send a reseller form to http://www.voipon.co.uk/index.php?category=Polycom_IP_Telephones&cPath=1_4`
`15:02.31`	`coppice`	`vaewyn: and what are the best DSPs?`
`15:02.31`	`vaewyn`	`man... I have been gone from here too long... time to play with * some more :P`
`15:02.33`	`ghenry`	`http://www.voipon.co.uk/ even`
`15:02.52`	`[TK]D-Fender`	`ghenry: Keep in mind the concept of LINES, vs CALLS. Any Polycom can shuffle over a dozen calls at a time....`
`15:02.54`	`ghenry`	`I see polycomm in just about every conference room`
`15:03.07`	`ghenry`	`[TK]D-Fender: Ah, yes, thanks for the clearup`
`15:03.14`	`coppice`	`[TK]D-Fender: they don't EC the handset. people complain about echo when they turn up the handset volume`
`15:03.24`	`brif8`	`if I replace host = dynamic in sip.conf with host = IP_Address I then get "Peer '650' is trying to register, but not configured as host=dynamic" AND "Registration from '<sip:650@Asterik_IP;user=phone>' failed for 'Remote_Public_IP'' - Username/auth name mismatch"`
`15:03.26`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`15:03.29`	`ghenry`	`[TK]D-Fender: We are also getting into the net4801 and Astlinux ;-)`
`15:03.40`	`[TK]D-Fender`	`coppice: I think I've heard of it once or twice... very rare. Must be people who are largely deaf :)`
`15:03.52`	`*** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net)`
`15:03.53`	`mitcheloc_`	`Popups or Pop-ups anyone?`
`15:04.06`	`RoyK`	`poopups`
`15:04.13`	`mitcheloc_`	`:(`
`15:04.16`	`L-info`	`ghenry, if you are getting the usual 25% of those prices, then you are pretty much bang on price-wise. We've bought a few times from there and not had any problems`
`15:04.32`	`ghenry`	`L-info: OK, thanks`
`15:04.39`	`L-info`	`np`
`15:04.50`	`[TK]D-Fender`	`ghenry: In any normal business scenario a phone is represented as a single "identity" or "line". How many CALLS that phone is capable of taking on is another matter.`
`15:05.12`	`vaewyn`	`coppice: ASPI... which owns "MELP" and a couple other fun patents`
`15:05.13`	`ghenry`	`[TK]D-Fender: understood.`
`15:05.47`	`ghenry`	`[TK]D-Fender: Average user should be jsut getting calls and then transfering them to the right no, if the misdial`
`15:06.08`	`ghenry`	`[TK]D-Fender: But I think it will depend on per project/client usage`
`15:06.19`	`ghenry`	`[TK]D-Fender: As with any project/client`
`15:06.33`	`[TK]D-Fender`	`ghenry: On my IP 501 at home I use 3 completely seperate SIP registrations, each allowed to handle up to 5 calls (at my discresion) for example. I wanted the independat identities for call identification purposes and VM indication.`
`15:06.40`	`ghenry`	`[TK]D-Fender: So, stay clear of Granstream?`
`15:06.48`	`[TK]D-Fender`	`ghenry: Dear God YES....`
`15:07.03`	`awannabe`	`[TK]D-Fender, and you can check different voicemail boxes from the polycom?`
`15:07.04`	`ghenry`	`[TK]D-Fender: ;-)`
`15:07.11`	`vaewyn`	`ghenry: they are ok for a home phone... in any business stear clear`
`15:07.16`	`rpm`	`Grandstreams like to just lock up and unregister themselves randomly, but they're nice phones when they work.`
`15:07.50`	`coppice`	`vaewyn: compandent owns the patents on MELP`
`15:07.58`	`[TK]D-Fender`	`awannabe: Yes, the messages button is progammable on a per-reg basis as well as multiple MWI.`
`15:08.24`	`ghenry`	`[TK]D-Fender, vaewyn that was our understanding ;-)`
`15:08.34`	`awannabe`	`[TK]D-Fender, can can you have it directly dial your vmbox? as in not just your extension?`
`15:08.35`	`[TK]D-Fender`	`vaewyn: I have an IP 301, IP 430, and IP501 in my bedroom alone ;)`
`15:08.40`	`*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`15:08.55`	`[TK]D-Fender`	`awannabe: Yup. variable per reg any which way I want it.`
`15:09.09`	`vaewyn`	`coppice: ASPI has development and distro rights on that patent though`
`15:09.19`	`awannabe`	`[TK]D-Fender, dang it, wtf cant i get it to work, lol`
`15:09.31`	`ghenry`	`thanks all, doing a price spreadsheet with 9133i and 9112i ;-) All points and advise have been greatly received! Cheers all!`
`15:09.44`	`vaewyn`	`[TK]D-Fender: You know... I hope those arn't all for phone sex lines ;P`
`15:09.49`	`[TK]D-Fender`	`I wouldn't even recomment GD for home. If you wanna be cheap for a home scenario, get an ATA (a DECENT ONE). Otherwise pay for quality.`
`15:10.04`	`[TK]D-Fender`	`vaewyn: Not ALL of them ;)`
`15:10.07`	`vaewyn`	`yeah... at home just run cheap ATA's`
`15:10.16`	`[TK]D-Fender`	`SPA-2002 is a GREAT value.`
`15:10.22`	`vaewyn`	`RTP-300 if you really like cheap and bashing vonage :P`
`15:10.38`	`vaewyn`	`takes some fun to set up the first time... but nice little boxes`
`15:10.49`	`vaewyn`	`like 25$ on ebay :P`
`15:10.50`	`[TK]D-Fender`	`and gives you most of what a user should need. You can still do local conferencing, 3-way calls, etc.... just not as friendly as a true SIP hard-phone.`
`15:12.05`	`vaewyn`	`So... it's been a long time... anyone done work on making voicemail just a standard dialplan while I was gone? :P`
`15:12.18`	`blitzrage`	`vaewyn: thought about it...`
`15:12.25`	`blitzrage`	`didn't get very far unfortunately`
`15:12.45`	`brif8`	`anyone aware of the causes for "481 "Call Leg/Transaction Does Not Exist" back from... message ?`
`15:12.47`	`blitzrage`	`busy with other things`
`15:12.49`	`vaewyn`	`nods I got like the first 2 levels of menus... then ran into a ton of functions we needed so...`
`15:12.52`	`ghenry`	`bye all`
`15:12.58`	`vaewyn`	`ghenry: adious`
`15:12.59`	`blitzrage`	`vaewyn: I hear that`
`15:13.15`	`blitzrage`	`ok... off to setup an svn repo`
`15:13.36`	`vaewyn`	`oh well... I should work on it some more... makes too much sense`
`15:13.37`	`*** join/#asterisk SplasPood (n=jwb@206.252.198.101)`
`15:14.37`	`awannabe`	`to bad the friggin polycoms take a while to boot up though`
`15:14.50`	`vaewyn`	`awannabe: agreed`
`15:15.07`	`vaewyn`	`minute+ sucks`
`15:15.09`	`blitzrage`	`vaewyn: totally agreed`
`15:16.44`	`jbalcomb`	`I would say as rare as it is that the phones reboot its not a big deal`
`15:17.18`	`awannabe`	`yeah, its just when doing testing it takes forever, cause u gotta reboot all the time, heh`
`15:17.46`	`creativx`	`my swissvoices like to reboot and then hang`
`15:18.12`	`awannabe`	`the xml files rock though, its a great way to configure them, i glad they were thinkin`
`15:19.13`	`vaewyn`	`and all phones should have a microbrowser :P`
`15:19.27`	`vaewyn`	`we now have phones for timeclocks :P`
`15:19.57`	`jbroome`	`apparently one of my clients is annoyed by the poly 501 beeping when there's a vm. :)`
`15:20.21`	`vaewyn`	`PEBKAC... answer the @#$$# message :P`
`15:20.26`	`*** join/#asterisk SplasPood (n=jwb@206.252.198.101)`
`15:22.08`	`awannabe`	`im pissed the 501s dont do microbroswer`
`15:22.14`	`awannabe`	`only 601 does i guess :(`
`15:22.19`	`vaewyn`	`yep`
`15:22.27`	`vaewyn`	`would be nice in the 501`
`15:23.03`	`L-info`	`brif8, a 481 is due to a response being recieved that references an existing call or transaction, that the UAS has no record or state information on`
`15:23.26`	`brif8`	`L-info: UAS ? not sure what that is ?`
`15:23.46`	`L-info`	`UAS=user agent server`
`15:24.15`	`brif8`	`oh ok http://pastebin.com/778900 has some debug taken from the console IP Addresses have been masked`
`15:24.23`	`L-info`	`the IP phone, in this scenario`
`15:24.34`	`brif8`	`L-info: anyway to fix it. the IP Phone is a cisco 7960`
`15:25.11`	`awannabe`	`thost 7960s are a damn beast, of junk!! lol`
`15:28.28`	`sxpert`	`moh related misfeature posted ;D`
`15:29.16`	`brif8`	`awannabe: yeah but that is all I have right now`
`15:29.23`	`*** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net)`
`15:29.33`	`aydiosmio`	`alright, I need your guys' help with a game plan`
`15:29.39`	`brif8`	`L-info: any ideas what may be wrong or what I could change ?`
`15:29.52`	`L-info`	`brif8, i dont know much about cisco phones, but a quick google on this shows quite a few instances where the 7960 issues a 481 when it shouldnt`
`15:30.09`	`L-info`	`brif8, what firmware version are you running?`
`15:30.31`	`awannabe`	`brif8, the ciscos are loaded with junk, and then 7960 is a pita, i gave up on them`
`15:30.53`	`aydiosmio`	`client is in a split-level apartment with no existing telephone or ethernet wiring. Wants 4 lines, 3 wired one cordless, with transfer and intercom capabilities. Probably using an asterisk box to connect to existing VoIP provider. Ideas?`
`15:31.07`	`brif8`	`L-info: P0S3-05-3-00`
`15:31.13`	`aydiosmio`	`2.4ghz spectrum is saturated.`
`15:31.50`	`coppice`	`filled with a mass of spread rectum signals`
`15:32.00`	`aydiosmio`	`I think they should go IP phone and wire for ethernet.`
`15:32.02`	`Dr-Linux\|work`	`brif8, your cisco 7960 is behind the nat?`
`15:32.02`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`15:32.02`	`*** mode/#asterisk [+o anthm] by ChanServ`
`15:32.06`	`aydiosmio`	`coppice: I was referring to the clinet site.`
`15:32.10`	`aydiosmio`	`:)`
`15:32.45`	`brif8`	`Dr-Linux\|work: yes but it's router has DMZ enabled to pass all to the Phones IP address`
`15:32.50`	`Dr-Linux\|work`	`brif8, use nat=yes in sip.conf for ip phone user`
`15:33.10`	`brif8`	`Dr-Linux\|work tried with nat=yes and not=no both give the same results`
`15:33.25`	`awannabe`	`brif8, we had to bust out etherreal and look at the packet flow`
`15:33.27`	`Dr-Linux\|work`	`brif8, the asterisk box is located on same subnet?`
`15:33.31`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`15:34.06`	`Dr-Linux\|work`	`brif8, what sip firmware you are using for cisco 7960?`
`15:34.18`	`aydiosmio`	`alright, how about recommendations on economical IP phones with intercom capability`
`15:34.21`	`Dr-Linux\|work`	`i'm using 7.4 for all`
`15:34.21`	`brif8`	`Dr-Linux\|work no the Asterisk <--> Router (Nat) <---> Internet <--> Router (Nat <--> Remote IP Phone`
`15:34.24`	`L-info`	`brif8, that is version 5.3, released in early 2004. current version is 7.5 (i think), and has a list of bug fixes as long as my arm. and i have loong arms`
`15:34.43`	`aydiosmio`	`brif8: haha!`
`15:34.44`	`awannabe`	`brif8, that will never work, thats a double nat`
`15:35.02`	`Dr-Linux\|work`	`brif8, upgrade to 7.4`
`15:35.17`	`brif8`	`ok I'll upgrade to 7.4 Thanks see if that helps`
`15:35.36`	`awannabe`	`_noone_ has gotten double nat to work, right?`
`15:35.36`	`Strom_M`	`good morning`
`15:35.50`	`aydiosmio`	`it can work`
`15:36.14`	`Strom_M`	`double nat works in the same sense that a beat-up old Yugo works`
`15:36.17`	`awannabe`	`and using sip?`
`15:36.42`	`Dr-Linux\|work`	`brif8, also set nat=yes`
`15:37.06`	`awannabe`	`cause i have to switch a server off nat tonight and put it on a public so i can use my phones over the internet, and id rather not, heh`
`15:37.59`	`brif8`	`Dr-Linux\|work I see I have 7.3 Can I put in the SIPMacAddress.cnf file just image_version: "POS-07-03-00" and it will upgrade just that one phone ?`
`15:39.04`	`Dr-Linux\|work`	`brif8, there will be file something like <version>.txt`
`15:39.07`	`juanjoc`	`Is anybody using T.38 in passtrough mode with Asterisk 1.2?`
`15:39.24`	`Dr-Linux\|work`	`that's how phone will grab the correct version from tftp server`
`15:40.02`	`aydiosmio`	`if you set asterisk as the DMZ, it should work no problem`
`15:40.26`	`aydiosmio`	`and set exernip/host`
`15:40.35`	`sevard`	`Strom_M: Stay out of the news`
`15:41.00`	`Strom_M`	`huh?`
`15:42.41`	`*** join/#asterisk lilalinux (i=e-trolle@deepthroat.deswahnsinns.de)`
`15:43.01`	`*** join/#asterisk mercestes (n=merceste@216.54.143.2)`
`15:43.59`	`lilalinux`	`I'm trying to setup a voicemailbox that should answer after 20 seconds`
`15:44.15`	`lilalinux`	`however, after 5 seconds, the CAPI decides to hangup without reason`
`15:44.20`	`lilalinux`	`ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 4`
`15:44.23`	`lilalinux`	`any idea?`
`15:46.01`	`[TK]D-Fender`	`awannabe: You should be able to disable the MWI reminder noise.`
`15:46.28`	`brif8`	`Dr-Linux\|work: in SIPDefault.cnf I have image_version: "P0S-05-03-00" and all the phones are running that version. I was wondering if I add image_version: "P0S-07-03-00" to just the SIPMacAddress.cnf of this phone, will it upgrade just this phone ?`
`15:46.45`	`[TK]D-Fender`	`awannabe: I run double NAT just fine. Plenty of places too....`
`15:47.46`	`*** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18)`
`15:47.52`	`techie`	`7.3 is old`
`15:48.09`	`[TK]D-Fender`	`aydiosmio: When you say 3 lines 1 cordless... you mean 3 CORDED PHONES, and 1 cordless, right? What about actually LINES?`
`15:49.11`	`Dr-Linux\|work`	`brif8, well, you should use MAC<mac>.cnf only for the phone you want to upgrade`
`15:50.28`	`*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)`
`15:51.08`	`mover`	`<PROTECTED>`
`15:52.47`	`sxpert`	`mover: start with copying the file over, and attempt to compile.`
`15:56.11`	`[TK]D-Fender`	`coppice: Still around? I found there IS an option to enable AEC on the handset for Polycoms....`
`15:56.50`	`sxpert`	`[TK]D-Fender: how many menu levels deep ?`
`15:57.05`	`coppice`	`really? well it seems its not the deault, and most people don't know about it. if it works, that might greatly reduce complaints`
`15:57.08`	`Dr-Linux\|work`	`brb dinner time`
`15:57.14`	`[TK]D-Fender`	`sxpert: Menus? What crazyness are you talking about? :)`
`15:57.26`	`[TK]D-Fender`	`coppice: No it is indeed not enabled by default.`
`15:58.23`	`coppice`	`its probably on the menu just below braised sea cucumber, where only the intrepid look :-)`
`15:58.45`	`[TK]D-Fender`	`coppice: Keep an eye out for land-mines ;)`
`15:59.55`	`phearless`	`Pickup does not work`
`15:59.59`	`phearless`	`Pickup() does not work`
`16:00.09`	`sxpert`	`we hear you`
`16:00.13`	`Strom_M`	`phearless, what are you trying to do`
`16:00.28`	`sxpert`	`gah, stupid moh bug...`
`16:01.04`	`*** join/#asterisk b1ch0 (n=aa@host-206-107-150-177.acelerate.net)`
`16:01.34`	`[TK]D-Fender`	`coppice: its under "<VOICE><AEC ..." Independant settings for hs, hf hd`
`16:01.37`	`aydiosmio`	`[TK]D-Fender: 4 lines, 3 corded, 1 cordless, yes`
`16:01.56`	`[TK]D-Fender`	`aydiosmio: not LINES, you seem to talking about PHONES....`
`16:02.07`	`aydiosmio`	`4 lines, 4 phones`
`16:02.54`	`aydiosmio`	`the guy wants to try powerline LAN for IP phones`
`16:02.55`	`[TK]D-Fender`	`aydiosmio: Why so many lines?`
`16:03.02`	`aydiosmio`	`I couldn't help but laugh at him`
`16:03.12`	`[TK]D-Fender`	`aydiosmio: You mean PoE?`
`16:03.17`	`aydiosmio`	`no`
`16:03.22`	`aydiosmio`	`NoP`
`16:03.26`	`aydiosmio`	`powerline ethernet`
`16:03.27`	`[TK]D-Fender`	`aydiosmio: You're terminology is very confusing....`
`16:03.31`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`16:03.38`	`Druken`	`morning everyone`
`16:03.39`	`[TK]D-Fender`	`aydiosmio: your*`
`16:03.42`	`vaewyn`	`that stupid in home networking crud`
`16:03.47`	`aydiosmio`	`yeah sorry, I just woke up`
`16:04.10`	`[TK]D-Fender`	`vaewyn: I pray to God you're mistaken`
`16:04.15`	`vaewyn`	`hehehe`
`16:04.18`	`Druken`	`anyone feel like troubleshooting a digium g729a codec problem for me?`
`16:04.26`	`aydiosmio`	`the problem is...`
`16:04.28`	`aydiosmio`	`wait for it`
`16:04.30`	`aydiosmio`	`YOU`
`16:04.37`	`aydiosmio`	`RMA that shit`
`16:04.37`	`Strom_M`	`Druken, is this the same problem you were having this weekend?`
`16:04.39`	`[TK]D-Fender`	`vaewyn: Those craptasic controllers are WORTHLESS and ran on pure hype a decade ago.`
`16:04.48`	`aydiosmio`	`yeah`
`16:04.51`	`vaewyn`	`[TK]D-Fender: exactly :P`
`16:05.00`	`aydiosmio`	`I can imagine how VoIP would perform`
`16:05.25`	`vaewyn`	`or not as the case may be :P`
`16:05.38`	`[TK]D-Fender`	`aydiosmio: He's seriously fucked up. Powerline lan = BAD and expensive, Telephone line based LAN = just as bad.`
`16:05.41`	`coppice`	`the better ethernet over power units actually seem to work now`
`16:05.41`	`Druken`	`Strom_C: uhmm... no... i have purchased the license, even registered it... however, i can't get asterisk to load the module`
`16:05.47`	`*** join/#asterisk Boggi (n=boggi@luleavnpc091-dot1x.studentby.uit.no)`
`16:05.53`	`[TK]D-Fender`	`aydiosmio: Total stupidity. So why as many lines as phones?`
`16:06.12`	`aydiosmio`	`they're on the phone constanly, some production house`
`16:06.23`	`Strom_M`	`Druken, ok`
`16:06.41`	`Qwell`	`[TK]D-Fender: Somebody brought one of those to Astricon...and it worked. :D`
`16:06.46`	`Strom_M`	`Druken, and calls in g729 format fail, right?`
`16:06.53`	`Qwell`	`EoP`
`16:07.00`	`Druken`	`well, of course, if i can't load the module...`
`16:07.04`	`Druken`	`[Aug 29 10:59:33] WARNING[29990]: loader.c:357 load_dynamic_module: /usr/lib/asterisk/modules/codec_g729a.so: undefined symbol: ast_register_translator`
`16:07.04`	`Druken`	`[Aug 29 10:59:33] WARNING[29990]: loader.c:601 load_resource: Module 'codec_g729a' could not be loaded.`
`16:07.07`	`Druken`	`is what i get`
`16:07.11`	`aydiosmio`	`the issue is finding IP phones with intercom for a reasonable price`
`16:07.14`	`Qwell`	`Druken: Are you trying to load it on trunk or something?`
`16:07.18`	`[TK]D-Fender`	`aydiosmio: AH. Ok well for intercom you'll need 3 SIP hardphones that support it, and the cordless is likely a problem. Not sure if there is one that will support paging on it. Plus the fact that SIP WiFi phones SUCK.`
`16:07.19`	`Qwell`	`because thats not gonna work`
`16:07.26`	`kb3ien`	`running anything over power tends to cause harmful interference, and gets you a cease-and-decist order when it is least convenient.`
`16:07.31`	`[TK]D-Fender`	`Qwell: "worked" is a relative term....`
`16:07.32`	`Druken`	`Qwell: are you fucken kidding me?`
`16:07.40`	`aydiosmio`	`[TK]D-Fender: yes, they're currently using wifi SIP phones, it's a mess`
`16:07.56`	`Qwell`	`Druken: no, not currently`
`16:08.05`	`aydiosmio`	`[TK]D-Fender: check this out http://www.simplycheap.com/aast480ict.html`
`16:08.06`	`[TK]D-Fender`	`aydiosmio: If you are talking about 1-way paging, there are plenty of decent choices. Speakerphone adds a litele extra.`
`16:08.36`	`Filar`	`interesting`
`16:08.37`	`[TK]D-Fender`	`aydiosmio: ICK`
`16:08.56`	`[TK]D-Fender`	`aydiosmio: Its bastardizes its lines to create those virtual "phones" off the main...`
`16:08.57`	`Druken`	`Qwell: what the hell do i need to run then? since the development has turned all gay`
`16:09.15`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.207)`
`16:09.17`	`Qwell`	`Druken: 1.2, until a module for trunk/1.4 is made available`
`16:09.22`	`sxpert`	`Druken: what version of gay is that ?`
`16:09.52`	`aydiosmio`	`[TK]D-Fender: I just need one line for that phone. if you have anything similar, feel free to suggest it`
`16:10.16`	`Druken`	`Qwell: is someone working on it? or is it a wet dream still ?`
`16:10.39`	`Filar`	`the new trunk is actually quite stable for me`
`16:10.44`	`Qwell`	`Druken: somebody may or may not already be working on it...`
`16:12.51`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:12.51`	`*** mode/#asterisk [+o mog] by ChanServ`
`16:12.57`	`Qwell`	`Druken: hang on, and I'll get you an official answer`
`16:12.59`	`Strom_M`	`mog!`
`16:13.06`	`[TK]D-Fender`	`aydiosmio: Well for paging on it you're in trouble most likely. Otherwise run a normal cordless on an ATA`
`16:13.09`	`mog`	`Strom_M,`
`16:13.13`	`Druken`	`Qwell: okie dokie`
`16:13.59`	`aydiosmio`	`[TK]D-Fender: the handset doesn't need to be paged, just the base`
`16:15.12`	`*** part/#asterisk vaewyn (i=ejabberd@jabber.andrews.edu)`
`16:16.39`	`aydiosmio`	`[TK]D-Fender: I'd appreciate any input you had on finding intercom on a cordless IP set`
`16:19.22`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`16:19.36`	`Qwell`	`Druken: The man with the official answer is away from his desk`
`16:21.14`	`[TK]D-Fender`	`aydiosmio: Doesn't exist I suspect.`
`16:21.28`	`aydiosmio`	`oh [TK]D-Fender http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom`
`16:21.35`	`aydiosmio`	`The phones most often mentioned supporting this are:`
`16:21.36`	`aydiosmio`	`<PROTECTED>`
`16:21.42`	`aydiosmio`	`interesting, no?`
`16:21.55`	`[TK]D-Fender`	`aydiosmio: The 480 BASE maybe, but that doesn't imply the handset`
`16:22.05`	`aydiosmio`	`yeah just need to page the base`
`16:22.12`	`*** join/#asterisk NewSole (n=NewSole@d226-106-228.home.cgocable.net)`
`16:22.41`	`[TK]D-Fender`	`aydiosmio: Well tghe base may as well be a seperate phone you know...`
`16:22.58`	`[TK]D-Fender`	`aydiosmio: Effectively has nothing to do with the cordless other than as a dependancy.`
`16:23.02`	`NewSole`	`Any one here have Canada DID's`
`16:23.20`	`*** join/#asterisk Givemelove (n=non@208.57.229.162)`
`16:24.05`	`aydiosmio`	`hm, well this seems to be the best solution.`
`16:24.21`	`aydiosmio`	`3 480i and 1 480i CT`
`16:25.01`	`[TK]D-Fender`	`aydiosmio: Not my choice for phone, but yeah, they'd work.`
`16:25.19`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`16:25.40`	`Dr-Linux\|work`	`Qwell, Sergio is still alive?`
`16:25.59`	`Qwell`	`Dr-Linux\|work: probably not`
`16:26.16`	`Qwell`	`he has 2 days to release a new version, per his promise over a month ago`
`16:27.07`	`Dr-Linux\|work`	`Qwell, but didn't relaease`
`16:27.20`	`*** join/#asterisk RoyK (n=roy@ti211310a080-10830.bb.online.no)`
`16:27.29`	`*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)`
`16:27.37`	`Boggi`	`is there any gui interface for asterisk?`
`16:27.38`	`RoyK`	`jbot: feature freeze?`
`16:27.56`	`RoyK`	`Boggi: vi`
`16:28.07`	`*** join/#asterisk Dr^Mouse (n=noneof@89-145-198-64.xdsl.murphx.net)`
`16:28.12`	`Dr-Linux\|work`	`vi?`
`16:28.12`	`Boggi`	`RoyK, vi ?`
`16:28.18`	`Qwell`	`Druken: "we have said multiple times that the g729 codec for trunk will not be available until trunk is ready for beta testing, which it is not"`
`16:28.20`	`RoyK`	`~vim`
`16:28.21`	`jbot`	`from memory, vim is better than emacs`
`16:28.23`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`16:28.28`	`Qwell`	`Druken: "we have said multiple times that the g729 codec for trunk will not be available until trunk is ready for beta testing, which it is not"`
`16:28.43`	`Boggi`	`RoyK, how is vi a gui?`
`16:28.49`	`Qwell`	`gvim is a gui`
`16:28.54`	`RoyK`	`:)`
`16:28.58`	`NewSole`	`lol`
`16:29.05`	`sxpert`	`Boggi: vi is a gui in an xterm ;D`
`16:29.06`	`Druken`	`so in otherwords.... sucks to be you... you shouldn't have upgraded, your dumb`
`16:29.20`	`*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)`
`16:29.32`	`Boggi`	`okey, so that's the best "gui" there is for asterisk?`
`16:29.34`	`[TK]D-Fender`	`Druken: that'd be "you're" ;)`
`16:29.39`	`phearless`	`<phearless> Pickup() does not work`
`16:29.39`	`phearless`	`<sxpert> we hear you`
`16:29.39`	`phearless`	`<Strom_M> phearless, what are you trying to do`
`16:29.41`	`Qwell`	`[TK]D-Fender++`
`16:29.48`	`[TK]D-Fender`	`Qwell: :D`
`16:29.52`	`phearless`	`err... I would like to pickup the phone call of a ringing phone`
`16:29.59`	`phearless`	`in the office`
`16:30.07`	`Dr^Mouse`	`ok simple question. is the jitter buffer for SIP availavle in the release version (1.2.11)? if so how do i use it (jitter is the biggest problem i currently have) and if not, any timescale on when?`
`16:30.08`	`Strom_M`	`phearless, pastebin your extensions.conf`
`16:30.12`	`phearless`	`ok`
`16:30.19`	`phearless`	`exten => _7.,1,Pickup(${EXTEN:2})`
`16:30.22`	`Dr^Mouse`	`*available`
`16:30.27`	`sxpert`	`the one thing I don't understand is this "yeah there's that stable version, but we don't really care about it, all new features / patches should be for trunk" attitude`
`16:30.40`	`RoyK`	`jbot: feature freeze is when a project like asterisk rejects new functionality, that is, new stuff that changes the old ways unless they are (a) parts of important bugfixes or (b) are submitted/written by important people or, implying (b), (c) just are cool enough`
`16:30.41`	`jbot`	`okay, RoyK`
`16:30.41`	`Strom_M`	`phearless, why chop off two digits?`
`16:30.57`	`eKo1`	`Dr^Mouse: are you having jitter within your network?`
`16:31.19`	`RoyK`	`Dr^Mouse: asterisk-backports.org :)`
`16:31.21`	`Dr^Mouse`	`no, out onto internet to my sip line supplier`
`16:31.31`	`RoyK`	`eKo1: if using voip over a wan, there's always jitter and some packet loss`
`16:31.44`	`Corydon-w`	`sxpert: it's release, not stable`
`16:31.49`	`eKo1`	`RoyK: well, that depends on the wan`
`16:32.01`	`eKo1`	`I have no jitter and packet loss on my wan`
`16:32.05`	`Dr^Mouse`	`ooo @ backports, didnt know that was there :)`
`16:32.15`	`RoyK`	`eKo1: are we arguing for the sake of arguing here, sir?`
`16:32.30`	*** join/#asterisk trelane` (n=trelane@unaffiliated/trelane)
`16:32.31`	`eKo1`	`I'm just refuting your statement`
`16:32.34`	`RoyK`	`Dr^Mouse: we paid for the jb to be written for 1.2 in the first place. we'be been running it stably for some three months now......`
`16:32.40`	`[TK]D-Fender`	`Boggi: Pretty much every GUI for * takes complete control over all aspects of * and you get a "cannerd" PBX out of it. If thats what you want you can take any OTHER solution out there.`
`16:32.49`	`Corydon-w`	`sxpert: we do not call it "stable" for a very good reason... often, trunk is more stable than release`
`16:32.49`	`RoyK`	`eKo1: well, MOST people are not that happy with their WANs`
`16:33.11`	`phearless`	`<Strom_M> phearless, why chop off two digits?`
`16:33.13`	`RoyK`	`eKo1: a 10x or 50x overbooked DSL link doesn't give all too good results to everyone`
`16:33.15`	`phearless`	`I copied from...`
`16:33.16`	`phearless`	`s1`
`16:33.16`	`Boggi`	`[TK]D-Fender, okey, thanks for the answer.`
`16:33.20`	`phearless`	`1second`
`16:33.27`	`eKo1`	`RoyK: well, I can't speak for most`
`16:33.32`	`phearless`	`from the wiki`
`16:33.39`	`phearless`	`ok shit somebody ask me something`
`16:33.56`	`NewSole`	`Any one here have Canada DID's`
`16:33.59`	`RoyK`	`eKo1: we have several thousand customers on crappy wan links, and living without a jitterbuffer was a real PITA`
`16:34.35`	`Strom_M`	`phearless, you dont have to repeat my question back to me before you answer it`
`16:34.44`	`eKo1`	`RoyK: yeah; having a jitter buffer really improves quality`
`16:34.55`	`Strom_M`	`phearless, blindly copying without understanding what you're doing will lead to problems`
`16:35.00`	`phearless`	`I agree`
`16:35.12`	`RoyK`	`eKo1: yes, and the pope IS indeed catholic :P`
`16:35.12`	`phearless`	`sorry I come back later !`
`16:35.22`	`RoyK`	`eKo1: the PLC in there really rocks as well`
`16:35.43`	`eKo1`	`PLC?`
`16:35.55`	`RoyK`	`~plc`
`16:35.56`	`jbot`	`hmm... plc is Packet Loss Concealment, a method of recreating lost packets by interpolating or merely retransmitting old ones`
`16:36.42`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`16:36.42`	`aydiosmio`	`haha`
`16:36.45`	`aydiosmio`	`interpolating`
`16:36.47`	`eKo1`	`Ah!`
`16:36.52`	`aydiosmio`	`that better be a damn small packet`
`16:37.06`	`RoyK`	`jbot: plc is also the new jitterbuffer has a rather good PLC that uses the interpoation method`
`16:37.07`	`jbot`	`RoyK: okay`
`16:37.17`	`RoyK`	`~plc`
`16:37.19`	`jbot`	`i heard plc is Packet Loss Concealment, a method of recreating lost packets by interpolating or merely retransmitting old ones, or the new jitterbuffer has a rather good PLC that uses the interpoation method`
`16:38.15`	`RoyK`	`jbalcomb: no, plc is Packet Loss Concealment, a method of recreating lost packets by interpolating or merely retransmitting old ones. The new 1.4 jitterbuffer has an interpolating PLC.`
`16:38.32`	`RoyK`	`jbot: no, plc is Packet Loss Concealment, a method of recreating lost packets by interpolating or merely retransmitting old ones. The new 1.4 jitterbuffer has an interpolating PLC.`
`16:38.33`	`jbot`	`RoyK: okay`
`16:38.37`	`RoyK`	`jbalcomb: sorry :P`
`16:39.04`	`Dr^Mouse`	`RoyK: ok, i know im being a complete retard here, but its been so long since ive had to do this - how do i use the patch (1.2.11-jb+lots.patch) on my 1.2.11 code?`
`16:39.32`	`RoyK`	`cat 1.2.11-jb+lots.patch \| patch -p0`
`16:39.36`	`RoyK`	`or -p1`
`16:39.41`	`RoyK`	`-p1 iirc`
`16:39.45`	`Qwell`	`why cat it?`
`16:39.48`	`RoyK`	`from the source dir`
`16:39.51`	`Qwell`	`waste of a process`
`16:39.51`	`blitzrage`	`jbot: no, plc is Packet Loss Concealment, a method of recreating lost packets by interpolating or merely retransmitting old ones. An interpolating PLC was introduced in the jitterbuffer of Asterisk 1.4.`
`16:39.53`	`jbot`	`blitzrage: okay`
`16:39.53`	`RoyK`	`Qwell: whatever :P`
`16:40.02`	`blitzrage`	`RoyK: that way we don't say "new"`
`16:40.05`	`RoyK`	`blitzrage: thanks`
`16:40.06`	`blitzrage`	`since 1.4 won't always be new :)`
`16:40.13`	`RoyK`	`right`
`16:40.16`	`blitzrage`	`minor tweak`
`16:40.27`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`16:40.27`	`RoyK`	`yep`
`16:40.47`	`Dr^Mouse`	`RoyK: so is it -p0 or -p1`
`16:41.10`	`RoyK`	`Dr^Mouse: let me check`
`16:41.25`	`RoyK`	`-p1`
`16:41.34`	`RoyK`	`from the asterisk-1.2.11 dir`
`16:41.44`	`seebs`	`Hey, anyone here know much about asterisk-bsd?`
`16:41.48`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`16:41.57`	`*** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net)`
`16:42.06`	`RoyK`	`seebs: apart from zaptel, it should be pretty much the same`
`16:42.09`	`seebs`	`Okay.`
`16:42.18`	`Dr^Mouse`	`RoyK: cheers`
`16:42.25`	`seebs`	`In particular, I am thinking to try to update the NetBSD zaptel port. It looks, uhm. Interesting. :)`
`16:42.39`	`RoyK`	`Dr^Mouse: you'll need spandsp, as mentioned`
`16:42.44`	`*** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982)`
`16:42.50`	`RoyK`	`the patch uses the spandsp g.726 codec`
`16:42.56`	`*** join/#asterisk bluecrow1976 (n=bluecrow@bigpdc.bignetworks.com)`
`16:43.15`	`*** part/#asterisk spr1te (i=spr1te@194.187.130.227)`
`16:43.17`	`*** join/#asterisk spr1te (i=spr1te@194.187.130.227)`
`16:43.18`	`Dr^Mouse`	`RoyK: thought that was just for using the G726 codec, which i dont plan to do (just yet at least)`
`16:43.49`	`Givemelove`	`Hey guys`
`16:43.58`	`Givemelove`	`I've got a weird issue with asterisk`
`16:44.15`	`Givemelove`	`when I have a phone connected onto the same network as asterisk`
`16:44.17`	`Givemelove`	`noprob`
`16:44.34`	`Givemelove`	`but when I connect the same phone (or softphone whatever) from a different network`
`16:44.38`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`16:44.39`	`Strom_M`	`when you press enter at the end of every phrase, big problem`
`16:44.58`	`RoyK`	`Dr^Mouse: that, and fax stuff. just remember to add a noload => codec_g726.so in modules.conf if you're not using it, or asterisk will fail to start`
`16:45.08`	`Givemelove`	`onto all the calls to the asterisk system, such as VoicemailMain or IVR, I hear no audio`
`16:45.12`	`Dr^Mouse`	`okies`
`16:45.15`	`smackus`	`using chanspy, can I set it up to specify what extension to listen to? right now I have it so that it is I think random. exten => 1234,1,Chanspy(\|q)`
`16:45.19`	`Strom_M`	`Givemelove, that's a classic SIP and NAT issue`
`16:45.20`	`Givemelove`	`but the outgoing calls works`
`16:45.36`	`Givemelove`	`Strom_M the asterisk is onto the DMZ`
`16:45.59`	`Strom_M`	`Givemelove, regardless, the box has to go through network address translation of some variety`
`16:46.17`	`Dr^Mouse`	`ok, obviously it needs the library to even compile. can i just delete the codec_g726.c, or do i have to modify makefiles or some such?`
`16:46.18`	`Givemelove`	`ok`
`16:46.48`	`Givemelove`	`any hint for resolving this?`
`16:47.01`	`Strom_M`	`put the asterisk box in front of the NAT?`
`16:47.31`	`RoyK`	`Givemelove: sounds like one or both are behind nat or something`
`16:47.37`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`16:47.42`	`RoyK`	`Givemelove: pretty usual problem, at least if using sip`
`16:47.42`	`Givemelove`	`yeah, they are`
`16:47.54`	`TripleFFFF`	`1.2.11 sucks balls`
`16:48.02`	`TripleFFFF`	`dtmf.. sip echo .. etc`
`16:48.06`	`RoyK`	`TripleFFFF: works well here`
`16:48.09`	`Strom_M`	`TripleFFFF, works for me`
`16:48.10`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.73)`
`16:48.12`	`TripleFFFF`	`weir`
`16:48.29`	`Givemelove`	`I'll give a try`
`16:48.31`	`Givemelove`	`BRB`
`16:48.36`	`RoyK`	`TripleFFFF: echo is really rare, unless you're using a speakerphone or something`
`16:48.44`	`TripleFFFF`	`hmmm`
`16:48.58`	`RoyK`	`with aucustic feedback and not an echocancellor from the midst of hell`
`16:48.59`	`_alex_mx_`	`smackus, try Chanspy(SIP/XXX\|q)`
`16:49.09`	`TripleFFFF`	`and dtmf`
`16:49.11`	`Dr^Mouse`	`im having no (aditional) problems in 1.2.11`
`16:49.17`	`TripleFFFF`	`got someone said they cant pass dtmf no more`
`16:49.20`	`TripleFFFF`	`since we on 1.2.11`
`16:49.23`	`RoyK`	`TripleFFFF: tried debugging the stuff?`
`16:49.28`	`Strom_M`	`what dtmfmode are you using?`
`16:49.34`	`TripleFFFF`	`sames as always`
`16:49.39`	`RoyK`	`TripleFFFF: tried rolling back to 1.2.10?`
`16:49.46`	`Dr^Mouse`	`cd codecs`
`16:49.54`	`TripleFFFF`	`cd ??`
`16:50.01`	`Dr^Mouse`	`damit i keep doing that - wrong window`
`16:50.14`	`Strom_M`	`TripleFFFF, what dtmfmode are you using?`
`16:50.21`	`TripleFFFF`	`hmmm bluetooth stacks are not gd..tried 4 diff usb dongles.. 5 diff headsets`
`16:50.25`	`*** join/#asterisk Gregabyte (n=greg@gateway.digium.com)`
`16:50.30`	`TripleFFFF`	`after a while it looks like its buffering`
`16:50.31`	`RoyK`	`find / -name Dr^Mouse -exec rm -f {} \;`
`16:50.49`	`*** join/#asterisk charles___ (n=charles@fw.invosat.com)`
`16:51.21`	`Dr^Mouse`	`lol`
`16:51.44`	`*** join/#asterisk rnovotny22 (n=Bob@198.57.19.126)`
`16:51.52`	`Druken`	`1.2.11 works for me, ASIDE FROM NO GOD DAMN G729!!!!!`
`16:51.58`	`Dr^Mouse`	`i need a faster machine to compile on`
`16:52.05`	`Qwell`	`Druken: g729 works fine on 1.2.11`
`16:52.05`	`RoyK`	`Druken: tried buying something from digium yet?`
`16:52.11`	`TripleFFFF`	`relaxdtmf=yes ; Relax dtmf handling`
`16:52.13`	`TripleFFFF`	`does what ?`
`16:52.18`	`Dr^Mouse`	`Drunken - g729 works fine for me on 1.2.11`
`16:52.19`	`Druken`	`RoyK: uhmm... yes`
`16:52.27`	`RoyK`	`it doesn't stress dtmf :)`
`16:53.11`	`Druken`	`oh.. apparently i'm using trunk.. not 1.2.11`
`16:53.13`	`Druken`	`my bad`
`16:53.15`	`TripleFFFF`	`<PROTECTED>`
`16:53.18`	`TripleFFFF`	`my default`
`16:54.44`	`*** join/#asterisk supjigatr (n=syslod@152.53.16.10)`
`16:55.04`	`aydiosmio`	`anyone know what 8ft lengths of cable channel for walls/baseboards costs retail?`
`16:55.16`	`supjigatr`	`Does anyone have an example of voicemail main for 86 dial to voicemail with no password.`
`16:55.25`	`aydiosmio`	`I'm such a installer noob.`
`16:55.38`	`supjigatr`	`U mean panduit?`
`16:55.50`	`Druken`	`conduit`
`16:56.02`	`aydiosmio`	`the channels you can snap closed ro surface routing`
`16:56.08`	`aydiosmio`	`s/ro/for`
`16:56.23`	`supjigatr`	`plastic or metal?`
`16:56.45`	`supjigatr`	`we use monorail and its like $23 per 10ft stick.`
`16:57.03`	`aydiosmio`	`plastic`
`16:57.15`	`aydiosmio`	`$23!?`
`16:57.18`	`aydiosmio`	`good lord.`
`16:57.33`	`Druken`	`i hear duct tape works nicely.... even comes in an easy to dispence roll`
`16:57.49`	`aydiosmio`	`yeah`
`16:57.55`	`*** join/#asterisk giesen (i=giesen@dirtypackets.net)`
`16:58.02`	`aydiosmio`	`but my client's office demanda s slightly higher standard.`
`16:58.05`	`giesen`	`Anyone have experience with astmanproxy?`
`16:58.07`	`*** join/#asterisk zedkatuf (n=zedkatuf@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk)`
`16:58.09`	`giesen`	`I authenticate fine to it`
`16:58.20`	`Strom_M`	`aydiosmio, you're in the u.s.?`
`16:58.22`	`aydiosmio`	`ah`
`16:58.23`	`giesen`	`but it doesnt seem to be passing commands to asterisk manager`
`16:58.26`	`aydiosmio`	`this is what I was looking for`
`16:58.28`	`aydiosmio`	`http://cableorganizer.com/surface-raceways/latching.html`
`16:58.32`	`aydiosmio`	`$6`
`16:58.36`	`aydiosmio`	`Strom_M: yes`
`16:58.53`	`giesen`	`Aug 29 12:56:41: set destination:`
`16:58.56`	`bluecrow1976`	`Any switch => gods around? Is there a way to get switch to handle dial arguments such as Dial(IAX2/provider/number,,tTdDwW) would?`
`16:58.57`	`Strom_M`	`aydiosmio, go to a Graybar counter and look at the available options`
`16:59.02`	`giesen`	`that's what I see after I try to initiate a call`
`17:03.30`	`*** join/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com)`
`17:03.50`	`supjigatr`	`How do I make voicemail skip the password?`
`17:04.48`	`Strom_M`	`you don't`
`17:04.53`	`Strom_M`	`er`
`17:04.59`	`Strom_M`	`wait, i looked at the wrong app`
`17:05.11`	`Strom_M`	`you pass it the s flag :)`
`17:05.28`	`*** join/#asterisk new2voip (n=chatzill@user-24-214-152-5.knology.net)`
`17:05.42`	`Strom_M`	`supjigatr, "show application voicemailmain" at the CLI`
`17:06.46`	`Dr^Mouse`	`hmm... ok then i dont think that jitter was my problem after all. anyone know what could be causing this? i make a call to my internet line provider (lvdx) and the sound is 'choppy'. i dont think silence suppression is enabled (it isnt on the phone), im using 2 frames per tx on g729. any ideas?`
`17:07.20`	`Strom_M`	`Dr^Mouse, does the choppiness go away if you use a different codec?`
`17:07.40`	`Dr^Mouse`	`hmm, ill check, one sec...`
`17:08.07`	`supjigatr`	`Strom_M: The instructions are confusing. Do I do a s in front of the extension or exten\|s ? I've tried both and neither seem to work.`
`17:08.10`	`new2voip`	`do I need zaptel, zapata, libpri if I am only going to use softphones?`
`17:09.08`	`Strom_M`	`supjigatr, read more carefully, please`
`17:09.08`	`Strom_M`	`the instructions tell you how`
`17:09.22`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`17:09.33`	`Strom_M`	`new2voip, no, unless you want to do meetme, in which case you should compile zaptel so you can use ztdummy`
`17:10.05`	`[TK]D-Fender`	`supjigatr: Indeed it is rather clear how......`
`17:10.06`	`Dr^Mouse`	`Strom_C: no, it is the same using ulaw`
`17:10.20`	`supjigatr`	`So no sample huh?`
`17:10.51`	`Dr^Mouse`	`supjigatr: VoicemailMain(s123)`
`17:11.24`	`*** part/#asterisk bethaud (n=bethaud@host-84-9-82-198.bulldogdsl.com)`
`17:11.37`	`Dr^Mouse`	`supjigatr: add an 's' to the beginning of the box number, but reading the documentation would have told you that`
`17:11.47`	`supjigatr`	`Dr^Mouse: I've tried that. Also tried "as the help for voicemail says" a \|s option at the end.`
`17:12.03`	`supjigatr`	`Actually the documentation now says \|s does it`
`17:12.11`	`supjigatr`	`The wiki says s at the front.`
`17:12.16`	`supjigatr`	`Neither seem to work.`
`17:12.43`	`Dr^Mouse`	`ah so it does, not done it in a while myself`
`17:12.49`	`[TK]D-Fender`	`supjigatr: pastbin what you are attempting.`
`17:12.53`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`17:13.48`	`*** join/#asterisk spr1te (i=spr1te@194.187.130.227)`
`17:14.07`	`Dr^Mouse`	`so then, 'choppy' sound.... not so sure if 'choppy' is right, it 'pops'.`
`17:14.16`	`Strom_M`	`Dr^Mouse, SIP or IAX?`
`17:14.26`	`supjigatr`	`http://pastebin.ca/153490`
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`17:14.45`	`Dr^Mouse`	`Strom_M: SIP (lvdx dont do IAX unfortunately :( )`
`17:15.06`	`Strom_M`	`supjigatr, you should be using ${CALLERID(num)}`
`17:15.16`	`Dr^Mouse`	`Strom_M: i wish they would start, itd make things SO much easier`
`17:15.24`	`NewSole`	`Any one here have Canada DID's`
`17:15.35`	`Strom_M`	`Dr^Mouse, without hearing it, it's tough to diagnose`
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`17:15.44`	`EyeCue`	`question strom`
`17:15.45`	`*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)`
`17:15.51`	`Strom_M`	`answer EyeCue`
`17:15.56`	`EyeCue`	`:)~`
`17:16.31`	`RoyK`	`~feature freeze`
`17:16.32`	`jbot`	`hmm... feature freeze is when a project like asterisk rejects new functionality, that is, new stuff that changes the old ways unless they are (a) parts of important bugfixes or (b) are submitted/written by important people or, implying (b), (c) just are cool enough`
`17:16.33`	`EyeCue`	`say i/we/some people were to create like an asterisk boxen with like, teh awesome hardware, and stuff. can you just arbitrarily broadcast your sip domain to the interweb?`
`17:16.53`	`Strom_M`	`EyeCue, what?`
`17:17.02`	`EyeCue`	`i mean, we have quite a few isp's in australia that provide voip accounts along with adsl right`
`17:17.11`	`[TK]D-Fender`	`supjigatr: now show us CLI output of a failed attempt`
`17:17.13`	`EyeCue`	`how do i 'link' in with that infrastructure?`
`17:17.28`	`EyeCue`	`well, not "how do i" but more rather "how does one"`
`17:17.33`	`Strom_M`	`EyeCue, what do you want to accomplish, exactly`
`17:17.40`	`EyeCue`	`well ok`
`17:17.42`	`[TK]D-Fender`	`NewSole: www.unlimitel.ca`
`17:17.49`	`EyeCue`	`iinet for example, with their dsl accounts, gives a user a phone number`
`17:17.59`	`EyeCue`	`which is dialable from any normal pstn phone service right`
`17:18.04`	`EyeCue`	`how do they 'get' the numbers`
`17:18.16`	`EyeCue`	`or is like kinda like an ip allocation, but for phone number`
`17:18.18`	`Strom_M`	`how does the telco get a number?`
`17:18.23`	`EyeCue`	`im trying to conceptualize`
`17:18.27`	`EyeCue`	`i dunno :)`
`17:18.33`	`*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)`
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`17:18.37`	`Strom_M`	`im clarifying, not asking`
`17:19.03`	`EyeCue`	`well im asuuming the telco can just make em up so to speak`
`17:19.03`	`Druken`	`EyeCue: with special "telephone lines" that costs lots of money and special hardware`
`17:19.34`	`EyeCue`	`hmm.`
`17:19.47`	`Strom_M`	`EyeCue, I don't know how it's accomplished in australia, but here in the US, telcos have to apply for office codes from the north american numbering plan administrator`
`17:20.01`	`EyeCue`	`yeh theres probably something similar`
`17:20.15`	`EyeCue`	`so the only thing anyone without $$$$$$$$$$$++ would be to create an 'internet only`
`17:20.19`	`EyeCue`	`' voip system`
`17:20.30`	`EyeCue`	`well, and outbound to normal ph numbers`
`17:20.45`	`*** part/#asterisk arisjr (n=arisjr@201.37.66.122)`
`17:20.53`	`Strom_M`	`EyeCue, I'm getting very irritated at the vagueness of your questions. Please outline the specific goal you are trying to accomplish and I can try to help you from there.`
`17:21.00`	`EyeCue`	`ponders`
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`17:21.13`	`EyeCue`	`If i knew the terminology i could probably be less vague.`
`17:22.13`	`EyeCue`	`Umm, say with a big fat pipe (100mbit+) and a suitable server sitting in colo, i could create a 'service' that allowed people to only talk to other 'sip' services/devices right?`
`17:22.27`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`17:22.29`	`Strom_M`	`why do you keep putting things in quotes?`
`17:22.32`	`[TK]D-Fender`	`Strom_ : He's pretty clear. He wants to know how telco's are assigned ranges within a structure like our NPA-NXX.`
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`17:22.44`	`Strom_M`	`[TK]D-Fender, that we already settled`
`17:22.48`	`[TK]D-Fender`	`Strom_ : like what is the process for "reserving" numbers and so on.`
`17:22.57`	`EyeCue`	`well kinda`
`17:22.58`	`[TK]D-Fender`	`Strom_ : I'm catchin up it seems :)`
`17:23.02`	`EyeCue`	`im trying to think of how to articulate it`
`17:23.13`	`Strom_M`	`EyeCue, something like free world dialup?`
`17:23.16`	`EyeCue`	`well yeh`
`17:23.18`	`eKo1`	`What does it mean for a context to lack a type. I'm getting warning messages saying things like: 'pstn' lacks type`
`17:23.18`	`Dr^Mouse`	`Strom_M: blessed poo - doing a mixmonitor on the channel, no popping. its between the phone and the system, not on the other side (coz i heard it in the phone)`
`17:23.30`	`EyeCue`	`they have like exten@domain type of a setup right ?`
`17:23.42`	`Strom_M`	`EyeCue, no, they assign numbers`
`17:23.57`	`EyeCue`	`ok, rephrase, this might be better`
`17:24.32`	`[TK]D-Fender`	`EyeCue: For YOU to become a "mini-telco" you'd need some sort of digital link (T1/E1 or bigger) upon which you can use DID's. you would then have a "pool" of lines to use for each connection you would like to terminate/originate and you could rig the callerID to the DID YOU assign to a given customer for instance`
`17:25.46`	`EyeCue`	`if i setup a colo box with asterisk/billing/mysql accounts and such, and assign each account an extension/number/whatever, register a domain for it, how do i 'broadcast/advertise' my service to the internet?`
`17:25.46`	`EyeCue`	`hows that?`
`17:25.46`	`Strom_M`	`much better`
`17:25.46`	`EyeCue`	`and how to other sip services know about me?`
`17:25.46`	`EyeCue`	`and my users`
`17:25.46`	`EyeCue`	`:D`
`17:25.46`	`[TK]D-Fender`	`EyeCue: Advertise in the normal ways, TV, radio, printed publications, search engines, etc. Start a web-page, etc.`
`17:25.46`	`EyeCue`	`it was a conceptual thing ;)`
`17:25.46`	`Strom_M`	`the answer is that I don't know enough about SIP to be able to tell you`
`17:25.47`	`EyeCue`	`well i dont mean advertise like that`
`17:25.58`	`EyeCue`	`i mean more in the context of 'discoverable'`
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`17:26.13`	`*** part/#asterisk new2voip (n=chatzill@user-24-214-152-5.knology.net)`
`17:26.21`	`Strom_M`	`Dr^Mouse, is the phone connected locally to the asterisk box?`
`17:26.40`	`EyeCue`	`technically people could just use sip uri's to contact my users`
`17:27.05`	`Dr^Mouse`	`Strom_M: yes, its in the same room, in the same 'bank' on a high quality switch.`
`17:27.26`	`Strom_M`	`Dr^Mouse, what kind of phone?`
`17:27.53`	`smackus`	`ok, so I am using: exten => *9,1,Chanspy(Agent\|q). How can I change which agent I am listening to?`
`17:27.58`	`EyeCue`	`can hard phones use sip uri's ?`
`17:28.06`	`Dr^Mouse`	`Strom_M: Grandstream GXP-2000. But its wierd, ive never had sound like this before, with any of them.`
`17:28.06`	`smackus`	`i tried to dial the agent number followed by the # sign. that did not work`
`17:28.33`	`Strom_M`	`smackus, thats because your extension will only match the string "*9"`
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`17:29.06`	`smackus`	`that is just the extension to launch chanspy i thought.`
`17:29.30`	`Strom_M`	`ChanSpy([chanprefix][\|options])`
`17:29.54`	`smackus`	`here is what I am reading from: (e.g. run Chanspy(Agent) and dial 1234# while spying to jump to channel Agent/1234)`
`17:30.25`	`smackus`	`so are you saying that i have to have exten => 1234,1,chanspy... for this to work?`
`17:31.14`	`Dr^Mouse`	`Strom_M: erm, ok, sorry about that, looks like its a phone problem. seeing as i just dialled into my milliwatt extension, and got no sound whatsovere unless i spoke. it appears the phone is using silence suppression even when i disable it.`
`17:31.34`	`smackus`	`Dialing a series of digits followed by # builds a channel name to append to <scanspec>`
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`17:34.41`	`Dr^Mouse`	`%*&$! i cannot beleive ive just spent an entire day debugging this problem, and in the end its that the phone isnt doing what is told!`
`17:35.06`	`sezuan`	`I see in some extensions.conf examples lines like 'exten => 49231/+4912334,1,foobar'`
`17:35.22`	`Qwell`	`sezuan: The second part matches the "incoming" callerid`
`17:35.37`	`Dr^Mouse`	`just to check, if i dial an ext which just puts me into the milliwatt() app, and i get no audio unless i speak, thats the phone doing silence suppression right?`
`17:35.55`	`Qwell`	`Dr^Mouse: probably, yes`
`17:36.20`	`sezuan`	`Qwell: Thanks.`
`17:37.12`	`_alex_mx_`	`smackus, what we do is Set(SPYGROUP=whatever) in the dialplan then using your example, exten => *9,1,Set(SPYGROUP=whatever exten => *9,2,Chanspy(SIP\|g(whatever)b)`
`17:37.46`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.73)`
`17:37.59`	`charles___`	`hey guys`
`17:38.02`	`Dr^Mouse`	`Qwell: is there any way yet of using a separate timing source, other than the incoming rtp stream? like the zaptel timer?`
`17:38.08`	`charles___`	`does anyone have cisco terminating calls into an Asterisk ?`
`17:38.15`	`Qwell`	`Dr^Mouse: in trunk/1.4, I believe, yes`
`17:38.55`	`Dr^Mouse`	`Qwell: :( i have been categoricaly told not to use trunk, as i am working on the office system and customers systems.`
`17:39.09`	`Qwell`	`Dr^Mouse: Then wait until 1.4 is released...`
`17:39.52`	`TripleFFFF`	`lol anyonw have app_Rxfax working on 1.2.11 ? if so can they pastebin email the .c file s and the makefilel for paps`
`17:39.54`	`TripleFFFF`	`apps`
`17:40.26`	`charles___`	`Cisco -> asterisk as a peer`
`17:40.36`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`17:40.50`	`charles___`	`but calls get from different TCP Ports and Asterisk calls the gateway unknown`
`17:41.06`	`charles___`	`is there any workaround ?`
`17:44.03`	`Dr-Linux\|work`	`charles___, cisco what?`
`17:44.42`	`Dr^Mouse`	`hmm... ok im gonna have to call it a day. i can beleive this. the goddam phone should do as its told!`
`17:44.49`	`Dr^Mouse`	`*cant, not can`
`17:45.42`	`Dr^Mouse`	`see you all later...`
`17:46.06`	`Dr^Mouse`	`GONE`
`17:46.41`	`j0`	`does anyone know how to transfer calls using the linksys pap2? i cant find any documentation on it`
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`17:50.36`	`rpm`	`anyone know if it is possible to hack a mitel 5215, its currently running MiNet firmware and i wanna put SIP firmware on it, but it is not one of their dual mode phones`
`17:50.46`	`rpm`	`MiNet is such a stupid protocol.`
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`17:55.39`	`Qwell`	`rpm: ask Juggie, he would know`
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`18:07.40`	`jhb`	`hi *. I am trying to get realtime running with unixodb->mysql. The unixodb setup works (I check the queries in the mysql.log), asterisk tell me "Binding sipusers to odbc/de/sipusers", but I never see queries from asterisk. How to proceed?`
`18:07.55`	`aydiosmio`	`I'm excited about these 480i phones`
`18:07.58`	`aydiosmio`	`they look like fun`
`18:10.49`	`*** join/#asterisk m4rkl4r (n=mark@c-71-203-182-234.hsd1.fl.comcast.net)`
`18:12.34`	`*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)`
`18:12.34`	`*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.11, Asterisk-addons 1.2.4, and Zaptel 1.2.8 released! (August 22, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx`
`18:12.36`	`m4rkl4r`	`but I am so far unable to get + or $ to work`
`18:12.41`	`*** join/#asterisk kpettit (n=keith@69.15.174.114)`
`18:13.09`	`*** join/#asterisk grabeez (n=gaving@grabes2.enter.net)`
`18:14.43`	`m4rkl4r`	`i tried the trick on suggests using: Set(dollar=$)`
`18:15.43`	`*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)`
`18:16.15`	`Dr-Linux\|work`	`m4rkl4r, what's docs for regex?`
`18:17.15`	`*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)`
`18:17.27`	`blitzrage`	`m4rkl4r: yah... you have to really place with it such as using \ to escape characters since the asterisk dialplan is also trying to parse those chars`
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`18:19.12`	`[TK]D-Fender`	`aydiosmio: 480i works, but I'd sooner suggest Polycoms.....`
`18:20.20`	`*** join/#asterisk elriah (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net)`
`18:20.51`	`elriah`	`Hi all. In my dialplan (* 1.2/extensions.conf), if I don't want to use queue but want to ring a bunch of phones, I just stack them as Dial commands, right?`
`18:21.34`	`aydiosmio`	`I'd prefer them as well`
`18:22.02`	`[TK]D-Fender`	`elriah: Dial(SIP/1&SIP/2&ZAP/1,30)`
`18:23.00`	`elriah`	`Hey, thanks!`
`18:23.07`	`m4rkl4r`	`Dr-Linux\|work, blitzrage: i'm using man 7 regex as well as the voip-info.org page for asterisk func regex`
`18:24.33`	`m4rkl4r`	`also, escape characters cause errors like the follwing: ast_yyerror: ast_yyerror(): syntax error`
`18:24.36`	`new2voip`	`where can I find required dependencies?`
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`18:26.09`	`m4rkl4r`	`i'm trying to uses regexes in the context of the REGEX function and the : operator`
`18:26.11`	`aydiosmio`	`new2voip: internet`
`18:26.13`	`Dr-Linux\|work`	`hhm...`
`18:26.38`	`Dr-Linux\|work`	`m4rkl4r, i'm also looking for regex func...`
`18:26.53`	`aydiosmio`	`you could easily add your own`
`18:27.15`	`Dr-Linux\|work`	`exten => _99XXXXXXXXXX,1,GotoIf($["${CALLERID(num)}" : "4..."]?5:2)`
`18:27.16`	`aydiosmio`	`compile in a .c or php/perl AGI`
`18:27.23`	`Dr-Linux\|work`	`any clude does it work?`
`18:28.10`	`Dr-Linux\|work`	`blitzrage, your thoughts?`
`18:29.08`	`blitzrage`	`Dr-Linux\|work, m4rkl4r: I remember doing this a while ago playing with them... and REGEX's were just a pain in the but... was only able to get the most simple ones working...`
`18:29.12`	`blitzrage`	`Qwell: little help?`
`18:29.17`	`Qwell`	`huh?`
`18:29.53`	`elriah`	`quit`
`18:30.00`	`m4rkl4r`	`well, the documentation rather seems incomplete`
`18:30.16`	`m4rkl4r`	`i also looked in doc/README.variables as suggested by the asterisk debug logs.`
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`18:30.35`	`blitzrage`	`Qwell: remember when we fought with Regex's a couple months back?`
`18:30.43`	`Dr-Linux\|work`	`blitzrage, well, regex works, but how .. mmmm`
`18:30.46`	`Qwell`	`I don't remember what I had for breakfast.`
`18:31.04`	`blitzrage`	`Qwell: lol... I do... but I don't remember what we did to get regex's working :)`
`18:31.08`	`Dr-Linux\|work`	`Qwell, do you remember your girlfriend?`
`18:31.21`	`Qwell`	`blitzrage: actually, I didn't have breakfast.`
`18:31.22`	`Qwell`	`:P`
`18:31.25`	`blitzrage`	`I don't think his wife would like him talking about his g/f :)`
`18:31.27`	`Qwell`	`yeah, I remember...umm`
`18:31.28`	`blitzrage`	`Qwell: lol`
`18:31.46`	`Dr-Linux\|work`	`blitzrage, i have 2 ways to do what i want`
`18:31.50`	`Qwell`	`we...did...`
`18:31.50`	`blitzrage`	`Qwell: I totally forget what we did... although I might have an example kicking around somewhere with what I was using`
`18:31.51`	`Qwell`	`something`
`18:31.58`	`blitzrage`	`Qwell: exactly! that was it!`
`18:31.59`	`Qwell`	`oh`
`18:32.01`	`Qwell`	`$[]?`
`18:32.06`	`blitzrage`	`maybe?`
`18:32.16`	`Qwell`	`I want to say it was`
`18:32.18`	`blitzrage`	`$[] is needed for any sort of evaluation...`
`18:32.24`	`Qwell`	`well, not needed, but yeah`
`18:32.26`	`blitzrage`	`let me check my script with the regex in it`
`18:32.29`	`Dr-Linux\|work`	`Qwell, what you say for >> GotoIf(${REGEX("4[0-9][0-9][0-9]","${CALLERID(number)}")?true:false)`
`18:32.35`	`blitzrage`	`not needed... but preferred :)`
`18:32.47`	`Qwell`	`Dr-Linux\|work: func REGEX uses a space, not a comma`
`18:32.56`	`Qwell`	`GotoIf(${REGEX("4[0-9][0-9][0-9]" "${CALLERID(number)}")?true:false)`
`18:32.57`	`Qwell`	`or something`
`18:33.10`	`Qwell`	`I've never actually used it, but the syntax is something like that. show function REGEX`
`18:33.43`	`Dr-Linux\|work`	`Qwell, i see`
`18:34.06`	`Dr-Linux\|work`	`Qwell, wht you say about this >> exten => _99XXXXXXXXXX,1,GotoIf($["${CALLERID(num)}" : "4..."]?5:2)`
`18:34.10`	`aydiosmio`	`REGEX("<regular expression>" <data>)`
`18:34.22`	`Qwell`	`Dr-Linux\|work: looks okay, but...`
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`18:34.34`	`Qwell`	`Dr-Linux\|work: my mind is elsewhere right now, sorry`
`18:35.07`	`Dr-Linux\|work`	`Qwell, .... << is just like pattern, like xxx everything`
`18:35.09`	`Dr-Linux\|work`	`oo ok`
`18:36.48`	`*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)`
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`18:37.54`	`clyrrad`	`Hi guys - I have made a script that needs to run each time asterisk is loaded - I have opened /etc/init.d/asterisk and added php /var/lib/asterisk/agi-bin/sync_queues.php to the bottom of the start function right before return $RETVAL - however when I stop and start asterisk the script does not run - can anyone point me in the right direction`
`18:38.32`	`kpettit`	`Anybody had any luck with Sphinx? I can't see any good examples on how to get it going, or weather it even works well`
`18:38.42`	`aydiosmio`	`init.d is what happens when the system starts, not asterisk`
`18:38.59`	`clyrrad`	`ah okay - so where "should" i be calling this script?`
`18:39.13`	`Dr-Linux\|work`	`kpettit, i don't think if anyone have luck to use sphinx3 with asterisk`
`18:39.37`	`kpettit`	`Say again?`
`18:40.13`	`Dr-Linux\|work`	`kpettit, sorry for my bad english, but i don't think sphinx3 works with asterisk`
`18:40.14`	`kpettit`	`I just got sphinx3 installed, and I'm trying to figure out what I can do with it`
`18:40.38`	`*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`18:40.38`	`kpettit`	`bummer. I have sphinx2 as well, but there are no good examples on how to integrate it`
`18:40.40`	`clyrrad`	`aydiosmio - Where should I be calling the script then? I need asterisk to be loaded before the script is executed....`
`18:40.41`	`Dr-Linux\|work`	`kpettit, you can't do anything, don't waste your time`
`18:40.57`	`kpettit`	`got ya. Just didn't know if it was something I could toy around with yet`
`18:41.14`	`*** part/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`18:41.15`	`aydiosmio`	`clyrrad: I don't know.`
`18:41.17`	`kpettit`	`I got tts working pretty good with Festival, was just tyring to get it going the other way now`
`18:41.30`	`Dr-Linux\|work`	`kpettit, please carryon, if you have done it, then please post it on my site`
`18:41.42`	`kpettit`	`the Festival stuff?`
`18:42.06`	`clyrrad`	`.. is /etc/init.d/asterisk not the script that is used for stop start and reload asterisk once the system is already running?`
`18:42.16`	`kpettit`	`I think the Festival stuff is only really usefull if your doing RTA. Updating the text is alot easier`
`18:42.21`	`Dr-Linux\|work`	`kpettit, never use fastival`
`18:42.23`	`*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`18:42.43`	`aydiosmio`	`clyrrad: no, it's not.`
`18:42.45`	`m4rkl4r`	`dr-linux, qwell: i'm trying to do a match on 10 digit phone numbers, using the $[] syntax, and even simple things like the following do not work:`
`18:42.54`	`kpettit`	`festival is text to speech. It sounds kind of cheesy but it's ok`
`18:43.00`	`clyrrad`	`... ok can someone please tell me what script is used?`
`18:43.07`	`kpettit`	`I like the cepertral stuff, I think that's the best sounding text to speech stuff.`
`18:43.10`	`clyrrad`	`or where its loacated... I have not been able to find it`
`18:43.10`	`m4rkl4r`	`execif $[ ${EXTEN} : [0-9][0-9][0-9] ]\|Dial\|SIP/19046268172@test-td1.starnetusa.net`
`18:43.19`	`m4rkl4r`	`(this is realtime db based)`
`18:43.35`	`m4rkl4r`	`um, that is suppose to match 3 numbers.`
`18:43.58`	`clyrrad`	`aydiosmio - the way I have it now would work only after a system reboot correct?`
`18:43.59`	`*** join/#asterisk Nix (n=Nix@81.213.125.220)`
`18:44.24`	`aydiosmio`	`clyrrad: right`
`18:44.31`	`aydiosmio`	`clyrrad: asterisk is just the binary`
`18:44.49`	`aydiosmio`	`when you do a reload it just reload the configuration fomr the binary`
`18:44.53`	`aydiosmio`	`there's no load script`
`18:44.56`	`clyrrad`	`makes sense - now I need to figure out how to make this script run after you do a stop now and asterisk -vvvvvvvvgc`
`18:45.22`	`aydiosmio`	`clyrrad: you could put it in a module, but it'd have to be compiled`
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`18:45.31`	`*** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)`
`18:45.40`	`clyrrad`	`make a module that calls the php script?`
`18:45.46`	`aydiosmio`	`yes`
`18:45.47`	`clyrrad`	`yea - but is there not an easier way?`
`18:45.54`	`aydiosmio`	`I don't know.`
`18:45.55`	`clyrrad`	`seems like extra overhead`
`18:46.00`	`aydiosmio`	`I doyes it does.`
`18:46.16`	`aydiosmio`	`yes it does.`
`18:46.16`	`sb_mx`	`clyrrad, or you could also make a script that does asterisk -rx "stop now" and then asterisk -vvvvvvgc and then run your script. something along those lines`
`18:46.46`	`clyrrad`	`how will that work when the system is already up unless I can find the script that is running asterisk`
`18:46.58`	`aydiosmio`	`there is no script`
`18:47.03`	`clyrrad`	`when you type asterisk -vvvvvvgc from the CML that is talking to the binary directly`
`18:47.11`	`clyrrad`	`is that correct?`
`18:47.21`	`aydiosmio`	`recompile asterisk to exec(php myscript.php) when it loads`
`18:47.29`	`aydiosmio`	`clyrrad: that's correct`
`18:47.39`	`clyrrad`	`so then the only way to do it is with a module then`
`18:47.46`	`clyrrad`	`becase asterisk auto loads the modules`
`18:47.46`	`aydiosmio`	`or recompile asterisk`
`18:48.05`	`*** join/#asterisk dasenjo_ (n=dasenjo@63.245.86.189)`
`18:48.08`	`clyrrad`	`If I recompile asterisk where would I add that exec line?`
`18:48.20`	`aydiosmio`	`I'm sure you can find out on your own`
`18:48.47`	`clyrrad`	`lol thanks`
`18:49.22`	`*** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net)`
`18:50.01`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`18:50.50`	`awannabe`	`is there a special crazy trick to be able to transfer a call to a cell phone and have the call bridge properly?`
`18:51.42`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`18:51.42`	`*** mode/#asterisk [+o mog] by ChanServ`
`18:51.49`	`*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)`
`18:51.55`	`kpettit`	`awannabe, you doing it through ami or just through the phone?`
`18:52.14`	`awannabe`	`was thru the phone`
`18:54.43`	`*** join/#asterisk Assid (i=assid@203.115.83.214)`
`18:54.45`	`kpettit`	`I'd watch it through the console, but I've never had a problem.`
`18:55.34`	`awannabe`	`i wonder if its cause im sending it back out a iax link`
`18:55.47`	`awannabe`	`actailly, my iax providers only allows one connection, that might be it! lol`
`18:56.29`	`kpettit`	`there ya go.`
`18:57.21`	`awannabe`	`thanks for kicking my brain into gear! heh`
`18:58.24`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.72)`
`18:59.30`	`aydiosmio`	`awe, that sucks`
`18:59.44`	`aydiosmio`	`my vonage softphone allows at least one out and one in`
`19:00.38`	`awannabe`	`yeah, i got a plan thru teliax that does only 1 line, its just usage based`
`19:01.49`	`*** join/#asterisk pnlarsson (n=niklas@83.248.0.248)`
`19:01.51`	`Luke-Jr`	`limited lines are so 1980s =p`
`19:02.26`	`Assid`	`tkd .. you there`
`19:02.39`	`Assid`	`anyone with a polycom?`
`19:02.42`	`Assid`	`i got a weird issue`
`19:02.48`	`awannabe`	`i got 2 in front of me`
`19:02.58`	`Assid`	`i click on dial.. on my placed call list.. and some of my calls dont go through.. i come back to the menu`
`19:03.03`	`Assid`	`this is on both 301 and 501s`
`19:03.05`	`awannabe`	`Luke-Jr, it is..i dont have my PRI in it yet :(`
`19:03.56`	`Assid`	`i dont even see it dial on the CLI`
`19:05.15`	`awannabe`	`what firmware?`
`19:06.08`	`Assid`	`me?`
`19:06.11`	`awannabe`	`yeah`
`19:06.12`	`*** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr)`
`19:06.15`	`jhiver`	`hi guys`
`19:06.28`	`awannabe`	`hi`
`19:06.58`	`jhiver`	`I wonder if you have run in this issue where you have two peers which support, say, g711 and g729, and despite that both do, asterisk decides to transcode anyway`
`19:07.02`	`Assid`	`bootrom 2.6.1.003`
`19:07.07`	`jhiver`	`from g729 to g711`
`19:07.08`	`Assid`	`one mroe 0 there`
`19:07.32`	`Assid`	`sip 1.6.6.0036`
`19:07.33`	`awannabe`	`Assid, what sip verr?`
`19:07.39`	`awannabe`	`hrmm, intresting`
`19:07.50`	`jhiver`	`i was wondering how you would solve this issue...`
`19:08.17`	`jhiver`	`i'm considering making a separate sip "peer" for each codec, i.e. peerA-g711, peerA-g729...`
`19:08.37`	`awannabe`	`jhiver, well it only transcodes if they are talking differnet codecs`
`19:08.44`	`jhiver`	`awannabe, nope`
`19:09.02`	`awannabe`	`it cant transcode if both are using same codec`
`19:09.09`	`jhiver`	`if I do disallow=all allow=g729 on peer B then it works fine`
`19:09.16`	`aydiosmio`	`I'm sure you have the codec preferences matched and are allowing all codecs on each peer`
`19:09.27`	`*** join/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8)`
`19:09.39`	`jhiver`	`codec preferences matched?`
`19:09.41`	`Scrye`	`can someone point me in the right direction to get a rhino dual port T1 card working in linux?`
`19:09.45`	`jhiver`	`what do you mean?`
`19:09.51`	`pnlarsson`	`A small Q: What version of SIP does * use?`
`19:10.03`	`Assid`	`awannabe: any clue?`
`19:10.13`	`Assid`	`hey polycom guru.. you there?`
`19:10.16`	`aydiosmio`	`jhiver: SIP clients usually allow you to sec a codec preference`
`19:10.22`	`jhiver`	`aydiosmio, ok`
`19:10.32`	`aydiosmio`	`set`
`19:10.51`	`*** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it)`
`19:11.08`	`jhiver`	`so you're saying one peer has, say, g729, g711 and the other one has g711, g729?`
`19:11.23`	`jhiver`	`and that's why asterisk starts transcoding?`
`19:11.29`	`awannabe`	`yeah`
`19:11.33`	`awannabe`	`they are different codecs`
`19:11.39`	`Assid`	`[TK]D-Fender: any clue on this issue? it only happens on placed calls..`
`19:11.40`	`awannabe`	`so it has to transcode, the phone cant do that`
`19:12.07`	`jhiver`	`awannabe, of course they can, both legs support g711 et g729`
`19:12.16`	`jhiver`	`and they are not phones, they are 'peers'`
`19:12.23`	`jhiver`	`that is, softswitches`
`19:12.25`	`awannabe`	`well u need to force the codes`
`19:12.29`	`awannabe`	`codecs*`
`19:12.58`	`jhiver`	`so that means i need to do something like disallow=all allow=g729 right?`
`19:13.30`	`Scrye`	`anyone have a url for configuring a data T1?`
`19:13.38`	`awannabe`	`yeah, that sould force it then`
`19:13.52`	`jhiver`	`well that sucks too, because if now I have a peer C that sends a call to peer A then I'm doing g729 -> g711 translation "simply" because i "forced" the codec...`
`19:13.54`	`awannabe`	`and then the softswitch should just use that codec, and wa la`
`19:14.14`	`awannabe`	`i think i see what your saying now`
`19:14.16`	`aydiosmio`	`jhiver: run ethereal and look at the codec negotiation`
`19:14.34`	`awannabe`	`you want * to know the proper codec and use it on all sides, gotcha`
`19:15.01`	`Assid`	`would a sip update take care of this problem?`
`19:15.13`	`jhiver`	`I want * to go, "haha, this is the intersection of the supported codecs by both legs, let's use one of them"`
`19:15.35`	`jhiver`	`only transcode if the intersection is empty, which would make some kind of sense then`
`19:16.02`	`[TK]D-Fender`	`Assid: Trying to redial previously successful calls?`
`19:16.06`	`jhiver`	`right now it's transcoding even though it shouldn't have to unless i "force" the codec`
`19:16.14`	`jhiver`	`kind of sucks doesn't it :)`
`19:16.23`	`[TK]D-Fender`	`Assid: What SIP ver you on now?`
`19:16.33`	`Assid`	`yes. but from the menu.. im just hitting the dial button`
`19:16.36`	`Assid`	`1.6.6`
`19:16.49`	`[TK]D-Fender`	`jhiver: Its a question about which side has codec PRIORITY.`
`19:17.17`	`jhiver`	`[TK]D-Fender, please explain`
`19:17.21`	`[TK]D-Fender`	`Assid: Well 1.6.7 did have a lot of little bug fixes....`
`19:17.36`	`Assid`	`is that the lates /greatest out there?`
`19:17.37`	`aydiosmio`	`indeed. priority.`
`19:18.02`	`[TK]D-Fender`	`jhiver: Lets say both sides CAN support a desired codec, it doesn't necessarily find a best fit if both are agressive on the choice and * is in the way.`
`19:18.29`	`[TK]D-Fender`	`jhiver: if you want to allow multiple options, make sure you put them in that order in your sip.conf "allows".`
`19:18.32`	`aydiosmio`	`your softswitches should have a priority setting jhiver`
`19:19.05`	`*** join/#asterisk rbordeaux (i=hidden-u@80.169.196.234)`
`19:19.09`	`jhiver`	`[TK]D-Fender, why shouldn't it? maybe it's my understanding of SIP that is a bit weak`
`19:19.24`	`jhiver`	`at any rate, i thought of a way of fixing this, i'd like your thoughts`
`19:19.54`	`aydiosmio`	`voipswitch has codec priority...`
`19:19.56`	`[TK]D-Fender`	`jhiver: I'm not a great expert in this dept, but unless you force a single codec per channel you have to be careful of your ordering. Play around with it a bit`
`19:20.02`	`jhiver`	`maybe have a [peerA-g729] [peerA-g711] [peerC-g723] sections, say`
`19:20.05`	`aydiosmio`	`it affects asterisk's selection`
`19:20.05`	`sivana`	`has anyone worked with queues?`
`19:20.11`	`jhiver`	`i think that would for for outgoing`
`19:20.14`	`aydiosmio`	`sivana: yes, it's a pain.`
`19:20.20`	`aydiosmio`	`and I quickly gave up`
`19:20.32`	`jhiver`	`now the issue is that all these peers would have essentially the same parameters (host=...)`
`19:20.34`	`sivana`	`aydiosmio: does the press any key to exit a queue actually work?`
`19:20.45`	`jhiver`	`so i wonder how asterisk would come with it for incoming calls?`
`19:20.46`	`aydiosmio`	`sivana: don't know`
`19:21.05`	`sb_mx`	`sivana, we have it working`
`19:21.10`	`aydiosmio`	`jhiver: run ethereal, watch the codec negotiations`
`19:21.13`	`sivana`	`sb_mx: with 1.2.10?`
`19:21.23`	`jhiver`	`say a call arrives from ip address 123.123.123.123, which happens to be in peerA-g711 peerA-723 peerA-g729`
`19:21.38`	`jhiver`	`which of these 'peer' will asterisk choose?`
`19:21.44`	`aydiosmio`	`jhiver: it doesn't have to be that complicated`
`19:21.50`	`sb_mx`	`sivana, 1.2.5 but i'm not sure any changes have been made regarding that suff (have yet to test tho)`
`19:22.02`	`sivana`	`sb_mx: what's the trick`
`19:22.07`	`sivana`	`I have a context specified`
`19:22.16`	`sivana`	`but nothing happens`
`19:22.20`	`sb_mx`	`sivana, in queues.conf ?`
`19:22.22`	`sivana`	`yes`
`19:22.28`	`*** join/#asterisk zoa (n=d@pirus.securax.be)`
`19:22.42`	`*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)`
`19:22.46`	`Aurs`	`anyone know what this is? Aug 29 14:50:44 DEBUG[12267] channel.c: Got a FRAME_CONTROL (5) frame on channel Zap/3-1`
`19:22.58`	`sivana`	`sb_mx: and in the context, I have exten => s,1,blah`
`19:23.09`	`zoa`	`hey ho`
`19:23.21`	`[TK]D-Fender`	`sivana: Yes, just fine`
`19:23.27`	`sb_mx`	`sivana, the way we have it working is we announce the caller that they can press 1 to leave a message. on that context we have exten => 1,1,Voicemail(bleh)`
`19:23.30`	`*** part/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8)`
`19:23.38`	`sivana`	`ah`
`19:23.41`	`[TK]D-Fender`	`sivana: What ver of *?`
`19:23.44`	`sivana`	`let me try that`
`19:23.46`	`sivana`	`1.2.10`
`19:24.00`	`*** join/#asterisk supjigatr (n=syslod@152.53.16.10)`
`19:24.08`	`[TK]D-Fender`	`sivana: If your next test fails, pastebin everything.`
`19:24.17`	`sivana`	`[TK]D-Fender: ok`
`19:24.23`	`supjigatr`	`Anyone have a working sample of voicemailmain with no password.`
`19:24.34`	`bkw_`	`prepend an 's'`
`19:24.39`	`bkw_`	`ie s100@context`
`19:25.01`	`supjigatr`	`bkw_: I tried that but the application help says to do as a \| option`
`19:25.11`	`sivana`	`sb_mx: thank you :)`
`19:25.18`	`sivana`	`apparently it has to match on the key pressed`
`19:25.19`	`*** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)`
`19:25.56`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`19:26.42`	`[TK]D-Fender`	`sivana: Just like it says to ;)`
`19:27.41`	`bkw_`	`supjigatr, chances are they broke it :P`
`19:27.43`	`bkw_`	`or changed it`
`19:27.52`	`supjigatr`	`Yea thats what I'm thinking.`
`19:28.05`	`bkw_`	`they are really efficient at breaking things sometimes :P`
`19:28.15`	`supjigatr`	`and changing for no reason.`
`19:28.30`	`bkw_`	`but i'm gonna shut up`
`19:28.38`	`bkw_`	`because having an opinion around here gets you called a troll`
`19:28.43`	`sivana`	`[TK]D-Fender: where?`
`19:28.52`	`sivana`	`context = queue-out ; Here we go when the caller presses a single digit, while in the queue`
`19:29.02`	`supjigatr`	`:)`
`19:29.06`	`bkw_`	`supjigatr, I'm doing VoicemailMain(s1000)`
`19:29.07`	`sivana`	`doesn't say you can't use 's' :)`
`19:29.07`	`bkw_`	`and it works`
`19:29.09`	`bkw_`	`without a password`
`19:29.20`	`supjigatr`	`I tried on an older box and it works.`
`19:29.26`	`bkw_`	`are you using 1.4?`
`19:29.28`	`bkw_`	`or what?`
`19:29.28`	`supjigatr`	`But with a current * release not so much.`
`19:29.39`	`bkw_`	`oh chances are its all messed up`
`19:29.39`	`sb_mx`	`sivana, well it does say "single" digit`
`19:29.45`	`supjigatr`	`1.2.10`
`19:29.45`	`sivana`	`:)`
`19:30.08`	`sivana`	`I was hoping to match any digit`
`19:30.24`	`sb_mx`	`exten => [0-9],1,bleh ?`
`19:30.45`	`Zodiacal`	`anyone know why my gotoif doesn't work when i try to check this variable? ${CALLERID(num)}) when i verbose that variable i get the correct ext. shown, but when i try to use gotoif() it never fires true. heres my gotoif statment: exten => 555,1,GotoIf($["${CALLERID(num)})" = 25]?9:11)`
`19:30.45`	`*** join/#asterisk ToyMan (n=stuq@pool-70-19-236-242.bos.east.verizon.net)`
`19:30.53`	`Zodiacal`	`it allways goes to 11`
`19:30.55`	`Zodiacal`	`never 9`
`19:30.57`	`sivana`	`trying that now`
`19:31.28`	`sivana`	`sb_mx: nope`
`19:31.36`	`Zodiacal`	`do i have to change that callerid variable into a integer or somthing?`
`19:32.13`	`sivana`	`sb_mx: _X`
`19:32.16`	`sivana`	`that works`
`19:32.54`	`sb_mx`	`sivana, i thought [0-9] would work. oh well`
`19:33.19`	`sivana`	`no worries.. thanks for putting me on the path :)`
`19:33.24`	`[TK]D-Fender`	`Zodiacal: 1 bracket too many`
`19:33.51`	`[TK]D-Fender`	`Zodiacal: exten => 555,1,GotoIf($["${CALLERID(num)})" = 25]?9:11) <- bad`
`19:34.11`	`[TK]D-Fender`	`Zodiacal: exten => 555,1,GotoIf($["${CALLERID(num)}"="25"]?9:11) <- good`
`19:34.21`	`Zodiacal`	`i'll try right now :)`
`19:35.19`	`Zodiacal`	`tkd-fender, works! Thank You!`
`19:35.36`	`Zodiacal`	`i copied and pasted from the variables wiki and it had an ).. grr..`
`19:35.49`	`Zodiacal`	`my fault tho`
`19:36.43`	`[TK]D-Fender`	`Zodiacal: And didn't put the other end of the equation in quotes to match either...`
`19:37.58`	`hmmhesays`	`hola`
`19:38.35`	`*** join/#asterisk AJaymn (n=boiwonde@71-36-133-35.mpls.qwest.net)`
`19:39.45`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`19:40.30`	`Zodiacal`	`is there a function to make the phone beep like no ext was found?`
`19:41.01`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`19:41.06`	`hmmhesays`	`yes in case ${CALLERID(num)} can be a string`
`19:42.54`	`Zodiacal`	`huh?`
`19:43.23`	`Zodiacal`	`i.e. StartRepeatingBeeping()`
`19:43.26`	`hmmhesays`	`thats why you need to the quotes`
`19:43.34`	`Zodiacal`	`im trying to make it beep`
`19:43.35`	`hmmhesays`	`in your gotoif statement`
`19:44.13`	`*** join/#asterisk DFiber (n=bwarner@65.172.101.235)`
`19:44.14`	`Zodiacal`	`yeah the statment works great with the quotes and by removing the ) like fender said.. im tryin to do somthin new..`
`19:44.36`	`Zodiacal`	`i guess i can just send it to a non existent ext..`
`19:47.11`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`19:47.29`	`*** join/#asterisk Stp1800 (n=Stp1800@atlsfl-bundle-69-167-93-164.atlsfl.adelphia.net)`
`19:47.35`	`Stp1800`	`Hi guys.`
`19:47.37`	`jbalcomb`	Should I be able to do something like `grep 4??? pri_in.conf` to get all my four digit extensions that start with 4?
`19:48.14`	`Stp1800`	`Does anyone have a Grandstream gxp2000 they can help me reset?`
`19:50.09`	`jbalcomb`	`Stp1800: instructions for that operation are on the voip-wiki`
`19:51.02`	`*** join/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net)`
`19:51.28`	`Stp1800`	`Ah, I see.`
`19:51.37`	`hmmhesays`	`Zodiacal what are you trying to do?`
`19:51.43`	`Stp1800`	`Anyone have one of these units can walk me through what buttons to hit on the device?`
`19:52.02`	`Stp1800`	`I knew the keypad and menu puttongs, just can't see the screen to do the reset.`
`19:52.10`	`Zodiacal`	`hmmhesays deny a phone from calling international numbers`
`19:52.16`	`Stp1800`	`I have the mac of the phone memorized.`
`19:52.30`	`Zodiacal`	`hmmhessays i should really play a message more specific to the error`
`19:52.36`	`*** join/#asterisk rbordeaux (i=hidden-u@80.169.196.234)`
`19:52.43`	`Zodiacal`	`i'll do that insted`
`19:52.46`	`Zodiacal`	`thanks tho!`
`19:52.49`	`charles___`	`Hey guys sorry for the delay, was fighting against the Cisco here`
`19:52.53`	`*** part/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net)`
`19:52.53`	`charles___`	`it still doesn't work`
`19:53.06`	`charles___`	`cisco 5400 -> Asterisk`
`19:53.15`	`charles___`	`trying to terminate Cisco calls onto the asterisk`
`19:53.37`	`*** join/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net)`
`19:53.43`	`jbalcomb`	`Stp1800: ah, round button; down, down, down, down, down, down, down; round button;`
`19:54.02`	`*** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com)`
`19:54.13`	`hmmhesays`	`Zodiacal: you could`
`19:54.25`	`*** part/#asterisk new2voip (n=chatzill@user-24-214-152-5.knology.net)`
`19:54.35`	`jbalcomb`	`Stp1800: oh, do you want factory reset?`
`19:55.02`	`*** part/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net)`
`19:55.19`	`*** join/#asterisk Assid (i=assid@203.115.83.215)`
`19:56.26`	`jbalcomb`	`Stp1800: round button; down, down, down, down, down; round button; down, down, down, down; round button; up; <enter MAC>;`
`19:56.31`	`*** join/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net)`
`19:56.56`	`jbalcomb`	`Stp1800: after mac; round button; round button;`
`19:57.14`	`AJaymn`	`Anyone know of a Termination Provider that offers flat rate unlimited US calling?`
`19:58.03`	`hmmhesays`	`pakcet 8 I think does`
`19:58.33`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`19:58.34`	`AJaymn`	`need reseller availability.. :P`
`19:59.13`	`aydiosmio`	`...`
`19:59.24`	`aydiosmio`	`you're going to resell unlimited calling?`
`19:59.34`	`hmmhesays`	`good luck with that`
`20:00.44`	`MikeJ`	`I know :P`
`20:01.10`	`Assid`	`120$`
`20:03.05`	`aydiosmio`	`$25/mo though vonage:)`
`20:03.08`	`aydiosmio`	`through`
`20:03.12`	`Stp1800`	`Ok, when I go to enter the mac I know the notation they described.`
`20:03.15`	`docelmo`	`I just bought 2 of them..`
`20:03.22`	`aydiosmio`	`but I'll give it you YOU for the LOW LOW price of your SOUL`
`20:03.24`	`Stp1800`	`How do I seporate the numbers / letters?`
`20:03.41`	`justinu\|laptop`	`Aug 28 20:35:39 DEBUG[30913] chan_sip.c: Failed to grab lock, trying again...`
`20:03.42`	`justinu\|laptop`	`Aug 28 20:35:39 DEBUG[30913] chan_sip.c: Failed to grab lock, trying again...`
`20:03.44`	`aydiosmio`	`depends on the system`
`20:03.44`	`justinu\|laptop`	`ad infinum`
`20:03.49`	`aydiosmio`	`usually colons :`
`20:03.53`	`bkw_`	`justinu\|laptop, punt`
`20:03.53`	`*** join/#asterisk spr1te (i=spr1te@194.187.130.227)`
`20:03.54`	`aydiosmio`	`sometines dots .`
`20:03.59`	`aydiosmio`	`sometimes dahses -`
`20:04.05`	`justinu\|laptop`	`bkw_: heh`
`20:04.45`	`bkw_`	`guess it doesn't have any logic in there to give up after it knows it can't get the lock`
`20:05.10`	`hmmhesays`	`and now the times are changing, look at everything thats come and gone`
`20:05.27`	`hmmhesays`	`bkw_ find me a new warehouse for my band to practice in`
`20:05.29`	`AJaymn`	`aydiosmio VONAGE SUCKS! :P`
`20:05.43`	`bkw_`	`no they don't.. they BLOW`
`20:06.05`	`hmmhesays`	`they're alright if you managed to get one of their old business plus test accounts`
`20:06.06`	`AJaymn`	`besides its against there AUP :P`
`20:06.25`	`justinu\|laptop`	`bkw_: nope, and it takes down the entire sip stack`
`20:06.26`	`*** part/#asterisk m4rkl4r (n=mark@c-71-203-182-234.hsd1.fl.comcast.net)`
`20:06.30`	`*** join/#asterisk flujan (n=flujan@internet.nube.com.br)`
`20:06.32`	`justinu\|laptop`	`so all call processing stops`
`20:06.38`	`bkw_`	`justinu\|laptop, what are you doing to the poor thing?`
`20:06.48`	`flujan`	`hi guys... I am trying to understand how queues work.`
`20:06.53`	`justinu\|laptop`	`not sure why it all of a sudden starting doing this`
`20:07.01`	`justinu\|laptop`	`someone mentioned they changed out an L2 switch`
`20:07.09`	`bkw_`	`doubt that would cause this`
`20:07.14`	`flujan`	`for instance, when I receive a call from my customer how do I send my constumer to a specific queue?`
`20:07.26`	`justinu\|laptop`	`yeah, so I enabled a bunch of debuging to try and figure out what packet it receives fubars it`
`20:07.35`	`flujan`	`and how do I define which extension will answer to this client?`
`20:07.54`	`hmmhesays`	`so uclibc buildroot is kicking my ass, i can't figure out where to turn on softFPU`
`20:09.04`	`jbalcomb`	`Stp1800: what do you mean? the : inbetween the octets?`
`20:09.33`	`aydiosmio`	`AJaymn: what is against their AUP?`
`20:09.40`	`aydiosmio`	`reselling? of course.`
`20:09.42`	`jbalcomb`	`Stp1800: ah, septets actually i guess`
`20:09.43`	`aydiosmio`	`and Vonage is fine`
`20:09.44`	`Stp1800`	`Yes.`
`20:10.00`	`Stp1800`	`Lets say the first two chars of the mac are 00`
`20:10.04`	`aydiosmio`	`ff:ff ff-ff ff.ff`
`20:10.07`	`aydiosmio`	`take your pick`
`20:10.08`	`ki2k`	`anyone getting "chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 2 (read/write = 64/64)"?`
`20:10.12`	`clyrrad`	`This is my first time compiling a custom asterisk module based off the app_skel.c - can someone please tell me the proper way to compile the module - or link me somewhere that I can read up on it...`
`20:10.33`	`jbalcomb`	`Stp1800: well, there is no seperation between alpha-numeric and that entry doesn't expect the :'s`
`20:10.49`	`Stp1800`	`Next two are`
`20:10.52`	`[TK]D-Fender`	`flujan: You choose which queue when you call the Queue application from extensions.conf. As to which extension will answer, well the point of queues is to give a NUMBER of people who CAN answer the call. If you only want to ring one person then Queue's are not for you...`
`20:10.56`	`Assid`	`ki2k: your native is gsm.. your trying slin`
`20:10.59`	`jbalcomb`	`aydiosmio: I choose :`
`20:11.01`	`Stp1800`	`Ah. so the mac of my phone is:`
`20:11.04`	`Assid`	`make both streams set to gsm`
`20:11.12`	`justinu\|laptop`	`<PROTECTED>`
`20:11.13`	`justinu\|laptop`	`<PROTECTED>`
`20:11.26`	`justinu\|laptop`	`so i guess the channel associated with that sip message is deadlocked`
`20:11.30`	`hmmhesays`	`any lawyers in the room?`
`20:11.31`	`jbalcomb`	`Stp1800: enter 000B8205ADD8 for instance`
`20:11.35`	`justinu\|laptop`	`too bad that takes down the entire sip stack`
`20:11.57`	`Assid`	`hmmhesays: and what would lawyers do with asterisk?`
`20:12.04`	`hmmhesays`	`there is all kinds in here usually, and I need some legal advice`
`20:12.15`	`Assid`	`dont do drugs`
`20:12.15`	`Qwell`	`hmmhesays: What you'll find here, are mock lawyers`
`20:12.30`	`flujan`	`[TK]D-Fender, i want to ring a group of attendants. I want to know which attendant is idle and them, send the call to he/she.`
`20:12.30`	`Assid`	`he didnt like my advice`
`20:12.39`	`jbalcomb`	`guess he need that advice pretty quick ;)`
`20:12.45`	`flujan`	`[TK]D-Fender, i see about agents... but I want to use realtime queues...`
`20:12.56`	`flujan`	`[TK]D-Fender, how do I set up realtime agents?`
`20:13.04`	`Stp1800`	`My mac is 00:0B:82:09:CC:F6`
`20:13.07`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`20:13.23`	`eKo1`	`is there a command that will play an audio file for a specified period of time?`
`20:13.30`	`jbalcomb`	`Stp1800: ok, so enter 000B8209CCF6`
`20:13.35`	`hmmhesays`	`so yeah`
`20:13.39`	`Qwell`	`hmmhesays: What you'll find here, are mock lawyers`
`20:13.54`	`Qwell`	`ie; "I'm not a lawyer, but I play one on IRC"`
`20:14.05`	`jbalcomb`	`s/IRC/TV/`
`20:14.13`	`hmmhesays`	`yeah`
`20:14.16`	`flujan`	`Realtime queue members is the same as agents from agents.conf?`
`20:14.18`	`hmmhesays`	`that sucks Qwell`
`20:14.19`	`*** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18)`
`20:14.20`	`Qwell`	`flujan: no`
`20:14.37`	`hmmhesays`	`I have an employer threatening legal action against me if I accept payment from one of their customers`
`20:14.47`	`Qwell`	`hmmhesays: call a real lawyer. :)`
`20:14.49`	`eKo1`	`ah, I see waitmusiconhold`
`20:15.10`	`_alex_mx_`	`anyone familiar enough with voicemail code to answer a quick question?`
`20:15.20`	`hmmhesays`	`problem is they screwed me out of 8.5k usd when I left`
`20:15.29`	`E-bola`	`_alex_mx_: impossible to answer`
`20:15.29`	`hmmhesays`	`so I have no money`
`20:15.34`	`[TK]D-Fender`	`_alex_mx_: Ask your question and we'll see how much "enough" takes :)`
`20:15.47`	`clyrrad`	`can anyone help me to compile an asterisk module?`
`20:15.52`	`jbalcomb`	`hmmhesays: yeah, that is pretty tricky`
`20:16.04`	`hmmhesays`	`i never signed a non-compete agreement`
`20:16.06`	`hmmhesays`	`of any sort`
`20:16.25`	`jbalcomb`	`hmmhesays: oh, well then what ground do they state for having to sue you?`
`20:16.33`	`flujan`	`Qwell, I imagine something like this: 1 - I receive a call, put int in the queue using the Queue command.`
`20:16.34`	`[TK]D-Fender`	`hmmhesays: Then threaten them with incessant hail! Equal threat!`
`20:17.02`	`hmmhesays`	`"fraudulant business practice" which I didn't understand at all`
`20:17.06`	`flujan`	`Qwell, then, will have queue members who will be associated with a specific queue.`
`20:17.32`	`*** part/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com)`
`20:17.33`	`_alex_mx_`	`hehe,ok when enabling operator escape from voicemail i get a pls hold while i try that extension recording...language is set to spanish, ....asterisk/sounds/es has a pls-hold-while-try.gsm and i've soft linked the english version to that file just for good measure...still getting it in english`
`20:17.37`	`clyrrad`	`has no one made any modules for asterisk?`
`20:17.45`	`*** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com)`
`20:17.46`	`flujan`	`Qwell, if I have a queue member and a member in my queue the queue dial and connect the caller to my queue member. Is it right?`
`20:18.07`	`jbalcomb`	`hmmhesays: the most likely implication with that charge is that you are scamming the client`
`20:18.44`	`_alex_mx_`	`any clue as to wich file it is using?`
`20:18.50`	`_alex_mx_`	`which`
`20:19.11`	`_alex_mx_`	`nothing on the console to indicate which file`
`20:19.22`	`jbalcomb`	`hmmhesays: is the customer paying you for work you did while employed by your former employer?`
`20:19.58`	`hmmhesays`	`jbalcomb: he was going to pay me for work I did after hours (i'm paid hourly) by my former employer`
`20:20.42`	`flujan`	`Qwell, is my thought right?`
`20:20.47`	`clyrrad`	`I have an C file like app_skel.c - and I just need to know where / how to compile it can someone please hlep me out?`
`20:20.52`	`jbalcomb`	`hmmhesays: off the clock sure but you were formally employed`
`20:20.54`	`*** part/#asterisk Cubber (n=Cubber@static-74-41-185-190.br1.glv.ny.frontiernet.net)`
`20:20.58`	`*** join/#asterisk RickNZ (n=rick@cust-202-89-145-162.quicksilver.net.nz)`
`20:21.36`	`jbalcomb`	`hmmhesays: that being the situation they do have grounds`
`20:21.54`	`Assid`	`if you were employed during the time you first worked with the client.. along witht he initial implementation of anything`
`20:22.04`	`jbalcomb`	`hmmhesays: non-competes are primarily regarding post-employment practices`
`20:22.05`	`hmmhesays`	`ok... jbalcomb, the work I did, I was never paid for, i merely sent out a service estimate. now I have had my former employer invoice for the work I ddi`
`20:22.11`	`Assid`	`then there is a good chance they can win`
`20:22.13`	`hmmhesays`	`*did`
`20:22.26`	`*** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`20:22.40`	`Assid`	`technically.. they could win`
`20:22.55`	`aydiosmio`	`here's the trick`
`20:22.58`	`Assid`	`since you were hired by them to work at clients location during business hours`
`20:23.01`	`Qwell`	`at which point, you countersue for overtime. :)`
`20:23.07`	`jbalcomb`	`hmmhesays: sounds like the thing to do is sue your former employer for lack of payment for the hours they are billing the client`
`20:23.09`	`aydiosmio`	`DON'T TELL YOUR EMPLOYER YOU'RE MOONLIGHTING ON THEIR DOLLAR`
`20:23.14`	`hmmhesays`	`so they got paid for the work I did, but are also threatening to sue if I work with this person at all in the future`
`20:23.22`	`Assid`	`as for the overtime..`
`20:23.25`	`aydiosmio`	`heh`
`20:23.40`	`hmmhesays`	`i don't care about the work I did, i'm worried about any future work, now that I am no longer employed`
`20:23.43`	`Assid`	`you tell the employer to pay you for the overtime.. and thereby he can charge to the client whatever they want`
`20:23.55`	`jbalcomb`	`hmmhesays: i doubt they have ground to sue for new work`
`20:24.04`	`Assid`	`new work.. not an issue`
`20:24.11`	`hmmhesays`	`like I said, I don't care about the them taking my money for work I did while I was employed, even if it was after hours`
`20:24.23`	`Assid`	`if you play your cards right.. you can walk out with it`
`20:24.40`	`jbalcomb`	`why does your former employer know about new work?`
`20:24.59`	`clyrrad`	`I have an C file like app_skel.c - and I just need to know where / how to compile it can someone please hlep me out? Or please link me to a document that shows how to compile a custom module for asterisk.....`
`20:25.00`	`Assid`	`just tel your employer A> Were you on the clock during the work done? YES/NO`
`20:25.24`	`Assid`	`If yes Pay for the overtime. If No, He cant charge for the overtime and therefore the work is done by you personally`
`20:25.41`	`jbalcomb`	`hmmhesays: final advice, pay a lawyer $150 to get an official answer.`
`20:25.47`	`Assid`	`yeah`
`20:26.27`	`jhiver`	`okay guys, u were right, reordering my codecs made the stuff behave correctly`
`20:26.34`	`jhiver`	`cool :)`
`20:27.24`	`hmmhesays`	`i don't care about the previous money... I care about them suing me now that i'm unemployed if I finish this guys project`
`20:28.07`	`smackus`	`has anyone else here had a problem with voicemail where every couple of voicemail left are a 20 second long busy signal? what causes this?`
`20:28.12`	`flujan`	`Qwell, am I right about the queue dynamics in realtime?`
`20:28.14`	`Assid`	`if i were you.. id call a lawyer`
`20:28.56`	`Assid`	`technically i doubt their case can stick.. since the only thing they have on their side is that you were working for them durig the time you worked extra hours`
`20:29.20`	`grabeez`	`Get a lawyer: Pay them now, or pay them a lot more later`
`20:29.42`	`Assid`	`however, because of the fact you were not paid for those hours.. you could technically turn it around and counter sue for being paid less than what was promised to you`
`20:29.54`	`[TK]D-Fender`	`jhiver: Quite welcome`
`20:30.12`	`Assid`	`wow.. updating sip version on 4.5K/sec connection is painful`
`20:30.21`	`Assid`	`stupid isp decides to do their maintainance :(`
`20:30.29`	`[TK]D-Fender`	`Assid: Why would you do something ridiculous like that?`
`20:30.33`	`hmmhesays`	`Assid: again I am not worried about that money at all, only being sued for working with this project in the future ( i am no longer employed with them)`
`20:31.03`	`[TK]D-Fender`	`hmmhesays: Without an NDA / anti-competitive clause they are blowing hot air...`
`20:31.24`	`Assid`	`well.. if you didnt sign any non-compete or NDA.. doubt they can do anything`
`20:31.49`	`Assid`	`people leave their jobs and start businesses all the time.. and many a times.. using the old clients of their jobs`
`20:31.54`	`hmmhesays`	`Assid, [TK]D-Fender, nothing of the sort`
`20:32.06`	`hmmhesays`	`[TK]D-Fender you know the worst part of this whole thing?`
`20:32.12`	`hmmhesays`	`I lost the warehouse the band practices in`
`20:33.04`	`eKo1`	`man, i which the background command would take a timeout option...`
`20:33.15`	`*** part/#asterisk Seyr (n=Seyr@cpe-67-10-139-141.houston.res.rr.com)`
`20:33.24`	`hmmhesays`	`set response timeout`
`20:33.37`	`Stp1800`	`Jbalcomb: Wrote you a msg in the chat window.`
`20:33.41`	`aydiosmio`	`set reply hazy`
`20:34.10`	`eKo1`	`hmmhesays: i do but after calling background, background never quits`
`20:36.30`	`*** join/#asterisk McLazarus (n=mcallist@pool-72-78-41-24.phlapa.east.verizon.net)`
`20:37.24`	`*** join/#asterisk seebq (n=cbq@207.59.210.210)`
`20:37.56`	`smackus`	`if disconnect supervision is not on my T1, is there something I can do within asterisk to do the same thing?`
`20:38.41`	`*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`20:38.59`	`seebq`	`hi all - am loving my new trixbox 1.1 setup, it is fantastic`
`20:39.34`	`[TK]D-Fender`	`seebq: Congrats. When that changes, you know where to ask....`
`20:39.52`	`[TK]D-Fender`	`smackus: Ditch that signalling and go PRI for crying out loud...`
`20:39.57`	`seebq`	`I do have a question: ordered a DID an trunk from telesip and am trying to do follow me finding but I hear no audio when it connects to an external line`
`20:40.24`	`seebq`	`no audio makes me think it's a firewall issue but it's not`
`20:40.27`	`[TK]D-Fender`	`seebq: Check your NAT settings.`
`20:40.32`	`seebq`	`routing out to a cell phone works`
`20:40.46`	`seebq`	`it's just under the follow me find that it just doesn't transfer over`
`20:41.01`	`*** join/#asterisk jontow (i=jontow@hijacked.us)`
`20:41.25`	`jontow`	`anyone run asterisk as non-root in freebsd (and is able to use ztdummy as a timing device?)`
`20:41.30`	`[TK]D-Fender`	`seebq: What does "routing out to a cell phone" mean?`
`20:41.37`	`eKo1`	`background with a timeout option or waitexten that plays a sound file is what I need`
`20:42.31`	`Assid`	`err.. anyone know where you can set the default app for sip: calls ?`
`20:42.36`	`Assid`	`in windows?`
`20:42.53`	`hmmhesays`	`eKo1: hmmm waitexten should work for you`
`20:42.54`	`seebq`	`so if I setup a Misc Extension of my cell phone # and have the autoattendent just dial that, calls in get transferred to my cell fine`
`20:42.59`	`flujan`	`guys, is it possible to remove a caller from a queue and connect it to a extension without using agents?`
`20:43.11`	`jontow`	`flujan; look at Redirect AMI app`
`20:43.28`	`[TK]D-Fender`	`seebq: Dial it HOW?`
`20:43.34`	`eKo1`	`hmmhesays: yeah but I need background music`
`20:44.37`	`hmmhesays`	`so you want backround to play a song while ... what?L`
`20:45.02`	`clyrrad`	`TKD- do you know how to compile an asterisk module?`
`20:45.25`	`seebq`	`[TK[D-Fender: trying to figure this out -- it's a ringall strategy group with my number listed as: 404xxxxxx# and it actually rings, but the person on the other line cannot hear me, i can hear them`
`20:45.37`	`aydiosmio`	`clyrrad: it's easier to recompile asterisk then to create a module.`
`20:45.41`	`*** join/#asterisk tdonahue-laptop (n=tdonahue@vonmail.vonworldwide.com)`
`20:45.55`	`*** part/#asterisk warmbowski (n=warmbows@dsl231-055-233.sea1.dsl.speakeasy.net)`
`20:46.06`	`clyrrad`	`aydiosmio - its a custom module I have made....`
`20:46.20`	`seebq`	`i wonder if it's because I don't have enough "trunks" -- i'm using a provider (telasip) that's unlimited, so max connections = 1`
`20:46.20`	`eKo1`	`clyrrad: I do. What do you need?`
`20:46.23`	`clyrrad`	`I just need to compile it like you would compile app_skel.c ....`
`20:46.42`	`tdonahue-laptop`	`hi all`
`20:46.45`	`eKo1`	`clyrrad: do you know how to compile dynamic modules?`
`20:46.45`	`clyrrad`	`I just need to know how / where to compile it so i get a .so that i can put in the modules directory`
`20:47.01`	`clyrrad`	`eK01 ... no thats what I am tryign to find out how to do`
`20:47.18`	`clyrrad`	`I have been searching all over to no avail`
`20:47.22`	`eKo1`	`clyrrad: google for it. there is a howto about dlls`
`20:47.22`	`tdonahue-laptop`	`anyone have any idea why even though my sip account has "dtmfmode=rfc2833`
`20:47.22`	`tdonahue-laptop`	`" set asterisk isn't negotiating rfc2833 for that peer in the SDP?`
`20:47.25`	`*** join/#asterisk saftsack (n=saftsack@p54A7D32C.dip.t-dialin.net)`
`20:47.31`	`saftsack`	`hi`
`20:47.38`	`flujan`	`jontow, Redirect AMI app? Is it from asterisk?`
`20:47.41`	`saftsack`	`has somebody here a patton gateway?`
`20:47.59`	`jontow`	`yes`
`20:48.03`	`clyrrad`	`eKo1 - do you know how to do it?`
`20:48.09`	`gmfm`	`how does one reload the manager config file without restarting?`
`20:48.11`	`jontow`	`http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect`
`20:48.18`	`eKo1`	`clyrrad: yes, I make custom modules all the time`
`20:48.28`	`clyrrad`	`could you please enlighten me :)`
`20:48.33`	`jontow`	`gmfm:`
`20:48.34`	`jontow`	`*CLI> reload manager`
`20:48.34`	`jontow`	`<PROTECTED>`
`20:48.48`	`jbalcomb`	`Is /usr/local/sbin the correct location for a cronjob script to bzip2 my log files?`
`20:48.48`	`*** join/#asterisk livesNbox (n=chadkous@165.236.120.14)`
`20:48.49`	`livesNbox`	`hey guys -- I'm dealing with Dynamic Members for my queues and trying to figure out a way to auto logoff members after they don't answer a call for so many seconds.. apparently the setting in agents.conf doesn't get applied to dynamic members?`
`20:48.57`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`20:49.03`	`gmfm`	`jontow: thanks much`
`20:49.05`	`jontow`	`np`
`20:49.07`	`livesNbox`	`I also tried passing option "n" with the queue call...`
`20:49.16`	`eKo1`	`clyrrad: http://www.tldp.org/HOWTO/Program-Library-HOWTO/shared-libraries.html`
`20:49.42`	`hmmhesays`	`god what a farking mess`
`20:50.23`	`clyrrad`	`eKo1 - wow thats lots of reading`
`20:50.29`	`clyrrad`	`guess its not a simple gcc myapp.c`
`20:50.40`	`eKo1`	`clyrrad: yes, but it will be worth it`
`20:50.46`	`[TK]D-Fender`	`livesNbox: Use a local channel to dial your agent for less time than the queue distributes for. if that times out use the dialplan tu unlog the agent.`
`20:50.47`	`Qwell`	`Why not just add it to the asterisk Makefile?`
`20:51.07`	`clyrrad`	`becase I didnt know that i could :)`
`20:51.23`	`clyrrad`	`and also its better to not have to recompile asterisk just for this one module....`
`20:51.35`	`clyrrad`	`would be nice to have a way to just compile it... and get the .so`
`20:51.38`	`Qwell`	`You don't have to recompile asterisk. It only recompiles what has changed`
`20:52.00`	`clyrrad`	`so I can just run make again and it will only do my module....?`
`20:52.07`	`Qwell`	`pretty much`
`20:52.12`	`livesNbox`	`[TK]D-Fender: Not quite sure I understand.. I don't want to make 2 simulatenous calls to the agent...`
`20:52.17`	`clyrrad`	`neat.... I didnt know that`
`20:52.23`	`eKo1`	`clyrrad: you should read about make too`
`20:52.25`	`clyrrad`	`I just put it in the sources with all the other modules then`
`20:52.36`	`flujan`	`jontow, does not exist a command in the dialplan?`
`20:52.43`	`[TK]D-Fender`	`livesNbox: Should be equally protected against that....`
`20:53.05`	`flujan`	`jontow, for what i see I must always use the admin interface to do this thing....`
`20:53.37`	`flujan`	`i just want to remove a caller from the queue and connect it to a extension... :)`
`20:53.43`	`clyrrad`	`Thanks Qwell`
`20:55.46`	`jontow`	`flujan: perhaps the voip-info.org wiki shall do you well.. if not.. maybe the flash operator panel`
`20:56.57`	`flujan`	`jontow, i already search the voip-info.org. I just find information about static queues using the agents.conf. But I am using asterisk with realtime...`
`20:57.43`	`flujan`	`I do not want to register agents... I just need a dial plan to remove a caller from the queue and connect it to a sip extension.`
`21:02.28`	`*** join/#asterisk r0d3nt\|m (n=RatMan@foster.stonedcoder.org)`
`21:05.58`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`21:09.10`	`*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)`
`21:10.07`	`flujan`	`hum... there are plans to add agents.conf into a realtime mode? Or make queues_members to support extensions directly?`
`21:12.26`	`seebq`	`ahah! I found my problem, but don't know how to fix -- my followme group is not doing a "native bridge" -- reading through the log files, the only difference between when I can hear audio and when I can't happens when my SIP dials my local # and then prints: "Attempting native bridge"`
`21:13.10`	`Assid`	`more providers need to start giving 729 passthroguh`
`21:13.38`	`*** join/#asterisk anthonyl (n=rmerz@c-24-12-179-217.hsd1.il.comcast.net)`
`21:15.51`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`21:16.11`	`seebq`	`the log when I can hear audio on a call transfer:`
`21:16.11`	`seebq`	`<PROTECTED>`
`21:16.11`	`seebq`	`<PROTECTED>`
`21:16.11`	`seebq`	`<PROTECTED>`
`21:16.14`	`seebq`	`<PROTECTED>`
`21:16.17`	`seebq`	`<PROTECTED>`
`21:16.20`	`seebq`	`<PROTECTED>`
`21:16.23`	`seebq`	`<PROTECTED>`
`21:16.25`	`seebq`	`<PROTECTED>`
`21:16.28`	`seebq`	`<PROTECTED>`
`21:16.30`	`Assid`	`damn.. cant seem to update on this speed`
`21:16.30`	`seebq`	`<PROTECTED>`
`21:16.38`	`seebq`	`the log when I cannot hear audio on a call transfer:`
`21:16.38`	`Assid`	`seebq: pastebin boy pastebin!!!`
`21:16.43`	`seebq`	`oh ok sorry heh`
`21:16.52`	`seebq`	`yeah irc was about to boot me for sure`
`21:17.51`	`flujan`	`hum... from the oreilly asterisk book... a queue member can be a agent or a channel...`
`21:18.04`	`flujan`	`so it could be a SIP/123456 for instance?`
`21:19.01`	`seebq`	`ok here we go, here's my problem: http://pastie.caboo.se/10800`
`21:19.31`	`seebq`	`that line 11 is missing on the one where i can't here audio: # -- Attempting native bridge of SIP/myaccount-ace7 and SIP/telasip-ab95`
`21:19.39`	`*** part/#asterisk ki2k (n=ki2k_@207.231.83.242)`
`21:19.48`	`flujan`	`guys... just a final question about definitions... To asterisk a channel could be a SIP/1234 or something similar?`
`21:20.22`	`flujan`	`So a queue-member could be a SIP/1234 extension right?`
`21:21.17`	`eKo1`	`Does doing a reload on the CLI reload musiconhold.conf?`
`21:22.18`	`tdonahue-laptop`	`no one has any idea why i wouldn't be getting rfc2833 DTMF even though "dtmfmode=rfc2833" is set?`
`21:22.57`	`CunningPike`	`flujan: Absolutely`
`21:23.33`	`CunningPike`	`flujan: Many people (and by that I mean me) use queues without using agents at all`
`21:25.09`	`[TK]D-Fender`	`seebq: Make sure your phones are "canreinvite=no"`
`21:25.48`	`seebq`	`[TK]D-Fender: will give it a shot`
`21:26.15`	`flujan`	`hi CunningPike ... thanks for the answer`
`21:26.17`	`seebq`	`what does that mean`
`21:26.23`	`CunningPike`	`flujan: np`
`21:26.45`	`rpm`	`is there a schema for func_odbc hints yet?`
`21:27.12`	`flujan`	`CunningPike, a long time ago i have problem with a digium board and you helped me... I have the answer now... The echo cancellation modules make the board have that strange behavior... :)`
`21:27.51`	`CunningPike`	`flujan: Glad you got it sorted out - we ditched our onboard e/c modules, too`
`21:30.58`	`*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)`
`21:32.23`	`*** join/#asterisk stympy (n=bencurti@dsl254-026-217.sea1.dsl.speakeasy.net)`
`21:35.40`	`eKo1`	`this moh stuff is annoying.`
`21:37.23`	`seebq`	`[TK]D-Fender: that canreinvite=no should go on the incoming trunk or outgoing ? not sure where to place that`
`21:37.52`	`[TK]D-Fender`	`seebq: GLOABLLY`
`21:38.10`	`clyrrad`	`ie... under [general]`
`21:38.23`	`Doctor`	`anyone know of a guide that explains each part of an asterisk configuration?`
`21:38.27`	`seebq`	`ahh interesting ok`
`21:38.46`	`clyrrad`	`seebq - search for "Asterisk The Future Of Telephony"`
`21:38.58`	`seebq`	`the book by O'Reilly`
`21:39.04`	`clyrrad`	`no`
`21:43.56`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`21:44.02`	`clyrrad`	`how do you set a module to auto load at start?`
`21:44.29`	`zoa`	`modules.conf`
`21:44.31`	`clyrrad`	`so that when you show modules your module will show with a 1`
`21:44.35`	`eKo1`	`Check your modules.conf file and make sure you have autoload`
`21:44.42`	`clyrrad`	`yes i have autoload`
`21:44.50`	`eKo1`	`then it should load automagically`
`21:44.58`	`clyrrad`	`I need something to envoke it though`
`21:45.03`	`eKo1`	`unless you have a noload => mymod.so`
`21:45.13`	`*** part/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu)`
`21:45.15`	`clyrrad`	`well if I show modules the module is shown with a 0`
`21:45.18`	`clyrrad`	`instead of a 1`
`21:45.19`	`eKo1`	`If you want to invoke it on the cli, enter load mymod.so`
`21:45.24`	`clyrrad`	`and it has not done its functionality`
`21:45.27`	`eKo1`	`clyrrad: that is a usage count`
`21:45.37`	`clyrrad`	`I want it to envoke automatically when Asterisk loads`
`21:45.39`	`eKo1`	`if it is there, your module is loaded`
`21:45.40`	`*** join/#asterisk DFiber (n=bwarner@65.172.101.235)`
`21:45.55`	`clyrrad`	`ah... ok so how do i make it run automatically?`
`21:46.19`	`clyrrad`	`wiht out having to call it in the CLI?`
`21:46.21`	`eKo1`	`clyrrad: you really need to read about modules`
`21:46.49`	`clyrrad`	`I been trying to do that all day - and playing with them to get it working`
`21:46.56`	`clyrrad`	`I got it compiled and have the .so`
`21:47.03`	`clyrrad`	`and I can see it is loaded`
`21:47.10`	`*** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`21:47.16`	`eKo1`	`ok. so?`
`21:47.19`	`clyrrad`	`I just need a way to have it envoke its actions automatically when asterisk loads`
`21:47.26`	`clyrrad`	`yes I have the .so`
`21:47.30`	`eKo1`	`what actions. what does your module do?`
`21:47.49`	`clyrrad`	`it syncs call queues by calling a PHP script by use of system() call`
`21:47.59`	`eKo1`	`Is it an app? Does it have cli commands associated with it? what?`
`21:48.07`	`tdonahue-laptop`	`have any ideas why i wouldn't be getting rfc2833 DTMF even though "dtmfmode=rfc2833" is set?`
`21:48.16`	`clyrrad`	`its main command is to call the php script`
`21:48.27`	`eKo1`	`tdonahue-laptop: are you pressing the buttons long enough?`
`21:48.49`	`tdonahue-laptop`	`eKo1, 2833 isn't even being requested in the SDP`
`21:48.50`	`eKo1`	`clyrrad: You have no idea what you're doing do you.`
`21:49.01`	`clyrrad`	`yes i certanlly do`
`21:49.03`	`*** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com)`
`21:49.15`	`eKo1`	`no because the questions you're asking make no sense`
`21:49.23`	`clyrrad`	`to you perhaps`
`21:49.27`	`eKo1`	`tdonahue-laptop: what is it requesting then?`
`21:50.18`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`21:50.25`	`clyrrad`	`It really is not that hard to understand - I have a module that calls a PHP script by using the system() function in C.... im not sure whats confused you...`
`21:50.47`	`clyrrad`	`in that modules load function is where the call is being made....`
`21:50.57`	`eKo1`	`yes, I understand that part just fine.`
`21:51.03`	`clyrrad`	`I am looking for a way to have that functionality ran automatically each time asterisk loads`
`21:51.06`	`clyrrad`	`and that is it`
`21:51.14`	`clyrrad`	`there is nothing more to my question then simply that`
`21:51.36`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`21:51.36`	`*** mode/#asterisk [+o mog] by ChanServ`
`21:51.37`	`eKo1`	`Well, does your module export a command?`
`21:51.45`	`clyrrad`	`no`
`21:51.54`	`tdonahue-laptop`	`eKo1, here is the 200 OK with SDP`
`21:51.54`	`clyrrad`	`becase I did not want to have to run it from the CLI`
`21:51.55`	`tdonahue-laptop`	`http://pastebin.ca/153837`
`21:51.59`	`eKo1`	`Does it export CLI commands?`
`21:52.15`	`clyrrad`	`no - I dont want it to be used from the CLI`
`21:52.22`	`eKo1`	`Does it export a dialplan command?`
`21:52.30`	`tdonahue-laptop`	`it isn't requesting any DTMF handling so our carrier is giving us the DTMF inband (doesn't work so well with G.729)`
`21:52.37`	`*** join/#asterisk hads (n=hads@mail.nice.net.nz)`
`21:52.39`	`clyrrad`	`I also do not need it to be accessable to the dial plan`
`21:52.48`	`*** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net)`
`21:52.52`	`clyrrad`	`I just need it ran one time when asterisk loads and thats it`
`21:53.00`	`*** join/#asterisk redondos (n=redondos@190.48.11.72)`
`21:53.11`	`eKo1`	`clyrrad: Is it a resourse?`
`21:53.18`	`clyrrad`	`when the PC reboots it runs fine becase it have linked it in the init.d script for asterisk`
`21:53.22`	`eKo1`	`You can do that with a resource.`
`21:53.32`	`clyrrad`	`the problem is to get it to run if I stop and re-start asterisk`
`21:53.42`	`clyrrad`	`ok....`
`21:53.47`	`clyrrad`	`that what I was trying to get at`
`21:53.48`	`Kerry_G`	`anyone know if there is a SIP firmware for the cisco 7936?`
`21:54.07`	`eKo1`	`clyrrad: how about making a script that runs your PHP script or what not before you start Asterisk?`
`21:54.14`	`eKo1`	`Using a module is overkill.`
`21:54.27`	`clyrrad`	`ok....`
`21:54.38`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`21:54.39`	`clyrrad`	`but how about handling when the server reboots`
`21:54.48`	`clyrrad`	`I need this script to run each and every time asterisk is started`
`21:55.07`	`clyrrad`	`even if it a manual stop and start or an automatic one when the server reboots`
`21:55.08`	`tdonahue-laptop`	`clyrrad, use that script as your init script...`
`21:55.09`	`eKo1`	`You can do that easily with some shell scripting.`
`21:55.25`	`tessier__`	`djpbx*CLI> zap show channels`
`21:55.25`	`tessier__`	`No such command 'zap' (type 'help' for help)`
`21:55.30`	`tessier__`	`Anyone know why zap would not work?`
`21:55.36`	`clyrrad`	`yes - shell scripting works if the server reboots..... but not if you manually stop and start asterisk`
`21:55.38`	`tessier__`	`Maybe that is why my PRI doesn't work. :)`
`21:56.02`	`tdonahue-laptop`	`tessier__, does lsmod show zaptel and the driver for the card loaded?`
`21:56.04`	`clyrrad`	`I need it to work automatic which is why I tried the module approach....`
`21:56.17`	`tessier__`	`zaptel 196868 1 wanpipe`
`21:56.20`	`tessier__`	`tdonahue-laptop: Yes indeed`
`21:56.32`	`eKo1`	`clyrrad: how are you manually starting asterisk?`
`21:56.34`	`clyrrad`	`I figured a module would load it and envoke it once..... but it appears thats not the case`
`21:56.40`	`clyrrad`	`eko1 2 ways....`
`21:56.56`	`clyrrad`	`1) automatically with init script when server reboots`
`21:56.56`	`clyrrad`	`or`
`21:56.56`	`clyrrad`	`2) manually with asterisk -vvvvvvvvgc`
`21:57.18`	`eKo1`	`ok, then run the script first before calling asterisk -vvvvvgc`
`21:57.27`	`clyrrad`	`it needs to run after it`
`21:57.51`	`clyrrad`	`it adds queue members dynamically so Asterisk must be running first`
`21:57.54`	`jontow`	`flujan: search bugs.digium.com -- i don't remember the id, but yes, i've posted a patch to make agents realtime-backed, a month or a few ago`
`21:57.59`	`eKo1`	`then just run asterisk without the -vgc arguments and call your script`
`21:58.24`	`clyrrad`	`but how do i get that to do it automatically with out me manually having to do it`
`21:58.35`	`clyrrad`	`i got the part working if the server reboots....`
`21:58.38`	`clyrrad`	`it starts asterisk`
`21:58.41`	`clyrrad`	`then runs the script`
`21:58.48`	`clyrrad`	`but if im at the CLI and type stop now`
`21:59.00`	`eKo1`	`Use a shell script`
`21:59.03`	`clyrrad`	`then asterisk -vvvvvvgc I would like the script to get started automatically`
`21:59.12`	`tdonahue-laptop`	`why don't you run the script that you use to start asterisk when the machine boots?`
`21:59.59`	`clyrrad`	`becase everyone who uses the server knows to start asterisk -vvvvvvgc by its binary....`
`22:00.03`	`clyrrad`	`so i wanted to automate it`
`22:00.06`	`clyrrad`	`if possible`
`22:00.15`	`*** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`22:00.18`	`eKo1`	`clyrrad: make an shell command then`
`22:00.21`	`TripleFFFF`	`hey guys hwats chan_phone crap ?`
`22:00.35`	`mog`	`its that phone jack`
`22:00.45`	`mog`	`or qnet linejack cards`
`22:00.49`	`clyrrad`	`eKo1 - then just tell everyone to use that script?`
`22:00.54`	`clyrrad`	`shell script i mean....`
`22:01.03`	`TripleFFFF`	`hmmm well its pretty stupd to have asterisk NEED the conf file no ?`
`22:01.21`	`*** join/#asterisk spr1te (i=spr1te@194.187.130.227)`
`22:01.46`	`clyrrad`	`Wish there was just a way to have asterisk load it up after it loads...`
`22:02.25`	`eKo1`	`There is, but you need to make your module behave as a resource. E.g. res_agi, res_odbc, etc...`
`22:02.34`	`eKo1`	`But it is overkill.`
`22:02.38`	`tdonahue-laptop`	`TripleFFFF, just noload => chan_phone.so in modules.conf`
`22:02.44`	`clyrrad`	`How can I set it as a resource?`
`22:02.59`	`eKo1`	`clyrrad: check the source code and start hacking one out.`
`22:03.10`	`*** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-221.dsl.pltn13.sbcglobal.net)`
`22:03.40`	`tdonahue-laptop`	`night all`
`22:03.43`	`tessier__`	`tdonahue-laptop: Any other ideas as to why zap show pri does not work?`
`22:04.04`	`clyrrad`	`looks similair to a module`
`22:04.31`	`eKo1`	`It is a module.`
`22:04.38`	`eKo1`	`Anything ending in .so is a module.`
`22:04.53`	`clyrrad`	`they just named different then...`
`22:04.57`	`clyrrad`	`app_ vs res_`
`22:05.33`	`clyrrad`	`would it work to put the module in the /res folder and rebuild asterisk?`
`22:05.37`	`clyrrad`	`or there is more to it then that`
`22:05.39`	`eKo1`	`There are other differences besides the name.`
`22:06.15`	`tessier__`	`bah, sangoma tech support is closed already`
`22:06.22`	`tessier__`	`I hate only being able to do one round of emails per day`
`22:06.31`	`tdonahue-laptop`	`tessier__, i would make sure that your zaptel.conf and zapata.conf are correct`
`22:06.55`	`tessier__`	`tdonahue-laptop: I have checked and double checked them. The setting seem to be correct for the pri. Should asterisk or ztcfg or something cough up an error somewhere if they are not?`
`22:07.12`	`eKo1`	`ztcfg should err`
`22:07.21`	`tessier__`	`It does not. ztcfg is fine.`
`22:07.24`	`TripleFFFF`	`lol`
`22:07.26`	`*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`22:07.57`	`eKo1`	`Does chan_zap load?`
`22:09.08`	`tessier__`	`eKo1: I don't see any errors but don't see it loading either in the asterisk startup either.`
`22:09.31`	`eKo1`	`What happens when you type "load chan_zap.so" from the CLI?`
`22:09.36`	`tessier__`	`eKo1: Do I have to manually add it to modules.conf?`
`22:09.45`	`eKo1`	`No you don't.`
`22:10.02`	`tdonahue-laptop`	`it should load as long as it isn't set to noload`
`22:10.17`	`tessier__`	`Unable to load module chan_zap.so`
`22:10.17`	`tessier__`	`Aug 29 15:10:41 WARNING[4459]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: cannot open shared object file: No such file or directory`
`22:10.18`	`tessier__`	`hmm`
`22:10.35`	`aptura`	`I am begining to wonder if my isp is blocking my registration packets. most of my registration are unregistered.`
`22:10.35`	`tessier__`	`It seems it didn't get built.`
`22:10.44`	`seebq`	`man i still can't figure this out -- this canreinvite=no was promising, but did not fix my problem, so strange`
`22:11.02`	`seebq`	`it's only on a ring-group for a "follow me" that i can't hear audio`
`22:11.24`	`tessier__`	`Maybe I need to rebuild asterisk`
`22:11.28`	`syzygyBSD`	`tessier__: do you use udev?`
`22:11.35`	`seebq`	`canreinvite=no is set globally and asterisk reloaded, and still it doesn't switch me over`
`22:12.13`	`syzygyBSD`	`does lsmod show zaptel?`
`22:12.43`	`tessier__`	`syzygyBSD: um....CentOS 4.3. I think that uses udev.`
`22:12.47`	`eKo1`	`tessier__: you need to rebuild zaptel`
`22:12.53`	`*** join/#asterisk wundapat (n=wonderbu@159.191.241.94)`
`22:12.54`	`tessier__`	`lsmod does show zaptel`
`22:12.56`	`syzygyBSD`	`and zaptel is distributed seperatly from asterisk, you need to compile it alone`
`22:13.00`	`*** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`22:13.03`	`eKo1`	`rebuild that and then asterisk`
`22:13.03`	`tessier__`	`eKo1: hmm...I already did that once today...`
`22:13.08`	`syzygyBSD`	`tessier try to start asterisk now`
`22:13.10`	`tessier__`	`ok, I am rebuilding asterisk now`
`22:13.14`	`syzygyBSD`	`no need to`
`22:13.29`	`syzygyBSD`	`you just need to add a delay after modprobe for udev to create the devices`
`22:13.31`	`tessier__`	`syzygyBSD: I have restarted asterisk a few times since I rebuilt zaptel an hour ago.`
`22:14.05`	`syzygyBSD`	`tessier__: what devices are in your /dev/zap/`
`22:14.18`	`syzygyBSD`	`or are they /dev/zap[1-9]`
`22:14.22`	`syzygyBSD`	`no directory`
`22:15.12`	`tessier__`	`syzygyBSD: There is a /dev/zap/ and a bunch of devices in there named by number and channel ctl pseudo timer etc`
`22:16.17`	`syzygyBSD`	`updatedb && locate chan_zap`
`22:17.43`	`tessier__`	`syzygyBSD: I just rebuilt and installed asterisk. Now there is a chan_zap.so. There was not before.`
`22:18.19`	`syzygyBSD`	`hmm, i wouldn't have thought it was in modules.conf till you recompiled asterisk`
`22:18.28`	`syzygyBSD`	`good to know`
`22:18.51`	`tessier__`	`ah, zap show channels works now`
`22:20.46`	`tessier__`	`ok, zap show channels works but doesn't list anything.`
`22:24.03`	`*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)`
`22:25.47`	`*** join/#asterisk docelm0 (n=vircuser@55-65.126-70.tampabay.res.rr.com)`
`22:27.13`	`Stp1800`	`Hi guys.`
`22:27.38`	`*** part/#asterisk seebq (n=cbq@207.59.210.210)`
`22:28.07`	`eKo1`	`tessier__: you need to configure zapata.conf`
`22:30.45`	`tessier__`	`eKo1: It is configured.`
`22:31.09`	`tessier__`	`http://pastebin.ca/153869`
`22:31.36`	`tessier__`	`Anything missing there?`
`22:31.48`	`tessier__`	`I copied that from another config which I used on a PRI a year ago which worked.`
`22:31.50`	`eKo1`	`What happens when you enter: reload chan_zap.so?`
`22:32.52`	`tessier__`	`<PROTECTED>`
`22:32.52`	`tessier__`	`Aug 29 15:32:35 WARNING[12054]: chan_zap.c:10873 setup_zap: Ignoring switchtype`
`22:32.52`	`tessier__`	`Aug 29 15:32:35 WARNING[12054]: chan_zap.c:10873 setup_zap: Ignoring signalling`
`22:32.59`	`tessier__`	`Ignoring switchtype?`
`22:33.16`	`tessier__`	`Switchtype seems like it would be important.`
`22:33.32`	`eKo1`	`it ignores those because those can't be reloaded`
`22:33.35`	`justinu\|laptop`	`it only reads those on startup`
`22:33.47`	`tessier__`	`ah, ok`
`22:33.51`	`tessier__`	`Well, there were no errors.`
`22:34.12`	`eKo1`	`and what does 'zap show channels' say?`
`22:34.42`	`tessier__`	`The usual header but does not list any channels. The switch probably has my PRI down at the moment so it is not in alarm.`
`22:34.58`	`tessier__`	`Since I now have chan_zap loaded I should probably call them and have them bring it up for another test`
`22:35.02`	`*** join/#asterisk new2voip (n=chatzill@user-24-214-152-5.knology.net)`
`22:35.09`	`tessier__`	`It is quite annoying that they will only have it up when we are actually working on it`
`22:39.57`	`new2voip`	`I installed Fedora C5 and asterisk. I didn't notice that the audio isn't working (snd-es1968). Asla project site says that capture isn't supported. I am thinking this isn't good because this is needed for VM? Would I be correct?`
`22:40.24`	`eKo1`	`No.`
`22:40.45`	`syzygyBSD`	`does anyone use rxfax?`
`22:41.17`	`new2voip`	`ekol: No, I would not be correct?`
`22:41.20`	`syzygyBSD`	`well, my real question is are there any problems with rxfax`
`22:42.48`	`eKo1`	`new2voip: What makes you think your sound card driver has anything to do with voicemail?`
`22:44.02`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`22:44.08`	`Druken`	`anyone in ontario that has a cheapo x100p that i can grab off them for $10 - $20 bux?`
`22:44.29`	`*** join/#asterisk tasat- (n=tasat-@c-67-180-181-221.hsd1.ca.comcast.net)`
`22:44.53`	`new2voip`	`guessing that a sound card was needed to record and playback of vm. glad that I am wrong`
`22:46.10`	`eKo1`	`No.`
`22:46.36`	`*** join/#asterisk h3x (n=h3xor@64.192.116.17)`
`22:46.40`	`Druken`	`don't need a soundcard for anything in asterisk, that i'm awear of`
`22:46.51`	`eKo1`	`You do for chan_alsa`
`22:46.53`	`h3x`	`local calling`
`22:46.57`	`h3x`	`Local/ channel`
`22:47.23`	`syzygyBSD`	`really? I don't have a sound card and local channels work great for me...`
`22:47.25`	`Druken`	`h3x: local can do done to anything... LOCAL/ext@context`
`22:47.37`	`h3x`	`er sorry`
`22:47.38`	`h3x`	`not Local`
`22:47.48`	`Druken`	`he's thinking console`
`22:47.52`	`h3x`	`yes`
`22:48.02`	`*** join/#asterisk _Vile (n=vile@90.b160.bendtel.net)`
`22:48.15`	`syzygyBSD`	`hmm, never used that, probably because I have never had a sound card in an asterisk box`
`22:49.00`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`22:49.14`	`shmaltz`	`anybody know if it's posible to flash the FXO on a SPA3000?`
`22:49.35`	`Druken`	`uhg... call waiting?`
`22:50.14`	`*** join/#asterisk batphone (n=bugz@cpe-70-123-122-41.houston.res.rr.com)`
`22:50.14`	`shmaltz`	`Druken, actulay for lagecy PBX integration`
`22:50.23`	`batphone`	`how do you set the timeout for musiconhold() ?`
`22:50.32`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`22:50.42`	`tasat-`	`hi -- ran into the file descriptor limit, doubled the limits in limits.conf per asterisk's README, rebooted, and see no difference -- is there something I'm missing?`
`22:51.13`	`Druken`	`shmaltz: ahh... then it's excuseable`
`22:51.20`	`*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)`
`22:51.41`	`shmaltz`	`Druken you make me laugh, do u know the answer?`
`22:52.24`	`eKo1`	`tasat-: how about using ulimit?`
`22:52.31`	`Druken`	`i don't actually... i'm sure i could look at the documentation for the box and see if it's possible...`
`22:52.49`	`Druken`	`i got one lying around somewhere....`
`22:53.44`	`shmaltz`	`Druken, thanks that I can do alone`
`22:54.10`	`Druken`	`:)`
`22:54.20`	`tasat-`	`eKo1: I checked ulimit -a after rebooting and it should the updated number... also, running as user 'asterisk' not root`
`22:54.25`	`Druken`	`good... cause i didn't feel like searching my office for it...`
`22:54.34`	`tasat-`	`...it showed...`
`22:54.50`	`shmaltz`	`Druken, I think you should go and rest a bit :P`
`22:55.23`	`Druken`	`shmaltz: are you my girlfriend?? she's always telling me that`
`22:56.49`	`shmaltz`	`Druken, nah, its just that my girlfriend just told me the same thing so I just passed it on :P`
`22:57.23`	`shmaltz`	`actualy my wife`
`22:57.26`	`eKo1`	`uh oh....could it be....you both share the same girl?!`
`22:57.40`	`shmaltz`	`eKo1, don't tell him :P`
`22:58.24`	`Druken`	`eKo1: E Gads!!!! hope not... is your girl lying in my bed watching tv? hehehe`
`22:59.56`	`*** join/#asterisk wglenncamp (n=5849451c@208.203.43.124)`
`22:59.58`	`*** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-221.dsl.pltn13.sbcglobal.net)`
`23:01.41`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`23:01.57`	`shmaltz`	`http://www.sipura.com/Documents/rnote/rn3k-3.1.3a.htm`
`23:01.58`	`shmaltz`	`the last to options tell me something :P`
`23:02.55`	`shmaltz`	`tzafrir ping`
`23:03.03`	`shmaltz`	`~seen tzafrir`
`23:03.20`	`jbot`	`tzafrir is currently on #asterisk, last said: 'nounoursfr, why change the GID?'.`
`23:03.21`	`shmaltz`	`~sleeping?`
`23:03.24`	`jbot`	`well, sleeping is the magical thing geeks have forgotten how to do`
`23:05.34`	`shmaltz`	`anybody here ever tried xircoms FXO products?`
`23:05.35`	`hmmhesays`	`bah according to the documentation this is a mips32 proc, but when I compile for mips323 it freaks out`
`23:05.53`	`shmaltz`	`hm, it's not xircom`
`23:06.00`	`shmaltz`	`what the name of the company tzafrir works for?`
`23:06.16`	`shmaltz`	`xorcom *****`
`23:07.32`	`Druken`	`a double hookflash....`
`23:07.57`	`Druken`	`i bet ya that has to be the phone that's connected to the device, and not a voip phone or asterisk`
`23:08.37`	`*** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net)`
`23:08.39`	`shmaltz`	`Druken, well I'm trying to do that from another SPA 3000 (or 1000) and I think it should work`
`23:09.01`	`Druken`	`i dunno.... might, might not....`
`23:10.00`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`23:10.16`	`gmfm`	`does anyone know if it's possible to do callerid routing with wildcards? for example exten => 1234/949XXXXXXX`
`23:10.27`	`Druken`	`looks like if you send it an out of band hookflash it'll do it`
`23:10.50`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
`23:12.42`	`Doctor`	`hmm if I hook up an FXO, asterisk etc, if the phone rings and then I pick up a normal phone, will asterisk do anything screwy?`
`23:13.04`	`tasat-`	`Can anyone help with my file descriptor problem?`
`23:13.13`	`Druken`	`no.. it'll continue to do that it normally does`
`23:13.22`	`Druken`	`er, what`
`23:15.28`	`tasat-`	`Druken: ran into a limit and tried quadrupling the limits in limit.conf as explained in Asterisk's README -- rebooted and see no difference...`
`23:16.43`	`*** join/#asterisk wundapat (n=wonderbu@159.191.241.94)`
`23:18.21`	`*** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`23:18.24`	`wundapat`	`when i am on a call this populates the console: Aug 29 00:06:15 WARNING[7602]: chan_iax2.c:7550 socket_read: Received mini frame before first full voice frame`
`23:18.38`	`wundapat`	`just keeps showing that`
`23:18.43`	`wundapat`	`how do i fix that?`
`23:21.30`	`*** part/#asterisk new2voip (n=chatzill@user-24-214-152-5.knology.net)`
`23:22.52`	`jmorgan`	`wundapat: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg29493.html`
`23:25.16`	`*** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com)`
`23:32.02`	`tasat-`	`jmargan: any idea about the file descriptors?`
`23:32.08`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`23:32.12`	`tasat-`	`sorry, jmorgan`
`23:36.51`	`jmorgan`	`tasat-: sorry, don't know about file descriptor limits`
`23:37.07`	`*** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk)`
`23:37.24`	`tasat-`	`thanks -- anyone else have any ideas?`
`23:37.29`	`Doctor`	`hmmm this dialplan stuff is :-/`
`23:37.48`	`*** join/#asterisk lordbaron (n=redbaron@host55-226.rancor.birch.net)`
`23:38.30`	`*** join/#asterisk \|dennis\| (n=dennis@shc.edu.bz)`
`23:39.09`	`Doctor`	`I want to hook up an FXO, and then when the phone ring, the normal phones ring, and connected VoIP phones ring, but after 4 rings voicemail picks up`
`23:44.53`	`nextime`	`tasat- a line with asterisk hard nofile 1024 and asterisk soft nofile 1024 shuld be enought, anyway, try to check if you load pam_limits.so module... and if it don't work as a work around you can try to put at the start of the asterisk init script a line with "ulimit -n 1024"`
`23:45.48`	`tasat-`	`nextime: thanks -- I've got that line there... I'll check the pam_limits modules`
`23:45.57`	`*** join/#asterisk JakBeatZ (n=JakBeatZ@beta.arionetworks.ca)`
`23:46.32`	`tasat-`	`nextime: I'm using safe_asterisk -- I see a line in there with ulimit -c -- does that have any affect on ulimit -n?`
`23:46.56`	`nextime`	`tasat- : no effect, -c is only for core file size`
`23:48.12`	`*** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`23:48.56`	`JakBeatZ`	`Anyone here got a 7970 working? I'm having a hard time getting it to pull the SEP<MAC>.cnf.xml off my TFTP server. The TFTP server log shows it always requesting the term70.default.loads file. I extracted that file from the cmterm 8.3.0 SIP firmware .cop file but it doesn't matter if that file exists or not, it still constantly requests it.. any ideas?`
`23:50.08`	`*** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com)`
`23:50.50`	`tasat-`	`nextime: I don't seem to have a pam_limits.so in /usr/lib/asterisk/modules -- is it someplace else? also mention of it in modules.conf`
`23:51.11`	`tessier__`	`djpbx*CLI> pri show span 1`
`23:51.12`	`tessier__`	`No PRI running on span 1`
`23:51.22`	`tessier__`	`the switch still has the pri up but asterisk does not see it`
`23:52.14`	`nextime`	`tasat- : it isn't an asterisk module, it is a pam module`
`23:52.15`	`mitcheloc_`	`JakBeatZ, Kerry_G is having the same issue over in #snap right now`
`23:52.31`	`tessier__`	`How/where do I define a pri spam?`
`23:52.41`	`JakBeatZ`	`mitcheloc_: Fun! I'll go say Hi..`
`23:53.07`	`nextime`	`you have to look for it in /etc/pam.d/login`
`23:53.20`	`tessier__`	`span=1,1,0,esf,b8zs`
`23:53.43`	`tessier__`	`That should create span 1`
`23:53.48`	`*** join/#asterisk roving_prole (n=Harper@c-71-199-16-110.hsd1.co.comcast.net)`
`23:54.11`	`tasat-`	`nextime: no mention of it there --`
`23:54.16`	`tessier__`	`wanpipe1 card 0 OK 0 0 0`
`23:54.21`	`tessier__`	`zap show status shows the card is ok`
`23:54.48`	`nextime`	`tessier__ this is the config in /etc/zaptel.conf, you need to configure /etc/asterisk/zapata.conf too to see pri running in asterisk`
`23:55.14`	`CunningPike`	`~seen dlynes_laptop`
`23:55.36`	`jbot`	`dlynes_laptop <n=dlynes@S01060016b6c052ee.vc.shawcable.net> was last seen on IRC in channel #asterisk, 3d 7h 40m 19s ago, saying: 'EmleyMoor: you can use friend for iax2 as well, yes'.`
`23:55.48`	`Doctor`	`I want to hook up an FXO, and then when the phone ring, the normal phones ring, and connected VoIP phones ring, but after 4 rings voicemail picks up. Anyone know how to do this?`
`23:56.12`	`nextime`	`tasat- : good, you found the problem. add a line with session required pam_limits.so`
`23:56.29`	`tasat-`	`nextime: no you found it -- I love you! thanks!`
`23:56.56`	`tasat-`	`nextime: can I avoid a reboot?`
`23:57.18`	`nextime`	`tasat- : i'm not sure about that, it is better to reboot to be sure`
`23:57.28`	`lunaphyte`	`has anyone experimented with tycho?`
`23:58.04`	`*** join/#asterisk smackus2 (n=smackus2@c-67-169-248-217.hsd1.ut.comcast.net)`
`23:59.09`	`tessier__`	`nextime: Ah, I think I was missing a channel => 1-23 line`
`23:59.09`	`smackus2`	`i am looking in the sip.cfg for my polycom 301s for somewhere to enable and disable the ability to forward the phone. I do not see anything logical to me... does anyone know what I should be looking for? I see all of the features, but unless i missed it there is nothing for forwarding.`
`23:59.47`	`nextime`	`tessier__ it seem to me that you are thinking right`

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