| 00:01.19 | [TK]D-Fender | ManxPower: Wind FTW |
| 00:05.19 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 00:07.18 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 00:07.55 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-237-15.balt.east.verizon.net) |
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| 00:10.04 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 00:15.54 | echelon | how do i set it up so that anyone can call sip:asterik_ip and asterisk would ask them to enter an extension? |
| 00:20.14 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:22.47 | *** join/#asterisk isamar (i=1000@voice.maxirede.net) |
| 00:22.58 | isamar | h folks |
| 00:25.55 | lmadsen | j folks |
| 00:26.06 | echelon | anyone? |
| 00:26.42 | isamar | hi |
| 00:26.43 | lmadsen | echelon: allowguest=yes I think is the option |
| 00:26.46 | isamar | here |
| 00:26.48 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 00:27.08 | echelon | lmadsen: and it will ask for an extension? |
| 00:27.14 | lmadsen | no idea |
| 00:27.17 | echelon | hmm |
| 00:27.28 | echelon | guests are already able to call in |
| 00:28.52 | *** join/#asterisk goobsoft2 (n=chad@cpe-24-167-97-202.satx.res.rr.com) |
| 00:30.33 | echelon | also.. does anyone know what voip protocol optimum online uses? |
| 00:31.50 | goobsoft2 | When I read about echo cancelation in asterisk it seems to be always associated with a hardware card and real phone line, but I'd like to tweak the echo cancellation that runs on normal SIP calls. Does anyone know where I need to start? |
| 00:32.03 | *** join/#asterisk NemesisX11 (n=jtharpe@pool-68-238-54-108.port.east.verizon.net) |
| 00:33.16 | NemesisX11 | Hi all I have a question about asterisk. I have it running on Slackware 12, and I have it up and running, but when I join my zoiper soft phones to it they register and can call but then they unregister and I must restart the computers to get them to re-register again |
| 00:34.19 | echelon | NemesisX11: try ekiga |
| 00:34.56 | NemesisX11 | ill give it a shot and let you know thanks echelon |
| 00:35.46 | echelon | NemesisX11: are you using slackbuilds? |
| 00:36.10 | *** join/#asterisk MrNaz (n=naz@ppp121-44-249-224.lns4.mel4.internode.on.net) |
| 00:36.38 | NemesisX11 | i followed the tutorial on astrikast when I installed it |
| 00:36.55 | NemesisX11 | and they where running slackware also so i assume they installed the slackware packages |
| 00:37.26 | echelon | NemesisX11: i mean from http://www.slackbuilds.org ? |
| 00:38.00 | NemesisX11 | no i went right to slackware.com |
| 00:38.17 | echelon | slackware.com doesn't have asterisk packages |
| 00:40.34 | NemesisX11 | when i installed the box i installed slackware 12 and got all my packages from there sources (astrikast) but i have never been to slackbuilds.org |
| 00:40.45 | *** join/#asterisk l2cache (n=chatzill@179.190.204.68.cfl.res.rr.com) |
| 00:42.54 | [TK]D-Fender | goobsoft: There is no EC for SIP calls. EC needs to be done only at the PSTN endpoint |
| 00:43.33 | goobsoft | Do you think my nokia phone my be doing EC? If I just hold a pitch into the echo test, it gets quieter. |
| 00:43.47 | [TK]D-Fender | NemesisX11: You install * from source on Slackware (or any platform if you know what's good for you) |
| 00:44.03 | jeev | [TK]D-Fender, pump pump the jam, pump it up |
| 00:44.29 | [TK]D-Fender | jeebjust cam back from a 40 minute high-speed bike-ride through the woods. |
| 00:44.38 | [TK]D-Fender | jeev: I'm as pumped as I'm getting tonight :) |
| 00:44.44 | jeev | damn |
| 00:44.47 | jeev | man i need to go ride a bike too |
| 00:44.50 | jeev | i'm so out of shape |
| 00:44.58 | [TK]D-Fender | jeev: Round is a shape :) |
| 00:45.02 | jeev | i'm like 203 right now, should be 195 |
| 00:45.09 | jeev | i'm not round you dorkus maximus |
| 00:45.16 | jeev | you call 4'11" 203 lbs round? |
| 00:45.19 | jeev | :D hahaha |
| 00:45.21 | [TK]D-Fender | jeev: I'm almost 210 (eek!) and should be 190 myself... |
| 00:45.23 | jeev | if i were 4'11 man, i'd cut myself |
| 00:45.27 | jeev | how tall? |
| 00:45.36 | [TK]D-Fender | jeev: 6'3" |
| 00:45.38 | jeev | bastard |
| 00:45.39 | l2cache | I'm 230 |
| 00:45.40 | jeev | i'm 6'1 |
| 00:45.40 | echelon | [TK]D-Fender: so is there a way to call sip:asterisk_ip and have asterisk ask for an extension? |
| 00:45.46 | jeev | you must be a tig |
| 00:45.49 | l2cache | and 6' 5" |
| 00:45.59 | jeev | damn l2cache |
| 00:46.00 | [TK]D-Fender | echelon: Yes, though its unusual. |
| 00:46.06 | echelon | oh |
| 00:46.08 | jeev | why am i short!!! |
| 00:46.12 | l2cache | lol |
| 00:46.13 | jeev | i wish i were 6'4 |
| 00:46.18 | l2cache | I'm not fat, just really tall |
| 00:46.20 | echelon | but what can i google for? |
| 00:46.21 | [TK]D-Fender | echelon: these "directed" calls are flat-out not normal. |
| 00:46.48 | [TK]D-Fender | echelon: why are you sending calls to * without a target #? |
| 00:47.02 | [TK]D-Fender | jeev: blame your parents. |
| 00:47.08 | jeev | i'm taller than my dad! |
| 00:47.24 | echelon | to create a directory |
| 00:47.43 | [TK]D-Fender | NemesisX11: And follow the instructions in your source tarball. And while you're at it, go grab THE BOOK, and get reading |
| 00:47.44 | [TK]D-Fender | ~book |
| 00:47.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 00:47.46 | [TK]D-Fender | ^^^^^^^^^^^ |
| 00:47.53 | [TK]D-Fender | echelon: huh? |
| 00:48.36 | [TK]D-Fender | echelon: BRB |
| 00:48.43 | echelon | i dunno.. i just want asterisk to request an extension, and have the person dial it from the pad |
| 00:52.51 | [TK]D-Fender | echelon: why not dial it proerly in the first place? |
| 00:52.53 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0fca22cf987b5c04) |
| 00:56.47 | *** join/#asterisk makkksimal (n=makkksim@e177210249.adsl.alicedsl.de) |
| 01:00.27 | [TK]D-Fender | echelon: Though I've already told you how to catch their "null" dials. |
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| 01:07.36 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-237-15.balt.east.verizon.net) |
| 01:14.16 | jeev | Fender, today.. no static, perfect DTMF.. he said the server he moved me to is 1.4.18.1 |
| 01:14.47 | [TK]D-Fender | jeev: ? |
| 01:15.14 | jeev | heh |
| 01:15.26 | jeev | i think i'm gonna order 20 polycom's now :/ |
| 01:15.31 | jeev | i'm so nervous |
| 01:15.57 | unpaidbill | send me one |
| 01:16.08 | unpaidbill | it will make you more calm to be generous |
| 01:16.21 | unpaidbill | or: being generous will calm you |
| 01:16.34 | unpaidbill | i speaka day een glesh |
| 01:17.09 | jeev | haha |
| 01:17.33 | unpaidbill | you made a wise choice to stay away from cisco phones. |
| 01:17.34 | unpaidbill | i salute you. |
| 01:18.06 | jeev | unpaidbill, i will gladly give you one if you were to become my monkey |
| 01:18.08 | jeev | and do EVERYTHING for me |
| 01:18.23 | unpaidbill | give me one + salary and you have yourself a deal! |
| 01:18.37 | jeev | salary will be the cost of a polycom 330 per month |
| 01:18.51 | jeev | so $105/month + phone |
| 01:18.55 | unpaidbill | haha |
| 01:19.00 | jeev | i love you too |
| 01:19.00 | jeev | hahaha |
| 01:19.35 | unpaidbill | i think you'd be better off with fender |
| 01:19.51 | jeev | fender wanted half anyway |
| 01:19.54 | jeev | he said he'd deal with $50/month |
| 01:20.05 | jeev | damn, shaq called out kobe.. again. that's messed up, as much as i hate kobe, that's not cool |
| 01:20.32 | unpaidbill | kobe has nice marbling though |
| 01:20.50 | jeev | marbling ? |
| 01:20.57 | jeev | like marble in his home? |
| 01:21.09 | unpaidbill | no, marbling of the fat in the meat.. kobe beef |
| 01:22.35 | jeev | ahh |
| 01:22.39 | *** join/#asterisk joobie (n=joobie@119.11.70.136) |
| 01:23.00 | unpaidbill | http://upload.wikimedia.org/wikipedia/commons/d/d6/4_Kobe_Beef%2C_Kobe_Japan.jpg ! |
| 01:23.06 | jeev | damn |
| 01:23.13 | jeev | he's like, "kobe ratted me out, that's why im getting divorced" |
| 01:23.14 | jeev | damn |
| 01:23.32 | jeev | that's a lot of fat |
| 01:25.31 | jeev | so i'm going to pick this up: http://www.newegg.com/Product/Product.aspx?Item=N82E16833122177 |
| 01:25.33 | jeev | i guess |
| 01:25.49 | jeev | i dont need managed since i do dual wan, with source and destination based routing |
| 01:25.56 | jeev | i'll just push the voice out it's dedicated internet |
| 01:29.06 | [TK]D-Fender | jeev: I'd sooner go with a D-Link DES-1228P |
| 01:29.28 | [TK]D-Fender | jeev: Only Netgear I'd use would be their 8 port (4port POE) switch for small installs. |
| 01:29.45 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 01:29.52 | [TK]D-Fender | jeev: Netgear is almost univerally frowned upon, so I'd keep their use to a minimum |
| 01:30.10 | [TK]D-Fender | jeev: http://www.newegg.com/Product/Product.aspx?Item=N82E16833127228&Tpk=DES-1228P |
| 01:30.26 | [TK]D-Fender | jeev: AND, its 50$ cheaper :) |
| 01:30.42 | magic_hat | hey everyone. I"m trying to get speak_text to work with an agi application. Everything's going swimmingly, according to the console messages. but the audio is not playing. |
| 01:30.44 | magic_hat | AGI Script Executing Application: (AGI) Options: (swift.agi|Please enter your poll id, followed by the pound sign) |
| 01:31.03 | x86 | HP switches are great |
| 01:32.57 | jeev | damn |
| 01:33.01 | jeev | i wanted the ipod, i dunno why, i hate them |
| 01:33.52 | unpaidbill | why the dlink over the netgear, past bad experiences? |
| 01:34.17 | magic_hat | THis is my swift.agi file. test.wav is being created, but it's not playing. http://pastie.org/220856 Thoughts? |
| 01:34.27 | jeev | fender, that thing does vlan? heh.. thought it was unmanaged |
| 01:34.56 | unpaidbill | both do vlans.. they have a gmanagement gui |
| 01:35.26 | jeev | i dont want vlan |
| 01:35.29 | jeev | i want just regular switch |
| 01:35.34 | jeev | and i'll handle the routing through my bsd router :D |
| 01:35.41 | unpaidbill | heh, just because it has a feature doesnt mean you're going to use it. |
| 01:37.58 | *** join/#asterisk makkksimal (n=makkksim@e177210249.adsl.alicedsl.de) |
| 01:38.24 | jeev | fender, 24 Port 10/100Mbps + 4 1000BASE-T + 2 Combo Ports & PoE only 2 poe ports? |
| 01:39.03 | jeev | http://www.dlink.com/products/resource.asp?pid=541&rid=2075&sec=0 ajhh nevermind |
| 01:39.15 | [TK]D-Fender | unpaidbill: No personal bad experience with Netgear, just plenty of other complaints in here. And I see a lot of them being sold as refurb, what does that tell you? For basic use I've been very happy with D-Link PoE. Wouldn't touch their routers though. |
| 01:39.23 | unpaidbill | on the newegg page you have to look under 'features' jeev :P |
| 01:39.26 | NovceGuru | fwiw those Dlinks can work with cisco 79*0 G's with a special cable |
| 01:39.29 | jeev | blah blahb lah |
| 01:39.53 | jeev | oh yea, cat5e can carry power, right? you dont need cat6 or anythign different |
| 01:39.59 | [TK]D-Fender | NovceGuru: Yeah... but does it include the sacrifical goats you'll need to make the Cisco's work with *? ;) |
| 01:40.01 | jeev | yea, NovceGuru, i think anything shoudl work, no ? |
| 01:40.07 | x86 | jeev: right |
| 01:40.15 | unpaidbill | fender, good to know, i havent messed with much netgear or dlink stuff.. i usually stick to linksys or foundry |
| 01:40.15 | *** part/#asterisk drdrain (n=kimmyd@cpe-066-057-105-080.nc.res.rr.com) |
| 01:40.22 | magic_hat | anyone got some wisdom on my agi/speak_text situation? |
| 01:40.25 | x86 | NovceGuru: any PoE switch can power a cisco phone with the proper cable |
| 01:40.28 | NovceGuru | [TK]D-Fender: I don't think denmark has enough goats... |
| 01:40.37 | unpaidbill | at least i used to stick to linksys, before they ruined the wrt series |
| 01:40.47 | NovceGuru | jeev: not with the 7940/60gs that are based on the cisco poe standard |
| 01:41.09 | NovceGuru | I had to put a 20k resistor across the data lines to get it to power the phone |
| 01:41.12 | unpaidbill | my foundry switch handles 7960s without a problem |
| 01:41.28 | x86 | NovceGuru: yeah man just change the pinout you're good ;) |
| 01:41.37 | NovceGuru | Some work, some don't. It's the ones that stick to the standards that don't |
| 01:41.43 | NovceGuru | I did, no workie without the resitor |
| 01:41.46 | NovceGuru | resistor*( |
| 01:41.50 | unpaidbill | no shit |
| 01:41.53 | x86 | you're doing it wrong then ;) |
| 01:41.55 | unpaidbill | how much power are you putting to them |
| 01:41.57 | unpaidbill | inline power legacy-powerdevice configurepower 15400inline power legacy-powerdevice configurepower 15400 |
| 01:42.07 | unpaidbill | oops i pasted twice.. that's what i use |
| 01:42.12 | unpaidbill | 15400mW or whatever |
| 01:42.21 | NovceGuru | unpaidbill: whatever the dlink feeds it |
| 01:42.22 | *** join/#asterisk hsv-al (n=sdfsdf@user-24-214-126-81.knology.net) |
| 01:42.25 | hsv-al | . |
| 01:42.27 | unpaidbill | all the cisco phones are running with that here |
| 01:42.30 | x86 | always works for me using cat6 patch cables with the X bar separating the pairs |
| 01:42.37 | unpaidbill | novce you cant change it at all? |
| 01:43.06 | NovceGuru | unpaidbill: if the (particular model I have atleast) doesn't sense a load across the data pairs it won't give it pair |
| 01:43.18 | NovceGuru | part of the 802.11af standard |
| 01:43.21 | unpaidbill | ah that sucks |
| 01:43.38 | unpaidbill | that's why i have the 'legacy-powerdevice' option |
| 01:43.46 | unpaidbill | :( |
| 01:43.59 | NovceGuru | yeah, the web interface on this dlink was from 1982 |
| 01:44.03 | unpaidbill | haha |
| 01:44.12 | unpaidbill | gopher gui |
| 01:44.32 | NovceGuru | wouldn't work with firefox OR ie7 |
| 01:44.39 | unpaidbill | wow |
| 01:44.47 | unpaidbill | how the hell did they manage that.. did you check for a firmware update? |
| 01:45.21 | NovceGuru | yeah, wasn't any I don't think, let me see what model it was |
| 01:46.12 | magic_hat | agi & speak_text? anyone? |
| 01:46.17 | NovceGuru | des-1316 |
| 01:46.31 | NovceGuru | http://www.newegg.com/Product/Product.aspx?Item=N82E16817111036 |
| 01:47.39 | [TK]D-Fender | NovceGuru: I run 2 x DES-1526's at the office. |
| 01:48.20 | NovceGuru | I have no complaints, it's not it's fault it doesn't support legacy cisco stuff |
| 01:49.20 | unpaidbill | yeah no firmware updates since '05 |
| 01:49.21 | unpaidbill | wee |
| 01:49.22 | unpaidbill | haha |
| 01:50.52 | NovceGuru | I was glad I got it working because I scratched it to hell putting it in the stupid rack :P |
| 01:51.31 | unpaidbill | dont abuse your networking equipment, it will be vindictive |
| 01:52.14 | NovceGuru | It probably will...tomorrow |
| 01:52.17 | NovceGuru | it's 2k miles away :( |
| 01:57.51 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:57.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:04.47 | seanbright | start seeing a bunch of these before the crash, too: |
| 02:04.51 | seanbright | [Jun 23 22:04:24] WARNING[26009]: file.c:658 ast_readaudio_callback: Failed to write frame |
| 02:08.40 | echelon | what are some stun servers? |
| 02:09.28 | [TK]D-Fender | echelon: stun.fwdnet.net:3478 |
| 02:10.02 | *** join/#asterisk blinky42 (n=blinky42@c-71-230-47-244.hsd1.pa.comcast.net) |
| 02:12.59 | echelon | [TK]D-Fender: i meant stun server software |
| 02:14.23 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 02:14.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:19.36 | [TK]D-Fender | echelon: As in a stun SERVER application |
| 02:19.40 | [TK]D-Fender | ? |
| 02:19.42 | echelon | yes |
| 02:19.57 | [TK]D-Fender | echelon: Not sure. Why do you ask? |
| 02:20.24 | echelon | i want to know how nat-traversal works and apply it to VPN |
| 02:20.55 | [TK]D-Fender | echelon: If you're on VPN, isn't everything "local"? |
| 02:22.03 | echelon | [TK]D-Fender: yeah, but a port needs to be open on the vpn server for inbound connections |
| 02:22.32 | [TK]D-Fender | echelon: * has no need or support of STUN. |
| 02:22.56 | [TK]D-Fender | echelon: And ditto for its clients 99.9% of the time |
| 02:22.57 | echelon | [TK]D-Fender: i'm not referring to * |
| 02:23.23 | echelon | i'm going off topic |
| 02:24.42 | [TK]D-Fender | echelon: Stuff better researched on SER |
| 02:24.57 | echelon | ser? |
| 02:25.01 | [TK]D-Fender | echelon: And stun is still not needed... |
| 02:25.05 | [TK]D-Fender | ~ser |
| 02:25.05 | jbot | from memory, ser is Sip Express Router - see http://www.iptel.org/ser/, or an old secret method of obtaining a havoc of NAT problems, or at #ser |
| 02:25.18 | [TK]D-Fender | ~sipnat |
| 02:25.19 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:25.21 | [TK]D-Fender | ^^^^ |
| 02:25.27 | echelon | k thanks :) |
| 02:46.51 | lmadsen | i don't want to know your name! |
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| 03:01.00 | jeev | lmadsen, what song was that |
| 03:03.13 | *** join/#asterisk damjan (n=damjan@217.16.95.15) |
| 03:04.40 | *** join/#asterisk mamacito62 (n=XD@190.72.246.125) |
| 03:05.40 | [TK]D-Fender | lmadsen: I just want... |
| 03:05.55 | mamacito62 | hi for all |
| 03:06.08 | jeev | [TK]D-Fender, i just called.. through sip, to say i love you |
| 03:06.08 | mamacito62 | anyone can help me to config one account of net2phone |
| 03:07.07 | [TK]D-Fender | mamacito62: http://www.voip-info.org/wiki/view/Asterisk+settings+Net2phone |
| 03:07.15 | mamacito62 | not work |
| 03:07.26 | *** part/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 03:07.51 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 03:07.55 | jeev | har har |
| 03:07.56 | [TK]D-Fender | ... wtf |
| 03:08.30 | jeev | Fender, you know what grinds my gears? i spend all this time on the phone system, asterisk.. annoying everyone.. and when i etll my friend i'm ready to order the phones, he just ignores it. |
| 03:10.32 | [TK]D-Fender | jeev: Since I don't know the backstory, care to explain its relevence? |
| 03:10.54 | jeev | why am i wasting my damn time with this |
| 03:10.58 | jeev | if he's not gonna order the damn phones |
| 03:11.04 | jeev | i'm trying to save his ass money and he's putting me off |
| 03:12.21 | [TK]D-Fender | jeev: What kind of "savings" for what kind of "expense"? |
| 03:12.47 | jeev | i think they'r doing from 1400-1800 a month |
| 03:12.53 | jeev | compared to a 4k expensive |
| 03:12.58 | jeev | and 200/month probably after |
| 03:15.56 | *** join/#asterisk colulu (n=jgreen@58.251.75.157) |
| 03:17.09 | [TK]D-Fender | jeev: And where do the savings come from? |
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| 03:24.54 | TJNII | Anyone familiar with a Leich Electric Series 600? |
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| 03:33.44 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 03:33.45 | andrewy | can anyone recommend a python AGI library? all of them are outdated, the most recent being a year old (starpy) |
| 03:34.02 | andrewy | s/outdated/appear to be unmaintained |
| 03:35.14 | ManxPower | andrewy: The AGI API has not changed in years, an old library is fine |
| 03:36.23 | andrewy | right, but it's still suspicious that no bugs have been found in 3+ years. i guess i'm trying to get an idea of what other people use |
| 03:37.38 | lmadsen | [TK]D-Fender: ! ! ! |
| 03:38.44 | [TK]D-Fender | andrewy: AGI's spec is pretty much the same. How wrong can you BE with such a limited interface? |
| 03:39.39 | andrewy | yeah, i suppose everything's simple enough for the agi libs not to need maintenance |
| 03:49.34 | jeev | fender, what do you mean, from the better rate i get with voip! |
| 03:51.21 | [TK]D-Fender | jeev: Who said you needed to replace everything to achieve that gain? |
| 03:52.24 | jeev | well, it'd be cool to keep the existing nortel phones.. but it's a massive headache |
| 03:52.45 | [TK]D-Fender | jeev: how many lines do you have now, and what kind? |
| 03:52.54 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.137) |
| 03:53.01 | jeev | i think that there are at least 16 lines in 2 offices, they're consolidating offices, moving to a single building |
| 03:53.06 | jeev | and 24 phones |
| 03:53.12 | jeev | on 16 #'s |
| 03:53.38 | [TK]D-Fender | jeev: And you've got an internet conenction that would survive that kind of usage? |
| 03:53.42 | jeev | yea |
| 03:53.44 | jeev | they dont use it often |
| 03:53.51 | jeev | trust me, and it's 99% outgoing. |
| 03:53.59 | [TK]D-Fender | jeev: .... ok, because that would max out a T1. |
| 03:54.09 | jeev | if all were used, correct? probably 1-3 at a time, MAX. |
| 03:54.20 | [TK]D-Fender | jeev: So why 16 lines NOW? |
| 03:54.29 | jeev | only time all would be used is if it was guaranteed that a comet would be hitting the earth.. and they'd be calling loved ones, these lazies dont work! |
| 03:54.36 | jeev | Fender, i dunno, i didn't know them when they got it |
| 03:54.52 | jeev | Fender, i think i'll do 4 DID's each |
| 03:54.59 | [TK]D-Fender | jeev: Guess nobody's been watching their needs. |
| 03:55.11 | [TK]D-Fender | jeev: 4 DID's each? |
| 03:55.20 | jeev | aint my job ;D |
| 03:55.20 | jeev | yea |
| 03:55.59 | jeev | all the terms i say are wrong, why, DID is wrong too? |
| 03:56.12 | [TK]D-Fender | jeev: explain what you mean and we'll see |
| 03:56.19 | jeev | bastard, why put me on the spot |
| 03:56.25 | jeev | 4 incoming numbers for each business. |
| 03:56.33 | jeev | so it's 2 businesses, i'll get 8 #'s |
| 03:56.40 | *** join/#asterisk gitguy (n=diego@adsl-137-142.click.com.py) |
| 03:56.45 | jeev | sup gitguy |
| 03:57.00 | [TK]D-Fender | jeev: If you can't explain your use of a word, what right do you have flinging it around? Thats escalating from a wrong idea to "no clue" immediately :p |
| 03:57.03 | gitguy | hey jeev |
| 03:57.09 | *** join/#asterisk BeeBuu (n=beebuu@59.38.97.192) |
| 03:57.30 | gitguy | jeev: hows your asterisk installation going |
| 03:57.32 | jeev | fender, what was i saying wrong then |
| 03:57.34 | jeev | it's goin great gitmo |
| 03:57.41 | gitguy | nice |
| 03:57.47 | [TK]D-Fender | jeev: Ok, fine, sure. so 8 #'s, some arbitrary # of fixed channels (or pay/min) |
| 03:57.56 | jeev | pay per min |
| 03:58.16 | [TK]D-Fender | jeev: Your business is primarily outbound LD? |
| 03:58.22 | jeev | yes sir |
| 03:58.33 | [TK]D-Fender | jeev: makes sense then. |
| 03:58.45 | jeev | Fender, although there are 3 fax lines doing 1000 pages per day.. |
| 03:58.49 | jeev | what you suggest for that;/ hard line ? |
| 03:58.53 | jeev | those are the most important |
| 03:59.31 | [TK]D-Fender | jeev: yes, hard line for sure. |
| 03:59.46 | [TK]D-Fender | jeev: As far away from * as possible. |
| 03:59.47 | jeev | ok |
| 03:59.51 | jeev | really |
| 03:59.52 | jeev | ok |
| 04:01.24 | echelon | can asterisk run on a machine with 24mb of ram and 152mhz cpu? |
| 04:01.58 | gitguy | i'm sure it will run, but i'm not sure how well it will perform |
| 04:02.02 | jer | barely, and only if you aren't doing any transcoding |
| 04:02.07 | jeev | Fender, where'd you go |
| 04:02.13 | [TK]D-Fender | and passing maybe 2 calls tops |
| 04:02.15 | [TK]D-Fender | :p |
| 04:02.24 | jeev | oh dood, the box i have it on has 128 mb ram ; |
| 04:02.28 | echelon | oh.. was going to use it for conferencing :\ |
| 04:02.42 | jeev | ultimately fender, for maximum stability.. does it have better support on linux or bsd? |
| 04:02.46 | [TK]D-Fender | echelon: get me some crack too while you're at it... |
| 04:02.53 | jeev | cause i'm gonna have it on a sexy server |
| 04:03.06 | echelon | :P |
| 04:03.08 | [TK]D-Fender | jeev: either will do, but Linux is better supported. |
| 04:03.17 | gitguy | Linux++ |
| 04:03.41 | axisys | is there any other site like misterhouse.. it seems to be not updated since last yr |
| 04:03.46 | jeev | yea, i guses i'll slap linux on to it |
| 04:03.53 | jeev | should i even bother with raid |
| 04:03.57 | jeev | or do a nightly tarball |
| 04:04.17 | [TK]D-Fender | axisys: I use(d) "heyu2" with my setup |
| 04:04.35 | [TK]D-Fender | jeev: Yes, raid, and yes, back it up too |
| 04:04.49 | jeev | crap |
| 04:04.51 | gitguy | i just got these two t-shirts http://www.thinkgeek.com/tshirts/itdepartment/59ce/ http://www.thinkgeek.com/tshirts/itdepartment/280d/ |
| 04:05.02 | jeev | Fender, would you laugh at me if i dont wanna build anythign and just get them a Q6600 dell box? :D |
| 04:05.05 | gitguy | it rocks |
| 04:05.15 | jeev | gitguy, welcome to 1997! |
| 04:05.16 | jeev | haha jk |
| 04:05.43 | gitguy | linux is the present and future ;) |
| 04:06.04 | gitguy | these t-shirts are a classic though |
| 04:06.05 | gitguy | :P |
| 04:06.30 | axisys | [TK]D-Fender: thnx |
| 04:06.35 | Nugget | Linux is poo. |
| 04:06.47 | gitguy | not |
| 04:08.09 | [TK]D-Fender | ... telnet |
| 04:08.13 | axisys | [TK]D-Fender: is there any other open source project that works on smart home setup? |
| 04:08.23 | [TK]D-Fender | darn... bot's fired off too recently :p |
| 04:08.41 | [TK]D-Fender | axisys: there were several, all old by now I'm sure. |
| 04:08.52 | [TK]D-Fender | axisys: What do you want to do? |
| 04:09.19 | axisys | [TK]D-Fender: why? people lost interest? no just look really fun.. may be slow get into it.. once get the asterisk lined up |
| 04:10.04 | axisys | [TK]D-Fender: mainly looking for way to save energy.. using many different matrixs .. |
| 04:10.18 | [TK]D-Fender | axisys: ok, any of these might do then. |
| 04:10.21 | axisys | and also make the home secure .. just for fun really |
| 04:10.23 | [TK]D-Fender | axisys: Get googling. |
| 04:10.47 | axisys | [TK]D-Fender: smart home gave me all commercial products.. |
| 04:11.06 | [TK]D-Fender | axisys: you aren't being specific enough. |
| 04:12.34 | axisys | [TK]D-Fender: i know.. lets see.. how to save energy .. that means turn the chiller on when it is too hot or when I am on my way home .. may be turn it on 1 hr before i get home.. |
| 04:12.49 | [TK]D-Fender | axisys: I meant your SEARCH. |
| 04:12.56 | axisys | [TK]D-Fender: oh |
| 04:13.25 | jeev | i want to play sim copter |
| 04:13.29 | jeev | we need to port it to multiplayer |
| 04:14.50 | *** part/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
| 04:18.07 | jeev | damn, freenode rules |
| 04:18.36 | *** join/#asterisk joobie (n=joobie@119.11.68.8) |
| 04:19.50 | gitguy | yeah |
| 04:24.48 | *** join/#asterisk mosty (n=mosty@ppp191-34.static.internode.on.net) |
| 04:25.33 | colulu | TK: hi |
| 04:25.41 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 04:26.41 | jeev | im' convinced someone pays fender 500k a year to have him put up with my shit |
| 04:27.17 | gitguy | lol |
| 04:29.57 | gitguy | some people do it for the love :p |
| 04:30.10 | jeev | i dont know how he could survive me |
| 04:30.14 | jeev | i think he's making a jeev doll to stab |
| 04:32.49 | *** join/#asterisk [cfdisk] (n=cfdisk@68-116-156-85.dhcp.ftwo.tx.charter.com) |
| 04:34.36 | TJNII | Stab? I'd think voodoo would be more likely. |
| 04:34.39 | [TK]D-Fender | goes back to his effigy mass-production run... |
| 04:34.57 | jeev | yea |
| 04:35.04 | jeev | vodoo |
| 04:35.06 | jeev | that's what i meant |
| 04:35.08 | jeev | voodoo banshee mofo's |
| 04:35.25 | jeev | goes back to amavisd on his new postfix server |
| 04:36.56 | TJNII | goes back to trying to figure out how a phone made before WWII whould be wired |
| 04:38.06 | TJNII | If the ringer wire wasn't floating this would be a lot easier..... |
| 04:38.17 | jeev | lol |
| 04:38.37 | jeev | i wonder if listening to R Kelly I believe i could fly makes me as bad as him |
| 04:46.13 | gitguy | lol |
| 04:49.59 | [TK]D-Fender | loads jeev into his trusty trebuchet and haelp him achieve his dreams. |
| 05:01.21 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:10.35 | gitguy | is there release date for 1.6.0? |
| 05:11.34 | gitguy | yawns |
| 05:12.57 | *** join/#asterisk dmz (n=dmz@ip67-94-22-68.z22-94-67.customer.algx.net) |
| 05:13.47 | [TK]D-Fender | gitguy: "When its done |
| 05:14.03 | gitguy | heh |
| 05:14.43 | TJNII | Well, my ATA didn't smoke, it rings, and I can talk on it. I guess I got it right. |
| 05:14.55 | TJNII | Though there is an unused coil in the hybrid, but meh. |
| 05:24.13 | jeev | heh |
| 05:24.21 | jeev | i hate the media and everyone else |
| 05:24.33 | jeev | people start telling the government, speculation is causing oil to rise, DUH |
| 05:24.44 | jeev | then people complain about not enough supply and it's not speculation |
| 05:24.54 | jeev | AND THEN, something is attacked, speculation calls it to go up |
| 05:24.57 | jeev | jesus christ how people are dumb |
| 05:26.57 | gitguy | lol |
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| 05:39.06 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 05:43.02 | slavon_net | hello all.. how use last 1.6.0 svn brunch with zap channels? svn 1.6.0 brunch update all code to DAHDI support but DAHDI not released... maybe DAHDI support need add only to trunk and create new brunch like 1.6.1 that support DAHDI and keep 1.6.0 for bugfixes? |
| 05:46.46 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 05:47.34 | ManxPower | did you try zapte.? |
| 06:00.11 | mosty | i'm trying to figure out how to use PICKUPMARK with app_directed_pickup- what channel variable does this look in? |
| 06:04.41 | *** join/#asterisk af_ (n=getsmart@88-149-230-105.dynamic.ngi.it) |
| 06:12.04 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
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| 06:47.40 | mosty | if i set a global variable PICKUPMARK=102 then my Pickup(102@PICKUPMARK) is able to pickup SIP/102 which is ringing. if i remove the global variable and set a channel variable __PICKUPMARK=102 in sip.conf for SIP/102 then i can't pick up this call. what am i doing wrong? |
| 06:55.27 | TJNII | SIP call picking is buggy |
| 06:56.22 | TJNII | But why would you want to set pickupmark in sip.conf? By its nature it needs to be dynamic. |
| 06:59.22 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 06:59.25 | mosty | my sip.conf is written by a program |
| 06:59.48 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 06:59.52 | mosty | if SIP/102 is ringing, i want to be able to pick it up no matter which extension it's ringing in |
| 07:02.34 | TJNII | I'm 99% sure that pickupmark needs to be set in the dialplan |
| 07:02.55 | TJNII | Pickupmark is really a kludge; we shouldn't need it at all |
| 07:03.09 | mosty | how can i do it without PICKUPMARK? |
| 07:03.15 | TJNII | But for whatever reason call picking doesn't like SIP. |
| 07:03.22 | TJNII | Until they fix the bug I don't know. |
| 07:03.50 | TJNII | It might be fixed now, I haven't researched it in a couple months. |
| 07:08.01 | *** join/#asterisk egypcio (n=korn@unaffiliated/egypcio) |
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| 07:09.32 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:09.34 | Chris-NB | hi |
| 07:10.05 | Chris-NB | anyone using a kirk wireless server 600v3 SIP with asterisk? |
| 07:10.28 | Chris-NB | my problem is, the dect phones can't register when there is a secret set for this user |
| 07:10.34 | jql | hmm |
| 07:10.45 | Chris-NB | if i comment the secret, everything is fine |
| 07:10.53 | Chris-NB | anyone had this problem? |
| 07:13.23 | mosty | is app_directed_pickup related to pickupexten in res_features? or are they completely different implementations of similar things? |
| 07:15.22 | Chris-NB | noone had these problems? |
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| 07:17.03 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:27.38 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 07:45.57 | *** join/#asterisk J4zen (n=Jeroen@a82-95-153-17.adsl.xs4all.nl) |
| 07:47.43 | J4zen | Hi there, a little while ago i read an article(review) about a Polycom desktop office phone, it returned the most reliable, has a color screen, voicemail indicator that works. if i remember correctly it was on voip-info.org. Does anyone remember a review like that? |
| 07:48.17 | J4zen | Or can someone recommend me a good solid business Sip phone, i'm currently using SNOM320's but i'm not too happy about them |
| 07:48.48 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:48.48 | mosty | polycom phones are ok, so long as you have a provisioning server and never ever waste your time using the polycom phone web interface |
| 07:49.18 | *** join/#asterisk RoyK (n=roy@box36.fortel.no) |
| 07:49.43 | J4zen | so i was told, we do have a provisioning server available |
| 07:49.48 | J4zen | what's your thoughts on the SoundPoint® IP 670 ? |
| 07:50.04 | unpaidbill | prrovisioning server? they dont just grab a dhcp address and a file from tftp? |
| 07:50.05 | J4zen | http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip670.html <-- |
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| 07:58.05 | J4zen | Can anyone recommend a SIP-phone in the price range of 150~300$ which doesn't require a seperate power-adapter? |
| 07:58.50 | jql | so... better than a grandstream? heh |
| 07:58.56 | mosty | unpaidbill, tftp or http, ftp or https. but you have to create all the settings files |
| 07:59.51 | J4zen | jql: I don't follow? |
| 08:01.19 | Maliuta | J4zen: you want something POE? |
| 08:01.28 | J4zen | yes |
| 08:01.37 | Maliuta | J4zen: get a cisco |
| 08:01.57 | unpaidbill | oh, yeah |
| 08:02.00 | unpaidbill | that's just like cisco phones |
| 08:02.03 | unpaidbill | except cisco phones suck |
| 08:02.07 | unpaidbill | suck big dongs |
| 08:02.11 | unpaidbill | big hairy dongs |
| 08:02.32 | unpaidbill | the SIP software for cisco phones is junk |
| 08:02.41 | J4zen | Ok so, other than Cisco. No manufacturer offers this AND manages to provide a solid phone? |
| 08:02.56 | unpaidbill | snom |
| 08:02.59 | unpaidbill | polycom |
| 08:03.11 | unpaidbill | they all do poe and you can get phones as cheap as 110 bucks |
| 08:03.18 | unpaidbill | sometimes cheaper (with snom especially) |
| 08:03.21 | jql | poe is everywhere |
| 08:03.31 | J4zen | didn't like SNOM too much |
| 08:03.40 | unpaidbill | cisco phones have a terrible UI, absolutely terrible, for SIP |
| 08:03.47 | J4zen | i see |
| 08:03.49 | unpaidbill | the skinny ui is ok |
| 08:04.04 | mosty | what don't you like about snom? maybe that will give us something to base suggestions on |
| 08:04.04 | unpaidbill | they gimped out the SIP stuff to be jerks. |
| 08:04.35 | unpaidbill | so, if you dont mind using chan_skinny or chan_sccp, cisco phones are OK. if you want to use sip, get something else is my advice |
| 08:04.38 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 08:04.47 | unpaidbill | but my advice normally sucks |
| 08:05.23 | J4zen | mosty: Basically it behaved extremely buggy while undergoing firmware upgrades, support from SNOM is terrible in my expierence and the extension modules generally look a bit cheap in my opinion |
| 08:06.00 | unpaidbill | polycom is the sexiest looking |
| 08:06.02 | J4zen | that along with several malfunctioning phones that had to be returned to our supplier |
| 08:06.03 | mosty | J4zen, i do lots of snom firmware upgrades- are you using the auto update URL or direct links to firmware files? |
| 08:06.14 | J4zen | auto-update URL |
| 08:06.29 | unpaidbill | http://www.newegg.com/Product/Product.aspx?Item=N82E16876129005 polycom IP 550 |
| 08:06.33 | unpaidbill | whistles |
| 08:07.02 | J4zen | part of the blame goes to the supplier as well i suppose, they delivered two different "series" of the 320's |
| 08:07.13 | J4zen | one with soft key-buttons, the other with hard edgy ones |
| 08:07.22 | J4zen | both delivered with different firmware versions |
| 08:07.45 | unpaidbill | http://www.newegg.com/Product/Product.aspx?Item=N82E16876129004 and that, a nice cheapy, 129 on newegg, most certainly cheaper anywhere else |
| 08:07.50 | unpaidbill | god they look nice. |
| 08:08.03 | J4zen | i was actually looking at the 670's |
| 08:08.04 | mosty | J4zen, the auto update URL works great here, you just have to leave the phones for like 10 minutes if it's starting from a really old firmware version |
| 08:08.18 | unpaidbill | 390 bucks for those things |
| 08:08.18 | unpaidbill | eesh |
| 08:08.26 | J4zen | aye, but very appealing ;) |
| 08:08.32 | unpaidbill | indeed, i would love one on my desk |
| 08:08.37 | unpaidbill | if only i could justify it to mr bossman |
| 08:09.05 | J4zen | mosty: i've tried, i had one of my phones completely hang in the update process even |
| 08:09.43 | mosty | J4zen, strange- i've never had that and i've done over a hundred of them |
| 08:09.48 | J4zen | it might have been a bad batch or so, since they obviously came from two different release batches |
| 08:09.59 | J4zen | what kind do you have? |
| 08:10.01 | J4zen | hard key-buttons |
| 08:10.04 | J4zen | or soft rubber ones |
| 08:10.35 | J4zen | might be in that, older versions of the 320 |
| 08:10.36 | mosty | i've done both. the hard button versions are newer |
| 08:10.57 | J4zen | ah |
| 08:11.16 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 08:11.59 | unpaidbill | sweet host dude |
| 08:12.26 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 08:12.38 | unpaidbill | oh friggen sweet, that 670 has a usb port |
| 08:12.58 | unpaidbill | although it doesnt do much, i like it. |
| 08:17.25 | J4zen | ye it looks awesome |
| 08:18.05 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 08:18.24 | J4zen | can you recommend an VoIP supplier in USA? |
| 08:18.29 | J4zen | or that operates in USD ;) |
| 08:21.16 | *** join/#asterisk lupino3 (n=andrea@217-133-45-108.b2b.tiscali.it) |
| 08:21.33 | Maliuta | J4zen: business or residential? |
| 08:21.41 | J4zen | business |
| 08:21.43 | lupino3 | hello, are there any known problems with Asterisk 1.4.19 and realtime voicemail? |
| 08:22.01 | lupino3 | CLI voicemail show users says there are no users defined |
| 08:22.18 | lupino3 | but I've configured them |
| 08:22.38 | J4zen | sorry, i should have been more clear |
| 08:22.59 | J4zen | i'm refering to a VoIP hardware supplier such as SIP-phones and additional asterisk Hardware such as PRI/BRI interfaces |
| 08:23.20 | Strom_C | J4zen: telephonydepot.com |
| 08:25.49 | J4zen | Strom_C: Thanks :) |
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| 08:35.26 | BeeBuu | ~~altitude |
| 08:36.43 | redax | hi, |
| 08:42.05 | redax | what is the usual registration expire time for lan sip phones? 1hour should work? |
| 08:42.42 | redax | or should I specify longer exiration? |
| 08:43.48 | mosty | it doesn't really matter, it's not much traffic |
| 08:44.06 | mosty | 30 mins? |
| 08:45.17 | redax | heh. somehow it matters ;-) I specified 120secs and our ISP link went down, so my asterisk played timeouts with 10 SIP trunks, so all of the extensions went offline ;-) |
| 08:45.29 | redax | ok, that's my bad... |
| 08:46.07 | redax | just wanted to ask what ppl use to set for register expiration... |
| 08:46.13 | mosty | at 120s it will just fill up your logs |
| 08:46.13 | redax | now I added 3600sec |
| 08:46.26 | mosty | i just leave the default |
| 08:46.54 | redax | yah. not mention the problem when internet link is down, and you have SIP trunks ;-) |
| 08:47.36 | redax | anyhow the default was 3600s |
| 08:48.13 | redax | ok, thanks |
| 08:48.17 | mosty | what does your internet connection matter to your lan sip clients? |
| 08:48.38 | mosty | they should still register to your pbx fine, you just won't be able to call out via the internet |
| 08:49.06 | redax | seems like, asterisk played (waiting for) sip trunk registrations, and meanwhile lan clients can't reregister themselves |
| 08:49.33 | redax | the default sip registrationtimeout is 20sec, I have 10SIP trunks, that's 200sec |
| 08:49.41 | mosty | that doesn't sound right at all |
| 08:49.44 | redax | the register expire was 120s :) |
| 08:49.48 | *** part/#asterisk BeeBuu (n=beebuu@59.38.97.192) |
| 08:50.09 | mosty | upstream sip registrations shouldn't stop sip clients from registering |
| 08:50.23 | mosty | you must have had lan problems at the same time |
| 08:51.00 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 08:51.46 | redax | Net gone at 3am, I arrived to the office at 7:30am and almost all of the lan clients was offline |
| 08:51.58 | *** join/#asterisk MrNaz (n=naz@124-168-81-27.dyn.iinet.net.au) |
| 08:52.19 | redax | almost == ~83 sip phones |
| 08:55.08 | mosty | the only way that might happen was if you had a sip deadlock, i haven't seen that with so few sip clients |
| 08:56.22 | redax | is there a way to specify different port for trunk registration than 5060, meanwhile the clients should register at 5060 |
| 08:56.36 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:59.09 | redax | hm. maybe DNS requests caused the sip deadlock... |
| 08:59.50 | redax | this box doesn't have local DNS server. and the specified is via the net conection |
| 09:02.48 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
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| 10:52.39 | cfh | hi all , i have succesfully configured distinctive ring with asterisk 1.4 and polycom 601 with alert info, but if i try with a polycom 550 doesnt work |
| 10:53.09 | cfh | are there different configuration with the new polycom to works with asterisk ? |
| 10:53.21 | d-k-t-2 | Anyone got any ideas for what I should do with external calls from someone who'd been dialling all numbers, leaving calls open to conference lines, trying to get employees to give them more numbers for other offices etc? Just rejecting their call could work, but it doesn't seem enough |
| 11:10.57 | *** join/#asterisk LuisTorres (n=chatzill@bl6-198-54.dsl.telepac.pt) |
| 11:19.25 | TheH | Hi guys anybody knows why X-lite does not register with MAC OS TIGER |
| 11:19.39 | TheH | not even incoming registrations are shown on asterisk |
| 11:28.12 | *** join/#asterisk ming_zym (n=ming_zym@222.128.39.136) |
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| 11:52.42 | Dave_K | *Phew* Java IRC is not friendly. Hi there folks. |
| 12:01.50 | *** join/#asterisk blinky42 (n=blinky42@64.9.15.189) |
| 12:03.19 | tgoodchild | Hi, perhaps someone has an idea... |
| 12:03.41 | tgoodchild | I'd like to use my phones (Snom) in a "Chef-Secretary" way |
| 12:04.57 | tgoodchild | Such that whenever the "Chef" picks up the phone (also when he's going to make a call) the secretary can see that he's using it |
| 12:05.02 | tgoodchild | Something like blf |
| 12:17.43 | tgoodchild | hum... ideas? |
| 12:19.49 | *** join/#asterisk stevie_ramjet (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 12:19.49 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
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| 12:33.46 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:33.46 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:34.21 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 12:37.58 | *** join/#asterisk oilinki (n=oil@ppp-124-121-249-197.revip2.asianet.co.th) |
| 12:38.32 | lmadsen | morning all |
| 12:38.52 | mvanbaak | yo lmadsen |
| 12:41.09 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 12:42.23 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:45.18 | *** join/#asterisk quazzmarsh (n=quazzmar@62.8.93.2) |
| 12:46.47 | lmadsen | how goes mvanbaak? |
| 12:47.31 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 12:47.32 | mvanbaak | good good |
| 12:47.38 | Mike8861 | hello everyone!!!! |
| 12:47.38 | mvanbaak | fixing bugs in our PHP application |
| 12:47.56 | Mike8861 | so glad to see all asterisk professional! |
| 12:48.23 | x86 | morning |
| 12:48.26 | x86 | heya lmadsen |
| 12:48.26 | tgoodchild | hi |
| 12:48.48 | lmadsen | Mike8861: oh oh oh.... we're not professionals |
| 12:48.54 | lmadsen | x86: howdy! |
| 12:49.03 | Mike8861 | heehe, whatever |
| 12:49.12 | lmadsen | mvanbaak: I just woke up :) |
| 12:49.17 | Mike8861 | you guy rules |
| 12:49.19 | lmadsen | cleaned the kitchen... thinking about breakfast |
| 12:49.26 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 12:49.30 | Mike8861 | i just spent many many rimes to enter asterisk IRC |
| 12:49.34 | Mike8861 | gotta headache |
| 12:49.36 | mvanbaak | lmadsen: damn, it's almost 3 PM here |
| 12:50.04 | lmadsen | mvanbaak: you've been busy for many an hour while I was sleeping, heh |
| 12:50.12 | mvanbaak | only on sundays I wake up around 3 PM |
| 12:50.23 | mvanbaak | normal days it's 6 AM |
| 12:50.32 | x86 | lmadsen: what's new? how's business? |
| 12:50.50 | lmadsen | mvanbaak: wow... I never wake up that late... almost always up by 8:30am even on weekends |
| 12:51.14 | lmadsen | x86: business is good... lots to do today... did 10 billable hours yesterday... which is a lot in one day |
| 12:51.19 | mvanbaak | lmadsen: you have kids ? |
| 12:51.22 | lmadsen | (at least for me -- about double) |
| 12:51.33 | lmadsen | mvanbaak: no kids, no animals, no wives, no girlfriends :) |
| 12:51.39 | mvanbaak | ah |
| 12:51.52 | lmadsen | livin' the single life |
| 12:52.05 | mvanbaak | well, on saturday I stay up pretty early. around 8 or 9 AM |
| 12:52.09 | lmadsen | although I do have a female friend of mine stop by about once a week |
| 12:52.16 | mvanbaak | but saturdays go on till 4 or sometimes 6 AM |
| 12:52.16 | lmadsen | mvanbaak: ahhh, well that makes sense then |
| 12:52.22 | x86 | lmadsen: 10 billable that's awesome |
| 12:52.26 | mvanbaak | never a saturday that ends before 4 AM |
| 12:52.33 | x86 | lmadsen: what do you charge an hour? $125 or so? |
| 12:52.37 | lmadsen | I used to be a night owl like that, but have gotten out of it |
| 12:52.41 | *** join/#asterisk oilinki3 (n=oil@ppp-124-121-249-197.revip2.asianet.co.th) |
| 12:52.45 | lmadsen | x86: not quite |
| 12:52.49 | Mike8861 | hello oilinki3 |
| 12:52.51 | lmadsen | x86: although I wish I was ;) |
| 12:53.01 | x86 | lmadsen: that's the going rate around here :P |
| 12:53.08 | Mike8861 | well, I got question on asterisk , can anyone help |
| 12:53.27 | lmadsen | ya I know... but I charge less because the company I work for gives me lots of hours |
| 12:53.33 | lmadsen | ~ask |
| 12:53.34 | jbot | ask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:53.51 | lmadsen | is against his will :) |
| 12:54.24 | *** join/#asterisk dieno (i=771e7892@gateway/web/ajax/mibbit.com/x-874ace33a1ae5327) |
| 12:54.31 | lmadsen | x86: I usually throw out the $200/hr to someone when I don't really want to do the project :) |
| 12:54.49 | Mike8861 | ok, get ready?! |
| 12:55.14 | dieno | does any one have idea how to install app_swift on Asterisk 1.4.21 with CentOS 5.0 |
| 12:55.22 | Mike8861 | We are running Trixbox 2.4.1, install from ISO image, is there any way to add IM function to this ? |
| 12:55.32 | tgoodchild | Question: I'd like to use my phones in a "Chef-Secretary" way, such that whenever the "Chef" picks up the phone (not only receiving, but making calls) the secretary can see that he's using it... Sth. like blf? |
| 12:55.36 | Mike8861 | i saw IM patch for asterisk, but donno how to install |
| 12:57.27 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:57.40 | Mike8861 | morning jaytee |
| 12:57.53 | jaytee | morning |
| 12:58.14 | mvanbaak | Mike8861: ask in trixbox channel |
| 12:58.18 | mvanbaak | ~trixbox |
| 12:58.18 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 12:58.57 | Mike8861 | thanks for all da help |
| 12:59.00 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 12:59.05 | Mike8861 | cya all |
| 12:59.07 | *** join/#asterisk fenlander (n=fenlande@80.169.36.194) |
| 12:59.13 | lmadsen | lates |
| 13:01.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:03.00 | x86 | 07:54 < Mike8861> ok, get ready?! |
| 13:03.07 | x86 | that's when I stopped paying attention to him ;) |
| 13:03.17 | Mike8861 | yup.thanks for helping |
| 13:03.24 | x86 | lmadsen: what's your hourly rate to a new client |
| 13:03.41 | x86 | lmadsen: not someone gauranteeing you a fixed amount of hours |
| 13:03.42 | lmadsen | x86: depends on the project, expected number of hours, and who's asking :) |
| 13:03.47 | x86 | ah |
| 13:03.59 | x86 | so you don't have a fixed price schedule |
| 13:04.08 | lmadsen | x86: no guarantee of hours usually means about $200/hr, but I don't really take one-off projects |
| 13:04.12 | lmadsen | I'm too busy as it is |
| 13:04.30 | lmadsen | I haven't really taken on new clients for 2 years now |
| 13:04.55 | x86 | man that's awesome |
| 13:05.16 | lmadsen | I work with new people all the time, but I do it all through one company. They do the billing, bill collecting, etc... so I don't have to worry about all the overhead |
| 13:05.21 | x86 | of course... when someone asks you for references, all you have to do is say "uh... me?" ;) |
| 13:05.29 | lmadsen | I let the sales people do their job, and then I can just focus on the technology |
| 13:05.37 | x86 | oh that's rad |
| 13:05.39 | lmadsen | x86: I just point them at the book :) |
| 13:05.45 | x86 | how did you work that out? |
| 13:05.52 | gr0mit | lmadsen, are you us-based? |
| 13:05.55 | x86 | you just pay them a percentage? |
| 13:05.55 | lmadsen | x86: very carefully :) |
| 13:06.04 | x86 | never knew such a service existed.... |
| 13:06.05 | lmadsen | gr0mit: I live in downtown Toronto |
| 13:06.12 | gr0mit | aaah, nice! |
| 13:06.17 | lmadsen | x86: no, they charge their rate, and I charge them my flat rate |
| 13:06.19 | x86 | I've heard of like SalesForce.com and what not that are basically lead generators |
| 13:06.20 | gr0mit | loved toronto |
| 13:06.42 | lmadsen | gr0mit: just bought a condo downtown, so I'm pretty excited... I take possession on Thurs. |
| 13:06.45 | x86 | lmadsen: so you're still a contractor though right? |
| 13:06.58 | gr0mit | how do you get your consulting leads? |
| 13:07.02 | lmadsen | x86: yes, I'm still an independent consultant. I send my invoices from LeifMadsen Enterprises, Inc. |
| 13:07.18 | lmadsen | gr0mit: people usually just come to me... I haven't had to actively look for work for some years now |
| 13:07.22 | x86 | lmadsen: what's this front company? I might be interested in checking them out |
| 13:07.26 | gr0mit | nice |
| 13:07.56 | lmadsen | x86: http://www.digium.com/en/services/ |
| 13:07.57 | lmadsen | :) |
| 13:07.59 | x86 | they provide sales and billing and all of that overhead, and all you have to do is do the work... that's totally rad... |
| 13:08.24 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:08.25 | lmadsen | I originally worked for Steve Sokol at Sokol & Associates |
| 13:08.29 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 13:08.35 | x86 | right I remember you telling me that |
| 13:08.36 | lmadsen | then the team transitioned over to Digium |
| 13:08.40 | x86 | gotcha |
| 13:08.48 | x86 | can I set something like that up? |
| 13:08.58 | x86 | do I have to get dCap certified first? |
| 13:08.59 | lmadsen | no idea... you can check I suppose :) |
| 13:09.14 | gr0mit | nice. i need to start ramping things up - my employer is looking very dodgy at the moment |
| 13:09.21 | x86 | yeah, I'll talk to mark about it next time we talk |
| 13:10.02 | lmadsen | x86: Mark probably has very little to do with that aspect anymore unfortunately. Company is getting too big. |
| 13:10.57 | lmadsen | which is a good thing! |
| 13:11.09 | x86 | yeah but he could throw in a word about me :) |
| 13:11.19 | lmadsen | I really like working in the professional services department. I get to work with a lot of cool people on a lot of neat projects |
| 13:11.24 | lmadsen | x86: true :) |
| 13:11.29 | *** join/#asterisk Maxous (n=Maxous@mobile-166-214-147-208.mycingular.net) |
| 13:11.35 | x86 | nifty |
| 13:11.38 | x86 | travel required? |
| 13:11.39 | lmadsen | x86: just ask him if you can use him as a reference |
| 13:11.47 | lmadsen | x86: I don't travel -- I work from home |
| 13:12.00 | x86 | see that's what's up |
| 13:12.10 | x86 | I want to work from home... and I want to live in downtown chicago :) |
| 13:12.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.59 | x86 | a condo down there is about $24k/yr on average |
| 13:12.59 | lmadsen | wtf |
| 13:13.04 | lmadsen | oh... $24k/yr heh |
| 13:13.06 | x86 | $2k/mo |
| 13:13.10 | lmadsen | that's a lot :) |
| 13:13.13 | x86 | (for a lease) |
| 13:13.14 | lmadsen | how many sqft? |
| 13:13.20 | x86 | 1 bedroom heh |
| 13:13.45 | gr0mit | usd or cdn $?? |
| 13:13.45 | lmadsen | I pay $1641/mth for a 1+1, about 750sqft. Just bought for $275k for 585sqft, 1 bedroom |
| 13:13.57 | x86 | not sure how many sq ft.... but my buddy lives on the 31st floor in an older condo building, 1 bedroom, 1 bath, $2k/mo, $200/mo extra for a parking space in the deck |
| 13:14.00 | gr0mit | or are they at parity now? |
| 13:14.11 | lmadsen | gr0mit: doesn't matter -- CAD is actually worth more |
| 13:14.17 | gr0mit | really!?!! |
| 13:14.27 | x86 | yeah I'm talking USD |
| 13:14.27 | lmadsen | gr0mit: within 0.01, so at parity |
| 13:14.33 | gr0mit | good grief! |
| 13:14.49 | lmadsen | gr0mit: it's been like that for months... you obviously aren't Canadian getting paid in USD and losing 25% on the exchange rate |
| 13:14.54 | x86 | gr0mit: yeah the USD is very weak right now... |
| 13:15.01 | *** join/#asterisk oilinki (n=oil@ppp-124-121-246-52.revip2.asianet.co.th) |
| 13:15.09 | lmadsen | I was LOSING 10% on every dollar exchanged at some point |
| 13:15.12 | gr0mit | nah - i am in UK getting paid in GBP by a poor american company |
| 13:15.16 | x86 | lmadsen: it's best to find work in canada and spend your money down here ;) |
| 13:15.17 | lmadsen | about a year and a half ago I was MAKING 20% |
| 13:15.27 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:15.35 | JT | i think the AUD is about to be worth more than the USD soon too |
| 13:15.43 | lmadsen | x86: no way... I like to import the money and spend it here :) |
| 13:15.50 | lmadsen | JT: wow... that's grazy |
| 13:15.54 | lmadsen | crazy even :) |
| 13:16.13 | gr0mit | sold some stock in USD a couple years ago - i thought i was clever - opened a USD account |
| 13:16.16 | JT | heh |
| 13:16.20 | x86 | lmadsen: I'm saying if the CAD is stronger, make CAD, then convert it to USD and you'll effectively have more money down here |
| 13:16.24 | gr0mit | now i have some worthless usd |
| 13:16.25 | lmadsen | JT: don't expect the prices of your commodities to drop unfortunately... only after about a year of a crappy exchange rate are prices finally starting to come down |
| 13:16.38 | lmadsen | x86: it's at par, it doesn't matter anymore |
| 13:16.47 | gr0mit | the only consolation was that if i had not sold them at all i would not have anything at all!!!! |
| 13:16.48 | x86 | it will as the USD continues to dive ;) |
| 13:16.50 | JT | lmadsen: hey i can still import stuff cheap from the us :) |
| 13:16.56 | lmadsen | JT: true :) |
| 13:17.29 | lmadsen | x86: it's down diving... it dropped to 1.00 USD -> 0.90 CAD at one point, but it is back up to par |
| 13:17.36 | lmadsen | s/down/done/ |
| 13:18.12 | gr0mit | not sure what is falling faster - the USD or the Motorola stock price |
| 13:18.24 | *** join/#asterisk dlu_ (n=dlu@dus-ix-pos-002.dus.net) |
| 13:18.42 | lmadsen | USD isn't falling anymore (at least in relation to CAD) |
| 13:19.26 | x86 | it will, watch |
| 13:19.43 | lmadsen | doubt it |
| 13:20.09 | jer | the CAD is the 4th strongest currency on the planet right now |
| 13:20.17 | lmadsen | oh ya |
| 13:20.24 | lmadsen | CAD is very strong economically right now |
| 13:20.28 | jer | nod |
| 13:20.28 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 13:20.30 | lmadsen | just look at housing prices in Saskatchewan |
| 13:20.33 | Maxous | lmadsen: why is motorola stock price falling? |
| 13:20.34 | jer | thanks to all our national resources |
| 13:20.48 | jer | tech stocks are a gamble, you want security -- natural resources |
| 13:20.51 | lmadsen | Maxous: because they haven't innovated anything since the Razr? |
| 13:20.52 | x86 | lmadsen: I thought Toronto was the most expensive |
| 13:20.56 | lmadsen | x86: it is :) |
| 13:21.04 | lmadsen | actually... vancouver is more expensive |
| 13:21.04 | jer | x86, Montréal is a little more expensive per capita |
| 13:21.11 | x86 | really? |
| 13:21.14 | lmadsen | yes |
| 13:21.14 | jer | nod |
| 13:21.17 | Maxous | lmadsen: Hah, gotcha. |
| 13:21.22 | lmadsen | Vancouver is about 25% more expensive than Toronto |
| 13:21.32 | x86 | I was looking at places in Toronto, and the cheapest house I could find was $1.3M |
| 13:21.37 | jer | lmadsen, but only about 8% more than Montréal |
| 13:21.44 | jer | x86, and that's a one bedroom shack in unionville |
| 13:21.45 | jer | lol |
| 13:21.47 | lmadsen | jer: ya, natural resources as long as our stupid gov't stops selling them to foreign companies |
| 13:21.57 | lmadsen | jer: ya, Montreal and Toronto are almost on par |
| 13:22.25 | lmadsen | x86: oh ya... houses are stupid expensive... but you're looking more like $2M to start in Vancouver |
| 13:22.34 | lmadsen | unless you move to a suburb, but then you're not in Toronto proper |
| 13:22.36 | jer | i live in the middle of nowhere, about an hour south of georgian bay, housing prices here are fairly reasonable for nice places, about 300k |
| 13:23.43 | lmadsen | I live here come thursday :) http://tinyurl.com/4ghg7w |
| 13:24.38 | jer | lmadsen, i could not live in toronto |
| 13:24.41 | jer | i just couldn't |
| 13:24.41 | x86 | my house was $50k.... ~2 acres of land, fenced in back yard, 2 bedrooms, 1 bath, ~950 sq. ft. |
| 13:25.03 | lmadsen | jer: I come from a town of 4500 people, and I love Toronto. Lived downtown for 2 years now, and finally bought a place |
| 13:25.15 | lmadsen | x86: ya, I could buy that too, but I hate living in the country :) |
| 13:25.26 | jer | x86, when the ex wife and i built our house (which neihter of us live in anymore), we bought the property (15 hectares of mostly forest) for 36k, built the house for 425k, but it was a VERY nice modern house |
| 13:25.39 | jer | lmadsen, same, except loving toronto |
| 13:25.40 | jer | hehe |
| 13:25.51 | lmadsen | what is there not to love about toronto?! |
| 13:26.19 | jer | lmadsen, umm, i'm not going to get into an argument with you over it... just my opinion of the city |
| 13:26.28 | lmadsen | I wasn't trying to argue |
| 13:26.32 | lmadsen | I was just curious |
| 13:26.35 | lmadsen | but c'est la vie |
| 13:26.41 | jer | but my response would have incited an argument, so i'll reserve my statement |
| 13:26.47 | lmadsen | I don't argue |
| 13:26.59 | lmadsen | people are way to fuckin' touchy on irc :) |
| 13:27.04 | jer | indeed |
| 13:27.22 | jer | is probably going to move to costa rica anyway in a year or so |
| 13:27.31 | lmadsen | I know lots of people who don't like Toronto -- they moved to Vancouver. I went to Vancouver a couple of times, and I like it, but could never live there |
| 13:27.31 | jer | all depends on what i can find for work down there |
| 13:27.46 | jer | i love Montréal |
| 13:27.58 | jer | downtown is especially fun after a hockey game |
| 13:28.30 | lmadsen | ya I bet. Only thing I don't like about Montreal is how cold the winters get |
| 13:29.10 | lmadsen | anyways, breakfast then work time |
| 13:29.26 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-2864a8a799a7d75e) |
| 13:30.00 | jer | lmadsen, meh, 7th generation Canadian, i guess winter is in my blood hehe |
| 13:30.21 | jer | snow is the only thing that bugs me... it could snow all it wanted to away from roads and sidewalks |
| 13:30.28 | x86 | lmadsen: I'm not in the country |
| 13:30.43 | lmadsen | actually I'm starting to like winter again because I started skiing this winter |
| 13:30.51 | x86 | lmadsen: it's a small metro area... ~250k people total in the area |
| 13:31.04 | jer | i like playing shinny outdoors, the best part of winter |
| 13:31.20 | x86 | shinny? |
| 13:31.22 | jer | gonna miss that when i move to CR |
| 13:31.27 | x86 | CR? |
| 13:31.31 | jer | x86, kinda like what US-ians call "road hockey" |
| 13:31.33 | jer | costa rica |
| 13:31.51 | jer | except shinny isn't limited to just roads, or parking lots, or hell, even ice |
| 13:32.29 | *** join/#asterisk ManxPower (n=manxpowe@150.sub-75-250-157.myvzw.com) |
| 13:33.13 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:34.12 | *** join/#asterisk ludan (n=daniele@192.167.215.122) |
| 13:34.24 | ludan | hello |
| 13:35.21 | x86 | ah ok |
| 13:35.45 | Cabel | I am running a fractional T1 with 16 voice channels. I know all the information about it except the Signaling. Would it be PRI-Net or PRI-CPE? |
| 13:37.08 | JT | depends |
| 13:37.15 | JT | if it's pri, pri_cpe |
| 13:37.20 | JT | but it might be RBS |
| 13:38.36 | *** join/#asterisk gramulhaozin (n=charles@c-65-34-131-58.hsd1.fl.comcast.net) |
| 13:38.45 | ludan | guys i do not understand why i can't create a conference |
| 13:38.56 | ludan | i can see the users with "sip show peers" |
| 13:39.18 | ludan | but i'm missing steps in order to get the conference up and running |
| 13:39.33 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) |
| 13:39.35 | ThoMe | hello! |
| 13:39.43 | ThoMe | how i can set the timeout higher? |
| 13:39.45 | ThoMe | Jun 24 15:38:11 NOTICE[17367]: chan_iax2.c:5815 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 900) |
| 13:39.51 | ThoMe | this is set to 60 seconds |
| 13:40.01 | ThoMe | is it bad/good if i set the to 600 |
| 13:40.02 | ThoMe | ? |
| 13:40.29 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 13:41.08 | ThoMe | Any ideas? |
| 13:42.42 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 13:43.58 | ludan | [Jun 24 15:43:00] WARNING[4665]: pbx.c:1832 pbx_extension_helper: No application 'Meetme' for extension (training, 8900, 3) |
| 13:44.02 | ludan | this is what I get |
| 13:47.55 | ManxPower | ludan: you did not have zaptel installed when you built Asterisk |
| 13:48.48 | ludan | ManxPower: so i only need to install it and that's it? |
| 13:49.18 | ManxPower | ludan: install zaptel, reinstall asterisk from scratch |
| 13:49.45 | ludan | argh |
| 13:50.01 | ludan | no less painfull way? |
| 13:50.07 | ManxPower | ludan: painful???? |
| 13:50.09 | *** join/#asterisk buzzdee (n=buzzdee@ogo.rapideye.de) |
| 13:50.15 | ludan | :D |
| 13:50.17 | ManxPower | the most painful part is downloading the source |
| 13:50.49 | ludan | :D |
| 13:50.51 | ludan | thanks |
| 13:51.03 | ManxPower | I/m sure you can find out how to tell asterisk to forget the list of libs it found, but it would be faster to unpack source, make install |
| 13:51.49 | ludan | i installad asterisk 1.4.21, which zaptel version do I need? |
| 13:52.03 | ludan | zaptel-1.4-current.tar.gz should be fine |
| 13:52.15 | ManxPower | looks at the /topic and says "1.4.11" |
| 13:54.07 | buzzdee | hi,I try to call a fax number, but I get the famous "all circuits are busy" I use asterisk 1.2.18, and I can call the number via my mobile and I hear the FAX answering, any idea what I can do to make it work? http://www.pastebin.ca/1055082 |
| 13:55.37 | buzzdee | I call out via a PRI ISDN line from the Asterisk, calling other fax machines from the Asterisk works fine |
| 13:55.57 | ludan | ManxPower: «checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no» is it important? |
| 13:56.27 | ManxPower | buzzdee: what country are you located in? |
| 13:56.29 | ludan | i did install zaptel from sources |
| 13:56.50 | buzzdee | ManxPower, in Germany |
| 13:57.14 | ManxPower | Then I think you need one or more 0 at the beginngin of the call, don't you/ |
| 13:57.36 | buzzdee | ManxPower, I know, and this is usually working |
| 13:57.54 | ManxPower | then why are you sending the call to the telco without a leading 0 |
| 13:58.06 | buzzdee | I can call other FAX machines here in germany, without problem, its just with this number |
| 13:58.26 | ManxPower | then show a call to a fax machine that works? |
| 13:58.33 | ThoMe | what this: Jun 24 15:55:00 WARNING[17360]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x8148938', 9 retries! |
| 13:58.52 | buzzdee | ManxPower, I need to find a fax number, ... |
| 13:59.00 | ManxPower | ThoMe: harmless unless it causes a problem. If it causes a problem there is a doc file in the source to help you report the issue |
| 13:59.17 | ThoMe | ManxPower: ah ok |
| 13:59.17 | ManxPower | I have to go move a building in a few mins so make it fast |
| 14:01.07 | anonymouz666 | ThoMe: recompile with don't optimize, debug_threads, and when that happens you need to grep the output of 'core show locks' and send it to bugs.digium.com. |
| 14:01.28 | anonymouz666 | make sure you are using the latest 1.4 version |
| 14:01.47 | buzzdee | ManxPower, here it is: http://www.pastebin.ca/1055090 |
| 14:01.54 | buzzdee | after some ringing, it answered |
| 14:03.27 | ManxPower | buzzdee: the two calls have different outgoing callerid |
| 14:03.41 | buzzdee | yes, the first I changed, the second not, to make it fast |
| 14:04.04 | buzzdee | the first was also the 451 search/replace |
| 14:04.27 | ManxPower | That is the only difference between the two calls that I can see. Also, is one of them local and one not? |
| 14:04.45 | ManxPower | carriers can be strict about outgoig callerid, as far as rejecting the call if they don't like it |
| 14:04.47 | buzzdee | no, both are in different cities, and not local to the town where I am |
| 14:05.00 | ManxPower | but at least in germany, right? |
| 14:05.33 | buzzdee | I know, I fiddled some weeks to figure out, what I have to send out, so that the callierid is send out that I want |
| 14:06.22 | buzzdee | my number is 03381/8904451 and I have to send out 33818904451 then my callerid is taken, otherwise the telekom is replacing it with the main number from our number block |
| 14:06.30 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:06.37 | ManxPower | (9:04:59 AM) ManxPower: but at least in germany, right? |
| 14:06.49 | buzzdee | they are both in germany yes |
| 14:07.10 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 14:07.23 | ThoMe | [TK]D-Fender: hello. only a hello :) |
| 14:07.34 | ManxPower | buzzdee: I don't know what is wrong. |
| 14:08.09 | buzzdee | then thanks, then I'll have to ask on the mailing list |
| 14:08.35 | buzzdee | yeah, I also do not see, why the first is not working the same way as the other |
| 14:09.40 | ThoMe | ManxPower: have now updated from 1.2.24 to 1.4.21 - hoho |
| 14:10.47 | ludan | ManxPower: ok, now when I dial 8900 (after logging in) a voice tells me: this is not a valid conference number... uhm |
| 14:11.59 | [TK]D-Fender | ludan: It'll do that if you don't have a valid zaptel timer installed. It doesn't always mean the conference # you entered was bad. |
| 14:12.18 | ludan | timer? |
| 14:12.20 | ludan | uhm |
| 14:12.40 | ludan | i've something likle exten => 8900,3,Meetme(9000|M) in extensions.conf |
| 14:12.48 | [TK]D-Fender | ludan: ZAPTEL <---- |
| 14:12.58 | ludan | and conf => 9000 in meetme.conf |
| 14:13.10 | [TK]D-Fender | ludan: this is not a CONFIG error. |
| 14:14.29 | ludan | and what is it about? |
| 14:14.58 | [TK]D-Fender | ludan: I've said it twice <- You don't have a valid zaptel timer installed |
| 14:15.34 | ludan | how can I check if this is true? and in case, where can I get a zaptel timer? |
| 14:15.47 | [TK]D-Fender | ludan: INSTALL ZAPTEL. |
| 14:16.19 | ludan | it is installed! |
| 14:16.25 | [TK]D-Fender | ludan: www.asterisk.org Go look at the downloads section, then go read the book on how to install ZTDUMMY. |
| 14:16.57 | ludan | but I did it already |
| 14:17.09 | [TK]D-Fender | ludan: clearly not initialized, or wasn't built before Asterisk was. Zaptel needs to be compiled and initialized before Asterisk is compiled in the first place or it won't be compiled with support for it. |
| 14:17.26 | ludan | let see |
| 14:18.04 | ludan | !!! it works! |
| 14:18.15 | ludan | i didn't start zaptel :( |
| 14:18.18 | ludan | what a idiot |
| 14:19.05 | coreyf52 | how do i go about suggesting a feature enhancement? The sender email address for voicemails currently can only be set in voicemail.conf, I'd really like to set the source address in the dialplan.. |
| 14:20.24 | [TK]D-Fender | coreyf52: You can ask on the mailing list to see if someone will take you up on it. |
| 14:21.23 | coreyf52 | [TK]D-Fender: http://lists.digium.com/mailman/listinfo/asterisk-dev/ or is someplace else better? |
| 14:21.35 | [TK]D-Fender | coreyf52: that'd be it |
| 14:21.47 | ludan | [TK]D-Fender: before the crash of my disk, the asterisk configuration was as follows: with my username I could join the queue (8900) and wait for people to get in, talk with them and afterwars move them into the conf (9000). is it difficult to set up again? |
| 14:22.31 | [TK]D-Fender | ludan: You've already done it before. You shouldn't ask questions you already know the answer to. |
| 14:22.53 | ludan | [TK]D-Fender: man what can i do, i do not remember what I did :( |
| 14:23.13 | [TK]D-Fender | ludan: Get a clue, and read the book. |
| 14:23.15 | [TK]D-Fender | ~book |
| 14:23.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 14:23.19 | coreyf52 | [TK]D-Fender: great thanks! i've made an attempt on my own but zero success, i'm not very good with C (spend too much time with C#) |
| 14:23.19 | ludan | thanks |
| 14:24.22 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
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| 14:25.28 | *** mode/#asterisk [+o mog] by ChanServ |
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| 14:40.57 | ludan | [TK]D-Fender: please, last thing! i've a file that I include in sip.conf... i'd like a webinterface to edit this file and create authorized users to join the conf... is it possible? |
| 14:42.15 | Kobaz | [Jun 24 10:41:52] NOTICE[18888]: channel.c:2227 __ast_read: Dropping incompatible voice frame on IAX2/2280-6 of format gsm since our native format has changed to ulaw |
| 14:42.18 | Kobaz | [Jun 24 10:41:52] NOTICE[18888]: channel.c:2227 __ast_read: Dropping incompatible voice frame on IAX2/2280-6 of format gsm since our native format has changed to ulaw |
| 14:42.21 | Kobaz | [so umm |
| 14:42.23 | Kobaz | how would i fix that? |
| 14:42.32 | ludan | i tried the one that comes with asterisk but it creates users in users.conf and then these users can't join the conf |
| 14:42.55 | Kobaz | i'm using ulaw on the iax2 peer, and i allow ulaw in the codecs |
| 14:43.15 | [TK]D-Fender | ludan: Of course its possible. Look at the other 50 GUI's out there. |
| 14:43.25 | [TK]D-Fender | ludan: Go right ahead and get coding. |
| 14:46.27 | *** join/#asterisk magic_hat (n=geoffdou@h-68-164-7-249.chcgilgm.dynamic.covad.net) |
| 14:47.43 | magic_hat | hey everyone. I'm using ruby and AGI to call a bunch of numbers in *. * then transfers control of the call back to my ruby script. I'm wondering if there's any way for me to know whether the number dialed is a valid one (or whether it's reached the telco error message), whether anyone's picked up, etc. |
| 14:48.31 | [TK]D-Fender | magic_hat: if the respones is inband audio... not really. |
| 14:48.57 | magic_hat | [TK]-D-Fender: eek. |
| 14:49.13 | magic_hat | What about CHANNEL STATUS? That doesn't fly w/ audio? |
| 14:50.22 | ThoMe | how i SET the status with devstate on the CLI ? |
| 14:50.27 | ThoMe | this works nto: Set(DEVSTATE(Custom:lamp1)=BUSY) |
| 14:50.37 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 14:50.37 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:50.39 | ThoMe | No such command |
| 14:50.56 | [TK]D-Fender | magic_hat: it doesn't speak english <- The best you can hope for is using indications.conf + "callprogress=yes", but don't forget, that also means "hanguponmycallsrandomly=yes" |
| 14:51.06 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 14:51.38 | magic_hat | wow, that sounds grand! lol |
| 14:51.42 | ddunavant | Could someone help me with why I can't seem to call into my number for about 20 min after a reload? My config(what I think is relevant) is at: http://pastebin.com/m34bdc184 |
| 14:51.46 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:52.52 | ddunavant | actually it's at: http://pastebin.com/m2b9621ba |
| 14:52.56 | [TK]D-Fender | ThoMe: Google gives the answer in the first result of my query. |
| 14:53.05 | [TK]D-Fender | ThoMe: Show some effort. |
| 14:53.06 | magic_hat | [TK]-D: does * hand control of the call back to the agi script if the # dialed is an invalid number? |
| 14:53.21 | [TK]D-Fender | ddunavant: You have not configured your system to work behind NAT. Read up : |
| 14:53.23 | [TK]D-Fender | ~sipnat |
| 14:53.23 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:53.25 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 14:53.39 | [TK]D-Fender | magic_hat: Dial will exit if it has a reason to |
| 14:53.42 | *** join/#asterisk smach (n=smach@207.35.173.122) |
| 14:53.45 | ThoMe | [TK]D-Fender: hello. i mean, can i use "SET" with php ? |
| 14:54.07 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:54.15 | ddunavant | [TK]D: would that matter for the server if the server wasn't NATing? |
| 14:54.39 | magic_hat | So if i've dialed something with agi, and then I never hear anything back, I can assume it was either a bad number, recipient immediately hung up or I forgot to pay my Teliax bill? |
| 14:54.40 | [TK]D-Fender | ThoMe: of course you can. How else would you set variables? My guess is your formatting is all wrong. AGI is NOT the same formatting as externsions.conf |
| 14:55.14 | ThoMe | [TK]D-Fender: hm. ok. how i can on demand this run: Set(DEVSTATE(Custom:lamp1)=BUSY) ? |
| 14:55.16 | [TK]D-Fender | ddunavant: You have nat=yes in there, so I'd be thinking it was behind NAT. Is it, or isn't it? |
| 14:55.24 | ThoMe | or [TK]D-Fender DEVSTATE(Custom:lamp1)=BUSY |
| 14:55.25 | ludan | boh |
| 14:55.32 | ludan | there a lot of problems like mine |
| 14:55.45 | [TK]D-Fender | ludan: GUI's suck. Deal with it. |
| 14:55.48 | ludan | the gui edits users.conf, i want to edit sip.conf :( |
| 14:55.51 | ludan | man i know |
| 14:55.53 | ludan | i never use it |
| 14:56.02 | ludan | but i've to provide an idiot with a gui |
| 14:56.02 | ddunavant | [TK]D: the clients are but the server isn't, let me see if that fixes it |
| 14:56.18 | mvanbaak | ludan: vim is a wonderfull gui |
| 14:56.24 | ludan | that's exactly what i use! |
| 14:57.46 | [TK]D-Fender | ddunavant: Also your register must appear at the end of everything else that should be under [general] |
| 14:58.11 | ddunavant | [TK]D: gotcha, thanks, I'll see if that gets what I want done |
| 14:58.13 | defswork | aastra snuck out a 2.2.1 release - but I can't find any release notes :O |
| 14:58.15 | [TK]D-Fender | users.conf is a flaming pile of shit. |
| 14:58.42 | [TK]D-Fender | ddunavant: And for the next, pastebin an actual call with sip debug enabled. |
| 14:59.05 | ddunavant | [TK]D: ok |
| 14:59.59 | Maliuta | [TK]D-Fender: you sound like a magician; "And for my next trick I will ..." |
| 15:00.04 | Maliuta | ;) |
| 15:00.31 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582656.dsl.bell.ca) |
| 15:00.36 | Maliuta | alright, 0100 is time for bed |
| 15:02.03 | seanbright | roger |
| 15:02.06 | seanbright | over and out |
| 15:04.05 | ThoMe | servetux*CLI> console send text SIP/10 test |
| 15:04.05 | ThoMe | Not in a call |
| 15:04.06 | ThoMe | hmm |
| 15:04.33 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:05.00 | ddunavant | [TK]D-Fender: It hasn't changed anything, The problem that I'm seeing is that the call never seems to get to my asterisk box after I perform a reload for about 20 minutes or so. I'm not sure if it's a problem with the provider or with my config(I assume my config) |
| 15:06.10 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:06.26 | Kobaz | do de do |
| 15:06.32 | Kobaz | [Jun 24 10:41:52] NOTICE[18888]: channel.c:2227 __ast_read: Dropping incompatible voice frame on IAX2/2280-6 of format gsm since our native format has changed to ulaw |
| 15:06.48 | magic_hat | so this app i'm working on is an IVR opinion polling deal. Call a random #, record answers. I'm only gonna be doing one call at a time, and it's going on its own server? How much RAM/processor do I need to handle one outbound call at a time? |
| 15:06.49 | Idle | Qwell: poke |
| 15:06.55 | Qwell | ? |
| 15:06.57 | Kobaz | [TK]D-Fender: any idea? |
| 15:07.01 | [TK]D-Fender | ddunavant: First you should not have "nat=yes" under [general] if * itself isn't behind NAT, and then you should still disable reinvites, etc. Check your firewall settings, etc. Pastebin a complete call attempt that fails with SIP DEBUG. Then ones you've restarted things, place a working call. |
| 15:07.05 | [TK]D-Fender | Kobaz: Nope. |
| 15:07.13 | Kobaz | hmm |
| 15:07.21 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:07.26 | defswork | magic_hat: 256mb |
| 15:07.30 | [TK]D-Fender | magic_hat: Virtually nothing. |
| 15:07.33 | Kobaz | it's like the codec is randomly change to gsm mid call |
| 15:08.00 | magic_hat | TK-D: lol that's what I hoped!. |
| 15:09.23 | magic_hat | TK-D: going back to my earlier?: So is there any way to know if the call's been picked up? Or will * only send the call back to agi after it's been answered? What i'm getting at is... how do I know when to start playing my outgoing message. |
| 15:09.27 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:11.08 | [TK]D-Fender | magic_hat: Use a PRI or SIP service that passes full prgress back |
| 15:11.18 | [TK]D-Fender | magic_hat: but you can basically forget about this on analog |
| 15:11.35 | magic_hat | Okay, so I'm doing this over a Teliax SIP connection. |
| 15:13.03 | *** part/#asterisk solar (n=solar@smtp.gentoo.org) |
| 15:13.47 | *** join/#asterisk RobH (n=RobH@72-254-5-174.client.stsn.net) |
| 15:14.06 | magic_hat | so how do I pull the progress info outta there? |
| 15:14.23 | [TK]D-Fender | magic_hat: You should already have it. Your dial should fail with a reason code. |
| 15:14.35 | [TK]D-Fender | magic_hat: otherwise they consider the call answered as its placed |
| 15:14.47 | defswork | notices aastra now do a wlan phone |
| 15:14.57 | ddunavant | [TK]D-fender: SIP debugging(failed call, incoming from cell): http://pastebin.com/m68a63fe8, SIP.conf(used for that call): http://pastebin.com/m7a49b6ae |
| 15:15.06 | *** part/#asterisk goobsoft (n=chad@cpe-24-167-97-202.satx.res.rr.com) |
| 15:15.30 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 15:15.33 | magic_hat | [TK]D: okay, gonna mess around w/ this and see how it goes. |
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| 15:21.53 | *** join/#asterisk kannan (n=kannan@123.201.60.110) |
| 15:21.58 | kannan | hello all |
| 15:23.00 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
| 15:25.44 | *** join/#asterisk ibob63 (n=ibob63@dsl-217-155-69-86.zen.co.uk) |
| 15:26.39 | *** join/#asterisk RobH (n=RobH@72-254-5-174.client.stsn.net) |
| 15:27.03 | NovceGuru | herro |
| 15:27.08 | *** join/#asterisk TedNJ38 (n=HungLad@ool-435336f7.dyn.optonline.net) |
| 15:27.19 | ibob63 | can anyone recommend a did / iax termination provider in france? |
| 15:27.23 | TedNJ38 | Can someone help me with a problem with Trixbox? |
| 15:27.37 | [TK]D-Fender | ddunavant: "secure=no" is not a valid option. What you should probably be doing is "insecure=port,invite" |
| 15:27.47 | [TK]D-Fender | ddunavant: update, apply, test |
| 15:28.03 | [TK]D-Fender | TedNJ38: Wrong channel, you know this already. Move along. |
| 15:28.24 | TedNJ38 | I know, nobody is replying in the other channels. |
| 15:28.29 | TedNJ38 | Thanks anyway. |
| 15:28.52 | *** part/#asterisk TedNJ38 (n=HungLad@ool-435336f7.dyn.optonline.net) |
| 15:29.00 | [TK]D-Fender | TedNJ38: Yes, and just because my mechanic isn't answering my calls doesn't mean I can ask the clerk at McDonalds either. |
| 15:29.20 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:30.45 | *** join/#asterisk l2cache (n=chatzill@179.190.204.68.cfl.res.rr.com) |
| 15:30.53 | ddunavant | [TK]D-Fender: Thank you so much. It works now |
| 15:33.26 | gr0mit | ibob63, what are you looking for? |
| 15:33.42 | gr0mit | calls in or out? |
| 15:35.28 | gr0mit | ibob63, ping |
| 15:35.49 | [TK]D-Fender | ddunavant: glad to help |
| 15:36.11 | ibob63 | hi gr0mit. I am looking for call in and out. Basically, I need a did somewhere in France for business. |
| 15:36.19 | *** join/#asterisk PepOSX (n=angeldav@190.78.221.113) |
| 15:36.24 | gr0mit | u based in UK? |
| 15:36.32 | ibob63 | yes. |
| 15:36.47 | gr0mit | i can offer Paris numbers |
| 15:36.55 | gr0mit | PM for details |
| 15:37.38 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:46.52 | *** join/#asterisk remibemol (n=remibemo@ASt-Lambert-151-1-14-203.w82-120.abo.wanadoo.fr) |
| 15:51.28 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:52.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:59.18 | *** join/#asterisk kensukeido (i=c829e4f4@gateway/web/ajax/mibbit.com/x-67964a4a8094f7a1) |
| 15:59.54 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:00.06 | *** join/#asterisk RobH (n=RobH@72-254-5-174.client.stsn.net) |
| 16:02.36 | x86 | hmm... I'm having quality issues randomly all of a sudden (as in, within the past week only) with inbound faxes going from POTS line to channel bank over T1 to Sangoma T1 card into Asterisk then IAXModem and finally Hylafax |
| 16:02.59 | x86 | seems like sometimes the fax is just garbled... |
| 16:03.38 | x86 | should I just use a fax modem direct to hylafax? would that solve the issues? |
| 16:07.32 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 16:07.35 | Katty | ello. |
| 16:08.42 | [TK]D-Fender | Katty: Mew. |
| 16:08.45 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-237-15.balt.east.verizon.net) |
| 16:08.48 | Qwell | ~roflmao |
| 16:08.48 | jbot | well, roflmao is rolling on the floor laughing my arse off, or painful, or http://www.youtube.com/watch?v=iEWgs6YQR9A |
| 16:10.40 | Qwell | Katty: ^^ |
| 16:11.06 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 16:11.21 | Katty | hugs [TK]D-Fender and Qwell |
| 16:11.46 | Katty | anyone know where to check if the auto answer for polycoms makes all the phones auto answer, but there's no audio... and regular calls are fine? |
| 16:12.03 | Katty | DNS? |
| 16:14.11 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 16:14.16 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 16:14.23 | *** join/#asterisk eharris_ (n=eharris@75-43-20-21.lightspeed.austtx.sbcglobal.net) |
| 16:17.37 | [TK]D-Fender | Katty: Pastebin..... |
| 16:17.58 | *** join/#asterisk CunningPike_ (n=arodgers@204.239.8.157) |
| 16:18.35 | kensukeido | hi, where can find the diff between asterisk 1.2 and 1.4? |
| 16:22.04 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:22.39 | [TK]D-Fender | kensukeido: plenty of articles out there, and there is the changelog, and the BOOKS. |
| 16:26.28 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 16:27.24 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:30.04 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-185-165.hsd1.wa.comcast.net) |
| 16:36.08 | Katty | [TK]D-Fender: where i'm going i don't have irc. |
| 16:36.14 | Katty | [TK]D-Fender: do you have any suggestions that i can look at? |
| 16:36.58 | *** join/#asterisk neurosys0 (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 16:40.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:40.59 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 16:42.56 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:47.53 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:48.36 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:53.10 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:54.20 | NovceGuru | Katty: that sounds like a terrible place |
| 16:54.27 | [TK]D-Fender | Katty: That could be one of a million things... |
| 16:54.39 | jeev | hi |
| 16:54.47 | [TK]D-Fender | Katty: Sorry, you're going to have to seriously narrow that down. |
| 16:57.56 | jeev | fender, anywhere you know that offers long term financing for a bunch of phones? |
| 16:58.31 | gr0mit | suggests a bank! |
| 16:59.10 | jeev | not for me, for someone else |
| 16:59.21 | jeev | i'm really getting annoyed by them and i might just tell them i'm not doing it anymore |
| 17:00.15 | [TK]D-Fender | jeev: If they are stupid chances are they have a lot of experience at it. Don't expect to change them instantly. |
| 17:00.16 | *** join/#asterisk spokra (n=spokra@gumby.sea0.speakeasy.net) |
| 17:00.19 | jjshoe | jeev I'd ditch them, sounds like they won't have cash left for you. |
| 17:01.02 | gr0mit | cash first, work later is what I have learned |
| 17:01.42 | gr0mit | it is very tough to get cash out of people after the event |
| 17:02.06 | [TK]D-Fender | jeev: And as I said before the project doesn't have to be a rip&replace. The can start with a minimal investment to go VoIP with thier current PBX and then once the savings have realized themselves, finish the onversion. |
| 17:02.44 | gr0mit | big-bang projects are usually high stress and doomed to mediocrity |
| 17:03.09 | jeev | fender, either way. they will have to pay some nortel expert to come in and move everything. |
| 17:03.15 | gr0mit | start small, piggyback asterisk in the middle |
| 17:03.59 | [TK]D-Fender | jeev: And what role do you play in all of this exactly? |
| 17:04.19 | *** join/#asterisk PepOSX (n=angeldav@190.78.221.113) |
| 17:04.48 | jeev | i'm just his friend, setting this thing up for them to save them money |
| 17:05.39 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 17:06.36 | Cabel | Is it possible to add an array after installing asterisk? If so how? |
| 17:06.48 | Qwell | an array of what? |
| 17:07.03 | Cabel | opps |
| 17:07.05 | Cabel | i meant raid |
| 17:07.07 | Cabel | raid arrya |
| 17:07.21 | Qwell | Asterisk has nothing to do with a RAID |
| 17:07.21 | Cabel | array* |
| 17:07.47 | Cabel | yea well i know, but I'm a noob with linux and was hopeing someone had suggestions |
| 17:08.10 | Qwell | ##linux |
| 17:08.17 | *** part/#asterisk kensukeido (i=c829e4f4@gateway/web/ajax/mibbit.com/x-67964a4a8094f7a1) |
| 17:09.00 | mog | Cabel, the answer is yes |
| 17:09.23 | mog | "its only software" |
| 17:11.02 | Cabel | ok. I just dont feel like reinstalling my system again just to add a raid |
| 17:11.08 | Cabel | thanks |
| 17:11.32 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 17:11.57 | *** join/#asterisk _MrSeb_ (n=SebaX@87.253.113.240) |
| 17:11.57 | _MrSeb_ | Hi to all |
| 17:12.09 | mog | hi to _MrSeb_ |
| 17:13.56 | NovceGuru | hai2u |
| 17:14.26 | _MrSeb_ | I've a nat problem and all the incoming call hangup after 10/15 seconds... some ideas? I've all traffic of router redirected on asterisk server as dmz |
| 17:14.57 | _MrSeb_ | in the debug I'd found some warning about a critical packet |
| 17:16.01 | Idle | Qwell: who do I contact to get an RMA on this card? the client is totally pissed right now |
| 17:16.09 | Qwell | support |
| 17:16.13 | Idle | aight |
| 17:17.52 | jeev | [TK]D-Fender, i dont know why he's asking for long term financing on 4k. i told him i'm not gonna deal with it till i know what's going on.. its either that or pay 1800 a month to att. |
| 17:24.30 | jjshoe | jeev tell him to get an american express card :D |
| 17:26.37 | *** join/#asterisk simond (n=simon@syria.uc.org) |
| 17:28.27 | flujan | hello guys. I am trying to use realtime sippeers with pgsql |
| 17:28.58 | flujan | I am having problems, i need the option rtcachefriends no sip.conf because i need the nat and qualify options |
| 17:30.32 | flujan | but when i update a field in the database it does not reflect in the asterisk configuration for instance the user context and so forth |
| 17:30.50 | flujan | if i issue a sip reload command asterisk drops all peers |
| 17:31.37 | ThoMe | hello. |
| 17:31.54 | ThoMe | in which variable is the source number if I have a incomming call? |
| 17:32.17 | ThoMe | ${CALLERID(num)} ? |
| 17:33.07 | jjshoe | the number of the person calling, or the number they dialed? |
| 17:33.17 | ThoMe | jjshoe: source. or the sip-id? |
| 17:34.59 | flujan | I do believe that I am having this issue because of the rtcachefriends option... There is a way to have asterisk periodically update the information in the database and keep using the rtcachefriend option? |
| 17:38.40 | *** join/#asterisk l2cache (n=chatzill@179.190.204.68.cfl.res.rr.com) |
| 17:38.55 | jjshoe | ThoMe I don't get what you're asking for, sorry. |
| 17:40.57 | *** join/#asterisk kombi (n=kombi@port-87-234-216-47.static.qsc.de) |
| 17:41.28 | kombi | Unable to create channel of type 'mISDN' (cause 66 - Channel not implemented)<- where must I implement it? |
| 17:42.38 | kombi | make menuconfig says [*]chan_misdn.. |
| 17:43.08 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 17:43.08 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:46.43 | _MrSeb_ | someone can say to me how to resolve this... WARNING[18855]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission C57688FF-3D8B11DD-A312CE3D-FD954DD@62.94.71.96 for seqno 102 (Critical Response)? |
| 17:47.15 | *** join/#asterisk implicit (n=implicit@dhcp-x216-194.mobile.uci.edu) |
| 17:47.51 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-112-rrdg-esr-2.dynamic.isadsl.co.za) |
| 17:49.13 | *** join/#asterisk glaz (n=strke@mofu.ca) |
| 17:51.03 | kombi | hmm, all I did was migrate from 1.4 to 1.6, now misdn seems gone.. |
| 17:51.35 | lmadsen | did you recompile the mISDN (or whatever it was) packages? |
| 17:52.10 | kombi | lmadsen: shoot, no.. just those four I always compile.. where do you get those from again? |
| 17:52.17 | lmadsen | I have no idea |
| 17:52.22 | kombi | great..;) |
| 17:52.24 | lmadsen | I don't use BRI devices |
| 17:52.38 | kombi | I did all that before, just can't remember.. |
| 17:53.00 | lmadsen | documentation is a good thing :) |
| 17:53.04 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 17:53.05 | lmadsen | that's why I have a personal wiki |
| 17:53.40 | kombi | lmadsen: funny you should say that, I do too, but I also forgot to write it in there..;) |
| 17:54.29 | kombi | but then again, this box has a completely standard digium card, do I need misdn at all? |
| 17:54.41 | Qwell | kombi: which card? |
| 17:54.49 | mvanbaak | if you have BRI you need it |
| 17:55.00 | kombi | Qwell: how do I tell from cli.. |
| 17:55.03 | lmadsen | "completely standard digium card" |
| 17:55.25 | Qwell | kombi: You can't. You have to kind of...just know. |
| 17:55.39 | lmadsen | dmesg |
| 17:55.41 | lmadsen | and ztscan |
| 17:55.49 | kombi | Qwell: ;) I'll screw the box open.. or wait.. |
| 17:55.52 | lmadsen | lspci helps too |
| 17:55.53 | x86 | what would you guys recommend for an external fax modem? looking to use it only for inbound faxing to hylafax... not sure if you guys would have any suggestions or not... |
| 17:55.54 | Qwell | kombi: lspci |
| 17:56.25 | lmadsen | Qwell: copy cat |
| 17:56.55 | kombi | Qwell: right! Digium, Inc. Uknown device b410 (rev 01)..;) so bri I take it |
| 17:57.03 | Qwell | so yeah, you need misdn |
| 17:57.56 | kombi | Qwell: such a shame, had it all working before but had to do a dist-upgrade which messed everything up.. so on to misdn, where are misdn? |
| 17:58.11 | kombi | I mean, where are you misdn! |
| 17:58.23 | x86 | lmadsen: no suggestions for fax modem |
| 17:58.24 | x86 | ? |
| 17:58.28 | lmadsen | x86: I don't do faxing |
| 17:58.32 | x86 | ah ok |
| 17:58.39 | Strom_C | x86: I use an old external serial modem |
| 17:58.39 | lmadsen | if I had a suggestion, I would have offered it :) |
| 17:58.49 | Strom_C | some USR thing |
| 17:58.53 | khronos | <PROTECTED> |
| 17:59.37 | kombi | Qwell, lmadsen: you just do make b410p in the zaptel source, that goes into my wiki pronto.. |
| 18:00.44 | x86 | lmadsen: figured ;) |
| 18:01.36 | *** join/#asterisk edoceo (n=edoceo@c-71-197-244-147.hsd1.or.comcast.net) |
| 18:01.59 | *** join/#asterisk RobH (n=RobH@72-254-5-174.client.stsn.net) |
| 18:02.02 | x86 | Strom_C: I'm thinking USR just because that's all I've ever had experience with besides Hayes as far as modems go.... but I've never done faxing with one before |
| 18:03.05 | Strom_C | x86: seriously dude, it's just a modem |
| 18:03.20 | *** join/#asterisk [reed] (n=reed@firefox/gnu.webmaster.reed) |
| 18:03.27 | x86 | yeah ;) |
| 18:03.33 | [reed] | what's the (T) next to the port number in `iax2 show peers` ? |
| 18:03.35 | Strom_C | this is a lot like freaking out because you're not really sure which end of the toothbrush to stick in your mouth |
| 18:03.36 | *** join/#asterisk RoyK (n=roy@ti0002a380-0029.bb.online.no) |
| 18:03.42 | lmadsen | [reed]: trunking |
| 18:03.44 | x86 | but I've never done faxing so I'm not sure if some modems handle it better than others / etc |
| 18:03.49 | lmadsen | or so I would guess |
| 18:04.10 | [reed] | lmadsen: ah, thanks |
| 18:04.14 | x86 | Strom_C: no, it's a lot like picking the right toothbrush that's not going to cause damage to your gums ;) |
| 18:04.15 | [reed] | lmadsen: and it's always UDP? |
| 18:04.18 | [reed] | not ever TCP? |
| 18:04.19 | lmadsen | yes |
| 18:04.21 | lmadsen | never TCP |
| 18:04.21 | [reed] | k |
| 18:04.33 | *** join/#asterisk MrNaz (n=naz@ppp59-167-94-76.lns2.mel6.internode.on.net) |
| 18:05.24 | kombi | and now make b410p throws an error.. shoot! |
| 18:06.12 | [reed] | any hints on debugging iax2 trunk problems? I have two asterisk servers -- the first one can connect to the second one just fine while the second one can't connect to the first one (gets unreachable)... pretty sure it's not a network issue since I can ssh between the two machines just fine, but maybe I'm just not looking at the right thing. |
| 18:06.12 | kombi | ok, back to 1.4... |
| 18:06.21 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 18:06.43 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:06.53 | Strom_C | [reed]: ssh isn't the same port |
| 18:07.03 | [reed] | correct |
| 18:07.10 | Strom_C | is there a firewall or router between the two machines? |
| 18:07.23 | [reed] | there is, and this has been working fine for months... just stopped working early this morning |
| 18:07.35 | kombi | anyone good with make files? "CFLAGS was changed. Fix it to use EXTRA_CFLAGS" only how? |
| 18:07.45 | Strom_C | check to see if anyone touched it :) |
| 18:07.55 | edoceo | I want my extension 620 to be able to monitor/view/intercept calls to extension 617, 615 and 613 - what dialplan commands do I look for? |
| 18:08.28 | edoceo | How can I indicate to x620 that x617 is ringing and give them ability to intercept? |
| 18:09.26 | [TK]D-Fender | edoceo: Go read up on pickup groups on the WIKI |
| 18:09.28 | [TK]D-Fender | ~wikis |
| 18:09.29 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 18:09.37 | edoceo | Thanks TK! |
| 18:10.06 | [TK]D-Fender | edoceo: And to know that they are ringing you might be able to use "presence" on your phone to see their state. |
| 18:11.35 | [TK]D-Fender | ~blf |
| 18:11.35 | jbot | hmm... blf is Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 18:12.57 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 18:14.11 | [reed] | Strom_C: 99% sure it isn't a networking issue, especially since the check_asterisk.pl nagios script I have isn't getting an IAX reply on localhost |
| 18:19.01 | edoceo | It looks like the Line Status and Presence won't work for my Aastra 480i - so no way to indicate to x620 the status of x617 huh? |
| 18:21.00 | [TK]D-Fender | edoceo: Yes, Aastras support presence on the 480i |
| 18:22.13 | *** join/#asterisk Segnale007 (n=Segnale0@sms5-pool120-0101.bmts.com) |
| 18:22.24 | *** part/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 18:23.01 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 18:23.02 | anonymouz666 | anyone know if I can reload just the file func_odbc.conf without restart everything? |
| 18:23.38 | [reed] | Strom_C: a full shutdown and startup of asterisk seemed to fix it |
| 18:23.38 | [reed] | weird |
| 18:24.02 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 18:24.34 | kombi | now let's see whether zaptel make b410p compiles against kernel 2.6.22.. |
| 18:24.56 | _MrSeb_ | someone can say to me how to resolve this... WARNING[18855]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on transmission C57688FF-3D8B11DD-A312CE3D-FD954DD@62.94.71.96 for seqno 102 (Critical Response)? |
| 18:25.20 | edoceo | TK: If I wanted you to help me with this how much would it cost (in USD please) and how could I share files with you w/o using SSH (can't give access to system :( ) |
| 18:26.03 | kombi | _MrSeb_: is your sip gateway accessable? |
| 18:26.49 | _MrSeb_ | kombi: all traffic for router is redirected to sip server as in dmz |
| 18:27.26 | [TK]D-Fender | edoceo: PM |
| 18:28.47 | kombi | _MrSeb_ can you ping it? |
| 18:29.11 | [TK]D-Fender | _MrSeb_: DMZ is not enough for NAT support. READ : |
| 18:29.15 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 18:29.15 | [TK]D-Fender | ~sipnat |
| 18:29.16 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:31.57 | axisys | i get this when I start x-lite http://pastebin.com/f1826799b .. my asterisk is running on same laptop.. |
| 18:32.08 | axisys | what do I do to get the sound working |
| 18:32.12 | axisys | this is on ubuntu |
| 18:32.32 | [TK]D-Fender | axisys: the local console of * is stealing your dev/dsp |
| 18:32.37 | axisys | no other app running that is using /dev/dsp .. except may be asterisk server itself ? |
| 18:32.54 | [TK]D-Fender | axisys: They certainly can't FIGHT over it |
| 18:32.55 | axisys | [TK]D-Fender: :-) |
| 18:34.15 | _MrSeb_ | [TK]D-Fender: I'm always the same person, all thing you have saw to me I've done, nat, canreinvite, externalhost and all the other things, the last things was only port redirection, now it's full redirection but the problem is not resolved, outgoing call hangup in 10/15 seconds (audio in and out is ok), incoming call don't hangup, but haven't audio, in nor out |
| 18:34.23 | axisys | [TK]D-Fender: the dial works fine.. and also from someone elses desk I tried xlite to my asterisk server's ip.. works awesome.. is there a way to get the sound working.. i cn play multiple music same time with mplayer.. no prob |
| 18:34.57 | Qwell | axisys: chan_oss/chan_alsa lock the device |
| 18:34.59 | [TK]D-Fender | _MrSeb_: pastebin is your friend.... |
| 18:35.04 | Qwell | add a noload line in modules.conf for them |
| 18:35.23 | _MrSeb_ | [TK]D-Fender: thanks, wait a while |
| 18:35.33 | [TK]D-Fender | axisys: Qwell has given you the specific modules to comment out. |
| 18:35.53 | Qwell | [TK]D-Fender: That'll be $19.94 |
| 18:35.55 | axisys | Qwell: sweet! |
| 18:36.00 | axisys | [TK]D-Fender: thnx |
| 18:36.04 | axisys | let me try that |
| 18:36.06 | [TK]D-Fender | Qwell: Didn't hemp ME :p |
| 18:36.11 | [TK]D-Fender | help* |
| 18:36.15 | Qwell | sure it did! |
| 18:36.23 | Corydon76-dig | Freudian slip? |
| 18:36.38 | jeev | Fender was admitting to a crack problem |
| 18:36.41 | *** join/#asterisk blackhole (n=Mishu@unaffiliated/blackhole) |
| 18:36.51 | [TK]D-Fender | Corydon76-dig: Nope, just really bad aim for the "l" :P |
| 18:36.54 | Corydon76-dig | hemp != crack |
| 18:37.10 | [TK]D-Fender | With a biochemical imbalance like mine, who needs drugs! |
| 18:37.10 | Qwell | http://en.wikipedia.org/wiki/Futurama:_The_Beast_with_a_Billion_Backs |
| 18:37.12 | [TK]D-Fender | ~whee |
| 18:37.12 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 18:37.12 | Qwell | Go buy it - now. |
| 18:37.14 | jeev | oh whatever, i dont know anything about drugs or alcohol, i've never drank :D |
| 18:37.28 | Qwell | jeev: uh huh |
| 18:37.38 | axisys | Qwell: restart or reload to actiavte the noload changes in modules.conf |
| 18:37.42 | jeev | Fender, is that why you survive my questions? |
| 18:37.43 | Qwell | restart |
| 18:37.51 | blackhole | Can i use asterisk to setup my own SIP gateway? |
| 18:37.55 | Qwell | blackhole: sure |
| 18:37.58 | jeev | my girlfriend's older cousin says i have a chemical unbalance cause i'm always tired |
| 18:38.05 | Qwell | imbalance |
| 18:38.12 | blackhole | Qwell, What would i need else than a good broadband line, Linux and asterisk |
| 18:38.12 | jeev | yea.. |
| 18:38.16 | jeev | im' tired, what do you expect |
| 18:38.24 | Qwell | blackhole: pretty much nothing |
| 18:38.33 | Qwell | depends on what you want to do |
| 18:38.46 | axisys | Qwell: i still get this http://pastebin.com/f1826799b |
| 18:38.54 | axisys | Qwell: do I need to manually unload them ? |
| 18:39.13 | Qwell | nope, but do the lsof like it says |
| 18:39.19 | Qwell | see what it thinks is using it |
| 18:39.19 | [TK]D-Fender | axisys: "module unload chan_oss.so" |
| 18:39.23 | [TK]D-Fender | axisys: "module unload chan_alsa.so" |
| 18:39.42 | [TK]D-Fender | axisys: just a 'reload' won't do it. You'd have needed a full restart. |
| 18:39.48 | _MrSeb_ | ~pastebin |
| 18:39.48 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:39.58 | axisys | [TK]D-Fender: i did restart now |
| 18:40.02 | jeev | Fender, lets start a asterisk business, i'll supply the webhosting, you'll do everything and we'll split the money down the middle |
| 18:40.03 | axisys | Qwell: lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system /home/iqbala/.gvfs Output information may be incomplete. |
| 18:40.15 | axisys | Qwell: taht was from lsof /dev/dsp |
| 18:40.16 | Qwell | axisys: with the device |
| 18:40.21 | jeev | and only webhosting, actual cage and stuff money will have to come out of your pocket! |
| 18:40.22 | jeev | hahaha |
| 18:40.23 | Qwell | that's unrelated |
| 18:40.29 | [TK]D-Fender | axisys: "show modules like chan" |
| 18:40.39 | axisys | Qwell: that is all I saw.. |
| 18:40.42 | axisys | [TK]D-Fender: ok |
| 18:40.51 | blackhole | Qwell, Can you point me to document to setup a SIP gateway using asterisk? |
| 18:40.56 | Qwell | ~book |
| 18:40.57 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:42.10 | axisys | [TK]D-Fender: module show like chan shows this http://pastebin.com/f2b31556d |
| 18:42.10 | _MrSeb_ | [TK]D-Fender: here my sip.conf, this is the last used, in the middle I've tryed many settings of nat-qualify-reinvite, but nothing is changed... http://rafb.net/p/Vp5lo592.html |
| 18:43.33 | _MrSeb_ | [TK]D-Fender: the strange things is that the results is always the same, is like the different settings don't change the results |
| 18:44.39 | [TK]D-Fender | _MrSeb_: I've told you before, all of your ITSP accounts should be nat= no <------ |
| 18:45.01 | _MrSeb_ | [TK]D-Fender: I've tried with this too... I go to restart |
| 18:45.45 | [TK]D-Fender | _MrSeb_: And pastebin a call attempt |
| 18:45.51 | [TK]D-Fender | _MrSeb_: With SIP DEBUG <- |
| 18:46.01 | _MrSeb_ | ok |
| 18:46.25 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 18:47.15 | _MrSeb_ | [TK]D-Fender: canreinvite=no for ITSP is correct if nat=no? |
| 18:47.53 | [TK]D-Fender | _MrSeb_: Correct no matter what. |
| 18:50.46 | _MrSeb_ | [TK]D-Fender: the incoming call hangup after 11 seconds, here debug info... http://rafb.net/p/6Y2xIE96.html |
| 18:50.47 | *** join/#asterisk froy (n=troy@manuel.dublan.net) |
| 18:50.58 | axisys | [TK]D-Fender: module show like chan gives http://pastebin.com/f2b31556d |
| 18:51.51 | axisys | any other suggestion on how to get rid of the dev/dsp error ? |
| 18:51.52 | froy | So does anyone here run asterisk in a vmware virtual host? My gut reaction is that since vmware can't keep reliable time, it'd probably suck for asterisk, but I thought I'd ask... |
| 18:52.23 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:52.39 | _MrSeb_ | [TK]D-Fender: the problem hangup was done on line 471 |
| 18:52.56 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 18:53.26 | *** join/#asterisk gitguy (n=diego@adsl-128-132.click.com.py) |
| 18:53.36 | kombi | zaptel's "make b410p" will NOT compile against kernel 2.6.25, it will however against 2.6.22 (in case anyone cares..;) |
| 18:54.54 | *** part/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 18:54.55 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:59.34 | [TK]D-Fender | _MrSeb_: Contact: <sip:01119838876@192.168.0.127> <- BAD! |
| 18:59.48 | [TK]D-Fender | _MrSeb_: <--- Transmitting (no NAT) to 83.211.227.21:5060 ---> <- MORE bad! |
| 19:00.12 | _MrSeb_ | [TK]D-Fender: yes, but the configuration file is correct |
| 19:00.28 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 19:00.29 | _MrSeb_ | [TK]D-Fender: how can I correct this? |
| 19:00.54 | *** join/#asterisk boddah (n=haddob@201.86.9.115.adsl.gvt.net.br) |
| 19:01.00 | [TK]D-Fender | _MrSeb_: go test picg your host yourself. |
| 19:01.19 | [TK]D-Fender | _MrSeb_: Maybe you've got a resolution error. when in doubt, set it manually as "externip" |
| 19:02.18 | _MrSeb_ | [TK]D-Fender: ok, thanks, I do this trial with external host and check the debug info for this error |
| 19:02.37 | _MrSeb_ | [TK]D-Fender: what's picg? |
| 19:03.32 | [TK]D-Fender | PING |
| 19:04.46 | _MrSeb_ | ah ok |
| 19:05.09 | kombi | is LookUpBlacklist gone in 1.6? |
| 19:06.04 | kombi | apperantly so ("No application LookupBlacklist"), what does on do instead? |
| 19:07.16 | spokra | is there a utility to create the spa xml config file that anyone knows of? I have spc and can create the default and hand edit.. just wondering if there is a better way! |
| 19:07.59 | _MrSeb_ | [TK]D-Fender: now i go to do more trial, with externip the sip client don't ring... for now very thanks for the help, and at the next time if I don't resolve this myself |
| 19:08.11 | [TK]D-Fender | kombi: that was an ancient crap application. Make your own using AstDB. Thats all it was anyways. |
| 19:08.19 | jaytee | the way they keep deprecating commands in * sooner or later the only one left will be reload |
| 19:08.50 | kombi | Fender: will do, just happens to sit in that dialplan.. |
| 19:09.04 | [TK]D-Fender | Kobaz: [del] |
| 19:09.25 | [TK]D-Fender | jaytee: plenty of apps that should be removed / remodelled to be more efficient. |
| 19:09.32 | _MrSeb_ | bye to all |
| 19:10.32 | jaytee | [TK]D-Fender, I only need 4 commands; Startup, Status, Fixit, Shutdown |
| 19:10.35 | jaytee | :-) |
| 19:11.13 | [TK]D-Fender | jaytee: No... if you do it right, you never have to shutdown :) |
| 19:11.30 | [TK]D-Fender | jaytee: It will power itself, and fix itself infinitely! |
| 19:12.06 | jaytee | who knows, maybe * is the precursor to Skynet and not some military defense computer? |
| 19:14.04 | [TK]D-Fender | jaytee: All part of Qwell's mater plan : chan_SKinnY.so bot-NET ;) |
| 19:14.16 | jaytee | lol |
| 19:14.18 | [TK]D-Fender | master* |
| 19:15.17 | jaytee | hehe, everytime I see the word bot nowadays I think of Basshunter's Boten Ana video on YouTube. |
| 19:16.22 | jaytee | the one with the swedish lyrics subtitled in English by someone who just did what he "thought" he was saying in English. :-) |
| 19:17.31 | *** part/#asterisk [reed] (n=reed@firefox/gnu.webmaster.reed) |
| 19:18.47 | kombi | what is wrong when in addons make menuconfig app_addon_sql_mysql appears with XXX in frontß |
| 19:18.50 | kombi | ? |
| 19:19.24 | SplasPood | kombi: missing dependency |
| 19:19.31 | SplasPood | something you needed to have isn't there, so you can't enable it |
| 19:19.38 | SplasPood | possibly it cannot find the mysql libs/headers |
| 19:20.48 | kombi | SplasPood: plausible, make menuconfig also says "Depends on: mysqlclient(E)", that is completely there though, I'll look for the header files next |
| 19:21.27 | [TK]D-Fender | kombi: mysql-devel <- as well |
| 19:21.36 | kombi | that's it, right! |
| 19:22.29 | *** join/#asterisk MitchM (n=mitch@unaffiliated/MitchM) |
| 19:22.30 | *** join/#asterisk cy3o3 (n=cy@it.was.otherkids.net) |
| 19:22.37 | SplasPood | yup |
| 19:22.38 | *** join/#asterisk isamar (i=1000@voice.maxirede.net) |
| 19:22.43 | isamar | hi folkz... |
| 19:22.58 | kombi | libmysqlclient15-dev.. |
| 19:23.25 | jaytee | mysql, mysql-devel and mysql-server |
| 19:23.29 | kombi | (that's even in my wiki, shame on me) |
| 19:24.12 | [TK]D-Fender | BBL |
| 19:24.57 | kombi | that's for blame big looser? |
| 19:27.33 | anonymouz666 | Corydon76-dig: is there any way to see in func_odbc the query that was being executed? |
| 19:28.35 | MitchM | On a scale of 1 - 10 how easy would it be for a newb to convert a dual T1 POTS NT (non-pbx system) over to an asterisk POTS / VoIP system ? |
| 19:29.35 | *** join/#asterisk angryuser (n=sdfsdf@d04m-89-83-111-62.d4.club-internet.fr) |
| 19:30.00 | Corydon76-dig | anonymouz666: turn on debugging |
| 19:30.26 | anonymouz666 | core set debug and verbose |
| 19:30.31 | anonymouz666 | both 10 |
| 19:30.36 | Corydon76-dig | anonymouz666: and logger.conf |
| 19:30.40 | isamar | Anybody playing with Forking Calls? |
| 19:30.49 | kombi | thing is back in shape finally, thanks everyone, over and out |
| 19:31.10 | anonymouz666 | Corydon76-dig: thanks |
| 19:31.48 | Corydon76-dig | If debug isn't enabled in logger.conf, you can fiddle with the level all day long and it won't do anything |
| 19:32.04 | isamar | cdr is not recording forked calls info (like Dial,SIP/100&SIP/200) |
| 19:32.08 | anonymouz666 | full => notice,warning,error,debug,verbose,dtmf |
| 19:32.10 | isamar | any workaround? |
| 19:32.20 | jjshoe | isamar it's seperate calls. |
| 19:32.45 | Corydon76-dig | isamar: that's not a forked call |
| 19:32.51 | isamar | jjshoe: actually the call goes into cdr but no destination recorded :-( |
| 19:33.17 | isamar | Corydon76-dig: what's that then? |
| 19:33.26 | Corydon76-dig | isamar: parallel dialing |
| 19:33.33 | anonymouz666 | Corydon76-dig, I am missing some ,odbc or something |
| 19:33.48 | isamar | Corydon76-dig: ok. Great. But how CDR it? |
| 19:33.53 | Corydon76-dig | anonymouz666: pastebin the log |
| 19:34.12 | Corydon76-dig | isamar: I'm not aware that you can. Are you using CDRfix4, even? |
| 19:34.31 | anonymouz666 | Corydon76-dig: -- Executing [305@redir-atendente:8] Set("SIP/305-0909b668", "audiofile=") in new stack |
| 19:34.43 | Corydon76-dig | anonymouz666: PASTEBIN |
| 19:35.05 | Corydon76-dig | ~pb |
| 19:35.05 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:35.25 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 19:35.33 | isamar | Corydon76-dig: not using CDRfix4... |
| 19:36.01 | Corydon76-dig | isamar: you might want to think about starting |
| 19:36.43 | isamar | Corydon76-dig: ok.. thanks |
| 19:38.20 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 19:42.23 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 19:45.20 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:47.46 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-114-251.32-151.iol.it) |
| 19:48.09 | *** part/#asterisk MitchM (n=mitch@unaffiliated/MitchM) |
| 19:53.21 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-114-251.32-151.iol.it) |
| 19:54.22 | dr_gogeta86 | hi to alle |
| 19:54.24 | dr_gogeta86 | *all |
| 19:57.00 | *** join/#asterisk RobH (n=RobH@72-254-5-174.client.stsn.net) |
| 19:59.31 | *** join/#asterisk deeperror (n=deeperro@76.226.176.21) |
| 19:59.52 | NovceGuru | holy hell im hating trixbox with a passion |
| 20:00.34 | dr_gogeta86 | yeah NovceGuru |
| 20:04.23 | jjshoe | NovceGuru ce or pro? |
| 20:04.38 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 20:04.52 | NovceGuru | pro free |
| 20:05.03 | jjshoe | NovceGuru what's wrong? |
| 20:05.27 | jjshoe | NovceGuru feel free to pm me |
| 20:06.17 | outtolunc | green and rage should go hand in hand <G> |
| 20:11.45 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
| 20:14.04 | *** part/#asterisk boddah (n=haddob@201.86.9.115.adsl.gvt.net.br) |
| 20:14.47 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 20:21.04 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 20:37.01 | *** join/#asterisk gitguy (n=diego@adsl-128-132.click.com.py) |
| 20:37.40 | *** join/#asterisk magic_hat (n=geoffdou@h-68-164-7-249.chcgilgm.dynamic.covad.net) |
| 20:37.52 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:38.17 | magic_hat | anybody know if i can SIP register two * boxes on the same teliax account? |
| 20:39.19 | jjshoe | sure, but they will constantly battle to answer the ringing line |
| 20:39.26 | jjshoe | whoever got the last re-register will get the calls |
| 20:39.33 | *** join/#asterisk mefistofelix (n=bekset@host251-42-dynamic.23-79-r.retail.telecomitalia.it) |
| 20:39.37 | mefistofelix | hi |
| 20:39.46 | magic_hat | even if they're to a # that only exists as an extension on one of the boxes? |
| 20:40.52 | *** join/#asterisk ManxPower (n=manxpowe@164.sub-75-250-138.myvzw.com) |
| 20:41.25 | mefistofelix | is possible to stream a wav file with agi? |
| 20:41.47 | ManxPower | mefistofelix: you can do anything with an AGI you can do in the dialplan |
| 20:42.04 | mefistofelix | the wav file is not on the pbx |
| 20:42.53 | mefistofelix | i mean stream trought the agi<->pbx tcp connection |
| 20:43.01 | ManxPower | no. |
| 20:43.08 | mefistofelix | argh :( |
| 20:43.45 | mefistofelix | i generate wav files on the fly on the agi machine |
| 20:43.51 | ManxPower | Your AGI could download the file using whatever method you want into a temp directory, use STREAM FILE |
| 20:44.03 | mefistofelix | ok |
| 20:44.06 | ManxPower | Don't you mean FAST AGI? |
| 20:44.19 | mefistofelix | oh yes sorry fastagi |
| 20:44.26 | ManxPower | So that download idea won't work |
| 20:44.32 | gitguy | asterisk sucks, i just found another fucking bug |
| 20:44.38 | ManxPower | next time be SPECIFIC. |
| 20:44.50 | mefistofelix | ManxPower: i can upload no? |
| 20:45.01 | ManxPower | gitguy: gitguy1.4.21 has no bugs! *grin* |
| 20:45.10 | gitguy | ManxPower: yeah right, and i'm linus torvalds |
| 20:45.14 | ManxPower | mefistofelix: where would you upload do |
| 20:45.20 | ManxPower | to |
| 20:45.35 | mefistofelix | to the pbx |
| 20:45.39 | gitguy | ManxPower: http://bugs.digium.com/view.php?id=12810 |
| 20:45.43 | ManxPower | gitguy: I assume you are using 1.4.21. There have been many bug reports with 1.4.21 |
| 20:45.47 | gitguy | ManxPower: http://bugs.digium.com/view.php?id=12628 |
| 20:45.48 | ManxPower | mefistofelix: FTP, etc? |
| 20:45.53 | gitguy | ManxPower: http://bugs.digium.com/view.php?id=12653 |
| 20:45.57 | mefistofelix | yes something like that |
| 20:45.59 | gitguy | ManxPower: 1.4.21 has no bugs? |
| 20:46.06 | mefistofelix | would be ok in theory? |
| 20:46.31 | ManxPower | gitguy: that's a DTMF issue |
| 20:46.43 | ManxPower | gitguy: many reports of crashes with 1.4.21 |
| 20:47.34 | gitguy | ManxPower: yeah... don't get me wrong, i love asterisk, but sometimes i spend more time working around problems than using my system... |
| 20:47.50 | gitguy | ................... |
| 20:47.56 | ManxPower | gitguy: stop upgrading |
| 20:47.57 | gitguy | anyway |
| 20:48.06 | ManxPower | We never find bugs in Asterisk anymore |
| 20:48.17 | ManxPower | But we have never upgraded past 1.2.24 or something like that. |
| 20:48.18 | mefistofelix | fastagi script get called, generate the wav, upload it to the pbx(ftp?), and the issue a STREAM FILE command... |
| 20:48.32 | ManxPower | mefistofelix: if you can get the file on the server, you can play it |
| 20:49.41 | mefistofelix | ManxPower: can't i install a ftp server on the pbx? |
| 20:49.56 | jjshoe | mefistofelix yes. |
| 20:50.04 | ManxPower | mefistofelix: this is not an asterisk issue. you can get the file onto the pbx in any way you want. |
| 20:50.08 | jjshoe | I wouldn't, but sure. |
| 20:51.00 | mefistofelix | also with eagi i can't stream directly the file? |
| 20:54.50 | ManxPower | NO, EAGI does audio Asterisk -> EAGI, but not EAGI->Asterisk |
| 20:55.59 | mefistofelix | thank you ManxPower |
| 20:56.02 | *** join/#asterisk mtown_nerd (n=JHester@fileserver.ghruaim.net) |
| 20:56.46 | [hC] | anyone recall, what exactly is the /var/spool/asterisk/voicemail/<context>/<mailbox>/tmp directory for? |
| 20:57.58 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
| 20:59.53 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 21:01.54 | *** join/#asterisk electus (i=electus@asphaleia.SuxOS.org) |
| 21:04.04 | ManxPower | [hC]: you don't want a message that is in the process of being recorded in the other dir |
| 21:04.54 | jjshoe | asterisk used to do that before |
| 21:05.05 | jjshoe | or well, maybe not |
| 21:05.06 | jjshoe | BUT |
| 21:05.13 | jjshoe | the mwi would be triggered before they finished |
| 21:05.21 | jjshoe | so you would call in and hear one second of silencer |
| 21:05.30 | jjshoe | and delete the file before it finished recording |
| 21:05.33 | jjshoe | that was a cute bug |
| 21:07.35 | mefistofelix | bleah this fucking extensions.conf webadmin is impossible |
| 21:08.01 | [hC] | ManxPower: gotcha. I just found a 2.5gb message in there and wasnt sure how it could have gotten there. I thought thats what it was for but i wasnt sure. |
| 21:09.10 | ManxPower | [hC]: a bug or a crash is the only time I saw files in there for more than a very short time |
| 21:10.43 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 21:10.57 | *** join/#asterisk echelon (n=echelon@ool-182cc7a4.dyn.optonline.net) |
| 21:10.58 | echelon | hi! |
| 21:11.07 | echelon | is it possible to connect someone from a conference? |
| 21:11.15 | echelon | like.. dial-out? |
| 21:11.25 | echelon | i'm using app_conference |
| 21:13.53 | mog | call files, or originate |
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| 21:14.44 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.131) |
| 21:14.47 | frieze | is there a dialplan intro that's less incomprehensible than the one in the oreilly book? |
| 21:17.19 | echelon | mog: what? |
| 21:17.28 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:17.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:17.43 | mog | you can use call files or the originate command |
| 21:18.00 | mog | lookout at sample call in your asterisk src dir |
| 21:20.29 | echelon | mog: i don't have a src directory.. is it in /usr/doc/asterisk/ ? |
| 21:20.52 | mefistofelix | i can't get Read(myvar||8) understand the third parameter (8) |
| 21:21.04 | mog | if you dont have a src directory i have no clue how you have asterisk installed |
| 21:21.11 | mefistofelix | it always wait some time before go ahead |
| 21:21.27 | mefistofelix | even if i've entered 8 digits |
| 21:22.56 | echelon | mog: what's the file called? i'll search it |
| 21:23.16 | mog | i cant remember something like sample.call |
| 21:23.19 | unpaidbill | just google for 'asterisk originate' |
| 21:23.22 | unpaidbill | and click the first link |
| 21:28.07 | echelon | mog: thanks, found it on voip-info.org :) |
| 21:28.12 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
| 21:28.14 | echelon | this is great |
| 21:28.22 | mefistofelix | lol |
| 21:29.25 | jeev | mog, any relation to murder on the government people? |
| 21:29.39 | mog | im sorry jeev ? |
| 21:30.07 | jeev | ok nevermind! sorry |
| 21:30.13 | mog | heh thats a new one |
| 21:30.14 | jeev | leif, where you at! |
| 21:30.15 | jeev | heh |
| 21:30.17 | mog | most people assume its ff |
| 21:30.25 | jeev | murder on the government m.o.g.! |
| 21:30.26 | jeev | what's ff? |
| 21:30.33 | mog | finaly fantasy |
| 21:30.36 | jeev | oh, i never played that |
| 21:30.41 | jeev | i remember boulderdash |
| 21:30.46 | mog | mog -> mogorman -> matt o'gorman |
| 21:30.53 | jeev | no idea |
| 21:31.01 | outtolunc | thought it was bow upsidedown <G> |
| 21:31.10 | jeev | i said it once, i'll say it again.. we need to port simcopter to a multiplayer system |
| 21:31.52 | outtolunc | was joking btw |
| 21:32.02 | mog | heh' |
| 21:32.04 | jeev | wob |
| 21:32.06 | jeev | what the hell is a wob |
| 21:32.21 | jeev | world of britney? |
| 21:32.47 | jeev | is confused. where is Fender, he'll de-confuse me |
| 21:35.30 | *** join/#asterisk jsmith (n=jsmith@72.21.36.138) |
| 21:35.32 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 21:35.55 | *** join/#asterisk frieze (n=frieze@pool-98-113-86-28.nycmny.fios.verizon.net) |
| 21:36.09 | frieze | if my dialplan has no patterns remotely starting in "10" why do I get a busy signal after typing the first 0 in "1000"? It's a polycom phone if that could possibly be relevant |
| 21:36.32 | frieze | *except for the one for 1000 obviously |
| 21:36.58 | jsmith | frieze: The phone doesn't send the digits one by one... it sends *all* the digits at once |
| 21:37.08 | Strom | what does your polycom phone digitmap look like? |
| 21:37.11 | jsmith | frieze: You could adjust the dialplan on the phone itself to match your Asterisk dialplan |
| 21:37.20 | frieze | jsmith: then why send after "10"? |
| 21:37.42 | Strom | frieze: also, that's not a busy signal; it's a reorder tomne |
| 21:37.42 | frieze | Strom: that all depends. what in the name of all that is holy is a phone digitmap? |
| 21:37.47 | Strom | s/mn/n/ |
| 21:38.05 | mog | frieze, it times out |
| 21:38.06 | Strom | frieze: it's in your polycom's configuration files |
| 21:39.03 | frieze | Strom: it's set to null right now |
| 21:39.19 | frieze | "" to be precise |
| 21:39.35 | Strom | you're doing it wrong, then :) |
| 21:39.41 | frieze | <sigh> |
| 21:39.48 | Strom | it should match your asterisk dialplan |
| 21:40.02 | frieze | okay, so I have my dialplan in extensions.conf. what goes in the digitmap? |
| 21:40.17 | Strom | well, what does your numbering plan look like? |
| 21:41.10 | frieze | is there some other book besides the oreilly one with the starfish on the cover I should be reading |
| 21:41.30 | frieze | it seems like every time I come to the chan I learn that there is a different name for everything and have to start over |
| 21:41.44 | frieze | or, to be more responsive, I have no idea what a numbering plan is or would be |
| 21:41.47 | Strom | you don't have to start over just because the terminology changes |
| 21:41.59 | frieze | learn over I should say |
| 21:42.10 | Strom | a numbering plan is the generic framework for how you assign numbers; how long extensions are, what ranges they're in, etc |
| 21:42.21 | frieze | and where would I specify that? |
| 21:42.41 | Strom | it's not something you "specify" in the config files |
| 21:42.42 | frieze | or would it be implicit in my dialplan? |
| 21:43.16 | Strom | your dialplan should conform to your numbering plan, but the numbering plan isnt somethig you directly configure |
| 21:43.36 | frieze | right. |
| 21:43.51 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 21:44.08 | Strom | so...what does your numbering plan look like? |
| 21:44.44 | frieze | short answer: http://pastebin.com/d382e9a9b |
| 21:45.21 | Strom | ok |
| 21:46.01 | Strom | i would recommend not using "1" as a starting digit for any thousands block you choose to assign extensions in |
| 21:46.16 | Strom | 1 and 0 tend to be reserved digits |
| 21:46.42 | frieze | okay |
| 21:47.18 | Strom | also, I would avoid having different applications and devices on the same extension number depending on where you're calling from |
| 21:47.32 | froy | my extensions start with 0, since I don't have an operator and nothing starts with 0 on the outside. That way, I don't have to dial 9 to get out. :) |
| 21:47.38 | frieze | do I? |
| 21:48.11 | Strom | frieze: you have an inbound context with echo tests on 1000 and 2000, and then an internal context with SIP phones on 1000 and 2000 |
| 21:48.23 | frieze | doesn't semicolon comment out? |
| 21:48.28 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 21:48.30 | ghenry | Hi |
| 21:48.41 | ghenry | When you see if (option_debug > 3) in an asterisk source file |
| 21:48.50 | ghenry | How do you set the debug level? |
| 21:48.59 | Strom | frieze: yeah, but i'm just warning you should you ever uncomment those lines :) |
| 21:49.02 | ghenry | asterisk -d and debug in logger.conf doesn't sow a level |
| 21:49.17 | frieze | Strom: ah, okay |
| 21:49.21 | frieze | hmm |
| 21:49.23 | Strom | ghenry: core set debug [whatever] |
| 21:49.41 | ghenry | doh, cheers Strom |
| 21:49.46 | frieze | I think I may have done my contexts wrong |
| 21:50.15 | Strom | frieze: one problem at a time |
| 21:50.42 | Strom | fix your digit map so that the phone only sets up the call after "2xxx" and "91xxxxxxxxxx" |
| 21:51.20 | frieze | that should be in the polycom admin docs somewhere, right? |
| 21:51.37 | Strom | yes, but look at digitmap in...sip.cfg, IIRC |
| 21:51.50 | frieze | cool, thanks |
| 21:52.01 | Strom | also... |
| 21:52.07 | Strom | have a read through this |
| 21:52.09 | Strom | ~101 |
| 21:52.10 | jbot | it has been said that 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 21:52.42 | Qwell | NT? |
| 21:52.54 | Strom | nortel |
| 21:52.57 | Qwell | ahh |
| 21:54.29 | ghenry | Im still not seeing any of these if (option_debug > 3) messages |
| 21:54.38 | ghenry | even with core set debug 10 on |
| 21:54.51 | ghenry | and debug shown on the console as set in logger.conf |
| 21:55.01 | ghenry | I'm seeing lots of DEBUG colors etc. |
| 21:55.17 | ghenry | but they dont' tie in with the DBUG messages in the soruce of app_voicemail.c |
| 22:00.15 | *** join/#asterisk YossiS (n=YossiS@ool-43509b9c.dyn.optonline.net) |
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| 22:00.56 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
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| 22:03.22 | *** part/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 22:10.21 | echelon | can someone please paste their sample.call file to rafb.net/paste? |
| 22:12.44 | outtolunc | http://svn.digium.com/view/asterisk/branches/1.4/sample.call?view=markup |
| 22:13.44 | Alowishus | I have a bone-stock Zaptel 1.4.11 and Asterisk 1.4.21 install on CentOS 5.2 (64 bit)... zaptel installs and ztcfg runs just fine... but upon running Asterisk it insists, "Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection." But without telling me *anything* else in the log. Ideas? |
| 22:14.07 | Alowishus | And the error occurs regardless of what's in zapata.conf |
| 22:16.41 | echelon | what's .. Channel: Zap/1 ? |
| 22:16.48 | echelon | for zaptel device? |
| 22:17.03 | unpaidbill | outlook hazy |
| 22:17.38 | *** join/#asterisk smach (n=smach@207.35.173.122) |
| 22:18.27 | outtolunc | if you are asking that question you have not been reading the |
| 22:18.34 | outtolunc | ~thebook |
| 22:18.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 22:19.23 | outtolunc | and yes, Zap is short for zaptel which is in newer versions dahdi |
| 22:20.54 | outtolunc | things that make you go hmmm |
| 22:22.02 | echelon | outtolunc: what channel should i use for not-zaptel? :\ |
| 22:22.37 | outtolunc | unpaidbill: shake that ball again |
| 22:23.16 | outtolunc | i (nor anyone else here) has any freakin clue what channels you have on your box |
| 22:23.26 | outtolunc | we can only guess |
| 22:23.35 | outtolunc | i suggest you read the book |
| 22:23.47 | outtolunc | it will truly help you answer these questons |
| 22:23.52 | outtolunc | er questions |
| 22:23.57 | unpaidbill | Perhaps. |
| 22:24.08 | unpaidbill | the ball has spoken! |
| 22:24.18 | outtolunc | claps |
| 22:24.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:27.49 | echelon | outtolunc: i'm just using sip softphones |
| 22:28.07 | echelon | outtolunc: so there's no external devices |
| 22:28.14 | echelon | just ethernet |
| 22:30.29 | outtolunc | do you really want me to answer your question |
| 22:30.53 | outtolunc | because honestly, i think you are just trying to 'play stupid' |
| 22:31.08 | Alowishus | I want an answer to my Zaptel config error question :) |
| 22:31.11 | outtolunc | channels are discussed A LOT in the book |
| 22:31.28 | outtolunc | if you don't want to read the BOOK, at least head to the wiki |
| 22:32.12 | echelon | there's a wiki? |
| 22:32.17 | outtolunc | haha |
| 22:32.35 | echelon | it takes forever to search through the pdf |
| 22:32.39 | outtolunc | dear lord please help me <G> |
| 22:32.42 | unpaidbill | read it from start to finish |
| 22:32.43 | unpaidbill | jesus |
| 22:32.46 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 22:33.15 | outtolunc | echelon: google 'asterisk wiki' |
| 22:33.22 | Qwell | ~wikis |
| 22:33.23 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 22:33.26 | outtolunc | and POOF there are the answers |
| 22:33.34 | echelon | oh, i've been there |
| 22:33.45 | outtolunc | visit it more often.. PLEASE |
| 22:33.53 | *** join/#asterisk RoyK (n=roy@ip-183-25-149-91.dialup.ice.no) |
| 22:33.54 | unpaidbill | You may rely on it. |
| 22:34.05 | unpaidbill | It is decidedly so. |
| 22:34.26 | echelon | so how do i disconnect a call that was initiated with a call file? |
| 22:34.56 | echelon | because i'm trying to connect the outgoing call to a conference |
| 22:34.58 | unpaidbill | you let it finish on its own or use soft hangup from the console |
| 22:35.23 | unpaidbill | i'm a fan of soft hangup |
| 22:35.32 | unpaidbill | i do it to people randomly |
| 22:35.36 | outtolunc | also read up one 'timeout' |
| 22:35.40 | outtolunc | er on |
| 22:35.46 | *** part/#asterisk RoyK (n=roy@ip-183-25-149-91.dialup.ice.no) |
| 22:35.58 | unpaidbill | ah yeah, that's good too |
| 22:38.16 | echelon | also, if i bind asterisk to listen on a certain ip, would it still be able to make outgoing calls that are only accessible from a different interface? |
| 22:38.44 | echelon | because i set it so only people on vpn can call reach it |
| 22:41.53 | outtolunc | one might think that listening and speaking are 2 diff things, but also understand a conversation requires both |
| 22:42.09 | echelon | yeah |
| 22:45.40 | echelon | so.. my channel would be.. SIP/1000? |
| 22:46.34 | *** join/#asterisk RobH (n=RobH@72-254-5-174.client.stsn.net) |
| 22:46.45 | outtolunc | if you have a sip device/user 1000, then you could use that as a channel (or you can use a channel that can masq it.. such as a Local channel) |
| 22:47.25 | outtolunc | see that wasn't so hard now was it |
| 22:47.26 | echelon | now i'm more confused.. i'm just trying to connect the call to an extension |
| 22:47.38 | echelon | i don't have any users registered |
| 22:47.49 | unpaidbill | ! |
| 22:47.59 | outtolunc | haha |
| 22:48.16 | echelon | i don't need to :P |
| 22:48.30 | echelon | the extension is to Conference() |
| 22:48.40 | echelon | and anyone can access the confernece |
| 22:48.57 | outtolunc | obviously not <G> |
| 22:49.30 | echelon | what? |
| 22:49.38 | outtolunc | huh? |
| 22:49.48 | outtolunc | see two can play that game |
| 22:49.54 | unpaidbill | haha |
| 22:50.09 | *** join/#asterisk rift0r (i=rift@420nugs.info) |
| 22:50.09 | echelon | what do you mean "obviously not"? |
| 22:50.13 | outtolunc | echelon: it is really quite simple |
| 22:50.20 | echelon | ok? |
| 22:50.24 | outtolunc | if you have created 'confs' |
| 22:50.38 | *** part/#asterisk Alowishus (n=jpenix@adsl-69-109-156-138.dsl.sndg02.pacbell.net) |
| 22:50.43 | outtolunc | they are there for the any channel that come into your dialplan pointed at them |
| 22:50.48 | echelon | Conference() doesn't use any conf files |
| 22:50.52 | rift0r | Can someone recommend a nice, sturdy, functional higher end sip or iax hardphone that works well with asterisk. |
| 22:50.59 | unpaidbill | echelon this is how i connect a person to a MeetMe conference from AMI |
| 22:51.00 | unpaidbill | Action: Originate\nChannel: Local/$num\@officeld\nApplication: MeetMe\nData: $from|cd\nAsync: true\n\n |
| 22:51.02 | rift0r | i don't need video conf or anything |
| 22:51.20 | echelon | unpaidbill: AMI? |
| 22:51.21 | rift0r | maybe in the $100-$150 max range |
| 22:51.22 | unpaidbill | $num = the number of the person i want to be in the conference, $from = the meetme room # |
| 22:51.22 | rift0r | per phone |
| 22:51.30 | echelon | i was just trying to use a calll file |
| 22:51.44 | unpaidbill | yes adam, i can. |
| 22:51.51 | unpaidbill | polycom ip 330 |
| 22:52.15 | unpaidbill | the manager interface |
| 22:52.22 | rift0r | how much do those run bill |
| 22:52.31 | unpaidbill | i think you can use the same commands in .call files |
| 22:52.36 | unpaidbill | 110 ish |
| 22:52.45 | unpaidbill | maybe a little more, maybe a little less |
| 22:52.51 | rift0r | k |
| 22:52.52 | rift0r | good |
| 22:52.53 | rift0r | thx |
| 22:52.56 | unpaidbill | they look nice too |
| 22:53.02 | unpaidbill | http://www.newegg.com/Product/Product.aspx?Item=N82E16876129004 |
| 22:53.57 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 22:53.58 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 22:55.25 | rift0r | ooh the 550 is nice |
| 22:55.26 | rift0r | heh |
| 22:56.45 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 23:00.35 | *** join/#asterisk marlow (i=marlow@sleipner.tuxbox.ie) |
| 23:02.10 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 23:05.54 | echelon | oh, i just realized something |
| 23:06.26 | echelon | if the asterisk is behind a firewall, it wouldn't be able to dial-out would it? |
| 23:06.34 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 23:06.47 | echelon | bah, i'll just try it |
| 23:07.05 | echelon | it's going to make the call through a proxy, so it shouldn't matter |
| 23:09.15 | unpaidbill | look at the 670 rift |
| 23:09.17 | unpaidbill | it's way nicer |
| 23:09.44 | echelon | so how is the asterisk manager accessed? |
| 23:09.50 | unpaidbill | you could have it display the latest post on supertangas if you wanted! |
| 23:10.22 | unpaidbill | http://www.voip-info.org/wiki/view/Asterisk+manager+API read that echelon |
| 23:10.34 | echelon | i read that |
| 23:10.45 | unpaidbill | short version: set up manager.conf, telnet to port 5038, type in commands |
| 23:10.45 | echelon | just tells you to edit manager.conf |
| 23:10.57 | unpaidbill | Opening a Manager Session and Authenticating as a User |
| 23:11.03 | unpaidbill | that header describes logging in |
| 23:11.34 | echelon | telnet? ^_- |
| 23:12.55 | echelon | isn't there something i could do from cli instead? |
| 23:18.26 | x86 | is 1.6.0-beta9 stable enough for production use? |
| 23:18.35 | x86 | I've not tried it out yet at home |
| 23:19.57 | *** part/#asterisk Mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:21.42 | x86 | hmm... also, can I use libpri-1.4.4 with asterisk-1.6.0-beta9? |
| 23:22.00 | x86 | and zaptel-1.4.11 with asterisk-1.6.0-beta9? |
| 23:24.23 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 23:25.29 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:28.04 | ThoMe | is it posible pickup a queue? |
| 23:28.05 | Strom_C | x86: it's a BETA |
| 23:28.11 | Strom_C | don't put beta into production |
| 23:28.20 | ThoMe | pickup(queue/bla) ? |
| 23:28.23 | x86 | Strom_C: well, quasi-production ;) |
| 23:28.29 | x86 | Strom_C: it's at home heh |
| 23:28.59 | rift0r | unpaidbill i got them to get the 550\ |
| 23:29.02 | rift0r | 2 of them |
| 23:29.02 | rift0r | =D |
| 23:29.05 | rift0r | for me to play wit |
| 23:29.06 | rift0r | h |
| 23:29.16 | rift0r | found them for 200 on ebay |
| 23:29.39 | unpaidbill | sweet, now go look at the 670 and pine over it |
| 23:29.42 | unpaidbill | hehe |
| 23:30.32 | ThoMe | hm |
| 23:30.33 | ThoMe | [Jun 25 01:29:55] NOTICE[20004]: app_directed_pickup.c:159 pickup_exec: No target channel found for Queue/hauptnummer. |
| 23:30.37 | ThoMe | is it not posoble? |
| 23:30.43 | ThoMe | try this: exten => 77,1,Pickup(Queue/hauptnummer) |
| 23:30.44 | ThoMe | :-( |
| 23:33.47 | echelon | unpaidbill: what if Local/$num\@officeld is an external sip address? |
| 23:33.48 | Strom_C | ThoMe: what are you trying to do, exactly |
| 23:34.48 | ThoMe | Strom_C: have a Queue called "hauptnummer" my agends logon this. now i would like, if the agent not login and incomming a call |
| 23:34.52 | wonderworld | hey, i try to compile chan_mobile. i followed these instructions -> http://www.chan-mobile.org/?page_id=5 but the patch can't find all blocks to patch and the make fails. |
| 23:35.01 | ThoMe | then pickup this call to the agent/user |
| 23:35.04 | unpaidbill | set up an extension in the officeld context and set set $num to that extension |
| 23:35.28 | unpaidbill | or just put in SIP/place/number or whatever you need |
| 23:36.15 | echelon | place/number? |
| 23:36.28 | unpaidbill | yeah like SIP/myvoiptelco/phonenumber |
| 23:36.37 | unpaidbill | or SIP/devicelistedinsip.conf |
| 23:37.41 | echelon | so this should be fine... SIP/sipphone.com/18004664411 ? |
| 23:38.54 | Strom_C | no. |
| 23:38.59 | Strom_C | echelon: go read the book |
| 23:39.00 | Strom_C | please |
| 23:39.01 | Strom_C | ~book |
| 23:39.02 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:39.24 | echelon | yeah yeah , what specifically? |
| 23:39.31 | unpaidbill | the whole thing |
| 23:39.33 | Strom_C | the whole thing |
| 23:51.14 | *** join/#asterisk shido6 (n=shido6@74-130-124-83.dhcp.insightbb.com) |
| 23:51.28 | ThoMe | how i can looking but cdr_mysql is active? |
| 23:56.04 | x86 | Strom_C: but will libpri-1.4 and zaptel-1.4 work ok with asterisk-1.6? |
| 23:56.17 | Strom_C | I wouldn't count on it |
| 23:57.36 | ThoMe | Strom_C: iss cdr_mysql in addons? |
| 23:57.42 | Strom_C | ThoMe: yes |
| 23:58.12 | *** join/#asterisk [Akemi] (n=akemi@206-248-133-30.dsl.teksavvy.com) |
| 23:59.33 | *** join/#asterisk kmshanah (n=kmshanah@cubit.disenchant.net) |
