| 00:05.45 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 00:13.38 | *** join/#asterisk chendy (n=chatzill@58.60.125.112) |
| 00:14.16 | *** join/#asterisk coppice (n=chatzill@48.169.232.220.dyn.pacific.net.hk) |
| 00:25.30 | *** join/#asterisk t0rrieri (n=Torrieri@nelug/crew/torrieri) |
| 00:33.56 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 00:35.21 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 00:55.31 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 00:58.29 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 01:10.08 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 01:19.22 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 01:41.25 | *** join/#asterisk geneticx (n=geneticx@adsl-10-104-182.mia.bellsouth.net) |
| 01:54.06 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 01:57.13 | *** join/#asterisk ruben23 (n=RPL@121.1.37.146) |
| 01:58.16 | ruben23 | hi can i installe asterisk with dynamic IP on my ISP |
| 01:59.35 | [TK]D-Fender | ruben23: Of course |
| 02:00.36 | ruben23 | <PROTECTED> |
| 02:01.42 | [TK]D-Fender | ruben23: is * behind NAT? |
| 02:03.45 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 02:04.02 | ruben23 | <PROTECTED> |
| 02:04.13 | ruben23 | where should i go better..? |
| 02:04.32 | [TK]D-Fender | ruben23: The one with a public IP does not need anything to operate just fine. the other should follow : |
| 02:04.34 | [TK]D-Fender | ~sipnat |
| 02:04.35 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:04.51 | *** join/#asterisk AlmightyOatmeal (n=jamie@76-255-16-163.lightspeed.mdsnwi.sbcglobal.net) |
| 02:05.08 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 02:05.09 | AlmightyOatmeal | if i set an interface to bind to in sip.conf, will all rtp traffic go through that interface as well? |
| 02:07.40 | carrar | You mean IP? |
| 02:08.29 | ruben23 | [TK]D-Fender: how about for my dynamic connection...? |
| 02:08.40 | carrar | and then I would assume it needs to be the most specific route to the destination in your route table |
| 02:09.13 | [TK]D-Fender | ruben23: .... |
| 02:09.20 | [TK]D-Fender | ruben23: READ THE DAMN INSTRUCTIONS |
| 02:09.34 | carrar | where's the candy |
| 02:10.37 | ChannelZ | takes off his pants |
| 02:11.51 | jaytee | pants candy? |
| 02:12.22 | ChannelZ | It's up to the reader to decide if it's a trick or a treat |
| 02:15.59 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 02:20.03 | AlmightyOatmeal | carrar: yes i meant ip |
| 02:20.17 | AlmightyOatmeal | sighs |
| 02:20.23 | AlmightyOatmeal | more routing table goodness to come |
| 02:22.53 | AlmightyOatmeal | sip traffic seems to go over the second interface, but rtp doesn't |
| 02:23.38 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 02:48.12 | *** join/#asterisk diatonic1 (n=diatonic@71-37-165-223.bois.qwest.net) |
| 02:49.19 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 02:54.11 | manxpower | Anyone know the command to turn the horrid .tex documentation files into something more useful like text? |
| 02:54.31 | manxpower | AlmightyOatmeal: you're not using bindip= are you? |
| 02:55.02 | manxpower | I'd be happy to just view those .tex files over an ssh session, I don't care how. |
| 02:55.18 | manxpower | [TK]D-Fender: they turned channelvariables.txt into a .tex file. |
| 02:57.08 | [TK]D-Fender | manxpower: Welcome to "OMG Years Ago" |
| 02:57.50 | [TK]D-Fender | manxpower: You should be able to read your way throught it in plain-text ignore what little syntax is actually in there |
| 02:58.18 | manxpower | seems like an additional barrier to n00bs |
| 03:00.16 | [TK]D-Fender | manxpower: And you'll notice a pre-compiled PDF in there too.... conglomerating all of the .tex's |
| 03:00.30 | [TK]D-Fender | manxpower: And back to n00b-friendly |
| 03:01.33 | manxpower | [TK]D-Fender: only if you call "n00b friendly" having to copy the file to a machine using a GUI to view the PDF |
| 03:02.01 | *** join/#asterisk rlarson90 (n=rlarson9@CPE001cdf73661f-CM001ceacec55e.cpe.net.cable.rogers.com) |
| 03:02.57 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
| 03:03.51 | *** join/#asterisk path (i=path@server1.bshellz.net) |
| 03:04.08 | [TK]D-Fender | manxpower: You say that as though most newbs aren't going to try just running it on a system running a GUI anyway |
| 03:04.20 | [TK]D-Fender | manxpower: let alone the FreePBX swarm. |
| 03:04.47 | [TK]D-Fender | manxpower: Seriously... its no impediment to the truly lazy, or the slightly intelligent. The problem is the ones in between :) |
| 03:04.57 | manxpower | [TK]D-Fender: I really could not care less about GUI people. |
| 03:05.24 | manxpower | If they are using a GUI I doubt they are looking at the docs in the Asterisk source tree anyway. |
| 03:06.46 | path | if I do sip show users from * CLI this only reads from *.conf files? |
| 03:07.08 | path | I'm playing around odbc+pgsql |
| 03:07.20 | path | and realtime :9 |
| 03:07.38 | path | found database show I can see some users |
| 03:11.27 | *** part/#asterisk darkpixel (n=darkpixe@2001:470:83b7:4242:6b45:3c76:409c:2f43) |
| 03:13.04 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 03:15.23 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 03:20.28 | Katty | peeks in |
| 03:21.05 | Katty | HAPPY HALLOWEEN :> |
| 03:27.11 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 03:42.45 | *** join/#asterisk Caplain (i=shayne@2001:470:5:fb:91aa:aa90:5754:8ae4) |
| 03:48.02 | _ShrikE | give Katty packs of smarties |
| 03:50.27 | ruben23 | awooow |
| 03:55.45 | Katty | meep. |
| 03:56.00 | Katty | how about an apple instead. |
| 03:56.13 | Katty | dark red please :> |
| 04:02.28 | ChannelZ | smarties are the best |
| 04:10.21 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 04:17.21 | *** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1176438379.dsl.bell.ca) |
| 04:17.37 | darkdrgn2k | hey guys, my asterisx process spiked to 80% buy as far as i know its idle... |
| 04:17.40 | darkdrgn2k | any idea why? |
| 04:17.45 | darkdrgn2k | *but as far.. |
| 04:17.52 | darkdrgn2k | or where i can see what its doing |
| 04:18.42 | jblack | clearly something is running. check out the top command |
| 04:18.56 | darkdrgn2k | jblack: as i said ASTERISX process |
| 04:19.04 | darkdrgn2k | its holding at about 80% |
| 04:19.14 | darkdrgn2k | ive restarted asterisk, it just spikes up again |
| 04:19.44 | ChannelZ | turn up the verbosity or debug and see if it's busy doing something over and over |
| 04:20.04 | darkdrgn2k | so what -r -vvvvvvvvv then debug on |
| 04:20.52 | darkdrgn2k | cause i see nothing :-D few registers here or thre.. |
| 04:20.54 | ChannelZ | if verbosity isn't showing anything happening then yes turn on debug and see what it might be spending so much time doing |
| 04:21.16 | Katty | core set verbosity 10 |
| 04:21.27 | Katty | or maybe it's verbose 10 |
| 04:21.28 | Katty | i forget |
| 04:21.34 | darkdrgn2k | verbose 10 :) |
| 04:21.36 | darkdrgn2k | still nothing |
| 04:21.44 | darkdrgn2k | did debug 10 as well |
| 04:21.46 | darkdrgn2k | nothing |
| 04:22.11 | darkdrgn2k | hmm well its trying to register to an account thats been disabled... |
| 04:22.17 | darkdrgn2k | but would that cause it to spike to 80% |
| 04:23.08 | Katty | why don't you disconnect the phone/software and see what happens. |
| 04:23.33 | darkdrgn2k | k |
| 04:23.38 | darkdrgn2k | but i dont see a flood of messages.. |
| 04:23.43 | darkdrgn2k | in fact thers like nothign |
| 04:23.56 | darkdrgn2k | 4 attempts to registered teh disabled account..... and then nothing |
| 04:24.35 | darkdrgn2k | WOW i think i just found the problem :-P |
| 04:24.47 | darkdrgn2k | 100% harddrive :-S |
| 04:24.57 | darkdrgn2k | wtf is using100% |
| 04:25.00 | ChannelZ | oops |
| 04:25.11 | ChannelZ | logfiles gone wild |
| 04:25.59 | darkdrgn2k | 35G ./log |
| 04:26.00 | darkdrgn2k | hmmm |
| 04:26.52 | darkdrgn2k | -rw-r----- 1 asterisk asterisk 6.6G Oct 31 04:03 full.1 |
| 04:26.54 | darkdrgn2k | yep .... |
| 04:27.24 | darkdrgn2k | hmmm |
| 04:27.27 | darkdrgn2k | [Oct 30 04:03:25] NOTICE[5080] chan_iax2.c: Host 127.0.0.1 failed MD5 authentication for '1000' (3e587562080cc044e14fdb8343cba75e != e42d63f64f63687dc52027c6 |
| 04:27.28 | darkdrgn2k | 6048fe64) |
| 04:31.12 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 04:32.24 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:32.25 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 04:35.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:39.06 | Katty | drops a pen |
| 04:39.55 | ChannelZ | takes a picture |
| 04:45.52 | ppc | yo |
| 04:46.12 | Katty | herro. |
| 04:46.21 | ppc | whats up |
| 04:46.28 | Katty | having snack |
| 04:46.58 | Katty | burning a cinnamon vanilla candle...watching virtual fireplace on youtube on one lcd screen, and restaurant city flash game on another. |
| 04:47.15 | ppc | sounds depressing |
| 04:47.30 | Katty | why does that sound depressing? |
| 04:47.56 | ppc | a virtual fire place? |
| 04:48.12 | Katty | it's soothing |
| 04:48.23 | ppc | I'm watching K-1 kickboxing |
| 04:48.47 | Katty | meh, kickboxing |
| 04:48.55 | Katty | meh, sports ;) |
| 04:49.16 | Katty | you watch kickboxing, i'll watch my fireplace |
| 04:49.22 | Katty | and the boyfriend is watching Supernatural behind me |
| 04:49.47 | ppc | that sounds familiar |
| 04:50.01 | Katty | Dean and Sammy fight the forces of evil. |
| 04:55.23 | Katty | doi want to buy an electric blanket? |
| 04:59.35 | Katty | http://ecx.images-amazon.com/images/I/511bt7lGTCL._AA260_.jpg <- pretty |
| 05:00.16 | ChannelZ | http://pixdaus.com/pics/fKbTWxPnVZ2E.jpg |
| 05:00.51 | Katty | that's good camo |
| 05:01.03 | Katty | i didn't know what i was supposed to be looking for there for a second |
| 05:01.21 | ChannelZ | hehe |
| 05:01.54 | Katty | ChannelZ: what kind of blanket do you keep on your bed in the winter? |
| 05:02.21 | ChannelZ | I just have a comforter that I use year round (I just sleep on top of everything in the summer) |
| 05:02.31 | Katty | ah, k |
| 05:02.39 | ChannelZ | It's just a tan color, no pattern.. I think from Nautica |
| 05:03.14 | Katty | nods |
| 05:03.40 | ChannelZ | it needs to be washed. I need to find a laundrymat that has big-ass washers |
| 05:04.07 | Katty | nods |
| 05:04.13 | Katty | you can spritz it with vodka in the meantime |
| 05:04.23 | Katty | the el cheapo brand will do fine |
| 05:04.51 | ChannelZ | hmm |
| 05:06.32 | Katty | http://www.thedailygreen.com/going-green/tips/fresh-laundry-vodka-460808 |
| 05:06.41 | Katty | vodka is a natural disinfectant. |
| 05:07.20 | leifmadsen | zup |
| 05:07.26 | Katty | hi leif |
| 05:07.29 | leifmadsen | I have been playing with ableton live far too much lately |
| 05:07.32 | Katty | what do you two keep on your bed in the winter? |
| 05:07.47 | ChannelZ | leifmadsen: do you know Kid Beyond? |
| 05:07.57 | leifmadsen | Katty: we keep a sheet, comfortor, and think blanket |
| 05:08.04 | leifmadsen | ChannelZ: can't say I do |
| 05:08.12 | leifmadsen | <-- deth nesdam! |
| 05:08.18 | Katty | what sort of blanket? |
| 05:08.26 | ChannelZ | He's a beatboxer who uses Ableton to do live looping, builds up tracks |
| 05:08.29 | leifmadsen | Katty: just a simple cotton one I think but we don't really use it |
| 05:08.35 | Katty | nods |
| 05:08.50 | Katty | i threw some flannel sheets on the bed earlier this week, along with the quilt...but i'm still freezing |
| 05:09.03 | Katty | another quilt, and it's too hiot |
| 05:09.04 | Katty | hot |
| 05:09.13 | Katty | perhaps a cotton blanket will do nicely. will give it a shot. |
| 05:09.15 | leifmadsen | ChannelZ: sounds fun! I'm just learning the software and playing around. I'm traditionally a drummer, and I just got it hooked up via MIDI and learned the Drum Rack effect so I can create my own kits now |
| 05:09.42 | leifmadsen | Katty: ya, the flannel SHOULD be enough typically, but i prefer things to be a bit cool |
| 05:09.47 | ChannelZ | cool |
| 05:10.09 | leifmadsen | Katty: cotten is good because it tends to breathe when too hot, but warm you up when too cold |
| 05:10.10 | Katty | leifmadsen: yeah..i'm always freezing :/ |
| 05:10.22 | leifmadsen | Katty: I'm always way too hot -- I'm like a furnace |
| 05:10.30 | Katty | :< |
| 05:10.31 | Katty | envy you |
| 05:10.50 | Katty | has space heater on right now. |
| 05:10.57 | ChannelZ | I think I'm going to head off to the range for a bit |
| 05:11.04 | Katty | kk, sleep well |
| 05:11.15 | leifmadsen | Katty: I'm sweating right now... and I don't even have the heat on |
| 05:11.15 | ChannelZ | no sleeping, gunfire! |
| 05:11.21 | leifmadsen | ChannelZ: nice :) |
| 05:12.04 | leifmadsen | man, there is almost an overwhelming amount of stuff I could learn about Ableton |
| 05:13.59 | drmessano | The Will Smith movie? |
| 05:14.21 | drmessano | Oh, nm |
| 05:16.21 | [TK]D-Fender | leifmadsen: And I jsut bought a bass yesterday :) |
| 05:16.38 | leifmadsen | [TK]D-Fender: awesome! |
| 05:16.45 | leifmadsen | drmessano: heh, no :) |
| 05:16.50 | leifmadsen | Ableton Live :) |
| 05:17.22 | Katty | hey fender. |
| 05:17.49 | [TK]D-Fender | leifmadsen: I have the Live Lite version from my earlier M-Audio purchases, but since I don't really run Windows I never bothered to use it. Plenty of alternatives here however |
| 05:18.12 | [TK]D-Fender | leifmadsen: Then again.. I'm a "live" kinda guy, and my jam book has almost hit 110 pgs |
| 05:18.34 | drmessano | You guys ever use Reaper? |
| 05:18.49 | leifmadsen | [TK]D-Fender: nice -- I don't run Windows typically either, but I installed it on my MacBook Pro, so now I'm able to triple boot between OSX, Linux, and Windows |
| 05:21.33 | *** join/#asterisk dkirker_ (n=dkirker@24-180-2-10.dhcp.snlo.ca.charter.com) |
| 05:44.05 | *** join/#asterisk ardor^ (n=IceChat7@ip72-193-201-128.lv.lv.cox.net) |
| 05:44.15 | ardor^ | I am trying to use the shell command in asterisk. |
| 05:44.34 | ardor^ | ERROR[26128]: pbx.c:1550 ast_func_read: Function SHELL not registered |
| 05:51.34 | [TK]D-Fender | ardor^: What version? |
| 05:51.56 | ardor^ | I am installing updates now, that might fix it |
| 05:52.00 | ardor^ | Thanks Fender |
| 05:52.24 | ardor^ | I guess I;ll go watch 40 mins of TV. |
| 05:54.48 | ppc | blah |
| 05:56.15 | *** join/#asterisk coppice (n=chatzill@202.64.176.34) |
| 06:43.49 | Katty | sighs. |
| 06:43.55 | Katty | shouldn't have had that caffeine around 2 :/ |
| 06:44.05 | Katty | how long does caffeine stay in your system anyway? |
| 06:44.31 | Katty | oh nice. 18 to 20 hours. |
| 06:44.34 | Katty | shoot me. |
| 06:45.06 | *** join/#asterisk puzzled (n=foobar@535335AA.cable.casema.nl) |
| 06:47.45 | ardor^ | Well doing the updates didnt work. |
| 06:48.39 | ardor^ | Asterisk 1.4.21.2 |
| 06:48.50 | ardor^ | sigh, I need Asterisk 1.6 for Shell to work.. huh. |
| 06:49.16 | ardor^ | SHELL: Returns output of a shell command. (1.6) |
| 07:02.05 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 07:17.32 | ardor^ | Removing my 1.4 installtion... |
| 07:24.08 | *** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net) |
| 07:24.12 | ramindia | hi |
| 07:24.15 | ramindia | any one around |
| 07:24.26 | ardor^ | not that i know of |
| 07:25.14 | ramindia | i want to use passthrough g729 bin from "http://asterisk.hosting.lv/" i have xeon processor, which one i need to download |
| 07:27.59 | ardor^ | no ideal what your asking, I know g729 is a codec. |
| 07:31.21 | *** join/#asterisk errotan (n=errotan@a1775.adsl.pool.eol.hu) |
| 07:31.22 | ramindia | never mind |
| 07:36.08 | ChannelZ | farts a little |
| 07:42.30 | *** join/#asterisk tzafrir__laptop (n=tzafrir@local.xorcom.com) |
| 07:55.08 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:01.48 | mchou | asterisk documentation is TERRIBLE |
| 08:03.02 | mchou | I'm reading through sip.conf.sample and the sample conflicts with itself |
| 08:04.16 | ChannelZ | yes... yes it is |
| 08:05.31 | mchou | ; Don't mix extensions with the names of the devices. Devices need a unique |
| 08:05.31 | mchou | ; name. The device name is *not* used as phone numbers. Phone numbers are |
| 08:05.31 | mchou | ; anything you declare as an extension in the dialplan (extensions.conf). |
| 08:06.39 | mchou | later on down the file they then do this: |
| 08:06.44 | mchou | ; and finally instantiate a few phones |
| 08:06.44 | mchou | ; |
| 08:06.44 | mchou | ; [2133](natted-phone,my-codecs) |
| 08:06.45 | mchou | ; secret = peekaboo |
| 08:06.45 | mchou | ; [2134](natted-phone,ulaw-phone) |
| 08:06.46 | mchou | ; secret = not_very_secret |
| 08:06.48 | mchou | ; [2136](public-phone,ulaw-phone) |
| 08:06.50 | mchou | ; secret = not_very_secret_either |
| 08:06.54 | mchou | WTF??? |
| 08:07.30 | mchou | dumbasses |
| 08:08.07 | mchou | FAILBUS |
| 08:08.24 | ChannelZ | I hate examples like that |
| 08:09.00 | mchou | they can't even get the freaking terminology straight |
| 08:09.23 | mchou | what the hell is a "device?" |
| 08:09.51 | mchou | Do they mean a "Device class" as in all Polycom phones? |
| 08:09.53 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 08:10.09 | mchou | an instance of a device class? |
| 08:10.33 | ChannelZ | they're equating a 'sip user' (*I* don't even know what to call it) as being a device I guess |
| 08:10.35 | mchou | a freakinga sip ua with an account?? |
| 08:11.01 | mchou | whoever wrote this shit should be shot |
| 08:11.15 | ChannelZ | I can help with that |
| 08:11.25 | ChannelZ | but you can submit changes :) |
| 08:11.35 | mchou | hell no |
| 08:11.52 | mchou | documentation like that is beyond help |
| 08:11.57 | ChannelZ | well then stop complaining. Such is the life of open source software |
| 08:12.14 | ChannelZ | You get what you pay for |
| 08:12.42 | mchou | indeed |
| 08:12.57 | ChannelZ | What's the saying? "If you're not part of the solution, you're trying to google for it.." |
| 08:14.11 | mchou | the sad part was this is so confused it's not even googlable |
| 08:14.20 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 08:15.13 | mchou | I sae someone using syntax like allow=ulaw,alaw, just wanted to know if that was legit |
| 08:15.23 | mchou | saw* |
| 08:16.12 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 08:16.24 | mchou | you go over to the asterisk site and none of the sip.conf parameters are even documented |
| 08:20.10 | ppc | Why cant' you complain about it? |
| 08:22.22 | mchou | cause I'll just get responses like: [01:11:25] <ChannelZ> but you can submit changes :) |
| 08:24.12 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 08:29.20 | *** join/#asterisk plundra (i=404@article.se) [NETSPLIT VICTIM] |
| 08:29.20 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 08:46.46 | mumtazah1 | hello, i would like to ask, how to handle asterisk through web |
| 08:46.49 | mumtazah1 | like web can see the status |
| 08:49.13 | trogs | mchou: all of the sip.conf parameters are documented here - http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-A-SECT-2.html (this is the asterisk book in html form, which is actually far bettery than trying to read any of the docu or examples |
| 08:50.01 | mchou | trogs: I'm not so sure that's current |
| 08:50.20 | mchou | trogs: the reason for this was I upgraded |
| 08:50.59 | mchou | needed to find out what new keywords/parameters meant |
| 08:57.11 | *** join/#asterisk plundra (i=404@article.se) [NETSPLIT VICTIM] |
| 08:58.39 | trogs | right |
| 08:58.43 | trogs | yeah teh book is quite old now |
| 09:02.21 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-71-238.link.net.pk) |
| 09:02.56 | *** join/#asterisk puzzled (n=foobar@535335AA.cable.casema.nl) |
| 09:11.09 | ChannelZ | Look I think the docs suck too and I said as much |
| 09:12.11 | ChannelZ | But then I suggested fixing some of it you basically said 'fuck that'. So if your time is better spent whining, well knock yourself out |
| 09:25.16 | trogs | hard to fix if the information that needs fixing is what you're actually looking for |
| 09:27.10 | ChannelZ | of course but he was lamenting about inconsistencies in terminology and such as well |
| 09:39.38 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 09:51.44 | *** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be) |
| 10:05.43 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 10:23.39 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 10:47.10 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 10:51.01 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 10:59.57 | *** join/#asterisk Godfather_ (n=godfathe@62.43.134.46.dyn.user.ono.com) |
| 10:59.59 | Godfather_ | hi |
| 11:01.56 | TSM2 | wot :) |
| 11:02.10 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 11:02.10 | Godfather_ | im trying to set up an asterisk server. It seems is already running ( Asterisk PBX is already running. Use restart ), but i cant connect with the softphones... |
| 11:02.43 | Godfather_ | i tried ekiga and linphone with no succeed |
| 11:02.52 | mchou | dude |
| 11:03.02 | mchou | ~thebook |
| 11:03.03 | infobot | from memory, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 11:03.54 | mchou | Godfather_: dont expect to install asterisk and have it "just work" |
| 11:04.05 | mchou | configuration is necessary |
| 11:04.19 | Godfather_ | mchou, i followed those configurations in /etc/astersik |
| 11:04.26 | Godfather_ | i created some users |
| 11:04.46 | Godfather_ | i'll read this book and will be back |
| 11:04.49 | mchou | you followed _which_ configurations? |
| 11:04.53 | Godfather_ | mmm |
| 11:05.10 | Godfather_ | http://www.alcancelibre.org/staticpages/index.php/como-ekiga-asterisk |
| 11:05.23 | Godfather_ | its in spanish, but should work |
| 11:05.52 | Godfather_ | i mean, i edited sip.conf, and manager.conf |
| 11:06.06 | Godfather_ | with the examples on that web (just copy & paste) |
| 11:06.21 | mchou | haha, that's rich |
| 11:06.32 | mchou | it should work, but isnt |
| 11:06.43 | Godfather_ | yep, you are right hehe |
| 11:07.03 | Godfather_ | i tried to connect from the same host, it could be a problem? |
| 11:07.24 | mchou | yup |
| 11:07.53 | mchou | you need different ports of softphone and asterisk is on the same host |
| 11:08.53 | mchou | something that's definitely not mentioned in "Configuración de cliente Ekiga." |
| 11:09.57 | mchou | that's not actually not a bad write-up but you still need to read the book |
| 11:09.59 | Godfather_ | mchou, well, i remember when i opened linphone it says ' UDP port 5060 seem already in use! cannot initzialize" |
| 11:10.04 | Godfather_ | or something similar |
| 11:10.08 | Godfather_ | ok |
| 11:10.12 | mchou | precisely |
| 11:10.23 | mchou | that is your problem |
| 11:10.42 | mchou | that page has good instructions |
| 11:11.01 | Godfather_ | now i will try from another host of my net i just try to connect, then ill read the book, seems pretty |
| 11:11.07 | mchou | you just need to change to port 5061 or whatever on the softphone |
| 11:18.00 | *** join/#asterisk Tim_Toady (n=moi@adsl95-119.kln.forthnet.gr) |
| 11:19.07 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 11:22.56 | lesouvage | Does any of you have an idea about the cost range of gsm picocells that can be used to set up your own inhouse gsm network (that can be integrated into an Asterisk based solution) |
| 11:27.53 | TSM2 | ive not seen any small picocells |
| 11:28.04 | TSM2 | i think its all regulated, well in the uk it is |
| 11:34.18 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 11:38.34 | *** join/#asterisk Godfather_ (n=godfathe@187.Red-83-38-229.dynamicIP.rima-tde.net) |
| 11:42.20 | *** join/#asterisk Godfather_ (n=godfathe@62.43.134.46.dyn.user.ono.com) |
| 11:52.44 | lesouvage | TSM2: in Holland it is allowed to set up an inhouse gsm with low power gsm picocells. My first impression is that it is the dreamed solution that avoids lots of points of failure. But I can't find any info about the costs of setting up an inhous gsm network. |
| 11:55.02 | lesouvage | That might be a major barrier for implementation. |
| 11:55.31 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
| 11:55.40 | *** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl) |
| 12:01.02 | mchou | damn |
| 12:01.14 | mchou | nortel fire sale looks really tempting |
| 12:02.00 | mchou | LG-Nortel 6812 for $30 |
| 12:02.19 | mchou | wonder if it plays well with asterisk |
| 12:10.06 | *** join/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net) |
| 12:12.38 | TSM2 | thats one horible phone |
| 12:13.23 | mchou | TSM2: hmm?? |
| 12:13.50 | mchou | TSM2: you speak from personal experience with this phone? |
| 12:14.06 | TSM2 | nop |
| 12:14.17 | TSM2 | looks horible |
| 12:14.21 | mchou | TSM2: lol |
| 12:14.33 | TSM2 | out of all the phones i thing that polys seem to be at the top |
| 12:14.40 | TSM2 | look wise |
| 12:14.46 | TSM2 | then Astra |
| 12:14.49 | mchou | TSM2: I haven'rt seen it in person but it's built like a tank, Ive heard |
| 12:15.08 | mchou | polycoms dont work fro me |
| 12:15.13 | mchou | volume is too low |
| 12:15.17 | mchou | for* |
| 12:15.53 | mchou | I had a polycom for all off two weeks and went back to my old phone |
| 12:16.06 | TSM2 | there are XML settings to sort that out |
| 12:16.35 | mchou | TSM2: what makes you think I didnt try that? |
| 12:17.00 | TSM2 | because ive seen a few people use the wrong settings, there are about 20 diffrent ones |
| 12:17.28 | mchou | TSM2: I used thr right settings all right |
| 12:17.32 | mchou | the* |
| 12:17.40 | mchou | polycoms still suck |
| 12:18.01 | TSM2 | naa they dont, im not having any probs with them |
| 12:18.13 | mchou | bullshit |
| 12:18.17 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 12:18.18 | TSM2 | im about to roll out a 35phone system in one of our offices tommorow |
| 12:18.25 | *** part/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 12:18.35 | mchou | for speakerphone the volume is so low to be unusable |
| 12:18.59 | TSM2 | naa, had no issues with that |
| 12:19.07 | mchou | it's lame ass junk |
| 12:19.12 | TSM2 | if i want it that loud ile have a conf phone |
| 12:19.20 | mchou | I took the shit apart |
| 12:19.33 | mchou | teeny lil speaker |
| 12:19.39 | TSM2 | which model |
| 12:19.40 | mchou | cheapskates |
| 12:20.05 | mchou | 501 |
| 12:20.11 | TSM2 | old series |
| 12:20.15 | mchou | so? |
| 12:20.24 | TSM2 | ive got 330/331/450/550 |
| 12:20.37 | mchou | sucked back then, still havent changed |
| 12:21.05 | TSM2 | keep deluding, they are the most popular phones, must be for a reason |
| 12:21.27 | mchou | 50 billion flies eat shit |
| 12:21.30 | TSM2 | most other manufs need to step up to the plate and create better looking phones |
| 12:21.42 | TSM2 | dont need that many files, there is a way around that |
| 12:21.43 | mchou | doesnt mean you should follow the fly |
| 12:21.53 | *** join/#asterisk voipmonk (n=voipmonk@69.172.83.148) |
| 12:22.21 | mchou | TSM2: you sound like someone who favors style over substance |
| 12:22.42 | TSM2 | nop, i look at both equaly, you dont want shit on your corporate desks |
| 12:22.47 | mchou | it's not as if polycoms are pretty |
| 12:23.16 | TSM2 | better than cisco/linksys etc.. i have to say only astra is on par with poly |
| 12:23.27 | mchou | corporate desks? |
| 12:23.37 | mchou | are you frigging daft? |
| 12:23.42 | TSM2 | less than you |
| 12:23.56 | mchou | which "visitors" are going to see them? |
| 12:24.08 | mchou | this is rich |
| 12:25.38 | mchou | "Oh, they've got shit looking phones, must be an awful place to work" |
| 12:25.54 | TSM2 | no its the whole package |
| 12:26.15 | mchou | TSM2: whole package? |
| 12:26.16 | TSM2 | at the end of the day i think polys work well, good config |
| 12:27.07 | mchou | TSM2: I aasked you at the beginning of this convo whether you had any personal experience with these phones |
| 12:27.48 | mchou | since you replied no, how are you coming to the conclusion that polycoms represent a superior "package?" |
| 12:28.31 | mchou | TSM2: or you just like to talk smack? |
| 12:29.22 | *** join/#asterisk naif (n=naif@93-32-138-134.ip33.fastwebnet.it) |
| 12:29.24 | naif | okhi all |
| 12:29.31 | naif | after in depth analisys |
| 12:29.49 | naif | i conclude that there is no way to have a "modem terminal" to establish even a v.21 300bps connection trough VoIP |
| 12:29.58 | naif | not even there are commercial software doing it |
| 12:30.02 | naif | it's incredible |
| 12:30.19 | mchou | naif: it's obvious |
| 12:30.33 | voipmonk | have you tried TIA or SliRP from back in the day, naif? |
| 12:30.45 | voipmonk | or SLIPPP |
| 12:30.56 | voipmonk | old skewl |
| 12:30.59 | naif | is not obvious, with ulaw and packet redundancy it may be possible to reach even 9600bps. I just was looking for a dirty low speed carrier |
| 12:31.20 | mchou | it aint about the bandwidth |
| 12:31.36 | naif | voipmonk: SliRP is not a DSP |
| 12:31.45 | naif | here the problem is the DSP software's available (spandsp) |
| 12:32.04 | naif | that does not speak the modem connections others than FAX ones |
| 12:32.19 | naif | and there's not even commercial software to do it. |
| 12:32.30 | naif | The only way is to get an ATA and use the RJ11 analog port trough a real modem |
| 12:32.37 | mchou | naif: you cant fit 15 lbs of crap in a 5 lb. bag |
| 12:32.53 | naif | it's plenty of ppl even reaching 19.200bps using sipura ATA and g711 making modem over VoIP |
| 12:32.59 | naif | but there's no software solution |
| 12:33.14 | mchou | naif: wideband(modem) and narroband(speech) |
| 12:33.25 | mchou | narrowband* |
| 12:33.28 | naif | i wrote to spandsp author to ask him if he can accept a consultancy to include some signaling |
| 12:33.32 | naif | even with commercial DSP libraries |
| 12:33.42 | naif | because most of the 'standard' modem are patented |
| 12:33.53 | naif | and so usually are only available as a commercial implementation |
| 12:34.05 | naif | a valid implementation would be http://fabrice.bellard.free.fr/linmodem.html |
| 12:34.09 | naif | to replace spandsp |
| 12:34.12 | naif | with IAXmodem |
| 12:34.14 | mchou | naif: you cant violate laws of physics |
| 12:34.29 | naif | mchou: modem over VoIP can be done. It works using VoIP ATA |
| 12:34.39 | florz | of course it can be done |
| 12:34.52 | mchou | naif: "works" would be putting it charitably |
| 12:34.53 | naif | is not a matter of laws of physics but of availability of software implementation of the modem |
| 12:35.00 | naif | damn |
| 12:35.23 | mchou | naif: ask vonage how they are doing with the class action lawsuit on failed fax |
| 12:35.57 | mchou | naif: you know anyone that thinks it's even close to RELIABLE? |
| 12:36.01 | naif | mchou: The problem with "SOME" fax machine is that only get carrier at 14400bps and 14400bps are difficult to reach on VoI |
| 12:36.08 | naif | and that' the only bug |
| 12:36.26 | naif | if remote fax modem stay 4800bps you would almost certanly get a carrier |
| 12:36.33 | naif | 'purely' technically speaking |
| 12:36.39 | mchou | yeah |
| 12:36.57 | mchou | except there is the real world outside the lab |
| 12:37.12 | naif | i know |
| 12:37.51 | naif | I am wondering how much could it costs and to which ask to integrate Linmodem ( that is made specifically for MODEM http://fabrice.bellard.free.fr/linmodem.html) as a DSP backend for IAXModem replacing the spandsp (that is made specifically for FAX) |
| 12:37.56 | florz | it simply depends on the quality of the network, the reduncancy used, and the jitter buffers at the ends - there is nothing magic about modem connections over VoIP |
| 12:38.00 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 12:38.35 | mchou | florz: jitter buffers? are you kidding me? |
| 12:38.51 | florz | mchou: no, why should I? |
| 12:39.21 | mchou | florz: cause if you have ANY jitter all bets are off on fax |
| 12:39.34 | florz | mchou: that's pretty obviously bullshit |
| 12:39.49 | mchou | florz: hell no |
| 12:39.55 | florz | mchou: hell yes |
| 12:40.12 | naif | florz: but in fact when T38 is in place it zeroize the jitterbuffer and provide packet redundancy at RTP level in asterisk |
| 12:40.59 | florz | naif: well, yeah, T.38 is a slightly different business, of course - I was speaking of G.711 |
| 12:41.07 | mchou | lol |
| 12:42.06 | florz | mchou: seriously, you obviously don't understand the problem |
| 12:42.19 | naif | does anyone know who could be asked/provided a consultancy to make such integration? |
| 12:42.43 | florz | mchou: the problem with jitter and fax is when you have jitter in the analog signal - which is exactly what you make disappear by using a jitter buffer |
| 12:43.29 | mchou | florz: riddle me this: G.711 is a vocoder. How you make modem tones fit inside a vocoder? |
| 12:43.42 | florz | mchou: a jitter buffer simply transforms jitter into latency - and since fax doesn't care much about latency, it's not too difficult to get rid of all jitter |
| 12:43.52 | mchou | oh lord |
| 12:44.18 | florz | mchou: whatever you mean by a "vocoder" - you mean, like, one cannot build modem connections through the PSTN? |
| 12:44.28 | mchou | oh my god |
| 12:44.39 | florz | yes, please? |
| 12:44.55 | TSM2 | G729 you will have a major issue, thats a proper vocoder |
| 12:45.21 | mchou | copper aint tuned just to accomdate speech. Speaker wires on your stereo prove that |
| 12:45.23 | florz | mchou: is it possible that you are not aware of the fact that the whole PSTN (well, pretty much) transfers everything in G.711? |
| 12:45.28 | TSM2 | G711 is just a standard audio codec without any phycoacustic encoding |
| 12:47.11 | florz | well, depends on whether you consider the logarithmic sensitivity of the ear a "psychoacoustic" effect, I guess |
| 12:47.40 | TSM2 | psychoacoustic encoding has voice properties coded in, in a way its predictive |
| 12:49.23 | TSM2 | you could never run fax/data using G729 |
| 12:49.40 | florz | well, depends on the kind of modem, obviously =:-) |
| 12:50.52 | TSM2 | true, but i dont think one exists |
| 12:51.19 | florz | does a human count? |
| 12:51.28 | TSM2 | well i was gona write that |
| 12:51.33 | florz | *g* |
| 12:53.02 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:56.39 | florz | mchou: oh, and just in case: the current telephone network does not use relays much anymore - so, you don't get a pair of copper wire from start to end anymore in most places (as in: everywhere), but just a 8 kHz/8bit G.711 audio channel that probably even has a low frequency limit (so, no DC voltage transfer) |
| 12:57.17 | mchou | are you daft? |
| 12:57.26 | florz | are you? |
| 12:57.34 | mchou | I wasnt talking about voltage transfer |
| 12:59.48 | florz | oh, IC, you wanted to use mechanical impulses through the copper wire? |
| 12:59.48 | florz | clever idea, saves you the microphone :-) |
| 12:59.48 | florz | but that would be some noisy modems, I guess ... |
| 13:07.28 | *** join/#asterisk gurra (n=gurra__@unaffiliated/gurra) |
| 13:07.55 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:09.37 | *** join/#asterisk kazaa_lite (n=msaleem@cpc6-slam5-2-0-cust171.2-4.cable.virginmedia.com) |
| 13:12.12 | *** join/#asterisk wierdo (n=chatzill@91.210.88.226) |
| 13:14.57 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 13:19.40 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-71-238.link.net.pk) |
| 13:23.31 | *** join/#asterisk blop (n=blop@217.112.179.12) |
| 13:23.35 | blop | hi |
| 13:23.43 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 13:25.17 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 13:26.18 | *** join/#asterisk wierdo (n=chatzill@91.210.88.226) |
| 13:29.32 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 13:34.51 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1096762451.dsl.bell.ca) |
| 13:36.34 | dlynes | If you get 'Beginning cache-load run for flavor 'nocona'...', or something similar when you run the benchtest for asterisk fax, and it sits there and never returns, is it going to do it every single time? |
| 13:37.29 | *** join/#asterisk lwh (n=lwh@66.212.183.122.tor.pathcom.com) |
| 13:37.37 | *** join/#asterisk Godfather_ (n=godfathe@62.43.134.46.dyn.user.ono.com) |
| 13:49.38 | *** join/#asterisk PoWeRKiLL (n=powerkil@bzq-79-178-125-15.red.bezeqint.net) |
| 13:50.22 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-66-71.link.net.pk) |
| 13:59.18 | *** join/#asterisk bsaxon (n=bryantsa@97.82.27.250) |
| 14:04.22 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167978043.dsl.bell.ca) |
| 14:04.28 | *** join/#asterisk dandate2 (n=mangy@112.202.196.127) |
| 14:04.43 | dandate2 | what are the stipulations against wardialing a competitors toll-free line to rack up their bill. any legalities? |
| 14:05.06 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:07.54 | *** join/#asterisk Zarag (n=J0ff@out.clearnet.com) |
| 14:07.57 | dandate2 | i ask because some losers in the philippines been directing their traffic to my 800# in hopes that i provide service for their fraudulent service. i have 24 dial up modems available, if i pound their toll free line with calls will comcast shut my service off? |
| 14:07.58 | wam | lol |
| 14:09.27 | Zarag | Here is what I want to do: I want to connect to my asterisk server with a DISA command to make an external call. Once that external call is connected with the person, I want to bridge them over a meetme conference. How can I do this? |
| 14:10.35 | *** join/#asterisk cosmicwombat_ (n=cosmicwo@69.7.44.68) |
| 14:10.42 | TSM2 | transfer them into the conference, then make another call into the system and put youself into the conference |
| 14:11.08 | Zarag | TSM2: how do I transfer them to the conference when the call is active? |
| 14:11.32 | TSM2 | ## |
| 14:11.47 | TSM2 | or *2 i think |
| 14:11.54 | TSM2 | ## is blind transfer |
| 14:12.02 | TSM2 | *2 is attended transfer |
| 14:12.26 | Zarag | but where do I configure it so that asterisk knows which conference to transfer to? |
| 14:12.30 | TSM2 | or just a DDI into the meetme app, then they need to put in pin code to getinto the room |
| 14:13.49 | dandate2 | i guess if comcast shuts my serviecv off i can tell em my vpn network got hacked |
| 14:17.23 | [TK]D-Fender | Zarag: You TRANSFER them. When you transfer the call you dial the extension in your dialplan to send them to. So pick one that leads to MeetMe. |
| 14:17.45 | [TK]D-Fender | dandate2: Riiiight... |
| 14:18.02 | [TK]D-Fender | dandate2: Bad idea mentioning VPN, because that should be a LOT harder to crack than anything else |
| 14:18.47 | Zarag | Ok makes sense thx |
| 14:20.59 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 14:21.01 | Zarag | TKD-Fender: What if my outside call is a standard (non voip) verizon bridge that from there I also need to enter pounds to get in it? |
| 14:21.08 | dandate2 | well im just wondering, if i war dial an 800# will comcast shut me down? am i breaking any laws? |
| 14:21.42 | TSM2 | why do you want to wardial 800 numbers |
| 14:21.54 | [TK]D-Fender | dandate2: Grow up |
| 14:22.21 | dandate2 | cuz thers this company in the philippines that is frauding me |
| 14:22.34 | dandate2 | they dont have much money so i figure i can finish em off with a few days of mass calling on their toll free |
| 14:22.42 | [TK]D-Fender | dandate2: And Iwar-dialing is an "attack" and they could shut you down for legal, or perhaps contract reason, etc. They could simply consider it "network abuse". |
| 14:23.02 | TSM2 | dandate2: they will just BL you number soon |
| 14:23.18 | dandate2 | prolly finish em off before they can do that |
| 14:23.28 | TSM2 | doubt it |
| 14:23.51 | dandate2 | for sure, they are housingfinancials.com , their website doesnt even work. but if u call their hotline they say they work at my home address, give you my toll free line |
| 14:24.25 | [TK]D-Fender | Zarag: "Use the "t" dial option to allow you to transfer the call you place when you dial out. |
| 14:24.50 | [TK]D-Fender | dandate2: Then call their ISP hand have them shut down. |
| 14:25.45 | dandate2 | well i complain and complain, i have recordings, but everyone wants "documentation" |
| 14:26.01 | dandate2 | whatever happened to good ol vigilante justice |
| 14:26.47 | [TK]D-Fender | dandate2: A bunch of them got sued and losing that battle after winning the "vigilange street-justice" battle was a bitter pill. |
| 14:26.57 | [TK]D-Fender | dandate2: So grow up. |
| 14:27.11 | dandate2 | alright then |
| 14:34.47 | *** join/#asterisk QaDeS_ (n=mklaus@p4FC72773.dip0.t-ipconnect.de) |
| 14:46.33 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:46.38 | ppc | dandate2: I don't think they would care |
| 14:46.56 | dandate2 | bahahahahahahahahahahahahahahahahahahahahahhahahahahahahah |
| 14:47.17 | ppc | As long as you aren't screwing up the network, they probably have other things to do |
| 14:47.35 | dandate2 | i figured the voip provider would be making money so |
| 14:47.43 | dandate2 | probably wouldnt even file anything |
| 14:47.49 | ppc | Everyone is making money in that scenario except the scammers |
| 14:47.56 | ppc | and who is going to go out on a limb for them? |
| 14:48.19 | ppc | It's not even an attack really, they setup a phone number whoever calls it is their problem |
| 14:48.43 | dandate2 | yeah thats what i thought, but mabye comcast wil lsee a bunch of dialed #s on an 800, redialing every 3 seconds and think something is up |
| 14:49.07 | dandate2 | not even waiting for the hangup cuz u can get em quicker redialing than waiting a whole minute |
| 14:50.02 | ppc | I'm not sure but I think they might get money everytime you dial that number |
| 14:50.12 | dandate2 | who comcast? |
| 14:50.19 | ppc | i might be completely wrong on that, yeah |
| 14:50.26 | dandate2 | damn |
| 14:50.31 | ppc | I just read a story about something like that w/ google |
| 14:50.46 | dandate2 | i know the outbound trunk providers charge u to call toll free |
| 14:50.52 | dandate2 | but my comcast digital voice lines no right |
| 14:52.52 | ppc | http://www.themoneytimes.com/featured/20091101/google-voice-accidently-reveals-secret-stats-id-1089485.html |
| 14:53.31 | ppc | just an interesting read |
| 14:53.37 | path | is it any worth trying to develop a web softpone? |
| 14:53.46 | path | s/softpone/softphone |
| 14:53.51 | ppc | path: for what? |
| 14:53.52 | dandate2 | only if u develop a stable browser |
| 14:54.05 | path | good answer |
| 14:54.12 | TSM2 | why not just do it as a java applet? |
| 14:54.25 | TSM2 | flash would be cool |
| 14:54.49 | path | indeed |
| 14:54.55 | TSM2 | but flash prolly does not have enough access to the TCP stack |
| 14:55.00 | path | though I hate flash :D |
| 14:55.07 | TSM2 | java would be the best probability |
| 15:00.15 | *** join/#asterisk bbeattie (n=bbeattie@208.53.57.89) |
| 15:02.20 | darkdrgn2k | umm i though there have been softphones deleoped in java already |
| 15:02.41 | path | open source? |
| 15:02.42 | bbeattie | What's a name/reference I can google to read up on how to calling a number when a user uses a phone history of calls and selects a number? The phone isn't smart enough to prepend a 9+ only on those calls. The phone supports auto adding numbers to a call but if this is set in the phone, the phone no longer is able to call internal extensions as every number has a 9 at the beginning. |
| 15:02.45 | darkdrgn2k | yep |
| 15:02.50 | path | interesting |
| 15:02.59 | darkdrgn2k | im not sure if this is the one but http://sip-communicator.org/ |
| 15:03.17 | darkdrgn2k | i know i found one at one point.. i dont think its that one |
| 15:04.01 | darkdrgn2k | might have been.. but google around i know i seen them |
| 15:04.14 | path | thanks darkdrgn2k :9 |
| 15:04.17 | path | :) |
| 15:04.37 | darkdrgn2k | i know Nortel in its MCS package has a java softphone bult in.. works pritty well to |
| 15:05.00 | [TK]D-Fender | bbeattie: Change your dialplan so you don't need a prefix |
| 15:05.19 | [TK]D-Fender | bbeattie: Dialout prefixes are so 1980 |
| 15:06.01 | darkdrgn2k | lol@1980 refrence |
| 15:06.02 | dlynes | dandate2: cooll...none of the links on their website work, either |
| 15:06.07 | dlynes | dandate2: was that your handiwork? |
| 15:06.24 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 15:11.23 | dlynes | If you get 'Beginning cache-load run for flavor 'nocona'...', or something similar when you run the benchtest for asterisk fax, and it sits there and never returns, is it going to do it every single time? |
| 15:11.38 | dlynes | Or any other status message in benchtest, for that matter? |
| 15:13.19 | dlynes | I'm having troubles getting it to actually complete, no matter how many times I run it...it keeps stalling in random spots |
| 15:14.31 | dlynes | If it helps at all, it's happening on an AMD Opteron 246 |
| 15:28.19 | *** join/#asterisk coppice (n=chatzill@233.166.232.220.dyn.pacific.net.hk) |
| 15:42.33 | *** join/#asterisk Prodego (n=Prodego@wikipedia/Prodego) |
| 15:49.56 | *** join/#asterisk ming_zym (n=ming_zym@124.14.74.50) |
| 15:54.24 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:5972:4356:7ec7:5d34) |
| 15:56.33 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:01.35 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 16:02.11 | *** join/#asterisk puzzled (n=foobar@535335AA.cable.casema.nl) |
| 16:04.11 | *** join/#asterisk karuru (n=karuru@p549CB093.dip.t-dialin.net) |
| 16:05.08 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 16:06.50 | *** part/#asterisk karuru (n=karuru@p549CB093.dip.t-dialin.net) |
| 16:10.57 | *** join/#asterisk undertuga (n=undertug@88.210.79.132.rev.optimus.pt) |
| 16:28.06 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 16:29.02 | *** join/#asterisk gurra (n=gurra__@unaffiliated/gurra) |
| 16:43.49 | *** join/#asterisk ardnat (i=4cf5eef9@gateway/web/freenode/x-paswzaixzxgtjmmn) |
| 16:43.53 | ardnat | hey guys |
| 16:43.56 | ardnat | quick question |
| 16:44.03 | ardnat | can i use CDR(durration) |
| 16:44.09 | ardnat | to check the total durration |
| 16:44.14 | ardnat | of the account |
| 16:44.17 | ardnat | 's calls |
| 16:45.12 | ardnat | anyone ? |
| 16:46.15 | TSM2 | ardinat = Johnny_ |
| 16:46.25 | TSM2 | ardnat = Johnny_ |
| 16:46.43 | ardnat | yeah |
| 16:46.58 | ardnat | this is my freenode account |
| 16:47.05 | TSM2 | it was not a question |
| 16:47.07 | ardnat | i need to auth to get into astrisk |
| 16:47.12 | ardnat | *asterisk |
| 16:47.17 | TSM2 | yup true |
| 16:47.34 | ardnat | TSM is this corrent |
| 16:47.43 | ardnat | *correct |
| 16:47.56 | ardnat | CDR(durration)=total durration |
| 16:48.00 | ardnat | or just of that call |
| 16:50.17 | [TK]D-Fender | <PROTECTED> |
| 16:50.31 | [TK]D-Fender | 1 call does not look at another |
| 16:51.12 | ardnat | how could you get it to make the total calls then |
| 16:51.41 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 16:51.53 | ardnat | beause you couldnt do the CDR(durration) after the dial command |
| 16:52.07 | ardnat | beause when you disconect it wont excute |
| 16:52.09 | ardnat | right? |
| 16:52.28 | ardnat | unless the reciving party disconnects |
| 16:53.38 | [TK]D-Fender | ardnat: "total calls"? What "total calls"? This looks at the CURRENT CHANNEL ONLY |
| 16:54.09 | ardnat | I would like the sum of all the calls the current accunt has made |
| 16:54.10 | *** join/#asterisk brightontez (n=tez@notleb.plus.com) |
| 16:54.17 | ardnat | the billsec durration |
| 16:54.21 | brightontez | good evening |
| 16:54.24 | ardnat | is there a command to do this |
| 16:54.24 | [TK]D-Fender | ardnat: that involes looking at CDR history |
| 16:54.34 | [TK]D-Fender | ardnat: and is a completely external process you have to program yourself |
| 16:54.40 | ardnat | dang |
| 16:54.48 | [TK]D-Fender | ardnat: No, there is no command, you have to write an APPLICATION to do it |
| 16:54.54 | ardnat | is the CDR is xml? |
| 16:54.58 | ardnat | or cvs? |
| 16:55.03 | [TK]D-Fender | ardnat: No, CSV / DB |
| 16:55.15 | ardnat | ah ok |
| 16:55.31 | [TK]D-Fender | ~book |
| 16:55.32 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:55.32 | brightontez | is this the right place for a noob to ask questions...? or is there a better channel? |
| 16:55.36 | [TK]D-Fender | ^^ get reading... |
| 16:55.38 | ardnat | ah that sucks |
| 16:55.46 | [TK]D-Fender | briDepends on the questions |
| 16:56.00 | [TK]D-Fender | brightontez: Depends on the questions |
| 16:57.17 | ardnat | TK can you give me an example CDR CVS file as I am unable to acess my server currently |
| 16:57.22 | brightontez | OK. Just installed asterisk now, got tdm410 card, just want to find somewhere to 'begin to learn how to get it going' as simple as poss to begin. |
| 16:57.59 | brightontez | i'm linux experenced since early slack. asterisk virgin |
| 16:58.22 | ardnat | so you want a sort of tutorial bright? |
| 16:58.47 | ardnat | http://members.optusnet.com.au/~bsharif/asterisk/AsteriskForDumbMe.htm |
| 16:58.57 | brightontez | please ardanat |
| 16:59.08 | [TK]D-Fender | ardnat: its all documented in the book and in the source tarball. Get reading. |
| 16:59.28 | ardnat | gotcha |
| 16:59.47 | brightontez | um, didn't use the tarball. built a box and shoved the cd in and then thought um |
| 17:00.23 | *** join/#asterisk debuggerboy (n=debugger@117.196.162.4) |
| 17:00.49 | [TK]D-Fender | brightontez: ... |
| 17:00.50 | [TK]D-Fender | ~freepbx |
| 17:00.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:00.53 | [TK]D-Fender | ^^^^ |
| 17:01.11 | [TK]D-Fender | brightontez: GUI's are not supported in this channel |
| 17:01.32 | brightontez | ok, i'm not scared at firing up vi :) |
| 17:01.38 | [TK]D-Fender | briToo late <- |
| 17:01.49 | brightontez | mc is my friend :D |
| 17:01.50 | [TK]D-Fender | brightontez: You're running a distro that completely manages things |
| 17:03.44 | *** join/#asterisk QaDeS_ (n=mklaus@213.157.13.70) |
| 17:03.47 | brightontez | so, what's the best way forward? not scared to use a shell (prefer it) |
| 17:08.15 | [TK]D-Fender | brightontez: We support only from-scratch installs here or custom bits that fall outside of GUI's etc |
| 17:09.13 | brightontez | D-Fender: thanks. i'm keen to do a from scratch install. |
| 17:10.46 | brightontez | D-Fender: i'm not into gui stuff at the best of times. box will only be for home use. |
| 17:12.14 | TSM2 | when doing URI dialing, will * lookup the DNS SVR record of the domain you are calling? |
| 17:12.37 | brightontez | I'll have a quick look at the link ardnat kindly gave. |
| 17:18.05 | *** join/#asterisk p3nguin (i=BuhPjX1J@24-171-73-146.dhcp.mtvr.il.charter.com) |
| 17:21.56 | *** join/#asterisk robl^laptop (n=robl@c-98-197-98-39.hsd1.tx.comcast.net) |
| 17:24.15 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 17:30.44 | brightontez | OK, I'll install a fresh copy of Centos and start again. |
| 17:34.33 | *** join/#asterisk kazaa_lite (n=msaleem@cpc6-slam5-2-0-cust171.2-4.cable.virginmedia.com) |
| 17:34.52 | *** join/#asterisk endemic (n=endemic@orion.onvox.net) |
| 17:40.25 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 17:43.19 | *** join/#asterisk sizzers (n=resin000@75.119.125.63) |
| 17:43.33 | sizzers | good day everyone |
| 17:43.52 | *** join/#asterisk Malkor (n=marco@hlle-d9ba5d75.pool.mediaWays.net) |
| 17:43.59 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 17:45.13 | sizzers | I would like some help understanding the skype-for-asterisk logic and call flow before I decide to do anything with it. Does anyone have a link to the details, or personal experience with it yet? |
| 17:54.08 | sizzers | first, the software license schema which involves simultaneous channels, how are the keys managed? |
| 17:54.19 | lesouvage | Does Asterisk supports setting up a private gsm network in some way as an alternative for http://www.privatemobilenetworks.com/products/? |
| 18:00.14 | *** join/#asterisk neurosys (n=vinix@c-71-196-8-127.hsd1.fl.comcast.net) |
| 18:03.20 | sizzers | lesouvage, realize that the key to such a network would be the actual cellular tower emulation equipment for the cell phones to communicate with |
| 18:03.44 | sizzers | asterisk has no such functionality |
| 18:05.07 | *** join/#asterisk shinao1 (n=shinao1@41.219.193.30) |
| 18:07.47 | lesouvage | sizzers: I understand that it has to be a combination between software and hardware. It has not been allowed for a long time tos setup a private gsm network but in Holland it is since short time. It would be great if Digium would offer the hardware to set it up combined with the software support within Asterisk. |
| 18:08.05 | *** join/#asterisk QaDeS_ (n=mklaus@213.157.13.70) |
| 18:09.14 | sizzers | indeed, it's extremely cool technology |
| 18:09.36 | sizzers | however this would probably take digium at least 2 years to get a product to market |
| 18:09.43 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
| 18:09.56 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 18:11.12 | sizzers | furthermore, they'd be delving into a brand new industry, where there's already industrial strength and mature solutions available |
| 18:11.34 | sizzers | i don't speak for digium, but i don't see the business case for them to get involve |
| 18:15.38 | *** join/#asterisk Dabian (n=morten@fsf/member/dabian) |
| 18:16.18 | Dabian | I've battered with asterisk configuration for a day |
| 18:16.29 | Dabian | or so now. Is this the right channel to ask for help? |
| 18:17.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:18.15 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-67-207.link.net.pk) |
| 18:19.15 | riddlebox | yes |
| 18:19.37 | Dabian | Cool. |
| 18:19.46 | Dabian | I managed to set up my |
| 18:20.22 | Dabian | phone-thing .. and I can dial the local echotest in my extension context .. I can also dial asterisk from my M |
| 18:20.26 | *** join/#asterisk shinao1 (n=shinao1@41.219.193.30) |
| 18:20.30 | Dabian | Mobile phone. |
| 18:20.36 | Dabian | (celluar phone) |
| 18:20.53 | Dabian | However, calling out troubles me. |
| 18:21.15 | riddlebox | have you read the book? |
| 18:21.16 | Dabian | I have two providers, and apparently I can register alright |
| 18:21.23 | Dabian | the book? |
| 18:21.37 | [TK]D-Fender | ~book |
| 18:21.38 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:21.39 | Dabian | No, I don't think so |
| 18:22.08 | Dabian | I went by some site with a quickstart guide .. since my setup is simple for now. |
| 18:22.13 | dlynes | I guess the 32-bit version of benchfax and res_fax and res_fax_digium won't work on 64-bit architectures? |
| 18:22.14 | riddlebox | thanks [TK]D-Fender ] |
| 18:22.20 | Dabian | Also, the asterisk I use is quite old, I guess. |
| 18:23.02 | Dabian | Asterisk 1.2.7.1 |
| 18:23.18 | [TK]D-Fender | Dabian: No matter. Perhaps you should describe your problem in more detail than "calling out troubles me." |
| 18:23.32 | Dabian | Yes ... |
| 18:23.36 | riddlebox | Dabian, if you read that book it will explain how to dial out |
| 18:24.30 | Dabian | riddlebox : Right .. I kinda assume its just a few lines in sip.conf and extention.conf though. |
| 18:24.43 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 18:25.07 | debuggerboy | sangoma or digium the best? |
| 18:25.12 | Dabian | somethinhg like this: |
| 18:25.14 | [TK]D-Fender | debuggerboy: Yes |
| 18:25.15 | Dabian | exten => _X.,1,Dial(SIP/musimi.dk/${EXTEN},120) |
| 18:25.24 | Dabian | exten => _X.,1,Dial(SIP/musimi.dk/${EXTEN},120) |
| 18:25.36 | Dabian | (my provider is musimi) |
| 18:25.52 | riddlebox | Dabian, pretty much the book will help you through |
| 18:26.20 | [TK]D-Fender | Dabian: You need to set up a peer entry in sip.conf with the correct auth, codecs, host, etc. |
| 18:26.33 | Dabian | riddlebox : You might be right ... I've consulted a lot of sites now, including the asterisk.org .. but I seem to be pretty thick. |
| 18:26.36 | debuggerboy | every where on asterisk web its digium described. What about sangoma? |
| 18:26.55 | Dabian | [TK]D-Fender : I setup musimi.dk as a friend ... |
| 18:27.39 | debuggerboy | where can I find some installation instructions regarding to Sangoma A200D |
| 18:27.55 | [TK]D-Fender | Dabian: You should not use hostnames as a peer entry in sip.conf |
| 18:28.10 | [TK]D-Fender | debuggerboy: On Sangoma's support WIKI |
| 18:28.19 | TSM2 | yup sangoma have lots of docs |
| 18:28.31 | TSM2 | even their cfg_dahdi tool does all the work for you |
| 18:28.38 | Dabian | [TK]D-Fender : The dot will trick asterisk? |
| 18:28.43 | debuggerboy | TSM2: do you have any good URLs? |
| 18:28.58 | TSM2 | just goto wiki.sangoma.com |
| 18:29.39 | lesouvage | sizzers: the businesscase for setting up Private Mobile Network is the same as the businesscase for the other hardware. Offering high quality hardware that in combination with Asterisk offers a high quality flexible and open communication solution and a vihicle for offering paid services |
| 18:30.03 | *** join/#asterisk Chodorenko (n=chodoren@ext.one.by) |
| 18:30.28 | [TK]D-Fender | Dabian: Not a good idea. Simply don't do it. |
| 18:30.36 | *** join/#asterisk brightontez (n=tez@notleb.plus.com) |
| 18:31.08 | Dabian | hmm .. guess I do need to read the book ... |
| 18:31.26 | brightontez | what book? |
| 18:31.50 | [TK]D-Fender | ~book |
| 18:31.51 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:32.43 | lesouvage | sizzers: btw, there aren't that many providers of this kind of hardware, at least I haven't find that many on the internet. |
| 18:33.07 | debuggerboy | TSM2: thanks gud docs. |
| 18:33.28 | Dabian | [TK]D-Fender : I guess the hostpart of the register line, doesn't have to correspond to the entry for peer/friend in sip.conf? |
| 18:33.53 | Dabian | register line, and peer entry are not connected (as such) |
| 18:34.03 | [TK]D-Fender | Dabian: these 2 aspects are completely separate of each other |
| 18:34.15 | Dabian | ahh .. that explains a lot. :) |
| 18:34.30 | [TK]D-Fender | ~sipregister |
| 18:34.31 | infobot | [~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register. Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently. Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW) |
| 18:36.24 | Dabian | [TK]D-Fender : So basicly I just need a register-line for incomming calls, and a peer-section for outgoing calls. |
| 18:37.49 | Dabian | [TK]D-Fender : Thanks. That makes a lot of sense! :) |
| 18:37.55 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 18:38.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:38.49 | brightontez | d-fender: I've reinstalled asterisk as per http://www.asterisk.org/applications/pbx and the asterisk -vvvgci |
| 18:38.49 | brightontez | <PROTECTED> |
| 18:38.50 | Dabian | exten => _X.,1,Dial(SIP/musimi/${EXTEN},120) |
| 18:38.53 | Dabian | does thaexten => _X.,1,Dial(SIP/musimi/${EXTEN},120) |
| 18:39.15 | Dabian | does that look functional, assuming everything else is good? |
| 18:40.31 | brightontez | ~help |
| 18:40.34 | [TK]D-Fender | Dabian: Sure, as long as the # is in an acceptable format, and the rest of your sip.conf is correct |
| 18:41.31 | Dabian | [TK]D-Fender : The line will send all digits typed before the poundkey to the peer "musimi", right? |
| 18:42.50 | *** join/#asterisk ardor^ (n=IceChat7@ip72-193-201-128.lv.lv.cox.net) |
| 18:42.51 | [TK]D-Fender | Dabian: No, that will send whatever was dialed starting with a digit and at least one more char to your provider |
| 18:43.18 | ardor^ | Can i pass Veriables into a ${SHELL(echo ${VAR}) |
| 18:43.30 | ardor^ | } |
| 18:44.08 | ardor^ | exten => _XX,1,Set(r=${SHELL(if [ -f ${ICSNDDIR}${EXTEN} ]; then echo -n 1; else echo -n 0; fi)}) |
| 18:45.53 | brightontez | ~book |
| 18:45.54 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:46.39 | kaldemar | ardor^: of course. |
| 18:46.58 | ardor^ | pbx.c:2970 pbx_substitute_variables_helper_full: Error in extension logic (missing '}') |
| 18:47.05 | ardor^ | But I dont think I am missing a '}' |
| 18:49.10 | ardor^ | Pretty werid, Dont you think Kaldemar |
| 18:49.27 | Dabian | [TK]D-Fender : How can I test if my PEER entry is functional? |
| 18:49.40 | [TK]D-Fender | Dabian: USE IT |
| 18:49.59 | kaldemar | ardor^: ; might be the source of your problem |
| 18:50.02 | Dabian | [TK]D-Fender : Can I use it from the console? |
| 18:50.15 | [TK]D-Fender | Dabian: use a softphone or some other device |
| 18:50.30 | ardor^ | Kaldemar what does a ; have to do with anything? (dont understand) |
| 18:50.36 | Dabian | [TK]D-Fender I have a Sipura2k connected to my asterisk |
| 18:50.45 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 18:51.10 | ardor^ | oh! |
| 18:51.21 | kaldemar | ardor^: the extension you posted has them. comment characters. |
| 18:51.22 | ardor^ | you think i have ; in my shell string and Asterisk is thinking they are commints |
| 18:51.25 | ardor^ | that make since. |
| 18:51.47 | ardor^ | Well, How do i send ';' in my shell statments. |
| 18:51.48 | kaldemar | escape them with \ |
| 18:51.52 | ardor^ | Thanks |
| 18:53.01 | Dabian | [TK]D-Fender : I can use the other extensions I have in my context, but when I try for instance an 8-digit number, I just get a busy tone. |
| 18:53.01 | *** join/#asterisk dandate2 (n=mangy@112.202.196.127) |
| 18:53.26 | ardor^ | Thanks Kaldemar that worked it says my shell statment is broken now so let me look into that. |
| 18:53.33 | dandate2 | any kayako tech support avail? my ticket is critical my pbx got hacked and they turned db authentication in my.cfg so i cannot access http gui |
| 18:53.36 | [TK]D-Fender | Dabian: PASTEBIN is your friend <- |
| 18:53.36 | Dabian | [TK]D-Fender : I am not sure how to debug that problem |
| 18:53.38 | [TK]D-Fender | ~pb |
| 18:53.39 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:54.42 | ardor^ | if [ -f /usr/share/asterisk/ic/snds/55 ]; then echo -n 1; else echo -n 0; fi |
| 18:54.42 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 18:54.47 | ardor^ | works fine in my bash shell |
| 18:55.02 | ardor^ | also works in sh |
| 18:55.14 | ardor^ | sh: Syntax error: end of file unexpected (expecting "then") |
| 18:56.28 | [TK]D-Fender | ardor^: "core show function STAT" |
| 18:57.04 | ardor^ | Thats a good work around. |
| 18:57.14 | ardor^ | I could also use a shellscript i guess. |
| 18:57.51 | [TK]D-Fender | ardor^: Workaround? more like the direct way, rather than reinventing the wheel |
| 18:58.15 | ardor^ | Yes, but not all wheels are invented. |
| 18:58.29 | ardor^ | I need to know how to make them |
| 18:58.53 | ardor^ | but I will use stat |
| 18:59.11 | ardor^ | after i slove this, (which is silly but I;ll learn 2 things.) |
| 19:03.24 | brightontez | I just started reading the asterisk book and there is no reference to dahdi... |
| 19:03.41 | [TK]D-Fender | ~dahdi |
| 19:03.42 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 19:03.49 | ardor^ | D-Fender: I gave up, I am using stat now, Thanks |
| 19:04.05 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 19:04.14 | ardor^ | D-Fender if i have to do multiline Shell Commands, I will use a simple script. |
| 19:04.32 | *** join/#asterisk lozarythmic (n=lpraties@e1-1.ns500-1.ts.milt.as9105.net) |
| 19:04.37 | brightontez | error 404 |
| 19:05.08 | Dabian | Hmm .. apparently it calls the number at my provider correctly, but then tries to transfer the call to same extension as I called locally? |
| 19:05.49 | Dabian | I have the feeling I should get rid of the "clutter" in my config files. :D |
| 19:07.45 | brightontez | who do I tell http://www.asterisk.org/zaptel-to-dahdi gives error 404 to? tnx |
| 19:09.45 | ChannelZ | Maybe the big giant link at the bottom that says "Feedback or Report Broken Links" ? |
| 19:11.01 | brightontez | sorry if I type ~dahdi the bot gives the broken link |
| 19:11.26 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 19:12.22 | Dabian | [TK]D-Fender : Hmmm .. the "domain=xxxxxx.com" .. I shouldn't set that to the FQDN of the asterisk server, but rather to the domain of my provider? |
| 19:12.49 | Dabian | (In sip.conf) |
| 19:13.04 | [TK]D-Fender | Dabian: Probably shouldn't touch |
| 19:13.23 | [TK]D-Fender | briIts just the new name. its configs are a tiny bit different, but thats all. |
| 19:13.32 | [TK]D-Fender | brightontez: Read the docs from its tarball |
| 19:14.32 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 19:17.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:21.00 | sizzers | Dabian, when trying to learn the basics, it's a good idea to get rid of clutter in sip.conf , however, save a copy of the original first (always best practices) |
| 19:21.10 | Dabian | Right |
| 19:21.13 | Dabian | Thanks |
| 19:21.36 | sizzers | also, the domain parameter is probably not relevant for your situation |
| 19:22.15 | sizzers | Wow d-fender, i haven't been on IRC in probably a year and a half , it's awesome to see that you're still the man around here |
| 19:25.38 | ChannelZ | !dahdi |
| 19:25.40 | ChannelZ | ero |
| 19:25.45 | ChannelZ | ~dahdi |
| 19:25.45 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 19:25.49 | ChannelZ | damnation I can't type today |
| 19:26.01 | brightontez | i know the feeloing |
| 19:27.02 | brightontez | D-Fender: I guess the docs are also in RPMs. |
| 19:27.42 | Dabian | [TK]D-Fender : Thanks! Now its working! |
| 19:29.31 | [TK]D-Fender | Dabian: Glad to hear |
| 19:29.57 | [TK]D-Fender | brightontez: Most will recommend installing from suorce |
| 19:34.41 | *** join/#asterisk niekie (i=quasselc@dreamworld.bergnetworks.com) |
| 19:41.43 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) |
| 19:42.00 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 19:46.47 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 19:46.47 | *** mode/#asterisk [+o ChanServ] by irc.freenode.net |
| 19:46.51 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 19:46.51 | *** mode/#asterisk [+o ChanServ] by irc.freenode.net |
| 19:47.30 | Fister | wow |
| 19:48.53 | sizzers | Dfender, you still around? |
| 19:51.00 | Dabian | hmmm ... aparently I can only call numbers local to my provider .. I guess thats an authorisation-problem. |
| 19:51.17 | [TK]D-Fender | yes |
| 19:51.24 | [TK]D-Fender | (here) |
| 19:51.24 | dandate2 | omg kayko support cannot help me |
| 19:52.01 | sizzers | So, i know it's new and it's probably hated among the people in this channel, but have you looked at the Skype for Asterisk documentation at all? |
| 19:52.39 | dandate2 | if someone has msql auth pass can they see what root pass is if i cahnge it? |
| 19:52.52 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 19:52.52 | *** mode/#asterisk [+o ChanServ] by irc.freenode.net |
| 19:53.07 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 19:53.08 | *** mode/#asterisk [+o file] by ChanServ |
| 19:53.12 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 19:53.12 | *** join/#asterisk endemic (n=endemic@209.20.75.129) |
| 19:53.13 | *** join/#asterisk Chinorro (n=Chino@91.117.226.19) |
| 19:53.14 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) |
| 19:53.17 | *** join/#asterisk cnu (n=cnu@63.80-203-44.nextgentel.com) |
| 19:54.43 | *** join/#asterisk brezular (n=brezular@adsl-dyn108.95-103-107.t-com.sk) |
| 19:56.50 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 19:57.08 | *** join/#asterisk drewster213 (n=ahanrath@ip72-213-25-243.om.om.cox.net) |
| 19:57.31 | [TK]D-Fender | sizzers: No, but I'm due to, before long... |
| 19:57.44 | sizzers | very interesting |
| 19:57.54 | [TK]D-Fender | dandate2: depends if they gave themselves rights to that table |
| 19:58.00 | sizzers | the manual is the single most well written documentation i've ever read |
| 19:59.59 | dandate2 | the hackers are blocking the ip of the kayako tech support but i have access still cuz i use dhcp |
| 20:00.02 | dandate2 | anything i can do to stop that? |
| 20:00.30 | *** join/#asterisk dkirker (n=dkirker@gateway0.openmobl.com) |
| 20:01.06 | [TK]D-Fender | dandate2: Kick them off the machine and firewallt them |
| 20:01.16 | dandate2 | how? |
| 20:01.30 | dandate2 | i have the ip that is fraudulent i just dont know how heh |
| 20:02.33 | *** join/#asterisk _bugz_ (n=bugz@99.129.28.165) |
| 20:03.07 | [TK]D-Fender | dandate2: "man iptables" |
| 20:04.02 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 20:05.39 | dandate2 | i am afraid that kayako.freepbx might just be a scam to drain you for hours of tech support, how could they not do anything.. |
| 20:07.22 | dandate2 | now they are saying there will be an additional 30% charge heh |
| 20:09.05 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 20:12.33 | *** join/#asterisk wierdo (n=chatzill@77.78.3.197) |
| 20:16.19 | *** join/#asterisk dbrotman (n=nonet@c-98-236-3-118.hsd1.pa.comcast.net) |
| 20:16.53 | dbrotman | Hi, I'm looking for recommendations for a call manager to use on a trixbox CE 2.8 install. Suggestions? |
| 20:24.43 | dandate2 | <PROTECTED> |
| 20:25.32 | [TK]D-Fender | dandate2: Check your firewall |
| 20:26.33 | dandate2 | i think he just got hit by fail2ban |
| 20:33.55 | *** join/#asterisk dandate3 (n=mangy@112.202.196.127) |
| 20:34.03 | dandate3 | doh got disconnected on a packetloss |
| 20:34.14 | dandate3 | now umm, i usually check my firewall by typing config |
| 20:34.27 | dandate3 | but that option has seemed to change, it now just lets me change firewall or keyboard table |
| 20:34.43 | dandate3 | errr not firewall but timezone |
| 20:34.50 | dandate3 | sorry i am most stressed in my life |
| 20:38.39 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 20:43.29 | *** join/#asterisk wierdo (n=wierdo@77.78.3.197) |
| 20:44.07 | *** join/#asterisk robl^laptop (n=robl@c-98-197-98-39.hsd1.tx.comcast.net) |
| 20:44.49 | *** part/#asterisk wierdo (n=wierdo@77.78.3.197) |
| 20:45.55 | *** join/#asterisk aclarkobn (i=4244aaaf@gateway/web/freenode/x-debac1d81595c600) |
| 20:47.53 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:49.28 | aclarkobn | Hello, Does anyone know how to block and unblock skype buddys with SFA? |
| 20:56.20 | *** join/#asterisk wierdo (n=chatzill@77.78.3.197) |
| 20:59.02 | aclarkobn | Hello, anyone know how to unblock skype buddys on skype for asterisk? |
| 20:59.39 | TSM2 | whats this for, the new digium plugin? |
| 20:59.48 | TSM2 | mabey its in astdb, just a guess though |
| 21:01.04 | *** join/#asterisk saftsack (n=oliver@p579DDA60.dip.t-dialin.net) |
| 21:02.04 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 21:02.37 | *** join/#asterisk gurra (n=gurra__@unaffiliated/gurra) |
| 21:03.38 | aclarkobn | yes Skype for Asterisk is the new digium plugin, I have looked over every since peace of documentation about SFA and it doesnt mention how to unblock users on that were blocked on the buddy list. |
| 21:04.02 | aclarkobn | I will take a look at astdb |
| 21:05.58 | *** join/#asterisk saftsack (n=oliver@p579DDA60.dip.t-dialin.net) |
| 21:08.54 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 21:09.08 | *** join/#asterisk rdude99 (n=infamis@adsl-76-192-184-243.dsl.chcgil.sbcglobal.net) |
| 21:11.01 | rdude99 | I have hardware that only works on windows(usb dongle, x-lite) and then digium hardware that only works on linux...can I actually creat an asterisk solution with one computer? |
| 21:12.11 | rdude99 | with windows as host & linux as guest os in vmware, I can't get the digium card to be seen. |
| 21:12.26 | rdude99 | with linux as host and windows as guest os in vmware, I can't get bluetooth stuff working |
| 21:13.09 | rdude99 | most pratical thing for me to do is to get asterisk running as guest in linux with a windows host |
| 21:13.11 | rdude99 | but how? |
| 21:14.14 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 21:14.22 | [TK]D-Fender | rdude99: Get an external SIP gateway |
| 21:15.21 | rdude99 | TK D-Fender, what will that enable me to do, and how? |
| 21:16.01 | [TK]D-Fender | rdude99: You asked about Digium's cards. I gave you an alternative. |
| 21:16.03 | *** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net) |
| 21:16.36 | rdude99 | is a SIP gateway connect to a telephone line? |
| 21:17.12 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 21:17.58 | [TK]D-Fender | rdude99: If thats the kind you buy, yes |
| 21:19.26 | rdude99 | so would that mean I would run asterisk under windows? |
| 21:20.41 | [TK]D-Fender | rdude99: No, that means you won't need PCI support for your PSTN interfaces. |
| 21:24.10 | rdude99 | I'm looking on eBay and I see those listed for >$100, so I can't use that. This is just a school project. We used asteriskwin32 for VoIP under windows & everything worked perfectly with the rest of the software we built, however, we didn't have telephone access. Then we bought an X100P card but learned we had to use it under linux. So we installed vmware on our windows box with linux as the guest os, however couldn't |
| 21:24.12 | rdude99 | access the card. Is there another alternative that doesn't require purchasing additional hardware? |
| 21:25.45 | [TK]D-Fender | rdude99: Dual boot and run Linux by itself for your learning |
| 21:26.18 | [TK]D-Fender | rdude99: And asteriskwin32 is not supported here. |
| 21:27.53 | *** join/#asterisk p3nguin (i=BuhPjX1J@24-171-73-146.dhcp.mtvr.il.charter.com) |
| 21:28.49 | rdude99 | I understand about asteriskwin32. But dual booting will only allow one active OS at a time, unless I'm not understanding what you mean. We need to be able to run a linux box(for asterisk&telehone access) & a windows box(for everything else) at the same time (and we're using tcp/ip for talking between the two). I'm just wondering if any virtualization solution can allow this without any extra hardware. |
| 21:29.16 | p3nguin | rdude99: That's what virtualization does. |
| 21:29.57 | p3nguin | rdude99: You have a host OS with guests (virtual OSs) inside. |
| 21:29.58 | [TK]D-Fender | rdude99: And what do you need Windows for precisely? |
| 21:30.09 | rdude99 | p3nguin: true, but which one can actually do it so that the guest OS can "see" my telephone fxo card |
| 21:30.39 | rdude99 | TK D-Fender: because all of our software is written in windows; we're doing bluetooth & audio processing on the windows side |
| 21:30.48 | p3nguin | rdude99: Depending on your overall requirements, you could install a Linux-based OS as the host and run Windows in the virtual machine. |
| 21:30.50 | [TK]D-Fender | rdude99: You seem to think that Linux should be the "guest". When looking for hardware support you seem to have a broken sense of priorities |
| 21:31.35 | rdude99 | p3nguin: that's what I'm hoping for, but have failed in making it work correctly |
| 21:31.40 | [TK]D-Fender | rdude99: And the price you found is not commensurate to the need. |
| 21:31.42 | rdude99 | TK: the problem is bluetooth only works on windows |
| 21:31.59 | rdude99 | ...not technically |
| 21:32.03 | rdude99 | but for our purposes |
| 21:32.10 | [TK]D-Fender | rdude99: http://www.telephonydepot.com/Catalog/FXS-FXO-Analog-Adapters/Linksys-SPA3102 |
| 21:32.22 | p3nguin | rdude99: If you are relying on both operating systems to use hardware, you might be better off setting up a second computer to run the PBX. |
| 21:33.44 | p3nguin | rdude99: I don't know what your call load will be, but Asterisk can run on some pretty low-spec'ed hardware. |
| 21:33.57 | rdude99 | yes, that seems the last resort. however that will be expensive (since this is a school project) |
| 21:34.14 | p3nguin | rdude99: I run mine on a PIII 933 MHz/512 MB computer. |
| 21:34.55 | rdude99 | The really limiting factor to having to use windows (for hardware purposes) is that we were unsuccessful in making linux utilize a bluetooth headset as a normal headphone/mic but were successful under linux. |
| 21:35.09 | rdude99 | sorry, we were successful under windows |
| 21:35.10 | [TK]D-Fender | rdude99: Change your approach to virtualization or buy hardware. Your choice |
| 21:36.09 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 21:37.03 | *** join/#asterisk kfife (n=Miranda@home.chicagoventure.com) |
| 21:37.15 | rdude99 | if we can make bluetooth work how we want on linux, we have no reason to force windows. windows as guest OS would work because everything else is just serial which VMWare can patch through. |
| 21:37.18 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 21:37.50 | rdude99 | but I was just seeing if there was a tiny possibility that a software-based alternative would work; guess not. |
| 21:38.12 | [TK]D-Fender | rdude99: And you seem to have issues running Linux as the host OS. |
| 21:38.41 | p3nguin | rdude99: VirtualBox should have relatively good USB support, so you might be able to use Windows as a guest with bluetooth on USB successfully. |
| 21:39.28 | *** join/#asterisk kfife (n=Miranda@kfife.com) |
| 21:39.51 | rdude99 | [TK]D-Fender: I don't have any issues with linux as the host...just that then the functionality provided by the bluetooth headset would only work on windows (which would be guest OS, but then arises the hardware patch-through problem of virtualization). |
| 21:40.12 | p3nguin | See above. |
| 21:40.22 | rdude99 | p3nguin: hmm, never heard of VirtualBox, will look into it. |
| 21:41.11 | p3nguin | As a project, and not necessarily production server, vbox might get the job done. |
| 21:41.16 | rdude99 | I don't wanna get too OT but VirtualBox runs under linux,windows, or both? |
| 21:41.29 | p3nguin | either |
| 21:41.49 | kfife | Urgent: I have a toll-fraud issue to debug on a legacy system RIGHT NOW. Asterisk is being used to record the DTMF sequences, but it is 'absorbing' them so they are unintelligible. Is there a way to have the tones recorded into the mixmonitor audio stream? Thanks! |
| 21:42.18 | p3nguin | Right on the front page: "Presently, VirtualBox runs on Windows, Linux, Macintosh and OpenSolaris hosts and supports a large number of guest operating systems including but not limited to Windows (NT 4.0, 2000, XP, Server 2003, Vista, Windows 7), DOS/Windows 3.x, Linux (2.4 and 2.6), Solaris and OpenSolaris, and OpenBSD." |
| 21:42.19 | rdude99 | p3nguin, [TK]D-Fender, thanks for the info. |
| 21:42.59 | rdude99 | will possibly come back after doing some research of my own. |
| 21:43.39 | kfife | Toll Fraud: It's locked down now, but trying to understand the exploit. Dirty bastard is still making attempts--trying to capture them. Calling mobile phones in Somalia! |
| 21:44.58 | kfife | Dirty fu¢ker$ |
| 21:46.41 | p3nguin | Under what circumstances could I get a Cisco 7940 to ring while I'm already on a call? Right now, call waiting beeps in, but I'm wanting to hear it ring (maybe not a full-volume ring) rather than the call waiting beep. |
| 21:47.18 | ChannelZ | kfife: you have an open SIP user or something? |
| 21:48.49 | saftsack | is it possible to tunnel an asterisk call through ssh? |
| 21:49.26 | kfife | ChannelZ: I don't understand your question. |
| 21:50.22 | TJNII | saftsack: Should be, if you really wanted to. Use port forwarding. |
| 21:50.32 | ChannelZ | kfife: how are these people making calls through your system? Are they coming in VoIP, or you have some badly-configured DISA setup that they are dialing in on and able to dial out of your system...? |
| 21:52.17 | kfife | It's all analog baby! Connected to a legacy Nortel system. I'm temporarily passing the calls through Asterisk to 'sniff' the attacks on a legacy system. It's a mailbox forward exploit, It's now locked down but I'm trying to understand the exact methodology, and level of sophistication. For example, have they guessed the admin pass? |
| 21:53.16 | kfife | They apparently reconfigure a mailbox somehow to forward off-site, then transfer themseles to it, and by doing so, call for free. |
| 21:53.17 | ChannelZ | so how is the * involved? |
| 21:53.31 | kfife | ChannelZ: I'm temporarily passing the calls through Asterisk to 'sniff' the attacks on a legacy system. |
| 21:53.41 | kfife | Otherwise, not at all |
| 21:53.46 | ChannelZ | via what, more analog? TDM card? |
| 21:54.29 | kfife | Redirect to a PRI, PRI loops back to antoher trunk on the same compromised system. |
| 21:54.36 | kfife | I can record all but the DTMF |
| 21:54.45 | kfife | it gets 'snubbed' as it gets absorbed |
| 21:54.58 | kfife | I want Mixmonitor to lay the audio down into the file. |
| 21:55.07 | kfife | ...so I can debug it |
| 21:55.27 | saftsack | TJNII, ok. so if i just want to register to the asterisk i just need to forward udp 5060, or? |
| 21:56.00 | [TK]D-Fender | ~sipnat |
| 21:56.01 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:56.03 | [TK]D-Fender | saftsack: ^^^ |
| 21:56.03 | thomas | saftsack? mussn deutscher sein.. *kopf schüttel* |
| 21:56.04 | TJNII | saftsack: SIP signalling is on 5060. Voice data is on a whole other block of ports (10000-20000 default, IIRC) |
| 21:56.26 | TJNII | saftsack: Follow [TK]D-Fender's link. The sipnat info will detail all the ports. |
| 21:56.41 | ChannelZ | kfife: I forget what it's called but there's a separate DAHDI tool that will let you record channels - outside of asterisk |
| 21:57.01 | *** join/#asterisk QaDeS_ (n=mklaus@p4FC72773.dip0.t-ipconnect.de) |
| 21:57.02 | kfife | user dials A - Redirected to a TN on our PRI, turn on mixmonitor, call another trunk on same system as A |
| 21:57.42 | saftsack | TJNII, yes thats true but i just want to see if signalling works. thomas japp ;) |
| 21:58.05 | kfife | ChannelZ: good info. Is this an asterisk applicaiton that interacts with Dahdi at the lower level or more like a chan_dahdi parameter? |
| 21:58.27 | ChannelZ | no it's a separate commandline utility |
| 21:58.42 | ChannelZ | I was trying to look but now my box at work is just locking up when I try to ssh in |
| 21:59.26 | *** join/#asterisk werdan7_ (n=w7@freenode/staff/wikimedia.werdan7) |
| 22:00.03 | ChannelZ | under zaptel it was called ztmonitor |
| 22:00.35 | ChannelZ | I think |
| 22:00.51 | saftsack | TJNII, my question is now why some phones register over port 2051. e.g. snom phones are using this port. but that's strange because 5060 is standard. |
| 22:01.47 | kfife | ChannelZ: BRB |
| 22:02.37 | TJNII | saftsack: The port is connects to on your server should be 5060. Otherwise it isn't sip. |
| 22:03.30 | TJNII | I don't remember the other ports, save IAX. |
| 22:03.47 | saftsack | http://www.trixbox.org/forums/trixbox-forums/trixbox-endpoints/snom-phones-port-2051 for example |
| 22:03.51 | ChannelZ | kfife: dahdi_monitor |
| 22:04.10 | ChannelZ | IAX is 4569 |
| 22:04.36 | TJNII | saftsack: That is the port on the CLIENT |
| 22:04.40 | TJNII | not the SERVER |
| 22:06.38 | kfife | ChannelZ: Thanks! Trying this now. |
| 22:09.02 | saftsack | ah ok |
| 22:09.38 | *** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net) |
| 22:10.10 | ChannelZ | damn it all |
| 22:10.16 | TJNII | saftsack: My understanding is the client opens a sip port as well. The server will connect to that port so signal incoming calls. It doesn't need to be on 5060, and sometimes can't be if there are other SIP devices on the same NAT. |
| 22:11.48 | TJNII | So actually, now that I really think about it, tunneling over ssh is more difficult that I originally thought. |
| 22:12.34 | saftsack | tried it before with 5060 didnt worked. that was the reason why i asked |
| 22:13.38 | *** join/#asterisk fluidicspace (n=AMP@c-69-181-42-133.hsd1.ca.comcast.net) |
| 22:17.02 | TJNII | saftsack: You would probably have better luck running the SIP/RTP data over a VTUN tunnel and running that through SSH. |
| 22:17.58 | TJNII | If nothing else it is something to try. |
| 22:31.25 | *** join/#asterisk shinao1 (n=shinao1@41.219.232.213) |
| 22:32.19 | manxpower | each UDP packet has a SOURCE port and a DESTINATION port. The SOURCE port does NOT (usually) matter. |
| 22:32.43 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 22:33.26 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:36.28 | *** join/#asterisk saftsack (n=oliver@p579DDEF1.dip.t-dialin.net) |
| 22:42.59 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 22:43.00 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:52.49 | _ShrikE | kfife: Very interesting situation. Have you determined the exact methodology? |
| 22:58.39 | kfife | _ShrikE: Yes. They log in and somehow (to be determined) grant themselves access to mailbox creation (normally not availble from off-site). The mailbox has a forwarding property which they populate with the desired TN. Then within the same call after 1 programming the mailbox, and 2. enabling the forward, the go BACK to the main menu and transfer themselves to said mailbox. PLS HOLD - as they get transferred. In our case they hear a reorder. |
| 22:58.49 | kfife | .. because we're now locked down. |
| 22:59.12 | kfife | A crusty old Nortel guy just spoke to me and said "Yeah, this is the second |
| 22:59.12 | kfife | case I've heard of one of these old NAM systems getting hacked recently--it |
| 22:59.12 | kfife | appears the newer CallPilots aren't vulnerable". |
| 22:59.47 | kfife | I'm still trying determine the precise nature of the vulnerability--whether a bona-fide hole, or just an unfortunate 'default' or an actual brute-force attack on an admin password. |
| 23:04.01 | TSM2 | kfife: which system?? the BCM stuff orthe MICS or CICS? |
| 23:05.45 | kfife | This is a MICS 4.1 with NAM - about ONE WEEK From tear-out. I kid you not!! |
| 23:06.04 | kfife | Isn't that the most ironic thing in the entire world? |
| 23:06.20 | kfife | Thing's been untouched for nearly a decade. Now a week from deprication it gets hacked! |
| 23:06.34 | TSM2 | kfife: i used to deal with thoes systems in the past, normaly dead reliable |
| 23:06.48 | TSM2 | you know what the NAM acutaly is? |
| 23:07.01 | kfife | Not without looking. Off site at moment. |
| 23:07.10 | kfife | I'm more than happy to share that with you. |
| 23:07.12 | TSM2 | OS/2 box |
| 23:07.23 | kfife | I believe so. It's the large pc-looking thing. |
| 23:07.30 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 23:07.32 | kfife | with a FDD |
| 23:07.36 | kfife | under the panel |
| 23:07.42 | TSM2 | well it looks like another main unit but has fiber connection between the two |
| 23:07.55 | kfife | Correct. Ours is a six-port fiber nam. |
| 23:07.56 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 23:08.20 | TSM2 | ours was just a single fiber, cant remember how many calls it owuld do with that |
| 23:08.34 | TSM2 | it has IP connection, they prolly found a way to it through your network |
| 23:08.37 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 23:08.37 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:08.51 | TSM2 | from what i remember it did not have a way to create mailboxes from the phones |
| 23:09.00 | TSM2 | only the normal CallPilots did |
| 23:09.12 | kfife | I've actually been really fond of setting up asterisk BETWEEN these old systems an the telco. You can create just about any feature--except wideband. |
| 23:09.29 | TSM2 | depending on the software rev |
| 23:09.35 | kfife | Use ADA for click-to-dial on an old Norstar system. Prety trick! |
| 23:09.59 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 23:10.03 | TSM2 | ADA cool, but thats piping via an * box? |
| 23:10.18 | kfife | COrrect 2-span PRI card - B2BUA |
| 23:10.27 | kfife | one span Telco, other span CPE |
| 23:10.40 | kfife | Nice thing is: Asterisk craps out? Just bypass. |
| 23:10.55 | kfife | Nice thiing is: NORTEL craps out: insert alternate dialing plan! |
| 23:11.12 | TSM2 | still rid of that NAM its some old shit |
| 23:11.12 | kfife | nice thing is: TELCO CIRCUIT craps out? Redirec via alternate routing plan to DID's at ITSP! |
| 23:11.48 | kfife | Yes, the migration is to a 7.1 with CallPilot using PRI rather than DID trunks. |
| 23:12.21 | TSM2 | 7.1, BCM unit? |
| 23:12.29 | kfife | 7.1 NORSTAR MICS |
| 23:12.31 | TSM2 | ive been out of nortels for the last 3yrs |
| 23:12.33 | kfife | I don't really like BCM |
| 23:12.45 | TSM2 | BCM is just a MICS in disguise |
| 23:13.10 | kfife | mmm sort of. Spinning HDD, running WinNT4 |
| 23:13.14 | TSM2 | even when the mobo died the phones just kept going |
| 23:13.29 | TSM2 | all the callprocessing was done on the main PCI card |
| 23:13.33 | kfife | I see - so sounds like some sort of hybrid |
| 23:13.36 | TSM2 | the Modules then connect to that card |
| 23:14.05 | kfife | So it's like a norstar brand software B2BUA all rolled into one box. |
| 23:14.06 | TSM2 | i had an early model with mobos that suffered from the bad cap issues |
| 23:14.34 | kfife | I like the "old iron" solid-state stuff. |
| 23:14.46 | kfife | especially when its functinality can be augmented with Asterisk. |
| 23:14.51 | TSM2 | yup, but that market is dying |
| 23:15.01 | kfife | you are right. |
| 23:15.13 | TSM2 | even nortel have moved to Linux |
| 23:15.18 | TSM2 | or had moved |
| 23:15.25 | TSM2 | now they are LG |
| 23:15.32 | kfife | I think it will die more slowly than people think. Some peopl just need phones. |
| 23:16.08 | TSM2 | Yup, but thoes all in one boxes the size of routers that can do small offices of 10, good value for money compared to old PBXs |
| 23:16.39 | TSM2 | i think we will see many of the main telcos starting to offer VoIP solutions but have QoS backed DSL lines for it |
| 23:16.45 | TSM2 | guaranteed serivce |
| 23:16.49 | kfife | Absolutely. NObody is going to install one today, but I think the attrition rate will stun the 'experts' |
| 23:17.08 | kfife | Even AT&T. It's a great idea. |
| 23:17.28 | kfife | PRI is amazing, but what the hell kind of product only comes in Crates of 23 channels?? |
| 23:17.37 | TSM2 | 32 in the UK :) |
| 23:17.44 | kfife | Imagine a coffee shop that only sold coffe by the 44 gallon barrel? |
| 23:17.57 | kfife | That's why I love SIP - you can get PRI without buying 23 channels |
| 23:18.11 | kfife | ...not to mention other codecs |
| 23:18.18 | kfife | ...wideband |
| 23:18.30 | TSM2 | asterisk still has to offer full support for wideband |
| 23:18.36 | TSM2 | ive found a couple of issues |
| 23:18.47 | TSM2 | especialy G722.1 |
| 23:19.19 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 23:19.26 | kfife | However I hate the latency of VoIP |
| 23:19.52 | TSM2 | this is true, but usualy only an issue if you are within hearing distance of the other person |
| 23:20.21 | TSM2 | oh it happens |
| 23:20.49 | TSM2 | we had one person test the new system in the office, she called her cell from the office phone, ear to ear and complained it had delay |
| 23:20.50 | TSM2 | grrr |
| 23:20.54 | TSM2 | duh |
| 23:21.55 | kfife | brb |
| 23:27.06 | TSM2 | kfife: do you have any info on the RAD unit, in the UK BT always installed one but never used it or told anyone how to use it |
| 23:31.59 | *** join/#asterisk wireddd (n=wired@unaffiliated/wireddd) |
| 23:32.40 | wireddd | can you use the magicjack usb dongle as an ata device for asterisk? |
| 23:34.40 | wireddd | I don't know if ata is the right term, I am thinking of signing up with a different sip provider, and just using the magicjack hardware |
| 23:37.15 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 23:45.09 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 23:48.02 | p3nguin | wireddd: For the price of the MagicJack, you can buy an actual ATA on ebay. |
| 23:49.31 | wireddd | well I already have one, and I hate the software/service, and I don't want to use the sip credentials from magicjack since they would probably just cancel my account.... |
| 23:53.07 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
