| 00:00.06 | terrapen | 2.6.2 |
| 00:00.09 | terrapen | dunno what is on the phone |
| 00:00.14 | terrapen | just opened the box |
| 00:00.26 | CunningPike | Check the phone - it won't let you downgrade |
| 00:00.36 | terrapen | ok |
| 00:00.51 | terrapen | isn't there something bad/nasty about the 3.x bootroms? |
| 00:00.53 | CunningPike | 2.x -> 3.x is a one-way street |
| 00:00.57 | terrapen | yeah |
| 00:01.03 | terrapen | and doesn't 3.x do something evil? |
| 00:01.20 | CunningPike | terrapen: No - we confirmed with Polycom. The only warning is that it is a one-way upgrade - there is no way back |
| 00:01.23 | sevard | Hmmhesays |
| 00:01.24 | sevard | you here man |
| 00:01.32 | terrapen | cool, thanks CP |
| 00:02.10 | *** join/#asterisk jsaunders (n=root@216.86.121.58) |
| 00:02.14 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 00:02.16 | CunningPike | Back in the days of SIP 1.5.x, there were incompatabilities between the SIP 1.5.x and the 3.x bootrom, but everyone should be on SIP 1.6.x by now |
| 00:02.17 | terrapen | the wiki says: |
| 00:02.17 | terrapen | If you do not require one of these boot protocols, DO NOT upgrade to BR 3.x and instead stick with BR 2.6.1. |
| 00:02.25 | CunningPike | It's ood |
| 00:02.31 | MrDigital | whats a good softphone? |
| 00:02.40 | CunningPike | And doesn't apply to SIP 1.6.x |
| 00:02.43 | jsaunders | is there a way "reload" zap changes w/out restarting *? |
| 00:02.51 | dlynes_ | MrDigital: windows or linux? |
| 00:02.53 | CunningPike | MrDigital: SJphone I like |
| 00:03.10 | MrDigital | windows |
| 00:03.11 | jsaunders | Like a SIGHUP or somethin'? |
| 00:03.33 | dlynes_ | MrDigital: Try snom360 (www.snom.de) |
| 00:03.47 | dlynes_ | MrDigital: there's others, too, but i've had the most luck with that one |
| 00:04.05 | dlynes_ | MrDigital: it worked the first time, out of the box |
| 00:04.58 | dlynes_ | CunningPike: you been to that new rice world yet? |
| 00:05.12 | dlynes_ | CunningPike: wicked prices, and good variety, too |
| 00:05.16 | CunningPike | On Garden City? |
| 00:05.20 | dlynes_ | yeah |
| 00:05.27 | CunningPike | No - I'm Irish |
| 00:05.30 | CunningPike | Spuds for me ;) |
| 00:05.33 | dlynes_ | lol |
| 00:05.48 | dlynes_ | It's a Chinese grocery store...it's not just rice |
| 00:05.59 | dlynes_ | super cheap meat there, to |
| 00:06.02 | dlynes_ | s/to/too |
| 00:06.08 | CunningPike | lol - here was me thinking there was a store just for rice |
| 00:06.20 | dlynes_ | Nah...it's called China World/Rice World |
| 00:06.34 | dlynes_ | same shop that's down at Gore and Cordova |
| 00:06.43 | dlynes_ | but much bigger |
| 00:06.52 | CunningPike | Haven't been into it - I don't do much grocery shopping |
| 00:07.01 | dlynes_ | ah |
| 00:07.17 | CunningPike | My darling wife does most of it |
| 00:07.25 | CunningPike | I buy beer and burgers :) |
| 00:07.26 | sevard | dlynes_: I'm 50% rebuilt |
| 00:07.39 | CunningPike | sevard: Phew - I thought you'd offed yourself |
| 00:07.44 | dlynes_ | sevard: cool |
| 00:07.50 | dlynes_ | sevard: did you recover your configs? |
| 00:07.52 | Weezey | what's the highest number of people you've had in a meetme conference? |
| 00:08.02 | sevard | dlynes_: can you help me with zapata.conf, Hmmhesays did it for me last time and I don't have any more money / can't get a hold of him |
| 00:08.05 | sevard | I did not recover my configs |
| 00:08.14 | sevard | i'm pulling the most valuable things out of my head |
| 00:08.26 | dlynes_ | sevard: what kinda card is it? |
| 00:08.29 | sevard | TDM400P |
| 00:08.40 | sevard | I still have my /etc/zaptel.conf |
| 00:08.41 | dlynes_ | how many fxo ports, and how many fxs ports? |
| 00:08.51 | sevard | four FXO(red) modules |
| 00:08.55 | dlynes_ | ah |
| 00:08.58 | dlynes_ | one sec |
| 00:09.00 | sevard | <3 |
| 00:09.04 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 00:09.14 | terrapen | <PROTECTED> |
| 00:09.14 | terrapen | oops |
| 00:10.19 | dlynes_ | sevard: try this: http://pastebin.ca/53045 |
| 00:10.50 | sevard | dlynes_: I will try that. the only thing i remembr from them is the gain levels we were talkign abou today :\ |
| 00:11.16 | dlynes_ | sevard: btw...the first thing you're going to do after you get all this up and running is what? |
| 00:11.28 | sevard | dlynes_: Back Up |
| 00:11.32 | Weezey | sevard: I still haven't got the gains correct |
| 00:11.39 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:11.40 | sevard | see the problems is i backed up, i just didn't back it up correctly ;\ |
| 00:11.41 | CunningPike | sevard: Good boy ;) |
| 00:11.48 | sevard | Weezey: I set both of my gains to 5 and it worked awesome |
| 00:11.55 | dlynes_ | sevard: :) |
| 00:12.00 | Weezey | really? i get lots of echo |
| 00:12.07 | dlynes_ | sevard: I've got it at -4 and it works perfectly |
| 00:12.33 | Weezey | if it's neg on mine it's too quiet |
| 00:12.49 | dlynes_ | Weezey: what kinda card? |
| 00:12.57 | Weezey | Zap 04b |
| 00:13.15 | sevard | I lost all my local prefixs :( |
| 00:13.17 | dlynes_ | clone, or digium board? |
| 00:13.19 | sevard | i spent a day getting those from carriers |
| 00:13.22 | Weezey | digium |
| 00:13.31 | dlynes_ | ah...yeah...mine's a digium x100p card |
| 00:13.38 | justinu | dlynes: gains all tuned right now? |
| 00:13.41 | Weezey | x100p 'eh? |
| 00:13.53 | dlynes_ | justinu: yep...you didn't see me thanking you a while back, I guess |
| 00:14.19 | justinu | good deal |
| 00:14.28 | dlynes_ | Weezey: yeah...works great, for the most part |
| 00:14.48 | dlynes_ | Weezey: just certain machines the driver refuses to load about 2 days after a fresh install |
| 00:15.56 | docelm0 | hay guys.. quick and dumb question.. to bind chan_sip to more than one IP you either A commment out the bindip= or use bindip=0.0.0.0? Cause I have tried both and they dont work.. What am I doing wrong/ |
| 00:16.04 | sevard | dlynes_: wait, i need phone.conf or something? |
| 00:16.37 | sevard | docelm0: I don' |
| 00:16.48 | sevard | docelm0: I don't know but in normal unix daemons 0.0.0.0 == all |
| 00:16.56 | jsaunders | restart when convenient |
| 00:17.00 | jsaunders | oops :D heheh |
| 00:17.02 | sevard | wait, yes i do know, use 0.0.0.0 |
| 00:17.05 | jsaunders | wrong console |
| 00:17.17 | dlynes_ | sevard: no idea...never used phone.conf |
| 00:17.34 | sevard | dlynes_: chan_zap failes to start * without a phone.conf |
| 00:18.16 | sevard | oh, apparently i also need dundi.conf which i have no clue about |
| 00:18.43 | dlynes_ | sevard: oh...just use the default file then |
| 00:18.47 | dlynes_ | sevard: dood |
| 00:18.55 | dlynes_ | sevard: edit your modules.conf file |
| 00:19.03 | justinu | friggen n00bs |
| 00:19.14 | justinu | 0:) |
| 00:19.34 | sevard | oh thank god /var/lib/asterisk/agi-bin still exists |
| 00:19.51 | sevard | yeah, i don't have a modules.conf |
| 00:20.04 | dlynes_ | well, if you give me ssh access and your root password, i'll fix the agi-bin problem for you |
| 00:20.10 | justinu | lol |
| 00:20.19 | sevard | dlynes_:agi-bin still exists |
| 00:20.25 | dlynes_ | exactly ;) |
| 00:20.26 | justinu | yeah - that's the problem |
| 00:20.29 | sevard | it wasn't overwritten |
| 00:20.29 | dlynes_ | I was gonna remove it for you :) |
| 00:20.33 | sevard | asshat! |
| 00:21.08 | *** part/#asterisk xcoyote (n=farfan@201.135.194.118) |
| 00:22.03 | *** join/#asterisk BadPacket (n=BadPacke@unaffiliated/badpacket) |
| 00:22.49 | dlynes_ | sevard: download the asterisk source code |
| 00:22.58 | *** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net) |
| 00:23.02 | dlynes_ | sevard: erm actually fix that |
| 00:23.11 | dlynes_ | sevard: look in /usr/lib/asterisk/modules |
| 00:23.35 | dlynes_ | sevard: add in [modules] at the top of a new modules.conf file |
| 00:23.47 | dlynes_ | then add autoload => yes |
| 00:24.03 | sevard | i just made it |
| 00:24.03 | dlynes_ | then add noload => modulename.so for every module in /usr/lib/asterisk/modules |
| 00:24.06 | sevard | i think i have all the modules i need |
| 00:24.11 | sevard | i'm fixing permissoin problems |
| 00:24.12 | sevard | permission* |
| 00:24.18 | dlynes_ | then change noloads to loads until you've got asterisk loading properly |
| 00:24.27 | sevard | that's what i'm doing |
| 00:24.34 | dlynes_ | make sure all the res_....so is at the beginning |
| 00:24.52 | dlynes_ | all codec_..., chan_..., app_.... have dependencies in the res_.... files |
| 00:25.11 | znoG | does anyone know how lucent digital phones work? |
| 00:25.28 | dlynes_ | znoG: similar to nortel digital phones |
| 00:25.52 | znoG | probably yeah :) i guess i can't make any use out of them with asterisk, eh? |
| 00:26.06 | znoG | being digital phones, they don't work with an ATA.. since the ATA does just that.. convert to digital signals :) |
| 00:26.22 | dlynes_ | znoG: you could if you have about $50,000 to spend on reverse engineering the protocol they use on the wire |
| 00:26.49 | sevard | why the heck does asterisk want to write to /var/run/asterisk.pid/ctl and not /var/run/asterisk/* |
| 00:26.50 | znoG | hehe |
| 00:27.02 | dlynes_ | znoG: adn then probably another $5000 or so to develop a box to provide that signal that asterisk can talk to |
| 00:27.05 | sevard | <PROTECTED> |
| 00:27.23 | dlynes_ | sevard: who knows...probably something fubar about your setup |
| 00:27.29 | znoG | dlynes_: could make a lot of people who use Lucent PBXs that want to switch to Asterisk a smoother move |
| 00:27.29 | dlynes_ | sevard: check your /etc/asterisk/asterisk.conf file |
| 00:27.49 | sevard | dlynes_: oh, that's missing too |
| 00:28.03 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:28.12 | dlynes_ | znoG: yeah, but for now, the best you can hope for is Dialogic boards (a crapshoot), or MCK's product that used to be Citel's product |
| 00:28.23 | dlynes_ | znoG: The old citel product is your best bet |
| 00:28.35 | znoG | what does the product do? |
| 00:28.59 | dlynes_ | znoG: Allows SIP devices to talk to digital phones if I remember correctly |
| 00:29.25 | dlynes_ | znoG: If you just want to front end the lucent pbx though, that's another story...you can already do that |
| 00:29.25 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 00:29.40 | dlynes_ | znoG: you don't need to be able to talk to the phones directly to do that |
| 00:29.50 | file | ha |
| 00:29.58 | file | I just found a NuFone network pen on my desk! |
| 00:30.08 | znoG | dlynes_: what do you mean by "front end" the lucent pbx? |
| 00:30.12 | dlynes_ | file: oh...thought you were gonna say your server was back online :) |
| 00:30.23 | file | pfft |
| 00:30.26 | dlynes_ | znoG: put asterisk in between the analog lines and the lucent pbx |
| 00:30.30 | docelm0 | What's wrong with NuFone |
| 00:30.42 | file | I found an octasic pen too! |
| 00:30.47 | dlynes_ | znoG: or the pri and the lucent pbx |
| 00:31.00 | znoG | dlynes_: yeah i do that already .. i use a few fxs lines from the lucent to a Digium TDM2400 card |
| 00:31.15 | file | you know - I don't go shopping for office supplies or even t-shirts, I just go to tradeshows/expos/and conferences! |
| 00:31.28 | dlynes_ | file: yeah, no kidding |
| 00:31.39 | dlynes_ | file: and all those promotional companies that mail you all those goofy pens |
| 00:32.30 | sevard | [chan_zap.so][May 2 19:32:18] WARNING[31783]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call |
| 00:32.34 | sevard | oh what the crap |
| 00:32.42 | dlynes_ | sevard: dood....do you not listen? |
| 00:32.45 | file | and 4 genband lanyards! |
| 00:32.53 | sevard | i try to listen |
| 00:32.56 | dlynes_ | sevard: i said make sure all the res_.....so files are loaded in your modules.conf file first |
| 00:33.10 | sevard | I totally missed when you said that |
| 00:35.20 | *** part/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 00:35.36 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 00:37.20 | sevard | great |
| 00:37.52 | terrapen | durrr stupid question....how can I send a user to voicemailmain when they call their own extension? |
| 00:38.54 | znoG | terrapen: you could do a GotoIf the caller ID = number dialed => voicemailMain |
| 00:39.28 | terrapen | yeah, that would probably work |
| 00:39.48 | terrapen | what i really would rather do is figure out how to map that Messages button on my Polycom phone to my voicemail extension |
| 00:40.04 | ManxPower | you can also just have an exten => a,1,Voicemailmain |
| 00:40.16 | znoG | infact it comes stock in the .ael file |
| 00:41.57 | terrapen | hrmm ok |
| 00:42.20 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 00:42.23 | Aurs | terrapen: put callerid in the conf |
| 00:42.27 | Aurs | on the polycom |
| 00:42.33 | Aurs | on the messages button |
| 00:42.34 | dlynes_ | sevard: so everything's working now? |
| 00:43.22 | sevard | no, it won't run |
| 00:44.07 | sevard | i can't find the error message |
| 00:44.08 | terrapen | manx, that didn't work |
| 00:44.31 | terrapen | aurs, i couldn't find anything in the manual for that |
| 00:46.00 | znoG | what I'd really like to find out is the full PAP2-NA XML file so I can setup remote provisioning |
| 00:46.03 | terrapen | ah, found it on the wiki, i think |
| 00:46.57 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 00:47.03 | Druken | hey guys... |
| 00:47.26 | *** join/#asterisk _-Jon-_ (n=jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 00:47.33 | _-Jon-_ | hey everyone |
| 00:47.51 | Druken | ok, who's the country buff here? i'm lookin for a song, it's called "your gone" but i don't remember who it's by |
| 00:48.01 | _-Jon-_ | Wonderinf if anyone knows what might cause this error: NOTICE[20082]: app_dial.c:1012 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 00:48.24 | sevard | Druken: how many country songs do you think that are out there with that title? |
| 00:48.28 | Druken | uhmmm... did you actually read the error? |
| 00:48.42 | _-Jon-_ | Yes |
| 00:48.44 | Druken | sevard: oh, i dunno, like 100 million? :) |
| 00:48.56 | _-Jon-_ | oh wait, i think i know why now :P |
| 00:48.57 | sevard | Druken: that number sounds like it's in the ballpark |
| 00:49.03 | Druken | does john micheal montgomery have one called that? |
| 00:49.28 | sevard | Druken: not sure who that even is.. i really dislike pop country |
| 00:49.37 | dlynes_ | sevard: instead of running safe_asterisk, run asterisk -vvvvvvvvvvvvvvg |
| 00:49.54 | sevard | dlynes_: I don't run safe asterisk |
| 00:50.12 | Druken | sevard: i dislike most country in general, but someone tried to send it to my earlier, but trillian didn't keep a record of the name... :( |
| 00:50.25 | dlynes_ | sevard: damn...you live life on the edge..you actually trust asterisk that much? |
| 00:50.38 | sevard | dlynes_: ha, no, i just hadn't gotten it to work yet |
| 00:50.55 | sevard | Druken: google for the lyrics |
| 00:51.10 | dlynes_ | I know the song he's talking about |
| 00:51.17 | dlynes_ | I'll be damned if I knwo who sings it though |
| 00:51.44 | dlynes_ | It's a pretty new song, too |
| 00:52.05 | dlynes_ | Druken: You get KIKX FM where you are, don't you? |
| 00:52.42 | dlynes_ | I think that's the Calgary country station, anyways |
| 00:52.46 | Druken | unfortunatly.... |
| 00:52.58 | Druken | calgary? |
| 00:53.06 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
| 00:53.08 | Druken | hell no... barrie, ontario... |
| 00:53.19 | Druken | i get KICX |
| 00:53.35 | dlynes_ | I used to live in that area |
| 00:53.38 | Druken | or something like that |
| 00:53.40 | dlynes_ | I'm so happy I moved out west |
| 00:54.30 | dlynes_ | I don't think I could spend another summer in Southern Ontario |
| 00:54.46 | sevard | i thought asterisk -vvvg was supposed to give me a cli |
| 00:54.49 | demigod2k | windsor is fun |
| 00:54.58 | Druken | dlynes: i like it here... |
| 00:54.59 | dlynes_ | sevard: it will if you don't get an error |
| 00:55.07 | demigod2k | I love going out to the bars in windsor, you can get away with anything |
| 00:55.10 | *** part/#asterisk phonic (i=phonic@antisocial.nu) |
| 00:55.14 | Druken | course, i've never been out west :) |
| 00:55.26 | dlynes_ | demigod2k: same thing for bars in Toronto...well...almost |
| 00:55.38 | dlynes_ | demigod2k: My brother's gotten kicked out of the Gasworks like ten times :) |
| 00:55.39 | demigod2k | ya toronto is probably better actually |
| 00:55.45 | demigod2k | windsor does have its limits |
| 00:55.51 | dlynes_ | each time was 'permanent' |
| 00:55.58 | demigod2k | last time I was there the strip clubs god busted with a no-touching penalty |
| 00:56.07 | demigod2k | windsor that is |
| 00:56.21 | dlynes_ | demigod2k: yeah...we have that policy in vancouver, too |
| 00:56.38 | dlynes_ | it was like that before i moved out here |
| 00:56.42 | dlynes_ | so I didn't cause it |
| 00:56.43 | dlynes_ | honest |
| 00:57.11 | Druken | uh huh.... |
| 00:57.30 | sevard | dlynes_: I don't get an error but I see my phones registering, I also don't get an input line though |
| 00:57.45 | dlynes_ | sevard: yeah, so youv'e got an issue somewhere |
| 00:57.46 | sevard | it says Asterisk Ready |
| 00:57.52 | dlynes_ | sevard: ah...then it works |
| 00:57.56 | sevard | dlynes_: mother fucker I have issues all over the place |
| 00:57.56 | dlynes_ | sevard: hit ctrl-c |
| 00:58.02 | dlynes_ | sevard: and then type safe_asterisk |
| 00:58.04 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:58.04 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:58.07 | Druken | welcome to asterisk :) |
| 00:58.10 | sevard | I don't have safe_asterisk :P |
| 00:58.13 | dlynes_ | sevard: then do an asterisk -r |
| 00:58.20 | dlynes_ | sevard: /usr/sbin/safe_asterisk |
| 00:58.25 | sevard | omg that worked! |
| 00:58.29 | sevard | WTF |
| 00:58.35 | dlynes_ | sevard: you're probably running some version of slackware that doesn't include /usr/sbin in your path |
| 00:59.22 | sevard | dlynes_: I know about paths and slax doesn't include /usr/sbin/ for normal users, wha tI would do with my old version was su asterisk - ; /usr/sbin/asterisk |
| 00:59.30 | sevard | I never used safe_asterisk before |
| 01:00.02 | Druken | oh for fuck sakes!, my god damn dvd player program associated mp3's.... |
| 01:00.11 | dlynes_ | sevard: yeah, but with root user on some slackware releases, /usr/sbin wasn't in the path, either |
| 01:00.16 | demigod2k | I hate when programs do that without asking |
| 01:00.22 | Druken | why do asshole programmers make their shit do EVERYTHING! |
| 01:01.17 | sevard | Druken: are you using windows? |
| 01:01.29 | Druken | i do for my work station, yes |
| 01:02.10 | russellb | well there's your problem! |
| 01:02.45 | *** join/#asterisk kavit (n=kavit@210-84-40-39.dyn.iinet.net.au) |
| 01:03.07 | Weezey | anyone a meetme expert? |
| 01:03.24 | MikeJ[Laptop] | no one |
| 01:03.25 | Druken | aside from it's a pain in the ass? |
| 01:03.34 | russellb | Druken: oh shush you |
| 01:03.39 | Druken | oh wait, that's asterisk in general :) |
| 01:03.49 | russellb | you're just bitter because you're using windows |
| 01:04.17 | Druken | russellb: hehe blow me :) i'd never be stupid enough to attempt servers on windows |
| 01:04.31 | sevard | alright, now dtmf is only reading two digits |
| 01:04.37 | justinu|laptop | Druken: lol |
| 01:04.46 | *** join/#asterisk I-MOD (i=opticron@68.62.165.168) |
| 01:04.50 | russellb | Druken: yeah, i was just trolling |
| 01:04.52 | demigod2k | or stupid enough to not use windows on the desktop |
| 01:04.57 | demigod2k | everything runs on windows :/ |
| 01:05.09 | Druken | basically |
| 01:05.19 | Druken | and everyONE can use windows... |
| 01:05.24 | russellb | let's not start an OS war, here |
| 01:05.39 | MikeJ[Laptop] | no? |
| 01:05.41 | MikeJ[Laptop] | ;) |
| 01:05.47 | Druken | OS/X baby :) hahahahahahaha |
| 01:05.48 | [hC] | OSX Rules! |
| 01:05.51 | [hC] | brb. |
| 01:05.51 | [hC] | :) |
| 01:05.54 | demigod2k | I have a mac. it just wont run shit :/ |
| 01:06.02 | demigod2k | at least it was a cute $500 collectible |
| 01:06.02 | MikeJ[Laptop] | I will take the side of totally cross platform |
| 01:06.16 | kavit | OSX Rules with the whip of vendor lockin..... |
| 01:06.41 | MikeJ[Laptop] | yay trolls... |
| 01:06.47 | demigod2k | I dont feel vendor lockin so much on my mini as lack of software |
| 01:06.50 | MikeJ[Laptop] | speaking of... where is RoyK? |
| 01:06.52 | demigod2k | no CAD, no engineering, no nothing |
| 01:06.53 | dlynes_ | sevard: just adjust your rxgain |
| 01:06.55 | Druken | hehe i remember people and os/2 |
| 01:07.04 | dlynes_ | Druken: dood...os/2 rules you |
| 01:07.20 | tainted- | lol os/2 |
| 01:07.28 | tainted- | that was a good platform |
| 01:07.32 | Druken | os/2 was good for multitasking back in the day |
| 01:07.32 | SwK | TRS-DOS RULES j00r A55! |
| 01:07.37 | [hC] | i remember installing os/2 form like... 68 floppies |
| 01:07.40 | dlynes_ | heh...i gave up on it before os/2 v5 came out though |
| 01:07.46 | dlynes_ | ibm gave up on it, so i gave up on it |
| 01:07.51 | demigod2k | haha 68 floppies |
| 01:07.57 | dlynes_ | that's when i started using linux more seriously |
| 01:08.02 | [hC] | praying that like #59 wasnt bad |
| 01:08.11 | tainted- | lol |
| 01:08.14 | kavit | dlynes_: you mean os/2 rules us posthumously |
| 01:08.21 | dlynes_ | lol |
| 01:08.21 | justinu|laptop | os/2 was interesting... anyone have a vmware image or iso images of that? |
| 01:08.29 | Druken | i remember downloading slack, took 2 days to download, and 12 hours to install |
| 01:08.31 | SwK | shit Netware used to sit on like 30 or 40 5.25" 360K floppies |
| 01:08.39 | tainted- | wow people trade vmware images now? |
| 01:08.40 | dlynes_ | justinu: I've still got originals for 2.0, 3.0, and 4.0 |
| 01:08.48 | tainted- | what was warp |
| 01:08.49 | tainted- | 3.0? |
| 01:08.52 | justinu|laptop | 3.0 was warp |
| 01:08.55 | dlynes_ | tainted-: 3.0 |
| 01:08.58 | kavit | first boot dos and then load netware :( |
| 01:08.59 | dlynes_ | tainted-: 4.0 was merlin |
| 01:09.08 | tainted- | yea warp was nice |
| 01:09.08 | dlynes_ | tainted-: 5.0 was server only...not sure what ibm called it |
| 01:09.14 | tainted- | they still run in in some industries |
| 01:09.22 | dlynes_ | 2.0 was buggy as all hell |
| 01:09.24 | tainted- | saw a atm 'bluescreen' some os/2 msg |
| 01:09.35 | dlynes_ | but you could still get more done in 2.0 with it crashing all the time than you could in windows 3.1 |
| 01:09.45 | SwK | OS/2 4.0 drives a stack of older ATMs |
| 01:09.51 | Druken | hehehe win 3.0 rawked! |
| 01:09.54 | sevard | dlynes_: i'm playing with the gain levels and I'm not geting any results |
| 01:09.57 | sevard | the same thing each level |
| 01:10.11 | Nugget | heh, I just got a random call from some guy who accidently dialed SIP:nugget@slacker.com because hsi dialplan was screwed up or something. |
| 01:10.26 | Druken | Nugget: hehe |
| 01:10.33 | Nugget | I saw the callerid showed "103@10.0.0.3" on my cisco |
| 01:10.37 | Nugget | nutty asterisk. |
| 01:10.42 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 01:10.55 | Nugget | he thought he was calling his girlfriend or something. |
| 01:11.18 | [hC] | Nugget: maybe you have a bigger problem on your hands! :P |
| 01:11.24 | dlynes_ | sevard: you remember it's from -100 to +100, right? |
| 01:11.30 | kavit | hey honey, come tame the wilderbeast ooops sorry wrong number |
| 01:11.31 | kavit | :(\ |
| 01:11.34 | dlynes_ | sevard: adjust it by 4 each time |
| 01:11.38 | sevard | you said -7 to +7 |
| 01:11.45 | sevard | I'm using a TDM400P |
| 01:11.50 | dlynes_ | sevard: yeah...I found out from justin that it's -100 to 100 |
| 01:12.04 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 01:12.11 | *** join/#asterisk kaz0358 (n=kurtzogl@asterisk.telecom.ksu.edu) |
| 01:12.14 | sevard | Last time Hmmhesays had set it to 12.0, i changed them both to 5.0 and it worked awesomely |
| 01:12.14 | Druken | ok, new movie just came off, later ppls |
| 01:13.01 | tainted- | Nugget pick up |
| 01:13.03 | tainted- | lol |
| 01:13.07 | Nugget | I did. I didn't hear you. |
| 01:13.17 | justinu|laptop | the miracle of voip |
| 01:13.22 | Nugget | indeed. |
| 01:13.42 | OloBola | what do you think a good per minute price is for 800 origination? 3centsish? 2.5ish? |
| 01:13.47 | Nugget | besides, why would I want to talk to "polycom"? |
| 01:13.49 | kavit | imagine if this was an emergency, imagine if tainted- was out of beer or something :( |
| 01:13.53 | dlynes_ | OloBola: free |
| 01:14.01 | tainted- | what! |
| 01:14.04 | tainted- | polycom's are the best |
| 01:14.09 | Nugget | OloBola: be daring -- price it in rubles or something. |
| 01:14.14 | sevard | OloBola: Charge .3 cents and i'll buy |
| 01:14.23 | kaz0358 | kinda strange question, but if you get a sip 300 redirect and it wording to Local/original-extension@original-context.. umm.. what does the channel local really mean? i have sipbroker as a part of my dialplan and if it fails the enum lookup, then it just continues down the dial plan.. put it is weird to see that message. its like it is redirecting back to my own box as "local" |
| 01:14.24 | tainted- | .3 cents? |
| 01:14.26 | tainted- | that's cheap |
| 01:14.55 | OloBola | free? I'll take free (thank you). |
| 01:14.57 | dlynes_ | kaz0358: asterisk is redirecting it to local, not the phone |
| 01:15.09 | dlynes_ | kaz0358: the phone is saying redirect to blehblehbleh |
| 01:15.09 | MrDigital | how do i setup xten for asterisk? its saying unauthorized |
| 01:15.23 | dlynes_ | kaz0358: asterisk translates that into local/... because your dialplan tells it to |
| 01:15.42 | dlynes_ | MrDigital: your username and/or password doesn't match |
| 01:15.52 | kaz0358 | dlynes, where in the dial plan am i specifying "local"? |
| 01:15.53 | MrDigital | can the pass be letters? |
| 01:15.58 | dlynes_ | MrDigital: or you're not allowed to access that sip resource from the given ip address |
| 01:16.10 | dlynes_ | kaz0358: no idea...I don't have a copy of your dialplan in front of me |
| 01:16.17 | dlynes_ | MrDigital: yes |
| 01:16.25 | *** join/#asterisk phonic (i=phonic@antisocial.nu) |
| 01:17.11 | phonic | when i try to make a call with dial(), i get the error Protocol error layer 1 (broken line or B-channel removed by signalling protocol). somebody who knows a solution? |
| 01:17.23 | kaz0358 | dlynes, well i have a "Local", but i thought contexts were case sensitive.. and the device that is making the call is in the "longdistance" context |
| 01:17.34 | MrDigital | the login is right still unauthrozied |
| 01:18.08 | sevard | dlynes_: dude, this isn't working at all |
| 01:18.33 | dlynes_ | sevard: so fix it, then |
| 01:18.44 | kaz0358 | dlynes, i meant to say i have a "local" context and not a "Local" context |
| 01:18.45 | justinu|laptop | haha |
| 01:19.05 | sevard | it only fricken reads two digits |
| 01:19.22 | dlynes_ | maybe you're only typing two digits |
| 01:19.31 | sevard | i'm putting in four |
| 01:19.34 | sevard | it reads one or two digits |
| 01:19.50 | dlynes_ | Did you do an answer, then Wait(2), and then do your ivr? |
| 01:20.01 | sevard | yes |
| 01:20.23 | dlynes_ | relaxdtmf=yes is set in your zapata.conf? |
| 01:21.06 | sevard | yes |
| 01:21.15 | dlynes_ | try changing it to no |
| 01:21.21 | dlynes_ | and then restarting asterisk |
| 01:21.40 | *** join/#asterisk gezick (n=gezick@c-68-50-25-85.hsd1.dc.comcast.net) |
| 01:22.16 | gezick | how tough would it be to set up streaming with asterisk? e.g. you call into a number, put in your extension, then fire up itunes and listen to that person on an mp3 stream |
| 01:22.45 | dlynes_ | gezick: take a look at the sample musiconhold.conf file |
| 01:22.49 | sevard | same results. |
| 01:22.59 | dlynes_ | sevard: now that you've changed that |
| 01:23.04 | dlynes_ | sevard: set both gains at 0 |
| 01:23.08 | sevard | alright |
| 01:23.10 | gezick | dlynes_: i mean in the other direction |
| 01:23.13 | dlynes_ | sevard: and try bring them up by 4 each time |
| 01:23.20 | gezick | take the audio that comes in over the telephone and stream it out |
| 01:23.22 | dlynes_ | sevard: restarting asterisk in between each change |
| 01:23.33 | dlynes_ | gezick: you mean paging?? |
| 01:23.53 | gezick | i don't understand what you mean by that |
| 01:23.57 | znoG | sevard: you shouldn't have to change gain levels with a TDM400.. i never had to |
| 01:24.17 | dlynes_ | gezick: look up on www.voip-info.org for the ALSA channel driver |
| 01:24.25 | dlynes_ | gezick: that's probably what you want |
| 01:24.47 | sevard | same results. |
| 01:24.51 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 01:24.54 | dlynes_ | sevard: doood |
| 01:24.56 | sevard | znoG: May I see your zapata.conf |
| 01:25.12 | dlynes_ | you couldn't have possibly done 0->100 and 0->-100 that fast and restarted asterisk every time |
| 01:25.20 | sevard | no, i'm doing that now |
| 01:25.39 | znoG | sevard: i don't have it here unfortunately, they're on the work servers which I can't ssh to from here |
| 01:26.36 | dlynes_ | sevard: if you want, i can try sshing in to see what's happening |
| 01:27.07 | gezick | dlynes_: i think i kinda want that, but i want it to go out to icecast |
| 01:27.14 | gezick | rather than to a sound card |
| 01:27.28 | dlynes_ | gezick: so write your own channel driver then |
| 01:27.57 | dlynes_ | gezick: there's a skeleton driver in the asterisk source code |
| 01:28.06 | gezick | dlynes_: is there a limit to the size of wav file that i can record to? |
| 01:29.42 | gezick | and can i record to a socket, because i'm thinking that that's what makes the most sense here |
| 01:30.04 | dlynes_ | gezick: i wouldn't have a clue |
| 01:30.11 | dlynes_ | gezick: never done anything like that before |
| 01:30.27 | dlynes_ | gezick: I'm planning to do paging eventually, but i'm not doing it yet |
| 01:30.35 | gezick | i should think that i could, since a socket is just a file... |
| 01:30.46 | gezick | rather, just like a file |
| 01:30.56 | dlynes_ | you mean a unix domain socket? |
| 01:31.30 | justinu|laptop | hahaha |
| 01:37.20 | ManxPower | ~docs |
| 01:37.26 | jbot | docs is, like, 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 |
| 01:37.26 | ManxPower | ~thebook |
| 01:37.27 | jbot | i guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 01:38.37 | justinu|laptop | ~suggestions |
| 01:38.39 | jbot | suggestions is, like, 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite and ... |
| 01:38.59 | file | and never end a sentence with ... |
| 01:40.20 | ManxPower | file, why not ... |
| 01:40.42 | znoG | anyone have a working 1.2.4 installation and upgraded to 1.2.7.1 and things don't work as well? |
| 01:42.14 | file | because ... |
| 01:44.30 | mds2 | I have a Digium TDM2400XXP which regularly stops talking to the analog lines on its FXO ports. Asterisk claims to be picking up the channel for outbound calls but nothing connects. No ring indication for inbound calls. Asterisk 1.2.6, Zaptel 1.2.5, kernel 2.4.32-grsec. Reboot/power cycle fixes the problem for a day or two. Any ideas? |
| 01:44.38 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 01:45.36 | xachen | ... makes me want to kill people unless justly used |
| 01:47.07 | file | ... o rly? ... |
| 01:47.08 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 01:47.45 | TheCops | Someone know if register server for g729 licensing is down for Digium ? |
| 01:51.21 | MrDigital | <PROTECTED> |
| 01:54.24 | Nivex | MrDigital: ouch. start killing off unused processes, bounce daemons, or you may even have to reboot |
| 01:54.38 | *** join/#asterisk trig_hm (i=jason@home.monkeypr0n.org) |
| 01:58.51 | demigod2k | there is a racoon in my chimney |
| 01:58.52 | demigod2k | this sucks |
| 01:59.32 | justinu|laptop | smoke it out |
| 01:59.45 | demigod2k | dont want to open the flue |
| 01:59.51 | demigod2k | it's still in the chimney not the house |
| 01:59.55 | justinu|laptop | heh |
| 02:00.17 | *** join/#asterisk anthm (n=anthm@CPE-69-76-83-52.wi.res.rr.com) |
| 02:00.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:01.46 | *** part/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 02:02.40 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 02:06.30 | kaz0358 | <PROTECTED> |
| 02:08.34 | file | eep it's [TK]D-Fender |
| 02:08.57 | Snake-Eyes | Any one know why grandstream gxp-2000 will take 40-30secs to kill/hang up over the linksys sp941 that hangs up right away? The phone says the call has ended yet the other party doesnt get the d/c signal for another 40-30sec |
| 02:09.30 | [TK]D-Fender | :O |
| 02:10.53 | xachen | demigod2k: Start a fire and open the flue :O |
| 02:11.25 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 02:15.02 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 02:15.22 | demigod2k | i dont have solid fireplace doors though :( |
| 02:15.51 | *** join/#asterisk mog_home (n=achika54@68.62.237.103) |
| 02:16.25 | xachen | :P |
| 02:16.47 | *** join/#asterisk Samoied (n=Samoied@200.175.75.225.adsl.gvt.net.br) |
| 02:18.49 | justinu|laptop | i wonder what would happen if you discharged a 12ga shotgun into your fireplace |
| 02:19.01 | demigod2k | it'd damage the brick |
| 02:19.05 | justinu|laptop | that sucks |
| 02:19.06 | demigod2k | I'm going to try mothballs |
| 02:19.13 | demigod2k | I've got a sheet of plywood in front of it which should help |
| 02:19.24 | demigod2k | hopefully it either climbs out or dies |
| 02:19.33 | demigod2k | because I dont really care to go pick up mothballs tonight |
| 02:19.33 | justinu|laptop | a .22LR rifle would do the trick |
| 02:19.58 | *** join/#asterisk The_Isle_of_Mark (n=mark@c-68-85-63-96.hsd1.ga.comcast.net) |
| 02:20.06 | The_Isle_of_Mark | why does asterisk take so long to ring an extension? |
| 02:20.59 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 02:21.05 | websae | Perhaps due to the way you have it configured |
| 02:21.37 | The_Isle_of_Mark | websae, default install and default sip settings |
| 02:23.03 | SwK | mothballs run everything off |
| 02:23.32 | [TK]D-Fender | The_Isle_of_Mark : It rings as long as YOU tell it to. |
| 02:24.05 | The_Isle_of_Mark | [TK]D-Fender, it takes a full 10 seconds to ring an extension on an inbound pstn call |
| 02:24.52 | *** join/#asterisk brockj49464 (n=chatzill@63.87.56.153) |
| 02:24.57 | [TK]D-Fender | The_Isle_of_Mark : Have you examined what it is you are doing, and watched the CLI as the call is being processed? |
| 02:26.07 | The_Isle_of_Mark | [TK]D-Fender, I have, but I don't know what they mean. Perhaps a link to a decent TFM would be helpful. Do you have one? |
| 02:26.17 | The_Isle_of_Mark | docs are very convoluted |
| 02:26.20 | [TK]D-Fender | ~thebook |
| 02:26.28 | jbot | i heard thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 02:26.52 | [TK]D-Fender | The_Isle_of_Mark : I guess the first question is : did you make your config yourself pretty much by hand? |
| 02:27.21 | [TK]D-Fender | "thebook" is the best beginners refernce, and when you're ready for jsut raw lists of commands, then there's the WIKI |
| 02:27.22 | [TK]D-Fender | ~docs |
| 02:27.25 | jbot | methinks 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 |
| 02:27.38 | The_Isle_of_Mark | [TK]D-Fender, nope, I didn't...I used amp...but that book is published by oreilly...right on! |
| 02:29.33 | *** join/#asterisk naS_- (n=andrew@182.136.233.220.exetel.com.au) |
| 02:31.06 | [TK]D-Fender | The_Isle_of_Mark : What kind of phones (soft/hard), and connectivity (analog/digital line equipment, VoIP providers, etc) do you have now? |
| 02:32.20 | The_Isle_of_Mark | [TK]D-Fender, using generic 2 zap interfaces and a sip trunk for trunking. Using 2 pap2-na ATAs for extensions |
| 02:32.20 | *** join/#asterisk gursikh (n=guriskh1@adsl-68-92-63-134.dsl.hstntx.swbell.net) |
| 02:32.32 | The_Isle_of_Mark | the sip is vonage |
| 02:32.40 | [TK]D-Fender | The_Isle_of_Mark : Excellent starting euipment. |
| 02:33.07 | [TK]D-Fender | Minus Vonage.... taht goes into one of the Zap cards, doesn't it? |
| 02:33.12 | The_Isle_of_Mark | seriously? I just found what would work and started testing lol |
| 02:33.16 | *** join/#asterisk _Sam-- (n=sam@mail.kneedraggers.com) |
| 02:33.26 | The_Isle_of_Mark | [TK]D-Fender, yeah it does |
| 02:34.00 | The_Isle_of_Mark | [TK]D-Fender, I use a european provider to allow cheap (free) calls to family in germany and holland |
| 02:35.30 | The_Isle_of_Mark | everything works fine outgoing, but it takes a long time incoming to ring the extension...once I figure that out I'll setup distinctive ring headers (wherever the hell I do that) and a proper voice mail |
| 02:36.04 | [TK]D-Fender | Ok, if everything is working now, take a GOOD look at how the provider is set up in sip.conf. Do the same for your zapata, zaptel, and then prepare to FLUSH extensions.conf and learn to build from scratch. |
| 02:36.07 | anthm | btw, [TK]D-Fender i granted you wish for chanspy http://bugs.digium.com/view.php?id=7072 |
| 02:36.19 | [TK]D-Fender | Which incoming is slow? |
| 02:36.33 | The_Isle_of_Mark | [TK]D-Fender, the same either way...pstn or sip |
| 02:36.39 | *** join/#asterisk CodyC (n=cody@cpe-70-112-210-245.austin.res.rr.com) |
| 02:37.00 | [TK]D-Fender | anthm : So you set before placing a call and thats it? |
| 02:37.37 | [TK]D-Fender | anthm : Nifty, not quite the approach I was thinking about, but more versitile. I was thinking in the channel definition actually, but this IS more fine tuned. |
| 02:37.51 | anthm | same as before but you can : them together so you can be in more that 1 group |
| 02:40.38 | [TK]D-Fender | anthm : Yup, added versatility for sure, but more occurances, but better per call control on a given interface ratch that fixed |
| 02:42.06 | naS_- | can anybody assist with a question on iax2 trunking using g729 codec? |
| 02:43.11 | [TK]D-Fender | naS_- : Just ask |
| 02:43.21 | *** part/#asterisk Samoied (n=Samoied@200.175.75.225.adsl.gvt.net.br) |
| 02:45.02 | naS_- | ok I have 2 asterisk boxes and I have setup an IAX2 trunk. asterisk shouldn't be doing any transcoding as the ip phones connected to each asterisk box are using g729 codec. As I understand this should be working in pass through mode and not using g729 licenses, however, I can see that they are (I purchased 2 on each side for testing). Have I got something wrong or is this how it works? |
| 02:45.14 | [TK]D-Fender | anthm : You know what I think would be SERIOUSLY useful? being able to define channel vars/constants in the channel declaration for processing during exectution of the dial-plan. |
| 02:45.43 | [TK]D-Fender | anthm : Similar to how you do in .call files. |
| 02:47.07 | [TK]D-Fender | anthm : And be able to reference like GotoIf($[${CHANVAR(SIP/123/myvariable)}=123 ]?5) |
| 02:47.38 | tainted- | that's fugly |
| 02:48.02 | [TK]D-Fender | anthm : Good for processing tech-specific features to things like Dial (tT for XLite, etc but not for REAL SIP phones) |
| 02:48.36 | The_Isle_of_Mark | [TK]D-Fender, where do I put alert_info for distinctive ring? I find a lot of pages referring to it but no instructions on which file it goes in..or even an example file |
| 02:48.39 | [TK]D-Fender | tainted- : Its the cleanest "easy" way I can imagine. I mean you wouldn't want to mangle ASTDB to do taht I would think. |
| 02:48.57 | [TK]D-Fender | The_Isle_of_Mark : Depends on the phone's method of indicating it. |
| 02:49.05 | tainted- | yea |
| 02:49.06 | anthm | CHANVAR is a proposed function ? |
| 02:49.10 | The_Isle_of_Mark | [TK]D-Fender, same as sipura |
| 02:49.14 | [TK]D-Fender | The_Isle_of_Mark : And if you're referring to the PAP2-NA, I could tell you. |
| 02:49.28 | [TK]D-Fender | The_Isle_of_Mark : Never tried ti personally... I'd say check the WIKI on that one. |
| 02:49.31 | tainted- | seems like dialplan is moving towards AGI type functionality |
| 02:50.14 | [TK]D-Fender | anthm : Yes, just an idea. more largely useful if we are able to add vars in a similar way as .call files do. |
| 02:50.44 | The_Isle_of_Mark | Anyone else have any GOOD info on the alert_info for distinctive ring on the pap2-na? |
| 02:50.49 | [TK]D-Fender | anthm : but imaging being able to mull the mailbox var from the cahnnel being dialed without having to hardcode it in a macro, etc? |
| 02:50.53 | [TK]D-Fender | pull* |
| 02:51.10 | dlynes_ | The_Isle_of_Mark: have you checked the documentation? |
| 02:51.12 | [TK]D-Fender | The_Isle_of_Mark : Oh, DO please try to look it up, just a litte, ok? :) |
| 02:51.58 | dlynes_ | The_Isle_of_Mark: i.e. the Sipura 2000 User's Guide, the Sipura 2000 Administrator's Guide? |
| 02:52.04 | sevard | AUDIO!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 02:52.12 | sevard | wow msn you have failed me |
| 02:52.22 | *** join/#asterisk kamileon (n=kamileon@68.62.190.253) |
| 02:52.24 | The_Isle_of_Mark | [TK]D-Fender, heheh ok since you asked nicely. I really don't have a lot of time to devote to this. I apologise. |
| 02:52.54 | [TK]D-Fender | The_Isle_of_Mark : You have no idea the restraint I've shown already from the moment you even MENTIONED AMP :) |
| 02:53.15 | [TK]D-Fender | The_Isle_of_Mark : Truely phenominal, let me just say... |
| 02:53.40 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 02:54.18 | The_Isle_of_Mark | I gotcha, amp was just a way to get up and running. I was hoping to tweak from the command line. features are tough in Asterisk I have found...and I am no command line newbie! |
| 02:54.33 | dlynes_ | The_Isle_of_Mark: I'm sure everyone on this channel has less time to devote to it than the person that needs it |
| 02:54.55 | tainted- | [TK]D-Fender what i've done is take the accountcode associated w/ sip user, and use agi to query db for variables |
| 02:55.27 | anthm | still not sure what the CHANVAR func does can you reitterate |
| 02:55.57 | [TK]D-Fender | tainted- : Yeah, that is sorta nice, but I was thinking of it more for constants that presistant data exp ones requiring ASTDB setup prior. |
| 02:56.20 | [TK]D-Fender | This way you can just port over a sip.conf with basic vaars to a new box and have a more function dial-plan follow it. |
| 02:58.03 | demigod2k | just FYI if you're having difficulties, I can recommend the VS-1 off-the-shelf server |
| 02:58.04 | [TK]D-Fender | anthm : Picture being able to add something like "set=var=value1" etc in channel definitions (SIP phones, zap channels, etc). Then when a call is placed by that channel the dialplan can pull that value out by use of the function. So for an x-lite softphone you could do "set=dialoptions=tT" and just add that do the dial-line and it'll be there for THAT phone and not others. |
| 02:58.09 | demigod2k | easy to configure, easy to setup,e tc |
| 02:58.25 | [TK]D-Fender | demigod2k : VS1 is an OLD EPIA platform isn't it? |
| 02:58.37 | demigod2k | no idea. I bought it for my office, relatively happy with it |
| 02:58.48 | demigod2k | I'm only running 4 lines |
| 02:58.55 | dlynes_ | [TK]D-Fender: vs1 is the xorcom asterisk server |
| 02:59.06 | demigod2k | you probably need more if you're planning to run a whole T1 but its ok. www.thevoipconnection.com |
| 02:59.23 | anthm | cant you already do that ? |
| 02:59.27 | anthm | i think i invented it |
| 02:59.35 | anthm | setvar= lines in peers |
| 03:01.07 | [TK]D-Fender | anthm : I may have missed it.... I know its in the .call files... |
| 03:01.36 | [TK]D-Fender | anthm : But I wouldn't want it restricted to just one tech/type (peer, friend, user) |
| 03:01.43 | [TK]D-Fender | anthm : Abstraction is my game. |
| 03:04.22 | anthm | there is abstraction and then there is getting asterisk to work |
| 03:05.02 | anthm | i can show you some abstraction that will being tears to your eyes but that dialplan shit can only be tweaked to a certian extent |
| 03:05.25 | anthm | I am the inventor of the entire concept of diaplan functions so i know firsthand =p |
| 03:06.27 | [TK]D-Fender | anthm : Very cool to hear... I can't code anything practical so I pass on what I can in teaching its USE :) |
| 03:07.02 | [TK]D-Fender | anthm : Do you thing that a function is indeed a good way to implement this? |
| 03:07.32 | gursikh | OMG i wish there wasn't like a bajillion VOIP providers, none of which A) Provide what i'm looking for or B) Have a decent reputation in working with * |
| 03:08.11 | [TK]D-Fender | gursikh : Good, Cheat, Fast.... choose two :) |
| 03:08.25 | gursikh | Exactly. |
| 03:08.39 | gursikh | 'Cheat=Cheap i'm assuming |
| 03:08.44 | xachen | hehe |
| 03:08.49 | anthm | well i had something like that at one point |
| 03:08.54 | [TK]D-Fender | "All ITSP's suck, some less than others in varying intervals" |
| 03:08.56 | anthm | it was a globalized hash |
| 03:08.56 | [TK]D-Fender | correct |
| 03:08.59 | xachen | how about DID guarantee? :O |
| 03:09.02 | xachen | then you can pick 3! |
| 03:09.07 | *** join/#asterisk triple-e (n=piaergnj@adsl-70-128-78-22.dsl.stlsmo.swbell.net) |
| 03:09.17 | [TK]D-Fender | xachen : I think its referred to as DID "wishful thinking" :) |
| 03:09.24 | xachen | naa |
| 03:09.31 | xachen | more like I wanna make sure my DID doesn't die for 3 weeks |
| 03:09.42 | *** join/#asterisk CANO-1982 (i=alejandr@190.48.70.192) |
| 03:10.10 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 03:11.16 | anthm | i had it and they nuked it |
| 03:11.49 | anthm | i also had these dialplan alises that they nuked over some pittly shit about the naming convention |
| 03:13.37 | [TK]D-Fender | anthm : *sigh* Feel the SUPPORT.... |
| 03:13.41 | anthm | http://bugs.digium.com/view.php?id=4323 |
| 03:14.11 | anthm | that one is actually really nifty the clownsil vetoed my for not wanting to recode it thier way |
| 03:14.56 | justinu|laptop | :\ |
| 03:15.24 | [TK]D-Fender | anthm : Actually thats frightening close to implementing FUNCTIONS as a dial-plan entity..... |
| 03:16.13 | anthm | it was indended to hide ugly ass combos of vars and functions |
| 03:16.25 | anthm | and dummy them down to little macros |
| 03:16.45 | [TK]D-Fender | anthm : Picture [function-mycustomfunc] in the same style as [macro-justamacro] with a return value... |
| 03:16.53 | anthm | it truly is a macro but they already have something called macro that of corse is not a real macro at all that took the name |
| 03:17.13 | [TK]D-Fender | anthm : the language that ALMOST was :) |
| 03:17.13 | justinu|laptop | heh |
| 03:17.20 | justinu|laptop | that's rediculous |
| 03:17.31 | [TK]D-Fender | Means to ends.. thats all we ask for. |
| 03:17.41 | anthm | well they also didnt like that it was self sufficient |
| 03:17.44 | anthm | an addon |
| 03:17.54 | anthm | that you could opt out of by not loading it |
| 03:18.25 | anthm | they wanted me to add it to the core instead and i felt i had done enough by inventing it in the first place and giving it to them for free |
| 03:18.27 | justinu|laptop | why? |
| 03:18.29 | [TK]D-Fender | anthm : That to me is the HIGHLIGH of a "feature". |
| 03:18.30 | anthm | so they closed it =D |
| 03:18.44 | justinu|laptop | anthm: that's interesting... |
| 03:18.46 | [TK]D-Fender | And we have the choice to do what we will. |
| 03:19.04 | anthm | i think you can probably install that the way it is in the bug and it works |
| 03:19.17 | anthm | cept they probably changed the names of the functions for fun along the way |
| 03:19.32 | justinu|laptop | lol |
| 03:20.18 | [TK]D-Fender | I still say we should implement my proudest acheivement in programming..... ILLOGICAL operators! Y = MAYBE(X) ! |
| 03:20.45 | justinu|laptop | uncertaintly? |
| 03:20.54 | justinu|laptop | uncertainty, that is |
| 03:20.59 | blitzrage | I wouldn't mind a macro that would return a value :) |
| 03:21.08 | blitzrage | instead of me adding it in manually |
| 03:22.13 | triple-e | hey --- gotta perplexing problem with iax trunking between two box's |
| 03:22.32 | triple-e | trunk is registering in both directions |
| 03:22.49 | triple-e | both box's have outbound routes to use the defined trunk |
| 03:23.33 | triple-e | but i get "Dial failed due to CHANUNAVAIL" on one side and "Dial failed due to CONGESTION" on the other side |
| 03:23.58 | anthm | oh yeah |
| 03:24.01 | anthm | here it is |
| 03:24.06 | anthm | more from the trash heap |
| 03:24.06 | *** join/#asterisk annonimous (n=annonimo@201.137.44.154) |
| 03:24.07 | anthm | http://www.freeswitch.org/asterisk_stuff/res_hashvar.c |
| 03:24.09 | annonimous | hello |
| 03:25.12 | anthm | that lets channels have associative arrays |
| 03:26.00 | [TK]D-Fender | anthm : I only wish I could read that code and understand what you just said :) |
| 03:27.57 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 03:28.04 | anthm | a 2 tired varialble |
| 03:29.02 | anthm | instead of FOO=bar FOO:${EXTEN}=bar |
| 03:29.17 | anthm | or FOO:section1=bar |
| 03:29.35 | anthm | sounded to me like what you were daydreaming about |
| 03:30.35 | [TK]D-Fender | anthm : Similar to how you'd do somthing like this Set(${var[${index}]}=value), no? |
| 03:32.31 | anthm | depends |
| 03:32.39 | anthm | is index a number or can it be a string |
| 03:33.36 | anthm | it's like in perl $var{$key} = "$val"; |
| 03:33.55 | anthm | instead of an array based on number index it's a word you store it in |
| 03:33.58 | [TK]D-Fender | anthm : In my sample it could be "anything" as well... |
| 03:34.08 | anthm | is you sample imaginary ? |
| 03:34.32 | *** join/#asterisk tekkno (n=tekkno@209.182.99.52) |
| 03:34.54 | [TK]D-Fender | anthm : It COULD work, I haven't actually tried witha var, but I have done it with a DB |
| 03:35.00 | [TK]D-Fender | (ASTDB that is) |
| 03:35.22 | anthm | this would be Set(HASHVAR(${var}:${index})=value) |
| 03:35.29 | tekkno | hello all, I had a question concerning incoming calls via PSTN via SPA-3000 and AAH 2.7 |
| 03:35.54 | anthm | it also fires a manager event so you can see when someone alters it and you can query for the vals on the cli |
| 03:36.16 | [TK]D-Fender | anthm : Wouldn't need a function at all, and even if, you could pass the var as the name withouth the ${} |
| 03:36.27 | anthm | yah have fun |
| 03:36.31 | tekkno | for some reason callers always receive: "the person at extension 200 is busy" |
| 03:36.38 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 03:37.01 | anthm | the code to parse vars is already as easy to follow as chineese algebra in braile |
| 03:37.20 | tekkno | but I did turn off the vertical Service Activation Codes in the SPA-3000 |
| 03:38.23 | justinu|laptop | i feel that way about all the code |
| 03:38.43 | tekkno | so now I am a bit lost, and was hoping one of you would be able to help |
| 03:39.24 | tekkno | in the asterisk console I see: |
| 03:39.25 | tekkno | <PROTECTED> |
| 03:39.25 | tekkno | <PROTECTED> |
| 03:39.26 | tekkno | <PROTECTED> |
| 03:39.31 | jql | I only understand sip. you'd have to pastebin a sip debug of it not working |
| 03:39.40 | [TK]D-Fender | anthm : You forgot "in snow 20ft high, uphill, BOTH ways!" |
| 03:39.52 | jql | .0.5 is the phone? |
| 03:41.17 | *** join/#asterisk flynux (i=zsp2n0x@cl-8.bru-01.be.sixxs.net) |
| 03:45.39 | tekkno | oops jql you were talking to me? |
| 03:45.46 | jql | yeah |
| 03:45.48 | tekkno | yes .5 is the SPA-3000 |
| 03:46.16 | tekkno | calling in from the DID works fine |
| 03:46.23 | tekkno | outgoing calls work too |
| 03:46.24 | jql | I checked my sip logs... I get a 486 when I have my phone call itself |
| 03:46.31 | tekkno | but incoming calls from the PSTN line |
| 03:46.38 | tekkno | get busy message |
| 03:47.13 | tekkno | actually the next line might be of interest: |
| 03:47.17 | tekkno | <PROTECTED> |
| 03:47.17 | tekkno | <PROTECTED> |
| 03:47.17 | tekkno | <PROTECTED> |
| 03:47.17 | tekkno | <PROTECTED> |
| 03:47.30 | tekkno | the last one says the extension 200 is busy too |
| 03:47.54 | tekkno | then it says: == Everyone is busy/congested at this time (1:1/0/0) |
| 03:50.04 | tekkno | yesterday I tried to check the time with *60 |
| 03:50.35 | tekkno | and that triggered the SPA-3000 busy mode |
| 03:50.42 | tekkno | I finally found info on a forum |
| 03:51.00 | tekkno | and I removed all the vertical Service Activation Codes |
| 03:51.18 | tekkno | rebootet the device and restartet AMP |
| 03:51.24 | tekkno | just to be save |
| 03:51.30 | tekkno | it worked then |
| 03:51.36 | *** join/#asterisk bmg505 (n=leon@c1-184-10.rndf.isadsl.co.za) |
| 03:51.39 | tekkno | but this morning it is again busy :-( |
| 03:53.01 | dlynes_ | tekkno: did you follow the sipura 3000 setup instructions on voxilla? |
| 03:53.27 | tekkno | I did follow those instructions initially, they do have a wizzard |
| 03:53.35 | dlynes_ | no...not that stupid thing |
| 03:53.53 | tekkno | but that didn't work well, so I followed the instructions on NerdVittles |
| 03:53.56 | dlynes_ | there's instructions on how to set sipura 3000 up specifically for asterisk |
| 03:54.12 | tekkno | I think I saw that too |
| 03:54.43 | dlynes_ | What's SIP/200? |
| 03:55.05 | tekkno | that is an extension |
| 03:55.22 | dlynes_ | And what's SIP/asterisk? |
| 03:56.00 | tekkno | hm, I don't know why that's in the logs |
| 03:56.19 | tekkno | I am assuming that the SPA is regitering to the asterisk server |
| 03:56.37 | dlynes_ | Because some sip device that you've got configured as 'asterisk' is calling the sip device you've got configured as '200' |
| 03:56.41 | tekkno | btw, the registraion shows correctly in the web interface of the SPA |
| 03:57.01 | tekkno | interesting... |
| 03:57.26 | dlynes_ | tekkno: you might be better off asking in #freepbx...they'll be able to tell you which config files to check |
| 03:57.46 | tekkno | good advice, are they here on freenode aas well? |
| 03:57.52 | dlynes_ | tekkno: correct |
| 03:57.58 | dlynes_ | tekkno: i guess you didn't read the topic :) |
| 03:58.08 | tekkno | thanks dlynes |
| 03:58.24 | *** join/#asterisk argos73 (n=mike@cpe-24-93-184-116.neo.res.rr.com) |
| 03:58.33 | tekkno | well, I did assume the "real" geeks were in here ;-) |
| 03:58.49 | dlynes_ | tekkno: yeah, but the vast majority of us don't use freepbx |
| 03:58.52 | dlynes_ | and probably never have |
| 03:58.55 | tekkno | and AAH is based on asterisk |
| 03:58.59 | tekkno | ok I understand |
| 03:59.02 | [TK]D-Fender | Real Geeks Use Punch Cards! |
| 03:59.07 | tekkno | thank you anyways for your help |
| 03:59.07 | dlynes_ | Yeah, but the configuration files are all screwed up |
| 03:59.15 | tekkno | lmao TK |
| 03:59.28 | dlynes_ | and A@Home adds a few customizations |
| 03:59.37 | dlynes_ | not to mention it runs on Centos |
| 03:59.49 | tekkno | is Centos that bad? |
| 03:59.55 | tekkno | I usually use Debian |
| 03:59.57 | dlynes_ | It's Redhat/Fedora |
| 04:00.01 | dlynes_ | 'nuff said :) |
| 04:00.28 | tekkno | but I am not that much of a geek that I could configure asterisk myself |
| 04:00.39 | tekkno | thanks for the advice, guys |
| 04:00.55 | dlynes_ | tekkno: Just trim all the cruft out of the config files such as commented out lines |
| 04:01.03 | dlynes_ | tekkno: and trim it down to the basic modules |
| 04:01.11 | dlynes_ | tekkno: it's much simpler then |
| 04:01.15 | tekkno | I started doing that already |
| 04:01.18 | tekkno | true |
| 04:01.30 | dlynes_ | tekkno: then you're not dealing with information overload |
| 04:01.35 | [TK]D-Fender | CentOS is just fine... just that AMP makes you a cookie-cutter system that noone here wants to debug . |
| 04:01.38 | tekkno | correct, I will do so |
| 04:02.07 | tekkno | it is a neat idea to pre-package a system that Joe Sixpack can use |
| 04:02.12 | dlynes_ | [TK]D-Fender: still...anything based on redhat i'd prefer to stay away from...they have a nasty history of including beta and prerelease binaries |
| 04:02.15 | [TK]D-Fender | Very regretably *'s samples are "too much of everything, all at once" be be counted as a good learning tool... |
| 04:02.17 | tekkno | that will help to make VoIP more popular |
| 04:02.43 | [TK]D-Fender | dlynes : CentOS is much the reverse to FC. FC = bleeding edge |
| 04:02.44 | tekkno | I hear you TK |
| 04:02.59 | dlynes_ | So Centos is more stable than Redhat, too? |
| 04:03.06 | [TK]D-Fender | tekkno : I teach this stuff and have made many converts |
| 04:03.22 | justinu|laptop | Centos == RH enterprise linux |
| 04:03.27 | justinu|laptop | with the name changed to protect the guilty |
| 04:03.35 | tekkno | lol |
| 04:03.39 | dlynes_ | with all the non-gpl crap stripped out |
| 04:04.18 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 04:04.21 | kernel20 | hi there |
| 04:04.23 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 04:04.56 | kernel20 | i have installed asterisk, any ideas where can i find good tutorial to create dial plan? |
| 04:05.01 | kernel20 | i use xlite |
| 04:05.16 | tekkno | ok, I am off, bugging the freepbx guys |
| 04:05.19 | kernel20 | i want to test if my asterisk is really working |
| 04:05.19 | tekkno | thanks for the insight |
| 04:05.23 | kernel20 | any ideas? |
| 04:05.35 | *** part/#asterisk tekkno (n=tekkno@209.182.99.52) |
| 04:06.13 | ghost99 | <PROTECTED> |
| 04:06.26 | dlynes_ | kernel20: dialplan.conf, www.voip-info.org, click on asterisk pbx on the left hand side, click on applications in the main section, and then click on 'Dial' |
| 04:06.35 | dlynes_ | ~docs |
| 04:06.43 | jbot | hmm... 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 |
| 04:07.49 | kernel20 | ghost99? |
| 04:07.50 | *** join/#asterisk joelsolanki (n=joelsola@202.160.163.144) |
| 04:08.24 | kernel20 | im kind have a hard time how to test it |
| 04:08.33 | kernel20 | please understand i am just a newbie |
| 04:08.47 | dlynes_ | everybody's a noob |
| 04:09.28 | dlynes_ | But try the howtos on the main voip-info wiki for asterisk |
| 04:09.35 | dlynes_ | and also try the howtos mentioned above |
| 04:09.46 | joelsolanki | Hi dlynes ...Good Morning |
| 04:09.57 | dlynes_ | heya joel |
| 04:10.04 | joelsolanki | how are u ? |
| 04:10.16 | dlynes_ | tired |
| 04:10.31 | dlynes_ | i'm getting sooooooooooooo bored of working on this billing system, it's not even funny |
| 04:10.47 | kernel20 | actually howtos at voip-info are not so detailed |
| 04:10.55 | kernel20 | any ideas where to find a good howto? |
| 04:11.27 | kernel20 | i am not concerned of any fxo nor fxs card in here |
| 04:11.44 | kernel20 | what i want is only ip 2 ip call |
| 04:12.00 | dlynes_ | then what do you need asterisk for? |
| 04:12.22 | kernel20 | to make a call? |
| 04:12.23 | kernel20 | huh] |
| 04:12.33 | joelsolanki | :) |
| 04:12.42 | dlynes_ | if you only need ip to ip call |
| 04:12.46 | dlynes_ | you don't need asterisk |
| 04:12.50 | kernel20 | ? |
| 04:12.57 | dlynes_ | just tell one xlite phone to call the other xlite phone directly |
| 04:13.27 | dlynes_ | you need asterisk if you want to set up a pbx or a softswitch |
| 04:13.38 | orlok | Hmmm... |
| 04:13.39 | kernel20 | yes later on |
| 04:13.59 | kernel20 | how can i make calls on xlite? |
| 04:14.08 | orlok | i am registering with a sip provider, i shouldent need to have them listed as an extension in sip.conf, correct? |
| 04:14.19 | dlynes_ | kernel20: maybe try their support number? |
| 04:14.27 | justinu|laptop | lol |
| 04:15.06 | dlynes_ | justinu: xlite is commercial software isn't it? |
| 04:15.26 | justinu|laptop | unsupported, i believe |
| 04:15.36 | [TK]D-Fender | dlynes : X-Lite is the free-ware striped down enticement for the payed product |
| 04:15.38 | dlynes_ | ah...thought they had two versions |
| 04:15.47 | dlynes_ | free, and not free...the not free had g729 support |
| 04:15.48 | justinu|laptop | the pay for is called eyebeam |
| 04:15.53 | dlynes_ | ah |
| 04:16.07 | [TK]D-Fender | and also X-Pro |
| 04:16.36 | kernel20 | dlynes: what? |
| 04:16.38 | dlynes_ | too confusing |
| 04:16.52 | kernel20 | ok how can i call my own IP? |
| 04:17.02 | dlynes_ | kernel20: try x-lite's mailing lists then, or something....or maybe they have a freenode channel? |
| 04:17.10 | kernel20 | huh |
| 04:17.16 | justinu|laptop | RTFM d00d |
| 04:17.19 | justinu|laptop | that's what he's trying to say |
| 04:17.39 | dlynes_ | asterisk != x-lite |
| 04:18.47 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 04:23.56 | kernel20 | <dlynes_> just tell one xlite phone to call the other xlite phone directly: what? |
| 04:24.17 | dlynes_ | kernel20: it's called peer to peer calling |
| 04:24.29 | dlynes_ | kernel20: but like I said...this is the asterisk channel, it's not the xlite channel |
| 04:24.43 | dlynes_ | kernel20: try using google to find some info on how to do it |
| 04:24.50 | kernel20 | huh |
| 04:24.51 | *** join/#asterisk scubes13 (n=klanders@cpe-071-068-198-068.sc.res.rr.com) |
| 04:24.53 | kernel20 | come on |
| 04:24.57 | kernel20 | just making calls |
| 04:25.15 | dlynes_ | kernel20: dood....I didn't even know what x-lite was |
| 04:25.21 | dlynes_ | How do you expect me to help you? |
| 04:25.26 | terrapen | woohoo, * is finally deployed here |
| 04:25.34 | kernel20 | ok |
| 04:25.41 | kernel20 | back to my asterisk |
| 04:25.44 | kernel20 | how can i test it |
| 04:25.47 | kernel20 | it is running now |
| 04:25.50 | terrapen | i'm using it as a bridge between a NEC system and an Avaya, 50 miles away |
| 04:26.17 | dlynes_ | kernel20: register two phones against, and set up a dial plan using hte Dial() command for them to call each other |
| 04:26.27 | terrapen | the bossman just got his first voicemail e-mailed to him and nearly lost it |
| 04:26.33 | terrapen | i mean, he was quite excited |
| 04:28.22 | terrapen | so now i need to figure out what to do for the Next Big Thing |
| 04:28.34 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 04:29.15 | dlynes_ | blacklists |
| 04:29.28 | dlynes_ | terrapen: so when his wife calls, it dumps her into voicemail right away :) |
| 04:29.28 | kernel20 | dlynes: any good website i can follow? |
| 04:29.35 | dlynes_ | ~docs |
| 04:29.41 | jbot | somebody said docs was 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 |
| 04:29.48 | kernel20 | huh |
| 04:29.50 | kernel20 | docs |
| 04:29.59 | kernel20 | ~diocs |
| 04:30.03 | kernel20 | ~docs |
| 04:30.13 | jbot | methinks 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 |
| 04:30.13 | dlynes_ | Read what jbot told you |
| 04:30.32 | terrapen | heh |
| 04:30.32 | kernel20 | its nothing there |
| 04:30.32 | kernel20 | i've been there so so so so many times |
| 04:30.37 | dlynes_ | kernel20: You must go there with your eyes closed then |
| 04:30.41 | terrapen | i guess i should get my extensions and sip.conf into an RDBMS |
| 04:30.42 | dlynes_ | I go to voip-info every day |
| 04:31.05 | kernel20 | dlynes: .|. |
| 04:31.20 | terrapen | kernel20, its not that damned hard |
| 04:31.37 | kernel20 | terrapen: can u help me? |
| 04:31.39 | dlynes_ | terrapen: I think he wants everyone to do it for him |
| 04:32.22 | dlynes_ | I'm not into helping peopel that can't even be bothered to help themselves |
| 04:33.01 | kernel20 | did i asked u? |
| 04:33.13 | dlynes_ | yes you did |
| 04:33.14 | kernel20 | u just want to be involved out of nothing |
| 04:33.27 | terrapen | kertnel20, i can't help you |
| 04:33.32 | kernel20 | to be acknowledge of nothin |
| 04:33.42 | kernel20 | did your mother ever confronted u? |
| 04:33.49 | terrapen | go away. |
| 04:34.01 | terrapen | if you have to ask, you aren't ever going to know |
| 04:34.59 | dlynes_ | Maybe #perl's sending trolls over here |
| 04:35.13 | dlynes_ | payback time for those trolls that went over to #perl |
| 04:36.18 | terrapen | I need to look and see if * supports sqlite for extensions.conf storage |
| 04:37.06 | websae | fax 2 email.....anyone had any experience? |
| 04:38.14 | argos73 | scenario - ast w/te405 hooked to 100D module in a merlin legend system... works, but got fairly regular d-chan errors (don't remember the message)... ran across a paradyne DSU (the one that Lucent recommends) and installed that in the line - the errors seem to have stopped. sound reasonable? |
| 04:38.42 | argos73 | or is something else going on? cable is only about 30 feet |
| 04:39.39 | terrapen | argos, it works now, right? |
| 04:39.46 | argos73 | seems to |
| 04:40.08 | terrapen | what I need to figure out is why this works: |
| 04:40.19 | terrapen | Dial(Zap/g1/12345) |
| 04:40.19 | argos73 | if I understand what a DSU does correctly (signal regeneration, among other things), it makes sense |
| 04:40.22 | terrapen | errr |
| 04:40.29 | terrapen | ok, let me start again |
| 04:40.31 | terrapen | this works: |
| 04:40.41 | terrapen | Dial(Zap/G1/12345) |
| 04:40.44 | terrapen | this doesn't: |
| 04:40.50 | terrapen | Dial(Zap/g1/12345) |
| 04:40.57 | terrapen | for some reason, channel 1 has issues |
| 04:41.03 | [TK]D-Fender | terrapen L there is a difference and it IS case sensitive.. |
| 04:41.09 | terrapen | yup i know |
| 04:41.15 | [TK]D-Fender | ascending vs descending I believe. |
| 04:41.30 | terrapen | when it starts from 1 ascending, the call immediately hangs up |
| 04:41.38 | terrapen | but when it starts from 23 descending, it works fine |
| 04:42.01 | terrapen | something is hosed up with channel 1 ... i think it could be my zaptel.conf or zapata.conf |
| 04:42.14 | orlok | terrapen: what sorta nex gear? |
| 04:42.17 | orlok | pabx or other? |
| 04:42.22 | terrapen | "nex"? |
| 04:42.28 | orlok | nec, sorry |
| 04:42.33 | orlok | force of habit |
| 04:42.35 | terrapen | oh, shit, hmmm lemme go see |
| 04:42.47 | orlok | we deal with a nec subsiduary a lot - they make dslam/wan gear too now |
| 04:42.59 | terrapen | are you just curious or are you asking in relations to this problem im having |
| 04:43.03 | orlok | just curious |
| 04:43.12 | terrapen | the problem i have is not on the PRI to the NEC but the PRI to the avaya |
| 04:43.17 | terrapen | orlok, lemme check... brb |
| 04:43.27 | orlok | ok, i can dial inbound to * |
| 04:43.31 | orlok | but it goes straight to messagebank |
| 04:44.41 | terrapen | electra elite 192 |
| 04:44.45 | terrapen | does that sound right? |
| 04:45.19 | terrapen | i'm running this all through a redfone fonebridge which seems to work rather nicely |
| 04:46.02 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:46.23 | terrapen | orlok, i'm only having one problem with the * <-> NEC combo |
| 04:46.33 | terrapen | the NEC is not sending callerID stuff |
| 04:46.47 | orlok | heh, funny, i'm getting callerid issues dialling out via nec |
| 04:46.51 | terrapen | so when an NEC phone calls an * phone, it just says "unknown caller" |
| 04:47.11 | orlok | * is just telling me "caller id is blocked" |
| 04:47.18 | orlok | but its purely an * i think |
| 04:47.22 | terrapen | its probably a configuration problem on the NEC but i don't really care much |
| 04:47.31 | terrapen | i want to replace that pile of shit anyway |
| 04:47.43 | terrapen | the NEC phones are junk. i have a huge box of broken ones |
| 04:52.27 | orlok | man |
| 04:52.27 | orlok | -- Got SIP response 406 "Not Acceptable" back from 192.168.1.82 |
| 04:52.31 | orlok | wat the hell! |
| 04:53.22 | terrapen | sounds like you have my girlfriend at 192.168.1.82 |
| 04:53.52 | justinu|laptop | d'oh |
| 04:54.15 | *** part/#asterisk CodyC (n=cody@cpe-70-112-210-245.austin.res.rr.com) |
| 04:59.32 | scubes13 | hi all, I am looking for a VPN device or router that I can hook up a voip phone and a pc to that can connects back to the * box at our office (offsite) - possibly even looking for QoS on the device to allow the user to get best performance on their end of the connection - I know that I could prob do this by using Windows' builtin VPN access and a USB phone w/ possible softphone/software.... however, the boss wants it done his way - |
| 04:59.32 | *** join/#asterisk mgob (n=goldenol@c-67-160-85-76.hsd1.wa.comcast.net) |
| 04:59.35 | mgob | hi |
| 04:59.41 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 04:59.50 | mgob | should i bother to list an old T100P on ebay or are people just not buying these anymore? |
| 05:00.45 | asterboy | sure list it |
| 05:01.08 | websae | anyone played around with faxing much in here over t.39 or g711u? |
| 05:01.10 | mgob | I don't want to list something people don't want :P |
| 05:02.15 | jql | faxing... what a pita |
| 05:03.06 | Qwell | scubes13: "his way" is the right way. He's a smart man...listen to him |
| 05:03.25 | scubes13 | lol, kewl deal Qwell :-P |
| 05:03.38 | scubes13 | willing to learn if I am wrong :) |
| 05:03.48 | websae | faxing=headache |
| 05:04.01 | orlok | terrapen: yeah, 192.168.1.82 has no issues though, hmmm... |
| 05:04.06 | jql | ahh... vpns. increasing the VoIP packet overhead by another 20% |
| 05:04.55 | Qwell | jql: There is that too ;) |
| 05:04.55 | scubes13 | Qwell - is there a suggested product that I need to look for to make that happen? like perhaps a router with QoS? |
| 05:05.01 | asterboy | go with Hylafax |
| 05:05.15 | Qwell | scubes13: any cisco vpn gear should do that, heh |
| 05:06.14 | dlynes_ | terrapen: your nec pbx uses OS/2 as well? |
| 05:06.57 | asterboy | what do you define as IP faxing |
| 05:07.49 | websae | using t.38 to go through sip provider |
| 05:08.17 | dlynes_ | websae: how does asterisk handle t.38 passthrough? |
| 05:08.28 | dlynes_ | websae: or does asterisk totally screw it up? |
| 05:08.40 | *** join/#asterisk Eggplant (i=No@dsl-332.cascadeaccess.com) |
| 05:08.46 | asterboy | I just use Hylafax for faxing, (regular faxing) |
| 05:08.59 | asterboy | Although, it does do it over a VOIP line. |
| 05:09.09 | dlynes_ | asterboy: using ulaw? |
| 05:09.14 | asterboy | mcgp |
| 05:09.31 | websae | I have a provider that supports t.38 |
| 05:10.07 | dlynes_ | websae: Yeah, but you've got a sip ata that does t.38 which uses t.38 passthrough on asterisk to deliver it to the sip provider, right? |
| 05:10.24 | jql | so do I, supposedly. I haven't installed the passthrough support on that provider-connected test server yet, though |
| 05:10.44 | dlynes_ | ~seen coppice |
| 05:11.01 | jbot | coppice <n=chatzill@153.192.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 11h 58m 22s ago, saying: 'I have two in this house :-)'. |
| 05:11.01 | asterboy | sounds like a lot of mucking around to send a fax |
| 05:11.01 | websae | dlynes_:yep |
| 05:11.08 | jql | well, they do actually attempt to send t.38 faxes to me, so I suppose it's more than "supposedly" |
| 05:11.42 | websae | how does t.38 work out in terms of reliability? |
| 05:11.42 | dlynes_ | websae: sipura 2000/sipura 2002/pap2-na? |
| 05:11.54 | websae | sipura 2002 |
| 05:12.06 | terrapen | dangut... there doesn't appear to be SQLite support in Asterisk RealTime |
| 05:12.09 | *** join/#asterisk mitcheloc (i=user@204.8.143.106) |
| 05:12.11 | dlynes_ | websae: ah...so is that setup working out for you, or still in the testing phase? |
| 05:12.22 | websae | wait...I have the 2100 |
| 05:12.23 | dlynes_ | terrapen: unixODBC |
| 05:12.24 | websae | Sipura |
| 05:12.31 | websae | I haven't tried anything yet |
| 05:12.46 | jql | <-- sipura 2100 |
| 05:12.46 | dlynes_ | websae: sipura 2100 is the same thing as a wrt54g with a sipura 2002 built in, isn't it? |
| 05:13.10 | terrapen | dlynes: ugly, hacky |
| 05:13.10 | dlynes_ | well...and qos |
| 05:13.13 | terrapen | slowwww |
| 05:13.31 | websae | 2100 supports t.38 only i think |
| 05:13.33 | dlynes_ | terrapen: shurg...that's what everyone suggests to me for real time |
| 05:13.42 | terrapen | sqlite is the perfect database for ART |
| 05:13.57 | dlynes_ | terrapen: why? |
| 05:14.36 | websae | no one here has tried t.38 faxing? |
| 05:14.37 | terrapen | its simple to set up, file-based, and it's fast |
| 05:14.48 | dlynes_ | i would think it's not perfect, considering how art doesn't even support it |
| 05:15.01 | dlynes_ | anything that doesn't work with something is far from being perfect |
| 05:15.07 | jql | I couldn't get the passthrough to compile last time I tried |
| 05:15.19 | websae | for asterisk? |
| 05:15.21 | jql | but I did attempt to try it. does that count? :) |
| 05:15.24 | jql | yes |
| 05:15.25 | websae | for your t.38 |
| 05:15.28 | jql | yes |
| 05:15.37 | jql | I have t.38 provider, and t.38 ata |
| 05:15.45 | websae | hrm |
| 05:15.46 | jql | just need asterisk to pass it along |
| 05:15.50 | terrapen | i'm saying that it could be perfect |
| 05:15.50 | websae | what was your problem? |
| 05:15.55 | jql | wouldn't compile |
| 05:16.01 | jql | this was 3 weeks ago |
| 05:16.04 | terrapen | ART is hardly the gold standard for good use of a db |
| 05:16.11 | websae | ....just wouldn't compile? |
| 05:16.17 | dlynes_ | terrapen: no kidding |
| 05:16.20 | websae | I wonder how some people are implementing this... |
| 05:16.30 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.17.180.telnor.net) |
| 05:16.32 | terrapen | i'm dreading setting up ART |
| 05:16.41 | terrapen | i hope that i can get by for a while w/o it |
| 05:16.41 | jql | yeah. just plain wouldn't compile |
| 05:16.48 | jql | errors galore related to faxy |
| 05:16.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:16.56 | websae | where did you download from? |
| 05:16.58 | jql | probably fixed in trunk/ by now |
| 05:18.07 | orlok | Does anybody have an idea on why my supura would be spitting back 406's, but the grandstream is fine? |
| 05:18.20 | dlynes_ | 406? |
| 05:18.49 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:20.02 | orlok | -- Got SIP response 406 "Not Acceptable" back from 192.168.1.82 |
| 05:20.09 | Qwell | codec |
| 05:20.14 | CunningPike | orlok: Codec? |
| 05:20.18 | orlok | ahh |
| 05:20.20 | orlok | i will check |
| 05:20.22 | orlok | thankyou :) |
| 05:21.18 | dlynes_ | that's just what i was gonna suggest, until i see cp already got it :) |
| 05:21.47 | orlok | ahh |
| 05:22.13 | jql | websae: you have any luck with t.38 yet? |
| 05:23.24 | orlok | hmm, phone that works is using pcmu/pcma, the linksys is set to g711u |
| 05:23.33 | websae | not yet |
| 05:23.38 | websae | i haven't tried anything yet |
| 05:23.42 | dlynes_ | orlok: pcmu is g711u |
| 05:23.55 | websae | when i did g711u faxing, about 2/3faxes went through |
| 05:24.27 | jql | well, I have about 95% success with my 9600bps fax |
| 05:24.43 | websae | really? |
| 05:24.46 | websae | with g711u? |
| 05:24.49 | jql | yep |
| 05:24.51 | websae | what type of fax machine |
| 05:24.52 | orlok | dlynes_: dang then! |
| 05:25.00 | orlok | but the sipura only allows you to specify one codec |
| 05:25.01 | jql | cheapest analog fax on amazon |
| 05:25.14 | dlynes_ | orlok: check your sip.conf for the two phones, too |
| 05:28.08 | *** join/#asterisk mxmasster (i=mxmasste@68-171-34-117.vnnyca.adelphia.net) |
| 05:28.09 | mxmasster | hi all |
| 05:28.12 | mxmasster | quick question |
| 05:28.32 | mxmasster | i have a sipura spa-2002 behind a linksys router |
| 05:28.36 | websae | ok |
| 05:28.57 | jql | okay... now to actually try sending a fax |
| 05:29.09 | jql | I wish I had a fax machine here at home... |
| 05:29.10 | mxmasster | when i make calls through it i don't get inbound (asterisk -> linksys router -> ata) audio, outbound works fine |
| 05:29.22 | mxmasster | i have tcp/udp port 5060 forwarding on the linksys to the ata |
| 05:29.35 | mxmasster | and nat=yes in my asterisk configuration |
| 05:29.36 | websae | and port 10,000-20000 |
| 05:30.06 | mxmasster | websae: is there anyway i can narrow that port range? |
| 05:30.28 | orlok | hahaha |
| 05:30.29 | orlok | omg |
| 05:30.40 | orlok | theres two miners trapped about 1k under the ground in tasmania |
| 05:30.40 | Qwell | mxmasster: change it in rtp.conf, and on your devices |
| 05:30.50 | orlok | the miners have been given ipods to listen to music |
| 05:31.03 | orlok | as well as glowsticks, magazines and digital cameras |
| 05:31.10 | justinu|laptop | how about oxygen? |
| 05:31.11 | orlok | rave in the center of the earth! |
| 05:31.21 | websae | how about water |
| 05:31.24 | mxmasster | Qwell: no way to do it specific to a device? |
| 05:31.24 | orlok | 90mm pvc pipe has been run down to them |
| 05:31.25 | websae | and food |
| 05:31.29 | orlok | yeah, and that |
| 05:31.35 | Qwell | mxmasster: no |
| 05:31.37 | orlok | first things they got were jelly beans and energy drinks |
| 05:31.39 | mxmasster | bummer |
| 05:31.40 | websae | geeze |
| 05:34.29 | dlynes_ | mxmasster: if you're using defaults you only need 16384-16482 for rtp on the sipura 2002 |
| 05:34.44 | mitcheloc | they should send them plastic tubes to breathe through that are linked to the surface in case the dirt collapses on them |
| 05:35.56 | websae | i think they did |
| 05:35.56 | websae | yep...orlok said that already |
| 05:35.56 | websae | 90mm pvc pipe |
| 05:36.05 | *** join/#asterisk NoRemorse (n=bah@203-214-92-100.dyn.iinet.net.au) |
| 05:36.09 | mitcheloc | ah, yes |
| 05:36.42 | NoRemorse | hello all, does anyone know of a PST to sip service where you can call a voice line from traditional telephony and enter in a sip address through the number pad? |
| 05:37.34 | Qwell | NoRemorse: No, but you could write one fairly easily |
| 05:38.27 | mitcheloc | NoRemorse: if you write one use sphinx or something to make it easier to dial ;) |
| 05:44.30 | NoRemorse | yeah dont wanna reinvent the wheel tho if someone is doing it |
| 05:45.30 | Qwell | wheels were made to be reinvented |
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| 05:49.49 | orlok | yo |
| 05:49.49 | orlok | NoRemorse |
| 05:54.11 | NoRemorse | hey man |
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| 05:55.41 | boddy | I am planing to buy digium one port E1 card which model I have to buy ? |
| 05:55.54 | Qwell | boddy: TE100p |
| 05:55.58 | Qwell | or something |
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| 06:07.27 | CunningPike | OK - you can put the plug back in now |
| 06:07.27 | mitcheloc | welcome back ;) |
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| 06:24.59 | snitt | i suppose it is midnight overthere |
| 06:25.09 | kernel20 | hi |
| 06:25.15 | kernel20 | where is the sound file name locations? |
| 06:25.56 | snitt | /var/lib/asterisk/sounds? depends on your install |
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| 06:31.00 | kernel20 | it is from source |
| 06:31.11 | kernel20 | how can i create a customized sounds? |
| 06:32.56 | wasim | fart |
| 06:33.39 | wasim | actually, that was rude, you can create sounds by using the Record() facility or use sox to convert existing wav files |
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| 06:42.47 | mitcheloc | anyone in the yorba linda, ca area got a blank cd? |
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| 06:47.15 | snitt | ;) |
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| 07:04.17 | jsaunders | Ladies... germs... evenin'. |
| 07:04.29 | mitcheloc | huh, ladies? |
| 07:04.40 | jsaunders | Yeah, I know... it was a stretch. Heh. |
| 07:04.49 | mitcheloc | yep |
| 07:04.52 | wasim | l-fy, katty |
| 07:05.03 | jsaunders | l-fy is hardly a lady. |
| 07:05.12 | wasim | point taken :P |
| 07:05.16 | jsaunders | heheh |
| 07:05.45 | mitcheloc | 2/314 = 0.63% (if i did my math correctly) |
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| 07:25.42 | CrashHD | anyone know of some freeware asterisk manager/monitor apps? |
| 07:25.52 | CrashHD | or just any in general? |
| 07:26.12 | mitcheloc | CrashHD: what exactly do you want it to do? |
| 07:26.24 | mbrooks | crash: qview |
| 07:26.35 | CrashHD | I'm not picky at this point |
| 07:26.42 | CrashHD | I just want to see what is out there |
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| 07:26.50 | mitcheloc | CrashHD: snap @ www.snapanumber.com |
| 07:27.00 | mitcheloc | shameless plug ;) ^ |
| 07:27.30 | mitcheloc | if it's missing something you need let me know |
| 07:27.34 | CrashHD | mbrooks: do you have a website? |
| 07:28.07 | mbrooks | http://svn.digium.com/view/qview/ |
| 07:28.08 | mbrooks | heh |
| 07:28.11 | CrashHD | mitcheloc: I'm more looking for a manager app...something that would allow me to monitor extensions etc.. |
| 07:28.16 | CrashHD | heh |
| 07:28.16 | russellb | mbrooks: everything alright over there? |
| 07:28.27 | mbrooks | russell: what do you mean? |
| 07:28.34 | mbrooks | rb ;) |
| 07:28.36 | mitcheloc | CrashHD: well keep your eye on it, i'll be adding full support for the management api soon |
| 07:28.51 | CrashHD | ahh ok mitcheloc sounds good |
| 07:29.16 | mbrooks | crash: we use qview to show who is logged in to what queue |
| 07:29.19 | mbrooks | and how many callers, etc |
| 07:29.28 | mbrooks | there is also gastman |
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| 07:30.04 | mitcheloc | mbrooks: are there any screen shots of qview? |
| 07:30.33 | CrashHD | is qview a win app? |
| 07:30.47 | mitcheloc | CrashHD: it looked like a cgi web app to me |
| 07:31.55 | CrashHD | interesting |
| 07:31.57 | mbrooks | qview is a linux console app |
| 07:32.24 | mitcheloc | ah i was wrong then i ended up seeing this http://asterisk.toad.net/qview.pl |
| 07:32.38 | mbrooks | qview actually looks pretty nice |
| 07:32.43 | mitcheloc | actually i dunno if they are related, but both look like they are for asterisk |
| 07:32.56 | mitcheloc | but qview.pl looks like it's webbased |
| 07:33.42 | mbrooks | nope |
| 07:33.46 | mbrooks | not related |
| 07:33.56 | mbrooks | qview is a digium sponsored project |
| 07:35.59 | CrashHD | how good are the hooks into the *? |
| 07:36.22 | mitcheloc | sorry? |
| 07:36.28 | CrashHD | would windows developer be able to build a pretty advanced manager app? |
| 07:37.14 | CrashHD | hmm |
| 07:37.34 | CrashHD | guess my question is how far along are the socket connectors |
| 07:37.37 | yxa | does a 2.6 kernel offer any significant advantage to a 2.4 on a fast p4? |
| 07:43.39 | stoffell | how can a "avoiding initial deadlock" be caused? (like" May 3 06:27:27 DEBUG[29855] channel.c: Avoiding initial deadlock for 'Zap/6-1') |
| 07:46.40 | mitcheloc | CrashHD: if i understand correctly....you might want to look at this: http://www.voip-info.org/wiki-Asterisk+manager+API |
| 07:50.40 | CrashHD | mitcheloc: thank you |
| 07:50.40 | mitcheloc | np |
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| 08:09.15 | brookshire | anyone know how to kill a process when kill -9 won't work? |
| 08:09.27 | darkskiez | brookshire: is it state D or Z ? |
| 08:09.35 | brookshire | i have no idea |
| 08:09.47 | darkskiez | brookshire: typically it is already dead, u just have to wait for it to get cleared by the system. |
| 08:10.17 | brookshire | anyway to move that along? |
| 08:11.07 | tzafrir_laptop | Z is a zombie: already dead and just waiting for the parent to check its status. Nothing to worry about. |
| 08:11.38 | brookshire | i think it's in a S+ state |
| 08:11.40 | tzafrir_laptop | D is uninterruptable sleep. If a process stays in that state for long time, it is bad |
| 08:11.59 | tzafrir_laptop | + means a process group leader. Irrelevant to the state |
| 08:12.04 | brookshire | root 7730 0.0 0.0 2540 568 pts/1 S+ 02:52 0:00 df -h |
| 08:12.11 | brookshire | heh |
| 08:12.26 | tzafrir_laptop | S is a simple waiting state. Normal. Should respond to SIGKILL |
| 08:12.53 | brookshire | a nfs mount was down when i ran that |
| 08:12.59 | darkskiez | D is usually disk sleep |
| 08:13.00 | darkskiez | aah |
| 08:13.05 | darkskiez | thats your problem |
| 08:13.19 | tzafrir_laptop | brookshire, in that case avoid running 'mount', 'df' and such |
| 08:13.26 | tzafrir_laptop | They my hang in D state |
| 08:13.48 | tzafrir_laptop | Try umount -f or umount -l , maybe (Read the man page first) |
| 08:14.01 | darkskiez | brookshire: look at the hard and intr options for nfs mounts on the mount manpage. |
| 08:15.28 | brookshire | yay! tab completion crashes now! |
| 08:15.29 | brookshire | haha |
| 08:15.48 | brookshire | oh well... i guess i'm going to drive |
| 08:15.53 | brookshire | brb |
| 08:16.08 | darkskiez | brookshire: did u add hard,intr to your mount options ? |
| 08:17.27 | brookshire | ssh died |
| 08:17.47 | brookshire | sshd rather |
| 08:19.36 | *** join/#asterisk parag7732 (n=root@de2-b15868.alshamil.net.ae) |
| 08:20.19 | parag7732 | Hi i want to know that how can i find out my all recorded Outgoing calls....I have enabled "Record Outgoing Calls".... |
| 08:20.27 | parag7732 | I m using FreePBX |
| 08:21.31 | mitcheloc | parag7732: try #freepbx |
| 08:23.11 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 08:24.30 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 08:24.47 | parag7732 | Hi i want to know that how can i find out my all recorded Outgoing calls....I have enabled "Record Outgoing Calls"....I m using free PBX....and ARI is also installed even though i m not able to see outgoing call records |
| 08:24.54 | *** join/#asterisk ghenry (n=ghenry@195.38.86.72) |
| 08:25.06 | dpryo | parag7732: You're in the wrong channel. |
| 08:25.11 | mitcheloc | um "parag7732: try #freepbx" |
| 08:26.20 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:26.34 | Dandre | hello all |
| 08:26.36 | parag7732 | I m there also...but people are not answering |
| 08:27.28 | mitcheloc | parag7732: asterisk is not freepbx, so you have to ask people that know freepbx and how it works, so it's best to be patient in that channel or try during the day |
| 08:27.54 | Dandre | I am trying to use asterisk at home, but I haven't found the default password so I can't try it :-( |
| 08:28.02 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 08:28.16 | dpryo | Dandre: Try out plain asterisk. Install debian, and then asterisk. |
| 08:28.39 | dpryo | Dandre: It's alot easier if you use a text editor and take a look at the configuration files. |
| 08:28.48 | hwt | can i dial from the asterisk console? how? |
| 08:28.54 | dpryo | hwt: help dial |
| 08:29.37 | mitcheloc | jbot should auto answer any line that mentions freepbx or a@home heh |
| 08:29.40 | hwt | dpryo: thanks. |
| 08:30.05 | dpryo | mitcheloc: Yeah, or autokick :D |
| 08:30.16 | mitcheloc | or just mute ;) |
| 08:30.16 | Dandre | I am already using asterisk but I wanted to see what freepbx was like and I have downloaded vmware distro of aat |
| 08:30.36 | hwt | is freepbx worth anything? |
| 08:30.50 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 08:30.52 | mitcheloc | i gave it a try today too, didn't install on my machine...supposedly i didn't have enough hard drive space... (11gb) hehe |
| 08:31.03 | mitcheloc | hwt: it's free... |
| 08:31.13 | mitcheloc | worth $0 |
| 08:33.13 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:35.50 | hwt | mitcheloc: duh. |
| 08:36.51 | mitcheloc | heh, i've never used it so i dunno |
| 08:39.29 | sevard | alright |
| 08:39.35 | sevard | is there anything wrong with this line |
| 08:39.37 | sevard | exten = _1XXXXXXXXXX,1,DIAL(ZAP/2/${EXTEN},20) |
| 08:39.46 | sevard | because it looks friggen fine to me |
| 08:42.03 | mitcheloc | isn't it =>? |
| 08:42.10 | sevard | > does not matter. |
| 08:42.20 | darkskiez | is it not matching? |
| 08:42.24 | darkskiez | or dialing wongly? |
| 08:42.55 | sevard | I get a fast busy, i can't tell what's up, the zap channel works, that trunk is for my long distance context and when I plug channel two into the local context it works |
| 08:43.41 | *** join/#asterisk apardo (n=apardo@62.97.121.93) |
| 08:45.28 | hwt | sevard: maybe you mean exten = _1XXXXXXXXXX,1,DIAL(ZAP/2/${EXTEN:1},20) ? |
| 08:47.11 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 08:49.00 | sevard | omg |
| 08:49.02 | sevard | i think it's all done |
| 08:49.09 | sevard | all my lost configs |
| 08:49.12 | sevard | i rewrote them al |
| 08:49.22 | sevard | it's 3:30 a.m. |
| 08:49.31 | sevard | 3:50, i can't tell time |
| 08:50.03 | *** join/#asterisk kavit (n=kavit@210-84-40-39.dyn.iinet.net.au) |
| 08:52.15 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:52.56 | pif | are iax2 connections between * servers know to be brittle? |
| 08:53.01 | pif | s/know/known/ |
| 08:53.49 | pif | sometimes the remote * is no longer seen , even though pings are perfect |
| 08:54.49 | *** join/#asterisk drray (n=drray@c-67-183-123-24.hsd1.wa.comcast.net) |
| 09:00.00 | *** join/#asterisk CKGL (n=Cglob@202.8.86.162) |
| 09:00.29 | CKGL | anyone using gastman? |
| 09:04.56 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 09:06.07 | CKGL | wondering if the CLI> on gastman ever works |
| 09:06.54 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:09.54 | chapeaurouge | hi all |
| 09:10.22 | chapeaurouge | how do i refer to a ISDN-BRI phone connected to asterisk? like extension and such? I am looking for clear documentation, but unsuccessful so far |
| 09:10.59 | *** join/#asterisk lorinc (n=ang@caracas-3086.adsl.interware.hu) |
| 09:11.43 | brookshire | chap: do you currently have a BRI card? |
| 09:11.46 | chapeaurouge | yes |
| 09:11.49 | chapeaurouge | QuadBRI |
| 09:11.54 | chapeaurouge | using bristuff |
| 09:12.00 | brookshire | k |
| 09:12.25 | brookshire | i think you would set all the bri lines up in zaptel, if i am not mistaken |
| 09:12.33 | brookshire | zapata.conf |
| 09:12.37 | *** part/#asterisk parag7732 (n=root@de2-b15868.alshamil.net.ae) |
| 09:12.46 | chapeaurouge | yes, these are set. ztcfg shows them as configured anyway |
| 09:12.55 | chapeaurouge | but in the extension.conf, how do i refer to these phones? |
| 09:13.09 | brookshire | Zap/channel |
| 09:13.15 | chapeaurouge | aaah |
| 09:13.16 | brookshire | channel being the number |
| 09:13.18 | chapeaurouge | cool thanks :) |
| 09:13.19 | brookshire | like |
| 09:13.21 | chapeaurouge | dead easy |
| 09:13.22 | brookshire | Zap/1 |
| 09:13.24 | brookshire | Zap/2 |
| 09:13.25 | brookshire | etc |
| 09:13.27 | chapeaurouge | ah yes |
| 09:13.28 | chapeaurouge | :) |
| 09:13.30 | chapeaurouge | thanks dude |
| 09:13.34 | brookshire | np! |
| 09:14.54 | *** join/#asterisk MGSsancho (n=user@adsl-67-125-156-130.dsl.irvnca.pacbell.net) |
| 09:15.11 | chapeaurouge | you guys know if there's a vim schema for * ? this sux not to have highlights :) |
| 09:15.20 | chapeaurouge | indeed there is |
| 09:15.25 | chapeaurouge | niceness |
| 09:17.54 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 09:25.38 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
| 09:27.26 | stoffell | anyone already encountered a polycom 501 not getting an IP from the DHCP server? |
| 09:27.36 | stoffell | (while the other 20 polycom's do :) ) |
| 09:27.51 | Zeeek | what does the boot look like? Normal? |
| 09:28.03 | Assid | stoffell: happens |
| 09:28.09 | Assid | try assigning it a static dhcp |
| 09:32.36 | stoffell | boot looks normale |
| 09:32.47 | stoffell | hm, static definition in dhcp ? that I can do... |
| 09:33.01 | stoffell | boots looks normal Zeeek |
| 09:33.37 | *** join/#asterisk dlynes (i=1000@S010600c09f9a0fc4.vc.shawcable.net) |
| 09:35.07 | Zeeek | about 90 seconds? |
| 09:35.52 | stoffell | hm, counting now.. |
| 09:39.11 | stoffell | Zeeek, after approx 90 seconds, failed to get boot paramaters (but the phone is on the PoE switch), i see no requests in dhcp daemon |
| 09:40.02 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 09:44.09 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 09:45.51 | Zeeek | the phone is set p for dhbp obviously? |
| 09:46.00 | Zeeek | dhcp |
| 09:46.29 | Assid | i get this very very weird issue on my poly301 |
| 09:46.34 | Assid | it keeps going into hold |
| 09:46.38 | Assid | when in a call |
| 09:46.52 | Assid | err.. some calls.. it just keeps doing that.. some .. it doesnt |
| 09:47.53 | stoffell | Zeeek, yes, dhcp is enabled. adding a lease in dhcp doesn't help, i can try a fixed ip on the phone? (but right now it is on a working connection..) |
| 09:48.33 | Zeeek | you don't it asking for dhcp by sniffing? |
| 09:48.51 | Zeeek | Interesting - I had a bad ip500 once like that |
| 09:49.22 | stoffell | hm Zeeek, oke, will try sniffing |
| 09:49.23 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 09:49.35 | Zeeek | I had to send mine back for replacement |
| 09:49.41 | stoffell | you don't happen to know the right syntax to sniff dhcp-only? :) |
| 09:49.57 | Zeeek | in ethereal |
| 09:50.00 | Zeeek | ? |
| 09:51.18 | stoffell | tcpdump :) |
| 09:51.33 | Zeeek | do a tcpdump and look at it in ethereal |
| 09:51.46 | stoffell | ah, oke :) |
| 09:52.02 | Zeeek | then you'll find it easy to make a filter or color the lines |
| 09:53.31 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:55.46 | mut | anyone awake that knows about sangoma cards? |
| 09:56.16 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 09:56.21 | tparcina | hi everybody |
| 09:57.27 | tparcina | it's quite today |
| 09:58.36 | dlynes | yep |
| 09:58.48 | dlynes | i'm busy finishing off my code, so i'm not really here |
| 09:59.02 | *** join/#asterisk clive- (n=pirch@dsl-146-83-29.telkomadsl.co.za) |
| 09:59.52 | dlynes | but you weren't even here a minute when you concluded that it was quiet today |
| 10:00.10 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:01.14 | stoffell | Zeeek, different phone, different PoE adpater, works on same connect, trying tcpdump no |
| 10:01.16 | stoffell | w |
| 10:01.26 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:01.33 | Pj_ | stoffell: sounds like poetry |
| 10:02.15 | *** join/#asterisk f-bucher (n=fbucher@251.9.39-62.rev.gaoland.net) |
| 10:02.17 | mut | quiet? |
| 10:02.17 | f-bucher | hi |
| 10:02.30 | mut | it's 6am or earlier in the US |
| 10:02.40 | mut | which makes up most of the chat in the channel |
| 10:04.52 | f-bucher | hi i am working for a french voip provider and we are searching for some peering with an american voip company |
| 10:04.58 | tparcina | dlynes, yes bacause usuly in minute I get several mesages :)) |
| 10:05.09 | dlynes | ah |
| 10:05.26 | dlynes | bon matin, f-bucher |
| 10:05.32 | stoffell | Zeeek,no dhcp requests whatsoever... oh boy.. trying different PoE, unless you have other suggestions? |
| 10:05.38 | f-bucher | salut dlynes |
| 10:05.44 | *** join/#asterisk astr (n=ts@59.93.56.163) |
| 10:05.47 | tparcina | anybody uses Linksys SPA-901 SIP Phone - http://www.voipsupply.com/product_info.php?products_id=1565 |
| 10:05.57 | mitcheloc | stoffell: thats a good point, your cable could be bad, try connecting your computer or something else? |
| 10:06.04 | tparcina | I'm planing to buy nine of them |
| 10:06.10 | clive- | does voicemail have a maximum of 100 messages only ? |
| 10:06.20 | dlynes | clive-: that's an old limit that's been removed |
| 10:06.42 | dlynes | clive-: I don't know which 1.2 version removed that limit, or if maybe it's 1.2.8 when it'll be removed |
| 10:06.44 | f-bucher | is there some people here working in a american telco company ? |
| 10:07.00 | astr | i am looking for pstn providers with gsm codec. I found 2 - teliax and iconnecthere. the first one is unrealiable and the second one is very expensive |
| 10:07.02 | clive- | dlynes well, it seems to be still in 1.2.5 |
| 10:07.14 | dlynes | clive-: but it's definitely gone...if it's not gone in the currently release, you'll need to wait for those changes to be merged into a release from trunk |
| 10:07.30 | clive- | thanks |
| 10:07.37 | astr | is there any good and decently priced pstn minutes provider supporting gsm codec? |
| 10:07.40 | stoffell | mitcheloc, no, connecting another phone on this cable works. just this 1 polycom doesn't sent out dhcp requests (or they don't get at the server) |
| 10:08.08 | dlynes | f-bucher: i'm guessing you're looking for a CLEC or an ILEC? |
| 10:08.30 | clive- | astr try nufone |
| 10:08.40 | astr | checking.. |
| 10:08.40 | dlynes | isn't nufone defunct now? |
| 10:09.00 | clive- | afaik jerjer is still going strong |
| 10:09.02 | dlynes | I've heard everyone bitching left right and center about nufone i think it was lately |
| 10:09.06 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 10:09.10 | dlynes | ah...maybe it was some other company then |
| 10:09.14 | f-bucher | dlynes : we can say that yes |
| 10:09.40 | dlynes | f-bucher: so you're looking for someone on a pretty big scale then, right? |
| 10:09.44 | f-bucher | we are searching some partners |
| 10:09.56 | f-bucher | yes |
| 10:10.02 | clive- | there are plenty itsp's about, are you looking for iax or sip |
| 10:10.29 | dlynes | f-bucher: try www.suntelecom.net |
| 10:10.40 | f-bucher | sip but we have huge traffic and i can count on the poor QOS of the majority of voip provider |
| 10:10.42 | dlynes | f-bucher: they also parles en francais |
| 10:11.01 | dlynes | f-bucher: Illes dans la belle province |
| 10:11.11 | clive- | f-boucher are you looking in the usa or europe? |
| 10:11.18 | f-bucher | in the usa |
| 10:11.28 | f-bucher | in europe we already have some peering |
| 10:11.42 | astr | we need reliable providers, nufone - I am not sure if they look reliable. Reliability is a must as we are looking to terminate ~100k mins per month |
| 10:11.52 | dlynes | f-bucher: they have their own networks in both the US and Canada |
| 10:12.06 | clive- | 100k is not so huge |
| 10:12.18 | *** join/#asterisk L|NUX (n=linux@202.5.145.57) |
| 10:12.19 | dlynes | f-bucher: they do SIP into the US, and H323 into Canada |
| 10:12.37 | clive- | you will easily find someone in the usa |
| 10:13.06 | stoffell | Zeeek, indeed, the ip501 doesn't do a dhcp discovery, damn.. will set it to fixed ip then :( |
| 10:13.30 | f-bucher | yes sure but a need a certain quality of service and i am searching to create a direct trunck between my cirpack and their equipment |
| 10:13.44 | dlynes | stoffell: Try hooking up a crossover cable to it instead of a straighthrough cable |
| 10:13.48 | dlynes | stoffell: see if that doesn't fix it |
| 10:13.49 | astr | clive: thats the initial estimate |
| 10:14.12 | dlynes | stoffell: I've noticed some of these voip phones someone times only work with a xover cable |
| 10:14.25 | dlynes | s/someone/some |
| 10:14.29 | clive- | I know of a guy in cleveland who may be able to help you |
| 10:14.48 | stoffell | dlynes, but it did work before.. :( and so do all the other 19 ip501's.. strange |
| 10:15.11 | dlynes | f-bucher: talk to them...you'll probably be talking directly to the cfo or the ceo...they're the ones that usually answer my emails |
| 10:15.16 | astr | how do you guys terminate mins - we are currently using a server in datacenter and it connects to termination provider over internet. Is there any other better way ? |
| 10:15.35 | dlynes | f-bucher: they sell their customers on their own networks |
| 10:15.58 | dlynes | f-bucher: they were finding everyone else's networks weren't up to the expectations they had, so they built their own network |
| 10:16.01 | f-bucher | thx a lot dlynes |
| 10:16.03 | astr | clive: who in cleveland? |
| 10:16.30 | dlynes | f-bucher: That's one of the biggest reasons we decided to start terminating through them |
| 10:16.31 | astr | website? |
| 10:16.45 | dlynes | f-bucher: I'll be testing on their network probably next week or late this week |
| 10:17.00 | dlynes | f-bucher: we're looking to make them our main carrier shortly |
| 10:18.07 | clive- | astr: http://www.cordialcom.com/ some guy who is freinds with my sister-in-law , but I dont know him personally |
| 10:18.07 | dlynes | f-bucher: the nice thing about them, is they also speak french, so you can express yourself perfectly, too |
| 10:18.42 | f-bucher | dlynes do you say that my english is not perfect ? :) |
| 10:18.53 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:19.06 | dlynes | f-bucher: No...just saying that if you talk to them in French, guaranteed there won't be any confusion :) |
| 10:19.30 | f-bucher | i am looking at the website i juste see a canadian number i will try .. |
| 10:19.32 | f-bucher | thx |
| 10:19.34 | dlynes | f-bucher: Assumed your name was probably Francois Boucher, or something, so i figured you must be french :) |
| 10:19.38 | astr | clive: checking.. |
| 10:20.19 | dlynes | f-bucher: one sec...i'll get their email address for you instead |
| 10:20.23 | astr | clive: website does not open, strange reason |
| 10:20.29 | dlynes | f-bucher: the email will get you a quicker response |
| 10:20.32 | clive- | astr your other option is to get a T1 line yourself |
| 10:22.03 | astr | clive: we have lot of voip clients who connect to our server, we need a way to terminate few mins to pstn. hence looking for a provider. We have a dedi server in datacenter which has 100MBs port. But I was checking if there is a better way to connect the box to the pstn provider gateway using trunks etc. |
| 10:23.08 | dlynes | f-bucher: "'Peter Lazaris'" <plazaris@suntelecom.net> (Director & CTO), "Steve Mann" <smann@suntelecom.net> (Director of Wholesale Operations) |
| 10:23.17 | f-bucher | thx dlynes |
| 10:23.28 | dlynes | f-bucher: just tell them Daniel at 24/7 Communications sent you, so they know you're not coming out of the blue |
| 10:24.27 | dlynes | Steve will be the one you probably make initial contact with |
| 10:24.48 | dlynes | Peter will be the one to set you up with an account and/or possibly discuss other agreements |
| 10:24.52 | f-bucher | thx for all |
| 10:25.26 | dlynes | Steve probably doesn't speak French, but I think Peter is bilingual |
| 10:25.51 | astr | dlynes: I looked at your site. 24x7, do you guys terminate and support gsm? |
| 10:26.30 | dlynes | astr: We haven't started doing termination yet; not until I'm finished our postpaid and prepaid billing system |
| 10:26.42 | dlynes | astr: well, and had a chance to fully test it :) |
| 10:27.28 | *** join/#asterisk phpboy (n=shane@196.26.21.106) |
| 10:27.55 | phpboy | hey all... I think I've installed my zap drivers correctly yet when I try dial through it I get the following error |
| 10:27.56 | astr | dlynes: understood. anyone who would recommend for 100-200k gsm mins? we will need ~500k g711/g729 mins |
| 10:28.05 | phpboy | dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 10:28.42 | dlynes | astr: Why is it that you only want to do gsm? |
| 10:29.26 | astr | dlyes: our app runs on limited hardware devices and hence can do only gsm or g711. but bw might not be good always, hence gsm :) |
| 10:29.55 | dlynes | ah |
| 10:30.46 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
| 10:30.50 | phpboy | why Would I get this error? |
| 10:30.53 | astr | for g711/729, I am sure we would get plenty for ~1c. BTW, does anyone know some reliable wholesale provider providing mins for 1c |
| 10:31.07 | phpboy | I've configured zaptel.conf properly from what I can tell and it should be working :< |
| 10:31.09 | astr | i am yet to try commpartners |
| 10:31.38 | astr | phpboy: just giving it a shot, did you try sip calls |
| 10:31.42 | dlynes | Yeah...even after I get the billing system finished |
| 10:31.52 | dlynes | There's no possible way I could make money at that rate |
| 10:31.57 | phpboy | astr: sip calls work |
| 10:31.58 | dlynes | I'd be losing money |
| 10:32.08 | dlynes | I'm housed in a class 3 facility |
| 10:32.15 | dlynes | The gig charges are too high there |
| 10:32.24 | astr | dlynes: what is the best rate you can expect for reliable service for g711/g729 |
| 10:33.07 | dlynes | astr: I would have to calculate it based on my gig charges, cost of g729 codec licenses, and my termination costs |
| 10:33.20 | dlynes | astr: I couldn't give you an answer off the top of my head |
| 10:33.31 | astr | since you would be passing the bw, you would not need licenses for 729 |
| 10:33.57 | astr | what do others pay for pstn termination mins? reliable |
| 10:34.01 | dlynes | astr: If you're passing me gsm, i'd have to convert it to g729 |
| 10:34.16 | dlynes | astr: my voip upstream only takes g729 and g723 |
| 10:34.42 | astr | dlynes: no way - impossible. gsm to g729 is not possible using asterisk realtime without call degradation and also not scalable |
| 10:34.44 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:35.05 | dlynes | ah |
| 10:35.27 | dlynes | so where the heck are you going to get someone that does termination on gsm? |
| 10:35.34 | dlynes | I can only do gsm to g729, or gsm to pri |
| 10:36.05 | astr | dlynes: thats the sad part, i tried inphonex, teliax (level3), iconnecthere |
| 10:36.07 | dlynes | and my pri ld minutes are a ridiculous rate |
| 10:36.33 | dlynes | north america is 2.9c/min Cdn |
| 10:36.37 | dlynes | and that's my cost |
| 10:36.53 | astr | I would say expensive as others beat it |
| 10:36.57 | dlynes | so i avoid pstn ld at all costs |
| 10:37.05 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 10:37.05 | astr | understood |
| 10:37.08 | astr | from what I see, iconnecthere is good but charges 3c |
| 10:37.49 | dlynes | Well, in Canada, the CRTC states that if you're offering PSTN termination for VoIP, that you must be housed in a class 3 facility |
| 10:38.01 | dlynes | that's part of what keeps our costs high |
| 10:38.38 | dlynes | there's only two class three facilities in Vancouver, and they're both bloody expensive |
| 10:38.51 | *** part/#asterisk mitcheloc (i=user@204.8.143.106) |
| 10:39.04 | dlynes | anyways |
| 10:39.10 | dlynes | I've gotta get back to writing some code |
| 10:39.14 | mut | god damned MP-11a's |
| 10:39.16 | dlynes | I've wasted too much time on irc already |
| 10:39.22 | mut | fukin screwing up again |
| 10:39.27 | astr | dlynes: thnks |
| 10:39.35 | wasim | dlynes: i can give you us BULK for 0.92 and retail for 0.99 |
| 10:40.06 | wasim | err ... that euro cents, not US cents |
| 10:42.12 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 10:42.15 | dlynes | wasim: damnit |
| 10:42.22 | CKGL | anyone using gastman? |
| 10:42.25 | dlynes | wasim: the euro's strong...the USD's getting weak |
| 10:42.35 | dlynes | wasim: i want my Canadian buck to stretch further :) |
| 10:42.40 | CKGL | wondering if the CLI> function on gastman ever works |
| 10:42.42 | mut | needs to do some more push ups and sit ups |
| 10:43.00 | astr | wasim: where you from? |
| 10:43.04 | dlynes | wasim: btw...I'm not having troubles getting cheap LD minutes |
| 10:43.15 | dlynes | wasim: Only getting cheap LD minutes on a PRI |
| 10:43.22 | dlynes | wasim: and cheap PRI rates, for that matter |
| 10:43.47 | mut | ah man |
| 10:44.05 | mut | makes me happy thinking of our possible future build out |
| 10:44.22 | wasim | astr: pk |
| 10:44.22 | mut | running fiber from our location here to the next town over and connect it to the SBC building |
| 10:44.29 | wasim | dlynes: ah |
| 10:44.33 | mut | running OC48 out here |
| 10:44.44 | *** join/#asterisk MrChimpy (n=MrChimpy@smtp-gw.amplefuture.com) |
| 10:45.51 | *** join/#asterisk jeffik (n=Jeff@Violet-98.222.ADSL.NetSurf.Net) |
| 10:47.56 | mut | omfg |
| 10:48.00 | mut | this thing pisses me off |
| 10:48.04 | mut | it's been down like.. |
| 10:48.07 | mut | 20 minutes now |
| 10:48.34 | mut | so i finally decide it won't come back up on it's own, i get on the phone with a guy to go fix it, i get the words hello out, and the damn thing comes back up |
| 10:49.01 | mut | garrrrrrrrrrrrrrr |
| 10:49.02 | AsteriskAlbania | I change from EL4 to FC5 now I am trying to install zaptel and have this error |
| 10:49.03 | AsteriskAlbania | You do not appear to have the sources for the 2.6.16-1.2096_FC5 kernel installed. |
| 10:49.33 | MrChimpy | aa: yeah, and what do you think that means then? |
| 10:49.50 | AsteriskAlbania | I dont know how to resolve it |
| 10:50.03 | dpryo | Install the sources for the 2.6.16-1.2096_FC5 kernel |
| 10:50.03 | clive- | install your sources |
| 10:50.08 | MrChimpy | how about installing the kernel sources for the.... |
| 10:50.09 | papo | Assaf: install the sources of the 2.6.16-1.2096_FC5 kernel? |
| 10:50.10 | AsteriskAlbania | I have creatyt sym link with ln -s as I did in EL4 |
| 10:50.20 | AsteriskAlbania | but it does not work yet |
| 10:50.34 | AsteriskAlbania | I have tried with src.rpm |
| 10:51.02 | MrChimpy | find the correct one then |
| 10:51.21 | papo | or put the symlink at the correct place |
| 10:51.42 | MrChimpy | use strace to find where it's looking if you're desperate |
| 10:51.59 | gaupe | AsteriskAlbania: install kernel-devel |
| 10:52.57 | AsteriskAlbania | thanks I will try it now |
| 10:53.40 | phpboy | -- Executing Dial("SIP/6940-f3de", "Zap/1/827863878") in new stack |
| 10:53.41 | phpboy | May 3 10:43:10 NOTICE[2140]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 10:54.03 | phpboy | I've done the ztcfg, zttool and zttest and it seems to be working fine |
| 10:54.14 | phpboy | I've loaded the extention in extentions.conf and it's still not working |
| 10:54.19 | phpboy | what could be the problem? |
| 10:55.53 | MrChimpy | permissions on zap device? |
| 10:55.55 | MrChimpy | just a guess. |
| 10:57.54 | phpboy | it doesn't seem so |
| 10:57.54 | phpboy | :/ |
| 10:58.42 | phpboy | bear in mind that this is a analogue card |
| 10:58.56 | phpboy | TDM400P |
| 10:59.54 | MrChimpy | do you have /etc/udev? |
| 11:00.01 | tparcina | Linksys SPA901, does anybody use it? |
| 11:00.42 | AsteriskAlbania | gaupe: installing kernel-devel works thank you |
| 11:02.01 | AsteriskAlbania | gaupe: sorry it fails again even that starts to make linux26 |
| 11:02.30 | phpboy | MrChimpy: no |
| 11:02.35 | phpboy | this is a FreeBSD system |
| 11:02.57 | MrChimpy | ahhh. dunno then |
| 11:03.09 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 11:03.35 | MrChimpy | aa: get your kernel building outside of any asterisk stuff first |
| 11:03.42 | gaupe | AsteriskAlbania: you might want to look into this, http://fedoraproject.org/wiki/Extras/SIGs/VoIP |
| 11:07.46 | *** part/#asterisk oej (n=oej@apollo.webway.se) |
| 11:10.52 | wasim | me mack in a mit? |
| 11:11.45 | AsteriskAlbania | I have /usr/src/kernels/2.6.16-1.2096_FC5-i686 |
| 11:12.01 | AsteriskAlbania | and zaptel says that there is no source kernel |
| 11:12.47 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-147.claranet.co.uk) |
| 11:15.25 | MrChimpy | are you building as root? |
| 11:15.31 | AsteriskAlbania | yes |
| 11:16.07 | MrChimpy | can you build the kernel outside of the zaptel build? |
| 11:20.05 | AsteriskAlbania | what to you mean |
| 11:20.19 | AsteriskAlbania | whoat do you mean sorry |
| 11:20.20 | MrChimpy | can you build your kernel/ |
| 11:20.33 | AsteriskAlbania | I can make menuconfig |
| 11:20.40 | AsteriskAlbania | I can make |
| 11:20.55 | MrChimpy | and it builds without errors? |
| 11:20.57 | AsteriskAlbania | I have installed the RPM source package for the kernel |
| 11:21.03 | AsteriskAlbania | yes no errorrs at all |
| 11:21.12 | MrChimpy | ok |
| 11:21.35 | MrChimpy | well do your make with zaptel and strace it, see what it's looking at before it stops. |
| 11:21.55 | AsteriskAlbania | what is the option for strace |
| 11:23.43 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 11:24.08 | *** join/#asterisk HoopyCat (n=rtucker@cpe-66-67-224-82.rochester.res.rr.com) |
| 11:24.40 | HoopyCat | good morning! |
| 11:26.13 | *** join/#asterisk apardo (n=apardo@87.217.146.98) |
| 11:28.03 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 11:28.05 | HoopyCat | alright, this PRI isn't bouncing, time to get off of IRC, sit down, and go back to work. :-) |
| 11:32.25 | Zeeek | . |
| 11:34.49 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 11:35.06 | mut | wow, this dialup user that just called in... |
| 11:35.17 | mut | before the 14th of last month, every connection was great |
| 11:35.27 | mut | from the 14th on, every connection has been a lost-carrier |
| 11:35.51 | mut | ofcourse, somehow it's our fault too >:| |
| 11:37.43 | tparcina | I have dialplan question, I don't know ca it be done |
| 11:38.02 | tparcina | incoming call to SIP user |
| 11:38.17 | tparcina | first, it his phone is ringing for 15 sec |
| 11:38.55 | tparcina | then it's ringing all phones that are in the same pickup group as the first sip user |
| 11:39.09 | tparcina | can it be done thrue dialplan? |
| 11:42.22 | tparcina | as I see it, for that I'll just need pickupgroup variable |
| 11:45.12 | ManxPower | tparcina, no, you can't. You would need two Dial lines, one for the 1 phone, and a second for the group of phones. |
| 11:45.47 | ManxPower | Dial(SIP/mrhappy&SIP/johnson&SIP/oneeyedsnake) |
| 11:46.04 | snitt | it can be done |
| 11:46.43 | snitt | ring a phone for 15 sec, then say a 'please standby' to the caller, and then ring all phones |
| 11:46.56 | ManxPower | The call won't be able to be picked up between the time the first dial ends, and the 2nd dial begins, but that should be a very short time. |
| 11:47.18 | ManxPower | snitt, and while "please standby" is played, nobody will be able to pick up the call. |
| 11:47.25 | snitt | i know |
| 11:47.41 | ManxPower | most people that ask this question don't want that. |
| 11:48.08 | snitt | okay, then use a musiconhold that says stadby during the ringing |
| 11:48.44 | *** join/#asterisk Sebb (n=sebastia@einstein.f0o.de) |
| 11:48.45 | Sebb | hi.. |
| 11:49.07 | Sebb | is there a way to submit a custom variable via iax2, like i can do with sip? |
| 11:50.09 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 11:51.25 | ManxPower | Sebb, Define "Custom Variable" |
| 11:53.51 | Zeeek | aren't all variables custom? |
| 11:53.54 | *** join/#asterisk the_magic_bean (n=the_magi@209.43.15.211) |
| 11:54.49 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 11:55.50 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 11:56.22 | Sebb | ManxPower: well, as i can do it with sipaddheader, just add some information and get it out of the call on the other side.. e.g. for transmitting information about callerid presentation.. |
| 11:56.24 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
| 11:57.03 | ManxPower | Sebb, Use Set/SetVar, prefix the variable name with __ and it should be transported between servers. See README.variables |
| 11:58.25 | Sebb | ManxPower: thanks, i will test that. with which protocols does that work? |
| 11:58.45 | ManxPower | Sebb, I have no idea. Should work for at least IAX2 |
| 11:59.01 | ManxPower | This is a 1.2.x feature |
| 12:05.13 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:05.30 | puzzled | hi |
| 12:05.58 | *** join/#asterisk saftsack (n=saftsack@p54A7F975.dip.t-dialin.net) |
| 12:08.54 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 12:08.58 | Ariel_ | morning folks |
| 12:09.24 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:13.22 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-76.cybersurf.com) |
| 12:13.24 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 12:17.36 | *** join/#asterisk oej (n=oej@host-49.homerun.telia.com) |
| 12:18.34 | *** join/#asterisk sfollo81 (n=stefano@81.88.224.6) |
| 12:20.10 | sfollo81 | hallo |
| 12:20.49 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:22.25 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 12:29.03 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:29.18 | sfollo81 | hallo |
| 12:30.05 | *** join/#asterisk ivanfm (n=ivanfm@201-27-67-81.dsl.telesp.net.br) |
| 12:31.20 | [TK]D-Fender | Hello |
| 12:31.29 | [TK]D-Fender | :O |
| 12:31.32 | *** join/#asterisk gezick (n=gezick@208.255.178.130) |
| 12:31.52 | sfollo81 | ok |
| 12:32.02 | sfollo81 | i'm here |
| 12:33.41 | sfollo81 | can you help me? |
| 12:33.41 | [TK]D-Fender | sfollo81 : just ask |
| 12:33.47 | [TK]D-Fender | ~tips |
| 12:33.52 | jbot | i heard tips is (Trillion Instructions Per Second) This is a rating of a REALLY FAST computer. 1 TIPS is 1,000,000,000 instructions per seccond |
| 12:33.56 | [TK]D-Fender | hmmm |
| 12:34.03 | [TK]D-Fender | ~faq |
| 12:34.06 | jbot | well, faq is frequently asked question... try asking me about "RTFM" |
| 12:34.13 | [TK]D-Fender | UGH |
| 12:34.25 | [TK]D-Fender | Oh yeah! |
| 12:34.28 | [TK]D-Fender | ~suggestions |
| 12:34.30 | jbot | somebody said suggestions was 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be polite ... |
| 12:34.36 | [TK]D-Fender | There it is!~ |
| 12:36.17 | sfollo81 | i'm new in asterisk |
| 12:36.20 | *** join/#asterisk epablo (n=epablo@WLL-24-pppoe196.t-net.net.ve) |
| 12:36.44 | sfollo81 | and i'm trying to register an ata sip to it |
| 12:36.55 | epablo | Hi guys. How's it going? |
| 12:37.26 | epablo | Has anyone worked with the TDM2400E cards? |
| 12:38.28 | sfollo81 | i cannot understand how to see if my ata is registered or not |
| 12:38.35 | epablo | I want to know if I can put a couple (2) in one machine and it will work |
| 12:38.59 | [TK]D-Fender | epablo : Yes, you can if your MB is cooperative. |
| 12:39.34 | [TK]D-Fender | sfollo81 : in CLI do "sip show peers". In your sip.conf entry you should try putting "qualify=yes" as well to back this up. |
| 12:39.34 | epablo | D-Fender: cooperative? Meaning that it has 2 irq available? |
| 12:40.57 | epablo | D-Fender: on the specs it recomends a P4 1.6 what hw do you recomend when using 2 on the same box. Dual P4's or one bigger CPU? |
| 12:41.41 | [TK]D-Fender | epablo : Yes, you definately want each on its own IRQ. In general it is suggested you have no more than 2 cards in your system. |
| 12:42.04 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 12:42.07 | [TK]D-Fender | epablo : Depends on what that system will do exactly. No I seriously doubt you'll need that kind of processing power. |
| 12:42.18 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 12:42.32 | [TK]D-Fender | epablo : Just a basic P4 would do nicely I'm sure. But why on earth are you considering using that many analog channels? |
| 12:42.56 | epablo | At the time I have an old Dial P3 700 Xeon Would that work? |
| 12:43.35 | *** part/#asterisk sfollo81 (n=stefano@81.88.224.6) |
| 12:43.39 | phpboy | how do I put a delay between a 0 and the number it dials trough pstn? 0,012345678 |
| 12:43.40 | phpboy | ? |
| 12:44.14 | epablo | Well I have a client that has 33 analog channels in its actual PBX. They need asterisk for a new queue sys they bought for a callcenter. But don't wan't to change to VoIP phones |
| 12:44.22 | [TK]D-Fender | epablo : Depends what the box will be doing. Transcoding would kill it quite likely.. |
| 12:45.07 | [TK]D-Fender | epablo : You don't need to switch to VoIP phone, I'd sooner suggest geting a gateway like Mediatrix, AudioCodes, or a ATA's. |
| 12:45.29 | phpboy | please help me :< |
| 12:45.48 | [TK]D-Fender | phpboy : Go look up "cmd Dial" on the WIKI. |
| 12:46.02 | [TK]D-Fender | phpboy : Obviously the answer should be in there |
| 12:46.07 | phpboy | thanks |
| 12:46.12 | *** join/#asterisk sfollo81 (n=stefano@81.88.224.6) |
| 12:46.13 | epablo | D-Fender: I'll look into it. Thanks |
| 12:46.20 | [TK]D-Fender | epablo : np |
| 12:46.27 | sfollo81 | hi all |
| 12:47.23 | Sebb | ManxPower: i tried that with __foo, but it didn't work :/ |
| 12:47.38 | sfollo81 | sorry, but i'm a newbie: i'm configuring asterisk to work with a cisco ata 188 but, where can i see if the ata is registered/registering? |
| 12:48.08 | sfollo81 | i've not been so lucky with google |
| 12:48.33 | tparcina | manxPower, yes, i know that I'll need two dial lines. I don't mind that, just I don't know how to call group |
| 12:48.47 | [TK]D-Fender | sfollo81 : Is there some sort of issue with the rather specific steps I already gave you? Otherwise you're just starting to repeat yourself... |
| 12:49.26 | tparcina | snitt, i prefer ManxPover sugestion, I just need to know how to dial pickup group |
| 12:51.01 | AsteriskAlbania | anyone tested astcc , is it good for billing ? |
| 12:52.22 | mut | anyone good with photoshop or illustrator? |
| 12:54.16 | Hmmhesays | one day left in the clink |
| 12:54.46 | [TK]D-Fender | Hmmhesays : And boy is your butt tired? ;) |
| 12:55.15 | Hmmhesays | funneh |
| 12:55.49 | [TK]D-Fender | Hmmhesays : I am teh bomb y0! |
| 12:55.58 | Hmmhesays | nice |
| 12:59.32 | *** join/#asterisk tdonahue (n=tdonahue@www.vonworldwide.com) |
| 13:00.03 | [TK]D-Fender | Anyone awake here have experience compiling SpanDSP? I'm getting ready to give it a shot and have a quick question about compile order. On the site's instructions it gives me the impression I have to recompile Asterisk as a whole after using "path". Is this correct? |
| 13:00.09 | AsteriskAlbania | Zaptel seems not to compile with kernel 2.6.16 FC5 |
| 13:00.22 | AsteriskAlbania | have any one succeded with ti |
| 13:00.51 | [TK]D-Fender | AsteriskAlbania : Do you have the kernel source AND headers for your current kernel? |
| 13:03.13 | AsteriskAlbania | yes |
| 13:03.46 | AsteriskAlbania | I am reinstalling EL4 |
| 13:04.07 | [TK]D-Fender | AsteriskAlbania : have you checked the WIKI for pointers on your specific release? "asterisk fedora" |
| 13:05.19 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 13:05.26 | AsteriskAlbania | it seems to have only failures on FC5 |
| 13:05.33 | AsteriskAlbania | http://forums.digium.com/viewtopic.php?t=5825& |
| 13:06.49 | [TK]D-Fender | AsteriskAlbania : Seems to have instructions in there as to how to fix it. |
| 13:07.13 | [TK]D-Fender | This is a known issue with the latest CentOS 4/RHEL 4 latest kernel and I assume its the same on FC5. See this thread below on how to edit the Makefile for Zaptel to work around the issue. |
| 13:07.19 | [TK]D-Fender | http://forums.digium.com/viewtopic.php?t=5681 |
| 13:08.04 | Katty | morning. |
| 13:08.04 | AsteriskAlbania | I see it now |
| 13:08.19 | AsteriskAlbania | but already formated :) |
| 13:08.32 | AsteriskAlbania | thanks [TK]D-Fender |
| 13:09.09 | epablo | D-Fender: Instead of using 2 TMD2400E would it be better to install a channel banks and setup T1 cards? |
| 13:10.50 | [TK]D-Fender | epablo : I'd say better off getting an Analog>SIP gateway like the Meditrix 1124 for 24 ports at a time, and Sipura ATA's for the odd bits |
| 13:11.05 | Hmmhesays | ahh the mediatrix 1124, wonderful, wonderful gateway |
| 13:11.30 | phpboy | hey guys, I call myself through the PSTN to my mobile... the phone rings I answer but I can't hear anything on either side :< |
| 13:11.30 | epablo | What does the mediatrix do exactlly? |
| 13:11.40 | Hmmhesays | its an fxs gateway with 24 ports |
| 13:11.50 | epablo | Nice |
| 13:11.50 | phpboy | what could be the problem? |
| 13:13.08 | [TK]D-Fender | epablo : Means you don't have to worry about card compatability and its portable in that you don't need to run so much wire directly to your * box. |
| 13:13.19 | *** join/#asterisk awlane (n=awlane@s01.parallaxsystems.com.au) |
| 13:13.27 | Hmmhesays | yeah the amphenol connector is nice |
| 13:13.35 | *** part/#asterisk awlane (n=awlane@s01.parallaxsystems.com.au) |
| 13:13.58 | [TK]D-Fender | Both need it, its just with a gateay your * box can be further away. |
| 13:14.45 | epablo | D-Fender: Ok.. nice.. Thanks |
| 13:15.05 | mut | i'de also rather go with a channel bank |
| 13:15.20 | mut | they can push a line farther aswell |
| 13:16.00 | ManxPower | tparcina, I already gave you an example Dial to call a group |
| 13:16.07 | ManxPower | Dial(SIP/mrhappy&SIP/johnson&SIP/oneeyedsnake) |
| 13:16.08 | epablo | Well I already got the channelbank connectors made.. All I have to do is move them from the actual PBX to the new solution.. |
| 13:16.16 | ManxPower | Sebb, what does README.variable say about the subject? |
| 13:16.34 | epablo | I'm checking out the Rhino Channel Bank. Anyone used it? |
| 13:16.35 | tparcina | ManxPower, yes, but how can I know who is in particular pickupgroup? |
| 13:16.57 | [TK]D-Fender | mut : A theoretical benifit that doesn't always matter. releiving a weak server of load and compatability issues while being PBX agnostic wins on most scenarios. |
| 13:16.58 | tparcina | i need something like Dial(SIP/g1) |
| 13:17.18 | ManxPower | tparcina, you do not want a pickup group. A pickupgroup allows you to pick up a ringing phone that is NOT the one you are dialing from. |
| 13:17.27 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 13:17.29 | ManxPower | tparcina, you cannot do that, except for Zap. |
| 13:17.31 | [TK]D-Fender | epablo : Either solution would use the amphenol you'd wired up. |
| 13:17.48 | tparcina | error, I need something like Dial(SIP/$pickupgroup($extension)) |
| 13:18.05 | ManxPower | tparcina, you cannot do that. |
| 13:18.22 | tparcina | I know, and I'm looking for something that will work that way |
| 13:18.28 | ManxPower | But you could set a variable in global section to define what extensions are in a group. |
| 13:18.38 | tparcina | is ther any workaround that will work taht way? |
| 13:18.53 | ManxPower | tparcina, I guess you could write a custom application. |
| 13:19.16 | ManxPower | Why can't you use the standard dialing of multiple devices? |
| 13:19.21 | tparcina | yes, but I would like it to be exacly like the groups defined in sip.conf :)) |
| 13:19.54 | ManxPower | tparcina, groups defined in sip.conf are not used for Dial |
| 13:20.10 | ManxPower | Groups in sip.conf are used for *8 pickups |
| 13:20.33 | tparcina | standard dialing of multiple devices isn't good enough to much coding, and (as far I'm concern) to dirty |
| 13:21.05 | tparcina | ManxPower, I know that, I'm just trying to explain what I would like to acomplish |
| 13:21.31 | ManxPower | tparcina, Best of luck. I cannot help you further. |
| 13:22.04 | tparcina | standard dialing of multiple devices isn't good for one more reason. if I change group for one sip user, I'll have to change dialplan as well... |
| 13:23.04 | *** join/#asterisk axarob (n=jbash@spc1-barn1-5-0-cust127.asfd.broadband.ntl.com) |
| 13:23.24 | tparcina | ManxPower, thank you. If I find the solution... |
| 13:23.59 | tparcina | is ther any variable, that can tell me in what pickupgroup is called person? |
| 13:26.37 | *** join/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 13:27.14 | *** join/#asterisk EnoCix (n=jsloan@gateway.digium.com) |
| 13:27.35 | *** part/#asterisk EnoCix (n=jsloan@gateway.digium.com) |
| 13:27.44 | *** join/#asterisk Icemon (n=icechat5@dsl-146-32-24.telkomadsl.co.za) |
| 13:28.58 | *** part/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 13:29.04 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 13:29.07 | TheCops | Hi |
| 13:29.12 | *** part/#asterisk Icemon (n=icechat5@dsl-146-32-24.telkomadsl.co.za) |
| 13:29.26 | TheCops | Someone using french prompts for Asterisk ? |
| 13:30.35 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 13:33.16 | *** join/#asterisk arcy (n=arcanum@ppp84-82.adsl.forthnet.gr) |
| 13:33.52 | phpboy | hey guys, I call myself through the PSTN to my mobile... the phone rings I answer but I can't hear anything on either side :< what could be the problem? |
| 13:34.44 | [TK]D-Fender | TheCops : I do |
| 13:34.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:34.45 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 13:34.45 | TheCops | [TK]D-Fender, hi :) |
| 13:34.45 | [TK]D-Fender | TheCops : y0 |
| 13:34.45 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 13:35.13 | TheCops | [TK]D-Fender, do you have the vm-leavemsg.gsm wav for French ? |
| 13:35.20 | TheCops | It missing that one, like all french lol |
| 13:35.45 | [TK]D-Fender | let me look.... |
| 13:38.18 | [TK]D-Fender | Nope, I don't have it either |
| 13:39.18 | [TK]D-Fender | TheCops : Just passed on the news to one of the coordinators |
| 13:39.20 | *** join/#asterisk docelmo (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 13:39.32 | TheCops | [TK]D-Fender, relaly ? Nice! |
| 13:39.50 | TheCops | There's a way to get voice from that girl but, from thevoice digium ? I see only english person |
| 13:39.55 | [TK]D-Fender | TheCops : Ok, its a known issue, amongst others and is in the process of being corrected |
| 13:40.03 | *** join/#asterisk fulgas (n=fulgas@209.8.233.168) |
| 13:42.38 | TheCops | thnaks [TK]D-Fender |
| 13:42.51 | hrhrhr | ERROR[8379]: chan_zap.c:10231 setup_zap: Signalling must be specified before any channels are. |
| 13:42.54 | hrhrhr | any ideas? |
| 13:43.19 | *** part/#asterisk epablo (n=epablo@WLL-24-pppoe196.t-net.net.ve) |
| 13:43.26 | hrhrhr | i've looked through zapata/zaptel.conf, nothing obvious, google aint too helpful either |
| 13:44.03 | wasim | you have to run google's tummy just right ... |
| 13:44.08 | wasim | s/run/rub |
| 13:44.45 | [TK]D-Fender | hrhrhr : Seems pretty clear. You don't have a signalling line before you issue a "channel =>" line in zapata.conf |
| 13:48.10 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 13:48.26 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:48.44 | phpboy | :< |
| 13:48.48 | hrhrhr | [TK]D-Fender: thanks for that |
| 13:48.50 | hrhrhr | fxs_ks=1 FXS |
| 13:48.51 | phpboy | Can somebody PLEASe help me :< |
| 13:48.54 | hrhrhr | would that be my signalling? |
| 13:49.06 | *** join/#asterisk niter3 (n=klutch@d57-102-239.home.cgocable.net) |
| 13:49.14 | niter3 | anyone around. I'm having an issue connected to my asterisk via internet. It establishes a connection, but it won't send sounds over. Any idea? |
| 13:49.16 | hrhrhr | my channel => is on line 500 something and that is waayyy before it :s |
| 13:49.39 | hrhrhr | the above appears below Signalling method |
| 13:49.56 | hrhrhr | niter3: firewall issue perhaps? |
| 13:50.52 | niter3 | i've forwarded 5060 and the range. |
| 13:50.56 | niter3 | like I said it connects just fine. |
| 13:51.10 | *** join/#asterisk fulgas (n=fulgas@209.8.233.236) |
| 13:51.13 | niter3 | i can dial and it shows it's connected, but I hear no sound. For instance I dial *43 |
| 13:51.17 | hrhrhr | sorry, i'm an asterisk noob |
| 13:51.19 | niter3 | It connects, but no sound. |
| 13:51.27 | hrhrhr | there's an option for gain in one of the conf files too |
| 13:51.31 | hrhrhr | may be worth looking at that |
| 13:51.33 | [TK]D-Fender | hrhrhr : that line is wrong for zapata.conf. should be signalling=fxs_ks |
| 13:52.08 | hrhrhr | i'll give it a try :) |
| 13:52.32 | niter3 | this is odd. Wonder why it isn't working... :s |
| 13:53.03 | hrhrhr | [TK]D-Fender: you rock :D |
| 13:53.43 | niter3 | i've set nat to yes and qualify to yes for my extension. |
| 13:53.51 | niter3 | i've also edied sip_nat.conf |
| 13:54.16 | niter3 | for externhost = domain |
| 13:54.26 | niter3 | localnet = 10.1.1.0/255.0.0.0 |
| 13:54.31 | niter3 | i've restarted asterisk |
| 13:54.36 | niter3 | hrm.. anything i'm forgetting? |
| 13:54.59 | phpboy | I love you guys |
| 13:55.19 | [TK]D-Fender | niter3 : "externrefresh" |
| 13:55.28 | niter3 | yep I set that to 120 |
| 13:56.07 | niter3 | externhost = klutch.gotdns.com |
| 13:56.07 | niter3 | externrefresh = 120 |
| 13:56.08 | niter3 | localnet = 10.1.1.0/255.0.0.0 |
| 13:56.10 | niter3 | that's my line |
| 13:56.43 | niter3 | i've also set sip.conf and uncommented 'include sip_nat.con' |
| 13:56.50 | niter3 | i've also set sip.conf and uncommented 'include sip_nat.conf' |
| 13:56.52 | niter3 | sorry |
| 13:57.14 | niter3 | anything i'm forgetting? |
| 13:57.44 | hrhrhr | [TK]D-Fender: i guess there's no actual guide around for setting up an fxo (x100p) card? |
| 13:57.51 | hrhrhr | start to finish job |
| 13:58.12 | [TK]D-Fender | hrhrhr : WIKI & TheBook |
| 13:58.14 | [TK]D-Fender | ~docs |
| 13:58.16 | jbot | hmm... 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 |
| 13:58.27 | hrhrhr | the sip stuff seems to work easy enough, the actual pstn side of things has been the nightmare |
| 13:58.36 | hrhrhr | ok, cheers |
| 13:58.46 | [TK]D-Fender | brilliant |
| 13:59.12 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 13:59.19 | niter3 | yah I have to set my FXO card up when I get a phone card. I've set the trunk up and set inbound and outbound already. Just haven't test that yet. I want to login via the PBX over the net. |
| 13:59.23 | niter3 | but I don't know what i'md oing wrong. |
| 13:59.24 | niter3 | :s |
| 13:59.58 | [TK]D-Fender | niter3 : Have you forwarded the appropriate ports to your * box? |
| 14:00.06 | niter3 | 5060 |
| 14:00.10 | niter3 | both udp and tcp |
| 14:00.19 | niter3 | then a range 5060-5082 i think it was |
| 14:00.21 | niter3 | that's it |
| 14:00.54 | xachen | don't see the point of even opening TCP |
| 14:01.05 | niter3 | neither do I, but i'm just doing it. |
| 14:01.16 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:02.05 | sevard | [May 3 09:00:53] WARNING[5404]: file.c:970 ast_writefile: Unable to open file /private/tmp/_var_spool_asterisk_voicemail_default_1024_INBOX_msg0000.WAV: No suhh file or directory |
| 14:02.09 | niter3 | checking sip_additional.conf it seems ok |
| 14:02.13 | niter3 | username=200 |
| 14:02.14 | sevard | why the heck would it be looking there? |
| 14:02.16 | niter3 | type=friend |
| 14:02.42 | [TK]D-Fender | niter3 : You'll need to open up ports for RTP as well or you will get no audio. |
| 14:02.48 | sevard | that's supposed to be /var/spool and it's most def supposed to be / not _ |
| 14:02.48 | niter3 | qualify=yes port=5060 nat=yes mailbox=200@device host=dynamic dtmfmode=rfc2833 context=from-internal |
| 14:02.56 | niter3 | [TK]D-Fender: really? |
| 14:02.58 | niter3 | didn't know that |
| 14:03.01 | [TK]D-Fender | niter3 : yes |
| 14:03.07 | [TK]D-Fender | 10000-20000 UDP typically |
| 14:03.40 | BadPacket | has anyone seen JerJer? I'm getting annoyed now. |
| 14:03.41 | sevard | does anyone know? i'm like 'wtfmate' |
| 14:04.25 | *** join/#asterisk doolph (i=doolph@200.46.148.43) |
| 14:04.47 | phpboy | I can't seem to find where how to put a slight delay between 0 and the number |
| 14:04.48 | phpboy | :< |
| 14:04.53 | hrhrhr | niter3: what asterisk distro do you use anyway |
| 14:04.59 | phpboy | I can't seem to find where how to put a slight delay between 0 and the number I'm trying to dial through the pstn :< |
| 14:05.08 | hrhrhr | i've read there's an a@h distro too |
| 14:05.12 | hrhrhr | is that some kind of gui version? |
| 14:05.51 | doolph | anyone can help me with dtmf detection problem? |
| 14:05.58 | sevard | doolph: i wish |
| 14:06.22 | doolph | you wish but you could? |
| 14:06.29 | [TK]D-Fender | A@H *is* a distro with Asterisk and several accessories all bundled together. |
| 14:06.32 | sevard | doolph: i have a problem with that, the only advice i've gotten is play with your rxgain levels |
| 14:06.48 | Sonderblade | in the asterisk client i get this message: NOTICE[799]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! what does that mean? |
| 14:06.52 | sevard | [TK]D-Fender: can you give any hints to my vm problem? |
| 14:06.59 | sevard | Sonderblade: if you google for that string you'll find out |
| 14:07.06 | doolph | rxgain level? |
| 14:07.08 | doolph | from where |
| 14:07.12 | sevard | doolph: in zapata.conf |
| 14:07.17 | doolph | I am not using zap |
| 14:07.20 | sevard | oh |
| 14:07.24 | doolph | all SIP |
| 14:07.27 | sevard | sorry, i misread |
| 14:07.32 | [TK]D-Fender | sevard : which? |
| 14:07.51 | sevard | [May 3 09:00:53] WARNING[5404]: file.c:970 ast_writefile: Unable to open file /private/tmp/_var_spool_asterisk_voicemail_default_1024_INBOX_msg0000.WAV: No such file or directory |
| 14:08.00 | [TK]D-Fender | Sonderblade : Thats MPG123 being a little flakey. Switch to using Native MoH |
| 14:08.07 | sevard | [May 3 09:00:53] WARNING[5404]: file.c:981 ast_writefile: No such format 'wav49' |
| 14:08.21 | Sonderblade | [TK]D-Fender: how do i do that? |
| 14:08.24 | sevard | [May 3 09:00:53] WARNING[5404]: app.c:730 ast_play_and_record: Error creating writestream '/var/spool/asterisk/voicemail/default/1024/INBOX/msg0000', format 'wav49' |
| 14:08.27 | sevard | sorry, three line paste |
| 14:08.28 | [TK]D-Fender | sevard : That is one screwed up path.... I'd check your asterisk.conf and a few other places... |
| 14:08.45 | sevard | [TK]D-Fender: my asterisk.conf's spool directory is /var/spool/asterisk though |
| 14:08.55 | [TK]D-Fender | Sonderblade : Go look up "music on hold" on the WIKI. It'll give you instructions. |
| 14:08.57 | sevard | astspooldir => /var/spool/asterisk |
| 14:10.39 | niter3 | hrm, sound isn't working still |
| 14:10.43 | Sonderblade | [TK]D-Fender: where's the wiki? |
| 14:10.45 | niter3 | i forwarded the port range |
| 14:10.46 | niter3 | hrm.... |
| 14:10.50 | niter3 | what the heck |
| 14:11.02 | sevard | Sonderblade: http://www.voip-info.org |
| 14:11.05 | *** join/#asterisk GerbilNut (i=GerbilNu@65.88.144.41) |
| 14:11.26 | sevard | niter3: did you forward UDP traffic |
| 14:11.35 | niter3 | i got a linksys here |
| 14:11.42 | sevard | niter3: okay.. |
| 14:11.45 | [TK]D-Fender | Sonderblade :.... |
| 14:11.46 | niter3 | i'm in applications & gaming under port triggering |
| 14:11.46 | [TK]D-Fender | ~docs |
| 14:11.48 | jbot | extra, extra, read all about it, 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 |
| 14:12.00 | niter3 | and you can't specifiy you just tell it to use a range of ports. that's it |
| 14:12.06 | niter3 | doesn't give an option for UDP or TCP |
| 14:12.17 | *** join/#asterisk cji (i=3000@66.80.146.7) |
| 14:12.26 | sevard | niter3: look for a firmware upgrade, what model number of a linksys |
| 14:12.53 | [TK]D-Fender | niter3 : Look elsewhere. |
| 14:13.20 | sevard | [TK]D-Fender: iirc on linksys's firmware under applications and gaming is the port forward page |
| 14:13.20 | [TK]D-Fender | niter3 : you need to head to the more general port forwarding menu, not that "quicky" one you're in. |
| 14:13.23 | niter3 | wrt54GS but I have an opens ource firmware on it |
| 14:13.56 | niter3 | k |
| 14:13.59 | sevard | niter3: if we dont' know the firmware we obviously can't help you |
| 14:14.15 | sevard | s/help you/tell you where to click/g |
| 14:14.46 | niter3 | true.. |
| 14:14.53 | niter3 | i'm doing what you said. the genereal port forwarding |
| 14:14.53 | niter3 | try this |
| 14:15.08 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.94.Dial1.SanJose1.Level3.net) |
| 14:15.08 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 14:15.18 | phpboy | can somebody please tell me how to do _0.,1,Dial(Zap/1/0,1337${EXTEN:1}) properly? |
| 14:15.30 | niter3 | hrm...... urg |
| 14:15.32 | niter3 | still not working. |
| 14:16.25 | phpboy | AH |
| 14:16.26 | phpboy | _ |
| 14:16.30 | phpboy | cool1!! |
| 14:16.37 | sevard | ... |
| 14:17.01 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 14:17.22 | Sonderblade | i have installed asterisk from the debian package for 1.2.7 and it seems like the mp3 files fpm-world-mix.mp3 fpm-calm-river.mp3 fpm-sunshine.mp3 is missing from the package, can anyone confirm that? |
| 14:18.02 | sevard | Sonderblade: I don't use packages but if you want those specific mp3s i'll send them |
| 14:18.04 | Hmmhesays | you can confirm that |
| 14:18.12 | niter3 | stupid.... no idea why this sin't working.... |
| 14:18.20 | Hmmhesays | Sonderblade: why do you need someone else to confirm that? |
| 14:18.27 | nahirean | sonder, ls /var/lib/asterisk/sounds/ |
| 14:18.32 | niter3 | port range forward if don't it to point to my machien via UDP ports.. |
| 14:18.41 | sevard | nahirean: /var/lib/asterisk/mohmp3 |
| 14:18.50 | nahirean | wewps :) |
| 14:19.01 | sevard | niter3: just freaking put your god damn machine on dmz |
| 14:19.01 | Hmmhesays | i think the path is different in the deb packages |
| 14:19.15 | Hmmhesays | cat /etc/asterisk/asterisk.conf |
| 14:19.33 | sevard | Hmmhesays: did you see my error about voicemail? i'm lost on that. |
| 14:19.39 | Hmmhesays | no |
| 14:19.41 | sevard | Hmmhesays: last night I rebuilt all the freaking confs |
| 14:19.41 | Hmmhesays | what did you do |
| 14:19.42 | sevard | did you hear |
| 14:19.52 | Hmmhesays | i was in jail last night |
| 14:19.55 | sevard | WHAT |
| 14:20.01 | Sonderblade | Hmmhesays: cause it seems like a bug in the debian package because the mp3 files are included in the tarball |
| 14:20.02 | Hmmhesays | well not jail, but close enough |
| 14:20.08 | sevard | your woman tied you up? |
| 14:20.16 | Hmmhesays | Sonderblade: two people told you what to do |
| 14:20.19 | C4T3l | Hmmhesays: was it a real jail or chroot? |
| 14:20.24 | sevard | haha |
| 14:20.39 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:20.42 | Sebb | ManxPower: it doesn't say anything to iax-channels, the documentation is a bit rare.. |
| 14:20.48 | Hmmhesays | what'd you do sevard |
| 14:21.11 | sevard | Hmmhesays: I upgraded last night, I was pretty sure I backed up my config files but boom, I upgraded and lost everything. So I spent till 4:00 a.m. last night rebuilding all the files out of my head |
| 14:21.11 | ManxPower | Sebb, your extensive search of the mailing list archives was not helpful |
| 14:21.13 | ManxPower | !mailinglist |
| 14:21.37 | ManxPower | sevard, the only reason it would blow up your config files is if you did a "make samples" |
| 14:21.39 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-38-dsl.end.estpak.ee) |
| 14:21.47 | sevard | ManxPower: that's what I thought but I didn't make samples |
| 14:21.52 | sevard | samples* |
| 14:22.03 | *** join/#asterisk niter3 (n=klutch@d57-102-239.home.cgocable.net) |
| 14:22.08 | sevard | I have no idea what happened |
| 14:22.09 | niter3 | ok i put it on dmz. no difference. |
| 14:22.12 | Hmmhesays | um, you know when you do make samples your old configs are backed up automatically? |
| 14:22.35 | sevard | Hmmhesays: where to (although I didn't make samples) |
| 14:22.37 | shiznatix | I have a context with many terminals in it. I need a way that no matter that extension was dialed it starts in this extension |
| 14:22.41 | C4T3l | niter3: what problem are you guys having... sorry just joined. |
| 14:22.42 | Hmmhesays | do it twice in a row you're farked |
| 14:22.46 | doolph | why my asterisk is not detecting SIP dmtf tones? |
| 14:22.54 | Hmmhesays | x.conf.old in /etc/asterisk/ |
| 14:22.58 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 14:23.02 | Hmmhesays | because you have your dtmf setting wrong |
| 14:23.03 | sevard | Hmmhesays: nope, they wern't there |
| 14:23.08 | Sebb | ManxPower: yes. i just found that long, old http://lists.digium.com/pipermail/asterisk-dev/2003-December/002407.html IAX2 call variable passing between servers thread, but without a solution |
| 14:23.12 | Hmmhesays | wtf did you do |
| 14:23.20 | doolph | no i have rfc2833 |
| 14:23.21 | Hmmhesays | remind me never to let you near any of my production machines |
| 14:23.53 | ManxPower | Sebb, Show me your SetVar |
| 14:23.54 | sevard | Hmmhesays: No idea bub, but I spent all night fixing the mistakes |
| 14:23.59 | Hmmhesays | that'll learn ya |
| 14:24.08 | Hmmhesays | ahh to be in my youth again |
| 14:24.23 | sevard | Hmmhesays: everything I paid you to help me with was lost, I had to pull everything out of my head, apparently it all works now except for this voiemail problem |
| 14:24.30 | sevard | Hmmhesays: http://pastebin.ca/53197 |
| 14:24.49 | Hmmhesays | sevard, thats good for you |
| 14:24.49 | sevard | voicemail* |
| 14:25.00 | Hmmhesays | all the moh stuff was on thelostpacket |
| 14:25.04 | sevard | Hmmhesays: good for a nubtard |
| 14:25.12 | sevard | Hmmhesays: ah yes, I haven't added that y et. |
| 14:25.13 | sevard | yet* |
| 14:26.22 | Sebb | ManxPower: http://pastebin.ca/53200 |
| 14:26.27 | Hmmhesays | i got some other fun, yet really useless stuff to put up |
| 14:26.44 | sevard | Hmmhesays: useless stuff is the best, practical shit is boring |
| 14:26.51 | sevard | Hmmhesays: did you check out that pastebin? |
| 14:27.40 | Hmmhesays | are you still using that slimmed down install? |
| 14:27.45 | sevard | Hmmhesays: no |
| 14:27.53 | Hmmhesays | asterisk have write access to /tmp? |
| 14:27.53 | sevard | Hmmhesays: sort of but no |
| 14:28.16 | sevard | Hmmhesays: it should but it should write directly to /var/spool/asterisk |
| 14:28.24 | Hmmhesays | not according to that |
| 14:28.25 | sevard | I have no idea where the /tmp is coming into play |
| 14:28.40 | sevard | right, and the spool directory is set up in asterisk.conf as /var/spool/asterisk |
| 14:28.47 | sevard | that's what is throwing me for a loop |
| 14:28.50 | shiznatix | I have a context with a lot of SIP phones in it. I am using a GoToIfTime call to goto a queue durring certain hours. The problem is that if they dial a existing extension then auto call that phone and ignore the GoToIfTime. Here is my pastebin: http://pastebin.com/696177 |
| 14:29.00 | ManxPower | Sebb, what verison of Asterisk? See http://pastebin.ca/53202 |
| 14:29.02 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:29.12 | Sebb | ManxPower: 1.2.7.1 on both side |
| 14:30.01 | sevard | Hmmhesays: asterisk.conf: record_cache_dir => /private/tmp |
| 14:30.33 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:30.59 | Hmmhesays | beats me |
| 14:31.03 | Hmmhesays | if you figure out the answer, do tell |
| 14:31.05 | sevard | i might have fixed it |
| 14:31.06 | *** join/#asterisk redondos (n=redondos@201.255.36.217) |
| 14:31.11 | Sebb | ManxPower: and i don't see that the variable is transmitted when using iax2 debug |
| 14:31.12 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:31.46 | [TK]D-Fender | Sebb : Since when do variables get passed over a call? |
| 14:32.01 | Sebb | [TK]D-Fender: well.. ask ManxPower ;) |
| 14:32.24 | redondos | Hello, guys. I'm having a problem with my Asterisk server. It is behind a router, and I would like to connect to it using SIP. So I forwarded some ports (5060, 5080). Now, when I make a call, the server responds, and the line gets picked up, but I can't hear anything as the time counter rolls. |
| 14:32.38 | [TK]D-Fender | Sebb : They get passes in a Local/ call dure, but to another box? Ummm... NO |
| 14:32.52 | Sebb | [TK]D-Fender: i just want to submit a variable when calling with iax like it is possible with sipaddheader when using sip. and ManxPower means it should work like that.. do u have a better idea? ;) |
| 14:33.13 | [TK]D-Fender | redondos : You need to forward ports for RTP as well. SIP initiates calls, but doesn't control VOICE. |
| 14:33.21 | triple-e | what do you guys do about QoS in an office environment |
| 14:33.33 | [TK]D-Fender | redondos : You typically should forward UPD 10000-20000 to it as well |
| 14:34.04 | [TK]D-Fender | Sebb : not really. You could do somehing really painful and try to encode it in the EXTEN you dial.... |
| 14:34.36 | redondos | [TK]D-Fender: Isn't it possible to limit the port range that asterisk uses for RTP connections? |
| 14:34.39 | Sebb | [TK]D-Fender: yes, that's what i do at the moment.. ;) |
| 14:34.58 | triple-e | redondos: yes |
| 14:34.59 | Sebb | and I hoped there was a better solution.. |
| 14:35.36 | *** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx) |
| 14:35.46 | salviadud | !seen _paulo_ |
| 14:35.50 | redondos | triple-e: That sounds like what I need. Can you point me in the right direction please? |
| 14:36.09 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.94.Dial1.SanJose1.Level3.net) |
| 14:36.16 | triple-e | hummmm--- looking |
| 14:37.29 | triple-e | rtp.conf |
| 14:37.37 | *** join/#asterisk motu (n=motu@192.165.166.190) |
| 14:38.01 | triple-e | you can cut it down quite a bit |
| 14:38.30 | redondos | triple-e: thank you very much |
| 14:38.45 | triple-e | [TK]-Fender: do you have any suggestions on QoS for a 8 extension office |
| 14:39.48 | Hmmhesays | 'say goobye' by theory of a deadman rocks |
| 14:40.42 | syzygybsd | anyone have check_auth: stale nonce received from <321@mydomain> messages |
| 14:41.00 | syzygybsd | don't know why I would be getting those the ping is only 15ms |
| 14:41.18 | sevard | Hmmhesays: SO! I fixed the problem but now when I dial *98 I get a 404, WTF?! I didn't even touch extensions.conf |
| 14:41.30 | sevard | My God, I'm going to get a tumor from * |
| 14:41.46 | triple-e | :-) |
| 14:42.01 | triple-e | <-- got to sleep at 3am |
| 14:42.13 | sevard | <4:20am |
| 14:42.24 | triple-e | ^^wins |
| 14:42.52 | triple-e | holy cow -- what are you working on |
| 14:43.05 | sevard | i lost all of my configs and had to rebuild everything from scratch |
| 14:43.07 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:43.07 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:43.09 | Hmmhesays | I'm still waiting for those attractive mexican women to come knocking on my door |
| 14:43.16 | motu | xlite client with asterisk, how can I just get into the default s extension, without dialing anything? |
| 14:43.25 | triple-e | production box that customers are using ? |
| 14:43.27 | sevard | Hmmhesays: did you get that spam too? |
| 14:43.33 | sevard | triple-e: yeah, sort of. |
| 14:43.40 | triple-e | ouch |
| 14:43.41 | Hmmhesays | no, that was directed at salviadud |
| 14:43.42 | sevard | triple-e: sev made a baddie. |
| 14:43.50 | triple-e | yeah kinda |
| 14:43.54 | sevard | yeah, kinda. |
| 14:44.53 | sevard | now all of my configs are backed up across three machines |
| 14:45.00 | sevard | that will never happen again. |
| 14:45.06 | triple-e | i have a customer who complains of call quality --- can anyone suggest a QoS solution ? |
| 14:45.18 | Hmmhesays | what is causing the problem? |
| 14:45.25 | doolph | upgrade your bandwidth? |
| 14:45.35 | sevard | triple-e: how is your network set up / do your router/firewall/switches support qos |
| 14:45.39 | triple-e | sevard: i use svn for config |
| 14:46.09 | triple-e | the guy has a pix 501 |
| 14:46.36 | sevard | iirc the pix does qos just fine |
| 14:46.50 | sevard | you have to do QoS all the way down the line though |
| 14:46.51 | triple-e | so im going to put some qos device between the 501 and cable modem |
| 14:47.18 | triple-e | this guy insisted on running issolated cable's for the phones |
| 14:47.19 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 14:47.21 | motu | how can I get into the s extension, without dialing anything with xlite? |
| 14:47.21 | sevard | you have to mark the rtp traffic and the signaling traffic high priority |
| 14:47.48 | triple-e | rtp is the onlything on the network |
| 14:47.56 | triple-e | this guy really is the customer from hell |
| 14:48.04 | sevard | if all you're doing is VoIP then it's not a QoS problem |
| 14:48.07 | Hmmhesays | so kick him in the nuts and tell him to stfu |
| 14:48.33 | Greek-Boy | How can i setup recording for all extensions? |
| 14:48.33 | Hmmhesays | cable modem huh? |
| 14:48.34 | sevard | QoS only comes into play if you have other traffic on the network, if you prioritize VoIP traffic on a VoIP only network.. it's useless |
| 14:48.41 | Greek-Boy | i'm trying to use sox and soxmix to record mp3 |
| 14:48.46 | Hmmhesays | traceroute to your destination |
| 14:48.58 | triple-e | there is another lan the pc lan that is using the pix as GW |
| 14:48.58 | Greek-Boy | exten => s,1,SetVar(CALLFILENAME=i${CALLERIDNUM}-${TIMESTAMP}) |
| 14:48.59 | Greek-Boy | exten => s,2,Monitor(wav,${CALLFILENAME},m |
| 14:49.09 | noname32 | anyone got recomedations for hard phones? i am lookign for a phone with buttons i can program for example *1 record *2 attend tran *70 park so far i have heard good things about ploycom any one else got an option? |
| 14:49.31 | dlynes | sevard: yeah...i'm running into the same problem....the three sipura units aren't even behind a firewall, and they have their own dedicated 2.5Mb ADSL connection, but still call quality issues |
| 14:49.37 | sevard | noname32: I recommend the Aastra 480i (the CT if you want wireless extensions) |
| 14:49.59 | sevard | dlynes: WTF |
| 14:50.08 | dlynes | noname32: polycom's good, aastra's fine, too |
| 14:50.25 | dlynes | sevard: yeah, no kidding |
| 14:50.27 | triple-e | dlynes: who's your provider |
| 14:50.39 | dlynes | triple-e: Soho Skyway (Skyway West) |
| 14:50.44 | Greek-Boy | I used the example at http://www.voip-info.org/wiki/view/Monitor+stereo-example but there is something i'm not getting right |
| 14:50.54 | dlynes | triple-e: not that you'd know who that is, but.... |
| 14:50.55 | sevard | dlynes: pick apart each packet |
| 14:51.13 | dlynes | sevard: yeah...the biggest problem is |
| 14:51.27 | dlynes | sevard: they won't give me an accurate description of the call quality issues |
| 14:51.37 | dlynes | sevard: they just say "call quality sucks" |
| 14:51.37 | sevard | dlynes: well that's they're issue isn't it |
| 14:51.53 | sevard | dlynes: if your client won't provide you with a detailed bug report that's their problem |
| 14:52.00 | triple-e | dlynes: i spent days with tethereal trying to find something wrong with telasip |
| 14:52.05 | sevard | how can you fix it when you don't even know what's happening |
| 14:52.23 | dlynes | sevard: no, it's our problem, because we're footing hte bill for the dsl, and the dsl installation bill, and the labour costs, .. |
| 14:52.26 | Hmmhesays | dylnes I avoid customers like that |
| 14:52.37 | sevard | for all you know it's the analoge phone attached to the ata |
| 14:52.38 | noname32 | sevard, can you program the keys on tha aastra? and do like line apearances? |
| 14:52.42 | dlynes | Hmmhesays: you dont know they're customers like that, until after they become customers |
| 14:52.48 | Hmmhesays | untrue |
| 14:52.50 | dlynes | sevard: It's a Mitel PBX |
| 14:52.53 | Hmmhesays | i'm a pretty good judge of character |
| 14:53.17 | dlynes | noname32: yes |
| 14:53.18 | sevard | noname32: I don't know what you mean by line appearences but you can program all of the softkeys via tftp + configuration files |
| 14:53.43 | dlynes | noname32: softkeys are configurable via autoprovisioning and/or web page |
| 14:53.45 | noname32 | see if some one is currently on the phone |
| 14:53.54 | redondos | Trying to register a SIP account with a softphone gives me a "403 forbidden". But the asterisk console (with a very high verbosity -- something like 100000) doesn't show a single thing. What could be happening? |
| 14:54.02 | sevard | I highly recommend against using the web interface to program the Aastra |
| 14:54.09 | noname32 | cool i bough a gxp 2000 and was mad when i relized i could not program keys |
| 14:54.15 | dlynes | noname32: You can configure up to 10 accounts on the aastra 9133i's...not sure about the 480i, or the 480iCT |
| 14:54.31 | noname32 | and i know my users = dumb and cant handle *2 ext for trans hehe |
| 14:54.39 | sevard | dlynes: you can have 9 lines on the 480i CT with four hard buttons and 24 softkeys |
| 14:54.58 | noname32 | we will only be using 1 account per phone i think |
| 14:55.06 | dlynes | sevard: dood...why is a more expensive phone capable of one less line? |
| 14:55.12 | sevard | then the Aastra 480i is a bit overkill |
| 14:55.26 | sevard | dlynes: ? |
| 14:55.28 | dlynes | noname32: you only need the 9112i then |
| 14:55.45 | dlynes | sevard: 480i is capable of 9 lines, but the 9133i is capable of 10 lines |
| 14:55.47 | sevard | dlynes: you mean the comparison between the 480i and the 480 CT? |
| 14:56.21 | sevard | dlynes: I've never used the 9133i but from talking to a guy who has the quality of the 480i is 1,000x better than the 9133i |
| 14:56.32 | sevard | the CT can have four programmable wireless handsets |
| 14:56.46 | dlynes | sevard: no idea...I've had almost zero problems with the 9133i |
| 14:56.49 | noname32 | thanks you guys it is much apreacted |
| 14:56.56 | dlynes | sevard: the customers are quite impressed with them |
| 14:56.58 | sevard | dlynes: no, i have no doubts it's an awesome phone |
| 14:57.26 | sevard | dlynes: I myself have never used them so I can't give a personal account on it. However, I do know that the 480i is probably the best phone I've ever used in my entire life |
| 14:57.32 | sevard | and that's saying a _lot_ :) |
| 14:57.40 | *** join/#asterisk Kokey (n=ubunture@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 14:57.52 | dlynes | sevard: yeah...i've heard another guy on here that berates the Aastras every chance he gets, too :) |
| 14:57.59 | dlynes | I've heard both ends of the scale |
| 14:58.15 | sevard | The only issues i've ever heard about the aastra phones are the price |
| 14:58.16 | dlynes | He can't understand why myself, or his partner for that matter like them so much |
| 14:58.16 | Greek-Boy | nobody can help me? |
| 14:58.20 | Greek-Boy | :P |
| 14:58.24 | dlynes | Greek-Boy: with? |
| 14:58.35 | dlynes | sevard: Yeah...they're too cheap :) |
| 14:58.42 | Greek-Boy | http://www.voip-info.org/wiki/view/Monitor+stereo-example |
| 14:58.44 | dlynes | sevard: Much cheaper than the polycoms |
| 14:58.53 | sevard | dlynes: they have awesome weight to them, you really get what you pay for. i mean.. it's expensive. but they feel like real phones with excellent sound and excellent weight |
| 14:58.56 | Greek-Boy | dlynes: I looked at that page on the wiki but i'm doing something wrong |
| 14:59.01 | sevard | I thought the polycoms were more expensive |
| 14:59.11 | doolph | there's anyway todebug dtmf incoming? |
| 14:59.12 | dlynes | Greek-Boy: you trying to record both sides of a phone conversation? |
| 14:59.17 | Greek-Boy | yes |
| 14:59.22 | Greek-Boy | with sox and soxmix |
| 14:59.23 | hrhrhr | daft question... what is wiki abbreviated for/mean |
| 14:59.28 | sevard | hrhrhr: hahaha |
| 14:59.30 | Greek-Boy | i setup the script as instructed |
| 14:59.35 | Greek-Boy | i changed permissions on the script |
| 14:59.36 | hrhrhr | im serious :o |
| 14:59.37 | Hmmhesays | totalfark needs more boobies today |
| 14:59.40 | dlynes | Greek-Boy: that's the old asterisk 1.0.x method |
| 14:59.41 | redondos | So... why could registration be failing everytime? I just get a 203 Forbidden response back from the server. |
| 14:59.42 | sevard | hrhrhr: wiki it :) |
| 14:59.45 | dlynes | Greek-Boy: Use mixmonitor |
| 14:59.45 | [TK]D-Fender | dlynes : Only cheaper with your inside pricing, and geography may play a large factror as well. |
| 14:59.49 | doolph | there's anyway todebug dtmf incoming? |
| 14:59.50 | hrhrhr | dayam you |
| 14:59.52 | hrhrhr | :P |
| 14:59.57 | sevard | :D hrhrhr you made my day |
| 15:00.01 | Greek-Boy | oh |
| 15:00.05 | Greek-Boy | dlynes: thanks |
| 15:00.21 | hrhrhr | .Wiki wiki. means "rapidly" in the Hawaiian language |
| 15:00.25 | dlynes | Greek-Boy: yeah...in the 1.2.7.1 release, mixmonitor is quite stable |
| 15:00.27 | hrhrhr | there we go |
| 15:00.36 | Greek-Boy | dlynes: when i refer to extension s does it mean all extensions? |
| 15:00.58 | dlynes | Greek-Boy: read the wiki for special extensions...one sec, and i'll get you a link |
| 15:01.15 | sevard | hrhrhr: that's awesome. you know that thought never crossed my mind |
| 15:01.21 | Greek-Boy | thanks |
| 15:01.42 | hrhrhr | it's been puzzling me a while actually |
| 15:01.46 | sevard | haha |
| 15:02.13 | hrhrhr | suddenly everything turned into a wiki overnight *wtf is a wiki* |
| 15:02.41 | sevard | Am I right by thinking the TE405P is the _only_ four port PRI card for 5.0 volt PCI slots? |
| 15:02.47 | sevard | from digium |
| 15:03.15 | dlynes | Greek-Boy: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 15:03.35 | dlynes | sevard: maybe...one sec...I'll see what mine is |
| 15:04.06 | Greek-Boy | dlynes: thanks bud |
| 15:04.40 | *** join/#asterisk Lucas| (n=mmcguire@193.111.227.220) |
| 15:05.33 | shiznatix | I have this line: exten => 333,1,GotoIfTime(0:0-23:55|*|2-31|apr?queue_testing,s,1:outgoing_local,333,1) but it does not go to outgoing_local,333,1 ever even if the time is wrong |
| 15:05.35 | shiznatix | why? |
| 15:06.48 | dlynes | sevard: yeah...that's probably what mine is....TE4xxP |
| 15:07.27 | Greek-Boy | dlynes: how do I get mixmonitor to do mp3? |
| 15:07.31 | sevard | but the 405 is the only four port 5 volt |
| 15:07.32 | sevard | right? |
| 15:07.34 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-220-183.telkomadsl.co.za) |
| 15:07.43 | Hmmhesays | trying to resolve a domain name that doesn't exist is cool |
| 15:07.51 | dlynes | no idea, but that's what mine is....four port 5 volt, te4xxp |
| 15:08.08 | dlynes | It's a PCI-X 5 volt |
| 15:08.30 | coppice | PCI-X is never 5V |
| 15:08.55 | sevard | coppice: I have a 5 volt PCO-X (eXtended) not Express |
| 15:08.57 | dlynes | ah....just know it's 5V, and the only slots I've got that support 5V are the 64-bit PCI slots |
| 15:09.16 | sevard | s/PCO/PCI/g |
| 15:09.21 | dlynes | I thought it was called PCI-X |
| 15:09.55 | *** part/#asterisk sfollo81 (n=stefano@81.88.224.6) |
| 15:10.02 | dlynes | btw, coppice |
| 15:10.13 | dlynes | Does spandsp 0.0.3 not work with asterisk? |
| 15:10.24 | sevard | PCI-X == PCI eXtended not express |
| 15:10.53 | coppice | PCI-X runs at 100 or 133MHz and is always 3.3V |
| 15:11.18 | sevard | and a regular PCI card will plug into a PCI-X slot even though it doesn't look like it'll fit. I have a 5.0 volt PCI-X slot on my 1U and I have a TDM400P which is a 5volt PCI card and it fits in there and works great off it |
| 15:11.35 | sevard | coppice: that can't be right based on what I am using right now. |
| 15:11.37 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:11.39 | coppice | spandsp-0.0.3 says it is for development users only, and it says that for a reason. nonetheless it works if you have the right version of app_rxfax.c |
| 15:11.54 | dlynes | ah...ok |
| 15:12.05 | dlynes | so otherwise just use the latest 0.0.2 then |
| 15:12.16 | dlynes | gotcha |
| 15:12.34 | dlynes | I'll try using 0.0.3 at the office then |
| 15:12.45 | vader-- | have any of you read the book "Practical VOIP Security" |
| 15:13.02 | coppice | i made a suitable version of app_rxfax.c available at one time, but peoplke kept whining, so I removed it |
| 15:13.35 | dlynes | seems kinda silly people would whine if you're donig them a favor |
| 15:13.39 | dlynes | people are strange sometimes |
| 15:13.52 | anthm | that is the motto here |
| 15:13.57 | Ahrimanes | people are strange, customers/users even more so |
| 15:13.57 | dlynes | btw, this PCI-X bus is 66MHz, if that makes any difference |
| 15:14.28 | coppice | 66MHz is always 3.3V. Only 33MHz slots run at 5V |
| 15:14.43 | sevard | Like I said, I have TDM400P which is a 5.0 Volt PCI card plugged into a 5.0 volt PCI-X slot on my 1U and it works. |
| 15:15.57 | dlynes | I just figured it was 5V by looking at the way the slot dividers were arranged |
| 15:16.31 | dlynes | I never bought the original server, so I didn't know for sure what voltage it was |
| 15:16.36 | coppice | TDM400P is a universal card |
| 15:17.49 | sevard | from the digium website: this card only works in a 5volt slot |
| 15:17.50 | noname32 | hey sevard question about the aastra phones can you change the operation of some of the predifined buttons? |
| 15:18.29 | dlynes | noname32: no |
| 15:18.37 | sevard | not that I know of |
| 15:18.50 | noname32 | ok |
| 15:18.56 | coppice | sevard: where? the TDM400P is a double slotted card |
| 15:18.58 | dlynes | noname32: I have made the suggestion to the development team, though |
| 15:19.10 | noname32 | ya i would love that |
| 15:19.22 | triple-e | Sevard: qos ? |
| 15:19.49 | noname32 | i got an astrict question to is there a patch or way to make it so that when a call is parked it displays it on the phone screen? |
| 15:23.00 | sevard | coppice: what |
| 15:23.06 | sevard | triple-e: huh |
| 15:23.08 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 15:23.23 | coppice | where does it say 5V only? the card is double slotted |
| 15:23.43 | triple-e | ok sorry |
| 15:24.26 | triple-e | sevard: i was asking if you had an opnion on the QoS thing we were talking about earlier. tell them to turn qos on in their pix ? |
| 15:24.59 | triple-e | sevard: i need this guy to refer me to his friends -- so i don't wont to tell him to stfu |
| 15:25.37 | sevard | triple-e: all he is doing is VoIP on his line right? |
| 15:25.44 | triple-e | no |
| 15:26.05 | triple-e | two networks converge at pix 1phone 2pc |
| 15:26.28 | triple-e | would be a perfect if he just got a second dsl |
| 15:27.59 | key2 | How can I manage to get asterisk play a sound once the callee has hangup |
| 15:28.21 | sevard | coppice: maybe I'm on crack |
| 15:28.22 | key2 | can I just do a Dial() then a Play ? |
| 15:28.28 | *** join/#asterisk mikefoo (n=mikefoo@64.124.169.2) |
| 15:28.53 | SpaceBass | arrruuuuuggg voipsupply.com still hasn't shipped my new wifi phone...its been over a month |
| 15:28.53 | mikefoo | Anyone in nyc area? We need a few voip techs to employe |
| 15:29.07 | Ahrimanes | mikefoo: how much do you pay ;)I |
| 15:29.19 | mikefoo | Ahrimanes: where you located? |
| 15:29.35 | Ahrimanes | mikefoo: denmark at the moment, but might be willing to relocate ;) |
| 15:29.37 | sevard | hahaha |
| 15:29.42 | Lucas| | hah :) |
| 15:29.45 | mikefoo | hah, naah |
| 15:29.47 | triple-e | ha |
| 15:29.50 | mikefoo | looking for native nyc people, heh |
| 15:29.55 | sevard | nah? dude he's willing to relocate |
| 15:29.58 | Ahrimanes | oh darnit |
| 15:30.05 | sevard | NATIVE |
| 15:30.06 | sevard | hahahaha |
| 15:30.13 | sevard | new york and native do not belong in the same sentence |
| 15:30.26 | mikefoo | native as is, knowing the area, obviously.. |
| 15:30.33 | C4T3l | mikefoo: live in houston, tx but visited NY what's the pay$$ will reloc |
| 15:30.34 | sevard | heh |
| 15:30.36 | Ahrimanes | sevard: native nyc = mix of spanish, chinese, indian etc..? |
| 15:30.39 | mikefoo | no one is born in nyc, they just move here :) |
| 15:30.54 | triple-e | lol |
| 15:31.02 | sevard | mikefoo: I've been to times square and spent a month in nyc, does that count? |
| 15:31.07 | mikefoo | hah |
| 15:31.20 | triple-e | whats the rate |
| 15:31.24 | SpaceBass | i spent a month in times square one night....that was a rough one |
| 15:31.32 | sevard | ... |
| 15:31.37 | Ahrimanes | a month in one night? |
| 15:31.41 | sevard | wtfshit |
| 15:31.41 | mikefoo | you discuss that with hiring.. |
| 15:32.06 | sevard | mikefoo: so you come into #asterisk offering jobs but don't know how much you're willing to pay? |
| 15:32.20 | sevard | come on man at least lie to us. |
| 15:33.08 | mikefoo | I am just letting it known that we need some more tech support staff.. |
| 15:33.17 | mikefoo | If you are interested I forward information on to you.. |
| 15:33.27 | triple-e | who you work for mikefoo |
| 15:33.28 | sevard | tech support staff != voip tech |
| 15:33.51 | mikefoo | voip company = voip tech support = voip tech |
| 15:34.09 | sevard | voip tech == installing and maintaining backbone stuff, you're looking for phone support. |
| 15:34.17 | mikefoo | voip tech is pretty broad term, how can you say that it doesn't make something, hah. |
| 15:34.35 | mikefoo | support DOES install |
| 15:34.44 | SpaceBass | I know Genworth Financial is looking for a head VoIP/telephony engineer |
| 15:34.45 | mikefoo | DOES maintain |
| 15:34.53 | sevard | I know myself and most techs I knwo would NOT want to do phone support for end users |
| 15:35.11 | triple-e | im looking for a guy to help me in St Louis |
| 15:35.18 | triple-e | but im not going to tell him what he gets paid |
| 15:35.23 | triple-e | :-) |
| 15:35.26 | Lucas| | haha |
| 15:35.29 | sevard | hahaha |
| 15:35.41 | Lucas| | just give him a bowl of fruit every week? |
| 15:35.53 | mikefoo | I am letting it be known we have position, if interested you tell me, I get you intouch with right people.. simple |
| 15:36.05 | mikefoo | why would you stretch it to something its not? |
| 15:37.26 | SpaceBass | here's the senior job from Genworth: http://genworth.apply2jobs.com/index.cfm?fuseaction=mExternal.showJob&RID=60461&CurrentPage=2 |
| 15:37.41 | SpaceBass | there is also a team lead: http://genworth.apply2jobs.com/index.cfm?fuseaction=mExternal.showJob&RID=60460&CurrentPage=2 |
| 15:38.04 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:39.49 | dlynes | heh...when i was desperate for work and applied for a tech support job, they told me i knew too much about the technology...they were afraid of intimidating the customer |
| 15:40.05 | triple-e | but then you would have to move to richmond to work for genworth |
| 15:40.32 | triple-e | if you have only 1 year left to live -- move to richmond -- that will be the longest year of your life |
| 15:40.33 | sevard | dlynes: you're intimidating. |
| 15:40.35 | dlynes | ewww....working and living in the city of many spooks |
| 15:40.38 | mut | http://cahanes.com/templates/reales/re_dishomes.cfm?classid=249258&pids=79,80,81&secure=&moreinfotarget=_parent |
| 15:40.40 | coppice | richmond on thames is quite a nice area :-) |
| 15:40.45 | mut | my prospective house |
| 15:40.48 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:40.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:41.19 | SpaceBass | triple-e, you got a problem wiht Richmond? |
| 15:41.31 | triple-e | <-- commuted from stl to richmond to work for CapOne for a year |
| 15:41.45 | SpaceBass | I quite like Richmond |
| 15:42.04 | triple-e | lived in hotel on west side adjacent to office park |
| 15:42.05 | redondos | I just installed an E200P card. I am getting this error when starting asterisk -> http://pastebin.com/696309 |
| 15:42.15 | SpaceBass | well that would be f'ing miserable |
| 15:42.29 | triple-e | spent more time in strip clubs than i did anywhere else |
| 15:42.34 | triple-e | strip clubs sucked too |
| 15:42.39 | triple-e | lol |
| 15:42.42 | SpaceBass | I travel a LOT for work, I fell like 90% of the towns that I wrote off as shit holes probably had really nice areas too...but i never saw them |
| 15:42.54 | triple-e | Zactly |
| 15:42.55 | SpaceBass | va does suck for strip clubs....not even topless |
| 15:43.03 | triple-e | i know whats up with that |
| 15:43.22 | triple-e | St Louis -- crazy good strip clubs |
| 15:44.09 | SpaceBass | I swear as bad as it sounds....West Va has the best! |
| 15:44.23 | *** join/#asterisk wunderkin (n=kev@mmds-216-19-40-108.mm.az.commspeed.net) |
| 15:44.23 | coppice | what exactly is a non-topless strip club? :-\ |
| 15:44.30 | triple-e | Space: im scared.. truely scared |
| 15:44.36 | SpaceBass | lol |
| 15:44.38 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 15:45.12 | triple-e | big fat ugly toothless richmond chicks dancing on a pole while you drink 8 dollar soda |
| 15:45.34 | SpaceBass | dude....lay off the hometown insults :) |
| 15:46.16 | triple-e | <-- pissing people off today --- sorry everyone |
| 15:46.21 | coppice | oh, come on, for $8 they must have *some* teeth |
| 15:46.46 | SpaceBass | coppice, state law requires strippers to have at least 16 |
| 15:48.44 | triple-e | btw -- capitol one has their shit together for a non tech company |
| 15:49.00 | Qwell[] | triple-e: BAHAHAHAHA!! |
| 15:49.04 | Qwell[] | triple-e: HELL no |
| 15:49.17 | SpaceBass | I know few people who for CapOne in a tech role...they say its a good place to work |
| 15:49.24 | SpaceBass | I just live here in Ric...I work in Atlanta |
| 15:49.31 | Qwell[] | oh, place to work? meh, maybe |
| 15:49.40 | Qwell[] | but it's a shitty place to do business with, so I would never work there |
| 15:50.05 | triple-e | customer service and the quality/integrity of their technolgoy are issolated |
| 15:50.23 | Qwell[] | sorry, but they have poor tech too. |
| 15:50.32 | Qwell[] | They can't get a simple address change right |
| 15:50.37 | triple-e | lol |
| 15:50.55 | Qwell[] | (neither can Sprint, or pretty much any other company) |
| 15:51.01 | triple-e | i know nothing about that side of their business |
| 15:51.15 | Qwell[] | address change == tech |
| 15:51.31 | Qwell[] | thus, capital one has poor tech |
| 15:52.35 | sevard | Qwell[]: http://pastebin.ca/53220 |
| 15:52.49 | redondos | Here's zapata.conf, zaptel.conf and the logs when asterisk starts up. For some reason, it's not being able to find my E200P card (zaptel module is loaded). Any suggestions? http://pastebin.com/696329 |
| 15:53.59 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 15:54.52 | russellb | redondos: did you run ztcfg after you loaded the module? |
| 15:55.14 | redondos | ztcfg says: ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 15:55.33 | russellb | do you see the card with lspci? |
| 15:55.56 | C4T3l | whats the output of lsmod |
| 15:56.42 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:57.09 | russellb | oh look, there's a digium support guy now! |
| 15:57.11 | *** join/#asterisk apardo (n=apardo@87.217.146.98) |
| 15:57.55 | redondos | I see the card with lspci. |
| 15:58.05 | redondos | modules: |
| 15:58.06 | redondos | zaptel 189060 0 |
| 15:58.07 | redondos | crc_ccitt 2176 1 zaptel |
| 15:59.18 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 16:00.31 | *** join/#asterisk apardo (n=apardo@87.217.146.98) |
| 16:01.27 | russellb | redondos: you don't have the module for the card loaded ... modprobe wcte2xxp |
| 16:01.52 | russellb | redondos: if you need further assistance, contact support@digium.com |
| 16:01.59 | redondos | russellb: Oh, I see. Thanks. |
| 16:02.44 | redondos | hehe |
| 16:03.01 | redondos | Just a question: can I have both an e200p and an x100p card working at the same time? |
| 16:03.08 | Qwell[] | redondos: sure |
| 16:03.29 | redondos | I just need to make sure they use different channels and that's it? |
| 16:03.49 | redondos | Also, the E200P card-- it doesn't use the zaptel module, does it? |
| 16:07.37 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:07.42 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 16:15.11 | ManxPower | redondos, Of ot |
| 16:15.40 | ManxPower | redondos, If it's a genuine Digium card, it will use zaptel.o/zaptel.ko and the card specific driver, as listed in the Zaptel README |
| 16:22.10 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 16:23.20 | *** join/#asterisk keyhack (n=keyhack@68.236.93.224) |
| 16:24.18 | keyhack | I love Asterisk! AMAZING! :-) |
| 16:24.31 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 16:24.32 | Hmmhesays | settle down n00b |
| 16:25.24 | *** join/#asterisk drfoomod2 (i=DrFooMod@ool-43501d9f.dyn.optonline.net) |
| 16:28.08 | *** join/#asterisk marcus2 (i=marcus@atlantis.outer.org) |
| 16:29.08 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:30.20 | file | pfft |
| 16:32.13 | *** join/#asterisk Netgeeks (n=chris@68-185-24-8.static.mdfd.or.charter.com) |
| 16:32.34 | Netgeeks | good morning (west coast here) folks. |
| 16:33.39 | starlein | anyone using asterisk with compiled "-DBUSYDETECT" flag? |
| 16:33.52 | jsharp | Darn west coasters. Always behind the times. |
| 16:34.42 | sevard | What can I do to stop my SIP phones from going status 'UNREACHABLE' |
| 16:35.03 | sevard | my Aastra 480i CT doesn't but all of my SIP 2002s and my HOP-1002 do |
| 16:35.13 | sevard | look for a keepalive in the client somewhere? |
| 16:35.45 | Hmmhesays | people with no troubleshooting skillz rock |
| 16:36.31 | *** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 16:38.17 | [TK]D-Fender | sevard : "qualify=yes" |
| 16:38.27 | sevard | that is set. |
| 16:38.31 | Hmmhesays | so don't set it |
| 16:38.37 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:38.39 | [TK]D-Fender | sevard : the the phone behind NAT relative to *? |
| 16:38.49 | generalhan | whats goin on everyone !? |
| 16:38.51 | sevard | Dude, if I don't set it the phones just stop responding without the error |
| 16:38.55 | Hmmhesays | "guy goes to the doctor and says it hurts when I do 'this'" doc says "don't do that" |
| 16:39.07 | sevard | [TK]D-Fender: yeah |
| 16:39.35 | sevard | Hmmhesays: I am troubleshooting mo*#&$ |
| 16:39.37 | [TK]D-Fender | sevard : Also set "nat=yes" for the phone, and there may be something to set on the phone as well. |
| 16:39.38 | coppice | "masochist goes to the doctor and says it hurts when I do 'this'" doc says "I imagine it does" |
| 16:39.54 | sevard | [TK]D-Fender: nat is yes |
| 16:39.56 | [TK]D-Fender | coppice : just the person I'm looking for..... |
| 16:40.14 | coppice | i'm going to bed now |
| 16:40.27 | [TK]D-Fender | coppice : I'm about to compile SpanDSP for the first time, and just wanted to verify something. |
| 16:40.35 | [TK]D-Fender | coppice : time for 2 quick questions? |
| 16:40.52 | coppice | ./configure |
| 16:40.54 | coppice | make |
| 16:40.56 | coppice | make install |
| 16:41.54 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 16:42.44 | [TK]D-Fender | do I need to recompile * or just restart it to load an .so? |
| 16:42.48 | CoffeeIV_ | The link to asterisk-sounds download on asterisk.org doesn't seem to have the asterisk-sounds in there. Where should I get it ? |
| 16:44.31 | coppice | you need to compile app_rxfax and app_txfax. that is all |
| 16:45.35 | Qwell[] | CoffeeIV_: You mean this asterisk-sounds, from asterisk.org? http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1.tar.gz |
| 16:47.33 | Hmmhesays | faxing ownz joo |
| 16:48.03 | generalhan | i need some help with some Monitor() questions, i have all my incoming calls to any extension recorded, and it is working great, but i have an issue when i dial all of the extensions at once (like though a queue). It records 1 second of the call on ALL the lines but no matter which line picks up, the call is not recorded anywhere. |
| 16:48.12 | Hmmhesays | 4 days no guitar my fingers are freaking out |
| 16:48.16 | generalhan | this is how i have it set up to record :: http://generalhan.pastebin.ca/53236 :: |
| 16:48.36 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 16:48.41 | backblue | hi all |
| 16:49.35 | CoffeeIV_ | Qwell: yes, thanks -- the download page had a link to the parent directory of that link you just gave |
| 16:49.44 | sevard | it's too bad the sipuras don't run like a hamachi vpn or something on them so you could get to the conf webserver behind nat |
| 16:50.52 | Hmmhesays | openvpn |
| 16:50.54 | Hmmhesays | all i have to say |
| 16:50.55 | generalhan | anyone have any ideas on how i can get it to wait until the call is answered to decide where to record too ? |
| 16:53.10 | Hmmhesays | get what to wait |
| 16:53.24 | Hmmhesays | gnomes? dracula? a tricycle? |
| 16:53.24 | sevard | Hmmhesays: yeah, but when you don't control where the ATA is or what NAT it's behind without a tftp you can't change settings |
| 16:53.32 | tzafrir_laptop | where can I buy a simple ISDN card? preferebly a HFC-s PCI card? |
| 16:53.38 | Hmmhesays | that's why you always use tftp config |
| 16:53.42 | Hmmhesays | or something similar |
| 16:53.44 | tzafrir_laptop | Kind of hard to get here in Israel |
| 16:53.50 | sevard | Hmmhesays: right. dweeb. |
| 16:54.16 | generalhan | Hmmhesays: i was asking about the recording ... i have it set to record the inbound calls, but when i ring all the lines it cant figure out where to record |
| 16:54.19 | tzafrir_laptop | (I've already searched for a while) |
| 16:54.58 | Hmmhesays | use local channels |
| 16:55.09 | generalhan | what do you mean ? |
| 16:55.20 | Hmmhesays | case of bud please |
| 16:55.34 | generalhan | oh god ... thats not fair ... if i could get you one i want one too |
| 16:55.37 | generalhan | stupid work |
| 16:55.54 | Hmmhesays | i'm dealing with a special breed of idiot right now on messenger |
| 16:56.00 | generalhan | haha |
| 16:56.19 | generalhan | special breed huh ? |
| 16:56.50 | Hmmhesays | ok explain what your problem is with that dialplan |
| 16:57.54 | Hmmhesays | cause I don't see any issue |
| 16:58.04 | Hmmhesays | unless there is something in there you didn't paste |
| 16:58.21 | generalhan | well it works GREAT when they are reciving calls from a direct extension dial. or a transfer. but when i want to have all of them ring from the queue using local/7103@extension-dial, and i have 15 of them ringing it records 1 sec to each persons folder but doesnt record the call after that |
| 16:58.51 | Hmmhesays | 15 ring, how many answer? |
| 16:59.17 | generalhan | 1 |
| 16:59.27 | Hmmhesays | ok |
| 16:59.45 | generalhan | but it doesnt record the call into that persons folder, or any folder for that matter |
| 17:00.06 | generalhan | it puts a 44kb file with that phone number in each person's folder and thats it |
| 17:00.22 | Hmmhesays | so you want me to fix it for you |
| 17:00.26 | Hmmhesays | is that it? |
| 17:00.28 | generalhan | lol |
| 17:00.29 | generalhan | no |
| 17:00.43 | generalhan | i need some ideas on how to make it wait till some one picks up to record |
| 17:00.48 | doolph | I am offering to pay $10-$25 anyone that can help me fix a dtmf problem |
| 17:00.59 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:01.07 | Hmmhesays | i'm going to go have a smoke and if its still slow here I'll give you a pointer generalhan |
| 17:01.11 | Hmmhesays | doolph what problem |
| 17:01.12 | generalhan | k |
| 17:01.13 | generalhan | thanks |
| 17:01.22 | Hmmhesays | after I ask this guy what the issue is |
| 17:01.28 | triple-e | MOH is really choppy - im trying to figure out how to see if i have an irq conflict-- how do i see that ? |
| 17:01.42 | doolph | my did provider gives me a SIP did |
| 17:02.09 | doolph | then I call in, it is supposed to send to a context to a calling card system, and ask me for a number |
| 17:02.19 | Hmmhesays | k |
| 17:02.35 | doolph | when I dial it just doesn't like my numbers |
| 17:02.47 | Hmmhesays | doesn't accept them at all? |
| 17:03.00 | Hmmhesays | asterisk@home or plain |
| 17:03.39 | doolph | doesnt accept them at all |
| 17:03.43 | Hmmhesays | asterisk@home or plain |
| 17:03.48 | doolph | freepbx |
| 17:03.53 | generalhan | lol |
| 17:03.58 | doolph | I am not using aah |
| 17:03.58 | Hmmhesays | give me access, i'll fix it |
| 17:04.04 | doolph | ok |
| 17:04.10 | Hmmhesays | i'm low on beer money |
| 17:04.17 | Hmmhesays | pm it |
| 17:04.18 | SpaceBass | lol |
| 17:04.53 | doolph | heh |
| 17:10.41 | sevard | Does anyone use chanspy? |
| 17:12.29 | generalhan | i do |
| 17:13.18 | generalhan | why ? |
| 17:15.19 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 17:17.43 | sevard | can I see your chanspy line in your extensions.conf? |
| 17:17.52 | Qwell[] | s,1,ChanSpy() |
| 17:18.21 | sevard | i just get beeps ;\ |
| 17:19.11 | noname32 | whats chanspy? |
| 17:19.39 | wasim | it lets you monitor an existing channel |
| 17:20.01 | noname32 | ahh |
| 17:20.28 | noname32 | does anyone know how to make a beeping for record lines lol? |
| 17:21.16 | CoffeeIV_ | Playback(beep) |
| 17:21.33 | noname32 | but will it be continues? |
| 17:21.39 | Qwell[] | noname32: could pay somebody to add it to app_record |
| 17:21.54 | CoffeeIV_ | continuous ? no |
| 17:22.03 | triple-e | sevard: im having music on hold cutting out on this box im building , any ideas |
| 17:22.06 | noname32 | ya thats the prob |
| 17:22.11 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:22.34 | Qwell[] | noname32: tell dlynes to put up $50, and you put up $50, and it'll get taken care of in short order |
| 17:22.38 | noname32 | we have trading deskings and normal standard when you call you hear a faint beep in the background of call |
| 17:23.02 | Qwell[] | I think it was dlynes that needed it.. |
| 17:23.03 | Hmmhesays | nothing like tsp's unreliably transmitting dtmf |
| 17:23.11 | noname32 | lol |
| 17:23.23 | sevard | triple-e: umm |
| 17:23.34 | noname32 | well i am still not sure if we going to move tradding off pots lines yet |
| 17:23.34 | sevard | triple-e: does regular voice work alright? |
| 17:23.48 | Qwell[] | this stock trading? |
| 17:23.50 | noname32 | ok brb lunch |
| 17:23.51 | triple-e | yes |
| 17:23.51 | noname32 | yes |
| 17:23.54 | Qwell[] | yeah, keep it analog... |
| 17:23.54 | Netgeeks | will $50 in monopoly money (the old board style) work? |
| 17:24.03 | noname32 | yes Qwell |
| 17:24.04 | Qwell[] | Netgeeks: No, $500 in that case |
| 17:24.33 | sevard | so you'll take $100 real monies or $500 fake monies |
| 17:24.33 | noname32 | lol haha dude i have a full it staff here its not an issue to have it done |
| 17:24.47 | noname32 | but i just getting idea for best method |
| 17:24.48 | noname32 | s |
| 17:24.58 | Qwell[] | sevard: No. $1000 monopoly money |
| 17:25.15 | Qwell[] | 10x inflation for foreign currency |
| 17:25.18 | sevard | triple-e: I don't know man, i'm not very good with moh, i have issues too. are you running ztdummy module? |
| 17:25.48 | triple-e | current version mpg123 .59r -- with a Digium wildcard |
| 17:25.52 | sevard | Qwell[]: how is monopoy money foriegn currency. it's not even currency |
| 17:25.56 | Qwell[] | sevard: pfft |
| 17:26.02 | sevard | triple-e: jebus |
| 17:26.20 | sevard | triple-e: are you using the stock mp3s or custom |
| 17:26.22 | dlynes | what did i need? |
| 17:26.23 | X-Gen | Monopoly money > Zimbabwean Dollar |
| 17:26.24 | brodiem | triple-e, are you testing with the mp3s that came boxed |
| 17:26.25 | Qwell[] | sevard: It's the national currency for the monopoly islands |
| 17:26.26 | triple-e | stock |
| 17:26.33 | brodiem | triple-e what codec |
| 17:26.41 | triple-e | ulaw |
| 17:27.00 | sevard | wtf you shouldn't have any issues at all unless you're running it on a 386 |
| 17:27.10 | triple-e | could it be the hyperthreading p4 ? |
| 17:27.21 | brodiem | triple-e is moh only choppy over zap, i.e. what about sip only? |
| 17:27.28 | triple-e | or the disk's irq |
| 17:27.44 | sevard | triple-e: are you noticing any irq conflicts? |
| 17:27.49 | triple-e | havn't tried zap -- |
| 17:27.57 | triple-e | how do i see irq conflicts |
| 17:28.00 | dlynes | I was wanting a long beep on recording? |
| 17:28.23 | triple-e | cat /proc/interrupts |
| 17:28.25 | dlynes | Qwell[]: ? |
| 17:28.28 | brodiem | triple-e do you hear the moh normal but just choppy, or does it sound like you can just hear enough to know there is noise in the background? |
| 17:28.51 | triple-e | it cuts in and out |
| 17:29.12 | Qwell[] | dlynes: dunno, didn't you say you live in a state that requires a beep every x seconds, while recording? |
| 17:29.26 | dlynes | dlynes: nope...I don't live in a state |
| 17:29.30 | dlynes | erm |
| 17:29.35 | sevard | he lives up in da nort wuds |
| 17:29.37 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 17:29.37 | dlynes | s/dlynes/Qwell[] |
| 17:29.37 | Qwell[] | must've been somebody else :p |
| 17:30.04 | brodiem | triple-e are you positive it's using .59r? do you have another version installed in a different location? i.e. check /usr/bin/mpg123 and /usr/local/bin/mpg123 |
| 17:30.14 | dlynes | Even if I did, if I needed it bad enough, I'd probably just write the code for it |
| 17:30.29 | sevard | i'm dlynes i can do anything |
| 17:30.44 | file | Qwell wouldn't dare hurt me |
| 17:30.48 | triple-e | double checked and certain |
| 17:30.49 | dlynes | Nah...most of that asterisk code is too damned confusing to understand |
| 17:31.04 | dlynes | It's a big huge mess |
| 17:31.35 | Qwell[] | file: ORLY?! |
| 17:31.40 | Qwell[] | file: Is that a dare? |
| 17:31.47 | file | yes, it is |
| 17:31.53 | Qwell[] | buahahaha |
| 17:32.00 | Qwell[] | I know where you'll be next week :P |
| 17:32.08 | file | yeah, some place where you won't be |
| 17:32.18 | Qwell[] | I know people |
| 17:32.20 | mog_work | next youll be telling me linux kernel code is confusing |
| 17:32.26 | Qwell[] | mog_work: such nubs |
| 17:32.48 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:32.50 | triple-e | anyone done the thing where you move the system to raw |
| 17:32.55 | mog_work | its not a nub thing but anything of that size is big |
| 17:33.55 | mog_work | and difficult to read at first |
| 17:33.55 | sevard | quote "anything of that size is big" |
| 17:33.55 | sevard | you sir are a scientist! |
| 17:33.55 | brodiem | triple-e: lsof -p `ps ax |grep mpg123 | head -n 1 | awk '{print $1}'` |
| 17:34.19 | brodiem | triple-e when it tells you the path its using for mpg123, do strings /path/to/mpg123 | grep 59 |
| 17:34.42 | brodiem | and verify its 0.59r |
| 17:34.52 | Qwell[] | strings? Why? |
| 17:34.54 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:35.06 | ManxPower | why not just do mpg123 -v |
| 17:35.07 | Qwell[] | mpg123 | grep -i version |
| 17:35.16 | brodiem | doesn't matter |
| 17:35.19 | brodiem | will accomplish the same thing.. |
| 17:35.21 | ManxPower | and Asterisk only looks in /usr/bin and /usr/local/bin IIRC |
| 17:35.33 | sevard | K.I.S.. |
| 17:35.34 | sevard | K.I.S.S. |
| 17:35.37 | sevard | :\ |
| 17:35.38 | *** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net) |
| 17:36.35 | dlynes | mog_work: Yeah...I just need some spare time that I can devote to trying to understand the code |
| 17:38.10 | mog_work | yup |
| 17:38.45 | triple-e | brodiem: mpg123 is not running -- mpg123 -v shows 0.59r |
| 17:38.59 | triple-e | whats playing vm if mpg123 isn't running |
| 17:39.20 | brodiem | if * is running mpg123 should be running |
| 17:39.30 | ManxPower | 1.2 does not require mpg123 |
| 17:40.12 | ManxPower | 1.2 supports several new ways of doing MOH |
| 17:40.45 | brodiem | ManxPower even if the moh points to a .mp3 file and you don't specify a different player? |
| 17:41.46 | ManxPower | brodiem, all that stuff should be documented in musiconhold.conf.sample or on the Wiki. |
| 17:41.52 | *** join/#asterisk dextro (n=dextro@cpe-70-116-10-201.austin.res.rr.com) |
| 17:41.59 | ManxPower | I have not personally switched away from 1.0 style MOH |
| 17:42.52 | brodiem | triple-e if you're running * as non-root verify that the user has exec perms on mpg123 |
| 17:43.34 | ManxPower | and verify it has permissions to read the MP3 files, etc. |
| 17:43.47 | ManxPower | All those things the Wiki lists on the page about running Asterisk as non-root |
| 17:44.15 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 17:45.43 | brodiem | look at the debug info also since it should print out any problems if its trying to launch mpg123 andcan't |
| 17:47.54 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 17:47.58 | diclophis | howdy all |
| 17:48.28 | diclophis | so.. does anyone know how to terminate a call after X duration? |
| 17:48.57 | Qwell[] | diclophis: there is an option to Dial() |
| 17:49.02 | Qwell[] | show application dial |
| 17:49.23 | diclophis | mm, well what about if the call is inbound, and running straight into some AGI code |
| 17:49.23 | diclophis | ? |
| 17:49.40 | sevard | while [ 1 ] ; do clear; asterisk -rx "sip show peers; sleep 3 ; done |
| 17:50.04 | sevard | heh |
| 17:50.06 | fourcheeze | how near to the bleeding edge are people using for production? |
| 17:50.10 | sevard | man, i'm so tired |
| 17:50.25 | timscott | Well, 1.2.7.1 blows massive chunks. |
| 17:50.25 | fourcheeze | anyong using the svn trunk? |
| 17:50.28 | timscott | So I use 1.2.5 |
| 17:50.29 | sevard | fourcheeze: i'm using 1.2.7.1 |
| 17:50.43 | fourcheeze | aha |
| 17:50.45 | timscott | 1.2.7.1 craps out after like 100 simeotaneous calls, it won't open any new SIP channels |
| 17:50.53 | fourcheeze | nasty |
| 17:50.54 | sevard | timscott: oh? 1.2.7.1 is way more stable for me |
| 17:50.59 | brodiem | really |
| 17:50.59 | brodiem | wow |
| 17:50.59 | timscott | :/ |
| 17:51.14 | sevard | 1.2.5 wouldn't let some phones register until i rebooted it like 5 times |
| 17:51.16 | timscott | stable yeah, but I can't put more than ~100 calls through on a box that used to serve like 400 |
| 17:51.19 | generalhan | Hmmhesays: you have any more pointers for me ?? |
| 17:51.22 | timscott | I had to downgrade to 1.2.5 to get the calls through |
| 17:51.24 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:51.25 | sevard | I don't average more than 50 sim. calls though |
| 17:51.35 | drfoomod2 | is there a way via the Manager API to get a list of extensions? |
| 17:51.35 | diclophis | i am running 1.2.5 |
| 17:51.48 | timscott | i dunno, maybe it's just be |
| 17:51.49 | timscott | *me |
| 17:52.04 | sevard | timscott: i'd go with what you're saying as I've never reached that call volume |
| 17:52.09 | timscott | half the time, I dunno whats really going on, I just know what I'm observing |
| 17:52.21 | Hmmhesays | sorry, was helping doolph |
| 17:52.24 | Hmmhesays | he better not stiff me |
| 17:52.27 | fourcheeze | I need a quick way to route calls between multiple servers using realtime |
| 17:52.28 | generalhan | lol |
| 17:52.29 | sevard | do you get a sig hup? |
| 17:52.30 | generalhan | thats cool |
| 17:52.30 | timscott | hee |
| 17:52.31 | fourcheeze | long term I'll be using SER |
| 17:52.49 | diclophis | so yea.. how can i enforce call limits? |
| 17:52.49 | generalhan | Hmmhesays: did you get him all squared away ? |
| 17:52.53 | fourcheeze | but for now I just want to use 2 asterisks for redundancy reasons |
| 17:53.02 | Hmmhesays | his provider isn't sending dtmf correctly |
| 17:53.05 | generalhan | i see |
| 17:53.07 | fourcheeze | any ideas? |
| 17:53.13 | diclophis | particularly for inbound calls |
| 17:53.17 | sevard | i've nevr connected more than one * server instead of iax trunks, is it pretty hard? |
| 17:53.19 | brodiem | drfoomod2 show manager commandSIPpeers |
| 17:53.34 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 17:53.45 | generalhan | Hmmhesays: well i have been playing with a couple of things, putting the monitor command after the dial instead of before .. i just have no idea what else to try |
| 17:53.48 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 17:53.52 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 17:54.15 | Hmmhesays | I'll help you |
| 17:54.19 | Hmmhesays | feel free to donate when I'm done |
| 17:54.27 | generalhan | !! |
| 17:54.29 | ManxPower | diclophis, the best way is to bill by the min |
| 17:54.47 | *** join/#asterisk arbius (n=arbius@c-67-173-45-34.hsd1.il.comcast.net) |
| 17:55.30 | generalhan | Hmmhesays: im really starting to think that the only way to get this done properly is with agents ... which i REALLY dont want to do |
| 17:55.36 | Hmmhesays | no |
| 17:55.40 | Hmmhesays | I have a way |
| 17:55.49 | Hmmhesays | but I need a smoke badly and I have to finish with a customer |
| 17:56.00 | diclophis | yea, i have the billing and stuff, the problem is that a call that idles for an extended period of time stalls the database connection for my AGI server... thus preventing any more calls coming in |
| 17:56.21 | generalhan | Hmmhesays: thats fine ... here is how im calling it via queues.conf though .. just so you have that to reference later ! http://generalhan.pastebin.ca/53260 |
| 17:56.39 | Zodiacal | anyone know what i should put as the channel when creating a .call file to test calling of internal voip phones? |
| 17:57.36 | diclophis | the channel would be the [name] in your sip.conf file no? |
| 17:57.36 | sevard | What do people here use to connect manager? |
| 17:57.40 | wasim | IAX2/phone |
| 17:57.53 | diclophis | or SIP/phone |
| 17:57.56 | [TK]D-Fender | generalhan : you shouldn't have "Voicemail" in the dialplan logic of a called agent.... |
| 17:58.23 | generalhan | [TK]D-Fender: why not ? |
| 17:58.41 | [TK]D-Fender | generalhan : if the agent doesn't respond then the call doesn't get passed to the next agent |
| 17:59.05 | [TK]D-Fender | therefor defeating the poing of being in a queue |
| 17:59.10 | generalhan | [TK]D-Fender: well its not set up like roundrobin or anything its ringall |
| 17:59.26 | Zodiacal | dclophis, sevard, wasim Thank You! ill try SCCP/EXT. |
| 17:59.38 | generalhan | and most of my queues are not setup this way ... this is a test that im working on from when you and i were talking about a way around callwaiting |
| 17:59.57 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 18:00.25 | generalhan | when i finally get the recording part figured out i will have a context for voicemails in the queue |
| 18:00.58 | *** part/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 18:02.22 | [TK]D-Fender | generalhan : so it rigs 4 different people, eachwith their own VM box... so who gets it? Thats not even a queue... it never ever rigs twice at all! |
| 18:02.45 | generalhan | what ? |
| 18:02.48 | [TK]D-Fender | generalhan : may as well have done Dial(Local/1&Local/2&Local/3) |
| 18:02.55 | generalhan | no |
| 18:03.07 | generalhan | cause the client gets to hear hold music while they wait for some one to get to them |
| 18:03.24 | generalhan | and calls are passed out in the order they were recieved .. there are many reasons why i didnt just do a Dial() |
| 18:03.29 | [TK]D-Fender | if no-one answers, the it hits VM after nobody answer according to that macro. |
| 18:03.45 | *** join/#asterisk MrDigital (n=VBDIGITA@pool-72-81-113-227.phlapa.east.verizon.net) |
| 18:03.53 | [TK]D-Fender | Yeah I guess it queues them, but they get 1 chance before the next caller in line. |
| 18:04.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:04.22 | generalhan | in this one test sit. yes ... but not in my production queues |
| 18:05.29 | generalhan | [TK]D-Fender: my production queues look like this .. http://generalhan.pastebin.ca/53264 |
| 18:05.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:06.17 | [TK]D-Fender | generalhan : no declared strategy, announcements, etc? |
| 18:06.28 | generalhan | yes |
| 18:06.39 | generalhan | but you would laugh if you saw how i had it setup so i left it out ! lol |
| 18:06.42 | generalhan | you wanna see it ? |
| 18:06.47 | x86 | what is OSP? |
| 18:06.48 | marcus2 | hm |
| 18:06.52 | x86 | what is it used for? |
| 18:06.55 | marcus2 | when is someone gonna make a bluetooth-enabled ip phone |
| 18:06.59 | [TK]D-Fender | generalhan : Nah, I've had a good amount of humour already today |
| 18:07.14 | [TK]D-Fender | marcus2 : There are smart-phones out there for that already |
| 18:07.14 | keyhack | Is there some way I can make asterisk pick from a set of outbound VoIP accounts based on some SQL query factor or something? Basically I have a few voip accounts all with 4,000 mins a month, and I want outbound calls to go out in the line with the most MINUTES_AVAILABLE (specified in a column). There may be other ways to do this, just looking for input as I google around |
| 18:07.21 | marcus2 | such as? |
| 18:07.21 | generalhan | [TK]D-Fender: here is it anyway !! http://generalhan.pastebin.ca/53265 |
| 18:07.34 | [TK]D-Fender | keyhack : Yes. |
| 18:07.45 | [TK]D-Fender | marcus2 : UT Starcom PPC 6700 |
| 18:08.00 | marcus2 | oh, no, sorry |
| 18:08.03 | keyhack | [TK]D-Fender: Can you point me in the right direction? |
| 18:08.03 | marcus2 | i meant a desk phone |
| 18:08.22 | *** part/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 18:08.49 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 18:08.51 | gbodemantv | hi all |
| 18:08.51 | [TK]D-Fender | generalhan : Well... its MOSTLY wasted lines, but a few things could be tweaked and you lost monitoring. |
| 18:09.01 | gbodemantv | so I am having a problem |
| 18:09.04 | [TK]D-Fender | marcus2 : Oh don't get picky with me now! |
| 18:09.41 | generalhan | [TK]D-Fender: what do you mean i lost monitoring ?? |
| 18:09.42 | *** join/#asterisk ToTo (n=ToTo@host62-231.pool870.interbusiness.it) |
| 18:11.32 | CoffeeIV_ | is there some place on the web or wiki that has the svn commands needed to checkout out asterisk, zaptel, libpri, etc |
| 18:12.25 | Qwell[] | CoffeeIV_: asterisk.org, download |
| 18:12.38 | Qwell[] | erm |
| 18:12.40 | Qwell[] | http://www.asterisk.org/download |
| 18:12.54 | CoffeeIV_ | I specifically want the lastest out of the repository, not the latest release |
| 18:13.07 | twisted[asteria] | http://www.asterisk.org/download |
| 18:13.14 | znoG | is this normal? when the SIP packet goes out to a Linksys device, on the way back the call ID has a "0" appended to it. For example: |
| 18:13.26 | *** join/#asterisk Xen^ (n=linux@202.5.145.58) |
| 18:13.29 | znoG | SIP packet TO the ATA: Call-ID: 03b24e1f0b1248d8018fe76725c14bf7@192.168.136.67 |
| 18:13.41 | znoG | SIP packet FROM the ata: Call-ID: 03b24e1f0b1248d8018fe76725c14bf7@192.168.136.670 |
| 18:13.46 | twisted[asteria] | hahaha |
| 18:13.50 | twisted[asteria] | .670? |
| 18:13.55 | twisted[asteria] | you must be joking. |
| 18:14.02 | generalhan | lol |
| 18:14.08 | znoG | and Asterisk retransmits the packet, presumably because it couldn't match the call-ids!? |
| 18:14.12 | keyhack | [TK]D-Fender: What am I looking for? |
| 18:14.12 | generalhan | you have the BEST netowrk EVER ! |
| 18:14.16 | generalhan | .670 ! |
| 18:14.28 | znoG | yeah, who knows why it is appending the 0 |
| 18:14.29 | [TK]D-Fender | keyhack : You'll need to make a very custom AGI script for it. |
| 18:14.31 | carrar | thats IPv4.5 |
| 18:14.38 | twisted[asteria] | znoG, beat the ATA with a cane |
| 18:14.45 | [TK]D-Fender | keyhack : in AGI you can do whatever you want with your DB etc... and the choose where to dial |
| 18:14.45 | znoG | you think it is the ATA too? |
| 18:14.55 | znoG | i'm thinking the call ID should be the same |
| 18:14.59 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 18:15.02 | twisted[asteria] | znoG, well, tcpdump it |
| 18:15.09 | znoG | ethereal, good idea |
| 18:15.11 | twisted[asteria] | znoG, if it's the ATA, you'll see the packet with the 0 at the end before asterisk |
| 18:15.16 | znoG | yeah |
| 18:15.29 | gbodemantv | xlite for linux not making any sound when ringing? |
| 18:15.30 | twisted[asteria] | if you get the packet back with the 0 at the end, beat the ATA with a cane. |
| 18:15.35 | keyhack | [TK]D-Fender: The AGI needs to run locally on the box because I think I read it uses STDIN and STDOUT for communication? Could you use the manager interface with TCP instead? |
| 18:15.36 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:15.39 | gbodemantv | just pops up on screen |
| 18:15.46 | gbodemantv | any idea how to change that |
| 18:15.55 | znoG | what seems to be happening is that asterisk keeps retransmitting to the ATA, i presume because of what I just mentioned.. not matching the call ID |
| 18:16.02 | twisted[asteria] | yeah |
| 18:16.06 | twisted[asteria] | the call ID's must match |
| 18:16.16 | twisted[asteria] | it's the only REAL identifier for that dialog |
| 18:16.23 | znoG | now to find out if the problem is the ATA or not .. |
| 18:16.40 | znoG | the strange thing is that it only seems to happen when dialing in via Zap |
| 18:16.49 | znoG | when dialing from extension to extension, it's fine. |
| 18:16.51 | twisted[asteria] | hmm |
| 18:17.03 | *** join/#asterisk saftsack (n=saftsack@p54A7F1F6.dip.t-dialin.net) |
| 18:17.12 | znoG | so it's probably something to do with Asterisk... time to ethereal it |
| 18:17.22 | [TK]D-Fender | keyhack : Possibly. I don't know the fine points |
| 18:17.30 | keyhack | [TK]D-Fender: Alright, thanks for the input |
| 18:17.42 | [TK]D-Fender | keyhack : Wish I knew more. |
| 18:17.48 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) |
| 18:18.27 | generalhan | [TK]D-Fender: im really interested in what "i lost monitor" means ... i want to have this as seamless as possible |
| 18:18.36 | keyhack | [TK]D-Fender: Well, I want my .NET application to really be able to say "Call out on VoIP line #17", which is account ZYX with say Broadvoice |
| 18:18.56 | *** join/#asterisk GolobTGG (n=GolobTGG@BSN-77-78-87.dsl.siol.net) |
| 18:18.59 | keyhack | [TK]D-Fender: But we're using this 3rd party app that doesn't let us specify details of the outbound line, so I think the logic is going to have to occur in the PBX |
| 18:20.15 | distortion | can someone recommend a t38 enabled ata device? It seems the sipura 2100 supports t38 |
| 18:20.40 | [TK]D-Fender | generalhan : You production queue doesn't do recordinglike the old oneused to it seems |
| 18:20.41 | redondos | I could install the E200P with the X100P at the same time. Thanks everyone who helped! (Qwell, russellb, ManxPower, [TK]D-Fender) |
| 18:21.16 | [TK]D-Fender | keyhack : Yup, which means your app will just place the call and ALL processing happens on your * box |
| 18:21.21 | generalhan | the "old one" is really my new one .. and i want it to record that way IF i can get it to work |
| 18:21.28 | keyhack | [TK]D-Fender: Which _sucks_, lol |
| 18:21.46 | [TK]D-Fender | *sigh* |
| 18:21.53 | keyhack | [TK]D-Fender: Because theres more to it than "round-robin" the different VoIP accounts, depends on destination too, which country we're calling, etc. etc. |
| 18:22.20 | [TK]D-Fender | keyhack : Yeah I know... several "weights" to consider in routing. |
| 18:22.31 | [TK]D-Fender | keyhack : All stuff that screams AGI and local DB's |
| 18:22.34 | VoicePulse | keyhack: You could generate the appropriate .call file and place it in the asterisk spool directory. I believe you can make it run an AGI and pass in some data (like which route to use). |
| 18:22.55 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 18:23.00 | redondos | Now I would like some guidelines about using the output that Asterisk generates in CSV to create usage reports. Perhaps a spreadsheet? Suggestions are welcome. :) |
| 18:23.35 | timscott | I'm trying to figure out the cdr_mysql module right now... |
| 18:23.36 | CoffeeIV_ | Qwell: I figured out that this command works: "svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk" |
| 18:24.17 | VoicePulse | keyhack: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 18:24.25 | Telamon | Anyone use Unlimitel for outgoing VOIP calls? I can't get the caller ID to work with them. I do SetCallerPres(allowed) and Set(CallerID(number)=9021234567) but their equipment only sees the 1234567, so it screws up the area code. |
| 18:25.19 | gbodemantv | the windows version rings just fine |
| 18:25.27 | Hmmhesays | boy that kids problem was crazy |
| 18:25.40 | Hmmhesays | anyone ever have double dtmf digits come in from a voip provider? |
| 18:25.47 | justinu|laptop | yep |
| 18:25.53 | Hmmhesays | what was the cause? |
| 18:25.58 | generalhan | oh really |
| 18:26.02 | justinu|laptop | there's some problem in asterisk's RFC2833 code, i think |
| 18:26.02 | generalhan | figured it out though huh ! |
| 18:26.15 | Hmmhesays | oh yeah? |
| 18:26.29 | justinu|laptop | yeah, happens to too many people |
| 18:26.35 | Hmmhesays | good cause I haven't got my smoke yet |
| 18:27.03 | generalhan | lol |
| 18:27.16 | *** join/#asterisk Thus0 (n=Thus0@86.73.49.22) |
| 18:27.23 | *** join/#asterisk tomcontr3 (n=gcontrer@49-76-246-201.adsl.terra.cl) |
| 18:27.28 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-136.apple.com) |
| 18:27.41 | tomcontr3 | hi, Im looking for someone that could help me configurating my asterisk... |
| 18:28.07 | tomcontr3 | I have been using freepbx, but, I have some problems, and some told me that I should try to config it manyally |
| 18:28.09 | eric_hill | what are you trying to configurate? |
| 18:28.22 | tomcontr3 | my trunks and local extentions |
| 18:28.29 | tomcontr3 | so the can make outbound calls |
| 18:28.45 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-136.apple.com) |
| 18:29.10 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:29.39 | tomcontr3 | eric_hill, do you think you can help me? |
| 18:30.46 | keyhack | VoicePulse: Checking it out now |
| 18:31.11 | tomcontr3 | ?? |
| 18:31.16 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 18:31.40 | [TK]D-Fender | tomcontr3 : describe your setup |
| 18:32.01 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:32.05 | tomcontr3 | you mean, what I have right now, or what I would like to do? |
| 18:32.34 | [TK]D-Fender | tomcontr3 : both. |
| 18:33.13 | keyhack | VoicePulse: Hmm, seems like it may work, I have to think about it more. (The call application doesn't reside on the same machine but I guess you can do this with the Manager API too) |
| 18:33.15 | znoG | twisted[asteria]: i think it was just Asterisk printing an extra 0.. the packets themselves have the right call-ID on the in/out |
| 18:33.27 | tomcontr3 | right now, I have 5 extentiones, 4 trunks, 4 incoudn routes and 4 outbound routes, but I can only use 1 |
| 18:33.31 | znoG | twisted[asteria]: .. but why.. why would Asterisk keep trying to invite when the ATA is replying with "ringing ..." ? |
| 18:33.52 | tomcontr3 | because a problem that some gut told me that was a configuration problem from freepbx |
| 18:34.18 | [TK]D-Fender | tomcontr3 : What kind of "trunks"? |
| 18:34.27 | tomcontr3 | SIP trunks |
| 18:34.50 | [TK]D-Fender | tomcontr3 : Describe them. Al from the same ITSP, just different lines? something else? |
| 18:35.24 | tomcontr3 | right, al from the same ITSP, but diferent lines |
| 18:36.02 | tomcontr3 | I mean, I can call to each of those trunks from a pstn line |
| 18:36.11 | [TK]D-Fender | tomcontr3 : with a roll-over for something like company use? |
| 18:36.32 | [TK]D-Fender | Wht do you use for your extensions? |
| 18:36.54 | tomcontr3 | sorry I dont understund the question, what do you mean by roll-over? |
| 18:38.18 | tomcontr3 | for my extentions I use SIP |
| 18:40.08 | *** join/#asterisk Renacor (n=kvirc@ip21.farheap.net) |
| 18:40.17 | [TK]D-Fender | tomcontr3 : What I mean is is each of those ITSP provided "lines" really independant of the others or are then in a telco-based hunt-group? And for your phones I mean exactly what are you using? |
| 18:41.11 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 18:43.04 | tomcontr3 | you mean if I use a Softphone? |
| 18:43.52 | CoffeeIV_ | when I compile the libpri from subversion trunk, I get the error message "chan_zap.c:73:2: error: #error "You need newer libpri"" |
| 18:44.39 | sevard | get newer. |
| 18:44.44 | tzanger | CoffeeIV_: that error message has to be one of Asterisk's EASIEST to understand errors |
| 18:44.50 | justinu|laptop | lol |
| 18:44.50 | tzanger | CoffeeIV_: what do you feel it's telling you to do? |
| 18:44.53 | sevard | gawd damn. |
| 18:45.02 | brodiem | lol |
| 18:45.18 | docelmo | man just make CoffeeIV_ feel even dumber than he does now |
| 18:45.19 | Zodiacal | anyone know of a phone that can list at least 5 calls on the screen at once? the cisco 7960 uses two rows of text per call, which only alows 3 calls to be displayed on the screen at once, its very anoying for receiptionists to sort out the calls using the up down keys... |
| 18:45.23 | CoffeeIV_ | I am trying to compile the newer libpri though |
| 18:45.30 | Qwell[] | Zodiacal: Get a 7914 |
| 18:45.42 | Qwell[] | those can display calls too, no? |
| 18:45.43 | Zodiacal | qwell does that show calls or just speed dials? |
| 18:45.45 | tzanger | CoffeeIV_: ok, and what error are you getting? |
| 18:45.49 | Qwell[] | dunno |
| 18:45.57 | CoffeeIV_ | see above -- that's the first error |
| 18:46.02 | Zodiacal | qwell i think its just speeddials.. :( |
| 18:46.10 | CoffeeIV_ | it's compiling chan_zap.c |
| 18:46.30 | tzanger | CoffeeIV_: I don't see a libpri error |
| 18:46.44 | tzanger | when you build and install the latest libpri, what do you get that prevents you from building or installing ti? |
| 18:47.30 | tomcontr3 | [TK]D-Fender, are you still there? |
| 18:48.34 | CoffeeIV_ | tzanger: the error message "chan_zap.c:73:2: error: #error "You need newer libpri"" is what prevents me from building the new libpri -- looking at the code, it's because of a #ifndef PRI_KEYPAD_FACILITY_TX -- is that something I have to specify on a ./configure command line or something ? |
| 18:48.58 | file | install the latest libpri. |
| 18:49.23 | Qwell[] | CoffeeIV_: chan_zap isn't IN libpri |
| 18:49.27 | Qwell[] | fix your checkout comand |
| 18:49.48 | file | libpri is a completely different... thing... |
| 18:49.56 | tzanger | CoffeeIV_: uh, chan_zap isn't in libpri |
| 18:49.59 | tzanger | libpri is *very* small |
| 18:50.08 | Qwell[] | $20 says he checked out zaptel to the libpri dir ;) |
| 18:50.46 | *** join/#asterisk terrapen_ (n=cjs@166.70.183.109) |
| 18:50.48 | CoffeeIV_ | I see channels/chan_zap.c in my libpri directory . . . I checked it out with the command "svn checkout http://svn.digium.com/svn/asterisk/trunk libpri" -- what command should I have used ? |
| 18:50.56 | Qwell[] | umm |
| 18:51.10 | Qwell[] | one that...you know...checks out libpri, instead of asterisk? |
| 18:51.23 | CoffeeIV_ | yes, which one is that ? |
| 18:51.28 | Qwell[] | ...libpri? |
| 18:51.34 | file | svn co http://svn.digium.com/svn/libpri/trunk libpri |
| 18:51.42 | CoffeeIV_ | ok, thanks |
| 18:51.44 | justinu|laptop | how about just download the release tarballs? |
| 18:51.51 | justinu|laptop | or do we need trunk for some reason? |
| 18:51.59 | CoffeeIV_ | I need the trunk for some reason |
| 18:52.02 | sevard | turn off your pc. |
| 18:52.03 | sevard | just do it |
| 18:52.05 | sevard | turn her off. |
| 18:52.16 | *** join/#asterisk jtodd (n=jtodd@reserve-64-79-115-18.wiline.com) |
| 18:52.28 | Hmmhesays | wow everything is asploding today |
| 18:52.31 | sevard | Katty isn't here to make a smart comment about me turning off women. |
| 18:52.38 | Hmmhesays | i do the same thing |
| 18:52.44 | Hmmhesays | with my spikey hair and rugged good looks |
| 18:53.08 | sevard | <PROTECTED> |
| 18:53.09 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 18:53.24 | sevard | s/hit/beat |
| 18:53.45 | Hmmhesays | funneh |
| 18:53.46 | tzanger | Qwell[]: yep, I've done that |
| 18:54.10 | Hmmhesays | ginormously hailarious in fact |
| 18:54.27 | keyhack | VoicePulse, [TK]D-Fender: I decided that it'd be easier to have a rule like "exten=>80001.,1,Dial(${EXTEN:5}@myvoipaccount1)" |
| 18:54.44 | keyhack | And then have my .NET application pick a prefix of "80001" and "80345" based on some business logic |
| 18:55.04 | keyhack | kinda like dialing 9 to get an outside line, you dial 8 and a 4 digit code to pick the specific outbound line |
| 18:55.32 | Qwell[] | keyhack: Your extension above won't work |
| 18:55.47 | keyhack | what? |
| 18:55.48 | Qwell[] | and even if you did fix it to do pattern matching, it's still very bad |
| 18:56.39 | keyhack | Qwell[]: Whats wrong with my logic? |
| 18:56.48 | generalhan | Hmmhesays: did you finally get your cig ? |
| 18:56.53 | Hmmhesays | yes |
| 18:56.53 | Qwell[] | keyhack: 1) It won't match anything but "80001." |
| 18:57.04 | tzanger | keyhack: you forgot a "_" |
| 18:57.11 | Qwell[] | 2) You have a huge security hole, waiting to be exploited, by using . |
| 18:57.15 | generalhan | Hmmhesays: lol i just came back in from having one and i was thinking how crappy it was that you still couldnt have one ! lol |
| 18:57.22 | Hmmhesays | ha |
| 18:57.23 | Hmmhesays | just got it |
| 18:57.26 | Hmmhesays | you figure your shiat out |
| 18:57.35 | keyhack | tzanger: What does the _ do again? |
| 18:57.40 | generalhan | Hmmhesays: no ... it doesnt make any sense to me at all |
| 18:57.43 | keyhack | Qwell[]: What is the huge security hole? |
| 18:57.44 | Qwell[] | pattern matching |
| 18:57.45 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
| 18:57.49 | Hmmhesays | aight |
| 18:57.58 | Qwell[] | keyhack: allowing anybody to dial internationally, for one |
| 18:57.59 | tzanger | keyhack: if you're doing ANY pattern match, you need to start the extension with "_" |
| 18:58.15 | keyhack | Qwell[]: Well, considering this is a system only used by my software, I'm not too concerned about it |
| 18:58.21 | Assid | heya tzanger, Qwell, Hmmhesays |
| 18:58.22 | keyhack | tzanger: Alright, thanks |
| 18:58.28 | Assid | sup tkd! |
| 18:58.28 | Hmmhesays | hey |
| 18:58.49 | Qwell[] | keyhack: and how do phone numbers get put into the system? |
| 18:58.50 | Hmmhesays | paste that link with your dialplan again |
| 18:58.54 | Qwell[] | My bet is on user input |
| 18:58.56 | generalhan | k |
| 18:59.25 | keyhack | Qwell[]: What? The phone numbers that are dialed out are dialed by my program |
| 18:59.35 | Qwell[] | yes, and how does your program get the numbers? |
| 18:59.45 | keyhack | Qwell[]: From a local DB |
| 18:59.52 | Qwell[] | and how does the local DB get the numbers? |
| 19:01.10 | keyhack | Qwell[]: The user specifies it through a webapp which is verified by the system for sanity and called to verify the authenticity of the user |
| 19:01.19 | Qwell[] | I rest my case. |
| 19:01.28 | generalhan | Hmmhesays: http://generalhan.pastebin.ca/53277 |
| 19:01.51 | keyhack | Qwell[]: And your concern is what again? |
| 19:02.52 | generalhan | Hmmhesays: the comments in the macro is what im hoping to accomplish by all this ... if i can get the recording figured out i will turn that on so that anyone on a call wont be disturbed by a stupid beep (since there is no way to turn off the call waiting on my phones) |
| 19:03.46 | *** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com) |
| 19:04.21 | Hmmhesays | you want to use monitor or mixmon |
| 19:04.26 | tomcontr3 | can any one help me configurim my asterisk, with SIP 4 SIP trunks from a same provider?= |
| 19:04.55 | generalhan | Hmmhesays: i would use which ever works the best |
| 19:05.00 | Hmmhesays | ok |
| 19:06.17 | keyhack | Qwell[]: You thinking they can type in "911" and make my system call 911 or something? |
| 19:06.49 | keyhack | Qwell[]: We have international phone accounts, and the number is verified and parsed, and the appropriate voip account will call them, so international calling is not a concern of mine either |
| 19:07.05 | key2 | What do I have to do if I want the communication to go through asterisk in SIP, and not having asterisk to tell the two SIP phone to communicate between themself |
| 19:08.16 | jaiger | key2, I *think* canreinvite=no or something like that |
| 19:08.39 | blitzrage | key2: canreinvite=no |
| 19:08.49 | *** join/#asterisk hydride (n=nathan@HSE-Montreal-ppp140355.sympatico.ca) |
| 19:09.16 | hydride | can anyone help me set up a basic asterisk setup so I can call from sip phone to sip phone? |
| 19:09.27 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 19:10.08 | hydride | I want to do more, but I'll play with it later when I got time |
| 19:11.46 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 19:12.21 | C4T3l | you need to config sip.conf and extensions.conf for your sip phones |
| 19:12.30 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
| 19:12.52 | hydride | what do I need in my sip.conf? |
| 19:12.58 | justinu|laptop | stuff |
| 19:13.11 | hydride | no kidding |
| 19:13.21 | C4T3l | thats a very vague question what type of phones?? |
| 19:13.34 | hydride | I have to zyxell 2000w |
| 19:13.42 | hydride | and one machine is gonna use x-lite |
| 19:14.16 | SpaceBass | hydride, check out archatechs.wordpress.com and nerdvittles.com |
| 19:14.21 | SpaceBass | lots of good start-up guides there |
| 19:16.26 | C4T3l | are both phones on the same network? |
| 19:17.19 | hydride | negative, I'm taking a 4 month military contract in bc, so I installed * on my VPS webhosting and am using that as the server |
| 19:17.43 | hydride | gonna have a zyxel with me in bc, a zyxel at my g/fs, and I'm installing x-lite on my mothers computer |
| 19:18.08 | *** join/#asterisk ddaeschl (n=ddaeschl@devit.rsaisp.com) |
| 19:18.45 | ddaeschl | Hello, has anyone sucessfully configured a TE205p on a T1 PRI? |
| 19:18.58 | harryvv | hydride are you in vancouver? |
| 19:19.04 | mikefoo | zyxel has by far been the best brand of wireless router I have used. |
| 19:19.11 | hydride | negative, will be in Victoria on Sunday though |
| 19:19.17 | harryvv | where are u now |
| 19:19.23 | hydride | Windsor Ontario |
| 19:19.25 | harryvv | i see |
| 19:19.36 | harryvv | how has that voip sip phone worked for you? |
| 19:19.49 | hydride | have not got it working yet |
| 19:19.56 | harryvv | problems? |
| 19:20.05 | hydride | yeah, I know nothing about asterisk :( |
| 19:20.14 | key2 | blitzrage: so I can put canreinvite=no in [general] in sip.conf ? |
| 19:20.18 | hydride | and I got 'till saturday evening to get this working |
| 19:21.12 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 19:21.27 | Blackthorn | Hi, is there anyone else having problems making calls thorugh nufone today? |
| 19:21.42 | SpaceBass | the zyxel wifi phone sucks |
| 19:21.44 | SpaceBass | i hate to say |
| 19:22.07 | hydride | SpaceBass, I'm not looking for anything of great quality |
| 19:22.17 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 19:22.21 | SpaceBass | well the sound quality is fine, its the missing featuires that bother me |
| 19:22.24 | SpaceBass | like WPA |
| 19:22.26 | hydride | I just want to make sip to sip calls from the same usual place |
| 19:22.28 | SpaceBass | so you have to run your AP in WEP |
| 19:22.38 | hydride | the wpa thing bugs me too |
| 19:22.44 | hydride | but I can live with wep |
| 19:22.49 | SpaceBass | so can most hackers :) |
| 19:23.01 | Qwell[] | please |
| 19:23.08 | SpaceBass | its also missing a hold, conf, transfer and flash |
| 19:23.09 | Qwell[] | wpa takes all of 2 seconds longer to crack than wep :P |
| 19:23.12 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:23.12 | hydride | yeah, but what kind of hacker cares about someones home computer? |
| 19:23.21 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 19:23.32 | Druken | afternoon everyone |
| 19:23.37 | twisted[asteria] | that kind |
| 19:23.38 | timscott | Good afternoon. |
| 19:23.46 | SpaceBass | anyone who wants to steal bandwidth for nefarious reasons |
| 19:23.54 | SpaceBass | sorry, I'm a little nuts about wifi security |
| 19:23.55 | justinu|laptop | anyone who wants a zombie network |
| 19:24.21 | hydride | security isn't a concern for me |
| 19:24.22 | SpaceBass | anyway, back to your zyxels...you should be able to do what you want easily |
| 19:24.37 | vader-- | anyone deal with the TDM2400P? |
| 19:24.39 | mikefoo | I don't use wpa or wep, I just allow access my mac addresses |
| 19:24.42 | vader-- | im getting this error upon boot |
| 19:24.46 | vader-- | Unable to do INITIAL ProSLIC powerup on module # |
| 19:24.49 | Qwell[] | mikefoo: also spoofable :P |
| 19:25.01 | twisted[asteria] | mac addresses are easily spoofable |
| 19:25.04 | SpaceBass | hydride, I'd suggest you use asterisk@home...will make it a lot easier |
| 19:25.06 | vader-- | <PROTECTED> |
| 19:25.07 | Ariel_ | mikefoo, ever heard of mac spoofing |
| 19:25.13 | Qwell[] | all they have to do, is sniff for the mac address over the air |
| 19:25.14 | SpaceBass | (and here is where I get flamed :) ) |
| 19:25.15 | mikefoo | Qwell: not going to be spoofable since one mac addy is always registered. |
| 19:25.28 | twisted[asteria] | vader--, did you remember to plug in the power cable? |
| 19:25.29 | mikefoo | you obviuosly can't login two macs |
| 19:25.33 | mikefoo | same time |
| 19:25.36 | vader-- | ya |
| 19:25.36 | Qwell[] | mikefoo: What is the router going to do, say "oh, no, I already have THIS mac." |
| 19:25.44 | mikefoo | yes, I tried it already. |
| 19:25.51 | SpaceBass | mac address lists are the worst possable security...in fact they are not secure...the traffic is still unencrypted |
| 19:25.54 | Qwell[] | No, it's gonna say "Hey Joe, you're already logged in...okay, here you go" |
| 19:26.01 | hydride | SpaceBass, unfortunatly I can't use asterisk@home :/ |
| 19:26.05 | Qwell[] | hell |
| 19:26.07 | Druken | the power plug is there for a reason, wouldn't ya say ? |
| 19:26.09 | SpaceBass | hydride why? |
| 19:26.09 | Qwell[] | You don't *NEED* to login :p |
| 19:26.17 | Qwell[] | "You" are already logged in |
| 19:26.25 | hydride | SpaceBass, cause I'm running * of a VPS running debian |
| 19:26.34 | mikefoo | ok :) |
| 19:26.36 | SpaceBass | ah |
| 19:26.39 | Ariel_ | hydride, then install freepbx |
| 19:26.42 | SpaceBass | you can still install freepbx |
| 19:26.44 | SpaceBass | LOL! |
| 19:26.48 | hydride | yeah I know, I tried |
| 19:26.50 | hydride | it's a pain in the ass |
| 19:26.51 | Ariel_ | SpaceBass, hello |
| 19:27.01 | SpaceBass | hey Ariel_ ! |
| 19:27.02 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 19:27.03 | *** join/#asterisk Muecke77 (n=muecke77@p54A9CE3F.dip.t-dialin.net) |
| 19:27.05 | Ariel_ | nothing free and good comes easy |
| 19:27.06 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 19:27.09 | gbodemantv | anyone gettoing this? |
| 19:27.24 | gbodemantv | format_wav.c:247 update_header: Unable to find our position |
| 19:27.33 | gbodemantv | just keeps looping in CLI |
| 19:27.53 | SpaceBass | hydride, then check out the asterisk handbook...its dated but will get you exactly what you need to set up two or three extens quickly |
| 19:28.15 | SpaceBass | then go ebay a $50 and load A@H :) |
| 19:28.17 | Assid | err.. can anyone try loading up this wav file into polycom 301 ? |
| 19:28.22 | Assid | www.pienotech.com/ctu.wav |
| 19:28.32 | harryvv | yea |
| 19:28.39 | Druken | a@h should only be used in home settings... impo |
| 19:28.43 | Blackthorn | Hi, is there anyone else having problems making calls thorugh nufone today? |
| 19:28.53 | SpaceBass | Druken, I think small business too |
| 19:29.00 | Druken | SpaceBass: i don't... |
| 19:29.06 | Assid | sup basss |
| 19:29.06 | harryvv | it can be used in a small bussiness enviroment. |
| 19:29.14 | SpaceBass | Hey Assid |
| 19:29.29 | harryvv | I would just take asterisk at home and cut out alot of the dialplan thats not needed. |
| 19:29.37 | C4T3l | what's the general consensus on asterfax? |
| 19:29.59 | OloBola | how do these wifi phones work..? Can you just punch in your Asterisk IP address or? |
| 19:30.02 | harryvv | btw, I can route incomming faxes out to a fax with asterisk? |
| 19:30.13 | Qwell[] | harryvv: sure |
| 19:30.30 | SpaceBass | OloBola, just like a hard phone |
| 19:30.38 | Druken | only restriction would be timing |
| 19:30.39 | OloBola | ok |
| 19:30.39 | SpaceBass | just use wifi for network rather than a wire |
| 19:31.00 | OloBola | that makes sense |
| 19:31.10 | hydride | thanks |
| 19:31.12 | harryvv | Qwell, I get a lot of request to either fax or recive fax info but dont have this setup. BTW, has the fax over the internet been resolved? |
| 19:32.04 | vader-- | any of you guys using a digium tdm2400p card? |
| 19:32.20 | Druken | vader--: why? |
| 19:32.25 | generalhan | Hmmhesays: you still here ? |
| 19:32.29 | Hmmhesays | yeah |
| 19:32.32 | vader-- | im wonderinw hat you put in your /etc/modules file |
| 19:32.42 | vader-- | if you put in wctdm or wctdm24xxp |
| 19:32.55 | harryvv | anyway i goto go. |
| 19:33.04 | Qwell[] | vader--: the latter |
| 19:33.15 | Hmmhesays | was having some softphone trouble |
| 19:33.16 | Druken | vader--: i would assume wctdm24xxp.... |
| 19:33.17 | vader-- | hmm i wonder why it would say it can't find the modules then |
| 19:33.23 | generalhan | any suggestions ? |
| 19:33.34 | Qwell[] | vader--: got a recent version of *? |
| 19:33.40 | vader-- | ya |
| 19:33.44 | Qwell[] | how recent? |
| 19:33.51 | vader-- | 1.2.7.1 |
| 19:34.01 | sevard | that's recent. |
| 19:34.12 | Qwell[] | and you have zaptel installed? |
| 19:34.17 | Qwell[] | what version? |
| 19:34.29 | *** join/#asterisk techie (n=gus@antibala.com) |
| 19:34.35 | vader-- | ya |
| 19:34.38 | vader-- | latest version |
| 19:34.41 | Qwell[] | which? |
| 19:36.36 | shido6 | how do you add a .so u want to compile in bsd? |
| 19:36.52 | shido6 | im in /usr/ports/net/asterisk/files/patch-apps::Makefile |
| 19:36.56 | sevard | when my boss comes in i'm going to squirt visine in my eyes and cry about missing him |
| 19:37.18 | vader-- | 1.2.5 |
| 19:37.27 | justinu|laptop | sevard, get your platform running again? |
| 19:37.33 | Qwell[] | vader--: Is this debian by chance? |
| 19:37.34 | vader-- | i just checked hte power connection and that might be the problem |
| 19:37.39 | vader-- | ya it's debian |
| 19:37.43 | sevard | justinu|laptop: yeah, i stayed up very last last night |
| 19:37.50 | Qwell[] | of course it is |
| 19:37.51 | sevard | justinu|laptop: that was hard :| |
| 19:38.00 | Qwell[] | vader--: ls -l /lib/modules/`uname -r`/misc/ |
| 19:38.02 | vader-- | the power is connected by the yellow pin in the extension cable i plugged in moved out slightly |
| 19:38.06 | Qwell[] | Is it empty? Of course it is... |
| 19:38.28 | vader-- | by the = but the |
| 19:38.33 | justinu|laptop | sevard: good deal... lesson learned, right? |
| 19:38.38 | vader-- | so im rebooting now ill see if that fixed it |
| 19:38.41 | justinu|laptop | copy your configs onto another machine, at least :) |
| 19:38.42 | Qwell[] | no |
| 19:38.44 | Qwell[] | vader--: it won't |
| 19:38.46 | sevard | Qwell[] the great and powerful Oz only asks questions with answers he already knows |
| 19:38.57 | Qwell[] | sevard: indeed |
| 19:39.04 | sevard | justinu|laptop: lesson learned 10000x over |
| 19:39.11 | vader-- | ok it found the modules |
| 19:39.12 | vader-- | yay |
| 19:39.23 | vader-- | it was the connection |
| 19:39.30 | Qwell[] | Then you gave us the wrong error |
| 19:39.42 | sevard | it's the end users fault, i swear! |
| 19:39.42 | Qwell[] | module not found is MUCH different than module could not be loaded |
| 19:39.54 | vader-- | i copied the exact error it threw at me |
| 19:40.12 | vader-- | Unable to do INITIAL ProSLIC powerup on module # |
| 19:40.15 | sevard | i'm willing to lay money that vader is right |
| 19:40.19 | vader-- | <PROTECTED> |
| 19:40.29 | Qwell[] | So, modprobe pondered? |
| 19:40.29 | Qwell[] | < vader--> hmm i wonder why it would say it can't find the modules then |
| 19:40.55 | Qwell[] | You never pasted that error before |
| 19:41.14 | vader-- | ya i did |
| 19:41.41 | Qwell[] | not in the last 15 minutes you didn't |
| 19:42.31 | vader-- | i dunno i thought i did |
| 19:42.42 | sevard | you didn't |
| 19:43.02 | _Sam-- | Qwell is always right, you will learn |
| 19:43.23 | sevard | Qwell[]: What's six by nine? |
| 19:43.43 | Qwell[] | _Sam--: indeed |
| 19:44.17 | timscott | 42? |
| 19:44.28 | sevard | FOURTY-TWO |
| 19:44.40 | timscott | :D |
| 19:44.46 | generalhan | ??? |
| 19:44.48 | sevard | the new champ |
| 19:44.58 | Hmmhesays | da da da da |
| 19:45.03 | Hmmhesays | ~seen generalhahn |
| 19:45.17 | jbot | i haven't seen 'generalhahn', Hmmhesays |
| 19:45.17 | sevard | timscott: i was at a trade show and i had to set up voip phones for this one co, so i named one trillian and another ford prefect |
| 19:45.17 | timscott | The ultimate answer, of life, the universe, and everything. |
| 19:45.17 | sevard | etc |
| 19:45.17 | timscott | HA |
| 19:45.19 | sevard | timscott: NOBODY got it |
| 19:45.20 | timscott | :( |
| 19:45.21 | sevard | I know |
| 19:45.27 | generalhan | Hmmhesays: haha |
| 19:45.35 | sevard | my life is a fricken waste of time |
| 19:45.37 | timscott | We are sorry for the inconvenience. |
| 19:45.40 | Hmmhesays | nice |
| 19:45.51 | timscott | I hated book 5. It was lammmeeee. |
| 19:45.57 | timscott | I wish I would have stopped at book 4. |
| 19:46.01 | sevard | timscott: I had the feeling nobody would even comment if I named a phone Don't Panic |
| 19:46.07 | Hmmhesays | so you want an answer to this? |
| 19:46.08 | timscott | Wow, zombies. |
| 19:46.17 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 19:46.25 | sevard | it actually made me depressed :| |
| 19:46.25 | generalhan | Hmmhesays: if you really know how to get it to moinotr correctly |
| 19:46.35 | sevard | it was the only thing i was looking forward to |
| 19:46.36 | sevard | heh |
| 19:46.39 | Hmmhesays | the way you want it to? |
| 19:46.54 | generalhan | Hmmhesays: haha ! yea .. i want it to work the way i want it to ! lol |
| 19:47.16 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 19:47.31 | sevard | timscott: the whole gang was there, arthur, zaphod, even the names of the mice that i can never recall offhand |
| 19:47.53 | sevard | should have named the PBX "Deep Thought" |
| 19:48.53 | Hmmhesays | i see |
| 19:49.07 | Strom_C | well at least they didnt name the PBX "Deep Throat" |
| 19:49.11 | generalhan | lol |
| 19:49.25 | jsharp | Your PBX sucks. |
| 19:49.31 | generalhan | HAHAHAHA |
| 19:49.37 | sevard | Your mom's PBX sucks. |
| 19:50.04 | vader-- | ok the tdm2400p recognized now the te110p isn't lighting up but linux says it found it |
| 19:50.06 | vader-- | :( |
| 19:50.17 | vader-- | TE110P: Setting up global serial parameters for T1 FALC V1.2 |
| 19:50.17 | vader-- | TE110P: Successfully initialized serial bus for card |
| 19:50.17 | vader-- | Found a Wildcard: Digium Wildcard TE110P T1/E1 |
| 19:50.23 | *** join/#asterisk xcoyote (n=farfan@dsl-201-144-0-184.prod-infinitum.com.mx) |
| 19:51.08 | xcoyote | question: after executing make progdocs , where does it set the documentation? /etc/ ??? somewhere ? |
| 19:52.50 | Hmmhesays | yeah this works |
| 19:53.01 | generalhan | oh yea ? |
| 19:53.19 | Hmmhesays | yeah |
| 19:53.21 | Hmmhesays | cause I'm awesome |
| 19:53.30 | generalhan | yea .. that sounds about right ! lol |
| 19:54.33 | xcoyote | does anyone know where make progdoc sets the documentation |
| 19:54.36 | C4T3l | anyone ever see this error before: config.c: parse error: No category context for line 14 of cdr_mysql.conf |
| 19:54.40 | Hmmhesays | i'll post it on my site in a few minutes |
| 19:54.52 | generalhan | what is your site ? |
| 19:55.07 | C4T3l | line 14 is my database host name |
| 19:55.46 | *** join/#asterisk IceManRISK (n=kart@201.66.80.69) |
| 19:55.49 | Qwell[] | xcoyote: I think it creates a doxygen/ dir |
| 19:56.25 | xcoyote | ok |
| 19:56.46 | sevard | wow, am i really this tired |
| 19:56.48 | sevard | `which asterisk` -rx "sip show peers" | grep -v Verbos; |
| 19:56.48 | sevard | Binary file (standard input) matches |
| 19:57.34 | Sebb | sevard: grep -va |
| 19:57.49 | Qwell[] | Sebb: I think you missed the problem :p |
| 19:58.05 | Sebb | Qwell[]: i didn't read ;) but that fixes that "binary file" foo ;) |
| 19:58.17 | vader-- | do you guys know any example zaptel.conf that has a te110p and a tdm2400 card configured? |
| 19:58.20 | sevard | i've never gotten that error |
| 19:58.24 | *** join/#asterisk xunil (n=wkurdzio@office1.visionpointsystems.com) |
| 19:58.27 | Sebb | 21:56:52 < sevard> Binary file (standard input) matches |
| 19:58.43 | sevard | infact that only happens with * |
| 19:59.22 | Hmmhesays | there you go generalhan |
| 20:00.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:00.39 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:01.29 | Hmmhesays | boobies |
| 20:01.35 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-97.buckeyecom.net) |
| 20:01.38 | sevard | pork chop sammiches |
| 20:01.44 | Qwell[] | yuck |
| 20:01.50 | Qwell[] | that doesn't sound good at all |
| 20:01.54 | sevard | oh look at you and your cute little hats |
| 20:01.56 | sevard | help computer |
| 20:02.13 | Dr-Linux | Hmmhesays: ufff i'm alone in room so please don't call such words :P |
| 20:02.33 | Hmmhesays | that always gets you riled |
| 20:02.36 | Hmmhesays | its so cute |
| 20:02.49 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 20:03.22 | Dr-Linux | Hmmhesays: yaeh, b/coz boobies are cute :P |
| 20:04.15 | Hmmhesays | some girls just need to eat a freaking sammich |
| 20:04.22 | *** join/#asterisk the_magic_bean (n=the_magi@cpe-24-166-27-13.indy.res.rr.com) |
| 20:04.49 | Dr-Linux | Hmmhesays: what about your girl? :) |
| 20:04.56 | gambolputty | Hi. Can someone dial me via SIP for a test? |
| 20:05.03 | Hmmhesays | she could probably use a sammich |
| 20:05.17 | Qwell[] | I could use a sammich right now |
| 20:05.19 | Dr-Linux | what the hell is sammich? |
| 20:05.26 | Hmmhesays | a sandwich |
| 20:05.38 | Qwell[] | a tasty mid-day treat |
| 20:05.48 | Hmmhesays | two or more slices of bread generally accompanied by something seperating them |
| 20:05.56 | Dr-Linux | Hmmhesays: oo i see, i thought something ... |
| 20:06.12 | Qwell[] | occasionally with liquid-like compounds, to add flavor |
| 20:06.19 | Hmmhesays | indeed |
| 20:06.39 | Hmmhesays | her knees are a little sharp, but i get over that |
| 20:06.46 | Dr-Linux | but what's sammich related to boobies? :S |
| 20:06.58 | Hmmhesays | use your imagination |
| 20:07.02 | Qwell[] | mmm... |
| 20:07.05 | Hmmhesays | they have those in pakistan right? |
| 20:08.03 | Dr-Linux | Hmmhesays: every girl own that :P |
| 20:08.09 | C4T3l | C4T3l is embarrased |
| 20:08.13 | Strom_C | no, pakistan banned their importation in 1997 :) |
| 20:08.38 | Dr-Linux | C4T3l: put a finger at your embarrsed place |
| 20:08.40 | Dr-Linux | hein |
| 20:08.47 | Dr-Linux | 1997? |
| 20:08.54 | Dr-Linux | what was happend? :) |
| 20:09.08 | Strom_C | <PROTECTED> |
| 20:09.17 | Strom_C | < Strom_C> no, pakistan banned their importation in 1997 :) |
| 20:09.23 | Hmmhesays | damnit |
| 20:09.23 | C4T3l | forgot to uncomment "[global]" in cdr_mysql.conf |
| 20:09.39 | vader-- | do you guys know what would make my te110p card stop working now that my tdm2400p card is registering |
| 20:09.44 | Strom_C | Dr-Linux: in reference to either "sammich" or "boobies" - take your pick |
| 20:09.47 | vader-- | linux is reporting that the card is recognized |
| 20:09.53 | vader-- | but it's not lighting up in the back |
| 20:10.03 | Strom_C | vader--: do you have zaptel.conf set up correctly? |
| 20:10.16 | vader-- | i believe so |
| 20:10.29 | vader-- | i am getting an error though when i run ztcfg -vvvc |
| 20:10.33 | Strom_C | in what order are the drivers loading? |
| 20:10.40 | Dr-Linux | :S |
| 20:10.51 | vader-- | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 20:10.57 | Dr-Linux | i came here with a question, but went for boobies . aww |
| 20:11.04 | *** join/#asterisk dlynes_ (n=dlynes@216.251.149.66) |
| 20:11.05 | justinu|laptop | Dr-Linux: how's the wife? |
| 20:11.11 | C4T3l | boobies will do that to ya |
| 20:11.16 | vader-- | well in my /etc/modules i load zaptel first, then wcte1xxp then wctdm24xxp |
| 20:11.38 | vader-- | but when linux boots it seems to load wxtdm24xxp first and then wcte1xxp next |
| 20:12.02 | Strom_C | vader--: so ok, make it such that channels 1-24 are your TDM card, and channels 25-48 are your T1 |
| 20:12.19 | Dr-Linux | justinu: she is abit angry, coudn't talk to her since 4 days |
| 20:12.24 | *** join/#asterisk stoffell_h (n=stoffell@d5153F9E0.access.telenet.be) |
| 20:12.31 | Dr-Linux | was much busy at work |
| 20:12.55 | justinu|laptop | :( |
| 20:13.14 | Qwell[] | Dr-Linux: Do you live in a palace yet? |
| 20:13.32 | vader-- | storm_c thats my zaptel.conf |
| 20:13.42 | Dr-Linux | Qwell[]: sorry friend i didn't understand your question , palace? |
| 20:13.48 | Qwell[] | much big house |
| 20:14.14 | Strom_C | vader--: ok one sec |
| 20:14.31 | Dr-Linux | Qwell[]: i live alone in a small room, in other state, basically i'm from tribals |
| 20:14.49 | Dr-Linux | my parents house is a big house |
| 20:14.58 | Qwell[] | alone? aren't you married? |
| 20:15.01 | Hmmhesays | some customers will never go away |
| 20:15.28 | Dr-Linux | Qwell[]: no, i'm not married yet, but i did Nikkah |
| 20:15.38 | nahirean | you did nikkah? |
| 20:15.43 | nahirean | how was it? |
| 20:15.58 | Dr-Linux | Qwell[]: hhm.. i don't know what you guys call Nikkah in English :S |
| 20:16.09 | [TK]D-Fender | Dr-Linux : Engagement? |
| 20:16.13 | Qwell[] | Dr-Linux: descript Nikkah |
| 20:16.15 | Dr-Linux | maybe justinu|laptop can explain, |
| 20:16.17 | Qwell[] | ..describe |
| 20:16.21 | justinu|laptop | it's like vows, sorta |
| 20:16.22 | justinu|laptop | a contract |
| 20:16.26 | justinu|laptop | (i think) |
| 20:16.30 | nahirean | ikkah is the contract between a bride and bridegroom and part of a Islamic marriage. Various traditions may differ in how nikkah is performed because different groups accept different texts as authoritative. Therefore, Sunnis will likely accept Bukhari hadith while Shi'ites will have their own collections thus producing different procedures. This contract requires the consent of both parties and allows both parties to add conditions. ... |
| 20:16.33 | nahirean | google ;) |
| 20:16.46 | Strom_C | vader--: what does ztcfg spit back at you? |
| 20:16.53 | Qwell[] | I..see.. |
| 20:16.59 | Qwell[] | add conditions? haha |
| 20:17.02 | vader-- | it reads back all the channels that are configured |
| 20:17.05 | vader-- | then at the end it says |
| 20:17.07 | vader-- | ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 20:17.17 | Qwell[] | Condition the first - "You cook. I eat." |
| 20:17.32 | Dr-Linux | i see |
| 20:17.45 | Dr-Linux | nahirean is kinda right |
| 20:17.48 | nahirean | Subcomponent: you earn your red wings once a month. :) |
| 20:17.50 | vader-- | here is my booting of the cards |
| 20:17.51 | vader-- | http://pastebin.ca/53297 |
| 20:18.01 | nahirean | Dr-Linux; I'm not right, google is ;) |
| 20:18.11 | justinu|laptop | condition 2: you wash the laundry, while I sit on the couch |
| 20:18.16 | Strom_C | vader--: I assume your PRI is using binary eight zero substitution and extended superframe, right? |
| 20:18.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:18.21 | vader-- | http://pastebin.ca/53298 |
| 20:18.38 | vader-- | here is the span config |
| 20:18.39 | vader-- | span=1,0,0,esf,b8zs |
| 20:19.23 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 20:19.41 | Dr-Linux | vader--: esf= framing and b8zs is coding 8 zeros |
| 20:20.01 | Dr-Linux | justinu|laptop: hows going your new married life? :) |
| 20:20.05 | Dr-Linux | is it cool? :) |
| 20:20.15 | Strom_C | no, b8zs means "eight zeros will be substituted with something else such that your T1 doesnt lose timing" |
| 20:20.34 | Dr-Linux | justinu|laptop: i dont' think you feel much difference after marriage? |
| 20:20.39 | vader-- | well it's not hooked up to a T1 right now |
| 20:20.44 | tzanger | B8ZS = two BPVs for eight 0s |
| 20:20.44 | vader-- | it's hooked up to a loop back plug |
| 20:20.47 | blitzrage | extended super frame! sexy :) |
| 20:20.55 | vader-- | it was working |
| 20:21.00 | vader-- | until the tdm2400p started to work |
| 20:21.08 | dlynes_ | blitzrage: yeah...it's telco geek speak for she's a fat cow :) |
| 20:21.31 | tzanger | T1s send zeroes as the lack of a pulse and ones as a pulse of the opposite polarity of the last pulse. This maintains a DC zero voltage which keeps the recovery circuitry happy. |
| 20:21.31 | Hmmhesays | LOL |
| 20:21.40 | Strom_C | vader--: I think this is the time you call Digium :) |
| 20:21.41 | Hmmhesays | fat cows need lovin' too |
| 20:21.47 | Hmmhesays | and after you can milk em |
| 20:21.55 | tzanger | with voice it was statistically impossible to keep a PCM channel at all zeroes for any length of time |
| 20:21.58 | tzanger | with data it's not |
| 20:22.08 | dlynes_ | yeah...fat chicks are like mopeds...fun to ride until your friends catch ya :) |
| 20:22.23 | Hmmhesays | so much hate |
| 20:22.42 | Dr-Linux | i'm thinking about my question again |
| 20:22.44 | tzanger | so they invented B8ZS as an "adder" to AMI... whenever the framer sees eight zeroes coming at it it sends out 00++00-- instead. The other side sees this very special form of a BPV and replaces it with eight zeroes instead of flagging an error |
| 20:22.56 | Hmmhesays | Dr-Linux: you can do anything at zombo.com |
| 20:23.43 | Dr-Linux | what the hell is zmbo, |
| 20:23.48 | Nugget | the infinite is possible at zombocom. |
| 20:23.48 | Hmmhesays | go there and see |
| 20:23.52 | tzanger | yeah I just read that on /. |
| 20:24.00 | Dr-Linux | Hmmhesays: wow i just got my question in my mind |
| 20:24.29 | Dr-Linux | well, i'm recording all outgoing calls, i set monitor on ouging zap channels |
| 20:24.44 | Dr-Linux | but now i don't wanna record all calls, but specific extensions |
| 20:24.47 | Dr-Linux | what should i do? |
| 20:25.01 | Dr-Linux | actually my all extensions are different |
| 20:25.09 | *** part/#asterisk xcoyote (n=farfan@dsl-201-144-0-184.prod-infinitum.com.mx) |
| 20:25.50 | *** join/#asterisk gr0mit_home (n=wendolen@extrt.txrx.org.uk) |
| 20:25.55 | Hmmhesays | set up monitoring on each extension |
| 20:26.21 | Dr-Linux | Hmmhesays: but i'm using pattern |
| 20:26.30 | tainted- | what is iax.conf equivalent of sip.conf's accountcode = |
| 20:26.42 | Hmmhesays | explicitly match the ones you want to monitor |
| 20:27.14 | Hmmhesays | set a dbkey , do a rain dance, there are a bunch of ways you can do it |
| 20:27.41 | *** join/#asterisk stoffell (n=PircBot@pot.catsanddogs.com) |
| 20:27.52 | Dr-Linux | i wish i can know damn dbkey things :( |
| 20:28.49 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:28.50 | Dr-Linux | hhm... |
| 20:29.17 | Dr-Linux | Hmmhesays: i understand the logic .. but i wanna record their incoming calls as well? |
| 20:29.31 | Hmmhesays | same logic appies |
| 20:29.34 | Hmmhesays | *applies |
| 20:31.08 | *** join/#asterisk busco_developer (n=root@OL33-83.fibertel.com.ar) |
| 20:32.49 | busco_developer | hi there, looking for developers @argentina |
| 20:33.01 | busco_developer | if someone interested, let me know |
| 20:33.19 | Hmmhesays | http://static.flickr.com/55/139430789_329f7bff7e_o.gif |
| 20:33.41 | tzanger | Hmmhesays: that is just not right |
| 20:33.58 | bkw_ | ok that makes even me sick |
| 20:34.08 | Hmmhesays | LOL, mission accomplished |
| 20:35.40 | bkw_ | Moussaoui is going to be put in prison for LIFE |
| 20:36.02 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:36.12 | C4T3l | bkw_: have they passed the sentence yet |
| 20:37.18 | bkw_ | yes |
| 20:37.22 | bkw_ | they are announcing it on CNN |
| 20:37.25 | bkw_ | he's not getting death |
| 20:37.34 | tzanger | wow destroying aircraft is punishable by death? |
| 20:37.43 | hydride | whenever I run asterisk -vvvvvc, it displays a bunch of information but doesn't seem to run, ie nothing regarding asterisk in ps -A, but I'm not getting any real hint to an error when I run asterisk, any help with where I should look? |
| 20:37.44 | *** join/#asterisk ToTo (n=ToTo@host62-231.pool870.interbusiness.it) |
| 20:38.01 | tzanger | hydride: it should tell you near the bottom of that list of information |
| 20:38.02 | Qwell[] | tzanger: isn't |
| 20:38.15 | tzanger | Three of the six conspiracy counts made him eligible for the death penalty: committing acts of terrorism transcending national boundaries, destroying aircraft and using planes as weapons of mass destruction. |
| 20:38.33 | keyhack | well |
| 20:38.35 | keyhack | he didn't get it |
| 20:38.37 | keyhack | life in prison |
| 20:38.59 | hydride | Asterisk Dynamic Loader Starting: |
| 20:39.00 | hydride | <PROTECTED> |
| 20:39.00 | hydride | <PROTECTED> |
| 20:39.00 | hydride | <PROTECTED> |
| 20:39.00 | hydride | <PROTECTED> |
| 20:39.00 | hydride | <PROTECTED> |
| 20:39.01 | tzanger | I was reading about lethal injection... 3 drugs... one for pain (?), one to paralyze, one to stop the heart |
| 20:39.02 | hydride | that's all I get |
| 20:39.06 | hydride | sorry for the flood :( |
| 20:39.06 | Qwell[] | ~pb |
| 20:39.07 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 20:39.15 | Qwell[] | tzanger: it's a "cocktail" |
| 20:39.25 | tzanger | so basically you asphyxiate for a while and then your heart stops |
| 20:39.37 | tzanger | that is a fairly cruel way to go I have to say |
| 20:40.18 | tzanger | not being able to breathe is no fun |
| 20:40.40 | file | tzanger: one knocks you unconscious, one to paralyze, one to stop the heart... yeah |
| 20:40.51 | C4T3l | Hmhesays: try www.lemonparty.org !!!AT YOUR OWN RISK!!! |
| 20:40.56 | Qwell[] | it took a few tries to get the order right |
| 20:41.13 | tzanger | oh ok you're not awake when you die |
| 20:41.18 | tzanger | that's different then |
| 20:41.22 | bkw_ | nope you're not awake |
| 20:41.57 | file | it would be cruel if you were awake |
| 20:42.01 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 20:42.26 | markit | hi :) anyone using visdn? (http://www.visdn.org/) |
| 20:42.27 | Qwell[] | I say... |
| 20:42.37 | Qwell[] | If you commit murder...you should be executed in the same method |
| 20:42.50 | Qwell[] | if you commit multiple, you should be executed in the most cruel way of them |
| 20:42.51 | Dr-Linux | ooo |
| 20:43.07 | file | the drug they use to knock you out is actually used to induce comas... rather cool |
| 20:43.16 | file | I don't know why I know this, but meh |
| 20:43.28 | Qwell[] | file: lies, you know why |
| 20:43.30 | Assid | Qwell[]: can you help me with a wav file for polycom ? i wanna use it as a ringtone.. just cant manage |
| 20:43.37 | Qwell[] | nope |
| 20:45.12 | tzanger | Qwell[]: I don't think they're going to fly a plane into a building with him in it.. |
| 20:45.38 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 20:45.45 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 20:46.18 | vader-- | when using a pri card during boot when will the light come on in the back? |
| 20:46.29 | vader-- | when the driver loads the car or will it turn on when the computer powers up? |
| 20:46.47 | Qwell[] | tzanger: well...he didn't really "commit" murder, now did he? |
| 20:46.48 | vader-- | the pri card im using is a digium te110p |
| 20:47.24 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:47.42 | Dr-Linux | [TK]D-Fender: WB |
| 20:48.01 | tzanger | Qwell[]: true |
| 20:48.05 | tzanger | conspiracy though |
| 20:48.23 | stoffell_h | markit, no.. but if you do, use the daily snapshots :) |
| 20:49.29 | *** join/#asterisk Souvent22 (n=chatzill@151.200.137.138) |
| 20:49.33 | Souvent22 | hello. |
| 20:49.43 | *** join/#asterisk stoffell_h (n=PircBot@pot.catsanddogs.com) |
| 20:50.01 | Souvent22 | My company is evaluating astersik and i have a quesiton about the phone.... |
| 20:50.10 | Souvent22 | does it support 3rd party ISDN contrlol? |
| 20:50.23 | Strom_C | Souvent22: which phone are you talking about? |
| 20:50.38 | Souvent22 | example:..... |
| 20:50.43 | *** join/#asterisk marcoprechel (n=marcopre@fl-69-34-76-200.sta.sprint-hsd.net) |
| 20:50.48 | Souvent22 | i have my office phone forwarded to my cell phone...... |
| 20:50.59 | marcoprechel | hello |
| 20:51.13 | Souvent22 | so when a call comes in, can asterisk show the callerID of the incoming call on my cell, instead of it showing that the call is coming from my office? |
| 20:51.26 | marcoprechel | i have a question |
| 20:51.36 | Strom_C | marcoprechel: just ask |
| 20:51.52 | marcoprechel | new to asterisk |
| 20:52.00 | marcoprechel | please excuse my .. numbness |
| 20:52.03 | Strom_C | Souvent22: sure, if your trunking allows you to set caller ID number |
| 20:52.24 | marcoprechel | i'm loking into getting a small starter kit - the dev kit from digium looks good |
| 20:52.43 | *** join/#asterisk DeeJayTwo (n=deejay2@37-179.sh.cgocable.ca) |
| 20:52.43 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 20:52.46 | marcoprechel | the FXS & FXO modules |
| 20:52.56 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 20:52.58 | sevard | Strom_C: can you help me with supervised transfers |
| 20:53.02 | marcoprechel | FXS is for connecting regular POTS lines, correct? |
| 20:53.04 | Souvent22 | Strom_C: ah, so Asterisk supports it, it's just that my CO/Phone Company must support it also correct? |
| 20:53.09 | Strom_C | Souvent22: yes |
| 20:53.24 | Strom_C | marcoprechel: FXO is for phone lines, FXS is for telephone sets |
| 20:53.34 | Strom_C | sevard: I can try |
| 20:53.37 | twisted[asteria] | implosion != explosion |
| 20:53.43 | sevard | I just made the features.conf file and added include => featuremap in a relevent context except I'm unsure about where to put the 't' stuff |
| 20:53.45 | Qwell[] | twisted[asteria]: the bits have got to go somewhere |
| 20:53.53 | twisted[asteria] | teh black hole |
| 20:53.53 | Qwell[] | You didn't end up in a singularity |
| 20:53.56 | sevard | Strom_C: I'm assuming it would go in my stdexten macro |
| 20:53.56 | Qwell[] | :P |
| 20:54.06 | twisted[asteria] | sevard, my bits would not go in your stdexten macro |
| 20:54.23 | sevard | your bits |
| 20:54.25 | marcoprechel | phone sets such as ? - If i chose to go with a LAN/WAN VoIP/SIP solution i would not need the FXO modules then, correct? |
| 20:54.39 | [TK]D-Fender | <twisted[asteria]> implosion != explosion <-- that depends if there is a soft bouncy core :) |
| 20:54.41 | Strom_C | marcoprechel: telephone set == boring old analog desk telephone |
| 20:54.45 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-90-106.telkomadsl.co.za) |
| 20:54.45 | twisted[asteria] | [TK]D-Fender, lol |
| 20:54.49 | sevard | twisted[asteria]: wtf? |
| 20:54.52 | twisted[asteria] | sevard, n/m |
| 20:54.59 | sevard | crazy bitch. |
| 20:55.06 | unmanaged | I have a Manager question.... After a " Action: Originate" I need to see if the call fails, if so do something else, I am doing this in PHP for the scripts.... Does a "Action: Originate" have a 'failed' exten or some kind of callstatus or am I going at this the wrong way? :) |
| 20:55.25 | Qwell[] | twisted[asteria]: SO...I'm told you're paying to fly me out to Digiumville in a few months? :p |
| 20:55.32 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 20:55.47 | sevard | Strom_C: I'm assuming stdexten macro but I really don't know where * wants to see the transfer stuff |
| 20:56.02 | twisted[asteria] | Qwell, i don't work for digium |
| 20:56.03 | twisted[asteria] | :P |
| 20:56.08 | Qwell[] | yeah, so? :P |
| 20:56.37 | twisted[asteria] | lol |
| 20:56.43 | file | ooh |
| 20:56.54 | Qwell[] | or, are you trying to say that file will pay? |
| 20:56.56 | file | Qwell[]: are you trying to take over HSV? |
| 20:57.00 | Qwell[] | file: By storm |
| 20:57.02 | file | I could pay... |
| 20:57.08 | file | and then invoice Digium for it ^_^ |
| 20:57.10 | Qwell[] | s/could/won't/ :D |
| 20:57.11 | Qwell[] | haha |
| 20:57.12 | twisted[asteria] | lol |
| 20:57.27 | Strom_C | sevard: the only transfers I do involve the "transfer" button on my SIP phone |
| 20:57.43 | Qwell[] | is there a good hotel guests usually stay in? |
| 20:58.01 | unmanaged | econ'o'hoe |
| 20:58.04 | sevard | Strom_C: I have 'transfer' and confrence buttons on my aastra 480i ct, which is awesome.. but for my atas they need the pbx to do the work |
| 20:58.05 | unmanaged | :P |
| 20:58.09 | file | I stayed at the Chateau Spencer |
| 20:58.13 | Qwell[] | file: heh |
| 20:58.14 | file | and Place de Fleming |
| 20:58.23 | Strom_C | sevard: no you dont |
| 20:58.23 | Qwell[] | tsk, tsk |
| 20:58.25 | twisted[asteria] | Qwell, uhm... there's always la punta |
| 20:58.28 | twisted[asteria] | or whatever it's called |
| 20:58.32 | twisted[asteria] | la quinta |
| 20:58.36 | Strom_C | sevard: flash, get second dial tone, dial, wait for call to answer, hang up. |
| 20:58.42 | Qwell[] | I don't stay in la quinta's |
| 20:58.50 | file | you could sleep under kp's desk |
| 20:58.51 | twisted[asteria] | why, the high punta rate? |
| 20:58.58 | Qwell[] | twisted[asteria]: because I lived in one for a while :P |
| 20:59.04 | twisted[asteria] | but |
| 20:59.06 | [TK]D-Fender | sevard : What model of ATA? |
| 20:59.10 | twisted[asteria] | this one is across the street from teh bar |
| 20:59.12 | sevard | sip 2002 |
| 20:59.16 | Qwell[] | touche |
| 20:59.16 | twisted[asteria] | and teh waffle hose |
| 20:59.21 | twisted[asteria] | house. |
| 20:59.25 | jsharp | awful house |
| 20:59.26 | file | haha |
| 20:59.35 | unmanaged | casa de waffle |
| 20:59.36 | Qwell[] | I'm not allowed into any place that serves waffles anymore |
| 20:59.39 | *** part/#asterisk phonic (i=phonic@antisocial.nu) |
| 20:59.46 | file | or alcohol |
| 20:59.48 | twisted[asteria] | Qwell, HSV != LAX |
| 20:59.50 | sevard | Strom_C: neat |
| 20:59.52 | Qwell[] | file: mostly waffles |
| 20:59.54 | twisted[asteria] | they don't know you here |
| 21:00.03 | unmanaged | go in there and ask for pancakes |
| 21:00.03 | Qwell[] | twisted[asteria]: if bkw_ was right...they do |
| 21:00.06 | stoffell | Qwell, choices are getting limited? ;) |
| 21:00.18 | unmanaged | and see what they do heheh *evil grin* |
| 21:00.40 | twisted[asteria] | i got a great idea then |
| 21:00.41 | twisted[asteria] | rent a car |
| 21:00.43 | twisted[asteria] | no |
| 21:00.43 | twisted[asteria] | a van |
| 21:00.47 | twisted[asteria] | and live in that |
| 21:00.51 | [TK]D-Fender | sevard : you can do transfers and conferencing direct in the ATA w/o DTMFing them |
| 21:00.51 | Qwell[] | Do you have a river? |
| 21:00.52 | Qwell[] | nearby |
| 21:00.55 | twisted[asteria] | <PROTECTED> |
| 21:00.59 | twisted[asteria] | yes |
| 21:01.02 | Qwell[] | excellent |
| 21:01.16 | file | perhaps... |
| 21:01.18 | Qwell[] | file: "vacation" :p |
| 21:01.23 | file | ohhhhhhh |
| 21:01.24 | Qwell[] | filno |
| 21:01.26 | Qwell[] | wtf |
| 21:01.35 | twisted[asteria] | roflmao |
| 21:01.47 | bkw_ | Qwell what? |
| 21:01.49 | bkw_ | right about what? |
| 21:02.12 | twisted[asteria] | uh oh |
| 21:02.20 | twisted[asteria] | laggered |
| 21:02.34 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:02.37 | Qwell[] | bkw_: waffle houses :p |
| 21:02.40 | bkw_ | oh yes |
| 21:02.42 | bkw_ | and over passes |
| 21:02.45 | Qwell[] | and me probably not being allowed into any |
| 21:02.49 | Dr-Linux | wow twisted[asteria] is woke up after 1000 of years :) |
| 21:03.01 | unmanaged | 79 dollars a nite at the hilton |
| 21:04.37 | twisted[asteria] | WHOA |
| 21:05.07 | twisted[asteria] | bkw_, what about wahos and overpasses? |
| 21:05.21 | Qwell[] | twisted[asteria]: you don't want to know |
| 21:05.29 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 21:05.34 | twisted[asteria] | why do you say that? |
| 21:06.01 | unmanaged | yes RoyK but don't tell anyone |
| 21:06.02 | Qwell[] | twisted[asteria]: You'll just have to trust me on this one :p |
| 21:06.08 | twisted[asteria] | no way |
| 21:06.13 | file | Qwell is the life of the party... |
| 21:06.27 | Qwell[] | I almost got us all kicked out of the waffle place, I guess |
| 21:06.36 | Qwell[] | and some unlucky car on the freeway...well... |
| 21:06.37 | tzanger | yeah you asked for a crepe |
| 21:06.44 | twisted[asteria] | ROFL |
| 21:06.46 | stack_ | I am trying to receive a fax via my PRI, and on the console I see "Redirecting Zap/5-1 to fax extension" but that is it. My fax extension is just a Goto to another context... any ideas? |
| 21:07.50 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 21:08.03 | tzanger | unmanaged: when I first got our PRI I called my ex-wife with a callerid of "HINDI HOTTIES" and a Toronto # |
| 21:08.24 | twisted[asteria] | tzanger, and this is why she now calls me. |
| 21:08.32 | twisted[asteria] | ;) |
| 21:09.05 | twisted[asteria] | (waits for it) |
| 21:09.21 | unmanaged | Qwelll go book ya a room ... http://hiltongardeninn.hilton.com |
| 21:10.09 | Qwell[] | I totally just stayed at a hilton garden inn |
| 21:10.24 | hydride | is the pcm module required for sip? |
| 21:10.45 | unmanaged | that is prob the best 'chain' hoe'tell |
| 21:11.27 | vader-- | in my zaptel.conf how would i configure a TE110P and a TDM2400P card to work together? |
| 21:11.34 | vader-- | i seem to be having an issue with that |
| 21:12.03 | Strom_C | vader--: did you contact digium tech support like I told you to? |
| 21:12.12 | vader-- | i got the card to work now |
| 21:12.16 | vader-- | i was using the wrong driver now |
| 21:12.18 | vader-- | name |
| 21:12.22 | vader-- | now = name |
| 21:12.59 | twisted[asteria] | tzanger, i was joking, just so you know... don't go yelling at the wifey :P |
| 21:14.26 | blitzrage | unmanaged: Extended Stay |
| 21:15.04 | vader-- | strom i had fxs instead of fxo |
| 21:15.06 | blitzrage | unmanaged: http://www.extendedstayhotels.com/ |
| 21:15.12 | vader-- | that was sending the wrong signal |
| 21:15.17 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 21:15.38 | twisted[asteria] | don't listen to blitzrage... he's canadian |
| 21:19.11 | vader-- | da da da |
| 21:19.22 | vader-- | now to try and figure out how to get a dial tone on these analog channels |
| 21:19.22 | vader-- | :) |
| 21:19.31 | Zodiacal | anyone know if i can use a .call file to playback a sound file to the asterisk sound card's paging function? i.e. console/dsp would i just have the .call file: Dial(console/dsp,,A(soundfile)) or would that try to connect to a line and keep it open, is there a way to just playback the file insted of dialing? |
| 21:19.43 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:20.01 | Souvent22 | Is it possible to network asterisk boxes togather? e.g. you have 5 asterisk boxes and different locations, and you want to 'intercom' to one of the other boxes. |
| 21:20.04 | *** join/#asterisk tomcontr3 (n=gcontrer@200.28.21.98) |
| 21:20.07 | sevard | Strom_C: alright, flash to transfer, but what about blind transfers or even confrences? |
| 21:20.09 | [hC] | Is it possible to lower the volume for music on hold when using the native mp3 stuff? |
| 21:20.10 | Souvent22 | i'm thinking i'll have to setup a VPN b/t the boxes. |
| 21:20.11 | *** join/#asterisk P4C0 (n=ash@200.124.22.34) |
| 21:20.38 | P4C0 | hello guys, is there a way to set a code like if one extension is ringing and there's noone there, other extension can pick up that call? |
| 21:20.47 | stoffell | Souvent22, at least you'll need a vpn yes. openpvn might be a good start |
| 21:20.59 | sevard | P4C0: what you're looking for is a hunt group |
| 21:21.11 | P4C0 | sevard, hunt group? |
| 21:21.23 | Strom_C | sevard: no no |
| 21:21.30 | Strom_C | sevard: he's talking about centrex-like features |
| 21:21.35 | [hC] | Any of you guys used sangoma a200's yet? |
| 21:21.43 | [TK]D-Fender | sevard : Yes you can do both blind & consultative transfers, and 3-way conferencing directly on the ATA. |
| 21:21.52 | sevard | P4C0: or forward on no answer |
| 21:21.54 | Strom_C | P4C0: you want to, for example, take extension 200's ringing call on extension 400? |
| 21:21.57 | [TK]D-Fender | [hC] : I've administered systems with them. |
| 21:22.03 | Souvent22 | stoffell: cool. just checking. we use SonicWalls, so I was thinking I could just work with our networking guy and get that setup. |
| 21:22.05 | timscott | call groups? |
| 21:22.09 | P4C0 | Strom_C, yes :D |
| 21:22.15 | [hC] | [TK]D-Fender: with or without hardware echo can? |
| 21:22.17 | timscott | ring groups |
| 21:22.24 | Strom_C | pickup groups :) |
| 21:22.31 | timscott | that's what I was looking for >_< |
| 21:22.32 | sevard | ring groups ring a bunch of extensions at once and the first to the call wins |
| 21:22.41 | P4C0 | sevard, that may do the trick as well :D |
| 21:23.04 | *** join/#asterisk froguz (n=alvaro@pc-95-155-104-200.cm.vtr.net) |
| 21:23.19 | BadPacket | damn you NUFONE! |
| 21:23.22 | *** part/#asterisk BadPacket (n=root@unaffiliated/badpacket) |
| 21:23.23 | P4C0 | any example about how to do that? ring groups? |
| 21:23.25 | sevard | P4C0: a ring group is the only thing I know how to do, in this example you dial 3001 and it dials 2001, 2002, 2003, and 2004 ;exten = 3001,2,Dial(SIP/2001&SIP/2002&SIP/2003&SIP/2004,20,tr) |
| 21:23.49 | twisted[asteria] | oh |
| 21:23.53 | twisted[asteria] | hunt groups are easy |
| 21:23.56 | P4C0 | sevard, thanks :D |
| 21:23.58 | twisted[asteria] | exten => s,1,Dial(SIP/1) |
| 21:24.02 | twisted[asteria] | exten => s,2,Dial(SIP/2) |
| 21:24.04 | Strom_C | P4C0: hey, look what I found |
| 21:24.06 | Strom_C | http://www.voip-info.org/wiki-Asterisk+callgroups+and+pickupgroups |
| 21:24.07 | twisted[asteria] | exten => s,3,Dial(SIP/3 |
| 21:24.09 | twisted[asteria] | and so on |
| 21:24.35 | Strom_C | google is a marvelous thing |
| 21:24.35 | sevard | Strom_C: so how the heck would you blindtransfer / confrence on an analog phone? |
| 21:24.35 | file | twisted[asteria]: QT4 DANCE! |
| 21:24.39 | P4C0 | thanks everyone |
| 21:24.43 | sevard | P4C0: from now on google site:voip-info.org <topic> |
| 21:24.54 | twisted[asteria] | file, no! |
| 21:24.56 | Strom_C | sevard: blind transfer I don't know about...what do you nean "conference"? |
| 21:25.15 | sevard | Strom_C: get a call, put him on hold, open another call, bridge these calls |
| 21:25.27 | Strom_C | sevard: the same way you do on analog phones |
| 21:25.36 | sevard | Strom_C: I have no experience with that :| |
| 21:25.38 | Strom_C | flash, dial at second dial tone, flash again once the call supervises |
| 21:26.03 | Strom_C | three-way calling has only been done like that since 1964 |
| 21:26.17 | Strom_C | so I know you might be kind of unfamiliar with it |
| 21:26.23 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:26.24 | Strom_C | :) |
| 21:26.34 | sevard | so confrence is get call, flash, dial second number, flash -- and supervised tranfer is get call, flash, dial second number, hang up |
| 21:26.49 | sevard | Strom_C: :P poor people don't have awesome pstn features |
| 21:26.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:27.01 | Strom_C | sevard: yes, but wait until the second call starts ringing |
| 21:27.28 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 21:27.35 | sevard | so much easier on an aastra, you hit confrence, dial the number, hit confrence |
| 21:27.46 | sevard | or transfer, dial the number, transfer. |
| 21:29.27 | [TK]D-Fender | [hC] : Both |
| 21:29.37 | vader-- | can anyone look at this and make sure it's correct |
| 21:29.37 | vader-- | http://pastebin.ca/53326 |
| 21:29.55 | vader-- | it's my zaptel.conf for 1 TE110P and 1 TDM2400P with 24 FXS channel |
| 21:29.57 | vader-- | s |
| 21:31.02 | generalhan | vader--: looks good |
| 21:31.02 | [TK]D-Fender | sevard : Conference is [flash], place 2nd call, [FLASH] |
| 21:31.02 | sevard | got it down |
| 21:31.02 | vader-- | generalhan is there anything i need to do to the zapata.conf now? |
| 21:31.02 | generalhan | you need to define those cahnnels in there too |
| 21:31.09 | [TK]D-Fender | sevard : Blind transfer is [flash] + some star code I don't recall (you'll see it in the web-admin), get different dialtone, dial 2nd number, done |
| 21:31.10 | [hC] | [TK]D-Fender: did it work decently well without echo can? I just have deployed a lot of tdm400p and they're absolutely horrible for echo |
| 21:31.21 | vader-- | i can't seem to find a good sample of a zapata.conf that would be similar to my setup |
| 21:31.23 | vader-- | any suggestions? |
| 21:31.26 | [TK]D-Fender | [hC] : So far pretty good with straight Zaptel |
| 21:31.32 | unmanaged | hey............ |
| 21:31.37 | generalhan | vader--: show me what you have so far |
| 21:31.39 | [TK]D-Fender | [hC] : And you can always add it later |
| 21:31.47 | sevard | [TK]D-Fender: I just tested it, strom is right, blind transfer is get call {flash} dial, wait for ring, hang up |
| 21:31.56 | vader-- | generalhan pretty much the default config file |
| 21:31.59 | vader-- | i haven't modified it yet |
| 21:32.02 | Zodiacal | anyone know if i can use a .call file with having to bridge it to a ext? i just wanta play a sound file to my sound card. |
| 21:32.28 | Strom_C | sevard: doubt not my phone-fu |
| 21:32.33 | sevard | haha |
| 21:32.37 | sevard | Zodiacal: mplayer <file> :) |
| 21:32.42 | stack_ | I am trying to receive a fax via my PRI, and on the console I see "Redirecting Zap/5-1 to fax extension" but that is it. My fax extension is just a Goto to another context... any ideas? |
| 21:32.54 | Strom_C | stack_: are you using AMP/FreePBX? |
| 21:33.00 | [TK]D-Fender | sevard : that might work too, I've jsut the *code method personally following their guide. Actually doesn't make sense for single flash + nagup = blind..... |
| 21:33.03 | Zodiacal | sevard omg, i had asterisk in my head too long.. direct like that would be better, thanks! :P |
| 21:33.05 | [hC] | [TK]D-Fender: do you have experience to compare it to the reliability of the tdm400p |
| 21:33.09 | stack_ | Storm_C: nope |
| 21:33.31 | [TK]D-Fender | [hC] : TDM400.... reliable.... ? ;) |
| 21:33.36 | Strom_C | stack_: pastebin your extensions.conf |
| 21:33.38 | [hC] | exactly |
| 21:33.39 | Strom_C | ~pb |
| 21:33.41 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:33.41 | generalhan | vader--: here is mine but i have a TDM40B so only 4 channels instead of your 24 ... http://generalhan.pastebin.ca/53327 |
| 21:33.44 | sevard | Zodiacal: wow :P |
| 21:33.51 | [TK]D-Fender | [hC] : Everything they make seems to be a ROCK. |
| 21:34.12 | [hC] | [TK]D-Fender: sangoma? |
| 21:34.16 | [TK]D-Fender | [hC] : I run an A104d at work, and consulted 2 people with A200's |
| 21:34.18 | unmanaged | Can someone give me an example of the correct way to bridge someone on hold? I think you just gave me a fix for a problem that I have been trying to hack at... |
| 21:34.20 | [TK]D-Fender | [hC] : yup |
| 21:34.30 | [TK]D-Fender | [hC] : Wildly positive |
| 21:34.43 | [hC] | [TK]D-Fender: yeah i use a102u's and i love them so far. |
| 21:34.45 | unmanaged | or point me to the info ... :) |
| 21:34.57 | *** join/#asterisk ckwall (n=ckwall@65.218.229.224) |
| 21:35.00 | vader-- | so generalhan i don't need all that other stuff in the zapata.conf? |
| 21:35.09 | generalhan | thats all i have |
| 21:35.14 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:35.29 | ckwall | ok, so I have confused myself trying to learn how to set up the sip.conf and the extensions.conf |
| 21:35.29 | [hC] | [TK]D-Fender: id obviously prefer the hardware echo can, but wondering if the $350 is ultimately necessary, as ive basically decided that tdm400p with no echo can is almost unusable. |
| 21:35.32 | vader-- | man they throw all that shit in there and it's kinda confusing and organized |
| 21:35.42 | [hC] | [TK]D-Fender: even after hours of hair pulling with ztmonitor, its still not what i would call 'good' |
| 21:35.57 | generalhan | vader--: i dont do anything too impressive with these channels so i dont need all that other stuff |
| 21:35.59 | ckwall | I tried setting things up with a polycom soundpoint 301, and then the xten xlite softphone. |
| 21:36.02 | [TK]D-Fender | [hC] : Basically all experience with it says take it "raw" and you can add the EC on after. |
| 21:36.05 | ckwall | I cannot get them to work. |
| 21:36.11 | [hC] | [TK]D-Fender: 10-4. |
| 21:36.18 | stack_ | Storm_C: http://pastebin.com/697014 those are the relevant parts |
| 21:36.21 | ckwall | can anyone coach me through this? |
| 21:36.32 | Strom_C | stack_: my name is Strom, not Storm |
| 21:36.36 | Strom_C | please get it right |
| 21:36.38 | [TK]D-Fender | ckwall : Pastebin what you've done so far. |
| 21:36.41 | [TK]D-Fender | ~pb |
| 21:36.43 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:36.53 | ckwall | cool |
| 21:36.56 | ckwall | hang tight. |
| 21:37.05 | macTijn | Strom_C: don't be picky ;) |
| 21:37.10 | sevard | hahahahahaa |
| 21:37.11 | stack_ | Strom_C: my bad... you should see my try to type "destroy", it comes out destory every time |
| 21:37.15 | sevard | stormy storm |
| 21:37.28 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:37.37 | macTijn | sevard: that won't be out for a couple of years ;) |
| 21:38.29 | Strom_C | stack_: well first off, I assume that you've already answered the call when it goes to fax, so you dont need to have a second answer statement in there |
| 21:38.39 | vader-- | general i pretty much copied your conf file and added a few channels to the fxs |
| 21:38.53 | stack_ | Strom_C, yeah I was just throwing them around left and right to try it out... |
| 21:39.00 | vader-- | now when i reload asterisk i should get a dial tone on my analog channels right? |
| 21:39.16 | sevard | macTijn: say what |
| 21:39.21 | *** part/#asterisk P4C0 (n=ash@200.124.22.34) |
| 21:39.31 | Strom_C | stack_: I have little experience with iaxmodem; what happens if you forego the menu and just send ALL calls to the iaxmodem? |
| 21:39.36 | macTijn | sevard: you made that sound like an ubuntu release name |
| 21:39.49 | generalhan | vader--: once you start messing with zaptel and zapata you should rmmod all the modules and then modprobe them again and then run ztcfg -vvvvvvvvvvvvv |
| 21:39.49 | stack_ | Strom_C: works fine |
| 21:39.51 | sevard | macTijn: ! sorry i don't know ubuntu at all, slax for lief brotha |
| 21:39.54 | macTijn | sevard: "warty warthog", "breezy badger" :) |
| 21:39.56 | generalhan | vader--: THEN start * again |
| 21:39.58 | macTijn | hahaha |
| 21:40.04 | macTijn | slack is the past for me :) |
| 21:40.08 | Strom_C | stack_: show me the config that works fine |
| 21:40.11 | macTijn | debian for teh win |
| 21:40.14 | C4T3l | whats your fav monitoring tool for asterisk? |
| 21:40.14 | macTijn | eh no |
| 21:40.16 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 21:40.18 | ckwall | ok, i posted the sip.conf |
| 21:40.22 | macTijn | what am I saying !? |
| 21:40.30 | sevard | C4T3l: CLI + sip debug |
| 21:40.31 | macTijn | Ubuntu for teh win ofcourse :) |
| 21:40.52 | C4T3l | sevard: no nagios or anything? |
| 21:41.23 | sevard | say what brotha |
| 21:41.33 | stack_ | Strom_C: http://pastebin.com/697024 same site with a "works_fine" section |
| 21:41.41 | C4T3l | nagios server monitor |
| 21:42.03 | sevard | C4T3l: i'm really just fskering, i don't know any of that |
| 21:42.11 | vader-- | general i keep getting this error in * now |
| 21:42.11 | vader-- | May 3 11:41:39 WARNING[3171]: chan_zap.c:8970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. |
| 21:42.36 | stack_ | Strom_C: if I let the timeout go through on the background, it just fails, so it says it is redirecting to the fax extension, but then it never does |
| 21:43.07 | Strom_C | stack_: you are coming on over a ZAP channel, right? |
| 21:43.28 | stack_ | Strom, yep... TE110p |
| 21:44.23 | *** join/#asterisk jffmriii (n=findme@66.244.161.19) |
| 21:44.36 | jffmriii | looking to employ some developers |
| 21:44.42 | jffmriii | anyone interested? |
| 21:44.44 | *** join/#asterisk xlyz (n=xl@213-140-17-96.ip.fastwebnet.it) |
| 21:44.49 | *** part/#asterisk xlyz (n=xl@213-140-17-96.ip.fastwebnet.it) |
| 21:44.57 | Strom_C | jffmriii: look in #asterisk-dev or post on voip-info.org |
| 21:45.09 | [TK]D-Fender | jffmriii : In a non-descript job? Where do I sign up!? ;) |
| 21:45.11 | ckwall | my sip.conf is http://pastebin.com/pastebin.php?dl=697022 |
| 21:45.26 | *** join/#asterisk stkn_ (n=foobar@gentoo/developer/pdpc.active.stkn) |
| 21:45.43 | froguz | jffmriii, are you looking for ppl aoutside usa? |
| 21:46.06 | Strom_C | stack_: I wish I could help you further, but I have no experience with asterisk and faxes |
| 21:46.20 | stack_ | Strom_C: ok thanks, I'll play around some more |
| 21:46.30 | [TK]D-Fender | ckwall : X-Lite is the only one you don't have commented out... |
| 21:47.03 | ckwall | right, I thought that is what I wanted if that is the applicaiton I was using to connect with |
| 21:47.03 | [TK]D-Fender | ckwall : and for that one ditch the username, you should be authenticating with "xlite1" as the username in your client |
| 21:47.05 | froguz | stack_, linksys PAP2 works good with asterisk |
| 21:47.19 | ckwall | ok. will try, hang on |
| 21:47.29 | [TK]D-Fender | ckwall : You should have a section for EACH phone in your PBX.... |
| 21:47.42 | [TK]D-Fender | ckwall : Explains why your Polycom isn't getting very far... |
| 21:47.56 | Druken | pap2's work awesome with asterisk |
| 21:48.04 | [TK]D-Fender | ckwall : Tell you what, scrap EVERYTHING commented out.... |
| 21:48.08 | *** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 21:48.15 | ckwall | well, i tried with the polycom profile first, and commented it all back out |
| 21:48.21 | ckwall | ok... scrapping. |
| 21:48.26 | CoffeeIV_ | I have 2 asterisk servers with Digium T1 cards in them, I can connect a T1 cable between them and pass channels from one to the other, right ? I have only connected thos T1 cards to ADIT 600s or other equipment before now. Does it have to be some kind of "crossover" T1 cable ? |
| 21:48.45 | [TK]D-Fender | ckwall : and give me the BASE pastebin link when you're done with that. |
| 21:48.46 | Iam8up|lpy | i've got several IP phones on this network, and the outbound callerid appears either as what we set it to, unknown, or our carrier's did |
| 21:49.04 | Druken | CoffeeIV_: why waste the cards? hand the calls over ip |
| 21:49.12 | ckwall | ok |
| 21:49.14 | ckwall | will do. |
| 21:49.59 | CoffeeIV_ | Druken: I don't know, they asked me to -- probably they asked about it because I've been taking a really long time to get a simple IAX2 connection between them going |
| 21:50.02 | vader-- | do you guys know if having a loopback plug in a PRI card will cause these errors |
| 21:50.03 | vader-- | May 3 11:41:39 WARNING[3171]: chan_zap.c:8970 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. |
| 21:50.06 | Dr-Linux | CoffeeIV_: put both servers on same local network and enjoy |
| 21:50.29 | Druken | CoffeeIV_: why? iax2 is easy :) |
| 21:50.50 | justinu|laptop | vader--: yeah, it would cause that. |
| 21:51.31 | [TK]D-Fender | CoffeeIV_ : IAX2 = your friend |
| 21:52.01 | CoffeeIV_ | I've been trying to follow the IAX2 example from voip-info and basing it off the one in the sample files that calls digium, but I haven't go it working -- that's a separate question |
| 21:52.13 | Dr-Linux | i heard IAX2 trunk is good, but SIP works fine for me |
| 21:52.46 | Dr-Linux | CoffeeIV_: 1234567890,1,Dial(IAX2/user:pass@123.123.123.123/1234,120) |
| 21:52.47 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 21:52.48 | Druken | Dr-Linux: you use sip between your server? icky.... |
| 21:52.55 | [TK]D-Fender | Dr-Linux : IAX does let you do a little more on the interesting side with context control, but for just passing calls on a basic leve SIP is just fine |
| 21:53.03 | justinu|laptop | bah, sip is fine |
| 21:53.08 | Dr-Linux | Dr-Linux: yes, icky..? |
| 21:53.20 | Druken | Dr-Linux: my personal opinion |
| 21:53.42 | Dr-Linux | icky... mean bad? |
| 21:53.53 | [TK]D-Fender | Dr-Linux : Correct |
| 21:53.54 | Druken | icky == don't like it |
| 21:54.23 | Dr-Linux | ~dict icky |
| 21:54.49 | Druken | it's like sex with an 80 year old woman, i know it can be done, but it'd be one cold day in hell before i did it.... |
| 21:54.51 | Dr-Linux | Druken: why bad ? |
| 21:55.01 | timscott | IAX > * |
| 21:55.13 | Iam8up|lpy | sip > iax =P |
| 21:55.21 | Iam8up|lpy | just kidding! |
| 21:55.23 | Dr-Linux | Druken: but i'm using it, like i'm doing sex with 20 years virgin girl :S |
| 21:55.32 | brodiem | Is there a way of executing an automated call (i.e. with .call files) but to NOT ring an actual extension in order to connect the call? It's for a DND on/off function, where the dial plan does everything it needs based on the caller ID number. |
| 21:56.24 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
| 21:56.24 | Dr-Linux | heh, friends honestly i didn't know how to make trunk with IAX2, that's why i done that using SIP |
| 21:56.26 | [TK]D-Fender | brodiem : set the channel as "Local" |
| 21:56.32 | [TK]D-Fender | brodiem : And script away |
| 21:56.41 | justinu|laptop | Dr-Linux: sip is fine |
| 21:56.51 | [TK]D-Fender | brodiem : or an ugly way and have you * register with itself and get circular :D |
| 21:56.56 | brodiem | [TK]D-Fender, that's what I just tried, using Local/422 and it rang ext 422 |
| 21:57.12 | brodiem | orjust "Local" w/o anything appended |
| 21:57.12 | [TK]D-Fender | brodiem : Don't forget the CONTEXT |
| 21:57.13 | justinu|laptop | make it call a different extension, that doesn't ring the phone |
| 21:57.42 | brodiem | [TK]D-Fender yeah I didn't forget the context. Everything works fine except nothing initiates until I pick up the ext |
| 21:57.45 | [TK]D-Fender | brodiem : Loca/422@notmyphonextensioncontext |
| 21:57.54 | Dr-Linux | justinu|laptop: i had only problem that caller was getting low voice. so i just increase rxgain to 2.0 it works fine now so far |
| 21:57.58 | [TK]D-Fender | brodiem : Put an "Answer in there" |
| 21:58.12 | Druken | CoffeeIV_: you want some help with the iax2? |
| 21:58.18 | brodiem | yup it does |
| 21:58.19 | justinu|laptop | Dr-Linux: that has nothing to do with sip |
| 21:58.22 | brodiem | trying with @somefakeextension now |
| 21:58.33 | froguz | can i have sip peers and users in Real Time AND use sip.conf? do i need to do something special to get it work? |
| 21:58.42 | vader-- | when using zap channels where does asterisk get the caller ID infomration from? from zapata.conf? |
| 21:59.02 | Druken | vader--: meaning? |
| 21:59.27 | brodiem | [TK]D-Fender, nada, chan_local.c: No such extension/context 422@fakeext |
| 21:59.41 | froguz | i think i'm doing somethig wrong... mi realtime sip friends can't loggin |
| 21:59.44 | Dr-Linux | justinu|laptop: okey let me know what you think, what's difference between IAX2 trunk and SIP trunk? |
| 21:59.44 | [TK]D-Fender | Doe it exast as named in extensions.conf? |
| 21:59.54 | vader-- | like say i have an analog extension plugged into a tdm2400p and i call a sip phone |
| 22:00.09 | vader-- | it shows up on the phone as Asterisk |
| 22:00.12 | froguz | Dr-Linux, IAX2 is NAT transversal |
| 22:00.14 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 22:00.14 | vader-- | in the caller id info |
| 22:00.17 | CoffeeIV_ | Druken: I'm going to paste bin what I tried with IAX2 and see if anyone has any suggestions |
| 22:00.17 | [TK]D-Fender | vader-- : Yes, you set up the caller ID in zapata.conf |
| 22:00.23 | Druken | in that case, yes... |
| 22:00.37 | brodiem | [TK]D-Fender Doe it exast? what?? lol |
| 22:00.41 | vader-- | so i have to setup each channel in zapata.conf |
| 22:00.42 | Druken | CoffeeIV_: aight |
| 22:00.58 | Druken | vader--: if you want seperate caller id, yep |
| 22:00.58 | [TK]D-Fender | brodiem : Does that exten exist in the context you named? |
| 22:01.07 | [TK]D-Fender | brodiem : Don't get picky with my typing! |
| 22:01.10 | Dr-Linux | froguz: your statement has gone over my head :S |
| 22:01.42 | justinu|laptop | iax2 is a binary protocol |
| 22:01.46 | justinu|laptop | like q931 or something |
| 22:01.50 | justinu|laptop | sip is more like HTTP |
| 22:01.52 | brodiem | [TK]D-Fender yeah, like I said if I specify a valid channel to use to connect the call, it works fine except the channel in question rings first and I must answer it to initiate |
| 22:01.52 | *** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz) |
| 22:02.21 | [TK]D-Fender | brodiem : who said that exten had to actually CONTAIN a Dial at all, hmm? |
| 22:02.37 | ghost99 | Morning Tk-defender :) we are awake now ! |
| 22:02.40 | [TK]D-Fender | brodiem : Think outside the box a little... |
| 22:03.02 | [TK]D-Fender | ghost99 : Good morning, SpanDSP (Faxing) left to add, and IVR to customize. |
| 22:03.04 | Druken | brodiem: why not use the CID from the .call file to let the extentsion know... so you can use exten=4321/1234 |
| 22:03.10 | [TK]D-Fender | ghost99 : Continue in PM |
| 22:03.36 | *** join/#asterisk Led_Zeppelin (n=dummy@cpe-24-31-182-121.columbus.res.rr.com) |
| 22:03.37 | justinu|laptop | "we are borg" |
| 22:03.40 | Dr-Linux | hhmm... |
| 22:04.08 | Led_Zeppelin | hello, I am new to vo/ip world. Can someone please recommend me a good service for it? I just need a service to make calls, the cheapest if possible. |
| 22:04.09 | ghost99 | [TK] d-defender: yeah .. I was playing last night and my headset is stuffed so will you be around for an hours or 2 while i get another headset and we finish off ? |
| 22:04.21 | Dr-Linux | justinu|laptop: i start making a website |
| 22:04.31 | justinu|laptop | about what? |
| 22:04.32 | [TK]D-Fender | ghost99 : Actually I'm out of here in about an hour :/ |
| 22:04.52 | Dr-Linux | i'll introduce asterisk in pakistan, it will be spread |
| 22:04.57 | justinu|laptop | nice |
| 22:05.14 | Dr-Linux | justinu|laptop: there is nothing i just start .. www.syednetworks.com |
| 22:05.16 | brodiem | [TK]D-Fender so you're saying as my channel I should just create a valid channel that does nothing but an Answer() basically |
| 22:06.18 | Druken | answer(),wait(5),hangup |
| 22:06.20 | Druken | hehe |
| 22:07.21 | *** part/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 22:08.38 | brodiem | haha that did it |
| 22:08.58 | brodiem | never think of the simple things sometimes =/ |
| 22:09.15 | brodiem | thanks for the advice tho guys |
| 22:09.38 | [TK]D-Fender | brodiem : No, not JUST an answer, but includes it and does whatever else you need to do your DND triggering |
| 22:09.52 | brodiem | i know... i got it |
| 22:11.05 | generalhan | [TK]D-Fender: is there a way that i can set a variable in a context in extensions.conf and then go to a queue, then come out of a queue with that variable still intact ? |
| 22:11.27 | brodiem | generalhan set a global var |
| 22:12.01 | brodiem | Set(var=value|g) |
| 22:12.04 | generalhan | i dont know if that will work in this situation ... |
| 22:12.17 | [TK]D-Fender | generalhan : It may survive "_" inheritance, if not change the callerID toa unque value and use it as a key. |
| 22:12.27 | generalhan | hmm |
| 22:13.15 | generalhan | [TK]D-Fender: see thats kinda the route i was thinking of but not quite sure how to do it. i just want to save the callerid(number) as a variable so that i can call on it after the queue. cause the callerid(number) changes to whatever local/ i call on |
| 22:13.30 | generalhan | from the wueue |
| 22:13.40 | generalhan | wueue = queue ..... i think ! lol |
| 22:14.34 | [TK]D-Fender | generalhan : Change the callerID to the current UNIQUEID *BEFORE* the calls hits the Queue, then push some values into ASTDB based on that, then enter the Queue |
| 22:14.47 | key2 | kwhen I use a SER, do I have to register asterisk on it if I wanna treat calls on my dialplan or it's not necessary ? |
| 22:15.01 | generalhan | [TK]D-Fender: so basically the only way for me to do it is to use a DB of sorts ? |
| 22:15.21 | justinu|laptop | asterisk doesn't need to register with SER |
| 22:15.22 | generalhan | put that number in there and then call on it from location later ? |
| 22:15.28 | justinu|laptop | unless the IP adress is dynamic |
| 22:15.34 | [TK]D-Fender | generalhan : Not the only way, but its a way I can tell you WILL work. Clumsy but effective. |
| 22:16.07 | generalhan | [TK]D-Fender: hmm cause im not using DB stuff at all right now |
| 22:16.25 | generalhan | i dont really want to learn it just for this ... if i had more use out of it it may be worth learning ... |
| 22:16.26 | [TK]D-Fender | generalhan : inside your Local channel created by an agent-callout, you can use that CID to lookup the STORED CID from the ASTDB so that the Agent can't tell the difference as well.... |
| 22:16.33 | Druken | oh god... i'd be screwed without my database calls |
| 22:16.43 | [TK]D-Fender | generalhan : ASTDB stuff is damn easy and built-in..... |
| 22:16.44 | generalhan | [TK]D-Fender: what youre saying is perfet |
| 22:16.50 | generalhan | [TK]D-Fender: LOL |
| 22:17.00 | generalhan | [TK]D-Fender: as long as its easy ! lol |
| 22:17.13 | [TK]D-Fender | generalhan : a handful of extra lines. |
| 22:17.15 | generalhan | do you have a newb's link to ASTDB ? |
| 22:17.15 | vader-- | thanks for your help guys |
| 22:17.25 | generalhan | vader--: its all working now ? |
| 22:17.30 | vader-- | general, d-fender, drunk ,strom, just |
| 22:17.34 | [TK]D-Fender | generalhan : lookup "asterisk functions" on the WIKI. |
| 22:17.39 | vader-- | ya i got the analog working |
| 22:17.40 | generalhan | k |
| 22:17.48 | vader-- | im not sure about the pri because i don't have it connected yet to test |
| 22:17.57 | vader-- | so i disabled the lines in the zapata.conf |
| 22:18.00 | generalhan | [TK]D-Fender: thanks ill go check it out now |
| 22:18.43 | vader-- | i made a successful zap to sip call |
| 22:18.57 | justinu|laptop | good deal |
| 22:18.59 | vader-- | i have to setup my extensions file to handle a sip to zap call |
| 22:20.28 | vader-- | hmmm no rining on my zap line |
| 22:20.36 | vader-- | i can pick it up and the call is there but the phone didn't ring |
| 22:20.39 | vader-- | weird |
| 22:20.48 | Druken | check the ringer on the phone? hehe |
| 22:20.50 | *** join/#asterisk gezick (n=gezick@c-68-50-25-85.hsd1.dc.comcast.net) |
| 22:21.11 | vader-- | hmm worked this time |
| 22:21.12 | vader-- | weird |
| 22:21.30 | Dr-Linux | vader--: is it registered with SIP? |
| 22:21.42 | vader-- | now say i do a sip to zap call and the sip phone hangs up the zap phone gets a busy signal blasted in their ear |
| 22:21.45 | vader-- | is that normal? |
| 22:21.50 | Dr-Linux | vader--: if yes, the put qualify=yes option in your sip.conf for this user |
| 22:22.08 | vader-- | ya the sip phone is registered with sip |
| 22:22.15 | Dr-Linux | ok then do that |
| 22:22.21 | vader-- | what does that do? |
| 22:22.22 | Dr-Linux | this problem will be fine |
| 22:22.23 | Druken | vader--: blasted? maybe check your tx ? |
| 22:22.34 | [TK]D-Fender | vader-- : Thats called a "reorder" tone |
| 22:22.53 | justinu|laptop | qualify makes asterisk "ping" the phone periodically to make sure it's alive and kicking |
| 22:23.04 | vader-- | is there anyway to make asterisk just send dead air? |
| 22:23.08 | vader-- | when the other end hangs up |
| 22:23.14 | vader-- | instead of the reorder tone |
| 22:23.22 | dlynes_ | vader--: change your reorder tone in indications.conf |
| 22:23.27 | Druken | why would you want that ? |
| 22:23.32 | dlynes_ | vader--: but that's not something to suggest |
| 22:23.56 | vader-- | when the other party hangs up it is putting this tone on their line |
| 22:24.00 | vader-- | i rather just have dead air |
| 22:24.06 | vader-- | like a real phone has |
| 22:24.14 | justinu|laptop | depends on where you are vader |
| 22:24.26 | Druken | even with a real phone, the silence only lasts a few seconds... |
| 22:24.28 | justinu|laptop | some PSTNs play disconnect supervisory tones, some dont |
| 22:24.44 | dlynes_ | vader--: on a "real" phone, it becomes dead air, and then shortly thereafter, you get a reorder tone |
| 22:25.32 | vader-- | ya |
| 22:25.35 | vader-- | thats kinda what i want |
| 22:26.00 | justinu|laptop | here in the US, the bitchy recording comes on |
| 22:26.08 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 22:26.08 | [TK]D-Fender | I'm feeling stupid today : can someone remind me where sendmails config file is by default? |
| 22:26.10 | justinu|laptop | "If you'd like to mall a call, please hang up and try your call again" |
| 22:26.14 | Druken | so use the g dial option... i think it's g... and have a 3 second wait |
| 22:26.21 | justinu|laptop | depends on your distro, fender |
| 22:26.22 | dlynes_ | Yeah, here in Canada, it depends on what province you're in, what happens |
| 22:26.22 | Qwell[] | mall a call? |
| 22:26.34 | justinu|laptop | <PROTECTED> |
| 22:26.37 | jffmriii | sorry about that I am back |
| 22:26.38 | CoffeeIV_ | Druken and anyone else who might know some IAX2 asterisk->asterisk stuff -- I pasted some configs that aren't working: http://pastebin.ca/53345 |
| 22:26.42 | [TK]D-Fender | justinu : Slackware |
| 22:26.42 | jffmriii | location is not an issue |
| 22:26.53 | justinu|laptop | it's called sendmail.cf if that helps |
| 22:27.03 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 22:27.04 | [TK]D-Fender | justinu : It does, thanks |
| 22:27.36 | Dr-Linux | jffmriii: i suggest deal with [TK]D-Fender and justinu|laptop |
| 22:27.39 | [TK]D-Fender | justinu : brute-force search :) |
| 22:27.51 | dlynes_ | [TK]D-Fender: /etc/mail/sendmail.cf on recent versions of slackware; /etc/sendmail.cf on older versions |
| 22:27.51 | justinu|laptop | find / -name sendmail.cf -print :) |
| 22:27.56 | jffmriii | employing asterisk programmers please offline me |
| 22:28.07 | Druken | CoffeeIV_: did you try taking out the password in the dial line? |
| 22:28.09 | Nugget | sounds kinky. |
| 22:28.13 | Druken | i don't see you needing it |
| 22:28.19 | CoffeeIV_ | Druken: trying that now |
| 22:28.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 22:29.29 | [TK]D-Fender | justinu : Just trying to get VM -> E-mail running on a server and it seemed pretty automatically functional on mine. |
| 22:29.43 | justinu|laptop | yeah, should "just work" |
| 22:29.45 | [TK]D-Fender | jffmriii : Again, some details may be nice... |
| 22:29.58 | dlynes_ | [TK]D-Fender: yeah, and on slackware, sendmail is relatively secure out of the box, too |
| 22:30.15 | froguz | [TK]D-Fender, i allways thought your nickname was related to fender guitars :$, now i see "de" fender |
| 22:30.23 | CoffeeIV_ | Druken: taking out the password didn't change anything |
| 22:31.13 | justinu|laptop | i think it is related to fender guitars |
| 22:31.36 | dlynes_ | justinu|laptop: or maybe he's just a tcl/TK fan :) |
| 22:32.02 | [TK]D-Fender | froguz : No, I play Dean personally. The nick goes back to my Tribes 1 CTF gaming..... |
| 22:32.10 | justinu|laptop | ah |
| 22:32.49 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 22:32.50 | [TK]D-Fender | And [TK] was my Action:Half-Life clan. I've always carried the primary nick however |
| 22:33.17 | generalhan | [TK]D-Fender: OMG i havent thought about Tribes in soooo long ~ |
| 22:33.24 | generalhan | god that was a GREAT game |
| 22:33.33 | froguz | hmmm... so, you're not david gilmour :( |
| 22:33.47 | [TK]D-Fender | Tribes rocked. Ski-mode FPS realy brought out the best FPS experience for me excetp for AHL. |
| 22:34.06 | [TK]D-Fender | froguz : No I'm more Eric Johnson / Neil Zaza :) |
| 22:34.14 | generalhan | HAHAHAHA |
| 22:34.44 | froguz | HAHAHAHA |
| 22:34.48 | froguz | m? |
| 22:34.51 | froguz | jeje |
| 22:35.12 | jffmriii | we are a macintosh computer consulting firm |
| 22:35.23 | jffmriii | that has asterisk inhouse |
| 22:35.38 | brookshire | on a mac? |
| 22:35.39 | [TK]D-Fender | jffmriii : Ok, and now for the USEFUL details on what you're looking to do.... |
| 22:35.39 | brookshire | :) |
| 22:36.06 | jffmriii | we are looking for someone to help us design an user friendly interface for the local server as well as user side |
| 22:36.29 | jffmriii | kind of what fonality.com is doing but we want to keep it open source not pripritory |
| 22:36.45 | brookshire | you mean like amp/freeswitch? |
| 22:36.55 | Qwell[] | brookshire: freepbx |
| 22:36.56 | jffmriii | we are also looking to integrate asterisk into our collaboration suite zimbra |
| 22:37.01 | brookshire | oh. .freepbx |
| 22:37.02 | brookshire | lol |
| 22:37.03 | [TK]D-Fender | brookshire : But our wheel will be BIGGER... and ROUNDER! |
| 22:37.12 | *** join/#asterisk ibob63 (n=hp@bb-87-82-7-89.ukonline.co.uk) |
| 22:37.16 | Qwell[] | [TK]D-Fender: wheels were made to be reinvented |
| 22:37.33 | [TK]D-Fender | Qwell : And to think all most people do with them here is SPIN ;) |
| 22:37.35 | jffmriii | amp and freebpx is not user friendly and is missing some feature we would like to implement |
| 22:37.42 | brookshire | jrr: there is already a plugin for zimbra somwhere |
| 22:37.49 | Qwell[] | jffmriii: like...not sucking? :) |
| 22:37.53 | jffmriii | lol yes |
| 22:38.13 | jffmriii | and we are willing to but the money into it as we already have customer waiting for the soluitons |
| 22:38.20 | dlynes_ | jffmriii: Zimbra's an Ajax alternative to Exchange or something, right? |
| 22:38.23 | jffmriii | we are zimbra developers |
| 22:38.26 | [TK]D-Fender | Take a look at ScopServ... best GUI for * in that all GUI = ass.... |
| 22:38.28 | jffmriii | yes |
| 22:38.37 | jffmriii | we are the developers for zimbra on mac |
| 22:38.45 | jffmriii | the exchange killer we like to call it |
| 22:38.59 | dlynes_ | jffmriii: Yeah...it's been written up in Linux Magazine |
| 22:39.00 | froguz | jffmriii, are you looking for a mac app developer, a web GUI developer? |
| 22:39.15 | jffmriii | both |
| 22:39.23 | jffmriii | we are looking for to run on bsd |
| 22:39.26 | jffmriii | or linux |
| 22:39.32 | jffmriii | but we are mac focused |
| 22:39.40 | jffmriii | we are based out of chicago |
| 22:39.43 | ibob63 | I can seem to find any documention on allowing people to transfer calls. Is there any way of doing this? |
| 22:39.57 | brookshire | i dunno, but please. .the next person who develops a gui, please use comments as markers so you can make changes to the .conf files |
| 22:40.01 | dlynes_ | ibob63: hit the transfer button on your sip phone |
| 22:40.03 | froguz | xNIX -> bsd darwin -> mac |
| 22:40.15 | [TK]D-Fender | ibob63 : Plent on that, also depends on what kind of interface and hardware yuo are talking about. |
| 22:40.20 | dlynes_ | ibob63: It's usually a phone feature...asterisk can do it, but it's easier on the phone |
| 22:40.24 | jffmriii | I would like to take this off line as I fell I am using or spaming the channel |
| 22:40.38 | jffmriii | incorrectly |
| 22:40.48 | jffmriii | if not I will continiue answering questions here |
| 22:41.02 | ibob63 | dlynes_: I can't see the button on the phone. |
| 22:41.12 | dlynes_ | Anyways...fwiw....Zimbra looks like a kick ass suite |
| 22:41.16 | ibob63 | It is a linksys phone. |
| 22:41.26 | dlynes_ | I was quite impressed with the review in Linux magazine |
| 22:41.27 | jffmriii | zimbra is the bomb |
| 22:41.33 | [TK]D-Fender | jffmriii : I'd suggest writing it up and pasting a link on the WIKI, and in the mailing lists |
| 22:41.35 | jffmriii | www.ondecktech.com/zimbra |
| 22:41.44 | generalhan | [TK]D-Fender: ok i set {CALLERID(number)} to a db location ... now how do i call on that once im in the context i want to be in ? |
| 22:41.45 | jffmriii | that is a great idea |
| 22:41.55 | jffmriii | where is the wiki |
| 22:42.04 | dlynes_ | jffmriii: www.voip-info.org |
| 22:42.06 | generalhan | ~wiki |
| 22:42.30 | dlynes_ | jffmriii: You can try the forums on voxilla.com, too |
| 22:42.47 | dlynes_ | jffmriii: And the asterisk-biz mailing list |
| 22:42.50 | ibob63 | I think I will need to do call transfer through asterisk. Can anyone give me some pointers to get started? |
| 22:43.00 | dlynes_ | ibob63: Sipura 941? |
| 22:43.07 | dlynes_ | ibob63: or Linksys SPA-941? |
| 22:43.18 | ibob63 | Linksys SPA-941 |
| 22:43.23 | brookshire | forums.digium.com |
| 22:43.24 | [TK]D-Fender | generalhan : Set(DB(queuevars/${UNIQUEID}-CALLERID)=${CALLERID(number)}) |
| 22:43.29 | brookshire | don't forget that one :) |
| 22:43.32 | dlynes_ | Yeah...I'm pretty sure that can do transfers, ibob63 |
| 22:43.46 | jffmriii | thank you again |
| 22:43.49 | ManxPower | ibob63, what does the linksys manual say about transfers? |
| 22:43.53 | dlynes_ | ibob63: gimme a sec |
| 22:43.59 | [TK]D-Fender | generalhan : Set(CALLERID(number)=${UNIQUEID}) |
| 22:44.02 | generalhan | [TK]D-Fender: thats what i have ... now how do i call on that location once im in the context where i need to use it |
| 22:44.03 | ManxPower | since in SIP the PHONE does the transfer, not Asterisk |
| 22:44.05 | generalhan | ohh |
| 22:44.19 | [TK]D-Fender | generalhan : in your queue dialing I'm sure you know what to do from there... |
| 22:44.20 | *** part/#asterisk jffmriii (n=findme@66.244.161.19) |
| 22:44.44 | ibob63 | Okay, I am going to have a look throught the phone manual :) |
| 22:45.34 | [TK]D-Fender | generalhan : Set(DB(queuevars/${UNIQUEID}-Var1)=some value) |
| 22:46.02 | [TK]D-Fender | generalhan : do MORE things like that before entereing the que and then use the callerid as your KEY like we did in setting those values before entering |
| 22:47.33 | generalhan | [TK]D-Fender: umm im not quite sure i know what you mean |
| 22:48.01 | generalhan | [TK]D-Fender: -Var1 what is that for ? |
| 22:48.08 | ibob63 | okay. You were all right. The phone does do transfer but it is called a blind transfer and its hidden in deep in the menu |
| 22:48.22 | [TK]D-Fender | generalhan : just a SAMPLE of some more values you'd set that you'd like to have available in your Local channel |
| 22:48.28 | generalhan | i see |
| 22:48.35 | generalhan | well the first one didnt work out for me too well |
| 22:48.44 | [TK]D-Fender | generalhan : pastebin it all up. |
| 22:48.54 | generalhan | [TK]D-Fender: gimme just a sec ! |
| 22:48.56 | [TK]D-Fender | generalhan : and fast.. i've got guests coming in real soon :) |
| 22:49.48 | generalhan | http://generalhan.pastebin.ca/53354 |
| 22:50.43 | generalhan | [TK]D-Fender: the first 4 lines arent really anything anymore .. they used to do stuff for me but i havent removed them yet |
| 22:50.45 | generalhan | sorry |
| 22:51.01 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 22:51.17 | [TK]D-Fender | generalhan : You put the DB line in the wrong spot.. that was supposed to be BEFORE the Queue call. |
| 22:51.27 | generalhan | it is ... |
| 22:51.30 | [TK]D-Fender | generalhan : I'll make you a sample out of it... |
| 22:51.50 | generalhan | [TK]D-Fender: [live_xfer] is the context that is dialed BEFORE it gets transfered to the queue |
| 22:52.10 | [TK]D-Fender | generalhan : Or I'm blind :) |
| 22:52.20 | generalhan | lol |
| 22:52.28 | generalhan | i may have figured it out |
| 22:52.38 | generalhan | i think i messed it up on account of my own ignorance |
| 22:52.50 | [TK]D-Fender | generalhan : You are using Agent/ or Local/ for your agents, right? i'd need to see that context as well... tahts where you retreive the info. |
| 22:53.18 | dlynes_ | ibob63: that's ludicrous |
| 22:53.27 | dlynes_ | ibob63: there should be a button on the front of the phone to do that |
| 22:53.27 | [TK]D-Fender | generalhan : I gave you good sample pieces, you just need to fill in the blanks, and we should be maniplating the NAME, not the NUMBER field actually... my bad on that one. |
| 22:53.47 | generalhan | [TK]D-Fender: ok |
| 22:53.51 | [TK]D-Fender | dlynes : SPA's use soft-keys for the transfer/conf features. |
| 22:54.08 | dlynes_ | [TK]D-Fender: ah....so no hard keys? |
| 22:54.22 | dlynes_ | [TK]D-Fender: isn't that a pain in the ass? |
| 22:54.43 | dlynes_ | i mean for such an overused business phone feature |
| 22:54.54 | dlynes_ | you'd think they'd dedicate a button to it |
| 22:54.57 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 22:55.51 | ibob63 | dlynes_: yeah it is pretty stupid. I have been using the phone for several weeks and never worked it out. |
| 22:57.17 | dlynes_ | ibob63: well, you could always try aastras or polycoms...they both have dedicated transfer and conference buttons |
| 22:58.10 | generalhan | [TK]D-Fender: here is everything ... please dont get caught up in the custom monitoring section .. i was looking for a way to record a call AFTER it was answered and this is the best way so far :: http://generalhan.pastebin.ca/53356 |
| 22:58.11 | dlynes_ | ibob63: and they're both probably around the same price as the linksys |
| 23:00.17 | *** join/#asterisk MrDigital (n=VBDIGITA@pool-72-81-113-227.phlapa.east.verizon.net) |
| 23:01.06 | *** join/#asterisk gezick (n=gezick@c-68-50-25-85.hsd1.dc.comcast.net) |
| 23:02.55 | *** join/#asterisk gmaruzz (n=Miranda@217-133-80-112.b2b.tiscali.it) |
| 23:04.25 | luke-jr_ | too bad there's no maintained Asterisk release =p |
| 23:07.04 | mog_work | what? |
| 23:08.23 | luke-jr_ | no release of Asterisk that is being maintained |
| 23:08.26 | mog_work | 1.2 |
| 23:08.41 | mog_work | as well as asterisk-be |
| 23:08.49 | luke-jr_ | no, since bugs in 1.2 are not being fixed |
| 23:09.02 | mog_work | umm do you subscribe to the commit list? |
| 23:09.11 | mog_work | there have been several even today |
| 23:09.18 | luke-jr_ | no, but I've had two bug reports closed saying they won't be fixed |
| 23:09.28 | mog_work | linkage? |
| 23:09.29 | luke-jr_ | just because they don't apply in HEAD |
| 23:09.31 | dlynes_ | luke-jr_: did you read their reasoning? |
| 23:09.43 | luke-jr_ | dlynes: bugs are bugs |
| 23:10.14 | luke-jr_ | http://bugs.digium.com/view.php?id=6825 is the most recent one to be closed unfixed |
| 23:10.51 | generalhan | can some one explain to me how to call on a variable i set in ASTDB? i have a variable stored as Set(DB(queuevars/CALLERID)=${CALLERID(number)}) and i need to know how to pull that now |
| 23:10.53 | mog_work | the ael code is deamed experimental |
| 23:10.57 | mog_work | as it is clearly noted |
| 23:11.06 | mog_work | and has been fixed in the soon to be 1.4 |
| 23:11.22 | generalhan | $CALLERID(number)}=DB(queuevars/CALLERID) ???? |
| 23:12.24 | dlynes_ | luke-jr_: they asked you to try it in trunk, and you didn't try it to see if the problem exists there as well....how can they help you? |
| 23:12.33 | luke-jr_ | dlynes: I doubt it does exist there. |
| 23:12.41 | mog_work | no dlynes it doesnt exist in trunk |
| 23:12.46 | mog_work | as its been rewritten |
| 23:13.02 | mog_work | do you have another one luke-jr_ |
| 23:13.06 | luke-jr_ | dlynes: my point is that *1.2* isn't having the bugs fixed |
| 23:13.41 | dlynes_ | luke-jr_: I've discovered a number of bugs in 1.2 that got fixed |
| 23:14.02 | dlynes_ | luke-jr_: I've only come across one that hasn't gotten fixed, but it's fixed in trunk |
| 23:14.09 | mog_work | luke-jr_, you can feel free to troll |
| 23:14.20 | mog_work | but bugs are being fixed |
| 23:14.41 | dlynes_ | luke-jr_: just because they're not being backported to 1.2 doesn't mean they're not being fixed |
| 23:14.57 | dlynes_ | luke-jr_: try hanging out in asterisk-dev sometime...you'll see just how hard the developers are working |
| 23:15.09 | luke-jr_ | not all bugs |
| 23:15.20 | justinu|laptop | i fixed a few bugs in 1.2 |
| 23:15.55 | ManxPower | I thought all bugs were supposed to be fixed in 1.2 before being fixed in the dev version |
| 23:16.01 | dlynes_ | Even the few minor bugs i see left in 1.2 that I've been able to isolate have workarounds |
| 23:16.11 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:16.20 | mog_work | heh ManxPower we will be sure to fix ALL bugs for 1.4 |
| 23:16.40 | dlynes_ | ManxPower: it sounds like that particular problem though was a design flaw, and the code was completely rewritten |
| 23:16.55 | ManxPower | *nod* |
| 23:16.57 | luke-jr_ | http://bugs.digium.com/view.php?id=7028 was the other one, for reference |
| 23:17.08 | file | we can't backport the AEL stuff into 1.2... because it's not AEL in trunk, it's actually AEL2 :D |
| 23:17.56 | mog_work | i think the response is fairly straight forward luke-jr_ |
| 23:18.07 | luke-jr_ | dlynes_: the 'default' case bug wasn't completely a design flaw |
| 23:18.19 | luke-jr_ | mog_work: indeed, that 1.2 is not being completely maintained anymore |
| 23:18.22 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 23:18.36 | dlynes_ | luke-jr_: yeah, but what they're getting at, is the whole ael project was flawed, so it was completely rewritten |
| 23:18.51 | *** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 23:18.54 | dlynes_ | luke-jr_: the bugs were probably there because the design flaws made them too hard to find |
| 23:19.03 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 23:19.12 | luke-jr_ | dlynes_: probably, but they were still bugs in *correct usage* |
| 23:19.17 | file | it's a way of time and effort to fix AEL1 in 1.2, when it's going to be deprecated totally soon by the AEL in trunk which will become 1.4 in a few months |
| 23:19.32 | justinu|laptop | but I want it NOW! |
| 23:19.48 | Qwell[] | justinu|laptop: feel free to backport it :p |
| 23:19.50 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 23:20.00 | dlynes_ | justinu|laptop: or download and install trunk :) |
| 23:20.07 | file | now you're more then welcome to fix it and submit patches if you so desire |
| 23:20.10 | luke-jr_ | file: so replace AEL1 in 1.2 with AEL2? =p |
| 23:20.14 | justinu|laptop | "this deep fryer can flash fry a buffalo in 45 seconds flat" |
| 23:20.15 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-76.cybersurf.com) |
| 23:20.22 | justinu|laptop | "45 seconds!!?? but I want it NOW!" |
| 23:20.25 | Qwell[] | luke-jr_: go ahead - it's open source |
| 23:20.34 | file | luke-jr_: I won't in the tree, but you can :P |
| 23:22.19 | luke-jr_ | anyway, like I said, 1.2 isn't being completely maintained-- even if there's good reasons for it =p |
| 23:23.36 | dlynes_ | That reminds me |
| 23:23.45 | dlynes_ | I need to submit a bug report for MixMonito |
| 23:23.47 | dlynes_ | I need to submit a bug report for MixMonitor |
| 23:24.01 | justinu|laptop | what's up with it? |
| 23:24.15 | dlynes_ | It creates a dump file that doesn't play |
| 23:24.30 | dlynes_ | If I use Monitor instead, everything's peachy keen |
| 23:24.32 | justinu|laptop | under what conditions? |
| 23:24.38 | justinu|laptop | i use it sucessfully |
| 23:25.31 | dlynes_ | When one call leg is SIP, and the other is on a PRI Zap channel |
| 23:25.51 | justinu|laptop | i've used it in those conditions sucessfully |
| 23:26.13 | dlynes_ | It records the file, and it looks to be approximately the right size |
| 23:26.36 | dlynes_ | But when I use ControlPlayback to play it back, it starts, and then stops right away |
| 23:26.49 | justinu|laptop | ah, i never tried to use that app to play the files back |
| 23:27.05 | dlynes_ | Well, ControlPlayback works just fine for the files produced by Monitor |
| 23:27.06 | justinu|laptop | i bet it's some silly header incompatibility between the two |
| 23:27.09 | dlynes_ | Just not for MixMonitor |
| 23:27.43 | dlynes_ | Yeah, that would make sense, considering it's exiting the file right away |
| 23:28.01 | dlynes_ | Doesn't even seem to be attempting to play it, because it returns control to the dialplan so fast |
| 23:28.08 | dlynes_ | But no errors are eschewed, either |
| 23:28.41 | OloBola | do any of the softphone's allow you to "hyperlink" phone numbers? It would be nice to be able click a number on a page and place a call. |
| 23:30.17 | Dr-Linux | i'm still thinking about what should i do to, monitor only specific extensions from specific context :S |
| 23:32.19 | dlynes_ | OloBola: a lot of the call center software does that |
| 23:32.39 | dlynes_ | OloBola: but I think it's all commercial software |
| 23:33.08 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 23:33.48 | *** part/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 23:33.50 | OloBola | dlynes_: It would be nice if xten or whatever could do this |
| 23:34.02 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 23:34.59 | dlynes_ | OloBola: yeah...they would probably need to write their own html/web container then |
| 23:35.28 | dlynes_ | OloBola: unless the internet explorer active X control allows you to intercept a hyperlink action |
| 23:35.35 | OloBola | oh silly me, forgot about that |
| 23:36.03 | dlynes_ | OloBola: or they could write a Firefox extension to be able to do that |
| 23:36.18 | dlynes_ | OloBola: But I don't know if Firefox extensions are allowed to spawn processes or not |
| 23:36.58 | dlynes_ | iow, for an application, the rewards for doing that much work might not be worth it |
| 23:37.07 | dlynes_ | i.e. for an application like xten |
| 23:37.26 | OloBola | dlynes_: it would be easy enough with a little activex control (internal use only) |
| 23:37.36 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:37.58 | dlynes_ | OloBola: yeah, but you'd have to do html rendering, image rendering, ... all within that active x control |
| 23:39.38 | dlynes_ | OloBola: or, you could register a new application type with internet explorer, write a handler for that application type, read in the web site with the phone numbers in it, preprocess it, and then hand the processed html off onto the IE active x control |
| 23:40.14 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-185-148.lsanca.fios.verizon.net) |
| 23:40.46 | OloBola | click here: <activex control> feed # to xten, dial number via api whatever </activex control> |
| 23:42.50 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:43.30 | MrDigital | anyone wanna help me put together a kick ass asterisk box? |
| 23:44.44 | blitzrage | but I already have several kick ass asterisk boxes :) |
| 23:47.36 | Druken | god, i am bored out of my tree |
| 23:48.00 | MrDigital | Druken: help me build a nice box |
| 23:48.18 | Druken | you pay for it, i'll build it :) |
| 23:48.27 | justinu|laptop | you live in a tree? |
| 23:48.31 | MrDigital | you tell me the parts i pay i build |
| 23:49.03 | Druken | justinu|laptop: well, the house i live in, is made of at least 40% wood... :) |
| 23:49.21 | Druken | MrDigital: well that's no god damn fun for me.... |
| 23:49.28 | *** join/#asterisk pcrook (n=pcrook@fw.latis.com) |
| 23:49.43 | pcrook | wow - big room |
| 23:50.27 | Druken | hmm, yeah... a volcano eruption in the next few days... |
| 23:51.28 | pcrook | Do people answer questions about Digium T1 cards in here? |
| 23:51.46 | Druken | they might.. depends if you ask them or not... hehehe |
| 23:51.50 | justinu|laptop | Druken: only 40%? |
| 23:52.23 | MrDigital | Druken: sure it is |
| 23:52.24 | Druken | justinu: well, it's got a concrete foundation, and a shitload of drywall, and carpet, and other things... glass |
| 23:52.38 | pcrook | I bought a TE110P that I want to use as a straight WAN interface on Fedora - is there a configuration tool? |
| 23:55.42 | justinu|laptop | sounds like my house :P |
| 23:55.58 | Druken | justinu|laptop: i would hope so.. hehe |
| 23:56.07 | redondos | Do you know of any web-based softphone? I know about SIP-communicator, which is a java phone. It is open source, but still under heavy development and still isn't very usable. Anything else you might know of? |
| 23:57.00 | Druken | redondos: i know there's basic ones out there, but you gotta look... |
| 23:57.05 | Druken | iax and sip |
| 23:57.14 | redondos | Druken: I know I have to look :) |
| 23:57.23 | redondos | Druken: I was wondering if you guys could recommend me one. |
| 23:57.38 | Druken | oh crap... i forgot to take that damn movie back to blockbuster again.... oh well |
| 23:58.34 | carrar | The men in black will be looking for you |
| 23:58.54 | Druken | i prefer the women in black :) |
| 23:59.59 | *** join/#asterisk orlok (i=[xq7RFuB@202-44-174-4.nexnet.net.au) |
