| 00:00.14 | drmessano | You can pipe it in, like foreign oil |
| 00:02.14 | [TK]D-Fender | drmessano: So ho big is your IP foot-print? |
| 00:02.28 | [TK]D-Fender | rates the bits in all his computiers.. |
| 00:02.35 | [TK]D-Fender | DRILL BABY DRILL!!! |
| 00:02.54 | [TK]D-Fender | twooten: You can ditch your home phone service entrely. |
| 00:03.07 | drmessano | I can see russia from my house! |
| 00:03.36 | [TK]D-Fender | twooten: Typically ITSP's cost a fair bit less for home-typ service. this would et you let go of your home line and you wouldn't need an FXO device |
| 00:05.15 | grndslm | twooten: Ma Bell requires fees to connect to her service... if you drop the triple play package, you'll prolly end up spending more money... however, you'll be missing a ton of neat features like what LinuxMCE offers |
| 00:06.58 | drmessano | Bundling adds $20 right now to my parents service |
| 00:07.03 | drmessano | and i think the same for Comcast |
| 00:07.15 | drmessano | Even Vonage is only $5 more than that |
| 00:07.22 | drmessano | and thats on the high end |
| 00:08.33 | twooten | user beek was trying to aswer some questions, then he said 'no, you aren't going to get free telephony' then he disconnected so I'm still confused. I thought I understood that I could get a SPA-3102, and use linuxmce/asterisk server to make local and long distance calls from my home. Really sorry for the basic questions here, but I don't want to spend a bunch of money on something that won't work for me. Also I don't want to cancel my home phone servic |
| 00:08.33 | twooten | e if I still need it. There is basically one provider in my backwoods town in alabama, but they offer cable tv, internet and phone service, so I'll still be using them at least for tv and internet. |
| 00:08.37 | grndslm | and bundling is much more likely to increase with time... whereas ITSPs will be more likely to stay the same or even decrease |
| 00:08.51 | *** part/#asterisk UnixDawg (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
| 00:09.06 | drmessano | No, you cannot "just make free calls" |
| 00:09.16 | grndslm | twooten: how will you connect to Ma Bell with just a box? |
| 00:09.44 | grndslm | do you connect to cable internet with just a box? |
| 00:09.59 | grndslm | do you get gps signals with just a receiver?? no... you need a provider |
| 00:10.45 | grndslm | drmessano: am i missing something about gizmo's free inbound calling? |
| 00:11.11 | drmessano | Gizmo doesnt have free inbound calling |
| 00:11.18 | drmessano | Not from PSTN |
| 00:11.29 | twooten | so i do still need the local phone service in order to make local calls and long distance to make long distance calls? If so, what are the benefits of asterisk? |
| 00:11.56 | drmessano | twooten: Its a PBX.. if you dont know what a PBX is, you dont need it |
| 00:11.59 | grndslm | twooten: if you're getting cable phone service... that *is* your ITSP |
| 00:14.21 | grndslm | drmessano: so gizmo is useless without grandcentral? (at least in terms of free anything) |
| 00:15.07 | drmessano | twooten: the benefits of Asterisk is that its a telephony toolkit from which you can create a very complex PBX, and also use its user agents to make SIP <> SIP calls, connect to an ITSP for "YAY, more options than Cable or Bell" phone service, and even connect extensions over the internet making a company or even a familys communication on one large platform |
| 00:15.24 | drmessano | grndslm: You dont not have free PSTN connection with Gizmo, no |
| 00:16.21 | drmessano | On the advanced side, it connects to GoogleTalk, you can use a Bluetooth dongle and connect a cell phone, or even use the PSTN and use it to split your phone line out across the house, give yourself free voicemail, etc |
| 00:16.58 | [TK]D-Fender | twooten: Not to get you lost, but the idea is for you to understand your options for how you acquire phone service, and how that choice influences the equipment you buy, etc |
| 00:17.02 | drmessano | Theres a lot of "points" to asterisk |
| 00:17.29 | drmessano | But if your needs can be filled with a Vonage hockey puck, any PBX will be a waste of your time |
| 00:19.09 | [TK]D-Fender | twooten: twooten we encourage you to look at your oprions and see how the service your cable co offers fits in and evaluate what the best overall direction is going to be. |
| 00:19.40 | [TK]D-Fender | twooten: for instance if you have call-waiting with your line with the cable co you will functionally lose that with * and an FXO device |
| 00:20.18 | [TK]D-Fender | twooten: when you're on a call and it beeps through, * can't take that call separately and provide its VM service |
| 00:20.43 | [TK]D-Fender | twooten: So you lose some of *'s advantages & functionaily as well as some on your line. |
| 00:21.57 | grndslm | =^O there's no way to do call waiting with *? |
| 00:24.34 | twooten | drmessano and [TK]D-Fender: thank you. This is what I want to do. Install linuxmce on my system. I want to be able to make cheap/free phone calls both locally and long distance. I also want to hook up a security system and have the functionality provided by linuxmce/asterisk so that if a break in occurs, the system will send a text message or call my cell phone while I'm at work. I also want to have voice mail. It's only me and my wife, we don't really |
| 00:24.35 | twooten | <PROTECTED> |
| 00:24.46 | twooten | am I making this harder than it needs to be?? |
| 00:24.52 | [TK]D-Fender | twooten: AH |
| 00:25.00 | [TK]D-Fender | twooten: your security system is the clincher |
| 00:25.34 | [TK]D-Fender | twooten: You should indeed keep your land line yet. Now be aware of the limitations I mentioned with regards to *'s use of your line. |
| 00:25.55 | [TK]D-Fender | twooten: Forget about securit systems being pumped over VoIP & the internet |
| 00:26.20 | [TK]D-Fender | twooten: 1 approach you could take : |
| 00:27.23 | [TK]D-Fender | twooten: Remove any extra option you pay for on that Cable phone line. use it as a failover & for your security system. PORT your current phone # to an ITSP which will support multiple calls. You can then use * to do whatever you want from there |
| 00:28.15 | Spirits-Sight | [TK]D-Fender: is there any other speech engin that is Free that allows you to use as many steams as needed, Crystal (spelt wrong) only allows one steam at a time under its 30$ license |
| 00:28.52 | [TK]D-Fender | Spirits-Sight: I don't do TTS |
| 00:29.27 | Spirits-Sight | What do you use for your IVR voice, do you record them or what? |
| 00:30.36 | [TK]D-Fender | Spirits-Sight: if it isn't TTS, its clearly human... |
| 00:30.54 | [TK]D-Fender | Spirits-Sight: Don't ask sequels to boolean questions. |
| 00:31.48 | twooten | [TK]D-Fender: I googled ITSP and found http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential#USA, do you have a recommendation or do I just pick one and go with that? |
| 00:31.58 | Spirits-Sight | well it could of been that you record your own or have them record, but your right the end result is the same human done |
| 00:32.12 | [TK]D-Fender | ~itsplist-us |
| 00:32.12 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 00:32.36 | [TK]D-Fender | twooten: Also consider les.net. pretty good for USA as well. |
| 00:33.07 | [TK]D-Fender | twooten: Vitelity is pretty good, and Voicepulse is as well... though they've had a few hiccups |
| 00:34.27 | Spirits-Sight | Is there a reason you don't do TTS [TK]D-Fender? |
| 00:35.08 | [TK]D-Fender | Spirits-Sight: Because I don't want some unnatural voice talking to my callers and the message isn't dynamic? Are these concepts not as obvious as they may sound? |
| 00:35.29 | [TK]D-Fender | Spirits-Sight: Why don't I have a plane? Maybe because I don't need to fly anywhere. |
| 00:38.45 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 00:38.49 | Spirits-Sight | Right, the menu is not dynamic! So I understand that. I don't understand whats wrong with asking why, some people may want to use TTS because the person voice does not sound good on recordings or even if they don't want to change a area of their system that may play event information or what not. Sorry I asked you. |
| 00:39.37 | [TK]D-Fender | Spirits-Sight: I'm just saying some answers are obvious. This was one of them. |
| 00:39.55 | [TK]D-Fender | Spirits-Sight: Are your menus dynamic? How dynamic? |
| 00:40.33 | [TK]D-Fender | Spirits-Sight: If they rarely change then you only need 1 channel and you can use that to record static prompts. People don't seem to realize these things... |
| 00:40.37 | Spirits-Sight | my menus are not, I will have a couple of areas where reading to a record is not going to be the greatest sound and thats why I would look at TTS |
| 00:41.37 | Spirits-Sight | I used the record() thing earlier today, so I know that this can be done |
| 00:42.33 | [TK]D-Fender | Spirits-Sight: Then your single channel is enough |
| 00:44.33 | Spirits-Sight | ok, I am going to move on, when I used Playback(${RECORDED_FILE}) it gave me a error saying that it needed an agurment (filename) which from what I read should be taken from Record(filename.ulaw) is this right or not, |
| 00:45.28 | [TK]D-Fender | Spirits-Sight: Well if you've got an actual * problem, feel free to show us. |
| 00:47.43 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 00:47.52 | Spirits-Sight | the two lines that I have are: Record(Greeting-DIS.ulaw) then next is Playback(${RECORDED_FILE}) the error is: WARNING[23612]: app_playback.c:397 playback_exec: Playback requires an argument (filename) I don't see whats wrong, what am I doing wrong |
| 00:48.16 | nextime | hello all. I'm investigating in how to put a voice file on a remote asterisk server from a fastagi/ami server |
| 00:48.47 | nextime | i'm thinking about try to use CURL(), and if it doesn't exists, try Exec(wget) or something else |
| 00:49.03 | Spirits-Sight | it records the file as if I dial another ext it plays it fine |
| 00:49.05 | nextime | or also if there is any hack that let me write a binary file from ami command updateconfig |
| 00:49.23 | [TK]D-Fender | Spirits-Sight: pastebint he WHOLE thing. Little bits and pieces hide the evidence |
| 00:49.35 | nextime | is there any other (and maybe better) way to do that, assuming that i can't add any additional module to the asterisk server? |
| 00:49.39 | Spirits-Sight | ok one sec please |
| 00:49.39 | [TK]D-Fender | nextime: SSH |
| 00:50.06 | [TK]D-Fender | nextime: you can SCP a file over to it |
| 00:50.07 | nextime | [TK]D-Fender : assume that i don't have any access to the asterisk server except ami and some extension calling my fastagi server |
| 00:50.41 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 00:51.07 | [TK]D-Fender | nextime: us AMI to add an exten that will call System to do the pickup and Originate a call to it |
| 00:51.17 | Spirits-Sight | Do you want the whole ext file or just the context area |
| 00:51.39 | [TK]D-Fender | Spirits-Sight: ALL of the evidence. |
| 00:51.42 | nextime | [TK]D-Fender : yes, this is what i'm thinking to so, but i'm investigating if there is any better way to do it |
| 00:52.09 | [TK]D-Fender | nextime: A better way to do something when you tell us your hands are tied? No, AMI is not some miracle server remote control tool. |
| 00:52.29 | [TK]D-Fender | nextime: 99% of people wouldn't even have come up with the hack I jsut suggested |
| 00:53.12 | [TK]D-Fender | nextime: What are you running for you * server that you are so locked out? |
| 00:54.14 | nextime | [TK]D-Fender : nothing. I'm just writing an application to be distributed to some asterisk ignorant and i want to let the application is more self-configurant that i can |
| 00:54.42 | nextime | and the default/extra sounds files aren't good for my needs |
| 00:55.11 | [TK]D-Fender | nextime: anybody installing a 3rd party tool should not be an idiot incapable of managing their own system. |
| 00:55.47 | nextime | [TK]D-Fender : i can agree with you, but i don't care about this things, i do what i'm payed for. |
| 00:55.55 | [TK]D-Fender | nextime: and this assumes they can even give it rights and that the AMI user will have the appropriate rights to do everything in the first place. A horrific chain of requirements to do something to wrong way |
| 00:56.16 | [TK]D-Fender | nextime: So yes... its a giant, mess "maybe". |
| 00:56.23 | [TK]D-Fender | messy* |
| 00:56.54 | twooten | [TK]D-Fender: I found http://www.chanskype.com/ it seems to be able to connect asterisk to a skype account. Do you think it would make sense to buy a license for $19.00, then use it to make my local and long distance calls using skypeout for .02 cents a minute? |
| 00:56.55 | Spirits-Sight | here is the pastebin: http://pastebin.com/d1836a443 |
| 00:57.19 | nextime | [TK]D-Fender : anyway, i'm not here to discuise about what i need to do, i'm here to see if anyone have a better idea on how to do it. |
| 00:57.24 | twooten | seems like a cheap way to go |
| 00:57.59 | nextime | twooten : you can find better and cheaper rates from many sip providers |
| 00:58.17 | [TK]D-Fender | nextime: nextime With AMI as the carrier? You see the API as well as we do (hopefully). WYSIWYG <- |
| 00:58.19 | nextime | use chanskype only if you really need to talk with skype accounts |
| 00:58.53 | [TK]D-Fender | twooten: No. Any decent ITSP will cost $.02 / min or less WITHOUT stupid limitations |
| 00:59.15 | nextime | [TK]D-Fender : if and when we will have FastEAGI i will not have those problems |
| 00:59.16 | nextime | :) |
| 01:00.21 | [TK]D-Fender | nextime: I won't hold my breath. |
| 01:00.39 | nextime | actually my *only* problem with * is the absence of sounds files |
| 01:00.52 | nextime | for every other thing i have a good and working solution |
| 01:01.02 | nextime | at least for my needs, of course |
| 01:01.16 | [TK]D-Fender | Spirits-Sight: Holy. Shit. Afer all this you failed to give me the ENTIRE call's CLI output so I can see EXACTLY WTF is being executed. |
| 01:02.00 | [TK]D-Fender | Spirits-Sight: And include ONLY the warning and not even the line that generated it. |
| 01:02.07 | [TK]D-Fender | GAH |
| 01:02.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:03.38 | [TK]D-Fender | Spirits-Sight: and the purpose of that variable is for something VERY specifi which is listed in the app's instructions. Something which you are not using it for in the first place. |
| 01:03.46 | twooten | [TK]D-Fender: I looked at http://www.vitelity.com/index.php?p=retailserv and they seem to charge $1.44 per minute. While http://www.voicepulse.com/plans/default.aspx?plan=lnkPlan1 has a $14.99 plan with 200 minutes per month of long distance. I just figured that a one-time payment of $19.00 and a .02 cent a minute skypeout call to any landline, local or long distance would be cheaper. |
| 01:04.21 | Spirits-Sight | [TK]D-Fender: that was the only thing the cli gave beside a timestamp |
| 01:04.56 | [TK]D-Fender | Spirits-Sight: I see no DIALPLAN execution in your CLI output. |
| 01:05.21 | Spirits-Sight | [TK]D-Fender: should it be giving me more because it not, what do I do to get it to give that information |
| 01:05.26 | [TK]D-Fender | Spirits-Sight: AND you are using a variable that A : Is NOT for the purpose you are trying to use it for. and B : NOT NEEDED in the first place |
| 01:06.22 | Spirits-Sight | I was using a toutal instead of keep ask people like your self, I am trying to learn it on more own |
| 01:06.28 | [TK]D-Fender | twooten: Look again http://www.voicepulse.com/connect/Rates.aspx |
| 01:07.09 | [TK]D-Fender | Spirits-Sight: And what did this tutorial say about that variable? And what does that have to do with not providing the COMPLETE call debug? |
| 01:07.41 | Spirits-Sight | THE CLI only gave me what I gave you |
| 01:08.08 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 01:08.18 | [TK]D-Fender | Spirits-Sight: Where is the CODE being called? Why no execution information? |
| 01:08.38 | [TK]D-Fender | Spirits-Sight: Did you completely forget to max out VERBOSE? |
| 01:10.00 | [TK]D-Fender | twooten: http://www.voicepulse.com/connect/default.aspx "US long distance starting at below-a-penny!" |
| 01:12.49 | *** join/#asterisk andresmujica (n=andresmu@190.25.193.216) |
| 01:13.23 | Spirits-Sight | http://pastebin.com/d7da4fbea ok, I think this is what you was asking for now |
| 01:13.23 | nextime | uhmm |
| 01:14.04 | nextime | maybe another option is to generate a call to a sip peer and redirect it to a Record() |
| 01:14.57 | nextime | this way i need to implement a sip stack on my external daemon, but this shuld work with any * installation, indipendently of version or build options |
| 01:15.31 | Spirits-Sight | [TK]D-Fender: it said the var was for calling the file that was recorded |
| 01:15.41 | nextime | i can assume to be in LAN with the * server, so, i can use a good codec without compression, so the quality shuld be acceptable |
| 01:15.50 | [TK]D-Fender | Spirits-Sight: "core show application record" |
| 01:17.03 | [TK]D-Fender | nextime: * works with SIP. You should already know if its broken in some way that preclues use with * |
| 01:17.09 | [TK]D-Fender | precludes* |
| 01:17.35 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:17.35 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:18.05 | [TK]D-Fender | nextime: In general one should be able to believe that it will do what its supposed to and that things will work... |
| 01:18.26 | [TK]D-Fender | nextime: without naning names of course there si no hope for any answer more precise than that.. |
| 01:18.30 | [TK]D-Fender | naming* |
| 01:18.44 | Spirits-Sight | [TK]D-Fender: I have read this, am I doing the Record() wrong? |
| 01:19.19 | nextime | [TK]D-Fender : yes, of course. I'm just doing some brainstorming to find which one of the options is the better one |
| 01:19.25 | [TK]D-Fender | Spirits-Sight: Read it again and again, and again until you realize that that variable has *ONE* speicific* use and it sure isn't yours. |
| 01:19.34 | nextime | where for better i mean "the most portable in any * installation" |
| 01:20.07 | Spirits-Sight | [TK]D-Fender: its only to be used if you have %d used right |
| 01:20.17 | Spirits-Sight | ? |
| 01:20.28 | [TK]D-Fender | nextime: if you have a UA that can reliably pass calls in & our, using Record in the dialplan (regardless fo your having to generate it) would indeed be a way to use "near" only AMI to do this |
| 01:20.40 | [TK]D-Fender | Spirits-Sight: yes... that's what we call BIG PRINT. |
| 01:21.14 | [TK]D-Fender | Spirits-Sight: So A ) that variable is uselees, and B ) you don't need ANY variable at all anyway. |
| 01:21.21 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:22.09 | Spirits-Sight | [TK]D-Fender: my throught was I just put the filename there, would this be the right way of doing it |
| 01:23.01 | [TK]D-Fender | Spirits-Sight: as I said, this isn't VARIABLE. There is no guessing. You provided a HARD name to RECORD, so do the same for your PLAYBACK. its not like you have to guess .... |
| 01:23.48 | Spirits-Sight | I was just following what the tourtal had said |
| 01:25.38 | [TK]D-Fender | Spirits-Sight: What tutorial? And Don't jsut copy & paste some random source of info. Go read the instructions on what you're doing. Otherwise someone will pass you a script with an "rm -rf /" instruction and watch you trash your server |
| 01:27.04 | Spirits-Sight | well I am not that dum I know rm is does either two things rename or remove so I would NOT just follow that blindly |
| 01:27.31 | Spirits-Sight | I am trying to fing the tutorial again to see what I may of done wrong from what they said |
| 01:28.44 | Spirits-Sight | it had the %d, which I did not notice the couple of times I read it |
| 01:28.55 | Spirits-Sight | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Record |
| 01:29.37 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.2) |
| 01:29.55 | [TK]D-Fender | Spirits-Sight: I'm goin to say this nicely (not) Please realize that you are not the target of this next comment.. FUCK THE FUCKING WIKI. |
| 01:30.58 | [TK]D-Fender | Spirits-Sight: Those are NOT official docs, and many of the samples in there are written by well-meaning TWITS. The information is also often dated (not appropriate for your cersion), or worse. Flat out wrong is sometimes the case regardless of everything else |
| 01:31.02 | Spirits-Sight | then where is a good place for me to read and learn from examples |
| 01:31.10 | [TK]D-Fender | spitThat is nearly the last place you should look for information. |
| 01:32.08 | [TK]D-Fender | Spirits-Sight: the BOOK, and the docs in your source folder. Go read the application & function list. one you know how priorities work, "core show applications" and "core show functions" tells you just about everything. The rest is in the * source DOC folder and in the sample files. |
| 01:34.03 | [TK]D-Fender | Spirits-Sight: And you should realize somethingis wrong with your variable when its coming up BLANK |
| 01:34.14 | Spirits-Sight | I don't mean to ask dumb quesion and what not, I am ONLY trying to understand what I need, I don't plan on doing great and big things, I will leave that upto developers in the future, I just want something setup to do what I would like |
| 01:34.34 | Spirits-Sight | I know there was something wrong with the var not working thats why I ask. |
| 01:35.32 | Spirits-Sight | I understand var and how they work, at less I hope see I have done so much reading on var with php and coldfusion in the past |
| 01:35.41 | *** join/#asterisk RobertLaptop (n=rmiddle@12.53.163.106) |
| 01:36.23 | Spirits-Sight | well I read the book a few times already, the areas that covered the dailplan and sip stuff I read a few times |
| 01:36.58 | fakhir | hey i got a question. I am testing out Asterisk 1.6 (actually upgraded to 1.6.1-beta2 to see if that would solve it) but when i try to dial an extension i get -> "WARNING[4540]: app_macro.c:205 _macro_exec: No such context 'macro-stdexten' for macro 'stdexten'" |
| 01:36.58 | fakhir | http://pastebin.com/m2db6c4bd |
| 01:37.05 | [TK]D-Fender | Spirits-Sight: When you assume that Record spits out that varilbe, go prove it. |
| 01:38.08 | [TK]D-Fender | fakhir: Well, where do YOU see that macro defined in your pastebin? |
| 01:38.21 | Spirits-Sight | well we know it does not do so with out the %d which adds a number to the end of the file name and the ${RECORDED_FILE}gives that name to you |
| 01:39.58 | [TK]D-Fender | fakhir: -- Executing [101@default:1] <-- in fact I see no way for that CLI output to match that CONTEXT at all |
| 01:40.07 | fakhir | [TK]D-Fender, ok i will look but i never had to touch any of the sample config in the past when using 1.4.* |
| 01:40.18 | *** join/#asterisk RB2 (n=RB2@pool-71-127-212-121.nwrknj.east.verizon.net) |
| 01:40.42 | [TK]D-Fender | fakhir: [default] soe not have such a line in it. You are showing us apples & oranges or somethin funky is happening behind the scenes |
| 01:41.05 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00159a03312c.cpe.net.cable.rogers.com) |
| 01:41.16 | [TK]D-Fender | Spirits-Sight: that variable is ONLY made when its needed, and it isn't |
| 01:41.35 | Spirits-Sight | correct, I now understand this |
| 01:49.59 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 02:12.04 | fakhir | [TK]D-Fender, heh i dont know what is going on because this is a near new ubuntu/asterisk install i do make samples -> create a user is users.conf and a context in extensions.conf -> reload asterisk -> try to dial myself and get that error :( |
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| 04:09.30 | fakhir | well that solves allot :p copied over the stdexten macro from 1.4 to 1.6 and that seemed to fix my original problem and | != , in 1.6 :p |
| 04:10.38 | fakhir | should read the 1.6 docs more carefully hehe |
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| 08:07.05 | icarrar | Near |
| 08:07.14 | icarrar | Err neat |
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| 08:46.30 | cvnet | hey, how do you set the callerId name on the asterisk? |
| 08:46.50 | mrj | Anyone decently familiar with Switchvox? |
| 08:50.48 | SwK | SET(CALLERID(name)=Foo) |
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| 10:13.38 | invalidrecord | hi guys, is there any good software for generating reports of calls on asterisk from the logs, or do i need to bash something up in ruby? |
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| 10:31.05 | loompek | i have problems compiling asterisk-addons-1.6.1-rc1 |
| 10:31.10 | loompek | [CC] chan_ooh323.c -> chan_ooh323.o |
| 10:31.13 | loompek | something to do with this... |
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| 11:37.02 | masus | is it possible to execute an agi script from the CLI ? Thanks. |
| 11:38.18 | masus | or is it possible only from extensions ? |
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| 13:26.22 | matyk | hey i wanna play with asterisk for some fun now will it work with a normal 56kps modem? |
| 13:26.36 | abalashov | What do you mean by work? |
| 13:26.48 | abalashov | You mean, will voice work over 56k dialup? |
| 13:27.25 | matyk | well i wanna plug the asterisk box into my phone line then make it go over the network |
| 13:27.41 | abalashov | So what part of your 56K connection is it going to utilise? |
| 13:27.51 | abalashov | The clients are not on a LAN, but connected over the modem? |
| 13:28.05 | matyk | well i was wondering if i could plug my phone line in via the modem |
| 13:28.10 | abalashov | Aah. |
| 13:28.17 | frawd | hello there, are all transfers from any channels using ast_channel_masquerade, or is there a common function called when a transfer takes place (ex: when receiving a REFER in SIP)? |
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| 13:28.43 | abalashov | To my knowledge, no, ordinary modem cards are not supported by Asterisk's hardware layer (Zaptel). Special analog interface cards are required, although for a single FXS/FXO port they are quite inexpensive. |
| 13:29.04 | abalashov | This is a little ironic because the analog cards are essentially stripped down and rebadged modem components. |
| 13:29.23 | matyk | so theres no way for me to play without buying parts :( |
| 13:29.34 | abalashov | Well, if you want to have an analog phone line interface, no. :/ |
| 13:29.41 | abalashov | But you can play without doing that. |
| 13:29.51 | abalashov | For instance if you have a home LAN you can run a soft phone on your PC(s)... |
| 13:29.56 | abalashov | And call between your computers. |
| 13:30.08 | abalashov | But as far as connecting to the telephone network at large, no. :/ |
| 13:30.21 | frawd | abalashov, I thought that any software controlable modems (winmodems) would do? |
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| 13:30.36 | abalashov | frawd: That is not my understanding. |
| 13:30.38 | abalashov | But I could be wrong. |
| 13:30.54 | abalashov | However, I've never heard of someone using ordinary winmodem cards with DAHDI/Zaptel. |
| 13:31.22 | abalashov | As I said, they have a lot of hardware that is very similar if not identical, but, it has to be a supported card expressly designed for that purpose. |
| 13:31.48 | matyk | well what i really want is a way to filter out unwanted calls |
| 13:31.58 | matyk | and maybe a voicemail service |
| 13:32.13 | abalashov | Well, what I can tell you is that a single analog line card is about $20 US if you get a Wildcard X100P. |
| 13:32.25 | frawd | abalashov, i take your word for it, no idea how dahdi works... |
| 13:32.28 | abalashov | So they're not expensive, but yes, you do have to buy cards. |
| 13:32.36 | matyk | um |
| 13:32.40 | abalashov | If you have a broadband connection, you can get SIP trunking service. |
| 13:32.46 | abalashov | And bring in calls via VoIP instead. |
| 13:32.56 | abalashov | (and place them as well) |
| 13:33.07 | abalashov | But if you want to connect to it to your home analog line, you need a card. |
| 13:33.16 | matyk | er |
| 13:33.23 | matyk | about the bradband connection |
| 13:33.34 | matyk | i have cable broadband? |
| 13:33.38 | matyk | will it still work |
| 13:33.51 | abalashov | Yes, VoIP should work over that quite well. |
| 13:34.03 | abalashov | So if you had ways of getting calls in and out of Asterisk as VoIP, you can do that. |
| 13:34.08 | abalashov | But you wouldn't be able to use your home phone number. |
| 13:34.15 | frawd | Anyways, is there a developer around that could explain me how transfers work (are they all channel specific or is there a common ground?)? |
| 13:34.33 | matyk | so how do i get the SIP trunking service |
| 13:34.44 | abalashov | frawd: I'm not a developer but I could look in the code for you. |
| 13:34.48 | matyk | and is it free |
| 13:34.57 | abalashov | frawd: But your best is to just look at channels/chan_sip.c and see how REFER etc. is handled. |
| 13:35.15 | abalashov | frawd: I mean, I am a developer, but I don't work on asterisk code (generally) :) |
| 13:36.08 | abalashov | matyk: It is possible to get it free, but in general it has a nominal cost. You seem to be in the UK and I am not sure of the options there, but you can sign up with an infinite number of providers here in the US and may a few cents/min for outgoing calls and a flat rate for inbound number usage. |
| 13:36.22 | frawd | abalashov, thanks, there are calls to some function called ast_channel_masquerade, but it seems to be used for other things like call pickup, ... |
| 13:36.44 | frawd | i was just wondering if someone knew already how it basically work for me to gain time :-) |
| 13:36.54 | frawd | but apparently, everyone sleeps |
| 13:37.01 | abalashov | frawd: Let me take a look. |
| 13:37.11 | abalashov | frawd: 1.6 or 1.4 |
| 13:37.19 | frawd | 1.4.22 |
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| 13:39.50 | abalashov | matyk: In general, when it comes to telephony nothing is completely free, excluding certain types of providers that make money off of intercarrier compensation arbitrage for inbound traffic (which doesn't have wide applicability outside the US). |
| 13:39.58 | abalashov | matyk: I am fairly sure there are some resources of free DIDs out there. |
| 13:40.21 | abalashov | matyk: You can certainly sign up for a trial to play with it. |
| 13:40.48 | abalashov | matyk: And terminating traffic into the PSTN always costs money, so the chances of finding that free are slim. |
| 13:41.01 | frawd | for what i saw, channels implement transfers in their own way (which is logical), but if i'm not wrong they don't seem to call any common pbx function... I want to be able to run something on any transfer situation, independently from the channels, and would prefer not having to patch each and every chan_???.c |
| 13:41.17 | abalashov | frawd: Ah, I see. |
| 13:41.42 | abalashov | frawd: That may be problematic because depending on the channel technology the exact notion of what constitutes a "transfer" in both protocol and higher-level semantics may be very different. |
| 13:42.02 | abalashov | frawd: But it might be reasonable to suppose that they do call some common PBX functions for things like CDR generation. |
| 13:43.09 | abalashov | frawd: But as far as common abstraction layers for accommodating the notion of a transfer beyond administrativia/utility stuff... I doubt it. |
| 13:43.13 | abalashov | frawd: However, I'm just speculating. |
| 13:43.17 | frawd | abalashov, I would have thought that, as you can transfer any channel independently from its technology, there would be at least some minimal common function (for common stuff like CDR) |
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| 13:43.34 | abalashov | frawd: For CDR, I'm sure. But aside from that kind of thing, I doubt it. |
| 13:45.02 | frawd | i didn't see any common things for CDR when transferring, it must be called when channels are bridged or something... |
| 13:45.44 | abalashov | frawd: This is possible. Take a peek at handle_request_refer() in chan_sip.c |
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| 13:46.04 | frawd | i'll look a bit further, thanks for pointing that out |
| 13:46.24 | abalashov | frawd: Or are you talking about when asterisk transfers a call, not when it receives a REFER request from a peer? |
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| 13:47.10 | abalashov | frawd: For the latter, try attempt_transfer() in chan_sip.c. |
| 13:47.11 | frawd | abalashov, yes, the channel specifics don't interest me much, but i'll start looking from there |
| 13:47.30 | abalashov | frawd: There are plenty of ast_cdr_append() calls there. |
| 13:47.42 | abalashov | frawd: And yes, it ultimately seems to call ast_channel_masquerade(). |
| 13:48.16 | abalashov | frawd: But there are calls to ast_quiet_chan() (not quite sure what that does), ast_cdr_append(), etc. |
| 13:48.34 | abalashov | frawd: Aside from that, don't see any calls to anything that looks global. |
| 13:49.19 | frawd | abalashov, where did you see calls to ast_cdr_append? in masquerade or in attempt_transfer of chan_sip? |
| 13:49.30 | abalashov | frawd: attempt_transfer() |
| 13:49.46 | abalashov | frawd: I am looking at masquerade now to see what else is going on. |
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| 13:51.28 | abalashov | frawd: ast_channel_masquerade() is in main/channel.c but there's nothing global that gets called there outside from stuff that manipulates channel-related data structures. |
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| 13:52.49 | abalashov | frawd: ast_channel_masquerade() is called out of chan_sip.c for call parking, attended transfers (INVITE + replaces header) and SIP REFER handling |
| 13:53.09 | abalashov | frawd: So it seems sufficiently idiosyncratic to transfer-related mechanisms. But no, nothing else particularly global. |
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| 13:55.08 | abalashov | frawd: I think you may be out of luck in the sense that there do not seem to be any global abstractions for transfers aside from calls to CDR stuff. |
| 13:55.57 | abalashov | frawd: Whatever exactly is going on in ast_channel_masquerade() seems like the closest to that because that seems to underlie the essence of bridging two channels of different (or similar) technologies. |
| 13:56.23 | abalashov | frawd: It does appear to be used only in transfer-related scenarios, but also call parking, which can be thought of as a form of transfer, and for that reason is reasonable. |
| 13:56.52 | abalashov | frawd: But for further insight I would consult someone who actually works on the core code - that's not me. |
| 13:57.01 | abalashov | frawd: There is a -dev IRC channel. |
| 13:57.13 | abalashov | frawd: Although I doubt many of them are going to be up right now. |
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| 13:58.55 | matyk | im just thinking why wont asterisk work with a modem, i mean voip software would be able to use it to dial out |
| 13:58.57 | abalashov | frawd: It is likely that aside from CDR stamps no global transfer-related functionality was envisioned. |
| 13:59.17 | _matt | hello, i have two connections with support for different audio codecs, when they try to call eachother asterisk says ' No audio format found to offer.' how can I make asterisk convert between the two ? |
| 13:59.21 | abalashov | matyk: It just has to do with the hardware driver layer. It was designed to work with Digium cards originally. |
| 13:59.38 | abalashov | _matt: Are the peers both SIP? |
| 13:59.55 | _matt | abalashov| no, one is sip the other is iax |
| 14:00.17 | abalashov | _matt: Make sure that in both the SIP and IAX peer both codecs are "allowed." |
| 14:00.22 | abalashov | _matt: Explicitly if necessary. |
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| 14:00.51 | _matt | i tried that, but i got Error 500 back from the SIP host |
| 14:01.03 | abalashov | _matt: Strange. What are the codecs? |
| 14:01.34 | _matt | erm... |
| 14:01.47 | _matt | i have used gsm with the SIP host |
| 14:02.09 | _matt | and i think the IAX host is asking for g729 |
| 14:02.23 | abalashov | _matt: Well, here's the thing. The "allow=" lines in the peers only control what Asterisk allows to appear in the SDP offer from a given peer, not what it will transcode necessarily. So you just have to make sure that the preferred codec being advertised by both sides is allowed on the respective peer. |
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| 14:02.55 | abalashov | _matt: But otherwise, I'm not sure. I hate IAX, for one, so I can't really say what's going on there. I'd be curious to see if the same thing happens with two SIP peers. :) |
| 14:03.27 | _matt | abalashov| ok, the SIP peer is an exchnage UM host |
| 14:03.45 | galeras | Weird: i need to make an inbound call to "enable" my zap fxo channels for outbound calls. (TDM410P Card). Any suggestion wellcome! THANKS! |
| 14:03.59 | _matt | i have had it working with an IAX connection from my asterisk box here and everything works fine |
| 14:04.54 | abalashov | _matt: I'm not really sure. But I think the key is to figure out if one of the peers is rejecting the codec offer or if it's Asterisk that's originating the unsupported feedback. |
| 14:05.03 | tzafrir_laptop | galeras, wow, this is with a TDM410P card as well? good to know |
| 14:05.09 | _matt | abalashov| ok |
| 14:05.13 | _matt | abalashov| http://pastebin.com/m1d6f310b .. |
| 14:05.41 | _matt | but asterisk will convert betwen codecs if needed ? |
| 14:06.10 | tzafrir_laptop | galeras, http://bugs.digium.com/view.php?id=13786 |
| 14:06.16 | tzafrir_laptop | What version do you use? |
| 14:06.29 | abalashov | _matt: In theory, yes. But I think the key is to figure out what SIP/EXUM is offering in its SDP offer. |
| 14:06.42 | tzafrir_laptop | I wonder if it is actually fixed (in 1.6.0) |
| 14:07.30 | _matt | abalashov| ok |
| 14:07.33 | galeras | tzafrir_laptop, thanks, trying the patch.... |
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| 14:07.52 | tzafrir_laptop | galeras, what version of asterisk is it? |
| 14:07.58 | galeras | 1.4.22 |
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| 14:17.03 | frawd | abalashov, thanks a lot for the help and research, I saw that not only is ast_channel_masquerade used for other things than transfers (pickup, ...), but it also seems that not all transfers situation call that function. Some of them seem to call ast_async_goto instead... I'll try the -dev channel. |
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| 14:27.54 | galeras | tzafrir_laptop, Thanks a lot, it worked!!! |
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| 14:47.28 | Magicblaze007 | Has anyone bought this from this vendor? http://www.apocalypsecomputers.net/linksys-54mbps-rout. The price looks too good to be true... |
| 14:49.05 | Magicblaze007 | Are there any good competing products to (WRT54GP2) this one that I should consider? |
| 14:49.22 | seanbright | farkus: gotta say... pick a g-d nickname and stick with it |
| 14:49.45 | hi365 | :) |
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| 15:33.43 | matyk | is it possible to plug a modem into my phone line and use some software to get the callers number? |
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| 15:40.30 | abalashov | matyk: Generally speaking, yes. |
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| 15:40.38 | abalashov | matyk: Although the type of software you'd use varies far and wide. |
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| 16:13.04 | simpat1zq | hi everybody. I need some help getting inbound calls working with asterisk. I'm actually using Elastix, but I'm guessing that it is about the same. let me know if it is not. I got a MagicJack, and I got the credentials from it and plugged it into Asterisk. I think that I did it almost right because when I dial the MJ number, it shows up in the logs, but I just get a message saying "The number you have dialed is not in |
| 16:13.17 | simpat1zq | anyone that can help me out with that? |
| 16:13.21 | abalashov | Hmm. |
| 16:16.08 | simpat1zq | This is a brand new instlal. I have no extensions setup either. Not sure if that is a problem |
| 16:21.51 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
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| 16:56.38 | feeds | omg farkus, farkus_ , farkus__ could you stop multi connecting? |
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| 17:04.01 | abalashov | Hehe. |
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| 17:04.52 | [TK]D-Fender | feeds: Strike a deal then... you stop connecting and he'll stop multi-connecting ;) |
| 17:05.22 | feeds | [TK]D-Fender: xD |
| 17:05.24 | feeds | ^^ again |
| 17:06.01 | feeds | [TK]D-Fender: I had to restart my crap PC, got stuck updating packages :D |
| 17:08.10 | drmessano | You rebooted a linux box? |
| 17:08.25 | feeds | yup |
| 17:08.55 | outtolunc | says 'do it again, do it again' |
| 17:09.07 | drmessano | I once swapped out a motherboard, 2 sticks of RAM, built a RAID5 array, replaced the power supply, CDROM drive and the CPU, without a reboot |
| 17:09.09 | [TK]D-Fender | drmessano: Is kinda required for a kernel upgrade... |
| 17:09.17 | drmessano | But then it wouldnt detect my Flash Drive, so I had to reboot |
| 17:09.25 | feeds | drmessano: xD |
| 17:10.46 | abalashov | Self-aggrandising smartass? Where? *looks* |
| 17:12.31 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 17:13.08 | feeds | get braced, have to reconnect (again), don't worry drmessano, I'm not rebooting ;) |
| 17:13.41 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 17:13.42 | drmessano | Yeah well |
| 17:13.54 | *** join/#asterisk feeds (n=feeds@85-135-245-223.adsl.slovanet.sk) |
| 17:13.56 | drmessano | Linux has a guaranteed 102% network uptime too |
| 17:14.09 | drmessano | WHERE YOUR SECOND NIC, FEEDS? |
| 17:14.12 | drmessano | errr |
| 17:14.15 | drmessano | WHERE'S YOUR SECOND NIC, FEEDS? |
| 17:14.22 | drmessano | 102% uptime |
| 17:14.24 | feeds | NIC? |
| 17:14.45 | drmessano | Network Interface Card. The little telephone looking thing in the back the intarweb plugs into |
| 17:16.59 | [TK]D-Fender | drmessano: He's using an ACTUAL telephone plug with an acoustic-coupled modem ;) |
| 17:17.20 | drmessano | ZOMFG, Asterisk on dialup |
| 17:17.24 | drmessano | I am 10-7 |
| 17:17.25 | *** part/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
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| 17:17.47 | drmessano | Oh, SVN is back up |
| 17:17.49 | abalashov | grins |
| 17:18.11 | feeds | has to hold on the desk not to rofl |
| 17:18.18 | drmessano | Did you call me another hyphenated thing again? |
| 17:18.25 | stintel | just let go and rofl ahead :P |
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| 17:23.37 | beek | In IMAP voicemail storage, how is asterisk notified that a voicemail (email) has been read? How does it know to turn off the MWI? |
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| 17:40.24 | [TK]D-Fender | beek: shold be using the "read" flag.. this is e-mail you know... |
| 17:41.43 | Akiyuki | sigh :( |
| 17:44.47 | beek | [TK]D-Fender: My problem is that when I use an email client to "read" the message, I'm wondering how Asterisk knows that I read it. |
| 17:45.38 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 17:45.40 | beek | [TK]D-Fender: I'm looking at the docs for 1.6 and now I see that they have modified the voicemail apps to set the time for polling, so I'm beginning to believe that 1.4 polls the email server for the information. |
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| 17:51.39 | RB2 | I'm having problems with voicemail (happens intermittently),,, the asterisk log shows a warning: func_db.c: DB requires an argument, DB(<family>/<key>) |
| 17:52.06 | RB2 | Any suggestions on where to start looking? |
| 17:52.20 | hardwire | can I please use Tool for my MOH? |
| 17:52.39 | hardwire | I've a feeling some good anti-sheeple music would do the world some good |
| 17:53.34 | *** join/#asterisk ManxPower (n=manxpowe@17.sub-70-221-193.myvzw.com) |
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| 17:54.53 | [TK]D-Fender | RB2: Thats an AstDB warning, not VM |
| 17:55.16 | [TK]D-Fender | hardwire: When in doubt : The Best Of Slayer |
| 17:56.18 | RB2 | [TK]D-Fender, ok. Weird intermittent issue with leaving a voicemail failing I'm trying to track down. |
| 17:56.39 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:57.04 | ManxPower | RB2: What version of Asterisk? |
| 17:57.44 | [TK]D-Fender | RB2: You haven't actually told us the problem... |
| 17:58.56 | RB2 | MaxPower, 1.4.22-rc5. |
| 17:59.21 | RB2 | If you leave voicemail, press # and 1 to save the message. It says the messages has been saved and then says that an error has occurred. The voicemail disappears into the void. |
| 17:59.24 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 17:59.27 | ManxPower | RB2: the bug I was thinking of was fixed much earlier n the 1.4 release cycle. |
| 17:59.35 | RB2 | But, it doesn't happen all the time. |
| 18:00.01 | ManxPower | RB2: how much disk space do you have free on the partition with the voicemail? |
| 18:00.13 | RB2 | 6GB |
| 18:01.08 | puppet | anyone know a good free stunserver for *? |
| 18:01.17 | hardwire | what distro you using? |
| 18:01.31 | hardwire | puppet: ^ |
| 18:01.39 | puppet | hardwire: pbxinabox so centos |
| 18:01.44 | ManxPower | RB2: You really need to get the cli output of a failed call. check in /var/log/asterisk for the logs |
| 18:01.49 | hardwire | debian has the stun package |
| 18:01.51 | hardwire | it works well |
| 18:01.56 | hardwire | dunno about centos |
| 18:02.06 | puppet | ok |
| 18:02.07 | hardwire | http://www.vovida.org/applications/downloads/stun/ |
| 18:02.25 | ManxPower | puppet: Any reason you want to use STUN instead of Asterisk's built in NAT traversal features? |
| 18:02.42 | puppet | ManxPower: Nokia N95 |
| 18:03.24 | ManxPower | That isn't a reason. 8-) |
| 18:03.28 | [TK]D-Fender | puppet: STUN helps the CLIENT fix its IP. * shouldn't need help for itself. If your phone has issues, then maybe your PHONE needs STUN |
| 18:03.33 | RB2 | MaxPower, these three lines are all that show up when the error occurs: http://pastebin.com/d75bd8d1c |
| 18:03.41 | puppet | [TK]D-Fender: exactly |
| 18:04.01 | RB2 | ManxPower, The AstDB error occurs as soon as the voicemail picks up. |
| 18:04.08 | [TK]D-Fender | puppet: So what do you need stun on your server for? |
| 18:04.19 | ManxPower | RB2: I suspect your AstDB is broken. You also need to fix that sip.conf error and start Asterisk as root. |
| 18:04.30 | [TK]D-Fender | RB2: that kind of error you'd be able to see from CLI. the log is unqualified |
| 18:04.48 | ManxPower | RB2: neither of the non-astdb errors should affect VM |
| 18:05.35 | [TK]D-Fender | log files = 99% useless |
| 18:05.36 | ManxPower | RB2: that is very weird. We should be seeing at least app_voicemail running |
| 18:05.41 | ManxPower | [TK]D-Fender: I disagree. |
| 18:05.42 | RB2 | ok, let me bring up the CLI and try the call again. |
| 18:05.52 | [TK]D-Fender | ManxPower: Compared to real live CLI? |
| 18:06.06 | ManxPower | The default logging logs most of the stuff you'd see on the CLI. |
| 18:06.25 | [TK]D-Fender | ManxPower: I don't want some unmatchable warning I'll never be able to trace. Show me something REAL and LIVE and maybe I'll care :) |
| 18:06.59 | ManxPower | [TK]D-Fender: he's not providing any context for the information. It doesn't matter where he gets the info if he does not provide context |
| 18:07.01 | [TK]D-Fender | But then again.. there's good reason why he wouldn't show whats going on "live" if you cath my drift... |
| 18:07.19 | [TK]D-Fender | ManxPower: I'm sure thats part of it |
| 18:07.19 | ManxPower | [TK]D-Fender: You smell a GUI? |
| 18:07.23 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:07.34 | [TK]D-Fender | ManxPower: "/etc/asterisk/manager_custom.conf" <- you think? |
| 18:07.50 | [TK]D-Fender | ~[TK]D-Fender |
| 18:07.51 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 18:08.12 | ManxPower | [TK]D-Fender: *nod* Yes, that does have the stench of a GUI user |
| 18:08.21 | ManxPower | The poor sod. |
| 18:08.38 | RB2 | :-O |
| 18:08.41 | RB2 | hehe |
| 18:09.22 | ManxPower | [TK]D-Fender: I still think #asterisk-cli is a good idea. |
| 18:10.08 | [TK]D-Fender | ManxPower: No, I think simply purging that other element from here is. Why make this a "go to A or B" channel.... there'd be nobody left but a door-man :) |
| 18:10.24 | [TK]D-Fender | ManxPower: Or maybe we could import #asterisk-dev and thats what this would become |
| 18:10.37 | ManxPower | [TK]D-Fender: the only way to purge the undesirables is with Digium's assistance and they won't do that. |
| 18:10.51 | drmessano | GUI users are undesriables? |
| 18:10.56 | drmessano | Thats a bit harsh |
| 18:11.03 | [TK]D-Fender | drmessano: Only the ones trolling for GUI support here. |
| 18:11.09 | ManxPower | drmessano: GUI users are the black plague of this channel. |
| 18:11.12 | [TK]D-Fender | drmessano: They do have their own channels... |
| 18:11.38 | drmessano | They're also fastly becoming the grand majority of the userbase of Asterisk |
| 18:12.11 | ManxPower | [TK]D-Fender: we both know the GUI channels are totally useless -- the blind leading the blind. Personally I don't care if they fall down stairs, I just don't want them polluting this technical channel |
| 18:12.25 | feeds | drmessano: unfortunately... |
| 18:13.06 | ManxPower | Makes about as much sense as a Debian user asking questions on the CentOS channel. They are both linux after all. |
| 18:13.14 | feeds | xD |
| 18:13.23 | RB2 | Yes, I use freepbx for this install. That's not to say I'm oblivious to anything going on beyond freepbx. I think to classify all users that use a gui for convenience as people who have no understanding beyond the gui would be a mistake. That MAY be a majority, but isn't true for everyone. |
| 18:13.25 | ManxPower | But I admit it is a losing battle. |
| 18:13.39 | ManxPower | RB2: It does not MATTER. |
| 18:13.41 | ManxPower | ~freepbx |
| 18:13.42 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:13.58 | feeds | bye gotta go |
| 18:14.06 | drmessano | Well, as much as I dislike Trixbox, so I hate to lump it in, but adoption of Asterisk based systems had increased exponentially thanks to the GUI's and only leads to more development and support of the platform as a whole |
| 18:14.28 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 18:14.39 | ManxPower | I even heard someone teaching an Asterisk class AT DIGIUM referring to "SIP trunks". That's when I realized these things are hopeless. |
| 18:14.41 | RB2 | ManxPower, I get that. |
| 18:14.53 | [TK]D-Fender | drmessano: I don't disagree... just that support should fall into the appropriate place... |
| 18:14.56 | drmessano | Thats about as bad of an argument as "Asterisk runs better on BSD" |
| 18:15.12 | ManxPower | drmessano: It's not the GUI, it's the venue |
| 18:15.33 | [TK]D-Fender | RB2: Anyway, if you want any kind of help for your current situation you'll have to show conclusive info on when that error is generated along with everything surrounding it |
| 18:15.54 | RB2 | btw, as usual, when I jumped into the CLI and left a voicemail, it worked flawlessly. ;) |
| 18:15.58 | ManxPower | I feel that running a GUI is pointless and stupid. But I feel the same way about the cable TV show "Jackass". More power to them, but don't make watch it. |
| 18:16.27 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:16.41 | ManxPower | RB2: doesn't really matter. The CLI output is pretty much useless when running a GUI. |
| 18:16.47 | drmessano | ManxPower: There's a surprising number of very technical users that use GUI's for things like everyday deployments where development time for complex dialplans can be cut dramatically and make the difference between selling someone a PBX and them buying a grandstream PBX.. |
| 18:17.23 | ManxPower | drmessano: You miss the point. This. Is. Not. A. Channel. For. GUI. Users. |
| 18:17.24 | drmessano | "The CLI output is pretty much useless when running a GUI." <-- except its the same CLI output |
| 18:17.28 | ManxPower | They have their own channels. |
| 18:17.34 | drmessano | ManxPower: Says you. |
| 18:17.59 | ManxPower | drmessano: More people say this than just me. |
| 18:18.04 | drmessano | Feel free to take up the Skype argument as well with the devs. |
| 18:18.18 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 18:18.30 | ManxPower | There's a Skype argument? |
| 18:19.14 | [T]ank | is anyone here using a t.38 provider directly to an ATA device? wondering how well it works and what equipment you are using. There are a few providers around offering it. Wanted to get some input before I spend the money on an account. |
| 18:19.18 | drmessano | Well according to Russell, this channel is suited for general VoIP discussion just as much as it is asterisk |
| 18:19.47 | frawd | who is Russell? |
| 18:19.58 | ManxPower | frawd: one of the Asterisk developers |
| 18:20.14 | drmessano | I really dont see the point in occupying twice as much bandwidth arguing about what is and is not acceptable than the actual occasional offtopic convo |
| 18:20.18 | drmessano | Unless there's boredom |
| 18:20.38 | drmessano | and really, it's not "offtopic" if it's encouraged |
| 18:20.47 | ManxPower | Maybe we should all go to the GUI channels and start talking about building your own configs? |
| 18:20.59 | ManxPower | That would be just as rude as the gui people coming here. |
| 18:21.05 | drmessano | This is IRC, you're free to do as you please |
| 18:21.32 | ManxPower | Ah well. It's a lost cause anyway. |
| 18:21.33 | drmessano | Well, I completely disagree |
| 18:21.39 | drmessano | Fucking ass |
| 18:22.06 | [TK]D-Fender | drmessano: Nobody likes the guy going "MMMMmmmM!!!" while eating his Burger King Whopper.... inside a McDonalds. |
| 18:22.14 | drmessano | Generally a user comes in here with a technical issue that they believe is more core asterisk related than help clicking on the GUI |
| 18:22.19 | drmessano | So I dont think its offtopic |
| 18:22.31 | drmessano | They go to the GUI channel for help with the GUI |
| 18:22.35 | drmessano | and many reside in both |
| 18:22.37 | [TK]D-Fender | drmessano: The problem with that guy is he's 99% wrong <- |
| 18:23.02 | [TK]D-Fender | drmessano: "It must be a core * problem!!!" = umm... BS |
| 18:23.19 | [TK]D-Fender | drmessano: Seen it here so many times. |
| 18:23.19 | Carlos_PHX | scratches balls, wonders what all the whining is about. |
| 18:23.39 | drmessano | Ok, so let me ask you this |
| 18:24.00 | drmessano | If I hand coded a dialplan IDENTICAL to one a FreePBX system spit out into config files.. |
| 18:24.16 | drmessano | Used identical contexts, extension setup, etc |
| 18:24.23 | drmessano | Which is all very readable by Asterisk |
| 18:24.29 | drmessano | But had NO freePBX involved |
| 18:24.31 | drmessano | and had a problem |
| 18:24.36 | drmessano | How is that not asterisk? |
| 18:24.46 | [TK]D-Fender | drmessano: And all of their scripts, and since you did it yoursewlf naturally you'd have to be nearly an expert on this.... |
| 18:25.01 | RB2 | drmessano, I had written the same thing, but thought better of it lest Manx go off the deep end. hehe |
| 18:25.17 | [TK]D-Fender | drmessano: Anybody THAT good would drill it down to only the barest of info needed in a PB we could decode with NO issues. |
| 18:25.45 | drmessano | Thats not the point |
| 18:26.03 | [TK]D-Fender | drmessano: If they're that good we wouldn't have reason to wonder wha's going on with their problem, it be so shockingly well layed out that we wouldn't need to care how complex the crap going on behind the scenes is. Problem is that situation simply doesn't happen here. |
| 18:26.18 | drmessano | If I have an issuse that is NOT related to the dialplan itself, which is obviously static due to FreePBX writing it, it is NO LESS an asterisk issue |
| 18:26.34 | drmessano | NAT is NAT, Astdb corruption is AstDB corruption |
| 18:26.46 | [TK]D-Fender | drmessano: Yes it is. They are clueless about whats really going on behind the scenes and we should have to drill though 10 tons of BS include DB's who's actual context we can't know jsut because he's too dumb to pull it himself. |
| 18:27.09 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:27.14 | frawd | i agree with your last statement drmessano |
| 18:27.16 | [TK]D-Fender | drmessano: Yes, NAT is NAT, and there are well written guides for where to set that up in hte GUI's. |
| 18:27.19 | frawd | NAT is indeed NAT |
| 18:27.26 | drmessano | If I wrote my dialplan with-lots-of-many-complicated-contexts it would be NO different than a GUI dialplan |
| 18:27.30 | frawd | cool, everyone agrees! |
| 18:27.37 | drmessano | Jst my lack of ability to the change the thigns that are staticly written |
| 18:28.05 | drmessano | I think there is a complete and total lack of understand what the GUI's actually do here |
| 18:28.14 | drmessano | and lots of stupid zealot bias |
| 18:28.21 | [TK]D-Fender | drmessano: GUI contexts follow their own logic and assume tons of other crap. Also they get BLOWN AWAY on rebuild so effecftively you aren't even steering your own ship. It OWNS your ass. |
| 18:28.29 | drmessano | SO? |
| 18:28.49 | [TK]D-Fender | drmessano: Since you can't even really change it then how to help when the GUI designer fucks up? |
| 18:29.04 | [TK]D-Fender | drmessano: And the next 10 chumps walking in with the same complaint? |
| 18:29.05 | drmessano | So I dont ask you how to fix my context, since I cant.. But when Zaptel is hosed, or I have a problem building Asterisk, I am supposed to ask in FreePBX or blah-GUI? |
| 18:29.07 | drmessano | Wrong |
| 18:30.34 | *** join/#asterisk atta (n=atta@p57A3DF53.dip.t-dialin.net) |
| 18:30.44 | [TK]D-Fender | drmessano: Tell you what, we can debate it on the next case that comes in. |
| 18:30.46 | drmessano | I hate to break it to you, but there are dozens upon dozens of users that flat out LIE when they come in here to get help with a problem they have already determined is outside the GUI, such as Zaptel, installing G729, hardware, compiling, etc, etc.. just to get help, knowing they will be stepped on and yelled at for having the wrong interface |
| 18:31.19 | [TK]D-Fender | drmessano: Oh yes, and that makes them so much more endearing... |
| 18:31.29 | drmessano | Which is incredibly stupid.. "If you are running FreePBX, we hope your asterisk doesnt brake or you need help installing a Digium card.. Go die in a fire" |
| 18:31.37 | drmessano | break* |
| 18:31.54 | drmessano | [TK]D-Fender: They have to LIE to get help |
| 18:32.03 | drmessano | [TK]D-Fender: That makes the CHANNEL less endearing |
| 18:32.07 | seanbright | holy crap... just installed something called FreePBX and... it. is. *awesome* |
| 18:32.21 | [TK]D-Fender | drmessano: And how much of what they need to get things working depends on the GUI & its tools to get them functional? |
| 18:33.11 | drmessano | Zaptel/Dahdi - Very little, compiling Asterisk on Ubuntu - None, Troubleshooting a TDM400 card - None |
| 18:33.39 | frawd | talking about GUIs, which one do you recommend? is FreePBX any good? |
| 18:33.53 | drmessano | Generally the GUI channels DO_NOT_SUPPORT_THE_CORE |
| 18:34.00 | drmessano | They support the _GUI_ |
| 18:34.11 | drmessano | So where do the users go? #asterisk to get browbeat |
| 18:34.14 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:34.40 | [TK]D-Fender | drmessano: And 99% of the config questions they have are answered on FreePBX's own site |
| 18:34.49 | [TK]D-Fender | drmessano: "how to setup card, etc) |
| 18:34.57 | *** part/#asterisk atta (n=atta@p57A3DF53.dip.t-dialin.net) |
| 18:35.01 | [TK]D-Fender | drmessano: And NAT... they did have a nice page on that.... |
| 18:35.10 | [TK]D-Fender | drmessano: So There is little left really.. |
| 18:35.26 | drmessano | [TK]D-Fender: Really? Because even the devs know the website needs a lot of work, and a lot of the info is outdated |
| 18:35.39 | [TK]D-Fender | drmessano: But as I said I'm willing to look at this on a case by case for the next few with you. |
| 18:35.54 | drmessano | You gonna tell me the voip wiki is a complete guide to asterisk too? |
| 18:36.38 | frawd | [TK]D-Fender, what does the [TK] part mean? |
| 18:36.41 | [TK]D-Fender | drmessano: I don't to be "the bad guy" on this and am not so closed-minded. I have seen little reason to change my approach yet however, but am willing to re-evaluate |
| 18:36.48 | seanbright | frawd: team kill |
| 18:36.54 | seanbright | :P |
| 18:36.54 | frawd | thanks |
| 18:37.01 | seanbright | doesn't really know |
| 18:37.07 | frawd | oh... |
| 18:37.13 | [TK]D-Fender | drmessano: WIKI... no... thats far worse for * than older GUI guides are to GUI users... |
| 18:37.27 | frawd | the K must be for Kill or Killer |
| 18:37.34 | frawd | Turtle Killer? |
| 18:37.47 | frawd | Tavuk Kebab? |
| 18:38.04 | drmessano | I just think the whole thing is really, really, really counterproductive for Asterisk.. If I was a new user and installed FreePBX and Asterisk, and got the GUI working, but had a hard as shit time with a Zaptel card, came in here and got beat the fuck up because my FXS didn't have dialtone, asterisk wasnt seeing the card on the CLI, but I happened to be using FreePBX, so "I am an idiot", I would say "Fuck asterisk" and go buy a black bo |
| 18:38.10 | [TK]D-Fender | drmessano: Because the core apps change. In a GUI if something * does changes behind the scenes the GUI viewable part that its related to probably looks almost the same for many cases. That being said the steps Jowe Blow goes through is the same and its a GUI design problem for not accounting for it |
| 18:39.23 | drmessano | Last time I checked, devs like the FreePBX devs for example, have Asterisk changes supported while they're still in early beta.. 1.6 is a good example |
| 18:39.34 | frawd | wikipedia says: .tk is the Internet country code top-level domain (ccTLD) for Tokelau, a territory of New Zealand located in the South Pacific. |
| 18:39.42 | frawd | could it be related? |
| 18:40.27 | [TK]D-Fender | drmessano: Yes, thaqts what I'm saying.. things the end user normally doesn't have to care about. |
| 18:40.28 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 18:40.56 | drmessano | [TK]D-Fender: So? if its a dialplan syntax change and the GUI takes care of it, I AM STILL NOT ASKING FOR DIALPLAN help |
| 18:41.02 | drmessano | your arguement is baseless |
| 18:41.06 | frawd | Tomato Ketchup? |
| 18:41.18 | [TK]D-Fender | drmessano: Any piece of dialplan processing aside from their own 100% hand made stuff is really a GUI config iissuue thought. NAT should be referred to its doc page, Zap cards... for the basics we DO support them mostly here. |
| 18:41.29 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:41.35 | drmessano | Again |
| 18:41.37 | drmessano | And again |
| 18:41.39 | drmessano | and again |
| 18:41.44 | drmessano | This is NOT about dialplan help |
| 18:41.47 | frawd | Tragic Kingdom |
| 18:41.52 | frawd | ? |
| 18:41.55 | [TK]D-Fender | drmessano: Ok, well lets just stop and see for the next case, alright? |
| 18:41.58 | drmessano | We've already gone over that the dialplan is fixed |
| 18:42.42 | *** join/#asterisk frieze (n=frieze@pool-98-113-86-38.nycmny.fios.verizon.net) |
| 18:42.52 | frawd | [TK]D-Fender, are you Tsuyoshi Kohsaka? |
| 18:42.59 | drmessano | [TK]D-Fender: No, I am not going to go there with you.. Because should someone come in here with a FreePBX problem, and be one of those cases where the question had to do with something static, there will be 20 paragraphs of a rant about how FreePBX is "ALWAYS WRONG" |
| 18:43.06 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 18:43.10 | drmessano | So no, I wont agree to some pretext about "the next user" |
| 18:43.30 | drmessano | I made my point, done with my end here |
| 18:43.59 | [TK]D-Fender | drmessano: No, I'm willing to play this out over a much longer caseload. How does 20 sound? 20 cases of GUI user problems? Get some real stats? Would that not be fair? |
| 18:44.07 | drmessano | Obviously this is about being a CLI zealot, and not about the good of asterisk as a whole |
| 18:44.16 | [TK]D-Fender | drmessano: We each have points, I'm now willing to take it to stats. |
| 18:44.44 | [TK]D-Fender | drmessano: The god of * as a whole VS the heath of this channel are 2 separate matters. |
| 18:44.46 | [TK]D-Fender | good* |
| 18:45.07 | frawd | I think I found, you're a member of the "The Killers" band!! |
| 18:45.13 | [TK]D-Fender | drmessano: So as I said I'm willing to see how this plays out in the longer term. |
| 18:46.35 | *** join/#asterisk abalashov (n=sasha@97.81.69.51) |
| 18:46.52 | drmessano | This isn't #asterisk-cli, and regardless of the "ITS ONLY FOR CLI PROBLEMS BECAUSE WE SAID SO" is not only ridiculous, but like women posing as men on IRC to not get trolled, I think you're being increadibly nieve about the number of GUI users that frequent here and are intentionally ambigious about their dialplan just to get core help. I frequent many of the channels, and many of the forums, I know the names |
| 18:47.23 | [TK]D-Fender | drmessano: That other channel was created by ManxPower simply because its hard to separate the wheat & chaff here. |
| 18:47.29 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:48.07 | abalashov | aye. |
| 18:48.50 | [TK]D-Fender | drmessano: No, I am well aware of how many GUI users are in this channel... I keep a LIST so that when they ask qquestion I don't run into surprises later <- |
| 18:49.09 | jblack | drmessano: That's a valid statement. That doesn't mean anything should change. |
| 18:49.11 | [TK]D-Fender | drmessano: You have underestimate me on that one... |
| 18:49.24 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 18:49.31 | jblack | Console setups can be debugged over irc. GUI setups cannot. |
| 18:50.11 | [TK]D-Fender | jblack: Technically yes, with Imagebin, etc... painful, but workable |
| 18:50.38 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:50.47 | [TK]D-Fender | jblack: Thing is how much hand-holding for the "toaster" parts should be done outside of the room dedicated to it? |
| 18:50.48 | jblack | If one is overlaying a gui on top of the core principles, they've got to be stripped away to a common point, where common knowledge can do some good. |
| 18:51.03 | drmessano | Nobody is asking to debug GUI dialplans |
| 18:51.10 | drmessano | I keep hearing this over and over |
| 18:51.17 | drmessano | and it sounds like blah blah zealot blah blah |
| 18:51.42 | [TK]D-Fender | drmessano: I'm dropping this and preparing a list to take stats from this point. |
| 18:51.59 | jblack | I think we're right where we should be. No help on extended products. Either one can remove those abstractions to get help here, or one should go to a channel dedicated to the tool that's being used. |
| 18:53.09 | [TK]D-Fender | jblack: Or more like for parts tied to to. a Corrupt AstDB file as drmessano made as an example? Sure, we can help with that. One should hope the guy can SHOW his problem however. |
| 18:53.13 | drmessano | What you fail to realize is that most GUI users with above-I-just-installed-it intelligence understand YOU CAN FIX THE DIALPLAN.. so guess what.. they go to the GUI channels for help with GUI setup. Nobody asks how to add an extension in FreePBX in here. Thats the RESPECT part that ManxPower hinted at, that he had BACKWARDS. When it comes to CORE problems, they do come here, get yelled at, and move on |
| 18:53.31 | jblack | Oh, sure, I'd think astdb as part of asterisk's core. |
| 18:53.45 | drmessano | Its not? |
| 18:54.29 | [TK]D-Fender | jblack: it is, and would be fully valid. |
| 18:54.36 | drmessano | Database show is not part of asterisk? |
| 18:54.41 | drmessano | help database? |
| 18:54.49 | drmessano | The astdb IS part of the CORE |
| 18:54.53 | [TK]D-Fender | drmessano: so thats it for now. I'm going to keep stats on this and let you know when I've got a substantial case load for comparison. |
| 18:55.08 | abalashov | It is interesting how much of the discursive space of the ecosystem is polluted with folks who don't really understand Asterisk at all - they just unroll Trixbox or FreePBX or what have you - but expect that they're going to get the same level of insight, understanding, debugging, verbosity, and accommodation as someone who actually knows how to configure the product. |
| 18:55.25 | drmessano | [TK]D-Fender: You honestly think I am gonna trust results compiled by someone with bias? No |
| 18:55.47 | jblack | By following that rule, keeping to the core, we're doing another good thing too. We're minimizing devergence. |
| 18:56.08 | drmessano | jblack: You still havent commented on the astdb statement you made that was wrong |
| 18:56.20 | drmessano | Nobody is asking for GUI help |
| 18:56.21 | jblack | I said I think of it as part of the core. How is that wrong? |
| 18:56.23 | [TK]D-Fender | drmessano: Bias? I'm pretty damn fair in splitting A from B. You rant and I'm willing to back it up and do it patiently over time. The person who rejects offers like mine show bias. |
| 18:56.41 | [TK]D-Fender | drmessano: You are the one locked in your belief. I'm willing to be proven wrong in my research. |
| 18:56.49 | [TK]D-Fender | drmessano: Do the same if you think you're "fair"\ |
| 18:57.19 | stencil | should Linux kernel hackers provide support for people having difficulty with gnome desktop |
| 18:57.19 | drmessano | [TK]D-Fender: You are the one that thinks configuring the basic SIP nat settings on a system that has FreePBX on it is a *freepbx problem*.. thats not gonna be unbiased |
| 18:57.22 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:57.28 | drmessano | Wow |
| 18:57.35 | drmessano | Talk about zealot |
| 18:57.46 | abalashov | drmessano: Probably depends on what is meant by "configuring." |
| 18:57.55 | drmessano | NOBODY ASKES FOR FREEPBX HELP IN HERE |
| 18:58.04 | abalashov | If it is being configured the "FreePBX way" and dependent on FreePBX metadata and all that good stuff, it is a FreePBX problem. |
| 18:58.09 | [TK]D-Fender | drmessano: I said I'd class the nature of each case. I never said I would say WHICH SIDE OF THE FENCE each blonged to. Stop assuming these things. |
| 18:58.13 | drmessano | I am talking about help with ZAPTEL, NAT CONFIG, ETC |
| 18:58.15 | abalashov | If it's a "my sip.conf is broken... and the original happened to be generated by FreePBX," probably not. |
| 18:58.16 | drmessano | All done at the CLI |
| 18:58.21 | drmessano | all NOT a part of the dialplan |
| 18:58.24 | drmessano | all NOT a part of the dialplan |
| 18:58.29 | drmessano | Whatever lol |
| 18:58.36 | [TK]D-Fender | drmessano: I'll shove it ALL out on the table fand we can pick the cases in better detail. |
| 18:58.56 | jblack | drmessano: I think you're just in the mood to fight. In the spirit of that, I don't think you're a real doctor. |
| 18:58.59 | [TK]D-Fender | drmessano: Calling us zealots? Look at you raving now... |
| 18:59.39 | *** join/#asterisk djin (n=djin@84-105-70-127.cable.quicknet.nl) |
| 19:00.00 | [TK]D-Fender | ~drmessano |
| 19:00.01 | jbot | [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 |
| 19:00.07 | drmessano | I am not in the mood to 'fight', I am making a valid case.... I make the point that NOBODY is asking for FreePBX GUI help, and someone mentions GNOME like they're not even fucking listening to the part about NOT asking for help in the GUI |
| 19:00.12 | [TK]D-Fender | jblack: jbot seems to think the same... |
| 19:00.15 | drmessano | Honestly, that point is not being taken |
| 19:00.16 | drmessano | At all |
| 19:00.20 | [TK]D-Fender | jblack: For what little that matters :) |
| 19:01.01 | jblack | And while we're at it, I'm not black, so HA! |
| 19:01.08 | [TK]D-Fender | drmessano: and I'm willing to take down every case that comes in while I'm here to qualify the nature of who comes in here and what the problems are. |
| 19:01.19 | jblack | Don't hate me cause I'm not black. |
| 19:01.20 | drmessano | The issue here is that as soon as someone mentions they are running a GUI, their problem is automatically a GUI dialplan problem, and they get trolled |
| 19:01.21 | drmessano | Which is lame |
| 19:01.31 | [TK]D-Fender | drmessano: and willing to have it prove how it really is |
| 19:01.34 | Carlos_PHX | is black from the waist down. |
| 19:01.38 | drmessano | Even if their house is on fire |
| 19:01.41 | drmessano | Its a GUI problem |
| 19:01.42 | jblack | drmessano: Just how I think things should be. |
| 19:01.54 | frieze | anyone had any luck building chan_mobile from asterisk-addons? |
| 19:02.21 | abalashov | The issue is that it often is a GUI problem if only because the problem seems to demand being addressed on a GUI level, mindful of the metadata and constructs used by the GUI. |
| 19:02.37 | abalashov | i.e. using the GUIs' metadata, template structures, etc. |
| 19:02.40 | frawd | frieze, are you using FreePBX or a GUI? |
| 19:02.41 | abalashov | and if you just provide a generic dial plan fix, it won't work |
| 19:02.47 | frieze | frawd: no |
| 19:02.52 | abalashov | I think that's the beef. |
| 19:02.59 | drmessano | This isn't about dialplan problems |
| 19:03.04 | abalashov | Or anything else. |
| 19:03.05 | drmessano | But lets keep saying it is |
| 19:03.09 | drmessano | Because its FUN |
| 19:03.19 | abalashov | dial plan, NAT config, SIP peers, features, voicemail, whatever |
| 19:03.28 | abalashov | same issue applies |
| 19:03.31 | frawd | frieze, what's the problem? missing dependencies? |
| 19:03.33 | x86 | oh... my.... god..... |
| 19:03.34 | x86 | http://images.encyclopediadramatica.com/images/5/5f/Fuckingowned.jpg |
| 19:03.37 | drmessano | Because a NAT issue (externhost, localnet, nat=yes) is surely a GUI problem |
| 19:03.40 | x86 | (potentially nsfw) |
| 19:03.46 | abalashov | It could be. |
| 19:03.46 | drmessano | It has to be a GUI problem |
| 19:03.52 | drmessano | abalashov: No, its not |
| 19:04.05 | SHUBDORRUSH | x86: stay away from there :P |
| 19:04.22 | frieze | frawd: compile problem when I build it -- I tried using it when I had 1.4 but since it was apparently being written for 1.6 I assumed that it would actually work when I installed 1.6 |
| 19:04.24 | drmessano | abalashov: But see, like many people who haven't used a GUI, you're talking out your ass |
| 19:04.31 | drmessano | abalashov: "It may be" |
| 19:04.32 | drmessano | No |
| 19:04.33 | frieze | chan_mobile.c:792: error: request for member ‘ptr’ in something not a structure or union |
| 19:04.37 | frieze | like that and more |
| 19:04.56 | x86 | SHUBDORRUSH: yeah lol |
| 19:05.10 | jblack | You should fork #asterisk, drmessano. Make a #asterisk-guisok |
| 19:05.14 | abalashov | No, I am not asserting that this is necessarily the case. But it could be. The problem is that the observational language - the metaphysics, if you will - of how the problem is formulated is often in the terms of the GUI and the abstraction and functional layers, therein. |
| 19:05.27 | abalashov | And don't invite help, even if the problem is unrelated to the GUI management layer. |
| 19:05.29 | drmessano | jblack: No need, open source darwinism will take care of the ignorance |
| 19:05.35 | [TK]D-Fender | jblack: Hold off on that... I made my offer. Do you agree with the collection? |
| 19:05.56 | frawd | frieze, 1.6.0? |
| 19:05.58 | jblack | I didn't see all the details. A survey of who's using what, right? |
| 19:06.06 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:06.19 | drmessano | A survery of whatever-it-ends-up-in-the-end |
| 19:06.21 | frieze | I checked addons out from svn trunk |
| 19:06.37 | frieze | is there a package I should have downloaded instead? |
| 19:06.42 | abalashov | frieze: Some sort of header that defines those data types missing? |
| 19:06.51 | abalashov | frieze: Other errors accompanying it, preceding it or following, in context? |
| 19:06.56 | [TK]D-Fender | jblack: A survey of the precise nature of every GUI-user's problems and what was needed to correct. This was we can say X NAT issues, Y GUI Route config issues, Z, Zaptel issues, etc |
| 19:07.26 | jblack | hmmm. |
| 19:07.40 | abalashov | lol HIIII im using trixbox & my problem is shits busted plz fix k thx bai!!!11 |
| 19:07.51 | drmessano | abalashov: Ever used a GUI? |
| 19:07.55 | drmessano | abalashov: Even tried one? |
| 19:07.56 | abalashov | drmessano: Sadly. |
| 19:08.10 | abalashov | drmessano: I've used FreePBX and Trixbox extensively. |
| 19:08.18 | abalashov | drmessano: But not really of my own volition no. |
| 19:08.32 | jblack | [TK]D-Fender: I'd trust results from you, but I don't think you'll find anything interesting. I think you'll end up with the same script for "problem/identification/solution" 95% of the time. |
| 19:08.37 | drmessano | abalashov: So you should be much less ignorant of where the line is between Dialplan and core |
| 19:08.45 | frieze | abalashov: there's a data structure (I can't remember which -- I looked it up last time about 8 months ago) that changed between asterisk versions. chan_mobile appears to be assuming that it has not |
| 19:08.58 | [TK]D-Fender | jblack: I'm willing to accept whatever the stats say. |
| 19:09.05 | abalashov | frieze: Well, on the line where the undefined data type is used, what are its contents? |
| 19:09.29 | jblack | They'd be the closest thing to facts anyone would have, so you should. |
| 19:09.39 | [TK]D-Fender | jblack: jblack Mind you we will be able to evaluate how each is class AFTER collection. I may only include a preliminary classification at best |
| 19:09.57 | abalashov | [TK]D-Fender: How did this discussion arise? What was the essence of the controversy? |
| 19:10.12 | [TK]D-Fender | abalashov: This ihas beena point of contention for YEARS |
| 19:10.26 | abalashov | [TK]D-Fender: Someone alleged that GUI users get categorically trolled without reference to the precise nature of their problem? |
| 19:10.34 | frieze | pvt->fr.data.ptr = pvt->io_buf + AST_FRIENDLY_OFFSET; |
| 19:10.40 | [TK]D-Fender | abalashov: Nothing new here, I'm just willing to help settle it and maybe help set a new tone or approach for the channel |
| 19:10.48 | [TK]D-Fender | abalashov: that too. |
| 19:10.54 | jblack | That, you won't do. I'd put money on it. |
| 19:10.59 | frieze | there's a frame data structure that changed the way it dealt with the data pointer in it |
| 19:10.59 | abalashov | [TK]D-Fender: What's the real point of contention, to sum it up as broadly as possible? |
| 19:11.03 | [TK]D-Fender | jblack: I'd try anyways. |
| 19:11.34 | abalashov | [TK]D-Fender: To me it seems like people who really understand Asterisk vs. users that use prepackaged distributions without really doing a whole lot of learning. The latter inevitably running into problems and requiring help, and the former being what the latter feel is "elitist" about it. |
| 19:11.45 | frawd | frieze, what version of asterisk do you try those trunk addons on? |
| 19:11.52 | abalashov | [TK]D-Fender: Is that a correct interpretation? |
| 19:11.56 | jblack | I'd like to save you 10 years of emotion pain by suggesting you adopt, WGAF. |
| 19:12.03 | drmessano | Nothing will change in here as long as the good of the future of Asterisk as a telephony toolkit comes second to how people are actually using it |
| 19:12.09 | drmessano | And that I will put money on |
| 19:12.12 | jblack | With IIWII. |
| 19:12.18 | [TK]D-Fender | abalashov: About this channel's suitability to assist those running a GUI install having no basic understanding of * whatsoever and havin g us debug their issues which they are largely incapable of properly demonstating, and the solution for which is documented in their GUI (or related resources), or in their own channel, etc |
| 19:12.37 | abalashov | I see. |
| 19:12.45 | drmessano | Because when it comes down to it, if you dont use asterisk how people think you should, youre a fucking idiot |
| 19:12.58 | frieze | frawd: and abalashov: thanks, but nevermind. when you asked I realized that I apparently didn't check out 1.6-current but some other package labelled similarly |
| 19:13.03 | frieze | it just built |
| 19:13.09 | jblack | I don't think that. I just don't care about their problem. |
| 19:13.27 | drmessano | jblack: then GTFO.. this is a chat/help channel |
| 19:13.41 | frawd | frieze, ok! |
| 19:13.53 | frieze | but while i have your ear I may as well ask: when modules in addons say they have mysqlclient(E) as a requirement, what should I be installing? |
| 19:13.55 | abalashov | Well, it seems to me that what I originally said is probably the most pragmatic policy - look at how the problem is formulated. If it's being articulated in GUI terms and requires a solution that is applied in the methodologies of that GUI manager and its way of doing things, probably not valid. If it's an essentially Asterisk problem, seems to be put that way, and the user is capable of appreciating lower-level discourse in response applicable to Asteri |
| 19:13.56 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:14.03 | jblack | So, you think I should help everyone, rather than picking and chosing problems I want to solve? |
| 19:14.12 | [TK]D-Fender | drmessano: Here's a though... FreePBX is being sponsored by bandwidth.com which is growing in strength with their adoption of FreeSWITCH. IIRC there is also talk of converting FreePBX over to it as a platform. Can you corroborate? |
| 19:14.41 | drmessano | jblack: No, picking and choosing for yourself is fine.. Imposing that on others is the problem |
| 19:15.10 | drmessano | [TK]D-Fender: Not converting.. Actually supporting BOTH as OPTIONS.. |
| 19:15.10 | jblack | If my disinterest is an imposition to someone, then that's their problem. |
| 19:15.29 | [TK]D-Fender | drmessano: So should they proceed down that route, does that help *? |
| 19:15.39 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 19:15.40 | drmessano | [TK]D-Fender: Dunno, but THIS isn't |
| 19:15.50 | [TK]D-Fender | drmessano: Keep it in mind.. |
| 19:15.55 | drmessano | [TK]D-Fender: THIS doesnt make the case for continuing to use ASTERISK |
| 19:15.58 | jblack | jameswf: Cat fight. /part while you still can. |
| 19:16.01 | drmessano | [TK]D-Fender: THIS is a good read to NOT |
| 19:16.08 | drmessano | rason |
| 19:16.11 | drmessano | grr |
| 19:16.12 | drmessano | reason too |
| 19:16.21 | [TK]D-Fender | drmessano: When the GUI is what people pick, and nowt whats underneath, then what is underneath loses importance. |
| 19:16.27 | jameswf | Dr Pepper didnt learn from open office or codeweavers thir site wont load :( |
| 19:16.44 | drmessano | [TK]D-Fender: Not true.. The GUI doesnt command all |
| 19:16.46 | seanbright | want your free can? |
| 19:16.57 | [TK]D-Fender | drmessano: Which mind you is one of the reasons I LOVE Asterisk, but fails the "strengthen *'s position" standing |
| 19:17.20 | [TK]D-Fender | drmessano: When the GUI is what people see, if that shifts, * loses market-gain. |
| 19:17.24 | drmessano | [TK]D-Fender: Since you know very little about FreePBX, I dont think you really understand what is needed to be configged under the hood.. Theres a 70/30 split and that 30 is the problem |
| 19:17.36 | [TK]D-Fender | drmessano: Because of GUI users who simply follow their interfaces evolution |
| 19:17.40 | drmessano | There is a lot that needs to be touched in the CLI |
| 19:18.02 | drmessano | But again, I have used the products, understand the line.. It's not all GUI |
| 19:18.08 | jblack | Guys, time for http://i6.photobucket.com/albums/y208/Japtman/retard.jpg |
| 19:18.10 | Roxton | spouts polemics about convention over configuration. |
| 19:18.31 | [TK]D-Fender | drmessano: Got a guess as to what % of FreePBX users in here are actually running Trixbox? I already know that #freepbx doesn't want to hear about them either... |
| 19:18.58 | [TK]D-Fender | drmessano: so of that "30", how many are left that we want to care about between both channels? |
| 19:19.01 | abalashov | Again, I would say the political question of how accommodating the channel should be is probably best addressed by looking at how the question is posed. Is it someone looking for FreePBX help, or for Asterisk help? And what sort of answer do they seem to be looking to receive, an Asterisk answer or a GUI answer? |
| 19:19.04 | drmessano | Trixbox doesnt use FreePBX anymore.. so no, its not supported there.. its forked and is DIFFERENT now |
| 19:19.05 | jameswf | I want my free dr pepper in 4-6 weeks damnit |
| 19:19.07 | drmessano | SO thats not accurate |
| 19:19.26 | drmessano | Apples and oranges |
| 19:19.56 | [TK]D-Fender | drmessano: in a way I agree... it is so much worse.. and does that help * in your eyes really? Knowing that Trixbox is so alienating the tools beneath them that sure * is getting out there now, jsut that few will want to support it? |
| 19:20.35 | [TK]D-Fender | drmessano: So thats part of the quest. Not jsut "gui, yes/no", but also "what distro, extra tools, from package? Whose repo?", etc |
| 19:21.02 | [TK]D-Fender | drmessano: I recall so many complaints about the FreeBSD ports tree and *'s poor record there... |
| 19:21.12 | [TK]D-Fender | drmessano: From people key in following it |
| 19:21.27 | drmessano | Trixbox is getting Asterisk platforms in places only blackbox PBX's and Skype boxes would have gone before.. I dont see how it hurts.. Forking FreePBX hurt the config side because its now a "kinda the same/kinda different" GUi that is supported by less of a community.. But its still asterisk |
| 19:21.45 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:23.09 | drmessano | So when Asterisk is running on 3COM, Avaya, Nortel, and Cisco boxes as "_The_ premiere PBX platform", then I suppose the only REAL asterisk user is one using Linux that downloaded the tarballs and compiled? That sounds a bit like debian vs redhat to me. |
| 19:23.27 | drmessano | Youre kinda digging your own hole here |
| 19:24.09 | abalashov | No, that's not the question. The question is whether those users should go to 3Com/Avaya/Nortel/Cisco for their support, or go to the Asterisk core users. |
| 19:24.32 | [TK]D-Fender | drmessano: When that day goes, I'll give you the shovel :) |
| 19:24.35 | [TK]D-Fender | comes* |
| 19:24.39 | drmessano | If its a NAT problem and its the same 3 config options, who care what the name of the fucking file is? |
| 19:24.55 | abalashov | drmessano: As long as the user knows how to appropriate the response, it is NOT a problem. |
| 19:25.06 | abalashov | drmessano: The problem, I think is when the user says, "What? sip.conf? I don't understand. I'm using Trixbox..." |
| 19:25.33 | drmessano | abalashov: THAT is the crux of my argument.. even when they DO, they get browbeat |
| 19:25.42 | abalashov | drmessano: When they DO, they shouldn't. |
| 19:25.52 | drmessano | God damnit |
| 19:25.55 | drmessano | "As long as the user knows how to appropriate the response" |
| 19:25.55 | abalashov | drmessano: When they don't, well.. I don't advocate browbeating anyone, but ... |
| 19:25.57 | [TK]D-Fender | damns it |
| 19:26.00 | drmessano | When they DO that ^^^^ |
| 19:26.12 | drmessano | When they DO appropriate the response |
| 19:26.24 | drmessano | They get beat up over it, just the same |
| 19:26.30 | outtolunc | coming from you that sounds funny |
| 19:26.34 | abalashov | drmessano: If they understand the response in core asterisk terms, good, then it is not much consequence they are running Trixbox or whatever - it is their problem. |
| 19:26.37 | jameswf | thinks he has missed the oppurtunity to troll |
| 19:26.39 | abalashov | drmessano: If they don't, well... |
| 19:26.43 | drmessano | I have supported FreePBX for a couple years now |
| 19:26.47 | abalashov | I'm sorry. |
| 19:27.04 | drmessano | I know the difference, and I know a lot of users that do |
| 19:27.59 | drmessano | and it doesnt make a difference.. if you have a NAT problem, or your Zaptel that download the tarball for wont compile, that FreePBX running on there makes you somebody elses problem with the compile |
| 19:28.25 | abalashov | I agree that is probably unreasonable. |
| 19:28.34 | drmessano | THAT sir, is my issue |
| 19:28.51 | abalashov | I'm sure it's not often as clear-cut and simple as that, but in principle I concur. |
| 19:28.55 | drmessano | Nobody is asking that FreePBX GUI gets supported on the GUI side |
| 19:29.01 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:29.09 | abalashov | What about when they want support on the non-GUI side that has hooks into the GUI's way of doing things? |
| 19:29.09 | drmessano | RARE has someone been so stupid as to expect that in here |
| 19:29.21 | drmessano | They go to #freepbx for that |
| 19:29.29 | drmessano | and I see that with AsteriskNOW as well |
| 19:29.33 | abalashov | I've run into plenty of folks who think FreePBX, Trixbox, Asterisk... it's all the same. |
| 19:29.34 | abalashov | :/ |
| 19:29.40 | abalashov | But not on here no. |
| 19:29.44 | drmessano | True, and I dont disagree that some users are problems |
| 19:30.04 | drmessano | But I think its VERY lame to tag users who happen to have a GUI when its a CORE problem |
| 19:30.10 | abalashov | Well, your position seems very reasonable. Why would someone disagree? |
| 19:30.27 | drmessano | Like chan_mobile.. FreePBX doesnt even pretend to touch chan_mobile yet |
| 19:30.34 | drmessano | All raw config |
| 19:30.40 | abalashov | *nod* |
| 19:30.44 | abalashov | Makes sense. So where's the controversy? |
| 19:31.02 | abalashov | If someone's disagreeing, is it because they disagree about the extent and the significance of the problem? Or that they simply think all GUI users should step. |
| 19:31.10 | drmessano | "Why would someone disagree?" <-- You tell me.. As I said, its the general tone |
| 19:31.37 | drmessano | "Or that they simply think all GUI users should step." "Oh, you have FreePBX? GTFO" |
| 19:31.37 | abalashov | The general tone would presumably embody prejudices accrued through actual experiences with presumptuous users... I would think. |
| 19:33.05 | outtolunc | i always thought the general tone was due to those gui users being unable to *any* support from the gui providers, then coming here and thinking we should support the gui products |
| 19:33.31 | drmessano | abalashov: Perhaps.. But the same can be said for users with REALLY bad hand config'ed dialplans.. But they can't be dismissed with "Youre using CLI? GTFO" as easily |
| 19:33.36 | drmessano | I think its convenient |
| 19:33.45 | drmessano | Nobody is asking for GUI support |
| 19:33.47 | abalashov | When GUI users post to the mailing lists they typically don't have these problems. |
| 19:34.02 | abalashov | If they ask a question that seems to be centered around freePBX's internal mechanics, they usually don't get much luck. |
| 19:34.24 | abalashov | If they post a question formulated in the parlance and structure of Asterisk as such... then the fact that they are using a GUI seems immaterial. |
| 19:34.25 | drmessano | abalashov: Thats a fine example of my problem.. The same reiteration about "GUI users coming here for GUI help".. when you seem to understand that is NOT the issue |
| 19:34.31 | drmessano | But it gets repeated, over and over |
| 19:34.44 | abalashov | So why are you taking this cause up? Has it impacted you deeply? |
| 19:34.45 | drmessano | and from my experience, help configuring a GUI is the LAST thing people come in here for |
| 19:35.02 | jameswf | GUI = Generaly u'r an idiot.... |
| 19:35.05 | jameswf | sorry all i had |
| 19:35.11 | drmessano | They come in here with needing help with some core part of asterisk that FreePBX or whatever-GUI doesnt touch |
| 19:35.14 | drmessano | and they get run off |
| 19:35.42 | abalashov | So their only liability is that they let the fact that they're using a GUI slip? If it's not materially relevant, why do they mention it? Just because it seems generally relevant somehow? |
| 19:35.55 | *** join/#asterisk saftsack (n=saftsack@p57924F90.dip.t-dialin.net) |
| 19:36.10 | drmessano | abalashov: I take this up because (1) I do use both GUI's and hand config and (2) It not only hurts those trying to set up a new PBX PERIOD, but it hurts asterisk too |
| 19:36.41 | abalashov | Is the channel considered a serious support forum? I don't know, I really haven't hung around here much. I'm mostly a mailing lists guy. |
| 19:37.14 | drmessano | abalashov: Because some people are too stupid to keep their mouth shut about details in Open source channels. Want help with Apache vhosts? Dont mention its on windows, even if the config is identical :) |
| 19:37.17 | outtolunc | i always have fun explaining to gui users that to apply a patch they need to use the asterisk source, when the gui they are one uses packages |
| 19:37.36 | outtolunc | then the their little world just implodes |
| 19:37.53 | abalashov | But generally I've always thought IRC channels surrounding projects are generally cabals of somewhat cliqueish insiders who have all known each other in various ways connected to the project... either have been users/contributors since the early years of the codebase, or met at conferences, or whatever. I really have no idea if that's the case here. I have never really gone to them looking for serious answers to serious questions - it all depends on thei |
| 19:38.01 | drmessano | outtolunc: Thats no different than installing straight asterisk from RPM or DEB |
| 19:38.02 | abalashov | But I don't know what this channel's prevailing culture is in regards to that. |
| 19:38.05 | drmessano | Thats NOT a GUI problem |
| 19:38.11 | drmessano | Has NOTHING to do with GUIs |
| 19:38.31 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:38.49 | drmessano | I hear that Digium has some RPMs floating around now thanks to the work that Qwell has done with Asterisknow.. I guess those RPMs wont be supported? |
| 19:39.11 | drmessano | They are just asterisk afterall |
| 19:39.29 | drmessano | No GUI |
| 19:39.33 | outtolunc | haha |
| 19:40.35 | abalashov | In other words, I don't get the sense this channel is structured to be a serious support forum that has some contextual relevance to the business environment surrounding project. It seems more like a social / hobbyist cultural artifact, although I don't really know. |
| 19:40.42 | drmessano | Theres an awful lot of FreePBX users that compile asterisk from source anyway |
| 19:41.05 | abalashov | So, why's it such a big deal? Maybe everyone here's just a stuck-up elitist who thinks if your setup isn't 100% raw you lack Chi. |
| 19:41.18 | Carlos_PHX | abalashov: That's not correct. There are often social aspects particularly nights and weekends. |
| 19:41.24 | drmessano | lol |
| 19:41.35 | abalashov | Carlos_PHX: Forgive me I really don't know. |
| 19:41.39 | abalashov | Carlos_PHX: Trying to find out. |
| 19:41.48 | Carlos_PHX | I know, that's why I'm just saying. |
| 19:42.03 | abalashov | The rest of the time, ordinary personal and business users are encouraged to come here to get real-time help? |
| 19:42.06 | abalashov | Like, in earnets? |
| 19:42.08 | abalashov | earnest |
| 19:42.09 | drmessano | abalashov: Maybe that is the problem.. maybe this is not meant to be a support forum in any sense, and it just another IRC garbage can like #do_me_baby |
| 19:42.23 | drmessano | So probably like asking them to support BeOS would be the same |
| 19:42.26 | Carlos_PHX | You can do that right now, but don't complain if there's also a debate or other stuff going on. |
| 19:42.46 | abalashov | drmessano: That's what I've generally found with project IRC channels. Some of them can be quite interesting and insightful - not necessarily garbage - but certainly not meant to be a support forum as such. |
| 19:43.16 | Carlos_PHX | The GUI stuff, I don't really care one way or the other, but GUI users do seem to have a lot of issues that can't be resolved unless you know the GUI, even if the issues seem to be "just Asterisk". |
| 19:43.34 | abalashov | Carlos_PHX: exactly. That's kind of the idea I was attaching to earlier. |
| 19:43.38 | drmessano | abalashov: #apache is about like that.. You are a fucking moron.. Period. Your problem is beneath them, IRC, and the world |
| 19:44.41 | jameswf | being a fucking moron way better than a virgin moron |
| 19:44.43 | drmessano | abalashov: the other thing is that this channel can be 115% helpful at times.. but mention some non-standard use or a GUI, and you may as well GFY |
| 19:44.43 | abalashov | drmessano: Yep. Well, if that's the prevailing incumbency, isn't it futile to challenge it? It seems like most serious support requests transpire elsewhere. The only thing that could motivate someone to come here would be a need for real-time support and/or possibly some aspiration to instant gratification. |
| 19:45.07 | abalashov | What other nonstandard uses are generally besmirched and derided besides GUI installs? |
| 19:45.11 | drmessano | abalashov: I come here for the dip |
| 19:45.16 | drmessano | abalashov: mmm spinach |
| 19:46.41 | jameswf | BLT dip is pretty good |
| 19:46.48 | drmessano | An example would like using asterisk to screen calls, or (at times yes, other times no) using asterisk to bridge a legacy PBX to a T1 or SIP |
| 19:46.57 | abalashov | Huh. |
| 19:47.33 | abalashov | Sociologically, it is acknowledged that hackers (and "elite users") are both somewhat technically elitist and motivated primarily by whether something seems intellectually and/or technically compelling, which I imagine run-of-the-mill Trixbox problems do not. |
| 19:48.04 | abalashov | So I suppose they're probably going to confine their enthusiasm for interesting-sounding problems (but still ones that fit within their conception of what is "standard" and therefore "appropriate") and be rather curt with others. |
| 19:48.41 | [TK]D-Fender | abalashov: non-standard? The only real point we have issues with are those where the user is not in control of their situation, can't describe whats going on, or pin down their problems at all. Those that fight about bugs in old unsupported versions that have been fixed alter yet refuse to upgrade, etc. |
| 19:48.55 | drmessano | I know I am gonna hear "oh, so thats what this is about".. but another example was asking about SIP TCP in 1.6.. Had a few problems with it.. Onyl TCP client I had handy was windows messenger to test with.. and it worked fine calling my IVR.. just couldnt transform from my TCP extension to a UDP.. I later tested with Xlite and had the same problem, but when I mentioned Messenger, I had a windows problem |
| 19:49.15 | drmessano | Even though it WORKED |
| 19:49.33 | [TK]D-Fender | abalashov: I've seen people do some pretty interesting things with * that don't resemble a PBX in many ways. I myself have stated that you can use * as a CRON replacement, a jukebox, or any other number of silly sounding things. I used to use * to make me COFFEE |
| 19:49.47 | abalashov | Well, Microsoft's SIP implementation isn't known for its.. er.. standards adherence. I could see the temptation to dismiss it as irrelevant. I certainly got that impression head-on trying to integrate with Office Communicator. |
| 19:50.02 | drmessano | abalashov: Sure, but that was the only thing that WAS working |
| 19:50.22 | drmessano | abalashov: Asterisk was clearly the problem.. and no amount of explanation gets you past that |
| 19:50.25 | drmessano | Even evidence |
| 19:50.40 | abalashov | So asterisk TCP<->UDP conversion wasn't working correctly? |
| 19:50.55 | drmessano | But if I was using GNU_SIP_TalkToMe-TCPEdition, I would have been spot on |
| 19:51.07 | drmessano | yeah.. Oddest damn problem |
| 19:51.21 | Carlos_PHX | abalashov: I like problems that are technically compelling and will make me money. |
| 19:51.24 | abalashov | [TK]D-Fender: I agree. |
| 19:51.27 | abalashov | Carlos_PHX: Me too! |
| 19:51.51 | abalashov | drmessano: So, what evidence did whoever was talking to you have for the assertion that the MS client is the culprit? |
| 19:52.02 | abalashov | drmessano: Did they look at some packet captures or output logs? |
| 19:52.28 | drmessano | abalashov: My TCP adventure started with wanting to use Asterisk with MS Exchange 2007 Unified Messaging.. its been done, theres forum posts on it, that all reflect the same. I didnt dare mention that bit, just the part I needed about TCP not working :) |
| 19:52.39 | abalashov | *nod* |
| 19:52.40 | drmessano | abalashov: I have done little since.. I got pissed and havent touched it |
| 19:52.52 | abalashov | Yeah, I was trying to get it to work with Office Communicator .. no sucess. |
| 19:52.54 | abalashov | success |
| 19:53.04 | abalashov | the SIP and particularly SDP implementation on the OCS side is far too broken |
| 19:53.18 | abalashov | but I was using 1.4 and OpenSER as a TCP<->UDP bridge |
| 19:53.28 | abalashov | unfortunately the mangling capabilities didn't help me any |
| 19:53.47 | drmessano | I agree.. OCS is dead end.. Exchange seems to be working though.. I believe that is due to the "PBX as you are" stuff, and the need to come to some middle ground with SIP based PBXs |
| 19:54.08 | drmessano | More middle would have been UDP, but you know :) |
| 19:55.34 | *** join/#asterisk ElCheapo (n=elcheapo@d199-126-36-20.abhsia.telus.net) |
| 19:55.40 | [TK]D-Fender | drmessano: MS : "Standards are whatever we delare with the name you thought you knew, its jsut really works the way we tell you is does and the rest of the wold has inter-op problems" |
| 19:55.48 | drmessano | lol |
| 19:55.54 | [TK]D-Fender | drmessano: Quite true |
| 19:56.07 | [TK]D-Fender | drmessano: Considering their attempts to steer WWW standards,e tc |
| 19:56.39 | drmessano | For the most part I agree.. But again, this goes back to "PhushMail doesn't connect to my POP server" "What it is?" "Exchange 2003" "Oh, fuck MS".. <-- Not necessary |
| 19:57.12 | drmessano | Actually, Exchange 2007's IMAP is faster than what GMAIL is using.. heh |
| 19:57.38 | seanbright | i'd assume gmail was homegrown |
| 19:57.51 | drmessano | Ehhh |
| 19:57.52 | seanbright | gmail's IMAP implementation, rather. |
| 19:58.13 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:58.23 | drmessano | I dunno about homegrown.. I would assume they started off with something, then hacked at it to make it more suitable.. I think thats what they do with most of their core |
| 19:58.48 | seanbright | yeah, good call. |
| 19:59.07 | drmessano | Grantid... they do more than most |
| 19:59.35 | drmessano | They may start off with postfix and end up with something you couldnt recognize in a car crash with an 18-wheeler |
| 19:59.45 | seanbright | i really should have updated my resume a couple times in the last 3 years |
| 19:59.53 | seanbright | i'm having a hard time remembering what i did 2 jobs ago |
| 20:00.40 | drmessano | You banged out widgets |
| 20:00.53 | outtolunc | you complained you were going to be fired alot |
| 20:00.55 | outtolunc | hehe |
| 20:01.02 | drmessano | When the widget banging needed improvement, you implemented |
| 20:01.12 | seanbright | outtolunc: have we met? ;) |
| 20:01.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:02.57 | abalashov | MS doesn't say the rest of the world has interop problems. |
| 20:03.09 | seanbright | they just don't talk about it |
| 20:03.10 | abalashov | they just believe themselves to have defined the "new" standard |
| 20:03.14 | abalashov | and call everything else "legacy" |
| 20:03.26 | abalashov | i.e. asterisk's "legacy" SIP stack |
| 20:03.37 | seanbright | well... |
| 20:03.51 | seanbright | there is definitely some room for improvement |
| 20:04.01 | abalashov | Didn't you hear? In the "non-legacy" SIP stack we throw 0.0.0.0 in the Contact address and SDP media endpoints no matter what you do. |
| 20:04.26 | abalashov | Yes, but room for improvement != what's existing is fundamentally noncompliant due to its out-of-dateness. |
| 20:06.31 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:06.47 | abalashov | There may definitely be some lacking in feature support, but the essence of the UAC core is very much RFC3261 compliant. |
| 20:07.12 | abalashov | Aside from some minor controversies about trivial details that don't fundamentally *break* interaction. |
| 20:07.30 | abalashov | Which is a whooole lot more than I can say for, say, the MS Office Communicator Mediation Server. |
| 20:08.07 | drmessano | abalashov: I wont even get into them pushing for SIP URI dialing and using "Your email address as your phone address" like, "a lot of years ago" |
| 20:10.22 | abalashov | Oh, I'm sorry, does the IETF owe them some patent settlements then? |
| 20:11.45 | [TK]D-Fender | Microsoft follows NFC's "No Fucking Comment"..... |
| 20:12.21 | [TK]D-Fender | And encourage others to do the same |
| 20:12.51 | jaytee | ooooh, [TK]D-Fender said a naughty word! I think I'm starting to have a bad influence on people in here :-) |
| 20:13.23 | [TK]D-Fender | jaytee: I sanitize most of my commentary, but this was not directed at a person :) |
| 20:13.35 | abalashov | OFCs ("Our Fucking Comments"). And there are no drafts. |
| 20:13.53 | [TK]D-Fender | jaytee: Most people who know me in person rarely see what I'm capable of... its a burden sometimes :p |
| 20:13.55 | drmessano | abalashov: I was referring to them being ahead of the game in that |
| 20:14.08 | jaytee | damn, I come back to the puter for a sec and have to leave soon and the topic gets interesting. |
| 20:14.13 | abalashov | drmessano: Zoom, zoom, look at them go. |
| 20:14.14 | drmessano | abalashov: Versus 2007 "Oh hey, SIP URIs are cool" |
| 20:14.27 | abalashov | I think that idea predated 2007 considerably. |
| 20:14.34 | jaytee | abalashov, have you worked with Mediation Server at all? |
| 20:14.37 | drmessano | abalashov: For MS it did |
| 20:14.40 | abalashov | But I still don't think anybody thinks they're cool, whether they're listening to MS or not. |
| 20:15.05 | drmessano | abalashov: MS was pushing for it back when Exchange messaging was their "IM" |
| 20:15.07 | abalashov | jaytee: What do you mean by "worked with?" My job was to try to figure out how to get any sane SIP UAS to talk to it. Someone else set it up. |
| 20:15.16 | jameswf | Fatal error: Maximum execution time of 30 seconds exceeded in c:\inetpub\websites\CodeAndTheory\freeDrPepper\account\VML.php on line 15 DAMNIT I want my drpepper in 4-6 weeks |
| 20:15.33 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:15.57 | abalashov | jaytee: I try to avoid touching MS stuff if possible, especially if anyone has the naive delusion they're going to speak standardised protocols. |
| 20:15.59 | jaytee | abalashov, ah. Ok was just wondering if you'd gotten Asterisk 1.6 to talk to it. |
| 20:16.04 | abalashov | Sadly, no. |
| 20:16.13 | abalashov | It has several bugs related to Contact and SDP handling. |
| 20:16.36 | abalashov | i.e. implementation FAIL |
| 20:16.58 | drmessano | abalashov: M$'s biggest problem 8 years ago was that they had a vision for SIP <> SIP URI dialing replacing PSTN, and implemented it in LCS, but they failed to address the legacy bits.. and its taken far too long to adopt SIP at a mass level |
| 20:17.22 | jaytee | so basically the patches for tcp in 1.6 are to fix bugs when using * with MS products, not bugs specifically in Asterisk since someone figured MS would take forever to fix their own "issues"? |
| 20:17.35 | abalashov | I do not know. |
| 20:17.36 | drmessano | "PBX as you are" has been a good marketing move.. telling me I CAN use Asterisk and not be "Wrong" |
| 20:17.54 | abalashov | drmessano: Aside from it being a lie. |
| 20:18.03 | drmessano | jaytee: Depends.. Is lack of TCP support a bug? |
| 20:18.49 | abalashov | jaytee: In my case I wasn't worried about Asterisk or TCP support - I used a SIP proxy in between that supports both. It was forwarding just fine. The problem is that OCS sent complete garbage further up the session establishment flow, as far as anyone with a cursory knowledge of the SIP or SDP specs is concerned. |
| 20:19.12 | abalashov | jaytee: Doesn't matter the direction of the call. |
| 20:19.17 | drmessano | abalashov: They're not ignorant to the fact that PSTN exists, and they dont natively support it (yet). I think some of it may be wishful avoidance, buying time until everyone is using VoIP services direct to the core of their UM |
| 20:19.41 | abalashov | the PSTN isn't going anywhere, and MS needs to stay out of telephony |
| 20:19.49 | abalashov | it's not their core competency, along with everything else that's also not |
| 20:20.24 | Carlos_PHX | The PSTN is dead, they just don't know it yet. |
| 20:20.26 | abalashov | Well, Windows can be okay. Except in the mornings. And in the daytime. And evenings. And night time. And always. |
| 20:20.34 | abalashov | No, unfortunately it is not. |
| 20:21.43 | abalashov | If I am to judge from OCS's SIP implementation, MS should get the fuck out of telecom. Way over their head. |
| 20:22.04 | abalashov | I hope for their sake they've done better, being the "visionaries" they are. |
| 20:22.31 | drmessano | Someone has a better SIP implementation? |
| 20:22.40 | drmessano | Wait.. that would require a standard |
| 20:22.43 | jaytee | abalashov, I'm using * 1.4 with sipX as a proxy to Exchange UM and it works fine in both directions as far as UM functions. |
| 20:23.17 | abalashov | There's enough of a standard that you can achieve basically viable interop, although I agree that interop is one of the most difficult and pervasive obstacles about SIP. |
| 20:23.21 | abalashov | But it's a matter of degree. |
| 20:23.35 | abalashov | And in the case of OCS it is basically noncompliant and throws manifest garbage. |
| 20:23.44 | abalashov | Not some subtle quirk. |
| 20:24.04 | abalashov | But basic, overt errors of semantics. |
| 20:24.23 | *** join/#asterisk ElCheapo (n=elcheapo@d199-126-36-20.abhsia.telus.net) |
| 20:25.01 | duztbunny | i think the asterisk-community should be carefull about being critical of things like sip complaince |
| 20:25.13 | drmessano | glass houses? |
| 20:25.17 | duztbunny | totally :> |
| 20:25.21 | drmessano | Indeed |
| 20:25.32 | duztbunny | asterisk rocks, but it's not perfect like most other things out there |
| 20:25.39 | abalashov | For good reason. Trust me, I know how SIP interop goes - I had a job at a telco where all I did was try to find SIP interop bug workarounds for proprietary platforms <-> each other, sometimes asterisk |
| 20:25.45 | abalashov | No, it's not. |
| 20:25.49 | abalashov | But it's a matter of degree. |
| 20:26.04 | abalashov | And the degree to which, say, OCS is "noncompliant" is... logarithmically different. |
| 20:26.06 | drmessano | Asterisk is less sucky than most |
| 20:26.21 | duztbunny | it's gotten much better |
| 20:26.27 | drmessano | I agree too |
| 20:26.57 | abalashov | There's a difference between 'basically functioning UAC perhaps with some interop issues' and 'complete FAIL.' |
| 20:27.03 | drmessano | Im actually using 1.6.0.. Who the hell would have thought a 1.x.0 asterisk would be useable? lol |
| 20:27.04 | abalashov | And MS has achieved nothing but the latter with OCS. |
| 20:27.05 | drmessano | But it is |
| 20:28.36 | abalashov | What's missing from the asterisk SIP stack at this point is support for a lot of extensions, extended / overlay methods, and draft features. |
| 20:28.48 | abalashov | messenging, presence stuff, yadda yadda |
| 20:29.39 | abalashov | oh, and "directrtpsetup" needs to work. WTF. It's not that hard to proxy two SDP offers to each other. |
| 20:30.01 | abalashov | (i.e. a non-reINVITE based mechanism for media release) |
| 20:31.45 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:37.26 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:39.49 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:41.56 | kb3ien | Is there a sleep() command in asterisk? i cannot seem to find it. |
| 20:42.29 | seanbright | no, there isn't. |
| 20:42.35 | seanbright | there's a Wait() |
| 20:42.50 | seanbright | core show application Wait |
| 20:42.51 | kb3ien | not for a pid or a mutex, just for time to pass. |
| 20:42.59 | seanbright | core show application Wait |
| 20:43.17 | seanbright | (type that in the asterisk CLI ^^^) |
| 20:43.44 | kb3ien | that a horrible misnomer. |
| 20:43.52 | kb3ien | s/that/that's/ |
| 20:43.55 | seanbright | what is? |
| 20:44.03 | kb3ien | Wait() for Sleep(). |
| 20:44.11 | seanbright | ok |
| 20:44.36 | seanbright | if you wait for 15 seconds, i'll see if i can find something else |
| 20:44.39 | *** join/#asterisk l0uis (n=l0uis@madmax.fitnr.com) |
| 20:44.42 | seanbright | zing! |
| 20:45.11 | *** part/#asterisk l0uis (n=l0uis@madmax.fitnr.com) |
| 20:45.19 | drmessano | I think you put the room to Sleep() |
| 20:45.26 | drmessano | :( |
| 20:45.36 | hardwire | lame |
| 20:45.49 | drmessano | I got lost on the double pun |
| 20:45.55 | drmessano | Horrid fail |
| 20:46.01 | abalashov | usleep(), isleep(), wsleep() |
| 20:46.26 | seanbright | is going to stab farkus |
| 20:46.36 | hardwire | drmessano: go right ahead, I'm often the lame one. |
| 20:46.38 | hardwire | I could use a break |
| 20:46.38 | drmessano | "Asterisk doesn't Sleep(), it Wait()s." <---- HA.. Chuck Norris-ism |
| 20:46.43 | drmessano | I WIN |
| 20:47.56 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:48.53 | hardwire | yup |
| 20:49.00 | hardwire | you're the weener. |
| 20:49.58 | hardwire | weener weener weener |
| 20:50.20 | *** part/#asterisk abalashov (n=sasha@97.81.69.51) |
| 20:50.59 | Carlos_PHX | So these two Asterisk guys walk into a bar... |
| 20:52.36 | [TK]D-Fender | (ouch) |
| 20:56.17 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:56.43 | drmessano | But the bar was non-compliant |
| 20:57.31 | drmessano | HA |
| 20:58.25 | Carlos_PHX | It was stupid, because you'd think the second guy would have seen the first guy do it. |
| 20:58.26 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 20:58.31 | [TK]D-Fender | drmessano: FOO! |
| 20:58.45 | [TK]D-Fender | Carlos_PHX: They were dead-locked |
| 20:58.54 | Carlos_PHX | Single-threaded |
| 20:58.59 | drmessano | Two asterisk guys walk into a bar.. One orders a beer, the other just eats peanuts.. The guy with the beer says to the other guy "Hey, you're not drinking?". The other guy says "No, I can't even take a SIP, i'm an alcoholic".. So the first guy says "Well you know, there's probably a PATCH for it." |
| 20:59.02 | drmessano | HA.. PATCH |
| 20:59.03 | creativx | j asp |
| 20:59.45 | Carlos_PHX | Two Fonality guys walk into a bar... |
| 20:59.52 | Corydon76-dig | OUCH! |
| 20:59.57 | drmessano | The two that didnt get fired? |
| 21:00.04 | Carlos_PHX | Are there two? |
| 21:00.29 | Carlos_PHX | Two Fonality guys walk into an open-source bar... |
| 21:00.43 | drmessano | Theres a rumor that Fonality is being run by imported asian slave labor while they try to hire engineers to work for free |
| 21:00.47 | drmessano | But I dont know |
| 21:01.15 | Carlos_PHX | notices all his Fonality contacts are offline in IM. Hmm |
| 21:01.19 | drmessano | I called them for support on a Trixbox and they offered me Peking Duck |
| 21:03.58 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:04.02 | drmessano | I hate you farkus |
| 21:04.05 | drmessano | oops |
| 21:05.56 | [TK]D-Fender | drmessano: Seems "/ignore" doesn't work on "joins", etc |
| 21:09.21 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 21:10.14 | seanbright | just ban him |
| 21:10.24 | seanbright | chanserv, get on that. |
| 21:11.08 | Carlos_PHX | ~chanserv |
| 21:11.09 | jbot | rumour has it, chanserv is your mother |
| 21:11.51 | seanbright | ~chanserv |
| 21:11.52 | jbot | i guess chanserv is Carlos_PHX's mother |
| 21:11.55 | seanbright | much better |
| 21:13.40 | drmessano | Ha |
| 21:13.52 | drmessano | Guess what me and chanserv did last night :P |
| 21:14.44 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:14.49 | seanbright | stab stab stab |
| 21:15.06 | Carlos_PHX | drmessano: Chanserv said something about a performance problem. |
| 21:15.24 | Carlos_PHX | Somebody's ports didn't probe or something. |
| 21:15.50 | seanbright | not enough RAM |
| 21:16.08 | [TK]D-Fender | lol |
| 21:17.26 | seanbright | dude... seriously? |
| 21:17.54 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 21:18.34 | Carlos_PHX | is trying to decide whether to go to Fry's for some parts or just stab self in the eye with a fork. Equal pain level most likely. |
| 21:20.47 | [TK]D-Fender | Carlos_PHX: Yes, but with Fry's you could potentially leave with the part you're looking for... your choice is clear. "just pain", vs "Pain & the part I'm looking for, or close enough" |
| 21:21.16 | Carlos_PHX | True, there is a slim but non-zero chance that I will find a non-defective part that I actually need. |
| 21:21.45 | Carlos_PHX | On the other hand, if I stab myself with a fork I will know what I'm getting. |
| 21:21.53 | Carlos_PHX | Question is, am I a gambling man? |
| 21:21.57 | [TK]D-Fender | Carlos_PHX: True, but there is pretty much a zero chance of the part falling out of thin air upon stabbing yourself with a fork. Slim > none |
| 21:22.15 | [TK]D-Fender | Carlos_PHX: Are you feeling lucky, punk? |
| 21:22.50 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:22.50 | Carlos_PHX | "If there is a god, may he strike me in the head with an 8800GT video card when I stick this fork in my eye" |
| 21:22.59 | *** join/#asterisk Segnale007 (n=Pietro@host142-255-dynamic.19-79-r.retail.telecomitalia.it) |
| 21:23.52 | *** join/#asterisk Segnale007 (n=Pietro@host142-255-dynamic.19-79-r.retail.telecomitalia.it) |
| 21:24.33 | *** join/#asterisk Segnale007 (n=Pietro@host142-255-dynamic.19-79-r.retail.telecomitalia.it) |
| 21:25.48 | *** join/#asterisk Segnale007 (n=Pietro@host142-255-dynamic.19-79-r.retail.telecomitalia.it) |
| 21:28.15 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:29.33 | [TK]D-Fender | Wow, farkus has found a soul-mate |
| 21:30.20 | seanbright | long distance relationships never work out |
| 21:35.52 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:44.22 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 21:45.10 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:46.02 | *** join/#asterisk root52 (n=root52@ip70-191-120-39.cl.ri.cox.net) |
| 21:47.43 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 21:48.06 | kb3ien | longdistance relationships dont tend to result in marriage, but they can be helpful and revealing to those who travel that path. (small observation) |
| 21:48.40 | Carlos_PHX | ~longdistancerelationships |
| 21:48.52 | Carlos_PHX | Huh, you'd think that would be there. |
| 21:49.25 | Akiyuki | ~ooma |
| 21:49.37 | *** join/#asterisk andresmujica (n=andresmu@190.25.106.65) |
| 21:49.38 | Akiyuki | Some new VoIP provider in the US. Selling equiptment at Bestbuy |
| 21:51.29 | kb3ien | oh? |
| 21:52.17 | Akiyuki | Just was checking on the offchance that there were any SIP devices at bestbuy in town. |
| 21:52.21 | kb3ien | 'voip provider' and 'selling equipment' in the same sentance scare me. BESTBUY was the icing on the icy-feeling cake. |
| 21:52.24 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:52.48 | Akiyuki | :> |
| 21:53.17 | Carlos_PHX | Pretty sure it's another P2P voice service. There are others. |
| 21:53.28 | Akiyuki | Those are lame. |
| 21:54.03 | drmessano | HEH |
| 21:54.09 | Akiyuki | Is it possible to use the vonage ATA with asterisk, by changing /etc/hosts to point the vonage service to your own ip? |
| 21:54.26 | drmessano | No |
| 21:54.52 | drmessano | Ooma is interesting |
| 21:54.57 | drmessano | I can see how they do it |
| 21:55.48 | Carlos_PHX | So I get like 50 Ooma devices and connect them to one our gateways... |
| 21:56.58 | drmessano | Basically, they take advantage of the fact that they can make money off inbound calls on every customer.. So the expense for them is outbound.. They basically wholesale out the termination and cut their load for the average residential user to under $10 a month.. and there's always saving from ENUM and P2P |
| 21:57.19 | drmessano | For $250, thats a lot of months of service for a $5 expense |
| 21:57.52 | Akiyuki | I need to start a scam like that |
| 21:58.19 | Akiyuki | Did you read the link, "Ok, how does this company make money?" at the bottom, drmessano ? :) |
| 21:58.27 | drmessano | Even $10 a month means they need to go 25 months on the same box before you end up losing on their purchase |
| 21:58.51 | [TK]D-Fender | regrets not going back in time to kill Hitler and decides to shoot Akiyuki NOW to save everyone future grief. |
| 21:59.13 | Akiyuki | falls over |
| 21:59.26 | Akiyuki | utters his last words... "if only i had read the manual" |
| 21:59.49 | drmessano | Well, if you dont buy premiere service, that leaves what I stated above |
| 22:00.33 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:00.36 | drmessano | And if you break down and buy a new box after two years, the counter resets to 0 |
| 22:01.28 | *** join/#asterisk write_erase (n=Olivier@royale.aixmarseille.com) |
| 22:01.35 | Akiyuki | I need to just break down and either buy a real hardphone, or a real ATA and use my current phone at home. |
| 22:01.51 | Carlos_PHX | Or both. |
| 22:02.05 | drmessano | or several |
| 22:02.07 | Akiyuki | Lets not go that far :) |
| 22:02.12 | Akiyuki | Its hard for me to part with money. |
| 22:02.25 | drmessano | Thats what we love to hear |
| 22:02.36 | carrar | build one then |
| 22:02.37 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 22:02.40 | write_erase | Any tools for tracking & profiling channel locks ? |
| 22:02.41 | Carlos_PHX | ~cheap |
| 22:02.41 | jbot | cheap is, like, a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 22:02.46 | Carlos_PHX | :-p |
| 22:03.25 | drmessano | Apache = free webserver, Asterisk =! free telephony |
| 22:03.29 | Akiyuki | I don't like buying online :/ |
| 22:03.42 | Carlos_PHX | Apache =! free internet |
| 22:03.44 | Akiyuki | I am scared to use my credit card, lol. I try to purchase things only in person. |
| 22:04.07 | seanbright | =! ? |
| 22:04.16 | seanbright | silly non-programmers. |
| 22:04.21 | Carlos_PHX | ~=! |
| 22:04.26 | Carlos_PHX | Damn |
| 22:04.36 | seanbright | jbot, a == b |
| 22:04.49 | seanbright | jbot, 3 + 9 |
| 22:04.50 | jbot | 12 |
| 22:04.53 | seanbright | good boy |
| 22:04.56 | Akiyuki | Do all GS products suck? Or just the phones? |
| 22:05.30 | Carlos_PHX | Usually, an ATA has the same basis as the phones. |
| 22:05.49 | Carlos_PHX | They are quite similar right up to the analog interface, then it's just a matter of a different interface. |
| 22:05.53 | Akiyuki | Found a GS ATA on telephony depot for around ~25 |
| 22:05.59 | Carlos_PHX | The underlying IP stuff is all the same. |
| 22:06.05 | Carlos_PHX | ~cheap |
| 22:06.06 | jbot | cheap is, like, a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 22:06.17 | drmessano | Whats wrong with =! ? |
| 22:06.27 | drmessano | oh |
| 22:06.27 | Carlos_PHX | != |
| 22:06.29 | drmessano | != |
| 22:06.33 | drmessano | doh |
| 22:06.44 | Carlos_PHX | Yeah, and what's worse is then I typed it like you did. |
| 22:06.46 | Carlos_PHX | We suck. |
| 22:06.53 | drmessano | :( |
| 22:06.59 | drmessano | ~kill me |
| 22:07.00 | jbot | ACTION shoots a magneto-ionized pseudotachyon gun at drmessano |
| 22:07.00 | ix33 | GS is fine to play with |
| 22:07.13 | ix33 | i have 2 that work just fine. just have to power cycle them every month or so ;) |
| 22:07.15 | Carlos_PHX | Ok, my punishment is to go to Fry's now. |
| 22:07.26 | Akiyuki | I don't want to spend $200 on an IP phone or ATA. |
| 22:07.36 | drmessano | GS is like a tinkertoy.. make a PBX out of tinkertoys, pretend the spoke is a FXO |
| 22:07.36 | seanbright | jbot, 2^32 |
| 22:07.37 | jbot | 34 |
| 22:07.42 | drmessano | Then make a real PBX |
| 22:07.43 | seanbright | nooooo |
| 22:07.43 | [TK]D-Fender | Akiyuki: Do you have a land-line you would use with *? |
| 22:07.51 | Akiyuki | no |
| 22:07.57 | ix33 | drmessano: that's pretty much what i did |
| 22:08.06 | ix33 | Akiyuki: buy a polycom IP 330 |
| 22:08.08 | seanbright | jbot, 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 * 2 |
| 22:08.09 | jbot | 2147483648 |
| 22:08.24 | [TK]D-Fender | Akiyuki: Then get a Linksys PAP2T-NA or SPA-3102 |
| 22:08.24 | drmessano | Out of tinkertoys? |
| 22:08.24 | drmessano | Really? |
| 22:08.24 | ix33 | you can find for around $80 |
| 22:08.24 | seanbright | will stop |
| 22:08.31 | drmessano | Dont ever buy a grandstream |
| 22:08.34 | ix33 | drmessano: out of GS's |
| 22:08.45 | ix33 | then i got my company to buy 40 polycomms and everything was good |
| 22:08.45 | drmessano | jbot, 250 + grandstream |
| 22:08.52 | drmessano | :( |
| 22:08.54 | seanbright | he does factorials too! |
| 22:09.02 | seanbright | jbot, 123! |
| 22:09.02 | jbot | methinks 123! is 12,146,304,367,025,329,675,766,243,241,881,295,855,454,217,088,483,382,315,328,918,161,829,235,892,362,167,668,831,156,960,612,640,202,170,735,835,221,294,047,782,591,091,570,411,651,472,186,029,519,906,261,646,730,733,907,419,814,952,960,000,000,000,000,000,000,000,000,000 |
| 22:09.06 | seanbright | heh |
| 22:09.20 | ix33 | jbot, 3! |
| 22:09.21 | jbot | from memory, 3! is 6 |
| 22:09.27 | Akiyuki | [TK]D-Fender: Thats not bad. That is affordable actually. And linksys makes a pretty good product. |
| 22:09.30 | drmessano | jbot, (6*6)+(6*6)-72 |
| 22:09.31 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:09.39 | drmessano | loser |
| 22:09.46 | seanbright | he can't handle that? lame. |
| 22:09.49 | drmessano | jbot, ( 6 * 6 ) + ( 6 * 6 ) - 72 |
| 22:10.03 | seanbright | he doesn't grok 0 |
| 22:10.05 | drmessano | jbot, 6 * 6 + 6 * 6 - 72 |
| 22:10.08 | seanbright | like the romans |
| 22:10.08 | drmessano | jbot, 6 * 6 + 6 * 6 - 71 |
| 22:10.08 | jbot | 1 |
| 22:10.11 | drmessano | AHH |
| 22:10.16 | drmessano | jbot, (6*6)+(6*6)-71 |
| 22:10.16 | jbot | 1 |
| 22:10.19 | drmessano | YAY |
| 22:10.21 | seanbright | wow... |
| 22:10.25 | seanbright | i was making that up |
| 22:10.28 | seanbright | but i guess it's true |
| 22:10.34 | ix33 | er IP 320 |
| 22:10.35 | drmessano | jbot, (6*6)+(6*6)-73 |
| 22:10.36 | jbot | -1 |
| 22:10.37 | ix33 | Akiyuki: http://www.voipsupply.com/polycom-ip-320 |
| 22:10.38 | seanbright | jbot, 1 - 1 |
| 22:10.39 | drmessano | Sweet |
| 22:10.45 | seanbright | he doesn't get 0. weird. |
| 22:10.47 | ix33 | if you don't need a phone w/built-in switch |
| 22:11.16 | ix33 | once you figure out how to set it up correctly, you'll have learned a marketable skill as well |
| 22:11.38 | Akiyuki | Have you purchased from that company before? |
| 22:11.43 | drmessano | Get a grandstream if you dont need an ATA with a built in ability to operate properly |
| 22:11.56 | ix33 | Akiyuki, actually yes |
| 22:12.11 | ix33 | drmessano: i've only ever bought GS phones now. i can't speak to their silly gateway boxes |
| 22:12.26 | drmessano | They're just as bad |
| 22:12.28 | drmessano | just as bad |
| 22:12.31 | drmessano | Oh sorry, echo |
| 22:12.35 | Akiyuki | drmessano: you purchased one? |
| 22:12.35 | ix33 | lol |
| 22:12.42 | drmessano | ~grandstream |
| 22:12.43 | jbot | well, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 22:13.00 | ix33 | ah my bt101 isn't that bad... |
| 22:13.01 | drmessano | Akiyuki: Fuck no, I would NEVER buy one and not kill myself |
| 22:13.11 | Akiyuki | lol |
| 22:13.32 | Akiyuki | Then how do you know they are bad? Just from trying to diagnose problems here? And user feed back? |
| 22:13.36 | drmessano | Akiyuki: See, I LIKE asterisk.. I dont hate it and like put Grandstreams on it |
| 22:13.54 | [TK]D-Fender | Akiyuki: I'f you're that cheap and willing to risk disappointment on a flakey product, then go right ahead. Just consider how little you're risking it for. |
| 22:14.11 | ix33 | Akiyuki: is this a home setup or what? |
| 22:14.15 | Akiyuki | yes |
| 22:14.29 | Akiyuki | ix33: A guy at my office has the bt101 its pretty neat.. its the black one w/ the blue interface |
| 22:14.36 | drmessano | Akiyuki: If you want to justify getting a grandstream by the whole "Doesnt sound like you guys really know a lot about them" argument, its been used here 172 times.. this week |
| 22:14.51 | kb3ien | hrm these polycoms 650s are being nice to me at long last. |
| 22:14.51 | drmessano | If you want to buy one, buy one.. we dont care. |
| 22:14.59 | drmessano | Have fun with the poor quality |
| 22:15.02 | Akiyuki | Uh, I think you have a chip on your shoulder or something? |
| 22:15.07 | drmessano | No |
| 22:15.09 | drmessano | Not at all |
| 22:15.11 | Akiyuki | I was just meerly asking what kind of problems you ran into with it. |
| 22:15.20 | drmessano | Never owned one.. never will |
| 22:15.27 | kb3ien | i'd like to find a 'reseller' who returns his calls, but hey. i'll get the upgraded licenses sooner or later. |
| 22:15.58 | ix33 | Akiyuki: i use 2 grandstreams at home. but i deployed a 40-line system at work using polycomms. there is no comparison on quality. |
| 22:16.00 | [TK]D-Fender | Akiyuki: there's a reason they're called BarbieTones... |
| 22:16.09 | Akiyuki | oh |
| 22:16.31 | ix33 | Akiyuki: i think i paid $40 for the GS's, and i wish i had paid double that for one ip320 |
| 22:16.38 | [TK]D-Fender | ix33: Sure there is. Things like "Night and day", etc... |
| 22:17.12 | Akiyuki | I will probably stay away from the GS brand. I was just wanting to see other people's nightmare stories, so I could see why they are so bad. |
| 22:17.52 | ix33 | Akiyuki: you don't know how much you take voice quality for granted day-to-day until you try to talk on a GS. |
| 22:18.02 | ix33 | Akiyuki: it will drive you nuts. |
| 22:18.06 | Akiyuki | Ok :) |
| 22:18.37 | joat | then again, i never had a problem with them (once the firmwares were updated) |
| 22:20.10 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:21.19 | ix33 | anybody ever do the snom 370 through its openvpn client? |
| 22:23.01 | kb3ien | exten => _91NXXNXXXXXX,1,Goto(outbound,${EXTEN:1},1); |
| 22:23.34 | seanbright | ? |
| 22:23.37 | kb3ien | any reason this is 'rejected because extension not found' ? when 1NXXNXXXXXX works in the line above? |
| 22:24.03 | seanbright | kb3ien: can you pastebin the output from the CLI? |
| 22:24.05 | seanbright | ~pb |
| 22:24.05 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 22:24.08 | kb3ien | i expect someone to REFUSE to stop dialing 9. |
| 22:25.47 | [TK]D-Fender | "9" prefixes are SOOO 80's. NONE of my clients do that. |
| 22:26.35 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:26.46 | seanbright | hey look, it's farkus |
| 22:28.00 | seanbright | wanders off |
| 22:28.12 | [TK]D-Fender | sticks his foot into farkus' revolving door and watches him face-plant |
| 22:29.24 | kb3ien | http://pastebin.com/m622f3741 |
| 22:29.47 | kb3ien | really arnt even 3 lines worth looking at. |
| 22:30.10 | kb3ien | its not the clients, its their staff. |
| 22:31.52 | kb3ien | is there an asterisk limit im hitting? |
| 22:33.18 | [TK]D-Fender | kb3ien: I have absolutley no faith that the CONTEXT its looking in has those extens in it... |
| 22:33.40 | [TK]D-Fender | kb3ien: I suggest you provide a more complete dialplan dump along with the SIP DEBUG of the failed call/. |
| 22:34.42 | kb3ien | the context is include=>ed |
| 22:35.36 | kb3ien | or i dont know what im talking about. |
| 22:35.44 | kb3ien | yep its the latter. |
| 22:36.22 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:38.12 | kb3ien | suspecting include=>outbound of brokeness i hacked _1NXXNXXXXXX into the customer's config file. |
| 22:38.32 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@77.207.64.172) |
| 22:38.34 | PoWeRKiLL | hi |
| 22:39.32 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:39.50 | PoWeRKiLL | I'm doing a goto in my extension.conf to another context but in my cdr I got the first context on not the next one anyone got this bug ? |
| 22:42.16 | [TK]D-Fender | PoWeRKiLL: CDR reflects the place the call was received in IIRC... |
| 22:43.39 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:44.02 | PoWeRKiLL | but it should reflect the last context it was when hangup |
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| 23:18.20 | PoWeRKiLL | I found that I got this problem http://bugs.digium.com/view.php?id=13797 |
| 23:18.30 | PoWeRKiLL | anyone know how to solve it ? |
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| 23:40.35 | [TK]D-Fender | PoWeRKiLL: Well the bug is listed as "open", so the answer seems to be a pretty clear "No". |
| 23:40.52 | *** join/#asterisk Cutlass (n=chatzill@c-67-176-208-15.hsd1.il.comcast.net) |
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| 23:42.41 | Cutlass | I want to define an extension in the dial plan which, when called directly, will park the caller and will NOT play back the orbit that (s)he was parked on...I'm using the Park() application, but from what I can tell, there is now way to make it suppress the orbit announcement. Does anyone know how to accomplish this? |
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| 23:45.08 | _matt | does asterisk have a sleep command ? |
| 23:45.19 | jql | you mean Wait? |
| 23:45.24 | _matt | yes :) |
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| 23:51.19 | *** join/#asterisk kripton1x (n=mike@apoc.digiport.com) |
| 23:51.24 | kripton1x | I have provider-canada and provider-usa, and am having issues with call forwarding. If i call to my canada-did, and it rings my polycom phone, i can pick up and get two way audio. When I set callforward on the SIP phone, i cannot hear from either side. And the same occurs when I try to dial my cellphone out of provider-usa |
| 23:51.49 | kripton1x | (instead of ringing the SIP phone) |
| 23:52.14 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.3) |
| 23:53.37 | drmessano | Asterisk does not Sleep(), it Wait()'s |
| 23:53.41 | drmessano | Much like chuck norris |
| 23:53.47 | stintel | lol |
| 23:54.03 | drmessano | <--- Twice in one day.. Who wins the intarwebs now |
| 23:54.04 | drmessano | ? |
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| 23:55.16 | [TK]D-Fender | kripton1x: So basically nothing on USA works? |
| 23:55.42 | kripton1x | it does, if I dial from a SIP phone, or call file out USA, it works |
| 23:55.50 | kripton1x | and I can receive calls on Canada DID and hear both ways as well... |
| 23:56.08 | kripton1x | whenever I seem to throw a method of forwarding in between is when I experience two-way audio issues |
| 23:58.39 | kripton1x | I have switched different nat settings as well |
