irclog2html for #asterisk on 20060530

`00:00.48`	`InfraRed`	`cool`
`00:00.51`	`InfraRed`	`thanks trixter`
`00:01.02`	`InfraRed`	`i think i wont bother tbh`
`00:01.25`	`InfraRed`	`I can get DIDs from magrathea. i suppose the 0871 is my best case for now`
`00:01.32`	`trixter`	`when you start reselling like that you end up with lower quality generally`
`00:02.14`	`InfraRed`	`its not reselling`
`00:02.33`	`trixter`	`do you own the number directly? if so then its reselling :P`
`00:03.02`	`InfraRed`	`fair point`
`00:03.03`	`trixter`	`when you get something from A to sell to B and are a middleman that is kinda the definition of reselling`
`00:03.05`	`trixter`	`:D`
`00:03.07`	`[TK]D-Fender`	`tomcontr3 : Looks like you're trying to fax over G729... NOT a good codec for faxing in as much as ANY VoIP codec is big trouble.`
`00:03.15`	`trixter`	`but I meant with the free ones`
`00:03.29`	`InfraRed`	`g729 is bad for fax :)`
`00:03.35`	`trixter`	`even though both stanaphone and ipkall are directly connected.. ipkall is even the telco (itltd)`
`00:03.37`	`InfraRed`	`711u is just about achieveable`
`00:03.45`	`trixter`	`jitter is the fax iller`
`00:03.47`	`trixter`	`killer`
`00:03.52`	`InfraRed`	`T.38 ftw`
`00:03.53`	`[TK]D-Fender`	`oops.. G726, NOT G729`
`00:04.13`	`tomcontr3`	`right...`
`00:04.16`	`tomcontr3`	`lolol`
`00:04.17`	`InfraRed`	`any codec compression is not healthy for faxing`
`00:04.17`	`trixter`	`t.38 uses 2x bandwidth and in some cases even more`
`00:04.18`	`InfraRed`	`:)`
`00:04.24`	`[TK]D-Fender`	`Still half-rate and not a harmonic compression, but still practically a seal of doom.`
`00:04.30`	`tomcontr3`	`I have been trying the hole day`
`00:04.49`	`InfraRed`	`trixter: but it gets through`
`00:04.49`	`[TK]D-Fender`	`tomcontr3 : Stop wasteing your time with Fax of SIP.`
`00:04.50`	`InfraRed`	`:)`
`00:04.53`	`[TK]D-Fender`	`over*`
`00:05.04`	`InfraRed`	`ya, you'll fail`
`00:05.10`	`tomcontr3`	`what should I do instead?`
`00:05.11`	`InfraRed`	`listen to [TK]D-Fender`
`00:05.23`	`*** join/#asterisk MoutaPT (n=MoutaPT@85.139.196.147)`
`00:05.42`	`[TK]D-Fender`	`tomcontr3 : If you want to do faxing, use a REAL line.`
`00:05.44`	`InfraRed`	`tomcontr3: look up T.38 it's about the only option for faxing over voip but like trixter suggested, it's not a free ride`
`00:05.47`	`trixter`	`I personally prefer t.37 (store and forward) from a network point of view.. I know the argument for t.38 that you lose capabilities of the end fax machine and delivery reports may not be quite what you expect, but meh`
`00:06.00`	`[TK]D-Fender`	`InfraRed : Don't expect much of him, he's using AMP anyways.`
`00:06.06`	`InfraRed`	`doh`
`00:06.16`	`InfraRed`	`you'll fail then`
`00:06.30`	`InfraRed`	`give up`
`00:06.53`	`MoutaPT`	`any one has experience with mISDN with beronet telephony carD?`
`00:06.59`	`[TK]D-Fender`	`T.38 is not always an agreed upon interoperable standard by those that claim to adhere to it...`
`00:07.07`	`trixter`	`T.37 is largely based on existing standards too, smtp and mime.. as in email attachments more or less.. way nicer on your network than t.38 :D`
`00:08.03`	`trixter`	`[TK]D-Fender: yeah why I said that it takes 2x bandwidth sometimes more.. when the t.38 reinvite is issued some devices dont stop the original codec so they have t.38 plus whatever the other was.. on top of that there are vague spots in the spec (what I think you were getting at) where different companies impleement it different, some are just lazy but the spec itself is not that good`
`00:11.03`	`[TK]D-Fender`	`trixter : So T.37 woul be better termed a "post transmission relay" which once the receiving end receives the transmission will repeat to the other side as a new call?`
`00:11.30`	`tomcontr3`	`where do I set the Fax Mode? T.38....etc..`
`00:11.31`	`tomcontr3`	`?`
`00:11.38`	`trixter`	`store and forward is what I always heard it called`
`00:13.02`	`trixter`	`yeah you have a gateway that receives typically via TDM but it can even be in the fax machine itself (ie a scanner and computer) which then using specific formatting uses smtp and mime will send over some network (maybe inet maybe not) and then at the remote end does a local call`
`00:13.10`	`trixter`	`there is a free fax gateway you can use on the internet`
`00:13.20`	`trixter`	`if you dont want anything too fancy and since its free ...`
`00:13.36`	`trixter`	`http://freefax.com/ff_whats.htm`
`00:14.56`	`InfraRed`	`i just print to it and it magically appears on the other side as fax :)`
`00:15.35`	`InfraRed`	`i call it magic`
`00:15.45`	`trixter`	`it would appear that it is`
`00:20.43`	`copland`	`does anyone know of any softphones that support g729`
`00:21.03`	`trixter`	`g729 is patented so it wont be free`
`00:21.12`	`copland`	`well yeah`
`00:21.15`	`trixter`	`becuase the person who distributes it has to pay the g729 consortium`
`00:21.19`	`trixter`	`but yes there are some`
`00:21.20`	`copland`	`but one that not too overly price`
`00:21.42`	`trixter`	`well the license only costs $0.10 in quantity`
`00:21.55`	`trixter`	`so more can do it if they choose to`
`00:22.04`	`copland`	`60 dollars for ip is a little excessive`
`00:22.28`	`copland`	`sorry`
`00:22.32`	`copland`	`<PROTECTED>`
`00:22.49`	`InfraRed`	`arsebeam`
`00:23.22`	`trixter`	`what platform do you want this softphone to run on?`
`00:23.30`	`trixter`	`that is a major component to whether or not there is one available`
`00:23.30`	`copland`	`windows`
`00:24.20`	`copland`	`I only need 6 or so channels so I will get those digium licences for g729 bought tommarow`
`00:24.48`	`copland`	`it just finding a reasonable softphone for the few people who are going to be using it outside our lan`
`00:24.56`	`[TK]D-Fender`	`InfraRed : I have 3 HP 4345 MFP's and with the DSS I can basically remote photocopy to any IP printer :)`
`00:25.03`	`mitcheloc`	`copland: eyebeam or idefisk...`
`00:25.07`	`trixter`	`too bad they are g729a only (but then asterisk cant really handle b ...) and they are hardware locked to one machine instead of floating licenses`
`00:26.24`	`InfraRed`	`l33t :)`
`00:26.29`	`InfraRed`	`this is a GP215`
`00:26.48`	`InfraRed`	`basically a photocopier + options`
`00:27.19`	`InfraRed`	`http://www.interactivesystems.co.uk/photocopiers/preused/gp215.jpg`
`00:27.44`	`[TK]D-Fender`	`InfraRed : nice set of trays on it`
`00:28.03`	`trixter`	`sounds like my ex`
`00:28.41`	`trixter`	`she had a nice set of trays`
`00:29.30`	`InfraRed`	`:)`
`00:30.24`	`MoutaPT`	`any one has here has experience with chan_misdn?`
`00:30.25`	`InfraRed`	`canon always wins`
`00:35.59`	`dlynes_office`	`Has anyone used any Konftel products with Asterisk?`
`00:36.42`	`InfraRed`	`why is debian so retarded when it comes to iptables`
`00:37.02`	`dlynes_office`	`InfraRed: iptables is pretty generic....has nothing to do with debian`
`00:37.34`	`InfraRed`	`when it comes in .deb it becomes a debian problem`
`00:37.53`	`De_Mon`	`InfraRed youre joking right?`
`00:37.54`	`InfraRed`	`the whole thing with startup scripts and config with iptbales`
`00:38.03`	`dlynes_office`	`InfraRed: so it's a debian iptables script then, not iptables :)`
`00:38.04`	`InfraRed`	`and the whole write your own rc scripts is bit silly`
`00:39.01`	`copland`	`what does everyone thing about the gsm codec`
`00:39.16`	`De_Mon`	`My computer says microsoft windows on it, does that make zonealarm's stupidness a windows problem?`
`00:39.18`	`dlynes_office`	`InfraRed: not terribly silly....if you want a consistent environment across all of your machines`
`00:39.32`	`InfraRed`	`it's pretty when it comes in purbple`
`00:39.51`	`De_Mon`	`its packaged for windows so, clearly it is!`
`00:40.12`	`InfraRed`	`it's not packaged by microsoft, when it is its a windows issue`
`00:40.13`	`dlynes_office`	`De_Mon: that made no sense, whatsoever`
`00:40.13`	`copland`	`De_Mon: yes if microsoft maintain a package for zonealarm`
`00:40.25`	`InfraRed`	`apache comes with its own startup script, exim does`
`00:40.27`	`InfraRed`	`why not iptables`
`00:40.32`	`copland`	`if iptables are foobared it because debian package maintainers foobared it up`
`00:40.36`	`InfraRed`	`it's retarded, stop defending it`
`00:40.54`	`dlynes_office`	`InfraRed: because then iptables team would need to write a script for 2000 something different linux distributions`
`00:40.57`	`dlynes_office`	`InfraRed: it's not feasible`
`00:41.16`	`InfraRed`	`i spifically said debian iptables package`
`00:41.19`	`De_Mon`	`dlynes_office you've never used a windows installer that didn't work?`
`00:41.26`	`InfraRed`	`specifically`
`00:41.27`	`copland`	`dlynes_office: no they would leave it up to the maintiners for the various distro to make a script`
`00:41.31`	`copland`	`which they normally do`
`00:41.32`	`dlynes_office`	`De_Mon: the odd time`
`00:41.51`	`dlynes_office`	`copland: that's exactly what I was saying`
`00:41.51`	`De_Mon`	`dlynes_office the same is true about debian install packages`
`00:41.53`	`copland`	`Slackwars iptables package comes with a proper script maintain by Patrick creator of slackware`
`00:42.06`	`copland`	`still iptables is stupid`
`00:42.07`	`mitcheloc`	`iptables is fine but it's not very flexible =/`
`00:42.16`	`copland`	`and ipstables as a whole can be a pita`
`00:42.16`	`*** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net)`
`00:42.18`	`dlynes_office`	`copland: yeah...I don't use pat's script though...I use shorewall...much easier to work with`
`00:42.41`	`InfraRed`	`it's clean and sensible`
`00:42.46`	`copland`	`dlynes_office: blah shorewall. PFSense/MonoWall here`
`00:43.05`	`mitcheloc`	`blah linux, windows here`
`00:43.13`	`mitcheloc`	`whoops, ignore that ^^`
`00:43.16`	`copland`	`mitcheloc: pfsense is bsd`
`00:43.16`	`InfraRed`	`bah windows, OS/2 here`
`00:43.16`	`dlynes_office`	`copland: well, whatever you're comfortable with, as long as it's secure`
`00:43.21`	`dlynes_office`	`copland: that's the main thing`
`00:43.39`	`mitcheloc`	`oooh, monowall looks ace`
`00:43.54`	`copland`	`pfsense is more active than monowall`
`00:43.57`	`dlynes_office`	`but, otoh I could probably run without a firewall and be just fine`
`00:44.15`	`copland`	`I use both byt I stop using monowall as they are slow on development`
`00:44.22`	`copland`	`PFSense team is very active`
`00:44.41`	`mitcheloc`	`pfsense looks like monowall?`
`00:44.59`	`mitcheloc`	`copland: did you ever donate money to monowall?`
`00:45.22`	`InfraRed`	`i donated to openvpn because they deserve it`
`00:45.23`	`InfraRed`	`they rock`
`00:45.56`	`InfraRed`	`i suggest everyone else do the same`
`00:46.01`	`mitcheloc`	`does aah come with some sort of pfsense style software?`
`00:46.05`	`copland`	`mitcheloc: Yes the company that I consulted did donate 200 dollars for the 5 installed we used`
`00:46.18`	`mitcheloc`	`copland: cool, okay, then i have no point`
`00:46.28`	`copland`	`AAH uses Freepbx which is a nice webgui`
`00:46.32`	`InfraRed`	`heh`
`00:46.45`	`copland`	`mitcheloc: I have also dumped money into pfsense`
`00:47.06`	`InfraRed`	`dump some money here <---`
`00:47.20`	`copland`	`I adovcate opensource but i also tell my clients that paying them at less a little something helps keep up with the updates`
`00:47.21`	`mitcheloc`	`okay, i was only going to say how do you expect them to move development quickly if they aren't makin money ;), but you already are ontop of that`
`00:47.34`	`InfraRed`	`the infra study fund, studying the effects of alcohol on human body, I volenteers`
`00:47.37`	`InfraRed`	`d`
`00:47.58`	`copland`	`copland needs a new headset fund`
`00:48.03`	`dlynes_office`	`InfraRed: get shared line appearance working on asterisk so that it's compatible with aastra and polycom phones, and i'm sure there's several of us that would dump some money your way :0`
`00:48.16`	`copland`	`my plantronics is falling apart after 3 years of hard useage`
`00:48.46`	`InfraRed`	`nobody uses aastra`
`00:48.59`	`dlynes_office`	`InfraRed: Are you kidding? Lots of people use Aastra`
`00:49.06`	`InfraRed`	`lies`
`00:49.07`	`copland`	`or add stun support to avaya 4600 series phones`
`00:49.11`	`dlynes_office`	`InfraRed: They wouldn't still be in business, otherwise`
`00:50.05`	`dlynes_office`	`The 9133i's are a pretty nice, cheap alternative to Polycom 501's`
`00:50.12`	`InfraRed`	`urban myth`
`00:50.44`	`InfraRed`	`:)`
`00:50.51`	`copland`	`I have 10 4602 Avaya phones brand new that i picked up for 36 dollars a piece at gov auction`
`00:50.59`	`dlynes_office`	`damn cheap`
`00:51.01`	`mitcheloc`	`jerk!`
`00:51.02`	`[TK]D-Fender`	`dlynes_office : And questionable vs the IP 430's ;)`
`00:51.20`	`*** join/#asterisk copantl (n=galel@207.13.77.20)`
`00:51.24`	`dlynes_office`	`[TK]D-Fender: maybe...but it remains to be seen until the pricing is out for them`
`00:51.44`	`copland`	`I own cisco stock yet i use there competiors ip phones`
`00:51.47`	`dlynes_office`	`[TK]D-Fender: i wish they'd fix their fscking software, though`
`00:51.53`	`[TK]D-Fender`	`dlynes_office : Atacomm has a price $10 below taht of the 501.`
`00:52.17`	`dlynes_office`	`[TK]D-Fender: atacomm has a $10 price below that of the 501 on what?`
`00:52.17`	`[TK]D-Fender`	`dlynes_office : Not broken for me...`
`00:52.27`	`[TK]D-Fender`	`dlynes_office : On the 430`
`00:52.42`	`dlynes_office`	`[TK]D-Fender: and how is that even remotely price competitive with the 9133i?`
`00:52.54`	`copland`	`I love these avaya phoens the only thign that i cant do with them is nat transversal`
`00:52.59`	`a1fa`	`would there be a way to limit the number of inbound/outbound minutes via (php) agi?>`
`00:53.05`	`[TK]D-Fender`	`dlynes_office : And Williams is supposed to be getting new kill pricing on them momentarily (the whole lineup)`
`00:53.25`	`a1fa`	`anybody doing pbx outsourcing?`
`00:53.28`	`InfraRed`	`eww php agi`
`00:53.29`	`dlynes_office`	`a1fa: Dr-Linux already answered you about 3 or 4 hours ago`
`00:53.30`	`[TK]D-Fender`	`dlynes_office : well atacomm is RETAIL.....`
`00:53.42`	`a1fa`	`dlynes_home: i missed it`
`00:53.45`	`[TK]D-Fender`	`a1fa : As in hosted PBX?`
`00:53.53`	`dlynes_office`	`[TK]D-Fender: i meant the 9133i's software, not the polycom software`
`00:54.05`	`a1fa`	`ok`
`00:54.11`	`a1fa`	`so it can be done via agi`
`00:54.25`	`[TK]D-Fender`	`dlynes_home : What don't you like about Aastra's? I've never had to work with them first-hand`
`00:54.36`	`dlynes_office`	`[TK]D-Fender: nothing...but their software is a bit flaky`
`00:54.47`	`dlynes_office`	`[TK]D-Fender: other than the software though, they're a great phone`
`00:55.04`	`dlynes_office`	`[TK]D-Fender: i would definitely recommend them any day over all the cheaper phones out there`
`00:55.06`	`InfraRed`	`i think nortel sells merdian phones here`
`00:55.08`	`InfraRed`	`they rock`
`00:55.15`	`[TK]D-Fender`	`dlynes_office : Yeah... that solid bell home-phone feel...`
`00:55.25`	`dlynes_office`	`InfraRed: yeah, they do, but they're digital phones, not analog or IP`
`00:55.39`	`a1fa`	`Dr-Linux : is there an example of how to limit number of minutes?`
`00:55.43`	`copantl`	`any body use varion t400P t/E1 card?`
`00:55.46`	`dlynes_office`	`[TK]D-Fender: no...more like that solid nortel office phone feel`
`00:56.03`	`InfraRed`	`dlynes_office: which model is the ip ones?`
`00:56.27`	`dlynes_office`	`InfraRed: 2000-2007`
`00:56.45`	`dlynes_office`	`InfraRed: I think they're call IP2000, IP2001, ... IP2007`
`00:57.01`	`dlynes_office`	`InfraRed: and they're not SIP`
`00:57.08`	`[TK]D-Fender`	`UNISTIM! yay`
`00:57.09`	`dlynes_office`	`InfraRed: they're Nortel's proprietary IP protocol`
`00:57.16`	`InfraRed`	`how poor`
`00:57.19`	`dlynes_office`	`yeah...what [TK]D-Fender said`
`00:57.37`	`dlynes_office`	`I think you can do SIP on them as well`
`00:57.46`	`dlynes_office`	`But you won't be able to do 100% of their features using it`
`00:57.57`	`[TK]D-Fender`	`the BCM is a Frankenstein's Monster inspired PBX.`
`00:58.03`	`dlynes_office`	`If you want full access to their features, you need UNISTM`
`00:58.08`	`dlynes_office`	`BCM is horrible`
`00:58.22`	`[TK]D-Fender`	`dlynes_office : SIP.. if you want to pay HUNDREDS per channel to use it for friggen licensing...`
`00:58.26`	`dlynes_office`	`BCM 2.0 and earlier was extremely buggy, too`
`00:58.36`	`dlynes_office`	`[TK]D-Fender: ah...lol`
`00:58.55`	`dlynes_office`	`[TK]D-Fender: their lawyers have everything all figured out, to do the vendor lockin thingy :_`
`00:59.15`	`[TK]D-Fender`	`dlynes_home : when I sold my company on * ($25K) the competition was BCM ($50K) and Avaya ($45K).`
`00:59.49`	`[TK]D-Fender`	`dlynes_office : And thats with 26 Polycom IP 600's and PAYING for a provided server and software`
`01:00.01`	`trixter`	`there is an asterisk module for unistim, afaik its free but I havent verified`
`01:00.19`	`copantl`	`i need to change a tormenta II card from t1 to E1, any idea?`
`01:00.29`	`dlynes_office`	`trixter: yeah, you're correct, but it's major alpha state, and only supports a small fraction of the features`
`01:01.00`	`dlynes_office`	`InfraRed: oh yeah..that's what it was I2000-2007`
`01:01.05`	`dlynes_office`	`not IP`
`01:01.17`	`copantl`	`hello?`
`01:01.32`	`[TK]D-Fender`	`dlynes_office. Yes, UNISTIM is IP, just not SIP/RTP`
`01:01.44`	`[TK]D-Fender`	`dlynes_office : Much more like MGCP....`
`01:01.49`	`trixter`	`its nortel proprietary like iax2 is digium proprietary`
`01:01.49`	`trixter`	`:P`
`01:01.51`	`dlynes_office`	`[TK]D-Fender: no...i meant the phone models`
`01:01.56`	`dlynes_office`	`[TK]D-Fender: not the protocol`
`01:02.15`	`a1fa`	`http://www.dynx.net/ASTERISK/AGI/CCARD/agi-ccard.agi`
`01:02.19`	`dlynes_office`	`[TK]D-Fender: i.e. the fancy looking phones with the super fancy looking displays that run at about $800 or $900`
`01:02.20`	`a1fa`	`anybody using this application?`
`01:04.21`	`dlynes_office`	`Here's an advertisement for a Nortel i2007 if anyone's interested: http://atlasphones.stores.yahoo.net/noi2inte.html`
`01:05.20`	`mitcheloc`	`damn...`
`01:05.33`	`mitcheloc`	`i want one =)`
`01:05.42`	`dlynes_office`	`it basically looks like an ipaq version of a voip phone`
`01:06.44`	`dlynes_office`	`anyways....i'm out of here`
`01:06.51`	`dlynes_office`	`time to go home`
`01:06.53`	`a1fa`	`[TK]D-Fender : you think it would be safe to run calling card AGI script?`
`01:08.23`	`[TK]D-Fender`	`a1fa : Have you searched for imbedded "rm -rf /" ;)`
`01:08.32`	`[TK]D-Fender`	`a1fa : write your own....`
`01:09.35`	`a1fa`	`nah`
`01:09.41`	`a1fa`	`i mean, you know`
`01:10.46`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`01:12.01`	`*** join/#asterisk operat0r (i=operator@adsl-152-132-93.asm.bellsouth.net)`
`01:12.38`	`operat0r`	`Anybody use FWD / IAX they seem to be having issues ?`
`01:13.04`	`KaBewM`	`heh, i gave up`
`01:13.13`	`KaBewM`	`witched to SIP FWD`
`01:13.20`	`*** join/#asterisk pabst (n=phil@c-71-200-102-241.hsd1.md.comcast.net)`
`01:13.23`	`operat0r`	`KaBewM humm`
`01:13.32`	`KaBewM`	`er switched`
`01:13.33`	`pabst`	`Any problems with CVS, or has it gone away?`
`01:13.40`	`dlynes_office`	`~cvs`
`01:13.46`	`jbot`	`methinks cvs is concurrent versions systems. more info here http://www.cvshome.org/. The asterisk CVS is no more. Please see svn.`
`01:13.53`	`dlynes_office`	`~svn`
`01:13.56`	`pabst`	`Make sense`
`01:13.57`	`pabst`	`:)`
`01:14.04`	`operat0r`	`KaBewM can you pastebin your sip.conf ? I have not tried sip`
`01:14.07`	`dlynes_office`	`stupid jbot`
`01:14.28`	`pabst`	`sorry, sure the question has come up multiple times now.`
`01:15.10`	`dlynes_office`	`yep...ever since they shut down the cvs server about a week ago`
`01:15.17`	`pabst`	`has anyone written an update script using SVN yet?`
`01:15.21`	`dlynes_office`	`Nobody ever seems to read the announcements list`
`01:15.38`	`dlynes_office`	`And when they switched to svn well over a year ago, they said they'd be phasing out cvs`
`01:15.46`	`dlynes_office`	`so everyone's had at least a year's warning`
`01:15.59`	`KaBewM`	`http://pastebin.ca/59767`
`01:16.10`	`dlynes_office`	`pabst: how about svn update?`
`01:16.25`	`pabst`	`haha... nice... I guess I should get on the lists...`
`01:16.28`	`dlynes_office`	`pabst: all commands that you're used to in cvs will also work on svn`
`01:16.31`	`operat0r`	`KaBewM Thanks`
`01:16.43`	`operat0r`	`do I leave extentions.conf the same ?`
`01:16.54`	`dlynes_office`	`pabst: yeah...at a minimum, you should subscribe to asterisk-announcements`
`01:17.06`	`dlynes_office`	`pabst: then you'll know when new versions come out and that kinda thing`
`01:17.07`	`KaBewM`	`its gotta say SIP/${EXTEN}@fwdnet`
`01:17.23`	`pabst`	`I will do that... I just started with asterisk about a month ago... loveing it, so i guess it is time i use the lists`
`01:17.47`	`dlynes_office`	`pabst: the users, dev, biz, and commits list are all extremely high volume`
`01:18.01`	`dlynes_office`	`pabst: like on the order of probably 500 new emails per week`
`01:18.34`	`pabst`	`yeah, ill avoid those, unless there is any greatly valuable information in them... or a digest version... I don't have time to add 500 emails to my list of reading per week :)`
`01:19.07`	`dlynes_office`	`pabst: i just archive them myself`
`01:19.22`	`dlynes_office`	`pabst: and when i get time, sort them out into subfolders according to content`
`01:19.35`	`dlynes_office`	`pabst: that way when i'm looking for particular types of information i know where to look`
`01:19.41`	`pabst`	`right...`
`01:19.53`	`pabst`	`sounds like a job for my secretary :P`
`01:20.36`	`operat0r`	`KaBewM http://pastebin.com/745993 can you correct what I I need ?`
`01:21.24`	`KaBewM`	`no`
`01:21.39`	`pabst`	`im concerned that since I downloaded the source manually, that running SVN right now will mess things up, or no?`
`01:21.57`	`operat0r`	`KaBewM ill check fwd forums for sip.conf configure thx`
`01:22.01`	`KaBewM`	`k`
`01:22.11`	`dlynes_office`	`pabst: just svn into a separate directory if you want to be on the safe side`
`01:22.13`	`operat0r`	`or if you know a url for it`
`01:22.24`	`dlynes_office`	`pabst: i.e. do an svn co asterisk-1.2.7.1`
`01:23.24`	`dlynes_office`	`anyways....i'm going home for sure now`
`01:23.25`	`dlynes_office`	`laters peeps`
`01:23.36`	`operat0r`	`KaBewM how long has iax been down for you ?`
`01:24.36`	`KaBewM`	`not sure, it worked about a year ago`
`01:25.04`	`KaBewM`	`then it got really intermittent and i switched`
`01:26.27`	`pabst`	`but when i compile from that isn't that going to overwrite my exisiting install? or i guess that won't matter if i don't make samples huh? (sorry to sound dumb, i used to use the asterisk-update.pl script that was written by Steve Szmidt)`
`01:29.21`	`pabst`	`i guess being that i am currently on 1.2.7.1, now would be a good idea to get my system running using SVN too...`
`01:31.24`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`01:36.58`	`*** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org)`
`01:38.19`	`*** join/#asterisk jahani (n=k@41.250.32.254)`
`01:38.19`	`operat0r`	`Ok so ipkall > FWD works because I see it listed in missed calls. How do I check to see if asterisk connected to fwd OK ? I ran asterisk in -vvvvv called my number and got busy sig`
`01:38.52`	`operat0r`	`and nothing logged by asterisk`
`01:40.02`	`trixter`	`ipkall will go without fwd`
`01:40.10`	`trixter`	`which is a little less latency and one less thing to break`
`01:41.34`	`operat0r`	`then what is the point in fwd ? I am just following a tut but I am running BSD http://www.techcentric.org/episodes/ep1/ep1-notes.html`
`01:42.10`	`operat0r`	`trixter I am using sip insted of iax because FWD IAX is down`
`01:42.35`	`[TK]D-Fender`	`operat0r : Please pastbin your entire sip.conf file masking PW's`
`01:42.39`	`[TK]D-Fender`	`~pb`
`01:42.48`	`jbot`	`pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`01:44.29`	`operat0r`	`[TK]D-Fender http://www.rmccurdy.com/stuff/asterisk/sip.conf`
`01:45.23`	`trixter`	`operat0r: even more reason to get ipkall to go direct`
`01:45.33`	`trixter`	`they will terminate sip directly to your box`
`01:45.55`	`trixter`	`then any fwd problems or latency that is introduced by hopping through another network are gone`
`01:46.29`	`pabst`	`biggest frustration - finding a local DID`
`01:47.18`	`[TK]D-Fender`	`operat0r : Your * box behind NAT?`
`01:47.42`	`operat0r`	`[TK]D-Fender Yes but static NAT for my BSD box`
`01:48.16`	`[TK]D-Fender`	`operat0r : So it only has a private IP?`
`01:49.57`	`operat0r`	`[TK]D-Fender Yes I could try passthrough later.`
`01:50.46`	`trixter`	`pabst: local free or local pay?`
`01:51.01`	`pabst`	`either or, i just need it local to Ocean City MD`
`01:51.03`	`[TK]D-Fender`	`operat0r : That is your problem. You have not put in any of the settings * needs to work from behind NAT`
`01:51.13`	`pabst`	`the best I could find was 2.49p/m incomming`
`01:51.14`	`trixter`	`voxbone.com $7.50/mo for 2 channels`
`01:51.21`	`operat0r`	`trixter I and just following tut for incoming calls. I do have the free sip from sipdiscount.com`
`01:51.30`	`pabst`	`im half tempted just to get a PRI`
`01:51.35`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`01:52.10`	`trixter`	`operat0r: its not free, the minutes expire after 120 days (if you use their client you will see that) so its basically $0.10/day`
`01:52.15`	`operat0r`	`[TK]D-Fender Ok I will check forums for NAT`
`01:52.38`	`[TK]D-Fender`	`operat0r : trash all that commented out junk and start atting in step by step`
`01:52.51`	`[TK]D-Fender`	`operat0r : Not the forums... the Wiki. Nice & concise`
`01:53.00`	`trixter`	`and they limit outbound calls now 7 fday running average`
`01:53.21`	`trixter`	`it was only 'free' to get customres, now that they have enough they are tightening down`
`01:53.56`	`*** join/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net)`
`01:57.42`	`*** join/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au)`
`01:58.35`	`sevard`	`man, it sucks balls that you can only have one CNAM tied to a DID`
`02:00.04`	`trixter`	`that depends`
`02:00.17`	`trixter`	`technically you can have more if you have the right access`
`02:01.17`	`sevard`	`do you know of any telco that would provide that sort of option?`
`02:01.29`	`mosty`	`i'm having trouble upgrading my install from asterisk 1.0 to 1.2- I have a SIP provider routing DID's to me, which works with 1.0 but not with 1.2. at the moment i have the 1.0 machine receiving the sip call and dial'ing it to the 1.2 machine, which works`
`02:01.55`	`mosty`	`but i would like to remove the asterisk 1.0 machine altogether. what could be wrong that i am missing?`
`02:02.14`	`trixter`	`options in sip.conf most likely`
`02:02.30`	`trixter`	`but since you havent provided much information on what is not working other than 'it' its really hard to say`
`02:03.21`	`mosty`	`trixter: ok, it's hard to describe, because somebody else was playing with settings while i was testing things. i will retrace my steps and get some more detailed info`
`02:04.53`	`*** join/#asterisk RF_MIA (n=unknown@24-55-227-232.miamfl.adelphia.net)`
`02:06.03`	`*** part/#asterisk marcus2 (i=marcus@atlantis.outer.org)`
`02:06.10`	`operat0r`	`http://www.rmccurdy.com/stuff/asterisk/sip.conf ?`
`02:10.23`	`[TK]D-Fender`	`operat0r : http://pastebin.ca/59780`
`02:11.49`	`RF_MIA`	`Anyone have any experience with the res_snmp module?`
`02:12.05`	`*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)`
`02:13.07`	`operat0r`	`[TK]D-Fender I am running asterisk no erros but number busy / asterisk not doing anything`
`02:14.17`	`operat0r`	`shoulw I see things ins asterisk -vvvvv -c if its working when I call the number ?`
`02:14.51`	`RF_MIA`	`yes. you should see something operator..even if it is an error`
`02:17.25`	`[TK]D-Fender`	`operat0r : What are you forwarding to *?`
`02:17.26`	`operat0r`	`no erros I can see for sip http://www.rmccurdy.com/stuff/asterisk/log1.txt but it does not appear FWD is talking to my server`
`02:18.54`	`[TK]D-Fender`	`operat0r : And you'll need to turn on SIP debugging "sip debug"`
`02:20.57`	`operat0r`	`[TK]D-Fender I am just seeing if it "does stuff" when I call the nubmer for now`
`02:21.29`	`*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)`
`02:24.44`	`[TK]D-Fender`	`....does stuff? And since you're still behind nat, what ports have you forwarded to *?`
`02:24.59`	`litage`	`if extensions.conf is huge (1000+ lines), will that significantly impact starting and/or reloading asterisk?`
`02:26.26`	`mitcheloc`	`who was it that uses pfsense?`
`02:26.51`	`mitcheloc`	`ah it was copland`
`02:27.00`	`mitcheloc`	`~seen copland`
`02:27.14`	`jbot`	`copland <n=stonecol@209.216.65.10> was last seen on IRC in channel #asterisk, 1h 34m 20s ago, saying: 'I love these avaya phoens the only thign that i cant do with them is nat transversal'.`
`02:27.15`	`RF_MIA`	`FreeBSD`
`02:27.23`	`operat0r`	`[TK]D-Fender I have static NAT enabled so all ports http://www.rmccurdy.com/stuff/asterisk/log2.txt`
`02:29.08`	`operat0r`	`[TK]D-Fender by does stuff I mean when I call my number it talks to my asterisk server then I plan to get a soft phone to work with it`
`02:30.32`	`mosty`	`ok, when i dial one of my DID's that I have with a sip provider, the sip extension rings, but when i pick it up neither end hears anything, and about 5-10 seconds later both ends get an engaged signal. this only happens with asterisk 1.2, with 1.0 it works fine. if i receive the calls with asterisk 1.0 and then forward from that machine to the 1.2 machine the call works fine. what could be wrong?`
`02:31.05`	`mosty`	`sip.conf on both asterisk machines is essentially the same (general section is the same), just the accounts on each box differ`
`02:31.29`	`[TK]D-Fender`	`operat0r : Ok, looks like you'r registering....`
`02:31.50`	`mosty`	`the sip.conf on the 1.2 machine forwards sip calls from the 1.0 machine to the same context of the dialplan that calls coming direct from my did sip provider does`
`02:32.17`	`mosty`	`is this a bug in asterisk 1.2? what should i check before submitting a bug report?`
`02:32.21`	`operat0r`	`[TK]D-Fender I guess I need to fix extentions.conf`
`02:32.41`	`operat0r`	`or should it do stuff if I call even it extentoins.conf is wrong`
`02:32.48`	`[TK]D-Fender`	`operat0r : that MIGHT just helpa little ;)`
`02:33.18`	`*** part/#asterisk RF_MIA (n=unknown@24-55-227-232.miamfl.adelphia.net)`
`02:33.28`	`[TK]D-Fender`	`operat0r :Don't expect to see much of anything (outside of debug) when a call comes in if your extensiosn.conf isn't ready`
`02:34.19`	`*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)`
`02:37.39`	`operat0r`	`[TK]D-Fender http://www.rmccurdy.com/stuff/asterisk/extensions.conf`
`02:38.06`	`mosty`	`anyone have any suggestions for me?`
`02:39.32`	`*** join/#asterisk Splat (n=Splat@220-253-102-19.TAS.netspace.net.au)`
`02:43.14`	`[TK]D-Fender`	`operat0r : Make sure your context matches the ones refernced in sip.conf`
`02:44.06`	`[TK]D-Fender`	`And remove the * from exten => 774851,1,Dial(SIP/*${EXTEN}@fwd1)`
`02:45.48`	`mosty`	`i have a sip provider that works with asterisk 1.0 but not with 1.2 using the same sip.conf, however receiving the sip calls on the 1.0 machine and forwarding them to the 1.2 machine works fine. what could be going wrong here?`
`02:48.00`	`[TK]D-Fender`	`most : Show us the setup.`
`02:48.33`	`mosty`	`my setup is quite large, is there a specific part i can show you?`
`02:49.12`	`operat0r`	`[TK]D-Fender here is what I got so far I am just trying to get to "do stuff" when I call my number then I will try to get something to actualy ring http://www.rmccurdy.com/stuff/asterisk/`
`02:50.27`	`operat0r`	`[TK]D-Fender Thx for your time I will leave you alone for a bit and figure out soft phones`
`02:50.56`	`mds2`	`I've got a peer in my SIP registry which goes from "Registered" to "Request Sent" state every few days. 'sip reload' brings it back to life every time. Both ends have registertimeout set to 120s. Any ideas what else might cause that behaviour?`
`02:52.06`	`[TK]D-Fender`	`operat0r : You didn't change the context in sip.conf......`
`02:52.19`	`[TK]D-Fender`	`operat0r : it needs to point to the one you created in extensions.conf`
`02:52.29`	`*** part/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net)`
`02:53.18`	`operat0r`	`[TK]D-Fender you see my entire extensions.conf that's all I have in it`
`02:53.55`	`operat0r`	`I not sure what I need to add or change`
`02:54.43`	`[TK]D-Fender`	`operat0r : in sip.conf you should change the line saying "context=pickoneplease" to "context=fwd" to match your extensions.conf. that is where calls will LAND. if they don't have anywhere to land POOF... nothing.`
`02:54.51`	`operat0r`	`I can't wait to toy with agi`
`02:55.16`	`a1fa`	`lol`
`02:55.29`	`a1fa`	`operat0r : you can't get context to work, and you want to toy with agi..`
`02:55.32`	`[TK]D-Fender`	`operat0r : believe me you are a while away from there. I can see we are starting from the very bottom up... you really should start working with getting 2 softphones talking first...`
`02:55.39`	`operat0r`	`[TK]D-Fender ok I get it`
`02:56.20`	`[TK]D-Fender`	`operat0r : Not jumping down your throat, just don't let the excitement overwhelm you :)`
`02:56.38`	`operat0r`	`I have endless docs but I have I hard time finding non a@h tutirlas that are itiot proof for noobs`
`02:58.10`	`[TK]D-Fender`	`operat0r : Download TEHBOOK...`
`02:58.13`	`[TK]D-Fender`	`~book`
`02:58.21`	`jbot`	`book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`02:59.22`	`a1fa`	`hehhe`
`02:59.28`	`a1fa`	`[TK]D-Fender`
`02:59.54`	`*** join/#asterisk IceManRISK (n=kart@201.66.47.72)`
`03:00.33`	`operat0r`	`with sipdiscount.com I shoud be able to make any US call for free ( assuming i live in us )`
`03:00.38`	`a1fa`	`what do you think about http://www.dynx.net/ASTERISK/AGI/ccard/agi-ccard.agi`
`03:00.39`	`[TK]D-Fender`	`And the is no such thing as "idiot-proof" because we all know how gosh-darned clever idiots can be...`
`03:01.34`	`[TK]D-Fender`	`operat0r .... um yeah... for 1 minute......`
`03:01.52`	`a1fa`	`[TK]D-Fender : what do you think`
`03:01.55`	`a1fa`	`bout that script`
`03:02.12`	`operat0r`	`ok I read some place for like 12USD it lets you call longer ? or ?`
`03:02.41`	`a1fa`	`$19.99/month`
`03:02.43`	`a1fa`	`unlimited`
`03:02.48`	`a1fa`	`usa/canada`
`03:02.51`	`trixter`	`sipdiscount and all finerea voip stuff let you with 10 EUR (about $12) make 1 hour calls`
`03:03.01`	`[TK]D-Fender`	`a1fa : Not a Perl guy and never actually did AGI`
`03:03.07`	`trixter`	`the money expires after 120 days, you get so many hours on a running 7 day average`
`03:03.19`	`a1fa`	`[TK]D-Fender : would it be dirty to control time available via agi?`
`03:03.25`	`trixter`	`internetcalls.com, voipbuster.com, etc are all the same company`
`03:03.37`	`trixter`	`pick the one that gives you the best routes to where you call :)`
`03:03.45`	`blitzrage`	`unlimted? whats the softcap? :)`
`03:03.54`	`operat0r`	`So there is not a free outgoing semi unlimited or cheap`
`03:03.55`	`a1fa`	`probably 2000 minutes`
`03:04.07`	`a1fa`	`fuck`
`03:04.12`	`a1fa`	`i whish there was a free incomming`
`03:04.32`	`[TK]D-Fender`	`a1fa : Set(TIMEOUT(absolute)=${timeleft})`
`03:04.35`	`trixter`	`sipdiscount isnt unlimited, the TOS of fineras companies recently changed, its a running 7 day average`
`03:05.05`	`blitzrage`	`why do you need unlimited? rarely do you really need an unlimited plan :) Per-minute is usually cheaper depending on usage.`
`03:05.09`	`[TK]D-Fender`	`a1fa : Overall I'm not going to be of much use to you in trying to become a CC telco...`
`03:05.12`	`a1fa`	`[TK]D-Fender : ?`
`03:05.13`	`trixter`	`but it works out to about $3/mo for service, and most residential users have no problems`
`03:05.21`	`a1fa`	`[TK]D-Fender : i am not gonna do cc`
`03:05.29`	`a1fa`	`[TK]D-Fender : i need to limit my friends account`
`03:05.34`	`operat0r`	`All I want really is to get incomming scripts to work`
`03:05.40`	`trixter`	`although fineras companies have some issues calling north american tollfrees, but trxtel.com lets you do that totally free`
`03:05.47`	`trixter`	`dont even register with em just send calls`
`03:05.58`	`a1fa`	`[TK]D-Fender : to 200 minutes a month.. he is paying me $5 to park his extension on my asterisk box`
`03:06.13`	`trixter`	`trxtel.com will even pay people to send them tollfree calls, as opposed to most companies trying to get money from customers they are backwards :)`
`03:06.26`	`*** join/#asterisk twisla (i=twisla@lutin.jard.in)`
`03:07.03`	`operat0r`	`So I guess jst give up on free semi unlimited outgoing local`
`03:07.14`	`operat0r`	`unless I can get a free local DID ?`
`03:07.43`	`trixter`	`having a did doesnt guarantee free outbound`
`03:07.49`	`trixter`	`its easy enough to seperate incoming from outgoing`
`03:08.19`	`operat0r`	`What makes incomming free and outgoing not free... just demand`
`03:08.32`	`[TK]D-Fender`	`operat0r : Free rides don't happen.....`
`03:08.43`	`trixter`	`largely cost`
`03:08.52`	`trixter`	`it costs money to send calls to other phone companies`
`03:09.12`	`operat0r`	`[TK]D-Fender ok so whats the catch with ipkall > FWD > my server ?`
`03:09.16`	`a1fa`	`what company sells $2 per did + 0.02 per call?`
`03:10.58`	`[TK]D-Fender`	`a1fa : www.vapourware.com`
`03:11.12`	`a1fa`	`thats not it`
`03:11.30`	`trixter`	`someone might have as a loss leader to get customers`
`03:12.03`	`operat0r`	`so am I wrong that I can setup asterisk for free incomming calls to run crazy AGI scripts ?`
`03:12.08`	`trixter`	`finera did that with their voip stuff then after they got enough customers they lock it down and reduce the quantity of minutes or increase costs or ...`
`03:12.27`	`trixter`	`operat0r: yes you can if you have someone that will give you free incoming`
`03:12.35`	`trixter`	`ipkall.com stanaphone.com and others give free DIDs`
`03:12.51`	`trixter`	`trxtel.com is gearing up to give national inbound free but they arent there yet`
`03:13.18`	`operat0r`	`trixter so the catch on outgoing is somebody said above is that phone Co's charge to take a call ?`
`03:13.33`	`trixter`	`basically yes`
`03:13.55`	`a1fa`	`dude`
`03:14.03`	`trixter`	`if you only need occasional outbound companies like plainvoip.com is $0.009/min and you can specify your caller id to match your free DID from somewhere else`
`03:14.06`	`a1fa`	`i cant remember the provider name`
`03:14.44`	`a1fa`	`$2/did + 0.02min`
`03:15.01`	`*** join/#asterisk TheCops (n=henri@got.securebinary.com)`
`03:15.26`	`operat0r`	`wish I could use my cell some how ? it has free nights etc ?`
`03:15.26`	`TheCops`	`someone ever seen this error: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown), I just rebooted my server and I've got that with my X100P card.`
`03:15.27`	`a1fa`	`its is a very popular provider`
`03:15.34`	`a1fa`	`termination and origination are separate`
`03:15.51`	`trixter`	`operat0r: why dont you?`
`03:16.08`	`trixter`	`there are a variety of ways, bluetooth, if you have a sim in your phone you can get a card for your system, etc`
`03:16.14`	`TheCops`	`In my zap show status I have a red alarm, but I dont know how to get more info, I dont see the alarm in dmesg`
`03:16.18`	`operat0r`	`trixter I dont know that you can I am a total noob`
`03:16.33`	`trixter`	`now you do know`
`03:16.55`	`[TK]D-Fender`	`a1fa : .02$/min is a lot more reasonable thanyour previously mentioned $.02/CALL.`
`03:17.14`	`[TK]D-Fender`	`a1fa : And that looked like VoicePulse Connect`
`03:17.16`	`operat0r`	`hummmm so I setup a open proxy to anybody can use my phone heh ?`
`03:17.19`	`a1fa`	`nah`
`03:17.21`	`a1fa`	`its not voice pulse`
`03:17.24`	`[TK]D-Fender`	`a1fa : Which is now CHEAPR.`
`03:17.26`	`a1fa`	`it is something else`
`03:17.36`	`[TK]D-Fender`	`a1fa : Maybe Broadvoice.`
`03:17.48`	`a1fa`	`NO!`
`03:17.55`	`a1fa`	`you buy origination and termination separate`
`03:18.09`	`a1fa`	`you can originate calls.. i think limit is 25 per account`
`03:20.14`	`a1fa`	`its a famous one`
`03:20.25`	`trixter`	`aparently not that famous if no one knows who it is`
`03:20.38`	`operat0r`	`humm still not doing anything when I call my number. Just gunna read a bit more and come back tomarrow. Thanks all`
`03:20.38`	`*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)`
`03:22.07`	`a1fa`	`i have the link @ home`
`03:22.09`	`a1fa`	`err`
`03:22.11`	`a1fa`	`@work`
`03:23.54`	`mosty`	`should sip invite requests stop being sent by the sip source when the destination has answered?`
`03:27.36`	`operat0r`	`So bisicly you can be your own long distance provider if you have the traffic and $$$`
`03:27.47`	`a1fa`	`lol`
`03:27.50`	`a1fa`	`i am out`
`03:28.12`	`operat0r`	`so long webhost hello long distance provider ?`
`03:28.18`	`operat0r`	`heh`
`03:28.34`	`operat0r`	`I would rather write uber agi scripts`
`03:28.35`	`TheCops`	`Wow, Asterisk found my line problem, a short!`
`03:29.18`	`*** join/#asterisk x86 (n=x86@p3m/member/x86)`
`03:29.21`	`TheCops`	`asterisk console own`
`03:30.50`	`*** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com)`
`03:37.45`	`dlynes_home`	`~seen docelmo`
`03:38.09`	`jbot`	`docelmo is currently on #asterisk, last said: 'The market will close soon.. vonage is currently down another $1.85'.`
`03:38.09`	`dlynes_home`	`~seen docelm0`
`03:38.13`	`jbot`	`docelm0 <n=docelmo@55-65.126-70.tampabay.res.rr.com> was last seen on IRC in channel #asterisk, 3d 6h 22m 2s ago, saying: '~mybutt'.`
`03:38.50`	`dlynes_home`	`~mybutt`
`03:38.54`	`jbot`	`extra, extra, read all about it, mybutt is HUGE and stands for some funky stuff...`
`03:43.40`	`killfill`	`how do i hangup?..`
`03:44.01`	`killfill`	`soft hangup 1 --> 1 is not a known channel`
`03:45.04`	`[TK]D-Fender`	`killfill : try providing it an actual channel.`
`03:45.19`	`*** join/#asterisk postel_ (n=jp@unaffiliated/postel)`
`03:47.20`	`*** join/#asterisk voipaster (n=25x8supp@203.167.120.9)`
`03:47.35`	`killfill`	`:-p`
`03:48.07`	`killfill`	`[TK]D-Fender: you know.. ive set my asterisk so, that when my clone x100p card recives a call, a voicemail gets up.`
`03:48.20`	`*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)`
`03:48.41`	`killfill`	`[TK]D-Fender: the problem is, that the calles hangsup and leave the message, asterisk keeps saving to the disk`
`03:48.48`	`killfill`	`its not detecting the call has finished`
`03:49.32`	`killfill`	`http://pastebin.com/746111 <--- thats my zapata.conf`
`03:50.04`	`killfill`	`busycount=3 should do it isnit?..`
`03:50.18`	`*** join/#asterisk bmg505 (n=leon@c1-151-5.rndf.isadsl.co.za)`
`03:50.50`	`[TK]D-Fender`	`killfill : Disconnect supervision is difficult on analog lines. Busycount has nothing to do with that, its for detecting if the telco responds busy to a dial attempt.`
`03:51.06`	`[TK]D-Fender`	`killfill : AndAMP isn't exactly supported around here.`
`03:51.18`	`*** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv)`
`03:51.41`	`killfill`	`[TK]D-Fender: oh i cannot prevent this happening on analog lines?`
`03:51.58`	`killfill`	`[TK]D-Fender: (yah i know.. but this has nothing to do with amp anyway.. :-p)`
`03:52.47`	`[TK]D-Fender`	`killfill : Ask your telo to supply "disconnect supervision" on the line9typically a polarity reversal or raw cut) to signal the far end termination of the call.`
`03:54.09`	`killfill`	`ok`
`03:54.12`	`*** join/#asterisk BugKham (i=BugKham@202.8.86.168)`
`03:54.14`	`killfill`	`got it`
`03:54.48`	`BugKham`	`hi, is it possible to set call-limit in the dialplan`
`03:55.03`	`Qwell`	`BugKham: don't think so, no`
`03:55.20`	`BugKham`	`Qwell: ok`
`03:55.21`	`[TK]D-Fender`	`BugKham : Clarify...`
`03:55.47`	`[TK]D-Fender`	`BugKham : And by every means I can think of so far its "yes"`
`03:55.49`	`Qwell`	`not unless you setup a complex system which reads/writes astdb or something, on every call`
`03:56.21`	`[TK]D-Fender`	`Qwell : I wouldn't think it that complex.`
`03:56.47`	`Qwell`	`[TK]D-Fender: it would be if it ever got out of sync :p`
`03:57.00`	`BugKham`	`I can see it's possible for realtime sip`
`03:59.20`	`bkw_`	`Qwell chan groups`
`03:59.23`	`bkw_`	`you can do it`
`03:59.23`	`[TK]D-Fender`	`BugKham : AstDB can do it all.....`
`03:59.31`	`bkw_`	`no astdb sucks ass`
`03:59.42`	`bkw_`	`groupcheck`
`03:59.46`	`bkw_`	`thats the way`
`03:59.47`	`[TK]D-Fender`	`bkw_ : True, but does't devalidate the point :)`
`03:59.58`	`bkw_`	`astdb will not keep state`
`04:00.04`	`bkw_`	`if it crashes the state is stuck`
`04:00.17`	`bkw_`	`not a way to go thru trusting something like that`
`04:00.34`	`[TK]D-Fender`	`bkw_ : And we've never heard the clarification as to what exactly is being limited about these "calls" :)`
`04:01.27`	`Qwell`	`bkw_: hey..mind a quick msg?`
`04:10.20`	`Nugget`	`Qwell and bkw are enjoying a quickie.`
`04:10.31`	`Qwell`	`Nugget: still waiting for him to say yes :P`
`04:11.30`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`04:25.02`	`*** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net)`
`04:43.59`	`*** join/#asterisk FinboySlick (n=FinboySl@c207.134.200-71.clta.globetrotter.net)`
`04:44.21`	`FinboySlick`	`Anybody care to help me getting a Sangoma A200 working?`
`04:45.32`	`mosty`	`ok, i have asterisk 1.0 and 1.2 setup side by side on two machines, each accepting sip calls from a particular provider. sip.conf is nearly identical on both (general section and the sip user's section is identical), but the 1.2 machine never sees ACK messages from the remote sip provider, which soon cancels the call (neither end can hear the other end). is this a bug, and if so where should i report it?`
`04:46.26`	`*** join/#asterisk watchy (n=watchy@h236.176.255.206.cable.cmdn.cablelynx.com)`
`04:46.36`	`watchy`	`anyone wanna help me tune a tdm interface to get rid of echo?`
`04:46.45`	`watchy`	`its memorial day i thought someone might be bored`
`04:51.58`	`Nugget`	`decent chance they'll be bored and drunk. sure you want to risk it?`
`04:52.27`	`FinboySlick`	`If you were a sexy girl, that could be a good prospect ;)`
`04:52.28`	`*** join/#asterisk ThaZZa_Work (n=me@124-254-82-17-dsl.ispone.net.au)`
`04:52.32`	`ThaZZa_Work`	`Hey Al..`
`04:52.33`	`ThaZZa_Work`	`All`
`04:52.50`	`ThaZZa_Work`	`I just had one of the stranges things happen to my asterisk box.`
`04:54.14`	`watchy`	`nugget: yea`
`04:55.46`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`05:04.44`	`*** join/#asterisk sPOiDar (n=sPOiDar@border.meloff.keypoint.com.au)`
`05:06.00`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`05:09.03`	`*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)`
`05:18.24`	`*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)`
`05:36.19`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`05:37.15`	`*** part/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au)`
`05:46.27`	`*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net)`
`05:49.51`	`*** join/#asterisk syedrizwanm (n=syedrizw@203.36.198.235)`
`05:49.57`	`syedrizwanm`	`hello`
`05:50.13`	`ThaZZa_Work`	`hello`
`05:50.28`	`syedrizwanm`	`I need help conifguring an Asterisk Server on Win32`
`05:50.36`	`syedrizwanm`	`configuring*`
`05:51.12`	`ThaZZa_Work`	`sorry i am no help. haven't tried that one yet.`
`05:51.24`	`watchy`	`<PROTECTED>`
`05:51.24`	`syedrizwanm`	`you mean win32?`
`05:51.36`	`syedrizwanm`	`I think the commands are same`
`05:51.53`	`syedrizwanm`	`I basically need to know what things need to be setup to achieve what I want`
`05:52.09`	`watchy`	`you should use linux`
`05:52.25`	`syedrizwanm`	`I dont want to fall into that debate`
`05:52.44`	`watchy`	`well im gonna tell you that it probably runs like shit on windows`
`05:53.07`	`syedrizwanm`	`I need to know the steps involved in setting up Asterisk`
`05:53.13`	`syedrizwanm`	`even on LInux`
`05:53.25`	`syedrizwanm`	`I have an ISDN Card AVM Fritz Card`
`05:53.30`	`syedrizwanm`	`asterisk installed`
`05:53.34`	`syedrizwanm`	`with CAPI enabled`
`05:53.40`	`syedrizwanm`	`what do I need to do next?`
`05:53.57`	`dlynes_office`	`~seen docelmo`
`05:54.08`	`jbot`	`docelmo is currently on #asterisk, last said: 'The market will close soon.. vonage is currently down another $1.85'.`
`05:54.12`	`dlynes_office`	`~seen docelm0`
`05:54.13`	`jbot`	`docelm0 <n=docelmo@55-65.126-70.tampabay.res.rr.com> was last seen on IRC in channel #asterisk, 3d 8h 38m 2s ago, saying: '~mybutt'.`
`06:00.18`	`*** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net)`
`06:00.28`	`asterboy`	`~recipies`
`06:00.36`	`asterboy`	`~tips`
`06:00.37`	`jbot`	`from memory, tips is (Trillion Instructions Per Second) This is a rating of a REALLY FAST computer. 1 TIPS is 1,000,000,000 instructions per seccond`
`06:00.46`	`asterboy`	`~docs`
`06:00.47`	`jbot`	`[docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html`
`06:01.06`	`*** join/#asterisk Kis (i=vlad@p5080D2D4.dip.t-dialin.net)`
`06:01.08`	`asterboy`	`what is the web site for aster recipies?`
`06:01.55`	`asterboy`	`got it: AstRecipes.net`
`06:02.23`	`asterboy`	`~recipies`
`06:02.29`	`asterboy`	`~recipes`
`06:02.31`	`jbot`	`i hope you have some good ones ;)`
`06:02.53`	`mitcheloc`	`aww nv is still down =/`
`06:03.22`	`asterboy`	`jbot, recipes is also but if you don't, go here for some good ones: http://astrecipes.net/`
`06:03.24`	`jbot`	`okay, asterboy`
`06:03.29`	`asterboy`	`~recipes`
`06:03.31`	`jbot`	`i hope you have some good ones ;), or but if you don't, go here for some good ones: http://astrecipes.net/`
`06:03.54`	`asterboy`	`good enough`
`06:05.23`	`asterboy`	`find / -name convert`
`06:05.31`	`asterboy`	`opps wrong window`
`06:14.17`	`*** part/#asterisk syedrizwanm (n=syedrizw@203.36.198.235)`
`06:15.45`	`*** join/#asterisk Assid (n=assid@203.115.83.214)`
`06:23.32`	`*** join/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`06:24.55`	`TripleFFFF`	`<PROTECTED>`
`06:28.20`	`*** join/#asterisk ptblank (n=MURDER1@ca-yorbalnd-cuda2-c5a-90.anhmca.adelphia.net)`
`06:35.21`	`*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)`
`06:39.26`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`06:44.30`	`*** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg)`
`06:51.23`	`*** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`06:51.39`	`*** join/#asterisk ptblank (n=MURDER1@ca-yorbalnd-cuda2-c5a-90.anhmca.adelphia.net)`
`06:52.24`	`*** join/#asterisk cjk_ (n=cjk@80.92.64.103)`
`06:52.48`	`cjk_`	`hi, was anyone able to compile mysal support for the asterisk trunk branch`
`06:53.57`	`X-Rob_`	`Anyone with polycomm clues here?`
`06:55.06`	`watchy`	`i aint no polycom god`
`06:55.10`	`watchy`	`but i might can help`
`06:55.21`	`watchy`	`i got like 30 polys in the field deployed`
`06:58.10`	`X-Rob_`	`I got a user that can't dial *xx`
`06:58.17`	`X-Rob_`	`he's getting a busy straight away`
`06:58.22`	`X-Rob_`	`it's in his digitmap`
`06:58.35`	`X-Rob_`	`hints?`
`06:58.54`	`watchy`	`hmm`
`06:59.16`	`watchy`	`does a *xx extension exist`
`06:59.25`	`X-Rob_`	`it's not reaching asterisk`
`07:00.00`	`watchy`	`hmm`
`07:00.27`	`watchy`	`no idea i got bitches dialing *XX to access voicemail`
`07:00.46`	`X-Rob_`	`yeah`
`07:00.48`	`X-Rob_`	`it was working yesterday`
`07:00.53`	`X-Rob_`	`he's done some fiddling`
`07:00.56`	`X-Rob_`	`and it's no longer working.`
`07:00.58`	`watchy`	`hmm`
`07:00.58`	`watchy`	`wtf`
`07:01.14`	`watchy`	`how could he change anything`
`07:01.28`	`X-Rob_`	`fiddling with the config file that is`
`07:01.40`	`watchy`	`oh`
`07:01.47`	`watchy`	`put in place a default conf`
`07:01.53`	`watchy`	`and tell him quit fucking arond`
`07:01.56`	`X-Rob_`	`heh`
`07:02.02`	`X-Rob_`	`I don't know anything about 'em, that's my problem.`
`07:02.05`	`watchy`	`im tring to figure out how to make a phone ring`
`07:02.06`	`X-Rob_`	`I s'pose I'll have to buy one`
`07:02.11`	`watchy`	`even tho a bitch is on the phone`
`07:02.28`	`watchy`	`cuz these chicks aint hearing the call waiting beep shit`
`07:02.35`	`watchy`	`buy a phone?`
`07:02.47`	`*** join/#asterisk X-Gen (n=X-Gen@dsl-145-193-121.telkomadsl.co.za)`
`07:02.48`	`*** join/#asterisk voipaster (i=25x8supp@203.215.73.197)`
`07:02.55`	`techie`	`i wonder if 1.66 fixed the NAT hell issue`
`07:03.40`	`watchy`	`dunno`
`07:03.48`	`watchy`	`i know it fixed buddy list shit with 601s`
`07:03.55`	`techie`	`yeah`
`07:04.17`	`watchy`	`i sold 4 601s to a business for it`
`07:04.19`	`techie`	`pisses you off when you have a $300 phone that doesnt support STUN and the like`
`07:04.28`	`watchy`	`and come to find out it only supports like 8 folks`
`07:04.32`	`watchy`	`thank god they came with that update`
`07:04.49`	`watchy`	`techie: you know how to make the phone actually ring when someone is already on it`
`07:04.56`	`watchy`	`these secretaries arent hearing the call waiting beep`
`07:05.09`	`techie`	`601s?`
`07:05.18`	`watchy`	`yea`
`07:05.31`	`watchy`	`say someone calls and they are on the line when a 2nd call comes in`
`07:05.37`	`watchy`	`it just beeps in the earpiece`
`07:05.43`	`techie`	`how many lines you have configured?`
`07:05.46`	`watchy`	`i want the damn phone to actually ring`
`07:05.57`	`watchy`	`1 line but i have linekeys set to i think 6`
`07:06.24`	`watchy`	`so she can get like 6 calls at once`
`07:06.28`	`watchy`	`but they ring the arpiece`
`07:07.34`	`*** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it)`
`07:08.14`	`*** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-8-158.w86-207.abo.wanadoo.fr)`
`07:09.45`	`mitcheloc`	`does anyone have an opinion on freebsd and asterisk?`
`07:10.00`	`watchy`	`i think its a bad idea`
`07:10.04`	`watchy`	`freebsd is the best unix os made`
`07:10.14`	`watchy`	`but i dont think asterisk is such a good choice for stability`
`07:10.26`	`Qwell`	`it works fine on freebsd`
`07:10.43`	`mitcheloc`	`and fine = better then linux?`
`07:10.52`	`Qwell`	`fine means...fine`
`07:10.52`	`watchy`	`when i was using it in freebsd zaptell didnt work to good`
`07:11.00`	`watchy`	`has all that shit been fixed`
`07:11.16`	`mitcheloc`	`Qwell: what would be the most stable OS for asterisk then?`
`07:11.34`	`Qwell`	`mitcheloc: most of the developers use Linux, and that's what's officially supported`
`07:12.49`	`mitcheloc`	`Qwell: hmm...okay, i thought so...`
`07:13.19`	`watchy`	`i might try it in fbsd again`
`07:13.22`	`watchy`	`i use it with gentoo`
`07:14.05`	`mitcheloc`	`well, if most the developers use linux, it would make sense to stick to it then no? i just want to pick the most stable OS for the asterisk system... i've been using fedora`
`07:14.41`	`watchy`	`id use linux personally unless you need bsd`
`07:14.56`	`watchy`	`if you got major issues with fbsd with a production box you might be fucked`
`07:21.22`	`*** join/#asterisk lorinc (n=ang@caracas-4809.adsl.interware.hu)`
`07:21.27`	`*** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com)`
`07:25.03`	`*** join/#asterisk qdk_ (n=qdk@213.237.44.34)`
`07:26.00`	`stephane_`	`reboot @+`
`07:29.27`	`*** join/#asterisk _4d4m_ (n=adam@62.69.102.99)`
`07:35.50`	`*** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net)`
`07:45.02`	`*** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it)`
`07:51.03`	`zparta`	`watchy: you MIGHT be fucked if you dont know bsd`
`07:51.28`	`zparta`	`you need some knowledge to run it in production`
`07:54.06`	`*** join/#asterisk stephane_ (n=stephane@merlin.cabale.net)`
`08:00.21`	`*** join/#asterisk drunkmaster (n=lucifer@81.169.228.68)`
`08:00.28`	`drunkmaster`	`heil`
`08:00.47`	`drunkmaster`	`i have a problem with building h323 on rhel4`
`08:01.48`	`drunkmaster`	`chan_h323 whants libh323_linux_x86_r.so.1, but symbols that need to it are in ..._d.so.1 library`
`08:02.09`	`drunkmaster`	`does anybody know how to avoid this problem?`
`08:17.34`	`*** join/#asterisk assert_true (n=Sunil@59.176.58.247)`
`08:21.44`	`*** join/#asterisk kernel20 (n=kernel20@203.160.223.26)`
`08:21.48`	`kernel20`	`hi`
`08:21.57`	`kernel20`	`got problems in eyebeam`
`08:22.14`	`kernel20`	`why is it when somebody picks up the call it will hang up`
`08:22.17`	`kernel20`	`any ideas?`
`08:22.32`	`*** join/#asterisk JohnJacob (n=JohnJaco@pool-71-127-110-89.aubnin.fios.verizon.net)`
`08:23.04`	`JohnJacob`	`anyone know why I'm having trouble controlling an IVR from a meetme conference?`
`08:23.18`	`JohnJacob`	`Is this something you should be able to do?`
`08:23.44`	`JohnJacob`	`I loop an application into the meetme using the local channel...`
`08:23.52`	`JohnJacob`	`and I hear the application in the meetme...`
`08:23.58`	`*** join/#asterisk parag7732 (n=root@de1-b15475.alshamil.net.ae)`
`08:24.12`	`kernel20`	`why is it when somebody picks up the call it will hang up`
`08:24.18`	`JohnJacob`	`but I can't get the application to read any DTMF from the conference partipants`
`08:24.27`	`kernel20`	`this is using eyebeam`
`08:24.30`	`kernel20`	`any ideas?`
`08:24.41`	`kernel20`	`but if i use xlite, there is no problem`
`08:24.45`	`kernel20`	`all went fine`
`08:25.24`	`kernel20`	`but when i use eyebeam when i call an account using xlite it will hang up`
`08:25.26`	`parag7732`	`I am using busydetect and busy count for hangup detection !!! I don't think so it's a good idea !!! but our isp doesn't provide hangup on detection..`
`08:25.29`	`parag7732`	`so what should I do`
`08:25.56`	`parag7732`	`I mean reverse polarity`
`08:26.04`	`*** join/#asterisk pbx1 (n=pbx1@58.69.102.72)`
`08:26.15`	`parag7732`	`reverse polarity is not provided by our server`
`08:26.56`	`kernel20`	`?`
`08:33.30`	`*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)`
`08:34.01`	`littleball`	`hello, how can i remove the last character from the dialed exten? example, the user key in 12345678#, i need to remove the #`
`08:34.08`	`*** join/#asterisk cfh (n=luca@host194-20.pool21757.interbusiness.it)`
`08:37.15`	`operat0r`	`Hello I think I got registerd to FWD how do I test it ?`
`08:37.21`	`tzafrir`	`${EXTEN:-1} ?`
`08:37.24`	`X-Rob_`	`littleball, http://www.voip-info.org/wiki-Asterisk+variables`
`08:37.25`	`X-Rob_`	`RTFM`
`08:37.53`	`operat0r`	`i tried *CLI> dial 613`
`08:38.00`	`kernel20`	`anybody here have use eyebeam?`
`08:39.09`	`cfh`	`hi all, i have an asterisk server with beronet on NT mode connect to a alcatel 4200 PBX and all it works good but when i try to make a call from a voip telephone to traditional phone, on my voip phone i cant heard the ring tone and the call then is established correctly.(on my server i see "mISDN/1-u23 is ringing" and the voip phone has no tone) any suggestions ?`
`08:41.04`	`littleball`	`X-Rob_, i read, cannot`
`08:46.29`	`*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)`
`08:47.59`	`*** part/#asterisk drunkmaster (n=lucifer@81.169.228.68)`
`08:50.11`	`kernel20`	`anybody here have use eyebeam?`
`08:50.45`	`kernel20`	`if i call my conference 8000, its hangs up`
`08:50.54`	`kernel20`	`where in fact it has no problem in xlite`
`08:50.56`	`kernel20`	`any ideas/`
`08:50.57`	`kernel20`	`>?`
`08:50.58`	`kernel20`	`?`
`08:52.44`	`SheriF_WorK`	`kernel20: eyebeam is a hardphone ?`
`08:52.48`	`kernel20`	`nop`
`08:52.50`	`kernel20`	`softphone`
`08:52.52`	`kernel20`	`from xten`
`08:53.00`	`SheriF_WorK`	`ahh may be codec issue`
`08:53.02`	`kernel20`	`xlite is the free onw`
`08:53.04`	`kernel20`	`xlite is the free one`
`08:53.07`	`kernel20`	`hmm`
`08:53.10`	`SheriF_WorK`	`yes i use xlite`
`08:53.22`	`SheriF_WorK`	`check what is the diff. in the 2 configurations file`
`08:53.26`	`kernel20`	`in my sip.conf?`
`08:53.34`	`kay2`	`is there anyway from the dialplan to easyly write into a mysql base ?`
`08:53.50`	`kernel20`	`SheriF_WorK: what makes u say that it is a codec issue?`
`08:54.34`	`kernel20`	`where in fact that settings are working in xlite`
`08:54.55`	`*** part/#asterisk cfh (n=luca@host194-20.pool21757.interbusiness.it)`
`08:55.05`	`*** join/#asterisk cfh (n=luca@host194-20.pool21757.interbusiness.it)`
`08:57.25`	`SheriF_WorK`	`kernel20: ~/.Xsrc in linux`
`08:58.06`	`*** join/#asterisk voipaster (i=25x8supp@203.215.73.200)`
`08:58.09`	`operat0r`	`im registerd with sip but I call my ipkall number and get a busy signal FWD says I missed the call`
`08:59.55`	`kernel20`	`?`
`09:02.09`	`*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)`
`09:02.25`	`*** join/#asterisk AsteriskAlbania (n=info@217.24.244.130)`
`09:02.56`	`*** part/#asterisk assert_true (n=Sunil@59.176.58.247)`
`09:03.09`	`AsteriskAlbania`	`what is the lowest bandwidth consumption codec for video on asterisk ?`
`09:05.21`	`SheriF_WorK`	`kernel20: the configuration file for Xlite is ant ur home ~/.Xscrc`
`09:07.42`	`dlynes_home`	`SheriF_WorK: Just give up....give up now :))`
`09:07.52`	`littleball`	`can i set EXTEN variable?`
`09:07.53`	`operat0r`	`what number to I put in ipkall ?`
`09:08.08`	`SheriF_WorK`	`dlynes_home: why ?`
`09:08.14`	`operat0r`	`if I want to use my FWD`
`09:08.21`	`dlynes_home`	`SheriF_WorK: he's very frustrating at the best of times :)`
`09:09.10`	`SheriF_WorK`	`dlynes_home: oh.. .i'm more than him :P i don't even know how to RTFM / STFW :P`
`09:09.29`	`dlynes_home`	`SheriF_WorK: lol...you're a perfect match, then :)`
`09:10.03`	`SheriF_WorK`	`dlynes_home: hehe yes i think that too :P and why ur at home !? u don' have work :-s?`
`09:10.31`	`dlynes_home`	`i just finished up at the office`
`09:10.35`	`dlynes_home`	`It's 2:10am`
`09:10.39`	`SheriF_WorK`	`dlynes_home: i'm trying to code a small page to get CDR out of asterisk .`
`09:10.47`	`SheriF_WorK`	`dlynes_home: 12:10 PM in egypt here`
`09:11.10`	`SheriF_WorK`	`dlynes_home: so i'm trying to learn php and at the same time messing around with asterisk alittle as i'm not into asterisk that much yet..`
`09:11.15`	`dlynes_home`	`SheriF_WorK: good luck`
`09:11.34`	`dlynes_home`	`SheriF_WorK: it might be easier to use perl with HTML::Template and CGI::Application`
`09:11.58`	`SheriF_WorK`	`dlynes_home: but it's like 1 year old asterisk .. 1.0.x and the CDR get me the call time with sec like 2343 so i want to do some math :-)...`
`09:12.02`	`dlynes_home`	`SheriF_WorK: worked great for me, but i needed a pretty complicated post-paid solution`
`09:12.16`	`SheriF_WorK`	`dlynes_home: nah i'm not a coder at all i'm trying to learnn so perl is a bitch for that :-) but i'll one day soon anyway.`
`09:12.30`	`dlynes_home`	`SheriF_WorK: if you know php, perl should be pretty easy to learn`
`09:12.41`	`dlynes_home`	`SheriF_WorK: the two, syntax wise are almost the same`
`09:12.42`	`SheriF_WorK`	`oh and i want to check the call recording featuer`
`09:12.49`	`SheriF_WorK`	`dlynes_home: yes i'm still learning php :-)`
`09:12.54`	`*** join/#asterisk A-Tuin\|work (n=A-Tuin@212.41.185.81)`
`09:12.59`	`*** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net)`
`09:13.16`	`littleball`	`hello, is it possible to set PRE defined channel variable like EXTEN?`
`09:15.04`	`dlynes_home`	`you mean override the variable's value?`
`09:15.53`	`hads\|home`	`yay, I just installed trunk on my test/home server and it fixed my niggling echo problems.`
`09:16.07`	`*** join/#asterisk Adentist (n=adentist@59.163.66.107)`
`09:16.12`	`X-Gen`	`...`
`09:16.47`	`kay2`	`Someone has ever worked with asterisk Real Time ?`
`09:16.47`	`littleball`	`yes`
`09:17.13`	`littleball`	`dlynes_home`
`09:17.14`	`littleball`	`<PROTECTED>`
`09:17.41`	`SheriF_WorK`	`kay2: yes me a little`
`09:21.09`	`dlynes_home`	`hads\|home: which ec were you using before?`
`09:22.08`	`hads\|home`	`MG2 on both but I've upped echocancel to 1024 since it's now possible.`
`09:22.40`	`hads\|home`	`Using a TDM400`
`09:22.47`	`operat0r`	`so if I setup siprox I dont need FWD ?`
`09:22.49`	`dlynes_home`	`you can set the echocancel level?`
`09:23.15`	`hads\|home`	`There was a commit to trunk the other day that lets you :)`
`09:23.27`	`dlynes_home`	`ah`
`09:23.28`	`dlynes_home`	`cool`
`09:23.40`	`hads\|home`	`Yeah, that's what I thought.`
`09:24.01`	`dlynes_home`	`i wonder how that'll affect sangoma cards`
`09:24.12`	`kay2`	`SheriF_WorK: did you use it for queue ?`
`09:24.20`	`hads\|home`	`More CPU, but for a low volume server it's not too major.`
`09:24.59`	`dlynes_home`	`hads\|home: well, any place i'm going to need echo help, is going to be a place with a small install`
`09:25.14`	`kay2`	`SheriF_WorK: because for some reason, it doesn't add people to the queue :(`
`09:25.28`	`hads\|home`	`It's only the software EC that the setting is for so I wouldn't of thought that it would affect the Sangoma's`
`09:25.45`	`dlynes_home`	`hads\|home: sangomas without a hardware ec`
`09:26.18`	`hads\|home`	`Ah sorry, I haven't had the pleasure of installing an A200 yet.`
`09:26.33`	`dlynes_home`	`hads: i've got one a200 with a hardware ec, and two without`
`09:26.55`	`dlynes_home`	`hads: i haven't had a chance to take them for a spin yet though`
`09:26.56`	`hads\|home`	`Like them?`
`09:27.02`	`hads\|home`	`Ah, bummer.`
`09:27.04`	`dlynes_home`	`hads: i've been too weighed down with other things`
`09:27.18`	`dlynes_home`	`hads i just got the driver installed for them last night`
`09:27.35`	`dlynes_home`	`now i need to figure out how to write a wanpipe1.conf file`
`09:27.57`	`SheriF_WorK`	`kay2: no`
`09:28.18`	`hads\|home`	`dlynes_home: Yeah, they're on my list of things to play with`
`09:29.06`	`*** part/#asterisk parag7732 (n=root@de1-b15475.alshamil.net.ae)`
`09:29.14`	`dlynes_home`	`hads\|home: yeah...i've got one four port fxo a200 w/o ec, one four port fxo a200 w/ec, and another 2 port fxs/2 port fxo w/o ec`
`09:30.34`	`hads\|home`	`dlynes_home: Cool, maybe I'll catch up with you on here and ask you how you got on sometime.`
`09:31.58`	`hads\|home`	`Thinking about it, the Sangoma cards use the zaptel driver in the end so you would have thought that the software EC would work the same way.`
`09:34.15`	`SheriF_WorK`	`hum any one played with record command ? monitor ?`
`09:34.33`	`SheriF_WorK`	`it's saving every side of the call in sperated file .. can't it record both sides in one file?`
`09:35.00`	`cjk_`	`hi, was anyone able to compile mysql support for the asterisk trunk branch?`
`09:37.00`	`*** join/#asterisk mr_horsepower (n=igor@82.102.1.42)`
`09:37.04`	`mr_horsepower`	`morning all`
`09:37.07`	`*** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com)`
`09:39.06`	`*** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se)`
`09:41.24`	`stoffell`	`hm, i'm doing a call to a queue, and there are a few hundred entries added in mysql.. only for this specific queue that is.. ?`
`09:46.23`	`stoffell`	`okay... got it.. the queue had 1 agent, being the queue number itself (this gave a loop).. so never mind :)`
`09:46.55`	`*** part/#asterisk cfh (n=luca@host194-20.pool21757.interbusiness.it)`
`09:47.29`	`*** join/#asterisk voipaster (i=25x8supp@203.215.73.200)`
`09:49.24`	`dlynes_home`	`SheriF_WorK: look up MixMonitor`
`09:49.30`	`operat0r`	`so if I want to not use FWD and have it goto my asterrisk box. I put in my ip for SIP Proxy but I get busy signal`
`09:49.37`	`dlynes_home`	`anyways...night, peeps`
`09:50.07`	`*** join/#asterisk Modcuts (n=bob@lan.proporta.com)`
`09:52.02`	`SheriF_WorK`	`dlynes_home: http://www.voip-info.org/wiki/view/Monitor+stereo-example`
`09:52.08`	`SheriF_WorK`	`i'm trying this but i'm a alittle lost ;-)`
`09:53.59`	`SheriF_WorK`	`dlynes_home: oh i'm using asterisk 1.0.x not 1.2 :-`
`09:54.03`	`SheriF_WorK`	`can't use MixMonitor`
`09:58.44`	`*** join/#asterisk RoyK (n=roy@213.160.242.91)`
`10:18.39`	`*** join/#asterisk michael-i (n=michael-@141.41.38.58)`
`10:20.36`	`marl`	`can anyone help me get iax working on my asterisk box? (please) i have asterisk with iax2 and have added an entry to the iax.conf file, udp5036 apears to be open when scanned from another machine on the network, but if i try and connect to * from my softfone 'idefisk' i dont see anything apearing on the asterisk console, evan after setting iax2 debug, and idefisk says timeout on registration, anyone got any pointers? ive read a ton of stuff, but i know iv`
`10:20.36`	`marl`	`e missed something stupid along the way :(`
`10:21.30`	`kaldemar`	`iax2 uses port 4569 by default.`
`10:21.39`	`marl`	`asterisk SVN-trunk-r7230, on Ubuntu 5.10`
`10:22.09`	`kaldemar`	`have you defined it to use 5036 in iax.conf and are you trying to register to that same port with idefisk?`
`10:23.21`	`marl`	`:) now im getting no registration for peer :)`
`10:23.27`	`marl`	`at leaste now its connecting`
`10:23.33`	`marl`	`or trying to`
`10:24.09`	`marl`	`i think maybe i had entered the 5036 port at some point the other day trying to get it working, thanks`
`10:24.21`	`marl`	`ill now go and try and get the registration working :)`
`10:25.59`	`kaldemar`	`set the host as X.X.X.X:5036 in your idefisk if you're using that port.`
`10:28.32`	`*** join/#asterisk _4d4m_ (n=adam@62.69.102.99)`
`10:30.05`	`operat0r`	`so for ipcall can I just put my hostname to bypass FWD ?`
`10:30.44`	`operat0r`	`sombody said I dont need FWD`
`10:32.44`	`*** join/#asterisk Ecio (n=eciostar@194.105.59.42)`
`10:32.53`	`Ecio`	`hi all`
`10:34.25`	`Ecio`	`i have installed poundkey (that afaik it's based on asterisk 1.2.5) and i want to install asterisk-addons in order to have mp3 moh. is it ok if i use latest addons (1.2.2) ?`
`10:35.59`	`Ecio`	`or maybe it's better if i upgrade to 1.2.7.1`
`10:36.00`	`Ecio`	`:)`
`10:37.24`	`*** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr)`
`10:37.26`	`RoyK`	`erm`
`10:37.30`	`RoyK`	`wtf is this??`
`10:37.30`	`RoyK`	`May 30 12:37:04 NOTICE[27343]: chan_zap.c:7395 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 4`
`10:37.53`	`RoyK`	`it worked before last reboot, now it doesn't`
`10:37.55`	`InfraRed`	`looks like a notice from zap channel`
`10:38.02`	`InfraRed`	`have you heard of google ?`
`10:38.10`	`RoyK`	`I have indeed`
`10:38.12`	`zoa`	`marl, idefisk connects to 4569`
`10:38.16`	`zoa`	`noy 5036`
`10:38.19`	`zoa`	`not`
`10:38.40`	`RoyK`	`InfraRed: but google did not show me anything relevant, just 'have you checked for interrupt sharing problems' etc`
`10:39.20`	`marl`	`thanks zoa, id added the 5036 port th eother day trying to solve this, and hadnt re-set it, as hlaf the docs i was reading were saying 5036 :( now im getting someware (i think)`
`10:39.33`	`zoa`	`royk, is it non stop ?`
`10:39.44`	`zoa`	`ive also seen it a few times`
`10:39.45`	`zoa`	`already`
`10:39.48`	`*** join/#asterisk whatisthat (n=va2003ch@203.119.9.9)`
`10:39.49`	`zoa`	`most of the time with bad cables`
`10:39.53`	`whatisthat`	`Hi`
`10:39.59`	`whatisthat`	`anyone can help me`
`10:40.06`	`whatisthat`	`I use meetme for voice conference`
`10:40.11`	`RoyK`	`zoa: every second or so`
`10:40.16`	`whatisthat`	`and I meet a problem of echo`
`10:40.21`	`whatisthat`	`how to resolve it`
`10:41.02`	`zoa`	`ouch`
`10:41.03`	`zoa`	`no good`
`10:43.17`	`RoyK`	`zoa: tried with another cable, same problem. tried with another box, no problem`
`10:43.26`	`mr_horsepower`	`damm, you ppl dont have troubles with tida?`
`10:43.27`	`mr_horsepower`	`disa?`
`10:44.18`	`RoyK`	`zoa: and the box used to work before we moved it :(`
`10:45.09`	`operat0r`	`running in asterisk -vvvvv -c I should see any call comming in correct ? I get busy sig still with ipkall`
`10:45.28`	`zoa`	`the card might be broken`
`10:45.30`	`operat0r`	`I took out FWD and its been 2+ hrs safter change`
`10:45.36`	`zoa`	`call digium support`
`10:45.38`	`RoyK`	`zoa: ?????????????`
`10:45.42`	`RoyK`	`hm`
`10:45.42`	`RoyK`	`ok`
`10:45.59`	`RoyK`	`yeah. fine. they'll send me a new one within three weeks`
`10:47.12`	`operat0r`	`I tried IAX SIP and direct to astrisk and still get busy`
`10:47.28`	`zoa`	`let me know how it goes roy`
`10:49.13`	`RoyK`	`zoa: will do`
`10:52.49`	`*** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag)`
`10:53.02`	`kmilitzer`	`Anyone every played with res_jabber and can help me a bit?`
`10:53.14`	`mr_horsepower`	`res_jabber?`
`10:53.16`	`kmilitzer`	`s/every/ever/`
`10:53.39`	`mr_horsepower`	`kmilitzer: what does it do?`
`10:53.51`	`kmilitzer`	`mr_horsepower: yes ... res_jabber in trunk ...`
`10:54.31`	`mr_horsepower`	`i see, but what does it provides?`
`10:55.08`	`kmilitzer`	`mr_horsepower: Notification ... like send a message with the incoming callerid to someone ...`
`10:55.53`	`mr_horsepower`	`we do that here, but with agi`
`10:56.07`	`mr_horsepower`	`i will try that module`
`10:56.26`	`mr_horsepower`	`sendtxt whould be very nice! :D`
`10:56.32`	`kmilitzer`	`mr_horsepower: I guess that is another way ... but I am in a playfull mood today and thought I can give it a test ... ;)`
`10:57.58`	`mr_horsepower`	`yes, test it.`
`10:58.14`	`kmilitzer`	`mr_horsepower: Thing is: I don't get it to work ... ;)`
`10:58.23`	`mr_horsepower`	`what's the problem?`
`10:58.57`	`*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6)`
`10:59.15`	`kmilitzer`	`I cannot get it to register ... I always get a 401 from the jabber server, but that's it ... jabberd does not tell me why the registaration failed`
`10:59.34`	`Dr-Linux`	`Hi`
`11:04.38`	`*** join/#asterisk faber3 (n=martin@berlin.programmfabrik.de)`
`11:04.43`	`faber3`	`hi everbody`
`11:05.04`	`faber3`	`is there somebody here from Germany? we are looking for professional asterisk consulting`
`11:05.39`	`kmilitzer`	`faber3: I am from germany ...`
`11:06.00`	`*** join/#asterisk andrebarbosa (n=andrebar@83.240.148.214)`
`11:06.15`	`*** join/#asterisk zotz (n=zotz@24.244.133.115)`
`11:08.02`	`qdk_`	`faber3: The Asterisk book is good. ;-)`
`11:10.21`	`SheriF_WorK`	`mawahahhaha i managed to get the stero call recording :-)`
`11:11.32`	`Ecio`	`i have a little problem with music on hold.. i've downloaded asterisk-addons, compiled format_mp3 and loaded it into asterisk (asterisk says "Loaded /usr/lib/asterisk/modules/format_mp3.so" and registered file format mp3)`
`11:11.50`	`Ecio`	`now im tryin to edit musiconhold.conf but it doesnt play`
`11:12.13`	`Ecio`	`i can see in the debug "started music on hold" and immediately "stopped music on hold"`
`11:17.47`	`marl`	`can someone tell me what extra config i need to enter into extesnions.conf to allow my iax to be used within the dialplan? i have an iax number (incoming only) and have set the context to 'incoming' (same as my normal land lines) and * is saying the context iax-fone-no@incoming does not exist, but i thought that by setting its context to incoming it would simply act the same as my land lines`
`11:18.38`	`*** join/#asterisk skeffling (n=chatzill@andrew.1ec.aaisp.net.uk)`
`11:22.22`	`hwt`	`http://www.voip-info.org/wiki/index.php?page=Asterisk+Voicemail+ODBC+storage`
`11:22.30`	`hwt`	`this does not appear to work in 1.2.7.1`
`11:22.45`	`hwt`	`grep ODBC Makefile`
`11:22.48`	`hwt`	`gives nothing.`
`11:31.09`	`*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)`
`11:32.38`	`puzzled`	`hi`
`11:33.41`	`russellb`	`hwt: grep ODBC apps/Makefile`
`11:34.24`	`*** join/#asterisk subdolus (n=subby@subby.afraid.org)`
`11:35.22`	`hwt`	`russellb: yeah, i noticed. thanks anyway. :)`
`11:36.33`	`marl`	`please anypointers on getting request 'phonenumber@incoming' does not exist, when trying to connect from an iax number to *`
`11:39.16`	`zoa`	`the number you are trying to dial`
`11:40.03`	`zoa`	`does not exist there`
`11:40.21`	`X-Rob_`	`make a exten => phonenumber,1,Something-to-happen in [incoming]`
`11:40.24`	`marl`	`but how do i add it? cus when i dial through the landline, i just specify the context inthe zap conf files, i dont have to specify the phonenumber`
`11:44.19`	`mr_horsepower`	`odbc sucks, i really dont like it.`
`11:44.35`	`marl`	`aaaaaaaagggggggggggggggggggggghhhhhhhhhhhhhhhhh, one of these days im going to chuck this box out of the window`
`11:45.11`	`marl`	`has anyone ever ported * to the BBC micro or the ZX Spectrum?`
`11:45.25`	`marl`	`lol`
`11:47.32`	`faber3`	`kmilitzer: sorry i was on the phone. hast du erfahrung mit asterisk als gateway f�r ISDN auf linux?`
`11:48.40`	`RoyK`	`faber3: det pleier å funke fint, det`
`11:48.41`	`qdk_`	`faber3: who made you king of the world?`
`11:49.38`	`hwt`	`qdk_: hitler.`
`11:50.00`	`qdk_`	`hehe... must be something like that. ;-)`
`11:50.30`	`*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)`
`11:51.08`	`kmilitzer`	`faber3: If you want to talk german, use private message, in here the language is german ...`
`11:51.14`	`kmilitzer`	`s/german/english/`
`11:51.16`	`*** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr)`
`11:51.18`	`jhiver`	`hi all`
`11:51.20`	`kmilitzer`	`I need a break I think ;)`
`11:51.28`	`jhiver`	`I have a pretty strange "no audio" issue`
`11:51.35`	`jhiver`	`Asterisk gives me this:`
`11:51.40`	`RoyK`	`faber3: i dunno if herr junghanns is doing anymore business with asterisk, but he used to be good`
`11:51.45`	`jhiver`	`sked to transmit frame type 4, while native formats is 256 (read/write = 256/256)`
`11:51.45`	`RoyK`	`faber3: junghannns.net`
`11:51.45`	`jhiver`	`e type 256, while native formats is 4 (read/write = 256/256)`
`11:51.45`	`jhiver`	`e formats is 256 (read/write = 256/256)cr*CLI>`
`11:51.45`	`jhiver`	`d/write = 256/256)cr*CLI>`
`11:51.45`	`jhiver`	`ed to transmit frame type 256, while native formats is 4 (read/write = 256/256)`
`11:51.46`	`jhiver`	`type 4, while native formats is 256 (read/write = 256/256)cr*CLI>`
`11:51.49`	`RoyK`	`faber3: junghanns.net`
`11:51.55`	`RoyK`	`~pb?`
`11:51.58`	`jbot`	`[pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`11:51.59`	`jhiver`	`have you seen this garbage before?`
`11:52.28`	`jhiver`	`sorry i've pasted too much`
`11:52.28`	`kmilitzer`	`Arghh ... I grab something to eat now ... I cannot think straight any more ;)`
`11:53.07`	`marl`	`exten => _440845xxxxxx,1,Answer`
`11:53.07`	`marl`	`exten => _44845xxxxxx,1,goto(incoming,s,1) in extensions.conf, and still iax dont work :( anyone tell me what stupid mistake ive made?`
`11:53.09`	`jhiver`	`it does it with asterisk 1.2.7.1 but used to work with 1.0.9`
`11:53.32`	`RoyK`	`zoa: ping`
`11:54.02`	`*** join/#asterisk coppice (n=chatzill@160.201.17.210.dyn.pacific.net.hk)`
`11:55.48`	`*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)`
`11:56.04`	`*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)`
`11:56.31`	`zoa`	`pong`
`11:56.33`	`zoa`	`on the phone`
`11:56.59`	`russellb`	`RoyK: we're working on it :-p`
`11:57.26`	`RoyK`	`russellb: i'm working on a far worse problem. one of my boxes fail to find its PRIs after moving it`
`11:57.40`	`RoyK`	`tried with two different te410p cards now`
`11:57.44`	`jhiver`	`nevermind it was a codec issue`
`11:59.03`	`RoyK`	`FUCK THIS SHIT`
`11:59.26`	`zoa`	`russel, quick question`
`11:59.39`	`zoa`	`a tdm card, does it speak ulaw / alaw internally ?`
`11:59.41`	`zoa`	`or slin ?`
`11:59.58`	`zoa`	`royk, is the carrier still the same ?`
`12:00.02`	`zoa`	`is this at your place or in a colo ?`
`12:00.12`	`RoyK`	`colo, am at the colo`
`12:00.22`	`zoa`	`did the pri change somehow ?`
`12:00.34`	`RoyK`	`not at all`
`12:00.42`	`zoa`	`what exactly did you move ?`
`12:00.46`	`zoa`	`the rack ?`
`12:00.48`	`zoa`	`or more ?`
`12:00.49`	`RoyK`	`the box`
`12:00.52`	`RoyK`	`from one rack to another`
`12:00.53`	`faber3`	`RoyK: junghanns.net does not provide service`
`12:00.54`	`russellb`	`zoa: i'm not sure, sorry ...`
`12:00.58`	`RoyK`	`faber3: ok`
`12:01.07`	`zoa`	`russellb: is matt there ?`
`12:01.20`	`zoa`	`RoyK: is it grounded ?`
`12:01.23`	`zoa`	`hmm`
`12:01.24`	`russellb`	`zoa: I don't work in town`
`12:01.28`	`RoyK`	`is the pope catholic?`
`12:01.28`	`zoa`	`how is that said in english`
`12:01.29`	`zoa`	`aha k`
`12:01.44`	`zoa`	`k`
`12:01.54`	`RoyK`	`everything is the same`
`12:01.57`	`RoyK`	`one box works`
`12:01.58`	`RoyK`	`one doesn't`
`12:02.04`	`RoyK`	`same make of the two boxes`
`12:02.07`	`RoyK`	`ibm 306`
`12:02.12`	`russellb`	`zoa: is slav on IRC?`
`12:02.16`	`RoyK`	`:%s/digium/sangoma/gi`
`12:02.58`	`russellb`	`RoyK: if you wait an hour, you can get digium support on the phone`
`12:03.05`	`zoa`	`i will ask him to come online`
`12:03.08`	`*** join/#asterisk normast (n=Norm@CPE0014bf80aeff-CM0012c90d3496.cpe.net.cable.rogers.com)`
`12:03.14`	`zoa`	`oh yes the other box worked`
`12:03.26`	`zoa`	`he is out for a cigarette now`
`12:03.48`	`russellb`	`ok, no problem`
`12:03.55`	`zoa`	`will be back in 5 minutes`
`12:04.38`	`*** join/#asterisk _Paulo_ (n=Paulo@c9064c64.virtua.com.br)`
`12:04.59`	`faber3`	`is there BRI stuff from digium ?`
`12:05.09`	`russellb`	`faber3: coming soon`
`12:06.06`	`faber3`	`russellb: how soon? our asterisk linux isdn gateway crashed for the third time now and we have to replace it or get some decent bri cards`
`12:06.30`	`*** join/#asterisk rleyba (n=root@60-241-132-21.tpgi.com.au)`
`12:06.31`	`russellb`	`i think it's pretty much ready, i'm not sure when it starts shipping ...`
`12:07.08`	`RoyK`	`russellb: Last tiime I tried asking them about this sort of problem, I got the reply I just had to return the cards to get them upgraded`
`12:07.24`	`RoyK`	`considering I need this server up within hours, that's not really an option`
`12:07.47`	`zoa`	`well if the card works in the different server i dont think they will propose that`
`12:07.47`	`RoyK`	`also, they told me the IBM hardware was crap and that I needed to test lots of other hardware instead`
`12:07.52`	`russellb`	`i understand, but they may be able to solve your problem if you give them a chance`
`12:07.58`	`rleyba`	`excuse me.....may I ask a newbie question about asterisk and vonage?`
`12:08.31`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`12:08.34`	`RoyK`	`zoa: I don't know if they do yet, only I've seen more or less the same problem before and that it was solved after trying dozens of different stuff`
`12:08.39`	`psk`	`ERASE`
`12:08.47`	`RoyK`	`DELETE`
`12:09.14`	`psk`	`arg! wrong window!`
`12:10.24`	`zoa`	`ive also seen the problem before`
`12:10.33`	`zoa`	`couldnt solve it in some occasions`
`12:10.48`	`coppice`	`zoa: the TDM card ought to work linear inside, but I seem to remember it uses ulaw. could be wrong. its a long time since I played with it`
`12:11.04`	`kay2`	`russelb: do you know why I get an "invalid" there : Members: >`
`12:11.04`	`kay2`	`<PROTECTED>`
`12:11.17`	`zoa`	`k`
`12:12.14`	`kay2`	`russellb: I added him with addqueuemember(queuename\|SIP/805@192.168.4.16:5060)`
`12:12.17`	`zoa`	`slav seems to have found something`
`12:12.18`	`coppice`	`zoa: what is your interest?`
`12:12.20`	`zoa`	`he is coming`
`12:12.23`	`*** join/#asterisk Skymarshal (n=Skymarsc@p54AF4C88.dip0.t-ipconnect.de)`
`12:12.30`	`zoa`	`slav asks, i now gave him a tdm card instead of a pri card`
`12:12.34`	`zoa`	`for the jb testing`
`12:12.44`	`zoa`	`and he things the path used in asterisk is different`
`12:12.49`	`*** join/#asterisk sturmflut (n=sraffein@mail.app.leitwerk.net)`
`12:12.53`	`sturmflut`	`hi`
`12:12.59`	`zoa`	`and thinks the codec might be different`
`12:13.02`	`zoa`	`he's checking it no`
`12:13.03`	`zoa`	`w`
`12:13.24`	`zoa`	`to be honest i dont get why it would make a difference :)`
`12:13.28`	`sturmflut`	`I updated my Asterisk to 1.2.7.1 now and it still crashes as soon as the Cisco IP Phone 7941G tries to register via Skinny`
`12:13.36`	`*** join/#asterisk slav_jb (n=k@pirus.securax.be)`
`12:13.43`	`zoa`	`there you go russellb`
`12:15.13`	`Skymarshal`	`Hi, I use Dial(SIP/10&SIP/20) to "split" a call to two phones. Problem: If I deny to answer e.g. the 10 by given it a busy it will ring again after some seconds. Why?`
`12:18.39`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`12:18.51`	`*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)`
`12:20.16`	`*** join/#asterisk FaithX (n=FaithX@mail.familyfirst.org.au)`
`12:21.25`	`kay2`	`MikeJ[Laptop]: there /`
`12:21.26`	`kay2`	`?`
`12:22.03`	`kay2`	`in a queue member, what is the penality for ?`
`12:25.03`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`12:26.39`	`*** join/#asterisk \|MxB\|aRKo (n=asd@213-140-17-110.ip.fastwebnet.it)`
`12:27.01`	`*** join/#asterisk chipdolingana (n=raymondm@bureaumw.demon.nl)`
`12:27.09`	`\|MxB\|aRKo`	`hi all :)`
`12:27.17`	`chipdolingana`	`hi everyone..`
`12:27.27`	`\|MxB\|aRKo`	`can i ask someone a question about digium card?`
`12:27.50`	`_Paulo_`	`\|MxB\|aRKo, just ask your question`
`12:28.01`	`chipdolingana`	`i don't know if i can help bet do ask your question... (i will olso do)`
`12:28.10`	`qdk_`	`\|MxB\|aRKo: ask 2 ask?!? you drunk?`
`12:28.18`	`\|MxB\|aRKo`	`i have a problem about impedence`
`12:28.20`	`[TK]D-Fender`	`kay2 : if 2 agents could be chosen equally, the one with the higher rating (lowest penelty) will be chosen. Its for skills based routing`
`12:28.36`	`\|MxB\|aRKo`	`i don't know what impedence in Ohm is supported`
`12:28.54`	`_Paulo_`	`\|MxB\|aRKo, what card model is yours?`
`12:29.03`	`\|MxB\|aRKo`	`TE110p`
`12:29.24`	`[TK]D-Fender`	`\|MxB\|aRKo : I don't know the #'s for it but thats what "loadzone" is in zaptel.conf if I'm not mistaken.`
`12:29.24`	`\|MxB\|aRKo`	`for PRI connection`
`12:29.31`	`coppice`	`aren't they always the same impedance?`
`12:29.43`	`_Paulo_`	`\|MxB\|aRKo, this isnt an E1/T1 card?`
`12:29.45`	`\|MxB\|aRKo`	`telecom ask me for calibrate the line`
`12:29.51`	`[TK]D-Fender`	`\|MxB\|aRKo : Digital links shouldn't have any kind of issue like that I would think....`
`12:29.54`	`\|MxB\|aRKo`	`yes T1/E1 card`
`12:30.12`	`\|MxB\|aRKo`	`on a PRI ISDN line`
`12:30.16`	`_Paulo_`	`\|MxB\|aRKo, better call another PSTN provider...`
`12:30.20`	`coppice`	`its 110 ohms. they always are if they use twisted pairs`
`12:30.41`	`chipdolingana`	`can someone tell me how to use the MSN numbers on my ISDN30 (te110p from digium) for outgoing calls in the netherlands. (at this moment i only show the main number to the person i call)`
`12:30.50`	`_Paulo_`	`\|MxB\|aRKo, this one seems to know nothing about what they busines`
`12:30.57`	`\|MxB\|aRKo`	`lol`
`12:31.00`	`\|MxB\|aRKo`	`sure :)`
`12:31.17`	`\|MxB\|aRKo`	`but here is the only provider :/`
`12:31.26`	`\|MxB\|aRKo`	`110 Ohm so`
`12:31.37`	`kay2`	`\|MxB\|aRKo: it's not important`
`12:31.47`	`kay2`	`\|MxB\|aRKo: plus depend of the lenght of the line`
`12:31.54`	`kay2`	`\|MxB\|aRKo: so they should know that better`
`12:32.03`	`kay2`	`\|MxB\|aRKo: they're just talkin shit :)`
`12:32.09`	`\|MxB\|aRKo`	`asd`
`12:32.17`	`\|MxB\|aRKo`	`but the card support all ?`
`12:32.28`	`_Paulo_`	`\|MxB\|aRKo, just plug it`
`12:32.32`	`kay2`	`[TK]D-Fender Do you know why I get "invalid" here : SIP/805@192.168.4.16:5070 (dynamic) (Invalid) has taken no calls yet`
`12:32.47`	`kay2`	`[TK]D-Fender: it's when I do a "show queue queuename"`
`12:32.49`	`RoyK`	`problem solved. s/digium/sangoma/gi solved it all`
`12:32.56`	`\|MxB\|aRKo`	`ok thanks for the answer :)`
`12:32.59`	`_Paulo_`	`\|MxB\|aRKo, I dont think you can damage the board, even if it somewhat fail`
`12:32.59`	`\|MxB\|aRKo`	`thanks all`
`12:33.07`	`\|MxB\|aRKo`	`yes ok`
`12:33.41`	`*** join/#asterisk LoRez (i=lorez@freenode/staff/lorez)`
`12:34.14`	`chipdolingana`	`incomming calls do work on the MSN numbers, outgoing calls always show the main number (over if i set the caller id to another number)`
`12:35.52`	`*** part/#asterisk satlan32 (n=pargit@212.150.142.211)`
`12:36.52`	`\|MxB\|aRKo`	`thank again`
`12:37.05`	`\|MxB\|aRKo`	`i try`
`12:37.18`	`*** part/#asterisk \|MxB\|aRKo (n=asd@213-140-17-110.ip.fastwebnet.it)`
`12:38.53`	`kay2`	`Is something wrong about that: exten=> 1900,2,AddQueueMember(mytest\|SIP/805@192.168.4.16:5070\|0)`
`12:41.19`	`[TK]D-Fender`	`kay2 : Why are you including an IP in your member detials?`
`12:41.35`	`[TK]D-Fender`	`kay2 : It inappropriate.`
`12:41.58`	`[TK]D-Fender`	`kay2 : Can you pastebin your entire extensions.conf for a sec.... I want to see how you're running things...`
`12:42.06`	`Ecio`	`which is the right config for musiconhold.conf with format_mp3?`
`12:42.33`	`[TK]D-Fender`	`Ecio : ... huh?`
`12:42.36`	`*** part/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)`
`12:42.47`	`Ecio`	`d-fender: i've compiled and loaded format_mp3 in asterisk`
`12:43.05`	`[TK]D-Fender`	`Ecio : Ok... so whats not working with it now?`
`12:43.35`	`Ecio`	`now im tryin to edit moh.conf in order to make it work... but when i call conference or an extension mapped to moh i see "started moh" and then immediately after "stopped moh" and no music is heard on the phone`
`12:43.47`	`*** join/#asterisk Ariel_ (n=Ariel@70.46.87.158)`
`12:43.54`	`Ecio`	`i suppose format_mp3 is correctly loaded`
`12:44.03`	`Ecio`	`cause asterisk said "loaded" and mp3 registered`
`12:44.48`	`Ecio`	`i've tried "mode=files" and "mode=quietmp3" (the default option)`
`12:45.11`	`[TK]D-Fender`	`Ecio : Are you using Native MoH or MPG123? Do you have MP3's in the folder specified? Didi you make sure they aren't VBR and have no ID3 tags?`
`12:45.32`	`chipdolingana`	`can someone tell me how to use the MSN numbers on my ISDN30 (te110p from digium) for outgoing calls in the netherlands. (at this moment i only show the main number to the person i call)`
`12:45.53`	`Ecio`	`i have the default /var/lib/asterisk/mohmp3 dir with fpm-.mp3 files so i think they're ok for `
`12:46.31`	`kay2`	`[TK]D-Fender: my SIP phones are registered on a Sip Express Router, so basically, how could asterisk do the AddQueueMember ?`
`12:46.32`	`Ecio`	`should i specity a "application=" parameter?`
`12:46.41`	`*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)`
`12:47.02`	`kay2`	`[TK]D-Fender: Otherwise I would have to add all the phones in sip.conf`
`12:47.02`	`[TK]D-Fender`	`~pb`
`12:47.08`	`jbot`	`i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`12:47.42`	`kay2`	`[TK]D-Fender: what do you want me to pastbin ?`
`12:47.58`	`kay2`	`[TK]D-Fender: there is just one AddqueueMember(mytest)`
`12:48.11`	`[TK]D-Fender`	`kay2 : Make a peer entry for them or use a Local/member method fo using your dial-plan to call them`
`12:48.12`	`kay2`	`[TK]D-Fender: but since the call comes from the SER, it doesn't get it properly`
`12:48.25`	`[TK]D-Fender`	`kay2 : the PB link was for Ecio`
`12:48.36`	`Ecio`	`k`
`12:49.08`	`kay2`	`[TK]D-Fender: what you mean by "make a peer entry for them" ?`
`12:49.17`	`*** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk)`
`12:49.39`	`[TK]D-Fender`	`kay2 : make dial entries in a context and add members like "member=Local/805@SER" and in [SER] do "exten => 805,1,Dial(SIP/805@192.168.4.16:5070)"`
`12:49.53`	`buzzyd`	`hi all, does anyone here use a audiocodes mp102 with asterisk?`
`12:49.56`	`Ecio`	`d-fender:`
`12:50.07`	`Ecio`	`oops... that's the link: http://pastebin.ca/59904`
`12:50.30`	`kay2`	`[TK]D-Fender: So I have to do a member=blabla for each phone`
`12:51.08`	`[TK]D-Fender`	`kay2 : just use the local channel for whatever "add" feature you're using to list/add your agents`
`12:51.42`	`*** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt)`
`12:52.01`	`buzzyd`	`or does anyone know where I can get some help setting up an Audiocodes MP102 with asterisk?`
`12:52.50`	`kay2`	`[TK]D-Fender: let say I have a SIP phone that register on the ser and it's id is "foo", Without doing a member=local/foo@ser, can I add "foo" to a queue ?`
`12:53.09`	`[TK]D-Fender`	`kay2 : Yes. I just told you what you need to do...`
`12:53.22`	`kay2`	`[TK]D-Fender: well I didnt get it very well`
`12:53.38`	`[TK]D-Fender`	`kay2 : exten=> 1900,2,AddQueueMember(mytest\|Local/805@SER\|0)`
`12:53.52`	`[TK]D-Fender`	`kay2 : And make that context the way I described.`
`12:54.02`	`[TK]D-Fender`	`kay2 : You can probably just use a pattern match as well.`
`12:54.06`	`kay2`	`[TK]D-Fender: but that means I know that there is a phone with "805" ?`
`12:54.37`	`[TK]D-Fender`	`kay2 : You are the one that gave me that example. pastbin your dialplan.....`
`12:55.43`	`*** join/#asterisk niter3 (n=klutch@d57-102-239.home.cgocable.net)`
`12:55.57`	`[TK]D-Fender`	`Ecio : You have MP3's in that folder? Checked for the VBR & ID3 like I mentioned?`
`12:56.29`	`Ecio`	`d-fender: the files are the predefined that comes with asterisk (pound key installation) so fpm-sunshine.mp3 etc...`
`12:56.37`	`Ecio`	`i suppose they are ok`
`12:56.50`	`Ecio`	`i've not tried (yet) to use my own mp3s`
`12:57.32`	`kay2`	`[TK]D-Fender: http://pastebin.com/746649`
`12:57.38`	`niter3`	`Hey guys, I'm looking for an IAX provider that allows you to set your own CPN.`
`12:57.48`	`niter3`	`And no, it's not for malicious use.`
`12:57.56`	`kay2`	`niter3: CPN ?`
`12:57.57`	`qdk_`	`CPN?`
`12:58.05`	`kay2`	`sda`
`12:58.06`	`kay2`	`?`
`12:58.23`	`kay2`	`[TK]D-Fender: see, my dialplan is quiet empty, that's for tests`
`12:58.39`	`qdk_`	`CallPartyNumber?`
`12:58.40`	`Ecio`	`uhm d-fender, im tryin to re-execute asterisk... and i see an error... [app_rxfax.so]Ouch ... error while writing audio data: : Broken pipe Warning, flexibel rate not heavily tested!`
`12:58.41`	`[TK]D-Fender`	`Ecio : Then I only have one susp[icion left... are you running * as non-root?`
`12:58.46`	`niter3`	`caller id`
`12:58.53`	`Ecio`	`maybe something's gone wrong installing the addons...`
`12:59.04`	`Ecio`	`d-fender: no im running it as root`
`12:59.19`	`qdk_`	`niter3: where did you come up with CPN?`
`12:59.30`	`niter3`	`sites`
`12:59.31`	`[TK]D-Fender`	`kay2 : Where's that line you pasted for me earlier with the IP embedded?`
`12:59.48`	`kay2`	`I don't have it, but that was just for a test`
`12:59.55`	`[TK]D-Fender`	`Ecio : Ok if its not a permissions thing and you're accurate in the rest of your claims I don't know what to tell you....`
`13:00.06`	`qdk_`	`niter3: CID doesnt work for you?`
`13:00.27`	`kay2`	`[TK]D-Fender: the thing is that when somebody dials a number, if I queue it, I just get "SIP/SER_DOMAIN" added in the queue`
`13:00.40`	`kay2`	`[TK]D-Fender: unless I have in sip.conf a context with the specific phone`
`13:00.48`	`[TK]D-Fender`	`kay2 : Well now you have NOTHING in there..... I can't fix NOTHING.`
`13:01.03`	`Ecio`	`d-fender: im tryin to reboot the machine, that app_rxfax error is suspect... i hadnt it before... i cant relaunch asterisk...im investigating it...`
`13:01.06`	`kay2`	`[TK]D-Fender: but I wasn't talkin about fixing something :)`
`13:01.12`	`[TK]D-Fender`	`kay2 : For dialing out to SER you don't NEED to make SIP.CONF entries for them.`
`13:01.35`	`kay2`	`[TK]D-Fender: If I do a dial(SIP/805@SER) it works find`
`13:01.38`	`*** join/#asterisk ToTo (n=ToTo@81.174.33.2)`
`13:01.42`	`SheriF_WorK`	`[TK]D-Fender: hiii ;-) today i managed to get a phone recording :P`
`13:02.26`	`kay2`	`[TK]D-Fender: but if I do a AddQueueMember(myqueue), then asterisk doesn't get the info from the contact header but just get "SIP/SER_DOMAIN"`
`13:02.34`	`Ecio`	`d-fender: lol it was another asterisk machine (another test) the one with that error... i suppose the beer and the fried pizza i've eaten are having some effect :)`
`13:02.50`	`kay2`	`unless I have the info about the specific phone in sip.conf, and I didn't wanted to have it in sip.conf`
`13:03.04`	`[TK]D-Fender`	`SheriF_WorK : congrats... not that hard was it?`
`13:03.23`	`Ecio`	`d-fender: i've reloaded asterisk (on the right machine) and now it seems to work.... /me dumb`
`13:03.24`	`SheriF_WorK`	`[TK]D-Fender: no it wasn't ;-)`
`13:03.47`	`SheriF_WorK`	`i coudn't use MixMonitor since i'm still in this crazy CVS 1.0.x asterisk :( can't upgrade .. don't have what it takes :D`
`13:04.25`	`SheriF_WorK`	`[TK]D-Fender: now my CDR show me Duration time but in seconds like 180 for 3 mints i want to convert it to something like 03:00`
`13:04.47`	`[TK]D-Fender`	`kay2 You have no interface on any of those empty AddQueueMemeber lines.... READ THE INSTRUCTIONS ON HOW TO USE IT.`
`13:05.22`	`[TK]D-Fender`	`SheriF_WorK : Don't have what it takes? You meana few minutes?`
`13:06.58`	`SheriF_WorK`	`[TK]D-Fender: nop the ballls cuz i'm sure it's will break the running system .. there is will down time donn how much :-s cuz i'm already want to upgrade the system too which will break too :D`
`13:07.18`	`[TK]D-Fender`	`SheriF_WorK : How big a setup are you running? Doing anything special on it?`
`13:09.57`	`*** join/#asterisk gandhijee (n=gandhije@host-66-202-34-162.spr.choiceone.net)`
`13:11.37`	`*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)`
`13:12.02`	`kay2`	`[TK]D-Fender: interface is optionnal!`
`13:12.37`	`kay2`	`[TK]D-Fender: if you do a AddQueueMember(queuename), it doesn't get the interface itself ?`
`13:13.07`	`SheriF_WorK`	`[TK]D-Fender: no but it's the company PBX .. so i have to do it over the night or in weekend :(`
`13:13.49`	`tamp4x`	`is there anything out there that allows recording of conversations with asterisk`
`13:13.58`	`SheriF_WorK`	`[TK]D-Fender: the other end hears me in very law sound .. is that a common problem ? or can u think about why it's might be like this ?`
`13:14.16`	`_Paulo_`	`tamp4x, sure.`
`13:14.29`	`tamp4x`	`how paulo`
`13:14.50`	`tamp4x`	`?`
`13:15.26`	`_Paulo_`	`mixmonitor`
`13:16.50`	`_Paulo_`	`http://www.voip-info.org/wiki/view/MixMonitor`
`13:17.49`	`*** join/#asterisk pa (n=Paolo@unaffiliated/pa)`
`13:19.12`	`*** join/#asterisk aze (n=aze@ACayenne-101-1-12-98.w81-248.abo.wanadoo.fr)`
`13:19.17`	`_Paulo_`	`tamp4x, that is what you were looking for?`
`13:19.20`	`*** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com)`
`13:22.02`	`*** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net)`
`13:25.12`	`[TK]D-Fender`	`kay2 : Use the callerID as the backtrace to find out where to go.`
`13:25.20`	`*** join/#asterisk Hmmhesays (i=negative@66.173.103.110)`
`13:25.43`	`[TK]D-Fender`	`SheriF_WorK : Depends... whats on both ends?`
`13:25.50`	`Hmmhesays`	`so I got a new phone`
`13:26.08`	`[TK]D-Fender`	`Hmmhesays : Which?`
`13:26.12`	`Hmmhesays`	`motorola E815`
`13:26.34`	`[TK]D-Fender`	`Hmmhesaysm :Great phone....`
`13:26.51`	`Hmmhesays`	`looks pretty crippled out of the box`
`13:26.59`	`[TK]D-Fender`	`Hmmhesays : I got a 1 gig card for mine and am about to load it up with MP3's`
`13:27.07`	`[TK]D-Fender`	`Hmmhesays : Crippled? How?`
`13:27.50`	`Hmmhesays`	`looks like some of the bluetooth stuff could be better`
`13:28.07`	`Hmmhesays`	`but from what I read there are some upgraded firmwares out there`
`13:29.19`	`*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)`
`13:29.53`	`[TK]D-Fender`	`Hmmhesays : BT works fine for me for file transfer, modem, and headset (tested once)`
`13:30.21`	`kay2`	`[TK]D-Fender: in sip.conf, I added [SER] as a peer, but still, when I do a AddQueueMember(mytest), it adds "SIP/ser_domain_name" and doesn't get the "805" from the contact or from header`
`13:30.47`	`*** join/#asterisk aze_ (n=aze@ACayenne-101-1-12-84.w81-248.abo.wanadoo.fr)`
`13:30.56`	`[TK]D-Fender`	`kay2 : I didn't say add it to SIP.CONF. its for EXTENSIONS.CONF for the local channel.`
`13:30.59`	`Hmmhesays`	`[TK]D-Fender can you do voice dialing on it without a headset?`
`13:31.03`	`Hmmhesays`	`i can't seem to find that anywhere`
`13:31.10`	`[TK]D-Fender`	`Hmmhesays : yup`
`13:31.21`	`[TK]D-Fender`	`Hmmhesays : button on right side`
`13:31.31`	`[TK]D-Fender`	`Hmmhesays : the lower one`
`13:31.47`	`Hmmhesays`	`speakerphone button`
`13:31.55`	`Hmmhesays`	`oh nm`
`13:31.58`	`Hmmhesays`	`too early yet`
`13:32.25`	`*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.59.Dial1.SanJose1.Level3.net)`
`13:32.37`	`Hmmhesays`	`how do I activate the prompt asking for a name, or whatever it says`
`13:32.52`	`*** join/#asterisk Conductor (n=thomas@62.8.240.185)`
`13:33.09`	`Conductor`	`hi! is there anything like setCallerPres for incoming calls?`
`13:33.18`	`[TK]D-Fender`	`Hmmhesays : Press the lower right side button quickly and say "call so-and-so:`
`13:33.26`	`kay2`	`[TK]D-Fender: don't get pissed :)`
`13:33.31`	`Hmmhesays`	`ahah! my bad, held it down`
`13:33.45`	`[TK]D-Fender`	`Hmmhesays : only a momentary press`
`13:33.57`	`[TK]D-Fender`	`Hmmhesays : holding it is for long term recording...`
`13:34.13`	`*** part/#asterisk awad (n=naoshige@avtomat.probsd.net)`
`13:34.14`	`SheriF_WorK`	`[TK]D-Fender: hehe on both ends me on SIP phone and a analog line then analog phone.`
`13:34.19`	`[TK]D-Fender`	`Hmmhesays : I can't wait to load mine up with MP3's..... I already filled the 40 meg it had.. 1 Gig ought to do me fine for a while :)`
`13:34.50`	`kay2`	`[TK]D-Fender: but I dunno what are the username of the phone registered on the SER, how can I add them on [local] if I don't have the username ? I'll get the username just once the call for being added in the queue comes!`
`13:34.56`	`[TK]D-Fender`	`SheriF_WorK : if your phone is well balanced for inter-sip talking and VM recording then its your gain on the PST that needs to be adjusted.`
`13:35.51`	`[TK]D-Fender`	`kay2 : Allow unauthenticated callers on your system and [local] isn't meanto to be a context, its for the Local/ channel!`
`13:35.55`	`Hmmhesays`	`[TK]D-Fender its microsd right?`
`13:36.04`	`[TK]D-Fender`	`Hmmhesays : yup. Dirt cheap these days`
`13:36.49`	`kay2`	`[TK]D-Fender: i've already done a "insecure=very", and not auth calls gets in, that's not the pb`
`13:37.33`	`kay2`	`s/not/now`
`13:38.01`	`kay2`	`[TK]D-Fender: If I do a dial, the call comes from the SER to asterisk and that works fine`
`13:38.25`	`*** join/#asterisk stack_ (n=stack@63.239.190.202)`
`13:38.28`	`[TK]D-Fender`	`Hmmhesays : Typical name is TransFlash`
`13:38.32`	`stack_`	`good morning, everyone`
`13:38.37`	`Hmmhesays`	`yeah`
`13:39.01`	`kay2`	`[TK]D-Fender: the only thing is that I don't see anytihng like SIP/PHONEID@my_ser_domain but only SIP/my_ser_domain ...`
`13:39.13`	`stack_`	`Every once in a while, I get the following on the console: -- Zap/22-1 is proceeding passing it to SIP/aohler-c5cb`
`13:39.14`	`stack_`	`<PROTECTED>`
`13:39.14`	`stack_`	`<PROTECTED>`
`13:39.14`	`stack_`	`<PROTECTED>`
`13:39.14`	`stack_`	`<PROTECTED>`
`13:39.23`	`stack_`	`any ideas as to why?`
`13:39.34`	`kay2`	`~pb`
`13:39.36`	`jbot`	`pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`13:39.43`	`[TK]D-Fender`	`stack_ : They're on the phone.`
`13:40.43`	`stack_`	`[TK]D-Fender, really... it's that simple?`
`13:40.54`	`stack_`	`[TK]D-Fender, sorry, I was out in the sun too much`
`13:41.06`	`*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)`
`13:41.10`	`[TK]D-Fender`	`kay2 :You are complicating this a LOT for nothing. Does your CallerID match a username that you can dial back to SER to reach them? So if a call comes in from 805, that means that if * dialed 805@serIP it'd ring?`
`13:41.58`	`[TK]D-Fender`	`stack_ : PRI has progress codes and the telco was able to detect the busy and instead of providing tone it passed back digital progress indications instead of the annoying busy signal... its up to * to annoy you now :)`
`13:42.19`	`stack_`	`[TK]D-Fender, gotcha`
`13:42.55`	`mut`	`<PROTECTED>`
`13:43.08`	`kay2`	`[TK]D-Fender: yeah`
`13:43.42`	`kay2`	`if I do a Dial(SIP/SER/805) or Dial(SIP/805@serip) it would ring`
`13:44.07`	`kay2`	`[TK]D-Fender: the only thing is from asterisk, how do I get the "805"`
`13:44.25`	`pollo`	`hi`
`13:44.30`	`kay2`	`[TK]D-Fender: since I don't have [805] anywhere`
`13:44.45`	`pollo`	`anyone can helpme with sip softphones`
`13:44.54`	`kay2`	`pollo: shoot`
`13:45.01`	`[TK]D-Fender`	`kay2 : So use the incoming callerid and add your queuemembers using Local/${CID}@ SER and make a catch-all to dial the actual phone.`
`13:45.07`	`pollo`	`ok`
`13:45.47`	`kay2`	`[TK]D-Fender: which means I have to have a member=something`
`13:46.13`	`pollo`	`I can�t call from my extension 201 from 203 into my lan , when i call i get these error : May 30 14:02:03 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 387c67d7788e0f277f9420af641490b6@192.168.1.44 for seqno 102 (Critical Request)`
`13:46.34`	`pollo`	`these are my conf files http://pastebin.com/745115`
`13:46.43`	`FaithX`	`anyone using pennytel`
`13:47.09`	`[TK]D-Fender`	`kay2 : NO. You need to WAKE UP! Add you querumebers EXACTLY LIKE THIS : exten => 1900,1,AddQueueMember(myqueue\|Local/${CALLERID(number)}@SER)`
`13:47.26`	`Ecio`	`sorry for the (maybe) dumb question, but is "include" in context multilevel ? i mean if B includes C and A includes B, does A includes also C ?`
`13:48.05`	`*** join/#asterisk C4T3l (n=rcall01@216.54.143.2)`
`13:48.12`	`[TK]D-Fender`	`kay2 : Then make a context named [SER] in extensions.conf and add "exten => _X.,1,Dial(SIP/${EXTEN}@1.2.3.4)" and change the IP to point to your SER`
`13:48.13`	`Ecio`	`(im used to use call manager and it uses two concepts, partitions and callingsearchspace in order to manage separations)`
`13:48.23`	`[TK]D-Fender`	`Ecio : Yes, full inheritance`
`13:48.30`	`kay2`	`[TK]D-Fender: ok 2s`
`13:49.43`	`*** join/#asterisk bprice20 (n=brandon@Dynamic-216.120.224.167.hrnoc.net)`
`13:49.49`	`Ecio`	`uhm... so i must separate my users in another way...`
`13:50.09`	`[TK]D-Fender`	`Ecio : Make a better combination.`
`13:50.16`	`*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)`
`13:50.52`	`[TK]D-Fender`	`Ecio : Pastebin what you've got. I doubt you need to do too much to make it very functional and clean.`
`13:51.08`	`Ecio`	`d-fender: im just doing some planning, i've not created anything yet`
`13:51.10`	`niter3`	`Hey guys, I'm looking for an IAX provider that allows you to set your own CPN.`
`13:51.13`	`niter3`	`CID`
`13:51.14`	`niter3`	`sorry`
`13:51.49`	`bkw__`	`CPN and CID really don't differ in 99% of the cases`
`13:52.06`	`Ecio`	`my idea is having two groups of users. group A: can call group A, group B and a sip trunk, group B can call only group A and B`
`13:52.22`	`bkw__`	`you have a few rare cases where a us cellphone roaming in canada.. they differ then`
`13:52.26`	`[TK]D-Fender`	`Ecio : very easy`
`13:52.37`	`kay2`	`[TK]D-Fender: ok like that it works :)`
`13:53.09`	`*** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.34)`
`13:53.25`	`Ecio`	`d-fender: i see that i can do it having two separate contexts... but the problem is... if i have to specify a line for every number, Dial(SIP/user)`
`13:53.36`	`Ecio`	`i have to replicate all of these on both contexts...`
`13:53.42`	`Hmmhesays`	`wow there are a lot of hacks for this phone`
`13:53.52`	`Ecio`	`maybe i can do it using variables and so on...`
`13:53.59`	`[TK]D-Fender`	`Ecio : Not at all... Pastebin what you;'ve got.`
`13:54.19`	`[TK]D-Fender`	`Hmmhesays : like?`
`13:54.31`	`qdk_`	`niter3: Some IAX peer might allow it, but why? it will be striped or denied going to a regular/old carrier.`
`13:54.50`	`Hmmhesays`	`http://www.howardforums.com/showthread.php?t=674803`
`13:55.09`	`niter3`	`qdk_: I don't think so..`
`13:55.33`	`bprice20`	`Is anyone besides myself using odbc for voicemail storage?`
`13:55.57`	`qdk_`	`niter3: i could test it and make sure, but i really dont have the time... but WHY do you need that "feature"?`
`13:56.19`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`13:56.47`	`SheriF_WorK`	`[TK]D-Fender: hum i'll look for that now i should go home :-)`
`13:56.50`	`SheriF_WorK`	`later guys`
`13:56.58`	`Conductor`	`can i change the callingPres for incoming calls?`
`13:57.02`	`qdk_`	`bprice20: no, but im thinking about it, so i can get the files of one server.`
`13:57.03`	`Conductor`	`does this make sense?`
`13:57.12`	`coppice`	`i remember when realplayer for linux used to work. WTF did they do to it?`
`13:58.14`	`bprice20`	`qdk_ I have had one problem with it, and suprisingly since using mysql 5 it hasn't been performance w/ blobs`
`13:58.59`	`bprice20`	`its that if you select out the blob and pipe that to a file then its not a wav file`
`13:59.31`	`bprice20`	`its raw data, I'm like how to you get the wav out of the database for use in say a web based interface`
`13:59.38`	`Ecio`	`d-fender: http://pastebin.ca/59919 \| but consider that im just starting and doing some test.. but my final obj is having around 500 SIP users (obviously not calling concurrently :D)`
`14:00.16`	`*** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr)`
`14:00.29`	`Ecio`	`i think that maybe it will be easier to use directly numbers for SIP account... so i can call SIP/number instead of having all those exten to map numbers to users...`
`14:00.30`	`bprice20`	`asterisk obviously is doing it because they playback fine via the voicemail app`
`14:00.38`	`niter3`	`qdk_: I want this feature so that when somebody calls and asterisk passes it on to my cell phone it will transfer the CID number, so I know if I want to answer it or not.`
`14:00.54`	`Ecio`	`and doing in that way i'll have only a couple of lines in the extensions.conf... (correct me if im wrong)`
`14:01.27`	`Ecio`	`(but the idea of having users instead of numbers is fascinating :))`
`14:02.11`	`sevard`	`Is anyone experencing that wakeup.php is not working after you upgrade to 1.2.7.1`
`14:02.51`	`*** join/#asterisk ToTo (n=ToTo@81.174.33.2)`
`14:03.01`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`14:06.40`	`mr_horsepower`	`everyone knows wakeup.php? i dont!`
`14:07.16`	`[TK]D-Fender`	`Ecio : Like this : http://pastebin.ca/59922`
`14:07.59`	`[TK]D-Fender`	`Ecio : And you need to learn about macros FAST....`
`14:08.11`	`mr_horsepower`	`Ecio: what's so fascinating about having users insted of numbers?`
`14:08.32`	`Ecio`	`d-fender.. and then i'll have that A users (like john and jane) have in their sip.conf that their context is GroupA (not GroupAPhones) right?`
`14:08.45`	`[TK]D-Fender`	`Ecio : Correct.`
`14:08.56`	`Ecio`	`mr_horse: that saying "you can call me at jdoe@mydomain.com"`
`14:09.04`	`Ecio`	`is far better than calling tu 6456@mydomain.com D:`
`14:09.42`	`Ecio`	`d-fender: i've used (err. copied) one simple macro for calling the cisco call manager sip trunk...`
`14:09.54`	`[TK]D-Fender`	`Ecio : Taht way you can includ GroupAPhones in IVR contexts etc without comprimising other functionality`
`14:10.18`	`mr_horsepower`	`ivr contexts? why dont use queues?`
`14:10.23`	`[TK]D-Fender`	`Ecio : What do you need CM for now that you have *?`
`14:10.41`	`[TK]D-Fender`	`mr_horsepower : Queues don't give you a menu....`
`14:10.48`	`Ecio`	`d-fender: cause my company (actually all the building im in) has a cisco infrastructure`
`14:10.57`	`Ecio`	`so cisco phones, cisco voice gw, cisco call manager`
`14:11.03`	`Ecio`	`all that f**kin expensive stuff :D`
`14:11.07`	`[TK]D-Fender`	`Ecio : Poor you....`
`14:11.12`	`mr_horsepower`	`offcourse not, and whats users have to do with ivr's anyway?`
`14:11.31`	`Ecio`	`d-fender: actually it's quite easy to use it and administer it... but of course i havent paid it :)`
`14:11.40`	`[TK]D-Fender`	`mr_horsepower : Maybe you might want to dial an EXTENSION direclty from a menu perhaps?`
`14:12.33`	`mr_horsepower`	`[TK]D-Fender: in a ivr? on a normal ivr, you dial queues, that have the members/extensions you mention.`
`14:12.35`	`Ecio`	`the idea is giving SIP accounts to our external agents (~500) so they can call us and we can call them via sip trunk, organize audio conferences with meetme and of course give them the opportunity to call each other without payin`
`14:12.39`	`[TK]D-Fender`	`mr_horsepower : IVR isn't all about "press 1 for customer server, press 2 for Joe (even though everyone on the inside dials 100 to call him). NO, you make it so you can dial peoples extension NORMALLY.`
`14:13.14`	`mr_horsepower`	`NORMALY no one dials extensions directly in ivr's.`
`14:13.25`	`[TK]D-Fender`	`mr_horsepower : Stop thinking like the entire world is a call center where names and people don't matter. When I want to call Joe at a friggen company I call them up and puch HIS extension.`
`14:13.39`	`[TK]D-Fender`	`mr_horsepower : You are very VERY wrong on that one....`
`14:14.18`	`[TK]D-Fender`	`mr_horsepower : Do the words "If you know the extensions of the person you want to reach, please dial it now" ring a bell?`
`14:14.26`	`mr_horsepower`	`[TK]D-Fender: normaly, dont have a ivr like "punch 1 to dial john, press 2 to dial joe, press 3 to dial ..."`
`14:14.29`	`Ecio`	`d-fender: err i mean, of course my company paid for the cisco infrastructure but i had no decision role in that...`
`14:14.42`	`buzzyd`	`Anyone here know anything about audiocodes MP102/4 and asterisk?`
`14:14.54`	`mr_horsepower`	`[TK]D-Fender: OFFCOURSE but if you want to dial the ppl directly, why do you need a fucking ivr?`
`14:15.00`	`[TK]D-Fender`	`mr_horsepower You mean YOU don't. Tell that to the rest of the world.`
`14:15.47`	`[TK]D-Fender`	`mr_horsepower : IVR is jsut something that accepts DTMF to route calls instead of paying a receptionist to pick up the phone. It has nothing to do with Queues my its nature. Perhaps you should look up the definition of IVR.`
`14:15.50`	`mr_horsepower`	`yeah "punch 1 to dial john, press 2 to dial joe, press 3 to dial ... OR you can just stop wasting time in this fucking useless ivr, and dial it directly"`
`14:16.20`	`[TK]D-Fender`	`mr_horsepower : The very sample menu you just gave IS AN IVR!`
`14:16.26`	`Conductor`	`what is CALLERID(ani) CALLERID(dnis) CALLERID(rdnis)?`
`14:16.33`	`mr_horsepower`	`yes, it is, useless, but it is.`
`14:16.40`	`[TK]D-Fender`	`mr_horsepower : Any automated menu is an ivr. Period.`
`14:17.02`	`mr_horsepower`	`yes, your right, it keeps behing useless anyway.`
`14:17.13`	`*** join/#asterisk froguz (n=alvaro@200.104.155.95)`
`14:17.40`	`[TK]D-Fender`	`mr_horsepower : Well thats LANGUAGE for you. Words have meanings. Respect their intent or go on mumbling by yourself...`
`14:17.52`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`14:18.02`	`sevard`	`Is anyone experincing that wakeup.php is not working after you upgrade to 1.2.7.1`
`14:18.56`	`[TK]D-Fender`	`mr_horsepower : Being able to dial an extension is no different that pressing a shoret entry that could possibly do the same thing. Its not the structiure of the menu (*like being able to dial what is conceptually a users extension) that counts. Its the mere ability to enter something at all and have it process it that counts.`
`14:19.28`	`Conductor`	`could anyone tell me what dnis and rdnis is for?`
`14:19.53`	`buzzyd`	`There has to be some here that has experience with audiocodes kit and Asterisk`
`14:19.56`	`mr_horsepower`	`[TK]D-Fender: that's why i sayed, NORMALY, and not ALLWAYS.`
`14:21.10`	`froguz`	`it's posible to share an E1's channels between 2 PBX? i have an E1 connected to a NEC Neax 2000 IPS and i want to connect 4 of those channles to * (sharing the same signaling channel)`
`14:21.57`	`blitzrage`	`anyone have a good example set of files for Cisco 7960's with all the features? I think this set of conf files I have is lacking.`
`14:22.24`	`*** join/#asterisk flujan (n=flujan@internet.nube.com.br)`
`14:22.29`	`_Paulo_`	`froguz, you can split the E1 with a MUX`
`14:22.47`	`flujan`	`coppice, Hi steves... I'm having problems using unicall . Could you help me? :)`
`14:23.10`	`coppice`	`ok`
`14:23.10`	`_Paulo_`	`froguz, but I dont know a mux that is less expensive than a second interface card.`
`14:23.14`	`asterboy`	`yuk, I'm getting Seg Faults on sox`
`14:23.23`	`asterboy`	`sux`
`14:23.24`	`hwt`	`where do i specify the host for odbc storage?`
`14:23.48`	`flujan`	`coppice, I start debuging the legacy pbx which I'm connecting asterisk with.`
`14:23.59`	`froguz`	`a MUX? mmm... haven't heard before. i'll look for some info in the wiki`
`14:24.13`	`hwt`	`i can't find anything relevant in either res_config.conf or extconfig`
`14:24.20`	`asterboy`	`multiplexer`
`14:24.27`	`*** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com)`
`14:24.52`	`_Paulo_`	`froguz, a multiplexer. you can find some used units cheap on ebay`
`14:24.53`	`asterboy`	`used them in the old days to transport groups of terminals from one building to another.`
`14:25.00`	`hwt`	`perhaps its odbc.ini?`
`14:25.26`	`flujan`	`coppice, The legacy pbx did not send the full number... for instance, If i dialed 12345678 it just send the number 4. I asked the company which made de software and they said that asterisk should "ask" for the other number... And it is not "asking"...`
`14:25.44`	`flujan`	`coppice, how can I debug and retrieve this information?`
`14:25.48`	`brif8`	`Can one set call forwarding by extensions. ie if extension 302 can be call forward to xxx-yyy-zzzz and 304 to aaa-bbb-cccc when 302 and 304 are set to DND ?`
`14:26.03`	`_Paulo_`	`flujan, its MFC-5C`
`14:26.22`	`_Paulo_`	`flujan, Brazilian variant of MFC-R2`
`14:26.48`	`froguz`	`_Paulo_, so i can't share the signalling channel between 2 pbx?`
`14:27.19`	`mr_horsepower`	`flujan: i have almost the same thing`
`14:27.22`	`_Paulo_`	`froguz, not without some gear like a mux.`
`14:27.23`	`mr_horsepower`	`with a matra pbx`
`14:27.34`	`coppice`	`flujan: do you have a log of a call with loglevel=255?`
`14:27.42`	`mr_horsepower`	`it sends the first digit, and then sends me everything else in dtmf`
`14:27.48`	`mr_horsepower`	`dont know why`
`14:27.54`	`flujan`	`coppice, Yes... I will pastebin it.`
`14:28.37`	`flujan`	`coppice, when I dial using one of my legacy pbx extensions... I route the call to asterisk. Asterisk detect the event as you can see: http://pastebin.com/746815`
`14:28.45`	`_Paulo_`	`flujan, You are talking about "Bina"`
`14:28.49`	`mr_horsepower`	`_Paulo_: do know why i'm having this problem? any clue?`
`14:29.06`	`flujan`	`coppice, but doesn't enter in the context I configured in the unicall.conf`
`14:29.17`	`_Paulo_`	`mr_horsepower, yes, I know... :-)`
`14:29.32`	`flujan`	`_Paulo_, yes Paulo. I'm trying to put asterisk working with my legacy pbx through a E1 link.`
`14:29.47`	`mr_horsepower`	`_Paulo_: we speak almost the same language, can you say me why? :D`
`14:30.08`	`_Paulo_`	`lets talk at #asteriskbrasil.org`
`14:30.08`	`flujan`	`_Paulo_, asterisk is detecting the events correctly but isn't entering the context I configured in the unicall.conf`
`14:30.50`	`coppice`	`flujan: what you have pasted is only the start of the call`
`14:31.35`	`Conductor`	`when a caller uses CLIP no screening, i always get his _real_ number. How can i get the number he set?`
`14:31.54`	`*** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg)`
`14:32.02`	`mut`	`damn phone`
`14:32.07`	`flujan`	`coppice, Yes... asterisk doesn't enter in the context. Just this event and then, when I hang up: http://pastebin.com/746826`
`14:32.12`	`mut`	`i missed american pie on the shoutcast stream`
`14:32.29`	`mut`	`curse our stupid wireless crew!`
`14:32.50`	`Conductor`	`doesn't anyone know how i can get both numbers of the CLIP no screening caller?`
`14:32.52`	`coppice`	`flujan: there are no tones coming from the other end`
`14:32.57`	`*** join/#asterisk The_X (i=chris@true.fiberpimp.net)`
`14:33.14`	`The_X`	`I have a weird problem`
`14:33.33`	`flujan`	`flujan, they said that asterisk should ask for another tone... something like. OK, old pbx i receive the digit, send me another...`
`14:33.53`	`yxa`	`guys is this possible? SIP clients/phones <-> M$ LCS <-> Asterisk <-> Cisco Call Manager <-> SCCP Phones`
`14:34.04`	`coppice`	`we are not seeing the first tone, according to the log you pasted`
`14:34.05`	`The_X`	`When someone calls a SIP phone from the outside, they hang up but the SIP phone wont drop the call`
`14:34.06`	`*** join/#asterisk Pigi (n=pigi@pdpc/supporter/active/Pigi)`
`14:34.14`	`The_X`	`then I get bridged another phone line`
`14:34.17`	`hwt`	`what does this mean?`
`14:34.17`	`hwt`	`May 30 16:33:16 WARNING[7890]: res_odbc.c:565 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified`
`14:34.18`	`Pigi`	`'morning all`
`14:34.18`	`The_X`	`and hear some random folks talking`
`14:34.28`	`sevard`	`To: Asterisk`
`14:34.31`	`sevard`	`Subject: DIE`
`14:34.36`	`sevard`	`Body: call files don't work`
`14:35.14`	`flujan`	`coppice, Moises Silva said I need to change the libmfrc2 to have this working propertly...`
`14:35.33`	`Pigi`	`is there anyone that has successfully got asterisk working with eicon diva pci cards (passive cards ) ?`
`14:35.48`	`coppice`	`which version of spandsp and libmfcr2 are you using?`
`14:35.49`	`*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)`
`14:36.38`	`[TK]D-Fender`	`sevard : I'm sure they work just fine....`
`14:36.50`	`sevard`	`I can't get them to work`
`14:36.52`	`sevard`	`they're gay.`
`14:36.56`	`sevard`	`gay like gay sex, gay.`
`14:37.01`	`[TK]D-Fender`	`sevard : Thats probably much more accurate :)`
`14:38.17`	`trelane_`	`?`
`14:38.29`	`trelane_`	`wtf`
`14:39.47`	`Hmmhesays`	`sevard is gay`
`14:39.50`	`flujan`	`coppice, here goes my unicall.conf: http://pastebin.com/746835`
`14:39.57`	`flujan`	`coppice, could you please take a look?`
`14:40.12`	`Hmmhesays`	`actually he's just retarded`
`14:40.25`	`Hmmhesays`	`but retarded people are generally happy right?`
`14:40.46`	`Conductor`	`can i set the callingpres for incoming calls?`
`14:40.54`	`sevard`	`Hmmhesays loves the cock`
`14:40.58`	`flujan`	`coppice, I'm using the lastest version I grab from your site.. :)`
`14:41.09`	`mitcheloc`	`sevard: that's not cool`
`14:41.14`	`Hmmhesays`	`my girlfriend smacked me in the face with her balloon weiner dog`
`14:41.18`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)`
`14:41.18`	`Conductor`	`is this an option in zapata.conf maybe?`
`14:41.19`	`yxa`	`anyone tried * with MS LCS?`
`14:41.24`	`Conductor`	`or in zaptel.conf?`
`14:41.35`	`flujan`	`coppice, mfcr2 0.0.3`
`14:41.47`	`*** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com)`
`14:42.03`	`flujan`	`coppice, spandsp 0.0.3pre6`
`14:42.27`	`Conductor`	`GUYS! you can't tell me that noone knows how to set the callingPres for incoming calls!`
`14:42.33`	`*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)`
`14:42.49`	`zoa`	`its on voip info for sure`
`14:42.54`	`zoa`	`and probably on asteriskguru too`
`14:43.03`	`sevard`	`mitcheloc: not cool man, not cool.`
`14:43.08`	`Conductor`	`obviously there are two different numbers transmitted. ${CallerId} is one of them. Where is the other one?`
`14:43.17`	`*** join/#asterisk fugitivo (n=ajf@190.48.162.70)`
`14:43.22`	`[TK]D-Fender`	`sevard : You need to calm down and just read the script to find out where you went wrong (aside from just trying to take some random piece of code and expect it to work out of the box)`
`14:43.50`	`[TK]D-Fender`	`Conductor : the other is the ${EXTEN} which is the number they dialed to land on your system`
`14:43.52`	`*** join/#asterisk vechers-away (i=vechers@64.61.117.138)`
`14:43.55`	`asterboy`	`looks like sox seg faults if there are spaces in the file name`
`14:44.26`	`sevard`	`[TK]D-Fender: i'm creating my own call files by hand, in asterisk 1.2.4 if you made your file 0943.ext.0.call it would call ext 0 at 9:43 A.M., now with 1.2.7.1 whenever you drop in any file in there at any time it will call out right away, regardless.`
`14:44.58`	`coppice`	`flujan: a few people have had problems with getting the line signals but not getting any tones, and it has turned out to be some configuration issue at the other end of the line. there was a bad version of spandsp, which didn't generate tones properly. your problem is you are not seeing tones, so that wouldn't be the issue, anyway`
`14:45.02`	`[TK]D-Fender`	`sevard : The WIKI clearly states its teh DATESTAMP that controls delayed execution, not the filename... where do you keep getting that idea from?`
`14:45.20`	`sevard`	`[TK]D-Fender: from the wakeup.php script that worked fine :\|`
`14:45.22`	`[TK]D-Fender`	`sevard : and I've corrected you on it previously.`
`14:45.31`	`tamp4x`	`Unable to open '/dev/zap/channel': No such file or directory anyone know what woul dbe the cause of this`
`14:45.37`	`Conductor`	`[TK]D-Fender, the problem is, that the ${CALLERID} from some callers is not their extension but their company's main number.`
`14:45.42`	`[TK]D-Fender`	`sevard : yes, and I clealy saw where it SETS THE TIMESTAMP`
`14:45.50`	`[TK]D-Fender`	`sevard : Link it.`
`14:45.52`	`Conductor`	`[TK]D-Fender, but only with our asterisk installation`
`14:45.57`	`coppice`	`tamp4x: you don't have permission to open it, most probably`
`14:46.27`	`The_X`	`anyone using Patton smartnode + asterisk?`
`14:46.35`	`flujan`	`coppice, so it should be a problem in the legacy pbx signalling tones?`
`14:46.52`	`coppice`	`i think so`
`14:47.08`	`Conductor`	`[TK]D-Fender, so i thought maybe callerid=asreceived and pritrustusercid=yes in zapata.conf would help but it did not.`
`14:47.27`	`[TK]D-Fender`	`Conductor : ummm... What exactly are you trying to do with presentation?`
`14:48.07`	`tamp4x`	`i have the persmissions and what not set in udev`
`14:48.08`	`yxa`	`anyone knows if I can use * to transcode btw CCM and ms live communications server (SIP) ?`
`14:48.26`	`[TK]D-Fender`	`yxa : sure`
`14:48.27`	`Conductor`	`[TK]D-Fender, just display it on our phones.`
`14:48.50`	`[TK]D-Fender`	`Conductor : should have "usecallerid=yes" and "callerid=asreceived"`
`14:48.55`	`[TK]D-Fender`	`Conductor : thats it.`
`14:49.01`	`Conductor`	`[TK]D-Fender, i have that.`
`14:49.25`	`Conductor`	`[TK]D-Fender, we have the telco features CLIP no screening and COLP no screening`
`14:49.26`	`[TK]D-Fender`	`Conductor : maybe your telco is blocking incoming CID info.`
`14:49.32`	`_Paulo_`	`coppice, in early days in Brazil, the engeneers devised a callerid protocol so they could mix analog and digital`
`14:49.39`	`Conductor`	`[TK]D-Fender, could this have anything to do with it?`
`14:49.45`	`yxa`	`[TK]D-Fender for CCM part, i need to use sergio's chansccp?`
`14:50.01`	`[TK]D-Fender`	`yxa : Depends if you can talk SIP to it instead.`
`14:50.02`	`sevard`	`[TK]D-Fender: -rw-r--r-- 1 asterisk nogroup 139 2006-05-30 09:50 0950.ext.0.call`
`14:50.25`	`sevard`	`that's set to call at 9:50 A.M. according to the time stamp`
`14:50.26`	`[TK]D-Fender`	`sevard : The sure LOOKS like a file that'll execute NOW....`
`14:50.28`	`yxa`	`[TK]D-Fender no, ccm side is fully sccp`
`14:50.45`	`sevard`	`[TK]D-Fender: you mean in 4 minutes.`
`14:50.56`	`[TK]D-Fender`	`yxa : Ok, I'm none too knowledgable with SCCP`
`14:50.57`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`14:50.57`	`*** mode/#asterisk [+o anthm] by ChanServ`
`14:51.07`	`[TK]D-Fender`	`sevard : more or less.`
`14:51.13`	`Ecio`	`yxa: u can establish a SIP trunk between CCM and *`
`14:51.17`	`sevard`	`it executed when i put it in the directory`
`14:51.19`	`*** join/#asterisk matt_ (n=mr245@amos.bath.ac.uk)`
`14:51.24`	`Ecio`	`without using SCCP, cant u ?`
`14:51.25`	`[TK]D-Fender`	`My clock says 09:51 here.`
`14:51.29`	`sevard`	`REGARDLESS (notice caps) of timestamp`
`14:51.32`	`yxa`	`Ecio can CCM do that?`
`14:51.38`	`sevard`	`I'll make it for freaking 11pm`
`14:51.42`	`Ecio`	`yxa: which version are u using?`
`14:51.51`	`Ecio`	`if it's 4.x u can`
`14:51.59`	`[TK]D-Fender`	`sevard : How about you PROVE your current system date before thinking you know what you're doing? :)`
`14:52.04`	`Ecio`	`one moment please`
`14:52.07`	`Ecio`	`telefone rining`
`14:52.08`	`Ecio`	`ring`
`14:52.16`	`[TK]D-Fender`	`sevard : Dont try and cheat the process!`
`14:52.28`	`sevard`	`Tue May 30 09:52:20 CDT 2006`
`14:52.31`	`ecio_tel`	`yxa: one moment`
`14:52.34`	`yxa`	`Ecio ok i'll check it later. that means CCM will do transcoding? i'm really interested to know how it'll perform :)`
`14:52.44`	`[TK]D-Fender`	`sevard : Ok, so set it for 11am.`
`14:52.49`	`[TK]D-Fender`	`sevard : Err 10am`
`14:52.50`	`ecio_tel`	`yxa 5 min and i'll beb ack`
`14:53.28`	`[TK]D-Fender`	`yxa : Well * will transcode if it can even bring up a channel.. in fact it will HAVE to since they are speaking different languages anyways`
`14:53.40`	`sevard`	`-rw-r--r-- 1 asterisk nogroup 139 2006-05-30 11:23 1123.ext.0.call`
`14:53.43`	`sevard`	`11:23`
`14:53.51`	`sevard`	`put in /var/spool/asterisk/outgoing`
`14:54.00`	`sevard`	`WOW WEIRD IT CALLS ME AS SOON AS IT'S IN THERE Mr. CAPS`
`14:54.18`	`sevard`	`:\|`
`14:54.19`	`sevard`	`sorry.`
`14:55.03`	`yxa`	`[TK]D-Fender i always thought CCM cant do SIP, until ecio_tel said otherwise`
`14:56.05`	`*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.95)`
`14:56.27`	`[TK]D-Fender`	`sevard : how are you putting it in there?`
`14:56.43`	`sevard`	`cp, mv, how should I put it in there`
`14:56.47`	`[TK]D-Fender`	`mv`
`14:56.57`	`fugitivo`	`[TK]D-Fender: what was the chipset you said is not compatible with digium cards?`
`14:57.18`	`[TK]D-Fender`	`fugitivo : One I'm aware of is the i7205`
`14:57.55`	`fugitivo`	`ok, thanks`
`14:59.12`	`Conductor`	`is there a way to cancel "stop gracefully"?`
`14:59.35`	`[TK]D-Fender`	`Conductor : Don't think so, but thats a great idea...`
`14:59.35`	`kaz0358`	`kill -9 ;)`
`14:59.41`	`flujan`	`coppice, could you develop a patch to make mcfr2 work with the brazilian protocol?`
`14:59.53`	`flujan`	`coppice, I can speak with my boss about a contribution...`
`15:00.16`	`coppice`	`it does work with the brazilian protocol. its your setup that isn't working. there is no audio being decoded`
`15:00.58`	`ecio_tel`	`yxa im back`
`15:01.17`	`Ecio`	`yxa: i've just setup a SIP trunk between CCM 4.1 and asterisk`
`15:01.22`	`Ecio`	`so i know it works :)`
`15:01.41`	`flujan`	`coppice, the problem is that the legacy pbx doesn't send all the digits of the number... if I dial 12345678 in a legacy pbx extension, it just send the number 1 and freeze`
`15:02.06`	`yxa`	`Ecio yep i definitely have 4.x`
`15:02.32`	`flujan`	`coppice, the guys said asterisk should "ask" for the other digits... You said it aren't receiving the digits but is asking... I dunno what can I do.`
`15:02.39`	`Ecio`	`yxa: i've successfully established the trunk between CCM and * and i can call from CM to * and viceversa`
`15:02.45`	`Ecio`	`dont know about office live communication server`
`15:03.04`	`coppice`	`the other end should send the first digit. then my software should ask for the next one`
`15:03.06`	`Ecio`	`what are u using on the office side, windows messenger?`
`15:03.33`	`yxa`	`Ecio some soft phones`
`15:03.37`	`flujan`	`coppice, yes... they are sending the first digit... But they said asterisk is not asking for the next...`
`15:03.49`	`flujan`	`coppice, at least, it was what they said.`
`15:03.54`	`flujan`	`coppice, :(`
`15:03.55`	`*** join/#asterisk assert_true (n=Sunil@59.176.2.162)`
`15:04.07`	`coppice`	`my software is not receiving the first digit, according to the log you pasted`
`15:04.08`	`Ecio`	`maybe u can directly connect office live comm. and ccm without passing from * (of course if u dont need some of the asterisk's features)`
`15:04.21`	`*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)`
`15:04.40`	`yxa`	`Ecio no, for sure ccm does not talk to lcs`
`15:05.08`	`Ecio`	`yxa: well, how can u say that if u didnt know until today that ccm can speak SIP ? :)`
`15:05.26`	`flujan`	`coppice, hum... but in the legacy pbx log it is been sent... Asterisk is at least detecting the event... do you think it can be a problem in my asterisk setup?`
`15:05.53`	`coppice`	`asterisk is not detecting the digit`
`15:06.00`	`yxa`	`Ecio i'm looking rite at it now. lcs uses SIP over TCP. how screwed up is that???`
`15:06.00`	`flujan`	`coppice, it will be easy if i isolate the problem... But it isn't happening.`
`15:06.26`	`*** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com)`
`15:06.35`	`flujan`	`coppice, I will pastebin my unicall.conf, extensions.conf and zaptel.conf could you please take a look?`
`15:06.37`	`Ecio`	`yxa: CCM SIP TRUNK supports SIP over UDP and over TCP, * only over UDP`
`15:06.50`	`PMantis`	`New call center... Polycom or SNOM phones?, why?`
`15:06.59`	`Ecio`	`so u have much more chances of making lcs to talk with ccm than with * imho :)`
`15:07.44`	`Ecio`	`yxa: btw windows messenger client supports both UDP and TCP, are u sure there's not some hidden UDP option in lcs ?`
`15:07.52`	`[TK]D-Fender`	`PMantis : Polycom. Solid firmware, great audio, great price`
`15:08.12`	`Ecio`	`try to create a sip trunk in the ccm, it's on device -> trunk`
`15:08.22`	`*** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl)`
`15:08.26`	`Tagor`	`Hi`
`15:08.33`	`Tagor`	`I've the following setup:`
`15:08.43`	`Ecio`	`and dont remember (as i did... and lost two days tryin to understand the problem) to specify the CSS for the incoming calls that are received from the CCM on the sip trunk`
`15:08.52`	`Tagor`	`Grandstream GXP 2000 -> asterisk -> SIP Provider -> Internet`
`15:09.13`	`Tagor`	`Now I need to set dtmfmode=inband for outgoing calls`
`15:09.19`	`Tagor`	`Else an external IVR doesn't work`
`15:09.20`	`*** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk)`
`15:09.29`	`Tagor`	`But when I do this, then my own IVR doesn't work anymore`
`15:09.33`	`[TK]D-Fender`	`Tagor : Set your ITSP peer entry to that mode.`
`15:09.36`	`*** join/#asterisk azzie (n=az@azzie.net)`
`15:09.39`	`hypnox`	hmm.. fresh compile of 1.2.7, mpg123 compile fails with *** No rule to make target `\
`15:09.44`	`flujan`	`coppice, http://pastebin.com/746901`
`15:09.50`	`yxa`	`Ecio i swear i have tried that. but let me reconfirm that tomorrow. will be back here to look for you :) thanks`
`15:09.52`	`azzie`	`does anybody has a recording of "dinars" and "piastres" ? :)`
`15:10.01`	`Tagor`	`Is there a proper way to use dtmfmode=auto for incoming calls and dtmfmode=inband for outgoing calls?`
`15:10.07`	`Ecio`	`yxa: you're welcome`
`15:10.07`	`hypnox`	`anyone have any idea? did someone break it?`
`15:10.14`	`Tagor`	`ITSP peer = SIP provider, [TK]D-Fender?`
`15:10.22`	`[TK]D-Fender`	`Tagor : correct`
`15:11.00`	`Tagor`	`[TK]D-Fender -> I only have one entry in sip.conf: type=friend`
`15:11.08`	`*** join/#asterisk SplasPood (n=jwb@206.252.198.100)`
`15:11.34`	`coppice`	`flujan: that looks OK`
`15:12.09`	`coppice`	`flujan: as a said. a number of people have had exactly what you have, and the problem has been something in the config at the other end`
`15:12.10`	`[TK]D-Fender`	`Tagor : well add the dtmfmode=inband then`
`15:12.27`	`Tagor`	`[TK]D-Fender >> That entry is for both incoming and outgoing calls`
`15:12.50`	`Tagor`	`[TK]D-Fender >> As said if I set it to dtmfmode=inband then my own IVR doesn't work anymore`
`15:13.18`	`flujan`	`coppice, shit... :( I was hoping I've made some mistakle... :) OK, i will ask the guys... thanks coppice. I will try another solution with the legacy pbx's guys.`
`15:13.42`	`[TK]D-Fender`	`Tagor : Don't do it in [general] do it in your ITSP peer entry`
`15:14.25`	`Tagor`	`[TK]D-Fender >> I don't have an [general]. I just contains: context=mysipprovider`
`15:14.34`	`[TK]D-Fender`	`Tagor : Pastebin the whole thing`
`15:14.37`	`[TK]D-Fender`	`~pb`
`15:14.39`	`jbot`	`i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca`
`15:14.41`	`Tagor`	`Ok, second :)`
`15:15.11`	`asterboy`	`ls`
`15:15.15`	`asterboy`	`ls -l`
`15:16.10`	`mikefoo`	`I am trying to figure out what processes is blocking processes from read/write on, I see in vmstat it jumps to 7 as process wait - what utility can tell me more information?`
`15:16.47`	`kaz0358`	`is it possible to do something like... GotoIf(condition?label:label+2) ?`
`15:16.52`	`froguz`	`somebody has connected asterisk to a nec neax 2000 ips?`
`15:16.54`	`Tagor`	`[TK]D-Fender >> http://pastebin.ca/59937`
`15:17.36`	`*** join/#asterisk salviadud (n=ralfalfa@201.133.207.93)`
`15:17.53`	`[TK]D-Fender`	`Tagor : What kind of phone is your SIP phone?`
`15:18.03`	`Tagor`	`Grandstream GXP 2000`
`15:18.32`	`[TK]D-Fender`	`tagor : http://pastebin.ca/59940`
`15:19.13`	`Tagor`	`[TK]D-Fender >> Problem is that if I set dtmfmode=inband then my IVR doesn't work`
`15:19.26`	`Tagor`	`[sipprovider]`
`15:19.31`	`Tagor`	`does also the incoming calls`
`15:19.53`	`[TK]D-Fender`	`Tagor : IVR doesn't work from where?`
`15:20.06`	`Tagor`	`If I call with my home phone to asterisk`
`15:20.19`	`[TK]D-Fender`	`Tagor : You don't sue 1 DTMF mode for outgoing, and another for incoming... that makes no sense`
`15:20.33`	`Tagor`	`Well, else it doesn't work`
`15:20.40`	`[TK]D-Fender`	`Tagor : What is "home phone"?`
`15:20.55`	`Tagor`	`Just a normal PSTN phone`
`15:21.01`	`Tagor`	`Same problem with my mobile phone`
`15:21.05`	`[TK]D-Fender`	`Tagor : Coming in on that SIP provider?`
`15:21.33`	`*** part/#asterisk vechers (i=vechers@64.61.117.138)`
`15:21.35`	`Tagor`	`Yes`
`15:21.52`	`[TK]D-Fender`	`Tagor : And with a SIP provider you should probably be using RFC2833. Check with them as to what they support.`
`15:22.07`	`Hmmhesays`	`what town do you live in?`
`15:22.10`	`sevard`	`owie my forhead`
`15:22.14`	`sevard`	`too bad you haven't figured that out`
`15:22.16`	`sevard`	`bizzilch`
`15:22.18`	`*** part/#asterisk a1fa (n=a1fa@207.210.210.202)`
`15:22.32`	`PMantis`	`[TK]D-Fender, tagor, I've seen that happen here...`
`15:22.55`	`[TK]D-Fender`	`Tagor : Could be your audio is so choppy that inband fails.`
`15:23.06`	`asterboy`	`Is there a way to test mp3 files for output levels and * quality?`
`15:23.13`	`[TK]D-Fender`	`Tagor : call them up to confirm RFC2833 support`
`15:23.28`	`[TK]D-Fender`	`asterboy : Play them.`
`15:23.48`	`*** join/#asterisk Splas (n=jwb@206.252.198.101)`
`15:24.38`	`asterboy`	`They play fine, but its a different story when it comes time for *`
`15:24.55`	`asterboy`	`some don't play well and some seg fault sox on conversion`
`15:25.15`	`asterboy`	`otherwise they play fine in windows`
`15:25.32`	`asterboy`	`4489 Segmentation fault sox -r 44100 -w -s -c 1 "$BASEFILE.raw" -r 8000 -c 1 "$BASEFILE.wav"`
`15:26.19`	`qdk_`	`assert_true: ${BASEFILE}`
`15:26.36`	`qdk_`	`ups`
`15:26.43`	`qdk_`	`asterboy: that was for you.`
`15:26.55`	`asterboy`	`ok, trying`
`15:27.18`	`Tagor`	`PMantis >> Have you got an idea how to fix this?`
`15:27.31`	`Tagor`	`[TK]D-Fender >> Just tried RFC2833 on both grandstream and SIP provider`
`15:27.35`	`*** join/#asterisk mega (n=mega@2001:618:400:7fe3:213:10ff:fe8a:f8dd)`
`15:27.38`	`Tagor`	`[TK]D-Fender >> Then incoming works, outgoing not`
`15:27.52`	`*** part/#asterisk mega (n=mega@2001:618:400:7fe3:213:10ff:fe8a:f8dd)`
`15:28.12`	`*** join/#asterisk MGSsancho (n=user@adsl-67-126-140-26.dsl.irvnca.pacbell.net)`
`15:28.28`	`PMantis`	`Tagor, If your provider does need a different setting for inbound vs outbound... then setup 1 peer and 1 user, instead of combining them into 1 friend.`
`15:28.56`	`Tagor`	`PMantis >> I was thinking of that too, but I have no idea how to relize that`
`15:29.22`	`Tagor`	`Is there a way to split it so it takes one for outgoing and one for incoming?`
`15:29.42`	`[TK]D-Fender`	`Tagor : PMantis just told you want to do.`
`15:30.01`	`[TK]D-Fender`	`Tagor : Peer = outgoing, user = incoming`
`15:30.17`	`[TK]D-Fender`	`PMantis : How big a call center you planning?`
`15:30.50`	`Tagor`	`Sorry, I don't understand that, [TK]D-Fender, what do you mean with user = incoming?`
`15:31.13`	`[TK]D-Fender`	`Tagor : You need to learn how users & peers work. go read the book.`
`15:31.15`	`[TK]D-Fender`	`~book`
`15:31.17`	`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`
`15:31.24`	`Tagor`	`I need to make two [sipprovider]'s with one type=user and one type=friend?`
`15:33.37`	`PMantis`	`[TK]D-Fender, About 15 agents`
`15:33.57`	`sevard`	`[TK]D-Fender: now it's not working at all`
`15:33.59`	`[TK]D-Fender`	`PMantis : PoE? Need speakerphone?`
`15:34.12`	`[TK]D-Fender`	`sevard : www.drphil.com`
`15:34.20`	`sevard`	`solves all my problems :)`
`15:34.35`	`*** join/#asterisk wunderkin (i=kev@69.26.192.234)`
`15:34.43`	`Ecio`	`pmantis: what about "friend" ?`
`15:35.07`	`Hmmhesays`	`sevard paypal me a 10 and i'll give you my rhinonews account for the next 11 days`
`15:35.18`	`sevard`	`10 dollars, 11 days.`
`15:35.21`	`Ecio`	`s/pmantis/tagor`
`15:35.43`	`asterboy`	`still seg fault.`
`15:35.44`	`Hmmhesays`	`woooo`
`15:35.45`	`sevard`	`how about I paypal you a bag of shit and you give me your account for 2 months`
`15:36.00`	`asterboy`	`has to be my lfs/hardware combo`
`15:37.30`	`PMantis`	`[TK]D-Fender, no PoE, only some need speakerphones.`
`15:37.35`	`Tagor`	`Thanks guys, especially [TK]D-Fender, got it working now with type=peer/user`
`15:37.51`	`Tagor`	`But correct me if I am wrong, peer = incoming not outgoing`
`15:38.26`	`PMantis`	`Tagor, You can make or take calls with a peer`
`15:38.30`	`PMantis`	`Same as with user`
`15:38.37`	`PMantis`	`:-)`
`15:38.57`	`[TK]D-Fender`	`PMantis : IP 301 w/o Speakerphone = $115 ; IP 501 = $170 w/ Speakerphone.`
`15:39.27`	`salviadud`	`you can't paypal a bag of shit, you need fedex for that`
`15:40.06`	`Ecio`	`is there a way to manipulate the callerid (sip -> sip) in order to dinamically add a digit at the beginning? i have this trunk A -> B and i use "6" on A to prefix the call that should be routed to B (6XXX), but when B calls A it shows only the XXX. I was wondering if i can fix it`
`15:40.08`	`PMantis`	`[TK]D-Fender, Just what I was thinking. Had to ask again to be sure... Thanks!`
`15:40.15`	`froguz`	`i have a nec pbx extension conected to a TDM400P, when i try to get his dial tone ( with exten => 9,1,Dial(Zap/1/,90) ) the channel answer, but i can't listen any tone. however, if i connect an ordinary phone directly to the extension it gives me dialtone.`
`15:41.00`	`froguz`	`i have connected the extension to the first tdm400p span (upper)`
`15:41.23`	`[TK]D-Fender`	`PMantis : New IP 430 is a great choice, but mostly if you have PoE in mind`
`15:41.24`	`asterboy`	`ya, I can't tell what file is going to work with * until I play it.`
`15:41.28`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)`
`15:41.37`	`froguz`	`do am i missing something?`
`15:42.31`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`15:42.40`	`asterboy`	`be nice to have some suggested parameters for mp3s, like bit and sample rates`
`15:43.00`	`asterboy`	`some are reported as having junk at the beginning.`
`15:43.22`	`asterboy`	`also , would be nice to have a utility to filter the files so that clipping does not occour during playback`
`15:44.09`	`*** join/#asterisk Ox0000 (n=null@84-72-173-86.dclient.hispeed.ch)`
`15:45.51`	`asterboy`	`this just soxs!`
`15:46.06`	`asterboy`	`4915 Segmentation fault sox -r 44100 -w -s -c 1 ${BASEFILE}.raw -r 8000 -c 1 ${BASEFILE}.wav`
`15:46.55`	`CunningPike`	`Ecio: Look at Set(CALLERID(number))`
`15:48.25`	`Ecio`	`thx`
`15:48.50`	`[TK]D-Fender`	`Ok, I"m outfor a bit`
`15:49.26`	`*** join/#asterisk Alric (n=nbowyer@ppp-db.1stel.com)`
`15:50.30`	`asterboy`	`Well seems my * likes Johny Cash - Ring of Fire.`
`15:50.35`	`asterboy`	`some clipping though.`
`15:50.51`	`asterboy`	`and the flames went higher....and it burns burns burns`
`15:51.04`	`iq`	`yo`
`15:51.12`	`asterboy`	`doesn't like Fat Britney Spears though`
`15:51.39`	`asterboy`	`Britney Spears looks like Fat Elvis now.`
`15:52.02`	`Tagor`	`Picture? :D`
`15:52.30`	`MrChimpy`	`don't be mean to elvis`
`15:52.40`	`MrChimpy`	`at least he could sing`
`15:53.14`	`asterboy`	`ya, he had an interesting 2 part life...Hollywood Elvis and Fat Las Vegas Elvis`
`15:53.36`	`*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)`
`15:53.40`	`MrChimpy`	`and fat las vegas elvis was his funkiest stage`
`15:54.09`	`asterboy`	`That is probably why impersonators choose to copy the fat Elvis.`
`15:54.35`	`*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)`
`15:55.43`	`asterboy`	`those big Herculean Belt Buckles`
`15:57.34`	`*** join/#asterisk Assid (n=assid@203.115.83.214)`
`15:58.08`	`*** part/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt)`
`15:58.21`	`*** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt)`
`16:01.24`	`froguz`	`please, can anybody give a clue on why can't i get dial tone from the NEC PBX with the method mentioned above?`
`16:01.44`	`froguz`	`i'm very, very confussed`
`16:03.25`	`*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)`
`16:05.53`	`froguz`	`it seems the whole channel is going to have lounch... =(`
`16:06.41`	`Sonderblade`	`when a line is busy and asterisk sends 486 to a phone, is there a way to control how many busy tones the ip-phone plays before ending the call?`
`16:07.48`	`*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)`
`16:09.09`	`*** join/#asterisk Martz (n=martz@pdpc/supporter/active/Martz)`
`16:09.22`	`froguz`	`Sonderblade, look for an option called busycount`
`16:09.49`	`Sonderblade`	`froguz: in the phone or in asterisk?`
`16:10.54`	`charles___`	`Hey anyone into REDUNDANT ASTERISK ?`
`16:11.04`	`*** part/#asterisk Alric (n=nbowyer@ppp-db.1stel.com)`
`16:11.32`	`MrChimpy`	`kinky`
`16:11.37`	`froguz`	`Sonderblade, http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf press ctrl+f and type busycount`
`16:13.20`	`Sonderblade`	`froguz: thanks but i don't see how that relates to sip-phones?`
`16:15.16`	`*** join/#asterisk Ariel_ (n=Ariel@70.46.87.158)`
`16:16.20`	`*** join/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226)`
`16:17.33`	`froguz`	`Sonderblade, do you want to change how many times the phone will play the busy tone? i think the phone will continue playing the busy tone until * detect it and finish the call`
`16:18.05`	`Sonderblade`	`froguz: yes i want to change that`
`16:19.19`	`froguz`	`what ip phone are you using?`
`16:20.51`	`*** join/#asterisk suma (n=suma@222.165.116.228)`
`16:22.11`	`*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)`
`16:25.13`	`Sonderblade`	`froguz: different models of grandstream`
`16:25.51`	`Sonderblade`	`froguz: my guess is that the # of busy tones is configured in the phone itself but im not sure`
`16:26.11`	`*** part/#asterisk _alex_mx_ (n=_alex_mx@200.94.154.226)`
`16:27.40`	`*** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com)`
`16:30.24`	`The_X`	`do I need to register my patton smartnode to Asterisk to make it forward disconnection notices?`
`16:31.49`	`*** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net)`
`16:32.49`	`*** join/#asterisk jtodd (n=jtodd@reserve-64-79-115-18.wiline.com)`
`16:32.54`	`*** join/#asterisk jahani (n=k@41.250.49.207)`
`16:33.08`	`CrashHD`	`anyone know a way to do linepark keys on a 942 (BLFS)?`
`16:33.49`	`*** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net)`
`16:39.33`	`obiwanmikenolte`	`Has anyone gotten distinctive ringing to work on a Polycom? I've been trying to follow the guidelines from voip-info, but I'm getting nowhere. Even using the "Visual" ring that's predefined in my sip.cfg doesn't work`
`16:39.47`	`*** join/#asterisk slobberknocker (n=ckwall@63.149.122.94)`
`16:39.49`	`kay2`	`Someone is familiar with asterisk Queue ?`
`16:43.20`	`Ecio`	`bye all`
`16:46.05`	`*** join/#asterisk bjohnson (n=bjohnson@216.58.51.69)`
`16:48.04`	`marl`	`can someone help with an incoming IAX problem? ive followed the documentation i can find, and help from here earlier, but i still cant get this working :( iax.con/extensions.con and iax debug output at : http://pastebin.com/747073`
`16:49.09`	`*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)`
`16:50.22`	`marl`	`i think ive tried so many things now, im totaly lost :(`
`16:50.54`	`kay2`	`marl: iax.conF`
`16:51.09`	`marl`	`ok, what did i do rong?`
`16:51.21`	`CrashHD`	`files need to be named .conf`
`16:51.35`	`CrashHD`	`(by default anyway)`
`16:51.37`	`marl`	`sorry all files are named .conf, miss typed`
`16:52.23`	`Nugget`	`That's Ms. Typed to you.`
`16:52.39`	`marl`	`lol :P`
`16:53.28`	`*** join/#asterisk Bamtang (n=Adam@200.121.189.241)`
`16:54.02`	`*** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it)`
`16:54.12`	`Bamtang`	`I have a small office, 12 users all on computers, a wireless LAN and 2 phone lines`
`16:54.20`	`marl`	`so anyone any idea what i did wrong?`
`16:54.41`	`Bamtang`	`Is it possible to use Asterisk to set up a PBX for all the users?`
`16:54.57`	`marl`	`Bamtang, yup`
`16:55.08`	`Bamtang`	`Is it just a matter of buying an analogue card`
`16:55.13`	`marl`	`got simalar setup here`
`16:55.17`	`Bamtang`	`and installing the software on the server?`
`16:55.25`	`mr_horsepower`	`Bamtang: damm, it should not be called pbx if it does not do that.`
`16:55.32`	`marl`	`thats the thery anyway :)`
`16:55.45`	`Bamtang`	`The digium site is not user friendly for non phone techies`
`16:56.04`	`Bamtang`	`I'm an engineer and we are a video game dev co`
`16:56.12`	`mr_horsepower`	`Bamtang: for non-phone techies, it's not on the click that you use asterisk.`
`16:56.19`	`*** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net)`
`16:56.27`	`The_X`	`anyone using a 3rd party sip gateway?`
`16:56.40`	`mr_horsepower`	`just have to take some time with it, not dificult, and you have a lot of documentation.`
`16:57.02`	`Bamtang`	`but there is no one in Peru that can help me`
`16:57.11`	`mr_horsepower`	`Bamtang: engineer dont help, do you know linux?`
`16:57.18`	`Bamtang`	`If I buy the card and business package do you think I will be able to get it going?`
`16:57.32`	`Bamtang`	`Some of the programmers here do`
`16:57.34`	`marl`	`Bamtang, take an old box, and install asterisk@home, it is a very good place to start from, once u know your way around a bit, flatten the box and install asterisk on its own`
`16:57.53`	`mr_horsepower`	`Bamtang: so buy the card, forget the business package, and you should be able to get it working.`
`16:58.31`	`Bamtang`	`oh, so each of the PC's would just need a mike/earpiece head set?`
`16:58.32`	`*** join/#asterisk sb_mx (n=sb_mx@200.94.154.226)`
`16:58.44`	`mr_horsepower`	`Bamtang: yes, and a software phone.`
`16:58.52`	`Bamtang`	`and a Windoz client program I suppose`
`16:59.08`	`mr_horsepower`	`windows, linux, console macosx`
`16:59.12`	`mr_horsepower`	`whatever you need`
`16:59.25`	`Bamtang`	`So why is it not more popular in small offices?`
`16:59.26`	`mr_horsepower`	`i missed a comma after console.`
`16:59.39`	`*** join/#asterisk fugitivo (n=ajf@190.48.164.43)`
`16:59.58`	`Bamtang`	`just one card and no cables to pull... and a server`
`17:00.35`	`Bamtang`	`seems too good to be true`
`17:01.36`	`Bamtang`	`Is there any way to install the Windoz client and try it out without having it all locally (aka Skype...)?`
`17:01.39`	`mr_horsepower`	`Bamtang: because usually ppl dont like to mess arround with nothing.`
`17:02.22`	`mr_horsepower`	`Bamtang: you can allways buy a sip account somewhere, but you are not testing nothing`
`17:02.26`	`*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.59.Dial1.SanJose1.Level3.net)`
`17:02.49`	`Bamtang`	`I mean to test the client. Does it pause your music or anything?`
`17:02.50`	`mr_horsepower`	`that's a lot diferent when testing accounts from somewhere, where someone have spend many month's woring on it.`
`17:02.53`	`mr_horsepower`	`working`
`17:03.07`	`mr_horsepower`	`test the client? who cares about the client? :D`
`17:03.12`	`Bamtang`	`Is voice mail manageable?`
`17:03.22`	`mr_horsepower`	`Bamtang: it does everything.`
`17:03.44`	`Bamtang`	`Can it be tested without setting the whole thing up on a local server?`
`17:03.58`	`mr_horsepower`	`please read more, and just ask tecnical questions, if you want everything done, not worry, pay someone to do it.`
`17:04.11`	`Bamtang`	`Is there a simulation app that shows you what it would be like to receieve a call or voice mail while you are working?`
`17:04.48`	`Bamtang`	`I've been through all the sites and I cant find a sim client or web client demo...`
`17:05.24`	`mr_horsepower`	`contact some company, that can show you everything, and next pay them.`
`17:06.03`	`Bamtang`	`So a client like that it does not exist? I'm in Peru - there's no one to call.`
`17:06.07`	`mr_horsepower`	`pbx areas, moves a lot of money all arround the world`
`17:06.34`	`mr_horsepower`	`there is no one in peru that can help you?`
`17:06.45`	`mr_horsepower`	`thats very weird.`
`17:06.49`	`Bamtang`	`cant find any companies running asterisk`
`17:07.54`	`mr_horsepower`	`Bamtang: http://www.asterisk-peru.com/`
`17:07.58`	`Bamtang`	`thanks for the help`
`17:08.16`	`Bamtang`	`ciao`
`17:09.24`	`*** join/#asterisk \|dennis\| (i=dennis@200.32.215.84)`
`17:10.29`	`CrashHD`	`how can I debug mwi?`
`17:11.23`	`CrashHD`	`ok this is odd`
`17:11.36`	`CrashHD`	`the sip simple messages that are being sent to the phone are 0/0`
`17:11.50`	`CrashHD`	`but I know there are atleast 2 messages at that extensions mailbox`
`17:12.13`	`Dr-Linux`	`hi`
`17:13.12`	`Dr-Linux`	`if the net is DHCP, should i only enable DHCP to YES on 7940, or i need to define DHCP server ip adress at the phone?`
`17:13.16`	`mut`	`http://img211.imageshack.us/img211/8007/howsmart4xg.jpg`
`17:13.33`	`mr_horsepower`	`Dr-Linux: do you know what means dhcp?`
`17:13.48`	`Dr-Linux`	`mr_horsepower: yes`
`17:13.50`	`mr_horsepower`	`dhcp it's layer2, there is no ip.`
`17:14.17`	`*** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net)`
`17:14.20`	`Dr-Linux`	`mr_horsepower: yes, i know, but not sure why the phone is not grabing one IP address from the DHCP server :(`
`17:14.31`	`mr_horsepower`	`check dhcpd log's`
`17:14.58`	`*** join/#asterisk saftsack (n=saftsack@p54A7F843.dip.t-dialin.net)`
`17:15.04`	`Dr-Linux`	`mr_horsepower: this system is kinda odd, i'm doing installation from remote end.`
`17:15.45`	`Dr-Linux`	`mr_horsepower: their system is something like >> Cable from wall >>> DSL modem >>> Hub >> "here all PC's"`
`17:16.03`	`Dr-Linux`	`mr_horsepower: every PC's grabs public dynamic IP address`
`17:16.36`	`Dr-Linux`	`and everytime new IP address`
`17:17.25`	`mr_horsepower`	`all public ip addr? damm weird.`
`17:17.48`	`mr_horsepower`	`so, there are ip addr avaliable for the phone?`
`17:18.15`	`slobberknocker`	`ok, I am not sure what terms or phrases to look for in the wiki and such for what I am trying to do. Can someone help me figure out what phrase to search for? I want to be able to change the way an extension works based on time of day. for example between the hours of 8-5 are open, and from 5-8 closed. what should i be looking for?`
`17:19.54`	`Dr-Linux`	`mr_horsepower: that's what strange to me`
`17:19.56`	`*** join/#asterisk jbailey (n=jbailey@modemcable139.249-203-24.mc.videotron.ca)`
`17:19.59`	`*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net)`
`17:20.14`	`Dr-Linux`	`mr_horsepower: no, the phone doesn't get an IP address from the DHCP, not sure why`
`17:20.22`	`PMantis`	`Is there really a benefit to echo cancelation on a Digium TE device?`
`17:22.06`	`*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)`
`17:22.24`	`Nugget`	`ppeerrhhaappss,, ddeeppeennddiinngg oonn yyoouurr ssiittuuaattiioonn.`
`17:23.50`	`*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)`
`17:25.02`	`[TK]D-Fender`	`slobberknocker : "show application GotoIfTime"`
`17:25.53`	`saftsack`	`hi`
`17:26.19`	`saftsack`	`are there some news concerning to the b410p card?`
`17:26.30`	`slobberknocker`	`gotoiftime... thanks`
`17:26.32`	`slobberknocker`	`found it`
`17:27.57`	`x86`	`gah, for some reason asterisk wont connect to my CDR database server`
`17:28.33`	`[TK]D-Fender`	`saftsack : The first rule of B410P is... you don't talk about B410P !!!!`
`17:28.43`	`x86`	`i do a "cdr mysql status" and it just tells me not currently connected to a MySQL server`
`17:28.53`	`x86`	`and no calls are being logged in CDR at all :(`
`17:29.01`	`x86`	`MySQL _or_ flatfile`
`17:30.14`	`*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)`
`17:31.10`	`x86`	`nothing showing up in the debug logs about it...`
`17:31.57`	`stephane_`	`re`
`17:31.58`	`PMantis`	`Is there any point to "echo canceling" a digital PRI line?`
`17:32.42`	`robl^`	`Digital echo? 0011 0011 0011 0011?`
`17:32.51`	`[TK]D-Fender`	`PMantis : HELL YEAH`
`17:33.03`	`[TK]D-Fender`	`PMantis : You end up analog SOMEWHERE along that call`
`17:33.18`	`[TK]D-Fender`	`PMantis : Keep in mind who you're CALLING as well...`
`17:33.36`	`saftsack`	`[TK]D-Fender, why that?`
`17:33.44`	`[TK]D-Fender`	`PMantis : And ask yourself about the myriad people here who've gone bald due to frustration with it.`
`17:33.49`	`PMantis`	`[TK]D-Fender, ok, point taken.`
`17:33.56`	`[TK]D-Fender`	`saftsack : Its a joke.....`
`17:34.07`	`[TK]D-Fender`	`saftsack : Based on the movie Fight Club.`
`17:36.01`	`x86`	`so no one knows how to fix my issue?`
`17:36.40`	`x86`	`it worked fine when mysql was on the same box as asterisk... now that i'm using a dedicated server for asterisk, and another for mysql, it's not wanting to connect...`
`17:37.07`	`x86`	`I'm using realtime which works fine after the switch, so I know mysql is accepting remote connections properly`
`17:37.30`	`x86`	`and i know the database, user, and password are good, because i use the same for realtime as I do CDR`
`17:39.16`	`*** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com)`
`17:39.20`	`CrashHD`	`how can you easily create holding patterns for extensions? so I can have multiple calls waiting and able to be handled by a user?`
`17:39.22`	`[TK]D-Fender`	`x86 : Bad connect credentials?`
`17:40.00`	`x86`	`[TK]D-Fender: err, no... same as realtime credentials... which work fine`
`17:40.25`	`[TK]D-Fender`	`x86 : bad table name / structure?`
`17:40.45`	`x86`	`err no, it used to work fine...`
`17:40.53`	`[TK]D-Fender`	`:/`
`17:40.59`	`x86`	`when it was local on the same box as asterisk, worked like a champ`
`17:41.04`	`x86`	`after moving it, it wont connect`
`17:41.22`	`blitzrage`	`Qwell: ping?`
`17:41.29`	`x86`	`i turned on mysql debugging and tracing on my mysql server, and cdr_addon_mysql.so is not even trying to connect to the remote box`
`17:41.36`	`*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)`
`17:42.02`	`x86`	`ah!`
`17:42.07`	`x86`	`nevermind, i found the prob :)`
`17:42.37`	`Sonderblade`	`my sip-provider sends inband dtmf using rfc2833, but when I try use my asterisk's IVR menu from a cell phone, it seems like asterisk doesn't recognize the dtmf tones, anyone know what the problem is?`
`17:43.17`	`*** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV)`
`17:43.34`	`russellb`	`inband dtmf and rfc2833 are 2 different things`
`17:43.38`	`[TK]D-Fender`	`Sonderblade : rfc2833 != inband`
`17:43.48`	`[TK]D-Fender`	`russellb : ! ! !`
`17:43.55`	`russellb`	`i beat you!`
`17:44.00`	`blitzrage`	`I don't want to know your name`
`17:44.06`	`[TK]D-Fender`	`russellb : You know I like it rough ;)`
`17:44.10`	`mut`	`edwardo!`
`17:44.16`	`Sonderblade`	`ok so s/inband dtmf/incoming dtfm tones/`
`17:44.39`	`[TK]D-Fender`	`And now... introducing "Sasso" ... formerly known as RON`
`17:44.40`	`x86`	`[TK]D-Fender: sicko ;)`
`17:45.19`	`*** part/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com)`
`17:45.29`	`[TK]D-Fender`	`:D`
`17:46.09`	`*** join/#asterisk chapeaurouge (n=chapeaur@user-85-201-82-146.tvcablenet.be)`
`17:46.14`	`DarKnesS_WolF`	`x86: ur still everywhere :P`
`17:46.33`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`17:47.29`	`blitzrage`	`Anyone ever use ExecIf() and try to pass multiple arguments to an application? i.e. ExecIf($[${FOO} = 1]\|Macro\|my_macro,arg1,arg2)`
`17:48.16`	`blitzrage`	`placing the values into an variable doesn't seem to make it work -- Asterisk turns the commas into literal, so it look for the macro-my_macro,arg1,arg2 literally`
`17:49.03`	`saftsack`	`[TK]D-Fender, i thought so too but whats going on with this card? why isnt it released yet?`
`17:49.30`	`[TK]D-Fender`	`saftsack : I don't work at Digium... don't ask me!`
`17:49.48`	`*** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com)`
`17:49.55`	`[TK]D-Fender`	`saftsack : I mean sure I'm here all the time, but thats besides the point!`
`17:51.55`	`marl`	`can someone help with an incoming IAX problem? ive followed the documentation i can find, and help from here earlier, but i still cant get this working :( iax.conf/extensions.conf and iax debug output at : http://pastebin.com/747073`
`17:53.35`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-17-113.red.bezeqint.net)`
`17:54.02`	`DarKnesS_WolF`	`[TK]D-Fender: sorry i had to leave work when u told me about this low volum solution ... the other end " which is PSTN analog line " always hears me really in a very low voice... what that could be the problem ?`
`17:54.26`	`[TK]D-Fender`	`DarKnesS_WolF : Your gain on that line obviously.`
`17:54.33`	`[TK]D-Fender`	`DarKnesS_WolF : Up it.`
`17:54.59`	`charles___`	`[TK]D-Fender: hey man`
`17:55.08`	`charles___`	`[TK]D-Fender: did you see the guy with 512 concurrent calls ?`
`17:55.16`	`*** part/#asterisk azzie (n=az@azzie.net)`
`17:55.59`	`DarKnesS_WolF`	`[TK]D-Fender: sorry i didn't understand ?`
`17:56.45`	`[TK]D-Fender`	`DarKnesS_WolF : INCREASE THE GAIN IN ZAPATA. Not clear?`
`17:57.08`	`[TK]D-Fender`	`charles___ : Nope... but that could be lot...`
`17:57.26`	`DarKnesS_WolF`	`[TK]D-Fender: now get it :-) i don't know what is the GAIN is .. but i'll google / read about it :-) thx ;-)`
`17:57.48`	`[TK]D-Fender`	`DarKnesS_WolF : GAIN = volume.`
`17:58.04`	`*** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)`
`17:58.07`	`[TK]D-Fender`	`DarKnesS_WolF : "rxgain" and "txgain" are the options to tweak.`
`17:58.51`	`DarKnesS_WolF`	`[TK]D-Fender: thx alot for ur help :-) will do so and let u know the results tomorrow at work`
`17:59.23`	`[TK]D-Fender`	`DarKnesS_WolF : np.`
`18:01.24`	`Vorondil`	`hi all, i'm setting up an asterisk box to replace an a@h machine we currently use with teliax. however, i can't get my new machine to dial out correctly. http://pastebin.com/747202 there is the error i get dialing out and copies of extensions.conf and iax.conf. anybody know what's up?`
`18:07.14`	`gandhijee`	`who was it here that said the Intel HMP software was crap?`
`18:07.15`	`[TK]D-Fender`	`Vorondil : I strongly suspect you need to dial 10 digit numbers with Teliax... you only dialed 7 as a local number. try it with your area code maybe.`
`18:07.51`	`Juggie`	`yeah thats probally it`
`18:07.53`	`*** join/#asterisk zotz (n=zotz@24.244.133.115)`
`18:08.07`	`Vorondil`	`[TK]D-Fender: hmm okay, i'll try that`
`18:09.33`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`18:11.18`	`salviadud`	`a@h sucks btw`
`18:11.34`	`Vorondil`	`salviadud: indeed :-P`
`18:12.33`	`[TK]D-Fender`	`Vorondil : So, did it work?`
`18:16.01`	`*** join/#asterisk littlejohn (n=little@host123-81.pool877.interbusiness.it)`
`18:16.35`	`Vorondil`	`well, i'm having networking issues atm too >:/`
`18:18.16`	`[TK]D-Fender`	`Vorondil : :/`
`18:20.18`	`*** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com)`
`18:20.49`	`charles___`	`[TK]D-Fender: I'm thinking about a Quad (Dual Core) Opteron at 2Ghz to run 512 call simultaneous`
`18:21.03`	`blitzrage`	`probably won't be enough CPU`
`18:21.09`	`blitzrage`	`definately not with transcoding`
`18:21.12`	`charles___`	`[TK]D-Fender: but the problem that I see is how to record all those calls`
`18:21.35`	`[TK]D-Fender`	`charles___ : Ummm... what are you using for phones & PSTN in this scenario of yours?`
`18:21.36`	`salviadud`	`mixmonitor?`
`18:21.54`	`*** join/#asterisk Blackthorn (i=blacktho@72.236.88.10)`
`18:21.57`	`[TK]D-Fender`	`blitzrage : a fraction of that would kill with transcoding....`
`18:23.01`	`charles___`	`[TK]D-Fender: cisco 5400 -> AST -> sip phones`
`18:23.30`	`Blackthorn`	`Hi, I would like to play a mp3 file as my music on hold. Is the proper command musiconhold(filename.mp3) ? or musiconhold(class) and then in the modemonhold.conf you set the filename.mp3? or niether?`
`18:23.35`	`[TK]D-Fender`	`charles___ : AST?`
`18:23.59`	`*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`18:24.05`	`charles___`	`[TK]D-Fender: asterisk`
`18:24.06`	`[TK]D-Fender`	`Blackthorn : make a dedicated MoH class with it own folder and in it only place 1 MP3`
`18:24.16`	`charles___`	`blitzrage: not enough CPU ?`
`18:24.22`	`Blackthorn`	`ok will do`
`18:24.30`	`[TK]D-Fender`	`charles___ : So * will only be a SIP passthrough with G.711 on all sides?\`
`18:24.55`	`*** part/#asterisk slav_jb (n=k@pirus.securax.be)`
`18:25.08`	`[TK]D-Fender`	`charles___ : And 512 SIMULTANEOUS? Is that a T3-> SIP gateway?`
`18:25.38`	`charles___`	`[TK]D-Fender: yes`
`18:25.50`	`charles___`	`[TK]D-Fender: asterisk will handle menu's and do the recording`
`18:25.54`	`charles___`	`[TK]D-Fender: also conferences`
`18:26.27`	`[TK]D-Fender`	`charles___ : I guess as long as * doesn't have to work to hard it'll all be fine.... make sure GBIT on all sides :)`
`18:26.32`	`charles___`	`[TK]D-Fender: no transcodec and no transprotocol`
`18:27.39`	`[TK]D-Fender`	`charles___ : Ok, its doable....`
`18:29.41`	`[TK]D-Fender`	`charles___ : How much recording?`
`18:29.46`	`charles___`	`[TK]D-Fender: I'm thinking about a scalable solution`
`18:29.52`	`charles___`	`[TK]D-Fender: recording all channels`
`18:29.58`	`[TK]D-Fender`	`charles___ : OMG!`
`18:30.13`	`[TK]D-Fender`	`charles___ : Ok, I'm not qualified to qualify your scenario then :)`
`18:30.31`	`[TK]D-Fender`	`charles___ : And I'd get a nasty RAID array for that setup`
`18:30.34`	`charles___`	`[TK]D-Fender: I'm thinking about using a separated storage server`
`18:32.27`	`charles___`	`[TK]D-Fender: do you know about anyone that works on large ast installs ?`
`18:32.44`	`charles___`	`[TK]D-Fender: for paid consulting`
`18:33.47`	`charles___`	`[TK]D-Fender: I'm reading the stuff that other people did .`
`18:33.54`	`charles___`	`[TK]D-Fender: for example recording all calls to RAM`
`18:34.19`	`[TK]D-Fender`	`charles___ : nOONE WHO DOES SETUPS LIKE THAT....`
`18:35.12`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`18:35.32`	`*** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr)`
`18:35.37`	`jhiver`	`hi all`
`18:35.46`	`jhiver`	`I have a really strange AGI problem`
`18:36.16`	`jhiver`	`I do this:`
`18:36.18`	`jhiver`	`<PROTECTED>`
`18:36.19`	`jhiver`	`<PROTECTED>`
`18:36.19`	`jhiver`	`<PROTECTED>`
`18:36.37`	`jhiver`	`but $answeredtime seems to differ from what's in the Master.csv file`
`18:36.49`	`Vorondil`	`[TK]D-Fender: okay, so dialing out w/ the area code seems to work`
`18:37.15`	`Vorondil`	`[TK]D-Fender: so do i just need to fix up the dial plan to prepend the local area code to 7 digit numbers?`
`18:37.17`	`[TK]D-Fender`	`Vorondil : THERE YOU GO. yOU JUST NEED TO MODIFY YOUR DIALPLAN TO ADD IT WHEN YOU DIAL 7 DIGITS`
`18:37.25`	`[TK]D-Fender`	`Vorondil : yup`
`18:37.39`	`Vorondil`	`hehe, kk`
`18:37.43`	`Vorondil`	`thanks much ^_^`
`18:40.09`	`The_X`	`how do you fix SIP/2.0 407 Proxy Authentication Required`
`18:40.19`	`The_X`	`my setup is t1 -> patton -> asterisk -> phone`
`18:40.27`	`The_X`	`I get that on every inbound call`
`18:40.36`	`file`	`The_X: who says it is broken? it's Asterisk asking the device to authenticate...`
`18:40.47`	`russellb`	`that's normal behavior`
`18:41.21`	`The_X`	`patton is registered to asterisk`
`18:41.22`	`The_X`	`same with phone`
`18:41.26`	`The_X`	`why is it giving me that`
`18:41.34`	`file`	`registration just tells the other side where to send calls if they need to send calls to you`
`18:42.10`	`The_X`	`now help me with something, when I disable sip registration between the patton and asterisk`
`18:42.12`	`The_X`	`everything works fine`
`18:42.16`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`18:42.21`	`The_X`	`when I enable it, incoming calls just keep on ringing`
`18:42.29`	`file`	`then debug the situation`
`18:42.44`	`file`	`do sip debug and pastebin what you see when you try to do whatever...`
`18:42.58`	`charles___`	`[TK]D-Fender: is there a way to load balance between two SIP servers ?`
`18:42.59`	`*** join/#asterisk stephane_ (n=stephane@merlin.cabale.net)`
`18:44.10`	`C4T3l`	`there is no "good" way to do that`
`18:45.22`	`The_X`	`second, why is asterisk taking 30 secs to hang up a call that's been closed by the remote phone`
`18:45.22`	`[TK]D-Fender`	`charles___ : Not inherently. I'd say use SER for that.`
`18:45.32`	`The_X`	`seems like it's not getting the disconnection notice`
`18:46.01`	`file`	`The_X: that depends on the technology... equipment...`
`18:46.19`	`The_X`	`t1 -> patton -> asterisk`
`18:46.19`	`*** join/#asterisk eivindtr (n=wingnut-@217.68.103.66)`
`18:46.28`	`file`	`get me a sip debug and I can tell you.`
`18:48.00`	`eivindtr`	`Hi all. Does anyone know a quick way to determine if a SIP account has an active channel (ie that for 2002, a channel SIP/2002-foo exists)? I need to determine that a user is busy even if he has a phone with multiple lines....`
`18:48.09`	`The_X`	`and often I get bridged another call from some random source`
`18:49.01`	`file`	`The_X: I'm more inclined to say this patton is the source of your issues... but until I see a sip debug I can't tell you anything`
`18:49.15`	`charles___`	`[TK]D-Fender: sorry to bother you`
`18:49.42`	`The_X`	`I'm pretty sure it's the patton`
`18:49.53`	`charles___`	`[TK]D-Fender: but having 2 Asterisk and 1 SER, is it possible for the SIP phone to send the call thru other server if one point crashes ?`
`18:50.09`	`harryvv`	`TK, you have the ip500 series?>`
`18:50.17`	`[TK]D-Fender`	`charles___ : Sounds about right.`
`18:50.24`	`[TK]D-Fender`	`harryvv : I own a 501, yes`
`18:51.01`	`[TK]D-Fender`	`harryvv : I have a few 301, 1 x501, a pile of 600, and 1 x 601`
`18:51.09`	`[TK]D-Fender`	`harryvv : 601 has 2 sidecars`
`18:51.46`	`harryvv`	`yea since yesterday got some of the things on this phone working but its not registering at least not in sip show peers.`
`18:51.49`	`blitzrage`	`can you expand the number of lines available on the 501?`
`18:51.55`	`charles___`	`[TK]D-Fender: if the SER crashes can the SIP phone send the call to the ASTERISK server directly ?`
`18:52.18`	`blitzrage`	`why not just have 2 SER boxes for redundancy?`
`18:52.30`	`blitzrage`	`or else then you have to authenticate the call from the phone instead of from SER`
`18:52.42`	`blitzrage`	`and add logic to handle all that (plus billing)`
`18:54.45`	`harryvv`	`ohh yea those extention side cars.`
`18:54.55`	`harryvv`	`most phones come with those on the base.`
`18:55.34`	`Blackthorn`	`ok got my moh working with mp3. I read the docs that said I should use a program called "lame" to convert my mp3's to a certain spec to reduce cpu... know where i can get lame from? dosn't seem to be on my fc2 box.`
`18:55.39`	`file`	`The_X: okay let's see here...`
`18:55.43`	`The_X`	`yes!`
`18:56.41`	`*** join/#asterisk boch (n=root@201.216.241.97)`
`18:56.55`	`file`	`The_X: it looks fine, the only thing is that a reinvite is occuring so audio is flowing directly... and not through Asterisk`
`18:57.40`	`The_X`	`so I should disable it?`
`18:57.55`	`file`	`you can try`
`18:57.58`	`file`	`canreinvite=no in sip.conf`
`18:58.12`	`The_X`	`I know :), I'm not THAT newbie`
`18:59.07`	`The_X`	`same crap, I'm sure it's the patton not fwd it to asterisk`
`19:03.02`	`[TK]D-Fender`	`harryvv : maybe the phone ISN'T registereing but its passing calls.`
`19:03.18`	`[TK]D-Fender`	`harryvv : I'd need to see your sip.cfg and phonexxx.cfg`
`19:04.47`	`Blackthorn`	`Where can i d/l the program lame to convert mp3 files?`
`19:05.06`	`[TK]D-Fender`	`Blackthorn : depends on your distro`
`19:07.26`	`*** join/#asterisk tsurk0 (n=tsurko@digsys226-159.pip.digsys.bg)`
`19:08.38`	`Blackthorn`	`fedora core 2`
`19:09.10`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`19:09.15`	`harryvv`	`Tk, okay i can send those to you as a file or as a pastebin?`
`19:09.41`	`harryvv`	`Also, what is the purpous of the line 1-3 options?`
`19:09.55`	`boch`	`if i dont specify a secret for a sip peer, will it register even if it sends a password?`
`19:10.21`	`boch`	`or tellme how to break an md5 hash`
`19:10.49`	`[TK]D-Fender`	`harryvv : WELL.. ITS A 3 LINE PHONE....`
`19:10.55`	`harryvv`	`Tk i know that`
`19:11.06`	`harryvv`	`I have never programing the three lines before :)`
`19:11.11`	`[TK]D-Fender`	`harryvv : Pretty self explanitory to me...`
`19:11.34`	`[TK]D-Fender`	`harryvv : And most users use 10% of Excel's functionaily. Join the club!`
`19:11.51`	`harryvv`	`Thats probebly true`
`19:11.54`	`boch`	`10% is too much`
`19:14.11`	`*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)`
`19:14.11`	`*** mode/#asterisk [+o anthm] by ChanServ`
`19:14.49`	`[TK]D-Fender`	`harryvv : I'm going to be using at least 2 lines on mine at home so I can have one key with a seperate dial-plan linked to a custoemr I'm debugging.`
`19:15.26`	`*** part/#asterisk TripleFFFF (n=Miranda@147-102.mc.cite.net)`
`19:15.32`	`*** join/#asterisk TripleFFFF (n=Miranda@147-102.mc.cite.net)`
`19:15.48`	`*** join/#asterisk ToTo (n=ToTo@host224-94.pool8260.interbusiness.it)`
`19:15.59`	`TripleFFFF`	`hey.. in a loop is there a way to make . like $I++; if I$>4 then goto\|bad\|1 ?`
`19:16.23`	`[TK]D-Fender`	`harryvv : So thats 1 reg (line) using 2 line keys @ 1 call/line. Reg 2 = 1 line key supporting a few calls at a time.`
`19:16.28`	`[TK]D-Fender`	`TripleFFFF : Sure`
`19:16.58`	`[TK]D-Fender`	`TripleFFFF : GotoIf = your friend`
`19:17.45`	`TripleFFFF`	`nevermind weird logic i had`
`19:17.49`	`TripleFFFF`	`ill just push to sales queue`
`19:17.49`	`TripleFFFF`	`lo,l`
`19:18.04`	`[TK]D-Fender`	`:/`
`19:18.21`	`*** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com)`
`19:18.58`	`*** join/#asterisk bugz (n=will@69.15.174.114)`
`19:19.04`	`TripleFFFF`	`but something not working`
`19:19.13`	`TripleFFFF`	`i press 0 and it ttrasnferes me to a wrong dialplan extencsion`
`19:19.26`	`bugz`	`anyone know of an issue with polycom phones losing the first 3 seconds of a call?`
`19:20.39`	`tzanger`	`mine don't`
`19:20.42`	`[TK]D-Fender`	`bugz : I've heard of it, but only by this one paranoid guy :)`
`19:20.44`	`harryvv`	`TK, since asterisk has a horrible time dealing with cicw I am in the hunt for a 604 did thats very affordable. We have a telus line but cicw does not pass though to the ip500 or the sipura ata.`
`19:20.56`	`tzanger`	`you aren't terminating to POTS and have echotraining set to some stupidly high value do you?`
`19:21.22`	`harryvv`	`tzanger, who are you asking`
`19:21.24`	`[TK]D-Fender`	`harryvv : Who cares about CICW when you can't even deal with CW? :)`
`19:21.27`	`bugz`	`[TK]D-Fender: tzanger: this seems to be a sipconnect related issue`
`19:21.29`	`tzanger`	`bugz: ^^`
`19:21.44`	`harryvv`	`TK, this is a side issue with the ..other person in my life`
`19:21.44`	`bugz`	`tzanger: no, this is all voip`
`19:21.49`	`TripleFFFF`	`hey.. why would 0 not work and 1 work ? they exavctly the same dialplans`
`19:21.53`	`harryvv`	`but that is another sory`
`19:21.57`	`harryvv`	`story`
`19:21.59`	`bugz`	`we are thinking of moving them to a PRI`
`19:22.04`	`[TK]D-Fender`	`tzanger : Yeah.. I hadn't thought about that factor in FOREVER.. thanks for the reminder.`
`19:22.05`	`bugz`	`im pretty sure that will solve the problem`
`19:22.25`	`sevard`	`harvvvyyyyyyyyyyyyyyyyyyyy birdmannnnnnnnnnnnnnnnnnnnnnnnnnnnnnn`
`19:22.25`	`[TK]D-Fender`	`harryvv : OH. Go VoIP 100% or get more lines...`
`19:22.34`	`tzanger`	`[TK]D-Fender: I never ever ever use echotraining`
`19:22.36`	`tzanger`	`I can't stand the pause`
`19:22.37`	`bugz`	`we suspect the phone is sending out some garbage, maybe at a really high volume, during the call setup`
`19:22.51`	`[TK]D-Fender`	`tzanger : What would I need it for? I don't GET echo ;)`
`19:22.53`	`bugz`	`and the cisco equipment on the providers end is doing the echo cancellation on it`
`19:22.55`	`TripleFFFF`	`ima d umbass`
`19:22.56`	`tzanger`	`[TK]D-Fender: :-)`
`19:23.05`	`tzanger`	`I am sofa king we todd it`
`19:23.12`	`harryvv`	`TK, yea thats the idea. Problem is we need to keep our number and so far no luck in finding anyone to port our number`
`19:23.17`	`bugz`	`effectively cutting off the first few seconds of the call`
`19:23.24`	`[TK]D-Fender`	`tzanger :D`
`19:23.45`	`TripleFFFF`	`harry what is the number ill check`
`19:23.56`	`[TK]D-Fender`	`harryvv : Guess its "good luck" then`
`19:24.06`	`[TK]D-Fender`	`harryvv : I don't know ITSP's in your area.`
`19:24.41`	`TripleFFFF`	`oh 604`
`19:24.44`	`TripleFFFF`	`aske teliax /`
`19:24.54`	`harryvv`	`thay do 604? and porting?`
`19:25.10`	`TripleFFFF`	`yeah ask for david ..`
`19:25.14`	`*** part/#asterisk The_X (i=chris@true.fiberpimp.net)`
`19:27.29`	`harryvv`	`TripleFFFF thay dont support 604`
`19:28.00`	`harryvv`	`thay dont do i.... grumble.`
`19:29.12`	`*** join/#asterisk techie (n=gus@brutus.voipops.net)`
`19:29.32`	`harryvv`	`Anyway i need to split.`
`19:30.22`	`*** join/#asterisk Damin (n=damin@nucleus.nacs.net)`
`19:31.50`	`harryvv`	`And [TK]D-Fender thanks for the help`
`19:36.47`	`*** join/#asterisk hellop (n=hellop@udp115314uds.hawaiiantel.net)`
`19:39.11`	`saftsack`	`hi is anyone here who is employed at digium?`
`19:43.32`	`mog_work`	`yes`
`19:43.38`	`_Sam--`	`er`
`19:44.20`	`lunk`	`saftsack: they will not hack the gibson for you, don't ask`
`19:44.48`	`file`	`yeah - we don't hack gibsons`
`19:44.58`	`russellb`	`i hack gibsons`
`19:45.04`	`file`	`lies!`
`19:45.09`	`Ahrimanes`	`hackers again.. hee`
`19:45.13`	`tzanger`	`hahaha`
`19:45.31`	`lunk`	`haha`
`19:45.41`	`Ahrimanes`	`man i'm sure i have that movie somewhere on dvd`
`19:45.45`	`*** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com)`
`19:45.49`	`Ahrimanes`	`damned moving boxes`
`19:46.05`	`cytrak`	`is there a way to write a for loop within the dialplan ?`
`19:46.17`	`russellb`	`cytrak: you can write a While loop :)`
`19:46.23`	`cytrak`	`ok`
`19:46.31`	`russellb`	`show application While EndWhile`
`19:46.36`	`cytrak`	`thanks`
`19:46.56`	`[TK]D-Fender`	`cytrak : GotIf !!!`
`19:46.59`	`[TK]D-Fender`	`GOTOIF*`
`19:49.07`	`cytrak`	`how do I increment a varialble though ? using Math ?`
`19:49.31`	`cytrak`	`Math application I mean`
`19:49.33`	`[TK]D-Fender`	`cytrak : Set(var=$[${var}+1])`
`19:49.52`	`cytrak`	`thanks`
`19:52.53`	`blitzrage`	`$[ ] is used for expressions... so anything like that is done within that`
`19:53.00`	`blitzrage`	`just like [TK]D-Fender says`
`19:53.45`	`Blackthorn`	`is the mp3player app/module an addon to the basic * install?`
`19:53.46`	`blitzrage`	`lol`
`19:55.10`	`CunningPike`	`Blackthorn: Yes, but you can install the correct version of mpg123 by doing a 'make mpg123' in your asterisk source folder`
`19:55.27`	`Blackthorn`	`i'm reading in the book here about exten => 123,x,mp3player(http://www.domain.com//server)`
`19:55.39`	`CunningPike`	`Blackthorn: Oh, that`
`19:55.47`	`*** part/#asterisk slobberknocker (n=ckwall@63.149.122.94)`
`19:55.50`	`Blackthorn`	`ok belive i have the right mp3 player working because the music on hold is working and playing mp3's`
`19:56.20`	`CunningPike`	`Blackthorn: Check out asterisk-addons - I believe you might need something from there`
`19:56.29`	`CunningPike`	`Gotta run - bbl`
`19:59.18`	`*** part/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com)`
`20:02.32`	`*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)`
`20:05.10`	`*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)`
`20:06.45`	`blitzrage`	`pick that up -- I'll probably end up stepping on it if you don't`
`20:06.49`	`blitzrage`	`then I'd have to kill you`
`20:09.48`	`hellop`	`awsome`
`20:09.50`	`*** part/#asterisk hellop (n=hellop@udp115314uds.hawaiiantel.net)`
`20:13.14`	`*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`20:17.01`	`cytrak`	`you guys know how on AGI we can use Get Data to play a file giving a timeout and the max number of digits that should be expected, is there a function that would do the same within the dialplan ? I'm browsing through show applications right now but don't see anything`
`20:17.44`	`file`	`cytrak: Read`
`20:18.59`	`*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)`
`20:27.53`	`PMantis`	`Man, I'm getting lots of mixed information...`
`20:28.47`	`PMantis`	`VoIP Call center phones... Polycom or SNOM? I've now heard that * can control the SNOM LCD... details?`
`20:29.23`	`*** join/#asterisk Assid (n=assid@203.115.83.214)`
`20:29.38`	`[TK]D-Fender`	`PMantis : What do you want to do with it?`
`20:30.07`	`PMantis`	`[TK]D-Fender, Like to update the LCD display with Queue information, etc.`
`20:30.32`	`PMantis`	`[TK]D-Fender, You an * developer ?`
`20:30.33`	`[TK]D-Fender`	`PMantis : Thats only viable on the IP60x series from Polycom.`
`20:30.39`	`bkw__`	`PMantis, if you read it on voip-info trust it about 25%`
`20:30.45`	`bkw__`	`PMantis, if you read it on -users don't even trust it`
`20:30.52`	`[TK]D-Fender`	`PMantis : Nope, just a user and I specialize in Polycom`
`20:30.55`	`bkw__`	`trial and error :P`
`20:31.16`	`bkw__`	`btw the 7970 with SIP is an awesome`
`20:31.21`	`bkw__`	`er awesome phone`
`20:31.32`	`[TK]D-Fender`	`ok, gtg, later all`
`20:31.35`	`tzanger`	`bkw__: yeah?`
`20:31.40`	`bkw__`	`tzanger, yeppers`
`20:31.43`	`PMantis`	`[TK]D-Fender, Oh? We were planning to implement the IP-301's for agents`
`20:31.46`	`PMantis`	`Ahhh`
`20:31.50`	`tzanger`	`have you used ohter wifi sip phones?`
`20:31.57`	`tzanger`	`I've used some cheap models, and they're lacking`
`20:32.03`	`bkw__`	`7970 isn't wifi.. the 7920 is`
`20:32.11`	`bkw__`	`I should get a 7920 also`
`20:32.12`	`tzanger`	`ohh`
`20:33.20`	`bkw__`	`7970 is the color display touch screen bad ass phone`
`20:33.26`	`*** join/#asterisk CoaxD (i=coax@shell1.cornernet.com)`
`20:33.38`	`bkw__`	`OMG its CoaxD`
`20:33.40`	`bkw__`	`ltns`
`20:33.42`	`CoaxD`	`indeed :)`
`20:33.44`	`CoaxD`	`hiya.. :)`
`20:33.48`	`bkw__`	`CoaxD, you coming ot cluecon this year?`
`20:33.57`	`CoaxD`	`bkw; Hah. Yeah, like i have time`
`20:34.11`	`CoaxD`	`bkw: I'm too busy supporting a gazillion acronyms in linux+oracle these days`
`20:34.15`	`Ariel_`	`7970 is a very nice color phone and finally it has Sip for it's firmware.`
`20:34.45`	`CoaxD`	`bkw: Know, off the top of your head, if any bugs exist in 1.2.7.1 that might cause messages to become un-deletable in voicemail queues?`
`20:35.06`	`Corydon-w`	`CoaxD: fixed in the latest 1.2 tree`
`20:35.07`	`bkw__`	`CoaxD, could be`
`20:35.15`	`bkw__`	`Ariel_, yes it is`
`20:35.30`	`CoaxD`	`Corydon: Got a patch against 1.2.7.1? Dont wanna go beta..`
`20:35.39`	`Corydon-w`	`1.2 isn't beta`
`20:35.45`	`Corydon-w`	`it's just fixes`
`20:36.05`	`Ariel_`	`there up to 1.2.8 are they not.`
`20:36.21`	`CoaxD`	`ariel: 1.2.7.1 is the last release i think`
`20:36.23`	`*** join/#asterisk Dovid (n=none@barak.cellcom.co.il)`
`20:36.25`	`Corydon-w`	`Note that that's different than SVN trunk. Trunk is about up to beta`
`20:36.43`	`CoaxD`	`hmmm. ok`
`20:37.27`	`*** join/#asterisk KranZ (n=user@sme.bestline.net)`
`20:37.36`	`KranZ`	`poop`
`20:37.43`	`KranZ`	`is chan_phone.so manditory?`
`20:38.30`	`Ariel_`	`CoaxD, looks like your correct. I am on drugs I guess.`
`20:38.51`	`CoaxD`	`ariel: Its okay. Drugs are sometimes good.`
`20:38.56`	`*** join/#asterisk low_rad (n=hibbert@h66-38-194-130.gtconnect.net)`
`20:38.57`	`Dovid`	`Can anyone help me`
`20:39.04`	`Dovid`	`I have a problem with real time`
`20:39.06`	`low_rad`	`hey I need some help too...`
`20:39.20`	`low_rad`	`a question really`
`20:39.23`	`Dovid`	`If I use static all works well`
`20:39.53`	`Dovid`	`But with real time the I and t option wont work`
`20:40.24`	`low_rad`	`I have a TDM400 and it was working last week`
`20:40.27`	`Dovid`	`??`
`20:40.47`	`low_rad`	`over the past weekend there was a power surge and the server was frozen`
`20:41.09`	`low_rad`	`so I rebooted it and now I can't make incoming/outgoing calls`
`20:41.32`	`low_rad`	`plus when other people connect to the server they get a busy signal`
`20:41.33`	`Ariel_`	`low_rad, you might have gotten a power surge via the phone lines? if that is the case then the board is bad.`
`20:41.43`	`*** join/#asterisk unmanaged (n=unmanage@64.89.118.139)`
`20:41.50`	`low_rad`	`Ariel_: none of the lights at the back of the card are on`
`20:41.59`	`Ariel_`	`fried`
`20:42.00`	`low_rad`	`bad card or modules?`
`20:42.08`	`CoaxD`	`low_rad: Probably both.`
`20:42.09`	`low_rad`	`can I still use the modules?`
`20:42.14`	`obiwanmikenolte`	`low_rad: is Asterisk started?`
`20:42.14`	`low_rad`	`oh damn!`
`20:42.22`	`CoaxD`	`low_rad: Lightning Does That[tm]`
`20:42.32`	`Ariel_`	`low_rad, the actually question is should you use it?`
`20:42.37`	`low_rad`	`obiwanmikenolte: yep, and no error messages in debian`
`20:42.52`	`obiwanmikenolte`	`Did you modprobe wctdm?`
`20:42.54`	`low_rad`	`obiwanmikenolte: debian sees all of the modules fine`
`20:42.55`	`Dovid`	`Anyone can help me with real time ?`
`20:42.58`	`CoaxD`	`low_rad: when you load the modules, does it actually work?`
`20:43.09`	`CoaxD`	`low_rad: did ztcfg -vv report anthying?`
`20:43.17`	`CoaxD`	`low_rad: does zttool fail or show reds?`
`20:43.27`	`low_rad`	`CoaxD: working according to the logs and ps`
`20:43.38`	`low_rad`	`1 sec... I'll check...`
`20:43.41`	`CoaxD`	`low_rad: logs and ps dont tell you bs`
`20:44.01`	`obiwanmikenolte`	`Catchy`
`20:45.17`	`*** join/#asterisk SplasPood (n=jwb@206.252.198.101)`
`20:45.20`	`*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)`
`20:45.23`	`Ariel_`	`Dovid, maybe someone might be able to help you later. But don't keep asking every minute. (I don't use realtime)`
`20:45.28`	`Dovid`	`ok`
`20:45.40`	`low_rad`	`CoaxD: will ztcfg work?`
`20:45.45`	`charles___`	`Anyone into Asterisk at Large ?`
`20:45.55`	`Ariel_`	`into asterisk at large`
`20:46.24`	`unmanaged`	`Asterisk at Large ?`
`20:46.36`	`low_rad`	`ztcfg reports 4 channels configured`
`20:47.10`	`CoaxD`	`low_rad: Good.`
`20:47.14`	`CoaxD`	`low_rad: now test them with zttool`
`20:47.32`	`low_rad`	`I don't think I have that compiled...`
`20:47.33`	`charles___`	`zoa: hey man`
`20:47.34`	`low_rad`	`:/`
`20:47.40`	`charles___`	`zoa: did you got your quad opteron ?`
`20:47.41`	`CoaxD`	`low_rad: You do if you installed zaptel`
`20:47.58`	`low_rad`	`default location? /usr/local/sbin?`
`20:48.05`	`CoaxD`	`uh`
`20:48.13`	`CoaxD`	`voip:~/asterisk# which zttool`
`20:48.17`	`CoaxD`	`<PROTECTED>`
`20:48.35`	`low_rad`	`not found`
`20:48.47`	`low_rad`	`i'm still using asterisk 1.x`
`20:48.55`	`CoaxD`	`so am I`
`20:49.06`	`CoaxD`	`not that it matters; zttool has been around a long while`
`20:49.15`	`CoaxD`	`zttool doesnt come with asterisk. it comes with zaptel`
`20:49.55`	`low_rad`	`hmm I'll check again...`
`20:50.14`	`CoaxD`	`low_rad: if it aint there, it aint there. might need to recompile zaptel of the same version`
`20:50.21`	`CoaxD`	`low_rad: To support zttool`
`20:50.36`	`CoaxD`	`(I forget if its an option or whatnot.) If this is a debian package, though.. God only knows if they apckaged it`
`20:51.33`	`low_rad`	`CoaxD: do I need root permissions? because the admin took a vacation and didn't tell me :/`
`20:51.43`	`CoaxD`	`Uh. yes.`
`20:51.54`	`CoaxD`	`unless of course your zaptel devices are owned by someone else`
`20:52.18`	`CoaxD`	`but if you arent root, you arent fixing this if its filesystem related`
`20:52.39`	`low_rad`	`well I'm in the same group (adm) just no root password`
`20:52.48`	`CoaxD`	`low_rad: And if you dont know how to gain root privs if your admin aint there... well, you're worse off than i thought`
`20:53.32`	`low_rad`	`CoaxD: :), seems I need a newt library to compile zttool ?`
`20:53.40`	`*** join/#asterisk MGSsancho (n=user@adsl-67-126-140-26.dsl.irvnca.pacbell.net)`
`20:53.44`	`CoaxD`	`that might well be the prob`
`20:54.11`	`CoaxD`	`low_rad: need root to install it globally, but might be able to compile it via a locally installed library by modding up the makefiles`
`20:54.38`	`Dr-Linux`	`low_rad: what's your login ID , 0:0?`
`20:55.00`	`*** join/#asterisk dsfr_ (n=dsfr@pdpc/sponsor/digium/dsfr)`
`20:55.02`	`Dr-Linux`	`low_rad: type >> id <user>`
`20:55.11`	`low_rad`	`CoaxD: hmm`
`20:55.34`	`low_rad`	`Dr-Linux: what will that prove?`
`20:56.14`	`CoaxD`	`'id' will tell you the id of who you are currently logged in as.`
`20:56.17`	`Dr-Linux`	`low_rad: root should have 0 ID`
`20:56.21`	`*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)`
`20:56.28`	`CoaxD`	`evidently, Dr-Linux missed the fact that you arent logged in as root`
`20:56.46`	`low_rad`	`:D`
`20:56.58`	`Dr-Linux`	`nope`
`20:57.05`	`low_rad`	`ok, well I guess I'll have to get in contact with the admin somehow`
`20:57.05`	`Dr-Linux`	`i didn't mean that`
`20:57.18`	`low_rad`	`thanks for all of your help`
`20:57.24`	`*** join/#asterisk epoch (n=epoch@octane.breakbeats.org)`
`20:57.29`	`Dr-Linux`	`low_rad: just type this command`
`20:57.32`	`Dr-Linux`	`:`
`20:57.42`	`Dr-Linux`	`id <your user>`
`20:58.07`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`20:58.21`	`low_rad`	`id keith`
`20:58.41`	`Dr-Linux`	`low_rad: i mean, at your shell`
`20:58.49`	`low_rad`	`:)`
`20:59.11`	`low_rad`	`groups=1000(keith),4(adm),29(audio),61(asterisk)`
`20:59.23`	`Dovid`	`Hope its not too soon that I am asking but any people here that know real time ?`
`20:59.31`	`tzafrir_laptop`	`I'm helping a guy install a tor2-compatible card . It seems to have a PCI product ID of 4000 . Anybody familiar with those?`
`21:00.40`	`TripleFFFF`	`darn`
`21:00.45`	`Dr-Linux`	`low_rad: so you don't have full privilages`
`21:01.01`	`TripleFFFF`	`Dovid me`
`21:01.02`	`TripleFFFF`	`lol`
`21:01.24`	`TripleFFFF`	`i live in realtime but i archive my memory with tequila every saturday`
`21:01.24`	`Dovid`	`TripleFF: u use real time ?`
`21:01.27`	`TripleFFFF`	`yes`
`21:01.32`	`Dovid`	`haha`
`21:01.36`	`Dovid`	`Can I pm u ?`
`21:01.43`	`TripleFFFF`	`sur`
`21:01.45`	`TripleFFFF`	`e`
`21:02.00`	`low_rad`	`bye`
`21:02.56`	`*** join/#asterisk noky (n=noky@200.69.211.18)`
`21:02.58`	`noky`	`hi buddies`
`21:03.21`	`noky`	`i want know if asterisk use threads or is a only gigant process? :D`
`21:05.36`	`noky`	`??`
`21:07.28`	`noky`	`because`
`21:07.34`	`noky`	`when i do a: ps -fea`
`21:07.40`	`noky`	`i see a only process for asterisk...`
`21:07.49`	`noky`	`but in the source code i have for example: loader.c: ast_mutex_unlock(&modlock);`
`21:07.54`	`noky`	`mutex?!`
`21:08.00`	`noky`	`use threads?`
`21:08.32`	`TripleFFFF`	`!jbot seen bkw ?`
`21:08.38`	`TripleFFFF`	`~seen`
`21:08.42`	`TripleFFFF`	`~seen bkw`
`21:08.45`	`jbot`	`bkw <n=bkw@k7j231-2.kam.afb.lu.se> was last seen on IRC in channel #debian, 146d 9h 9m 54s ago, saying: 'Anyone who can explain why a nic sometimes become eth0, others eth1. This really confuse dhclient during bootups.'.`
`21:09.21`	`KranZ`	`noky: #asterisk-dev`
`21:09.50`	`C4T3l`	`~seen C4T3l`
`21:09.52`	`jbot`	`c4t3l is currently on #asterisk (7h 21m 47s). Has said a total of 2 messages. Is idling for 2s, last said: '~seen C4T3l'.`
`21:10.03`	`*** part/#asterisk jbailey (n=jbailey@modemcable139.249-203-24.mc.videotron.ca)`
`21:10.11`	`TripleFFFF`	`~seen linksys`
`21:10.14`	`jbot`	`i haven't seen 'linksys', TripleFFFF`
`21:10.30`	`TripleFFFF`	`~seen the size of it`
`21:10.32`	`jbot`	`i haven't seen 'the size of it', TripleFFFF`
`21:10.39`	`noky`	`ok`
`21:13.07`	`bugz`	`anyone know of an issue with polycom phones losing the first 3 seconds of a call?`
`21:13.55`	`Dr-Linux`	`anyone is using AT&T internet?`
`21:14.48`	`*** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70)`
`21:15.09`	`Dovid`	`nope`
`21:15.16`	`Dovid`	`I am connected via gprs :(`
`21:16.50`	`*** join/#asterisk schuylerdigium (n=schuyler@gateway.digium.com)`
`21:16.52`	`CrashHD`	`when running multiple asterisk instances on the same machine (with a digium board installed in the machine) what precautions should I take to make sure both instances are not trying to utilize the board?`
`21:17.22`	`*** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net)`
`21:17.37`	`ceeto`	`Hi all. If I'm using rxfax(${faxnum}\|debug) where does the "debug" info go?`
`21:17.39`	`TripleFFFF`	`oh god`
`21:17.47`	`TripleFFFF`	`goes to consol maybe`
`21:17.55`	`TripleFFFF`	`show application debug`
`21:17.58`	`TripleFFFF`	`i mean txfax`
`21:18.29`	`*** join/#asterisk MoutaPT (n=MoutaPT@85.139.196.147)`
`21:18.49`	`ceeto`	`That's it. Thanks.`
`21:19.02`	`TripleFFFF`	`<PROTECTED>`
`21:19.03`	`MoutaPT`	`hi doesn any one here with experience with BRI card and Asterisk?`
`21:19.09`	`TripleFFFF`	`there`
`21:19.22`	`Dovid`	`Dont sorry`
`21:19.32`	`Dovid`	`We dont use ISDn in the US :(`
`21:19.37`	`TripleFFFF`	`<PROTECTED>`
`21:19.37`	`TripleFFFF`	`<PROTECTED>`
`21:19.37`	`TripleFFFF`	`<PROTECTED>`
`21:20.10`	`TripleFFFF`	`i assume it overides the basic asterisk startup and adds a verbose flag to it`
`21:20.15`	`MoutaPT`	`thks Dovid, unfortunately one client with BRI :(`
`21:20.30`	`Dovid`	`U can get isdn here just real hard`
`21:20.44`	`TripleFFFF`	`CrashHD .. precaution #1.. not use 2 process on same board ;)`
`21:21.01`	`MoutaPT`	`i've been informed from Beronet that i need kernel 2.6.12 ...`
`21:21.23`	`MoutaPT`	`i'm trying to make it in Debian... any distro with this kernel already?`
`21:21.34`	`ceeto`	`show application debug didn't work..`
`21:21.38`	`ceeto`	`did I miss something?`
`21:21.41`	`ceeto`	`Thanks for the help, btw.`
`21:22.03`	`sevard`	`CrashHD: Not that I know your answer but as a careful bystander I have to wonder to myself why somebody would do something that insane`
`21:22.50`	`Dr-Linux`	`MoutaPT: almost all distro's`
`21:23.11`	`MoutaPT`	`centos doesn't and debian neither, or am I wrong?`
`21:23.19`	`*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)`
`21:23.41`	`tzafrir_laptop`	`MoutaPT, need kernel 2.6.12 for Debian Sarge?`
`21:23.48`	`saftsack`	`MoutaPT, i have experiences`
`21:23.51`	`MoutaPT`	`yes`
`21:24.18`	`tzafrir_laptop`	`deb http://updates.xorcom.com/rapid stable main`
`21:24.38`	`MoutaPT`	`Debian Sarge is the same of just Debian? or something else? dummie question`
`21:24.42`	`tzafrir_laptop`	`Or browse that that URL to grab the kernel packages`
`21:24.55`	`tzafrir_laptop`	`Let me know if anything is missing`
`21:25.20`	`MoutaPT`	`tzafrir_laptop: installing debian from netinst first`
`21:25.34`	`MoutaPT`	`<tzafrir_laptop> asap i will give u some feedback`
`21:26.28`	`tzafrir_laptop`	`MoutaPT, also consider the ISOs from http://rapid.tzafrir.org.il/iso/`
`21:27.04`	`MoutaPT`	`asterisk ready?`
`21:27.08`	`*** join/#asterisk Sammich (n=brian@elk-en0.intercom.net)`
`21:28.29`	`SplasPood`	`MoutaPT: Debian sarge is the stable version of debian`
`21:34.39`	`*** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com)`
`21:35.22`	`vooduhal`	`Hello all. I've remapped the physical "Transfer" button on the Polycom 601 to be '#' for asterisk transfer, but does anyone know how to disable the damned softkey transfer button?`
`21:37.39`	`[TK]D-Fender`	`vooduhal : WHY on earth would you want to remap SIP hard keys like that?`
`21:38.49`	`vooduhal`	`Because QueueMetrics can't register a SIP redirect instead of asterisk transfering or we can't figure out a way to do it. Management is bitching that the reports don't show the transfers properly. But if they press # to transfer it is registered properly.`
`21:39.16`	`vooduhal`	`So the easiest solution is to force the polycom to play with asterisk.`
`21:39.33`	`vooduhal`	`But we still have that obnoxious softkey.`
`21:40.12`	`vooduhal`	`Quite simple to to reprogram the physical button.`
`21:40.25`	`[TK]D-Fender`	`AH.... freakish....`
`21:40.32`	`vooduhal`	`We also had a great time with one of our VPs on 4/1. :)`
`21:40.41`	`[TK]D-Fender`	`just about the ONLY slightly valid reason I can imagine :)`
`21:40.48`	`vooduhal`	`Thank you. :)`
`21:41.13`	`[TK]D-Fender`	`You're welcome...`
`21:43.23`	`*** join/#asterisk _Paulo_ (n=Paulo@c9064c64.virtua.com.br)`
`21:43.41`	`_Paulo_`	`I have some strange problem...`
`21:43.55`	`x86`	`we might have a strange solution`
`21:44.04`	`[TK]D-Fender`	`strangely appropriate ;)`
`21:44.17`	`vooduhal`	`Ah, and if anyone needs to know. In the 1.6.1 Admin guide from polycom, they finally including button mappings. We had to harrass one of their engineers to get them out of them (Ok, all we had to do was ask) but now they've included it in the guide.`
`21:44.20`	`[TK]D-Fender`	`People come out in the rain!`
`21:44.53`	`x86`	`teh mud peoples`
`21:44.58`	`[TK]D-Fender`	`vooduhal : Since we're at 1.6.6 official, and 20.beta going gold momentairly, there's more good stuff to come.`
`21:45.02`	`_Paulo_`	`when I call my boss gs286 from PSTN it works`
`21:45.29`	`[TK]D-Fender`	`vooduhal : I stronly hope that they yank the web interface out completely and leave room for more real functionality :D`
`21:45.30`	`x86`	`_Paulo_: but he can not dial out?`
`21:45.42`	`vooduhal`	`[TK]D-Fender, Agreed.`
`21:46.08`	`_Paulo_`	`when I call from a DID in USA, and canreinvite=yes, I lost 1 leg...`
`21:46.16`	`vooduhal`	`[TK]D-Fender, didn't realize they were up to 1.6.6. Our sales rep told us 1.6.1 was the current.`
`21:46.23`	`_Paulo_`	`when I call from a DID in USA, and canreinvite=no, it works...`
`21:46.30`	`[TK]D-Fender`	`vooduhal : New rep time!`
`21:46.30`	`vooduhal`	`[TK]D-Fender, got a changelog by any chance?`
`21:46.36`	`vooduhal`	`[TK]D-Fender, Agreed.`
`21:46.46`	`[TK]D-Fender`	`vooduhal : I'm running 2.0 Beta at home personally.`
`21:46.57`	`vooduhal`	`[TK]D-Fender, I think we use VoIP supply.`
`21:47.07`	`[TK]D-Fender`	`vooduhal : go here for it : http://www.polycom.com/products_services/0,1443,pw-34-182-15672,00.html`
`21:47.13`	`_Paulo_`	`x86, when he dials everything works fine.`
`21:47.22`	`[TK]D-Fender`	`vooduhal : They're supposed to be better than that...`
`21:47.31`	`_Paulo_`	`I got some DIDs from netcyber`
`21:47.32`	`ceeto`	`Hi all. How would I specify a specific ZAP channel to always receive faxes through rxfax?`
`21:47.56`	`[TK]D-Fender`	`ceeto : Set their incoming context accordingly to one that does only that`
`21:48.02`	`vooduhal`	`[TK]D-Fender, Sweet.`
`21:48.08`	`Dr-Linux`	`does cisco phone work with its PPPoE connection?`
`21:48.12`	`[TK]D-Fender`	`vooduhal : Oh, and a new model to oggle :)`
`21:48.32`	`[TK]D-Fender`	`vooduhal : I'm going to get one ASAP.`
`21:48.32`	`vooduhal`	`[TK]D-Fender, We just bought a shit ton of 601s to replace about 50 phones with.`
`21:48.37`	`[TK]D-Fender`	`!`
`21:48.44`	`[TK]D-Fender`	`overkill for most...`
`21:48.54`	`_Paulo_`	`when his wife call him from the netcyber DID, and canreinvite=yes, she hears well but he cant.`
`21:48.54`	`ceeto`	`Can you give me an example? I'm a noob.`
`21:49.09`	`[TK]D-Fender`	`ceeto : Just change the context in zapata for that channel!`
`21:49.28`	`vooduhal`	`They just dumped some Linksys deskphones on me to test for a 7 office IP centrex design we may do for a realty company. Any experience with them?`
`21:49.55`	`[TK]D-Fender`	`vooduhal : Yeah... they work, and I'd say they're nice, but not comperable.`
`21:50.06`	`[TK]D-Fender`	`vooduhal : I owned a 941.`
`21:50.13`	`_Paulo_`	`is there something else that afects canreinvite=yes ?`
`21:50.21`	`[TK]D-Fender`	`vooduhal : had it less than 2 months and sold it off.`
`21:50.32`	`vooduhal`	`Luckily they don't need much in the way of functionality. Just trying to avoid toll charges in their 7 locations all long distance.`
`21:50.33`	`ceeto`	`How do I specify a specific context for a specific zap channel in zapata.conf?`
`21:50.36`	`ceeto`	`(Thanks for the help)`
`21:50.48`	`_Paulo_`	`x86, any guess?`
`21:50.50`	`vooduhal`	`These are SPA942s.`
`21:51.06`	`[TK]D-Fender`	`vooduhal : Same shit w/ backlight & PoE.`
`21:51.22`	`[TK]D-Fender`	`vooduhal : Far too inflexible for my tastes.`
`21:51.43`	`vooduhal`	`Haven't played with them yet, but we needed PoE and that was pretty much it.`
`21:52.00`	`vooduhal`	`Well, I was wrong. We're running 1.6.5 on these things.`
`21:52.46`	`[TK]D-Fender`	`vooduhal : The real advantage of 1.6.6 is the effective removal of the buddy watch limitation on the IP 601 for use of sidecars.`
`21:53.00`	`vooduhal`	`Ah...`
`21:53.06`	`vooduhal`	`We've actually just went with FOP.`
`21:53.20`	`[TK]D-Fender`	`vooduhal : Not any more! Sidecar is uber nice now.`
`21:53.26`	`hads`	`Cool, didn't realise they'd removed that limit.`
`21:53.37`	`ceeto`	`How do I specify a specific context for a specific zap channel in zapata.conf?`
`21:53.45`	`vooduhal`	`May have to give it another shot. That's what they wanted originally but it didn't test out so well and we sold them on FOP.`
`21:53.58`	`[TK]D-Fender`	`When * 1.4 adds SIP-B and Shared Line support they;ll be Godly...`
`21:54.16`	`CunningPike`	`ceeto: Read the examples - everything above a channels => statement affects those channels`
`21:54.20`	`vooduhal`	`Plus, we use agents instead of phones so I'm not sure how that will affect the side car now.`
`21:54.25`	`vooduhal`	`We have all roaming users.`
`21:54.39`	`vooduhal`	`And FOP supports Agent status and not just line status.`
`21:54.39`	`[TK]D-Fender`	`vooduhal : :/`
`21:54.49`	`vooduhal`	`MWI was a bitch.`
`21:55.02`	`vooduhal`	`Thank god the polycoms don't seem to care where the MWI on and off packets come from.`
`21:55.04`	`[TK]D-Fender`	`vooduhal : I don't know if there is any practical way to manage that scenario...`
`21:55.20`	`[TK]D-Fender`	`Oh GOD... MWI with mobile users? EEK`
`21:55.26`	`vooduhal`	`I've got it working well. :)`
`21:55.36`	`[TK]D-Fender`	`What'd you do for it?`
`21:56.15`	`vooduhal`	`Part of the voicemail extern notification just calls a perl script to locate which phone an agent is logged on to and turns it on and off as needed. When the user logs in, it checks if they have messages and lights it and turns it off when they log off.`
`21:56.20`	`vooduhal`	`Thank god for ngrep.`
`21:56.31`	`vooduhal`	`Literally just dumped a MWI message and resend it as needed.`
`21:56.41`	`Dr-Linux`	`anybody answer my question?`
`21:56.44`	`vooduhal`	`It's scary how the Polycom will just do what its told and doesn't seem to check anything.`
`21:56.44`	`Dr-Linux`	`does cisco phone work with its PPPoE connection?`
`21:56.57`	`vooduhal`	`Dr-Linux which model? I've got one sitting right here.`
`21:57.22`	`Dr-Linux`	`vooduhal: 7940`
`21:57.51`	`vooduhal`	`I modified the code for extern notification that I wrote for SMDI to a coppercom we have.`
`21:58.03`	`vooduhal`	`Dr-Linux, one sec.`
`21:58.03`	`Dr-Linux`	`vooduhal: we have many Cisco 7940/60's all are fine`
`21:58.12`	`*** join/#asterisk Qwell (n=north@unaffiliated/qwell)`
`21:58.15`	`[TK]D-Fender`	`vooduhal : You mean yo do a RAW packet transmission manually?`
`21:58.24`	`Dr-Linux`	`but this one client has some odd connection, i never understand`
`21:58.47`	`[TK]D-Fender`	`vooduhal : Perhaps it doesn't consider MWI a threat ;)`
`21:59.06`	`Dr-Linux`	`vooduhal: it's something like Phone line >>> Modem >>> Hub >>> PC's (with dynamic IPs)`
`21:59.08`	`vooduhal`	`Well, since its just UDP. :)`
`21:59.22`	`vooduhal`	`Dr-Linux, running to the wiring closet to test the cisco.`
`22:00.18`	`[TK]D-Fender`	`vooduhal : PM.`
`22:00.36`	`[TK]D-Fender`	`Ok, I've got to get moving ; class awaits.`
`22:00.39`	`Dr-Linux`	`:S`
`22:02.01`	`[TK]D-Fender`	`Back in a few hours. Later all...`
`22:02.29`	`vooduhal`	`Ok, 7940G does not do PoE.`
`22:02.50`	`carrar`	`does if you wire it right`
`22:03.42`	`vooduhal`	`Lmao. I don't consider using 2 of the 8 pins in Cat5 wired right. :(`
`22:04.06`	`carrar`	`swap 4 with 7, and 5 with 8`
`22:04.41`	`vooduhal`	`I prefer 48 port PoE production injector. If it turns on, it works, if not, it doesn't. :)`
`22:04.43`	`*** join/#asterisk Bishoy (n=CodeGuru@62.139.87.122)`
`22:04.59`	`carrar`	`cisco 6500 blades will do both`
`22:05.33`	`vooduhal`	`carrar, Or are you saying Ciscos are using proprietery wiring for PoE?`
`22:06.17`	`Bishoy`	`Gentlemen, im in trouble with asterisk with a critical error @ customer's site and a little hand here`
`22:06.36`	`vooduhal`	`Shoot.`
`22:06.48`	`Dr-Linux`	`vooduhal: i mean PPPoE internet connection from AT&T.`
`22:07.04`	`Bishoy`	`ok, here is my problem`
`22:07.48`	`Bishoy`	`i have a Mega server with asterisk@home installed v2.7 with 2 x TDM400P (4 FXS)`
`22:08.03`	`KranZ`	`woot, mega!`
`22:08.29`	`vooduhal`	`There is your problem. You are using digium. :)`
`22:08.30`	`vooduhal`	`j/k`
`22:08.33`	`vooduhal`	`Kind of. :)`
`22:08.33`	`Bishoy`	`the calls comes through the TDM card to an IVR and queued for the support agents (8 SIP agents using xlite)`
`22:08.40`	`ceeto`	`Thanks all, got it! :)`
`22:08.52`	`KranZ`	`vooduhal: the cards are a bit fickle yes`
`22:09.17`	`Bishoy`	`after 3-4 hours of normal operation everything just stops and calls just stop comming and all the lines become busy`
`22:09.19`	`MoutaPT`	`sangoma are the best cards currently?`
`22:09.27`	`vooduhal`	`KranZ, fickle is definitely a word for it. I love the digium people that I've met, but have fallen in love with Sangoma equipment.`
`22:09.53`	`Dr-Linux`	`vooduhal: did you mean Cisco 7940 phone doesn't work with PPPoE internet connection?`
`22:10.16`	`Bishoy`	`any clue where to start nailing the problem down ?`
`22:10.34`	`vooduhal`	`Dr-Linux, I meant PoE. If you are talking about PPPoE, are you trying to hook your phone up to the modem itself or do you at least have a dialing router?`
`22:11.02`	`vooduhal`	`Bishoy, what revision of the TDM400P cards are you using?`
`22:11.04`	`*** join/#asterisk JASON99 (n=jason@jason.unitz.ca)`
`22:11.19`	`Bishoy`	`i dont know, we bought them a month ago`
`22:11.41`	`Dr-Linux`	`vooduhal: i have dialing router, but the phone can't grab an IP address from the DHCP, either if i put the same cable in the a PC, it works`
`22:12.09`	`*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)`
`22:12.54`	`vooduhal`	`Dr-Linux, Have you configured the phone factory defaults or can you at least guarantee that it's not doing 802.1q?`
`22:12.58`	`JASON99`	`is it normal to get the following warnings?`
`22:12.58`	`JASON99`	`May 30 18:12:34 WARNING[4525]: chan_zap.c:10879 setup_zap: Ignoring switchtype`
`22:12.58`	`JASON99`	`May 30 18:12:34 WARNING[4525]: chan_zap.c:10879 setup_zap: Ignoring signalling`
`22:13.28`	`vooduhal`	`Stupid question, can Cisco phones support ISL? Never tried, or would I ever, just a funny thought. :)`
`22:13.43`	`*** join/#asterisk bjohnson (n=bjohnson@i216-58-51-69.cybersurf.com)`
`22:14.51`	`Dr-Linux`	`vooduhal: this phone was already working at other location with 7.4 SIP firmware, so as i moved to this client, it doesn't get any IP from PPPoE DHCP`
`22:15.20`	`Dr-Linux`	`JASON99: yeah don't worry about that`
`22:15.26`	`vooduhal`	`Dr-Linux, is it plugged into a multiport modem or into an actual router with built in switch?`
`22:16.15`	`Dr-Linux`	`vooduhal: it's somethign like this Phone line >>> Modem >>> Hub >>>> PC's and 7940 phone`
`22:16.25`	`vooduhal`	`That won't work.`
`22:16.38`	`Dr-Linux`	`vooduhal: why?`
`22:16.59`	`Dr-Linux`	`vooduhal: but why it works when i plugg same ethernet cable in a PC ?`
`22:17.31`	`vooduhal`	`If you are plugging a hub to a DSL modem port you still need something to dial the PPPoE connection. Unless by Hub you mean router. .`
`22:18.16`	`Dr-Linux`	`vooduhal: yes, but how it works when i plugg the same cable into a PC?`
`22:18.50`	`vooduhal`	`Are you automatically getting an IP when you plug the PC in or are you having to dial a PPPoE connection?`
`22:19.00`	`Dr-Linux`	`vooduhal: after hub there are 4 PC's all can get an IP from DHCP, but if i plugg the same cable in to the phone, then phone looks for IP but can't grab one`
`22:19.35`	`vooduhal`	`Are these PCs getting private IPs or public?`
`22:19.46`	`Dr-Linux`	`vooduhal: automatically get a Public IP, and each PC get different IP on each reboot`
`22:19.52`	`vooduhal`	`Trying to figure out if you mean, router/switch instead of hub.`
`22:20.00`	`Dr-Linux`	`vooduhal: Public IP's`
`22:20.36`	`Dr-Linux`	`vooduhal: i'm in Pakistan and the client is in USA, so she told me she is using a Hub after modem`
`22:20.48`	`vooduhal`	`Hmmm..`
`22:21.00`	`vooduhal`	`Can you get the model of the hub?`
`22:23.17`	`*** join/#asterisk adker (n=adker@74-33-201-18.br1.glv.ny.frontiernet.net)`
`22:23.19`	`Dr-Linux`	`vooduhal: she is away right now :S :(`
`22:23.51`	`Dr-Linux`	`vooduhal: it's not a dialing modem i think, she is using DSL`
`22:24.10`	`Dr-Linux`	`and Its PPPoE (PPP over ethernet)`
`22:24.31`	`*** join/#asterisk Smi\|k (n=smilk@netblock-72-25-103-165.dslextreme.com)`
`22:25.23`	`vooduhal`	`And you are sure she is using PPPoE?`
`22:25.45`	`Dr-Linux`	`vooduhal: yes, bcoz i talked to the her provider AT&T`
`22:25.46`	`vooduhal`	`I work for an ISP that does DSL but we choose not to do PPPoE in favor of MAC registration.`
`22:26.29`	`Dr-Linux`	`vooduhal: my question is that, if CIsco phone supports PPPoE connection or not?`
`22:26.43`	`JASON99`	`Dr-Linux: Thanks`
`22:27.01`	`file`	`why would you put a PPPoE implementation in a VoIP phone?`
`22:27.31`	`Dr-Linux`	`file: i don't, one of our client has this type of damn connection`
`22:27.42`	`vooduhal`	`1. I doubt it, but 2. If these PCs are not having to dial a virtual connection as they would with PPPoE, I'm not sure they are using PPPoE and without knowing if the "hub" is actually a hub or a router I'm stuck.`
`22:27.46`	`Dr-Linux`	`and phone doesn't grab an IP`
`22:28.08`	`_Paulo_`	`Dr-Linux, there are many adsl routers out there`
`22:28.18`	`_Paulo_`	`Dr-Linux, they are inexpensive.`
`22:29.30`	`*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)`
`22:29.33`	`Dr-Linux`	`_Paulo_: yes, i know but i should have an answer to her :) if the phone works with PPPoE connection or not`
`22:29.34`	`vooduhal`	`And if it's a polycom and that damned built in browser, much fun can be had. :)`
`22:29.48`	`Dr-Linux`	`and we are stuck in this problem since last week`
`22:30.05`	`vooduhal`	`Dr-Linux, if the question is will the phone work connected to a PPPoE network via a DSL modem, I would say no.`
`22:30.22`	`file`	`you shouldn't... just... UGH`
`22:30.23`	`_Paulo_`	`What cisco is that?`
`22:30.24`	`vooduhal`	`But if you can get the details of the equipment I can probably help you.`
`22:30.32`	`vooduhal`	`7940.`
`22:31.19`	`_Paulo_`	`I have some cisco atas and none have pppoe.`
`22:31.21`	`*** join/#asterisk opus_ (n=opus@68.216.187.60)`
`22:31.28`	`*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)`
`22:31.33`	`opus_`	`if I have multipele T1s and I want to Dial out on the first available one, how do i do it?`
`22:31.55`	`opus_`	`Dial(Zap/g1/%{EXTEN}&Zap/g2/${EXTEN})`
`22:31.56`	`_Paulo_`	`opus_, put them in the same group`
`22:32.04`	`opus_`	`oh ok g1`
`22:32.38`	`Dr-Linux`	`Vorondil: i'm trying to call her USA`
`22:33.54`	`TripleFFFF`	`hey`
`22:33.58`	`TripleFFFF`	`anyone can help me out ?`
`22:34.16`	`vooduhal`	`Dr-Linux, one other thing to look for while you wait is to make sure you aren't tagging the traffic from the phone.`
`22:34.24`	`Dr-Linux`	`vooduhal: she said, it's a HUB and she is sure.`
`22:34.29`	`*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)`
`22:34.35`	`TripleFFFF`	`the directory app is looking for NAME field but i got FULLNAME field.. i dont want to rechange my crap.. anyway to Alias a field name in mysql ?`
`22:34.38`	`vooduhal`	`Can you get a model of the hub and the modem?`
`22:35.18`	`opus_`	`do I just put the same "group = 1" by each channel = block?`
`22:35.29`	`Dr-Linux`	`vooduhal: i asked her`
`22:35.31`	`TripleFFFF`	`im on 1.2.7.1`
`22:35.32`	`TripleFFFF`	`<PROTECTED>`
`22:35.37`	`TripleFFFF`	`even says it should work`
`22:35.48`	`Dr-Linux`	`vooduhal: it's not dialing modem,`
`22:36.37`	`_Paulo_`	`opus_, I would try that.`
`22:36.48`	`vooduhal`	`Dr-Linux, I'm going to teach you a very important lesson about dealing with users. Never trust them. Ask her what the manufactorer and model is of both devices.`
`22:37.14`	`_Paulo_`	`opus_, but sorry, I never tested this.`
`22:37.25`	`vooduhal`	`Because if the modem doesn't dial for them, and the computers are not dialing a virtual connection, then PPPoE is not in use.`
`22:37.37`	`*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)`
`22:37.49`	`vooduhal`	`It's like sticking a computer with a modem on a phone line with a 56k modem and it automatically getting an IP address and internet connection.`
`22:37.50`	`TripleFFFF`	`??`
`22:38.05`	`Dr-Linux`	`vooduhal: PPPoE is only dialing connection? not a DSL?`
`22:38.18`	`Qwell[]`	`dsl doesn't need to use pppoe`
`22:38.22`	`*** join/#asterisk tsurk0 (n=tsurko@digsys226-159.pip.digsys.bg)`
`22:38.43`	`_Paulo_`	`TripleFFFF, you can create a view`
`22:38.48`	`Qwell[]`	`and if yours does, you just need to get a pppoe router`
`22:38.58`	`Qwell[]`	`(one smart enough to get multiple ips)`
`22:38.58`	`_Paulo_`	`TripleFFFF, rename your table and create a view.`
`22:39.12`	`vooduhal`	`Dr-Linux, the whole point of PPPoE is to provide a dialup interface like dial up. There is no real reason for an ISP to require it, but some just like to provide that old fashoined feeling.`
`22:39.34`	`Dr-Linux`	`Qwell[]: yes, it's multiple IP's , and public`
`22:39.40`	`vooduhal`	`Without the model of the modem and "hub" I can't really help you anymore.`
`22:40.36`	`TripleFFFF`	`no matter hide from dir=yes`
`22:40.41`	`Dr-Linux`	`vooduhal: i asked her, she is just checking`
`22:40.49`	`*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)`
`22:41.08`	`TripleFFFF`	`anyway to make this talked by festival ?`
`22:41.37`	`vooduhal`	`K.`
`22:42.23`	`TripleFFFF`	`source says VoiceMail2`
`22:42.24`	`TripleFFFF`	`i think`
`22:42.25`	`Dr-Linux`	`vooduhal: it's a autosensing switch >> NSH510 is the model #`
`22:42.30`	`TripleFFFF`	`<PROTECTED>`
`22:42.35`	`TripleFFFF`	`./* Check for the VoiceMail2 greeting first */`
`22:42.44`	`TripleFFFF`	`would mean it could be festival based ?`
`22:43.11`	`*** join/#asterisk bobman (n=bobman@24-53-5-197.agstme.adelphia.net)`
`22:43.16`	`*** part/#asterisk opus_ (n=opus@68.216.187.60)`
`22:43.39`	`*** part/#asterisk epoch (n=epoch@octane.breakbeats.org)`
`22:44.31`	`vooduhal`	`And the modem?`
`22:44.56`	`Dr-Linux`	`vooduhal: she is checking`
`22:44.59`	`vooduhal`	`K`
`22:45.16`	`Dr-Linux`	`broadxent by creative`
`22:45.29`	`Dr-Linux`	`Model# 8012-V`
`22:46.12`	`*** join/#asterisk operat0r (i=operator@adsl-152-132-93.asm.bellsouth.net)`
`22:46.55`	`TripleFFFF`	`join #asterisk-dev`
`22:47.28`	`Dr-Linux`	`vooduhal: hopfully you will give me feedback postive , like it can work or it can't :)`
`22:47.39`	`vooduhal`	`I agree. :)`
`22:48.09`	`vooduhal`	`Also, have you verified that she is getting public IPs or is that just what she as said?`
`22:48.25`	`vooduhal`	`(Sorry, did tech support for too many years to trust a thing a user says)`
`22:48.53`	`Dr-Linux`	`Vagabond: i access her machines' via VNC and i checked my self, it's public IPs and always changed`
`22:49.01`	`vooduhal`	`K.`
`22:51.37`	`*** join/#asterisk opus_ (n=opus@68.216.187.60)`
`22:51.48`	`*** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.8, Zaptel 1.2.6, Libpri 1.2.3 Released! (May 30, 2006) -=- FreePBX/AMP/Asterisk@Home support in #freepbx`
`22:51.53`	`opus_`	`how do I match an area code? exten => _206NXXNXXXX,1, ?`
`22:52.05`	`vooduhal`	`Yes.`
`22:52.08`	`_Paulo_`	`opus_, yes`
`22:52.13`	`opus_`	`are you sure?`
`22:52.28`	`vooduhal`	`1.2.8 is out???!??!`
`22:52.30`	`vooduhal`	`Woot!!!!`
`22:52.30`	`opus_`	`if I had _X. after it, but that was before it, it would still catch the 206 right?`
`22:52.33`	`_Paulo_`	`opus_, the trick with the group dialing worked?`
`22:52.39`	`opus_`	`Paulo never tried`
`22:52.55`	`vooduhal`	`Did they include the updates to app_queue like autopause and the parallel distribution?`
`22:53.16`	`vooduhal`	`Sounds like I'm staying late to upgrade. :)`
`22:53.28`	`Dr-Linux`	`what's new features in 1.2.8?`
`22:53.37`	`Qwell[]`	`Dr-Linux: none`
`22:54.00`	`vooduhal`	`Checking app_queue now.`
`22:54.11`	`Dr-Linux`	`Qwell[]: what's difference than previous version?`
`22:54.13`	`Qwell[]`	`vooduhal: was it a bug fix, or a feature?`
`22:54.15`	`Qwell[]`	`Dr-Linux: bug fixes`
`22:54.17`	`_Paulo_`	`Time to go home. Bye!`
`22:54.38`	`*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)`
`22:54.55`	`vooduhal`	`Feature added to app_queue in CVS head about a month ago.`
`22:55.10`	`Qwell[]`	`vooduhal: then no`
`22:55.13`	`russellb`	`The ChangeLog between 1.2.7.1 and 1.2.8 is about 500 lines`
`22:55.15`	`Qwell[]`	`ONLY bug fixes were included`
`22:55.18`	`vooduhal`	`The parallel call distribution should have been added a long, long time ago.`
`22:55.20`	`russellb`	`tons of bug fixes`
`22:55.30`	`Qwell[]`	`(unless there was a very compelling reason for a feature to be included)`
`22:55.57`	`opus_`	`bah.`
`22:56.17`	`opus_`	`i'll wait until 1.2.8.1`
`22:56.20`	`Qwell[]`	`If you want new features, either test trunk, or wait for 1.4 beta`
`22:56.28`	`opus_`	`which should be out in about 20 minutes from now, hehe`
`22:56.29`	`*** join/#asterisk Delta239 (n=paparapa@cpe-0014bfab77da.cpe.cableonda.net)`
`22:56.30`	`file`	`no new features for you!`
`22:56.33`	`Dr-Linux`	`one day my callers will play games while waiting in queues :P`
`22:56.35`	`vooduhal`	`Lol.`
`22:56.51`	`Qwell[]`	`Dr-Linux: Tell file to give you his blackjack dialplan :P`
`22:56.58`	`Qwell[]`	`Which I still want, btw!`
`22:57.10`	`vooduhal`	`Do you think trunk is stable enough for a production call center of 30 users and 50 other non queue users?`
`22:57.12`	`file`	`you could actually rework it for DTMF`
`22:57.20`	`Dr-Linux`	`Qwell[]: sorry, don't have money for file :S`
`22:57.21`	`Qwell[]`	`file: tis the plan`
`22:57.35`	`Qwell[]`	`that, and to get ideas for...yep...you guessed it`
`22:57.39`	`Qwell[]`	`hold'em!`
`22:58.00`	`russellb`	`you could have a CARDDECK dialplan function`
`22:58.03`	`Dr-Linux`	`file: you will give me that for free? :P`
`22:58.08`	`Qwell[]`	`russellb: That would be sweet`
`22:58.15`	`file`	`noooooooooooooooooooo`
`22:58.16`	`russellb`	`that could keep track of which cards have been dealt, etc`
`22:58.33`	`Dr-Linux`	`file: ok sorry :)`
`22:58.51`	`Qwell[]`	`russellb: yeah...Set(newcard=${CARDDECK(1)})`
`22:58.51`	`file`	`russellb: datastores... yesssss`
`22:59.03`	`russellb`	`file: yup :)`
`22:59.08`	`Dr-Linux`	`when will be 1.4 beta out?`
`22:59.14`	`Qwell[]`	`Dr-Linux: soon my child...soon`
`22:59.15`	`file`	`soon.`
`22:59.19`	`file`	`like, very soon`
`22:59.24`	`russellb`	`Dr-Linux: like, monday at the latest`
`22:59.34`	`russellb`	`it was going to be tomorrow, but there are some important things to finish this week`
`22:59.46`	`Dr-Linux`	`oo :S`
`23:00.00`	`russellb`	`rtp jitterbuffer is pretty much ready, though`
`23:00.01`	`russellb`	`w00t`
`23:00.11`	`file`	`so no more complaining is allowed`
`23:00.16`	`russellb`	`none`
`23:00.38`	`Dr-Linux`	`i bought new dual core server for asterisk, i'll recieve it on 1/6, what version should i load. 1.2.8?`
`23:01.20`	`Dr-Linux`	`russellb: what's special stuff in 1.4 beta?`
`23:02.10`	`operat0r`	`Hey guys I finaly got ipkall to talk to my asterisk box bypassing FWD but I am on NAT do I need to set that some place even if I dont have anything in sip.conf ? http://pastebin.com/748014`
`23:02.30`	`Ariel_`	`I knew they were working on releasing 1.2.8 version of asterisk...`
`23:02.39`	`Dr-Linux`	`vooduhal: you forgot my problem? :P`
`23:02.58`	`vooduhal`	`No, I'm still here.`
`23:03.06`	`vooduhal`	`DId you get the model on the modem?`
`23:03.20`	`Dr-Linux`	`vooduhal: i aleady give you`
`23:03.29`	`vooduhal`	`Sorry, I must have missed it.`
`23:03.40`	`Dr-Linux`	`broadxent by creative`
`23:03.40`	`russellb`	`Dr-Linux: oh geez, i don't even know, we'll have to work on a list sometime soon`
`23:03.57`	`Dr-Linux`	`Model# 8012-V`
`23:04.04`	`vooduhal`	`Manu?`
`23:04.18`	`vooduhal`	`Broadxent?`
`23:04.41`	`Dr-Linux`	`russellb: it will be not a gui?`
`23:04.46`	`Dr-Linux`	`vooduhal: yes dude.`
`23:06.10`	`russellb`	`Dr-Linux: no.`
`23:06.27`	`mitcheloc`	`oh just a thought for everyone here... it seems counterpath has forums on their website, so it might be good to point x-lite/eyebeam users to support.counterpath.net`
`23:06.53`	`*** join/#asterisk rmayorga (n=churro@168.243.89.17)`
`23:07.10`	`Dr-Linux`	`mitcheloc: is that your web?`
`23:07.53`	`vooduhal`	`Ok, you said you have VNC access to their desktop correct?`
`23:08.18`	`rmayorga`	`help`
`23:08.21`	`Dr-Linux`	`yes, i used always when i'm at work`
`23:08.26`	`rmayorga`	`hi guys`
`23:08.37`	`Dr-Linux`	`vooduhal: did you findout something?`
`23:08.42`	`operat0r`	`http://www.rmccurdy.com/stuff/twat_SC_VNC.mp3 my VNC over firewall/ NAT guide`
`23:09.13`	`rmayorga`	`I have question, there is anyway that I can do somethink like DIAL(SIP/XXXXX@foo) from my asterisk CLI`
`23:09.37`	`Qwell[]`	`rmayorga: with the Dial command, and add extension, if you have chan_oss or chan_alsa`
`23:10.08`	`vooduhal`	`Just reading the docs on it. I'm still having trouble believing that they are not dialing a PPPoE connection on each PC though. The last thing I can suggest is VNCing in, checking their network connections and see if they just have a "Local Area Network" Connection or if they actually have a PPPoE dialup connection. If they do have a PPPoE dialup connection, you are screwed.`
`23:10.32`	`Az_au`	`hello... what card(s)? would be recommended for use with 2 BRI lines?`
`23:10.35`	`operat0r`	`raspppoe`
`23:10.48`	`Qwell[]`	`operat0r: good luck putting ras on a phone`
`23:10.52`	`operat0r`	`altern pppoe drivers`
`23:10.58`	`TripleFFFF`	`wats is changelog on 1.8 ?`
`23:11.04`	`TripleFFFF`	`ioi mean 1.2.8`
`23:11.04`	`rmayorga`	`Qwell[]: thanks`
`23:11.19`	`*** join/#asterisk Lord_Drachenblut (n=Lord@12.210.112.14)`
`23:11.22`	`operat0r`	`http://pastebin.com/748014`
`23:11.30`	`vooduhal`	`Anyway, back to getting access to our multicast video streams for watching world cup next week. Back later all.`
`23:11.32`	`operat0r`	`monkeys won't play :(`
`23:11.39`	`Dr-Linux`	`vooduhal: i have checked that already, while accesing her PC, she has PPPoE connection, when i do >> run >> cmd >> ipconfig at her windows machine`
`23:12.22`	`Qwell[]`	`Dr-Linux: You need to get a router that can do pppoe and multiple IPs, and let it do your dhcp`
`23:13.01`	`vooduhal`	`Dr-Linux, in that case, you are going to need a router that will do PPPoE ....`
`23:13.07`	`vooduhal`	`As Qwell said. :)`
`23:13.08`	`Lord_Drachenblut`	`anyone ever get a lucent phone working with asterisk`
`23:13.09`	`Dr-Linux`	`Qwell[]: it's already doing that. all PC's grab dynamic IPs ..`
`23:13.15`	`Dr-Linux`	`but this cisco phone can't`
`23:13.32`	`Qwell[]`	`Dr-Linux: but, it's getting it straight from the modem...you need a router that can do the pppoe connections for you`
`23:13.34`	`vooduhal`	`Dr-Linux, yes, but they are establishing PPPoE connections on their own.`
`23:13.59`	`vooduhal`	`I wonder if anyone makes a single port PPPoE gateway. :)`
`23:14.13`	`Qwell[]`	`vooduhal: Linux`
`23:14.18`	`vooduhal`	`Lmao.`
`23:14.19`	`Dr-Linux`	`vooduhal: so what's difference between a PC and cisco phone?`
`23:14.24`	`Dr-Linux`	`just wanna confirm`
`23:14.25`	`rmayorga`	`Qwell[]: With the add extension command I can add a new extension to a context`
`23:14.26`	`Qwell[]`	`Dr-Linux: the PC has a dialer`
`23:14.33`	`vooduhal`	`The PC has a PPPoE client.`
`23:14.46`	`rmayorga`	`That I need is to try to make a Automatic, call from the CLI`
`23:15.02`	`rmayorga`	`doing something like asterisk -rx COMMAND`
`23:15.15`	`Qwell[]`	`rmayorga: Just use a call file`
`23:15.16`	`Dr-Linux`	`i see`
`23:15.19`	`operat0r`	`some other stuff you may need to set to make PPPOE to work`
`23:15.29`	`operat0r`	`not MTU but something else`
`23:15.41`	`Qwell[]`	`BTU?`
`23:15.53`	`Dr-Linux`	`vooduhal: what if she buy a linksys router and that do NAT. it will work right?`
`23:16.08`	`vooduhal`	`Yes.`
`23:16.12`	`Qwell[]`	`Dr-Linux: You need to make sure that the linksys can handle multiple public IPs`
`23:16.15`	`vooduhal`	`Just be prepared for the SIP fun.`
`23:16.21`	`Qwell[]`	`unless you want to switch to NAT..which is silly`
`23:16.29`	`vooduhal`	`Dr-Linux, is there a need for multiple public IPs?`
`23:16.35`	`vooduhal`	`NAT is silly?`
`23:16.46`	`Qwell[]`	`When you're paying for public IPs, of course it is`
`23:16.48`	`MikeJ__`	`nat is fine... stun is good`
`23:16.57`	`Qwell[]`	`You've got em...use em`
`23:17.08`	`vooduhal`	`Qwell, agreed, but it doesn't sound like they are.`
`23:17.11`	`*** join/#asterisk iq\|mobile (n=iq@71-215-34-237.omah.qwest.net)`
`23:17.21`	`Qwell[]`	`why let them go to waste?`
`23:17.24`	`MikeJ__`	`what's up with this sip doesn't handle nat well myth?`
`23:17.28`	`Qwell[]`	`It'd be simple to get a router that can deal with it`
`23:17.33`	`Qwell[]`	`MikeJ__: got me`
`23:17.43`	`vooduhal`	`Qwell, they are getting them dynamically through DSL/DHCP and changing often so what's the point?`
`23:17.48`	`MikeJ__`	`it's up there with the h323 sucks myth`
`23:17.50`	`vooduhal`	`Do you get your boxes owned?`
`23:17.56`	`Qwell[]`	`MikeJ__: no, that one is true :p`
`23:18.14`	`vooduhal`	`Qwell, agreed on the router.`
`23:18.36`	`vooduhal`	`Dr-Linux, Tell her for $1k, I'll fly out and fix everything for her. :)`
`23:21.52`	`Dr-Linux`	`vooduhal: heh if i'd have handly access i would be able to fix it,`
`23:22.08`	`vooduhal`	`I know. :) You said you are in Pakistan right?`
`23:22.10`	`Dr-Linux`	`vooduhal: i configured a bunch of Cisco phones, but never seen ONE in real`
`23:22.17`	`Dr-Linux`	`vooduhal: yes`
`23:22.25`	`vooduhal`	`What part of the US is she in?`
`23:22.53`	`Dr-Linux`	`vooduhal: CA`
`23:23.14`	`vooduhal`	`Ah, on the other side of the country. :)`
`23:23.15`	`Dr-Linux`	`vooduhal: but now i understand everything with 2 lines`
`23:23.25`	`Dr-Linux`	`i appritiate your help`
`23:23.37`	`Dr-Linux`	`you know what was 2 lines?`
`23:23.55`	`Dr-Linux`	`<Qwell[]> Dr-Linux: the PC has a dialer`
`23:23.57`	`Dr-Linux`	`<vooduhal> The PC has a PPPoE client.`
`23:24.08`	`vooduhal`	`Glad to help. :)`
`23:24.51`	`Dr-Linux`	`vooduhal: you don't often come to this channel, right? :)`
`23:25.09`	`vooduhal`	`Some times. I don't spend a lot of time on IRC.`
`23:27.44`	`*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)`
`23:27.56`	`JASON99`	`How would I convert 7 digits to 10 digits before dialing out?`
`23:28.20`	`vooduhal`	`JASON99, you meaning appending an area code?`
`23:28.24`	`JASON99`	`yes`
`23:28.34`	`JASON99`	`I should have specified :P`
`23:28.57`	`vooduhal`	`exten => _NXXXXXX,1,Dial(Zap/g0/234${EXTEN})`
`23:29.17`	`vooduhal`	`Will dial 234 and your 7 digit number.`
`23:30.01`	`operat0r`	`grrr`
`23:30.03`	`operat0r`	`Reliably Transmitting (NAT) to 66.54.140.46:5060:`
`23:30.04`	`operat0r`	`SIP/2.0 404 Not Found`
`23:30.04`	`JASON99`	`ok you're right.. I tried that and it works.. but here is my problem... once I add the area code.. I want to lookup the first 6 digits and match that to decide which trunk to send it on...`
`23:30.41`	`vooduhal`	`${EXTEN:1:6} I believe.`
`23:30.48`	`vooduhal`	`That's after you've created the 10 digit number.`
`23:30.52`	`vooduhal`	`Or more clearly`
`23:31.08`	`vooduhal`	`Set(MYEXTEN=423${EXTEN})`
`23:31.24`	`vooduhal`	`${MYEXTEN:1:6} should refer to the first 6 digits.`
`23:32.02`	`vooduhal`	`So GotoIf($[${MYEXTEN} = 123435]?2:3)`
`23:32.07`	`vooduhal`	`Something like that.`
`23:32.12`	`JASON99`	`Do I have to use a GotoIf or is there a way to do exten => _234111XXXX,1,Dial(${EXTEN})`
`23:32.50`	`JASON99`	`ok your example will work.. Thanks :)`
`23:33.27`	`vooduhal`	`Yep.`
`23:33.49`	`operat0r`	`so I am ipkalll > my asterisk box I dont need a sip.conf ? just extentions.conf ?`
`23:33.53`	`vooduhal`	`I think you could do something like:`
`23:34.06`	`vooduhal`	`What's that damned place to paste code at?`
`23:34.17`	`operat0r`	`vooduhal pastebin`
`23:34.23`	`vooduhal`	`Url?`
`23:34.26`	`*** join/#asterisk omarc55 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net)`
`23:34.35`	`JASON99`	`http://pastebin.ca/`
`23:34.42`	`vooduhal`	`Thank you.`
`23:34.45`	`vooduhal`	`Time to bookmark.`
`23:34.49`	`JASON99`	`hehe`
`23:34.55`	`operat0r`	`time to smack asterisk`
`23:35.07`	`omarc55`	`Hi all. how can I find out how many kbps an IAX2 call is taking up, is there a tool to test it from the asterisk console? I am using 1.2.7.1`
`23:35.16`	`operat0r`	`omarc55 bmon`
`23:35.33`	`*** part/#asterisk opus_ (n=opus@68.216.187.60)`
`23:35.39`	`omarc55`	`ah ok. thanks.`
`23:35.43`	`vooduhal`	`Hey * gurus, if EXTEN changes mid dialplan, does the priority reset to 1?`
`23:38.23`	`nextime`	`vooduhal : no`
`23:38.47`	`vooduhal`	`Didn't think so.`
`23:40.32`	`vooduhal`	`JASON99, This will do what you want without gotoifs: http://pastebin.ca/60106`
`23:41.39`	`*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)`
`23:41.42`	`operat0r`	`at least I got it to do something when I get a call`
`23:41.52`	`JASON99`	`vooduhal: Perfect.. Let me try that out.. Thanks`
`23:42.12`	`vooduhal`	`Np.`
`23:43.53`	`*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)`
`23:44.15`	`JASON99`	`Perfect.. it worked`
`23:44.17`	`rmayorga`	`Qwell[]: Thanks, It Works nice`
`23:45.42`	`vooduhal`	`JASON99, good to hear.`
`23:46.56`	`JASON99`	`Thanks again`
`23:51.20`	`vooduhal`	`Did someone earlier say that you could swap some pins in a standard Cat5 for a 7940 to use PoE?`
`23:51.36`	`Qwell[]`	`vooduhal: yes`
`23:51.47`	`Qwell[]`	`on a newer one`
`23:51.49`	`vooduhal`	`What was the pinout. I want to try. I don't care if I brick it. :)`
`23:51.54`	`vooduhal`	`I've got a 7940g`
`23:52.03`	`vooduhal`	`The docs say it supports it.`
`23:52.12`	`Qwell[]`	`I think some of the g's support 802.3 poe`
`23:52.14`	`vooduhal`	`But my PoE injector doesn't do shit for it.`
`23:52.26`	`Qwell[]`	`vooduhal: there is a thing on the wiki, I believe`
`23:52.29`	`Qwell[]`	`~wikis`
`23:52.30`	`jbot`	`somebody said wikis was http://www.voip-info.org`
`23:52.38`	`vooduhal`	`Lol.`
`23:52.41`	`mitcheloc`	`~food`
`23:52.42`	`jbot`	`[food] essential to life`
`23:52.42`	`vooduhal`	`Let me chceck.`
`23:52.53`	`mitcheloc`	`~givemefood`
`23:54.35`	`vooduhal`	`Woot, found the doc.`
`23:55.57`	`JASON99`	`When you have a context with includes.. does it go through the includes starting from the top or bottom?`
`23:56.04`	`*** join/#asterisk MoutaPT (n=MoutaPT@85.139.196.147)`
`23:57.26`	`znoG`	`guys, can anyone think of a way one could setup a dialplan conf or a AGI script that auto-redials a number when busy and when it rings, connect the call to an extension?`
`23:58.15`	`znoG`	`ie. extension 5 rings number 123 and hangs up. Asterisk dials 123 and if not busy, dial extension 5 and bridge. If busy, keep retrying until its not`
`23:59.36`	`Qwell[]`	`znoG: You mean like...RetryDial()?`

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