| 00:01.13 | Neptu | no application meetme for extension (home,9000,3) |
| 00:01.43 | ChannelZ | Is MeetMe loaded? module load app_meetme |
| 00:01.57 | *** part/#asterisk kl4m (~klamontag@gw2.noc1.sys-tech.net) |
| 00:02.04 | WIMPy | Before we get to the specific questions: What Version of Asterisk? |
| 00:06.15 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 00:08.10 | *** join/#asterisk s[X]_ (~mark@eth589.qld.adsl.internode.on.net) |
| 00:10.05 | Neptu | ChannelZ: gonna try |
| 00:11.01 | Neptu | ChannelZ: seems fails to load |
| 00:11.25 | ChannelZ | as WIMPy asks, what version of asterisk? |
| 00:11.34 | WIMPy | Does it exist? Do you have dahdi running? |
| 00:11.35 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 00:11.55 | *** join/#asterisk lal00 (~eduardo@201.151.40.230) |
| 00:12.13 | Neptu | core version?? |
| 00:12.36 | ChannelZ | core show version |
| 00:12.46 | Neptu | 1.8.7.1 |
| 00:12.57 | ChannelZ | hmm should be good .. |
| 00:13.11 | Neptu | I did loadmodule and was screaming |
| 00:13.14 | ChannelZ | what about module show like timer |
| 00:13.32 | WIMPy | You could use (the old) ConfBridge instead of MeetMe. |
| 00:13.43 | ChannelZ | ConfBridge is new, MeetMe is old. |
| 00:14.06 | ChannelZ | (albeit even newer and shinier in 10) |
| 00:14.28 | WIMPy | Yes, it's not compatible with the 1.8 version. |
| 00:14.33 | Neptu | res_timing_timerfd.so |
| 00:14.35 | WIMPy | That's wy I said old. |
| 00:14.49 | Neptu | ok so how I do the conference room with version 1.8? |
| 00:14.53 | ChannelZ | Well timerfd should work as a timing source for meetme regardless |
| 00:15.04 | ChannelZ | What error do you get when you try to load app_meetme? |
| 00:15.12 | Neptu | w8 |
| 00:15.21 | WIMPy | And who is to do the mixing? |
| 00:16.07 | Neptu | http://pastebin.com/jgWhScYJ |
| 00:16.10 | ChannelZ | it's kind of built-in now |
| 00:16.28 | WIMPy | For MeetMe? Since when? |
| 00:16.33 | ChannelZ | Did you compile Asterisk yourself or is this a package? |
| 00:16.56 | Neptu | as a package for gentoo |
| 00:17.06 | Neptu | maybe I need some extension? |
| 00:17.23 | ChannelZ | maybe, no idea how they busted it up |
| 00:17.33 | lal00 | In PBX in a Flash, how can I make a DID Goto a specific context? |
| 00:17.36 | Neptu | dalhi? |
| 00:18.10 | WIMPy | Might be easiest to use ConfBridge. |
| 00:18.45 | Neptu | WIMPy: can u explain how to do it? |
| 00:19.07 | WIMPy | 'core show application ConfBridge' |
| 00:19.48 | ChannelZ | My mistake, yeah you do need DAHDI running it's just dahdi_dummy which is more or less obsolete. Fuzzy brain |
| 00:20.29 | ChannelZ | Which is probably why meetme is missing on his system, they maybe bundled it with the DAHDI package |
| 00:20.46 | WIMPy | Would make sense. |
| 00:20.59 | Neptu | ChannelZ: gonna try to recompile |
| 00:21.02 | Neptu | w8 |
| 00:21.27 | WIMPy | Oh, right. That's source packages. |
| 00:21.56 | Neptu | another question |
| 00:22.22 | ChannelZ | Neptu: for a basic conference ConfBridge works just as well without having to jump through additional hoops |
| 00:22.25 | Neptu | I do in the account disallow=all allow=g722 (trying to get a better sound quality codec) |
| 00:23.00 | WIMPy | MeetMe won't do G.722. You need ConfBridge for that. |
| 00:23.06 | Neptu | ChannelZ: I will take a look |
| 00:23.23 | Neptu | WIMPy: seems nothing works with g722 |
| 00:23.42 | ChannelZ | Won't it just transcode to/from ulaw (or really 'slin') |
| 00:23.43 | Neptu | I whould like to increase sound quality but does not really seems to be working |
| 00:24.04 | WIMPy | Yes, it should transcode. |
| 00:24.15 | Neptu | mmmmmm |
| 00:24.22 | ChannelZ | (you just get no benefit of a wideband codec) |
| 00:24.38 | Neptu | ok |
| 00:24.56 | Neptu | I tried with saydigits and did not worked was complaining about the codec |
| 00:24.58 | WIMPy | Probably not noticible, but it will probably be worse. |
| 00:24.59 | Neptu | ... |
| 00:25.11 | Neptu | ok |
| 00:25.24 | Neptu | so ulaw is already quite good quality then |
| 00:25.46 | WIMPy | The samples aren't available in G.722, either, but they too should ge transcoded. |
| 00:26.02 | WIMPy | that's the normal telephone quality. |
| 00:26.10 | Neptu | so no benefit, maybe some more load to the cpu |
| 00:26.29 | Neptu | WIMPy: there is any other codec taht gives me better quality? |
| 00:26.35 | Neptu | any recomendation? |
| 00:26.36 | ChannelZ | well g.722 can use lets bits |
| 00:26.39 | WIMPy | Not with playing any of the original spmples. |
| 00:27.03 | WIMPy | G.722 in a conference could make some difference. |
| 00:27.23 | Neptu | ok |
| 00:27.42 | Neptu | maybe I need to enconde my own samples in g722 |
| 00:27.49 | ChannelZ | ding! |
| 00:28.01 | ChannelZ | MOH and all that |
| 00:28.09 | Neptu | ding!? |
| 00:28.11 | ChannelZ | though I think Asterisk 10 has much better wideband support |
| 00:28.24 | ChannelZ | In general anyway |
| 00:28.32 | ChannelZ | ding = you are correct sir |
| 00:28.44 | Neptu | ok |
| 00:28.46 | WIMPy | Yes, you can have full audio-DVD quality conferences now. |
| 00:28.56 | ChannelZ | Suck it, Skype! |
| 00:29.00 | Neptu | hehehe |
| 00:29.08 | Neptu | meaning 1.10 version?? |
| 00:29.20 | WIMPy | 10.0 |
| 00:30.22 | Neptu | ok I see Im somehow behing with 1.8 |
| 00:30.37 | Neptu | ... |
| 00:30.40 | ChannelZ | 10 just came out like 5 days ago |
| 00:30.50 | ChannelZ | It's not like you're really that far behind or anything |
| 00:31.00 | Neptu | ok |
| 00:31.11 | Neptu | just wanted to get better quality |
| 00:31.18 | Neptu | that's all |
| 00:31.22 | ChannelZ | (there is no 2.0, 3.0, etc.. we went from 1.8 to 10.0) |
| 00:31.40 | Neptu | Yes I saw that on the web just now |
| 00:31.57 | *** join/#asterisk prologic (~prologic@unaffiliated/prologic) |
| 00:32.01 | ChannelZ | Wideband should work just fine in 1.8 assuming everything in your path supports it |
| 00:32.07 | Neptu | compiling dalhdi with one core takes a while |
| 00:32.39 | ChannelZ | there wasn't a package for it? (if you're using a packaged Asterisk, might as well use a packaged DAHDI already built too..) |
| 00:32.41 | prologic | Hey all. For some reason if I try to set an Outbound Router's CID to say 214, making an outbound calls I get 403 Forbidden from the pbx |
| 00:32.43 | Neptu | The major problem will be the samples I do not feel like pend the time to recode them |
| 00:33.23 | Neptu | ChannelZ: in gentoo u need to add the use to add it as part of the compiling process... just added the "use" and recompiled |
| 00:33.27 | Neptu | will be ready soon |
| 00:33.29 | ChannelZ | then you'll just have 'normal' quality standard IVR. Just make sure you install the ulaw sound set and don't just have gsm or something crap |
| 00:34.01 | Neptu | I added the allow ulaw so I guess that will do |
| 00:34.35 | ChannelZ | prologic: From what PBX? You obviously have more than just an Asterisk setup we probably cant diagnose by guessing |
| 00:35.02 | ChannelZ | Neptu: well that's just what codecs are allowed, I'm talking about the actual sound file set on disk |
| 00:35.30 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 00:35.32 | Neptu | ChannelZ: I added all of them I think... just to try |
| 00:35.37 | prologic | ChannelZ, it's hard to say really. but from the pbx I'm dialing out from. I get "[2011-12-20 19:02:12] WARNING[4958] chan_sip.c: Received response: "Forbidden" from '"214" <sip:214@172.18.42.8>;tag=as5be6ccdc'" |
| 00:35.43 | ChannelZ | If you only installed the GSM set for instance (which sounds like poo) it doesn't matter what codecs your phones are capable of, it will always sound like poo |
| 00:35.57 | prologic | my only guess is, it's not liking the CID I'm presenting |
| 00:36.10 | ChannelZ | prologic: yes but WHO is the response coming from? An ITSP? |
| 00:36.25 | ChannelZ | (and if so probably yes they don't like your CID) |
| 00:36.25 | prologic | it's coming from asterisk I think |
| 00:37.02 | Neptu | ChannelZ: net-misc/asterisk-core-sounds-1.4.19 USE="alaw g722 g729 gsm siren14 siren7 sln16 ulaw wav" |
| 00:37.07 | ChannelZ | You need to describe your setup. The warning you pasted is from Asterisk, but it's talking about a response it received from somewhere else (that you haven't told us) |
| 00:37.08 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 00:37.50 | WIMPy | Since when does Asterisk tell you about the responses it receives? |
| 00:38.15 | ChannelZ | well the slin16 are of much higer quality for wideband use, assuming it happens to pick those files to transcode them to 722 or whatever |
| 00:38.20 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 00:38.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 00:39.08 | *** join/#asterisk nix8n82 (~hmg@75-174-156-77.chyn.qwest.net) |
| 00:39.58 | Neptu | ChannelZ: ok so Maybe I can try sln16... :) why not |
| 00:40.19 | SeRi | p3nguin: found the issue |
| 00:40.30 | ChannelZ | WIMPy: it will tell you a lot of things if you ask it to. :) |
| 00:40.38 | ChannelZ | heads for home |
| 00:41.11 | WIMPy | The only way to get responses seems to be sip debug, wich is a real PITA. |
| 00:42.09 | Neptu | NOTICE[10484]: chan_sip.c:8901 process_sdp: No compatible codecs, not accepting this offer! |
| 00:42.20 | Neptu | used allow=slin16... |
| 00:42.30 | ChannelZ | Again, you're only limiting the codecs the DEVICE can use |
| 00:42.46 | Neptu | softphone? |
| 00:42.52 | ChannelZ | And I doubt a lot of hardware phones know what slin16 is |
| 00:43.14 | ChannelZ | You can have your device use g.722 (if they indeed support it) but I am talking about using slin16 as a sound source that * will transcode to g.722 |
| 00:43.18 | WIMPy | I'd gues round about none. |
| 00:43.39 | ChannelZ | although, it seems like there IS a g.722 sound set? hmm |
| 00:44.28 | Neptu | aha 3cx on android only allow ulaw alaw gsm and speex |
| 00:45.44 | ChannelZ | the 'slin' formats are really more of local raw formats that get encoded to something else |
| 00:45.58 | *** join/#asterisk fipu (~fipu@fipu.usr.sh) |
| 00:46.21 | ChannelZ | But in any event if that's the only phone you're testing with, it's a bit pointless |
| 00:46.41 | ChannelZ | Although I guess speex can be wideband. |
| 00:47.07 | Neptu | speex sounds a bit better yes, just tested it... |
| 00:47.34 | p3nguin | seri: I just got my D-Link router today, and I've already ordered an RP-TNC and an RP-SMA (because I'm not sure which one will fit the best) to convert it over to a removable antenna. |
| 00:48.39 | *** join/#asterisk l0uis (~l0uis@madmax.fitnr.com) |
| 00:48.51 | Neptu | WARNING[27401]: loader.c:829 load_resource: Module 'app_meetme' already exists. |
| 00:49.00 | Neptu | now seems the app_meetme is there... |
| 00:49.14 | *** part/#asterisk l0uis (~l0uis@madmax.fitnr.com) |
| 00:49.36 | *** join/#asterisk ltd (~z@zx.io) |
| 00:50.27 | Neptu | http://pastebin.com/qnQNPDtt mmmmmmm now i get this evil error |
| 00:52.41 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 00:54.14 | Neptu | ChannelZ: http://pastebin.com/ZR4nbXZC |
| 00:55.47 | Neptu | I donno why the room exits with 0 |
| 00:57.16 | WIMPy | You don't have dahdi running. |
| 00:57.54 | WIMPy | modprobe dahdi |
| 00:58.14 | Neptu | chech the pastebin under that one |
| 00:58.28 | Neptu | seems i needed to do 9000@rooms is a context problem |
| 00:58.42 | Neptu | butn then it exits the conference |
| 01:01.30 | Neptu | mmm module not found |
| 01:03.45 | WIMPy | Hmm. Without installed dahdi you wouldn't have been able to compile app_meetme. So I guess something must have gon wrong on dhdi installation. |
| 01:03.51 | Neptu | http://pastebin.com/Gj8gd9Vn |
| 01:03.59 | Neptu | sorry seems is thre |
| 01:04.28 | Neptu | :) |
| 01:04.43 | Neptu | but conference is exiting real quick |
| 01:05.03 | WIMPy | Is the kernel module loaded? |
| 01:05.23 | Neptu | http://pastebin.com/ZR4nbXZC |
| 01:05.29 | Neptu | looking |
| 01:07.15 | Neptu | asterisk # lsmod |grep dah |
| 01:07.17 | Neptu | dahdi 187536 0 |
| 01:07.19 | Neptu | crc_ccitt 1192 1 dahdi |
| 01:07.29 | Neptu | seems is there or I just loadead it |
| 01:07.37 | Neptu | not really sure... fast hands |
| 01:08.46 | Neptu | makes no difference I exit the conference room real quick |
| 01:08.50 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 01:09.01 | Neptu | maybe need to add some parameters to meetme? |
| 01:09.37 | WIMPy | should all be optional. |
| 01:10.43 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:11.02 | paulc | If I Gosub an extension, then in there Goto somewhere else instead of using Return, does it clear the stack? |
| 01:11.07 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 01:12.03 | Neptu | WIMPy: then I do not know why I cannot not stay in the conference room and crashes... how was the code to see a deep debug? |
| 01:14.37 | Neptu | http://pastebin.com/6QgyKJGk |
| 01:14.50 | Neptu | not finding any problem with the debug... |
| 01:17.06 | Neptu | dammi seems its a bye bit comming... |
| 01:17.20 | Neptu | that might be the reason... but now this becomes weird |
| 01:20.06 | Neptu | dahid |
| 01:29.38 | SeRi | p3nguin: Mine arrives friday! :D |
| 01:30.34 | SeRi | My phone should be here tomorrow or Thursday. the guy said he has not been able to ship it :/ |
| 01:31.14 | p3nguin | I fixed the patch that was broken in asterisk-1.8.8.0-1.src.tar.gz if you're ready to give it a try. |
| 01:31.32 | p3nguin | He's going to be refunding the $25 for next day shipping, right? |
| 01:32.07 | p3nguin | I have not tested the software, but I did test the build of the package. |
| 01:32.33 | SeRi | Sure ill give it a go. my sys tem is down right now after a nasty storm last night |
| 01:32.42 | SeRi | system* |
| 01:32.59 | SeRi | p3nguin: Yes he will refund me the shipping money |
| 01:33.13 | SeRi | He has not reply to me though |
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| 01:46.04 | SeRi | p3nguin: the jabber error coems from users that sign in via web |
| 01:46.08 | SeRi | :) |
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| 02:04.41 | macroevolve | hi guys, I have a somewhat off-topic question i was trying to get an snwer to. was wondering if anyone here might know an answer. |
| 02:05.27 | lauris | you should focus more on your question :) |
| 02:05.48 | macroevolve | I had just brought back two colocated servers from los angeles and am planning on running them at home using my comcast home isp. I currently have one router with four ports. Three of these four ports are being used by my personal computers, and i only have one port left to do a wired connection to one of the servers. I was wondering whether it would be ok to use the remaining open port |
| 02:05.48 | macroevolve | from my router and then connect it to a regular network switch and use that network switch sort of like an extension to buy me a few more ports so that I can plug my two servers directly to that switch? |
| 02:06.01 | macroevolve | sorry i don't have the quickest hands |
| 02:06.16 | woleium | yes |
| 02:06.41 | lauris | macroevolve, yes, you can do that |
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| 02:09.27 | woleium | the long answer: yes to a point. as long as you don;t put more than 100 machines (rough guess, as it varies) on a port you'll not overflow the cheap home router's CAM table and you'll be OK :-) |
| 02:10.00 | cyford | i would hook my servers up too the router instead and hook my computers up to the switch |
| 02:10.42 | woleium | makes little difference - aa chain is only as good as it's weakest link and your weakest link is router-internet not swaths -router |
| 02:10.53 | woleium | s/swaths/switch |
| 02:11.11 | woleium | unless of course you have a 100mb connection to the internet :-) |
| 02:11.41 | cyford | no... it it is less points of failure |
| 02:12.26 | cyford | router goes out everything goes... switch goes out .. and your away what do you want to run, serverd or computers |
| 02:15.28 | woleium | what is the make and model of your router modem box? |
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| 02:54.39 | p3nguin | seri: Good to know, I guess. I don't have my chat enabled, so that is probably why I do not see the messages. |
| 02:55.10 | SeRi | :) |
| 02:55.16 | SeRi | p3nguin: bussy? |
| 02:55.25 | p3nguin | Not really. |
| 02:55.42 | SeRi | p3nguin: If you ever see this: Got presence packet from 0zhgn7v4q8sm83a262ghs3l8fg@public.talk.google.com, someone not in our roster!!!! |
| 02:55.55 | SeRi | That means you are logged in twice from an asterisk server |
| 02:56.01 | p3nguin | That's got to do with the auto register, I think. |
| 02:56.33 | p3nguin | How do you log in the same account more than once from the same asterisk? |
| 02:57.04 | SeRi | I copy over my conf files from one asterisk over and for got to disable jabber in one account :) |
| 02:57.25 | SeRi | so I had to asterisk setup with presence of 25 |
| 02:57.28 | SeRi | two* |
| 02:57.30 | p3nguin | You meant two asterisks. |
| 02:57.35 | SeRi | yes sr. |
| 02:57.46 | p3nguin | I thought you meant two times from the same one asterisk. |
| 02:57.58 | p3nguin | I didn't see that happening by accident. |
| 02:57.59 | SeRi | oh no :) |
| 02:58.02 | SeRi | sorry |
| 03:04.57 | SeRi | we had a massive TS come in this morning somewhere around 1:30AM and power went out..... my atom machine went down everything else I caught on time... |
| 03:05.52 | p3nguin | Why aren't you using UPS? |
| 03:06.24 | SeRi | I am for my systems. but I have the atom running from its own battery since is a netbook |
| 03:06.56 | p3nguin | You should be able to tell it to shut down in the event of a power outage. |
| 03:07.05 | woleium | heh, so it's not just me that recommends using old laptops for * servers |
| 03:07.46 | SeRi | p3nguin: didnt thought about. Thanks for the idea |
| 03:07.46 | woleium | dead screen laptops from ebay - so cheap, built in UPS! |
| 03:09.31 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 03:10.31 | p3nguin | If your laptop and OS don't have some way to communicate the loss of power and shut down, you should be able to use snmp or something from one of the other systems on an UPS to tell it to shut down, as long as the interconnect is also on backup. |
| 03:10.46 | p3nguin | I assume your switches are on backup, too. |
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| 03:14.55 | SeRi | p3nguin: yes everything is on backup. I have a nut server but I have not had time to test it. |
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| 04:12.21 | SeRi | any body used adtran before? |
| 04:16.56 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 04:24.38 | [TK]D-Fender | SeRi: No |
| 04:28.07 | SeRi | [TK]D-Fender: I sort of figure that :P |
| 04:28.18 | SeRi | hehehe..... I was looking at it.... |
| 04:28.25 | SeRi | Interesting equipment |
| 04:30.29 | p3nguin | It? |
| 04:30.42 | SeRi | ? |
| 04:30.51 | p3nguin | "I was looking at it." |
| 04:31.02 | p3nguin | You were looking at the company? |
| 04:31.23 | SeRi | at the equipment/company |
| 04:31.45 | SeRi | I stumble accross them cross referencing data |
| 04:31.54 | p3nguin | What kind of equipment are you interested in? |
| 04:32.36 | SeRi | I am not interested. I was looking at faxing solutions over voip and came across adtran |
| 04:33.11 | SeRi | analog to voip |
| 04:33.25 | [TK]D-Fender | SeRi: they make phones, switches, routers, VoIP servers, and possibly kitchenware. |
| 04:33.31 | SeRi | lmao |
| 04:33.37 | [TK]D-Fender | serSo please don't ask if we've tried ... "a company: |
| 04:34.30 | SeRi | is that bad to ask if somebody had experience with gear? |
| 04:34.57 | p3nguin | I have an Adtran duck press. |
| 04:35.55 | SeRi | lol I see what you guys mean now. |
| 04:40.17 | SeRi | :/ |
| 04:40.55 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:42.20 | p3nguin | http://imagebin.org/index.php?mode=image&id=189639 |
| 04:42.27 | p3nguin | There are a couple Adtrans in this pic. |
| 04:42.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 04:43.19 | SeRi | I get your point. |
| 04:43.51 | [TK]D-Fender | <PROTECTED> |
| 04:44.32 | p3nguin | The PBX is Nortel stuff. |
| 04:44.34 | [TK]D-Fender | p3nguin: with the smaller VM module and I think an ATA (or was that a CID module?) |
| 04:45.05 | [TK]D-Fender | p3nguin: I had one before I migrated my company to * 6 years ago |
| 04:45.16 | p3nguin | The smaller boxes with the white labels all over them are Adtrans for the T1s. |
| 04:46.05 | SeRi | just curious does adtran voice equipment excluding phones or ata are geared to T1's? |
| 04:46.46 | [TK]D-Fender | SeRi: The have all kinds of products. EVERYTHING. Analog, digital, LAN, WAN, wireless, VOIP. |
| 04:47.37 | [TK]D-Fender | SeRi: Just like Cisco. Large telecom / networking product lines |
| 04:47.50 | [TK]D-Fender | SeRi: They aren't a "1-trick-pony" |
| 04:48.33 | SeRi | :/ ok..... |
| 04:50.25 | p3nguin | In addition to this wall of goodies, they also have a nice channel bank. http://imagebin.org/index.php?mode=image&id=189640 |
| 04:52.01 | SeRi | p3nguin: Is that one of your jobs? |
| 04:52.36 | p3nguin | Yes, Ameren Illinois power company. |
| 04:52.48 | SeRi | looks like fune. |
| 04:52.51 | SeRi | cool. |
| 04:57.14 | p3nguin | I don't think the cell phone tower has anything in it that you'd recognize, or I'd share some of those, too. |
| 04:59.55 | p3nguin | You might recognize the Cisco router and the USR modem attached to it, but that's about it. |
| 05:00.10 | p3nguin | The rest is all RF equipment and batteries. |
| 05:01.03 | SeRi | cool. |
| 05:01.09 | ChannelZ | Hrmph. I need some torrents from some old Bob Barker-era 80s The Price Is Right episodes |
| 05:01.10 | p3nguin | I love how the entire place runs on DC. |
| 05:02.06 | p3nguin | That was the first place I ever encountered a battery powered Cisco router. |
| 05:02.28 | SeRi | interesting |
| 05:03.10 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 05:04.59 | SeRi | well my failover setup defently works. It was properly tested in a live situation all morning and threw out the day. nice to know. |
| 05:05.32 | SeRi | s/defently/definitely/ |
| 05:05.40 | p3nguin | tyco powar! http://imagebin.org/index.php?mode=image&id=189643 http://imagebin.org/index.php?mode=image&id=189644 |
| 05:06.31 | SeRi | nice |
| 05:06.35 | SeRi | kneat |
| 05:08.49 | p3nguin | battery powered router http://imagebin.org/index.php?mode=image&id=189645 |
| 05:10.22 | SeRi | Interesting. |
| 05:10.28 | SeRi | never seen one battery powered |
| 05:11.50 | SeRi | res_jabber.c:2291 aji_handle_presence: Got presence packet from 0zhgn7v4q8sm83a262ghs3l8fg@public.talk.google.com, someone not in our roster!!!! <----- thats getting to be annoying. |
| 05:14.25 | p3nguin | Turn off chat in gmail? |
| 05:14.46 | SeRi | I did. |
| 05:14.59 | SeRi | I even disable it on my cell. |
| 05:15.10 | SeRi | looks like it lingers on the system some how |
| 05:15.35 | p3nguin | I don't think I've ever seen that message. |
| 05:17.40 | SeRi | I am sure ill get use to it... I hope :/ |
| 05:20.54 | SeRi | voip.ms told me that their fail over system does not wait for a system not to be register to failover to a different number... If the "server" is unreachable they will fail over the call. |
| 05:21.10 | SeRi | good to know |
| 05:21.35 | p3nguin | That's the typical cause of being unreachable, though. |
| 05:22.28 | SeRi | I would thought that the system seen you online and register would not fail over the call. but the system trys to send the call to you. if it can not deliver it will fail over |
| 05:23.03 | SeRi | regardles of you been register or not. so I guess I misunderstood how it worked |
| 05:24.12 | p3nguin | If your registration goes away, you're unreachable. |
| 05:24.21 | p3nguin | But there are other reasons to not be reachable. |
| 05:25.04 | p3nguin | Think of the "sequential dial" concept for the fail-over. |
| 05:25.43 | SeRi | I see. |
| 05:31.56 | p3nguin | If the channel is not available, dial something else. |
| 05:35.46 | *** join/#asterisk Kroms (~Kroms@rrcs-173-197-77-200.west.biz.rr.com) |
| 05:35.48 | Kroms | hi |
| 05:41.39 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
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| 05:54.10 | SeRi | I autoprune=yes and all the flodding stop. I dont use any type of chat so to hell with them |
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| 06:14.26 | SeRi | well shit |
| 06:14.35 | p3nguin | pewps |
| 06:14.51 | SeRi | still there |
| 06:15.31 | SeRi | cant get rid of it. I wonder if they have to remove me from there google chat |
| 06:15.37 | SeRi | their* |
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| 07:49.07 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:49.10 | schmidts | good morning |
| 07:50.13 | Yourname` | ax1600p: FATAL: Module ax1600p not found. -> How can I fix this? |
| 07:52.13 | schmidts | Yourname` module load ax1600p ? |
| 07:56.24 | Yourname` | That error comes when I type service dahdi start |
| 07:57.03 | schmidts | Yourname` check your dmesg if you see something hardware related |
| 07:57.47 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 07:58.01 | Yourname` | dahdi: Telephony Interface Registered on major 196 |
| 07:58.02 | Yourname` | dahdi: Version: 2.5.0.2 |
| 07:58.02 | Yourname` | rxt1: no version for "dahdi_hdlc_putbuf" found: kernel tainted. |
| 07:58.02 | Yourname` | rxt1: Unknown symbol dahdi_transmit |
| 07:58.02 | Yourname` | rxt1: Unknown symbol dahdi_receive |
| 07:58.05 | Yourname` | and more... |
| 07:58.13 | Yourname` | I mean more of those unknown symbols.. |
| 08:09.06 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 08:12.06 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:18.47 | kaldemar | Yourname`: install ax1600p driver module. |
| 08:20.00 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 08:21.20 | kaldemar | Yourname`: your init script is trying to load a module by that name. it is not a part of DAHDI. if you don't have a AX1600P card in your system, remove the attempt to load the module from /etc/dahdi/modules or your init script. |
| 08:21.56 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-byhpnsnrsrdcybhq) |
| 08:21.58 | IsUp | hello |
| 08:25.01 | *** join/#asterisk ollii (~risker@vpn.nbank.de) |
| 08:25.43 | Yourname` | kaldemar: I do have a card AX1600P tho |
| 08:26.24 | WIMPy | I guess the module version doesn't fit your dahdi version. |
| 08:26.36 | Yourname` | ah |
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| 08:32.59 | *** join/#asterisk nicola_pav (~chatzilla@62.219.238.196) |
| 08:34.17 | nicola_pav | hello. I want to use Fax For Asterisk (FFA) in order to send faxes. anyone familiar with a client that maybe runs as a module printer? |
| 08:35.31 | Yourname` | I think, kaldemar, the card is not being detected now. :S |
| 08:35.43 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:37.28 | kaldemar | Yourname`: yes, DAHDI sure does not detect it if you don't have a working module for it. |
| 08:38.03 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:39.33 | Yourname` | kaldemar: Is there a way to uninstall all dahdi on the system and start fresh? |
| 08:40.39 | kaldemar | Yourname`: yes, depending on the way you installed it. |
| 08:41.10 | kaldemar | Yourname`: if you installed from source, "make uninstall" |
| 08:42.55 | Yourname` | kaldemar: Gonna give it a try now.. |
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| 08:56.23 | *** join/#asterisk shido6_ (~shido6@nat/yahoo/x-akruxftkwzvvgdge) |
| 08:57.31 | Yourname` | Is there anything like Elastix, but WAY easier to run as on GUI? And gets stuff like dahdi, etc running easy? |
| 08:58.05 | *** join/#asterisk Nasga (~Nasga@112.4.118.78.rev.sfr.net) |
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| 09:27.46 | flyankur | Is there a php-agi command to execute a asterisk command line and return its output |
| 09:28.00 | flyankur | I want see all the channels running in phpAGI file, using exec(" /usr/sbin/asterisk -rx 'core show channels'") ,but php's exec does not seems to work in it. Thanks |
| 09:28.20 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:29.42 | Nasga | flyankur: try with $output = array(); exec(" /usr/sbin/asterisk -rx 'core show channels'",$output); var_dump($output); |
| 09:39.04 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 09:57.59 | petern_ | hmm, need to test moh :S |
| 09:59.37 | ChannelZ | sounds easy enough |
| 10:00.25 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 10:01.41 | petern_ | file.c:650 ast_openstream_full: File /usr/share/asterisk/moh/fpm-calm-river.wav does not exist in any format |
| 10:01.46 | petern_ | /usr/share/asterisk/moh/fpm-calm-river.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 10:01.49 | petern_ | i see :S |
| 10:02.16 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 10:02.34 | ChannelZ | does it have proper file permissions such that Asterisk can read it? |
| 10:03.04 | petern_ | yes |
| 10:03.13 | petern_ | i fixed it, it was confused |
| 10:06.44 | petern_ | ("moh reload" doesn't necessarily give the same state as restarting asterisk) |
| 10:08.14 | WIMPy | From my experience you have to unload and load res_musiconhold. |
| 10:12.41 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 10:14.53 | ChannelZ | yeah it doesn't notice files on a reload |
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| 11:07.53 | MariusAgon | Hello. Is there any way to have multiple position announcements for different queues (different languages)? |
| 11:17.01 | schmidts | MariusAgon IMHO you can use the Setlanguage function but i am not sure about this, and different announcements should work when you set them in the queues conf file |
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| 12:00.18 | *** join/#asterisk Yourname`` (~Yourname@121.97.255.134) |
| 12:00.42 | Yourname`` | What does it mean when I try to stop dahdi service, it tells me modules are in use echo in use, etc.. how can I stop it clean? |
| 12:01.39 | IsUp | Yourname``: your asterisk is working, stop it first |
| 12:01.57 | Yourname`` | IsUp: ah! |
| 12:02.33 | Yourname`` | IsUp: Another thing. I want loadzone to change, so I run dahdi_genconf.. but it generates chan_dahdi.conf as just with loadzone.. |
| 12:02.51 | Yourname`` | IsUp: AND it says Empty configuration -- no spans |
| 12:04.01 | Yourname`` | So, I change /etc/dahdi/genconf_parameters and add the loadzone, and run dahdi_genconf.. but again, a) loadzone still shows us b) loadzone is the only one in existence |
| 12:05.18 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 12:05.56 | *** join/#asterisk ffon (~ffon@host-2-100-62-127.as13285.net) |
| 12:07.34 | ffon | Hi, I'm having a few issues with chan_iax and I can't figure out what has happened, this was working fine, although as of yesterday inbound calls work, but outbound comes up with an error |
| 12:11.26 | ffon | After adding qualify=1000 I'm getting "__ast_request_and_dial: Unable to request channel IAX2/financesolvers/447708", prior to this I got a "Auto-congesting call due to slow response". The iax trunk is registered but I can't figure out a way around this. |
| 12:15.32 | IsUp | ffon: do 'iax2 show peers', what you see in your peer, OK (xxx ms)? |
| 12:16.44 | ffon | can't remember now, I've just lost connection to the server |
| 12:16.59 | IsUp | Yourname``: where you set your loadzone in "genconf_parameters"? |
| 12:17.17 | Yourname`` | IsUp: third line.. |
| 12:17.36 | Yourname`` | IsUp: It had only two lines, and I added the loadzone on the 3rd line |
| 12:17.49 | IsUp | Yourname``: in genconf_parameters, you should set "lc_country" |
| 12:18.22 | Yourname`` | IsUp: lc_country and no loadzone? |
| 12:19.09 | ffon | IsUp: iax show registry shows registered, yet iax show peers shows unreachable |
| 12:19.32 | IsUp | Yourname``: just lc_country, and run dahdi_genconf then see loadzone in /etc/dahdi/system.conf |
| 12:20.13 | ffon | IsUp: whilst at the other end shows 40ms |
| 12:20.18 | Yourname`` | IsUp: I guess that's why genconf was creating an empty system.conf cuz it didn't reconize "loadzone" and is set to recognize "lc_country". So this time, it will do full config AND add "loadzone", right? |
| 12:20.33 | IsUp | ffon: set qualify to 5000, reload and see whats going on. i think you have problems with your provider/peer |
| 12:20.52 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 12:21.23 | IsUp | Yourname``: yes, but first of all, please do 'dahdi_hardware' to test your hardware. are you able to run your card here? check it first |
| 12:21.43 | Yourname`` | IsUp: Yup. It shows up. |
| 12:21.50 | ffon | IsUp: sorry, it's already set as 5000 |
| 12:22.03 | IsUp | Yourname``: great, run dahdi_genconf |
| 12:22.39 | Yourname`` | Thing is, hangup isn't detected well so I know loadzone will help it. But when I edited genconfparams and added loadzone, it broke it by adding only 2-3 lines to the conf. Gonna try it now with your suggestion |
| 12:22.41 | IsUp | ffon: okay, seems like a firewall/connection problem to me. you have to talk with your provider. |
| 12:23.43 | Yourname`` | IsUp: Ugh. Did it, but again, empty configuration error and now the conf file is again empty except for loadzone |
| 12:24.12 | ffon | IsUp: as far as I'm aware nothing has changed there but I shall take a look at the router, thanks |
| 12:24.21 | IsUp | ffon: good luck |
| 12:24.54 | IsUp | Yourname``: pastebin your current genconf_parameters, system.conf and chan_dahdi.conf file please |
| 12:25.01 | Yourname`` | one sec |
| 12:26.54 | Yourname`` | IsUp: http://pastebin.com/F57aMkyF |
| 12:28.14 | Yourname`` | PS: I want to enable hanguponpolarity.. |
| 12:28.49 | IsUp | Yourname``: okay please do 'dahdi_scan' and pastebin output |
| 12:28.58 | Yourname`` | one sec |
| 12:29.10 | Yourname`` | isup: Unable to open /dev/dahdi/ctl: No such file or directory |
| 12:29.44 | IsUp | Yourname``: do you have DAHDI init script? try '/etc/init.d/dahdi start' first. |
| 12:30.29 | Yourname`` | isup: http://pastebin.com/KA8nrPAL |
| 12:30.58 | IsUp | Yourname``: ok great now run dahdi_scan and pastebin output. |
| 12:31.29 | Yourname`` | isup: http://pastebin.com/bunwA0pi |
| 12:32.23 | IsUp | Yourname``: ok now run dahdi_genconf and see your system.conf again |
| 12:32.34 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 12:32.42 | Yourname`` | isup: wtf |
| 12:32.44 | Yourname`` | lol |
| 12:32.49 | Yourname`` | it's clean and done now |
| 12:32.53 | Yourname`` | What was wrong isup/ |
| 12:33.41 | IsUp | Yourname``: if DAHDI is not started, it doesnt recognize modules, so you dahdi_genconf cant generate spans |
| 12:34.38 | Yourname`` | IsUp: ah! duh! now I have one way audio on outgoing call :/ |
| 12:38.19 | Yourname`` | IsUp: On default configuration, audio is just great. But otherwise, it doesn't work. :S |
| 12:40.00 | IsUp | Yourname``: whats default configuration? give me more details, whats your provider schema plan feelings hardware? you are here for 2 days and you are still generating a conf file. |
| 12:40.09 | Yourname`` | LOL |
| 12:40.56 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:40.56 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:41.10 | Yourname`` | That comment gave me an idea. |
| 12:41.12 | Yourname`` | One sec.. |
| 12:45.52 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 12:47.49 | Yourname`` | IsUp: After changing chan_dahdi.conf, do I need to do anything other than dahdi restart in cli? |
| 12:48.57 | IsUp | Yourname``: if you are changing chan_dahdi.conf, restart your Asterisk. i am always following this way. |
| 12:49.06 | Yourname`` | IsUp: Thanks! |
| 12:51.58 | ffon | IsUp: sorted partially, was a stray port pointing to 4567 instead of the default (long story). Audio RX is quite choppy (the original problem), enabeling jb doesn;t seam to help |
| 12:53.12 | Yourname`` | IsUp: When called party hangs up, it takes about 2 seconds before I see the hangup on CLI.. why so? |
| 12:54.49 | IsUp | Yourname``: its about with hangup detection. this could be a problematic case, i suggest you to take a look on voip-info.org |
| 12:55.55 | Yourname`` | IsUp: ok, cool, thanks |
| 12:56.18 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
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| 13:03.53 | *** join/#asterisk kriegerod (~krieger@79.135.222.22) |
| 13:04.38 | Yourname`` | IsUp: Also, when a call does out via dahdi, and before the called party even rings, it shows on cli as DAHDI answered.. |
| 13:05.30 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 13:07.09 | [TK]D-Fender | Yourname``, Analog DAHDI does that unless you do "callprogress=yes" |
| 13:07.29 | Yourname`` | [TK]D-Fender: chan_dahdi? |
| 13:07.39 | [TK]D-Fender | Yourname``, Which is often synonymous with "hangupmycallsonmeatrandom=yes" |
| 13:08.07 | Yourname`` | Ugh. Nevermind then, lol |
| 13:10.10 | Yourname`` | Hmm, but then mixmonitor will always cause some trouble :( |
| 13:11.03 | beaver_rrr | how can it be: local sip calls transfer no voice, but echo test working, outbound/inbound sip calls too |
| 13:11.50 | IsUp | beaver_rrr: NAT? |
| 13:12.18 | beaver_rrr | hm, two networks |
| 13:12.39 | beaver_rrr | one with direct connection to sip provider, and second one with local phones |
| 13:13.24 | *** join/#asterisk Dovid (~Dovid@host-78-158-94-213.wlan-guest.nycmny02.us.sargasso.net) |
| 13:13.32 | beaver_rrr | oh |
| 13:14.39 | beaver_rrr | do i need to explicitly show asterisk that local peers use no nat? i have nat=route in global section of sip.conf, because sip provider uses ip-address auth |
| 13:14.45 | beaver_rrr | gonna try |
| 13:15.53 | beaver_rrr | hm, they all have nat=no setting |
| 13:22.02 | IsUp | beaver_rrr: set localnet= |
| 13:22.44 | *** join/#asterisk luckman212_ (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 13:25.02 | Dovid | I have some one using "friendly-scanner" on me. Anyone know of a way of "sending a present back to them" or some other response to get it to stop sending packets? |
| 13:26.01 | [TK]D-Fender | Dovid, "DROP" |
| 13:30.00 | beaver_rrr | localnet helped me not :< |
| 13:34.04 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.2) |
| 13:34.08 | Dovid | TK: I did that aleady. i jsut wanted to send them a "present' |
| 13:34.10 | Dovid | present" |
| 13:35.19 | IsUp | Dovid: for christmas? |
| 13:35.26 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 13:35.51 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:36.04 | Dovid | IsUp: for them being A** holes |
| 13:38.53 | beaver_rrr | http://img.leprosorium.com/1324259 - do i need any NATs here? |
| 13:41.24 | [TK]D-Fender | beaver_rrr, No, just "directmedia=no"on everything |
| 13:41.49 | Dovid | if all the RTP goes via asterisk then no. |
| 13:42.11 | Dovid | the main advantage of no NAT is having the rtp go direct which here you can't do. |
| 13:42.28 | Dovid | oh how i long for the day when we all have IPv6 |
| 13:42.28 | fprior | hi all, what about Predictive Dialers ? what is better: pre-armed solution (Vicidial & Co) or "manually" dialplan ? "the definitive guide" don't focus on this function |
| 13:45.05 | blitzrage | fprior: because we're not even bastards who call people |
| 13:45.09 | [TK]D-Fender | fprior, Most of us hate telemarketers, and this isn't a "function", it's an "application" |
| 13:45.11 | blitzrage | s/even/evil/ |
| 13:45.28 | blitzrage | highly requested, highly rejected feature in the book |
| 13:45.31 | schmidts | Dovid i know the guy who wrote sipvicous, he will be happy if you sent him a present. btw he likes beer so be kindly to him ;) but to be honest, you will only sent a present to some zombie machine out there and not the real hacker guy so just secure your server and you will not have any problems anymore |
| 13:45.53 | [TK]D-Fender | fprior, also unclear what you mean about : "manually" dialplan |
| 13:46.13 | blitzrage | [TK]D-Fender: he just means programming it himself instead of a drop in solution |
| 13:46.43 | blitzrage | fprior: there is more than enough information in the book about how to use the tools we do focus on to build what you want -- but we don't go to the extent of showing you exactly how to build it. |
| 13:46.59 | blitzrage | just look at any of the Originate functionalities in asterisk |
| 13:52.29 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 13:55.52 | *** join/#asterisk lcat (~lcat@187.45.254.64) |
| 14:00.44 | fprior | @blitzrage, [TK]D-Fender: wow wow wow, keep calm, please. 1) I'm NOT telemarketer, my call-center place calls for clients who have debts with us; 2) for me "the definitive guide" is the bible, saying "don't focus on this function" I don't mean "f**king guida that not explain me perfectly the application", I would repeat what the bible say on page 261 (http://pastebin.com/raw.php?i=f47BRtbD ),3) I don't searching any Dialplan to co |
| 14:01.04 | blitzrage | I was super calm |
| 14:01.34 | schmidts | blitzrage :D |
| 14:01.58 | blitzrage | fprior: yes I know what the book says, I co-authored it |
| 14:02.03 | pabelanger | blitzrage: n00b |
| 14:02.15 | blitzrage | pabelanger: I know right? |
| 14:02.29 | kaldemar | no, the atmospehere was definitely getting tense. now it's ready to blow up. |
| 14:02.34 | blitzrage | calmly drinks the rest of his Tim Hortons coffee |
| 14:02.41 | blitzrage | kaldemar: I'm freaking out man! |
| 14:03.01 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:03.05 | *** kick/#asterisk [kaldemar!~pabelange@asterisk/contributor-and-bug-marshal/pabelanger] by pabelanger (ka-boom! ) |
| 14:03.12 | [TK]D-Fender | fprior, And asterisk has app_queue for inbound call center type use. It does not have anything vaguely complex in the direction of outbound campaigns |
| 14:03.26 | [TK]D-Fender | fprior, This is entirely up to you |
| 14:03.53 | *** join/#asterisk justdave_ (~dave@unaffiliated/justdave) |
| 14:03.54 | brad_mssw | the packages.asterisk.org doesn't appear to have any ubuntu 1.8.8 releases .... is that still in the works? |
| 14:03.55 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 14:03.56 | pabelanger | kaldemar: <3 |
| 14:04.14 | kaldemar | pabelanger: that's it! |
| 14:04.24 | blitzrage | anyways, as I eluded to previously -- you can use the Originate type functionality for outbound call centres. Most of that functionality needs to be built externally using scripts/programs to control the outbound calls and connect them to agents. The book focuses on Asterisk functionality (look at the AMI chapter for information about originating calls via external scripts). So all the tools are explained, but since there |
| 14:04.24 | blitzrage | is no dialplan application for "outbound call centres" then we don't discuss it. |
| 14:04.27 | pabelanger | brad_mssw: they should exist, what version |
| 14:04.49 | blitzrage | pabelanger: 1.8.8.0 |
| 14:05.01 | brad_mssw | http://packages.asterisk.org/deb/pool/main/a/asterisk/ |
| 14:05.16 | brad_mssw | there's 1.8.8.0~rc4 ... but no final packages |
| 14:05.16 | *** join/#asterisk serafie (~erin@nat/digium/x-zdnoapfxokpucmtk) |
| 14:05.26 | pabelanger | Hmm, must be a syncing problems |
| 14:05.30 | pabelanger | standby |
| 14:05.50 | brad_mssw | debian is also affected, I should say |
| 14:09.31 | fprior | [TK]D-Fender, @blitzrage: ok, thanks. And sorry if I was confuse with my question and terms. |
| 14:13.03 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 14:17.39 | pabelanger | brad_mssw: they are built, it is a sync'ing issue with the mirror. Working on fixing it how |
| 14:17.50 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-msnqnhmyfxedojca) |
| 14:17.50 | brad_mssw | pabelanger: thanks! |
| 14:21.04 | beaver_rrr | http://pastebin.ru/ARAExrkA what's wrong with these peers, in case of my previous scheme? |
| 14:21.13 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 14:24.59 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-plznwpqidfovrzgr) |
| 14:25.36 | Yourname`` | IsUp: Thanks a ton for your help! |
| 14:27.01 | [TK]D-Fender | beaver_rrr, canreinvite was replaced by directmedia in 1.6.0 |
| 14:27.08 | [TK]D-Fender | beaver_rrr, change the parm name |
| 14:27.15 | [TK]D-Fender | beaver_rrr, And check your firewalls |
| 14:27.51 | beaver_rrr | okay, thanks |
| 14:28.15 | *** join/#asterisk resno (~resno@unaffiliated/resno) |
| 14:29.15 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 14:31.21 | beaver_rrr | yes, iptables was misconfigured a little. |
| 14:32.45 | resno | hello folks. my job is looking to move to asterisk. i have a question about using a potn. id need an ata right? |
| 14:34.10 | kaldemar | or an analog interface card. or a MUX. |
| 14:35.07 | [TK]D-Fender | potn? |
| 14:35.37 | [TK]D-Fender | And an ATA is to plug in an analog phone, not a a line |
| 14:46.20 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 14:47.53 | schmidts | does someone ever noticed a problem with MOH on a sip channel if also a local channel was dialed at the same time? |
| 14:48.25 | schmidts | i have some like Dial(SIP/phoneA&LOCAL/123@test) and when phoneA answer the call and puts the call on hold, moh doesnt work anymore |
| 14:48.31 | schmidts | without the local channel it works fine |
| 14:50.53 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:55.47 | *** join/#asterisk Dovid (~Dovid@host-78-158-94-213.wlan-guest.nycmny02.us.sargasso.net) |
| 14:57.01 | *** join/#asterisk DaRooster (bennettr@rooster.satexas.com) |
| 14:58.12 | DaRooster | I'm an old trixbox user, so I'm semi-familar with how asterisk works.... but it's been a few years. I have AASTRA 6731i phones and a fresh install of asterisknow. Where can I read info about how to program these to talk to asterisk? (Yes, I knkow of phone web interface) |
| 14:58.18 | schmidts | btw it looks like MOH doesnt work at all .... :( |
| 14:58.38 | DaRooster | I've googled my ass off :( |
| 14:59.07 | resno | kaldemar: are there any advantages in going ip phones over analog ones? |
| 15:02.20 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 15:03.29 | jacc0 | @resno: remote management is one of them |
| 15:04.19 | resno | are there any sources i can look at to learn more about this? |
| 15:04.32 | resno | id rather not ask 10 billion questions here |
| 15:04.48 | jacc0 | and you can use the existing utp cables; so you need les cables |
| 15:04.49 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:05.39 | resno | do the ip phones split off the utp? or will i need a hub? |
| 15:06.07 | jacc0 | linksys and cisco phones do |
| 15:06.20 | jacc0 | some others also |
| 15:06.27 | resno | thats pretty cool |
| 15:06.47 | jacc0 | you should also consider peo (power over ethernet) |
| 15:06.52 | jacc0 | *poe |
| 15:07.05 | jacc0 | so you don't need extra power outlets |
| 15:07.14 | jacc0 | that will also greatly reduce cost |
| 15:07.17 | resno | our current phones need power |
| 15:07.35 | jacc0 | cisco has poe phones |
| 15:07.41 | resno | reduce cost were in new outlets? |
| 15:07.55 | jacc0 | yeah, because the power comes over the ethernet utp cable |
| 15:08.09 | jacc0 | you would only need a poe switch/router |
| 15:08.11 | resno | does that cause any issues with other networked devices? |
| 15:08.15 | jacc0 | nope |
| 15:09.03 | jacc0 | not that I know of |
| 15:10.11 | schmidts | resno if you have some analog wires directly laying besides the network cable you could have more interferences between both but thats not a very common problem |
| 15:11.10 | resno | schmidts: ill just wrap them in tin foil :) |
| 15:11.48 | schmidts | resno where are you located? in usa? cause mostly everywhere else in the world common network cable are shielded and grounded so this wouldnt be such a big problem ;) |
| 15:12.00 | resno | yes, USA |
| 15:13.21 | DaRooster | Where can I find info on how to connect AASTRA 6731i phones to a fresh AsteriskNow install (Digigium card, analog trunks) |
| 15:13.53 | schmidts | resno ok then you should try to use Cat7 cables for it, not only Cat5e |
| 15:14.00 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:14.07 | resno | i didnt know they made cat7 cables |
| 15:15.01 | *** join/#asterisk mocker (~ksexton@206.55.118.84) |
| 15:15.02 | schmidts | Cat7 are more shielded and the wiring is better than in Cat5e or even Cat6 but they are more expensive |
| 15:16.53 | *** join/#asterisk Wil_Syd (~androirc@c211-30-0-250.rivrw2.nsw.optusnet.com.au) |
| 15:17.31 | resno | for a small office... 5 people you dont need a high powered server do you? |
| 15:17.54 | schmidts | resno only for asterisk? not really ;) |
| 15:20.38 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
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| 15:27.21 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 15:27.32 | SeRi | does anybody know how I can I silence this in asterisk? http://pastebin.com/4tDgdfhU |
| 15:32.05 | Naikrovek | what asset tracking software do you sysadmins use? (non-homegrown solutions) |
| 15:33.07 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:33.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:34.29 | Dovid | my brain |
| 15:34.34 | Dovid | or google docs |
| 15:35.29 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 15:35.34 | schmidts | Naikrovek if you mean something like an issue tracker take a look at bestpractical tracker system |
| 15:35.40 | Naikrovek | no asset tracking. |
| 15:35.46 | Naikrovek | i have an issue tracker. |
| 15:36.01 | Naikrovek | i have LOTS of little bits of hardware (IT and non-IT related) that I need to track |
| 15:36.11 | Naikrovek | wondering what would be recommended by the group, here. |
| 15:36.31 | Naikrovek | s/no asset tracking/no, asset tracking/ |
| 15:38.20 | Naikrovek | something like Wasp MobileAsset probbaly |
| 15:38.23 | Naikrovek | probably* |
| 15:38.35 | Naikrovek | ... wish they had an android app, though. |
| 15:46.08 | *** join/#asterisk chasing`Sol (~cS@41.206.155.159) |
| 15:49.01 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.29.44) |
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| 16:03.13 | *** part/#asterisk joshaidan (~brianj@24.109.210.41) |
| 16:03.21 | *** join/#asterisk mintee (~mintee@2001:470:7:a41::2) |
| 16:03.23 | mintee | Hey all |
| 16:03.41 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 16:03.45 | mintee | So I'm trying to hand off a DAHDI dialed number to a SIP and I'm getting cause 27 |
| 16:03.52 | mintee | destination out of order |
| 16:04.11 | mintee | i'm using the exact same setup as another box with the same hardware |
| 16:04.15 | WIMPy | From SIP? |
| 16:04.19 | mintee | ya |
| 16:04.19 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 16:05.20 | mintee | [globals] |
| 16:05.21 | mintee | OUTBOUNDTRUNK=DAHDI/G0 |
| 16:05.21 | mintee | [outbound] |
| 16:05.21 | mintee | ;Outbound number without a 1 |
| 16:05.21 | mintee | exten => _9NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) |
| 16:05.23 | mintee | exten => _9NXXNXXXXXX,n,Playtones(tt-monkeys) |
| 16:05.25 | mintee | exten => _9NXXNXXXXXX,n,Congestion(10) |
| 16:05.28 | WIMPy | I get 20 if SIP fails |
| 16:05.28 | mintee | exten => _9NXXNXXXXXX,n,Hangup() |
| 16:05.30 | WIMPy | ~pb |
| 16:05.30 | 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. |
| 16:05.41 | mintee | yes infobot, i appologize |
| 16:05.43 | voipeng | yea they will boot ya if ya dont use PB |
| 16:06.04 | WIMPy | My client already did. |
| 16:08.04 | mintee | http://pastebin.com/0JqRfPFV |
| 16:08.28 | mintee | There... now I'm compliant |
| 16:08.29 | WIMPy | Nice. If it tries, but gets no reply I get 111. |
| 16:08.51 | mintee | this is a new PRI |
| 16:09.05 | mintee | and the provider says it's my problem |
| 16:09.09 | mintee | but i'm wondering... |
| 16:09.13 | WIMPy | So it's from SIP to DAHDI, not the other way round? |
| 16:09.15 | mintee | i can get inbound calls no prob |
| 16:09.25 | mintee | WIMPy: yup |
| 16:10.01 | WIMPy | Wrong number type? |
| 16:10.16 | WIMPy | Do you have prilocaldialplan set in chan_dahdi.conf? |
| 16:11.00 | [TK]D-Fender | mintee, that dialplan you showed us doesn't match the call even. |
| 16:11.02 | WIMPy | And is that a NANP destination? |
| 16:11.27 | [TK]D-Fender | mintee, And you aren't looking at PRI debug |
| 16:11.57 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:12.45 | mintee | prilocaldialplan is not set, but it's not set on the good known box |
| 16:13.15 | WIMPy | Same provider? |
| 16:13.27 | mintee | [TK]D-Fender: oh, ya... i didn't copy the right dialplan... sorry. it's there though |
| 16:13.34 | mintee | no, different provider |
| 16:13.51 | WIMPy | And is that a NANP destination? |
| 16:14.05 | WIMPy | have you tried without the 1? |
| 16:14.38 | *** join/#asterisk woleium (~woleium@email.tecglobal.net) |
| 16:14.58 | mintee | ya, tried with and without the 1 |
| 16:15.04 | mintee | i have 2 seperate dialplans |
| 16:17.13 | [TK]D-Fender | mintee, I don't see matching configs & debug, nor complete debug. |
| 16:17.32 | mintee | http://pastebin.com/xbtRGvVq |
| 16:18.19 | mintee | line 52 |
| 16:18.37 | WIMPy | No new information. |
| 16:18.58 | WIMPy | Try to set prilocaldialplan and pridialplan to unknown. |
| 16:19.16 | mintee | in chan_dahdi.conf? |
| 16:19.25 | WIMPy | yes |
| 16:19.54 | WIMPy | Hmm. Remote public network? Not sure if that should be take too serious. |
| 16:21.20 | mintee | prilocaldialplan didn't fix anything |
| 16:21.28 | mintee | err, both set to unknown |
| 16:22.45 | mintee | the PBX and the SIP are both on local private ips |
| 16:22.50 | mintee | same subnet |
| 16:23.06 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 16:23.10 | WIMPy | oops |
| 16:23.39 | WIMPy | "custserv2" is not a valid caller ID. |
| 16:24.32 | WIMPy | And not every switch likes the display IE. But I guess it would moan immediately about that one. |
| 16:24.47 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 16:25.56 | mintee | hum... i donno where i would change the caller id part |
| 16:26.41 | WIMPy | In sip.conf, on the device or in your dialplan. |
| 16:28.19 | WIMPy | wonders why they accept that setup at all. |
| 16:29.07 | mintee | ok, fixed that |
| 16:29.12 | mintee | but still no outbound |
| 16:29.26 | mintee | i setup an inbound to the same SIP from the PRI and it worked fine |
| 16:30.17 | WIMPy | I don't think I can spot more that those. |
| 16:31.09 | mintee | hum... |
| 16:32.25 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 16:34.37 | *** join/#asterisk mattsqz (~luser@67-61-162-124.cpe.cableone.net) |
| 16:38.18 | hackeron | heh, weird, #freeswitch and #asterisk have exactly 201 people at the moment :) |
| 16:40.43 | *** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 16:41.06 | mintee | 201 is my default extension for everything |
| 16:41.11 | DelphiWorld | hackeron: xD |
| 16:41.22 | mintee | so, i removed all nat related shit and it's still not working |
| 16:41.28 | mintee | maybe lunch will fix it |
| 16:42.32 | tamiel | mintee: coffee/tea/smoking/whatdrugUneed fix ;) |
| 16:43.15 | WIMPy | mintee: You don't have a sip issue there. |
| 16:44.41 | *** part/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 16:44.55 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 16:47.22 | mattsqz | see, i knew irc was good for something other than begging for help and triggering questionably legal dcc transfers |
| 16:47.28 | mattsqz | i had no idea freeswitch existed |
| 16:47.34 | mattsqz | learned something |
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| 17:24.27 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 17:24.29 | wcselby | o/ |
| 17:24.35 | *** join/#asterisk vassilux (~vassilux@LPuteaux-156-14-48-156.w82-127.abo.wanadoo.fr) |
| 17:25.05 | [TK]D-Fender | <PROTECTED> |
| 17:25.35 | wcselby | doh! |
| 17:25.43 | wcselby | just typed my password into the login box |
| 17:25.50 | wcselby | trying to remote into a linux server |
| 17:26.12 | wcselby | that's why I like settng up ssh keys |
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| 17:30.39 | wcselby | r0m|u ping |
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| 17:44.31 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 17:48.35 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:49.16 | *** join/#asterisk DagMoller (~aguirre@unaffiliated/dagmoller) |
| 17:49.33 | *** part/#asterisk DagMoller (~aguirre@unaffiliated/dagmoller) |
| 17:52.46 | *** join/#asterisk mmaki (~mmaki@pool-71-109-218-186.lsanca.dsl-w.verizon.net) |
| 17:53.35 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:58.06 | *** join/#asterisk fipu (~fipu@fipu.usr.sh) |
| 18:15.02 | anonymouz666 | anyone in here already used SHARED function? It is possible to use a variable created in called channel to become visible in calling channel? |
| 18:15.09 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 18:16.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:17.13 | *** join/#asterisk kikohnl (~kotis@72.253.138.39) |
| 18:17.31 | [TK]D-Fender | anonymouz666, Standard inheritance rules apply |
| 18:17.40 | *** join/#asterisk mmaki (~mmaki@pool-71-109-218-186.lsanca.dsl-w.verizon.net) |
| 18:19.03 | anonymouz666 | what |
| 18:19.54 | anonymouz666 | calling to called is easy _ __ |
| 18:20.33 | anonymouz666 | to make it global on called channel is also a way to become visible in calling channel |
| 18:20.43 | anonymouz666 | but I want to work with channel variables |
| 18:23.40 | *** join/#asterisk Azrael808 (~peter@178.102.211.63) |
| 18:40.19 | *** join/#asterisk corretico (~luis@190.211.94.6) |
| 18:40.37 | corretico | somebody know about this line: CSeq: 102 CANCEL |
| 18:41.04 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 18:41.05 | corretico | I'm try to conect my asterisk with a Nortel/Avaya BCM 450 using a sip trunk |
| 18:46.44 | [TK]D-Fender | corretico, Single lines like that are meaningless... you need to loko at the entire SIP conversation. |
| 18:46.47 | [TK]D-Fender | ~pb |
| 18:46.47 | 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. |
| 18:46.49 | [TK]D-Fender | ^^^ |
| 18:47.27 | corretico | <[TK]D-Fender>ok. let try to copy a full sip debug for this trunk |
| 18:51.38 | corretico | http://pastebin.com/KzPtEa7C |
| 18:51.57 | *** join/#asterisk gravin (~gravin@57.70.50.60.brf01-home.tm.net.my) |
| 18:53.24 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-212.ks.ks.cox.net) |
| 18:54.46 | [TK]D-Fender | corretico, The other side isn't answering at all |
| 18:55.00 | [TK]D-Fender | corretico, INVITE sip:8000@10.100.100.14:1720 SIP/2.0 <- and 1720 is an odd port # to be sending SIP to |
| 18:56.24 | [TK]D-Fender | infact ... 1720 is an H.323 port not SIP |
| 18:56.35 | corretico | yes |
| 18:56.49 | corretico | let me try to change |
| 18:56.51 | corretico | 5060 |
| 18:56.59 | corretico | 1720 is an H.323 |
| 18:57.10 | [TK]D-Fender | So why are you sending to that? |
| 18:57.46 | corretico | 'cost the avaya admin, toll me that 1720 is a correct port |
| 18:57.55 | wcselby | i was about to say |
| 18:58.06 | wcselby | because most interconnects with avaya that I've seen have been done on h323, not SIP |
| 18:58.24 | wcselby | so someone is probably reading from the wrong config guide |
| 19:06.30 | corretico | <wcselby>yes, I was reading about avaya and asterisk with h.323 but not with SIP |
| 19:08.49 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 19:10.04 | corretico | http://pastebin.com/P8Czg7eL |
| 19:11.37 | [TK]D-Fender | corretico, that isn't even a complete packet from * SIP debug let allone a complete communication. |
| 19:12.57 | corretico | <[TK]D-Fender>how i cant get only a debug for this trunk |
| 19:12.59 | corretico | ? |
| 19:13.11 | [TK]D-Fender | corretico, Funny you got it to me the time before.... |
| 19:13.42 | corretico | jejeje yes, but know I see all debug for all ext and trunks |
| 19:13.42 | [TK]D-Fender | corretico, Stop worrying about filtering and let us parse it out |
| 19:20.31 | corretico | http://pastebin.com/BpqXeTQk |
| 19:20.34 | corretico | complete log |
| 19:21.01 | corretico | my ext is 3204 and the destination ext is 8000 |
| 19:22.13 | [TK]D-Fender | corretico, BCM is still not answering back at all |
| 19:22.59 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 19:23.38 | corretico | the problem is on the other side |
| 19:25.21 | [TK]D-Fender | corretico, Retransmitting #4 (no NAT) to 10.100.100.14:5060: <- make sure the IP is right, that the networking in between is right, verify that your server can see any networking resource at that IP at all, etc. |
| 19:25.42 | [TK]D-Fender | corretico, make sure you even configured it to be listening to SIP.... |
| 19:26.12 | [TK]D-Fender | corretico, Considering your first H.323 mixup I would not be surprised if this is also part of the issue |
| 19:27.11 | corretico | <[TK]D-Fender>thanks. the network is ok. I can see other resources at 10.100.100.0/24, including 10.100.100.14 |
| 19:27.30 | [TK]D-Fender | corretico, Prove that it is even listening. |
| 19:28.08 | *** join/#asterisk tapout (~karen@unaffiliated/tapout) |
| 19:28.15 | corretico | <[TK]D-Fender>but I'll talk with the avaya admin for make another test... |
| 19:28.33 | corretico | I'll try to use a softphone to connect with the avaya |
| 19:28.41 | tapout | for Faxes, which codec do I want to use? Right now I have... G.711U and G.729a enabled. |
| 19:36.45 | Micc | tapout, don't use g729 for faxing. ulaw is probably your best bet. |
| 19:37.24 | Micc | tapout, you want less compressed audo, not more. |
| 19:40.38 | Qwell | tapout: T.38, if you can |
| 19:41.02 | tapout | voip.ms only has g711u and g729 |
| 19:41.06 | tapout | that is listed for my account |
| 19:43.07 | tapout | one asterisk box, can register multiple accounts right? |
| 19:43.07 | Qwell | well, it's not really a codec, so it wouldn't be listed there. |
| 19:43.16 | Qwell | You'd probably have to ask if they support T.38 |
| 20:12.05 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:26.50 | tapout | can you not have multiple register lines in sip? |
| 20:27.00 | tapout | for some reason, I am trying to have one asterisk box satisfy 2 accounts. |
| 20:27.02 | *** join/#asterisk shido6 (~shido6@c-76-21-0-126.hsd1.ca.comcast.net) |
| 20:27.08 | tapout | when calling the second account, it routes thru the first account |
| 20:27.28 | tapout | i'm expecting it to be... 123456_fax but it comes thru as 111111 |
| 20:27.54 | Qwell | tapout: it can, yes |
| 20:28.02 | Qwell | How are you registering? |
| 20:28.19 | tapout | [general] section |
| 20:28.32 | tapout | register => 111111:pass@host:5060 |
| 20:28.43 | tapout | register => 123456_fax:pass@host:5060 |
| 20:28.50 | tapout | then i have 2 sections |
| 20:28.58 | tapout | [voipms] and .. [richard] |
| 20:29.20 | tapout | oh i gotta tell 123456 which box to call ee? |
| 20:29.26 | tapout | register => 123456_fax:pass@host:5060/richard |
| 20:29.29 | tapout | lemme try that |
| 20:30.54 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 20:31.05 | tapout | it's so weird, the calls from his account are coming thru on mine |
| 20:33.33 | *** join/#asterisk bmg505 (~leon@196-209-101-126.dynamic.isadsl.co.za) |
| 20:33.41 | carrar | all from the same IP? |
| 20:33.48 | carrar | Your provider? |
| 20:34.13 | sunfone | Hi - I am trying to track back the "source" of random hangups... I am getting reports of calls terminating, and at verbose 3 I don't see any difference in a call ending normally to a call that apparently ended unintentionally... |
| 20:34.19 | carrar | Probably maching on IP |
| 20:34.29 | carrar | but you haven't shown us enough |
| 20:35.02 | sunfone | So I enabled verbose 100 and debug 100, and see "Didn't get a frame from channel: SIP/100-00000f73" when the hangup happens locally... |
| 20:35.20 | sunfone | And the other channel (trunk) if it happens remotely... |
| 20:35.34 | sunfone | Is that the only indication I will get about hangups? That a frame is missed? |
| 20:37.03 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:37.14 | carrar | Welcome to the Micky's FUN HOUSE TK! |
| 20:37.19 | *** join/#asterisk brezular (~brezular@adsl-dyn129.78-98-48.t-com.sk) |
| 20:40.44 | [TK]D-Fender | YOU BLOW MY MIND |
| 20:46.12 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 20:52.54 | *** part/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 20:59.07 | mmaki | Seem to have a problem with voicemail.conf option sendvoicemail=yes |
| 20:59.35 | mmaki | when I'm in voice mail and select 3 for advanced options and 5 to leave a message |
| 21:00.02 | mmaki | I get the users name recording then it immediatly dumps me back into my mailbox |
| 21:00.27 | mmaki | No option to record a message |
| 21:00.56 | *** join/#asterisk bmg505 (~leon@196-209-101-126.dynamic.isadsl.co.za) |
| 21:01.06 | pabelanger | sunfone: you only need to set verbose / debug to 15 |
| 21:09.31 | sunfone | pabelanger: ok, but still is there an easier way to log the side that hungup? Like a return from Dial or Answer that I could verbose into the log or something? |
| 21:10.00 | sunfone | I have smallish disks in the customer PBXes (4G flash cards) and they would fill up quickly, even at 15 |
| 21:10.17 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 21:11.14 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:13.09 | chuckf | sunfone: is a remote syslog an option for sending the logs to? |
| 21:14.06 | sunfone | chuckf: yes, I suppose so, but with hundreds of them in the field even collecting level 15 from all of them centrally, just for the purposes of detecting and logging the hangup side, seems a bit severe |
| 21:14.42 | sunfone | I guess I could dig into the "Dial" code and try to send out a log line or a return code that indicates which channel terminated? |
| 21:14.51 | sunfone | I'd be going out on a limb for sure :) |
| 21:18.39 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 21:32.39 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 21:39.12 | [TK]D-Fender | sunfone: "g" |
| 21:40.05 | sunfone | Well, I found the spot in channel.c that spits out the debug line, anyway |
| 21:40.47 | sunfone | But channel.c doesn't know who is remote and who is local |
| 21:40.49 | [TK]D-Fender | Dialplan offers this already |
| 21:41.00 | sunfone | how so? |
| 21:41.43 | sunfone | is there a variable set that indicates which channel terminated? |
| 21:43.02 | [TK]D-Fender | "g" |
| 21:43.13 | [TK]D-Fender | + DIALSTATUS |
| 21:44.28 | sunfone | Ha! I thought you were grinning at me :) |
| 21:44.55 | sunfone | Thanks! I'll play with that |
| 21:48.09 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 21:50.14 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 21:53.49 | *** join/#asterisk CyfordTechnologi (~allen@c-24-98-175-41.hsd1.ga.comcast.net) |
| 21:54.37 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.57.16.dsl.dyn.forthnet.gr) |
| 21:55.54 | akrohn | hello. i was wondering if anyone could tell me how to best utilize my CDRs for billing purposes. Currently, all I have to go on is the call's context which contains the customer name and the two phone numbers involved in the call. I would like to use the accountcode field to give clients a UID. I guess I'm asking where the best place in the dialplan is to define this field? |
| 21:55.56 | *** join/#asterisk XnOSX (~XnOSX__@62.83.235.60.dyn.user.ono.com) |
| 21:58.22 | XnOSX | hello everybody, anybody have an script to automatically start asterisk? |
| 21:58.50 | akrohn | XnOSX, what platform? |
| 21:59.45 | *** join/#asterisk schmidts (~sst@86.59.31.4) |
| 21:59.51 | schmidts | hi folks |
| 22:00.00 | [TK]D-Fender | akrohn: Any time before the call ends |
| 22:00.32 | [TK]D-Fender | * comes with a variety of init scripts. that's what "make config" is for |
| 22:00.35 | schmidts | could someone please help me and try normal music on hold when a call between two sip phones is put on hold? |
| 22:00.42 | akrohn | [TK]D-Fender, thanks. that's all? i guess i was over-complicating things |
| 22:00.44 | schmidts | with asterisk 1.8 |
| 22:02.00 | wcselby | akrohn Set(SIPPEER(accountcode)=...) maybe? |
| 22:02.31 | *** join/#asterisk woleium (~woleium@208.73.56.64) |
| 22:02.42 | wcselby | schmidts rephrase your question, I don't really understand what you're needing help with? |
| 22:03.14 | XnOSX | akrohn, Debian Lenny! |
| 22:03.23 | akrohn | wcselby, !!!! that's awesome. I didn't know I could do that! thanks =) |
| 22:03.35 | wcselby | i don't know if it's read / write or just read only |
| 22:03.51 | schmidts | wcselby the following dialplan is given: exten 123,1,Set(CHANNEL(musicclass)=1234) or default or whatever and a simple Dial(SIP/phone123) |
| 22:04.07 | schmidts | sorry i meant Dial(SIP/phone123,30,m(1234)) |
| 22:04.07 | XnOSX | akrohn, before had a script. sh for that, but I can not find it! |
| 22:04.18 | schmidts | i hear the music until phone123 answer the phone |
| 22:04.30 | akrohn | XnOSX, you should have a startup script in /etc/init.d called 'asterisk' |
| 22:04.39 | schmidts | phone123 puts the call on hold, i see in the cli started music on hold class 1234 but i dont hear anyhting |
| 22:04.55 | schmidts | and atleast i dont even see any rtp packet being sent from asterisk |
| 22:05.17 | wcselby | schmidts please pastebin a copy of the CLI logs and the relevant configs when this is happening |
| 22:05.19 | wcselby | ~pb |
| 22:05.19 | 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. |
| 22:05.46 | akrohn | schmidts, you're certain about the location of the music file? That * is looking in the right place for it? |
| 22:06.13 | XnOSX | akrohn, no i don't have it! i have looked at the directory |
| 22:06.36 | schmidts | akrohn no as i said the music works, audio between the phones works but i dont hear any music when the call is put on hold |
| 22:07.22 | akrohn | XnOSX, The "/usr/src/asterisk/contrib/init.d directory contains some sample scripts" ... see if there's one in there |
| 22:10.31 | XnOSX | akrohn, ok hold on! |
| 22:11.52 | [TK]D-Fender | [17:06]schmidtsakrohn no as i said the music works, audio between the phones works but i dont hear any music when the call is put on hold<- you just the music works. and doesn't work.. all in the same sentence |
| 22:14.49 | schmidts | [TK]D-Fender sorry i dont know how this is called when playing music before a call is answered, as the m parameter of the Dial application. this music works. Maybe i will call this Music before answer. the real Music on Hold while the call is on hold doesnt work |
| 22:15.49 | wcselby | schmidts still waiting on that pastebin |
| 22:16.32 | schmidts | wcselby here is a pastebin of the sip trafic i am working on the debug output |
| 22:16.41 | schmidts | <PROTECTED> |
| 22:16.44 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:16.46 | wcselby | i don't really need that right now, I want to see the CLI output |
| 22:16.54 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:17.00 | schmidts | wcselby you dont see anything specific there |
| 22:18.21 | schmidts | wcselby here you are: http://pastebin.com/QsM1dbFX |
| 22:18.28 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 22:19.28 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 22:21.59 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:22.00 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 22:22.22 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 22:24.09 | wcselby | schmidts what version of asterisk are you using? |
| 22:24.15 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:24.25 | schmidts | Asterisk SVN-branch-1.8-r348735 |
| 22:31.27 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 22:35.58 | schmidts | wcselby with 1.8.7.1 it works :( looks like a regression somewhere |
| 22:36.14 | wcselby | schmidts there's a newer version of svn trunk, maybe try that? |
| 22:36.16 | wcselby | or submit a bug report |
| 22:36.36 | wcselby | i dn't knwo the reviusion numbers very well, is that newer than 1.8.8.0 ? |
| 22:37.07 | schmidts | wcselby this rev was from today in the morning, the changes to 1.8 today doesnt have anything do to with that |
| 22:37.12 | schmidts | yes its newer ;) |
| 22:37.15 | WIMPy | Yes, would be extremely helpful if the svn versions had version numbers. |
| 22:37.53 | schmidts | 1.8.8 was rev r348304 |
| 22:39.48 | wcselby | WIMPy :P |
| 22:43.55 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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| 23:05.25 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-msnqnhmyfxedojca) |
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| 23:13.58 | *** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 23:14.01 | DelphiWorld | hey |
| 23:14.07 | DelphiWorld | anyone have a gxv3140? |
| 23:19.51 | *** join/#asterisk F2Knight (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 23:20.06 | F2Knight | Q: I need some help setting up TDMoE |
| 23:20.15 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 23:23.33 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 23:24.16 | corretico | hi |
| 23:24.25 | corretico | this is my new log http://pastebin.com/HK4B8qPt |
| 23:31.39 | DelphiWorld | F2Knight: how doe tdmOe work ? |
| 23:35.05 | F2Knight | TDMoE, uses a ethernet card to connect with either another ethernet card, or external T1/E1 (TDM-device) |
| 23:35.27 | F2Knight | it uses TDM protocol like a T1 rather then a SIP account or the like |
| 23:36.03 | F2Knight | I have an external T1 (redfone) but am not fimular with how to configure DAHDI in general so Not sure what I should be looking at or for. |
| 23:38.00 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 23:41.40 | *** join/#asterisk neurosys (~neurosys@75-149-243-177-Miami.hfc.comcastbusiness.net) |
| 23:44.35 | *** join/#asterisk giesen_ (giesen@204.14.17.135) |
| 23:45.06 | giesen_ | Is there any way to have asterisk originate a call programatically without a local extension/device |
| 23:45.42 | giesen_ | basically I want to use either a call file or the originate command to perform a dialout/dialin check without connecting it to a local extension |
| 23:45.52 | giesen_ | either that or some sort of null extension that always answers |
| 23:45.55 | WIMPy | You need an extension, but not a device. |
| 23:47.58 | giesen_ | According to http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out, I need some sort of channel |
| 23:48.34 | WIMPy | Or an extension. |
| 23:50.36 | giesen_ | So I can use an extension for the Channel argument? |
| 23:58.10 | *** join/#asterisk gigawatts121 (~gigawatts@c-24-13-241-203.hsd1.il.comcast.net) |