| 00:01.13 | Qwell[] | convergence technologies...with trixbox... |
| 00:01.19 | thock | anyone have an idea what would cause this? |
| 00:01.20 | thock | Write to 64 failed: Unknown error 500 |
| 00:01.20 | thock | Short write: 0/5 (Unknown error 500) |
| 00:01.21 | hads | heh |
| 00:03.34 | *** join/#asterisk znoG (n=gs@205-17-235-201.fibertel.com.ar) |
| 00:06.13 | *** part/#asterisk angom_w (n=angom@red-corp-201.130.138.29.telnor.net) |
| 00:06.26 | *** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk) |
| 00:06.34 | *** join/#asterisk rpm (n=none@128-64-151-2.priorityevents.net) |
| 00:06.42 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 00:06.56 | rpm | im having some problems with a x100p, i can dial out fine, but incoming calls are all static and disconnect right away |
| 00:10.23 | P-NuT | hmm, probably broken, send it to me, I'll give you $10 |
| 00:10.26 | P-NuT | deal? |
| 00:10.31 | P-NuT | LOL |
| 00:10.39 | P-NuT | anyway, what country? |
| 00:12.34 | generalhan | ~generalhan |
| 00:12.36 | jbot | i guess generalhan is THE MAN |
| 00:13.01 | generalhan | jbot, no, generalhan is a big dumb retard ... why is he here anyway ? |
| 00:13.04 | jbot | okay, generalhan |
| 00:13.12 | generalhan | ~generalhan |
| 00:13.13 | jbot | well, generalhan is a big dumb retard ... why is he here anyway ? |
| 00:13.21 | coppice | ~george bush |
| 00:13.25 | jbot | methinks george bush is eehhh..ehhhh...dunno, or a short-dicked, draft dodging, weed smoking, rubber bellied hillbilly |
| 00:13.25 | generalhan | that sounds about right |
| 00:13.38 | thock | if i'm showing something stating that "HOOKSTATE(FXS only): OffHook" and this number is constantly opening a channel and calling my DID |
| 00:13.41 | thock | can i do something to put it on hook |
| 00:13.44 | thock | or set those values somewhere? |
| 00:13.46 | coppice | that sounds about right |
| 00:13.50 | generalhan | lol |
| 00:14.15 | generalhan | s/lol/LOL |
| 00:14.24 | generalhan | i can never get that to work for me |
| 00:14.29 | generalhan | jbot HATES me |
| 00:14.31 | jbot | yes. |
| 00:14.33 | generalhan | see |
| 00:14.34 | generalhan | lol |
| 00:15.19 | generalhan | jbit: why are you so mean to me ? |
| 00:15.24 | generalhan | jbot: why are you so mean to me ? |
| 00:15.26 | jbot | generalhan: what are you talking about? |
| 00:15.39 | generalhan | jbot: forget it then ... im outta here |
| 00:15.39 | jbot | i didn't have anything called 'it then ... im outta here' to forget, generalhan |
| 00:15.56 | *** join/#asterisk withaynoti (n=withayno@c-68-49-245-183.hsd1.dc.comcast.net) |
| 00:16.20 | *** part/#asterisk withaynoti (n=withayno@c-68-49-245-183.hsd1.dc.comcast.net) |
| 00:19.08 | generalhan | ebonify generalhan |
| 00:19.19 | generalhan | aww that would have been cool |
| 00:19.24 | generalhan | abuse generalhan |
| 00:19.42 | generalhan | where is everyone ??? |
| 00:19.42 | Qwell[] | generalhan: need a / at the end too :P |
| 00:19.49 | generalhan | ohh |
| 00:20.23 | generalhan | i wanted the 'ebonify' to work |
| 00:20.27 | generalhan | that sounds pretty good ! |
| 00:20.57 | generalhan | ~abuse generalhan |
| 00:20.58 | jbot | ACTION smacks generalhan across the face. "Take that, sucker!" |
| 00:21.06 | pjchilds | ~spank generalhan |
| 00:21.08 | jbot | ACTION bends generalhan over his knee and tatoos 'ibot' on generalhan's pasty white buttocks. |
| 00:21.08 | generalhan | haha jbot said "sucker" |
| 00:21.26 | generalhan | hey ... hes good ... knows i have a pasty white buttocks ! lol |
| 00:21.33 | generalhan | ~ebonify generalhan |
| 00:21.55 | generalhan | whats the good one ... ~larf or ~lart or something like that |
| 00:22.06 | coppice | ~lart generalhan |
| 00:22.18 | coppice | ~lart generalhan |
| 00:22.32 | generalhan | wow ! |
| 00:22.33 | coppice | ~lart generalhan |
| 00:22.35 | generalhan | thats a good one ! |
| 00:22.43 | generalhan | lol |
| 00:22.45 | coppice | ~lart generalhan |
| 00:22.48 | Qwell[] | ~lart myself |
| 00:22.51 | generalhan | AHHHHHH |
| 00:22.53 | coppice | nasty |
| 00:23.06 | coppice | ~lart generalhan |
| 00:23.08 | generalhan | jbot: no, i REFUSE to use Outlook Express ... and you cant make me |
| 00:23.17 | pjchilds | A Luser Attitude Readjustment Tool (abbreviated LART) is "something large, heavy and painful, used to respond appropriately to particularly annoying lusers." A LART can by definition be anything from a pat on the back, to a small tactical nuclear attack, depending on how much adjustment the attitude needs. Other names for a LART include cluebat, clue stick, clue-by-four and Board of Education. |
| 00:23.20 | Qwell[] | AOL cds explode at 700 RPM |
| 00:23.46 | Qwell[] | ~cluebat |
| 00:23.48 | jbot | *WHACK* *WHACK* *WHACK* |
| 00:23.53 | generalhan | WOah |
| 00:23.55 | generalhan | lol |
| 00:24.05 | generalhan | jbot: you rock .. and you know it |
| 00:24.05 | jbot | generalhan: aw, gee |
| 00:24.06 | coppice | the entire AOL seems to have self destructed |
| 00:25.03 | generalhan | jbot: bullshit |
| 00:25.04 | jbot | well, bullshit is If you want to speak bullshit, please go to #debian.bullshit. sdf dflkj Linux sucks sfg yo momma dfg #debian.bullshit |
| 00:25.20 | generalhan | WOAH ! hahaha |
| 00:25.23 | generalhan | good 'ole jbot |
| 00:26.24 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 00:27.24 | generalhan | hmm that sux that some of the commands i found dont work |
| 00:27.31 | generalhan | ~ebonify I'm going to the store to buy some cigarettes |
| 00:28.31 | *** join/#asterisk Noky (n=jZcript2@200.68.127.100) |
| 00:29.27 | *** part/#asterisk Noky (n=jZcript2@200.68.127.100) |
| 00:33.50 | philv | Heh, I love the i2004... seems the firmware from early models is as buggy as shit |
| 00:36.54 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 00:39.09 | *** join/#asterisk Flauto (n=zhao@adsl-75-5-128-101.dsl.chcgil.sbcglobal.net) |
| 00:39.14 | Flauto | hi all |
| 00:40.24 | generalhan | ~lart generalhan |
| 00:40.30 | generalhan | lol |
| 00:40.31 | generalhan | im out guys |
| 00:40.33 | generalhan | have a good one ! |
| 00:40.39 | generalhan | ~lart jbot |
| 00:40.47 | generalhan | HAHAHAHAHAHAHAHAHAHA |
| 00:42.58 | Flauto | anyone here is using icall.com? |
| 00:43.38 | Flauto | i am wondering if anyone is able to use icall.com with asterisk |
| 00:44.19 | Flauto | icall.com offers free calls to usa and canada |
| 00:44.20 | *** join/#asterisk CrashHD (n=crashhd@c-67-182-167-222.hsd1.ca.comcast.net) |
| 00:44.30 | Flauto | it would be nice to be able to use it with asterisk |
| 00:44.30 | CrashHD | anyone know how to set the timezone on the astra 9133 phones? |
| 00:54.21 | thock | can anyone tell me how to get some debug info for an E&M wink based t1 line that's not accepting incoming calls? |
| 00:54.58 | *** join/#asterisk BZBW (i=BZBW@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 00:55.23 | CrashHD | you tried debug level ## from command line? |
| 00:55.31 | thock | that's for asterisk though, right? |
| 00:55.39 | CrashHD | I mean the asterisk CLI |
| 00:55.54 | CrashHD | zttool |
| 00:55.55 | CrashHD | ztcfg |
| 00:55.57 | CrashHD | zttest |
| 00:56.01 | CrashHD | from the zapata dir |
| 00:56.32 | thock | could ou refresh my memory on what zttest does? |
| 00:59.30 | CrashHD | tests the interface |
| 00:59.40 | CrashHD | should be 99.87% or higher |
| 00:59.42 | CrashHD | never lower |
| 01:00.08 | *** join/#asterisk terrapen (n=cjs@mauritius.island.nu) |
| 01:00.23 | thock | it's only opening the psudo interface |
| 01:00.29 | thock | can you select it on a specific channel or device? |
| 01:00.36 | CrashHD | I'm not sure |
| 01:00.41 | CrashHD | make sure your modules are loaded that you need |
| 01:00.44 | CrashHD | modprobe -l |
| 01:01.05 | CrashHD | make sure zaptel is in your init.d or you are manually loading your modules |
| 01:01.18 | CrashHD | I think it's make config |
| 01:01.21 | CrashHD | for zaptel |
| 01:01.25 | CrashHD | you'd have to check me on that though |
| 01:12.54 | *** part/#asterisk terrapen (n=cjs@mauritius.island.nu) |
| 01:13.02 | *** join/#asterisk VoIP4ever (n=myhome@PX65.neoline.com.br) |
| 01:13.16 | VoIP4ever | hi |
| 01:13.20 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 01:13.32 | VoIP4ever | anyone from brazil here? |
| 01:15.40 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 01:15.45 | SkramX | sorry yo |
| 01:16.37 | *** join/#asterisk razu (n=razu@80-235-89-115-dsl.prn.estpak.ee) |
| 01:17.05 | VoIP4ever | does anyone have articles, books or others about asterisk using as virtual pabx? |
| 01:17.23 | Flauto | anyone uses icall.com with asterisk? |
| 01:17.41 | SkramX | VoIP4ever: I do it for income.. what's your question? |
| 01:22.21 | VoIP4ever | I'm doing a academic work with asterisk and I need more sources to use as foundaments |
| 01:22.24 | VoIP4ever | do you understand? |
| 01:22.55 | SkramX | foundaments? |
| 01:23.05 | SkramX | You are looking for learning resources? |
| 01:24.07 | VoIP4ever | it's that! I'm sorry for my poor english :) |
| 01:24.31 | SkramX | No problem :) |
| 01:24.36 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 01:24.57 | SkramX | Uhmm... there are a couple books (written in English) about asterisk which I am sure you could learn from.. |
| 01:25.07 | VoIP4ever | nice |
| 01:25.34 | VoIP4ever | can you tell me how a get them? |
| 01:25.39 | SkramX | One is published by Oreilly.. |
| 01:25.45 | SkramX | Google, maybe even froogle :) |
| 01:25.49 | SkramX | "Asterisk book" |
| 01:25.57 | VoIP4ever | i've oreilly |
| 01:26.06 | SkramX | i have never read any of them |
| 01:26.19 | SkramX | I started using asterisk and learned much of what I know before the books were published |
| 01:26.39 | VoIP4ever | cool |
| 01:27.37 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 01:28.06 | VoIP4ever | but in a academic work, a I need many resources, so I'm trying to find anything more |
| 01:28.18 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 01:30.06 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 01:38.08 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe196.t-net.net.ve) |
| 01:38.12 | *** join/#asterisk _agt (n=hyperion@cpe-72-226-231-51.rochester.res.rr.com) |
| 01:39.05 | epablo | hi people |
| 01:39.40 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe196.t-net.net.ve) |
| 01:39.59 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe196.t-net.net.ve) |
| 01:39.59 | Flauto | hi eq |
| 01:40.02 | Flauto | ep |
| 01:40.04 | Flauto | sorry |
| 01:40.14 | epablo | How are you doing? |
| 01:40.22 | Flauto | good, thanks, yourself? |
| 01:40.41 | epablo | Fine.. |
| 01:40.44 | epablo | thanks |
| 01:41.12 | epablo | I've been busting my brain all day with a "routing" problem. |
| 01:42.22 | epablo | I have an asterisk and a gnugk running on one box.. With 2 Ip's and 2 if. But I need asterisk to register on the gk with it's IP and not the other one.. Any ideas? |
| 01:42.38 | *** join/#asterisk batphone (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
| 01:42.41 | batphone | greetings asterisklings |
| 01:42.48 | epablo | hi |
| 01:43.21 | batphone | im trying to put together some php interface that just displays some output from simple ami commands |
| 01:43.39 | batphone | my preg_replace is a little lacking |
| 01:43.50 | batphone | anyone here bored enough not to tell me to try #php? |
| 01:43.52 | *** join/#asterisk john_van_ostrand (n=john@CPE0050baaa8222-CM014300001579.cpe.net.cable.rogers.com) |
| 01:44.00 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 01:44.35 | batphone | # Agent/8731 (Unavailable) has taken no calls yet |
| 01:44.37 | batphone | # Agent/8732 (Not in use) has taken 1 calls (last was 6094 secs ago) |
| 01:44.52 | batphone | im trying to parser this |
| 01:44.56 | john_van_ostrand | A TDM400P card is giving me problems. Does anyone get feedback when forwarding calls from zap to zap? |
| 01:44.58 | P-NuT | hey all, will asterisk handle 2 x100p cards at the same time? |
| 01:45.13 | P-NuT | coomuter says no? |
| 01:45.32 | epablo | I don't see why not |
| 01:45.38 | batphone | P-NuT: i have 4 te405p cards in a box |
| 01:45.40 | john_van_ostrand | I've had two working. It was tough. I had to make sure that they got distinct IRQs. |
| 01:45.52 | epablo | If you have an IRQ for each one |
| 01:46.08 | batphone | john_van_ostrand: epablo: its a little more complicated than that |
| 01:46.14 | john_van_ostrand | And it helps if the IRQs are not in use by another device too. |
| 01:46.21 | batphone | you have to make sure there is not a bottleneck on your pci bus |
| 01:46.33 | batphone | you can put the cards on their own irq and still have crappy sounds |
| 01:46.37 | Mavvie | batphone: I had two, but the third E1 on the second card acted strange on it. |
| 01:46.53 | batphone | its a matter of allocating bandwidth across the bus |
| 01:47.15 | batphone | im using the PDSMA from supermicro with some pretty good luck |
| 01:47.22 | john_van_ostrand | Hold on. Aren't we talking about 128kbits for two channels? |
| 01:47.39 | batphone | zttest with 4 cards shows 100% about 95% of the time |
| 01:47.49 | epablo | Well I talked to digium for a similar setup a couple cards and they said: np |
| 01:49.41 | P-NuT | hmm.. |
| 01:49.49 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 01:50.06 | *** join/#asterisk NormSteel (n=nathank@69.17.44.81) |
| 01:50.18 | P-NuT | what is the cheapest option for 2 port FXO cards? |
| 01:51.12 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 01:51.53 | NormSteel | why do i keep getting a "dereferencing pointer to incomplete type" error when compling zaptel.c? |
| 01:52.20 | *** join/#asterisk iq (n=iq@71-215-58-212.omah.qwest.net) |
| 01:52.50 | philv | OH GOD |
| 01:52.57 | philv | I'm ready to pitch this i2004 out the window. |
| 01:55.25 | Qwell | P-NuT: TDM400P |
| 01:55.44 | Qwell | NormSteel: error, or warning? |
| 01:56.18 | P-NuT | Qwell, how much are they? |
| 01:56.25 | Qwell | like $200 |
| 01:56.33 | Qwell | telephony ain't cheap...get used to it |
| 01:56.47 | NormSteel | qwell: lots of errors and warnings |
| 01:59.55 | P-NuT | ok ok, calm down. |
| 02:00.01 | P-NuT | How much to ship it to Oz? |
| 02:00.18 | hads|home | P-NuT: Buy one in au. |
| 02:00.42 | hads|home | Then it will be A-ticked. |
| 02:03.44 | X-Rob | And dont' buy an X100 |
| 02:03.50 | X-Rob | they're the goats scrotum. |
| 02:05.15 | Qwell | X-Rob: Do I want to know? |
| 02:05.22 | X-Rob | Qwell, no 8) |
| 02:05.28 | X-Rob | The X100's are evil in .au |
| 02:05.29 | P-NuT | k |
| 02:05.30 | Qwell | I usually regret asking that question |
| 02:07.12 | *** part/#asterisk VoIP4ever (n=myhome@PX65.neoline.com.br) |
| 02:07.15 | speedwagon | x100p are evil every where not just in AU |
| 02:08.04 | hads|home | True |
| 02:09.59 | Qwell | X-Rob: alzheimer's |
| 02:10.07 | X-Rob | Qwell, heh |
| 02:10.58 | speedwagon | X-Rob, it's cold down were your at now. You have to watch out for those colds.. |
| 02:11.17 | X-Rob | S'not a cold |
| 02:11.30 | X-Rob | it's some nasty vomit-causing brain-fogging virus |
| 02:11.44 | X-Rob | Xrob.BrainFuck.AB |
| 02:11.44 | speedwagon | Ahh virus stuff |
| 02:11.52 | X-Rob | (AirBorne) |
| 02:12.50 | X-Rob | heh |
| 02:12.55 | X-Rob | unlikely |
| 02:14.13 | hads|home | It's bloody cold here. It was -3 deg C this morning. |
| 02:14.32 | X-Rob | I dunno what it is here |
| 02:14.42 | X-Rob | I've turned off all the airconds and am sitting in the computer room |
| 02:14.48 | speedwagon | ~ktmb |
| 02:14.52 | X-Rob | I'm guessing about 15c |
| 02:15.00 | X-Rob | and it's raining |
| 02:15.01 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 02:15.05 | X-Rob | which usually isn't that bad |
| 02:15.07 | speedwagon | ~weather ktmb |
| 02:15.20 | X-Rob | ~weather myplace |
| 02:15.26 | hads|home | lol |
| 02:15.27 | X-Rob | 8) |
| 02:15.48 | speedwagon | I was just wondering what it was here... 27.2 c |
| 02:15.54 | speedwagon | 81.0 sounds better |
| 02:16.01 | hads|home | X-Rob: You in BNE? |
| 02:16.21 | X-Rob | ~weather ybgl |
| 02:16.32 | X-Rob | pfft. it says it's th ere |
| 02:16.37 | X-Rob | hads|home, Gladstone. |
| 02:16.43 | X-Rob | 500km north of BNE |
| 02:16.52 | hads|home | Cool. |
| 02:17.48 | X-Rob | ~weather 94381 |
| 02:17.54 | X-Rob | pfft. |
| 02:19.04 | X-Rob | http://www.nws.noaa.gov/tg/siteloc.shtml that works |
| 02:19.32 | Qwell | it thinks it's a zipcode :p |
| 02:19.39 | X-Rob | aaah |
| 02:20.04 | X-Rob | http://mirror.bom.gov.au/products/IDR233.shtml |
| 02:20.08 | X-Rob | thats our radar image |
| 02:20.10 | Qwell | ~weather ybgl |
| 02:20.15 | Qwell | wtf |
| 02:20.30 | Qwell | now I know he's full of shit :p |
| 02:20.35 | speedwagon | ~weather KMIA |
| 02:21.17 | X-Rob | http://www.bom.gov.au/products/IDQ65519/IDQ65519.94380.shtml |
| 02:21.18 | speedwagon | works |
| 02:21.20 | X-Rob | 17c |
| 02:21.47 | speedwagon | you were close with your guess of 15 |
| 02:21.54 | X-Rob | bloody cold. |
| 02:22.18 | orlock | yes |
| 02:22.19 | Qwell | 17c is cold? |
| 02:22.29 | orlock | you know those standing gas heaters you see at cafes and stuff? |
| 02:22.34 | X-Rob | Qwell, for someone who lives in the tropics, yes, 17c is cold. |
| 02:22.39 | hads|home | For auzzies it is :) |
| 02:22.43 | Qwell | ~gmath 17c in f |
| 02:22.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:22.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:22.50 | orlock | we have one of those heaters _inside_ |
| 02:23.01 | X-Rob | orlock, hah. That's not bad. |
| 02:23.09 | Qwell | 62.6F...wtf |
| 02:23.11 | speedwagon | I have no heaters in my house or work place... |
| 02:23.13 | orlock | combine all the xeons and the aircon to cool the xeons, no juice left to heat the people, heh |
| 02:23.38 | X-Rob | orlock, which is why I've turned _off_ the airconds in the computer room |
| 02:23.38 | X-Rob | 8) |
| 02:23.48 | orlock | X-Rob: its good, its down the back near the fire escape/smokers corner, so whenevr we wander back in from the cold.. ahh, toasty warm |
| 02:23.59 | Qwell | wait... |
| 02:24.06 | orlock | X-Rob: heh, we did have ours turned off, but theres no through-ventilation in there |
| 02:24.07 | Qwell | orlock: You're using a gas heater...indoors? |
| 02:24.10 | X-Rob | orlock, you're in VIC are't you? |
| 02:24.12 | Qwell | umm |
| 02:24.16 | orlock | or a window |
| 02:24.19 | orlock | Qwell: yea |
| 02:24.24 | Qwell | that's like... |
| 02:24.31 | orlock | X-Rob: yup |
| 02:24.34 | Qwell | kinda dangerous? :p |
| 02:24.38 | orlock | Qwell:" apparently they are fine for indoor use |
| 02:24.47 | X-Rob | will be heading down to leongatha again in july, prolly. |
| 02:24.52 | orlock | cool |
| 02:24.57 | Qwell | besides the whole carbon monoxide poisoning thing |
| 02:25.09 | hads|home | We used to do that at a workshop I worked at untill the council told us off. |
| 02:25.15 | X-Rob | Qwell, which is why you can't get coal gas, only natural gas, in AU. |
| 02:25.49 | X-Rob | (eg, Liquefied Petrolem Gas) |
| 02:26.02 | X-Rob | very little CO1 in it |
| 02:26.09 | orlock | Hmm.. food.. what to get.. |
| 02:26.17 | hads|home | Aparently those patio heaters aren't designed for indoor use (over here at least). |
| 02:26.20 | Qwell | Appliances fueled with natural gas, liquified petroleum (LP gas), oil, kerosene, coal, or wood may produce CO. Burning charcoal produces CO. Running cars produce CO. |
| 02:26.26 | orlock | X-Rob: yeah, petrolheads who need to get by the EPA will run LPG |
| 02:26.38 | orlock | cos then they can run their blown 454 big blocks without hassles |
| 02:26.58 | X-Rob | Qwell, yeah, but not as much as the coal gas - that's what generates insane amounts of CO |
| 02:27.02 | orlock | *shrug* i'm not near it :) |
| 02:27.10 | Qwell | yes, but it still produces it :p |
| 02:27.18 | X-Rob | well yes. But not as much. |
| 02:27.24 | orlock | so do all the people in here |
| 02:27.33 | Qwell | CO is CO.. |
| 02:27.54 | speedwagon | Qwell, what do you do then with the fireplaces that are in allot of homes that run off gas??? |
| 02:28.07 | Qwell | speedwagon: umm...open the flute? |
| 02:28.47 | speedwagon | Qwell, hummm I see... I don't have any need for a fireplace in fact not to many homes here with them...Just was wondering. |
| 02:29.01 | X-Rob | Reverse cycle airconds are the best |
| 02:29.17 | X-Rob | more than 100% efficiency is nice. |
| 02:42.49 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 02:43.13 | batphone | is there a way to really crank up the volume on a polycom chassis speaker? |
| 02:43.22 | batphone | the sip.cfg file appears to be useless |
| 02:43.56 | batphone | are there any plans to make ami output more useful? like XML maybe???? |
| 02:44.15 | batphone | i have to write 4 pages of php to parse the output for a simple web page with 1 table.... |
| 02:44.42 | batphone | test, test.. is this thing on.. |
| 02:47.31 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 02:48.42 | JoseBravo | May be anoye know how works, Billing Increment and Conection Charge in AstBill? |
| 02:50.17 | dlynes_home | JoseBravo: have you tried the astbill wiki and/or forums? |
| 02:50.41 | *** join/#asterisk loather (i=[U2FsdGV@ip70-187-180-140.oc.oc.cox.net) |
| 02:50.59 | loather | good afternoon. |
| 02:51.19 | loather | i'm trying to get asterisk running with my shiny new teliax account. anyone got a few minutes? |
| 02:59.44 | *** part/#asterisk cj (n=cjcollie@unaffiliated/daman/x-0000001) |
| 03:04.48 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 03:05.40 | *** join/#asterisk Strom_C (n=strom@m215e36d0.tmodns.net) |
| 03:05.51 | P-NuT | So if I wanted to play music while dialing an extension, what command would I be wanting? |
| 03:09.49 | hads|home | Dial :) |
| 03:12.40 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 03:17.48 | JoseBravo | dlynes_home yes |
| 03:17.52 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 03:18.10 | JoseBravo | dlynes_home AstBill don't have good docs. |
| 03:18.45 | P-NuT | thanks hads. |
| 03:18.49 | P-NuT | I had no idea. |
| 03:18.50 | P-NuT | LOL |
| 03:18.58 | hads|home | :) No problem. |
| 03:19.12 | P-NuT | another question, 2 x100p cards. Will they work with asterisk? |
| 03:19.41 | hads|home | Yes, but as everyone has told you - go for a TDM02B. |
| 03:20.08 | hads|home | You are just asking for trouble using X100 cards |
| 03:21.19 | dlynes_home | JoseBravo: probably why nobody on this channel seems to use it |
| 03:21.24 | P-NuT | tdm02b you say |
| 03:21.32 | P-NuT | I thought everyone was saying tdm400? |
| 03:21.49 | hads|home | Yes, the same as an hour ago. AU$380 from ATP |
| 03:21.54 | dlynes_home | tdm02b is a tdm400p with 2 fxo ports |
| 03:22.04 | dlynes_home | tdm400p is a card without any modules on it |
| 03:22.22 | hads|home | Oh yeah. Thanks dlynes_home |
| 03:23.20 | P-NuT | oh right. |
| 03:23.49 | P-NuT | Sorry, I'm just not made of cash and it's for a home project. That's why I keep asking about the x100ps. |
| 03:24.51 | P-NuT | so even though the TDM02b would be the CORRECT way to go, it's 5 times the price of 2 x100ps. |
| 03:24.52 | dlynes_home | P-NuT: tdm400 and tdm02 are the same card; it's called tdm400 in a general sense, and a tdm02b specifically if it has 2 fxo modules installed |
| 03:25.07 | P-NuT | k point taken. |
| 03:25.19 | hads|home | That's fine. I don't know anyone that's made of cash. Use X100 cards if you like but you WILL have problems, it's just not worth it IMHO. |
| 03:25.54 | dlynes_home | P-NuT: do you have lots of free pci slots? |
| 03:27.33 | P-NuT | yes |
| 03:27.42 | P-NuT | I have 3 |
| 03:27.50 | dlynes_home | thought you said you had lots? |
| 03:27.55 | dlynes_home | 3 isn't a lot |
| 03:28.05 | X-Rob | 3 isn't bad |
| 03:28.08 | P-NuT | umm.. I only have 4 on the board. |
| 03:28.09 | X-Rob | most pc's only have 4 |
| 03:28.12 | dlynes_home | ah |
| 03:28.23 | dlynes_home | every pc i've had has had 5 or 6 slots |
| 03:28.47 | dlynes_home | P-NuT: well, your mileage may vary |
| 03:29.02 | dlynes_home | P-NuT: but my best guess is that you might only be able to get one x100p card to work |
| 03:29.09 | P-NuT | hmmm. |
| 03:29.30 | P-NuT | has anyone on here got it to work?> |
| 03:29.39 | dlynes_home | yes, i have |
| 03:30.03 | dlynes_home | but what i'm getting at, is that if more than one of those free pci slots you've got shares an irq with something else on your motherboard |
| 03:30.08 | dlynes_home | you're going to be hosed |
| 03:30.14 | hads|home | And you will most likely have echo problems since you are in .au |
| 03:30.47 | P-NuT | ok... |
| 03:30.54 | P-NuT | what about 2 SPA3000's? |
| 03:30.56 | dlynes_home | mmmm |
| 03:30.58 | dlynes_home | pad thai :) |
| 03:31.10 | dlynes_home | P-NuT: yeah...you shouldn't have any issues with a couple of spa-3000's |
| 03:32.24 | hads|home | Pad Thai rocks |
| 03:32.40 | dlynes_home | yeah...found the most delicious thai restaurant the other day |
| 03:32.53 | dlynes_home | a couple of super cute waitresses in there, too :) |
| 03:32.56 | P-NuT | pad thai with satay sauce on it is the bomb. |
| 03:33.03 | dlynes_home | one malay girl, and one pinay, too :) |
| 03:33.10 | hads|home | Nice. My town is small - we only have one to choose from. |
| 03:33.23 | dlynes_home | ah...i think we have about 20 or so |
| 03:33.51 | dlynes_home | but then again |
| 03:33.55 | dlynes_home | we're on the west coast |
| 03:34.00 | dlynes_home | so lots of asians here |
| 03:35.05 | *** part/#asterisk loather (i=[U2FsdGV@ip70-187-180-140.oc.oc.cox.net) |
| 03:40.17 | P-NuT | my little town has 6 thai restaurants. |
| 03:40.21 | P-NuT | and 20 streets. |
| 03:40.23 | P-NuT | LOL |
| 03:49.16 | sivana | anyone here set up hylafax and iaxmodem? |
| 03:49.49 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 03:52.19 | dlynes_home | P-NuT: damn....where do you live? |
| 03:52.40 | dlynes_home | Alice Springs? |
| 03:53.09 | *** join/#asterisk bmg505 (n=leon@196.209.33.199) |
| 03:53.10 | dlynes_home | P-NuT: 6 thai restaurants and only 20 streets? |
| 03:53.21 | dlynes_home | P-NuT: shiet...is the whole town people from thailand? |
| 04:02.28 | *** part/#asterisk pythos (i=pythos@unaffiliated/pythos) |
| 04:05.13 | P-NuT | dlynes_home: No, Aussies just love thai. |
| 04:06.02 | dlynes_home | P-NuT: thai women, or thai food? |
| 04:06.30 | znoG | sivana: yup, whats the prob? |
| 04:14.59 | P-NuT | haahaa, thai food. |
| 04:16.33 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 04:22.02 | Idle | how well will a pentium 120, w/ 32 mb ram handle asterisk with 1 wildcard, and 1 sip client (wildcard has 2 fxo and 2 fxs) |
| 04:22.17 | sivana | znoG: I think I got it... I'll have to test tomorrow.. I'm exhausted :) |
| 04:23.10 | Corydon76-home | Idle: fine if you do no transcoding whatsoever |
| 04:23.52 | Idle | hm, |
| 04:24.01 | Idle | its literally doing _nothing_ else |
| 04:24.22 | Idle | 99% of calls are going to be just direct bridged between the wildcard ports |
| 04:24.52 | Idle | the other 1% will be calls between the SIP phone and work (also asterisk and sip phones) |
| 04:25.06 | dlynes_home | Idle: ulaw? |
| 04:25.16 | Idle | yes, almost indefinately |
| 04:25.28 | Idle | my sip only does ulaw and alaw |
| 04:25.33 | znoG | sivana: i don't have a high success rate with big faxes, at least using a standard modem... but with fax machines it's not too bad .. |
| 04:25.39 | Idle | (I think its alaw... G729 A and B |
| 04:26.12 | dlynes_home | Idle: don't even think about g729 on that machine |
| 04:26.15 | Idle | yea, one thing I want to do is setup fax bridging automaticly |
| 04:26.20 | Idle | I'm not :D |
| 04:26.47 | Idle | no need really, lan has enough speed, and my ISP does QOS (1 or 2 calls _at most_) via the net |
| 04:28.04 | dlynes_home | Idle: i'm actually setting up a P75 at the office with an x100p card |
| 04:28.23 | dlynes_home | Idle: going to throw it in at my boss's house for blacklisting calls |
| 04:28.23 | Idle | yea, i think thats what I ahve.... :) |
| 04:28.40 | Idle | I've been meaning to setup this box for quite some time |
| 04:28.43 | dlynes_home | Idle: so it won't actually function as a pbx |
| 04:28.50 | dlynes_home | Idle: just an overglorified call screener |
| 04:28.56 | Idle | if I could get a 700mhz machines (where the hell does anyone actually find old machines) |
| 04:29.08 | Idle | yea, this is gonna be a pbx doing minor stuff |
| 04:29.09 | orlock | bin |
| 04:29.12 | orlock | :) |
| 04:29.17 | Idle | think it can do voicemail ok? like email voicemail? |
| 04:29.35 | dlynes_home | Idle: I just got a Dual PIII-700 MHz for $300 :) |
| 04:29.43 | dlynes_home | Idle: email voicemail, you can forget |
| 04:29.50 | dlynes_home | Idle: you might be able to do regular voicemail |
| 04:30.06 | dlynes_home | Idle: just make sure you record it in wav format or something, not gsm |
| 04:30.10 | Idle | well, isn;t email voicemail just the same, cept it just emails it over? |
| 04:30.23 | Idle | I thought gsm was the native format |
| 04:30.39 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:30.51 | dlynes_home | Idle: the machine takes a huge hit on voicemail email on fast machine, never mind on something that slow |
| 04:31.03 | dlynes_home | It's the huge hit on your memory and your hard drive |
| 04:31.07 | dlynes_home | not so much the cpu usage |
| 04:31.18 | dlynes_home | not to mention your nic |
| 04:31.23 | Idle | hm |
| 04:31.52 | Idle | well, the nic isn't bad... my server has the same one, and I can get my full 7mbps downstream on it, with 10mbps half duplex |
| 04:32.29 | dlynes_home | anyways |
| 04:32.35 | dlynes_home | i'm off to watch a movie |
| 04:32.37 | dlynes_home | ttfn |
| 04:32.43 | Idle | ciao |
| 04:36.26 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 04:38.50 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:40.04 | *** join/#asterisk flujan (n=flujan@201-27-89-144.dsl.telesp.net.br) |
| 04:41.46 | flujan | guys, is possible to use chanspy specifying a channel? something like this: ChanSpy(IAX2/123456)? |
| 04:43.14 | denon | flujan: stick that channel in it's own group |
| 04:43.19 | denon | <PROTECTED> |
| 04:43.19 | denon | <PROTECTED> |
| 04:44.39 | *** join/#asterisk feld (n=feld@ruc-mwt-gw-1.ruc.mwt.net) |
| 04:44.46 | feld | hey guys. |
| 04:45.04 | flujan | denon, hum... something like ChanSpy(|g(IAX2/12345))? |
| 04:45.11 | flujan | hi feld |
| 04:45.12 | feld | anyone have a quick pointer to where I find a setting that notifies when you have a voicemail? something the phones pick up? |
| 04:45.25 | flujan | hasnewvoicemail()? |
| 04:45.34 | feld | maybe a blinking light, etc, type of thing. my email on voicemail works perfect. |
| 04:45.54 | denon | flujan: nah, not quite what I had in mind .. |
| 04:45.57 | feld | flujan: lol u make that up or are u actually a code worm? :) |
| 04:46.37 | flujan | denon, please, provide me a example... |
| 04:46.50 | flujan | denon, I grab that from the voip-info... :) |
| 04:47.11 | flujan | feld, a little bit from both... ;) |
| 04:47.13 | denon | oh, if it's on voip-info, then yeah, may work |
| 04:47.24 | denon | give it a shot, I dont use chanspy at all |
| 04:47.27 | denon | so I wouldnt know |
| 04:47.43 | flujan | denon, thanks |
| 04:47.57 | flujan | denon, what do u use to monitor clals? |
| 04:48.07 | denon | I dont |
| 04:48.19 | denon | Ive got no reason to snoop |
| 04:48.20 | flujan | :) |
| 04:48.45 | flujan | :) |
| 04:51.47 | feld | flujan: so this isn't a notifier of sorts... it's a command put into an extension in your dialplan that you configure your phones to monitor? |
| 04:52.32 | flujan | denon, I cannot understand why I should stick the channel to it's own group to make a specific spy. |
| 04:53.08 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 04:54.30 | rainkid | for some reason, when voicemail prompts for the password, it sounds like the p is missing and it is asking for 'assword'. i do not notice this in any other voice prompts. even when i use a wait statement, it does the same thing. anyone else noticing this? |
| 04:54.42 | *** join/#asterisk cryptnix (n=andrew@64.25.198.123) |
| 04:54.47 | feld | rainkid: i miss the first chunk of some of mine too |
| 04:54.54 | feld | "sterisk mail. Mailbox?" |
| 04:54.58 | rainkid | why is that? |
| 04:55.04 | rainkid | is there a fix? |
| 04:55.10 | X-Rob | Do an Answer, then a wait. |
| 04:55.16 | feld | i've yet to finish my asterisk setup yet. that's something I need to solve. |
| 04:55.17 | cryptnix | hmm, we got like 16 lines coming in via a bunch of sipura boxes into our NBX do you guys know of a rack solution that'll take the place of multiple devices? |
| 04:55.27 | cryptnix | we really don't want to get rid of the nbx system |
| 04:55.32 | rainkid | even with a wait, it still does this |
| 04:55.45 | X-Rob | Do an Answer, then a wait. |
| 04:55.49 | X-Rob | ^^, no, really. |
| 04:56.18 | feld | how long of a wait |
| 04:56.25 | feld | like 2 seconds or something? |
| 04:56.28 | X-Rob | Oh, in passwod? |
| 04:56.33 | rainkid | yes |
| 04:56.34 | X-Rob | Maybe your password.gsm file is fubar. |
| 04:56.41 | X-Rob | download the high quality ones |
| 04:56.46 | X-Rob | they're linked from www.aussievoip.com |
| 04:56.47 | X-Rob | front page |
| 04:57.12 | *** part/#asterisk flujan (n=flujan@201-27-89-144.dsl.telesp.net.br) |
| 04:57.46 | feld | another thought X-Rob -- I once saw a snippet of handy code that pointed out how to make it so when you do a "dial 9 to get out" dialplan your users wouldn't be confused because it would continue dialtone after you press 9. |
| 04:58.06 | feld | ever seen that? i lost my bookmark on that one I guess..... |
| 04:58.08 | X-Rob | Playtones(dialtone) |
| 04:58.17 | X-Rob | or playtones(dial) even |
| 04:58.25 | feld | oh maybe that was what it was |
| 04:58.38 | feld | i thought it was more complex, but i've seen so much in so little time it gets to be a blur sometimes. :) |
| 04:58.53 | hads | feld: ignorepat |
| 04:59.02 | *** join/#asterisk pc55 (i=pc55@66-233-8-212.boi.clearwire-dns.net) |
| 04:59.12 | feld | hads: what did pat ever do to me? hah..... |
| 04:59.23 | hads | Heh |
| 04:59.27 | pc55 | Can anyone help me diagnose a client will not connect using xlite issue? |
| 04:59.34 | pc55 | Or sorry, can connect, can't make call. |
| 04:59.43 | pc55 | I have console on asterisk box, and am at x-lite box. |
| 04:59.52 | feld | pc55: most likely a firewall issue absolutely not letting sip work. |
| 04:59.54 | pc55 | I don't see any errors or anything else useful |
| 05:00.08 | pc55 | feld - client hooked straight to ISP's modem with external IP |
| 05:00.22 | feld | strange. |
| 05:00.42 | feld | do more network troubleshooting first and then move to asterisk. that's how i'd approach it. |
| 05:00.44 | pc55 | x-lite just says calling... |
| 05:02.03 | pc55 | Every time I call I don't see any traffic getting to the server with ethereal other than a SIP register request. |
| 05:02.09 | pc55 | It re-registers. |
| 05:02.17 | denon | hmm, anyone know if Rob Thomas (from AussieVoip) is ever in here? |
| 05:02.41 | orlock | denon: yes |
| 05:02.48 | orlock | he was just talking :) |
| 05:02.50 | denon | orlock: what nick? |
| 05:02.56 | denon | ooh, X-Rob |
| 05:02.56 | denon | heh |
| 05:03.00 | denon | guess I shoulda figured that one out eh |
| 05:03.08 | X-Rob | Heh |
| 05:03.09 | X-Rob | That's me |
| 05:03.23 | X-Rob | And, I'm officially no longer unwell. I think. |
| 05:03.27 | denon | X-Rob: you know, I dont think you can call Gladstone "rural AU" |
| 05:03.41 | X-Rob | Well, it's pretty rural. |
| 05:03.53 | denon | bit south and a lot west, I believe |
| 05:04.13 | X-Rob | gundy = goondiwindi? Or gundagai? |
| 05:04.21 | denon | goondiwindi |
| 05:04.31 | X-Rob | 500km south and 500km west |
| 05:05.00 | X-Rob | coz gundagai is a _lot_ south, and not that far west 8) |
| 05:05.05 | denon | and not a drop of water for 500km either :) |
| 05:05.13 | X-Rob | It's raining here!@# |
| 05:05.17 | denon | oh yeah? |
| 05:05.17 | orlock | wow! |
| 05:05.34 | feld | alright, bedtime. |
| 05:05.38 | feld | later guys. |
| 05:05.52 | X-Rob | http://www.bom.gov.au/products/IDQ65519/IDQ65519.94380.shtml |
| 05:05.57 | X-Rob | 10mm of rain so far |
| 05:06.10 | denon | switch the tank :) |
| 05:07.41 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:07.45 | P-NuT | pc55: an IAX2 connection from outside is 100 times easier to get working. |
| 05:08.02 | P-NuT | you coul duse iaxcomm instead of xlite. |
| 05:11.49 | neilbags-work | anyone using asterisk with bluetooth channel? |
| 05:14.02 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 05:19.48 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:19.49 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:19.58 | CunningPike | seen dlq |
| 05:20.09 | CunningPike | ~seen dlynes_home |
| 05:20.25 | jbot | dlynes_home is currently on #asterisk (1d 1h 33m 14s). Has said a total of 62 messages. Is idling for 47m 48s, last said: 'ttfn'. |
| 05:20.29 | asterboy | ~seen myself |
| 05:20.32 | jbot | myself <n=wguru@67.63.84.74> was last seen on IRC in channel #wowi-lounge, 58d 20h 51m 33s ago, saying: 'omg yay'. |
| 05:20.51 | Sedorox | ~seen sedorox |
| 05:20.53 | jbot | sedorox is currently on #asterisk (3d 4h 40m 9s). Has said a total of 49 messages. Is idling for 2s, last said: '~seen sedorox'. |
| 05:21.02 | Sedorox | lol |
| 05:21.13 | CunningPike | It never fails........ :S |
| 05:21.24 | denon | ~seen ` |
| 05:21.26 | jbot | ` <n=ew@port-212-202-8-197.dynamic.qsc.de> was last seen on IRC in channel #openzaurus, 40d 15h 42m 45s ago, saying: 'hello'. |
| 05:21.56 | denon | ~seen |`cat /etc/passwd` |
| 05:21.58 | jbot | i haven't seen '|`cat /etc/passwd`', denon |
| 05:22.01 | denon | some things never get old :) |
| 05:22.10 | Sedorox | lol |
| 05:22.31 | Sedorox | ~seen `cat /etc/passwd | grep root` |
| 05:22.32 | jbot | Sedorox: i haven't seen '`cat /etc/passwd | grep root`' |
| 05:22.51 | Sedorox | </bad_attempts_at_expliots> |
| 05:23.10 | denon | the past decade has brought a lot of quality to irc bot coding, unfortunately :) |
| 05:23.14 | asterboy | ~jbot, dialupVoIP is this new way to use VoIP services over dialup modems. Check out Fusion Telecommunications in this article: http://www.techweb.com/showArticle.jhtml?articleID=189500404&cid=RSSfeed_TechWeb |
| 05:23.17 | jbot | asterboy: okay |
| 05:23.43 | Sedorox | jebot, useless info |
| 05:23.49 | Sedorox | jbot, useless info |
| 05:23.59 | Sedorox | ~useless_info |
| 05:24.03 | Sedorox | :/ |
| 05:24.11 | asterboy | jbot is Sederox useless info? |
| 05:24.23 | asterboy | :| |
| 05:24.35 | Sedorox | eh? no... |
| 05:24.42 | Sedorox | :p |
| 05:24.49 | *** join/#asterisk Beighto (n=Kry5ta1@adsl-71-155-143-8.dsl.scrm01.sbcglobal.net) |
| 05:25.03 | denon | asterboy: so I wonder when someone is going to tell them that a modem often has 2-300ms of latency |
| 05:25.14 | denon | doesnt matter what codec you use, that's going to be a delay |
| 05:25.50 | denon | especially funny that they use SIP, the one place they could streamline a bit |
| 05:28.04 | pjchilds | well the SIP messaging isn't going to impact on the voice quality is it :) |
| 05:28.51 | denon | well, they say they support "standard SIP phones" |
| 05:29.03 | denon | which means it's probably G729 or such |
| 05:30.05 | pjchilds | sounds like a load of market speak to me... what they have failed to get through to anyone in that article is what they offer that skype doesn't ? |
| 05:30.20 | denon | bleh, skype |
| 05:30.50 | pjchilds | isn't that like saying 'yuck... microsoft....' |
| 05:30.58 | pjchilds | or yurk... at&t.... |
| 05:31.09 | denon | nah, it's more like saying "yuck.. apple" |
| 05:31.13 | pjchilds | or geeze Siemens / Lucent / Alcatel suck ;) |
| 05:31.22 | pjchilds | yumm.... apple... |
| 05:31.32 | denon | whoa wait a second, you're comparing skype to att and lucent? |
| 05:31.42 | denon | how quickly we forget the .com boom and collapse |
| 05:32.27 | pjchilds | I thought your 'bleh, skype' comment was just a 'bashing' type comment |
| 05:32.40 | pjchilds | one of those 'not invented here' type comments ;) |
| 05:32.57 | denon | well. . |
| 05:33.08 | denon | they're very vonage-like |
| 05:33.22 | pjchilds | ;) |
| 05:33.32 | denon | why, do you work for Skype? |
| 05:33.39 | pjchilds | ah... your just trolling ;) I see... sorry I was a bit slow ;) |
| 05:34.22 | pjchilds | no but people bag skype... but most of the reasons are pretty lame... |
| 05:34.48 | pjchilds | technically its a pretty impressive bit of work... |
| 05:35.16 | pjchilds | distributed peer-2-peer meshed directory services, security, presence, https tunnelling... video, conferencing... |
| 05:35.34 | denon | well .. |
| 05:35.34 | pjchilds | network centric preference storage... |
| 05:35.45 | denon | the one thing that tends to aggrevate me with skype in particular .. |
| 05:36.13 | denon | is that its found it's way to the clueless masses .. and they have a hosed network, "yeah, I tried that voip thing .. it doesnt work very well.." |
| 05:36.34 | denon | of course, skype doesnt explain that they dont equal voip .. that voip is a technology, but the public Internet still blows |
| 05:37.23 | pjchilds | thats just the nature of trying to make a packet based network delivery a circuit switched service ;) |
| 05:37.53 | pjchilds | however the dynamic codecs that skype run are pretty cool technically too... dynamically adjusting to network conditions.... |
| 05:38.02 | denon | well, it tends to disillusion people about the technology |
| 05:38.16 | denon | making it harder for the next generation to gain support |
| 05:38.25 | pjchilds | I use skype from australia to sanfrancisco all the time and the quality is better than landline.... |
| 05:38.27 | asterboy | lol |
| 05:38.55 | orlock | we dont have landlines |
| 05:39.01 | orlock | been 100% pure voip for years |
| 05:39.07 | asterboy | I'm in a sea of disillusioned people |
| 05:39.11 | pjchilds | i find that skype-out quality is quite poor and sucks a turd most of the time.... |
| 05:39.18 | orlock | even at home i use voip now |
| 05:39.20 | stephane_ | jour |
| 05:39.37 | orlock | no landline |
| 05:39.47 | asterboy | all day long...but hey, it pays the bills |
| 05:39.51 | denon | people are always astonished when they hear that we use voip, "wow, you sound so clear!" .. uh, yeah, I use voip all the way to our call switch, connected via gig-e, then it goes to pstn |
| 05:40.06 | denon | it's skype and vonage that are giving that bad name |
| 05:40.10 | orlock | yeah |
| 05:40.16 | orlock | our voip provider is also our dsl provider |
| 05:40.18 | asterboy | buy vonage stock! |
| 05:40.21 | denon | hah |
| 05:40.24 | orlock | and they also make the dslam hardware |
| 05:40.26 | denon | asterboy: ive got a bridge to sell you |
| 05:40.43 | orlock | so they are in control of the network all the way, which is handy |
| 05:40.45 | pjchilds | I think in the long run it will be other things that gives VOIP a bad name... |
| 05:40.50 | asterboy | ya but if you buy it now, it's sure to go down |
| 05:40.56 | orlock | fibre, dsl, voip.. whatever the issue, its one place to call |
| 05:41.26 | denon | some day we'll have dedicated channels .. |
| 05:41.28 | denon | just for voice |
| 05:41.32 | denon | and switch them at a really low level |
| 05:41.35 | pjchilds | things like.... its much easier to implement really bad RFC3261 compliance than get it _right_ |
| 05:41.38 | denon | so as to avoid the packets |
| 05:41.42 | denon | and we can build a while network of it |
| 05:41.50 | denon | call it the pstn or something |
| 05:41.51 | pjchilds | ... computer geeks don't understand 99% of the technology they use... |
| 05:42.15 | orlock | our provider is also owned by one of he larger pabx manufacturers |
| 05:42.17 | orlock | :) |
| 05:42.24 | pjchilds | ... all those nice DTMF tones in INFO packets...great for getting someones phone banking details... |
| 05:42.41 | pjchilds | very limited support for SRTP... |
| 05:43.08 | pjchilds | no really good way at authenticaing a user (oh I am sip:bill.gates@microsoft.com ... for sure!!) |
| 05:43.16 | asterboy | buy Vonage stock that let's you sell now and buy back later to cover...then you get the difference. |
| 05:43.27 | denon | just short it |
| 05:43.30 | asterboy | lol |
| 05:43.31 | asterboy | yep |
| 05:44.03 | asterboy | not too many people understand that concept. |
| 05:44.29 | asterboy | so it doesn't matter if the price goes up or down...just as long as you know what way it's going. |
| 05:44.45 | asterboy | Cue in the directors |
| 05:44.48 | denon | asterboy: if you could short an IPO, I would have shorted vonage |
| 05:45.08 | asterboy | for sure...but easy to say now that we are on the downturn |
| 05:45.25 | denon | I also dont believe you can short an IPO on it's opening |
| 05:45.32 | *** join/#asterisk pc55 (i=pc55@66-233-8-212.boi.clearwire-dns.net) |
| 05:45.38 | pc55 | What does thsi mean in sip debug? |
| 05:45.38 | pc55 | <-- SIP read from 66.233.8.212:60746: |
| 05:45.38 | pc55 | --- (0 headers 1 lines)--- |
| 05:45.46 | pc55 | I get it every once in a while form a host which I can't get to work. |
| 05:48.42 | asterboy | Interesting read: http://www.washingtonpost.com/ac2/wp-dyn?pagename=article&contentId=A22099-2004Mar24¬Found=true |
| 05:51.37 | pc55 | Ok, what gives. x-lite and ethereal show traffic to my asterisk server. ethereal on asterisk side shows nothing |
| 05:51.50 | pc55 | yet, using netcat, I can pass traffic on 5060 just fine |
| 06:00.13 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-182-66.dsl.pltn13.sbcglobal.net) |
| 06:02.24 | h3x0r | anybody played with unisniff yet |
| 06:04.33 | *** join/#asterisk VictorWoon (n=victor@mail.netregy.com) |
| 06:05.08 | VictorWoon | hi |
| 06:05.14 | VictorWoon | i need help |
| 06:05.20 | VictorWoon | asterisk G 729 |
| 06:05.53 | VictorWoon | harlo |
| 06:06.44 | VictorWoon | i need help |
| 06:06.53 | pjchilds | ~questions |
| 06:06.56 | jbot | remember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html> |
| 06:07.29 | pjchilds | whoops... I thought that was the 'Just say the question' thing ... |
| 06:07.35 | VictorWoon | i already buy 2 set of G 729 codec from digium |
| 06:07.50 | VictorWoon | Total : $20 |
| 06:08.09 | VictorWoon | how many capacity i can use?? |
| 06:08.30 | pjchilds | 2 ? |
| 06:09.18 | pjchilds | amulrg01*CLI> show g729 |
| 06:09.18 | pjchilds | 0/0 encoders/decoders of 5 licensed channels are currently in use |
| 06:10.15 | VictorWoon | i can use 3 lines to call out but after 4th.. my asterisk show " codec_g729.c: Out of G.729 Encoder Licenses!" |
| 06:10.42 | VictorWoon | 0/0 encoders/decoders of 2 licensed channels are currently in use |
| 06:10.50 | Strom_C | VictorWoon, one license per call excluding pass-through |
| 06:11.11 | VictorWoon | really? |
| 06:11.14 | Strom_C | yes |
| 06:11.19 | Strom_C | read the license |
| 06:11.34 | pjchilds | ~license |
| 06:11.36 | jbot | it has been said that license is something that micro$~1 came up with ....It's evil |
| 06:11.47 | VictorWoon | hmm |
| 06:12.02 | VictorWoon | but i buy 2 licensed , i can make 3 calls |
| 06:12.15 | Strom_C | what kind of telephone set are you using? |
| 06:12.19 | VictorWoon | i using IAX sip phone to call |
| 06:12.28 | VictorWoon | IAX to SIP |
| 06:12.34 | Strom_C | um |
| 06:12.36 | Strom_C | ok? |
| 06:12.45 | Strom_C | I don't understand what that's supposed to mean |
| 06:13.36 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
| 06:13.46 | Strom_C | are you calling from an IAX phone to a SIP phone? |
| 06:14.23 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:14.31 | VictorWoon | calling from IAX sip phone > Asterisk server > external SIP server |
| 06:14.50 | Strom_C | VictorWoon, there is no such thing as an IAX SIP phone |
| 06:14.55 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:14.56 | Strom_C | it speaks IAX or it speaks SIP |
| 06:15.02 | Strom_C | they are two separate protocols |
| 06:15.18 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 06:15.33 | VictorWoon | OK |
| 06:15.44 | _m_ | And the transport protocol isn't closely related to the encoding. |
| 06:16.02 | Strom_C | yes, I know that _m_, but still |
| 06:16.24 | Strom_C | VictorWoon, in all three cases, are you placing the call the exact same way? |
| 06:16.27 | _m_ | If the external SIP Server and your phone both use the same encoding then * doesn't need to encoding anything. |
| 06:16.35 | _m_ | (and you wouldn't need a license then) |
| 06:16.51 | _m_ | s/ding/de/ |
| 06:17.19 | _m_ | Strom_C: I didn't intend to object to anything you were saying. |
| 06:17.36 | Strom_C | alright :) |
| 06:17.46 | VictorWoon | no .i' using IDEFISK to make call |
| 06:18.04 | Strom_C | VictorWoon, then that's IAX, not SIP |
| 06:18.17 | VictorWoon | Yes |
| 06:18.20 | VictorWoon | correct |
| 06:18.22 | VictorWoon | then.. |
| 06:18.52 | VictorWoon | Ii have making a routing rules |
| 06:19.16 | VictorWoon | when ppl call start with number 0.... |
| 06:19.25 | Strom_C | is the SIP endpoint also using G.729 in all cases? |
| 06:19.34 | pjchilds | _m_: are you saying that IAX using g729 to SIP using g729 would not need an encoder/decoder ? :) |
| 06:19.45 | VictorWoon | it will go though our external SIP provider |
| 06:20.06 | VictorWoon | SIP endpoint is using G 729 and G 723 |
| 06:20.29 | VictorWoon | If ppl going to local call . will go to zap trunk |
| 06:20.55 | Strom_C | VictorWoon, in the case where you are running out of licenses, are you placing all calls through the SIP provider? |
| 06:21.07 | VictorWoon | YEs |
| 06:21.39 | Strom_C | Set verbose to 5 and pastebin your console output from the first call to the point where you run out of licenses |
| 06:21.42 | Strom_C | ~pb |
| 06:21.50 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 06:23.47 | pc55 | I change the port= line on sip.conf, and it still listens on 5060 even thoguh I told it 5062 |
| 06:23.48 | pc55 | any ideas? |
| 06:23.50 | VictorWoon | I cant get your means , MR Strom |
| 06:24.06 | Strom_C | pc55, did you do a sip reload? |
| 06:24.16 | Strom_C | VictorWoon, at the console type "set verbose 5" |
| 06:24.27 | pc55 | Strom_C - did a stop now and restarted asterisk |
| 06:24.45 | Strom_C | then take the console output from that point until you run out of licenses |
| 06:24.52 | Strom_C | and paste it onto pastebin.com |
| 06:24.56 | VictorWoon | Sorry , MR Strom , i cant understand console type "set verbose 5" |
| 06:25.10 | pc55 | storm - I'm thinking my isp is blocking voip or something |
| 06:25.13 | Strom_C | VictorWoon, the asterisk console |
| 06:25.24 | Strom_C | VictorWoon, the command line interface |
| 06:25.29 | pc55 | Strom_C - I can pass traffic on 5060/udp through a netcat test. |
| 06:25.45 | pc55 | Strom_C - but my SIP traffic just keeps retransmitting according to ethereal on source box and never gets to destination. |
| 06:25.45 | Strom_C | VictorWoon, the thing you get when you type "asterisk -r" |
| 06:25.56 | VictorWoon | Ya.. ok |
| 06:25.57 | pc55 | Strom_C - however, I can register just fine. no calls |
| 06:26.30 | Strom_C | pc55, is the problem with your SIP traffic or with the RTP media stream? |
| 06:26.46 | pc55 | Strom_C - Well, I don't see the sip traffic from call intiation hitting th eothe box |
| 06:26.50 | pc55 | with ethereal or sip debug |
| 06:26.57 | pc55 | but I see it being transmitted eith ethereal on source box |
| 06:27.02 | VictorWoon | Mr Strom , u wan the the output command? |
| 06:27.17 | VictorWoon | When i making a first call |
| 06:27.17 | Strom_C | VictorWoon, listen to me |
| 06:27.25 | Strom_C | I am getting tired of explaining this again and again |
| 06:27.36 | pc55 | Strom_C - What's really strange, is using netcat, I can pass 5060 udp just fine. |
| 06:27.41 | VictorWoon | sorry Mr strom |
| 06:28.06 | Strom_C | paste the complete console output starting with the first call and ending with the point where you run out of available licenses |
| 06:28.29 | Strom_C | pc55, describe your network setup |
| 06:28.41 | pc55 | Strom_C - linux server, at isp, colo |
| 06:28.50 | pc55 | Strom_C - clinet plugged straight into ISP modem, public ip |
| 06:28.53 | pc55 | no firewall |
| 06:28.54 | pc55 | router |
| 06:28.55 | pc55 | etc |
| 06:29.06 | Strom_C | what type of client? |
| 06:29.09 | pc55 | x-lite |
| 06:29.25 | Strom_C | maybe the client is screwed up. Try a different one. |
| 06:29.49 | pc55 | which one? |
| 06:30.02 | Strom_C | sjphone perhaps? |
| 06:30.15 | pc55 | googling/installing |
| 06:30.16 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 06:30.54 | rainkid | trying to bypass voicemail passwords for internal calls. VoicemailMain(${CALLERID(number)|s) does not appear to work in my dialplan. |
| 06:30.58 | rainkid | any tips? |
| 06:31.25 | Strom_C | because you forgot the closing brace? |
| 06:31.44 | Strom_C | and it's ${CALLERID(num)} IIRC |
| 06:31.49 | rainkid | you know, that happens at 2:30 in the morning more often than i would like to admit |
| 06:32.32 | Strom_C | Don't worry...I've nubbed it up far worse |
| 06:34.27 | pc55 | Strom_C - ok, it registered |
| 06:34.39 | pc55 | Strom_C - But keeps trying at "creating outgoing cal" |
| 06:35.01 | Strom_C | are you sure that perhaps the PC isn't at fault |
| 06:35.18 | rainkid | does anyone know if page ('autoanswer') is possible with the polycom 301 phones? |
| 06:35.21 | pc55 | It's worked in the past, but no |
| 06:35.29 | pc55 | Strom_C - Other than what ethereal shows |
| 06:35.35 | pc55 | Strom_C - on both the server and client. |
| 06:35.46 | pc55 | Strom_C - packets leaving my machine never arriving on remote machine |
| 06:36.21 | Strom_C | SIP packets, or RTP packets? |
| 06:36.24 | pc55 | sip |
| 06:36.26 | Strom_C | hm |
| 06:36.35 | Strom_C | use iax2? :) |
| 06:36.48 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:37.03 | pc55 | Request: invite lines in ethereal |
| 06:37.05 | pc55 | transmitting over and over |
| 06:37.14 | VictorWoon | Jun 19 17:16:08 DEBUG[2873] chan_sip.c: Stopping retransmission on '22d729dd1a4eabc301d5978e020ed68d@192.168.100.6' of Request 103: Match Found |
| 06:37.14 | VictorWoon | Jun 19 17:16:09 VERBOSE[2870] logger.c: -- Accepting AUTHENTICATED call from 192.168.100.113: |
| 06:37.14 | VictorWoon | <PROTECTED> |
| 06:37.14 | VictorWoon | <PROTECTED> |
| 06:37.14 | VictorWoon | <PROTECTED> |
| 06:37.21 | pc55 | pastbin.com! |
| 06:37.26 | Strom_C | idiot |
| 06:37.43 | pc55 | hmmm |
| 06:37.46 | pc55 | testyourvoip.com |
| 06:37.51 | pc55 | The test agent could not connect to Boston using the standard SIP port (5060), but was able to connect and run a test to another location via an alternate port (6000). This indicates that one of the following may be true: |
| 06:37.59 | *** join/#asterisk VictorWoon (n=victor@mail.netregy.com) |
| 06:38.07 | Strom_C | VictorWoon, PASTEBIN |
| 06:38.10 | Strom_C | ~pb |
| 06:38.12 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 06:38.13 | VictorWoon | oh.. ok |
| 06:38.17 | Strom_C | jeez |
| 06:38.21 | VictorWoon | sorry |
| 06:38.28 | Strom_C | READ what I say to you |
| 06:38.44 | pc55 | Strom_C - Think it might be my ISP? |
| 06:38.56 | Strom_C | pc55, who is your isp? |
| 06:39.03 | pc55 | clearwire |
| 06:39.11 | Strom_C | never heard of them |
| 06:39.20 | pc55 | ya, small rural wireless outfit |
| 06:39.26 | VictorWoon | Yes . Mr Strom |
| 06:39.31 | pc55 | did you see what testmyvoip.com said earlier? |
| 06:39.51 | pc55 | it's so strange... netcat on port 5060 passes fine |
| 06:39.54 | pc55 | anything else it needs? |
| 06:40.02 | Strom_C | pc55, try iax2 |
| 06:40.18 | pjchilds | pc55, you run netcat on your x-lite machine? |
| 06:40.28 | pc55 | pjchilds - yes |
| 06:40.44 | pc55 | And shut down asterisk on th eother site with netcat -l -u 5060 |
| 06:40.48 | pc55 | and text passed fine |
| 06:40.54 | pjchilds | pc55, isn't netcat for TCP services.... and asterisk only supports UDP ... |
| 06:41.06 | pc55 | pjchilds - netcat -u |
| 06:41.50 | pc55 | How come asteirsk won't listen on another port? |
| 06:41.54 | pc55 | even after cahnging sip.conf? |
| 06:41.55 | pjchilds | so your x-lite client talks fine to your asterisk server, but not when asterisk is running on it ... ? |
| 06:42.06 | pc55 | pjchilds - it Registers fine |
| 06:42.19 | pc55 | <PROTECTED> |
| 06:42.20 | pc55 | <PROTECTED> |
| 06:42.27 | pc55 | But can't make a call |
| 06:42.39 | pjchilds | pc55, well in asterisk 1.2 its 'bindport=' not 'port=' |
| 06:42.40 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-182-66.dsl.pltn13.sbcglobal.net) |
| 06:43.04 | pc55 | pjchilds - let me check on that |
| 06:43.16 | darkgamer20 | what is a channel? and how many are needed for a hobby project with asterisk? |
| 06:43.17 | pc55 | pjchilds - But ethereal shows it sending the call intiating packet out |
| 06:43.35 | pjchilds | pc55, and when you do a 'sip debug' on the asterisk console you get nothing ?? |
| 06:43.41 | pc55 | pjchilds - ethereal sniffing on remote side; never arrives |
| 06:43.46 | pc55 | pjchilds - nothing but the register |
| 06:43.55 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 06:44.46 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 06:44.57 | pjchilds | ahh.... well then you have done something really foobar ;) |
| 06:45.16 | Strom_C | I'm willing to blame the PC |
| 06:45.27 | pjchilds | pc55, didn't x-lite have some sort of strange STUN, DNS PTR lookup crap and outbound proxy junk... |
| 06:45.30 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:45.32 | Strom_C | pc55, do you not have a terminal adapter or something you can try? |
| 06:46.02 | *** join/#asterisk yacyac (n=yac@202.189.231.82) |
| 06:46.04 | pc55 | storm - nope |
| 06:46.05 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 06:46.18 | pc55 | but there are others working on thsi server just fine |
| 06:46.20 | Strom_C | why do you keep calling me storm? |
| 06:46.31 | yacyac | where do i find that oreally's asterisk book to download |
| 06:46.41 | pjchilds | ~thebook |
| 06:46.47 | pjchilds | ~book |
| 06:46.48 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:46.53 | pjchilds | ~the-good-book |
| 06:46.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:46.59 | Strom_C | O RLY |
| 06:47.04 | qdk | the very good book |
| 06:47.05 | pjchilds | ~the-not-so-good-book |
| 06:47.09 | darkgamer20 | lol |
| 06:47.11 | yacyac | lol |
| 06:47.17 | yacyac | thanks pjchilds |
| 06:47.25 | yacyac | so howz everyone doing |
| 06:47.30 | Strom_C | ~the-moderately-decent-but-not-quite-as-good-as-the-original-book |
| 06:48.12 | pjchilds | pc55, why don't you do a trace with ethereal on the client side, and then either send us the tcpdump, or print the whole thing to a text file and pastebin it... |
| 06:48.18 | darkgamer20 | hey guys I am reading the book now, in chapter 2 theres a table thats says I need a less than 5 CHANNELS for a hobby system. what are channels? |
| 06:48.28 | Strom_C | darkgamer20, a channel is a single telephone call |
| 06:48.38 | Strom_C | darkgamer20, read the book more |
| 06:49.17 | pjchilds | pc55, or is that 'File -> Export -> as Plain Text file ' and select Packet Details <*> All Expanded... |
| 06:49.58 | darkgamer20 | Strom_C: oops, but there dosent seem to be an explanation that a channel is a phone call, thanks for your help |
| 06:50.11 | Strom_C | darkgamer20, then perhaps you missed that |
| 06:50.14 | SwK | xlite is easy to set up if you know what fields to put what in |
| 06:50.21 | Strom_C | I seem to recall that being explained |
| 06:50.28 | pc55 | The path to file "" does not exist |
| 06:50.30 | pc55 | when I try to safe |
| 06:50.31 | pc55 | yuck |
| 06:50.41 | pc55 | Any way to just copy/paste th thing out? |
| 06:50.41 | darkgamer20 | Strom_C: well maybe it is later on in the book, im on chapter 2 only |
| 06:51.10 | SwK | a channel is 1 call leg, not nessecarily 1 phone call (but it could be 1 phone call) |
| 06:51.20 | pjchilds | pc55, only thru the export -> plain text etc... you might need to put in a file name to save it to (foobar.txt) |
| 06:51.25 | yacyac | hey i am new to asterisk |
| 06:51.31 | yacyac | what excatly does asterisk do |
| 06:51.36 | SwK | its a PBX |
| 06:51.39 | darkgamer20 | SwK: now whats a call leg? |
| 06:51.49 | pjchilds | ~asterisk |
| 06:51.50 | jbot | asterisk is the best free PBX in the world. |
| 06:51.54 | yacyac | pbx ? |
| 06:51.56 | pc55 | pjchilds - ethereal is broken :( |
| 06:51.57 | yacyac | ok |
| 06:52.01 | Strom_C | oh god, it's nub night in #asterisk |
| 06:52.03 | SwK | a call leg is from a phone to the switch |
| 06:52.03 | yacyac | but you anyways need hardware |
| 06:52.06 | darkgamer20 | SwK: a channel is an entire PBX? |
| 06:52.16 | yacyac | then why dont get a real pbx hardware |
| 06:52.21 | pjchilds | Storm *grin* whats a nub? :) |
| 06:52.29 | yacyac | what makes asterisk different |
| 06:52.35 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 06:52.36 | yacyac | sorry for my silly question |
| 06:52.39 | Strom_C | yacyac, read the book |
| 06:52.42 | Strom_C | ~thebook |
| 06:52.44 | Strom_C | ~bok |
| 06:52.45 | Strom_C | er |
| 06:52.46 | Strom_C | ~book |
| 06:52.48 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:52.49 | yacyac | ok |
| 06:52.50 | yxa | Failed to authenticate user "polycom" <sip:polycom@202.176.210.86>;tag=144D91D4-7FFD2BD7 |
| 06:52.53 | yacyac | yeah |
| 06:52.53 | SwK | yacyac: because $5000 worth of asterisk hardware is like $20K worth of avaya or nortel |
| 06:52.53 | yacyac | lol |
| 06:53.07 | h3x0r | no more like $200k |
| 06:53.14 | yacyac | wow |
| 06:53.15 | yacyac | ok |
| 06:53.31 | darkgamer20 | SwK: thanks for you explanation |
| 06:53.34 | SwK | darkgamer20: and no 1 channel isnt really a PBX it typcally takes more then 1 |
| 06:53.44 | yacyac | but recently i found a hardware which can connect your pstn to voip |
| 06:53.46 | SwK | the PBX is just the switching until |
| 06:53.52 | pjchilds | well.... $5000 of asterisk gear is a pc designed to last 2 years and some junk... and $200,000 of nortel gear will need a oxy torch and 10 hours just to remove it from site... |
| 06:54.00 | yxa | why cant my polycom 601 register with asterisk no matter what i do? |
| 06:54.01 | yacyac | you just voip account... skype would do too |
| 06:54.04 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:54.07 | pjchilds | and it will probably still be switching your calls *grin* |
| 06:54.35 | pjchilds | yxa: probably your configuration in sip.conf ? |
| 06:54.38 | SwK | pjchilds: you've never done a forklift upgrade to a G1, G3, Opt 61 or Opt 81 have you? |
| 06:54.52 | darkgamer20 | SwK: is a channel something you rent from the telephone company? |
| 06:54.59 | yacyac | does asterisk hardware works in india |
| 06:55.03 | yacyac | btw i am from india |
| 06:55.09 | SwK | yxa: post the <reg> section of you poly's config to a pastebin |
| 06:55.15 | pjchilds | SwK, don't tell anyone but I'm a NEC guy... so I don't touch nortel gear, but I have seen a few beasts... |
| 06:55.24 | yxa | pastebin is slow. |
| 06:55.30 | orlock | pjchilds: cool. |
| 06:55.31 | yxa | i think its dead |
| 06:55.39 | orlock | pjchilds: our ISP is owned by NEC |
| 06:55.55 | SwK | darkgamer20: well yes and no... 1 channel = 1 SIP connection, or 1 POTS line, or 1 H323 connection etc |
| 06:56.20 | SwK | pjchilds: i'm a old school G1/G3 guy |
| 06:56.21 | yxa | http://pastebin.com/720297 |
| 06:56.21 | orlock | pjchilds: NEC are getting into SIP now |
| 06:56.24 | yacyac | what kind of hardware do one requires for asterisk |
| 06:56.32 | pc55 | pjchilds - best I can do. http://img161.imageshack.us/img161/3651/pack10ux.jpg |
| 06:56.56 | SwK | yacyac: any old Pentium PC will do if you just want to play with it |
| 06:57.06 | hads|home | yacyac: it depends what one wants to do with Asterisk |
| 06:57.07 | pc55 | pjchilds - image2: http://img225.imageshack.us/img225/8453/pack20gn.jpg |
| 06:57.08 | pjchilds | orlock, yeah I was testing some interconnects between Asterisk and the NEC CX6xxxx stuff that NeXtep have |
| 06:57.11 | darkgamer20 | SwK: damn this telephony stuff is complicated, lol, but I think i got it, a channel is a telephone line? |
| 06:57.30 | SwK | darkgammer20: yeah close enuff |
| 06:57.37 | yxa | Failed to authenticate user "polycom" <sip:polycom@202.176.210.86>;tag=144D91D4-7FFD2BD7. please see http://pastebin.com/720297 |
| 06:57.39 | darkgamer20 | SwK: lol ok thanks |
| 06:58.04 | SwK | darkgammer20: but with VoIP there is not "telephone line"... VoIP is NOT telephony |
| 06:58.12 | orlock | pjchilds: you in aus then? |
| 06:58.21 | yacyac | SwK , pjchilds .. what is the most advance thing you can do with asterisk ? |
| 06:58.24 | SwK | VoIP just emulates telephony on a data network |
| 06:58.33 | pjchilds | orlock, adelaide.. but I work for NEC BS in melbourne... |
| 06:58.50 | orlock | pjchilds: Ahh... Cool! |
| 06:58.50 | pc55 | pjchilds - Hmm. Changed to port 5061 and it works fine. |
| 06:58.55 | *** join/#asterisk lorinc (n=ang@caracas-0691.adsl.interware.hu) |
| 06:58.58 | orlock | pjchilds: you probably know some of the people i hassle then :) |
| 06:59.00 | SwK | yacyac: the most advanced... thats pretty hard to say, if it relates to so called phone calls asterisk can do pretty much anything |
| 06:59.21 | SwK | that pastebin isnt opening for me |
| 06:59.27 | pjchilds | yacyac, some people think meetme conferencing is pretty cool ... and if you've had to pay $80,000 for a 30 channel conference bridge it is pretty cool... |
| 06:59.35 | SwK | hah |
| 06:59.38 | SwK | no shit pj |
| 06:59.54 | hads|home | There's a few aussies around today. |
| 06:59.56 | darkgamer20 | SwK: yea i know a little about VoIP its just this asterisk things i dont understand, so a channel is from the actual phone to the thing that distributes all the calls (the switch)? |
| 07:00.08 | SwK | especially when you can build a much larger conference bridge for a fraction of that cost |
| 07:00.12 | orlock | pjchilds: love your dslams! :) |
| 07:00.13 | yxa | anyone help me with polycoms please? |
| 07:00.15 | yacyac | whats meetme conference |
| 07:00.27 | SwK | darkgammer20: yes |
| 07:00.39 | SwK | yacyac: www.voip-info.org |
| 07:00.50 | yacyac | ok |
| 07:00.58 | pjchilds | orlock, well they aren't mine... NeXtep is actually a whole different company believe it or not... |
| 07:00.59 | SwK | yza: the polycom's configs |
| 07:01.06 | yxa | SwKhttp://pastebin.com/720297 |
| 07:01.13 | SwK | yza: and if you are using the web interface on the phone I have NEVER gotten it to work |
| 07:01.31 | SwK | yxa you posted the stuff in sip.conf |
| 07:01.52 | pjchilds | orlock, NEC has 356 subsidiaries... I just work for one *grin* |
| 07:01.58 | SwK | and nuke that progress inband crap... poly's havent needed that in forever |
| 07:02.02 | orlock | pjchilds: heh, i thought nextep were 100% owned by nec? |
| 07:02.08 | pjchilds | orlock, 154,180 staff worldwide... |
| 07:02.13 | orlock | pjchilds: all the poeple i've been hassling about voip are @nec.com.au |
| 07:02.27 | yacyac | so i will need 2 asterisk server right if i want to call to usa from india |
| 07:02.42 | yxa | swk i'm trying to configure it on the phone itself |
| 07:02.43 | SwK | yacyac: not nessecarily |
| 07:02.46 | pjchilds | orlock, sure... but in Australia there is NEC Home Electronics, NEC Australia , NeXtep, NEC Business Solutions (and probably some others...) |
| 07:02.47 | yacyac | then |
| 07:02.50 | yacyac | swk |
| 07:02.51 | yacyac | then |
| 07:02.54 | orlock | heh |
| 07:03.08 | SwK | yxa: get the the ftp boot stuff from polycoms website |
| 07:03.16 | orlock | pjchilds: Nexvoice :) |
| 07:03.27 | SwK | and boot it from a (t)ftp or www server |
| 07:04.03 | SwK | yacyac: there are a ton of services you can use on the US side and just put a server in india or possibly even the other way around |
| 07:04.32 | yacyac | services like skype ? |
| 07:04.38 | SwK | and if you are just calling "site to site" (ie: US Office to India Office) you might only need a phone or multi-line ATA at one end |
| 07:05.05 | yacyac | what if i want to call multiple people in multiple cities |
| 07:05.08 | SwK | not real VoIP Services like Asterlink, cxmusa.com, et al |
| 07:05.44 | pc55 | pjchilds - Jitter, what's the best solution? It's pretty bad |
| 07:05.50 | pc55 | pjchilds - choppy anyways |
| 07:06.12 | SwK | yacyac: do you need a consultant to help you determine what you really need? |
| 07:06.34 | pjchilds | pc55, hmm... not sure... I think ethereal can analyse jitter... I read somewhere someone has ported the jitterbuffer patches into 1.2.9.1 |
| 07:06.38 | yacyac | SwK i dont need anything... i just doing it for knowladge |
| 07:06.46 | yacyac | to learn |
| 07:06.57 | yacyac | maybe understand it better |
| 07:07.07 | yacyac | maybe get something brillent out of it |
| 07:07.18 | SwK | I understand that.... you might want to start at asterisk.org and voip-info.org they have tons of info on what asterisk can do at those 2 websites |
| 07:07.47 | mitcheloc | did anyone see that verizon is suing vonage over voip patents? |
| 07:07.52 | yacyac | ok thanks buddy.. you were a great help |
| 07:07.57 | SwK | and by all means download asterisk and install it on a linux or BSD or ${SOME_OTHER_UNIX} box and have fun with it |
| 07:07.58 | *** join/#asterisk r0d3nt (n=RatMan@ip68-108-184-243.lv.lv.cox.net) |
| 07:08.00 | yacyac | and sorry to bug you all like this |
| 07:08.01 | yacyac | lol |
| 07:08.15 | yacyac | SwK which is the best distro to get it on |
| 07:08.26 | yacyac | i have used linux and bsd both |
| 07:08.35 | yacyac | so what do you think is best |
| 07:08.46 | yacyac | any recommmandation |
| 07:10.21 | pjchilds | pc55, http://www.laimbock.com/asterisk/ has some RPMs with jitterbuffer merged in... (plus other stuff...) |
| 07:10.39 | pc55 | pjchilds - any docs? |
| 07:10.54 | SwK | pretty much your choice... legacy PSTN (PRI, E&M, POTS) interfaces are best supported under linux (pick one) FreeBSD has some hardware support, every thing else is good from straight VoIP only (but there are hardware gateways like Audiocodes, Cisco AS5300s, Sipura SPA2000 and 3000 series ATAs etc you can use |
| 07:11.16 | SwK | if you want to use BSD |
| 07:11.17 | hads|home | Anyone around that knows a bit about fxotune? I always seem to get 1=5,0,0,0,0,0,0,0,0 which seems wrong. |
| 07:11.48 | yacyac | i have been using arch for a while |
| 07:11.53 | Strom_C | hads, what seems wrong with it |
| 07:12.03 | yacyac | i will be getting a server ready for that in my office |
| 07:12.11 | yacyac | maybe you guys can connect to it if you want |
| 07:12.15 | yacyac | once its ready |
| 07:12.32 | hads|home | Well, possibly nothing, but it seems odd. |
| 07:12.38 | Strom_C | hads: looks fine to me |
| 07:12.49 | pc55 | pjchilds - any way to set aside a greater allowance for jitter in asterisk? |
| 07:12.57 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 07:13.05 | hads|home | Strom_C: Fair enough, I just expected the line to need more tuning than that. Cheers. |
| 07:13.31 | pjchilds | pc55, no idea.. standard 1.2.x doesn't have any jitterbuffer support... as for the patched versions... not sure how that works either... |
| 07:14.45 | pc55 | pjchilds - Ok, thanks. i thought there was a way you could tell asterisk to increase latency at the cost of some packet queueing and reassembly on a crappy link. maybe it's a factor fo the codec. |
| 07:14.47 | pc55 | of |
| 07:15.08 | X-Rob | pjchilds, SIP doesn't have a jitterbuffer, IAX does. |
| 07:15.19 | X-Rob | and generic jitterbuffer is in 1.4 |
| 07:16.04 | X-Rob | hads|home, that's right - all the zeros are meant to be there. But there is a patch, search for fxotune on bugs.digium.com |
| 07:16.41 | hads|home | X-Rob: Are you talking about the patch that made it way into trunk the other day. |
| 07:16.53 | X-Rob | hads|home, dunno |
| 07:17.00 | X-Rob | possibly |
| 07:17.07 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 07:17.07 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:17.24 | hads|home | I'm using trunk, and there was a major patch to fxotune that got merged the other day. |
| 07:17.51 | pc55 | pjchilds - Is it posible to pump a SIP packet overe netcat, so say a firewall would recognize it as SIP traffic and not just trafficeo n 5060/udp/ |
| 07:18.01 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:19.44 | hads|home | I read reports from people after using the 'new' version of fxotune that they get much better results. Mine are the same as before so I was just wondering. |
| 07:20.40 | hads|home | I guess that could just be the correct settings for that line. |
| 07:21.27 | pjchilds | X-Rob, generic jitterbuffer has been backported by a few people to various RPM releases of 1.2.9.1 |
| 07:22.03 | X-Rob | pjchilds, cool. |
| 07:22.17 | pjchilds | pc55, in theory yes ;) |
| 07:23.10 | pc55 | pjchilds - theory sounds good :p |
| 07:24.33 | pjchilds | well in theory there is no difference between theory and practice... but in practice there is.... |
| 07:24.42 | pc55 | rofl |
| 07:25.05 | *** join/#asterisk Szolke (n=Szolke@22-36.adsl.etel.hu) |
| 07:25.57 | pjchilds | pc55, google 'sipp' or 'sipsak' for sip message injection tools... |
| 07:27.46 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:29.09 | pc55 | thanks |
| 07:30.27 | Szolke | Hi everybody. I would like to get some information my asterisk agents. I found a command: exten => XXX,n,AGENT(....) this command need func_agent.so. Where can i find it? It is not in the funcs folder. |
| 07:32.18 | pjchilds | Szolke, no such function exists... |
| 07:34.25 | Szolke | :o I found it on web. :( how can i get some information from my agent's status? I would like to use it in dial plan. |
| 07:40.45 | pjchilds | Szolke, hmm.. its on voip-info.org but I can't see anything in /functions/ or chan_agent... sorry.. i've never used it... |
| 07:42.03 | Szolke | pjchilds, Yes it was on voip-info , thanks |
| 07:43.18 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 07:43.41 | *** join/#asterisk Shoragan (n=shoragan@134.169.175.72) |
| 07:43.51 | *** join/#asterisk darkgamer20 (n=chatzill@adsl-71-146-182-66.dsl.pltn13.sbcglobal.net) |
| 07:44.01 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 07:46.08 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 07:46.11 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 07:46.33 | yxa | can polycom phones that are behind a nat connect to a sip server also behind a nat? |
| 07:46.58 | yxa | soft phones like xlite can but i do not have success so far with a polycom 601 |
| 07:48.49 | darkgamer20 | I have a regular home phone line from SBC without caller ID, can asterisk server as something: that will play music instead of a ringing sound, can it direct calls to a specific phone in my house based on the extension pressed on the other end of the line, can it filter calls based on the person calling (assuming I do get caller ID in the future)? |
| 07:51.33 | mitcheloc | yes |
| 07:51.57 | *** join/#asterisk lorinc (n=ang@caracas-0691.adsl.interware.hu) |
| 07:52.02 | darkgamer20 | mitcheloc: you rock! |
| 07:52.26 | mitcheloc | thanks? |
| 07:52.27 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 07:53.46 | darkgamer20 | lol i just didnt think asterisk can do that much, i thought it was for like a business phone system, you know for routing calls through the office |
| 07:55.49 | mitcheloc | well, asterisk can do anything |
| 07:55.59 | darkgamer20 | i have been reading the book but everything is kinda complicated |
| 07:56.26 | darkgamer20 | lol some guy wrote in a guide that it can open his door if he locked him self out? |
| 07:57.27 | mitcheloc | i don't get the "locked him self out" part, but my door unlocks when i call my house... |
| 08:00.09 | darkgamer20 | wow how did you set that up? |
| 08:00.30 | mitcheloc | it's really not worth the trouble... |
| 08:00.53 | darkgamer20 | oh but i want to know, was there a guide you followed or something? |
| 08:01.55 | mitcheloc | denon: you would have to know my phone number to my door =P, nobody but me & bv knows it ;) |
| 08:03.05 | mitcheloc | denon: didn't we have a chat a long time ago about win32 gui's for asterisk? |
| 08:03.11 | denon | yeah |
| 08:03.25 | mitcheloc | ahh the bounty is still there |
| 08:03.30 | denon | finding your door number would be easy .. binoculars and watch you on a lazy day |
| 08:03.47 | denon | I doubt you cover the LCD every day |
| 08:03.49 | denon | :) |
| 08:03.56 | mitcheloc | denon: meh, you could just kick in my door, that'll do the trick too |
| 08:04.04 | denon | sure, but then you'd know someone was there <G> |
| 08:04.24 | mitcheloc | what ever happened with the bounty? maybe i'll take you up on it, i could use $460 |
| 08:04.28 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:04.44 | denon | eh, nobody ever actually delivers a good app |
| 08:04.51 | mitcheloc | denon: have you seen snap? |
| 08:04.56 | mitcheloc | i'm sure i can add whatever is missing |
| 08:04.56 | denon | there's a million half-assed ones out there |
| 08:05.02 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:05.02 | denon | url? |
| 08:05.03 | mitcheloc | snap damnit! |
| 08:05.07 | mitcheloc | www.snapanumber.com |
| 08:05.16 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 08:05.26 | mitcheloc | look at the "more screenshots" |
| 08:05.55 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:06.29 | denon | meh, that's only one of the things I wanted :) |
| 08:06.37 | denon | most of what I wanted is in that flash thing |
| 08:06.45 | denon | FOPwhatever |
| 08:06.48 | *** join/#asterisk Ikarus (n=ikarus@c109100.upc-c.chello.nl) |
| 08:06.53 | denon | Flash Asterisk Panel or whatever it's called |
| 08:06.55 | mitcheloc | well i've put a lot of love into it, so i can add whatever is missing |
| 08:06.57 | *** join/#asterisk CerealVore (n=Draco@203.194.19.141) |
| 08:07.02 | nick125_lappy | flash operator panel IIRC |
| 08:07.02 | CerealVore | Hi guys |
| 08:07.10 | denon | ah, it's yours |
| 08:07.23 | Ikarus | hey, I am using BRIstuff ISDN drivers, and my telephone provider refuses to turn off the bounce of their D channel, any suggestions for a work around |
| 08:07.40 | CerealVore | If i'm using queues/agents, does the 'agent' have to be logged in the whole time? ie, the agent can't hang up the phone and make calls? |
| 08:07.40 | mitcheloc | denon: you'll get a free license for the bounty, lol |
| 08:07.47 | denon | hahaha |
| 08:07.50 | denon | um, thanks but uh nah |
| 08:08.15 | denon | man, you suck at sucking up .. most people say "here's a 10 pack, let me know what ya think" |
| 08:08.43 | CerealVore | i'm wondering if there is any way i can have queues, without agents basically |
| 08:08.46 | mitcheloc | sure, why not? |
| 08:09.17 | mitcheloc | denon: do you work at digium? |
| 08:09.44 | denon | nope |
| 08:10.01 | denon | dont think I'd want voip to be my whole life heh |
| 08:12.30 | mitcheloc | i asked because i'm visiting there soon |
| 08:13.46 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:13.55 | Dandre | Hello all, |
| 08:15.12 | Dandre | I am using a sip phone and asterisk 1.2.7.1 |
| 08:16.14 | Dandre | when I blind transfer a call, if the destination extension is unavailable, the call is lost. Is there any workaround? |
| 08:17.19 | mitcheloc | Dandre: i'm sure you can use an "i" extension to catch it, and redirect it? |
| 08:18.13 | Dandre | I am using freepbx, I don't know where to look for this |
| 08:18.52 | pjchilds | ~amp |
| 08:18.57 | jbot | somebody said amp was NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 08:19.43 | Dandre | Ok |
| 08:20.53 | yxa | can polycom phones that are behind a nat connect to a sip server also behind a nat? |
| 08:20.56 | yxa | soft phones like xlite can but i do not have success so far with a polycom 601 |
| 08:22.51 | *** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com) |
| 08:24.05 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 08:27.00 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 08:27.29 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:33.16 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 08:39.17 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 08:48.52 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 08:49.04 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl) |
| 08:49.24 | QuAtRo[NL] | I have a problem with logging in/on to VoipBuster |
| 08:50.02 | x86 | that's nice |
| 08:50.04 | *** join/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 08:50.26 | QuAtRo[NL] | The problem seem to be that the logon fails |
| 08:51.13 | n3glv | hello all |
| 08:51.23 | n3glv | konichi-wa |
| 08:51.24 | x86 | n3glv: you in pittsburgh? |
| 08:51.27 | n3glv | yes |
| 08:51.33 | n3glv | pittsburgh |
| 08:51.34 | x86 | n3glv: i used to live in penn hills ;) |
| 08:51.45 | n3glv | I am in Plum, on 286 |
| 08:51.48 | n3glv | ;-) |
| 08:51.50 | x86 | ah cool |
| 08:51.56 | x86 | not too far from where i used to live ;) |
| 08:51.58 | n3glv | right behind kings |
| 08:52.22 | n3glv | every late night Mt Dew junkie knows kings here |
| 08:52.42 | n3glv | u know darwin? |
| 08:52.45 | x86 | nope |
| 08:53.00 | x86 | the OS? :) |
| 08:53.02 | n3glv | Rich used to live here too, met him at wplug |
| 08:53.09 | n3glv | no, irc/asterisk man |
| 08:53.13 | x86 | rich? |
| 08:53.15 | n3glv | unixbher |
| 08:53.15 | x86 | kings? |
| 08:53.20 | x86 | dont know any of them ;) |
| 08:53.22 | n3glv | or behr |
| 08:53.35 | x86 | you know duane from wplug? |
| 08:53.39 | n3glv | wplug is linux grp at CMU |
| 08:53.43 | x86 | right |
| 08:53.45 | n3glv | yeah, think I know him |
| 08:53.50 | x86 | he's a cock monkey ;) |
| 08:53.52 | n3glv | no most to see em |
| 08:54.16 | x86 | dont trust him man |
| 08:54.16 | n3glv | did a little wifi thing with some of the guys, a separate group |
| 08:54.24 | x86 | a cool |
| 08:54.27 | x86 | ah* |
| 08:54.31 | n3glv | if he's who I think, I would not |
| 08:54.36 | x86 | duane wells |
| 08:54.42 | n3glv | he what's her name's side kick? (the pres) |
| 08:54.57 | x86 | i dunno ;) |
| 08:55.03 | n3glv | tall? |
| 08:55.17 | x86 | yep |
| 08:55.21 | n3glv | early 30's, works for CMU as tech support or something |
| 08:55.27 | x86 | no, he owns his own company |
| 08:55.32 | n3glv | ahhhhh yes |
| 08:55.34 | x86 | interservers |
| 08:55.37 | x86 | something like that |
| 08:55.38 | n3glv | know Eactly |
| 08:55.59 | n3glv | he's in Penn Hills or something too I think |
| 08:56.15 | n3glv | did some work for Jack Earl and then tried to snipe the client |
| 08:56.15 | x86 | he moved to squirrel hill |
| 08:56.20 | n3glv | that's it |
| 08:56.32 | x86 | yeah man he's schiesty |
| 08:56.37 | n3glv | was working on a wifi nusecall system |
| 08:57.06 | n3glv | anybody running trixbox? |
| 08:57.18 | n3glv | heard something about an iax upgrade? |
| 08:57.57 | RoyK | ~nickometer n3glv |
| 08:58.09 | RoyK | ~nickometer x86 |
| 08:58.13 | x86 | ~nickometer RoyK |
| 08:58.21 | x86 | pfft... what a joke ;) |
| 08:58.27 | RoyK | :) |
| 08:58.33 | RoyK | ~nickometer 4x90823y40r7__ |
| 08:58.41 | RoyK | you see.... |
| 08:59.10 | sternn | ~nickometer jbot |
| 08:59.16 | sternn | heh |
| 08:59.29 | x86 | biased mofo ;) |
| 08:59.45 | n3glv | anyway, anyone wanna talk about trunks betwen * machines? |
| 09:00.05 | *** join/#asterisk muezzin (n=muezzin@81-174-13-104.f5.ngi.it) |
| 09:00.13 | n3glv | been doing iax trunks, but have to have them log into each other, is there a better way? |
| 09:00.29 | muezzin | hi |
| 09:00.30 | x86 | you can do anonymous trunking with iax |
| 09:00.37 | x86 | use guest ;) |
| 09:00.39 | n3glv | is:each machine has account on the other |
| 09:00.56 | QuAtRo[NL] | Does someone know if sip1.voipbuster.com has problems? |
| 09:00.58 | n3glv | guest? 200? |
| 09:01.03 | QuAtRo[NL] | Or doesn't exist anymore? |
| 09:01.49 | n3glv | I think I am set up for inboun anon sip, possibly iax, would like to get a test call here, if somone would care to |
| 09:02.02 | n3glv | inbound |
| 09:02.10 | x86 | n3glv: e164.org |
| 09:02.21 | n3glv | I have outbound enum trunk |
| 09:02.32 | n3glv | they never called me to reg my numbers |
| 09:02.35 | x86 | you want a test call... e164.org can do that ;) |
| 09:02.57 | n3glv | did over and over, they never did the verify call to me |
| 09:03.22 | muezzin | I've a problem with MusicOnHold : it's full of skips, silences... instead IVR's audio and callings' quality is fine... what I have to look at to fix it? |
| 09:03.41 | x86 | n3glv: how long ago? they were having problems at one point... been working fine for me for the last 2 months |
| 09:03.43 | n3glv | would LIKE to use enum between friends boxes, but so far only one of them has a reg on there |
| 09:04.01 | x86 | n3glv: i have ~80 US numbers registered with them, a few numbers in the UK, and one in PK |
| 09:04.31 | n3glv | I think it was sat or sunday when I was trying e.164 |
| 09:04.42 | x86 | hmm i dunno man |
| 09:04.45 | x86 | works fine for me |
| 09:04.52 | x86 | what NPA/NXX? |
| 09:04.59 | n3glv | the outbound 'trunk' seems to work (enum) |
| 09:05.14 | n3glv | don't know those two |
| 09:05.25 | x86 | enum doesnt use trunks, it's just a DNS lookup |
| 09:05.29 | x86 | NPA == area code |
| 09:05.32 | x86 | NXX == prefix |
| 09:05.40 | n3glv | 412-528 |
| 09:05.45 | n3glv | sri |
| 09:05.47 | muezzin | i'm running asterisk@home anyway |
| 09:05.50 | n3glv | newbie is showing |
| 09:06.03 | n3glv | I am trixbox , aah/freepbx |
| 09:06.38 | x86 | freepbx is lame |
| 09:06.43 | x86 | be a man use asterisk ;) |
| 09:07.06 | n3glv | plan on going to something smaller and tighter, this was easy |
| 09:07.09 | CerealVore | hey, i'm having some difficulty trying to register an agent using AgentCallbackLogin(), it keeps telling me my extensions are invalid, however, if i put in the correct context for the phones, ie 101@internal, it lets me register the agent, but it fails with: Jun 20 19:01:28 WARNING[20729]: res_features.c:450 ast_bridge_call: Bridge failed on channels Local/101@internal-19b0,2 and SIP/101-a805. when a calls comes through the queue for an agent |
| 09:07.10 | n3glv | test system |
| 09:07.56 | n3glv | callback is something that I have had no success with either |
| 09:08.00 | muezzin | anyone who can help me with MusicOnHold issue? it plays badly |
| 09:08.01 | muezzin | :( |
| 09:08.15 | x86 | muezzin: you have a timing device? |
| 09:08.17 | n3glv | muezin how much ram, how much free? |
| 09:08.27 | CerealVore | n3glv: have you had similar experiences? |
| 09:08.44 | muezzin | <n3glv>: HP e-vectra PIII 1 Ghz 256 mb ram |
| 09:08.48 | CerealVore | there doesn't seem to be a lot documented about this, whether contexts are applicable, etc |
| 09:08.50 | n3glv | I have not even got that far with callback |
| 09:08.57 | n3glv | it' mostly ignores me |
| 09:09.11 | muezzin | x86: i'm running astersik@home (last release) i don't know if it has RTC compiled in |
| 09:09.16 | n3glv | 256mb? what install? |
| 09:09.18 | x86 | uh |
| 09:09.28 | x86 | muezzin: you need a timing device for MOH to work |
| 09:09.33 | x86 | muezzin: you have any zap hardware? |
| 09:09.43 | x86 | muezzin: if not, you need the ztdummy kernel module loaded |
| 09:09.49 | muezzin | no.. i'm running zaptel module only |
| 09:09.58 | x86 | you need ztdummy then |
| 09:09.58 | muezzin | yes dummy zap |
| 09:10.09 | x86 | zaptel != ztdummy |
| 09:10.24 | muezzin | yes..it's zaptel module |
| 09:10.39 | n3glv | muezzin what install of asterisk? |
| 09:10.41 | x86 | it's part of the zaptel package |
| 09:10.45 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:11.02 | muezzin | <n3glv>: latest release of astersik@home |
| 09:11.14 | n3glv | u have the AMP? |
| 09:11.17 | muezzin | yep |
| 09:11.29 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 09:11.33 | n3glv | can u tell how much ram is in use? system free/in use? |
| 09:11.43 | x86 | n3glv: has nothing to do with ram |
| 09:11.53 | x86 | n3glv: he needs a timing device, and he's not listening to me ;) |
| 09:11.54 | *** part/#asterisk Szolke (n=Szolke@22-36.adsl.etel.hu) |
| 09:11.56 | n3glv | my trixbox was GASPING till I went to 300+Mb ram |
| 09:12.00 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 09:12.04 | n3glv | ok |
| 09:12.18 | muezzin | x86: how can i do? have i to recompile kernel with rtc ? |
| 09:12.44 | x86 | muezzin: you need ztdummy!!@$GYH@hb |
| 09:12.56 | n3glv | test with insmod ztdummy???? |
| 09:13.38 | muezzin | <x86> : lsmod output says i've ztdummy loaded as module |
| 09:13.53 | x86 | ok now test |
| 09:13.53 | muezzin | but musiconhold sucks |
| 09:14.48 | muezzin | music skips |
| 09:14.50 | muezzin | :( |
| 09:14.59 | muezzin | ztdummy is loaded |
| 09:16.16 | l-fy | hello |
| 09:16.22 | l-fy | who is from Budapesta? |
| 09:17.02 | x86 | muezzin: what mode? |
| 09:17.13 | x86 | muezzin: use mode=files in musiconhold.conf |
| 09:17.17 | x86 | muezzin: or mode=native |
| 09:18.34 | muezzin | i've noted that the first time music starts it works fine... instead when music is buffered it skips.. I've seen mpg123 process start with 2 Mb buffer options..so I would change it to 0 buffer at all.. where can I look to change default mpg123 parametres? |
| 09:19.01 | muezzin | in musiconhold.conf i have: |
| 09:19.13 | QuAtRo[NL] | Does someone have a working configuration with Voipbuster for outgoing calls? |
| 09:19.29 | muezzin | default => mp3:/var/lib/asterisk/mohmp3 |
| 09:19.32 | muezzin | only this line |
| 09:19.58 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 09:20.59 | muezzin | i must add the line "mode=files" before that line? |
| 09:22.30 | x86 | remove everything in the file |
| 09:22.34 | x86 | add this |
| 09:22.37 | muezzin | ok |
| 09:22.37 | x86 | [default] |
| 09:22.37 | *** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com) |
| 09:22.39 | x86 | mode=files |
| 09:22.50 | x86 | directory=/var/lib/asterisk/mohmp3 |
| 09:22.55 | x86 | save the file |
| 09:22.59 | x86 | go into CLI and do a reload |
| 09:23.01 | muezzin | ok i try |
| 09:32.29 | yxa | anyone has a SPA-84x or SPA-94x |
| 09:35.24 | hads|home | yes |
| 09:36.34 | CerealVore | has anyone had queues/agents/AgentCallbackLogin() work successfully? |
| 09:37.34 | yxa | hads|home does it work over NAT? |
| 09:38.18 | hads|home | I don't see why not, most SIP devices do. I use it with a local Asterisk server though. |
| 09:39.53 | *** join/#asterisk zepmantra (i=wahhh@203.215.100.96) |
| 09:40.33 | QuAtRo[NL] | I have a problem with my outgoing calls.. Asterisk seem to end all calls |
| 09:41.13 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 09:41.24 | QuAtRo[NL] | CerealVore; yes mine works fine |
| 09:41.44 | QuAtRo[NL] | yxa: I do |
| 09:41.57 | QuAtRo[NL] | SPA942 |
| 09:42.24 | QuAtRo[NL] | But not working to a public IP |
| 09:42.35 | QuAtRo[NL] | Didn't try so far |
| 09:43.54 | CerealVore | QuAtRo[NL]: what did you specify in your extensions.conf for the extension that has the AgentCallbackLogin() application? |
| 09:44.48 | CerealVore | no matter what i seem to do, i can't get it right |
| 09:45.02 | CerealVore | i think that's where the problem lies, but i'm not 100% on that |
| 09:45.32 | QuAtRo[NL] | exten => 2,1,AgentCallbackLogin(${CALLERIDNUM},@Support) |
| 09:45.47 | QuAtRo[NL] | Be carefull with spaces between arguments |
| 09:46.04 | QuAtRo[NL] | Most of the time it prevents the function from working |
| 09:47.24 | yxa | QuAtRo[NL] you have any experience for working over NAT? |
| 09:48.00 | yxa | QuAtRo[NL] for example, you bring the phone to a hotel, get dhcped and reg with a sip server somewhere |
| 09:48.03 | *** join/#asterisk kegg (i=Vitz@p5084D52F.dip.t-dialin.net) |
| 09:48.05 | n3glv | my system runs fine behind a consumer NAT router |
| 09:48.41 | n3glv | however, IAX was a lot easier |
| 09:48.46 | n3glv | all my clients are iax |
| 09:49.05 | kegg | is anybody here using analog cards? |
| 09:49.14 | n3glv | can connect via 'anyones' wifi, heehee |
| 09:49.59 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 09:50.01 | muezzin | x86: changing the musiconhold.conf didn't work... musiconhold still skips... :( |
| 09:50.25 | x86 | muezzin: upgrade to real asterisk then ;) |
| 09:50.36 | P-NuT | Hi all. I'm having some issues with my cisco 7905. I'm hearing an echo of myself talking in the reciever. |
| 09:50.41 | QuAtRo[NL] | yxa: No, i'm sorry :( |
| 09:51.15 | muezzin | x86: Asterisk@home use real Astersik to.,.. |
| 09:51.27 | muezzin | packages are those |
| 09:52.04 | kegg | is somebody experienced with fxo/fxs cards here |
| 09:52.07 | n3glv | muezzin I am on trixbox, works great, seems to want a lot of ram though |
| 09:52.32 | x86 | muezzin: old versions though |
| 09:52.49 | muezzin | what is tribox? |
| 09:52.59 | muezzin | sorry but i'm new to pbx |
| 09:53.27 | kegg | muezzin, asterisk@home moved to trixbox i think |
| 09:54.04 | *** join/#asterisk tparcina (n=tparcina@lns02-0072.dsl.iskon.hr) |
| 09:56.05 | kegg | i wonder how many ppl ar assleep in this channel :) |
| 09:56.55 | muezzin | <n3glv>: trixbox is a distribution dedicated to asterisk? is it manageble trough web interface? |
| 09:57.07 | muezzin | i see an iso to download on the site |
| 09:57.17 | muezzin | is it a standalone distro? |
| 09:57.47 | muezzin | only 24 mb.... m... |
| 09:58.02 | muezzin | what the hell is there in only 24 mb? |
| 09:58.02 | hads|home | ~trixbox |
| 09:58.07 | jbot | [trixbox] NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 10:01.26 | muezzin | thanks bye |
| 10:01.42 | CerealVore | QuAtRo[NL]: can i private you? |
| 10:02.16 | CerealVore | or just about what yo said before, that @Support, is that a context in extensions.conf? or is that the name of the queue? |
| 10:03.20 | QuAtRo[NL] | CerealVore: yes you can |
| 10:03.40 | QuAtRo[NL] | Does someone know the error: Everyone is busy/congested at this time |
| 10:05.10 | *** part/#asterisk n3glv (n=Omega__@monrovll-cuda1-24-53-251-235.pittpa.adelphia.net) |
| 10:15.37 | tparcina | is anybody on this channel from quatar (or katar) |
| 10:17.35 | Pj_ | It happens to the best |
| 10:17.42 | Ahrimanes | has anyone here thought of a way to make an automated test call, to verify iax/sip links? |
| 10:18.09 | motu | is it possible to set up and monitor a call between two zap devices through sip? |
| 10:19.07 | tparcina | i need SIP connection in quatar, anybody knows for any quatar SIP provider? |
| 10:24.48 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 10:29.42 | tparcina | RoyK, norway is close to Quatar, right? ;)) |
| 10:31.45 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:35.35 | RoyK | tparcina: just around the corner |
| 10:40.14 | QuAtRo[NL] | Is there really nobody with the error: == Everyone is busy/congested at this time (1:0/0/1) |
| 10:40.24 | RoyK | QuAtRo[NL]: lots |
| 10:41.12 | QuAtRo[NL] | And nobody knows the solution? |
| 10:42.04 | Ahrimanes | QuAtRo[NL]: there could be many reasons you get busy/congested.. what are you calling from and what are you calling to? |
| 10:43.10 | QuAtRo[NL] | Calling from a voipbuster account to a Dutch landline (0031534364624) |
| 10:43.27 | QuAtRo[NL] | And using the VoipBuster application works fine |
| 10:44.01 | Ahrimanes | but now you're trying to use asterisk with voipbuster? |
| 10:44.07 | QuAtRo[NL] | Yes |
| 10:44.11 | QuAtRo[NL] | The register works |
| 10:44.17 | QuAtRo[NL] | Cause incomming calls do work fine |
| 10:44.28 | QuAtRo[NL] | also through voipbuster |
| 10:45.45 | Ahrimanes | http://www.voip-info.org/wiki/view/Asterisk+VoIPBuster <- around the middle of the page, have you seen that? |
| 10:47.17 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 10:48.57 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 10:51.40 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 10:54.33 | *** join/#asterisk [pyro] (i=pyro@tor/regular/bracketed-pyro) |
| 10:56.31 | *** join/#asterisk sevard (n=sev@24-179-179-47.dhcp.dlth.mn.charter.com) |
| 10:58.10 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 11:00.47 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 11:03.23 | *** join/#asterisk tRSS (n=tRSS@pk-isb-trg-sc01-019.speedcast.com) |
| 11:03.46 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 11:03.50 | tRSS | quick question: my asterisk server is working fine. but my users are unable to register with it with a softphone? |
| 11:03.55 | tRSS | is this a firewall issue? |
| 11:04.00 | *** join/#asterisk Maan (n=mbsat@190.128.77.83.cust.bluewin.ch) |
| 11:07.32 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl) |
| 11:13.14 | QuAtRo[NL] | Still can't solve the problem with the == Everyone is busy/congested at this time (1:0/0/1) |
| 11:13.30 | QuAtRo[NL] | I have the settings from the page you suggested me |
| 11:19.18 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 11:23.35 | *** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk) |
| 11:24.37 | Maan | hello, I'm having problems with connecting my asterisk box to FWD. Calling from FWD shows "Don't know any of 0xf800 formats" on my * log. |
| 11:24.49 | Ahrimanes | where could i get a 1 port pci card to connect to a single phone line? |
| 11:26.38 | RoyK | Ahrimanes: google :) |
| 11:26.46 | RoyK | Ahrimanes: google for x100p |
| 11:27.01 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 11:30.00 | Ahrimanes | RoyK: ah, forgot the name of the card, thx :) |
| 11:34.57 | P-NuT | Hi all, has anybody got a Grandstream BT101 |
| 11:38.44 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 11:39.39 | *** join/#asterisk userdefined (n=jross@cpe-24-169-142-23.rochester.res.rr.com) |
| 11:39.59 | *** join/#asterisk jmacz (n=jmacz@201.244.240.60) |
| 11:40.07 | jmacz | Hi everyone |
| 11:41.13 | jmacz | I'm having a problem to compile Asterisk 1.2.9.1 in a Debian Sarge box |
| 11:41.40 | jmacz | I got "error: termcap support not found" |
| 11:41.46 | QuAtRo[NL] | Configuration files for the not working outgoing calls: http://pastebin.com/720641 |
| 11:42.13 | jmacz | I tried installing termcap-compat 1.2.3, but I still got that message |
| 11:42.42 | jmacz | any ideas what might be wrong? |
| 11:44.04 | jmacz | I'm running kernel 2.6.8-2-686 |
| 11:44.21 | userdefined | jmacz: debian has packages for asterisk (though it's v.1.2.7.1 in 'testing') ... could see what deps it's got and see if installing them helps |
| 11:44.33 | qdk | jmacz: termcap libs? |
| 11:44.51 | jmacz | userdefined, good idea, I'll check that out |
| 11:46.02 | tzafrir | jmacz, apt-get builddep asterisk |
| 11:46.14 | jmacz | qdk, don't see any package like that :( |
| 11:46.26 | qdk | jmacz: me neither. :-D |
| 11:46.31 | tzafrir | jmacz, apt-get build-dep asterisk # that is |
| 11:47.02 | tzafrir | (will work if you have a standard deb-src source) |
| 11:47.23 | tzafrir | The build-deps of asterisk 1.2 ar not that different of those of the Sarge package |
| 11:47.59 | tzafrir | also, apt-get install kernel-image-2.6-686 |
| 11:48.38 | tzafrir | (for if you don't like typing kernel-image-2.6.8-3-686 ) |
| 11:49.13 | jmacz | tzafrir, :) |
| 11:49.39 | jmacz | tzafrir, thanks I'm trying with the builddep |
| 11:49.46 | tzafrir | build-dep |
| 11:50.00 | tzafrir | (my typo) |
| 11:50.28 | jmacz | that's right |
| 11:58.00 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:58.09 | RoyK | hm. any idea what 'rtp payload type' is supposed to be set to on my ATA? |
| 11:58.22 | Ahrimanes | which ata? |
| 11:58.49 | RoyK | a yoda.com.tw ata |
| 11:58.55 | RoyK | rfc2833 stuff |
| 11:59.23 | Ahrimanes | a yoda.. :P |
| 11:59.31 | Ahrimanes | hm mine are set to 101 or something like that afaik |
| 12:00.36 | *** join/#asterisk saftsack (n=saftsack@p54A7D5D0.dip.t-dialin.net) |
| 12:00.44 | saftsack | hi |
| 12:00.56 | saftsack | are there some good working and available fax modems? |
| 12:01.38 | saftsack | because i want to do hylafax -> hardwarefaxmodem -> fxs card -> asterisk -> idn |
| 12:01.39 | saftsack | isdn |
| 12:01.52 | QuAtRo[NL] | My problem is solved |
| 12:01.59 | QuAtRo[NL] | I had user what should be friend :D |
| 12:04.36 | *** join/#asterisk heath__ (n=heath__@71-87-34-39.dhcp.stcd.mn.charter.com) |
| 12:06.09 | sivana | saftsack: why have hardware modem in there? |
| 12:06.24 | sivana | saftsack: hylafax -> iaxmodem -> asterisk -> idn |
| 12:07.18 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:14.22 | *** join/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net) |
| 12:19.00 | userdefined | is there a way to have * use tcp/5060 rather than udp? |
| 12:20.28 | [TK]D-Fender | userdefined : No. * Does not support SIP over TCP yet |
| 12:20.44 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
| 12:22.23 | userdefined | i suspected that was the case =/ |
| 12:23.03 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 12:25.01 | userdefined | in light of that, i've been considering using netcat to forward udp/5060 -> tcp/5060 via a fifo. any downsides to doing that come to mind (other than the obvious "it's a kludgy hack" of course ;-) |
| 12:25.12 | Dovid | Can anyone help me with 101 |
| 12:26.08 | Dovid | If I have Exten => 100,1,Dial(SIP/100) and its busy whats the next line that I can put in so it jumps to it if its busy ? Exten => 101,2,102 Dial(SIP/300) ? |
| 12:28.08 | [TK]D-Fender | Dovid : Go read about ${DIALSTATUS} to see the different return codes so that you can shoose how you want to deal with a non-completed dial. |
| 12:28.38 | [TK]D-Fender | userdefined : What so adamant on TCP? |
| 12:28.42 | Dovid | Url ? |
| 12:28.50 | [TK]D-Fender | Dovid : The WIKI |
| 12:28.52 | [TK]D-Fender | ~docs |
| 12:29.03 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:29.03 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:29.03 | Dovid | On the wiki. |
| 12:29.03 | Dovid | :) |
| 12:30.08 | userdefined | [TK]D-Fender: trying to get * talking to MS LCS which only does SIP via tcp |
| 12:30.11 | RoyK | i heard someone talk about the necissity of rtp over tcp for nat/firewall issues.... |
| 12:30.59 | coppice | yeah. RTP over TCP is definitely needed. That and toothache |
| 12:31.21 | Dovid | RoyK: yes if u r using nat with sip and ur box is behind the nat u need to forward some ports over |
| 12:31.36 | coppice | TCP == Telephony Corruption Protocol |
| 12:31.37 | Dovid | I think 5060,5061 and 5000 as well as 10000 - 20000. UDP |
| 12:31.49 | Dovid | heee |
| 12:31.51 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 12:34.30 | userdefined | well, in my case i don't have nat to worry about (yet) ... both hosts are behind the firewall |
| 12:34.52 | *** join/#asterisk dwildes2 (n=dwildes@209.164.237.195) |
| 12:34.54 | userdefined | but of course, MS has to do things just differently enough to make it miserable for anything non-MS to work with it |
| 12:34.57 | Dovid | Eh. U will soon |
| 12:35.08 | Dovid | We need more hardphones that support IAX |
| 12:37.03 | *** join/#asterisk teknoprep (n=teknopre@c-68-44-169-116.hsd1.pa.comcast.net) |
| 12:37.13 | teknoprep | hm anyone here get a build error from ports building asterisk ? |
| 12:37.15 | teknoprep | on freebsd? |
| 12:37.33 | Ahrimanes | hm not last time i tried |
| 12:39.09 | UlbabraB | teknoprep: yes, after the last commit |
| 12:39.19 | Dovid | I am waiting till 1.4 is released b4 I upgrade. Anyone know why they are going from 1.2 to 1.4 and not to 1.3 first ? |
| 12:39.42 | iDunno | I'd assume that 1.3 is a development branch |
| 12:39.47 | iDunno | that 1.4 would be based from |
| 12:40.08 | iDunno | kinda like the linux kernel is 2.0, 2.2, 2.4 |
| 12:40.13 | iDunno | 2.6 |
| 12:40.17 | UlbabraB | teknoprep: just remove files/patch-asterisk.c and it will compile (and run) fine |
| 12:40.40 | RoyK | omg. a customer kindly informed us that he had a broadband router integrated in his hard drive..... |
| 12:41.12 | Ahrimanes | RoyK: i hate those new hybrid products |
| 12:41.22 | sep | RoyK coool where can i get one ? :) |
| 12:43.19 | [TK]D-Fender | Luge-Bowling! Now on ESPN! |
| 12:43.50 | coppice | the japanese probably make one. think of any weird combination of functions, and someone in japan will be making it, probably in a version painted with Hello Kitty |
| 12:44.48 | Pj_ | coppice: and a variable water spray directed with a joystick |
| 12:44.57 | Pj_ | with luckwarm water |
| 12:44.58 | Pj_ | hmmmm |
| 12:45.31 | coppice | Pj_ stick that up your ass |
| 12:47.31 | *** join/#asterisk Ciber311 (n=Ciber@user-1087e94.cable.mindspring.com) |
| 12:49.47 | *** join/#asterisk Henk (n=Henk@s5593c2e9.adsl.wanadoo.nl) |
| 12:50.52 | *** join/#asterisk Lino` (n=Lino@i577BD6B1.versanet.de) |
| 12:51.15 | Henk | calling out with asterisk is done with a .call fie right? any good tutorial on that ? |
| 12:51.24 | mitcheloc | or a phone ;) |
| 12:51.50 | Pj_ | coppice: :'( |
| 12:52.51 | coppice | Pj_ having used those things I still can't work out if they are a good idea, or just a desperate attempt to launch a new product |
| 12:53.21 | zamba | [TK]D-Fender: it's an alcatel 4200 |
| 12:53.27 | coppice | they probably need a camera in the toilet pan, so you can direct the spray effectively. |
| 12:53.30 | zamba | [TK]D-Fender: we talked yesterday about the telephony system |
| 12:53.34 | Pj_ | I have too, found out the "idea" is "neat", though they may have elaborated too much on it |
| 12:53.51 | Pj_ | the joystick might be overkill :/ |
| 12:53.52 | Henk | mishehu, I mean i want to inplement system generated triggers (e.g. hook it into a systems monitoring system) |
| 12:54.16 | userdefined | Henk: there's a couple of places that discuss the .call files (voip-info and asteriskdocs.org each have some useful info) |
| 12:54.32 | userdefined | magic words are "auto-dialout" |
| 12:54.36 | userdefined | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 12:54.48 | [TK]D-Fender | zamba : Ok, it wasn't the system brand that mattered, its the kind of interfaces you have available on it., |
| 12:55.11 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 12:55.18 | Henk | userdefined, i found some info ... I made a typo "astrisk" (missing e) so google gave me some crap on my qeury for "astrisk call file" |
| 12:55.48 | zamba | [TK]D-Fender: what am i looking for? what kind of words/abbrievations? |
| 12:55.49 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 12:56.24 | zamba | UA/I dedicated telephone, seems to be four of them.. then T0 ISDN trunk, two of them.. then Z analog telephone.. two of them as well.. |
| 12:58.06 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 12:58.50 | Dr-Linux | my pri is still down :( |
| 13:05.44 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.159.Dial1.SanJose1.Level3.net) |
| 13:06.19 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.159.Dial1.SanJose1.Level3.net) |
| 13:07.41 | Dr-Linux | RoyK: is this something wrong, i can see it when i do "dmesg" |
| 13:07.42 | Dr-Linux | 2G: Got interrupt, status = 0000000a, GIS = 0080 |
| 13:07.44 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:09.56 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 13:10.12 | *** join/#asterisk kSquared (i=kkaitan@68-65-51-183.chvlva.adelphia.net) |
| 13:10.23 | kSquared | howdy all |
| 13:11.00 | kSquared | quick question: |
| 13:11.14 | kSquared | I have a legacy setup that looks something like this |
| 13:11.36 | kSquared | [Cisco H.323 gateway] <---> [trunk lines to PSTN] |
| 13:11.53 | kSquared | how amenable is Asterisk to getting that setup to look like this: |
| 13:11.55 | Dr-Linux | question, what does Yellow alram mean? : |
| 13:11.56 | Dr-Linux | wct2xxp: Setting yellow alarm on span 1 |
| 13:11.56 | Dr-Linux | TE2XXP: Span 1 configured for ESF/B8ZS |
| 13:12.26 | kSquared | [VoIP SIP hardphones] <---> [Asterisk PBX as SIP] <---> [Cisco H.323 gateway] <---> [trunk lines to PSTN] |
| 13:14.01 | qdk | kSquared: fairly easy. |
| 13:15.18 | *** join/#asterisk ivanfm[wrk] (n=ivanfm@cumulus.saisp.br) |
| 13:15.20 | kSquared | so the SIP-to-H.323 crossover isn't too bad? |
| 13:16.02 | kSquared | I haven't looked into what's involved |
| 13:16.28 | kSquared | and I just finished getting a very simple 4-phone SIP setup working last week, so I'm still pretty much an Asterisk newbie |
| 13:16.53 | qdk | kSquared: i never used H.323, but its just like any other channel when its set up. |
| 13:17.41 | kSquared | alrighty |
| 13:17.53 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 13:18.09 | kSquared | I'd heard rumblings that Asterisk didn't play very well with H.323 |
| 13:18.18 | kSquared | but perhaps that was only if Asterisk itself was the H.323 gateway |
| 13:18.43 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 13:18.59 | *** join/#asterisk frenzy (n=frenzy@196.41.57.3) |
| 13:19.03 | *** join/#asterisk aetius (n=aetius@cpe-071-077-064-039.nc.res.rr.com) |
| 13:19.08 | frenzy | hello folks |
| 13:20.20 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 13:20.42 | qdk | kSquared: well H.323 is not an internal module, as far as i remember. |
| 13:20.42 | frenzy | I just setup the x100p FXO card from x100p.com.. I'm using Grandstream'd Budgetone... when calling out the zaptel interface I hear my own voice too |
| 13:20.49 | *** join/#asterisk Strom_C (n=strom@m615e36d0.tmodns.net) |
| 13:21.21 | [TK]D-Fender | frenzy : Start reading up on zaptel echo cancellation... and the fun begins |
| 13:21.40 | frenzy | I've go all the echo cancellations enabled |
| 13:24.13 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 13:25.14 | Dr-Linux | [TK]D-Fender: do you know what Yellow alarms mean? |
| 13:25.17 | Dr-Linux | wct2xxp: Setting yellow alarm on span 1 |
| 13:25.23 | frenzy | where can I find good documentatin on zaptel echo cancellation? |
| 13:25.24 | Dr-Linux | TE2XXP: Span 1 configured for ESF/B8ZS |
| 13:25.31 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 13:26.22 | santoshr | if a queue has a certain set og agent as members but none of them logged in.. wht hppens then |
| 13:27.09 | santoshr | will it go to the next line in the extensions or.. |
| 13:27.14 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 13:27.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 13:27.55 | stack_ | I upgraded to 1.2.9.1 and now AgentCallbackLogin stopped working... any ideas? |
| 13:29.53 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:29.57 | Dandre | hello, |
| 13:30.50 | Dandre | I have difficulties using isdn eicon cards with asterisk. Is there a good installation documentation for it and chancapi configuration? |
| 13:33.35 | tzafrir | frenzy, what echo do you try to cancel? |
| 13:35.03 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 13:35.06 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 13:36.23 | Dr-Linux | tzafrir: what does yellow alram mean? |
| 13:36.44 | coppice | yellow == afraid :-) |
| 13:37.33 | Dr-Linux | coppice: what do you mean by afraid? |
| 13:38.02 | qdk | lol |
| 13:38.58 | *** join/#asterisk imbrandon (n=brandon@ubuntu/member/pdpc.active.imbrandon) |
| 13:39.19 | Henk | hi, please bear with me i'm pretty new at asterisk and voip in general. I've configured my sip.conf like this http://pastebin.com/720816 (calling into that is working).. I'm trying to create a call file that will use this account to place calls to my mobile phone. I keep getting 'forbidden' errors in the logs though. I'm trying to use channel SIP/mobilephonenumber@budgetphone.nl |
| 13:39.35 | tzanger | yellow alarm is the state when you can see the other side but they cannot see you. |
| 13:39.47 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@dsl-083-247-051-039.solcon.nl) |
| 13:39.52 | tzanger | i.e. you receive yellow alarm -- you're seeing a signal but it's screaming "I CANNOT SEE YOU" |
| 13:40.15 | tzanger | you send red alarm when you cannot see the other side |
| 13:40.22 | tzanger | (I think it's been a while) |
| 13:40.51 | *** join/#asterisk cokehabit (n=Oscar_Wi@gentoo/contributor/pdpc.active.cokehabit) |
| 13:41.03 | Strom_C | I thought yellow alarm was "physical link up; data link down/unsynchronized" |
| 13:41.17 | tzanger | I cannot see you = send unframed all-1s signal. When that is received the receiver is in yellow alarm state |
| 13:41.42 | tzanger | red alarm means that you are not seeing a valid signal at all so you send out an all-1s pattern |
| 13:42.22 | tzanger | so sender is in red, receiver is in yellow. |
| 13:42.29 | tzanger | Strom_C: let's use google. I think we're saying the same thing :-) |
| 13:43.15 | Dr-Linux | tzafrir: what does this mean :S |
| 13:43.16 | Dr-Linux | ivr1*CLI> pri show span 1 |
| 13:43.16 | Dr-Linux | Primary D-channel: 24 |
| 13:43.16 | Dr-Linux | Status: Provisioned, Down, Active |
| 13:43.18 | tzanger | yeah, AIS = blue alarm = I can't see you |
| 13:43.23 | Dr-Linux | it's down? |
| 13:43.46 | tzanger | RAI = yellow alarm = I can see you but you're saying you can't see me |
| 13:44.06 | Dr-Linux | ivr1*CLI> zap show status |
| 13:44.06 | Dr-Linux | Description Alarms IRQ bpviol CRC4 |
| 13:44.06 | Dr-Linux | T2XXP (PCI) Card 0 Span 1 OK 0 0 0 |
| 13:44.06 | Dr-Linux | T2XXP (PCI) Card 0 Span 2 OK 0 0 0 |
| 13:44.13 | tzanger | red alarm = no signal whatsoever (can't sync) |
| 13:44.52 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 13:44.58 | tzanger | RoyK: it means that the LT (switch) side of the PRI has the PRI turned down |
| 13:45.00 | *** join/#asterisk znoG (n=speedy@webtastic.com.au) |
| 13:45.18 | tzanger | it means everything will be good (should be) once you call the switch tech and have her turn up the PRI |
| 13:45.21 | znoG | does anyone know if the Grandstream HT386 has some codes for call forwarding? (like *69<num> to activate) |
| 13:45.32 | *** join/#asterisk sumdingwong (n=sumdingw@adsl-66-137-178-195.dsl.spfdmo.swbell.net) |
| 13:45.55 | Dr-Linux | tzanger: you see my messages? :S |
| 13:46.06 | tzanger | ? |
| 13:46.37 | Dr-Linux | tzanger: i'm keeping asking but, i though you can't see my messages? |
| 13:46.51 | Dr-Linux | <Dr-Linux> Primary D-channel: 24 |
| 13:46.51 | Dr-Linux | <Dr-Linux> Status: Provisioned, Down, Active |
| 13:46.52 | tzanger | which messages? |
| 13:47.02 | Dr-Linux | it's down? |
| 13:47.03 | tzanger | yes I just told you what that means |
| 13:47.13 | tzanger | 09:45 < tzanger> it means everything will be good (should be) once you call the switch tech and |
| 13:47.18 | tzanger | <PROTECTED> |
| 13:47.54 | Dr-Linux | i see |
| 13:48.08 | Dr-Linux | tzanger: so you mean, it's telco side issue? |
| 13:48.12 | stack_ | I upgraded from 1.2.7.1 to 1.2.9.1 and now my queues and agents are acting crazy... The queues are hung and agent logins hang... The channels stay active too and can't be hungup... any ideas? |
| 13:48.25 | tzanger | the D channel is likely administratively down |
| 13:48.34 | sumdingwong | Does anyone know the processes for MWI lights with sip in Asterisk? |
| 13:48.47 | Dr-Linux | tzanger: you remeber i had some issues with my previous asterisk server, card were not being detected, so we bought new server and plugged the cards IN |
| 13:48.57 | znoG | stack_: i have an idea, go back to 1.2.7.1 until you figure out whats up with 1.2.9.1 :) |
| 13:49.19 | Dr-Linux | tzanger: i just want to verify if it's not my configuration issue.. |
| 13:49.24 | Dr-Linux | there looks something wrong |
| 13:49.43 | Dr-Linux | tzanger: can you have a look at my "dmesg" out put >> http://pastebin.com/720836 ? |
| 13:50.17 | tzanger | Dr-Linux: as I said twice now, it looks as if the switch has the D chan turned down on their side. |
| 13:51.02 | Dr-Linux | tzanger: yes i understand that, but i don't wanna miss this chance so i wanna verify another thing as well |
| 13:51.20 | Strom_C | tzanger, that wouldnt necessarily cause a yellow alarm though, would it? he'd still be getting frame sync on the T1 |
| 13:51.26 | Dr-Linux | tzanger: i can see some intrrupts in "dmesg" output |
| 13:51.51 | tzanger | yellow alarm just means that I am seeing the other side, and the other side is sending an unframed, all-1s pattern |
| 13:51.51 | Dr-Linux | http://pastebin.com/720836 << here, so wanna make sure if thre is nothing wrong :S like irq etc |
| 13:52.21 | tzanger | instead of a red alarm, which is when *I* send out an all-1s unframed pattern because I cannot see *anything* on the rx pair |
| 13:52.46 | tzanger | pastebin doesn't tell anything good or bad there. it looks normal |
| 13:53.48 | Dr-Linux | tzanger: ok, i just called datacenter voice support, she said, after sometime their technician will call me back. |
| 13:54.58 | Dr-Linux | tzanger: when pri turned up, will i need to restart the asterisk, or it will auto change it's status to UP ? |
| 13:55.33 | tzanger | it will automatically go |
| 13:59.21 | Dr-Linux | tzanger: thanks |
| 14:00.43 | Henk | I'm getting this error but i'm pretty dure the password is correct Forbidden - wrong password on authentication for INVITE to "3113xxxx" |
| 14:01.08 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 14:01.52 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:01.52 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:02.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:02.58 | *** join/#asterisk vechers (n=svecher@64.61.117.139) |
| 14:03.10 | *** part/#asterisk vechers (n=svecher@64.61.117.139) |
| 14:04.20 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 14:05.52 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 14:06.38 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 14:08.14 | *** join/#asterisk Spy000007 (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 14:11.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:13.48 | *** join/#asterisk znoG_ (n=gs@205-17-235-201.fibertel.com.ar) |
| 14:14.04 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 14:14.25 | smackus | using the polycom phones, is there a way to display on the phone that an agent is logged in? |
| 14:15.43 | smackus | for example, is there something that can be displayed on the lcd screen, or can a light be displayed to show that the agent is in fact logged in? |
| 14:16.03 | smackus | can it show the name of the queue that they are logged into? |
| 14:18.03 | smackus | looks like everyone is asleep |
| 14:18.43 | vader-- | does anyone know if there is a way to get the cisco 7940G phones to display caller ID name instead of just the caller ID number? |
| 14:20.23 | *** join/#asterisk frenzy (n=frenzy@196.46.104.92) |
| 14:20.58 | *** join/#asterisk jake1932 (n=Administ@pool-68-163-52-92.phil.east.verizon.net) |
| 14:21.07 | frenzy | just wondering what the correct syntax is it for zconfig.h |
| 14:21.12 | frenzy | #define AGGRESSIVE_SUPPRESSOR or define AGGRESSIVE_SUPPRESSOR |
| 14:21.23 | Pj_ | # |
| 14:21.49 | Henk | hmm I'm at a loss here. Asterisk seems to have lost its capability to register to budgetphone. but i't not telling me anything in the logfiles. where can ask asterisk to log more ? |
| 14:22.06 | frenzy | I just cant seem to get rid of the echo |
| 14:22.12 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:22.13 | frenzy | I keep hearding myself |
| 14:22.14 | jake1932 | anyone know if the polycom IP*01 series phones work ok behind NAT? |
| 14:22.17 | frenzy | when talking |
| 14:22.27 | frenzy | what echo canceller is best rated? |
| 14:24.32 | frenzy | is there a way I can change impendience? |
| 14:25.12 | *** join/#asterisk W4 (n=W4@212.58.232.254) |
| 14:25.59 | smackus | jake1932: when configured correctly those phones work very well |
| 14:26.06 | frenzy | ? |
| 14:26.49 | jake1932 | smackus: only reason i asked was i read some horror stories about polycom + NAT. |
| 14:26.50 | Sonderblade | instead of hardcoding all extensions in extensions.conf and voicemail.conf, i want to fetch them from a database or something, is that possible? |
| 14:27.12 | Ahrimanes | Sonderblade: look at realtime |
| 14:27.57 | jake1932 | Sonderblade: it's only advisable with a large number of extensions as realtime adds complexity |
| 14:28.33 | frenzy | are there specific codecs that improves echo cancellation? |
| 14:28.56 | RoyK | frenzy: no, but echo cancellors are nice |
| 14:29.40 | Sonderblade | jake1932: how else can i for example change the email address of an extension? |
| 14:29.40 | frenzy | it doesnt work for me |
| 14:29.40 | frenzy | I keep hearding myself |
| 14:29.40 | frenzy | hearing** |
| 14:29.40 | jake1932 | Sonderblade: for vm? |
| 14:29.53 | RoyK | frenzy: what sort of clients? |
| 14:29.54 | Sonderblade | jake1932: yes and so on |
| 14:30.13 | frenzy | it personal |
| 14:30.17 | frenzy | my own pbx |
| 14:30.18 | jake1932 | Sonderblade: just change it in the config files. unless you have a large amount of extens |
| 14:30.35 | jake1932 | Sonderblade: how many users? |
| 14:30.38 | frenzy | both phone and * are on my local network |
| 14:30.42 | RoyK | frenzy: what sort of clients? softphones? ATAs? |
| 14:30.44 | Sonderblade | jake1932: the customer wants to be able to change it using a gui |
| 14:30.56 | frenzy | LoL --- ATAs |
| 14:31.04 | frenzy | Grandstream |
| 14:31.06 | Strom_C | Sonderblade, my argument against the GUI is this: |
| 14:31.16 | Strom_C | you wouldnt let the secretary admin the webserver |
| 14:31.29 | Strom_C | so why the hell are you letting the secretary admin the PBX? |
| 14:31.49 | Strom_C | leave the configuration in the hands of a competent professional (i.e. you) |
| 14:31.53 | jake1932 | Sonderblade: do they have a network admin? |
| 14:31.57 | RoyK | Sonderblade: just use realtime |
| 14:32.19 | frenzy | ? |
| 14:32.26 | Sonderblade | jake1932: yup, but noone competent enough to edit /etc/asterisk/extensions.conf |
| 14:32.46 | RoyK | Sonderblade: well, use agi with a db - that works |
| 14:32.47 | RoyK | ] |
| 14:32.51 | RoyK | we do that |
| 14:33.05 | jake1932 | Sonderblade: that's pretty bad |
| 14:33.09 | Sonderblade | Strom_C: thats different, you allow the secretary to upload content using a cms |
| 14:33.58 | jake1932 | Sonderblade: you could try a gui based installation of asterisk like asterisk at home, but I personally don't like it |
| 14:34.17 | Sonderblade | RoyK: how? why that instead of realtime? |
| 14:34.28 | *** join/#asterisk umay (n=chris@71-208-188-148.hlrn.qwest.net) |
| 14:35.35 | sumdingwong | Royk: Ahrimanes: how many clients do you guys run on realtime? |
| 14:35.53 | RoyK | a few thousands |
| 14:35.54 | Ahrimanes | sumdingwong: i dont use realtime.. just know it's there :) |
| 14:36.09 | sumdingwong | Royk: Have you ever had any problems with it? |
| 14:36.25 | RoyK | no |
| 14:36.32 | RoyK | not with realtime |
| 14:36.36 | jake1932 | Sonderblade: how many extens? |
| 14:36.39 | *** join/#asterisk Dovid (n=none@barak.cellcom.co.il) |
| 14:36.43 | sumdingwong | I have a small ISP with 500-600 clients now |
| 14:36.43 | Sonderblade | jake1932: developing the gui isn't hard, i just want to know the best way to separate customer config (email addresses, names etc) from "hard config" like codecs, protocols and special extensions |
| 14:37.00 | Sonderblade | jake1932: ~30 |
| 14:37.04 | sumdingwong | sip.conf is getting hard to manage |
| 14:37.05 | *** part/#asterisk imbrandon (n=brandon@ubuntu/member/pdpc.active.imbrandon) |
| 14:37.14 | smackus | ok, so back to my question about displaying on the phone that the agent is logged in, I have come across this in the polycom configs, feature.15.name="acd-login-logout" has anyone used this? i have enabled it, and have seen no difference in my display. |
| 14:37.15 | sumdingwong | Do you use realtime for extensions.conf also? |
| 14:37.57 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 14:38.07 | Dovid | Having a problem with IAX real time |
| 14:38.24 | Dovid | Iax2 show registry show that I am registerd but when I wana make a call I get an error 3 for no route |
| 14:38.40 | sumdingwong | Dovid: Do you use realtime for extensions.conf? |
| 14:38.58 | Dovid | yes |
| 14:38.59 | jake1932 | Sonderblade: my advice - seperate the special extens into a separate file, include it, and only have them edit that file. even a bad admin should get the idea of how the file should be laid out. otherwise, go with a GUI based asterisk |
| 14:39.11 | Dovid | And for iax users, sip users etc. |
| 14:39.32 | jake1932 | oops - should be - seperate the regular extens into a separate file |
| 14:40.08 | sumdingwong | Dovid: I assume you are using a gui for this ... is it a homebrew, commercial or foss? |
| 14:40.49 | frenzy | Is there a way i can change impendience? I belive default is 600ohm |
| 14:40.54 | *** join/#asterisk kpettit (n=keith@adsl-70-241-118-252.dsl.hstntx.swbell.net) |
| 14:41.10 | Dovid | lol |
| 14:41.15 | kpettit | Having a problem getting musiconhold to seound decent. |
| 14:41.18 | kpettit | right now i"m doing "sox mohSPAN.wav -r 44100 -c1 span.mp3" for a 16 bit, stereo 44100 Hz file |
| 14:41.21 | Dovid | No gui using ssh :) |
| 14:41.24 | Sonderblade | jake1932: to add a users you have to edit lines in three files: extensions.conf, sip.conf, voicemail.conf it's just way to much |
| 14:41.25 | *** join/#asterisk hensema (n=erik@scrat.hensema.net) |
| 14:41.47 | kpettit | it sounds pretty crappy though. Any ideas on a better way to encode it? I'm using * 1.2.4 on this box |
| 14:41.57 | feld_ | Sonderblade: that's the consequence of having so much power and configurability, though. |
| 14:42.10 | Dovid | For some reason it wont send the user id's and pass to the IAX providers, I think I am gona go with static |
| 14:42.13 | nortex | Is there a way to set the caller id when someone transfers a call? |
| 14:42.21 | feld_ | You could always look at Asterisk@Home or AMP or whatever else is floating around with a GUI wrapped around * |
| 14:42.35 | feld_ | nortex: yes |
| 14:42.47 | feld_ | you can set caller IDs in your dialplan |
| 14:42.55 | hensema | when I'm called with a blocked caller-id, I can't set $CALLEDID(name) or $CALLERID(number) to anything, asterisk seems to ignore it and send unknown/unknown to my sip phone |
| 14:42.56 | sumdingwong | Dovid: just ssh .... sounds like a lotta typin to me.... |
| 14:43.00 | Sonderblade | feld_: do they use realtime? |
| 14:43.01 | hensema | any clue what's going on? |
| 14:43.07 | feld_ | Sonderblade: I'm not sure |
| 14:43.55 | Dovid | Yup. |
| 14:44.10 | Dovid | Its the real way to do it. Once u get used to it its not hard |
| 14:44.37 | sumdingwong | I guess it would definately polish the old MySQL syntax ;) |
| 14:44.56 | *** join/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 14:45.33 | sumdingwong | I'm just lazy |
| 14:45.43 | sumdingwong | I only work hard to keep from working |
| 14:46.07 | variable_office | i have realtime setup, but i always get "Requested contexts didn't get merged" in the messages log, and when stuff is dialed it fails saying it cant find main1,s,1 even though i have that in the db |
| 14:46.12 | nortex | feld_, I currently am setting it on all incomming calls, but where do I put the set command for a transfer? Before the inside extensions? |
| 14:46.18 | variable_office | is there something i need in extensions.conf? |
| 14:46.55 | saftsack | does hylafax work with every external modem? |
| 14:47.03 | feld_ | nortex: i'm not 100% sure. maybe someone else has more experience with that. |
| 14:47.53 | Dovid | Dont know. I am having issues with IAX real time as well |
| 14:48.21 | variable_office | i cant seem to find anything past setting realtime up, but not actually how to get it to work |
| 14:48.40 | Dovid | Real time for what ? Extensions or IAX ? |
| 14:48.51 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 14:48.58 | variable_office | Dovid extensions, at least for now |
| 14:49.05 | Dovid | I did it for extensions, its real simple once u get used to it |
| 14:50.21 | *** join/#asterisk Jick (n=chatzill@68-118-131-224.dhcp.sprn.tx.charter.com) |
| 14:50.25 | Qb3rt | polycom ip 4000 i have a problem with the configuration files.... iam trying to used the as i have for 501 and 301... it says application image error |
| 14:50.38 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 14:51.18 | Qb3rt | application image error... i think its more a problem with the bootrom... no??? what do u think? |
| 14:52.12 | smackus | does anyone have any experience making the acd agents phone display that they are logged in? |
| 14:52.14 | [TK]D-Fender | Qb3rt : You need to make sure that the 4000 is provisioning from its own set of files because its requirements are different that the desktop series |
| 14:52.25 | smackus | i am using the polycom soundpoint ip phones |
| 14:52.27 | Jick | Does anyone know what happens to call logs when an Asterisk server using MySQL for its CDR records can't make a connection to that database? |
| 14:52.51 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 14:52.52 | Jick | Are logs queued to be inserted when the next connection is made? |
| 14:53.04 | Ahrimanes | Jick: either you have a backup copy from cdr_csv in Master.csv or you lose the records i guess |
| 14:53.22 | _problem_ | can anbody help me related to http://pastebin.ca/67728? |
| 14:53.27 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 14:53.30 | Jick | eh... that's no good. :) |
| 14:53.50 | [TK]D-Fender | Qb3rt : Go grab 1.6.6. and set a seperate upgrade folder for the 4000 and then in your <MAC>.cfg files do NOT tell it what SIP / BR to pick up after you get it functioning. that will stop it from autoupdating to a potentially bad version |
| 14:53.58 | Ahrimanes | Jick: enable cdr_csv and feel good knowing that you can always sync your mysql from Master.csv |
| 14:54.10 | Jick | yeah |
| 14:54.42 | [TK]D-Fender | _problem_ : Pastebin your dialplan |
| 14:54.55 | Ahrimanes | Jick: backup plans ftw :) |
| 14:54.57 | Jick | Ahrimanes: how sure are you that cdr_mysql won't try to account for socket disconnects? |
| 14:55.00 | _problem_ | ok |
| 14:55.32 | Ahrimanes | Jick: am not completely sure, but in the event that the mysql stays down, master.csv is _good_ to have |
| 14:55.54 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 14:55.55 | Jick | right. but awking it for missing records isn't :) |
| 14:56.05 | Ahrimanes | hehe |
| 14:56.10 | Ahrimanes | perl ftw |
| 14:57.21 | [TK]D-Fender | ~pb |
| 14:57.22 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 14:57.55 | _problem_ | [TK]D-Fender: i have a very big and complicated extensions.conf..do u want me to paste whole file ..if u wants i can paste the required things. |
| 14:58.57 | nortex | Does the variable TRANSFER_CONTEXT go in a specific section of features.conf? |
| 14:59.00 | flujan | guys, how can I use chanspy to spy a specific channel? |
| 14:59.01 | *** join/#asterisk wunderkin (n=wunderki@69.26.192.234) |
| 14:59.06 | flujan | for instance, IAX/123456 |
| 14:59.12 | Ahrimanes | nortex: it goes in extensions.conf |
| 14:59.18 | [TK]D-Fender | _problem_ : Shoev the whole thing. I want to make sure you don't overlook handing me something necessary |
| 14:59.48 | _problem_ | ok |
| 14:59.52 | nortex | Ahrimanes, Thanks, I wonder why it is the wiki on the features.conf section. Do you use it? |
| 15:00.12 | *** join/#asterisk wintix (n=tobias@pegel-neuburg.de) |
| 15:00.14 | vader-- | could anyone tell me why this dial plan isn't working past the last like exten => s,n,Set(TIMEOUT..... |
| 15:00.15 | vader-- | http://pastebin.ca/67732 |
| 15:00.23 | vader-- | it just disconnects the call |
| 15:00.26 | Ahrimanes | nortex: yes i use it, and its on that page because it related to features blind transfer and attended transfer i guess |
| 15:02.13 | nortex | Ahrimanes, Can I use it to change the caller id and possibly distinctive ring on calls transfered? I'm thinking a couple lines to set things and then an include to my inside extensions. Will that work? |
| 15:03.49 | [TK]D-Fender | vader-- "autofallthrough=no" <- required |
| 15:06.13 | Dovid | Anyone here using queuese and real time ? |
| 15:06.17 | Dovid | Queues* |
| 15:07.38 | smackus | so i have the agen stuff turned on for the polycom phones. It shows the graphic on the lines, the little head, but it does not change weather or not it is logged in or out. Does anyone know how to set this up? |
| 15:08.15 | Ahrimanes | nortex: should work fine yes, it just tells * to go to ${TRANSFER_CONTEXT} on a transfer and execute whatevers there |
| 15:09.23 | hensema | why are all assignments to $CALLERID() ignored when an anonymous call is answered? |
| 15:09.34 | *** join/#asterisk Kte2 (n=Grumoz@209.151.130.10) |
| 15:09.43 | hensema | eg whatever I do, I don't get to see a callerid |
| 15:10.06 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 15:11.58 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 15:12.33 | flujan | I'm having difficult to spy a specifig IAX2 channel using chanspy. |
| 15:13.02 | variable_office | here are the confs i had for asterisk w/ realtime, but i still get the same cannot find main1,s,1 error |
| 15:13.03 | variable_office | http://pastebin.ca/67739 |
| 15:13.15 | flujan | using the command ChanSpy(IAX2/123456|q), asterisk show that I'm spying the channel, but I have no audio output in my channel. |
| 15:15.41 | asterboy | When talking on a call, IF you talk while the other party is talking, it gets garbled...how in the heck do you fix that??? |
| 15:16.35 | asterboy | Sangoma A200 ---> * ---> Polycom IP600 |
| 15:17.03 | [TK]D-Fender | asterboy : what kind of EC in between? |
| 15:17.10 | asterboy | no hardware |
| 15:17.21 | [TK]D-Fender | asterboy : echotraining? |
| 15:17.22 | asterboy | just * with typical EC |
| 15:17.27 | asterboy | yes echo training |
| 15:17.34 | [TK]D-Fender | asterboy : try disabling |
| 15:17.35 | asterboy | but that's only at the beginning of a call |
| 15:17.47 | [TK]D-Fender | # taps? |
| 15:17.48 | asterboy | ok, I'll try it and see what happens. |
| 15:17.57 | asterboy | as in pots? |
| 15:18.27 | asterboy | what do you mean by taps? |
| 15:18.33 | [TK]D-Fender | asterboy : Pastebin your zapata |
| 15:21.09 | asterboy | http://pastebin.ca/67740 |
| 15:22.03 | asterboy | Zaptel has Mark2 with Aggressive enabled |
| 15:22.10 | Qb3rt | do anyone know where can i find the boot rom 3.1.2 polycom |
| 15:22.22 | asterboy | ~polycom |
| 15:22.23 | jbot | from memory, polycom is the manufacturer of one of the best IP phones in the market. http://polycom.com - Note: Here is where you can get some downloads: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 15:23.18 | l-fy | snom are the best phones on the market |
| 15:23.20 | l-fy | not polycom |
| 15:23.20 | asterboy | Dam annoying when trying to talk with someone and you get tripped up when you talk while they talk. |
| 15:23.46 | asterboy | snom? ah...does it have a microbrowser? |
| 15:23.56 | *** join/#asterisk Beighto (n=chatzill@64.160.113.130) |
| 15:24.00 | asterboy | if no --> snom = crap |
| 15:24.06 | *** join/#asterisk nothinman (i=shakey@83.11.253.182) |
| 15:24.22 | Qb3rt | snom = crap anyway! |
| 15:24.30 | asterboy | lol |
| 15:24.31 | nortex | asterboy, Only one Polycom has the microbrowser |
| 15:24.40 | asterboy | And I love it. |
| 15:24.52 | asterboy | albiet it is a very regimented browser |
| 15:25.20 | tzanger | asterboy: what do you use the browser for |
| 15:25.23 | asterboy | Digg, Weather, Calendar, Pictures |
| 15:25.33 | Sonderblade | is it possible to get the dialplan both from extensions.conf and from realtime? |
| 15:25.34 | tzanger | interesting |
| 15:25.56 | tzanger | I want to pull apart this 501 and see how hard it is to add a white-led backlight/sidelight |
| 15:26.16 | asterboy | Wife loves the weather report, kids love seeing themselves on a phone, I love Digg RSS feed. |
| 15:26.25 | nothinman | hello... |
| 15:26.31 | Dr-Linux | tzanger: the telco said, PRI is turned on, and also they given me DID's, but same happend at my end |
| 15:26.39 | asterboy | The Calendar is kick ass cause it can show you your appointments |
| 15:27.08 | asterboy | backlight on a Polycom would be sweet |
| 15:27.18 | nothinman | could you please tell me whether of not it is normal that bridge call is being answered immediately (when still ringing) and sip call is just... ringing? |
| 15:27.29 | nothinman | timeoute doesn't seem to work on two zap's.. |
| 15:27.35 | nothinman | timeout* |
| 15:27.58 | asterboy | nothinman, you should pastebin your extensions.conf |
| 15:28.21 | nothinman | em, can I just paste few lines from asterisk command line? |
| 15:28.32 | asterboy | ~pb |
| 15:28.33 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 15:28.45 | tzanger | Dr-Linux: do you have the correct switchtype emulation? Is the D channel provisioned on their end or is it administratively down? You have to specifically ask them if they've got the D channel held down to silence alarms |
| 15:29.32 | [TK]D-Fender | asterboy : Get rid of immediate=yes, beyond that I'm not sure. As a factor could be that you switch the EC algorythm |
| 15:30.28 | *** join/#asterisk `lyme (n=Lyme@manufacturerstransportation.com) |
| 15:31.00 | Dr-Linux | tzanger: thanks friend, now my PRI is up and i can make dialout calls, |
| 15:31.05 | smackus | ok, I am reading around trying to figure out how to get acd agents to display that they are logged in... I have come across a post on a mailing list, but I do not understand what it is saying to do. Can someone more familiar with how to do patches against asterisk read this and see if you can explain to me what I need to do: http://www.threebit.net/mail-archive/asterisk-users/msg00721.html |
| 15:31.09 | nothinman | asterboy: http://pastebin.ca/67763 |
| 15:31.43 | Dr-Linux | tzanger: but when i dial the DID"s that i was given by telco, it says "the number you are dialing is disconnected or no longer in use" |
| 15:31.58 | tzanger | Dr-Linux: :-) a lot of times they get everything ready and then set the D chan down until you call and ask why it's not working... and usually they forget they've done that and sendd you on a wild goose chase because it's obviously your end that is incorrectly configure |
| 15:32.16 | smackus | i can set the agent login up on the phone, but no matter what the logged in status of the agent is, it shows the same graphic on the phone. |
| 15:32.28 | tzanger | Dr-Linux: do you have the zaptel channels dumping into a working contest? |
| 15:32.30 | tzanger | er context? |
| 15:32.33 | tzanger | like this? |
| 15:32.36 | tzanger | [PRI] |
| 15:32.40 | nothinman | asterboy: zap to sip works fine. zap to zap seems to be "answering" but it's still ringing... |
| 15:32.53 | Dr-Linux | tzanger: yes, but i can't see anything on the CLI |
| 15:32.53 | tzanger | exten => _X.,1,NoOp(Exten is ${EXTEN}) |
| 15:33.22 | tzanger | exten => _X.,n,Dial(SIP/${EXTEN}@DrLinux) |
| 15:33.23 | tzanger | ? |
| 15:33.37 | Dr-Linux | tzanger: here is my zapata.conf > http://pastebin.com/719239 |
| 15:33.46 | Dr-Linux | and the context is ivr |
| 15:34.17 | tzanger | put what I had in there |
| 15:34.27 | tzanger | _X. where it shows you what ${EXTEN} is |
| 15:35.12 | smackus | ACD... no one |
| 15:35.35 | nothinman | asterboy: any ideas? :-) |
| 15:36.31 | asterboy | ya, I had this too at the beginning. |
| 15:37.06 | asterboy | I think it had to do with zaptel monitoring the calls |
| 15:37.28 | asterboy | callprogress=no, iirc is how I solved that |
| 15:37.48 | Dr-Linux | tzanger: i'm sorry i didn't understand your dialplan :( |
| 15:38.22 | asterboy | nothinman, you need to paste your extensions.conf, not just a few lines. |
| 15:38.35 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 15:38.40 | asterboy | kill any passwords and context sensitive info with **** |
| 15:38.51 | nothinman | asterboy: man, it's over 500KB in size... |
| 15:38.59 | asterboy | why so big? |
| 15:39.12 | nothinman | asterboy: were you talking to me about this callprogress=no? |
| 15:39.18 | asterboy | yep |
| 15:39.25 | asterboy | min is 6Kb |
| 15:39.26 | nothinman | asterboy: because I didn't split IVR in few files yet ;) |
| 15:39.36 | nothinman | okay, let me check. 2 mins. |
| 15:43.35 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:43.37 | Dr-Linux | tzanger: when i call from my same system where PRI connected it gives me this: |
| 15:43.37 | Dr-Linux | Zap/1-1 is proceeding passing it to SIP/4092-74f5 |
| 15:43.38 | Dr-Linux | <PROTECTED> |
| 15:43.38 | Dr-Linux | <PROTECTED> |
| 15:43.57 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 15:44.21 | nothinman | nah, I added callprogress=yes to zapata.conf, and looks like the problem is gone! |
| 15:44.25 | asterboy | so does 2109 exist in context "ivr" from 4092? |
| 15:44.43 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 15:44.46 | asterboy | nothinman, so it works now? Great! |
| 15:45.19 | asterboy | callprogress when bridging is usually what I've had to play with. |
| 15:45.20 | variable_office | is there a way to check to make sure asterisk can see that the contexts/extensions are properly loaded from realtime |
| 15:45.21 | variable_office | ? |
| 15:45.30 | nothinman | asterboy: lovely! with =yes it does! |
| 15:45.39 | nothinman | asterboy: thanks a lot dude :-) |
| 15:45.49 | asterboy | variable_office, watch CLI when you turn verbose up |
| 15:46.01 | asterboy | ok, glad I could help. |
| 15:46.14 | variable_office | right now it is at default 3, how can i turn it up to max? |
| 15:46.34 | asterboy | I just use asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 15:46.37 | asterboy | :P |
| 15:46.42 | asterboy | verbose 10 |
| 15:47.33 | asterboy | set verbose 10 |
| 15:47.54 | asterboy | at clie |
| 15:47.56 | asterboy | cli |
| 15:48.16 | asterboy | and then do an extensions reload to see the file being read |
| 15:48.22 | variable_office | well at 3 it would say, cannot find main1,s,1 main1 being the context i set for zap/1 in zapata.conf |
| 15:48.39 | asterboy | well it's not lieing |
| 15:48.50 | asterboy | it can't find it. |
| 15:48.55 | *** join/#asterisk salviadud (n=ralfalfa@201.145.29.99) |
| 15:49.10 | Dr-Linux | [TK]D-Fender: can you have a look on this. |
| 15:49.20 | Dr-Linux | <PROTECTED> |
| 15:49.35 | variable_office | when i did "extensions reload" it said Requested contexts didnt get merged |
| 15:49.38 | Dr-Linux | http://pastebin.com/721133 |
| 15:50.37 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:51.36 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 15:52.48 | variable_office | asterboy what could be causing the contexts to not be merged |
| 15:57.44 | [TK]D-Fender | Dr-Linux : means what it says.... |
| 15:59.03 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 15:59.29 | Blackthorn | Hi, could someone tell me about a message that I get sometimes "Jun 19 17:11:02 WARNING[2850] chan_iax2.c: Received mini frame before first full voice frame" |
| 15:59.45 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 16:01.39 | Sonderblade | Blackthorn: i can't, but I've also received that message and it seems mostly harmless |
| 16:01.43 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:03.08 | Blackthorn | umm ok |
| 16:03.12 | *** part/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
| 16:04.02 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 16:05.34 | frenzy | what is rxwink ? |
| 16:06.03 | justinu|laptop | that's for E&M signalling |
| 16:06.07 | blebleble | Sets the receive wink timing |
| 16:06.33 | *** join/#asterisk MamboKing (n=mambo@d38-23-38.commercial1.cgocable.net) |
| 16:06.35 | MamboKing | hey all |
| 16:06.39 | frenzy | I'm using x100p |
| 16:06.39 | Blackthorn | hi |
| 16:06.47 | frenzy | and I keep hearing myself |
| 16:07.04 | justinu|laptop | frenzy: adjust gains |
| 16:07.08 | frenzy | its very fustrating I've setup all echo cancellation |
| 16:07.08 | justinu|laptop | ~fxotune |
| 16:07.14 | justinu|laptop | ~gains |
| 16:07.19 | [TK]D-Fender | frenzy : Go read up on analog echo cancellation |
| 16:07.26 | justinu|laptop | ~txgain |
| 16:07.28 | frenzy | I did fxotune.. |
| 16:07.39 | justinu|laptop | if you hear yourself, txgain is too high |
| 16:07.40 | blebleble | anyone have much experience with hylafax? does it do t.38? |
| 16:07.45 | MamboKing | has anyone used asterisk for setting up a small dialup server? or alternately if anyone knows of some good documentation out there for doing so? |
| 16:07.53 | Blackthorn | This question is a bit off-topic to asterik but maybe somone can help. I'm using webmin and the file manager I can move around and edit most anyting but It won't allow me to enter the etc/asterisk director? any suggestion? (i'm loged in as root at this time) |
| 16:08.24 | blebleble | see what the permissions of the directory are |
| 16:08.26 | blebleble | stat dir |
| 16:08.56 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:09.08 | flujan | guys, someone can help me using chanspy app? |
| 16:09.15 | flujan | I want to monitor a specific channel... |
| 16:09.23 | flujan | for instance IAX2/123456 |
| 16:09.45 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 16:10.58 | flujan | someone here use it? |
| 16:11.10 | uwe | hello, i have asterisk 1.0.10, and for sometime now there is no entries of the calls in the database! any idea why this could happen? any hints on debugging this is appreciated |
| 16:13.20 | Blackthorn | says modify access and change all st to -400 |
| 16:14.39 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:17.10 | variable_office | do i possibly need to have something inside extensions.conf to make realtime extensions work? |
| 16:18.13 | *** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
| 16:18.15 | *** part/#asterisk mfedyk (n=mfedyk@adsl-63-194-240-129.dsl.lsan03.pacbell.net) |
| 16:18.45 | *** join/#asterisk NewSole (n=dave@d226-105-226.home.cgocable.net) |
| 16:19.09 | frenzy | ive read and tried alot on echo cancellation but it simply wont work |
| 16:19.58 | [TK]D-Fender | variable_office : Yes. |
| 16:20.42 | kegg | HI there |
| 16:20.58 | kegg | i've got a problem with my sangoma a200 card |
| 16:21.11 | variable_office | [TK]D-Fender what must be in extensions.conf? |
| 16:21.20 | kegg | or it may be a a problem in chan_zap, im not sure |
| 16:21.31 | *** part/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 16:21.49 | kegg | has anybody experiences with zap and or sangoma and callerID ? |
| 16:22.38 | kegg | my asterisk segfaults as soon as i make an inbound call |
| 16:22.49 | kegg | any ideas? |
| 16:23.30 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:23.51 | flujan | hi [TK]D-Fender :) |
| 16:24.07 | flujan | do you have experience using chanspy? |
| 16:25.26 | kegg | okay, so i've to solve my own |
| 16:25.30 | kegg | cu all |
| 16:25.58 | flujan | I don't know how to specify a channel to monitor... For instance, chanspy ( IAX2/123456) |
| 16:26.05 | flujan | i just listen to the beeps... :( |
| 16:26.19 | [TK]D-Fender | flujan : Nope |
| 16:27.02 | [TK]D-Fender | variable_office : You need to do all of your "switch" statements by context. Go read the instructions again |
| 16:27.32 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
| 16:27.51 | variable_office | i havent found anything that had talked about how to do stuff like that. do you know a good site [TK]D-Fender ? |
| 16:27.55 | Hmmhesays | Q: Why do prostitutes demand pandas pay up front? |
| 16:28.07 | Hmmhesays | A: Because a panda eats bushes and leaves. |
| 16:28.25 | Hmmhesays | LOL |
| 16:28.28 | [TK]D-Fender | variable_office : Go to the WIKI |
| 16:28.28 | blebleble | wow |
| 16:28.44 | [TK]D-Fender | Hmmhesays : Bastardized short version of an otherwise good joke... |
| 16:28.56 | Hmmhesays | this is IRC what the hell do you expect |
| 16:29.09 | variable_office | [TK]D-Fender this http://www.voip-info.org/wiki-Asterisk ? |
| 16:29.11 | [TK]D-Fender | Hmmhesays : And its supposed to be koala's. Panda's wouldn't screw to save their own species... |
| 16:29.31 | [TK]D-Fender | variable_office : Yes, and THE BOOK |
| 16:29.32 | [TK]D-Fender | ~book |
| 16:29.34 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:29.34 | Hmmhesays | give me a freaking break |
| 16:29.54 | Hmmhesays | time to watch some tortured souls on youtube |
| 16:31.31 | variable_office | [TK]D-Fender thanks, so i have to switch for each context i have |
| 16:31.32 | variable_office | amazing |
| 16:33.07 | frenzy | my echo is one sided only |
| 16:33.40 | frenzy | using the sip ata i hear myself |
| 16:33.50 | frenzy | however on the otherend its perfect |
| 16:34.49 | frenzy | i think its more like sidetone |
| 16:34.56 | frenzy | how do i eliminate that? |
| 16:36.28 | variable_office | [TK]D-Fender i thank you much, i finally progressed |
| 16:38.03 | [TK]D-Fender | variable_office : np |
| 16:41.26 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 16:42.32 | salviadud | Hmmhesays, tortured souls? |
| 16:42.45 | Hmmhesays | some chick reading poetry on youtube |
| 16:43.07 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 16:43.18 | *** join/#asterisk grabowski (i=grabowsk@i.use.efnut.com) |
| 16:43.57 | MRH2 | hi anybody happen to know the state of play with srtp provision on polycom phones? |
| 16:44.51 | Dr-Linux | [TK]D-Fender: my first PRI is working perfectly fine, outgoing/incoming both :) but my 2nd PRI status is still down |
| 16:45.05 | *** join/#asterisk mixi (n=mixi@p5491FB51.dip.t-dialin.net) |
| 16:45.07 | [TK]D-Fender | MRH2 : SIP 2.0 |
| 16:45.15 | mixi | hi everybody |
| 16:45.22 | Dr-Linux | [TK]D-Fender: my telco says, it's problem at my end, he adviced me to restart my CSU |
| 16:45.29 | Dr-Linux | what's CSU |
| 16:45.34 | Dr-Linux | ~CSU |
| 16:45.37 | mixi | i've a four quadBRI (cologne chipset), which chan_ is most recommended for usage? |
| 16:45.39 | MRH2 | so it is a standard as from SIP 2.0 (?) |
| 16:45.53 | mixi | i've one |
| 16:45.57 | mixi | not four |
| 16:46.30 | userdefined | flujan: i think you just set the extension to chanspy(IAX2), then once you're in dial 123456 (to use your example) |
| 16:46.33 | mixi | four isdn b channels running te and four runnin nt mode |
| 16:47.19 | mixi | chan_capi or bristuff (chan_zap) ? |
| 16:48.30 | userdefined | flujan: correction, 123456# to monitor that specific one (i forgot the #) |
| 16:48.34 | mixi | bristuff worked fine so far, but as i am updating the whole asterisk box, i've the option to switch |
| 16:48.40 | mixi | that's why i'm asking |
| 16:49.29 | flujan | userdefined, is not possible to pass this information? |
| 16:50.15 | userdefined | doesn't seem so (reading the doc from running 'show application chanspy' at the CLI) ... i just have mine set to chanspy(all) and use * to cycle through |
| 16:50.52 | websae | anyone know where one could get a nice pbx box with p4 1gb ram ?...just curious |
| 16:51.11 | [TK]D-Fender | www.justfindalocalretailerandwhiteboxit.com :) |
| 16:52.22 | [TK]D-Fender | and don't give me any lip about the URL not resolving! |
| 16:52.28 | flujan | userdefined, I tried without success |
| 16:52.31 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
| 16:52.41 | flujan | every time I press a number in the dialpad I just hear other beep. |
| 16:52.42 | flujan | :( |
| 16:52.48 | *** join/#asterisk dant (n=dan@host-84-9-188-2.bulldogdsl.com) [NETSPLIT VICTIM] |
| 16:52.50 | *** join/#asterisk toBee (i=ts@wayfinder.blazing.de) |
| 16:52.54 | toBee | moin |
| 16:52.58 | mixi | no one runnin asterisk with isdn? |
| 16:53.00 | MattH | about every 5 seconds I have a "Remote UNIX connection" and then disconnecting... how can I figure out where this is coming from? the full debug doesn't give an IP |
| 16:53.05 | SplasPood | hrm... |
| 16:53.18 | [TK]D-Fender | MattH : AMI <- |
| 16:53.24 | *** part/#asterisk variable_office (n=variable@Adv-Proprietary-Systems.s7-0-0.2-15-0.ar4.CHI1.gblx.net) |
| 16:53.25 | mixi | MattH that's a daemon trying to access asterisk |
| 16:53.35 | MattH | I realize that |
| 16:53.41 | MattH | how can I get an IP though? |
| 16:53.44 | MattH | or figure out what is trying to access it |
| 16:53.52 | SplasPood | s/Voicemail/VoicemailMain/ |
| 16:53.55 | mixi | a certain ps should do |
| 16:54.01 | mixi | something like ps -aux |
| 16:54.07 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 16:54.07 | mixi | you might change the params |
| 16:54.22 | mixi | not sure what's param for unix sockets |
| 16:54.56 | TESTER2 | Where can I find the last version of fxotune.c (because I found many patchs...) ? |
| 16:55.52 | MattH | odd |
| 16:55.57 | MattH | apparently asterisk never cleanly shut down |
| 16:56.03 | MattH | so safe_asterisk was polling it every 4 seconds |
| 16:56.18 | mixi | killall safe_asterisk |
| 16:56.22 | mixi | nohup safe_asterisk |
| 16:56.46 | MattH | that brings me to my next question... I upgraded to 1.2.9.1 today.. and it was working fine.. but just a moment ago I couldn't send new calls to the queues...just went to dead air... dial plan didn't work.. and if I did a 'show agents' 'show queues' 'zap show channels', etc nothing happened |
| 16:57.15 | MRH2 | oic a new "SIP2.0" phone |
| 16:57.41 | [TK]D-Fender | MRH2 : No, SIP Application 2.0 Due for imminent release. |
| 16:58.34 | MRH2 | I mean looks like the new Soundpoint 430 does srtp |
| 16:59.21 | MRH2 | sip 2.0 as in a firmware update from 1.6.x? |
| 16:59.40 | toBee | someone in here who has got experiences in transfering calls via capiHOLD in a queue-setup? |
| 17:00.49 | MRH2 | TK - sorry for being dense lol |
| 17:01.17 | MRH2 | found a press release now |
| 17:01.28 | flujan | userdefined, no success. |
| 17:02.17 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 17:02.28 | uwe | hello, im having trouble with committing the calls logs to database, it appears that i dont have cdr_mysql module! im using asterisk package from xorcom on debian, and when itry to apt-get install asterisk-mysql it cant find asterisk-mysql_1.0.9-2_i386.deb, any suggestions how i can get crd_mysql for my system? |
| 17:02.59 | _Sam-- | get asterisk-addons from source and compile it |
| 17:03.27 | MikeJ[Laptop] | or find the package maintainer and get them to fix it.. |
| 17:03.41 | uwe | _Sam--, thank you, would the latest one work for me? im using asterisk 1.0.10 ! |
| 17:03.42 | _Sam-- | debian asterisk packages suck...although there is a newer one than he is trying. |
| 17:03.57 | _Sam-- | uwe: im not positive, but i think it will. |
| 17:04.20 | uwe | :) thank you very much _Sam-- , MikeJ[Laptop] |
| 17:04.35 | _Sam-- | you may also want to check/change your apt source(s) to get a newer asterisk package |
| 17:04.36 | MikeJ[Laptop] | wait.. you are using asterisk 1.0 ? |
| 17:04.54 | MikeJ[Laptop] | you are going to want to pull the 1.0 branch of addons then.. the modules have changed |
| 17:05.00 | MikeJ[Laptop] | I beleive |
| 17:06.14 | uwe | well ... |
| 17:06.30 | _Sam-- | it only takes 2 seconds to compile and try |
| 17:06.35 | [TK]D-Fender | MRH2 : Yes. The IP430 comes with SIP 2.0 by default. |
| 17:06.47 | MRH2 | TK do you happen to know if the upcomming release will add srtp to the IP500 as well? or only the phones with more memory? |
| 17:06.49 | uwe | the problem is i cant risk breaking the current system, so i have to test the upgrade on another machine ... when i have one |
| 17:06.52 | [TK]D-Fender | MRH2 : Its a synchronized release |
| 17:06.59 | [TK]D-Fender | MRH2 : Yes, to all. |
| 17:07.10 | MRH2 | absolutely fantastic! |
| 17:07.12 | _Sam-- | there is nothing to break...compile asterisk-addons...copy cdr_something_mysql to /usr/lib/asterisk/modules |
| 17:07.45 | _Sam-- | if your cdr_mysql_whatever isnt working now...how can you break it? |
| 17:07.46 | uwe | _Sam--, im in the process :) |
| 17:07.55 | [TK]D-Fender | MRH2 : I've been running 2.0 Beta for a while now. Its zippier, but I wish they ported the IP 430's LCD style to the other models... |
| 17:08.47 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 17:09.03 | *** join/#asterisk ringhals (i=fwuser@firewall.drgutah.com) |
| 17:09.05 | mixi | anyone running asterisk with isdn? |
| 17:09.19 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 17:09.33 | Mw3 | mixi: yes |
| 17:09.39 | MRH2 | TK does it work fine with srtp in asterisk? |
| 17:09.53 | _problem_ | mixi: i m running with misdn..what is ur problem ? |
| 17:09.57 | [TK]D-Fender | MRH2 : Never tried that part personally.. |
| 17:10.14 | [TK]D-Fender | MRH2 : * SRTP is trunk no? I only do Relase versions. |
| 17:10.31 | MRH2 | i think it is moved into trunk now |
| 17:10.39 | Beighto | looking for a succesful sip.conf example for broadvoice |
| 17:11.19 | ringhals | I am sure I have just missed a setting somewhere but my iax (softphones) will not stay registered. the become UNREACHABLE even though ping times are sub 20 milli to the server .. any ideas? |
| 17:11.20 | MRH2 | i was messing about with a branch |
| 17:11.49 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 17:11.50 | Nugget | Beighto: http://www.google.com/search?q=broadvoice%20sip.conf%20example would be a great start. |
| 17:12.02 | Nugget | the #1 hit seems to be exactly what you're asking for |
| 17:12.02 | tzafrir | uwe, what do you have on your sources.list? |
| 17:12.29 | *** join/#asterisk DMark_ (n=kk7cu@pool-70-105-217-25.scr.east.verizon.net) |
| 17:12.37 | _Sam-- | tzafrir : the current debian package is still 1.2.1? |
| 17:12.57 | tzafrir | no. My current package is 1.2.7.1 |
| 17:13.07 | tzafrir | (the "unstable" one) |
| 17:13.25 | _Sam-- | just curious...i dont use packages, but my apt-cache showpkg says 1.2.1 |
| 17:13.55 | _Sam-- | you still doing the zap modules ? |
| 17:14.03 | _Sam-- | or i forget what you maintened |
| 17:14.32 | tzafrir | But it doesn't have asterisk-mysql yet. I forgot about it |
| 17:14.53 | tzafrir | _Sam--, what do you have on your sources.list? |
| 17:15.15 | Dr-Linux | tzafrir: do you know what's CSU? |
| 17:15.33 | Dr-Linux | <PROTECTED> |
| 17:15.45 | Dr-Linux | <PROTECTED> |
| 17:15.57 | _Sam-- | csu = unit before your T1 connection |
| 17:16.11 | _Sam-- | digium cards have built in csu |
| 17:16.19 | _Sam-- | old t1 type hardware, you need external csu |
| 17:16.35 | tzafrir | Dr-Linux, google tells me that this is the name of a number of universities... |
| 17:17.24 | _Sam-- | tzafrir : trying to paste my source to pastebin...but they're running slow. |
| 17:17.28 | _Sam-- | sources |
| 17:17.59 | ringhals | I am sure I have just missed a setting somewhere but my iax (softphones) will not stay registered. the become UNREACHABLE even though ping times are sub 20 milli to the server .. any ideas? |
| 17:18.10 | MamboKing | does anyone know if there is a good howto on setting up asterisk as a dial-up server I've seen some posts stating that it is possible but nobody has posted any solutions or suggestions on how to go about doing so |
| 17:18.12 | Dr-Linux | tzafrir: WIKI says : SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 17:18.13 | _Sam-- | firewall/nat? |
| 17:18.49 | MRH2 | thanx TK g2g |
| 17:18.50 | *** join/#asterisk Joshaidan (n=icechat5@thunderbay-voip-4.vianet.ca) |
| 17:18.52 | ringhals | same subnet here if that wasfor me |
| 17:19.13 | _Sam-- | ringhals : same subnet, and nothing between the clients and the server, like no router or anything? |
| 17:19.14 | Dr-Linux | tzafrir: it's show in "zatcfg -vv" |
| 17:19.16 | Dr-Linux | SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 17:19.16 | Dr-Linux | SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 17:19.23 | ringhals | right |
| 17:19.41 | Dr-Linux | my SPAN 1 works fine, but SPAN 2 is still down |
| 17:20.11 | _Sam-- | ringhals : what iax client? |
| 17:20.19 | _Sam-- | and are you sure the clients are registered to the server? |
| 17:20.45 | tzafrir | _Sam--, apt-get update |
| 17:20.58 | _Sam-- | i would, but i dont want to disrupt my voip traffic! |
| 17:21.02 | _Sam-- | after hours i will |
| 17:21.17 | tzafrir | See also http://packages.debian.org/asterisk |
| 17:21.37 | [TK]D-Fender | Use the Source Luke! |
| 17:21.41 | _Sam-- | lol |
| 17:21.46 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 17:21.47 | dlynes_office | ringhals: you can do a registertimeout=n, where n is the number of seconds you want to reregister in |
| 17:22.01 | dlynes_office | ringhals: but your problem seems to be nat related |
| 17:22.09 | dlynes_office | ringhals: i would set qualify=300 on those phones |
| 17:22.13 | _Sam-- | ringhals : from the asterisk server CLI...if you do 'iax2 show peers' do you see your clients? |
| 17:22.19 | _Sam-- | with an ip? |
| 17:22.31 | dlynes_office | ringhals: that means that it'll try to keep a session open with the phone every 300ms |
| 17:23.54 | dlynes_office | ringhals: ah...it doesn't take long to get used to irc |
| 17:24.31 | ringhals | I hope not |
| 17:24.51 | dlynes_office | Dr-Linux: what's your pri show span 1 and 2 looking like now? |
| 17:25.07 | Dr-Linux | dlynes_office: welcome :) |
| 17:25.12 | dlynes_office | huh? |
| 17:25.22 | Dr-Linux | dlynes_office: my Span 1 is up,active and running fine |
| 17:25.35 | dlynes_office | Dr-Linux: so it was a telco problem then, right? |
| 17:25.43 | Dr-Linux | dlynes_office: i can dialout and dialin, |
| 17:25.50 | Dr-Linux | but SPAN 2 is still down |
| 17:25.55 | dlynes_office | Dr-Linux: i.e. they just hadn't activated the pri yet? |
| 17:26.03 | Dr-Linux | dlynes_office: yes, correct |
| 17:26.12 | dlynes_office | Dr-Linux: is span 2 also a pri from the telco? |
| 17:26.21 | Dr-Linux | dlynes_office: but looks like SPAN 2 is down, it's my configuration problem |
| 17:26.31 | Dr-Linux | dlynes_office: yes, |
| 17:26.48 | dlynes_office | Dr-Linux: do you have dids assigned on the pri on span 2 as well? |
| 17:26.56 | Dr-Linux | dlynes_office: bcoz i can't see any activity when i stop/start :S |
| 17:27.13 | _Sam-- | ~seen zoa |
| 17:27.28 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 6d 2h 20m 18s ago, saying: 'roy, any updates for your one way jitter buffer ?'. |
| 17:27.54 | Dr-Linux | dlynes_office: i got 23 DID's |
| 17:27.55 | dlynes_office | ~seen flauto |
| 17:27.57 | jbot | flauto <n=zhao@adsl-75-5-128-101.dsl.chcgil.sbcglobal.net> was last seen on IRC in channel #asterisk, 15h 47m 35s ago, saying: 'good, thanks, yourself?'. |
| 17:28.05 | dlynes_office | ah...cool |
| 17:28.08 | Dr-Linux | but not sure if they are for PRI 1 or PRI 2 |
| 17:28.12 | dlynes_office | Guess flauto's back from Beijing :) |
| 17:28.27 | Dr-Linux | dlynes_office: but i configured them and i can get calls in my asterisk |
| 17:28.50 | Dr-Linux | dlynes_office: can you help me to make sure if my span 2 is configured fine in zaptel.conf? |
| 17:28.56 | _Sam-- | Dr-Linux : on the PRI card....are the lights green or what color |
| 17:29.07 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 17:29.11 | _Sam-- | there are lights (usually?)...look at the color |
| 17:29.28 | dlynes_office | Dr-Linux: pastebin your zaptel.conf, zapata.conf, pri show span 1, pri show span 2, zap show status |
| 17:29.56 | Dr-Linux | _Sam--: i asked datacenter suppor that LED's question, they will tell me later, but they said, it's my end problem or cable problem. |
| 17:30.02 | dlynes_office | Dr-Linux: also pastebin your dmesg output when you fire up the driver |
| 17:30.07 | Dr-Linux | dlynes_office: okey hold on |
| 17:30.19 | mixi | which one would you recommend: chan_capi or bristuff (chan_zap)? |
| 17:30.36 | _Sam-- | hey if you cant even look at the lights on where the PRI plugs into your asterisk server...then you get no help! |
| 17:30.39 | dlynes_office | mixi: both are being used; I think most people on here use bristuff though |
| 17:31.06 | dlynes_office | _Sam--: It might be on his server in California, to which he has no physical access |
| 17:31.14 | mixi | has one any advantage in comparison? |
| 17:31.22 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 17:31.38 | dlynes_office | _Sam--: he's got a call center in California, and another in Lahore; he's at the one in Lahore |
| 17:31.40 | _Sam-- | the way he said it made me think he was having the other side (the telco side) tell him the color of his lights |
| 17:31.49 | mixi | bristuff worked brilliant so far for me, but as i am updating the whole asterisk, i would be able to switch |
| 17:32.10 | mixi | i guess i simply stay |
| 17:32.20 | dlynes_office | _Sam--: his English isn't too good, either |
| 17:32.42 | dlynes_office | _Sam--: kinda have to read between the lines |
| 17:33.00 | mixi | you mean me? |
| 17:33.01 | dlynes_office | _Sam--: but i think in this case, it's at the lahore location where he's having problems |
| 17:33.13 | dlynes_office | mixi: nope |
| 17:33.18 | dlynes_office | mixi: are you in lahore? |
| 17:33.18 | mixi | oh, okay, |
| 17:33.19 | _Sam-- | dlynes_office : me and DrL go way back! |
| 17:33.28 | mixi | where is lahore? |
| 17:33.33 | dlynes_office | mixi: Pakistan |
| 17:33.38 | dlynes_office | mixi: Punjab province |
| 17:34.26 | _Sam-- | honestly, im amazed that even a single pri is working :) |
| 17:34.34 | dlynes_office | _Sam--: heh |
| 17:34.45 | dlynes_office | _Sam--: Well, I helped him out with it yesterday |
| 17:34.50 | dlynes_office | _Sam--: it was a telco problem |
| 17:34.51 | _Sam-- | what PRI card? |
| 17:35.05 | dlynes_office | He's got a couple of dual port digium cards |
| 17:35.18 | dlynes_office | The second card is for a backup server |
| 17:35.27 | Dr-Linux | dlynes_office: go here >> http://pastebin.ca/67821 |
| 17:35.48 | _Sam-- | nice, im sure you will get it sorted...sounds like you have a good handle on it! |
| 17:35.50 | Dr-Linux | dlynes_office: i can't see any activity for span 2 in "dmesg" |
| 17:36.17 | dlynes_office | Dr-Linux: when you do a dmesg, you should see something about the two spans getting configured |
| 17:36.27 | dlynes_office | Dr-Linux: also, is this an e1, or a t1 pri? |
| 17:36.44 | _Sam-- | how come there is no span 2 in zaptel.conf? just span 3? (i am by no means a pri expert) |
| 17:36.49 | mixi | never been to pakistan |
| 17:36.56 | Dr-Linux | dlynes_office: but i see only 1 span 1 |
| 17:37.01 | Dr-Linux | dlynes_office: it's PRI T1 |
| 17:37.06 | dlynes_office | Dr-Linux: yeah...._sam-'s already pointed out your problem |
| 17:37.25 | dlynes_office | Dr-Linux: your channels are configured correctly, but your spans are not |
| 17:37.41 | dlynes_office | Dr-Linux: iow, you've got a yellow light on the back of your pri card |
| 17:37.46 | Dr-Linux | dlynes_office: how they suppose to be? |
| 17:38.13 | dlynes_office | change span=3,1,0,esf,b8zs to span=2,1,0,esf,b8zs |
| 17:38.18 | Dr-Linux | dlynes_office: yes, looks like something wrong with my zaptel.conf |
| 17:38.39 | dlynes_office | Dr-Linux: and why are you defining group 1 and group 3? |
| 17:38.43 | *** join/#asterisk drray (n=drray@c-67-183-123-24.hsd1.wa.comcast.net) |
| 17:38.48 | dlynes_office | Dr-Linux: why not keep it simple with group 1 and group 2? |
| 17:38.50 | Dr-Linux | dlynes_office: no, i'm not using 2nd port of my first card |
| 17:38.59 | dlynes_office | only port 1 and 3? |
| 17:39.06 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 17:39.06 | Dr-Linux | dlynes_office: yes |
| 17:39.08 | dlynes_office | i thought you were using a couple of 2 port pri cards? |
| 17:39.23 | *** join/#asterisk babyju___ (n=babyju@ool-4352274f.dyn.optonline.net) |
| 17:39.37 | Dr-Linux | dlynes_office: i'm using two TE210P card, total 4 port |
| 17:39.49 | Dr-Linux | but PRI is plugged in 1 and 3 |
| 17:40.03 | dlynes_office | Dr-Linux: ok, in zaptel.conf change bchan=25-47 to bchan=49-71 and dchan=48 to dchan=72 |
| 17:40.22 | Dr-Linux | dlynes_office: okey wait |
| 17:40.31 | dlynes_office | Dr-Linux: then in zapata.conf change chan => 25-47 to chan => 49-71 |
| 17:41.00 | dlynes_office | Dr-Linux: and in zaptel.conf, make sure you have span=1,1,0,esf,b8zs and span=3,1,0,esf,b8zs |
| 17:41.06 | *** join/#asterisk Eggplant (i=No@dsl-72-19-44-253.cascadeaccess.com) |
| 17:42.00 | *** join/#asterisk s0lid (n=jlq@210.213.242.39) |
| 17:42.07 | dlynes_office | After you've got that all squared away, shutdown asterisk |
| 17:42.18 | Dr-Linux | dlynes_office: okey |
| 17:42.19 | dlynes_office | then unload your zaptel drivers |
| 17:42.23 | dlynes_office | then reload your zaptel drivers |
| 17:42.32 | dlynes_office | and then do a dmesg, to make sure everything provisioned fine |
| 17:42.42 | Dr-Linux | Asterisk cleanly ending (0). |
| 17:42.42 | Dr-Linux | [root@ivr1 ~]# |
| 17:42.53 | Dr-Linux | ok |
| 17:43.35 | dlynes_office | reloaded the drivers yet? |
| 17:43.39 | Dr-Linux | [root@ivr1 ~]# rmmod zaptel |
| 17:43.40 | Dr-Linux | ERROR: Module zaptel is in use by wct4xxp |
| 17:43.40 | Dr-Linux | [root@ivr1 ~]# |
| 17:44.01 | Qwell[] | rmmod wct4xxp first |
| 17:44.11 | dlynes_office | modprobe -r wct4xxp ; modprobe wct4xxp |
| 17:44.15 | Dr-Linux | hhm.. ok |
| 17:44.28 | Joshaidan | Which * supports h.264? |
| 17:44.45 | dlynes_office | Joshaidan: all of them? |
| 17:44.55 | dlynes_office | Joshaidan: but afaik, only in passthrough mode |
| 17:45.02 | Joshaidan | hmm... k |
| 17:45.07 | dlynes_office | Joshaidan: it doesn't support it internally |
| 17:45.11 | Joshaidan | I was having issues getting a GVX-3000 to work |
| 17:45.20 | Joshaidan | audio is fine, just doesn't start the stream |
| 17:45.29 | Joshaidan | video stream |
| 17:45.50 | Qwell[] | Joshaidan: got the right codecs allowed? |
| 17:45.56 | _Sam-- | did you turn on support for video in sip.conf? |
| 17:46.03 | Qwell[] | and that.. |
| 17:46.03 | _Sam-- | ;videosupport=yes ; Turn on support for SIP video |
| 17:46.44 | tzafrir | modprobe -r only works for kernel 2.4 |
| 17:46.51 | tzafrir | checkout genzaptelconf -u |
| 17:46.58 | Qwell[] | tzafrir: it works for 2.6.. |
| 17:46.58 | dlynes_office | works fine for me in 2.6.15.5 |
| 17:47.01 | Dr-Linux | dlynes_office: dmesg , still doesn't show anything about SPAN 2 :S |
| 17:47.03 | *** part/#asterisk sevard (n=sev@24-179-179-47.dhcp.dlth.mn.charter.com) |
| 17:47.12 | Qwell[] | assuming you have module unloading support :p |
| 17:47.14 | Joshaidan | yup, videosupport is enabled |
| 17:47.15 | dlynes_office | Dr-Linux: check the back of the computer |
| 17:47.24 | Dr-Linux | dlynes_office: i did all as you said |
| 17:47.25 | dlynes_office | Dr-Linux: do you have any yellow lights there? |
| 17:47.47 | vader-- | hmmm do you guys know if asterisk has a sound file that says "Please record your message", "Press 1 to continue or 2 to change your message"? |
| 17:47.53 | Dr-Linux | dlynes_office: sorry, my server is in US datacenter :( |
| 17:48.00 | dlynes_office | Dr-Linux: ah...it's that one |
| 17:48.10 | dlynes_office | Dr-Linux: ok, pastebin your zaptel.conf and zapata.conf again |
| 17:48.22 | Dr-Linux | dlynes_office: but it should show something about SPAN 2 but it doesn't |
| 17:48.32 | Dr-Linux | but when i do ztcfg -vv then it does |
| 17:48.34 | Dr-Linux | SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 17:48.34 | Dr-Linux | SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 17:49.24 | Dr-Linux | dlynes_office: i started asterisk again and: |
| 17:49.26 | Dr-Linux | Asterisk Ready. |
| 17:49.26 | Dr-Linux | *CLI> -- B-channel 0/1 successfully restarted on span 1 |
| 17:49.26 | Dr-Linux | <PROTECTED> |
| 17:49.30 | Dr-Linux | ... so on |
| 17:49.38 | Dr-Linux | but it doesnt' show anything about SPAN 2 |
| 17:49.39 | dlynes_office | there ya go |
| 17:49.44 | dlynes_office | zap show status |
| 17:49.47 | Dr-Linux | ok |
| 17:49.49 | dlynes_office | what does that tell you? |
| 17:50.12 | dlynes_office | You should see span 1 and span 3, not span 1 and span 2 |
| 17:50.16 | vader-- | is putting on autofallthrough a good idea or bad? |
| 17:50.23 | dlynes_office | vader--: bad |
| 17:50.24 | Dr-Linux | aww |
| 17:50.36 | Dr-Linux | dlynes_office: span 3 is red now :( |
| 17:50.41 | dlynes_office | Dr-Linux: pastebin your zapata.conf and zaptel.conf again |
| 17:50.52 | Dr-Linux | dlynes_office: check your pvt the output |
| 17:50.58 | Dr-Linux | okey hold on |
| 17:51.33 | vader-- | hmmm |
| 17:51.48 | vader-- | well it seems i need autofallthrough set to no to create a menu type system |
| 17:51.54 | dlynes_office | vader--: the reason it's bad, is because it can mask issues with your dialplan |
| 17:52.01 | dlynes_office | vader--: and some features won't work with it enabled |
| 17:52.11 | vader-- | if not asterisk doesn't wait for a user to enter an extension |
| 17:52.20 | Dr-Linux | dlynes_office: here you go >> http://pastebin.ca/67836 |
| 17:52.32 | dlynes_office | vader--: You just found one feature that won't work with it enabled |
| 17:52.50 | dlynes_office | vader--: You're probably using Set(TIMEOUT(response)=10) or something right? |
| 17:52.59 | vader-- | hehe well i wanna create a menu that callers get when they dial a certain extensino |
| 17:53.00 | vader-- | ya |
| 17:53.03 | vader-- | http://pastebin.ca/67732 |
| 17:53.17 | dlynes_office | vader--: yeah...background i think will still work |
| 17:53.24 | dlynes_office | vader--: but not response timeout |
| 17:53.43 | vader-- | so autofallthrough should be set to yes |
| 17:53.54 | dlynes_office | vader--: should be set to no, not yes |
| 17:53.56 | vader-- | and i should use background for menus |
| 17:54.13 | dlynes_office | vader--: you'll probably still have a need for response timeout |
| 17:54.25 | dlynes_office | vader--: especially if you want to wait longer than the background lasts |
| 17:54.56 | vader-- | ok |
| 17:55.03 | [TK]D-Fender | vader-- : I told you want to do... if it hangs up immediately then you need to set "autofallthrough=no" in [general] |
| 17:55.17 | vader-- | ya |
| 17:55.30 | vader-- | i was just asking some other questions |
| 17:55.36 | vader-- | :) |
| 17:55.43 | vader-- | my next issue is with festival |
| 17:55.52 | vader-- | damn thing gets all jibberishy |
| 17:55.57 | [TK]D-Fender | vader-- : Typically you want response times at 10, digiat at 3. |
| 17:56.10 | vader-- | works once fine and then the next time it's all jibberish |
| 17:56.14 | [TK]D-Fender | vader-- : How fortunate, I'm fluent in gibberish! |
| 17:56.24 | Dr-Linux | dlynes_office: are the zap configs fine now? |
| 17:56.25 | dlynes_office | [TK]D-Fender: this is #asterisk, where people ask for suggestions and ignore htem |
| 17:56.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.150) |
| 17:56.44 | vader-- | hehe |
| 17:59.49 | florz | Hmmm. Does anyone know what the values output by "show translation" in the asterisk CLI are supposed to mean? |
| 18:01.44 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.150) |
| 18:01.48 | [TK]D-Fender | florz : It says hom much CPU power is reuired to translate from one codec to another. |
| 18:01.57 | *** join/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net) |
| 18:02.13 | Dr-Linux | dlynes_office: looks like you are gone :S |
| 18:02.17 | florz | [TK]D-Fender: Well, so far I did understand it =:-) |
| 18:02.31 | florz | [TK]D-Fender: So, it's in watts? =:-) |
| 18:02.36 | dlynes_office | Dr-Linux: it looks fine to me |
| 18:02.49 | dlynes_office | Dr-Linux: assuming that's what it's supposed to look like when you've got two cards |
| 18:03.17 | Dr-Linux | dlynes_office: but there is one thing, in my privious setting, the span 2 was "OK" in the "zap show status" |
| 18:03.18 | dlynes_office | Dr-Linux: can you reboot the box? |
| 18:03.31 | Dr-Linux | dlynes_office: but now Span 2 is RED |
| 18:03.54 | Dr-Linux | dlynes_office: should i reboot? |
| 18:04.07 | Dr-Linux | dlynes_office: if it helps, then sure i can do reboot |
| 18:05.03 | Dr-Linux | dlynes_office: should i? |
| 18:05.07 | vader-- | is there a way to tell the person who is trying to make a call that all available lines in the group are busy and to try again later? |
| 18:05.22 | vader-- | i think right now it just gives them a busy signal |
| 18:05.58 | [TK]D-Fender | vader-- : "show application playback" |
| 18:06.11 | dlynes_office | Dr-Linux: well, let's try one thing first |
| 18:06.19 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 18:06.20 | dlynes_office | Dr-Linux: rearrange everything for span2, instead |
| 18:06.37 | Dr-Linux | dlynes_office: how to rearrange? i didn't get ya |
| 18:06.38 | dlynes_office | Dr-Linux: maybe with two cards, span 1 is first span on first card, span 2 is first span on 2nd card, ... |
| 18:06.59 | [TK]D-Fender | dlynes_home : News to me.... |
| 18:07.05 | Dr-Linux | dlynes_office: yes |
| 18:07.14 | dlynes_office | [TK]D-Fender: ah..so I did have it right for him, then? |
| 18:07.28 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 18:07.28 | dlynes_office | [TK]D-Fender: i've never set up a machine with 2 dual port pri cards in it before |
| 18:07.40 | dlynes_office | [TK]D-Fender: seems kinda pointless to use two dual ports instead of a quad port |
| 18:08.06 | CunningPike | Dr-Linux: Is the second PRI for telco failover? |
| 18:08.12 | *** join/#asterisk hohum (n=dcorbe@12.195.58.236) |
| 18:08.17 | Dr-Linux | CunningPike: yes |
| 18:08.35 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 18:08.55 | CunningPike | Dr-Linux: Well, maybe they would have it turned down then - it'll only get turned up when calls failover to it? |
| 18:09.06 | CunningPike | Dr-Linux: Did you actually speak to your telco yet? |
| 18:09.50 | Dr-Linux | CunningPike: yes, they said, both PRI's are turned on, and they said there is some problem at my end. |
| 18:10.22 | Dr-Linux | CunningPike: i THINK there is some problem with my configuration i.e. zaptel.conf |
| 18:10.37 | Dr-Linux | bcoz it's not showing any acitivty while doing "dmesg" |
| 18:10.48 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 18:11.23 | Dr-Linux | CunningPike: i'm still not sure, what's span 1 and what's spam 3 |
| 18:11.46 | *** join/#asterisk albanach2000 (n=albanach@va-69-68-190-44.dyn.sprint-hsd.net) |
| 18:12.22 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 18:13.06 | Dr-Linux | dlynes_office: should i revert back all the setting? |
| 18:13.55 | albanach2000 | I just installed asterisk for the first time in over a year. When I divert to voicemail the 'beep' doesn't go away and the recorded message is very quiet. Any clue as to what might cause that? |
| 18:14.18 | CunningPike | Dr-Linux: Is the second card even recognized? I have no idea how to configure two cards in a server.... |
| 18:15.04 | Dr-Linux | CunningPike: both cards are recognized and installed |
| 18:16.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:18.30 | CunningPike | OK - I can see that from your 'zap show status' - it wouldn't show RED unless that span existed in your config |
| 18:19.01 | CunningPike | Dr-Linux: Did you pastebin your zaptel and zapata recently? |
| 18:19.14 | Dr-Linux | yes |
| 18:19.16 | Dr-Linux | i did |
| 18:19.32 | Dr-Linux | what if i change this: |
| 18:19.34 | Dr-Linux | span=3,1,0,esf,b8zs |
| 18:19.34 | Dr-Linux | bchan=49-71 |
| 18:19.34 | Dr-Linux | dchan=72 |
| 18:19.42 | Dr-Linux | to: |
| 18:19.56 | Dr-Linux | span=2,1,0,esf,b8zs |
| 18:20.05 | Dr-Linux | bchan=49-71 |
| 18:20.11 | Dr-Linux | dchan=72 |
| 18:20.55 | Dr-Linux | dlynes_office: thoughts? |
| 18:21.33 | CunningPike | Dr-Linux: You can try it, but it will activate Span 2 on your first card - at least it should |
| 18:22.05 | CunningPike | Dr-Linux: What does ztcfg -vvvv say? |
| 18:22.26 | Dr-Linux | that i pasted above |
| 18:22.56 | Dr-Linux | <Dr-Linux> SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 18:22.57 | Dr-Linux | <Dr-Linux> SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) |
| 18:23.00 | nortex | Ahrimanes, Can I ask you a few more questions about the TRANSFER_Context? |
| 18:23.21 | Dr-Linux | CunningPike: look it still says: span 2 |
| 18:23.26 | Dr-Linux | where is span 3 ? |
| 18:24.47 | CunningPike | Dr-Linux: I think ztcfg numbers its spans sequentially as they are configured....... - your 'pri show' is correct. Have you double-checked the cable for the second span? |
| 18:25.29 | rene- | if i reload chan_agent, do they lose their place in queue? their registration? |
| 18:25.36 | Dr-Linux | CunningPike: i just change span=3,1,0,esf,b8zs to span=2,1,0,esf,b8zs |
| 18:25.41 | Dr-Linux | lets see what happends |
| 18:26.12 | *** join/#asterisk Iamtheman222 (n=fddfffsa@modemcable237.246-83-70.mc.videotron.ca) |
| 18:26.16 | Iamtheman222 | Hi |
| 18:26.32 | Iamtheman222 | I have a question regarding Mediatrix 2102 config for working with Asterisk |
| 18:27.37 | *** join/#asterisk speedwagon (n=FreePBX7@70.46.87.158) |
| 18:27.42 | speedwagon | hello everyone |
| 18:28.59 | CunningPike | speedwagon: Are your initials REO? |
| 18:29.25 | Iamtheman222 | Does someone knows the 2102 ? |
| 18:30.06 | nortex | [TK]D-Fender, Can I ask you some questions about the transfer_context? |
| 18:30.20 | [TK]D-Fender | nortex : Never used.... |
| 18:30.41 | nortex | [TK]D-Fender, ok ... |
| 18:32.19 | speedwagon | CunningPike, funny. |
| 18:33.10 | CunningPike | I crack myself up |
| 18:33.26 | speedwagon | I have used this nick since my early days of fido... |
| 18:33.33 | *** join/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com) |
| 18:33.56 | harpermood | so i support impossibly backlogged.. I have been on hold for 25 minutes |
| 18:34.24 | speedwagon | are they at least playing good music? |
| 18:34.29 | harpermood | rocking |
| 18:34.30 | file | support where? |
| 18:34.33 | harpermood | Digium |
| 18:34.53 | file | probably |
| 18:34.57 | harpermood | I have been in the queue for 25 minutes, even though I have my ticket number |
| 18:35.13 | speedwagon | queue support hell. |
| 18:35.46 | speedwagon | you did call them via there exten => 500 guest call. |
| 18:36.05 | harpermood | I called the number on the website... |
| 18:36.09 | harpermood | my phone system isn't setup. |
| 18:36.17 | harpermood | which is why I am calling ;) |
| 18:36.30 | file | what ticket #? |
| 18:36.30 | speedwagon | OK so what is your issue? |
| 18:36.39 | *** join/#asterisk radhios (n=radhios@bue215-194.is.net.ar) |
| 18:36.47 | harpermood | can dial out (sort of) not in. |
| 18:36.53 | harpermood | been with support all day yesterday |
| 18:36.59 | harpermood | digium and telco |
| 18:37.08 | file | exciting |
| 18:37.10 | harpermood | I am supposed to get on a conf call with both of those at Noon Central |
| 18:37.16 | harpermood | to no avail. |
| 18:38.03 | *** join/#asterisk clive- (n=pirch@dsl-145-43-208.telkomadsl.co.za) |
| 18:38.06 | speedwagon | t1/e1 board setup? |
| 18:38.19 | harpermood | T1 board ... TE110P |
| 18:38.31 | harpermood | doing E&M with wink |
| 18:38.37 | harpermood | 6 channels on T1 assigned. |
| 18:38.49 | harpermood | they send the wink... my card doesn't respond |
| 18:38.52 | radhios | Hi all, How can i route an incoming call from zap interface(pstn) to an extension in other box connected using iax2??? |
| 18:39.08 | speedwagon | dial rules |
| 18:39.32 | nortex | radhios, With the dial command |
| 18:41.15 | speedwagon | radhios, are all the calls from this one line needing to be sent to other box? |
| 18:41.48 | radhios | the escenario is as follow, i have a box in location A with only zap interfaces, without any extension there and in a site B |
| 18:42.16 | radhios | i havo other box with all sip extensions. |
| 18:42.43 | Dr-Linux | what's difference between PRI cable's color and network cables color? |
| 18:43.23 | MikeJ[Laptop] | Dr-Linux? huh |
| 18:43.25 | speedwagon | pri cable has colors? Network cables have definded colors???? hum new info for me. |
| 18:43.41 | file | PRI cables are bright pink |
| 18:43.42 | nortex | radhios, Same setup here. All zap interfaces are routed with a dial command Dial(IAX2/[iaxpeername]) |
| 18:43.51 | MikeJ[Laptop] | I just use regular eth cables for pri usually |
| 18:44.06 | MikeJ[Laptop] | crossovers are pinned differently than eth cross |
| 18:45.24 | clive- | file you should try those illuminated cables that glow in the dark |
| 18:45.56 | speedwagon | yes lets makes some neon color cables... |
| 18:46.39 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 18:46.42 | *** join/#asterisk rainkid (n=rainkid@gemini.os5.com) |
| 18:47.23 | rainkid | i'm running asterisk for a home, just a few phones. im not familiar with dundi and mgcp.. but it appears as i do not need these features. can i disable them? |
| 18:47.47 | speedwagon | yes |
| 18:47.47 | clive- | :) |
| 18:48.43 | rainkid | correct me if im wrong, but mgcp is another channel similiar to sip and iax, but used mostly by cisco phones, correct? |
| 18:49.01 | rainkid | dundi is used for distributed management of asterisk? |
| 18:49.15 | rainkid | the question is, how do i disable them? |
| 18:49.24 | *** part/#asterisk harpermood (n=harpermo@24-180-0-138.static.snlo.ca.charter.com) |
| 18:49.54 | malverian | Anyone using asterisk-1.2.9.1 and doesn't mind reproducing a bug I'm experimenting? |
| 18:50.06 | *** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com) |
| 18:50.07 | radhios | speedwagon: Yes i need rote incoming call from site A to an estension on site B |
| 18:50.19 | speedwagon | mgcp is not for Cisco's phones, it can be used for some of them that have mgcp. The dundi is like asterisk own enum setup. |
| 18:50.22 | malverian | I am experiencing a regression with 1.2.9.1 (known to not be there in 1.2.7.1) where if you do something like "asterisk -r -x 'sip show peers'" it only shows the first line. |
| 18:50.43 | malverian | Can anyone confirm this bug before I submit a report? |
| 18:50.53 | Dr-Linux | dlynes_office: i change to span=2,esf,... now i can see things in "dmesg" |
| 18:52.10 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
| 18:52.21 | speedwagon | radhios, send the call to a context like [inbound] then there so exten => s,1,Dial(IAX2/yourotherbox/${exten}) But you should do some reading and learn how to setup your accounts. |
| 18:52.44 | TESTER2 | I get this error with the last fxotune.c (HEAD): |
| 18:52.45 | TESTER2 | Tuning module /dev/zap/4 |
| 18:52.45 | TESTER2 | Could not fill input buffer - got -1 bytes, expected 4000 bytes |
| 18:52.45 | TESTER2 | Failure! |
| 18:53.22 | malverian | No one? |
| 18:53.30 | Dr-Linux | CunningPike: good news! problem solved ;) |
| 18:54.26 | CunningPike | Dr-Linux: Great! What was the fix - span=2? |
| 18:54.39 | *** join/#asterisk xunil (n=wkurdzio@office1.visionpointsystems.com) |
| 18:54.40 | Dr-Linux | CunningPike: yes ;) |
| 18:54.46 | Dr-Linux | CunningPike: odd? |
| 18:55.09 | xunil | do i need festival.conf configured before the "say digits" AGI command will work? |
| 18:55.10 | CunningPike | Dr-Linux: Great - it is a little odd, but what the hey - it works :D |
| 18:55.16 | Dr-Linux | CunningPike: even in my IM everyone told me that, it doesn't make sense don't do that, but i did |
| 18:55.34 | Dr-Linux | lol look here: /msg |
| 18:55.48 | CunningPike | What does 'pri show spans' say now? |
| 18:56.13 | Dr-Linux | pri show status |
| 18:56.18 | Dr-Linux | i pasted in your pvt |
| 18:56.34 | CunningPike | Dr-Linux: Whatever. Oh, I see - you _added_ a fake span 2 |
| 18:56.40 | CunningPike | Cool |
| 18:56.50 | Dr-Linux | *CLI> pri show span 3 |
| 18:56.51 | Dr-Linux | Primary D-channel: 72 |
| 18:56.51 | Dr-Linux | Status: Provisioned, Up, Active |
| 18:56.57 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net) |
| 18:57.06 | Dr-Linux | pri show span 2 , is not runnig :P |
| 18:58.35 | *** join/#asterisk digime (n=digime@user-0cdf0g7.cable.mindspring.com) |
| 18:58.46 | digime | is there a polycom expert in here? |
| 18:58.57 | digime | i have an ip400 that is bricking using ip500 software. |
| 18:58.59 | Dr-Linux | CunningPike: how can i verify if span 2 is working fine? |
| 18:59.26 | CunningPike | Dr-Linux: Span 2??? |
| 18:59.33 | radhios | speedwagon: thanx!!! Currently I using freePBX 2.x, is more easy setup this routing with freePBX??? |
| 18:59.45 | Dr-Linux | CunningPike: lol span 3 |
| 19:00.01 | Dr-Linux | CunningPike: it status is just fine. |
| 19:00.23 | CunningPike | Dr-Linux: Create an extension in your dialplan that has Dial(Zap/g3/whatever) |
| 19:00.28 | speedwagon | radhios, it's about the same with creating the trunks and inbound rules. But you should ask on #freepbx |
| 19:00.45 | Dr-Linux | ok, lemme try |
| 19:01.06 | radhios | speedwagon: thnx a lot for your help!! |
| 19:01.30 | CunningPike | digime: What are the symptoms? |
| 19:01.42 | digime | cunningpike: freezing up after bootup |
| 19:01.52 | digime | and a 5 tone error sound |
| 19:02.02 | digime | i am having it login and pull a 500/501 config |
| 19:02.21 | digime | i am trying to establish if there is some compatibility between 400 and 500 |
| 19:02.25 | CunningPike | digime: Does it load the bootrom and SIP OK? |
| 19:02.29 | digime | yes |
| 19:02.37 | CunningPike | digime: What version? |
| 19:02.39 | malverian | Any Asterisk-1.2.9.1 users in here? |
| 19:02.46 | digime | it loads then freezes |
| 19:02.48 | CunningPike | malverian: About 500 |
| 19:02.57 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 19:02.58 | malverian | CunningPike, are you using asterisk 1.2.9.1? |
| 19:03.02 | xunil | do i need festival.conf configured before the "say digits" AGI command will work? |
| 19:03.05 | CunningPike | malverian: In test |
| 19:03.12 | malverian | CunningPike, Do you have access to it right now? |
| 19:03.23 | CunningPike | malverian: I sure hope so........ |
| 19:03.35 | malverian | CunningPike, Do you mind running: asterisk -r -x "show applications" |
| 19:03.35 | digime | BootROM |
| 19:03.35 | digime | 2.6.1 |
| 19:03.44 | malverian | And tell me if it shows more than just the first line of the output for you? |
| 19:03.48 | CunningPike | digime: SIP? |
| 19:04.03 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 19:04.07 | asterboy | df -v |
| 19:04.08 | asterboy | w |
| 19:04.10 | asterboy | clear |
| 19:04.12 | malverian | CunningPike, Any CLI command will work fo course.. but you need to use "-r -x ..." |
| 19:04.15 | asterboy | oops |
| 19:04.34 | CunningPike | malverian: OK - stand by |
| 19:04.35 | digime | sip: 1.6.5.0043 |
| 19:05.17 | CunningPike | malverian: OK - and? |
| 19:05.29 | malverian | # asterisk -r -x "show applications" | wc -l |
| 19:05.29 | malverian | 1 |
| 19:05.36 | malverian | It should be more like 50 |
| 19:05.37 | CunningPike | digime: I think you need 1.6.6 and the associated sip.cfg for the IP403 |
| 19:05.48 | digime | wow, great info. couldnt find that anywhere |
| 19:05.51 | malverian | CunningPike, I"m just making sure it's not specific to my configuration. |
| 19:05.59 | malverian | CunningPike, Are you also getting only 1 line of output? |
| 19:06.12 | digime | cunningpike: what is a 403? |
| 19:06.13 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 19:06.17 | malverian | CunningPike, Notice that if you perform the same command on a <1.2.9.1 it prints out all of the lines. |
| 19:06.25 | CunningPike | malverian: No - the whole enchilada |
| 19:06.30 | malverian | CunningPike, Really... hm. |
| 19:06.30 | digime | cunningpike: i have never heard of that model. the software is showing an ip400 |
| 19:06.38 | malverian | Peculiar... |
| 19:06.55 | CunningPike | digime: What does it look like - what label is on the back? |
| 19:07.08 | malverian | CunningPike, When I connect manually with 'asterisk -r' and execute the command I get all lines.. but if I use -x, it only prints the first line of any command. |
| 19:07.09 | digime | cunningpike: I have not found any ip40x configuration file. i have looked for weeks. |
| 19:07.24 | smackus | can anyone help me figure out why I cannot install zaptel on my redhat enterprise linux box... I am getting alot of incompatible pointer type errors. here is some of my last output when typing make linux26: http://pastebin.ca/67890 |
| 19:07.44 | digime | cunningpike: there is no "label" on the back. they dont brand these things. they make it confusing like cisco. it's just labeled as a polycom. that's it. |
| 19:08.04 | CunningPike | digime: Does it look like a IP300? |
| 19:08.12 | digime | cunningpike: it looks like an ip500 |
| 19:08.23 | digime | cunningpike: black, with the 3 line presences, etc. |
| 19:08.50 | digime | cunningpike: i had a few seconds before the phone locked up, i was able to pull up in the phone that its an ip400 |
| 19:09.03 | CunningPike | digime: Hmm - hang on a sec |
| 19:09.22 | digime | cunningpike: great, thanks. this is a high priority |
| 19:09.27 | file | digime: http://www.truedataonline.com/xq/asp/file.preBuilt/preBuiltSearchID.40156/qx/images/products/ip400.jpg |
| 19:09.28 | file | like that? |
| 19:09.31 | speedwagon | ? centosbug |
| 19:09.40 | speedwagon | ~centosbug |
| 19:09.41 | jbot | it has been said that centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 19:09.50 | digime | file! regards |
| 19:09.57 | file | hi hi |
| 19:10.11 | digime | YES that is the phone |
| 19:10.16 | digime | this thing is driving us all crazy |
| 19:10.28 | speedwagon | smackus, see the CentOS bug |
| 19:10.30 | digime | we need to get it working with asterisk asap but it's locking up. we are using ip500 software with it |
| 19:11.18 | *** part/#asterisk DMark_ (n=kk7cu@pool-70-105-217-25.scr.east.verizon.net) |
| 19:11.53 | smackus | speedwagon: no such file or directory |
| 19:12.01 | nortex | Does anyone have a working example of transfer_context? |
| 19:13.38 | CunningPike | digime: I have never seen that phone before - where are you based? |
| 19:13.46 | digime | i am currently in new york |
| 19:13.58 | CunningPike | digime: :S |
| 19:14.09 | malverian | eh... while (!ast_el_read_char(el, &tempchar)); |
| 19:15.09 | *** join/#asterisk userdefined (i=jr000430@shell1.phx.gblx.net) |
| 19:15.38 | CunningPike | digime: The only models I have come across are 301, 430, 501, 601 and 4000 |
| 19:15.59 | smackus | if i do a locate on rwlock, here is the output... http://pastebin.ca/67897 |
| 19:16.08 | digime | a simple google search displays a page on polycom's site about how wonderful the ip400 is |
| 19:16.16 | digime | that is the only information i can find |
| 19:16.33 | smackus | speedwagon: do I just need to manipulate the command a little? I do not understand what that is supposed to do. |
| 19:16.54 | *** join/#asterisk scorn (n=tim@eclipse.sigint.com) |
| 19:17.35 | scorn | I've got a Digium TE205P board and I'm wanting to know what impedance the output port is currenty set to (if it is even configurable) |
| 19:17.55 | CunningPike | digime: I can't even find that much - got a URL? |
| 19:18.49 | digime | http://www.polycom.com/company_info/1,1412,pw-9393-3024,00.html |
| 19:19.29 | smackus | speedwagon: you there? |
| 19:20.33 | CunningPike | digime: Weird - where'd you get it? |
| 19:20.53 | digime | google search polycom 400 ip |
| 19:21.00 | digime | anyhow.. any solutions/ideas? |
| 19:21.12 | CunningPike | digime: The phone, man, the phone! ;) |
| 19:21.18 | digime | sorry |
| 19:21.22 | [TK]D-Fender | ip 400 = old MODEL |
| 19:21.26 | digime | brain freeze. eGay I think |
| 19:21.56 | *** part/#asterisk SRCR (n=Peter@a80-127-87-78.adsl.xs4all.nl) |
| 19:22.58 | CunningPike | digime: The only sip.cfg file/SIP application I have seen that mentions the IP400 family is 1.6.6 - I would try and get that and see if it works |
| 19:23.33 | [TK]D-Fender | CunningPike : That'd be IP 4000, no? IP 400 is an ancient modfel. |
| 19:23.41 | digime | okay. "get that". that is the hard part for me. |
| 19:23.56 | CunningPike | [TK]D-Fender: Thanks - digime, pay attention to [TK]D-Fender |
| 19:24.26 | *** join/#asterisk pnlarsson (n=niklas@c83-248-7-150.bredband.comhem.se) |
| 19:24.40 | CunningPike | digime: And now I look, the sip.cfg is specific to the IP430 |
| 19:25.05 | digime | ok. but you keep saying ip430, do you mean 400? |
| 19:25.10 | *** join/#asterisk Elshar (n=Elshar@I.wants.your.cheekan.org) |
| 19:26.44 | CunningPike | digime: I don't think so - [TK]D-Fender has said that the IP400 is a very old model |
| 19:27.01 | CunningPike | digime: The 430 is a new model - looks like a 300/301 |
| 19:27.06 | xunil | do i need festival.conf configured before the "say digits" AGI command will work? |
| 19:29.13 | malverian | xunil, No. |
| 19:29.38 | *** join/#asterisk W4 (n=W4@212.58.232.254) |
| 19:29.39 | malverian | xunil, SayDigits uses the sounds in /usr/lib/asterisk/sounds/digits/ |
| 19:29.42 | florz | Well, just in case somebody didn't know yet: show translation shows the time transcoding of one second worth of audio takes (in milliseconds), regardless of the amount of data you let it process for measuring ... |
| 19:30.01 | MikeJ[Laptop] | florz, correct |
| 19:30.12 | digime | cunningpike: old model, yes, but there should still be the right firmware out there for it to get it working. someone else must be using this phone...? |
| 19:30.27 | florz | MikeJ[Laptop]: I know, the source says so ;-) |
| 19:30.31 | scorn | Anyone know where I can find a good hardware manual for the Digium TE205P? |
| 19:30.47 | florz | MikeJ[Laptop]: But the output isn't all that clear, after all ... |
| 19:31.04 | scorn | the Digium website doesn't appear to go into much depth as far as hardware goes. |
| 19:31.22 | smackus | still trying to figure out what the heck is the problem installing the zaptel to my new redhat server. |
| 19:31.25 | smackus | can anyone help me |
| 19:31.35 | smackus | i tried the centos bug, but I am not getting anywhere with that |
| 19:32.50 | [TK]D-Fender | IP 430 = really nice generl-purpose budget phone for PoE. Without need of PoE I'd suggest the 501 typically. |
| 19:34.28 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 19:35.05 | shmaltz | has there been a major change in the options for the Queue command from 1.0.x? |
| 19:36.24 | xunil | malverian: thanks |
| 19:37.03 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 19:37.17 | [TK]D-Fender | shmaltz : Yes, the dialplan parameter order has changed and your extensions.conf may need to be tweaked |
| 19:37.27 | *** join/#asterisk blackmousepad (n=blackmou@24-180-194-145.dhcp.aldl.mi.charter.com) |
| 19:37.53 | shmaltz | [TK]D-Fender, realy let me check |
| 19:38.59 | shmaltz | [TK]D-Fender I still got that ok |
| 19:39.12 | shmaltz | i only use nr and then a timeout value |
| 19:39.20 | shmaltz | which is good in my current dp |
| 19:39.48 | CunningPike | digime: You can get SIP 1.5.3 here: http://www.polycom.com/resource_center/0,,pw-6812-12612,00.html |
| 19:39.55 | CunningPike | digime: That might work |
| 19:40.05 | *** join/#asterisk momelod (n=momelod@HSE-Montreal-ppp133997.qc.sympatico.ca) |
| 19:40.08 | momelod | hello people |
| 19:40.11 | digime | cool! i will try |
| 19:40.43 | [TK]D-Fender | CunningPike : nOT ACCORDING TO THE CHANGELOG... |
| 19:40.50 | CunningPike | smackus: We have zaptel on RHEL - what's the problem |
| 19:41.03 | CunningPike | [TK]D-Fender: OK - THANKS FOR SHARING ;) |
| 19:41.37 | blackmousepad | I'm trying to setup disa so I can dial items like *61 from outside. I've got it sorta working right now, IF I call from my sip phone out a zap trunk to the other zap trunk hit 9 on the menu which connects you to my custom-mydisa app prompts for a password, accepts the password (or hangs up on fail). I then have dialtone and can dial any extension that doesn't start with a *. Why? Also if I call from a pots line to the zap |
| 19:41.38 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-80-10.red.bezeqint.net) |
| 19:41.52 | blackmousepad | the passcode and get dialtone but can't dial anything. |
| 19:42.09 | [TK]D-Fender | Shit, not even 1.4.1 lists the IP 400..... |
| 19:43.31 | smackus | CunningPike: hang on, let me send you a link to the output |
| 19:43.35 | AlexCTI | Hi, someone know how can I increase the priority for a Incoming call when it get into the queue? |
| 19:43.54 | smackus | http://pastebin.ca/67897 |
| 19:44.04 | digime | [TK]D-Fender: Im not even sure how to factory reset the phone |
| 19:44.30 | shmaltz | has anybody seen the follwoing problem in queues: |
| 19:44.31 | shmaltz | call comes in goes to queue sales like this: queue(sales,nr,,,18) |
| 19:44.33 | shmaltz | the call rings, to 3 members using Sip/sipresourece as memeber in queues.conf, member1 picks up noone there, and the call continues to ring to memeber2 and 3, member2 and 3 have the same problem if they try to pick up |
| 19:44.35 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 19:44.39 | CunningPike | smackus: Huh? |
| 19:45.00 | flujan | guys, are someone here using chanspy ? |
| 19:45.05 | Qb3rt | polycom... i need bootrom 3.1.2 or greater and on their website it says that you need to be reseller to download this bootrom... the only one i can get is the older version 3.1.0 |
| 19:45.29 | CunningPike | smackus: All I got was locate output.... |
| 19:46.40 | dlynes_office | damnit...forgot to install a kernel with scsi support :( |
| 19:47.06 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 19:47.16 | SexyKen | Hey guys -- how does one make asterisk mail voicemail instead of leave it on the phone |
| 19:47.16 | blackmousepad | http://pastebin.ca/67923 |
| 19:47.35 | dlynes_office | SexyKen: attach=yes |
| 19:47.56 | dlynes_office | SexyKen: take a look at your sample voicemail.conf file |
| 19:48.38 | SexyKen | OKay and if attach=yes is there why wouldn't it send it? |
| 19:49.43 | dlynes_office | SexyKen: several reasons: you haven't specified an email address to send it to, you don't have sendmail, or a reasonable facsimile installed on your machine, or you don't have sendmail or your reasonable facsimile configured correctly, or your dns server is whacked, or you don't have an outbound route (read default gateway) set up |
| 19:49.46 | nortex | SexyKen: There is an email address option |
| 19:49.58 | SexyKen | serveremail = is set |
| 19:50.18 | SexyKen | I have everything setup there... |
| 19:51.05 | dlynes_office | SexyKen: i would start looking for MTA, mail server, and/or dns problems then |
| 19:51.23 | dlynes_office | SexyKen: for instance, type 'which sendmail' at the linux command line |
| 19:51.34 | dlynes_office | SexyKen: if that doesn't return a path, that's your problem right there |
| 19:51.52 | SexyKen | I've got sendmail |
| 19:51.56 | SexyKen | It's at /usr/bin/sendmail |
| 19:51.56 | dlynes_office | SexyKen: otherwise, your problems are deeper rooted, and this won't be the channel to get help in |
| 19:52.09 | dlynes_office | SexyKen: yeah, so you need try asking on #sendmail then |
| 19:52.17 | smackus | CunningPike: crap... wrong link: |
| 19:52.26 | dlynes_office | SexyKen: your sendmail setup is probably fubar |
| 19:52.38 | Dr-Linux | dlynes_office: thanks for your help, my both PRI's are UP and Active |
| 19:52.46 | dlynes_office | SexyKen: if you check your /var/log/maillog, you might be able to figure out what your issue is |
| 19:52.56 | Dr-Linux | dlynes_office: as i said, the problem was in my configs |
| 19:52.56 | AlexCTI | Does anyone know how can put a call on top of the line of the queue? |
| 19:53.12 | dlynes_office | Dr-Linux: so what was the problem then? |
| 19:53.24 | smackus | CunningPike: http://pastebin.ca/67890 |
| 19:53.51 | Dr-Linux | dlynes_office: i just changed span=3 to span=2 in zaptel.conf |
| 19:53.55 | Dr-Linux | and worked |
| 19:54.10 | dlynes_office | Dr-Linux: ummm...I asked you to change that, and then [TK]D-Fender jumped all over me :p |
| 19:55.00 | dlynes_office | Dr-Linux: so span 1 on card 1 is span 1, and span 1 on card 2 is span 2, and span 2 on card 1 is span 3, and span 2 on card 2 is span 4? |
| 19:55.31 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:56.02 | Dr-Linux | dlynes_office: yes correct |
| 19:56.11 | flujan | someone could point me another resource to find information about chanspy? |
| 19:56.19 | dlynes_office | Dr-Linux: ok, now i know for the next time someone else has such a fubar setup then |
| 19:56.29 | dlynes_office | Dr-Linux: so my original suspicion was correct then |
| 19:56.29 | Dr-Linux | dlynes_office: when i was trying this, folks told me do not do that, it doesn't make any sense |
| 19:56.40 | Dr-Linux | but i did, and it worked for me |
| 19:56.40 | dlynes_office | Dr-Linux: yeah...that's exactly what [TK]D-Fender told me :) |
| 19:56.49 | dlynes_office | Dr-Linux: when i made the suggestion to you originally |
| 19:56.59 | Dr-Linux | dlynes_office: yeah |
| 19:57.13 | CunningPike | smackus: Hmmm - I gotta run to a meeting, but I've never seen that before. I see you've tried the spinlock patch? |
| 19:57.31 | smackus | well... it does not work like i expected |
| 19:57.32 | dlynes_office | flujan: have you tried voip-info? |
| 19:57.39 | smackus | i get a file not found error |
| 19:57.52 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 19:57.56 | flujan | dlynes_home, yes... the example there doesn't work for me |
| 19:58.03 | flujan | I want to spy a channel... |
| 19:58.05 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 19:58.05 | dlynes_office | flujan: ah |
| 19:58.08 | CunningPike | smackus: The patch? |
| 19:58.19 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 19:58.21 | Wazb^ | hi to all |
| 19:58.21 | flujan | even dialing the number I can spy it... |
| 19:58.22 | dlynes_office | CunningPike: maybe he hasn't downloaded the patch |
| 19:58.27 | smackus | yeah... if I did what I was told... which was... |
| 19:58.31 | dlynes_office | CunningPike: maybe he's trying to apply it without downloading it |
| 19:58.44 | smackus | sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h |
| 19:58.56 | CunningPike | dlynes_office: COuld be - any case, gotta run. Back in an hour |
| 19:58.56 | dlynes_office | oh...that |
| 19:58.58 | dlynes_office | ~redhatbug |
| 19:58.59 | jbot | rumour has it, redhatbug is is a problem with the latest RedHat Enterprise Linux and CentOS kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 19:59.03 | smackus | sed: can't read /usr/src/kernels/2.6.9-34.0.1.ELsmp-x86_64/include/linux/spinlock.h: No such file or directory |
| 19:59.17 | raidenz | Is it possible to do SQL commands in the dialplan in the latest Asterisk 1.4 SVN? |
| 19:59.25 | CunningPike | smackus: You're not in /usr/src - amend the paths as necessary |
| 19:59.36 | dlynes_office | smackus: you don't have linux-2.6.9-34.0.1.ELsmp-devel.rpm installed or whatever |
| 19:59.49 | smackus | hmmm |
| 20:00.00 | dlynes_office | smackus: you have the regular kernel-dev installed, not the smp kernel-dev |
| 20:00.01 | smackus | ok |
| 20:00.10 | smackus | gotcha... makes sense. |
| 20:00.26 | CunningPike | smackus: There ya go - listen to dlynes_office - he sometimes knows what he's talking about ;) |
| 20:00.50 | smackus | hehe |
| 20:01.00 | smackus | he has never led me astray |
| 20:01.03 | smackus | thanks all |
| 20:01.06 | dlynes_office | lol |
| 20:01.11 | dlynes_office | I never use redhat |
| 20:01.19 | dlynes_office | but I seen someoen else in here yesterday with the same problem |
| 20:01.34 | smackus | and that took care of the problem? |
| 20:01.38 | dlynes_office | yep |
| 20:01.40 | smackus | cool |
| 20:01.45 | Wazb^ | i am getting problem in cisco 12.3T & Asterisk connectivity, i am able to answer the call from ip phone but Call still receive Ring tone |
| 20:01.45 | dlynes_office | he had the wrong dev rpm installed |
| 20:01.53 | smackus | having the admin install it as we speak. |
| 20:02.10 | dlynes_office | good admin |
| 20:02.16 | dlynes_office | installed the wrong rpm :p |
| 20:02.18 | TESTER2 | Is there a procedure for calibrating tdm400p (ex: fxotune -s at each reboot and fxotune -i 5 in a crontab once a week...) |
| 20:02.31 | *** join/#asterisk FANCHO (n=fancho@201.216.202.113) |
| 20:02.33 | dlynes_office | TESTER2: it's set and forget, usually |
| 20:02.40 | dlynes_office | TESTER2: unless you've got a really bad line noise issue |
| 20:02.57 | dlynes_office | TESTER2: and then you're probably better off getting carrier grade hardware echo cancellation |
| 20:03.25 | dlynes_office | TESTER2: then you don't need to adjust gains |
| 20:03.31 | TESTER2 | ok so only fxotune -s at each reboo |
| 20:03.48 | Wazb^ | any document or link regarding Asterisk & Cisco 3800 (12.3T) connectivity ? |
| 20:03.48 | dlynes_office | wtf is fxotune -s, anyways? |
| 20:04.00 | TESTER2 | apply fxotune.conf |
| 20:04.06 | dlynes_office | ah |
| 20:04.07 | dlynes_office | i guess so |
| 20:04.19 | dlynes_office | why not just edit zapata.conf? |
| 20:04.27 | dlynes_office | it would seem a lot simpler |
| 20:05.36 | TESTER2 | there a lot of numbers in fxotune.conf and only 2 in zapata (rx and tx gain) |
| 20:06.00 | dlynes_office | TESTER2: ah...never really used fxotune |
| 20:06.03 | Wazb^ | any document or link regarding Asterisk & Cisco 3800 (12.3T) SIP call connectivity ? |
| 20:06.06 | dlynes_office | TESTER2: never had a card that could use it |
| 20:06.10 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 20:06.16 | dlynes_office | TESTER2: I wrote my own fxotune for x100p's |
| 20:06.22 | TESTER2 | good :) |
| 20:06.27 | dlynes_office | TESTER2: and sangoma a200d's don't need it |
| 20:06.40 | TESTER2 | with the EC? |
| 20:06.53 | dlynes_office | yeah...'d' means it has a carrier grade hwec |
| 20:06.59 | TESTER2 | 700$ :) |
| 20:07.02 | dlynes_office | Maybe |
| 20:07.04 | dlynes_office | No idea |
| 20:07.08 | dlynes_office | i buy wholesale |
| 20:07.55 | TESTER2 | even on ebay I can't find a a200 with EC |
| 20:07.56 | kpettit | I'm using the voicemail system, leaving a unavialbe message for a certian exten but it never plays it, it jsut defaults to the generic. I can see that the unavail.wav is there. Why wouldn't it be playing it? |
| 20:08.04 | dlynes_office | on fleabay? |
| 20:08.07 | TESTER2 | so I bougth a new tdm400p (tdm11p) |
| 20:08.07 | dlynes_office | of course not |
| 20:08.21 | dlynes_office | the a200d's just came out in March I think it was |
| 20:09.49 | *** join/#asterisk charles___ (n=charles@fw.invosat.com) |
| 20:09.59 | Wazb^ | any help? |
| 20:10.15 | dlynes_office | I've already bought three of those, and two of their pri cards in the last three weeks or so |
| 20:10.30 | dlynes_office | they're quite good |
| 20:12.49 | vader-- | hmmm |
| 20:13.04 | vader-- | anyone in here using cisco 7940/7960G phones without cisco switches? |
| 20:13.10 | TESTER2 | I would have prefer a s200 + EC but for a home setup is a little bit $$$$ |
| 20:13.24 | vader-- | im trying to specify the operational vlan for the pc to use but from what im reading it only works with cisco switches |
| 20:13.29 | vader-- | is there a work around? |
| 20:13.39 | dlynes_office | vader--: lots of peeps are using the 7940 without cisco switches for sure |
| 20:13.51 | dlynes_office | vader--: first time i've seen the 7960g mentioned though, i think |
| 20:14.22 | dlynes_office | vader--: but qwell's pretty much the residential expert on cisco, here |
| 20:14.33 | dlynes_office | vader--: he's the chap that wrote chan_skinny |
| 20:14.33 | vader-- | is he around? |
| 20:14.35 | denon | residential expert? |
| 20:14.37 | Qwell[] | nope |
| 20:14.45 | vader-- | hehe |
| 20:14.52 | Qwell[] | dlynes_office: fixed...not wrote |
| 20:14.54 | dlynes_office | s/denon/null/ |
| 20:14.57 | *** join/#asterisk Vagabond (i=andrew@pdpc/supporter/active/Vagabond) |
| 20:15.03 | denon | residential cisco expert == linksys expert <G> |
| 20:15.09 | dlynes_office | erm yeah...fixed/rewrote |
| 20:15.20 | denon | kind of an insult heh |
| 20:15.29 | vader-- | qwell know much about setting the operational vlan on cisco 7940g without having to have a cisco switch |
| 20:15.36 | Qwell[] | noe |
| 20:15.38 | Qwell[] | nope too |
| 20:16.22 | dlynes_office | oh the joys of compiling asterisk on a p75 :p |
| 20:16.29 | denon | ick |
| 20:16.35 | Qwell[] | distcc |
| 20:16.47 | denon | hehe yeah, load balance it over lots of p75s :) |
| 20:16.50 | dlynes_office | lol |
| 20:17.03 | dlynes_office | It had a load average of 1.07 :p |
| 20:17.05 | Qwell[] | I'm going to distcc install gentoo on my sunfire tonight :D |
| 20:17.16 | *** join/#asterisk digime (n=digime@user-0cdf0g7.cable.mindspring.com) |
| 20:17.23 | dlynes_office | 94.6% cpu usage by cc :p |
| 20:17.31 | dlynes_office | and 50% memory usage :) |
| 20:17.48 | philv | Hrmn, so, don't tell Cedric but I'm doing a total rewrite of chan_unistim, heh :) |
| 20:17.50 | denon | not bad, considering it's probably only got 32mb |
| 20:17.50 | *** part/#asterisk Vagabond (i=andrew@pdpc/supporter/active/Vagabond) |
| 20:17.56 | dlynes_office | philv: cool |
| 20:18.05 | dlynes_office | philv: so did you get it working with your i2004 yet? |
| 20:18.15 | philv | Nope... it's a problem with the i2004's firmware. |
| 20:18.23 | philv | Early firmwares didn't use Reliable UDP, as it would turn out |
| 20:18.27 | dlynes_office | ah...thought you said it worked on a bcm? |
| 20:18.36 | philv | Yup, it does :) |
| 20:18.39 | dlynes_office | or it was flaky on the bcm, too? |
| 20:18.40 | philv | It knows about these phones. |
| 20:18.48 | dlynes_office | ic |
| 20:18.59 | dlynes_office | so you upgraded the firmware and now it works fine? |
| 20:19.31 | philv | Not yet |
| 20:19.40 | dlynes_office | anyways...if you want any help with the code, i'd love to help |
| 20:19.41 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
| 20:19.42 | philv | I will tonight, and I'm going to watch the unistim traffic. |
| 20:19.54 | philv | So that way I can make chan_unistim able to push new firmwares :) |
| 20:20.00 | dlynes_office | That woudl be kick ass to see Nortel phones working with asterisk |
| 20:20.21 | philv | It's just tftp, but there's a special packet you need to send to initiate it. |
| 20:21.00 | philv | dlynes_office: I agree :) |
| 20:21.21 | philv | A few of the firmware guys are a floor up from mine, so I can bug them every now and then for advice, heh |
| 20:21.49 | *** join/#asterisk nothinman (i=shakey@adar182.neoplus.adsl.tpnet.pl) |
| 20:21.56 | dlynes_office | heh |
| 20:22.18 | dlynes_office | as i understand it, the Meridian 1 can talk unistm? |
| 20:23.47 | dlynes_office | philv: i guess the meridian 1's that it's compatible with would be newer than the Option 11E, eh? |
| 20:24.12 | philv | Dunno, to be quite honest. That completely isn't my line of business, heh |
| 20:24.19 | dlynes_office | ah |
| 20:24.21 | philv | I know Contivity |
| 20:24.23 | *** part/#asterisk xunil (n=wkurdzio@office1.visionpointsystems.com) |
| 20:24.32 | philv | Contivity and Metro is what I do, heh |
| 20:25.12 | philv | I'm gonna work at the whole Meridian/Norstar line eventually. I should learn that. |
| 20:25.46 | dlynes_office | philv: contivity is vpn? |
| 20:26.08 | philv | I think some of the Meridian gear did UNIStim. The BCM telephony guys stole a lot of code from meridian, heh |
| 20:26.14 | philv | dlynes_office: Sure is |
| 20:26.27 | philv | VPN/Router/Firewall/Kitchen Sink |
| 20:26.29 | dlynes_office | ah...just never heard of contivity or metro before |
| 20:26.35 | philv | Ahh |
| 20:26.48 | dlynes_office | I guess metro is probably a low end Bay Networks switch? |
| 20:26.58 | philv | Metro is the carrier and high-end enterprise grade optical networking (SONET, heh) |
| 20:27.03 | dlynes_office | oh |
| 20:27.08 | philv | The low-end Bay Networks switches were BayStack |
| 20:27.13 | vader-- | Does anyone in here know a way to set the operational and admin vlans on the Cisco 7940G/7960G phones without having to have a Cisco Catalyst switch? I currently have extreme network switches and i can get the phones to work on the voice vlan but that doesn't make for good practice with data flowing across the voice vlan from the PC. Any suggestions would be mucho appreciated. |
| 20:27.19 | dlynes_office | Yeah...I've got some baystacks |
| 20:27.19 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 20:27.23 | philv | Though we call them ERS (for the routing switches) and ES (for the non-routing managed switches) |
| 20:27.25 | dlynes_office | They're actually pretty good switches |
| 20:27.33 | philv | I think so, yeah :) |
| 20:27.39 | philv | I'm biased though, haha |
| 20:27.41 | dlynes_office | i'm connected through one right now, actually ;) |
| 20:27.55 | philv | Haha, cool. |
| 20:28.05 | philv | I've got one at home. |
| 20:28.24 | dlynes_office | the biggest problem i had with the baystack hubs though, is that they had problems connecting to a cisco 10Mb network |
| 20:28.42 | philv | Hrmn, really? |
| 20:28.52 | dlynes_office | Yeah...they weren't configurable either, afaik |
| 20:29.00 | dlynes_office | The switch I'm connected to atm though |
| 20:29.03 | dlynes_office | is quite configurable |
| 20:29.05 | philv | Strange... very strange. |
| 20:29.20 | dlynes_office | It's a 350F-HD |
| 20:29.25 | dlynes_office | It's a pretty nice switch |
| 20:29.30 | philv | Yup yup |
| 20:29.32 | dlynes_office | picked it up for $75Cdn : |
| 20:29.35 | philv | :D |
| 20:29.41 | dlynes_office | damned good deal :0 |
| 20:30.00 | philv | I guess the benefit of Nortel gear not being used for training practice and such is that it gets cheaper quicker |
| 20:30.01 | dlynes_office | Got two sunsparcs from the same guy for $180Cdn :) |
| 20:30.17 | philv | Whereas a Catalyst 2900 goes for an arm-and-a-leg STILL |
| 20:30.19 | philv | :) |
| 20:30.25 | *** part/#asterisk FANCHO (n=fancho@201.216.202.113) |
| 20:30.29 | dlynes_office | training practice? |
| 20:30.41 | philv | Yeah... CCNA people buy up the old Cisco gear like crazy. |
| 20:30.53 | dlynes_office | ah |
| 20:30.57 | philv | so they can train for CCNA, CCIE, CCSE or whatever |
| 20:31.17 | dlynes_office | they should buy bay networks instead |
| 20:31.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:31.24 | dlynes_office | give the canadians the money instead :) |
| 20:32.13 | philv | Haha :) |
| 20:32.36 | philv | There are Nortel certifications... just nobody has heard of them because Nortel marketing is lame right now :( |
| 20:33.04 | dlynes_office | Well, and nortel probably charges an arm and a leg for them |
| 20:33.18 | philv | Hrmn, I think the exams are about 150 bucks |
| 20:33.20 | philv | But yeah |
| 20:33.24 | philv | The stuff ain't cheap |
| 20:33.26 | philv | :( |
| 20:33.26 | dlynes_office | oh...that's not too bad then |
| 20:33.36 | dlynes_office | I was thinking $2000-3000 |
| 20:33.52 | philv | Hrmn, that might be the Employee price... one sec |
| 20:33.59 | dlynes_office | hahahaa |
| 20:36.27 | philv | It _looks_ like it's 149 CAD for anyone, |
| 20:36.41 | philv | That's the Nortel Networks Design Specialist |
| 20:36.46 | dlynes_office | ah...so employees are getting screwed :) |
| 20:37.06 | philv | Hrmn, I should do the upcoming one on Contivity... that'd be a freebie |
| 20:38.20 | *** join/#asterisk Vagabond (i=andrew@pdpc/supporter/active/Vagabond) |
| 20:38.54 | *** part/#asterisk Vagabond (i=andrew@pdpc/supporter/active/Vagabond) |
| 20:39.20 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 20:39.25 | smackus | ok, zaptel happy :-D |
| 20:39.41 | philv | bbl, gonna head home |
| 20:40.05 | dlynes_office | have a good trip |
| 20:40.18 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 20:40.49 | smackus | so now, I was asking about this earlier this morning, but is there anyone here who has played with getting the ACD agent stuff to work with the polycom phones? i have enabled the stuff in the phone1.cfg and it displays the ACD agent icon on then phone. But no matter if I am logged in or not, the ACD display does not change. Has anyone had any experience with this? |
| 20:40.57 | smackus | wow.. .that was long. sorry |
| 20:42.36 | vader-- | Does anyone in here know a way to set the operational and admin vlans on the Cisco 7940G/7960G phones without having to have a Cisco Catalyst switch? I currently have extreme network switches and i can get the phones to work on the voice vlan but that doesn't make for good practice with data flowing across the voice vlan from the PC. Any suggestions would be mucho appreciated. |
| 20:42.55 | Qwell[] | vader--: asking every 2 minutes isn't going to get you an answer |
| 20:43.27 | vader-- | sorry :( |
| 20:43.30 | smackus | Qwell : was that directed to me? |
| 20:43.36 | smackus | or sorry... vader |
| 20:43.37 | vader-- | im just sitting with 60 of these things |
| 20:46.22 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.150) |
| 20:46.53 | *** join/#asterisk copantl (n=galel@190.4.22.82) |
| 20:47.14 | copantl | any body know what happen with a2billing site? |
| 20:48.53 | [TK]D-Fender | smackus : You you using bweskhe's patched SVN tree? |
| 20:50.07 | copantl | any body use a2billing? |
| 20:51.55 | charles___ | vader--: are you having performance problems ? |
| 20:52.02 | charles___ | vader--: quality problems ? |
| 20:53.34 | clive- | copantl hi, I know someone who uses it,,,why do you ask ? |
| 20:54.57 | copantl | clive-: because , the website is out and the irc is empty |
| 20:55.44 | [TK]D-Fender | copantl : Careful, or the ninjas will get you too! ;) |
| 20:56.15 | znoG | does anyone know what the call forward activate/deactivate codes are for the grandstream handytone range? |
| 20:56.22 | clive- | copantl dunno,....use astcc rather then |
| 20:56.37 | copantl | astcc.org? |
| 20:56.42 | copantl | or irc? |
| 20:57.12 | clive- | its an addon i think |
| 20:59.50 | znoG | question: i have my dial app configured to use a voip provider, failing that use the TDM card. Now, if my provider is down, the dial command takes like a minute or so to timeout... is there a way to make the timeout around 10 seconds? |
| 21:00.09 | sivana | anyone here using IAXModem and Hylafax? |
| 21:00.47 | pdunkel | znoG: Try using qualify=yes in sip.conf |
| 21:01.06 | pdunkel | That should mark Your VoIP Provider unavail sooner |
| 21:01.25 | pdunkel | Like Timeout Immediately |
| 21:02.33 | pdunkel | Also fiddle with the qualify options, because it can get tricky if your intervals go too low. |
| 21:02.56 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 21:03.04 | pdunkel | If you have line lags, also calls can get killed. But again only if your intervals are too low |
| 21:04.24 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
| 21:04.46 | znoG | pdunkel: nice, just tried that and it auto-congested the call due to slow response! |
| 21:05.05 | funxion | I know this will sounds slightly retarded but I'm trying to copy the source files of asterisk libpri and zaptel from one server to another mainly to have the exact same version wwhile asterisk and lib pri will compile and run fine zaptel will not I'm getting |
| 21:05.06 | funxion | In file included from zaptel.c:42: |
| 21:05.06 | funxion | /usr/include/linux/kernel.h:72: error: syntax error before "size_t" |
| 21:05.06 | funxion | /usr/include/linux/kernel.h:74: error: syntax error before "size_t |
| 21:05.40 | funxion | I hve the sources |
| 21:05.46 | funxion | kernel source |
| 21:05.52 | funxion | anyone? |
| 21:06.00 | pdunkel | znoG: As I said it does take some tuning. Because if you lag too much, your peer is marked as UNREACHEABLE. |
| 21:06.22 | pdunkel | If that happens while a call is in Progress, there are situations when the call is stopped |
| 21:07.00 | znoG | uh, thats not good |
| 21:07.06 | znoG | but then, if its laggy, probably best it does that :) |
| 21:07.25 | angler | funxion, what kernel version? |
| 21:07.25 | pdunkel | znoG: Just set the Intervals high enough and it stops being a problem! |
| 21:08.01 | funxion | kernel-source-2.4.27 |
| 21:08.24 | pdunkel | znoG: Took me about FOREVER to figure out what was going on with that. So :) is not what I felt at the time. So I do agree now. |
| 21:08.53 | rene- | i need quality, and competitive pricing on routes to central america, over IAX, who provides? |
| 21:08.54 | funxion | which has never given me this issue in the past |
| 21:09.10 | pdunkel | rene-: Try voipjet.com |
| 21:10.39 | twisted[asteria] | OH GNOES |
| 21:10.43 | twisted[asteria] | MY DIXIE WRECKED! |
| 21:10.43 | tzanger | haha yeah try voipjet |
| 21:10.47 | tzanger | twisted[asteria]: hahaha |
| 21:10.55 | pdunkel | funxion: Sounds like it is missing includes which might be due to stale ./configure stuff, But is zaptel auto-conf yet? |
| 21:11.15 | Qwell[] | twisted[asteria]: eh? |
| 21:11.28 | [TK]D-Fender | twisted[asteria] : Life is like that.... if it gets to hard, FUCK IT :) |
| 21:11.35 | funxion | stale ./configure |
| 21:11.37 | twisted[asteria] | [TK]D-Fender, :P |
| 21:11.39 | funxion | I dont get it |
| 21:11.45 | tzanger | funxion: say it out loud |
| 21:11.49 | funxion | there isn't ./configure in zaptel |
| 21:11.55 | marv[work] | twisted[asteria]: did your anus stop bleeding yet? |
| 21:11.56 | *** join/#asterisk Meaty (n=meaty3@66.254.41.11) |
| 21:11.57 | tzanger | say "I am we todd it, I am sofa king we todd it" out loud |
| 21:11.59 | funxion | lol @ tzafrir |
| 21:12.02 | funxion | lol @ tzanger |
| 21:12.05 | twisted[asteria] | marv[work], yours is the one that was bleeding, mk? |
| 21:12.05 | funxion | sry tzafrir |
| 21:12.30 | twisted[asteria] | btw |
| 21:12.33 | twisted[asteria] | when you say marv[work] |
| 21:12.37 | twisted[asteria] | his computer makes noises |
| 21:12.52 | pdunkel | tzanger: haha? I really like people that are unwilling to google IAX Provider .... :( |
| 21:12.52 | twisted[asteria] | everyone should say it |
| 21:12.52 | Qwell[] | twisted[asteria]: say what, marv[work]? |
| 21:13.04 | twisted[asteria] | yeah, marv[work] |
| 21:13.12 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
| 21:13.30 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
| 21:13.59 | marv[work] | yes, keep doing it, maybe someone will hear and save me from the man who's DIXIE WRECKED |
| 21:14.12 | twisted[asteria] | it's not like it crashed into you |
| 21:15.03 | marv[work] | this is true, who did it crash into again? and where's gavin at? |
| 21:15.12 | funxion | anyone know how I would tell what version of zaptel Im using? |
| 21:15.29 | pdunkel | funxion: Then I guess I was wrong. Do you have the kernel sources only, or have you also compiled it. Check if your kernel includes are in /usr/include |
| 21:15.48 | funxion | I did compile it |
| 21:16.37 | pdunkel | funxion: I hate these cases. You usually spend AGES figuring out what is wrong, and then it's usually something stupid and little. :) |
| 21:16.49 | funxion | pdunkel I agree |
| 21:17.07 | funxion | do you know how I would be able to tell what versin of zaptel Im using |
| 21:18.28 | x86 | I have a remote device setup on my asterisk server with both SIP and IAX. I have the device in the same exact context in both sip.conf and iax.conf, yet it only works via SIP. with IAX it reports "No such extension/context" |
| 21:18.33 | x86 | why would that be? |
| 21:18.55 | pdunkel | funxion: cat version.h |
| 21:19.02 | funxion | tnx |
| 21:19.44 | *** part/#asterisk speedwagon (n=FreePBX7@70.46.87.158) |
| 21:19.55 | funxion | pdunkel there isnt a verion.h |
| 21:20.17 | pdunkel | funxion: But there is a version.h |
| 21:20.28 | pdunkel | :) |
| 21:20.35 | funxion | I spelled it right when i was looking for it |
| 21:20.36 | Henk | Hi, I've set up the default asterisk woth nothing but a modified SIP.CONF. If i call the number i registered with (budgetphone) I get the asterisk demo. So that works. But if I drop a call file in the spooldir I get an error saying that the login is incorrect to INVITE . what am i doing wrong ? |
| 21:20.41 | funxion | but its not in th efodler |
| 21:20.48 | funxion | Its a slightly odler version |
| 21:20.51 | funxion | older |
| 21:21.32 | pdunkel | funxion: How did you copy the sources? Cause I think version.h was there in 0.0.0a |
| 21:21.39 | funxion | ftp |
| 21:22.12 | pdunkel | funxion: What did you transfer? You seem to be missing files |
| 21:22.34 | pdunkel | funxion: Maybe retransfer after archiving |
| 21:22.43 | funxion | onna try that now |
| 21:22.44 | funxion | thnx |
| 21:25.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:29.55 | rene- | a quick one: how do i include a file from within an asterisk configuration file? |
| 21:30.43 | rene- | nah #include path |
| 21:31.20 | hads | rene-: #include "other.conf" |
| 21:31.23 | hads | Yes. |
| 21:33.18 | rene- | my agent definition names may include unicode characters could this be a problem in asterisk? |
| 21:33.58 | *** join/#asterisk Xen^ (n=linux@203.101.168.2) |
| 21:34.03 | hads | Not sure, quite possibly. |
| 21:34.08 | rene- | :( |
| 21:34.22 | hads | I don't use any so I couldn't tell you sorry. |
| 21:35.38 | Xen^ | hello every one |
| 21:35.41 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 21:36.19 | Xen^ | can some one tell me I have added my telephone number in e164.org and when i do enum lookup in asterisk it will give resutl TEL:number |
| 21:36.28 | Xen^ | how can we dial TEL using enum ? |
| 21:36.32 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 21:38.03 | smackus | [TK]D-Fender: Sorry, I stepped away for a second. You asked me if I was using weskhe's patched SVN tree. Where do I find this? |
| 21:38.41 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 21:38.45 | viler | Exists some client written i |
| 21:38.58 | viler | Exists some sip or iax client written in Java ? |
| 21:39.29 | rene- | thereis moziax and yes i believe there are java iax softphones |
| 21:39.33 | rene- | ~google |
| 21:39.35 | jbot | rumour has it, google is a search engine found at http://www.google.com/ |
| 21:40.06 | dlynes_office | smackus: you mean bweschke? |
| 21:40.11 | Xen^ | rene- : an you help me |
| 21:40.20 | Xen^ | rene- : can some one tell me I have added my telephone number in e164.org and when i do enum lookup in asterisk it will give resutl TEL:number |
| 21:40.49 | rene- | i am sorry Xen^: i havent used e164.org ever, and i know i should ve |
| 21:40.59 | Xen^ | no problem |
| 21:41.04 | Xen^ | atleast you replied :) |
| 21:41.14 | heath__ | jiaxclient |
| 21:42.13 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 21:42.16 | [TK]D-Fender | smackus : Base * does not support Polycom's ACD signalling, but there is a specific SVN tree that includes patches for support, but they aren't complete. |
| 21:42.30 | [TK]D-Fender | smackus : Check the bug tracker for info on it. |
| 21:42.52 | *** join/#asterisk lars-ut (n=lars-ut@70.103.228.158) |
| 21:43.34 | *** join/#asterisk Wazb^ (n=wazb@199.243.74.220) |
| 21:44.20 | Wazb^ | anyone have idea about ASterisk & Cisco 3800 (12.3T) SIP connectivity |
| 21:45.03 | rene- | Wasb^: try the wiki |
| 21:45.15 | Wazb^ | i looked there |
| 21:46.22 | rene- | were is the free music on hold? that doesnt come with * |
| 21:49.46 | rene- | http://puck.nether.net/pipermail/cisco-voip/2005-January/002822.html for a thread of a dude that plugged asterisk to a 3825 |
| 21:49.52 | AlexCTI | Does anyone know how can put a call on top of the line of the queue? |
| 21:50.18 | rene- | i dont think that is possible |
| 21:50.23 | [TK]D-Fender | AlexCTI : Set their priority before they go in. |
| 21:50.27 | rene- | ahh |
| 21:50.38 | TESTER2 | someone has gain problem with the MG2 EC algo? |
| 21:51.04 | rene- | i as going to suggest using multiple queues with different weights, but then I DIDNT KNEW ASTERISK DID THAT. TM |
| 21:51.20 | AlexCTI | TK-Fender, I have tryied many options but I don't get it works |
| 21:53.38 | AlexCTI | TK-Fender, That my sentence exten => s,1,Set(${PRIORITY}=$[${PRIORITY}+1]) |
| 21:54.01 | [TK]D-Fender | AlexCTI : exten => whatever,1,Set(QUEUE_PRIO=10) |
| 21:56.45 | AlexCTI | TKD-Fender, That's my scenario: I send the call to the queue and after timeoit i give and option to leave a VM, and if the people don;t want to do that I send them back to the queue, so, that sentense should me there at the begining? |
| 21:57.47 | [TK]D-Fender | AlexCTI : You shouldn't take them out of the queu on timeout only to throw them back in if they ddon't want to leave a VM. The only reason to leave a queue is to increment the priority or to leave a VM forcably. |
| 21:58.17 | [TK]D-Fender | AlexCTI : Although what you SHOULD be doing is using the exit context to allow them to SHOOSE to quit and leave a VM if they don't feel like waiting. |
| 21:58.32 | *** part/#asterisk zamba (i=marius@2001:730:5:21e0:0:0:0:1) |
| 21:58.54 | vader-- | ok i figured out my cisco 7940G vlan problem |
| 21:59.15 | vader-- | i sent the pc vlan untagged and the voice vlan tagged and set the admin vlan on hte phone to the voice vlan and life it good |
| 21:59.16 | vader-- | :) |
| 21:59.19 | [TK]D-Fender | AlexCTI :Queue's are automatically FIFO, so the only reason to escalate a priority is if you are already doing that by default through other means. |
| 21:59.56 | vader-- | it is |
| 22:01.25 | Wazb^ | i am using Cisco 3800 (12.3T) , i am forwarding SIP call from cisco3800 to ASterisk box, prob. is call get disconnect after 1 sec upon answer |
| 22:03.43 | [TK]D-Fender | OK, ITS CHECKOUT TIME... MAYBE BACK LATER TONIGHT |
| 22:04.28 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 22:11.37 | rene- | BYEBYE |
| 22:11.45 | *** join/#asterisk gromm{CA} (n=me@xx081151026.cipherkey.com) |
| 22:11.49 | rene- | Wazb^ sounds like a codec issue |
| 22:12.44 | rene- | i would look into a codec issue first |
| 22:12.59 | rene- | what your asterisk logs are sayig |
| 22:12.59 | rene- | saying |
| 22:15.47 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:16.43 | *** join/#asterisk tdonahue (i=tdonahue@207.138.151.58) |
| 22:17.30 | *** join/#asterisk Kokey (n=jramirez@201.123.192.227) |
| 22:18.16 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 22:18.53 | marv[work] | Like a Rainbow! |
| 22:20.46 | robin_sz | what? bent and hard to track down? |
| 22:23.09 | *** join/#asterisk jets (i=jets@ownsu.com) |
| 22:25.01 | jmacz | Hi everyone |
| 22:26.04 | jmacz | I would like to know what do you guys think about monitor vs mixmonitor from your experience |
| 22:26.31 | dlynes_office | i much prefer mixmonitor, myself |
| 22:26.54 | dlynes_office | however, i've talked to a few people that have told me mixmonitor will hang the machine after being run about 1000 times or so |
| 22:26.58 | *** join/#asterisk los415 (n=los415@sfca-office.corp.race.com) |
| 22:26.58 | jmacz | dlynes_office, for some special reason? does it has lesser CPU consumption or something? |
| 22:27.03 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:27.32 | dlynes_office | for the one individual, 1000 calls was less than a day |
| 22:27.33 | jmacz | I understand |
| 22:27.34 | dlynes_office | jmacz: no need to create all the extra environment for running a shell |
| 22:27.41 | dlynes_office | jmacz: that's a huge overhead, imo |
| 22:27.51 | jmacz | dlynes_home, that's right |
| 22:28.03 | *** part/#asterisk W4 (n=W4@212.58.232.254) |
| 22:28.22 | dlynes_office | and for me, that's reason enough to use mixmonitor over monitor right there |
| 22:28.28 | jmacz | at asterisk-guru they talk about mixmonitor as "an upgraded version" for monitor so I was curious |
| 22:29.09 | dlynes_office | jmacz: yeah...basically instead of having to spawn a sox subprocess now, mixmonitor does everything natively in asterisk |
| 22:29.15 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 22:29.40 | dlynes_office | same for native music on hold...no need to run mpg123 (buggy as all hell), or madplay as a subprocess |
| 22:30.41 | dlynes_office | for native music on hold you'll need to install format_mp3 from asterisk-addons |
| 22:30.45 | jmacz | yes, that's the point with native moh |
| 22:30.57 | jmacz | dlynes_office, thanks for sharing your point of view with me :) |
| 22:31.02 | dlynes_office | It was written by the chap on here with the nick 'anthm' |
| 22:31.23 | dlynes_office | jmacz: well, i'm a programmer |
| 22:31.35 | dlynes_office | jmacz: and programmers don't necessarily live in the real world, either :) |
| 22:33.10 | zwelch | i read a HOWTO about joining the real world. it seemed like a lot of work |
| 22:33.15 | jmacz | dlynes_office, he he, I understand you. BTW don't let the Matrix catch you again ;) |
| 22:33.42 | CunningPike | zwelch: There's a HOWTO??? |
| 22:33.45 | *** join/#asterisk MarcPtz (n=MarcPtz@18.Red-80-35-146.staticIP.rima-tde.net) |
| 22:33.51 | jmacz | zwelch, have the link or something? |
| 22:34.13 | zwelch | lol. it was a joke, folks. :) laugh. it was supposed to be funny |
| 22:34.14 | MarcPtz | Hi , is ok to ask support questions here ? |
| 22:34.28 | denon | yes, just ask |
| 22:34.54 | jmacz | zwelch, guess I'll have to write one by myself :P |
| 22:35.09 | CunningPike | I wonder is there a wiki......... :( |
| 22:35.20 | zwelch | i was just about to say: this is not to say that some programmers could use such a document :) |
| 22:35.47 | MarcPtz | thanks! I'm really confused with the error "Got SIP response 482 "Loop Detected" back from x" , does it mean than extension is trying to call itself? |
| 22:36.14 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 22:36.28 | zwelch | MarcPtz: that sounds strange, because i know that soft phones can happily call themselves |
| 22:37.19 | zwelch | can you describe the simplest scenario that generates that response? |
| 22:37.31 | jmacz | MarcPtz, do you have a phone with multiple-lines support? |
| 22:37.44 | MarcPtz | yes , I'm using X-Lite |
| 22:37.47 | dlynes_office | MarcPtz: maybe you've got forward when busy back to itself on the phone? |
| 22:38.20 | dlynes_office | MarcPtz: which would in that case be a sip redirect, not the phone calling asterisk and then asterisk calling the phone |
| 22:39.00 | *** join/#asterisk knarfly (n=Knarf980@c-69-180-98-189.hsd1.fl.comcast.net) |
| 22:39.32 | knarfly | Can anyone help me with remote access calling using my Asterisk server? |
| 22:41.12 | MarcPtz | I don't know what more to try .. may be a problem with NAT ? |
| 22:41.39 | dlynes_office | MarcPtz: did you try my suggestion about call forwarding turned on, for the phone? |
| 22:42.42 | MarcPtz | dlynes_office : how can I do that? |
| 22:42.53 | dlynes_office | MarcPtz: you're the one running x-lite, not I |
| 22:42.57 | dlynes_office | MarcPtz: I have no idea |
| 22:43.14 | dlynes_office | MarcPtz: all i'm suggesting is for you to disable call forwarding |
| 22:43.22 | *** join/#asterisk iq (n=iq@71-215-54-112.omah.qwest.net) |
| 22:44.06 | l-fy | being wild and being free |
| 22:44.12 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 22:44.20 | l-fy | this what you get when you have a brain |
| 22:44.58 | dlynes_office | you do? |
| 22:44.59 | jmacz | knarfly, you mean registering _in_ your server? |
| 22:45.16 | l-fy | yes dlynes_home |
| 22:45.20 | dlynes_office | jmacz: no...he wants someone to r00t him |
| 22:45.21 | l-fy | dlynes_office sorry :) |
| 22:45.55 | knarfly | jmacz: i'm interested in learning how to make a call using my * server from a remote location |
| 22:45.58 | l-fy | dlynes_office > actually what you get is that you are allow to be wild and free |
| 22:46.16 | dlynes_office | l-fy: i thought that was brains combined with alcohol? |
| 22:46.34 | dlynes_office | erm no wait |
| 22:46.40 | l-fy | nope |
| 22:46.42 | jmacz | he he |
| 22:46.43 | dlynes_office | the two are kinda mutually exclusive :((( |
| 22:47.23 | gromm{CA} | knarfly: that depends. The simple answer is "take your softphone/ata with you". :) |
| 22:47.51 | jmacz | knarfly, sure, well you can do that in many ways, the easiest is to let extensiones register from Internet in your box |
| 22:48.27 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:48.37 | *** join/#asterisk reticent (n=reticent@207-232-98-114.ip.van.radiant.net) |
| 22:49.01 | dlynes_office | kiwi! |
| 22:49.13 | reticent | does anyone know of a way to change the permissions set on voicemail files by asterisk when new voicemail is left? |
| 22:49.21 | dlynes_office | reticent: ah...another fellow bc'er |
| 22:49.31 | reticent | currently its 700 |
| 22:49.36 | reticent | yeah, vancouver =P |
| 22:49.40 | reticent | where are you from? |
| 22:49.41 | dlynes_office | reticent: yeah...same here |
| 22:49.46 | dlynes_office | reticent: 49th and vic |
| 22:49.54 | reticent | gastown |
| 22:50.04 | knarfly | jmacz: okay, think I got it. What about getting two * servers to register with each other. |
| 22:50.35 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:50.38 | knarfly | jmacz: how do I go about telling on server to let the other register with it? |
| 22:50.55 | dlynes_office | reticent: until 1.4 comes out, you're going to have to use umask |
| 22:51.07 | dlynes_office | reticent: or change the voicemail code |
| 22:51.19 | dlynes_office | reticent: erm |
| 22:51.24 | dlynes_office | reticent: why is it 0700? |
| 22:51.38 | Qwell[] | dlynes_office: because executable voicemail == fun |
| 22:51.42 | jmacz | knarfly, then the best solution is to use iax |
| 22:51.54 | jmacz | that makes it easier |
| 22:52.07 | knarfly | jmacz: what conf file do i setup the login information in? |
| 22:52.16 | dlynes_office | Qwell[]: weird...looks like mine's that way too |
| 22:52.29 | dlynes_office | Qwell[]: another thing for me to fix, while i'm modifying app_voicemail |
| 22:52.35 | Qwell[] | already fixed |
| 22:52.54 | dlynes_office | Qwell[]: ah...so what insane elf set it that way to begin with? a windows programmer? |
| 22:52.57 | knarfly | i'll be back |
| 22:53.11 | reticent | dy: i dont know ;P |
| 22:53.23 | dlynes_office | oh....Qwell[] did? |
| 22:53.27 | Qwell[] | no :p |
| 22:53.30 | dlynes_office | heh |
| 22:53.41 | dlynes_office | kram? |
| 22:53.47 | Qwell[] | got me |
| 22:53.50 | dlynes_office | ah |
| 22:53.51 | reticent | dy: i've tried setting the umask, but no luck |
| 22:53.51 | jmacz | knarfly, iax.conf, check this out http://www.voip-info.org/wiki/index.php?page=Asterisk+iax+rsa+auth |
| 22:54.23 | reticent | dy: oddly enough, when it moves them, the perms change to 0600 |
| 22:57.18 | *** join/#asterisk SgtDitt (n=mike@63.251.157.162) |
| 22:57.25 | reticent | so configuring the permissions on voice messages is a feature for 1.4? |
| 22:57.34 | Qwell[] | no |
| 22:58.27 | dlynes_office | reticent: it's just fixed to sane values |
| 22:58.31 | dlynes_office | reticent: for 1.4 |
| 22:59.43 | dlynes_office | reticent: i believe the changes are committed to trunk if you want to try it |
| 23:00.05 | reticent | down to 0027 or some such? |
| 23:00.16 | dlynes_office | I don't know, offhand |
| 23:00.22 | dlynes_office | I'd have to look at the code |
| 23:00.42 | reticent | i'll have a look, thanks |
| 23:00.55 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:03.00 | reticent | hmmm |
| 23:03.01 | reticent | that could work |
| 23:03.32 | reticent | right in the beginning of app_voicemail.c there are a few defines that hold umasks |
| 23:03.45 | reticent | i was wondering why the env umask wasn't working |
| 23:03.51 | reticent | hopefully this will work |
| 23:04.31 | dlynes_office | reticent: hrm...I can't even find a bug report/feature request...I guess it was fixed without prompting |
| 23:06.44 | *** join/#asterisk adker (n=adker@67-136-213-243.dsl1.glv.ny.frontiernet.net) |
| 23:09.26 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 23:09.56 | reticent | i wonder if honouring the environment umask would be appropriate than focing a more restrictive one |
| 23:10.31 | reticent | might cause more inadvertant shot toes ;P |
| 23:10.43 | dlynes_office | or, you could honor umask |
| 23:10.54 | dlynes_office | and still allow it to be overridden in your voicemail.conf file |
| 23:11.05 | hads | Bug 6334 |
| 23:11.23 | *** join/#asterisk iq|mobile (n=iq@71-215-54-112.omah.qwest.net) |
| 23:11.57 | dlynes_office | hads: ah...that's why didn't find it...it's under core asterisk, not app_voicemail |
| 23:12.23 | hads | Yeah, I rembered seeing some discussion about it on asterisk-dev |
| 23:15.10 | reticent | welll |
| 23:15.13 | reticent | that half works ;/ |
| 23:15.27 | reticent | the directory permissions are set to 770 |
| 23:15.39 | reticent | files are still 700, at least theres hope ;P |
| 23:16.03 | *** join/#asterisk IMG-SD (n=IMG-SD@as2.imperialgroup.ca) |
| 23:18.00 | IMG-SD | Any "obvious" things that I should check for when receiving this status when doing a "pri show span 1" command: Status: Provisioned, Down, Active. I've been working with the telco for the past three hours and are no closer to a solution. How do I bring the D channel up? |
| 23:21.05 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 23:22.47 | *** join/#asterisk terracon (n=tc@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 23:23.15 | *** part/#asterisk terracon (n=tc@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 23:23.28 | *** join/#asterisk asterboy (n=kevin@S01060204ee2b6007.ed.shawcable.net) |
| 23:23.55 | gromm{CA} | IMG-SD: i feel your pain man. :) |
| 23:24.09 | asterboy | interesting that my Dell 450MHz, 64Mb system can outperform my P4 1.4GHz IBM machine on zttest scores |
| 23:24.34 | gromm{CA} | heh |
| 23:25.35 | asterboy | sure wish Digium et al, would come up with a database of motherboards whose PCI bus works 5 9s. |
| 23:25.58 | dlynes_office | asterboy: if it helps |
| 23:26.12 | dlynes_office | asterboy: my p75 only had one zttest score below 100.00% |
| 23:26.29 | *** join/#asterisk Kokey (n=jramirez@201.123.192.227) |
| 23:26.41 | asterboy | wonder what makes the difference. |
| 23:26.42 | dlynes_office | I'll sell you my P75 if you want |
| 23:26.51 | asterboy | sure isn't the cpu/memory |
| 23:28.34 | Qwell[] | asterboy: There are far too many for them to test... |
| 23:29.31 | dlynes_office | Dood...I'm telling you....I'll give you a great deal on this P75 Compaq Deskpro...it works grate with digium hardware |
| 23:29.37 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 23:30.23 | ariel_ | ~centosfix |
| 23:30.31 | dlynes_office | ~centosbug |
| 23:30.32 | jbot | [centosbug] a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 23:30.32 | dlynes_office | ~redhatbug |
| 23:30.34 | jbot | redhatbug is, like, is a problem with the latest RedHat Enterprise Linux and CentOS kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 23:31.12 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:37.14 | dlynes_office | asterboy: the P4 1.4GHz IBM machine...how many pci slots does it have? |
| 23:37.58 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.150) |
| 23:38.06 | Qwell[] | dlynes_office: 12 |
| 23:38.21 | dlynes_office | yeah, ok |
| 23:38.37 | Qwell[] | don't laugh..they exist :p |
| 23:38.53 | Qwell[] | ~lart dlynes_office |
| 23:39.21 | dlynes_office | ~wedgie qwell[] |
| 23:39.22 | jbot | ACTION hangs qwell[] from the flag pole by his underwear |
| 23:39.29 | dlynes_office | cute |
| 23:40.06 | dlynes_office | i miss my mud scripts for ircii :( |
| 23:40.50 | *** join/#asterisk coppice (n=chatzill@18.162.17.210.dyn.pacific.net.hk) |
| 23:42.49 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 23:43.06 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.150) |
| 23:44.29 | asterboy | dylnes_office, it has 3 |
| 23:46.57 | dlynes_office | asterboy: and i bet your 450MHz dell system has more than 3 pci slots |
| 23:47.11 | dlynes_office | asterboy: and it probably doesn't have some crappy intel nic on it, either |
| 23:52.29 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 23:52.38 | TESTER2 | tdm11p => low volume on the fxo even if rxgain=6 (higher than that=distortion)... any idea? |
| 23:53.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.150) |
| 23:53.58 | TESTER2 | very little difference between rxgain=0 and 6 |
| 23:55.59 | *** join/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
