| 00:00.04 | kramba | ah ok |
| 00:00.30 | kramba | will have to try back monday then I guess |
| 00:00.55 | kramba | been searching in google for a couple hours now not finding an answer to my problem |
| 00:01.16 | rob0 | It's often busy after hours (N.American). |
| 00:01.33 | rob0 | ~docs |
| 00:01.34 | jbot | i heard docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 00:03.08 | *** part/#asterisk angom_w (n=angom@red-corp-200.79.158.218.telnor.net) |
| 00:03.20 | Un1x | ~extensions |
| 00:03.23 | Un1x | ~extensions.conf |
| 00:03.25 | jbot | [extensions.conf] at http://voip-info.org/wiki-Asterisk+config+extensions.conf, or know as dialplan, or known as extensions, or known as exten |
| 00:03.25 | Un1x | :P |
| 00:03.36 | Un1x | ~iax.conf |
| 00:03.39 | Un1x | ~iax2.conf |
| 00:03.41 | Un1x | :S |
| 00:06.21 | toerkeium | ~ami |
| 00:07.06 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 00:07.07 | Kumba_ | or known as a pain in the ass... |
| 00:07.14 | Kumba_ | atleast, that's how it's known to me :) |
| 00:08.13 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:08.19 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:12.35 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:16.54 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 00:17.04 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 00:17.20 | Mattwj2005 | hey guys....everyone have freeworlddialup running using iax? |
| 00:17.55 | Mattwj2005 | if so is it working? |
| 00:18.02 | Mattwj2005 | *is it working |
| 00:18.18 | Qwell[] | I doubt *everyone* has it working |
| 00:18.23 | Qwell[] | but, I'm sure at least a couple do |
| 00:19.34 | Un1x | hey is it physcialy possibel tio have incoming calls via IAX and going via SIP or visa versa? |
| 00:19.40 | toerkeium | seems that I am not buffering with my php script, since if I pause the script for 2 seconds before echoing the results, it works |
| 00:19.49 | toerkeium | anyone has experienced this? |
| 00:19.54 | Qwell[] | Un1x: physically? no, it's all virtual |
| 00:19.56 | file | physically possible? |
| 00:20.21 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:20.37 | Qwell[] | bad file, bad |
| 00:21.12 | Kumba_ | Is it normal for ztcfg to make asterisk get red alarms on all channels even tho i'm not changing the zaptel.conf? |
| 00:21.15 | file | eep |
| 00:22.44 | *** join/#asterisk Kirds (n=chatzill@ip-93-4.dsl.newel.net) |
| 00:23.32 | *** join/#asterisk AJaymn (n=FreePBX8@156-77.dsl.scc.net) |
| 00:23.44 | *** part/#asterisk Kirds (n=chatzill@ip-93-4.dsl.newel.net) |
| 00:27.06 | Mattwj2005 | it is okay |
| 00:27.18 | Mattwj2005 | I just set it up as sip :) |
| 00:27.39 | *** join/#asterisk j0 (i=dan@S01060016b6b541be.va.shawcable.net) |
| 00:29.55 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 00:33.59 | Un1x | wtf is these gay FreePBx bots :S |
| 00:36.09 | Mattwj2005 | what are you talking about un1x? |
| 00:37.48 | Mattwj2005 | is that for telemarketers or something? |
| 00:42.19 | *** join/#asterisk MikeJ (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net) |
| 00:42.35 | turbo- | i actually havent set asterisk up yet (waiting for TDM card), but im planning on just using it as a pstn phone for local calls, and eventually route long distance through sipphone.com or something... |
| 00:43.08 | turbo- | i'll be using a cisco 7960 as a phone. if i put people on hold using that phone, will asterisk know if another phone is picked up and turn the hold music off? |
| 00:45.22 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 00:47.39 | hads | turbo-: No, each phone has it's own extension |
| 00:48.47 | turbo- | well, it's just going to have a single line using an FXO module on a TDM, and for now ive only got 1 ip phone to play with |
| 00:49.18 | hads | So you have nothing to worry abot then :) |
| 00:49.25 | hads | s/abot/about/ |
| 00:49.48 | turbo- | except that if i put someone on hold on the phone downstairs and go upstairs, asterisk will probably still be putting hold music out on the line |
| 00:50.08 | hads | So you do have more than one phone. |
| 00:50.28 | turbo- | only 1 ip phone |
| 00:51.11 | hads | Asterisk won't play nice if you have other phones plugged into your POTS line. |
| 00:52.04 | turbo- | hmm. they'd need some FXS ports to connect to, wouldnt they? |
| 00:52.30 | hads | Yes. |
| 00:53.37 | budmang | i need a good toll free per minute company. Instant toll free. iax |
| 00:54.49 | *** join/#asterisk wvtrixbox (n=roger@216.115.192.22) |
| 00:55.20 | *** join/#asterisk nailbags (n=nailbags@c220-237-12-224.randw1.nsw.optusnet.com.au) |
| 00:55.43 | *** part/#asterisk wvtrixbox (n=roger@216.115.192.22) |
| 00:55.44 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-120-130-206.gdrpmi.dsl-w.verizon.net) |
| 01:03.32 | Un1x | hey is it physcialy possibel tio have incoming calls via IAX and going via SIP or visa versa? |
| 01:04.15 | *** join/#asterisk andrew` (n=andrew@adsl-69-236-174-180.dsl.pltn13.pacbell.net) |
| 01:05.11 | Un1x | hey is it possible tio have incoming calls via IAX and going via SIP or visa versa? |
| 01:05.47 | turbo- | do TDM cards happen to monitor line voltage on the FXO modules? |
| 01:05.47 | andrew` | hey, i'm trying to use fxotune and I keep getting Unable to set impedance on fd 4 |
| 01:05.47 | andrew` | any ideas? |
| 01:06.11 | hads|home | turbo-: What do you mean? |
| 01:06.59 | turbo- | well, i want to tinker around with a mixed environment of IP phones and regular phones at home and ive got a TDM400 card sitting here to play with |
| 01:07.42 | turbo- | and if the TDM card can see increases or drops in voltage (a regular phone picking up a POTS line), i could probably make it a little bit more sane |
| 01:07.59 | hads|home | turbo-: No. |
| 01:09.08 | andrew` | ah, apparently it didn't like my older zaptel modules...but i cna't use the latest ver in production as it randomly hangs up the line |
| 01:09.29 | turbo- | hads|home: how does it detect hangups in certain areas then? |
| 01:10.10 | hads|home | turbo-: You'll need disconnect supervision form your telco, i.e. line reversal or open loop |
| 01:10.25 | hads|home | Or you can use busydetect if you really must. |
| 01:12.38 | toerkeium | guys, anyone knows what does the Async header? I tried it but I don't see the difference when calling |
| 01:14.03 | toerkeium | I want to make a call in both directions (caller, called) at the same time, so both phones ring at the same time |
| 01:14.54 | *** join/#asterisk SwK (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 01:15.36 | *** join/#asterisk Mike (n=mike@dsl-201-129-122-168.prod-infinitum.com.mx) |
| 01:16.07 | Mike | anyone knows why i could be getting alot of cause 16 hangups? |
| 01:16.40 | *** join/#asterisk iq|mobile (n=IQ@unaffiliated/iq) |
| 01:16.53 | *** join/#asterisk Trakkasure (n=Sgemtum@69-163-145-203.atlsfl.adelphia.net) |
| 01:18.37 | hads|home | Mike: Isn't cause 16 normal. |
| 01:20.50 | *** join/#asterisk Skarmeth (n=Skarmeth@201009109103.user.veloxzone.com.br) |
| 01:28.03 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 01:37.18 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 01:37.50 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:42.09 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 01:45.44 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 01:49.34 | brookshire | HOLD ON, READY OR NOT! |
| 01:49.42 | brookshire | I CAN MAKE IT I SWARE! |
| 01:49.47 | brookshire | or is it we |
| 01:51.46 | hads|home | living in a prayer |
| 01:51.55 | hads|home | s/in/on/ |
| 01:52.01 | twisted[asteria] | hahah |
| 01:52.02 | hads|home | bah |
| 01:52.09 | brookshire | DING! you win! |
| 01:52.13 | twisted[asteria] | loving on a player? |
| 01:52.13 | twisted[asteria] | what? |
| 01:52.21 | hads|home | lol |
| 01:52.37 | twisted[asteria] | "WHO LET THESE HOES IN MY ROOM?" |
| 01:52.45 | brookshire | *caugh* Gavin.is.gay *caugh* |
| 01:53.23 | brookshire | HEY FILE! |
| 01:53.27 | twisted[asteria] | FILE!~ |
| 01:53.30 | brookshire | OMG HI! |
| 01:53.34 | file | I KNOW WHERE YOU ARE!!! |
| 01:53.40 | file | are you at the bar with wifi? |
| 01:53.41 | brookshire | WHERE AM I G? |
| 01:53.47 | twisted[asteria] | hahahaha |
| 01:53.47 | brookshire | sorta |
| 01:53.50 | twisted[asteria] | kinda |
| 01:53.50 | brookshire | asteria.bar |
| 01:54.09 | file | well yeah, Asteria is just the front for a big bar organization |
| 01:54.16 | twisted[asteria] | suuuure. |
| 01:54.20 | file | *obviously* |
| 01:54.33 | brookshire | file: guess what |
| 01:54.47 | file | ummmm, whattttttttttt |
| 01:55.00 | brookshire | file: there is a fridge, with beer here.. *gasp* |
| 01:55.04 | twisted[asteria] | omg |
| 01:55.05 | file | uh oh |
| 01:55.09 | file | that's dangerous |
| 01:55.11 | twisted[asteria] | wtfbbq |
| 01:55.12 | file | you might... |
| 01:55.13 | file | drink it! |
| 01:55.18 | brookshire | possibly |
| 01:55.26 | twisted[asteria] | it's alright |
| 01:55.31 | twisted[asteria] | she moves in mysterious ways |
| 01:55.50 | brookshire | omg this song is sooo old |
| 01:55.57 | brookshire | we need asteria radio (tm) |
| 01:56.07 | file | for shizzle! |
| 01:56.09 | twisted[asteria] | this is such a remake by KMFDM though |
| 01:56.35 | brookshire | file: candian folk music asap |
| 01:56.40 | twisted[asteria] | my music totals 17:07:59:17 |
| 01:56.41 | file | noooooooooo |
| 01:57.28 | brookshire | "boom boom boom" |
| 01:57.38 | brookshire | 'tit tit tit tit tit tit tit tit |
| 01:58.35 | brookshire | "unf unf unf unf unf unf unf.. ah baby" |
| 01:58.39 | twisted[asteria] | Uhn Tiss Uhn Tiss Uhn Tiss baby |
| 01:58.47 | brookshire | yeah.. i guess it is more like that |
| 01:59.32 | twisted[asteria] | bloodhound gang owns my anus |
| 01:59.38 | dan__t | hed pe > * |
| 01:59.50 | brookshire | is that norsk? |
| 02:00.09 | dan__t | eh? |
| 02:00.13 | brookshire | hei! |
| 02:00.17 | twisted[asteria] | canuk? |
| 02:00.27 | dan__t | er, no, not really but ok. |
| 02:00.29 | brookshire | matt j: what do you make of that? |
| 02:00.38 | twisted[asteria] | matt j esta aqui? |
| 02:02.59 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 02:04.57 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 02:09.24 | *** join/#asterisk ctaloi (n=Chris@nat-66-218-1-226.usadatanet.com) |
| 02:13.54 | *** join/#asterisk lunaphyte (n=lunaphyt@pool-71-115-150-237.gdrpmi.dsl-w.verizon.net) |
| 02:22.53 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 02:23.51 | *** join/#asterisk WhoDaMan (n=none@161.91.171.66.subscriber.vzavenue.net) |
| 02:23.57 | *** part/#asterisk Skarmeth (n=Skarmeth@201009109103.user.veloxzone.com.br) |
| 02:24.18 | *** join/#asterisk Skarmeth (n=Skarmeth@201009109103.user.veloxzone.com.br) |
| 02:24.23 | WhoDaMan | can't find anyone at asterisk-biz, but does anyone here offer Burma (Myanmar) termination? |
| 02:24.58 | Mattwj2005 | hey did anyone check this out? http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk |
| 02:26.10 | brookshire | that's been the case for a while |
| 02:26.33 | Mattwj2005 | oh |
| 02:26.44 | Mattwj2005 | well I was impressed :) |
| 02:27.01 | *** join/#asterisk mitcheloc (n=mitchelo@205.134.225.18) |
| 02:28.12 | quid2478 | I tell you all the * Open-Source billing systems look like crap... not that I could code better. |
| 02:28.33 | Mattwj2005 | a buddy and I are doing Asterisk though a VPN |
| 02:32.45 | brookshire | matt: hotness |
| 02:33.23 | Mattwj2005 | yeah it is pretty sexy |
| 02:33.58 | brookshire | openvpn i imagine |
| 02:34.10 | Mattwj2005 | nope....IPCop |
| 02:34.19 | brookshire | what is that? |
| 02:34.55 | Mattwj2005 | http://www.ipcop.org/ A Linux firewall/vpn |
| 02:39.58 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 02:40.07 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 02:40.07 | *** join/#asterisk tengulre (n=hbma23@219.144.249.8) |
| 02:41.48 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 02:43.11 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 02:43.15 | rene- | hello |
| 02:43.21 | tengulre | hello |
| 02:43.27 | Mattwj2005 | hey |
| 02:45.14 | rene- | asterisk is dying me producing what ManXpower refers to as 'ECFO' the net result is asteisk going down silently, i was told by Manxpower to upload a bug report to mantis, i am compiling asterisk using make dont-optimize and getting this: http://pastebin.ca/116104 |
| 02:45.30 | *** part/#asterisk budmang (n=budman@12.206.134.162) |
| 02:45.36 | rene- | i meant that asterisk is dying on me |
| 02:45.43 | ManxPower | ECFO does not cause crashes. |
| 02:45.50 | ManxPower | It causes something that sounds like feedback |
| 02:45.58 | rene- | well what i have is loud bad noise |
| 02:46.05 | rene- | and asterisk going down |
| 02:46.10 | *** join/#asterisk andrew` (i=andrew@69-12-136-56.dsl.dynamic.sonic.net) |
| 02:46.19 | rene- | i am on debian sarge with asterisk 1.2.9.1 |
| 02:46.29 | brookshire | yay! |
| 02:46.36 | brookshire | debian > * |
| 02:46.36 | twisted[asteria] | Darwin jockstrap.local 8.7.0 Darwin Kernel Version 8.7.0: Fri May 26 15:20:53 PDT 2006; root:xnu-792.6.76.obj~1/RELEASE_PPC Power Macintosh powerpc |
| 02:46.38 | twisted[asteria] | HAH |
| 02:46.42 | rene- | heh |
| 02:46.45 | brookshire | MACOSX BLOWS |
| 02:46.56 | brookshire | jockstrap.local, lol |
| 02:46.58 | Qwell | brookshire: Isn't that...a good thing? |
| 02:46.58 | twisted[asteria] | so does your mom |
| 02:47.05 | twisted[asteria] | but i'm not complaining |
| 02:47.08 | brookshire | hmm.. maybe! |
| 02:47.21 | Qwell | drunkards :p |
| 02:47.26 | twisted[asteria] | not really |
| 02:47.30 | twisted[asteria] | not even really buzzing yet |
| 02:47.48 | rene- | brookshire: debian > * ? |
| 02:47.53 | brookshire | yes |
| 02:47.58 | twisted[asteria] | NO |
| 02:48.01 | brookshire | apt-get install asterisk |
| 02:48.19 | brookshire | of course it is old and shit and exploitable, but who cares! |
| 02:48.20 | twisted[asteria] | gentoo > *(&!OSX) |
| 02:48.21 | rene- | brook but that will get me a very old asterisk build |
| 02:48.34 | rene- | i am relying in 1.2 stuff |
| 02:49.02 | rene- | i think debian builds 1.07 or 1.09 |
| 02:49.59 | brookshire | rene-: 1.0.9 is pretty stable though |
| 02:50.00 | brookshire | :) |
| 02:50.42 | rene- | i have built stuff around realtime |
| 02:50.53 | rene- | i dont want to ditch that |
| 02:51.00 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 02:51.03 | rene- | so well |
| 02:51.19 | rene- | i need to build this core dump or whatever |
| 02:51.34 | tengulre | which OS is best for building asterisk server? |
| 02:51.39 | rene- | does anybody knows why i cant build asterisk with make dont-optimize ? |
| 02:51.40 | brookshire | debian of course |
| 02:51.44 | twisted[asteria] | tengulre, linux |
| 02:51.46 | Qwell | gentoo, nub |
| 02:51.48 | twisted[asteria] | brookshire, debian != OS |
| 02:51.49 | brookshire | AAH |
| 02:51.52 | dlynes_laptop | tengulre: OS/2 |
| 02:51.59 | rene- | aah? |
| 02:52.02 | brookshire | window 2.0 |
| 02:52.02 | rene- | really |
| 02:52.08 | twisted[asteria] | netware. |
| 02:52.08 | twisted[asteria] | honest. |
| 02:52.13 | brookshire | OPENVMS |
| 02:52.18 | Qwell | beos |
| 02:52.19 | dlynes_laptop | MS-DOS |
| 02:52.21 | dlynes_laptop | honest. |
| 02:52.22 | twisted[asteria] | SYSTEM V |
| 02:52.24 | twisted[asteria] | BITCHES |
| 02:52.26 | rene- | is debian for noobs? |
| 02:52.32 | Qwell | twisted[asteria]: it *might* actually compile on that |
| 02:52.32 | twisted[asteria] | gawd yes |
| 02:52.33 | dlynes_laptop | rene-: no |
| 02:52.37 | dlynes_laptop | rene-: it's for total noobs |
| 02:52.45 | twisted[asteria] | Qwell, yea, except system V is not totally posix compliant |
| 02:52.49 | rene- | what would a real man use? |
| 02:52.50 | Qwell | so? :P |
| 02:52.54 | Qwell | rene-: gentoo |
| 02:52.54 | twisted[asteria] | a condom |
| 02:52.56 | dlynes_laptop | rene-: his fingers |
| 02:52.59 | twisted[asteria] | er wait |
| 02:53.01 | twisted[asteria] | haha |
| 02:53.33 | rene- | ha |
| 02:53.37 | rene- | so gentoo |
| 02:53.38 | rene- | mmm |
| 02:53.53 | rene- | does gentoo bundles ages old software? |
| 02:53.54 | brookshire | commodore 64 |
| 02:54.03 | twisted[asteria] | bah |
| 02:54.05 | brookshire | gentoo breaks |
| 02:54.08 | twisted[asteria] | c64 has no fsck |
| 02:54.12 | twisted[asteria] | use fortran |
| 02:54.14 | twisted[asteria] | er wait |
| 02:54.16 | twisted[asteria] | no |
| 02:54.20 | twisted[asteria] | GEOS |
| 02:54.23 | rene- | old computers are lame and so it is the people that raves about them |
| 02:54.49 | twisted[asteria] | hmm |
| 02:54.58 | JunK-Y | yo yo yo |
| 02:55.00 | twisted[asteria] | i can find you |
| 02:55.03 | hads|home | heh |
| 02:56.42 | file | yooooooooo |
| 02:56.59 | JunK-Y | hey hey mr. file! |
| 02:57.09 | file | yo yo Junky, you're hurting me |
| 02:57.10 | rene- | hey junky |
| 02:57.20 | JunK-Y | hey rene! whats up? |
| 02:57.39 | quid2478 | anyone know where I can find an "up to the minute" list of country codes and prefixes (ie. for mobile, landline, etc)? |
| 02:57.44 | JunK-Y | file: i just get back, from cluecon, guess who wants to see all pics+video! |
| 02:57.46 | rene- | not much, what have you been up to |
| 02:57.50 | rene- | junky cool |
| 02:57.51 | file | JunK-Y: :D |
| 02:57.58 | rene- | show us the girls |
| 02:57.58 | twisted[asteria] | JunK-Y, yea sure |
| 02:57.59 | file | JunK-Y: put 'em up online, I'd like to see them toooooooo |
| 02:58.01 | twisted[asteria] | CHICAS |
| 02:58.15 | JunK-Y | rene: nothing, same routine, same job, same gf, same everything! |
| 02:58.31 | JunK-Y | file: i'll soon |
| 02:58.46 | rene- | cool man.. |
| 02:58.50 | twisted[asteria] | JunK-Y, your gf still likes me, right? |
| 02:59.10 | *** join/#asterisk tempest1 (n=asf@adsl-150-198-149.chs.bellsouth.net) |
| 02:59.16 | JunK-Y | she says: yeah of course and forever! |
| 02:59.24 | JunK-Y | how can she forgets ya! |
| 02:59.31 | twisted[asteria] | yay! |
| 02:59.36 | twisted[asteria] | although I have never met her |
| 02:59.41 | rene- | weird |
| 03:00.02 | JunK-Y | twisted[asteria]: thanks godness, she fantasms too much on ya, i'll be scared! |
| 03:00.02 | rene- | wtf this means: /usr/lib/libc_nonshared.a(stat.oS)(.text.__i686.get_pc_thunk.bx+0x0): In function `__i686.get_pc_thunk.bx': |
| 03:00.03 | rene- | : multiple definition of `__i686.get_pc_thunk.bx' |
| 03:00.28 | JunK-Y | shes a great girl, ask file, he met her. |
| 03:00.40 | file | oh did I ever meet her... |
| 03:00.40 | rene- | i am using Linux srv5 2.6.8-3-686 #1 Sat Jul 15 10:32:25 UTC 2006 i686 GNU/Linux |
| 03:00.43 | twisted[asteria] | JunK-Y, haha |
| 03:00.44 | file | she's GREAT |
| 03:00.50 | twisted[asteria] | JunK-Y, i would like to meet her |
| 03:00.52 | twisted[asteria] | in all seriousness |
| 03:01.10 | JunK-Y | come spend some days in montreal! |
| 03:03.35 | rene- | manx: so i am not experiencing ECFO. is it normal to hear ugly and loud noises and then see poor asterisk die by it self? i am using AMI but i am being conservative i think. |
| 03:03.46 | rene- | in its usage |
| 03:04.55 | rene- | i am just going to reinstall the funcking thing over |
| 03:05.02 | rene- | what OS is it going to be? |
| 03:05.23 | brookshire | debian |
| 03:05.32 | rene- | what kernel? |
| 03:05.37 | brookshire | default |
| 03:05.41 | brookshire | heh |
| 03:05.51 | brookshire | unless you have 64-bit or multiprocessors |
| 03:06.00 | twisted[asteria] | MPC |
| 03:06.09 | brookshire | ppc |
| 03:06.14 | rene- | i have an intel 3.2 i think it does HT |
| 03:06.15 | Qwell | sparc |
| 03:08.11 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
| 03:08.42 | asterisk-dud | is there a way to make asterisk ring back when a local extention becomes available after being busy |
| 03:10.04 | twisted | uhhh... specify ring back... you mean like try until free? |
| 03:10.13 | twisted | then originate from that extensin? |
| 03:10.19 | twisted | s/extensin/extension |
| 03:10.41 | brookshire | rene: use smp then |
| 03:11.21 | rene- | ah |
| 03:11.23 | rene- | cool |
| 03:11.29 | rene- | ok |
| 03:11.31 | rene- | mmm |
| 03:11.47 | rene- | why asterisk screams and yells at me on a friday night |
| 03:11.57 | asterisk-dud | twisted: lets say the lines busy, i dial a number, like 6, and them aterisk alerts me when the line becomes available |
| 03:11.58 | rene- | i think it has to do with AMI |
| 03:12.05 | rene- | it happened to me on a different PC |
| 03:12.47 | twisted | asterisk-dud, uhh... well, not really |
| 03:12.49 | rene- | well at least if i submit the trace i can get someone to tell me wtf is going on |
| 03:13.01 | twisted | but you could just repeatedly try until it's available and have it originate the call from that channel |
| 03:13.05 | twisted | or write an app to do it :) |
| 03:13.27 | rene- | use the asterisk database |
| 03:14.03 | twisted | uhh |
| 03:14.11 | twisted | how is astdb going to know when the channel is free? |
| 03:14.15 | twisted | oh wait |
| 03:14.15 | twisted | it isn't |
| 03:14.51 | asterisk-dud | so if i make that call, its busy, i can somehow tell asterisk to keep trying and ring me back when available? |
| 03:15.01 | rene- | well you have to reeschedule the call |
| 03:15.09 | rene- | every time it fails |
| 03:15.25 | rene- | i dont know if you could write a .call file entirely from the dial plan |
| 03:15.40 | twisted | you can't |
| 03:15.52 | rene- | then it is an AGI |
| 03:16.00 | twisted | yep. |
| 03:16.03 | twisted | OR |
| 03:16.19 | twisted | you could code an app that watches the channel structure and waits for the inuse flag to set back to 0 |
| 03:16.42 | rene- | or you could code it and sell it to the guy who asked for it |
| 03:16.47 | twisted | or something along those lines :P |
| 03:16.49 | asterisk-dud | ok, is there a way to capture events from asterisk |
| 03:16.55 | twisted | AMI |
| 03:17.06 | asterisk-dud | because when a certain channel hangs up, asterisk records it |
| 03:17.18 | rene- | you can do that in the dialplan |
| 03:17.19 | asterisk-dud | if i could capture that i can make it work |
| 03:17.41 | twisted | you can, sure |
| 03:17.45 | rene- | then spawn a script using agi to write a call file for asterisk |
| 03:17.47 | twisted | but that's highly inefficient |
| 03:18.20 | rene- | what are the new options using AJAM, twisted? |
| 03:18.39 | hads|home | An app would be way cooler |
| 03:18.48 | rene- | an app is 1337 |
| 03:19.04 | rene- | btw old school pbx have that function |
| 03:19.12 | twisted | rene-, AJAM? |
| 03:19.13 | asterisk-dud | i know |
| 03:19.20 | asterisk-dud | its handy |
| 03:21.47 | asterisk-dud | how can i test if a line is available with ringing it? |
| 03:21.53 | asterisk-dud | from a dial plan |
| 03:21.59 | asterisk-dud | without |
| 03:22.05 | asterisk-dud | without ringing it |
| 03:22.05 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 03:22.14 | twisted | ChanIsAvail() |
| 03:22.16 | twisted | ;) |
| 03:22.20 | rene1 | AJAM: asterisk xml-rpc manager interfase |
| 03:22.21 | asterisk-dud | cool |
| 03:22.25 | brookshire | ajam is like.. xml manager |
| 03:22.42 | Mattwj2005 | how long does it take e164.org to update on sipbroker.com? |
| 03:22.44 | brookshire | basically manager over http |
| 03:22.46 | twisted | xml ... the only language you have to use > 256 bytes to describe NULL |
| 03:23.42 | file | if at first it doesn't work, use XML - at least that way it won't work and it'll look complicated |
| 03:24.08 | file | OR |
| 03:24.18 | file | if at first it doesn't work, remap the SIGSEGV signal handler so it doesn't matter |
| 03:25.15 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
| 03:25.33 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 03:26.28 | Mattwj2005 | any ideas? |
| 03:28.27 | brookshire | i would ask the mailing list! |
| 03:29.43 | Mattwj2005 | naw....it'll be updated by then :P |
| 03:30.42 | rene1 | bookshire: is there some more clever advice ? i am reluctant to backup a debian-asterisk-rails system reinstall it and then send a core dump to bugs.digium.com.... what can be causing a sudden terrible sound and then killing asterisk silently (using 1.2.9-10 series and SJphones on windows mainly |
| 03:31.33 | brookshire | you're using rails with asterisk? |
| 03:31.47 | brookshire | that's probably what is making it core dump |
| 03:31.53 | rene1 | how come |
| 03:32.05 | twisted | asterisk != ruby |
| 03:32.14 | rene1 | i am using a proxy called RAMI |
| 03:32.19 | rene1 | it is in ruby |
| 03:32.36 | rene1 | rails for the realtime interfase should be doing no harm |
| 03:32.44 | twisted | RAMI |
| 03:32.51 | twisted | sounds like a sex position |
| 03:32.52 | bkw__ | I swear if I hear more rails crap.... i'm gonna scream |
| 03:32.56 | twisted | "Get in the RAMI position" |
| 03:32.59 | brookshire | sounds like a product from asteria |
| 03:33.05 | *** kick/#asterisk [brookshire!n=twisted@pdpc/supporter/active/twisted] by twisted (twisted) |
| 03:33.07 | rene1 | it is written in ruby |
| 03:33.09 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 03:33.11 | brookshire | haha |
| 03:33.17 | hads|home | It's maddness I tell you! |
| 03:33.18 | *** kick/#asterisk [twisted!n=north@unaffiliated/qwell] by Qwell (brookshire) |
| 03:33.18 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 03:33.18 | *** mode/#asterisk [+o twisted] by ChanServ |
| 03:33.25 | *** kick/#asterisk [Qwell!n=twisted@pdpc/supporter/active/twisted] by twisted (twisted) |
| 03:33.30 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 03:33.30 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:33.46 | twisted | BOTO |
| 03:33.46 | rene1 | rails is cool and so is ruby i pity poor souls that work with php, but that is not why i am here for |
| 03:33.53 | brookshire | omg. .you're going to make the server ops appear |
| 03:34.16 | Nugget | rene1 is to php as nugget is to mysql. |
| 03:34.23 | hads|home | telnet boi! |
| 03:34.27 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 03:34.27 | *** mode/#asterisk [+o brookshire] by Qwell |
| 03:34.32 | *** kick/#asterisk [twisted!i=mbrooks@hijacked.us] by brookshire (HA.. LAME!) |
| 03:34.33 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 03:34.33 | *** mode/#asterisk [+o twisted] by ChanServ |
| 03:34.34 | *** mode/#asterisk [-o brookshire] by twisted |
| 03:34.38 | Qwell | ha |
| 03:34.41 | *** mode/#asterisk [-o Qwell] by twisted |
| 03:34.49 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:34.51 | Qwell | nub |
| 03:34.52 | *** mode/#asterisk [-o Qwell] by twisted |
| 03:34.56 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:34.58 | *** mode/#asterisk [-o Qwell] by twisted |
| 03:35.11 | Qwell | lame |
| 03:35.19 | *** mode/#asterisk [+o Qwell] by twisted |
| 03:35.23 | Qwell | still lame :p |
| 03:35.29 | *** mode/#asterisk [+b %Qwell!*@*] by twisted |
| 03:35.30 | brookshire | qwell: /cycle |
| 03:35.34 | Qwell | pfft |
| 03:35.36 | twisted | he has ops |
| 03:35.53 | *** mode/#asterisk [-b %Qwell!*@*] by Qwell |
| 03:36.59 | sumasuma | hi |
| 03:37.04 | brookshire | hello |
| 03:37.07 | twisted | howdy |
| 03:37.08 | rene1 | ok so asterisk != ruby, i think we all knew that, it could make a good topic line |
| 03:37.21 | sumasuma | always my asterisk stucks up at loading chan_sip |
| 03:37.31 | twisted | suma, remove all hostnames from sip.conf |
| 03:37.53 | brookshire | cat /dev/null > /etc/asterisk/sip.conf |
| 03:37.55 | brookshire | heh |
| 03:37.59 | brookshire | no.. don't do that |
| 03:38.00 | brookshire | really |
| 03:38.01 | brookshire | don't |
| 03:38.04 | twisted | yea no |
| 03:38.12 | twisted | brb smokey treat |
| 03:38.18 | sumasuma | ok |
| 03:38.27 | brookshire | it will blank your sip.conf if you do |
| 03:38.41 | L|NUX | hello every one |
| 03:38.41 | L|NUX | salam |
| 03:38.44 | L|NUX | O_o |
| 03:38.50 | sumasuma | hi L|NUX |
| 03:39.03 | sumasuma | how r u doing ? |
| 03:39.11 | L|NUX | sumasuma : i am fine :) |
| 03:39.22 | L|NUX | sumasuma : you tell me about your self ? |
| 03:39.59 | rene1 | twisted: i am a boto? |
| 03:40.17 | rene1 | damn |
| 03:40.20 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 03:41.13 | rene1 | so it is my damn proxy what is causing asterisk to die? the thing is not supported anymore apparently |
| 03:41.24 | Nugget | http://www.last.fm/user/MacNugget * Nugget discos |
| 03:41.52 | rene1 | Nugget, so you dislike mysql? |
| 03:42.03 | Nugget | hate it with the fire of a thousand suns. |
| 03:43.06 | rene1 | so you are what a PG guy? |
| 03:43.11 | file | Nugget: SLACKER |
| 03:43.18 | Nugget | I'm just a database guy. |
| 03:43.57 | twisted | rene1, no |
| 03:44.06 | twisted | rene1, i was saying BOTO as in Battle Of The Operators |
| 03:44.22 | rene1 | ahh i thought you meant the weird dolphin |
| 03:44.30 | twisted | haha no |
| 03:44.48 | DrkShdw | him and Decibull |
| 03:44.56 | twisted | pg is the sheezy |
| 03:45.18 | Nugget | I use all of them. mysql holds a special place in my bowels, though. |
| 03:45.32 | twisted | I use them all too |
| 03:45.34 | twisted | but that's because i'm forced ot |
| 03:45.36 | twisted | s/ot/to |
| 03:46.33 | DrkShdw | everyone is pretty much forced to use mysql, because most of the OS community caters their software around it.. so unless you wanna spend your time patching/porting software to another DB.. guess what? |
| 03:46.47 | twisted | write custom funcitons? |
| 03:46.52 | twisted | s/funcitons/functions |
| 03:46.59 | twisted | alcohol......taking....hold |
| 03:47.20 | twisted | you can make PG act like My |
| 03:47.31 | twisted | why you would want to is beyond me |
| 03:47.52 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:47.52 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:47.59 | twisted | russellb!! |
| 03:48.04 | rene1 | ok so i have to choose between rewriting the proxy i am using (which i dont fully understand how it work but i have expanded it a bit to add some extra events) or backing up, capture a trace, send it to digium and pray to not get something like ' you are crashing it via the AMI' |
| 03:48.06 | Nugget | Reminds me of http://thedailywtf.com/forums/thread/84491.aspx (which is a real mind-boggler) |
| 03:48.13 | russellb | twisted: !! |
| 03:48.24 | twisted | russellb, i'm counting the days until you move back to hsv |
| 03:48.27 | brookshire | or use a db extraction layer that supports multiple servers :) |
| 03:48.33 | russellb | twisted: :D |
| 03:48.37 | twisted | brookshire, abstraction. |
| 03:48.40 | Nugget | I think you mean "abstraction" layer. |
| 03:48.41 | russellb | twisted: less than 5 months, not too bad |
| 03:48.43 | brookshire | abstration |
| 03:48.46 | brookshire | asdfj;aksdf |
| 03:48.51 | brookshire | abstraction i mean, lol |
| 03:48.52 | twisted | russellb, yeah |
| 03:48.59 | Nugget | extracting the database is when you use flat files. ;) |
| 03:49.15 | twisted | hehe |
| 03:49.22 | twisted | brookshire is a dbnub |
| 03:49.39 | brookshire | i don't have google to check my answers, lol |
| 03:49.39 | DrkShdw | nugget: you still working for ud? |
| 03:49.44 | brookshire | stupid console |
| 03:50.08 | Nugget | nope, I left UD in spring '04. I work for one of my customers now (from when I was at UD) and in my spare time I hack on flightaware.com |
| 03:50.16 | brookshire | russell: firmware status? |
| 03:50.33 | DrkShdw | flightaware, wasn't that dbakers pet project at one point? |
| 03:50.39 | Nugget | it's a "thing" now |
| 03:50.44 | DrkShdw | nice |
| 03:50.51 | Nugget | real revenue and we're about to hire some people |
| 03:50.55 | russellb | brookshire: idle |
| 03:51.03 | *** join/#asterisk adfsadfae (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
| 03:51.04 | file | thing with stuff |
| 03:51.04 | brookshire | no updates? |
| 03:51.09 | brookshire | *pfft* |
| 03:51.10 | twisted | awww |
| 03:51.13 | russellb | no, i've been busy, sheesh |
| 03:51.38 | *** join/#asterisk bmg505 (n=leon@c1-199-8.rndf.isadsl.co.za) |
| 03:51.54 | adfsadfae | can some one help me with a mysql/php asterisk problem? |
| 03:51.58 | Nugget | yeah, d.net's pretty stagnant these days |
| 03:52.18 | DrkShdw | Nugget: sounds cool. last I looked at the site, it was sorta like a blog/forum/whatnot.. apparently it's evolved into much more now |
| 03:52.35 | Nugget | nah, that was a different site. ityt.com. we sold that. |
| 03:52.45 | DrkShdw | oh yeah, ityt. that was it. |
| 03:52.54 | adfsadfae | can some one help me with a mysql/php asterisk problem? Ive been working on this for a week now \ |
| 03:53.12 | DrkShdw | adfsadfae: it's probably better to ask the actual question :) |
| 03:53.29 | brookshire | ALL THE LEAVES ARE BROWN! |
| 03:53.48 | Nugget | and the sky is gray. |
| 03:54.04 | brookshire | DROP DOWN TO MY KNEES! |
| 03:54.46 | adfsadfae | this is what i keep getting |
| 03:54.58 | adfsadfae | Couldn't connect to 67.52.187.22:3306/tcp: IO::Socket::INET: connect: Connection refused at /home/cron/AST_VDhopper.pl line 141 |
| 03:55.28 | brookshire | what the hell is that |
| 03:56.01 | twisted | VDhopper |
| 03:56.16 | brookshire | obviously |
| 03:56.30 | twisted | adfsadfae, looks like VICIDIAL |
| 03:56.41 | brookshire | /etc/init.d/mysql start |
| 03:56.42 | brookshire | :) |
| 03:58.30 | adfsadfae | mysql/php is 1 server and asterisk/vicidial is another server |
| 03:59.32 | brookshire | adfsadfae: your mysql server isn't started, or firewalled, or is not responding on that ip address |
| 04:00.02 | adfsadfae | Its on a public ip |
| 04:00.14 | adfsadfae | how do i open the port for it |
| 04:00.46 | brookshire | it's open.. i just connected to it |
| 04:00.51 | brookshire | so.. something it's right |
| 04:00.59 | brookshire | s/it's/isn't |
| 04:01.14 | adfsadfae | how come its still refusing |
| 04:01.28 | brookshire | i have no idea |
| 04:01.33 | brookshire | millions of reasons |
| 04:01.56 | brookshire | /j mysql :) |
| 04:01.59 | adfsadfae | ??? Ive been working on this all week |
| 04:02.31 | adfsadfae | need help |
| 04:02.33 | twisted | looks like your mysql server isn't accepting connections |
| 04:02.54 | adfsadfae | how do I make it accept connections |
| 04:03.01 | twisted | rtfm |
| 04:03.09 | twisted | OH, and make mysql listen on the IP |
| 04:03.20 | adfsadfae | how |
| 04:03.30 | adfsadfae | i m new to this |
| 04:03.32 | twisted | see previous statement |
| 04:03.34 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.55) |
| 04:03.41 | adfsadfae | i bit off more then i can chew |
| 04:03.53 | *** join/#asterisk dorphalsig (i=Dorphals@pcsp168-254.supercabletv.net.co) |
| 04:04.01 | dorphalsig | Hi |
| 04:04.02 | brookshire | adfsadfae: we're pointing you in the right direction |
| 04:04.07 | brookshire | this is not an asterisk problem |
| 04:04.10 | brookshire | this is a mysql problem |
| 04:04.27 | brookshire | dorphalsig: hi! |
| 04:04.30 | dorphalsig | Can I trigger an event once a call has been answered? |
| 04:04.44 | twisted | dorphalsig, sure |
| 04:04.59 | dorphalsig | I mean its more or less like this: call comes in -> goes to queue -> gets answered -> operator gets a nice webpage telling her who the caller is |
| 04:05.17 | dorphalsig | so I'd need to know which channel the call went to |
| 04:05.22 | rene1 | for a couple of hours until asterisk dies |
| 04:05.27 | dorphalsig | How can I do that? |
| 04:05.30 | rene1 | with a loud noise |
| 04:06.16 | dorphalsig | twisted --> do you know how to do that? |
| 04:06.16 | brookshire | rene: what are you doing? |
| 04:06.24 | brookshire | just agi? |
| 04:06.27 | brookshire | how many calls? |
| 04:06.31 | twisted | dorphalsig, well, kinda.... we use AMI for that stuff though |
| 04:06.33 | rene1 | i have a ruby process that loops forever |
| 04:07.00 | rene1 | it harvest information from asterisk and mysql to count the number of calls it is going to originate, |
| 04:07.06 | twisted | dorphalsig, but you *COULD* run a macro that calls an AGI that drops the channel var ${CHANNEL} into a db somewhere that your agent pulls for |
| 04:07.10 | brookshire | soo.. if you don't use that ruby script.. does asterisk die? |
| 04:07.10 | twisted | or the agi could push it |
| 04:07.11 | rene1 | it also harvest agentcomplete and agentconnect events |
| 04:07.32 | dorphalsig | twisted ... how would AMI be? |
| 04:07.39 | rene1 | brookshire: if i dont use the script. then i got nothing but a plain asterisk system |
| 04:07.40 | dorphalsig | monitor the channels untill they are answered |
| 04:07.54 | dorphalsig | ? |
| 04:08.03 | brookshire | rene: so you're saying this isn't a problem with asterisk? |
| 04:08.07 | twisted | dorphalsig, right, you'd need to know what agent is on what channel, then monitor the outbound channel that's associated with your agent channel |
| 04:08.15 | rene1 | asterisk is the one dying tho |
| 04:08.21 | rene1 | and making a horrible sound |
| 04:08.28 | twisted | asterisk makes sound? |
| 04:08.37 | brookshire | well yeah.. if you chew up enough resources.. that definitely can happen |
| 04:08.38 | rene1 | it is probably corrupted audio packets |
| 04:08.46 | dorphalsig | twisted --> yeah, it should go like arrrgh! and then drops dead :P |
| 04:08.48 | rene1 | or whatever |
| 04:08.53 | twisted | dorphalsig, hahahah |
| 04:09.08 | rene1 | brookshire usage never goes beyong 5% |
| 04:09.20 | rene1 | i have a big fat sleep 1 (second) in each loop |
| 04:09.26 | twisted | ooh yeah. |
| 04:09.35 | brookshire | rene1: you mean a load of 5? |
| 04:09.36 | rene1 | beyond * |
| 04:09.37 | twisted | sleep me bigger baby |
| 04:09.39 | dorphalsig | Speaking of weird asterisk problems ... some channels 'die' on me |
| 04:09.49 | dorphalsig | zap |
| 04:09.50 | twisted | 5.00? |
| 04:10.00 | rene1 | vmstat shows 95% idle |
| 04:10.13 | rene1 | my dialer script is using like 5% cpu |
| 04:10.14 | brookshire | that's memory stats |
| 04:10.21 | brookshire | asterisk doesn't use much memory |
| 04:10.27 | twisted | vmstat == virtual memory stat |
| 04:10.29 | twisted | not cpu stat |
| 04:10.36 | rene1 | ok |
| 04:10.36 | brookshire | cpu matters |
| 04:10.38 | rene1 | top |
| 04:10.42 | dorphalsig | they're working and then they stop giving dialtone |
| 04:10.45 | rene1 | shows 5% usage for my main script |
| 04:11.01 | twisted | dorphalsig, what hardware/ |
| 04:11.03 | brookshire | what is the load? |
| 04:11.06 | rene1 | and less than 1.0 for mysql, asterisk |
| 04:11.07 | twisted | s/\//? |
| 04:11.31 | dorphalsig | TE410P connected to a couple of Zhone CBs |
| 04:11.36 | brookshire | s/twisted/naked strippers/ |
| 04:11.37 | twisted | OOOH ZHONES |
| 04:11.58 | rene1 | bookshire havent measured |
| 04:11.58 | twisted | yeah, zhones are fun. I have one at home that half the channels act that way |
| 04:12.04 | twisted | but when i hook up ta750's they dont' |
| 04:12.15 | rene1 | right now with no script is 0.07, 0.05, 0.01 |
| 04:12.18 | dorphalsig | so you say its a CB issue? |
| 04:12.20 | rene1 | let me fire those scripts |
| 04:12.24 | twisted | dorphalsig, possibly, yes. |
| 04:12.30 | twisted | i'm not saying that for sure, but it's quite possible |
| 04:12.35 | dorphalsig | but a few months ago we were not having those issues |
| 04:12.38 | twisted | yeah |
| 04:12.43 | twisted | dorphalsig, zhones suck, and they die. |
| 04:12.51 | twisted | mine was fine for the first 18 months or so |
| 04:12.55 | twisted | then it started shitting the bed |
| 04:13.05 | dorphalsig | lol shitting the bed |
| 04:13.13 | dorphalsig | either way, when the channel dies i do a zap show channel |
| 04:13.16 | dorphalsig | and I get its onhook |
| 04:13.20 | twisted | yep |
| 04:13.21 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.13.Dial1.SanJose1.Level3.net) |
| 04:13.23 | dorphalsig | but I get an owner |
| 04:13.35 | dorphalsig | and i have to restart * to get it working again |
| 04:13.40 | twisted | ooh hmm |
| 04:13.45 | brookshire | twisted: i blame gentoo |
| 04:13.57 | twisted | dorphalsig, that sounds bizzare.. .my channels don't recoup that wya |
| 04:14.00 | twisted | s/wya/way |
| 04:14.05 | twisted | brookshire, gentoo owns you |
| 04:14.15 | rene1 | brookshireL just after starting the proxy and the script i got load average: 0.65, 0.16, 0.04 |
| 04:14.25 | rene1 | the actual rails app runs from another host |
| 04:14.32 | brookshire | is that when it is dying? |
| 04:14.35 | dorphalsig | twisted --> restart * = restart zapata and * |
| 04:14.37 | rene1 | no |
| 04:14.43 | rene1 | i need to generate calls to make it die |
| 04:14.46 | brookshire | what is the load when asterisk dies |
| 04:14.55 | rene1 | just a sec |
| 04:15.17 | twisted | dorphalsig, hmm... running release or svn trunk/ |
| 04:15.20 | twisted | s/\/? |
| 04:15.25 | dorphalsig | 1.2.7 |
| 04:15.28 | twisted | eek |
| 04:15.28 | twisted | update |
| 04:15.36 | twisted | 1.2.7 has security flaws too |
| 04:15.39 | brookshire | isn't that the one with the timebomb? |
| 04:15.49 | brookshire | no.. that was 1.2.2 |
| 04:15.50 | brookshire | heh |
| 04:15.56 | twisted | heh |
| 04:16.02 | brookshire | sip go boom |
| 04:16.12 | twisted | more like rtp |
| 04:16.18 | brookshire | tru |
| 04:16.20 | brookshire | e\ |
| 04:16.25 | twisted | brb smoke |
| 04:16.27 | Un1x | i cant wait to learn more of asterisk going to buy the book sometime soon :D |
| 04:16.38 | dorphalsig | I'm not using * outside my firewall, so I guess I'm kinda safe for the moment |
| 04:16.44 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 04:16.54 | dorphalsig | Un1x ... just download and print it |
| 04:16.57 | dorphalsig | its creative commons |
| 04:17.04 | dorphalsig | so its free (like a bee) |
| 04:17.08 | Un1x | yea i know but i dont have a rpinter... |
| 04:17.12 | *** part/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 04:17.13 | Un1x | well i do i got 2 |
| 04:17.20 | Un1x | but it requires alot of INK! |
| 04:17.30 | Un1x | it's going to cost me roughly same ammount 40$ for new ink |
| 04:17.33 | Un1x | and 40$ for book |
| 04:17.34 | brookshire | it's worth the money for the sexy cover |
| 04:17.38 | Un1x | so id rather bye the book :D |
| 04:17.41 | Un1x | yea hehe :) |
| 04:17.58 | dorphalsig | hehehe |
| 04:18.20 | Qwell | and I hear that for every copy you buy, blitzrage can like...I don't know...afford to eat, or something |
| 04:18.26 | Qwell | ie; buy one :P |
| 04:18.39 | dorphalsig | lol |
| 04:18.43 | file | eating is overrated |
| 04:18.48 | russellb | so is sleeping |
| 04:18.59 | file | Qwell: !!! |
| 04:19.01 | file | russellb: !!! |
| 04:19.03 | russellb | zomg |
| 04:19.06 | brookshire | file! |
| 04:19.09 | brookshire | russellb! |
| 04:19.12 | russellb | brookshire: !!! |
| 04:19.13 | russellb | file: !! |
| 04:19.15 | brookshire | qwell! |
| 04:19.15 | russellb | Qwell: !!!! |
| 04:19.29 | dorphalsig | russellb: ¡¡¡¡¡ |
| 04:19.35 | russellb | dorphalsig: !!!!!!!!!!! |
| 04:19.51 | file | yay |
| 04:19.56 | *** join/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 04:19.57 | russellb | brookshire: GUESS WHAT I AM DOING |
| 04:20.04 | russellb | brookshire: polycom is updating as we speak. |
| 04:20.10 | brookshire | russellb: updating your firmware? |
| 04:20.12 | brookshire | OPMG!!@# |
| 04:20.12 | brookshire | !@# |
| 04:20.13 | brookshire | !@ |
| 04:20.14 | brookshire | 1231231 |
| 04:20.24 | file | hi |
| 04:20.26 | brookshire | finally! |
| 04:20.33 | twisted | hehe |
| 04:20.41 | brookshire | then autoprov in 200 easy steps |
| 04:21.00 | dorphalsig | 200 not-so-easy steps?? |
| 04:21.55 | russellb | zomg, IT'S ALIVE |
| 04:24.46 | rene1 | brookshire: i have sent like 8 connected calls out of 30 generated and this is my load |
| 04:24.48 | rene1 | 0.06, 0.07, 0.04 |
| 04:26.02 | twisted | 184062 |
| 04:26.40 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 04:26.51 | rene1 | it is not that loaded is it?? |
| 04:27.00 | brookshire | did asterisk crash? |
| 04:27.04 | rene1 | not yet |
| 04:27.12 | dorphalsig | I'm off |
| 04:27.14 | dorphalsig | bye |
| 04:27.16 | brookshire | then it means nothing |
| 04:27.17 | rene1 | bye bye |
| 04:27.20 | rene1 | ok |
| 04:28.38 | twisted | hehe |
| 04:28.42 | twisted | upc codes are fun |
| 04:28.54 | twisted | 080432500170 |
| 04:30.05 | rene1 | mmm was there a bug in digium regarding the usage of show queues???? |
| 04:30.10 | brookshire | that means 'nub' in upc |
| 04:30.14 | twisted | hah |
| 04:30.21 | brookshire | oh.. you're using queues |
| 04:30.30 | brookshire | and agents too? |
| 04:30.48 | rene1 | yes |
| 04:30.55 | brookshire | agents = bad |
| 04:30.57 | rene1 | it is meant to be a predictive dialer |
| 04:31.01 | rene1 | i have been told that |
| 04:31.11 | brookshire | use dynamic queue members |
| 04:31.12 | rene1 | that add queue members |
| 04:31.13 | russellb | the issues we are seeing lately are related to the use of chan_agent in combination with transfers. |
| 04:32.27 | rene1 | i am not transfering directly but i would guess that queues transfer calls to agents |
| 04:32.40 | russellb | no ... |
| 04:32.47 | russellb | what I'm talking about is the agents transferring calls |
| 04:32.51 | rene1 | ok |
| 04:32.53 | brookshire | do your agents transfer anyone?> |
| 04:32.53 | rene1 | not doing that |
| 04:32.58 | twisted | hehe |
| 04:33.00 | brookshire | okay |
| 04:33.22 | CunningPike | ~seen dlynes_laptop |
| 04:33.42 | jbot | dlynes_laptop is currently on #asterisk (1h 42m 42s). Has said a total of 6 messages. Is idling for 1h 40m 46s, last said: 'rene-: his fingers'. |
| 04:33.48 | brookshire | well you can try converting your agents to dynamic queue members and see if that fixes stuff |
| 04:34.05 | rene1 | i generate calls then the good calls show up in the dialplan, i then send those calls to the queue they belong, (i read that queuename from an asterisk channel variable) |
| 04:34.06 | twisted | yay |
| 04:34.30 | rene1 | i would lose permanently logged in agents |
| 04:34.47 | rene1 | but could skip the login process (obviate with the login from the rails app) |
| 04:35.04 | rene1 | permanently -> agentlogin vs agentlogincallback |
| 04:37.03 | rene1 | i could try with that |
| 04:41.01 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 04:50.44 | *** join/#asterisk madlynx (n=sleever@195.206.63.10) |
| 04:50.51 | *** part/#asterisk madlynx (n=sleever@195.206.63.10) |
| 04:51.33 | rene1 | brook and crashed it did |
| 04:51.34 | rene1 | 2.03, 1.52, 0.75 |
| 04:51.40 | rene1 | are those bad? |
| 04:51.53 | hads|home | rene1: That is high |
| 04:52.18 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 04:52.20 | *** part/#asterisk tlow (n=tlowe@bgp.terrorist.net) |
| 04:52.42 | twisted | that's real high |
| 04:53.00 | rene1 | ahh iam seeing a 98% cpu usage on the proxy |
| 04:53.06 | twisted | wheeee |
| 04:53.06 | rene1 | not in the script that uses it |
| 04:53.15 | twisted | the proxy looks like it's fuxored |
| 04:53.27 | rene1 | i am not much of a network programmer |
| 04:53.50 | rene1 | but the proxy did seem to skip a lot of asterisk events |
| 04:53.54 | rene1 | and in general was slow |
| 04:54.03 | brookshire | rene1: how's sound quality now? |
| 04:54.20 | rene1 | asterisk is long dead |
| 04:54.38 | rene1 | <PROTECTED> |
| 04:54.44 | rene1 | this is real bad |
| 04:54.58 | twisted | dur |
| 04:55.13 | rene1 | i have two insances of the proxy, one serves my ruby script and the other servers my web app |
| 04:55.38 | rene1 | at least i know now that i dont need to reinstall asterisk nor call digium |
| 04:56.19 | rene1 | i could use ajam and forget about using a proxy at all |
| 04:57.03 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 04:58.39 | brookshire | ajam just does manager stuff, nothing more than original manager |
| 04:59.20 | hads|home | rene1: Basic load average overview - http://www.wlug.org.nz/LoadAverage |
| 05:00.27 | twisted | haha |
| 05:00.44 | brookshire | twisted: irony police |
| 05:00.48 | twisted | you ran? |
| 05:00.50 | twisted | s/ran/rang |
| 05:00.58 | brookshire | is that like ren? |
| 05:01.04 | twisted | no |
| 05:01.12 | twisted | it's not like ringer equivalency number |
| 05:01.16 | brookshire | how many nubs can *YOU* ring? |
| 05:01.24 | brookshire | nen |
| 05:01.33 | twisted | rofl |
| 05:01.34 | rene1 | brookshire: |
| 05:01.38 | twisted | sounds like a new metric at digium |
| 05:01.51 | rene1 | ajam would probably wont crash at me as much as rami |
| 05:02.58 | rene1 | rami is the proxy i am using |
| 05:02.58 | twisted | RAMI in the ARSE |
| 05:02.58 | brookshire | i think you can separate the proxy from the actually asterisk box |
| 05:02.58 | twisted | you *SHOULD* be able to |
| 05:02.58 | rene1 | was it good for you twisted? |
| 05:02.59 | brookshire | i would probably try that first |
| 05:03.04 | twisted | rene1, no... it's load average was way to high |
| 05:03.33 | rene1 | the proxy should be running on another box |
| 05:03.46 | rene1 | than asterisk |
| 05:05.07 | brookshire | yes.. nothing but asterisk should be running on the asterisk box ;) |
| 05:05.20 | *** join/#asterisk DeVilSouLBlacK (n=chatzill@gea-gye-internet.telconet.net) |
| 05:05.50 | rene1 | twisted: what language/proxy are you using for ami stuff |
| 05:05.57 | brookshire | asterisk is just ultrasensitive to cpu |
| 05:06.05 | russellb | i386 assembly |
| 05:06.10 | Mattwj2005 | anyone know anything about e164.org? |
| 05:06.18 | linagee | who is strom carlson? StromC? |
| 05:06.28 | brookshire | linagee: brandon :) |
| 05:06.39 | linagee | brookshire: ? |
| 05:06.44 | twisted | rene1, we've written our own |
| 05:06.45 | brookshire | who is |
| 05:06.47 | brookshire | haha |
| 05:07.38 | rene1 | twisted: i wont give up ruby because of this, the dialer script still goes strong without signs of being leaking memory and shit, it is the proxy that is crap |
| 05:07.56 | *** join/#asterisk zonalnet (n=chickens@ip68-96-233-180.lv.lv.cox.net) |
| 05:08.01 | rene1 | i wonder should i pull all the asterisk events into a mysql table? |
| 05:08.05 | russellb | telemarketers aren't meant to have it easy |
| 05:08.21 | hads|home | hehe |
| 05:08.28 | twisted | rene1, i never said give up on ruby |
| 05:08.36 | brookshire | i did! |
| 05:08.37 | twisted | rene1, mysql will explode |
| 05:08.40 | brookshire | and you should :) |
| 05:08.41 | twisted | asterisk generates TONS of events |
| 05:08.42 | Mattwj2005 | can you associate a server e164.org number with a no-ip.org entry? |
| 05:08.50 | rene1 | brookshire: and use what |
| 05:09.07 | brookshire | i'm all for anything but python and ruby |
| 05:09.08 | rene1 | twisted: i am basically only monitoring for agent[connect|complete] |
| 05:09.13 | hads|home | scripting languages shouldn't leak. |
| 05:09.22 | twisted | rene1, well, as long as you filter |
| 05:09.31 | twisted | and can assure that mysql is constantly up to date |
| 05:09.33 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 05:09.44 | Mattwj2005 | back |
| 05:09.46 | twisted | but that's really an extreme way of doing it |
| 05:10.07 | rene1 | i have only this weekend |
| 05:10.16 | rene1 | i was sent to another country to get this crap working |
| 05:10.19 | twisted | oooh nice |
| 05:10.23 | twisted | did you look at astmanproxy? |
| 05:10.38 | rene1 | i did, but it didnt looked as rubish as the rami thing |
| 05:10.51 | rene1 | ruby-like i meant |
| 05:10.54 | brookshire | we use astmanproxy |
| 05:11.01 | Qwell | astmanproxy rocks... |
| 05:11.04 | twisted | astmanproxy is quite stable |
| 05:11.05 | Qwell | rene1: It isn't to replace rami |
| 05:11.14 | Qwell | It's a layer between rami and asterisk |
| 05:11.18 | rene1 | no no |
| 05:11.25 | rene1 | rami is a layer between asterisk and ruby |
| 05:11.31 | hads|home | yes |
| 05:11.31 | rene1 | astmanproxy would replace rami for me |
| 05:11.35 | hads|home | no |
| 05:11.35 | Qwell | yes, and this is another layer :p |
| 05:11.38 | Qwell | no, not at all |
| 05:11.45 | Qwell | astmanproxy is just a manager proxy... |
| 05:11.49 | twisted | but it's another layer that would keep rami from crashing asterisk :) |
| 05:12.01 | rene1 | that is interesting |
| 05:12.03 | brookshire | i think maybe he's talking about ruby agi |
| 05:12.04 | brookshire | ?? |
| 05:12.06 | twisted | quite possibly |
| 05:12.07 | Qwell | ragi? |
| 05:12.08 | rene1 | i get to keep my code |
| 05:12.10 | rene1 | RAMI |
| 05:12.16 | rene1 | ragi _> ruby agi |
| 05:12.22 | rene1 | rami -> ruby ami |
| 05:12.22 | twisted | ragi sounds like what a girl is on once a month |
| 05:12.26 | Qwell | yeah, look into using astmanproxy |
| 05:12.28 | brookshire | god.. anyways.. he's like the first to ever use ruby for anything with asterisk |
| 05:12.49 | twisted | i think i'm going to use that |
| 05:12.53 | Qwell | brookshire: give me time...I'm gonna get somebody to use asterisk, that will put everything ruby related that currently exists...to shame |
| 05:12.55 | brookshire | i wouldn't even use php for asterisk :) |
| 05:12.55 | twisted | girl being bitch: asshole |
| 05:12.59 | twisted | me: are you on the ragi? |
| 05:13.13 | rene1 | ha |
| 05:13.19 | Qwell | he's like the ruby master... |
| 05:13.35 | tengulre | hi,all |
| 05:13.35 | brookshire | so he would be the rastmaster |
| 05:13.37 | rene1 | you have a ruby wiz friend Qwell? |
| 05:13.42 | Qwell | rene1: yeah, several |
| 05:13.45 | rene1 | cool |
| 05:14.24 | ManxPower | http://www.nop.org/inkling/logo.sys |
| 05:14.27 | Qwell | it's scary what they do with ruby...really |
| 05:14.28 | rene1 | Qwell: so asterisk-astmanproxy-rami-ruby |
| 05:14.31 | tengulre | why my sip phone only ringing one and can not pickup? |
| 05:14.34 | Qwell | rene1: exactly |
| 05:14.53 | rene1 | today has to be my lucky day |
| 05:15.13 | twisted | rene1, the worst that could happen is that it doesn't make any difference at all |
| 05:15.20 | twisted | the best that could happen is it solves the problem |
| 05:15.34 | Qwell | ManxPower: That is so 1996 |
| 05:15.56 | twisted | ManxPower, what's up man? How you been fairing since katrina? |
| 05:16.02 | ManxPower | I never said I'm a trendsetter |
| 05:16.13 | ManxPower | twisted, temp living in Tuscaloosa. |
| 05:16.20 | twisted | oooh, you're in the same state |
| 05:16.24 | ManxPower | Designing and building a CATV system for the campground. |
| 05:16.38 | Qwell | fyi: AA could not have skimped more on their FF cards |
| 05:16.41 | twisted | ManxPower, cool, sounds like fun |
| 05:16.55 | Qwell | this thing might as well say "Hi, my name is ____________" |
| 05:17.07 | twisted | ManxPower, we should get together sometime soon... you're less than 3h away from here |
| 05:17.10 | hads|home | slim shady? |
| 05:17.21 | ManxPower | twisted, You in Huntsville? |
| 05:17.25 | twisted | ManxPower, yep |
| 05:17.39 | ManxPower | I was in Steele, AL (near Rainbow City/Gadsden) until this week. |
| 05:17.45 | twisted | Yeah, that's what I hear |
| 05:17.48 | twisted | but i've been out of the loop |
| 05:17.54 | ManxPower | Heck, I even came up and saw OEJ when he was there once. |
| 05:18.01 | twisted | OMG |
| 05:18.11 | tengulre | Hi,my name is ZhangTengHong |
| 05:18.33 | Qwell | Hi ZhangTengHong! |
| 05:18.34 | twisted | I hereby dub the ZTH |
| 05:18.57 | twisted | s/the/thee |
| 05:19.09 | Qwell | twisted: ...creative |
| 05:19.18 | twisted | Qwell, yeah. blame the BAC |
| 05:19.22 | brookshire | Manx: you did? i soo didn't meet you |
| 05:19.43 | ManxPower | brookshire, I had planned on being up there more often, but have not really needed to. |
| 05:20.48 | twisted | ManxPower, well, you need to next weekend |
| 05:21.54 | twisted | ManxPower, especially if you shoot pool at all ;) |
| 05:22.40 | ManxPower | I have no depth perception |
| 05:22.54 | ManxPower | Anyway, life is pretty stressful at the moment. |
| 05:23.22 | brookshire | what's up? |
| 05:23.22 | ManxPower | I miss the 50 acers, big pool, hottub, etc where I was at in Steele, AL. Oh well, I'll be back there again in a few months. |
| 05:23.23 | twisted | well, you could use it as a means to get away and enjoy yourself for a weekend |
| 05:30.57 | brookshire | TAKE ME HOME! |
| 05:31.00 | twisted | or not, just thought I'd offer to have ya come up and have a good time |
| 05:31.04 | brookshire | YEAHAHH!!@!@# |
| 05:31.16 | twisted | (drinking, pool, etc) |
| 05:39.46 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 05:39.59 | *** join/#asterisk tempest1 (n=asf@adsl-150-198-149.chs.bellsouth.net) |
| 05:41.56 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:43.12 | ManxPower | twisted, perhaps later in the month |
| 05:43.22 | ManxPower | or after I move back to Steele |
| 05:45.00 | twisted | okay |
| 05:53.17 | ManxPower | Has anyone here put more than 2 video cards into a linux box? |
| 05:53.41 | brookshire | yes |
| 05:53.48 | brookshire | you're better off not |
| 05:53.56 | ManxPower | Why? |
| 05:53.57 | brookshire | if you even try, use xorg |
| 05:54.34 | ManxPower | Um, I'm looking at sending the Svideo or RCA out from each card into a modulator, so it's not silly mulpiple screen desktop stuff. |
| 05:56.16 | ManxPower | I was mostly curious about the cards conflicting |
| 05:56.32 | brookshire | manxpower: that's up in the air |
| 05:56.40 | brookshire | sometimes they work beautifully |
| 05:56.43 | ManxPower | Hence my question. |
| 05:56.56 | brookshire | if your curious, google for both of your cards |
| 05:57.02 | brookshire | s/your/you're |
| 05:57.20 | ManxPower | Back in The Old Days, VGA cards used C000-CFFF (I think) so you could not put more than 1 in a system. |
| 05:57.43 | ManxPower | brookshire, I was hoping to use all the same make/model of card. |
| 05:59.34 | *** join/#asterisk dthreatt (n=dthreatt@nv-71-49-135-53.dhcp.embarqhsd.net) |
| 05:59.49 | dthreatt | hello everyone |
| 05:59.56 | dthreatt | :) |
| 06:01.18 | dthreatt | does anyone know if there is a macro writen that read a prompt takes that input stores it in a variable, reads it back....and ask u to press 1 if its good or 2 if it isnt?? |
| 06:01.49 | dthreatt | or a site that has dialplan example other than voip-info.org |
| 06:02.42 | sumasuma | i want to pass arguments from a text file, how would i do that ? |
| 06:03.05 | sumasuma | i mean for Application(arguments) |
| 06:03.20 | sumasuma | Do I have to do that in AGI or possible within dial plan itself ? |
| 06:08.27 | dthreatt | from a text file |
| 06:08.28 | dthreatt | ? |
| 06:08.33 | dthreatt | hmm |
| 06:08.38 | sumasuma | yes |
| 06:08.40 | dthreatt | what do u wanna do with the arguments |
| 06:08.44 | sumasuma | cool |
| 06:08.49 | dthreatt | how do they get in the file? |
| 06:08.52 | sumasuma | i was playing with cepstral |
| 06:09.02 | sumasuma | i will write to the file using php from the web |
| 06:09.06 | dthreatt | ok... |
| 06:09.14 | sumasuma | asterisk need to play that using text to voice |
| 06:09.29 | dthreatt | which text to voice engine |
| 06:09.30 | dthreatt | ? |
| 06:09.35 | dthreatt | festival |
| 06:09.36 | sumasuma | cepstral |
| 06:09.41 | dthreatt | ahhh |
| 06:09.43 | dthreatt | is that better |
| 06:09.45 | dthreatt | than |
| 06:09.53 | dthreatt | festival |
| 06:10.04 | sumasuma | yep |
| 06:10.07 | dthreatt | cool |
| 06:10.09 | dthreatt | i am new |
| 06:10.14 | sumasuma | brilliant and easy to use |
| 06:10.20 | dthreatt | cool |
| 06:10.25 | dthreatt | maybe u can let me hear it |
| 06:10.27 | dthreatt | :) |
| 06:10.36 | sumasuma | you can call singapore number ? |
| 06:10.40 | dthreatt | sure |
| 06:10.53 | dthreatt | or just give me a sip address |
| 06:10.54 | dthreatt | lol |
| 06:11.06 | sumasuma | no sip address here |
| 06:11.11 | dthreatt | ummmm if u are using php |
| 06:11.12 | sumasuma | it is connected out of internet |
| 06:11.16 | dthreatt | o ok |
| 06:11.27 | sumasuma | one second |
| 06:11.41 | dthreatt | i would just use |
| 06:12.09 | sumasuma | +6568299771 |
| 06:12.20 | sumasuma | it will just say how are you! |
| 06:12.22 | dthreatt | the system command |
| 06:12.30 | dthreatt | to run the php file |
| 06:12.56 | dthreatt | but i dont know if u can pass the contents back into asterisk that way |
| 06:13.08 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
| 06:13.14 | sumasuma | it is possible with AGI |
| 06:13.25 | dthreatt | u could do an include in your extensions |
| 06:13.29 | dthreatt | as a variable |
| 06:13.34 | dthreatt | and call a php to do a reload |
| 06:13.39 | dthreatt | that would work |
| 06:13.47 | dthreatt | i never used AGI yet |
| 06:13.58 | sumasuma | thanks for your help |
| 06:13.58 | dthreatt | i olny been messing with asterisk about a month |
| 06:14.06 | sumasuma | i c |
| 06:14.31 | dthreatt | i am callin |
| 06:14.37 | dthreatt | nice |
| 06:14.42 | dthreatt | sound great |
| 06:14.46 | dthreatt | wow |
| 06:14.47 | tengulre | how many at least baudwidth when building asterisk server? |
| 06:15.14 | dthreatt | whats the website for that voice soft? |
| 06:15.34 | sumasuma | how is the quality ? |
| 06:15.42 | dthreatt | excellent |
| 06:15.53 | dthreatt | sounds way better than mine |
| 06:16.09 | dthreatt | festival sounds like a darn talking wrist watch |
| 06:16.22 | dthreatt | speak and spell |
| 06:16.23 | dthreatt | lol |
| 06:22.23 | *** join/#asterisk frodonbash (n=frodon@AMarseille-153-1-84-85.w86-209.abo.wanadoo.fr) |
| 06:24.02 | frodonbash | hello world |
| 06:24.10 | brookshire | hi? |
| 06:24.19 | dthreatt | hi |
| 06:33.24 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 06:40.29 | *** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg) |
| 06:41.34 | yxa | how can I do it in such a way that after someone rings for 2 times, do not answer and hangup and do something else like create a callfile after Hangup? it seems to stop after I issue Hangup |
| 06:48.17 | tengulre | sumasuma: qualify=6000 |
| 06:48.24 | tengulre | is it right? |
| 06:49.20 | sumasuma | tengulre: what is that ? |
| 06:49.25 | tengulre | how many at least baudwidth when building asterisk server? |
| 06:49.39 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 06:49.40 | tengulre | I can not dial out with my sip phone! |
| 06:50.22 | tengulre | sometimes the asterisk notice 'No route to destination'! I don't konw why? |
| 06:50.40 | sumasuma | you have vnc ? |
| 06:50.43 | sumasuma | or ssh ? |
| 06:50.58 | tengulre | yes |
| 06:51.00 | tengulre | ssh |
| 06:51.23 | sumasuma | share the info in private, i will just have a look |
| 06:52.26 | tengulre | 61.152.252.29 |
| 06:53.01 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
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| 07:50.56 | *** part/#asterisk tlow (n=tlowe@cypher.punk.net) |
| 07:51.47 | *** join/#asterisk amer (n=ame@fiber-gw3-stc-100.brain.net.pk) |
| 07:53.40 | amer | Hi! Can I restrict the number of simultaneous calls from a SIP peer? |
| 07:54.56 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 07:55.24 | UlbabraB | amer: http://www.voip-info.org/wiki/view/Asterisk+cmd+CheckGroup |
| 07:55.39 | amer | thanks man |
| 07:56.01 | UlbabraB | you're welcome |
| 07:59.14 | tasat- | Hi... anyone know how call forwarding is handled by the carriers? specifically, I believe they charge the user who does the forwarding to the new destination -- does the new destination receive any indication who the call was forwarded by? or only the originating number? |
| 08:00.19 | *** join/#asterisk docelmo (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 08:03.15 | *** join/#asterisk xxoxxx (n=xxoxxx@70.91.242.21) |
| 08:11.23 | *** join/#asterisk [LucaS] (n=lucas@217-133-99-153.b2b.tiscali.it) |
| 08:11.35 | [LucaS] | Ola! |
| 08:15.16 | *** join/#asterisk mitcheloc (n=mitchelo@205.134.225.18) |
| 08:16.35 | [LucaS] | Hi, how stop asterisk? just kill it? |
| 08:16.54 | hads|home | 'stop now' from the CLI |
| 08:17.08 | [LucaS] | hads|home: thx! |
| 08:17.14 | hads|home | np |
| 08:18.30 | [LucaS] | hads|home: sorry.. and to reload configurations without restart it? |
| 08:18.56 | hads|home | reload |
| 08:19.28 | [LucaS] | hads|home: I've found help command!!! ;) |
| 08:19.41 | hads|home | That's a good place to start :) |
| 08:20.32 | hads|home | [LucaS]: THe CLI also has tab completion which will help you out |
| 08:21.51 | Un1x | :O |
| 08:22.41 | Un1x | hey hads|home when you get into cli via, `rasterisk r` command how can you exit ive tried the regular Exit and Quit, but nor work ofcourse without the caps :) |
| 08:23.09 | hads|home | erm... quit |
| 08:24.16 | tengulre | !ask |
| 08:24.35 | Un1x | i tried.. |
| 08:25.03 | *** join/#asterisk ^cLuBk|d^ (n=clubkid@ACCD01BC.ipt.aol.com) |
| 08:25.30 | ^cLuBk|d^ | asterisk can be run also in FreeBSDE ? |
| 08:25.30 | ^cLuBk|d^ | asterisk can be run also in FreeBSD ? |
| 08:26.03 | *** join/#asterisk UlbabraB (n=filippo@host241-43-static.72-81-b.business.telecomitalia.it) |
| 08:26.17 | Un1x | how do you run something IN FreeBSD? |
| 08:27.10 | ^cLuBk|d^ | i mean is it compatible in freebsd Os ? |
| 08:27.18 | Un1x | :| |
| 08:27.29 | Un1x | how is it `in` |
| 08:27.47 | Un1x | its not inside, the Operating system is now... |
| 08:27.53 | Un1x | it runs On the operating system... |
| 08:28.52 | Un1x | yes, it runs on FreeBSD perfectly fine however, you wont be able to have a great QoS, due to...Zaptel drivers being in Beta for Digium cards, thus recommend Linux, anywho... yes you could... |
| 08:29.08 | Un1x | !freebsdbug |
| 08:29.09 | Un1x | !freebsd |
| 08:29.11 | Un1x | etrr |
| 08:29.14 | Un1x | ~freebsd |
| 08:29.18 | jbot | A stable secure open source operating system.. URL: http://www.freebsd.org/ FreeBSD: Nothing runs like a daemon with a pitch fork. |
| 08:29.18 | ^cLuBk|d^ | oic |
| 08:29.25 | Un1x | ~freebsdasterisk |
| 08:29.38 | Un1x | bah, we should really get to adding that :) |
| 08:29.41 | ^cLuBk|d^ | what about in FC5 ? |
| 08:29.57 | Un1x | Yes, sure... but why not go with CentOS... |
| 08:30.13 | ^cLuBk|d^ | im not familiar with CentOS |
| 08:30.22 | ^cLuBk|d^ | coz im planning to setup asterisk on a callcenter |
| 08:30.36 | Un1x | Redhat develops both Centos and Redhat i beleive not too sure on centos :p |
| 08:30.41 | Un1x | but yes, same architecture, etc.. |
| 08:31.00 | ^cLuBk|d^ | i think ill go with fc5 |
| 08:31.06 | ^cLuBk|d^ | im more familiar with it |
| 08:31.11 | ^cLuBk|d^ | its stil d same? |
| 08:31.15 | Un1x | yes... |
| 08:31.20 | ^cLuBk|d^ | thank you |
| 08:31.21 | ^cLuBk|d^ | :) |
| 08:31.22 | Un1x | but depends on wich card youre using as well... |
| 08:31.37 | ^cLuBk|d^ | how about zaptel for |
| 08:31.42 | ^cLuBk|d^ | fc5? |
| 08:31.55 | Un1x | it's good :) |
| 08:32.25 | ^cLuBk|d^ | nice |
| 08:34.05 | ^cLuBk|d^ | in your own opinion |
| 08:34.22 | ^cLuBk|d^ | is it advisable that you use xwindows for the servers? |
| 08:35.21 | hads|home | ^cLuBk|d^: You don't want to run anything except Asterisk on an Asterisk server. Especially X |
| 08:35.48 | ^cLuBk|d^ | oic |
| 08:35.53 | ^cLuBk|d^ | asterisk only? no more no less? |
| 08:43.33 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 08:44.10 | *** join/#asterisk Skymarshal (n=Skymarsc@pD9E1484C.dip0.t-ipconnect.de) |
| 08:46.24 | Skymarshal | Is it possible to use a call file to trigger a special extension? I'd like to start something every hour and though of solving this by copying a call file which connects to a special extension. But I have to dial out to something real and that I don't have. Can I use a call file to just call one extensions and not to open a channel? |
| 08:50.30 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:52.33 | hads|home | Skymarshal: I don't really know what you are asking? WHat do you mean by 'special extension' |
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| 08:54.09 | Un1x | dlynes :p |
| 08:54.18 | Skymarshal | hadslhome: Lets assume I'd like to generate a wake up call. The times for the wake up calls are stored in an Asterisk database. But I have to have a trigger which starts an extension every minute. Than the extension can check in the database if there is a wake up call to be done. Got the idea and the problem? |
| 08:56.26 | hads|home | Hmm... sort of. You could probably use the Local channel. TBH it sounds like a strange way to solve the problem. |
| 08:57.53 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.12) |
| 08:58.10 | Skymarshal | hdslhome: I know that I could solve this problem easier with e.g. the API but it is for documentation purposes (what can one do with call files). |
| 08:58.12 | Skymarshal | What do you mean with local channel? Can you give me an example? |
| 08:59.49 | hads|home | OK sure. You can Dial(Local/123@context) which you could use in a call file too. |
| 09:00.49 | hads|home | http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels |
| 09:01.19 | Skymarshal | thanks. I'll try that. |
| 09:01.24 | hads|home | np |
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| 09:05.31 | *** mode/#asterisk [+o twisted[asteria]] by ChanServ |
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| 09:44.00 | zebulunz | bonjour |
| 09:44.51 | Taker | Hi! I compiled zaptel-1.2.7. How to get it working with my asterisk? Do I have to edit the /etc/asterisk/modules.conf file? |
| 09:45.00 | Taker | hello zebulunz |
| 09:53.19 | zebulunz | hello taker |
| 09:54.24 | hads|home | Taker: WHat are you tring to 'get working'? |
| 09:55.41 | Taker | My ISDN-Card with HFC-Chip. |
| 09:56.23 | hads|home | Ah, sorry I don' tknow much about BRIstuff |
| 09:56.35 | Taker | hads|home: I don't use Bristuff |
| 09:57.21 | hads|home | Then what are you doing with zaptel? |
| 09:59.01 | Taker | I just downloaded this tarball: http://ftp.digium.com/pub/zaptel/zaptel-1.2.7.tar.gz - First I used asterisk just with Softphones, but I want to connect an ISDN-Phone. |
| 09:59.41 | Taker | Doesn't it work this way? |
| 10:00.54 | hads|home | Well zaptel is only for Digium (and Sangoma) cards and BRIstuff. Also ztdummy for timing. |
| 10:02.30 | Taker | Well, I have to use Bristuff? |
| 10:03.21 | hads|home | Like I said I don't know much about BRIstuff. You should have got some docs with the card, or there will be some on the wiki or somthing |
| 10:12.33 | zebulunz | c koi un LFS? |
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| 10:13.10 | hads|home | zebulunz: ? |
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| 10:30.58 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 10:41.06 | zebulunz | qui peut me livrer 1 lien vers 1 doc en francais qui traite asterisk |
| 10:41.09 | zebulunz | svp |
| 10:43.17 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 10:44.12 | hads|home | If you are talking about Asterisk docs in French I don't know of any. |
| 10:44.41 | zebulunz | yes that's what i'm talking about |
| 10:45.06 | zebulunz | cause i search really so much with no result |
| 10:45.38 | zebulunz | so i'm trying to find some help |
| 10:45.54 | hads|home | Sorry, I don't know of any either. |
| 10:46.58 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
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| 10:59.18 | zebulunz | thk's anyway |
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| 11:10.18 | xheliox | Quiet this morning. |
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| 11:21.05 | jeanmi_ | I have set MP3Player(foo.mp3) for when someone calls me |
| 11:21.14 | jeanmi_ | but the sounds is extremely loud .... |
| 11:21.22 | jeanmi_ | is there anything I can do about that ? |
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| 12:02.02 | jeanmi_ | when I see something like |
| 12:02.04 | jeanmi_ | <PROTECTED> |
| 12:02.16 | jeanmi_ | does that mean that something went wrong ? |
| 12:02.28 | jeanmi_ | because the call seemed to be OK |
| 12:07.19 | xheliox | jeanmi_: Nope. |
| 12:07.33 | *** join/#asterisk NoRemorse (n=bah@203-214-13-168.dyn.iinet.net.au) |
| 12:07.36 | NoRemorse | hi all |
| 12:07.56 | NoRemorse | how do I get realtime extensions to take priority over static ones? |
| 12:08.34 | NoRemorse | I have number configured in mysql, but when i dial them , a _X. overides them |
| 12:09.08 | NoRemorse | ah silly me |
| 12:09.17 | xheliox | lol |
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| 12:53.01 | xheliox | Does the IAXy not support transfer? |
| 12:54.13 | xheliox | IAX transfer, I mean, reinvite, if you will... |
| 12:58.32 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 13:02.12 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 13:06.00 | queuetue | file: Are you around? |
| 13:12.04 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 13:12.07 | PakiPenguin | hello everyone |
| 13:16.15 | asterisk-dud | i have two incoming lines, one voip, and one pstn, asterisk always answers the voip but all of a sudden stopped answering the pstn, can anyone please help me |
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| 14:02.13 | effectiveape | can anyone stop me jumping off a cliff in trying to work out transfer on my cisco phones! :/ |
| 14:02.21 | Maroder | can somebody help me with asterisk + oh323 ? |
| 14:02.55 | PakiPenguin | Maroder i can :D |
| 14:03.05 | Maroder | PakiPenguin thats good ;) |
| 14:03.36 | PakiPenguin | Maroder whats the issue |
| 14:04.09 | Maroder | PakiPenguin i read from http://www.voip-info.org/wiki/view/Asterisk+config+h323.conf |
| 14:05.02 | Maroder | i have the following situation |
| 14:05.11 | PakiPenguin | yes? |
| 14:05.16 | Maroder | asterisk and gnugk |
| 14:05.34 | Maroder | for the moment gnugk is the main gk |
| 14:05.56 | Maroder | but i want the calls from gnugk to forward to asterisk |
| 14:06.46 | Maroder | and cals from asterisk to forward to gnugk ... |
| 14:07.41 | Maroder | but i can't understand the ideology |
| 14:08.02 | PakiPenguin | on phone , holdon |
| 14:08.21 | Maroder | okay |
| 14:10.23 | viperdude | hi anyone know why playing the moh natively in * instead of using mpg123 would push the load averages up on a box? |
| 14:18.08 | effectiveape | someone must be able to help me - i just kep getting transfer failed :/ |
| 14:18.55 | viperdude | effectiveape: WHATS UP@ |
| 14:19.06 | viperdude | whats up even lol |
| 14:19.58 | effectiveape | viperdude: hehe. When i call in from Zap to Sip and then attempt to use the cisco transfer softkey to transfer it to another Sip extension i just get transfer failed on the first SIP extension |
| 14:20.28 | viperdude | do you have options t and T in the dial command to call the cisco phone? |
| 14:20.34 | effectiveape | viperdude: however if i call sip to another sip then transfer it to another sip it works finr |
| 14:20.54 | effectiveape | yeah but i thought those only applied to *2 features |
| 14:21.14 | viperdude | you need the for all transfer capability AFAIK |
| 14:21.29 | viperdude | we have cisco 7940's here and we need those options |
| 14:21.46 | effectiveape | aha cool. I think i have them in all.... Just checking |
| 14:23.23 | effectiveape | viperdude: here is my ext.conf anything look wrong? http://pastebin.ca/116892 |
| 14:25.14 | *** join/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 14:25.30 | viperdude | effectiveape: look at [inbound-from-isdn] i assume this is the context your ZAP calls come in on |
| 14:25.37 | effectiveape | yeah |
| 14:25.39 | viperdude | there is no t or T in the dial commansd |
| 14:26.04 | effectiveape | i noticed that - just added it in. Still nothing |
| 14:26.46 | viperdude | there is also no t or T in macro-stfPrivacyexten |
| 14:26.57 | viperdude | macro-stdPrivacyexten |
| 14:27.03 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 14:27.11 | effectiveape | yeah not using that |
| 14:27.23 | viperdude | ok so still no transfer from pstn incoming? |
| 14:27.31 | effectiveape | nope |
| 14:27.51 | viperdude | if you crank the verbose up what does the CLI show when try to transfer the zap call? |
| 14:28.15 | effectiveape | what command is it to do that? |
| 14:28.22 | viperdude | set verbose 99 |
| 14:28.41 | viperdude | btw can u repost your extensions.conf now you have made the changes? |
| 14:31.43 | effectiveape | viperdude: new conf - http://pastebin.ca/116899 output from cli - http://pastebin.ca/116901 |
| 14:32.04 | effectiveape | the other weird thing is that the zip channel doesn't get music on hold which might be related? |
| 14:34.14 | *** join/#asterisk jtar (n=john@cpc3-mapp3-0-0-cust366.nott.cable.ntl.com) |
| 14:34.38 | viperdude | just a thought, i know the dial command should answer the call when it connects but try answering the call before dialling the extension |
| 14:34.49 | viperdude | see if that gives MOH? |
| 14:35.10 | effectiveape | what do you mean? |
| 14:36.43 | viperdude | in your example u answered the incoming call on exten 12, correct/ |
| 14:36.53 | effectiveape | yep |
| 14:37.11 | viperdude | and what extension did u try to transfer to? 16? |
| 14:37.30 | effectiveape | yep |
| 14:37.50 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
| 14:38.23 | viperdude | you sure not the other way round? |
| 14:39.19 | effectiveape | Nope called in and answered 12 then pressed transfer on 12 to dial 16 - answered 16 then pressed transfer on 12 again which is where i get called failed |
| 14:39.21 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:40.06 | viperdude | err but in your CLI post 16 answers 12 |
| 14:40.11 | viperdude | oops |
| 14:40.22 | viperdude | sorry 12 answers 16 |
| 14:41.16 | effectiveape | oh sorry might have answered that one the other way round - currently all exts ring |
| 14:41.44 | *** part/#asterisk viperdude (n=jon@195.74.96.117) |
| 14:53.04 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 15:00.19 | queuetue | I have a digium TDM400P board in a machine with two FXO cards in it. zap show status sees it, and zap show channels shows three entries - pseudo, 3 and 4. 4 has a local PSTN line plugged into it, and when I call that line, nothing happens on the console, and the line just rings. Is there some important step I've missed? |
| 15:00.38 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 15:03.08 | effectiveape | ok this is just mental! |
| 15:04.06 | effectiveape | It's completely different when i hit *2 instead of transfer on the phone. When i use *2 i get the music on hold in the external call but the call doesn't transfer the calling party just get's continuous music on hold |
| 15:21.06 | clyrrad | Anyone know why the monitor folder keeps having calls recorded in there? Is there a way to disable this? |
| 15:23.01 | clyrrad | I thought you had to say monitor=yes to enable this feature - but its doing it automatically |
| 15:24.20 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 15:24.23 | *** join/#asterisk xxoxxx (n=xxoxxx@70.91.242.21) |
| 15:25.53 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 15:31.45 | jeanmi_ | when I use MP3Player(somefile.mp3) it is extremely loud and pretty much inaudible ..... |
| 15:31.51 | jeanmi_ | is there anything I can do about that ? |
| 15:32.42 | queuetue | Lower the volume of the mp3? |
| 15:34.48 | puppet | i live aig-php |
| 15:36.23 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 15:38.19 | *** join/#asterisk constfilin (n=cf@204.147.182.37) |
| 15:41.30 | clyrrad | anyone farmiliar with how asterisk records calls that come in from the queues? |
| 15:41.45 | queuetue | What does "Unable to create channel of type 'Zap' (cause 0 - Unknown)" indicate? |
| 15:42.11 | RoyK | queuetue: usually a configuration fsckup of some kind |
| 15:44.20 | queuetue | RoyK: Any suggestions where i start to look? |
| 15:45.06 | RoyK | doublecheck zaptel config first |
| 15:45.09 | RoyK | ztcfg -vvvvv |
| 15:45.46 | queuetue | Channel 03: FXS Kewlstart (Default) (Slaves: 03) \ Channel 04: FXS Kewlstart (Default) (Slaves: 04) ... looks right. |
| 15:46.03 | RoyK | *shrug* |
| 15:46.12 | RoyK | pot is nice, though |
| 15:49.24 | *** join/#asterisk adorah (n=Administ@87.68.156.188.cable.012.net.il) |
| 15:50.38 | ManxPower | "Pot is proof that God loves us." |
| 15:51.16 | queuetue | And also that the government does not have your best interests in mind. |
| 15:51.48 | constfilin | greetings |
| 15:53.12 | constfilin | there was a discussion on the mailing list about optimizing RTP traffic by changing iptables rules so that RTP packet routing is happening entirely in the kernel, without user more. I wonder if anyone has tried to do that. |
| 15:53.51 | RoyK | http://www.fishtank.org.uk/humour/pictures/Computer/Microsoft%20Bug%20List.gif |
| 15:56.50 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
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| 16:23.50 | charles___ | constfilin: routing is done at the kernel level |
| 16:24.01 | charles___ | constfilin: user level generates but all routing on the kernel already |
| 16:27.05 | *** join/#asterisk XeSolutions (n=Steve@68.225.80.195) |
| 16:27.35 | XeSolutions | can someone help me whats the default /etc/asterisk/manager.conf suppose to have in it as I accidently deleted this file. |
| 16:29.14 | queuetue | XeSolutions: [general] \enabled = yes \ port = 5038 \ bindaddr = 127.0.0.1 |
| 16:29.46 | queuetue | And then #include "manager.d/*.conf" |
| 16:30.07 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 16:32.15 | XeSolutions | hmm thats what I put in there... any idea why this clicktocall script I put in isnt working? |
| 16:33.18 | XeSolutions | whats suppose to be in manager_custom.conf ? |
| 16:35.07 | Qwell | XeSolutions: #freepbx |
| 16:35.44 | XeSolutions | Qwell been there no help |
| 16:37.06 | RoyK | XeSolutions: the question you asked is not an asterisk question..... so wrong channel |
| 16:37.08 | Qwell | XeSolutions: read the channel topic |
| 16:38.43 | charles___ | XeSolutions: why don't you just download it again ? |
| 16:40.30 | XeSolutions | Qwell I did read the Channel Topic Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- |
| 16:40.42 | Qwell | XeSolutions: try reading the rest |
| 16:40.51 | XeSolutions | already did read the rest... |
| 16:41.07 | XeSolutions | Qwell and your point is? |
| 16:41.16 | Qwell | that FreePBX isn't supported here... |
| 16:41.29 | XeSolutions | lets see this is 1.2.10 or Zaptel 1.2.7 or FreePBX, AMP, Asterisk@Home, Trixbox |
| 16:41.35 | XeSolutions | this isn't |
| 16:41.42 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 16:41.45 | XeSolutions | this is Asterisk Business |
| 16:41.54 | Qwell | well, hell, call Digium |
| 16:41.57 | Qwell | it comes with support |
| 16:42.17 | XeSolutions | they support 3rd party scripts though? |
| 16:42.26 | Qwell | I do not know |
| 16:42.29 | *** part/#asterisk XeSolutions (n=Steve@68.225.80.195) |
| 16:44.39 | constfilin | RoyK: have you been looking into iptables for RTP routing? |
| 16:46.01 | RoyK | not more than you it seems..... |
| 16:46.22 | RoyK | but I've been thinking about perhaps mangle will be a better approach |
| 16:46.31 | constfilin | well, this makes two of us the most interested around here. |
| 16:46.32 | RoyK | or what do you think? |
| 16:46.35 | bkw__ | btw doing iptables rtp stuff doesn't work very well |
| 16:46.45 | RoyK | bkw_: why? |
| 16:46.56 | bkw__ | we did it last year at cluecon with SER |
| 16:46.57 | constfilin | bkw_: I know it by now. |
| 16:47.04 | bkw__ | and it just didn't work out well |
| 16:47.06 | file | it "depends" if NAT is involved |
| 16:47.10 | bkw__ | yep |
| 16:47.10 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 16:47.15 | bkw__ | that evil NAT |
| 16:47.43 | RoyK | could it be possible to use mangle instead? |
| 16:47.50 | puppet | wewt i just did my first public service; P |
| 16:47.53 | constfilin | Well, if there is no NATed clients then (with SIP anywya) you do re-INVITE and RTP performance is not an asterisk issues. |
| 16:48.15 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 16:48.40 | constfilin | RoyK: with mangle you can change packet flags, TTL, TOS or some such, not NAT them. |
| 16:48.45 | xai | where should I paste my config files for the channel? |
| 16:48.58 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-177-207.bflony.east.verizon.net) |
| 16:49.01 | RoyK | argh. udp is ugly..... |
| 16:49.08 | SuPrSluG | hello all |
| 16:49.52 | SuPrSluG | i'm trying to get call forwarding to work. anyone use this feature? |
| 16:50.06 | constfilin | It is when NATed clients are present, you need to route RTP through PBX. I did mess with iptables, changing their configuration on the flight (through pipe to iptables-restore STDIN), but this didn't work out well. |
| 16:51.03 | RoyK | a possible solution to making asterisk scale better would perhaps be to use reinvites if, but only _if_ client is not behind nat |
| 16:51.34 | constfilin | ... it didn't work because NAT table in iptablessess only packets in the new connections, byt the time when I was making the rules the first RTP packets from the phones have already arrived and the NAT table wads ignoring them. |
| 16:52.50 | constfilin | So I am now thinking baout chaning iptables to reset its conntrack module and make the NAT table work again. |
| 16:53.52 | constfilin | And, Roy, I've tried it with epoll as well to optimize RTP performance. To my tests epoll was performing a notch worse than poll(). |
| 16:56.46 | SuPrSluG | here's my config http://pastebin.ca/117034 . If I manually enter CLI>database put CFIM 2002 (cell number) it works. when I use *72 it says DBget: varname=temp, family=CFIM, key=2002-- DBget: Value not found in database. Any Ideas? |
| 17:05.11 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 17:08.01 | *** part/#asterisk chexum (i=chexum@gateway/tor/x-c4872b55e08a8232) |
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| 17:09.04 | clyrrad | I have a question about context = qoutcon - Where do you actually define that qoutcon context? |
| 17:09.30 | queuetue | clyrrad: extensions.conf, generally. |
| 17:09.45 | clyrrad | It never seems to go there though.... |
| 17:10.14 | RoyK | constfilin: ping |
| 17:13.54 | clyrrad | Yea if your in a Q - and you press a key on the phone it does not jump to qoutcon |
| 17:13.57 | clyrrad | any idea why? |
| 17:14.20 | clyrrad | I have context = qoutcon in queues.conf and [qoutcon] in extensions.conf |
| 17:17.35 | clyrrad | anyone? |
| 17:23.54 | RoyK | emr. wtf? make: *** No rule to make target `menuselect/menuselect_stub.c', needed by `menuselect/menuselect'. Stop. |
| 17:23.59 | RoyK | with trunk |
| 17:24.16 | RoyK | never mind |
| 17:25.44 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-74b680b619529409) |
| 17:29.23 | *** join/#asterisk viperdude (n=viperdud@195.74.96.120) |
| 17:29.34 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:33.57 | constfilin | RoyK: I am here. So what do you think of iptables solution for RTP routing? |
| 17:34.05 | *** join/#asterisk folsson_ (n=filip@h100n2fls33o985.telia.com) |
| 17:35.27 | constfilin | As for SIP, NAT and reinvites - the configuration solution that is currently in chan_sip is perfectly fine for my needs anyway. However there are situations where clients are just NATed. I'd like to explore every possible way to make PBX work weel in this case. |
| 17:37.11 | RoyK | constfilin: the problem with the current implementation is that you can only set canreinvite statically |
| 17:37.21 | constfilin | Hey, anyone, have uyouu ever played with iptables, NATing and stuff like that? |
| 17:37.28 | RoyK | which is bad if you have several thousand users |
| 17:37.51 | RoyK | then you'd rather want to set it dynamically, detecting if user is behind nat |
| 17:37.53 | constfilin | RoyK: I've long since modifief chan_sip to have all the configs in DB and since then it is no longer an issue. |
| 17:38.04 | RoyK | it is indeed. |
| 17:38.11 | RoyK | since still it needs to be set per user |
| 17:38.22 | RoyK | and you don't want to keep updating a list of several thousand users manually |
| 17:38.39 | RoyK | we're also using a database, of course. |
| 17:38.45 | constfilin | Besides - you are right, you can change chan_sip to detect NAT situation internally and route accrodingly. This is not a problem/. |
| 17:39.07 | RoyK | I think it detects nat already |
| 17:39.28 | RoyK | but I'm unsure of where to set the reinvitet flag. chan_sip isn't really documented too well |
| 17:39.44 | constfilin | RoyK: it may very well be. I am talking about a scenario when the client is NATed. |
| 17:40.06 | denon | RoyK: you talking about Reinvite=yes in sip.conf? |
| 17:40.11 | constfilin | RoyK: use the FORCE, read the SOURCE. |
| 17:41.17 | constfilin | Yes, I think, it is about chaning reinvite flag in sip_user or sip_peer structure. Although I haven't been looking into SVN chan_sip code for more than a year. |
| 17:41.18 | RoyK | force what? |
| 17:41.22 | RoyK | what part of the source? |
| 17:41.26 | RoyK | 1.2 or trunk? |
| 17:41.43 | RoyK | denon: yes |
| 17:41.48 | RoyK | denon: or in realtime, per user |
| 17:41.53 | denon | oh, in realtime |
| 17:41.58 | RoyK | doesn't matter |
| 17:42.15 | constfilin | either. Look into sip_user or sip_peer structure. There is a flag about whether it can make re-invites. Just change this flag depending on the result of NAT detection and you are done. |
| 17:42.19 | RoyK | denon: i want this set dynamically based on wheather the user is behind nat or not |
| 17:42.33 | RoyK | constfilin: do you know where nat detection is done? |
| 17:42.39 | denon | cant you just turn it on .. and it'll fail if they arent reinvite-capable? |
| 17:42.52 | constfilin | RoyK: I haven't been looking into SVN chan_sip code for more than a |
| 17:42.56 | jbalcomb | wow, my backup server is pushing 12MB/s for 4 hours every night |
| 17:43.10 | ManxPower | Um, I don't think Asterisk has any NAT detection. |
| 17:43.11 | constfilin | year. I use my own chan_sip, heavily customized. |
| 17:43.23 | constfilin | BRB |
| 17:43.24 | RoyK | denon: invites work, but audio disappear, and i don't want hundereds of users hitting customer support ever hour |
| 17:43.36 | denon | RoyK: ah, for me they've always just failed |
| 17:43.41 | denon | but I guess I havent done it often |
| 17:43.44 | RoyK | ManxPower: it doesn't? |
| 17:44.24 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 17:44.28 | RoyK | <PROTECTED> |
| 17:44.28 | RoyK | <PROTECTED> |
| 17:44.29 | RoyK | <PROTECTED> |
| 17:44.29 | RoyK | <PROTECTED> |
| 17:44.54 | RoyK | or is that just based on whatever's set in the config? |
| 17:45.01 | ManxPower | RoyK, isn't that just using the value in the config? |
| 17:45.06 | ManxPower | I don't know for sure. |
| 17:45.12 | RoyK | erm. yes |
| 17:45.12 | RoyK | it is |
| 17:45.13 | RoyK | sorry |
| 17:46.13 | RoyK | anyway. looking up the addresses in the SDP and matching them with rfc1918 and rfc3927 networks should give an answer |
| 17:46.27 | RoyK | or even better, to match the address in the sdp with the one connecting |
| 17:47.40 | ManxPower | Unless, of course, the remote side is doing NAT fixups..... |
| 17:47.46 | ManxPower | i.e. STUN, etc |
| 17:49.03 | *** join/#asterisk asteriskbaby (n=undercov@58.65.193.69) |
| 17:49.24 | RoyK | stun won't help if you're doing a reinvite... incoming call *1 -> *2 -> user, then *2 sends a reinvite to *1 upon getting the call, user's nat router doesn't know anything abouot *1s address and ditches the packets |
| 17:49.32 | ManxPower | NAT is a complex thing and RTP just makes it that much more complicated. |
| 17:50.07 | RoyK | so if the client is _not_ behind nat, it would, in most cases, work with reinvites, that is unless the client is behind some paranoid firewall |
| 17:50.25 | ManxPower | RoyK, Yes, but if the client is using STUN, then it knows what it's external IP address is and will prolly put THAT address in the SDP, and so the header IP and SDP IP will be the same so Asterisk will think it's not NOTed |
| 17:50.32 | ManxPower | and not NATed either |
| 17:51.11 | harryvv | What would be the steps needed to create a call back option in the vm ? |
| 17:51.32 | RoyK | ManxPower: quite right... perhaps not use stun then |
| 17:51.34 | harryvv | That would be a nice feature for a customer or for my self. |
| 17:51.41 | ManxPower | harryvv, that should be documented in voicemail.conf.sample |
| 17:51.52 | harryvv | okay |
| 17:52.19 | ManxPower | RoyK, STUN is just 1 example of a type of NAT traversal that would break Asterisk's detection of NAT. Any sort of SIP aware router could do similar things. |
| 17:52.38 | clyrrad | I have context = qoutcon in queues.conf and [qoutcon] in extensions.conf - however when you are in a queue and you press a key it does not jump to the qoutcon in extensions.conf - how can I correct this? |
| 17:52.40 | ManxPower | harryvv, as I understand it you just set the outgoing context for callbacks in voicemail.conf |
| 17:52.50 | denon | harryvv: ; callback=fromvm ; Context to call back from |
| 17:52.50 | denon | <PROTECTED> |
| 17:53.25 | harryvv | the vm would anonounce, "to call this number press 1" after listening to the vm of that party that left the message. |
| 17:53.45 | harryvv | so I would not have to remember the number and then redial it. |
| 17:53.48 | RoyK | ManxPower: sure, but most routers are not sip aware, and most of our clients are not using stun... |
| 17:53.57 | RoyK | so this would help in _most_ cases |
| 17:54.02 | clyrrad | anyone on my queues questions? |
| 17:54.08 | RoyK | instead of bridging rtp when it's not necessary |
| 17:54.10 | *** join/#asterisk saftsack (n=saftsack@p54A7F03F.dip.t-dialin.net) |
| 17:54.15 | ManxPower | RoyK, in that case, why not just disable reinvites? |
| 17:54.23 | RoyK | scalability |
| 17:54.34 | ManxPower | But if most of your clients are behind NAT..... |
| 17:54.43 | RoyK | i'd say 60% or so |
| 17:54.57 | harryvv | ManxPower I have a hardware voip engineer looking for work. He has not used asterisk so will send him here. |
| 17:55.02 | RoyK | currently we're not using reinvites, but traffic is increasing |
| 17:55.06 | ManxPower | Another issue is that nat routers tend to change the source port of packets. |
| 17:55.25 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 17:55.37 | ManxPower | RoyK, Are that many of hour clients REALLY calling other clients? |
| 17:55.45 | harryvv | BTW is there a quick way to see if a clients network has sip/rtp ports blocked by hooking in a pocket port reader into there network? |
| 17:56.22 | ManxPower | harryvv, define "pocket port reader" |
| 17:56.29 | RoyK | ManxPower: nono... that's not a problem. the problem is the middle asterisk server connecting the clients to outer asterisk pstn-gateways |
| 17:56.44 | RoyK | client (sip) * (sip) * (pstn) |
| 17:57.17 | ManxPower | RoyK, Why the middle (sip)? |
| 17:57.22 | RoyK | first * server does authentication and all sorts of things. second ones (several boxes) does sip/pstn |
| 17:57.35 | ManxPower | why not client (sip) * (iax2) PSTN |
| 17:57.55 | harryvv | ManxPower something thats small like what fluke makes. |
| 17:57.57 | ManxPower | trunk the two asterisks |
| 17:58.10 | RoyK | because with sip/iax2 bridging timestamps are lost, and there's nothing such as a jitterbuffer for sip/iax2 |
| 17:58.11 | RoyK | yhet |
| 17:58.12 | RoyK | yet |
| 17:58.14 | ManxPower | harryvv, All I've heard of are packet sniffers |
| 17:58.34 | RoyK | but with sip/sip and keeping the original rtp timestamps from the client, dejittering at the sip/pstn gateway, this works well |
| 17:58.43 | ManxPower | RoyK, Ah, yes. lack f SIP jitterbuffer is why we don't support off-WAN phones. |
| 17:58.56 | RoyK | asterisk-backports.org |
| 17:59.02 | RoyK | check out the jb there |
| 17:59.12 | RoyK | we've been running on that for months now |
| 17:59.15 | RoyK | without problems |
| 17:59.20 | ManxPower | RoyK, I'll wait for something that's well tested and well known. |
| 17:59.20 | *** join/#asterisk jarg (n=jarg@dsl-201-129-100-218.prod-infinitum.com.mx) |
| 17:59.34 | ManxPower | and supported. |
| 17:59.45 | RoyK | ManxPower: we have a rather large site... so take my word for it, it works |
| 17:59.52 | RoyK | or test it |
| 18:00.20 | *** join/#asterisk s0lid (n=jlq@210.213.127.98) |
| 18:00.29 | ManxPower | test it? HAHAHAHA! Maybe when the client is more willing to pay for even MORE R&D |
| 18:00.40 | RoyK | hehe |
| 18:00.40 | RoyK | ok |
| 18:00.47 | RoyK | anyway. works for me (tm) |
| 18:00.55 | ManxPower | We have to get QoS working on the WAN first. |
| 18:00.55 | RoyK | heavily loaded servers too |
| 18:01.10 | RoyK | you don't really need that much QoS with that jb |
| 18:01.16 | RoyK | it's got a rather good PLC |
| 18:01.16 | harryvv | fluke makes really sweet and advanced network analysers |
| 18:01.29 | ManxPower | RoyK, on a 384k frame relay network with 0 CIR you do. |
| 18:01.38 | RoyK | hehe |
| 18:01.38 | RoyK | ok |
| 18:01.47 | RoyK | still, the PLC is very handy on lossy links |
| 18:01.51 | denon | 0 CIR? |
| 18:02.06 | denon | so the contract reads "We may provide you service, if we have extra bandwidth available"? |
| 18:02.07 | ManxPower | We are slowly upgrading the speeds on the Frame, and moving the high traffic users off the Frame. |
| 18:02.34 | ManxPower | denon, correct. But we have not had issues with this carrier. |
| 18:02.34 | denon | we used to have some frame T1s, they had a 768 CIR, but I swore I got 110% out of them 24/7 |
| 18:02.37 | RoyK | i thought frame relay was a 1990 thing |
| 18:02.39 | harryvv | ManxPower you mean QOs on the routers right? |
| 18:02.49 | RoyK | or ninetees-thing |
| 18:02.50 | ManxPower | harryvv, correct. |
| 18:02.55 | denon | RoyK: nah, frame is still pretty cool |
| 18:03.07 | denon | it's like a multi-site VPN, but RELIABLE, when done right |
| 18:03.11 | harryvv | Cisco routers and you need to update the cards |
| 18:03.16 | ManxPower | We can get a 384k frame relay for $350/month or we can go with a T-1 for $2,000/month |
| 18:03.24 | harryvv | tftp the flash |
| 18:03.27 | RoyK | denon: i know... but it's not really that fast |
| 18:03.36 | harryvv | its been years since cisco for me :) |
| 18:03.37 | denon | RoyK: you can do full t1 frame relay |
| 18:03.42 | ManxPower | harryvv, I've been doing basic Cisco router management for 10 years. |
| 18:03.43 | RoyK | sure |
| 18:03.44 | denon | RoyK: 1.544 is enough for most offices |
| 18:03.56 | harryvv | I see |
| 18:04.00 | denon | RoyK: our regional frame cloud was fed by an oc48 :) |
| 18:04.07 | RoyK | oc48? |
| 18:04.10 | ManxPower | Hell, the nearest CO with IXCs in it is 40 miles away from the corporate HQ |
| 18:04.13 | denon | si senior |
| 18:04.16 | denon | its not all peer to peer |
| 18:04.24 | denon | you can feed a cloud with a massive pipe |
| 18:04.32 | *** join/#asterisk edorsey (i=edorsey@216-63-192-130.ded.swbell.net) |
| 18:04.58 | ManxPower | denon, at OC48, I would think ATM would be a better choice, unless most of your far ends are T-1 or less speeds. |
| 18:05.08 | RoyK | s/4800/4800bps/ |
| 18:05.41 | denon | ManxPower: many, many T1s and fractional T1s |
| 18:05.46 | RoyK | the 4800bps links were fed into 12 64kbps links into a minicomputer |
| 18:06.27 | ManxPower | After Katrina one of our offices demanded we drop the CLEC we were using and move to BellSouth (the ILEC). We told them the phone bills would go up, but they did not care. They started caring when they got their first phone bill from Bell, 4x what the CLEC was charging. |
| 18:07.20 | ManxPower | The CLEC was giving them free LD to any destination in Louisiana and Mississippi. |
| 18:07.51 | RoyK | erm. oc-48 is - how fast? |
| 18:08.04 | ManxPower | This is good for IT, however, the free calling was the main thing that prevented Asterisk from being installed, because Asterisk would not save them enough money. |
| 18:08.04 | clyrrad | I have context = qoutcon in queues.conf and [qoutcon] in extensions.conf - however when you are in a queue and you press a key it does not jump to the qoutcon in extensions.conf - how can I correct this? |
| 18:08.12 | clyrrad | has nobody done this? |
| 18:08.19 | ManxPower | RoyK, 192Mbps, I think. |
| 18:08.42 | ManxPower | RoyK, I would have to look it up. |
| 18:09.06 | ManxPower | clyrrad, perhaps nobody currently active on the channel uses queues |
| 18:09.09 | RoyK | @ |
| 18:09.10 | RoyK | OC-48 is a network line with transmission speeds of up to 2488.32 Mbit/s (payload: 2405.376 Mbit/s; overhead: 82.944 Mbit/s).@ |
| 18:09.25 | clyrrad | hrm..... |
| 18:09.36 | clyrrad | I cant find any answer on Google either :s |
| 18:09.46 | ManxPower | We use queues, but we don't allow people to get out of the queue. |
| 18:10.12 | ManxPower | clyrrad, google site:lists.digium.com queues context |
| 18:10.21 | clyrrad | queues.conf file has a context= options os that if the caller presses a number on the keypad they can exit the queue |
| 18:10.28 | clyrrad | but this option does not seem to work for me |
| 18:10.37 | denon | RoyK: sure, but its also only $19.95/mo |
| 18:10.46 | ManxPower | clyrrad, have you confirmed that Asterisk is seeing the DTMF? |
| 18:10.47 | denon | you've got an extra 40 grand a month :) |
| 18:11.03 | clyrrad | ManxPower - No - How could I confirm that? |
| 18:11.35 | ManxPower | clyrrad, surning on debugging might help, or editing /etc/asterisk/logger.conf to enable debug messages |
| 18:12.02 | clyrrad | I have my verbose set to 20 on the CLI - nothing reports there... |
| 18:12.55 | ManxPower | verbose != debug |
| 18:13.07 | ManxPower | look in /etc/asterisk/logger.conf |
| 18:13.14 | clyrrad | ok |
| 18:14.34 | clyrrad | there is a ;queue_log = no option - if changed to yes where would it write the log? |
| 18:15.33 | RoyK | cluetrain: it works for me.... |
| 18:15.48 | RoyK | cluetrain: i press 9 and it jumps to [asdf] to extension nine in there |
| 18:16.28 | ManxPower | clyrrad, you want to debug DTMF at this point, not queues. |
| 18:16.30 | RoyK | clyrrad: sorry. that was for you |
| 18:16.41 | clyrrad | RoyK - where is your [asdf]? |
| 18:16.45 | clyrrad | is that in extensions.conf? |
| 18:16.49 | RoyK | yes |
| 18:17.01 | clyrrad | DAMN - I am doing the same thing |
| 18:17.09 | RoyK | pb your config |
| 18:17.10 | RoyK | ~pb |
| 18:17.17 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 18:17.17 | RoyK | brb |
| 18:17.20 | clyrrad | can you paste your queues.conf? |
| 18:18.55 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 18:20.02 | clyrrad | OK.... Here is the config http://pastebin.ca/117124 |
| 18:22.11 | clyrrad | also how do you call Queue? I am doing it like this Queue(sales|tw|||300) |
| 18:26.38 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 18:26.56 | RoyK | clyrrad: try this http://pastebin.ca/117132 |
| 18:27.45 | RoyK | clyrrad: it jumps to queue_out,<whatevertheuserpressed>,1 |
| 18:27.46 | RoyK | not s |
| 18:30.31 | clyrrad | Ah Ha! |
| 18:30.33 | clyrrad | that was it |
| 18:30.42 | clyrrad | it works now - thanks bro :) |
| 18:31.02 | clyrrad | That is not documented anywhere that I could find..... |
| 18:31.28 | RoyK | it is, in queues.conf |
| 18:31.44 | clyrrad | in the samples? |
| 18:31.48 | RoyK | yep |
| 18:31.53 | clyrrad | not in mine..... |
| 18:31.57 | clyrrad | what does yours say? |
| 18:32.04 | RoyK | ; A context may be specified, in which if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this context. |
| 18:32.04 | clyrrad | that it goes to whatever was pressed? |
| 18:32.18 | clyrrad | ahhhhhhhh |
| 18:32.24 | clyrrad | yea makes sense |
| 18:32.36 | clyrrad | cool its working now :) |
| 18:32.43 | clyrrad | thanks alot :) |
| 18:32.59 | RoyK | 'if you do not want to wait any longer, since this might take a few hours, please press nine and we will return your call one day' |
| 18:33.10 | clyrrad | LOL |
| 18:33.25 | *** join/#asterisk frodonbash (n=frodon@AMarseille-153-1-84-237.w86-209.abo.wanadoo.fr) |
| 18:33.49 | RoyK | s/will/may/ |
| 18:34.41 | clyrrad | glad thats done - that was furstrating me - fix was pretty simple |
| 18:35.10 | denon | s/snickers/laughs |
| 18:35.15 | denon | s/snickers/laughs/ |
| 18:35.20 | denon | hah |
| 18:35.27 | RoyK | :) |
| 18:35.37 | denon | blah blah blah blah |
| 18:35.43 | denon | s/blah/doh/g |
| 18:35.49 | denon | oooh, buggy bot :) |
| 18:35.55 | denon | er, nm |
| 18:35.57 | denon | blah blah blah blah |
| 18:36.00 | denon | s/blah/doh/ |
| 18:36.20 | denon | coo' |
| 18:36.30 | charles___ | is that thing really smart ? |
| 18:36.36 | denon | hehe |
| 18:36.37 | charles___ | s/a/Z/g |
| 18:36.59 | RoyK | testing 123 |
| 18:37.26 | RoyK | s/(testing)\s+(.*)/$2 oh no $1/ |
| 18:37.44 | RoyK | ~lart himself |
| 18:38.28 | denon | so it seems to be moderately smart |
| 18:38.29 | denon | hehe |
| 18:38.49 | RoyK | testing 123 |
| 18:38.53 | RoyK | s/.*// |
| 18:39.48 | RoyK | s/denon//g :P |
| 18:40.02 | denon | psha |
| 18:40.42 | denon | what would this channel be like without RoyK? |
| 18:40.44 | denon | perroykfect |
| 18:40.50 | denon | s/royk// |
| 18:40.59 | RoyK | :P |
| 18:41.18 | denon | hehe |
| 18:42.20 | *** join/#asterisk tempest1 (n=asf@adsl-150-198-149.chs.bellsouth.net) |
| 18:42.50 | charles___ | RoyK: what do you know about wine ? |
| 18:42.56 | RoyK | just a little |
| 18:42.58 | charles___ | :P |
| 18:43.11 | charles___ | RoyK: are you a drunk bot ? |
| 18:43.15 | luke-jr_ | do broadvoice devices come locked? |
| 18:43.28 | RoyK | but this a very special wine. they don't make too much of it, and it tastes almost more like a red, still being a white |
| 18:43.49 | charles___ | this thing is making me hungry |
| 18:43.51 | charles___ | BRB |
| 18:46.41 | *** join/#asterisk amer (n=ame@202.163.121.116) |
| 18:48.44 | *** join/#asterisk Mike (n=mike@201.112.50.158) |
| 18:48.56 | Mike | how can i reload my iax2 configs without making a "reload" |
| 18:49.01 | Mike | kind of like sip reload |
| 18:49.03 | amer | hi, What is the best way to limit the number of simultaneous calls from a sip peer? |
| 18:49.16 | Mike | amer, call-limit=20 |
| 18:49.17 | Mike | ? |
| 18:49.18 | harryvv | I was scanning though the sounds.gsm files and came across a please-press-the-spacebar.gsm what is that one for. |
| 18:49.22 | amer | in version 1.2 |
| 18:49.28 | RoyK | Mike: iax2 reload ? |
| 18:49.28 | Mike | amer, yes it works |
| 18:49.35 | Mike | RoyK, it doesnt work |
| 18:49.43 | RoyK | hm |
| 18:49.46 | amer | but is it for peers for friends |
| 18:49.49 | file | reload chan_iax2.so |
| 18:49.51 | RoyK | Mike: but it works with a full reload? |
| 18:49.55 | amer | I mean or friends |
| 18:50.12 | RoyK | file: iirc that should do exactly the same |
| 18:50.26 | file | there once was a time when there was no "iax2 reload" |
| 18:50.34 | denon | not so long ago |
| 18:50.52 | RoyK | added post 1.2? |
| 18:51.11 | denon | wasnt it right around 1.2? maybe in 1.2? |
| 18:52.05 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 18:52.54 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 18:53.31 | russellb | i added it last summer |
| 18:54.04 | RoyK | is it possible to 'reload' zap yet? |
| 18:54.26 | russellb | RoyK: yeah, to some degree |
| 18:54.37 | russellb | reload chan_zap.so reloads everything except signalling (in theory) |
| 18:54.46 | russellb | and in trunk, there is some new command ... zap restart ... |
| 18:54.52 | RoyK | ok |
| 18:55.00 | russellb | i'm not quite quire what that does, but i think it is supposed to redo everything |
| 18:55.17 | RoyK | without breaking connections? |
| 18:55.47 | russellb | RoyK: I doubt it ... |
| 18:56.04 | russellb | how often are you hotswapping zap cards anyway? :) |
| 18:56.10 | RoyK | hehe |
| 18:56.22 | *** join/#asterisk adorah (n=Administ@87.68.159.65.cable.012.net.il) |
| 18:56.23 | denon | maybe with IBM's hardware... :) |
| 18:57.05 | RoyK | still, pstn can't failover a call between PRIs |
| 18:57.21 | amer | Sangoma vs diguim, which one should I go for |
| 18:57.27 | denon | digium |
| 18:57.40 | RoyK | amer: I'd say digium for asterisk, since wanpipe generates more overhead |
| 18:57.45 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 18:57.52 | RoyK | as long as there are no native drivers |
| 18:58.02 | amer | digium cards are very resource hungry |
| 18:58.06 | RoyK | still, i like the sangomas more |
| 18:58.14 | RoyK | amer: not more than sangoma+wanpipe |
| 18:58.37 | amer | latest sangoma cards have echo cancl. on board |
| 18:58.46 | amer | anyways I will try both |
| 18:58.48 | russellb | so do the digium cards ... |
| 18:59.26 | *** join/#asterisk n838901_2 (n=name@24-117-16-198.cpe.cableone.net) |
| 18:59.37 | n838901_2 | hello |
| 19:00.01 | amer | We are planning to deploy a 70 million minutes / month prepaid solution using asterisk |
| 19:00.10 | n838901_2 | having problems compiling asterisk from source |
| 19:00.28 | RoyK | amer: keep in mind that sangoma echocancel doesn't yet work with x86_64 |
| 19:00.49 | amer | but need to choose the right tdm card |
| 19:00.52 | russellb | n838901_2: what's the problem |
| 19:00.59 | amer | or leave tdm stuff to external gateway |
| 19:01.11 | amer | has anyone tried asterisk with Sonus? |
| 19:01.12 | RoyK | tdm? pots? |
| 19:01.23 | denon | amer: I like the digium cards -- if there's an issue, you only need to deal with one company, not "oh, that's an asterisk problem" |
| 19:01.27 | n838901_2 | russellb: sorry, i meant asterisk-addons |
| 19:01.31 | denon | or "oh, that's a sangoma issue |
| 19:01.33 | denon | " |
| 19:01.44 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:01.47 | denon | especially important in a mission-critical environment |
| 19:01.59 | russellb | n838901_2: what's the problem? |
| 19:02.00 | RoyK | denon: sangoma works, no problem with that |
| 19:02.08 | RoyK | denon: we run several boxes with A104s |
| 19:02.14 | amer | thanks denon, will keep this in mind |
| 19:02.15 | denon | RoyK: well, im just saying when some weird quirk comes up |
| 19:02.16 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.150) |
| 19:02.30 | denon | RoyK: and you've got 5 trunks down, no clue why |
| 19:02.34 | n838901_2 | russellb: i would pastebin the output if i could get it to format correctly |
| 19:02.41 | russellb | heh |
| 19:02.44 | RoyK | denon: that hasn't happened yet, so somehow I doubt it will |
| 19:03.03 | amer | we are going for a modular solution where our nortel switch will round robin b/w asterisk servers |
| 19:03.12 | amer | no single point of failure |
| 19:03.18 | russellb | RoyK: and obviously, many thousands of people are running digium cards without problems and would say the same thing. |
| 19:03.26 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.160) |
| 19:04.40 | RoyK | russellb: still, I've had more problems with the digium cards we've been having than with the sangomas, except that the sangomas, that is, wanpipe, eats too much cpu |
| 19:04.42 | amer | I hope asterisk will not let me down as I am single person responsible for introducing asterisk in our company |
| 19:05.03 | denon | amer: lots of high volume asterisk out in the world, I wouldnt be too concerned |
| 19:05.09 | RoyK | russellb: but then again, that was two years old te410p cards, and the firmware there wasn't as good as it should've been |
| 19:05.30 | russellb | RoyK: the firmware got a serious upgrade a year ago |
| 19:05.56 | RoyK | still, you gotta return the cards to get one :P |
| 19:05.58 | denon | amer: just make sure to spend a lot of time testing things before foolishly throwing it into prod. good advice for any solution |
| 19:05.59 | RoyK | but i'm doing that |
| 19:06.17 | RoyK | we have five unusable te410p cards now. they don't work with post 1.0.7 |
| 19:06.21 | charles___ | anyone want some corn in the cob ? |
| 19:06.23 | amer | I am expecting that if I remove the TDM cards asterisk server's calls handling capacity will be quadrupled |
| 19:06.35 | russellb | RoyK: well talk to tech support ... |
| 19:06.51 | RoyK | i have. need to return the cards |
| 19:06.59 | amer | are there any plans of making asterisk IMS compliant? |
| 19:07.12 | RoyK | wtf is IMS? |
| 19:07.34 | charles___ | I hate acronyms |
| 19:07.45 | charles___ | amer: IMS is another STD ? |
| 19:08.08 | amer | its the new standard for soft switches |
| 19:08.10 | amer | :) |
| 19:08.16 | folder | hmmm. VoIP mangling. Anyone read this yet? http://www.newscientisttech.com/article.ns?id=dn9647&feedId=online-news_rss20 |
| 19:08.28 | charles___ | amer: who wrote the STD ? |
| 19:09.03 | amer | IP multimedia subsystem |
| 19:09.34 | amer | everything SIP |
| 19:09.39 | charles___ | never heard about them |
| 19:09.43 | russellb | everything SIP ... of course |
| 19:09.59 | russellb | SIP is the answer to every possible situation that could exist! |
| 19:10.02 | amer | IMS is the session enabler and application servers reside on top of it |
| 19:10.46 | amer | Sylantro, broadsoft for calls 5, iperia for voicemail, ubiquity for conf |
| 19:11.01 | russellb | that just sounds like a bunch of buzzwords |
| 19:11.13 | amer | I think a single asterisk box can replace all the above |
| 19:11.14 | charles___ | I agree |
| 19:11.23 | russellb | and you could do that easily, just use ser as a proxy (your "session enabler") |
| 19:11.59 | russellb | then asterisk as your application server for whatever parts you want |
| 19:12.05 | amer | yes |
| 19:12.21 | n838901_2 | russellb: pastebin is having problems apparently..so i put in on my webserver here: http://n8box.homelinux.net/debug/output |
| 19:12.24 | russellb | or you could skip the whole mess and just use asterisk :) |
| 19:12.53 | file | russellb: TPS REPORT!!! |
| 19:13.00 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
| 19:13.08 | RoyK | SIP is created solemly in the thought of that no NAT exists anywhere and that all people are good and nice and noone would think a bad thought |
| 19:13.40 | n838901_2 | lol |
| 19:13.46 | russellb | n838901_2: that is not asterisk addons. those files don't exist there :) |
| 19:13.50 | russellb | file: OK! |
| 19:13.58 | n838901_2 | uhh... |
| 19:14.00 | n838901_2 | hmm |
| 19:14.51 | n838901_2 | russellb: what is goin on? |
| 19:15.10 | russellb | i don't know. i just woke up and i am confused |
| 19:15.16 | n838901_2 | I'm installing freepbx, but i thought this was an asterisk issue |
| 19:15.32 | russellb | well this is some kind of patched asterisk tree |
| 19:15.38 | russellb | because the files are all from the main asterisk tree |
| 19:15.44 | russellb | but it includes mysql stuff ... |
| 19:15.51 | russellb | so, you need to report the problem to them |
| 19:19.57 | n838901_2 | ahh |
| 19:20.02 | n838901_2 | thanx..will do |
| 19:20.33 | n838901_2 | would anyone recommend a 'hassle-free' gui for asterisk? |
| 19:20.46 | n838901_2 | ive tried destar, but it is unstable |
| 19:20.52 | RoyK | n838901_2: vi |
| 19:21.00 | n838901_2 | lol ^_^ |
| 19:21.07 | RoyK | "menus belong in restaurants" |
| 19:21.28 | *** join/#asterisk jolo (n=jolo@khepri.openbios.org) |
| 19:21.33 | jolo | hi |
| 19:21.36 | n838901_2 | i would actually rather do it that way, but there are some hellacious config files for asterisk |
| 19:22.04 | RoyK | n838901_2: it takes a little time learning how to do stuff, but after then it's quite straight-forward |
| 19:22.30 | n838901_2 | i am having trouble trying to understand and hand-code the dialplans |
| 19:23.13 | RoyK | ~docs |
| 19:23.15 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 19:23.20 | RoyK | ~book |
| 19:23.22 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:23.26 | RoyK | the book is rather good |
| 19:23.30 | n838901_2 | really? |
| 19:24.08 | n838901_2 | thanx for the info |
| 19:24.12 | harryvv | nufone has changed its page layout. |
| 19:24.32 | RoyK | how wonderfully delightful to hear, harryvv |
| 19:24.59 | harryvv | yes RoyK, it is. |
| 19:25.30 | RoyK | although it might be so, I can hardly see the value of it for asterisk |
| 19:26.25 | harryvv | I just wonder if there customer services has also improved :) |
| 19:27.01 | RoyK | I somehow doubt it |
| 19:27.10 | *** join/#asterisk Trakkasure (n=Sgemtum@69-163-145-203.atlsfl.adelphia.net) |
| 19:27.51 | harryvv | Hard to run a voip company one two people but because its such a razor thin margin line of work its probebly not going to happen any time soon. Vonage press release stated thay will finally make a profit in 2008. |
| 19:28.02 | harryvv | :) |
| 19:29.07 | harryvv | btw I am comparing sangoma and digium. Echo has always been a headach in the past has Digium made improvments where I should consider more of there cards? |
| 19:29.16 | folder | My copy of 'The Book' arrived yesterday. It's so much easier reading the printed book than a stash of A4 printed sheets. |
| 19:29.41 | *** join/#asterisk adker (n=adker@74-33-205-58.br1.glv.ny.frontiernet.net) |
| 19:31.38 | harryvv | Anyone here review the Sangoma A200? |
| 19:33.34 | RoyK | harryvv: what is that? tdm card? |
| 19:34.01 | harryvv | yes |
| 19:37.58 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 19:39.05 | konfuzed | hey how can I setup asterisk to receive a fax and send it to CUPS default print queue |
| 19:39.05 | *** join/#asterisk St1ckm4n (n=shortes9@c-71-193-166-111.hsd1.or.comcast.net) |
| 19:39.27 | St1ckm4n | Hi |
| 19:40.40 | St1ckm4n | can anyone point me to some decent documentation on asterisk database show "/AMPUSER/XXX/recording" I want to see if I can enable recording for a specific extension easily |
| 19:41.36 | St1ckm4n | I'm not even sure that this recording key will do that, but I thought it would be easier than editing my extensions.conf |
| 19:41.49 | [Airwolf] | St1ckm4n, read topic about AMP support |
| 19:42.34 | St1ckm4n | sorry, I didn't think about going there because I don't use AMP to configure anything, it was installed on our box by the vendor that made it but I really don't like it |
| 19:43.03 | ManxPower | St1ckm4n, Your config files are still tortured and mamed by AMP. |
| 19:43.36 | St1ckm4n | yeah, I'm see AMP is thrown all over the conf files in comments and crap |
| 19:43.38 | ManxPower | harryvv, Just go with Tellabs |
| 19:44.01 | St1ckm4n | is that database key specific to AMP? |
| 19:44.05 | ManxPower | Tellabs external EC is hell to get working, but once you do, you'll never have echo again. |
| 19:44.16 | ManxPower | St1ckm4n, Yes, that is specific to AMP. |
| 19:44.17 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 19:44.29 | St1ckm4n | ok, thanks I'll try #freepbx |
| 19:45.02 | *** join/#asterisk AJaymn (n=Ya@70.59.126.206) |
| 19:45.07 | *** join/#asterisk beyond (n=beyond@201-1-206-248.dsl.telesp.net.br) |
| 19:45.13 | ManxPower | When I installed the Tellabs EC at a larger office (80 phones), turned off Asterisk EC, set rxgain and txgain back to 0, I never had a complaint about volume levels or EC again. |
| 19:45.19 | ManxPower | And that was 6 months ago. |
| 19:45.34 | denon | tellabs EC adjusts gain too? |
| 19:45.42 | ManxPower | Tellabs is what companies like AT&T use. |
| 19:46.11 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 19:46.13 | ManxPower | denon, no, but most people have to lower their txgain so much to get rid of echo that callers complain outbound audio is too soft. |
| 19:46.28 | denon | nod |
| 19:46.37 | denon | though some pstn is just quiet because it's a long ways from the CO |
| 19:46.40 | ManxPower | no need to do that with Tellabs |
| 19:46.48 | ManxPower | denon, Not on PRI |
| 19:46.53 | denon | obviously |
| 19:47.03 | denon | figured you had the tellabs between the CB and * |
| 19:47.10 | ManxPower | denon, gads no. |
| 19:47.17 | denon | people do it |
| 19:47.21 | denon | and supposedly like it |
| 19:47.26 | ManxPower | IP Phones <-> Asterisk <-> Tellabs <-> Telco |
| 19:48.05 | ManxPower | we also have a CB hanging off the asterisk box, but it's not used much. We had to move all our fax lines off of Asterisk because we had to turn down the txgain so much faxes were having problems. |
| 19:48.18 | ManxPower | We have not moved them back to Asterisk since the Tellabs EC install. |
| 19:49.49 | konfuzed | whats this third party OSS fax package |
| 19:50.47 | ManxPower | konfuzed, I don't know, what is it? |
| 19:53.35 | konfuzed | it was listed as the feature for Fax send and receive on the asterisk.org site |
| 19:53.45 | konfuzed | or it still is listed ;^) |
| 19:57.04 | konfuzed | well what is the functional way to use asterisk to receive the occasional fax 2 or 3 a month |
| 19:58.14 | quid2478 | hmm.. anybody here using RealTime with MySQL? |
| 19:58.20 | konfuzed | maybe ill just give him this usr 28,800 external faxmodem to use with his windows desktop |
| 19:59.28 | ManxPower | We use fax machines for sending/receiving faxes. |
| 20:00.05 | Mike | hey guys how can i know if trunk mode is working or not? |
| 20:03.17 | *** join/#asterisk niter3 (n=niter3@d57-102-239.home.cgocable.net) |
| 20:03.50 | niter3 | i'm a bit confused.. I've setup a mailbox for one of my users and set a number to dial up voicemailmain pointing to the mailbox # and the voicemail context. |
| 20:03.57 | niter3 | well my password will not get accepted. |
| 20:06.19 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 20:10.19 | jbalcomb | Anyone know what the brackets of numbers after the MAC address are for? GS_LOG: [00:0b:82:04:d5:89][708][FF71][01010010] File Not Found: boot55a.bin |
| 20:11.38 | *** join/#asterisk iq|mobile (n=IQ@unaffiliated/iq) |
| 20:11.47 | niter3 | yah my voicemail password will not work.... |
| 20:11.49 | niter3 | what the heck |
| 20:12.19 | niter3 | i've just edited the scripts.. Do I have to run a command.... addmailbox or something |
| 20:12.20 | niter3 | then add the one |
| 20:12.22 | niter3 | ? |
| 20:12.45 | niter3 | hrm.... |
| 20:12.58 | niter3 | ok well it's in my /var/spool/asterisk/voicemail/default |
| 20:15.18 | St1ckm4n | niter3: does your password match the one in /etc/asterisk/voicemail.conf? |
| 20:15.25 | niter3 | yes |
| 20:15.43 | niter3 | that one I defined i try dailing and it says it's incorrect |
| 20:17.34 | St1ckm4n | does the line look somewhat like this |
| 20:17.35 | St1ckm4n | 555 => 1234,User Name,email@email.com,,attach=yes|saycid=no|evelope=no|delete=no |
| 20:17.47 | St1ckm4n | 555 being your extension and 1234 being the pass |
| 20:18.49 | niter3 | i don't have the additional attach=yes|saycid=n etcc. |
| 20:18.54 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 20:19.04 | St1ckm4n | what version of * are you running? |
| 20:19.08 | niter3 | in voicemail |
| 20:19.09 | niter3 | yes |
| 20:20.11 | niter3 | 1.2.9.1 |
| 20:20.41 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:20.50 | St1ckm4n | can you pm me that line in your voicemail.conf |
| 20:23.02 | niter3 | St1ckm4n: hey |
| 20:23.10 | niter3 | 555 would be my voicemail # |
| 20:23.17 | St1ckm4n | yes |
| 20:23.27 | niter3 | ok, sec i'll msg you it |
| 20:23.52 | jbalcomb | Shouldn't this be showing essentially what the consolse at set verbose 99 does? |
| 20:23.53 | jbalcomb | logger.conf: /var/log/asterisk/rediculous.rstar File Enabled - Debug DTMF Verbose Warning Notice Error |
| 20:24.42 | jbalcomb | file: is that eyebrow for me? |
| 20:25.11 | file | perhaps |
| 20:26.26 | jbalcomb | I'm looking to see the usage of my call queues by thier extension and was hoping to just grep a log file for them |
| 20:26.46 | jbalcomb | something like grep 854854 rediculous.asterisk | wc -l |
| 20:27.04 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 20:27.27 | jbalcomb | s/854854/|854854| |
| 20:27.37 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 20:28.57 | *** join/#asterisk nailbags|work (n=nailbags@149.171.94.134) |
| 20:29.36 | jbalcomb | ok, nm, it works now for reasons beyond my comprehension. |
| 20:31.50 | Mattwj2005 | anyone know how to get e164 servers to work behind NAT? |
| 20:32.49 | Cybertoy | you mean asterisk? |
| 20:33.11 | Cybertoy | or the e164 dns server? |
| 20:35.09 | niter3 | hey guys.. I've just setup a voicemail for one of my extensions and when trying to listen to my voicemail my password keeps getting rejected.. |
| 20:35.19 | niter3 | everything seems to be placed under the correct context.. |
| 20:36.14 | Mattwj2005 | I want to get a range of numbers to connect to my asterisk server |
| 20:37.14 | niter3 | yeah this makes no sense |
| 20:37.46 | *** join/#asterisk jterrero (n=that@200.42.248.7) |
| 20:39.24 | jterrero | Can someone help me out with a simple home setup, just need help with extensions.conf, I already did the zap and asterisk install... Have an analog phone connected to tdm card... i just want the user to have to dial a predefined # (2121) before being able to dial out to real world |
| 20:39.27 | jterrero | using pstn line |
| 20:39.37 | Mattwj2005 | from within my local network I can connect through a guest account |
| 20:42.09 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 20:43.21 | *** part/#asterisk becks (n=becks@stepbuild.com) |
| 20:45.29 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 20:52.14 | niter3 | jterrero: Have you got voicemail to work? |
| 20:55.14 | niter3 | wow this is pissing me off.. It won't freeking accept my voicemail password |
| 20:55.17 | niter3 | urg.. why |
| 20:58.01 | *** part/#asterisk jarg (n=jarg@dsl-201-129-100-218.prod-infinitum.com.mx) |
| 21:05.09 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 21:08.22 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 21:08.26 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:08.28 | hads|home | niter3: WHat type of phone are you using? |
| 21:08.55 | niter3 | sip phone |
| 21:09.28 | hads|home | Your DTMF mode could be wrong |
| 21:09.52 | niter3 | in the sip phone? |
| 21:09.55 | *** join/#asterisk s0lid (n=jlq@210.213.127.56) |
| 21:09.57 | hads|home | Yes |
| 21:09.59 | niter3 | how do I know what it is normally? |
| 21:10.10 | hads|home | What phone is it? |
| 21:10.16 | niter3 | twinkle |
| 21:11.02 | hads|home | Well, I don't know about Twinkle, you should be able to set the DTMF mode in the settings somewhere |
| 21:11.13 | niter3 | well how do I know what i'm setting it to? |
| 21:12.58 | niter3 | yes that's what it is |
| 21:13.03 | niter3 | ok i gotta figure out where to configure ti |
| 21:13.31 | niter3 | i tried with another softphone and it worked |
| 21:13.33 | niter3 | so it's the DTMF |
| 21:13.37 | *** join/#asterisk apardo (n=apardo@87.217.146.61) |
| 21:14.46 | hads|home | jterrero: You may want to look into the DISA applicationm |
| 21:15.00 | niter3 | hrm..... what should it be set to? |
| 21:16.05 | niter3 | i have dtmf payload type, dtmf duration, dtmf pause, dtmf volume.. and dtmf i have the options Auto RFC 2833 and inband. |
| 21:16.11 | niter3 | I tried both options and neither worked |
| 21:16.44 | niter3 | brb |
| 21:21.29 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 21:22.13 | Ciber311 | hmm |
| 21:24.58 | *** join/#asterisk niter3 (n=niter3@d57-102-239.home.cgocable.net) |
| 21:25.18 | niter3 | ok, well I know the DTMF is set to RFC 2833 on asterisk and it's set on my client |
| 21:25.23 | niter3 | yet it still bitchs. However, xten works fine |
| 21:26.34 | *** part/#asterisk Mattwj2005 (n=Matt@user-12l3nck.cable.mindspring.com) |
| 21:29.20 | frodonbash | hi |
| 21:30.00 | frodonbash | i would like more time before switch on zaptel ! how can i do ? |
| 21:33.31 | frodonbash | please :-p |
| 21:35.02 | Mike | hey guys how can i know if trunk mode is working |
| 21:35.03 | Mike | ? |
| 21:35.59 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 21:36.26 | jterrero | ug 5 10:37:38 NOTICE[4654]: app_dial.c:1029 dial_exec_full: Unable to create channel of type 'IAX' (cause 66 - Channel not implemented) |
| 21:36.27 | jterrero | <PROTECTED> |
| 21:36.31 | jterrero | can someone help me out with that |
| 21:36.38 | Mike | i have 20 calls using g729 and im getting RX(KB/sec) 50.975 TX(KB/sec) 28.029 |
| 21:36.43 | jterrero | i can recieve calls, cant make calls through my iax providor |
| 21:42.23 | *** join/#asterisk leejohn (n=burnproo@210.213.198.75) |
| 21:46.24 | frodonbash | i want change the time before zaptel switching ! please help me |
| 21:46.31 | _4d4m_ | jterrero: normally means you have an error in your call to Dial() in your dialplan |
| 21:49.11 | *** join/#asterisk psymon101 (i=psymon10@24-236-222-55.dhcp.aldl.mi.charter.com) |
| 21:52.33 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 21:57.29 | *** join/#asterisk tempest1 (n=asf@adsl-150-198-149.chs.bellsouth.net) |
| 21:58.27 | *** join/#asterisk anthm (n=anthm@000-397-985.area4.spcsdns.net) |
| 21:58.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:59.02 | frodonbash | i want change the time before zaptel switching ! please help me |
| 22:00.51 | frodonbash | ok good night all |
| 22:03.37 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 22:07.00 | Rowter | I need a funcionality on queues on asterisk trunk, I could compile it with zaptel 1.2.7 and lastest stable libri? or I'll need to work with trunk zaptel and libpri? |
| 22:08.25 | Mike | Rowter, Armando? |
| 22:08.44 | Rowter | Mike, hey!!!! |
| 22:12.21 | niter3 | hey guys, what's a good softphone for linux? |
| 22:24.28 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 22:32.40 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 22:49.52 | *** join/#asterisk Paavum (n=chiardon@200.71.58.39) |
| 22:51.03 | Paavum | Hello? |
| 22:51.47 | Paavum | Kinda wondering who could help me look at a small config issue |
| 22:51.57 | Paavum | I think I blew it, yet I am unsure where |
| 22:52.05 | Paavum | and I cant make or recieve any calls |
| 22:59.29 | psymon101 | from where? |
| 22:59.49 | psymon101 | how are you getting incomming calls? |
| 23:00.03 | psymon101 | IAX provider? |
| 23:00.04 | psymon101 | etc. |
| 23:01.03 | Paavum | psymon101... Colombia, ETB |
| 23:01.11 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:01.11 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 23:03.51 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 23:04.21 | Paavum | extensions.conf: http://pastebin.ca/117435 |
| 23:12.28 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 23:13.08 | Paavum | Hello? |
| 23:22.23 | Paavum | Anybody here |
| 23:23.57 | docelmo | no go away |
| 23:23.57 | docelmo | :P |
| 23:24.03 | Paavum | docelmo ... can ya gve me a hand? |
| 23:24.25 | docelmo | maybe whats the issue? |
| 23:24.27 | Paavum | I have just redone my extensions.conf |
| 23:24.40 | Paavum | and I cant get my system to accept incoming calls |
| 23:24.58 | Paavum | I believe I have some kind of mixup in my contexts |
| 23:26.43 | docelmo | Where are you trying to recieve calls from? |
| 23:26.49 | docelmo | provider or ata? |
| 23:26.54 | Paavum | provider |
| 23:26.55 | docelmo | or zap? |
| 23:27.00 | docelmo | ok.. SIP/IAX? |
| 23:27.04 | Paavum | zap |
| 23:27.21 | docelmo | so you calls come in on Zap channels? |
| 23:27.32 | Paavum | yeah |
| 23:27.39 | Paavum | I get this onthe cli |
| 23:27.39 | Paavum | <PROTECTED> |
| 23:28.09 | docelmo | ok so go into your entrada context and fix it. |
| 23:28.16 | docelmo | thats a no brainer.. |
| 23:28.17 | nextime | Paavum : exten => _5922999,1,.... |
| 23:28.26 | nextime | on [entrada] context. |
| 23:28.28 | file | don't need _ |
| 23:28.34 | docelmo | if your matching no |
| 23:28.45 | Paavum | I'm matching from 97 to 99 |
| 23:28.48 | docelmo | if your going to guess for a number you need the _ |
| 23:29.04 | docelmo | ok so put _592299X,1,.... |
| 23:29.16 | *** join/#asterisk derekS (n=dereks@unaffiliated/dereks) |
| 23:29.28 | docelmo | You catch all of them with one extension |
| 23:29.38 | Paavum | just 97-99 :P |
| 23:29.44 | Paavum | 90 is a pbx |
| 23:29.53 | Paavum | 91 is another ... |
| 23:30.00 | derekS | hi. i am reading the trixbox config info for my 7960 phone (http://www.trixbox.org/modules/smartsection/item.php?itemid=15)... and it talks about getting the updated firmware... where do i find that? |
| 23:30.11 | Paavum | here I'll pastebin the context |
| 23:30.16 | docelmo | derekS READ THE FRICKEN TOPIC! |
| 23:30.55 | derekS | docelmo: sorry |
| 23:30.56 | *** part/#asterisk derekS (n=dereks@unaffiliated/dereks) |
| 23:31.46 | Paavum | http://pastebin.ca/117477 |
| 23:34.49 | docelmo | ok other than its not in english where is the problem? |
| 23:36.31 | Paavum | When I call the call gets dropped |
| 23:36.33 | Paavum | look |
| 23:37.04 | Paavum | <PROTECTED> |
| 23:37.22 | docelmo | try a _ |
| 23:37.36 | docelmo | I have found sometimes it needs it. |
| 23:37.42 | docelmo | asterisk is flaky like that |
| 23:37.43 | Paavum | already did |
| 23:37.48 | docelmo | amongst other things |
| 23:37.59 | nextime | ehm, extension reload? |
| 23:38.00 | nextime | :) |
| 23:38.05 | docelmo | ohh lord |
| 23:38.11 | docelmo | do a show dialplan |
| 23:38.16 | docelmo | and looking for that extension |
| 23:39.49 | Paavum | I cant see it |
| 23:40.11 | docelmo | there ya go |
| 23:40.14 | docelmo | RELOAD.. :) |
| 23:40.42 | Paavum | I already did that! |
| 23:40.44 | Paavum | 20 times |
| 23:41.30 | docelmo | 1 more time for shits and giggles |
| 23:41.44 | anthm | try PUNT |
| 23:42.14 | docelmo | Im looking at yer dialplan again hold |
| 23:42.16 | Paavum | anthm.... me??? |
| 23:42.41 | Paavum | docelmo . . yepppp |
| 23:43.30 | docelmo | when you dial the other extension to that person does it work the 60XX number? |
| 23:43.46 | anthm | you probably have that extension.ael shit taking precedence over extensions.conf |
| 23:43.55 | Paavum | nope |
| 23:44.09 | Paavum | anthm ...no extensions.ael exist anymore+ |
| 23:44.21 | Paavum | docelmo...nope |
| 23:44.35 | docelmo | try seperating your exten => bla.. |
| 23:45.42 | Paavum | alrfeady did that |
| 23:46.01 | *** join/#asterisk saftsack (n=saftsack@p54A7F03F.dip.t-dialin.net) |
| 23:46.02 | Paavum | can you likecome in the system and help me out? |
| 23:46.05 | docelmo | ok then yer fuk'd.. someone needs to get into the box.. |
| 23:46.55 | Paavum | docelmo... can ya come in? |
| 23:47.05 | docelmo | ya I have a couple minutes.. |
| 23:47.10 | docelmo | waiting on phone calls anyway.. |
| 23:48.30 | Ciber311 | anyone using axvoice here? |
| 23:49.41 | Paavum | docelmo . . .check your private!!please |
| 23:51.04 | anthm | is this really the place for him to check his privates ? |
| 23:52.46 | Paavum | i can`t think better place to doit tha his own PC! |
| 23:52.58 | *** join/#asterisk Dorphalsig (n=asdasdas@200.71.58.39) |
| 23:54.23 | anthm | your problem is dialplan not reloading ? |
| 23:54.23 | *** join/#asterisk jeffgus (n=jeffgus@shaper.zimage.com) |
| 23:54.48 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 23:55.03 | Paavum | anthm . . .i reloaded a couple of times! |
| 23:55.36 | Paavum | (including shuting down asterisk and restarting it again) |
| 23:55.44 | Dorphalsig | anthm, .... my prob is incoming calls are rejected |
| 23:56.02 | anthm | you said you changed your diaplan and it didnt take effect ? |
| 23:56.54 | Dorphalsig | uhh no |
| 23:57.15 | Dorphalsig | I'm saying I cant find my context 'entrada' when I do the show dialplan |
| 23:57.35 | Dorphalsig | antm ... Should i pastebin the whole extensions.cofn? |
| 23:57.41 | anthm | yep |
| 23:57.46 | Dorphalsig | oki hold on |
| 23:59.22 | Paavum | http://pastebin.ca/117505 |
