irclog2html for #asterisk on 20061017

`00:01.00`	`*** join/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net)`
`00:05.22`	`salaud`	`dumb question that gets asked all the time... but, what's the cheapest outbound only per minute rate ... at the 40,000 minute level?`
`00:05.25`	`salaud`	`any clues?`
`00:06.28`	`saftsack`	`why do all buy SIP phones? sip phone: 160$ analog phone: 20$ + about 50$ per channel`
`00:06.30`	`TripleFFFF`	`Adjusting skew to -10900.`
`00:06.30`	`TripleFFFF`	`[2006-10-16 20:06:11] Adjusting skew to -10950.`
`00:06.31`	`TripleFFFF`	`nice`
`00:07.17`	`Juggie`	`because, sip phones are multi line and easily support transfering, etc... and use existing network infrastructure`
`00:08.03`	`saftsack`	`thats an good argument :9`
`00:12.42`	`[TK]D-Fender`	`saftsack : because we like having the ability to put people on hold, handle more than 2 calls with out waiting for CID. So that we can transfer without screing around with DTMF with messes up IVR's`
`00:12.50`	`*** part/#asterisk salaud (n=salaud@h-66-166-226-2.sttnwaho.covad.net)`
`00:13.08`	`[TK]D-Fender`	`saftsack : Normal companies don't like haivng to fidgit with phones to do what should be 'easy:"`
`00:13.39`	`saftsack`	`bri phones cost 25$ here in europe and they have the ability to do most things sip phones are doing too`
`00:13.58`	`justinu\|laptop`	`$25? nice`
`00:14.00`	`saftsack`	`bri phones are cripples beetwen analogues and sip phones`
`00:14.30`	`saftsack`	`yes we got some phones for 25eu here. middle quality ware.`
`00:14.32`	`manalive`	`I'm trying to use module-assistant in Debian to install zaptel drivers. I'm trying to use module-assistant but it's generating errors http://paste.uni.cc/10867 saying the kernel sources aren't there; but the sources are there, this application is mv'ing it`
`00:21.38`	`benjk`	`saftsack, cause nothing sucks as bad as analog does`
`00:23.14`	`TripleFFFF`	`RECV FAX: Failure to train modems`
`00:23.18`	`TripleFFFF`	`is what my log says`
`00:23.45`	`benjk`	`why would you want to use a train modem?`
`00:23.52`	`TripleFFFF`	`i dont`
`00:23.58`	`TripleFFFF`	`its just saying that lol`
`00:24.08`	`benjk`	`either you take the train, or you use a modem`
`00:24.20`	`TripleFFFF`	`lol`
`00:24.24`	`TripleFFFF`	`wat a nice uhumor tonight ;)`
`00:24.40`	`benjk`	`tonight?`
`00:24.44`	`benjk`	`which planet are you on?`
`00:24.55`	`benjk`	`the one I'm on it is 9:23 am now`
`00:25.09`	`justinu\|laptop`	`it's afternoon in the center of civilization`
`00:25.11`	`justinu\|laptop`	`get a clue`
`00:25.17`	`benjk`	`heh`
`00:25.41`	`benjk`	`isn't that in the middle of the Pacific?`
`00:25.49`	`justinu\|laptop`	`well, i guess it's going on evening here now`
`00:26.04`	`justinu\|laptop`	`getting dark early again`
`00:26.17`	`benjk`	`gotta move south then`
`00:26.43`	`justinu\|laptop`	`i think i like the colder climates away from the equator`
`00:26.55`	`justinu\|laptop`	`i'd have to move way south, like melbourne`
`00:27.02`	`benjk`	`gotta move ver ver far south then`
`00:27.11`	`JT`	`equator climates are evil`
`00:27.11`	`benjk`	`very`
`00:27.33`	`justinu\|laptop`	`like singapore... great city, but the climate there is harsh`
`00:27.49`	`JT`	`yeah`
`00:27.56`	`JT`	`wake up and you want to die every day`
`00:28.07`	`benjk`	`there is nothing like crossing the equator in the middle of nowhere for the first time in a yacht`
`00:28.23`	`justinu\|laptop`	`heh, i've like to go on a yacht`
`00:28.29`	`justinu\|laptop`	`sail across an ocean`
`00:28.55`	`JT`	`get rescued by rthe navy...`
`00:29.01`	`justinu\|laptop`	`heh, i think i could make it`
`00:29.04`	`TripleFFFF`	`yeah on the isalnd that lost is in`
`00:29.12`	`justinu\|laptop`	`be nice to have an experienced captain though`
`00:29.24`	`benjk`	`I'll be your skipper`
`00:29.32`	`benjk`	`you pay the bareboat charter`
`00:29.39`	`justinu\|laptop`	`don't tell me, a 3 hour tour?`
`00:29.44`	`*** join/#asterisk chmod007 (n=chmod007@ua-83-227-238-221.cust.bredbandsbolaget.se)`
`00:29.54`	`benjk`	`more like 3 weeks`
`00:29.54`	`*** part/#asterisk chmod007 (n=chmod007@ua-83-227-238-221.cust.bredbandsbolaget.se)`
`00:29.54`	`justinu\|laptop`	`i don't think i can afford a yact`
`00:30.00`	`benjk`	`bareboat charter?`
`00:30.11`	`justinu\|laptop`	`no idea what it costs`
`00:30.15`	`benjk`	`its about 300-500 USD per day or so`
`00:30.26`	`benjk`	`depending on location and size of the yacht`
`00:30.37`	`justinu\|laptop`	`3 weeks to sail LA to Tokyo?`
`00:30.42`	`benjk`	`50 ft is where it starts getting interesting for ocean sailing`
`00:30.52`	`benjk`	`depends on the season`
`00:30.56`	`benjk`	`bit tough`
`00:31.09`	`TripleFFFF`	`100 footer yach called the temrinator`
`00:31.40`	`JT`	`it can have the yacht called The Typo following closely`
`00:31.41`	`justinu\|laptop`	`sail only, no motor?`
`00:31.44`	`justinu\|laptop`	`lol`
`00:31.44`	`benjk`	`let's ask Jim "Netscape" Clarke if he lets us sail Hyperion`
`00:31.47`	`TripleFFFF`	`saw that when i lived in puerto vallarta`
`00:32.01`	`TripleFFFF`	`guy had a chopper on it and flew to work for 2 weeks then sailed away`
`00:32.17`	`TripleFFFF`	`JT :) LOL`
`00:32.31`	`TripleFFFF`	`yeah my keyboard disagrees with my finger all the time`
`00:32.35`	`benjk`	`http://www.charterworld.com/index.html?sub=yacht-charter&charter=sailing-yacht-hyperion-1095`
`00:32.37`	`benjk`	`he does`
`00:32.45`	`benjk`	`for only 125.000 USD per week`
`00:32.48`	`TripleFFFF`	`actualy speech recognition is more accurate in my case ;)`
`00:32.51`	`benjk`	`well then`
`00:33.00`	`benjk`	`peanuts`
`00:33.20`	`benjk`	`I guess 50 ft is all we can afford for now justin`
`00:34.39`	`justinu\|laptop`	`oh, that's very affordable`
`00:35.08`	`benjk`	`1 Siemens Hicom 130 wireless telephone exchange with 21 extensions`
`00:35.26`	`TripleFFFF`	`http://www.charterworld.com/index.html?sub=yacht-charter&charter=luxury-motor-yacht-clarity-1054`
`00:35.29`	`TripleFFFF`	`that my style`
`00:35.37`	`benjk`	`I don't like motor yachts`
`00:35.41`	`benjk`	`its awful`
`00:35.45`	`benjk`	`noisy`
`00:35.47`	`benjk`	`and stinks`
`00:35.59`	`TripleFFFF`	`50k per week is nothing i used to make that @ 19`
`00:36.12`	`benjk`	`50ft not 50k`
`00:36.15`	`TripleFFFF`	`actlualy 30k usd but exchange rate was 1.54 at the time`
`00:36.34`	`TripleFFFF`	`its not 1.02`
`00:36.38`	`benjk`	`I used to charter a Feeling 486 (just under 50 ft) in New Caledonia`
`00:36.43`	`benjk`	`that was about 400 per day`
`00:36.44`	`TripleFFFF`	`nice USDollar;.lol im thinking of charging euros`
`00:36.52`	`TripleFFFF`	`like all the oil guys.. fuck usd it sucsk now`
`00:37.01`	`JT`	`drug dealing @ 19? :P`
`00:37.07`	`TripleFFFF`	`nah had porn sites`
`00:37.13`	`TripleFFFF`	`in the begining when thereswaas 4`
`00:37.33`	`benjk`	`if you go on a holidy in a nice resort you spend easily 400 USD with 2 people on hotel and entertainment`
`00:37.39`	`TripleFFFF`	`like persiankitty , webyoung me and another one till sharsawky the penthouse goon started to fuck everyone wih 12 cents checks and rapped us all`
`00:38.04`	`TripleFFFF`	`good ol days.. then the chargeback started.. nic trend that we still se in voip these days`
`00:38.17`	`TripleFFFF`	`jus got another one`
`00:38.19`	`TripleFFFF`	`from paypal`
`00:38.24`	`benjk`	`but on a yacht, you have no other expenses`
`00:38.33`	`TripleFFFF`	`yeah.. beer women and gas`
`00:38.45`	`TripleFFFF`	`ah sail .. no gas`
`00:38.46`	`[1]BigBadHoss`	`haha`
`00:38.48`	`TripleFFFF`	`wind.. ;)`
`00:39.06`	`TripleFFFF`	`or a giant tsunami to push you to destination .. now thats SURFING.`
`00:39.15`	`[1]BigBadHoss`	`wouldnt be surprised if the gov started a wind tax`
`00:39.24`	`benjk`	`on the ocean you won't recognise a tsunami`
`00:39.29`	`benjk`	`its under water`
`00:39.35`	`JT`	`TripleFFFF: did you actually have original content or what?`
`00:39.40`	`benjk`	`only barely noticeable`
`00:39.58`	`benjk`	`only when it gets close to shore will the water build up`
`00:40.22`	`TripleFFFF`	`yeah`
`00:40.22`	`TripleFFFF`	`lol`
`00:40.30`	`TripleFFFF`	`streams and all`
`00:41.18`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`00:41.22`	`JT`	`nice`
`00:41.31`	`JT`	`how did you start that venture?`
`00:43.41`	`TripleFFFF`	`luck`
`00:43.52`	`justinu\|laptop`	`yeah, I want a real sailboat`
`00:43.54`	`JT`	`i see`
`00:44.09`	`justinu\|laptop`	`i want to experience total quiet and see the sky without light polution`
`00:45.25`	`benjk`	`south pacific`
`00:45.25`	`TripleFFFF`	`move to canada`
`00:45.27`	`TripleFFFF`	`lol`
`00:45.53`	`justinu\|laptop`	`great white north?`
`00:45.54`	`benjk`	`sailboat is for kids`
`00:45.57`	`TripleFFFF`	`actulay this reminded me ibill dot com still owes me 50k`
`00:46.13`	`TripleFFFF`	`form the old time.. they fucked 20 mill of out webmasters.. and are still in business it seem`
`00:46.17`	`benjk`	`a yacht is what you want`
`00:46.19`	`TripleFFFF`	`thanks for reminder all`
`00:46.41`	`*** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`00:48.46`	`justinu\|laptop`	`what is the diff between a sailboat and a yacht`
`00:49.46`	`benjk`	`a sailboat is a small boat with a sail`
`00:51.47`	`*** join/#asterisk Samoied (n=Samoied@201.21.232.88)`
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`01:02.31`	`justinu\|laptop`	`hyperion is kinda of a shitty yacht, where's the pool?`
`01:02.56`	`benjk`	`outside`
`01:03.21`	`benjk`	`in soviet russia, you don't carry a pool, the pool carries you`
`01:04.07`	`benjk`	`btw, Hyperion is powered by a cluster of 24 SGI servers`
`01:04.17`	`justinu\|laptop`	`lol`
`01:04.36`	`justinu\|laptop`	`i dunno if i'm ready to swim out with sharks and stuff`
`01:04.41`	`benjk`	`Jim Clarke had a ship management system designed for him`
`01:04.48`	`TripleFFFF`	`anyone seen this ? regisration from '<sip:1231231234@domeina;m;transport=UDP>' failed for`
`01:04.49`	`TripleFFFF`	`?`
`01:04.54`	`TripleFFFF`	`trasnport=udp ?`
`01:04.56`	`TripleFFFF`	`seesm weird`
`01:05.03`	`justinu\|laptop`	`it's part of the rfc`
`01:05.21`	`justinu\|laptop`	`it's a method for specifying protocol when the SRV records aren't setup, iirc`
`01:06.27`	`*** join/#asterisk tengulre (n=tengulre@222.90.66.4)`
`01:06.48`	`TripleFFFF`	`hmm`
`01:06.57`	`TripleFFFF`	`weird..`
`01:11.57`	`*** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw)`
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`01:17.29`	`*** mode/#asterisk [+o mog] by ChanServ`
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`01:20.25`	`nortex`	`Anybody here used SkyNet?`
`01:24.25`	`Juggie`	`The terminator did`
`01:24.54`	`*** join/#asterisk fholmes (n=fholmes@cpe-72-177-253-50.houston.res.rr.com)`
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`01:26.53`	`nortex`	`haha`
`01:27.06`	`nortex`	`Took me a minute to put that together :)`
`01:36.53`	`tengulre`	`hi,all`
`01:36.56`	`tengulre`	`anybody alive?`
`01:37.27`	`ManxPower`	`.part #asterisk-undesd`
`01:37.28`	`ManxPower`	`oops`
`01:37.34`	`ManxPower`	`.part #asterisk-undead`
`01:37.45`	`*** join/#asterisk lule (i=lule@host166.201-252-90.telecom.net.ar)`
`01:38.27`	`mog`	`heh`
`01:43.48`	`Iamnacho`	`lol`
`01:48.26`	`*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)`
`01:54.31`	`*** join/#asterisk shmaltz (n=mybox@mail.dmaven.com)`
`01:54.44`	`shmaltz`	`anybody here using hint?`
`01:55.22`	`[TK]D-Fender`	`yup`
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`01:56.30`	`shmaltz`	`D-Fender, if I have exten => 200,hint,Sip/200 and sip/200 makes a phone call, will it show up on the console (like a polycom sidecar)?`
`02:00.04`	`[TK]D-Fender`	`shmaltz : If things are configured right on the phone yes`
`02:00.39`	`shmaltz`	`[TK]D-Fender, lets take a very simple dialplan something like this:`
`02:00.49`	`[TK]D-Fender`	`shmaltz : And assuming you don't have SIP 2.0.1. on the phone and * 1.2.11 - 1.2.12`
`02:01.17`	`shmaltz`	`I'll be using 1.2.6 *, and sip 1.6.7`
`02:01.33`	`X-Rob`	`yay`
`02:01.38`	`X-Rob`	`lets hack his box!`
`02:01.45`	`X-Rob`	`and crash`
`02:01.52`	`X-Rob`	`hack, crash, hack, crash, this is fun.`
`02:01.53`	`shmaltz`	`[out]`
`02:01.55`	`shmaltz`	`exten => _X.,1,Dial(Zap/g1/${EXTEN})`
`02:02.04`	`shmaltz`	`X-Rob, it's not open to the interweb`
`02:02.37`	`[TK]D-Fender`	`shmaltz indavisable dialplan....`
`02:03.07`	`X-Rob`	`looks OK to me, what's your problem with it?`
`02:03.07`	`shmaltz`	`[TK]D-Fender, in the above context - which is for out, how do I make sure that when sip/200 dials 8005558355 that the presence is upadeted?`
`02:03.25`	`shmaltz`	`X-Rob, he is afraid you might be able to hack it :P`
`02:03.30`	`X-Rob`	`heh`
`02:04.04`	`[TK]D-Fender`	`shmaltz : just make sure that the hint is linked to whatever context the phones point to and that its enabled in your firmware.`
`02:04.27`	`shmaltz`	`[TK]D-Fender in egnlish please :(`
`02:04.44`	`[TK]D-Fender`	`shmaltz : That WAS english.....`
`02:05.02`	`shmaltz`	`I usualy have 2 contexts, 1 for internal like:`
`02:05.03`	`[TK]D-Fender`	`shmaltz : Gota "buddies" soft-key visible on your phone?`
`02:05.04`	`shmaltz`	`exten => 200,1,Dial(Sip/200)`
`02:05.35`	`shmaltz`	`and the other for out, like the one posted above, now I know for the internal I just do:`
`02:05.36`	`shmaltz`	`exten => 200,hint,sip/200`
`02:05.46`	`shmaltz`	`but how do I do it for out? like the one above?`
`02:05.56`	`[TK]D-Fender`	`shmaltz : yes, your hit format is fine`
`02:06.05`	`[TK]D-Fender`	`What do you mean for out?`
`02:06.22`	`shmaltz`	`how will the state be updated when sip/200 dials 8005558355?`
`02:06.32`	`shmaltz`	`which matches exten => _X.`
`02:07.00`	`[TK]D-Fender`	`shmaltz : You've got things a little mixed up. Presence tracks the DEVICE,, not that tart it is dialing.`
`02:07.34`	`[TK]D-Fender`	`shmaltz : So you would tak ext 201 and add 200 as a buddy. the moment 200 make a call to ANYWHERE he'll light up`
`02:07.34`	`shmaltz`	`[TK]D-Fender I know I'm mixed up, and at the moment I can't test it, thats why I'm asking :(`
`02:07.47`	`[TK]D-Fender`	`Why can't you test it?`
`02:07.58`	`shmaltz`	`just by adding hint,sip/200?`
`02:08.00`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)`
`02:08.16`	`shmaltz`	`because I'm missing a swithc to setup my lab for this one`
`02:08.26`	`shmaltz`	`tomorrow I'll test it`
`02:08.32`	`[TK]D-Fender`	`shmaltz : Yes. the hint just tells * that anyone subscribing to it that they want status updates the moment that device doe anything.`
`02:08.55`	`*** part/#asterisk lule (i=lule@host166.201-252-90.telecom.net.ar)`
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`02:09.16`	`Aces1Up`	`anyone here use Bandtel?`
`02:09.18`	`shmaltz`	`oh, so exten => 200,hint,sip/200 the first 200 is realy bogus, but needed so when the button is pressed it dials the right exten :P`
`02:09.22`	`shmaltz`	`I got it`
`02:09.26`	`shmaltz`	`thanks D-Fender`
`02:10.03`	`shmaltz`	`I was just confused about that, I thought that the 200 tells asterisk when anybody dials 200 then update the subscription states`
`02:10.07`	`shmaltz`	`stupid me`
`02:11.30`	`[TK]D-Fender`	`<shmaltz> oh, so exten => 200,hint,sip/200 the first 200 is realy bogus, but needed so when the button is pressed it dials the right exten :P <- no`
`02:11.51`	`shmaltz`	`:(, whats wrong?`
`02:12.00`	`[TK]D-Fender`	`the first 200 is the # the phone will ASK for. the sip/200 tells * which devices status to report back`
`02:12.22`	`shmaltz`	`D-Fender, yes but its bogus because it could be anything`
`02:12.25`	`[TK]D-Fender`	`so 202 will ASK to look at 200. * will look at SIP/200 and report back.`
`02:12.50`	`shmaltz`	`in fact it has nothing to do with exten => 200,1,WHatever`
`02:12.55`	`shmaltz`	`which was my mistake that it does`
`02:13.05`	`[TK]D-Fender`	`shmaltz : Anything can be anything in *. What if your sip device was names "sip/fred" instead? who says you need to use #'s for sip peer names in the first place?`
`02:13.37`	`shmaltz`	`D-Fender, I know that part, I'm just trying to explain where my mistake was`
`02:13.41`	`[TK]D-Fender`	`shmaltz : it indeed has nothing to do with actual dialplan... mind you ployoms do kinda "figure" that if you can watch someone, you can dial them too....`
`02:13.44`	`shmaltz`	`it doesn't matter now, I understand it`
`02:13.58`	`shmaltz`	`exactly`
`02:14.06`	`[TK]D-Fender`	`I think you've got it then.`
`02:14.10`	`[TK]D-Fender`	`good`
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`02:30.35`	`manalive`	`http://paste.uni.cc/10876 any help is appreciated, trying to get zaptel working on debian.`
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`02:58.30`	`drcode`	`I can use regular modem with ast to send fax?`
`03:05.25`	`*** join/#asterisk Newbie___ (n=me@211.24.146.11)`
`03:05.55`	`Newbie___`	`hi all, how do i get asterisk to play music on hold and simultaneous`
`03:05.57`	`Newbie___`	`ops`
`03:06.06`	`Newbie___`	`hi all, how do i get asterisk to play music on hold and simultaneously dial a number`
`03:06.35`	`Corydon76-home`	`option m`
`03:07.18`	`Newbie___`	`Corydon76-home: please show me an example`
`03:07.35`	`Corydon76-home`	`CLI> show application dial`
`03:08.53`	`Newbie___`	`Corydon76-home: thanks`
`03:14.04`	`Newbie___`	`Corydon76-home: option m in dial application will stop music when zap answered. and it takes another 8 sec for zap to call a cell phone. is it possible to play music for another 8 sec when zap is trying to connect to a cell ?`
`03:14.15`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`03:16.50`	`Corydon76-home`	`only if Zap is a digital line. If it's analog, then no.`
`03:17.02`	`kilobit2001`	`i get: Unable to create/find SIP channel for this INVITE`
`03:17.09`	`kilobit2001`	`anyone knows why?`
`03:17.30`	`Newbie___`	`Corydon76-home: my zap is analog, darn`
`03:18.57`	`pepepedo`	`when I use h323 chan my asterisk sometimes crash sudenly with no core dump , how can i force to asterisk create coredumps?`
`03:19.58`	`pepepedo`	`may someone help me pls?`
`03:20.41`	`Corydon76-home`	`ulimit -c unlimited`
`03:21.28`	`Corydon76-home`	`Note that Asterisk should also be compiled non-optimized or else the backtrace may be useless`
`03:21.45`	`pepepedo`	`great! thxs!`
`03:22.53`	`pepepedo`	`i'll try it`
`03:23.47`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`03:25.49`	`*** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com)`
`03:31.00`	`*** join/#asterisk BlepsoaF (n=pbaker@ool-4356d6c2.dyn.optonline.net)`
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`03:37.41`	`kronic`	`does anyone have any examples of using the manager api from withint a dial plan, bascially I want to retrieve all the agents for a particular queue`
`03:48.36`	`*** join/#asterisk rausosf (n=rausosf@c-69-181-224-186.hsd1.ca.comcast.net)`
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`03:51.49`	`tengulre`	`hi,all`
`03:52.17`	`tengulre`	`how to play the /var/lib/asterisk/moh/fpm-sunshine.wav when dialing in?`
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`03:57.06`	`heison`	`~seen JerJer`
`03:57.16`	`jbot`	`jerjer <n=jj@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #debian, 11h 24m 12s ago, saying: 'apt-file search lspci'.`
`03:57.16`	`tuck3r`	`tengulre: MusicOnHold?`
`03:57.23`	`*** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com)`
`04:05.34`	`kronic`	`how can I obtain the output of the QUEUE_MEMBER_LIST function in 1.1.12 (equivalent)`
`04:07.34`	`BlepsoaF`	`exten => s,1,Set(foo=${QUEUE_MEMBER_LIST(q1)})`
`04:07.46`	`BlepsoaF`	`that stores the output to foo`
`04:07.52`	`kronic`	`returns a function not registered in 1.1.12`
`04:09.11`	`Corydon76-home`	`In 1.2, that's QUEUEAGENTLIST`
`04:09.32`	`Corydon76-home`	`I think`
`04:09.36`	`kronic`	`arh.. cheers`
`04:09.37`	`kronic`	`I'll check`
`04:10.00`	`kilobit2001`	`<PROTECTED>`
`04:10.05`	`kilobit2001`	`anyone knows why?`
`04:10.34`	`BlepsoaF`	`kilobit2001: thats not a whole lot of info`
`04:10.35`	`kronic`	`Corydon76-home: nope`
`04:10.45`	`BlepsoaF`	`kronic: use AGI, or goto 1.4`
`04:13.31`	`Corydon76-home`	`should be QUEUE_MEMBER_LIST in 1.2`
`04:14.55`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net)`
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`04:34.04`	`xai`	`does a dual-homed * box work well with dedicated phone system? I want to use route all * traffic from outsie to the 3rd firewall interface, and then to a dual-nic * box. The other nics talk exclusevely to phones..`
`04:35.27`	`*** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue)`
`04:35.37`	`EyeCue`	`Can anyone summarise the ILBC license for me?`
`04:35.38`	`EyeCue`	`confused`
`04:35.52`	`kilobit2001`	`Unable to create/find SIP channel for this INVITE`
`04:35.57`	`kilobit2001`	`why do i get this?`
`04:36.06`	`*** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net)`
`04:57.34`	`*** part/#asterisk kilobit2001 (n=locid@206-248-160-247.dsl.teksavvy.com)`
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`05:14.55`	`*** join/#asterisk CT234 (n=c@203-59-82-197.dyn.iinet.net.au)`
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`05:16.36`	`CT234`	`Hi geniuses. I've just started out with Asterisk; read up on a zillion how-to's etc.. I'm running FreeBSD and Asterisk installed successfully.. but should I see it bound to UDP/5060 in 'netstat'?`
`05:16.52`	`EyeCue`	`For the iaxclient, hows does it deal with a codec that is preferred but no allowed`
`05:16.56`	`EyeCue`	`?`
`05:18.00`	`bsdfreak`	`CT234: yes, that's for SIP`
`05:18.16`	`CT234`	`it bound to tcp/2000 but not udp/5060`
`05:19.09`	`CT234`	`it didn't load the h232 module; would that stop it opening the SIP port?`
`05:20.41`	`CT234`	`..h323 [typo]`
`05:24.46`	`JT`	`CT234: what?`
`05:26.33`	`CT234`	`JT: i've installed Asterisk on FreeBSD successfully. Its bound to tcp/2000 [ie, I can connect to the CLI console]..`
`05:26.54`	`CT234`	`..but it hasn't opened the sip port [udp/5060]..`
`05:27.13`	`CT234`	`its a shockingly simple question, but how do i start the sip 'daemon'?`
`05:28.58`	`*** join/#asterisk AJaymn (n=boiwonde@24-159-236-181.dhcp.mdsn.wi.charter.com)`
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`05:33.20`	`CT234`	`turning on 'debug' in the CLI console, it appears that it IS accepting SIP requests, but trying to fwd them to 10.1.98.100 - an IP that I have never configured. [and I can't find any reference to that IP in the .conf's] ..ideas pls?`
`05:43.32`	`*** part/#asterisk AJaymn (n=boiwonde@24-159-236-181.dhcp.mdsn.wi.charter.com)`
`05:45.04`	`*** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)`
`05:45.22`	`kronic`	`does anyone have any examples or tips on returning agents associated with a queue (in the dialplan) in 1.2.12`
`05:46.03`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
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`05:51.24`	`*** mode/#asterisk [+o Corydon-w] by ChanServ`
`05:54.20`	`*** join/#asterisk VibroMax (n=phisto@83.229.70.154)`
`05:55.18`	`*** join/#asterisk kilobit2001 (n=locid@206-248-160-247.dsl.teksavvy.com)`
`05:55.23`	`kilobit2001`	`hello`
`05:56.43`	`*** join/#asterisk angler_ (i=angler@nat/digium/x-51a0f75122a76a5c)`
`05:58.28`	`*** join/#asterisk lorinc (n=ang@caracas-3355.adsl.interware.hu)`
`06:00.44`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`06:04.48`	`EyeCue`	`What does it mean by 'toll quality' speech ? when discussing codecs ?`
`06:05.34`	`JT`	`the voice quality is sufficiently to good that it can be charged for in a telco capacity`
`06:05.46`	`*** join/#asterisk sjsobol (n=sjsobol@amethyst.justthe.net)`
`06:05.53`	`sjsobol`	`'lo`
`06:08.23`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`06:11.51`	`JT`	`s/to//`
`06:12.05`	`*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)`
`06:13.16`	`*** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)`
`06:14.23`	`EyeCue`	`hmm, charged for`
`06:14.29`	`EyeCue`	`and bit-robust?`
`06:14.36`	`JT`	`bit-robust?`
`06:14.42`	`EyeCue`	`oh, degradation tolerance?`
`06:14.53`	`JT`	`what`
`06:14.55`	`JT`	`?`
`06:15.01`	`EyeCue`	`http://www.speex.org/comparison/`
`06:15.03`	`EyeCue`	`sok`
`06:15.03`	`JT`	`it means the voice doesn't sound like shit`
`06:15.04`	`EyeCue`	`:)`
`06:15.06`	`EyeCue`	`got it`
`06:15.09`	`JT`	`to put it simply`
`06:15.17`	`EyeCue`	`nah theres a reference to the term below the comparison`
`06:15.21`	`EyeCue`	`Robust to corruption at the bit level, as found on wireless networks`
`06:15.23`	`EyeCue`	`" "`
`06:19.49`	`*** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net)`
`06:20.42`	`*** join/#asterisk tengulre (n=tengulre@222.90.66.4)`
`06:20.44`	`tengulre`	`hi,all`
`06:21.08`	`tengulre`	`how to play the /var/lib/asterisk/moh/*.wav in extensions.conf?`
`06:22.33`	`tengulre`	`anybody alive??`
`06:22.36`	`tengulre`	`pls!`
`06:26.34`	`JT`	`EyeCue: that sounds like a diversion`
`06:26.53`	`JT`	`the real term "toll quality" refers to the perceived audio quality`
`06:27.04`	`tengulre`	`JT: are you answer me?`
`06:27.08`	`tengulre`	`;)`
`06:27.10`	`EyeCue`	`yah i know, it was answered :)`
`06:28.05`	`tengulre`	`I using Mp3Player(/var/........wav) to play these files, but failed! my asterisk 's version is 1.4.0.`
`06:37.07`	`kilobit2001`	`is there a way to test odbc connection in cli?`
`06:38.50`	`*** join/#asterisk VTD (n=td@adsl-68-92-197-123.dsl.hstntx.swbell.net)`
`06:40.55`	`*** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r)`
`06:51.11`	`JT`	`anyone know if the pinout for an ISDN2 BRI crossover cable is the same as a T1 crossover cable?`
`06:53.53`	`*** join/#asterisk af_ (n=af@ip-201-210.sn2.eutelia.it)`
`06:55.10`	`kaldemar`	`JT: don't think so. T1 uses pins 1245 and BRI (S interface) 3456.`
`06:55.19`	`qdk`	`JT: i guess not since that a T1 is like a ISDN24 link.`
`06:56.05`	`kaldemar`	`JT: 4 and 5 are one pair and 3 and 6 the other.`
`06:58.11`	`JT`	`kaldemar: ah true, i had all that info in front of me, it just didn't click :/`
`06:58.26`	`JT`	`qdk: that in itself wouldn't preclude it from being the same`
`06:58.43`	`JT`	`kaldemar: do you know which pair is rx, and which is tx, and the polarities?`
`06:59.54`	`*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)`
`07:02.38`	`*** join/#asterisk parag_ast (n=root@dxb-b112207.alshamil.net.ae)`
`07:02.51`	`parag_ast`	`sometimes I do in need to hangup my incomming calls on TDM400P myself...it dosn't hagup automatically....and when i soft hangup Zap/1-1 then i use to get this error "Call legs/trasaction dosn't exists"`
`07:03.16`	`kaldemar`	`JT: http://www.zytrax.com/tech/layer_1/cables/telephone.html`
`07:06.26`	`JT`	`thanks`
`07:07.57`	`*** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com)`
`07:18.07`	`*** join/#asterisk xnon (n=xnon@200.8.31.93)`
`07:18.33`	`xnon`	`hello`
`07:26.33`	`*** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net)`
`07:27.22`	`*** join/#asterisk oQPa (i=Ftv@144.Red-83-41-136.dynamicIP.rima-tde.net)`
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`07:31.14`	`*** mode/#asterisk [+o Corydon76-home] by ChanServ`
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`08:01.07`	`*** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)`
`08:01.30`	`*** join/#asterisk chat_jokey (n=chat_jok@202.63.175.78)`
`08:03.41`	`chat_jokey`	`hello`
`08:04.10`	`chat_jokey`	`can any one tell me .. if there is any easy way to record all the DTMF tones which are gettig passed on the asterisk channels .`
`08:04.21`	`chat_jokey`	`like Origination --> SIP .. and Destination --> IAX2`
`08:04.46`	`chat_jokey`	`so i wanna write some app which can just record thse DTMF and save it onto a file on disk ..`
`08:12.28`	`*** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr)`
`08:12.35`	`tparcina`	`good morning list`
`08:12.51`	`shellshark`	`list?`
`08:14.03`	`tparcina`	`ok, channel :))`
`08:14.10`	`shellshark`	`:)`
`08:14.16`	`shellshark`	`morning :)`
`08:14.21`	`tparcina`	`and, like I said, ti's morning, what do you expect? :))`
`08:14.52`	`*** join/#asterisk darkskiez (n=mbryars@195.11.205.216)`
`08:15.27`	`tparcina`	`two days ago I head problem installing BeroNet ISDN card. Yesterday I head problem configuring it. Today I have problem receiving a calls :))`
`08:15.37`	`shellshark`	`lol`
`08:15.41`	`shellshark`	`you are full of issues ;)`
`08:15.58`	`*** join/#asterisk Asterisk-User (n=Asterisk@port-83-236-235-195.static.qsc.de)`
`08:15.58`	`*** join/#asterisk darkskiez (n=mbryars@195.11.205.216)`
`08:15.58`	`tparcina`	`I can establish outgoing calls, but incoming cals don't work.`
`08:16.22`	`Asterisk-User`	`tparcina : just joined, what configuration ?`
`08:16.42`	`tparcina`	`shellshark: yes, defenetly seams ishues like me ;)`
`08:17.23`	`tparcina`	`I'm not so experienced, and when I install/configure/test something for the first time, I do get some troubles`
`08:18.04`	`tparcina`	`Asterisk-User: BeroNet ISDN card, outgoing calls work fine, but can't receive incoming call`
`08:18.31`	`Asterisk-User`	`BeroNet = hfc ?`
`08:19.25`	`*** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw)`
`08:19.38`	`Asterisk-User`	`bristuff, misdn, visdn ?`
`08:20.14`	`tparcina`	`Asterisk-User: I beleve beronet=hfc`
`08:20.19`	`tparcina`	`misdn`
`08:20.37`	`Asterisk-User`	`Got two HFC Pci cards here, Incoming said Cannot forward voice, outgoing everyone�s busy stuck on this with visdn`
`08:21.20`	`Asterisk-User`	`no experience with misdn... sorry`
`08:22.07`	`tparcina`	`everyone's busy - do you have L1 conectivity?`
`08:22.22`	`Asterisk-User`	`L1 ? what do u mean ?`
`08:23.18`	`Asterisk-User`	`Cards are in TE P2MP mOde`
`08:24.22`	`tparcina`	`I have asked do you have layer 1 connectivity. Are your lights blinking red or solid green?`
`08:24.48`	`Asterisk-User`	`No Leds on my Card :(`
`08:26.04`	`*** join/#asterisk darkskiez (n=mbryars@195.11.205.216)`
`08:26.35`	`Asterisk-User`	`also got an TDM400P with 2 FXS ports which works fine`
`08:27.22`	`Asterisk-User`	`Telephone connected to NTBA works fine an USB ISDN Interface also`
`08:27.37`	`Asterisk-User`	`but those hfc Cards in TE sucks...`
`08:28.14`	`chat_jokey`	`any one knows an easy way to record DTMFs that are being sent on already established channels ??`
`08:29.22`	`Asterisk-User`	`sniff your VoIP traffic and use some Etheral Plugin to extract the Frames to .wav`
`08:30.34`	`chat_jokey`	`Asterisk-User umm is there any ready AGI script or something ?`
`08:30.46`	`chat_jokey`	`because i cant code :P (not a coder)`
`08:31.03`	`Asterisk-User`	`you dont have to.. just use ethereal to capture packets`
`08:31.08`	`*** join/#asterisk Whisk (n=whisk@82-35-164-20.cable.ubr04.croy.blueyonder.co.uk)`
`08:31.16`	`Asterisk-User`	`but i just thought of an easier way`
`08:31.25`	`Asterisk-User`	`use an Sip SoftPhone`
`08:31.27`	`xnon`	`i have a some down calls in my asterisk server`
`08:31.28`	`Asterisk-User`	`like X-Lite`
`08:31.40`	`xnon`	`when a movil call to a extension in my asterisk`
`08:31.41`	`Asterisk-User`	`there�s a Record button while talking`
`08:32.22`	`*** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net)`
`08:34.13`	`tparcina`	`yes, I have solve it. The problem was that my ISDN provider has send me number in national format. I needed some time to figure this out.`
`08:34.30`	`Asterisk-User`	`how did you get it ?`
`08:35.10`	`tparcina`	`Asterisk-User: buy BeroNet card, it seams fine, but I havent use it in production jet`
`08:35.33`	`Asterisk-User`	`@tparcina Both cards are in TE, when dialing : Everyones busy....Auto fallthrough, channel`
`08:35.40`	`Asterisk-User`	`CHANUNAV`
`08:35.51`	`xnon`	`Asterisk-User, i have a asterisk server with some extensions, and all calls work good but when a movil call to my asterisk server and go to any extension, the call breakout!`
`08:36.01`	`tparcina`	`Asterisk-User: I have put X. in extension conf, so it matched all, and I put NoOp ({EXTENSION}) - so I got result in CLI :))`
`08:36.47`	`Asterisk-User`	`hmm i�m testing with _X. doesnt seem to work either`
`08:37.16`	`Asterisk-User`	`xnon : whats a movil ?`
`08:37.17`	`chat_jokey`	`Asterisk-User: the thing is .. i am running a service and i want to track some user ... as to what DTMF he is sending .. and recrod them ..`
`08:37.38`	`tparcina`	`Asterisk-User: you need to check why asterisk thinks that your channels are unvailable`
`08:37.46`	`Asterisk-User`	`chat_jokey : ah you mean which numbers`
`08:37.47`	`xnon`	`Asterisk-User, sorry my english is not so good`
`08:37.52`	`xnon`	`a movil phone`
`08:38.07`	`Asterisk-User`	`xnon : is it a Hard Phone ?`
`08:38.17`	`xnon`	`Asterisk-User, when a movil phone number Telefonica`
`08:38.37`	`xnon`	`yes fisical movil phone`
`08:38.40`	`xnon`	`PDA PHONE`
`08:38.46`	`xnon`	`Cellular Phone`
`08:38.58`	`Asterisk-User`	`ok got it :)`
`08:38.58`	`xnon`	`Any movil phone`
`08:39.04`	`xnon`	`jejeje ok`
`08:39.30`	`xnon`	`if call any global PSTN number to my asterisk ists so good`
`08:39.51`	`xnon`	`but when a movil called to my asterisk the call breakout`
`08:40.02`	`*** join/#asterisk bitter_vn (n=bitter_c@202.134.16.202)`
`08:40.05`	`xnon`	`alguien aca habla español?`
`08:40.13`	`xnon`	`siento que no me puedo hacer entender en ingles`
`08:40.28`	`Asterisk-User`	`xnon : sorry no idea`
`08:40.46`	`xnon`	`Asterisk-User, thanx anyway friend`
`08:42.16`	`bitter_vn`	`Anyone can help with reinvite?`
`08:44.06`	`bitter_vn`	`Hello`
`08:44.09`	`bitter_vn`	`Anyone can talk?`
`08:44.38`	`bitter_vn`	`I have posted on Asterisk-Users a topic about reinvite with wrong media address`
`08:44.43`	`bitter_vn`	`Any one can help?`
`08:48.19`	`Asterisk-User`	`no sorry`
`08:48.31`	`Asterisk-User`	`so many logged in users all idle :(`
`08:50.38`	`bitter_vn`	`Hello`
`08:50.40`	`bitter_vn`	`Any one can help`
`08:51.00`	`*** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net)`
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`09:13.49`	`*** join/#asterisk Tili (n=tili@202.133.65.54)`
`09:16.08`	`wasim`	`the Lord helps those who help themselves ...`
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`09:20.25`	`bitter_vn`	`Any one experienced with reinvite?`
`09:20.42`	`*** join/#asterisk angryuser (n=uk@i03v-213-44-169-43.d4.club-internet.fr)`
`09:20.48`	`shellshark`	`STOP REPEATING!`
`09:21.36`	`angryuser`	`~pb`
`09:21.39`	`jbot`	`hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste`
`09:55.08`	`*** join/#asterisk Ebola (n=Ebola@host86-138-123-50.range86-138.btcentralplus.com)`
`09:55.42`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`10:01.20`	`*** part/#asterisk ernie_ (i=jgeraert@193.202.9.42)`
`10:05.32`	`RoyK`	`morning`
`10:07.44`	`hads`	`Evening`
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`10:20.59`	`RoyK`	`whatevering`
`10:21.05`	`bsdfreak`	`hehing`
`10:23.06`	`*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)`
`10:27.21`	`festr__`	`RoyK: hello`
`10:32.37`	`*** join/#asterisk M_at (n=matt@lrfadsl01.demon.co.uk)`
`10:40.42`	`RoyK`	`festr__: hi`
`10:40.55`	`festr__`	`RoyK: see query`
`10:41.27`	`*** join/#asterisk DataCompBoy (n=datacomp@gw6.lvs.ru)`
`10:42.32`	`DataCompBoy`	`Hi everybody!`
`10:43.45`	`DataCompBoy`	`have anybody success with background (Async) playing of sounds from AGI?`
`10:47.14`	`M_at`	`Anyone got experience running within VMware in a production environment?`
`10:47.30`	`stugster`	`vmware sux`
`10:48.17`	`festr__`	`M_at: my friend has expirience with XEN virtualization`
`10:49.35`	`*** join/#asterisk inspired (n=mikael@85.221.7.59)`
`10:52.33`	`M_at`	`Trouble is we need reliable multi platform virtualisation`
`10:52.50`	`DataCompBoy`	`M_at: what multi-platform you need?`
`10:52.54`	`DataCompBoy`	`windows+linux?`
`10:53.08`	`DataCompBoy`	`Asterisk in vmware will give you choppy sound...`
`10:53.28`	`M_at`	`Yup`
`10:53.45`	`M_at`	`Running vmware on a dual dual-code opteron with 8Gb RAM ontop of Linux`
`10:53.57`	`M_at`	`Need to run both Windows and Linux guests`
`10:54.19`	`DataCompBoy`	`and every guest should have runned asterisk?`
`10:54.21`	`M_at`	`had hoped that * within a vmware guest wouldn't be too bad for some simple use`
`10:54.26`	`RoyK`	`festr__: I may start billing you quite soon`
`10:54.31`	`M_at`	`No - only one guest will be an * server`
`10:54.41`	`DataCompBoy`	`M_at: move asterisk to host.`
`10:55.00`	`M_at`	`Sort of defeats the object of virtualisation :)`
`10:55.04`	`RoyK`	`DataCompBoy: asterisk needs to run alone. forget vmware for everything except testing`
`10:55.09`	`*** part/#asterisk Samoied (n=Samoied@201.21.232.88)`
`10:55.17`	`DataCompBoy`	`If you need to be sure that asterisk doesn't affect anything on host, use linux-vserver for virtualization`
`10:55.35`	`DataCompBoy`	`RoyK: i have asterisk in linux-vserver, works excellent :)`
`10:55.43`	`festr__`	`RoyK: there is serious problem with PLC i though that you could be interested, so why billing`
`10:56.49`	`M_at`	`I'll try it on vmware while I'm the only one in this office with a VoIP phone but will move it to a dedicated blade later`
`10:57.09`	`*** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-166-122.columbus.res.rr.com)`
`10:58.19`	`DataCompBoy`	`M_at: you will have choppy sound even if the only runned on whole box will be that vmware with that asterisk.`
`10:58.39`	`DataCompBoy`	`Asterisk need to have precise time measurement, but in vmware it can't`
`10:59.32`	`*** join/#asterisk Barnes (n=barnes@c193-150-214-238.cm-upc.chello.se)`
`11:00.09`	`*** join/#asterisk oQPa (i=Ftv@144.Red-83-41-136.dynamicIP.rima-tde.net)`
`11:00.17`	`*** join/#asterisk backblue (n=igor@82.102.1.42)`
`11:01.05`	`*** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm)`
`11:01.05`	`stephane_`	`re`
`11:01.12`	`DataCompBoy`	`re`
`11:01.31`	`backblue`	`mi`
`11:02.55`	`M_at`	`DataCompBoy: Yup but for initial testing that's something I can live with - I don't have a budget for new hardware for this project`
`11:03.13`	`M_at`	`Need to re-use other physical machines as they get transfered to virtual ones.`
`11:03.27`	`DataCompBoy`	`M_at: develop on your workstation :) why not`
`11:03.44`	`DataCompBoy`	`M_at: I was started with asterisk4win for initial development`
`11:04.06`	`M_at`	`Because my workstation is slow enough without having people pass calls through it`
`11:06.08`	`DataCompBoy`	`M_at: then, if you don't worry about sound, you can run in in vmware.`
`11:06.18`	`DataCompBoy`	`it will work. in someway`
`11:06.21`	`*** part/#asterisk darkskiez (n=mbryars@195.11.205.216)`
`11:08.30`	`*** join/#asterisk soda_ (n=soda@www.dat.demokritos.gr)`
`11:09.46`	`soda_`	`is it possible to redirect incoming call to an internal sip extension`
`11:09.55`	`soda_`	`all calls`
`11:11.34`	`RoyK`	`festr__: it's not a serious problem with PLC. it's a configuration issue. if there were serious problems with the PLC, I'd seen it. we've sent hundreds of thousands of calls throught this and it works`
`11:12.00`	`kristalino`	`DataCompBoy, would we get the same choppy sound with xen ?`
`11:13.26`	`*** join/#asterisk Cyt (n=danielcy@122.Red-81-32-219.dynamicIP.rima-tde.net)`
`11:14.07`	`DataCompBoy`	`kristalino: nevery tried xen. only vmware and linux-vserver`
`11:14.18`	`Cyt`	`Hi, as soon as I dial out I'm getting this error. I tryied to google on it but I could not find an good answer. Does anybody knows what is this error? http://pastebin.ca/206330`
`11:14.37`	`*** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com)`
`11:14.42`	`DataCompBoy`	`kristalino: never tried xen. only vmware and linux-vserver. agh, also once tried virtuozzo -- under virtuozzo works fine`
`11:14.43`	`kristalino`	`DataCompBoy, and what about l-vserver ? Does it work better than vmware ?`
`11:15.26`	`DataCompBoy`	`kristalino: linux-vserver & virtuozzo only virtualize linux machines. all use one kernel (specially-patched host kernel), all other can be different.`
`11:16.51`	`kristalino`	`so maybe more precise about precise time measurement...`
`11:17.15`	`festr__`	`RoyK: check mail, you do not understand me again. i'm trying to explain it more clearly. i'm explaining why i think it is not configuration problem.`
`11:17.20`	`*** join/#asterisk ramtha (n=tk@195.14.234.162)`
`11:17.30`	`ramtha`	`hi`
`11:20.56`	`DataCompBoy`	`M_at: what happen? :)`
`11:22.03`	`RoyK`	`festr__: answered`
`11:27.20`	`*** join/#asterisk M_at (n=matt@lrfadsl01.demon.co.uk)`
`11:37.15`	`*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)`
`11:42.37`	`*** join/#asterisk jmls (n=asterisk@62.49.235.130)`
`11:44.17`	`*** join/#asterisk tparcina (n=tparcina@19-209.dsl.iskon.hr)`
`11:46.09`	`*** join/#asterisk xxxAGENTExxx (n=x@201.65.48.130)`
`11:46.27`	`xxxAGENTExxx`	`[brookshire]: what this problem? http://pastebin.ca/206342 ??`
`11:46.59`	`xxxAGENTExxx`	`WARNING[84615]: translate.c:88 powerof: Powerof 0: No power??`
`11:47.08`	`xxxAGENTExxx`	`translate.c:133 ast_translator_build_path: No translator path from gsm to unknown`
`11:51.57`	`DarKnesS_WolF`	`xxxAGENTExxx: seems codecs to me`
`11:52.19`	`DarKnesS_WolF`	`xxxAGENTExxx: to debug try to do this ... in ur sip.conf in [general] section do this disallow=all`
`11:52.22`	`DarKnesS_WolF`	`allow=ulaw`
`11:52.33`	`DarKnesS_WolF`	`the bothe ends supports ulaw ? or G.711U ?`
`11:52.38`	`xxxAGENTExxx`	`[DarKnesS_WolF]: ok testing now`
`11:52.39`	`xxxAGENTExxx`	`...`
`11:53.44`	`*** join/#asterisk fmueller (n=user@p548F6C8D.dip.t-dialin.net)`
`11:55.26`	`xxxAGENTExxx`	`DarKnesS_WolF thanks!`
`11:55.36`	`DarKnesS_WolF`	`xxxAGENTExxx: works ?`
`11:55.41`	`xxxAGENTExxx`	`yes`
`11:55.48`	`DarKnesS_WolF`	`ok great glad to help ...`
`11:55.53`	`DarKnesS_WolF`	`now i'll go home ;-)`
`11:55.55`	`DarKnesS_WolF`	`later guys`
`11:56.12`	`xxxAGENTExxx`	`ok see more later!`
`11:56.30`	`*** part/#asterisk hyphen (n=hyphen@71.224.213.97)`
`11:56.32`	`DarKnesS_WolF`	`sure always here ;-)`
`11:56.41`	`xxxAGENTExxx`	`:)`
`12:01.36`	`*** join/#asterisk ramtha (n=tk@195.14.234.162)`
`12:02.54`	`marexz`	`for totaly beginner which is better GUI for asterisk`
`12:04.53`	`*** join/#asterisk jbsolutios (n=jbenson@83.217.104.19)`
`12:05.54`	`jbsolutios`	`Hi everyone. When a call comes in to a queue using ringall, the periodic-announce is never played. Is this by design? Many thanks`
`12:05.58`	`ramtha`	`what should the guor you?i do f`
`12:06.17`	`ramtha`	`hmpf`
`12:06.24`	`ramtha`	`what should the gui do for you`
`12:07.51`	`*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)`
`12:09.14`	`marexz`	`configure pbx`
`12:09.19`	`*** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk)`
`12:09.26`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`12:09.50`	`*** join/#asterisk xezz (n=xez@serial.trust-it.ondsl.gr)`
`12:10.55`	`xezz`	`hi`
`12:11.55`	`ramtha`	`marez: you can configure 1001 part on your pbx, there is no gui, that supports all 1001 funktions`
`12:12.04`	`ramtha`	`so you must decide what you wan�t to do`
`12:12.18`	`ramtha`	`and THEN decide wich gui is the right for you`
`12:12.19`	`xezz`	`anybody knows about bristuffed versions of asterisk ? are they different from normal asterisk version ? cause when i try to update the 1.2.10-BRIstuffed ver to asterisk 1.2.12. asterisk could not start`
`12:12.26`	`ramtha`	`at the beginning try AMP`
`12:12.44`	`marexz`	`ramtha, gui not for me gui for user who not use linux and not use cli :)`
`12:14.46`	`ramtha`	`marexz: what ever, the gui must do a job, you must define this job`
`12:14.52`	`*** join/#asterisk gerhard7 (n=gerhard@82-171-119-13.dsl.ip.tiscali.nl)`
`12:15.40`	`ramtha`	`marexz: AMP does a good job for the most things...BUT there ist mostly no mysql support. guis write conf files`
`12:16.13`	`marexz`	`ramtha, ok thanks`
`12:21.27`	`*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)`
`12:21.38`	`JT`	`xezz: of course the bristuffed versions are different...`
`12:23.06`	`*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)`
`12:23.33`	`*** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home)`
`12:23.33`	`*** mode/#asterisk [+o Corydon-w] by ChanServ`
`12:25.32`	`RoyK`	`oej: ping`
`12:26.04`	`RoyK`	`or anyone, really, how can I add a SIP domain in Dial() ??`
`12:27.38`	`oej`	`RoyK Pong`
`12:28.03`	`oej`	`dial(SIP/royk@roykdomain.com)`
`12:28.15`	`xezz`	`JT: can u tell me the point of bristuffed versions ? and how am i going to update now ? everytime im going from a bristuffed ver to a standar (1.2.12 i.e) i get errors from the isdn card`
`12:28.19`	`[TK]D-Fender`	`RoyK: Dial(SIP/something@domain.com) ?`
`12:28.26`	`JT`	`xezz: the bristuffed version has changes made by klaus junghanns of junghanns.net`
`12:28.34`	`JT`	`mainly better support for isdn cards`
`12:28.39`	`JT`	`such as the ones he produces`
`12:28.43`	`JT`	`BRI cards, that is`
`12:28.55`	`benjk`	`unless you have [domain.com] in your sip.conf`
`12:28.58`	`xezz`	`jt : so im doomed to use the bristuffed ver once i have isdn card ?`
`12:29.19`	`JT`	`unlike misdn and the other isdn access methods available in asterisk, he has integrated the functionality with zaptel`
`12:29.33`	`JT`	`not made seperate commands just because it's isdn`
`12:29.34`	`RoyK`	`benjk: where?`
`12:29.48`	`xezz`	`so isdn has its own ver right ?`
`12:29.53`	`JT`	`xezz: if you want it to be consistent with zaptel, i guess so`
`12:30.00`	`JT`	`mm`
`12:30.15`	`JT`	`it's one implementation of BRI...`
`12:30.22`	`xezz`	`yes ok`
`12:30.24`	`xezz`	`got it`
`12:30.48`	`xezz`	`are they updating bristuffed vers ?`
`12:30.56`	`JT`	`umm`
`12:31.03`	`benjk`	`if you have a section [foobar.com] in your sip.conf defined, then if you dial SIP/jdoe@foobar.com it will target the host specified under [foobar.com] in sip.conf and not the actual domain foobar.com`
`12:31.18`	`xezz`	`i havent seen any tuto/mirror something for updating those vers`
`12:32.20`	`RoyK`	`oej: did you see my last email?`
`12:32.21`	`benjk`	`JT its actually not an implementation of BRI, it uses Libpri to avoid to have to use yet another ISDN iomplementation`
`12:32.40`	`JT`	`it's an implementation that allows you to use bri`
`12:32.58`	`JT`	`xezz: http://www.junghanns.net/downloads/`
`12:33.38`	`JT`	`0.2.0-RC8s is the latest stable`
`12:35.55`	`xezz`	`yes i got it`
`12:36.30`	`xezz`	`thanx for your help i had messed up bristuffed vers with standard vers`
`12:36.38`	`*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net)`
`12:36.59`	`benjk`	`xezz, you don't need anyting other than BRIstuff`
`12:37.31`	`benjk`	`it contains a script that will download the right asterisk for you, apply all the patches and then build/install`
`12:37.54`	`xezz`	`yes`
`12:38.00`	`xezz`	`thanx a lot`
`12:39.46`	`xezz`	`it has a script name install-ZAPHFC`
`12:40.06`	`xezz`	`wich configures the isdn hfc based card`
`12:40.42`	`xezz`	`and also downloads the latest asterisk wich is right for the system`
`12:40.47`	`benjk`	`make sure you red the readmes`
`12:41.50`	`xezz`	`so if i run that script at any time will download the latest patch right ?`
`12:42.31`	`xezz`	`i've run it now , the script was updated and get me to the latest bristuffed ver`
`12:42.50`	`[TK]D-Fender`	`benjk: It'll only taget that peer if the "host"-like name is a peer entry (why would you do this?) but then again you can just shove it in there as well as "host=domain.tld"`
`12:49.53`	`*** join/#asterisk ambriento (n=ambrient@201-27-81-64.dsl.telesp.net.br)`
`12:54.21`	`benjk`	`[TK]D-Fender, I am not saying you would want to do this, I am only saying that there is a scenario where you might think all is fine when you look at the dial statement, but its behaviour is altered by an entry in sip.conf, it could be entirely accidental, its something to be aware of so you can check if things don't go as you expect them`
`12:55.35`	`xxxAGENTExxx`	`ERROR[84615]: chan_sip.c:2258 update_call_counter: Call from user '107' rejected due to usage limit of 1`
`12:55.36`	`xxxAGENTExxx`	`Oct 17 14:48:55 NOTICE[84615]: chan_sip.c:10582 handle_request_invite: Failed to place call for user 107, too many calls`
`12:55.44`	`xxxAGENTExxx`	`whats this problem?`
`12:55.50`	`JT`	`hmm`
`12:55.56`	`[TK]D-Fender`	`benjk: TRhen again... think of the * admin who doesn't know what they've entered into sip.conf screwing up on something like that :) Would be kinda sad....`
`12:56.02`	`JT`	`benjk: just had a bristuff install fail`
`12:56.08`	`JT`	`at the zaptel part`
`12:56.09`	`JT`	`You do not appear to have the kernel sources for your current kernel installed.`
`12:56.13`	`JT`	`make: *** [linux26] Error 1`
`12:56.15`	`JT`	`You do not appear to have the kernel sources for your current kernel installed.`
`12:56.18`	`JT`	`make: *** [linux26] Error 1`
`12:56.21`	`JT`	`but i do have the sources..`
`12:56.44`	`[TK]D-Fender`	`JT: I'd be guessing "not where they are supposed to be" at best`
`12:56.51`	`benjk`	`do you have the link in your /usr/src dir?`
`12:57.31`	`benjk`	`[TK]D-Fender, I think ambiguous grammar is always a bad thing`
`12:58.02`	`JT`	`lrwxrwxrwx 1 root src 13 2006-10-17 16:08 linux -> linux-2.6.18/`
`12:58.02`	`JT`	`lrwxrwxrwx 1 root src 13 2006-10-18 08:50 linux-2.6 -> linux-2.6.18/`
`12:58.02`	`JT`	`drwxr-xr-x 20 root root 4096 2006-10-17 16:10 linux-2.6.18`
`12:58.06`	`JT`	`what more does it need`
`12:58.20`	`[TK]D-Fender`	`benjk: Talks does Yoda funny hmmmmmMMMMMMM!?>?!?!`
`12:59.08`	`benjk`	`[TK]D-Fender the SIP related dial argument in Dial has an ambiguous grammar`
`12:59.28`	`benjk`	`that is a bad thing, no matter what`
`12:59.43`	`xxxAGENTExxx`	`whats reason this problem?? : ERROR[84615]: chan_sip.c:2258 update_call_counter: Call from user '107' rejected due to usage limit of 1 \| NOTICE[84615]: chan_sip.c:10582 handle_request_invite: Failed to place call for user 107, too many calls`
`13:00.24`	`benjk`	`JT looks alright to me`
`13:00.35`	`JT`	`hrm`
`13:00.37`	`benjk`	`JT what distro?`
`13:00.42`	`JT`	`debian`
`13:00.56`	`DataCompBoy`	`ok, since there are skilled guys now, i'm try to ask again:) have anybody success with background (i mean -- Async, don't tangle with Background()) playing of sounds from AGI?`
`13:00.57`	`benjk`	`hmm, I am using BRIstuff stable on Ubuntu`
`13:01.06`	`[TK]D-Fender`	`benjk: argv[-1]`
`13:01.34`	`benjk`	`went like a charm, but I did have to upgrade the kernel to bring it in line`
`13:01.43`	`JT`	`hrm`
`13:02.13`	`[TK]D-Fender`	`xxxAGENTExxx: You clearly have a call limit of some kind for [107] 's entry in sip.conf`
`13:03.52`	`benjk`	`[TK]D-Fender, if you don't understand what a grammar is in the context of IT, I don't know waht to suggest to help you .... oh wait ... try some of Donald Knuth's books`
`13:04.49`	`benjk`	`JT, another thing that may help is rebuild the kernel deps`
`13:05.01`	`JT`	`? in what way?`
`13:05.08`	`JT`	`you don't make dep in kernel 2.6`
`13:05.18`	`*** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net)`
`13:06.08`	`benjk`	`when I ran into this, I asked for help in #debian or #ubuntu and somebody walked me trhough some reconfiguration of kernel dependencies information`
`13:06.19`	`tzanger`	`ok total brainfart... what's wrong with this?? GotoIf(${LEN(${CALLERID(num)})} != 3?gotmbox)`
`13:06.58`	`[TK]D-Fender`	`benjk: ...... </sarcasm> :) I understand full well.........`
`13:07.01`	`DataCompBoy`	`so, nobody =(`
`13:07.05`	`*** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35)`
`13:07.07`	`ramtha`	`is call limit not working for iax connections?`
`13:07.09`	`benjk`	`I don't recall exactly now what I was told to do but the background was that the installed kernel did not match some information about the kernel`
`13:07.13`	`tzanger`	`aha... $[`
`13:07.26`	`JT`	`ah gotchya, will look into it`
`13:07.37`	`Juggie`	`if you dont know what you doing, why the hell are you building your own kernels`
`13:07.55`	`[TK]D-Fender`	`GotoIf("${${LEN(${CALLERID(num)})}"!="3"?gotmbox)`
`13:07.56`	`benjk`	`Juggie I didn't build the kernel, I tried to build zaptel drivers`
`13:07.57`	`tzanger`	`Juggie: the same reason why people who know nothing about mechanics try to change their own oil`
`13:08.03`	`JT`	`Juggie: that's a bit rude`
`13:08.11`	`JT`	`are you saying i don't know what i'm doing?`
`13:08.25`	`Juggie`	`building zaptel should be as simple as making sure kernel-devel/kernel-smp-devel are installed`
`13:08.31`	`JT`	`i can rebuild an engine and i change my own oil :)`
`13:08.32`	`benjk`	`who can say they always know exactly what they are doing`
`13:08.34`	`Juggie`	`if it doesnt work then somethings broken with debian`
`13:08.35`	`[TK]D-Fender`	`GotoIf($["${LEN(${CALLERID(num)})}"!="3"]?gotmbox)`
`13:08.37`	`[TK]D-Fender`	`sadjklfhlkjasd`
`13:08.55`	`[TK]D-Fender`	`there you go`
`13:08.57`	`benjk`	`Juggie, in this case zaptel complains about a mismatch`
`13:09.00`	`Juggie`	`assuming of course, those installed dep's match the running kernel.`
`13:09.09`	`JT`	`and anyway an oil change is a simple operation and if done correctly requires minimal understanding of the internal combustion engine`
`13:09.17`	`benjk`	`it should be easy as make ; make install, but that is no guarantee that it always will be`
`13:09.22`	`tzanger`	`[TK]D-Fender: got it... $[${LEN(${CALLERID(num)})} != 3]`
`13:09.44`	`tzanger`	`hwoever now it's gotoif'ing correctly but voicemailmain is still asking for a mailbox number even though it's got 851@default\|a`
`13:09.53`	`Juggie`	`you need the kernel headers/source present which is commonly not installed by default.`
`13:09.59`	`[TK]D-Fender`	`tzanger: Considering how * freaks about empty args on sides of an expression I'd suggest "quoting" it...`
`13:10.02`	`benjk`	`he has got it installed`
`13:10.18`	`tzanger`	`[TK]D-Fender: true`
`13:10.19`	`[TK]D-Fender`	`tzanger: Pastebin the whole deal up`
`13:10.24`	`benjk`	`read the backlog before you comment`
`13:10.25`	`tzanger`	`[TK]D-Fender: no I got it... no mailbox 851 existed :-)`
`13:11.00`	`Juggie`	`is there a full compile log in a pastebin?`
`13:11.30`	`benjk`	`JT, I did have a fresh install at the time so my kernel wasn't up to date, thereofr the easiest to do for me was to upgrade the kernel and sources via apt-get`
`13:11.46`	`benjk`	`it was fine after that`
`13:12.10`	`Juggie`	`i can pretty much guarantee this is a case of just exactally what you said, the kernel compile not matching the source.`
`13:12.17`	`ramtha`	`hmm is there a way to limit iax calls?`
`13:12.49`	`JT`	`Juggie: why would it not match?`
`13:13.06`	`Juggie`	`where did you get the source from?`
`13:13.11`	`Juggie`	`via apt-get?`
`13:13.15`	`JT`	`it's probably some pointer somewhere with the wrong version`
`13:13.57`	`JT`	`kernel.org`
`13:13.57`	`Juggie`	`right`
`13:13.58`	`Juggie`	`so thats exactally why`
`13:13.58`	`JT`	`ok, where's the problem?`
`13:13.58`	`Juggie`	`the configs dont match`
`13:13.58`	`Juggie`	`there for, no love.`
`13:13.58`	`JT`	`where is the config you speak of?`
`13:13.58`	`benjk`	`JT you need to have the kernel and sources from the same exact source`
`13:14.08`	`benjk`	`or configure/build it yourself`
`13:14.13`	`Juggie`	`trash that directory and all your symbolic links`
`13:14.14`	`JT`	`benjk: the kernel running is the kernel i compiled`
`13:14.30`	`JT`	`with the correct ./config`
`13:14.48`	`Juggie`	`pastebin.ca the full compile log from zaptel.`
`13:14.55`	`benjk`	`well some information about the kernel config on your system is inconsistent`
`13:15.20`	`Juggie`	`it could even be as simple as running the kernel config after and saving changes`
`13:15.22`	`JT`	`Juggie: is that really necessary? it yielded little useful`
`13:15.25`	`Juggie`	`but not implementing them`
`13:15.27`	`Juggie`	`JT, yes.`
`13:15.41`	`Juggie`	`if you want my help, pastebin.ca the full zaptel compile log.`
`13:16.34`	`Juggie`	`compiling your own kernel was cool 5 years ago :) i did it too`
`13:16.51`	`Juggie`	`but, now its a million times easier to use the predone binary unless you have a specific reason not to`
`13:17.14`	`JT`	`the predone one in deb stable was 2.4`
`13:17.17`	`Juggie`	`just because the package management tools have come so far along.`
`13:17.20`	`*** join/#asterisk JunoD (n=Junodixo@216-188-237-188.dyn.grandenetworks.net)`
`13:17.30`	`JT`	`and i wanted to make sure some stuff specific to my ibm server hardware was compiled in`
`13:17.43`	`Juggie`	`well there you go :) let me see the compile log.`
`13:18.07`	`Juggie`	`until i see that, were just running in circles here.`
`13:19.28`	`*** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com)`
`13:19.30`	`Stephnie`	`hi`
`13:19.49`	`JunoD`	`whats up`
`13:20.12`	`Stephnie`	`how to keep the peers all the time available that are behind NAT?`
`13:20.16`	`Stephnie`	`I get this`
`13:20.17`	`Stephnie`	`Oct 17 05:18:52 NOTICE[17930]: chan_sip.c:11613 sip_poke_noanswer: Peer '9715055' is now UNREACHABLE! Last qualify: 276`
`13:20.21`	`JT`	`http://pastebin.ca/206537`
`13:20.35`	`Juggie`	`Stephnie, qualify=yes`
`13:21.00`	`*** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar)`
`13:21.01`	`Stephnie`	`Juggie: the msg I receive is due to qualify=yes`
`13:21.35`	`Juggie`	`hah, so it is, .. whats the nat device?`
`13:21.55`	`Juggie`	`JT, what version of zaptel is this?`
`13:22.05`	`JT`	`lemme check`
`13:22.14`	`JT`	`it's the one from the latest bristuff stable`
`13:22.15`	`Stephnie`	`Juggie: X-lite`
`13:22.17`	`benjk`	`JT, what's "ZAPTEL installed." about?`
`13:22.32`	`JT`	`benjk: i assume it's a script with poor error detection :P`
`13:22.39`	`Juggie`	`this looks like 1.0.x`
`13:22.42`	`benjk`	`yeah, that's what I suspect too`
`13:22.43`	`JunoD`	`does any one know if i can run http, ftp, and asterisk on the same box`
`13:22.52`	`Stephnie`	`Juggie: I have checked all the softphones...but same prb..Peer goes down and then re-register aftter a while..`
`13:22.54`	`benjk`	`which means you might actually have succeeded to build it`
`13:22.56`	`[TK]D-Fender`	`JunoD: Sure`
`13:23.02`	`JunoD`	`thanx`
`13:23.04`	`benjk`	`check if you have the kernel mods`
`13:23.27`	`Juggie`	`Stephnie, you can lower the registration time, i set mine to 60 seconds.`
`13:23.43`	`Juggie`	`none the less w/ qualify=yes and nat=yes theres not much else * can do to make nat work.`
`13:23.53`	`JT`	`zaptel 1.0.10`
`13:23.57`	`Juggie`	`eww.`
`13:24.20`	`Juggie`	`try bristuff-unstable or whatever its called`
`13:24.22`	`Juggie`	`it uses 1.2.x`
`13:24.32`	`benjk`	`bristuff unstable is rather troublesome`
`13:24.36`	`Stephnie`	`Juggie: that means...peer behind NAT will keep on UNREACHABLE and then REACHABLE :-/ ?`
`13:25.00`	`Juggie`	`Stephnie, it means that * lost the sip connection with the peer because the nat device closed the connection.`
`13:25.18`	`Juggie`	`which means the nat device is being too agressive closing connections usually.`
`13:25.23`	`*** join/#asterisk feld_ (n=feld@nat.ccwis.com)`
`13:25.44`	`feld_`	`....`
`13:25.48`	`Stephnie`	`hmmmm`
`13:26.01`	`feld_`	`so my boss and his friend decided to rip out my asterisk install this weekend because i wouldn't work for free and finish it`
`13:26.10`	`Juggie`	`what kind of nat device`
`13:26.12`	`feld_`	`all it needed was a fixed incoming call plan`
`13:26.31`	`feld_`	`they put in Asterisk@home, not even the newer trixbox`
`13:26.35`	`Stephnie`	`Juggie: X-lite`
`13:26.44`	`Juggie`	`Stephnie, thats not my question`
`13:26.49`	`Juggie`	`what is the NAT device`
`13:26.51`	`Juggie`	`not the client`
`13:27.02`	`benjk`	`JT, did you check if you have the kernel mods?`
`13:27.17`	`Stephnie`	`Juggie: oh....it's a SIMPLE CABLE Modem....`
`13:27.26`	`benjk`	`I say there is a good chance is succeeded and the messages are errant`
`13:27.51`	`JT`	`benjk: be more specific, you mean ccitt?`
`13:28.08`	`Stephnie`	`Juggie: cable modem also works as DHCP`
`13:28.44`	`Juggie`	`well, the cable modem is being too agressive w/ nat/udp it would seem.`
`13:29.00`	`benjk`	`JT I mean there is a chance your zaptel build succeeded and the script echoed those complaints in error`
`13:29.21`	`JT`	`hrm, but they were Errors not warnings`
`13:29.36`	`JT`	`they were echoed by make or cc`
`13:30.06`	`*** join/#asterisk flujan_ (n=flujan@internet.nube.com.br)`
`13:30.18`	`flujan_`	`hi guys... :)`
`13:30.18`	`benjk`	`well, it doesn't show that way on your pastebin`
`13:30.20`	`Juggie`	`benjk, no kernel modules are compiled in that log.`
`13:30.36`	`fmueller`	`hello, does astersik (1.0.9 :-)) do some kind of silence suppression on zaptel channels which are connected to meetme room?`
`13:30.44`	`JT`	`i've compiled enough software to know what a make/cc thrown error looks like :P`
`13:30.50`	`JT`	`worked out what the problem is i think`
`13:30.52`	`JT`	`fingers crossed`
`13:30.56`	`flujan_`	`I now that jitter the lower is the best... but.... Which is the better value to the jitter and jitterbuffer on a LAN network using U-LAW?`
`13:30.58`	`fmueller`	`i am asking because it sounds like asterisk is doing it but actually i do not want it ...`
`13:31.19`	`Juggie`	`* does not do silence supression of any kind.`
`13:31.26`	`benjk`	`JT, I am just not confident when I see output that says "failed" and "succeeded" at the same time`
`13:31.33`	`*** join/#asterisk ast_freak (n=jesse@h69-130-160-41.69-130.unk.tds.net)`
`13:31.42`	`JT`	`i can see the succeeeded error was likely in error`
`13:31.46`	`soda_`	`when i call from outside i get in the log "Checking SIP call limits for device xxxxxx`
`13:31.50`	`JT`	`succeeded message i mean`
`13:31.56`	`benjk`	`fair enough`
`13:31.57`	`soda_`	`but nothing rings, any idea?`
`13:32.02`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`13:32.13`	`Juggie`	`benjk, there is clearly nothing in the log which shows modules being compiled, the succeeded is likely from bristuff.`
`13:32.37`	`Juggie`	`bristuff install script perhaps?`
`13:32.48`	`benjk`	`Juggie, my point was I didn't trust the script becaue of the inconsistent messages`
`13:32.58`	`benjk`	`yeah`
`13:33.04`	`Juggie`	`i agree, but it looks like zaptel is being run frmo within another script.`
`13:33.06`	`fmueller`	`Juggie: ah, that's suprising to me since it really sounds like silence suppression. when i whisper into one phone i cannot hear the voice in the other.`
`13:33.34`	`Juggie`	`fmueller, what kind of phones, sip, tdm?`
`13:33.53`	`fmueller`	`Juggie: the phones are normal isdn phones ...`
`13:34.13`	`*** join/#asterisk riksta (n=rick@89.240.134.21)`
`13:34.19`	`Juggie`	`using bristuff?`
`13:34.37`	`fmueller`	`no, zaptel-1.0.9 ...`
`13:35.05`	`Juggie`	`the isdn phones are calling into * over a pri?`
`13:35.12`	`Juggie`	`give me a little more info`
`13:35.28`	`fmueller`	`there is a digium E1 card in the server.`
`13:36.07`	`Juggie`	`and the isdn phones are serviced by who, the telco or locally provided.`
`13:36.10`	`JT`	`/lib/modules/2.6.18/build and /lib/modules/2.6.18/source had incorrect symlinks to the kernel source`
`13:36.28`	`fmueller`	`the telco`
`13:36.37`	`benjk`	`like i said, something inconsistent ;)`
`13:36.38`	`Juggie`	`fmueller, you can try turning up the gain`
`13:36.48`	`JT`	`it became abundantly clear when i tried to compile the unstable bristuff, which as well as throwing the error and success messages, also threw the exact path of the reference`
`13:36.53`	`Juggie`	`in zaptel.conf i believe /etc/asterisk/zaptel.conf`
`13:37.05`	`Juggie`	`JT: i thought it would, that zaptel log was not as much information as i would have hoped.`
`13:37.14`	`fmueller`	`you mean one of rxgain / txgain?`
`13:37.20`	`benjk`	`JT, well it should be all straightforward from here on`
`13:37.20`	`Juggie`	`fmueller, yes.`
`13:37.33`	`JT`	`Juggie: yeah, i had even tried stracing it to little avail`
`13:37.43`	`Juggie`	`turn up your rxgain a little.`
`13:37.46`	`JT`	`benjk: heh, don't speak too soon, i just got another error :P`
`13:38.13`	`benjk`	`what's the error?`
`13:38.33`	`fmueller`	`Juggie: ok, great. thanx for the hint. does an "asterisk restart" suffice to activate the change?`
`13:38.41`	`Juggie`	`UMmMm....`
`13:39.00`	`Juggie`	`its in zaptel.conf right?`
`13:39.03`	`benjk`	`fmueller, not if it is zaptel.conf`
`13:39.11`	`fmueller`	`zapate.conf`
`13:39.14`	`*** join/#asterisk kattmat (n=kattmat@c213-100-64-47.swipnet.se)`
`13:39.17`	`kattmat`	`hello.`
`13:39.19`	`fmueller`	`zapata.conf`
`13:39.29`	`Juggie`	`ah, if its zapata then you'll need to stop asterisk`
`13:39.31`	`kattmat`	`anyone got a link to an first-time asterisk user tutorial?`
`13:39.32`	`Juggie`	`run ztcfg -vvv`
`13:39.34`	`Juggie`	`and then restart.`
`13:39.42`	`JT`	`http://pastebin.ca/206565`
`13:39.52`	`Juggie`	`zaptel.conf however, is just for chan_zap and would have only required a restart.`
`13:40.42`	`Juggie`	`class_device_create is a linux kernel function`
`13:40.59`	`benjk`	`JT, there was a change in some kernel APIs from 2.6.14 to 15`
`13:41.43`	`benjk`	`seems like you have older code which doesn't have the changes for the newer kernel APIs`
`13:42.08`	`Juggie`	`benjk, was zaptel 1.0.x even modified to support that?`
`13:42.21`	`benjk`	`there was some patch for this`
`13:42.44`	`benjk`	`its not so difficult to do`
`13:42.54`	`benjk`	`this function has a different parameter list`
`13:43.05`	`benjk`	`you can fix it by a single #define`
`13:43.39`	`fmueller`	`Juggie: thank you very much for your help. I will try to increase the gain.`
`13:43.43`	`Juggie`	`theres a zaptel 1.0.10 that may have the fix included.`
`13:43.50`	`Juggie`	`fmueller, np`
`13:43.52`	`JT`	`can you point out what it is, benjk? :)`
`13:43.53`	`benjk`	`strange thing is that I am using BRIstuff-stable on Ubuntu with a .18 kernel and I don't recall having had this problem`
`13:44.11`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.150)`
`13:44.16`	`benjk`	`in fact I recall that I had this problem with the pre-.15 kernel and I then had to upgrade to not run into this`
`13:44.29`	`Juggie`	`benjk, ubuntu may have rolled in a fix.`
`13:44.38`	`Juggie`	`or some sort of change, who knows`
`13:44.46`	`benjk`	`but I think the fix was for zaptel`
`13:45.05`	`benjk`	`maybe they added a compatibiliuty #define somewhere, yes`
`13:45.13`	`benjk`	`JT that function you see in your error log`
`13:45.25`	`benjk`	`search for places in your zaptel code where this fucntion is called`
`13:45.44`	`benjk`	error: too few arguments to function `class_device_create'
`13:46.01`	`benjk`	`you need to add a missing argument in the parameter list`
`13:46.04`	`benjk`	`I think NULL`
`13:46.30`	`benjk`	`check the kernel docs for this function and see what the new parameter list looks like`
`13:46.42`	`Juggie`	`heres the fix, http://lists.digium.com/pipermail/zaptel-commits/2005-December/000013.html`
`13:46.42`	`benjk`	`then you can add some macro like this`
`13:47.15`	`*** join/#asterisk zzxxcc (n=zzxxcc@219.140.34.169)`
`13:47.32`	`benjk`	`#define old(foo, bar) new(foo, bar, NULL)`
`13:47.39`	`benjk`	`along those lines`
`13:49.56`	`*** join/#asterisk flandero (n=theratbo@213-140-21-230.fastres.net)`
`13:49.57`	`flandero`	`hi there`
`13:50.10`	`Juggie`	`benjk, i bet this fix is probally commited to 1.0 trunk`
`13:50.37`	`*** join/#asterisk lorinc (n=ang@caracas-3355.adsl.interware.hu)`
`13:51.11`	`Juggie`	`right i am.`
`13:51.59`	`Juggie`	`JT, you could just download zaptel 1.0 trunk and convince bristuff to use that`
`13:52.08`	`Juggie`	`not much has changed since 1.0.10 just some bug fixes (like this one)`
`13:52.34`	`flandero`	`hi there, someone got goal with asterisk + skypho (italian voip provider) + NAT ? i'm really in trouble`
`13:52.39`	`benjk`	`JT, I can give you my zaptel.c`
`13:52.54`	`benjk`	`patched by bristuff and with those patches in`
`13:52.56`	`*** join/#asterisk ellisdee (n=ellisdee@69.15.174.114)`
`13:53.06`	`benjk`	`in case anything else fails`
`13:53.42`	`*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net)`
`13:54.16`	`JT`	`what version does it suit?`
`13:54.34`	`*** join/#asterisk M_at (n=matt@lrfadsl01.demon.co.uk)`
`13:57.12`	`sp0n9e-`	`for voicemail realtime, what's the customer_id for?`
`13:57.56`	`JT`	`benjk: ?`
`13:58.28`	`benjk`	`back`
`13:58.30`	`Juggie`	`JT, just use svn and download 'zaptel-old/trunk'`
`13:58.45`	`Juggie`	`rename that to 1.0.10 stick it where bristuff is looking for it, and you'll be off to the races.`
`13:58.47`	`benjk`	`JT, I used the same BRIstuff`
`13:59.03`	`JT`	`bristuff-0.2.0-RC8s`
`13:59.08`	`benjk`	`and I am using with with Ubuntu 6.06 kernel 2.6.18`
`13:59.20`	`iCEBrkr`	`<PROTECTED>`
`13:59.23`	`iCEBrkr`	`MAKE IT STOP!!!!`
`13:59.28`	`iCEBrkr`	`WTF is that shit all of a sudden`
`14:03.23`	`*** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr)`
`14:04.05`	`backblue`	`JT: dont use bristuff, use mISDN`
`14:05.43`	`kattmat`	`anyones using digisip and asterisk here?`
`14:06.36`	`JT`	`backblue: err, why?`
`14:09.47`	`backblue`	`JT: because it's badly designed, even worst than asterisk chan_sip.`
`14:10.23`	`benjk`	`the Junghanns drivers are actually quite good`
`14:10.34`	`benjk`	`Zaptel is of course another matter`
`14:11.49`	`*** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35)`
`14:12.29`	`RoyK`	`benjk: how do you DARE question the quality of Digium's code????`
`14:14.58`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`14:15.54`	`JT`	home/jon/bristuff-0.2.0-RC8s/zaptel-1.0.10/wcusb.c:1451: error: unknown field `owner' specified in initializer
`14:15.58`	`JT`	`/home/jon/bristuff-0.2.0-RC8s/zaptel-1.0.10/wcusb.c:1451: warning: initialization from incompatible pointer type`
`14:16.01`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`14:16.02`	`JT`	`make[2]: *** [/home/jon/bristuff-0.2.0-RC8s/zaptel-1.0.10/wcusb.o] Error 1`
`14:16.03`	`JT`	`make[1]: *** [_module_/home/jon/bristuff-0.2.0-RC8s/zaptel-1.0.10] Error 2`
`14:16.43`	`Juggie`	`you dont need wcusb so remove it from the makefile`
`14:18.53`	`bkw_`	`maybe he needs it`
`14:18.58`	`bkw_`	`:P`
`14:19.54`	`benjk`	`JT while you are at it, get rid of all the other wc crap too`
`14:20.07`	`benjk`	`wcfxo and co`
`14:20.20`	`bkw_`	`oh why not just nuke all of zaptel`
`14:20.31`	`benjk`	`because BRIstuff needs some of it`
`14:20.48`	`backblue`	`benjk: no they are not.`
`14:20.59`	`bkw_`	`not crap?`
`14:21.18`	`backblue`	`misdn it's far way better`
`14:21.25`	`benjk`	`Junghanns has more clue about BRI than most other players`
`14:21.26`	`JT`	`far way better`
`14:21.29`	`JT`	`like your english`
`14:21.35`	`backblue`	`you are wrong`
`14:21.36`	`benjk`	`for data, not for voice`
`14:21.41`	`JT`	`OK GOOD ARGUMENT`
`14:21.48`	`backblue`	`i only work with BRI cards`
`14:21.51`	`backblue`	`i have tryed all`
`14:21.55`	`backblue`	`i know what i'm saying.`
`14:21.55`	`JT`	`hrm, bristuff 0.3 compiles`
`14:21.57`	`JT`	`tempting :)`
`14:22.24`	`JT`	`sorry, i listen to structured reasoning, not blind handwaving, backblue`
`14:22.25`	`backblue`	`bristuff does not concert about some points in the rfc`
`14:22.44`	`JT`	`what rfc?`
`14:22.47`	`backblue`	`JT: and i dont care.`
`14:22.52`	`backblue`	`maybe isdn rfc?`
`14:22.53`	`JT`	`bri is not based on an rfc technology`
`14:22.55`	`JT`	`then don't comment`
`14:22.58`	`JT`	`you're an idiot`
`14:23.06`	`backblue`	`omfg`
`14:23.07`	`JT`	`bri is not an internet technology`
`14:23.08`	`backblue`	`LOL`
`14:23.11`	`benjk`	`ISDN is not defined by any RFCs`
`14:23.18`	`JT`	`it is NOT based on an RFC`
`14:23.19`	`benjk`	`its defined by ITU-T`
`14:23.30`	`RoyK`	`the ISDN RFC!`
`14:23.36`	`wasim`	`them silly documents with a . in them ...`
`14:23.38`	`JT`	`do telcomms in a tertiary institution then come back with some clue`
`14:23.38`	`backblue`	`it's just a way to speak`
`14:23.51`	`JT`	`i can say blueberries when i meant apples`
`14:23.57`	`JT`	`it's not exactly clear though`
`14:23.58`	`benjk`	`ITU-T Q.920, Q.921, Q.930, Q.931`
`14:24.11`	`RoyK`	`benjk: STFU and read RFC2324`
`14:24.20`	`*** join/#asterisk TexasJay (n=me@ns1.accu-com.com)`
`14:24.38`	`M_at`	`Cockney rhyming slang is a way to speak. Stating that something is defined by an RFC when it isnt is being inaccurate.`
`14:24.39`	`benjk`	`not unless you can recite those Q series documents without error`
`14:24.43`	`benjk`	`;)`
`14:25.16`	`RoyK`	`http://www.faqs.org/rfcs/rfc3057.html <-- ISDN Q.921-User Adaptation Layer`
`14:25.18`	`RoyK`	`:)`
`14:25.23`	`TexasJay`	`are 'periodic-announce' and 'periodic-announce-frequency' the correct options to set when I want to play a message to users waiting in a queue?`
`14:25.30`	`benjk`	`UA is a totally different thing`
`14:25.38`	`benjk`	`its is not even doing any ISDN`
`14:25.41`	`benjk`	`it is like ssh`
`14:25.45`	`RoyK`	`I'm just arguing :)`
`14:25.49`	`RoyK`	`you know`
`14:25.55`	`wasim`	`TexasJay: yes`
`14:25.55`	`RoyK`	`for the sake of the argument`
`14:25.57`	`RoyK`	`tamtitam`
`14:25.58`	`RoyK`	`:D`
`14:26.09`	`benjk`	`it establishes a remote connection to an API that sits on top of an ISDN stack and sends messages to it`
`14:26.32`	`backblue`	`what kind of moron, tells me i'm a moron/idiot, and cant compile bristuff? (it's a just or something?)`
`14:26.37`	`TexasJay`	`wasim: ok, what is the correct syntax for 'periodic-announce' options then? Do I need to include a full path to the file?`
`14:26.37`	`benjk`	`those messages have to be translated to ISDN messages by the adaptation layer API`
`14:27.12`	`benjk`	`in other words it is a protocol how to talk to a remote machine over IP and have that machine talk to ISDN`
`14:27.19`	`backblue`	`you guys, should listen, i'm only trying to help, i have lost so many time with bristuff, use misdn.`
`14:27.22`	`TexasJay`	`or, more directly, what path does 'periodic-announce' assume?`
`14:27.35`	`benjk`	`backblue exact opposite here`
`14:27.40`	`JT`	`backblue: what sort of moron says "look at the rfc" for a telco standard, after saying they know heaps about bri, but can't back up their recommentation with structured arguments, just "USE BLAH, IT'S HEAPS BETTER"`
`14:27.46`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`14:27.55`	`benjk`	`I have lost so much time with mISDN, BRIstuff works like a swiss clock for me`
`14:27.55`	`wasim`	`TexasJay: prolly /var/lib/asterisk/sounds`
`14:28.18`	`backblue`	`JT: because allmost of the ppl that comes here, just are using bristuff, because it was the first match.`
`14:28.29`	`TexasJay`	`wasim: I thought so too, but it's not playing the file I have there.`
`14:28.29`	`backblue`	`and i will try to send them directly to misdn`
`14:28.38`	`backblue`	`insteed of losing time with bristuff`
`14:28.41`	`TexasJay`	`So I tried a full path, no dice. Added the file extension, no dice.`
`14:28.42`	`bkw_`	`ok you know its pointless to sit here and call people morons and such.`
`14:28.50`	`bkw_`	`counter productive`
`14:28.58`	`JT`	`no, they mostly use bristuff because they've reviewed data froma number or sources and determined it to be the best fit for their needs`
`14:29.16`	`*** join/#asterisk ambriento (n=ambrient@201-27-81-64.dsl.telesp.net.br)`
`14:29.23`	`TexasJay`	`It doesn't even look like it's attempting to play it.`
`14:29.23`	`coppice`	`don't call people morons. just know it to be true in your heart :-)`
`14:29.39`	`*** join/#asterisk anthm (n=anthm@66.250.69.110)`
`14:29.39`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:29.39`	`sevard`	`Does anyone know much about the 2100hz detection / echo can disable in zaptel?`
`14:29.44`	`benjk`	`BRIstall was not the first match, CAPI was`
`14:29.55`	`bkw_`	`chan_crappy?`
`14:29.57`	`bkw_`	`haha`
`14:29.57`	`coppice`	`sevard: nope. nobody`
`14:29.58`	`*** join/#asterisk alkiser (n=email@bdsl.66.14.163.224.gte.net)`
`14:29.59`	`backblue`	`bristuff does not even concert about PTMP PTP in the right way`
`14:30.02`	`sevard`	`coppice: lame.`
`14:30.23`	`JT`	`concerts can be cool`
`14:30.33`	`sevard`	`Mine apparently stopped working`
`14:30.40`	`bkw_`	`sevard, um coppice knows more about this stuff than I think he cares to tell :P`
`14:30.50`	`bkw_`	`sevard, don't let him fool ya`
`14:30.52`	`TexasJay`	`Anyone else care to put their $0.02 in on my queue messages problem, please? :)`
`14:31.11`	`sevard`	`coppice: :) I used to see "echo can disabled on chan XX" in dmesg when I sent a fax and zaptel detected a 2100hz tone`
`14:31.15`	`*** join/#asterisk jarg (n=jarg@200.56.225.61)`
`14:31.18`	`sevard`	`I don't see that anymore....`
`14:31.39`	`JT`	`hrm so asterisk is meant to have a 1000hz compiled kernel? damn i set it to the default :(`
`14:31.40`	`sevard`	`It's 'not disabled' in zaptel.h which hopefully means 'not not enabled'`
`14:31.44`	`coppice`	`did you change the config? if not, they must have broken something`
`14:31.46`	`alkiser`	`would anyone have a suggestion on a good way to make a "pool" of zap trunks to use for making outbound calls`
`14:31.58`	`backblue`	`concern`
`14:32.01`	`backblue`	`bah, lame`
`14:32.11`	`backblue`	`damm kids`
`14:32.13`	`RoyK`	`wasim: nice guy?`
`14:32.24`	`JT`	`kids, you're the one talking like a kid to be honest, backblue`
`14:32.26`	`sevard`	`coppice: I didn't change the configs, I had made a backup of everything, and an HD died, I downloaded libpri, zaptel, and asterisk and installed on a new hdd`
`14:32.43`	`bkw_`	`coppice, then break things? NEVER!!!`
`14:32.49`	`bkw_`	`s/then/them/`
`14:32.50`	`backblue`	`JT: i'm writing the best english i know, you should not joke, and help me out.`
`14:33.00`	`backblue`	`because my english it's not the best`
`14:33.02`	`sevard`	`and IIRC there isn't anything in the configs about 2100hz detection`
`14:33.14`	`backblue`	`:)`
`14:33.55`	`TexasJay`	`Hi. Having a queue problem here. 'periodic-announce' isn't periodically announcing the file I want to my users in queue. Do I need to set a full path to the file? If not, what folder does it assume?`
`14:34.39`	`backblue`	`connect misdn to a nortel/matra pbx, with isdn, and check out what i'm saying.`
`14:34.43`	`sevard`	`coppice: any advice man? :)`
`14:36.23`	`backblue`	`bristuff does not follow the standart TEI defenicion in ITU isdn standards`
`14:36.30`	`*** join/#asterisk foRza (n=forza@firewall.hikt.no)`
`14:36.44`	`backblue`	`ptmp = variable tei, ptp = fixed tei.`
`14:37.14`	`TexasJay`	`Anyone?`
`14:37.35`	`backblue`	`well i dont know how it's bristuff now, but in March it was a joke.`
`14:37.43`	`*** part/#asterisk fmueller (n=user@p548F6C8D.dip.t-dialin.net)`
`14:38.20`	`*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)`
`14:39.59`	`*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)`
`14:40.34`	`TexasJay`	`Bueller?`
`14:41.20`	`sevard`	`Ferris?`
`14:41.54`	`Nivex`	`Wheel?`
`14:43.12`	`TexasJay`	`Hi. Having a queue problem here. 'periodic-announce' isn't periodically announcing the file I want to my users in queue. Do I need to set a full path to the file? If not, what folder does it assume?`
`14:43.49`	`sevard`	`Jesus man, you posted that a good 15 lines ago`
`14:44.02`	`sevard`	`How about you pastebin your conf`
`14:44.07`	`TexasJay`	`Yes I did, sevard :P`
`14:44.11`	`*** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty)`
`14:44.13`	`TexasJay`	`Hehe, shut up Whisk :P`
`14:44.33`	`sevard`	`TexasJay: I don't see a URL anywhere, bub.`
`14:44.40`	`spaghetty`	`hi`
`14:44.56`	`*** part/#asterisk anthm (n=anthm@66.250.69.110)`
`14:44.58`	`spaghetty`	`I'm new to asterisk (really new =))`
`14:45.08`	`ellisdee`	`how do i reload voicemail.cfg?`
`14:45.09`	`TexasJay`	`Didn't think I needed to make my conf public for that question :/`
`14:45.20`	`wasim`	`TexasJay: it should be /var/lib/asterisk/sounds`
`14:45.25`	`sevard`	`ellisdee: reload app_voicemail.so`
`14:45.31`	`wasim`	`TexasJay: its probably something else, and not the path of the file`
`14:45.34`	`TexasJay`	`But the relevant lines:`
`14:45.34`	`TexasJay`	`periodic-announce = _test_rec1`
`14:45.34`	`TexasJay`	`periodic-announce-frequency = 10`
`14:45.45`	`bkw_`	`the _`
`14:45.46`	`wasim`	`TexasJay: if you think its the file, replace the default with it`
`14:45.46`	`spaghetty`	`I don't know why now my sip channel results unreachable`
`14:45.48`	`bkw_`	`bet taht fucks it up`
`14:45.48`	`wasim`	`TexasJay: _`
`14:46.14`	`ellisdee`	`sevard, thank you`
`14:46.21`	`TexasJay`	`really? Asterisk has no problems with other files that start with the underscore.`
`14:46.32`	`sevard`	`TexasJay: Try it.`
`14:46.33`	`TexasJay`	`I'll change it quick and see if that fixes it. Thanks :)`
`14:46.45`	`sevard`	`and make sure user asterisk has read privs to the file`
`14:46.50`	`sevard`	`and the file is in the correct directory`
`14:46.51`	`*** join/#asterisk santiago (n=santiago@debian/developer/santiago)`
`14:46.57`	`sevard`	`and make sure you're reloading correctly.`
`14:47.07`	`wasim`	`i.e. Playback(file) and test it first too ...`
`14:47.42`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.170)`
`14:47.54`	`*** join/#asterisk edwar64896 (n=medwards@72.83.233.220.exetel.com.au)`
`14:47.58`	`edwar64896`	`hi`
`14:48.22`	`TexasJay`	`Yup, already ensured Asterisk could play it.`
`14:48.28`	`TexasJay`	`And no dice on removing the _`
`14:49.16`	`spaghetty`	`I've error in dial_exec Unable to crate channel of type 'SIP'`
`14:50.17`	`wasim`	`TexasJay: does it play the default file in periodic announce?`
`14:50.25`	`*** join/#asterisk toxap (i=toxap@194.187.128.88)`
`14:51.43`	`TexasJay`	`I don't believe I have a default file set to play.`
`14:51.58`	`*** join/#asterisk toxap (i=toxap@194.187.128.88)`
`14:52.36`	`TexasJay`	`I do not have a default setting. It's currently commented out. Let me uncomment it and see what happens.`
`14:52.42`	`wasim`	`TexasJay: try it ...`
`14:54.07`	`*** join/#asterisk toxap (i=toxap@194.187.128.88)`
`14:54.20`	`TexasJay`	`Drat. Nope. :(`
`14:54.49`	`sevard`	`you're putting it in queues.conf right?`
`14:54.56`	`TexasJay`	`correct`
`14:54.58`	`*** join/#asterisk toxap (n=toxap@213.227.193.75)`
`14:55.16`	`sevard`	`are you reloading correctly?`
`14:55.29`	`TexasJay`	`I have an included file however. The rest of the queue works fine.. Just not the periodic message.`
`14:55.33`	`TexasJay`	`I'm restarting asterisk`
`14:55.46`	`*** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net)`
`14:56.41`	`TexasJay`	`Lemme get a public file that has the pertinent info in it for you guys...`
`14:57.24`	`alkiser`	`anyone mind commenting on my dial plan for outbound calls.. is that the best way to make a "pool"?`
`14:57.24`	`alkiser`	`http://pastebin.ca/206691`
`14:59.11`	`*** join/#asterisk qdk_ (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)`
`15:01.32`	`TexasJay`	`http://www.ulpriorityone.com/queue.txt`
`15:01.39`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`15:01.54`	`spaghetty`	`could be a problem change port into iax.conf !!`
`15:02.01`	`TexasJay`	`I snipped a bunch of info. Hopefully none of it is related to my problem. If it is, let me know and I'll update it.`
`15:02.28`	`TexasJay`	`(other extensions and agents and such)`
`15:02.50`	`edwar64896`	`ls`
`15:03.02`	`RoyK`	`benjk: ping`
`15:03.05`	`spaghetty`	`seem that if i change from 5036 to 4569 sip channel can work`
`15:03.38`	`benjk`	`RoyK pong`
`15:05.05`	`edwar64896`	`alkiser: what is it you are trying to achieve?`
`15:05.52`	`*** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com)`
`15:06.34`	`alkiser`	`when making an outbound call to use the first available Zap, otherwise jump to the next until it find one available for dialing out`
`15:06.57`	`edwar64896`	`why don't you just use a group?`
`15:07.16`	`Cyt`	`[- OS Uptime: 4h 10m 3s -]`
`15:07.26`	`alkiser`	`im un familiar with groups, which conf can I do that it?`
`15:07.37`	`edwar64896`	`set up a group in zapata.conf using the 'group' command`
`15:07.42`	`edwar64896`	`hang on...`
`15:07.56`	`DarKnesS_WolF`	`alkiser: if SIP then sip.conf zaptel channel then zapata.conf`
`15:08.17`	`*** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se)`
`15:08.25`	`jeremy_g`	`tjena grabba`
`15:08.55`	`jeremy_g`	`is anybody in there?`
`15:08.58`	`jeremy_g`	`:)`
`15:09.06`	`edwar64896`	`alkiser: http://pastebin.ca/206713`
`15:09.42`	`edwar64896`	`alkiser: this gives me a group on 60 Zap channels.`
`15:09.44`	`*** join/#asterisk cbm11211 (n=Administ@66.28.182.170)`
`15:10.42`	`edwar64896`	`alkiser: to dial out, I simply use Dial(Zap/g1/${EXTEN})`
`15:10.50`	`edwar64896`	`and it will pick the first available channel from the group`
`15:11.22`	`TexasJay`	`Any input on my queue question?`
`15:11.58`	`*** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net)`
`15:12.07`	`[TK]D-Fender`	`TexasJay: Do you see it attempting to play it back?`
`15:12.21`	`TexasJay`	`No. That's what's confusing me the most.`
`15:15.18`	`TexasJay`	`I've updated the .txt file to show the console output`
`15:15.20`	`TexasJay`	`http://www.ulpriorityone.com/queue.txt`
`15:17.00`	`*** join/#asterisk shmattie (n=mms@66-193-183-130.static.twtelecom.net)`
`15:18.29`	`shmattie`	`I am attempting to get a Avaya 4610 mwi to work. With a sip debug I see the error "SIP/2.0 400 Invalid Subscription-State". Any suggestions?`
`15:18.51`	`[TK]D-Fender`	`TexasJay: Go into queue, and just sit there for 2 minutes.`
`15:19.55`	`DataCompBoy`	`Re all! is anyone have success with background playing/stopping of specific sounds from AGI application? like PlayTones/StopPlayTones but for sound files ?`
`15:23.15`	`sp0n9e-`	`i need help debugging some voicemail trouble`
`15:23.39`	`*** join/#asterisk oQPa (i=Ftv@144.Red-83-41-136.dynamicIP.rima-tde.net)`
`15:23.41`	`sp0n9e-`	`i've got an asterisk realtime voicemail table set up`
`15:24.06`	`*** join/#asterisk RoyK (n=roy@80.239.107.70)`
`15:24.10`	`sp0n9e-`	`but i can't authenticate with the voicemail using the information in the table`
`15:24.40`	`sp0n9e-`	`where can i look to see what i'm doing wrong? (i already have asterisk running very verbosely)`
`15:26.10`	`sevard`	`[TK]D-Fender: Do you know anything about zaptel's 2100hz detection + echo can disable?`
`15:28.03`	`De_Mon`	`mixmonitor lets me adjust the "heard" and "spoken" volumn. which one is the caller and which is the callee?`
`15:28.23`	`TexasJay`	`[TK]D-Fender: That worked. It played the periodic announcement after the first ringthru timed out. 30 sec to the first agent, and when they didn't answer, it played the announcement.`
`15:35.06`	`*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`15:35.48`	`hoobastooba`	`i reinstalled asterisk last night because of some problems i was having. now my music on hold will not work.`
`15:35.53`	`hoobastooba`	`here is the error i get:`
`15:35.55`	`hoobastooba`	`Oct 17 09:34:20 WARNING[20921]: file.c:512 ast_openstream_full: File /var/lib/asterisk/moh-native/fpm-sunshine does not exist in any format`
`15:36.11`	`hoobastooba`	`ls -l shows: -rwxrwxrwx 1 root root 2582496 Oct 17 09:30 fpm-sunshine.mp3`
`15:36.15`	`hoobastooba`	`any thoughts.`
`15:36.21`	`hoobastooba`	`asterisk is running as root`
`15:36.46`	`shmattie`	`did you reinstall the asterisk add-ons?`
`15:37.20`	`hoobastooba`	`no, just asterisk. is that required? i did reinstall asterisk-sounds this morning trying to fix this issue.`
`15:37.21`	`alkiser`	`edwar64896: that appears to have worked, thanks for the help`
`15:37.39`	`Qwell[]`	`hoobastooba: asterisk-addons includes format_mp3, which is needed to play mp3`
`15:37.52`	`shmattie`	`hoobastooba, I believe the mp3 player is part of asterisk-addons`
`15:37.53`	`hoobastooba`	`good to know... reinstalling now. thanks`
`15:37.58`	`shmattie`	`hoobastooba, looks like you got it`
`15:38.11`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.150)`
`15:40.00`	`hoobastooba`	`does asterisk have to be restarted for that to take effect or will simply doing a moh reload do the trick?`
`15:40.36`	`Qwell[]`	`hoobastooba: load format_mp3.so`
`15:40.59`	`hoobastooba`	`uh-oh: what does this mean: Oct 17 09:40:38 WARNING[19735]: config.c:499 process_text_line: parse error: No category context for line 1 of modules.conf`
`15:41.31`	`[TK]D-Fender`	`TexasJay: It only plays outside the dial timeout between agents. And if I had an announcement playing every 10 seconds I'd friggen KILL someone...`
`15:42.44`	`*** part/#asterisk jmls (n=asterisk@62.49.235.130)`
`15:42.50`	`*** join/#asterisk kannan (i=1001@61.8.147.164)`
`15:42.58`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`15:42.58`	`*** mode/#asterisk [+o mog] by ChanServ`
`15:43.02`	`TexasJay`	`lol I just have it set to 10 seconds for testing :P`
`15:44.25`	`TexasJay`	`Besides, since my dial timeout is 30 seconds, it actually plays every 40 sec. ;)`
`15:45.08`	`TripleFFFF`	`anyother site with mailing lists you know of ?`
`15:47.54`	`*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)`
`15:48.12`	`kattmat`	`my asterisk is connected ("Registered") with my sip service provider`
`15:48.31`	`kattmat`	`but when i try to call to it from a "normal" phone, it says the number currently is unavailable`
`15:49.07`	`TripleFFFF`	`kattmat that means the number not formated right`
`15:49.13`	`sp0n9e-`	`hrm vm_authenticate: Couldn't read username <<< anything obvious about that?`
`15:49.17`	`TripleFFFF`	`if senindg a 1 try without if not try adding one`
`15:49.30`	`TripleFFFF`	`full 10 digits .. easy one to test is 800 go fedex`
`15:49.36`	`TripleFFFF`	`and i WOULD HATE to get that bill ;)`
`15:49.37`	`kattmat`	`TripleFFFF: I'm calling from my cell phone to the normal phone number i got from my sip provider. i`
`15:49.46`	`TripleFFFF`	`ok`
`15:50.01`	`kattmat`	`if i test the same account in, for example, x-lite, it works just fine.`
`15:50.03`	`*** join/#asterisk JunoD (n=Junodixo@216-188-237-188.dyn.grandenetworks.net)`
`15:50.05`	`TripleFFFF`	`on your server do "sip debug peer INSERTPEERNAMEOFPROVIDERHERE`
`15:50.06`	`CunningPike`	`kattmat: Your CLI will tell you which number is being presented`
`15:50.30`	`TripleFFFF`	`then call and see if you see traffic.. .. you probably using trixshit.. try to enable anonymous inbound for now to debug`
`15:52.33`	`kattmat`	`ha! what the hell`
`15:52.35`	`kattmat`	`it worked now :)`
`15:53.19`	`kattmat`	`it says "Welcome" and hangs up, just like it should. lovely.`
`15:54.19`	`kattmat`	`how can i try it out without having to use my cellular phone all the time?`
`15:54.34`	`kattmat`	`can i call to it with a free sip phone or something?`
`15:54.43`	`TripleFFFF`	`yeah`
`15:55.04`	`kattmat`	`know of any i can use (os x) and how to use them to call to my asterisk server?`
`15:55.19`	`CunningPike`	`kattmat: SJPhonre`
`15:55.34`	`CunningPike`	`s/SJPhonre/SJPhone/`
`15:55.49`	`kattmat`	`gonna have a look, cheers guys`
`15:55.52`	`*** join/#asterisk lorinc (n=ang@caracas-3355.adsl.interware.hu)`
`15:57.10`	`*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net)`
`15:58.20`	*** join/#asterisk Lino` (n=Lino@i577BC72C.versanet.de)
`15:59.23`	`kattmat`	`ok, SJPhone is running. how to call my asterisk?`
`15:59.54`	Lino`	`hmmm`
`16:00.17`	`kattmat`	`hm`
`16:00.19`	Lino`	`is there a way to set the CID displayed when dialing out?`
`16:00.28`	`kattmat`	`tried calling 0366800010@proxy.digisip.net`
`16:00.38`	`kattmat`	`got "Number not available"`
`16:00.39`	`[TK]D-Fender`	Lino`: Yes.
`16:00.45`	Lino`	`how do i do it?`
`16:00.54`	Lino`	`this is a special case`
`16:01.01`	Lino`	`i dont want to change my CID, i know how to do that`
`16:01.10`	Lino`	`but eg when i dial 110 (police)`
`16:01.15`	Lino`	`i want the phone to display police`
`16:01.24`	Lino`	`i got a database, i got the lookup scripts`
`16:01.46`	Lino`	`when someone calls me, the lookup is done, everything is allright. but i want the lookup to work when i dial out`
`16:01.46`	Lino`	`:D`
`16:02.33`	Lino`	`the lookup done, that works fine. even when i dial out. the only thing i need is the command to set the name displayed on my phone`
`16:02.45`	Lino`	`(cisco 7960 via chan_sccp2)`
`16:03.28`	`[TK]D-Fender`	`Then You asked the wrong quesiotn. you wan CPID (Called Party ID), hence the name of the person you are dialing.`
`16:04.07`	`kattmat`	`ah, worked when i called the server through the local network`
`16:04.26`	`[TK]D-Fender`	Lino`: Few phones support this, and * doesn't yet. Cisco has the best odds of being able to do a lookup, but I don't know of any others that work yet. There is a Polycom patch in the works IIRC
`16:04.48`	Lino`	`aaah`
`16:04.58`	Lino`	`ok i'm from germany, so i don't know the exact english abbr.`
`16:05.33`	Lino`	`yeah, the lookup is working great, all i need is a way to display text on the phones display. the phone doesn't need to do anything ;)`
`16:06.10`	`groogs_`	Lino`: i know the linksys phones will display the phonebook name when you dial a number, so if you program it into the phonebook it should do it
`16:07.03`	Lino`	`oh that is not an option`
`16:07.21`	Lino`	`a) 7960 != linksys b) 7960 does not have an internal directory`
`16:07.31`	Lino`	`at least not the SCCP firmware`
`16:07.40`	`pif`	`app_voicemail complains: Unable to launch "/usr/sbin/sendmail -t"`
`16:07.54`	`pif`	`but /usr/bin/sendmail exists and is executable`
`16:07.56`	`alkiser`	`anyone know if there is a syntax file for textpad for asterisk configs`
`16:08.11`	`sp0n9e-`	`pif: you need to set the alias then`
`16:08.18`	`[TK]D-Fender`	`pif: Obvious question becomes "Who are you running * as?"`
`16:08.19`	`sp0n9e-`	`/usr/sbin isn't /usr/bin`
`16:08.36`	`[TK]D-Fender`	`that too :)`
`16:09.04`	`sp0n9e-`	`why doesn't it let me log in?`
`16:09.21`	`[TK]D-Fender`	`sp0n9e-: DETAILS please...`
`16:09.28`	`sp0n9e-`	`[TK]D-Fender: which ones do you need?`
`16:09.30`	`pif`	`[TK]D-Fender : as "asterisk" user`
`16:09.44`	`*** join/#asterisk ShadowHntr (i=sentinel@c-68-52-48-169.hsd1.tn.comcast.net)`
`16:09.57`	`sp0n9e-`	`pif: in the voicemail.conf set the sendmail path to /usr/bin not /usr/sbin`
`16:10.00`	`[TK]D-Fender`	`pif: So fix your alias issue and make sure "asterisk" has rights to sendmail`
`16:10.20`	`pif`	`what alias?`
`16:10.20`	`sp0n9e-`	`or just ln -s`
`16:10.36`	`sp0n9e-`	`pif: asterisk is looking for sendmail in /usr/sbin but it exists in /usr/bin`
`16:10.44`	`[TK]D-Fender`	`sp0n9e-: You have not told me ANYTHING about your problem. How the hell am I supposed to know what to ask for?`
`16:10.45`	`sp0n9e-`	`not the s before bin`
`16:10.59`	`[TK]D-Fender`	`sp0n9e-: And I might not need any .conf files posted depending on the question.`
`16:11.09`	`*** part/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net)`
`16:11.11`	`pif`	`no, /usr/sbin/sendmail exists and is executable by asterisk`
`16:11.47`	`sp0n9e-`	`[TK]D-Fender: i can leave a message in the voicemail, but when i invoke VoiceMailMain,@default i have trouble logging in...it says it doesn't recognize the username`
`16:12.19`	`[TK]D-Fender`	`sp0n9e-: Ok, pastebin your voicemail.conf and your dialplan that calls it.`
`16:12.27`	`sp0n9e-`	`pif: which isn't what you said earlier`
`16:12.42`	`sp0n9e-`	`[TK]D-Fender: table dumps of realtime alright?`
`16:14.49`	`*** join/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net)`
`16:14.54`	`DasTech`	`good morning`
`16:14.54`	`intralanman`	`anyone had problems using the LIMIT_PLAYAUDIO_CALLE[ER] vars with the dial command?`
`16:15.02`	`DasTech`	`anyone here using astbill ?`
`16:17.21`	`*** join/#asterisk Tili (n=tili@202.133.65.130)`
`16:18.03`	`*** part/#asterisk foRza (n=forza@firewall.hikt.no)`
`16:19.07`	`sp0n9e-`	`[TK]D-Fender: http://paste.devpipe.com/1151`
`16:21.24`	`[TK]D-Fender`	`sp0n9e-: Pastebin the CLI output of a call.`
`16:21.59`	`*** join/#asterisk Egonis (n=Egonis@CPE001346405687-CM00080da4da41.cpe.net.cable.rogers.com)`
`16:24.03`	`sp0n9e-`	`it seems it isn't even reading the password`
`16:24.13`	`sp0n9e-`	`Incorrect password '' for user 123`
`16:25.34`	`[TK]D-Fender`	`sp0n9e-: What I'm suspecting is you don't have your DTMF mode right at all....`
`16:27.16`	`sp0n9e-`	`the phone's Send DTMF is "in-audio"`
`16:27.16`	`sp0n9e-`	`should that be changed?`
`16:28.24`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`16:28.34`	`*** join/#asterisk fiber0pti (n=John@207.114.199.107)`
`16:28.37`	`sp0n9e-`	`aha`
`16:28.55`	`sp0n9e-`	`what is better to use? RFC2833 or SIP INFO?`
`16:28.59`	`Qwell[]`	`rfc`
`16:30.47`	`*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)`
`16:32.27`	`TrixVox`	`is it true that plainvoip is shutting down?`
`16:32.56`	`*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)`
`16:33.42`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`16:34.20`	`hmmhesays`	`so, I'm looking for a good termination provider to russia`
`16:34.53`	`*** join/#asterisk jmls (n=asterisk@62.49.235.130)`
`16:36.08`	`*** join/#asterisk roving_prole (n=Harper@72-254-127-111.client.stsn.net)`
`16:36.16`	`jmls`	`OT: (sortof) Does anyone know how the auto-merge scripts used by digium work ? If not, a lookin' I will go`
`16:36.27`	`Qwell[]`	`jmls: What do you need to know?`
`16:36.42`	`Qwell[]`	`and this would probably be better answered in -dev`
`16:37.04`	`jmls`	`ok. didn't want to get murdered in there ;)`
`16:38.56`	`*** join/#asterisk ambriento (n=ambrient@201-27-81-64.dsl.telesp.net.br)`
`16:38.58`	`[TK]D-Fender`	`hmmhesays: In Soviet Russia providers terminate YOU.`
`16:39.51`	`*** join/#asterisk oej (n=oej@apollo.webway.se)`
`16:40.46`	`hmmhesays`	`[TK]D-Fender: haha yeah`
`16:41.02`	`hmmhesays`	`voipjet and vitelity both suck with russian termination`
`16:41.07`	`hmmhesays`	`I was thinking about trying out gafachi`
`16:42.00`	`*** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl)`
`16:42.17`	`*** join/#asterisk dexteruk (n=charper@adsl-68-90-35-43.dsl.hstntx.swbell.net)`
`16:42.47`	`*** join/#asterisk lorinc (n=ang@caracas-3355.adsl.interware.hu)`
`16:43.36`	`dexteruk`	`I can an incomming call on sip:+number@ipaddress then i want to pass that to another switch using IAX but i want to pass the +number as the from Id and the caller ID`
`16:43.44`	`dexteruk`	`can anyone help?`
`16:46.39`	`dexteruk`	`I want to pass the origination and the caller id to another switch`
`16:47.20`	`*** part/#asterisk Egonis (n=Egonis@CPE001346405687-CM00080da4da41.cpe.net.cable.rogers.com)`
`16:48.41`	`dexteruk`	`How can i set the Origination ID?`
`16:50.26`	`*** join/#asterisk lorinc (n=ang@caracas-3355.adsl.interware.hu)`
`16:50.44`	`ManxPower`	`define Origination ID`
`16:51.16`	`ManxPower`	`dexteruk: what you want would be handled by a SIP proxy if you want to pass the various SIP stuff unchanged.`
`16:51.20`	`ManxPower`	`Asterisk is not a SIP proxy`
`16:52.09`	`*** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-0ff4bc843a76c38a)`
`16:52.51`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`16:54.04`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`16:54.18`	`dexteruk`	`I wanted to pass an incomming SIP call to another office and push that into a Queues based on its origination ID`
`16:55.06`	`*** join/#asterisk p1p (i=pip@mail.comp911.com)`
`16:55.21`	`p1p`	`Is it possible to kill a sip channel without restarting asterisk?`
`16:55.27`	Lino`	`origination id meaning what?`
`16:55.29`	Lino`	`caller id?`
`16:55.38`	Lino`	`called party id?`
`16:55.43`	`p1p`	`Ive got an open channel that doesnt seem to want to clear`
`16:55.44`	Lino`	`(see: i learn ;))`
`16:56.08`	`DaveCanoe`	`g729 question: when it says I have 2/2 encoders/decoders out of 30 ... does that mean I have 28 remaining calls of capacity or 13 remaining calls of capacity?`
`16:56.08`	`dexteruk`	`the origination id that i see on the sip header sip:+number@ipaddress`
`16:57.08`	`dexteruk`	`if i just pass the call you see the callers ID but not the Origination ID`
`16:58.08`	`dexteruk`	`i need this as i wanted to move call that came into a paticular number into a paticular queue`
`16:58.35`	`dexteruk`	`but still maintain the caller id for reporting`
`16:59.59`	`[TK]D-Fender`	`DaveCanoe: Depends on each side of the call.`
`17:00.03`	`*** join/#asterisk gerhard7 (n=gerhard@82-171-119-13.dsl.ip.tiscali.nl)`
`17:00.49`	`*** join/#asterisk boch (n=root@201.216.241.97)`
`17:01.17`	`p1p`	`[TK]D-Fender: Is there any way to destroy an open sip channel without killing asterisk?`
`17:01.28`	`DaveCanoe`	`well... in a 'round about way, I'm asking if each encoder and each decoder uses a licence ... or if a pair of them uses a licence.`
`17:01.36`	`[TK]D-Fender`	`p1p: "soft hangup [channel]"`
`17:01.41`	`dexteruk`	`ManxPower: so the sip:+number is what i want to pass and the Origination ID`
`17:01.42`	`boch`	`do you know why im getting -- Got SIP response 481 "Call/Transaction Does Not Exist" back from x.x.x.x every about 60sec, in the CLI ?`
`17:01.51`	`*** join/#asterisk kilobit2001 (n=locid@206-248-129-141.dsl.teksavvy.com)`
`17:01.55`	`kilobit2001`	`when does the ${exten} variable gets set?`
`17:02.06`	`[TK]D-Fender`	`DaveCanoe: More or less the pair. if you bought 30 licenses, then thats 30 transcoded calls roughly`
`17:02.34`	`[TK]D-Fender`	`kilobit2001: ${EXTEN} is wherever you are in your dialplan.`
`17:03.57`	`dexteruk`	Lino`: when i get passed a sip call i get passed sip:+number this is what i want to pass as the origination address
`17:04.10`	`dexteruk`	`origination id sorry`
`17:04.55`	`kilobit2001`	`how is the exten var set to the dialed number ?`
`17:06.10`	`*** join/#asterisk lorinc (n=ang@caracas-3355.adsl.interware.hu)`
`17:06.49`	`[TK]D-Fender`	`kilobit2001: its always set. When you dial a number, * dumps that call into the device's context on whatever pattern match fits and the exten is the exten. Plain and simple.`
`17:08.04`	`*** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net)`
`17:09.19`	`kilobit2001`	`for exten to be set, do i need to have a pattern _ right before?`
`17:09.26`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.150)`
`17:09.40`	`[TK]D-Fender`	`kilobit2001: For a pattern yes. fixed ones, no.`
`17:10.09`	`[TK]D-Fender`	`kilobit2001: And thats "yes" you need "_" for a pattern. not for the var to be set. The var is set by the fact you dialed.`
`17:11.40`	`kilobit2001`	`in my app, i do [default] exten => s,1,goto(${exten},s,1) -- jump to a context that matches the exten.`
`17:13.08`	`*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net)`
`17:13.30`	`[TK]D-Fender`	`kilobit2001: The exten IS "s".`
`17:14.03`	`*** part/#asterisk toxap (n=toxap@213.227.193.75)`
`17:14.52`	`kilobit2001`	`no, i use the value of var ${EXTEN} to goto a context with name=${EXTEN}`
`17:16.12`	`*** join/#asterisk justinu\|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)`
`17:16.30`	`*** join/#asterisk budairc (n=chatzill@200.215.57.174)`
`17:21.17`	`*** join/#asterisk eidolon (n=eido@pdpc/supporter/monthlybronze/eidolon)`
`17:21.33`	`*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)`
`17:22.08`	`*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)`
`17:24.16`	`*** join/#asterisk pifiu (n=someone@216.5.79.1)`
`17:26.10`	`*** join/#asterisk xxxAGENTExxx (n=x@201.65.48.130)`
`17:26.12`	`xxxAGENTExxx`	`bird*CLI> sip show registry`
`17:26.12`	`xxxAGENTExxx`	`No such command 'sip' (type 'help' for help)`
`17:26.18`	`xxxAGENTExxx`	`????`
`17:26.37`	`xxxAGENTExxx`	`Whats this?? problem??`
`17:27.05`	`sevard`	`OMFG`
`17:27.07`	`justinu\|laptop`	`sip module not loaded`
`17:27.10`	`sevard`	`YOU HAVE SO MANY X`
`17:27.52`	`*** join/#asterisk soylentgreen (n=fgast@nebukadnezar-em0.only640k.org)`
`17:28.31`	`DarKnesS_WolF`	`xxxAGENTExxx: load chan_sip.so`
`17:29.13`	`ManxPower`	`xxxAGENTExxx: are you using some asterisk gui like FreePBX/AMP/Asterisk@Home/Trixbox`
`17:31.07`	`*** join/#asterisk eliphas (n=user@BHE200150026016.res-com.wayinternet.com.br)`
`17:31.20`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)`
`17:31.27`	`*** join/#asterisk Gunde (n=spamyous@82.153.170.213)`
`17:32.45`	`xxxAGENTExxx`	`ok`
`17:32.50`	`*** join/#asterisk jmorgan (n=jack@static-71-117-242-5.ptldor.dsl-w.verizon.net)`
`17:34.12`	`[TK]D-Fender`	`kilobit2001: When you call that goto, the exten IS "s". tahts where you are, so thats what it is!`
`17:34.18`	`*** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net)`
`17:34.43`	`ManxPower`	`kilobit2001: 's' extension is NOT "match any"`
`17:35.34`	`pifiu`	`hey fender`
`17:37.17`	`pifiu`	`can someone help me installing *?`
`17:37.42`	`DarKnesS_WolF`	`pifiu: what is the problem ?`
`17:38.01`	`DarKnesS_WolF`	`[TK]D-Fender: u always say mysters to the guys :P i swear i didn't get what i mean with s is s ?`
`17:38.08`	`*** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net)`
`17:38.18`	`ManxPower`	`These days, I just shut my brain off and when I turn it back on Asterisk is installed.`
`17:38.25`	`[TK]D-Fender`	`DarKnesS_WolF: ... huh?!`
`17:38.29`	`pifiu`	`i downloaded *, the sound package and installed it`
`17:38.40`	`*** part/#asterisk jmls (n=asterisk@62.49.235.130)`
`17:38.40`	`pifiu`	`but when i type asterisk -vvvvvvvg or w/e it doesnt load`
`17:38.46`	`ManxPower`	`DarKnesS_WolF: what do you now undetsand about exten => s`
`17:38.46`	`pifiu`	`it deosnt recognize asterisk`
`17:38.54`	`pifiu`	`says command not found`
`17:39.11`	`ManxPower`	`pifiu: Asterisk is usually installed in /usr/sbin`
`17:39.12`	`Gunde`	`have you tried with the complete path?`
`17:39.22`	`pifiu`	`what would that be?`
`17:39.25`	`ManxPower`	`or "safe_asterisk"`
`17:39.28`	`pifiu`	`yeah both`
`17:39.31`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)`
`17:39.50`	`[TK]D-Fender`	`pifiu: Forget to do "make install" by any chance?`
`17:39.54`	`pifiu`	`i think it migt have something to do with the make config command i made for it to be able to start on startups`
`17:40.01`	`pifiu`	`lol no`
`17:40.03`	`ManxPower`	`[root@pbx-1 ~]# whereis asterisk`
`17:40.03`	`ManxPower`	`asterisk: /usr/sbin/asterisk /etc/asterisk /usr/lib/asterisk /usr/include/asterisk /usr/share/man/man8/asterisk.8`
`17:40.03`	`ManxPower`	`[root@pbx-1 ~]#`
`17:40.04`	`pifiu`	`but i have a question`
`17:40.18`	`pifiu`	`is there anyw ay to completely delete asterisk automatically? with a single command or something? maybe try from scratch again?`
`17:40.34`	`ManxPower`	`pifiu: no.`
`17:40.48`	`ManxPower`	`pifiu: Very few programs provide that feature.`
`17:40.51`	`pifiu`	`and if i re-install it or re-make it, will it make for example "etc/asterisk2/?`
`17:40.56`	`pifiu`	`or will it just replace the original?`
`17:40.59`	`ManxPower`	`pifiu: no.`
`17:41.08`	`[TK]D-Fender`	`pifiu: It will overwrite where appropriate`
`17:41.08`	`ManxPower`	`it will overwrite the original by default`
`17:41.11`	`DarKnesS_WolF`	`ManxPower: taht S is teh start exten`
`17:41.15`	`Gunde`	`pifiu: nope, but you could check during installation where it was installed, and then manually do that or write your own script. Been there, done that.`
`17:41.25`	`pifiu`	`it seems like its in /etc/asterisk`
`17:41.34`	`ManxPower`	`This is rocket science, this is standard unix admin stuff.`
`17:41.50`	`ManxPower`	`pifiu: Asterisk config files are in /etc/asterisk by defauilt. The binary is in /usr/sbin by default`
`17:41.56`	`pifiu`	`when i type whereis asterisk`
`17:41.58`	`[TK]D-Fender`	`ManxPower: You go Marjor Tom!`
`17:42.00`	`pifiu`	`i get this`
`17:42.00`	`pifiu`	`asterisk: /usr/src/asterisk-1.2.12.1/asterisk.sgml /usr/src/asterisk-1.2.12.1/asterisk.c /usr/src/asterisk-1.2.12.1/asterisk.8 /usr/src/asterisk-1.2.12.1/asterisk.o /usr/src/asterisk-1.2.12.1/asterisk /etc/asterisk`
`17:42.00`	`pifiu`	`root@unisoft [/usr/src]#`
`17:42.10`	`DarKnesS_WolF`	`pifiu: make uninstall on the src dire`
`17:42.12`	`DarKnesS_WolF`	`dir*`
`17:42.22`	`ManxPower`	`DarKnesS_WolF: when was THAT added?`
`17:42.32`	`ManxPower`	`pifiu: "make install" again`
`17:42.39`	`[TK]D-Fender`	`pifiu: Sure LOOKS like you didn't do "make install". Go do it again`
`17:42.47`	`pifiu`	`ok`
`17:42.50`	`pifiu`	`i did it twice fender`
`17:43.00`	`pifiu`	`but i ran "make config" after since its on a redhat box so it could start up again`
`17:43.03`	`pifiu`	`with init.d`
`17:43.05`	`ManxPower`	`pifiu: well look at the output if make install`
`17:43.16`	`pifiu`	`ok so darkness how do i make uninstall?`
`17:45.01`	`ManxPower`	`pifiu: you type make uninstall`
`17:45.16`	`pifiu`	`inside the download folder of asterisk?`
`17:45.42`	`jmorgan`	`i'm not able to play background files for my IVR this morning. I'm getting WARNING[4572] file.c: File custom/asterisk-prospectus_IVR-main-day does not exist in any format errors. has anyone seen this before?`
`17:45.53`	`pifiu`	says *** No rule to make target `uninstall'. Stop.
`17:47.10`	`DarKnesS_WolF`	`ManxPower: i didn't get what u mean ?`
`17:48.24`	`*** join/#asterisk ccesario (n=ccesario@matriz.isic.com.br)`
`17:50.27`	`kattmat`	`ok, so i've got my asterisk answering to calls and saying "Welcome"`
`17:50.39`	`kattmat`	`i want it to go straight to the voicemail`
`17:50.57`	`kattmat`	`how do i create a voicemail?`
`17:51.24`	`*** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk)`
`17:51.28`	`ManxPower`	`kattmat: first you read The Book`
`17:51.29`	`ManxPower`	`~book`
`17:51.31`	`jbot`	`somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`17:51.33`	`ManxPower`	`~thebook`
`17:51.35`	`jbot`	`i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`17:51.42`	`sevard`	`really really need to read the book.`
`17:51.43`	`kattmat`	`thanks`
`17:52.38`	`*** join/#asterisk r0d3nt\|m (n=RatMan@foster.stonedcoder.org)`
`17:55.23`	`*** join/#asterisk iCallAndy (n=iCallAnd@cpe-74-72-215-143.nyc.res.rr.com)`
`17:57.46`	`kattmat`	`that book is a bit overkill for my home setup`
`17:59.00`	`[TK]D-Fender`	`kattmat: No its not. If you don't understand the basics and don't want to do the bit of fluff reading that the book lays out for you nicccce & neat, then pay a consultant.`
`17:59.27`	`*** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net)`
`17:59.56`	`ManxPower`	`kattmat: PBXs are COMPLEX.`
`18:00.03`	`ManxPower`	`Even a home system is complex`
`18:01.26`	`*** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu)`
`18:02.27`	`sp0n9e-`	`if i hang up during certain parts of a voicemail call, i get "Ouch ... error while writing audio data: : Broken pipe\nBus error"`
`18:03.11`	`pifiu`	`the book is nice, but i setup * using the book, and look at where im at`
`18:03.36`	`ManxPower`	`sp0n9e-: That is correct.`
`18:03.52`	`sp0n9e-`	`well, that's a very bad thing.`
`18:03.58`	`sp0n9e-`	`why does it fail so "hard"`
`18:04.36`	`sp0n9e-`	`is there anything i can do to mitigate?`
`18:04.48`	`eidolon`	`this is totally pie in the sky chattering here - but has anyone considered using VOIP over wifi clients on PDAs and handhelds as a sort of commercial radio system? ala, the radio connects up to the WIFI network and stays connected, then the client is switched into a sort of simplex mode... listening to multiple chatterers. I don't think a normal 'conference room' model would work - you don't want ambient noise, and don't want "we'`
`18:04.48`	`eidolon`	`ll hear you unless you mute". want a keyed chat mechanism. with good enough wifi coverage, is this possible, say, in a commercial installation?`
`18:06.01`	`[TK]D-Fender`	`eidolon: Sounds like a simple MeetMe conference.`
`18:07.32`	`DarKnesS_WolF`	`eidolon: meetme == conferece app. in *`
`18:07.55`	`eidolon`	`yah, i'm reading on the voip-info.org wiki.`
`18:08.41`	`jedaustin`	`I've been pulling my hair out trying to resolve zaptel issues (TDM400P) with Centos 4.4, kernel 2.6.9-42.0.3.ELsmp, newest stable asterisk/libpri/zaptel. I get low volume on calls that connect zap channels together (one side can barely hear the other). When I monkey with rxgain/txgain it does get louder but the audio is distorted too. Anyone fixed this issue?`
`18:08.47`	`sp0n9e-`	`[TK]D-Fender: thanks for the help earlier`
`18:08.53`	`eidolon`	`welll. that's close. that seems like a more traditional conference tool though`
`18:09.11`	`eidolon`	`i'm looking for "the room is silent until someonekeys to talk" situations.`
`18:09.22`	`FuriousGeorge`	`jedaustin: zapata.conf txgain and rcgain`
`18:09.27`	`FuriousGeorge`	`check the example file`
`18:09.39`	`FuriousGeorge`	`*rxgain`
`18:09.59`	`eidolon`	`xref: similar to how teamspeak works.`
`18:10.27`	`eidolon`	`(gr. back in a few)`
`18:11.34`	`jedaustin`	`I think my audio issue is because one line is the DSL line. fxotune has one line at 5 and the other at 14 in fxotune.conf`
`18:13.42`	`jedaustin`	`Anyone dealth with that sort of issue where adjusting rxgain up so that you can hear causes the audio quality to degrade significantly?`
`18:13.50`	`FuriousGeorge`	`ive never needed to use fxotune, so im not quite sure what to tell you about that`
`18:14.12`	`FuriousGeorge`	`the louder it is the greater the chance of echo, but you only need to nudge it just a little`
`18:14.23`	`FuriousGeorge`	`then test`
`18:14.38`	`kilobit2001`	`is a sip user dial 123 -- does the ${exten} automatically set to 123?`
`18:14.42`	`jedaustin`	`I wish I didn't. I consistantly get 'I can't hear you!' when people call me and it bridges to dial out to a pstn number`
`18:14.45`	`*** join/#asterisk eidoLaptop (n=eido@pdpc/supporter/monthlybronze/eidolon)`
`18:14.52`	`pifiu`	`ok i ran the make and make install again and still nothing`
`18:15.10`	`FuriousGeorge`	`jedaustin: how much are you increasing the gain by`
`18:15.21`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`18:15.21`	`*** mode/#asterisk [+o mog] by ChanServ`
`18:15.43`	`SplasPood`	`NICE! Beta of eyeBeam for X86 macs!`
`18:16.22`	`jedaustin`	`FuriousGeorge:I've tried everything from 0.5 to 17. about 8 loud enough but by then the audio quality is horrible`
`18:16.52`	`jedaustin`	`FuriousGeorge: if txgain is too high then I get the 'I'm sorry that's not a valid extension" issue.`
`18:17.49`	`*** join/#asterisk marta (n=a@host238-200-dynamic.17-87-r.retail.telecomitalia.it)`
`18:17.56`	`*** join/#asterisk [Outcast] (n=bill@222-154-84-190.jetstream.xtra.co.nz)`
`18:18.00`	`FuriousGeorge`	`who did the wiring at the location`
`18:18.18`	`jedaustin`	`FuriousGeorge: It's been redone about 4 times. I did it the last time`
`18:18.39`	`jedaustin`	`FuriousGeorge: cat5 straight to pole with no breaks now.`
`18:18.51`	`FuriousGeorge`	`also, ive noticed issues like echo at the beginning of calls that goes away can be solved by upgrading a cpu and motherboard`
`18:19.15`	`FuriousGeorge`	`also cat /proc/interrupts`
`18:19.24`	`jedaustin`	`FuriousGeorge: I did that too :) Now dual core Pentium M board. Didn't help.`
`18:19.24`	`FuriousGeorge`	`is your tdm sharing an irq with something`
`18:20.07`	`eidoLaptop`	`eww. teamspeak is closed source. no documented api or anything. ew.`
`18:20.15`	`eidoLaptop`	`okay, so hacking up a client for teamspeak isn't an option`
`18:20.37`	`pifiu`	`how do i know if i get errors during the doing of 'make' in an * install? how do i know whats normal, and whats not?`
`18:20.55`	`FuriousGeorge`	`pifiu: if it fails you will know`
`18:20.56`	`heison`	`anyone here using the X option in meetme?`
`18:21.09`	`heison`	`~seen moc`
`18:21.24`	`jbot`	`moc <n=mochouin@modemcable173.101-70-69.mc.videotron.ca> was last seen on IRC in channel #asterisk, 371d 13h 25m 21s ago, saying: 'gota check it'.`
`18:21.24`	`pifiu`	`how so furious?`
`18:21.25`	`pifiu`	`i mean it doesnt start when i do asterisk -vvvvvc`
`18:21.34`	`pifiu`	`it says unknown command`
`18:21.53`	`FuriousGeorge`	`pifiu: it will say someting like "build failed" "post topmost error when blah blah blah"`
`18:21.57`	`FuriousGeorge`	`it will be obvious`
`18:22.01`	`*** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca)`
`18:22.20`	`FuriousGeorge`	`pifiu: make install?`
`18:22.23`	`FuriousGeorge`	`did you do that`
`18:22.36`	`pifiu`	`yes`
`18:22.48`	`FuriousGeorge`	`are you root`
`18:22.51`	`pifiu`	`yes`
`18:23.06`	`pifiu`	`i downloaded *, and the sounds pacakge`
`18:23.09`	`pifiu`	`i extracted both`
`18:23.14`	`pifiu`	`i went into the * one and did`
`18:23.19`	`pifiu`	`make clean`
`18:23.20`	`FuriousGeorge`	`is asterisk in /usr/sbin?`
`18:23.21`	`pifiu`	`make`
`18:23.22`	`pifiu`	`make install`
`18:23.27`	`FuriousGeorge`	`is /usr/sbin in your path`
`18:23.27`	`pifiu`	`and then make samples`
`18:23.30`	`pifiu`	`let me see`
`18:23.34`	`Gunde`	`I suggested to pifiu redirecting the output into two files so he can read it step by step`
`18:23.45`	`pifiu`	`what do you mean if its in my path george?`
`18:23.48`	`jedaustin`	`FuriousGeorge: I made sure it was on an irq by itself.. I just noticed a service running irqbalance.. maybe related?`
`18:24.12`	`FuriousGeorge`	`bmy brian # echo $PATH`
`18:24.24`	`FuriousGeorge`	`jedaustin: could be`
`18:24.43`	`pifiu`	`when i run "whereis asterisk" i get the following http://pastebin.ca/206927`
`18:24.44`	`FuriousGeorge`	`jedaustin: digium telephony hardware has a reputation for being fussy with irqs`
`18:25.20`	`FuriousGeorge`	`pifiu: what happens when you echo $PATH like i showed you`
`18:25.48`	`FuriousGeorge`	`pifiu: if /usr/sbin is there, check in that dir for asterisk`
`18:26.37`	`FuriousGeorge`	`cd /usr/sbin`
`18:26.39`	`FuriousGeorge`	`ls -la`
`18:26.56`	`FuriousGeorge`	`make that ls -lc`
`18:27.34`	`pifiu`	`ok and what do i look for george?`
`18:27.42`	`pifiu`	`that list is too big to copy`
`18:27.46`	`FuriousGeorge`	`asterisk`
`18:28.05`	`pifiu`	`it doesnt let me scroll up past the D`
`18:28.10`	`pifiu`	`any way to format it different?`
`18:28.21`	`FuriousGeorge`	`yeah ls -lc\|less`
`18:28.25`	`pifiu`	`nevermind, asterisk isnt there`
`18:28.29`	`*** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr)`
`18:28.44`	`FuriousGeorge`	`changedir to you asterisk source and type make install`
`18:29.16`	`pifiu`	`ok done`
`18:29.24`	`pifiu`	`then?`
`18:29.31`	`FuriousGeorge`	`did you see a bunch of files being copied`
`18:29.39`	`pifiu`	`it said "nothing to be done" in a lot of the lines`
`18:29.46`	`pifiu`	`let me copy a portion`
`18:29.58`	`jedaustin`	`FuriousGeorge: yes :) I just bought the ebook: Trixbox Made Easy by Kerry Garrison. It has a section where a bunch of services are disabled. irqbalance is one of them.`
`18:30.19`	`pifiu`	`http://pastebin.ca/206988`
`18:30.34`	`pifiu`	`thats what i get in a lot of the lines`
`18:30.35`	`FuriousGeorge`	`jedaustin: im curious to know if it helps your issues`
`18:30.36`	`pifiu`	`it ran really quickly`
`18:30.39`	`jedaustin`	`Too bad this book is DRM'd so you can't print or cut and paste.`
`18:30.50`	`jedaustin`	`FuriousGeorge: I'll know in a minute :)`
`18:31.23`	`FuriousGeorge`	`jedaustin: you may want to try unloading the zap modules and reloading them, just as a sanity check`
`18:31.43`	`jedaustin`	`FuriousGeorge: I rebooted.`
`18:32.15`	`FuriousGeorge`	`pifiu: what version of asterisk is this`
`18:32.27`	`pifiu`	`the most stable i believe from their site`
`18:32.28`	`joe`	`~book`
`18:32.34`	`jbot`	`well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`18:32.34`	`pifiu`	`1.2.12`
`18:32.36`	`pifiu`	`1.2.12.1`
`18:32.50`	`*** join/#asterisk oQPa (i=Ftv@144.Red-83-41-136.dynamicIP.rima-tde.net)`
`18:32.55`	`FuriousGeorge`	`pifiu: ok we can assume you made and installed libpri and zaptel already right`
`18:33.03`	`pifiu`	`no`
`18:33.06`	`pifiu`	`i am not using any hardware`
`18:33.09`	`pifiu`	`just iax`
`18:33.30`	`FuriousGeorge`	`i thought libpri was required no matter what`
`18:33.36`	`pifiu`	`oh`
`18:33.39`	`FuriousGeorge`	`and the zaptel module helped with the timing on meetme`
`18:33.41`	`pifiu`	`i thought not?`
`18:33.48`	`pifiu`	`in that case i havent done that`
`18:34.02`	`pifiu`	`i skipped it, since the description in the * book said its really used for the hardware`
`18:34.02`	`FuriousGeorge`	`give that a whirl then make && make install asterisk again`
`18:34.12`	`pifiu`	`make &&?`
`18:34.31`	`FuriousGeorge`	`that means after a successfully completed make, make install`
`18:34.32`	`pifiu`	`i thought the order was critical to the install process? let me try it again`
`18:34.34`	`FuriousGeorge`	`in bash`
`18:34.35`	`toerkeium`	`guys, what could be the reason that when I ping my * box it gives me 3ms, but when I do a "sip show peers" shows my extension with 110ms. Any idea about it ?`
`18:34.49`	`pifiu`	`ok george, bb in a bit`
`18:35.18`	`*** join/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`18:35.38`	`hoobastooba`	`how do i set my console dsp device?`
`18:35.57`	`FuriousGeorge`	`toerkeium: are the clients on the same lan`
`18:36.49`	`toerkeium`	`no, clientes/softphones are behind a nated lan next to the * server`
`18:37.21`	`toerkeium`	`so, client/lan > DHCP Router > LAN - * Box`
`18:37.53`	`FuriousGeorge`	`and you ping the clients from the same lan`
`18:38.19`	`toerkeium`	`yes`
`18:38.33`	`FuriousGeorge`	`or rather you ping asterisk from the same lan as the clients?`
`18:39.54`	`toerkeium`	`the same machine running x-lite shows pings output of about 3ms, then I go to the server and I do a "sip show peers" and show the extension which is that x-lite connected with 110ms more or less`
`18:39.55`	`FuriousGeorge`	`well if it makes you feel any better, my * server and my clients are on the same lan, and my ping to them from sterisk is around .5 ms, but in sip show peers its 14ms`
`18:39.55`	`jedaustin`	`FuriousGeorge: I rebooted. Didn't help :(`
`18:40.21`	`FuriousGeorge`	`jedaustin: what about now setting all the gains and all the everything back to 0 and starting again`
`18:40.28`	`toerkeium`	`FuriousGeorge: I ping the * box from the cliets`
`18:40.29`	`toerkeium`	`clients`
`18:40.31`	`pifiu`	`hey george when i run make in the asterisk source, i get a lot of these lines that look like this http://pastebin.ca/207002`
`18:40.32`	`kattmat`	`got my voicemail running. anyone has a link on how to build menus (perl or php) ?`
`18:41.45`	`FuriousGeorge`	`toerkeium: im pointing out that its normal for the ping to be slower`
`18:41.50`	`*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)`
`18:41.51`	`FuriousGeorge`	`err faster`
`18:41.57`	`stugster`	`florz:D`
`18:41.58`	`FuriousGeorge`	`what you see in asterisk cli is the latency`
`18:42.01`	`stugster`	`IPv6 :D`
`18:42.09`	`FuriousGeorge`	`its not the same as ping`
`18:42.14`	`stugster`	`you're the first i've seen!`
`18:42.22`	`FuriousGeorge`	`increase packet size on ping and see what happens to latency`
`18:42.53`	`toerkeium`	`which size of packet is sending * to know the latency to the clients?`
`18:43.04`	`florz`	`stugster: Maybe, others simply do have reverse resolution configured? =:-)`
`18:43.35`	`FuriousGeorge`	`toerkeium: i dunno, but let me know when you find out :)`
`18:43.56`	`toerkeium`	`sure`
`18:43.57`	`toerkeium`	`thanks`
`18:44.06`	`stugster`	`yeh, but ohhhh`
`18:44.09`	`stugster`	`first one i've seen :DL`
`18:44.19`	`FuriousGeorge`	`pifiu: looks normal`
`18:44.23`	`pifiu`	`ok`
`18:44.26`	`FuriousGeorge`	`so you made libpri and zaptel right`
`18:44.29`	`pifiu`	`well ok im done so now what`
`18:44.31`	`pifiu`	`yep`
`18:44.43`	`FuriousGeorge`	`then you did a make && make install in the asterisk source?`
`18:44.47`	`pifiu`	`zaptel first then lipri,then *`
`18:44.57`	`ManxPower`	`Gads! The numbers shown in "sip show peers" or "iax2 show peers" are NOT NETWORK LATENCY`
`18:45.02`	`FuriousGeorge`	`i usually do libpri first buy i dont know that it matters`
`18:45.15`	`pifiu`	`wait no... lol i didnt know where teh && was`
`18:45.16`	`*** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il)`
`18:45.25`	`pifiu`	`so go to the * source folder and type make &&?`
`18:45.28`	`FuriousGeorge`	`you can just do make`
`18:45.32`	`pifiu`	`i did make`
`18:45.33`	`FuriousGeorge`	`wait for it to finish`
`18:45.34`	`pifiu`	`and then make install`
`18:45.35`	`FuriousGeorge`	`then make install`
`18:45.39`	`pifiu`	`yeah i did those`
`18:45.47`	`pifiu`	`what should i do next?`
`18:45.52`	`drcode`	`hi al`
`18:45.55`	`ManxPower`	`They are the amount of time it takes for the remote device to accept, process, and reply to a specific type of packet. For SIP it is a SIP OPTIONS packet. Many SIP devices don't have a lot of CPU to spare and so make responses to SIP OPTIONS packets a VERY VERY low priority.`
`18:45.59`	`drcode`	`I have two qustions`
`18:46.14`	`drcode`	`1. I saw some web site with sip:333@... number`
`18:46.22`	`FuriousGeorge`	`ManxPower: yeah, thats what i said. just not in so many words :)`
`18:46.25`	`drcode`	`how I can call them , do I need to setup my ast. box?`
`18:46.33`	`FuriousGeorge`	`pifiu: see if you can run asterisk now`
`18:46.41`	`*** join/#asterisk flujan_ (n=flujan@internet.nube.com.br)`
`18:46.48`	`FuriousGeorge`	`use tab complete like ast<tab>`
`18:46.50`	`flujan_`	`hi all :)`
`18:46.51`	`ManxPower`	`pifiu: then you look at the end of the messages from make install and confirm that there were no problems`
`18:47.03`	`flujan_`	`I am trying to get the channel status`
`18:47.11`	`flujan_`	`I am using idefisk so for instance`
`18:47.27`	`ManxPower`	`FuriousGeorge: I'm starting to wonder if there is a policy where only people that know nothing about linux or networking are allowed to install Asterisk`
`18:47.35`	`flujan_`	`so I am trying to use channel status IAX2/40003`
`18:47.38`	`FuriousGeorge`	`:)`
`18:47.39`	`flujan_`	`but i have no success...`
`18:48.14`	`FuriousGeorge`	`ManxPower: i guess that is a good thing for both linux and asterisk in the end`
`18:48.18`	`flujan_`	`so, how can I get a channel status?`
`18:48.29`	`*** part/#asterisk hoobastooba (n=ckwall@63.149.122.93)`
`18:48.43`	`FuriousGeorge`	`flujan_: you know tab complete works in asterisk cli, right?`
`18:49.18`	`FuriousGeorge`	`zap show channel, sip show channel, show channel`
`18:49.25`	`pifiu`	`yeah i did that george and nothing =(`
`18:49.31`	`pifiu`	`=(`
`18:49.33`	`FuriousGeorge`	`try all those with tab complete and check out your options`
`18:49.43`	`FuriousGeorge`	`pifiu: echo $PATH`
`18:49.49`	`FuriousGeorge`	`do you see /usr/sbin`
`18:49.52`	`ManxPower`	`pifiu: put the last screen of the output of "make" on pastebin`
`18:50.20`	`pifiu`	`yeah i think somewhere. that line is hard to read`
`18:50.22`	`ManxPower`	`same with "make install"`
`18:50.22`	`FuriousGeorge`	`then do what he said`
`18:50.41`	`pifiu`	`my options dont have the asterisk in any of them when i start to type ast (tab) (tab)`
`18:51.05`	`ManxPower`	`pifiu: I believe asterisk is NOT being built.`
`18:51.08`	`pifiu`	`i get this http://pastebin.ca/207012`
`18:51.17`	`FuriousGeorge`	`pifiu: do what ManxPower is recommending`
`18:51.22`	`FuriousGeorge`	`he's probably right`
`18:51.31`	`pifiu`	`ok so make and make install again for *?`
`18:51.32`	`flujan_`	`FuriousGeorge, humm... ok... but how to get it without tab complete?`
`18:51.47`	`flujan_`	`beacuse I am using it throught a agi script`
`18:51.47`	`pifiu`	`ill copy the last 10 or so lines from each one?`
`18:51.57`	`flujan_`	`and asterisk generally puts a minus and a number after the channel`
`18:51.59`	`flujan_`	`for instance`
`18:51.59`	`ManxPower`	`pifiu: the last 60 lines would be better.`
`18:52.00`	`FuriousGeorge`	`asterisk -rx "command here"`
`18:52.11`	`FuriousGeorge`	`flujan_: that was for you ^`
`18:52.25`	`flujan_`	`IAX2/56982 became IAX/56982-56`
`18:52.30`	`flujan_`	`os something like that`
`18:52.38`	`FuriousGeorge`	`pifiu: yeah, on pastebin like you did before`
`18:52.46`	`FuriousGeorge`	`but this time the last lines, not in the middle`
`18:53.32`	`flujan_`	`FuriousGeorge, because of that I can call it without the tab complete feature`
`18:53.37`	`flujan_`	`:(`
`18:53.45`	`FuriousGeorge`	`flujan_: im not familiar enough with the agi to help you, i dont think`
`18:53.55`	`ManxPower`	`flujan_: what exactly are you trying to do?`
`18:54.14`	`flujan_`	`ManxPower, get a channel status to allow or not a call to be placed to that channel.`
`18:54.30`	`pifiu`	`ok after i run "make" i get this, http://pastebin.ca/207022 and then when i run "make install" I get this http://pastebin.ca/207023`
`18:54.42`	`flujan_`	`FuriousGeorge, even with asterisk -rx "show channel IAX2/teste" for instance returns nothing`
`18:54.44`	`pifiu`	`that includes everything`
`18:54.52`	`pifiu`	`from the make command to the make install, its quite short`
`18:54.59`	`flujan_`	`since asterisk knows IAX2/teste as IAX2/teste-2`
`18:55.01`	`pifiu`	`however the very first time i built it yesterday i think it was longer?`
`18:55.04`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`18:55.04`	`ManxPower`	`pifiu: asterisk is not being built.`
`18:55.14`	`ManxPower`	`flujan_: what is wrong with ChanIsAvail?`
`18:56.00`	`jedaustin`	`FuriousGeorge: looks like I have to configure the gains of the two channels separately. fxotune.conf for channel 3 is 14, chanel 2 is 5. I wish there was an automatic gain control in zap cards`
`18:56.05`	`ManxPower`	`flujan_: channel status and stuff like that is normally something you would use the Manager Interface to get`
`18:57.23`	`flujan_`	`flujan_, hum...`
`18:57.28`	`flujan_`	`ops`
`18:57.35`	`FuriousGeorge`	`pifiu: your make failed`
`18:57.42`	`flujan_`	`ManxPower, hum... I will try to use it then...`
`18:57.58`	`ManxPower`	`flujan_: at least look into it. 8-)`
`18:58.13`	`FuriousGeorge`	`ManxPower: says there that pifiu needs to upgrade his TDS`
`18:58.22`	`ManxPower`	`flujan_: you could actually use the Manager Interface from within your AGI.`
`18:58.25`	`flujan_`	`ManxPower, yes... I will first try to use the ChanIsAvail?`
`18:58.32`	`*** join/#asterisk dasenjo (n=dasenjo@63.245.86.245)`
`18:58.33`	`FuriousGeorge`	`then make fails. ive never seen that before`
`18:58.35`	`flujan_`	`ManxPower, really? How?`
`18:58.41`	`ManxPower`	`FuriousGeorge: Yes, but I don't do well with people that can't even build asterisk. He is all your'se.`
`18:58.42`	`pifiu`	`so what causes it to fail, or what can i do to fix it?`
`18:58.43`	`boch`	`do you know why im getting -- Got SIP response 481 "Call/Transaction Does Not Exist" back from x.x.x.x every about 60sec, in the CLI ?`
`18:58.57`	`ManxPower`	`flujan_: Asterisk Management Interface (AMI) is accessed using a TCP connection.`
`18:58.57`	`FuriousGeorge`	`lol`
`18:59.12`	`pifiu`	`lol sorry manxpower, im trying but i always run into problems!`
`18:59.20`	`FuriousGeorge`	`pifiu: ok so you gotta upgrade your tds`
`18:59.29`	`pifiu`	`english plz?`
`18:59.30`	`FuriousGeorge`	`at least thats the first thing i would try`
`18:59.41`	`ManxPower`	`pifiu: your problem is that you are trying to learn many things at once. Linux, unix build process, Asterisk`
`18:59.52`	`FuriousGeorge`	`tabular datastreams? lemme ask google`
`19:00.03`	`pifiu`	`thats something related to databases isnt it?`
`19:00.13`	`pifiu`	`well said manx`
`19:00.22`	`pifiu`	`but is there something wrong with that =P?`
`19:00.55`	`pifiu`	`i had actually just installed freetds yesterday. maybe that has something to do with it?`
`19:02.00`	`FuriousGeorge`	`http://bugs.digium.com/view.php?id=2194`
`19:02.02`	`*** join/#asterisk manalive (n=yashy@d66-183-170-14.bchsia.telus.net)`
`19:02.09`	`FuriousGeorge`	`pifiu: check that link out`
`19:02.12`	`ManxPower`	`pifiu: Yes there is something wrong with that. You are on #asterisk expecting to learn the basics of building software.`
`19:02.14`	`pifiu`	`ok`
`19:02.38`	`ManxPower`	`That is like going to a medical doctor conference and expecing to learn how to perform heart transplants`
`19:02.42`	`pifiu`	`well doing a make clean, make and make install doesnt seem complicated. it just does when it fails and you have no idea waht to do`
`19:02.57`	`*** part/#asterisk jmorgan (n=jack@static-71-117-242-5.ptldor.dsl-w.verizon.net)`
`19:03.17`	`*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)`
`19:03.18`	`FuriousGeorge`	`pifiu: well, look, if you dont know what tds is then you dont need it, right? i use asterisk i dont have freetds installed`
`19:03.30`	`FuriousGeorge`	`so try upgrading or uninstalling it`
`19:03.35`	`flujan_`	`ok ManxPower I will try it... thanks for the help`
`19:04.06`	`pifiu`	`well i needed it for some php script i was messing with that had nothing to do with *`
`19:04.10`	`FuriousGeorge`	`as the words that appeared concurrently with the failed compile suggest`
`19:04.16`	`pifiu`	`something else i was doing. so i got that to work, but apparently broke this`
`19:04.40`	`kilobit2001`	`on what port is audio sent from * to sip device?`
`19:04.46`	`FuriousGeorge`	`pifiu: so if you do need it, and your sure you cant just upgrade it, apply the patch i linked you to`
`19:05.42`	`manalive`	`I just setup asterisk on a x100p card and I'm hearing a high pitched tone now when talking on the telephone. I do have DSL but I have the card filtered`
`19:06.15`	`pifiu`	`i will give that a look george. i appreciate the help, brb then`
`19:06.17`	`sbingner`	`manalive: cat /proc/interrupts -- is it sharing an IRQ?`
`19:06.50`	`FuriousGeorge`	`pifiu: and btw, i just googled 'You have older TDS asterisk' and that bug report was the first thing that came up, that's a little trick i learned in 'Nam that has proved quite useful`
`19:07.09`	`manalive`	`sbingner: yes, thanks :)`
`19:07.12`	`FuriousGeorge`	`for when charlie would break a dependency`
`19:07.14`	`*** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net)`
`19:08.20`	`joe`	`so I reset the time on a box and now things go directly to voicemessage`
`19:08.27`	`joe`	`suggestions?`
`19:08.29`	`drcode`	`sorry`
`19:09.29`	`*** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net)`
`19:09.53`	`drcode`	`I have some qustion`
`19:10.05`	`drcode`	`I saw some web site that I can call them by sip number`
`19:10.06`	`elriah`	`Hi all. Anyone else having problems with g729 decoders not freeing themselves after being used on 1.2.12.1?`
`19:10.37`	`sbingner`	`joe: restart *?`
`19:10.48`	`drcode`	`I can connect my ast. box to call world sip number`
`19:10.51`	`sbingner`	`messing with the time coulda made it a little unhappy`
`19:11.12`	`joe`	`sbingner: tried that no good`
`19:11.23`	`elriah`	`Also -- Anyone know of a good IAX provider? I need some 800 numbers via IP ...`
`19:11.28`	`elriah`	`toll free rather...`
`19:11.38`	`sbingner`	`elriah: I hear voxbone is decent`
`19:12.09`	`elriah`	`Tnx.`
`19:14.48`	`*** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net)`
`19:14.57`	`flujan_`	`ManxPower, I am trying to use chanisavail... I shall use it in the agi or in the cli?`
`19:18.34`	`jedaustin`	`FuriousGeorge: doesn't seem to work. Trying the newest svn branch of zaptel.`
`19:18.35`	`*** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal)`
`19:18.48`	`*** join/#asterisk adorah (n=admin@87.68.173.22.cable.012.net.il)`
`19:20.50`	`drcode`	`I understand that astriks can't be before fw?`
`19:20.52`	`drcode`	`like nat?`
`19:21.13`	`FuriousGeorge`	`jedaustin: i only mess with stable and the new beta. if u use svn zaptel im pretty sure you gotta use svn everything`
`19:21.30`	`FuriousGeorge`	`droops: i have several servers behind nat with remote clients`
`19:21.45`	`FuriousGeorge`	`err drcode`
`19:21.55`	`drcode`	`k`
`19:22.04`	`jedaustin`	`FuriousGeorge: that may be true.`
`19:22.22`	`drcode`	`with nat , I mean lan ip inside and realy ip out side?`
`19:22.31`	`FuriousGeorge`	`jedaustin: still it doesnt make sense that you should have this manny issues`
`19:23.02`	`jedaustin`	`FuriousGeorge: I think the root cause is my line, but I've had Qwest out 20 times at least.. they say 'nothing wrong'`
`19:23.31`	`drcode`	`any way, I can connect my ast. box into world sip phone`
`19:23.42`	`drcode`	`so I can call any sip nuber on the interner?`
`19:27.27`	`FuriousGeorge`	`jedaustin: i gotta run, but try bringing that server elsewhere and seeing if the problem exists behind those lines too`
`19:27.30`	`kilobit2001`	`on what port is audio sent to sip phone?`
`19:27.45`	`bsdfreak`	`5060`
`19:27.45`	`jedaustin`	`FuriousGeorge: good Idea`
`19:28.19`	`FuriousGeorge`	`jedaustin: its about that time to start swapping different elements and seeing what makes the problem disappear. if we know one thing to be true its that not every server has this issue`
`19:28.56`	`jedaustin`	`Any known issues with revision I of TDM400 cards?`
`19:29.05`	`FuriousGeorge`	`so some combination of server, tdm hardware, and lines must work`
`19:29.16`	`FuriousGeorge`	`good question`
`19:29.18`	`FuriousGeorge`	`but i gotta run`
`19:29.30`	`FuriousGeorge`	`good luck, and lemme know how it goes when i see you again`
`19:31.02`	`sbingner`	`jedaustin: what is your problem?`
`19:32.23`	`*** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au)`
`19:33.03`	`TripleFFFF`	`anywayt to disable hint and subscribe ?`
`19:33.17`	`TripleFFFF`	`i ge hammered with xlite crap users who polle me avery sec`
`19:33.30`	`elriah`	`Hi all. Anyone else having problems with g729 decoders not freeing themselves after being used on 1.2.12.1?`
`19:33.33`	`boch`	`do you know why im getting 481 from a host all the time ?`
`19:34.09`	`*** join/#asterisk Tako-san (n=Tako-san@24.108.162.254)`
`19:34.38`	`*** join/#asterisk jedaustin (n=chatzill@austin-j.its.dist.maricopa.edu)`
`19:36.04`	`sp0n9e-`	`hrm, random calls are starting again.`
`19:37.12`	`TripleFFFF`	`sp0n9e- ? random ?`
`19:38.21`	`sp0n9e-`	`TripleFFFF: yeah, for some reason it detects ringing when there is no ringing`
`19:38.29`	`sp0n9e-`	`and executes the dialplan then hangs up`
`19:40.18`	`boch`	`my asterisk is sending my peers a NOTIFY with a message-summary, how can i stop this? i dont want any summary`
`19:40.30`	`*** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-152-190-223.red.bezeqint.net)`
`19:42.01`	`justinu\|laptop`	`boch: remove the "mailbox=" line from sip.conf`
`19:42.03`	`TripleFFFF`	`<PROTECTED>`
`19:48.17`	`boch`	`justinu\|laptop: awesome, thanks`
`19:48.48`	`justinu\|laptop`	`np`
`19:49.56`	`sbingner`	`Unable to do INITIAL ProSLIC powerup on module 3`
`19:49.56`	`sbingner`	`Module 3: FAILED FXS (FCC)`
`19:49.58`	`sbingner`	`ut oh`
`19:51.02`	`sbingner`	`(yes, the power is connected heh)`
`19:53.22`	`drcode`	`any one can help me with some qustions?`
`19:54.08`	`drcode`	`I want to call from my ast. box into world sip numbers , any idea ?`
`19:56.31`	`*** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)`
`19:57.07`	`*** join/#asterisk s0lid (n=jlq@210.213.245.128)`
`19:57.15`	`*** join/#asterisk Primer (n=vi@sh.nu)`
`19:57.48`	`Primer`	`Anyone try this PSGw asterisk -> skype gateway for Linux? Does it work? Is there a better, freer one?`
`19:58.00`	`*** join/#asterisk vader-- (n=johndoe@204.183.88.101)`
`19:58.01`	`drcode`	`yes`
`19:58.11`	`drcode`	`Primer: from what I read it work perfect`
`19:58.24`	`*** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca)`
`19:58.30`	`drcode`	`did u download or bout it?`
`19:59.09`	`vader--`	`have any of you guys setup asterisk for recording calls when they come into your system?`
`19:59.43`	`vader--`	`im looking to record all calls to a certain extension and then if the call is trasnfered from that extension it would stop recording`
`20:01.48`	`kattmat`	`anyone's got a link on how to setup your own menues in asterisk?`
`20:02.22`	`*** join/#asterisk tzafrir (n=tzafrir@62.90.10.53)`
`20:03.15`	`*** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`20:03.39`	`*** join/#asterisk pingwin[work] (i=pingwin@gateway/tor/x-30667c91e82e2eeb)`
`20:03.44`	`*** join/#asterisk fjean5 (n=fjean5@207.107.208.137)`
`20:03.54`	`fjean5`	`hi everybody`
`20:03.56`	`pingwin[work]`	`what command can I issue to record a channel while in conversation?`
`20:03.59`	`Primer`	`drcode: I'm going to evaluate it now, and it if works, buy it`
`20:04.20`	`drcode`	`k`
`20:04.26`	`drcode`	`great`
`20:04.40`	`drcode`	`I saw some web site that let u call sip:// number`
`20:04.57`	`fjean5`	`question; anyone would know a motherboard model with 3.3v pci ?`
`20:05.22`	`drcode`	`let say I connect my x-lite into ast .box (at home) can I call those world wide number?`
`20:07.43`	`[TK]D-Fender`	`fjean5: Why? Got a 3.3V specific card already?`
`20:08.00`	`fjean5`	`tk-d-fender: exactly`
`20:08.09`	`CrazyTux`	`Hello, can someone here help me understand the whole under lying logic of going from PSTN <-> VoIP, I mean I get the basics as far as needing a gateway, but just a few things from the knowledge bank seem like they are missing?`
`20:08.16`	`[TK]D-Fender`	`fjean5: My condolences....`
`20:08.37`	`fjean5`	`[tk]d-fender: hehe`
`20:09.11`	`[TK]D-Fender`	`fjean5: Thats why I don't touch those cards :)`
`20:09.24`	`fjean5`	`<PROTECTED>`
`20:09.28`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`20:10.54`	`fjean5`	`:-)`
`20:11.01`	`*** join/#asterisk japerry (n=falc0n@216.231.51.209)`
`20:14.27`	`kilobit2001`	`is it possible to load the gsm audio from db as in odbcstorage?`
`20:18.25`	`TripleFFFF`	`hey all`
`20:18.31`	`TripleFFFF`	`TE410P is worth wat ?`
`20:19.14`	`fjean5`	`trippleFFFF: I recently sold a used one for 350$us....`
`20:19.37`	`TripleFFFF`	`k`
`20:19.43`	`TripleFFFF`	`so 900 can is high`
`20:20.02`	`fjean5`	`oh sorry`
`20:20.04`	`fjean5`	`wrong model`
`20:20.12`	`TripleFFFF`	`hehehe`
`20:20.15`	`fjean5`	`hehe`
`20:20.18`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`20:24.16`	`Primer`	`bah, wtf? this skype gateway requires skype and a sound card`
`20:26.12`	`*** part/#asterisk fjean5 (n=fjean5@207.107.208.137)`
`20:28.24`	`*** join/#asterisk RoyK (n=roy@ti211310a080-12885.bb.online.no)`
`20:29.13`	`*** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw)`
`20:29.50`	`*** join/#asterisk RoyK (n=roy@ti211310a080-12885.bb.online.no)`
`20:31.27`	`*** join/#asterisk mrg82 (n=na@85-210-133-44.dsl.pipex.com)`
`20:32.04`	`CrazyTux`	`Ok can someone help answer this for me....?`
`20:32.15`	`sbingner`	`TripleFFFF: www.digium.com -- has prices`
`20:32.27`	`japerry`	`CrazyTux: what is it?`
`20:32.44`	`CrazyTux`	`INVITE sip:bob@Biloxi.com SIP/2.0 <---- ok it looks like the format is sip:user@hostname correct?`
`20:32.56`	`CrazyTux`	`japerry, what would the user@hostname part be if it was a PSTN ?`
`20:33.04`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
`20:33.14`	`CrazyTux`	`japerry, would it be sip:xxxxxxxxx being my #?`
`20:33.18`	`japerry`	`one sec, let me look`
`20:33.20`	`japerry`	`not sip`
`20:33.35`	`CrazyTux`	`japerry, how does the whole interaction between client <-> server <-> land line work?`
`20:33.49`	`hmmhesays`	`any Bloody Brits in here right now?`
`20:33.51`	`japerry`	`through the extensions.conf file`
`20:34.05`	`japerry`	`CrazyTux: you use this file to map your dialplan and your extensions`
`20:34.10`	`CrazyTux`	`japerry, I not actually using asterisk, I'm using openser, but I just need to know this 'in general'`
`20:34.21`	`japerry`	`uhh well I can only help with asterisk`
`20:34.21`	`CrazyTux`	`japerry, I'm*`
`20:34.33`	`CrazyTux`	`japerry, so asterisk, I would map out the dial plan?`
`20:34.35`	`hmmhesays`	`anyone anyone?`
`20:34.43`	`CrazyTux`	`japerry, how would this work with "outgoing" requestrs?`
`20:34.55`	`RoyK`	`hmmhesays: no bloody brits in here, no :)`
`20:35.39`	`sp0n9e-`	`_9NXXXXXX would allow you to call out locally, right?`
`20:35.53`	`mrg82`	`One of my telecoms providers currently forwards calls to my sip account. I've recently built and asterisk server and plan to use the iax protocol for incoming calls. What information would i need to provide my provider with? Will a iax.domain.co.uk be sufficient?`
`20:35.57`	`japerry`	`Crazytux: if you're doing an outside call ,you'd do something like: exten => _1800NXXXXXX,1,Macro(outgoingdial25,${EXTEN})`
`20:36.14`	`japerry`	`sp0n9e: correct`
`20:36.42`	`hmmhesays`	`There is usually some`
`20:36.43`	`CrazyTux`	`japerry, what if I don't want an extension?`
`20:36.48`	`CrazyTux`	`japerry, what if I just wanted to dial the #`
`20:36.56`	`CrazyTux`	`japerry, also how would the actual SIP message / header look?`
`20:37.03`	`japerry`	`CrazyTux: --ALL--numbers are exten`
`20:37.21`	`japerry`	`thats how the dialplan works, you use the NX's for wildcards`
`20:37.38`	`CrazyTux`	`N being, and X being?`
`20:37.43`	`japerry`	`then I have it piping out to a Macro which is my digium T1 card, to send the DTMF digits`
`20:38.14`	`japerry`	`CrazyTux: read this: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf`
`20:38.37`	`stephane_`	`back`
`20:39.24`	`CrazyTux`	`japerry, ok one last question`
`20:39.40`	`CrazyTux`	`japerry, I don't get it... if I'm dialing out, why would I need to create that 'exten' in the dial plan?`
`20:39.48`	`CrazyTux`	`japerry, why can't I just say .... "dial this number entered"`
`20:40.11`	`sbingner`	`should I call digium if I have a failed module, or is there somebody here I can talk to?`
`20:41.37`	`japerry`	`CrazyTux: because thats how asterisk works. The dial plan is what you're talking about`
`20:41.43`	`japerry`	`dialing a number is part of the dial plan`
`20:41.51`	`japerry`	`so are phone extensions internally`
`20:42.06`	`CrazyTux`	`japerry, so do you see where I am confused?`
`20:42.44`	`japerry`	`CrazyTux: yes, asterisk's configuration, imho is fairly covoluted and unnecissarly complicated. but alas, thats how it works`
`20:43.03`	`CrazyTux`	`japerry, ok here is my thought process right now..`
`20:43.32`	`CrazyTux`	`japerry, user A, has a phone, he hooks it up, its configured to talk to my "server".... it sends over the SIP data of who he dialed / what he wants to connect to`
`20:43.37`	`CrazyTux`	`japerry, so far with me?`
`20:43.47`	`japerry`	`...`
`20:43.53`	`japerry`	`yes`
`20:44.00`	`CrazyTux`	`japerry, ok, now how does my server find out who to connect him to "externally"`
`20:44.08`	`CrazyTux`	`japerry, the sip packet right?`
`20:44.38`	`CrazyTux`	`japerry, so at this point, how does my server see the sip packet? the user entered a number...`
`20:44.43`	`CrazyTux`	`sip:phone_number_here ?`
`20:44.48`	`japerry`	`if the phone is registered with the server, then when the phone sends out what numbers it wants to dial, the server will see those numbers and dial them accordingly`
`20:44.52`	`CrazyTux`	`I don't get where the sip:user@hostname comes from.`
`20:45.03`	`japerry`	`well, you need the phone to be registered to the server`
`20:45.12`	`CrazyTux`	`Ok lets say the phone is registered to the server`
`20:45.15`	`CrazyTux`	`the phone calling out that is.`
`20:45.29`	`japerry`	`yah if its registered, then its talking to the server`
`20:45.39`	`CrazyTux`	`yes, so what about the 'unregistered phone'`
`20:45.52`	`japerry`	`when the phone sends out digits to call, if it matches an 'exten' then it'll go out through that path`
`20:46.02`	`CrazyTux`	`japerry, i.e. 1-800-play-boy`
`20:46.11`	`japerry`	`yah, it goes out via an external line`
`20:46.24`	`CrazyTux`	`japerry, I as user "a" use my voip phone to call 1-800-play-boy...`
`20:46.30`	`japerry`	`mine uses the digium card to goto verizon, sending DTMF tons`
`20:46.37`	`CrazyTux`	`japerry, data is sent to my server, now my server forwards this request to that actual #?`
`20:46.43`	`japerry`	`did you read that page?`
`20:46.50`	`CrazyTux`	`japerry, reading it right now`
`20:47.00`	`pingwin[work]`	`hey anyone know anything about 1-800-wet-bobs? it's something I "found" but don't really understand what it's purpose is`
`20:47.14`	`CrazyTux`	`pingwin[work], lol`
`20:47.44`	`pingwin[work]`	`CrazyTux: hehe have you called it yet?`
`20:47.53`	`CrazyTux`	`pingwin[work], nope not yet.`
`20:47.56`	`pingwin[work]`	`it's not my box or anything. don't have a clue what it is`
`20:48.14`	`pingwin[work]`	`it just replies with a list of what appears to be random numbers and then a series of tones`
`20:48.18`	`*** join/#asterisk JunoD (n=Junodixo@rrcs-71-40-42-130.sw.biz.rr.com)`
`20:49.11`	`JunoD`	`anyone know of a free or real cheap voip com. to test my box with`
`20:50.12`	`pingwin[work]`	`digium has a iax you can connect to if your wanting to test`
`20:50.34`	`*** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep)`
`20:50.53`	`CrazyTux`	`japerry, well thanks for the help, be back in a bit :)`
`20:51.18`	`JunoD`	`thanks pingwin[work]`
`20:53.02`	`pingwin[work]`	`JunoD: IAX2/guest@misery.digium.com`
`20:53.10`	`*** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net)`
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`20:54.46`	`*** mode/#asterisk [+o Qwell[]] by ChanServ`
`20:55.16`	`JunoD`	`thanks`
`20:55.30`	`*** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone)`
`20:57.06`	`teknoprep`	`exten => *25,1,Set(DB(ivr/custom-mode)=ivr-5)`
`20:57.26`	`teknoprep`	`is what i have now.. i need it to be exten => *25,1,Set(DB(ivr/custom-mode)=ring group 900)`
`20:57.27`	`*** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net)`
`20:57.27`	`*** mode/#asterisk [+o mog] by ChanServ`
`20:57.41`	`teknoprep`	`i don't understand how to setup a "ring group" onto this type of setup`
`21:02.40`	`*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`21:03.04`	`*** part/#asterisk eKo1 (n=eKo1@190.4.7.90)`
`21:04.37`	`kattmat`	`how to play an audio file in a directory of my own in the standard sound directory?`
`21:04.47`	`kattmat`	`standard_dir.../../my_folder/intro.wav`
`21:04.57`	`kattmat`	`Playback(intro) doesn't work, neither do Playback(my_folder/intro)`
`21:05.07`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`21:05.18`	`teknoprep`	`any help ?`
`21:05.44`	`kattmat`	`nvm worked now`
`21:05.52`	`pingwin[work]`	`teknoprep: what are you trying to get to happen?`
`21:05.53`	`teknoprep`	`no for me`
`21:05.56`	`teknoprep`	`ok`
`21:06.17`	`teknoprep`	`well right now i have it setup for 25 26 and *27 to change the IVR's`
`21:06.41`	`teknoprep`	`one ivr is for .. hi we are open... next ivr is out to lunch... and the last IVR is for closed`
`21:06.49`	`teknoprep`	`since the times for this varry so much i had to set this up`
`21:07.09`	`pingwin[work]`	`okay`
`21:07.21`	`teknoprep`	`now... the client want to change the hi we are open IVR to just make it ring the Operator or Receptionts phones.. the ring group`
`21:07.38`	`teknoprep`	`so basically i have ring group 900 setup for that`
`21:07.43`	`teknoprep`	`exten => *25,1,Set(DB(ivr/custom-mode)=ivr-5)`
`21:07.57`	`teknoprep`	`i can't figure out what to do to that.. to make it ring 900 which is a ring group`
`21:08.09`	`teknoprep`	`i have a combonation of FreePBX and custom asterisk macro's`
`21:08.29`	`teknoprep`	`this would be the custom stuff not able to be done in freepbx`
`21:08.30`	`pingwin[work]`	`drop the set and just have Dial`
`21:08.47`	`teknoprep`	`oh thats it?`
`21:09.29`	`teknoprep`	`exten => 900,n(DIALGRP),Macro(dial,25,${DIAL_OPTIONS},101-102-103-104-105-106-107-108-109-110-111-112-113-114)`
`21:09.52`	`teknoprep`	`that won't work`
`21:09.59`	`pingwin[work]`	`if that's how your macro works`
`21:10.20`	`pingwin[work]`	`well dial isn't a macro (to my knowlege) it's an application function of asterisk`
`21:10.27`	`teknoprep`	`i need to set the DB(ivr/custom-mode)=ivr-5 for this reason`
`21:10.31`	`pingwin[work]`	`and I have zero experience with freepbx`
`21:10.43`	`pingwin[work]`	`for what reason?`
`21:11.19`	`*** join/#asterisk Dr-Linux\|work (n=Linux@202.59.73.131)`
`21:12.18`	`teknoprep`	`onse second`
`21:12.20`	`teknoprep`	`looking forit`
`21:13.12`	`teknoprep`	`exten => 14843396707,n,Goto(${DB(ivr/custom-mode)}\|s\|1,)`
`21:13.21`	`teknoprep`	`i have that for my inbound calls`
`21:13.44`	`teknoprep`	`sooo`
`21:13.50`	`*** join/#asterisk dasenjo (n=dasenjo@208.195.215.170)`
`21:13.59`	`teknoprep`	`i need *25 to ring on exten => 900 RING Group`
`21:14.29`	`pingwin[work]`	`teknoprep: *25,1,Dial(SIP/111&SIP/112&SIP/123) there`
`21:14.33`	`pingwin[work]`	`that's one way`
`21:14.48`	`teknoprep`	`pingwin[work], that won't set it in the DB tho`
`21:15.00`	`teknoprep`	`that will just dial ALL extensions if i dial @25`
`21:15.03`	`teknoprep`	`*25`
`21:16.02`	`pingwin[work]`	`are you trying to record it to the call log?`
`21:16.02`	`teknoprep`	`nope`
`21:16.02`	`pingwin[work]`	`for historical reference?`
`21:16.02`	`teknoprep`	`no`
`21:16.03`	`teknoprep`	`as i have said before`
`21:16.03`	`teknoprep`	`*25 is dialed on a phone`
`21:16.03`	`pingwin[work]`	`then why are you requiring it to write to the database?`
`21:16.03`	`teknoprep`	`this puts the system into a so called 'we are open mode'`
`21:16.03`	`teknoprep`	`as of now`
`21:16.06`	`teknoprep`	`it brings you to an IVR that says hi we are open..`
`21:16.07`	`teknoprep`	`pretty much`
`21:16.42`	`pingwin[work]`	`are you talking about an IVR for a different extension?`
`21:16.44`	`teknoprep`	`NOW... if you dial *27 it changes the DBset of the ivr/custom-mode -> closed-ivr`
`21:16.47`	`teknoprep`	`nope`
`21:16.52`	`teknoprep`	`stop it and listen i will explain better`
`21:17.28`	`RoyK`	`new releases of asterisk never^Wrarely^Ware not meant to include new features unless good developers^W^Wfriends of digium have written them, and that the code is flawless^Wwell^Wmostly^Wsomehow tested`
`21:18.06`	`*** kick/#asterisk [RoyK!i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net] by mog (enough of that ^_^)`
`21:18.07`	`*** join/#asterisk RoyK (n=roy@ti211310a080-12885.bb.online.no)`
`21:19.05`	`*** join/#asterisk xnon (n=xnon@200.8.31.68)`
`21:19.33`	`pingwin[work]`	`what was wrong with what RoyK said?`
`21:19.34`	`teknoprep`	`i dial 25 from any phone... when someone on the outside calls 4843396707 from now on ... the caller will here... Hi welcome to 'company name" and so on(ivr)... now at lunch time... the ppl inside the building dial 26... this sets the incomming IVR for the number 4843396707 to... "hi we are in today but away from our desks and can not answer the phone" and so on(ivr)... now when somone in the office dials *27 it puts the ivr into "we are now`
`21:19.34`	`teknoprep`	`closed Blah BLAH BLAH"... my client LOVES this... except for one thing... when they are in the office.. which is *25... they DO NOT WANT and IVR... they want it to just ring every single phone in the office.....`
`21:19.53`	`teknoprep`	`i use this to set it up`
`21:19.56`	`mog`	`its inflammatory`
`21:19.59`	`teknoprep`	`i have 3 different IVR's of course`
`21:20.06`	`RoyK`	`pingwin[work]: I have no idea, really.`
`21:20.12`	`teknoprep`	`exten => *25,1,Set(DB(ivr/custom-mode)=ivr-5)`
`21:20.12`	`teknoprep`	`exten => *26,1,Set(DB(ivr/custom-mode)=ivr-6)`
`21:20.12`	`teknoprep`	`exten => *27,1,Set(DB(ivr/custom-mode)=ivr-4)`
`21:20.32`	`teknoprep`	`and exten => 14843396707,n,Goto(${DB(ivr/custom-mode)}\|s\|1,)`
`21:20.51`	`pingwin[work]`	`mog he's just making a point of view about the subject of this channel`
`21:21.01`	`teknoprep`	`now for my dilema.. how the F do i set *25 to an already working "ring group" using DBset`
`21:21.05`	`pingwin[work]`	`and as an asterisk developer I am interested in what he may have to say`
`21:21.21`	`mog`	`pingwin[work], if you dont think that was an attempt at upsetting specific people in this channel your pretty silly`
`21:21.38`	`pingwin[work]`	`i dunno I wouls have just set variables to maintain the status`
`21:21.38`	`mog`	`pingwin[work], what have you developed for asterisk? im not familiar with your work`
`21:22.23`	`pingwin[work]`	`well all I can say is I work for a company that is developing software to interact with asterisk.`
`21:22.45`	`mog`	`what have you given to the community?`
`21:22.46`	`teknoprep`	`no comments on me ?`
`21:22.46`	`pingwin[work]`	`but the business end of things isn't solid so I am unfortunately unable to discuss any kind of details`
`21:22.47`	`teknoprep`	`lol`
`21:22.51`	`mog`	`zilch?`
`21:22.58`	`mog`	`okies`
`21:22.59`	`pingwin[work]`	`mog: nothing for asterisk.`
`21:23.17`	`mog`	`okies`
`21:23.44`	`pingwin[work]`	`mog: but I'm still interested in what goes into asterisk and how things get included into the distribution of the software`
`21:24.10`	`Qwell[]`	`pingwin[work]: we have very clearly defined policies on what gets put into a release branch`
`21:24.46`	`pingwin[work]`	`thats good`
`21:25.56`	`teknoprep`	`bleh i typed that all out for nothing ?`
`21:25.57`	`teknoprep`	`lol`
`21:32.11`	`[TK]D-Fender`	`teknoprep : just have that context dial the Local channel entry for that ring-group`
`21:32.25`	`teknoprep`	`you mean DIAL 900?`
`21:32.36`	`teknoprep`	`DIAL(900) ?`
`21:33.20`	`teknoprep`	`but it needs to be setup in a DBset`
`21:34.07`	`teknoprep`	`i am busy of my own`
`21:34.44`	`teknoprep`	`oh the context`
`21:34.53`	`teknoprep`	`wait?`
`21:34.56`	`teknoprep`	`i stil don't get it`
`21:36.11`	`kattmat`	`hm`
`21:36.13`	`kattmat`	`i have an`
`21:36.25`	`kattmat`	`exten => 9999,1,VoicemailMain(1234@default)`
`21:36.34`	`kattmat`	`and this works from a software sip phone`
`21:36.43`	`kattmat`	`but not from my normal phone hooked up to an ata-box`
`21:37.20`	`[TK]D-Fender`	`teknoprep : you are just jumping to a context, so make one for that DB value and dial from there`
`21:37.43`	`[TK]D-Fender`	`teknoprep : [ivr-9]`
`21:38.09`	`[TK]D-Fender`	`teknoprep : exten => s,1,Dial(Local/900@whereverAMPshovesyourringgroups)`
`21:38.09`	`teknoprep`	`[ivr-9] being my ring all context ?`
`21:38.27`	`teknoprep`	`ohhh`
`21:38.35`	`[TK]D-Fender`	`teknoprep : Arbitrary # I came up with`
`21:38.45`	`teknoprep`	`got ya`
`21:38.55`	`teknoprep`	`i understand it was just a ficticious config`
`21:39.07`	`*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)`
`21:39.35`	`[TK]D-Fender`	`teknoprep : 99% Ready-to-serve`
`21:39.46`	`teknoprep`	`hehe`
`21:40.18`	`*** join/#asterisk crochat (i=crochat@84-74-145-66.dclient.hispeed.ch)`
`21:40.29`	`teknoprep`	`so in your example i would change .... exten => 25,1,Set(DB(ivr/custom-mode)=ivr-5) to exten => 25,1,Set(DB(ivr/custom-mode)=ivr-9)`
`21:40.36`	`*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)`
`21:41.52`	`teknoprep`	`or i could do this`
`21:41.59`	`teknoprep`	`since all ring groups are under ext-group`
`21:42.01`	`teknoprep`	`context`
`21:42.01`	`[TK]D-Fender`	`teknoprepYou see what I'm doing, just run with it :)`
`21:42.34`	`teknoprep`	`exten => *25,1,Set(DB(ivr/custom-mode)=ext-group,900,1)`
`21:43.41`	`teknoprep`	`ty again [TK]D-Fender`
`21:44.00`	`kattmat`	`i can't get it to call out`
`21:44.02`	`teknoprep`	`i need to read the damn book before i do my next install.. i want to get away from FreePBX`
`21:44.07`	`[TK]D-Fender`	`[17:13] <teknoprep> exten => 14843396707,n,Goto(${DB(ivr/custom-mode)}\|s\|1,) <----- see this? It jumps to a context and starts you at "s,1". So jsut make a context that immediately dials a local channel`
`21:44.37`	`[TK]D-Fender`	`teknoprep How big a setup do you have? What kind of phones / cards?`
`21:44.50`	`teknoprep`	`mix between the need of good phones.. and just phones`
`21:44.57`	`teknoprep`	`about 20 Linksys Spa942`
`21:45.02`	`teknoprep`	`and 60 gxp-2000`
`21:45.10`	`teknoprep`	`a few sipurar ata's`
`21:45.11`	`[TK]D-Fender`	`SPA-942 aren't "bad", just generally not worth it.`
`21:45.18`	`teknoprep`	`i really like them alot`
`21:45.23`	`[TK]D-Fender`	`Though infinitely better than the SPA's :)`
`21:45.25`	`teknoprep`	`they have a great feel and sound to them`
`21:45.57`	`p1p`	`coughgxp's suckcough`
`21:45.58`	`[TK]D-Fender`	`er, GXP's`
`21:46.12`	`teknoprep`	`i actually like the GXP for area's that only need intercom functionality`
`21:46.17`	`p1p`	`teknoprep has been known to launch into 15 minute tirades defending those phones`
`21:46.19`	`teknoprep`	`or say just random low use calls`
`21:46.27`	`p1p`	`I love busting his chops over it`
`21:46.28`	`teknoprep`	`lol p1p`
`21:46.34`	`[TK]D-Fender`	`In North America its either Polycom or Aastra for all of my installs`
`21:46.56`	`teknoprep`	`are polycom really that much better then the Linksys SPA 942 ?`
`21:47.02`	`teknoprep`	`for sound quality`
`21:47.13`	`kattmat`	`can somebody explain this to me`
`21:47.15`	`teknoprep`	`i just can't see how it could be THAT much better... ppl are amazed with the 942`
`21:47.22`	`kattmat`	`when i connect to my asterisk server from a sip software client`
`21:47.30`	`kattmat`	`and call 9999 i go to voicemailmain`
`21:47.48`	`kattmat`	`but when i hook up an ata-box and register it with the same login stuff, i can't call 9999 (nothing happens)`
`21:48.16`	`carrar`	`teknoprep, do you use VLAN's on the 942?`
`21:48.26`	`[TK]D-Fender`	`teknoprep : You haven't worked with Polycoms. I've owned all sort of phones, including an SPA-941 which is functionally equivalent to the 942. Believe me the SPA's are NOT impressive by comparison.`
`21:48.34`	`teknoprep`	`what do you mean VLAN's on the 942 ? i have a VoIP vlan`
`21:48.45`	`carrar`	`phone one vlan, data port on back of phone on a different vlan`
`21:48.48`	`teknoprep`	`VLAN 101 is my VoIP vlan`
`21:48.59`	`[TK]D-Fender`	`teknoprep : They'd flip over what a 501 or 601 could offer them by contrast. much more solid choice.`
`21:49.35`	`*** join/#asterisk Chicago (n=Chicago@c-71-201-141-218.hsd1.il.comcast.net)`
`21:49.42`	`[TK]D-Fender`	`Aastra 480i is also a very solid choice.`
`21:49.57`	`teknoprep`	`carrar in a different situation i would consider that.. don't have enough ether outlets for that stuff`
`21:50.18`	`carrar`	`You mean you have too many outlets?`
`21:50.21`	`*** join/#asterisk hyphen (n=hyphen@71.224.213.97)`
`21:51.05`	`carrar`	`So you are not using the data port on the back in a different vlan then the voip traffic?`
`21:51.22`	`*** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do)`
`21:51.23`	`[TK]D-Fender`	`kattmat : Could be that you have a dialplan on your ATA that does not like "9999" as a number and isn't even bothering to pass it on to *.`
`21:51.43`	`kattmat`	`the dialplan setup on the ata is just empty`
`21:51.58`	`[TK]D-Fender`	`kattmat : That could actually be a BAD thing you know... what model?`
`21:52.06`	`kattmat`	`vood 322`
`21:52.08`	`Adam12`	`d-fender: What's your thoughts on the Aastra 9133I (I believe that's the model)? I'm in a toss up over the Polycom 430 and the Aastra 9133`
`21:52.23`	`Chicago`	`Would I need something as full blown and complete as asterisk to simply record calls?`
`21:52.28`	`[TK]D-Fender`	`kattmat : Ok, you are also picking a "fringe" choice model.... can't help you there.`
`21:52.44`	`[TK]D-Fender`	`Adam12 : 9133 is "decent", but I'd pick any Polycom over it.`
`21:52.45`	`kattmat`	`i got it from some phone operator.`
`21:52.48`	`kattmat`	`not using their service though`
`21:52.53`	`Adam12`	`d-fender: Thanks.`
`21:52.58`	`[TK]D-Fender`	`Adam12 : depends on your scenario.`
`21:53.05`	`[TK]D-Fender`	`Adam12 : Got PoE?`
`21:53.16`	`paolob-parroquia`	`Hi guys! I can't get sound moh when transfering a call. When I perform the transferring the CLI tells me: "WARNING[18976]: file.c:509 ast_openstream_full: File /usr/share/asterisk/mohmp3/espinosa-danos_un_corazon does not exist in any format", but the correponding .mp3 file is there. Any hint? v. 1.2.7.1 ubuntu`
`21:53.17`	`Adam12`	`d-fender: No, and it's my first voip installation (soho here)`
`21:53.30`	`[TK]D-Fender`	`Adam12 : then forget the IP430 and go for the IP 501.`
`21:53.53`	`[TK]D-Fender`	`Adam12 : bigger screen, 3 line-keys, etc.`
`21:53.54`	`kattmat`	`ah works now. great.`
`21:54.04`	`kattmat`	`it was set not to dial if there wasn't 5 or more digits`
`21:54.18`	`[TK]D-Fender`	`kattmat : Yup... dialplan crazyness`
`21:54.29`	`Adam12`	`d-fender: I looked at that model earlier. I'll have to look at it again. Thanks.`
`21:54.51`	`[TK]D-Fender`	`Adam12 : I own ever desk model out to date`
`21:55.17`	`[TK]D-Fender`	`Adam12 : And have worked with the Aastra 480i. The 9133 is "ok", but I'd rather spend more to get a better phone.`
`21:55.45`	`kattmat`	`fender, cheers man`
`21:55.56`	`Adam12`	`d-fender: I agree.`
`21:56.04`	`p1p`	`[TK]D-Fender: Has Aastra been keeping pretty consistent on software updates? I tested a bunch of their stuff and I liked it but it all seemed a little to new for me so i stuck with Polycom`
`21:57.45`	`[TK]D-Fender`	`p1p : The 480i has impressed me and is an interesting option. its a mixup choice. I still love Polycom's superior call-handling, but Aastra's soft-keys are rather remarkable and the configuration quite simple.`
`21:58.00`	`[TK]D-Fender`	`p1p : Both are worth, though I still love my Poly's :)`
`21:58.05`	`[TK]D-Fender`	`worthy*`
`21:58.44`	`kattmat`	`Fender, http://rafb.net/paste/results/iULZ8H72.html that's my extensions.conf`
`21:58.59`	`kattmat`	`but i just get "beep-beep-beep" fast when i try calling out`
`21:59.47`	`teknoprep`	`[TK]D-Fender, this just isn't working for me`
`22:00.22`	`[TK]D-Fender`	`kattmat : I'd need to see some SIP debug for that call. do "sip debug peer [yourphonesentryhere]" and then pastebin it. Need to move fast, I'm outta here momentarily`
`22:00.55`	`kattmat`	`oh. i got it working now.`
`22:00.59`	`kattmat`	`or well`
`22:01.02`	`kattmat`	`the outbound stuff gets called now`
`22:01.07`	`kattmat`	`but it doesn't call out`
`22:01.12`	`kattmat`	`can paste my sip.conf`
`22:01.54`	`kattmat`	`http://rafb.net/paste/results/hYKL3O35.html`
`22:01.56`	`kattmat`	`there it is`
`22:02.15`	`kattmat`	`hm.. don't read it all :)`
`22:02.52`	`*** join/#asterisk zotz (n=zotz@24.244.163.225)`
`22:03.04`	`kattmat`	`http://rafb.net/paste/results/6qbJrg90.html new extensions.conf`
`22:03.06`	`kattmat`	`fender, any ideas?`
`22:04.12`	`[TK]D-Fender`	`kattmat : apstebin a failed call.`
`22:04.36`	`kattmat`	`eeh.`
`22:04.39`	`kattmat`	`hm this may help`
`22:04.52`	`kattmat`	`<PROTECTED>`
`22:04.52`	`kattmat`	`<PROTECTED>`
`22:05.05`	`kattmat`	`it just gets an s instead of the number`
`22:05.28`	`[TK]D-Fender`	`kattmat : actually... of COURSE ti will fail. you made "outbound" a context with "s" as the starting exten. Your dial statement calls ${EXTEN} which no longer contains the # you dialed but rather "s". this leads to instant rejection by your ITSP`
`22:06.03`	`kattmat`	`aha. trying to fix it`
`22:06.19`	`teknoprep`	`[TK]D-Fender, this is how i would dial directly into a Ring Group`
`22:06.27`	`teknoprep`	`exten => 14843351444,n,Goto(ext-group,900,1)`
`22:06.36`	`teknoprep`	`now... if i wanted to DBset something for this to work withthat`
`22:06.47`	`teknoprep`	`exten => 14843396707,n,Goto(${DB(ivr/custom-mode)}\|s\|1,)`
`22:06.53`	`teknoprep`	`wtf do i do.. i have tryed everything i can think of`
`22:07.28`	`[TK]D-Fender`	`kattmat : Merry Christmas : http://pastebin.ca/207249`
`22:07.43`	`kattmat`	`hm`
`22:07.44`	`kattmat`	`Oct 18 00:07:12 WARNING[16465]: chan_sip.c:9720 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"15399" <sip:0366800010@proxy.digisip.net>;tag=as57b477cd'`
`22:07.47`	`kattmat`	`what the hell?`
`22:07.53`	`[TK]D-Fender`	`teknoprep : I just finished gift-wrapping the answer for you earlier....`
`22:07.59`	`kattmat`	`i have a [digisip] in the sip.conf`
`22:08.01`	`teknoprep`	`yeah i tryed that`
`22:08.10`	`kattmat`	`so it should authenticate alright, why doesn't it?`
`22:08.18`	`[TK]D-Fender`	`teknoprep : Well I have to head out. Back in 4 hours or so.`
`22:08.31`	`teknoprep`	`np have fun`
`22:09.20`	`[TK]D-Fender`	`teknoprep : If you're around when I'm back gimme a shout`
`22:09.29`	`teknoprep`	`yeah i will be here at 10pm my time`
`22:09.32`	`teknoprep`	`thats about 4 hours`
`22:09.35`	`teknoprep`	`probably later too`
`22:09.59`	`kattmat`	`fender, thanks man. but it still appears as though it can't authenticate correctly with the server for some strange reason?`
`22:12.54`	*** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
`22:13.07`	`*** join/#asterisk docelmo (n=vircuser@c-69-138-91-104.hsd1.de.comcast.net)`
`22:14.04`	`*** join/#asterisk wangster (n=wangster@static-64-201-170-129.ptr.terago.ca)`
`22:14.20`	`kattmat`	`it tries to authenticate on just number@proxy.digisip.net`
`22:14.33`	`kattmat`	`when i want number:password:accountid@proxy.digisip.net`
`22:14.34`	`kattmat`	`how to?`
`22:14.46`	`wangster`	`Any docs on how to use the new jitterbuffer in 1.4? I seem to see any sign that its working on the console.`
`22:15.14`	`wangster`	`" don't see any sign" is what i ment.`
`22:16.04`	`*** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw)`
`22:19.36`	`*** join/#asterisk sjsobol (n=sjsobol@amethyst.justthe.net)`
`22:19.44`	`sjsobol`	`hey`
`22:22.56`	`TripleFFFF`	`anyone need a te410p ?`
`22:24.41`	`*** join/#asterisk damanivu (n=damanivu@ip68-4-207-173.oc.oc.cox.net)`
`22:24.42`	`Sauron	`Hmm`
`22:24.59`	`Sauron	`Wonder how much * has changed between CVS-HEAD-03/08/05-23:05:41 and now...`
`22:26.12`	`mog`	`svn diff`
`22:26.44`	`Sauron	`rofl`
`22:29.03`	`*** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net)`
`22:29.16`	`*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)`
`22:33.36`	`dlynes_laptop`	`TripleFFFF: for free?`
`22:34.47`	`TripleFFFF`	`free ?`
`22:34.48`	`TripleFFFF`	`nah`
`22:34.52`	`TripleFFFF`	`but cheaper then new`
`22:34.56`	`TripleFFFF`	`its 1 month old..`
`22:35.13`	`TripleFFFF`	`asking around 1000 cdn .. check xchange rate on xe.com`
`22:35.24`	`TripleFFFF`	`not sure.. prolly 900 .. so save 500 from retail`
`22:36.22`	`Sauron	`Hmm..`
`22:36.23`	`robin_sz`	`sigh ...`
`22:36.29`	`Sauron	`* is 1.4 for devel and 1.2 for stable?`
`22:36.31`	`Sauron	`Phew`
`22:37.42`	`TripleFFFF`	`Sauron somethign liek that.. dwhere you killed when they destroyed the ring ?`
`22:38.30`	`Sauron	`yawn`
`22:38.41`	`Sauron	`you know how old those jokes get after a while?`
`22:38.52`	`TripleFFFF`	`if the hat fits`
`22:39.07`	`dlynes_laptop`	`man...that chick in the new p. diddy video is damned hot`
`22:39.14`	`TripleFFFF`	`i get people asking me DAILY what happened to my DoubleFF nick`
`22:39.26`	`dlynes_laptop`	`you mean TripleFFFFFFFFFFFFFF?`
`22:39.37`	`TripleFFFF`	`heheh dlynes_laptop i knew youd remmeber ;)`
`22:40.06`	`TripleFFFF`	`im still wondering why no one tried a bot that register every combo ...`
`22:40.21`	`*** join/#asterisk oQPa (i=Spain@236.Red-83-38-109.dynamicIP.rima-tde.net)`
`22:40.25`	`justinu\|laptop`	`better things to do? :P`
`22:40.37`	`teknoprep`	`any suggestion on daylight savings time config and the spa-942 ?`
`22:40.52`	`TripleFFFF`	`yep.. but youd be amazed. i was debuggin a client the other day.. remote desktoped and saw..`
`22:41.08`	`dlynes_laptop`	`you mean you were buggin' a client?`
`22:41.09`	`TripleFFFF`	`that she was playing cards.. now that nice corp salary wasted`
`22:41.31`	`dlynes_laptop`	`Nicole Schwerzinger`
`22:41.34`	`TripleFFFF`	`hehehe not really. but i was annoyed yeah..`
`22:41.37`	`dlynes_laptop`	`damn, she's a hotty`
`22:41.59`	`TripleFFFF`	`pussy cat doolls ?`
`22:42.19`	`dlynes_laptop`	`don't know...is she?`
`22:42.26`	`TripleFFFF`	`yes.. singer`
`22:42.28`	`dlynes_laptop`	`ah`
`22:42.29`	`TripleFFFF`	`google says`
`22:42.39`	`dlynes_laptop`	`anyways....major droolage :0`
`22:43.06`	`TripleFFFF`	`got a link im trying to find`
`22:43.28`	`*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)`
`22:44.27`	`TripleFFFF`	`oh`
`22:44.37`	`TripleFFFF`	`Scherzinger its spelled`
`22:44.48`	`dlynes_laptop`	`TripleFFFF: yeah...check out that 'Come to Me' video`
`22:45.03`	`dlynes_laptop`	`TripleFFFF: P Diddy Combs f. Nicole Schwerzinger`
`22:45.42`	`fiber0pti`	`Anyone want to try a new operator panel for Asterisk? Any feedback and beta testers are welcome.`
`22:45.59`	`TripleFFFF`	`http://ca.askmen.com/specials/2006_top_99/lindsay-lohan-18.html`
`22:46.43`	`Adam12`	`triplef: I like the first picture! :)`
`22:46.45`	`TripleFFFF`	`klum is hot too`
`22:46.49`	`dlynes_laptop`	`fiber0pti: sure`
`22:46.59`	`dlynes_laptop`	`fiber0pti: windows or linux based?`
`22:47.14`	`fiber0pti`	`dlynes_laptop: The client is windows based and the server will work on windows or linux.`
`22:47.15`	`Sedorox`	`fiber0pti: I'd be willing to check it out.. dunno when I would be able to tho.. probably not till weekend`
`22:47.23`	`TripleFFFF`	`omg.. people are nuts.. i think its reversed`
`22:47.42`	`fiber0pti`	`The server and client can be downloaded from i9technologies.com/isymphony`
`22:47.49`	`fiber0pti`	`We just got done with the new user interface.`
`22:47.53`	`dlynes_laptop`	`TripleFFFF: dood`
`22:48.04`	`dlynes_laptop`	`TripleFFFF: nicole is like 100x better than lindsay lohan`
`22:48.08`	`dlynes_laptop`	`fiber0pti: thanks`
`22:48.31`	`TripleFFFF`	`yeah but dumber.. and angelina is #3.. that nice.. you know Jolie = Beautifull in french`
`22:48.55`	`TripleFFFF`	`and our litle jessica alba is #1.. oh well`
`22:49.27`	`TripleFFFF`	`man i would sue having put this pic on a site.. http://ca.askmen.com/women/galleries/actress/jessica-alba/picture-1.html`
`22:49.32`	`fiber0pti`	`dlynes_laptop: Don't hesitate to post in the forum or e-mail us if you need any installation help. Contact information can be found on the website`
`22:50.03`	`Sedorox`	`TripleFFFF: why? pics like that happen all the time`
`22:52.01`	`TripleFFFF`	`hehe yeah.. but you pay people to change it ;) its not good for a top 100.. for a blog on the other hand its funny.. would you use your worst photo for a portfolio ?`
`22:52.02`	`TripleFFFF`	`lol`
`22:52.19`	`Sedorox`	`lol, I wouldn't say its her worst :p`
`22:52.49`	`TripleFFFF`	`well she looks like she woke up in a jail and a huge gorilla stuck on up her a..`
`22:53.10`	`TripleFFFF`	`kind of politicaly correect smile while something is going terribly wrong`
`22:53.11`	`Sedorox`	`ahah`
`22:55.17`	`*** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net)`
`22:55.40`	`*** join/#asterisk kronic (n=gnorman@mail.stabat.com)`
`22:56.10`	`riddlebox`	`ok I have a dumb question, but if a keyboard was bad could it cause my pc to lock up?`
`22:56.46`	`justinu\|laptop`	`probably`
`22:57.51`	`*** join/#asterisk Chicago (n=Chicago@c-71-201-141-218.hsd1.il.comcast.net)`
`22:57.56`	`riddlebox`	`justinu\|laptop, its wierd I have a new motherboard and cpu and new ram in this system and about 10 minutes after I boot it locks up`
`22:59.02`	`justinu\|laptop`	`could be lots of other stuf`
`22:59.27`	`Chicago`	`I want to connect my voice modem to asterisk. I have a telephone adapter from my voip provider which I can only connect to my computer using the regular telephone line. Am I being stupid, or can I use my telephone adapter and voice modem to use this line with asterisk?`
`22:59.29`	`lesouvage`	`fiber0pti: I would like to try it on my mac is 10.2.8 Ibook`
`22:59.35`	`*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)`
`23:02.53`	`riddlebox`	`Chicago, who is your VOIP provider?`
`23:03.09`	`Chicago`	`riddlebox: I use AT&T Callvantage.`
`23:03.21`	`fiber0pti`	`lesousvage: I will be more than happy to get you a beta version for your mac. What is your asterisk server running on?`
`23:04.06`	`riddlebox`	`Chicago, most of the time you setup asterisk to connect directly to the voip provider in sip.conf, then you setup your adaptor that they gave you to connect to asterisk`
`23:04.36`	`riddlebox`	`Chicago, you do not plug the adaptor into the pc, asterisk will go out and connect to the voip provider itself`
`23:05.33`	`Chicago`	`riddlebox: I saw that Packet8 and Vonage and Free World Dialup all support that. What about AT&T CallVantage?`
`23:05.43`	`ManxPower`	`Chicago: Most providers with unlimited plans activly prevent you from using 3rd party devices to connect to them.`
`23:06.00`	`ManxPower`	`riddlebox: packet 8 is not supported.`
`23:06.18`	`ManxPower`	`vonage is also not supported unless you have their "sfotphone service" which does not have unlimited.`
`23:06.28`	`wangster`	`does anyone know how to get the sip jitter buffer running in 1.4? I've enabled the options but I see nothing on the console.`
`23:06.59`	`Chicago`	`ManxPower: Would using the telephone line out of the telephone adapter and into a voice modem work? I suppose there must be a good use for US Robotics Voice Modems...`
`23:08.24`	`*** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net)`
`23:09.21`	`riddlebox`	`ManxPower, the only thing to do then is buy a sipura adaptor that will allow you to take the analog dial tone from the voip provider and convert it back to ip so asterisk can use it right?`
`23:10.41`	`Chicago`	`Is just the fax part of the modem supported by asterisk?`
`23:11.37`	`brookshire`	`just fax passthrough i believe`
`23:15.39`	`Chicago`	`It looks like there are voice modem channel drivers.`
`23:15.42`	`wangster`	`Chicago: there are many providers that do support asterisk directly so no point in jumping through hoops to get it working via an adapter.`
`23:25.58`	`ManxPower`	`Chicago: voicemodem drivers were removed from Asterisk because they did not work`
`23:26.31`	`Chicago`	`ManxPower: For real??? I know it's a full-duplex US Robotics voice modem speakerphone.`
`23:26.42`	`*** join/#asterisk kattmat (n=kattmat@c213-100-64-47.swipnet.se)`
`23:26.45`	`ManxPower`	`Chicago: For real.`
`23:26.53`	`ManxPower`	`If you want to write new ones, feel free.`
`23:27.55`	`ManxPower`	`Anyone that wanted to use a voicemodem with Asterisk was not qualified to write or maintain the required drivers. Anyone that was qualified to write or maintain the drivers just goes out and buys a compatable card.`
`23:38.33`	`teknoprep`	`anyone here using the spa-942 and one to many intercom ?`
`23:39.07`	`teknoprep`	`the intercom puts the call on hold if someone is using there phone and then lets them hear the intercom.. i don't want this.. if someone is on a call i want them to be able to stay on the call .. this is a major bug`
`23:42.21`	`wangster`	`teknoprep: yup. look on the voip-wiki for the 942.`
`23:42.52`	`teknoprep`	`yes i know its a bug.. i saw that on the voip-wiki.. is there a fix?`
`23:44.21`	`wangster`	`teknoprep: someone posted a potential work-around which I have not tested.`
`23:44.26`	`*** part/#asterisk kattmat (n=kattmat@c213-100-64-47.swipnet.se)`
`23:45.55`	`teknoprep`	`hmmm`
`23:46.04`	`teknoprep`	`linksys says its a feature not a bug`
`23:46.05`	`teknoprep`	`LOL`
`23:46.33`	`wangster`	`yup.`
`23:46.41`	`*** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com)`
`23:46.42`	`wangster`	`Thats my posting btw ;)`
`23:47.23`	`teknoprep`	`lol`
`23:49.14`	`*** join/#asterisk keltorsori (n=keltorso@adsl-074-245-072-225.sip.msy.bellsouth.net)`
`23:50.01`	`wangster`	`teknoprep: jfyi, they have now acknowledged its a bug and "I'll put that on the sugestions list."`
`23:50.14`	`teknoprep`	`good`
`23:50.21`	`wangster`	`well, not actually that its a bug but that it could work differently.`
`23:50.29`	`teknoprep`	`heh`
`23:50.33`	`wangster`	`That was from Sep 9.`
`23:50.42`	`teknoprep`	`just putting someone on random hold isn't all that great`
`23:50.55`	`wangster`	`So I have no idea if that means it will make it into the next firmware or not.`
`23:50.58`	`teknoprep`	`how would you setup the screen differently on the spa-942?`
`23:51.07`	`teknoprep`	`is there a good user manual for the spa-942 ?`
`23:51.11`	`wangster`	`teknoprep: "isn't all that great" ?!? its freakin stupid.`
`23:51.12`	`teknoprep`	`i can't seem to find one on the linksys site`
`23:51.54`	`teknoprep`	`also... setting up a button on the screen that you just push to setup intercom without dialing the Extension would be nice`
`23:51.57`	`teknoprep`	`stuff like that`
`23:52.07`	`wangster`	`You're talking to your million a year client on the phone who is about to trade 10,000 shares of X stock and someone pages and suddenly throws the client on hold...`
`23:52.26`	`teknoprep`	`lol`
`23:52.27`	`teknoprep`	`yup`
`23:52.37`	`wangster`	`the client is listening to free-river music on hold instead of you. thats stupid.`
`23:52.44`	`teknoprep`	`LOL`
`23:52.47`	`teknoprep`	`!!!!!!!!!`
`23:53.14`	`wangster`	`I can not comprehend in what situation you would want it to work that way...`
`23:53.28`	`wangster`	`oh, btw, 1.4 of asterisk might have a better fix`
`23:53.33`	`teknoprep`	`k00`
`23:53.54`	`teknoprep`	`dude`
`23:53.57`	`robin_sz`	`whats a good log viewer CGI to go with mysql CDR`
`23:53.58`	`robin_sz`	`??`
`23:54.01`	`teknoprep`	`i just got that mx revolustion mouse from logitech`
`23:54.03`	`teknoprep`	`very f'n awesome`
`23:54.11`	`wangster`	`1.4 has something called "whisper paging". I haven't looked into it yet but my understanding is it will insert the audio into the existing conversation.`
`23:54.28`	`teknoprep`	`wow`
`23:54.33`	`teknoprep`	`thats just crazy`
`23:54.39`	`wangster`	`robin_sz: Asterisk CDR`
`23:54.42`	`robin_sz`	`bah, old Strowger exchanges did that by defualt`
`23:55.43`	`wangster`	`ya, so whats your point? phones from the 60s had lights that lit up when you were on a line. Thats a major challenge for SIP`
`23:55.57`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`23:56.10`	`dan__t`	`Why?`
`23:56.17`	`teknoprep`	`anyone else here run there asterisk inside of vmware ?`
`23:56.34`	`teknoprep`	`i am about to run a patch on asterisk.. i take a snapshot.. update it.. if it fails i revert to snapshot`
`23:56.38`	`teknoprep`	`got i love vmware`
`23:56.55`	`wangster`	`teknoprep: you can do it but you can't use any of the hardware with it.`
`23:56.56`	`robin_sz`	`wangster, no, it was a sarcastic comment that old Strowger exchanges where shit full of crosstalk between circuits`
`23:57.09`	`wangster`	`robin_sz: lol. ok.`
`23:57.16`	`teknoprep`	`wangster, i use all voip.. BUT.. you can use the USB channel banks`
`23:57.45`	`robin_sz`	`especially when a uniselector got bridged halfway between two positions ...the old "crossed line" which we dont get these days`
`23:57.54`	`wmandra`	`does anyone know if there's a way to place a bridged call on hold through the manager api??`
`23:58.01`	`wangster`	`teknoprep: there was an posting on VMware & asterisk on the Ontario Asterisk users group mailing list recently.`
`23:58.15`	`robin_sz`	`so, CGI log viewers for CDR?`
`23:58.23`	`teknoprep`	`wangster, yeah i run it in vmware.. i love it`
`23:58.33`	`teknoprep`	`have to reboot the vmware box`
`23:58.34`	`teknoprep`	`brb`
`23:58.38`	`teknoprep`	`my firewall is on it also`
`23:58.50`	`wangster`	`robin_sz: AsteriskCDR is a web log viewer for mysql_cdr.`

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