| 00:00.12 | mattzerah | i move the phone to our internal asterisk box, phone has no issue |
| 00:00.16 | [TK]D-Fender | mattzerah: You'll have to show us something useful. |
| 00:00.22 | Nest0r | [TK]D-Fender: eact? |
| 00:00.24 | mattzerah | what would you like :) |
| 00:00.37 | [TK]D-Fender | Nest0r: "exact" |
| 00:00.49 | mattzerah | its hard because when i do a traffic sniff, the dial traffic doesn't seem to get to asterisk |
| 00:01.05 | mattzerah | i automatically blamed the network, but i have it running on a network internally that I know works |
| 00:01.06 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 00:01.20 | mattzerah | obviously noone else has this issue (or reported it) :/ |
| 00:01.39 | mattzerah | i might roll back to 1.4.21 and see if the issue resolves |
| 00:02.25 | Nest0r | [TK]D-Fender: thanks :) It is an IVR. When a user calls and enter a number say 3 is defined as 33 |
| 00:02.25 | Nest0r | <PROTECTED> |
| 00:03.59 | [TK]D-Fender | Nest0r: I do not understand " 3 is defined as 33" |
| 00:04.21 | [TK]D-Fender | Nest0r: are you trying to say that they press it once and * receives it twice? |
| 00:04.49 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 00:04.55 | [TK]D-Fender | mattzerah: I don't see how * can magically ward off the packets.... |
| 00:05.18 | mattzerah | neither can i - but i have isolated everything else! |
| 00:05.34 | Nest0r | [TK]D-Fender: yes |
| 00:05.39 | mattzerah | new network, new phones, i can verify my test works on one we have on our network, test fails everytime on that one box |
| 00:06.38 | mattzerah | only difference between the setups (only major difference i should say) is the * version - as far as i can tell - i'm going to roll back versions patch by patch, i'll report if/when i get the problem fixed, perhaps that is the most useful thing i can do |
| 00:07.10 | Nest0r | only happens with 4 ports not the case with TE121PF |
| 00:07.12 | mvanbaak | drmessano: wooohooooooo. You have voice on google today, while all the rest here does have voice for years with SIP/IAX/SKINNY/H323/MGCP ;) |
| 00:07.19 | [TK]D-Fender | Nest0r: try "relaxdtmf=no", and then after testing (with restarting * in between), try "yes" afterwards if that didn't work |
| 00:07.40 | drmessano | mvanbaak: Google PHONE service |
| 00:07.43 | mvanbaak | <--- doesn't believe in the 'google is the best' stuff |
| 00:07.52 | drmessano | Not Google over GTALK |
| 00:07.56 | drmessano | Err |
| 00:07.58 | drmessano | Voice |
| 00:08.03 | mvanbaak | I'm one of the few that thinks google is evil |
| 00:08.15 | drmessano | You should also read more |
| 00:08.54 | mvanbaak | drmessano: how about the next step. every call will start with a spoken advertisement based on your previous gtalk voice calls |
| 00:09.32 | drmessano | Just like Gmail? |
| 00:09.34 | drmessano | Oh, wait |
| 00:09.36 | mvanbaak | yup |
| 00:09.52 | Nest0r | [TK]D-Fender: No, I tried this and not work for me: ( Any other idea? |
| 00:09.55 | jblack | who's less evil than google? |
| 00:10.17 | jaytee | yahoo |
| 00:10.37 | docidu | gmail has ads? |
| 00:10.42 | docidu | never seen um |
| 00:10.43 | jblack | do you seriously think that, jaytee? |
| 00:10.54 | Nest0r | [TK]D-Fender: I also have a Quintum and this presents no problem, only the card with 4 ports |
| 00:11.06 | jaytee | jblack, I don't involve myself with anything remotely resembling serious |
| 00:11.08 | mvanbaak | jaytee: anyone that has a contract where they state they will never use your stuff for their own usage and where they guarantee your stuff is private and never shown to anyone else |
| 00:11.49 | drmessano | Thats a fairytale |
| 00:11.58 | jaytee | I do think Google is a kickass search engine and I certainly wish more of the "noobs" that come in channel would learn to use it once in awhile. |
| 00:12.03 | mvanbaak | jaytee: in fact, every company that outlines what they do with your data is better then google |
| 00:12.20 | drmessano | Privacy policies can and do change when every company gets bigger |
| 00:12.32 | drmessano | They all exploit their users |
| 00:12.43 | mvanbaak | drmessano: maybe in .us, but here in .eu we actually have laws and contracts that are very clear about it |
| 00:12.58 | jaytee | as for them being evil? I'm not sure. Do they torture detainees? do they push for the company they used to be CEO of to get no-bid contracts? |
| 00:13.14 | docidu | any 1 sided contract makes a loser out of the side that abides by it |
| 00:13.47 | mvanbaak | I always find it funny to have this conversation after 1 AM |
| 00:13.59 | drmessano | Well, since the EU did such a good job with making MS take Media player out of Windows (N, HA!), and did such a good job breaking up MS, oh, and pretty much every effort where they pussied out and slapped MS on the wrist, I trust the EU |
| 00:14.14 | jaytee | jblack, besides, privacy is dead. The NSA knows everything in every email and phone call you've ever made and they even know which partition and folder on which hard drive you store your pr0n |
| 00:14.38 | jblack | Maybe they know that about you. :) |
| 00:15.00 | mvanbaak | drmessano: read up bitch. here in .nl you cannot get windows with media player embedded without a signed contract of the buyer |
| 00:15.12 | drmessano | The EU: "Protecting our citizens! NO FEAR! As long as thats ok with you..." |
| 00:15.30 | jaytee | did he just call you bitch? oooooooh, there's gonna be a cat fight, i can tell |
| 00:15.45 | docidu | actually would have probally been worse if they banned M$ from europe, then every kid with a net connection would run all their software out of spite |
| 00:15.50 | mvanbaak | drmessano: same as buying a new computer. if you dont 'need' windows you get 250 euro discount |
| 00:16.40 | jaytee | I loved the picture on Digg the other day of the Iranian nuclear plant control room computer with a license expired dialog box on the screen. |
| 00:16.45 | drmessano | jaytee: I LOL'ed |
| 00:17.14 | mvanbaak | drmessano: be brave and respond to my comments ! |
| 00:17.22 | drmessano | Since I can get a PC without Windows easily, and get it discounted from the price, I dont see that as being something the EU has that no one else has.. |
| 00:17.26 | jaytee | now he's callin you a chicken! |
| 00:17.37 | mvanbaak | I am |
| 00:17.43 | jblack | I wonder if obama's DOJ has the cajones to chase M$ for perjury, if the EU gets M$ to split IE off. |
| 00:17.58 | jaytee | sticks and stones ...... YO MOMMA! |
| 00:17.59 | apeiron | We'll see, eh? |
| 00:18.16 | apeiron | That would be quite lulzy indeed. |
| 00:19.15 | mvanbaak | the big difference is, is that us is ruled by the companies with money, while most countries in .eu actually care for the ppl living in their country |
| 00:19.29 | drmessano | Second, if you READ UP, the EU has backed down on every fucking chest puffing they have done.. Is IE still in Windows? Did M$ get split up? Is the M$ source code open? No, M$ has given the EU nothing any other company wouldnt release and nothing they havent had a contingency plan in place for since Windows 1.0 hit the shelves |
| 00:19.33 | jaytee | oh that's a stretch!!! |
| 00:19.38 | drmessano | So victory? Uh, no |
| 00:20.18 | jaytee | the big difference is, is that us is ruled by the companies with money, while most countries in .eu actually care for the ppl living in their country <=== Ya mean like Fiat? or Peugot? |
| 00:20.28 | jaytee | tell it to the French |
| 00:20.39 | mvanbaak | drmessano: here in .nl you get a machine without mediaplayer and it has firefox installed by default. it's by law |
| 00:20.44 | jblack | didn't the french just do 3 strikes infringement? |
| 00:21.19 | jblack | I wonder what they're gonna do five years from now when they realize none of their college students are allowed to touch the intarweb. |
| 00:21.22 | drmessano | mvanbaak: They have laws giving you value added software? Shit, Dell does that for free here |
| 00:21.25 | drmessano | So does HP |
| 00:21.30 | drmessano | and Toshiba |
| 00:21.34 | drmessano | and .... |
| 00:21.37 | mvanbaak | jaytee: the french just moved police and department of justace to openoffice |
| 00:22.00 | jaytee | and surrendered to Lichtenstein on the same day no doubt |
| 00:22.22 | mvanbaak | drmessano: no, you said .eu did nothing about it. we actually have laws to force it ! |
| 00:22.33 | drmessano | jaytee: Using the German translation of OpenOffice, no doubt |
| 00:22.54 | jaytee | it's why the streets of Paris are lined with trees, the germans like to march in the shade |
| 00:23.02 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 00:23.13 | drmessano | mvanbaak: Forcing value added software? thats toolish.. Sounds like LACK OF CHOICE to me |
| 00:23.48 | jaytee | I think they should force the .eu to bundle WinAMP on every copy of Windows |
| 00:24.10 | mvanbaak | know what? forget about it. you guys are so freaking narrowminded |
| 00:25.08 | mvanbaak | "oh no! they are not accepting our us-based uber power company to force the user to use certain software! you are evil !" |
| 00:25.16 | drmessano | I want someone to force me, by law, to put Firefox on a PC.. thats not Orwellian |
| 00:25.18 | jaytee | all I know is if IBM buys SUN then we're all doomed |
| 00:25.23 | drmessano | Thats worse that M$ |
| 00:25.41 | drmessano | than* |
| 00:25.47 | coppice | jaytee: no, it will only doom IBM |
| 00:26.19 | coppice | mvanbaak: quite a few of these things happen, but then a manager changes and they go back to MS |
| 00:26.45 | mvanbaak | "if a pc ships with MS IE, Mozilla Firefox and Opera should be installed as well, and the default browser should be not specified" |
| 00:27.15 | docidu | OS/2 JavaVM edition? |
| 00:27.28 | drmessano | "Narrowminded" is laughable.. Forcing M$ to remove bits and pieces of their OS and include other software has nothing to do with choice.. thats imposing someones value on others by regulating what someone can and can not include in a product they sell.. thats LESS FREEDOM, and you sheep have no clue |
| 00:27.28 | mvanbaak | that's a word-by-word translation of the dutch law |
| 00:27.52 | drmessano | Good thing the EU is protecting you from orwellian US companies with Orwellian polices on software distribution |
| 00:27.59 | drmessano | You had me confused there |
| 00:28.19 | mvanbaak | drmessano: you clearly did not read the .eu law. |
| 00:28.24 | coppice | drmessano: the banning of bundling has consistently been found to be an important way to keep markets open |
| 00:28.26 | jblack | drmessano: Oh come on. |
| 00:29.00 | Nest0r | [TK]D-Fender: I also have a Quintum and this presents no problem, only the card with 4 ports |
| 00:29.04 | mvanbaak | it states that if $some_vendor_stuff is installed by default they should at least provide 2 alternatives and not set a default |
| 00:29.12 | drmessano | Fine, my next car need not come with a factory stereo.. Those bastards at GM can KEEP their fucking stereo.. I want CHOICE |
| 00:29.42 | mvanbaak | drmessano: here in .nl you dont get a default stereo |
| 00:29.44 | drmessano | I demand, by law, they remove all stereos and include iPods instead |
| 00:29.46 | docidu | well, you should be able to either get the car with no stereo or return it for a refund |
| 00:29.51 | drmessano | Lets call them N edition GM cars |
| 00:29.51 | docidu | and easilly removeable |
| 00:29.59 | drmessano | As in No stereo |
| 00:30.02 | jblack | actually, I believe there's case law on car stereos, and you can force them to tell you a care without one. |
| 00:30.15 | docidu | fun |
| 00:30.22 | mvanbaak | drmessano: like I said, audio system is an option here in .nl |
| 00:30.25 | drmessano | jblack: I think the US needs to force GM to not include stereos |
| 00:30.29 | *** join/#asterisk Eduardo_Assis (n=Eduardo_@189-55-3-106-nd.cpe.vivax.com.br) |
| 00:30.47 | drmessano | Not a choice, but not offer cars with Stereos in the US anymore |
| 00:31.01 | coppice | drmessano: I expect Alpine and Pioneer would find that a very wise move |
| 00:31.02 | jblack | Oh, I see your argument now |
| 00:31.09 | mvanbaak | hhmm |
| 00:31.24 | mvanbaak | are you actually seeing the stuff i say here ? |
| 00:31.29 | drmessano | Sell your damn Suburbans in Canada, GM.. I want a Suburban B |
| 00:32.01 | docidu | you actually want a rattle trap that pulls all of 6mpg? |
| 00:32.38 | mvanbaak | drmessano: here in .nl you dont get a default stereo <--- read that again |
| 00:32.41 | coppice | he feels for the plight of the oil companies with falling sales |
| 00:32.47 | drmessano | mvanbaak: Of course not.. You said earlier when no one agreed with you that we were narrowminded and not reading anything you said.. why start now, right? |
| 00:33.20 | drmessano | mvanbaak: I read what you said, clearly you cant read what I am saying, so I will drop it |
| 00:33.28 | jblack | that was the point I stopped reading what he was saying too. |
| 00:33.43 | mvanbaak | stupid yanks |
| 00:33.47 | drmessano | Of course |
| 00:33.47 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:34.23 | drmessano | Different opinion <--- Dumbasses.. We also have freedom of speech and lack of government issues opinions |
| 00:34.35 | drmessano | I think Wikipedia has an article on it |
| 00:34.47 | jaytee | and if we want to and have half a brain we can run linux instead of windows! |
| 00:34.48 | drmessano | issued |
| 00:35.10 | drmessano | jaytee: No, we must be mandated to use Linux.. |
| 00:35.11 | mchou | drmessano: what do you mean? where you been the last 8 yrs? |
| 00:35.18 | docidu | freedom of speech? i kinda remember when i was a little kid, they sort of had that in the US |
| 00:35.23 | jblack | anyways, your point, if I understand it correctly, is that you're fine on them having to offer nothing bundled as an option, but not ok on forcing them to offer nothing bundled at all. Correct? |
| 00:35.52 | mvanbaak | jblack: that's meant for me ? |
| 00:36.01 | jblack | Nope. |
| 00:36.03 | *** join/#asterisk MainMax (n=MainMax@72.243.24.201) |
| 00:36.06 | mchou | drmessano: Bush and Co. offered may opinions on Torture and wiretapping |
| 00:36.07 | drmessano | jblack: Freedom of choice... not forced choice through government regulation |
| 00:36.17 | jblack | Ok. I can understand that position. |
| 00:36.25 | jaytee | it irritates our Corporate Masters when we run linux but they are powerless to stop us without totally shredding the Constitution. Fortunately, Bush and Cheney bought their shredder at Office Depot and it broke before they could shred it totally. |
| 00:36.35 | mvanbaak | you guys should google for John Pilger |
| 00:36.39 | jblack | I think it works, but I can understand it. |
| 00:36.47 | jblack | Pardon, I think it doesn't work, but I can understand it. |
| 00:36.56 | mvanbaak | and see for yourself how 'free' .us is |
| 00:37.22 | mchou | mvanbaak: nothing in the US comes for 'free' |
| 00:37.26 | drmessano | Forcing M$ to remove Media Player was more orwellian than manufacturers just putting fucking Real Player, WinAMP, or whatever else in the bundle |
| 00:37.34 | MainMax | Hi. Is it a way to have a shared address book in asterisk that would get automatically imported into soft-phone (like X-Lite) |
| 00:37.45 | mchou | mvanbaak: someone's goota pay. It's just the nature of capitalism :) |
| 00:37.51 | drmessano | Windows N edition is a fucking laughable version of Freedom |
| 00:38.06 | mvanbaak | http://video.google.com/videoplay?docid=-3739500579629840148 |
| 00:38.35 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-ebf02a0d2d1e31e7) |
| 00:38.44 | jblack | drmessano: That's where I think your logic fails. That company isn't exactly the most open, and has a proven record of having dual apis. The internal api that they use, and less efficient public api that others can use. |
| 00:38.45 | mvanbaak | drmessano: there's no Windows N edition in .nl |
| 00:38.58 | *** join/#asterisk RobH (n=RobH@120-118.186-72.tampabay.res.rr.com) |
| 00:39.34 | jblack | I don't think that either approach (may vs. can't) solves the actual problem -- a level playing field. |
| 00:39.35 | mvanbaak | drmessano: what i forced here, is that when a vendor ships an OS by default they should at least provide two alternatives with no default set |
| 00:40.00 | mvanbaak | what i forced -> what is forced |
| 00:40.21 | jblack | well, only solves half of it. I think the other half required is that microsoft should be forced to use the same api for IE that they force on others. |
| 00:40.30 | docidu | of corse theuldny be banned from bundleing software however they like, they should jsut be required to offer a clean product without that garbage as well, and not burry it or mae t impossible to get |
| 00:40.32 | drmessano | jblack: Whether or not they include Media Player or IE has nothing to do with their closed nature.. I value that M$ is a closed secretive bunch of SOBs, but forcing removal of an application so others have a chance to compete in a playground YOU built is not the job of the govt |
| 00:40.46 | mvanbaak | so if you buy a pc with windows, there are at least 2 alternatives and none of them is set as default |
| 00:40.48 | jblack | That's less free for them, but more free for competitors and ultimately provides more choice. |
| 00:41.20 | jblack | I agree with you that's the crucial issue; lack of openness. |
| 00:42.11 | jblack | I also know that they have a habit of sham openness... public api's that shift so quickly, no one can reasonably keep up. |
| 00:42.27 | drmessano | What if they forced PC manufacturers to leave out the CPU, offer an externally available socket for the CPU, just to give more freedom at purchase time over the processor you end up with? Who gives them the fucking right? |
| 00:43.02 | jblack | I think a la carte should always be a viable option for consumers. I hate lockin. |
| 00:43.34 | mvanbaak | drmessano: they have that here |
| 00:43.39 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 00:43.43 | drmessano | But it is.. I can get machines both ways.. or I can buy it included and not fuck with it |
| 00:43.54 | mchou | you guys seen this?? http://news.bbc.co.uk/1/hi/world/europe/7949018.stm |
| 00:44.00 | mchou | lol |
| 00:44.08 | drmessano | I have a choice.. not govt mandated exlusion of parts to open competiton |
| 00:44.10 | mchou | ultimate geek |
| 00:44.10 | jblack | drmessano: No, actually, you can't. |
| 00:44.13 | mvanbaak | drmessano: if you offer machineA with intel CPU you have to offer machineB with AMD with the same specs for roughly the same price |
| 00:44.25 | jblack | You can get close, but there's a divide there, and it's grown. |
| 00:44.42 | drmessano | mvanbaak: and thats completely too much govt intervention.. |
| 00:44.52 | mvanbaak | no it's not |
| 00:45.23 | drmessano | Oh hell yes it is.. |
| 00:45.41 | drmessano | Forced regulation at that level is insane |
| 00:46.07 | drmessano | Do they also force you to have a boy and a girl child? |
| 00:46.15 | mvanbaak | ah, so you agree with the fact that the company with the most money can force what users get eh ? |
| 00:47.01 | drmessano | Survival of the fittest.. its how competition has always worked since the day the earth cooled and life formed |
| 00:48.00 | mvanbaak | so why the fuck aren't you using CCME ? |
| 00:48.57 | drmessano | Better question is, why isnt Asterisk use mandated by the govt? |
| 00:49.12 | jblack | nothing like an evening of computerchair philosophy. |
| 00:49.22 | drmessano | I should have a choice of IOS or Asterisk on all Cisco boxes |
| 00:49.33 | jblack | I would agree with that. |
| 00:49.34 | drmessano | Its just the right thing to do |
| 00:49.36 | drmessano | More Kool Aid pls |
| 00:49.57 | drmessano | jblack: By govt mandate? |
| 00:49.58 | jblack | Well, almost, anyways. |
| 00:50.12 | jblack | I'd agree you should have the choice to replace IOS with Asterisk. |
| 00:51.04 | jaytee | "people should not fear their government, government should fear their people" |
| 00:51.15 | drmessano | If the govt mandated tomorrow that Cisco needed to open up their telephony products to allow something other than IOS to run on them, why the hell should anyone build a better mousetrap anymore? |
| 00:51.37 | jblack | to make money. |
| 00:51.42 | drmessano | The govt will just mandate the other guys shit can run on what you build when you get too big |
| 00:52.03 | *** join/#asterisk Spirits-Sight (n=christop@c-71-192-91-123.hsd1.ma.comcast.net) |
| 00:52.06 | mvanbaak | you guys just dot get it |
| 00:52.07 | jblack | i think it should always be that way. |
| 00:52.18 | jaytee | I built a better mousetrap once, but there were all those pesky NRC regulations and red tape regarding the transport and storage of fissionable material. |
| 00:52.38 | Spirits-Sight | quick question, I reset my router and tryng to remember the ports that I should forward to my * box |
| 00:52.40 | jblack | Being big is fine. Preventing others from being big is not. if you want to be big, you'll just have to consistantly do a better job. |
| 00:53.01 | drmessano | Seems to me the goal would be to build something only slightly better than the other guy, since you are going to have govt mandated competition anyway.. Like the car of tomorrow in Nascar |
| 00:53.03 | mvanbaak | Spirits-Sight: default asterisk install ? |
| 00:53.22 | jblack | Spirits-Sight: The various ports will be defined in /etc/asterisk, typically in iax.conf, sip.conf and rtp.conf |
| 00:53.22 | Spirits-Sight | yes for sip |
| 00:53.33 | jaytee | 5060 |
| 00:53.38 | drmessano | Joe is building a better car than you and winning all the races? Fuck em, we'll mandate everyone use the same car |
| 00:53.44 | mvanbaak | Spirits-Sight: udp port 5060 and udp portrange 10000 to 20000 |
| 00:53.45 | jaytee | 10000-20000 for rtp |
| 00:53.45 | Spirits-Sight | thats what I throught but wanted to make sure |
| 00:53.49 | *** join/#asterisk mattzerah (n=matt@CPE-124-177-67-59.qld.bigpond.net.au) |
| 00:54.10 | jblack | drmessano: I think I've been treating your opinions fairly and with respect. |
| 00:54.19 | Spirits-Sight | jaytee: I for get whats the 10000-20000 do? |
| 00:54.26 | drmessano | Reminds me of something I read on a comic once.. |
| 00:54.32 | mvanbaak | Spirits-Sight: it's RTP |
| 00:54.38 | jaytee | for the RTP audio streams |
| 00:54.39 | jblack | It would be nice if you returned the favor. |
| 00:54.47 | drmessano | "Everyone is unique, just like the rest of us" |
| 00:54.50 | mvanbaak | Spirits-Sight: it's the actual audio |
| 00:54.57 | drmessano | jblack: Havent disrespected your comments whatsoever |
| 00:55.31 | jblack | Oh, well, somewhere between what you mean to say, and what I read, caustic sarcasm is intruding. |
| 00:55.49 | Spirits-Sight | jaytee: which one should I put it under UDP / TCP or both for sip / rtf? |
| 00:55.54 | jblack | goes off to read a book |
| 00:56.25 | mvanbaak | I'm going to bed |
| 00:56.27 | mvanbaak | latero all |
| 00:56.27 | jaytee | Spirits-Sight, usually udp only but if you're running * 1.6 and want to use tcp as well then add both |
| 00:56.33 | jaytee | nite mvanbaak |
| 00:56.42 | drmessano | I guess I need to go to bed too, since I am mandated |
| 00:56.52 | drmessano | Cant sleep longer/shorter |
| 00:56.54 | *** part/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 00:56.57 | jaytee | drmessano, exactly how many men have you dated? |
| 00:57.09 | Spirits-Sight | jaytee: so if I pick both on both of them it won't be a issue I am running 1.4 I believe it was |
| 00:57.10 | drmessano | jaytee: Just you, sweet cheeks |
| 00:57.39 | jaytee | Spirits-Sight, then just use udp, 1.4 doesn't support SIP/TCP |
| 00:57.50 | coppice | jaytee: come on. he's never dated men. who'd go out with him? |
| 00:58.10 | Spirits-Sight | jaytee: thansk |
| 00:58.16 | drmessano | Your wife? |
| 00:58.20 | drmessano | .... |
| 00:58.28 | KavanS | ohhh SNAP! |
| 00:58.30 | drmessano | That was kinda funny |
| 00:58.38 | KavanS | *SNAP* |
| 00:58.39 | drmessano | But 5th grade-ish |
| 00:58.48 | KavanS | that was a double snap....no question about it |
| 00:59.06 | KavanS | these situations are quite uncommon.....however they do occur |
| 00:59.36 | jaytee | "so can anyone tell us who first suggested the idea of asexual reproduction?...... David?" "Um, your wife?" |
| 00:59.53 | drmessano | Nice one |
| 01:00.03 | jaytee | that was the best line in War Games |
| 01:00.25 | drmessano | ------> pencil |
| 01:00.32 | drmessano | Bets password ever |
| 01:00.34 | drmessano | Best |
| 01:00.36 | jaytee | even better than John Spencer (rest in peace) going on about the primo bud his buddhist girlfriend was growing |
| 01:01.22 | jaytee | best place to store passwords, a 3x5 index card taped to the pullout sideboard of the desk the computer is on |
| 01:01.40 | drmessano | With the old passwords crossed out |
| 01:01.48 | drmessano | Pattern? HELO? |
| 01:01.59 | jblack | better yet, a text file with a list. |
| 01:02.17 | drmessano | Sticky note under the keyboard |
| 01:02.24 | drmessano | Ultimate fail |
| 01:02.41 | drmessano | NO ONE will look there |
| 01:02.45 | coppice | jaytee the latest LCD monitors with narrow surrounds really limit the space for proper post it note password management |
| 01:03.11 | jaytee | I have plenty of room with my two 19" widescreens |
| 01:03.36 | jaytee | it's finding all the good 2880x900 wallpaper that's a bitch |
| 01:03.42 | drmessano | ahahha |
| 01:04.11 | Chainsaw | Even just 1376x768 wallpapers are rare. |
| 01:04.28 | coppice | I have no problem with 3840x1200 wallpaper, and windows really look like windows with that bar down the middle |
| 01:04.32 | Chainsaw | Especially if you don't want a city skyline reflected in the water. |
| 01:04.42 | Chainsaw | Because they're all like that. |
| 01:04.45 | jaytee | <PROTECTED> |
| 01:05.24 | jaytee | as a matter of fact one of them is Seattle from the water with the city's reflection :-) |
| 01:05.26 | drmessano | jaytee: do what I do.. when you find roadkill, take a side by side shot of each half of whats left |
| 01:05.48 | drmessano | You can recreate it at home on the dual screens |
| 01:05.57 | drmessano | Head <-----------> Body |
| 01:06.36 | drmessano | Then for giggles, mix and match the parts |
| 01:06.44 | drmessano | Deer Head <-----> Cow body |
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| 01:10.19 | [TK]D-Fender | grabs his rifle and looks for rue_afk |
| 01:13.09 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 01:13.37 | ZX81 | hey all - does anyone know what would cause an IAX2 peer to be Ok (20ms) at one end and UNKNOWN at the other end? |
| 01:13.53 | ZX81 | the customer pbx sees the link as 20ms |
| 01:14.02 | *** part/#asterisk Mulvane_ (n=_@75-132-201-105.dhcp.stls.mo.charter.com) |
| 01:14.03 | ZX81 | and the server it is connected to says UNKNOWN |
| 01:14.28 | ZX81 | other peers with identical config show as OK on both server and CPE |
| 01:15.12 | ZX81 | :) oh well |
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| 01:31.32 | deltaray2 | Someone gave a review of the Linksys SPA942 phone I'm using and said that there are things that can be done on the VoIP server side to increase the volume of the phone. Is that true? |
| 01:33.55 | ZX81 | no |
| 01:33.56 | ZX81 | :) |
| 01:33.57 | ZX81 | kinda |
| 01:34.08 | ZX81 | you could use func_volume |
| 01:34.13 | ZX81 | in asterisk 1.6 |
| 01:34.17 | ZX81 | not sure if its in 1.4 |
| 01:34.33 | ZX81 | have you got the phone set to +6db? |
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| 01:35.18 | ifnotwhynot | anyone active? |
| 01:38.57 | jaytee | I'm usually sedentary |
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| 01:40.53 | carrar | pushes jaytee up a hiking trail |
| 01:41.18 | jaytee | whoa! way to much movement for my tastes |
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| 02:03.23 | Nest0r | ok, bye |
| 02:03.28 | Nest0r | thanks for all :) |
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| 02:07.15 | ifnotwhynot | my snom 370 hangs the whole time and the buttoms that monitors activity on extensions stops working, does anyone know how to resolve this? |
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| 03:08.35 | hadronzoo | Hello, I'm experiencing a delayed echo with both internal and external calls through asterisk to a PAP2T. Where can I change the echo cancellation settings for Asterisk? Also, what is the algorithm? Adaptive LMS? |
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| 03:47.32 | p1mrx | hi, so I've got Asterisk 1.6.0.6, and I'm trying to take 2 incoming SIP calls, and connect them to each other. |
| 03:48.17 | p1mrx | I see that there's a Bridge() i can use in extensions.conf, but I'm not sure how to use it |
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| 03:51.32 | ectospasm | p1mrx: normally you'd just accept the incoming call, then Dial() the other channel you want to connect to... |
| 03:51.55 | p1mrx | ectospasm: but I want to connect with a phone that's already "off the hook" |
| 03:52.09 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 03:52.24 | ectospasm | p1mrx: afaik it doesn't work like that. |
| 03:52.38 | lmadsen | Bridge() does |
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| 03:52.46 | ectospasm | Ah, OK |
| 03:52.50 | lmadsen | you need to provide the channel you want to 'bridge' together |
| 03:52.55 | ectospasm | I'm unfamiliar with Bridge() |
| 03:53.17 | p1mrx | how do you specify a channel? does the channel have to be in some certain state in order to allow something to bridge with it? |
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| 03:53.22 | *** mode/#asterisk [+o Mog] by ChanServ |
| 03:53.25 | lmadsen | so for example... if you had a channel that was sitting in MusicOnHold(), then you could dial an extension from another phone, and call Bridge() with the channel name of the device bridged to MusicOnHold() |
| 03:53.38 | lmadsen | it needs to be up |
| 03:53.46 | p1mrx | can it be in Echo() mode? |
| 03:53.49 | lmadsen | SIP/dwayne-045569a |
| 03:53.51 | lmadsen | for example |
| 03:54.03 | lmadsen | yes, it can be in any application as long as the channel exists |
| 03:54.11 | p1mrx | ok, do I get that channel name using "sip show channels" ? |
| 03:54.15 | lmadsen | yes |
| 03:54.24 | lmadsen | or some other dialplan method |
| 03:54.44 | lmadsen | such as setting the channel name in the AstDB or something |
| 03:55.19 | lmadsen | so when a device placed a call, you could Set(DB(phones/100/current_channel)=${CHANNEL}) for example |
| 03:55.28 | p1mrx | I'm just trying to do it manually right now; dial into Echo() mode, capture the channel name, type the channel name into the dial plan, and then take an incoming call |
| 03:55.30 | lmadsen | then another channel could retrieve the current channel name from the DB |
| 03:55.57 | lmadsen | the last part of your sentence confuses me |
| 03:56.19 | p1mrx | er, well I've got this google voice account, and I can receive incoming SIP calls through it via Gizmo |
| 03:56.33 | p1mrx | and, I can click2call from the web interface |
| 03:56.56 | p1mrx | I'm trying to create a script which, after I dial a number, it does some web magic to trigger the incoming click2call, and bridge it into my handset. |
| 03:58.00 | p1mrx | normally, with click2call, I type a number into the web interface, and my phone rings. when I answer, it completes the call. |
| 03:58.34 | lmadsen | so when the call comes in... why not just do a Dial()? |
| 03:58.40 | hadronzoo | btw, I fixed my echo problem by setting the FXS Port Impedance to 220+820||120nF |
| 03:58.41 | lmadsen | I don't understand why you need Bridge() |
| 03:58.59 | p1mrx | hm... can I Dial() a phone that's currently in the process of placing an outbound call? |
| 03:59.42 | hadronzoo | I am still interested in the algorithm Asterisk uses for echo cancellation, though |
| 03:59.59 | lmadsen | p1mrx: sure you can.. it'll just be a separate call to the device |
| 04:00.24 | lmadsen | hadronzoo: it had several software echo cans built in for hardware devices... I just forget where they are exactly... :) |
| 04:00.33 | *** join/#asterisk thelordmortis (n=lordmort@203.8.160.250) |
| 04:00.34 | p1mrx | lmadsen: but won't that not be transparent to the device? I want the device to think it just placed a normal outbound call. |
| 04:00.42 | ectospasm | they've been moved to DAHDI, haven't they? |
| 04:00.51 | *** join/#asterisk jpeeler-home (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 04:01.49 | hadronzoo | lmadsen: but no filters for sip channels? |
| 04:02.01 | p1mrx | ok, so I've got this rule: exten => s,1,Bridge(SIP/softphone-30092c10) |
| 04:02.19 | p1mrx | and softphone-30092c10 is currently in Echo() mode |
| 04:02.36 | p1mrx | that *should* cause an incoming call to patch into the channel, right? |
| 04:03.08 | lmadsen | p1mrx: that would bridge the channel executing Bridge() to be bridged to the SIP/softphone-..... channel |
| 04:03.14 | joobie | aiksa[LV], u there? |
| 04:03.41 | lmadsen | hadronzoo: you can check google... I'm not an echo expert... I think you can get some software echo cans, but that might only apply to hardware. Not sure. |
| 04:03.49 | lmadsen | OSLEC is one type of software echo can |
| 04:04.32 | p1mrx | lmadsen: okay, so with that rule, when my incoming call arrives, the Echo stops, but then I just get silence on the line. |
| 04:04.39 | hadronzoo | lmadsen: I was googling, and I found OSLEC, but it sounded like it only worked for Zaptel-compat hardware. I know the theory, but I unfortunately don't know what's available |
| 04:05.00 | hadronzoo | lmadsen: but thanks for the response |
| 04:05.15 | p1mrx | maybe I should try putting the two calls into different contexts... |
| 04:05.31 | drmessano | EC for SIP? |
| 04:05.58 | lmadsen | you shouldn't need EC for digital channels though |
| 04:06.02 | drmessano | No |
| 04:06.14 | drmessano | By the time its digital, theres nothing to cancel |
| 04:06.20 | drmessano | You cant cancel it at that point |
| 04:06.46 | hadronzoo | drmessano: if someone's mic is feeding back what's coming out of the speaker, won't that be sent back? |
| 04:06.47 | drmessano | You need EC where echo is generated, which is on the analog side |
| 04:06.53 | drmessano | Thats not echo |
| 04:06.59 | drmessano | Thats feedback on the client end |
| 04:07.06 | drmessano | They need to fix their setup |
| 04:07.15 | drmessano | Use headphones instead of speakers |
| 04:07.20 | drmessano | Get a better softphone |
| 04:07.21 | mmlj4 | anyone ever hear of this? I can't get ${EXTEN} to work on one * box, but it's of course fine on another box (it evaluates to an empty string on the first box)... thoughts? |
| 04:07.21 | drmessano | or phone |
| 04:07.26 | hadronzoo | drmessano: but you could kill it easily with an LMS filter |
| 04:07.32 | drmessano | IN the device |
| 04:07.37 | drmessano | Not in the SIP channel |
| 04:07.46 | drmessano | There is no echo in the SIP channel |
| 04:08.07 | hadronzoo | drmessano: feedback by your definition |
| 04:08.11 | hadronzoo | drmessano: not echo |
| 04:08.14 | drmessano | Not by my definition |
| 04:08.25 | drmessano | That IS what it IS |
| 04:08.45 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:08.49 | *** part/#asterisk kn0x (n=pinochle@67.159.48.101) |
| 04:08.54 | drmessano | Echo is generated on analog lines by improper impedence matching or other line matching problems |
| 04:09.13 | p1mrx | hm... after the bridged call arrives, "sip show channels" indicates that one channel is "gsm", and hte other is "ulaw". Could that be preventing them from communicating? |
| 04:09.21 | ectospasm | ...or at least, the type of echo that Asterisk and related hardware can cancel... |
| 04:09.25 | drmessano | Feedback in a client is generated by someone with the speakers too close to the mic |
| 04:09.31 | hadronzoo | drmessano: when people hear themselves, they informally call that echo. I'm referring to feedback in this case |
| 04:09.49 | drmessano | hadronzoo: That is NOT an asterisk problem |
| 04:09.52 | ectospasm | hadronzoo: that's a problem in the SIP phone, most likely. |
| 04:09.57 | ectospasm | Asterisk can't do anything about that. |
| 04:09.59 | drmessano | hadronzoo: That is a problem with the phone |
| 04:10.32 | drmessano | You can NOT filter that in the SIP channel.. its alread digital and is NOT feedback, echo, shamwow, or whatever you want to call it by your own definition |
| 04:10.33 | hadronzoo | drmessano: yes, but you coud correct it regardless of whose problem it is by characterizing the channel |
| 04:11.00 | drmessano | You need to correct it at the CLIENT end |
| 04:11.04 | drmessano | Not using a cheap phone |
| 04:11.08 | drmessano | or cheap softphone |
| 04:11.21 | drmessano | or getting a headset instead of using speakers |
| 04:11.52 | hadronzoo | drmessano: This is a theoretical discussion. I solved my problem. However, it is possible to characterize a channel to remove feedback |
| 04:12.22 | drmessano | This isnt theory.. this is about client problems with feedback being generated |
| 04:12.29 | drmessano | Which is not a PBX problem |
| 04:12.36 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 04:12.51 | drmessano | Its a problem with the device/client and needs to be corrected there through user intervention or not using cheap hardware |
| 04:13.00 | hadronzoo | drmessano: my point is that a PBX could remove feeback if it needs to be used with cheap hardware |
| 04:13.17 | drmessano | Using X-lite with speakers at full blast and a $5 mic is not how you use a softphone |
| 04:13.18 | p1mrx | lmadsen: awesome! Bridge() worked once I forced my client to use the ulaw/pcmu codec. Is it a known limitation that you can't bridge 2 different formats? |
| 04:14.05 | drmessano | hadronzoo: That makes no sense.. if your cheap hardware is generating feedback, it needs to be trashed.. not even a $9 Wal Mart POTS phone will do that most of the time |
| 04:14.22 | drmessano | Only heard that really with Grandstreams and people using speakers with softphones |
| 04:14.24 | p1mrx | oh, lmadsen isn't here anymore. anyone else know if it's supposed to be possible to use Bridge() with different formats? |
| 04:15.42 | hadronzoo | drmessano: change the hardware or characterize the channel and adaptively filter. |
| 04:16.04 | drmessano | hadronzoo: Pointless |
| 04:16.10 | hadronzoo | drmessano: :) |
| 04:16.56 | mmlj4 | grandstreams are one step removed from worthless |
| 04:17.15 | hadronzoo | drmessano: anyway, it's a moot point. Luckily my hardware isn't cheap enough to need that and I just had an impedance mismatch which was the cause of my problems |
| 04:17.24 | hadronzoo | drmessano: It's just an interesting problem |
| 04:17.42 | drmessano | hadronzoo: It is a moot point.. and doesnt need to be solved in the SIP channel, where its pointless to try anyway |
| 04:17.55 | drmessano | hadronzoo: Fix the shitty analog side of the problem |
| 04:17.59 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.63) |
| 04:18.25 | drmessano | hadronzoo: $5 solution looking for a $500 problem |
| 04:18.32 | hadronzoo | drmessano: most hardware (including speakerphones) have sufficient noise cancellation? |
| 04:18.59 | hadronzoo | drmessano: it's just interesting to me because it's a common PHY layer problem in wireless (ISI) |
| 04:19.06 | drmessano | hadronzoo: A $10 speakerphone from Wal Mart does.. Why not expect that in a VoIP phone |
| 04:19.34 | drmessano | hadronzoo: If a $40 VoIP phone doesnt work as well as a $10 GE phone from Wal Mart, it needs to be burned |
| 04:22.14 | hadronzoo | drmessano: You may not control the phone on the other end, but I take your point. I guess feedback has been efficiently solved by the phone companies long ago, so it's not really a problem. |
| 04:24.51 | drmessano | hadronzoo: On the contrary, it can be a problem.. In the scenarios I mentioned.. EC on an analog line is fixed either as a Zaptel addon or hardware EC, which fixes it AT the point of A-D conversion.. On the client side, "echo" (feedback) is common with people who are too dumb to use a softphone properly, or using cheap phones like grandstreams |
| 04:25.14 | drmessano | All with easy solutions long before hitting a SIP channel |
| 04:28.13 | hadronzoo | drmessano: but what if you're not in control of the user generating the feedback (like someone calling you from a softphone)? |
| 04:28.32 | *** join/#asterisk Kyosh (i=48595da5@gateway/web/ajax/mibbit.com/x-555ca052e00c2412) |
| 04:29.43 | drmessano | Again, they need to fix their shit.. What if they have a loud BUZZING sound from a flourescent light 2 inches from the $6 chinese VoIP phone.. do you need a filter for that too? |
| 04:31.51 | hadronzoo | drmessano: right. |
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| 05:17.18 | mattzerah | hi, i'm in desprarate need for some help - occasionally calls made from our asterisk box has a delay in dialling (sometimes up to 20 seconds) - this happens regularaly, i found this in the logs when the delays occur, can someone tell me whats going on ???? |
| 05:17.22 | mattzerah | [Mar 19 15:13:46] DEBUG[7744] sched.c: Request to schedule in the past?!?! |
| 05:17.27 | mattzerah | [Mar 19 15:13:46] DEBUG[7744] sched.c: Request to schedule in the past?!?! |
| 05:17.37 | mattzerah | what would sched.c be trying to schedule? |
| 05:17.55 | mattzerah | i dont even know where to start looking :/ |
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| 05:26.27 | carrar | mattzerah: http://tinyurl.com/dc763h |
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| 05:38.52 | bobsaccamano | hi..is it possible to disallow call hold for a partcular sip channel? |
| 05:42.45 | drmessano | carrar: LOL |
| 05:46.45 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-36-113.w86-215.abo.wanadoo.fr) |
| 05:55.51 | the_5th_wheel | Does anyone know how I can run asterisk 1.2 and 1.4 in parralel? |
| 05:55.55 | the_5th_wheel | (on the same Box) |
| 05:56.22 | Mog | yes |
| 05:56.44 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-36-113.w86-215.abo.wanadoo.fr) |
| 05:56.47 | the_5th_wheel | Mog: Elucidate? |
| 05:56.48 | Mog | -C you can specify where /etc/asterisk/asterisk.conf is located |
| 05:57.01 | the_5th_wheel | But that would mean Its two of the same versions |
| 05:57.09 | Mog | you just need to use different interfaces |
| 05:57.16 | Mog | well instal to a different spot |
| 05:57.37 | Mog | like i installed everything into /usr/local/asterisk-1.2 and /usr/local/asterisk/1.4 |
| 05:57.48 | Mog | er asterisk-1.4 |
| 05:58.19 | Mog | i run asterisk-1.4 and asterisk-1.6 in that way |
| 06:01.36 | *** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 06:01.39 | the_5th_wheel | Mog: How do I get to that asterisk's console/ |
| 06:02.20 | Mog | i run them as different users so that isnt a problem for me, i think if you pass the -C flag when you pass the -r flag it should work but i dont know to be honest |
| 06:03.59 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 06:05.45 | the_5th_wheel | Mog: COuld you possibly paste me your Make.conf for asterisk 1.4? |
| 06:06.12 | Mog | the boxes i did this for i dont have access to, you can pass install prefix to ./configure |
| 06:07.55 | the_5th_wheel | Mog: Out of interest, why did you do 1.4 and 1.6 |
| 06:08.10 | Mog | i wanted the autoprovisioning stuff from 1.6 |
| 06:08.12 | Mog | but thats it |
| 06:08.17 | the_5th_wheel | aha |
| 06:08.27 | Mog | so i run the manager console for 1.6 and the webserver |
| 06:08.31 | Mog | thats it |
| 06:08.38 | Mog | symlinked users.conf and i was golden |
| 06:14.19 | the_5th_wheel | Mog: do I just set prefix and exec_prefix? |
| 06:18.03 | Mog | i think --prefix in the configure is what you want |
| 06:18.10 | Mog | and set it to /usr/local |
| 06:18.12 | Mog | or anything |
| 06:19.05 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-074dae74dcdfc4f4) |
| 06:21.25 | the_5th_wheel | But If I look at the Make.conf, bindir='${exec_prefix}/bin', which would mean it would overwrite the executables, Not? |
| 06:23.19 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 06:36.10 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 06:36.37 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) |
| 06:38.39 | pif | hi, what does "pbx_extension_helper: No application 'SetVar' for extension" mean |
| 06:38.57 | pif | i'm really lost on this one |
| 06:45.33 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128744958.dsl.bell.ca) |
| 06:46.04 | bobsaccamano | how can i set distinctive ring tones on asterisk for two different lines? |
| 06:46.50 | aiksa[LV] | hmm, I was wondering - increasing "nice"ness of the asterisk process to lets say -2, what problems I might run into. |
| 06:47.20 | aiksa[LV] | if the kernel related tasks run at -5 |
| 06:47.28 | aiksa[LV] | shouldnt be any right |
| 06:47.46 | aiksa[LV] | and the asterisk will have a greater priority than anything else |
| 06:49.08 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 06:51.33 | bobsaccamano | hi..can i pass a ringtone as an argument to Ringing function? |
| 06:56.20 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 06:59.25 | *** join/#asterisk Daejeo (n=chatzill@110.9.169.216) |
| 07:04.45 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:12.56 | *** join/#asterisk joobie (n=joobie@203-217-82-215.dyn.iinet.net.au) |
| 07:16.17 | p1mrx | how would I determine why asterisk is generating a "401 unauthorized"? : http://pastebin.ca/1365013 |
| 07:16.25 | *** join/#asterisk tjz (n=tjz@bb116-14-182-232.singnet.com.sg) |
| 07:16.39 | p1mrx | line 38, specifically |
| 07:17.57 | bobsaccamano | p1mrx, your authentication is failing |
| 07:18.15 | bobsaccamano | are you using sip channels? |
| 07:18.25 | p1mrx | yes, I'm trying to receive a sip call |
| 07:18.41 | bobsaccamano | so check you sip channel config |
| 07:18.46 | p1mrx | and it appears that my server is responding to the call with "unauthorized" |
| 07:18.55 | bobsaccamano | specifically secret and username params |
| 07:19.03 | bobsaccamano | otherwise |
| 07:19.21 | bobsaccamano | just do insecure=very to disable authentication |
| 07:19.26 | bobsaccamano | but its not advisable |
| 07:21.13 | p1mrx | [Mar 19 00:19:37] WARNING[13336]: chan_sip.c:20124 set_insecure_flags: Unknown insecure mode 'very' on line 22 |
| 07:22.00 | kaldemar | p1mrx: insecure=port,invite |
| 07:22.18 | kaldemar | very is removed |
| 07:22.42 | p1mrx | kaldemar: okay, I was able to accept a call after setting that |
| 07:22.57 | bobsaccamano | yeah its removed in 1.6 |
| 07:23.04 | p1mrx | so, I need to create credentials for incoming calls, somehow? |
| 07:24.09 | *** join/#asterisk techie (n=techie@253.sub-75-212-247.myvzw.com) |
| 07:24.24 | p1mrx | I'm obviously able to register with gizmo (proxy01.sipphone.com), but what sort of authentication do I need for incoming calls, to avoid needing to set insecure? |
| 07:24.50 | kaldemar | p1mrx: your call matches to peer proxy01.sipphone.com, now you just have to fix the authentication. |
| 07:25.20 | p1mrx | authentication with whom, though? I wouldn't be receiving the call if I didn't register properly? |
| 07:25.29 | p1mrx | -? |
| 07:25.54 | kaldemar | registering has little to do with incoming calls. an incoming call has to be authenticated despite any registers. |
| 07:26.02 | *** join/#asterisk kerx (n=kerx@adsl-69-104-17-222.dsl.irvnca.pacbell.net) |
| 07:26.33 | kaldemar | registering is basically just a way to tell the other end where you are. |
| 07:27.33 | kaldemar | you need to define a secret for the peer and the other end needs to use the same secret calling you. |
| 07:27.57 | p1mrx | hm, well I don't have any way to tell gizmo a secret |
| 07:32.04 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 07:35.10 | p1mrx | ok, well at this point I'd rather have a system accept calls from anywhere than from nowhere. thanks. |
| 07:38.29 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 07:40.26 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 07:51.29 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128744958.dsl.bell.ca) |
| 07:59.24 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 08:11.46 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 08:11.46 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:16.19 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:16.37 | *** join/#asterisk obruT (n=turbo@griffin.linux.hr) |
| 08:18.43 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:20.36 | *** join/#asterisk stoffell (n=kristof@220.41-201-80.adsl-dyn.isp.belgacom.be) |
| 08:23.15 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:26.51 | *** join/#asterisk kaou_ (n=kaou@aghnar.doowan.net) |
| 08:30.40 | *** join/#asterisk tokozedg (n=slA@85.118.98.122) |
| 08:31.56 | tokozedg | hi, i have php script end when i open it gives me i number, so how can i set this variable in asterisk, i mean VAR=(the number which php script has echoed)? |
| 08:41.36 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 08:48.23 | *** join/#asterisk r0d3nt (i=astrutt@pinky.ratman.org) |
| 08:48.24 | joobie | tokozedg, as a variable to be used in the dialplan? how does the php script interact with the dialplan? |
| 08:56.58 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 08:58.40 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 08:59.04 | *** part/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 09:00.00 | *** part/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 09:00.03 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 09:00.46 | shyam_k | is 1.4 and 1.6 two "Current" versions with difference in some structure or just "stable" and "unstable" versions? |
| 09:02.10 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 09:02.18 | *** join/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 09:02.36 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) |
| 09:02.53 | joobie | shyam_k, there's a stable 1.4 and stable 1.6 |
| 09:02.59 | joobie | there's a unstable 1.6 |
| 09:03.04 | joobie | duno if there's an unable 1.4 |
| 09:03.08 | joobie | dont thin kthere is |
| 09:03.14 | joobie | 1.6 went stable relatively recently |
| 09:03.22 | joobie | if that carries any weight |
| 09:04.15 | joobie | guys im looking at using txfax() in a dialplan to send a fax.. i looked at 'show application txfax' and the syntax takes the tiff filename as the first arguement in the syntax.... there doens't seem to be a way to specify the channel to use |
| 09:04.41 | joobie | any ideas why? .. i have been using this to send via the call file, but i want this to dial via the dialplan now |
| 09:05.21 | shyam_k | joobie: thanks! will try both, was thinking which one to take first;-) |
| 09:05.57 | joobie | no worries |
| 09:07.45 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-58-80.w86-215.abo.wanadoo.fr) |
| 09:07.55 | NMR_1122 | Hi everyone, when I connected my SPA 2102 (Linksys VoIP Adapter) to my Voip provider, I had to setup a "STUN Server". Now that I'm going through Asterisk, do I need to set a STUN server somewhere so that Asterisk can connect to my Voip provider? |
| 09:08.32 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-58-80.w86-215.abo.wanadoo.fr) |
| 09:09.57 | joobie | NMR_1122, are your clients all connecting to the asterisk box and then asterisk does the routing to your voip provider? |
| 09:10.37 | *** join/#asterisk Vec (n=Vec@87.74.7.50) |
| 09:12.12 | NMR_1122 | yes, right now, all my voip phones can call each other internally, via extensions, but astersik can't get incomming calls or make outgoing calls to the actual provider. |
| 09:12.41 | joobie | ive setup similar without the use of stun |
| 09:12.54 | joobie | with asterisk behind nat |
| 09:13.08 | joobie | look at your sip (presuming you are using sip to the voip provider) nat settings for that specific profile |
| 09:13.18 | joobie | focus on getting your asterisk talking to the voip provider first |
| 09:13.23 | joobie | then look at your phones |
| 09:13.51 | joobie | nat=yes |
| 09:13.58 | joobie | i had to specify that in my sip.conf for the voip provider |
| 09:14.06 | joobie | to indicate the connection is coming from behind nat |
| 09:16.37 | p1mrx | so I've hacked together this horrible blob which lets me dial outbound through google voice: http://pastebin.ca/1365101 |
| 09:16.50 | NMR_1122 | I think it's sort of talking to my viop provider, but not quite... If I try to call out, the terminal shows the error: [Mar 19 05:15:21] NOTICE[21255]: chan_sip.c:12573 handle_response_invite: Failed to authenticate on INVITE to '"OLD FCM Phone" <sip:aquagear@192.168.5.10>;tag=as4c8349a2' |
| 09:16.56 | *** part/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 09:17.17 | NMR_1122 | I added nat=yes to sip.conf and reloaded sip |
| 09:18.04 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:18.05 | p1mrx | goes to sleep |
| 09:19.07 | joobie | my knowledge of this is limited NMR_1122 |
| 09:19.16 | joobie | though what does sip show peers say for that provider? |
| 09:19.36 | joobie | soudns like u need to get ur auth right in sip.conf |
| 09:19.44 | joobie | but sip show peers and ship show peer blah |
| 09:19.47 | NMR_1122 | It shows up as a peer |
| 09:19.52 | joobie | are good commands to show u the status of a sip peer |
| 09:20.03 | joobie | it should tell u if it's registered or not, so u can base ur success on that :P |
| 09:20.11 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 09:20.12 | NMR_1122 | "unmonitored" |
| 09:20.17 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 09:21.33 | joobie | add qualify=yes to that sip peer |
| 09:21.45 | joobie | then it will monitor it and show u the status |
| 09:21.58 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-58-80.w86-215.abo.wanadoo.fr) |
| 09:22.19 | joobie | .. tho that auth failure is probably because u havent specified the correct fromuser/username/secret/host |
| 09:22.25 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-58-80.w86-215.abo.wanadoo.fr) |
| 09:23.01 | fc1979 | Hi 2all. I have "extension not found" when i make a call, in realtime config mode. I try to set 2 number, 1501 and 1401 for testing purpose. This is the asterisk log with command "sip set debug on" |
| 09:23.02 | fc1979 | http://pastebin.ca/1365105 |
| 09:23.23 | NMR_1122 | I have fromuser=myusername, secret=password, host=216.143.130.36 |
| 09:23.40 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 09:23.52 | joobie | u may need to specify username= |
| 09:23.57 | joobie | but try that qualify line first |
| 09:24.09 | joobie | then do ur sip peers |
| 09:24.12 | joobie | show |
| 09:24.36 | fc1979 | As you can see at line 35, i obtain "401 unauthorized" |
| 09:24.58 | NMR_1122 | status says "OK (74 ms)" |
| 09:25.50 | fc1979 | the type set in sip.conf is "friend" |
| 09:26.23 | fc1979 | i don't know how to debug this trouble... |
| 09:27.14 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 09:27.54 | WeazelON | anyone knows whats these mean ? |
| 09:27.55 | WeazelON | Mar 19 10:47:06 WARNING[30517] frame.c: Cannot disallow unknown format '' |
| 09:27.55 | WeazelON | Mar 19 10:47:06 WARNING[30517] frame.c: Cannot allow unknown format '' |
| 09:27.59 | fc1979 | if can help, i find in postgresql "LOG: incomplete startup packet". Postgresql is the backend connected via odbc. I use asterik 1.6 |
| 09:28.40 | fc1979 | i try to make this test with x-lite sofphone. |
| 09:31.43 | WeazelON | anyone ? |
| 09:36.40 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 09:39.37 | WeazelON | i need help from someone guys |
| 09:39.46 | WeazelON | my asterisk crashes and what i see in full is this |
| 09:39.47 | WeazelON | Mar 19 11:24:29 DEBUG[2315] chan_local.c: Not posting to queue since already masked on 'Local/211@default-7916,2' |
| 09:39.48 | WeazelON | Mar 19 11:24:30 DEBUG[1331] channel.c: Got clone lock for masquerade on 'SIP/211-0847c278' at 0x8325834 |
| 09:39.48 | WeazelON | Mar 19 11:24:30 DEBUG[2315] channel.c: Building translator from unknown to SLINEAR for spies on channel Local/211@default-7916,2 |
| 09:39.48 | WeazelON | Mar 19 11:24:30 WARNING[2315] codec_g726.c: Out of buffer space! |
| 09:40.27 | WeazelON | there are about 20 lines of the "Not posting to queue since" |
| 09:40.57 | WeazelON | 2 lines before the first of the "not posting to queue since" are |
| 09:40.58 | WeazelON | Mar 19 11:24:29 DEBUG[2315] channel.c: Planning to masquerade channel SIP/211-0847c278 into the structure of Local/211@default-7916,1 |
| 09:40.58 | WeazelON | Mar 19 11:24:29 DEBUG[2315] channel.c: Done planning to masquerade channel SIP/211-0847c278 into the structure of Local/211@default-7916,1 |
| 09:41.07 | WeazelON | help ! |
| 09:44.04 | WeazelON | can anyone shed some light ? |
| 09:46.21 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 09:47.45 | WeazelON | noone ? |
| 09:49.36 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 09:51.05 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:53.03 | WeazelON | no one ? |
| 09:54.22 | joobie | hey guys.. anyone know if there's a way to set the T38CALL Channel Variable within a dialplan? im submitting the variable now via a Call File, though i want to overwrite it in the dialplan for txfax() |
| 09:54.29 | *** join/#asterisk SamaelA (n=samael@innotel.kiev.farlep.net) |
| 09:54.37 | joobie | my txfax() doesnt accept the channel as the first arg, only accepts the tiff file |
| 09:54.39 | SamaelA | Hello ) |
| 09:54.45 | joobie | so presuming this is read in from some var that i can set / read |
| 09:55.50 | SamaelA | Has anyone got http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO to work? |
| 10:01.53 | *** join/#asterisk synthetiq (n=roger@193.79.224.62) |
| 10:02.49 | synthetiq | if you are using a digium card with built in echo can, and you run the asterisk command echo on that call, will the echo can cancel the echo() app |
| 10:05.53 | *** join/#asterisk AJFisher (n=alex@82-70-11-70.dsl.in-addr.zen.co.uk) |
| 10:05.56 | *** join/#asterisk shyam_k (n=shyam@unaffiliated/shyam-k/x-8459115) |
| 10:06.01 | joobie | http://markmail.org/message/vtj362wpz762g4mw |
| 10:06.09 | joobie | damn.. that is exactly what i want to know - but no answer :/ |
| 10:06.39 | shyam_k | i see the codec_speex option in menuselect unavailable.. does it means, am missing some package, or that its default.. |
| 10:06.45 | WeazelON | i need help from someone guys |
| 10:06.45 | WeazelON | my asterisk crashes and what i see in full is this |
| 10:06.45 | WeazelON | Mar 19 11:24:29 DEBUG[2315] chan_local.c: Not posting to queue since already masked on 'Local/211@default-7916,2' |
| 10:06.45 | WeazelON | Mar 19 11:24:30 DEBUG[1331] channel.c: Got clone lock for masquerade on 'SIP/211-0847c278' at 0x8325834 |
| 10:06.45 | WeazelON | Mar 19 11:24:30 DEBUG[2315] channel.c: Building translator from unknown to SLINEAR for spies on channel Local/211@default-7916,2 |
| 10:06.47 | WeazelON | <PROTECTED> |
| 10:06.49 | WeazelON | ***there are about 20 lines of the "Not posting to queue since" |
| 10:06.51 | WeazelON | 2 lines before the first of the "not posting to queue since" are |
| 10:06.53 | WeazelON | Mar 19 11:24:29 DEBUG[2315] channel.c: Planning to masquerade channel SIP/211-0847c278 into the structure of Local/211@default-7916,1 |
| 10:06.56 | WeazelON | <PROTECTED> |
| 10:07.18 | joobie | WeazelON, pastebin.com is a great site |
| 10:07.32 | shyam_k | i mean the XXX means its already there by default right? |
| 10:08.10 | shyam_k | same with almost everything related to ogg.. |
| 10:13.31 | tzafrir_laptop | The XXX means it is missing some dependency |
| 10:13.42 | tzafrir_laptop | It is not related directly to ogg |
| 10:14.14 | tzafrir_laptop | Ogg is a container. Commonly used with the codec vorbis. |
| 10:14.34 | tzafrir_laptop | (Asterisk's "ogg" is an ogg container with vorbis codec) |
| 10:14.54 | tzafrir_laptop | shyam_k, what distro do you use? |
| 10:15.13 | AJFisher | Asterisk 1.6.0.6 and 1.6.0.7-rc1 are seg faulting on me :( http://pastebin.com/m1c90fab3 |
| 10:15.27 | shyam_k | tzafrir_laptop: gnewsense(ubuntu 8.04) |
| 10:15.28 | tzafrir_laptop | you may need either just libspeex-dev or also libspeexdsp-dev |
| 10:15.48 | shyam_k | didn't care it much, i have already compiled and installed.. |
| 10:16.20 | tzafrir_laptop | shyam_k, generally use distro packages for low-level packages. It will save you troubles |
| 10:16.42 | tzafrir_laptop | speex is used by asterisk, ekiga, and whatnot |
| 10:17.02 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 10:17.08 | SamaelA | M... Can anyone say, which must be paramstrs of Dial(), that asterisk will accept DTFM during the call? Cause it fully ignores my features.conf? |
| 10:17.12 | shyam_k | but i wanna study the system..;-) anyway that was just an extra-extra-care.. asterisk compiled so cool.. |
| 10:17.20 | AJFisher | ah, perhaps I should be asking about the seg fault on #asterisk-dev instead |
| 10:17.35 | *** join/#asterisk fbnts (n=root@89.16.176.40) |
| 10:17.47 | tzafrir_laptop | shyam_k, have you written your own compiler yet? |
| 10:18.01 | fbnts | Hi, I have a Cisco 7941 that is constantly saying Registering |
| 10:18.06 | shyam_k | tzafrir_laptop: ? |
| 10:18.09 | fbnts | I have checked the config file and all seems ok |
| 10:18.36 | fbnts | the status doesn't report any error in the config file either |
| 10:18.38 | tzafrir_laptop | shyam_k, for speex, openh323, and such, stick to distro packages unless you have a good reason |
| 10:18.41 | *** part/#asterisk kaou_ (n=kaou@aghnar.doowan.net) |
| 10:18.48 | tzafrir_laptop | Normally distro packages do a good work |
| 10:19.00 | shyam_k | yeah sure.. |
| 10:19.03 | tzafrir_laptop | (though with the ubuntu masters of the universe you can never tell) |
| 10:19.14 | shyam_k | yeah;-) |
| 10:19.50 | tzafrir_laptop | however a surprising number of the dependencies of asterisk are in their main |
| 10:20.57 | shyam_k | grr install got stuck as it can't get net to download the sound-en-wav from digium.. i can just make install again with the net right? |
| 10:21.27 | joobie | guys im using a call file that uses a local channel in order to then run through the dialplan and use txfax() to dial a specific number.. trouble is, i cant specify a different channel to use for txfax(), so it ends up inherting the channel that was used in the call file (local) and never ends up sending the fax.. is there a way to get around this? |
| 10:21.32 | synthetiq | if you are using a digium card with built in echo can, and you run the asterisk command echo on that call, will the echo can cancel the echo() app? |
| 10:28.18 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 10:29.31 | shyam_k | ok first asterisk installation completed.. |
| 10:34.16 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:36.00 | *** join/#asterisk FreezeS (n=kvirc@82.208.157.125) |
| 10:36.03 | FreezeS | hey guys |
| 10:36.11 | FreezeS | I have a weird problem with DeadAGI |
| 10:36.42 | FreezeS | I'm running it in h but the call is only hung up after the agi exits |
| 10:36.59 | FreezeS | (I'm running this in 1.4.20) |
| 10:37.26 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 10:37.30 | FreezeS | tried running hangup in h before deadagi but then the agi is not ran at all |
| 10:38.20 | nfi|ermes | hi all |
| 10:41.37 | *** join/#asterisk stoffell_w (n=kristof@220.41-201-80.adsl-dyn.isp.belgacom.be) |
| 10:42.32 | joobie | guys is there a way to set the ${CHANNEL} var within the dialplan? I try with set() and it doesn't seem to change it..... |
| 10:44.00 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 10:47.38 | nfi|ermes | i use externnotify parameter in voicemail.conf; is there a way to be notified on ly when a new message is arrived, and not when someone log in tho the voicemail ??? |
| 10:56.47 | joobie | damn |
| 10:56.53 | joobie | docs say channel var cant be set in the dialplan :/ |
| 11:00.05 | the_5th_wheel | Can anyone confirm that this would cut the call after 40 Minutes? Dialing SIP/Number@provider-a|30|L(2.4E+6:61000:30000) |
| 11:03.28 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:04.20 | *** join/#asterisk mizerydearia (n=mizeryde@CPE-65-27-127-37.new.res.rr.com) |
| 11:08.27 | SamaelA | M... Can anyone say, which must be paramstrs of Dial(), that asterisk will accept DTFM during the call? Cause it fully ignores my features.conf? |
| 11:09.57 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 11:11.01 | *** join/#asterisk Eduardo_Assis (n=Eduardo_@200-207-61-133.dsl.telesp.net.br) |
| 11:14.48 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 11:14.52 | fcois93 | hello all |
| 11:15.05 | fcois93 | I work on the voicemail configuration |
| 11:15.24 | fcois93 | I need to have ${VM_DATE} in french ! how to do? |
| 11:16.08 | *** join/#asterisk Gopaul (n=Miranda@61.17.185.118) |
| 11:19.43 | Eduardo_Assis | fc1979, search in google |
| 11:19.52 | Eduardo_Assis | fcois93, search in google |
| 11:20.04 | *** part/#asterisk Eduardo_Assis (n=Eduardo_@200-207-61-133.dsl.telesp.net.br) |
| 11:20.06 | *** join/#asterisk commandata (n=commanda@p5087A17B.dip0.t-ipconnect.de) |
| 11:20.42 | fcois93 | easy to say that and go out! |
| 11:25.40 | *** join/#asterisk destructure (n=de@67-23-12-32.static.slicehost.net) |
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| 11:50.34 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 11:53.40 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
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| 12:07.01 | fc1979 | Eduardo_Assis: i've searched in google, but i didn't find anything useful.. |
| 12:07.36 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:08.30 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 12:14.29 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 12:14.36 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 12:20.18 | joobie | guys how do u read an application return code in the dialplan? |
| 12:20.22 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:24.24 | synthetiq | if you are using a digium card with built in echo can, and you run the asterisk command echo on that call, will the echo can cancel the echo() app? |
| 12:28.16 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:30.41 | *** part/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
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| 12:32.20 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:11.22 | *** join/#asterisk blackmousepad (n=blackmou@intranet.shantycreek.com) |
| 13:11.42 | blackmousepad | I've built an agi script and I need to be able to have the caller disconnect a call by pressing *** or ### but when I add H to the Dial command a single * disconnects the call. How do I customize the disconnect feature code to be *** instead of * Anyone got any ideas? |
| 13:12.05 | *** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-12-127.w90-56.abo.wanadoo.fr) |
| 13:12.44 | merlin8282 | Hi. I now have a problem with a fax and an ISDN telephone, both connected to the same line (2 channels). |
| 13:13.15 | merlin8282 | When I call the fax with a phone, it responds. Ok. But when I call the ISDN phone, the fax also answers ! |
| 13:13.41 | merlin8282 | In fact it's a fax-server, with a Fritz! Card |
| 13:13.55 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 13:13.57 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 13:14.00 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:14.09 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 13:14.28 | merlin8282 | I tried to configure the MSN of the faxserver, but it always shows 14 or 15 as calling MSN in the serverlog |
| 13:14.59 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:15.01 | merlin8282 | Zap/14 is the line where the faxserver is connected, on my *, and Zap/15 is the ISDN phone. |
| 13:15.13 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 13:15.53 | merlin8282 | So if I call Zap/15, faxserver responds, and Zap/14 it responds too, where it should be the ISDN phone answering here. |
| 13:16.01 | merlin8282 | Anybody has an idea ? |
| 13:17.13 | *** join/#asterisk SeicherlBoB (n=seicherl@85-126-76-170.work.xdsl-line.inode.at) |
| 13:18.06 | blackmousepad | I don't have a clue and I may not be understanding your situation correctly but it sounds like a wiring problem |
| 13:18.21 | *** part/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 13:19.49 | merlin8282 | It isn't a wiring problem. The port on the ISDN card (on the *) is connected to an ISDN hub, thus allowing to connect 2 devices. |
| 13:20.06 | merlin8282 | the fax server and the ISDN phone are connected to this hub. |
| 13:20.46 | merlin8282 | Zap/14-15 are one ISDN line. |
| 13:21.28 | [TK]D-Fender | blackmousepad: features.conf <----- |
| 13:22.38 | blackmousepad | TK -- disconnect => *** for *** instead of * right? |
| 13:22.45 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 13:23.55 | phix | hi |
| 13:25.11 | [TK]D-Fender | blackmousepad: Head of the class with you... |
| 13:25.34 | phix | [TK]D-Fender: <3 |
| 13:25.40 | blackmousepad | merlin I'm not really familiar with ISDN (our local telco never bothered to offer isdn or dsl even) |
| 13:26.25 | *** join/#asterisk shyam_k (n=shyam@unaffiliated/shyam-k/x-8459115) |
| 13:27.07 | merlin8282 | ok blackmousepad thanks anyway |
| 13:27.17 | *** join/#asterisk cm_5_1_2 (n=cm_5_1_2@58.252.229.215) |
| 13:29.52 | blackmousepad | TK - Thanks! Now I'm off to figure out what freepbx did with the content of features.conf (I don't want to reinstall my mod every time I apply changes in freepbx) Thanks again! |
| 13:31.10 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
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| 13:34.46 | SeicherlBoB | hi there! I've read the O'reiley books a couple of time and i have a running asterisk 1.4.17 more or less running as a SIP-only PBX... now i'm running into trouble about basic understanding... What I do now is: sip.conf: Register the PBX at a server (several times at the same server for several accounts, all have the extension 0), define extensions in sip.conf for all accounts, everyone with different credentials and a contex |
| 13:34.54 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:38.12 | *** join/#asterisk davevg (n=davevg__@74.94.3.214) |
| 13:38.44 | *** join/#asterisk dni (n=f00kj00@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 13:40.24 | leifmadsen | SeicherlBoB: ...different credentials and a conte <-- where you cut off |
| 13:41.36 | SeicherlBoB | leifmadsen: thanks |
| 13:41.48 | SeicherlBoB | ... everyone with different credentials and a context of their own for the dialplan. What happens is: all incoming calls of one of the sip-accounts are resolved to belong to the same user (which should not happen!)... how can i tell asterisk to "check" for the right user and forward to the right context in the dialplan? |
| 13:42.30 | leifmadsen | SeicherlBoB: depends.... basically the provider needs to send the call as a unique user so that Asterisk can match on the user name instead of the IP |
| 13:42.50 | leifmadsen | right now it sounds like it isn't matching on any user names, and is just matching on IP, which will be the first one it finds |
| 13:42.59 | SeicherlBoB | i thought so too... but they are obviously resolved by domain |
| 13:43.21 | leifmadsen | there is no domain matching in sip.conf |
| 13:43.24 | SeicherlBoB | leifmadsen: actually its always the last one configured as peer in the sip.conf |
| 13:43.40 | leifmadsen | SeicherlBoB: right... which will be the first one you see at the top of your 'sip show peers' list |
| 13:43.51 | SeicherlBoB | leifmadsen: the sip debug tells something about "for domain" and actually states the ip adress |
| 13:44.10 | leifmadsen | just a wording thing |
| 13:44.19 | leifmadsen | but it is matching on IP, no user name |
| 13:44.22 | leifmadsen | not* |
| 13:44.35 | SeicherlBoB | looks like. |
| 13:44.44 | SeicherlBoB | may the fromuser help? |
| 13:44.55 | leifmadsen | possibly... but that is really for outgoing |
| 13:45.04 | SeicherlBoB | thought so too. |
| 13:45.06 | leifmadsen | the registration is what is telling the end point to send calls to you |
| 13:45.15 | [TK]D-Fender | SeicherlBoB: what exten are your calls landing on? |
| 13:45.20 | leifmadsen | actually I take that back -- I don't think fromuser will help at all |
| 13:45.30 | leifmadsen | [TK]D-Fender: most likely 0 |
| 13:45.30 | SeicherlBoB | [TK]D-Fender: all on extension 0 (by the provider) |
| 13:45.32 | [TK]D-Fender | leifmadsen: It won't, thats for outgoing |
| 13:45.38 | [TK]D-Fender | SeicherlBoB: Show us |
| 13:45.43 | leifmadsen | [TK]D-Fender: isn't that what I just said? :) |
| 13:45.47 | SeicherlBoB | moment |
| 13:45.58 | [TK]D-Fender | leifmadsen: I just clarified for why :) |
| 13:45.59 | leifmadsen | goes to make breakfast now that the boss is in :) |
| 13:46.14 | [TK]D-Fender | OMG Springsteen!!!! |
| 13:46.19 | [TK]D-Fender | runs in circles |
| 13:46.30 | leifmadsen | [TK]D-Fender: do you just have a day job, or do you do asterisk consulting? |
| 13:46.30 | phix | [TK]D-Fender: yay |
| 13:46.37 | phix | [TK]D-Fender: can I join you? |
| 13:46.45 | [TK]D-Fender | leifmadsen: both |
| 13:46.57 | phix | [TK]D-Fender: you like to asterisk? |
| 13:47.03 | SeicherlBoB | register => userid:pass:userid@host.server.at:5060/0 |
| 13:47.18 | [TK]D-Fender | phix: * != verb :p |
| 13:47.21 | SeicherlBoB | i have like 5 or more of them. all at the same host.server |
| 13:47.23 | phix | register => SeicherlBoB => <3 |
| 13:47.34 | [TK]D-Fender | SeicherlBoB: Set a unique exten for each and that is how you will separate them |
| 13:47.41 | phix | [TK]D-Fender: :) neither do you :) |
| 13:47.49 | phix | then again |
| 13:47.50 | *** join/#asterisk aksyn (n=aksyn@94-193-98-124.zone7.bethere.co.uk) |
| 13:47.56 | phix | I like to [TK]D-Fender now and then :P |
| 13:48.01 | [TK]D-Fender | phix: "I Am America And So Can You"? |
| 13:48.13 | SeicherlBoB | [TK]D-Fender: but how can i then put them to different contexts? |
| 13:48.18 | phix | [TK]D-Fender: no, its more like all your bas belong to us |
| 13:48.24 | [TK]D-Fender | SeicherlBoB: Might not be able to |
| 13:48.32 | [TK]D-Fender | phix: Take of every Zig |
| 13:48.35 | [TK]D-Fender | off* |
| 13:48.43 | phix | [TK]D-Fender: :) |
| 13:48.45 | [TK]D-Fender | FOR GREAT JUSTICE! |
| 13:48.54 | phix | [TK]D-Fender: cause some one send us the bomb |
| 13:49.09 | SeicherlBoB | [TK]D-Fender: but the provider will send additional extensions with the call... i dont think i can overwrite the extension that way. |
| 13:49.12 | [TK]D-Fender | YOU KNOW WHAT YOU ARE DOING |
| 13:49.40 | [TK]D-Fender | SeicherlBoB: register = user:pass@host/extensiongoeshere <------ |
| 13:49.40 | fc1979 | ok i'll come back with some news...i try to set an extension (in realtime configuration) with a fixed number i want call,1501. Now, when i make call,asterisk daemon die.The asterisk version is 1.6.1. Thii is the log: |
| 13:49.41 | fc1979 | http://pastebin.com/m206be13d |
| 13:49.41 | SeicherlBoB | when i call 555123 it arrives at exten 0, when i call 555123-99 it will come in on -99 |
| 13:49.44 | phix | [TK]D-Fender: hehe |
| 13:49.46 | [TK]D-Fender | seYou make it unique |
| 13:49.51 | [TK]D-Fender | SeicherlBoB: You make it unique |
| 13:50.13 | SeicherlBoB | [TK]D-Fender; I know... but when the caller calls extension 99 manually.. where does it go then? |
| 13:50.14 | leifmadsen | fc1979: the backtrace would be more useful if it is crashing |
| 13:50.32 | fc1979 | really, asterisk don't crash. |
| 13:50.35 | fc1979 | Simply stop. |
| 13:50.37 | leifmadsen | oh |
| 13:50.40 | leifmadsen | sounds like a deadlock |
| 13:50.57 | phix | fc1979: my computer crashes before asterisk does |
| 13:51.13 | phix | then again my TDM digium card crasheses first |
| 13:51.17 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 13:51.20 | leifmadsen | enable MALLOC_DEBUG in Compiler Flags in menuselect, 'make install', then when that happens, run 'core show locks' |
| 13:51.23 | phix | it doesnt like my wifi cad |
| 13:51.24 | phix | card |
| 13:51.38 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 13:51.44 | phix | teknoprep111 yaayayayya |
| 13:51.51 | teknoprep | woooooohoooooooooo |
| 13:52.00 | teknoprep | i said yaaaayaya |
| 13:52.01 | teknoprep | lol |
| 13:52.01 | phix | yayaya |
| 13:52.03 | phix | :D |
| 13:52.04 | phix | hehe |
| 13:52.09 | phix | sup? |
| 13:52.17 | teknoprep | nm... i llove how i am loved |
| 13:52.38 | fc1979 | i've only binary asterisk package,not the source. |
| 13:53.09 | phix | fc1979: use the source luke |
| 13:53.28 | leifmadsen | fc1979: ahhh... you should use source :) |
| 13:53.40 | leifmadsen | goes off to get breakfast... FINALLY |
| 13:54.12 | phix | fc1979: the source luke the source!!!!1!! |
| 13:54.28 | fc1979 | what's the meaning of source luke? |
| 13:54.35 | phix | I am watching the sheild, it is great |
| 13:54.44 | mmlj4 | fc1979: it's from star wars, mainly |
| 13:54.48 | teknoprep | so i have a question about the 79xx series phone and the 7914 expansion module... whats the "best" way to get the 7914 to work ? |
| 13:55.00 | fc1979 | :) |
| 13:55.01 | phix | I am going to murder my next door neigbours dog, or at least open the gate so it can fuck off |
| 13:55.03 | teknoprep | can i have the 7970 firmware with SIP and the 7914 Firmware with SCCP ? |
| 13:55.10 | mmlj4 | and the fact that this is open-source software... you can read the source code yourself, to see how stuff works |
| 13:55.14 | [TK]D-Fender | teknoprep: last I checked it only worked when you used the phone with SCCP |
| 13:55.35 | teknoprep | [TK]D-Fender, the 7914 has SIP firmware now |
| 13:55.44 | teknoprep | [TK]D-Fender, but you have to use SIP TCP |
| 13:55.53 | phix | I like TCP |
| 13:56.05 | teknoprep | yeah it works with M$ OCS |
| 13:56.40 | phix | yay |
| 13:57.07 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 13:57.58 | teknoprep | hmm |
| 13:58.50 | phix | <3 |
| 13:59.40 | mmlj4 | </3 |
| 14:00.12 | teknoprep | does switchvox use asterisk 1.6 yet ? |
| 14:00.40 | [TK]D-Fender | teknoprep: Extremely doubtful |
| 14:00.41 | phix | mmlj4: <3<3<3<#<#<#<#<# |
| 14:00.53 | leifmadsen | teknoprep: not to my knowledged |
| 14:01.50 | mmlj4 | </3 M$ |
| 14:01.51 | *** join/#asterisk rgsteele||work (n=rgsteele@75.147.74.137) |
| 14:02.02 | phix | <3 mmlj4 |
| 14:02.03 | *** part/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 14:02.11 | SeicherlBoB | [TK]D-Fender: when i set the extension myself, how can i check on the extension the caller dialed himself? where does it go then? |
| 14:02.16 | mmlj4 | get away from me, you freak :-) |
| 14:03.01 | [TK]D-Fender | SeicherlBoB: you register for multiple DID's correct? As in each registration is for a unique DID that you pay them for, no? |
| 14:03.16 | rgsteele||work | Hey folks. Can you can up a call/kill an open SIP connection? Asterisk thinks a certain extension is on the phone, when the phone is powered off right next to me :) |
| 14:03.28 | rgsteele||work | s/can up a call/hang up a call/ |
| 14:04.10 | mmlj4 | rgsteele||work: you might find that your extensions.conf or sip.conf is pointing to the wrong phone |
| 14:04.18 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:04.20 | [TK]D-Fender | rgsteele||work: "soft hangup [channel]" |
| 14:04.26 | *** part/#asterisk blackmousepad (n=blackmou@intranet.shantycreek.com) |
| 14:04.29 | SeicherlBoB | [TK]D-Fender: I have multiple real phonenumbers |
| 14:04.34 | [TK]D-Fender | mmlj4: No impact |
| 14:04.43 | mmlj4 | aye |
| 14:04.53 | [TK]D-Fender | SeicherlBoB: And each reg represents one number, correct? |
| 14:05.01 | SeicherlBoB | [TK]D-Fender: right. |
| 14:05.09 | [TK]D-Fender | mmlj4: Not a config error, the call is simply hung |
| 14:05.24 | SeicherlBoB | [TK]D-Fender: but i dont get the number from my provider, i only get the extension |
| 14:05.32 | [TK]D-Fender | SeicherlBoB: then you should do "/thenumbertheregrepresents" at the end instead of "/0" |
| 14:05.45 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 14:05.45 | *** part/#asterisk zepolen (n=zepolen@adsl89-1.kln.forthnet.gr) |
| 14:05.55 | [TK]D-Fender | SeicherlBoB: And then you make an exten in that common context for each DID you are paying them for and set in your reg's |
| 14:06.04 | SeicherlBoB | [TK]D-Fender: so when my number is 555123 i make it "/555123" |
| 14:06.11 | [TK]D-Fender | SeicherlBoB: Yes |
| 14:06.41 | [TK]D-Fender | SeicherlBoB: then you do "exten => 555123,1,DoSomethingHerLikeMaybeJumptToAnotherContextEtc |
| 14:06.52 | SeicherlBoB | [TK]D-Fender: ok, I'll try if that works... I think i already did so in the first place, but maybe I messed up with something |
| 14:07.16 | [TK]D-Fender | SeicherlBoB: If you are using "/0" already then that works and a unique number each should as well |
| 14:07.22 | mmlj4 | ah. |
| 14:07.38 | SeicherlBoB | [TK]D-Fender: ah... ok . I could make it jump and only send the extension of the number (dialed by the caller) .... ah |
| 14:07.55 | SeicherlBoB | ok, i'll try that. |
| 14:08.04 | rgsteele||work | [TK]D-Fender: Thanks, worked like a charm. |
| 14:08.10 | *** join/#asterisk usam (n=alx@193.214.109.195) |
| 14:08.17 | [TK]D-Fender | rgsteele||work: You're welcome |
| 14:08.49 | phix | leo!! |
| 14:09.03 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:09.34 | usam | any1 with ideas for how I can manipulate the LED on SIP phones with asterisk? .. We would like to have the LED on if the agent is logged in' |
| 14:09.50 | [TK]D-Fender | usam: WHAT LED? |
| 14:09.58 | [TK]D-Fender | usam: What phone? |
| 14:10.02 | usam | polycom |
| 14:10.20 | [TK]D-Fender | usam: You can do something custom with : |
| 14:10.23 | [TK]D-Fender | ~devstate |
| 14:10.23 | jbot | [~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=- http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/, or http://www.asterisk.org/node/48325 |
| 14:11.08 | usam | Ok,... I will try do so |
| 14:11.09 | usam | thx |
| 14:12.07 | SeicherlBoB | [TK]D-Fender: now i remember why that did not work.... the PBX is not answering at all... on verbosity 5 i got no reaction at all |
| 14:12.26 | [TK]D-Fender | SeicherlBoB: pastebin a failed call with SIP debug. |
| 14:12.28 | [TK]D-Fender | ~pb |
| 14:12.29 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 14:12.46 | SeicherlBoB | [TK]D-Fender: what do you mean "a failed call"? |
| 14:12.49 | riddlebox | Katty: you awake? |
| 14:13.01 | *** join/#asterisk stoffell (n=kristof@220.41-201-80.adsl-dyn.isp.belgacom.be) |
| 14:13.09 | [TK]D-Fender | SeicherlBoB: place a call and pastebint he complete output with SIP DEBUG enabled so we can see what's happening. |
| 14:13.22 | SeicherlBoB | so you mean the sip debug... ok. |
| 14:13.52 | teknoprep | hey [TK]D-Fender do you have any programs you recommend for testing for where VoIP problems with RTP are probably comming from ? |
| 14:14.09 | [TK]D-Fender | teknoprep: nope |
| 14:15.28 | SeicherlBoB | [TK]D-Fender: although i have put an extension (not 0) at the end of the register statement, It still tries the extension 0 in the context. |
| 14:15.42 | [TK]D-Fender | SeicherlBoB: PASTEBIN |
| 14:16.00 | SeicherlBoB | ok |
| 14:16.36 | phix | yaya bizlatz1 |
| 14:17.52 | jplank | [TK]D-Fender: my phone system doesn't work, I think it might be the flux capacitor, can you help? |
| 14:18.40 | SeicherlBoB | [TK]D-Fender: http://pastebin.com/d30537cbf |
| 14:18.41 | [TK]D-Fender | fries jplank with 1.21 Jiggawatts of electricity |
| 14:19.07 | merlin8282 | [TK]D-Fender: wasn't it 2,21 gigawatt ? |
| 14:19.14 | [TK]D-Fender | SeicherlBoB: I don't see your config in there... |
| 14:19.23 | [TK]D-Fender | merlin8282: Nope |
| 14:19.25 | SeicherlBoB | ... |
| 14:19.38 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-1c704a4c8c8440f5) |
| 14:19.48 | jplank | its gigawatts, just its pronounced jiggawatts :) |
| 14:20.00 | [TK]D-Fender | jplank: not quite actually. |
| 14:20.10 | merlin8282 | [TK]D-Fender: hmmm, funny... in the french version it's 2,21 GW ^^ |
| 14:20.25 | [TK]D-Fender | merlin8282: Math got lost in translation. |
| 14:20.34 | merlin8282 | :) |
| 14:20.51 | jplank | http://en.wikipedia.org/wiki/DeLorean_time_machine#.22Gigawatt.22 |
| 14:21.02 | aiksa[LV] | hmm, I was wondering - increasing "nice"ness of the asterisk process to lets say -2, what problems I might run into. |
| 14:21.05 | aiksa[LV] | ? |
| 14:21.11 | synthetiq | if you are using a digium card with built in echo can, and you run the asterisk command echo on that call, will the echo can cancel the echo() app? |
| 14:21.22 | [TK]D-Fender | jplank: Was already there... yeah, they caught the phonetic up elsewhere :) |
| 14:21.31 | jplank | :) |
| 14:22.19 | jplank | the only reason I knew that is I know guys who pronounce it jigawatts |
| 14:22.44 | [TK]D-Fender | jplank: and the only rason I do on occasion is because of BTTF :) |
| 14:22.59 | coppice | of course it should be pronounced jiggle watts |
| 14:23.06 | [TK]D-Fender | jplank: And both are supposedly acceptable |
| 14:23.09 | SeicherlBoB | [TK]D-Fender: http://pastebin.com/d8a42afc |
| 14:23.13 | jplank | so Ive read |
| 14:23.15 | teknoprep | jiggle what .. moron |
| 14:23.17 | phix | hi |
| 14:23.21 | jplank | lol |
| 14:23.51 | teknoprep | watt is a NYC what |
| 14:23.55 | phix | sup |
| 14:23.59 | [TK]D-Fender | coppice: Sign on operahouse door : "The Handel is Baroquen. If I wiggle Handel, will Handel wiggle Bach?" |
| 14:24.01 | jplank | synthetiq: I'm not 100% sure, but I heard its not good to have hardware echo and software echo on at the same time |
| 14:24.20 | jplank | actually its wut wut |
| 14:24.26 | coppice | well, more people pronounce giga correctly than pronounce cepstrum correctly |
| 14:24.36 | mmlj4 | sep-strum |
| 14:24.44 | jplank | +1 |
| 14:24.45 | mmlj4 | no? |
| 14:24.50 | coppice | kep-strum |
| 14:25.05 | jplank | damn english and its crazy rules |
| 14:25.20 | coppice | unless you pronounce spectrum as spestrum :-\ |
| 14:25.35 | mmlj4 | cello |
| 14:25.39 | [TK]D-Fender | celestial |
| 14:25.47 | [TK]D-Fender | celery |
| 14:26.24 | *** join/#asterisk propellerhead (n=yogurt2u@host108.190-30-184.telecom.net.ar) |
| 14:26.37 | phix | propellerhead you suck |
| 14:26.51 | [TK]D-Fender | however : http://en.wikipedia.org/wiki/Cepstrum |
| 14:27.28 | [TK]D-Fender | Which offers a real reason to understand why coppice knows best |
| 14:27.55 | phix | [TK]D-Fender: <3 |
| 14:27.57 | propellerhead | thanx phix |
| 14:28.01 | phix | you are so wise |
| 14:28.33 | phix | propellerhead you represent a band? |
| 14:28.50 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 14:28.52 | propellerhead | nop |
| 14:29.17 | propellerhead | I don't like the band |
| 14:30.15 | [TK]D-Fender | propellerhead: Gota beanie? :) |
| 14:30.18 | phix | great |
| 14:30.26 | phix | propellerhead: in that case your nick is stupid |
| 14:30.36 | phix | propellerhead: change it to LambOfGod :) |
| 14:30.50 | [TK]D-Fender | phix: no, could well be in reference to a "propeller beanie"" |
| 14:30.52 | phix | if your nick is going to mimic a band it may as well be a good one :P |
| 14:31.13 | phix | [TK]D-Fender: <3 |
| 14:31.20 | phix | like i said, you are wise |
| 14:31.26 | propellerhead | phix I used this nick from 1998.... |
| 14:32.53 | phix | propellerhead: and? |
| 14:33.57 | phix | lol |
| 14:34.17 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 14:34.26 | keith4 | according to wikipedia, the propellerheads released their first album in 1996 |
| 14:35.21 | phix | keith4: offically |
| 14:35.37 | phix | keith4: they were around before that |
| 14:35.48 | keith4 | shrugs |
| 14:35.51 | keith4 | never head of 'em |
| 14:41.03 | SeicherlBoB | [TK]D-Fender: any suggestions? |
| 14:42.28 | mmlj4 | how much adoption has ael seen? |
| 14:43.30 | [TK]D-Fender | SeicherlBoB: register => user2:passwd2:user2@node5.voicenode.at:5060/0 ,-- you left this with 0 |
| 14:43.37 | [TK]D-Fender | SeicherlBoB: register => user2:passwd2:user2@node5.voicenode.at:5060/0 <-- you left this with 0 |
| 14:43.46 | SeicherlBoB | yes, but i did not call that number |
| 14:44.04 | SeicherlBoB | i called user1 |
| 14:44.23 | [TK]D-Fender | SeicherlBoB: did you completely restart *? |
| 14:44.43 | SeicherlBoB | [TK]D-Fender: ... hmmm... guess i reloaded dialplan and sip |
| 14:46.08 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 14:46.24 | [TK]D-Fender | SeicherlBoB: Well change BOTH reg's, and the only reason the call goes to * is because thats the last thing to provider was told to do |
| 14:46.36 | [TK]D-Fender | SeicherlBoB: You may have to wipe the SIP registry from astdb and reload |
| 14:47.04 | SeicherlBoB | how to "wipe the SIP registry from astdb" ? |
| 14:47.29 | mmlj4 | ok, lemme redirect... who here used ael? |
| 14:47.32 | [TK]D-Fender | SeicherlBoB: "help database" "database show" |
| 14:48.10 | SeicherlBoB | [TK]D-Fender; there is only the sip client and dundi stuff |
| 14:49.13 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 14:49.35 | SeicherlBoB | [TK]D-Fender: should there be more? |
| 14:50.48 | [TK]D-Fender | SeicherlBoB: change both of your register statemtns and reload SIP, check "sip show registry" |
| 14:50.59 | SeicherlBoB | ok |
| 14:52.51 | SeicherlBoB | ok. changed all registries and reloaded. registry looks good |
| 14:53.47 | *** join/#asterisk Snublefot_Wrk (n=kjell@89.191.9.147) |
| 14:54.58 | SeicherlBoB | ok... debugging just flooded my console completely |
| 14:55.17 | phix | [TK]D-Fender: what time is it there? |
| 14:55.22 | phix | -15? |
| 14:55.32 | phix | 11AM? |
| 14:58.33 | [TK]D-Fender | phix: EST |
| 15:00.48 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 15:02.24 | SeicherlBoB | [TK]D-Fender: ok... now i get an attempt to get extension 0 in a context where 0 is not defined, so it answers to the provider with an "address incomplete" which leads to another invite with the given number but in the default context... |
| 15:02.38 | SeicherlBoB | this is really sick! |
| 15:02.57 | [TK]D-Fender | SeicherlBoB: your provider is still remembering the "0" you put in there before. they did not make this up themselves. |
| 15:03.09 | SeicherlBoB | maybe... |
| 15:03.10 | propellerhead | sorry phix, meetings |
| 15:03.15 | [TK]D-Fender | SeicherlBoB: Not sure what you'll have to do to speed up re-registration |
| 15:03.26 | SeicherlBoB | but i get some strange behaviour on register commands |
| 15:03.55 | *** join/#asterisk dellnec (n=sysadmin@mail.controlsg.com) |
| 15:04.08 | SeicherlBoB | the provider sends "SIP/2.0 401 Unauthorized" |
| 15:04.40 | dellnec | Question. Does anyone know how to activate the speakerphone a sip phone when the user hits the number nine? |
| 15:04.51 | [TK]D-Fender | SeicherlBoB: you've messed something else up then because the exten merely changes the "contact" header in the invite and has no impact on auth |
| 15:05.08 | [TK]D-Fender | dellnec: thats up to your PHONE to do. |
| 15:05.30 | [TK]D-Fender | dellnec: And i cannot even imagine this being created by anyone |
| 15:05.52 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.24.187) |
| 15:06.24 | [TK]D-Fender | dellnec: You have a speakerphone button on that phone already. use it |
| 15:06.49 | dellnec | i want the phone to open up when the user dials nine to get an outside line |
| 15:08.58 | SeicherlBoB | [TK]D-Fender: ok. this now works... but the calls are placed to the default context defined in the general section. |
| 15:09.11 | merlin8282 | bye |
| 15:09.15 | *** join/#asterisk dvdevel (n=devel@wiggum.digitalcoven.com) |
| 15:09.17 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:09.20 | *** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-12-127.w90-56.abo.wanadoo.fr) |
| 15:09.58 | [TK]D-Fender | SeicherlBoB: And i don't see your updated configs or your failed call with SIP debug |
| 15:10.11 | [TK]D-Fender | dellnec: Not happening. |
| 15:10.31 | dellnec | does anyone know of a sip phone that will activate the speakerphone when the user dials nine to get an outside line? |
| 15:10.42 | [TK]D-Fender | dellnec: So far I can pretty much say NONE |
| 15:10.49 | dellnec | kk |
| 15:11.06 | [TK]D-Fender | dellnec: Want speakerphone, press the BUTTON for it. |
| 15:11.10 | SeicherlBoB | [TK]D-Fender: it works better, now that i fixed the registration string. |
| 15:11.16 | [TK]D-Fender | dellnec: No phone will change mode jsut because of a digit you dial |
| 15:11.34 | dellnec | digital phones work like that... |
| 15:11.50 | SeicherlBoB | [TK]D-Fender: when I call the number, the call is forwarded to the default context with the extension i defined. |
| 15:12.02 | dellnec | my users will have to cope with this |
| 15:12.30 | rue_mohr | ok, anyone know how to turn off this freaking sip debug info? |
| 15:12.33 | dellnec | just out of curiosity what flavor of asterisk to you run? |
| 15:12.46 | dellnec | do * |
| 15:13.37 | rue_mohr | you know what would be cool for the CLI is to have a 'capture on' 'capture off' and 'search capture' features |
| 15:13.37 | SeicherlBoB | [TK]D-Fender: but when I call the number + an extension, this is not working. lets say, i call 555123 => works; but when i call 555123-99 I have an incoming call on exten 99 (and not 55512399) so i have the same problem as before. I can not make a difference between 555123-99 and 555999-99... the same problem as i had with the -0 extension |
| 15:15.12 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:15.29 | *** join/#asterisk surfdue (n=anthonyu@unaffiliated/surfdue) |
| 15:15.32 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:15.36 | surfdue | Long time no talk! |
| 15:15.56 | [TK]D-Fender | dellnec: Sweet&Sour Edition |
| 15:16.16 | tzafrir_laptop | hmm... when snom phones are sccidentally told to use pxe boot they boot from the network? |
| 15:16.20 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:16.27 | [TK]D-Fender | SeicherlBoB: Where's my pastebin? |
| 15:16.31 | dellnec | TK]D-Fender> dellnec: Sweet&Sour Edition |
| 15:16.33 | dellnec | [11:16] <t |
| 15:16.35 | surfdue | So how is asterisk doing :} |
| 15:17.43 | SeicherlBoB | omg... ok |
| 15:17.44 | dellnec | [TK]D-Fender: what is that? |
| 15:17.45 | rue_mohr | ~pb |
| 15:17.46 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:17.58 | *** join/#asterisk InfoNutz (n=what@204.50.209.225) |
| 15:18.00 | rue_mohr | you know what would be cool for the CLI is to have a 'capture on' 'capture off' and 'search capture' features |
| 15:18.04 | [TK]D-Fender | dellnec: Sarcasm. there are only 2 "flavours" of Asterisk. OSE & ABE. |
| 15:18.12 | tzafrir_laptop | snom 320 |
| 15:18.21 | SeicherlBoB | [TK]D-Fender: which call? the +99 call? |
| 15:18.26 | SeicherlBoB | or the working one |
| 15:18.42 | [TK]D-Fender | dellnec: and ABE is just an outdated backported version with some closed source stuff tacked on. |
| 15:18.44 | dellnec | [TK]D-Fender now im lost....i just bought a sangoma t1 card looking to install trixbox...im on the fence about it |
| 15:18.54 | rue_mohr | [TK]D-Fender, any more ideas on my persistant sip debug info? |
| 15:18.56 | [TK]D-Fender | dellnec: trixbox is not a version of * |
| 15:19.00 | surfdue | Does anyone know any of these? 1. a cheap/good toll-free number provider for Asterisk (sip forwarding), or 2. a free (sip forwarding) |
| 15:19.07 | [TK]D-Fender | dellnec: it is a garbage distro that just happens to INCLUDE IT |
| 15:19.17 | [TK]D-Fender | rue_mohr: upgrade |
| 15:19.28 | *** join/#asterisk JimmyTrivette (n=JimmyTri@75.76.165.34) |
| 15:19.29 | rue_mohr | [TK]D-Fender, what you talking about!? |
| 15:19.41 | dellnec | [TK]D-Fender which distro is good with sangoma? |
| 15:19.53 | rue_mohr | my understanding is that 1.6 is not reliable for production right now |
| 15:19.55 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:20.05 | [TK]D-Fender | rue_mohr: Its a single self-explanitory word. You seem to get debug you shouldn't Could be a bug. UPGRADE |
| 15:20.19 | [TK]D-Fender | rue_mohr: What version are you on? |
| 15:20.23 | rue_mohr | and, I just spent over a month getting it working as is |
| 15:20.30 | InfoNutz | hello all, if i were to keep a voicemail.conf entry but delete the voicemails that are associated with it what would be the shell cli for do do that? |
| 15:20.35 | rue_mohr | Asterisk 1.4.22 built by root @ debian on a i686 running Linux on 2009-03-17 19:03:01 UTC |
| 15:20.42 | [TK]D-Fender | rue_mohr: UPGRADE |
| 15:20.58 | rue_mohr | to what for production? |
| 15:21.07 | [TK]D-Fender | InfoNutz: go wipe out the files inyour voicemail folder |
| 15:21.16 | [TK]D-Fender | rue_mohr: Look in the damn topic |
| 15:21.33 | [TK]D-Fender | facepalms |
| 15:21.38 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:21.41 | dellnec | [TK]D-Fender: does wanpipe work with asterisk |
| 15:21.47 | *** join/#asterisk thelordmortis (n=lordmort@124-169-248-10.dyn.iinet.net.au) |
| 15:21.49 | [TK]D-Fender | dellnec: Yes. |
| 15:22.02 | [TK]D-Fender | dellnec: Zaptel/dahdi more specifically |
| 15:22.03 | dellnec | [TK]D-Fender: so my sangoma t1 card will work |
| 15:22.12 | rue_mohr | those may be the lastest versions, dosn't mean their appropriot for a system that needs to work |
| 15:22.14 | [TK]D-Fender | dellnec: Its its their AFT series, yes |
| 15:22.34 | dellnec | [TK]D-Fender: how would i be able to tell? |
| 15:22.52 | [TK]D-Fender | rue_mohr: You have an issue, FFS get off your ass and stop being the problem. |
| 15:23.02 | [TK]D-Fender | dellthe MODEL NAME perhaps? |
| 15:23.24 | rue_mohr | wow, sorry, I was just asking if anyone had thought why the sip debug info wouldn't turn off |
| 15:23.27 | *** join/#asterisk Chex (n=Stefan@bas1-montreal48-1176173096.dsl.bell.ca) |
| 15:23.43 | [TK]D-Fender | rue_mohr: I do. Looks like a BUG. Go try and do something about it now. |
| 15:23.50 | dellnec | [TK]D-Fender: A102DE |
| 15:24.05 | [TK]D-Fender | dellnec: Yes, it will work |
| 15:24.06 | rue_mohr | I didn't look to see if its a known bug cause I dont care, this system took me about a month to get togethor and I'm not about to do something STUPID like overhaul the entire thing now |
| 15:24.20 | rue_mohr | cause it would take me another month to get it all working again |
| 15:24.29 | dellnec | [TK]D-Fender: you sound very confident in *. i will give it a try |
| 15:24.33 | [TK]D-Fender | rue_mohr: And you are blowing the size of a minor recompile out of proportion. |
| 15:24.40 | rue_mohr | I know just with the drivers alone there would be a few weeks work trying to get them to work |
| 15:24.52 | [TK]D-Fender | rue_mohr: Don't whine about the problem and then refuse to follow suggestions to fix it. |
| 15:25.14 | [TK]D-Fender | rue_mohr: And * has nothing to do with your drivers. |
| 15:25.16 | rue_mohr | I wasn't whining, I asked if anyone had ideas on why it wouldn't turn off |
| 15:25.28 | rue_mohr | the dahdi drivers |
| 15:25.29 | [TK]D-Fender | rue_mohr: that is a separate layer that you don't need to upgrade along-with |
| 15:25.52 | rue_mohr | I have NO confidence I woulnd't have to rebuild everything from the ground up |
| 15:25.56 | rue_mohr | none |
| 15:26.37 | InfoNutz | where is the collection of voicemails for each number on asterisk? |
| 15:26.40 | [TK]D-Fender | rue_mohr: http://tinyurl.com/2t5chy |
| 15:26.47 | InfoNutz | lol |
| 15:26.48 | rue_mohr | is there any reason I cant start a dialplan with #? |
| 15:27.04 | [TK]D-Fender | InfoNutz: typically /var/spool/asterisk/voicemail |
| 15:27.17 | [TK]D-Fender | rue_mohr: clarify "a dialplan" |
| 15:27.29 | InfoNutz | thank you sir |
| 15:27.34 | InfoNutz | you have finished my project for me |
| 15:27.37 | rue_mohr | ~pb |
| 15:27.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 15:27.55 | rue_mohr | that comma is annoying |
| 15:28.44 | rue_mohr | http://rafb.net/p/YMAvQR31.html |
| 15:28.48 | rue_mohr | see line 67 |
| 15:29.09 | [TK]D-Fender | rue_mohr: thats fine |
| 15:29.22 | *** join/#asterisk chrisb (n=chrisb@98.111.176.88) |
| 15:30.41 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 15:30.55 | dni | How many bytes does a SIP call utilize ?? If i remember correcttly its about 40k ber sip call ? |
| 15:31.19 | dni | **g.729 |
| 15:31.33 | dni | but i would like to know ulaw |
| 15:32.09 | [TK]D-Fender | dni: 85kbit |
| 15:32.23 | chrisb | is there a teleconference provider using asterisk? |
| 15:32.40 | [TK]D-Fender | chrisb: Certainly a few of them |
| 15:32.53 | rbd | easy question... what's the easiest way to get the sending IP of a SIP message (e.g. the same IP as what's sent in the "<--- SIP read from 216.82.224.202:5060 --->" line of the sip debugs)...would I just be best parsing it out from the SIP channel name, or is there a better way? |
| 15:32.56 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 15:33.58 | rue_mohr | aha, its just the aastra that wont do it, wont dial * or # alone |
| 15:34.33 | [TK]D-Fender | rue_mohr: Yup, stupid dialplan reserves # AS END-OF-DIAL |
| 15:35.05 | rue_mohr | hmm ok yea the polycom will do * and not #, k |
| 15:35.17 | [TK]D-Fender | rue_mohr: No, polycom can do both |
| 15:35.19 | rue_mohr | thats local phone dialplan |
| 15:35.25 | [TK]D-Fender | Polycom > all |
| 15:35.29 | [TK]D-Fender | rue_mohr: Yes, it CAN. |
| 15:35.32 | rue_mohr | hmm |
| 15:35.38 | rue_mohr | I wonder what failed |
| 15:35.42 | rue_mohr | I'll try again |
| 15:35.47 | mltlnx | Hello, So.... our asterisk server started to hang up on callers when they are checking "old" messages in voicemail |
| 15:35.53 | mltlnx | any ideas? |
| 15:36.03 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:36.15 | [TK]D-Fender | mltlnx: Perhaps a corrupted messaage? |
| 15:36.43 | mltlnx | Thats what I though....but it seems to be happening to everyones "Old" folder. |
| 15:36.54 | chrisb | [TK]D-Fender: bargainconference.com is offline |
| 15:37.08 | [TK]D-Fender | mltlnx: What ver of *? |
| 15:37.18 | mltlnx | 1.4.12 |
| 15:37.23 | [TK]D-Fender | ANCIENT |
| 15:37.23 | chrisb | [TK]D-Fender: i'm looking for a teleconference service provider |
| 15:37.37 | [TK]D-Fender | chrisb: Why not run your own? |
| 15:37.38 | *** join/#asterisk whitehat (n=whitehat@unaffiliated/whitehat) |
| 15:37.49 | [TK]D-Fender | chrisb: and they needn't run * if you are outsourcing anyways |
| 15:37.51 | rue_mohr | http://rafb.net/p/mcEhlh67.html |
| 15:38.00 | rue_mohr | can you tell if the phone tried to dial #? |
| 15:38.06 | rue_mohr | I dont see it |
| 15:38.32 | mltlnx | not too old |
| 15:38.49 | whitehat | hello group. I just download asterisk-gui from svn and followed all instructions. I logged in and it's apparnetly in a lop in channel.c unable to request channel local/executecommand@asterisk_quitools. ideas? thank you |
| 15:39.51 | dellnec | [TK]D-Fender: asterisknow is that good? |
| 15:40.13 | [TK]D-Fender | dellnec: For what? |
| 15:40.39 | dellnec | [TK]D-Fender: for an install for a mid size company 50 users with a sangoma card? |
| 15:41.08 | [TK]D-Fender | dellnec: Its a DSITRO once again, it depends how you configure your system |
| 15:41.16 | *** join/#asterisk javar (n=javar@190.157.100.110) |
| 15:41.18 | rbd | no one knows the best way to get a SIP message sending IP as an asterisk variable? |
| 15:41.21 | [TK]D-Fender | dellnec: Your choice does not particularly matter |
| 15:41.26 | chrisb | finds [TK]D-Fender 's responses orthogonal |
| 15:41.30 | *** part/#asterisk javar (n=javar@190.157.100.110) |
| 15:41.34 | [TK]D-Fender | rue_mohr: No, it is clearly sending no exten and defaulting to "s" |
| 15:41.35 | rue_mohr | I dont suggest the digium card, I had all sorts of echo problems, the sangoma cards are apparently ok out of the box |
| 15:41.42 | rue_mohr | hmm |
| 15:41.49 | dellnec | [TK]D-Fender: i have installed trixbox and it seems to be fine... |
| 15:41.51 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 15:42.06 | [TK]D-Fender | dellnec: Yeah, fine, whatever |
| 15:42.25 | dellnec | [TK]D-Fender: you seem to be the expert...whats your favorite distro? |
| 15:43.06 | [TK]D-Fender | dellnec: I use CentOS personally, Slackware in the past, many use Debian, etc, but jsut about anything will do if you're competant to install *'s prerequisites |
| 15:44.20 | dellnec | [TK]D-Fender: thats linux...what about full installs like trixbox, pbx in a flash, etc? anyone that you like? |
| 15:44.30 | dellnec | trixbox runs on centos |
| 15:44.33 | [TK]D-Fender | dellnec: Surprise... that LINUX |
| 15:45.14 | *** join/#asterisk NOT_guru (n=NOT_guru@24-241-103-142.static.stls.mo.charter.com) |
| 15:45.16 | [TK]D-Fender | dellnec: My impression of the pre-made+Asterisk distros : FUCK 'EM |
| 15:45.33 | rue_mohr | heh, the things I havn't done yet |
| 15:45.53 | dellnec | [TK]D-Fender10-4 |
| 15:47.09 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 15:47.10 | *** join/#asterisk littlerock (n=littlero@58.31.43.131) |
| 15:47.12 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.27 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 15:47.28 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 15:48.47 | rue_mohr | huh sipdebug was on in sip.conf... |
| 15:48.55 | rue_mohr | wonder if that would have done it |
| 15:50.28 | littlerock | anyone here use fax functionality of asterisk ? |
| 15:51.01 | whitehat | hello group. I'm trying to login to the gui and receiving "creating a config file to store gui perferences" in a loop. ideas? |
| 15:51.29 | *** part/#asterisk stmaher (n=stephen@mateus.province5.tv) |
| 15:51.41 | rue_mohr | hey [TK]D-Fender |
| 15:51.46 | rue_mohr | guess what? |
| 15:52.12 | rue_mohr | seems the cli cant override the setting in sip.conf for turning debug messages off |
| 15:52.38 | rue_mohr | hah |
| 15:53.33 | rue_mohr | ok, anyhow, for my speeddial system, am I right that I need to use dial to get the dahdi channel connected and then use ... senddigits to do the dialing on the line? |
| 15:53.47 | rue_mohr | dosn't dial connect the call through? |
| 15:55.00 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:55.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:55.05 | rue_mohr | I must be wrong, dial does everything, one of these options must do it |
| 15:55.35 | rue_mohr | [TK]D-Fender, that would have been funny for me to have spent hours recompiling asterisk and had that same debug message problem, hah |
| 15:55.42 | littlerock | I hope the user in front of the desktop can call me from a my website, I mean he doesnt need to install any VoIP softphone, except having a web browser, any resources about this ? |
| 15:55.52 | littlerock | s/this/it |
| 15:55.55 | *** join/#asterisk mizery (n=mizeryde@CPE-65-27-107-31.new.res.rr.com) |
| 15:56.38 | rue_mohr | aha |
| 15:57.59 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 15:59.36 | *** join/#asterisk t_corr (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:00.22 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 16:00.24 | bougyman | i'm having the hardest time finding a voip provider. |
| 16:00.26 | rue_mohr | if I say dial(zap/1/12345&zap/2/12345) then only the first available channel will actaully be dialed, right? |
| 16:00.46 | rue_mohr | bougyman, I'll do it for $1400/mo |
| 16:00.54 | t_corr | Hi all, I'm fixing my system's 911 handling and I found this example on the voip-info wiki: http://pastebin.com/m3702fe6d |
| 16:00.54 | tzafrir_laptop | rue_mohr, use groups |
| 16:01.00 | bougyman | hah. |
| 16:01.03 | rue_mohr | ok |
| 16:01.13 | bougyman | rue_mohr: with 500 inbound dids and 1m outbound minutes? |
| 16:01.15 | tzafrir_laptop | in zapata.conf, put both in, say, group=4 |
| 16:01.18 | t_corr | On lines 13 and 17, what exactly is the s+2 supposed to do? |
| 16:01.23 | bougyman | (unlimited inbound on the 500 dids) |
| 16:01.24 | t_corr | Shouldn't that be n+2? |
| 16:01.30 | tzafrir_laptop | and dial: Zap/g4/12345 |
| 16:01.30 | rue_mohr | tzafrir_laptop, but it will only do the first one thats available right? |
| 16:01.43 | rue_mohr | bougyman, 23 channels |
| 16:02.00 | tzafrir_laptop | yes, the first available will be dialed |
| 16:02.02 | bougyman | oh, so you mean 3x the price of a t1? |
| 16:02.06 | rue_mohr | tzafrir_laptop, perfect |
| 16:02.12 | ajohnson | s+2 would be the same priority as the line above +2 |
| 16:02.16 | rue_mohr | bougyman, 1.5x what we pay for a t1 |
| 16:02.21 | rue_mohr | :) |
| 16:02.31 | bougyman | we pay $416 for T1s |
| 16:02.42 | rue_mohr | we pay about $1100 |
| 16:02.42 | t_corr | ajohnson: I thought s would be the extension? |
| 16:02.59 | rue_mohr | ok groups, this might make more of this easier |
| 16:03.19 | ajohnson | sorry, maybe I didn't read enough. s can also be a priority though |
| 16:04.01 | t_corr | Hmm. I don't understand it. It looks like you could change the s+2 to n since both lines have (labels) and are referenced by them |
| 16:04.34 | Qwell | bougyman: at 1m minutes, you might as well talk to Level3 |
| 16:05.48 | rue_mohr | tzafrir_laptop, I dont think the groups will work, the lines are all for different businesses |
| 16:06.14 | tzafrir_laptop | rue_mohr, you want them all to ring? |
| 16:06.20 | tzafrir_laptop | or just one? |
| 16:06.33 | rue_mohr | no its for outgoing speed dials |
| 16:06.49 | rue_mohr | the outgoing line dosn't matter, but I only want it to choose the first available one |
| 16:07.11 | bougyman | Qwell: heh, broadvox says 1m minutes doesn't even qualify for wholesale. |
| 16:07.15 | tzafrir_laptop | still does not understand why groups won't work |
| 16:07.35 | rue_mohr | maybe I'm misunderstanding how groups are set up? |
| 16:08.22 | rue_mohr | aren't the groups set up in zaptel.conf (dahdi.conf) |
| 16:08.46 | tzafrir_laptop | with groups (g, G, r, R) asterisk dials on one specific line. It chooses it among the lines in that group. The first one available |
| 16:08.58 | tzafrir_laptop | The letter only sets the scanning order |
| 16:09.04 | rue_mohr | and their defined in? |
| 16:10.32 | rue_mohr | hey now, please tell |
| 16:10.37 | rue_mohr | cant leave me in suspense |
| 16:10.50 | rue_mohr | cant find it in dahdi/system.conf |
| 16:11.04 | rue_mohr | and there are no relivent hits in THE BOOK |
| 16:11.09 | tzafrir_laptop | g: first one (from bottom). G - first one (from top) |
| 16:11.23 | rue_mohr | but where are the groups defined? |
| 16:11.32 | tzafrir_laptop | r - round robin (asccending order), R - round robin (decending order) |
| 16:11.37 | tzafrir_laptop | zapata.conf |
| 16:11.49 | tzafrir_laptop | group = 4 |
| 16:12.00 | tzafrir_laptop | before channel => 3 |
| 16:12.06 | rue_mohr | zapata.conf became which? chan_dahdi.conf? |
| 16:12.11 | tzafrir_laptop | means channel 3 is in group 4 |
| 16:12.26 | rue_mohr | aha |
| 16:12.35 | tzafrir_laptop | chan_dahdi.conf, yes (though if you use zaptel, chances are it is still zapata.conf) |
| 16:12.49 | rue_mohr | its fun converting between so many systems |
| 16:13.02 | rue_mohr | okok, so I assign channels to a group then |
| 16:13.14 | rue_mohr | I dont assign a group channels |
| 16:14.26 | rue_mohr | so I can go group 1,2, 4, 7 ? |
| 16:15.20 | *** join/#asterisk zaafouri (i=c4cb33ee@gateway/web/ajax/mibbit.com/x-3614dd4c6370d765) |
| 16:16.00 | rue_mohr | huh, looks like THE BOOK skips groups |
| 16:17.53 | tzafrir_laptop | a channel can be in any number of groups |
| 16:18.13 | rue_mohr | right, and its assigned on a per-channel basis |
| 16:18.14 | tzafrir_laptop | (well, you only have 64 groups to begin with) |
| 16:18.19 | rue_mohr | ok |
| 16:18.28 | tzafrir_laptop | group = 4,30 |
| 16:18.30 | rue_mohr | so can I use comma seperated? |
| 16:18.32 | rue_mohr | ok |
| 16:18.35 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-58-80.w86-215.abo.wanadoo.fr) |
| 16:19.29 | whitehat | hello group. my asterisk-gui is in a loop. displaying verifying dialplan contexts needed for gui....creating a config file to store gui preferences. ideas? thank you. |
| 16:19.32 | *** join/#asterisk angryuser_ (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 16:19.53 | rue_mohr | dont usea gui, sorry |
| 16:20.46 | *** join/#asterisk tawker (n=ahuman@wikipedia/Tawker) |
| 16:23.41 | angryuser_ | . |
| 16:26.50 | kaldemar | whitehat: #asterisk-gui |
| 16:28.01 | rue_mohr | tzafrir_laptop, do you know how I set what order the channels are filed in the group? |
| 16:28.31 | whitehat | kaldemar: thank you. didn't know that channel existed. |
| 16:28.44 | tzafrir_laptop | rue_mohr, I know of no such way. But how many channels do you have in that group? |
| 16:28.52 | rue_mohr | 3 |
| 16:29.10 | rue_mohr | I need to make sure they are tried in a particular order |
| 16:29.58 | *** join/#asterisk Amazinzay (n=isaiah@67.108.187.186.ptr.us.xo.net) |
| 16:31.57 | *** join/#asterisk CieD (n=ciemon@unaffiliated/cied) |
| 16:32.57 | kaldemar | rue_mohr: http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels <-- see dialing a group |
| 16:33.10 | kaldemar | other than that, you have to make the logic in the dialplan channel by channel. |
| 16:35.29 | rue_mohr | ok so in the order they appear in I need to do 2, 1, 3 :) |
| 16:36.42 | rue_mohr | k, I can show a group in cli right? |
| 16:36.49 | rue_mohr | see what I have |
| 16:37.11 | rue_mohr | I dont have conflicting groups so I can re-order chan_dahdi if needbe |
| 16:37.47 | kaldemar | dial 2 first, then check what DIALSTATUS says and so on. |
| 16:37.51 | rue_mohr | oh dahdi dosn't have a group command |
| 16:38.49 | Amazinzay | 1626307 |
| 16:38.52 | rue_mohr | hmm cli knows NOTHING of dahdi groups... |
| 16:39.14 | rue_mohr | I cant dialthem cause the system isn't active cause I'm not done programming it yet |
| 16:40.35 | rue_mohr | group show channels |
| 16:40.59 | rue_mohr | or not |
| 16:41.14 | BlargMaN00 | I need some help with the devstate function, and a polycom phone... anyone have experience with that?? |
| 16:41.26 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-58-80.w86-215.abo.wanadoo.fr) |
| 16:41.32 | rue_mohr | BlargMaN00, not me |
| 16:41.50 | BlargMaN00 | rue_mohr: your no help... lol ;-)\ |
| 16:42.51 | rue_mohr | what are groups part of? |
| 16:44.13 | kaldemar | what do you mean by part of? |
| 16:44.21 | rue_mohr | ok I think groups are a writeoff, I need to specify the order |
| 16:44.45 | rue_mohr | well I need to see what channels are members of a group, but it seems there is no means for that |
| 16:45.04 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:45.24 | rue_mohr | I dont understand why there are functions for doing operations to groups in different orders if you cant specify the order of the channels in a group anyhow |
| 16:46.59 | rue_mohr | http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels |
| 16:47.12 | rue_mohr | "if you have defined channel group 2 as containing Zap channels 1, 2, 5 and 8," but it dosn't work like that |
| 16:47.20 | rue_mohr | you specify what groups a channel is in |
| 16:49.36 | rue_mohr | ok, after the * I need to get another digit to use for the speeddial lookup |
| 16:50.46 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 16:52.53 | rue_mohr | anyone know if there is an example that captures digits and plays them back to you? |
| 16:55.05 | rue_mohr | found it |
| 16:57.20 | tamiel | hello, what is exactly relaxdtmf option in chan_dahdi.conf ? |
| 16:57.47 | rue_mohr | iirc is widens the match paramiters incase its not detecting your dtmf right |
| 17:03.06 | *** join/#asterisk khronos (n=none@c-76-110-120-247.hsd1.fl.comcast.net) |
| 17:06.57 | *** part/#asterisk khronos (n=none@c-76-110-120-247.hsd1.fl.comcast.net) |
| 17:07.50 | *** join/#asterisk rvhi (n=chatzill@udp100194uds.hawaiiantel.net) |
| 17:08.36 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 17:09.44 | rue_mohr | http://bin.cakephp.org/view/1367421738 |
| 17:09.51 | rue_mohr | does that look reasonable? |
| 17:10.48 | rue_mohr | it must be it worked |
| 17:10.53 | rue_mohr | sweeeet |
| 17:12.27 | [TK]D-Fender | rue_mohr: What sip.conf option left it stuck? |
| 17:12.42 | rue_mohr | debug was on it there |
| 17:12.52 | [TK]D-Fender | rue_mohr: where? |
| 17:13.04 | rue_mohr | sipdebug = yes was what did it |
| 17:13.11 | [TK]D-Fender | SMRT |
| 17:13.23 | rue_mohr | I'm guessing that the cli just cant override it |
| 17:13.31 | [TK]D-Fender | rue_mohr: Congratulations on not needing to challenge your fear of minor upgrades. |
| 17:13.38 | rue_mohr | hah |
| 17:14.05 | rue_mohr | since when it 1.4 to 1.6 minor? |
| 17:14.09 | rue_mohr | :/ |
| 17:20.19 | masus | hi all, newbie question. is it possible "send a text" with asterisk to eyebeam's Display or a similar softphone? Thanks All. |
| 17:20.20 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:20.27 | masus | hi all, newbie question. is it possible "send a text" with asterisk to eyebeam's Display or a similar softphone? Thanks All. |
| 17:20.46 | *** join/#asterisk dvdevel (n=devel@wiggum.digitalcoven.com) |
| 17:20.55 | *** join/#asterisk stewbaby (n=stewart@ip-217-204-65-78.easynet.co.uk) |
| 17:21.08 | masus | sorry for writing twice ... |
| 17:21.35 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 17:25.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:26.25 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 17:29.48 | *** join/#asterisk aksyn (n=aksyn@94-193-98-124.zone7.bethere.co.uk) |
| 17:30.05 | t_corr | Hmm. Is the 'h' extension local to the context? So if I have a [emergency-services] with an 'h' extension within and I 'include =>' that in another context, will the 'h' then apply to the whole including context, or only to calls that end up matching extensions in the included context? |
| 17:33.29 | leifmadsen | only the context the call is in |
| 17:33.50 | leifmadsen | although I think it may try to fall back down through the contexts that it was included from to "find" an 'h' |
| 17:33.57 | leifmadsen | it's been a bit since I experimented with that |
| 17:34.18 | rue_mohr | ok I think their excited when I told them the phone system could do the "ITS A GREAT DAY AT.." intro on the phone calls before they pick them up |
| 17:37.40 | [TK]D-Fender | rue_mohr: From your 1.4.22 to 1.4.24 schmuck... |
| 17:37.47 | rue_mohr | oh |
| 17:37.57 | [TK]D-Fender | ~cluebat rue_mohr |
| 17:37.59 | jbot | ACTION pulls out a ClueBat (tm) and thwaps rue_mohr. |
| 17:38.02 | rue_mohr | wonder how I ended up with .22 |
| 17:38.13 | rue_mohr | I thought I was using .22 at home |
| 17:38.18 | [TK]D-Fender | rue_mohr: My guess would be "downloading" |
| 17:38.22 | rue_mohr | or maybe thats 1.2 |
| 17:38.41 | guax | rue_mohr, i swear i though you had a gun .22 in your home, then i searched for context =P |
| 17:39.06 | rue_mohr | hehe |
| 17:39.37 | rue_mohr | I object to having to walk to a phone to operate it |
| 17:41.58 | *** join/#asterisk stoffell (n=kristof@d51A4D629.access.telenet.be) |
| 17:44.04 | rue_mohr | ok I cant test this next line, so please look and see if it looks ok, it tires to get one of the pots lines in a particular order (chanels 6, then 5, then 7) |
| 17:44.06 | rue_mohr | http://bin.cakephp.org/view/805256122 |
| 17:44.17 | rue_mohr | line 6 in question |
| 17:44.32 | rue_mohr | ok I forgot a } |
| 17:44.45 | rue_mohr | anything else? |
| 17:44.54 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:45.09 | rue_mohr | will in try them in that order? |
| 17:45.21 | rue_mohr | or will it try them all at once and give me a random result? |
| 17:46.55 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 17:50.50 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 17:53.10 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-d5bdf9fd985bba88) |
| 17:54.09 | rue_mohr | whate hte app for making dtmf noise ? |
| 17:54.18 | rue_mohr | its not playtones, or dtmf |
| 17:54.25 | rue_mohr | not saydigits |
| 18:00.26 | rue_mohr | ARG, the dtmf I'm sending to the phone is being muted |
| 18:00.29 | rue_mohr | by someone |
| 18:00.40 | rue_mohr | -- Incoming call: Got SIP response 482 "Loop Detected" back from 192.168.1.2 ?? |
| 18:06.13 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-58-80.w86-215.abo.wanadoo.fr) |
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| 18:08.43 | rue_mohr | whats the name of the background app that isn't background cause it dosnt', seems to not be play |
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| 18:09.04 | rue_mohr | oh playback |
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| 18:13.36 | rue_mohr | ok if I want to user to enter a bunch of digits and store them I should have them end the string in a # and use a pattern of some sort I suppose, READ has no 'terminate on this character' control |
| 18:13.49 | [TK]D-Fender | rue_mohr: Horrible pile of devices to call that way. Setup a zaptel group for them |
| 18:14.06 | rue_mohr | yea but I cant control the order of the channels in the group |
| 18:14.20 | [TK]D-Fender | rue_mohr: and that dial does not control the order either |
| 18:14.26 | rue_mohr | right? |
| 18:14.27 | rue_mohr | oh |
| 18:14.29 | rue_mohr | hmm |
| 18:14.34 | [TK]D-Fender | rue_mohr: That would take 3 SEPARATE dials |
| 18:14.36 | rue_mohr | suggestions? |
| 18:14.39 | rue_mohr | k |
| 18:14.53 | [TK]D-Fender | rue_mohr: and READ ends on # or a length you determine |
| 18:14.58 | rue_mohr | so I use a busy fallback to the next dial? |
| 18:15.09 | rue_mohr | oh it does end on # |
| 18:15.15 | rue_mohr | cool |
| 18:15.29 | rue_mohr | haha |
| 18:15.29 | rue_mohr | I missed that , first line |
| 18:15.35 | rue_mohr | hands [TK]D-Fender the clue-bat |
| 18:16.13 | t_corr | i prefer the batclue, batman's themed clue board game |
| 18:16.33 | rue_mohr | ok am I wrong that the order of the channels in a group cant be defined and right to use the busy fallback in a string of dial commands? |
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| 18:17.26 | [TK]D-Fender | rue_mohr: Yes, You could stand in front of a bar rotating at 1 RPM with a submachine-gun w/ 5000 round clip firing at 900 RPM and still miss it |
| 18:17.31 | [TK]D-Fender | barn* |
| 18:17.43 | [TK]D-Fender | rue_mohr: dial them back to back already... |
| 18:18.06 | rue_mohr | yo mean exten-> dial ... \n exten-> dial ... |
| 18:18.24 | rue_mohr | cause the first one that gets it will hang up and leave the rest to die in peace? |
| 18:18.39 | rue_mohr | the errors from the others will cause them to fall through and not matter? |
| 18:18.58 | rue_mohr | end the string of dial commands in a playtones(game-over) ? |
| 18:20.14 | rue_mohr | that explains why I can get the vm to git on wit it if I press # after my user and pass... |
| 18:22.06 | rue_mohr | hmm wonder how I trigger a odbc write |
| 18:23.18 | rue_mohr | not like that |
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| 18:31.13 | rue_mohr | read isn't capturing ANY of the digits I dial?!? |
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| 18:35.40 | rue_mohr | it says by default read will wait forever, but it quits and says "user entered nothing" |
| 18:35.46 | rue_mohr | so it only caught the #? |
| 18:36.08 | rue_mohr | maybe I have an inband/outofband setting wrong? |
| 18:38.43 | rue_mohr | http://bin.cakephp.org/view/705586462 <- like that [TK]D-Fender |
| 18:40.15 | rue_mohr | wouldn't it cause the person in the office to get the person their calling again on another line if the out of office party hangs up first? |
| 18:40.32 | rue_mohr | or is a hangup on either end the end of it all |
| 18:41.30 | rue_mohr | dtmf method: rtp |
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| 18:48.10 | rue_mohr | any reason read cant ...read>? |
| 18:48.30 | rue_mohr | it says I didn't enter anything |
| 18:49.43 | rue_mohr | http://bin.cakephp.org/ |
| 18:49.54 | rue_mohr | http://bin.cakephp.org/view/1916790272 |
| 18:50.13 | rue_mohr | I cant see anything I did wrong, I pressed numbers like there was no tommorow |
| 18:51.52 | rue_mohr | read isn't reading |
| 18:54.35 | rue_mohr | maybe I can get around it by not using read |
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| 19:02.19 | anonymouz666 | anyone in here can suggest a good GSM gateway? I am thinking something that can talk SIP to Asterisk and goes to PSTN through the GSM.... |
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| 19:14.49 | msetim | hello guys. I discovered this ticket: http://bugs.digium.com/view.php?id=11962 How do you check if it is using this buffer in fact. I have been teste with/without this ticket and I really have been noted less cpu load however I don't belive that it is the correct way to check it :) |
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| 19:23.39 | rbd | hey guys, how would I compile in app_meetme.c without running make menuselect (I need to do so in a fully automated fashion) |
| 19:23.46 | rbd | asterisk 1.4.x BTW |
| 19:27.39 | Corydon76-dig | rbd: it's already selected by default |
| 19:28.00 | Corydon76-dig | rbd: if it's not selected, then you're missing a dependency |
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| 19:29.03 | Corydon76-dig | rbd: or you explicitly disabled it |
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| 19:30.52 | rbd | Corydon76-dig, yeah, I see that asterisk will include it in the build if I have zaptel loaded... problem is I need to build the module in isolation (e.g. zaptel won't be installed on the buildbox), while still building in meetme |
| 19:31.39 | rbd | basically, force meetme to be compiled in regardless of whether zaptel is loaded in the kernel (which is what asterisk is looking for I'd guess...that or zaptel being installed) |
| 19:31.44 | Corydon76-dig | rbd: then you cannot compile meetme, because it needs headers defined in dahdi/zaptel |
| 19:32.06 | Corydon76-dig | rbd: it's the equivalent of compiling kernel modules without kernel headers |
| 19:32.28 | rbd | ah, ok. any idea if this changes with asterisk 1.6. due to its move away from zaptel? |
| 19:32.57 | Corydon76-dig | rbd: sure, but it still requires dahdi headers |
| 19:33.08 | Qwell | rbd: How are you building anything if dependencies won't be on the build box? |
| 19:33.11 | Qwell | ie...gcc |
| 19:33.43 | rbd | Qwell, all deps will be on the build box, except I was trying to avoid having zaptel have to be built _and_ installed on there. |
| 19:34.00 | Qwell | You'll have to |
| 19:34.03 | Corydon76-dig | rbd: install just the dahdi headers, then |
| 19:35.00 | rbd | Corydon76-dig, any idea of a path..not seeing them in the usual suspects... |
| 19:35.45 | Corydon76-dig | rbd: /usr/include/dahdi |
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| 19:37.07 | rbd | with 1.4? I only see /usr/include/zaptel/zaptel.h and timezone.h being installed from the zaptel source packages at least .... then /usr/include/asterisk/dahdi_compat.h from the asterisk package |
| 19:37.13 | Corydon76-dig | rbd: once that directory exists and the files within it, then the configure check will pass the dependency |
| 19:37.35 | Corydon76-dig | rbd: why are you building against zaptel and not dahdi? |
| 19:38.26 | Corydon76-dig | rbd: zaptel is dead. No further releases or bugfixes. Dahdi is the way forward. |
| 19:38.38 | rbd | ah ok...I wasn't aware of that |
| 19:39.42 | rbd | so I should ditch the zaptel stuff (only using ztdummy anyhow) and move to dadhi.... does dadhi have something like ztdummy ... I would think meetme under 1.4 still requires some kind of zaptel-ish timing source (maybe the name has changed as dahdi is providing it now) |
| 19:40.03 | Corydon76-dig | dahdidummy |
| 19:40.05 | [TK]D-Fender | rbd: solution = SRPMS or equivalent <- |
| 19:40.38 | Corydon76-dig | rbd: meetme doesn't depend upon dahdi/zaptel for TIMING. It's for audio MIXING |
| 19:41.33 | rbd | ok, this is enlightening. I'll start building up against dahdi. thanks guys |
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| 19:42.45 | DavidR2008 | hello all! |
| 19:43.21 | hardwire | wot. |
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| 19:44.24 | DavidR2008 | I've got a faxing question: Will I run into any problem with this config: PRI<->Asterisk<-(IAX across LAN)->Asterisk<->fax machine |
| 19:44.50 | DavidR2008 | and/or what do I need to do to make this work |
| 19:45.55 | Poincare | DavidR2008: if you are verrrryyyy lucky |
| 19:46.05 | Poincare | you will need t38 |
| 19:46.54 | mercutioviz | DavidR2008: I guess he's saying you won't have any problems as long as you don't actually try to send a fax over that setup... :P |
| 19:47.29 | [TK]D-Fender | pointNeato... you invented T.38 for IAX2? SHARE! :p |
| 19:47.31 | DavidR2008 | it's all in-house, I have an asterisk server that terminates all my Voice Ts and PRIs. currently it is tied to a legacy PBX via T1 and the fax machines hang off the PBX, I want to replace the PBX with an asterisk server |
| 19:48.07 | [TK]D-Fender | Poincare: Neato... you invented T.38 for IAX2? SHARE! :p |
| 19:48.31 | Poincare | DavidR2008: you will have jitter that way. fax doesn't like that |
| 19:48.42 | Poincare | [TK]D-Fender: did i? cool :-) |
| 19:48.43 | DavidR2008 | so my config pretty flexable the one I stated above is how it's configure now while I test the new Asterisk PBX. I can make any changes necessary. |
| 19:49.45 | Poincare | what hardware are you using? |
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| 20:06.50 | DavidR2008 | sangoma A104s for the PRI/T1 interface I will be using sangoma FXO/FXS cards for the fax ports |
| 20:06.57 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 20:07.58 | KavanS | DavidR2008: get fax from asterisk, then email to other server ;) |
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| 20:08.06 | KavanS | probably your best route to success ? |
| 20:08.15 | KavanS | waits for impending correction from l33t asterisk guru's |
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| 20:13.44 | jmodigb | has anyone here had problems compiling asterisk on a 64 bit system? |
| 20:14.25 | Qwell | jmodigb: explain your problems |
| 20:14.31 | jmodigb | getting error /usr/bin/ld: skipping incompatible /usr/lib/libidn.so when searching for -lidn |
| 20:14.31 | jmodigb | <PROTECTED> |
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| 20:16.08 | Qwell | jmodigb: Do you have a 64-bit version of it installed? |
| 20:16.13 | `paul | maybe off topic: do you know a good toll free provider in the us? |
| 20:16.23 | `paul | any recommendations? |
| 20:16.36 | Qwell | ~itsplist-us |
| 20:16.37 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:16.38 | jmodigb | i followed the book... |
| 20:16.50 | jmodigb | probably should have checked for 64 bit veersion... |
| 20:16.55 | Qwell | jmodigb: You're on CentOS, aren't you? |
| 20:16.57 | Qwell | 5.2? |
| 20:17.03 | jmodigb | yes for both |
| 20:17.07 | Qwell | sigh |
| 20:17.12 | Qwell | Why have they not fixed this yet? |
| 20:17.24 | Qwell | jmodigb: Go through your yum logs. Remove everything that was just installed. |
| 20:17.38 | Qwell | When you install whatever development package you need, make sure to add .x86_64 at the end |
| 20:17.50 | Qwell | so, `yum install unixODBC-devel` becomes `yum install unixODBC-devel.x86_64` |
| 20:18.22 | anonymouz666 | Qwell: in some cases this shouldn't install both versions? |
| 20:18.37 | Qwell | anonymouz666: No, in no cases ever should it install .i386 |
| 20:18.44 | Qwell | It's broken. Badly. |
| 20:19.07 | anonymouz666 | Well, that was not what CentOS guys said to me.... |
| 20:19.11 | Qwell | They're wrong. |
| 20:19.13 | Corydon76-dig | One wonders if it's also broken in RHEL |
| 20:19.22 | Qwell | Corydon76-dig: RHEL doesn't use yum. |
| 20:19.29 | Corydon76-dig | Ah |
| 20:19.46 | Qwell | I mean, it can use yum, but it wouldn't be supported because it would be third party repos |
| 20:19.49 | Corydon76-dig | unless you install it from centos |
| 20:19.56 | Corydon76-dig | right |
| 20:19.58 | Qwell | it includes yum, but there are no repos |
| 20:20.07 | Qwell | it's just wrong. period. |
| 20:20.17 | Qwell | here's what happens... |
| 20:21.01 | Qwell | You install unixODBC-devel, and it says "Okay, I need to install the .i386 and .x86_64 packages of unixODBC-devel" (which is fine), then it says "Okay, these depend on unixODBC. Okay, I found a unixODBC package. I'm installing unixODBC.i386" |
| 20:21.08 | Qwell | No, that's dumb and broken. |
| 20:22.10 | Qwell | (that goes for any package - not just that one) |
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| 20:22.37 | file | Qwell isn't bitter over it or anything... |
| 20:22.42 | Corydon76-dig | wonders how difficult that would be to fix... |
| 20:22.58 | Qwell | Corydon76-dig: trivial... |
| 20:23.06 | s14ck | Qwell: hi |
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| 20:23.40 | s14ck | Qwell: i have this |
| 20:23.52 | DavidR2008 | I could duplicate what I have today, but I'm trying to avoid the cost of putting a T1 card in my * PBX just to connect to another * server. The two services are about five feet apart, would using a crossover cable reduce latency? |
| 20:24.15 | DavidR2008 | *servers |
| 20:24.31 | *** join/#asterisk jsmith (n=nnnjsmit@asterisk/training-and-documentation-guru/jsmith) |
| 20:24.31 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 20:24.40 | s14ck | *CLI> dahdi show status Wildcard TE121 Card 0 RED 1 0 0 ESF B8ZS YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 20:24.50 | s14ck | Qwell: *CLI> dahdi show status Wildcard TE121 Card 0 RED 1 0 0 ESF B8ZS YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 20:24.55 | DavidR2008 | or is it one of those things where I just have to try it? and YMMV |
| 20:25.39 | jmodigb | Qwell: thx, i think at this point it would be easier to reinstall centos since asterisk is the first thing i've set up |
| 20:25.57 | s14ck | Qwell: i dont know how set Alarms in green |
| 20:26.54 | Qwell | s14ck: I don't know why you're directing your question at me... |
| 20:28.15 | jaytee | I'm getting HDLC FCS(8) errors and HDLC Abort(6) errors, could someone check my configs? my pastebin includes the pertinent messages from the asterisk logs, my zaptel and zapata configs and the output of cat /proc/interrupts: http://pastebin.com/m824fa93 |
| 20:28.27 | s14ck | Qwell: well, I'm asking you |
| 20:28.31 | anonymouz666 | Qwell: The CentOS stuff won't be fixed, because from their perspective is totally right to have both version of the "same" package. |
| 20:28.49 | jsmith | jaytee: Those usually indicate a problem with the line (or a poorly crimped cable), not a config problem |
| 20:29.11 | jsmith | jaytee: Look at Layer 1 first :-) |
| 20:30.17 | jaytee | these two spans have been stable without errors since I corrected the timing parameters for them back on Mar 2nd. I was removing my old Nortel Option 11C from the computer room this morning but around the time of the alarm I don't think I jarred the cables to my csu/dsu's |
| 20:30.48 | Qwell | s14ck: why? |
| 20:30.48 | s14ck | Qwell: do you think you can help me? |
| 20:31.04 | jaytee | jsmith, Layer 1 meaning physical layer i.e. cabling? |
| 20:31.08 | Qwell | No, but had you not directed your question at me, the other 316 people in the channel would have not ignored it... |
| 20:31.45 | Qwell | anonymouz666: It's broken. |
| 20:31.56 | Qwell | anonymouz666: Tell them to install unixODBC-devel, and then try to actually *use it*. |
| 20:32.01 | Qwell | It won't work. Ever. |
| 20:32.09 | s14ck | Qwell: thank you for your help |
| 20:32.49 | Qwell | anonymouz666: And no, it's absolutely not right to have multiple versions of all packages installed at the same time. |
| 20:33.07 | anonymouz666 | Qwell: "not everything is 64-bit" |
| 20:33.22 | Qwell | Install a package (let's use subversion as an example). Now run a `file /usr/bin/svn` |
| 20:33.26 | Qwell | That is *WRONG*. |
| 20:34.13 | Qwell | now, hell, remove subversion.i386 so you're left with subversion.x86_64. run `file /usr/bin/svn` - now what happens? Oh, the file is gone. Oops! |
| 20:34.41 | Qwell | You can't successfully argue with me that it isn't broken. :) |
| 20:35.06 | anonymouz666 | Qwell: "Do you wish to run CentOS?" :) |
| 20:35.21 | Qwell | I've worked around the issue entirely. |
| 20:35.46 | Qwell | but, I'm sick of people reporting bugs about the same thing repeatedly, because it's A) a change in behavior, and B) clearly broken. |
| 20:36.57 | anonymouz666 | I gotta go. |
| 20:37.02 | anonymouz666 | but I think you're right |
| 20:37.09 | Qwell | If you're talking to CentOS devs, feel free to have th... |
| 20:37.10 | Qwell | bleh |
| 20:37.31 | jaytee | bleh, bleh :-) |
| 20:37.54 | jsmith | jaytee: Yes, Layer 1 = Physical layer |
| 20:38.11 | jaytee | Qwell, somedays ya just gotta tilt your head back and yell, "F*&# IT!!!" |
| 20:38.14 | Qwell | jsmith: There should be a layer 0 |
| 20:38.17 | jsmith | jaytee: Also known as the "It works better if you plug it in" layer... |
| 20:38.27 | jaytee | jsmith, domo arrigato gozimashita, Sensei!!! |
| 20:38.30 | jsmith | Qwell: Yeah... that's the "It works better if you turn it on" layer... |
| 20:38.32 | Qwell | jsmith: That would be the consultant who did the physical install |
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| 20:38.51 | jsmith | Qwell: You heard they added layers 8 and 9 to the OSI layer, right? |
| 20:38.56 | Qwell | no? |
| 20:38.57 | jaytee | that would be me then |
| 20:39.05 | jsmith | Layer 8 is the financial layer, and the Layer 9 is the political layer |
| 20:39.10 | Qwell | heh |
| 20:39.15 | jaytee | since I'm a one-stop shop for all the crap they wanna do around here. |
| 20:39.55 | Qwell | http://en.wikipedia.org/wiki/Layer_8 |
| 20:40.33 | s14ck | jsmith or jaytee, do you can help me? I'm trying configuring E1 card with dahdi at asterisk 1.6 |
| 20:40.46 | jaytee | so Layer 8 is where we get bonuses for screwing up? |
| 20:41.01 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 20:42.13 | jaytee | sl4ck, sorry dude, I have no experience with E1's. just T1 and not that much of that either. |
| 20:42.43 | jsmith | s14ck: Just ask the question out to the entire group -- don't single people out |
| 20:42.56 | jsmith | s14ck: Give us a bit more detail, and somebody may answer you |
| 20:43.02 | s14ck | jaytee: do you have config an interface with R2 signalling? |
| 20:43.04 | mmlj4 | can't be a lot different from T1s |
| 20:43.19 | jsmith | s14ck: Is this a PRI? Channelized E1? |
| 20:43.38 | jsmith | s14ck: R2 over E1? |
| 20:43.39 | tzafrir_laptop | s14ck, R2 is something that can be run on E1 |
| 20:43.49 | tzafrir_laptop | Is that what you have? |
| 20:44.43 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 20:45.35 | deadpigeon | when I look at the datasheet for the TE110P, it says it supports hardware echo cancellation. if this is enabled by default, it definitely isn't functioning for me. |
| 20:45.55 | deadpigeon | TE412P, TE407P, TE212P, TE207P, and B410P. |
| 20:46.09 | deadpigeon | The datasheet says that TE110P supports it, but digium's site lists only the above cards. |
| 20:46.10 | tzafrir_laptop | s14ck, IIRC MFC/R2 is only supported with trunk / 1.6.2 (unless you use the external chan_unicall) |
| 20:46.46 | Qwell | deadpigeon: The hardware echo can is a module that plugs into the board. |
| 20:46.58 | deadpigeon | ah. |
| 20:47.03 | Qwell | deadpigeon: You'll need to figure out whether you bought one or not |
| 20:47.12 | deadpigeon | surely a 400 dollar dsp module? =/ |
| 20:47.28 | Qwell | I don't know the pricing, but yeah, something like that |
| 20:47.44 | deadpigeon | im assuming, no we havn't if im placing calls with echo. |
| 20:49.14 | jsmith | deadpigeon: Look in dmesg (or /var/log/messages) for VPM found or VPM not found |
| 20:52.31 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
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| 20:58.58 | deadpigeon | anyhoo, is MG2 the best solution for software echo cancellation? |
| 20:59.07 | deadpigeon | or is it worth to toy around with the others? |
| 20:59.41 | jsmith | deadpigeon: It only costs you time to try the others out, but I'm perfectly happy with MG2. |
| 20:59.49 | *** join/#asterisk ayeso (n=chatzill@216.65.195.52) |
| 21:00.02 | jsmith | deadpigeon: I don't know if I'd try to do echo cancellation on entire T1 with it, but it's great for low numbers of channels |
| 21:00.14 | deadpigeon | Hm, that's what I was trying... |
| 21:00.18 | deadpigeon | 1-23 |
| 21:00.25 | ayeso | If I want to change the greetings in comedian mail, what file do i edit? |
| 21:00.42 | jsmith | ayeso: Which greetings? |
| 21:01.10 | deadpigeon | And I honestly didn't see much of a difference when changing the settings around... it seems to work sometimes and then not. |
| 21:01.17 | ayeso | voicemail main options, like she says "press whatver for this" where can I go to change that? |
| 21:01.38 | jsmith | ayeso: You want to change just the sound prompt, or the option numbers as well? |
| 21:01.48 | jsmith | ayeso: For the sound prompts, they're in /var/lib/asterisk/sounds |
| 21:01.49 | ayeso | both |
| 21:02.02 | jsmith | ayeso: If you want to change the option numbers, you get to change app_voicemail.c and recompile |
| 21:02.07 | ayeso | jsmith: yeah, i want to edit the actual file |
| 21:02.16 | jsmith | ayeso: Or use minivm (in newer versions of Asterisk) |
| 21:02.28 | ayeso | jsmith: what is minivm? |
| 21:02.40 | s14ck | tzafrir_laptop: yes, i have a alcatel 4400 with one E1 interface, but i need interconect with de arterisk 1.6 |
| 21:03.01 | jsmith | ayeso: http://www.asterisk.org/node/48326 |
| 21:03.16 | tzafrir_laptop | s14ck, do you connect with R2? PRI? |
| 21:04.06 | s14ck | tzafrir_laptop: the alcatel people tellme R2, so, i need configured asterisk with r2 too |
| 21:04.12 | ayeso | jsmith: thanks |
| 21:05.02 | tzafrir_laptop | s14ck, if you're brave, the first beta of 1.6.2 is about to be released ... |
| 21:05.33 | s14ck | tzafrir_laptop: Asterisk 1.6.0.3 built |
| 21:05.41 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 21:06.00 | s14ck | tzafrir_laptop: some idea about the r2? |
| 21:06.27 | tzafrir_laptop | I think that there's a branch with the mfcr2 branch to for other asterisk versions. not sure |
| 21:07.06 | s14ck | tzafrir_laptop: do you say than asterisk 1.6 is not r2 compilance? |
| 21:07.59 | tzafrir_laptop | s14ck, I'm saying that I need to go now and I hope someone else will lend a hand here |
| 21:08.32 | s14ck | tzafrir_laptop: well... |
| 21:09.18 | ayeso | is there a functions to querry comedian mail for # of message or # of unread messages? |
| 21:09.37 | moy | s14ck: there are branches here http://svn.digium.com/view/asterisk/team/moy/ |
| 21:09.52 | moy | and patches here http://code.google.com/p/openr2/downloads/list |
| 21:10.23 | moy | questions can be sent to asterisk-r2 mailing list |
| 21:11.07 | mercutioviz | moy: nice work on the r2 stuff. Now you need to hack away on OpenZAP! ;) |
| 21:11.42 | s14ck | moy: thank you |
| 21:12.19 | moy | mercutioviz: indeed :) |
| 21:12.38 | mercutioviz | You'll earn the same amount of money! :P |
| 21:13.09 | *** join/#asterisk mizerydearia (n=mizeryde@CPE-65-27-127-37.new.res.rr.com) |
| 21:13.36 | moy | then I hope is twice as fun! hehe |
| 21:13.40 | *** join/#asterisk ESCulapio__ (n=Error@adsl-224-238.tricom.net) |
| 21:15.57 | ESCulapio__ | who can help me configure asterisk with ss7 to head office |
| 21:16.33 | BlargMaN00 | does anyone in here have some experience with the hints feature and polycom phones??? specifically the ringinuse hint??? |
| 21:16.59 | ESCulapio__ | who can help me configure asterisk with ss7 to head office |
| 21:17.26 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:18.45 | *** join/#asterisk dvdevel (n=devel@wiggum.digitalcoven.com) |
| 21:20.16 | DavidR2008 | I've been out of the asterisk loop evidently. I'm trying to install a sangoma card on asterisk 1.4.23.1 and I can't find zaptel, does asterisk still use zaptel? and/or can I use an older version of zaptel with 1.4.23.1 (I've got one I downloaded for * 1.4) |
| 21:21.45 | jsmith | DavidR2008: Asterisk 1.4 will use Zaptel or DAHDI |
| 21:21.54 | jsmith | DavidR2008: Asterisk 1.6 will use DAHDI |
| 21:21.59 | DavidR2008 | thx! |
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| 21:27.13 | ESCulapio__ | who can help me configure asterisk with ss7 to head office |
| 21:28.52 | jsmith | ESCulapio__: Your best bet is to ask on the asterisk-ss7 mailing list... I don't think very many people in the channel use SS7 |
| 21:30.26 | ESCulapio__ | jsmith, ok thank |
| 21:34.55 | DavidR2008 | bbl |
| 21:35.10 | *** join/#asterisk af_ (n=getsmart@88-149-240-30.dynamic.ngi.it) |
| 21:57.08 | EmleyMoor | I'm using LCDial.sh (obsolete-ish but it does what I want better than the nearest alternative) and I am wondering if it only uses the cheapest route or whether it should fail over to a more expensive one if, for example, there is no spare outgoing capacity on the cheap provider? |
| 21:58.31 | hardwire | Anybody else here using Affinity? |
| 21:58.36 | hardwire | Their CDR is.. weird. |
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| 22:13.57 | *** join/#asterisk WeazelON (n=WeazelON@mail2.tikalnetworks.com) |
| 22:15.27 | WeazelON | whats the command for "pri show span" in asterisk 1.4.23.1 ? |
| 22:15.36 | WeazelON | please help me i need an answer asap |
| 22:17.18 | *** join/#asterisk Valect (n=Valect@unaffiliated/valect) |
| 22:17.26 | Valect | awesome |
| 22:18.01 | Valect | so, not sure if this is related to asterisk or not - i actually had our works voip system dumped on my lap with zero documentation, and i'm not totally familiar with voip |
| 22:18.08 | Valect | anyway, we're having issues dialing international numbers |
| 22:18.18 | Valect | receiving "three one, please leave a message fater the tone" |
| 22:18.20 | Valect | ideas? |
| 22:18.22 | jsmith | WeazelON: It's "pri show span 1" |
| 22:18.37 | WeazelON | not working for me for some reason |
| 22:18.42 | WeazelON | not even Tab completing for me |
| 22:18.44 | jsmith | WeazelON: Did you compile libpri? |
| 22:18.53 | jsmith | (compile and install, that is) |
| 22:18.56 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 22:19.10 | jsmith | Valect: Sounds like the number is being misdialed for some reason. |
| 22:19.25 | jsmith | Valect: You could try reading the Asterisk book to help get up to speed |
| 22:19.27 | jsmith | ~thebook |
| 22:19.28 | jbot | from memory, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 22:20.06 | Valect | so is this something that could be related to asterisk then? |
| 22:21.04 | jsmith | Valect: Could be... might be your provider as well |
| 22:21.07 | WeazelON | @smith it was working before.... i just got a call from a client that his PBX is down, and i go in, run rc-status and i see only zaptel stopped but i'm not sure about this and asterisk 1.4.23.1 |
| 22:21.12 | WeazelON | not sure it needs it |
| 22:21.20 | jsmith | Valect: Could be a *lot* of things |
| 22:21.25 | Valect | i'm guessing it's probably related to the remote location, but i don't have any way to verify |
| 22:21.39 | jsmith | WeazelON: If you're connecting a PRI, you must compile and install libpri prior to building Asterisk |
| 22:21.49 | Valect | reads |
| 22:21.59 | WeazelON | is there a way to find out if its already installed and compiled ? |
| 22:22.02 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 22:22.02 | *** mode/#asterisk [+o Mog] by ChanServ |
| 22:22.11 | WeazelON | cause i'm guessing it is, since it was working for months |
| 22:22.16 | Valect | find / -name libpri\* |
| 22:23.05 | WeazelON | it finds some stuff regarding that line |
| 22:23.14 | WeazelON | alot |
| 22:24.27 | Chuggs | Would anyone know how to fix this -> "WARNING[3213]: translate.c:654 __ast_register_translator: plc_samples 160 format f" ? |
| 22:26.04 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
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| 22:42.31 | darnoq | l |
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| 22:44.44 | phix | lol |
| 22:44.57 | *** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
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| 23:04.00 | *** join/#asterisk LeandroLeal (n=linux@187.14.8.144) |
| 23:04.07 | LeandroLeal | Hello! |
| 23:04.20 | LeandroLeal | Algum brasileiro aqui? |
| 23:04.50 | Chainsaw | Not really, no. But you could try in english? |
| 23:05.01 | LeandroLeal | yes! |
| 23:05.08 | LeandroLeal | i can! |
| 23:05.10 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 23:05.13 | Chainsaw | Excellent. Phone problems? |
| 23:05.20 | LeandroLeal | no ! |
| 23:05.29 | LeandroLeal | i`m developer |
| 23:05.37 | Chainsaw | Ah, alright. |
| 23:05.40 | LeandroLeal | in php |
| 23:05.45 | LeandroLeal | and mysql |
| 23:05.50 | Chainsaw | C here. But PHP is fun too. |
| 23:06.06 | LeandroLeal | i understand |
| 23:06.15 | LeandroLeal | yes C too |
| 23:06.39 | LeandroLeal | no more, but i to do anything |
| 23:07.06 | LeandroLeal | do you know any software billing freeware? |
| 23:08.55 | Chainsaw | I'm hoping some others will pipe up with information. |
| 23:09.16 | Chainsaw | If our office users are logged in, they can call anywhere. British Telecom do the billing for us. |
| 23:09.24 | Chainsaw | (Calls go out over ISDN) |
| 23:13.03 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 23:13.27 | *** join/#asterisk Doc (n=scott@willers.employees.org) |
| 23:13.36 | Doc | are there any good web interfaces for meetme to show things like who's talking/mute users/etc ? |
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| 23:27.10 | lesouvage | <PROTECTED> |
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| 23:51.00 | lesouvage | Solved it, was a not continuing ivr kind a thing. |
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| 23:59.17 | *** join/#asterisk martyn-dev (n=admin@190.24.134.154) |
| 23:59.21 | martyn-dev | hi |
| 23:59.25 | martyn-dev | every1 :D |
| 23:59.36 | martyn-dev | somebody has used JAGIClient to connect to AGI asterisk ? |