00:00.14 | xeet2 | rocketraid has alot of linux issues, they use the system cpu for some things, kind of like a winmodem did |
00:00.16 | Juxt | no, i don't touch anything made by highpoint |
00:00.27 | xeet2 | good. |
00:00.32 | X-Rob | I've never had a problem with highpoint stuff |
00:00.40 | xeet2 | x-rob: give it time |
00:00.41 | Juxt | i had better luck with software raid under linux |
00:00.42 | X-Rob | I've got a few sites running their 8 port SATA cards without problems. |
00:00.46 | xeet2 | its like buying a kia |
00:01.11 | X-Rob | Admittedly, I'm just using the card to provide ports, it's not doing any RAID stuff itself. |
00:01.17 | xeet2 | ah |
00:01.25 | xeet2 | yeah then it should work without any major issues |
00:01.29 | xeet2 | just don't turn on raid |
00:01.37 | X-Rob | I hate doing that - if the card dies you're fucked unless you can find exactly the same card. |
00:01.38 | xeet2 | or if you do, use software raid =) |
00:01.42 | Juxt | actually my card is made by maxtor |
00:01.50 | Juxt | but i don't think the chipset is maxtor |
00:02.00 | Slup | drumkilla mmm ok i'm talking to someone who says that if an external address is recieved by * it just drops it on the floor and to talk to users on an external provider you need to set it up in the dial plan |
00:02.01 | X-Rob | use mdadm and it doesn't matter what you plug the drives into. |
00:02.09 | xeet2 | right |
00:02.28 | Juxt | i have a box that runs 2 raid1 arrays using mdadm |
00:02.34 | Juxt | so far it's been freaking fantastic |
00:02.39 | xeet2 | slup: sip? you need to set up localnet, etc... |
00:02.44 | jackx | Slup: http://slacker.com/~nugget/asterisk7.php |
00:02.46 | Juxt | i am seriously considering not buying raid cards anymore |
00:02.52 | Slup | drumkilla but i was saying to that if you did that the callee would be presented with the id of the account you used in the dial plan |
00:03.07 | xeet2 | they work really well =) |
00:03.22 | Juxt | yep, they are also quite pricey |
00:03.31 | xeet2 | well, yeah, you get what you pay for |
00:03.36 | Juxt | i concur |
00:03.57 | *** join/#asterisk telephoneman (~chatzilla@67-41-211-223.brbn.qwest.net) |
00:04.15 | xeet2 | combining them with western digital raptor drives, its quite speedy |
00:04.33 | Juxt | so i have this problem where my box has to transcode some calls and in peak hours it gets pegged out |
00:04.50 | xeet2 | juxt: between what codecs? |
00:04.59 | Juxt | ulaw and ilbc |
00:05.10 | Juxt | it's a call center |
00:05.14 | Juxt | so lots of calls |
00:05.28 | jackx | ... |
00:05.40 | jackx | why you need conversion for? |
00:05.54 | Juxt | i use ilbc for trunking |
00:05.55 | jackx | i assume all the agents in the call center are calling to "normal" extension |
00:06.20 | Juxt | well all internal calls are ulaw |
00:06.24 | jackx | you connected it to an other pbx? |
00:06.34 | Juxt | yes |
00:06.40 | xeet2 | juxt: how many concurrent calls are we talking about? |
00:06.47 | jackx | trought voip? |
00:06.48 | Juxt | 150-200 |
00:06.59 | jackx | big center :) |
00:07.00 | xeet2 | and how much bandwidth is available? |
00:07.05 | jackx | we have only 60 seats |
00:07.11 | Juxt | a T1 and 2 DSL |
00:07.27 | Juxt | i put about 110 calls thru a data T1 using ilbc |
00:07.39 | xeet2 | how are you load balancing between the t1 and the dsl? |
00:07.44 | jackx | T1 is like 1.5Mbit up/down fd right? |
00:08.39 | xeet2 | juxt: and what kind of box is it? |
00:09.13 | Juxt | xeet2: i use a linux based load balancer |
00:09.23 | Juxt | the box is a dual P3 |
00:09.31 | xeet2 | juxt: how many boxes? |
00:09.38 | Juxt | 1 |
00:09.46 | jackx | respect |
00:09.52 | *** join/#asterisk afaict (~afaict@201.6.255.86) |
00:10.02 | jackx | are all the phones IP phones? |
00:10.08 | xeet2 | yeah, I think you'd need a bigger box than that to handle 200 calls |
00:10.12 | xeet2 | 200 calls of ilbc |
00:10.13 | afaict | hi folks! |
00:10.26 | xeet2 | I would also be quite concerned about your bandwidth |
00:10.37 | Juxt | bandwith is cheap |
00:10.40 | xeet2 | you need another t1 regardless |
00:10.45 | Juxt | yeah |
00:10.55 | Juxt | actually we're moving to a new location that has metro-ethernet |
00:10.55 | afaict | I would like to know. I have a x.25 line connected in my FXO digium card. Can I call more then once if I have multiple X.25 channels right ? |
00:10.56 | shido | . |
00:11.03 | jackx | Juxt: what company terminates your calls? |
00:11.09 | xeet2 | yes you could use just under the maximum but when you start having jitter issues your circuit will be useless |
00:11.18 | shido | its HOT |
00:11.23 | shido | I need a new A/C |
00:11.32 | Juxt | jackx: i currently use 4 carriers |
00:11.39 | xeet2 | shido: in mi? |
00:11.48 | shido | yes in michigan |
00:11.50 | Juxt | have least cost dial too |
00:11.51 | Juxt | :-) |
00:11.57 | xeet2 | I thought it was always cold out there =) |
00:11.58 | jackx | Juxt: can you name some? |
00:12.02 | shido | we get everything here |
00:12.07 | shido | hail in summer |
00:12.11 | shido | and summer in winter |
00:12.15 | shido | and contruction |
00:12.19 | jackx | other question: What do you pay for the connection and the termination/ |
00:12.20 | xeet2 | earthquakes? landslides? |
00:12.23 | shido | no |
00:12.28 | xeet2 | how about tsunamis? |
00:12.28 | Juxt | RNK, experient |
00:12.29 | xeet2 | =P |
00:12.30 | shido | 1 little tremor every 10 yrs |
00:12.31 | jackx | excl. the callkosts |
00:12.43 | harryvv | shido: want some of our cool BC air? |
00:12.44 | harryvv | :) |
00:12.48 | shido | landslides? not unless they're man made |
00:12.51 | shido | BC? |
00:12.52 | shido | err |
00:12.55 | harryvv | vancouver |
00:13.01 | shido | I know something else you can send me from BC |
00:13.13 | shido | xpress post |
00:13.18 | harryvv | whats that? |
00:13.30 | afaict | hi folks. |
00:13.31 | shido | xmas colors grown natural around there |
00:13.47 | harryvv | no idea what that is |
00:13.47 | harryvv | :) |
00:13.52 | shido | I bet you dont |
00:13.55 | xeet2 | hehe |
00:14.03 | Juxt | jackx: level3 in 2 months :-) |
00:14.04 | shido | now pass the zigzags |
00:14.24 | afaict | if I have a X.25 line with 4 channels connected in a FXO Digium card. Will I able to do 4 calls at the same time with asterisk |
00:14.32 | harryvv | Juxt what relation do you have with L3? |
00:14.37 | xeet2 | juxt: you should look at getting a bigger box or two, more bandwidth, and using g726 |
00:14.40 | shido | me? |
00:14.47 | Micc | Ok, I'm running stable and I've reinstalled on linux, ztdummy seems to be fine, but I've still got a bit of a delay in meetme. hmmm I should try the echo test. |
00:14.54 | Juxt | harryvv: can't disclose that yet |
00:15.09 | Juxt | g726? |
00:15.14 | harryvv | are you a customer or reseller of there service? |
00:15.25 | Juxt | going thru a reseller |
00:15.28 | shido | I can jump rope with level3 connections |
00:15.30 | harryvv | i see |
00:15.43 | Juxt | i can't meet their monthy commit yet |
00:15.45 | Juxt | but soon |
00:16.00 | Juxt | why would i use g729 over say g729 or ilbc? |
00:16.08 | shido | Juxt, drop me a line at greg@nufone.net |
00:16.08 | xeet2 | g726, in voice conversations, sounds exactly like ulaw... and its not bad on the cpu |
00:16.31 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
00:16.33 | Juxt | oh hey greg :-) i was looking for you at VON |
00:16.50 | Juxt | too bad you weren't there, i was gonna throw a huge chunk of loot your way |
00:16.57 | xeet2 | lol |
00:17.03 | jackx | Juxt: can you mean a company? |
00:17.13 | jackx | mean = name |
00:17.13 | afaict | I have the IAXy behind IPSEC network. I connected two offices with IPSEC. One of the offices has asterisk and another office has IAXy. It works fine right ? |
00:17.25 | Juxt | jackx: name of my company? |
00:17.32 | Juxt | or reseller? |
00:17.34 | jackx | Juxt: no. name of a reseller |
00:17.34 | xeet2 | afaict: yes, barring any network issues |
00:17.40 | Juxt | no i can't just yet |
00:17.43 | Juxt | sorry |
00:17.45 | shido | was... cuz... |
00:17.55 | jackx | I'm in holland, no competition here ;) |
00:17.57 | afaict | xeet2, thanks. |
00:18.06 | Juxt | i have an NDA |
00:18.23 | jackx | none disclosure Agreement? |
00:18.23 | afaict | and for that question do you have the anwser? if I have a X.25 line with 4 channels connected in a FXO Digium card. Will I able to do 4 calls at the same time with asterisk |
00:18.24 | harryvv | jack..you lucky dog |
00:18.25 | harryvv | :) |
00:18.38 | jackx | i am not into business or something |
00:19.01 | afaict | xeet2 do you know something about ? X.25 lines and asterisk |
00:19.11 | xeet2 | afaict: I didn't know zaptel could do anything x.25 |
00:19.58 | afaict | xeet2, really ? |
00:20.28 | xeet2 | really, I didn't know, don't know anything about that |
00:20.37 | xeet2 | haven't touched x.25 in quite a while =) |
00:21.13 | Juxt | so what i was thinking is to put another box and a load balancing proxy in front of the 2 |
00:21.22 | Juxt | this way i could scale my rig |
00:21.30 | X-Rob | X25 is 1970's technology. We've moved on since then. |
00:21.37 | X-Rob | moved on a -long- way. |
00:22.10 | X-Rob | Altho, sadly enough, in australia you can dial 01924 and get a 2400bps connection to Austpac, Australias X25 network. |
00:22.35 | X-Rob | _still_ |
00:22.40 | xeet2 | haha |
00:23.03 | afaict | X-Rob, im from brasil ... and we use x.25 lines when we have multiple dial connections at the same time |
00:23.13 | afaict | X-Rob, :-/ |
00:23.14 | xeet2 | juxt: look at bigip load balancers |
00:23.29 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
00:23.32 | newmedian | Aren't a number of EFTPOS style terminals still using x.25? |
00:24.05 | xeet2 | 7-11 still has a ton of x.25 stuff |
00:24.12 | X-Rob | afaict, you poor thing. |
00:24.22 | *** part/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
00:24.25 | X-Rob | xeet, and that's why it's still around |
00:24.33 | afaict | X-Rob, :D |
00:24.39 | Juxt | xeet - looking right now |
00:24.44 | Juxt | looks magical :-) |
00:24.55 | xeet2 | juxt: they are |
00:24.58 | Micc | can someone call my asterisk system and test my conference meetme. 1 (888) 206-4443 room #1234 |
00:25.02 | xeet2 | juxt: and white reliable if you can afford them |
00:25.10 | X-Rob | even though you can get IPSEC,TCP/IP and PPPoE tunnel on a single chip now, they still use X25. |
00:25.15 | X-Rob | because it's too hard to change. |
00:25.21 | X-Rob | (which is why 01924 still works) |
00:25.30 | xeet2 | x-rob: doex x25 run on that single chip too? |
00:25.37 | X-Rob | probably |
00:25.38 | Juxt | i just don't see the advantage of that over SER, except i can keep IAX |
00:25.49 | X-Rob | Linux supports X25, so it should. |
00:26.15 | X-Rob | The difference is, IPSEC is actually secure, whilst X25 can be intercepted anywhere inside the telco. |
00:26.34 | xeet2 | juxt: there's alot more you can do with a bigip with sip than what ser can do |
00:27.18 | harryvv | X-Rob that VERY important to govermental agencies. |
00:27.32 | Juxt | xeet2: that's probably true, alas i just want to accomplish only 1 thing :-) |
00:27.40 | harryvv | here in canada the privacy laws are very strict...almost to strict. |
00:27.41 | shido | no |
00:27.46 | shido | hello |
00:27.51 | shido | yes hello |
00:27.52 | xeet2 | hello |
00:28.04 | shido | are you still having echo problems or delay problems with this thing |
00:28.10 | shido | choppy? |
00:28.13 | shido | hrmmm |
00:28.25 | shido | ok let me change some things... |
00:29.10 | harryvv | btw does anyone know of any voip testing software that can simulate say 100,1000 or more calls on my server? |
00:29.19 | shido | show audio codecs |
00:29.33 | *** join/#asterisk file[desk] (~jcolp@mctn1-4092.nb.aliant.net) |
00:29.40 | shido | I sound bad? |
00:29.41 | shido | choppy? |
00:29.49 | shido | let me try a differnet phone |
00:30.43 | Micc | maybe I have firewall issues. |
00:30.47 | shido | hrmm |
00:30.53 | shido | naah i think its me and limewire |
00:31.08 | shido | just cuz you kill the limewire interface doesnt mean java is dead |
00:31.16 | shido | let me kill limewire and call you back |
00:31.37 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
00:31.37 | Micc | It still seems that there is a bit of a delay. But when I try the echo test it isn't too bad. |
00:31.38 | shido | mic chec |
00:32.52 | Umaro | Hey guys.. I currently have a area code 971 DID from broadvoice, but i'm finally frustrated to the point where i'm looking for someone else to provide it. Does anyone have any suggestions? |
00:33.10 | shido | odd |
00:33.15 | *** join/#asterisk IQ (~IQ@70-59-165-178.omah.qwest.net) |
00:33.34 | IQ | hi all |
00:34.33 | xeet2 | umaro: where is 971? |
00:34.46 | afaict | X-Rob, im trying to figure out how it will works. I mean guess that I have in one office 5 phones connected in a single IAXy. What IAXy will do if I have 2 phones calling at the same time ? |
00:35.09 | harryvv | umaro, first time here? |
00:35.17 | Umaro | xeet2: portland, oregon |
00:35.20 | Umaro | harryvv: no, why? |
00:35.37 | X-Rob | I don't know anythign about the IAXy's, cept they have a bad habit of catching on fire. |
00:35.46 | xeet2 | mmm, wish I could but I can't port oregon =) |
00:35.46 | X-Rob | You can't buy 'em in australia, so I've never seen on. |
00:35.47 | harryvv | okay there have been complaints about broadvoice discussed here ;) whats the problem waiting to buy one? |
00:36.01 | afaict | X-Rob, why u can not but it in australia ? |
00:36.11 | harryvv | portland oregon my original home ;) |
00:36.13 | X-Rob | not approved. |
00:36.21 | Umaro | harryvv: do they include people saying their calls sound like there's a helicopter in the background? |
00:36.31 | afaict | X-Rob, hows that ? " they have a bad habit of catching on fire." |
00:36.32 | harryvv | Uramo, thats jitter |
00:36.34 | afaict | X-Rob, they burn ? |
00:36.36 | afaict | :D |
00:36.46 | afaict | sorry.. it burns ? |
00:36.54 | X-Rob | Well |
00:36.55 | harryvv | Thay have a jitter problem |
00:37.00 | X-Rob | not actually catching on fire |
00:37.09 | X-Rob | but they get extremely hot and I've heard of a couple of the cases melting. |
00:37.27 | afaict | X-Rob, nice. :D |
00:37.28 | X-Rob | This is only a rumour, by the way. Totally unsubstanciated. |
00:37.35 | afaict | ok. |
00:37.39 | X-Rob | If you want IAX, get some PA1688 handsets. |
00:37.44 | X-Rob | they're as cheap as anything. |
00:37.46 | harryvv | I have only experainced jitter once on my asterisk install. that was 6 months ago. I dont know enough about it but is it related to bandwith problems? |
00:38.31 | ptiggerdine | harryvv, use mrtg to find out |
00:38.32 | *** join/#asterisk SwK (~ken@12-219-156-206.client.mchsi.com) |
00:38.35 | X-Rob | harry - jitter can be related to _lots_ of things, but yes, lack of bandwidth is one of the things that will cause jitter |
00:38.41 | X-Rob | Gah |
00:38.43 | *** join/#asterisk Strom_C (~strom@66.159.243.60) |
00:38.47 | X-Rob | I have this dual xeon to build |
00:38.57 | xeet2 | bandwidth probelms, out of order packets, corrupted packets |
00:39.04 | Umaro | harryvv: I don't think it's a bandwidth problem.. T1 in las vegas, using their LAX proxy server |
00:39.09 | Umaro | T1 is at like 25% max |
00:39.26 | xeet2 | umaro: what about the other end? |
00:39.49 | Umaro | xeet2: what do you mean? |
00:39.57 | Umaro | xeet2: I have the problem with those two end points |
00:40.15 | *** join/#asterisk Eduardo_Henrique (~ehst@201.19.9.10) |
00:40.36 | xeet2 | this is the problem with voip over the internet |
00:40.42 | xeet2 | so many variables |
00:40.47 | harryvv | X-ROB um running asterisk on my opteron system works fine. |
00:40.50 | Eduardo_Henrique | Exist a IAX ActiveX for .NET plataform ? |
00:41.12 | xeet2 | eduardo: blasphemy |
00:41.20 | Umaro | maybe i'll just forward my broadvoice DID |
00:41.34 | xeet2 | umaro: oh its to broadvoice? |
00:42.03 | xeet2 | broadvoice has had alot of quality issues lately |
00:42.12 | Eduardo_Henrique | xeet2: link ? |
00:42.15 | Eduardo_Henrique | =P |
00:44.26 | Micc | anyone know what might cause a delay of over 1 second with meetme? I"m using ztdummy |
00:44.39 | Micc | and I'm using the stable tree. |
00:46.12 | X-Rob | Harry, yeah, I know asterisk works extremely well. |
00:46.16 | X-Rob | I'm tossing up what to do with it. |
00:46.34 | X-Rob | I've just put Fedora Core 4pre3 on it. |
00:47.18 | *** join/#asterisk drooth (~drooth@ip68-107-113-76.sd.sd.cox.net) |
00:47.33 | drooth | does anyone know how to change the hostname for CentOS |
00:47.35 | X-Rob | do I a: Go with CVS HEAD, or b: stick with 1.0.7.. They're not going to be running any TDM hardware in it, so I probably want to go with CVS's ztdummy. Gah. Decisions. |
00:47.42 | afaict | X-Rob, what internet rate connection to use with asterisk office-internet-office ? |
00:47.58 | X-Rob | drooth - edit the file /etc/sysconfig/network |
00:47.59 | afaict | X-Rob, 128k up/downlink link is fine ? |
00:48.08 | xeet2 | afaict: how many concurrent calls? |
00:48.16 | X-Rob | xeet2 beat me 8) |
00:48.17 | afaict | xeet2, 2 |
00:48.19 | harryvv | X-ROB thats what I am running is FC3 x86_64 but if you want to run it as a work station I would avoid it. There are still alot of binaries and source code that have alot of missing files that problebly work on a i386 system |
00:48.28 | drooth | x-rob: you are the man! htanks |
00:48.29 | drooth | thanks |
00:48.41 | xeet2 | 128k what, dsl? and is that dsl pppoe? |
00:48.52 | afaict | xeet2, yes. dsl and ipsec |
00:48.55 | Juxt | has anyone done any benchmarking with the dual-core opterons? |
00:49.03 | X-Rob | harry, this is a 32bit xeon machine, not the new funky (expensive) 64bit xeons. |
00:49.19 | afaict | X-Rob, 64bit xeons... ;~ |
00:49.23 | xeet2 | afaict: you'd be lucky to get 1 ulaw call out of that then. go with g729, ilbc, gsm, or speex, but yeah you could get 2 calls |
00:49.30 | X-Rob | I'm gunna build up a nice athlon64 as the gateway box shortly. |
00:49.44 | X-Rob | afaict, rare as hens teeth, and twice as spensy. |
00:49.57 | afaict | X-Rob, :D |
00:49.58 | X-Rob | 128k shoudl get two ilbc calls. |
00:50.18 | drooth | x-rob: uname -a still shows the old name, do i need to reset server? |
00:50.21 | Juxt | eeh 128k should do more than that |
00:50.33 | X-Rob | drooth, if you want to change it -now- then type 'hostname new.host.name' |
00:50.36 | Micc | is 99.95% accuracy with the zttest good? |
00:50.50 | X-Rob | micc, 99.98% is what digium say you should have |
00:50.57 | xeet2 | juxt: for almost anything but ulaw yeah you can get more than 2 hopefully |
00:51.03 | afaict | X-Rob, Juxt thanks. |
00:51.13 | Micc | X-Rob, I've got 99.95 average with ztdummy |
00:51.24 | X-Rob | micc, are you using linux 2.4 or 2.6? |
00:51.31 | Juxt | http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html |
00:51.44 | Micc | X-Rob: 2.6 |
00:52.01 | harryvv | X-Rob my opteron works smoothly. for a long time i did not know if the orginal motherboard.cpu ram was bad when it was new..i replaced it with a warentee board and booted right up. |
00:52.19 | *** join/#asterisk xcore (~rmaraschi@200.180.111.2) |
00:52.19 | harryvv | msi neo far 2 had a bad back of boards. |
00:52.25 | harryvv | bach |
00:52.34 | *** part/#asterisk Eduardo_Henrique (~ehst@201.19.9.10) |
00:53.00 | file[desk] | batch |
00:53.12 | X-Rob | <PROTECTED> |
00:53.19 | X-Rob | micc ^^ do that |
00:53.24 | X-Rob | that'll get the latest CVS of asterisk |
00:53.38 | X-Rob | you want the ztdummy from there |
00:56.55 | *** part/#asterisk Juxt (~Juxt@sfl-dsl-64-135-113-4-cust.host.net) |
00:58.00 | Micc | X-Rob, is this going to build and install everything? |
00:58.08 | X-Rob | yeah |
00:58.26 | X-Rob | Then at line 48 |
00:58.27 | Micc | it dies on No description for sounds/1-yes-2-no.gsm |
00:58.37 | X-Rob | Wierd |
00:58.43 | X-Rob | I just did exactly that on my FC4 machine |
00:58.52 | X-Rob | ant it worked flawlessly |
00:58.59 | Micc | Do I need to reboot or restart anything before I try it again? |
00:59.02 | X-Rob | nope |
00:59.09 | Micc | I'm using FC3 |
00:59.18 | X-Rob | this is FC4pre3 I think |
00:59.54 | X-Rob | anyway. Line 48 of /usr/src/zaptel/ztdummy.c change '#if 0' to '#if 1' |
01:01.18 | xeet2 | what does that do? |
01:01.29 | X-Rob | enables the 2.6 RTC support for ztdummy |
01:01.39 | xeet2 | oh cool |
01:01.39 | X-Rob | doesn't use USB, uses POSIX timers, smegloads more accurate |
01:01.54 | xeet2 | is that documented anywhere? |
01:02.00 | X-Rob | Not really. |
01:02.08 | X-Rob | http://aussievoip.com.au/wiki-1.10.008-Install-MDK10.1 |
01:02.15 | X-Rob | That's the mandrake 10.1 docco |
01:02.29 | X-Rob | I'm going to copy most of that to fedora core, coz it's very similar. |
01:07.37 | Micc | My line ztdummy.c line 48 is #else |
01:08.03 | X-Rob | around there |
01:08.05 | X-Rob | is #if 0 |
01:09.45 | Micc | it is #ifdef LINUX26 |
01:10.51 | *** join/#asterisk telephoneman (~chatzilla@67-41-211-223.brbn.qwest.net) |
01:11.11 | X-Rob | I don't know what you've downloaded |
01:11.48 | X-Rob | . /usr/src/zaptel/ztdummy.c |
01:11.55 | *** part/#asterisk Moc (~Moc@modemcable165.109-70-69.mc.videotron.ca) |
01:14.32 | X-Rob | Hrm. |
01:14.41 | X-Rob | It's sucking. |
01:15.07 | X-Rob | --- Results after 4 passes --- |
01:15.07 | X-Rob | Best: 99.975586 -- Worst: 99.938965 -- Average: 99.960327 |
01:15.09 | Micc | X-Rob, I did that thing you said, so I should have that version. |
01:16.36 | X-Rob | line 69 has LINUX26 |
01:17.40 | X-Rob | this is the first time I've tried this |
01:17.59 | Micc | yeah I am using hyperthreading too. |
01:18.21 | h3x | FC3 dosent work too well with zaptel |
01:18.47 | X-Rob | This is strange |
01:19.32 | X-Rob | --- Results after 9 passes --- |
01:19.32 | X-Rob | Best: 99.987793 -- Worst: 99.987793 -- Average: 99.987793 |
01:19.37 | X-Rob | that's _without_ use rtc |
01:19.37 | Micc | h3x, does anything work well with zaptel? |
01:20.25 | Strom_C | zaptel and debian seem to play nicely for me |
01:20.51 | Hmmhesays | debian plays nice with lots |
01:21.56 | Micc | X-Rob, what is rtc? |
01:22.00 | X-Rob | real time clock |
01:22.14 | *** join/#asterisk jpablo (~jpablo@200.94.130.194) |
01:22.26 | X-Rob | 99.987793 is optimal |
01:22.29 | Micc | is rtc supposed to be better? |
01:22.36 | ptiggerdine | yes |
01:22.36 | X-Rob | yeah, rtc is supposed to be perfect |
01:23.25 | Micc | I'm getting 99.95 with latest source. what line do I need to change in ztdummy.c? |
01:23.58 | jpablo | hi, anyone knows how to solve this: I'm not getting ring tone when doing a blind transfer with a spa-2000 in asterisk. |
01:24.06 | Micc | Average: 99.832266 |
01:24.10 | jpablo | the blind transfer is done using 'flash'. |
01:24.28 | X-Rob | Micc, ok. You should change the line near the top that says '#if 0' and underneath that says '#define USE_RTC' |
01:24.35 | X-Rob | change the '#if 0' to '#if 1' |
01:27.14 | Micc | ok, I found that. |
01:27.27 | Micc | I must not have had the right cvs version before. |
01:27.53 | X-Rob | OK |
01:28.03 | X-Rob | I am goign to have to turn off hyperthreading |
01:28.08 | X-Rob | becuase with RTC, it's all over the place |
01:28.10 | X-Rob | and it shouldn't be. |
01:28.24 | X-Rob | --- Results after 8 passes --- |
01:28.24 | X-Rob | Best: 99.963379 -- Worst: 99.938965 -- Average: 99.951172 |
01:28.30 | Micc | I think I need to build a non smp kernel. |
01:28.46 | Micc | Best: 99.963379 -- Worst: 96.923828 -- Average: 99.699910 |
01:28.52 | Micc | look at that worst. 96 |
01:29.16 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
01:29.56 | X-Rob | I'm turning off HT now |
01:30.04 | X-Rob | will try with a non-smp kernel next |
01:30.09 | X-Rob | (This is a dual xeon 3.0 machine) |
01:30.18 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
01:31.10 | Micc | X-Rob, let me know how that works. |
01:31.20 | Micc | I've got a dual xeon 2.4 |
01:31.21 | X-Rob | rebooting... rebooting. |
01:31.30 | X-Rob | problem with server boards is that they take _ages_ to boot |
01:33.09 | telephoneman | anyone interested in a MB that works with spandsp? |
01:35.03 | PatrickDK | server boards don't take ages |
01:35.09 | PatrickDK | ecc memory takes along time to check |
01:35.21 | X-Rob | OK. It's _better_ but it's still not right. |
01:35.23 | PatrickDK | my server takes 2.2min to boot, from start to freebsd running |
01:35.42 | PatrickDK | 4gig ecc memory in it |
01:35.45 | X-Rob | It's alternating between 8192 in 8195 and 8187 |
01:35.48 | kimo_sabe | PatrickDK: most do, even without ECC |
01:36.05 | X-Rob | You try probing 4 SCSI cards quickly |
01:36.07 | X-Rob | it doesn't happen |
01:36.26 | PatrickDK | ya, expectually if you have 2 4port scsi cards :) |
01:36.41 | X-Rob | Hrm. |
01:36.45 | PatrickDK | next upgrade it gets fibre |
01:36.45 | X-Rob | Micc, I dunno. |
01:37.04 | *** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar) |
01:37.09 | Micc | X-Rob, I'm gonna use non smp kernel |
01:37.17 | SwK | try 2 scsi cards, RAID Controller, 2 FC hbAs, and remote management card |
01:37.26 | faqu | i've a problem i've an ata 186 connected to my switch and i can't set the ip |
01:37.39 | faqu | i really tried to do it manually and nothing |
01:37.42 | SwK | faqu: hard set the duplex |
01:37.44 | SwK | and the speed |
01:38.01 | faqu | sorry? |
01:38.24 | SwK | on the ethernet port on the switch and the ata186 turn off aut speed/duplex negotiation and hard set it |
01:38.53 | faqu | yep the lnk is active but the 100 isn't |
01:38.56 | faqu | how may fix it? |
01:39.07 | SwK | dunno |
01:39.28 | SwK | are you sure the ata is still functioning? |
01:39.35 | faqu | why not? |
01:39.44 | faqu | everything looks okey with it |
01:39.50 | SwK | cant get dhcp or set an ip address manually... |
01:40.03 | SwK | can you get the IVR on the phone attached to it? |
01:40.08 | Slup | jackx thanks for that link |
01:40.14 | faqu | how can i check it? |
01:40.31 | SwK | pre the button on top of it (assuming you are talking abou a cisco ata186) |
01:40.39 | faqu | yes |
01:40.42 | faqu | the function button |
01:40.44 | faqu | red one |
01:40.50 | SwK | yeah |
01:40.55 | faqu | then? |
01:41.02 | SwK | listen to the phone |
01:41.07 | faqu | yep its working ok |
01:41.10 | SwK | got to cisco.com and get the manual for it |
01:41.10 | shido | :) |
01:41.14 | SwK | theres a tone of info there |
01:41.19 | *** part/#asterisk faqu (pn@OL187-166.fibertel.com.ar) |
01:41.25 | X-Rob | ah crap. |
01:41.27 | SwK | hah |
01:41.35 | X-Rob | [root@dhcp5 zaptel]# make |
01:41.35 | X-Rob | You do not appear to have the sources for the 2.6.11-1.1286_FC4 kernel installed. |
01:41.35 | X-Rob | make: *** [linux26] Error 1 |
01:41.37 | X-Rob | Sigh. |
01:41.48 | SwK | hah |
01:41.53 | SwK | i hate it when that happens |
01:42.16 | SwK | and for the record... FC sucks |
01:42.26 | X-Rob | yeah |
01:42.35 | X-Rob | I'm coming to that conclusion myself. |
01:42.40 | SwK | get ricer linux |
01:43.30 | X-Rob | 'ricer'? |
01:43.42 | shido | so install the kernel source, X-Rob |
01:43.43 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
01:43.57 | X-Rob | I dunno what you yanks call a ricer, but there's no way I'd want a customer of mine knowing that I installed 'ricer' linux. |
01:44.03 | X-Rob | shido, yeah |
01:44.04 | X-Rob | doing that |
01:44.08 | YoWoudi | lolol |
01:44.22 | Sedorox | lol |
01:44.27 | Micc | X-Rob, make a symlink linux-2.6 -> /lib/modules/2.6.9-1.667smp/build/ |
01:45.43 | SwK | X-Rob its gentoo |
01:46.21 | SwK | ricer linux being a reference to www.funroll-loops.com |
01:46.22 | ptiggerdine | i thought ricer was LFS |
01:46.37 | SwK | errr .org |
01:47.00 | Sedorox | lol |
01:47.04 | Sedorox | no.. ricer is Gentoo |
01:47.42 | *** part/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
01:47.46 | X-Rob | gentoo's a pain in the arse. |
01:47.56 | X-Rob | compiling everything everywhere |
01:48.10 | SwK | thats only cause you think you have to do that |
01:48.25 | X-Rob | *shrug* |
01:48.30 | X-Rob | I've become lazy in my old age. |
01:48.36 | SwK | its like ports on bsd... |
01:48.55 | SwK | build it if you want (or have a fast machine on slow link) else grab the binary packages |
01:49.57 | ptiggerdine | what and rpm's better |
01:50.30 | SwK | rpms are a pain in the ass |
01:50.47 | SwK | atleast emerge will install dependancies for you |
01:51.43 | newl | and apt or urpmi won't? ;) |
01:52.43 | X-Rob | Well |
01:52.45 | X-Rob | fuck this |
01:52.57 | X-Rob | Fedora can blow me. |
01:53.00 | bjohnson | would adding externip in the general section of sip.conf be responsible for screwing up my lan voip calls? ie no audio |
01:53.47 | Micc | X-Rob, my FC3 came with two kernels, I just switched to the on smp version. |
01:53.53 | X-Rob | Yep |
01:53.55 | X-Rob | So did I |
01:54.03 | X-Rob | but the kernel source they've supplied is all over the place. |
01:54.08 | shido | its simple... dont use fc for production |
01:54.35 | X-Rob | Mandrake 10.1 I find incredibly good |
01:54.47 | X-Rob | but the _name_ 'Mandriva' just sounds .... slimy. |
01:55.12 | X-Rob | So. FC is crap. I've got a ubuntu CD here |
01:55.14 | X-Rob | maybe I'll throw that on |
01:55.21 | X-Rob | Micc, you got some time while I do a reinstall? |
01:56.59 | *** join/#asterisk Astermigraine (~psolomon@69-165-217-96.atlsfl.adelphia.net) |
01:57.05 | *** join/#asterisk sudhir492 (~sudhir@pool-71-114-74-131.washdc.dsl-w.verizon.net) |
01:57.10 | sudhir492 | Hi all |
01:57.49 | *** join/#asterisk mne (~mne@chello080108001212.35.11.tuwien.teleweb.at) |
01:58.10 | sudhir492 | For most of the callers, we get only the phone number in caller id. Is it possible to get names too ? |
01:58.53 | Astermigraine | If I set my IP phone to use inband DTMF and notify as the type, it causes asterisk to abend. While I can simlpy change it, wouldnt a bug like that possb lead to a possible DOS? |
01:59.17 | Micc | X-Rob, I'll be here most of the night. |
01:59.28 | Astermigraine | sudhir492 depens what your provider is delivering. |
01:59.43 | mne | hi there ! I'm completely new to asterisk. I looked through some available asterisk documentation, but for now it is far to complex for me. Can you suggest anything ? I want to set up a SIP server so that the registered SIP users can talk to each other |
01:59.49 | Micc | X-Rob, it looks I'm getting better results with non smp kernel I think, but they are still not the best. |
02:00.55 | sudhir492 | Astermigraine: does it depend on the telco from which I have PRI, or on the caller's provider? |
02:01.38 | Astermigraine | Could be both |
02:01.58 | Astermigraine | Depends on what they will pass |
02:02.51 | Astermigraine | The Telco can charge for Caller ID and also passing the ANI's associated account name |
02:02.58 | Astermigraine | With reguard to a SIP client |
02:03.01 | X-Rob | mne - best thing for you to do is download asterisk@home |
02:03.08 | Astermigraine | Thats going to be different |
02:03.41 | Astermigraine | Depending upon who they are going through, they may block client configured CID |
02:03.49 | mne | X-Rob, Astermigraine already suggested it to me, I'm already downloading it ;) thx |
02:03.52 | Astermigraine | and pass account based CID |
02:05.42 | sudhir492 | thx. I will check with my telco on Monday |
02:07.12 | *** join/#asterisk anti (~russ@anti.developer.gentoo) |
02:10.28 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
02:11.13 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
02:12.09 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
02:16.40 | *** join/#asterisk midnight- (midnight@after.12am.net) |
02:17.23 | *** join/#asterisk zeus232 (~zoose@203.49.209.210) |
02:17.45 | X-Rob | Grr |
02:17.52 | X-Rob | OK, now no distribution will detect the CDROM |
02:17.56 | X-Rob | *sigh* |
02:18.32 | shido | ? |
02:18.42 | X-Rob | Stupid new server. |
02:19.16 | *** part/#asterisk mne (~mne@chello080108001212.35.11.tuwien.teleweb.at) |
02:19.41 | shido | what server? |
02:20.34 | X-Rob | It's one of the new intel boards with onboard SATA |
02:21.09 | Micc | X-Rob, supermicro? |
02:21.15 | X-Rob | no, intel intel 8) |
02:21.32 | shido | no drivers? |
02:21.40 | shido | modprobe is your friend |
02:21.44 | shido | insmod is too |
02:21.49 | Sedorox | modprobe > insmod |
02:21.56 | X-Rob | Yeah, it's more a lack of care |
02:22.00 | Sedorox | lol |
02:22.08 | Sedorox | ya gotta care man.... ya gotta care... |
02:22.18 | X-Rob | If I cared, I'd figure out why it can't detect a stadard smegging IDE DVD burner |
02:22.43 | shido | molex plugged in? :) |
02:23.01 | Sedorox | lol |
02:23.10 | X-Rob | It _boots_ off the damn CD |
02:23.18 | shido | errr |
02:23.20 | X-Rob | but the installers are going 'no CD found' |
02:23.20 | shido | k |
02:23.32 | Sedorox | lol |
02:23.33 | shido | do you have to remount the new cd? |
02:23.34 | X-Rob | Fedora loaded an 'ata_piix' modeule |
02:23.35 | X-Rob | module |
02:23.39 | X-Rob | which I'm guessing needs to be done |
02:23.47 | X-Rob | but, again, lack of care. |
02:23.51 | shido | you picked fedora cuz? |
02:24.01 | X-Rob | because it was the handiest DVD |
02:24.16 | X-Rob | I should download SUSE |
02:24.30 | Sedorox | fedora... suck... |
02:24.52 | X-Rob | http://www.intel.com/design/servers/boards/se7520af2/index.htm |
02:24.55 | X-Rob | That's the board |
02:25.02 | *** join/#asterisk rcw (~rcw@206-124-146-40.blarg.net) |
02:25.04 | Micc | X-Rob, let me know what gets your accuracy up. I'm going to have to install another version of linux too. |
02:26.33 | X-Rob | I probably should do this sensibly and just upgrade the kernel |
02:26.55 | X-Rob | I'll download the latest 2.6 and see how I go |
02:29.31 | *** join/#asterisk nemisus (~nemisus@203-217-85-104.dyn.iinet.net.au) |
02:29.37 | *** join/#asterisk wfu (~wfu@203.131.175.66) |
02:31.15 | Umaro | what's the best way to do callerid lookups? I've got an 800 DID, so I don't get callerid name, but i'd really like to. |
02:32.08 | Micc | Ok, this might be a bit better. |
02:32.15 | Micc | Shido, you want to give it a try again? |
02:32.51 | Micc | (888) 206-4443 room# 1234 |
02:33.51 | shido | non smp did the trick |
02:33.52 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
02:33.52 | *** mode/#asterisk [+o anthm] by ChanServ |
02:33.52 | shido | ok |
02:33.57 | kimo_sabe | Umaro: I search our existing LDAP addressbook |
02:34.55 | nemisus | is there a tar.gz of 10.0.8rc? |
02:34.59 | Umaro | kimo_sabe: hmm, smart |
02:35.20 | Umaro | brb |
02:35.48 | kimo_sabe | Umaro: that way we get meaningful caller id info instead of "US Government" for everybody |
02:36.45 | *** join/#asterisk squirrelv5 (~squirrelv@202.57.81.146) |
02:42.15 | *** join/#asterisk CoolAcid (~jk@216.99.98.39) |
02:43.27 | wfu | hi, can i ask anyone to send me samples of sip.conf/extensions.conf/voicemail.conf |
02:49.13 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
02:50.03 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
02:50.03 | *** mode/#asterisk [+o bkw_] by ChanServ |
02:51.58 | zeus232 | i have fresh fedora1 install and i can't get wcfxs driver to load, although modprobe zaptel works and appropriate output is show in /var/log/message ... error msg is: |
02:51.59 | zeus232 | /lib/modules/2.4.22-1.2115.nptl/misc/wcfxs.o: init_module: No such device |
02:52.00 | zeus232 | Hint: insmod errors can be caused by incorrect module parameters, including |
02:52.00 | zeus232 | invalid IO or IRQ parameters. |
02:52.06 | zeus232 | anyone got any ideas? |
02:52.59 | SwK | what fxs card are you using? |
02:53.02 | SwK | a freshmaker? |
02:53.43 | bkw_ | haha |
02:53.52 | bkw_ | go rm -rf all kernel drivers |
02:53.52 | SwK | bkw_ |
02:53.53 | bkw_ | and try again |
02:53.58 | YoWoudi | question |
02:54.06 | SwK | or just rm -rf / and try again |
02:54.08 | zeus232 | umm it's digium tdm400P |
02:54.14 | *** part/#asterisk wfu (~wfu@203.131.175.66) |
02:54.16 | SwK | modprobe wctdm |
02:54.20 | bkw_ | NO |
02:54.22 | YoWoudi | what does the F parameter do swk? |
02:54.24 | bkw_ | you people aren't listening to me |
02:54.30 | SwK | hah |
02:54.38 | SwK | ignore bkw he doesnt know shit ;) |
02:54.38 | bkw_ | cd /lib/modules/2.4.22-1.2115.nptl/misc/ |
02:54.43 | bkw_ | rm -rf the modules |
02:54.48 | SwK | heh |
02:54.49 | bkw_ | zaptel was changed for a week |
02:54.53 | bkw_ | to install in a different location |
02:54.55 | SwK | YoWoudi what -F ? |
02:55.03 | SwK | you men th e f in rf -rf? |
02:55.03 | bkw_ | even if you do modprobe wctdm it might still pickup wcfxs |
02:55.04 | bkw_ | no go |
02:55.10 | SwK | errr rm -rf |
02:55.11 | h3x | god damn tivo-to-go transfer is slow |
02:55.12 | YoWoudi | PERFECT |
02:55.13 | zeus232 | can't locate wctdm... should it be included in the zaptel |
02:55.16 | YoWoudi | rm -rf is perfect! |
02:55.19 | zeus232 | file? |
02:55.22 | YoWoudi | what does the rf stand for though? |
02:55.23 | bkw_ | I give up |
02:55.27 | bkw_ | if you don't know what rm -rf is |
02:55.35 | bkw_ | YOUR TOTALLY LOST.. you'll need to come to cluecon |
02:55.38 | zeus232 | r = recursive f = files ? |
02:55.39 | YoWoudi | i was trying to go do -rm -r withou the F |
02:55.42 | bkw_ | you can go signup at www.cluecon.com ;) |
02:55.43 | YoWoudi | oh no wonder |
02:55.44 | bkw_ | f ix force |
02:55.45 | YoWoudi | lol |
02:55.46 | SwK | YoWoudi try rm -rf / and figure it out |
02:55.47 | *** join/#asterisk Klar (~null@libertarians.STUDENT.CWRU.Edu) |
02:55.48 | bkw_ | f is force |
02:55.54 | YoWoudi | yeah i wasnt sure what the F stood for |
02:56.02 | bkw_ | -f = FUCKED |
02:56.03 | bkw_ | haha |
02:56.06 | YoWoudi | it was trying to get me to delete one by one if i didnt use F |
02:56.06 | SwK | hah |
02:56.08 | YoWoudi | lol |
02:56.10 | YoWoudi | thanks |
02:56.18 | YoWoudi | lol i havent done linux in ages, bear with me |
02:56.20 | SwK | rm -rf bkw_ |
02:57.30 | Micc | I'm cool. I have 3 asterisk servers. |
02:58.04 | YoWoudi | are most of you guys recomending to use the stable release over the standard one? |
02:58.27 | YoWoudi | of asterisk that is |
02:58.44 | Micc | YoWoudi, I like to live on the edge. |
02:58.55 | Micc | It gets me screwed sometimes. |
02:58.57 | YoWoudi | is there much difference? |
02:59.01 | Micc | But it makes me cool. |
02:59.04 | YoWoudi | lol |
02:59.15 | Micc | I get to hang with the cool crowd. |
02:59.48 | MikeJ[Laptop] | SwK, some things even -f won't work on ;) |
03:00.02 | YoWoudi | i gotta change this name |
03:00.10 | YoWoudi | let me see if some other one is taken |
03:00.14 | SwK | unlink works on those tho |
03:00.37 | SwK | unlink /dev/zero give about 20% performance boost on solaris |
03:02.24 | *** join/#asterisk pifiu (~myassisbi@208.205.181.170) |
03:02.27 | pifiu | ok back |
03:02.33 | pifiu | yo woudi=pifiu now |
03:04.16 | Micc | PRI and single span card is on its way. yah! |
03:08.33 | Micc | hmmm |
03:10.04 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
03:11.27 | *** join/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au) |
03:13.57 | *** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 |
03:16.39 | Micc | I thought cluecon already happened? |
03:19.05 | niZon | yeah last year |
03:21.12 | drumkilla | um. |
03:21.17 | drumkilla | you're making stuff up :p |
03:21.39 | anthm | no! |
03:22.06 | drumkilla | it's coming soon, folks |
03:24.39 | *** join/#asterisk edy2 (anonymous@202.134.128.49) |
03:28.43 | pifiu | i need to get an ethernet phone to test my implentation of asterisk into our office |
03:28.53 | pifiu | any phones you guys recommend in general that is good all around? |
03:29.07 | pifiu | i would need to buy around 8 or so eventually |
03:29.29 | pifiu | but i would start with one to test then buy the rest if i am happy with it |
03:29.40 | *** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net) |
03:29.56 | mosty | i'm playing with a linksys pap2 with a regular telephone attached, it's pretty simple to setup |
03:29.58 | telephoneman | would a Java programmer get anything out of cluecon??? |
03:30.42 | telephoneman | someone not developing code for Asterisk? |
03:31.59 | telephoneman | test 1 2 3 test 1 2 3 |
03:32.32 | Syncros | clues |
03:32.34 | Syncros | i guess |
03:33.25 | X-Rob | pifiu - The GXP 2000 is pretty shithot at the moment |
03:33.30 | X-Rob | best bang-for-buck |
03:33.38 | X-Rob | http://www.aussievoip.com.au/wiki-GXP-2000 |
03:36.23 | pifiu | let me see rob one sec |
03:38.36 | *** join/#asterisk squirrelv5 (squirrelv5@202.57.81.149) |
03:44.39 | pifiu | hey rob question |
03:44.47 | pifiu | in the review in the end it says this |
03:45.07 | pifiu | The layout of the phone implies that these can be used for BLF's, but they can't |
03:45.10 | pifiu | what are BLF's? |
03:45.17 | X-Rob | Busy Lamp Fields |
03:45.23 | X-Rob | eg, program them to be othe rpeoples phones |
03:45.25 | pifiu | which does what? |
03:45.29 | X-Rob | and they light up when they're on the phone |
03:45.38 | X-Rob | like a key system |
03:45.46 | X-Rob | the extensions are mapped to a lighted button |
03:46.10 | pifiu | lol so confused |
03:46.17 | pifiu | can you give me an example in use? |
03:46.23 | Slup | just read an artical on asterisk 2 is it true it's out? i can't seem to find it anywhere |
03:46.27 | pifiu | give me a sceneraio |
03:46.29 | X-Rob | Nearly every cheap pbx does it |
03:46.36 | X-Rob | slup, no, it's an april fools joke |
03:46.54 | X-Rob | pif - ok. Have you ever seen a phone system that has buttons marked 'line1', 'line2' 'line3' etc? |
03:47.30 | X-Rob | and the buttons light up when the line is in use? |
03:47.36 | pifiu | yeah |
03:47.53 | Slup | X-Rob thanks did sound a little to much |
03:48.00 | X-Rob | The odds are, those phones would -also- have buttons for extensions |
03:48.08 | X-Rob | that light up when that extension is in use, or is ringing |
03:48.29 | X-Rob | -that's- a busy lamp |
03:48.40 | X-Rob | it lights up when the extension/line/whatever is busy |
03:49.29 | edy2 | i saw ying only |
03:50.15 | bkw_ | telephoneman, sure |
03:50.33 | bkw_ | telephoneman, I know of some people that have setup a java fast agi server |
03:50.51 | bkw_ | we'll be releasing res_js soon too |
03:51.27 | bkw_ | http://bkw.dyndns.org/test.txt |
03:51.28 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
03:53.32 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
03:53.35 | shido | hrmm |
03:54.24 | *** join/#asterisk Kaos76k (~chatzilla@CPE-24-163-140-163.new.res.rr.com) |
03:54.39 | Kaos76k | Hey I received a Zyxel 2000W V2 today and it works great! |
03:54.49 | bkw_ | and? |
03:55.40 | Kaos76k | The wiki says they don't work at all. |
03:55.49 | Kaos76k | All kinds of issues. |
03:55.55 | bkw_ | the wiki says alot of shit that isn't true |
03:56.30 | bkw_ | those phones are pieces of shit |
03:56.35 | bkw_ | feel like NOTHING |
03:56.42 | bkw_ | too light weight |
03:56.42 | Kaos76k | ?? |
03:56.55 | bkw_ | if you want a real wifi phone get a Hitachi |
03:56.56 | Kaos76k | too light weight? |
03:57.12 | bkw_ | the phone doesn't have any meat on its bowns |
03:57.15 | bkw_ | er bones |
03:58.19 | bkw_ | the funny part is |
03:58.19 | bkw_ | CODEC: G711, G.729a(G.729 is recommended for better voice quality) |
03:58.27 | bkw_ | thats an exact quote from zyxels website |
03:58.51 | *** join/#asterisk cluecon (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
03:58.51 | bkw_ | thats funny as hell |
03:58.56 | bkw_ | cluecon, my man |
03:58.57 | bkw_ | whats up |
03:59.01 | bkw_ | where at in Oklahoma are you? |
03:59.10 | cluecon | Tulsa |
03:59.17 | drray | tulsa from his address |
03:59.21 | drray | oops |
03:59.29 | bkw_ | drray, you would say the same about me |
03:59.35 | bkw_ | but i'm not in tulsa |
03:59.44 | Godsey | at last I have exim setup w/ sql and cyrus sasl |
03:59.48 | bkw_ | adsl-69-154-144-52.dsl.tulsok.swbell.net |
03:59.51 | cluecon | ip addresses can be faked...not like i would. |
03:59.51 | pifiu | I <3 asterisk, its changed so much |
03:59.52 | pifiu | i LOVE IT |
03:59.54 | bkw_ | and i'm 110 miles away |
03:59.58 | Godsey | auto-create auto filtering spam into Spam, viruses in Virus :) |
04:00.07 | drray | well, having grown up in austin, I don't trust anything that SWBELL says |
04:00.08 | Godsey | dspam is slick bkw, have you used it? |
04:00.23 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
04:01.30 | Godsey | trust goes as far as someone is willing to pay me ;) |
04:03.13 | *** join/#asterisk waz- (~tjs@208.218.27.24.cfl.res.rr.com) |
04:09.28 | X-Rob | And verily, rob did say, GCC 4.0.0 is an abomination undo Zog, and it should be stricken from the filesystem. |
04:09.46 | anti | X-Rob: whys that. |
04:09.53 | X-Rob | it's giving me the shits, that's why |
04:10.07 | Micc | X-Rob, which side of the bed did you wake up on this morning? |
04:10.13 | X-Rob | The normal one. |
04:10.26 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
04:10.33 | X-Rob | my wife gets upset when I clamber over her to get out of bed. |
04:10.34 | Micc | do your machines normally give you this much trouble? |
04:10.37 | X-Rob | No |
04:10.39 | X-Rob | no they don't. |
04:10.45 | X-Rob | Because usually I don't use fedora. |
04:11.01 | cluecon | X-Rob: what do you normally use? |
04:11.07 | X-Rob | Mandrake 10.1 |
04:11.13 | X-Rob | however |
04:11.23 | X-Rob | I'm not impressed with 'Mandriva' |
04:11.34 | Slup | anyone got any info on when the 1.0.1.9 firmware for the gxp 2000 is comming? |
04:11.39 | cluecon | ah, therein lies the issue. Fedora is a beast which, much like Debian, takes some tweaking to make it happy. |
04:11.42 | X-Rob | Slup, it's on aussievoip.com.au now |
04:11.43 | Godsey | I've been using CentOS 4 and Gentoo |
04:11.48 | X-Rob | I have it in my phone |
04:11.51 | X-Rob | and it's _nice_ |
04:12.04 | Slup | cheers X-Rob |
04:12.37 | X-Rob | If you have a fast net connection, you can download them from this website — set the 'Firmware Upgrade via HTTP' to point to 68.178.145.188/gxp2000/ |
04:13.10 | Slup | X-Rob do you have one of these, just they still look a bit toy like in the photos wondering if they are like that in the flesh? |
04:13.35 | X-Rob | Yeah, I've got one. It -is- reasonably plastic compared to a cisco, but, so is everything else |
04:13.49 | X-Rob | if I was them, I'd be sticking a big chunk of metal in the handset |
04:14.00 | X-Rob | because it just feels too light. You want to have a bit of weight in it |
04:14.27 | *** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar) |
04:14.27 | Slup | yea and for the money all you can ask is that it dose what it says without complaints |
04:14.58 | Slup | i'm sure i cn find a bit somewhere to stick in it ;) |
04:15.07 | X-Rob | Hehehe |
04:15.26 | X-Rob | Actually, there's no screw on the handset to easily pull it apart |
04:15.26 | Slup | the bugitones wernt even good as paperweights |
04:15.42 | Godsey | why take it apart? |
04:15.46 | Godsey | gold plate it |
04:15.47 | Slup | hmmm |
04:15.59 | Slup | lolololol pmsl |
04:16.05 | *** part/#asterisk faqu (pn@OL187-166.fibertel.com.ar) |
04:16.15 | Slup | the gold plating would double the value of the phone |
04:16.47 | Slup | cluecon thats about the general concensus |
04:17.29 | Slup | i'm trying to find didtant relatives i can palm the few i have on to |
04:17.38 | Slup | *distant |
04:17.45 | *** join/#asterisk dynaqueue (~wbevan@adsl-63-205-16-227.dsl.mtry01.pacbell.net) |
04:18.24 | X-Rob | looking at the prices, I'm paying AU$120 for 10+ BT101's, or $160 for 10+ GXP2000's |
04:18.34 | X-Rob | You can't beat that. |
04:18.50 | X-Rob | uh. That's $170 for 10+ GXP's even |
04:19.03 | Slup | nope and if they work they make an excelent entry level phone |
04:19.48 | X-Rob | Obviously, someone at grandstream heard everyone going 'God, these budgettone's look like $9.95 specials. I'm not putting that in an office!' |
04:20.11 | X-Rob | SO they thought' Make it black with shiny bits. Give it grey buttons. And lots of lights' |
04:20.16 | X-Rob | *tadaah* |
04:20.35 | Godsey | with a cp3o ring? |
04:21.01 | X-Rob | Haven't heard a cp3o ring? |
04:21.04 | Micc | does babblefish have a r2d2 translator yet? |
04:21.22 | Godsey | oh ya |
04:21.26 | Godsey | r2d2 not cp3o |
04:21.26 | Slup | i wish they had scratched the shiny bits befor they get scratched |
04:21.37 | Godsey | or that little buck rogers wiggy thing |
04:23.31 | dynaqueue | So anyone tried sox and the GXP to get their own ring tones on it ? |
04:23.46 | X-Rob | apparently it works with 1.0.1.9 |
04:24.15 | dynaqueue | Just got hold of one this week works a treat especially with the beta firmware |
04:24.49 | dynaqueue | although managed to lock it up I think cos my DHCP server renews IP's every 24hrs so need to look into that |
04:26.00 | Godsey | I have my dhcp server hand out 1 hour leases :) |
04:26.05 | X-Rob | yeah, so do I |
04:26.52 | pifiu | what does hanging out ips every hour do? |
04:26.57 | dynaqueue | weird I went to the GXP this morning and went off hook and that was it, light on but no body at home |
04:27.00 | *** join/#asterisk L|NUX (~linux@202.63.195.147) |
04:27.26 | Godsey | keeps a clean lease.db |
04:27.29 | *** join/#asterisk jasonkw (~me@CPE000cf18d47f6-CM00137188afce.cpe.net.cable.rogers.com) |
04:27.58 | pifiu | i still have no idea what that would do, but if it works ill take it as a tip i guess |
04:28.18 | X-Rob | pifiu, if you have lots of different machines that aren't always on |
04:28.23 | jasonkw | does any one here have any idea on how to setup a D-Link DVG-1402S w/ Asterisk? |
04:28.26 | X-Rob | it doesn't make sence to have a long lease |
04:28.44 | X-Rob | Either my machines are always on, so they'll never lose it, or they're in here temporarily |
04:29.07 | X-Rob | ooh |
04:29.08 | X-Rob | oooh |
04:29.11 | X-Rob | I've just found a new bug |
04:29.18 | X-Rob | who here is running 1.0.1.9? |
04:29.26 | X-Rob | (dyna?) |
04:29.35 | dynaqueue | Well for me the 24hrs works fine for the dhcp renewal besides I have plenty of IP's to doll out |
04:29.45 | dynaqueue | me...... |
04:29.49 | X-Rob | You got two phones there? |
04:29.54 | X-Rob | two of anything |
04:29.57 | *** join/#asterisk DIEMACO (~diemaco@Toronto-HSE-ppp3751032.sympatico.ca) |
04:30.21 | dynaqueue | Yeah external and the GXP |
04:30.29 | dynaqueue | waiting for more to come :-( |
04:30.43 | X-Rob | OK, GXP line 1 is 306, and I have a sipura 2000 as 300 |
04:30.45 | X-Rob | line1 |
04:30.47 | X-Rob | 300 [send] |
04:30.55 | X-Rob | answer 300 |
04:31.02 | X-Rob | push 'TRNF' |
04:31.13 | X-Rob | I get dialtone on the GXP and nothing on the sipura |
04:31.22 | X-Rob | hrm |
04:31.23 | *** join/#asterisk starsoldier (~joey@laptop.arsenic.org.uk) |
04:31.26 | X-Rob | didn't happen that time |
04:31.53 | X-Rob | but what happened a couple of times whas that after you pushed 'TRNF' then pushed 'Line2' then pushed 'line2' again to put the sipura on hold, the sipura didn't get MOH. |
04:32.00 | X-Rob | maybe I need to wait a bit longer |
04:33.04 | dynaqueue | May be but I'm still not convinced they have the transfer sequence right. On the previous firmware you had the issue of hit transfer and both parties heard the DTMF tones.... So I still think that there is a lot to be done in that area. |
04:33.12 | *** join/#asterisk brc-tux (~brc-tux@212.70.34.110) |
04:33.20 | X-Rob | Bah, it's working now |
04:33.21 | *** part/#asterisk brc-tux (~brc-tux@212.70.34.110) |
04:33.26 | X-Rob | but it definately happened twice. |
04:33.27 | dynaqueue | I worked for a ACD company, and I'll tell you the hardest thing to get right are transfers and conferences... ! |
04:33.43 | X-Rob | Heh. Their problem was that they weren't expecting people just to push what they watn to do |
04:33.52 | X-Rob | you weren't ever meant to just push trnf |
04:34.03 | dynaqueue | LOL..... |
04:34.08 | pifiu | when is a new update for the gxps coming out? |
04:34.15 | X-Rob | you were meant to put the call on hold, then use another line, then push trnf and the on hold line |
04:34.28 | dynaqueue | Cant wait for them to electrify the chair we are using !.....if we push the wrong button ! |
04:34.28 | X-Rob | pifiu, it's on aussievoip now, grandstream shoudl be releasing it on monday our time. |
04:34.41 | pifiu | oh wow not bad |
04:34.46 | pifiu | so these updates are pretty often? |
04:34.59 | X-Rob | these phones are brand new |
04:35.06 | X-Rob | like -really- brand new |
04:35.15 | X-Rob | their first run of 5000 phones sold out in about 3 weeks |
04:35.17 | pifiu | so that means a lot of updates? |
04:35.20 | dynaqueue | Pifiu its hands better than what is there at the moment on the firmware that coems with the phones... |
04:35.25 | pifiu | damn so where can i get one? |
04:35.31 | X-Rob | they've just finished another run of 10,000 phones (apparently) and are shipping new ones now |
04:35.44 | pifiu | where are the phones from? |
04:35.48 | X-Rob | china |
04:35.54 | pifiu | and where can i purchase one? |
04:36.00 | X-Rob | grandstream dealer? |
04:36.01 | pifiu | i want to try one out |
04:36.06 | X-Rob | depends where you are |
04:36.08 | X-Rob | I'm in australia |
04:36.09 | pifiu | miami, fl |
04:36.13 | pifiu | USA |
04:36.16 | pifiu | online website? |
04:36.37 | X-Rob | if you can't type 'grandstream distributor' into google, I'm so not going to help you any more. |
04:37.29 | pifiu | lol wait |
04:37.36 | dynaqueue | So Rob... other than the transfer issue have you seen any other weird behaviour as yet. The speakerphone definitly has improved. |
04:37.50 | X-Rob | Hurm. |
04:38.15 | X-Rob | I want them to be able to support SIP ANSWER_IMMED, so they can actually be used as an intercom |
04:38.18 | Micc | Now all I need to do is build freetds and unixodbc and I'll be all set. |
04:38.23 | *** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar) |
04:39.03 | dynaqueue | Isnt that what the auto answer feature is for. Not tried it, but wondered if that would work ? |
04:39.30 | *** join/#asterisk michael1234 (~mick@staff.tsn.cc) |
04:39.36 | X-Rob | It does work |
04:39.39 | X-Rob | but it's not really what you want |
04:39.39 | michael1234 | does anyone in here run ser |
04:39.56 | X-Rob | You don't want to configure a seperate account on there purely for intercom |
04:39.57 | michael1234 | or is there a channel for ser |
04:40.43 | X-Rob | because then the user may pick the intercom line up inadvertently |
04:40.56 | MikeJ[Laptop] | who uses head? |
04:41.05 | dynaqueue | Thats where the electrified chair comes in handy ! |
04:41.15 | Micc | I use current cvs head |
04:41.32 | MikeJ[Laptop] | can you test a patch for me? |
04:41.51 | Micc | sure. |
04:42.04 | pifiu | wow rob |
04:42.11 | dynaqueue | Rob, well I would like for Grandstream to allow more access to the LCD they have...... |
04:42.12 | pifiu | they arent that expensive at all! |
04:42.20 | X-Rob | Micc - by the way, I haven't had any luck with using RTC |
04:42.20 | pifiu | i found one for 94 US |
04:42.25 | X-Rob | yeah, they're cheap as. |
04:42.29 | X-Rob | which is why they're so popular. |
04:42.35 | dynaqueue | Damn... Where Pifio.... $114 for my one ! |
04:42.46 | Micc | X-Rob, yeah its broken, but I got it to be good enough with non smp kernel. |
04:42.54 | pifiu | http://www.myphonecall.co.uk/voip/iptelephones/grandstream/default.aspx |
04:42.57 | pifiu | +shipping though |
04:43.01 | MikeJ[Laptop] | http://bugs.digium.com/view.php?id=4511 |
04:43.07 | pifiu | so it might be more since shipping is from europe |
04:43.10 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
04:43.17 | dynaqueue | yeah true |
04:43.49 | MikeJ[Laptop] | it just give background an option to stop streaming only on certain keys, instead of any dtmf |
04:44.31 | MikeJ[Laptop] | I need outside feedback as to if it works or not |
04:45.48 | *** join/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com) |
04:47.58 | Slup | pifiu is that not GBP? |
04:49.14 | MikeJ[Laptop] | if you can comment feedback on the bug that would be wonderful |
04:52.14 | *** join/#asterisk darwinfish (~funxion@69-165-200-17.miamfl.adelphia.net) |
04:53.54 | darwinfish | anyone still up |
04:54.24 | Pete_Largo | if that is British pounds (GBP) then it's more like $144 |
04:54.39 | Pete_Largo | but I did find them on ebay for $109 free shipping. |
04:55.26 | cluecon | darwinfish: who says we ever go down? |
04:56.52 | darwinfish | lol |
04:57.04 | darwinfish | just checking cuz I didnt see any activity |
04:57.15 | darwinfish | guess I didnt wait long enuff |
04:58.05 | Micc | MikeJ[Laptop], I've never applied a patch before, what do I do with this file? I've got it in my source directory now. |
04:58.21 | riksta | patch -p0 < patch.diff |
04:58.25 | MikeJ[Jayden] | thanks |
04:58.32 | Micc | actually, thats not true, but its been about 10 years since I've ran patch. |
04:58.39 | riksta | might have to adjust the -p value |
04:58.57 | MikeJ[Jayden] | Micc, then compile and install |
04:59.16 | dynaqueue | OffTopic < Anyone aware of the largest Asterisk deployment running in an ACD configuration ? |
04:59.33 | MikeJ[Jayden] | the patch adds a 4th option to background |
05:00.01 | MikeJ[Jayden] | you just set the 4th option to be the list of digits you want to make background stop playing and try that ext |
05:00.14 | MikeJ[Jayden] | any other digit, it will just ignore |
05:00.24 | riksta | if he cant apply a patch how is he going to test it :) |
05:01.11 | darwinfish | anyone using astcc |
05:01.25 | MikeJ[Jayden] | anyone else happy to test :D |
05:01.36 | Micc | I'm on the phone, hold on. |
05:01.38 | riksta | i use stable, dont like to walk on the wild side |
05:01.44 | Juggie | mikej, doesnt playback already do that |
05:01.57 | Juggie | i know in agi stream file will break on what you specify |
05:02.08 | MikeJ[Jayden] | allow you to break when hit a key? |
05:02.19 | MikeJ[Jayden] | background does |
05:02.24 | MikeJ[Jayden] | playback does not |
05:02.42 | Juggie | well if background does, what is the extra patch for |
05:03.00 | MikeJ[Jayden] | to specify which digits to break for, istead of any digit |
05:03.58 | MikeJ[Jayden] | it's for a project a friend is working on, but I have no box to test on at the moment |
05:04.07 | Juggie | oh |
05:04.10 | Juggie | agi already does that |
05:04.19 | Juggie | weird that you cant do it with background |
05:04.45 | MikeJ[Jayden] | in agi you do start and stop stream manually don't you? |
05:04.57 | MikeJ[Jayden] | I am not an agi user |
05:05.01 | Juggie | no |
05:05.06 | Juggie | its blocking |
05:05.10 | Juggie | it will return @ the end of file |
05:05.18 | Juggie | or when one the dtmf's in your list is pressed |
05:05.46 | MikeJ[Jayden] | hmmm |
05:06.09 | MikeJ[Jayden] | well... I need to go to bed.. if anyone tests that can you post in the bug please :) |
05:06.20 | Micc | ok. I'm off the phone. |
05:06.53 | Micc | ok it built. |
05:09.22 | michael1234 | what versions of ser do you guys run |
05:10.08 | pifiu | ok i think enough for tonight |
05:10.16 | pifiu | my brain is getting overloadead with info |
05:10.20 | pifiu | MUST SLOW DOWN lol |
05:11.04 | mosty | i'm following this walkthrough http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html and i'm trying to listen to my voicemail. i get a menu system that states when my messages were, but i can't figure out how to actually hear the messages |
05:14.43 | mosty | ahh, the messages were just zero sized files. fixed now |
05:16.47 | *** part/#asterisk Cresl1n (~matt@216.207.245.23) |
05:17.42 | darwinfish | anyone using astcc? |
05:23.19 | MikeJ[Jayden] | Micc, now background needs to be tried out both to make sure that it works like it used to with the 4th param ommited, and that when you specify the 4th param it works as it should... |
05:23.29 | MikeJ[Jayden] | ok.. I'm really going to sleep now |
05:26.25 | Micc | talk to me tomorrow mikej |
05:27.39 | Micc | I don't think I'll get to it tonight. I built and am using it now but I won't test it until later |
05:31.00 | X-Rob | Ah |
05:31.01 | X-Rob | smeg |
05:31.09 | *** join/#asterisk outtolunc (~me@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
05:31.14 | X-Rob | Micc, with the RTC patches, it looks like they -were- right |
05:31.34 | X-Rob | http://bugs.digium.com/view.php?id=4301 |
05:33.04 | outtolunc | updating |
05:33.06 | outtolunc | oops |
05:33.37 | *** part/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com) |
05:34.08 | Micc | X-Rob, do I need to apply these patches or are they already in CVS-HEAD? |
05:34.26 | Nugget | meetme and ztdummy need to die. |
05:34.26 | *** join/#asterisk Astermigraine (~psolomon@69-165-217-96.atlsfl.adelphia.net) |
05:35.19 | X-Rob | they're already there |
05:35.22 | X-Rob | that's what you enabled |
05:35.27 | X-Rob | you can go back to hyperthreading and SMP |
05:35.29 | X-Rob | I think |
05:35.32 | X-Rob | *checking now* |
05:35.44 | X-Rob | hyperthreading and SMP breaks ztdummy badly -without- RTC |
05:35.51 | Micc | No, I tried that and it didn't work. |
05:36.01 | X-Rob | it should |
05:36.12 | X-Rob | you won't get 100&, nor will you get 99.98 |
05:36.18 | Micc | I might need to apply that patch still, there was more to it than just the #if |
05:36.21 | X-Rob | but you'll get it bouncing between more and less than per seconds |
05:36.37 | X-Rob | Micc, that patch is allready in ztdummy |
05:36.46 | X-Rob | that ifdef just comments it out |
05:36.59 | Micc | Ok, well then, it doesn't help with smp kernel. |
05:37.04 | riksta | does anyone in here use ADM ? |
05:37.15 | X-Rob | Rebooting now and verifying this fact. |
05:37.26 | X-Rob | I've been researching this in my long quiet period here |
05:37.35 | Micc | X-Rob, I've got pretty good response time now with my meetme I think. |
05:38.03 | Micc | X-Rob, I'll give you my 800 number if you want to see how mine sounds. |
05:38.09 | X-Rob | I'm in australia 8) |
05:38.14 | Micc | doh |
05:38.17 | X-Rob | And FWD won't let me dial 1-800 numbers at the moment |
05:38.20 | X-Rob | (for some reason) |
05:38.34 | Micc | can you dial regular US numbers? |
05:38.36 | X-Rob | Actually, I should be able to |
05:38.42 | Micc | I've got about 10 DIDs |
05:38.50 | X-Rob | Give me an 800 number and I'll see what I can do |
05:39.02 | Micc | (888) 206-4443 |
05:39.57 | Nugget | if you enable enum lookups there's some host out there that will grab any US toll free number you dial and let you route via iax. |
05:39.57 | X-Rob | whichroom no? |
05:39.59 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
05:40.02 | Micc | 1234 |
05:40.14 | Nugget | I dunno who they are, but they want you to route toll free calls through them |
05:40.40 | Nugget | enum lookups are good karma anyway. |
05:41.03 | Nugget | http://www.e164.org/config.php |
05:41.08 | X-Rob | heh |
05:42.33 | Nugget | if you had enum lookups you could call me, and about 320,000 other phone numbers, for free. |
05:42.53 | X-Rob | Zapata Telephony Interface Registered on major 196 |
05:42.53 | X-Rob | ztdummy: using RTC |
05:42.53 | X-Rob | Registered tone zone 0 (United States / North America) |
05:49.19 | pifiu | quick question |
05:49.22 | pifiu | in the sip.conf |
05:49.36 | pifiu | the callerid= |
05:49.42 | pifiu | when you write it for example like this |
05:49.54 | pifiu | callerid=pifiu <?> what is ? |
05:49.56 | Slup | one quick Q befor i go to bed .... |
05:50.19 | Micc | I reboot that machine. |
05:50.33 | Slup | anyone here got good ethernet and phone cabeling knowledge? |
05:51.04 | X-Rob | what do you want to know? |
05:51.58 | Slup | on a ethernet cable only half the pairs are used and convienently the middle two pins on the blue pair are free which is what a rj11 phone lead would connect too .... |
05:52.50 | X-Rob | That's correct, unless you're using gigabit |
05:52.56 | X-Rob | gigabit uses all 4 pairs. |
05:53.12 | Slup | but the local earth 'rinning' cable would normaly be on orange white which is used |
05:53.13 | Slup | yes |
05:53.38 | X-Rob | That last line made no sence to me whatsoever. |
05:53.40 | Slup | not got that luxaury yet :( |
05:54.00 | Slup | mayby you don't have it in Oz |
05:54.22 | X-Rob | Convetnion has it that blue/white is a pstn line |
05:54.33 | X-Rob | or red/wite is voice on a MDF |
05:55.08 | Slup | on the UK system you have a master socket with the ringing cap and resister so it is not needed in every phone or secondary socket |
05:55.23 | Slup | standard pstn lines that is |
05:55.28 | X-Rob | OPoh yeah, this was mentioned on asterisk-users today I think |
05:55.35 | X-Rob | Never had that in oz |
05:55.39 | X-Rob | we have two wire phones |
05:55.43 | X-Rob | no ground thing. |
05:56.11 | Slup | which goes out on orange white which is one of the ground wires on the ethernet side of things |
05:57.00 | Slup | you can have that in the uk too but you need to use phones that have the ringing cap and resister in the phone or it won't ring |
05:58.29 | X-Rob | don't use colours, use pin numbers 8) -- RJ45/100BaseT uses 1,2,3 and 6. |
05:58.54 | Slup | so i'm just wondering if you had ringing on the orange and white wire shaired with the orange and white of the network am i going to start shafting things |
05:59.16 | Slup | mmm just sent my other pc whith all that kind of stuff on into hybernate |
05:59.18 | X-Rob | I'm guessing that sending a 48v signal into a network card is going to be bad. |
05:59.26 | X-Rob | smoke and sparks bad. |
05:59.28 | Slup | i thought the coulours were an int standard |
05:59.42 | Slup | lol yea thats that i was thinking |
05:59.45 | Slup | hmmm |
05:59.57 | X-Rob | Micc, how'd your reboot go? |
06:00.28 | Micc | machine freaked out and didn't boot. |
06:00.36 | Micc | Had to call up and have someone at the datacenter reboot it. |
06:00.41 | X-Rob | ick |
06:00.42 | Micc | my boss just happened to be there still. |
06:00.51 | X-Rob | That could be good or bad |
06:01.21 | Micc | ok, zttest looks ok. |
06:01.42 | Slup | looks like i'd have to either use a master mod tap with the ringing cap in it or move the ringing wire to the brown pair and make moded rj11 cables |
06:01.44 | jerlique | does anyone have a url to a flow diagram of how SIP and RTP sessions work with SER and say an * media proxy?? |
06:02.04 | Micc | how do I see how many cpus my machine thinks it has? |
06:02.46 | X-Rob | cat /proc/cpuinfo |
06:02.51 | Slup | X-Rob do you know of any site where i can find info on the various rj11 standards i've had a google but didn't find much |
06:03.28 | Slup | well not amoungst the multitude of component catalogues which didn't have much info |
06:04.32 | Slup | or anyone else whos still alive |
06:04.50 | Pete_Largo | pifiu the extension number: callerid=pifiu <1002> for example |
06:06.01 | X-Rob | http://www.faqs.org/faqs/LANs/cabling-faq/ |
06:06.31 | *** join/#asterisk fugitivo (~ajf@201.255.104.46) |
06:06.33 | fugitivo | hello |
06:06.56 | Slup | robs the man !!! |
06:07.46 | Slup | rob i hope while your not on here your goign out and earning good money and not just wastiing your talents on us |
06:08.00 | Pete_Largo | putting ring voltage on a UTP cable that has data running on it is probably not a good idea re: the continuity of the data.... probably be less of a headache to run a second cable. |
06:08.14 | Micc | X-Rob, could you call in again? |
06:08.17 | fugitivo | what's the best codec for a adsl 256kbit/s conection? |
06:08.41 | Pete_Largo | ~codec |
06:08.42 | jbot | somebody said codecs was http://snipurl.com/wiki_codecs |
06:09.03 | fugitivo | thanks |
06:09.21 | Slup | Pete_Largo yup thats what i was thinking but you never know so i thought i'd stick the question in |
06:10.11 | fugitivo | is iLBC a good option? |
06:10.12 | Pete_Largo | I've never done that before Slup, that's my story and I'm stickin' to it :) |
06:13.28 | Micc | Thanks for the help, X-Rob. |
06:13.46 | X-Rob | no probs |
06:13.55 | X-Rob | I'd like to find out where all that latency is coming from tho |
06:13.57 | X-Rob | coz that was crud |
06:14.15 | X-Rob | well |
06:14.28 | X-Rob | actually it would be voip -> us -> pstn -> voip -> you |
06:14.38 | X-Rob | at least 4 conversions |
06:15.18 | Micc | I'll test it some more later. |
06:15.28 | X-Rob | ah |
06:15.37 | X-Rob | fwd is now letting me dial 1800's |
06:16.26 | *** join/#asterisk drooth (~drooth@ip68-107-113-76.sd.sd.cox.net) |
06:16.31 | drooth | at long last i have QoS!!!!!!!! |
06:16.38 | drooth | I am using latest PFsense |
06:17.08 | fugitivo | anyone is using qos with linux? |
06:17.31 | drooth | i am maxing out at 3 calls however, 4th call and it's choppy |
06:18.11 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
06:19.44 | *** join/#asterisk loud (~roots@cypher.punk.net) |
06:19.48 | Pete_Largo | Hey, I just had a brainstorm... IP Phone with an analog port on the side... so you can plug in your regular house phones like an ata... |
06:42.50 | *** join/#asterisk drooth (~drooth@ip68-107-113-76.sd.sd.cox.net) |
06:42.57 | drooth | how do you transfer a call in * |
06:44.04 | X-Rob | Um |
06:44.10 | X-Rob | DEpends |
06:44.19 | X-Rob | look at /etc/asterisk/features.conf |
06:44.26 | X-Rob | that's where you configure the hotkey |
06:44.35 | X-Rob | and you need to have the dial string with 'Tt' on it |
06:44.46 | X-Rob | usually, you use your phone to do transfers. |
06:45.09 | drooth | yeah i am using a spa-841 |
06:45.16 | drooth | display is kinda crappy |
06:47.16 | drooth | X-Rob: I don't have any hotkeys there |
06:47.25 | drooth | X-Rob: do you have an example you could paste |
06:47.38 | drooth | X-Rob: like hitting # twice or..? |
06:48.32 | drooth | to put the caller on hold and then allow to transfer |
06:48.51 | X-Rob | [featuremap] |
06:48.51 | X-Rob | ;blindxfer => #1 ; Blind transfer |
06:48.51 | X-Rob | ;disconnect => *0 ; Disconnect |
06:48.51 | X-Rob | ;automon => *1 ; One Touch Record |
06:48.52 | X-Rob | ;atxfer => *2 ; Attended transfer |
06:49.01 | X-Rob | That's possibly only in CVS HEAD |
06:49.04 | drooth | wow |
06:49.13 | drooth | cool |
06:49.41 | drooth | CVS HEAD? |
06:50.16 | drooth | let me paste that in and restart |
06:50.34 | X-Rob | you'll need to uncomment them |
06:50.37 | drooth | right |
06:50.38 | X-Rob | ; == comment |
06:51.00 | X-Rob | heh |
06:51.04 | X-Rob | new to IRC are you? 8) |
06:51.10 | drooth | no |
06:51.12 | drooth | tired |
06:51.14 | X-Rob | hehehe |
06:51.16 | drooth | been on for 8 yrs |
06:51.20 | drooth | approx |
06:51.34 | drooth | so those are universal hotkeys i can use when people call |
06:51.46 | drooth | but can only be used if logged into queue |
06:51.55 | drooth | and i have answered a call |
06:52.38 | X-Rob | http://www.voip-info.org/wiki-Asterisk+cmd+dial |
06:52.41 | X-Rob | read that too |
06:53.04 | drooth | thx |
07:00.23 | *** join/#asterisk cmk (~cmk_@p54A3F060.dip.t-dialin.net) |
07:01.00 | drooth | read it... |
07:01.07 | drooth | interesting |
07:02.16 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
07:02.29 | drooth | Attempts to establish a new outgoing connection on a channel, and then link it to the existing input channel. not quite sure what that means actually |
07:02.34 | drooth | conference? |
07:03.14 | X-Rob | where's that? |
07:03.19 | X-Rob | conferencing is done with apps |
07:03.22 | X-Rob | app_meetme and such |
07:03.28 | drooth | i have conferencing |
07:03.34 | drooth | that is from the link ou gave me |
07:03.58 | X-Rob | Oh |
07:04.10 | X-Rob | That's a complex way of saying 'It dials this number' |
07:04.21 | X-Rob | but |
07:04.22 | X-Rob | uh |
07:04.27 | X-Rob | lemme rephrase that |
07:04.27 | drooth | so why did you give me that link? |
07:04.39 | X-Rob | BEcause on the Dial command, you need to have 't' and/or 'T' options |
07:04.45 | X-Rob | before you can transfer a call |
07:05.54 | drooth | i hit # and i was able to transfer! wow |
07:06.06 | drooth | guess I have it already |
07:06.09 | drooth | using a modified config |
07:07.14 | X-Rob | You're using Asterisk@Home I guess? |
07:07.50 | X-Rob | (It helps to say that at the start 8) |
07:08.59 | drooth | yeah |
07:13.55 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
07:15.38 | jerlique | why is something like rtp proxy used for helping traverse nat'ed clients? My understanding is that SER can re-write the packets to show the correct ip addresses.... |
07:16.09 | drooth | seems that if I hit # I can do a blind transfer. dope |
07:16.30 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
07:18.42 | X-Rob | you'll need CVS-HEAD to do attended transfers |
07:19.28 | *** join/#asterisk cmk (~cmk_@p54A3F060.dip.t-dialin.net) |
07:21.33 | drooth | ok |
07:22.46 | xeet2 | attended transfers aren't in stable? |
07:24.52 | X-Rob | afaik not |
07:24.55 | X-Rob | Not that I've tried it |
07:38.28 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
07:38.29 | drooth | where are the voice messages located for each extension? |
07:39.26 | mosty | drooth, they should be in /var/spool/ somewhere |
07:39.31 | Hmmhesays | joan jett is scarey now |
07:40.05 | drooth | umm /var/spool/asterisk/vm |
07:42.57 | *** join/#asterisk loud (~roots@cypher.punk.net) |
08:04.54 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
08:08.30 | drooth | anyone know how i can change the codec per call? |
08:08.42 | drooth | like the first 2 calls get highest codec 80k and then all calls after get lower quality |
08:10.31 | drooth | how do you record a greeting by calling into * via a phone |
08:12.52 | jerlique | 80k?? what codec is that? |
08:15.09 | *** join/#asterisk dasuberdavid (~david@pcp01540029pcs.huntsv01.al.comcast.net) |
08:15.11 | drooth | highest quality |
08:15.21 | drooth | including overhead |
08:15.29 | drooth | 64k + IP overhead |
08:23.02 | drooth | how do you dial a SIP URL from firefly? |
08:30.47 | *** join/#asterisk Godsey (lanny@204.17.223.9) |
08:46.57 | X-Rob | drooth, I don't think you can. |
08:47.18 | drooth | hmm i just installed new version |
08:47.28 | drooth | it looks like they took out features to dial your own IAX network?? |
08:57.59 | drooth | http://www.virbiage.com/firefly/download/firefly-thirdparty.exe |
09:07.36 | *** join/#asterisk zoa (zoa@82.103.76.147) |
09:50.07 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
09:51.07 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
09:53.28 | *** join/#asterisk core-ix (~Administr@2001:618:495:3:2d0:b7ff:fe0f:ba6) |
10:20.47 | *** join/#asterisk mcra (~alutiu@satlink.ro) |
10:49.06 | *** join/#asterisk ruski (ruski@adsl-160-122.swiftdsl.com.au) |
10:50.16 | ruski | I've setup asterisk to work with my SIP provider, they dont know about it, when i make an incoming call from the outside world, it bounces through my provider then into my asterisk then onto the voip phone, when im calling i cant hear any ringing on the handset, it's just silent .. any ideas? |
10:51.07 | Mavvie | you should tell them about it. |
10:51.25 | *** join/#asterisk twisty (twisty@ndn-165-143-17.telkomadsl.co.za) |
10:51.30 | twisty | hi all. |
10:51.34 | Mavvie | are you registering? |
10:52.06 | ruski | Mavvie: they'd most probably not be very happy, yeh i get registerd, i can even make calls to the voip phone |
10:52.14 | ruski | it's just the calling party doesnt hear any ringing |
10:52.14 | twisty | this will probably sound stupid but how do you tell what version of asterisk you are running other than the show version |
10:52.28 | ruski | just silence until the reciever answers the call |
10:52.59 | Mavvie | it's the local phone who makes the ringing sound until it's picked up. |
10:53.40 | Mavvie | until the call is answered that is |
10:54.42 | twisty | ok. is there anything that one should specificaly do when updateing. other than backup the config files. |
10:55.04 | cochi | r-option to the Dial command or Ringing command in the extension on lower prio? ;) |
10:55.13 | *** join/#asterisk outofjungle (~outofjung@61.17.134.230) |
10:58.08 | *** join/#asterisk meppl (mephisto@p54AABBEC.dip.t-dialin.net) |
10:59.28 | twisty | are there many people following this chat ? |
10:59.47 | Mavvie | no |
11:00.26 | twisty | Mavvie : am about to update asterisk installation. is there anyhting i should watch out for or do specificaly to ensure that it comes up again without problem |
11:00.50 | Mavvie | just make sure you have the old binaries still, just in case. |
11:00.59 | twisty | have done that thanks/ |
11:12.50 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
11:13.12 | twisty | Mavvie : this is not an urgent one but i set the loadzone and defaultzone to za. for south africa but asterisk still does not pickup on the engaged tone and runs through all options before hanging up the line. do you have an idea why this is happening or is there something else to change the zone ? |
11:16.49 | *** join/#asterisk nazgool (~nazgool@p62.246.134.116.tisdip.tiscali.de) |
11:16.50 | nazgool | hi |
11:19.22 | nazgool | i just tried to make a modules.conf with autoload=no. after solving my module dependency issues so far, i got asterisk starting up without errors, but: |
11:20.15 | nazgool | when i say "extensions reload" i get the error "No such command extensions'" |
11:20.56 | nazgool | what module must i load to have the extensions command? |
11:22.10 | *** join/#asterisk nicox (~nicox@h082218027030.host.wavenet.at) |
11:22.33 | nazgool | (the modules loaded so far are: res_features.so; app_parkandannounce.so; res_musiconhold.so; chan_sip.so; format_mp3.so; format_wav.so; codec_ulaw.so; chan_capi.so) |
11:22.58 | nazgool | any really crucial module i would be missing to run asterisk with sip + capi? |
11:23.38 | nazgool | (well, obviously there's at least one missing: the one that would give me the 'extensions' command) |
11:23.42 | nazgool | ? |
11:24.51 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
11:27.44 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
11:33.59 | ruski | is there a way in asterisk to work out what codecs a peer // another sip server has available ? |
11:36.24 | Romik_ | ruski: ty hot' ponyal chto sam to skazal? |
11:40.53 | nicox | is there any specialist with extension.conf? |
11:42.33 | Romik_ | nicox: there a lot of guru's of this file...;) |
11:43.11 | djin | nicox, define specialist. |
11:44.10 | Romik_ | djin: specialist is 1 steps before guru the step name is expert ;) |
11:46.03 | djin | so nicox, phat's the wroblem? |
11:49.45 | *** join/#asterisk stkn_ (nobody@stkn.developer.gentoo) |
11:51.15 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
11:51.52 | *** join/#asterisk Yellow_Fuzzy (~yellow@c211-31-41-107.wavrl1.nsw.optusnet.com.au) |
11:52.03 | Yellow_Fuzzy | hi |
11:53.05 | Yellow_Fuzzy | anyone here got or used a Draytek 2900V |
11:54.08 | djin | I use a 2600VG, not sure if that covers your question. |
11:55.01 | newl | Actually, the specialist would probably be a grunt, then the SME (subject matter expert), then the GURU. :D |
11:55.20 | Yellow_Fuzzy | basically i bought a 2900V today and connected it to my optus cable modem and have tested and discovered that my internet speed is about 200KB/s slower when through the 2900V then when i plug directly into a pc |
11:55.57 | newl | Blame Optus. Everyone else does. haha |
11:56.46 | djin | Yellow_Fuzzy, I don't seem to have that problem. Download is running at maximum. |
11:56.55 | Yellow_Fuzzy | i know its not the 2900V that is slowing it down because as i said it works 200KB/s faster running the same speed test |
11:57.04 | Yellow_Fuzzy | and ive updated to the latest firmware |
11:57.55 | Yellow_Fuzzy | and ive tried turning off QoS & all the firewall features |
11:58.14 | *** join/#asterisk nemisus (~nemisus@203-206-228-13.dyn.iinet.net.au) |
11:58.22 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
11:58.54 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
11:59.25 | Yellow_Fuzzy | any ideas of what i could try to get it to work properly |
12:00.36 | djin | No, I'm sorry + I think this is not the best place to find answers. |
12:01.30 | Yellow_Fuzzy | yeh i know its probably not the best place, but ive posted on a few forums & am waiting for answers from them |
12:02.39 | djin | nicox, did you try setting the DigitTimeout? |
12:02.40 | djin | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DigitTimeout |
12:03.33 | djin | and ResponseTimeout |
12:03.33 | djin | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ResponseTimeout |
12:05.48 | nicox | how do i set this for a PRI? |
12:06.33 | djin | For a PRI? You can set this in the extensions.conf |
12:07.03 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
12:07.03 | *** mode/#asterisk [+o bkw_] by ChanServ |
12:09.13 | MikeJ[Laptop] | morning bkw_ |
12:11.06 | nicox | (msg djin i read this "If not specified in the dialplan, the default is 5 seconds. " on the website but i have no 5 seconds to dial the correct number. so i think this is no solution!? |
12:11.12 | *** join/#asterisk gres (~serg@81.222.48.242) |
12:19.19 | *** join/#asterisk dudes (~dudes@12-215-34-84.client.mchsi.com) |
12:34.07 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
12:35.47 | *** join/#asterisk asterbeast (~acidburnn@cust.7.204.adsl.cistron.nl) |
12:35.50 | asterbeast | ppl... |
12:36.16 | asterbeast | i have some probs trying to call out to the 612 number for time |
12:36.32 | asterbeast | Dial prefix is 393... |
12:36.43 | asterbeast | can someone help? |
12:39.48 | *** join/#asterisk Cybertank (~cybertank@CPE000dbd0f269c-CM00111ae6ff9c.cpe.net.cable.rogers.com) |
12:42.23 | asterbeast | i have some probs trying to call out to the 612 number for time.. my prefix is 393 but get -- IAX2/fwd/3 is busy |
12:42.36 | asterbeast | can someone point me to the right direction? |
12:45.43 | djin | what is your exten line? |
12:47.02 | asterbeast | in extensions_additional.conf? |
12:47.31 | asterbeast | [outrt-002-test393] |
12:47.31 | asterbeast | include => outrt-002-test393-custom |
12:47.31 | asterbeast | exten => _393.,1,Macro(dialout-trunk,2,${EXTEN:3}) |
12:47.31 | asterbeast | exten => _393.,2,Macro(outisbusy) ; No available circuits |
12:47.53 | asterbeast | is this my prob?? "No available circuits" |
12:48.58 | asterbeast | no.. i dont think so.. |
12:49.28 | *** join/#asterisk destro_dm4 (~rdsaffell@dsl092-161-155.wdc2.dsl.speakeasy.net) |
12:49.39 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
12:50.11 | destro_dm4 | anyone know how to automatically add a prefix (9) to numbers in the call list on Polycom 500 phones? |
12:56.50 | *** join/#asterisk wooden (zaiwot@p54AA03DD.dip0.t-ipconnect.de) |
12:56.55 | wooden | hi |
12:57.05 | wooden | ive a problem with meetme, conferences |
12:57.15 | wooden | chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory |
12:58.08 | JerJer | you don't have zaptel timing |
12:58.33 | wooden | i use the debian 3.1 asterisk |
12:58.44 | JerJer | that was your first mistake |
12:59.13 | wooden | oh there are zaptel package in debian |
12:59.35 | JerJer | use the source - that's why it was written |
13:00.18 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
13:00.25 | RoyK | ~seen inspired |
13:00.26 | jbot | inspired is currently on #asterisk |
13:01.38 | wooden | i think the support for chan_zap isnt build in debian? |
13:01.42 | wooden | u |
13:02.14 | asterbeast | i have some probs trying to call out to the 612 number for time.. my prefix is 393 but get -- IAX2/fwd/3 is busy |
13:03.14 | JerJer | then the number is busy |
13:03.29 | JerJer | wooden: use the source |
13:04.30 | asterbeast | ok.. JerJer... but this number is the standard for time isnt it? |
13:07.11 | asterbeast | everythig i call (when using the prefix 393 ) ends up with: "Executing Playback("SIP/200-e685", "allison7/all-circuits-busy-now") in new stack" |
13:08.22 | Ariel_ | asterbeast, are you using amp for your setup? |
13:08.37 | asterbeast | and the console yes... |
13:09.32 | Ariel_ | did you setup fwd as sip or iax2? |
13:10.05 | Romik_ | any one uses bluetooth channel in asterisk? |
13:11.19 | *** join/#asterisk WeezeyD (Weezey@206.210.111.117) |
13:12.11 | *** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
13:13.36 | WeezeyD | cluecon: you have a PBX Developers Conference in Chicago from Aug 3rd to 5th. |
13:13.40 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
13:14.16 | wooden | i need for zaptel kernel drivers? |
13:14.20 | Cybertank | Anyone else install asterisk and digium card in Fedora Core?? |
13:15.09 | *** join/#asterisk Laibsch (~Laibsch@G1dfb.g.pppool.de) |
13:17.00 | asterbeast | Ariel_ : iax2 |
13:17.47 | Cybertank | What distro is most recommended for asterisk and digium? |
13:19.21 | asterbeast | CENTOS??? from the asterisk@home project |
13:22.12 | cluecon | Cybertank: I use FC3 and it works. |
13:23.46 | dwmw2_gone | Romik_: I do |
13:24.11 | *** join/#asterisk nain (nain@61.5.143.19) |
13:24.20 | nain | Hi Every body |
13:24.32 | WeezeyD | Hi Doctor Nain! |
13:24.35 | Ariel_ | asterbeast, did you follow the hand book on setting up fwd on with Asterisk@home? |
13:24.55 | nain | WeezeyD: I am Not Doctor :) |
13:25.10 | WeezeyD | But you play one on TV? |
13:25.15 | Ariel_ | Cybertank, most will work. There are post on the wiki on how to setup your choice. |
13:25.27 | WeezeyD | how do I grep through a directory full of gzipped files? |
13:25.30 | Ariel_ | Cybertank, which Distro do you like for your other use? |
13:26.08 | nain | WeeseyD: That may be some one else |
13:26.29 | Yellow_Fuzzy | Weezey: as in you want to search through the contents of thoes gzipped files? |
13:26.36 | nain | Well Any one Can help me ? |
13:26.59 | nain | I want to recieve IP call in SIP using asterisk ????? |
13:27.19 | Ariel_ | nain, that is what asterisk does |
13:27.38 | newl | wees: something akin to: for i in `ls .`; do gzip -l -c $i|grep foo; done |
13:27.41 | file | ^^^ that is what chan_sip does |
13:28.35 | nain | Ariel_: yes but my problem is that carrier is sending in H323 and i have SIP account So i want to directly calls to SIP not in H323?? |
13:29.04 | nain | I don't want to use chan_h323 or chan_oh323.so |
13:29.25 | Ariel_ | nain, then get a different carrier |
13:29.37 | file | uh well, there's another channel driver in asterisk-addons too... but if they only do H323, nothing you can do about it |
13:30.45 | nain | Actually problem is that i was using h323 too but H323 driver has lot of problem to me like no ringtone or dead air, Protocol Error, So i decided to recieve call only in SIP as my provider is USING SIP |
13:31.21 | file | you're contradicting yourself |
13:31.36 | newl | indeed. |
13:31.57 | nain | for the above said reason i don't want to more use H323 . I HAVE DISABLE H323 DRIVER and only want call in SIP |
13:32.18 | *** join/#asterisk Rez (lorez@lorez.staff.freenode) |
13:32.22 | file | yeah... so only use SIP |
13:32.45 | nain | but customer is sending call in h323 how to recieve in SIP ? |
13:32.57 | file | you have to tell them to do it... |
13:33.05 | file | that's something between you and your provider |
13:33.06 | file | not asterisk |
13:33.09 | nain | They are not ready to do it |
13:33.18 | newl | or you and the customer. :) |
13:33.32 | file | well then what are you asking? |
13:33.40 | file | "I want to do SIP... but they're using H323, and I don't want to" |
13:33.41 | file | boo who |
13:33.45 | nain | Actually in Customer and me there is VOIPESCROW which is doing this conversion but calls are failed |
13:34.39 | newl | Sell the customer a SIP enabled device, set them up with a provider, collect the commission, problem solved. |
13:35.22 | nain | But Whole Sale Carrier Only like to Use H323 not SIP |
13:35.48 | RoyK | Guten Tag |
13:35.58 | file | nain: and what do you want us/asterisk to do? |
13:37.00 | nain | file: Recieve call from particular IP in SIP and Dial to my Provider |
13:37.31 | file | that's what asterisk does... go Google, tons of examples |
13:37.35 | file | literally TONS |
13:38.22 | nain | but that examples are with chan_oh323 or chan_h323. I don't want to involve h323 |
13:38.25 | RoyK | ~docs |
13:38.28 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
13:39.06 | Ariel_ | nain, your going to have to setup h323 of some type on your system. if not then your not going to get the calls. Again if you don't want h323 get a different carrier there are many that do sip or even iax2 |
13:39.31 | RoyK | nain: no, just configure a sip account in sip.conf with context=asdf and in extensions.conf add [asdf] with exten => dial(SIP/${EXTEN}@provider) |
13:39.36 | RoyK | nain: rtfm |
13:39.52 | file | Ariel_: thx |
13:39.59 | file | today is not turning out to be a good day |
13:40.08 | file | Swiss Chalet got rid of my chicken strips |
13:40.09 | file | SOOOOOOO |
13:40.11 | file | I'm cheating |
13:40.20 | file | and getting two kids chicken strip dinners :P |
13:40.22 | RoyK | nain: and if your provider only runs h.323, there's no fucking way you can avoid using it yourself |
13:40.56 | nain | RoyK: i have a context like [voip-escrow] host=carrier-ip ...... in context=sip-call in sip.conf and have extenions in extension.conf which is dialing to my provider |
13:41.21 | RoyK | nain: have you read the asterisk docs yet? |
13:41.50 | nain | RoyK: yes ! But some one tell me that it is still possible without using h323 . |
13:42.02 | newl | heh |
13:42.04 | RoyK | what isn't possible without h323?? |
13:42.17 | nain | IP to SIP |
13:42.26 | RoyK | SIP to SIP? |
13:42.46 | nain | ofcourse SIP to SIP is no problem but problem is IP to SIP without involvind h323 |
13:42.51 | RoyK | iirc IP is a layer three protocol and SIP a layer 7 protocol, so IP to SIP is quite hard.... |
13:43.15 | newl | 'e's lost the plot gov! |
13:43.38 | file | RoyK: you're mean |
13:43.44 | RoyK | SIP over NetBEUI :) |
13:43.53 | RoyK | file: i'm well aware of that |
13:44.03 | Ariel_ | nain, h323 is like spanish and sip is like english the don't talk to each other unless there is a translator(asterisk) but for the translator to do this it needs to speak both spanish and english |
13:44.04 | nain | SOme one tell me that in sip account set host=IP not dynamic and qualify=yes, insecure=very will work but it is not working |
13:44.32 | file | food ordered, yay |
13:44.36 | Ariel_ | nain no no no sip is different the h323 |
13:45.01 | file | nah, german! |
13:45.08 | newl | javole! |
13:45.11 | nain | I KNOW SIP IS DIFFERENT THEN OH323 BUT FEW PEOPLES ARE DOING AND THEY SAID IT IS TRICK. JUST GO AND FIND |
13:45.33 | Ariel_ | nain, there pulling your legs |
13:45.39 | nain | Strange |
13:45.45 | newl | both of them. :) |
13:46.17 | WeezeyD | damn, why does mgcp do *67 by default, but SIP doesn't. That's irritating |
13:46.28 | nain | They are very expert and Professional. What do u say to VOIPESCROW TECH. An online billing solution ? |
13:46.46 | file | WeezeyD: SIP doesn't do it, the phone does it - on MGCP, the server (asterisk) does it |
13:46.55 | file | nain: SIP |
13:46.56 | file | gah |
13:46.58 | nain | I m using Asterisk as Server |
13:47.19 | file | nain: SIP isn't H323, let's just get that down pat... if you are using H323 in some form, you need an H323 channel driver (chan_h323, chan_oh323, chan_ooh323) |
13:47.32 | newl | or a gatekeeper |
13:47.45 | Ariel_ | nain, if they send you signal in h323 the only way asterisk is going to know it is with the h323 drivers one of 3 out there. You have to have a translator. |
13:47.49 | file | or something else that'll do H323 and SIP on it's own and do the conversion of signalling for you |
13:47.56 | WeezeyD | file: the SIP phone does it 'eh? Hmm... |
13:48.11 | file | WeezeyD: if it doesn't, you write dialplan logic to |
13:48.24 | file | SIP phones do most everything |
13:48.41 | WeezeyD | file: I think it does, I just have sip.conf setting it, so that would be over-writing it. |
13:48.42 | Ariel_ | Weezey, you can setup rules to do that. Also like some sip device sipura it's in there configuration for many *67 and other codes built-in |
13:49.14 | nain | ok you are right |
13:50.28 | nain | but they said that by setting host=IP SIP.conf is still able to recieve and dial the extensions. I have succeeded lit bit but calls are failing. The person who guide me is not available due to Weekend and he is using the Trick. So i can't wait for him for 2 days Therefor i m here to ask u for help ???? |
13:50.50 | file | if you want SIP to work that's easy, like I said there's TONS of examples out there |
13:50.51 | RoyK | ~lart nain |
13:50.55 | nain | I DON'T KNOW WHAT LOGIC HE IS USING, HOWEVER HE IS DOING |
13:50.58 | newl | heh I've got dialplan logic for all that fun stuff. Didn't take but a couple of days from start to finish. It'd be nicer to have built in though and controlled through features.conf. :) |
13:51.19 | file | SIP and H323 are two completely different configured things |
13:51.34 | file | so, go find examples and don't give me "oh I have... there's none" |
13:51.36 | file | because I know there's TONS |
13:51.37 | nain | Hmmmmmmmmmmmmm |
13:51.46 | file | so go use your brain, common sense, and read |
13:51.49 | RoyK | SIP is a nice protocol designed with headers like HTTP. H.323 is more like ISDN over IP |
13:51.51 | nain | THANKS |
13:53.29 | nain | OK ONE MORE THING DID GnuGk can convert h323 calls to SIP ? |
13:54.21 | RoyK | nain: no, nothing can convert between SIP and H.323 |
13:54.51 | nain | SO I MUST USE CHAN_OH323.SO DRVIER FOR DOING THIS |
13:55.01 | RoyK | nain: but asterisk and possibly yate can perhaps bridge between them |
13:55.10 | RoyK | nain: don't use caps you moron |
13:55.41 | nain | OK |
13:55.44 | RoyK | nain: you will have to use some channel driver for h.323 to use h.323, yes |
13:57.03 | nain | OK I have also used asterisk_oh323.0.7.1 and asterisk_oh323.0.7.2pre1 driver but there are some problems with using these drivers for examples, noringtone to carrier, dead air, or protocol errors etc. |
13:57.40 | nain | As these are the latest Channels Drivers for oh323 |
13:58.29 | RoyK | i beleive chan_h323 with asterisk cvs head is the best you get |
13:58.31 | cluecon | nain, welcome to the world of h323 |
13:58.31 | RoyK | by now |
13:58.35 | RoyK | although i'm not sure |
13:58.57 | newl | So isolate the cause of the bugs, fix, submit patches, enjoy the open source wave. :) |
13:58.59 | Ariel_ | nain, h323 does work. But there not easy to setup. That is the main reason poeple are going away from h323. |
13:58.59 | nain | I have used latest CVS head version of 3 days before of asterisk |
13:59.04 | RoyK | nain: but, there is something called aefirion that is an asterisk fork and that has an OPAL-based h323 driver |
13:59.09 | RoyK | that should perhaps work better |
13:59.10 | RoyK | or |
13:59.12 | RoyK | Yate |
13:59.16 | RoyK | yate.null.ro |
13:59.25 | RoyK | that uses h323 natively |
13:59.34 | RoyK | which again may work better than asterisk on h323 |
13:59.39 | RoyK | but then |
13:59.53 | nain | hmmmmmmmmm but is Yate do translation from h323 to SIP? |
14:00.01 | RoyK | no, nain |
14:00.14 | RoyK | but as i said, it may gateway between them |
14:00.16 | nain | So the problem is same still |
14:00.22 | RoyK | listen to what i say |
14:00.24 | RoyK | idiot |
14:00.37 | nain | ok |
14:00.51 | RoyK | you don't translate protocols |
14:00.56 | RoyK | you gateway between them |
14:02.23 | nain | that' it |
14:02.53 | nain | Well so Yate will perform well as a Gateway between h323 and SIP |
14:03.01 | RoyK | yes |
14:03.15 | RoyK | just as asterisk or aefirion may do with chan_h323, chan_oh323 |
14:04.11 | nain | i have seen lot of support for asterisk . What about Yate ? |
14:04.13 | RoyK | or aefirion's woomera/opal support |
14:04.21 | RoyK | nain: ask on #yate |
14:04.52 | RoyK | l-fy may be bitchy, but she knows a lot if you treat her well |
14:05.17 | nain | Where is she ? |
14:05.45 | RoyK | usually hanging out on #yate |
14:05.48 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
14:06.10 | nain | Thanks |
14:11.09 | *** join/#asterisk HaHaOok (~norman@60-240-147-63.tpgi.com.au) |
14:15.42 | WeezeyD | RoyK; I can't get h323 to run with CVS HEAD and I can't get oh323 0.7.1 to compile. :( |
14:16.18 | WeezeyD | I wouldn't care, but I can hook up to my buddy's phone system with oh323. |
14:16.25 | WeezeyD | or h323 |
14:17.05 | RoyK | don't ask my |
14:17.08 | RoyK | i only use iax2 and sip |
14:22.47 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
14:23.00 | *** join/#asterisk nain (nain@61.5.143.19) |
14:23.10 | nain | HI Every BOdy |
14:23.15 | WeezeyD | you again. |
14:23.37 | nain | uffffffffff you are also here |
14:24.01 | nain | Actually i didn't find someone in Yate channel so i thought to search out more on other channels |
14:26.39 | WeezeyD | and died. |
14:26.40 | *** join/#asterisk nain (nain@61.5.143.19) |
14:28.27 | wooden | ive installed ent extracted the zaptel-source and edited the Makefile how to install this no correct? |
14:28.37 | wooden | ups wrong channel |
14:32.58 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
14:56.04 | *** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se) |
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15:00.32 | `Sauron | Are there any good CDR post-processing scripts/things? |
15:03.20 | *** join/#asterisk coppice (~chatzilla@14.198.17.210.dyn.pacific.net.hk) |
15:03.40 | ersila | hey all. can anyone reference me to somewhere in the dial plan where i can add a 911 feature? i'm using sip. |
15:05.24 | *** join/#asterisk Kernel_core (Raph@22.230.dial-up.xter.net) |
15:07.11 | *** join/#asterisk Godsey (lanny@Godsey.sustaining.supporter.pdpc) |
15:07.11 | *** join/#asterisk jackx (none@dslam7-21-59-81.dyndsl.versatel.nl) |
15:07.46 | jackx | Does someone know what a standard gsm-gateway (1 sim) may cost? |
15:08.52 | *** join/#asterisk ArkyLady (ArkyLady@adsl-208-191-242-96.dsl.ltrkar.swbell.net) |
15:10.39 | eKo1 | ersila: eh, exten => 911,dial... |
15:11.11 | eKo1 | jackx: you mean for cellular? |
15:11.33 | jackx | yes |
15:11.43 | WeezeyD | is it possible to dial a pause to a PSTN somehow? |
15:11.50 | Godsey | exten => 911,1,Background(not-so-pro-gun-control-now-are-you) |
15:12.08 | tzanger | Godsey: hahaha |
15:12.15 | tzanger | that's not nice |
15:13.10 | Godsey | I'm a firm believer that people kill people, not guns |
15:13.17 | Godsey | and god helps those who help themselves ;) |
15:13.44 | *** join/#asterisk natox (pn@32-76-235-201.fibertel.com.ar) |
15:14.16 | Godsey | WeezeyD: it is, I'm trying to recall how |
15:15.33 | WeezeyD | Godsey: Within the phone number, you may use the special modifier w to indicate a half-second pause. |
15:15.36 | eKo1 | define 'dial a pause' |
15:15.41 | WeezeyD | Godsey: sound right? |
15:15.52 | natox | i've a problem when i work with my offline ip my cisco ata 186 is responding but when i try to assign one of my 3 ips isn't working how may i fix that? |
15:16.03 | Godsey | that I don't know |
15:16.11 | Godsey | eKo1: pause is standard feature |
15:16.24 | Godsey | WeezeyD: I did it a hackish way then |
15:16.30 | eKo1 | really? I've never heard of it. |
15:16.39 | Godsey | your phone has pause |
15:16.51 | WeezeyD | Godsey: really? what did you do? |
15:17.17 | eKo1 | It does? All I see is redial and recall buttons. |
15:17.17 | Godsey | I used an agi |
15:17.32 | pigpen | could someone post what that cheap fxo pci card is that asterisk is cool with (wildcard or something?) |
15:17.34 | Godsey | and had it just continue the dial plan |
15:17.52 | eKo1 | pigpen: x101p |
15:18.05 | pigpen | thanks...I have a bad memory. |
15:18.40 | Godsey | oh i see :) |
15:18.53 | Godsey | mine was a bit different |
15:18.59 | Godsey | I did it w/ a connection macro |
15:19.13 | Godsey | you put pause=yes in zapata.conf? |
15:19.16 | eKo1 | does 'pause' mean 'hold'? |
15:19.32 | Godsey | no, delay |
15:20.04 | eKo1 | aren't delays just added with exten => ...,wait(...) |
15:20.13 | WeezeyD | Godsey: no I didn't put pause=yes, but I can. |
15:20.30 | WeezeyD | eko1: not if you're dialling |
15:21.18 | eKo1 | Why would I want a delay when I'm dialing? |
15:21.27 | pigpen | eKo1, is the Astralis X101P clone ok? |
15:21.32 | WeezeyD | because you need time to scratch your nuts before the call connects |
15:21.49 | eKo1 | pigpen: no clue, i use the wildcards from digium. |
15:21.55 | pigpen | k |
15:22.04 | Godsey | exten => 911,1,Dial(Zap/911) |
15:22.10 | Godsey | but I'm not positive that's correct |
15:22.14 | Godsey | I don't use zap devices |
15:22.14 | tzanger | no that is not correct |
15:22.15 | *** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca) |
15:22.21 | eKo1 | Weezey: really? i do after the call connects. |
15:22.24 | tzanger | Zap/[channel_number]/911 |
15:22.37 | tzanger | and maybe a soft hangup before to make sure the line's free |
15:22.51 | jackx | [channel_number' |
15:22.57 | jackx | can't that be like a group of lines? |
15:23.00 | tzanger | yes |
15:23.02 | tzanger | if you have a group defined |
15:23.03 | jackx | so it figures out wich is free by itself? |
15:23.09 | tzanger | but still, you want to make sure *A* line is free |
15:23.12 | tzanger | or better |
15:23.13 | jackx | true |
15:23.22 | tzanger | dial, and then if all are busy, soft hangup, wait a second and goto(1) again |
15:23.24 | Godsey | i only said that because someone took my joke seriously about gun control |
15:23.41 | tzanger | you can make it more complex not to cut off an emergency call but IMO that's getting a little overboard |
15:23.44 | tzanger | but not impossible at all |
15:23.52 | tzanger | Godsey: who? |
15:24.16 | Godsey | anon unless they want to say :) |
15:24.31 | tzanger | ha |
15:24.38 | WeezeyD | how can I set EXTEN ? |
15:24.39 | tzanger | alison should do that phrase |
15:24.51 | WeezeyD | I do Setvar(EXTEN= but it seems to ignore me. |
15:24.55 | tzanger | WeezeyD: Goto(context,exten,1) |
15:25.22 | WeezeyD | cool |
15:25.23 | WeezeyD | thanks |
15:25.57 | Godsey | well |
15:26.01 | Godsey | there is an easy way |
15:26.07 | Godsey | pick channel 1 |
15:26.15 | Godsey | and kill any call on it :) |
15:26.17 | Godsey | then dial |
15:26.20 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
15:26.35 | Godsey | or drop all zap channels and dial 911 out the group |
15:26.48 | tzanger | no need to drop all |
15:26.52 | Godsey | unless they're talking to 911 ;P |
15:26.59 | ersila | tzanger: me :) |
15:27.01 | Godsey | depends on how fast people are w/ redial right? :) |
15:27.10 | tzanger | "a 911 call is already in progress, please look for the panicked person and inform them of the new emergency." |
15:28.04 | Godsey | there could be more than 1 person having a problem |
15:28.17 | tzanger | "please look for the panicked parties" then :-) |
15:28.33 | jackx | lol |
15:28.49 | Godsey | what is to happen to nufone and others now that fcc mandates they provide e911 service? |
15:29.12 | tzanger | I don't know about hte others but I'm fairly confident jerjer will have zero issue with that |
15:30.07 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
15:30.53 | Godsey | tzanger: 0 issue how? |
15:31.00 | Godsey | you mean easily provide e911? |
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15:31.24 | tzanger | Godsey: he terminates directly to TDM himself and it's not little PRIs here and there... so I don't think doing e911 is going to be a major issu efor him |
15:32.05 | Godsey | I'm not so sure |
15:32.20 | Godsey | dont you have to connect to every ilec in every serviced area? |
15:32.29 | Godsey | and we have ss7 connection also |
15:32.39 | Godsey | mainly for dial IN :) |
15:32.44 | tzanger | Godsey: I don't think so, you just need to route to the correct PSAP |
15:33.16 | tzanger | I don't think you need to be connected to every PSAP's tandem, just be able to get a call to them (even over standard links) |
15:33.36 | tzanger | I'm speaking right at the very fringe edge of my knowlege of "real" telco though so I may stop making sense at any moment |
15:33.45 | Godsey | humm |
15:33.57 | Godsey | I thought you had to have direct connection w/ each 911 center |
15:35.04 | tzanger | could be but at least for what I do I just call the emergency line of the PSAP (the standard # that 911 routes to) |
15:35.11 | tzanger | but I'm very much small fry :-) |
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15:37.07 | nain | Hi every body |
15:37.40 | Zeeek | eeeek un nain ! |
15:38.17 | nain | yes me nain |
15:38.28 | nain | well i have little problem |
15:38.56 | Zeeek | please state your case |
15:39.09 | Zeeek | an agent will be onlie to help soon |
15:39.27 | nain | Is it possible to recieve IP call in SIP without asterisk_oh323 channel driver |
15:39.28 | nain | ? |
15:40.06 | nain | Only in sip not h323 involvement |
15:40.15 | Zeeek | If you use SIP you probbly don't need h323 |
15:40.37 | nain | good but what if carrier is sending in h323 and my provider is SIP? |
15:41.01 | Zeeek | then you are forced to use 323 |
15:41.14 | nain | so i want that i recieve call from carrier in h323 and asterisk dial an sip account to my sip provider |
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15:42.58 | nain | Any solution Plz |
15:43.26 | Zeeek | maybe someone can help, I know nothing about h323 |
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15:43.34 | nain | thanks |
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15:48.10 | Sparkie | anyone ever setup a TDM04B before? |
15:48.35 | Sparkie | nifty. |
15:49.01 | SarahEmm | Sparkie: what's the issue? |
15:49.21 | Sparkie | well. |
15:49.26 | Sparkie | I got the card installed, got the module built |
15:49.34 | Sparkie | now I'm at setting up zapata.conf |
15:49.46 | SarahEmm | okay.. |
15:49.53 | Sparkie | and I'm not overly certain what to put in there. |
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15:50.17 | Sparkie | i'm still rereading the manual ;) |
15:50.26 | Zeeek | Sparkie there's a LOT of the wiki (if it's working) |
15:50.27 | Sparkie | yea |
15:50.27 | Sparkie | That's where I'm at right now |
15:50.28 | Zeeek | sample configs for zapata stuff |
15:50.40 | SarahEmm | Sparkie: ahh. 04b is 4fxs no fxo, or the other way? |
15:50.40 | Zeeek | zaptel has about 4 lines |
15:50.47 | Sparkie | but the only examples it lists (that i see) are for T-1/E-1 cards |
15:51.10 | Sparkie | SarahEmm: the other way. |
15:51.10 | Sparkie | my zaptel.conf is just 3 lines |
15:51.17 | Zeeek | well that sounds great |
15:51.23 | cluecon | ~seen bkw_ |
15:51.24 | jbot | bkw_ is currently on #asterisk (3h 44m 21s) |
15:51.24 | Sparkie | fxsks=1-4 |
15:51.24 | Sparkie | then my loadzone and default zone |
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15:51.32 | SarahEmm | sparkie: okay, 4fxo no fxs... |
15:51.38 | SarahEmm | Sparkie: okie, and ztcfg doesn't like it? |
15:51.56 | nain | HI |
15:52.38 | Zeeek | Sparkie there is something for you here: http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
15:52.38 | nain | is there any h323 expert ? |
15:52.38 | Zeeek | a few pages and you'll be there |
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15:54.03 | Sparkie- | ride the way |
15:54.03 | Sparkie- | wave |
15:54.03 | Sparkie- | lol |
15:54.16 | nain | any h323 expert ? |
15:54.53 | SarahEmm | Sparkie: what's ztcfg doing when you try using that config file? |
15:55.46 | Sparkie- | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
15:55.46 | Sparkie- | gives me 4 channels |
15:55.46 | Zeeek | Sparkie here's another one that will help (save 'em for later) http://automated.it/guidetoasterisk.htm |
15:55.46 | Sparkie- | 4 channels configured. |
15:55.53 | SarahEmm | sparkie: okay, what does zttool show after you do that? |
15:55.57 | SarahEmm | should show OK on the spans |
15:56.15 | Sparkie- | whoa |
15:56.15 | Sparkie- | nice |
15:56.26 | SarahEmm | mew? |
15:56.35 | Sparkie- | No alarms, Internally clocked, then it gives me 1234 |
15:56.42 | Sparkie- | and a series of transmit / receive lines |
15:56.43 | SarahEmm | ok, no alarms is good |
15:56.46 | SarahEmm | so what's the problem? :) |
15:56.49 | RoyK | splittetisplat |
15:57.00 | Sparkie- | I dunno |
15:57.13 | SarahEmm | Sparkie-: okay..... it looks like it's configured right to me |
15:57.19 | SarahEmm | Sparkie-: what problem are you having? |
15:57.20 | Sparkie- | well |
15:57.29 | Sparkie- | I'm not having a 'problem' really I know it's just something I haven't done |
15:57.35 | Sparkie- | to add the 4 lines into the trunk |
15:57.39 | Sparkie- | so that I can make outgoing calls |
15:57.51 | SarahEmm | ahh |
15:57.53 | SarahEmm | okay :) |
15:58.13 | SarahEmm | have you edited the zap cfg file in asterisk itself? |
15:58.14 | SarahEmm | that'd be the next step |
15:59.02 | Sparkie- | zapata.conf right? |
15:59.20 | SarahEmm | yeppers |
15:59.25 | ruski | what "give aways" should i be looking for im only hearing 1 way audio on a sip call ? |
15:59.40 | Zeeek | ruski NAT on either side ? |
15:59.56 | Zeeek | ruski and what phones? |
16:00.20 | Sparkie- | Ok, so if I have the 4 fxo cards on there, it's gonna start at g1 or g0 ? |
16:00.38 | Zeeek | g1 |
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16:01.17 | Sparkie- | wonder if that's been my problem all this time. |
16:01.34 | ruski | Zeeek: no nat |
16:02.02 | ruski | Zeeek: sjphone and a mobile phone, im using a sip provider and have setup asterisk to recieve incoming calls from them and route them to my sjphone |
16:02.36 | bkw_ | yo yo yo |
16:03.53 | cluecon | morning brian. |
16:04.17 | ariel_ | hello everyone |
16:04.41 | ariel_ | I guess some of us got disconnected by freenode... |
16:04.42 | SarahEmm | hihi |
16:05.05 | nain | HI every body |
16:05.27 | Sparkie- | show channels at the asterisk cli should only show channels that are currently in use right? |
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16:05.48 | nain | There is a tricky question regarding SIP |
16:06.00 | bkw_ | SIP tricky? NEVER!! |
16:06.04 | SarahEmm | lol |
16:06.06 | nain | yes |
16:06.24 | bkw_ | guess you didn't catch the sarcasim |
16:06.36 | nain | What if SIP client sent only his IP not username neither secret or no registration how to treat it ? |
16:06.57 | nain | where this will gone in sip.conf ? |
16:07.08 | nain | bkw_: is it not tricky? |
16:07.19 | bkw_ | no |
16:07.26 | bkw_ | it hits the context in [general] |
16:07.38 | bkw_ | all unauthenticed sip calls hit that context |
16:08.07 | nain | but my calls are not attaching to general context |
16:08.12 | nain | they are failing |
16:08.21 | bkw_ | they are if they don't auth |
16:08.35 | bkw_ | you can also match him by ip |
16:08.39 | bkw_ | [bob] |
16:08.42 | nain | but i don't need auth i just want call successful |
16:08.42 | bkw_ | type=user |
16:08.45 | bkw_ | host=hisip |
16:08.54 | bkw_ | context=bob |
16:08.58 | nain | i have done host=IP and it's context but :( |
16:09.00 | bkw_ | do what I said |
16:09.09 | bkw_ | you can't have a secret or username |
16:09.18 | bkw_ | switch it to type=peer |
16:09.44 | nain | type=peer ??? i have set it to type=friend bcz i also want to dial it to sip provider |
16:09.57 | bkw_ | again |
16:10.00 | bkw_ | LISTEN TO ME |
16:10.07 | nain | OK |
16:10.22 | bkw_ | its evil |
16:10.23 | bkw_ | but I understand it |
16:10.31 | bkw_ | if you look at the code you'll understand it too |
16:10.38 | nain | ok i m listening |
16:10.46 | bkw_ | you can't have ANY auth data on the entry you want it to match |
16:10.53 | bkw_ | otherwise it will fail |
16:11.08 | bkw_ | so you will have to have a user entry with NO username or secret |
16:11.10 | bkw_ | just for inbound |
16:11.12 | bkw_ | or |
16:11.17 | bkw_ | make them match the general context |
16:11.20 | bkw_ | which is what it will do |
16:11.44 | bkw_ | unless you have a user and/or peer without auth data |
16:11.46 | Kernel_core | hi bkw_ |
16:11.47 | bkw_ | and host=ip |
16:11.59 | bkw_ | yo Kernel_core |
16:12.34 | bkw_ | go look at find_user and find_peer in chan_sip.c |
16:12.40 | bkw_ | then look how they are used |
16:12.49 | bkw_ | that will spell it all out clear as day if you can read and understand the code |
16:13.27 | nain | OK |
16:13.30 | bkw_ | if I were on the box.. watching calls come in and looking at a sip debug |
16:13.33 | bkw_ | I could fix it in two seconds |
16:13.46 | nain | sure i can give u access if u can |
16:14.08 | bkw_ | but you'll never learn if I do it for you |
16:14.12 | bkw_ | give it a few more tries |
16:14.20 | bkw_ | if you don't get it then i'll look at it |
16:14.40 | Kernel_core | bkw_: is CVS-Nv1-0-7-06/11/05-17:43:44 the lastest version of Asterisk ?! |
16:14.51 | bkw_ | LIAR |
16:14.52 | bkw_ | no |
16:15.02 | bkw_ | becuase that no longer had the version string in it |
16:15.05 | nain | i will do it ok tell me one more thing that the context name will be ignore if host=ip define ? |
16:15.24 | bkw_ | you need need context= also |
16:15.41 | nain | OK |
16:15.41 | bkw_ | Asterisk CVS-HEAD built by root@shinzon on a i686 running Linux on 2005-06-11 15:58:51 |
16:15.47 | bkw_ | thats what the version string looks like now as of late |
16:15.55 | bkw_ | you also have a new cli command 'show version files' |
16:16.09 | bkw_ | that will show you the revision of all files |
16:17.20 | Kernel_core | I use cvsup what should I put instead of " *default release=cvs tag=." to get the latestet version of Asterisk ?! ( CVS ) |
16:17.38 | bkw_ | your running 1.0.7 |
16:17.45 | bkw_ | that is by no means the latest CVS-HEAD |
16:17.50 | bkw_ | stable is a pile of shit |
16:17.54 | bkw_ | you call from stable to head |
16:17.57 | bkw_ | it will sound like total ass |
16:18.19 | nain | bkw_: see it is right ? |
16:18.23 | nain | [general] |
16:18.24 | nain | context=global-calls |
16:18.24 | nain | port=5060 |
16:18.24 | nain | bindaddr=0.0.0.0 |
16:18.24 | nain | [no-auth] |
16:18.24 | nain | host=207.113.96.114 |
16:18.26 | nain | context=global-calls |
16:18.28 | nain | type=peer |
16:18.30 | nain | nat=yes |
16:18.32 | nain | canreinvite=no |
16:18.34 | nain | insecure=very |
16:18.36 | nain | qualify=yes |
16:18.38 | nain | disallow=all |
16:18.40 | nain | allow=ulaw |
16:18.42 | nain | allow=g729 |
16:18.43 | bkw_ | turn auqlify off |
16:18.44 | jontow | dude.. pastebin.ca |
16:18.46 | SarahEmm | gah |
16:18.46 | jontow | :) |
16:18.55 | bkw_ | and don't paste in here again or i'll slap you |
16:18.57 | file | bkw_: I want my Powerbook :( |
16:18.57 | bkw_ | :P |
16:19.00 | nain | OK |
16:19.00 | bkw_ | and I mean that with love |
16:19.12 | bkw_ | file what is this you msged me about |
16:19.13 | nain | every thing is OK instead qualify=yes ? |
16:19.13 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
16:19.15 | bkw_ | me no understand |
16:19.25 | file | bkw_: and I never messaged you anything |
16:19.32 | bkw_ | yes you did |
16:19.34 | bkw_ | on my phone |
16:19.34 | file | atleast not that I can remember |
16:19.39 | file | oh that was AGES ago |
16:19.43 | bkw_ | haha |
16:19.45 | bkw_ | I got it today |
16:19.45 | file | when DNS went kaboom |
16:19.48 | nain | OK |
16:19.57 | bkw_ | funny |
16:21.18 | bkw_ | ouch |
16:21.35 | pigpen | ok..I just ordered some of the x100p's for cheap fxo's...can one of these ports be an fxs ? if not, any cheap fxs's out there? |
16:21.47 | bkw_ | pigpen, no and no |
16:21.50 | bkw_ | NEXT!!! |
16:22.23 | *** join/#asterisk pjz (~pj@place.org) [NETSPLIT VICTIM] |
16:22.25 | pigpen | Ok..so just get an IAXy or someting... |
16:22.40 | cluecon | bkw_: you do plan to run around cluecon screaming NEXT!!! every 5 minutes right? |
16:22.44 | Godsey | if you're doing that, why not get SPA-3000? |
16:22.46 | Godsey | :P |
16:22.58 | bkw_ | cluecon, na |
16:23.12 | pigpen | I have a 3000...it is kinda clunky since I have an asterisk box at location. |
16:23.13 | bkw_ | i'm not a bitch in real life.. I just play one on IRC |
16:23.15 | Godsey | AND THEN? |
16:24.10 | *** join/#asterisk netofsickcoder (~netofsick@200.121.129.178) |
16:24.43 | *** part/#asterisk mosty (mostynm@adsl-137-244.swiftdsl.com.au) |
16:25.28 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
16:25.48 | cluecon | the least you could do is run around screaming next between speakers. |
16:26.00 | cluecon | it would liven it up a bit. |
16:26.52 | pigpen | anyone know how the IAXy would do with fax? |
16:28.09 | *** join/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
16:29.01 | *** join/#asterisk nitram (foo@superblob.com) |
16:29.17 | *** join/#asterisk wooden (ppvray@p54AA03DD.dip0.t-ipconnect.de) |
16:29.40 | wooden | <PROTECTED> |
16:29.46 | wooden | knows anyone about this problem? |
16:29.58 | *** join/#asterisk dsvanlund (~david@ua-83-227-224-57.cust.bredbandsbolaget.se) |
16:32.54 | eKo1 | Did you change from stable to head or vice versa? |
16:35.42 | WeezeyD | how can I block my number in xlite? |
16:40.18 | bkw_ | HAHAH I pissed off the neighbors |
16:40.22 | bkw_ | they are mad as hell |
16:40.40 | bkw_ | http://homepage.mac.com/brian.west/PhotoAlbum13.html <-- this is them... I called da cops |
16:41.49 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4075317.sympatico.ca) |
16:42.21 | outtolunc | .. /. |
16:42.41 | outtolunc | oops |
16:42.45 | outtolunc | cute |
16:42.58 | outtolunc | did you also take the wheels and tires first <G> |
16:44.13 | bkw_ | haha |
16:44.23 | bkw_ | they did that |
16:44.28 | bkw_ | get |
16:44.29 | bkw_ | toe |
16:44.32 | WeezeyD | they did that to who? |
16:44.39 | bkw_ | thats their own car |
16:44.41 | bkw_ | they did that |
16:44.49 | bkw_ | guess they needed to sell the wheels to make some money |
16:44.50 | WeezeyD | and they're trying to claim? |
16:44.52 | bkw_ | I wish they would just move |
16:44.59 | bkw_ | no they are just white trash |
16:45.02 | WeezeyD | heh |
16:45.11 | bkw_ | I wish they would LEAVE |
16:45.15 | MikeJ_ | so they sold used tires for cash? |
16:45.17 | bkw_ | they are loud |
16:45.20 | WeezeyD | they needed beer money |
16:45.23 | bkw_ | no clue |
16:45.27 | MikeJ_ | you think at that point they would see the car |
16:45.28 | bkw_ | but the care is not in the road today |
16:45.29 | MikeJ_ | boom |
16:45.31 | WeezeyD | my new neighbours better not be loud. |
16:45.43 | file | bkw_ is uber-sexy in real life |
16:45.47 | bkw_ | they are the only trouble people in the neighbor hood |
16:46.07 | WeezeyD | file: and his middle name is Kelvin |
16:46.19 | file | haha |
16:46.21 | bkw_ | well they drive property value down |
16:47.07 | MikeJ_ | awww |
16:47.12 | MikeJ_ | it's a slow dance |
16:47.19 | MikeJ_ | isn't that sooo cute |
16:47.38 | MikeJ_ | :p |
16:48.10 | MikeJ_ | is it hot as hell down there too? |
16:48.11 | file | :P |
16:48.38 | MikeJ_ | uhhhh |
16:48.45 | bkw_ | hahaha |
16:48.54 | bkw_ | i've been thinkin bout you.... |
16:49.02 | file | thinking about me eh? |
16:49.05 | MikeJ_ | it's my high school reunion this weekend |
16:49.07 | MikeJ_ | hehe |
16:49.16 | bkw_ | i'll never get to go to one of those |
16:49.16 | MikeJ_ | scary shit |
16:49.21 | bkw_ | since I didn't actually graduate |
16:49.24 | MikeJ_ | I am.. it's kinda funny |
16:49.27 | MikeJ_ | soo |
16:49.30 | MikeJ_ | who cares |
16:49.38 | file | MikeJ_: you're old |
16:49.43 | bkw_ | I don't care to see the people that made my teen years a living hell |
16:49.45 | file | I care about my Powerbook! |
16:49.56 | MikeJ_ | I barely did.. I was getting an a in one of the required classes, and I never did this one paper.. |
16:50.02 | MikeJ_ | ethics i think it was |
16:50.08 | bkw_ | I quit and took my GED |
16:50.29 | bkw_ | MikeJ_, neither did alot of folks it would seem |
16:50.35 | MikeJ_ | and my prof said a week before graduation that he would give me an incomplete if I didn;t write it and I would not graduate |
16:50.59 | MikeJ_ | my girl finished at an alternative school |
16:51.26 | bkw_ | sad part is now I have to get a security camera for the house |
16:51.34 | MikeJ_ | the people attarcted to this kind of project are probably all non traditional learners |
16:51.40 | bkw_ | yep |
16:51.48 | MikeJ_ | bkw_, that's not sad.. that's an excuse to buy toys |
16:51.56 | Sparkie- | Hrm. I keep getting a 503 error from my softphone, what have I done wrong now? :) |
16:52.04 | bkw_ | MikeJ_, hahah |
16:52.12 | MikeJ_ | 996 |
16:52.14 | bkw_ | well I fear those pricks will fuck with our shit |
16:52.25 | MikeJ_ | you know what you need |
16:54.20 | drumkilla | MikeJ_: you've really got to stop using outlook, haha |
16:54.25 | drumkilla | you're killing my threading :p |
16:57.37 | MikeJ_ | sorry |
16:57.48 | drumkilla | it's all good, just pickin' on you |
16:57.49 | MikeJ_ | tis my work e-mail... |
16:57.50 | opus__ | yo |
16:58.14 | MikeJ_ | and the house is running a test exchange server for the office too... |
16:58.18 | MikeJ_ | so I have what I have |
16:58.24 | ariel_ | Sparkie-, 503 means service is not available. |
16:58.38 | drumkilla | evolution can talk to exchange ;( |
16:58.40 | drumkilla | er, ;) |
16:58.57 | Sparkie- | yea I fixed that |
16:59.03 | MikeJ_ | can't do the integration with sharepoint |
16:59.13 | Sparkie- | it's not letting me pickup the outside line, just keeps telling me all circuits are busy. |
16:59.36 | ariel_ | Sparkie-, are you using amp? |
16:59.45 | MikeJ_ | Sparkie-, are all the circuits busy :D |
16:59.49 | drumkilla | MikeJ_: ah. well... I don't really know what I'm talking about. When I worked at an office that used exchange, i just used outlook. :) |
16:59.50 | Sparkie- | ariel_: yes. |
17:00.01 | MikeJ_ | :p |
17:00.12 | MikeJ_ | wassup... anything happy going on today |
17:00.14 | Sparkie- | They shouldn't be busy. Granted, I only have 1 line plugged into the fxo, but no one is in the office so the phone shouldn't be on |
17:00.22 | ariel_ | Sparkie-, make sure your trunks are registered. |
17:00.42 | ariel_ | Sparkie-, do you have your dialing rules set correctly? |
17:01.10 | Sparkie- | I have 1 trunk setup, ZAP/g1 with 4 maximum channels, dial rules are 1NXXNXXXXXX, NXXNXXXXXX, and NXXXXXX |
17:01.13 | MikeJ_ | ariel_, if he has no trunks set up, then all 0 of his circuits would be busy ;) |
17:01.18 | Sparkie- | outbound dial prefix is 8 |
17:01.19 | Sparkie- | err 9 |
17:01.27 | Sparkie- | and the zap identifier is g1 |
17:01.34 | ariel_ | also there is an area here just for amp #amportal |
17:01.47 | ariel_ | Sparkie-, you have 4 fxo's? |
17:02.12 | ariel_ | The default for amp is Zap/g0 |
17:02.17 | Sparkie- | yes |
17:02.19 | Sparkie- | I have 4 fxo's |
17:02.25 | Sparkie- | yea ... I deleted g0 |
17:02.25 | Sparkie- | lol |
17:02.32 | Sparkie- | woops |
17:04.40 | *** join/#asterisk trasherr (trasher@dsl-084-058-015-087.arcor-ip.net) |
17:04.44 | trasherr | hi |
17:04.48 | ariel_ | Oh boy it's mango season down here again. Hummm good |
17:05.45 | *** join/#asterisk bofh42 (~bofh42@p548231DA.dip0.t-ipconnect.de) |
17:06.27 | nain | bkw_: are you there ? |
17:07.03 | trasherr | i am using asterisk with sip + capi, forwarding incoming isdn calls via sip to my workstation, and outgoing sip calls through isdn... but: the called-person can't see my ISDN number. how to i configure that? |
17:08.14 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
17:08.40 | ariel_ | trasherr, are you setting up your callerID when your dialing out? |
17:09.50 | trasherr | if you mean the "callerid=..." in sip.conf -> yes i do |
17:10.35 | Sparkie- | from the asterisk cli, should show channels show me something to let me know the channels are configured properly? |
17:11.18 | ariel_ | Sparkie-, show channels will only show you channels in use. |
17:11.48 | Kernel_core | No description for sounds/digits/minus.gsm |
17:11.48 | Kernel_core | make: *** [datafiles] Error 1 |
17:11.57 | Kernel_core | during compile I got this ! what is wrong ?! |
17:11.57 | Sparkie- | hrm. |
17:12.02 | *** part/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca) |
17:12.05 | Sparkie- | welp. i'm lost lol |
17:13.25 | ariel_ | Sparkie-, ztcfg -vvvv will show you what channels are configured. from the linux prompt |
17:13.38 | Sparkie- | yea, it shows me the 4 channels. |
17:15.33 | ariel_ | trasherr, you can set your caller ID in your dial command look up SetCallerID on the wiki for your outbound dialing rules |
17:15.43 | trasherr | ariel_: oks, thank you |
17:16.33 | ariel_ | Sparkie-, ok how did you configure them in amp which file did you put the settings in? |
17:17.06 | Sparkie- | the trunks? |
17:17.13 | *** join/#asterisk Hmmhesays (~Neg@24-117-130-121.cpe.cableone.net) |
17:18.02 | ariel_ | Sparkie-, well in which files yes are you using amp 1.10.08? |
17:19.12 | Sparkie- | I'm using the AMP that came with asterisk at home |
17:20.01 | ariel_ | Sparkie-, I see. ok did you do genzaptelconf -s -d at the linux command prompt to configure your zap devices? |
17:20.22 | Sparkie- | no |
17:20.27 | Sparkie- | genzaptelconf errors out |
17:20.30 | Sparkie- | I had to manually set it up |
17:21.01 | Sparkie- | the zaptel modules that came with asterisk at home caused problems under linux 2.6, so I manually built the modules and installed them |
17:21.53 | ariel_ | asterisk at home is preset for 3.4 CentOS 2.6 you need to follow the udev setup. did you? |
17:22.17 | ariel_ | which is linux 2.4 version |
17:22.32 | Sparkie- | yes I did. |
17:22.41 | Sparkie- | the stuff in /dev/zap is setup properly. |
17:22.54 | ariel_ | can you post your zapata.conf on pastebin.ca for us to see? |
17:23.02 | Sparkie- | yea .... |
17:23.03 | Hmmhesays | is *@home stable enough for a small production setup? |
17:23.04 | Sparkie- | uhm |
17:23.06 | Sparkie- | there's nothing there |
17:23.14 | ariel_ | Hmmhesays, yes |
17:23.26 | Sparkie- | That's probably part of my problem. |
17:23.32 | Hmmhesays | for someone who knows what they are doing |
17:23.41 | ariel_ | Hmmhesays, yes |
17:23.43 | Sparkie- | oh woops i was in zapata-auto |
17:24.33 | ariel_ | Hmmhesays, they now also include a great CRM called SugarCRM. |
17:24.41 | Sparkie- | http://pastebin.ca/13969 |
17:24.59 | Hmmhesays | oh really?, i've not watched the dev very closely |
17:25.38 | ariel_ | Sparkie-, ok what does the additional and the auto have? |
17:25.42 | Sparkie- | nothing |
17:25.46 | ariel_ | your default group is 0 |
17:26.02 | ariel_ | ahh then it's not going to work then. |
17:26.05 | Sparkie- | yea. I reset everything back to g0 |
17:26.59 | ariel_ | you need something to say channel=1-4 if you want to use them |
17:27.25 | Sparkie- | ok |
17:28.30 | Kernel_core | maye so stupid question , but how do I get the yesterday asterisk Snapshot ?! |
17:31.40 | MikeJ_ | Kernel_core, head or stable |
17:32.05 | *** join/#asterisk Total-Net (~pbudden@user-1121i7q.dsl.mindspring.com) |
17:32.12 | MikeJ_ | cvs co -D .... |
17:32.41 | MikeJ_ | followed by the date of vode you want |
17:32.43 | MikeJ_ | code |
17:33.03 | MikeJ_ | if you want a specific date of stable you would add -r v1-0 |
17:33.45 | *** join/#asterisk iq (~iq@65-103-165-110.omah.qwest.net) |
17:33.55 | iq | hi all |
17:34.38 | MikeJ_ | high |
17:34.53 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
17:35.13 | wooden | [app_capiCD.so]Jun 11 18:29:32 WARNING[10529]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/app_capiCD.so: undefined symbol: capidebug |
17:35.14 | *** join/#asterisk Sparkie- (sparkie@24-50-118-84.lkwnny.adelphia.net) |
17:35.27 | wooden | knows anyone whats wrong? |
17:36.04 | Hmmhesays | say hi |
17:38.10 | Kernel_core | MikeJ_: HEAD , not stable |
17:40.25 | *** join/#asterisk brookshire (~matt@207.111.174.1) |
17:41.29 | drumkilla | i think there are nightly snapshots on the ftp site, as well ... |
17:43.34 | Kernel_core | where do I get this nightly snapshot ? ftp.asterisk.org ?! or digium !? |
17:44.27 | file | a Matt has been detected |
17:46.32 | Hmmhesays | a Matt? |
17:47.05 | file | he is a Matt |
17:47.15 | Hmmhesays | heh, me too |
17:48.44 | *** join/#asterisk dynaqueue (~wbevan@adsl-63-205-16-227.dsl.mtry01.pacbell.net) |
17:49.13 | dynaqueue | Anyone else running the latest firmware on a grandstream xps 2000 ? |
17:51.48 | trasherr | ariel_: i tried it with setting the callerid before dialing the number.. but it doesn't work :| google can't help me |
17:54.14 | *** join/#asterisk faqu (pn@32-76-235-201.fibertel.com.ar) |
17:59.16 | Hmmhesays | yeah ariel_ sugarcrm looks pretty sweet |
18:00.04 | Hmmhesays | actually I was looking for something very similar |
18:03.56 | *** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
18:04.00 | cluecon | is iaxtel down? |
18:05.37 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
18:08.00 | Hmmhesays | i do not know |
18:08.06 | brookshire | hi! |
18:08.46 | Hmmhesays | hello person of the same name |
18:08.53 | brookshire | are you a matt? |
18:09.04 | cluecon | Somebody got an iaxtel number i can call to test? |
18:09.11 | brookshire | cluecon: iaxtel is always down |
18:09.12 | wooden | where to get a good example for capi.conf and extensions.conf for dialin and dialout |
18:09.16 | *** join/#asterisk meppl (mephisto@p54AABBEC.dip.t-dialin.net) |
18:09.22 | wooden | isdn |
18:09.22 | brookshire | russell has been working on it, trying to make it stable |
18:09.32 | brookshire | :( |
18:10.17 | *** join/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net) |
18:11.39 | *** join/#asterisk ManxPower (~eric@n34.infopact.nl) |
18:13.52 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
18:16.09 | *** part/#asterisk Matt-E- (~Matt-E-@66-224-125-137.atgi.net) |
18:17.06 | shido | boink |
18:17.16 | Hmmhesays | hey jerk |
18:17.23 | Hmmhesays | whats up ;) |
18:19.02 | shido | hey num nuts |
18:19.04 | shido | :) |
18:19.10 | Hmmhesays | haha |
18:19.20 | Hmmhesays | yeah I gotta quite putting the icy hot on them |
18:19.26 | shido | LOL |
18:19.36 | *** join/#asterisk Xumpi (~SysOp@a81-84-69-130.netcabo.pt) |
18:19.57 | Hmmhesays | you know anyone using sugarcrm? |
18:20.06 | shido | whats that? |
18:20.07 | Hmmhesays | I was tinkering around with the demo today |
18:20.11 | Xumpi | hi |
18:20.23 | Xumpi | any kiax-alike for freeBSD ? |
18:20.30 | Hmmhesays | http://demo.sugarondemand.com/sugarcrm_os/index.php?action=Login&module=Users |
18:20.32 | Xumpi | any kiax-alike-program for freeBSD ? |
18:21.09 | Hmmhesays | freeBSD is definately not my strong suite |
18:21.37 | Xumpi | OpenBSD also's available here |
18:23.00 | shido | wow |
18:23.09 | *** join/#asterisk ManxPower (~eric@n34.infopact.nl) |
18:23.31 | Hmmhesays | my employers crm software blows |
18:23.38 | shido | this is cool |
18:23.51 | Hmmhesays | yeah ariel pointed it out to me earlier |
18:24.17 | Hmmhesays | it's integrated into the latest release of asterisk@home too |
18:25.19 | Hmmhesays | I find that a little bit odd but, whatever |
18:25.26 | Xumpi | anyone's using voipbuster.com ? |
18:28.50 | *** join/#asterisk cinix (~The@24-52-166-190.lndnnh.adelphia.net) |
18:29.16 | trasherr | Xumpi: hmm sounds intresting |
18:29.27 | cinix | What's a name of an IAX2 linux client? |
18:30.00 | Xumpi | got it |
18:30.01 | Xumpi | http://david.dw-perspective.org.uk/VoipBuster-Linux-MacOS-Solaris-FreeBSD.html |
18:30.02 | Xumpi | brb |
18:30.11 | Xumpi | see you guys l8er's |
18:30.45 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
18:31.35 | *** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net) |
18:33.27 | *** join/#asterisk RoyK (~roy@host-n39-135.homerun.telia.com) |
18:34.33 | *** join/#asterisk wacky_ (~wacky@modemcable011.4-37-24.mc.videotron.ca) |
18:34.39 | wacky_ | hey boys and girls :) |
18:35.31 | wacky_ | I'm having a hard time configuring Asterisk with my SIP provider. When the provider sends in a call (from the PSTN network), asterisk reply with a: 407 Proxy Authentication Required, thus dumping the incoming call |
18:35.42 | wacky_ | I'd like to have a little insight about how I could work around this |
18:35.49 | *** join/#asterisk waz (~tjs@208.218.27.24.cfl.res.rr.com) |
18:36.03 | Romik_ | hmmmmmmmmmm load average: 179.91, 163.94, 136.86 |
18:36.16 | Hmmhesays | wacky_ |
18:36.17 | cinix | out of... |
18:36.19 | cinix | hehe wow |
18:36.21 | Hmmhesays | I could tell you the answer |
18:36.28 | waz | yikes |
18:36.31 | Xumpi | DoS prolly |
18:36.43 | Kernel_core | wacky_: I have the same problem with my SIP phone to asterisk |
18:36.43 | Xumpi | some kind of fork(); bomb prolly |
18:36.55 | waz | or hitting something in inetd |
18:37.06 | file | who wants a lesson in SIP authentication? |
18:37.14 | file | if so, everyone else shutup and tell me when I should start speaking |
18:37.15 | Romik_ | CPU Usage: u12974.4 s297.61 cu.04 cs.02 - 62.9% CPU load |
18:37.16 | Romik_ | 2.58 requests/sec - 37.8 kB/second - 14.7 kB/request |
18:37.16 | Romik_ | 135 requests currently being processed, 90 idle servers |
18:37.23 | wacky_ | I read it may be caused by the 'secret' field being present, or 'auth' or 'md5secret' |
18:37.37 | Hmmhesays | insecure=very |
18:37.44 | file | Hmmhesays: quiet - I wanna explain why/etc |
18:37.46 | shido | does your provider tell you to use a secure password? |
18:37.47 | waz | anyone have asterisk working with AT&T CallVantage? |
18:37.52 | file | oh nevermind |
18:38.15 | Hmmhesays | sorry file |
18:38.18 | wacky_ | file - go ahead, come on :)) |
18:38.47 | file | If you're using a user entry, then asterisk will challenge for credentials if it finds that the incoming INVITE is going to match against a user... that's standard SIP |
18:39.02 | file | normally the other side will respond with the same invite, except a Proxy-Authorization header is there |
18:39.07 | file | (this is the case with phones) |
18:39.30 | file | now, depending on how you set it up - your asterisk will do the same thing with a provider, and not all (okay... very few) providers support doing this |
18:39.38 | file | instead you have to match based on the IP address of the INVITE |
18:39.46 | file | this is done using a peer entry with host, context, and insecure set |
18:39.53 | file | insecure=very liek Hmmhesays said |
18:40.10 | file | this will cause asterisk to not doing any kind of authentication but simply trust the INVITE based on the source IP address |
18:40.40 | file | mmmkthx |
18:40.45 | Hmmhesays | good job file |
18:40.57 | citats | ughh stupid gcc3isms |
18:40.57 | file | SIP happens. |
18:41.02 | Hmmhesays | :D |
18:41.34 | *** join/#asterisk Pazzo (~Pazzo@host130-250.pool8172.interbusiness.it) |
18:41.36 | Hmmhesays | I figured an explanation would have fallen on deaf ears... I was mistaken |
18:42.04 | file | giving an answer is one thing, but explaining the answer and why is another :) I'd rather you folks learn the why, then just 'here is how to do it' |
18:42.33 | Hmmhesays | I forget people actually listen in here most of the time though |
18:42.46 | file | it's rare |
18:42.56 | Hmmhesays | as opposed to paid support customers |
18:43.01 | wacky_ | hey good, thanks file.. |
18:44.13 | Kernel_core | file: what should I do to authenticate My Eyebeam SIP Client to get authenticated with asterisk ?! |
18:44.16 | Pete_Largo | thanks file |
18:44.25 | file | Kernel_core: set it up right? because you obviously didn't |
18:44.52 | file | as I've got Eyebeam working fine and dandy |
18:44.53 | Kernel_core | file: with this config , X-lite can connect , so it seems that my conf is OKey |
18:45.13 | *** join/#asterisk cougar101 (Molad@adsl-69-224-94-166.dsl.irvnca.pacbell.net) |
18:45.48 | wacky_ | file - only now.. my def contains type=peer, insecure=very, a host= field and a context= field |
18:45.53 | wacky_ | and I still get a 407.. |
18:46.00 | wacky_ | now maybe I could check the IP adresses ? |
18:46.20 | file | wacky_: do a sip reload, and yes - check the IP address |
18:46.26 | Kernel_core | file: what should I specify to get Eyebeam work ? |
18:46.31 | Kernel_core | file: what should I specify to make Eyebeam work ? |
18:46.36 | file | Kernel_core: there's nothing special in asterisk you have to specify |
18:46.37 | Hmmhesays | if you are using a host name to register, make sure it matches the incoming IP |
18:46.45 | Kernel_core | file: so ? |
18:46.47 | wacky_ | I reloaded, and restarted a few times, waiting for the registration to expire. |
18:46.53 | wacky_ | I'll check IP addresses |
18:47.04 | file | Kernel_core: you probably have something configured wrong |
18:48.39 | Kernel_core | file: http://pastebin.com/299002 here is my conf file |
18:48.50 | Kernel_core | for X-Lite |
18:49.00 | file | okay some of this is just unneeded |
18:49.00 | Kernel_core | so what could be misconfigured ? |
18:49.07 | Kernel_core | example ? |
18:49.12 | wacky_ | file - which field of the request must match the host= def ? |
18:49.38 | file | wacky_: the origin IP address |
18:49.45 | file | Kernel_core: http://pastebin.com/299004 use that |
18:49.57 | wacky_ | oh, okay |
18:51.12 | wacky_ | file - I'll explain something, see my provider requires the use of an outbound proxy.. so someone found a trick to bypass that.. by placing a def. in the /etc/hosts file and fake the whole thing |
18:51.17 | Hmmhesays | it's odd he's get a proxy authentication required if the origin ip didn't match is registration ip |
18:51.26 | Hmmhesays | because the call would just go into the default context |
18:51.35 | wacky_ | Has support been added in latest releases ? |
18:51.40 | *** join/#asterisk ska (~ska@68-234-182-91.snbrca.adelphia.net) |
18:51.58 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
18:52.04 | file | Hmmhesays: almost makes me think a user entry is mucking with it |
18:52.12 | file | wacky_: no |
18:52.14 | shido | grrr |
18:52.38 | Hmmhesays | indeed |
18:52.42 | ska | is it possible with some services to have more incoming calls that you have t1 channels for? or do you need a real voip solution to do that? |
18:52.48 | Hmmhesays | pastebin your sip.conf |
18:53.06 | Hmmhesays | down to where you define your proxy |
18:53.08 | Kernel_core | file: got the same result ! |
18:54.15 | dsvanlund | I am unable to call dundi_lookup from my own application module. It compiles without any problem. However when I load my module I get "undefined symbol: dundi_lookup". What could I be doing wrong? dundi.h is included, and pbx_dundi.so loaded (if that matters). |
18:54.20 | file | Kernel_core: and your eyebeam is setup right? |
18:54.37 | Kernel_core | Display name : blank |
18:54.41 | file | dsvanlund: you can't, dundi is an isolated module |
18:54.42 | Kernel_core | username :54321 |
18:54.48 | Kernel_core | pass: 4511793 |
18:55.01 | Kernel_core | and ip address of domain |
18:55.05 | file | ska: uhhhhh no |
18:55.06 | Kernel_core | is it enough? |
18:55.21 | file | Kernel_core: fill out the rest, realm and sip proxy with your address for asterisk |
18:55.34 | Kernel_core | doiman proxy: register with domain and use as outbound proxy |
18:55.53 | Kernel_core | I don't see any realm |
18:55.56 | trasherr | i am using asterisk with sip + capi, forwarding incoming isdn calls via sip to my workstation, and outgoing sip calls through isdn... but: the called-person can't see my ISDN number. how to i configure that? i tried it with SetCallerID before dialing, but it does not work.. |
18:56.02 | Kernel_core | I am useing the latest version ! |
18:56.17 | file | Kernel_core: you're going to make me set it up aren't you? |
18:56.21 | file | and it's going to work |
18:56.41 | Kernel_core | if kindly you come :) |
18:57.08 | *** join/#asterisk PakiPenguin (~pingu@202.147.163.11) |
18:57.24 | file | worked fine |
18:57.26 | file | :P |
18:57.46 | file | Display name, User name, Password, Authorization user name, and Domain |
18:57.52 | file | that's what I filled out |
18:58.01 | file | and checked "Register with domain" and "Use as Outbound Proxy" |
18:58.17 | Kernel_core | I get this message in my debug : Ignoring too old SIP packet packet 1 (expecting >= 2) |
18:58.17 | Godsey | I wonder how many drug kingpins use asterisk w/ ssh tunnels for toll bypass and DEA subversion U.S.<->Columbia |
18:58.25 | Kernel_core | what's wrong ?! |
18:58.26 | Godsey | that is a $$ business I bet |
18:58.47 | dsvanlund | file: OK... I believed I could use dundi_lookup if I included dundi.h, where it is declared. |
18:58.52 | file | dsvanlund: no. |
18:59.10 | infi | asteriskDOTbz: *THE* Choice of Money Launderers Everywhere. |
18:59.14 | infi | er |
18:59.17 | file | Kernel_core: make sure what I told you to have set (see above) is set, close Eyebeam, and then open it |
18:59.23 | infi | stupid nick completion |
18:59.35 | *** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
18:59.44 | *** part/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
18:59.51 | *** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
19:00.05 | Kernel_core | file: got the same result |
19:00.19 | Kernel_core | useing the lateset version of Eyebeam ! |
19:00.21 | Godsey | narcotel :) |
19:00.26 | file | well I just set it up, and it works fine |
19:00.52 | Kernel_core | file: I have near 700ms latency |
19:00.59 | Kernel_core | does it affect to my SIP ? |
19:01.02 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
19:01.03 | Kernel_core | I get this message in my debug : Ignoring too old SIP packet packet 1 (expecting >= 2) |
19:01.22 | Kernel_core | maybe some SIP packets reject |
19:01.33 | Godsey | 700? |
19:01.39 | *** join/#asterisk at561 (~oiasdf@68.71.213-254.atlsfl.adelphia.net) |
19:01.42 | Godsey | directway? :) |
19:01.49 | Kernel_core | : |
19:01.50 | L|NUX | evening |
19:01.51 | Kernel_core | :| |
19:02.34 | at561 | is there a way to get two asterisk servers registered as peers to a third, with one operating as primary and the other as backup? |
19:02.39 | cluecon | afternoon. |
19:02.41 | Godsey | Kernel_core: I'm not positive but i think asterisk defaults to 300ms qualify= |
19:03.03 | Kernel_core | Godsey: how do I change this value ?! |
19:03.15 | Kernel_core | and where do I ? |
19:03.19 | Godsey | wait for someone smart to confirm it |
19:03.21 | Godsey | :) |
19:03.26 | wacky_ | does asterisk consider the /etc/hosts file while doing name resolutions ? |
19:03.29 | Godsey | I'm not so good w/ ASterisk yet |
19:03.29 | PakiPenguin | anyone here using areski calling card system?? |
19:03.30 | Kernel_core | OKey |
19:03.35 | cluecon | at561: so you want server a to use b and c as gateways with failover to c if b is down? |
19:03.41 | at561 | pakipenguin, what's that |
19:03.43 | Godsey | wacky_: you may need to disable srv lookup |
19:03.44 | Kernel_core | file: what do you think ? |
19:03.47 | at561 | yes cluecon |
19:04.00 | file | Kernel_core: I think you're a potato |
19:04.05 | at561 | access to a is minimal, too |
19:04.09 | PakiPenguin | at561, its a calling card( prepaid calling card system ) written in php and pgsql |
19:04.22 | at561 | in other words changing its config would require a huge favor from the admin |
19:04.42 | cluecon | some dialplan logic might accomplish this. i think if b is down when you attempt the dial it will return with a congested status which you could trap and then pass the call over to c. |
19:04.56 | Kernel_core | file :and you act as salt :) |
19:05.24 | wacky_ | file - is /etc/hosts considered ? |
19:05.28 | cluecon | at561: that could complicate things. |
19:05.30 | at561 | yeah. i'm thinking of making c try to dial b if it receives any of its calls |
19:05.43 | at561 | and have them both register as the same peer |
19:06.53 | file | yes. |
19:06.58 | Hmmhesays | the areski cdr thing is pretty good, I can't imagine he would have done a bad job on the cc agi's |
19:08.20 | wacky_ | ok thanks guys |
19:11.20 | *** join/#asterisk bkw__ (~brian@adsl-69-154-144-52.dsl.tulsok.swbell.net) |
19:13.18 | *** join/#asterisk fugitivo (~ajf@168.226.246.239) |
19:13.28 | fugitivo | hiu |
19:16.16 | Hmmhesays | yo |
19:17.03 | cluecon | eh |
19:17.21 | Hmmhesays | heh |
19:17.43 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
19:17.47 | Hmmhesays | so joan jett was pretty scary last night |
19:18.53 | Godsey | nearly everything is :) |
19:19.03 | citats | WilliamK: should be possible to do with extensions.conf processing and astdb |
19:20.31 | Hmmhesays | i second citats |
19:20.33 | citats | hmmm, although unless theres an app to authenticate off voicemail passwords and return then that could be a sticky part |
19:20.51 | Hmmhesays | a simple agi could auth voicmail passwords |
19:20.56 | Hmmhesays | *voicemail even |
19:21.13 | Hmmhesays | there's about a million ways you could do it |
19:21.26 | citats | yeah, but thats not the elegant solution i was thinking of |
19:22.13 | Hmmhesays | http://www.voip-info.org/tiki-index.php?page=Asterisk+authenticate+using+voicemail+passwords |
19:22.27 | citats | something already? cool |
19:22.44 | Hmmhesays | i dunno if it's "elegant" enough for you |
19:23.15 | Hmmhesays | i joke |
19:23.24 | *** join/#asterisk zoa (zoa@82.103.76.147) |
19:24.58 | Chuji | Anyone fool with sphinx at all? |
19:25.08 | Chuji | Do I have train the damn thing before I can use it? |
19:25.24 | citats | definitely not the elegant solution i would use, but it uses my perl module so good enough for me :) |
19:25.47 | Hmmhesays | I only go for elegant if I have time to do so |
19:25.54 | hardwire | I love you all |
19:25.55 | hardwire | really |
19:25.58 | Hmmhesays | if it's stable and it works... |
19:25.58 | hardwire | you are all so very cool |
19:25.59 | hardwire | and |
19:26.05 | hardwire | I would be heartbroken if all of you left me |
19:26.14 | Hmmhesays | been drinkin' hardwire? |
19:26.15 | hardwire | just thinking about it gives me insecurity issues. |
19:26.23 | hardwire | Hmmhesays: no.. too much coffee. |
19:26.27 | Hmmhesays | haha |
19:26.32 | hardwire | my poor brain |
19:26.42 | hardwire | hah |
19:26.43 | Hmmhesays | i got my drink on last night... i'm moving pretty slow today |
19:26.47 | Hmmhesays | ribfest was last night |
19:26.48 | hardwire | ugh |
19:26.54 | fugitivo | i don't like the quality of my gsm files, is any trick to have more quality on the files? |
19:26.55 | hardwire | I just got done being on a boat for 5 days |
19:27.01 | hardwire | and we headed to the first bar in sight |
19:27.10 | hardwire | drank around 5 pitchers of amber inbetween 3 of us |
19:27.20 | Hmmhesays | fugitivo: use different format? |
19:27.20 | fugitivo | with sox -r 8000, quality is really poor |
19:27.24 | hardwire | I fell over when we got back to the hotel and was drunk for the next day.. poor! |
19:27.36 | hardwire | I had a meeting with the us coast guard that day too.. ugh |
19:27.38 | Hmmhesays | haha nice |
19:27.39 | fugitivo | Hmmhesays: the default asterisk gsm files have good quality |
19:27.56 | hardwire | I got 11 last night.. thats twice as much as usual |
19:27.58 | Hmmhesays | check the wiki and the mailing list, i'm sure there is info there |
19:28.00 | hardwire | I am all happy today |
19:28.14 | hardwire | so |
19:28.14 | cluecon | fugitivo: quality will depend heavily on the source file. |
19:28.18 | hardwire | voice over starband |
19:28.37 | hardwire | http://bogomip.com/~hardwire/Starband_Install.jpg |
19:28.39 | hardwire | works great |
19:28.47 | cluecon | how about voice over microwave? think it'll work? |
19:29.25 | fugitivo | cluecon: the source file is a studio recorded audio, great quality |
19:29.57 | hardwire | I'm the big one in that picture. |
19:30.02 | Hmmhesays | so use the same software to turn it convert to gsm |
19:30.04 | cluecon | fugitivo: what is the original format and rate? |
19:30.05 | hardwire | everybody else is so very tiny |
19:30.27 | Hmmhesays | awesome |
19:30.55 | Hmmhesays | http://www.grumpy-monkey.com/users/3853/huh.jpg <-- Me |
19:31.05 | hardwire | jesus christ |
19:31.06 | fugitivo | cluecon: wav, 44100hz, 16bit |
19:31.10 | hardwire | it would be like cuddling a porcupine |
19:31.24 | hardwire | cluecon: hah |
19:31.24 | Hmmhesays | cluecon where in ok are you from? |
19:31.38 | hardwire | cluecon: I would say move somewhere other than anchorage |
19:31.39 | cluecon | fugitivo: what command are you using to do the conversion? |
19:31.40 | hardwire | like homer |
19:31.52 | hardwire | otherwise the misquitos will carry off your pets |
19:31.57 | Hmmhesays | cuddling porcupine? |
19:32.00 | fugitivo | cluecon: i tried "sox file.wav -r 8000 -c 1 file.gsm" |
19:32.02 | hardwire | I just had to spray my entire lawn to get rid of all them blood suckers |
19:32.38 | Chuji | cluecon : you fool with sphinx at all/ |
19:32.40 | Chuji | ? |
19:32.42 | hardwire | Hmmhesays: whats funny is.. you fit the demographic I would concider saying things like "Monkey Poo" |
19:32.48 | Xumpi | oh man |
19:33.02 | Hmmhesays | hardwire wtf are you talking about? lol |
19:33.09 | Xumpi | "You can also catch some of the developers online at irc.freenode.net, #iaxclient, or #asterisk" @ http://iaxclient.sourceforge.net/ |
19:33.10 | hardwire | http://www.grumpy-monkey.com/index.php |
19:33.16 | Xumpi | where can i get the source code ? |
19:33.26 | cluecon | Hmmhesays: Tulsa, thunderstorms this morning (lightning struck the backyard) and now it's about 85 out and 60% humidity makes it feal like closer to a hundred. |
19:33.28 | Hmmhesays | oh lol |
19:33.30 | hardwire | heh |
19:33.48 | Hmmhesays | i once knew a girl from tiawah? |
19:33.52 | Hmmhesays | i think that's spelled right |
19:34.00 | hardwire | atleast you have good taste in guitars. |
19:34.27 | Hmmhesays | she tore my heart out and kicked it across the floor |
19:34.29 | cluecon | Chuji: no, haven't made it to sphinx yet but it is on my list. |
19:34.31 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
19:34.56 | Hmmhesays | hardwire: it's a epiphone, couldn't afford a gibson/gibson when I graduated highschool |
19:35.08 | hardwire | heh! |
19:35.19 | hardwire | I had a peavey nitro so don't be ashamed |
19:35.25 | Hmmhesays | but gibson is epiphones parent company |
19:35.33 | Hmmhesays | so it plays very well |
19:35.42 | cluecon | fugitivo: let me see what it is I use, that doesn't look right. |
19:35.43 | hardwire | epiphones aren't terrible at all |
19:35.47 | hardwire | just upgrade the strings and be happy. |
19:35.56 | hardwire | its all in the strings |
19:36.00 | Hmmhesays | yeah, I use erney ball on those 10's |
19:36.01 | hardwire | afaic |
19:36.15 | hardwire | I can't play a amn thing now |
19:36.18 | hardwire | damn |
19:36.27 | hardwire | I never took lessons |
19:36.31 | hardwire | just played and cried a lot |
19:36.34 | hardwire | it was that sad |
19:36.45 | Hmmhesays | me either, i just played a lot.... that acoustic has seen many a drunken fire pits |
19:36.46 | hardwire | but my friends played very well. |
19:37.28 | Hmmhesays | good or bad doesn't really matter, if you can play a tune that people can sing along to... everyone will think you're awesome |
19:37.35 | hardwire | haha |
19:37.36 | hardwire | yeh |
19:37.54 | Hmmhesays | well i've been sitting here for 2 hours... on my day off |
19:38.01 | Hmmhesays | haha time to jet |
19:38.06 | hardwire | GOQ |
19:38.08 | hardwire | err |
19:38.09 | hardwire | go! |
19:38.19 | Hmmhesays | oh and just for you hardwire .......... |
19:38.22 | hardwire | anyhoot.. I have questions about speex for anybody who has mastered it :) |
19:38.24 | Hmmhesays | "MONKEY POO" |
19:38.29 | hardwire | Hmmhesays: YAAAY |
19:38.37 | Hmmhesays | ok, later |
19:39.35 | hardwire | the speex implementation in codec_speex is all vbr right? |
19:40.40 | cluecon | fugitivo: i use sox filename.wav -r 8000 -c 1 filename.gsm resample -ql |
19:41.21 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
19:46.09 | wooden | how to define a default phone in the extensions.conf for all incoming calls? |
19:46.33 | hardwire | I have a house to clean |
19:48.03 | cluecon | wooden: create a s extension in the default context that your calls come into. |
19:48.44 | wooden | please give me example for it |
19:51.03 | wooden | cluecon: can u give me a exten for a standard phone? |
19:51.19 | cluecon | wooden: have you setup an s extension before? |
19:51.24 | wooden | yes |
19:51.31 | wooden | and it works |
19:51.36 | wooden | but dont know it |
19:51.46 | wooden | for standard phone |
19:52.10 | wooden | exten => *,1,Dial(SIP/snom190,,r) |
19:52.12 | wooden | ? |
19:52.32 | *** join/#asterisk mafia (pure@mother.phuck.in) |
19:52.35 | mafia | hi |
19:52.38 | cluecon | wooden: yes. put that in the incoming context for your calls. |
19:52.52 | wooden | ok |
19:52.58 | mafia | im new to asterisk, tryin to get it working |
19:53.14 | mafia | lol |
19:53.16 | *** join/#asterisk drumkilla_ (~russell@user-24-214-77-225.knology.net) |
19:53.16 | *** mode/#asterisk [+o drumkilla_] by ChanServ |
19:53.18 | mafia | webpage |
19:53.24 | mafia | :P |
19:53.39 | mafia | uhm |
19:53.45 | mafia | can someone help me get it werkin? |
19:54.09 | mafia | lol |
19:54.11 | cluecon | depends on what the problem is mafia. step 1 is always read the docs. |
19:54.19 | mafia | right |
19:54.22 | trasherr | cluecon: you're talking about an "incoming context"... how do i define something like that? i mean.. e.g. if i had a "context=capi" in capi.conf, isn't it the context for incoming AND outgoing calls? |
19:54.30 | mafia | well i changed the one line |
19:54.38 | mafia | exten line |
19:54.46 | mafia | to what nufone told me to |
19:54.59 | *** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net) |
19:54.59 | *** mode/#asterisk [+o drumkilla] by ChanServ |
19:55.02 | mafia | but Unable to connect to remote asterisk |
19:55.05 | mafia | ;/ |
19:55.14 | cluecon | trasherr: by incoming context, i mean the following |
19:55.20 | *** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net) |
19:55.20 | *** mode/#asterisk [+o drumkilla] by ChanServ |
19:56.38 | cluecon | on my zap lines: all incoming calls get dropped into an 'incoming' context because I've set the context=incoming line. i'm not familiar with capi, but with zap, outbound calls are made using dial commands and the outbound context doesn't matter (at least not in my case). |
19:57.05 | zoa | contexts are not used for outgoing calls |
19:57.13 | zoa | you have two call legs all the time |
19:57.22 | zoa | so for every call you have an incoming context anyway |
19:57.24 | cluecon | mafia: pastebin your cli output and your extensions.conf with sensitive info removed and i'll take a look. |
19:57.30 | zoa | sip to zap -> sip context is used |
19:57.40 | zoa | zap to iax2 -> zap context is used |
19:57.46 | *** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net) |
19:57.46 | *** mode/#asterisk [+o drumkilla] by ChanServ |
19:57.54 | cluecon | yeah, what zoa said...as long as he is right. |
20:00.45 | mafia | hm |
20:02.18 | trasherr | zoa: ok, i see.. thanks |
20:02.53 | zoa | thanks for the question, i just added it to my astricon tutorial :) |
20:03.19 | cluecon | hey zoa: think i could get a copy of that tutorial for my reference purposes? |
20:03.55 | *** join/#asterisk drumkilla (~russell@user-24-214-77-225.knology.net) |
20:03.55 | *** mode/#asterisk [+o drumkilla] by ChanServ |
20:04.07 | zoa | of course |
20:04.14 | zoa | it will be online after the speech |
20:04.59 | cluecon | thanks zoa. just point me in the right direction when it's up. |
20:05.19 | cluecon | if i thought i could get back in the country and had the money, i would be at astricon. |
20:05.43 | zoa | it will be at asteriskguru.com |
20:07.18 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
20:07.23 | *** join/#asterisk mtgh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
20:08.58 | cluecon | bookmarked. thank you. |
20:12.51 | trasherr | could anybody say me, why i can't login to sipgate with asterisk? Jun 11 22:12:14 NOTICE[3580]: -- Registration for '4213871@sipgate.de' timed out, trying again |
20:14.20 | *** join/#asterisk syle (~blah@wnpgmb06dc1-43-135.dynamic.mts.net) |
20:14.58 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
20:16.16 | *** join/#asterisk bjohnson (~bjohnson@66.11.165.163) |
20:16.41 | shido | finally |
20:16.43 | shido | some rain |
20:17.29 | PakiPenguin | send us some too shido! |
20:18.03 | cluecon | we got slammed yesterday and early this morning. Not cool when your wakeup call is a lightning bolt in the backyard. |
20:18.12 | shido | hehehe |
20:18.18 | shido | we desperately needed some here |
20:18.21 | shido | it was HOT! |
20:18.23 | mtgh | shido: are you in MI |
20:18.26 | shido | and I have no working A/C |
20:18.30 | shido | yes |
20:18.36 | cluecon | we needed it too. but it didn't help the heat. |
20:18.36 | mtgh | shido: So am I, where are you? |
20:18.44 | PakiPenguin | its not hotter then where i am |
20:18.46 | shido | Detroit |
20:19.14 | PakiPenguin | The current temperature in Multan, Pakistan is 91°F (11:00 PM PKT on June 11, 2005). Conditions: Blowing Sand. Humidity: 28%. Dew Point: 62°F. Pressure: 29.55 in 1001 hPa. Visibility: 0.0 miles 0.5 kilometers. |
20:19.46 | citats | shido: you living here now? |
20:19.58 | mtgh | Conditions: Blowing Sand .... nice |
20:20.12 | shido | "here" ? |
20:20.16 | PakiPenguin | yeah , we had a furious sand storm earlier |
20:20.18 | citats | michigan |
20:20.21 | shido | dodo |
20:20.28 | shido | I've always been a michigander |
20:20.42 | citats | thought you were from canada |
20:20.45 | shido | no |
20:20.48 | cluecon | Paki: we are close, would be pretty equal if we hadn't started the day with rain. |
20:20.48 | shido | I visited Canada |
20:20.55 | shido | and got stuck |
20:20.58 | shido | due to sept 11 |
20:21.35 | shido | when we have voip issues |
20:21.42 | shido | I head over to our secret location |
20:21.56 | shido | and reboot a few things or get the crash cart and fix0r them |
20:22.21 | citats | shido: i don't have any gear at that place anymore, so i wont ever see ya there |
20:22.33 | shido | where are you citats ? |
20:22.43 | citats | i live in howell for now |
20:23.12 | shido | why'd u move your gear? |
20:23.46 | citats | shido: i was laid off by the company that had gear there, and i moved my personal cabinet to a diff location |
20:24.22 | shido | citats, you know you coulda just asked us |
20:24.47 | shido | our cabinets are relatively empty - due to jerjer's Linux Kung Fu |
20:25.08 | shido | if we filled them we could take over the world finally - but he keeps tellin us "in due time" |
20:25.23 | citats | shido: i was paying for the one there. i moved my own stuff to another location where i get the cabinet comp'd for helping out every once in a while |
20:26.02 | shido | well we have a ton of bandwidth at each and every one of our locations |
20:26.29 | PakiPenguin | anyone care to look at this http://pastebin.ca/13983 <-- i am using areskicc , giving me channel status : 6 , from the agi and then exits |
20:26.49 | shido | talk to areski, PakiPenguin he's very helpful when u drop him some cash |
20:27.01 | PakiPenguin | :) alright |
20:28.19 | shido | tell him shido6@gmail.com sent you |
20:28.50 | shido | Hostans friend |
20:29.25 | shido | engrish not so good but understand him you will, yes |
20:29.50 | shido | hell, my engrish isnt so good either |
20:30.01 | asterbeast | what port do i need to open in my firewall to let an X-lite softphone connect with my PBX? |
20:30.11 | zoa | lots of them :) |
20:30.55 | X-Rob | I'd actually say 'most' |
20:30.57 | X-Rob | 8) |
20:30.58 | hardwire | anybody found a good voice processing card for non circut interfacing? |
20:31.11 | asterbeast | why? |
20:31.23 | shido | hardwire, softwire |
20:31.25 | asterbeast | 5060 udp is for sip ok... |
20:31.29 | shido | no need for hardwiring |
20:31.42 | shido | hardwiring will cost you $$$$$$ |
20:31.45 | hardwire | shido: its what I doooo :) |
20:31.50 | shido | hehehe |
20:31.56 | zoa | http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html |
20:31.59 | zoa | read this tutorial |
20:32.03 | shido | how many simultaneous calls, hardwire ? |
20:32.08 | zoa | it might explain something |
20:32.11 | zoa | such as, use iax2 |
20:32.13 | zoa | :) |
20:32.23 | hardwire | shido: no idea.. I just need to work on low power voip systems |
20:32.43 | hardwire | so I need to use 266mhz hardware for the most part |
20:32.52 | shido | no problem |
20:32.53 | shido | how much ram |
20:32.55 | shido | ? |
20:33.12 | hardwire | 128 :) |
20:33.16 | hardwire | actually.. 64 |
20:33.18 | hardwire | hah |
20:33.19 | shido | for 60 calls? |
20:33.25 | X-Rob | hardwire, look at the VIA EPIA boards, they use bugger all power, and you can get up to 1ghz |
20:33.29 | shido | what else do u need it to do? |
20:33.33 | hardwire | X-Rob: I already use them |
20:33.42 | shido | just calls |
20:33.43 | shido | no ivrs |
20:33.46 | shido | no musiconhold |
20:33.50 | hardwire | well |
20:33.50 | shido | no db? |
20:34.01 | hardwire | I need voice prompts |
20:34.01 | hardwire | and moh |
20:34.05 | hardwire | and usb for timing for meetme |
20:34.15 | shido | hrmm |
20:34.19 | shido | ur gonna have to cut the kernel |
20:34.28 | hardwire | with scissors? |
20:34.32 | shido | and build your own distro |
20:34.35 | hardwire | I already have |
20:34.40 | shido | no not with scissors |
20:34.42 | shido | with "rm" |
20:34.43 | shido | :) |
20:34.47 | hardwire | its called bulletdebian :) |
20:34.49 | shido | and vi |
20:34.54 | shido | bulletdebian |
20:35.01 | hardwire | I need to make a homepage for it |
20:35.08 | X-Rob | hardwire - USe 2.6 and use RTC. No need for USB |
20:35.15 | hardwire | X-Rob: ztrtc? |
20:35.15 | shido | tell me more about bulletdebian |
20:35.19 | shido | I'm running ubuntu here |
20:35.26 | hardwire | shido: what I did was pretty simple |
20:35.43 | hardwire | custom kernel based around debians build system for kernels.. targetted for the geode processor |
20:35.53 | hardwire | or any target processor.. even Via C3 |
20:36.02 | X-Rob | hardware - cvs ztdummy.c has support for using 2.6's RTC, rather than USB |
20:36.07 | X-Rob | is much MUCH more accurate |
20:36.08 | hardwire | I have a build system for the distro that has a target drive.. |
20:36.14 | hardwire | it makes a boot drive |
20:36.15 | X-Rob | Real Time Clock |
20:36.23 | hardwire | a squashfs for the root that is partition 2 |
20:36.29 | hardwire | partition 3 is the backup storage |
20:36.33 | hardwire | partition 3 is the unionfs mount |
20:36.38 | *** join/#asterisk Sparkie- (sparkie@24-50-118-84.lkwnny.adelphia.net) |
20:36.39 | hardwire | 4 I mean |
20:36.48 | hardwire | every boot it populates partition 4 with partition 3 |
20:36.51 | hardwire | and mounts that on top of / |
20:37.05 | hardwire | a script rsyncs p4 to p3 as perminant storage |
20:37.11 | hardwire | hence.. bullet"proof" debian |
20:37.15 | asterbeast | zoa: I want to connect to my pbx from the internet... |
20:37.18 | Sparkie- | How do you make asterisk answer the phone faster ... currently it's waiting 2 rings to direct my call to the digital receptionist. Is there a way to speed that up to 1 ring? |
20:37.26 | hardwire | you can disconnect its power at any time and no problems occur |
20:37.44 | X-Rob | Sparkie - turn off caller ID detection |
20:37.49 | shido | Sparkie-, dialplan logic |
20:38.18 | X-Rob | IT's waiting for caller ID to be sent before it answers. it waits ~10 secs before it gives up and answers |
20:38.38 | Sparkie- | k |
20:39.47 | hardwire | ugh |
20:39.51 | hardwire | never drink blue sky energy drink |
20:39.57 | trasherr | does anybody know a sip program (windows), where i can answer a call with a hotkey? |
20:39.57 | hardwire | it tastes like pepto |
20:40.22 | shido | heh |
20:40.31 | X-Rob | Australian Mountain Dew tastes like Diesel. |
20:40.35 | X-Rob | It's _horrible)_ |
20:44.12 | bkw__ | muhahha |
20:44.17 | bkw__ | my switch kit ROCKS |
20:44.51 | mafia | switch kit ? |
20:45.12 | zoa | i just calculated that an asterisk server could cost you 1.7 million euro per 24 hours if hacked |
20:45.22 | zoa | if its has 2x4 e1 cards |
20:45.29 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
20:45.40 | kajtzu | how do you come up with that number? |
20:46.13 | zoa | calls to inmarsat are 5 euro per minute |
20:46.28 | *** join/#asterisk cluecon (~asterisk-@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
20:47.11 | kajtzu | right. you're still gonna have a problem calling someone over inmarsat with 240 channels 24x7 |
20:47.26 | *** join/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
20:48.00 | zoa | yeah true |
20:48.01 | zoa | :) |
20:48.02 | zoa | but still |
20:49.02 | zoa | some belgian toll numbers would be half that price |
20:49.05 | zoa | and thats possible:) |
20:50.18 | kajtzu | hmm last time I used inmarsat was in a comms room on a cruise liner ;) |
20:52.22 | zoa | i never used it |
20:52.28 | zoa | must be extremely expensive to call with it |
20:53.37 | kajtzu | mortals use mtn or one of the other satellite bandwidth providers |
20:53.56 | kajtzu | inmarsat tends to be company business or emergency use |
20:54.04 | kajtzu | (mortals = paying customers) |
21:03.15 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
21:10.48 | bkw__ | OMG BECKY realtime OWNZ |
21:11.53 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
21:11.56 | zoa | brian |
21:11.57 | zoa | mr mr mr |
21:12.59 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
21:24.12 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
21:28.16 | *** join/#asterisk loick (~loick@APuteaux-151-1-28-174.w82-124.abo.wanadoo.fr) |
21:31.44 | WilliamK | anyone know howto get the Verticle Service Activation codes to work with *? |
21:32.01 | X-Rob | Which one would you like to work? |
21:32.49 | WilliamK | how about all of them? =) |
21:32.55 | WilliamK | *67, 82, etc.. |
21:32.55 | WilliamK | =) |
21:33.02 | X-Rob | 'all' of them are usually implemented by your handset |
21:33.15 | X-Rob | any specific one I might be able to hint at using with a dialplan |
21:33.31 | WilliamK | Sipura has it in there, however it gives a busy signal right after and sends nothing to * |
21:33.40 | shido | hehe |
21:33.44 | shido | dialplan logic WilliamK |
21:34.07 | WilliamK | gotta create a dialplan per service that I want to enable then? =) |
21:34.09 | X-Rob | eg, call waiting is a function of your phone, and of asterisk |
21:34.15 | shido | errr |
21:34.24 | shido | just dialplan entries |
21:34.24 | X-Rob | asterisk can either not send a call and redirect it to voicemail |
21:34.31 | X-Rob | or it can send a call and get a busy back from your phone |
21:34.37 | WilliamK | X-Rob, I'm speaking about outbound services, not the inbound ones =) |
21:34.39 | shido | not an entire plan for each service unless you are creating your services that way - to enable and disable services, etc |
21:34.41 | X-Rob | or it can send a call and your phone can generate call waiting tones |
21:35.06 | WilliamK | I want the user to be able to like hit *67 and make it work |
21:35.08 | WilliamK | =) |
21:35.11 | X-Rob | What's *67 meant to do? |
21:35.23 | WilliamK | disable callerID per call outbound |
21:35.35 | WilliamK | *82 does it perm. |
21:35.41 | X-Rob | That's a phone and asterisk function again |
21:35.59 | X-Rob | look at AMP's dialer agi |
21:36.08 | X-Rob | it uses astdb to figure out wether to send caller ID or not |
21:36.16 | WilliamK | ah |
21:36.24 | X-Rob | but most phones can handle not sending caller id |
21:36.44 | WilliamK | the sipura should be able to, not sure why it's giving back a busy signal |
21:37.04 | X-Rob | probably coz it's not enabled |
21:37.08 | X-Rob | RTFM? |
21:37.24 | X-Rob | my 2000 does it |
21:37.27 | X-Rob | I think |
21:37.29 | X-Rob | (checks) |
21:37.41 | opus__ | hi x-rob |
21:37.41 | WilliamK | I am using a 2000 and it is enabled |
21:37.51 | blitzrage | zoa: have you seen the hotel that Astricon Europe is at? |
21:37.55 | X-Rob | Well, it gave me a dial tone, but it still sent caller ID |
21:38.22 | X-Rob | I dialed *67, got a dialtone, dialed the GXP2000, and I got caller ID |
21:38.38 | X-Rob | Don't know why. |
21:39.25 | blitzrage | zoa: http://www.hotelauditorium.com/ |
21:41.25 | zoa | yeah i know it |
21:41.28 | zoa | will be cool |
21:41.37 | zoa | i hope im not speaking in that very big room :) |
21:42.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
21:43.55 | WilliamK | okie, I factory defaulted my sipura and now it dials without the busy but gives the callerID |
21:45.08 | ska | Sorry, I missed the respones to my question about incoming calls.. Is it possible to recieive more incoming calles than you have t1 channels for? |
21:46.36 | zoa | blitzrage: see private, fast i need to go to bed |
21:46.50 | cluecon | ska: the number of calls will always be limited by the number of available channels. |
21:46.55 | Blissex | ska: depends what you mean by «receive» |
21:47.34 | ska | Blissex: i mean incoming calls to a set of finite numbers. |
21:48.10 | Blissex | ska: thats not that useful |
21:48.39 | Blissex | ska: for example, if you get calls routed to you over IAX2, of course many many more. |
21:49.22 | zoa | all extra calls will give you busy |
21:49.23 | cluecon | Blissex: he is asking bout T1. |
21:49.27 | ska | Blissex: so if we use IAX we are only limited by available bandwidth? |
21:49.45 | zoa | bandwidth and cpu |
21:50.03 | Blissex | ska: if you mean routed to you over POTS or ISDN lines, the routing telephone company cannot route more than N, and will return some ''system busy'' to the caller. |
21:50.10 | ska | zoa: good point. Never thought about cpu limits.. |
21:50.26 | *** join/#asterisk apolinux (~carlos@ppp-105-36.telesat.com.co) |
21:50.35 | cluecon | ska: T1 is limited to either 23 or 24 channels. Practically everything else is strictly limited by the power of the machine and available bandwidth. |
21:50.40 | ska | Are there any decent estimates for cpu/call numbers? |
21:50.40 | Blissex | ska: CPU limits matter in particular if you use some compressed scheme... But not just because of that. |
21:50.47 | apolinux | ehlo pe0ple |
21:51.08 | Blissex | apolinux: you speaking SMTP or just a typo? :-) |
21:51.52 | apolinux | jejejeje yes.. |
21:53.28 | ska | I suppose there are special cards with compression and codecs that can speed things along.? |
21:53.50 | apolinux | someone tried to configure a voip like sua <-> asterisk <-> gnugk <-> gw <-> pstn , to call from sip phone to pstn? |
21:54.32 | Blissex | ska: I think Digium themselves sell something like that. |
21:54.45 | Blissex | ska: but it can be also ''just buy a bigger PC''. |
21:54.53 | zoa | ska, no such cards exist for asterisk (yet) |
21:55.02 | Blissex | ska: also it is not that difficult to use multiple PCs. |
21:55.06 | apolinux | i'm talking alone.. <:-| |
21:55.20 | zoa | apolinux: i did before |
21:55.29 | zoa | but there is no good h323 implementation for asterisk |
21:55.33 | Blissex | apolinux: be patient... |
21:55.34 | zoa | i could not get anything stable |
21:55.37 | zoa | and i gave up |
21:55.41 | zoa | after months of trying |
21:55.56 | Blissex | apolinux: just dtop H323 it is the spawn of the devil and part of the plan for apocalypse... |
21:56.08 | apolinux | i make it works.. |
21:56.11 | apolinux | but |
21:56.12 | ska | blissex: like in high availability setup with multiple * systems? |
21:56.40 | apolinux | only the voice goes one way..from sip to pstn |
21:56.48 | blitzrage | zoa: replied to PM |
21:56.55 | Blissex | ska: «high availability» is a bit of an ambitious phrase. |
21:57.03 | apolinux | no..i mean from pstn to sip |
21:57.28 | apolinux | but doesn't works from the sip to pstn.. |
21:57.31 | ska | ska: perhaps simple load balancing? |
21:57.33 | Blissex | ska: but for example if you setup a nice shared user database, and route calls between the various PCs, its not that hard. |
21:57.50 | apolinux | zoa: which software do you use it? |
21:58.28 | zoa | i tried everything available |
21:58.30 | zoa | chan_h323 |
21:58.33 | zoa | chan_oh323 |
21:58.37 | zoa | and chan_ooh323 |
21:58.38 | apolinux | jejejejej |
21:58.41 | apolinux | hm.. |
21:59.02 | apolinux | which problems do you had? |
22:00.21 | apolinux | :-O ... |
22:01.06 | X-Rob | Give up on h323 |
22:01.18 | X-Rob | as was recently pointed out on asterisk-users, it causes cancer. |
22:03.52 | bkw__ | h323 works |
22:03.53 | bkw__ | chan_woomera works |
22:04.07 | bkw__ | the one in asterisk-addons works |
22:04.09 | bkw__ | kinda |
22:05.03 | zoa | bkw, i had bad luck with it a while ago |
22:05.06 | zoa | with the ooh323 |
22:05.20 | zoa | but maybe it was the carriers fault |
22:05.22 | bkw__ | it works and makes a call |
22:05.28 | bkw__ | I called from it to woomera |
22:05.29 | bkw__ | worked |
22:05.35 | bkw__ | woomera to it worked |
22:06.02 | apolinux | zoa.. |
22:06.08 | tzanger | bkw__: did we ever figure out a definitive answer about what entailed "linking" to Asterisk? |
22:06.17 | apolinux | what is the ooh323? where i can get it? |
22:06.23 | bkw__ | no but i'm 100% sure agi and talking to the manager does NOT count |
22:06.43 | tzanger | files don't explode |
22:06.51 | tzanger | bkw__: yeah that's the common sense answer :-) |
22:06.52 | bkw__ | depends on the file system |
22:07.14 | bkw__ | file my dear |
22:07.20 | bkw__ | you wanna test my new stuff I put up today? |
22:07.26 | zoa | what what what |
22:07.29 | zoa | gimme the goodies |
22:07.32 | file | ohhhhh not that |
22:07.39 | bkw__ | its a service goodie for asterlink customers |
22:07.50 | bkw__ | hehe |
22:08.07 | zoa | and what is it ? |
22:08.07 | file | btw, did you try put international into the standard routing stuff, or do I need to add a service flag to the account? |
22:08.16 | file | s/try// |
22:08.25 | bkw__ | file, the way it is now you don't have to bother with rating calls or billing records |
22:08.28 | zoa | im going to be dnow |
22:08.31 | bkw__ | thats still done on the asterisk side |
22:08.36 | file | yeah i know that |
22:08.40 | file | I just meant for my regular calling :P |
22:08.47 | bkw__ | ah call 996 |
22:09.48 | *** join/#asterisk yaaar (~todd@12-216-227-245.client.mchsi.com) |
22:09.49 | yaaar | word |
22:10.09 | *** join/#asterisk ptblank (~MURDER1@68-65-93-235.lmdaca.adelphia.net) |
22:10.15 | *** join/#asterisk diablopico (~russ@ip68-8-227-7.sd.sd.cox.net) |
22:10.41 | diablopico | i need help |
22:13.12 | diablopico | is anyone able to help with a h323-gateway config with 2 digium/zaptel e1 cards /connecting with CAS/r2 |
22:15.03 | Hmmhesays | r2 yuck |
22:23.54 | SwK | shit shit shit |
22:24.18 | SwK | something busted my box to hell in a head check out from about 20 minutes ago |
22:24.52 | SwK | iax no worky work... dont even get debug output hah |
22:24.57 | *** join/#asterisk fiber0pti (~john@209-234-169-232.gen.twtelecom.net) |
22:25.25 | fiber0pti | What kind of tools does Digum offer with their "Asterisk Business Edition?" |
22:27.26 | drumkilla | nothing additional |
22:27.37 | SwK | you just get to pay for it |
22:27.50 | drumkilla | it's a commercially licensed and supported version of Asterisk |
22:27.53 | SwK | (well I think they throw in some support to) |
22:28.42 | SwK | drumkilla what all support do they provide under that lic? |
22:29.03 | bkw__ | drumkilla, we need a RAND function |
22:29.08 | file | omg, HI |
22:29.14 | bkw__ | I need a random number between 1 and 7 |
22:29.20 | bkw__ | and no real good way to do it in the dialplan |
22:29.23 | drumkilla | bkw__: I think there is an app_rand |
22:29.31 | bkw__ | that will not give me a result |
22:29.35 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
22:29.35 | bkw__ | it will only branch |
22:29.55 | drumkilla | write a rand dialplan function |
22:29.58 | tzanger | I love getting "deleted your email because it contained sensitive words" emails |
22:30.08 | bkw__ | tzanger, who sent that? |
22:30.14 | tzanger | what's the matter, can't take me saying "fuck" or "bitch" |
22:30.18 | tzanger | dunno |
22:30.22 | tzanger | somoene subscribed to -users |
22:30.34 | tzanger | my ex's old job had the same kind of fucked up filter |
22:30.53 | Hmmhesays | I got no filter at work |
22:30.55 | tzanger | I can certainly understand it for OUTGOING mail, but to block customer's email without notifying the person it was intended for is just as bad if not worse |
22:31.21 | WilliamK | anyone know about the outdated modules warning messages? I keep seeing them show up at random when I recompile |
22:31.26 | asterbeast | does anyone have experience with handling firewall ports trough asterisk? |
22:31.27 | Hmmhesays | my email says I can add 4 1nch3s to my p3n0r daily |
22:31.43 | SwK | bkw_ you know what I dont even see it sending a register packet in tcpdump |
22:32.08 | SwK | Hmmhesays d00d sure fire way to get in to the guiness book of records! |
22:32.39 | asterbeast | i would like to open or close ports in my firewall with a script. when i call asterisk with my mobile. |
22:33.09 | SwK | "Hmmhesays: man with the worlds largest p3n0r... with a t00l that measures 56 inches!" |
22:33.23 | bkw__ | WHACK |
22:33.24 | SwK | asterbeast: so write an AGI for that |
22:33.26 | bkw__ | is iax broken? |
22:33.28 | Hmmhesays | SwK LOL |
22:33.30 | Syncros | moose penis ! |
22:33.37 | Hmmhesays | where! |
22:33.49 | file | Hmmhesays: that's sexy |
22:33.50 | SwK | bkw__ I dunno |
22:34.31 | SwK | but i can get calls out now just by adding in app_math which i commeneted out cause it would build... it has some f'd up error w/ gcc 2.95 in if(!foo){ |
22:34.44 | SwK | gcc repoprts it as a syntx error heh |
22:35.04 | asterbeast | Swk doenst anybody already did something like that? |
22:35.12 | tzanger | asterbeast hahaha love the nick |
22:35.18 | *** join/#asterisk ToyMan (~stuq@user-12lcqur.cable.mindspring.com) |
22:35.19 | SwK | asterbeast: dunno |
22:35.27 | Hmmhesays | strange conversations in here sometimes |
22:35.38 | tzanger | Hmmhesays: you don't know the half of it :-) |
22:35.57 | SwK | hah |
22:35.58 | shido | :) |
22:36.07 | tzanger | power flicker? wtf |
22:36.09 | Hmmhesays | i can only imagine |
22:36.15 | SwK | bkw__: tcpdumping IAX port dont show shit |
22:36.20 | tzanger | too bad I'm not gonna be here for this storm |
22:36.37 | SwK | bkw__ till I dial out hah |
22:36.43 | asterbeast | tzanger thanks!@ |
22:36.48 | Hmmhesays | It just rained hardcore here, I"m going to take pictures of the river |
22:36.58 | tzanger | were's here? |
22:37.04 | Hmmhesays | fargo/moorhead |
22:37.05 | SwK | its been raining here for like 3 days |
22:37.14 | Hmmhesays | the river is the red river of the north |
22:37.27 | SwK | my yard looks like ass cause the yardboy wont come and cut it if it looks likes its gonna rain today |
22:37.34 | SwK | (and its looked like that for about 2 weeks now) |
22:37.43 | tzanger | tell him to get his ass over and mow it BEFORE it fucking rains |
22:37.46 | Hmmhesays | most of the streets here are flooded |
22:38.48 | SwK | i'm just gonna go buy the wife a lawnmower and say "get out there and cut the grass bIatch |
22:39.02 | tzanger | SwK: heh |
22:39.04 | tzanger | I was out mowing today |
22:39.07 | tzanger | about 1/2 done |
22:39.44 | SwK | my yard wouldnt be that bad... cept for the damned johnson grass all over the place... that shit grows like foot tall in a week |
22:39.48 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
22:39.49 | xbmodder | I hope you mean mowing the lawn |
22:40.03 | tzanger | sounds like my johnson, now I know how it got its name :-) |
22:40.05 | xbmodder | lol |
22:40.16 | tzanger | it's up before you realize it |
22:40.31 | SwK | tzanger you on that same p3n0r plan as Hmmhesays? |
22:40.36 | tzanger | hahaha |
22:40.39 | xbmodder | bash.org? |
22:40.45 | tzanger | yeah mine's on consignment though |
22:40.52 | tzanger | I go by and visit it every day |
22:40.58 | SwK | hah |
22:41.15 | tzanger | er |
22:41.17 | tzanger | not consignment |
22:41.19 | tzanger | layaway |
22:41.26 | tzanger | dammit I hate hwen I fuck up a good joke |
22:41.49 | SwK | tzanger: does that mean "it's in tzanger's wifes purse until she lets you visit it"? |
22:42.10 | tzanger | no, I'm wifeless. Had one, she's WAS on consignment but someone fially picked her up |
22:42.17 | SwK | hah |
22:42.18 | tzanger | only she got 50% |
22:42.19 | Hmmhesays | someone want to sponsor me for cluecon? i just found out work can't pay it |
22:42.34 | tzanger | Hmmhesays: get in line, bitch. I'm first in line for sponsorship :-) |
22:43.10 | Hmmhesays | they said "we have enough to pay transportation, from old airline miles" |
22:43.37 | Hmmhesays | sad thing is, it's only like 600 miles away for me |
22:44.04 | tzanger | damn |
22:44.13 | tzanger | will they spring for half? |
22:44.21 | Hmmhesays | i'm going to see |
22:44.26 | tzanger | anyway gotta go |
22:44.27 | tzanger | later all |
22:44.30 | *** join/#asterisk cjk (~cjk@80.92.75.120) |
22:44.31 | Hmmhesays | yeah same here |
22:44.34 | shido | hrmm |
22:44.39 | shido | Hmmhesays, where are you? |
22:44.42 | Hmmhesays | fargo |
22:44.51 | shido | now for the cheap seats |
22:44.53 | shido | where is fargo? |
22:45.02 | Hmmhesays | right next to moorhead mN |
22:45.06 | Hmmhesays | fargo ND |
22:45.26 | cjk | hi, lets say i need some small custom asterisk things to be changed in the source code? who could i contract? is there a chance that they will be put into the asterisk tree? anyone interested, mostly basic things... |
22:45.29 | Hmmhesays | i'd drive there no problem |
22:45.41 | Hmmhesays | we just restructered though and money is really really tight |
22:46.48 | Hmmhesays | I gotta run, BBIA |
22:50.31 | *** join/#asterisk outtolunc (~me@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
22:58.58 | harryvv | really tight mmm |
23:00.23 | *** join/#asterisk cmk (~cmk_@p54A3FB87.dip.t-dialin.net) |
23:01.57 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
23:02.02 | diablopico | is anyone able to help with a h323-gateway config with 2 digium/zaptel e1 cards /connecting with CAS/r2 |
23:03.28 | diablopico | nobody ? |
23:05.13 | shido | yeah |
23:05.17 | shido | whats up diablopico ? |
23:05.28 | shido | frustrated? |
23:05.31 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
23:10.12 | diablopico | what happened shido |
23:12.58 | harryvv | anyone here know where voipsupply is located at? |
23:16.03 | X-Rob | 716 area code |
23:16.06 | X-Rob | wherever that is |
23:17.47 | X-Rob | Buffalo NY |
23:18.10 | harryvv | mm thats really to far from the pacific nw. |
23:18.26 | harryvv | so voipsupply and atacomm are the two major phone resellers. |
23:18.28 | X-Rob | specifically, 454 Sonwil Drive, Buffalo NY |
23:18.33 | X-Rob | gotta love google. |
23:21.10 | hardwire | you ever pick up the phone |
23:21.17 | hardwire | and somebody is just heavy breathing at you |
23:21.28 | hardwire | I swear to god it wasn't me |
23:23.33 | hardwire | it was him. |
23:25.36 | X-Rob | who, him? |
23:26.18 | hardwire | :) |
23:26.28 | hardwire | the commercials in anchorage are the worst |
23:26.37 | hardwire | the airtime must be incredibly cheap |
23:26.38 | X-Rob | Ahh. TV |
23:27.23 | X-Rob | http://www.nodedb.com/australia/qld/gladstone/? |
23:27.27 | Sedorox | What is this... T.V. You speak of ? |
23:27.27 | Sedorox | :p |
23:27.33 | X-Rob | Zoom out from there and you'll see where I live |
23:27.55 | X-Rob | TV ads here are so cheap that there are at least 3 companies that have ads made in the middle 70's still on TV |
23:27.58 | X-Rob | currently. |
23:28.17 | hardwire | haha |
23:28.28 | hardwire | we have a guy here that has a place called the matress ranch |
23:28.30 | hardwire | he sells beds |
23:28.37 | X-Rob | We changed our phone numbers from 049-72-xxxx to 07-4972-xxxx about 7 years ago |
23:28.39 | hardwire | and he has some of the worst tv ads ever made |
23:28.51 | X-Rob | two of those ads haven't even bothered updating the phone number on their ads |
23:28.56 | hardwire | wow.. you are way the hell up there |
23:29.11 | X-Rob | I'm 42kms south of the tropics |
23:29.23 | hardwire | I should come visit :) |
23:29.31 | X-Rob | feel free! |
23:29.33 | hardwire | we will go 4ms north |
23:29.34 | hardwire | err |
23:29.36 | X-Rob | We have a 6br house 8) |
23:29.39 | hardwire | 42kms |
23:29.43 | hardwire | holy crap |
23:29.59 | X-Rob | Uh |
23:29.59 | hardwire | did it grow from the ground up? |
23:30.02 | X-Rob | it's not a mansion |
23:30.05 | X-Rob | it's a highset place |
23:30.07 | X-Rob | built in underneath |
23:30.08 | X-Rob | 8) |
23:30.33 | X-Rob | where are you? |
23:30.39 | hardwire | nowhere near you |
23:30.44 | hardwire | I am in Alaska |
23:30.48 | X-Rob | Ahha. |
23:30.55 | X-Rob | Cold. Mooses. |
23:31.00 | hardwire | the mooses haunt me |
23:31.07 | hardwire | we are trying to escape this weekend actually |
23:31.11 | X-Rob | damn mooses. |
23:31.12 | hardwire | and have no idea where to go |
23:31.38 | X-Rob | it's a 14hour flight from LAX to sydney |
23:31.47 | X-Rob | then 3 hours flight north |
23:32.02 | hardwire | there goes the weekend? |
23:32.04 | hardwire | err |
23:32.05 | hardwire | ! |
23:32.08 | *** join/#asterisk Cybertoy (user@ool-4357da9f.dyn.optonline.net) |
23:34.19 | X-Rob | heh |
23:34.28 | Cybertoy | I'm implementing the telemarketer torture... anyone happen to have a nice women's voice prompts for the example in voip-info.org ready? |
23:34.42 | Cybertoy | otherwise I'll use my own unsexy voice... :) |
23:35.14 | X-Rob | You guys have -no- idea how good it is to not live in the US, really. |
23:35.26 | hardwire | oh don't start |
23:35.27 | X-Rob | I can't remember the last time someone tried to call me and sell me shit |
23:35.44 | Cybertoy | actually I do have an idea... I just moved to the states from Switzerland. |
23:35.48 | hardwire | it helps living in alaska |
23:35.54 | hardwire | it costs 5 times as much to call us |
23:35.56 | hardwire | so nobody does |
23:35.58 | Cybertoy | and once in a while some monkey would call me there to sell me a newspaper... |
23:35.59 | X-Rob | Cybertoy - is it as bad as you expected? |
23:36.20 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
23:36.29 | Cybertoy | the telemarketers here are worse... but my own phone number is not spoiled yet... but the one in the corporate appartment I'm in now is. |
23:36.49 | Cybertoy | actually ... the other day someone called on my swiss voip number and wanted to sell a newspaper... I was talking to him for about 15 minutes... |
23:36.52 | hardwire | sell sell sell guys |
23:36.53 | *** join/#asterisk file[desk] (~jcolp@mctnnbsah25-142166147147.nb.aliant.net) |
23:36.56 | Cybertoy | and then wanted to give him my address... |
23:36.57 | hardwire | don't you know the US rule of thumb? |
23:37.04 | Cybertoy | he didn't like my us address... |
23:37.16 | X-Rob | heh |
23:37.17 | Cybertoy | and was pretty annoyed that I wasted 15 minutes of his time. |
23:37.17 | |Vulture| | Cybertoy: who do you have a swiss DID from? |
23:37.23 | Cybertoy | www.sipcall.ch |
23:37.34 | hardwire | Cybertoy: a friend of mine forwarded to his sat phone from his home phone |
23:37.38 | hardwire | and he was in Iran |
23:37.41 | hardwire | and he did the same thing |
23:37.48 | hardwire | the newspaper salesman was pretty torqued |
23:37.49 | Cybertoy | my old phone company was nice enough to make an announcement on my previous phone number to call the new swiss did... |
23:37.52 | |Vulture| | wow... can't read that |
23:38.07 | Cybertoy | vulture, it's german... |
23:38.14 | hardwire | yerman |
23:38.20 | Cybertoy | vulture, but try www.netphone.ch ... there's a bunch of links there to others. |
23:38.50 | Cybertoy | sorry... that should be www.nettel.ch |
23:38.57 | hardwire | Cybertoy: no LNP? |
23:38.57 | |Vulture| | Cybertoy: okay cause I know one of my clients would love a swiss DID |
23:39.25 | Cybertoy | vulture, nettel lists a bunch of them... maybe one of them is in english ... |
23:39.41 | hardwire | engrish |
23:39.53 | shido | u don like my engrish, boi? |
23:40.00 | Cybertoy | vulture, if you want though I can talk you through setting up sipcall.ch ... |
23:40.11 | hardwire | ugh |
23:40.13 | hardwire | I need to lik |
23:40.15 | hardwire | fall over |
23:40.17 | hardwire | and sleep |
23:40.21 | |Vulture| | How do those voip providers send 911 different data? I thought that was via the hardware location |
23:40.47 | Cybertoy | now why would a swiss did provider worry about 911 ?? |
23:41.08 | |Vulture| | no I got a link from there |
23:41.15 | |Vulture| | and it was talking about 911 calls |
23:41.16 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
23:41.23 | |Vulture| | nikotel |
23:41.58 | Cybertoy | dunno them... |
23:42.07 | Cybertoy | use sipcall .. they'll give you a swiss did for free |
23:42.24 | |Vulture| | I use VPC and Asterlink |
23:42.33 | |Vulture| | but neither offer DIDs in remote areas |
23:44.07 | Cybertoy | what's vpc? |
23:44.10 | RoyK | ~seen inspired |
23:44.11 | jbot | inspired <mikael@213.197.167.61> was last seen on IRC in channel #asterisk, 8d 10h 18m 50s ago, saying: 'if it's made at the wood alcohol factory, you'll go blind ;)'. |
23:44.15 | |Vulture| | voicepulse connect |
23:44.18 | Cybertoy | ah |
23:45.09 | Cybertoy | well ... the offer stands... I can guide you through sipcall if you want. the setup is free. no need for credit-card .. nothing ... |
23:45.48 | |Vulture| | nice, thanx Ill see if he wants it right now he can just dial but I am sure he will want to have a DID |
23:46.22 | Cybertoy | that's what you get |
23:47.52 | Cybertoy | www.voipbuster.com is nice to make free phone calls in usa and switzerland ... |
23:48.07 | Cybertoy | very good quality. |
23:53.57 | bkw__ | any asterlink customers here? |
23:54.03 | SwK | no |
23:54.05 | SwK | asterlink sucks |
23:54.12 | SwK | hah |
23:54.17 | nemisus | lol |
23:54.18 | bkw__ | thats bullshit.. NEXT!!!! |
23:54.19 | SwK | god damned register issues |
23:54.30 | SwK | </asshole> |
23:54.37 | SwK | really asterlink is a good provider |
23:54.37 | bkw__ | freebsd ain't my problem :P |
23:54.41 | bkw__ | haha |
23:54.45 | bkw__ | but we gotta fix it |
23:54.45 | SwK | they are just bsd clueless |
23:54.55 | bkw__ | no we are not.. asterisk is |
23:55.02 | bkw__ | me luvs BSD |
23:55.02 | SwK | heh |
23:55.04 | bkw__ | :P |
23:55.05 | *** join/#asterisk calvinhp (~calvinhp@cpe-65-29-88-222.indy.res.rr.com) |
23:55.25 | Cybertoy | I thought there's a port for openbsd for asterisk |
23:55.34 | Cybertoy | but probably an older version |
23:55.35 | bkw__ | it compiles on Mac OS X |
23:55.37 | SwK | bkw__ I wont tell anyone about your chuck the daemon butt plug |
23:55.39 | *** join/#asterisk R3DB0x (nobody@66.142.28.36) |
23:55.54 | SwK | Cybertoy: sometimes it will compile |
23:55.58 | hardwire | heh |
23:56.00 | file[desk] | bkw__: I can't call toll-frees d00d |
23:56.09 | SwK | Cybertoy just depends on the when you check out head |
23:56.22 | bkw__ | file[desk], i'll fix that in a few dear |
23:56.24 | bkw__ | :P |
23:56.29 | SwK | Cybertoy try it and lemme know id it doesnt we'll see what we can do to patch it up |
23:56.30 | hardwire | thats sweet |
23:56.52 | Cybertoy | I'm not gonna... I have a nice working installation on redhat linux. |
23:56.53 | |Vulture| | bkw_: yea |
23:56.57 | file[desk] | bkw__: my service doesn't work, *WHINE* :P |
23:56.59 | |Vulture| | bkw_: I am one |
23:58.42 | *** join/#asterisk dasuberdavid (~chibski@user-24-214-126-76.knology.net) |
23:58.56 | harryvv | We just had a major down poor and asked if my dog wanted to go out side he was exceited. he then ran anywhere he could to get away from the huge downpoor :) |
23:59.06 | bewest | has anyone got their sipura 3000 to get the pstn line to call * properly? |
23:59.18 | bewest | the only way I could manage it was to mangle the caller id stuff in sipura |
23:59.19 | |Vulture| | harryvv: where? |
23:59.41 | dasuberdavid | does anyone know what context a _9NXXXXXXX extensions should be in |
23:59.42 | dasuberdavid | ? |
23:59.55 | |Vulture| | the outbound one... |
23:59.58 | RoyK | ~lart himself |