IRC log for #asterisk on 20090620

`00:00.57`	`[TK]D-Fender`	`misyel: Looks like a star...`
`00:01.18`	`lanning`	`*`
`00:01.27`	`[TK]D-Fender`	`lanning: Yeah, kinda like that`
`00:01.28`	`misyel`	`if I run it on my dedicated server box, how can I use it to call someone?`
`00:02.02`	`Docteh`	`~book`
`00:02.02`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`00:03.08`	`[TK]D-Fender`	`misyel: Where would * sit in this solution?`
`00:03.19`	`[TK]D-Fender`	`misyel: And what are you expecting * to do for you?`
`00:03.45`	`misyel`	`to call my mom to u.s for free`
`00:04.13`	`misyel`	`I suppose asterisk is like a voip server right?`
`00:04.27`	`misyel`	`so if I run it on my dedicated server, how can I and how can my mom connect to it?`
`00:04.43`	`misyel`	`does it compile a program where she runs from her computer`
`00:06.45`	`Docteh`	`asterisk is not a C compiler`
`00:08.23`	`seanbright`	`misyel: she can connect with a SIP softphone`
`00:08.27`	`seanbright`	`as can you`
`00:08.42`	`Docteh`	`misyel: http://en.wikipedia.org/wiki/Asterisk_(PBX) <-- you'll figure stuff out way faster if you read some basic information on what asterisk is and does`
`00:10.24`	`[TK]D-Fender`	`misyel: * does not make calling free`
`00:10.54`	`[TK]D-Fender`	`misyel: * lets you connect varios devices and services together and act in a PBX toolkit capacity`
`00:11.32`	`[TK]D-Fender`	`misyel: * does not provide free services. * merely lets you utilitize them in various ways`
`00:11.52`	`[TK]D-Fender`	`misyel: Ever used a phone system in a company before?`
`00:22.02`	`carrar`	`Wait!! I can't make free calls!!`
`00:22.03`	`carrar`	`WTF!!`
`00:34.53`	`misyel`	`[08:10] <[TK]D-Fender> misyel: * does not make calling free << so what makes it popular then if it's not free`
`00:35.39`	`misyel`	`for example, I run asterisk on my dedicated server, I and my mom connect to it through SIP softphone right? and my my calls me, how is she gonna pay for that call? or is that call free?`
`00:35.55`	`misyel`	`and how can she call me? what would be my number`
`00:36.44`	`engien`	`asterisk isnt skype`
`00:36.51`	`lanning`	`cost of server, cost of power, cost of internet connection, cost of upkeep...`
`00:36.56`	`engien`	`it would depend on what you set up`
`00:37.03`	`engien`	`sounds like you want skype or equivalent`
`00:38.51`	`misyel`	`can someone please enlighten me please`
`00:39.11`	`engien`	`i just did`
`00:39.29`	`lanning`	`with asterisk, you are running everything, and paying for all low level services`
`00:39.39`	`lanning`	`skype, just download app and run.`
`00:39.56`	`[TK]D-Fender`	`misyel: Are you going to complain that just because I give you a free PHONE that you still hve to pay for the line to use it with?`
`00:40.14`	`[TK]D-Fender`	`misyel: How about is Exxon gave you a free CAR?`
`00:40.30`	`[TK]D-Fender`	`misyel: You don't seem to understand *'s FUNCTION`
`00:41.03`	`[TK]D-Fender`	`misyel: it is a PBX toolkit, not a "invent free service to wherever the hell I ple that I can't do in some other way without"`
`00:41.36`	`misyel`	`ok, right now...we're using mymplus voip...if I run asterisk, that means we don't have to pay mymplus anymore right? cause we're running our own server`
`00:41.50`	`[TK]D-Fender`	`misyel: You using SIP to talk to your mother is as free as the means by which packets can frow between you`
`00:41.58`	`misyel`	`[08:40] <[TK]D-Fender> misyel: You don't seem to understand *'s FUNCTION << that's the point, I don't understand...`
`00:41.59`	`lanning`	`try reading the book, that was posted earlier. You will need to learn about VoIP issues, asterisk's channel and extension logic.`
`00:42.12`	`misyel`	`so what is this channel for?`
`00:42.19`	`[TK]D-Fender`	`misyel: For your simplistic approach, think of * as a PBX.`
`00:42.23`	`lanning`	`misyel, my power bill is $400 a month, because I run my own servers.`
`00:42.28`	`misyel`	`I wouldn't have come here if you just tell me a book to read`
`00:42.35`	`[TK]D-Fender`	`misyel: Its for those implementing * to acheive some sort of goal.`
`00:42.37`	`misyel`	`how about closing this channel instead`
`00:42.52`	`lanning`	`misyel, we would have to recite the book`
`00:42.53`	`[TK]D-Fender`	`misyel: Do you understand what a PBX is?`
`00:43.41`	`lanning`	`this channel is to answer direct questions, not abstract questions.`
`00:44.15`	`carrar`	`You FAILED to enter in your CC`
`00:44.22`	`[TK]D-Fender`	`lanning: I'm sorry, you do not win the showcase showdown because you forgot to phrase your response in the form of a haiku`
`00:44.37`	`lanning`	`heh`
`00:48.48`	`misyel`	`sigh ok, can someone tell me what to do so I and my mom can make calls for free using asterisk`
`00:49.52`	`carrar`	`Buy two SIP phones, ensure you both have decent speed internet, get a linux box to run asterisk on`
`00:50.15`	`[TK]D-Fender`	`misyel: You don't need * to talk to your mother`
`00:50.16`	`carrar`	`then read the book to learn how to configure it`
`00:50.42`	`[TK]D-Fender`	`misyel: You can use a softphone on your system and on heres and use them to talk to each other`
`00:50.51`	`[TK]D-Fender`	`misyel: * isn't required to do that`
`00:51.00`	`carrar`	`but he noted he wanted to use asterisk`
`00:51.29`	`engien`	`~book`
`00:51.29`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`00:51.52`	`engien`	`you couldve read the first chapter or two by now`
`00:51.53`	`engien`	`i have :p`
`00:52.03`	`misyel`	`I don't care if I don't need asterisk to do that, I know I can make calls through skype, yahoo messenger but like I said....USING ASTERISK`
`00:52.09`	`misyel`	`can it be done through asterisk?`
`00:52.26`	`misyel`	`if yes, I want to know how can it be done`
`00:52.37`	`misyel`	`like tell me >> 1.) you need a server to run asterisk from`
`00:52.44`	`misyel`	`2.) download SIP softphone`
`00:52.46`	`engien`	`youre pretty damned arrogant for someone requesting help and refusing to do any research on their own`
`00:52.53`	`misyel`	`3rd connect to it bla bla`
`00:53.08`	`[TK]D-Fender`	`misyel: Are you following that you don't need * to talk to her for free over the internet?`
`00:53.16`	`carrar`	`<carrar> Buy two SIP phones, ensure you both have decent speed internet, get a linux box to run asterisk on`
`00:53.45`	`misyel`	`[08:53] <[TK]D-Fender> misyel: Are you following that you don't need * to talk to her for free over the internet? << are you also following that I mean to say >> using asterisk? <<`
`00:54.06`	`[TK]D-Fender`	`misyel: * = ASTERISK`
`00:54.07`	`lanning`	`misyel, there is a problem with step 1`
`00:54.16`	`[TK]D-Fender`	`misyel: I've been saying ASTERISK this whole time`
`00:54.18`	`lanning`	`there are about 200 sub-steps in that`
`00:54.25`	`[TK]D-Fender`	`misyel: * is not REQUIRED for what you want.`
`00:54.45`	`engien`	`has been in the channel as long as misyel and is halfway through configuring the centos install as suggested in the oreilly book`
`00:54.53`	`*** join/#asterisk ruben23 (n=RPL@124.107.3.178)`
`00:55.14`	`misyel`	`<carrar> Buy two SIP phones, ensure you both have decent speed internet, get a linux box to run asterisk on << ok, I do have a dedicated server with centos distro installed...after I buy two sip phones, how can we call each other?`
`00:55.14`	`[TK]D-Fender`	`engien: Congratulations on your initiative`
`00:55.21`	`misyel`	`what would be our numbers?`
`00:55.30`	`ruben23`	`[TK]D-Fender: hi`
`00:55.32`	`carrar`	`misyel, You will need to instal Asterisk`
`00:55.34`	`carrar`	`and`
`00:55.35`	`lanning`	`misyel, read the book, then come back here for clarifications.`
`00:55.35`	`carrar`	`configure it`
`00:55.38`	`[TK]D-Fender`	`misyel: Just dial by IP. Or use a free service like ekiga.net`
`00:55.52`	`carrar`	`The book will tell you how to do that, as you won't get people here doing it for you`
`00:56.11`	`[TK]D-Fender`	`carrar: Actually.. the book won't tell him that....`
`00:56.16`	`carrar`	`heh`
`00:56.41`	`carrar`	`it will give him the fundamentals on how to do it`
`00:56.57`	`misyel`	`oky, I think I'm on it...so, where to buy that SIP phones?`
`00:56.58`	`carrar`	`he'll have to connect the dots`
`00:57.12`	`carrar`	`You can use free xlite for now`
`00:57.12`	`[TK]D-Fender`	`misyel: Just install a SOFTPHONE on each of your computers.`
`00:57.14`	`[TK]D-Fender`	`~softphone`
`00:57.15`	`infobot`	`[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga`
`00:57.15`	`carrar`	`till you find a phone`
`00:57.30`	`[TK]D-Fender`	`~xlite`
`00:57.30`	`infobot`	`[~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/`
`00:57.32`	`[TK]D-Fender`	`^^^`
`00:58.09`	`carrar`	`misyel, buy two polycom 430's`
`00:58.12`	`carrar`	`OR`
`00:58.13`	`carrar`	`actually`
`00:58.16`	`[TK]D-Fender`	`carrar: Nah`
`00:58.17`	`carrar`	`buy two of their HD phones`
`00:58.24`	`[TK]D-Fender`	`carrar: Linksys ATA at most...`
`00:58.42`	`[TK]D-Fender`	`carrar: wouldn't want to complicate configs or use PBX features... he doesn't ahve a PBX for :)`
`00:58.50`	`carrar`	`heh`
`00:59.04`	`misyel`	`can I use the Global Softphone?`
`00:59.17`	`misyel`	`can I use the Global Softphone to connect to asterisk server?`
`00:59.21`	`carrar`	`test it`
`00:59.24`	`carrar`	`let us know`
`00:59.29`	`[TK]D-Fender`	`'s egg smashes carrar's chicken with a +5 Broadsword for 3D + 70,000,000 Damage!!!`
`01:00.09`	`carrar`	`Don't make me start to drink MR!!`
`01:00.10`	`[TK]D-Fender`	`misyel: What is this "Global Softphone", and what protocols does it use?`
`01:00.33`	`ruben23`	`hi i got this sip debug for my asterisk server when callers cant hear the client on call ive test this with 3 voip telco same thing happens, asterisk server is not behind NAT, its using public IP.http://pastebin.com/m41dd551d`
`01:00.34`	`misyel`	`in network configuration, what do you need to connect? the IP of asterisk's server, the server SIP port, local SIP port and what else?`
`01:01.09`	`*** join/#asterisk dacs (n=chatzill@unaffiliated/dacs)`
`01:01.09`	`[TK]D-Fender`	`misyel: Server IP, port is default unless you change it, and a user & pass`
`01:02.23`	`dacs`	`~pd`
`01:05.29`	`ruben23`	`how to interpret sip debug`
`01:05.49`	`carrar`	`how to type english`
`01:06.17`	`misyel`	`[TK]D-Fender - where to get the user and pass?`
`01:06.27`	`[TK]D-Fender`	`misyel: Huh?!`
`01:06.42`	`carrar`	`he's not allowed to give that to you`
`01:06.51`	`carrar`	`it's secret`
`01:07.02`	`lanning`	`misyel, you make them up, and put them in the asterisk config and the phone`
`01:07.16`	`misyel`	`where to get the user and pass to connect to my asterisk's server`
`01:07.21`	`misyel`	`ohh okey`
`01:07.25`	`[TK]D-Fender`	`misyel: What is this talk about * for right now?`
`01:07.44`	`[TK]D-Fender`	`misyel: You do not require it for what you have described wanting to do.`
`01:07.54`	`misyel`	`[09:07] <lanning> misyel, you make them up, and put them in the asterisk config and the phone << how many users can I add to the config file that comes in asterisk`
`01:08.02`	`carrar`	`where to get enlightenment`
`01:08.09`	`[TK]D-Fender`	`misyel: I asked you if you knew what a PBX was and didn't get an answer.`
`01:08.11`	`lanning`	`misyel: 1000000000000`
`01:08.44`	`[TK]D-Fender`	`misyel: I can only take this as a "no", at which point you seem to spit "asterisk" as being the solution to your needs without understanding what it is.`
`01:09.15`	`lanning`	`misyel, btw, there are about 20ish config files for asterisk...`
`01:09.22`	`carrar`	`Asterisk is the gateway to enlightenment`
`01:09.25`	`misyel`	`no, I don't know what it is but what I know is asterisk is like a voip server or something like that`
`01:09.43`	`lanning`	`it is a voip framework`
`01:09.55`	`[TK]D-Fender`	`carrar: And .conf files the sharp rocks at the bottom that will break your fall when we push people through it!`
`01:10.00`	`lanning`	`which means you have to piece a lot of it together yourself.`
`01:10.02`	`carrar`	`haha`
`01:10.04`	`carrar`	`YES`
`01:10.11`	`[TK]D-Fender`	`miyYou don't know what a PBX is?`
`01:10.16`	`[TK]D-Fender`	`misyel: You don't know what a PBX is?`
`01:10.21`	`carrar`	`VoIP stuff is cool`
`01:10.27`	`misyel`	`no`
`01:10.29`	`[TK]D-Fender`	`misyel: I asked if you eve used a phone system at a company before.`
`01:10.39`	`misyel`	`yes from mymplus`
`01:11.35`	`[TK]D-Fender`	`miThats what * is. The central piece of hardware that rules what one of the phones attached to it is allowed to dial, and lets a regular boring shit phone connected to is ustilize the resoureces conencted to the PBX is any way it is configured to.`
`01:12.21`	`misyel`	`then that's what exactly I need...`
`01:12.21`	`[TK]D-Fender`	`misyel: Can a 1 line phone normally get calls from 500 lines? No its plugged to jsut 1 wire. but the PBX can be connected to 500 lines and can be told to direct a call to the port the phone is plugged into.`
`01:12.31`	`lanning`	`[TK]D-Fender: you drinking?`
`01:13.04`	`lanning`	`your typing is deteriorating... :)`
`01:13.13`	`[TK]D-Fender`	`misyel: Sure you need something to get your call to your mother, but her IP address alone is enough. And ther are free services that do not require you having to set up your own central system`
`01:13.38`	`[TK]D-Fender`	`misyel: You do not need to do all of this for the minimum of what you're asking.`
`01:13.43`	`*** join/#asterisk dacs (n=chatzill@unaffiliated/dacs)`
`01:13.54`	`misyel`	`ohh well, I am paying 600 usd for my 3 dedicated servers and I don't use much resources...wouldn't it be waste if I don't run something from it?`
`01:13.59`	`[TK]D-Fender`	`misyel: Just set up an account for each of you at ekiga.net and set your softphone up to use it.`
`01:14.39`	`*** join/#asterisk brookshire (i=mbrooks@hijacked.us)`
`01:15.09`	`[TK]D-Fender`	`misyel: And you're here asking about Asterisk when you could be learning astrophysics. You probably don't need a Phd in astrophysics any more than you need * to ctalk with your mother so why go through the pain?`
`01:15.51`	`misyel`	`I'll have all my relatives connect from my server instead getting from mymplus`
`01:16.01`	`lanning`	`misyel, do you want to learn telecommunications or do you want to talk to your mom?`
`01:16.09`	`[TK]D-Fender`	`misyel: Do you need to take a course in electrical engineering so you can learn to build a cell tower, or is it better to just by a stupid cell phone?`
`01:16.12`	`carrar`	`I wanna talk to his mom`
`01:16.18`	`carrar`	`sounds HOT++`
`01:16.21`	`lanning`	`heh`
`01:16.36`	`dacs`	`[TK]D-Fender: so from your experience , which sip provider is low cost for inbound and outbound. i mean do you recommend one`
`01:16.50`	`[TK]D-Fender`	`dacs: Depends on your calling needs`
`01:17.12`	`misyel`	`well my parent is paying 60 usd / month for our voips`
`01:17.33`	`dacs`	`[TK]D-Fender: just for testing purpouses`
`01:17.34`	`[TK]D-Fender`	`misyel: And I gave you the website name for a free service TWICE.`
`01:17.45`	`dacs`	`for right now [TK]D-Fender`
`01:17.57`	`[TK]D-Fender`	`dacs: what does "testing" tell you?`
`01:18.25`	`dacs`	`nothing [TK]D-Fender`
`01:19.30`	`[TK]D-Fender`	`dacs: Then pick anyone, it doesn't really matter, does it?`
`01:20.15`	`misyel`	`why use asterisk if we can have ekiga.net?`
`01:20.42`	`lanning`	`asterisk is for those that need a pbx`
`01:20.45`	`[TK]D-Fender`	`dacs: if you just want to terminate a call to the PSTN to say "yay I did it... don't really have a need, I just wanna say that I did it", then there are providers that will terminate calls to US toll-free for free.`
`01:21.12`	`[TK]D-Fender`	`misyel: Because * is a lot MORE than that and you don't seem to understand why a person would want to run their own server`
`01:22.38`	`[TK]D-Fender`	`misyel: You know how a company's PBX (you'd better have a clue what a PBX is after that last description of mine) allows each 'extension" to have their own voicemail box? And that when you call a company they can have a fancy auto-attendant? And do other funky shit? Well * can do all that and MORE.`
`01:22.48`	`[TK]D-Fender`	`misyel: But this isn't what you asked to do.`
`01:23.28`	`[TK]D-Fender`	`misyel: Yes you can put your groceries in a car to bring them home OR you could just ask the damn clerk for a plastic bag and be done with it.`
`01:23.53`	`dacs`	`i know you told me this earlier, and you said its easy! but i also want to make sure to call my * and test that calling list i was telling you about earlier.`
`01:24.05`	`[TK]D-Fender`	`misyel: You don't need a car to carrly an armload's worth of groceries`
`01:24.35`	`[TK]D-Fender`	`dacs: Calling list? No recollection of what you're referring to..`
`01:24.47`	`carrar`	`You can hire people outside of home depot to carry your groceries!!`
`01:24.57`	`[TK]D-Fender`	`dacs: But again, for what little you intend to use you could pick just about anybody.`
`01:25.06`	`[TK]D-Fender`	`dacs: Here, just pick one :`
`01:25.09`	`[TK]D-Fender`	`~itsplist-us`
`01:25.10`	`infobot`	`[itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net`
`01:25.46`	`[TK]D-Fender`	`carrar: Yes, that would be like hiring one of us to set up * for his so he can OWN his own equivalent to ekiga.net`
`01:26.33`	`[TK]D-Fender`	`misyel: And as I mentioned, you don't even need a central server, you can dial by IP. Or you could just go and install SKYPE and be done with it.`
`01:28.18`	`*** join/#asterisk Pwn-BoFH (n=Coto@pc-86-7-239-201.cm.vtr.net)`
`01:31.32`	`dacs`	`[TK]D-Fender: i was referring to the church and the priest recording a message and * will send it to a distribution list!`
`01:32.41`	`[TK]D-Fender`	`dacs: Well that sounds like you have some kind of requirements that make the cost effectiveness of your provider an important factor`
`01:34.51`	`dacs`	`[TK]D-Fender: try cheap :)`
`01:35.58`	`misyel`	`how come mymplus voip able to call to landline/cellphone? if I use asterisk, will I be able to call my mother to her landline?`
`01:36.12`	`[TK]D-Fender`	`dacs: Go shop because you'll have to consider how many simultansous calls you want to place, to where, how long per call, etc`
`01:36.52`	`lanning`	`misyel, for that you will need to purchase a termination service.`
`01:37.00`	`lanning`	`~itsp`
`01:37.00`	`infobot`	`[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs`
`01:37.27`	`[TK]D-Fender`	`misyel: becasue THEY let you call a land-line number because THEY have the eqipment to. They provide you SERVICE. * does NOT. Is is jsut like buying an answering machine, you don't get the stupid PHONE LINE for free just because you own an answering machine`
`01:38.10`	`misyel`	`~itsplist-us`
`01:38.10`	`infobot`	`somebody said itsplist-us was Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net`
`01:38.12`	`[TK]D-Fender`	`misyel: You can use their service WITH * is you want, but McDonalds won't give me food for free jsut because I want to put it in my bag, and not theirs`
`01:38.47`	`dacs`	`[TK]D-Fender: do you know magic jack usb usb -ata with voip sp`
`01:39.25`	`*** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr)`
`01:39.51`	`carrar`	`harro`
`01:40.32`	`[TK]D-Fender`	`dacs: Trying to use MagicJack with * is a violation of their terms of service and willg et you ejected once caught and possibly worse`
`01:41.53`	`dacs`	`opps wanna stay away from that then, i was just thinking to test with it`
`01:42.30`	`[TK]D-Fender`	`dacs: again test doesn't mean anything. If you're looking for a free ride you are largely wasting your time.`
`01:42.49`	`[TK]D-Fender`	`dacs: Are you desperately looking for a way to prove your planned system to them?`
`01:43.22`	`[TK]D-Fender`	`dacs: Sure looks like it. Either that or you're jsut looking for a free ride.`
`01:44.32`	`dacs`	`[TK]D-Fender: NO, its just taking so long, and i am really busy ! but you are right, i will just get me a provider pay $20 /month better than doing some stupid and getting caught :)`
`01:45.07`	`[TK]D-Fender`	`dacs: Now since when has the thought that you might be doing somethign stupid ever slowed you down? ;)`
`01:45.07`	`drmessano`	`Youre selling someone a PBX?`
`01:46.10`	`dacs`	`i barely can setup my sip.cong....you expect me to sell a whole PBX drmessano rotflmao`
`01:46.25`	`drmessano`	`Actually, yes I do..`
`01:46.28`	`drmessano`	`Sadly..`
`01:47.03`	`dacs`	`i could careless what you think .. i hope you know that`
`01:47.04`	`[TK]D-Fender`	`Delusions the likes of which imaginary pink unicorns are made of...`
`01:47.06`	`dacs`	`;)`
`01:47.47`	`drmessano`	`I could care less what you think, dacs.. and with you coming in every 4 months asking the same questions, I can even care less before you know I care less.`
`01:49.28`	`dacs`	`you know drmessano the is an option called '/ignore <nick name> ...i will make it easy for you ...copy and paste this please /ignore dacs`
`01:49.52`	`[TK]D-Fender`	`this please /ignore dacs`
`01:50.12`	`drmessano`	`If you were an application, you would be a predictive disconnected number dialer`
`01:51.08`	`dacs`	`and if you think you are funny, you are not even the same zipcode a funny`
`01:51.13`	`[TK]D-Fender`	`dacs: ... doesn't work!`
`01:51.30`	`dacs`	`lol [TK]D-Fender`
`01:51.37`	`[TK]D-Fender`	`dacs: I just pasted "this please /ignore dacs" and I still see you!`
`01:51.46`	`drmessano`	`dacs: maybe in a few years of not learning asterisk you would learn what those 4 words mean together`
`01:51.57`	`drmessano`	`dacs: Its ok though, we understand`
`01:52.03`	`drmessano`	`nods and smiles`
`01:53.02`	`dacs`	`quotes " I understand ...said the dumb fuck, to his stupid friend"`
`01:53.40`	`drmessano`	`In that scenario, you are both?`
`01:53.50`	`drmessano`	`or all three?`
`01:54.23`	`drmessano`	`lost count after taking 2 months to not unlock a PAP2`
`01:54.24`	`[TK]D-Fender`	`drmessano: Sure is crowded up there...`
`01:54.29`	`drmessano`	`Wait, sorry.. that was you`
`01:54.50`	`*** join/#asterisk twisted (n=twisted@m205e36d0.tmodns.net)`
`01:54.50`	`*** mode/#asterisk [+o twisted] by ChanServ`
`01:55.06`	`twisted`	`OMG`
`01:55.15`	`brookshire`	`hi`
`01:55.16`	`dacs`	`dude you are distracting me , but to make you happy, no i don't sell * . it is true since the last time my box crashed and i got busy and alot of shit happen .`
`01:55.17`	`drmessano`	`ZWTF?`
`01:55.33`	`twisted`	`OMGHI`
`01:55.55`	`drmessano`	`ZOMG TORRENT PLZ!!!!`
`01:56.04`	`twisted`	`torrent for...?`
`01:56.22`	`carrar`	`LIBPRI`
`01:56.28`	`drmessano`	`Wait, isnt OMGHI the new Digiu... oh nevermind..`
`01:56.33`	`twisted`	`to do what?`
`01:56.51`	`carrar`	`to make FREE CALLS!!`
`01:56.53`	`brookshire`	`are you trying to torrent asterisk?`
`01:57.00`	`drmessano`	`OMGHI sounds like a great name for the Skype app`
`01:57.15`	`brookshire`	`no.. that is OMGFAIL`
`01:57.22`	`twisted`	`no, the skype app is ZOMGHAIFAIL`
`01:57.23`	`drmessano`	`Has the whole DAHDI, DUNDI thing going on, and sums up Skype in 2 awesome words`
`01:58.12`	`twisted`	`i haven't been here in 1.4-ever`
`01:58.31`	`brookshire`	`me either, what are even up to anyways?`
`01:58.32`	`drmessano`	`dacs, please do not PM me.. I will not have kinky wild mansex with you`
`01:58.49`	`twisted`	`hmmm`
`01:58.54`	`twisted`	`1.4.26-rc1`
`01:58.59`	`twisted`	`1.6.99999999999`
`01:59.02`	`twisted`	`shrug`
`01:59.10`	`brookshire`	`1.fail`
`01:59.16`	`engien`	`im trying to follow the asterisk install guide for centos in the oreilly book. when i do make on zaptel, it says kernel source not found. any suggestion`
`01:59.21`	`dacs`	`drmessano: you are basterd :)`
`01:59.26`	`twisted`	`download the kernel source`
`01:59.36`	`drmessano`	`dacs: You are gud speeler`
`01:59.51`	`engien`	`its downloaded and in /usr/src/kernels/2.6.18-128.1.14.el5-i686`
`02:00.03`	`brookshire`	`yum search kernel-headers`
`02:00.07`	`twisted`	`you probably should link it to /usr/src/linux then`
`02:00.22`	`drmessano`	`yum install kernel-devel would do it for you`
`02:00.30`	`twisted`	`brb i need moar bear`
`02:00.32`	`twisted`	`*beer`
`02:00.34`	`engien`	`did yum install kernel-devel`
`02:00.38`	`engien`	`make still says no source headers`
`02:00.43`	`drmessano`	`uname -r`
`02:00.54`	`engien`	`2.6.18-128.el5`
`02:01.03`	`engien`	`so wrong v ?`
`02:02.14`	`drmessano`	`I wouldnt think so`
`02:02.29`	`drmessano`	`you did a configure?`
`02:02.45`	`engien`	`yeah`
`02:03.00`	`drmessano`	`and then make fails`
`02:03.04`	`twisted`	`Darwin 94.28.208.25.in-addr.arpa 10.0.0b1 Darwin Kernel Version 10.0.0b1: Fri May 29 00:02:02 PDT 2009; root:xnu-1456~1/RELEASE_I386 i386 i386`
`02:03.05`	`twisted`	`er`
`02:03.09`	`twisted`	`oops`
`02:03.17`	`engien`	`make says 'you do not appear to have the sources for the 2.6.18-128.el5 kernel isntalled'`
`02:03.28`	`carrar`	`2.6.18-128.el5, centos 5.3?`
`02:03.30`	`brookshire`	`just install asterisknow :)`
`02:03.35`	`engien`	`yes`
`02:03.44`	`twisted`	`cat /var/lib/asterisk/mohmp3/fpm-calm-river.mp3 > /dev/sda`
`02:03.46`	`carrar`	`2.6.30!!`
`02:03.48`	`twisted`	`DO NOT DO THAT`
`02:04.02`	`drmessano`	`2.6.18-128.1.10.el5`
`02:04.33`	`engien`	`?`
`02:04.39`	`*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)`
`02:04.41`	`twisted`	`ew asterisknow`
`02:04.50`	`twisted`	`oh wait`
`02:04.55`	`twisted`	`i should be punished for that statement`
`02:04.55`	`drmessano`	`engien: Seems like I needed to reboot last time I had that problem`
`02:04.57`	`*** mode/#asterisk [-o twisted] by twisted`
`02:05.16`	`engien`	`tries`
`02:05.51`	`drmessano`	`Im not sure of the correct answer.. but the sources didnt match the kernel I was using.. i rebooted to the new kernel, and bam.. or something like that or whatever or kinda or dunno`
`02:05.51`	`*** mode/#asterisk [+o [TK]D-Fender] by ChanServ`
`02:06.00`	`*** mode/#asterisk [+o twisted] by [TK]D-Fender`
`02:06.06`	`twisted`	`d'oh`
`02:06.06`	`[TK]D-Fender`	`twisted: I forgive you :)`
`02:06.23`	`[TK]D-Fender`	`goes to heal the sick`
`02:06.39`	`drmessano`	`points to dacs`
`02:06.42`	`brookshire`	`trixbox?`
`02:06.42`	`twisted`	`drinking + IRC = fun`
`02:06.57`	`twisted`	`OMG I do not wish trixbox on my worst enemy`
`02:07.27`	`drmessano`	`thinks the foncore asterisk binaries really rock`
`02:07.37`	`jaytee`	`actually, I wish an entire cluster of trixbox servers on my worst enemy`
`02:07.48`	`[TK]D-Fender`	`twisted: I would :) See as far as I know Trixbox won't actually kill you, just strip your soul to the bone.... that's why I don't give out death threats anymore.... I give out LIFE THREATS`
`02:07.54`	`twisted`	`hahaha`
`02:07.56`	`drmessano`	`~happyclownpbx`
`02:07.56`	`infobot`	`[HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone`
`02:08.12`	`twisted`	`awww... is that what happened to jbot?`
`02:08.24`	`drmessano`	`Same bot, new nick`
`02:08.29`	`twisted`	`ah`
`02:08.31`	`twisted`	`~twisted`
`02:08.31`	`infobot`	`hmm... twisted is toastido@gmail.com, for paypal and chatter. He is also known in some circles as toastido.`
`02:08.36`	`engien`	`hm, reboot did not fix it`
`02:08.59`	`drmessano`	`engien: Yum update`
`02:09.30`	`engien`	`running`
`02:10.29`	`drmessano`	`engien: maybe some of dacs dumbassedness rubbed off on you.. if so, I apologize`
`02:10.40`	`twisted`	`btw`
`02:10.47`	`twisted`	`snow leopard for mac > your mom`
`02:10.56`	`brookshire`	`no`
`02:11.11`	`engien`	`been a long time since ive worked in linux`
`02:12.02`	`drmessano`	`engien: youre fine.. Ive never actually used linux.. I am here for the LULZ and popcorn`
`02:12.53`	`[TK]D-Fender`	`drmessano: You got popcorn? All I got was this lousy T-Shirt!`
`02:14.05`	`drmessano`	`[TK]D-Fender: yes, but I had to compile the kernels myself..`
`02:14.30`	`drmessano`	`[TK]D-Fender: Felt like I was using debian`
`02:14.56`	`[TK]D-Fender`	`drmessano: http://www.zyra.org.uk/os-air.htm`
`02:15.56`	`drmessano`	`Ubuntu: Big mac + fries ... Debian: A cow, flour, dirt, some bees, potato sprout, and some water binaries.`
`02:15.57`	`twisted`	`~bkw`
`02:15.57`	`infobot`	`bkw is, like, wants to eat file's muffin`
`02:16.06`	`twisted`	`yep. that's jbot`
`02:16.14`	`drmessano`	`~drmessano`
`02:16.15`	`infobot`	`[drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily`
`02:17.24`	`engien`	`yum update didnt fix it.. but yum update + reboot did. thanks`
`02:17.52`	`engien`	`one step closer to getting this going`
`02:18.11`	`*** topic/#asterisk by twisted -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.26rc1 (2009/06/18), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits`
`02:18.35`	`*** join/#asterisk Deeewayne (n=dwayne@76.29.245.9)`
`02:18.35`	`*** mode/#asterisk [+o Deeewayne] by ChanServ`
`02:19.38`	`[TK]D-Fender`	`twisted: We don't tend to really put RC's in the topic...`
`02:19.58`	`twisted`	`oh. hah`
`02:20.15`	`*** topic/#asterisk by twisted -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.25 (2009/06/05), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits`
`02:20.35`	`twisted`	`fied`
`02:20.37`	`twisted`	`*fixed`
`02:20.42`	`[TK]D-Fender`	`twisted: Gives newbs the impression its a good idea to run in production and other sillyness.`
`02:20.56`	`twisted`	`but...isn't it!?`
`02:21.15`	`drmessano`	`RC = really close`
`02:21.28`	`path`	`ahaha`
`02:21.34`	`drmessano`	`B = broken`
`02:21.44`	`[TK]D-Fender`	`twisted: Its just a point of view. Now technically 1.6.2.0 hasn't been released yet and there are those wanting to try the brach, then that RC would be as viable a release as the last buggy RC... so go for it :)`
`02:21.51`	`ISO9001`	`I asked this earlier but I'm trying again -- I'm using followme in my asterisk setup, but I want a ring tone instead of musiconhold. Is there any way I can do that short of recording an mp3 of ring tones? Playtones doesn't seem to work.`
`02:21.54`	`twisted`	`yeah.. i didn't add 1.6.2 because it's beta`
`02:21.57`	`[TK]D-Fender`	`twisted: But I wouldn't recommend an RC from a stable branch`
`02:22.28`	`twisted`	`heh, nah`
`02:22.36`	`twisted`	`i'm not around enough to assume responsibility`
`02:22.40`	`[TK]D-Fender`	`ISO9001: make an MoH class with 1 sound recording in it of ringing`
`02:23.55`	`misyel`	`ekiga runs on windows?`
`02:24.08`	`twisted`	`ew windows`
`02:24.09`	`[TK]D-Fender`	`misyel: Yes`
`02:24.20`	`[TK]D-Fender`	`twisted: You don't want in on this.... trust me...`
`02:24.32`	`twisted`	`haha`
`02:24.36`	`ISO9001`	`[TK]D-Fender: yeah.. trying to avoid that, audio quality on MoH isn't always great though, audio quality on playtones has been perfect. If that's the only way though, I gues that's what I've got to do.`
`02:24.39`	`ISO9001`	`guess, even.`
`02:25.01`	`[TK]D-Fender`	`ISO9001: Sorry, but audio is audio.....`
`02:25.36`	`[TK]D-Fender`	`ISO9001: ISO9001 Now if your recording sucks and you're transcoding well... brains are in a bin on the left ;)`
`02:25.49`	`ISO9001`	`haha.`
`02:26.06`	`ISO9001`	`neither of those SHOULD be the problem, but I'll check.`
`02:26.07`	`ISO9001`	`thanks.`
`02:26.55`	`[TK]D-Fender`	`ISO9001: Bits thrown down a pipe dont care what generated them. Only reason Playtones would be any different is because your recording's transformation into what is being sent isn't the same`
`02:32.17`	`Docteh`	`what format is your recording in and what format is the call in?`
`02:37.54`	`Docteh`	`I wonder how hard it'd be to run voicemailmain with cepstral so that "message 2 recieved at 4:50pm" has better flow :-/`
`02:41.22`	`misyel`	`where to get OPAL and PTLIB?`
`02:42.21`	`[TK]D-Fender`	`misyel: Depends what you are getting them for`
`02:44.25`	`misyel`	`I downloaded ekiga-3.2.4.tar.gz`
`02:44.51`	`[TK]D-Fender`	`misyel: For what OS?`
`02:44.59`	`misyel`	`windows`
`02:45.01`	`misyel`	`xp`
`02:45.16`	`[TK]D-Fender`	`misyel: that pacgake isn't FOR windows.`
`02:45.23`	`[TK]D-Fender`	`package*`
`02:45.25`	`[TK]D-Fender`	`dangit`
`02:46.06`	`misyel`	`I don't see any other package`
`02:46.35`	`[TK]D-Fender`	`misyel: http://wiki.ekiga.org/index.php/Windows_Users`
`02:46.42`	`[TK]D-Fender`	`misyel: Didn't look hard enough`
`02:47.43`	`misyel`	`yeah, thought what I thought...well, I don't see it from their mainsite but anyways, thank you.`
`02:47.57`	`[TK]D-Fender`	`misyel: I drilled it off of their main-site`
`02:48.38`	`[TK]D-Fender`	`"Release binaries are available below. " <-`
`02:48.43`	`[TK]D-Fender`	`on their downloads page`
`02:48.48`	`Docteh`	`aha! surfing the internet IS a skill!`
`02:49.13`	`[TK]D-Fender`	`"download binary Ekiga releases " <- Hmmmm`
`02:49.31`	`[TK]D-Fender`	`"Note that there is a different page to download for Windows." <- HMMMMMM`
`02:49.55`	`Docteh`	`well that is kinda vague`
`02:50.08`	`Docteh`	`i'm going to email them about that even though I dont use ekiga`
`02:50.10`	`[TK]D-Fender`	`Docteh: that text is a LINK on the page <-`
`02:50.26`	`[TK]D-Fender`	`Docteh: Which leads to the "OMG you FOUND ME!?!?!"`
`02:51.40`	`Docteh`	`huh maybe it doesn't work in chrome`
`02:51.49`	`misyel`	`well, I still don't se it`
`02:51.52`	`[TK]D-Fender`	`Docteh: Links?`
`02:52.01`	`[TK]D-Fender`	`misyel: Don't see what?`
`02:52.48`	`misyel`	`a link for windows`
`02:53.02`	`Docteh`	`I see the link to http://snapshots.ekiga.net/ and http://ekiga.org/index.php?rub=5&path=sources/ekiga_3.2.1 ideally they should have a link to thier wiki`
`02:53.10`	`Docteh`	`they're not as bad as bluez.org though`
`02:53.34`	`[TK]D-Fender`	`misyel: http://www.ekiga.org/ <- look for the SERIES of links to other pages I just C&P'd.`
`02:53.47`	`[TK]D-Fender`	`misyel: And I linked you to the end result already`
`02:54.16`	`[TK]D-Fender`	`Docteh: Apparently you can't read my C&P and find the text ont he page either.`
`02:54.46`	`[TK]D-Fender`	`thinks there is a "web-browsing-temporary blindness" disease going around these days...`
`02:54.49`	`Docteh`	`well`
`02:54.51`	`[TK]D-Fender`	`contacts the WHO`
`02:55.01`	`[TK]D-Fender`	`Nope, not N1H1`
`02:55.05`	`[TK]D-Fender`	`Wait....`
`02:55.09`	`[TK]D-Fender`	`BREAKING NEWS!`
`02:55.10`	`misyel`	`here's what I got from download page >> www.ungab.us/ekiga.jpg`
`02:55.28`	`[TK]D-Fender`	`It's been identified as "ID ten T"`
`02:56.17`	`layne`	`issue lies between keyboard and chair =p`
`02:56.19`	`[TK]D-Fender`	`misyel: http://wiki.ekiga.org/index.php/Windows_Users <--- I gave you the precise page full of F-ing EXE's.`
`02:56.35`	`Docteh`	`well yea`
`02:56.45`	`misyel`	`I know, but you said to me that you can find it in their mainsite download page duh`
`02:57.20`	`[TK]D-Fender`	`misyel: [22:48]<[TK]D-Fender>"Release binaries are available below. " <- <-- I said find this text in the middle of their main page. then follow the NEXT line I gave you/.`
`02:57.29`	`[TK]D-Fender`	`misyel: Gah`
`03:02.23`	`[TK]D-Fender`	`Yup, some people you can't just hand answers to on a silver platter. No you have to spoon feed it to them one spoonful at a time prying their moths open with the jaws-of-life and forcing them to swallow.`
`03:21.19`	`jblack`	`[TK]D-Fender: If you spoonfeed, then it's with a spoon coated with diamond encrusted sandpaper.`
`03:21.21`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
`03:23.06`	`jblack`	`You're a nice guy, you help a lot. But sometimes you get so worked up, that if you had my age and weight, you would be dead of a heart attack, stroke, or both at the same time`
`03:25.42`	`[TK]D-Fender`	`Cool my final argument would be enacted through dualing coroner's reports!`
`03:25.44`	`[TK]D-Fender`	`AWESOM!`
`03:25.51`	`[TK]D-Fender`	`+E`
`03:26.15`	`[TK]D-Fender`	`jblack: So was that slamming my stree or your physical state? ;)`
`03:26.23`	`[TK]D-Fender`	`</selfdeprecation>`
`03:26.33`	`[TK]D-Fender`	`Stress*`
`03:31.36`	`freenose`	`Is there something wrong with this? http://pastebin.com/m5d332c6a , when dialing out I get les.net message saying number doesn't exists, where receiving a call a get the error at the bottom of the paste`
`03:31.48`	`freenose`	`I'm using 1.6.1 + bria`
`03:32.20`	`*** join/#asterisk rvhi (n=chatzill@207.2.110.6)`
`03:32.33`	`rvhi`	`hi, anyone knows how to transfer a call on linksys pap2t?`
`03:32.53`	`freenose`	`configuring the les.net account directly on bria works ok`
`03:34.22`	`[TK]D-Fender`	`freenose: Look what * is dialing : 174.143.27.77 <-- this is not a phone number, it is an IP address. And you showed us the error message and not the line that GENERATED it.`
`03:34.54`	`[TK]D-Fender`	`freenose: And you really should have SIP DEBUG enabeld at * CLI to see whats really going on.`
`03:36.36`	`freenose`	`Ok, I'm just in chapter four of the book, trying to get basic things working, what error message do you refer to?`
`03:37.38`	`[TK]D-Fender`	`[Jun 20 03:26:25] NOTICE[21149]: chan_sip.c:18160 handle_request_invite: Call from 'lesnet_peer' to extension '174.143.27.77' rejected because extension not found.`
`03:38.12`	`[TK]D-Fender`	`freenose: we also don't see your register statement in sip.conf (mask only passwords please)`
`03:38.39`	`[TK]D-Fender`	`freenose: Go enable SIP debug, try another call and pastebin it and include your register statement`
`03:39.50`	`freenose`	`[TK]D-Fender: I'm not using a register statement cause I chose public IP instead of register in the les.net trunk, should I chose register for the trunk?`
`03:39.56`	`*** join/#asterisk blkry (n=blkry@96.37.27.72)`
`03:40.18`	`[TK]D-Fender`	`freenose: Well you filled in yout IP on their webpage where they asked you for the number to dial.`
`03:40.56`	`[TK]D-Fender`	`freenose: Look for that field and replace it with teh DID they provide you.`
`03:41.48`	`freenose`	`[TK]D-Fender: well peer address must be my IP`
`03:42.20`	`freenose`	`[TK]D-Fender: not my DID, I think...`
`03:42.45`	`freenose`	`ok finally got SIP debug on ;)`
`03:43.15`	`[TK]D-Fender`	`freenose: You likely filled in an IP in 2 fields, only needing 1.`
`03:43.31`	`[TK]D-Fender`	`freenose: SIP debug will confim more for you`
`03:45.11`	`freenose`	`[TK]D-Fender: hmm that's alot of output heh, I'm on screen, how do you copy all that?`
`03:45.31`	`[TK]D-Fender`	`freenose: Sorry, can't help you there...`
`03:46.02`	`freenose`	`[TK]D-Fender: Ok, one sec`
`03:46.21`	`freenose`	`fires man screen`
`03:53.36`	`freenose`	`[TK]D-Fender: http://pastebin.com/m659fabeb`
`03:53.44`	`freenose`	`that's for the outgoing call`
`03:55.21`	`[TK]D-Fender`	`freenose: Looks OK, and you cancel the call.`
`03:56.14`	`freenose`	`[TK]D-Fender: yeah, the les.net message show up`
`03:56.50`	`[TK]D-Fender`	`freenose: what message? Where?`
`03:58.06`	`freenose`	`[TK]D-Fender: voicemessage, 'you have reach a les.net that is currently not in service, you can own this number visiting les.net'`
`03:58.22`	`freenose`	`[TK]D-Fender: I got the number from the net, is a hotel`
`03:58.28`	`[TK]D-Fender`	`freenose: then they accept your call and your config does not have an issue`
`03:58.51`	`[TK]D-Fender`	`freenose: Call ANOTHER number`
`04:00.01`	`freenose`	`[TK]D-Fender: same with all numbers, just tried 1-800-my-apple`
`04:00.08`	`freenose`	`:/`
`04:00.32`	`[TK]D-Fender`	`freenose: Or perhaps your setup of your account is short on funds, or otherwise incompletely activated on their side`
`04:00.51`	`[TK]D-Fender`	`freenose: because the audio shows they accept the call to start which doe not indicate any failure on *'s side`
`04:02.18`	`freenose`	`[TK]D-Fender: Ok, let me try something`
`04:06.18`	`freenose`	`[TK]D-Fender: the les.net account works directly on bria, weird, I'm gonna set up the peer for asterisk to register as like the bria one, instead of public IP`
`04:06.43`	`jplank`	`is fxotune for dahdi different?`
`04:09.01`	`jplank`	`ahhh using the older one for zaptel :)`
`04:12.04`	`ISO9001`	`[TK]D-Fender: tried out your suggestion from before, ulaw ringtone file on a ulaw connection... but the sound is 'cracking' when voice doesn't. Any idea why that would be?`
`04:12.46`	`[TK]D-Fender`	`ISO9001: Can you get it to not crackle on a local subnet call?`
`04:14.25`	`freenose`	`same issue setting the trunk to register but incoming works now!`
`04:14.58`	`[TK]D-Fender`	`freenose: You just said you weren't registering...`
`04:15.14`	`*** join/#asterisk luckyaba (n=Lucky@ip72-194-215-55.sb.sd.cox.net)`
`04:16.22`	`freenose`	`[TK]D-Fender: yeah but I change the les.net trunk to register, change the peer type`
`04:16.43`	`[TK]D-Fender`	`freenose: freenose And I still don't see a REGISTER statement....`
`04:16.44`	`ISO9001`	`[TK]D-Fender: don't have access to the local subnet. I'm calling it from a cell, the machine in question is in a datacenter halfway across the country.`
`04:17.31`	`[TK]D-Fender`	`ISO9001: Over a cell.... to a tower, then through X amount of interconnection.. then to your server through its internet conenctio + delay + jitter... colour me UNSURPRISED`
`04:18.11`	`freenose`	`[TK]D-Fender: I just added is not on the paste, 'register => user:pass@did.voip.les.net/user'`
`04:18.15`	`[TK]D-Fender`	`ISO9001: if direct voice is better it could be that * is not JB'ing the call wheras the clients are covering it up`
`04:18.36`	`[TK]D-Fender`	`freenose: Ok, well feel free to show new debug & configs.`
`04:18.52`	`ISO9001`	`[TK]D-Fender: haha. My server is ~2ms from the provider's. I don't think that (particular part) is the issue. What I don't get is that it doesn't affect voice. When I'm on a call I don't crackle.`
`04:19.27`	`freenose`	`[TK]D-Fender: kk`
`04:19.43`	`ISO9001`	`hrm.`
`04:20.17`	`[TK]D-Fender`	`ISO9001: Well you haven't proven the recording is intact yet or how a local * would play it back`
`04:20.40`	`ISO9001`	`intact?`
`04:21.35`	`[TK]D-Fender`	`ISO9001: "not a busted up POS"`
`04:21.59`	`ISO9001`	`haha.`
`04:22.08`	`ISO9001`	`fair enough.`
`04:22.28`	`ISO9001`	`it's just a 440+480 tone generated with sox though, there's nothing inherently special about it.`
`04:22.56`	`ISO9001`	`let me see if I can test it out on a local asterisk setup...`
`04:23.13`	`[TK]D-Fender`	`ISO9001: sorry, only "reality" will really cut it`
`04:24.04`	`ISO9001`	`beg pardon?`
`04:31.58`	`ISO9001`	`plays fine on local asterisk, plays very wrong on remote... oddly, not just crackling, the delay between rings is too long as well. I guess I've broken something else somewhere. Ah well. Thanks again for your help.`
`04:32.57`	`[TK]D-Fender`	`ISO9001: Got to be network conditions / cell issues`
`04:34.20`	`ISO9001`	`happening on two different cell providers, so I would guess network, but that shouldn't be possible.`
`04:34.40`	`ISO9001`	`I mean, it's in literally the same cage as the provider.`
`04:34.51`	`jplank`	`anyone have an opinion on rockbochs?`
`04:35.07`	`freenose`	`[TK]D-Fender: sip debug, sip.conf, extensions.conf - http://pastebin.com/m41d3eba0`
`04:37.12`	`freenose`	`[TK]D-Fender: Look at line 36,37 - is that ok?`
`04:37.38`	`freenose`	`37 mostly`
`04:38.03`	`shido6`	`how do you use regular expressions to eat the last 3 digits of a column of 3000 numbers if they look like 1204936 and all the numbers are different. The only thing that is similar is the amount of digits which is 7`
`04:38.14`	`[TK]D-Fender`	`freenose: Yes. The initial call comes in without auth, but is ID'd & challenged. They try again with auth and is accepted`
`04:39.11`	`*** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net)`
`04:40.02`	`freenose`	`hmm ok, I don't why the same peer works ok directly in bria`
`04:40.12`	`jplank`	`shido are you talking sql?`
`04:40.45`	`shido6`	`nope`
`04:41.19`	`jplank`	`your talking ASTDB?`
`04:41.43`	`*** join/#asterisk mcneelycorp (n=chatzill@c-67-175-45-48.hsd1.in.comcast.net)`
`04:41.54`	`shido6`	`if it were mysql how would I eat the last 3 digits ?`
`04:42.00`	`[TK]D-Fender`	`freenose: Well.. your lesnet peer has no username or secret specified which usualy sounds like it should lead to certain failure.`
`04:42.14`	`[TK]D-Fender`	`freenose: You should probably fill those in based on your REGISTER`
`04:42.23`	`jplank`	`shido your not giving enough information on what your trying to do`
`04:42.31`	`jplank`	`and what you are using to try and do it`
`04:43.25`	`freenose`	`[TK]D-Fender: Ok, let me try that, I just copy the lesnet peer from the les.net web, that was the asterisk config they show to use`
`04:43.30`	`*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)`
`04:43.43`	`mcneelycorp`	`hi all, i am new to asterisk... i am looking to see if there is a tutorial available on how to user asterisk and what is needed to make a phone verification, on unix and php`
`04:44.03`	`shido6`	`I have 3000 numbers each row has 7 numbers in it I want to remove the last 3 digits. Lets say these numbers are in mysql in table ACME coulmn name lottery_number`
`04:44.27`	`*** join/#asterisk engien (n=mark@c-71-199-107-125.hsd1.pa.comcast.net)`
`04:44.44`	`shido6`	`so an example number is 1234567`
`04:44.58`	`shido6`	`567 must come off`
`04:45.06`	`jplank`	`your still not giving enough information, HOW are you trying to drop digits, what language are you using? not including the fact this is #asterisk not #dba or something like that`
`04:45.10`	`shido6`	`the last 3 digits arent all 567.`
`04:45.46`	`jplank`	`your saying you have a column and using regexp, all I know your trying to do it in excel`
`04:45.49`	`shido6`	`in mysql or using grep and regular expressions or bbedit or vi`
`04:46.17`	`jplank`	`try #mysql or #grep or #regexp or #bbedit or #vi`
`04:46.24`	`shido6`	`thank you`
`04:46.38`	`[TK]D-Fender`	`mcneelycorp: What is a "phone verification", and what does it have to do with "unix" or "php"?`
`04:47.42`	`mcneelycorp`	`[TK]D-Fender: from my research, it seems that you can make a phone verification system with asterisk and php. it is used to verify someone exists. is used on sites like google local, sitepoint, and i think paypal`
`04:47.53`	`jplank`	`shido6: hint: if it is a myswl db, might be easier to use PHP or something like that`
`04:48.05`	`jplank`	`/s/myswl/mysql`
`04:48.28`	`[TK]D-Fender`	`mcneelycorp: That would involve having * call out. Lookup "call files" and "AMI originate" on the WIKI, and in the book`
`04:48.31`	`[TK]D-Fender`	`~wikis`
`04:48.31`	`infobot`	`[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners`
`04:48.33`	`[TK]D-Fender`	`~book`
`04:48.33`	`infobot`	`[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook`
`04:48.48`	`mcneelycorp`	`[TK]D-Fender: you visit a page on a site, put in your phone number. asterisk calls the number and gives a 4 digit code, for example, and the receiver would input that code into the form field...`
`04:49.00`	`jplank`	`sounds pretty simple`
`04:49.26`	`mcneelycorp`	`[TK]D-Fender: sorry to ask... where is the wiki and book?`
`04:49.31`	`jplank`	`ROFL`
`04:49.44`	`mcneelycorp`	`i see now... is above :)`
`04:49.53`	`[TK]D-Fender`	`mcneelycorp: Please look UP and see the obvious BOT responses with equally obvious links & descriptions.`
`04:50.26`	`jplank`	`mcneelycorp: borders might be a good start`
`04:50.41`	`jplank`	`~buybook`
`04:50.41`	`infobot`	`[~buybook] You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY`
`04:51.12`	`mcneelycorp`	`thanks`
`04:52.40`	`carrar`	`ERIOUSLY!!`
`04:52.41`	`mcneelycorp`	`i am surprised there is not some sort of script available for this functionality.. phone verification... sites like www.called.in www.maxmind.org and ttp://www.reducefraud.com all seem to be using this setup`
`04:52.43`	`carrar`	`+S`
`04:54.33`	`*** join/#asterisk kamanashisroy (n=kamanash@119.30.36.74)`
`04:54.41`	`jplank`	`thats the beauty of asterisk, and open source in general, if you can't find it, you can make it`
`04:55.28`	`jplank`	`I believe nerdvittles has a click to dial script that can be easily adapted for that use`
`04:56.34`	`xuser`	`[TK]D-Fender: using 'fromuser=' and 'secret=' in the lesnet peers destroys the call`
`04:57.23`	`mcneelycorp`	`jplank: good to see asterisk sites using drupal... i am a drupal dev`
`04:57.58`	`jplank`	`? nerdvittles uses wordpress`
`04:58.15`	`jplank`	`dont get me wrong, when it comes to CMS' drupal is my favorite`
`04:58.53`	`mcneelycorp`	`yeah... drupal is a developers paradise, once you learn the system`
`05:00.20`	`jplank`	`yea, except the fact that I cant find a prepackaged wedding site type theme, making me build one from the ground up`
`05:01.13`	`[TK]D-Fender`	`xuser: I didn't say "fromuser".`
`05:01.20`	`freenose`	`[TK]D-Fender: what xuser said`
`05:01.35`	`freenose`	`[TK]D-Fender: looks like you can't use that in lesnet`
`05:01.44`	`freenose`	`[TK]D-Fender: what do you mean then?`
`05:02.38`	`[TK]D-Fender`	`[00:42]<[TK]D-Fender>freenose: Well.. your lesnet peer has no username or secret specified which usualy sounds like it should lead to certain failure.`
`05:02.41`	`[TK]D-Fender`	`USERNAME`
`05:02.45`	`[TK]D-Fender`	`wake-up time...`
`05:03.37`	`freenose`	`[TK]D-Fender: you mean in the register line?`
`05:04.09`	`[TK]D-Fender`	`freenose: I told you to fix your peer filling in those 2 values based on your REGISTER`
`05:06.09`	`freenose`	`[TK]D-Fender: there is a 'username' a parameter?`
`05:06.35`	`[TK]D-Fender`	`.........`
`05:06.46`	`freenose`	`heh`
`05:09.58`	`freenose`	`[TK]D-Fender: can you show me a example with this paste: http://pastebin.com/m56af2aa5`
`05:10.21`	`[TK]D-Fender`	`....`
`05:10.25`	`[TK]D-Fender`	`:\|`
`05:10.35`	`[TK]D-Fender`	`username=user`
`05:10.38`	`[TK]D-Fender`	`secret=pass`
`05:11.17`	`[TK]D-Fender`	`tosses freenose's imagination in a burlap sack and beats it senseless with a ClueBat (tm)`
`05:11.22`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)`
`05:13.25`	`Docteh`	`freenose: is your * registering as a peer or are the calls being delivered to a static sip address?`
`05:13.55`	`engien`	`the oreilly book mentions .loads and .sb2 files to put in my tftpboot dir. What are these?`
`05:14.01`	`Docteh`	`whoops may mean to address to someone else`
`05:14.08`	`Docteh`	`engien: what page?`
`05:14.16`	`Docteh`	`tftp is used for settings for phones`
`05:14.18`	`engien`	`page 94`
`05:14.47`	`*** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk)`
`05:15.46`	`Docteh`	`sounds like firmware`
`05:16.05`	`freenose`	`[TK]D-Fender: same issue`
`05:16.09`	`freenose`	`Docteh: I tried with both`
`05:16.19`	`engien`	`i loaded the firmware with the SPxxxx.sbin file. no idea what loads and sb2 are though`
`05:16.36`	`freenose`	`calling a lesnet number works, this gets weirder`
`05:18.16`	`[TK]D-Fender`	`freenose: -- Executing [18006927753@phones:2] Dial("SIP/1000-5c01d138", "SIP/lesnet_peer/18006927753") in new stack`
`05:18.31`	`[TK]D-Fender`	`freenose: [1748142717]`
`05:18.41`	`[TK]D-Fender`	`freenose: Why are you showing me apples & oranges?`
`05:18.45`	`Docteh`	`are you working on inbound or outbound right now?`
`05:18.52`	`freenose`	`outbount`
`05:19.04`	`freenose`	`[TK]D-Fender: what do you mean? that's the user id`
`05:19.37`	`[TK]D-Fender`	`freenose: not the peer you are dialing out!`
`05:20.12`	`Docteh`	`so 17481 is your user id for them?`
`05:20.35`	`freenose`	`yes, to register`
`05:20.40`	`*** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)`
`05:21.08`	`freenose`	`[TK]D-Fender: so you mean to change 'lesnet_peer' with 1748142717`
`05:21.20`	`Docteh`	`ohhhh`
`05:21.28`	`[TK]D-Fender`	`..............`
`05:21.44`	`freenose`	`correct?`
`05:21.59`	`Docteh`	`freenose: [thisy] matches with SIP/thisy/1900sexyhos`
`05:22.59`	`freenose`	`[TK]D-Fender: ah that last paste of sip.conf was wrong, I have [lesnet_peer]`
`05:23.34`	`freenose`	`that was something I tried and forgot to change it before pasting`
`05:26.16`	`freenose`	`the correct one is: http://pastebin.com/m2699ac1a`
`05:30.06`	`freenose`	`[TK]D-Fender: this what lesnet tells me to use: http://pastebin.com/m7ab6e1b9`
`05:46.49`	`*** join/#asterisk mchou (n=quassel@unaffiliated/mchou)`
`05:47.28`	`*** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)`
`05:48.03`	`mchou`	`I'm in the process of upgrading * from 1.4 to 1.6. What's the easiest way to backup and restore the astDB?`
`05:49.29`	`[TK]D-Fender`	`mchou: cp <-`
`05:49.45`	`engien`	`When using SCCP, I could see asterisk rejecting my phone trying to connect. I upgraded to SIP, and I no longer see this. With SCCP, I could see callmanager IP was set to my asterisk server. With SIP, I don't see any setting like that. How do I tell it where the asterisk server is? Does it just use same as tftp server?`
`05:53.41`	`freenose`	`[TK]D-Fender: Ok, this weird, instead of using [1000] for the phone I used a [name] and now it works`
`05:54.06`	`ehsjoar`	`engine: in your tftp directory there should be files like SIPDefault.cnf. This is where the asterisk server is specified (proxy1_address)`
`05:55.29`	`*** join/#asterisk af_ (n=getsmart@88-149-230-49.dynamic.ngi.it)`
`05:56.17`	`ISO9001`	`is there any compelling reason to pick one of the versions in the topic over another? I need to apply a patch anyway.`
`05:59.15`	`*** join/#asterisk GlobeTrotter (n=GlobeTro@201.218.90.155)`
`05:59.31`	`Docteh`	`looks at topic`
`06:00.10`	`Docteh`	`looks like it just lists the newest versions for things`
`06:11.51`	`freenose`	`amazing`
`06:12.38`	`*** join/#asterisk micols (n=mio@rlogin.dk)`
`06:17.10`	`*** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com)`
`06:17.11`	`rue_mohr`	`why does the dahdi driver insist on trying to find /etc/zaptel.conf ?`
`06:20.33`	`Docteh`	`you're probably running an inbetween version?`
`06:21.07`	`rue_mohr`	`the latest stable`
`06:21.16`	`rue_mohr`	`I cant see whats telling it to look there`
`06:21.30`	`Docteh`	`is it the asterisk bit or the kernel module?`
`06:21.37`	`rue_mohr`	`I'm running the same ver on antoher machine and its happy`
`06:21.41`	`rue_mohr`	`its the kernel module`
`06:22.03`	`Docteh`	`huh maybe it compiled wrong`
`06:22.14`	`rue_mohr`	`I just cleaned and recompiled it, same thing`
`06:22.26`	`Docteh`	`strings /the/.ko \|grep zaptel.conf on both comps and see if they both mention it?`
`06:22.53`	`*** join/#asterisk CodeWork (n=Miranda@p5083D495.dip.t-dialin.net)`
`06:23.12`	`Docteh`	`<PROTECTED>`
`06:23.21`	`rue_mohr`	`there is a config menu here somewhere`
`06:23.27`	`engien`	`ehsjoar: thanks`
`06:23.34`	`engien`	`is there something i can type in console to test ring an ext ?`
`06:23.47`	`Docteh`	`you could call it from an extension`
`06:24.00`	`Docteh`	`err from the console like dial 1000@internal`
`06:24.07`	`engien`	`sweet, thanks`
`06:24.27`	`Docteh`	`huh, well i could in 1.4 :-/`
`06:28.46`	`*** part/#asterisk da__d00d (n=Dana@dsl-vlan422-66-18-194-227.nucleus.com)`
`06:32.50`	`freenose`	`[TK]D-Fender: thanks for you help`
`06:32.56`	`freenose`	`bedtime`
`06:34.16`	`rue_mohr`	`ztcfg, -c <filename> -- Use <filename> instead of /etc/zaptel.conf`
`06:34.24`	`rue_mohr`	`what am I missing here?`
`06:34.38`	`rue_mohr`	`why is that the default`
`06:35.01`	`rue_mohr`	`this is from the old driver`
`06:48.36`	`ISO9001`	`Docteh: it does, I'm just wondering if there's a reason to pick one branch over another. Other projects tend to have dev and stable branches, I'm just wondering if I should be avoiding any of those.`
`06:57.35`	`*** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)`
`06:58.10`	`Docteh`	`oh, well the rc's and beta's are considered dev, as well as svn trunk`
`06:59.14`	`ISO9001`	`but 1.4, 1.6.0 and 1.6.1 are considered stable?`
`06:59.48`	`Qwell`	`ISO9001: Those are the current release branches.`
`07:01.40`	`jplank`	`wget http://downloads.digium.com/pub/asterisk/asterisk-1.4-current.tar.gz or 1.6.1 should get him the latest stable right?`
`07:02.02`	`ISO9001`	`heh, yeah, I built 1.6.1, I just want to make sure it's not something I should regret ;)`
`07:03.40`	`Docteh`	`jplank: i'd be leery to download a -current`
`07:03.56`	`jplank`	`thats not latest stable?`
`07:04.02`	`rue_mohr`	`"ERROR[1422] asterisk.c: Asterisk has detected a problem with your DAHDI configuration and will shutdown for your protection" <---grrr, it'd be REALLY nice to know what it thinks the problem IS so I can FIX it!`
`07:04.11`	`Docteh`	`jplank: well its hopefully a link to the latest stable`
`07:04.18`	`jplank`	`hopefully?`
`07:04.36`	`Docteh`	`personally i would take the time to get the one its supposed to link to`
`07:04.45`	`jplank`	`I see what your saying`
`07:04.50`	`Docteh`	`1.4.25?`
`07:04.54`	`jplank`	`yea`
`07:05.32`	`jplank`	`I guess its like a yum -y update`
`07:05.56`	`*** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com)`
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`07:10.52`	`ISO9001`	`rue_mohr: It's not getting a response from /dev/dahdi/timer`
`07:11.17`	`rue_mohr`	`heh :) why cant it just say so`
`07:11.47`	`ISO9001`	`well that's the thing. There's an error case right above it that DOES say that. I'm not sure why they're split into two cases.`
`07:12.35`	`rue_mohr`	`hu, ok it looks like it was my clock source option then, I have phones again`
`07:12.43`	`ISO9001`	`go team.`
`07:12.47`	`ISO9001`	`HALPS.`
`07:12.55`	`rue_mohr`	`I'll be able to use them after changing all my 'Zap' to 'DAHDI'`
`07:12.56`	`rue_mohr`	`:/`
`07:17.27`	`rue_mohr`	`hmm`
`07:18.02`	`rue_mohr`	`powercycling the channelbank always used to fix that gain problem`
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`07:40.48`	`rue_mohr`	`arg, I had it for a sec`
`07:41.02`	`rue_mohr`	`the channelbank is supposed to be the clock source`
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`07:44.06`	`Gnoll`	`hi`
`07:44.35`	`Gnoll`	`I have a problem, I see this error every call: chan_sip.c: 14703 handle_request_invite: Call from''to extension 'MY_TELEPHONE_NUMBER' rejected because extension not found.`
`07:44.36`	`rue_mohr`	`no its not working either way I set the clock source`
`07:46.06`	`rue_mohr`	`its not even working if I use genconf`
`08:01.21`	`rue_mohr`	`ISO9001, it wont go again... so it cant find a clock source?`
`08:01.44`	`rue_mohr`	`what IS /dev/dahdi/timer`
`08:04.08`	`jplank`	`sounds like dahdi's timer`
`08:06.03`	`rue_mohr`	`http://www.pastebin.ca/1467358 <- lets go with the dahdi card as the clock source, look ok?`
`08:06.09`	`rue_mohr`	`cause it wont load it`
`08:07.23`	`rue_mohr`	`the dahdi tools seem to say its just not configured`
`08:07.52`	`rue_mohr`	`no dahdi_cfg -vv is right`
`08:10.12`	`rue_mohr`	`shoudln't dahdi_tool show me its configured if dahdi_cfg -vv shows what it shoudl?`
`08:12.04`	`*** part/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com)`
`08:18.03`	`coolthreads`	`still trying to find a means to the echo i hear from my end the asterisk end, im using a x100p, the other end doesnt hear no echo.`
`08:19.07`	`b14ck`	`coolthreads, x100p isn't a reliable card`
`08:19.31`	`b14ck`	`we can't really help do advanecd troubleshooting with an x100p. it is known to be a crappy card, and not worth using for actual telephony`
`08:19.36`	`b14ck`	`maybe for testing, but that's it`
`08:21.56`	`coolthreads`	`thanks for my first ever reply in here. :)`
`08:22.08`	`b14ck`	`sorry, lol`
`08:22.15`	`b14ck`	`im testing code so not really looking =p`
`08:23.37`	`jplank`	`lol`
`08:25.20`	`jplank`	`rue_mohr: whats the problem your having?`
`08:25.52`	`rue_mohr`	`I'm gradually getting thru them`
`08:26.03`	`jplank`	`upgrade from zaptel to dadhi?`
`08:26.22`	`rue_mohr`	`its working again, I think the key was that I was only reloading wct1xxp and not dahdi`
`08:27.00`	`jplank`	`I dont know what the lock was, so I could only say good job`
`08:27.09`	`rue_mohr`	`:)`
`08:27.24`	`coolthreads`	`anyone know the next best thing to a x100p?`
`08:29.59`	`rue_mohr`	`* you know what REALLY interesting about having gone from the zaptel drivers to the dahdi drivers??? THE ASTERISK GENERATED 1MW IS -11db instead of the 0db is was with the zaptel drivers!!!! *`
`08:30.31`	`rue_mohr`	`the dahdi drivers have an 11db loss somewhere!!!!`
`08:30.45`	`rue_mohr`	`this is serious`
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`08:31.10`	`rue_mohr`	`its common to the wct1xxp driver and the tdm800 driver`
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`08:33.10`	`b14ck`	`coolthreads, sangoma a200 is good`
`08:33.31`	`b14ck`	`coolthreads, your basic choices are: sangoma, rhino, digium, openvox`
`08:34.19`	`coolthreads`	`thats for your advise`
`08:34.23`	`coolthreads`	`thanks`
`08:34.24`	`coolthreads`	`lol`
`08:34.29`	`b14ck`	`no prob, lol`
`08:34.43`	`rue_mohr`	`I'm not crazy, the dahdi drivers are losing 11db!`
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`08:43.14`	`rue_mohr`	`I can almost start to pin this down`
`08:43.44`	`rue_mohr`	`its nothing to do with bridged channels`
`08:44.06`	`rue_mohr`	`it effects sip interfacing, and the playtones`
`08:44.54`	`rue_mohr`	`its common to the wct1xxp and the tdm24xxp drivers`
`08:45.17`	`rue_mohr`	`who can find the missing 11db?`
`08:46.27`	`rue_mohr`	`11db is almost 4 bits`
`08:46.57`	`rue_mohr`	`no sorry, 2 bits`
`08:58.01`	`pa`	`i have a TA ISDN pci.. can i send fax with it with asterisk?`
`08:58.26`	`pa`	`i mean to normal telephones`
`08:59.40`	`*** join/#asterisk chendy (n=chatzill@58.251.102.197)`
`09:19.39`	`pa`	`i mean, without any voip involved`
`09:21.39`	`*** join/#asterisk af_ (n=getsmart@88-149-230-49.dynamic.ngi.it)`
`09:26.16`	`*** join/#asterisk RUMMY (n=RUMMY@212.58.114.85)`
`09:26.53`	`RUMMY`	`Hi therem, Im going to learn Asterisk and which free soft phone can you consult for win32?`
`09:28.02`	`Gnoll`	`I have a problem, I see this error every call: chan_sip.c: 14703 handle_request_invite: Call from''to extension 'MY_TELEPHONE_NUMBER' rejected because extension not found. someone can help me please?`
`09:31.42`	`*** part/#asterisk RUMMY (n=RUMMY@212.58.114.85)`
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`09:54.24`	`b14ck`	`i noticed that in 1.6 the AGI fucntion SAY NUMBER has an optional gender argument. i tried putting in MALE, M, m, and male, but none of those make it read the number in a male voice`
`09:54.35`	`b14ck`	`any ideas what i need to put there to signal a male voice?`
`09:56.20`	`techie`	`Look at the code and find out`
`09:56.38`	`b14ck`	`any idea what file that would be in ? :(`
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`11:32.14`	`xfighter`	`hello`
`11:32.16`	`xfighter`	`what is the command used to reaload the data and update the .conf files from the "asterisk" database and then apply them??`
`11:33.14`	`xfighter`	`anybody?`
`11:34.35`	`*** join/#asterisk VJFROMGT (n=vjfromgt@user-12ldutg.cable.mindspring.com)`
`11:34.46`	`xfighter`	`HELLO`
`11:34.47`	`xfighter`	`?`
`11:35.26`	`VJFROMGT`	`hi, my iax bandwith jumps from 500k to 800k with barely any chnage of total calls`
`11:36.07`	`xfighter`	`everybody is asleep VJFROMGT`
`11:36.16`	`*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)`
`11:36.38`	`VJFROMGT`	`grr`
`11:36.46`	`VJFROMGT`	`sun woke me up`
`11:36.54`	`xfighter`	`hehehehehe`
`11:37.00`	`xfighter`	`lsn I wanna ask u`
`11:37.12`	`VJFROMGT`	`?`
`11:37.34`	`xfighter`	`do u know the command used to reload the data from the asterisk database and then apply them to .conf files and on asterisk in general?`
`11:38.09`	`VJFROMGT`	`the command is reload`
`11:38.25`	`xfighter`	`you mean asterisk reload?`
`11:38.33`	`xfighter`	`no it doesn't work`
`11:38.44`	`xfighter`	`I want to reload the new data from the database`
`11:38.55`	`xfighter`	`then reload asterisk in general`
`11:39.04`	`VJFROMGT`	`from db,,, hmm,, asterisk or freepbx?`
`11:39.11`	`xfighter`	`freepbx`
`11:40.05`	`VJFROMGT`	`hmm, u know these guys will eat you alive when they know ure here for freepbx?`
`11:40.20`	`VJFROMGT`	`give me a few mins,, i will check name of command`
`11:40.33`	`xfighter`	`yea I know :D`
`11:44.01`	`VJFROMGT`	`got it`
`11:44.02`	`VJFROMGT`	`http://www.trixbox.org/forums/trixbox-forums/help/apply-configuration-changes-command-line`
`11:45.52`	`xfighter`	`Thanx :)`
`11:46.00`	`VJFROMGT`	`np`
`11:52.33`	`riddlebox`	`is hungry.......`
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`12:28.47`	`dacs`	`Goof Morning Folks`
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`14:27.26`	`HorizonXP`	`hey guys, i'm trying to get my Linksys PAP2T to work with my public-facing * server. I can get my ekiga softphone to work just fine, but my pap2t is giving me problems.`
`14:28.29`	`HorizonXP`	`i think i've properly configured it for NAT traversal, as it is finding its external IP. It's even sending AND receiving SIP messages, but for its Registration State, it's still saying "Can't connect to login server."`
`14:28.33`	`HorizonXP`	`any ideas?`
`14:30.05`	`[TK]D-Fender`	`HorizonXP: Enable SIP DEBUG on yor * server and look at the traffic`
`14:32.01`	`*** join/#asterisk Chinorro (n=Chino@54.225.117.91.dynamic.mundo-r.com)`
`14:33.32`	`HorizonXP`	`[TK]D-Fender: ok, i see a lot of NAT messages`
`14:33.51`	`HorizonXP`	`they're directed to my IP, and to the LAN IP of the PAP2T`
`14:35.13`	`*** join/#asterisk AsteriskBeginner (n=me@p57970760.dip.t-dialin.net)`
`14:36.34`	`AsteriskBeginner`	`Hello i am new to asterisk! I downloaded the asterisknow iso cd and installed it. Now I only have a black console window. How can I connect to the gui? http://xxx.xxx.xxx.xxx:8088/ does not work. Haven't found a tutorial on the web for this. Please help me..`
`14:38.43`	`[TK]D-Fender`	`HorizonXP: Sorry, GUI's are not supported here. please refer to #freepbx and #asterisknow for support on that`
`14:41.13`	`HorizonXP`	`[TK]D-Fender: i got it working, thanks!`
`14:43.05`	`*** join/#asterisk ingenius (n=alektro@35-214-17-190.fibertel.com.ar)`
`14:43.11`	`AsteriskBeginner`	`i don't necessarily need an gui, is there some guide to configurate asterisknow that the basic stuff works? just receiving & sending calls?`
`14:46.24`	`McL0VIN`	`app_dial.c:1272 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown) i am not using IAX thu ?!!!`
`14:46.38`	`*** join/#asterisk ingenius (n=alektro@35-214-17-190.fibertel.com.ar)`
`14:49.15`	`*** join/#asterisk kamanashisroy (n=kamanash@119.30.36.79)`
`14:51.06`	`McL0VIN`	`AsteriskBeginner: take it from me, you will have to read the book, i battled here with these guys for a year! i thought they are *&^%$ but found they actually want you to benefit. and thats what i am doing right now and i come here to ask question that make sense to them :)`
`14:51.08`	`McL0VIN`	`[TK]D-Fender: it me dacs :P`
`14:51.08`	`AsteriskBeginner`	`do you mean "future of telephony"?`
`14:51.09`	`McL0VIN`	`AsteriskBeginner: yes sir, Once and old fart advised me and said its a guide for you!`
`14:52.00`	`McL0VIN`	`[TK]D-Fender: i setup an account with VoicePluse, and i am able to receive calls but not sending out..`
`14:52.18`	`AsteriskBeginner`	`problem ist, i want to use asterisk with ISDN... haven't found a book on this yet`
`14:52.30`	`AsteriskBeginner`	`problem is,..`
`14:53.47`	`McL0VIN`	`AsteriskBeginner: what you mean by connect to ISDN?!`
`14:54.03`	`AsteriskBeginner`	`well, most of the books concentrate only on VOIP`
`14:54.34`	`AsteriskBeginner`	`but my provider doesn't provide VOIP, only analog or isdn`
`14:54.37`	`AsteriskBeginner`	`i have isdn`
`14:55.05`	`McL0VIN`	`AsteriskBeginner: i can not give you any technical opinion or help as i am still in the first road of learning *, the guy u need to talk to is [TK]D-Fender`
`14:55.26`	`[TK]D-Fender`	`AsteriskBeginner: BRI or PRI?`
`14:55.48`	`rob0`	`AsteriskBeginner: see also /topic, "... Related channels: #asterisknow #asterisk-gui ..."`
`14:56.05`	`McL0VIN`	`AsteriskBeginner: it talk about that in the box, i think that why * was built for :)`
`14:56.11`	`[TK]D-Fender`	`McL0VIN: PASTEBIN is your friend.....`
`14:56.30`	`McL0VIN`	`AsteriskBeginner: FXO n FXS`
`14:56.39`	`AsteriskBeginner`	`i guess bri, i live in europe`
`14:57.57`	`[TK]D-Fender`	`AsteriskBeginner: Ok, interfacing with a PSTN interface like that isn't very difficult. Remeber that * = 90% dialplan, 10% interface configuration`
`14:58.24`	`[TK]D-Fender`	`AsteriskBeginner: FreePBX might not be made to do good configs for a BRI interface.`
`14:59.17`	`McL0VIN`	`AsteriskBeginner: you are lucky man!! [TK]D-Fender is in a good mood today`
`14:59.23`	`McL0VIN`	`runs and hide`
`15:00.28`	`AsteriskBeginner`	`:-)`
`15:01.29`	`tzafrir_laptop`	`BRI? DAHDI or mISDN?`
`15:03.01`	`AsteriskBeginner`	`i am struggling now, which isdn driver to use, any recommendations? (most information on the web seems old, and isdn is not popular in the us, so there's not much support for isdn in general on the web)`
`15:05.04`	`McL0VIN`	`[TK]D-Fender: http://pastebin.ca/1467669`
`15:06.02`	`[TK]D-Fender`	`McL0VIN: You should be pastebinning your filed call attempt at CLI with SIP DEBUG enabled...`
`15:06.43`	`[TK]D-Fender`	`AsteriskBeginner: Other will better be able to advise on whic ISDN BRI hardware and drivers work the best, but the rest of the book applies.`
`15:11.27`	`AsteriskBeginner`	`the book just explains how to install/compile asterisk, and in chapter 3 it starts creating a dialplan, without configruing voip configurations or isdn configurations`
`15:12.16`	`*** join/#asterisk talirk81 (n=talirk@rrcs-67-78-39-22.sw.biz.rr.com)`
`15:12.48`	`McL0VIN`	`[TK]D-Fender: the sip debug will have some info i don;t feel to share`
`15:12.57`	`McL0VIN`	`in public`
`15:13.42`	`talirk81`	`In an AGI i am Executiing "EXEC GoTo Error\|s\|1" and Asterisk shows GoTo (Error,s,1) but it doesnt seem to be running the items in the Dialplan context, can you not switch Dialplan contexts from an agi scrit?`
`15:14.15`	`[TK]D-Fender`	`McL0VIN: PM me the link and PW the post if you must.`
`15:14.43`	`McL0VIN`	`[TK]D-Fender: cool`
`15:14.59`	`[TK]D-Fender`	`talirk81: Yes, that might change where * will resume dialplan, but AGI does not happen in the background.`
`15:15.15`	`talirk81`	`McL0VIN, pastabin.ca has a nice encrypt fucntion btw`
`15:16.45`	`McL0VIN`	`talirk81: i know but i didn't want to violate [TK]D-Fender privacy by just pm him without him telling me`
`15:17.13`	`talirk81`	`[TK]D-Fender, right what i want to happen is the Dial Plan to switch to the error context so when the AGI script exits , it plays a message then hangs up. But its not playing the message. and i dont really want to have EXECIF($[${ERROR} = "TRUE"] Goto(Error,s,1) on every other line of my dial plan`
`15:17.39`	`[TK]D-Fender`	`talirk81: And I don't see you showing me the problem`
`15:18.34`	`[TK]D-Fender`	`McL0VIN: I would not have objected if you sent a 1 line PM with the link already prepared and the PW provided. I help those who want to help themselves.`
`15:19.02`	`talirk81`	`The AGI script issues a "EXEC GoTo Error\|s\|1) which has exten => s,1,Background("LevelCall/CallCenter_Error") but when the AGI script has exits the system doesnt play that message it just hangs up with is pri 2 in the Context`
`15:19.18`	`talirk81`	`its like its skipping pri 1`
`15:19.58`	`*** join/#asterisk chendy (n=chatzill@58.61.197.107)`
`15:24.59`	`Corydon76-dig`	`talirk81: Don't use EXEC goto, then. Use: set context, set extension, set priority`
`15:33.39`	`*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)`
`15:37.25`	`*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)`
`15:54.05`	`*** join/#asterisk leejohn (i=bsdnewbi@124.106.129.146)`
`15:55.17`	`leejohn`	`good day guys, has anyone tried the latest svn 1.4 branch, i'm having a problem to load chan_sip it has something to do with recent commit by mnicholson Review: https://reviewboard.asterisk.org/r/287`
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`16:04.07`	`McL0VIN`	`[TK]D-Fender: thank you i am able to receive and make call from my * now...now i have all the tools set correctly , i will be read for a few and will come back if i have questions :)`
`16:04.45`	`[TK]D-Fender`	`McL0VIN: And I suggest you trash 90% of that bloated sample dialplan that VP provided you`
`16:08.29`	`McL0VIN`	`[TK]D-Fender: no doubt , i was thinking to use Visual DP , what do you think?`
`16:10.35`	`*** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe)`
`16:23.53`	`rue_mohr`	`http://eds.dyndns.org/~ircjunk/asterisk/missing_11db.html`
`16:26.23`	`[TK]D-Fender`	`McL0VIN: Don't be a retard...`
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`16:27.56`	`McL0VIN`	`[TK]D-Fender: why?`
`16:28.35`	`*** join/#asterisk CodeWork (n=Miranda@p5083F081.dip.t-dialin.net)`
`16:28.43`	`[TK]D-Fender`	`McL0VIN: Someone else's bloat being added to the mix and it limits what you can do to what it was designed to offer`
`16:29.36`	`[TK]D-Fender`	`rue_mohr: So, how long have you spent on the phone with support trying to get this to work?`
`16:32.10`	`*** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl)`
`16:32.26`	`rue_mohr`	`this isn't a 'it dosn't work' issue, this is a 'there is something broken in asterisk between these versions' issue`
`16:32.50`	`rue_mohr`	`and I didn't buy that card, it was given to me by kb1`
`16:33.27`	`rue_mohr`	`BUT I'm sure this is the same -11db problem I'm having at work`
`16:33.50`	`*** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl)`
`16:34.23`	`rue_mohr`	`as my system at home uses native channel bridging, I'd never encounter this being a problem at home`
`16:35.19`	`rue_mohr`	`I think somewhere there is a 1 bit oops, that happens twice, its a shift or divide...`
`16:35.35`	`rue_mohr`	`but for me, that still leaves it as a needle in a haystack`
`16:37.15`	`rue_mohr`	`I cant really even mix and match to prove if its the drivers or asterisk`
`16:37.40`	`rue_mohr`	`I dont think those old zaptel drivers work with the new asterisk version`
`16:37.54`	`*** join/#asterisk \|Cybex\| (n=John@80.100.126.176)`
`16:39.21`	`[TK]D-Fender`	`rue_mohr: Then how can you say its an Asterisk issue?`
`16:39.31`	`rue_mohr`	`I'm lumping`
`16:39.37`	`*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)`
`16:39.56`	`rue_mohr`	`as far as I'm concerned, the digium drivers are part of asterisk`
`16:40.06`	`rob0`	`Hello lumping, how are you? :)`
`16:40.27`	`[TK]D-Fender`	`rue_mohr: I can see that. Sorry, there isn't magical FUBAR between Zap & Dahdi otherwise every card would be screwed and there'd be a shit-storm of people in here screaming`
`16:40.27`	`rue_mohr`	`so far I think Qwell is the only one who wants to drive me 6 feet under for it`
`16:40.53`	`rue_mohr`	`I think there is and nobody is pointing fingers cause they cant prove it`
`16:41.07`	`rue_mohr`	`how many people have a channelbank and a dbm meter?`
`16:41.23`	`jaytee`	`headdesks`
`16:41.25`	`rue_mohr`	`and just went from zaptel to dahdi`
`16:41.49`	`rue_mohr`	`jaytee, if you know something, I'd love to hear it`
`16:41.58`	`[TK]D-Fender`	`rue_mohr: Our systems are all fine and you are screwed up.`
`16:42.21`	`rue_mohr`	`no, there is a problem with asterisk or the dahdi drivers that causes an 11db attnuation somewhere`
`16:42.34`	`rue_mohr`	`I'v proven it`
`16:42.47`	`rue_mohr`	`look at the page, everything worked PERFECT before I upgraded it`
`16:42.51`	`[TK]D-Fender`	`rue_mohr: Thats the thing. Its only you. No-one else here. I can't think of a single other person with a problem at all like yours.`
`16:43.04`	`rue_mohr`	`I was thrilled that the asterisk milliwatt REALLY was 0dbm`
`16:43.16`	`[TK]D-Fender`	`rue_mohr: The card has never worked. You have no other versions to test against you say.`
`16:43.23`	`[TK]D-Fender`	`rue_mohr: Noone else has your problems with it`
`16:43.30`	`[TK]D-Fender`	`rue_mohr: Its jsut your card. Get over it`
`16:43.33`	`rue_mohr`	`this card does work`
`16:43.40`	`rue_mohr`	`[TK]D-Fender, this isn't the system at work`
`16:43.43`	`rue_mohr`	`this is my home system`
`16:43.53`	`[TK]D-Fender`	`rue_mohr: If your idea of work is "flaming piece of shit I struggle with daily", then sure`
`16:43.55`	`rue_mohr`	`the 11db loss happened when I upgraded`
`16:44.45`	`jaytee`	`those basement dwelling trolls that have an underground cave full of old ham radio gear, old telco equipment, tons of old computers and every back issue of Popular Electronics since 1959 suddenly discover this thing called the Internet.`
`16:45.11`	`rue_mohr`	`look, there is a real problem here, I'm standing on it`
`16:45.52`	`rue_mohr`	`for most cases I'm sure that people just write off the 11db as not knowing what their doing 200%`
`16:46.14`	`rue_mohr`	`I know what I'm doing, there is 11db loss here`
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`16:51.37`	`Juggie`	`rue_mohr, what hardware card is the problem occuring on?`
`16:51.41`	`Juggie`	`the t100p or another one?`
`16:55.29`	`tzafrir_laptop`	`rue_mohr, hmm.. the t100p is digital, right. It shouldn't really have any a2d in it`
`17:00.04`	`Juggie`	`the t100p is not supported, but if you have anything else call digium`
`17:01.36`	`*** join/#asterisk pif (n=ldm@zenon.apartia.fr)`
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`17:30.38`	`saxa`	`hi, if I want to make a Zap/1 ring in a local context I use just Dial() and the extension nr as an argument ?`
`17:30.59`	`saxa`	`on a local machine but a different context , sorry`
`17:32.43`	`*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)`
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`17:44.52`	`saxa`	`maybe I expressed myself wrongly,`
`17:45.01`	`*** join/#asterisk Loki (i=loki@unaffiliated/loki)`
`17:45.56`	`saxa`	`how do i connect a Zap/1 when it rings to a extension on my local machine ?`
`17:46.15`	`saxa`	`I should use Dial()`
`17:46.22`	`*** join/#asterisk ruben23 (n=RPL@124.107.3.178)`
`17:46.30`	`saxa`	`but just plain, Dial(ext_nr) doesnt work`
`17:47.25`	`saxa`	`[Jun 20 14:49:46] WARNING[22041]: app_dial.c:1156 dial_exec_full: Dial argument takes format (technology/[device:]number1) == Spawn extension (incoming, s, 2) exited non-zero on 'Zap/1-1'`
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`17:55.53`	`alonzo`	`any body used asterisk?`
`17:56.40`	`Docteh`	`nope, its a channel full of BeOS users that WISH they could run asterisk`
`17:57.25`	`*** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net)`
`17:58.29`	`Loki`	`Hehe`
`17:59.19`	`alonzo`	`it's cool.. why it's so quietly?`
`18:00.03`	`alonzo`	`Loki, hello!!`
`18:01.02`	`Loki`	`Hallo.`
`18:01.39`	`*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)`
`18:01.55`	`alonzo`	`listen you are master of asterisk? :)`
`18:02.17`	`Loki`	`No I am here with a question myself`
`18:02.37`	`alonzo`	`sorry for my english, it's not my native..`
`18:03.15`	`alonzo`	`what question you have?`
`18:04.02`	`Loki`	`I mean, I am not a master of Asterisk. I am here because I have a question reguarding asterisk as well`
`18:04.46`	`alonzo`	`you are developer or just user?`
`18:05.09`	`Loki`	`I am having issues getting the Asterisk-GUI to getting up and running so, I am here to ask a question, as well am I in #asterisk-gui`
`18:05.12`	`Loki`	`I am just a user.`
`18:05.21`	`rob0`	`Masterisk`
`18:05.32`	`alonzo`	`:)`
`18:06.00`	`alonzo`	`Loki, how for you trixbox?`
`18:06.20`	`Loki`	`trixbox?`
`18:06.50`	`alonzo`	`it's beter then asterisk-GUI or not?`
`18:08.38`	`Loki`	`That looks like the freePBX interface`
`18:08.55`	`Loki`	`But I am learning the asterisk system, so I am learning how to set it up`
`18:10.30`	`alonzo`	`do you have expirience?`
`18:12.40`	`alonzo`	`can i make like this Dial(some_app()/some_app) in dialplan to resolve tehnology and number?`
`18:15.26`	`Loki`	`I don't know.`
`18:16.17`	`alonzo`	`it's possible, any body knows?`
`18:21.50`	`alonzo`	`Loki? where you use asterisk?`
`18:22.25`	`*** join/#asterisk smps (n=smps@193.170.53.51)`
`18:22.30`	`*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)`
`18:30.50`	`*** join/#asterisk ruben23 (n=RPL@124.107.3.178)`
`18:31.43`	`*** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com)`
`18:32.13`	`*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)`
`18:36.38`	`*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)`
`18:38.53`	`Docteh`	`alonzo: you could do an AGI call and set a variable, before the dial`
`18:39.42`	`alonzo`	`and if i want to use my own module?`
`18:40.50`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)`
`18:41.02`	`Docteh`	`you could use a module to set a variable but im not sure on Dial(callingfunctionhere()/callanotherfunction())`
`18:41.36`	`alonzo`	`it's interisting..`
`18:42.16`	`alonzo`	`I need this.. in my system..`
`18:43.05`	`*** join/#asterisk justdave (n=dave@unaffiliated/justdave)`
`18:43.38`	`Docteh`	`its not critical`
`18:44.01`	`Docteh`	`call a function to set a variable then Dial(${variablename})`
`18:44.24`	`alonzo`	`hmm..`
`18:47.29`	`alonzo`	`thanx.. but it's interesting.. it's was be match transperent..`
`18:47.44`	`alonzo`	`like in other languges.. call in call..`
`18:48.44`	`Docteh`	`well it might be possible but i have no information on it`
`18:49.14`	`alonzo`	`who knows? developers?`
`18:50.22`	`Docteh`	`yea`
`18:51.43`	`*** join/#asterisk phurl (n=mdupont@82.114.75.252)`
`18:52.22`	`Loki`	`I am trying to allow allow calls to come in over my SIP line, but I can't seem to set it up. What config file is that in? To tell them that it is coming over the sip trunk?`
`18:54.31`	`Docteh`	`sip.conf and extensions.conf i think`
`18:54.32`	`alonzo`	`docteh right..`
`18:54.51`	`rob0`	`What do you mean, "allow calls to come in"? That's pretty broad.`
`18:56.19`	`Docteh`	`alonzo: changing from having two functions that set variables to two functions that return stuff, shouldn't be a big rewrite`
`18:58.27`	`Loki`	`rob0: I mean. I want to be able to accept calls from my sip provider, off of my DID, right now, I can't seem to be able to figure out how`
`18:58.31`	`alonzo`	`but it's much better..`
`18:59.02`	`Docteh`	`well is it better if you have to add the feature to asterisk? ;)`
`18:59.56`	`rob0`	`Does the SIP/DID provider give you a sample config?`
`19:00.20`	`alonzo`	`docteh, i want to write some applications.. something like vm..`
`19:01.49`	`alonzo`	`but, i don't wanna invent my own bike.. :)`
`19:03.20`	`Docteh`	`well`
`19:03.30`	`Docteh`	`then Dial() from inside your app then`
`19:05.15`	`Loki`	`rob0: yes they do`
`19:05.43`	`alonzo`	`developer say Dial(${MYFUNCTION(parm1,parm2)})`
`19:05.47`	`Loki`	`but not with DID, just SIP`
`19:07.34`	`alonzo`	`it's looking good..`
`19:19.38`	`*** part/#asterisk juanIMP (n=juan@200.71.41.254)`
`19:30.49`	`*** join/#asterisk DavidR2008 (n=chatzill@nc-71-54-156-206.dhcp.embarqhsd.net)`
`19:36.13`	`DavidR2008`	`I have an * box that uses Sangoma hardware to terminate POTS PRIs. From this the calls are routed to: Old IVR system over T1, Old PBX over T1, New IVR * via IAX2, or New PBX * via IAX2`
`19:36.46`	`DavidR2008`	`occasionally one of the PRIs stops transfering touchtones`
`19:37.30`	`*** join/#asterisk ruben23 (n=RPL@124.107.3.178)`
`19:37.35`	`DavidR2008`	`I would greatly appreciate any suggestions on how to troubleshoot this`
`19:45.54`	`*** join/#asterisk \|Cybex\| (n=John@80.100.126.176)`
`19:47.09`	`*** join/#asterisk xfighter (n=xfighter@a95-203.adsl.paltel.net)`
`19:47.13`	`xfighter`	`hello guys`
`19:47.19`	`xfighter`	`I wanna ask something`
`19:47.27`	`xfighter`	`do I have a fatal error here :`
`19:47.32`	`xfighter`	`http://pastebin.com/m4e093681`
`19:50.47`	`*** join/#asterisk af_ (n=getsmart@88-149-230-49.dynamic.ngi.it)`
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`20:21.45`	`rue_mohr`	`Juggie, its not a hardware problem its a T100P, I also have a missing 11db on a tdm800`
`20:23.08`	`rue_mohr`	`tzafrir_laptop, correct, so I'm saying its a error in the digital handling, thats why this is to convienient to be using a channelbank, especially when I know that the 1mw worked perfectly on the old zaptel drivers with the old asterisk version`
`20:23.08`	`rue_mohr`	`http://eds.dyndns.org/~ircjunk/asterisk/missing_11db.html`
`20:23.29`	`rue_mohr`	`I can positivly say its something thats between the 1mw source and the T1`
`20:24.02`	`tzafrir_laptop`	`rue_mohr, what test do you have to show this 11db difference?`
`20:24.12`	`rue_mohr`	`a dbm meter`
`20:24.19`	`rue_mohr`	`remember, it tested perfectly before the upgrade`
`20:24.38`	`rue_mohr`	`I can show you the meter`
`20:25.05`	`rue_mohr`	`I cant roll back my system to show you the 1mw working properly, it turned into a one way upgrade`
`20:25.44`	`rue_mohr`	`but I'm happy to do any experiments you can think of to try to isolate this`
`20:26.11`	`rue_mohr`	`I dont think the old asterisk will work with the new dahdi, but the new asterisk might work with the old zaptel?`
`20:26.27`	`rue_mohr`	`there are some hoops, I'm not versed`
`20:27.26`	`rue_mohr`	`my dbm readings are with a meter on the analog lines`
`20:33.48`	`rue_mohr`	`2008-11-06 22:38 +0000 [r5266] Doug Bailey <dbailey@digium.com> * zonedata.c: set DTMF twist levels for listed EU countries to meet TBR-21 standard of -9/-11 dB`
`20:33.50`	`rue_mohr`	`hmmm`
`20:36.18`	`rue_mohr`	`funny that in particular, cause milliwatt uses the playtones for its signal`
`20:37.33`	`af_`	`I can't set the language of a gxp2020, any hint?`
`20:37.46`	`rue_mohr`	`I dont know`
`20:42.30`	`rue_mohr`	`in the old drivers zonedata.c has no level specification`
`20:53.51`	`*** join/#asterisk [pnp]tomas (n=xanith@96bus5.tampabay.res.rr.com)`
`20:55.57`	`tzafrir_laptop`	`you can confirm that by editing zonedata.c, rebuilding ztcfg, and re-running the tests`
`20:56.29`	`rue_mohr`	`you mean run the new asterisk with the old zaptel driver?`
`20:56.32`	`rue_mohr`	`I could try`
`20:56.59`	`rue_mohr`	`or the current ones with the dtmf_high_level adjusted`
`20:57.56`	`rue_mohr`	`oh I see it only comes up in dahdi_config`
`20:58.01`	`rue_mohr`	`er _cfg`
`20:58.02`	`rue_mohr`	`hmm`
`20:58.36`	`[pnp]tomas`	`shouldn't 'dahdi show channels' be showing me more than: pseudo default default In - with a full PRI plugged into it?`
`20:58.39`	`*** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net)`
`21:00.08`	`rue_mohr`	`tzafrir_laptop, I'll mod the current drivers, try, and get back to you`
`21:00.40`	`tzafrir_laptop`	`rue_mohr, note that this is not a change in the actual drivers. just zonedata.c in dahdi_tools`
`21:01.10`	`rue_mohr`	`yes, I changed dtmf_high_level and dtmf_low_level to 0, will recompile, install and try`
`21:01.12`	`tzafrir_laptop`	`no need to reload any driver. just re-run dahdi_cfg`
`21:01.42`	`buttons840`	`I have a simple spool file, all it does is makes an outbound call (which rings ok), but it starts playing the sound file before the call is answered?`
`21:07.44`	`[pnp]tomas`	`is show channeltypes in 1.6?`
`21:30.03`	`*** join/#asterisk simNIX (n=simNIX@217.120.171.101)`
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`22:01.40`	`*** join/#asterisk s14ck (n=s14ck@190-76-79-224.dyn.movilnet.com.ve)`
`22:08.11`	`*** join/#asterisk McL0VIN (n=chatzill@unaffiliated/dacs)`
`22:09.47`	`*** join/#asterisk ruben23 (n=RPL@124.107.3.178)`
`22:09.54`	`ruben23`	`hi`
`22:10.10`	`McL0VIN`	`ok i am reading Chapter 5 Basic dialplan, and we are(me & the book) setting up a basic DP where incoming call will be answer(), Playback(hello-world), then Hangup()`
`22:11.01`	`b14ck`	`cool`
`22:11.03`	`b14ck`	`:)`
`22:11.33`	`McL0VIN`	`i changed my extension.com from [voicepulse-in] exten => _XX.,1,NoOp(Call received from VoicePulse) exten => _XX.,n,Dial(SIP/sipuser&IAX2/iaxuser)`
`22:12.09`	`b14ck`	`the best way to do it, is to modify your outbound context for phones`
`22:12.12`	`b14ck`	`eg:`
`22:12.27`	`McL0VIN`	`to exten => s,1,Answer() and exten => s,n,Playback(hello-workd) and then Hangup`
`22:12.35`	`b14ck`	`[from-internal] exten => 123,1,Answer() exten => 123,n,Playback(hello-world) exten => 123,n,Hangup()`
`22:12.44`	`b14ck`	`that way, you can pick up a phone, dial '123', and it'll run the code`
`22:12.48`	`b14ck`	`instead of calling into your server`
`22:13.36`	`McL0VIN`	`i get rejected because extension not found`
`22:13.45`	`b14ck`	`asterisk -rx 'reload'`
`22:14.01`	`b14ck`	`you need to reload asterisk after making changes to your dialplan`
`22:14.09`	`rob0`	`dialplan reload`
`22:14.19`	`McL0VIN`	`well i did dialplan reload and sip reload`
`22:14.21`	`b14ck`	`^ what he said`
`22:14.34`	`b14ck`	`McL0VIN, change what i said, dont do it for incoming`
`22:14.41`	`rob0`	`s,n,Playback(hello-workd) ?`
`22:14.43`	`b14ck`	`it'll be easier to test`
`22:14.57`	`AlmightyOatmeal`	`has anyone had experience with the polycom soundpoint ip 301 sip phone? how does one configure the phone to login to asterisk?`
`22:15.04`	`AlmightyOatmeal`	`voip-info.org wasn't much help`
`22:16.52`	`McL0VIN`	`bl4ck : i don't understand it`
`22:17.11`	`McL0VIN`	`sorry i am still learning and not that savy with * yet`
`22:19.24`	`b14ck`	`McL0VIN, no problem`
`22:19.32`	`b14ck`	`so basically, what extension are you using to test calls now?`
`22:19.39`	`b14ck`	`I'm assuming you have at least one phone hooked up.`
`22:19.54`	`McL0VIN`	`x-lite`
`22:20.11`	`b14ck`	`ok, so real quick. open up your sip.conf and look at your configuration for that phone`
`22:20.18`	`b14ck`	`look for the context= line`
`22:20.21`	`b14ck`	`and tell me what that says`
`22:20.36`	`McL0VIN`	`k i sec`
`22:21.15`	`McL0VIN`	`b14ck: outgoing`
`22:21.42`	`b14ck`	`ok, so what that means, is that whenever you dial a number on your x-lite phone, inside the file extensions.conf there will be some dialplan code under [outgoing] which your phone will go to`
`22:21.53`	`b14ck`	`so whenver you dial a number on your phone, it'll run the code in [outgoing]`
`22:22.05`	`Docteh`	`dialplan show outgoing <-- also handy`
`22:22.11`	`b14ck`	`The -easiest- way to test out custom code, is just to modify [outgoing] in extensions.conf`
`22:22.26`	`b14ck`	`That way, you can execute your custom code (for testing) by just dialing a number directly on your x-lite phone.`
`22:22.42`	`b14ck`	`The way to do this, (for example), would be to edit the [outgoing] to say:`
`22:23.00`	`b14ck`	`exten => 123,1,Answer() exten => 123,n,Playback(hello-world) exten => 123,n,Hangup()`
`22:23.08`	`McL0VIN`	`b14ck: i am not talking about outgoing`
`22:23.15`	`b14ck`	`what this means, is that if you dial 123 on your xlite phone, it will run your code`
`22:23.19`	`McL0VIN`	`i am taking for incoming`
`22:23.33`	`b14ck`	`McL0VIN, I know, but it is easier to test outgoing instead of incoming.`
`22:23.37`	`b14ck`	`It will work the same way.`
`22:23.51`	`b14ck`	`After you have tested the code, and understand it, then you can move it to incoming.`
`22:23.55`	`McL0VIN`	`ok wait a sex`
`22:23.58`	`b14ck`	`k`
`22:24.00`	`McL0VIN`	`sec`
`22:24.03`	`Docteh`	`McL0VIN: the dialplan code works either way, asterisk doesn't see a call as incoming or outgoing, and testing it with a softphone is faster than dialing a phone :)`
`22:25.27`	`Docteh`	`I wonder if i can work channel-insecure into infront of Voicemail() and have it sound good`
`22:25.46`	`*** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net)`
`22:25.59`	`b14ck`	`what is channel-insecure?`
`22:26.24`	`McL0VIN`	`b14ck: am back`
`22:26.36`	`b14ck`	`McL0VIN, so what happened? :)`
`22:27.15`	`McL0VIN`	`b14ck: am lost`
`22:27.21`	`b14ck`	`McL0VIN, lol`
`22:27.28`	`b14ck`	`McL0VIN, open up extensions.conf`
`22:27.32`	`Docteh`	`channel-insecure-warn: Attention! this voice path is not secure, do not discuss classified information and do not use project code words`
`22:27.39`	`b14ck`	`go down to where it says [outgoing]`
`22:27.42`	`b14ck`	`let me know once you are there`
`22:28.47`	`McL0VIN`	`b14ck: there`
`22:28.59`	`b14ck`	`McL0VIN, now, directly below that line, add the following:`
`22:29.05`	`b14ck`	`exten => 123,1,Answer()`
`22:29.09`	`Docteh`	`http://pastebin.com/meb46183`
`22:29.12`	`b14ck`	`exten => 123,n,Playback(hello-world)`
`22:29.17`	`b14ck`	`exten => 123,n,Hangup()`
`22:29.20`	`b14ck`	`let me know once you've done that`
`22:30.08`	`McL0VIN`	`b14ck: ok , but b4 i do i have a quick question: i have exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=1234567890) and some more`
`22:30.16`	`ruben23`	`hi`
`22:30.24`	`b14ck`	`McL0VIN, don't delete any of that, leave that part alone`
`22:30.31`	`b14ck`	`McL0VIN, just add the stuff i told you :)`
`22:30.42`	`McL0VIN`	`b14ck: o\|< :)`
`22:31.05`	`b14ck`	`McL0VIN, once you've done that, save the file. and run: asterisk -rx 'reload'`
`22:31.27`	`b14ck`	`Then, from your x-lite phone, dial 123`
`22:31.28`	`b14ck`	`:)`
`22:33.55`	`McL0VIN`	`b14ck: hahahahahahahahahah it worked , it worked !!!!!`
`22:34.00`	`McL0VIN`	`sweetttt`
`22:34.46`	`b14ck`	`=)`
`22:35.26`	`McL0VIN`	`b14ck: but how, i don't have a context in my sip for that 123 extension`
`22:35.31`	`*** join/#asterisk bmg505 (n=leon@196-209-235-22-rndf-esr-5.dynamic.isadsl.co.za)`
`22:35.49`	`b14ck`	`McL0VIN, you dont need one. All of the context's are referring to extensions.conf`
`22:35.59`	`b14ck`	`sip.conf contexts are only for devices (like phones, etc)`
`22:36.09`	`b14ck`	`extensions.conf contexts are for code`
`22:44.52`	`comfrey`	`anyone have a sec to help me debug sip call faiure.`
`22:45.25`	`comfrey`	`i am getting "the person at extension x is unavail"`
`22:47.06`	`comfrey`	`sip show peers shows 14 sip peers`
`22:47.14`	`comfrey`	`with status ok`
`22:48.16`	`McL0VIN`	`b14ck: ok, why it didn't work for the incoming?`
`23:01.30`	`*** join/#asterisk tjz (n=tjz@bb116-15-73-38.singnet.com.sg)`
`23:19.52`	`*** join/#asterisk buttons840 (n=buttons8@c-24-10-149-58.hsd1.ut.comcast.net)`
`23:20.45`	`buttons840`	`Are there any SIP soft phones which support multiple lines? The more lines the better. Open source is also prefered.`
`23:21.18`	`McL0VIN`	`buttons840: x-lite , but not the free one`
`23:22.13`	`buttons840`	`McL0VIN, yeah, i saw that one, but i don't want to pay, i'm on a low budget`
`23:22.31`	`McL0VIN`	`buttons840: you have 2 computers`
`23:22.37`	`buttons840`	`yes`
`23:22.56`	`McL0VIN`	`buttons840: then install x-lite in both and there you have it`
`23:23.00`	`McL0VIN`	`;)`
`23:24.08`	`buttons840`	`yeah, i've done that, although not with x-lite. would it be possible to create a softphone that supported and unlimited number of lines? or does the nature of sip prevent this?`
`23:26.49`	`buttons840`	`been looking and zoiper is free and supports 6 lines`
`23:29.41`	`buttons840`	`i also read that an older free version of x-lite supported 8 lines, although it's oudated now, it should still be freely available?`
`23:43.27`	`*** join/#asterisk gunter (n=user@87.127.97.39)`
`23:46.15`	`*** join/#asterisk Trionnis (n=rboggs@s233-51-251.nap.wideopenwest.com)`
`23:46.19`	`ruben23`	`hi anyone have idea setting IAX2 trunk on asterisk`
`23:46.29`	`*** part/#asterisk Trionnis (n=rboggs@s233-51-251.nap.wideopenwest.com)`
`23:47.29`	`McL0VIN`	`ruben23: yeah , chapter 4, page 101 on the book`
`23:52.09`	`McL0VIN`	`i have my dialplan set up to Playback(main-menu) for incoming, but when i call i hear "ain menu"`
`23:54.19`	`*** join/#asterisk chi6IT41 (n=chigital@tmo-105-125.customers.d1-online.com)`
`23:54.27`	`lanning`	`put a wait(1) before playback()`

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