00:08.44 | dlynes | damn....dead in here today |
00:10.38 | Druken | yep |
00:11.24 | dlynes | probably everyone's out enjoying the nice weather like i was :) |
00:11.34 | dlynes | good day for a drive and rollerblading all that stuff |
00:11.51 | dlynes | If it was a little warmer outside, I'd go kayaking instead |
00:12.51 | *** part/#asterisk kamileon (n=kamileon@68.62.190.253) |
00:14.43 | *** join/#asterisk kamileon (n=kamileon@68.62.190.253) |
00:15.36 | Druken | raining here |
00:17.24 | dlynes | i wonder why if amp/freepbx/... peeps are told to go to #freepbx, why aren't centos users told to go to #centos? |
00:19.07 | Druken | because centos is a distro, wheres amp is an asterisk related "package" with alot of crap that only the amp/freepbx knows.... we support the basic asterisk program |
00:19.48 | dlynes | I thought centos wasn't just a distro? I thought it was a distro with a bunch of asterisk cruft with it such as AMP, Flash operator panel, ..? |
00:20.47 | Druken | dunno, i'm a slackman |
00:20.48 | dlynes | Or is that just when it's part of Asterisk@Home? |
00:20.59 | dlynes | Yeah....same here |
00:21.13 | dlynes | i'm just itching to get my hands on Slackware 11.0 |
00:28.19 | Delmar | dlynes, that problem yesterday went away after i did a full package update, and installed the latest kernel and headers... the kernel and headers were installed correctly according to dpkg but who knows.. musta got messed up i reckon |
00:28.38 | Delmar | dlynes, was u i was talkin to wasnt it? think it was |
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00:51.13 | ariel_ | hello everyone |
01:00.08 | ariel_ | Seems very slow here tonight |
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01:24.20 | Hmmhesays | a bit |
01:24.22 | Hmmhesays | i'm going to the bar |
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01:41.39 | Druken | i'm gonna open a bar, i'll call it the drunken monkey... think anyone would go?? hehe |
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01:49.36 | ariel_ | Druken, are you sure you don't want to call it spank the monkey. |
01:50.08 | Druken | positive :) |
01:50.38 | Druken | not unless it's a ladies only club.... |
01:50.58 | Druken | course, then i'd never leave :) |
01:54.58 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
01:56.40 | brodiem | I'm trying to configure a TDM400P w/ FXS modules. I'm wondering what it is I need to do so that I get a dial tone when a telephone is plugged into one of the ports. Basically when I pick up the phone it does whatever is configured as an incoming call to the context specified for that channel in zapata.conf |
01:57.36 | Qwell | brodiem: immediate=no |
01:59.30 | brodiem | ahh thanks |
02:00.31 | Druken | you can use a fxs as a hotline? |
02:00.41 | Druken | sweet, i didn't know that |
02:00.43 | Qwell | Druken: batphone? sure |
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02:00.51 | brodiem | Druken I guess I just found out that you can :) |
02:01.01 | brodiem | immediate=yes lol |
02:01.01 | Qwell | s,1,Dial(SIP/batman) |
02:01.09 | Druken | course, fxs sucks ass |
02:04.33 | brodiem | best thing for fax besides a seperate pots though.. |
02:06.43 | Druken | agreed, however, my fxs ports die all the time |
02:06.58 | Druken | fucken fax very rarely has a dialtone |
02:07.40 | tzanger | Druken: on what |
02:07.51 | Druken | on my tdm card |
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02:08.19 | Druken | fxs port usually lasts about 10 days... then i have to reboot to fix it |
02:09.22 | coppice | some people seem to get that, and some don't. nobody seems to investigate the cause, though |
02:09.44 | Druken | personally, i'd never buy a tdm card ever again |
02:10.02 | Druken | i have a tdm and 2 fake x100p's in the same machine |
02:10.12 | Druken | and the fake cards blow the shit out of the tdm |
02:10.43 | Druken | the fxo on the tdm, i get echo on incoming calls, and feedback, and static, and it's just a peice of crap |
02:11.04 | Druken | yet with the exact same settings, the fake cards work like a dream |
02:11.10 | coppice | depends which clone you get. some have poor analogue front ends. the good ones are pretty good, though |
02:12.10 | Druken | i guess |
02:13.40 | tzanger | Druken: what rev of TDM card do you have? |
02:13.52 | Druken | no idea. purchased it about a year ago |
02:13.56 | tzanger | dmesg should tell you |
02:14.04 | tzanger | does it have RJ45 jacks or RJ11? |
02:15.00 | Druken | i belive it has the rj45 jacks |
02:15.21 | Druken | it's a pain in the ass removing the rj11 clips from the card |
02:15.52 | tzanger | that's an older card, but generally it's not the card that has the issues, it's the FXO modules. You can try putting a 0.22uF cap across the reset and ground pins on ONE of the modules (they're all in parallel),I can dig up the pinouts if you want |
02:15.52 | Druken | any other way to find the rev number aside from dmesg? |
02:15.56 | tzanger | it was fixed in a later version |
02:16.03 | tzanger | Druken: yeah, pull the card and read the board :-) |
02:16.23 | Druken | not feasable at the moment in time... |
02:16.38 | tzanger | Druken: actually "zap show status" should show it on a recent version of * |
02:16.53 | tzanger | *CLI> zap show status |
02:16.53 | tzanger | Description Alarms IRQ bpviol CRC4 |
02:16.54 | tzanger | Wildcard TDM400P REV E/F Board 1 OK 0 0 0 |
02:16.59 | tzanger | *CLI> show uptime |
02:16.59 | tzanger | System uptime: 10 weeks, 5 days, 34 minutes, 50 seconds |
02:17.04 | Druken | hehe |
02:17.12 | Druken | command not found |
02:17.18 | tzanger | I have *zero* trouble with my TDM400s |
02:17.23 | tzanger | but then again, NONE have FXO |
02:17.34 | Druken | rev h |
02:17.51 | Druken | good ol zttool told me :) |
02:17.56 | drray | hey Druken |
02:17.57 | tzanger | ok it's a newer carrier, I wonder if you have old FXO modules. they were extraordinarily sensitive to noise on the RST/ line |
02:18.12 | Druken | hey drray |
02:18.12 | tzanger | which is what the cap is for (not a great way to do it but works) |
02:18.22 | drray | did you ever get your payphone going? |
02:18.42 | Druken | nope... the board you sent me was for a newer phone... |
02:19.00 | Druken | mines an old western electric type |
02:19.23 | drray | damn, that's too bad |
02:19.32 | Druken | tzanger: what is that cap your talking about? |
02:19.54 | Druken | drray: yeah, i was a bit disapointed, but oh well, storey of my life lately |
02:21.10 | tzanger | Druken: 0.22uF cap across the RST/ and common pins of ONE of the modules (any one) |
02:21.19 | tzanger | I think it's pins 2 and 20 but I'll have to look |
02:21.36 | Druken | you lost me... |
02:21.40 | tzanger | Druken: :-) |
02:21.51 | coppice | TDM400s don't really work for FAX. I think its a driver problem, but for most people they keep frame slipping. |
02:21.52 | Druken | hehe i can put the cards together, but my understanding of them stop there |
02:21.54 | drray | tzanger - if you could look and tell me that pinout |
02:22.35 | Druken | coppice: when the fxs works, the fax works fine... |
02:23.02 | Druken | i'd prefer to have a t100p and use my channelbank, but i'm poor |
02:23.11 | coppice | seems OK for some people, but not for most. maybe its depends on PCI settings or something. |
02:23.46 | drray | I had a tdm400 that worked for 6 months for me, then I lend the card out and get Drukens 10 day reboot lockup |
02:24.09 | Druken | isn't that a fun "feature" ? |
02:24.23 | coppice | I have a little test program that looks for slipping. It seems to show slipping for most people who tried it. Is your faxing really solid, or are you relying on ECM to correct until it works. i.e. if faxing far slower than it should be |
02:24.31 | Druken | i gave up after a while, just vowed to never buy one again |
02:25.04 | coppice | the snag with most reports of rock solid behaviour is the reporter has actually never bothered to look and notice all the problems |
02:25.07 | drray | I've since moved on to a govarion tor card and a adit 600 |
02:25.12 | Druken | coppice: no idea, it's an older fax machine.... but generally works good |
02:25.48 | drray | g2 or g3? |
02:26.52 | coppice | if you have a g2 fax, donate it to the smithsonian :-) |
02:27.02 | tzanger | coppice: I tried that, I get no slipping ... that's the one that listens to the line, right? |
02:27.05 | tzanger | measures delay? |
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02:27.47 | coppice | yeah. I can run it for 10-15 minutes sometimes without a slip. then it will start slipping again |
02:27.52 | tzanger | hmm |
02:27.55 | tzanger | I've never run it that long |
02:27.58 | tzanger | I'll have to give it another shot |
02:28.31 | coppice | its doing autocorrelation on the card's own echo to detect the loop delay |
02:28.53 | tzanger | yeah it's spitting out white noise and listening, right? I think... it's been a while |
02:29.32 | Druken | uhg.... transfering data takes too damn long... |
02:33.14 | coppice | right. i looks for the peak in the autocorrelation of an AWGN signal it pumps out |
02:36.52 | Druken | man... i am so bored it's not even funny |
02:38.34 | xachen | So yeahhmm |
02:38.37 | xachen | woops |
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02:45.04 | dlynes | Delmar: which problem was that again? |
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02:55.32 | coppice | life. life is always the biggest problem |
02:56.51 | Druken | no... life isn't the problem... it's generally the women in your life.. hehe |
02:57.20 | coppice | aren't the women much less of a problem if either you or they are not alive? |
02:58.05 | Druken | very true |
02:58.30 | Druken | however, i am not planning on being not alive anytime soon.... |
02:58.47 | coppice | i try to take my mind off beautiful women, but I find it hard :-) |
02:59.55 | Druken | hehe beautiful women are usually good, cause being not a beautiful man, i would have no chance in smacking it anyways... |
03:00.41 | Sedorox | hehe.. I'm playing dominoes online with a beautiful women right now :p |
03:00.58 | Druken | i was playing pool, but it got boring |
03:01.01 | Druken | hehe |
03:01.02 | dlynes | coppice: How can you possibly take your mind off beautiful women, when you live in hong kong? |
03:01.17 | Sedorox | lol |
03:01.18 | Qwell | Sedorox: It's a guy |
03:01.21 | Druken | maybe he doesn't like orientals? |
03:01.26 | coppice | dlynes: duh! why do you think I live here? |
03:01.31 | dlynes | lol |
03:01.32 | Sedorox | Qwell: considering I know them in life... naaa :p |
03:01.44 | Qwell | Sedorox: are you SURE? :p |
03:01.46 | dlynes | Hong Kong's got some of the most beautiful chinese girls around |
03:01.52 | dlynes | Same with Vancouver (where I live) |
03:02.13 | Sedorox | Qwell: hehe, yes... :p we've spent a lot of time together (in fact.. thinking of asking her out soon, but anyway) :p |
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03:06.46 | coppice | dlynes: try singapore. similar girls to HK, but they dress sexier :-) |
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03:07.50 | Qwell | Try CA |
03:08.01 | Qwell | Can't go but 10ft without running into one :p |
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03:08.28 | coppice | they don't look so good though |
03:09.00 | Qwell | I'm just talking in general. Can't go 10 ft without seeing something worth looking at :p |
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03:10.05 | coppice | american chinese don't look like other chinese |
03:10.07 | coppice | american europeans don't look like other europeans |
03:10.08 | coppice | american blacks don't look like other blacks |
03:10.10 | coppice | is it just the diet? |
03:10.26 | dahunter3 | I'm looking for a function to strip all characters except for numbers, does one exist or should I roll my own? |
03:10.41 | Qwell | dahunter3: REGEX, I believe |
03:11.36 | Druken | coppice: fat looks like fat :) |
03:12.05 | coppice | fat is certainly a part of it, but only a part. |
03:12.35 | coppice | Qwell: due to our higher population density, I don't need to travel as far as you :-) |
03:12.47 | dlynes | coppice: Yeah, but most of the Chinese here are fresh from China, Hong Kong, or Taiwan |
03:12.48 | Nugget | "A lot of people, when they have a problem, think 'I know, I'll use regular expressions!' Now they have two problems." -- jwz |
03:12.58 | dlynes | coppice: Most of them can barely speak a word of English |
03:13.39 | Qwell | Nugget: :p |
03:13.48 | xachen | Anybody know of a good way to merge Monitor sessions into one? |
03:13.49 | dlynes | coppice: But the Chinese that have been here a while longer are usually a lot fatter, and don't seem to care what they look like |
03:14.12 | dlynes | xachen: sox...it's in the wiki for Monitor on how to do it |
03:14.26 | coppice | dlynes: I know a few american chinese who went there is their teens, and still look very american a few years later |
03:14.35 | xachen | I'm doing call audits on our support staff and need to record them ;) |
03:14.50 | dlynes | coppice: Yeah..it all depends on how fast they blend in |
03:14.59 | dlynes | coppice: And how young they are when they come here, too |
03:15.23 | dlynes | xachen: You could always use MixMonitor, too...apparently that nasty core dump/segfault problem has been fixed now |
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03:16.18 | coppice | of course people always tend to blend in. when I first came to asia I had a mass of blond hair, and really stood out. now i have less of that :-) |
03:16.18 | opus_ | hello |
03:16.19 | dlynes | coppice: The one girl I was engaged to, after she'd been here for three years, I was more Chinese than she was Canadian |
03:16.38 | dlynes | coppice: iow, she hadn't Canadianized at all |
03:17.01 | coppice | in vancouver? of course not |
03:17.10 | dlynes | coppice: Heh |
03:17.25 | dlynes | coppice: Yeah...you can live here for years with only knowing either Cantonese or Mandarin |
03:17.28 | coppice | same would happen in toronto |
03:17.46 | dlynes | coppice: You don't need to know English to survive in Vancouver |
03:18.12 | coppice | some mainlander moving to toronto find they don't need to learn english. they need cantonese |
03:18.23 | dlynes | coppice: Yeah...easier to find a job |
03:18.51 | dlynes | coppice: Another Chinese person will give you a job much faster than a Canadian will, because the Canadian doesn't know the value of your education |
03:19.19 | dlynes | coppice: Nobody here has ever heard of Tsinghua University, or Nanjing University, ... |
03:20.03 | dlynes | Well, that and they wouldn't know the difference between Tsinghua and Beijing universities |
03:27.53 | coppice | i was in nanjing this week |
03:28.52 | dlynes | heh...you're making me jealous |
03:28.59 | dlynes | I want to go back to china again |
03:29.23 | dlynes | Just no desire to live there...the pollution was way too bad |
03:29.27 | coppice | i'm not sure of the value of a chinese education. I've met some real dummies who graduated from tsinghua |
03:29.44 | coppice | dlynes: where are you from? |
03:29.59 | dlynes | coppice: I don't know....most of the guys I know that graduated from Tsinghua are quite smart, but they're way too arrogant |
03:30.06 | dlynes | coppice: Ontario |
03:30.26 | coppice | i thought you said you want to go back to china? |
03:30.32 | dlynes | Yeah, I do |
03:30.35 | dlynes | For another visit |
03:30.38 | dlynes | =) |
03:30.47 | dlynes | I've been there before, and loved it |
03:31.00 | dlynes | Just don't think I could handle it for long stretches because the pollution is so bad |
03:31.07 | coppice | oh. most of the cities are much cleaner than a few years ago, except beijing |
03:31.28 | dlynes | Beijing has a mandate to clean it up by 2008(?), too |
03:31.38 | coppice | beijing is suffering huge dust problems, due to the drier climate to the west |
03:31.43 | dlynes | Including getting rid of all the coal fired furnaces |
03:31.56 | coppice | you might have seen that in the news this week |
03:32.01 | dlynes | When I was there, the coal dust on the balcony in the morning was absolutely horrible |
03:32.14 | dlynes | It was usually about 1/8" thick |
03:32.17 | coppice | you don't get much of that now. |
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03:32.25 | dlynes | You could wipe it off, and the next morning it would be back |
03:32.39 | dlynes | All the poorer tenements used coal |
03:33.07 | coppice | many cities a few years ago were thick with dust from construction. now they use nets to control that |
03:33.49 | coppice | well, beijing has a big problem with the olympics, because there is no way they are going to control these dust storms |
03:33.51 | dlynes | Hangzhou apparently still has a major pollution problem, though |
03:34.07 | dlynes | That;'s what a couple of my friends tell me that live there |
03:34.19 | coppice | hangzhou is a beautiful place. |
03:34.30 | coppice | the air is really clear there. |
03:34.32 | dlynes | Yeah...I've seen pictures of West Lake |
03:34.40 | dlynes | It's one of the most beautiful lakes I've seen |
03:35.17 | coppice | west lake was a horrible mess a couple of years ago. they were constructing hotels and conference centres and masses of touristy things there. its settling down now |
03:35.18 | dlynes | Well, besides |
03:35.35 | dlynes | Hangzhou is famous for many beautiful things :) |
03:36.09 | coppice | they are trying to turn hangzhou into a very academically oriented area |
03:36.39 | dlynes | Maybe that'll make it more voip friendly then |
03:36.48 | dlynes | Government-wise |
03:36.52 | coppice | voip is outlawed |
03:37.08 | dlynes | That's my point :) |
03:37.23 | dlynes | But it's not totally outlawed |
03:37.27 | dlynes | It's still allowed |
03:37.37 | dlynes | Just not for outbound calls |
03:37.42 | coppice | its filtered in most situations |
03:38.13 | dlynes | So it's not allowed for domestic traffic, and international inbound traffic? |
03:38.32 | dlynes | Someone I know at Shanghai Bell Alcatel was telling me it was |
03:38.50 | coppice | they don't really need to ban it. they've just filtered every protocol you can think of :-) |
03:39.35 | dlynes | Make it go through on a different port then :) |
03:39.43 | dlynes | Skype works just fine in China |
03:40.14 | dlynes | It actually sounds quite clear coming from China, too |
03:41.21 | coppice | broadband in china is generally excellent, so the connections should be pretty clean, unless the international part gets congested |
03:42.04 | coppice | the free broadband in my nanjing hotel room this week let me download something at about 1MB/second |
03:43.09 | coppice | but they'd filtered UDP to the point where VPN wouldn't work, and I needed to use dialup :-( |
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03:44.46 | rogercharlie | anyway to change group count with a sip transfer? |
03:46.08 | rogercharlie | updates the correct channel with an asterisk transfer |
03:47.33 | *** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder) |
03:47.38 | xbmodder_lappy | how does E911 work? |
03:51.02 | dlynes | xbmodder_lappy: If I remember correctly, your PSTN/PRI upstream service provider provides that information on the ring delivery to the 911 call center |
03:51.46 | xbmodder_lappy | dlynes, Yeah, I understand that part of it, but I would like to know how they send it to the 911 call center... |
03:51.47 | dlynes | xbmodder_lappy: Of course, when you get that line/pri, you have to provide your SP with all that information for every new subscriber that needs E911 capability |
03:52.13 | dlynes | xbmodder_lappy: It's sent on the wire similarly to the caller id info |
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03:53.58 | dlynes | man....i need to do something about this beer...been sitting here since Christmas, and I've only drank two of them :( |
03:57.22 | xbmodder_lappy | i can help |
03:57.34 | rogercharlie | email me the beer |
03:59.02 | coppice | I wonder if any 911 call centres have been outsourced to bangalore :-) |
03:59.34 | Qwell | coppice: no doubt |
04:00.26 | coppice | it would be one of those homeland insecurity things :-) |
04:01.15 | xbmodder_lappy | lol |
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04:06.34 | *** mode/#asterisk [+o russellb] by ChanServ |
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04:16.11 | dlynes | email beer...hrm....that would be....kinda difficult....mmmmkay? |
04:18.01 | coppice | BoIP |
04:18.23 | coppice | or the more generic AoIP |
04:19.47 | xbmodder_lappy | a? |
04:19.52 | xbmodder_lappy | MoIP |
04:19.57 | coppice | alcohol |
04:20.03 | xbmodder_lappy | matter over IP |
04:20.28 | coppice | rather than usual immaterial over IP? |
04:25.33 | kamileon | anyone in here interested in a used tdm40b digium card? |
04:28.13 | xbmodder_lappy | how much? |
04:28.33 | kamileon | 250 |
04:28.51 | kamileon | all check out 100.0% |
04:29.32 | xbmodder_lappy | why are you selling it? |
04:33.41 | rogercharlie | he beat up a geek for it |
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04:48.17 | Qwell | kamileon: Sell it to the guy who posts every day on asterisk-biz |
04:54.09 | kamileon | im not familiar with that |
04:55.02 | dlynes | kamileon: it's a mailing list |
04:55.03 | kamileon | xbmodder_lappy: i need to purchase other toys to p;lay with |
04:55.03 | kamileon | and someone is posting trying to buy gear? |
04:55.03 | kamileon | sign up at asterisk.org or digium.com? |
04:55.40 | dlynes | kamileon: asterisk.org |
04:55.50 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
04:56.02 | dlynes | kamileon: people post on asterisk-biz regularly selling routes, buying routes, selling hardware, buying hardware, ... |
04:57.40 | rogercharlie | is OUTBOUND_GROUP able to be updated with a SIP transfer? |
04:58.12 | rogercharlie | it no workey even with __OUTBOUND_GROUP |
05:00.09 | kamileon | oh cool |
05:00.48 | dlynes | kamileon: Yeah...you should sign up on there...the crew on voipsupply regularly blows out specials on there |
05:01.24 | Qwell | kamileon: lists.digium.com |
05:02.05 | Qwell | kamileon: Just send all messages with "Rehan", "didx", or "used digium hardware" to the bit bucket :p |
05:02.13 | *** join/#asterisk jake1932 (n=Administ@pool-70-16-137-123.phil.east.verizon.net) |
05:02.20 | dlynes | lol |
05:02.29 | Qwell | well...the first two, keep those if you like to laugh |
05:02.45 | dlynes | or like to get annoyed :) |
05:02.47 | Qwell | You'll get some classic quotes, such as hungry dids, or tool free dids |
05:03.01 | Qwell | haha, I just got tool free, also |
05:03.18 | kamileon | lol |
05:07.38 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
05:14.24 | *** part/#asterisk felixdacat (n=fholmes@cpe-72-177-253-50.houston.res.rr.com) |
05:22.30 | drfoomod2 | for a ta750, is this all that is needed in the zaptel.conf file? |
05:22.30 | drfoomod2 | span=1,0,0,esf,b8zs |
05:22.30 | drfoomod2 | fxoks=1-4 |
05:22.30 | drfoomod2 | fxsks=5-8 |
05:28.03 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.97.246) |
05:29.34 | m_a_g_o | good evening guys... I'm installing DBD::mysql in order to run AGI scripts and I'm getting: Can't locate loadable object for module DBD::mysql in @INC |
05:29.44 | m_a_g_o | any idea where I can modify @INC? |
05:30.31 | dlynes | m_a_g_o: it's probably looking for a mysql client library...do you have the mysql client libraries installed? |
05:32.23 | m_a_g_o | I downloaded and compiled DBD-mysql-3.0002 with the option --mysql_config= and the path... |
05:32.43 | m_a_g_o | but I can see there are perl versions installed from 5.8.5 to 5.8,7 |
05:32.44 | coppice | anyone used the rhino T1 card? it seems to have been around for a while, but I never heard from anyone using it |
05:32.52 | dlynes | m_a_g_o: but do you have the mysql client libraries installed? i.e. libmysql.so |
05:33.38 | dlynes | m_a_g_o: erm libmysqlclient.so |
05:33.42 | m_a_g_o | from asterisk-addons? |
05:33.50 | dlynes | m_a_g_o: From your linux distribution |
05:34.24 | dlynes | m_a_g_o: It would help to know which distribution you're using; then someone can let you know what package you need to install |
05:34.36 | m_a_g_o | '/usr/lib/libmysqlclient.so.10 |
05:34.47 | dlynes | m_a_g_o: there ya go...is it there? |
05:34.48 | m_a_g_o | FC3 |
05:35.11 | *** join/#asterisk tzafrir_laptop (n=tzafrir@80.178.4.164.adsl.012.net.il) |
05:35.27 | m_a_g_o | there and in /usr/lib/mysql/libmysqlclient.so.10 |
05:35.40 | dlynes | m_a_g_o: Have you tried asking on #perl? |
05:36.09 | dlynes | m_a_g_o: I just did perl -MCPAN -e shell, and the install DBD::Pg for Postgres and it works just fine |
05:36.34 | dlynes | I'm using perl 5.8.7 |
05:40.15 | m_a_g_o | nothing... tried compiling with --mysql_config=/usr/local/mysql/bin/mysql_config --cflags=-I/usr/local/mysql/include/mysql/mysql.h, also from CPAN and got some errors... |
05:40.47 | dlynes | m_a_g_o: like i said...did you try asking on #perl? |
05:41.47 | m_a_g_o | asking right now |
05:43.22 | drfoomod2 | should a autogenerated zaptel.com have at least one span= in it? |
05:43.26 | drfoomod2 | mine does not |
05:44.00 | dlynes | you mean zaptel.conf? |
05:44.12 | drfoomod2 | yes |
05:44.25 | dlynes | only thing it usually has uncommented in it is loadzone and defaultzone |
05:44.32 | drfoomod2 | right |
05:44.53 | drfoomod2 | i added a span=1,1,0,esf,b8zs |
05:44.59 | dlynes | the span= line will be dependant upon what you've got it hooked up to |
05:45.01 | drfoomod2 | and a fxsks=1-4 |
05:45.08 | drfoomod2 | and fxoks=5-8 |
05:45.16 | drfoomod2 | and i get an error when i try ztcfg |
05:46.03 | dlynes | You probably have your fxsks/fxoks backwards then |
05:46.20 | dlynes | You've got a tdm04b and a tdm40b installed? |
05:46.24 | drfoomod2 | i get invalid argument |
05:46.52 | drfoomod2 | te110p |
05:47.00 | dlynes | and the span= afaik is for the te400p/te110p/te410p cards |
05:47.18 | dlynes | Ok, so why are you trying to define fxsks/fxoks? Isn't the te110p a t1 card? |
05:47.24 | drfoomod2 | it is |
05:47.35 | drfoomod2 | and it;s connect to a ta750 channel bank |
05:47.37 | dlynes | You don't define fxoks/fxsks on a t1 card |
05:47.44 | drfoomod2 | oh? |
05:47.49 | dlynes | no |
05:48.14 | drfoomod2 | drop those lines? |
05:48.28 | dlynes | span=, bchan, dchan, loadzone, and defaultzone for a t1 card |
05:48.34 | dlynes | yes |
05:48.34 | drfoomod2 | i did and i still get invalid argument |
05:48.46 | drfoomod2 | may i flood? |
05:48.48 | dlynes | because you haven't specified bchan, dchan more than likely |
05:49.15 | dlynes | bchan is 1-23 and dchan is 24 for most north american t1's |
05:49.29 | drfoomod2 | this is not a PRI |
05:49.34 | coppice | dlynes: he's using a channel bank, not ISDN. he doesn't have any B channels or D channels |
05:49.54 | dlynes | coppice: ah...thought you had to specify all that stuff for channel banks, too |
05:50.02 | drfoomod2 | coppice: tx |
05:50.23 | dlynes | drfoomod2: but regardless, the fxoks/fxsks aren't appropriate for a t1 card |
05:50.26 | drfoomod2 | the ztcfg -v shows span 1, 8 channels configured, and then barfs out a invalid argument |
05:50.47 | drfoomod2 | dlynes: yes they are |
05:50.50 | kamileon | cat tums > asterisk |
05:51.21 | drfoomod2 | ns |
05:51.46 | drfoomod2 | and what is ZTDUMMY/1? |
05:52.05 | dlynes | drfoomod2: a timing interface that relies on your rtc kernel driver |
05:53.43 | drfoomod2 | reboot |
05:53.47 | drfoomod2 | * |
05:56.00 | opus_ | anyone here use asterisk with large amount of simutaneous calls? |
05:56.07 | opus_ | like, more then 40 at once? |
05:56.21 | Qwell | opus_: yes, many people, I'm sure |
05:56.45 | opus_ | Qwell do you? |
05:57.00 | Qwell | To save time, let's say "sure" |
05:57.18 | coppice | opus_ 40 would be a rather small number of calls |
05:57.48 | opus_ | well, how do people stop asterisk from locking up? for example if it was in pure SIP mode |
05:57.53 | opus_ | eventually I get a locked up * |
05:58.12 | drfoomod2 | how can i tell what * thinks i have for a t1 card? |
05:58.13 | Qwell | opus_: What type of hardware, and how much transcoding and such? |
05:58.31 | coppice | opus_: do you cross your fingers when you start asterisk? |
05:58.35 | opus_ | g711, vmware, and sipp |
05:58.39 | drfoomod2 | oh boy |
05:58.43 | drfoomod2 | i have to go to sleep |
05:58.50 | Qwell | opus_: That's why. |
05:58.57 | drfoomod2 | i just remembered i tool the digium card out of this box |
05:58.59 | Qwell | asterisk requires a realtime environment |
05:59.02 | Qwell | vmware simply cannot provide that |
05:59.03 | opus_ | vmware. yeah yeah |
05:59.04 | dlynes | drfoomod2: dmesg |
05:59.09 | drfoomod2 | and put an old ethernet card back in its place |
05:59.17 | coppice | drfoomod2: what does /proc/zaptel/<whatever> say you have? |
05:59.25 | opus_ | Qwell, ok. so if I run it in a realtime environment it will be OK |
05:59.27 | opus_ | ? |
05:59.41 | drfoomod2 | coppice: it aint; gunna say shit |
05:59.43 | opus_ | nothing in my dial plan requires realtime stuff btw. |
05:59.47 | Qwell | opus_: Yes. Don't run it in any type of vm |
05:59.48 | opus_ | answer, play wave file, hangup. |
05:59.52 | Qwell | realtime != real time |
05:59.58 | opus_ | yeha |
06:00.02 | Qwell | I meant real time |
06:00.04 | opus_ | i mean zaptel hardware |
06:00.06 | drfoomod2 | coppice: the card is in another machine on the other side of the room |
06:00.13 | drfoomod2 | it';s later |
06:00.14 | tainted- | bengay != ben gay |
06:00.15 | drfoomod2 | g'nite |
06:00.37 | Qwell | tainted-: helping your uncle jack..nevermind |
06:00.49 | tainted- | Qwell i got my box up to ~350 calls |
06:00.49 | opus_ | i'm coming to the belief that anyone who saids they have huge amounts of simutaneous calls setup with asterisk is lying |
06:01.00 | Qwell | tainted-: I got my box up to ~2500 calls :p |
06:01.03 | opus_ | tainted: with RTP audio? |
06:01.06 | tainted- | did u ever get sipp to push media through? |
06:01.07 | rogercharlie | anway to update destination of OUTBOUND_GROUP with a SIP transfer? |
06:01.08 | Qwell | and that's with rtp |
06:01.11 | Qwell | tainted-: yeah |
06:01.14 | tainted- | what no way |
06:01.20 | Qwell | yep |
06:01.27 | Qwell | 2500 without any retransmits |
06:01.29 | opus_ | Qwell: you have a screen shot with 'show channels' |
06:01.31 | opus_ | ? |
06:01.35 | tainted- | what kind of sipp scenario |
06:01.40 | coppice | its amazing how many claim they pass RTP through when they don't |
06:01.50 | Qwell | tainted-: -sf uac_pcap.xml |
06:01.53 | opus_ | coppice: exactly |
06:01.53 | tainted- | opus_ cli is pretty useless at that point |
06:01.57 | Qwell | coppice: I assure you, I was |
06:02.16 | opus_ | Qwell were you doing RTP 100% of the time or just once for a few seconds.. |
06:02.17 | tainted- | i gave up on sipp and started loading the box up with call files |
06:02.32 | Qwell | opus_: 100% of the time |
06:02.36 | coppice | In the early days of T.38 investigation I was assured by lots of people T.38 was passing through their * box. of course, it wasn't :-) |
06:02.56 | opus_ | 169 active channels |
06:02.56 | opus_ | 86 active calls |
06:03.14 | opus_ | that will run OK for about half an hour then it will bomb . |
06:03.21 | coppice | 2500*50 packets a second*2 directions == a lot of rescheduling of threads |
06:03.22 | tainted- | really? |
06:03.34 | tainted- | hmm |
06:03.52 | Qwell | coppice: two words. ultrasparc T1 |
06:03.54 | tainted- | is it accurate when i use call files and the local/ channel |
06:04.13 | opus_ | tainted : no :( |
06:04.17 | tainted- | i just drop it in a looping background context |
06:04.31 | opus_ | tainted: because its all inmemory |
06:04.32 | tainted- | how do i accurately load test it then |
06:04.38 | opus_ | if you had two servers, it would be valid |
06:04.39 | coppice | Qwell: so its pure IP? no PSTN? why do you want the audio to pass through? |
06:04.52 | tainted- | opc0de i have two servers |
06:04.52 | opus_ | then the test would go through the network statck and be more legit |
06:04.57 | tainted- | opus_ |
06:04.59 | Qwell | why do I want it to pass through? |
06:05.05 | *** join/#asterisk Shaun222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
06:05.21 | opus_ | Qwell: so.. you have a server that does 2500 simutaneous calls at once? |
06:05.28 | Qwell | opus_: 2500 channels, yes |
06:05.42 | Qwell | with rtp flowing both directions |
06:05.55 | tainted- | opus_ so what if i used call file and connected two servers to each other |
06:05.59 | opus_ | is there anyway you could give me a screen shot.. or something? |
06:06.05 | Qwell | opus_: tomorrow |
06:06.07 | opus_ | tainted: I would say that would be valid |
06:06.23 | tainted- | why would the network stack involvement be more legit |
06:06.23 | dlynes | Qwell: I'm guessing you're running modified asterisk code, if it's running on a SPARC? |
06:06.23 | opus_ | Qwell: cool |
06:06.37 | Qwell | dlynes: only slightly modified, and really only the Makefile |
06:06.54 | dlynes | Qwell: ah...but no hardware, right? pure software? |
06:07.07 | Qwell | dlynes: right |
06:07.27 | dlynes | Qwell: And you're using the Sun compiler, or the GNU compiler? |
06:07.34 | Qwell | gcc |
06:07.39 | dlynes | ah |
06:07.46 | Qwell | gmake, ginstall, etc |
06:07.52 | dlynes | Sun produces more optimized code though, doesn't it? |
06:08.04 | Qwell | yeah, but it won't compile asterisk. it would take a lot more work |
06:08.08 | dlynes | ah |
06:08.09 | opus_ | one thing about the SIP stack, I don't think its fast enough to create/destory so many call stacks, like it needs to be refactored/nmore multithreaded |
06:08.23 | Qwell | opus_: I was getting 100+ cps |
06:08.31 | dlynes | Qwell: So you're not taking advantage of Solaris threads then? |
06:08.39 | Qwell | dlynes: oh, I most certainly am :p |
06:08.41 | opus_ | Qwell: i can't get over *4* cps |
06:08.50 | Qwell | opus_: dude, my amd64 can get 25 |
06:09.09 | Qwell | stop trying to run it in vmware |
06:09.25 | opus_ | 636 active channels |
06:09.25 | opus_ | 342 active calls |
06:09.46 | opus_ | Now the system is showing about 1 page of warning: per second |
06:09.47 | Qwell | That's without rtp, I'm sure |
06:09.51 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
06:09.55 | dlynes | Qwell: How do you think an Ultra2 Creator dual cpu would handle it, then? |
06:09.57 | opus_ | no this is with rtp |
06:10.01 | Qwell | dlynes: dunno |
06:10.39 | tainted- | <Qwell> stop trying to run it in vmware <--- lol |
06:10.43 | opus_ | <PROTECTED> |
06:10.50 | Qwell | 34k/s? rofl |
06:10.56 | tainted- | opus_ stop trying to connected to your access farm dude |
06:11.08 | Qwell | opus_: That is NOT with media |
06:11.08 | tainted- | s/connected/connect it |
06:11.11 | opus_ | wtf |
06:11.21 | Qwell | That's like...not even half a ulaw channel |
06:11.47 | opus_ | sipp is buggy |
06:11.56 | Shaun222 | i need the number to a international pizza hut or somthing to test int calls, lol |
06:12.14 | Qwell | Shaun222: bkw__ called Hilton hotels around the world |
06:12.17 | Qwell | I think it was Hilton |
06:12.17 | opus_ | Well anyway asterisk crashed.. just after 350 calls |
06:12.46 | Shaun222 | Qwell: not a bad idea.. |
06:12.49 | Shaun222 | gotta find the number... |
06:14.02 | tainted- | Qwell where'd u get uac_pcap.xml |
06:14.18 | Qwell | tainted-: I think that was the filename. I got it from the sipp site...it's in the faq |
06:14.22 | tainted- | u created it? |
06:14.23 | tainted- | oh |
06:14.30 | opus_ | tainted: sipp automatically generates it when you run ./sipp -sd uac_pcap |
06:14.34 | Qwell | no |
06:14.38 | opus_ | tainted: rtfm :) |
06:14.39 | tainted- | really? |
06:14.39 | Qwell | http://sipp.sourceforge.net/doc1.1/reference.html#PCAP+Play |
06:14.41 | tainted- | k |
06:14.59 | Qwell | http://sipp.sourceforge.net/doc1.1/reference.html#uac_with_media |
06:15.02 | Qwell | those two |
06:15.02 | tainted- | i need to get a better handle on sip |
06:15.12 | tainted- | opus_ stop loading testing against your desk phone |
06:15.23 | tainted- | lol |
06:15.43 | opus_ | i'm stress testing it |
06:15.52 | tainted- | i get seg fault |
06:16.00 | tainted- | when i do the -sd uac_pcap |
06:16.01 | opus_ | welcome to the club |
06:16.14 | tainted- | hmm |
06:16.31 | opus_ | tainted: hint, its also in the .tar.gz file from the download site |
06:16.49 | tainted- | did u guys 'make pcapplay'? |
06:16.58 | tainted- | or just 'make' |
06:17.02 | opus_ | yeah with oss |
06:17.21 | Qwell | make pcapplay, yeah |
06:17.29 | Qwell | which unfortunately didn't work on solaris... |
06:17.38 | Qwell | so I had to generate my calls from my amd64, and send them over the wire |
06:18.26 | opus_ | qwell, what timing device is used in solaris? h is that a problem you are going to need to tackle? |
06:18.26 | Qwell | iirc, it was about 200mbit/s? |
06:18.46 | Qwell | 176mbit/s, I guess |
06:18.54 | Qwell | wait, wrong number |
06:18.59 | distortion | qwell you arent human |
06:19.01 | Qwell | yeah, just at 200 |
06:19.03 | tainted- | opus_ i'm positive it wasn't in tarball |
06:19.08 | Qwell | distortion: yes, we've established this |
06:19.12 | tainted- | but gonna try the one in faq |
06:19.17 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
06:19.20 | Qwell | tainted-: it isn't. :p |
06:19.22 | distortion | just making sure.. |
06:19.29 | Qwell | the .pcap file is there though |
06:19.33 | websae | good evening all, how fair you? |
06:19.50 | websae | anyone have any good provisioning concepts? |
06:20.06 | distortion | websae you nerd |
06:20.12 | dlynes | opus_: not if you dont' need music on hold or conferencing |
06:20.13 | Shaun222 | any reason i couldnt use dial() to ring/find a agent with out using queue() ? |
06:20.25 | websae | distortion: how are you doing? |
06:21.02 | distortion | good, had a big day, almost 750k mou today mmm |
06:22.25 | rogercharlie | anyway to change OUTBOUND_GROUP to the SIP transfered username? |
06:25.55 | *** join/#asterisk gmaruzz (n=maruzz@217-133-80-112.b2b.tiscali.it) |
06:26.16 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-60-235.dsl.irvnca.pacbell.net) |
06:26.48 | tainted- | Qwell do u put actual values in the xml [keyword] or leave them? |
06:26.59 | opus_ | tainted, you have to create a csv file |
06:27.03 | Qwell | tainted-: huh? |
06:27.05 | tainted- | INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0 |
06:27.15 | Qwell | tainted-: no, it fills them in. Leave it alone :p |
06:27.16 | tainted- | should i leave that alone or put in actual values |
06:27.24 | Qwell | ./sipp someargs -sf uax_pcap.xml |
06:27.26 | tainted- | <PROTECTED> |
06:27.40 | Qwell | tainted-: That's with `make pcapplay`? |
06:27.47 | tainted- | yea |
06:27.51 | Qwell | fun |
06:28.02 | tainted- | http://pastebin.ca/50663 |
06:28.10 | tainted- | i just took that out of the faq uac section |
06:28.32 | Qwell | looks different than the one I had |
06:29.04 | opus_ | tainted you need to root when you run it, because on some distros the pthread method it uses for scheduling is blocked for security reasons |
06:29.06 | tainted- | looks like a generic invite, response 200, ack, then bye |
06:29.15 | tainted- | yea i'm root |
06:29.29 | opus_ | ok, then, its broken for sure:) |
06:29.42 | Qwell | ahh, heh |
06:29.45 | Qwell | http://sipp.sourceforge.net/doc1.1/uac_pcap.xml.html |
06:29.49 | Qwell | Get that, and save it |
06:30.05 | *** join/#asterisk {tasker-} (n=ghes@modemcable252.110-83-70.mc.videotron.ca) |
06:30.06 | tainted- | oh look at that |
06:30.21 | Qwell | You need the uac_pcap one, not uac.xml :p |
06:30.28 | Qwell | http://sipp.sourceforge.net/doc1.1/uac.xml.html Thats what you got |
06:30.52 | {tasker-} | anyone have success getting chan_h323.to load in svn trunk? |
06:30.55 | tainted- | hmm.. but the xml should still be valid |
06:30.56 | tainted- | weird |
06:31.01 | Qwell | tainted-: dunno |
06:31.49 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:32.00 | {tasker-} | all I get is this when I load asterisk -> loader.c:731 __load_resource: missing mod_data for chan_h323.so |
06:32.13 | {tasker-} | trunk |
06:32.44 | Qwell | is chan_h232 in the asterisk repository? |
06:32.58 | Qwell | It's in -addons, isn't it? |
06:33.06 | {tasker-} | no |
06:33.09 | distortion | no, its in channels/h323 |
06:33.20 | distortion | ooh323 is in addons |
06:33.20 | {tasker-} | yup |
06:33.24 | Qwell | well, wait for the loader changes to calm down |
06:33.30 | opus_ | tasker, grep for mod_data in the asterisk sources and make sure whatever module proves mod_data loaded before chan_h323.so in modules.conf? Or maybe its because in SVN trunk right now loader.c is being drammatically changed |
06:33.32 | Qwell | Things that aren't in the base will not work |
06:33.40 | opus_ | tasker: what Qwell said |
06:33.46 | Qwell | If you run svn trunk, you MUST subscribe to the -dev list |
06:33.57 | Qwell | not doing so is pretty stupid... |
06:34.19 | {tasker-} | crap |
06:35.00 | {tasker-} | the 1.2.7.1 version locks up after 15 minutes |
06:35.04 | {tasker-} | forcing a restart |
06:35.07 | opus_ | tasker you can probably create a mod_data structure .. or splice in the new module requirements into the top of chan_h323 |
06:35.11 | {tasker-} | and ooh323 is another useless piece of garbage |
06:35.19 | {tasker-} | under heavy load it stops answering calls after 1 hr |
06:35.29 | {tasker-} | issuing a debug shows nothing |
06:35.43 | {tasker-} | i can telnet into port 1720 of chan_ooh323.so and it does nothing |
06:35.49 | {tasker-} | when it's dead |
06:36.11 | {tasker-} | opus_: i suppose i can try that |
06:36.24 | opus_ | tasker: crash? really, I had some code that no longer works in 1.2.7.1 , it was .. lemme look |
06:36.45 | {tasker-} | chan_h323.so has a bad deadlock problem |
06:36.58 | {tasker-} | chan_ooh323 has another problem that I can't seem to trace |
06:37.03 | {tasker-} | no locks, it just stops |
06:37.22 | {tasker-} | they keep closing the channel deadlock issues on bug tracker |
06:37.35 | {tasker-} | without resolving anything |
06:38.01 | Qwell | {tasker-}: When bug marshals close bugs, they give a reason. What is the reason they give? |
06:38.09 | {tasker-} | and I'm getting a headache trying to trace through all the mutex locks & unlocks in JerJer's chan_h323.c |
06:38.19 | {tasker-} | fixed |
06:38.22 | distortion | tasker- i know your pain |
06:38.38 | distortion | I've had to resort to a load balancer and 3 servers to get ooh323 to work |
06:38.39 | {tasker-} | Qwell: they always report it fixed |
06:38.40 | Qwell | pfft, nobody knows the pain of dealing with JerJer's code more than I do :P |
06:38.47 | {tasker-} | lol |
06:38.59 | distortion | one server goes down and then is taken out of hunt to be reset |
06:39.02 | Qwell | (in 6 months, somebody will say that about my code) |
06:39.07 | opus_ | distortion HAHA that was going to be my next step :) |
06:39.18 | distortion | even a restart of asterisk doesnt fix it, i have to reboot the faking server |
06:39.22 | {tasker-} | distortion: what do you use to load balance |
06:39.23 | Qwell | h323 sucks anyhow |
06:39.32 | distortion | h323 is unavoidable |
06:39.37 | {tasker-} | i agree |
06:39.43 | Qwell | no it's not :p |
06:39.48 | {tasker-} | the majority of the world's carrier minutes still run through h323 |
06:40.02 | Qwell | Are you trying to run a carrier? |
06:40.08 | {tasker-} | not trying |
06:40.12 | distortion | I have a lame proprietary app that does the load balancing now, the very app i was hoping asterisk(h323) would help me get rid of |
06:40.15 | {tasker-} | we have over 250 of them on our network |
06:40.35 | {tasker-} | distortion: sigh :( |
06:40.36 | Qwell | h323 was irrelevent by the time the spec was finished |
06:40.48 | distortion | I have about 45 active h323 carriers in our network that "cant" do sip |
06:40.58 | {tasker-} | Qwell: whatever your opinion, the carriers adopted it first and it's very widespread |
06:41.04 | {tasker-} | not in the retail space |
06:41.08 | {tasker-} | SIP dominates there |
06:41.10 | Qwell | it was designed by telecom guys...so they used telecom issues |
06:41.15 | distortion | yeah, retail == sip |
06:41.16 | {tasker-} | but the retail space is nothing compared to carrier space |
06:41.36 | *** join/#asterisk hads|home (n=hads@203.109.245.87) |
06:41.37 | {tasker-} | H.323 was designed to mimick SS7 in some respects with very poor planning |
06:41.55 | distortion | well, i disagree, it has q931 which is quite nice |
06:42.00 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
06:42.01 | {tasker-} | yes |
06:42.09 | distortion | although h225 sucks balls |
06:42.21 | {tasker-} | and that's one of the driving factors keeping H.323 alive with carriers |
06:42.28 | {tasker-} | q931 |
06:42.35 | Qwell | What is q931? |
06:43.06 | distortion | its isdn signaling in voip (in lamens terms) |
06:43.11 | Qwell | why? |
06:43.33 | opus_ | its like SS7 for pbx systems that runs on the D channel from what I understand (?) |
06:43.40 | {tasker-} | it can help in diagnosing an endpoint's TDM problems |
06:43.42 | {tasker-} | for example |
06:43.47 | {tasker-} | SIP -> 404 not found |
06:43.50 | {tasker-} | can mean a dozen things |
06:43.57 | {tasker-} | q931 is very specific |
06:44.27 | {tasker-} | you get a cause code reflecting the actual error (to some degree, although some networks have gotten very loosey-goosey about that) |
06:44.29 | Qwell | besides that the requested resource couldn't be found, what else does 404 mean? |
06:44.47 | distortion | not only specific, but more importantly very relavent to TDM (isdn/ss7) messages |
06:44.47 | *** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org) |
06:44.48 | {tasker-} | hang on, i had a translation list handy |
06:45.01 | {tasker-} | some SIP errors translate into 3-4 q931 errors |
06:45.27 | dlynes | Qwell: I've got one sip error I get back from my upstream that can mean ten different things |
06:45.37 | Qwell | dlynes: which? |
06:45.40 | {tasker-} | oh |
06:45.52 | dlynes | Qwell: They don't feel like returning specific errors, because most of the switching eq doesn't understand it |
06:45.53 | {tasker-} | i know that pain all too well |
06:46.17 | Qwell | They probably aren't following the SIP spec either :p |
06:46.23 | opus_ | tasker are you running h323 and q931? |
06:46.38 | distortion | http://pastebin.com/676427 |
06:46.52 | distortion | thats the sip -> q931 mapping from my lucent gateway |
06:47.04 | dlynes | Qwell: Can't remember offhand...don't have any of those errors in my recent logs |
06:47.22 | {tasker-} | yes |
06:47.27 | {tasker-} | opus_: yes |
06:48.06 | {tasker-} | good lord |
06:48.14 | {tasker-} | i forgot how stupid error 503 was |
06:48.27 | distortion | grr 503 |
06:48.34 | distortion | 503 can mean anything |
06:48.51 | {tasker-} | "no circuit available" is a far cry from "network out of order" |
06:48.58 | *** join/#asterisk cian (n=cian@g5.cian.ws) |
06:49.12 | {tasker-} | does anyone remember the SIP -bis extension? |
06:49.19 | dlynes | Yeah...no circuit available...that's the stupid error I get once in a while |
06:49.26 | dlynes | It can mean many different things |
06:49.29 | {tasker-} | it was meant to integrate some ISDN idioms into SIP |
06:49.47 | {tasker-} | i.e. q931 cause codes |
06:49.57 | dlynes | It's error 403 or something is what my provider has it mapped into |
06:49.58 | {tasker-} | somewhere along the lines it was dropped |
06:50.01 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
06:50.11 | distortion | well i think the new thing is sigtran which i know little about but am very interested in |
06:50.16 | dlynes | Almost everything they have mapped into 403 |
06:50.23 | {tasker-} | ugh |
06:50.51 | tainted- | Qwell can u pastebin your uac_pcap.xml file? the one u pointed out generates 'No 'scenario' section in xml scenario file.' even though it validates, and ./sipp -sd uac_pcap segfaults |
06:50.53 | firestrm | anyone here good at wired iax problems? |
06:50.54 | dlynes | So whenever I have a problem with a route, I get on the msn blower to them and bitch |
06:51.12 | tainted- | firestrm what's the problem |
06:51.39 | distortion | tasker- you use * in your wholesale environment? |
06:51.49 | firestrm | im using sixtel (ya dont laugh) for long distance termination, i can make calls to cellular numbers, but not landlines.. |
06:51.56 | tainted- | the thing about asterisk is -- even when u get a crazy sip response, u can't even access it from AGI |
06:52.15 | Qwell | firestrm: That's on their side, no doubt |
06:52.29 | tainted- | firestrm sixtel is run by a joker |
06:52.39 | {tasker-} | no, just for some anciliary services |
06:52.39 | firestrm | thats what im afrade of... :( that means it will NEVER get fixed.. |
06:52.40 | dlynes | {tasker-}: another thing that really sucks about asterisk for connecting to upstreams is the lack of a g723 codec transcoder |
06:52.57 | dlynes | {tasker-}: a lot of the same guys that do h323 also seem to do g723 |
06:53.01 | {tasker-} | dlynes: allow me to fix that issue for you right now |
06:53.02 | {tasker-} | sec |
06:53.18 | dlynes | {tasker-}: ? |
06:53.22 | tainted- | dlynes yea i agree on that!! especially for intl termination |
06:53.38 | firestrm | tainted-, ya i know.. joker is one of the nicer words i would use for him, but he is one of the few that offer 250 and canadian 1-800 did's |
06:53.56 | dlynes | tainted-: 75% of my available north american termination uses g723 |
06:54.04 | tainted- | firestrm that's not termination though, that's origination |
06:54.09 | dlynes | tainted-: I have to ask my terminator to force g729 |
06:54.15 | tainted- | firestrm why don't u terminate through someone else |
06:54.40 | firestrm | ive tried, cant find anyone who gives as good quality connection as sixtel |
06:54.51 | {tasker-} | http://kvin.lv/pub/Linux/Asterisk/ |
06:54.53 | dlynes | firestrm: Group Telecom offers 1-800 dids |
06:54.54 | {tasker-} | have fun |
06:55.01 | {tasker-} | g.723.1 and g.729a |
06:55.32 | firestrm | dlynes, any good? as in , choppy audio, and delays like your on the moon? |
06:55.49 | Qwell | {tasker-}: Are you in the US? |
06:55.55 | {tasker-} | canada |
06:56.14 | distortion | tasker, have good routes to canada? :) |
06:56.16 | Qwell | mmhmm |
06:56.20 | distortion | <-- has minutes |
06:56.36 | {tasker-} | what's your volume? |
06:56.46 | {tasker-} | :) |
06:56.48 | distortion | 950k/mo |
06:56.53 | distortion | ~4t1's |
06:57.01 | {tasker-} | small volume |
06:57.26 | Qwell | {tasker-}: That violates both Intel copyright, and the codec patents... |
06:57.26 | dlynes | firestrm: They offer them on their pri's and landlines...so you'd have to find someone that can voip them for you |
06:57.34 | {tasker-} | Qwell: it violates nothing |
06:57.45 | dlynes | firestrm: try www.calltermination.com...there's probably someone on there that can do it for you |
06:57.52 | {tasker-} | Qwell: Intel provides the code base free on their website to developers |
06:57.57 | {tasker-} | Qwell: and for internal use |
06:57.58 | Qwell | WRONG |
06:58.01 | {tasker-} | nope |
06:58.05 | {tasker-} | go read their site |
06:58.08 | {tasker-} | I did |
06:58.15 | {tasker-} | already consulted a lawyer about it, too :) |
06:58.17 | Qwell | They provide it for RESEARCH ONLY, or you can get a license for non-commercial use |
06:58.18 | opus_ | Qwell: I think taske ris right |
06:58.39 | opus_ | from reading their site i came to the same conclusion |
06:58.57 | Qwell | You *MUST* get a license if you intend to USE the code, and you STILL have to pay the patent royalties |
06:59.19 | Qwell | the README file on the same damn link says the same |
06:59.24 | {tasker-} | sure |
06:59.34 | distortion | tasker: 1mm/mo customer may be small but they are the best kind ;) |
06:59.34 | {tasker-} | if you plan to use it commercially |
06:59.53 | {tasker-} | distortion: i agree |
06:59.56 | Qwell | {tasker-}: Go read the license at intel.com |
07:00.00 | {tasker-} | don't have to |
07:00.09 | {tasker-} | i already did that whole exercise |
07:00.13 | distortion | but my canada rates are dog cheap, i doubt you could touch em |
07:00.19 | {tasker-} | ITU's patent on g.729 is almost up |
07:00.24 | tainted- | woah are those 723/729 binaries? |
07:00.32 | {tasker-} | distortion: try me |
07:00.39 | {tasker-} | tainted: yup |
07:00.47 | tainted- | woah |
07:01.08 | dlynes | distortion: I'm currently getting $0.08/mi for Canada |
07:01.23 | Qwell | http://www.intel.com/cd/software/products/asmo-na/eng/perflib/ipp/219689.htm |
07:01.32 | {tasker-} | Qwell: dude, it's cool |
07:01.42 | {tasker-} | Qwell: I already did that whole exercise a year ago with the lawyers |
07:01.49 | Qwell | well, your lawyers were wrong. |
07:01.53 | tainted- | {tasker-} what kind of volume do you push? |
07:01.54 | Qwell | RTFEULA |
07:01.56 | {tasker-} | Qwell: I'm not violating anything the way we're set up |
07:02.04 | {tasker-} | Qwell: take a chill pill, dude |
07:02.04 | Qwell | {tasker-}: Do you have a license to use the code? |
07:02.19 | tainted- | Qwell well if he's doing pass-through it's legal eagle |
07:02.29 | Qwell | tainted-: and he doesn't need binaries |
07:02.30 | {tasker-} | tainted: exactly :) |
07:02.49 | Qwell | asterisk supports g723 and g729 passthrough |
07:03.00 | dlynes | Qwell: Just not very well |
07:03.03 | {tasker-} | tainted: except in the case where we play back wav files, but encoding prompts in g.729 and g.723 fixes that, too |
07:03.06 | tainted- | maybe just for the stray ulaw channel? |
07:03.09 | Qwell | dlynes: What "well"? |
07:03.14 | dlynes | Qwell: because the codec autonegotiation blows in asterisk |
07:03.20 | Qwell | dlynes: It's passthrough. It's 100% transparent |
07:03.38 | tainted- | {tasker-} what kind of volume do you do? |
07:03.41 | {tasker-} | dynes: agreed |
07:03.46 | dlynes | Qwell: Yeah...the passthrough works just fine, but the codec autonegotiation to make it all come together blows |
07:03.49 | {tasker-} | to Canada |
07:03.50 | {tasker-} | ? |
07:03.54 | {tasker-} | or in total? |
07:03.55 | tainted- | anywhere |
07:03.57 | tainted- | total |
07:04.05 | {tasker-} | about 4 million / day |
07:04.08 | dlynes | {tasker-}: pardon? |
07:04.12 | distortion | nice. |
07:04.14 | tainted- | woah |
07:04.19 | tainted- | that's small volume |
07:04.32 | distortion | we did about 750k today.. 500 on asterisk :DDD |
07:04.32 | tainted- | i do that between my polycom and next door |
07:04.42 | tainted- | distortion over how many boxes |
07:04.47 | distortion | 6. |
07:04.53 | tainted- | damn some heavy hitters in here tonight |
07:04.59 | opus_ | distortion, do they dead lock and lock up? |
07:05.14 | opus_ | how many simutanous calls can your asterisk box get? |
07:05.19 | distortion | h323 boxes, yes- sip boxes on 1.2.6 == rock solid |
07:05.26 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
07:05.32 | tainted- | distortion are u pure voip? |
07:05.33 | {tasker-} | lol |
07:05.59 | distortion | no, we have cisco/lucent gateways that pipe minutes into * |
07:06.13 | kamileon | where can i get a 1800 did ? |
07:06.35 | opus_ | distortion, really.. do you do anything in the dial plan or do you just route the calls around? |
07:06.50 | distortion | in response to taskers 'lol' asterisk has its issues but sip only the service is stable |
07:07.01 | dlynes | dood...rehan walla wallah agar has like tonnes of 1-800 dids |
07:07.12 | tainted- | lol |
07:07.14 | {tasker-} | distortion: i agree |
07:07.18 | kamileon | wtf |
07:07.19 | tainted- | rehan wallawalla? |
07:07.35 | distortion | we use asterisk for least cost routing through mysql. not much dial plan work |
07:07.36 | dlynes | tainted-: hehe...that clown in asterisk-biz :) |
07:07.49 | dlynes | tainted-: the one that's reselling other peeps dids |
07:07.50 | {tasker-} | distortion: my next try is to use a SIP/H.323 gateway between asterisk and the rest of the H.323 world |
07:07.56 | {tasker-} | has anyone looked at YATE? |
07:07.57 | opus_ | distortion whats the most one of the asterisk box can take? |
07:08.02 | distortion | yate is good |
07:08.12 | tainted- | distortion how are u balancing load |
07:08.17 | opus_ | distortion in numjber of simutaneous calls |
07:08.30 | tainted- | opus_ send me your uac_pcap.xml |
07:08.31 | kamileon | why when i switch to g729 over ulaw people tell me my calls sound worse |
07:08.33 | websae | distortion: i told tasker you would be talking to him about our rates |
07:08.35 | distortion | today, about 800 between 6 servers |
07:08.40 | tainted- | opus_ my sipp segfaults on -sd |
07:09.04 | opus_ | distortion: 200 calls on each server? |
07:09.06 | distortion | load balancing on sip side by ser |
07:09.16 | tainted- | thought so |
07:09.21 | *** part/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
07:09.29 | {tasker-} | i'm planning to use YATE as a proxy between asterisk and the h.323 providers |
07:09.38 | {tasker-} | if i could get the damn thing to compile |
07:09.39 | tainted- | i've yet to see a non ser involved * load balance solution |
07:09.50 | opus_ | tainted whats your email |
07:10.00 | distortion | ultramonkey anyone? |
07:10.12 | distortion | i purchased a f5 load balancer |
07:10.17 | distortion | faking thing worked amazing |
07:10.28 | distortion | but then it would drop sip sessions after 10 minutes |
07:10.29 | opus_ | distortion when you load balance to the box do you get 200 calls per server? is your load average on the server pretty low? |
07:10.35 | distortion | *sigh* |
07:11.33 | {tasker-} | sucks |
07:11.38 | distortion | so far i've seen about 320 calls max on a single server (g729 passthrough), cpu was at about 20% on a dual zeon 2.8 dell |
07:11.53 | {tasker-} | on asterisk? |
07:12.32 | {tasker-} | i've brought it up as high as 1600 channels on passthrough, dual xeon 3.2 800 fsb |
07:12.51 | distortion | tahts fkn pimp |
07:12.53 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
07:13.03 | {tasker-} | it's not ideal, though |
07:13.05 | opus_ | distortion: ah ha. what happeneds if you go over that amount? |
07:13.15 | {tasker-} | max 1200 channels to keep it cruising |
07:13.34 | opus_ | distorition, or.. how often does your asterisk box go down? |
07:13.51 | {tasker-} | h.323 -> every hour |
07:13.54 | {tasker-} | SIP - never |
07:14.00 | *** part/#asterisk rogercharlie (n=adrian@c-69-181-20-122.hsd1.ca.comcast.net) |
07:14.01 | distortion | agreed |
07:14.03 | distortion | sip never |
07:14.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
07:14.23 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
07:14.33 | opus_ | crap.. you guys have it easy, all you are doing is routing calls:) |
07:14.40 | opus_ | (you could do that in SER!) :) |
07:14.43 | distortion | ... |
07:15.22 | {tasker-} | SER can proxy media or pass it to the other end, right? |
07:15.29 | distortion | yes. |
07:15.33 | {tasker-} | allow two endpoints to connect media channels |
07:15.38 | {tasker-} | just like GnuGK |
07:15.53 | distortion | its much better than * on the media direct connect, they call it "record route" |
07:16.02 | {tasker-} | on both GnuGK and SER, we've never been able to hit as of yet |
07:16.06 | opus_ | coppice, are you still around |
07:16.11 | distortion | asterisk fucks itself on billing if it doesnt proxy media |
07:16.11 | {tasker-} | to hit a limit as of yet |
07:16.15 | {tasker-} | i missed a word :( |
07:16.43 | {tasker-} | GnuGK and SER can easily take several DS3s |
07:16.59 | {tasker-} | the problem is connecting SER and GnuGK clients together :( |
07:17.05 | {tasker-} | but GnuGK also never goes down |
07:17.19 | opus_ | GnuGK ?? never heard of it |
07:17.27 | {tasker-} | gnu gatekeeper |
07:17.30 | {tasker-} | http://gnugk.org |
07:17.33 | {tasker-} | open gatekeeper |
07:17.38 | {tasker-} | built on openh323 |
07:17.39 | distortion | i dislike gnugk |
07:17.46 | distortion | but it works |
07:17.47 | {tasker-} | it's flawless |
07:17.57 | distortion | its h323 implementation is very specific |
07:18.02 | distortion | and euro based |
07:18.02 | {tasker-} | stock it's missing stuff |
07:18.09 | {tasker-} | we added some code mods |
07:18.14 | distortion | i dont like euro h323 |
07:18.18 | {tasker-} | lol |
07:18.25 | {tasker-} | i've never heard of euro h.323 |
07:18.26 | distortion | its impossible to interop with |
07:18.34 | {tasker-} | we have no issues |
07:18.57 | {tasker-} | we have carriers going through it from europe, usa, canada, latin america, asia and north africa |
07:18.58 | distortion | connect with something that doesnt re-write h225 and you'll have fun |
07:19.00 | {tasker-} | no interop issues |
07:19.25 | {tasker-} | rewrite h225? |
07:19.55 | distortion | ok, pop quiz: q931 dialed number: 19492744000 h225 dialed digits: 200519492744000 what will gnugk route off of? |
07:20.20 | opus_ | the one that i am not familar the most with of course:) |
07:20.26 | {tasker-} | h225 |
07:20.39 | {tasker-} | i've seen that issue |
07:21.08 | {tasker-} | iv'e seen mixed prefixes on q931 and h225 |
07:21.28 | distortion | hoho |
07:21.31 | {tasker-} | but part of the mods we added manage all of that in the routing algorithm |
07:21.50 | {tasker-} | we don't use the stock static routing garbage |
07:21.51 | {tasker-} | lol |
07:22.04 | distortion | well, i dont hate you then |
07:22.05 | {tasker-} | those guys aren;t carrier guys, though they've put together a nice tool |
07:22.14 | {tasker-} | give me a break, man |
07:22.28 | {tasker-} | even SER needed a few sprinkles for us |
07:22.33 | opus_ | is there anyway to do intelligent routing in SER? like , these two calls are related: send them to the same server |
07:22.39 | {tasker-} | our routing matrix is fairly advanced |
07:22.48 | distortion | I've interconnected with some fools out of San Fran that claim to have written gnugk and they ouldnt figure out the q931/h225 routing issues |
07:22.56 | distortion | i think they scarred me |
07:23.06 | {tasker-} | you can input a million codes and routing decision still only takes an average of 3ms - 8 ms |
07:23.30 | {tasker-} | ouch |
07:23.31 | {tasker-} | lol |
07:26.51 | opus_ | so, is there anyway to load balance between servers with SER with some degree of intelligence(like my example) |
07:27.17 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-242.claranet.co.uk) |
07:27.53 | {tasker-} | open_: yes |
07:28.03 | websae | tainted- are you alive? |
07:28.10 | {tasker-} | open_: but you'd have to build the intelligence through scripting |
07:28.55 | {tasker-} | damn |
07:29.04 | {tasker-} | i meant opus_ |
07:29.34 | opus_ | is there a way to do it after asterisk has already picked up the line? |
07:30.09 | opus_ | can the scripting do database look ups or would i have to hack that in |
07:30.53 | {tasker-} | i don't recall off-hand |
07:33.21 | opus_ | SER doesn't require any crazy timing device, i could safely run it in vmware right? |
07:33.44 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
07:33.44 | tainted- | opus_ yes, i have |
07:34.41 | opus_ | do I want to get SER or OpenSer? |
07:34.48 | opus_ | probably openser right? |
07:34.54 | tainted- | yea |
07:35.08 | tainted- | more activity |
07:36.09 | {tasker-} | ser is pretty much dead |
07:36.16 | opus_ | ? |
07:36.21 | opus_ | what happened |
07:36.28 | {tasker-} | it's moved to openser |
07:36.34 | {tasker-} | openser is at release 1.1.x |
07:38.00 | {tasker-} | the openser repository has the SER revisions up to OpenSER |
07:38.13 | tainted- | no |
07:38.22 | tainted- | ser != dead |
07:38.34 | {tasker-} | for all intents and purposes, it is |
07:38.46 | {tasker-} | openser continues from where it left off on revisions |
07:40.39 | *** join/#asterisk {tasker-} (n=ghes@modemcable252.110-83-70.mc.videotron.ca) |
07:40.57 | {tasker-} | my wireless router is acting up |
07:41.02 | {tasker-} | stupid linksys |
07:41.12 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
07:50.41 | *** part/#asterisk {tasker-} (n=ghes@modemcable252.110-83-70.mc.videotron.ca) |
07:50.52 | tainted- | master of minutes enslaved by wifi |
07:51.17 | distortion | he has no minutes :) |
07:51.54 | tainted- | u took them all already? |
07:51.55 | tainted- | haha |
07:52.09 | distortion | not yet, i think im going to send all mine to him |
07:52.29 | distortion | he's trying to convince me that i should ditch asterisk |
07:52.34 | distortion | im starting to agree |
07:52.37 | distortion | :( |
07:52.37 | tainted- | and go with |
07:52.46 | distortion | ser/openser |
07:52.55 | tainted- | i'm heading to freeswitch as soon as it's stable for production environment |
07:53.08 | tainted- | that is a slippery slope |
07:53.17 | tainted- | ser is amazing |
07:53.19 | tainted- | BUT |
07:53.34 | tainted- | doing billing w/ ser is a serious commitment |
07:53.50 | distortion | we havent invested enough time into seeing if it will work for our routing/billing needs |
07:54.02 | distortion | into "ser" |
07:54.12 | tainted- | what do u use for LCR |
07:54.38 | distortion | i use both "rate-engine" and a modified "lcdial" |
07:55.10 | distortion | i love the concepts, they are pretty close |
07:55.34 | distortion | rate-engine is good for routes that have congestion because it returns multiple route lists |
07:55.59 | distortion | lcdial is cool because its logic is based on sql which is customizable |
07:57.23 | tainted- | i see |
07:57.51 | tainted- | i'd recommend something db-driven if ur headed into SER land |
07:58.22 | distortion | i would love it, im much more db wise than programming wise |
07:59.15 | tainted- | i built a lcr engine from scratch |
07:59.24 | websae | tainted- you did? |
07:59.49 | tainted- | yea |
07:59.52 | distortion | i built one as well- ms sql based- very nice. but the switch that uses it sucks |
08:00.09 | tainted- | and a web-based billing solution to match |
08:00.22 | websae | tainted- can I buy that from you? |
08:00.25 | tainted- | distortion how is * talking to mssql in your setup right now |
08:00.29 | tainted- | websae i don't sell it |
08:00.33 | distortion | and as i told twisted- i would shoot myself if i used tds w/asterisk |
08:01.15 | tainted- | freetds is based on old sybase tsql lol |
08:01.21 | distortion | tainted- i dont, i use it for our old h323 platform which im trying to replace w/asterisk (as soon as asterisk's h323 becomes worthy) |
08:01.26 | tainted- | but works w/ mssql 7.0/2000 |
08:01.52 | tainted- | *'s h323 is shit |
08:02.11 | distortion | well, it could work, yes- but would require an application to use it and im not a programmer |
08:02.15 | tainted- | how do they talk then? gw & mssql |
08:02.38 | distortion | no, h323 session boarder controller runs on MS server 2003 (gasp) |
08:02.42 | tainted- | my solution is fastagi based and scales beautifully |
08:03.11 | tainted- | i can't wait to do clustering w/ freeswitch though |
08:03.24 | tainted- | they intend on release in win32 platform |
08:04.01 | distortion | man, i cant wait until this niche overpowers the wholesale switch makers (nextone, sansay ect) |
08:04.13 | distortion | frakin nextone wants 100k for 1000 ports |
08:04.24 | tainted- | it will be very very soon |
08:04.29 | distortion | and people buy them left/right |
08:04.35 | tainted- | there is so much activity right now |
08:04.37 | tainted- | it's great |
08:04.38 | distortion | i have like 15-20 nextones in my datacenter |
08:04.45 | tainted- | damn |
08:05.14 | tainted- | i guess i'm fortunate to run sip end-to-end |
08:05.15 | distortion | some jackass has 3 of them fully loaded, (which is like $800k) |
08:05.23 | distortion | yes you are |
08:05.27 | tainted- | not really into playing the penny pinching wholesale game |
08:05.37 | distortion | :'( |
08:05.51 | distortion | <-- is :( |
08:06.26 | tainted- | why? |
08:06.31 | tainted- | did u start out in that market? |
08:06.41 | distortion | yeah, i think its all i know |
08:06.46 | distortion | sad |
08:06.52 | tainted- | only folks making money in that is intl gray routes nowadays |
08:07.11 | distortion | well, that's my co's #1 initiative |
08:07.12 | tainted- | grey too lol |
08:07.17 | distortion | which is sad |
08:07.23 | distortion | cause i hate grey routes |
08:07.24 | tainted- | what do u mean |
08:07.29 | tainted- | yea |
08:07.40 | opus_ | is there anyway SER can stop an already SIP session, like "oh shit, that asterisk box is blown !" |
08:07.41 | distortion | they make lots of $$ but its fraking way too much maintenance |
08:07.51 | opus_ | and reroute it to another server? |
08:08.31 | distortion | existing session? |
08:08.36 | tainted- | distortion what co? |
08:08.43 | opus_ | yes |
08:08.45 | tainted- | do u work for an itsp? |
08:09.21 | distortion | yeah, comsolo is the international side- zues is the domestic. |
08:09.59 | distortion | opus: i think you need to run sip(tcp) to do that which means it cant be done w/asterisk yet |
08:10.54 | distortion | but it will work for "next call" which is how we use it |
08:11.44 | opus_ | is there any way of doing a transparent proxy so that all my rtp data comes from one IP? |
08:11.53 | tainted- | distortion i don't see how pure routing is a long term viability |
08:12.51 | tainted- | distortion i guess right now the protocols are still settling so there's a niche for transcoders.. |
08:12.52 | distortion | opus: use ser and a "virtual ip" solution like ultramonkey- there are several out there |
08:13.11 | opus_ | ok.. hm |
08:13.20 | distortion | tainted- pure routing? |
08:13.35 | tainted- | opus_ i'm telling u man.. voip slashdot is the next big thing lol |
08:13.42 | tainted- | distortion origination/termination |
08:14.02 | distortion | tainted- with this industry its less becomming about what you know and how much money you have, and more about who you know |
08:14.33 | distortion | tainted- $$ was always the big barrier to entry and with these new open techs, that barrier is fading. |
08:14.49 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
08:15.13 | tainted- | yea no kidding! |
08:15.21 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
08:15.32 | distortion | tainted- so i agree, those who are closer to the end user will prevail, but there will always be room for arbitrage since the big guys dont want to deal with the small-mid customers |
08:15.41 | tainted- | $1/DID, 0.01/US org/term |
08:16.27 | distortion | .01 w/ani? :) |
08:16.51 | tainted- | dunno.. just numbers i see on whatever peering fabrics/exch etc |
08:17.12 | distortion | there are always catches |
08:17.37 | tainted- | i figured |
08:17.39 | distortion | i came accross this lame company offering .008 flat us |
08:17.43 | tainted- | but it's headed that direction |
08:18.05 | tainted- | there's calltermination.com, voipmatch.com, ipxc.com etc etc |
08:18.06 | distortion | i signed up, sent minutes and they charged me .03/min cause i was sending "off-net" |
08:18.08 | tainted- | it's the trend |
08:18.27 | tainted- | thevpf.com |
08:18.35 | distortion | im a member :) |
08:18.47 | distortion | i went to the conference in miami |
08:18.48 | websae | it's all about the quality......the support........and the follow through |
08:19.21 | distortion | i actually think that the vpf (or the like) is the future for voip |
08:19.34 | distortion | frak the internet |
08:20.35 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
08:20.47 | distortion | brb |
08:22.19 | *** join/#asterisk Kernel_Core (n=I@193.251.135.118) |
08:22.37 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
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08:30.52 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
08:31.06 | opus_ | cool |
08:37.11 | *** join/#asterisk lorinc (n=ang@caracas-2717.adsl.interware.hu) |
09:02.16 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
09:06.34 | *** join/#asterisk ketil (n=chatzill@217-131-74.5001.adsl.tele2.no) |
09:07.31 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
09:07.47 | websae | so where's everyone from here? |
09:07.51 | websae | United States here |
09:10.55 | OliverX | G. W. Bush |
09:11.03 | websae | hahaha |
09:11.04 | websae | yes |
09:11.11 | websae | where are you from OliverX? |
09:11.40 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
09:20.27 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
09:22.20 | [Airwolf] | 'morning |
09:22.48 | *** join/#asterisk pengyong (n=lala@218.93.119.110) |
09:22.53 | [Airwolf] | I want to give users a SIP accounts and IAX2 account. But woth with the same extension. |
09:24.14 | [Airwolf] | But I want to connect to the account where the user is logged on. |
09:24.43 | [Airwolf] | Now I solve this like this: exten => s,1,Dial(SIP/user&IAX2/user,20) |
09:25.22 | [Airwolf] | It works, but i'm not really satisfied with it, because it will generate an error on the account where the user isn't logged in. |
09:25.40 | [Airwolf] | And I was wondering if someone knows another way of handeling this situation. |
09:28.40 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
09:34.27 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
09:34.44 | Shaun222 | [Airwolf]: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail might do the trick, not sure... |
09:34.54 | Shaun222 | seams like you could check the channel first and then send the call... |
09:39.58 | *** join/#asterisk blop (i=blop@openbeer.be) |
09:40.55 | *** part/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
09:41.43 | *** join/#asterisk CrashHD (n=timf@c-67-182-168-37.hsd1.ca.comcast.net) |
09:44.19 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-168-37.hsd1.ca.comcast.net) |
09:45.39 | [Airwolf] | Shaun222, that was what I was looking for. |
09:45.43 | [Airwolf] | Thanks |
09:52.26 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
09:53.00 | int | hi |
09:53.04 | int | i got a new warning when reloading, Apr 23 11:51:45 WARNING[8280]: chan_zap.c:11050 setup_zap: Ignoring signalling => any idea where it does come from, |
09:53.07 | int | ? |
09:56.40 | blop | k its nothing :) |
10:04.29 | diLLec | int: if you reload asterisk those parameters are ignroed |
10:04.32 | diLLec | ignored |
10:05.19 | diLLec | do a fully restart and they will be applied |
10:11.16 | blop | i also get : |
10:11.16 | blop | Apr 23 12:04:40 NOTICE[8352]: chan_zap.c:6184 ss_thread: Got event 18 (Ring Begin)... |
10:11.16 | blop | Apr 23 12:04:42 NOTICE[8352]: chan_zap.c:6184 ss_thread: Got event 2 (Ring/Answered)... |
10:11.32 | blop | which i did not before, and the caller id isnt working anymore on the fxo :( |
10:11.40 | blop | cant figure out why |
10:23.38 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
10:25.44 | *** join/#asterisk ToTo (n=ToTo@host12-137.pool879.interbusiness.it) |
10:29.06 | *** join/#asterisk RoyK (n=roy@cD9088681.inet.catch.no) |
10:57.37 | *** join/#asterisk BearMan (i=karsten@freenode/staff/sourcemage.wizard.BearPerson) |
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11:15.54 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
11:16.36 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
11:31.16 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
11:34.21 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
11:34.30 | jhiver | ~seen tzanger |
11:34.35 | jbot | tzanger is currently on #asterisk (9d 12h 25m 5s). Has said a total of 492 messages. Is idling for 9h 5m 42s, last said: 'yeah it's spitting out white noise and listening, right? I think... it's been a while'. |
11:36.49 | tzafrir_laptop | someone asked me if "asterisk supports Radius". I can't find almost any trace for direct support. I figure that there are some workarounds for billing, right? |
11:41.43 | *** part/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org) |
11:45.00 | tzafrir_laptop | anybody? |
11:46.28 | RoyK | sp,ebpdy |
11:46.31 | RoyK | somebody |
11:47.34 | RoyK | tzafrir_laptop: ask jerjer. he loves radius ;) |
11:51.35 | Druken | radius.... pfft |
11:54.59 | RoyK | tzafrir_laptop: are you in .il? |
11:55.07 | tzafrir_laptop | RoyK, yes |
11:55.35 | tzafrir_laptop | (and not in il.us) |
11:55.43 | RoyK | didn't think you were israeli..... |
11:57.41 | *** join/#asterisk saftsack (n=saftsack@p54A7C470.dip.t-dialin.net) |
12:01.09 | coppice | some people are just diametrically opposed to radius |
12:11.13 | loonacy | radius is for squares. |
12:23.25 | *** join/#asterisk pengyong (n=lala@222.185.18.214) |
12:31.56 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
12:32.04 | PakiPenguin | hello everyone |
12:39.23 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
12:45.12 | *** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca) |
12:46.14 | Ateboy | short question about options to Dial()... how do you separate multiple options? I want to use m and A()... |
12:50.06 | Ateboy | I've read 'show application dial' and I'm at page 244 of the book, but I can't see examples of more than one option |
12:58.02 | RoyK | aM |
12:58.13 | RoyK | no separation or anything |
13:01.36 | Druken | uhg... |
13:02.13 | Druken | i remember bitching about dial-up, but now i'm bitching about 800k up.... pretty sad... |
13:02.41 | Ateboy | so it would be a(file)m... ok thanks RoyK |
13:03.12 | Ateboy | I'll try that |
13:06.22 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-189-92.dsl.chcgil.sbcglobal.net) |
13:07.09 | Flauto | -- Got SIP response 500 "Server Internal Error" back from 209.47.41.48 |
13:07.22 | Flauto | i got a lot of this kind of stuff |
13:07.30 | Flauto | is there anything i can do |
13:07.38 | Flauto | everything seems working |
13:12.35 | Ateboy | are you using vbuzzer.com? |
13:15.11 | Ateboy | I'm trying to use the 'w' option now... doesn't seem to be working |
13:15.52 | Ateboy | I've uncommented the line 'automon => *1' in features.conf, but whey I do *1, nothing happens (* CLI shows nothing) |
13:19.42 | Ateboy | hmmm, sunday morning doesn't seem like the best time to ask questions ;) |
13:23.06 | Flauto | ateboy, i am using vbuzzer |
13:23.22 | Flauto | i don't know why |
13:23.40 | Flauto | it is giving me so many these |
13:26.21 | Ateboy | flauto: maybe you should ask them... |
13:27.32 | Ateboy | flauto: you have connected your * box to vbuzzer? |
13:28.03 | *** join/#asterisk Skymarshal (n=Skymarsc@p54AF3DC8.dip0.t-ipconnect.de) |
13:28.13 | Flauto | yes |
13:28.16 | Flauto | it is connected |
13:28.18 | Flauto | and it is working |
13:30.03 | Ateboy | sound like a very good way to get cheap long-distance :) |
13:30.23 | Ateboy | what protocol do they support? |
13:30.36 | Flauto | you mean vbuzzer? |
13:30.40 | Skymarshal | Hi, what is the best why to search for a year in a timebased include? |
13:30.45 | Ateboy | flauto: yes |
13:30.47 | Flauto | i am really not using it much for long distance |
13:30.59 | Ateboy | flauto: what do you use it for? |
13:31.03 | Flauto | i use it mostly for inbound |
13:31.16 | Ateboy | then, a cheap did :), works well? |
13:31.17 | Flauto | and calling toronto area code 416 |
13:31.33 | Flauto | it has been working pretty well |
13:31.53 | Ateboy | well, the 500 error you get, I think only them can tell you what is going on... |
13:31.57 | Flauto | this problem started only a few days ago |
13:32.09 | Ateboy | possible |
13:32.31 | Flauto | it seems that now, they have two ip addresses |
13:32.31 | Ateboy | but since it is their servers that return this message, I think they're the only one that can answer you... |
13:33.15 | Flauto | but they don't support anything other than their softphone |
13:33.30 | Flauto | ateboy, there is a new service is called icall.com |
13:33.44 | Flauto | that one provides free unlimited long distance |
13:33.54 | Flauto | but the problem is that it does not work with asterisk |
13:33.54 | Ateboy | thanks |
13:34.05 | Flauto | you have to use their softphone as well |
13:34.17 | Ateboy | ok, so you're telling me you're not using asterisk? |
13:34.30 | Flauto | i tired to config is to my asterisk |
13:34.34 | Flauto | it did not work at all |
13:34.42 | Flauto | i am using it |
13:34.55 | Flauto | but can not make icall to work on my asterisk |
13:35.18 | Ateboy | if they force you to use their softphone, they're probably using a proprietary protocol, like skype |
13:35.48 | Flauto | they are using sip |
13:35.58 | Ateboy | hmmmi |
13:36.41 | Ateboy | they must be doing something else |
13:37.29 | Ateboy | but if you didn't get the 500 error on your asterisk, you shouldn't be asking on an asterisk list.... or at least say it (maybe you said it before I got in the room, though) |
13:39.01 | Flauto | that is okay |
13:44.24 | Flauto | hope someone will soon figure out how to config icall to asterisk |
13:44.37 | Mw3 | hm, is there any problem with fwd at the moment? it seems to be down :( |
13:45.23 | RoyK | ~seen wasim |
13:45.33 | jbot | wasim <n=wasim@pdpc/supporter/active/wasim> was last seen on IRC in channel #asterisk, 3d 23h 43m 33s ago, saying: 'guigouz: an fxo/sip ata'. |
13:45.33 | Flauto | at beginning, i had problem with vbuzzer too but later, i figured it out and then, i saw the config on the wiki as well |
13:45.44 | Flauto | mw3 |
13:45.46 | Flauto | let me try |
13:47.45 | Flauto | mw3, you are right |
13:47.47 | Flauto | not working |
13:48.52 | *** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
13:48.53 | Mw3 | even the web site is down as i see |
13:49.23 | Mw3 | dns cannot resolve the addresses (iax.fwdnet.net fwd.pulver.com www.freeworlddialup.com) |
13:51.54 | RoyK | Mw3: I can't resolv them from here either |
13:52.13 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
13:52.19 | Mw3 | ok thanks :) i hope it will be back soon |
13:58.47 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
14:01.40 | RoyK | hej hej |
14:01.56 | coppice | is this news? |
14:02.06 | Flauto | hehe |
14:02.15 | RoyK | :) |
14:02.39 | *** join/#asterisk jhiver_ (n=jhiver@89-114.206-83.static-ip.oleane.fr) |
14:02.44 | jhiver_ | ~seen tzanger |
14:02.48 | jbot | tzanger is currently on #asterisk (9d 14h 53m 18s). Has said a total of 492 messages. Is idling for 11h 33m 55s, last said: 'yeah it's spitting out white noise and listening, right? I think... it's been a while'. |
14:03.03 | jhiver_ | ~seen jhiver |
14:03.04 | jbot | jhiver <n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr> was last seen on IRC in channel #asterisk, 2h 28m 34s ago, saying: '~seen tzanger'. |
14:03.19 | jhiver_ | :) |
14:03.45 | jhiver | hi all |
14:04.06 | RoyK | ~lart jhiver |
14:04.30 | jhiver | blue or red? |
14:04.33 | jhiver | the pill :) |
14:05.00 | coppice | sounds like an ad for 7-up :-) |
14:06.34 | jhiver | I've been playing with SER recently |
14:06.46 | jhiver | it's pretty good but its config files are such a mess |
14:06.57 | coppice | ah, everyone has a sad story to tell |
14:07.00 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
14:07.10 | jhiver | and also the documentation is piss-poor / out of date... I guess I should contribute :) |
14:07.38 | jhiver | I'm thinking of writing a ser.cfg Perl generator but I'm wondering what a good syntax might bze |
14:07.48 | coppice | refactoring code is fine, but most project need to much more seriously refactor their configuration scheme |
14:08.00 | jhiver | because actually this software doesn't seem to do very much at all :) |
14:08.20 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
14:08.30 | [TK]D-Fender | Godd morning all..... |
14:08.32 | jhiver | mind you that's also a strengh... do something simple and do it well |
14:08.33 | jhiver | hi |
14:09.21 | [TK]D-Fender | Quick question : has a recent release of * broken unauthenticated incoming SIP calls? My config used to work, and those of a few people I talk with and none of ours seem to work any more... |
14:09.59 | jhiver | oh, sounds dangerous |
14:10.12 | jhiver | maybe I'll wait a little before I upgrade from 1.0.9 :) |
14:10.35 | russellb | [TK]D-Fender: when upgrading to 1.2 from 1.0, you need allowguest=yes |
14:10.39 | Druken | i'm afraid to upgrade... i'd have to rebuild my entire dialplan... :( |
14:10.41 | russellb | in sip.conf |
14:10.45 | [TK]D-Fender | russellb : Always been doing that... |
14:10.58 | jhiver | mind you the 'n' priority is just so right :) |
14:11.19 | russellb | I hope more and more people start using AEL in 1.4 |
14:11.30 | Druken | ael? |
14:11.44 | Maxxed | morning' folks :) |
14:11.46 | jhiver | russellb, why? it doesn't look very good, too much like a programming language |
14:11.51 | [TK]D-Fender | All Efforts Lost :) |
14:11.59 | jhiver | As if SER configs wasn't enough :) |
14:12.09 | russellb | jhiver: it is a language |
14:12.14 | Maxxed | hey, is there a way to have a caller thats been in a queue for a long enuff time to get a voicemail box? |
14:12.19 | jhiver | well |
14:12.24 | Druken | ser's config is fuct |
14:12.38 | Maxxed | ie, caller sits in queue for 10min, they get vmail |
14:12.52 | jhiver | if you want a language for a dialplan then maybe use python or perl or something |
14:12.57 | Maxxed | i know i can gotoif a var n such |
14:13.06 | russellb | Druken: http://svn.digium.com/view/asterisk/team/murf/AEL2/doc/ael.txt?view=markup |
14:13.08 | jhiver | not some other stupid _new_ language |
14:13.18 | Maxxed | but is there an easy, or rather nicer way to do it |
14:14.08 | jhiver | I think dial plans should look like access list |
14:14.41 | jhiver | as in: condition1 condition2 condition3 action |
14:14.41 | [TK]D-Fender | russellb : I get "SIP/2.0 407 Proxy Authentication Required" when I get the call.... though I have "allowguest=yes, and there is an exten for his "to:" header |
14:16.38 | russellb | too early to look a SIP trace |
14:17.32 | *** join/#asterisk mutilator (i=WebChat@65.111.201.122) |
14:17.33 | [TK]D-Fender | Well there is a context in [general], there is a matching exten in extensions.conf, and in it an exten named "andrew" which is what he's dialing |
14:17.47 | [TK]D-Fender | That sounds like all 3 pieces to me... |
14:18.00 | [TK]D-Fender | (and the fact it used to work...) |
14:22.49 | blitzrage | madness |
14:23.30 | *** join/#asterisk uski (n=uski@ALagny-151-1-83-209.w86-198.abo.wanadoo.fr) |
14:25.53 | [TK]D-Fender | ugh.. |
14:26.43 | RoyK | ugh.. |
14:27.44 | Maxxed | eh.. |
14:28.11 | Druken | blah.... |
14:36.20 | [TK]D-Fender | Yeah, its happening on 3 servers now... |
14:36.23 | [TK]D-Fender | dammit |
14:36.54 | *** join/#asterisk Assid (n=assid@203.115.64.8) |
14:40.35 | [TK]D-Fender | Might have something to do with SIP domain? |
14:52.00 | RoyK | http://freewlan.org/ <-- My latest project :) |
14:56.01 | Maxxed | so yeah, any idea how to time out a queue |
14:56.15 | Maxxed | sombody in the queue for 5min gets voicemail |
14:56.37 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
14:56.43 | Maxxed | i have the time out set in the queue.conf |
14:56.54 | *** join/#asterisk asteriskmonkey (n=phil@69.158.144.16) |
14:56.54 | Maxxed | but that dont seem to work the way i think it would |
14:57.19 | asteriskmonkey | hey , has anyone ever ordered the g729 codec licenses from digium before? |
14:58.04 | Maxxed | i havent |
14:58.06 | Maxxed | i thought about it |
14:58.13 | Maxxed | but i dont know what all the hype is about |
14:58.21 | Assid | i have |
14:58.22 | Maxxed | like really.. what makes it so great? |
14:58.24 | Assid | takes around 10 mins |
14:58.26 | *** join/#asterisk apardo (n=apardo@87.217.144.163) |
14:58.29 | asteriskmonkey | im trying to find out if they email you the key or you have to wait for it in the mail |
14:58.36 | Assid | wait for the eamil |
14:58.40 | asteriskmonkey | sweet |
14:58.46 | Assid | i think they send out the email manually |
14:58.47 | Maxxed | Assid: is it really all that badass? |
14:58.54 | Assid | badass? wouldnt know |
14:58.57 | Assid | just a damn codec |
14:59.07 | Maxxed | yeah.. bump paying for just a codec |
14:59.14 | asteriskmonkey | Maxxed: just a goode codec to use if you have little bandwidth |
14:59.24 | Maxxed | ah |
14:59.41 | asteriskmonkey | so if you wanna do 20 lines of voip on a dsl you would want g729 |
14:59.58 | Maxxed | damn, 20 lines on dsl |
14:59.58 | Assid | im thinking of that |
15:00.01 | Maxxed | is that even posible |
15:00.04 | asteriskmonkey | yes |
15:00.08 | Assid | there is only 1 issue |
15:00.14 | Assid | your carrier should be 729 compliant |
15:00.16 | asteriskmonkey | have lots of people using audiocodes gatways with that :) |
15:00.41 | asteriskmonkey | im a carrier trying to become g729 compliant lol |
15:01.00 | Assid | i actually wanna be a carrier |
15:01.06 | Assid | maybe on a retail level |
15:01.09 | asteriskmonkey | do you have a t1 or ds3 |
15:01.11 | Assid | i know ic an get tons of business |
15:01.19 | asteriskmonkey | Assid where are you located |
15:01.27 | Assid | me? personally ?india |
15:01.40 | asteriskmonkey | ah |
15:01.49 | Assid | hence why good opportunity to start up |
15:01.50 | asteriskmonkey | to many grey routes for voip there |
15:01.57 | Assid | well |
15:02.01 | Assid | im not gonna link into PSTN |
15:02.06 | Assid | so im doing it all legal |
15:02.30 | asteriskmonkey | i guess but then you cant sell termination to pstn to other providers |
15:02.38 | Assid | err.. no |
15:02.43 | Assid | dont wanna do that |
15:03.10 | Assid | instead of letting the west communicate with east.. im gonna let east communicate with west |
15:03.27 | ManxPower | Communications is far overrated. |
15:03.32 | Assid | well |
15:03.39 | Assid | so long as it pays my phone bills |
15:03.40 | Assid | hehe |
15:03.48 | [TK]D-Fender | Just got it... I have an ext configured mathcing the caller ID of the inbound call and it thinks its a LOCAL phone.... |
15:04.08 | Assid | huh?!?!? |
15:04.13 | [TK]D-Fender | note to self : Make all []'s in sip.conf RANDOM looking... |
15:04.26 | Assid | random looking huh? |
15:04.28 | Assid | why |
15:04.53 | [TK]D-Fender | Assid : read up... |
15:05.10 | ManxPower | a MAC works |
15:05.33 | [TK]D-Fender | ManxPower : Good for hardphones, less so for softphones, but at least it'd be meaningful where applicable... |
15:05.49 | ManxPower | all softphones suck! |
15:06.05 | Ateboy | lol |
15:06.19 | Assid | how is your pbx getting a matching caller id as incoming call |
15:06.42 | Assid | you using whole numbers per extension? |
15:07.27 | Assid | i wanna start playing with realtime soon |
15:08.29 | asteriskmonkey | if you have a large did range just use a matchign expression |
15:09.12 | Assid | you know.. we really should have a universal dundi and open access system.. to try and make things cheaper for everyone |
15:09.22 | asteriskmonkey | example _4165983XXX,1,Dial(SIP/${exten}) |
15:09.38 | asteriskmonkey | assuming you name your sip accounts as the did |
15:09.58 | asteriskmonkey | assid: its called msn |
15:10.06 | Assid | huh? |
15:10.29 | Assid | msn doesnt do jack |
15:10.52 | asteriskmonkey | makign a global dundi network would be a disaster |
15:10.55 | [TK]D-Fender | ManxPower : True, but its what my mother can use in her case :) |
15:11.07 | ManxPower | all softphones suck! |
15:11.08 | asteriskmonkey | you could 1)never assure quailty 2)never prevent idiots abusing it |
15:12.01 | asteriskmonkey | if you want an example of uber crudy dundi check out voip discoutn with over 32 countrys to call for free and all you get is pops and audio drops |
15:12.22 | Assid | well. quality should theoretically be better .. as its a voip-voip link without any traditional phones in the middle to mess it up |
15:12.33 | Assid | hrmm |
15:12.40 | Assid | audio hasnt been droppng for me |
15:12.43 | Assid | although |
15:12.54 | Assid | they arent supposed to be charging me for my calls to hongkong |
15:13.15 | coppice | Assid: nobody charges me for calls to Hong Kong |
15:13.31 | Assid | coppice: know any free providers? |
15:13.38 | asteriskmonkey | lots |
15:13.50 | asteriskmonkey | sipdiscount, voipdiscount, voipbuster |
15:13.51 | coppice | Yeah. the local telco. i live in HK :-) |
15:13.52 | asteriskmonkey | lots |
15:14.04 | Assid | err.. thats all the same.. |
15:14.14 | asteriskmonkey | yep |
15:14.15 | Assid | sipdiscount/voipdiscount and voipbuster belong to the same guys |
15:14.19 | *** join/#asterisk Hali_303 (n=surfk@dsl540096E3.pool.t-online.hu) |
15:14.21 | asteriskmonkey | woo just got my g729 codecs |
15:14.38 | Assid | nice |
15:14.42 | Assid | enjoy:D |
15:14.44 | Hali_303 | asteriskmonkey: how much do they cost? |
15:14.52 | asteriskmonkey | 10$ per channel |
15:15.04 | Hali_303 | and how is a channel defined? |
15:15.11 | asteriskmonkey | a call |
15:15.39 | asteriskmonkey | an audio speech path :P |
15:15.41 | Assid | active line is a channel |
15:15.42 | Hali_303 | hmm i see. and why is it better than speex at low bitrate? |
15:15.48 | Assid | speex sucks |
15:15.55 | Assid | sounds like robots |
15:15.57 | asteriskmonkey | better quailty |
15:16.09 | asteriskmonkey | g729 sounds pretty good and only eats up 8kbps |
15:16.14 | Jacke | interoperability ;> |
15:16.17 | asteriskmonkey | +overhead |
15:16.25 | Hali_303 | I used speex@7600bps and it was a bit robotic but not much |
15:16.29 | asteriskmonkey | ualw use nearly 90k with overhead |
15:16.35 | coppice | if speex sounds bad at 8k, something is wrong with your setup |
15:16.45 | asteriskmonkey | try explaining codecs to custoemrs.. they only care about quailty |
15:16.51 | coppice | g729 uses 30K with overheads |
15:16.58 | Hali_303 | asteriskmonkey: yeah that is true.. |
15:17.03 | Assid | G.729 8 31.2 |
15:17.12 | Hali_303 | asteriskmonkey: no hardware phones w/ speex inside? |
15:17.18 | Assid | ethernet bandwith = 31.2 |
15:17.34 | Assid | around 3.9K/sec |
15:17.47 | Assid | so effectively 4K/sec up/down you have a clear call |
15:17.56 | Assid | thats good enough for even a 56kbps dialup |
15:17.57 | RoyK | coppice: huh? IP+UDP+RTP overhead with 20ms packetization is 16kbps |
15:18.01 | coppice | asteriskmonkey: If your customer care about quality, how come that will accept G.729? |
15:18.02 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
15:18.02 | RoyK | coppice: so 24 in total |
15:18.12 | coppice | RoyK: try again |
15:18.26 | RoyK | why? |
15:18.28 | Assid | coppice: effectively .. 729 is lossless codec |
15:18.34 | Assid | so its better than ul |
15:18.42 | asteriskmonkey | coppice: all my customers are ulaw :) the g729 is for high volume places that want more than 6 concurrent calls on a dsl |
15:18.51 | coppice | 729 is a *very* lossy codec |
15:18.56 | Assid | its lossy ? |
15:19.13 | asteriskmonkey | 729 is good but you have to keep it under 150ms otherwise it goes to crap |
15:19.15 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
15:19.21 | coppice | 729 is *far* worse than ulaw or alaw, too |
15:19.28 | Assid | weird |
15:19.30 | asteriskmonkey | ulaw and alaw are the best |
15:19.35 | asteriskmonkey | there the least compressed |
15:19.36 | Assid | i do a few calls on 729 |
15:19.37 | Assid | works fine |
15:19.42 | coppice | Assid: where on earth did you get the idea 729 is lossless |
15:19.56 | Assid | some stupid forum i read longggggg ago |
15:20.00 | Assid | guess it stuck with me |
15:20.29 | asteriskmonkey | now i can take out my intel testing g729 ones :) |
15:20.44 | Assid | hehe |
15:20.49 | asteriskmonkey | 729 is good if the distance is short |
15:20.51 | Assid | see any different in translation times |
15:20.54 | Hali_303 | Assid: you need 64kbps for telephone quality, which is lossless audio sampled at 13 bits (compressed into 8 bits) at 8khz |
15:21.14 | asteriskmonkey | 729 however adds a lot of cpu usage to your box so you cant exactly go mad with it |
15:21.22 | Assid | err... |
15:21.27 | Hali_303 | I think the problem w/ speex was not with quality, but with delay! |
15:21.29 | Assid | asteriskmonkey: only if you transcode |
15:21.43 | coppice | Hai_303: not really. you can get *very* similar quality to ulaw at something like 25kbps. you can't get it at 8kbps, though |
15:21.48 | Assid | if caller/callee is 729.. no issues |
15:21.52 | asteriskmonkey | Assid: voip client > pstn = transcode |
15:22.05 | Assid | oh.. you got a fxs interface |
15:22.11 | asteriskmonkey | lol no |
15:22.11 | Assid | err.. fxo even |
15:22.15 | asteriskmonkey | try a ds3 |
15:22.17 | Assid | aah |
15:22.22 | Assid | thatd exaplain it |
15:22.25 | Hali_303 | coppice: yes. very similar, but not the same! |
15:22.39 | Assid | so how many licenses did you buy |
15:22.53 | coppice | Hali_303: well you can do a lot better than ulaw at about 30kbps |
15:23.16 | Assid | how the hell do you set ulaw to eat less bandwith? |
15:23.46 | Hali_303 | Assid: you cannot. ulaw is a standard, which makes audio sampled at 13 bits into 8 bits |
15:23.47 | asteriskmonkey | you dont |
15:23.50 | Hali_303 | you cannot set that |
15:24.06 | asteriskmonkey | you get a bigger internet connection |
15:24.08 | Assid | asteriskmonkey: when you gong live? |
15:24.18 | asteriskmonkey | Assid: i am live |
15:24.34 | Assid | retailing? |
15:24.44 | asteriskmonkey | wholesaling to dealers |
15:24.58 | Assid | site? |
15:25.07 | asteriskmonkey | its uber plain www.massivetel.com |
15:25.23 | Assid | 11:25:15 (1.28 MB/s) - `asterisk-1.2.7.1.tar.gz' saved [10554037/10554037] |
15:25.27 | asteriskmonkey | ill probably rejig it sometime soon but has been ok for past 8months :D |
15:25.27 | Assid | i think thats enough bandwith? |
15:25.50 | asteriskmonkey | thats downstram |
15:25.55 | asteriskmonkey | it upstream that counts |
15:26.11 | Assid | hrmm |
15:26.33 | Assid | well.. its a dedicated server.. |
15:26.37 | asteriskmonkey | so |
15:26.56 | asteriskmonkey | dosnt matter if its dedicated matters what connectivity is plugged into it and if its throttled or not |
15:27.16 | asteriskmonkey | try sending something to another backbone server that will give you an idea |
15:27.40 | asteriskmonkey | then you can just div you upstream by 90k to see how many users you can handle |
15:27.48 | asteriskmonkey | ttyl |
15:28.04 | Assid | hrmm around 800K up |
15:28.17 | Assid | > 1 if i do to my other box |
15:28.22 | Assid | oh well |
15:28.48 | Assid | hrmm.. as i was saying.. i wanna play with RT |
15:31.25 | *** join/#asterisk viperdude (n=viperdud@84-45-168-60.no-dns-yet.enta.net) |
15:31.58 | viperdude | hi anybody got experience of using SER with Asterisk? |
15:34.28 | Assid | okay time to upgrade one of these old boxes |
15:35.13 | Assid | err.. is it worth upgrading from 1.2.4 to current? |
15:45.54 | dpryo | Is it possible to get the number a call was transfered from? (c-number?) |
15:47.18 | *** join/#asterisk Grubs (n=Miranda@c220-239-223-90.eburwd3.vic.optusnet.com.au) |
15:47.36 | Assid | well.. you could set a variable before transferring the call |
15:47.41 | Assid | i think that could do it |
15:48.05 | dpryo | Well, I'm not doing the transfer, the telco is transfering a number in to me. |
15:48.11 | Assid | oh |
15:48.18 | Assid | doesnt the caller id change then |
15:48.32 | dpryo | Nope, I get the correct callerid |
15:48.43 | dpryo | I guess the telco sets it up.. |
15:48.49 | Assid | then i dont think you can |
15:49.05 | dpryo | I've seen some talk about 'c-number' containing it |
15:49.12 | Assid | not sure.. |
15:49.16 | Assid | ask around |
15:49.22 | dpryo | But i'm not sure how to debug it |
15:51.24 | Grubs | Just upgraded from 1.2.5 to 1.2.7.1 and I am seeing a new 2-3 second delay after dialing before asterisk responds (dialing and voicemail prompts). Any ideas on what to look for to eliminate it? |
15:51.51 | Grubs | useCallerID=no so it isnt caller ID detection. |
15:53.08 | Strom_C | this is going to seem like a really silly question, but how do you get asterisk to start at boot-time? |
15:54.45 | Grubs | Strom_C: - http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x430.html |
15:55.36 | Strom_C | thanks :) |
15:55.57 | Flauto | fwd is down |
15:56.09 | Flauto | can not evenopen their website |
15:57.50 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
15:58.09 | Jacke | anyone running asterisk on freebsd here? |
15:59.02 | Nugget | I do. I don't recommend it. |
15:59.39 | Jacke | why not? |
15:59.50 | Jacke | and which version of asterisk are you using. |
16:00.32 | Nugget | the zaptel-bsd driver lags the real zaptel stuff and has always been really flaky for me. |
16:00.48 | Nugget | if you can live without zaptel it's pretty viable. |
16:01.31 | Jacke | well, with my TE410P i'm not in the position to live without it ;) |
16:01.46 | Nugget | it's also hard to get support from the community if you have problems. You'll be forced to listen to a 15 minute diatribe on how ubuntu is going to change the world and how asterisk works just great if you'd just switch to their favorite distro. |
16:02.11 | dpryo | Nugget: It's true! You know it! |
16:02.15 | Jacke | the problem is i don't like their distros ;) |
16:02.21 | Sedorox | Nugget: ahahhahaha |
16:02.24 | Jacke | I don't like their operating system at all :) |
16:02.33 | Jacke | Nugget: Which version of asterisk are you running? |
16:02.36 | Nugget | neither do I, but I tolerate it for asterisk. |
16:02.57 | Sedorox | I like fbsd for networking shit... |
16:03.01 | DoktorGreg | Dogma is the one true religion! |
16:03.01 | Sedorox | not for normal workstation... |
16:03.12 | Jacke | on the FreeBSD i mean. |
16:03.17 | Nugget | of course not. os x is for normal workstations. ;) |
16:03.28 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-210.telkomadsl.co.za) |
16:03.30 | Nugget | freebsd for servers, openbsd for firewalls, and linux for asterisk. |
16:03.35 | Jacke | well, as if an asterisk machine was going to be a workstation ;) |
16:03.42 | Sedorox | Nugget: if I had a mac.. thats what I would run (of if osx86 ran on my laptop...) |
16:03.44 | coppice | * has *real* problems with OS/X :-) |
16:03.57 | Nugget | real problems spelling it? |
16:04.04 | DoktorGreg | osx has 'realtime' problems i hear |
16:04.16 | Sedorox | and look.. here we go on a tangent about operating systems :p |
16:04.22 | coppice | all OSes have realtime problems |
16:04.27 | Nugget | hey! I'm being a grammar troll, not an os troll! |
16:04.33 | Sedorox | lol |
16:04.33 | Nugget | get it right, Sedorox. |
16:04.38 | Sedorox | no no.. I mean in general |
16:04.40 | DoktorGreg | which begs the question |
16:04.41 | Sedorox | not you alone :p |
16:04.46 | DoktorGreg | what is realtime? |
16:04.54 | Sedorox | we're going on about fbsd.. obsd.. linux... osx... etc.. :p |
16:05.03 | DoktorGreg | can someone splain it to me in a nutshell? |
16:05.08 | coppice | realtime is something OSes don't do properly :-) |
16:05.28 | Nugget | in an asterisk context, "realtime" means "database-backed asterisk" |
16:05.39 | Nugget | in an os context, "realtime" means guaranteed response time |
16:06.34 | coppice | and guarantees of response time are nearly impossible to achieve these days |
16:06.35 | Nugget | general purpose operating systems can't do true "realtime" processign, no matter how much ugly crap you bolt onto them, whcih is the point I believe coppice is making. |
16:06.50 | Nugget | and even if it were feasible, you probably wouldn't want to |
16:07.04 | *** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net) |
16:07.11 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
16:07.15 | coppice | the processors can't do real time either, unless you turn off their caches |
16:07.33 | tier_1 | http://pastebin.ca/50702 |
16:07.38 | tier_1 | have fun |
16:07.51 | tier_1 | NANPA config |
16:08.16 | Corydon76-home | Well, if you had enough spare cores, you could guarantee response time |
16:08.39 | Nugget | not with a bus to mitigate. |
16:08.53 | coppice | Corydon76-home: sounds like you don't even begin to understand the problem |
16:09.01 | Corydon76-home | but master/slave CPUs has already been demonstrated to be quite inefficient |
16:09.03 | Nugget | you have to stop servicing interrupts if you want to guarantee a response time |
16:09.27 | DoktorGreg | so if i wanted an application to be 'realtime' I would have to design something from the ground up? |
16:09.45 | Nugget | or you could use an off-the-shelf realtime system. |
16:09.47 | DoktorGreg | or rather get an architecture that is designed to be realtime? |
16:09.49 | Sedorox | clockless cpus!!!!!111elevenone |
16:10.14 | Nugget | heh |
16:10.31 | Sedorox | man... amd64 but clockless.... yummy.... |
16:10.33 | Sedorox | anyway... |
16:10.33 | Sedorox | :p |
16:10.47 | *** join/#asterisk ketil (n=chatzill@217-131-74.5001.adsl.tele2.no) |
16:11.29 | x86 | wow.... 3 year wedding anniversary today for my wife and I |
16:11.38 | x86 | 3... long..... years ;) |
16:11.41 | Sedorox | congrats |
16:11.49 | x86 | thanks :) |
16:12.30 | Corydon76-home | You can have an efficient system or you can have a realtime system |
16:12.47 | Sedorox | anyone want to wite my persaisive (sp) speech on why rfid tagging is bad? :p |
16:13.05 | coppice | you can have a highly efficient real time system, but its far from general purpose |
16:13.27 | Corydon76-home | True enough |
16:13.29 | DoktorGreg | well googling around |
16:13.45 | DoktorGreg | it looks like fly by wire flight controll systems have to be real time |
16:13.54 | Corydon76-home | So, Realtime/Efficient/General Purpose... pick any two... |
16:14.09 | DoktorGreg | and frame grabbers for science experiments have to be real time |
16:14.17 | Nugget | no! you're not allowed to pick. everyone must run my favorite distro of linux otherwise I lose. |
16:14.24 | Sedorox | lol |
16:14.40 | Sedorox | let me gues.s.... your a debian user? :p |
16:15.05 | DoktorGreg | i am contemplating an OSX for my desktop |
16:15.23 | Corydon76-home | I have a few nub's who have picked Debian for their systems and then expect me to be able to advise them how to administrate it |
16:15.26 | DoktorGreg | but i play a game a week out of every three months |
16:15.33 | Nugget | I'm a slut. I use just about everything. |
16:15.39 | Corydon76-home | Homey don't play dat. |
16:15.39 | Sedorox | lol |
16:15.47 | Sedorox | I'm basically gentoo or fbsd now... |
16:16.00 | file | Nugget: SLACKER! |
16:16.05 | Nugget | that's me! |
16:16.10 | DoktorGreg | All will bow to the Hurd! |
16:16.17 | Corydon76-home | Somebody sold them on Debian, then PROMPTLY DISAPPEARED. |
16:16.19 | file | no, all will bow to OS/2 |
16:16.38 | blitzrage | hail OS/2! |
16:17.01 | Nugget | DoktorGreg: OS X runs world of warcraft and civilization. Those are the only two games I need. :) |
16:17.12 | Sedorox | ahahah |
16:17.19 | DoktorGreg | Nugget, last game i played was DnD online |
16:17.22 | Sedorox | and enemy territory... can't forget that one :p |
16:17.36 | Nugget | ah, yeah, I play call of duty when I want to shoot nazis. |
16:17.46 | Sedorox | hehe |
16:17.58 | DoktorGreg | I havent played a nazi shooting game for a while |
16:17.59 | file | blitzrage: you're supposed to be working out |
16:18.21 | DoktorGreg | oh i know i totally burned out on nazi shooting on bf1942 |
16:18.39 | x86 | Nugget: niether WoW or Civ have x86 binaries for OS X |
16:18.44 | Nugget | wow does. |
16:18.51 | x86 | Nugget: you sure? |
16:18.54 | tier_1 | Freeciv |
16:18.55 | Nugget | 100% |
16:18.56 | Sedorox | yes |
16:19.06 | Nugget | wow has been a universal binary for a few months now |
16:19.06 | Sedorox | I know a lot ofp eople with wow on osx86 |
16:19.42 | DoktorGreg | I more or less refuse to play games like wow since i lost a couple of friends to EQ |
16:19.51 | Nugget | yeah, that's a tangible risk. |
16:20.28 | Nugget | I mitigated it by getting my partner hooked too, so we're both equally hopeless and it works out just fine. |
16:20.34 | Sedorox | my roommates tried to get me to play it when it was in beta |
16:20.35 | Sedorox | :p |
16:20.37 | Sedorox | and free... |
16:20.47 | Sedorox | Nugget: lol |
16:21.00 | Nugget | she's playing right now, in fact. I can hear the music. |
16:21.02 | Sedorox | I wouldn't play tho.. I know I would get addicted to it.. so I didn't bother... |
16:21.27 | DoktorGreg | I get bored with a particular game after two weeks max |
16:21.45 | DoktorGreg | so problem with muds for me |
16:21.46 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net) |
16:21.53 | Sedorox | we play ET here at school a lot.. but its weird.. we'll go on kicks where for like 3 weeks straight we play nothing but it.. solid.. whenver we can... |
16:21.56 | DoktorGreg | is they take about a month to really get into them |
16:22.01 | Sedorox | then we'll go three weeks where no one plays |
16:23.02 | Nugget | I never liked W:ET. the controls and gameplay are too arcade style. I like DoD a lot better where the combat is less "point and spray" and gameplay is more deliberate. |
16:23.25 | DoktorGreg | apex of shooters was bf1942 |
16:23.42 | DoktorGreg | IMO |
16:24.26 | DoktorGreg | and single combat was irrelevant |
16:24.41 | DoktorGreg | you had to work as team or het pwned |
16:26.09 | Nugget | I'm in an endgame guild in WoW which is a really neat experience. 20 and 40 player teams who all have to work in lockstep to even have a chance of succeeding at something |
16:26.33 | Nugget | it's a real rush when everyone's focused |
16:33.19 | *** join/#asterisk stoffell_h (n=stoffell@81.164.209.43) |
16:44.27 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
16:47.09 | *** join/#asterisk Kernel_Core (n=I@193.251.135.118) |
16:51.40 | Flauto | is anyone here using icall.com |
16:51.48 | Flauto | i mean using it with asterisk |
16:53.12 | *** join/#asterisk jovan (n=jovan@host151-99.pool8711.interbusiness.it) |
16:53.21 | jovan | hi |
16:53.56 | Flauto | hi |
17:04.43 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
17:08.27 | *** join/#asterisk ramo (n=ramo@59.92.200.207) |
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17:15.09 | *** join/#asterisk SkramX (n=mark@admins.sentiensystems.net) |
17:15.31 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
17:15.36 | SkramX | what would i need to do, to be able to handle a lot of calls and just do passthrough/call accounting? SER? no codec changes needed. |
17:24.31 | *** join/#asterisk Skarmeth (n=Skarmeth@201008202192.user.veloxzone.com.br) |
17:28.42 | *** join/#asterisk IceManRISK (n=kart@201.10.94.122) |
17:31.22 | OliverX | == Everyone is busy/congested at this time (1:0/0/1) // Can anyone help me? |
17:32.16 | Assid | well |
17:32.34 | Assid | if you just do SER, that would be exactly what you need |
17:35.40 | SkramX | Assid: okay. |
17:35.49 | SkramX | what kind of server resoirces are we talking? |
17:37.45 | Assid | well.. not sure.. can handle couple hundred.. on a xeon |
17:37.47 | Assid | thats for sure |
17:37.53 | Assid | 150 odd + |
17:38.09 | Assid | atleast thats what i read |
17:38.12 | Assid | i could be mistaken |
17:38.27 | SkramX | hrmm |
17:40.02 | *** join/#asterisk Samoied (n=Samoied@201-3-227-215.fnsce7002.dsl.brasiltelecom.net.br) |
17:41.22 | SkramX | a xeon for only 150+? |
17:41.33 | Assid | umm.. whats the makefile config for K8 ? |
17:41.36 | Assid | athlon64? |
17:41.47 | Assid | not sure SkramX |
17:42.15 | *** join/#asterisk mmmmmToop (n=chatzill@dsl-165-166-229.telkomadsl.co.za) |
17:42.27 | Assid | PROC=k8 ? |
17:42.38 | Samoied | hello all! |
17:43.06 | Samoied | anyone have tested miax with bluetooth? |
17:43.44 | Samoied | I have tried with a nokia 6600 |
17:44.12 | Samoied | but miax try to use the phone as modem, not a handset |
17:46.40 | Assid | guess i will just set it to athlon |
17:47.42 | IceManRISK | Anyone here uses JIAX / |
17:49.46 | SkramX | so.. |
17:49.48 | IceManRISK | Anyone here uses JIAX ? |
17:50.51 | *** join/#asterisk Alric (n=nbowyer@ppp-db.1stel.com) |
17:51.11 | *** join/#asterisk stakk (i=sted@85.10.196.41) |
17:51.56 | Assid | okay i gotta run |
17:51.58 | Assid | laterz |
17:53.40 | *** part/#asterisk BearPerson (i=karsten@freenode/staff/sourcemage.wizard.BearPerson) |
17:54.31 | websae | anyone from Canada here? |
17:55.09 | *** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
17:56.12 | Druken | yep |
17:56.48 | Druken | websae: why ya lookin for canadians? |
17:57.17 | websae | just to say, hello :) |
17:57.31 | Druken | hello..... |
17:57.33 | websae | I am in Wisconsin -- pretty close to Canada |
17:57.42 | websae | do you have wholesale routes there? |
17:57.55 | Druken | how's the cheese? hehe |
17:58.05 | websae | moldy :) |
17:58.47 | Druken | websae: msg me priv with what your lookin for |
18:09.34 | NewSole | hmm |
18:21.53 | *** join/#asterisk FreddyFeuer (n=email@p54AE01A6.dip0.t-ipconnect.de) |
18:33.17 | dpryo | Is it possible to "traceroute" PRIs? .. like see how many hops there is to the destination? |
18:33.46 | macTijn | mostly that's just 1 |
18:33.52 | macTijn | what would you count as a hop ? |
18:33.58 | dpryo | A pbx |
18:34.02 | dpryo | or something |
18:34.34 | dpryo | My calls are going through at least 3 different pbxes |
18:35.00 | dpryo | cisco -> asterisk -> avaya -> .. |
18:35.17 | tainted- | dpryo that is pretty cool |
18:41.45 | xbmodder_lappy | how is the stability of chan_bluetooth |
18:43.53 | dlynes | websae: I'm from Canada, too |
18:44.00 | FreddyFeuer | kann mir jemand bei meiner asterisk konfiguration helfen? |
18:44.11 | Hmmhesays | put your hand up on your hip, i dip, you dip, we dip |
18:45.15 | dlynes | websae: Not able to talk right now, though...I'll be back in about an hour if you want to msg me about what you want |
18:45.48 | VoIPMasta | FreddyFeuer: Ich zweifele, daß Sie jemand finden, das Deutsches innen hier spricht |
18:46.55 | FreddyFeuer | hmm. |
18:53.32 | *** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net) |
18:53.45 | firestrm | hello |
19:01.36 | firestrm | anyone here good at wierd IAX problems? |
19:01.46 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
19:03.15 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
19:10.18 | *** join/#asterisk doolph (n=doolph@HACKERMACHINE.cpe.cableonda.net) |
19:11.25 | doolph | anyone know to fix this problem, when I try to enter any number in a ivr menu the system doesn't accept it |
19:11.59 | *** part/#asterisk SkramX (n=mark@admins.sentiensystems.net) |
19:18.25 | *** part/#asterisk Skarmeth (n=Skarmeth@201008202192.user.veloxzone.com.br) |
19:19.43 | ManxPower | doolph, many possible causes. you may need to increase or decrease the rxgain or txgain on your ZAP ports. |
19:20.06 | doolph | I am using sip trunk and sip extension |
19:20.09 | ManxPower | You may also be trying to do inband DTMF on a call that uses a compressed codec and EVERYONE knows that won't work. |
19:20.44 | [Airwolf] | I both have users with IAX2 and SIP accounts. |
19:21.21 | [Airwolf] | But I want to create a macro that dials the account (iax2 or sip) on which the user has logged in. |
19:21.49 | ManxPower | [Airwolf], I know of no easy to do that. |
19:21.50 | [Airwolf] | This morning I got the tip to use the ChanIsAvail function. |
19:21.55 | ManxPower | Why not just dial them both? |
19:22.11 | Hmmhesays | set a variable in sip.conf and use that |
19:22.24 | [Airwolf] | ManxPower, that is the solution, the problem is it isn't really good. |
19:22.25 | *** join/#asterisk espino (i=espino@srv4.v-expressa.com.br) |
19:22.38 | [Airwolf] | exten => s,1,Dial(${ARG2}&${ARG3},${ARG4}) |
19:22.39 | ManxPower | Hmmhesays, not going to do much good for OUTGOING calls. |
19:22.47 | [Airwolf] | I can solve it something like that |
19:22.50 | Hmmhesays | Manx, why not? |
19:23.02 | ManxPower | Hmmhesays, because that variable won't be set. |
19:23.08 | [Airwolf] | But it creates an error on the channel where the user isn't logged in. |
19:23.12 | Hmmhesays | nm |
19:23.36 | ManxPower | Say you have a Setvar for a SIP account. A call comes in via Zap and gets sent to that SIP account. The variable will not be set. |
19:23.39 | Hmmhesays | seperate out your extensions numbers, make 6XX sip and 7XX iax or osmething like that |
19:24.04 | ManxPower | Hmmhesays, the SAME user is allowed to register BOTH IAX and SIP. |
19:24.28 | [Airwolf] | That is a possibility, but that doesn't fit in my local dialplan. |
19:24.40 | *** join/#asterisk wikityler (n=yo_tyler@d66-183-163-151.bchsia.telus.net) |
19:25.02 | [Airwolf] | I think it just has to be the quick & dirty solution by calling the channels at the same time and just leave the error to be. |
19:25.08 | wikityler | yay, im finaly in. |
19:25.30 | ManxPower | [Airwolf], other providers either 1) make a user PICK if they want IAX or SIP (and only 1 can be selected at a time) or 2) send all calls to BOTH SIP and IAX registration for that user |
19:26.11 | wikityler | quich and dumb voip question: if im using an ata, can it be on the switch that the server is on, or does it have to be connected directly to a nic on the sever? |
19:26.36 | VoIPMasta | wikityler: you can connect it to the switch |
19:26.46 | wikityler | ok, thankyou. |
19:26.54 | VoIPMasta | wikityler: you're welcome |
19:27.18 | [Airwolf] | ManxPower, I get it. |
19:27.52 | [Airwolf] | Maybe there should be a function that just returns a true of false on the question if a channel is avalible. |
19:28.05 | [Airwolf] | Then the problem could be solved. :) |
19:30.42 | mmmmmToop | hi...anyone here used Sirrix BRI cards before? |
19:36.09 | *** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org) |
19:40.05 | mindwarp | Hello, I have a quick Asterisk question: I read a lot in tutorials about using SIP (which most IP phones use), but also about its problems with NAT. I was wondering if Asterisk allows me to use one protocol (SIP) for connecting my phone to the network and a different protocol (IAX) to carry that line over the net for use with an ITSP or such. |
19:40.53 | mindwarp | or do I have to commit to one protocol? Any clarifications appreciated, thanks! |
19:41.52 | xbmodder_lappy | you can use both IAX and SIP |
19:42.56 | mindwarp | So I can use a SIP phone but actually use IAX to terminate calls etc. That's great, thanks. |
19:47.50 | [Airwolf] | mindwarp, no, then you have to buy a phone that supports IAX2 |
19:48.31 | xbmodder_lappy | [Airwolf], I think he means having his asterisk box use both IAX2 and SIP |
19:49.47 | mindwarp | ah, no, [Airwolf] got it |
19:49.56 | mindwarp | thanks, that's what I was wondering |
19:50.25 | mindwarp | whether there is any way to have Asterisk bridge the two protocols |
19:50.36 | Maxxed | damnit |
19:50.47 | Maxxed | im getting this callerid@ipaddr of the pbx now |
19:51.01 | Maxxed | that i upgraded my 7960's to the lastest sip firmware |
19:51.16 | Maxxed | anybody know how to turn that crap off |
19:51.50 | [hC] | cisco sure is getting a kick out of fucking up sip loads on their phones these days |
19:52.00 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
19:52.07 | Maxxed | lol |
19:54.11 | techie | Maxxed: I have the same problem |
19:55.04 | Maxxed | figure out a resolution yet? |
19:55.19 | techie | no, no docs on it either |
19:55.24 | Maxxed | gayness |
19:56.45 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
19:58.42 | xbmodder_lappy | any VoIP providers here? |
20:04.18 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
20:05.05 | firestrm | anyone here use *@home? |
20:05.16 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
20:05.27 | firestrm | xbmodder_lappy, talk to websae.. |
20:05.37 | xbmodder_lappy | Yeah, I already havve |
20:05.38 | xbmodder_lappy | have |
20:05.58 | firestrm | he has been really good to me.. |
20:07.25 | firestrm | ok no asterisk@home people huh.. any one know of any paid support?.. im about to loose my sanity on this thing.. and even though i cant afford it, im still willing to pay to get it fixed.. |
20:08.47 | NewSole | whats wrong |
20:10.06 | *** join/#asterisk hohum (i=corbe@snoop.burghcom.com) |
20:10.42 | NewSole | o well.... I am going for some food |
20:11.40 | hohum | how can I set up my asterisk box so it only does RTP pass thru? |
20:11.44 | firestrm | NewSole, i keep getting "the number you called is temporarly disconencted" when dialing outbound |
20:12.05 | firestrm | even though the connection is going through.. |
20:14.04 | firestrm | and the #)$(#*$)@#($* thing doesnt give any usefull debug information.. i have no idea why it is winding up there.,. |
20:17.07 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
20:17.16 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
20:17.58 | RoyK | ~seen wasim |
20:18.08 | jbot | wasim <n=wasim@pdpc/supporter/active/wasim> was last seen on IRC in channel #asterisk, 4d 6h 16m 8s ago, saying: 'guigouz: an fxo/sip ata'. |
20:18.51 | bonfire1 | Is there any way to use an old voice-modem as an FXO card? |
20:20.18 | Igbothom_III | not if you want good quality |
20:20.28 | *** join/#asterisk synaptic (i=synaptic@68.62.176.196) |
20:21.01 | bonfire1 | well and if I just want to play with asterisk ? |
20:21.19 | bonfire1 | is it just like - installing the modem and configuring something in the zaptel.conf ? |
20:21.23 | hohum | how can I set up my asterisk box so it only does RTP pass thru? |
20:21.56 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) |
20:22.06 | Igbothom_III | bonfire1, not that easy - you need drivers for your modem. best way would be to look on ebay for cheap 1-port FXO clone cards - they cost around US$10 - 15 |
20:22.22 | bonfire1 | yeah but I live in israel |
20:22.25 | bonfire1 | shipping is a bitch |
20:22.37 | bonfire1 | they sell the digium ones for 49$ (!!) |
20:22.41 | Igbothom_III | then ask locally :) |
20:22.54 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
20:22.57 | Igbothom_III | then spend $49 and be done with it |
20:23.01 | bonfire1 | there isn't much of a market for voip supply. |
20:23.13 | [Airwolf] | bonfire1, there are enough cheap pci modems availible that are just the X100P, but you need to chage the driver a little bit. |
20:23.54 | Igbothom_III | after all, the clone cards are basically a modem themselves |
20:24.05 | synaptic | you just need to change the drivers to the chipset of the winmodem to make it a x100p heh |
20:24.50 | bonfire1 | so I basically plug a modem in, install winmodem's driver, and I'm good to go? |
20:25.41 | Igbothom_III | if you get the right modem, yes |
20:26.38 | synaptic | i rather just buy a dev kit. tdm11 |
20:26.57 | Igbothom_III | as I suggested, get cards you know will work |
20:27.03 | Igbothom_III | saves the buggerising around |
20:27.17 | bonfire1 | maybe I'll do that... in my next trip to the US... |
20:27.41 | bonfire1 | or with a company that supplies a US address and then ships it to Israel |
20:31.15 | Igbothom_III | like the guys who supply you with weapons :) |
20:31.32 | Igbothom_III | they generally ship straight from the 'States :) |
20:32.37 | bonfire1 | yeah US army stuff is great |
20:32.47 | [Airwolf] | hmm, 2 nukes, 3 stingers, 1 X100P, 200 granades |
20:32.49 | [Airwolf] | :P |
20:33.56 | hohum | how can I set up my asterisk box so it only does RTP pass thru? |
20:34.23 | `Kevin | asterisk should be able to become a slave via pri to a shortel ? we have this scenario and the d channel will not come up |
20:35.27 | *** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
20:35.32 | *** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue) |
20:35.50 | brookshire | hohum: by default |
20:36.27 | brookshire | noreinvte=yes (would make all traffic go through asterisk and not between endpoints) |
20:37.31 | brookshire | maybe it's canreinvite=no |
20:37.33 | distortion | canreinvite=no |
20:37.38 | brookshire | i forget |
20:38.01 | distortion | canreinvite=no forces rtp to pass through asterisk |
20:38.34 | brookshire | rtp pass thru is kind of a bad question though |
20:38.51 | brookshire | because pass through simply means asterisk isn't doing anything with transcoding |
20:39.30 | distortion | rtp passthrough is needed to maintain billing acuracy |
20:39.51 | distortion | i've had horrible billing issues by turning it off when connecting to session controllers |
20:40.04 | distortion | err by using canreinvite=yes rather |
20:40.17 | file | in Asterisk? |
20:40.24 | distortion | yep. |
20:40.36 | distortion | particularly bad when connecting to nextones |
20:40.45 | file | that makes no sense |
20:40.50 | brookshire | it doesn't make sense |
20:40.58 | brookshire | because the control should still be up and monitoring |
20:41.04 | file | reinvites might not work, so your audio might not flow... but signalling still goes through Asterisk |
20:41.17 | hohum | well this asterisk box is being set up as a signaling gateway |
20:41.24 | Maxxed | ok im out for now fellas |
20:41.25 | ManxPower | Yeah. reinvites only does AUDIO. |
20:41.25 | hohum | need to take SIP and spit out H323 to a vendor |
20:41.27 | hohum | so |
20:41.30 | distortion | for some reason there are always hung calls on the provider's end which cause billing issues |
20:41.31 | Maxxed | il harass yall later ;) |
20:41.33 | hohum | it doesn't need to touch the RTP stream |
20:41.49 | file | then device might not like reinvites for whatever reason |
20:41.52 | brookshire | h323 = ewh! |
20:41.53 | brookshire | :) |
20:41.56 | file | er the device |
20:42.11 | distortion | nextones are very good with re-invites |
20:42.26 | file | it still makes no sense that it would affect billing |
20:42.54 | file | oh well |
20:43.19 | distortion | i'd love to find out too- then i could stop spending thousands on my bandwidth bill since i proxy all the damn rtp |
20:43.26 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
20:47.01 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
20:51.07 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
20:55.15 | ghenry | What do you guys think of this: http://cgi.ebay.co.uk/Asterisk-PBX-Hardware-and-Full-Support_W0QQitemZ9716930873QQcategoryZ34165QQrdZ1QQcmdZViewItem ? |
20:55.20 | ghenry | Recommended? |
20:55.25 | ghenry | Or have go myself? |
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20:59.43 | *** part/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
21:00.50 | lesouvage | . |
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21:05.34 | *** part/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-251.claranet.co.uk) |
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21:25.28 | NoName32 | hi all i am still pretty new to asterisk got a problem i cant figure out .. working on the one touch record using the featuremap with automon => *1 but it doesnt seem to be reconizing the *1 if i change it to ** it works any ideas/ sugestions i am using asterisk 1.2.6 |
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22:17.39 | Druken | what is it about icecream that no matter what, it makes ya feel better? |
22:20.16 | SpaceBass | anyone using freedigts.com? |
22:20.29 | SpaceBass | freedigits that is |
22:22.12 | *** join/#asterisk saftsack (n=saftsack@p54A7E109.dip.t-dialin.net) |
22:23.34 | Druken | looks like a virtual peering with upgradeable dids |
22:23.58 | SpaceBass | know what it is, but just having problems routing my incoming numbers |
22:24.46 | Druken | oh, wuts the problem? |
22:24.54 | SpaceBass | they come in as guest sip calls |
22:25.03 | SpaceBass | with out a DID |
22:25.24 | Druken | really... thats strange.... |
22:25.35 | SpaceBass | so I can have all guest calls go to my incoming context but I'd really prefer to treat each one differently |
22:26.11 | Druken | are the seperate accounts per did? or multipul dids per account? |
22:26.24 | SpaceBass | seperate accounts |
22:26.40 | *** part/#asterisk opus_ (n=opus@dahphish.org) |
22:26.56 | Druken | alrighty |
22:28.10 | Druken | are you registering? |
22:28.35 | SpaceBass | no, nerdvittles.com suggested that you could, and I tried their settings, but it didnt work |
22:28.49 | SpaceBass | freedigits's tech support said to just forward the sip calls |
22:29.01 | SpaceBass | so thats why they are coming in as guest calls |
22:29.53 | Druken | forward the sip calls? how so ? |
22:29.58 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
22:30.27 | SpaceBass | like I put in the IP of my * box and the port (5060) |
22:34.42 | IceManRISK | Anyone here uses JIAX ? |
22:35.27 | X-Rob | SpaceBass, if you care about calls from a specific guest, they should have a type=peer, which means they'r eno longer a guest. |
22:35.27 | Druken | SpaceBass: you got dialing with them working? |
22:35.36 | X-Rob | a guest means, by definiton, something you don't know about and dont' care about. |
22:36.52 | *** join/#asterisk saftsack (n=saftsack@p54A7E109.dip.t-dialin.net) |
22:38.04 | Druken | i just grabbed a number from them, not sure if it's working, i'd need a test call... if it works, then i'll let ya know how to fix it :) |
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22:45.24 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
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22:52.04 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
22:52.27 | chiardon | Hi |
22:53.41 | chiardon | I have an E1 connected to an * box, yet my E1 provider seems to be signalling me with PCM-31. |
22:53.55 | Qwell | chiardon: Have them change it.. |
22:53.57 | chiardon | Strangely enough the link sometimes works |
22:54.00 | Nugget | My cat's breath smells like cat food. |
22:54.21 | chiardon | Qwell --> Yes, but in the meantime (they say that takes about 1 week) |
22:54.29 | chiardon | is there anything that can be done? |
22:54.36 | Qwell | chiardon: You could pay somebody to add support to * |
22:54.41 | Qwell | (which would take more than a week) |
22:54.52 | Qwell | Or, you could have tested before you went into production. :) |
22:55.31 | chiardon | There is a BRI patch around ... could that work for my pri? |
22:55.41 | Qwell | no |
22:56.27 | DoktorGreg | BRIStuff does not work on PRI |
22:56.31 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
22:56.36 | DoktorGreg | in no way shape or form |
22:57.13 | chiardon | Oh well thanks. |
22:58.14 | chiardon | Btw I am looking for a CHEAP SIP (physical) phone |
22:58.32 | DoktorGreg | get an ATA |
22:58.36 | Qwell | CHEAP as in crap, or cheap as in inexpensive? |
22:58.43 | DoktorGreg | lol |
22:59.00 | Qwell | grandstream for the former, linksys spa941 for the latter |
22:59.07 | chiardon | how crappy can 'crap' be? |
22:59.17 | Qwell | two words... |
22:59.20 | Qwell | barbie tones |
22:59.55 | chiardon | and in the inexpensive range... we are talking about how much? |
23:00.03 | Qwell | $100? |
23:00.07 | Qwell | $150 tops |
23:00.13 | chiardon | for one? |
23:00.18 | Qwell | yes |
23:00.23 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
23:00.24 | Qwell | telephony aint cheap |
23:00.35 | chiardon | Uuuh I think I joined the wrong channel ... you guys are expensive here :P |
23:00.47 | DoktorGreg | ATA is also option |
23:00.47 | Qwell | chiardon: Try #nortel. |
23:00.58 | Qwell | You'll wet yourself when you hear what a nortel phone costs |
23:01.14 | chiardon | DoktorGreg --> How much can a decent ATA cost? |
23:01.21 | Qwell | decent? ~$90 |
23:01.25 | DoktorGreg | I am contemplating selling all my nortel equiptment and replaceing it with the inexpensive voip stuff |
23:01.31 | SplasPood | Qwell: Is there even a #nortel? :) |
23:01.35 | Qwell | SplasPood: maybe :p |
23:01.42 | Qwell | probably on something silly, like dalnet |
23:02.01 | chiardon | DoktorGreg, --> What nortel thingies are you selling? |
23:02.04 | SplasPood | Heh, that just sounds so funny being said on freeload |
23:02.12 | DoktorGreg | playing with idea |
23:02.17 | DoktorGreg | I have about 40 phones |
23:02.29 | DoktorGreg | plus a MICS system with all the bells and whisles |
23:02.34 | chiardon | DoktorGreg, --> And I hope you're selling them at a barbie-toned price :P |
23:02.46 | Qwell | DoktorGreg: Only $25k for the base, and $300 for the phones? used? |
23:02.52 | Qwell | if so, good deal |
23:03.01 | DoktorGreg | Thats what i was thinking |
23:03.22 | chiardon | as I said before when you guys talk about money I get scared |
23:03.29 | chiardon | so I think I'll just run home to mommy =) |
23:03.33 | chiardon | Have a nice day |
23:03.40 | chiardon | Good bye =) |
23:03.46 | SplasPood | hahahaa |
23:03.49 | SplasPood | that was hillarious. |
23:03.52 | DoktorGreg | what about softphone? |
23:03.56 | DoktorGreg | oh he left |
23:04.04 | SplasPood | it was hilarious too |
23:04.19 | DoktorGreg | well a nice analog desk phone still cost ~80-90 |
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23:06.29 | project_2501 | is it possible to chang the ${EXTEN} channel variable? |
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23:31.19 | Ariel_ | hello eveyone |
23:31.28 | Ariel_ | eveyone/everyone |
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23:55.27 | *** join/#asterisk camonz (n=camonz@200.8.70.67) |
23:55.33 | camonz | hi |
23:55.55 | camonz | i was wondering if i could get some help registering * with a sip proxy |
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23:57.21 | camonz | i'm doing register => name@201.211.65.2/1006 |
23:57.27 | camonz | in my sip.conf |
23:57.56 | camonz | i sucesfully register but i'm not getting any calls from that server |
23:58.30 | tainted- | maybe no one is calling |
23:58.37 | tainted- | try sip debug |
23:58.46 | tainted- | which sip proxy |
23:58.46 | camonz | they where calling |
23:58.56 | tainted- | are u behind a nat |
23:58.59 | camonz | it's a setup between some friends and i, we're testing it now |
23:59.03 | tainted- | what does 'sip show peers' say |
23:59.09 | camonz | nope.., i'm in the dmz of my lan |
23:59.17 | *** part/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
23:59.27 | tainted- | ok |
23:59.28 | camonz | plus i've got correctly configured externip and localnet |
23:59.39 | tainted- | what does sip debug say |
23:59.52 | tainted- | when a call is placed through the sip proxy |