| 00:02.08 | `md | can someone help me with my asterisk setup? i cant get calls in or out of my sip provider, internal calls work tho :/ i am not sure what i am doing wrong, tried several guides for my sip provider where people said they have it working |
| 00:05.40 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 00:09.02 | `md | i am getting SIP/2.0 603 Declined when i try an outgoing call for example |
| 00:09.17 | *** join/#asterisk gravin (~gravin@57.70.50.60.brf01-home.tm.net.my) |
| 00:10.23 | `md | also it seems asterisk tries to map the external number i am calling to itself instead of to my sip provider |
| 00:10.42 | `md | i am really at a loss here |
| 00:11.02 | p3nguin | Pastebin your sip.conf and extensions.conf. |
| 00:11.03 | p3nguin | ~pb |
| 00:11.04 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 00:11.14 | `md | ok |
| 00:12.06 | p3nguin | Don't forget to hide your passwords. |
| 00:12.38 | *** join/#asterisk Korolev (~Korolev@nmd.sbx13940.miamifl.wayport.net) |
| 00:13.46 | `md | http://pastebin.com/p2pnwqFY sip.conf |
| 00:14.32 | `md | http://pastebin.com/VPA4ihh8 extensions.conf |
| 00:16.14 | `md | btw it registers just fine with the sip provider |
| 00:16.21 | *** join/#asterisk k-man (~k-man@unaffiliated/k-man) |
| 00:16.35 | `md | sip show registry shows all 3 as registered |
| 00:16.58 | k-man | any idea what the differences between cisco 7940, 7941 and 7942 phones are? |
| 00:17.37 | p3nguin | First thing, you don't need an inbound and outbound entry. One entry of type peer is good enough. |
| 00:17.44 | `md | ok |
| 00:18.20 | p3nguin | The extension in the incoming context is wrong. |
| 00:18.31 | `md | ok |
| 00:18.33 | p3nguin | You have: exten => 051198xxxx,1,Dial,SIP/11,30,r |
| 00:18.38 | `md | yes |
| 00:18.49 | p3nguin | It should be: exten => 051198xxxx,1,Dial(SIP/11,30) |
| 00:18.56 | `md | oh! |
| 00:19.26 | p3nguin | Line number 19, you have: exten => _0.,2,Dial(SIP/${EXTEN}@t-online,30,tr) |
| 00:20.02 | p3nguin | It should be: exten => _0.,n,Dial(SIP/t-online/${EXTEN},30) |
| 00:20.07 | p3nguin | Wait, no... |
| 00:20.11 | p3nguin | It should be: exten => _0.,n,Dial(SIP/t-online/${EXTEN}) |
| 00:20.12 | p3nguin | no timeout. |
| 00:20.16 | *** join/#asterisk rdegges (xeXsTIKg0a@208.100.5.111) |
| 00:20.22 | p3nguin | Let them deal with the timeout. |
| 00:20.30 | rdegges | Question: what's the maximum and minimum values you can use in the ``VOLUME`` function? |
| 00:20.52 | p3nguin | Lines 7, 11, and 15... take off the r in the dial options. |
| 00:21.22 | p3nguin | Line 18, you have: exten => _0.,1,SetCallerID(051198xxxx) |
| 00:21.40 | p3nguin | It should be: exten => _0.,1,Set(CALLERID(num)=051198xxxx) |
| 00:22.04 | `md | ah :D |
| 00:22.09 | carrar | Nigel says to put the volumn at 11 |
| 00:22.23 | p3nguin | Actually, just let me fix everything. You have more things wrong. |
| 00:22.37 | carrar | It's ONE LOUDER |
| 00:23.40 | `md | oh yea thats already better! |
| 00:23.48 | `md | now it at least tries to use the sip provider |
| 00:24.33 | `md | but i am getting an error 400 back, it complains about RFC3261 |
| 00:24.45 | `md | the error codes are in german :/ |
| 00:25.15 | p3nguin | http://pastebin.com/BTGZf4V8 |
| 00:25.20 | p3nguin | There. Much better. |
| 00:26.48 | `md | ok |
| 00:26.53 | `md | let me reload that |
| 00:27.47 | p3nguin | And that's just patching up yours. If you had me set it up, it would be different still. |
| 00:29.24 | p3nguin | Your sip.conf could use a couple changes, too. |
| 00:29.38 | `md | yea i am sure it could |
| 00:29.47 | p3nguin | t-online has nat=yes, but they are not behind NAT. Should be nat=no. |
| 00:30.13 | p3nguin | insecure=very is no longer valid. Should be insecure=port,invite if you even need it at all. |
| 00:30.32 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 00:30.47 | p3nguin | dtmfmode=info ? Why info? rfc2833 is preferred. |
| 00:31.04 | `md | i read it on forums that i might need it for my provider, not too sure about it tho |
| 00:31.50 | p3nguin | There's a chance that you will need it. It depends on how they have things set up. If you do need it, it's "port,invite", not "very". |
| 00:31.55 | WIMPy | p3nguin: linux-stable has been 3.2.0-r1 for five weeks. |
| 00:32.02 | p3nguin | *gasp* |
| 00:32.17 | `md | ok |
| 00:32.23 | p3nguin | I ran Linux 2.6 until yesterday. |
| 00:33.17 | WIMPy | I just learned from sruffell that you need at least 2.6.39 if you want IRQ time accounting. |
| 00:33.41 | WIMPy | Without it you don't see why dahdi with SWEC fails. |
| 00:34.36 | WIMPy | I tried the same configuration with misdn today. It didn't sound any better but unlike dahdi it was able to keep the link stable. |
| 00:37.37 | *** join/#asterisk nighty- (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 00:38.09 | `md | holy shit |
| 00:38.13 | `md | outgoing works now :DDDD |
| 00:39.35 | `md | incoming still wont work tho |
| 00:40.11 | `md | i see the call incoming in the asterisk console |
| 00:40.22 | `md | but extension 11 isnt ringing and i get a busy tone |
| 00:40.38 | p3nguin | Of course not. Extensions don't ring. Phones ring, though. |
| 00:40.44 | `md | uhm yea |
| 00:40.49 | `md | thats what i meant |
| 00:41.09 | p3nguin | Have you fixed the two peer entry thing in sip.conf yet? |
| 00:41.15 | `md | yes |
| 00:41.19 | `md | there's just one now |
| 00:41.27 | p3nguin | Is the context still set to incoming? |
| 00:41.30 | `md | yes |
| 00:42.05 | p3nguin | You see the call coming in, so what extension is it going to? |
| 00:43.09 | *** join/#asterisk gravin (~gravin@57.70.50.60.brf01-home.tm.net.my) |
| 00:43.14 | p3nguin | You have only one extension in the incoming context, but you have three phone numbers. |
| 00:43.18 | p3nguin | at least three, anyway. |
| 00:43.42 | `md | i am calling the number that i have specified in incoming |
| 00:43.48 | p3nguin | If you didn't filter out something I didn't ask you to filter out, I would have fixed that part for you as well. |
| 00:44.19 | `md | like my phone numbers? |
| 00:44.48 | p3nguin | I don't really care what your phone numbers are, but you asked for help. Help fixing phone calls requires knowing what extensions calls are going to. |
| 00:44.52 | p3nguin | I'll have to see the evidence of what's going on. |
| 00:44.52 | `md | i see something like this == Spawn extension (incoming, 051198xxxx, 1) exited non-zero on 'SIP/t-online-00000012' |
| 00:45.23 | p3nguin | I'll need to see the sip debug with core verbose included. |
| 00:45.29 | p3nguin | core set verbose 3 |
| 00:45.33 | `md | i have that |
| 00:45.33 | p3nguin | sip set debug on |
| 00:45.38 | `md | i have that too |
| 00:45.48 | p3nguin | Make a call. Pastebin it. Don't filter, or I can't (and won't try) to help you. |
| 00:45.49 | `md | should i paste the output |
| 00:45.54 | `md | ok |
| 00:46.49 | *** join/#asterisk sam555 (~sysadmin@udp124488uds.hawaiiantel.net) |
| 00:46.56 | sam555 | hello all! |
| 00:47.23 | sam555 | does anyone know what these are called? http://i42.tinypic.com/azfift.jpg |
| 00:47.33 | sam555 | I'm trying to order them and they are not called phone crimps |
| 00:47.51 | p3nguin | splice connectors |
| 00:47.58 | sam555 | p3nguin: thanks!!! |
| 00:48.45 | `md | p3nguin: http://pastebin.com/Xru685p9 |
| 00:49.31 | WIMPy | sam555: 3M scotchlock |
| 00:50.13 | p3nguin | I'm not seeing the verbose output in this debug. |
| 00:50.29 | `md | Verbosity is at least 3 |
| 00:50.29 | `md | Core debug is at least 1 |
| 00:50.37 | `md | didnt you say 3? |
| 00:50.58 | p3nguin | I need at least 3. |
| 00:51.07 | p3nguin | I don't see it, though. |
| 00:51.17 | p3nguin | If you can see it in that debug, please point it out to me. |
| 00:51.45 | `md | should i do it again with 4? 5? |
| 00:52.28 | p3nguin | Go ahead and turn off sip debug for a minute. Increase verbose to at least 3. Make a call. |
| 00:53.42 | `md | ok |
| 00:54.04 | `md | p3nguin: http://pastebin.com/7DYPPk7q |
| 00:54.08 | `md | thats with 5 and sip debug off |
| 00:54.47 | p3nguin | ooooookay... so there's no call happening. |
| 00:54.48 | p3nguin | How odd. |
| 00:55.03 | `md | yes |
| 00:55.04 | `md | odd |
| 00:55.12 | WIMPy | agrees. |
| 00:55.27 | p3nguin | Are you sure you have that extension in the incoming context? I didn't see it. |
| 00:55.33 | WIMPy | A call ending without an error or something happening looks wrong. |
| 00:55.39 | `md | [incoming] |
| 00:55.39 | `md | exten => 051198436418,1,Dial,(SIP/11,30) |
| 00:55.41 | `md | is what i have |
| 00:55.59 | p3nguin | Looks the same to me. |
| 00:56.04 | `md | to me as well |
| 00:56.20 | p3nguin | You remembered to "sip reload" and "dialplan reload" after making the changes? |
| 00:56.47 | `md | i did a core reload, and i think i even completely restarted it, but let me make sure |
| 00:57.02 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 00:57.58 | `md | ok i completely shut down asterisk and started it again |
| 00:58.06 | `md | and i get exactly the same behaviour as before |
| 00:59.01 | p3nguin | I honestly don't get it. How can a call to an extension exit when there was no call to an extension showing up before it? |
| 00:59.16 | `md | that's exactly what puzzles me as well |
| 00:59.26 | `md | is there anything else anywhere that can influence this? |
| 01:00.34 | WIMPy | I'd say this shouldn't be possible. |
| 01:00.47 | WIMPy | What Asterisk version are you running? |
| 01:00.55 | `md | Asterisk 1.8.7.1~dfsg-3 |
| 01:00.59 | p3nguin | Maybe the phone that is trying to be called isn't available or something. Add a Verbose() before your Dial(). |
| 01:01.27 | p3nguin | Something like Verbose(The call made it here) |
| 01:01.31 | WIMPy | Hmmm. Dies that version perhaps include the remote console loglevel changes that were bad? |
| 01:01.51 | `md | hmmmm |
| 01:01.54 | `md | i think i have it now |
| 01:01.59 | `md | there was a typo |
| 01:02.01 | `md | how strange |
| 01:02.07 | p3nguin | Where was the typo? |
| 01:02.14 | p3nguin | Oh... |
| 01:02.19 | `md | there was a , between Dial and () |
| 01:02.19 | p3nguin | Dial,(SIP/11,30) |
| 01:02.21 | p3nguin | I see. |
| 01:02.22 | `md | yes |
| 01:02.34 | `md | now it works :) |
| 01:02.34 | p3nguin | So the extension died before it could run. |
| 01:02.38 | `md | indeed |
| 01:02.48 | WIMPy | Ok, so now we know what that means. |
| 01:02.49 | p3nguin | I would have expected more info. |
| 01:03.23 | `md | does asterisk have the concept of "groups"? |
| 01:03.32 | p3nguin | Maybe. What kind of group? |
| 01:03.36 | WIMPy | Wht kind of groups? |
| 01:03.50 | `md | a group that includes extensions, and where i can route external numbers to |
| 01:04.08 | p3nguin | I don't understand. |
| 01:04.10 | WIMPy | You can dial as may peers as you like in a single Dial() |
| 01:04.14 | `md | aha |
| 01:04.20 | p3nguin | Are you talking about dialing phones? |
| 01:04.23 | p3nguin | Dial(SIP/10&SIP/11&SIP/12,30)? |
| 01:04.32 | `md | well that would work |
| 01:04.44 | *** join/#asterisk Micc_ (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 01:05.02 | p3nguin | It works, but I don't know if that's what you're asking or not. I didn't understand your question. |
| 01:05.10 | *** part/#asterisk sam555 (~sysadmin@udp124488uds.hawaiiantel.net) |
| 01:05.13 | `md | well i dont usually work with asterisk, but with alcatel-lucent... there you can define groups and put extensions inside, and have the group number in the external numbering plan |
| 01:05.39 | `md | you call one external number, and several phones ring |
| 01:05.40 | p3nguin | Asterisk doesn't do things like that. If you want that, you'll write it yourself. |
| 01:05.50 | p3nguin | Dial(SIP/10&SIP/11&SIP/12,30) <---- this |
| 01:05.52 | `md | yea well that is what i was asking :D |
| 01:05.56 | p3nguin | One extension. Three phones. |
| 01:05.57 | `md | so asterisk doesnt have that, ok |
| 01:06.00 | WIMPy | In asterisk an "extension" is a target in the dialplan. |
| 01:06.11 | `md | ah |
| 01:06.18 | WIMPy | And that can do or call whatever you want. |
| 01:06.36 | `md | that sounds interesting |
| 01:06.41 | `md | yea i can see how you mean |
| 01:06.42 | `md | yea |
| 01:06.48 | `md | that's actually pretty neat |
| 01:06.48 | `md | :D |
| 01:06.53 | p3nguin | exten => 051198436418,1,Dial(SIP/10&SIP/11&SIP/12,30) |
| 01:07.07 | WIMPy | It's flexible, but not neccessarily convenient. |
| 01:07.13 | p3nguin | Extension 051198436418 dials SIP/10, SIP/11, and SIP/12. |
| 01:07.19 | p3nguin | (1905.56) <p3nguin> One extension. Three phones. |
| 01:07.54 | p3nguin | You can also set a global which contains all the phones, and then Dial() the global variable. |
| 01:08.15 | p3nguin | In globals, ALL_PHONES=SIP/10&SIP/11&SIP/12 |
| 01:08.32 | p3nguin | then exten => 051198436418,1,Dial(${ALL_PHONES},30) |
| 01:08.39 | `md | or i could do this? |
| 01:08.40 | `md | exten => 20,1,Dial(SIP/10&SIP/11SIP/12,30) |
| 01:08.40 | `md | exten => 20,n,Hangup |
| 01:08.47 | p3nguin | Sure. |
| 01:08.50 | `md | cool |
| 01:08.55 | p3nguin | Extension 20 would dial the three phones. |
| 01:09.00 | `md | exactly |
| 01:09.06 | `md | and i can chain that right |
| 01:09.14 | p3nguin | As in...? |
| 01:09.22 | `md | so when i call my external number i can have it call 20, and that calls all the phones |
| 01:09.35 | p3nguin | Yes, using a Local channel. |
| 01:09.36 | `md | exten => 051198436418,1,Dial(SIP/20,30) |
| 01:09.38 | `md | like so |
| 01:09.39 | p3nguin | no |
| 01:09.43 | `md | no? |
| 01:09.43 | WIMPy | Yes, but that's probably not what you want. |
| 01:09.44 | `md | ok |
| 01:09.52 | WIMPy | no |
| 01:09.54 | p3nguin | Dial(Local/20@whatever-context) |
| 01:10.16 | p3nguin | extension 20 has nothing to do with SIP |
| 01:10.20 | `md | ah it's Local not SIP |
| 01:10.21 | WIMPy | There is no link between extensions and devices except for Dial commands in you dialplan. |
| 01:10.25 | `md | of course |
| 01:10.26 | p3nguin | SIP/20 is a phone, named 20. |
| 01:10.31 | `md | it's not a sip peer of course :D |
| 01:10.52 | p3nguin | Local channels are wonderful. They turn points in your dialplan into devices. |
| 01:11.02 | WIMPy | But a simple Goto might be easier. |
| 01:11.47 | p3nguin | Yeah, might be better. |
| 01:11.56 | p3nguin | Goto(whatever-context,20,1) |
| 01:12.01 | p3nguin | rather than Dial() |
| 01:12.07 | `md | ah interesting |
| 01:12.27 | p3nguin | Then you aren't creating a new channel when you don't *really* need one. |
| 01:12.55 | p3nguin | The end result would be the same in both cases though -- the three phones would ring. |
| 01:13.02 | `md | indeed |
| 01:13.42 | WIMPy | So for a classic aproach, you could have a groups context defining extensions that call multiple devices and in your incomming context you goto the group extensions. |
| 01:14.14 | p3nguin | Your imagination is the limit. You create the concepts. |
| 01:14.48 | `md | yea i slowly start to understand just HOW powerful asterisk can be |
| 01:15.52 | `md | works pretty well |
| 01:31.59 | luckman212 | whoa ..... asterisk 10 is released! |
| 01:33.25 | *** join/#asterisk kotis__ (~kotis@72.253.138.39) |
| 01:43.10 | SeRi | Mhhhhhhh...... I might wait a few months before I put my hands on it |
| 01:44.55 | p3nguin | Wait until next year, at least. |
| 01:45.33 | WIMPy | Wil next year be a good Asterisk year? |
| 01:45.48 | p3nguin | Every year is a good Asterisk year. |
| 01:46.01 | SeRi | :) |
| 01:46.11 | `md | i wonder how well asterisk works as a sip trunk of another pbx |
| 01:46.19 | p3nguin | ~trunk |
| 01:46.19 | infobot | i guess trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 01:46.25 | `md | i shall try that soon |
| 01:47.06 | `md | alcatel calls it that :/ |
| 01:47.09 | p3nguin | But asterisk to asterisk using SIP works just fine. Asterisk to asterisk using IAX2 works pretty well, too. |
| 01:47.12 | WIMPy | In ethernet, LACP is also called trunk. |
| 01:47.26 | `md | i was thinking asterisk to alcatel |
| 01:47.48 | p3nguin | There's no SIP trunk, only SIP peers. |
| 01:48.09 | `md | noted |
| 01:48.16 | p3nguin | Asterisk doesn't really care what kind of device peers with it, as long as it understands SIP pretty much the same as Asterisk understands it. |
| 01:48.49 | `md | sounds like it could work then |
| 01:49.01 | p3nguin | If it does SIP, it will most likely work. |
| 01:50.16 | `md | it does sip |
| 01:56.57 | *** join/#asterisk francisvgarcia (~francisvg@186.1.68.198) |
| 02:04.49 | SeRi | p3nguin: |
| 02:05.09 | SeRi | I am bore |
| 02:05.14 | SeRi | I am bored* |
| 02:06.36 | SeRi | compiling some times gets boring.... |
| 02:07.08 | francisvgarcia | boring is when u got a phone working |
| 02:07.12 | WIMPy | Get a faster box. |
| 02:07.27 | francisvgarcia | and suddenly start to have one way audio |
| 02:07.33 | francisvgarcia | for no reason like me now |
| 02:07.39 | SeRi | WIMPy: buy me one :) I am still sitting on my dual core xtream from like 6yrs a go... LOL |
| 02:07.53 | SeRi | francisvgarcia: LOL! |
| 02:08.07 | WIMPy | My Asterisk test box is a dual PIII. |
| 02:09.00 | SeRi | my test box is an atom |
| 02:09.01 | francisvgarcia | My personal machine is a Intel Celeron :D |
| 02:09.27 | francisvgarcia | with 512 of RAM DDR1 533 |
| 02:09.41 | SeRi | my mom ask for christmas if I could buy her a new latop.... I felt bad for her.... she still runs on a PIII :( bad son |
| 02:09.42 | francisvgarcia | running ubuntu |
| 02:10.11 | SeRi | ops sorry she has a sempron |
| 02:10.14 | WIMPy | I still have an old c2d. Maybe I should get a board for it. |
| 02:10.20 | SeRi | no better though.... lol |
| 02:10.27 | SeRi | LOL |
| 02:14.24 | SeRi | any body care to be my monkey for a few days or maybe more? I am going to be building a modified version of astlinux..... If you want to test let me know. |
| 02:15.04 | SeRi | I am almost done compiling :) |
| 02:15.16 | WIMPy | I think "testing Asterisk distributions" was somewhere on my to do list. |
| 02:15.41 | SeRi | WIMPy: Ill take that as you want in :) |
| 02:16.00 | WIMPy | I guess it might be pretty interesting what the hardware support looks like in them. |
| 02:16.14 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 02:16.19 | WIMPy | Can I do a netinstall? |
| 02:16.36 | SeRi | Sure :) |
| 02:16.43 | SeRi | Ill just have to prep it for that |
| 02:17.06 | SeRi | you have a static IP? |
| 02:17.24 | WIMPy | That would certainly make it easier. I don't think I can moticat myself to carry media around. |
| 02:17.50 | WIMPy | Yes, but I was on about doing it via PXE. |
| 02:18.14 | SeRi | well I have to allow your IP in. |
| 02:18.46 | SeRi | msg me your ip. |
| 02:18.50 | WIMPy | The BIOS won't search the internet. |
| 02:18.58 | SeRi | oooo |
| 02:19.05 | SeRi | Indeed |
| 02:19.12 | WIMPy | It's about how I get it on to a blank PC. |
| 02:19.24 | SeRi | got it. ok ill pass you the files when done |
| 02:19.42 | SeRi | astlinux leaves on the ram. |
| 02:19.54 | SeRi | I am wondering if load it via pxe would work. |
| 02:19.58 | SeRi | we shall try |
| 02:20.20 | WIMPy | loves PXE. |
| 02:20.40 | WIMPy | No more CDs, DVDs or pendrives. |
| 02:20.42 | SeRi | :} |
| 02:23.32 | SeRi | New on this build: mpg123, wget, option for new or existing persistant storage, ipkg capabilities, encryption, psql, gtal, etc..... |
| 02:23.35 | *** join/#asterisk master_of_master (~master_of@p57B540FE.dip.t-dialin.net) |
| 02:24.04 | SeRi | Ill get a list put together |
| 02:24.50 | SeRi | I am going to try and make a full install option as well. |
| 02:25.03 | SeRi | which will inlcude more options |
| 02:25.11 | WIMPy | Is it a good idea to have mpg123 in a distribution? |
| 02:25.25 | SeRi | It depends |
| 02:25.31 | SeRi | how you use it I guess |
| 02:25.39 | WIMPy | Instead of what? Does it run "live"? |
| 02:26.09 | SeRi | what do you mean live? |
| 02:26.20 | WIMPy | Yes, it does. But if you include it I guess you have to inform the user. |
| 02:26.33 | SeRi | ooo yes. |
| 02:26.42 | WIMPy | What's the other option to "full install"? |
| 02:26.48 | SeRi | My idea behind it is for streaming audio options |
| 02:27.18 | SeRi | WIMPy: astlinux runs on memory. It does not install on your system is a live OS |
| 02:27.33 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:27.42 | SeRi | persistant storage is made as per the users option |
| 02:28.04 | WIMPy | ok. Should make it easy to try. |
| 02:28.27 | SeRi | but astl wont install on the persistant fs... It's only for storage of files and media |
| 02:29.35 | WIMPy | But a live system is hard to patch. |
| 02:29.54 | SeRi | for example. if you do persistant storage all the files you edit teh most will leave there |
| 02:30.06 | SeRi | Yes. Thats why I want to make an option |
| 02:30.22 | WIMPy | had to patch a lot so far. |
| 02:30.42 | WIMPy | Especially to get hardware working. |
| 02:32.09 | SeRi | Yea I saw your posts |
| 02:32.10 | SeRi | brb |
| 02:32.33 | WIMPy | Probably not all of them :-) |
| 02:33.10 | WIMPy | Unfortinately I even lost one of them. |
| 02:37.32 | p3nguin | As for the mpg123 thing, just make users of your ast distro agree to be bound by the same license that he would agree to to install mpg123 himself. Wouldn't that be sufficient? |
| 02:38.48 | WIMPy | Don't know what is says, but probably yes. |
| 02:40.49 | SeRi | p3nguin: I am looking in to it. |
| 02:44.58 | p3nguin | Optionally, you could have two branches: one with mpg123 with license terms, and one without mpg123 for those who don't want to agree. |
| 02:46.32 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 02:49.35 | *** join/#asterisk F2Knight (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 02:50.02 | F2Knight | Q: Just upgraded a box to 1.8.8.0 to check it out.. and got this error about NAT. |
| 02:50.33 | WIMPy | That has been discussed a lot the last days. |
| 02:50.38 | SeRi | p3nguin: I thin I like that better |
| 02:50.40 | F2Knight | http://pastebin.com/hGuHM5Ve |
| 02:50.59 | p3nguin | f2knight: That is not an error. |
| 02:51.09 | F2Knight | no its a warning.. |
| 02:51.12 | p3nguin | Right. |
| 02:51.23 | p3nguin | warning != error |
| 02:51.25 | F2Knight | but it takes for ever to load asterisk now. |
| 02:51.41 | p3nguin | Fix sip.conf according to the information provided in the warning. |
| 02:51.48 | F2Knight | and I am not sure what they are implying to fix. |
| 02:52.19 | WIMPy | Only one nat= line per asterisk. |
| 02:52.33 | p3nguin | It says do not set nat per peer. Set it in general only. |
| 02:53.19 | p3nguin | Basically, turn on nat for asterisk and every peer, or turn it off entirely. |
| 02:54.27 | p3nguin | This change is going to make me redo my templates for my phones. |
| 02:54.40 | F2Knight | okay so ... heres my question then.. |
| 02:54.47 | F2Knight | Not all device will be natted.. |
| 02:54.55 | F2Knight | by setting nat = yes in global. |
| 02:54.55 | p3nguin | Yep. |
| 02:55.05 | F2Knight | will that effect non nated clients? |
| 02:55.12 | p3nguin | It isn't supposed to. |
| 02:55.40 | F2Knight | I wonder what the devs are trying to do with this.. .where is the ... end goal so to speak |
| 02:55.51 | p3nguin | It's stated in the warning. |
| 02:56.51 | F2Knight | no I mean.. in the next few versions.. what is going to be the end result..? like is Nat going to be on by default only? no nat support? external nat control? |
| 02:57.03 | F2Knight | just trying to understand what the change was for |
| 02:57.05 | SeRi | p3nguin: Is there a way to know which number is been dialed FROM the DB... right now I get MYNUMBERHERE is calling s |
| 02:57.38 | p3nguin | I don't understand your question. The database doesn't dial numbers. |
| 02:57.52 | SeRi | ok one sec... |
| 02:59.08 | SeRi | http://pastebin.com/9sxymrWX |
| 03:00.41 | SeRi | when I dial 1101 I see in the cli MYNUMBERHERE is calling s |
| 03:00.50 | WIMPy | Looks like I have a session-timer issue with an SPA962. Is there a solution other than disabling them on Asterisk? |
| 03:01.50 | p3nguin | seri: That's what line 5 says to show you. What more do you want? |
| 03:02.02 | p3nguin | I don't understand the point of this macro in the first place. |
| 03:02.16 | SeRi | Is there a way to make it show the number instead of "s" |
| 03:02.25 | SeRi | Is a speed dial |
| 03:02.32 | SeRi | I show this before |
| 03:02.39 | SeRi | and you said it was welld one |
| 03:02.40 | p3nguin | Yes, there is. Don't use a macro when you don't need one. |
| 03:02.53 | SeRi | :( |
| 03:03.06 | SeRi | ok so how can I convert it? |
| 03:04.21 | SeRi | p3nguin: http://pastebin.com/NVifEHJs |
| 03:04.25 | p3nguin | What is your exact goal? Just have speed dials in astdb? E.g., speed dial 10 = 3145551212 |
| 03:05.48 | SeRi | /sysdial/04 :numberhere |
| 03:05.54 | SeRi | <PROTECTED> |
| 03:06.05 | p3nguin | Why sysdial? Why not speeddial? |
| 03:06.25 | F2Knight | sip shois there a way of setting the make menuselect options with out running make menuselect? for scripted builds |
| 03:06.29 | WIMPy | Getting 481 Call Leg/Transaction Does Not Exist from the SPA. |
| 03:06.32 | SeRi | Is an astlinux thing I guess... I do use the webui to create db's |
| 03:06.40 | SeRi | its simpler |
| 03:07.00 | p3nguin | because database put speeddial/04 5551212 is hard? |
| 03:07.15 | p3nguin | At any rate, you don |
| 03:07.19 | p3nguin | 't need a macro. |
| 03:07.36 | p3nguin | It's because of the macro that your extension is s. |
| 03:07.59 | SeRi | p3nguin: I dont have access to cli all the time. webui simplifies this for me. |
| 03:08.24 | p3nguin | Give me an example of how you would dial a speeddial. |
| 03:08.50 | SeRi | 1101 |
| 03:09.18 | p3nguin | Is that considered speeddial number 1? |
| 03:09.36 | p3nguin | So your speed dial prefix is 110? |
| 03:09.55 | p3nguin | Or it is speed dial 01, and your speed dial prefix is 11? |
| 03:10.21 | SeRi | is o1 and prefix is 11 |
| 03:10.25 | SeRi | 01* |
| 03:10.37 | p3nguin | okay, one moment |
| 03:10.42 | SeRi | yes sr. |
| 03:13.50 | Netgeeks | if I'm getting zttest results of -29193%, what does that mean? I've never seen negative percentages as a result of zttest |
| 03:14.41 | WIMPy | Your timing is screwed? |
| 03:14.58 | Netgeeks | yeah, horribly screwed... |
| 03:15.16 | Netgeeks | maybe a hardware problem? ztdummy is the timer |
| 03:16.07 | WIMPy | But you are aware that zaptel was discontinued 3 years aog? |
| 03:16.17 | WIMPy | Or is it already 4? |
| 03:16.36 | SeRi | 4* almost there :P |
| 03:16.53 | Netgeeks | *shrug* I'm not using it, it's someone calling for help, and he's stuck with asterisk 1.4 and zaptel |
| 03:17.08 | SeRi | tell him to upgrade :P |
| 03:17.16 | Netgeeks | I did |
| 03:17.17 | Netgeeks | lol |
| 03:17.21 | SeRi | hehe |
| 03:17.37 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:17.39 | Netgeeks | he asked his boss, and his boss said no..... |
| 03:17.48 | Netgeeks | I'm going to tell him to get different hardware |
| 03:19.34 | p3nguin | seri: http://pastebin.com/yZKjDpf8 |
| 03:21.44 | SeRi | is on monkey mode! testing power up! |
| 03:25.37 | WIMPy | Didi you already break it? |
| 03:25.41 | WIMPy | -i |
| 03:28.56 | p3nguin | seri: I have a typo... |
| 03:29.04 | SeRi | I know |
| 03:29.06 | p3nguin | line 2, I missed the n, |
| 03:29.07 | SeRi | is been fixed |
| 03:29.11 | SeRi | :P |
| 03:29.13 | *** join/#asterisk radic (~radic@dslb-094-216-240-242.pools.arcor-ip.net) |
| 03:29.21 | p3nguin | It's hard for me to write dial plan IN the pastebin. |
| 03:29.37 | p3nguin | I have a much easier time writing it in vim and then pastebinning it. |
| 03:29.58 | p3nguin | I miss simple errors such as that. |
| 03:30.15 | WIMPy | Dou you have a dialplan syntax coloring mode? |
| 03:30.45 | SeRi | WIMPy: that would be nice |
| 03:30.52 | SeRi | good idea |
| 03:30.58 | SeRi | p3nguin: Thanks. It works. |
| 03:31.32 | p3nguin | I don't in the pastebin, but I do in vim. |
| 03:31.45 | SeRi | p3nguin: want to share it? |
| 03:31.46 | WIMPy | Nice |
| 03:31.57 | p3nguin | It's built in. :syntax on |
| 03:32.11 | SeRi | ooo I thought you had your own colors |
| 03:32.19 | p3nguin | Don't need my own. |
| 03:32.20 | SeRi | I know about color on and off |
| 03:32.25 | p3nguin | The default ones work just fine. |
| 03:32.30 | SeRi | ah |
| 03:33.58 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:33.59 | WIMPy | Freaky |
| 03:34.04 | SeRi | lol |
| 03:34.31 | *** join/#asterisk newsham (~chat@udp216902uds.hawaiiantel.net) |
| 03:34.43 | WIMPy | But it doesn't color same, switch or lswitch. |
| 03:35.12 | newsham | hi. I have sipdroid on my android phone. I have asterisk configured and running. I have it configured to do incoming and outgoing calls through google voice. mostlyworks.. however I tried to reconfigure it today to send incoming calls to my sipdroid and its not working. |
| 03:35.23 | newsham | although if i change it to dial a sip soft phone instead, it works! |
| 03:35.36 | newsham | only difference is the name "SIP/twinkle1" vs. "SIP/droid" for my diff sip accounts |
| 03:35.40 | newsham | any ideas? |
| 03:36.18 | p3nguin | Pastebin both peer entries. |
| 03:36.46 | newsham | they're both "[twinkle1](office-phone)\ncallerid=..." |
| 03:36.48 | newsham | using the same template |
| 03:36.50 | p3nguin | Also pastebin the extension used where you are changing the phone being dialed. |
| 03:37.24 | newsham | the extensions plan is "Anwer();Wait(2);SendDTMF(1);Dial(SIP/droid,20)" |
| 03:37.30 | newsham | or "SIP/twinkle1" |
| 03:37.36 | SeRi | ~pb |
| 03:37.36 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 03:37.42 | newsham | yah i know pastebin |
| 03:37.49 | SeRi | use it :) |
| 03:37.58 | newsham | i already gave you all the info in two lines on irc. |
| 03:38.01 | newsham | pastebin seems overkill |
| 03:38.11 | p3nguin | ~everything |
| 03:38.11 | infobot | If you actually know everything there is to know about this topic, you wouldn't be here asking *us* for help. Just give us the information we're asking for, and we'll do our best to help you solve the problem. Resistance is futile. And it just pisses people off. |
| 03:38.21 | SeRi | p3nguin: good luck on this one ;) |
| 03:38.24 | newsham | i didnt say i know everything |
| 03:38.33 | newsham | i said i gave you all the info |
| 03:38.43 | newsham | literally the only diff in these two entries is s/droid/twinkle1" |
| 03:39.13 | SeRi | sip debug? core verbose? nat? over cell? over wifi? |
| 03:39.17 | SeRi | :/ |
| 03:39.35 | newsham | fwiw, i can call the SIP/droid from my other handsets. |
| 03:39.47 | newsham | its all over wifi. i had debugging on and didnt notice anything.. lemme try again |
| 03:40.03 | p3nguin | I guess twinkle1 isn't online or something. |
| 03:40.09 | SeRi | pb your sip debug and your core verbose |
| 03:40.15 | newsham | both twinkle1 and droid are online. i can call both from a third handset. |
| 03:40.32 | newsham | jsut from google voice i can only call one :( |
| 03:41.46 | SeRi | you need ubuntu |
| 03:41.51 | p3nguin | haha |
| 03:43.31 | newsham | funny.. asterisk isnt running on ubuntu but i'm typing from ubuntu now |
| 03:44.27 | SeRi | figures |
| 03:47.15 | newsham | http://pastebin.com/jr3RECDX |
| 03:47.42 | newsham | i see a SIP invite, but i get no ringing. |
| 03:48.37 | newsham | and like i said, changing to another siphone extension (which is configured nearly identically) works perfectly :( |
| 03:49.19 | newsham | and calls between my handsets all work fine |
| 03:50.25 | SeRi | p3nguin: I won the 560 for 125.00 dollars |
| 03:50.38 | p3nguin | Cool. How much is shipping? |
| 03:50.42 | SeRi | free |
| 03:50.52 | p3nguin | Even better. |
| 03:50.56 | SeRi | +1 |
| 03:51.07 | SeRi | I might pay for next day air |
| 03:51.09 | SeRi | :P |
| 03:51.23 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 03:52.24 | SeRi | [TK]D-Fender: I got the 560. :D |
| 03:52.39 | SeRi | I am going to annouce it to the world! |
| 03:52.44 | SeRi | lol |
| 03:53.41 | newsham | well thanks for the help gents. |
| 03:53.44 | *** part/#asterisk newsham (~chat@udp216902uds.hawaiiantel.net) |
| 03:55.00 | SeRi | he left :/ |
| 03:55.10 | SeRi | just when I was going to help him |
| 03:55.12 | p3nguin | Remember, resistance is futile... and it just pisses people off. |
| 03:55.18 | SeRi | lol |
| 03:55.39 | SeRi | +1 |
| 03:59.00 | *** join/#asterisk TheCops (~mdb@72.55.132.180) |
| 04:11.14 | *** join/#asterisk Micc_ (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 04:18.49 | *** join/#asterisk ks3 (~ks3@cpe-184-57-153-87.cinci.res.rr.com) |
| 04:20.35 | SeRi | https://www.eff.org/deeplinks/2011/12/internet-inventors-warn-against-sopa-and-pipa |
| 04:21.40 | SeRi | took a little over 3hrs to build it all. nice |
| 04:25.36 | *** join/#asterisk timeshell (~timeshell@206.248.136.108) |
| 04:30.00 | *** join/#asterisk gravin (~gravin@57.70.50.60.brf01-home.tm.net.my) |
| 04:32.50 | *** join/#asterisk timeshell (~timeshell@206.248.136.108) |
| 04:37.30 | Kobaz | http://www.kobaz.net/misc/cell_uploads/2011-12-15_23-34-45_674.jpeg |
| 04:37.33 | Kobaz | my new office setup |
| 04:38.36 | SeRi | a working office |
| 04:38.50 | Kobaz | yeap yeap |
| 04:39.12 | SeRi | http://www.dslreports.com/forum/r26216477-Home-Office <----- old pics of my office |
| 04:39.25 | SeRi | It looks different now.... :) |
| 04:39.37 | Kobaz | just got the quad head working properly just a bit ago |
| 04:39.47 | SeRi | nice |
| 04:39.47 | Kobaz | i had three monitors up but adding the third would break x |
| 04:40.28 | Kobaz | er i mean, adding the fourth |
| 04:44.00 | WIMPy | only has a really old pic of times whe monitors were big and the desk a lot tidier. http://wimpy.yeti.dk/p/ww |
| 04:44.10 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 04:44.14 | WIMPy | +n |
| 04:45.02 | WIMPy | Even with a Mac keyboard... |
| 04:45.23 | SeRi | :) |
| 04:47.40 | WIMPy | Oh, and the good old ZyXELs. |
| 04:48.21 | Kobaz | woah crts |
| 04:49.04 | Kobaz | SeRi: your office is way too clean |
| 04:49.19 | WIMPy | Indeed :-) |
| 04:49.48 | SeRi | I know. blame that on my ocd |
| 04:49.54 | SeRi | :( |
| 04:56.24 | p3nguin | I bought another D-Link product today. |
| 04:56.37 | Kobaz | any good? |
| 04:56.51 | p3nguin | $20 for a b/g/n router. |
| 04:56.57 | SeRi | p3nguin: ebay? |
| 04:57.07 | p3nguin | I have no desire to use it as a router... I'll connect it to my switch and use it as an AP. |
| 04:57.11 | p3nguin | Newegg.. |
| 04:57.34 | SeRi | I use a EAP-3660 as AP |
| 04:57.40 | p3nguin | If you're interested, I'll get the link and give you the promo code. |
| 04:57.47 | SeRi | yes |
| 04:58.07 | p3nguin | http://www.newegg.com/Product/Product.aspx?Item=N82E16833127288 |
| 04:58.19 | p3nguin | Actually, if you are logged in, the code will show on that page. |
| 04:59.06 | p3nguin | I figured for $20, I should give it a try. |
| 04:59.17 | Kobaz | i got a trendnet a/b/g/n router black friday special from tigerdirect for $15 |
| 04:59.18 | p3nguin | I needed another AP. |
| 04:59.35 | p3nguin | That would have been even better, but I didn't know about it. |
| 05:00.04 | p3nguin | This is an impulse buy. |
| 05:00.25 | Kobaz | yeah that was too |
| 05:00.29 | p3nguin | s/ i/ wa/ |
| 05:00.50 | Kobaz | it was so cheap i couldn't resist the deals |
| 05:01.11 | WIMPy | knows that problem. |
| 05:01.26 | Kobaz | should have bought two |
| 05:01.30 | p3nguin | I should have looked around on all the regular sites, but I didn't even bother. |
| 05:01.49 | SeRi | lol |
| 05:01.51 | SeRi | for 20 |
| 05:01.56 | SeRi | shit I am buying onw |
| 05:02.00 | p3nguin | I don't have any wireless A devices anyway. |
| 05:02.03 | SeRi | s/onw/one/ |
| 05:02.26 | p3nguin | I figured for $20, you're not going to get a b/g/n AP or bridge anywhere else. |
| 05:02.52 | Kobaz | i have the shoprunner thing for newegg too |
| 05:02.56 | p3nguin | Or if you can, it won't be any better or any worse than this one. |
| 05:03.00 | Kobaz | figured might as well, i order enough stuff |
| 05:03.53 | p3nguin | What's the benefit of it? |
| 05:03.54 | SeRi | p3nguin: http://infodepot.wikia.com/wiki/D-Link_DIR-601_vA1 |
| 05:04.01 | Kobaz | free two day shipping |
| 05:04.05 | Kobaz | it's like amazon prime |
| 05:04.09 | SeRi | like that amazon stuff |
| 05:04.12 | SeRi | yea |
| 05:04.13 | SeRi | <PROTECTED> |
| 05:04.33 | Kobaz | especially on little things the shipping kills |
| 05:04.57 | Kobaz | it's like you want some moxel power splitters, they are like $1.99 each and $1.99 shipping each |
| 05:05.00 | Kobaz | or something stupid |
| 05:05.08 | Kobaz | er. molex |
| 05:07.11 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
| 05:08.01 | p3nguin | I think I might upgrade the 601 to use a removable antenna. Then I can connect my yagi to it. |
| 05:08.23 | p3nguin | As long as I can get the right RP-TNC jack for it, I'll do it. |
| 05:09.21 | p3nguin | It has an Atheros radio? That's useful. |
| 05:09.33 | p3nguin | Did not know that when I bought it. |
| 05:09.49 | SeRi | :) |
| 05:09.52 | p3nguin | I had no intention of modifying it, but now I probably will. |
| 05:09.59 | SeRi | +1! |
| 05:10.11 | SeRi | nothing I own is not modified |
| 05:10.20 | p3nguin | heh |
| 05:13.20 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 05:13.48 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 05:17.49 | SeRi | was that even right? |
| 05:17.52 | SeRi | lol |
| 05:18.45 | p3nguin | *shrug* |
| 05:20.24 | SeRi | lol. well the first img went slpendid |
| 05:23.50 | SeRi | I am happy again. now working on a second one and from there ill test and pass it to WIMPy I have 3. with mpg123 + modifications/no mpg123 + modifications/vanilla |
| 05:24.40 | WIMPy | What modifications? |
| 05:25.04 | SeRi | ill post them as soon as I am done. most are new binary's and addons |
| 05:25.41 | SeRi | p3nguin: you will be around? |
| 05:26.56 | p3nguin | When? |
| 05:27.35 | p3nguin | Tonight? |
| 05:27.45 | p3nguin | I'll be here until at least 2AM. |
| 05:28.15 | SeRi | yes. can I test call your conf? |
| 05:28.30 | p3nguin | As I said before, unless it is offline, you can call it. |
| 05:28.37 | SeRi | o cool. thanks |
| 05:29.13 | p3nguin | Don't expect anyone to be on it, though, unless you recruit people to conf, or unless you let me know so I can join it. |
| 05:29.22 | p3nguin | But if you want to listen to the music, it's always on. |
| 05:30.37 | p3nguin | I just charged my phone, too, so maybe it won't go dead mid-sentence. |
| 05:31.20 | SeRi | cool |
| 05:31.22 | SeRi | Thanks |
| 05:31.33 | SeRi | I have about 1 more hour to go before I can test |
| 05:32.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 05:34.13 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:38.18 | ponyofdeath | hi, guys is tehre an good wiki or guid on how to secure asterisk? currently i dont have port 5060 open to the outside but would like to do so? |
| 05:39.01 | SeRi | ~book |
| 05:39.01 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 05:39.38 | p3nguin | Do you need a bunch of random people calling you via SIP? |
| 05:40.44 | ponyofdeath | p3nguin: no but i want to have my phone be able to use my sip server |
| 05:41.02 | p3nguin | Then adjust your firewall accordingly. |
| 05:41.10 | ponyofdeath | p3nguin: so what ur saying is that there is no need to open it up |
| 05:41.51 | p3nguin | You are going to use an ITSP, right? |
| 05:42.06 | ponyofdeath | dont know what that is so no |
| 05:42.10 | p3nguin | ~itsp |
| 05:42.11 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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. |
| 05:43.52 | kaldemar | ponyofdeath: http://svn.digium.com/svn/asterisk/tags/10.0.0-rc3/README-SERIOUSLY.bestpractices.txt |
| 05:47.38 | ponyofdeath | kaldemar: cool ty |
| 05:54.35 | SeRi | people is so missinformed ont voip :( |
| 05:54.46 | SeRi | s/ont/on/ |
| 05:55.14 | WIMPy | Yes. |
| 05:55.49 | SeRi | some guy was arguing with me that voip can not provide better quality than gsm (Cell) |
| 05:56.20 | SeRi | sad.... |
| 05:56.46 | p3nguin | That's too much of a generalization. |
| 05:57.18 | p3nguin | "VoIP" covers a broad spectrum of qualities. |
| 05:57.33 | SeRi | in many ways voip can be better than gsm. |
| 05:57.39 | WIMPy | Usually negative ones. |
| 05:57.48 | SeRi | they guys had no clue about codecs in voip |
| 05:58.09 | WIMPy | Yes, but the better codec is about where it ends as well. |
| 05:58.55 | SeRi | WIMPy: what do you mean? If I make a call using g722 point to point thats all ready better |
| 05:59.26 | WIMPy | Yes. |
| 05:59.31 | WIMPy | But that's about the only thing that will be better. |
| 05:59.48 | SeRi | well we are talking voice quality. |
| 06:00.11 | WIMPy | That's not what you said initially. |
| 06:00.14 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-wdrskypbelkzdhjb) |
| 06:00.56 | SeRi | some guy was arguing with me that voip can not provide better quality than gsm (Cell) <------ gsm is a codec and we are talking quality |
| 06:01.08 | WIMPy | And even then: GSM has (practically) no jitter. |
| 06:01.25 | SeRi | true. |
| 06:01.41 | WIMPy | The way you wrote it it looks like you're talking about GSM, not one of the codecs made for that. |
| 06:02.05 | SeRi | I am not talking gsm bands |
| 06:02.15 | SeRi | I am talking gsm codec |
| 06:02.32 | kaldemar | i.e. AMR |
| 06:02.36 | WIMPy | Which one? |
| 06:03.13 | WIMPy | There are at least three associated with the name GSM and indeed, AMR would also be an option. |
| 06:03.37 | SeRi | That detail I did not ask. |
| 06:04.24 | WIMPy | The one you find in Asterisk and voip phones has been obsolete for like ten years. |
| 06:04.34 | SeRi | lol yes |
| 06:04.46 | *** join/#asterisk Sean-Der (~Sean-Der@NW1-DSL-74-215-64-154.fuse.net) |
| 06:05.55 | SeRi | wow looks like this build will take longer.... :( |
| 06:07.45 | WIMPy | Ooops. looks like EFR is from 1995. |
| 06:07.56 | WIMPy | So 10 years was not quite exact. |
| 06:08.21 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 06:10.54 | SeRi | you know what sounds good right now? |
| 06:11.07 | SeRi | nachos. |
| 06:17.36 | p3nguin | Damn, I almost had that. |
| 06:17.56 | p3nguin | I just assembled a beef, cheddar, and horseraddish sandwich. |
| 06:19.11 | SeRi | jesus!!!!! relax! man that sounds soooooo goooood! |
| 06:19.12 | p3nguin | I decided not to have nachos because I had tacos for lunch. |
| 06:19.29 | SeRi | lol |
| 06:19.34 | SeRi | tacos are good |
| 06:30.51 | Sean-Der | Is there anyway to record a calls length, even if the server is only a trunk |
| 06:32.07 | p3nguin | I don't understand the second part of your sentence, but the answer is CDR. |
| 06:32.52 | p3nguin | And that was a good sandwich. |
| 06:33.37 | p3nguin | I think after I eat up all my beef, I should switch to ham and swiss. |
| 06:34.07 | SeRi | perfect... i like honey ham |
| 06:35.08 | p3nguin | I've got enough beef to last all next week. |
| 06:35.31 | SeRi | nice |
| 06:36.02 | SeRi | i am a carnivour |
| 06:36.08 | SeRi | salads are for rabits |
| 06:36.42 | p3nguin | If it oinks or moos, it probably tastes good. |
| 06:36.48 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 06:36.57 | SeRi | hahahaha! |
| 06:37.35 | WIMPy | What does the beef eater say to the vegetarian? |
| 06:37.46 | WIMPy | My food shits on yours. |
| 06:38.18 | SeRi | hahahahahaha |
| 06:39.27 | SeRi | I am not sure psql is going to fly on this build :( maybe in the generic i586 |
| 06:39.46 | SeRi | p3nguin: you are running psql on your atom? |
| 06:39.56 | p3nguin | I don't have an atom. |
| 06:40.14 | SeRi | oh.... sorry |
| 06:40.17 | SeRi | ? |
| 06:40.33 | SeRi | what are you running? |
| 06:40.59 | p3nguin | VIA Nehemiah CentaurHauls |
| 06:41.15 | SeRi | ah.... mhhhhhhh 800Mhz? |
| 06:41.18 | p3nguin | yes |
| 06:41.22 | SeRi | psql? |
| 06:41.33 | p3nguin | pgsql? yes. |
| 06:41.53 | SeRi | s/psql/psgsql/ |
| 06:42.00 | SeRi | there |
| 06:42.15 | p3nguin | If you're talking about postgresql, yes I use it. |
| 06:42.32 | SeRi | ok.... mhhh ill have to test and see how dose it do on an alix.... |
| 06:46.58 | SeRi | -=[2.22 2.01 1.34]=- |
| 06:47.37 | SeRi | cpu is pegged at 100% on first core and second @34% |
| 06:49.35 | *** join/#asterisk gravin (~gravin@175.136.216.153) |
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| 06:59.55 | SeRi | dijib: is now all ways online.... lol irssi |
| 07:00.36 | p3nguin | idle |
| 07:01.16 | SeRi | yeap |
| 07:01.33 | SeRi | ~root@ :P |
| 07:01.45 | p3nguin | What a weirdo. |
| 07:01.50 | SeRi | hahahahahaha |
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| 07:12.22 | Sean-Der | http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf |
| 07:12.41 | Sean-Der | This talks about a little, I am trying to set notransfer = yes |
| 07:13.01 | Sean-Der | I don't want my server to step out of the Mediapath |
| 07:14.31 | ectospasm | for Asterisk 1.8 you should set directmedia = no |
| 07:14.38 | ectospasm | ...prolly 10 too |
| 07:14.39 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 07:14.50 | IsUp | morning |
| 07:14.53 | p3nguin | iax2? |
| 07:15.19 | p3nguin | There's directmedia in iax2? |
| 07:15.27 | ectospasm | good question |
| 07:15.31 | p3nguin | There isn't. |
| 07:15.36 | SeRi | I thought iax2 there was no directmedia |
| 07:15.53 | SeRi | o ok. I thought so :) |
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| 07:16.22 | ectospasm | didn't see the part about iax.conf... I should go to bed... |
| 07:18.22 | Sean-Der | transfer=no |
| 07:18.25 | Sean-Der | in my iax.conf |
| 07:18.30 | Sean-Der | Does exactly as I wanted it too |
| 07:18.30 | Sean-Der | ! |
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| 08:13.13 | schmidts | good morning |
| 08:14.45 | IsUp | morning |
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| 10:07.21 | phix | It's the morning? |
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| 10:34.08 | IsUp | in Tuırkey, yes. it's 12:34 atm |
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| 10:49.12 | ollii | hey i have a question about * 1.8 and t.38 fax with a grandstream ht502...i'm testing my grandstream with * APP ReceiveFax as an endpoint, so t38 capable <-> t38 capable...in a sip trace, the invite with t38 looks fine except to mixed rtp ports...grandstream offers rtpport 5004 and * 34658... here is my sip.conf: http://pastebin.com/Un0YE2yR |
| 10:49.52 | ollii | with direct media receivefax and ht502 should have the same rtp port ? |
| 10:50.19 | ollii | and yeah sip name = extension is a bad idea...i know, just testing purpose |
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| 10:55.03 | catphish | is it possible to permit or deny SIP 302 redirects on a per-call basis? |
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| 12:26.11 | IsUp | i want to monitor my PBX 7/24, if anything goes wrong (net connection, hardware error, anything) i want to see it in logs |
| 12:26.14 | IsUp | any ideas? |
| 12:28.12 | Dovid[Laptop] | IsUp: What you are asking is real broad |
| 12:28.37 | Dovid[Laptop] | for hardware it depends on the server. i use dell and they have something called openamange. i am sure there are other solutions to watch your hardware |
| 12:28.38 | IsUp | maybe i should write some custom cronjobs or something |
| 12:28.53 | Dovid[Laptop] | for the OS etc. have a look at nagios |
| 12:29.09 | Dovid[Laptop] | i believe nagios has a plugin for asterisk so you can watch that as well |
| 12:29.27 | IsUp | '[Dec 16 00:48:00] NOTICE[7074] chan_sip.c: Peer 'england' is now UNREACHABLE!' -> for example maybe i should grep 'UNREACHABLE' and dump them to a txt file |
| 12:29.55 | IsUp | oh, i dont know why but all my peers went to "UNREACHABLE" |
| 12:30.05 | IsUp | does it means my connection lost for a sec? |
| 12:30.46 | Dovid[Laptop] | IsUp: You have qualify !=no |
| 12:31.12 | Dovid[Laptop] | if you have qualify = yes or qualify = 2000 in sip.conf asterisk will send options every x amount of time to the peer etc |
| 12:31.28 | IsUp | no it's not normal 'unreachable' output |
| 12:31.42 | Dovid[Laptop] | if there is no response it marks it as unreachable so if say you call a phone you get a busy right away instead of asterisk sending an invite a few times to the device before it gives up |
| 12:31.42 | IsUp | 2000ms is enough for me, i have qualify=yes, i have 3 Asterisk setups on different locations |
| 12:33.08 | Dovid[Laptop] | IsUp: You can be getting the error for many reasons |
| 12:33.10 | *** join/#asterisk fobus912 (~fobus912@41.214.229.46) |
| 12:33.19 | fobus912 | Hi all |
| 12:33.21 | Dovid[Laptop] | internet ar your location/remote location etc. |
| 12:33.32 | Dovid[Laptop] | fobus912: hi there |
| 12:33.45 | fobus912 | Is it possible through asterisk to set a different expiration for Registeration |
| 12:33.48 | fobus912 | and subscription |
| 12:34.12 | fobus912 | Hi Dovid[Laptop] |
| 12:34.45 | Dovid[Laptop] | fobus912: I know there is a setting in the general sip.conf section. try setting it for an individual peer and see. in general the end device sets how long it watns to register with asterisk for |
| 12:35.33 | fobus912 | Dovid[Laptop] this expiration will be for both the register/Subscribtion |
| 12:35.47 | fobus912 | But if you want to have for example the register to be 3600 sec |
| 12:35.57 | fobus912 | and for the subscribe to set it to 300 sec |
| 12:36.16 | Dovid[Laptop] | fobus912: I have not poked around enough to see if you can split it. afaik it's one setting |
| 12:36.32 | fobus912 | Yes this is all i found as well |
| 12:36.37 | fobus912 | not sure if this can be done |
| 12:38.23 | fobus912 | Also i have another question |
| 12:38.42 | fobus912 | Can Asterisk be considered as an IP-PBX or a Softswitch ?? |
| 12:38.47 | fobus912 | It's quit confusing |
| 12:39.05 | leifmadsen | fobus912: PBX |
| 12:39.16 | leifmadsen | although it likely also fits into some regards as a softswitch |
| 12:40.20 | Dovid[Laptop] | leifmadsen: morning. why are you up this early? |
| 12:40.34 | leifmadsen | Dovid[Laptop]: what do you mean? |
| 12:40.35 | Dovid[Laptop] | fobus912: It was built as a PBX but you can do almost anything with it |
| 12:40.38 | *** join/#asterisk voipeng (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 12:40.39 | leifmadsen | I always start at 7:30am |
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| 12:40.54 | Dovid[Laptop] | leifmadsen: ouch. just looked at the clock. didnt realize how late it was |
| 12:40.58 | leifmadsen | :) |
| 12:41.05 | leifmadsen | what time did you THINK it was? |
| 12:41.06 | leifmadsen | <PROTECTED> |
| 12:41.15 | voipeng | haha |
| 12:41.54 | Dovid[Laptop] | 6:15 EST |
| 12:47.14 | fobus912 | @leifmadsen so it can be considered as both ? right ? |
| 12:47.37 | leifmadsen | fobus912: it really depends on how you implement it I guess, but foremost Asterisk is a PBX platform |
| 12:50.12 | fobus912 | I still don't see exactly the big difference between a softswitch and an IP-PBX |
| 12:50.22 | fobus912 | since all the article are just confusing |
| 12:50.41 | fobus912 | In one sentence how can distinguish the difference between them ? |
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| 13:03.00 | *** join/#asterisk TonyM (~TonyM@nat.softins.co.uk) |
| 13:03.37 | TonyM | haven't been able to reach digium.com or asterisk.org for several hours. known problem? |
| 13:03.44 | leifmadsen | fobus912: http://en.wikipedia.org/wiki/Softswitch <-- good job of describing a softswitch, which is something asterisk can "act like", but Asterisk contains PBX like functionality, thus it is a PBX |
| 13:03.55 | leifmadsen | TonyM: yes, apparently nameserver issue |
| 13:04.01 | TonyM | ok, thx |
| 13:13.52 | *** part/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 13:16.49 | schmidts | leifmadsen any idea which softphones would be the best to test the new video confbridge? :D |
| 13:17.00 | leifmadsen | jitsi |
| 13:17.12 | leifmadsen | Bria probably also works well, but costs money of course |
| 13:17.20 | schmidts | bad idea ;) |
| 13:17.23 | schmidts | x-lite? |
| 13:17.28 | leifmadsen | confbridge was heavily tested with jitsi |
| 13:17.38 | leifmadsen | I dont' know if x-lite has the appropriate codecs or not |
| 13:17.42 | leifmadsen | maybe |
| 13:18.08 | schmidts | ok i will check |
| 13:18.12 | schmidts | h264 i guess? |
| 13:19.09 | leifmadsen | yes |
| 13:19.20 | leifmadsen | asterisk won't do video transcoding |
| 13:19.28 | leifmadsen | so all endpoints must use the same video codec |
| 13:22.50 | schmidts | as i thought ;) |
| 13:23.28 | anonymouz666 | video transcoding seems to be way way way more complex than audio |
| 13:23.41 | leifmadsen | and it is |
| 13:23.42 | leifmadsen | :) |
| 13:24.20 | schmidts | hehe i really can understand this |
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| 14:25.48 | schmidts | leifmadsen do you know if need anything else for the video confbridge to work? |
| 14:26.05 | schmidts | i can make video calls between all 3 clients direclty but in the confbridge i only get audio |
| 14:26.23 | jaytee | can someone confirm that this is the right syntax to allow SIP traffic from a specific address? |
| 14:26.26 | jaytee | iptables -A INPUT -s 216.115.69.144 -p udp --dport 5060 -j ACCEPT |
| 14:26.34 | schmidts | channel shows me h264 and gsm on all three sip calls from the client but only slin on the confbridge in and out channels |
| 14:26.51 | schmidts | jaytee looks fine |
| 14:27.06 | jaytee | thanks, trying to tighten down security on my box |
| 14:30.28 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 14:30.58 | schmidts | leifmadsen maybe enable video in confbridge.conf is a good idea :D |
| 14:31.18 | sunfone | Howdy... anyone recall what the command is to check errors on a dahdi controlled TE410? |
| 14:31.18 | leifmadsen | schmidts: sounds like it could help :) |
| 14:32.58 | *** join/#asterisk Sean-Der (~Sean-Der@NW1-DSL-74-215-64-154.fuse.net) |
| 14:33.28 | Sean-Der | Is there anyway I could log incoming calls did with CDR? |
| 14:34.45 | sunfone | Hrm... nevermind... dahdi_maint -s <span? |
| 14:34.54 | sunfone | ^?^> |
| 14:35.08 | schmidts | leifmadsen yes works now :D any chance to get something like split screen directly out of asterisk? and please dont tell me, i have the code :P |
| 14:35.37 | [TK]D-Fender | Sean-Der, Callare already logged with CDR |
| 14:35.43 | [TK]D-Fender | calls are* |
| 14:36.29 | Sean-Der | What column? |
| 14:36.42 | Sean-Der | I am using Freepbx so I asked over in the right channel |
| 14:37.23 | [TK]D-Fender | column? Pardon? |
| 14:37.44 | Sean-Der | in my MySQL cdr table |
| 14:37.56 | [TK]D-Fender | Sean-Der, Indeed, wrong channel |
| 14:38.01 | leifmadsen | schmidts: no |
| 14:38.13 | leifmadsen | schmidts: that would require transcoding |
| 14:39.02 | schmidts | leifmadsen hmm ok i see is there any plan of bring this into asterisk maybe version 12 or something? |
| 14:39.22 | leifmadsen | schmidts: there is no plan because it is non-trivial |
| 14:39.43 | leifmadsen | and using an external library causes potential issues due to licensing requirements |
| 14:39.56 | leifmadsen | and would be a significant hit on CPU usage as well |
| 14:40.19 | schmidts | yes thats true ;) but still a very nice feature |
| 14:40.27 | leifmadsen | winning the lottery is nice too :) |
| 14:40.44 | leifmadsen | anyways, for multiple pane video, you'll likely have to look into that yourself to get it any time soon |
| 14:41.30 | schmidts | leifmadsen ha how many hours does a day have in canada? must be more than the days here have :D |
| 14:41.50 | leifmadsen | ya, that's the same problem here too :) |
| 14:45.54 | *** join/#asterisk CGMChris (~chatzilla@74.143.228.142) |
| 14:46.45 | CGMChris | I haven't been able to get to asterisk.org all morning. Anybody know what's up? |
| 14:47.07 | schmidts | CGMChris Digiums nameserver plays hard to get today |
| 14:47.08 | [TK]D-Fender | Title <- |
| 14:47.22 | [TK]D-Fender | Topic even. |
| 14:47.55 | schmidts | leifmadsen maybe you know this, is there any app or func to disable faxdetect for a call? |
| 14:48.05 | leifmadsen | not sure |
| 14:48.07 | CGMChris | [TK]D-Fender: I agree, something should be included in the topic. |
| 14:48.35 | leifmadsen | it's not even 9am at Digium right now |
| 14:48.49 | [TK]D-Fender | No, i was beeting its due to flooding for the release of * 10 :) |
| 14:48.57 | [TK]D-Fender | betting* |
| 14:49.05 | *** topic/#asterisk by leifmadsen -> Yes asterisk.org and digium.com are down -=- Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0 (2011/12/15), 1.8.8.0 (2011/12/15), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 14:50.07 | CGMChris | hmm, I guess I'll go try to hack my way around using IP addresses to install asterisk from the digium/asterisk YUM repos.... bbl |
| 14:52.53 | leifmadsen | you all realize there is an easy way to find out the answer to these types of questions? |
| 14:52.54 | leifmadsen | http://www.downornot.com/digium |
| 14:53.23 | [TK]D-Fender | Interestingly the WIKI is still up there. |
| 14:53.48 | CGMChris | I knew it was down... the real problem, for me at least, is I need to have a working PBX by end of day and all I have now is a fresh CentOS install. |
| 14:54.19 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 14:54.55 | sereal-work | Can someone tell me if there is anything special about creating AGI scripts? |
| 14:55.18 | sereal-work | Are they just making a socket connection to asterisk? |
| 14:56.21 | sereal-work | If I remember correctly from my expermentation of the asterisk manager stuff, it was just a socket connection I believe. |
| 14:57.08 | TonyM | fastagi uses a socket: AGI(agi://hostname/action) |
| 14:57.35 | TonyM | normal AGI just calls a program and communicates with its stdin/stdout: AGI(program) |
| 14:58.24 | CGMChris | sereal-work: http://www.voip-info.org/wiki/view/Asterisk+AGI lots of examples in all sorts of programming languages. |
| 15:01.59 | sereal-work | TonyM, so really it doesn't mater the language, there is nothing special going on behind the scenes. |
| 15:02.00 | sereal-work | As long as you can write to stdin/stdout |
| 15:02.20 | *** join/#asterisk andygraybeal (~andy.gray@obsidian.casanueva.com) |
| 15:02.36 | TonyM | @serial-work - that's correct |
| 15:02.57 | TonyM | I write mine in C - many people use php and some use perl or python |
| 15:03.23 | CGMChris | php ftw! |
| 15:08.15 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0 (2011/12/15), 1.8.8.0 (2011/12/15), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:08.21 | leifmadsen | services restored |
| 15:08.49 | CGMChris | awesome. Thanks ! |
| 15:09.16 | leifmadsen | don't thank me, I'm just delivering the msg :) |
| 15:09.59 | TonyM | cool! |
| 15:13.19 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:14.41 | *** join/#asterisk CGMChris (~chatzilla@74.143.228.142) |
| 15:15.14 | CGMChris | New problem.... http://packages.asterisk.org/centos/6/ = 404. I guess I'm ahead of my time. Best bet is I guess to use CentOS 5? |
| 15:15.31 | r0m|u | search for the rpm |
| 15:16.21 | CGMChris | finding the RPM isnt difficult, but yum support is great.... i dont want to end up in rpm dependency purgatory and spend a half day searching. |
| 15:16.49 | r0m|u | lazy :/ :P |
| 15:17.08 | CGMChris | I guess I can try my luck at hardcoding the path in the yum repo file, instead of letting it use the OS version...and see how the v5 ones do. |
| 15:17.24 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:19.17 | CGMChris | nope, doesnt like the kernel version. back to v5 I go ! |
| 15:20.03 | [TK]D-Fender | CGMChris, You didn't think that would be the very first thing that would blow up in your face? :) |
| 15:20.21 | [TK]D-Fender | CGMChris, ARCH-BOMB! |
| 15:20.25 | CGMChris | [TK]D-Fender: Hope blinded me. :-\ |
| 15:20.34 | [TK]D-Fender | assplodes |
| 15:27.38 | *** join/#asterisk irroot (~gregory@41.51.78.197) |
| 15:32.41 | leifmadsen | CGMChris: ya no centos 6 packages yet if I remember correctly |
| 15:34.52 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 15:35.02 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 15:36.01 | TonyM | g729 licences: anyone confirm whether a g729 call into Meetme uses up one licence, or whether it uses up two (one for Rx and one for Tx)? |
| 15:36.51 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-377107.home.otenet.gr) |
| 15:37.28 | r0m|u | To what I understand is one license per channel |
| 15:38.25 | TonyM | that's what I would have thought... but wanted to be sure |
| 15:41.58 | [TK]D-Fender | TonyM, Every G.729 leg into Meetme sucks up one encoder & 1 decoder |
| 15:45.11 | leifmadsen | each license provides 1 encoder, 1 decoder |
| 15:45.23 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:47.08 | TonyM | thanks! |
| 15:47.33 | *** join/#asterisk serafie (~erin@nat/digium/x-pysbwbsvfzixfwmb) |
| 15:47.55 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 15:51.04 | p3nguin | r0m|u: VUC today! |
| 15:52.16 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:54.00 | r0m|u | p3nguin: yES |
| 15:54.05 | r0m|u | Yes :) |
| 15:55.35 | r0m|u | no meeting and I brought lunch in. The GSM Gateway should txt me with a reminder :P |
| 15:56.20 | r0m|u | pinto beans, white rice, and beef stew is what is for lunch today. |
| 15:57.12 | p3nguin | I'm not sure what I'm going to have for lunch. |
| 15:57.38 | *** join/#asterisk Roadblock_RVA (~Roadblock@office.neteasyinc.com) |
| 15:59.55 | r0m|u | is 10AM and I am thinking about lunch.... damn |
| 16:00.59 | *** part/#asterisk TonyM (~TonyM@nat.softins.co.uk) |
| 16:09.10 | r0m|u | my stickers are done. |
| 16:09.17 | r0m|u | yezzzz |
| 16:10.24 | p3nguin | pot stickers? Those are delicious! |
| 16:10.39 | r0m|u | lol |
| 16:10.50 | p3nguin | Just like a fried dumpling. |
| 16:10.58 | r0m|u | now does are good |
| 16:10.58 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:11.34 | r0m|u | got an email that my requested "stickers" where done.... they did gave me some shit about it but they where cool enough to do it :) |
| 16:11.40 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:11.50 | p3nguin | Who did you have make them? |
| 16:12.21 | r0m|u | a shop down the road from my house..... flea market d00d could not do it withouth a high res img |
| 16:12.39 | r0m|u | funny.... they knew aout it. |
| 16:13.00 | r0m|u | they ask me what where my intentions. lol.... I said personal? |
| 16:13.24 | p3nguin | "I'm planning to stick them to my butt." |
| 16:13.31 | p3nguin | "And to my wife's butt." |
| 16:13.31 | r0m|u | rofl!!!!!!!!! |
| 16:13.35 | r0m|u | hahahahahah |
| 16:14.44 | p3nguin | "And I might even give some to my kids, so they can stick them to their butts, as well." |
| 16:14.53 | r0m|u | hahahahahahhahaha |
| 16:20.32 | r0m|u | any body used askozia? |
| 16:21.57 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 16:23.01 | *** join/#asterisk casix (~casix@117.168.17.95.dynamic.jazztel.es) |
| 16:24.48 | casix | hello, I'm trying to make and recieve calls from gmail but doesn't work. With the windows gtalk exe is working perfect but when I try from the gmail web doesn't. Any idea of the differeces from the configurations? |
| 16:25.11 | *** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 16:25.25 | krotos | hi all |
| 16:28.29 | krotos | there is a way to bind asterisk on multiple ip |
| 16:28.35 | krotos | and not using bind=0.0.0.0 ? |
| 16:28.35 | *** part/#asterisk catphish (~catphish@2001:9d8:2005:11:222:15ff:fe88:aae2) |
| 16:28.46 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.16.145) |
| 16:29.30 | anonymouz666 | did you know that jeep = gp = general purpose |
| 16:29.32 | anonymouz666 | lol |
| 16:30.01 | *** join/#asterisk bmg505 (~leon@196-209-101-126.dynamic.isadsl.co.za) |
| 16:30.58 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 16:31.33 | p3nguin | krotos: You need to bind them all or bind one. If you bind them all, then you can firewall off any that you don't want to be used. |
| 16:32.29 | krotos | p3nguin: thankyou :) |
| 16:33.15 | anonymouz666 | it would be nice to bind one, two or all and even to choose between those binds to tcp/tls or udp. |
| 16:33.25 | anonymouz666 | or a random port |
| 16:33.51 | p3nguin | Choosing a random port to bind might be a bit useless. |
| 16:34.22 | p3nguin | Starting asterisk would be like spinning the wheel of fortune... you never know what port it will bind, so you can't configure anything else to use it. |
| 16:34.34 | *** join/#asterisk greenwolf (~greenwolf@cpe-67-241-181-90.buffalo.res.rr.com) |
| 16:34.39 | anonymouz666 | Non rare I have seem systems running away from 5060 |
| 16:35.04 | *** join/#asterisk gokulnath (~gokulnath@117.216.80.33) |
| 16:37.01 | r0m|u | p3nguin: how can I get access to view the VUC panel? |
| 16:37.18 | anonymouz666 | "spinning the whell of fortune" hehe I just remembered the same term in portuguese |
| 16:37.19 | anonymouz666 | it's fun |
| 16:37.33 | p3nguin | log in |
| 16:37.57 | r0m|u | link |
| 16:38.17 | p3nguin | zipdx.com |
| 16:39.04 | r0m|u | conf code? |
| 16:39.17 | p3nguin | 200109 |
| 16:39.23 | p3nguin | I think |
| 16:39.52 | Roadblock_RVA | Is there a simple way to receive a phone call notification instead of an email for voicemail? |
| 16:39.57 | r0m|u | starts at 10:45 AM? |
| 16:40.09 | r0m|u | CST |
| 16:40.53 | casix | Roadblock_RVA: via gmail chat |
| 16:40.56 | casix | hello, I'm trying to make and recieve calls from gmail but doesn't work. With the windows gtalk exe is working perfect but when I try from the gmail web doesn't. Any idea of the differeces from the configurations? |
| 16:41.01 | p3nguin | It starts at 11, but you can join 15 minutes early. |
| 16:43.43 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu) |
| 16:49.08 | p3nguin | r0m|u: Oops. My dyslexia got me again. 200901 |
| 16:49.19 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 16:51.14 | *** join/#asterisk bmg505 (~leon@196-209-101-126.dynamic.isadsl.co.za) |
| 16:53.26 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu) |
| 16:54.51 | p3nguin | roadblock_rva: I use jabber for that. |
| 16:55.35 | Roadblock_RVA | I'll look into that ty. |
| 16:56.26 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 17:02.05 | r0m|u | p3nguin: The Organizer has not yet started the meeting. Please try again in a few minutes. |
| 17:02.14 | p3nguin | (1049.08) <p3nguin> r0m|u: Oops. My dyslexia got me again. 200901 |
| 17:03.58 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 17:04.15 | r0m|u | man my cell pops up :( |
| 17:04.34 | p3nguin | What does that mean? |
| 17:04.52 | r0m|u | That I am calling from my cell to the gsm gateway to the cong |
| 17:04.59 | r0m|u | s/cong/conf/ |
| 17:05.14 | p3nguin | Oh, I see you. You're the only one with an 11-digit USA caller id number. |
| 17:05.26 | r0m|u | I m not sure I can overwrite it since the gateways forces it |
| 17:06.16 | p3nguin | Don't expect anyone to call you. |
| 17:07.03 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:07.40 | r0m|u | I hope |
| 17:08.16 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 17:10.44 | p3nguin | Configure your account with your name and then use the call me button. Enter in a phone number or SIP URI for the conf to call you. That will hide your cell phone number. |
| 17:11.05 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-181-48.netvisao.pt) |
| 17:11.11 | [sr] | welcome 10.0.0! |
| 17:14.16 | r0m|u | Ill do it for the next one. hope fully I dont get an ass calling me |
| 17:14.50 | p3nguin | You won't. |
| 17:22.20 | r0m|u | I am so ammazed how clear it is over my cell system |
| 17:23.52 | r0m|u | p3nguin: what is the link for that site that has the numbers of spammers? |
| 17:29.38 | r0m|u | I doubt they use something like watson |
| 17:30.21 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 17:33.03 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:33.47 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 17:33.48 | p3nguin | r0m|u: There are several: whocallsme.com, callercomplaints.com, mrnumber.com, callerr.com, 800notes.com, and others. |
| 17:36.11 | r0m|u | Thanks |
| 17:44.22 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 17:44.24 | wcselby | o/ |
| 17:44.39 | r0m|u | qaz up wcselby |
| 17:44.48 | wcselby | not much |
| 17:44.48 | r0m|u | s/gaz/waz/ |
| 17:44.56 | wcselby | how's it going for you? |
| 17:45.21 | r0m|u | nothing much... listening on vcu.... on holiday mode at work :P |
| 17:45.42 | wcselby | Qwell or leifmadsen - asterisk-1.8-current still references 1.8.7.2, you have to manually download 1.8.8.0. just fyi |
| 17:45.55 | leifmadsen | oops, my bad |
| 17:45.57 | leifmadsen | wcselby: fixing |
| 17:46.02 | wcselby | leifmadsen thanks :) |
| 17:46.08 | leifmadsen | thank you |
| 17:46.14 | wcselby | np |
| 17:46.16 | leifmadsen | for some reason I missed that step yesterday :( |
| 17:48.22 | r0m|u | wcselby: bussy today? |
| 17:48.40 | leifmadsen | wcselby: fixed -- will populate in the next 5-10 mins |
| 17:48.52 | wcselby | r0m|u nah, have to install asterisk on one server and mysql on another and follow up on a couple items |
| 17:48.57 | wcselby | r0m|u i'm working from home |
| 17:49.17 | r0m|u | cool. fun :) |
| 17:49.26 | wcselby | leifmadsen cool beans :) |
| 17:50.01 | r0m|u | I am going to try and build astlinux with ast 1.8 and 10. |
| 17:50.20 | r0m|u | not sure how card it can be.... I am sure 1.8 should not be that hard. |
| 17:50.29 | r0m|u | s/crad/hard/ |
| 17:50.32 | wcselby | what's up with astlinux, you and p3nguin have been all about that build for a while now. i had never heard anything good about it before |
| 17:50.34 | r0m|u | damn |
| 17:50.50 | r0m|u | p3nguin: ? not him... lol |
| 17:50.54 | p3nguin | I've never cared about it. |
| 17:51.03 | wcselby | oh heh |
| 17:51.12 | wcselby | well then I guess it was just r0m|u that I saw talking about it |
| 17:51.18 | r0m|u | lol |
| 17:51.27 | p3nguin | r0m|u/seri |
| 17:51.32 | r0m|u | :) |
| 17:51.41 | wcselby | okay, i thought it wa smore than just r0m|u :) |
| 17:51.54 | r0m|u | wcselby: so you heard negative things? |
| 17:52.00 | wcselby | nah |
| 17:52.04 | wcselby | well i dunno |
| 17:52.14 | r0m|u | I see. |
| 17:52.15 | wcselby | all i'd ever seen were people asking how to make this part work or that part work |
| 17:52.26 | r0m|u | ah. I see. |
| 17:52.28 | wcselby | not really negative things, but i only ever saw it when people were having issues with it |
| 17:55.38 | r0m|u | ah looks like I dont have to do anything. its there. all I have to do is add the source :) |
| 17:57.11 | r0m|u | annoying |
| 17:58.24 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:59.16 | r0m|u | hahahahaha |
| 18:01.46 | r0m|u | intergrate it to pr0n videos |
| 18:03.32 | *** join/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 18:03.59 | wcselby | so, if I don't ever use ael, am I safe in simply unselecting pbx_ael on the pbx modules of make menuselect? |
| 18:04.16 | p3nguin | should be safe. |
| 18:04.43 | [TK]D-Fender | more than. |
| 18:04.54 | [TK]D-Fender | wcselby, Mind you I always just disable it in modules.conf |
| 18:05.06 | wcselby | [TK]D-Fender i've always done that too |
| 18:05.11 | [TK]D-Fender | I don't see the point in crippling an option that I can opt out of in config. |
| 18:05.13 | wcselby | same with like chan_mgcp and other things I don't need |
| 18:05.21 | wcselby | [TK]D-Fender that's a good point |
| 18:05.23 | [TK]D-Fender | Worst thing is with yo had something that takes real work to get back |
| 18:06.08 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 18:07.01 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 18:08.02 | p3nguin | I disable as much as possible to keep the package size as small as I can get it. |
| 18:08.02 | wcselby | i always found it weird that chan_mgcp is enabled by default, but res_pktccops is not. if you don't have res_pktccops installed, chan_mgcp will throw errors on startup (or at least it used to...I guess I'll find out in a minute if that's the case still) |
| 18:09.26 | r0m|u | gt22 is 64kbps? |
| 18:09.31 | r0m|u | g722* |
| 18:10.13 | wcselby | 48, 56, or 64 kbps accordign tot he first line of wikipedia |
| 18:10.13 | [TK]D-Fender | YES |
| 18:10.26 | wcselby | http://en.wikipedia.org/wiki/G.722 |
| 18:10.51 | wcselby | from that page -> Whilst G.722 allows for bitrates of 64, 56 and 48 kbit/s, in practice, data is encoded at 64 kbit/s, with bits from the lower sub-band being used to encode auxiliary data. The greater the number of bits allocated to aux data, the lower the bit rate. |
| 18:11.52 | r0m|u | how come it consumes more bandwitdth than ulaw? |
| 18:12.37 | wcselby | ulaw in practice consumes about 80-90 kbps |
| 18:13.44 | r0m|u | ok so I guess I got confused.... I thought that g722 been a HD codec would consume more data than ulaw |
| 18:14.24 | wcselby | i dunno, never really played with it |
| 18:15.12 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:15.51 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:15.51 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:17.41 | WIMPy | No, it's exactely yhe same. |
| 18:18.08 | r0m|u | WIMPy: Thanks for the answer |
| 18:18.12 | *** join/#asterisk drmessano-lt (~nonya@pdpc/supporter/active/drmessano) |
| 18:18.15 | WIMPy | .. as it was made for exactely the same old 64kbps pcm channels. |
| 18:19.49 | drmessano-lt | stretches |
| 18:21.06 | *** join/#asterisk BlackBishop (dexter@ipv6.d3xt3r01.tk) |
| 18:21.54 | *** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew) |
| 18:24.47 | wcselby | is there a way to disable reading users.conf by using a noload => line inside modules.conf? |
| 18:26.13 | p3nguin | Just delete the file. |
| 18:26.25 | wcselby | yeah that's what I usually do |
| 18:26.28 | wcselby | i was just curious |
| 18:30.23 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu) |
| 18:31.13 | r0m|u | wcselby: everything simmer down @ your job? |
| 18:31.29 | wcselby | r0m|u not really |
| 18:31.34 | r0m|u | :( |
| 18:31.38 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 18:32.12 | wcselby | r0m|u in fact, they told me they don't want me to get on IRC anymore during the days I'm in the office |
| 18:32.27 | *** join/#asterisk lopz (be6806d4@gateway/web/freenode/ip.190.104.6.212) |
| 18:32.28 | wcselby | mostly because they don't understand it, and it'd be too difficult to explain it to them |
| 18:32.35 | r0m|u | wtf! nazi's! |
| 18:32.37 | lopz | hi |
| 18:32.55 | p3nguin | wcselby: I should have answered your question more directly... pbx_config loads the file, so there's really nothing you can noload to keep it from being loaded. You basically just have to delete the file to keep it from being used. |
| 18:33.34 | r0m|u | wcselby: d00d that really sucks. irc is how I make my day go by.... I do most of my work in the morning unless I have to be @ the PDC |
| 18:33.50 | wcselby | p3nguin I think I got it just now by disabling res_phoneprov.so |
| 18:35.18 | p3nguin | hmm... pbx_config uses it, chan_sip uses it, res_phoneprov uses it, chan_agent uses it, chan_iax2 uses it, app_directory and app_voicemail use it... |
| 18:36.16 | wcselby | p3nguin i know, but res_phoneprov was the only thing throwing up an error for the file not being there at startup time. so I noload'ed it and I'm not getting the error / warning anymore |
| 18:36.27 | p3nguin | I see. |
| 18:36.40 | wcselby | i've also deleted the file :) |
| 18:38.45 | dijib | i have life |
| 18:38.49 | dijib | how is everyone |
| 18:39.01 | dijib | im facking starving, sitting in my automobile. |
| 18:39.16 | dijib | waiting for the GF |
| 18:41.14 | r0m|u | waz up dijib |
| 18:41.32 | WIMPy | And chan_dahdi. |
| 18:41.36 | dijib | nothin |
| 18:41.37 | dijib | cold |
| 18:41.52 | p3nguin | Yes, and chan_dahdi. |
| 18:42.01 | r0m|u | same here..... not as bad though..... |
| 18:42.16 | p3nguin | and chan_h323 |
| 18:42.24 | dijib | dude i should send you guys a picture of what im looking at. |
| 18:42.33 | dijib | i feel like rotting from the inside out |
| 18:42.49 | p3nguin | Do you know her name? |
| 18:43.12 | dijib | so im building from trunk asterisk on another box. |
| 18:43.28 | dijib | cheated a bit with dependencies |
| 18:43.31 | dijib | dahdi |
| 18:43.34 | dijib | swift |
| 18:43.41 | dijib | lumenvox (not yet) |
| 18:43.48 | dijib | what else. mp3 |
| 18:43.57 | dijib | digium fax |
| 18:44.02 | dijib | what else iam i missing?> |
| 18:44.13 | dijib | and i know i flood, thanks. |
| 18:44.19 | anonymouz666 | wants an oakley sunglass |
| 18:46.10 | wcselby | why does "service dahdi status" not return any output (running latest dahdi-linux-complete-current) ? |
| 18:47.43 | dijib | here you go r0m|u http://i.imgur.com/qFngY.jpg |
| 18:48.40 | anonymouz666 | where is it ? |
| 18:49.27 | wcselby | afk a bit |
| 18:51.48 | r0m|u | dijib: nice |
| 18:52.14 | p3nguin | I didn't see much nice about it. |
| 18:52.19 | r0m|u | looks boring but peaceful and I like peaceful |
| 18:52.37 | r0m|u | is trying to be nice. |
| 18:52.37 | dijib | not nice cold |
| 18:52.51 | dijib | its been snowing on and off all day, wet crap |
| 18:53.04 | dijib | this urun down crap town |
| 18:53.17 | r0m|u | looks like TX is not having a white christmass this year |
| 18:53.23 | r0m|u | at least Houston |
| 18:53.29 | dijib | jet streams |
| 18:53.52 | dijib | http://images.intellicast.com/WxImages/Radar/usa.gif |
| 18:54.25 | r0m|u | man cant wait for the 20th |
| 18:57.22 | *** join/#asterisk hfb (~hfb@pool-96-247-114-143.lsanca.dsl-w.verizon.net) |
| 19:00.05 | r0m|u | what to do next.... :/ I am now doing the waiting game with the phone.... |
| 19:02.43 | *** join/#asterisk brainiac (~mhauss@24-183-217-180.dhcp.jcsn.tn.charter.com) |
| 19:04.19 | *** join/#asterisk Korolev (~Korolev@nmd.sbx08290.northfl.wayport.net) |
| 19:04.40 | *** part/#asterisk brainiac (~mhauss@24-183-217-180.dhcp.jcsn.tn.charter.com) |
| 19:05.00 | SuperNull | Hey guys, Anyone able to help me with user_id not being filled in cdr_mysql ? i have the setting in cdr_mysql.conf |
| 19:06.20 | *** join/#asterisk brainiac (~brainiac@24-183-217-180.dhcp.jcsn.tn.charter.com) |
| 19:08.49 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 19:12.51 | BlackBishop | anyone with a ht503 around here any ideas why when I pickup my fxo, it still rings (weird ring sound though) ? ( the other party doesn't see me answering ) |
| 19:14.13 | *** join/#asterisk libryder (~david@209.33.214.243) |
| 19:15.37 | libryder | i'm not sure where to ask this, but i have a rails app that makes a connection to the AMI using adhearsion's ManagerInterface class and because rails isn't closing the sockets when it is finished in the AMI, asterisk finally fills up the log with the following error message: WARNING[6076] asterisk.c: Accept returned -1: Too many open files |
| 19:16.20 | libryder | i found this patch: https://issues.asterisk.org/jira/secure/attachment/39460/0001-prevent-flodding-logs-when-too-many-open-files.patch but i'm not sure what version asterisk it's for and if it will solve my problem |
| 19:17.06 | WIMPy | That's obviouselt a bug in your software and not in asterisk. |
| 19:17.52 | WIMPy | Keeping a connection open may be a good idea, but only if you reuse it. |
| 19:19.06 | libryder | it's ok that asterisk runs out of connections until i fix the application, but asterisk writes a 7gb log file until the machine finally locks up |
| 19:19.31 | libryder | so i can't really do any testing safely |
| 19:21.21 | WIMPy | For me, Asterisk has been automagically starting a new log file, once it hit the 2GB mark. |
| 19:21.55 | libryder | i wonder if that's a setting or if asterisk is just getting caught in an unstoppable loop |
| 19:22.14 | p3nguin | blackbishop: Clarify what you mean when you say, "I pickup my fxo." |
| 19:22.16 | WIMPy | Don't be a politician. Fix the part that's wrong, not the one that tells you about it. |
| 19:25.20 | r0m|u | does any body know of an affordable wireless headset? |
| 19:25.47 | WIMPy | What kind of wireless? |
| 19:27.15 | r0m|u | bluetooth is prefeer |
| 19:27.35 | BlackBishop | p3nguin: the fxo line of the ht503 thing, is connected to my phone. |
| 19:27.46 | BlackBishop | grrrr, I meant fxs |
| 19:28.22 | p3nguin | blackbishop: So what you meant before was that you pick up your phone (which is connected to the FXS port)? |
| 19:29.30 | BlackBishop | yup |
| 19:29.44 | BlackBishop | and I hear a weird sound like .. ringing .. |
| 19:30.13 | BlackBishop | and the other end ( probably because the ht ) doesn't notice me picking up |
| 19:30.38 | [TK]D-Fender | libryder, astmanproxy <- |
| 19:32.00 | libryder | [TK]D-Fender: awesome, thanks |
| 19:32.30 | r0m|u | WIMPy: what fs? |
| 19:33.18 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:33.18 | WIMPy | fs? |
| 19:33.52 | r0m|u | I think there was a limit on the file size on ext |
| 19:33.59 | r0m|u | I for got if it was ext2 or 3 |
| 19:34.29 | r0m|u | adnd I think it had to do with 32Bit |
| 19:34.47 | r0m|u | shit I dont remember is been so long |
| 19:34.47 | WIMPy | Ah. Hmm. I have NFI, what FS that box used. I didn't set it up and it has long gone. |
| 19:34.49 | r0m|u | never mind |
| 19:35.11 | WIMPy | So yes, maybe that was just because it couldn't write any more. |
| 19:35.28 | r0m|u | ah I see. |
| 19:39.54 | *** join/#asterisk rpluto (~rpluto@a95-92-60-159.cpe.netcabo.pt) |
| 19:44.44 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 19:45.37 | wcselby | okay |
| 19:45.44 | wcselby | so i just opened notepad to write down some commands |
| 19:45.51 | wcselby | and the first thing I typed was "conf t" |
| 19:45.57 | wcselby | as if I needed to enter config mode |
| 19:46.05 | wcselby | .... |
| 19:46.18 | wcselby | has a headache |
| 19:46.38 | WIMPy | Sue Cisco for mental torture. |
| 19:47.04 | p3nguin | That's a good one. |
| 19:47.35 | wcselby | i guess it's better than me trying to use vi commands inside notepad |
| 19:48.13 | WIMPy | That would at least stay in the "editor" category. |
| 19:49.46 | wcselby | if i noload chan_dahdi.so, does it screw anything up with dahdi timing? |
| 19:50.43 | *** part/#asterisk rpluto (~rpluto@a95-92-60-159.cpe.netcabo.pt) |
| 19:58.12 | *** join/#asterisk nix8n82 (~hmg@75-174-150-98.chyn.qwest.net) |
| 20:03.09 | wcselby | lol, i love the fact that if you google for "asterisk wiki", the actual wiki.asterisk.org is the 4th result, below voip-info and two wikipedia entries |
| 20:06.13 | wcselby | if I want to pull the IP address of an inbound SIP call, is there a variable for that? |
| 20:11.49 | *** join/#asterisk mocker (~mocker@72.165.148.230) |
| 20:23.38 | *** join/#asterisk mocker (~mocker@72.165.148.230) |
| 20:23.57 | p3nguin | Timing, probably not... but you'll want chan_dahdi for things like meetme. |
| 20:26.45 | WIMPy | wcselby: CHANNEL(peerip) |
| 20:32.51 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 20:36.06 | wcselby | WIMPy will that give a better result than SIPCHANINFO(recvip) ? |
| 20:38.28 | *** join/#asterisk mocker (~mocker@72.165.148.230) |
| 20:42.17 | justdave | anyone know what the prefered way is to get the music-on-hold files for asterisk on rhel6? None of the asterisk packages in EPEL seem to contain them |
| 20:42.48 | justdave | the main asterisk package creates /usr/share/asterisk/moh but doesn't put anything in it |
| 20:43.11 | WIMPy | wcselby: I have no idea what the differences are. Apart from that SIPCHANINFO is obviousely SIP only. |
| 20:46.09 | dijib | k back |
| 20:49.12 | malcolmd | justdave: maybe just grab the tarballs then from downloads.digium.com/pub/telephony/sounds ? |
| 20:50.31 | justdave | malcolmd: yeah, probably. Got several asterisk servers to deploy, guess I need to drop that tarball on an internal webserver somewhere puppet can get at it. |
| 20:50.46 | malcolmd | no idea why it's not in the epel packages, sorries |
| 20:52.51 | r0m|u | my sugar lavels are going crazy! |
| 20:53.03 | r0m|u | to much cookies |
| 20:55.26 | *** join/#asterisk kresp0 (~kresp0@200.168.16.95.dynamic.jazztel.es) |
| 20:56.25 | kresp0 | hi all, im trying to filter telemarketers using "LookupBlacklist", but it seems that doesnt work: |
| 20:56.28 | kresp0 | [Dec 16 21:53:05] WARNING[9226]: pbx.c:3680 pbx_extension_helper: No application 'LookupBlacklist' for extension (amigos, 25, 2) |
| 20:57.05 | kresp0 | i'm using Asterisk 1.6.2.9-2+squeeze3 |
| 20:57.17 | leifmadsen | doesn't work because it's not loaded |
| 20:57.26 | leifmadsen | LookupBlacklist doesn't exist apparently in your version |
| 20:58.12 | kresp0 | thanks leifmadsen. any clue on how to load LookupBlacklist? |
| 20:58.27 | leifmadsen | if your package doesn't supply it, then you can't unless you comile |
| 20:58.30 | leifmadsen | compile* |
| 20:59.02 | kresp0 | ok, if i didnt found an alternative i'll compile * |
| 20:59.07 | kresp0 | thank you again leifmadsen |
| 20:59.08 | r0m|u | must compile the world! |
| 20:59.23 | leifmadsen | just look in /usr/lib/asterisk/modules/ and make sure it's not there -- maybe your modules.conf doesn't load it |
| 20:59.37 | leifmadsen | not sure, I don't normally use packages, especially ones crazy old like that one |
| 20:59.57 | r0m|u | lol |
| 21:00.15 | kresp0 | it's not there :/ |
| 21:00.40 | kresp0 | i like to use packages because others took care when a security update apears |
| 21:02.10 | *** join/#asterisk afink (~afink@207.106.66.194) |
| 21:02.13 | r0m|u | Is that right? is that why you on 1.6x? |
| 21:02.50 | kresp0 | yep r0m|u |
| 21:03.11 | kresp0 | also, im on 1.6 because that is the version that comes with debian stable |
| 21:03.32 | drmessano | Asterisk 10 rocks.. it's like 9 better than 1.x |
| 21:03.38 | leifmadsen | kresp0: you realize you're about 3-4 security issues behind right? |
| 21:03.52 | r0m|u | leifmadsen: apperently he does not |
| 21:04.05 | afink | Hello everyone, I am trying to block some incoming callers. See anything wrong here? http://pastebin.com/uPwj5TFC |
| 21:04.05 | kresp0 | omg, no! |
| 21:04.27 | r0m|u | drmessano: Ill catch the boat next year :P |
| 21:05.08 | drmessano | It was a painless upgrade.. I can't get any of the text messaging stuff to work yet, but everything else seems stable |
| 21:05.57 | WIMPy | afink: You realize that the 2nd GotoIf will never be executed? |
| 21:06.31 | leifmadsen | kresp0: you're missing every security issue raised in 2011 |
| 21:06.39 | leifmadsen | I count 14 |
| 21:06.41 | r0m|u | out for the week end |
| 21:06.42 | leifmadsen | http://downloads.asterisk.org/pub/security/ |
| 21:07.54 | afink | WIMPy: that is my issue, thanks! |
| 21:08.15 | justdave | huh, the Haruhi S2 box set has a CD in it with the OP/ED on it |
| 21:08.18 | afink | too many left-handed cigarettes at lunch |
| 21:08.23 | kresp0 | thank you leifmadsen, im adding http://packages.asterisk.org/deb to my repos |
| 21:11.04 | wcselby | later r0m|u |
| 21:14.37 | kresp0 | leifmadsen, r0m|u: i've updated to 1.8.7.2 using the asterisk.org repos, and LookupBlacklist still doesn't exist :/ |
| 21:14.53 | leifmadsen | then you're not loading it, or that application doesn't exist |
| 21:15.06 | leifmadsen | was that a production server? |
| 21:15.14 | p3nguin | Okay, ipv6 tunnel officially moved from old router to vyatta router. |
| 21:15.21 | WIMPy | I haven't read that name before. |
| 21:15.21 | kresp0 | hehe an amateur one leifmadsen, dont worry :D |
| 21:15.22 | leifmadsen | did you just upgrade from 1.6.2 to 1.8 on production? that isn't smart. |
| 21:15.47 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 21:16.30 | kresp0 | i've made a config backup and i can revert all the changes if needed |
| 21:18.06 | dijib | hey p3nguin did you ever get that $5 |
| 21:18.39 | p3nguin | will look in 2 minutes |
| 21:19.05 | p3nguin | The last time I saw, I had not. |
| 21:19.10 | dijib | lol |
| 21:19.16 | dijib | it will be there soon |
| 21:19.19 | p3nguin | It was "processing," though. |
| 21:19.20 | dijib | dont worry about checking |
| 21:19.24 | dijib | im good for it |
| 21:19.24 | dijib | lol |
| 21:19.58 | dijib | so what software do i need for a full blown asterisk install? |
| 21:20.05 | p3nguin | Pending Until Dec 19, 2011 |
| 21:20.13 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 21:20.19 | dijib | canadian banks are slow |
| 21:20.30 | *** join/#asterisk mocker (~mocker@72.165.148.230) |
| 21:20.44 | p3nguin | Good thing I'm not starving waiting for it. :) |
| 21:21.25 | p3nguin | Since, you know, everyone buys food with PayPal. |
| 21:21.47 | dijib | lol |
| 21:21.53 | dijib | grocerygateway.com man |
| 21:22.46 | dijib | you can deposit into your el banco |
| 21:25.23 | kresp0 | leifmadsen, and for the record: finally I was able to made a LookupBlacklist alternative (without compiling asterisk) |
| 21:25.36 | leifmadsen | ok |
| 21:25.49 | leifmadsen | "how" would be better for the record though |
| 21:25.52 | kresp0 | get the idea from here: http://bahjons.com/stuff/lookupblacklist-alternative |
| 21:25.57 | kresp0 | xD |
| 21:25.58 | kresp0 | sure |
| 21:26.15 | leifmadsen | oh so you just wrote some dialplan... ya, that is usually the solution |
| 21:26.28 | leifmadsen | I can think of at least 2 things I would fix in that dialplan :) |
| 21:26.47 | dijib | database add |
| 21:27.14 | kresp0 | the line i'm using instead of the LookupBlacklist one: |
| 21:27.15 | kresp0 | exten => 25,n,Gotoif($["${DB(blacklist/${CALLERID(num)})}" = "1"]?blacklisted,s,1) |
| 21:29.43 | p3nguin | i would probably use GotoIf($[${DB_EXISTS(blacklist/${CALLERID(num)})}]?wherever) to mimic the blacklist function. |
| 21:30.43 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 21:32.15 | kresp0 | thank you p3nguin, lets try it! |
| 21:34.52 | *** join/#asterisk Micc (~Micc@c-24-19-33-189.hsd1.wa.comcast.net) |
| 21:35.01 | kresp0 | p3nguin: it works ok, thanks! |
| 21:47.57 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 21:54.43 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:54.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:55.54 | pabelanger | I though LookupBlacklist was deprecated in 1.6, and even removed. |
| 21:57.56 | p3nguin | I'm not familar with it. I just use BLACKLIST(). |
| 21:59.32 | p3nguin | GotoIf($[${BLACKLIST()}]?someplace) |
| 21:59.58 | p3nguin | But if he doesn't have that, the DB_EXISTS() function can mimic the behavior. |
| 22:00.46 | *** join/#asterisk mocker (~mocker@72.165.148.230) |
| 22:02.04 | *** join/#asterisk philippel_mac (~p_lindhei@c-24-19-228-236.hsd1.wa.comcast.net) |
| 22:02.42 | philippel_mac | wondering if anyone knows what specific version the CLI format changed from "core set global ..." to "dialplan set global ..." ? |
| 22:03.26 | kresp0 | pabelanger, p3nguin: that DB_EXISTS solution its working like a charm, now im waiting for the telemarketers to call me ]:D |
| 22:04.18 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:04.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:04.36 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 22:05.02 | lauris | hi, can asterisk work under nat? |
| 22:05.08 | kresp0 | yes lauris |
| 22:05.23 | lauris | both server and clients at the same time? |
| 22:05.45 | kresp0 | lauris, take a look here: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:05.51 | lauris | thanks |
| 22:06.07 | WIMPy | lauris: Usually yes, but it depends on the involved devices. |
| 22:06.36 | lauris | basically situation is the following |
| 22:07.00 | lauris | i'm moving abroad and want to have access to my landline |
| 22:07.45 | WIMPy | If it is for yourself, I'd suggest using a tunnel. That way you avoid any NAT issues. |
| 22:08.07 | WIMPy | And it may be handy for other things as well. |
| 22:08.17 | lauris | i didn't thought about that :) thanks for idea |
| 22:08.29 | lauris | i already have pptp running on my router |
| 22:08.41 | lauris | problem closed :) |
| 22:10.47 | lauris | what version would you recommend me to install? |
| 22:10.54 | lauris | last time I played with asterisk it was 1.4.12 |
| 22:11.08 | lauris | or no. maybe some earlier versions |
| 22:13.31 | p3nguin | kresp0: What was your reason for not using the BLACKLIST() function? |
| 22:14.18 | p3nguin | lauris: The current LTS is 1.8.8.0. I suggest you start there. |
| 22:14.27 | *** join/#asterisk Gokee2 (~gokee2@204-195-14-78.wavecable.com) |
| 22:18.22 | lauris | does it use the same configuration syntax as in 1.4? |
| 22:18.45 | p3nguin | In all aspects, no. |
| 22:19.02 | WIMPy | For the most parts. See the UPGRADE*.txt |
| 22:19.05 | lauris | if talking only about clients and extensions |
| 22:19.07 | p3nguin | There are changes to sip.conf and several dial plan apps/functions, at least. |
| 22:19.08 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:19.13 | lauris | i don't need any special features |
| 22:19.22 | kresp0 | p3nguin: tought that BLACKLIST function was included in the LookupBlacklist app. i was wrong! |
| 22:19.40 | p3nguin | Extension syntax remains the same. exten => 123,1,App(data) |
| 22:19.42 | kresp0 | p3nguin: now i'm using the BLACKLIST() function |
| 22:19.47 | kresp0 | simpler, better :D |
| 22:20.16 | p3nguin | It isn't a hard change to go from 1.4 to 1.8 if you just pay attention. |
| 22:20.23 | lauris | WIMPy, using tunnel is really great idea! as it will allow to prevent hackers from scanning asterisk ;) |
| 22:20.31 | lauris | and trying to bruteforce password of a client |
| 22:21.05 | WIMPy | likes tunnels |
| 22:21.20 | lauris | with one exception |
| 22:21.35 | lauris | that user has to have some skills to use tunnel ;) |
| 22:22.21 | WIMPy | Not much. |
| 22:22.27 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 22:23.22 | lauris | can I run asterisk in openvz container? |
| 22:24.38 | kresp0 | yes lauris: http://wiki.openvz.org/Asterisk_in_container_with_Debian_stable |
| 22:25.10 | lauris | i don't need zaptel |
| 22:25.12 | lauris | have spa3102 for this purpose |
| 22:25.21 | lauris | but in general answer is yes, cool ;-) |
| 22:27.27 | *** join/#asterisk mocker (~mocker@72.165.148.230) |
| 22:27.31 | kresp0 | anyone here is routing calls to +883 (international networks country code)? |
| 22:27.42 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 22:27.52 | lauris | what country is that? |
| 22:27.57 | lauris | haven't seen such a prefix |
| 22:28.09 | kresp0 | is not a country, tought it has a country code |
| 22:28.34 | kresp0 | is like the +881 country code used for satellite phones |
| 22:28.44 | kresp0 | but for voip ("international networks") |
| 22:28.56 | p3nguin | I don't know what you mean by routing them, but I dial them and get calls to mine. |
| 22:29.06 | p3nguin | It's iNum. |
| 22:29.53 | kresp0 | p3nguin: how do you make calls to a inum # using * ? |
| 22:30.06 | p3nguin | dial it |
| 22:30.30 | carrar | Use the magic function |
| 22:30.57 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 22:31.13 | p3nguin | Depending on your ITSP, you may need to send 00 or 011 on the front of the number. |
| 22:31.34 | p3nguin | 00 883 1234 .... |
| 22:31.57 | p3nguin | And with that, I gotta run. |
| 22:32.10 | kresp0 | p3nguin: i'm my own itsp |
| 22:32.40 | kresp0 | tried with voipbuster but it doesnt work |
| 22:33.16 | kresp0 | also tried with my traditional pots provider and with my mobile provider, with no luck |
| 22:33.28 | carrar | Your own ISP |
| 22:33.37 | carrar | Being a Teir 1 is great |
| 22:33.55 | carrar | I AM THE INTERNET!!!!! |
| 22:34.03 | kresp0 | yes, you are carrar :) |
| 22:34.11 | WIMPy | You are Cisco? |
| 22:34.25 | carrar | Cisco is not a ISP |
| 22:34.39 | carrar | Maybe Sisco |
| 22:34.50 | WIMPy | No, but they are the internet. At least according to their TV ads some years ago. |
| 22:35.01 | carrar | Isn't that SUN? |
| 22:35.36 | carrar | err no, SUN is "The network is the computer" |
| 22:35.55 | carrar | Cisco is "Welcome to the human network" |
| 22:36.38 | carrar | "Juniper Your Net" |
| 22:36.40 | carrar | heh |
| 22:37.19 | carrar | Which brings us to Asterisk's slogan |
| 22:38.25 | kresp0 | just found a way to call to a +883 number using SIP: using a ENUM query i've got the sip address for that number: |
| 22:38.30 | kresp0 | http://enumquery.com/lookup?e164=883510000000094&source=form&query=%C2%BB%3E |
| 22:39.30 | carrar | "The Open Source PBX" |
| 22:40.09 | WIMPy | still can't really see a PBX in Asterisk. |
| 22:40.53 | carrar | Asterisk is a Polymer-bonded explosive |
| 22:41.40 | WIMPy | So does using Asterisk make you a terrorist then? |
| 22:43.00 | darkbasic | do someone know how to debug not working early b3? |
| 22:43.16 | WIMPy | What channeltype? |
| 22:43.22 | darkbasic | BRI for dahdi |
| 22:43.29 | darkbasic | (Sangoma) |
| 22:43.52 | WIMPy | Current libpri, dahdi and Asterisk? |
| 22:44.51 | darkbasic | libpri svn 2273 (Wed Aug 17 15:48:54 2011), dahdi 2.5.0.1, asterisk 1.8.7, wanpipe-3.5.23.2 |
| 22:45.21 | WIMPy | Should work, I think. |
| 22:45.40 | darkbasic | I keep hearing the ring tone |
| 22:45.51 | darkbasic | my telco said they do support early b3 |
| 22:46.13 | darkbasic | I do progress() followed by playback with the noaswer opetion |
| 22:46.19 | WIMPy | What's the whole call like. From where and to where are you calling? |
| 22:46.54 | darkbasic | I tried to call from my mobile phone and from my messagenet sip account to my ISDN BRI attached to a sangoma A500 |
| 22:47.00 | WIMPy | So the call is coming from the PST via BRI to your Asterisk and you want to send early media? |
| 22:47.33 | WIMPy | And you are sure you are allowed to SEND early media? |
| 22:47.46 | darkbasic | so said my telco... |
| 22:47.52 | WIMPy | Sounds unusual to me. |
| 22:48.36 | darkbasic | sounds inusual when something actually works :( |
| 22:48.52 | WIMPy | coughs |
| 22:49.43 | WIMPy | If they allow you to send early media, they allow yo to make toll free calls, even is only for short times. That's why I doubt they do. |
| 22:49.57 | WIMPy | s/is/if/ |
| 22:52.21 | darkbasic | a telco technician will call me tomorrow to try to track down the issue, I hope they aren't so stupid to tell me they support something they actually don't :| |
| 22:54.09 | WIMPy | They usually tell you what you want to hear. Facts don't matter. |
| 22:54.28 | WIMPy | Have you looked at d-channel trace? |
| 22:54.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:56.07 | darkbasic | no I didn't |
| 22:57.54 | tompaw | Evening guys, I need an idea. I have a dynamic queue with agents signing up and unpausing themselves using dialplan/web interface. The problem is, some of them do not sign themselves off before they shut down their voip phone. |
| 22:58.23 | tompaw | So I got a bunch of unpaused (active) entries in the queue that cannot be reached, because those people simply turned off their softphones. |
| 22:59.16 | tompaw | So the question is - how can I automatically check if the queue entries are live (they're just Local/1234 SIPs) and kick the ghosts? |
| 23:00.34 | darkbasic | WIMPy: Progress Description: Inband information or appropriate pattern now available. |
| 23:00.37 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 23:01.24 | WIMPy | darkbasic: So unless Asterisk plays something other than you told it, you know where the issue is. |
| 23:02.48 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-lcuecnvqqdynzchc) |
| 23:03.13 | tompaw | Is there a way for Asterisk to test an extension - to check if it's reachable? |
| 23:03.30 | tompaw | Because if yes, I can simply test all members of the queue and kick the unreachables out. |
| 23:03.48 | WIMPy | The queue is the extension. |
| 23:04.04 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 23:04.07 | WIMPy | And if you have qulify enabled, you can check the reachability of devices as well. |
| 23:04.53 | tompaw | Well, what I meant was - I have a queue with a bunch of Local/123456 agents. I want to write a script that checks if every agent is SIP-reacable. |
| 23:04.56 | tompaw | reachable. |
| 23:05.26 | WIMPy | How could a local channel be unreachable? |
| 23:05.38 | WIMPy | So you need to check the SIP peer. |
| 23:06.04 | tompaw | I cannot, because they're dynamic peers (pgsql - based, they don't appear in sip show peers). |
| 23:06.46 | tompaw | WIMPy: easily, an agent first REGISTERs - * checks the pgsql, then the agent sings in to the queue, but then he simply shuts down his sipphone. |
| 23:07.03 | tompaw | Therefore his extension is still registered as an agent, even though it's unreachable. |
| 23:09.35 | SeRi | waz up WIMPy |
| 23:10.03 | SeRi | working on an img for 1.8.8.0 here |
| 23:10.16 | WIMPy | Hi SeRi |
| 23:10.49 | SeRi | I see you been bussy |
| 23:11.16 | WIMPy | Not much, no. |
| 23:11.31 | SeRi | oh ok. |
| 23:12.41 | tompaw | rtcachefriends=yes << looks like that might help |
| 23:15.20 | tompaw | I don't really understand how rtautoclear works. |
| 23:18.07 | tompaw | OK, let's hope rtcachefriends + rtautoclear + qualifyfreq do the job... |
| 23:19.34 | tompaw | Whoa, one sec. |
| 23:20.23 | tompaw | regcontext! |
| 23:21.09 | tompaw | (I feel like talking to myself) |
| 23:21.12 | *** join/#asterisk nighty- (~nighty@69-165-220-105.dsl.teksavvy.com) |
| 23:29.40 | tompaw | Hm... no. |
| 23:30.04 | tompaw | Is it possible to trigger an action by someone (un)registering? |
| 23:30.37 | WIMPy | You can listen on AMI. |
| 23:32.30 | tompaw | Is there something like regcontext, which dials an extension instead of adding it? |
| 23:34.50 | tompaw | Listening on AMI... hm... |
| 23:34.55 | tompaw | never thought of it to be honest |
| 23:36.24 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 23:42.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:42.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 23:44.18 | tompaw | WIMPy: that's a good idea thanks. |
| 23:45.28 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 23:45.42 | SeRi | p3nguin: I need some help |
| 23:46.01 | *** join/#asterisk raden_work (~star@66-191-96-75.static.eucl.wi.charter.com) |
| 23:46.11 | raden_work | how is asterisk 10 been running ? |
| 23:46.17 | raden_work | how has ? |
| 23:47.06 | raden_work | where is everyone ? |
| 23:48.01 | leifmadsen | working great on my pbx for the last 4-5 months :) |
| 23:48.14 | raden_work | leifmadsen, no issues at all ? |
| 23:48.58 | leifmadsen | none for me |
| 23:48.59 | raden_work | Idiot electrician hooked 220 up to a 120 outlet in fuse panel .... trying to get things back up and running was on 1.6 |
| 23:49.14 | raden_work | Well I will give it a try |
| 23:51.41 | raden_work | leifmadsen, what OS do you generally favor ? |
| 23:52.49 | leifmadsen | yes |
| 23:52.55 | leifmadsen | Linux generally |
| 23:53.02 | raden_work | what distro ? |
| 23:53.15 | leifmadsen | CentOS and Ubuntu |
| 23:53.24 | leifmadsen | asterisk doesn't care |
| 23:55.03 | raden_work | i have lil bitty issues with suse at times |
| 23:55.14 | raden_work | was thinking about loading next box with ubuntu |
| 23:55.30 | raden_work | just hard to take jump , been on suse for like 10 years |
| 23:55.56 | SeRi | voip.ms is having issues |
| 23:57.21 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
