| 00:04.40 | beek | [TK]D-Fender: The skull fucking threat may have done something. I just got a call from Level3 who claims that Verizon is dispatching a tech to replace the "working" card. The Verizon guy was unable to ascertain if they really did replace it the last time. |
| 00:05.30 | [TK]D-Fender | beek: Stab and thrust my friend... stab and thrust... |
| 00:06.00 | beek | I'm skeptical, of course. Time will tell. |
| 00:08.18 | beek | cepstral's site is back up. I left a voice mail and told them that there was no connectivity. The site responds with "we're moving, please try back later." |
| 00:08.18 | jaytee | you use cepstral? |
| 00:08.49 | beek | Hi jaytee: I want to, with Allison's voice. |
| 00:08.52 | beek | I have the demo here right now. |
| 00:09.38 | jaytee | All Digium sells is just the Allison voice license, right? if you want the Cepstral engine you have to buy it from Ceptstral's site. |
| 00:10.28 | beek | Yep. I have the Cepstral demo... Everything it says starts with "This voice is unlicensed". I can live with that for testing. As soon as Cepstral is back online I'm going to purchase both the license for it and for Allison. |
| 00:11.42 | jaytee | from what I was able to determine, spanish is supported with Cepstral's software but not with Digium but I can't find anything specific like if it's just the Allison voice that doesn't come with a spanish text to speech. |
| 00:11.53 | beek | Damned PRI just dropped again. It was down for seven seconds so let's hope it's because they just put in a new working card. |
| 00:12.22 | beek | jaytee: Fortunately I'm only having to deal with English. |
| 00:16.01 | *** join/#asterisk rdgr (n=rich@82-46-0-91.cable.ubr01.aztw.blueyonder.co.uk) |
| 00:18.04 | beek | jaytee: It must be that the Allison voice doesn't come with Spanish. Digium doesn't really provide the speech engine. |
| 00:18.14 | beek | :w |
| 00:18.28 | jaytee | that's what I was thinking |
| 00:19.17 | jaytee | but it may also be that there currently isn't a method in Asterisk to tell Cepstral which engine to use, English or Spanish |
| 00:19.33 | jaytee | I need to do some more readin |
| 00:21.29 | *** join/#asterisk andresmujica (n=andresmu@201.244.110.136) |
| 00:25.20 | beek | jaytee: What kinds of things are you planning for Cepstral? |
| 00:29.49 | JJ2110 | Does anyone have a good link to how to setup Asterisk, so that you can have an IVR where keypresses launch scripts on the server? :) |
| 00:30.02 | JJ2110 | Wow that was bad english |
| 00:30.06 | JJ2110 | on how to setup* |
| 00:30.30 | beek | JJ2110: core show application system |
| 00:33.05 | JJ2110 | beek, forgive me but that makes no sense to me at all :/ |
| 00:33.28 | beek | JJ2110: On your Asterisk console type: core show application system |
| 00:33.39 | beek | Or if you're not at 1.6 try show application system |
| 00:33.59 | beek | You want the System() application |
| 00:34.03 | beek | ~book |
| 00:34.03 | jbot | from memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 00:34.15 | beek | JJ2110: ^^^^^^^^^^^^^^^^ That explains it too |
| 00:34.44 | JJ2110 | Thank you :) |
| 00:42.39 | jaytee | beek, rather than record spanish prompts for my IVR or code it for Spanish voice recognition I was just going to use spanish text through the Cepstral engine for prompts. |
| 00:44.26 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 00:44.58 | beek | jaytee: Ah... that would work well. Essentially, that's what I want the Allison voice for. |
| 00:46.06 | jaytee | I have to dig into it more. It may be that I can use it with another voice and it's just Allison's voice that isn't available in spanish. |
| 00:47.00 | jaytee | bbiab |
| 00:47.43 | JJ2110 | exten => 123,1,System(echo hello > /tmp/hello.txt) |
| 00:48.17 | JJ2110 | beek, can you please explain whats happening in the system() part? :) |
| 00:49.33 | JJ2110 | Im wanting to launch a file with it |
| 00:49.45 | JJ2110 | Perl / Python / PHP script preferably |
| 00:50.49 | *** join/#asterisk ScribbleJ (n=sj@c-67-172-6-141.hsd1.il.comcast.net) |
| 00:51.46 | JJ2110 | Then I'll leave everyone alone |
| 00:57.38 | beek | JJ2110: I was away... I'm not sure of your question. |
| 00:59.02 | JJ2110 | Sorry, Umm |
| 00:59.40 | JJ2110 | Im wondering what I'd have to put into System([here]) to run say /var/www/vlc/play.php |
| 01:00.04 | beek | Just that. |
| 01:00.11 | beek | System(/var/www/vlc/play.php) |
| 01:00.40 | beek | JJ2110: have you done any programming before? |
| 01:02.36 | beek | JJ2110: You asked me to explain what '(echo hello > /tmp/hello.txt)' did... you don't know what that simple BASH snippet does? |
| 01:04.08 | beek | pfffft-pfffft (blows into mike) |
| 01:04.12 | beek | Is this thing on? |
| 01:04.30 | beek | <taps mike a few times> |
| 01:06.17 | JJ2110 | Sorry |
| 01:06.20 | JJ2110 | Phone rang |
| 01:06.31 | JJ2110 | Ive done C / C++ and PHP |
| 01:06.36 | JJ2110 | But Im new to Linux |
| 01:06.38 | JJ2110 | :/ |
| 01:06.54 | JJ2110 | So no, Dont know what a simple BASH snippet does |
| 01:07.06 | JJ2110 | Am learning though, doing RHCT soon |
| 01:07.46 | beek | JJ2110: Try it from a command line. Type "echo hello" and see what it does. |
| 01:07.56 | JJ2110 | says Hello |
| 01:08.06 | beek | Now do the echo hello > /tmp/hello.txt |
| 01:08.08 | JJ2110 | As it would in PHP etc :) |
| 01:08.37 | JJ2110 | Writes hello to /tmp/hello.txt |
| 01:08.41 | beek | There you go. |
| 01:08.42 | JJ2110 | I follow :) |
| 01:09.15 | JJ2110 | Cool |
| 01:09.29 | JJ2110 | Now I just need to figure out the scripting to control VLC remotely.. |
| 01:09.47 | beek | What is VLC? |
| 01:10.28 | beek | Is this script to do something external to Asterisk or is it supposed to control the call? |
| 01:10.36 | JJ2110 | http://www.videolan.org/ |
| 01:10.42 | JJ2110 | Media Player |
| 01:10.55 | beek | Is this supposed to provide audio back to Asterisk? |
| 01:10.58 | JJ2110 | I want to be able to dial a number and it starts playing |
| 01:10.58 | JJ2110 | No |
| 01:11.04 | JJ2110 | Just a general control interface |
| 01:11.13 | beek | Okay. Then System() is what you want. |
| 01:11.14 | ScribbleJ | Media PLayer + Transcoder + Streaming... it's extremely cool and you shoudl know about it. :) |
| 01:11.15 | JJ2110 | Start / Stop / Next / Previous |
| 01:11.28 | JJ2110 | I agree with ScribbleJ |
| 01:12.16 | beek | Looks interesting... |
| 01:12.23 | ScribbleJ | In theory, you could configure it to play the streams to an RTSP-capable SIP client. |
| 01:12.30 | JJ2110 | I have a server setup with a shared folder of music videos, VLC loads the folder and starts playing them on an old laptop I have setup to look like a screen on the wall |
| 01:12.31 | ScribbleJ | Since it does do RTSP, although not well. |
| 01:12.39 | JJ2110 | Connected to 7.1 speakers |
| 01:12.47 | JJ2110 | So its a nice way to have music playing |
| 01:13.00 | JJ2110 | The issue being I dont want to have a keyboard / mouse plugged in |
| 01:13.13 | JJ2110 | And loading the Web Interface isnt always the best way to control it |
| 01:13.19 | ScribbleJ | Ooooh |
| 01:13.33 | ScribbleJ | You just gave me the brilliantest idea for my girlfriend's old busted macbook. |
| 01:13.35 | beek | JJ2110: Sounds like an interesting project. You may want to investigate AGI(), as well. That will allow you more flexibility in programming. |
| 01:13.51 | ScribbleJ | Yes, I'd agree, with a little AGI and a little Perl you could have VLC doing backflips. |
| 01:13.53 | [TK]D-Fender | beek: No need |
| 01:14.07 | beek | [TK]D-Fender: Depends on how sophisticated he wants to get. |
| 01:14.09 | [TK]D-Fender | beek: He jsut wants to control VLC, not dilaplan / input flow |
| 01:14.10 | JJ2110 | ScribbleJ, Ill get the URL to some photos for you ;) |
| 01:14.24 | [TK]D-Fender | JJ2110: All you need is "System" |
| 01:14.35 | [TK]D-Fender | JJ2110: Though I don't knwo why you aren't using a web interface for this |
| 01:14.51 | JJ2110 | http://uploads.jamesforman.co.nz/blog/2008/frame/ |
| 01:15.01 | JJ2110 | I am using a web interface |
| 01:15.13 | JJ2110 | I just want to extend that interface |
| 01:15.34 | ScribbleJ | Hahaha, brilliant. |
| 01:15.46 | ScribbleJ | I'm so digging out that old macbook, only problem is a busted hinge on the display. |
| 01:16.03 | JJ2110 | ScribbleJ, thats why I did this to it, both the hinges were broken |
| 01:16.13 | ScribbleJ | Yeah.... dang cats. |
| 01:16.15 | JJ2110 | Just took all the casing off the screen, |
| 01:16.42 | JJ2110 | Put a frame around it and glued the screen on, flipped the base around and used the screw holes provided to screw into the wooden frame |
| 01:17.04 | JJ2110 | Plug in a keyboard and mouse and install an OS |
| 01:17.13 | JJ2110 | Install software to control it over the network |
| 01:17.16 | JJ2110 | :) |
| 01:18.00 | ScribbleJ | Yeah, well, windows is an odd choice, but the basic idea is sound. I've already got openssh-server on the macbook so I Think I'm all set, mnaybe use x2x from my mythtv box to really freak people out. |
| 01:18.12 | ScribbleJ | "YOur tv controls your picture frame?" |
| 01:18.20 | *** join/#asterisk adker_ (n=chatzill@74-33-209-209.dr02.glvv.ny.frontiernet.net) |
| 01:18.24 | JJ2110 | I used XP just because I had it lying around and the most experience with it |
| 01:18.34 | JJ2110 | Drivers were an issue too |
| 01:19.21 | ScribbleJ | Ugh, bother. I wrote a speech engine for Asterisk that integrates to Sphinx2, now I want to try pocketsphinx and/or sphinx3 instead and the APIs are, of course, all differnt. Yay me. |
| 01:19.22 | JJ2110 | Kernal errors really put a dent in your party |
| 01:19.35 | JJ2110 | Ouch |
| 01:20.04 | JJ2110 | My network switch just died again, lost my phone server, general server and photo frame... anything not on wifi |
| 01:20.24 | ScribbleJ | Well, I'm stealing your idea, thanks. Time fo rme to get coding. |
| 01:20.38 | JJ2110 | Which one? :) |
| 01:20.39 | beek | JJ2110: Unplug switch. Open window. Toss switch out window. Close window. Buy new switch. |
| 01:20.49 | JJ2110 | beek, I have a stack of switches |
| 01:21.01 | beek | s/Buy new switch/replace switch/ |
| 01:21.04 | JJ2110 | Its more a matter of not being able to get into the back of the server cabinet without ripping my stitches |
| 01:21.09 | mchou | JJ2110: which is the switch that crapped out? |
| 01:21.26 | beek | JJ2110: Stitches? |
| 01:21.39 | JJ2110 | This is an old photo |
| 01:21.43 | JJ2110 | But the one on top http://uploads.jamesforman.co.nz/blog/2008/network/PICT0001.JPG |
| 01:21.51 | JJ2110 | Have many more up to date switches now |
| 01:21.57 | JJ2110 | Those Cisco ones are 10BaseT |
| 01:22.20 | JJ2110 | D-LINK DES1016 |
| 01:22.23 | mchou | JJ2110: the dlink one? :) |
| 01:22.39 | JJ2110 | Yeah, explains it all? :) |
| 01:22.52 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 01:22.52 | JJ2110 | Could also be the 30degreeC room and the switches fans both being unplugged |
| 01:22.52 | mchou | JJ2110: what's that running? openwrt? |
| 01:23.03 | `Sean | lmao JJ2110 you got alot of switches running uselessly :P |
| 01:23.07 | JJ2110 | Bog standard os |
| 01:23.15 | JJ2110 | 'Sean, that was just me testing them :) |
| 01:23.27 | JJ2110 | Was loads of fun though |
| 01:23.30 | `Sean | why D-LINK they suck! |
| 01:23.39 | beek | JJ2110: Ah, but they look pretty in a darkened room... |
| 01:23.40 | JJ2110 | Because someone gave it to me? |
| 01:23.46 | JJ2110 | beek, that they do |
| 01:23.55 | JJ2110 | The cisco ones make great rack shelves too |
| 01:24.03 | mchou | haha, that's why they GAVE it to you |
| 01:24.17 | JJ2110 | Its worked fine for a year |
| 01:24.20 | `Sean | JJ2110 aslong as there for free :p |
| 01:24.21 | mchou | freebie from the dump |
| 01:24.30 | JJ2110 | shrugs, free is free |
| 01:24.38 | beek | If it's free, it's for me.... unless it is VD |
| 01:24.41 | JJ2110 | This is where I play the poor student card |
| 01:24.47 | `Sean | LOL! |
| 01:24.58 | `Sean | ive played that one alot too lmao |
| 01:25.09 | JJ2110 | 5 more exams to go and Im done :) |
| 01:26.03 | `Sean | thats cool |
| 01:26.07 | JJ2110 | beek, stitches yes :) I had 70, now 25, get them out in 6 months after I get another surgery and get new ones put in... |
| 01:26.21 | JJ2110 | Life is fun |
| 01:26.38 | beek | JJ2110: Yeesh, that doesn't sound good. I see you're from NZ so you could tell us all it's from a shark bite. |
| 01:26.54 | JJ2110 | Seen a shark in my 9 years in aussie, never one here in NZ though |
| 01:27.20 | JJ2110 | Nah, surgery was for rejected tissue |
| 01:27.25 | JJ2110 | :/ |
| 01:27.38 | beek | Well, if you have that many stitches you could play the wounded card and have the GF replace the switch for you. |
| 01:27.42 | JJ2110 | And with that, I am obliged to go to the supermarket |
| 01:28.08 | JJ2110 | beek, I could, if she was not on holiday to visit her father / step mother / half brothers + sisters |
| 01:28.09 | *** join/#asterisk andresmujica (n=andresmu@201.244.110.136) |
| 01:28.25 | beek | JJ2110: bummer. |
| 01:28.38 | beek | JJ2110: have fun with your project. |
| 01:28.40 | JJ2110 | I dont know if she could move the full height server cabinet either |
| 01:28.42 | JJ2110 | Will do :) |
| 01:45.05 | *** join/#asterisk gambolputty (n=BC43599@cpe-76-186-231-222.tx.res.rr.com) |
| 02:00.05 | *** join/#asterisk Tommy3 (n=tom@c-76-29-235-137.hsd1.al.comcast.net) |
| 02:00.35 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 02:02.26 | Tommy3 | I have a sip phone that shows idle, but message says "on the phone". Is there some way to reset. Other places show the phone as idle. |
| 02:05.24 | Tommy3 | (rephrasing) When I call the sip phone I get the "on the phone" message. The asterisk gui that I am using indicates the phone is available and idle. Tried restarting asterisk. no change. |
| 02:11.06 | Tommy3 | Is there possibly a status file in asterisk where I can change the state if the sip phone? |
| 02:12.16 | joat | depending on your phone you might be talking about "hint" and BLF |
| 02:12.50 | joat | then again, mebbe not (if it's a single line phone) |
| 02:13.10 | joat | what phone? |
| 02:22.51 | Tommy3 | it is a pap2. here is something from the log that seems fishy |
| 02:22.53 | Tommy3 | DEBUG[2829] func_db.c: DB: AMPUSER/5105/vmx/busy/state not found in database. |
| 02:24.39 | Tommy3 | joat: sorry for the delay. I was digging around in asterisk logs. |
| 02:25.15 | drmessano | "On the phone" in what GUI? Spark? |
| 02:25.27 | Tommy3 | this is evb |
| 02:25.37 | drmessano | evb? |
| 02:25.54 | drmessano | Oh |
| 02:25.54 | Tommy3 | easy vox box |
| 02:25.57 | drmessano | EasyVoxBox |
| 02:25.58 | drmessano | ha |
| 02:26.29 | joat | eeuuu! |
| 02:26.32 | joat | no clue |
| 02:27.28 | joat | mysql backend? |
| 02:27.32 | drmessano | I had no idea anyone was developing that |
| 02:27.53 | drmessano | I wonder if its real dev now or tweaks to appear like dev |
| 02:27.59 | drmessano | Since its been dead for ages |
| 02:28.23 | Tommy3 | yes, mysql. EVB is being used by a bunch of radio system developers that I am trying to keep up with. |
| 02:28.43 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:28.44 | joat | have noted a similar issue with realtime meetme in that if you attempt to transfer a call (e.g., to another conference room) the count will "stick". it goes away if i dial back in and then hang up |
| 02:28.46 | Tommy3 | I dont know why they are using evb. |
| 02:28.48 | drmessano | Which radio systems? |
| 02:29.44 | joat | that's something external to * though |
| 02:30.00 | Tommy3 | there is a group in california, and another in atlanta that are working on drivers for ham radio. There is also a small group here in Huntsville. |
| 02:30.23 | drmessano | Ouch |
| 02:30.29 | Tommy3 | I have not tried the drivers yet. |
| 02:30.42 | joat | for remote control of the radios? |
| 02:30.59 | Tommy3 | I was just in a call to the sip phone and it dropped off and I could not get it to clear. |
| 02:31.07 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 02:31.21 | drmessano | Tommy3: Does EVB use FreePBX? |
| 02:32.23 | Tommy3 | re remote... the radios are treated like a type of phone. there are control circuits from the driver to control the radios. I have not dug into the code yet |
| 02:32.48 | drmessano | Tommy3: Its just using app_rpt which is part of Asterisk |
| 02:32.52 | drmessano | Any distro will work |
| 02:33.09 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 02:33.28 | Tommy3 | Yes, I think it is freepbx. |
| 02:33.35 | joat | drmessano: it does |
| 02:33.55 | drmessano | So you could using the AsteriskNOW 1.5 beta just the same |
| 02:33.58 | drmessano | be* |
| 02:34.33 | Tommy3 | if you are interested in the radio control, ham radio operator w4wwm has a website documenting a lot of their work. |
| 02:35.10 | drmessano | I've got a controller based on Asterisk I am getting ready to install on my 440 machine |
| 02:37.28 | beek | drmessano: That sounds interesting... is that to be the repeater controller or something else? |
| 02:37.36 | Tommy3 | these guys are using a cm103 chip. it is a sound card on a usb dongle. they wire into the dongle to pick up two unused digital lines off the chip that are not used. I think they are like the old rs232 control signals. costs about 10 bucks |
| 02:38.13 | beek | Tommy3: have a URL for w4wwm? |
| 02:38.19 | Tommy3 | it ports into a repeater controller. i think w4wwm has a cat controller. let me grab the website for you... |
| 02:38.22 | drmessano | beek: It does it all.. let me find the link |
| 02:40.42 | drmessano | http://app-rpt.qrvc.com/ |
| 02:42.56 | Tommy3 | http://searn.alertradio.net/blog/archives/category/rfgateway/regionalrfgateway/ser |
| 02:43.55 | drmessano | It doesnt handle the PL bit, but that's easy enough |
| 02:44.00 | beek | They both look interesting projects. |
| 02:44.21 | Tommy3 | He's got a lot of stuff on that page, but he has an asterisk section. |
| 02:44.32 | coppice | what is the legal position of these things in the US? In most .eu countries there seem to be lots of restrictions on amateur radio interconnecting with anything else |
| 02:45.03 | drmessano | Well let see |
| 02:45.18 | beek | coppice: Provided that there's a control operator I don't think that there'll be any problem. |
| 02:45.20 | jaytee | the US has a much broader vision that places lots and lots of restriction on anything interconnecting to anything |
| 02:45.25 | drmessano | Over the air is a big deal.. Theres band restrictions on remote control |
| 02:45.33 | drmessano | But over internet, few |
| 02:45.53 | Tommy3 | the alertradio reference is to N0PCO in kansas. He has a lot of stuff too. They come up on asterisk and also on echolink and talk about it a lot |
| 02:45.54 | drmessano | Control operation needs to be available to kill the link at a moments notice blah blah blah |
| 02:46.17 | drmessano | We had an Echolink node set up for a while |
| 02:46.21 | drmessano | I hate echolink lol |
| 02:46.29 | drmessano | Took that crap off the repeater |
| 02:47.25 | Tommy3 | They dont have echolink tied to asterisk. just use it to let folks talk about asterisk and radio to build interest and ask questions. |
| 02:49.06 | Tommy3 | I have to go now. good luck on the repeater stuff. 73 , de wa4fyn, huntsville |
| 02:49.27 | beek | TTYL -- I need to check out to. 73 de NJ3N |
| 02:50.26 | *** part/#asterisk Tommy3 (n=tom@c-76-29-235-137.hsd1.al.comcast.net) |
| 02:50.57 | drmessano | <-- KE4RAP |
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| 04:23.04 | drmessano | "Man Hit By GET Bus Dies" |
| 04:23.13 | drmessano | Should have used WGET |
| 04:23.15 | drmessano | :( |
| 04:32.17 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 04:33.01 | frogonwheels | Can anybody tell me why ${ARG1} doesn't work when I Gosub() ? |
| 04:33.24 | frogonwheels | asterisk 1.4 - minimal set of modules loaded.. |
| 04:34.52 | [TK]D-Fender | frogonwheels: "core show application gosub" <- thats why |
| 04:35.31 | frogonwheels | .. as in my version doesn't allow parameters? |
| 04:35.50 | [TK]D-Fender | frogonwheels: As in, what does it SAY? |
| 04:36.11 | frogonwheels | well it doesn't say it handles parameters. |
| 04:36.18 | frogonwheels | so I guess that's it. ok |
| 04:37.20 | frogonwheels | guess I'll use a macro instead. |
| 04:37.55 | jaytee | yeah, cuz that other way would be just way too much harder |
| 04:41.20 | frogonwheels | [TK]D-Fender: so I'm assuming that ARGS are added in a later version (and hence it's in the doco on voipinfo).. rather than I'm missing a module.. |
| 04:43.06 | [TK]D-Fender | frogonwheels : For sanity's sake ( and I don't want you to take this the wrong way), "fuck the fucking wiki" :) |
| 04:43.16 | [TK]D-Fender | fromOr more gently, YMMV :p |
| 04:44.05 | [TK]D-Fender | frogonwheels: and you should NEVER look there first for syntax |
| 04:44.54 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-4-110.phlapa.east.verizon.net) |
| 04:48.59 | frogonwheels | [TK]D-Fender: well I often _do_ look their first 'cause it helps me work out which modules to load. |
| 04:49.07 | frogonwheels | .. but I take your point. |
| 04:49.31 | frogonwheels | however I was assuming that somehow the () was handled by some other part of asterisk. |
| 04:49.35 | frogonwheels | .. obviously wrongly. |
| 04:51.49 | [TK]D-Fender | frogonwheels: Other parts don't control App Args.. Thats nuts |
| 04:52.20 | frogonwheels | [TK]D-Fender: well it wouldn't be the only slightly weird thing about asterisk.. . like $[ ] .. but still, I take your point. |
| 04:53.04 | frogonwheels | [TK]D-Fender: so I'm correct in saying that the only way of having parameters to a 'function' like context is to use macro() ? |
| 04:53.15 | [TK]D-Fender | frogonwheels: Expressions are clearly more global than apps. |
| 04:53.28 | [TK]D-Fender | frogonwheels: For 1.4 yes |
| 04:53.55 | frogonwheels | [TK]D-Fender: Cool.. and in 1.6 ? |
| 04:53.57 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 04:54.14 | [TK]D-Fender | frogonwheels: Macro is deprecated, and Gosub has args |
| 04:54.28 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 04:54.29 | frogonwheels | arghhh |
| 04:54.33 | frogonwheels | ok. cool. |
| 05:00.08 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 05:00.14 | dlewis | [TK]D-Fender: you around? |
| 05:00.54 | frogonwheels | Is there a better expression to use in place of _X. that includes things like #*+ ... I can use _[0-9#*]. of course.. |
| 05:01.28 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 05:02.13 | [TK]D-Fender | dlewis: Yup |
| 05:02.19 | frogonwheels | hmm.. actually, in this case I think include would be better. |
| 05:02.40 | dlewis | [TK]D-Fender: we had a conversation the other day about the Sangoma A200D vs. the B600D |
| 05:02.41 | [TK]D-Fender | frogonwheels: _. |
| 05:02.48 | dlewis | what was your issue again with the A200D? |
| 05:02.59 | [TK]D-Fender | frogonwheels: though your very first priority had better be to jump OUT of that context. |
| 05:03.15 | [TK]D-Fender | dlewis: that the B600D represented a better dollar value |
| 05:03.50 | dlewis | ah, ok |
| 05:04.00 | dlewis | so, feature wise, no difference besides the A200D being modular. |
| 05:04.22 | frogonwheels | [TK]D-Fender: but doesn't that generate warnings?? and isn't there some kind of precedence trap with _. ? |
| 05:06.52 | [TK]D-Fender | frogonwheels: You're talking about a bloody catch-all. Thats already considered "nearly stupid" |
| 05:07.21 | [TK]D-Fender | dlewis: B600 coming with a lot more for the same base $ to meet any expected expansion... just makes sense. |
| 05:07.27 | [TK]D-Fender | dlewis: Do. The. Math |
| 05:08.41 | dlewis | [TK]D-Fender: I did... the B600 comes with 2 more FXO ports. |
| 05:08.57 | [TK]D-Fender | dlewis: Same $ pretty much though. More for the same $. |
| 05:09.10 | dlewis | right |
| 05:09.19 | dlewis | ok, thanks! |
| 05:11.05 | dlewis | If I go with the B600D, I'll just have to live with the 4FXO and 1FXS |
| 05:11.24 | [TK]D-Fender | dlewis: Sounds like TORTURE, considering you needed only 40% of it... |
| 05:11.49 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 05:11.59 | dlewis | if I go with the A200D, then I'll know that I can expand... |
| 05:12.43 | [TK]D-Fender | dlewis: You said yourself that you have no expectations to even fill a single B600 as it is... |
| 05:13.13 | dlewis | lol |
| 05:13.14 | dlewis | i did |
| 05:13.15 | dlewis | but |
| 05:13.18 | dlewis | you never know |
| 05:13.29 | dlewis | I can see myself needing more FXS |
| 05:13.35 | dlewis | if I wanted to connect more than one ATA |
| 05:13.41 | drmessano | No you dont |
| 05:13.43 | drmessano | Get IP phones |
| 05:13.59 | dlewis | drmessano: i'm using IP phones right now |
| 05:14.01 | [TK]D-Fender | dlewis: You're right, you should clearly go for the A108d + Channel bank RIGHT NOW. |
| 05:14.01 | dlewis | but |
| 05:14.07 | drmessano | So why do you need so many FXS? |
| 05:14.19 | dlewis | drmessano: for faxing or ATA's... |
| 05:14.21 | dlewis | you never know |
| 05:14.25 | [TK]D-Fender | dlewis: ATA's have NOTHING to do with that card |
| 05:14.25 | drmessano | ATA's dont need FXS |
| 05:14.27 | drmessano | They are IP |
| 05:14.35 | dlewis | sorry |
| 05:14.36 | drmessano | They ARE FXS to IP |
| 05:14.37 | dlewis | not ATA |
| 05:14.39 | dlewis | POTS phones |
| 05:14.40 | drmessano | ... |
| 05:14.45 | [TK]D-Fender | dlewis: "you never know" = RETARD statement |
| 05:14.46 | drmessano | Dont get Pots phones |
| 05:14.49 | drmessano | Get IP phones |
| 05:14.53 | drmessano | As I said |
| 05:15.02 | dlewis | right |
| 05:15.12 | [TK]D-Fender | dlewis: With that kind of thinking you'd better go BUY YOUR TELCO |
| 05:15.21 | dlewis | lol |
| 05:15.22 | dlewis | ok |
| 05:15.41 | drmessano | You're using IP phones, but may need FXS for POTS phones. CONFUSED MUCH? |
| 05:15.46 | drmessano | Your PBX is.. |
| 05:16.13 | dlewis | drmessano: i can think about one use for FXS right now... faxing |
| 05:16.33 | drmessano | and that one FXS port will work fine for that |
| 05:16.40 | drmessano | Done |
| 05:18.29 | dlewis | was thinking about also using an FXS port for a door phone |
| 05:19.03 | dlewis | eh |
| 05:19.07 | dlewis | that would use an FXO though |
| 05:19.34 | dlewis | any of you guys ever install a door phone? |
| 05:20.15 | dlewis | these SIP enabled door phones are expensive |
| 05:20.20 | dlewis | might just go with this setup: |
| 05:20.25 | dlewis | http://nerdvittles.com/wp-content/doorbell.pdf |
| 05:20.36 | drmessano | A door phone would NOT use an FXO |
| 05:20.56 | dlewis | drmessano: according to the instructions in that link, it would... |
| 05:21.04 | dlewis | unless you know of a more efficient way. |
| 05:22.17 | drmessano | Because its interfacing on the station <> station side |
| 05:22.28 | drmessano | Better way is to gut a phone and use an ATA |
| 05:22.37 | drmessano | Much cheaper |
| 05:23.04 | drmessano | Oh |
| 05:23.08 | drmessano | That one uses POTS |
| 05:23.10 | jaytee | some PA equipment and intercom stuff use FXO instead of FXS |
| 05:23.14 | drmessano | Great if you dont have POTS |
| 05:23.45 | drmessano | jaytee: If they're creating their own 48v talk circuit |
| 05:23.56 | jaytee | yep |
| 05:23.57 | drmessano | If you're use Asterisk, the talk circuit is the PBX |
| 05:24.01 | drmessano | using* |
| 05:24.41 | frogonwheels | http://pastebin.com/d12fca8e4 in this case, is there a better way of catching the 101 (which is dialing itself.. local_101 is the context for the 101 line). |
| 05:24.48 | dlewis | drmessano: that one actually uses asterisk |
| 05:25.00 | dlewis | (it states "instead of using POTS..." |
| 05:25.11 | frogonwheels | I have a feeling that _. isn't really correct here... |
| 05:25.25 | drmessano | ... |
| 05:25.26 | drmessano | dlewis |
| 05:25.43 | drmessano | Stop term grabbing |
| 05:26.17 | drmessano | The device generates a 48v talk circuit to use with POTS phones |
| 05:26.23 | drmessano | Rather than use a PHONE |
| 05:26.28 | drmessano | It uses an FXO |
| 05:26.34 | drmessano | Which is still a waste |
| 05:26.44 | dlewis | ok |
| 05:26.51 | dlewis | and your alternative is to gut a phone? |
| 05:27.17 | jaytee | or pontificate ad nauseum |
| 05:27.25 | drmessano | Actually, I like the idea of putting a wall phone outside |
| 05:27.32 | drmessano | and have it autodial on pick up |
| 05:29.10 | dlewis | that would look very awkward for a home (imho) |
| 05:29.37 | dlewis | a multi-tenant dwelling, possibly |
| 05:29.39 | drmessano | Well, if you have the $200, go get that box |
| 05:29.52 | dlewis | drmessano: agreed... |
| 05:30.40 | drmessano | You can say what you want about how something looks, but unless you have a better idea or are willing to pay the money for something proper, you're just passing gas |
| 05:37.12 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 05:39.40 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 05:49.07 | *** join/#asterisk adamw (n=AdamW@mandriva/channel-support/AdamW) |
| 05:49.18 | adamw | hey - don't suppose anyone familiar with iaxclient is around? |
| 05:49.51 | Maliuta_CA | no, everyone is on holidays right now |
| 05:49.53 | Maliuta_CA | ;P |
| 05:50.13 | adamw | bah, workshy...:D |
| 05:51.15 | cvnet | if a calls comes in sip g729 and you send it h323 g729 would it work with * ? |
| 05:53.18 | Maliuta_CA | cvnet: in that case you need to buy at least 2 channels of g729 license from digium |
| 05:54.17 | Maliuta_CA | cvnet: there is another implementation, but I can't speak for it. I use the digium codec |
| 05:54.40 | Maliuta_CA | adamw: wor???? that's a for letter word :) |
| 05:54.53 | cvnet | hum |
| 05:55.15 | Maliuta_CA | cvnet: g729 is only free for pass-through |
| 05:55.15 | cvnet | i get the calls coming in via DID and i send the calls to terminiation provider, why do i need lic from digium? |
| 05:55.49 | [TK]D-Fender | cvnet>if a calls comes in sip g729 and you send it h323 g729 would it work with * ? <- Yes |
| 05:55.50 | Maliuta_CA | if you are going to transcode it to h323 then you need a license |
| 05:56.08 | [TK]D-Fender | cvnet: "via DID ": = ambiguous term |
| 05:56.17 | [TK]D-Fender | cvnet: DELIVERY MEDIUM counts |
| 05:56.30 | cvnet | DID i mean phone number which ppl calls and it gets transfered to sip |
| 05:56.45 | cvnet | im totally lost about the licensing |
| 05:56.59 | [TK]D-Fender | cvnet: Describe only the call coming in, and the call going out. DID is NOT a "thng" in this case |
| 05:57.00 | cvnet | right now i get calls coming in and i send it sip both way no problem |
| 05:57.43 | [TK]D-Fender | cvnet: Yes you can take SIP G.729 as pass it through as G.729 without a license so long as you aren't changing the frame-size (at worst) |
| 05:57.50 | [TK]D-Fender | (to H.323 |
| 05:57.51 | cvnet | [TK]D-Fender so it would transfer the calls from sip to h323? |
| 05:58.07 | [TK]D-Fender | cvnet: RTP is the same, there is no encoding or decoding... or at least, there shouldn't be |
| 05:58.39 | cvnet | [TK]D-Fender thanks, your always very helpfull, appreciate it |
| 05:59.47 | cvnet | here is a joke: there is a compitition of muscians, drummer/gitar player ... |
| 06:01.05 | cvnet | winner is the one who plays the longest, gitar player, pino player all give up after 6 hours, but the drum player goes on and on, after 12 the tell him stop and tell you are the winner, please tell us what made you play this long? |
| 06:01.21 | cvnet | He answers: I had picture of my wife and mother in low in the drums |
| 06:01.28 | cvnet | low = law |
| 06:01.44 | cvnet | in = on |
| 06:01.50 | cvnet | im drunk btw |
| 06:04.41 | [TK]D-Fender | cvnet: You are hard to comprehend, AND not funny... |
| 06:05.06 | [TK]D-Fender | cvnet: Poor attempt at musical humour, and my nick has nothing to do with the guirtar brand. |
| 06:05.08 | cvnet | sorry, too much whisky |
| 06:05.33 | [TK]D-Fender | cvnet: Here's a better one : Do you know what Eric Clapton & bad coffee have in common? |
| 06:05.52 | cvnet | hum, no |
| 06:05.53 | [TK]D-Fender | cvnet: They both suck without Cream. |
| 06:05.58 | cvnet | lol |
| 06:12.21 | *** join/#asterisk pif (n=ldm@AToulouse-256-1-36-209.w90-38.abo.wanadoo.fr) |
| 06:21.34 | eppigy | hello |
| 06:21.37 | eppigy | i am dave |
| 06:29.57 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 06:35.38 | C4colo | what is this asound package asterisk wants me to install? |
| 06:35.47 | C4colo | yum doesn't have anything like that |
| 06:46.40 | justdave | asound == alsa |
| 06:47.33 | C4colo | I just needed alsa-devel |
| 06:47.48 | justdave | yeah, that sounds right. |
| 06:47.54 | C4colo | I did a yum list all | grep alsa |
| 06:48.02 | C4colo | everything but -devel was installed, so I added that and it liked it |
| 06:51.13 | C4colo | the only two I need to track down now are "resample(E)" and "speex_preprocess(E)" |
| 06:52.23 | C4colo | although I don't think I need SLIN resample codec, I was thinking that was res_convert |
| 06:53.04 | C4colo | and I probably won't use speex anyway, so I guess I'll leave it at that |
| 06:54.30 | [TK]D-Fender | C4libspeex |
| 06:55.06 | C4colo | I think I have that installed |
| 06:55.14 | C4colo | yea |
| 06:55.19 | C4colo | and libspeex-devel |
| 06:55.40 | C4colo | anyway, got an interesting error now: app |
| 06:56.29 | C4colo | app_fax.c:312 error: storage size of 'fax' isn't known // app_fax.c:483 error:storage size of 't38' isn't known |
| 06:56.45 | C4colo | apps error 2 |
| 06:56.50 | C4colo | and it bailed out |
| 06:57.05 | Gokee2 | Hello everyone :) I have been looking around for some documentation... I found a whole book pdf, nice idea but I really kinda want a quickstart guide... I want to be able to get it to answer the phone. (two lines) and have voicemail. I think everything else is pretty cool and I might play around with it sometime but I kinda want to have something work now. Any pointers? Thanks |
| 06:58.41 | C4colo | Gokee2 if you want something simple, you can write a pretty basic dialplan that does what you want... however if you want something easy, you might look at a pre-made solution like elastix |
| 06:59.12 | C4colo | you can download an iso from elastix.org and just pop it in and install, everything is installed for you and you just have to configure it |
| 06:59.51 | Gokee2 | C4colo, Hmmm |
| 06:59.54 | C4colo | however you get a much more complex setup since there are all sorts of features and packages installed when you do it that way. If you aren't interested in really learning how asterisk works, you just want a basic pbx for your home, that's the way I'd go |
| 07:00.11 | C4colo | if you want to learn asterisk, a pre-made solution like that is just chaos |
| 07:00.14 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
| 07:00.33 | C4colo | however, that's how I started |
| 07:00.56 | C4colo | I installed trixbox and got some experience with asterisk... now I work for a phone company managing their asterisk servers |
| 07:02.01 | Gokee2 | Hmm |
| 07:02.02 | C4colo | I started hacking dialplan into extensions_custom.conf (one of the files you can edit on a freepbx system) and before too long I had more in extensions_custom.conf than I did in the freepbx gui |
| 07:02.03 | Gokee2 | I se |
| 07:02.04 | Gokee2 | see* |
| 07:02.39 | Gokee2 | Whats a dialplan do? |
| 07:02.58 | C4colo | that is a good question |
| 07:03.00 | Gokee2 | I don`t mind learning a little while setting stuff up :) |
| 07:03.11 | C4colo | dialplan is everything on an asterisk system |
| 07:03.24 | C4colo | all the modules are installed and all that, but they don't know what to do until you give it dialplan |
| 07:03.56 | C4colo | you define what to do when a call comes in, when someone tries to dial out, when someone calls an extension on the pbx, how they get to voicemail, what happens when another call comes in when someone is already on the phone ... |
| 07:04.03 | C4colo | everything the pbx "does" is determined by dialplan |
| 07:04.10 | C4colo | that's what that pdf book is all about |
| 07:04.17 | C4colo | how to write dialplan |
| 07:04.21 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 07:04.43 | Gokee2 | I see |
| 07:06.07 | C4colo | if you have no experience with phone systems and programming, a GUI-based system like elastix or "pbx in a flash" .. something like that will give you the ability to start using asterisk without having to actually learn all about asterisk just to make a phone call |
| 07:06.32 | Gokee2 | Are there any example dialplan`s to look at? |
| 07:06.42 | C4colo | however, if you choose to go that way you will want to ask your questions in #freepbx since the guys here just work with asterisk (as far as I know, unless that has changed since asterisk chose freepbx as their alternate GUI) |
| 07:06.42 | Gokee2 | Hmm |
| 07:07.25 | C4colo | still, it is best to ask over there since those guys in #freepbx are more familiar with freepbx |
| 07:07.36 | Gokee2 | I have can do some programing (mostly in perl) but I have never worked with a pbx |
| 07:07.59 | C4colo | it is quite different |
| 07:09.53 | Gokee2 | So does freepbx just spit out asterisk dialplans? |
| 07:10.14 | C4colo | kinda |
| 07:10.29 | C4colo | it builds the dialplan in a mysql database, then when you hit "apply" it writes the dialplan |
| 07:10.39 | Gokee2 | I see |
| 07:10.40 | C4colo | so you can make many changes and then commit them all at once |
| 07:11.02 | C4colo | in case one step would break the phone system, and you need to do more than one change at a time |
| 07:11.31 | C4colo | but yea, you use a web-based GUI to program your pbx, then it builds your dialplan for you |
| 07:11.35 | UQlev | Gokee2: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf there are examples of dialplan for simple excercises |
| 07:13.39 | Gokee2 | Well I think I will give freepbx a try |
| 07:13.48 | Gokee2 | I can look at whatever it makes for me :) |
| 07:13.59 | C4colo | yea, it's a bit scary though |
| 07:14.19 | C4colo | if you want something that works easily, set up a freepbx system, then set up another one and you can play with the dial plan |
| 07:14.24 | C4colo | I had 3 pbxes at my house at one point |
| 07:17.00 | Gokee2 | Haha that must have been fun :) |
| 07:17.11 | [TK]D-Fender | Gokee2: No, you WON'T. GUI's build dead-end dialplans you can only add tiny bite-sized bits on to under its limited scope |
| 07:17.35 | [TK]D-Fender | Gokee2: Make no mistake that once you head down that road, it owns your ass |
| 07:18.17 | Gokee2 | [TK]D-Fender, Hmm, do you know of a way to get Asterisk up and running without reading a whole book first then? |
| 07:18.22 | C4colo | I agree, my ass was owned by trixbox... I'll never get a job at a CLEC as a network engineer |
| 07:18.30 | C4colo | oh wait, nevermind |
| 07:18.49 | Gokee2 | CLEC? |
| 07:18.57 | [TK]D-Fender | Gokee2: Read this for some inspiration on how simple a setup can be |
| 07:19.00 | [TK]D-Fender | ~jerjerguide |
| 07:19.00 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 07:19.05 | C4colo | I manage an asterisk-based telephony network that handles 15,000+ calls per day |
| 07:19.27 | [TK]D-Fender | C4colo: And is it running off Trixbox? |
| 07:19.32 | C4colo | obviously not |
| 07:19.40 | [TK]D-Fender | C4colo: Way to sell it... |
| 07:19.44 | C4colo | I'm just saying I started with Trixbox |
| 07:19.50 | C4colo | learned what I could from it |
| 07:20.15 | C4colo | and started building dialplan on my own, hacking it into trixbox... eventually dumping the gui and writing it from scratch |
| 07:20.23 | [TK]D-Fender | C4colo: GUI's don't teach you crap except the exact point at which they fall short because they build toaster-grade configs |
| 07:20.31 | C4colo | I agree |
| 07:20.37 | C4colo | however guis are good for some people |
| 07:20.57 | [TK]D-Fender | C4colo: And the effort you spend fighting it is considerably worse than just learning * for yourself from scratch, and far less valuable. |
| 07:21.03 | C4colo | above I said, if you want to learn asterisk, you won't learn it from frepbx |
| 07:21.12 | C4colo | obviously |
| 07:21.21 | C4colo | but if it weren't for trixbox I probably wouldn't have gotten into asterisk |
| 07:21.27 | [TK]D-Fender | C4colo: Like every other twit coming in here saying "can't I just..." and having NO clue because they've have their hand held for so long. |
| 07:21.45 | [TK]D-Fender | C4colo: Well congratulations on your motivation at least. |
| 07:21.50 | C4colo | thanks |
| 07:21.55 | C4colo | I'm a very motivated person |
| 07:21.56 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-51-125.w86-215.abo.wanadoo.fr) |
| 07:22.08 | [TK]D-Fender | C4colo: I'm sure you certainly are at this point... |
| 07:22.34 | C4colo | in any case I asked if he wanted to learn asterisk, or if he just wanted a quick and easy setup |
| 07:22.38 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 07:22.41 | C4colo | and explained the difference |
| 07:22.49 | C4colo | some people don't need anything more than freepbx can offer |
| 07:23.03 | C4colo | which is why Digium chose to offer a gui |
| 07:23.25 | Maliuta_CA | is glad he "ain't people" |
| 07:23.26 | C4colo | which, interestingly enough, is now FreePBX |
| 07:23.46 | [TK]D-Fender | C4colo: No, Digium offered a GUI because somebody would want it and it helps them make money. |
| 07:24.02 | C4colo | so we agree |
| 07:24.34 | C4colo | at this point I don't really care, I'm trying to figure out why the latest 1.6 won't compile app_fax |
| 07:24.39 | C4colo | I think I have too new of a version of spandsp |
| 07:24.40 | justdave | I finally got around to trying to switch my asterisk over from zaptel to dahdi... |
| 07:25.24 | justdave | and the dahdi drivers are failing to load, complaining that /dev/zap/channels doesn't exist |
| 07:25.42 | [TK]D-Fender | justdave: What versions? |
| 07:25.43 | justdave | which doesn't make sense when /dev/dahdi/channels does, and one would think it should be using that :) |
| 07:25.56 | justdave | 2.1.0.3 |
| 07:25.58 | [TK]D-Fender | justdave: perhaps you compiled chan_zap and left stuff BEHIND |
| 07:26.33 | justdave | asterisk is 1.4.22 |
| 07:26.42 | [TK]D-Fender | justdave: Trast your modules folder and restart from scratch |
| 07:26.51 | C4colo | now I have removed app_fax from menuconfig and ilbc won't compile |
| 07:27.40 | justdave | this is when I do "service dahdi start" |
| 07:27.46 | justdave | that doesn't depend on asterisk stuff, does it? |
| 07:28.01 | justdave | (the dahdi initscript) |
| 07:28.43 | justdave | oh, I bet I know... had this problem once before with an old version of zaptel versus a newer one... |
| 07:28.58 | justdave | wc* drivers in multiple places in /lib/modules |
| 07:29.20 | [TK]D-Fender | ok, checkout time, back later |
| 07:32.06 | Ericounet | hello! it's morning here and it's only -2° C brrrrr |
| 07:35.49 | justdave | ok, that didn't help |
| 07:36.13 | justdave | nuked all of misc from /lib/modules (because it was all dahdi/zaptel stuff) |
| 07:36.27 | justdave | and did make install from the dahdi source directory |
| 07:36.36 | justdave | trying to start dahdi still fails |
| 07:36.47 | justdave | Loading DAHDI hardware modules: wcte12xp: Notice: Configuration file is /etc/zaptel.conf |
| 07:36.50 | justdave | line 0: Unable to open master device '/dev/zap/ctl' |
| 07:37.00 | justdave | do I need to remove zaptel.conf to get it to find /etc/dahdi/system.conf ? |
| 07:37.26 | justdave | hmm, yep, that's exactly what it needed |
| 07:38.31 | frogonwheels | C4colo: Any tips for this? http://pastebin.com/d12fca8e4 Is there a better way of catching the 101 (which is dialing itself.. local_101 is the context for the 101 line). |
| 07:40.12 | frogonwheels | C4colo: btw, agree with you totally on the gui thing. They are useful in themselves as simple configuration tools.. but also can be quite useful as learning tools. |
| 07:41.23 | frogonwheels | C4colo: I'm actually writing an OpenWRT UCI interface for asterisk which will be used to configure asterisk from a web UI in an 'OpenWRT' way. |
| 08:00.06 | C4colo | frogonwheels: what I use in my dialplans to detect if an extension is calling itself is a simple macro that compares ${EXTEN} to ${CALLERID(number)} |
| 08:00.39 | C4colo | that works on our network because every user has the callerid set on the system, overrides the callerid on the device |
| 08:00.51 | C4colo | and our callerid numbers are always full DIDs |
| 08:00.55 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 08:01.06 | frogonwheels | C4colo: hah. in this case I do NOT have the extensions set :) |
| 08:01.34 | frogonwheels | .. and not relying on that..actually I was just extending it to SET the CALLERID(number) as well :) |
| 08:01.49 | C4colo | we also have a prefix to call your own voicemail box for our SIP phone customers that want more "pbx-style" configuration |
| 08:02.44 | frogonwheels | C4colo: ooh.. just thinking - I don't have that on the configs. |
| 08:03.10 | frogonwheels | anyway - easy enough from what I've got. |
| 08:03.46 | frogonwheels | I'm going to use _. which is annoying 'cause it gerates a warning.. but I don't _know_ of any problems in this case. |
| 08:03.52 | C4colo | something like exten => *98XXX,1,VoicemailMain(${EXTEN}@default) |
| 08:04.19 | C4colo | that's what we put in the phones for the "messages" button extension |
| 08:04.23 | frogonwheels | http://pastebin.com/d51e4270 |
| 08:04.36 | C4colo | oh sorry |
| 08:04.36 | frogonwheels | C4colo: oh yeah -that's a simple way. |
| 08:04.43 | C4colo | something like exten => _*98XXX,1,VoicemailMain(${EXTEN}@default) |
| 08:04.53 | C4colo | forgot the _ on that one |
| 08:05.08 | frogonwheels | in this case, (in a home), there's likely to be only 1 or 2 mailboxes. |
| 08:06.01 | C4colo | oops |
| 08:06.07 | C4colo | let me try that again |
| 08:06.12 | C4colo | something like exten => _*98XXX,1,VoicemailMain(${EXTEN:3}@default) |
| 08:06.16 | C4colo | strip the first three digits |
| 08:06.24 | C4colo | I really shouldn't write dialplan examples while distracted and tired |
| 08:07.04 | C4colo | I wonder if pattern matching works for CID matches |
| 08:07.39 | *** join/#asterisk xuser_ (n=xuser@unaffiliated/xuser) |
| 08:07.43 | C4colo | would be limited in use |
| 08:08.54 | C4colo | we actually have an inbound call macro that checks to see if the user is calling their own number, checks to see if call forwarding is enabled, sets the *69 data to the CID of this caller, etc, etc |
| 08:09.09 | C4colo | runs on every inbound call, takes care of all the redundant dialplan for each call |
| 08:09.53 | frogonwheels | C4colo: just been working on 'last dialed' 'last incoming' 'last missed' queues |
| 08:10.39 | C4colo | if you want more features like that, it is probably a good idea to create a macro for inbound and outbound calls |
| 08:10.56 | C4colo | then you can modify that one macro as much as you like and it will apply to all inbound/outbound calls |
| 08:12.12 | frogonwheels | C4colo: Yeah.. added difficulty - my dialplans are all generated by my openwrt uci to asterisk config converter .. |
| 08:13.21 | frogonwheels | .. but sure - I can set up common incoming /outgoing contexts and add the configs to that |
| 08:14.03 | C4colo | yea |
| 08:16.40 | frogonwheels | C4colo: one thing I'm doing, but wish it could be done nicer.. is to be able to have an incoming call - then after 10 secs answer it, but keep ringing, and allow a keypress to go through to a menu for access to internal menus |
| 08:16.58 | C4colo | our dialplan isn't as clean because we have to seperate everything out for multiple customers |
| 08:17.21 | frogonwheels | C4colo: well this thing is running on a router - so it's gotta be kept small anyway :) |
| 08:17.22 | C4colo | if you have a single-tennant setup you can use one outbound context and one inbound context with everything in it |
| 08:17.51 | C4colo | yea |
| 08:18.06 | C4colo | you could play a recording of ringing as the background for an IVR |
| 08:18.30 | C4colo | that takes up space though |
| 08:18.44 | frogonwheels | C4colo: problem is you actually have to stop the ringing briefly, answer the call, then start the ringing again. |
| 08:18.48 | C4colo | I don't know if you can play an indication like "ringing" as the sound for Read() |
| 08:18.54 | C4colo | yep |
| 08:19.10 | frogonwheels | C4colo: you can play an indication as the sound for Dial() |
| 08:19.20 | *** join/#asterisk lesouvage (n=lesouvag@dc51478cb8.adsl.wanadoo.nl) |
| 08:19.46 | frogonwheels | C4colo: so yeah- got it working.. but there's a brief pause.. and a number of times I've picked up during the pause.. and had to put down the phone and pick up again! |
| 08:20.24 | justdave | looks like I have to recompile asterisk with the dahdi drivers present, too |
| 08:20.52 | justdave | chan_dahdi picks which one to try to interact with at compile time |
| 08:21.51 | drmessano | Hmmm |
| 08:22.01 | drmessano | All my workstations here seem to have some bug in the IP stack |
| 08:22.06 | drmessano | They all have the same IP |
| 08:22.09 | drmessano | 127.0.0.1 |
| 08:22.11 | drmessano | Weird |
| 08:22.35 | drmessano | Wonder if they got hacked |
| 08:22.56 | drmessano | Ok, this is messed up, friend of mine has the same IP |
| 08:23.54 | C4colo | you probably did get hacked, because I don't think that IP is routable |
| 08:24.19 | drmessano | Hmmm |
| 08:28.34 | C4colo | check if someone put it in your /etc/hosts file |
| 08:28.47 | C4colo | that's the first place a hacker will put that ip address |
| 08:28.59 | C4colo | from what I can find on google anyway |
| 08:29.22 | *** join/#asterisk af_ (n=getsmart@88-149-241-15.dynamic.ngi.it) |
| 08:29.27 | drmessano | Oh crap |
| 08:29.32 | drmessano | Who the hell is "LOCALHOST" |
| 08:29.42 | drmessano | Some hacker group? |
| 08:29.52 | drmessano | DELETED! |
| 08:30.08 | C4colo | yea, they host their ripped movies and xbox games on your local machine |
| 08:30.17 | C4colo | that's why they call themselves 'localhost' |
| 08:30.34 | drmessano | Well, I am getting the last laugh now. |
| 08:30.45 | C4colo | http://letmegooglethatforyou.com/?q=127.0.0.1+hacked |
| 08:30.52 | drmessano | I noticed they also put my DNS servers in some "resolv.cfg" file |
| 08:31.03 | drmessano | Blew that out too |
| 08:31.19 | justdave | assumes you guys are kidding around and stays the hell out of the conversation |
| 08:31.36 | C4colo | see if you have files in /boot like vmlinuz-<some numbers> |
| 08:31.45 | C4colo | those are usually hacked files too |
| 08:32.27 | drmessano | Ah |
| 08:32.36 | drmessano | Those are gone too now |
| 08:33.04 | C4colo | I think they got my yum repos, probably DNS poisoning, because everytime I do a yum -y update I get more of those vmlinuz and initrd files |
| 08:33.40 | drmessano | I was just looking at the DNS server, and I dont know how Linux does it, but there's some crap called BIND on here. I thought NAMEd or some crap was the DNS server? |
| 08:34.06 | drmessano | I noticed it has some 127.0.0.1 stuff in it too |
| 08:34.12 | C4colo | you got hit with the bound-Ntied |
| 08:34.17 | C4colo | bound-n-tied/32 |
| 08:34.22 | drmessano | IS that how they spread the worm thing? |
| 08:34.33 | C4colo | probably |
| 08:34.44 | drmessano | I can just remove that with Yum |
| 08:34.55 | drmessano | It says its in the list ???? |
| 08:35.08 | drmessano | I will just remove the configs |
| 08:35.10 | *** join/#asterisk morglum (n=morglum@modemcable081.84-20-96.mc.videotron.ca) |
| 08:35.16 | *** part/#asterisk kerx (n=prepro@adsl-68-125-34-97.dsl.irvnca.pacbell.net) |
| 08:36.19 | C4colo | haha |
| 08:36.23 | C4colo | http://jim.blogsome.com/2005/06/06/i-hacked-127001/ |
| 08:37.10 | morglum | Hey everyone. I've got a "guaranteed to fail" setup that just failed on me and I could use some help regarding NAT and SIP. I've got a working asterisk behind a nat and a x-lite client behind another nat. The x-lite client can register and dial, but I eventually get a "maximum retries exceeded on transmission blablahblah -- see doc/sip-retransmit.txt' -- where can I get that file, or even better, can someone help me |
| 08:43.07 | drmessano | Um |
| 08:43.13 | drmessano | Not "guaranteed to fail" |
| 08:43.20 | drmessano | Easy |
| 08:43.33 | morglum | glad to hear that :) |
| 08:43.35 | drmessano | Do you have Asterisk configured properly, ports open? |
| 08:43.49 | drmessano | localnet, externhost/externip/nat=yes set? |
| 08:44.07 | morglum | from inside the network asterisk works fine -- this is the first time i try to access it from inside |
| 08:44.16 | drmessano | ?? nat |
| 08:44.16 | morglum | i probably havent set anything like that yet |
| 08:44.19 | drmessano | errr |
| 08:44.22 | morglum | from out side that is |
| 08:44.23 | drmessano | ~sipnat |
| 08:44.24 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 08:44.41 | morglum | drm: off to check |
| 08:45.19 | morglum | drm: that's in sip.conf, right? |
| 08:49.16 | drmessano | yeah |
| 08:50.03 | morglum | alright, under [general] i put the following: |
| 08:50.07 | morglum | nat=yes |
| 08:50.32 | morglum | externip = (my wan ip) |
| 08:50.37 | morglum | localnet=192.168.2.200 |
| 08:51.06 | morglum | same results ( can connect and dial, but get an error afterwarD) |
| 08:51.30 | drmessano | No |
| 08:51.40 | drmessano | localnet is a masked subnet |
| 08:51.56 | drmessano | localnet=192.168.2.0/255.255.255.0 |
| 08:52.08 | drmessano | local network |
| 08:52.23 | drmessano | Do you have the proper ports open? |
| 08:52.34 | morglum | oops, changed that will try. |
| 08:53.09 | morglum | ports-wise, i guess I'm fine : calls from inside asterisk's network work fine, and calls from the client using another network, say sip.nonoh.net, work fine |
| 08:53.10 | morglum | too |
| 08:53.34 | drmessano | What ports do you have open? |
| 08:53.49 | drmessano | calls from inside asterisk's network work fine <--- Inside is NOT the problem |
| 08:54.15 | morglum | 5060 and 10000-20000 for the client. the client works fine when connected to nonoh |
| 08:55.06 | drmessano | You dont need client ports open |
| 08:55.11 | drmessano | You need ports open on the PBX side |
| 08:55.12 | morglum | ok |
| 08:55.18 | drmessano | The client tranverses NAT fine |
| 08:55.22 | morglum | pbx also had these ports forwarded to it |
| 08:55.24 | drmessano | You can leave that closed |
| 08:55.29 | drmessano | Ok |
| 08:55.44 | morglum | (I'll close the client's one though - thanks) |
| 08:55.54 | Talkradio | shouldn't you be using vpn for outside client |
| 08:56.06 | drmessano | Why? |
| 08:56.18 | drmessano | People use client externally all the time |
| 08:56.46 | Talkradio | i was told they can be hacked and someone run up your phone bill |
| 08:56.53 | drmessano | ROFL |
| 08:57.06 | drmessano | Using weak passwords, yes |
| 08:57.11 | drmessano | Anything can be hacked |
| 08:57.30 | Talkradio | the info came from fonality :( |
| 08:57.36 | drmessano | They're idiots |
| 08:57.48 | drmessano | They are half the reason their systems get hacked.. |
| 08:58.13 | Talkradio | ok so it's not just voip in general |
| 08:58.20 | morglum | lol good to know that too |
| 08:58.51 | Talkradio | they told me it would crazy to open up those ports so i setup a vpn connection for the remote users |
| 08:59.16 | drmessano | LOL |
| 08:59.18 | morglum | drm: should I put NAT AND EXTERNIP AND EXTERNHOST, or only 1 or 2 of those? |
| 08:59.36 | drmessano | externip OR externhost... |
| 08:59.41 | morglum | ok |
| 08:59.41 | drmessano | Not both |
| 08:59.49 | morglum | that's what i did |
| 09:00.34 | drmessano | Talkradio: Fonality causes half of their own security holes. MANY, MANY, MANY of us have PBX's open to the outside without incident |
| 09:01.02 | morglum | drm: I basically only hear the first few rings-- I get silence when I expect my work's voicemail to have answered |
| 09:01.17 | morglum | that's still an issue with my nat, right? |
| 09:01.28 | drmessano | Did you reload? |
| 09:01.33 | Talkradio | ok thx for clearing that up drmessano |
| 09:01.45 | Talkradio | i'm learning little by little |
| 09:01.54 | drmessano | Talkradio: Use decent passwords and dont listen to Fonality |
| 09:02.08 | drmessano | Talkradio: Thats good advice to take you down the road |
| 09:02.12 | morglum | yup, killed and restarted |
| 09:02.37 | drmessano | Talkradio: They have a history of snooping on their own users without consent too |
| 09:03.10 | Talkradio | guess what we can't set the remote users or local users passwords since devices and mac addy's are set by fonality.. so now i understand what they mean it'll be insecure |
| 09:03.31 | morglum | x-lite says "call established" too .. |
| 09:03.32 | drmessano | Talkradio: Trixbox PRO? |
| 09:03.42 | Talkradio | call center edition |
| 09:03.47 | drmessano | morglum: Check SIP debug |
| 09:03.52 | drmessano | Talkradio: You poor bastard |
| 09:04.00 | Talkradio | yep |
| 09:04.07 | Talkradio | atleast it wasn't my decision |
| 09:04.08 | morglum | drm: command = sip set debug ? |
| 09:04.31 | drmessano | morglum: Yeah, and watch the call progress |
| 09:05.30 | Talkradio | so if you know someone who runs fonality tell them they setup mac addy as usersname and password on both my customers that have they same edition |
| 09:06.05 | Talkradio | so if someone reads the mac on the phone they instantly have a sip account |
| 09:07.16 | *** join/#asterisk lesouvage (n=lesouvag@dc51478cb8.adsl.wanadoo.nl) |
| 09:07.48 | drmessano | Talkradio: I dont know anyone using a paid Fonality product. Thankfully running and screaming from Trixbox CE was enough. |
| 09:08.00 | drmessano | But good to know |
| 09:08.06 | drmessano | Thats insecure as shit |
| 09:08.19 | morglum | drm : asterisk gets sip reads from myclientip:62951 --- isnt that a weird port for sip? |
| 09:08.28 | drmessano | No |
| 09:08.45 | drmessano | sports and dports |
| 09:08.52 | morglum | i cant really decipher it, but I cant see any blatant error message |
| 09:09.12 | morglum | quite a few "scheduling destruction of sip dialog blahblah@ |
| 09:09.27 | drmessano | can you call an extension that goes to a live person? |
| 09:09.49 | morglum | i could try that tomorrow, just not at 4am :) |
| 09:10.29 | drmessano | lol |
| 09:10.37 | drmessano | http://trends.google.com/websites?q=trixbox.org&sa=N <-- amusing |
| 09:11.01 | morglum | ahah |
| 09:11.14 | morglum | same for digium.com |
| 09:11.28 | drmessano | http://trends.google.com/websites?q=fonality.com&sa=N |
| 09:13.34 | morglum | curse you, sound! |
| 09:13.46 | morglum | guess i'll just go get some sleep |
| 09:13.50 | morglum | thanks for trying :) |
| 09:14.04 | drmessano | No probs.. I dont think youre far off here |
| 09:14.11 | drmessano | What sort of firewall on each end? |
| 09:14.20 | drmessano | Dont say "PIX" or "PFsense" |
| 09:14.22 | drmessano | Please |
| 09:14.51 | morglum | actually nothing but the router (tomato on a wrt54gl) |
| 09:15.11 | drmessano | on both ends? |
| 09:15.39 | C4colo | ok that's creepy |
| 09:15.47 | morglum | for the pbx .. the client also only uses the router for firewall an old usr 8054 |
| 09:15.51 | C4colo | search digium.com then avaya.com right after |
| 09:15.59 | C4colo | hit "back" and "forward" a couple times |
| 09:16.00 | drmessano | USR 8054? |
| 09:16.06 | C4colo | the graphs are almost identical |
| 09:16.18 | morglum | old router --- anyway, i thought the firewall on the client's end didnt matter |
| 09:16.31 | drmessano | Oh god |
| 09:16.44 | drmessano | If the firewall isn't handling SIP properly it does matter |
| 09:16.50 | drmessano | Open ports do not matter |
| 09:16.55 | drmessano | NAT transversal does |
| 09:17.02 | morglum | well the client can connect to sip.nonoh.net and place calls -- so I guess it does work? |
| 09:17.20 | *** join/#asterisk grndslm (n=grndslm@24-119-80-142.cpe.cableone.net) |
| 09:17.42 | drmessano | Does the Asterisk box have 2 NICS? |
| 09:18.07 | morglum | (internet) --> wrt54gl with tomato --> asterisk box with 1 nic |
| 09:18.25 | morglum | (internet) -->dsl modem--> wrt54gl with tomato --> asterisk box with 1 nic |
| 09:18.25 | drmessano | ok |
| 09:18.27 | morglum | asctually |
| 09:18.44 | drmessano | and the DSL modem is in passthru? |
| 09:19.36 | morglum | think so -- i actually didnt do anything to the modem.. just plugged it in, and put the pppoe setting in the router |
| 09:19.44 | morglum | the router is the dhcp server |
| 09:20.17 | drmessano | Does the router have the public IP on its WAN interface? |
| 09:20.38 | morglum | yup |
| 09:20.44 | drmessano | ok cool |
| 09:21.09 | drmessano | Well, two things here |
| 09:21.23 | morglum | (listens) |
| 09:21.31 | drmessano | One is misconfig in the dialplan somewhere.. which shouldnt be the case if its working internally |
| 09:21.46 | drmessano | Another may be an issue with Tomato and SIP.. |
| 09:22.02 | morglum | uh oh |
| 09:22.14 | drmessano | May be worth Googling for that |
| 09:22.16 | morglum | off for some googling |
| 09:22.17 | morglum | :) |
| 09:22.21 | morglum | tomorrow anyway :) |
| 09:27.57 | *** join/#asterisk cuco (n=diego@CBL217-132-108-208.bb.netvision.net.il) |
| 09:28.06 | *** part/#asterisk cuco (n=diego@CBL217-132-108-208.bb.netvision.net.il) |
| 09:38.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 09:42.38 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.157) |
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| 11:00.06 | hi365 | cnn.com |
| 11:17.47 | *** join/#asterisk andresmujica (n=andresmu@201.244.110.136) |
| 11:30.33 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-51-125.w86-215.abo.wanadoo.fr) |
| 11:31.37 | *** join/#asterisk markwaters (n=markwate@80.118.244.23) |
| 11:33.04 | markwaters | when I use the monitor command with the 'm' switch it concatenates the 2 wav files instead of mixing them |
| 11:33.28 | markwaters | yet when I run soxmix file1 file2 file3 it creates the mix correctly |
| 11:33.38 | markwaters | anyone else seen that / |
| 11:33.44 | markwaters | I`am on debian 4 |
| 11:34.02 | *** join/#asterisk lesouvage (n=lesouvag@dc51478cb8.adsl.wanadoo.nl) |
| 11:34.09 | tzafrir_laptop | markwaters, you have the wrong mixing command set? |
| 11:34.13 | markwaters | tried it as root and asterisk user to same effect |
| 11:34.25 | *** join/#asterisk sm7syx (n=marten@147.113.227.87.static.ld.siw.siwnet.net) |
| 11:34.38 | markwaters | I thought soxmix command was 'built-in' |
| 11:35.35 | sm7syx | My * is working properly BUT I can't connect to builtin http server |
| 11:35.39 | markwaters | where is that set ? |
| 11:44.51 | tzafrir_laptop | sm7syx, is it running? netstat -lntp | grep 8080 |
| 11:45.26 | *** join/#asterisk dieguito84 (n=diego@host26-186-dynamic.23-79-r.retail.telecomitalia.it) |
| 11:45.26 | tzafrir_laptop | markwaters, it's part of the package sox . See also 'core show application Monitor' |
| 11:45.56 | markwaters | tzafrir_laptop: yea I got sox package installed |
| 11:46.05 | markwaters | soxmix is available in the shell |
| 11:46.35 | markwaters | when I run it in the shell with soxmix file1 file2 file3 the file3 is created / mixed correctly |
| 11:47.04 | sm7syx | tzafrir_laptop, Nope can't find it! How do I start it?? I have done reload and start / stop of asterisk |
| 11:47.05 | markwaters | yet when asterisk runs the monitor command with 'm' to mix it , it concatenates the file instead of mixing it |
| 11:49.12 | markwaters | tzafrir_laptop: just defined it with :- MONITOR_EXEC=/usr/bin/soxmix |
| 11:49.16 | markwaters | and it works now |
| 11:49.29 | markwaters | yet leaves the 2 in and out files behind |
| 11:49.31 | markwaters | *sigh* |
| 11:49.35 | markwaters | oh well |
| 11:49.41 | markwaters | thanks for the clue |
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| 12:35.52 | asterisk_user2 | hello |
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| 12:59.49 | phaeron | hello everyone , I just upgraded to asterisk debian 1.4.21.2 , and it refuses to work , I can't see what's the problem in the logs. Any pointers on how to start debugging this ? |
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| 13:15.45 | tzafrir_laptop | phaeron, what problem do you actually see? |
| 13:15.57 | tzafrir_laptop | also try: /etc/init.d/asterisk debug |
| 13:20.27 | *** part/#asterisk asterisk_user2 (n=jim@195-240-249-117.ip.telfort.nl) |
| 13:24.12 | phaeron | tzafrir_laptop: ok one sec |
| 13:24.56 | phaeron | tzafrir_laptop: it outputs a lot of stuff and then dies with 4 warnings at the end |
| 13:25.13 | tzafrir_laptop | Could you please pastebin those warnings? |
| 13:25.57 | phaeron | tzafrir_laptop: ok |
| 13:26.35 | phaeron | tzafrir_laptop: http://pastebin.com/m55e16209 |
| 13:28.53 | tzafrir_laptop | phaeron, make sure that your /etc/asterisk/modules.conf includes: noload => app_directory_odbc.so |
| 13:29.32 | tzafrir_laptop | and likewise for app_voicemail_odbc.so and app_voicemail_imap.so |
| 13:31.19 | phaeron | tzafrir_laptop: done , still same warning and error |
| 13:32.34 | tzafrir_laptop | could you please pastebin /etc/asterisk/modules.conf ? |
| 13:32.41 | phaeron | tzafrir_laptop: all errors and warnings http://pastebin.com/m1d1884a |
| 13:33.47 | phaeron | tzafrir_laptop: http://pastebin.com/m4d725d6b heres the .conf file |
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| 13:38.57 | phaeron | tzafrir_laptop: I disabled autoload and it runs , but the phones don't work |
| 13:39.31 | tzafrir_laptop | do you have any modules not installed from packages? |
| 13:40.02 | phaeron | maybe leftover from the guy who worked on it |
| 13:40.20 | phaeron | should I remove all plugins and reinstall the package ? |
| 13:40.37 | tzafrir_laptop | hmm... you disabled it in modules.conf? this means that modules.conf is readable by asterisk |
| 13:41.57 | tzafrir_laptop | dpkg -S /usr/lib/asterisk/modules/*.so | sort # should give you an idea |
| 13:42.16 | phaeron | tzafrir_laptop: already doing for i in *.so ; do dpkg -S $i ; done |
| 13:42.20 | phaeron | tzafrir_laptop: one sec |
| 13:42.52 | phaeron | dpkg: *app_saycountpl.so* not found. |
| 13:42.52 | phaeron | dpkg: *codec_ilbc.so* not found. |
| 13:42.55 | phaeron | removed them |
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| 13:43.46 | phaeron | reverted to autoload and it still fails |
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| 14:07.21 | Mahmoud | which variable stores dialed exten/number when using dial() application? |
| 14:09.41 | phaeron | tzafrir_laptop: It started working when I added noload for everything and then selectively enabled modules |
| 14:09.56 | phaeron | I guess I can find the broken module one by one |
| 14:10.06 | phaeron | by exclusion that is |
| 14:12.18 | tzafrir_laptop | phaeron, those two could be from a source installation. codec_ilbc has long been blacklisted by Debian for patent issues. saycountpl has not been packaged because, well, nobody really needed it |
| 14:12.43 | phaeron | tzafrir_laptop: but it still failed after I removed them |
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| 15:28.00 | tzafrir_laptop | phaeron, any news? |
| 15:28.12 | phaeron | tzafrir_laptop: yeah it's working now |
| 15:28.23 | tzafrir_laptop | what module made the difference? |
| 15:29.12 | phaeron | tzafrir_laptop: I think it is chan_capi.so |
| 15:30.19 | tzafrir_laptop | interesting. That's from a separate package |
| 15:30.25 | tzafrir_laptop | or supposed to be |
| 15:32.39 | phaeron | tzafrir_laptop: yeah it is |
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| 15:54.28 | phaeron | tzafrir_laptop: thanks for the help |
| 15:54.30 | phaeron | gtg |
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| 18:29.16 | Assid | heya |
| 18:29.41 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 18:29.53 | lesouvage | . |
| 18:30.10 | Assid | i have a question regarding dial cmd.. or a hack around this. Currently If the channels /sip devices i am trying to call arent reachable. It goes straight to voicemail. It doesnt wait for the 30 seconds. I have set it to |
| 18:30.21 | Assid | is there a way to to force it to wait for the 30 seconds no matter what |
| 18:30.40 | Assid | i.e. even if the channels are not available |
| 18:30.47 | [TK]D-Fender | Assid: Dial your SIP channel, and a local channel that only does a "wait 30" |
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| 18:41.40 | Assid | [TK]D-Fender: congestion after that ? |
| 18:42.48 | [TK]D-Fender | <PROTECTED> |
| 18:43.46 | jaytee | I like Sudafed for congestion, it really helps clear out the sinuses |
| 18:44.51 | [TK]D-Fender | jaytee: If you can find the real stuff... |
| 18:45.09 | Assid | im not getting a ring yet |
| 18:45.13 | Assid | just empty ... |
| 18:45.17 | [TK]D-Fender | jaytee: Pathetic "War On Drugs" has seen the real deal drop off of shelves and its replacement is little better than a placebo |
| 18:46.09 | sm7syx | My * is working properly BUT I can't connect to builtin http server |
| 18:46.35 | jaytee | [TK]D-Fender, yes but look at all the good it's done! now we don't have to spend as much time on finding small meth labs here in the US. We can just try to interdict the massive flood of the crap as it comes across the Mexican border. |
| 18:47.19 | [TK]D-Fender | jaytee: Pseudophedrine Hydrochloride is my friend... my very bestest friend EVAR |
| 18:47.32 | [TK]D-Fender | jaytee: that & Ibuprofen |
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| 18:55.49 | *** join/#asterisk Borai (i=Bora@S0106001c109e98db.no.shawcable.net) |
| 18:55.51 | Borai | hi guys |
| 18:56.11 | *** join/#asterisk Mahmoud (n=foo@unaffiliated/mahmoud) |
| 18:56.32 | Mahmoud | any how to fix this error? i want to call external sources: [Dec 28 22:47:24] NOTICE[4562]: chan_iax2.c:7500 socket_process: Rejected connect attempt from 10.1.0.123, who was trying to reach 'iax2:888@conference.freeswitch.org' |
| 18:57.42 | Borai | anyone here with a Linksys Spa3102 |
| 18:57.44 | [TK]D-Fender | Mahmoud: Maybe if you actaully showed te complete callwith debug... |
| 18:58.16 | [TK]D-Fender | Borai: Move ahead to your actual question... |
| 18:59.36 | Borai | Well I am trying to configure one of those remotely, the problem is that I cant get to the pstn from the phone the handset connected to it nor from asterisk |
| 18:59.45 | Borai | but inbound works for some reason |
| 19:01.33 | [TK]D-Fender | Borai: pastebin your failed attempt with SIP debug enabled |
| 19:01.34 | [TK]D-Fender | ~pb |
| 19:01.35 | 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/ |
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| 19:03.47 | Borai | ok ill do that when i have access its night at the area right now, sorry. |
| 19:05.46 | [TK]D-Fender | Borai: Yes, try asking for help when you're in a position to DO something about it. |
| 19:06.19 | [TK]D-Fender | Borai: I'd love to know what you thought you were going to get asking like that... |
| 19:09.35 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 19:11.06 | Borai | Honestly, I have read that its a common problem that is why I asked if someone was using it at the first place. |
| 19:13.16 | jaytee | I've used the SPA-3102 with Asterisk. The default config it comes with won't work but........Google is your friend!!! |
| 19:13.55 | [TK]D-Fender | Borai: "it doesn't work" = tells us absolutely nothing |
| 19:14.27 | Borai | I know that, and you are right it was stupid of me saying that. |
| 19:14.29 | [TK]D-Fender | Borai: You can't look at its configs, or do tests, so you're running blind |
| 19:14.47 | [TK]D-Fender | Borai: Do come back when you've got access to it... |
| 19:14.58 | [TK]D-Fender | drmessano: All about being "accessible", right? |
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| 19:21.05 | *** join/#asterisk AJayMN (n=me@66-191-127-82.dhcp.mdsn.wi.charter.com) |
| 19:21.32 | AJayMN | I found a ClicktoCall script once that called the person then connected it to an Asterisk Extension Anyone know of this script I cant find it |
| 19:26.31 | [TK]D-Fender | AJayMN: You speak as if there weren't DOZENS of scripts like this. |
| 19:26.43 | [TK]D-Fender | AJayMN: there ARE,a dn we have no idea about which one YOU are thinking of. |
| 19:28.50 | AJayMN | im looking for any that work.. |
| 19:29.06 | AJayMN | but i need it to first call the number entered. then connect it to an internal exten |
| 19:29.51 | [TK]D-Fender | AJayMN: go read up on AMI / CLI Originate & call files |
| 19:44.59 | [TK]D-Fender | Translation : Clearly knows very little, wants to know even less and expects a ready-made functional hand-out |
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| 19:46.39 | jaytee | http://www.osburn.com/asterisk-sign.png |
| 19:46.56 | drmessano | lol |
| 19:47.34 | Borai | lol |
| 19:47.39 | drmessano | yep |
| 19:47.57 | drmessano | I am gonna put a NAT guide on my wiki later |
| 19:48.07 | drmessano | Im sick of the crap docs out there.. |
| 19:49.05 | x86 | better than sipnat? |
| 19:49.05 | x86 | ~sipnat |
| 19:49.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:49.11 | drmessano | I need to update my PBX build guide too |
| 19:49.28 | [TK]D-Fender | drmessano: Whats off on that one? |
| 19:49.46 | [TK]D-Fender | drmessano: I did want to go into better detail concerning externhost as well... |
| 19:50.15 | [TK]D-Fender | drmessano: Aside from that, whats missing? perhaps some SIP debug to show the contact ehader people miss? |
| 19:50.42 | drmessano | Even the first one needs to be trimmed a little and laid out more [TK]D-Fender style |
| 19:50.59 | jaytee | you two should collaborate on a non-O'Reilly Asterisk book |
| 19:51.03 | [TK]D-Fender | drmessano: the first link IS my guide. |
| 19:51.09 | drmessano | I know |
| 19:51.22 | [TK]D-Fender | drmessano: So you eman "bludgeon them over the head" style? |
| 19:51.29 | [TK]D-Fender | mean* |
| 19:51.36 | drmessano | Yeah, minus some expletives |
| 19:51.48 | [TK]D-Fender | drmessano: Will see what I can do :) |
| 19:52.00 | drmessano | Hang on |
| 19:52.03 | drmessano | I will pastebin my idea |
| 19:52.12 | drmessano | Copying your text as a base |
| 19:53.50 | jaytee | [TK]D-Fender, you're example is just way too clear. You need to add more "vagueness" otherwise every noob on the planet will become an overnight "expert" and then there will be no end to the madness! |
| 19:54.58 | jaytee | and drmessano, I'm a little pissed you haven't accepted my Mafia Wars invite. I've got a criminal empire to run here and you're holding up the works! |
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| 20:02.15 | drmessano | http://www.pastebin.ca/1295293 |
| 20:02.59 | drmessano | Little head beating added, and inclusion of externhost, which I would argue if you ignore it, they will find it mentioned somewhere else and add it anyway. |
| 20:03.08 | drmessano | So may as well explain it |
| 20:03.29 | drmessano | Especially if they are in Peru and their IP changes every 30.2 seconds |
| 20:04.07 | jaytee | yeah, and the VOIP service on Macchu Picchu really sucks |
| 20:04.24 | drmessano | I cant remember if it was indeed Peru or not |
| 20:04.37 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 20:04.40 | drmessano | But I remember something obnoxious like that.. Every 3 minutes |
| 20:08.07 | drmessano | sip,conf.. ewww |
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| 20:09.35 | drmessano | http://www.pastebin.ca/1295295 |
| 20:09.35 | drmessano | Took out one part of a sentence.. |
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| 20:10.43 | frogonwheels | drmessano: presumably canreinvite=nonat is just as good? |
| 20:13.27 | drmessano | That was new for 1.4.. |
| 20:14.58 | frogonwheels | oh. of course. |
| 20:17.21 | frogonwheels | hey, that arcomputing link above has diallow=all allow=ulaw - any reason _that_ example would do that? |
| 20:17.33 | frogonwheels | (I use the same- but that's so my router doesn't have to transcode) |
| 20:18.06 | [TK]D-Fender | frogonwheels: Yes, because it shows someone not being an idiot by trying to allow EVERYTHING and getting FUBAR'd when the call decides to agree on G.723 |
| 20:18.38 | [TK]D-Fender | frogonwheels: "Can't I jsut use NONAT" <- more people asking for trouble |
| 20:18.49 | frogonwheels | .. and G.723 is the licensed one yeah? what else is bad aboutit? |
| 20:19.26 | frogonwheels | [TK]D-Fender: why is nonat asking for trouble? is it broke? |
| 20:19.54 | [TK]D-Fender | frogonwheels: the part where there is no legal software codec for it and * can't transcode it. |
| 20:20.22 | [TK]D-Fender | frogonwheels: Unless you feel like buying a TC400 |
| 20:20.42 | frogonwheels | [TK]D-Fender: oh, that part. fair enough. |
| 20:20.43 | [TK]D-Fender | frogonwheels: You exhibit every reason these comments are added |
| 20:23.00 | frogonwheels | [TK]D-Fender: Well that was a dumb and unhelpful comment. |
| 20:23.47 | [TK]D-Fender | frogonwheels: what is there to "help"? |
| 20:24.12 | [TK]D-Fender | frogonwheels: You haven't stated a problem, just keep exhibiting the kind of tendencies that generate these problems. |
| 20:24.41 | frogonwheels | [TK]D-Fender: was asking about why nonat is trouble. |
| 20:25.47 | frogonwheels | [TK]D-Fender: presumably somebody went to the bother of implementing it, and I'm kinda assuming that the asterisk devs aren't stupid. |
| 20:29.02 | [TK]D-Fender | frogonwheels: Go try |
| 20:29.30 | frogonwheels | [TK]D-Fender: I use it - it seems to work. |
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| 21:08.51 | Borai | ips in peru change every 30.2 secs :O |
| 21:09.39 | frogonwheels | Borai: that'd be entertaining.. |
| 21:21.17 | ManxPower | Especially with downloads that take longer than 30.2 seconds |
| 21:27.20 | Borai | 565,1,Dial(SIP/18@pstn-spa3k,60,) how would you add a pause before it calls 18 after connecting to spa3k |
| 21:27.36 | hardwire | 30.2? |
| 21:29.12 | [TK]D-Fender | Borai: I doubt you can. You'd need to tell the SPA to delay the dial so the telco has time to fully register going off-hook |
| 21:29.29 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 21:29.39 | *** join/#asterisk bradleyprice86 (n=bradleyp@99-206-197-198.pools.spcsdns.net) |
| 21:30.27 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 21:30.48 | Borai | ok let me check that |
| 21:31.32 | ManxPower | Borai: See the D option to Dial |
| 21:31.53 | [TK]D-Fender | ManxPower: Not sure there's a way to manage that with an SPA... |
| 21:32.09 | [TK]D-Fender | ManxPower: To basically doa null-dial THEN do the digits |
| 21:32.18 | [TK]D-Fender | ManxPower: the "null" is the hard part in that |
| 21:32.32 | [TK]D-Fender | ManxPower: but the idea is sound, though an ugly thought |
| 21:32.38 | ManxPower | you would put , in front of the digits in D. i.e. D(,,,,555) |
| 21:32.48 | ManxPower | , means .5 second pause. |
| 21:35.04 | ManxPower | I assume the SPA is connected to a POTS line, if so, the call is considered answered as soon as dialing is finished, so you'll have to play around with the number of , |
| 21:37.28 | Borai | i can get calls, i can call but not out of pstn |
| 21:37.46 | *** join/#asterisk outtolunc (n=me@c-67-164-8-168.hsd1.ca.comcast.net) |
| 21:37.53 | Borai | so you cant do a dialtone detection? |
| 21:37.57 | Borai | to get it off-hook state |
| 21:38.18 | drmessano | Why do you think you need this pause? |
| 21:39.56 | Borai | because i always get circuit is busy |
| 21:40.07 | [TK]D-Fender | drmessano: Sometimes FXO side needs a sec to fully register or you lose a digit, etc |
| 21:40.20 | [TK]D-Fender | Borai: pardon? PASTEBIN IT |
| 21:40.29 | [TK]D-Fender | ~pb |
| 21:40.29 | 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/ |
| 21:41.01 | Borai | ok |
| 21:41.55 | drmessano | I didnt want to know why it needs to be done |
| 21:42.09 | Borai | http://pastebin.com/m6d1bb00d |
| 21:42.10 | drmessano | I wanted to know why HE thinks HE needs it in HIS situation and what indication HE had |
| 21:42.18 | drmessano | However, it sounds like your config is wrong |
| 21:42.53 | drmessano | http://wiki.2l2o.com/index.php/Linksys_SPA-3102 |
| 21:42.55 | drmessano | Follow that |
| 21:43.03 | drmessano | Including factory reset and set up the options again |
| 21:43.11 | Borai | ok |
| 21:44.13 | Borai | most likely its an issue with the config i never said my config is perfect. |
| 21:45.06 | drmessano | 105 out of 100 people set up SPA-3102's wrong the first time |
| 21:45.19 | Borai | Remove Feature Codes? what do you have to remove |
| 21:45.39 | [TK]D-Fender | Borai: Enable SIP DEBUG and next time pastebin the ENTIRE CALL |
| 21:45.47 | [TK]D-Fender | Borai: You have shown us nothing |
| 21:45.49 | joat | the vertical feature codes |
| 21:45.55 | Borai | ok |
| 21:46.21 | joat | he's trying to follow the link that drmessano gave him |
| 21:46.25 | Borai | Vertical Service Activation Codes |
| 21:46.28 | Borai | anything else i have to remove there |
| 21:46.36 | drmessano | READ the DOC |
| 21:47.02 | [TK]D-Fender | Borai: Stop. Now. Go pastebin a COMPLETE CALL with SIP DEBUG as requested and we'll see from there. |
| 21:47.23 | drmessano | He's better off starting off with a properly configged adapter |
| 21:48.22 | Borai | wouldn't it be better if i do as drmessano said? (honestly i am guessing that the problem is at the spa not *) |
| 21:48.25 | *** join/#asterisk ManxPower (n=manxpowe@adsl-157-16-222.msy.bellsouth.net) |
| 21:48.48 | drmessano | Make sure you FACTORY RESET it FIRST |
| 21:48.57 | drmessano | So the ugly can be removed |
| 21:49.58 | Borai | i did a factory reset |
| 21:50.25 | Borai | then i cleared Vertical Service Activation Codes but that page is full of items |
| 21:50.45 | drmessano | Ok, move along |
| 21:53.53 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 21:54.43 | [TK]D-Fender | Yup... I think I'm done helping the hopeless today... |
| 21:58.43 | Borai | http://pastebin.com/m5a8a59a3 |
| 21:58.47 | Borai | here sip debug |
| 22:02.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:04.39 | [TK]D-Fender | Borai: -- Executing [565@home:1] Dial("SIP/100-08327080", "SIP/||18@pstn-spa3k|60|") in new stack <- formatting is junk |
| 22:05.42 | Borai | Mind pointing me out what part of my formating is wrong please? |
| 22:07.02 | hardwire | sure.. why is 18@pstn-spa3k the flags for the call? |
| 22:07.31 | hardwire | did you already post the dialplan calling 18@pstn-spa3k |
| 22:08.22 | ManxPower | The line [TK]D-Fender pasted indicates that your Dial line is something like Dial(SIP/,,18@pstn-spa3k,60,) which is totally wrong. |
| 22:10.03 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:10.16 | [TK]D-Fender | Borai: You added ","s or "|" where they do not belong |
| 22:10.30 | ManxPower | And not a D() in sight |
| 22:10.38 | ManxPower | Borai: you're not using a GUI are you? |
| 22:13.37 | Borai | no i am not using a gui |
| 22:14.41 | Borai | , not | |
| 22:15.48 | *** join/#asterisk BajaEd (n=ednagy@72.170.62.90) |
| 22:16.34 | Borai | i cant find the usage of D( oin voip-info |
| 22:16.43 | [TK]D-Fender | Borai: well you know what you should not have done, go fix your formatting |
| 22:17.16 | Borai | i think the one after SIP/ shouldn't be there am i right? |
| 22:17.26 | [TK]D-Fender | Borai: neither |
| 22:17.42 | [TK]D-Fender | Borai: You put TWO of those where you should have none. |
| 22:17.51 | [TK]D-Fender | Borai: "core show application dial" |
| 22:18.05 | drmessano | WTF |
| 22:18.41 | drmessano | BajaEd: I don't accept unsolicited private messages, and certainly not unsolicited DCC chats |
| 22:18.55 | drmessano | BajaEd: I will not have sex with you |
| 22:19.25 | Borai | D([called]) |
| 22:19.37 | JJ2110 | Never have sex with anyone from IRC |
| 22:19.41 | JJ2110 | Thats just asking for HIV |
| 22:19.41 | Borai | so I have to dial SIP/pstn-spa3k and then use D(18)? |
| 22:21.35 | Borai | or better something like DIAL(SIP/pstn-spa3k,60,D18)? |
| 22:21.44 | drmessano | What is D18? |
| 22:21.54 | drmessano | throws a D20 |
| 22:22.10 | drmessano | Yay, i'm a wizard! |
| 22:23.06 | Borai | well i want to dial 18 using the pstn port on my spa3102 |
| 22:25.19 | [TK]D-Fender | Borai: DIAL(SIP/pstn-spa3k/18,60) |
| 22:27.49 | Borai | Thank you. Now I want to go do some research why I cant use the 18@pstn-spa3k sytax. |
| 22:29.07 | Borai | thats the syntax that was shown on voxilla |
| 22:30.16 | [TK]D-Fender | Borai: You never showed us you doing it RIGHT |
| 22:31.33 | Borai | Well I learned my lesson I wont follow guides from now on. |
| 22:31.44 | [TK]D-Fender | Borai: Last statement stands... |
| 22:31.46 | Borai | and try figuring it out by myself |
| 22:32.05 | [TK]D-Fender | Borai: And Feel free to show us the guide you're following, because I've never seen one with the errors you just showed us |
| 22:32.15 | Borai | i followed the guides i found on the internet. |
| 22:32.23 | Borai | not with the ,, error |
| 22:32.39 | [TK]D-Fender | Borai: &^#$%ING LINK IT |
| 22:32.52 | [TK]D-Fender | Borai: And as I said you never showed us 1 VALID attempt. |
| 22:33.02 | [TK]D-Fender | boriSo stop running around like a headless chicken |
| 22:33.33 | Borai | https://voxilla.com/voxilla_tools/config_wizard/spa3kasterisk.pl |
| 22:33.56 | [TK]D-Fender | Borai: bad link |
| 22:34.02 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 22:34.11 | Borai | ok |
| 22:34.21 | Borai | http://voxilla.com/tools/device-configuration-wizard/linksys-spa3102-spa3000-configuration-wizard-for-asterisk |
| 22:34.52 | drmessano | Oh christ |
| 22:34.55 | drmessano | A wizard |
| 22:35.04 | jaytee | did someone call me? |
| 22:35.06 | [TK]D-Fender | Borai: Next that configures the DEVICE, not ASTERISK |
| 22:35.17 | Borai | click submit |
| 22:35.21 | drmessano | WIIIIZARD |
| 22:35.25 | Borai | look at the next page |
| 22:35.33 | Borai | exten => 911,1,Dial(SIP/911@pstn-spa3k,60,) |
| 22:35.45 | Borai | thats where i got the idea of dialing SIP/18@pstn-spa3k,60,) |
| 22:35.54 | joat | there's a picture, jaytee on the side of a van... |
| 22:36.16 | Borai | then I asked how i could put a pause there before dialing 18 because I thought that the second the call connected with the spa it tried to connect it and couldnt because of the hook state |
| 22:36.27 | Borai | so i wanted to put a pause and i asked how i could put a pause on this channel |
| 22:36.43 | Borai | and you said i should use , and that each , means .5 pause |
| 22:36.52 | Borai | thats when i added the 2 ,, there |
| 22:37.03 | jaytee | yep, me dressed as Jesus on the side of a van with the caption underneath the picture saying, "Would you like fries with that while you burn in Hell?" |
| 22:37.08 | [TK]D-Fender | Borai: http://pastebin.com/m737240f1 |
| 22:37.16 | [TK]D-Fender | Borai: this did NOT get extra garbage commas |
| 22:37.39 | Borai | [TK]D-Fender |
| 22:37.44 | [TK]D-Fender | Borai: Borai So lets stop wasteing time and show us a VALID attempt |
| 22:37.51 | Borai | yes that guide doesnt give it but the guide didnt work |
| 22:38.01 | [TK]D-Fender | Borai: I just walked through that wizard myself to end up with that reult |
| 22:38.06 | Borai | ok |
| 22:38.11 | Borai | i said that i ended up with |
| 22:38.15 | *** join/#asterisk etfonhomey (n=chatzill@74-143-196-254.static.insightbb.com) |
| 22:38.32 | [TK]D-Fender | Borai: Go try a PROPERLY formatted call attempt |
| 22:39.47 | Borai | Im sorry that I wasted your time on a stupid issue. |
| 22:42.56 | Borai | I seriously need more sleep =(. |
| 22:43.07 | [TK]D-Fender | Borai: You're wasting our time if you don't get off your ass and show us a valid attempt |
| 22:44.05 | Borai | I will but I can not right now. |
| 22:44.13 | [TK]D-Fender | LOVELY |
| 22:44.36 | Borai | no because there is a connection problem with the other party right now |
| 22:44.47 | [TK]D-Fender | Borai: What other party? |
| 22:44.57 | Borai | where the spa is |
| 22:45.11 | [TK]D-Fender | Borai: What does this other party have to do? |
| 22:45.40 | Borai | well i am assuming you want to see the debug of a call going thru |
| 22:45.46 | Borai | am i wrong? |
| 22:45.53 | [TK]D-Fender | Borai: Going though? You're not even CLOSE yet. |
| 22:46.07 | [TK]D-Fender | Borai: You haven't gotten the SPA to ACCEPT a call yet |
| 22:46.16 | [TK]D-Fender | Borai: Get that far THEN I'll care about the rest |
| 22:46.34 | Borai | ok |
| 22:46.52 | Borai | Ill work on it and update you. |
| 22:47.45 | Borai | I got the call routing from pstn, and calling from voip to spa working as well as from spa to voip but not the voip-to-pstn part |
| 22:47.58 | [TK]D-Fender | Borai: Its 1 DIAL LINE... whats this "work on it" story? remove the junk chars and DIAL |
| 22:48.15 | [TK]D-Fender | Borai: I should have had the answer in the time it took me to type that last line. |
| 22:49.02 | *** join/#asterisk Decepticon (n=Deceptic@ks361773.kimsufi.com) |
| 22:49.28 | Borai | Ok honestly speaking if I dial I will wake up my sister which wont be a real good idea. |
| 22:49.51 | Borai | else i would have done it by the time i removed the garbage |
| 22:50.09 | [TK]D-Fender | Well this was just a cataclysmic waste of time... |
| 22:50.16 | [TK]D-Fender | goes on to do something productive |
| 22:50.22 | Borai | exten => 565,1,DIAL(SIP/pstn-spa3k/18,60) |
| 22:50.25 | Decepticon | where can i learn what asterisk is and how i can use it to make cheaper calls with my cellphone, (sip client) |
| 22:50.56 | drmessano | You cant use it for cheaper calls |
| 22:51.06 | [TK]D-Fender | Decepticon: Just look atht e call rates of any ITSP and look at a plan combination that calling in to * and back out again costs you less |
| 22:51.09 | drmessano | Its a PBX, not a ouji board |
| 22:51.14 | Borai | [TK]D-Fender: I'm sorry for stealing your time. |
| 22:51.23 | Decepticon | hm, tell me more |
| 22:51.36 | [TK]D-Fender | Decepticon: * does not make your call cheaper. Some service you might use WITH * might be more economical |
| 22:51.46 | [TK]D-Fender | ~itsplist-us |
| 22:51.47 | 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 |
| 22:51.58 | [TK]D-Fender | Decepticon: Go shop around |
| 22:52.05 | Borai | Nufone is good I like them |
| 22:52.32 | Decepticon | i am new to voip, what is a ITSP and what is the model around which asterisk works (what connects to what and over what mediums) |
| 22:52.40 | [TK]D-Fender | ~itsp |
| 22:52.40 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:56.13 | *** part/#asterisk xorl (n=xorl@li30-130.members.linode.com) |
| 23:09.50 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
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| 23:15.58 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:16.47 | *** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil) |
| 23:18.27 | *** join/#asterisk eppigy (n=Dave@plasticlobster.com) |
| 23:31.03 | JJ2110 | core show application system |
| 23:31.04 | JJ2110 | hmm |
| 23:32.13 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 23:32.31 | drmessano | doesnt work in IRC |
| 23:32.43 | JJ2110 | I know :) |
| 23:35.07 | eppigy | trippin |
| 23:36.03 | JJ2110 | Cool, call failed 603 |
| 23:42.56 | JJ2110 | drmessano, care to help me with the System command? |
| 23:43.12 | drmessano | System(Cool Stuff Here) |
| 23:43.46 | JJ2110 | System(http://192.168.1.65/play.php) |
| 23:43.48 | JJ2110 | Spot the error? |
| 23:44.03 | JJ2110 | Do I need "http://..."? |
| 23:44.11 | drmessano | What are you trying to get that to do? |
| 23:44.24 | [TK]D-Fender | JJ2110: It calls a UNIX CLI. Who said anything about being able to hand it a URL? |
| 23:44.29 | JJ2110 | run the php script on the user |
| 23:44.59 | JJ2110 | People did yesterday :) |
| 23:45.04 | [TK]D-Fender | JJ2110: System(wget http://192.168.1.65/play.php) |
| 23:45.11 | JJ2110 | Ah |
| 23:45.56 | JJ2110 | Sweet :D That works, thanks :) |
| 23:53.48 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
