00:02.01 | *** join/#asterisk camerin (~camerin@elite.bshellz.net) |
00:39.29 | *** join/#asterisk Qwell (north@asterisk/developer/Qwell) |
00:39.29 | *** mode/#asterisk [+o Qwell] by ChanServ |
01:18.09 | moe` | so, myes, what's this with setting up skype as a gateway |
01:19.13 | moe` | I keep getting the congrats from the asterisk bitch |
01:19.28 | moe` | exten => _9XXXXXXXXXXX,1,Dial(SIP/skype/${EXTEN}) |
01:19.33 | moe` | is that not correct? |
01:25.02 | moe` | unless that causes asterisk to pass the 9 prefix to skype, in which case I change it to... |
01:25.12 | moe` | exten => _XXXXXXXXXXX,1,Dial(SIP/skype/${EXTEN}) |
01:25.56 | moe` | drmessano, what say you? |
01:26.16 | [TK]D-Fender | That line clearly doesn't matter here |
01:26.22 | [TK]D-Fender | So top staring at it' |
01:26.25 | [TK]D-Fender | stop* |
01:26.37 | moe` | ok, so where does one go from there? |
01:26.47 | [TK]D-Fender | <moe`> unless that causes asterisk to pass the 9 prefix to skype, in which case I change it to... <- which it does and we told you FIVE TIMES |
01:26.56 | [TK]D-Fender | look at what it IS hitting and where |
01:28.20 | moe` | I keep getting the asterisk default "congrats" |
01:28.30 | [TK]D-Fender | If you are sitting in * CLI and WATCHING this happening.. then you aren't really looking or trying |
01:28.43 | [TK]D-Fender | Forget your EARS and start using your EYES |
01:28.48 | [TK]D-Fender | ASTERISK CLI |
01:28.57 | [TK]D-Fender | No if's and's or but's |
01:29.13 | [TK]D-Fender | connect and LOOK at what is getting executed |
01:29.38 | [TK]D-Fender | And there is no "asterisk default". |
01:29.52 | moe` | [Jul 5 21:27:28] WARNING[101283] chan_unistim.c: Your OS does not support IP_PKTINFO, you must set public_ip. |
01:29.54 | [TK]D-Fender | You may have put SAMPLE CONFIGS into your config folder.. but that's on you |
01:29.56 | moe` | what's this fresh hell |
01:30.09 | [TK]D-Fender | We don't care about unistim |
01:30.52 | moe` | yes I have sample configs in my config directory |
01:31.01 | moe` | that's what pkg install asterisk13 on FreeBSD does |
01:31.02 | [TK]D-Fender | then your dialplan is full of GARBAGE |
01:31.06 | [TK]D-Fender | Which you shouldn't have |
01:31.14 | [TK]D-Fender | But before tr4ashing it |
01:31.18 | [TK]D-Fender | time to LEARN what is happening |
01:31.26 | [TK]D-Fender | go LOOK at your call |
01:31.39 | moe` | yeah that's the problem, I don't understand what dial plans and contexts are yet |
01:31.48 | [TK]D-Fender | Then you're screwed |
01:31.57 | [TK]D-Fender | because that's the step you're DOING right now |
01:32.33 | [TK]D-Fender | You don't say "how do I steer?" while you're behind the wheel on the highway. You're gonna die. This will no please your driving instructor either |
01:32.38 | [TK]D-Fender | READ THE BOOK. |
01:32.52 | [TK]D-Fender | Don't know what contexts are? Don't know what the patterns fully represent? |
01:32.56 | [TK]D-Fender | There is a FREE BOOK for this |
01:32.58 | [TK]D-Fender | READ IT |
01:33.10 | moe` | yeah, I hear you, RTFM |
01:34.00 | [TK]D-Fender | All call processing = dialplan. You BETTER learn this part because it's 90% of Asterisk. |
01:34.08 | moe` | just I was rather hoping there'd be a quick and dirty way to get local extensions and a skype gate going |
01:34.48 | [TK]D-Fender | Setting up so you can call out to a SIP provider, or call to/from a SIP phone of some sort is a tiny number of lines. Processing what is DIALED it everything |
01:35.11 | [TK]D-Fender | There is no agic. Every single thing your system does when your soft-phone PLACES a call is dialplan |
01:35.20 | [TK]D-Fender | There is "quick and dirty". |
01:35.27 | [TK]D-Fender | There is also "you need to UNDERSTAND this" |
01:35.56 | [TK]D-Fender | I linked you a specific chapter the other day that even showed you both pieces TOGETHER |
01:36.25 | moe` | yep, I hear that, I need to stop bitching about stuff and RTFM and make it work |
01:37.32 | [TK]D-Fender | And as a primer there was the 5 times I told you to show us the call |
01:37.35 | [TK]D-Fender | You still haven't done. |
01:37.53 | [TK]D-Fender | Asking why it's doing what it's doing is poitnless when you aren't showing us the EVIDENCE |
01:45.44 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
02:02.55 | *** join/#asterisk D30 (~D30@58.71.19.178) |
02:02.55 | [TK]D-Fender | heads out for a while |
02:14.59 | *** part/#asterisk Mp5shooter (~Mp5@unaffiliated/mp5shooter) |
02:33.12 | moe` | oh c'mon, now I have another client that can connect to asterisk but extensions aren't working |
02:33.26 | moe` | <--- whining bitch |
02:49.01 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
02:57.19 | ChannelZ | "aren't working" is of no help |
02:58.04 | moe` | someone is being an asshole... 156.34.37.48 |
03:00.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
03:02.24 | ChannelZ | the net is full of em |
03:02.30 | moe` | well yeah |
03:02.58 | moe` | ok, so I have ekiga clients connecting |
03:03.18 | moe` | so now, how to setup extensions |
03:03.34 | moe` | this should not be so complex. there is something stupid I'm misunderstanding |
03:03.49 | ChannelZ | Did you read my primer? |
03:03.55 | moe` | ~primer |
03:03.55 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
03:04.05 | ChannelZ | Because if you did you'd have already discovered how NOT complex it is. |
03:04.05 | *** join/#asterisk bananabas (~bananabas@unaffiliated/bananabas) |
03:06.57 | moe` | yeah I see all that |
03:07.03 | moe` | well, not *all* |
03:07.05 | moe` | but ok |
03:07.18 | moe` | its pissing me off, these things |
03:07.34 | ChannelZ | "Creating a Disalplan" |
03:07.44 | ChannelZ | s/Dis/Dia |
03:08.17 | moe` | very well written howto dude |
03:08.25 | moe` | but its not getting me there |
03:08.59 | ChannelZ | We're not psychic. Show us a failure |
03:09.21 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
03:10.04 | drmessano | Actually, start at "Configuring SIP" |
03:10.08 | drmessano | then the dialplan |
03:10.17 | drmessano | Because both of those pages together answer ALL your questions |
03:10.31 | moe` | my god I must be annoying the hell out of you guys. |
03:10.40 | drmessano | Read the TWO fucking pages |
03:10.48 | ChannelZ | true, they relate to each other with peer names and things |
03:11.07 | ChannelZ | Should be easy to see the correlation |
03:11.43 | ChannelZ | and explains the elusive contexts you're having problems getting |
03:13.06 | drmessano | moe`, i'm trying to give you the benefit of the doubt here.. You have already admitted how you "hardheaded" you are, which is basically "I dont read, I just keep trying shit til it works".. We've wasted tens of hours combined showing you document after document in how to set up what takes any n00b an hour to setup |
03:13.39 | drmessano | So intentional or not, you've crossed into the "Do it for me" AKA "Help Vampire" phase of this operation |
03:13.43 | drmessano | Just read the documents!!! |
03:13.47 | drmessano | Come on man |
03:14.15 | ChannelZ | Heh, I like that.. "help vampire" |
03:14.19 | drmessano | I want you to succeed here, but all you're doing is glancing at shit and making us repeat |
03:15.00 | drmessano | I've never looked at ChannelZ's docs before, but OMG.. it's pretty much ALL THERE in regard to the VERY BASIC part of dialplan you're missing |
03:20.27 | moe` | yeah, likely |
03:21.26 | moe` | so when SIP users are defined, do they have to be included into a dialplan (sip.conf) or is that a function of extensions.conf? |
03:23.16 | drmessano | Thats what the context is for.. |
03:23.49 | moe` | see this is what I'm missing |
03:23.52 | moe` | WTF is a context |
03:24.11 | ChannelZ | he hasn't read a thing |
03:24.18 | drmessano | I know |
03:24.24 | moe` | yeah, actually, I have |
03:24.31 | moe` | but I'm just not getting it |
03:24.41 | drmessano | Because on the Configuring SIP page, he clearly notes context=internal for each user |
03:24.48 | ChannelZ | "Finally, we set context=internal for the device. The context refers to a section of the dialplan, something we haven't talked about yet (but will in the next section.) In short, the dialplan is what actually makes Asterisk do something with calls, and contexts allow you to separate what happens to calls that originate from different places." |
03:25.12 | drmessano | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
03:25.18 | drmessano | FTFA |
03:25.23 | ChannelZ | "Contexts are like containers for extensions; they serve to separate extensions from each other in the dialplan. In sip.conf we configured our TestPhone-A peer with context=internal, so any calls it makes will wind up in the [internal] context of the dialplan." |
03:25.24 | drmessano | FTFM? |
03:25.28 | moe` | ok |
03:25.49 | drmessano | Did you read any of that ? |
03:25.55 | drmessano | Prior to him pasting it? |
03:28.50 | moe` | I'm about to fall back to mumble, really. |
03:28.57 | drmessano | lol |
03:28.59 | moe` | <--- failure |
03:29.25 | ChannelZ | shrugs |
03:29.25 | moe` | I have my own mumble (mumble.unixninja.net) but yeah |
03:29.36 | moe` | this is starting to be an epic failure |
03:31.42 | drmessano | Thats unfortunate, because all the pieces are right in front of you |
03:31.49 | drmessano | Pretty much hand-fed |
03:31.49 | ChannelZ | Your original plan seemed to be to call out through Skype. Why not just use Skype? |
03:32.17 | moe` | cuz I don't want the give the US/NSA the contents of my calls? |
03:32.35 | moe` | I was planning to go TLSv1 and I have my own proper cert and stuff |
03:33.22 | drmessano | Oh lord |
03:34.45 | moe` | right now for mumble I use openvpn to talk with friends and it's SSL 4096bit |
03:35.01 | moe` | so it's SSL in SSL |
03:35.16 | moe` | nazi crypto :) |
03:35.16 | drmessano | I will tell you this.. I consider myself very proficient at setting up SIP TLS and SRTP, and if you can't even grasp the basic concepts of SIP.conf and extensions.conf (difficulty 1), you're not going to be able to set up TLS+SRTP (difficulty 2) |
03:35.52 | moe` | well I get sip.conf, I think, I have clients connecting, and I can register with sip.skype.com |
03:36.57 | drmessano | No, you have clients registering. You can't even grasp the concept of adding context=blah to each SIP user and having a [blah] context in extension.conf with your dialplan |
03:36.59 | moe` | the TLS options for sip.conf are pretty simple |
03:37.14 | moe` | well now |
03:37.15 | moe` | see |
03:37.21 | moe` | you just explained a lot |
03:37.28 | drmessano | WHAT?! |
03:37.57 | drmessano | So basically you're read NOTHING we've pasted |
03:38.26 | moe` | well, read, but not properly understood |
03:38.30 | drmessano | "Finally, we set context=internal for the device. The context refers to a section of the dialplan, something we haven't talked about yet (but will in the next section.) In short, the dialplan is what actually makes Asterisk do something with calls, and contexts allow you to separate what happens to calls that originate from different places. |
03:38.33 | drmessano | ^^^^^^ |
03:38.41 | moe` | I'm an old school C programmer and /bin/sh is my bitch |
03:38.46 | moe` | so I need it clear |
03:38.47 | drmessano | What do you think the DIALPLAN IS? |
03:38.48 | moe` | :) |
03:38.54 | drmessano | EXTENSION.fucking.conf |
03:38.58 | drmessano | EXTENSIONS.fucking.conf |
03:39.17 | drmessano | No, it sounds like you need it done for you, and I am not doing that |
03:39.33 | moe` | I would not want you to waste your time on that |
03:39.40 | drmessano | You already have |
03:39.51 | drmessano | By not reading whats been presented |
03:39.54 | drmessano | So i'm done |
03:40.42 | moe` | so in sip.conf I have per user, "context=unixninja" |
03:40.56 | moe` | and an extensions.conf, I have a [unixninja] block |
03:41.01 | moe` | is this sane? |