| 00:01.38 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-nvwfwxusojxiwsqg) |
| 00:05.37 | *** join/#asterisk twodogs (~twodogs@hendra.biohazard.seattle.wa.us) |
| 00:17.13 | *** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au) |
| 00:17.15 | phix | hey |
| 00:22.22 | *** join/#asterisk tapout (~karen@unaffiliated/tapout) |
| 00:22.54 | Henchman21 | so ugh how do i tell asterisks to goto voicemail if they dont answer in blah blah sec/rings |
| 00:23.02 | tapout | hey guys, I am looking for a voip phone that is wireless or runs off dect+base station. I was looking at kirk 5020, and it looks like it requires a kirk server. Is that the case? I just want a phone I can wifi connect or have a base station plugged into my router and use voip.ms |
| 00:24.32 | Henchman21 | exten => 1337,1,Dial(SIP/asteriskhacker) |
| 00:24.32 | Henchman21 | exten => 1337,2,VoiceMail(asteriskhacker@default) |
| 00:24.46 | Henchman21 | as it is rings for like 20-30 rings before droping me into voicemail |
| 00:26.47 | WIMPy | Henchman21: Add a timeout to your Dial. |
| 00:27.49 | Henchman21 | exten => 1337,1,Dial(SIP/asteriskhacker,timeout 10) ? |
| 00:28.14 | Henchman21 | or just 10 |
| 00:28.34 | [TK]D-Fender | 10 |
| 00:28.57 | [TK]D-Fender | Henchman21exten => 1337,2,VoiceMail(asteriskhacker@default) <--- alpha box names not allowed |
| 00:29.15 | WIMPy | Since when? |
| 00:29.18 | [TK]D-Fender | Henchman21: and the timeout is in seconds |
| 00:29.35 | Henchman21 | really? |
| 00:29.42 | Henchman21 | cause the mail is working for me |
| 00:30.00 | [TK]D-Fender | Last I heard repeated from devs was otherwise. |
| 00:30.13 | Henchman21 | ill be sure to change it |
| 00:30.28 | WIMPy | It has been working for me as well. |
| 00:31.59 | Henchman21 | we're special |
| 00:32.02 | Henchman21 | ;) |
| 00:32.55 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 00:34.37 | tapout | what's the cheapest wifi hardware phone that I can use in my house to use voip? I want to interface asterisk but cheaply.. looking at polycom solutions, they're looking for 500 dollars. I'm looking for the cheap cheap cheap method |
| 00:35.00 | Henchman21 | dang |
| 00:35.14 | Henchman21 | ive been using my android smarphone liek that |
| 00:35.27 | Henchman21 | got sip built in |
| 00:35.34 | Henchman21 | (latest firmware) |
| 00:35.49 | Henchman21 | its got wifi and such |
| 00:36.12 | Henchman21 | 50-100$ ish few years ago |
| 00:37.25 | tapout | which android app do you use? |
| 00:38.12 | Henchman21 | no app |
| 00:38.31 | Henchman21 | its got sip/voip stack builtin |
| 00:38.59 | tapout | how do you do it then? I've got a google samsung nexus .. android based, any ideas how I would set it up like you have it? |
| 00:39.14 | Henchman21 | android version 2.3.7 |
| 00:40.09 | Henchman21 | settings>call settings> internet calls settings |
| 00:40.26 | Henchman21 | i point it to my asterisk server and make a sip entry for it |
| 00:40.53 | Henchman21 | this is cyanogenmod firmware though dunno not all firmwares are created equal |
| 00:44.08 | Henchman21 | good luck |
| 00:52.10 | Henchman21 | hope i saved you 5 hundo |
| 00:57.43 | *** join/#asterisk RickCogley (~anonymous@EM114-48-53-253.pool.e-mobile.ne.jp) |
| 00:59.24 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:03.56 | tapout | awesome, i'm using csipsimple |
| 01:13.58 | Micc | a little off topic but, does anyone know a good wireless n router that allows you to have a web page that they have to accept terms and conditions before it allows wireless access to the network? |
| 01:22.05 | *** join/#asterisk kdmessano^ (~nonya@unaffiliated/kdmessano) |
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| 01:29.26 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
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| 01:38.37 | tzanger | coppice: thank you for the hint |
| 01:38.58 | tzanger | coppice: every 8th sample was wrong when it was fuzzing out |
| 01:39.15 | tzanger | and wrong as in correct in magnitude but incorrect in sign |
| 01:40.20 | *** join/#asterisk ketas- (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
| 01:42.45 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:43.52 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
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| 02:12.28 | *** part/#asterisk nny (~Scott@174.107.223.14) |
| 02:13.11 | Henchman21 | Micc: try openwrt and nocatsplash type stuff |
| 02:14.39 | Henchman21 | nocatauth whatever they're called |
| 02:15.22 | Micc | yeah I think I noticed dd-wrt does it too. |
| 02:15.23 | Henchman21 | captove portals |
| 02:15.31 | Henchman21 | captive* |
| 02:16.27 | Henchman21 | could probably use with any router using dedicated client to find/grab the new poeaple |
| 02:30.55 | phix | I want to set a SIP header when I dial out, so I use set to do this before I call dial? |
| 02:31.17 | phix | does the header reset at another invocation to dial? |
| 02:33.17 | *** join/#asterisk mintos (~mvaliyav@114.143.167.66) |
| 02:36.50 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 02:36.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 02:42.21 | phix | hey |
| 02:52.57 | phix | so what you rekcon? |
| 02:58.48 | Henchman21 | anyone familair with the linksys ata dialplan syntax? example ( <9,:>[2-9]xx[2-9]xxxxxx ) |
| 02:59.20 | Henchman21 | how do i tell it to just send whatever to asterisk and not trying to act smart ;) |
| 02:59.44 | ketas- | including ****? |
| 03:00.32 | Henchman21 | i have no idea |
| 03:00.47 | Henchman21 | didnt come with docs or help |
| 03:01.16 | ketas- | reminds me i have linksys ata here and i should hack it |
| 03:01.23 | Henchman21 | that i got from somones page saying how to mimic dial 9 for an outside line |
| 03:01.35 | ketas- | Henchman21: well docs are on internet of course |
| 03:02.02 | Henchman21 | i just want it to send everything to asterisk for routing maybe i should just leave it empty |
| 03:02.13 | ketas- | i guess so |
| 03:02.49 | ketas- | so you have problem where something is parsed by ata itself? |
| 03:03.52 | Henchman21 | found the docs |
| 03:04.19 | Henchman21 | http://www.jmgtechnology.com.au/spa_3000_guide.pdf#search=%22spa%203000%20dial%20plan%20syntax%22 |
| 03:22.59 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:32.17 | phix | any ideas gang? |
| 03:45.00 | autofsckk | good night, i was in a middle of a call and all of a sudden i stopped hearing, how can i see what happened? |
| 03:46.48 | autofsckk | how can i debug that? |
| 04:06.02 | phix | autofsckk: look at the log files or run asterisk -rvvvvdddd in a screen session and re-attach to it when it happens again |
| 04:06.30 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 04:09.11 | autofsckk | phix: while i was talking i did sip set debug on, could that be useful? |
| 04:09.41 | autofsckk | i got something like SIP/2.0 404 Not Found |
| 04:11.34 | [TK]D-Fender | the error message means little.... without know what it is in response to.... |
| 04:18.18 | autofsckk | http://pastebin.com/x6GvcLbU this is a weird one for example, 184.... is my providers ip, and 66.28.... is my vps, in the To: <sip:userfromprovider@my-vps's-ip> is that correct? isnt it mixing the info from my provider and my vps? |
| 04:18.52 | [TK]D-Fender | That is in response to an OPTIONS packet. This is of no concern whatsoever |
| 04:21.27 | [TK]D-Fender | This is not a call attempt, registration or anything else of any significance |
| 04:21.39 | autofsckk | ok |
| 04:25.45 | phix | yay [TK]D-Fender is back! |
| 04:26.21 | phix | [TK]D-Fender: I am setting up asterisk for hosting two seperate "businesses/purposes" |
| 04:27.10 | phix | [TK]D-Fender: I have the incoming and internal extensions / contexts setup good, now working on sounds / busy / DnD messages etc... |
| 04:28.02 | phix | My question is, can I look in a different directory to get a sound based on the extension I am in? or set a flag to say what the parent directory is of the custom sounds / messages I want to play? |
| 04:28.28 | autofsckk | sometimes i really hate asterisk, when it seems that everything is working right, then something happens and it stops receiving or sending voice or whatever error you could think of, im learning a lot trying to debug it, but i really hate when that happens :S |
| 04:29.05 | phix | generally if voice just stops or doesn't happen it usually means firewall is blocking it or issues with codec, in my experience any way |
| 04:29.06 | [TK]D-Fender | phix: The apps look at whatever file you tell them to... |
| 04:29.51 | phix | [TK]D-Fender: ok, I would like them to try a default location if the file doesn't exist |
| 04:30.01 | [TK]D-Fender | Not happening. |
| 04:30.12 | phix | what? |
| 04:30.16 | [TK]D-Fender | Tell it right, or it will fail |
| 04:30.25 | [TK]D-Fender | There is no "fallback" |
| 04:30.36 | phix | ok, so I need to use a macro or script to test if file exists? |
| 04:31.44 | [TK]D-Fender | No, you need to actually put the files there. Why is their very existance in doubt? You seem to be thinking backwards to detect and fix problems instead stopping yourself from making them. |
| 04:31.46 | phix | [macro-playWithFallback] take in 2 args, context / business name, and file name, if it doesnt exist try the default location, sound like a good or bad idea? |
| 04:32.05 | [TK]D-Fender | Do the full job <------- Good idea |
| 04:32.33 | phix | [TK]D-Fender: because end users are going to be putting files in here, if they delete it I want a set of defaults to play so the system doesnt break |
| 04:33.17 | phix | They will be making their own recordings by typing in a number, listening to prompt, recording, then using samba copy and rename the file to the correct name and still it in their business directory |
| 04:33.58 | phix | still = store |
| 04:34.34 | [TK]D-Fender | Then cover for it whatever way you'd like |
| 04:34.38 | phix | I mean I could create a full blown webapp or equiv to do this, but the end user is cheap and wants the bare basis |
| 04:34.41 | phix | basics |
| 04:35.03 | phix | [TK]D-Fender: and using macros sounds resonable? or should i be doing this another way? |
| 04:40.34 | [TK]D-Fender | Whatever works |
| 04:51.21 | phix | I see |
| 04:51.27 | phix | How would you do it? |
| 04:52.52 | [TK]D-Fender | Shoot the arsonist :) |
| 05:08.25 | phix | That makes sense :) |
| 05:09.33 | *** part/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 05:11.23 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 05:30.13 | p3nguin | I think I would create these fallback items within the script that is requiring user interaction. |
| 05:30.21 | p3nguin | Let dial plan work like dial plan works. |
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| 05:58.43 | phix | Is there a GotoIf that allows matching a regex pattern? |
| 05:58.54 | phix | expression I can use in a gotoId even |
| 05:58.58 | phix | GotoIf |
| 06:00.52 | phix | eg, GotoIf(${SIP_HEADER("To")} ~= /1234567@someDomain.tld/)?s-1234567) |
| 06:01.11 | phix | (or does that perl type syntax work?) |
| 06:01.50 | phix | haha nm, I just found the section in the docs |
| 06:09.59 | phix | How's this look? s,1,gotoIf(${SIP_HEADER("To")} =~ "1234567@someDomain.tld" ? 1234567,1) |
| 06:10.23 | kaldemar | bad |
| 06:10.28 | phix | :( |
| 06:10.34 | phix | what did I do wrong? |
| 06:10.47 | kaldemar | GotoIf($[]?) |
| 06:10.50 | phix | ah |
| 06:11.17 | phix | How's this look? s,1,gotoIf($[${SIP_HEADER("To")} =~ "1234567@someDomain.tld"] ? s-1234567,1) |
| 06:12.00 | phix | I am assuming else means read the next line in the extension |
| 06:12.09 | phix | (if I dont specify it) |
| 06:13.43 | phix | ok if I am testing ${SIP_HEADER("To")} but the field is not set is the field blank, NULL or some special noun or is an error thrown? |
| 06:13.49 | kaldemar | looks better, but remove spaces around ? and quotes around the assumed regex. |
| 06:14.13 | phix | I like my quotes and spaces for readability, will it still match? |
| 06:14.18 | kaldemar | quote it, then you won't get an error. |
| 06:14.55 | phix | can I quote it when SIP_HEADER("To") already has a quote in it? |
| 06:15.12 | kaldemar | try it, on some occasions extra spaces have caused unexpected behavior. |
| 06:15.17 | phix | ah ok |
| 06:16.13 | kaldemar | i meant "${SIP_HEADER("To")}" |
| 06:16.27 | *** join/#asterisk jkroon (~jkroon@dsl-244-34-207.telkomadsl.co.za) |
| 06:23.02 | phix | kaldemar: yes so the " before the $ wont close at the next occorence of " (that is the one before the T in To) |
| 06:31.30 | phix | hmmm how does one dial a number from asterisk console in 1.6? Looks like it was changed |
| 06:32.04 | phix | ah append console to it |
| 06:34.25 | Henchman21 | anyone familair with spa3000 |
| 06:35.12 | Henchman21 | im trying to get asterisk to dial out its FXS |
| 06:35.25 | Henchman21 | and im failing miserably |
| 06:40.42 | *** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2) |
| 06:40.42 | jpsharp | You should just be able to use a plain dial command for it. Dial(SIP/18005551212@spa300) |
| 06:41.05 | jpsharp | Treat it as a single port SIP trunk. |
| 06:41.58 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 06:48.28 | Henchman21 | well like both ports register to asterisk sip |
| 06:49.18 | Henchman21 | i can call the fxo "port" and send fxo calls to asterisk but when i try to send a number to fxo it craps |
| 06:49.25 | Henchman21 | er fxs |
| 06:49.29 | Henchman21 | damn |
| 06:50.03 | Henchman21 | now my configs are alls screwing trying different things |
| 06:51.43 | ChannelZ | You dial out the FXO, an analog phone connects to the FXS. But the SPA doesn't call it that anyway |
| 06:52.49 | Henchman21 | calls the FXO line1 and FXS PSTN |
| 06:53.28 | Henchman21 | right FXS produces rung voltage to phoines? |
| 06:53.40 | ChannelZ | Yes. But you are backwards |
| 06:53.48 | Henchman21 | crap |
| 06:53.57 | ChannelZ | Line 1 is the FXS - what you'd connect an analog phone into |
| 06:54.09 | Henchman21 | produces the ring voltage |
| 06:54.11 | Henchman21 | ok |
| 06:54.15 | ChannelZ | PSTN is the FXO - what you'd connect a telephone line to |
| 06:54.42 | Henchman21 | i have the FXO in sip.conf it picks up calls and sends them to asterisk fine, but i cant get it dialout |
| 06:54.52 | ChannelZ | How so |
| 06:54.56 | Henchman21 | exten => _9.,1,Dial(SIP/pstn/${EXTEN:1},20,r) |
| 06:55.26 | ChannelZ | So what happens? |
| 06:55.37 | Henchman21 | drops |
| 06:55.47 | ChannelZ | can you be more vague? |
| 06:56.12 | Henchman21 | <PROTECTED> |
| 06:56.26 | Henchman21 | spa3000 is the fxs |
| 06:56.56 | ChannelZ | And where's the drop part? |
| 06:57.03 | Henchman21 | right after that |
| 06:57.11 | Henchman21 | nothing happens |
| 06:57.28 | Henchman21 | im like sooo close |
| 06:57.32 | ChannelZ | sighs |
| 06:57.49 | ChannelZ | Does it actually hang up the channel, or you just hear silence until you hang up, or what actually happens? |
| 06:58.01 | Henchman21 | hear a few fast beeps |
| 06:59.45 | ChannelZ | What kind of beeps |
| 06:59.55 | ChannelZ | Do they sound like telco tones or what |
| 07:00.19 | Henchman21 | yea |
| 07:00.25 | Henchman21 | <PROTECTED> |
| 07:00.51 | ChannelZ | a fast busy |
| 07:00.56 | Henchman21 | yea |
| 07:02.00 | ChannelZ | what are you dialing? |
| 07:02.08 | Henchman21 | a local number |
| 07:02.13 | Henchman21 | you know what dont worry about it |
| 07:02.24 | Henchman21 | ill figure it out |
| 07:03.10 | ChannelZ | Do you have Voip-to-PSTN Gateway enabled on the PSTN Line? |
| 07:03.22 | Henchman21 | yea |
| 07:03.41 | Henchman21 | just got it today its brand new :P |
| 07:04.02 | Henchman21 | first time interfacing asterisk to the house line |
| 07:05.54 | ChannelZ | Do you have an appropriate dial plan on 'Line 1'? |
| 07:06.31 | Henchman21 | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 07:06.33 | Henchman21 | default |
| 07:07.34 | ChannelZ | Do you get the fast busy immediately after dialing or does it hang? |
| 07:08.17 | ChannelZ | I'd turn verbose up a little (core set verbose 3) and make sure * is getting the number you think it's getting from the SPA. |
| 07:08.50 | ChannelZ | It depends on where the fast busy is coming from.. the SPA/Asterisk due to error or from the telco because you're dialing something invalid |
| 07:11.47 | ChannelZ | But I can't make any more guesses since you haven't really shown enough console output, or even said what number you're dialing. |
| 07:19.32 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
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| 07:33.42 | schmidts | good morning |
| 07:38.43 | kleszcz | morning |
| 07:39.30 | Henchman21 | get this the spa3k's voip -> pstn dialplan is (xx.<:@gw0>) |
| 07:40.18 | Henchman21 | i make a call its info page says looks legit in that if i picked up that house line and dialed the number it got it will complete |
| 07:48.12 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:49.28 | Henchman21 | this is driving me crazy im pretty sure everything is set up right |
| 07:50.54 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:52.36 | Henchman21 | ok get this |
| 07:53.42 | Henchman21 | i make the voip call to pstn insta drop, i call the pstn line from my cellphone and it picks up redials the number i tried dialing via voip and continues to route me to my test/asterisk extension of music on hold |
| 07:54.03 | Henchman21 | something is deff wrong |
| 07:54.14 | Henchman21 | and its totally not my fault |
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| 08:38.12 | v0lZy | hi guys |
| 08:38.19 | v0lZy | some problems with incoming calls |
| 08:38.46 | v0lZy | people dialing in just get a dead lnie |
| 08:45.01 | kaldemar | v0lZy: pastebin CLI output of a call. |
| 08:45.45 | v0lZy | kaldemar: im in cli now, what should i do? |
| 08:46.03 | v0lZy | Im trying with sip debug |
| 08:46.06 | v0lZy | but too many data |
| 08:46.12 | v0lZy | i think its something to do with routing sip calls |
| 08:46.42 | v0lZy | is there any way to limit sip debug or output it to a file or something? |
| 08:47.55 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
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| 08:50.33 | gladier | v0lZy: you can limit it to a specific peer |
| 08:50.45 | v0lZy | how? |
| 08:51.36 | gladier | sip debug peer xxxxx |
| 08:51.55 | gladier | sip set debug peer xxxxx * |
| 08:55.36 | v0lZy | http://pastebin.com/gquL1gS3 |
| 08:55.44 | v0lZy | darnit, i have to run |
| 08:55.48 | v0lZy | ill be back in 1 hour |
| 08:56.25 | kaldemar | v0lZy: paste a call trace then. you didn't this time. |
| 08:56.32 | v0lZy | how do i trace that? |
| 08:56.43 | kaldemar | enable the debug commands and make a call. |
| 08:56.55 | v0lZy | i did. |
| 08:56.56 | kaldemar | and pastebin all output. yes, all. |
| 08:57.04 | v0lZy | i did sip set debug peer 16 (my internal) |
| 08:57.13 | v0lZy | ah |
| 08:57.15 | kaldemar | there is no call in your pastebin. |
| 08:57.15 | v0lZy | ok |
| 08:57.18 | v0lZy | ill do that as soon as i get back |
| 08:57.24 | v0lZy | yeah thats the problem, no phone rings in this case... |
| 08:57.31 | v0lZy | but yeah, ill do it again once im back |
| 08:57.32 | v0lZy | 1 hour |
| 08:57.34 | kaldemar | and don't limit the debug to a phone of yours. |
| 08:57.40 | v0lZy | oh... |
| 08:57.45 | kaldemar | it is not the point of interest. |
| 08:57.49 | v0lZy | (brb) |
| 08:57.58 | kaldemar | actually, don't limit the debug at all. |
| 08:58.03 | kaldemar | just use "sip set debug on". |
| 08:59.06 | *** join/#asterisk krotos (~d3v1l@host21-96-dynamic.11-79-r.retail.telecomitalia.it) |
| 08:59.10 | krotos | hi all :) |
| 09:02.26 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 09:02.39 | zkn | Hi, how to interpret the following error message: chan_sip.c:19829 handle_response_invite: just did sched_add waitid(1011820) for sip_reinvite_retry for dialog QFib4qT9EIWAN2oSo4GYc4mLfNxPoyvu in handle_response_invite |
| 09:02.54 | zkn | what does it mean? |
| 09:39.30 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 09:39.33 | v0lZy | ok im back |
| 09:40.13 | v0lZy | kaldemar, gladier: are you here? |
| 09:41.39 | v0lZy | kaldemar, gladier: for some reason its working perfectly normal now. I havent touched anything at all, i wasnt even in the office. |
| 09:41.50 | v0lZy | kaldemar, gladier: but now calls come through normally |
| 09:42.29 | v0lZy | What could be the issue for people spontaneously not being able to call in and then being able to call in? |
| 09:45.56 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 09:49.17 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 09:50.29 | kaldemar | v0lZy: crappy network, provider, pbx, phones or whatever. |
| 09:51.23 | v0lZy | i think we can rule out the network provider |
| 09:51.33 | v0lZy | i see stuff coming in on my firewall |
| 09:52.06 | v0lZy | could this be a NAT issue? theres 1 nat (mine) between me and the provider |
| 09:53.11 | kaldemar | hard to say since you haven't really provided any proper information on the symptoms, but yes, your router might be the issue as much as any other piece of the puzzle. |
| 09:55.32 | v0lZy | kaldemar: when I do debug on |
| 09:55.39 | v0lZy | i get a lot of stuff |
| 09:55.48 | v0lZy | so much it actually flies out of my terminal history |
| 09:55.55 | v0lZy | is there a way to redirect to a file or something? |
| 09:56.32 | v0lZy | or alternatively |
| 09:56.37 | v0lZy | i can unplug all the other phones or something' |
| 09:57.45 | kaldemar | v0lZy: for example "asterisk -vvvvvr | tee output.txt" |
| 09:58.24 | kaldemar | v0lZy: or "script" or some other tool that grabs output to a file. |
| 09:58.43 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 09:58.48 | v0lZy | i think it might work with tee |
| 09:58.50 | v0lZy | i have it here |
| 09:59.09 | v0lZy | ok... so lets see how would i get it to not work agian so we can trouble shoot where the issue is... |
| 09:59.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:00.03 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 10:01.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:03.51 | v0lZy | ok |
| 10:04.00 | v0lZy | theres about a 1000 lines of text in a few seconds |
| 10:04.08 | v0lZy | what would i be looking for with an incoming call? |
| 10:04.26 | kaldemar | you need to know SIP to be able to look for it. |
| 10:05.01 | kaldemar | if you don't, someone else can take a look at it if you pastebin the output. |
| 10:05.49 | v0lZy | well |
| 10:05.52 | v0lZy | this is now that its working |
| 10:05.53 | v0lZy | http://pastebin.com/s9FKPwUe |
| 10:06.43 | kaldemar | no use looking at a working call, is there? |
| 10:07.00 | kaldemar | you also need output for a call that does not work. |
| 10:10.17 | v0lZy | I dont know how to get it to not work |
| 10:10.54 | kaldemar | is that your output from CLI? |
| 10:11.13 | v0lZy | ok |
| 10:11.16 | v0lZy | now i can call out |
| 10:11.27 | v0lZy | cant here the person im calling |
| 10:11.31 | v0lZy | seems more and more like a firewall thing |
| 10:11.50 | v0lZy | kaldemar: its the next best thing i can do, not exactly cli, but i have a 'log' |
| 10:12.00 | v0lZy | that updates live .. so sort of like a cli |
| 10:12.04 | v0lZy | i can ssh in |
| 10:12.08 | v0lZy | and do the cli thing |
| 10:12.20 | v0lZy | but have to put it into | tee output.txt or something |
| 10:12.25 | v0lZy | saving that for when it doesnt wokr |
| 10:12.54 | kaldemar | logs are a huge pain to read because of stamping. |
| 10:14.30 | v0lZy | ill try and get a cli |
| 10:14.32 | v0lZy | btw |
| 10:14.37 | v0lZy | right now im in a situation where i can call outside |
| 10:14.41 | v0lZy | but i only get 1 beep |
| 10:14.44 | v0lZy | then no ringtone |
| 10:14.50 | v0lZy | while the phone im calling is ringing |
| 10:14.54 | v0lZy | if i answer |
| 10:14.56 | v0lZy | one way audio |
| 10:15.18 | v0lZy | now its ringing normally.. |
| 10:15.31 | v0lZy | and 2 way audio... |
| 10:36.15 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:44.17 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 10:46.02 | *** join/#asterisk jkroon (~jkroon@dsl-244-34-207.telkomadsl.co.za) |
| 10:47.05 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 10:51.59 | v0lZy | ok |
| 10:52.00 | v0lZy | right now |
| 10:52.02 | v0lZy | i have a call initiated |
| 10:52.14 | v0lZy | where i can send audio aut |
| 10:52.19 | v0lZy | but i dont get any audio in |
| 10:52.31 | *** part/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 10:52.38 | v0lZy | i initiated the call |
| 10:52.41 | v0lZy | so .. outgoing |
| 10:52.47 | v0lZy | how best to debug this? |
| 10:55.58 | v0lZy | kaldemar: i captured this: http://pastebin.com/wWptariW |
| 10:58.31 | v0lZy | this looks more and more like a nat issue to me |
| 11:00.42 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 11:03.09 | kaldemar | v0lZy: that looks like a few SUBSCRIBE messages, not a call. |
| 11:03.38 | v0lZy | i first ran the call |
| 11:03.45 | v0lZy | when it got through and i noticed i only have 1 way audio |
| 11:03.53 | v0lZy | i started the debuging on the peer i was calling from |
| 11:03.55 | v0lZy | thats all i got |
| 11:06.05 | kaldemar | v0lZy: completely useless. you need to grab debug during a call, bot after it. |
| 11:06.09 | kaldemar | s/bot/not/ |
| 11:06.59 | v0lZy | during a call u mean... as it rings? |
| 11:07.13 | v0lZy | or while the conversation is ongoing' |
| 11:07.22 | v0lZy | (i kind of get the when it rings part.. .sip headers and all) |
| 11:08.12 | kaldemar | during a whole call. setup, bridging, hangup. a whole call. |
| 11:09.21 | v0lZy | i guess my best go then is to simply disconnect all other phones |
| 11:09.31 | v0lZy | theres way too much traffic here im looking at |
| 11:10.42 | v0lZy | i suppose the easiest way to do that is to change the switch... |
| 11:10.43 | v0lZy | hold on |
| 11:11.04 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 11:11.07 | *** join/#asterisk cstachris (~chrismylo@114.73.81.192) |
| 11:12.22 | v0lZy | now im 1 switch short |
| 11:12.26 | v0lZy | damnation |
| 11:12.47 | v0lZy | brb, gotta go buy one |
| 11:15.07 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 11:19.53 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 11:23.18 | jkroon | hi all, is it possible to Read() without a channel being Up? |
| 11:24.18 | jkroon | i'm looking for a way to deal with requesting pin codes, making the channel go into an Up state causes problems, so I was thinking of sending Progress() and then using Read() with the channel in a Down state - but not sure if this is possible. The channel will ALWAYS be a SIP channel. |
| 11:25.05 | Dovid | anyone know if I can use Read in an agi with cepstral or some other way to use cepstral from an agi and get user input? |
| 11:26.41 | kaldemar | Dovid: you can use every application from AGI. |
| 11:27.05 | *** join/#asterisk FainaUkraina (~BuenGenio@221.125.202.119) |
| 11:27.39 | jkroon | nm, can pass the "n" option to Read |
| 11:28.23 | v0lZy | back |
| 11:28.38 | v0lZy | now to unplug all the phones... |
| 11:30.53 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 11:32.37 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 11:33.38 | Dovid | kaldemar: I know i can and I have. what i want to do is get DTMF input based on a cpestral message from the AGI |
| 11:33.45 | Henchman21 | man this spa3k is a piece of junk |
| 11:34.05 | Henchman21 | cant get it to dialout for crap, least it answers the phone and sends it to asterisk |
| 11:34.46 | Henchman21 | all i really wanted |
| 11:34.52 | Henchman21 | to fuck with telemarketers |
| 11:35.45 | Henchman21 | :P |
| 11:36.01 | kaldemar | Dovid: and? |
| 11:43.23 | *** join/#asterisk FainaUkraina (BuenGenio@221.125.202.113) |
| 11:45.35 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:48.24 | *** join/#asterisk cadey (5a980315@gateway/web/freenode/ip.90.152.3.21) |
| 11:50.00 | cadey | Hi guys, any one in here close to the south west of london (UK, surry) area with a good dev (php and .net) experience looking for work in relation to an asterisk project? |
| 11:58.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:58.43 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 11:59.23 | v0lZy | kaldemar: im reading an article |
| 11:59.43 | v0lZy | it mentions canreinvite |
| 11:59.46 | Chainsaw | leifmadsen: Any chance of the patch on https://issues.asterisk.org/view.php?id=18010 being applied now that there is a test report and detailed bug linked in? |
| 11:59.59 | v0lZy | if i have that set to "no", all calls are routed through the pbx, right? |
| 12:02.58 | *** join/#asterisk MarKsaitis (~MarKsaiti@88.96.60.78) |
| 12:08.12 | kaldemar | v0lZy: routed is a term that does not apply here. |
| 12:10.16 | v0lZy | ... the pbx relays for the phones. |
| 12:10.22 | v0lZy | btw... |
| 12:10.24 | v0lZy | i have |
| 12:10.33 | v0lZy | nat=yes in my sip.config for each number |
| 12:10.41 | v0lZy | but i dont see a canreinvite=no |
| 12:10.51 | v0lZy | are these 2 synonyms or separate settings? |
| 12:11.56 | v0lZy | ok seems these canreinvite thing and direct media changes names a few times |
| 12:12.57 | kaldemar | canreinvite was renamed to directmedia in some version. nat is a different setting. |
| 12:13.22 | v0lZy | ok |
| 12:13.33 | v0lZy | i have directmedia=no in teh context of each extension |
| 12:13.41 | v0lZy | but not in the context of general or provider |
| 12:20.06 | v0lZy | hm.... |
| 12:24.31 | v0lZy | kaldemar: the tee command didnt caputre anything... |
| 12:24.40 | v0lZy | or i used it wrong ithink |
| 12:24.43 | v0lZy | hold on, i try agani |
| 12:25.58 | v0lZy | eh |
| 12:26.03 | v0lZy | teh call doesnt even hang up when i hang up |
| 12:28.35 | v0lZy | http://pastebin.com/GepqstMv |
| 12:28.55 | v0lZy | this is what i captured |
| 12:28.57 | v0lZy | me calling out |
| 12:29.14 | v0lZy | and not hearing the audio from the number i dialed |
| 12:30.05 | *** join/#asterisk b14ckp34r1 (~PAPER@58.236.243.130) |
| 12:30.16 | *** part/#asterisk b14ckp34r1 (~PAPER@58.236.243.130) |
| 12:36.17 | kaldemar | v0lZy: sip debug is missing. there is no useful information in your pastebin. |
| 12:38.26 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:44.23 | *** join/#asterisk coppice (~coppice@m121-203-211-131.smartone.com) |
| 12:47.00 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 12:47.35 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 12:52.36 | *** join/#asterisk RickCogley (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 12:58.27 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 13:00.18 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 13:01.16 | AdvoWork | Anyone here ever set up so an extension can accept faxes on trixbox? I've got an extension set up, and have just setup an inbound route, which does the fax detection(system) and type nvfax and zaptel, neither work, it just rings out. Any suggestions please? |
| 13:03.31 | akrohn | we can't support trixbox =/ |
| 13:04.19 | WIMPy | And for zaptel you're 4 years late or so. |
| 13:04.24 | WIMPy | ~trixbox |
| 13:04.24 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 13:04.30 | *** join/#asterisk l0st-soul (~nqezhw@nat.msfsupply.be) |
| 13:04.34 | l0st-soul | hello |
| 13:05.05 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 13:05.08 | v0lZy | ok im back |
| 13:05.25 | v0lZy | theres way too much information in this debug and all stuff |
| 13:05.26 | v0lZy | i need to disconnect other phones |
| 13:05.28 | v0lZy | and try again |
| 13:05.30 | v0lZy | hold on |
| 13:05.37 | l0st-soul | i'm looking for a PRI gateway (not a card) but i'm lost in all the brands and models .. Can anyone point me to a known asterisk-friendly model (with 1 PRI port) ? |
| 13:06.14 | l0st-soul | i had a look at patton but they seem to have only models with multiple ports |
| 13:06.15 | WIMPy | l0st-soul: Are you sure you don't want a card? |
| 13:07.09 | l0st-soul | WIMPy: i have a card already, but i will need to handle a 2nd PRA in another city, and i'd like to avoid having 2 asterisk |
| 13:07.15 | beardy | We run Patton. They seem to work. (I haven't fiddled with them much. Linuxbased, and not much whining from customers running them so..) |
| 13:07.37 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:07.56 | l0st-soul | i have a patton gw for regular pstn already, and it works quite well.. Only, i dont find any 1-port patton PRI gw |
| 13:10.11 | v0lZy | ok |
| 13:10.17 | v0lZy | disconnected all othe rphones but this one |
| 13:10.20 | v0lZy | now ill give it a shot |
| 13:12.00 | l0st-soul | i take back what i said, it seems like patton smartnode 4120 does the job |
| 13:12.17 | l0st-soul | uh no - its not PRI |
| 13:12.55 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:13.14 | *** part/#asterisk cstachris (~chrismylo@114.73.81.192) |
| 13:13.31 | v0lZy | ...!="!%=(!%!( |
| 13:13.36 | v0lZy | and now it works normally... |
| 13:13.45 | *** join/#asterisk RickCogley_ (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 13:13.57 | v0lZy | apparently at random |
| 13:14.37 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:19.40 | TSM | im using asterisk 1.4, ive just downloaded a core audio pack, all the files are in sln16 format and sound all slowed down, ive got other files but they are only sln, what is the diffrence? can they be converted or asterisk made to play them correctly |
| 13:20.07 | v0lZy | finally i have the darn file |
| 13:21.14 | v0lZy | http://pastebin.com/yVQLXbUm |
| 13:21.17 | v0lZy | kaldemar: http://pastebin.com/yVQLXbUm |
| 13:21.22 | v0lZy | this is what i captured |
| 13:21.29 | v0lZy | i was able to have a conversation calling out.. normally |
| 13:21.31 | v0lZy | butcouldnt call in |
| 13:21.42 | v0lZy | whats worse, doesnt seem like there was an incoming call at all |
| 13:22.39 | mirelab | Does anyone know if I can trigger en Event (call a curl function or shell script) when * sends an Unregistered sip trunk NOTICE ? |
| 13:24.09 | mirelab | when NOTICE[4068] chan_sip.c: Peer '...' is now UNREACHABLE! Last qualify: 41 |
| 13:24.25 | WIMPy | mirelab: Listen on AMI. |
| 13:24.35 | mirelab | message appears but without changing sip.c |
| 13:25.02 | mirelab | WIMPy: Yes that is 1 of possibilities |
| 13:25.43 | mirelab | WIMPy: I can grep logs too, but i was hoping there is a elegant solution that doesn't use too much resources :) |
| 13:26.05 | WIMPy | Then go for AMI. |
| 13:26.32 | v0lZy | hi WIMPy |
| 13:27.01 | mirelab | both AMI and grepping logs trough cron , or sending calls that will return PEERSTATUS=OK* can be too demanding :| |
| 13:27.02 | WIMPy | hi v0lZy |
| 13:27.22 | WIMPy | In what way? |
| 13:30.05 | *** join/#asterisk Faustov (madrid@gentoo/user/faustov) |
| 13:31.29 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.35.63) |
| 13:35.00 | kaldemar | v0lZy: have you forwarded UDP 5060 and 10000-20000 in your router to your asterisk box? |
| 13:35.10 | v0lZy | kaldemar: can you throw an eye at my pastebin? http://pastebin.com/eSAxZ9h2 The first call was from outside in, it went ok, me calling out, I dont ge tthe audio from the person i called |
| 13:35.38 | v0lZy | kaldemar: i forwarded UDP 5060 and 10000 to 10020 |
| 13:36.13 | v0lZy | and restircted my pbx to advertising from 10000 to 10020 |
| 13:36.28 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:38.17 | v0lZy | right now i can call in |
| 13:38.22 | v0lZy | and when i do, i can talk fine |
| 13:38.25 | v0lZy | but when i call out |
| 13:38.53 | v0lZy | i don hear the phone number ringing (just a quick chirp) then i dont hear anything from the called number at all, but they hear me well |
| 13:39.19 | v0lZy | the more i consider it, the more it seems like a firewall issue to me |
| 13:40.04 | *** join/#asterisk albertoandrade (~albertoan@186.212.200.89) |
| 13:41.26 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:43.32 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 13:45.28 | kaldemar | v0lZy: your provider seems to be using a different address for SIP and RTP. |
| 13:45.50 | v0lZy | yes |
| 13:46.00 | v0lZy | its using 1 ip for rtp traffic and 1 for sip traffic |
| 13:46.29 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 13:47.26 | kaldemar | check with "rtp set debug on" that you're getting RTP from the provider, and confirm the address it comes from. |
| 13:49.53 | v0lZy | i actually dont see getting rtp packets from them |
| 13:49.58 | v0lZy | just me sending rtp packets to them |
| 13:50.02 | v0lZy | and getting them from my phone |
| 13:50.24 | v0lZy | they are either sending to my ip |
| 13:50.28 | v0lZy | (internal) |
| 13:50.32 | v0lZy | or somethings wrong with the prot |
| 13:50.35 | v0lZy | port* |
| 13:56.15 | kaldemar | your asterisk told them to send to port 10020. |
| 13:56.37 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ueynovhexhwgnckz) |
| 13:56.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:59.00 | v0lZy | ok.. WTF.. now its working again |
| 13:59.05 | v0lZy | yeah |
| 13:59.08 | v0lZy | thats what its supposed to do |
| 13:59.19 | v0lZy | i told my asterisk to advertise 10000 to 10020 |
| 13:59.46 | *** part/#asterisk mintee (~mintee@2001:470:7:a41::2) |
| 13:59.46 | v0lZy | funny thing though, right now it works both ways |
| 14:00.45 | v0lZy | and just now |
| 14:00.47 | v0lZy | no outgoing |
| 14:00.57 | [TK]D-Fender | v0lZy, * can use up to 4 port per single call and ports may not free up as fast as you'd like. You are asking for trouble offering so few |
| 14:02.05 | v0lZy | hm |
| 14:02.14 | v0lZy | i figured i only need 10 concurrent calls... |
| 14:02.23 | v0lZy | and no video |
| 14:02.26 | v0lZy | so just voice |
| 14:02.37 | v0lZy | i was under the assumption that in this case, i use only 2 ports at the same time |
| 14:02.39 | v0lZy | btw |
| 14:03.05 | v0lZy | Feb 28 15:00:59 WAN ISP'sIP:12566 myIP:10004 UDP |
| 14:03.08 | v0lZy | in my firewall |
| 14:03.09 | v0lZy | as passed |
| 14:03.25 | v0lZy | so.... source being port 12566 |
| 14:03.46 | v0lZy | and my internal port that translated to being 10004 |
| 14:03.56 | v0lZy | by the way things are set up |
| 14:03.59 | v0lZy | this shouldnt happen? |
| 14:04.59 | v0lZy | im not allowing port 12566 |
| 14:05.05 | v0lZy | nothing above 10020 |
| 14:06.05 | *** join/#asterisk serafie (~erin@nat/digium/x-venkhymawzxmupcc) |
| 14:06.21 | [TK]D-Fender | v0lZy, I suggest you stop choking your system so much |
| 14:06.49 | v0lZy | ok, so, expand port range |
| 14:07.01 | v0lZy | to ... 10000 - 10200 ? |
| 14:08.09 | [TK]D-Fender | far better already... |
| 14:08.35 | v0lZy | u think that was the issue? |
| 14:09.00 | [TK]D-Fender | v0lZy, I don't see the real debug. |
| 14:09.09 | [TK]D-Fender | v0lZy, I just know that was a psycho range |
| 14:10.55 | v0lZy | funny thing it worked with that |
| 14:10.59 | v0lZy | now i increased the range |
| 14:11.11 | v0lZy | it doesnt even hookup to the provider |
| 14:14.43 | v0lZy | which is spooky. |
| 14:15.31 | *** join/#asterisk MarKsaitis (~MarKsaiti@88.96.60.78) |
| 14:16.57 | v0lZy | [TK]D-Fender: if i want 10 concurrent lines |
| 14:17.03 | v0lZy | how many ports would you recommend? |
| 14:17.05 | [TK]D-Fender | v0lZy, Stop asking for trouble... |
| 14:17.20 | [TK]D-Fender | v0lZy, You are sitting in here failig constantly and TRYING to make life even harder |
| 14:17.25 | [TK]D-Fender | v0lZy, STOP |
| 14:17.54 | v0lZy | ... well thanks. |
| 14:18.26 | [TK]D-Fender | v0lZy, I recommend just leaving the nice and healthy 10000-20000 range the sample config shows like the vast majority of us |
| 14:19.11 | v0lZy | It didnt work with that range ... im using pfsense as a router, and it probably has something to do with outgoing port randomization. |
| 14:19.24 | v0lZy | (or at least that is what google hinted at) |
| 14:19.35 | v0lZy | so i dropped the port range and it started working. |
| 14:20.28 | kaldemar | change your router to something that works or configure it to behave. |
| 14:21.02 | [TK]D-Fender | pfsense(less) |
| 14:21.27 | v0lZy | well, there is an option to disable this outgoing port randomization thing i think |
| 14:21.56 | v0lZy | its tied to outbound NAT... |
| 14:22.00 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:22.30 | kaldemar | v0lZy: if pfsense changes the port from what is in the SDP, things will break. |
| 14:22.45 | kaldemar | v0lZy: so disable such stupidity or change the router. |
| 14:23.46 | kaldemar | you seem to have random issues with both inbound and outbound calls now. |
| 14:24.00 | v0lZy | right now no issues with call, issues hooking up to the provider. |
| 14:24.16 | v0lZy | or better yet |
| 14:24.24 | v0lZy | it just hooked up |
| 14:24.34 | v0lZy | seems it needs a reboot to write to smoe |
| 14:24.36 | v0lZy | some files* |
| 14:27.17 | v0lZy | however.. what does the rewrite matter? I figure if i open port X, and pass a sip message through whatever randomized port that port X is open.. and packets get sent to port X ... |
| 14:28.29 | [TK]D-Fender | SIP != RTp |
| 14:28.47 | leifmadsen | I have pfsense, and I've never had a router software that has so many problems dealing with multiple SIP phones behind it |
| 14:28.54 | leifmadsen | I'll be removing it as soon as I have time |
| 14:28.58 | [TK]D-Fender | And it's been over an hour... and I haven't seen a call. |
| 14:29.27 | v0lZy | well i can call out fine |
| 14:29.30 | kaldemar | [TK]D-Fender: it's been six hours. |
| 14:29.33 | v0lZy | calling in right now doesn twork. |
| 14:29.41 | leifmadsen | v0lZy: the only way to get it close is to set the firewall to manual mode, and to set the rule state to static |
| 14:29.42 | v0lZy | its been a few days actually |
| 14:29.56 | v0lZy | ill try that liefmadsen, thanks |
| 14:30.05 | leifmadsen | either way, it's not an Asterisk issue -- you're going to have to figure out how to configure pfsense to act nicer, and when you do, let me know, because I tried everything. |
| 14:30.24 | leifmadsen | m0n0wall worked fine |
| 14:30.26 | leifmadsen | btw |
| 14:30.31 | v0lZy | have to run. |
| 14:30.41 | leifmadsen | it's the way pfsense does its nat and firewall; it's not voip friendly |
| 14:30.45 | leifmadsen | imho |
| 14:30.52 | v0lZy | ok il set it to static |
| 14:30.58 | v0lZy | will test tomorrow, let you know how it goes |
| 14:31.09 | v0lZy | thanks for the help kaldemar, [TK]D-Fender, leifmadsen |
| 14:31.15 | v0lZy | gettin gthere bit by bit. |
| 14:31.24 | v0lZy | actually.. day fter tomorrow, argh. |
| 14:31.26 | v0lZy | later |
| 14:33.29 | shihan | howdy folks, i have a question about the ami side of asterisk, if a call is parked, is there a way to reset/change its timeout? |
| 14:34.46 | shihan | so far the only thing i've come up with is transfer it to an extension, force it to answer then repark it with a new timeout |
| 14:40.12 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 14:41.15 | [TK]D-Fender | shihan, pretty much |
| 14:43.21 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 14:44.18 | shihan | ahhh, fair enough then :) |
| 14:44.27 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
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| 14:46.34 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:46.44 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 14:47.36 | *** join/#asterisk ChIlEnU (c8a21584@gateway/web/freenode/ip.200.162.21.132) |
| 14:47.46 | mirelab | Is there a way to trigger an Event (call some script or curl func) when NOTICE[4068] chan_sip.c: Peer '...' is now UNREACHABLE! is triggered |
| 14:47.58 | *** part/#asterisk ChIlEnU (c8a21584@gateway/web/freenode/ip.200.162.21.132) |
| 14:48.35 | fireman_biff | I'm checking out Asterisk appliances for several offices with between 10 - 20 each. Any recommendations? So far I'm starting off by looking at what Xorcom and Digium have to offer. |
| 14:52.00 | [TK]D-Fender | mirelab, vi chan_sip.c |
| 14:55.37 | leifmadsen | or use some sort of log analyzer to trigger things when events happen |
| 14:56.10 | leifmadsen | (fail2ban does that kind of thing with iptables for example) |
| 15:00.32 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 15:00.51 | mirelab | [TK]D-Fender: thx :) |
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| 15:05.20 | *** join/#asterisk anny__ (~chatzilla@77.42.241.114) |
| 15:06.26 | anny__ | there's a registered sip user always showing in my asterisk box, probably because the corresponding client didn't properly un-register his account, how can i unregister him from Asterisk |
| 15:08.38 | akrohn | maybe a silly question, but is the phone's registration entry still in sip.conf? |
| 15:09.24 | [TK]D-Fender | akrohn, "registration entry"? |
| 15:09.49 | akrohn | [TK]D-Fender, umm... idk? is it called a context in reference to a peer? |
| 15:11.29 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:11.43 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:13.18 | akrohn | [TK]D-Fender, OH! i was replying to anny__ 's question! |
| 15:13.47 | [TK]D-Fender | And i'm wondering what you're referring to. there is no "registration entry" for phones in sip.conf |
| 15:13.48 | akrohn | did not specify |
| 15:14.38 | akrohn | hmm... I have a context for each sip phone in an include file that is included in sip.conf ... what terminology should i call that? or is it in the wrong place? |
| 15:15.12 | [TK]D-Fender | akrohn, There is nothing about "registration" there |
| 15:16.19 | shihan | anny: you might want "sip unregister" from the cli |
| 15:20.32 | *** join/#asterisk grandpapadot (~grandpapa@70.148.53.190) |
| 15:22.40 | grandpapadot | Hey guys, I have a sip server that does nothing but place and receive calls, only a single dialplan context, really simple. My question is, there is the possibility of having a couple thousand sip peers, not all being used, probably only about 1/3 in use, but they will exist in my sip.conf and in asterisk, think asterisk 1.8 will handle this gracefully without any major performance issues |
| 15:22.41 | grandpapadot | ? (quad core + 4gb + 64-bit centos) |
| 15:23.17 | grandpapadot | When I say not all being used, I mean not all registered. At no point in time will there be more than 50-100 simultaneous calls. |
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| 15:27.00 | [TK]D-Fender | grandpapadot, Depends on call load. |
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| 15:32.57 | shihan | heh, i would of thought an atom would handle that kind of load myself :) |
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| 16:03.23 | *** join/#asterisk wiit (~qtt@213.87.240.64) |
| 16:03.32 | wiit | hi all! |
| 16:04.34 | wiit | have a question: there is an eventfilter option in manager. Should it work in Asterisk 1.8? |
| 16:09.00 | Qwell | wiit: sure, why not? |
| 16:11.42 | wiit | I don't know, but it doesn't work:( I have Event:FullyBooted at every login to AMI and I try to filter it. I added eventfilter for this event in my manager.conf, but nothing changed. |
| 16:13.32 | wiit | and one more question: what does that event mean? |
| 16:14.11 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 16:15.24 | chuckf | wiit: are you saying that you want an event, one that you don't know what it is or does, filtered in a particular way? |
| 16:16.23 | Qwell | wiit: I doubt you can filter that specific event. You really aught to be checking that one, before continuing. |
| 16:17.13 | wiit | slaps chuckf around a bit with a large trout |
| 16:18.49 | Qwell | problem solved, I guess? |
| 16:19.13 | chuckf | I guess slapping people with trouts solves the world |
| 16:19.43 | shihan | i wish i knew this many years ago |
| 16:26.07 | *** join/#asterisk Deeewayne (~dwayne@c-76-29-230-45.hsd1.al.comcast.net) |
| 16:26.07 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:35.18 | saxa | p3nguin: after i removed that Answer() I'm expecting a ring and then the line drops off. |
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| 16:59.29 | Qwell | throws a taco at _Corey_ |
| 16:59.36 | _Corey_ | ducks |
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| 17:33.12 | *** join/#asterisk matelko (~ABC@unaffiliated/matelko) |
| 17:34.45 | matelko | Hi. I want to buy Digium card for Asterisk PBX - It would be hybrid card. It will be possible to record selective calls with it or I need some type of card? |
| 17:35.05 | Qwell | matelko: What kind of connectivity do you have? |
| 17:35.21 | Qwell | I misread your question. No, recording doesn't require hardware. |
| 17:35.22 | matelko | I will have ISDN or maybe analog |
| 17:36.18 | matelko | Qwell - I dont know If it is possible to record calls incoming calls from ISDN inside PBX allways or I need some ISDN card type. |
| 17:37.38 | matelko | So it is allways possible to record calls? I take it as yes. Am I right 100%? |
| 17:38.14 | Qwell | matelko: correct |
| 17:38.34 | matelko | good :) |
| 17:38.44 | Qwell | There is very little that Asterisk can't do all by itself. |
| 17:38.47 | [TK]D-Fender | matelko, Recording isn't dependant on a card. Any kind of call going through * can be recorded |
| 17:39.26 | matelko | I see. One more question for now - It is possible to do it with Asterisk GUI too or I need do it manually from command line? |
| 17:40.14 | [TK]D-Fender | AsteriskGUI (the Digium one) may have some features but its development is at a near stand-still and lacks a lot of polish. FreePBX is the most mature free one |
| 17:40.24 | [TK]D-Fender | Then there are commercial GUI's available |
| 17:41.03 | matelko | So its better to use FreePBX? It is problemless with Digium cards? |
| 17:41.13 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 17:42.36 | matelko | ok, I will start with Free PBX. |
| 17:42.59 | matelko | Thank you all for answers. |
| 17:43.07 | matelko | Bye. |
| 17:44.06 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 17:49.12 | *** part/#asterisk damjan (~damjan@unaffiliated/damjan) |
| 17:50.55 | *** join/#asterisk tmrhmdv (457279e9@gateway/web/freenode/ip.69.114.121.233) |
| 17:58.11 | tmrhmdv | Hi everyone! I'm having problem connecting to my asterisk account over TCP, I have enabled TCP and its bindaddr but getting: |
| 17:58.12 | tmrhmdv | KERBEROS + NTLM, Default Creditals, Cannot establish SSL tunnel to HTTP proxy. |
| 17:58.23 | Qwell | http proxy? O.o |
| 17:58.27 | tmrhmdv | Do I have to tell where my SSL certificates are located in? |
| 17:58.33 | Qwell | yes |
| 17:59.10 | tmrhmdv | OK, yes, the softphone is saying "HTTP proxy" |
| 18:00.51 | tmrhmdv | add tlscertfile and tlscafile in sip.conf only? |
| 18:02.06 | Qwell | What do the docs say? |
| 18:02.58 | Qwell | https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport |
| 18:03.43 | tmrhmdv | Thank you Qwell |
| 18:03.49 | tmrhmdv | sorry for being lazy |
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| 19:08.35 | p3nguin | saxa: I need more info. |
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| 20:03.49 | cvance | I am having some trouble with clock drift. I am running asterisknow as a virtual server on a xen hvm virtual machine. The clock keeps drifting about 15 minutes a day. I am running the ntpd service, but it is unable to synchronize the time. |
| 20:04.45 | [TK]D-Fender | problem is pretty much entirely due to your VM |
| 20:07.41 | cvance | I agree that the vm causes the drift. I am concerned why ntp doesn't keep the time correctly |
| 20:08.28 | Qwell | It has to be synced fully first. ntpd will not make large adjustments. |
| 20:08.40 | Qwell | If the drift is too large, ntpd will do nothing. |
| 20:09.53 | cvance | I have, I shut down the ntpd daemon, run ntpdate pool.centos.org, then restart the ntpd daemon, but drifts still occur. |
| 20:10.21 | cvance | sorry, ntpdate 0.centos.pool.ntp.org |
| 20:10.30 | Qwell | Fix your host. |
| 20:12.01 | cvance | any other suggestions while I am looking into the xen setup? |
| 20:12.23 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:12.23 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:12.45 | Qwell | pabelanger: what caused the problem you were having recently? |
| 20:12.51 | Qwell | ^^^ |
| 20:14.43 | pabelanger | Qwell: seems the VMware server was overriding the clients, which had the wrong time |
| 20:15.09 | p3nguin | Start ntpd with option -g |
| 20:15.44 | p3nguin | Use "ntpdate -u <your ntp server>" to update the time while ntpd is running. |
| 20:16.41 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
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| 20:18.01 | kfife | Hmmm... DISA behaving strangely since 1.6.2 --> 1.8.9.3 |
| 20:18.10 | p3nguin | How so? |
| 20:18.16 | kfife | Good question. |
| 20:18.19 | kfife | Still investigating. |
| 20:18.31 | kfife | The symptom is "Shit don't work" |
| 20:18.50 | kfife | c/"S/"The S" |
| 20:19.50 | kfife | App starts, but no dial tone. Have the options or dependencies changed between versions? |
| 20:20.07 | kfife | ...no 'scecond' dial tone, that is. |
| 20:20.29 | kfife | I hadn't noticed anything to that effect in the doc. |
| 20:20.41 | kfife | nor "the google" |
| 20:20.49 | kfife | :-) |
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| 20:28.09 | scubes13 | anyone here have a cisco 3550 PoE switch powering Polycom 501 units? |
| 20:28.33 | Naikrovek | people do that yes |
| 20:28.56 | Naikrovek | one guy in here even got a non-standard PoE switch to do correct PoE through a couple config changes |
| 20:29.11 | Naikrovek | and i believe that was a 3550 |
| 20:29.48 | scubes13 | just checking, had a vendor tell me it would not work with those two specific devices… I can see that people have some ip-401 and 320/330's, but didnt see 501 specifically mentioned via google |
| 20:31.16 | scubes13 | but even then seems like it is a matter of making the cisco switch wait before cutting power to a downed interface... |
| 20:31.20 | scubes13 | thanks Naikrovek |
| 20:31.41 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 20:32.29 | [TK]D-Fender | There is no IP 401 |
| 20:32.30 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 20:32.49 | [TK]D-Fender | There was an ancient IP 400.... but that is extremely old |
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| 20:33.08 | scubes13 | may be def wrong ;) was off the cuff |
| 20:55.48 | Qwell | kfife: Any changes would be in UPGRADE.txt |
| 20:55.55 | kfife | Thanks |
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| 20:59.45 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 21:02.45 | *** join/#asterisk Gestahlt (~chatzilla@p5DCD18B3.dip0.t-ipconnect.de) |
| 21:02.48 | Gestahlt | Hi |
| 21:03.01 | Gestahlt | I´ve got a asterisk 1.6.2 running on ubuntu |
| 21:03.10 | Qwell | congratulations |
| 21:03.11 | Gestahlt | and i got a high latency between 2 SIP clients |
| 21:03.29 | Gestahlt | Both softphones on samsung galaxy S2 |
| 21:03.37 | Gestahlt | csip or something like that |
| 21:03.43 | Gestahlt | connection and such works fine |
| 21:03.47 | Gestahlt | tried different codecs |
| 21:03.52 | Gestahlt | but the delay is a bit too much |
| 21:03.56 | kfife | Buffer Bloat |
| 21:03.56 | Qwell | ~enter |
| 21:03.56 | infobot | the enter key is not a substitute for punctuation. Use a period '.', exclamation mark '!', question mark '?', comma ',', colon ':', semi-colon ';' emdash '--', or ellipsis '...' instead. |
| 21:04.22 | Gestahlt | kfife: what is a buffer bloat? |
| 21:04.45 | kfife | It's where your wifi buffers packets instead of dropping them, which it probably should. |
| 21:04.54 | kfife | IP networks are designed with packet loss as part of the design. |
| 21:05.22 | kfife | SOme well-meaning radio engineers and networking engineers design thtem to deliver packets at all cost |
| 21:05.28 | kfife | resulting in shit. |
| 21:05.39 | Gestahlt | kfife: is there a workaround? I had once a WLAN config running (like 2 years ago) with a.. dunno, a real WLAN handset (just for voip) and there was latency no issue at all |
| 21:06.37 | kfife | There could be buffering in your S2. You also may be using a different AP that has a different policy. SOme WiFi APs have some QOS settings that MAY, just MAY improve the issue. |
| 21:06.57 | kfife | I'd not spend any time tweaking asterisk until you can re-create the issue on a wired lan. |
| 21:07.49 | kfife | Faster AP's such as 802.11n tend to be more guilty of overbuffering |
| 21:08.00 | kfife | you may have had better performanc eon your ' |
| 21:08.16 | kfife | 'worse' 802.11b |
| 21:08.22 | kfife | Wi-Fi AP |
| 21:08.55 | kfife | Your 802.11n thinks it's got a 100M link, buffers accordingly. |
| 21:09.14 | kfife | You may not have 100mbs to your ISP |
| 21:09.51 | kfife | VUC Podcast link coming... |
| 21:10.33 | kfife | The microfamous Jim Gettys talks about buffer bloat here: |
| 21:10.37 | kfife | http://www.voipusersconference.org/2011/bufferbloat/ |
| 21:10.50 | kfife | jim gettys' of x-window fame. |
| 21:19.10 | kfife | oops |
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| 21:19.53 | Qwell | Way to go kfife. |
| 21:19.54 | *** join/#asterisk VinceAntw (~VinceAntw@91.176.56.82) |
| 21:20.07 | VinceAntw | Hi all |
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| 21:21.00 | kfife | Qwell: Thanks |
| 21:21.08 | kfife | Qwell: With regard to what? |
| 21:21.10 | [TK]D-Fender | *BOOM* |
| 21:21.16 | Qwell | kfife: Splitting the nets. |
| 21:21.26 | kfife | I try |
| 21:21.26 | *** join/#asterisk cstachris (~chrismylo@42.241.23.191) |
| 21:21.33 | kfife | I though it was my bitching about buffer bloat |
| 21:21.34 | [TK]D-Fender | kfife, You just completely the internet ... |
| 21:21.42 | _Corey_ | quietly rolls his disruptor ray back into the garage... |
| 21:21.59 | kfife | The buffers were like "Ungrateful bastard" |
| 21:22.03 | kfife | We'll show 'em |
| 21:22.11 | VinceAntw | I have a question about hangup on a channel |
| 21:22.33 | kfife | I don't like to anthrompomorphize network buffers. They hate it when I do that. |
| 21:22.54 | Gestahlt | kfife: seems like they were.. listening. |
| 21:23.05 | cmendes0101 | Anyone familair modifying app_originate.c? Trying to let ring for longer then 30sec but when anything longer then 30 it still hangs up. When less then 30sec it seems to work. |
| 21:23.27 | kfife | :-) |
| 21:23.31 | Gestahlt | kfife: anyway, is there a workaround / or a good solution for VoIP via WiFi? maybe LAN -> DECT? |
| 21:24.37 | p3nguin | Using an ATA and a DECT handset is probably a better idea than using VoIP over Wi-Fi. |
| 21:24.40 | Gestahlt | ...test? |
| 21:25.09 | kfife | P3nguin is right. Dect is a purpose-built radio transport. |
| 21:25.16 | p3nguin | At least then you only have to worry about the quality of the LAN and the DECT as opposed to worrying about 802.11. |
| 21:25.18 | Gestahlt | p3nguin: Any recommendations? For multiple DECT hotspots and like 50+ handsets? |
| 21:25.27 | VinceAntw | Gestahlt I don't know your situation but I use Siemens Gigaset 610 IP, pretty good VoIP DECT |
| 21:25.30 | Gestahlt | or can we use existing ones? |
| 21:25.48 | kfife | You may be in Polycom Sip-Dect territory. |
| 21:25.53 | p3nguin | If you have a DECT cordless phone, use it with an ATA such as the Linksys PAP2T. |
| 21:26.12 | kfife | A native sip-dect phone is preferable howerver. |
| 21:27.01 | Gestahlt | p3nguin, we actually want to replace our crappy dect hardware with something better. Im not sure how good our current DECT setup works with different phones |
| 21:27.15 | kfife | Native SIP-DECT phones can support wideband telephony by binding two wireless channels. Normally SIP-DECT will transcode to g.726 |
| 21:27.47 | Gestahlt | kfife, what hardware is recommended? |
| 21:27.55 | kfife | Gestahlt: You're looking for GAP-compliance? |
| 21:28.03 | kfife | for existing bases? |
| 21:28.10 | Gestahlt | kfife: btw, we would just have loved to use all our wonderful smartphones as voIP clients |
| 21:28.17 | drudge` | wo |
| 21:28.18 | Gestahlt | kfife, if possible |
| 21:28.19 | kfife | I know. YOu're so right about hat. |
| 21:28.34 | kfife | Too bad the performance sucks ass. |
| 21:28.39 | Gestahlt | yeah |
| 21:28.47 | kfife | ...not to get too technical |
| 21:28.52 | Gestahlt | possible to do anything with a GSM tranciever? *G* |
| 21:29.02 | Gestahlt | well, i rather not |
| 21:29.03 | kfife | You mean like OpenBTS? |
| 21:29.12 | Gestahlt | in the end im the poor bitch who gets layed |
| 21:29.18 | Gestahlt | Like that |
| 21:29.51 | Gestahlt | OpenBTS-> im checking on it, gimme a few |
| 21:29.56 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 21:29.59 | kfife | If you don't make your facility a Faraday Cage, you may have the FCC on your case. |
| 21:30.14 | Gestahlt | Yeah most likely, i checked. That is a no go for us |
| 21:30.35 | Gestahlt | But... thats a awesome project |
| 21:30.39 | Gestahlt | thanks *bookmarked* |
| 21:31.17 | VinceAntw | Does anyone know if it is possible to play "silence" instead of hangupcall for one extension? |
| 21:31.17 | kfife | Ain't it? |
| 21:32.12 | kfife | http://www.polycom.com/products/voice/wireless_solutions/dect_communications/index.html |
| 21:32.14 | Gestahlt | well, i also read about gsm trunks and using certain Siemens or Nokia mobile phones as ptsn |
| 21:32.42 | kfife | I haven't used any of these first hand, but you can build an entire network of base stations with handoff, just like a GSM BTS. |
| 21:32.45 | Gestahlt | that sounded very nice to me, since you can use flatrates for D2->D2 or whatever |
| 21:33.27 | Gestahlt | which makes it a real goot LCR if you got a lot of ppl running the roads (service techs) |
| 21:33.38 | Gestahlt | anyway |
| 21:34.08 | kfife | I'm experiencing the same thing with my Wi-Sip clients on my Galaxy Nexus |
| 21:34.23 | kfife | "Whore-able" latency |
| 21:34.58 | Gestahlt | kfife: Hmm.. i was thinking of using some alternative firmware for my Linksys AP/Routers. dd-wrt or open-wrt and fiddle with the settings |
| 21:35.39 | kfife | When I get some time I'm going to monkey around with the Ubiquiti AP's and see if I can get some reasonable performance. I like the convenience of the unified handset, but want to eliminate the GSM layer when possible to improve fidelity, lower latency. |
| 21:35.46 | kfife | Latency? Me? |
| 21:36.22 | kfife | Android 4 has a built in SIP client. |
| 21:36.55 | Gestahlt | kfife: when i heard of ice cream sandwich as byname.. i thought of a sexual practice |
| 21:36.56 | kfife | but it doesn't have a concept of a dial-plan to opt-in to VoIP routes. Hmmmph |
| 21:37.16 | kfife | Gestahlt. It's that too. |
| 21:37.25 | Gestahlt | kinky |
| 21:37.47 | kfife | Can be. Depends on what you like. Let's say you're an android... |
| 21:38.09 | VinceAntw | kfife: can you help me with a problem? |
| 21:38.12 | *** join/#asterisk nitram (nitram@korben.fhloston.org) |
| 21:38.16 | Gestahlt | But what happens if i am an Apple? |
| 21:38.19 | kfife | Eye Can't rye. |
| 21:38.21 | Gestahlt | am i in the middle then? |
| 21:38.30 | kfife | Oops. Spell check. I can try. |
| 21:38.56 | kfife | An apple and an Android walk into a bar... |
| 21:39.27 | kfife | VinceAntw: What 'cha doing? |
| 21:39.56 | Gestahlt | and order a ice cream sandwich? |
| 21:40.22 | VinceAntw | thanks, I have an analog doorphone hooked to my asterisk, everything works exept when I terminate the call the doorphone keeps beeping for about a minute (doesn't detect hangup) |
| 21:41.05 | drmessano | Androids can't eat ice cream... they freeze up enough without it |
| 21:41.24 | VinceAntw | is there any way to play silence for some time instead of going straigt to hangup |
| 21:41.37 | *** join/#asterisk ajg (~inaki@wah.dreamchaos.net) |
| 21:41.45 | kfife | VinceAntw: connected via Dahdi or an ATA? |
| 21:41.52 | VinceAntw | Dahdi |
| 21:43.06 | kfife | Is the door phone powered by the analog bus? |
| 21:43.15 | kfife | by the dahdi loop itself, or is it externally powered? |
| 21:43.27 | VinceAntw | FYI: I noticed that the hangup beeps are much shorter then on the old PBX |
| 21:43.33 | VinceAntw | externally powered |
| 21:43.54 | kfife | by hangup you mean that you hang up the channel, but the phone thinks it's off hook? |
| 21:44.24 | VinceAntw | I see in the asterisk console hangup |
| 21:44.40 | kfife | Is the issue that the phone thinks it's still off hook? |
| 21:44.43 | VinceAntw | and on the street the doorphone goes beep beep beep |
| 21:45.05 | kfife | I get that it's beeping, but what are the beeps indicitave of? |
| 21:45.09 | VinceAntw | yes I think so |
| 21:45.09 | kfife | being off hook? |
| 21:45.55 | kfife | The first thing that comes to mind is that the door phone might need a quick polarity reversal to detect hangup. I think it's sometimes called 'Disconnect Supervision" |
| 21:46.40 | VinceAntw | yes I read something about that with loop start or kewl start |
| 21:46.44 | VinceAntw | right? |
| 21:47.02 | kfife | the Telco can explicitly signal remote party dignal disconnect. |
| 21:47.08 | kfife | DOes so with a reversal. |
| 21:47.20 | kfife | THat's a dahdi configuration. I'd confirm that it's correct. |
| 21:47.22 | Gestahlt | kfife: im checking the open-wrt manual, any idea what setting could reduce buffer bloat or latency? |
| 21:48.06 | kfife | Gestahlt: Maybe look for QOS settings under wireless? Not internet qos/shaping. |
| 21:48.12 | kfife | It would be a wi |
| 21:48.14 | kfife | fi setting |
| 21:48.33 | Gestahlt | kfife: will try tomorrow |
| 21:48.42 | Gestahlt | kfife: if you are interested, i can give you some results |
| 21:49.04 | kfife | More often QOS is used to refer to internet traffic shaping. |
| 21:49.23 | kfife | IIRC more correctly it's a layer 2 frame tagging specification. |
| 21:49.28 | Gestahlt | kfife: we want to try to consolidate hardware, so i rather avoid getting new stuff because the asterisk box is hanging for that test on our current pbx box |
| 21:50.03 | kfife | I'd love to hear yoru results. Wi-SIP makes the most sense long-term, but SIP-DECT works today. |
| 21:50.03 | Gestahlt | kfife: as test for outgoing calls i got a bintec bingo as capi server |
| 21:50.36 | Gestahlt | kfife: asterisk worked great with a avm b1 before.. i wonder how good it would be with the bingo |
| 21:50.57 | kfife | I'm going to have to look up bintec bingo to understand |
| 21:50.58 | Gestahlt | kfife: do you have any recommendations on BRI/ISDN hardware? Preferably no PCI cards |
| 21:51.06 | Gestahlt | kfife: its a LAN ISDN router |
| 21:51.09 | kfife | ISDN is dead in the US |
| 21:51.13 | Gestahlt | kfife: with remote-capi |
| 21:51.16 | ajg | hey folks! I have a question about a new 1.8 deployment if I may |
| 21:51.20 | kfife | Though I really love ISDN, |
| 21:51.31 | _Corey_ | kfife: PRI=ISDN |
| 21:51.51 | kfife | _Corey_: ISDN = PRI & BRI |
| 21:52.24 | _Corey_ | yeah, the reason i mentioned it is that PRIs are widely used, therefore ISDN can't be called "dead in the US" |
| 21:52.32 | kfife | You are correct. |
| 21:52.45 | kfife | I should have been more specific. |
| 21:52.52 | kfife | ISDN BRI is dead in the US> |
| 21:53.01 | kfife | Which kinda sucks |
| 21:53.08 | ajg | https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality I'm following this guide to do some basic tests with a softphone, however I can't get the softphone to register with the server. I get: handle_request_register: Registration from '"Aaron"<sip:6021@10.10.1.48:5060>' failed for '10.10.8.75:53805' - No matching peer found |
| 21:53.31 | ajg | I'm probably missing something stupid. any thoughts? :) |
| 21:53.34 | kfife | ajg: Have you configured a peer in sip.conf |
| 21:53.35 | kfife | ? |
| 21:54.40 | kfife | Gestahlt: Pls keep me posted with your success. I hope 802.11AC improves this some way. |
| 21:54.44 | ajg | sip.conf has a SIP account in it, and extensions.conf has a "users" contect |
| 21:54.50 | ajg | context, sorry |
| 21:55.02 | Gestahlt | kfife: sure thing, you helped me today a lot |
| 21:55.08 | kfife | My pleasure |
| 21:55.18 | kfife | I'm going to try Counterpath BRIA on my Nexus. |
| 21:55.29 | kfife | It's possible that they've been able to tighten up the local buffers |
| 21:55.39 | Gestahlt | kfife: and i like to take that challenge ;) I really want to get that WLAN SIP working properly and im not scared of hacks |
| 21:55.42 | kfife | or hook an internal packet queue |
| 21:55.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:56.29 | kfife | My biggert reason for wanting WI-SIP to work is that it reduces the repeated penalties of packet queueing. |
| 21:56.46 | kfife | You ideally would only packetize once, and de-jitter ONCE on the remote side. |
| 21:56.51 | ajg | and sip show peers does list one entry that matches the SIP account |
| 21:57.10 | kfife | If your route is all VoIP, you'll have the best performance. |
| 21:57.31 | kfife | Even SIP-dect introduces additional latency. |
| 21:57.37 | Gestahlt | kfife: btw, can i check in the asterisk CLI somehow the latency? Its years ago i used it and when i did, i didnt need to check |
| 21:58.11 | kfife | If asterisk is not the endpoint, it won't be buffering |
| 21:58.32 | kfife | if correctly configured, it should be just bridging packets for your endpoint (your sofphoen) to de-jitter. |
| 21:59.46 | kfife | ajg: you would need two peers. One for your ITSP (if you're terminating to the PSTN) and another for your local softphone. YET ANOTHER for your second softphone. Each should be registering to a unique definition. Extensions.conf controls their interaction. |
| 22:00.17 | kfife | (the peers' interaction) |
| 22:00.28 | ajg | kfife: okay so this is a sip.conf problem then.. |
| 22:00.44 | ajg | where the softphone can't register with asterisk |
| 22:00.48 | kfife | Sounds like it from here. |
| 22:01.06 | kfife | The peer much match the softphone's credentials exactly. |
| 22:01.22 | VinceAntw | kfife: I looked it up, if I am not mistaken it is the kewl start/loop start setting, it was set to KS (which I think is right) |
| 22:01.45 | kfife | I'm guessing your phone is LSDS |
| 22:01.51 | kfife | Loop Start Disconnect Supervision. |
| 22:02.03 | kfife | VinceAntw: ^^ |
| 22:02.32 | VinceAntw | okay so I should put it to loop start? |
| 22:02.56 | kfife | Not sure. I would think so, but I'd have to check. |
| 22:03.07 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 22:03.07 | kfife | hold on. |
| 22:03.09 | VinceAntw | I will try |
| 22:03.12 | VinceAntw | ok |
| 22:03.38 | ajg | I'm using 3CX for my softphone. in sip.conf I'm doing [softphone] \ type=friend \ host=dynamic \ secret=pass \ context=users |
| 22:05.03 | kfife | VinceAntw: My bad. Kewlstart is LSDS |
| 22:05.16 | VinceAntw | no problem |
| 22:05.31 | VinceAntw | actually it is a problem |
| 22:05.34 | kfife | LSDS: The CO switch will remove battery voltage from the loop for about 250 ms, within 6 seconds after the far-end party disconnects |
| 22:06.05 | kfife | but battery reverse is also specified for DS |
| 22:06.18 | kfife | Your FXO device may need battery reverse. |
| 22:06.51 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 22:06.59 | VinceAntw | and how do I do that? |
| 22:07.45 | kfife | You have to slog through the DAHDI doc. |
| 22:07.49 | kfife | :-) |
| 22:07.53 | kfife | Not sure specifically. |
| 22:08.07 | kfife | Read the sample configs. I'm sure it's called out there. |
| 22:08.12 | VinceAntw | oh joy |
| 22:08.37 | VinceAntw | will give it a try |
| 22:09.49 | kfife | Anyone here have specific experinence with VMWare ESXI |
| 22:10.11 | VinceAntw | yes kfife I have |
| 22:10.14 | kfife | Great. |
| 22:10.26 | kfife | I'm tryign to create a virtual DOS host. |
| 22:10.30 | kfife | Believe it or not :-) |
| 22:10.38 | kfife | It's for a production line |
| 22:10.59 | VinceAntw | ok |
| 22:11.05 | kfife | Can I run ESXI on a little embedded fanless box, and present the DOS machine to the local console? |
| 22:11.20 | kfife | The performance requirement is so close to zero that you can safely round down. |
| 22:11.23 | kfife | :-) |
| 22:12.46 | ajg | aha! got it to register |
| 22:12.58 | kfife | ajg: great! |
| 22:13.07 | ajg | it helps to follow the users.conf sample, not the online docs, apparenlty! |
| 22:13.12 | VinceAntw | by embedded you mean? does it has a x86 processor? |
| 22:13.20 | ajg | kfife: thanks for pointing me in the right direction! |
| 22:13.23 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 22:13.26 | kfife | ajg: sure! |
| 22:13.49 | kfife | VinceAntw: exactly. It's got a regular VGA console, not a COM Bios |
| 22:13.58 | kfife | and yes, x86/32 |
| 22:14.17 | ajg | I have been tasked with learning asterisk at work so we can replace our Trixbox server and save the contract costs on it. Whee! |
| 22:15.05 | VinceAntw | does the embedded machine have ethernet? |
| 22:15.05 | kfife | Building a full-featured PBX from scratch is not a trivial undertaking |
| 22:15.13 | kfife | Yes. |
| 22:15.20 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:15.54 | kfife | VinceAntw: the idea is to make it standalone. |
| 22:16.02 | kfife | off-net as it were. |
| 22:16.14 | VinceAntw | but you want to show the dos console on the vga? |
| 22:16.21 | kfife | Correct. |
| 22:16.33 | kfife | I don't want the line operators to know anything has changed. |
| 22:16.37 | VinceAntw | that is not possible with esxi |
| 22:16.53 | kfife | The idea is to virtualize the required floppy drive. |
| 22:17.07 | kfife | and to create a high-availablility fanless, diskless appliance. |
| 22:17.39 | kfife | The device can be ON-NET for maintenance, configuration, setup etc. |
| 22:18.26 | VinceAntw | does it have local storage? |
| 22:18.37 | kfife | Yes. It would have a local SSD |
| 22:18.56 | VinceAntw | can't you install dos there |
| 22:19.01 | kfife | i.e. a wear-leveling SLC SATA drive |
| 22:19.04 | VinceAntw | native? |
| 22:19.13 | kfife | SSD |
| 22:19.45 | kfife | Can't install the dos app natively without also installing a 34 pin mechanical floppy disk drive |
| 22:19.57 | kfife | That's a possibility, but a second choice. |
| 22:20.05 | kfife | The other consideration is that DOS doesn't know how to throttle. |
| 22:20.22 | kfife | So a 1.5 GHZ atom runs at 1.5 GHZ all the time. |
| 22:20.32 | kfife | noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, |
| 22:20.39 | kfife | faster faster! |
| 22:20.41 | kfife | Hurry |
| 22:20.42 | kfife | noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, noop, |
| 22:20.48 | VinceAntw | hehe |
| 22:20.58 | kfife | Sucker gets hot. |
| 22:21.21 | kfife | All of that heat may work counter to the high avialabiity design priority. |
| 22:22.09 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ueynovhexhwgnckz) |
| 22:22.23 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 22:22.30 | kfife | Ain't she pretty http://www.logicsupply.com/products/ms_9a19 |
| 22:22.42 | kfife | 4 serial ports for the production line |
| 22:22.50 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 22:23.20 | VinceAntw | pretty nice |
| 22:23.29 | kfife | and cheap! |
| 22:23.31 | Dovid | hi all |
| 22:23.56 | VinceAntw | you could install vmware server on win xp or something |
| 22:24.12 | kfife | Right. I just hate the windows wrapper. |
| 22:24.14 | VinceAntw | that gives you access to the graphics |
| 22:24.27 | kfife | Less is more in this case. |
| 22:24.40 | VinceAntw | yeah I understand |
| 22:24.50 | kfife | The less that is presented to the user, the less the likelihood of them fucking something up. |
| 22:25.03 | kfife | Opps. I didn't meant to say Fuck in the channel. |
| 22:25.08 | ajg | hehe |
| 22:25.14 | Qwell | kfife: I, for one, am fucking offended. |
| 22:25.16 | kfife | I'll stop saying fuck now. |
| 22:25.24 | kfife | Shit, that is offensive. |
| 22:25.33 | kfife | Qwell: :-) |
| 22:25.45 | VinceAntw | then shut the f*** up |
| 22:26.37 | kfife | So what do you think? Can you present the DOS VM to the local console? |
| 22:26.44 | VinceAntw | why do you need dos in the first place? |
| 22:27.45 | kfife | It's either that or re-code a "perfectly good" piece of control software from scratch ONLY to maintain status quo if everythign goes perfectly. |
| 22:27.53 | kfife | It will cost us about $40,000 |
| 22:28.08 | kfife | so you see.... |
| 22:28.11 | kfife | :-) |
| 22:28.13 | VinceAntw | damn |
| 22:28.17 | Qwell | kfife: use virtualbox and allocate 5% of the CPU |
| 22:28.41 | kfife | Qwell: hmmmm..... |
| 22:28.48 | VinceAntw | I hate virtualbox |
| 22:29.11 | Qwell | VinceAntw: ...why? virtualbox is amazing |
| 22:29.13 | kfife | Why the hate? |
| 22:29.19 | Qwell | If you don't like it, you're using it wrong. |
| 22:29.28 | kfife | It's like a penis. |
| 22:30.26 | VinceAntw | almost everyone at the university is using it, they all got usb and network problems, I use VMWare, never have problems |
| 22:30.45 | Qwell | VinceAntw: umm, install the extension |
| 22:31.05 | kfife | Qwell: Not being familiar with VirtualBox, How feasible would you say it's to make the Virtual-osity totally transparent to the end user? |
| 22:31.21 | kfife | Also, do you know if we can buy support from Oracle if we need it? |
| 22:31.23 | Qwell | kfife: how are they connecting to it? |
| 22:31.25 | VinceAntw | why is there an extension?? |
| 22:31.32 | Qwell | and I'm sure you can. How else would they make money on it? |
| 22:31.35 | VinceAntw | so hard to build that in? |
| 22:31.45 | Qwell | VinceAntw: Because it can't be open source. |
| 22:31.50 | Qwell | vendor licensing issues |
| 22:32.07 | kfife | Qwell: Certianly a reasonable choice, but remember the early days of digium and Asterisk :-) |
| 22:32.09 | VinceAntw | oh I see |
| 22:32.21 | Qwell | kfife: This is Oracle we're talking about here. They bought VirtualBox, so they could milk it. :p |
| 22:32.29 | kfife | LOL |
| 22:32.40 | kfife | Actually we have paid hundreds of thousands to Oracle over the years. |
| 22:32.42 | Qwell | no, but really, I'm a big fan |
| 22:32.50 | Qwell | Everything else Oracle does sucks, but vbox is done right |
| 22:33.20 | kfife | I like Oracle's licensing model. They get paid no matter what, and let you run whatever version of the software you want. |
| 22:33.34 | VinceAntw | I will stick to my VMWare |
| 22:33.49 | kfife | MS makes you upgrade to pay them $1000 even if costs your busindes $100,000 worth of disruption. |
| 22:34.22 | VinceAntw | but virtualbox may indeed be a good solution for kfife |
| 22:34.26 | kfife | Qwell: Any idea how feasible it would be to make the virual-ness of the DOS instance invisible to the end-user? |
| 22:34.34 | Qwell | kfife: how are they connecting to it? |
| 22:34.36 | saxa | p3nguin: do you remember few days ago you told me that I can cut off few parts from my [incoming] in the dial plan ? |
| 22:34.47 | kfifeVIRTUALBOXF | just kidding |
| 22:34.52 | kfifeVIRTUALBOXF | FAN |
| 22:34.57 | p3nguin | saxa: You'll have to pastebin the dial plan for me. |
| 22:35.12 | saxa | p3nguin: ok |
| 22:35.15 | KarlFife | Qwell: They'd be on the local console in my beautiful dream. |
| 22:35.42 | Qwell | KarlFife: well, you can enable vnc to it |
| 22:36.00 | KarlFife | If the operator is asked about virtualbox, I'd like them to say "What's virtualbox" |
| 22:36.02 | saxa | p3nguin: http://pastebin.com/wYUz1TbZ |
| 22:36.02 | KarlFife | ? |
| 22:36.03 | Qwell | otherwise, you could run it locally and have it launch fullscreen or something |
| 22:36.28 | KarlFife | Qwell: That would be perfect. Is that feasible? |
| 22:36.33 | Qwell | shrugs |
| 22:36.39 | saxa | p3nguin: i reput that Answer() in, because after removing it I had many calls just ring one time and then got disconnected |
| 22:36.47 | KarlFife | Any idea about floppy virutalization? |
| 22:36.54 | KarlFife | I guess I will just have to play with it. |
| 22:36.55 | Qwell | it can |
| 22:36.58 | KarlFife | Great. |
| 22:37.13 | p3nguin | saxa: Without seeing the call trace, I can't say why. |
| 22:37.21 | Qwell | the only question is whether DOS will actually run on hardware as new as it would offer |
| 22:38.24 | KarlFife | DOS doesn't know from drivers. |
| 22:38.28 | saxa | p3nguin: I'm not 100% sure this is the cause, but I expected one ring only calls after that. But will verify it. Then I will be back here. |
| 22:39.18 | KarlFife | except maybe oak technologies CD-ROM driver, and himem.sys :-) |
| 22:39.31 | KarlFife | I'll try this out. |
| 22:39.35 | KarlFife | Thanks Qwell |
| 22:40.15 | KarlFife | Does VirtualBox run on the bare metal or does it need a host operating system? |
| 22:40.28 | KarlFife | I've run VBox on Win |
| 22:40.34 | KarlFife | never on bare metal. |
| 22:41.29 | jkroon | tzafrir, you have a moment to discuss a patch we're trying to build to split oct612x stuff into it's own .ko module in order to fix parallel build issues (in particular when dahdi-linux is patched with the non-digium patches) |
| 22:41.57 | malcolmd | KarlFife: you need a host operating system. windows, osx linux, solaris |
| 22:42.20 | KarlFife | malcomd: thanks |
| 22:43.07 | KarlFife | I like that esxi runs on the bare metal. |
| 22:44.12 | VinceAntw | that is because esxi is build on a very small linux |
| 22:44.42 | KarlFife | VinceAntw: Precisely. very small linux = very few patches :-) |
| 22:44.55 | VinceAntw | true |
| 22:45.17 | KarlFife | Very few kernel updates that might "break your junk. |
| 22:45.36 | KarlFife | causing panic calls at midnight about the production line being down. |
| 22:45.38 | KarlFife | :-/ |
| 22:46.12 | VinceAntw | that would be a bit unpleasant |
| 22:46.26 | KarlFife | amen. |
| 22:46.54 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 22:46.57 | KarlFife | Have you ever tried with ESXI to present a host to the local console? |
| 22:47.18 | VinceAntw | yes |
| 22:47.36 | KarlFife | so there's no reason this shouldn't work then. Woohoo! |
| 22:47.56 | VinceAntw | ?? |
| 22:48.26 | KarlFife | ... no reason why I couldn't present the DOS machine to the local console, and make the virutal-ness invisible. |
| 22:48.37 | KarlFife | or? |
| 22:48.38 | VinceAntw | no |
| 22:48.52 | VinceAntw | I said I tried |
| 22:49.00 | VinceAntw | not I succeeded |
| 22:49.04 | KarlFife | got it |
| 22:49.08 | *** join/#asterisk Rewris (~Rewris@187.127.128.156) |
| 22:50.09 | VinceAntw | maybe if you hack deep in the very small linux that runs ESXi |
| 22:50.28 | VinceAntw | but I don't think it is possible |
| 22:50.40 | KarlFife | I can put 2 fanless machines together linked with an ethernet cable :-) |
| 22:51.01 | VinceAntw | so? |
| 22:51.20 | KarlFife | That'd be UUUUhhhh-Gleee |
| 22:51.32 | VinceAntw | lol |
| 22:51.50 | KarlFife | unless i put them both in the same chassis:-) |
| 22:52.48 | VinceAntw | and your plan is? |
| 22:53.00 | Gestahlt | phew |
| 22:53.11 | KarlFife | ..To play aroud with it. |
| 22:53.20 | woleium | KarlFife: you can do it, but you have to cheat a little |
| 22:53.20 | Gestahlt | i forgot my password to my linksys ap |
| 22:53.29 | Gestahlt | where i had my dd-wrt firmware installed |
| 22:53.31 | KarlFife | woleium: do tell |
| 22:53.43 | KarlFife | Gestahlt: ouch |
| 22:53.48 | Gestahlt | hehe i can test now tomorrow. Lots of settings to play with |
| 22:53.52 | Gestahlt | i cracked it open |
| 22:54.11 | Gestahlt | there seems to be special QoS settings for VoIP ;) |
| 22:54.18 | woleium | if you setup your vm to allow serial login, you can get the linux of esxi to open a vtty session to the machine at startup |
| 22:54.42 | Gestahlt | After reading a bit, i think i know where i might put the lever |
| 22:54.46 | KarlFife | woleium: You're my hero |
| 22:54.52 | KarlFife | I'm going to try this. |
| 22:54.57 | KarlFife | maybe I'll get a raise. |
| 22:55.00 | Gestahlt | man im so excited |
| 22:55.16 | KarlFife | Gestahlt: me too. |
| 22:55.19 | woleium | so it's actually still eesxi, but it looks like it's not |
| 22:55.31 | VinceAntw | nice woleium |
| 22:55.34 | Gestahlt | ha, i also had to "hack" esxi once |
| 22:55.42 | VinceAntw | would like to see that |
| 22:55.43 | KarlFife | woleium: That's precisely exactly what I want. |
| 22:55.55 | Gestahlt | had to add a kernel module for that scsi driver |
| 22:55.59 | woleium | and you can get back to esxi by using alt+f1 |
| 22:56.02 | *** join/#asterisk serafie (~erin@nat/digium/x-qltovxuclmmnkkjd) |
| 22:56.20 | KarlFife | We'll keep the alt+f1 a secret from the line oper. |
| 22:56.30 | Gestahlt | that was around esxi 3.xx |
| 22:56.38 | *** join/#asterisk RickCogley_ (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 22:56.40 | VinceAntw | good thinking Karl |
| 22:57.16 | KarlFife | woleium: ESXI can present a virutal hardware "floppy drive", right? |
| 22:57.39 | Gestahlt | KarlFife: if you need help with virtualbox, i can help you with that |
| 22:57.54 | woleium | KarlFife: oh, that may be more tricky |
| 22:57.55 | Gestahlt | while esxi is too long ago |
| 22:58.18 | woleium | I'd suggest looking into usb passthrough of a usb floppy drive maybe? |
| 22:58.48 | KarlFife | The problem is that DOS apps can't talk to FDD's even if the HOST BIOS can. |
| 22:58.51 | woleium | I don't know if you can allocate physical floppy disks with esxi |
| 22:59.20 | KarlFife | DOS actually manipulates the hardware parallel 34 pin interface to the FDD |
| 22:59.27 | woleium | and I don't know if you can get a usb floppy driver for dos |
| 22:59.28 | Gestahlt | KarlFife: btw, i work closely with oracle. Im managing a huge pool of hardware, including exalogics, exadata, supercluster and so on |
| 22:59.30 | KarlFife | Without a driver abstractionlayer |
| 22:59.44 | KarlFife | Gestahlt: This may be very helpful. |
| 23:00.20 | Gestahlt | KarlFife: Asterisk Server is running on a X4470 lol |
| 23:00.22 | KarlFife | Gestahlt: Have you ever done Asterisk--> OracleRDBMS via unixODBC? |
| 23:00.37 | Gestahlt | KarlFife: Nope, asterisk i just started recently. |
| 23:00.43 | KarlFife | funny |
| 23:00.52 | Gestahlt | KarlFife: means.. i hadnt used it in a long time |
| 23:01.02 | Gestahlt | When i used it first time it was a P4 @ 2 ghz |
| 23:01.05 | Gestahlt | with 1gb ram |
| 23:01.16 | KarlFife | Wow, that'sl like a billion bytes! |
| 23:01.18 | Gestahlt | now its 2x X7550 and 128 GB RAM |
| 23:01.29 | Gestahlt | but i had it spare |
| 23:01.42 | Gestahlt | so.. |
| 23:01.50 | KarlFife | Gestahlt: What's your email address? |
| 23:01.52 | Gestahlt | they are obsolted by the m2 model |
| 23:02.42 | KarlFife | If VB looks like a good fit, I may take you up on your offer. |
| 23:08.48 | *** part/#asterisk VinceAntw (~VinceAntw@91.176.56.82) |
| 23:11.22 | Gestahlt | hmmm |
| 23:13.24 | Gestahlt | Anyone tried asterisk on SPARC? |
| 23:15.26 | carrar | I got a Sparc 20 I'll sell ya |
| 23:16.06 | carrar | $20+plus shipping! |
| 23:16.09 | carrar | heh |
| 23:16.44 | carrar | even toss in a 5c keyboard |
| 23:17.04 | Gestahlt | lol |
| 23:17.09 | Gestahlt | Nah thanks |
| 23:17.14 | carrar | hahah |
| 23:18.04 | Gestahlt | Whats a SPARC good for anyway? LOL |
| 23:18.27 | Gestahlt | Ive got enough of them |
| 23:18.37 | Gestahlt | Maybe i can use a SPARC as my asterisk box |
| 23:18.43 | Gestahlt | but when its too much trouble..# |
| 23:18.52 | carrar | They ran the ISP's of yesterday |
| 23:19.03 | Gestahlt | Yeah the Netra models |
| 23:19.17 | carrar | no |
| 23:19.20 | Gestahlt | No? |
| 23:19.24 | carrar | well they probably did |
| 23:19.45 | carrar | only after ultra sparcs were released |
| 23:20.09 | carrar | Internet in a box was Sparc 20's |
| 23:20.15 | carrar | the original spry |
| 23:20.50 | Gestahlt | cool. |
| 23:21.04 | Gestahlt | We got a few T4 Superclusters |
| 23:21.17 | Gestahlt | I can only shake my head at them |
| 23:21.32 | carrar | they make good text processing boxes |
| 23:21.38 | Gestahlt | ROFL |
| 23:21.44 | Gestahlt | yeah.. well, with 1 TB RAM |
| 23:21.49 | Gestahlt | 4 in total |
| 23:21.54 | carrar | no I mean sparc 20's |
| 23:22.13 | Gestahlt | Whats good about the SPARC CPUs? |
| 23:22.25 | Gestahlt | I read a bit and i never had bad experiences with it |
| 23:22.26 | carrar | What good about cars |
| 23:22.27 | Gestahlt | but i dont get the point |
| 23:22.32 | carrar | wtf kinf of question is that |
| 23:22.37 | Gestahlt | Hmmm |
| 23:22.41 | Gestahlt | Okay |
| 23:22.48 | carrar | their day is over |
| 23:22.50 | carrar | obviously |
| 23:22.51 | Gestahlt | Why use Sparc when u got x86, arm and whatever |
| 23:23.07 | *** join/#asterisk Rewris (~Rewris@187.127.128.156) |
| 23:23.20 | Gestahlt | Well carrar, Oracle released the new T4 systems which are under heavy demand |
| 23:23.28 | Gestahlt | they have crypto engines |
| 23:23.40 | Gestahlt | and so on |
| 23:23.40 | carrar | I hate Oracle |
| 23:24.05 | Gestahlt | is not allowed to make a statement there |
| 23:24.07 | carrar | <- hater |
| 23:24.08 | carrar | heh |
| 23:24.09 | Gestahlt | grins |
| 23:24.36 | Gestahlt | Its crazy what these things cost |
| 23:24.41 | carrar | indeed |
| 23:24.52 | Gestahlt | I was banging my head when oracle bought sun |
| 23:24.59 | Gestahlt | i was scared about virtual box and open office |
| 23:25.08 | carrar | TeX!! |
| 23:25.09 | Gestahlt | which is very good software |
| 23:25.14 | Gestahlt | Java |
| 23:25.24 | Gestahlt | and so on |
| 23:25.34 | Gestahlt | and now.. im fixing these systems |
| 23:25.51 | Gestahlt | and they throw hardware at me like there is no tomorrow |
| 23:26.10 | Gestahlt | They got money |
| 23:26.12 | Gestahlt | thats for sure |
| 23:26.29 | Gestahlt | but they are good quality |
| 23:26.35 | Gestahlt | the hardware |
| 23:26.38 | Gestahlt | well engineered |
| 23:26.42 | carrar | So why do you want to run Asterisk on SUN hardware? |
| 23:26.42 | Gestahlt | really |
| 23:26.51 | carrar | err Oracle hardware |
| 23:26.51 | Gestahlt | Because i got plenty SUN hardware |
| 23:26.56 | Gestahlt | err Oracle |
| 23:26.58 | Gestahlt | hehe |
| 23:27.04 | carrar | does it compile? |
| 23:27.11 | Gestahlt | I dunno, didnt try yet |
| 23:27.13 | carrar | try it |
| 23:27.27 | Gestahlt | As said, right now asterisk is running on a X4470 |
| 23:27.36 | Gestahlt | Sun Fire X4470 |
| 23:27.45 | Gestahlt | With ubuntu 10.04 as base |
| 23:27.57 | carrar | go get the src http://www.asterisk.org/downloads |
| 23:28.07 | carrar | whirl it up! |
| 23:28.13 | Gestahlt | Will try it tomorrow |
| 23:28.18 | Gestahlt | Looks like fun |
| 23:28.32 | Gestahlt | i already abused a T3-2 as fileserver / storage head |
| 23:28.38 | Gestahlt | installed solaris 11 with napp-it |
| 23:28.50 | Gestahlt | got a 20 TB Diskshelf attached to it, raidz 3 and zfs |
| 23:28.56 | Gestahlt | with compression and dedup on |
| 23:28.59 | Gestahlt | runs fine so far |
| 23:29.10 | Gestahlt | ZFS is really sweet |
| 23:29.14 | carrar | I'm happy with my Synology |
| 23:29.20 | Gestahlt | While i dont like Solaris very much |
| 23:29.25 | carrar | 30TB |
| 23:30.05 | Gestahlt | Hehe, i got 4x 18GB Zeus iops SDDs as cache |
| 23:30.05 | carrar | bbl, must go my afternoon latte |
| 23:30.13 | Gestahlt | enjoy |
| 23:30.13 | carrar | make |
| 23:31.34 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 23:37.59 | *** join/#asterisk wonderworld (~ww@dsdf-4db5142a.pool.mediaWays.net) |
| 23:38.28 | carrar | mmm |
| 23:38.30 | carrar | latte |
| 23:38.33 | carrar | IT'S GOLD |
| 23:38.38 | paulc | carrar: I've been thinking about getting a NAS to replace my home built ancient Linux software RAID box.. you love the synology? |
| 23:39.00 | carrar | yeah I looked at doing it myself, QNAP and Synology |
| 23:39.35 | carrar | I also tested both iSCSI and NFS on the Synology since I've gotten it |
| 23:39.49 | *** join/#asterisk wonderworld (~ww@dsdf-4db5142a.pool.mediaWays.net) |
| 23:39.49 | carrar | iSCSI is a little faster but NFS is much more "usage" |
| 23:39.56 | carrar | useable |
| 23:40.33 | Gestahlt | carrar: have you ever tried ZFS? |
| 23:40.58 | carrar | could only use that with iSCSI |
| 23:41.09 | carrar | I don't care so much what FS it uses as long as it works |
| 23:41.21 | carrar | as a NFS server it uses EXT4 |
| 23:41.31 | carrar | I can max out a 1gig link |
| 23:41.47 | carrar | my mac pro can't load balance across two gig links |
| 23:41.48 | Gestahlt | ZFS has raidz, which is very sweet |
| 23:42.02 | carrar | but I had it with some other machines doig 1.5gbps |
| 23:42.06 | Gestahlt | and OpenSolaris / Solaris can do link aggregation |
| 23:42.30 | Gestahlt | we got 4x 1gbe aggregated, very sweet performance with failover |
| 23:42.50 | Gestahlt | im not a big fan of solaris itself, but some features are really nice |
| 23:43.14 | Gestahlt | redunancy on file system level |
| 23:43.49 | carrar | paulc, I use the DS3611xs |
| 23:43.58 | carrar | with 4 bonded gige links |
| 23:44.03 | Gestahlt | i spanned 20 1TB drives over raidz3, i have like 3TB less aviable and have a fault tolerance of 3 drives |
| 23:44.12 | carrar | love it |
| 23:44.35 | carrar | paulc, https://www.osburn.com/ss/120131/Screen_shot_2012-01-31_at_11.29.52_AM.jpg |
| 23:59.39 | *** part/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
