| 00:06.43 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 00:10.01 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 00:29.08 | [TK]D-Fender | thansen|laptop: "core show application read" |
| 00:43.53 | lucky|aba | w00t!! |
| 00:44.02 | lucky|aba | all the extensions are working with voicemail |
| 00:44.08 | lucky|aba | how fun is this |
| 00:50.20 | thansen|laptop | [TK]D-Fender: so we're reverting from | back to , it looks like |
| 00:50.42 | [TK]D-Fender | thansen|laptop: Yes, long since documented. |
| 00:50.51 | thansen|laptop | and the options have changed :D |
| 00:51.11 | thansen|laptop | hasn't kept up too well...I appreciate the help |
| 00:51.19 | thansen|laptop | where is it documented? |
| 00:53.25 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 00:58.06 | lucky|aba | Do i need to have any hardware to get inbound and outbound service? |
| 00:58.17 | lucky|aba | like a live # i can call and call out from? |
| 00:59.52 | *** join/#asterisk gones (n=gones@203.193.37.251) |
| 01:06.52 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:09.06 | jblack | lucky|aba: No, if you use soft phones and voip. |
| 01:09.35 | lucky|aba | sweet, know a cheap service i can get? |
| 01:10.18 | jblack | I've found as cheap as 1.2 cents a minute outgoing as reliable. |
| 01:10.39 | jblack | Incoming, the cheapest I have found is 2 cents a minute and $5 a month. |
| 01:14.32 | thansen|laptop | lucky|aba: http://www.vitelity.com/?p=retailserv has worked pretty well for me |
| 01:15.01 | jblack | lucky|aba: Basically, ten bucks a month for me. |
| 01:15.27 | jblack | For everything |
| 01:15.58 | lucky|aba | thats not too shabby |
| 01:16.22 | jblack | Yeah. Callerid, did, everything. |
| 01:16.36 | jblack | Oh, and you can get incoming calls for free if you don't mind a washington state number at ipkall.com |
| 01:16.58 | lucky|aba | no kidding |
| 01:17.19 | jblack | serious. DID and all. That's the number I usually hand out. :) |
| 01:18.53 | lucky|aba | So could i use that to ring in to the house and have an IVR which would allow people to dial extensions for different people? |
| 01:19.17 | jblack | yup. That's what I have. |
| 01:19.24 | lucky|aba | that is just too sick |
| 01:19.26 | lucky|aba | lol |
| 01:19.31 | jblack | Oh, then don't do it. |
| 01:19.47 | lucky|aba | man... how didn't i jump on this VOIP bandwagon before |
| 01:19.50 | lucky|aba | sick = awesome |
| 01:20.00 | lucky|aba | west coast... we're all jacked up out here :P |
| 01:20.08 | Strom | where on the west coast? |
| 01:20.13 | lucky|aba | Santa Barbara |
| 01:20.15 | jblack | Moronoville? |
| 01:20.15 | lucky|aba | Cali |
| 01:20.19 | jblack | damn. I tried. |
| 01:20.22 | lucky|aba | haha |
| 01:20.29 | Strom | lucky|aba: I'm in los angeles |
| 01:20.37 | Strom | and ffs, don't call it "cali" :) |
| 01:20.47 | Strom | Cali is a city in Colombia -- this is California :) |
| 01:20.53 | lucky|aba | haha |
| 01:20.54 | lucky|aba | true |
| 01:21.20 | jblack | strom: Are you all jacked up, getting too sick from cali? |
| 01:21.41 | Strom | jblack: I'm barely getting the hang of "dude" |
| 01:21.54 | Strom | and I've lived in this part of the country my whole life |
| 01:22.25 | jblack | Yeah. when I left L.A., I didn't exactly look back |
| 01:22.40 | Strom | I love it here |
| 01:23.10 | jblack | Not me. I couldn't cope with the 200 word vocabularies. |
| 01:23.26 | Strom | there are 200 word vocabularies wherever you go |
| 01:23.35 | Strom | it's a matter of who you choose to associate with |
| 01:23.39 | jblack | Oh yeah? In _this_ town, there are _three_ words for you. |
| 01:23.40 | lucky|aba | seriously.... try living in Minnesota |
| 01:24.03 | jblack | "You", as in a single person, "Yous" as in a group of people, and "yousses", as in two or more groups of people. |
| 01:24.13 | Strom | lucky|aba: the closest I've come is Las Vegas |
| 01:24.18 | Strom | and even that place feels small to me |
| 01:24.27 | academy | (slightly off-topic question) Does anyone knoe if you're allowed to use the UK Mobile Broadband packages with a phone rather than a usb mobile dongle? |
| 01:24.28 | jaytee | "Did you say yute?" |
| 01:24.39 | Strom | jblack: I picked up "Y'all" when I was in Alabama |
| 01:25.13 | hsv-al | strom |
| 01:25.25 | hsv-al | when you come to hsv, ask for some "fixins & vittles" |
| 01:25.28 | hsv-al | luls |
| 01:25.38 | jaytee | "I'm sawry your honaah, I meant the two yooothhhs" |
| 01:25.39 | hsv-al | chicken, and dinner, with "all the fixins" |
| 01:25.54 | jblack | fixins and vittles sounds like what's left over after a carcass is disemboweled. |
| 01:26.00 | hsv-al | heh |
| 01:26.13 | hsv-al | more like the insides of the snow camel |
| 01:26.13 | jaytee | "No self-respectin southerner would eat instant grits" |
| 01:26.17 | hsv-al | that han solo stuffed luke skywalker in |
| 01:26.19 | hsv-al | when he was frozen |
| 01:26.34 | hsv-al | ;-]] |
| 01:26.39 | Strom | I didn't quite get grits |
| 01:26.53 | Strom | now...barbecue is a different story |
| 01:27.01 | hsv-al | strom i moved here from NJ, so ive picked up, well not using myself |
| 01:27.02 | jaytee | or fried chicken |
| 01:27.03 | jblack | hsv-al: This'll be "too sick" for you, but Hans disemboweled the camel. |
| 01:27.05 | hsv-al | but know what the "weird phrases" are |
| 01:27.20 | jblack | Han, that is. |
| 01:27.26 | jaytee | either of those two foods and the south is unsurpassed |
| 01:27.34 | jaytee | but I never saw the appeal of grits |
| 01:27.41 | hsv-al | "hanging of the greens", "Pot luck dinner", "fixins & Vittles", "mudding" |
| 01:27.43 | hsv-al | hmmm what else |
| 01:27.48 | jblack | Speaking of the south, how does one deep fry ice cream? Doesn't it melt? |
| 01:28.04 | jaytee | I eat oatmeal cuz it's healthy but even then I'm not dancing a jig about it. |
| 01:28.19 | hsv-al | been stressing over this iphone last 5 hours |
| 01:28.26 | hsv-al | cant figure out how to setup ipsec in the cisco client |
| 01:28.28 | Strom | here in los angeles there's a small local chain called "Roscoe's House of Chicken & Waffles. It's the closest thing we have to a deep south soul food heart attack on a plate, I think. It's awesome, but you can't go there more than once a month or so. |
| 01:28.31 | hsv-al | you cant import PCF files |
| 01:28.43 | Strom | hsv-al: didn't anyone tell you yet? the iphone sucks. |
| 01:28.57 | Strom | also, brb |
| 01:28.57 | hsv-al | its good, once i get the vpn / ipsec setup |
| 01:29.03 | hsv-al | im putting ssh client on it |
| 01:29.11 | jblack | I heard about Roscoe's. Never got there. |
| 01:29.12 | hsv-al | then i can start engaging in IOSsex with our hardware |
| 01:29.39 | jaytee | I think you need some serious therapy, fanboi |
| 01:29.47 | x86 | there is a cisco vpn client for iphone? |
| 01:29.52 | hsv-al | its native |
| 01:29.53 | hsv-al | built in |
| 01:29.58 | hsv-al | they worked hand in hand w/ cisco on it |
| 01:30.03 | x86 | that's not the cisco vpn client then ;) |
| 01:30.08 | hsv-al | not native |
| 01:30.10 | x86 | that's an ipsec l2tp client |
| 01:30.11 | hsv-al | i should say: Cisco Native |
| 01:30.30 | x86 | generic, standardized client |
| 01:30.33 | hsv-al | well, then maybe a proper wording would be |
| 01:30.37 | hsv-al | cisco IPSec implementation |
| 01:30.43 | hsv-al | because its cisco logo'ized |
| 01:30.45 | hsv-al | when you click: Ipsec |
| 01:30.49 | hsv-al | isntead of pptp or l2tp |
| 01:30.58 | jaytee | who cares what you call it? does it even work? |
| 01:30.59 | x86 | hmm not here it's not |
| 01:31.05 | x86 | works for me :) |
| 01:31.10 | x86 | going to a PIX 501 |
| 01:31.11 | hsv-al | when i click IPsec, when i choose between the 3 prots |
| 01:31.14 | hsv-al | cisco's logo comes up |
| 01:31.19 | jaytee | cool |
| 01:31.22 | x86 | not mine hsv-al |
| 01:31.26 | hsv-al | 2.0? 3g? |
| 01:31.35 | Strom | back |
| 01:31.40 | jaytee | so does the actual phone part of it work too or are you still waiting on activation? |
| 01:31.53 | hsv-al | mine activated as soon as we did it at the store |
| 01:31.54 | x86 | but i'm running an ipod touch jailbroken and with some iphone apps on it :) |
| 01:32.01 | hsv-al | dunno |
| 01:32.03 | hsv-al | whats the website about that |
| 01:32.08 | hsv-al | some douchebag was showing me jailbreak on his |
| 01:32.13 | hsv-al | outside teh store today |
| 01:32.24 | x86 | jailbreak on iphone 2.0? |
| 01:32.28 | hsv-al | yes |
| 01:32.32 | x86 | wow that was fast |
| 01:32.39 | hsv-al | well 2.0? i dont know |
| 01:32.40 | x86 | oh prolly same firmeware though |
| 01:32.46 | hsv-al | n/m, it was an old one |
| 01:32.52 | hsv-al | dont think there is a 2.0 hack yet |
| 01:32.56 | x86 | ah ok |
| 01:32.59 | jaytee | wait a wee |
| 01:33.02 | jaytee | week |
| 01:33.06 | x86 | was about to say.... one day... that's quick :) |
| 01:33.17 | hsv-al | in a bit ill be targeting wire shark on it |
| 01:33.30 | hsv-al | when I use a usb to ethernet converter to give it router data, instead of self 3G(when I turn it off) |
| 01:33.34 | jaytee | I think they busted the first one within 10 days of the iPhone 1.0 release IIRC |
| 01:33.35 | hsv-al | im gonna try to capture what it filters |
| 01:33.44 | hsv-al | for now, its just public knowledge of proxie'd http |
| 01:34.12 | x86 | jaytee: yeah but the itouch had been out for a while before iphone 1.0 |
| 01:34.26 | hsv-al | x86, is there a flash client for it? |
| 01:34.31 | hsv-al | so we can do streetview(google) in safari |
| 01:34.39 | hsv-al | 3d map rotation by dropping pegman |
| 01:34.59 | jaytee | will it make phone calls? |
| 01:35.10 | hsv-al | jaytee, the devs on apples site were saying that |
| 01:35.16 | hsv-al | when you use streetview(when the official plugin comes out) |
| 01:35.26 | hsv-al | you can click on a house in streetview on the google map |
| 01:35.31 | hsv-al | and it dials that houses # |
| 01:35.33 | hsv-al | if its public listing |
| 01:35.44 | hsv-al | screwed up shit |
| 01:36.10 | hsv-al | you do know what street-view is right? |
| 01:36.29 | hsv-al | http://maps.google.com/help/maps/streetview/conversion.html |
| 01:38.00 | x86 | hsv-al: nope, and there will never be flash, unless someone ports some non-adobe distrobution of flash |
| 01:38.11 | x86 | same with Java... it'll never happen |
| 01:38.14 | hsv-al | thats creepy as hell |
| 01:38.18 | x86 | it's retarded |
| 01:38.19 | hsv-al | using the streetview w/ that capability |
| 01:38.29 | x86 | it's against Apple's policy |
| 01:38.34 | hsv-al | ? |
| 01:39.08 | x86 | sun came right out and announced via press release that they were going to make a version of java for the itouch / iphone... |
| 01:39.30 | x86 | apple said they'd sue sun if they tried it, as it's against the EULA or some crap |
| 01:39.40 | x86 | huge mistake, imho |
| 01:39.43 | hsv-al | how the hell would google do that |
| 01:39.49 | hsv-al | linking gps data to phone #'s |
| 01:39.56 | x86 | would love to play java games on yahoo on my itouch ;) |
| 01:41.34 | mvanbaak | OpenMoko |
| 01:41.50 | x86 | doesnt run on an itouch though, that's the thing ;) |
| 01:42.08 | hsv-al | openmoko in theory is good |
| 01:42.11 | hsv-al | but gprs only = ftl |
| 01:42.34 | mvanbaak | here in .nl the iPhone is gprs only as well |
| 01:42.53 | mvanbaak | they call it 3G |
| 01:43.04 | mvanbaak | but in practice it's gprs only |
| 01:44.07 | mvanbaak | so if it's gprs only, I prefer OpenMoko |
| 01:44.25 | mvanbaak | way better platform for development |
| 01:44.50 | *** join/#asterisk gnutoo (n=gnutoo@host54-133-dynamic.41-79-r.retail.telecomitalia.it) |
| 01:45.10 | hsv-al | x86 |
| 01:45.15 | hsv-al | ipsec on 2.0 requests "secret" |
| 01:45.21 | hsv-al | is that grouppwd = "$string" |
| 01:45.22 | hsv-al | in the pcf? |
| 01:46.54 | *** join/#asterisk ftp5 (n=ftp3@pool-71-117-212-7.ptldor.fios.verizon.net) |
| 01:47.28 | ftp5 | is there a way to connect skype so that if someone calls my skype number it goes into my asterisk pbx? |
| 01:47.57 | x86 | hsv-al: usually |
| 01:48.06 | x86 | hsv-al: depends on how AAA is setup |
| 01:48.19 | hsv-al | its like 120 chars |
| 01:48.23 | hsv-al | theres no way im typing that manually |
| 01:48.23 | Strom | ftp5: no |
| 01:48.24 | hsv-al | rotating cases |
| 01:48.31 | Strom | ftp5: nothing that works worth shit, anyway |
| 01:48.44 | x86 | if you've actually got individual users setup, you'll need to put in the group name, user name, and user password |
| 01:48.56 | ftp5 | i saw skip2pbx but it looks really expensive |
| 01:49.00 | hsv-al | we have individual users |
| 01:49.04 | hsv-al | but all the pcf's seem to have the same |
| 01:49.08 | hsv-al | rotating case 120char+ string |
| 01:49.13 | hsv-al | groupPwd="bleh" |
| 01:49.24 | x86 | hmm |
| 01:49.29 | x86 | shrugs |
| 01:49.39 | x86 | been a while since I've set mine up honestly |
| 01:49.42 | hsv-al | thats bs, you cant import files |
| 01:49.47 | hsv-al | to it via itunes, or some other method |
| 01:49.50 | hsv-al | and "point to pcf" etc |
| 01:50.09 | x86 | yeah that'd be too cool |
| 01:50.16 | x86 | it'll never happen though : |
| 01:50.17 | x86 | :( |
| 01:50.32 | x86 | Apple is being stingy about their control over itouch / iphone |
| 01:50.50 | hsv-al | my goal is just to simply vpn to work,and get jailbreak |
| 01:50.55 | hsv-al | so i can ssh into asa's/7500's |
| 01:51.00 | hsv-al | and ios on iphone etc |
| 01:52.00 | hsv-al | x86, theres a group working on a nice voice conversion app for it, so imagine speaking IOS on the fly to your hardware :) |
| 01:52.40 | x86 | yeah that's cool ;) |
| 01:52.46 | gnutoo | hello, i've a stupid question...all my contacts have ekiga wich doesn't support encryption...i have a freebsd -7 router(piii500,320MB ram 30GB)...if i install an asterisk server on my router is there a way to make a tunnel that would encrypt the communication from the computer to my asterisk server? i know it would be difficult because sip ports are random(are they?)... |
| 01:53.36 | x86 | SIP ports aren't random, RTP ports can be at times |
| 01:53.53 | gnutoo | ok so is it possible? |
| 01:53.54 | x86 | asterisk can be configured for a range of RTP ports |
| 01:54.00 | gnutoo | wow... |
| 01:54.03 | mvanbaak | I'm off to bed |
| 01:54.09 | gnutoo | so it may be possible... |
| 01:54.11 | x86 | mvanbaak: night |
| 01:54.15 | x86 | gnutoo: sure |
| 01:54.16 | mvanbaak | latero |
| 01:54.43 | hsv-al | thats my dream for this device |
| 01:54.46 | hsv-al | full sip riding on 3G |
| 01:54.49 | gnutoo | how large the range of port have to be? |
| 01:54.54 | hsv-al | and voice commands over ssh to 7500's |
| 01:54.54 | x86 | gnutoo: if you can run asterisk with the SRTP patch(es) |
| 01:55.43 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 01:57.06 | gnutoo | x86, i can't use srtp as ekiga doesn't support it but i could run stunnel or similar software to provide the encryption tunnel....(i could run openvpn but that would gives access to my network) |
| 01:57.17 | *** join/#asterisk routerweasel (n=4stfed@core.spokanecomputing.com) |
| 01:57.33 | gnutoo | because i already run openvpn and it's configured in a way that gives access to my network |
| 01:57.55 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 01:58.36 | routerweasel | im having some trouble with LCD4LNUX on my asterisk box. I have the display working and have defined 4 widgets. But since i only have a 20x2 display, i cannot get it to scroll between widgets. all that is on the display is the first two. |
| 01:59.59 | x86 | gnutoo: I was under the assumption you were going to another server somewhere... |
| 02:00.41 | gnutoo | x86, no basically i have the following configuration: ekiga->encrypted tunnel->my asterisk server->me(ekiga) |
| 02:00.55 | gnutoo | s/have/want |
| 02:02.52 | x86 | why not just run openvpn or something? |
| 02:03.42 | *** join/#asterisk Alpha_AI (n=Ben@210.11.97.57) |
| 02:04.02 | gnutoo | x86, because i would have to run 2 openvpns...because i don't think that it's possible to remove the client-to-client directive for a particular client |
| 02:05.09 | x86 | you're either running 2 openvpns, 2 stunnels, or 2 sane SRTP-capable clients |
| 02:06.57 | gnutoo | why 2 stunnel? the problem is that i can't change the ekiga software because 1)i need video 2)some of my contacts run windows and ekiga run on windows |
| 02:07.54 | x86 | because ekiga is not capable of terminating an encrypted tunnel, as your diagram suggests |
| 02:08.27 | x86 | also check out a product called eyeBeam |
| 02:08.43 | x86 | I think it supports SRTP |
| 02:08.52 | x86 | or X-Lite might be another |
| 02:08.56 | x86 | (free) |
| 02:11.04 | gnutoo | ah ok i understand why you told 2 stunnels...that's because you think the connection between me and my router is unencrypted...but i use openvpn for 2 things...encrypt my wifi connection and road-warrior |
| 02:12.34 | gnutoo | ah great...the freebsd script supports multiple instances of openvpn...wow |
| 02:13.17 | gnutoo | mabe i'll use openvpn+asterisk server... |
| 02:13.54 | gnutoo | thanks a lot for your help |
| 02:24.26 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 02:29.21 | gnutoo | hello,does anyone runs freebsd? i asterisk want to remplace libslang2-2.1.3 by libslang-1.4.9 what should i do? |
| 02:32.24 | *** join/#asterisk coppice (n=chatzill@6.196.17.210.dyn.pacific.net.hk) |
| 02:44.23 | *** join/#asterisk crudpuppy (n=someone@71-14-97-085.dhcp.gnvl.sc.charter.com) |
| 02:45.07 | crudpuppy | anyone got an idea why incoming connections would get staticy signal(like hard pops) once connect to the asterisk system |
| 02:45.22 | Strom | crudpuppy: what kind of entrance facilities? |
| 02:45.40 | crudpuppy | Strom, I'm unsure what your asking |
| 02:45.53 | crudpuppy | incoming calls is what I ment |
| 02:45.59 | crudpuppy | other side of call only |
| 02:46.02 | crudpuppy | don't here on this side |
| 02:46.04 | Strom | what kind of circuit is the call coming in on? |
| 02:46.11 | Strom | i.e. entrance facilities |
| 02:46.12 | crudpuppy | broadvoice sip |
| 02:46.30 | crudpuppy | that what you mean? |
| 02:46.33 | Strom | yes |
| 02:46.35 | crudpuppy | sip/iax etc |
| 02:46.40 | crudpuppy | k |
| 02:47.00 | Strom | hard pops? or is it more like dropouts? how frequently do they occur? |
| 02:47.35 | crudpuppy | sounds like pops to me, but could be slight drops rather frequent at the start sometimes stops shortly after connection is made(maybe 20-30 second into call) |
| 02:47.39 | crudpuppy | sometimes does not stop |
| 02:48.00 | Strom | ...how frequently do they occur? |
| 02:48.01 | crudpuppy | voice still passes though |
| 02:48.26 | crudpuppy | seems random but close together ie as I said rather frequent |
| 02:49.01 | Strom | but is it once every five seconds? twenty times per second? |
| 02:49.12 | Strom | I really love having to dig for basic information like this |
| 02:49.12 | crudpuppy | more like 20 per sec |
| 02:49.25 | Strom | are you sharing your connection with anything else? |
| 02:49.27 | crudpuppy | sorry, not up on terminology sometimes |
| 02:49.33 | crudpuppy | yeah |
| 02:49.36 | crudpuppy | my entire network |
| 02:49.37 | crudpuppy | hehe |
| 02:49.39 | Strom | um |
| 02:49.42 | Strom | theres your problem |
| 02:49.51 | crudpuppy | so its a bandwidth issue? |
| 02:49.52 | Strom | you need to shape your outbound traffic |
| 02:49.53 | Strom | yes |
| 02:50.01 | Strom | turn off filesharing / uploading crap |
| 02:50.15 | crudpuppy | so need to get my router setup better for presendence to voip traffic etc |
| 02:50.25 | Strom | that's "precedence" |
| 02:50.33 | crudpuppy | sorry, bad spelling |
| 02:50.33 | crudpuppy | hehe |
| 02:50.57 | crudpuppy | I'm a coder so I don't have to spell correctly just incorrectly consitantly |
| 02:51.42 | Strom | stuff it. I've heard every lame excuse for "I'm too lazy to read books and learn English." |
| 02:51.59 | crudpuppy | its more of a joke then anything man |
| 02:52.16 | crudpuppy | and english as a spoken langauge is easier then written language |
| 02:52.36 | crudpuppy | your quite upity arent you |
| 02:52.56 | crudpuppy | bad day? or just always like that? |
| 02:54.02 | crudpuppy | welp, thanks for the advice anyway |
| 02:54.22 | crudpuppy | the simplest things are often forgotten |
| 02:55.11 | coppice | the simplest things are often customers |
| 02:58.36 | gnutoo | can asterisk be compiled without Newt that depends on slang1? |
| 02:59.01 | *** join/#asterisk bsaxon (n=bsaxon@220.sub-75-250-50.myvzw.com) |
| 03:00.09 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 03:07.02 | tzanger | coppice: hahahahahha |
| 03:09.43 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 03:28.37 | lucky|aba | Do i have to specify the musiconhold for each user? I am trying to set it to use a shoutcaststream but it seems whatever i do it just plays that default sound |
| 03:31.42 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:04.32 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
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| 05:10.12 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 05:10.17 | Mike8861 | hello all |
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| 05:27.18 | thansen|laptop | I'm having problems connecting to my sip provider with 1.6...can anyone lend me some testing ideas? |
| 05:29.05 | [TK]D-Fender | thansen|laptop: go look at the SIP debug for your failed calls. |
| 05:29.34 | thansen|laptop | [TK]D-Fender: I'm not sure it's even connecting...I can't dial in |
| 05:30.43 | Jameno123 | Kernel Panic: Not Syncing: Machine Check Exception: 000000000000000000000004 |
| 05:30.53 | thansen|laptop | [TK]D-Fender: how can I test if I'm even registered/connected with my provider |
| 05:31.06 | Jameno123 | well, found the reason my box dies when i load the TDM400B driver ;) |
| 05:31.13 | Jameno123 | on the Intel Xeon 5130 CPU's |
| 05:31.30 | Jameno123 | well, the reason, but not the fix |
| 05:32.15 | Jameno123 | boot the kernel with nomce (disable machine check) == resolves the kernel panic :( but not a good method |
| 05:32.50 | [TK]D-Fender | thansen|laptop: "sip show registry" |
| 05:33.31 | Jameno123 | who's the best person to talk to about Zaptel causing CPU Register corruption? :) |
| 05:33.50 | Jameno123 | or something to that extent ;) |
| 05:34.30 | Jameno123 | crash happens with the TE420P but "random" rather, than instant like on the TDM400B |
| 05:35.08 | Jameno123 | Does digium accept hardware donations to fix a problem :) :? |
| 05:36.16 | Jameno123 | be nice to get somoene the physical hardware to help diagnose the problem :( -- doesnt matter which box/mobo, reproducable on every single motherboard, with those CPU's |
| 05:36.38 | thansen|laptop | [TK]D-Fender: I just have this provider listed as a friend...how can I check that? |
| 05:36.59 | [TK]D-Fender | thansen|laptop: if they are just a "friend" then you have not registered. |
| 05:37.19 | [TK]D-Fender | thansen|laptop: "sip show registry" <- if this is blank then they will not send you calls. |
| 05:37.26 | d-k-t | Jameno123, give them a call and suggest it |
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| 05:37.46 | thansen|laptop | [TK]D-Fender: ok, I never needed to before (with 1.4), now I've lost incoming and outgoing both |
| 05:38.17 | [TK]D-Fender | thansen|laptop: pastebin the CLI output of a filed outbound attempt at verbose 10 an sip debug enabled. |
| 05:38.19 | [TK]D-Fender | ~pb |
| 05:38.20 | 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 |
| 05:38.23 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 05:43.01 | *** join/#asterisk bp423 (n=bp423@firewall-outside.thesterlingco.net) |
| 05:53.41 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 05:54.03 | thansen|laptop | [TK]D-Fender: I'm kinda with my hands tied right now...why would I be getter this on an inbound |
| 05:54.18 | thansen|laptop | <PROTECTED> |
| 05:54.18 | thansen|laptop | destination) |
| 05:54.41 | [TK]D-Fender | thansen|laptop: And the reason you have not provided what I asked for 20 minutes ago is...? |
| 05:55.04 | thansen|laptop | I'm in a data center and not sitting with a phone I can call out on |
| 05:55.24 | thansen|laptop | and I don't know how to initiate a 'call' via the cli |
| 05:55.46 | [TK]D-Fender | thansen|laptop: Connect with a soft-phone then. "help originate" |
| 05:55.47 | Mike8861 | thansen|laptop: use the DIAL command |
| 05:55.58 | [TK]D-Fender | thansen|laptop: "help dial" |
| 05:56.03 | bp423 | where can I find good documentation on configuring 1.6 as realtime |
| 05:56.54 | [TK]D-Fender | bp423: the docs included with it, and 1.4 as a reference : |
| 05:56.56 | [TK]D-Fender | ~book |
| 05:56.57 | jbot | 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://tfot.leifmadsen.com, or see ~buybook |
| 05:56.58 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 05:57.27 | bp423 | awsome thanks |
| 05:58.21 | Mike8861 | thansen|laptop: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 05:58.55 | Mike8861 | [TK]D-Fender: do u use DIDww ? |
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| 05:59.22 | [TK]D-Fender | Mike8861: no |
| 05:59.34 | *** join/#asterisk blq (i=Bl@dslb-088-064-158-017.pools.arcor-ip.net) |
| 06:00.03 | Mike8861 | [TK]D-Fender: the account is not even activated, however tracstion are completed. |
| 06:01.25 | Mike8861 | [TK]D-Fender: which DID orgaination provider u recommand ? |
| 06:01.48 | [TK]D-Fender | Mike8861: Teliax & les.net |
| 06:03.02 | Mike8861 | [TK]D-Fender: thank you |
| 06:07.22 | Mike8861 | [TK]D-Fender: les.net only offer Canada and US DID ?? |
| 06:07.40 | [TK]D-Fender | Mike8861: what do they say? |
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| 06:13.59 | Mike8861 | [TK]D-Fender: IPDID USA & IPDID Canda as they say |
| 06:16.33 | [TK]D-Fender | Mike8861: Mike8861 Good, you can read it seems. |
| 06:16.55 | [TK]D-Fender | Mike8861: I have no advice for non-North American DID's |
| 06:17.17 | [TK]D-Fender | Mike8861: Go look up the WIKi for providers and see if you can Google up recommendations. |
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| 06:23.11 | Mike8861 | [TK]D-Fender: thank you. |
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| 07:08.25 | Ast001 | if I see this [Jul 13 08:58:30] ERROR[28176]: chan_zap.c:9470 start_pri: Unable to open D-channel 16 (Device or resource busy) |
| 07:08.44 | Ast001 | when I try module load chan_zap.so |
| 07:09.12 | Ast001 | does it mean my card is bad ? |
| 07:11.48 | MCooper | anyone here familiar with AGI PHP MySQL stuff.... |
| 07:12.16 | x86 | PHP sucks |
| 07:12.24 | x86 | perl is much better for AGI |
| 07:12.25 | Maliuta | amen |
| 07:12.27 | MCooper | x86, Ok... |
| 07:12.44 | MCooper | I am working what I got.... |
| 07:13.07 | MCooper | I am passing stuff to the PHP, but it appears I am not getting information back... |
| 07:13.24 | Maliuta | anyone who thinks php is appropriate for anything other than web app prototyping needs to be shot |
| 07:14.13 | MCooper | Maliuta, Again.. I am picking up the project that is 6 months behind.... |
| 07:14.30 | MCooper | I just need to bandage it until I can get the new stuff written.. that is all |
| 07:15.08 | x86 | prolly be faster to re-write |
| 07:15.11 | x86 | chop chop |
| 07:23.40 | *** join/#asterisk Gnutoo (n=gnutoo@host54-133-dynamic.41-79-r.retail.telecomitalia.it) |
| 07:24.43 | Gnutoo | hello, can i put 2 bind address in sip.conf? because it would simplify a lot... |
| 07:24.57 | Gnutoo | because otherwise we have to be 2 on the vpn |
| 07:25.50 | Gnutoo | i need it so i can have: ekiga->openvpn->asterisk-server->wifi's openvpn->me |
| 07:26.10 | Gnutoo | and of course openvpn and wifi's openvpn aren't on the same network |
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| 07:35.17 | *** mode/#asterisk [+o mog] by ChanServ |
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| 07:49.02 | Gnutoo | anyone? by the way with the default config i have that: Jul 13 09:48:39 router kernel: pid 34411 (asterisk), uid 0: exited on signal 11 (core dumped) mabe a bad config? |
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| 07:59.56 | Keypad | If you have a SPA-3102 is there a program you can download to use it like a phone as long as your attached to the network ? |
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| 08:18.11 | d-k-t-2 | ~centos52bug |
| 08:18.12 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
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| 09:29.07 | MCooper | I am having a little trouble with my dialplan - wondering if someone might have a look at it? |
| 09:29.31 | MCooper | Its rough, but functional... |
| 09:29.38 | *** part/#asterisk Keypad (n=Keypad@125.238.132.197) |
| 09:32.59 | lesouvage | MCooper: past it on psatebn somewhere please (f.i. www.pastebin.be) |
| 09:33.15 | lesouvage | psatebin=pastebin |
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| 09:35.46 | lesouvage | MCooper: If your project is overtime and under presure, make sure that you have ringing phones as soon as possible. With just a few lines you can offer a dialplan that cover must of the basic requirements ad ringing phones make other people happy ;-) |
| 09:39.34 | MCooper | lesouvage, http://pastebin.com/m32dc12f6 |
| 09:39.57 | MCooper | lesouvage, That is just part of it.. and yes.. .i have commented out the offending line to make sure phones ring. |
| 09:41.24 | jblack | people can tolerate a lot, so long as phone, email and http work. |
| 09:41.37 | MCooper | jblack, Yes they can... |
| 09:42.31 | MCooper | The thought is really simple.. call 130, and if you are blacklisted hang up... and not ring... The BLACKLIST is coming from an MySQL database |
| 09:46.16 | lesouvage | MCooper: how long is the blacklist |
| 09:47.34 | MCooper | one number now.. and it returns either 1 or 0 |
| 09:48.00 | MCooper | 1 is continue on, and 0 is the number is in the blacklist. |
| 09:53.27 | lesouvage | MCooper: just an idea to save your ass. Forn now forget about mysql, forget about agi scripts and all the ther things that makes it hard to debug. Go for the basics with a basic dialplan and a basic blacklist maganism. Yo can finish this within half an hour. |
| 09:53.57 | MCooper | what is your suggestion? |
| 09:54.26 | MCooper | I mean I have seen the asterisk database.. but we need to be able to log the # of tires |
| 09:54.51 | MCooper | if the tries go over 5 with 1 hour, we block it. |
| 09:55.28 | *** join/#asterisk [netman] (n=netman@68.Red-88-8-164.dynamicIP.rima-tde.net) |
| 09:55.43 | lesouvage | MCooper: How many call are we taking about? |
| 09:56.17 | MCooper | thousands |
| 09:56.35 | MCooper | this is the handle wardialing by insurgents.... |
| 09:56.40 | MCooper | to a tips line... |
| 09:56.55 | MCooper | (Yes I am serious) |
| 09:57.08 | lesouvage | I checked what you paste but I can not make anything out of it. It depends on what is in the php script. |
| 09:57.24 | *** join/#asterisk ikevin (n=kevin@ANancy-256-1-2-229.w90-6.abo.wanadoo.fr) |
| 09:57.33 | ikevin | morning |
| 09:57.53 | MCooper | I can post the PHP.... (not the first way I would have written it..) |
| 09:58.03 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:58.06 | lesouvage | What is more important: to have it up and running asap or the feature to blacklist someone who tries more then 5 times in an hour. |
| 09:58.17 | ikevin | i have a problem while i receive a sip call from my voip provider |
| 09:58.17 | MCooper | blacklisting. |
| 09:58.26 | ikevin | i receive this log: chan_sip.c:13952 handle_request_invite: Call from 'illu88' to extension 's' rejected because extension not found. |
| 09:58.46 | MCooper | because even if it is up.. they are calling it with 1000+ numbers over and over again.. and overwhelming the system |
| 09:58.50 | lesouvage | MCooper: I'm not a php man o I will not be of much help |
| 09:58.52 | ikevin | i've tryed to create an extension called s in the extensions.conf so that don't work |
| 09:59.01 | ikevin | anyone have an idea? |
| 09:59.09 | MCooper | lesouvage, Neither am I... haha |
| 10:00.30 | lesouvage | ikevin: start with exten => s,1,Answer() and then exten => s,n,Dial(SIP/<your internal sip phone>,30,r) and then exten => s,n,Hanhup(). It should work if you add this to the proper context. |
| 10:00.40 | MCooper | what I am curious about is how to show what the Variable is in the dialplan. Is there a way to display variables? |
| 10:01.58 | lesouvage | MCooper exten = s, n,NoOp(This is the value of the variable I want to show ${variableyou_want_to_show}) will show the value in the cli |
| 10:03.42 | MCooper | Ok... |
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| 10:08.32 | MCooper | lesouvage, http://pastebin.com/mbf4ab0e |
| 10:08.45 | MCooper | That is the output from the CLI |
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| 10:14.22 | ikevin | i use that on a macro: exten => s,1,Answer() exten => s,2,Dial(${ARG2},20,r) exten => s,3,Goto(s-${DIALSTATUS},1) |
| 10:14.54 | ikevin | and have that on my context: exten => illu88,1,Macro(stdexten,304,SIP/kevin) |
| 10:15.25 | lesouvage | Is this a blacklisted number? btw, the ringing line looks a bit strange to me. shouldn't there be a dial statement so the inbound channel is bridged to a channel of an internal phone? The cli output doesn't give lots of useful info. |
| 10:15.45 | lesouvage | or a queue |
| 10:16.08 | MCooper | If they number is allowed, we have a queue that it goes to... if it is a bad number we hangup |
| 10:21.13 | lesouvage | But now the agi script returns 0 and I assume that if you tried to call 5 times in a row this ought to change to 1 and based on that the hangup() is executed. I don't understand the line exten => 130,n(keep_going),Ringing . |
| 10:22.55 | MCooper | That is can actually go away.. |
| 10:23.22 | MCooper | I am not sure what it was doing there... i am taking over a project that is 6 months behind,... Lucky me. |
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| 10:35.11 | lesouvage | MCooper: is keeping records of all the numbers that has been blocked relevant or is it just important to block those that try 5 times in an hour. |
| 10:37.41 | MCooper | No its relevent |
| 10:37.50 | MCooper | relevant |
| 10:37.56 | MCooper | We have to track the numbers. |
| 10:41.09 | pputman- | MCooper, so the problem is it's not hanging up? |
| 10:41.13 | pputman- | or it's taking a while to hang up? |
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| 10:41.55 | Mike8861 | hello |
| 10:47.01 | lesouvage | MCooper: My advice is to hire someone to fix the problem. I'm afraid it is to complicated to have it done based on info on the irc channel. I can fix a solution but I can't do that for free. |
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| 11:02.17 | *** join/#asterisk nomad_cz (n=michal@r6db65.net.upc.cz) |
| 11:03.35 | nomad_cz | Hi. Could anyone tell me how the h extension is supposed to work ? In macro it sometimes jump to h defined there and sometimes to h in calling context :/ |
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| 11:21.15 | ikevin | can i put a waiting music while phone is ringing? |
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| 11:30.58 | nomad_cz | ikevin, I guess so |
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| 11:32.01 | nomad_cz | ikevin: check out dial() options I think there is 'm' option for playing music |
| 11:32.53 | ikevin | ok thx |
| 11:33.16 | nomad_cz | ikevin: u r welcome ;) |
| 11:33.38 | ikevin | :) |
| 11:46.49 | angryuser | hello al |
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| 11:50.40 | angryuser | i am searching a simple solution for a case when internet is down, i would like asterisk switch for ISDN lines then, i imagined a simple script of ping of next gateway, and when it is down ==> write a value in astDB and verify it before call, is there any simplier soution ? |
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| 12:36.40 | Gnutoo | hello, asterisk has a segmentation problem....where should i look for resolving the problem... |
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| 12:39.36 | MaliutaLap | segfault? in your compilation |
| 12:42.11 | Gnutoo | ah running it manually doesn't make him segfault |
| 12:42.38 | Gnutoo | mabe it's just that it doesn't find the config file in daemon mode.... |
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| 12:46.10 | Gnutoo | by the way how do i activate sip.conf? |
| 12:46.31 | Gnutoo | because it doesn't seems to be binding on the port that is defined inside sip.conf |
| 12:46.47 | tompaw | Morning! |
| 12:48.37 | Gnutoo | ah i made it segfault trying to connect to it with asterisk -r or asterisk -R |
| 12:51.57 | Gnutoo | http://pastebin.com/m4457cd8f |
| 12:52.37 | Gnutoo | but you can't differenciate the segfault when starting it and when connecting to it in this log |
| 12:52.51 | Gnutoo | MaliutaLap, no it's at runtime |
| 12:53.16 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 12:53.37 | MaliutaLap | segfaults are normally the result of something done wrong at compile time |
| 12:53.59 | Gnutoo | MaliutaLap, ah ok so what should i do? i realy need asterisk |
| 12:54.48 | MaliutaLap | have you hand compiled it? |
| 12:55.29 | MaliutaLap | i.e. you're not using a precompiled, possible packaged, binary? |
| 12:55.51 | Gnutoo | MaliutaLap, no i've used freebsd-7 ports to compile it |
| 12:57.03 | MaliutaLap | you can compile it again, but before you do make sure you have everything you need in place first |
| 12:57.16 | MaliutaLap | to enable sip look at modules.conf |
| 12:57.53 | Gnutoo | by the way asterisk -r tells me that: "Broken pipe" |
| 12:58.39 | tompaw | which command restart asterisk's web gui? |
| 13:00.23 | MaliutaLap | tompaw: read the topic. then go to #asterisk-gui |
| 13:01.05 | MaliutaLap | Gnutoo: you've fecked up the compile, build it properly |
| 13:01.21 | Gnutoo | MaliutaLap, how do i do that? |
| 13:02.14 | *** join/#asterisk k3mp (n=k3mp@p5B39653C.dip.t-dialin.net) |
| 13:03.19 | k3mp | hello there, could someone support me configuring my asterisk 1.4 with mysql? i got problems with my mysql-based-dialplan. |
| 13:03.34 | Gnutoo | ouch,,,,[Jul 13 15:03:10] WARNING[35421]: loader.c:416 load_dynamic_module: Error loading module 'chan_sip.so': /usr/local/lib/asterisk/modules/chan_sip.so: Undefined symbol "ast_park_call" |
| 13:04.30 | k3mp | hello there, could someone support me configuring my asterisk 1.4 with mysql? i got problems with my mysql-based-dialplan. |
| 13:06.17 | Gnutoo | i update my ports |
| 13:06.30 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 13:09.06 | MaliutaLap | Gnutoo: read the docs, it's all covered there |
| 13:09.09 | MaliutaLap | ~docs |
| 13:09.09 | jbot | [~docs] Asterisk documentation can be found at http://www.digium.com/index.php?menu=documentation , http://www.asteriskdocs.org , http://www.asteriskguru.com , the WIKI (~wiki), or the BOOK (~book) |
| 13:09.27 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:12.34 | MaliutaLap | alright, sleepy time |
| 13:13.09 | Gnutoo | thanks a lot |
| 13:18.05 | tompaw | Damn, I thought Digium sells G729 using some automated platform. |
| 13:20.57 | tompaw | What's with those http://asterisk.hosting.lv/ codecs? Do they achieve the same performance as those binary ones sold by Digium? |
| 13:21.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:25.47 | *** join/#asterisk bsaxon (n=bsaxon@119.sub-75-202-208.myvzw.com) |
| 13:26.09 | coppice | they've been optimised by Intel, but Intel has not always proven to be very good at performance optimisation of DSP code. |
| 13:26.38 | *** join/#asterisk ManxPower (n=manxpowe@201.sub-75-202-187.myvzw.com) |
| 13:29.40 | angryuser | has anyone used skip2pbx ? they clain to convert sip to skype and vise versa, their price is hight also |
| 13:29.46 | angryuser | claim* |
| 13:31.22 | tzafrir_laptop | the codec itself is in a binary blob you need to install from intel separately, IIRC |
| 13:31.34 | tzafrir_laptop | (the Asterisk module is merely a wrapper around it) |
| 13:31.52 | tzafrir_laptop | tompaw, ==^ |
| 13:32.50 | *** join/#asterisk simprix (n=simprix@65.209.146.209) |
| 13:34.42 | ManxPower | tzafrir_laptop: The illegal "G729 codec"? |
| 13:36.37 | tzafrir_laptop | ManxPower, it is a g729 (and g723) codec. And yes: requires a license that most people who consider it "free" dismiss for some strange reason |
| 13:38.23 | ManxPower | tzafrir_laptop: As I understand it, Intel has a library that does G729/G723. Intel's library can be licensed free for non-commercial use, but (and the Intel license even says this), the license is only for the Intel code, not the codecs themselves, which are licensed separately. |
| 13:40.22 | ManxPower | tzafrir_laptop: About once a year I report the asterisk unlicensed G729 web site to the patent holders of G729 |
| 13:43.42 | *** join/#asterisk aksyn (n=aksyn@jayfenton.plus.com) |
| 13:54.41 | coppice | I don't think there is any action they can take against a web site like that. |
| 13:56.02 | ManxPower | OT: I'm looking for recommendations for a phone that has a real (even if tiny) keyboard and supports SSH over some form of high speed internet access. /msg me so we don't clutter the channel |
| 13:57.16 | coppice | do you mean a cellphone? |
| 13:58.07 | ManxPower | coppice: or "mobile" as they say in your part of the world. |
| 13:58.32 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 13:58.49 | coppice | do we? that's news |
| 13:59.30 | coppice | i thought a mobile was something that fluttered above a baby's cot |
| 13:59.34 | ManxPower | I just assumed as HK is a former UK area |
| 14:00.36 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 14:01.14 | *** join/#asterisk zapp-branigan (n=malebolg@9.218.216.87.static.jazztel.es) |
| 14:03.13 | zapp-branigan | hi i have a problen registering an ata spa3102 in asterisk every time 4 times after a register he receive diferents nonce in the Register message |
| 14:03.29 | zapp-branigan | and in the code in the asterisk do : |
| 14:03.52 | zapp-branigan | Verify nonce from request matches our nonce. If not, send 401 with new nonce */ |
| 14:04.09 | zapp-branigan | function static enum check_auth_result check_aut |
| 14:04.21 | zapp-branigan | but this is important ? |
| 14:05.00 | zapp-branigan | because i have remover the check of nonce and is most farter |
| 14:05.35 | zapp-branigan | somebody know what i'm speaking ? |
| 14:06.48 | zapp-branigan | the challenge who i send and i receive in the sip protocol in the register is diferent |
| 14:07.02 | zapp-branigan | but the response is true |
| 14:07.03 | Kyoshi | Dialplan help with realtime please... http://pastebin.com/m1c339312 |
| 14:09.13 | ManxPower | Kyoshi: I don't see an actual problem |
| 14:09.43 | ManxPower | In any case, I can't really help with Realtime issues |
| 14:10.53 | Kyoshi | then you dont see the question |
| 14:11.56 | ManxPower | well, what DO you know about the include statement. |
| 14:11.58 | Kyoshi | an INCLUDE statement is not defined the same but i think i just got it. |
| 14:12.12 | Kyoshi | i will be trying something |
| 14:12.39 | Kyoshi | dude.. what i know about the include statement is prolly the same aS you, includes another context's dialplan info |
| 14:12.44 | Kyoshi | thats NOT the problem |
| 14:12.58 | Kyoshi | what IS the problem is how to put that INCLUDE statement into the ARTDB |
| 14:13.07 | Kyoshi | or ARADB |
| 14:13.40 | ManxPower | Using Realtime makes things to very much more complicated. |
| 14:14.01 | Kyoshi | considering usually there is ext, pri, app, args, an include only has include and context... doesnt exactly match the pattern |
| 14:14.11 | Kyoshi | really, much more complicated? |
| 14:14.38 | Kyoshi | well sometimes you need realtime for your application to work, even if it's way beyond your understanding |
| 14:15.00 | *** join/#asterisk simprix (n=simprix@65.209.146.221) |
| 14:15.09 | Kyoshi | usually if you answer a question in here it means you have some knowledge of the subject matter, why you answered me is a mystery |
| 14:15.46 | ManxPower | Just wondering why you would put the dialplan into a database. |
| 14:16.55 | ManxPower | None of the 2,470 hits of a google search of the mailinglist archive was not helpful? |
| 14:17.51 | ManxPower | http://www.google.com/search?hl=en&q=site%3Alists.digium.com+realtime+%22include+%3D%3E%22&btnG=Google+Search |
| 14:17.57 | ManxPower | ~mailinglist |
| 14:17.57 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 14:18.01 | Kyoshi | none at all because they either describe how to create the database, or what its for |
| 14:18.23 | Kyoshi | not how to import an include statement into an aradb |
| 14:18.27 | ManxPower | But many of them say you cannot use include => with Realtime |
| 14:19.51 | Kyoshi | manxpower: the reason you may want SIP users into a DB is to dynamically be able to change user info on the fly, without having to reload. the reason you have diaplans in a DB is the same reason. also aradb means NO config files for thosed items are preloaded into memory and when you will have over 1M dialplan functions and nearly 1M users, it helps conserve memory |
| 14:20.03 | Kyoshi | well if i cannot, then i cannot is all |
| 14:22.07 | ManxPower | With a larger userbase, it can be helpful. Most people seem to use Realtime only because of the "coolness factor" |
| 14:32.06 | tompaw | sorry if that's a stupid question, but are the codecs from http://asterisk.hosting.lv legit or not? |
| 14:32.15 | tompaw | I just installed g729 and it seems to be working fine. |
| 14:34.19 | lesouvage | MCooper: are you still around? |
| 14:36.04 | Kyoshi | manx, seems like you assumed my need |
| 14:36.13 | Kyoshi | sorry but i do NEED it |
| 14:36.24 | Kyoshi | dynamic users and dialplans are a HUGE factor of my work |
| 14:58.07 | angryuser | i installe it for coolness |
| 14:58.15 | angryuser | mhahaha ;) |
| 15:03.30 | tzafrir_laptop | tompaw, at least in most contries: no |
| 15:09.21 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
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| 15:18.50 | tompaw | tzafrir_laptop: ok, what is the legit way to get g729 on freebsd then? |
| 15:29.22 | tompaw | anybode here using a2billing? |
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| 16:06.16 | tompaw | I need an advice - which way is better regarding assigning extension numbers: http://www.tompaw.pl/numbers.TXT ? |
| 16:08.32 | [TK]D-Fender | tompaw: How many locations? How many users each? |
| 16:09.16 | tompaw | [TK]D-Fender: let's say... 6 locations, mostly up to 100 users, but one must be able to serve thousands. |
| 16:11.15 | [TK]D-Fender | tompaw: then XXYYYY - XX - Site #, YYYY= logical exten |
| 16:11.44 | [TK]D-Fender | tompaw: 1 less digit than yours |
| 16:12.47 | tompaw | [TK]D-Fender: so in other words - local extension numbers, right? |
| 16:14.06 | [TK]D-Fender | tompaw: yes |
| 16:14.23 | tompaw | [TK]D-Fender: just out of curiosity - why? |
| 16:14.28 | *** join/#asterisk th1 (n=th@pdpc/supporter/sustaining/th1) |
| 16:14.35 | [TK]D-Fender | tompaw: Why make yours bigger? |
| 16:14.40 | MCooper | lesouvage, I am here.,,,, |
| 16:14.46 | tompaw | [TK]D-Fender: fair enough ;-) |
| 16:14.57 | [TK]D-Fender | tompaw: One might figure you want to handle dialing lengths in a consistent way. |
| 16:15.22 | [TK]D-Fender | tompaw: And throwing in an extra digit doesn't seem to be offering your anything |
| 16:15.37 | tompaw | [TK]D-Fender: right, and with the 'local' attempt I can have extensions length variable! |
| 16:16.13 | MCooper | To everyone that has had to listen to newbie questions, and misunderstandings on my part.. thank you... we got the project to its first milestone... pputman / pputman- was very instrumental as was lesouvage (Thank you for the code) and others... Thank you... |
| 16:16.51 | [TK]D-Fender | tompaw: You can do whatever you want. Technically you could just do "_*XX.", and assume anything starting with a "*" is off-site, take 2 digits to figure out which site, and pass a variable length # to it. |
| 16:17.02 | [TK]D-Fender | tompaw: Based on digits from the 4th onwards |
| 16:19.58 | tompaw | [TK]D-Fender: great, thanks for the explaination. |
| 16:20.46 | tompaw | [TK]D-Fender: I wasn't trying to achieve anything extraordinary with the global one, it's just that I don't have any experience with * yet, and I wanted to confirm if I won't get in trouble by local extensions. |
| 16:24.50 | *** join/#asterisk moy (n=moy@189.169.86.155) |
| 16:26.09 | tompaw | regarding dialplans. if - during a call - asterisk goes through all defined exten = blahs.... and doesn't find a match, it then STILL calls the local extension (if there is one equal to EXTEN called), right? |
| 16:32.39 | [TK]D-Fender | tompaw: No, * does EXACTLY what you tell it to. |
| 16:32.50 | [TK]D-Fender | tompaw: If you don't have a match then you get nothing. |
| 16:33.13 | [TK]D-Fender | tompaw: there is not "automatic" anything |
| 16:35.53 | tompaw | [TK]D-Fender: I guess I shouldn't bring this subject in here, but I am reconfiguring one *Now installation manually changing extensions.conf. I defined a dialplan with only 2 extens (_4! and 553). |
| 16:36.16 | tompaw | I assigned my SIP account to this dialplan, yet still if I dial 6050 for example, I am able to reach my voicemail. |
| 16:36.59 | tompaw | (the other 2 extensions also DO work) |
| 16:39.10 | [TK]D-Fender | tompaw: GUI's own your ass |
| 16:40.59 | mvanbaak | gui's are evil |
| 16:42.15 | tompaw | gui doesn't even work no more since I messed the .confs myself ;-) |
| 16:45.57 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.143) |
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| 16:58.07 | lesouvage | gui's will fill your days with debugging until the end of days |
| 16:59.41 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 17:00.53 | lesouvage | tompaw: just out of curiousity, you are implementing Asterisk on 6 location with 100 phones each and this is your first Asterisk project? |
| 17:05.07 | tompaw | lesouvage: I am re-configuring one * which was based on *Now. |
| 17:05.13 | tompaw | lesouvage: yes. |
| 17:07.20 | lesouvage | Tompaw: It is just a suggestion but I think you save yourself a lot of trouble if you use a fresh asterisk install as a starting point instead of an Asterisk Now based asterisk box. |
| 17:15.18 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 17:16.15 | tompaw | lesouvage: thanks, will do. |
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| 17:34.57 | tzafrir_laptop | anybody feels like adapting a silly script to FreeBSD / OpenBSD / Solaris / whatever? |
| 17:35.13 | tzafrir_laptop | http://bugs.digium.com/13065 |
| 17:35.38 | ManxPower | tompaw: as I understand it, Digium has a FreeBSD G729 codec |
| 17:37.42 | ManxPower | Oh look! Digium G729 codec for Solaris 10, FreeBSD 6.1 and 7.0! |
| 17:40.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 17:47.44 | [TK]D-Fender | load chan_zomgmaybeishouldlookalittleharder.so! |
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| 18:12.04 | jeev | stabs Fender |
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| 18:27.43 | [TK]D-Fender | shoots jeev |
| 18:27.54 | [TK]D-Fender | jeev: Never bring a knife to a gun fight :p |
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| 18:42.38 | jeev | Fender |
| 18:42.40 | jeev | i;'m wearing a vest |
| 18:42.42 | jeev | you weren't! |
| 18:43.40 | [TK]D-Fender | jeev: thats why I shoot for the face :) |
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| 18:44.19 | jeev | i'm wearing a roman helmet |
| 18:45.26 | jeev | fender, i'm so tired andi t's freakin 12 |
| 18:45.30 | jeev | i dunno what to do man |
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| 18:45.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:46.29 | [TK]D-Fender | jeev: Get your hands off your nuts.... and SEIZE THE DAY! |
| 18:47.14 | jeev | how could i seize the day? the only 2 options i have right now, stay home and do nothing.. or go to my little cousins house where there's an endless supply of calories (because they own 2 baskin robbins) |
| 18:47.58 | [TK]D-Fender | jeev: You might want to consider drugs. Your imagination is so remarkably limited it seems. Careful not to overdo it though... |
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| 18:48.18 | [TK]D-Fender | jeev: Expand your horizons, not your waist-line! |
| 18:49.12 | jeev | you suck, i've never drank or smoked.. what makes you think i'd try drugs |
| 18:49.16 | jeev | i COULD fly to vegas but my friend is here so |
| 18:49.18 | jeev | i have no options |
| 18:49.30 | jeev | i dont want to expand my waist line since i need to decrease it like 5 lbs and have been saying it for 2 years now |
| 18:49.47 | [TK]D-Fender | jeev: Go outside and do something |
| 18:50.02 | [TK]D-Fender | jeev: Get some exercise. Go see a show or something. |
| 18:50.24 | [TK]D-Fender | jeev: if the weather were better here I'd be off biking or boating |
| 18:50.36 | jeev | i ahven't exercised since the beginning of this year and was for a basketball tournament |
| 18:50.41 | jeev | i want to bike but nobody else wants to |
| 18:50.47 | jeev | my mountain bike has been sittin here gathering rust |
| 18:50.52 | jeev | or rusting shall i say |
| 18:50.56 | jeev | i want to go boxing but i'm too lazy |
| 18:51.05 | jeev | i _cant_ exercise right now, i'm not in shape to even start! |
| 18:51.23 | [TK]D-Fender | jeev: First step to changing your life is.... change your life. |
| 18:51.43 | [TK]D-Fender | jeev: I'm about to head out and do groceries before the GF heads over for dinner & movie. |
| 18:51.47 | jeev | bah |
| 18:51.58 | jeev | my gf is going back to school today.. i barely get to see her, her studies = suck |
| 18:52.13 | [TK]D-Fender | jeev: and my GF's psycho call-ceenter hours suck |
| 18:52.21 | jeev | damn |
| 18:52.41 | [TK]D-Fender | jeev: fast & hard lesson : Life sucks, but rarely swallows |
| 18:53.02 | jeev | mine just goes through a very tough time.. in school, she is making up classes she had to suspend during her last quarter since her mom passed away |
| 18:53.07 | jeev | so she's always in school :/ |
| 18:54.35 | jeev | i wish i could go ride the bike now |
| 18:54.40 | jeev | then i'd go back to boxing again bah! |
| 18:55.17 | jeev | and my boxing gym is right next door to my office.. like i walk out of the building, cross 20 foot street <15 mph traffic and viola! |
| 18:57.51 | jeev | you left me |
| 18:57.52 | jeev | :;/ |
| 18:58.56 | implicit | i found a pretty nice bug that's been around in chan_sip for a LONG time :) |
| 18:59.52 | tzafrir_laptop | So let it lurk there |
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| 19:00.05 | tzafrir_laptop | is from the bug protection league |
| 19:00.08 | implicit | it has to do w/ an call that comes in and is Dialed back out, when asterisk is behind a NAT and both source and destination have comedia support |
| 19:00.24 | tzafrir_laptop | (sp) |
| 19:01.18 | implicit | asterisk doesn't send any RTP back after the SDP is received from first leg OR any RTP out to the destination |
| 19:01.33 | implicit | so they can never send RTP back (cause it has to be symmetric to get past the NAT) |
| 19:01.39 | implicit | and both ends of the call get no audio |
| 19:01.53 | implicit | if you do the same call but you do a Playback(something|noanswer) before Dial() it would work |
| 19:02.02 | implicit | :) |
| 19:02.51 | implicit | Asterisk needs to send out silence after it gets the SDP from the first and second legs |
| 19:03.12 | implicit | doing it for one leg is actually sufficient but both legs is correct and reduces RTP setup lag-time |
| 19:03.33 | implicit | ideally it would just not touch the SDP and send it out to the next INVITE, because it knows it doesn't need to touch it |
| 19:03.57 | implicit | but if it wants to use the RTP, that's fine, it can still relay it when it's behind a NAT, just needs to send rtp as soon as it can |
| 19:04.01 | implicit | and not wait for the other leg |
| 19:04.30 | implicit | i think that's why a lot of people end up using the 'r' option when behind NAT on sip calls |
| 19:04.31 | implicit | cause of that bug |
| 19:04.35 | implicit | otherwise they get no audio |
| 19:04.51 | implicit | if they haven't done anything w/ the audio on the call before Dial() |
| 19:06.05 | implicit | i was helping one of my sip trunking clients who had this problem and was behind a nat the other day and figured it out |
| 19:06.36 | implicit | have to have them play a second of silence with noanswer before sending the call back out to make audio work |
| 19:07.01 | implicit | tzafrir_laptop: i don't think this has been reported has it? |
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| 19:30.26 | Aquahallic | afternoon folks |
| 19:30.31 | mvanbaak | hey |
| 19:30.53 | Aquahallic | anyone played much with the unistim channel for asterisk? |
| 19:34.25 | Aquahallic | well.. maybe I can state it like this... cause I actually have the nortel phone working properly... just can't call to other extensions |
| 19:35.26 | *** join/#asterisk jack_sparo (n=eddy@91.73.203.98) |
| 19:35.30 | jack_sparo | hey guys, where is this file located? iax2.h anybody knows? |
| 19:35.49 | Aquahallic | I saw a couple wiki's and setup a firefly phone via SIP and registered it... edited sip.conf and extensions.conf (put the extension in a new context and pointed to that context in the sip.conf) |
| 19:36.53 | Aquahallic | then I installed the unistim channel driver and edited the unistim.conf with the phone info and then extensions.conf with the extension info in the same context as the firefly phone |
| 19:37.52 | Aquahallic | from the nortel softphone I can dial my own extension and it will say busy... but if I dial the extension of the firefly phone it says it's an invalid extension yet I can dial the firefly phone from another firefly phone setup the same way on my network |
| 19:38.54 | Aquahallic | so.. the nortel phone can dial itself fine.. and dial to like vmail... but it can't dial to the other sip phones on the network |
| 19:38.55 | *** join/#asterisk tinloaf (n=tinloaf@d90-134-96-21.cust.tele2.de) |
| 19:40.14 | tinloaf | hi. i have this setup: an asterisk box with an ISDN-card. i connect via SIP to this box and then dial out into PSTN. the idsn card is an AVM Fritz!Card. following problem ocurrs: while i'm talking i can't hear what the person on the other end hears. is this an asterisk problem? |
| 19:40.28 | Strom_C | tinloaf: sounds like a SIP/NAT problem |
| 19:40.30 | Strom_C | ~sipnat |
| 19:40.31 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:40.45 | tinloaf | thanks |
| 19:41.14 | tinloaf | NAT can mute incoming traffic if there is outgoing traffic? |
| 19:44.56 | tinloaf | i mean - i can hear incoming voice when i'm not talking, so it can't be a general NAT problem, can it? |
| 19:45.21 | kamanashisroy | Hi, my agi script is not working in asterisk 1.2 though I tested it working in 1.4 .. can I get any change log or something ? |
| 19:47.22 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 19:53.47 | mog | kamanashisroy, there is a changelog in in 1.4 callled changelog... |
| 19:54.08 | kamanashisroy | mog: thanks .. watching ... |
| 20:00.26 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:01.58 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 20:03.24 | mchou | are there tools to block calls (such as from telemarketers/800#) via asterisk? |
| 20:04.18 | mog | mchou, look at ex-girlfriend example |
| 20:04.21 | mog | and zapateller |
| 20:04.51 | mchou | ~ex-girlfriend |
| 20:05.13 | mchou | mog, is ex-gf example in the book? |
| 20:05.35 | mog | and voip-info, and the sample docs |
| 20:05.50 | mog | you can route calls based on caller id or lack their of |
| 20:06.38 | mchou | mog: alright, thx |
| 20:06.51 | mog | np, if you cant find it ill look it up for ya |
| 20:10.47 | *** join/#asterisk nephfl (n=no@wsip-68-110-130-57.ga.at.cox.net) |
| 20:11.16 | nephfl | I have a tdm400 card, i can receive incoming but when i try to dial out my whole system locks, any ideas what might cause that? |
| 20:11.33 | *** join/#asterisk implicit_ (n=bayan@ip72-211-221-209.oc.oc.cox.net) |
| 20:23.35 | kamanashisroy | hi [TK]D-Fender |
| 20:24.20 | [TK]D-Fender | mchou: "core show function CALLERID" , "core show application gotoif" |
| 20:24.26 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 20:25.30 | kamanashisroy | [TK]D-Fender: you are drunk :D |
| 20:25.51 | [TK]D-Fender | kamanashisroy: ? |
| 20:26.42 | kamanashisroy | [TK]D-Fender: no you are not .. :-P .. found mchou's problem :) |
| 20:39.17 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 20:41.03 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 20:49.09 | *** join/#asterisk angryuser (n=angryuse@88.140.144.209) |
| 20:49.14 | angryuser | hello |
| 20:49.38 | angryuser | cant remember the diff from type=user and type=friend ... same thing ? |
| 20:49.53 | sartan | what am i doing on #asterisk |
| 20:49.54 | *** part/#asterisk sartan (n=JP@S0106000f66a59cb0.cg.shawcable.net) |
| 20:50.17 | angryuser | ok remembered |
| 20:50.30 | angryuser | ;) memory holes |
| 20:55.51 | *** join/#asterisk stevie_ramjet (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 20:55.51 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 20:59.25 | *** part/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 21:07.38 | drfreeze | Hello |
| 21:08.03 | drfreeze | I'm having a bit of trouble getting the vsftpd setup working |
| 21:08.31 | drfreeze | For a default polycom setup, is the username PlcmSpIp or polycom? |
| 21:08.32 | *** join/#asterisk th1 (n=th@pdpc/supporter/sustaining/th1) |
| 21:09.05 | drfreeze | [TK]D-Fender: hi |
| 21:10.43 | drfreeze | anyone around? |
| 21:15.23 | [TK]D-Fender | drfreeze: the former. Mind you syou should be setting your own user/pass for these |
| 21:17.28 | drfreeze | [TK]D-Fender: I usually do that, but this client wanted to use default info on phones |
| 21:17.32 | TrentCreek | NOTICE[5758] chan_iax2.c: Rejected connect attempt from 64.34.181.47, who was trying to reach |
| 21:18.15 | drfreeze | [TK]D-Fender: problem I am hiaving is that phones cannot write the 0000...cfg file to the directory, but the <mac>-app.log and <mac>-boot.log files are being written |
| 21:19.18 | [TK]D-Fender | drfreeze: they should never be touching 00000000 for anything |
| 21:19.32 | [TK]D-Fender | drfreeze: Check your permissions as well |
| 21:19.36 | drfreeze | [TK]D-Fender: hmm, ok |
| 21:20.10 | drfreeze | how do I get the cfg, sip.ld and the boot.cfg files for the latest phones |
| 21:20.32 | [TK]D-Fender | ? |
| 21:20.43 | [TK]D-Fender | DrYou get them from your reseller |
| 21:21.20 | drfreeze | [TK]D-Fender: haven't asked for them |
| 21:21.24 | drfreeze | just got new 550's |
| 21:21.56 | j0 | drfreeze: using trixbox? |
| 21:23.23 | drfreeze | j0: nope |
| 21:23.53 | j0 | drfreeze: ah.. i just installed their polycom package which came with firmware and all the config files |
| 21:24.09 | drfreeze | installed 1.4.21.1 |
| 21:29.45 | TrentCreek | Zzzzzzzzzzzzzzzz |
| 21:45.59 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 21:47.45 | *** part/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 21:51.46 | j0 | do the polycom 501 or 601's support answer/hangup from a wireless headset? |
| 21:56.37 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 21:57.09 | lesouvage | drfreeze: If you mean the default username and password for the phone it is polycom and 456 (at least it was a couple of weeks ago.) |
| 21:58.02 | j0 | lesouvage: i think he was thinking of the ftp default |
| 21:59.42 | mvanbaak | hey lesouvage |
| 21:59.58 | lesouvage | mvanbaak: alles goed? |
| 22:00.04 | mvanbaak | yup |
| 22:01.04 | mvanbaak | still recovering from the shock after I found out what needed to be fixed on my car before it passed APK |
| 22:01.25 | mvanbaak | 612 euro |
| 22:01.58 | lesouvage | mvanbaak: I think it is time to start organizing some nice asterisk event again or at least have a gathering of the Asterisk incrowd. |
| 22:02.27 | mvanbaak | lesouvage: that would be a good idea indeed |
| 22:02.42 | mvanbaak | lesouvage: maybe we can get some sponsors so we can all go to astricon |
| 22:03.33 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 22:03.50 | lesouvage | mvanbaak: that was allmost the price of my citroen bx chicque and I drove 15.000 km (for the non Europeans around 10.000 miles) without problems. |
| 22:04.18 | mvanbaak | lesouvage: yeah, my micra is not worth that much money |
| 22:04.47 | Strom_C | lesouvage: I know how to convert kilometers to miles, thank you very much |
| 22:05.06 | mvanbaak | 1 mile is 1.6 km |
| 22:05.22 | mvanbaak | 1 foot is 33 cm |
| 22:05.22 | lesouvage | mvanbaak: that really would be a great idea. If I go I'm afraid I have to be my own sponsor. But business isn't going that bad so who knows. |
| 22:05.27 | Strom_C | makes it closer to 9000 miles than 10,000 |
| 22:05.59 | mvanbaak | lesouvage: I cant afford to go to astricon. the plane ticket alone is around 700 euro |
| 22:06.13 | lesouvage | Strom_C: dollar euro conversion is much harder because it is changing every day. |
| 22:06.23 | mvanbaak | 5 days of hotel (100 euro/day) |
| 22:06.25 | mvanbaak | blegh |
| 22:07.32 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:08.13 | lesouvage | mvanbaak: Yes we definitly needs a sponsor. Lets write a project plan with some nifty innovative aspects in it and raise some of the European subsidy and make the trip to astricon part of the project. ;-) |
| 22:09.06 | mvanbaak | whehehehehe |
| 22:09.17 | lesouvage | mvanbaak: I raised milions of European money by writing projectplans but the money was for the municipal. |
| 22:09.50 | mvanbaak | hhmm, your name was not mentioned in there as one of the major receivers of the money ? |
| 22:10.27 | mvanbaak | I wish I could go to the astricon. Ah well, maybe next year |
| 22:10.48 | lesouvage | mvanbaak: compare to what is available we just need a tiny drip to make the astricon tour |
| 22:11.01 | mvanbaak | yup |
| 22:11.06 | lesouvage | Will there be no European Astricon? |
| 22:11.15 | mvanbaak | nope |
| 22:11.31 | mvanbaak | well, nothing planned yet |
| 22:12.14 | mvanbaak | maybe you can setup one ? |
| 22:12.16 | mvanbaak | hides |
| 22:12.25 | lesouvage | Why not, maye it is something for media plaza to take a role. It would be nice to have it at the Asterisk hotel i Amsterdam. |
| 22:12.27 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 22:12.54 | lesouvage | I would love to take a role, but I'm afraid I can not do it on my own. |
| 22:13.02 | mvanbaak | I can help |
| 22:13.04 | mvanbaak | and nancy |
| 22:13.55 | mvanbaak | maybe it can be part of her OS program |
| 22:20.08 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 22:25.27 | *** join/#asterisk _mm_ (n=mmclain@cpe-67-49-233-178.dc.res.rr.com) |
| 22:28.47 | *** join/#asterisk raytruz` (n=raytruz_@96.28.43.212) |
| 22:35.51 | *** join/#asterisk raytruz` (n=raytruz_@96.28.43.212) |
| 22:36.51 | _mm_ | is away. Automatically set away [SZon] |
| 22:41.11 | *** join/#asterisk mvanbaak_ (n=michiel@vanbaak.xs4all.nl) |
| 23:02.28 | TrentCreek | Now what???? NOTICE[5758] chan_iax2.c: Rejected connect attempt from 64.34.181.47, who was trying to reach |
| 23:04.52 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:14.13 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:14.13 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:15.27 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:19.21 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 23:19.27 | *** join/#asterisk VIPCarrier (n=vipcarri@ool-44c65236.dyn.optonline.net) |
| 23:19.44 | ZX81 | ~ZX81 |
| 23:19.44 | jbot | rumour has it, zx81 is the creater of the Daily Asterisk News (see ~adn) |
| 23:20.36 | ZX81 | jbot, no zx81 is the creator of the Daily Asterisk News (see ~adn) and the author of the AsteriskWatch FaceBook Application (see ~asteriskwatch) |
| 23:20.37 | jbot | ZX81: okay |
| 23:20.44 | ZX81 | ~ZX81 |
| 23:20.44 | jbot | you are, like, the creator of the Daily Asterisk News (see ~adn) and the author of the AsteriskWatch FaceBook Application (see ~asteriskwatch) |
| 23:20.52 | ZX81 | ~asteriskwatch |
| 23:21.08 | VIPCarrier | hi |
| 23:21.13 | ZX81 | hi |
| 23:21.24 | VIPCarrier | does any one know a some good billing software that works with asterisk/mysql |
| 23:21.32 | ZX81 | a2billing |
| 23:21.39 | VIPCarrier | no way too mutch for me |
| 23:21.42 | VIPCarrier | i dont need all of it |
| 23:21.52 | ZX81 | so just write agi scripts |
| 23:21.56 | VIPCarrier | i need to be able to parce cdr's |
| 23:22.04 | VIPCarrier | i'm not a programmer |
| 23:22.14 | ZX81 | jbot, no asteriskwatch is the FaceBook application for Asterisk Users ( see http://apps.facebook.com/asterisk/ ) |
| 23:22.15 | jbot | ZX81: okay |
| 23:22.17 | VIPCarrier | match with a prices and create an invocie |
| 23:22.31 | ZX81 | check the wiki maybe? |
| 23:22.33 | ZX81 | ~wiki |
| 23:22.40 | ZX81 | hmm |
| 23:22.42 | ZX81 | I meant |
| 23:22.45 | ZX81 | ~voip-info |
| 23:22.46 | jbot | rumour has it, voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 23:24.02 | VIPCarrier | any one wanna give me a tip? it don't have to be a free |
| 23:24.06 | VIPCarrier | it can coast money |
| 23:24.55 | ZX81 | what's wrong with a2billing? |
| 23:25.06 | ZX81 | can you not just forget about some features? |
| 23:29.44 | mvanbaak | that's what I would do |
| 23:29.53 | mvanbaak | go with a2billing and only use what I need |
| 23:32.44 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 23:35.27 | *** join/#asterisk jakedahs (n=jakedahs@dynamic-acs-24-144-136-137.zoominternet.net) |
| 23:38.27 | jakedahs | Should I just post a question to the room? new to asterisks having difficulty getting inbound calling to work properly |
| 23:39.38 | mmlj4 | nah, call us instead |
| 23:41.41 | *** join/#asterisk diegoviola (n=diego@adsl-132-64.click.com.py) |
| 23:42.02 | jakedahs | peer firefly phone on local lan always shows status as UNREACHABLE. when trying to route incoming call receive "dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)" @ console |
| 23:42.22 | TrentCreek | Now what???? NOTICE[5758] chan_iax2.c: Rejected connect attempt from 64.34.181.47, who was trying to reach |
| 23:42.43 | ZX81 | firefly?! |
| 23:42.48 | ZX81 | haven't seen that in years |
| 23:42.53 | ZX81 | try zoiper |
| 23:43.10 | jakedahs | freebee too? |
| 23:43.11 | TrentCreek | there was a firefly movie last year ;-) |
| 23:43.17 | *** join/#asterisk VIPCarrier (n=vipcarri@ool-44c65236.dyn.optonline.net) |
| 23:43.48 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 23:43.52 | VIPCarrier | can I use a2billing with multiple servers? |
| 23:43.57 | ZX81 | TrentCreek: check that the context= line in the iax2 peer is pointed at the same context as the end of that message use posted |
| 23:44.08 | ZX81 | VIPCarrier: dunno |
| 23:44.10 | VIPCarrier | can I run a2billing not on a same server where is asterisk |
| 23:44.30 | ZX81 | I'd say so - you'd probably need the AGI scripts on the right machine though |
| 23:44.46 | ZX81 | If you post to the asterisk-biz mailing list someone will probably do it all for you |
| 23:45.00 | VIPCarrier | they can create for me a billing? |
| 23:46.47 | jakedahs | ZX81: thank you, thank you, thank you. |
| 23:46.59 | ZX81 | :) |
| 23:47.14 | jakedahs | chalk another back up for sanity this weekend! :) |
| 23:47.26 | ZX81 | :) - midday monday here :) |
| 23:47.37 | VIPCarrier | sent |
| 23:47.42 | pcrane | hi ZX81 ;) |
| 23:47.48 | ZX81 | heh hi man |
| 23:47.56 | pcrane | how goes it? |
| 23:48.13 | TrentCreek | ZX81: Okay...thanks..looking now |
| 23:48.23 | jakedahs | thought this was just going to be a weekend project last week, 1 week, 4 installs, and a few headaches later...all i had to do was ask to find out it was ....firefly... :) |
| 23:48.49 | ZX81 | I've got 126590, 128639, 128737 not crashing, and 128795, 129803, 130169 crashing |
| 23:48.50 | ZX81 | :) |
| 23:49.00 | ZX81 | although I'm not 100% sure about 128737 |
| 23:49.07 | ZX81 | so I'm testing it again |
| 23:49.48 | ZX81 | this would be great to set up on someone's machine you don't like: |
| 23:49.50 | ZX81 | tc qdisc add dev eth0 parent 1:1 handle 10: netem loss 60% |
| 23:49.52 | ZX81 | :D |
| 23:50.14 | ZX81 | go 60% packet loss :) |
| 23:51.24 | ZX81 | I've had rev 128737 running for 7 minutes without crash so far - should crash around 5-10 mins if its going to |
| 23:51.42 | ZX81 | although I did have one crash at 14 mins |
| 23:52.48 | ZX81 | pcrane, is melbourne/sydney still crashing? if so, what revision are they running? |
| 23:52.53 | *** join/#asterisk Fuzix (n=fuzix@250-118.citynet.ftth.internl.net) |
| 23:53.12 | pcrane | gimme a sec |
| 23:53.13 | pcrane | ;) |
| 23:53.26 | ZX81 | sweet |
| 23:53.47 | Fuzix | Does anyone know what the following could mean? : |
| 23:53.47 | Fuzix | l 13 23:48:14] ERROR[11834]: /usr/src/asterisk-1.4.20/include/asterisk/lock.h:479 __ast_pthread_mutex_unlock: chan_local.c line 189 (local_queue_frame): mutex '&us->lock' freed more times than we've locked! |
| 23:53.47 | Fuzix | [Jul 13 23:48:14] ERROR[11834]: /usr/src/asterisk-1.4.20/include/asterisk/lock.h:496 __ast_pthread_mutex_unlock: chan_local.c line 189 (local_queue_frame): Error releasing mutex: Operation not permitted |
| 23:54.00 | ZX81 | heh |
| 23:54.00 | Fuzix | I have it in 1.4.20, 1.4.21, 1.4.21.1 |
| 23:54.03 | ManxPower | Fuzix: try 1.4.21.1 |
| 23:54.04 | ZX81 | double free |
| 23:54.23 | ManxPower | Fuzix: then report it as an error |
| 23:54.31 | ManxPower | bugs.digium.com |
| 23:54.48 | ZX81 | heh ok |
| 23:54.51 | ZX81 | ty |
| 23:54.57 | ZX81 | will check the rev on that |
| 23:55.50 | ZX81 | yeah - is too early for that bug |
| 23:56.02 | ZX81 | 1.4.21.1 was ok |
| 23:56.37 | ZX81 | at least I think so - unless it was a regression |
| 23:57.02 | ZX81 | what does show uptime say? |
| 23:57.03 | pcrane | ZX81: both are Asterisk 1.4.21-rc1 |
| 23:57.10 | ZX81 | yeah - same master |
| 23:57.16 | pcrane | ye |
| 23:57.17 | pcrane | p |
| 23:57.54 | pcrane | Sydney: System uptime: 2 days, 20 hours, 6 minutes, 24 seconds |
| 23:58.19 | ZX81 | ok cool |
| 23:58.22 | *** join/#asterisk [netman] (n=netman@68.Red-88-8-164.dynamicIP.rima-tde.net) |
| 23:58.44 | *** join/#asterisk moy (n=moy@189.169.86.155) |
| 23:59.14 | pcrane | Melbourne: System uptime: 2 days, 20 hours, 3 minutes, 38 seconds |
| 23:59.23 | ZX81 | sweet |
| 23:59.29 | pcrane | but that's the weekend |
| 23:59.32 | ZX81 | yeah |
| 23:59.35 | ZX81 | good though |
| 23:59.36 | pcrane | so, no-body will have made calls ;) |
| 23:59.39 | ZX81 | means its call based |
| 23:59.41 | ZX81 | snap |
| 23:59.42 | ZX81 | :) |