00:01.27 | justnulling2 | crash3m: http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 There are currently no files for this type. |
00:01.31 | duke149 | thats just standard mobile phone isnt it? a phone company actually has one of their towers on our site |
00:02.11 | newmedian | duke149: that's just for cellular. You set up your own cell repeater, to give yourself perfect or near perfect signal strength; It repeats the closest tower to you. |
00:02.24 | duke149 | yea |
00:02.29 | crash3m | justnulling2: I dont have a username/password ;) |
00:03.06 | justnulling2 | crash3m: well i have one and it says There are currently no files for this type. |
00:03.10 | duke149 | did look at mobile phones once but it just costs 2 much |
00:04.01 | crash3m | justnulling2: an associate just got P0S3-07-4-00 from cisco...I dont know where, but he did |
00:04.47 | justnulling2 | hmm |
00:07.10 | ariel_ | ok is anyone here good with the polycom phones. I have a question about there qos settings |
00:08.18 | newmedian | jbot seen ManxPower |
00:08.20 | jbot | manxpower <~eric@adsl-6-105-205.msy.bellsouth.net> was last seen on IRC in channel #asterisk, 9h 34m 25s ago, saying: 'you asked about RTP always passing thru Asterisk.'. |
00:09.18 | *** join/#asterisk bumperhosting (shankland@82-41-86-32.cable.ubr01.dund.blueyonder.co.uk) |
00:09.20 | bumperhosting | Hello |
00:09.29 | cursor | hello there |
00:09.32 | cursor | UK, I see |
00:09.35 | *** join/#asterisk ChkDigit (~mike@static65-87-226-124.regina.accesscomm.ca) |
00:09.36 | cursor | You're up late |
00:09.37 | bumperhosting | Yeh:D |
00:09.40 | ariel_ | hello |
00:09.41 | bumperhosting | lol |
00:09.45 | *** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
00:09.50 | bumperhosting | Not that late its only 1.09 |
00:10.06 | cursor | wow - that's the same time as on my clock :-) |
00:10.13 | ariel_ | 1 am hummm I would be sleeping by then. |
00:10.32 | cursor | sleep is for the weak |
00:10.39 | *** part/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
00:10.43 | NewSole | Mine is Flashing 12:00 |
00:10.49 | ariel_ | cursor, then let me be week. |
00:10.54 | bumperhosting | lol |
00:10.54 | ariel_ | weak... |
00:11.05 | GyrosGeier | Hrm |
00:11.09 | cursor | sleep for a week |
00:11.09 | bumperhosting | I dont sleep till about 8am every morning |
00:11.10 | ariel_ | NewSole, you have my vcr then |
00:11.18 | duke149 | any recomendations for what distro is good to run Asterisk on? |
00:11.25 | cursor | Gentoo |
00:11.28 | ariel_ | CentOS |
00:11.41 | cursor | MS Windows XP |
00:11.42 | cursor | err |
00:11.43 | cursor | oops |
00:11.47 | cursor | :-) |
00:11.50 | GyrosGeier | What would be a good way to connect a USB phone (HID+Soundcard) to a Linux box? |
00:11.58 | duke149 | wtf |
00:12.01 | cursor | haha |
00:12.21 | ariel_ | duke, what OS are you normally used to? |
00:12.33 | cursor | GyrosGeier: Via the USB port |
00:13.15 | bumperhosting | god alll so technical |
00:13.17 | *** join/#asterisk theHub (~karl@ool-182cba82.dyn.optonline.net) |
00:13.19 | bumperhosting | :( i would not know where to start |
00:13.26 | newmedian | Newsole.. there's this .mov, and you watch it, and one week later you get a phone call.... |
00:13.38 | GyrosGeier | cursor, obviously. |
00:13.44 | cursor | :-) |
00:14.30 | GyrosGeier | cursor, I was wondering whether there is any soft phones that can deal with this hardware |
00:14.41 | shido6 | ? |
00:14.43 | NewSole | I go to bathroom and do a mov in there and a few min's I get a call for neibors... wondering what died |
00:14.56 | *** join/#asterisk duke149 (~ca07b782@cgiirc-gateway.ictoadd.com) |
00:15.00 | cursor | I don't know - sorry |
00:15.01 | newmedian | thank you for airing your concerns |
00:15.09 | NewSole | lol |
00:15.12 | duke149 | bah dam CGI IRC client. |
00:15.12 | shido6 | you did a movie in the bathroom while you were crapping? |
00:15.34 | cursor | newmedian: would that be thering.mov ? |
00:15.45 | newmedian | :) |
00:15.45 | duke149 | as i was sayin.. anyone got any recomendations for what distro to run Asterisk on? |
00:15.50 | NewSole | the-bowl-ring.mov |
00:16.14 | shido6 | redhat works duke149 |
00:16.32 | cursor | duke: I always recommend Gentoo, so that makes me very predictable |
00:16.38 | duke149 | hehe |
00:17.48 | newmedian | Here's a Linux Distribution Chooser: http://www.zegeniestudios.net/ldc/index.php |
00:17.57 | crash3m | justnulling2: http://www.voip-info.org/tiki-index.php?page=Cisco+Phones |
00:18.41 | justnulling2 | crash3m: thanks let me take a look at it |
00:20.01 | bmd | hey, I have a fairly technical question about HEAD and new features - any -dev types around? |
00:21.19 | cursor | ask about HEAD and they will come |
00:21.30 | bmd | ok here's my question |
00:21.36 | duke149 | newmedian: cool. it says Debian for me. |
00:21.44 | bmd | functions and agis. how does one call a function from an agi |
00:21.52 | bmd | is there some identity app that I don't know about? |
00:21.59 | jakepdev[work] | EXEC |
00:22.05 | cursor | yes - exec |
00:22.11 | bmd | exec runs apps |
00:22.16 | bmd | does it run functions too? |
00:22.26 | jakepdev[work] | functions? |
00:22.29 | cursor | functions are apps |
00:22.31 | cursor | like dial |
00:22.33 | cursor | etc. |
00:22.45 | jakepdev[work] | oh - where's that damn translator |
00:22.46 | bmd | looking at pbx.c they look like they're different |
00:23.13 | *** join/#asterisk bofh42 (~bofh42@p548234A3.dip0.t-ipconnect.de) |
00:23.35 | *** kick/#asterisk [cursor!~mark@kram.digium.sponsor.pdpc] by kram (turn off your silly script) |
00:23.39 | *** join/#asterisk hypa7ia (~leigh@HSE-London-ppp291210.sympatico.ca) |
00:24.20 | kram | hi file |
00:24.25 | file | hi krammy boy |
00:24.31 | file | how are you? |
00:25.28 | *** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz) |
00:25.31 | cursor | That's nice |
00:25.34 | kram | i'm otay |
00:25.47 | kram | i've mentioned it to you before, cursor :) |
00:25.55 | kram | it was maybe funny 3 or 4 months ago |
00:25.56 | bmd | cursor: can I bug you a bit more about apps vs. functions? |
00:25.57 | kram | but it's time to move along |
00:26.18 | jakepdev[work] | bmd - what are you calling a function? |
00:26.44 | cursor | bmd: if you like |
00:27.30 | NewSole | hi kram |
00:27.33 | kram | hi sole |
00:27.36 | *** join/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
00:27.39 | bmd | jake: anything registered by ast_custom_function_register |
00:27.53 | bmd | cursor: looking at the code, it looks like functions and apps differ |
00:27.57 | bmd | at least in HEAD |
00:28.00 | NewSole | got a question about your TE405 cards if u got a moment |
00:28.02 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
00:28.14 | drumkilla | NewSole: what's the question? |
00:28.14 | cursor | Can you give an example of a function vs. an app? |
00:28.15 | bmd | functions are registered with ast_custom_function_register, apps are registered with ast_register_application |
00:28.18 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-133.modem.logical.net) |
00:28.31 | cursor | ok |
00:28.35 | NewSole | is there a special setup for Echo cancel on those cards kram |
00:28.43 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
00:28.45 | bmd | funtion SIP_HEADER, app SipGetHeader |
00:28.56 | kram | sole: you can e-mail support@digium.com |
00:29.03 | kram | they can assist you with configuration issues |
00:29.10 | kram | also we have a new echo can board you can use if you want to offload it |
00:29.19 | jakepdev[work] | bmd - you can call SIP_HEADER from the dialplan? |
00:29.38 | bmd | jake: yes it's new in HEAD |
00:29.41 | jakepdev[work] | ok |
00:29.46 | MikeJ[Laptop] | for the echo can board email sales@digium.com ;) |
00:29.47 | bmd | functions are a new addition |
00:29.58 | NewSole | well.. ok.... |
00:30.36 | bmd | actually, looking at the code, it looks like nobody has committed any functions yet |
00:30.47 | bmd | sorry, I've been following mantis a bit more than CVS |
00:30.52 | bmd | there's a lot of them in mantis |
00:31.01 | cursor | I didn't notice the functions stuff go in |
00:31.02 | bmd | SIP_HEADER seems to be the only one right now |
00:31.06 | MikeJ[Laptop] | bmd, there are a bunch of them, they are in head, not 1.0.x brance |
00:31.08 | kram | sole: if you can't get your question answered through tech support, let meknow |
00:31.09 | MikeJ[Laptop] | branch |
00:31.09 | cursor | it probably had a CVS log like "oops" |
00:31.28 | cursor | I'll see if I can find it |
00:31.41 | NewSole | well last time I sent an email to tech... I waited a week almost for anser back |
00:31.53 | kram | what's your ticket number? |
00:31.54 | drumkilla | cursor: if you didn't notice, then you weren't paying any attention |
00:31.56 | bmd | MikeJ: all I see is SIP_HEADER, grepping for ast_custom_function_register |
00:32.03 | drumkilla | there has been a ton of commits realted to dialplan functions |
00:32.07 | cursor | drum - possibly :-) |
00:32.12 | bmd | ok, so back to my question |
00:32.13 | drumkilla | certainly |
00:32.15 | bmd | agis and functinos |
00:32.24 | bmd | doesn't look like you can call a function from an AGI |
00:32.27 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
00:32.30 | NewSole | I would have to look up its on office computer |
00:32.36 | bmd | unless I'm missing something |
00:32.43 | bmd | so there's either two solutions |
00:32.55 | bmd | one is an identity app, that just echos what the function args sent to it return |
00:33.14 | bmd | the other is a new agi command outside of EXEC that evaluates functions |
00:33.27 | bmd | or am I totally out of it here |
00:33.38 | jakepdev[work] | that sounds right |
00:33.46 | kram | newsole: i'll have matt o contact you then |
00:34.11 | jakepdev[work] | or implement the function code in your agi app |
00:34.11 | NewSole | k |
00:34.28 | bumperhosting | anyone wants to do some contracting for me, installing asterisk on a FreeBSD system, hosted remotely. |
00:34.32 | bumperhosting | oops |
00:34.41 | bumperhosting | does anyone want to... " " etc etc |
00:34.44 | kram | if you don't get the assistance you need, let me know |
00:35.07 | jakepdev[work] | bumperhosting - try shido6 |
00:35.19 | NewSole | ok thank |
00:35.26 | bumperhosting | jakepdev, thanks I will. |
00:36.00 | bmd | jakepdev: well, things like SIP_HEADER are impossible to implement in an agi (after SipGetHeader is removed) |
00:36.09 | bmd | so there's a gap here in functionality (pardon the pun) |
00:36.38 | *** join/#asterisk lilneon (~tj_r3@cuscon14794.tstt.net.tt) |
00:36.41 | cursor | I see a commit notice from 05 May 2005 related to functions |
00:36.46 | lilneon | hi everyone good night |
00:37.07 | asteriskn00b | evening all, new asterisk install on centos 3 Whenever I try to start asterisk, it randomly dumps and will not load, I have tried asterisk -vv but I get no "error" messeges. Any ideas on what to look for? |
00:37.21 | *** join/#asterisk hypa7ia (~leigh@HSE-Montreal-ppp143550.sympatico.ca) |
00:37.24 | lilneon | anyone can point me to a voip provider who can provide me with a toll free inbound number to my asterisk box? |
00:37.32 | cursor | nufone.net |
00:37.52 | cursor | Assuming USA |
00:38.12 | lilneon | cursor: assumption incorrect.. want outside the US, UK.. |
00:38.34 | cursor | My mind-reading skills are obviously getting rusty :-) |
00:38.56 | lilneon | cursor: nah.. prob no mind to read... i left it at the office |
00:39.01 | cursor | haha |
00:39.06 | lilneon | :) |
00:39.13 | lilneon | thnx though,... n e others? |
00:39.19 | cursor | calluk sell UK freephone numbers |
00:39.26 | cursor | calluk.com, I think |
00:39.33 | bumperhosting | uk |
00:39.44 | bumperhosting | go with sipgate.co.uk they give you free uk numbers. |
00:39.46 | lilneon | yeah but that's tol lfree for callers in the UK |
00:40.22 | bumperhosting | go with sipgate... they give u 0845, 0800, 0124 where ever u are... every area in the uk go with sipgate its completly free |
00:40.25 | cursor | freephone numbers are only free within the country the service is provided, unless you get an international toll-free |
00:40.28 | cursor | +800 |
00:40.51 | lilneon | which is what i kinda need right nwo |
00:41.08 | cursor | I don't know any +800 providers - I haven't looked |
00:41.15 | cursor | Google should scare up a couple of them |
00:41.30 | cursor | They are probably expensive |
00:41.50 | cursor | It might be cheaper to get a number for each country you want to deal with |
00:42.40 | bumperhosting | sipgate is free if u want uk man;) |
00:43.49 | Damin | From: markster@digium.com "Actually apply timestamp bug (bug #3961)" |
00:44.02 | Damin | So now we're actually ADDING bugs to CVS Head? ;) |
00:44.19 | kram | damin: sorry :) |
00:44.25 | *** join/#asterisk rene- (~root@200.78.176.114) |
00:44.38 | cursor | Hello rene |
00:44.44 | rene- | hey! |
00:45.18 | PatrickDK | we like bugs |
00:45.22 | PatrickDK | gives us something to do all day |
00:45.33 | *** join/#asterisk ilium007 (~ilium@203.94.178.214) |
00:45.38 | ilium007 | hi guys |
00:45.51 | ilium007 | "Unable to connect to remote Asterisk" |
00:45.55 | ilium007 | any ideas ??? |
00:46.06 | ilium007 | I am ssh'd into my * box at home form work |
00:46.16 | PTG123 | cvs head is where bugs live :) its like an ant hill |
00:46.21 | Cardoe | Any reliable outbound calling providers? |
00:46.34 | PTG123 | Cardoe, how many minutes per month? |
00:47.01 | Cardoe | quite a few |
00:47.04 | rene- | iirc digium used to sell developer kits based in t100p and zhone zplex back in the day, i have now stumbled on some cheap units on ebay, they claim to have V35 interfaces so how does one connects one to a Digium Rj48 interfase? i have no telecom background so i dont know whether v35 is a protocol or a physical interfase |
00:47.20 | PTG123 | cardoe: pm so i can help you out.. |
00:47.57 | file | PTG123: you! |
00:48.09 | PTG123 | file: you! :) |
00:48.18 | file | PTG123: status report. |
00:48.28 | PTG123 | heh pm me :) |
00:49.44 | *** part/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com) |
00:50.54 | *** join/#asterisk NewSole2 (dave@i216-58-44-245.avalonworks.net) |
00:53.25 | *** join/#asterisk shakuhashi (~luciux@200.101.54.31) |
00:56.14 | ariel_ | rene-, the zhone units were ok but they had major problems. I had 8 of them and ended up giving them away. |
00:58.01 | *** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no) |
00:58.34 | file | ariel_: Got any left?!? |
00:59.25 | *** join/#asterisk TransAM (me3@adsl-68-92-124-249.dsl.lgvwtx.swbell.net) |
00:59.29 | TransAM | good evening |
00:59.38 | TransAM | I have a question |
00:59.47 | b0ef | what happened to the transfer command?; I can't find it in recent CVS. I got ztdummy and zaptel on a 2.6 kernel and able to dial into it, but I can't find the transfer command to transfer a call from the asterisk cli. Is this function removed and only found in the manager interface? |
01:00.20 | TransAM | what is wrong when, I try an echo test from more then one computer and even the phone line and I dont get anything back at all except for the numberpad keypress? |
01:00.53 | TransAM | I can call out, hear the called party but I cannot be heard, any idea? |
01:01.30 | meppl | gute nacht |
01:01.33 | b0ef | TransAM: how do you know the keypresses are being echoed? |
01:01.55 | TransAM | because I press the keys and her it a split second later |
01:01.57 | b0ef | are you sure they are being echoed from the remote server? |
01:02.02 | meppl | good night |
01:02.12 | TransAM | yes |
01:02.18 | b0ef | meppl: sleep well, send enemies to hell |
01:02.25 | TransAM | Ihear them twice |
01:02.32 | meppl | ;) |
01:02.36 | b0ef | TransAM: what protocol? |
01:02.38 | *** join/#asterisk Johnsie (~john@acs-24-154-32-12.zoominternet.net) |
01:02.41 | TransAM | sip |
01:02.45 | TransAM | hmm |
01:02.51 | TransAM | what ever xlite uses |
01:03.03 | TransAM | and when I dail in with my cell phone, same thing |
01:03.28 | b0ef | how about when you dial the echo application from the asterisk cli? |
01:03.43 | TransAM | cli? |
01:03.47 | TransAM | ohh |
01:03.54 | bkw_ | have chan_oss/alsa |
01:03.56 | bkw_ | and a sound card |
01:03.58 | bkw_ | :P |
01:04.07 | TransAM | I never tried |
01:04.13 | TransAM | let me give it a whirl |
01:04.54 | b0ef | now, where did the transfer command go.. |
01:05.06 | b0ef | is it just me? |
01:09.00 | Damin | I need chan_pringle-can |
01:09.09 | Damin | And chan_ting-can-and-string |
01:10.13 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
01:10.13 | *** mode/#asterisk [+o bkw_] by ChanServ |
01:11.15 | *** join/#asterisk bobx (~bobx@lowfreq.trancemitter.org) |
01:13.02 | TransAM | I am too new to this, I cant figure it out |
01:13.20 | TransAM | I thought I Was doing good when I got it to finally awnser the phone |
01:13.49 | TransAM | of course it picks up , then says there has been a terrible error and hangs up |
01:14.49 | b0ef | well, you know what you need to do |
01:15.07 | TransAM | buy a bullet, rent a gun? |
01:15.09 | b0ef | get cli up and dial the echo app locally from the cli |
01:15.27 | TransAM | still trying to figure out how to do that |
01:16.18 | b0ef | just execute asterisk -vvvvpc |
01:16.45 | *** join/#asterisk buu (foobar@69.56.150.34) |
01:16.55 | b0ef | setup an extension to dial the echo app, I've set mine to extension 600 |
01:17.15 | b0ef | when in the cli, just do a dial 600 |
01:17.30 | *** join/#asterisk netvulture (0@63.174.172.245) |
01:17.31 | b0ef | make sure you've setup asterisk to use a sound api, in modules.conf |
01:17.49 | buu | How do I call a function, like say, "SetLanguage(foo)" from an AGI script? |
01:20.08 | *** join/#asterisk Flav (~jmm@user-10lfdgg.cable.mindspring.com) |
01:20.11 | netvulture | can anybody explain the function of the ./makeconfig.sh 7123 2123 000F905796D9 192.168.1.123 63.174.172.123 "(650) 403-7123 " |
01:20.22 | netvulture | whoops - bad paste |
01:20.25 | *** join/#asterisk Sato1 (~rauleli@sato1.wizardteam.com) |
01:21.03 | netvulture | can anybody explain the function of the proxy_backup on a 79xx cisco phone? |
01:21.53 | netvulture | at what point would the phone use this proxy instead of the defaul outbound_proxy and line1_proxy? |
01:22.05 | PTG123 | if it can't reach the primary i am guessing |
01:23.00 | netvulture | i gather that would be the idea of it - however cisco has no documentation worth reading that explains it |
01:23.06 | buu | Bah |
01:23.08 | buu | Anyone? |
01:23.18 | netvulture | thus cisco proboably doesn't even know what it does |
01:24.07 | *** part/#asterisk bumperhosting (shankland@82-41-86-32.cable.ubr01.dund.blueyonder.co.uk) |
01:24.41 | TransAM | what file holds the SIP passwords? |
01:24.56 | *** join/#asterisk dave2_ (~dave2@leibniz.catalyst.net.nz) |
01:25.49 | Mavvie | start with sip.conf, work through all included files. |
01:26.54 | dave2_ | Hi all, any objections if I launch into a long question? :) |
01:27.09 | Mavvie | that was the first. |
01:27.31 | newmedian | dave2 if you're going to cut and paste, use pastebin |
01:27.33 | newmedian | ~pastebin |
01:27.34 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca |
01:27.50 | dave2_ | wasn't going to paste a lot, more that I know it's going to be fairly involved.. |
01:28.09 | dave2_ | I have a bunch of UIP200s on the same LAN as my * box.. |
01:28.28 | cursor | Time for me to go |
01:28.32 | cursor | Later, guys |
01:29.17 | dave2_ | transfers using the XFER hard button on the phone sort of work, but * loses the call into a void if we try and transfer to an extension which ends up in voicemail.. |
01:29.56 | dave2_ | I suspect it's a UIP200 bug, but before making noises in that direction, I hope someone else has run into it.. :) |
01:31.12 | rene- | ariel, i have learned that too, is there an alternative that is as cheap to get multiple FXOs in asterisk |
01:34.42 | rene- | the cheapest way im aware of getting multiple fxos in asterisk are clipcomm 4 port sip channel banks |
01:34.52 | rene- | they are like $350 or so |
01:36.24 | asteriskn00b | while trying to start asterisk I am receiving the following err "Ouch ... error while writing audio data: : Broken pipe" any ideas? |
01:36.29 | *** join/#asterisk forrestc (~fwc@206.127.78.199) |
01:36.50 | forrestc | Are there any SIP protocol level experts on and listening? |
01:37.14 | forrestc | I need to find out about the nonce parameter and when it should change |
01:37.31 | forrestc | I've got a sip box which is trying to register with the same nonce parameter over and over. |
01:38.45 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
01:39.02 | jets | Is thre major queue development going on? |
01:40.35 | *** join/#asterisk forrestc (~fwc@206.127.78.199) |
01:40.55 | forrestc | Oooppss.. hit wrong key.. Was there someone one who knew about SIP and the nonce parameter? |
01:41.02 | *** join/#asterisk duke149 (~ca07b782@cgiirc-gateway.ictoadd.com) |
01:46.51 | *** join/#asterisk heison (~heison@216.191.251.226) |
01:51.12 | duke149 | . |
01:51.19 | Damin | --- Results after 594 passes --- |
01:51.20 | Damin | Best: 100.000000 -- Worst: 99.987793 |
02:05.30 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
02:06.29 | *** mode/#asterisk [+o Cresl1n] by drumkilla |
02:06.41 | jets | FYI do an answer before a queue. |
02:06.42 | jets | Duh. |
02:06.43 | jets | haha |
02:07.51 | Cresl1n | drumkilla!!!!! |
02:07.55 | drumkilla | Cresl1n: !!!!!!!!!!!!! |
02:07.59 | drumkilla | we are having a mad party in here. |
02:08.09 | drumkilla | music should be much louder, though |
02:08.12 | Cresl1n | yeah |
02:08.16 | Cresl1n | here, I'll turn it up |
02:08.23 | drumkilla | there we go. |
02:08.31 | MikeJ[Laptop] | ok, that's funny |
02:08.41 | MikeJ[Laptop] | it is a mad party over there |
02:09.04 | Cresl1n | hey.... is somebody looking at us? |
02:09.24 | Cresl1n | we need to get some mad lights in here |
02:09.34 | MikeJ[Laptop] | hehe |
02:10.13 | Cresl1n | woohoo! |
02:10.15 | *** join/#asterisk file[mac] (~jcolp@mctn1-3494.nb.aliant.net) |
02:10.20 | Qwell | omg its file! |
02:10.21 | *** join/#asterisk Inv_arp (junya@adsl-3-248-195.mia.bellsouth.net) |
02:10.22 | Cresl1n | we need some lights in #asterisk too :-) |
02:10.25 | file[mac] | yes it's me |
02:11.49 | Qwell | file[mac]: Just asked my boss about cluecon...let's see how he takes it. :D |
02:12.21 | Cresl1n | how can we get some lights in #asterisk? |
02:12.34 | Qwell | what kind of lights? |
02:12.42 | Qwell | leds, christmas lights, what? |
02:13.46 | Cresl1n | dance club lights |
02:13.52 | Qwell | strobe lights? |
02:14.12 | file[mac] | Cresl1n makes me happy! |
02:14.21 | jets | yay for cluecon |
02:15.06 | Cresl1n | woohoo! |
02:15.16 | Cresl1n | I'm hyper right now :-) |
02:15.21 | Cresl1n | I just drank a can of redbull |
02:15.27 | Cresl1n | (a.k.a. happy juice) |
02:15.29 | drumkilla | REDBULLREDBULLREDBULL |
02:16.01 | Cresl1n | woohoo! |
02:16.02 | Cresl1n | kram!!! |
02:16.17 | drumkilla | let's have a dance party |
02:16.19 | Cresl1n | good music |
02:16.38 | Juggie | hah |
02:16.49 | drumkilla | i'll just dance in my chair |
02:16.53 | drumkilla | while I code |
02:17.00 | Juggie | a dance party with only guys |
02:17.02 | file[mac] | can I dance too?!? |
02:17.05 | Juggie | doesnt sound like much of a party |
02:17.10 | file[mac] | unfortunately drumkilla and Cresl1n don't swing that way |
02:17.10 | Cresl1n | oooohhh..... |
02:17.17 | Cresl1n | there's got to be a chick somewhere here |
02:17.27 | Juggie | i dont believe it |
02:17.27 | Cresl1n | :-) |
02:17.34 | Cresl1n | hrm.... |
02:17.35 | drumkilla | hrm ... it's looking pretty rough |
02:17.40 | Cresl1n | yeah, kind of bleak |
02:17.42 | file[mac] | Cresl1n: come over to the dark side! |
02:17.46 | Cresl1n | no! |
02:17.47 | Juggie | Cresl1n, i got the quote from Bill Hall, its gone to management now. |
02:17.50 | Cresl1n | =) |
02:17.52 | Cresl1n | cool |
02:18.19 | *** join/#asterisk Moc[Toronto] (~mochouina@142.131.66.185) |
02:18.20 | file[mac] | hippity hoppity! |
02:18.36 | Moc[Toronto] | hi everyone |
02:18.42 | file[mac] | hey Moc |
02:18.45 | drumkilla | wheeeeeeee |
02:18.47 | MikeJ[Laptop] | hey moc.. |
02:18.49 | Juggie | file, you going to cluecon? |
02:18.57 | MikeJ[Laptop] | what are you doing in toronto... |
02:18.58 | file[mac] | yes |
02:19.02 | MikeJ[Laptop] | file, of course he is |
02:19.02 | Cresl1n | drumkilla, are you going to cluecon? |
02:19.10 | Juggie | file, you swinging by to get me on the way? |
02:19.12 | file[mac] | look at http://www.cluecon.com/, specifically the schedule |
02:19.13 | Moc[Toronto] | doing a national intranet site.. |
02:19.25 | file[mac] | Juggie: haha |
02:19.33 | Juggie | well your comming from NB :) |
02:19.35 | Moc[Toronto] | I gota get brian to change my name on the schedule |
02:19.40 | Juggie | you gotta pass through ontario :P |
02:19.58 | file[mac] | I might go through Pearson airport |
02:20.14 | file[mac] | Moc[Toronto]: I can get it changed |
02:20.16 | PTG123 | hey moc hows the fax client coming? |
02:20.23 | file[mac] | what do you want it to say? |
02:20.33 | Juggie | file, drive all the way to pearson from nb? why not just fly from nb if your gonna fly |
02:20.40 | file[mac] | Juggie: I am :) |
02:20.46 | drumkilla | Cresl1n: if I am sent there, then yes |
02:20.47 | file[mac] | but I will probably have a connection at Pearson |
02:20.49 | Juggie | ahhhh |
02:21.10 | Juggie | i looked @ a flight today, it was 400$ tax in |
02:21.18 | file[mac] | mine are coming out to $530 |
02:22.17 | Juggie | driving would take 12hrs of straight driving |
02:22.21 | Juggie | which i'd like to avoid |
02:22.44 | Qwell | Whats a good travel site? |
02:22.47 | *** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca) |
02:22.57 | Qwell | hotels.com is the only one I've ever used...but hotel is included with cluecon |
02:23.11 | Juggie | www.expedia.com www.travelocity.com |
02:23.11 | Moc[Toronto] | rr |
02:23.17 | Qwell | expedia, thats the one |
02:23.39 | file[mac] | travelocity is what is giving me the best prices |
02:23.52 | Qwell | What is the airport code? |
02:24.00 | Qwell | it's like, o'hare airport, right? |
02:24.08 | NewSole2 | travelocity are assholes |
02:24.17 | file[mac] | Moc[Toronto]: I'm getting it fixed for you |
02:24.17 | asterisk99 | Does anyone here use Sipura 1001? I have a new unit and it will not ring the phone, yet I can go offhook and take the call :( |
02:24.37 | Moc[Toronto] | getting what fixed ? |
02:24.42 | file[mac] | your name |
02:24.48 | Moc[Toronto] | oh ok thanks |
02:24.52 | file[mac] | yw |
02:25.33 | Moc[Toronto] | Marc Chouinard is my father brother |
02:25.43 | Qwell | yeah, is it midway, or o'hare? |
02:25.50 | asterisk99 | O'Hare = ORD |
02:26.57 | asterisk99 | Midway = MDW and Meigs (a la Flight Sim) = CGX |
02:26.58 | Juggie | i doubt i could get work to pay for cluecon |
02:27.07 | Juggie | i dont have passport either |
02:27.10 | Juggie | i should do that |
02:27.29 | Qwell | I'm gonna try, but I need exact costs first...sucks working for a bank |
02:27.40 | MikeJ[Laptop] | asterisk99, meigs does not take aircraft anymore.. it got bulldozed |
02:27.54 | Juggie | i already have them semi convinced to go to san francisco |
02:28.09 | asterisk99 | Now ass for ass*****, they are the ones who named the Canadian airports (like YYZ = Toronto?????????????) |
02:28.13 | MikeJ[Laptop] | that was a great airport to fly into.. second fav to toronto city senter |
02:28.24 | MikeJ[Laptop] | center |
02:28.37 | Moc[Toronto] | ?? |
02:28.44 | Juggie | airport codes are fine |
02:28.45 | asterisk99 | Centre ---- Canadian spelling ;) |
02:28.54 | MikeJ[Laptop] | hehe |
02:28.58 | Juggie | dont need to give away the name of the city |
02:29.01 | Juggie | its just a code |
02:29.22 | MikeJ[Laptop] | great airport, point at the CN tower and go till you almost hit it, then land... |
02:29.25 | Moc[Toronto] | toronto is boring... |
02:30.12 | Juggie | i like visiting toronto |
02:30.14 | asterisk99 | true true... but YEG for Edmonton???? |
02:30.14 | Juggie | but i woudnt live there |
02:30.39 | Juggie | asterisk99, st. john's is YOW its all the same :) |
02:31.52 | asterisk99 | I worked in the RBC tower in toronto when the Blue Agels came screaming down Yonge St, tunred right on Bay St and headed toowards the CN tower... I was on the 18th floor, they were level with the 17th floor |
02:32.24 | asterisk99 | Does anyone here use Sipura 1001? I have a new unit and it will not ring the phone, yet I can go offhook and take the call :( |
02:32.25 | *** join/#asterisk kb1_kanobe (~krisbouti@d154-20-43-155.bchsia.telus.net) |
02:32.33 | *** join/#asterisk iq (~iq@70-59-166-10.omah.qwest.net) |
02:34.20 | heison | moc[toronto]: when are u leaving? |
02:35.08 | *** join/#asterisk IsMe (~me@218.111.224.205) |
02:35.48 | IsMe | hi, i am looking for a developer that can do CID callback service |
02:35.58 | *** part/#asterisk rene- (~root@200.78.176.114) |
02:36.06 | asterisk99 | Juggie: "YOW" is the Scam Capital of Canada... Ottawa St. John's (Nfld) is a really obscure YYT |
02:38.57 | Qwell | This site really needs to list what the nearest airport is... |
02:39.33 | Qwell | oh, it does :D |
02:41.14 | iCEBrkr | bkw_: wake up |
02:42.45 | Juggie | asterisk99, ahh YOW is halifax is it |
02:43.03 | asterisk99 | Ottawa |
02:43.13 | Juggie | ahh |
02:43.22 | Juggie | thats where i am |
02:43.24 | Juggie | and i go to st. johns alot |
02:43.29 | Juggie | i get the codes mixed up |
02:44.09 | asterisk99 | I guess all the Sipura users are waitig in lline for the 1st showing of "Revenge of the Sith" |
02:44.25 | asterisk99 | 01:15:00 to go |
02:44.45 | *** join/#asterisk thetalon (~Administr@pcp05736786pcs.norstn01.pa.comcast.net) |
02:45.36 | asterisk99 | juggie: Been to St. John's --- great town!!!! ---- also been to Memorial U campus |
02:46.15 | Juggie | asterisk99, i am from st. johns :) its all about the beer |
02:46.27 | Moc[Toronto] | heison: Friday at 5pm |
02:46.35 | Moc[Toronto] | btw hi there ;) |
02:46.38 | asterisk99 | Juggie: Beer?!?!?!?!?!?!? You mean Screech :) |
02:47.02 | Juggie | screech is just what we give to the people who visit, no one from nf drinks that :) |
02:47.28 | Qwell | sweet, $275 round trip |
02:47.33 | Qwell | nonstop :) |
02:47.39 | asterisk99 | Juggie: Just so lomg as you're not a spilly-billy with your beer |
02:47.47 | Juggie | i dont spill |
02:48.18 | iCEBrkr | Anyone know of any eagi examples that work? |
02:51.20 | mmlj4 | anyone do wireless VoIP? i have a few questions (I know, just ask, etc.) |
02:51.25 | Juggie | wireless how |
02:51.30 | Juggie | wireless voip phones? |
02:51.57 | mmlj4 | i need to bridge 2 offices, 50 meters apart |
02:52.16 | mmlj4 | thinking of using linksys or d-link wireless bridges |
02:52.22 | asterisk99 | can someone see if voxilla.com is working??? |
02:52.28 | mmlj4 | WET11, for example |
02:52.32 | Juggie | that should be easy 50 m is nothing |
02:53.02 | mmlj4 | i'm worried about latency... and also packet aggregation, say with G gear (dunno if it uses that) |
02:56.46 | *** join/#asterisk jape (~jpb@sixshooter.v6.thrupoint.net) |
02:57.25 | *** join/#asterisk DrAk0 (~Luis@200.109.195.19) |
02:57.40 | DrAk0 | hello, can i make asterisk to use my external modem? |
02:57.51 | DrAk0 | so recive and make the phonecalls from it? |
02:58.04 | thetalon | don't go there DrAk0 |
02:58.12 | thetalon | you' |
02:58.22 | thetalon | you'll be better off with pure VoIP solution |
02:58.33 | thetalon | get a NuFone account |
02:58.42 | DrAk0 | is formy house i have no money for special hardware so i would like to use what i have in my hands |
02:58.44 | thetalon | or some other guy like teliax |
02:59.21 | *** part/#asterisk jape (~jpb@sixshooter.v6.thrupoint.net) |
02:59.37 | DrAk0 | thetalon, but it is possible what i want? |
03:00.15 | *** join/#asterisk ilium (~ilium@203.94.178.214) |
03:00.19 | asterisk99 | DrAk0: Look at the Digium WildCard... it'll do it for ya |
03:00.27 | ilium | hi |
03:00.49 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
03:01.00 | mmlj4 | DrAk0: you can buy specialized modems for $17 USD delivered, on ebay (anyone wanna say how good those are?) |
03:01.01 | asterisk99 | DrAk0: I;d send you a link, but it looks like digium.com is down |
03:01.11 | implicit | wow this is fun |
03:01.15 | asterisk99 | Drak0: No it's back up |
03:01.27 | ilium | can someone have a look at this output from asterisk -c: http://pastebin.ca/12121 |
03:01.37 | ilium | i have just installed freebsd 5.4 |
03:01.46 | ilium | updated via cvsup to STABLE |
03:01.56 | ilium | installed zaptel libpri and asterisk |
03:02.07 | ilium | i dont know how to troubleshoot this problem |
03:02.17 | mmlj4 | hey kram: you *might* see me at the bootcamp in florida in december, if I can find the cash to go |
03:02.24 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
03:02.52 | ilium | i have also built a custom kernel to include timing |
03:02.56 | asterisk99 | Drak0: You would need just 1 FXO daughter-card on a Digium TDM400P ($133 from Digium) |
03:03.49 | ilium | anyone ? |
03:04.03 | DrAk0 | the problem is im in south america that means shipping is very expensive plus my country has problem with us$ and getting them is really hard and well im like out of money too, however i have a motorola external modem thas why i was asking if i could use it |
03:04.40 | mmlj4 | DrAk0: no, those won't work |
03:04.42 | Nugget | ilium: do you actually need skinny or oss? |
03:04.48 | mmlj4 | the problem is drivers |
03:05.00 | DrAk0 | aahh |
03:05.11 | Nugget | skinny is for cisco phones and oss is for running a local soundcard. neither are very worthwhile. |
03:05.12 | DrAk0 | =/ |
03:05.14 | ilium | nope |
03:05.32 | Nugget | clean up your modules.conf then. |
03:05.42 | ilium | how do i disable them - i have installed asterisk on this box just last week and it all worked |
03:05.46 | ilium | i dont know what chaned |
03:06.12 | ilium | actually i have a sneaking suspicition i had 1.0.6 installed last week, and this time round i have ended up with 1.0.7 |
03:06.19 | Nugget | noload => chan_oss.so |
03:06.23 | Nugget | ^ in modules.conf |
03:06.25 | *** join/#asterisk loud (~ariel@omfg.wtf.no) |
03:06.59 | ilium | what about skinny |
03:07.00 | Nugget | although I'm not sure why you'd need to say that specifically |
03:07.20 | Nugget | did you install asterisk from ports or build it by hand? |
03:07.26 | ilium | from ports |
03:07.32 | ilium | is that bad ? |
03:07.35 | Nugget | I've only ever done it by hand in freebsd, so I'm unsure what the port might do. |
03:07.42 | Nugget | I don't know if it's bad or not, I've just never done it |
03:07.56 | ilium | ive been working woth the guy that ports it to freebsd |
03:08.03 | ilium | he has been helpful, so i was useing his ports |
03:08.09 | Nugget | makes sense. |
03:08.17 | ilium | how do i remove skinny |
03:08.29 | ilium | it starts at least after removing oss |
03:08.31 | ilium | thatnks! |
03:08.38 | Nugget | cool. |
03:09.04 | ilium | <PROTECTED> |
03:09.04 | ilium | odule: Unable to open IAX timing interface: No such file or directory |
03:09.14 | ilium | i would assume i should be using IAX |
03:09.29 | ilium | is modules.comf the only place this stuff loads form |
03:09.42 | Nugget | I don't know what that means, sorry. |
03:09.58 | Nugget | on my freebsd asterisk servers I have never had to change modules.conf at all, so I dunno. |
03:10.12 | ilium | what versio nr u running |
03:10.18 | ilium | im on 5.4 STABLE |
03:10.30 | Nugget | 5.4 stable, but it's several months old |
03:10.39 | *** join/#asterisk Avrice (~James@mmds-216-19-41-194.mm.az.commspeed.net) |
03:10.43 | Nugget | 5.3 actually |
03:10.53 | Nugget | and an asterisk stable (1_0) checkout from the same time period |
03:10.59 | ilium | hmmmm |
03:11.14 | *** part/#asterisk Cresl1n (~matt@216.207.245.23) |
03:11.16 | ilium | there is nothing in modules.conf about skinny or IAX2 |
03:11.24 | Nugget | I used ztdummy for a while, but I don't do anything zaptel now. |
03:11.37 | mmlj4 | DrAk0: look here, at the Intel modems: http://www.voip-info.org/wiki-Asterisk+Hardware |
03:11.59 | mmlj4 | you can probably find those (be cafeful of the exact chipset model) |
03:12.27 | DrAk0 | mmlj4, let me check |
03:14.48 | sudhir492 | help |
03:15.01 | sudhir492 | oops, wrong window |
03:16.52 | ilium | can someone point me to a known good freebsd build doc |
03:17.16 | Nugget | I dunno. |
03:17.24 | Nugget | I checkout the source, make, and then make install. |
03:17.34 | Nugget | I've never had a problem doing it that way |
03:17.38 | ilium | can you tell me how to do that ? |
03:17.40 | niZon | ~seen boris |
03:17.41 | jbot | boris <boris@S01060040ca1e5b54.wp.shawcable.net> was last seen on IRC in channel #asterisk, 5d 5h 51m 37s ago, saying: 'what?'. |
03:17.42 | Nugget | I just did. |
03:18.01 | ilium | ~seen darwin35 |
03:18.03 | jbot | darwin35 is currently on #asterisk (1d 17h 34m 55s). Has said a total of 35 messages. Is idling for 1d 5h 43m 31s |
03:18.58 | ilium | Nugget: can you tell me how you check out the sauce |
03:19.17 | Nugget | set CVSROOT to the asterisk cvs server |
03:19.23 | Nugget | cvs co -r v1-0 asterisk |
03:19.51 | ilium | ok so you run 1.0 ?? isnt it up to 1.0.7 ? |
03:19.59 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
03:19.59 | Nugget | v1-0 is "stable" |
03:20.01 | Nugget | not 1.0.0. |
03:20.09 | ilium | ok cool |
03:20.10 | Nugget | it |
03:20.15 | Nugget | it's like releng_5 |
03:20.24 | Avrice | where/what is the best place/doc to read on hooking up sjphone to asterisk over the internet |
03:20.28 | ilium | ok |
03:21.05 | *** join/#asterisk tessier (~treed@203.210.216.1) |
03:21.17 | Kumbang | guys, i have problem sending fax using txfax |
03:21.21 | Kumbang | *CLI> -- Attempting call on Zap/g0/801 for out_fax@ext-fax:1 (Retry 1) |
03:21.21 | Kumbang | <PROTECTED> |
03:21.21 | Kumbang | <PROTECTED> |
03:21.21 | Kumbang | DIS with final frame tag |
03:21.21 | Kumbang | In state 9 |
03:21.21 | Kumbang | Start tx document |
03:21.23 | Kumbang | CFR with final frame tag |
03:21.25 | Kumbang | In state 4 |
03:21.27 | Kumbang | Start tx page |
03:21.31 | Kumbang | RTN with final frame tag |
03:21.33 | Kumbang | In state 14 |
03:23.10 | *** part/#asterisk lilneon (~tj_r3@cuscon14794.tstt.net.tt) |
03:28.23 | DrAk0 | mmlj4, a friend who lives near me... has a Intel 537EP |
03:28.26 | DrAk0 | it works? |
03:31.34 | blitzrage | Kumbang: generally with that number of lines, etiquette says to use pastebin |
03:31.42 | blitzrage | Kumbang: for future ref. |
03:31.53 | MikeJ[Laptop] | ~pastebin |
03:31.55 | jbot | somebody said pastebin was a place to paste your stuff without flooding the channel - try http://pastebin.ca |
03:32.44 | blitzrage | ilium: checking out stable from CVS is not the same as grabbing a 1.0.x release tarball |
03:33.53 | blitzrage | ilium: CVS -r v1-0 is the latest patches to the stable branch, post release - but there CAN be problems, so its not considered STABLE - always use the releases for production systems |
03:34.09 | Kumbang | sorry |
03:34.34 | MikeJ[Laptop] | Nugget, v1-0 is release, not stable... don't call it stable |
03:34.48 | MikeJ[Laptop] | it jsut confuses people |
03:35.11 | MikeJ[Laptop] | cmon blitz |
03:35.14 | MikeJ[Laptop] | :) |
03:36.03 | Nugget | digium calls it "the current stable release" |
03:36.08 | Nugget | so I do the same. |
03:36.58 | Nugget | it's equivalent to freebsd's "stable", so the term is hardly unfamiliar to ilium. |
03:37.48 | Nugget | perhaps you're just unfamiliar with how us freebsd nuts use the term "stable" |
03:37.52 | MikeJ[Laptop] | everyone is too serious in here tonight |
03:38.00 | blitzrage | MikeJ[Laptop]: screw you! |
03:38.05 | MikeJ[Laptop] | ass |
03:38.09 | blitzrage | hat |
03:38.16 | MikeJ[Laptop] | don't make me kick your tired ass |
03:38.27 | MikeJ[Laptop] | you wanna take this outside? |
03:38.29 | blitzrage | I'll beat you on my worst day, and your best! |
03:38.38 | blitzrage | MikeJ[Laptop]: meet me at the backstop at recess, you're dead! |
03:38.43 | MikeJ[Laptop] | alright, cmon over... |
03:38.51 | MikeJ[Laptop] | backstop... |
03:38.53 | blitzrage | MikeJ[Laptop]: pffft - you come here |
03:38.55 | MikeJ[Laptop] | you are tired |
03:39.11 | MikeJ[Laptop] | chicken |
03:39.20 | blitzrage | backstop - fencing to stop the baseball behind the catcher |
03:39.28 | blitzrage | must be a Canadian term... :) |
03:39.40 | MikeJ[Laptop] | canuk chicken! |
03:40.01 | blitzrage | you can't even spell canuck right you lousy yank |
03:40.15 | MikeJ[Laptop] | 1/2 breed, get it straight |
03:40.19 | blitzrage | lol |
03:40.22 | MikeJ[Laptop] | eh? |
03:40.45 | blitzrage | damn straight |
03:40.50 | MikeJ[Laptop] | my folks grew up in windsor ontario, eh. |
03:41.01 | blitzrage | I grew up in Sarnia, ON |
03:41.07 | MikeJ[Laptop] | poor bastard |
03:41.12 | blitzrage | quite |
03:41.15 | *** part/#asterisk thetalon (~Administr@pcp05736786pcs.norstn01.pa.comcast.net) |
03:41.16 | blitzrage | now I live in Toronto :) |
03:41.21 | MikeJ[Laptop] | eh? |
03:41.32 | blitzrage | ha |
03:41.46 | Juggie | usa sucks :P |
03:41.47 | blitzrage | aye |
03:41.50 | blitzrage | aye! |
03:42.11 | MikeJ[Laptop] | I have lots of family in toronto |
03:42.20 | NewSole2 | yankies.... yank... yank... yank.... |
03:42.43 | Nuxi | sounds like a lot of usa-envy going on |
03:43.04 | blitzrage | envy? lol |
03:43.13 | MikeJ[Laptop] | it's an insecurity thing... it's ok |
03:43.14 | Juggie | Nuxi, at first glance phpagi 2 may have solved the defuncts |
03:43.15 | MikeJ[Laptop] | :D |
03:43.17 | Juggie | i woudnt call it envy |
03:43.24 | Juggie | i think i like my health care system |
03:43.32 | Juggie | and lack of RIAA/MPAA gayness |
03:43.40 | NewSole2 | there is no envy looking at shot spots on the walls |
03:43.43 | blitzrage | ok, I have to go before I say something you regret |
03:43.48 | NewSole2 | yankies.... yank... yank... yank.... |
03:43.51 | blitzrage | lol |
03:43.57 | blitzrage | I like that guy :) |
03:44.20 | NewSole2 | :P |
03:44.24 | MikeJ[Laptop] | Juggie, you mean the one that meany my gradmother had to come to the states in order to get compatent care, instead of waiting 9 months for a surgey and duying.. yeah, ontario's medical care rocks. |
03:44.49 | MikeJ[Laptop] | till you acutally need care |
03:45.01 | Juggie | MikeJ[Laptop], i've been in the health care system its been fine for me |
03:45.08 | MikeJ[Laptop] | #asterisk politics ... |
03:45.09 | MikeJ[Laptop] | hehe |
03:45.31 | blitzrage | woohoo! finally got that damn windows sticker off my laptop |
03:45.36 | MikeJ[Laptop] | well, good luck, they have a bad habbit of not having specialist cuz they all moved to the states |
03:45.38 | blitzrage | those things are on there GOOD |
03:45.45 | MikeJ[Laptop] | where they can get paid. |
03:45.49 | Nuxi | studies show that removing the widows sticker makes the hardware more stable. |
03:45.54 | Juggie | MikeJ[Laptop], funny, the guy i needed was available no problem. |
03:46.14 | tangel | how can i let people enter an extension after i answer an inbound analog call? |
03:46.23 | MikeJ[Laptop] | dunno, she needed spinal surgury, and there was a 9 month+ wait. |
03:46.24 | blitzrage | tangel: show application background |
03:46.32 | MikeJ[Laptop] | she got in in a week in the US. |
03:46.41 | Nuxi | This is the same kind of talk my parents have about which state is better. It's good to see so much patriotism. |
03:46.53 | Juggie | of course, because in the usa the people with money get help and the people who have no money die |
03:47.06 | MikeJ[Laptop] | not patriotism, really |
03:47.06 | Juggie | theres no perfect system |
03:47.21 | Juggie | you can have money rules, or wait your turn |
03:47.28 | Juggie | either way someone wont be happy |
03:47.36 | tangel | blitzrage, those are the commands? |
03:52.31 | *** join/#asterisk wvbroadband (~User@pool-151-205-40-176.clrk.east.verizon.net) |
03:57.14 | ilium | this sux |
03:57.16 | *** join/#asterisk onet (~JohnM@VDSL-151-118-4-13.DNVR.QWEST.NET) |
03:57.40 | ilium | can someone tell me how to download asterisk v1.0 stable and compile it on freebsd 5.4 STABLE |
03:58.37 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
03:58.50 | Mavvie | ilium: cvsup your ports collection, then cd /usr/ports/net/asterisk and make install |
03:59.53 | ilium | yeah had that installed but it wouldnot start, errors about IAX2 module not found |
04:00.03 | ilium | grrrr this is gettign so frustrating |
04:00.36 | Mavvie | ilium: mind your bloodpressure. |
04:00.57 | *** join/#asterisk Kernel_Core (Raph@236.228.dial-up.xter.net) |
04:02.38 | *** join/#asterisk openfly (mjoyce@judecca.aculei.net) |
04:02.42 | openfly | howdy. |
04:02.57 | openfly | anyone here ever worked with cisco 7902 ip phones and asterisk? |
04:02.57 | ilium | yeah well its hard when youre delaing with freebsd and asterisk ! |
04:03.11 | ilium | all i want to do is see it work with 1 * box and two SIP sopftphones |
04:03.16 | ilium | 2 weeks later ........ |
04:03.34 | |Vulture| | ilium: whats your problem? |
04:03.42 | ilium | i just cant get * to compile |
04:03.49 | ilium | and now i dont know where i am |
04:03.50 | |Vulture| | in BSD? |
04:03.55 | ilium | yeah in freebsd |
04:04.03 | |Vulture| | sorry Im a linux guy |
04:04.09 | ilium | i have made a custom kernel with extra commands to enable timing |
04:04.16 | |Vulture| | you tried the walkthrough on the wiki? |
04:04.16 | ilium | what linux do you reccomend |
04:04.25 | ilium | yeah i think so |
04:04.28 | |Vulture| | ilium: I use FC3 works wonderful for me |
04:04.31 | ilium | i ve tried liek 10 times now |
04:04.34 | openfly | icky. |
04:04.34 | Mavvie | ilium: try contacting the ports maintainer. |
04:04.43 | ilium | yeah richard was helping me |
04:04.44 | openfly | asterisk compiled fine on debian stable for me... |
04:04.48 | openfly | hasn't done much else yet |
04:04.50 | ilium | and at one stage i had it sort of workign |
04:04.56 | openfly | but seems to be going well thus far =P |
04:04.58 | |Vulture| | install... import fedora keys, yum update... |
04:05.10 | Mavvie | I don't know any richard, I know a maxim as port maintainer. |
04:05.15 | |Vulture| | and then the udev tweak for zaptel and your good to go |
04:05.22 | ilium | hmmmmm |
04:05.27 | ilium | any else knwo freebsd ? |
04:05.45 | *** part/#asterisk mogorman (~mogorman@207.111.174.1) |
04:05.59 | openfly | so yeah... i am looking at a cme setup on a 1760 and link to asterisk unless someone can point me to a working cssp driver for a 7902g. |
04:06.07 | Mavvie | if you relax for a second, and tell us exactly what yuo have and what doesn't work, it would be much easier. |
04:06.21 | ilium | ok |
04:06.27 | ilium | i have installed freebsd 5.4 |
04:06.33 | ilium | updated the ports with cvsup |
04:06.43 | ilium | updated src with cvsup |
04:06.55 | ilium | make buildworld |
04:06.59 | ilium | amke installworld |
04:07.01 | ilium | rebooted |
04:07.27 | ilium | installed zaptel |
04:07.30 | ilium | installed libpri |
04:07.30 | Mavvie | with regarding to the asterisk port. |
04:07.33 | ilium | installed asterisk |
04:07.42 | ilium | went to start asterisk |
04:07.55 | ilium | and had errors with IAX2 |
04:08.15 | Mavvie | brilliant! now take the verbosity of the first part of your story and use it on your last line. |
04:08.28 | openfly | you ran in verbose mode asterisk? |
04:08.33 | ilium | yes |
04:08.36 | openfly | and did you get a copy of said errors? |
04:08.41 | ilium | http://pastebin.ca/12121 |
04:09.13 | Mavvie | well, that's a FAQ. |
04:09.21 | Mavvie | ~ztdummy |
04:09.22 | jbot | methinks ztdummy is zaptel timing source which uses a usb-ohci compatible usb controller as source. (part of zaptel cvs) |
04:09.22 | openfly | did you "make samples" |
04:09.42 | ilium | huh |
04:09.47 | *** join/#asterisk Defraz (~t0tal@65.103.222.4) |
04:09.48 | Flav | what cards are people using for their home asterisk setups? tdm400? |
04:10.09 | Mavvie | ilium: did you google for "Unable to open IAX timing interface" ? |
04:10.29 | openfly | well thing is i don't see it loading any valid config files |
04:10.29 | ilium | no i cam to the asterisk irc channel to ask if people had seen these error before |
04:10.31 | openfly | at all.. |
04:10.38 | Mavvie | ilium: try that first. |
04:10.42 | ilium | ? |
04:10.59 | ilium | isnt this irc channel supposed to be used for help too ? |
04:11.10 | Mavvie | no, you came to the asterisk irc channel bitching about that asterisk didn't work on FreeBSD. |
04:11.27 | Mavvie | without giving any relevant information. |
04:11.31 | ilium | this is the exact reason i stay away from these channels |
04:11.37 | ilium | being flamed by people liek you |
04:11.44 | Mavvie | I haven't even started flaming yet. |
04:11.46 | ilium | i wasnt bitchin i was asking for assistance |
04:11.49 | nine76 | a valid answer has been given though ilium |
04:11.55 | ilium | whatever |
04:12.04 | openfly | ilium msg |
04:12.05 | Mavvie | You are the weakest link... Goodbye! |
04:12.13 | openfly | mavvie you are being a jerk. |
04:12.17 | Mavvie | (not a real great one, but that was a flame) |
04:12.17 | openfly | and antagonizing him |
04:15.48 | ilium | hey mavvie - to add insult to injury it would seem you are harrasing a fellow aussie ! |
04:15.58 | ilium | barnet.com.au ???? |
04:16.16 | Mavvie | I'm not an aussie, I'm a dutchman, stuck on this goddesserted island. |
04:16.17 | Mavvie | :-) |
04:16.32 | ilium | that explains everythign then |
04:16.39 | Mavvie | are you in the Sydney area? |
04:16.42 | ilium | brisbane |
04:16.57 | Mavvie | too far, otherwise I would have come over and helped you. |
04:17.09 | Mavvie | nice counter flame :-) |
04:17.11 | ilium | it would have been easy to find |
04:17.31 | ilium | driev along corronation drive and look for the freebsd server lying in the middle of the road |
04:18.33 | *** join/#asterisk Godsey (lanny@goofball.md5.com) |
04:19.05 | Mavvie | Now that shows you're been in the sun too long... You could have reused the harddisk. |
04:19.35 | ilium | no that was destroyed from me punching the box so hard before i threw it on the road |
04:20.19 | openfly | right... was that before or after you too hunted crocodiles and fought kangaroos with dead wallabee skins? |
04:20.20 | openfly | =P |
04:20.32 | Mavvie | Impressive. These things are supposed to survive 5G. Shows that you're more muscles than brains. |
04:20.42 | ilium | it was a big hit |
04:20.50 | openfly | that's not a server.... |
04:20.54 | openfly | this is a server! |
04:20.56 | openfly | =P |
04:21.02 | Mavvie | hehe |
04:21.19 | ilium | actually........i htought it woudl be funny to scare a fellow work mate, and dropped a phone book on his desk |
04:21.31 | ilium | clunl cluck clunk |
04:21.32 | openfly | kill him? |
04:21.38 | ilium | hard drive died |
04:21.53 | ilium | i fent bad |
04:21.54 | openfly | there was a harddrive in the phone book? |
04:22.05 | openfly | damn the outback has some crazy technology... |
04:22.08 | Mavvie | openfly: they're very advances there in .au |
04:22.31 | ilium | we only got power on over here last month, so we are still learning all about computers |
04:22.41 | openfly | must be those aboriginees i heard they have been stealing alien technology for years. |
04:23.35 | openfly | yeah so question. |
04:24.06 | openfly | should i a. try to link my 7902 ip phones to asterisk directly, or b. run them through cisco call manager on a 1760 and link that to asterisk? |
04:24.10 | openfly | which will be easier? |
04:25.04 | MikeJ[Laptop] | openfly, which protocol? |
04:25.22 | openfly | 7902 has to be skinny |
04:25.25 | openfly | =( |
04:25.49 | openfly | sccp driver is confirmed as far as i know on the 7902 |
04:25.55 | openfly | but may work on 7905. |
04:26.10 | openfly | err |
04:26.14 | openfly | unconfirmed |
04:26.15 | openfly | * |
04:29.44 | openfly | meh... |
04:29.59 | openfly | i guess i should try the cme way first.. |
04:30.02 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
04:30.12 | openfly | good study for ccnp =P |
04:31.12 | *** join/#asterisk iswm (iswm@iswm.user) |
04:32.06 | *** join/#asterisk [hC] (~hardcore@c-65-34-203-137.hsd1.fl.comcast.net) |
04:32.14 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
04:32.21 | MikeJ[Laptop] | the skinny drivers are there, but are immature for some phones, I have been using sip firmware to *, but you loose some of the functionality like that... |
04:32.24 | MikeJ[Laptop] | sooo.... |
04:32.27 | [hC] | Is anyone here using a dial macro that handles both their incoming and outgoing calls? Most peoples examples have one macro for in, and one for out. |
04:33.20 | Mavvie | [hC]: for PRI channels? |
04:33.26 | [hC] | For anything and everything |
04:33.33 | [hC] | I have written one, but ive run into a bit of a snag. |
04:34.15 | Mavvie | [hC]: I know it can be done for PRIs, but it's hard for with FXO/FXS's because you don't know where they are coming from except for the context assigned with them. |
04:34.26 | [hC] | Right, and thats what i test based on |
04:34.26 | ManxPower | . |
04:34.28 | [hC] | source context |
04:34.39 | [hC] | which is alright, as long as your pbx doesnt originate calls from inside services |
04:34.55 | [hC] | (call return, call forwarding, etc) cause then you have to keep track of every context that a call can originate from |
04:35.17 | [hC] | i might just split it out into two macros again, I was just wondering if people had really done it, and if so, i wanted to take a look at how. |
04:38.56 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
04:39.59 | openfly | hrmm... |
04:40.01 | openfly | odd... |
04:40.34 | openfly | 7902 + netgeat fs8p 4 port poe switch != powered 7902... |
04:40.42 | sudhir492 | how to save a bunch of variables in userfield in CDR? |
04:41.23 | sudhir492 | documentation is very incomplete in wiki |
04:41.44 | openfly | hehe its a wiki when you find out.. complete it =P |
04:42.24 | sudhir492 | sure. I have updated wiki before |
04:47.24 | niZon | openfly: cisco uses some weird POE protocol |
04:48.03 | Juggie | not anymore |
04:48.04 | Juggie | they did |
04:48.12 | Juggie | they use the standard now |
04:48.15 | niZon | ah ok |
04:48.21 | niZon | maybe he has an old 7902 :P |
04:48.36 | Juggie | my 7960's support both |
04:48.43 | Juggie | they run on cisco poe and normal poe |
04:48.50 | niZon | I wouldn't mind a 7960 |
04:48.50 | *** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com) |
04:48.57 | niZon | do you use chan_sccp? |
04:49.11 | Juggie | no |
04:49.15 | niZon | SIP? |
04:49.17 | Juggie | 7960 has a sip firmware |
04:49.27 | niZon | ah |
04:49.31 | niZon | yes |
04:49.48 | Juggie | i dont like the 7960 anymore tho |
04:49.57 | Juggie | the mitel 5220's are alot easier to manage/maintain |
04:50.13 | niZon | do they have the XML stufF? |
04:50.24 | Juggie | nope |
04:50.29 | Juggie | but i dont care about that |
04:50.36 | niZon | ah |
04:51.01 | Juggie | if it was my decision everyone would have softphones |
04:51.03 | niZon | I want a phone that supports XML stuff |
04:51.13 | Juggie | xml stuff isnt that exciting |
04:51.21 | niZon | yeah but I have a use for it |
04:51.23 | Juggie | the good thing is it can do a directory |
04:51.25 | Juggie | other then that |
04:51.30 | Juggie | whats your use |
04:51.46 | niZon | autiomation, lighting controls and stuff |
04:51.50 | niZon | -i |
04:52.04 | Juggie | write an attendant |
04:52.22 | soundguy | I didnt know lighting controllers used XML |
04:52.26 | soundguy | well..none that I have used have |
04:52.31 | niZon | pff |
04:52.42 | niZon | soundguy: I write/build my own controllers :P |
04:52.47 | Juggie | why would u use a phone |
04:52.54 | niZon | why not? |
04:52.55 | Juggie | to do something you could do on a real computer |
04:52.57 | Juggie | and easier |
04:53.00 | soundguy | heh..i just use a hog1000 :) |
04:53.02 | ManxPower | ~docs |
04:53.03 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
04:53.24 | niZon | why would you want to go to a computer, turn on the monitor, pull up IE or a control app and do stuff when you can press 2 buttons on a phone |
04:53.35 | niZon | (this is in a home setting btw) |
04:53.53 | Juggie | you could argue either way ;) |
04:54.10 | niZon | yeah |
04:54.21 | Juggie | i would argue a phone just wants to be a phone |
04:56.09 | niZon | or it could be a control terminal and a phone |
04:56.10 | niZon | lol |
04:56.44 | niZon | it's a matter of prefrence :P |
04:58.22 | |Vulture| | anyone know how you use ifconfig to set an adapter to tagged on a VLAN? |
04:59.54 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:00.58 | openfly | hrmmm |
05:02.53 | [hC] | I thought that if you had a type=user and host=dynamic in iax.conf, that that user HAD to register to you, but that does not seem to be the case. Is this true? |
05:04.40 | *** join/#asterisk HA (~mirc@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
05:04.54 | Mavvie | |Vulture|: ifconfig myvlan create, ifconfig myvlan ip.ad.dr.ess netmask ne.tm.as.k vlan tag vlandev interface |
05:04.57 | Mavvie | but that's FreeBSD. |
05:05.44 | |Vulture| | okay ill try that |
05:08.35 | ManxPower | niZon, I think some people get aroused by using a softphone. It's the only logical explaination. |
05:08.55 | ManxPower | Personally I think hardphones are better. |
05:09.12 | Juggie | arnt you going to europe :) go and stop bugging us |
05:11.15 | openfly | just spent mad bank on a poe switch... |
05:11.26 | openfly | and the darned thing doesn't appear to work with these freaking phones. |
05:11.28 | openfly | =( |
05:11.32 | openfly | imma go cry i think |
05:12.44 | ManxPower | Juggie, I leave tomorrow night |
05:13.11 | Juggie | nice.. |
05:13.13 | Juggie | have fun |
05:14.50 | *** part/#asterisk hellop (~LeeHarvey@cpe-70-93-40-171.hawaii.res.rr.com) |
05:18.22 | *** join/#asterisk shepherd (matt@pcp01541028pcs.huntsv01.al.comcast.net) |
05:20.03 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
05:25.33 | *** join/#asterisk bewest (~bewest@h48.204.255.206.cable.srcy.cablelynx.com) |
05:27.30 | bewest | which features of asterisk require a working sound card? |
05:27.57 | Qwell | dialing from the console |
05:28.06 | bewest | does playing a sound require alsa/soundcard to be working? |
05:28.13 | Qwell | don't think so |
05:29.05 | Juggie | no |
05:31.26 | *** join/#asterisk kimo_sabe (nick@zappa.azrackspace.net) |
05:31.45 | shepherd | i think it requires oss /dev/dsp |
05:32.54 | shepherd | for like music on hold maybe |
05:33.13 | *** join/#asterisk felipex (~dsfdsf@host162-91.pool8533.interbusiness.it) |
05:34.03 | Juggie | no |
05:34.05 | Juggie | it does not |
05:34.18 | Juggie | absolutely no sound card required |
05:36.24 | *** join/#asterisk jeffgus (~jeffgus@alpha.zimage.com) |
05:37.00 | bewest | when you do something like sip show channels, and it says 0 channels active. what exactly does that mean? that nothing is currently being used or nothing is set up to work? |
05:37.11 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
05:47.06 | *** join/#asterisk lehel (~root@82.79.20.17) |
05:47.28 | *** join/#asterisk rajo (~rajo@bfs.cs.uni-sb.de) |
05:47.44 | *** part/#asterisk lehel (~root@82.79.20.17) |
05:54.35 | nine76 | bewest: nothing in use |
05:54.41 | Flav | can asterisk record phone calls as .wav files or the like? |
05:54.47 | nine76 | yes |
05:54.52 | nine76 | search wiki for "monitor" |
05:54.56 | Flav | nine76: thanks |
05:55.10 | nine76 | np |
05:57.03 | *** join/#asterisk kb1_kanobe (~krisbouti@d154-20-43-155.bchsia.telus.net) |
06:04.47 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com) |
06:17.47 | bewest | how do I know for sure if my sip phone got registered with my asterisk server? and is there a way to do a simple hello world type thing |
06:18.09 | bewest | like, I pick up the phone, and asterisk simply plays a sound and hangs up |
06:18.21 | nine76 | sip show registry |
06:18.33 | nine76 | err |
06:19.16 | nine76 | as for the hello world thing, create an exten which plays a sound. i.e. exten => 1234,1,Playback(beep) |
06:19.28 | nine76 | then dial 1234 |
06:19.42 | bewest | ok |
06:19.48 | nine76 | if your connected to your * with verbosity enabled you can see the phones register |
06:20.11 | bewest | if it was exten => s, 1, Playback(beep), would that play without dialing anything? |
06:20.32 | nine76 | "-- Registered SIP '1002' at 192.168.0.1 port 5060 expires 1800" |
06:20.43 | nine76 | exten => s,no |
06:20.49 | bewest | ok |
06:21.09 | nine76 | I do not know of a way to have * do what your saying,without having the phone dial some type of number. |
06:21.18 | bewest | I see |
06:21.48 | bewest | it's moot cause my phone hasn't registered :-( |
06:22.09 | nine76 | "sip debug" may help,as does a packet sniffer |
06:22.50 | bewest | the verbosity that you mentioned, is that * setting or a phone setting? |
06:23.19 | nine76 | that would be starting * with v's. or connecting to an already running * with v's. i.e. asterisk -vvvvvvvvvvvr |
06:23.27 | *** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net) |
06:25.06 | bewest | ok |
06:25.16 | *** join/#asterisk jo3sm1th (Incogn1t0@200.77.206.43) |
06:26.08 | jo3sm1th | Does Firefly have a version that supports call waiting the one I got now only allows one line at a time which is weird since most VOIP providers allow you more than 3 lines at a time |
06:29.19 | justnulling2 | can't get sipura spa-1001 connect to astersk, any ideas? |
06:30.05 | nine76 | you followed the configuration example on the wiki? |
06:31.18 | justnulling2 | nine76: can't find it any more do yo uhave url? |
06:32.59 | nine76 | I'm looking. Cant seem to find exact docs relating to the 1001. Have you tried the 2000 example? http://www.voip-info.org/wiki-Asterisk+phone+SPA-2000 |
06:33.30 | nine76 | I would imagine they are very similar...just 1 less line on the 1001. |
06:34.39 | justnulling2 | makes sense, let me try |
06:36.32 | *** join/#asterisk north_ (faith@once.athief.net) |
06:37.04 | Qwell | hmm |
06:39.17 | north_ | Oh, apparently we share the same ~ |
06:39.46 | *** join/#asterisk rue_mohr (~dan@ip-215-222.ppp.ucc-net.ca) |
06:39.58 | rue_mohr | kb1_kanobe: hi |
06:40.08 | rue_mohr | ? |
06:40.42 | kb1_kanobe | vening |
06:40.51 | rue_mohr | ahh, excellent |
06:40.55 | rue_mohr | I have words for you |
06:41.07 | rue_mohr | wireless context |
06:41.26 | *** join/#asterisk af_ (~af@ip-132-224.sn2.eutelia.it) |
06:41.27 | rue_mohr | soekris net4801 and monowall |
06:41.35 | rue_mohr | heard?, know of? |
06:41.52 | *** join/#asterisk af_ (~af@ip-132-224.sn2.eutelia.it) |
06:43.43 | kb1_kanobe | back - sorry, was stirring soup. :-) |
06:44.00 | jo3sm1th | I can't believe Firefly apparently doesnt have call waiting features |
06:48.11 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk) |
06:48.15 | ellvis | good morning |
06:48.47 | rue_mohr | good night |
06:48.58 | *** join/#asterisk Corydon76-home (grey@pcp08665860pcs.500ash01.tn.comcast.net) |
06:49.48 | ellvis | i am experiencing no tones when i call the numbers. no ringing tone, no busy tone and so. where should i look to find a problem? i am using hisax for ISDN |
06:51.40 | nine76 | console output provide any clues? |
06:52.54 | ellvis | nine76: no. when i'll call you, i'll hear nothing, when you'll pick up the phone then the call work normally, we can speak and so. |
06:53.27 | *** join/#asterisk hellop (~hellop@cpe-70-93-40-171.hawaii.res.rr.com) |
06:53.32 | ellvis | nine76: audio is just fine, except the the ring/busy tones (is that what "signaling" mean?) |
06:54.14 | nine76 | strange behavior. I dont know the fix:-/ |
06:55.09 | *** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net) |
06:55.25 | ellvis | ok, i'll keep on digging. thanks for a try anyway |
06:57.39 | bewest | <PROTECTED> |
06:58.03 | nine76 | standard * install, /var/lib/asterisk/sounds |
06:58.20 | *** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
06:59.49 | bewest | yay :-) it works |
07:00.05 | nine76 | progress,excellent:) |
07:00.24 | tzafrir | anybody using the "classical moh" files from signate? http://www.signate.com/moh.php |
07:00.51 | tzafrir | That pag is not very clear about their copyright/license status: are those files in the public domain? |
07:01.05 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
07:01.12 | lehel | hi |
07:01.27 | nine76 | hello |
07:01.42 | lehel | do you use AMP? |
07:02.03 | nine76 | no sir |
07:02.13 | nine76 | tzafrir: notice the "Signate Download Terms for Free Music on Hold" paragraph? |
07:02.45 | RaYmAn-Bx | tzafrir: it seems to me that they basically don't have any rights for it..So if some of the music/names is otherwise protected by the copyright owners (i.e. it hasn't expired) then you can't use it without license |
07:02.47 | nine76 | Isnt very clear,but I'm not a lawyer:-/ |
07:03.38 | RaYmAn-Bx | I would guess that the copyright on that music has expired though.. |
07:05.05 | tzafrir | downloading them now... |
07:05.52 | hellop | Ok, I almost got Asterisk Working! |
07:06.08 | hellop | But, this time around my X100P is not responding. |
07:06.18 | nine76 | "zap show channels" ? |
07:06.26 | hellop | How can I check it? |
07:06.27 | bewest | I can't even get to signate |
07:06.29 | bewest | I've been trying to all day |
07:06.38 | hellop | Who wants to help me? eh? |
07:06.48 | justnulling2 | i got sipura spa-1001 to work with fwd but not with astersik, can this be b/c i am behind nat? |
07:06.48 | hellop | Come on, whoooo want to help? |
07:07.03 | Jas_Williams | hellop: try ztcfg -vvvv |
07:07.07 | hellop | Output of Asterisk here: http://pastebin.ca/12060 |
07:07.09 | nine76 | possibly just,is your * server also behind nat? |
07:07.14 | Qwell | justnulling2: What is behind NAT, ata or *? |
07:07.18 | hellop | Jas_Williams, output of ztcfg at that site |
07:07.23 | justnulling2 | ata |
07:07.57 | hellop | Jas_Williams, 1 channels configured. |
07:08.14 | justnulling2 | qwell: sipura is behind nat |
07:08.25 | Jas_Williams | hellop: type zap show channels in the cli |
07:09.19 | hellop | No such command 'zap' (type 'help' for help) |
07:09.47 | hellop | Jas_Williams, so like, zap is not loading |
07:10.04 | Jas_Williams | hellop: Zaptel module is not loading compile libpri and zaptel then make clean make install in asterisk to ensure zaptel built into asterisk |
07:10.53 | hellop | Jas.. ahh I didn't do libpri, because the howto at Sineapps didn't say to. |
07:11.02 | hellop | Zaptel howto did... |
07:11.16 | Jas_Williams | you need libpri |
07:11.22 | hellop | whats it for? |
07:11.24 | brad[] | If I send a fax or run a debit machine on the same PSTN line as an asterisk Zap channel, the asterisk server seems to ring all the connected IP phones while the data transmission is taking place. Anyone run into this? |
07:11.32 | nine76 | I didnt compile libpri either,just zaptel & *. scary |
07:12.03 | hellop | nine76, huh? |
07:12.05 | Jas_Williams | It is used to set reason codes and decodes for PRI based zap channels without is chan_zap will not compile |
07:12.49 | hellop | Ok, hey thanks Jas_Williams |
07:13.00 | nine76 | What functionality have I lost by not having libpri compiled? |
07:13.32 | Jas_Williams | nine76: do you have chan_zap ? |
07:13.53 | Qwell | I've never installed libpri, and I use zap |
07:13.58 | [hC] | I think ive forgotten a step someplace. For some reason when i try to launch MeetMe from my dialplan, it doesnt go anywhere, and a show application MeetMe says there is no meetme application available. |
07:14.07 | [hC] | Is there a trick to meetme? |
07:14.16 | Qwell | [hC]: are you noloading some stuff you need? |
07:14.24 | nine76 | yes,chan_zap is there,and all seems to work fine. |
07:14.34 | kapejod | [hC]: do you have a zaptel timing source? |
07:14.40 | [hC] | kapejod: yes. |
07:14.47 | [hC] | Qwell: i believe im using the default modules.conf |
07:14.53 | Qwell | [hC]: check |
07:14.53 | [hC] | Maybe i need to specify |
07:15.03 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
07:15.03 | Jas_Williams | [hC]: meetme.conf ? |
07:15.04 | kapejod | [hC]: ls /usr/lib/asterisk/modules/app_meetme.sol |
07:15.34 | hellop | The FAQ at digium just does cvs checkout zaptel asterisk.. no libpri, thats how I got the X100P working before. |
07:15.34 | [hC] | kapejod: negative. Do i have to specially mark it as enabled at compile time? |
07:15.50 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
07:16.15 | Jas_Williams | [hC]: Meetme will not copile without zaptel being compiled |
07:16.25 | [hC] | Aha. Thats why. I did zaptel after. |
07:16.27 | [hC] | Thanks! |
07:16.40 | [hC] | I knew it was something silly i'd missed. |
07:17.55 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
07:19.06 | justnulling2 | any idea how to get sipura work from behind nat with asterisk? |
07:19.28 | nine76 | mine works fine justnulling2. added nat=yes....thats about all... |
07:20.40 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
07:20.41 | justnulling2 | nine76: tried that no luck there is nothing in the * logs so something is really wrong and looks like nat |
07:20.46 | marcus5 | anyone know of outbound sip/iax providers that support 900 calls? |
07:21.12 | nine76 | have you enabled "sip debug" and tried to trace down an error? |
07:22.29 | *** join/#asterisk IsMe (~me@218.111.224.205) |
07:22.32 | bewest | what's the difference between host and permit? |
07:22.37 | IsMe | hi guys |
07:22.44 | nine76 | hello |
07:23.34 | IsMe | any developer here who can do a SMS callback ? |
07:23.54 | *** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com) |
07:24.23 | justnulling2 | nine76: on sip debug has something there |
07:24.31 | kapejod | IsMe: that is trivial with some little scripts |
07:25.03 | IsMe | kapejod: i cant do it, not this year nor the next million year |
07:25.04 | IsMe | hehhe |
07:25.33 | kapejod | are you talking about fixed line sms, or sms in general? |
07:26.12 | IsMe | something like GSM user trigger a SMS , * attach to some kinda GSM device, receive the SMS and call back to the originator and wait for destination number |
07:26.41 | kapejod | why wait for a destination number? |
07:26.47 | kapejod | you can send that in the sms.... |
07:27.06 | IsMe | kapejod: ok, that will do too, but user are generally too lazy |
07:27.25 | justnulling2 | nine76: it is 401 unathorized, why is that? |
07:27.27 | kapejod | but this way they can use the details from their phonebook |
07:27.41 | nine76 | perhaps a username/pass mismatch? |
07:28.03 | IsMe | kapejod: well, i am assuming my users are lazy bunch |
07:28.21 | IsMe | either that or a ANI call back |
07:28.24 | *** join/#asterisk cmk (~cmk_@p54A3BF70.dip.t-dialin.net) |
07:29.15 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
07:30.18 | justnulling2 | nine76: used the same user/pass ones with xten which works |
07:30.20 | kapejod | if you can receive the sms on fixed line than you can do it all with just * |
07:30.30 | kapejod | and 1 script |
07:32.35 | *** join/#asterisk at561 (~oishf@68.71.213-37.atlsfl.adelphia.net) |
07:33.13 | IsMe | kapejod: sms on gsm |
07:33.17 | at561 | has the e911 crisis been discussed to death already |
07:33.18 | IsMe | not fixed line |
07:33.36 | bewest | what's the difference between host and permit? |
07:33.45 | tengulre | who can give me some tips? |
07:34.24 | hellop | What's the word on getting some yes/no 0-9 Voice recognition in Asterisk? |
07:34.27 | kapejod | IsMe: than it's just a matter of how you connect the gsm devices to your box |
07:34.46 | kapejod | hellop: try sphinx |
07:35.26 | tengulre | smstools? |
07:35.56 | IsMe | i just bought a ericsson fixed celluar terminal |
07:36.15 | *** join/#asterisk Cheetah (~Snak@62.217.48.111) |
07:36.18 | IsMe | been googing around, no luck yet |
07:36.20 | hellop | kapejod, k.. I've been playing with Voice Recognition for a while.. I have yet to find something that will just match yes/no, instead of looking up entire festival phenome dictionary. |
07:36.21 | Cheetah | hey folks :) |
07:36.37 | Jas_Williams | bewest: host=XXX means that is the IP Address of the host, Permit gives a range of permitted IP Adreesses when you don't know what the IP address is. |
07:37.01 | Qwell | Jas_Williams: to expand on that. When you know what the different hosts the user will connect from |
07:37.05 | Cheetah | i just installed the speex codec on my server and was wondering if 1) how much hardware phones are compatible with it, 2) where I can configure the codec on the server side? |
07:37.21 | kapejod | hellop: i used to work with some siemens and falcon gsm modems for sms |
07:37.39 | bewest | ok thanks |
07:37.40 | Qwell | Jas_Williams: unless its one range, cdr...in which case, I'm quite wrong |
07:37.44 | IsMe | tengulre: sms tool is a sms server, no ? |
07:37.52 | hellop | kapejod, does sms have something to do with voice recognition? |
07:38.07 | kapejod | hellop: sorry, wrong nick ;) |
07:38.25 | Jas_Williams | Qwell: The range is based on the mask value |
07:38.46 | Qwell | Jas_Williams: ahh, I just assumed it could be comma/space seperated |
07:38.51 | IsMe | kapejod: nothing related to sms/ani call back anywhere, care to point me to the right direction ? |
07:39.45 | IsMe | i already have * running as prodution |
07:40.29 | Qwell | hmm, if it isn't, maybe I'll make a patch |
07:40.40 | kapejod | IsMe: i have some simple, old outdated callback example on my website. with sms it is then just a matter of generating the outbound call (manager interface or pbx_spool) |
07:41.05 | hellop | <PROTECTED> |
07:41.21 | hellop | I just ran that command without the -lz and then asterisk compiles. |
07:41.31 | hellop | Oh the error is:/usr/bin/ld: cannot find -lz |
07:41.38 | hellop | Is there a more correct way? |
07:41.49 | justnulling2 | nine76: ok see the problem asterisk is sending packets to local ip of sipura and no wonder they don't get to sipura |
07:41.50 | IsMe | kapejod: mind showing me ? |
07:42.03 | Flav | there's a list of cheap providers and also a list that usually mentions whether IAX is supported, but "cheapest providers of IAX" doesn't seem to exist at the moment - that'd be a nice list :) |
07:42.05 | Qwell | hellop: looks like zlib or something |
07:42.18 | Qwell | Flav: cheapest is usually crap |
07:42.46 | Qwell | hellop: have you tried googling it? |
07:42.48 | kapejod | IsMe: http://www.junghanns.net/asterisk/page14.html |
07:42.54 | kapejod | IsMe: it's for capi though |
07:42.55 | Flav | Qwell: hmm, true - cheapest that isn't crap, perhaps? like, what's used by many/most of the people in here? |
07:43.03 | Qwell | Flav: I like nufone |
07:43.14 | hellop | Qwell yeah.. but.. gcc -lz doesn't give too good of results |
07:43.27 | Zeeek | Flav : nufone, voipjet, voicepulse connect |
07:43.29 | hellop | I omit the lz and it compiles. |
07:43.36 | Qwell | hellop: well, google it |
07:43.42 | hellop | I then noload dundi.. some mayeb for now its not needed. |
07:43.56 | Qwell | I'd bet you're missing zlib or something |
07:44.06 | justnulling2 | what ports do i need to open in my firewall for astersik to work? |
07:44.17 | Qwell | justnulling2: depends on what you're doing with it |
07:44.18 | IsMe | kapejod: site seems to be down |
07:44.31 | nine76 | 5060 for sip,10000-20000 for rtp,4569 for iax...probably others |
07:44.41 | kapejod | IsMe: nope, it's fine here :) |
07:44.54 | IsMe | kapejod: is ok now |
07:45.20 | kapejod | mind my slow backup system. the main server recently died ;) |
07:45.42 | IsMe | kapejod: so you are the creator, google always point me to that site together with digium user list. |
07:45.59 | justnulling2 | qwell: i am trying to get sipura from behind nat to work and asterisk is sending packets to local ip of sipura over the internet so it doesn't work |
07:46.05 | kapejod | IsMe: creator of what? :) |
07:46.25 | IsMe | kapejod: of that callback script |
07:46.26 | justnulling2 | nice76: all re udp? |
07:46.28 | Cheetah | any someone help me with this codec issue? ;) |
07:46.33 | kapejod | IsMe: yes |
07:46.35 | nine76 | yes |
07:46.49 | IsMe | kapejod: still could not figure out wtf is that. hehhe |
07:46.55 | Zeeek | justnulling2 : did you check the wiki page about NAT? |
07:46.57 | Qwell | justnulling2: sounds like you need to read up on how to get nat working |
07:47.09 | kapejod | lol |
07:48.44 | hellop | Qwell, before, when compiling pbx_dundi, I got error: zlib.h not found. I fixed that by copying zlib.h from my kernel source to /usr/include/linux. Now, this -zl error. Any ideas? |
07:48.58 | kapejod | RoyK: you are way too friendly this morning |
07:49.39 | Qwell | hellop: different file. install zlib |
07:49.40 | justnulling2 | yes i have to read lots of stuff on nat and * in general |
07:49.53 | Zeeek | justnulling2 and what have you done so far? |
07:49.56 | hellop | Qwell, pretty sure it is, maybe I need devel |
07:50.05 | Qwell | probably |
07:50.09 | Zeeek | justnulling2 and with what results? |
07:50.11 | RoyK | kapejod: always friendly :) |
07:51.03 | hellop | There is only zliblg-dev... |
07:51.22 | Flav | hellop: you installed that already? |
07:51.49 | Qwell | hellop: for the 3rd time. Have you googled the error? |
07:51.59 | justnulling2 | zeek: changed nat=yes to nat=0 as nine76 said it might help and well it didn't help, any thing else you can think of? |
07:52.01 | at561 | does 2.6.11.8 lock up for anyone else |
07:52.28 | hellop | Qwell.. well yes. But, I'm trying different searchs now. |
07:52.50 | Qwell | ~google /usr/bin/ld: cannot find -lz |
07:52.55 | Qwell | I swear to god, if its the first hit... |
07:53.16 | hellop | It sems I need a kernel recompile. |
07:53.58 | Zeeek | justnulling2 : check this out http://willypick.mindsay.com/?entry=10 |
07:54.32 | Qwell | "you have to install the zlib1g-dev and/or zlib1g package" |
07:54.45 | *** join/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu) |
07:54.59 | hellop | Qwell, yeah thats what I figured thanks for the help. |
07:55.06 | justnulling2 | zeek: i don't have double nat, * is located on the internet only sipura is behind firewall |
07:55.35 | Zeeek | yes, and that page doesn't mention the lines needed in sip.conf |
07:55.37 | Jas_Williams | justnulling2: enable stun on the sipura |
07:56.12 | Cheetah | Question: Where can I set up the configuration for codecs like speex? ;) |
07:56.33 | Cheetah | i read that there was something like codecs.conf, but it seems to have become obsolete |
07:58.46 | Zeeek | justnulling2 to answer your original query, 5060 forwarding should work on the phone end, but what result do you get? Does the phone register, is there one-way audio? |
07:59.49 | Jas_Williams | Zeeek: justnulling2's phone does not post its external IP address in the sip packets, he needs to enable stun correctly on the sipura |
08:00.54 | Zeeek | Jas_Williams 1, hello, 2, I just realized he said that and it didn't sink in and 3 I don't use STUN but the GS somehow figures it out... |
08:01.06 | Zeeek | justnulling2 sorry as I say I didn't read you right |
08:01.57 | Zeeek | other than using STUB there should be a NAT setting of some kind in the phone setup? |
08:02.02 | Zeeek | STUN |
08:05.08 | nine76 | on my spa-2000 there is a NAT setting,under "line X". stun settings are under "sip". I have not needed to enable stun nor nat on the sipura side to get my setup working. 2000 behind nat -> * no nat. just a nat=yes in * sip.conf :-/ |
08:06.36 | hellop | Qwell, it's compiling now, tks. |
08:06.44 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk) |
08:06.47 | ellvis | re |
08:08.20 | ellvis | i am having problem with ISDN line, i have no ring/busy tones at all. the voice is good in both direction, the signaling is not working at all. anyone can help me where i should take a look for solution? |
08:08.30 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
08:08.53 | Jas_Williams | ellvis: is this PRI ? |
08:09.38 | *** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net) |
08:11.01 | Qwell | bed |
08:11.07 | Qwell | hellop: I accept paypal |
08:11.27 | *** part/#asterisk buu (foobar@69.56.150.34) |
08:11.55 | hellop | $20 if you can get my SIP phone to ring/dial. |
08:13.08 | hellop | Wow, Asterisk is running with no error msgs.. |
08:13.39 | ellvis | Jas_Williams: no, BRI |
08:14.00 | ellvis | Jas_Williams: Eicon DIVA 2.01 with HiSax |
08:14.41 | hellop | well it rings but doesn't call out, yet. |
08:14.48 | Jas_Williams | ellvis: use chan_capi or get a card that supports zaphfc |
08:15.10 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
08:15.32 | ellvis | Jas_Williams: i'd like to, in next box for sure, i am just curious right now if HiSax itself is not supporting it or if i have a mistake somewhere |
08:17.22 | ellvis | Jas_Williams: anyway, those eicon cards are not usable with capi under linux at all |
08:18.47 | shido6 | hellop |
08:18.49 | shido6 | $20? |
08:18.51 | *** join/#asterisk oej (~oej@213.204.186.40) |
08:18.55 | bewest | ' |
08:20.40 | bewest | . |
08:20.46 | hellop | shido6, I would be willing to pay for support. |
08:20.46 | bewest | 7 |
08:21.09 | Jas_Williams | hellop: what state are we now in ? |
08:21.16 | hellop | Hawaii.. |
08:21.29 | bewest | confusion |
08:21.50 | Jas_Williams | :-) no I'm speaking Uk english ;-P what is the state of your X100P now |
08:22.29 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
08:22.37 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
08:22.54 | ellvis | :) |
08:23.34 | bewest | how do you know whether to use sip or iax? I'm trying both and registered for both |
08:23.44 | Flav | anyone know of a place that provides listings of voip providers by area codes they have coverage in? filtering the ones I could use LNP of my existing number manually is slow going :) |
08:24.24 | hellop | Jas_Williams, oh, it accepts calls to voicemail and the phone, but outgoing just rings busy. I can probably figure it out. I need help with some scripting things tho. |
08:24.41 | bewest | I mean, I'm paying for a service that I'm trying to connect to using my own *, when I do iax show register and sip show register, both appear to be registered |
08:25.05 | bewest | however, when I call the number that the service provides, my * says it's rejecting that ip |
08:25.26 | hellop | I would be willing to pay (cheaply, I'm just a student) for figuring out how to do automated outbound calling, determine weather voice or machine, and leave a msg or interact with user. |
08:25.36 | shido6 | something is really wrong bewest , pastebin.ca your dialplan and iax.conf/sip.conf |
08:25.42 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
08:25.55 | *** join/#asterisk lichen (lichen@aequitas.ipclassless.com) |
08:26.23 | *** join/#asterisk makkia (~pippo@nat.xsec.it) |
08:26.25 | makkia | hello |
08:27.16 | hellop | BTW, this is not for malicious. This is for my own business. When a customer calls. I'm trying to make an automated voice attendent. |
08:29.13 | hellop | Is there a way to have extensions.conf run a shell script? |
08:30.06 | RestLessGemini | Hellop: use System command, i've never tried it before though, but i used it to reload asterisk config |
08:30.23 | fenlander | http://www.voip-info.org/wiki-Asterisk+cmd+System |
08:30.32 | lichen | i'm guessing the cisco ata 186's that are specified for use with vonage are locked and cann't be used with any other services? |
08:30.41 | hellop | great, tks |
08:31.06 | *** join/#asterisk Cheetah (~Snak@62.217.48.111) |
08:31.08 | RestLessGemini | lichen: yes you are rite, but you can unlock them if you have the password |
08:31.29 | *** join/#asterisk Heretic666 (~Miranda@195.128.153.131) |
08:31.40 | RestLessGemini | ask vonage how to get them unlocked, i think you need to pay them for ata 186, then they will give you the password to reset it |
08:31.59 | Cheetah | where do I configure codec settings? |
08:32.30 | lichen | if it's cisco equipment why can't you just do a password recovery? |
08:32.39 | RestLessGemini | Cheetah: in sip.conf or iax.conf |
08:32.44 | lichen | or is such a feature not availble becuse it's "vonage" |
08:32.55 | RestLessGemini | lichen: you can not do this until you have the password its locked with |
08:33.39 | Jas_Williams | hellop: post your extensions.conf to pastebin.ca for me to look at |
08:34.12 | bewest | ok shido6: http://pastebin.ca/12131 |
08:34.14 | Cheetah | RestLessGemini, i installed the speex codec, and read that there is some kind of codecs.config; this is not the case anymore, right? because I can't find this file in the asterisk distri or somewhere else anymore |
08:34.30 | lichen | ahh i gotcha.. also instead of buying an FXS card for my box to plug into my NID could i just buy an ata186 and route calls destined for the analog netowrk through that? |
08:35.37 | RestLessGemini | codecs.conf is there, i've never worked with speex.. sorry :) |
08:35.39 | lichen | such as i have an incoming voip call, could i have a ring a certain call group that would route everything to a 186? |
08:36.01 | RestLessGemini | lichen: yes you can do this |
08:36.04 | hellop | Jas_Williams, http://pastebin.ca/12132 |
08:36.12 | RestLessGemini | lichen: I do this all the time |
08:36.16 | Cheetah | RestLessGemini, so, do you have any idea how to configure speex, then? ;) |
08:36.27 | RestLessGemini | Cheetah: Nop :) |
08:36.32 | lichen | would i need to have the ata login to the asterisk box? |
08:36.36 | Cheetah | snap :/ |
08:36.54 | RestLessGemini | lichen: yes .. just like anyother IP-Phone |
08:37.14 | lichen | gotcha.. it's been such a long time since i've dealt with voip.. i'm trying to pick up all the nuances again |
08:37.38 | RestLessGemini | lichen: wishing you good luck |
08:37.42 | RestLessGemini | :) |
08:37.45 | IsMe | any developer here who can do a SMS callback ? |
08:38.09 | IsMe | gsm callback |
08:38.24 | Jas_Williams | hellop: what device are you using to place the outbound calls and what context is it in ? |
08:39.06 | lichen | my friend has voip service through broadvoice, we're trying to setup something along the lines of having his asterisk box login to the broadvoice server, and then if they're destined for his apt they're route to his ata, otherwise they'll go across a secure tunnel terminating at my router, which then sends it to my asterisk box (the two boxes will be IAX trunked) which will in turn send it to my ATA and then into the NID and ring my analog phones |
08:39.09 | jeffik | all: need some help with x-lite |
08:39.10 | *** join/#asterisk Ahrimanes (~aron@hobbes.bsd-dk.dk) |
08:41.00 | RestLessGemini | lichen: its very much possible as i've done similar stuff |
08:42.19 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
08:42.41 | lichen | good deal.. i'm just curious whether or not i'll need to have his ata login to hte broadvoice service or if i should have the asterisk box do it.. i'm guessing we won't have full access to configure his ata, plus i want a voice prompt so the person can choose between either my apt or his, so i guess my best bet is to have the asterisk box pick up the call |
08:43.08 | lichen | then on his end he can either route calls to the ATA or get an FXS port for hte box |
08:43.12 | Mimmus | hi, I'm just to buy an E1 card for Asterisk and I need some help |
08:43.20 | RestLessGemini | no, make ATA login to your *, and * login to Broadvoice |
08:43.38 | lichen | okay that's what i figured would be the best bet |
08:44.07 | RestLessGemini | accept calls from broadvoice to your *, then route them to ATA or to other * box |
08:44.09 | lichen | another area of concern is we can't make multiple concurrent calls.. so i'm hoping we'll be able to stop such a thing in * |
08:44.30 | RestLessGemini | lichen: dont even think about it :) |
08:44.31 | Mimmus | why are different Digium cards for 3.3 and 5 V PCI slots while Sangoma offers a 'universal' card? |
08:45.02 | af_ | how could I check fo r 3.3 or 5 Volt pci slot? |
08:45.08 | RestLessGemini | they will close your account and send you an overage bill, they've done this with me :) |
08:45.27 | hellop | Jas_Williams, it's a Budgetone100 |
08:45.38 | Ahrimanes | af_: motherboard manual? |
08:45.47 | Jas_Williams | hellop: what is it's context |
08:45.52 | hellop | context is local-access |
08:45.52 | lichen | exactly, that's what i'm worried about.. so i'm hoping like if i pick up a line from my * and try to make a call out of our presonal network it will realize that a call to broadvoice is already in progress and not allow it |
08:45.52 | af_ | Ahrimanes: I don't have it it's an epia 5000 |
08:45.55 | RestLessGemini | i've tried concurrent calls through my * box and in the end I got my account terminated and they've billed me more then $200 |
08:46.02 | Ahrimanes | af_: 2 sec |
08:46.40 | Mimmus | recent motherboards usually have 1 or 0 5V slots :-( |
08:47.12 | af_ | and a dell 2400 |
08:47.16 | af_ | uhm. |
08:47.30 | RestLessGemini | well, while making account for outgoing through broadvoice, limit it to just 1 outgoing call, this way it will not let another call go out and gives you a congestion signal, capture it in dialplan and play appopriate msg |
08:47.31 | hellop | Jas_Williams, I just got casterisk to compile. so, I haven't had a chance to RTFM on the SIP fone yet. |
08:47.31 | Ahrimanes | af_: dell.com should be able to tell you about the dell |
08:47.37 | hellop | whoops asterisk |
08:47.40 | af_ | Ahrimanes: nope :( |
08:47.47 | Ahrimanes | af_: then call support, hehe |
08:47.53 | af_ | ..... |
08:48.08 | Mimmus | af_: Dell 2850 has only 3.3 slots |
08:48.27 | lichen | nice sounds good.. if he ever gets ip phones i can make it so i can still dial is ip phones even if his fxs is off the hook |
08:48.38 | Mimmus | af_: Dell 2800 (tower) has 1 5V slot |
08:48.54 | makkia | mmm with MP3Player(...) the quality is of music is good, but with MusiOnOld() is very very bud |
08:48.57 | Jas_Williams | hellop: what is the number you are dialling from the phone ? |
08:48.59 | lichen | so here's the question.. do you suggest FXS card or ATA186 (or perhaps some other brand of ATA?) |
08:49.02 | makkia | is a know problem ? |
08:49.02 | Ahrimanes | af_: via is 3.3 |
08:49.16 | af_ | what happens if I put a board set to 5V in a 3.3 slot? |
08:49.21 | af_ | thank Ahrimanes |
08:49.32 | hellop | Jas_Williams, 641-3333 |
08:49.45 | jeffik | anyody: help with x-lite config |
08:50.02 | Mimmus | Ahrimanes: why do I need to buy a board that fits only in 3.3 or 5V slot (Digium), if I can have one (Sangoma) fitting in both? |
08:50.31 | Ahrimanes | Mimmus: no idea, buy sangoma if this is important to you? |
08:50.53 | RestLessGemini | ATA work good, and besides you can not take FXS card anywhere you want, but you can move ATA within the network |
08:51.04 | Mimmus | Ahrimanes: I'd like to buy a Digium card but I'm fighting with this problem |
08:51.25 | Ahrimanes | Mimmus: why, do you need to move the card around between servers alot? |
08:51.36 | lichen | good point.. plus i'm partial to just additional network hardware and basically more blinky lights :) |
08:51.50 | Jas_Williams | hellop: change exten => _9NXXXXXX,1,Dial(Zap/1/${EXTEN:1}) |
08:51.50 | Jas_Williams | exten => _9NXXXXXX,2,Congestion to exten => _9X.,1,Dial(Zap/1/${EXTEN:1}) |
08:51.50 | Jas_Williams | exten => _9X.,2,Congestion reload extensions and try again |
08:51.52 | RestLessGemini | :) |
08:51.55 | Mimmus | my test server is older and has only 5V slots, production server will be newer and probably it will not have 5V slot :-( |
08:52.36 | Ahrimanes | Mimmus: hm, upgrade test server or buy 2 cards? |
08:52.37 | RestLessGemini | brb |
08:53.08 | Mimmus | Ahrimanes: ok, bit it is annoying (I hope this the right word) :-) |
08:53.42 | Ahrimanes | Mimmus: i know it is.. might face the same kind of problem, but really a new test server can bring more benefits :) |
08:55.30 | hellop | Jas_Williams, same, busy signal. But thanks for hint. It was setup for australia. |
08:55.42 | bewest | I have this voip service that I'm trying connect to using *. iax show registry shows that I'm registered, however, when I call the number, I see that * is rejecting the IP |
08:55.45 | Mimmus | Ahrimanes: thanks for support, Sangome offers also a 2-ports card instead Digium only 4-ports |
08:56.15 | Jas_Williams | hellop: sounds like your sip phone is not registered |
08:56.17 | Mimmus | Ahrimanes: but I'd like to feel supporting Asterisk creator anyway! |
08:56.19 | Ahrimanes | Mimmus: hehe np.. ok we have 1 4 port digium, but chose to outsource pstn to another company so i just iax now :) |
08:56.28 | *** join/#asterisk tessier (~treed@203.210.216.1) |
08:56.36 | hellop | Jas_Williams, even tho it rings on incoming? |
08:56.44 | Ahrimanes | anyone here doing videotelephony with asterisk? |
08:57.21 | tessier | Ahrimanes: asterisk has nothing to do with video |
08:57.58 | Ahrimanes | tessier: oh but it does, videosupport=yes etc |
08:58.20 | fenlander | Ahriamanes: yes - we use it for video |
08:58.25 | hellop | Yannick Tessier? |
08:58.34 | Jas_Williams | hellop: do you see any output in the cli |
08:58.34 | Ahrimanes | fenlander: ok, with which devices? |
08:58.49 | hellop | Jas_Williams, on incoming I do, but not outgoing. |
08:58.55 | *** part/#asterisk dieck (~dieck@saengerschafter.de) |
08:59.03 | fenlander | Ahrimanes: softphones - eyebeam and one developed inhouse |
08:59.36 | Ahrimanes | fenlander: ah ok, we use eyebeam as well.. but am working on getting some hardphones working now.. but having lots of problems.. heh |
08:59.42 | Jas_Williams | hellop: turn on sip debug and then try again post the results to paste bin |
08:59.43 | jeffik | fenlander: may i ask you a question avout x-lite? |
08:59.56 | fenlander | jeffik: you can try |
09:00.12 | fenlander | Ahrimanes: would be interested to hear how you get on - which phones? |
09:00.28 | Ahrimanes | fenlander: innomedia, wooksung and leadtek for now |
09:01.03 | jeffik | ok, my system sits in chicago, I am i Toronto, i have x-lite and no prob. works fine. I just set up a friend in Tblisi Georgia with x-lite, he is not behind a firewall or router, he is diredt in |
09:01.36 | jeffik | he can call me, but when i call him i get recording the extension is not availabler |
09:02.27 | fenlander | jeffik: could be a dialplan problem, or maybe his xlite is not registering - what does sip show registry say? |
09:02.48 | jeffik | says logged-in enter number |
09:02.54 | jeffik | im runnig aah |
09:03.13 | jeffik | i wonder if we should say yes on nat |
09:03.26 | fenlander | I don't know much about aah - you could switch to a shell and run asterisk -r then sip show registry |
09:03.43 | hellop | this bug mean anything? WARNING[9908]: cli.c:967 ast_cli_register: Command 'sipdebug' already registered |
09:04.00 | jeffik | i can get to the shell i should run sip registry? |
09:04.39 | fenlander | if there is no nat going on anywhere yous shouldn't need any nat options |
09:05.05 | fenlander | run asterisk -r from the shell, then "sip show registry" will show you the registered clients |
09:05.11 | Jas_Williams | hellop: only a warning do not worry |
09:05.27 | jeffik | ok |
09:06.10 | fenlander | jeffik: hang on, no it won't ;-) wrong command |
09:06.21 | Ahrimanes | sip show peers? |
09:06.30 | hellop | SIP/2.0 404 Not Found |
09:06.52 | fenlander | that's what I meant :-) sip show peers |
09:07.01 | Ahrimanes | fenlander: priv msg.. |
09:07.40 | hellop | http://pastebin.ca/12135 <-Sip fone error. |
09:08.00 | *** join/#asterisk tass (~tassi@ppp163-93.static.internode.on.net) |
09:08.56 | Jas_Williams | hellop: proceed the 6407133 with a 9 for an outbound call |
09:09.08 | Jas_Williams | hellop: dial 96407133 |
09:09.18 | hellop | it works |
09:09.21 | tass | hello... was wondering if anyone can help me with a digium quad-span E1? |
09:09.29 | hellop | hehe |
09:09.31 | Jas_Williams | tass: in a moment |
09:09.38 | Jas_Williams | ~jas_williams |
09:09.39 | jbot | i guess jas_williams is jason@redskycomputing.co.uk, paypal donations accepted |
09:09.59 | Jas_Williams | tass: what is your question |
09:10.57 | tass | we have an E1 confirmed working - australian OnRamp10 service, but don't get any lights on the digium card other than the 'knightrider' lights up until the module is loaded |
09:10.58 | hellop | Jas, can I PM you? |
09:11.15 | Jas_Williams | hellop: sure |
09:11.15 | tass | as soon as the module is loaded, all lights switch off |
09:11.25 | kapejod | tass: did you run ztcfg? |
09:12.00 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
09:12.05 | tass | yes... doesn't display any errors |
09:12.17 | tass | zttool shows the first 3 spans as 'OK' and no alarms |
09:12.26 | Cheetah | i get this in asterisk log: |
09:12.27 | Cheetah | speextolin_framein: Out of buffer space |
09:12.35 | kapejod | tass: cat /proc/zaptel/* | grep Span |
09:12.42 | tass | asterisk 'zap show channel X' always shows 'InAlarm: 1' |
09:13.35 | tass | do you want me to paste the Grep output? |
09:13.38 | Cheetah | any ideas? ;) |
09:13.53 | kapejod | tass: yes, sir! |
09:13.56 | tass | Span 1: TE4/0/1 "TE410P (PCI) Card 0 Span 1" HDB3/CCS/CRC4 |
09:13.56 | tass | Span 2: TE4/0/2 "TE410P (PCI) Card 0 Span 2" HDB3/CCS |
09:13.56 | tass | Span 3: TE4/0/3 "TE410P (PCI) Card 0 Span 3" HDB3/CCS |
09:13.56 | tass | Span 4: TE4/0/4 "TE410P (PCI) Card 0 Span 4" |
09:14.20 | tass | span 2 and 3 i've tried connecting together with a loopback cable which didn't seem to do anything |
09:14.35 | tass | span 1 has been set up for the PRI |
09:14.37 | Jas_Williams | tass: post a pri debug intense span 1 to pastebin.ca |
09:14.52 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
09:15.01 | kapejod | tass: did you jumper the card to E1? |
09:15.14 | tass | yes, it's jumpered as E1 |
09:17.14 | tass | under dev/zap, i can see up to 124 which seems right for E1 which hopefully means its jumpered correctly |
09:17.48 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
09:18.38 | cjk | hi, does anyone know a way to get turkish did's ? |
09:18.51 | tass | Jas_Williams: i've enabled intense debug... is there anything in particular i should do other than putting the cable in? |
09:19.18 | Jas_Williams | tass: No just catch some output if any |
09:19.28 | *** join/#asterisk jackthe (~jesse@d594f03e.ftth.concepts.nl) |
09:20.48 | lichen | the hell.. i don't see any of the standard just FXS/FXO cards on digium's site anymore |
09:20.55 | lichen | now they're all the modular ones.. did they do away with the others? |
09:21.00 | Flav | sorry for a spammy question, but any pros/cons of TelaSIP/VoipXpress? |
09:21.47 | Jas_Williams | lichen: 1 port FXO cards AKA X100P is no more. You need to buy the modular cards |
09:22.59 | bewest | I don't understand: iax show registry shows that I am registered with my voip provider, yet when I try to make a call to *, it rejects the connection attempt |
09:23.03 | lichen | ah you're right.. TDM10B is only $125 .. that's nifty |
09:27.06 | Flav | how much was the X100P before? |
09:27.17 | lichen | almost $300 or so i thought? |
09:27.22 | lichen | or i could be way off |
09:27.30 | hellop | Should my SIP fone be doing 174kbits/s constant to the Asterisk Server? |
09:27.53 | hellop | lichen, I think it was about 100 |
09:29.40 | RoyK | hellop: about 80kbps full duplex |
09:29.46 | RoyK | so around 160 total |
09:30.11 | RoyK | plus, of course, some SIP control traffic, but that'll be minor |
09:30.37 | tass | Jas_Williams: i haven't seen any debug output at all |
09:31.43 | lichen | so i haven't heard much about anything except for the cisco ATA's... are there any others out there or is the cisco the best affordable one |
09:32.40 | Jas_Williams | tass: that is strange what is the output of zapcfg -vvv |
09:33.48 | bewest | does anyone have service with asterlink.com ? |
09:33.58 | Flav | stupid question, but any chance you could use a FXO card as an FXS card? |
09:34.05 | hellop | RoyK, so I can only have 2-3 calls on my 256kbit upstream? |
09:34.23 | Jas_Williams | Flav: not without some kind of converter |
09:34.24 | tass | jas: it's fairly long, can i pm it to you? |
09:34.41 | Jas_Williams | tass: use pastebin.ca |
09:35.28 | Flav | Jas_Williams: i just noticed the 100X single FXO card is < $10 on ebay |
09:35.31 | RoyK | hellop: with g.711, yes |
09:35.36 | Jas_Williams | tass: also post zaptel.conf and zapata.conf whilst your there |
09:35.57 | tass | jas: done (ztcfg @ http://pastebin.ca/12137) |
09:36.04 | Jas_Williams | Flav: Yes use a ata adapter for fxs ports |
09:36.08 | bewest | in order for * to recieve calls from another iax, do you have to register it? |
09:37.12 | Flav | Jas_Williams: i was kind of hoping to find a cheapish fxs card so i could do asterisk on the linux box then fxs out to my existing phones |
09:37.28 | lichen | flav... i was just looking at that too. unfortunately i need an FXS card as well.. i think i'm just going to buy an ATA instead though |
09:37.37 | tass | jas: config files @ http://pastebin.ca/12138 |
09:39.33 | Jas_Williams | tass: config files look good |
09:39.48 | Jas_Williams | tass: are you connected to the pri at the moment ? |
09:39.53 | tass | yes... |
09:40.23 | tass | i have the pri through a crossed cable... 1-4 2-5 i think it is |
09:40.43 | tass | same one we have working on a dialogic E1 |
09:40.48 | Jas_Williams | tass: |
09:41.10 | Jas_Williams | tass: is the span 1 going to a carrier or another pbx ? |
09:41.18 | tass | jas: carrier |
09:41.34 | Jas_Williams | tass: then you need a straight cable to the carrier. |
09:41.56 | tass | jas: i'll try a straight cable and check the debug output... |
09:43.19 | *** join/#asterisk kapejod (~kapejod@e178005154.adsl.alicedsl.de) |
09:44.47 | tass | jas: swapping the cable didn't change anything, however i did get the following before the swap: http://pastebin.ca/12140 |
09:46.32 | *** join/#asterisk Blackvel (~blackvel@dsl-084-057-126-189.arcor-ip.net) |
09:46.35 | jeffik | fenlander: it worked |
09:46.37 | Jas_Williams | tass: Your PRI is getting nothing from the network.... |
09:46.48 | jeffik | finlander: thanks |
09:46.55 | fenlander | jeffik: great |
09:47.11 | jeffik | and the connectio is really good, |
09:47.21 | Blackvel | morning |
09:47.27 | tass | jas: no configuration could affect this? we had it working just a couple of minutes ago out a separate application (not asterisk) on a dialogic card |
09:48.46 | Blackvel | how can I use xlite to do a direct ip call to *? #2 proxy is configured to use direct ip calling. I type in 192.168.1.2 and also try with 66@192.168.1.2 but I get this error message: Failed to authenticate user Blackvel <sip:192.168.1.3:5070> |
09:49.14 | Blackvel | is it possible to set insecure=very in sip.conf [general]? context=directip is there |
09:49.18 | Jas_Williams | tass: From what I see in your intense debug there is a cable problem, Has the debug changed with a straight through cable |
09:49.58 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
09:50.07 | Jas_Williams | Blackvel: make sure you have a relm defined in second proxy |
09:50.09 | tass | jas: i'll swap the cable and restart asterisk |
09:50.12 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
09:50.45 | *** join/#asterisk strafbomber (~strafi@p54A738F6.dip.t-dialin.net) |
09:50.48 | strafbomber | hello |
09:52.04 | Blackvel | Jas_Williams: oh in xlite? and what should that be? any name? ip of asterisk? |
09:52.27 | tass | jas: i received 1 frame that looks the same as the last 2 i pasted |
09:53.01 | Blackvel | same message but now with : failed ... Blackvel...sip:directip:5070 |
09:53.06 | Blackvel | realm is now directip |
09:53.46 | Blackvel | I can remember there was realm setting also for sip.conf. I don't have that set. is that required? |
09:53.52 | strafbomber | is it possible to use the Wildcard TE410P whit linux & capi is there a driver for that? |
09:54.01 | Cheetah | hmm |
09:54.25 | Cheetah | which is the best codec if you want good quality? (except ulaw, that one sucks ;) |
09:54.36 | Blackvel | alaw |
09:54.41 | Blackvel | it suxx? |
09:54.54 | Blackvel | shouldnt be |
09:54.55 | Ahrimanes | alaw works fine for me |
09:55.08 | Blackvel | everything expect ulaw/alaw is worse |
09:55.15 | Cheetah | well, you can clearly hear that the sound quality has a HUGE loss compared to normal phones |
09:55.16 | hellop | ass |
09:55.37 | hellop | such a short cord on the budgetone, it dropped, and now has static.. |
09:55.55 | Cheetah | i tried speex, but that one doesnt seem to be too compatible with other things |
09:55.56 | Jas_Williams | tass: Do you have the cable connected to the correct port You may be plugging int span 4 |
09:56.22 | tass | jas: it is definately numbered 1... |
09:56.25 | Jas_Williams | tass: leave intense debug on and plug straight cable into each in turn |
09:56.48 | Ahrimanes | Cheetah: with alaw i get similar or better sound than wired phones |
09:57.17 | Blackvel | Cheetah: HUGE loss? |
09:57.18 | bewest | could someone please take a look at this and possibly help me out? http://pastebin.ca/12141 |
09:57.20 | Cheetah | Ahrimanes, are there settings I can optimize? because bandwidth is not a problem here (100Mbit connected softphones/phones) |
09:57.31 | Blackvel | Cheetah: your softphone is the problem |
09:57.32 | Cheetah | Blackvel, yeah, it sounds worse than GSM |
09:57.37 | Cheetah | xlite? |
09:57.41 | Blackvel | prolly |
09:57.50 | Blackvel | I always have problems |
09:57.52 | Blackvel | use hardware |
09:58.09 | tass | jas: i didn't get anything from moving the cable |
09:58.11 | Cheetah | well, thats not an option here ;) |
09:58.14 | Blackvel | why |
09:58.23 | Blackvel | get sipura2000 |
09:58.25 | Cheetah | because exactly thats why we move to softphones |
09:58.33 | Cheetah | they need to be mac compatible |
09:58.34 | Blackvel | well |
09:58.36 | Blackvel | :) |
09:58.44 | Blackvel | bluetooth headsets? |
09:58.46 | Cheetah | yep |
09:58.49 | strafbomber | is this possible? i want to use this card for an dialin access server |
09:58.57 | Blackvel | hmm too bad |
09:59.00 | RoyK | Cheetah: I don't think there are any good softfons around for mac :( |
09:59.10 | Blackvel | xlite works on mac? |
09:59.12 | Blackvel | interesting |
09:59.15 | Cheetah | it does |
09:59.19 | Cheetah | sjphone? |
10:00.07 | *** part/#asterisk Cheetah (~Snak@62.217.48.111) |
10:00.17 | *** join/#asterisk Cheetah (~Snak@62.217.48.111) |
10:00.18 | Cheetah | er |
10:00.25 | Cheetah | i should stop pressing red buttons |
10:00.27 | strafbomber | has noone an answer? |
10:00.36 | bewest | no one can help me out? |
10:00.46 | Jas_Williams | tass: No idea what your problem is then unless you provider has the cables wrong an E1 cross over is 1-5 2-4 |
10:00.51 | Cheetah | strafbomber, weird nickname.. especially if you can speak german :D |
10:01.15 | Ahrimanes | eek german |
10:01.28 | strafbomber | :) |
10:01.48 | strafbomber | yes i can speak german, und das besser als englisch ;-) |
10:02.22 | *** join/#asterisk bsunil (~bsunil@202.54.37.182) |
10:02.27 | Blackvel | und das typische klische wird wieder wach |
10:02.28 | *** join/#asterisk robin_sz (~robin@adsl.redpoint.org.uk) |
10:02.33 | robin_sz | meep? |
10:02.58 | robin_sz | so .. for added fun, I'm back to trying to build ztdummy on debian again ... |
10:03.03 | strafbomber | which one? |
10:03.07 | robin_sz | sarge |
10:03.16 | bsunil | hello can any onle tell me how to trace a call through asterisk |
10:03.47 | robin_sz | asterisk-1.0.7 on sarge to be precise |
10:03.48 | Sato1 | bsunil, with its cdr |
10:04.03 | Blackvel | okay, what was now the correct way to do direct ip calling into asterisk? |
10:04.06 | bewest | is anyone familiar with this message from *: Rejected connect attempt from 66.250.68.190 |
10:04.21 | Sato1 | bsunil, over here, please |
10:04.34 | bsunil | ok |
10:04.36 | bewest | I'm registered with a voip provider through iax |
10:04.43 | bewest | but that is the message I get |
10:04.53 | bewest | I've tried dozens of things, searching via the wiki and all over |
10:04.55 | robin_sz | to save me added pain, does ztdummy actually work with a 2.4 SMP kernel? |
10:04.56 | bewest | and I am completely stuck |
10:04.56 | Blackvel | hm I try to register a xlite to an asterisk behind a firewall |
10:05.15 | Sato1 | bsunil, check for log files in the directory /var/log/asterisk |
10:05.18 | Blackvel | does xlite use automatically for registering port 5060 or will it use listening sip port 5070 (on my system its 5070) |
10:05.40 | Sato1 | bsunil, or install the addon to store all call records in mysql or pgsql |
10:05.49 | bsunil | Sato1 ok |
10:05.49 | bewest | my iax.conf, sip.conf, and extensions.conf are at http://pastebin.ca/12141 |
10:06.04 | bewest | and again, I am really really stuck, so if anyone could take a look and help me out |
10:06.15 | bewest | I would really appreciate it |
10:07.27 | tass | jas: RCV_RING<->XMIT_TIP & RCV_TIP<->XMIT_RING ?? this is according to the dialogic instructions. |
10:08.36 | bewest | wow no help |
10:09.40 | Jas_Williams | tass: sounds about right |
10:10.12 | Cheetah | i would help, if I could ;) |
10:10.16 | Jas_Williams | tass: should be 1-4 2-5 |
10:10.17 | Cheetah | but I'm a bloody newbie |
10:10.32 | Jas_Williams | tass: see http://www.gcom.com/home/support/t1crossover.html |
10:10.44 | Blackvel | Cheetah: you could try to do sip show channels |
10:10.52 | Blackvel | to see if you are really using ulaw/alaw |
10:11.02 | Sato1 | robin_sz, not sure if ztdummy works on SMP kernels, but you can give it a try, just follow the instructions in zaptel directory for compiling ztdummy support, it will take you 5 minutes |
10:11.40 | tass | jas: thanks for that... i'm fairly certain that's what we've got |
10:11.52 | tass | jas: especially since it works for the other card... |
10:12.05 | tass | jas: could it at all be a hardware fault do you think? |
10:12.08 | Zeeek | bewest ? |
10:12.13 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
10:12.47 | Zeeek | isn't there a little something funny about those first three lines in you context? |
10:13.30 | Jas_Williams | tass: possibly. |
10:14.03 | *** join/#asterisk ellvis (~ellvis@adsl-flat-basic-105.84-47-116.telecom.sk) |
10:14.07 | ellvis | re |
10:14.50 | ellvis | anyone have any experience with troubleshooting isdn bri and hisax and signaling? |
10:15.16 | tass | jas_williams: Thanks for your help... i might try set up the other spans instead and see if they work, and if i get the same problems i'll see if i can try a new card. |
10:15.45 | tass | jas_williams: i think i've tried just about everything I can otherwise! |
10:16.13 | bewest | Zeeek, dunno, very new at this and been staring at this stuff for hours |
10:16.54 | robin_sz | Sato1: I seem to be having lots pf trouble on debain Sarge compiling zaptel ... it just throws out pages of errors |
10:17.27 | Zeeek | bewest, looking at your first three lines, I would have to recommend you do a couple days research and read the doc at http://asteriskdocs.org and the wiki sections about IAX2, extensions etc. |
10:18.09 | Sato1 | robin_sz, see where it starts throwing errors, and paste the first 2 lines |
10:19.10 | onkeltimm | bewest: you have to put in your asterlink number in the pattern section of your first three lines. |
10:19.33 | robin_sz | Sato1: if I use buildpackage it compiles OK, but withut ztdummy .. if I just run make I get: |
10:19.39 | robin_sz | Sato1: In file included from zaptel.c:42: |
10:19.39 | robin_sz | /usr/include/linux/kernel.h:72: error: parse error before "size_t" |
10:19.39 | robin_sz | /usr/include/linux/kernel.h:74: error: parse error before "size_t" |
10:20.43 | onkeltimm | bewest: http://pastebin.ca/12143 |
10:21.53 | onkeltimm | but i do not know asterlink. |
10:22.00 | Zeeek | onkeltimm whatd'you make of the deny/permit at the end? |
10:22.13 | onkeltimm | they should have config info in their members section for sure |
10:22.43 | onkeltimm | Zeeek: din't read that far as the first three lines were so dead wrong... |
10:22.54 | Sato1 | robin_sz, dont know in debian, but in rh, it solves lot of those problems by changing /usr/include/linux/* to the real path to the kernel source (usually in /usr/src/linux-2.4/include/linux) |
10:23.07 | robin_sz | oh, ok ta |
10:23.34 | onkeltimm | oh yeah somebody forgot the netmasks there... |
10:23.45 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
10:23.49 | Mother_ | greetings |
10:24.36 | Zeeek | Hi Mom! |
10:24.44 | Mother_ | heya Zeeek |
10:24.53 | Mother_ | how's things? coming to Madrid? |
10:25.25 | Zeeek | bewest, it took a while for me to realize that many of the files in /usr/src/asterisk/*.sample contain almost all options possible |
10:25.40 | Zeeek | Mother_ I may be going to Barcelona |
10:25.54 | Mother_ | well let me know then, I live in Barcelona |
10:25.58 | onkeltimm | bewest: go reading. look closely at the examples on voip-info.org |
10:26.05 | Mother_ | we can have a paella and talk SIP |
10:26.07 | Mother_ | :) |
10:26.16 | bewest | are you referring to the extensions.conf |
10:26.19 | bewest | or all of them |
10:26.19 | Zeeek | and try to separate the obviously wrong ones from the valid :) |
10:26.56 | Mother_ | bewest: also look here http://www.loligo.com/asterisk/current/ |
10:27.08 | bewest | heh I've been looking at all of those |
10:27.09 | Mother_ | some comprehensive examples for quite a few things * can do |
10:27.29 | bewest | I don't go off shooting for help very quickly |
10:27.31 | Zeeek | bewest I think your configs are pretty far off; you need to study harder if you haven't been able to understand them |
10:27.33 | Mother_ | I found them quite useful when I started - print out the long ones however |
10:27.50 | Zeeek | possible a step by step approach might help |
10:27.52 | Zeeek | Installing linux, asterisk X100P, TDM400, SIP config, Vmail, parking and many other sexy topics |
10:27.52 | Zeeek | http://automated.it/guidetoasterisk.htm |
10:27.58 | Zeeek | The above is very good |
10:28.03 | Mother_ | yep, good one too that is |
10:28.07 | bewest | ok haven't seen that one yet |
10:28.11 | Zeeek | aha! |
10:28.25 | Mother_ | since it has wifi - free calls :) |
10:28.49 | onkeltimm | Zeeek: uargh decaf... |
10:28.53 | Zeeek | Cheetah why not IAX if you wanna dream? |
10:29.02 | Mother_ | well that too |
10:29.11 | Cheetah | Zeeek, |
10:29.13 | Cheetah | ? |
10:29.28 | Zeeek | no that was for Mother_ sorry |
10:29.33 | Cheetah | heh |
10:29.46 | Mother_ | I really hate the problems with hangup detection, I have this callback system working with the slight problem that the Zap channels just hang there forever |
10:30.51 | Mother_ | Zeeek: well, maybe porting some IAX client would be easier than writing a SIP one from scratch, who knows :) |
10:30.56 | Zeeek | annoying and potentially costly |
10:31.04 | Zeeek | the hanging ZAP I mean |
10:31.06 | Mother_ | indeed, very annoying |
10:31.07 | Mother_ | yep |
10:31.28 | Mother_ | does * monitor DTMF on Zap channels while on a call? |
10:31.47 | Mother_ | maybe I could get it to recognise a # or * to hang up the calls |
10:37.55 | Mother_ | maybe with the Read command something could be cooked up...grrr |
10:39.23 | fenlander | How about the h or H option to Dial? |
10:39.47 | onkeltimm | fenlander: bad if * does not detect the hengup correctly. |
10:40.09 | onkeltimm | Mother_: but what about a simple exten => #,1,Hangup |
10:40.53 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
10:42.31 | ellvis | Cheetah: not really a SIP phone, but who knows, maybe one day... http://www.taptarget.com/products/iphone.html |
10:46.12 | Mother_ | onkeltimm: I could try that, but I understand that between s,1,Dial..... and s,2,Hangup asterisk is waiting |
10:46.31 | Mother_ | if I dial # during a call, will it get thrown into the context again? |
10:46.35 | *** join/#asterisk newbien (~e@147.241.33.65.cfl.res.rr.com) |
10:47.45 | Mother_ | i.e. in [context] I would have s,1,Dial..... s,2,Hangup then #,1,Hangup <- will it ever get here once a call in the context is taking place¿ |
10:47.45 | onkeltimm | this is like a menu, isn't it? you have the s extensions. as soon as you dial #, asterisk should execute the hangup there? or am i braindead |
10:48.17 | Mother_ | yes, but you are dialing # on an active channel - do you mean that if I dialed another number during a call, asterisk will go through the whole smash? it will put the call into a context, start the call, etc??? |
10:48.18 | onkeltimm | i do something similar to let callers dial a known extension befor sending them to our operator |
10:48.38 | fenlander | Won't the H option to Dial just let you hangup with *? What don't I understand? |
10:50.02 | Mother_ | fenlander: I see what you mean now, sorry |
10:50.19 | onkeltimm | fenlander: misunderstood you, sorry. that may be an option. but you need to use h as this is callback. |
10:50.21 | Mother_ | adding H/h to the dial command, I misunderstood |
10:50.33 | Mother_ | let me try that... |
10:51.10 | fenlander | np - it might not work, but worth a try |
10:53.42 | *** join/#asterisk tessier (~treed@203.210.216.1) |
10:54.01 | Mother_ | no joy :( |
10:54.27 | onkeltimm | guys: http://voip-info.org/tiki-index.php?page=Asterisk+cmd+SIPCallPickup can I do something similar with IAX??? |
10:56.41 | Mother_ | what's wrong with pickupgroups etc.? |
11:01.03 | darwin35 | http://pastebin.ca/12068 if some with knowledge can look at this I have a issue with thr 3010150 and 4715227 not going to the vm mailbox and just going to falloff . |
11:02.11 | darwin35 | the 866 nmbr does like it should and goes to the vm |
11:03.02 | onkeltimm | Mother_: I cannot get it to work with Iax clients. |
11:03.08 | Mother_ | hmmkay |
11:03.30 | Mother_ | I think it works OK on one of my boxes with an IAX2 trunk to another * |
11:03.32 | RoyK | http://news.bbc.co.uk/1/hi/world/americas/4560847.stm <--US mulls ban on women in combat |
11:03.37 | RoyK | rotfl |
11:03.53 | Mother_ | i.e. the calls that come bridged from PSTN on the remote * can be picked up OK |
11:04.56 | onkeltimm | actually i am dumbstruck why this does not work. perhaps my config's just wrong... but http://voip-info.org/tiki-index.php?page=Asterisk%20callgroups%20and%20pickupgroups tells me it only works with mgcp, sip and sccp |
11:05.23 | *** join/#asterisk mattbridges (~mattb@mattbridges.plus.com) |
11:05.54 | mattbridges | hello all, can anyone help me to get UKCLID working on my X101P and BT Line? I've patched it but it's still not playing ball. |
11:06.01 | onkeltimm | and zap, of course |
11:06.42 | Mother_ | ah OK, well I use SIP phones and Zap channels, I've never tried with a native IAX client |
11:07.36 | onkeltimm | Mother_: seems like nobody else here did. have this problem for the whole week, it's perhaps costing me my job, and i am glad you at least answered. |
11:08.19 | onkeltimm | Mother_: the chan_iax.c code has nothing of callgroup/pickupgroup so it might not even be implemented |
11:08.55 | onkeltimm | s/iax/iax2/ |
11:08.55 | Mother_ | that kinda sucks |
11:09.09 | darwin35 | I cant even get feed back on one small issue dont feel bad |
11:10.05 | onkeltimm | and guys, i am ready to DO THIS if anyone of you would just help me get started. (explain the callgroup/pickupgroups internal workings to me) |
11:11.23 | darwin35 | from reading the wiki its all there |
11:11.38 | onkeltimm | darwin: url? |
11:11.55 | darwin35 | it talks about using *8 to puckup calls from outher phones |
11:12.35 | Mother_ | I've never really looked at the innards |
11:12.40 | onkeltimm | yeah, as i wrote before, it works only on SIP, SCCP, MGCP, and Zapata |
11:12.59 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
11:13.05 | Mother_ | darwin35: yes, but if you look here http://www.voip-info.org/wiki-Asterisk+config+iax.conf in the table, it doesn't say pickupgroup can be used anywhere |
11:13.22 | *** join/#asterisk m0f0x (m0f0x@m0f0x.user) |
11:13.28 | darwin35 | nope not supported IAX@ from what I know |
11:13.47 | darwin35 | if bkw comes back ask him |
11:13.52 | onkeltimm | ... why always me... ;) |
11:13.58 | darwin35 | IAX2 sorry |
11:15.26 | Mother_ | and you have tried adding pickupgroup and callgroup to iax.conf? just in case... |
11:15.35 | onkeltimm | Mother_: sure |
11:16.23 | onkeltimm | but it is not in the code. chan_sip.c has callgroup and pickupgroup in the user/peer structs. chan_iax2.c hasn't |
11:16.24 | Mother_ | damn... |
11:16.48 | Mother_ | hmkay - maybe I'll have to start learning C for real :) |
11:17.33 | Makenshi | does anyone know of a sip client for windows mobile that supports video? |
11:18.09 | onkeltimm | ~seen bkw |
11:18.10 | jbot | bkw <~bkw@u201.udal.afb.lu.se> was last seen on IRC in channel #debian, 88d 11h 17m 13s ago, saying: 'uhm, the latter one or both of them?'. |
11:18.46 | bewest | yay much progress |
11:21.11 | *** join/#asterisk mjlinda (~mjlinda@221.204.104.96) |
11:22.28 | Zeeek | it works (back from lunch) |
11:22.32 | Zeeek | ?? |
11:24.57 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
11:25.17 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
11:26.21 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
11:26.37 | tzafrir | How do I play a short "different" sound (to indicate an invalid operation)? |
11:26.52 | tzafrir | (in the dialplan) |
11:29.17 | Zeeek | you mean like windows? |
11:29.36 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
11:29.49 | *** join/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl) |
11:29.56 | tzafrir | Zeeek, I mean like \a :-) |
11:31.16 | Zeeek | what's wrong with <BELL> ? |
11:39.43 | tzafrir | Zeeek, how can I use <BELL> from the dialplan? |
11:42.14 | Zeeek | hok up a TTY as console? |
11:42.15 | Zeeek | that has the BELL key |
11:42.15 | Zeeek | but I assumed you could play a short sound file? |
11:42.15 | *** join/#asterisk ctooley (~ctooley@pc51.utati.net) |
11:42.15 | onkeltimm | tzafrir: what do you mean by invalid operation? in a menu, you could just define a fallthrough exten |
11:44.10 | langals | hi there...I am using IAX2 softphones, connecting to Meetme....when there are 3 or more users connecting from behind the same lan (.i.e going through the same NAT), then there are 3 second breaks every 30 seconds or so.....wondering if this problem might be because users all going through the same NAT?... |
11:44.20 | langals | ...bandwidth should not be a problem.... |
11:44.54 | langals | ...when I have 3 users, but all from behind different LANS, then don't have this problem.....any ideas? |
11:45.28 | tzafrir | onkeltimm, I want to put it in the i extension of an IVR. However I asked *what* to put and not *where*. |
11:46.00 | Zeeek | Whenever I call a friend from ZAP to Firefly (asterisk 1.0.6) the IAX2 the fact that the side answeres is not detected. Asterisk says "no one picked up afetr 25000ms" and goes to the "no one is available..." message. The connection is normal for both ends for the first n seconds, the n being the time in Dial() |
11:46.59 | onkeltimm | tzafrir: PlayBack(invalid) ? |
11:48.59 | tzafrir | This is what I currently use. But it may sound strange in an locally-recorded IVR. Oh, well |
11:49.26 | Zeeek | tzafrir can't you record the file you wish? |
11:49.40 | Zeeek | (with <BELL> in it) |
11:50.37 | *** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl) |
11:57.16 | *** join/#asterisk doughecka_ (~Tad@doughecka.user) |
11:59.07 | ctooley | langals, What kind of NAT box do you have? If it's slightly underpowered it might have problems keeping up with 3 constant streams of data like IAX |
11:59.42 | ctooley | Using NAT by itself shouldn't be an issue per se, but using NAT can lead to problems if the NAT box can't keep up. |
12:04.47 | Zeeek | langals if you mean all three are behind NAT on the same LAN, it would seem to be a bandidth issue on their side, no? |
12:06.50 | Blackvel | hmm question is still open for me. how the heck can I do direct ip calling into asterisk with xlite? :) I get Failed to authenticate user msg |
12:07.24 | langals | ctooley - not sure what kind of NAT it is - will try and find out |
12:07.38 | Zeeek | Blackvel - you mean yermutha@123.234.1.23 ? |
12:08.12 | Blackvel | jepp |
12:08.15 | langals | Zeeek - no - the asterisk box is actually also on the same network (altough outside the lan) - so there is plenty of bandwidth |
12:08.32 | Zeeek | langals FWIW, my SPeedtouch router/modem froze when I tried to run asterisk on it. Couldn't keep up apparently |
12:08.36 | robin_sz | quick question asterisk 1.0.7 on a 2.6 kernel .. do I need ztdummy to run a meetme room or do I remember it not being needed on 2.6 kernels? |
12:08.41 | *** join/#asterisk Dovid (~hirisk@pool-138-89-178-170.mad.east.verizon.net) |
12:08.42 | Blackvel | Zeeek: my sip.conf [general] contains context=directip |
12:08.54 | Blackvel | but there seems to be still a security issue |
12:08.57 | Zeeek | and in that context? |
12:09.09 | Blackvel | just failed msg, nothing more |
12:09.18 | Zeeek | what is in that context? |
12:09.26 | Blackvel | a dial |
12:09.35 | Zeeek | and what is in that dial ? :) |
12:09.48 | Zeeek | s,1,Dial... ? |
12:09.55 | Blackvel | haha |
12:10.13 | Blackvel | a user which asks me what the hell is zeeek asking you :) |
12:10.14 | tzafrir | robin_sz, ztdummy in 2.6 kernels is generally good enough (or at least better than the one in 2.4). But you still need one zaptel timing source as before |
12:10.27 | Blackvel | Zeeek: s,1,Dial + 66,1,Dial |
12:10.34 | robin_sz | tzafrir: OH. pooh. |
12:10.46 | Blackvel | so I expect that 66@192.168.1.2 and 192.168.1.2 is working |
12:10.55 | tzafrir | robin_sz, those are just modules, they don't require a reboot |
12:11.12 | Blackvel | but it does not execute the dail |
12:11.18 | robin_sz | tzafrir: but building ztdummy in debain sarge seems next to impossible |
12:11.26 | Blackvel | how can I send for direct ip something like insecure=very |
12:11.37 | Blackvel | can I do that in general section? |
12:12.03 | Zeeek | you have any disallow mask in general? |
12:12.33 | Zeeek | I'm pretty sure I can dial anything in my guest context from outside without id |
12:12.53 | Zeeek | but wait, maybe I've never tried |
12:13.22 | Blackvel | is the guest context that context=directip thing in sip.conf [general]? |
12:13.35 | Blackvel | sure I have disallow but also allow |
12:13.39 | Blackvel | g711 should work |
12:13.50 | Zeeek | I call it guest - you can call it [oral-sex] if you like :) |
12:14.12 | Zeeek | I'm referring to a netmask - sorry meant deny= |
12:15.17 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
12:15.21 | robin_sz | has anyone got any more clues about building zaptel/ztdummy on debian? someone must have doen this before .. I think even I did once ..!! .. sigh |
12:15.40 | Blackvel | upps |
12:15.46 | Blackvel | now it works with 66@192.168.1.2 |
12:15.47 | robin_sz | but that was months ago :( brain fade. |
12:15.48 | Blackvel | dunno why |
12:16.10 | *** join/#asterisk guyee (~izomtriko@nextra.nudli.equitas.hu) |
12:17.16 | guyee | does NE knows why my SIP client indicates ringing immediately without waiting for a result from the h323 gw? |
12:17.45 | cjk | robin_sz: i did it |
12:17.48 | cjk | its working |
12:19.09 | *** join/#asterisk ilium007 (~brantwint@220-253-92-177.QLD.netspace.net.au) |
12:19.34 | ilium007 | hi all |
12:19.37 | tzafrir | Is there a reason why in the CLI output the 'secret' of a SIP peer not shown and the 'secret' of an IAX peer is shown? |
12:19.53 | kapejod | guyee: you dont have the "r" dial option, do you? |
12:20.16 | ilium007 | i have jst managed to get my * box to route accall from 1 sip softphone to another. problem is i am getting an error stating: |
12:20.27 | tzafrir | (unless you use '<Set>' as your secret, that is |
12:20.37 | ilium007 | chan_sip.c process_sdp: No compatible codecs! |
12:20.38 | jakepdev[work] | tzafrir - is your secret in IAX.conf or are you specifying it in your dialplan? |
12:20.42 | ilium007 | can someone help ? |
12:21.05 | tzafrir | I refer to the secret entries for peers in sip.conf and iax.conf |
12:21.25 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
12:21.32 | guyee | kapejod: Grrrr... I had. thx. problem solved, I owe you a beer :) |
12:21.33 | jakepdev[work] | ilium007 - sounds like you need two codecs that are compatible for your two endpoints. what are your two endpoints? |
12:21.50 | ilium007 | i have x-lite on my mac osx and firefly on pc |
12:21.53 | ilium007 | windozwe |
12:22.05 | ilium007 | both have gsm i assume |
12:22.18 | ellvis | re |
12:22.24 | jakepdev[work] | do you have allow=gsm in sip.conf? |
12:22.39 | ilium007 | yes i do |
12:22.49 | ilium007 | dissallow=all |
12:22.52 | ilium007 | allow=gsm |
12:23.19 | jakepdev[work] | try allow=all (take out the disallow) and see what it comes up with |
12:23.34 | *** join/#asterisk dgillson (~dgillson@rrcs-24-97-7-126.nys.biz.rr.com) |
12:24.08 | ilium007 | k |
12:24.33 | kapejod | guyee: ok. when? where? ;) |
12:24.44 | RoyK | jakepdev[work]: no, just do disallow=all, allow=alaw |
12:24.48 | RoyK | start with that |
12:24.53 | ilium007 | dammm |
12:24.54 | RoyK | _always_ start with that |
12:25.00 | ilium007 | its all scratchy and foul |
12:25.04 | guyee | kapejod: anytime... er... what about Budapest, Hungary? :) |
12:25.25 | kapejod | guyee: will you cover travel and hotel expenses, too? ;-) |
12:25.27 | jakepdev[work] | ilium007 - try Roy's suggestion |
12:26.20 | ilium007 | same all scratchy and broken |
12:26.22 | ilium007 | :( |
12:26.33 | RoyK | wtf??? |
12:26.47 | kapejod | RoyK: he already agreed on that. |
12:26.51 | RoyK | even without a good SIP login, this host is able to make calls!! |
12:26.51 | guyee | kapejod: not really... I mean... I should look for a few more questions, and then maybe :) |
12:27.22 | jakepdev[work] | ilium007 - try ulaw |
12:27.29 | *** join/#asterisk nvrs (RUR@London-HSE-ppp3545191.sympatico.ca) |
12:27.34 | ilium007 | take out allow=all |
12:27.40 | jakepdev[work] | yes |
12:27.55 | jakepdev[work] | use disallow=all allow=alaw |
12:28.15 | ilium007 | ulaw or alaw ? |
12:28.16 | robin_sz | oh foo. sigh. damn. |
12:28.17 | jakepdev[work] | then try disallow=all allow=ulaw |
12:28.24 | ilium007 | ok |
12:28.45 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
12:29.09 | robin_sz | so anyother really good suggestions as to how to install a asterisk with a working meetme? |
12:29.29 | robin_sz | its looking too damn hard on debian (again) sigh |
12:29.31 | jakepdev[work] | robin - rapid? |
12:29.38 | jakepdev[work] | oh wait - that's debian |
12:29.38 | robin_sz | rapid? |
12:29.45 | tzafrir | ~rapid |
12:29.49 | jbot | [xorcom rapid] at http://www.xorcom.com/rapid.html . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir |
12:30.13 | robin_sz | tzafrir: AND IT HAS A WORKING ZTDUMMY? |
12:30.14 | RoyK | wtf________ |
12:30.21 | robin_sz | oops, didnt mean to shout :) |
12:30.24 | RoyK | I have no peers defined in sip.conf |
12:30.26 | RoyK | none |
12:30.28 | RoyK | whatsoever |
12:30.29 | vaewyn | tzafrir: bad link dude :} |
12:30.44 | tzafrir | robin_sz, sure. If the installer doesn't detect a TDM card installed, it will modprobe ztdummy |
12:30.49 | kapejod | RoyK: but you have a context in the general section. |
12:31.26 | tzafrir | <PROTECTED> |
12:31.44 | tzafrir | How do I update jbot information? |
12:31.45 | vaewyn | :} |
12:31.45 | onkeltimm | vaewyn, tzafrir : s/.html// |
12:31.55 | jakepdev[work] | jbot, rapid is [xorcom rapid] at http://www.xorcom.com/rapid . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir |
12:31.56 | jbot | ...but rapid is already something else... |
12:31.57 | *** join/#asterisk jetdotnet (jetdotnet@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
12:32.06 | jakepdev[work] | jbot - but i don't care |
12:32.15 | jakepdev[work] | jbot change it anyway |
12:32.26 | onkeltimm | jobot, forget rapid |
12:32.31 | onkeltimm | fsck |
12:32.34 | onkeltimm | jbot, forget rapid |
12:32.34 | jbot | onkeltimm: i forgot rapid |
12:32.41 | jakepdev[work] | jbot, rapid is [xorcom rapid] at http://www.xorcom.com/rapid . A Debian based distro that features an auto-installer which installs and configures both Debian, and Asterisk. Maintainer: tzafrir |
12:32.42 | jbot | okay, jakepdev[work] |
12:32.50 | jakepdev[work] | tnx |
12:33.03 | onkeltimm | np |
12:33.13 | tzafrir | thnkas |
12:33.36 | robin_sz | tzafrir: so .. the box I have has 2.4.27-2-686-smp already, would the zaptel-modules-2.4.27-2-686-smp_1.0.4-1_i386.deb solve my ztdummy problem you think??? |
12:33.52 | tzafrir | robin_sz, it should work, for that specific asterisk/zaptel version |
12:34.05 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
12:34.10 | *** join/#asterisk Jas_Williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com) |
12:34.18 | tzafrir | My "unstale" packages are at http://tzafrir.org.il/rapid/ |
12:34.44 | vaewyn | unstale bwahahaha... more correct than unstable these days |
12:35.24 | vaewyn | can't wait for the new debian... cause then testing and unstable will get 'unbunched' again |
12:36.18 | tzafrir | vaewyn, I currently use Sarge. But there are simply too much stuff waiting for Etch to start |
12:37.51 | *** join/#asterisk GrahamC (hidden-use@213-131-100-29.onyx.net) |
12:37.52 | vaewyn | I can never remember which names are which... but I use all the way to testing in 90% of the machines |
12:37.55 | ilium007 | can anyone suggest a good mac osx SIP client > |
12:38.07 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
12:40.34 | GrahamC | hi everyone. Think I need to load ztdummy on a suse 9.2 system to get meetme working. I get a compile error o rule to make target `modules'. Stop. Any ideas. I am newbie! |
12:41.12 | GrahamC | PS - Am using chan_capi on fritz card |
12:42.06 | darwin35 | xten works on the mac |
12:42.23 | robin_sz | tzafrir: thnaks, I installed the modules and was able to insmod zaptel and ztdummy .. now I get to see if they will actually work with my asterisk, its the current debian asterisk, 1.0.7 .. so it might be OK |
12:44.11 | darwin35 | http://pastebin.ca/12068 if some with knowledge can look at this I have a issue with thr 3010150 and 4715227 not going to the vm mailbox and just going to falloff . |
12:45.17 | vaewyn | darwin35: hehehe... 471 is our local exchange in 269 areacode :} |
12:45.38 | jakepdev | darwin - you say "Goto(autoattend,s,1)", but you don't show s,1 |
12:45.43 | darwin35 | this is in the 479 area code |
12:46.25 | vaewyn | jakepdev: yeah... is in the includes... but not sure it is correct |
12:46.50 | jakepdev | sorry - wrong person |
12:47.03 | jakepdev | vaewyn - show the include also |
12:47.08 | syle | http://www.sipgate.co.uk/catalog/product_info.php?products_id=10 |
12:47.20 | vaewyn | jakepdev: no... right person... I was just replying before he did |
12:47.24 | vaewyn | and is he showing them |
12:47.25 | darwin35 | it does |
12:47.26 | syle | can configure 2 sip accounts on same router :) |
12:47.40 | darwin35 | it shows with a incude day |
12:47.58 | darwin35 | it reads the tiiem first |
12:48.07 | darwin35 | to see if they are open or closed |
12:48.17 | vaewyn | 8667871709 does or does not work also? |
12:48.38 | darwin35 | the 866 works it goes to the vm |
12:48.52 | darwin35 | the other 2 nmbrs drop off |
12:49.39 | darwin35 | thats what confuses me |
12:49.44 | jakepdev | darwin - maybe i'm oversimpliying things here, but it would seem to me that you'd have a main context for the incoming and GotoIf based off of time to your seperate contexts |
12:49.59 | vaewyn | just out of curiosity... what does [selection] have in it? |
12:50.26 | vaewyn | anything with _XXXXXXX? |
12:50.27 | darwin35 | there is a selection part for at night h |
12:50.28 | vaewyn | or such? |
12:51.33 | darwin35 | we have 10 direct nmbrs |
12:51.42 | darwin35 | 1051 -0160 |
12:51.51 | darwin35 | they work fine |
12:52.20 | vaewyn | I was just thinking that since it is included first if there was a matching exten in there it would snatch it before those night/day ones would |
12:52.40 | darwin35 | but they ring the lines |
12:52.50 | darwin35 | its just when they hit the vm they fall off |
12:53.13 | darwin35 | am i going to have to manual set each one |
12:53.30 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
12:53.34 | darwin35 | to fix this |
12:53.48 | vaewyn | That's just odd... it should handle them all the same |
12:53.59 | darwin35 | its not |
12:54.01 | vaewyn | voicemail doesn't care where stuff is coming from |
12:54.20 | vaewyn | unless you do like s${EXTEN} or such |
12:54.24 | vaewyn | but that ain't in there |
12:55.09 | darwin35 | whe the 0150 and 5227 hit the s,3, line they fall off and you get the fall threw message that the call can not be completed at this time |
12:55.33 | mAsH` | sorry, anyone used h323 with * ? |
12:55.50 | robin_sz | tzafrir: are you the sarge asterisk maintainer? |
12:56.09 | *** part/#asterisk grolloj (~grolloj@dsl254-116-106.nyc1.dsl.speakeasy.net) |
12:56.09 | vaewyn | darwin35: try putting a s,104,Goto(4) in there |
12:56.10 | darwin35 | if I halt to I can set a dial line for each nmr |
12:56.14 | *** join/#asterisk HA (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
12:56.34 | robin_sz | hmm |
12:56.34 | vaewyn | just out of curiosity |
12:56.36 | robin_sz | res_musiconhold.c:278 monmp3thread: unable to spawn mp3player |
12:56.58 | robin_sz | I guess I need an mp3 library, any ideas which one |
12:57.16 | robin_sz | sorry about he random questions .. stressed here :) |
12:57.36 | vaewyn | type 'make mpg123' in your asterisk source dir.... it will do everything for you |
12:58.19 | vaewyn | gets the correct version and everything |
12:59.29 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
13:00.32 | robin_sz | vaewyn: well, it seems to have done someting ... |
13:02.13 | darwin35 | nope |
13:02.29 | darwin35 | now it loops and dies |
13:02.50 | vaewyn | did it hit the Goto? |
13:02.53 | darwin35 | am jus going to hard set this one |
13:02.59 | darwin35 | yes |
13:03.29 | vaewyn | that's just freeky |
13:03.33 | darwin35 | yeah |
13:03.39 | darwin35 | its pissing me off |
13:03.48 | MikeJ[zzzzzzz] | darwin35, did you get your compiling prob fixed? |
13:03.50 | robin_sz | wee |
13:04.58 | vaewyn | darwin35: also... might want to change your comments... you have both a day and night saying 'OPEN w00t!' |
13:05.18 | *** part/#asterisk n4y (~frodo7@host-ip237-209.crowley.pl) |
13:05.22 | vaewyn | or change the included context :P |
13:05.30 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
13:05.52 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
13:07.13 | tzafrir | robin_sz, no, but I'm on the pkg-voip mailing list and keep in touch with them |
13:07.55 | tzafrir | The "maintainer" of the asterisk packages in Debian is the "pkg-voip" team |
13:08.44 | tzafrir | robin_sz, mpg123 is available from non-free |
13:08.54 | tzafrir | http://packages.debian.org/mpg123 |
13:09.15 | vaewyn | tzafrir: some people have been having trouble with the newer ones... |
13:09.27 | vaewyn | is why the make mpg123 thing showed up :P |
13:09.33 | tzafrir | This is the latest 0.59*r( |
13:09.39 | tzafrir | s/(/*/ |
13:10.03 | tzafrir | It should work fine, from what I understand. |
13:10.06 | Mother_ | HAH! |
13:10.09 | Mother_ | it verks!!! |
13:10.15 | vaewyn | Su Madre! |
13:10.17 | Mother_ | lol |
13:11.06 | tzafrir | robin_sz, I have also packaged separately the moh files removed from the debian package. They are in my "unstable" directory in the package asterisk-sounds-moh |
13:11.24 | tzafrir | I do hope to package alternative sounds soon to avoid this silly license issue |
13:11.48 | *** join/#asterisk PandemiK (~PandemiK@62.2.255.42) |
13:11.52 | PandemiK | hi every1 |
13:15.16 | Mother_ | tzafrir: there was a CD in Wired a couple months ago with music one could freely distribute, package, sample, etc. |
13:15.36 | Mother_ | some of it was OK for moh if you don't want to pay royalties |
13:15.36 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
13:15.52 | km- | howdy! |
13:15.55 | Mother_ | in any case, it would be überlame if the RIAA started suing people for their moh music :D |
13:16.29 | Mother_ | hiya |
13:16.37 | km- | hey, I hear there are issues with the 7.x series of 7960 firmwares, anyone having the same issues? |
13:17.18 | darwin35 | thats for testing |
13:17.45 | darwin35 | so I can dial in at any time |
13:18.22 | sudhir492 | can someone explain callgroup and pickup group to me? |
13:18.27 | HA | is there a dev call today? if so, what time? |
13:18.28 | Mother_ | km-: I have 7.3 and no issues so far |
13:18.44 | Mother_ | I think 7.4 is out but I've not tried it |
13:18.47 | km- | mother: cool. My supplier threw 6.3 onto the phone, I'll give that a go for a while |
13:18.53 | km- | if it ain't broke don't fix it, I guess! |
13:18.54 | darwin35 | around 1 cst 2 est |
13:18.57 | Mother_ | yep, indeed |
13:19.13 | Mother_ | gotta go eat, cya all |
13:19.15 | HA | ty darwin. i might actually be able to listen in today. |
13:19.18 | km- | later Mother |
13:19.26 | darwin35 | cool |
13:21.06 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
13:21.27 | km- | I hate it when I misplace stuff! |
13:21.40 | km- | does anyone have the program that allows you to convert graphics to logos for the 7960? |
13:21.46 | HA | Unless of course we manage to get the financing approved for this new system and then I'll be tied up trying to buy all this crap. I really do enjoy spending other peoples money but having to spend $21,000 in a single day gets pretty tiring. |
13:22.57 | ariel_ | hello everyone |
13:23.09 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
13:23.09 | *** mode/#asterisk [+o bkw_] by ChanServ |
13:23.27 | iCEBrkr | km-: You need to start maintaining your own personal wiki |
13:23.48 | iCEBrkr | km-: That way you just dump all your how-to's and other configs/setup files there |
13:23.50 | km- | icebrkr: that's not a half bad idea |
13:23.55 | km- | bkw: moose penis? |
13:24.02 | *** join/#asterisk riksta (~rick@host81-155-216-40.range81-155.btcentralplus.com) |
13:24.05 | km- | bkw: dude, do you still have the 7960 configuration goodies like the logo creation program? |
13:24.15 | PandemiK | HA: what are you calling a dev call ? |
13:24.35 | iCEBrkr | km-: I started one last night actually, cuz I'm tired of hacking out asterisk confg file and scripts only to forget them 3wks later. |
13:24.43 | km- | icebrkr: hahaha, awesome |
13:24.48 | km- | I've started hacking on asterisk-java |
13:24.57 | km- | it gives me a semi |
13:25.01 | Jas_Williams | km-: it is not required in the latest release it reads standard graphics files |
13:25.08 | iCEBrkr | dude, 2.5hrs trying to get my iax.conf stuff working with Voicepulse.. It was working one day and then just stopped.. I dunno WTF I did.. |
13:25.13 | km- | jas_williams: i.e., version 7? |
13:25.16 | iCEBrkr | hahha |
13:25.23 | *** join/#asterisk astoria (~haydenth@66.235.201.217) |
13:25.29 | km- | icebrkr: huh, that sucks pretty hard |
13:25.33 | iCEBrkr | and the voip-info wiki's iax.conf foe Voicepulse is WRONG |
13:25.40 | iCEBrkr | s/foe/for |
13:25.48 | km- | I actually can't stand voicepulse |
13:25.56 | iCEBrkr | It works. *Shrug* |
13:26.00 | astoria | Hey, i got a stupid problem. two polycom phones, both can make outgoing IAX calls okay, but sip calls between the two bridge okay, but no sound ... sorry to be vague.. |
13:26.03 | km- | good luck getting support |
13:26.03 | Jas_Williams | km-: POS3-07-4-00 |
13:26.04 | iCEBrkr | Tho, I think my config broke cuz they changed something. |
13:26.12 | km- | jas_williams: ahh, thanks for the tip. |
13:26.18 | iCEBrkr | km-: I never call support :P |
13:26.19 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
13:26.26 | km- | jas_williams: I hear there are misc oddities for the 7 series though |
13:26.35 | km- | at least, that's what the guy at gts was tellin me |
13:26.51 | iCEBrkr | km-: You ever tinker with eagi? |
13:27.01 | Jas_Williams | km-: works fine for me and has done for a while now. Upgrading can be fun th |
13:27.02 | iCEBrkr | km-: I got sphinx to compile and actually work :) |
13:28.20 | iCEBrkr | I'm trying to clean up my asterisk install.. Got voicepulse working again, inbound and outbound FWD working again too. |
13:29.13 | HA | PandemiK: IIRC there is a * developers conference call on Thursdays. And according to darwin, my memory is working fine at the moment. |
13:29.25 | iCEBrkr | haha |
13:29.36 | iCEBrkr | Darwin is on vacation. |
13:29.37 | PandemiK | ok |
13:29.46 | PandemiK | thx |
13:29.55 | HA | Even though I did stay up all night to go see a movie at 3:30 in the morning. |
13:30.11 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
13:30.18 | km- | icebrkr: what's sphinx? |
13:30.25 | km- | word, episode 3 |
13:30.29 | iCEBrkr | km-: Speech recognition |
13:30.33 | *** join/#asterisk toot (~chris@217.30.126.10) |
13:30.33 | vaewyn | *spoiler* Anakin is Vader!!! */spoiler* |
13:30.35 | vaewyn | :} |
13:30.38 | km- | icebrkr: nice, did it actually recognize speech? |
13:30.43 | km- | vaewyn: NOOOOOOOOOOOOOOOOOOOOOOOOOOOOOO!!!! |
13:30.46 | iCEBrkr | km-: I got it working on my workstation. |
13:30.50 | toot | hey just had 2 call/hangups. then the next call i got the following Detected alarm on channel 1: Red Alarm |
13:30.51 | km- | icebrkr: cool |
13:31.03 | iCEBrkr | km-: I just dunno how I'm gonna get it to integrate with asterisk. |
13:31.09 | mishehu | *spoiler* Chewbacca is what a wookie! */spoiler* |
13:31.12 | vaewyn | and Padme is made out of Hot Grits... |
13:31.14 | toot | now nuffing seems to happen when a call comes in - has it hung the card in some way? |
13:31.16 | iCEBrkr | the eagi-sphinx-test is just a framework |
13:31.19 | PandemiK | anyone using budgetone here ? (although my problem is perhaps not from the GS) |
13:31.20 | Pete_Largo | vaewyn, I think that was spoiled in episode 6 - Return of the Jedi... |
13:31.30 | mishehu | vaewyn: those are some hot grits |
13:31.35 | mishehu | I'd eat those hot grits. |
13:31.37 | iCEBrkr | PandemiK: Yea, I got a BT100 |
13:31.37 | HA | now we just need 7, 8, and 9 so we can see if the republic is restored or if it remains an empire. |
13:31.37 | vaewyn | mishehu: amen! |
13:31.54 | PandemiK | iCEBrkr: I have some problem, when I'm called, there's no sound |
13:32.00 | vaewyn | HA: Anarchy rules... ;P |
13:32.11 | iCEBrkr | PandemiK: NAT'd? |
13:32.13 | mishehu | bastard friend of mine who studies at the hebrew u had no idea that natali portman was studying there... |
13:32.18 | mishehu | heh. |
13:32.27 | PandemiK | iCEBrkr: no, my asterisk is on the same subnet |
13:32.27 | sudhir492 | <mAsH`: I am using H323 with * |
13:32.47 | PandemiK | iCEBrkr: it was working great and ... nothing |
13:32.48 | iCEBrkr | PandemiK: So Asterisk and your BT have public IPs? |
13:33.09 | PandemiK | iCEBrkr: no, they're on the same private address subnet |
13:33.10 | toot | hmm show channels now shows 0 channels, but ztcfg -v shows 1 channel configured? |
13:33.17 | PandemiK | iCEBrkr: a LAN ;-) |
13:33.22 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
13:33.31 | iCEBrkr | Oh, you're making extension->extension local calls.... |
13:33.44 | PandemiK | yeah |
13:33.49 | iCEBrkr | Well, hell.. |
13:33.52 | iCEBrkr | That should be cake. |
13:34.01 | PandemiK | I have my BT102 and a softphone (X-lite) |
13:34.16 | iCEBrkr | Well, make sure X-Lite is actually 'hearing' your mic. :P |
13:34.37 | PandemiK | When I call from BT102, it works, and when I call from X-lite, no sound |
13:34.54 | iCEBrkr | Pretty interesting. |
13:34.57 | PandemiK | yeah |
13:34.58 | iCEBrkr | I don't think it's your BT tho. |
13:35.10 | PandemiK | I forced the codec to ulaw |
13:35.16 | PandemiK | and the extensions are the same |
13:35.19 | PandemiK | pretty simple |
13:35.23 | PandemiK | Dial, Hangup |
13:35.24 | *** join/#asterisk doolph (doolph@200.46.148.35) |
13:35.36 | PandemiK | hello doolph |
13:35.45 | doolph | what's new |
13:35.59 | iCEBrkr | PandemiK: you try 'sip debug' at the console? |
13:36.09 | PandemiK | not actually |
13:36.11 | iCEBrkr | PandemiK: also set verbose 9 |
13:36.18 | PandemiK | I only run asterisk with verbose 3 |
13:36.29 | iCEBrkr | I dunno what the upper limit is. :P |
13:36.34 | iCEBrkr | But 9 works for me |
13:38.58 | PandemiK | hmm |
13:40.25 | PandemiK | I can't get anything from sip debug |
13:40.41 | iCEBrkr | It's showing debug info or no? |
13:40.45 | PandemiK | yeah |
13:40.50 | kdayn | can somebody give me a sample CDR file, I want to test some things, but I dont have one to do that |
13:40.54 | PandemiK | but reading it don't give me any clue |
13:41.08 | iCEBrkr | kdayn: It's just a CSV file. |
13:41.15 | kdayn | i know |
13:41.26 | iCEBrkr | make one up |
13:41.30 | kdayn | but i would like to see real copy from asterisk |
13:42.15 | kdayn | matbe somebody can send it via dcc? :) |
13:42.19 | PandemiK | eeeh, what does that mean : == Spawn extension (sip, 1002, 1) exited non-zero on 'SIP/1001-91a3' |
13:42.54 | iCEBrkr | kdayn: |
13:42.54 | iCEBrkr | "","kphone","1234","default","""iCE Breaker"" <kphone>","SIP/kphone-1bd3","","VoiceMail","b1234","2004-03-21 22:35:14","2004-03-21 22:35:14","2004-03-21 22:35:33",19,19,"ANSWERED","DOCUMENTATION" |
13:42.59 | iCEBrkr | have fun. |
13:42.59 | kdayn | one question, asterrski uses udp, what linux based clients would you suggest? |
13:43.05 | kdayn | iCEBrkr: thx |
13:43.24 | vaewyn | iaxcomm for iax connections... |
13:43.40 | vaewyn | not sure on sip cause I don't use sip softphones :P |
13:43.45 | PandemiK | linphone, gnophone for sip |
13:43.47 | onkeltimm | asterrski, that the russion l10n of * ? |
13:43.59 | onkeltimm | sncr |
13:44.01 | PandemiK | I've heard that x-lite is available for linux too |
13:44.42 | vaewyn | If you have a choice use IAX and iaxcomm though... it will work a ton more places |
13:47.27 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
13:48.35 | zoa | dont forget idefisk !!! |
13:48.45 | zoa | ah idefisk is not yet for linux |
13:48.56 | zoa | although it works with wine |
13:50.06 | PandemiK | I have to go, but I'll be back in one hour with my problems ;-) |
13:53.43 | iCEBrkr | blah blah blah |
14:07.15 | *** join/#asterisk ManxPwr (~eric@dsl-209-205-172-111.i-55.com) |
14:07.34 | ManxPwr | Does anyone know what company makes the tbird brand of test equipment? |
14:08.30 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
14:09.37 | Pete_Largo | Acterna |
14:09.43 | Pete_Largo | use to be called TTC |
14:10.02 | Pete_Largo | and it's T-BERD (Bit Error Rate Detector?) |
14:10.12 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
14:10.42 | Pete_Largo | aren't you supposed to be in europe? |
14:10.51 | Umaro | hey guys, anyone else having problems with there inbound DIDs from nufone atm? |
14:12.24 | ManxPwr | Pete_Largo, I leave this evening |
14:13.31 | robin_sz | hmm m.. weird ... the musiconhold plays for about 0.5seconds and then craps out |
14:14.39 | ManxPwr | robin_sz, That would be expected if your SIP client has VAD enabled. |
14:14.44 | *** join/#asterisk file[class] (~jcolp@66.199.241.90) |
14:14.48 | file[class] | meep meep |
14:14.56 | ManxPwr | Pete_Largo, Do you know if the same company makes test equipment for analog lines? |
14:15.02 | robin_sz | ManxPwr: its an IAX client |
14:15.21 | robin_sz | file[class]: meep! |
14:15.33 | file[class] | hiiiiiii |
14:15.56 | robin_sz | and .. if I use a web based client, can I use the same user/pass on all the clients ?? |
14:16.18 | Pete_Largo | ManxPwr - you mean like for your home phone? |
14:16.31 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:16.31 | *** mode/#asterisk [+o anthm] by ChanServ |
14:16.40 | robin_sz | I mean for like web users calling into our sales team |
14:16.50 | robin_sz | via a web phone thing .. |
14:17.02 | file[class] | hi anthm |
14:17.11 | robin_sz | can I just set a single, hardwired "webuser:webpass" ? |
14:17.12 | ManxPwr | Pete_Largo, Well yes, but specifically for analog loops (i.e. wall jack to channel bank) |
14:17.13 | anthm | hi file with class |
14:17.13 | *** join/#asterisk freat (~freat@c-67-162-109-129.hsd1.il.comcast.net) |
14:17.19 | robin_sz | or does it need to be unique? |
14:17.20 | freat | anyone else Teliax? |
14:17.26 | freat | I've got 2 sites down right now |
14:17.27 | ManxPwr | freat, I do. |
14:17.35 | freat | try calling in |
14:17.48 | freat | or out |
14:18.03 | Pete_Largo | you'll need what's called a "Butt Set" don't laugh. the most common brand is Harris, but I use something called a 'lil Buttie' |
14:18.35 | freat | ManxPwr: let me know if it's working for you |
14:18.40 | ManxPwr | freat, looks broken |
14:18.44 | freat | :( |
14:18.47 | freat | crap |
14:19.01 | file[class] | you all need to attend ClueCon |
14:19.03 | file[class] | it's where it's at! |
14:19.10 | ManxPwr | Pete_Largo, A butt set isn't going to measure db levels of audio, loop length (cap kick), ohms |
14:19.19 | robin_sz | cluecon is gay! |
14:19.27 | ManxPwr | freat, teliax is usually pretty reliable. |
14:19.33 | file[class] | and whoever owes me stuff for help better go! |
14:19.42 | ManxPwr | freat, go to their site, go to support, click on the web chat app |
14:19.43 | file[class] | so you can buy me drinks :P |
14:19.53 | robin_sz | its in the USA? |
14:19.54 | HA | cluecon is in august right? if so, then i try to swing by if things work out financially. |
14:20.02 | doolph | what is Queues |
14:20.04 | file[class] | yes |
14:20.11 | file[class] | Cluecon is August 3rd to 5th in Chicago, IL |
14:20.32 | ManxPwr | robin_sz, Not everyone in the USA likes bad beer |
14:20.34 | Zeeek | teliax is "unreachable" here too |
14:20.36 | *** join/#asterisk Qiang-zh (~Qiang-zh@CPE00112f93a8b1-CM00003964ca63.cpe.net.cable.rogers.com) |
14:20.47 | robin_sz | ManxPwr: true, but its all they can get |
14:21.08 | ManxPwr | Pete_Largo, Right now we need something to generate 1000hz / 0db test tone, and measure the db level |
14:21.22 | file[class] | anthm: I looked at the design studies Chad did :) some of the stuff looks great |
14:21.25 | robin_sz | ManxPwr: in analogue? |
14:21.32 | ManxPwr | robin_sz, no, the isa does have some good beer, but only from the smaller breweries |
14:21.43 | anthm | which ones you like |
14:21.52 | ManxPwr | anthm, Sierra Navada is good. |
14:21.54 | robin_sz | Theakstons Old Peculiar |
14:22.01 | file[class] | anthm: I can't remember now, I looked lastnight |
14:22.07 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
14:22.19 | robin_sz | I fed that to a 'mercan once ... he failed to finish his pint. |
14:22.27 | file[class] | anthm: all the entry slips looked great, and the ID tags were great too... hard to decide on the best |
14:22.44 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
14:22.46 | anthm | oh we talking beer and cluecon logos at the same time lol |
14:23.03 | morris | how can i confirm installation of x100p clone ? |
14:23.08 | file[class] | well there was a pic where beer was associated with cluecon... |
14:23.12 | morris | ive installed zaptel |
14:23.23 | morris | i have no idea how i check it working |
14:23.47 | robin_sz | ManxPwr: if you are in the UK and want a good solid audio test set, get an ex-bbc EP14 test set. utterley dependable. |
14:23.49 | morris | i have rang the pbx line and asterisk doesnt show anything relating to it being dialed.. i guessed it should say something even if there was no rule |
14:23.59 | Pete_Largo | http://www.acterna.com/global/products/finder/products_by_technology.html#3 |
14:24.02 | ManxPwr | rob-, usa |
14:24.06 | anthm | Grolsch and Hacker Pschorr Weiss ! |
14:24.09 | freat | robin_sz: Old Peculier (I believe this is how it's spelled) is quite nice. molassas kind of flavor (mmmmmm....) |
14:24.12 | ManxPwr | robin_sz, usa |
14:24.22 | freat | and I'm in Chicago, USA |
14:24.25 | freat | ;) |
14:24.42 | anthm | yah you are in chicago , come to clue con bring your friends |
14:24.45 | file[class] | freat: you should attend Cluecon |
14:24.53 | ManxPwr | Linksys routers suck |
14:24.56 | freat | really it's in Chicago!!! |
14:24.58 | freat | ?? |
14:25.04 | file[class] | yes |
14:25.07 | Zeeek | Linksys routers rock |
14:25.08 | freat | yes! |
14:25.08 | file[class] | http://www.cluecon.com/ for details |
14:25.15 | Ahrimanes | who talked about beer?? |
14:25.24 | ManxPwr | Zeeek, not when they start crashing when put plug a SIP device into them |
14:25.26 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
14:25.32 | freat | I'm gonna have to go drinking if Teliax doesn't come back up |
14:25.37 | Zeeek | my routers don't do SIP! |
14:25.42 | freat | 2 corporate offices are down now... |
14:25.46 | Zeeek | freat have 10 providers |
14:25.49 | ManxPwr | Zeeek, neither does the linksys |
14:25.52 | file[class] | anthm: I wish we had more confirmed people ;( |
14:26.06 | freat | Zeeek: how do you handle inbound calls though? |
14:26.10 | ManxPwr | freat, Dude, if you rely on VoIP for mission critical stuff...well just kill yourself now. |
14:26.21 | Zeeek | ah, that's tough. Nufone has some kind of alternate routing |
14:26.28 | freat | hmm |
14:26.36 | Zeeek | be sure and use voip for all your 911 nees |
14:26.43 | freat | haha |
14:26.44 | ManxPwr | freat, What did teliax say when you talked to them? |
14:26.50 | Zeeek | heh |
14:26.52 | freat | you can't call them |
14:26.53 | freat | hehe |
14:26.54 | *** join/#asterisk jbjb (ito@justine.donck.net) |
14:26.56 | anthm | we just gotta keep reminding the ppl with so many questions that we have all the answers ... at , you guesed it! ClueCon! |
14:26.56 | Zeeek | email? |
14:27.03 | freat | trying |
14:27.10 | ManxPwr | freat, um, I always got very goos uspport via their web chat support app |
14:27.12 | freat | they just went down maybe 20 minutes ago when I was on the phone |
14:27.12 | vaewyn | I should come to cluecon even if work won't pay for it... is so close by and I get to meet more devel types :} |
14:27.18 | freat | yeah nobody is online there |
14:27.29 | ManxPwr | freat, ah. wait a few mins and try again |
14:27.32 | Zeeek | mayeb the dialup link went down? |
14:27.34 | syle | yeah i had this linksys , netgear router fight with people before, just go dlink all works good |
14:27.43 | jbjb | Hello ... Ik have build errors on mac os 10.3 ld: Undefined symbols: |
14:27.44 | jbjb | _res_9_dn_expand |
14:27.44 | jbjb | _res_9_nclose |
14:27.44 | jbjb | _res_9_ninit |
14:27.44 | jbjb | _res_9_nsearch |
14:27.44 | ManxPwr | syle, naw, get cisco |
14:27.44 | jbjb | Makenshi: *** [asterisk] Error 1 |
14:27.48 | Zeeek | netgear sucks! |
14:27.49 | jbjb | any one ? |
14:27.50 | freat | gonna switch outbound to voipjet now... |
14:27.57 | file[class] | come to Cluecon where ALL your questions will be answered, such as "Why can't I use _." |
14:27.58 | syle | cisco, whos gonna pay that price lol |
14:28.06 | vaewyn | ManxPwr: linksys is cisco now :} |
14:28.08 | anthm | work should pay for it! , we offer a free service where file and bkw will call your boss and urge him to send you |
14:28.13 | ManxPwr | syle, someone that wants something reliable. |
14:28.15 | file[class] | indeed |
14:28.23 | vaewyn | anthm: hehehehe :} |
14:28.23 | ManxPwr | vaewyn, Perhaps I should have said "an IOS router" |
14:28.24 | file[class] | we are VERY good at that |
14:28.42 | HA | anthm: just how much encouragement are we talking here? |
14:29.01 | Zeeek | anyone have any vintage "Cisco Kid" eposodes? It would be a great CluCon opener :) |
14:29.10 | ManxPwr | I suspect that a rev2 or rev3 of the linksys would work |
14:29.30 | vaewyn | Heck... I love WRT54G units... they rock |
14:29.34 | syle | oiww fuck i hate this, i get a popup from MS after its update saying would you like to restart now?....i say no...it keeps popping up and saying that every couple minutes, most annoying thing ever |
14:29.55 | Pete_Largo | that does suck syle |
14:30.08 | syle | have to reboot i guess what a joke |
14:30.15 | vaewyn | syle: Your only 1 knoppix disk away from fixing that permanently :P |
14:30.22 | astoria | Isn't that the slammer virus, syle? |
14:30.38 | astoria | :) |
14:30.49 | vaewyn | nah... just the M$ virus |
14:30.54 | HA | is this overkill or just right for doing ds3 to t1 conversions? http://www.tritechcoa.com/product/291967.html |
14:31.04 | file[class] | Cluecon, you MUST and you WILL go to it |
14:31.25 | HA | file: what's the cost? |
14:31.59 | file[class] | $650 |
14:32.05 | tzanger | more than my ass can afford |
14:32.05 | file[class] | but that includes your hotel and lunches |
14:32.18 | astoria | anyone going to supercomm? |
14:32.31 | anthm | just email sellmyboss@cluecon.com with the phone # and we will call and ask him to send you |
14:33.23 | astoria | damn, i want to go to cluecon now! |
14:33.24 | ManxPwr | The cost is: one beer for ManxPower |
14:33.39 | ManxPwr | It's one of the lower cost VoIP events |
14:33.45 | Chuji | Too many conventions |
14:33.59 | HA | anthm: oh yeah, that's exactly what i need. i've already asked my boss to spend $21,000 to put in a fancy new * system, and now I'm gonna ask him to pay airfare and con costs for me to go hang out with a bunch if assgeeks. |
14:34.26 | anthm | http://www.cluecon.com/images/shirtdesign/Carcrash.jpg |
14:34.28 | tzanger | anthm: one patch from 4222 got in |
14:34.31 | astoria | $21,000 for an asterisk system? thats gotta be some pretty hardcore hardware |
14:34.31 | ManxPwr | HA: No, it's "training courses" |
14:34.33 | tzanger | anthm: but not hte CNG one yet |
14:34.37 | tzanger | anthm: and my bridge is working now |
14:34.48 | anthm | yay |
14:34.58 | tzanger | anthm: I spoke with Mark; all sounds in zaptel are interruptible, including DTMF. It's intentional |
14:35.01 | HA | 'Training Courses'...that might work. |
14:35.03 | anthm | hmm who charges 21k to learn asterisk ? |
14:35.12 | jbjb | nevermind ... |
14:35.13 | tzanger | anthm: so I fix my bridge app such that it doesn't write until I get the DTMF complete event and the bridge works as intended |
14:35.21 | tzanger | anthm: but unfortunately it does not exhibit the problem <grrr> |
14:35.34 | anthm | still 200ms shit ? |
14:35.48 | tzanger | anthm: so... for my next trick, I am going to dlopen() chan_zap.so and try to recreate it with that |
14:36.00 | ctooley | Uh, $21,000 for a complete system, implemented and training courses sounds pretty cheap to me. |
14:36.00 | file[class] | tzanger: wow, good luck with that |
14:36.00 | doolph | 21k? |
14:36.05 | Pete_Largo | HA - http://www.interlinkweb.com/systemics/product.asp?sku=ADT+4205290L11 |
14:36.17 | anthm | step right up |
14:36.20 | *** join/#asterisk syle (~blah@wnpgmb02dc1-61-40.dynamic.mts.net) |
14:36.33 | doolph | ctooley what's a good price for you |
14:36.43 | file[class] | oh as well at Cluecon there will be uber-secret stuff that anyone who attends will see |
14:36.50 | astoria | It's the phones that really kill ya on cost.. |
14:36.52 | file[class] | so see it first, at Cluecon! |
14:37.04 | ctooley | doolph, depends on the number of phones, lines, and level of flexibility |
14:37.05 | doolph | i cant go to cluecon |
14:37.30 | tzanger | anthm: my bridge app does not show any data loss from zaptel when DTMF is being played |
14:37.45 | vaewyn | Anyone got experience with checking/setting MWI lights on a norhell system from a * system? |
14:37.46 | Pete_Largo | HA - http://www.telepricing.com/wwwboard/messages/11541.html |
14:37.49 | tzanger | I get audio from both fds, alternating, every 20ms |
14:37.59 | file[class] | c'mon folks - it's not that expensive! |
14:38.01 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
14:38.05 | tzanger | vaewyn: with an ATA on the nor*, *1<exten> |
14:38.07 | tzanger | toggles it |
14:38.11 | tzanger | no way to check it though |
14:38.24 | file[class] | 3 nights hotel, lunches, plus the actual conference for $650! |
14:38.25 | doolph | because you are a rich guy |
14:38.51 | *** join/#asterisk Andy-AAH (~piekoff59@CPE00508b0d60a7-CM000039b80a5c.cpe.net.cable.rogers.com) |
14:39.17 | HA | Pete_Largo: In that case...http://www.tritechcoa.com/product/291959.html |
14:39.27 | file[class] | j/k |
14:39.42 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
14:40.13 | astoria | how long do you have to sign up for cluecon? |
14:40.20 | HA | file:i also have to come up with money for getting there and back from Tulsa, OK...and I'm not in the mood to hitch 700 miles. |
14:40.21 | astoria | how many attendees are there? |
14:40.24 | anthm | until it arrives! |
14:40.28 | doolph | file can I msg you |
14:40.42 | anthm | ask bkw for a ride ! |
14:40.55 | doolph | its for business |
14:40.59 | astoria | is the con anywhere near the L? |
14:41.07 | HA | anthm: where is bkw coming from and do I have to listen to him scream NEXT! all the way there and back? |
14:41.07 | file[class] | HA: you should talk to bkw_ |
14:41.12 | anthm | elk grove village near the airport |
14:41.23 | file[class] | HA: he's coming from Mcallister |
14:41.45 | HA | Son-of-a...no wonder he keeps yelling NEXT!!! |
14:41.48 | file[class] | (however you spell it) |
14:42.49 | Pete_Largo | HA - did you look for http://www.carrieraccess.com/products/index.cfm/fa/widebank28.htm these as well? |
14:42.59 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
14:43.17 | *** join/#asterisk szw2001 (~vip@web1.ningo.net) |
14:43.51 | HA | Pete_Largo: What exactly is that? |
14:43.57 | Pete_Largo | M13 Mux |
14:45.05 | Pete_Largo | DS3 in DS1 (T1) out |
14:45.05 | *** part/#asterisk Qiang-zh (~Qiang-zh@CPE00112f93a8b1-CM00003964ca63.cpe.net.cable.rogers.com) |
14:45.46 | HA | Isn't that the same as this? http://www.tritechcoa.com/product/291959.html |
14:45.55 | darkskiez | if you have a E100P and a tdm400 card, what determines what order the channel numbers are, eg, what makes the pri channels 1-32 and the tdm 33-36 ? |
14:45.59 | morris | Anyone available to spend some time with me to understand what needs doing to get my ptsn line working with asterisk? |
14:46.29 | anthm | come to cluecon and learn how to get your pstn line working with asterisk |
14:46.36 | morris | i would love too |
14:46.38 | file[class] | and how to make your asterisk bake muffins! |
14:46.39 | morris | where is it btw? |
14:46.39 | HA | morris: about $2,000 USD will get it up and running. |
14:46.53 | morris | how about your lick my sweaty nut sack ? |
14:47.00 | darkskiez | deal |
14:47.02 | morris | you* |
14:47.05 | darkskiez | :) |
14:47.08 | morris | lol |
14:47.10 | morris | good lad ;p |
14:47.12 | anthm | see the topic |
14:47.17 | HA | we charge more for that. |
14:47.21 | morris | ah i have no passport ;/ |
14:47.26 | Zeeek | morris and after all the promise you showed yesterday! |
14:47.30 | anthm | PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm |
14:47.37 | morris | zeeek ;p |
14:47.43 | HA | this is America. You don't need a passport, just come thru Mexico. |
14:47.46 | Pete_Largo | HA - that would work fine, Adtran and Carrier Access are direct competitors |
14:47.50 | morris | i have been trying ;p |
14:47.54 | tzanger | HA: hahahaha |
14:47.58 | morris | *goes back to the books* ;p |
14:48.20 | Zeeek | morris there is a special program I can get you in if you qualify as a triple minority |
14:48.37 | morris | wtf lol |
14:48.56 | [TK]D-Fender | file[class] : Muffins? Eay enough. I've got my * setup running my X-10 modules, so you'd just need a oven that is 110v switchable like a Barbie EZ-Bake oven and you're set! |
14:48.59 | HA | Pete_Largo: OK. Just making sure it's the right equipment. Didn't know we needed that until late yesterday right before we were about to tell the boss how much it was actually gonna cost to get his new 'high priority' project in place. |
14:49.06 | file[class] | [TK]D-Fender: yay |
14:49.10 | Zeeek | other possibilities include being in the UK in BT or trying to use Broadvoice |
14:49.29 | Pete_Largo | fast learning on your part :) |
14:49.32 | [TK]D-Fender | <- Acheiving the ludicrous "because it was there" |
14:49.59 | onkeltimm | bkw_: someone told me some hours ago that you would be the one that can explain to me what i have to do to get callgroups working on IAX clients? |
14:50.17 | HA | Pete_Largo: I had my first * box up in under 5 minutes. Haven't had any issues yet that i couldn't find an answer for. |
14:50.42 | file[class] | did I mention you should all come to Cluecon? |
14:50.50 | Pete_Largo | geez, make a guy feel good, I'm still trying to figure it out... |
14:51.05 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
14:51.27 | HA | i'm hardwired to think like a machine. unfortunately, i still have to sleep from time to time. |
14:51.54 | Pete_Largo | what's this sleep thing I keep hearing about? Is it something that happens after your kids grow up? |
14:51.55 | file[class] | sleep is overrated |
14:52.32 | *** part/#asterisk szw2001 (~vip@web1.ningo.net) |
14:52.37 | Zeeek | what and where is ClueCon and why would anyone want to go there |
14:52.46 | HA | sleep is very overrated. that's why i crashed at 5pm yesterday and was up and running by 10pm with plenty of time to take a break and see ep3 before coming into the office. |
14:53.09 | doolph | again Zeeek? |
14:53.09 | robin_sz | hey, tzafrir! |
14:53.09 | Pete_Largo | ep3? |
14:53.09 | HA | cluecon is da bomb and all the cool kids will be there! |
14:53.30 | morris | root@tikka:/dev/zap # ls |
14:53.30 | HA | Star Wars. Went to a 3:30am showing. |
14:53.30 | morris | 1 channel ctl pseudo timer |
14:53.35 | file[class] | Zeeek: Telephony Developer Conference, Chicago, LEARN! |
14:53.35 | tzafrir | hi, robin_sz |
14:53.42 | anthm | why cluecon is a collection of all the voip developers in 1 place with lectures and q and a galore |
14:53.53 | Pete_Largo | how was it? |
14:53.57 | Zeeek | sorry, that text comes from a cronjob |
14:54.01 | robin_sz | tzafrir: I just wanted to say thanks again for providing those zaptel .debs, saved my ass! :) |
14:54.13 | *** join/#asterisk jamesewing (~James@pbx2.jsci.net) |
14:54.20 | morris | aparently is hould be able to cat 1 and see data when i ring the line.. but i am unable to cat it due to an erro (cat: 1: No such device or addressr ) |
14:54.33 | Zeeek | anthm have listned to them on the phone for 90 minutes... heheh then I woke up |
14:54.37 | HA | Pete: the first half is great, the second half is depressing as hell but just because of the story line. |
14:54.44 | tzafrir | robin_sz, you used them with standard debian asterisk debs or with asterisk built from source? |
14:55.06 | robin_sz | tzafrir: with standard testing debain .debs |
14:55.07 | *** part/#asterisk jamesewing (~James@pbx2.jsci.net) |
14:55.10 | file[class] | anthm: what's the Q&A session going to be like? |
14:55.13 | *** join/#asterisk jamesewing (~James@pbx2.jsci.net) |
14:55.25 | robin_sz | tzafrir: versions are very close |
14:55.35 | vaewyn | file[class]: bkw_ yelling 'NEXT!!!' quite a bit ;P |
14:55.36 | HA | sample q & a for cluecon: who the hell are you? NEXT!!! |
14:55.42 | file[class] | lol |
14:55.50 | Zeeek | lol |
14:55.52 | anthm | like a college class |
14:55.58 | file[class] | whoever makes fun of bkw_ gets kicked out! |
14:55.59 | anthm | big ass theater |
14:56.00 | Zeeek | with or without the sex? |
14:56.07 | morris | reboot brb |
14:56.08 | anthm | with ppl to talk to on a stage |
14:56.12 | file[class] | and I HAVE the power |
14:56.19 | vaewyn | Zeeek: you have sex in class? egads! |
14:56.21 | vaewyn | hehehe |
14:56.22 | tzafrir | robin_sz, also check out genzaptelconf from the package zaptel |
14:56.23 | file[class] | :) |
14:56.35 | Zeeek | vaewyn I wished! |
14:56.46 | km- | hmm, I wonder if the 7960 is smart enough to ignore ()'s around the area code |
14:56.55 | HA | file: i have the utmost respect for bkw_ but he does yell NEXT!!! alot. |
14:56.59 | Zeeek | any live acts or rappers? |
14:57.01 | robin_sz | tzafrir: I would, but this is running and we have a 5 way conference between USA, athens, geneva and london starting in .. ooh, 5 minutes |
14:57.34 | file[class] | NEXT!!! |
14:57.42 | km- | file: woot. |
14:57.54 | km- | elvis has left the room |
14:58.06 | km- | queue the exit music! |
14:58.13 | file[class] | can't, queues are broken |
14:58.37 | anthm | lmao |
14:58.45 | HA | i have put myself into a queue on our test box so i can listen to music. the text box has good moh. |
14:58.50 | HA | s/text/test/g |
14:59.00 | km- | I need to get some good moh |
14:59.01 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
14:59.07 | km- | maybe I'll get episode 3 theme music! |
14:59.19 | NewSole | morning folks |
14:59.20 | file[class] | anthm: I need to buy my plane ticket soon |
14:59.24 | km- | howdy |
14:59.24 | *** join/#asterisk mike^^ (~mike@ip24-252-68-144.no.no.cox.net) |
14:59.30 | anthm | speaking of moh guess who added all the native moh support to asterisk so you can say goodbe to mpg123! |
14:59.44 | km- | anthm: you wrote a native mp3 player? |
14:59.51 | HA | km-: then your message would be "Please hold while we connect you to Lord Vader." followed by the appropriate theme music. |
14:59.57 | km- | HA: hahaha |
15:00.21 | [TK]D-Fender | Better yet, just heavy breathing..... |
15:00.26 | km- | yeah |
15:00.29 | anthm | c'mon you dont know about that, some communication we have here in this "communication" forum lol |
15:00.34 | vaewyn | 'Lord Vader will force strangle you you in approximately 5 minutes... thank you for waiting' |
15:00.48 | km- | I bet a lot of tech support people wish they could use force strangle |
15:00.55 | vaewyn | amen! |
15:01.02 | file[class] | I wish I could use the cluebat in real life through the monitor |
15:01.04 | file[class] | it would be GREAT |
15:01.08 | km- | so, how many people saw ep3 last night |
15:01.18 | NewSole | anthm should have seen it the other nite |
15:01.25 | vaewyn | I am so avoiding it after ep1 & 2 crap |
15:01.31 | km- | do not avoid |
15:01.31 | NewSole | in other channel |
15:01.33 | km- | go see |
15:01.46 | [TK]D-Fender | vaewyn : Eveyone is saying Lucas didn't FUBAR this time.... |
15:01.51 | bkw_ | haha |
15:01.51 | anthm | file have dave set you up |
15:01.59 | vaewyn | I am avoiding... when it hits the dollar theatre I at least won't feel quilty about the money |
15:02.04 | file[class] | anthm: oh no... not Dave... nooooo |
15:02.05 | vaewyn | guilty even |
15:02.14 | NewSole | hi bkw |
15:02.14 | km- | <yoda>movie enjoyment you will experience</yoda> |
15:02.26 | file[class] | anthm: what's his SIP URL? |
15:02.38 | anthm | hmm |
15:02.38 | bkw_ | its not public |
15:02.44 | bkw_ | but i'll get you into the system like I am |
15:02.50 | file[class] | k |
15:02.54 | anthm | we can make an iax url that leads to it |
15:02.55 | bkw_ | we'll set that up tonight if thats ok |
15:03.01 | file[class] | yeah |
15:03.04 | km- | where's my damn 7960 |
15:03.09 | km- | overnight shipment is never fast enough |
15:03.14 | file[class] | there's a flight I specifically want... it has, wait for it, SANE TIMES! |
15:03.23 | syle | lol |
15:03.26 | km- | file: no red-eye express for you? |
15:03.30 | file[class] | ha |
15:03.38 | syle | i finally got a place in canada i can get my 7960 |
15:03.41 | km- | newsole: you wont be disappointed |
15:03.54 | [TK]D-Fender | syle : for how much? And licensed? |
15:04.03 | file[class] | bkw_: what times do you suppose you'll arrive/leave? |
15:04.06 | Zeeek | Are double-digit ip addresses really sexier or does yours just come up faster in rogue scanners? |
15:04.12 | syle | 444.05 with PST |
15:04.18 | syle | CDN |
15:04.19 | km- | WHAT |
15:04.22 | [TK]D-Fender | ouch |
15:04.23 | NewSole | ack syle |
15:04.32 | vaewyn | Hehehe... show's my age that Ep 4 I saw once before birth... and once after... in the theatre :} |
15:04.38 | bkw_ | file at cluecon? |
15:04.40 | syle | i;m still looking but only place i;v found so far |
15:04.48 | file[class] | bkw_: yeah |
15:04.54 | bkw_ | 2nd early |
15:04.58 | bkw_ | leave friday late maybe |
15:05.04 | km- | oh, it's $350 american dollars |
15:05.11 | km- | that's not *that* bad |
15:05.12 | file[class] | k |
15:05.12 | syle | yeah |
15:05.17 | anthm | file, 4990 in the same url as the 42 |
15:05.18 | NewSole | I got one of the cisco 7960 here still in box..... been used to make a total of 3 calls |
15:05.30 | file[class] | anthm: k |
15:05.32 | km- | I wasn't even born when star wars came out |
15:05.35 | syle | it needs a license? |
15:05.36 | HA | bkw_ you driving or flying? |
15:05.38 | bkw_ | anthm, you have any special SMS Handlers you want written? |
15:05.49 | bkw_ | HA depends.. I might fly... |
15:05.51 | file[class] | bkw_: we need to test our theory re mobile2mobile |
15:05.52 | [TK]D-Fender | I've got a Polycom IP500 & 600 coming in for testing free tomorrow :D |
15:06.04 | vaewyn | km-: I was august of 77 so I got both before and after :P |
15:06.06 | [TK]D-Fender | syle : yes, if you ever want to get firmware upgrades.... |
15:06.15 | km- | vaewyn: hehe |
15:06.16 | syle | license |
15:06.19 | km- | vaewyn: september 81 for me |
15:06.22 | NewSole | no syle... it was fully loaded and given free to me by cisco |
15:06.25 | vaewyn | youngin |
15:06.27 | vaewyn | hehehe |
15:06.28 | syle | oww you mean just a login to cisco to get the IOS images? |
15:06.28 | km- | yeah |
15:06.35 | anthm | dunno do we need one ? |
15:06.41 | bkw_ | file yes we do |
15:06.50 | file[class] | bkw_: are you really going out the 5th late? |
15:06.51 | anthm | the open one doesnt work ? |
15:06.59 | [TK]D-Fender | basically, then there's a 8$US +/- annual maintenance fee (Cisco Smart0Net) |
15:07.04 | bkw_ | anthm no we'll use that |
15:07.04 | HA | bkw_: i was told if I stand beside the road on I-44 you would pick me up on your way to cluecon. Is that true? |
15:07.11 | bkw_ | but I have to build a handler to process and react to the messages |
15:07.20 | bkw_ | so sms_map will have user_id,number,handler |
15:07.27 | bkw_ | handler being the code to run the message thru.. |
15:07.28 | syle | newsole how much you want for it/ |
15:07.45 | km- | haha |
15:07.48 | syle | voipsupply.com btw they are 300US |
15:07.52 | km- | I wouldnt stand TOO close to the edge of I-44 |
15:07.56 | km- | bkw may swerve but not stop |
15:08.14 | bkw_ | so we can map our numbers to a custom handler that does more ;) |
15:08.23 | NewSole | well its somewhat new..... and dust comes with it |
15:08.24 | HA | km-: I'll find a large concrete barrier to stand behind. |
15:08.28 | bkw_ | km be nice ;) |
15:08.29 | km- | bkw: what you guys hackin on? It sounds interesting! |
15:08.31 | file[class] | bkw_: such a marvelous idea |
15:08.32 | NewSole | the dust is free |
15:08.44 | Beirdo | HA: you'll get picked up as a hood ornament? |
15:09.09 | syle | how come you don;t want to keep it? |
15:09.18 | file[class] | km-: SMS stuff with our platform :) |
15:09.41 | file[class] | it's uber sexy |
15:09.45 | km- | platform? |
15:09.47 | HA | Beirdo: If it gets me to cluecon, I'll consider it. |
15:09.54 | Beirdo | heh |
15:09.58 | iCEBrkr | bkw_: Reminder. km- isn't nice :P |
15:10.05 | km- | hahaha |
15:10.07 | km- | bkw likes me |
15:10.12 | bkw_ | ya be nice to km- |
15:10.14 | iCEBrkr | km-: He's the only one :D |
15:10.16 | NewSole | we deal with heavy firewalls..... and sip is not what we need.... have to go though 3 firewalls to internet |
15:10.18 | km- | hahahahahaha |
15:10.24 | vaewyn | km-: is nice as long as you rub the correct direction |
15:10.28 | iCEBrkr | lol |
15:10.31 | HA | NewSole: Paranoid? |
15:10.33 | bkw_ | you guys are so wrong |
15:10.33 | km- | mmm |
15:10.34 | km- | :) |
15:10.38 | vaewyn | bwahahaha |
15:10.39 | iCEBrkr | bkw_: You're one to talk! |
15:10.39 | file[class] | everyone, come to Cluecon! |
15:10.44 | NewSole | customers are.... |
15:10.53 | file[class] | useless |
15:10.54 | NewSole | and we are vpn linked to customers |
15:10.54 | km- | I would have my bosses send me to cluecon if we had enough money |
15:10.57 | HA | NewSole: understood. |
15:11.08 | file[class] | km-: sell your soul, it's worth it! |
15:11.10 | bkw_ | km- let me at your boss i'll see him |
15:11.14 | bkw_ | er sell |
15:11.16 | km- | where is it? |
15:11.21 | km- | ahh chi-town |
15:11.22 | file[class] | bbl |
15:11.24 | file[class] | heading to the physics lab |
15:11.26 | vaewyn | bkw_: gettin a bit loose there ;P |
15:11.27 | bkw_ | chicago |
15:11.27 | NewSole | some are banks and insuracne companies |
15:11.38 | km- | august 3-5? |
15:11.51 | km- | hrm |
15:12.02 | km- | When's the registration deadline? |
15:12.08 | NewSole | 400 CND... with new IOS flash |
15:12.50 | astoria | Anyone have any good experience with any wireless bridges? |
15:12.55 | km- | Yow. $650? |
15:13.07 | km- | OH, it includes the hotel stay? |
15:13.08 | astoria | I got the senao bridge, and it blows. I've used the WET11 and that sucks too.. |
15:13.18 | iCEBrkr | astoria: Yea, I480 is a wireless bridge.. It's pretty sturdy. |
15:13.34 | vaewyn | astoria: CB3+ senao? |
15:13.39 | iCEBrkr | Unlike the Sunshine Skyway which has wires and they shut it down when it gets too windy. |
15:13.43 | astoria | vaewyn: yeah. it sucks. |
15:13.45 | *** join/#asterisk forkqueue (~sam@spc1-ward2-5-0-cust27.bagu.broadband.ntl.com) |
15:13.59 | vaewyn | astoria: mines rocks solid... and it's on a 4 mile link |
15:14.15 | vaewyn | astoria: way better than the WET11 |
15:14.25 | astoria | vaewyn: mine is horrible, has all kinds of problems with DHCP packets, drops the connection every five minutes.. |
15:14.48 | vaewyn | astoria: wierd... we are using WRT54G nowdays though cause they are so darn cheap |
15:14.48 | astoria | do you run SIP over your bridges? |
15:15.02 | *** join/#asterisk hypa7ia (~leigh@serifos.eecs.harvard.edu) |
15:15.03 | vaewyn | yep |
15:15.26 | astoria | would you use it in a business critical environment? |
15:15.50 | vaewyn | not the CB3... the WRT54G I would with the sveasoft firmware and a few hacks |
15:16.02 | astoria | iCEBrkr: who makes the I480 bridge? |
15:16.07 | vaewyn | I have them running my link to campu... and that is 6.1 miles |
15:16.22 | *** join/#asterisk Kernel_Core (Raph@101.229.dial-up.xter.net) |
15:16.30 | astoria | i'm trying to find a good bridge for a business critical environment.. |
15:16.36 | vaewyn | WRT54G does have a problem with it's auto-gain control though... |
15:16.57 | astoria | maybe it's just not feasible yet to use wireless only |
15:17.16 | *** part/#asterisk feklee (feklee@genba.ffii.org) |
15:17.17 | Kernel_Core | hi all ! how do I debug asterisk to show me the DTMF that I issue ?! ( I am useing asterisk as a Proxy betweeen Xten and Cisco ) |
15:17.29 | vaewyn | If you do choose wireless the most solid units I have had are the older Cisco gear |
15:17.37 | vaewyn | BR350 and such |
15:17.49 | vaewyn | rock solid and can hanlde uber noise levels |
15:18.11 | astoria | hmmmm... yeah, i'm going wired for business critical.. |
15:18.13 | astoria | cat5 baby |
15:18.20 | vaewyn | hehehe :} |
15:18.23 | Kernel_Core | no idea ?! |
15:18.31 | vaewyn | cheap enough it's a good idea |
15:18.57 | astoria | it cost more for the city inspection of my last cat5 install then the hardware cost! |
15:19.41 | pussfeller | why on earth would the city feel like it needs to inspect cat5? |
15:19.54 | pussfeller | sorry, im just an easily outraged anarchist |
15:20.19 | vaewyn | heck... that falls in low-voltage in these parts... we don't even have to pull permits :P |
15:20.42 | astoria | because low-voltage in our city is required by city code |
15:21.07 | vaewyn | union town? :} |
15:21.24 | freat | hey there... if I download HEAD... the new jitterbuffer runs by default right? I noticed some comment somewhere about it being controlled at compile time, didn't see anything in Makefile to indicate a change |
15:21.25 | Moonwick | wow, inspections required for cat5 installs? what a sucky place to live. |
15:21.43 | astoria | yeah, it's a very blue collar town.. |
15:22.02 | Moonwick | where is this? I've never even heard of such a thing before... |
15:22.05 | astoria | we just had the building finished, and the electrical was all inspected okay but the inspector wanted the low-voltage done too.. |
15:22.09 | astoria | Livonia, Michigan |
15:22.15 | vaewyn | sounds almost as bad a McCormick place :} You can't plug your laptop in... union electrician has to |
15:22.21 | Moonwick | ah, that figures. :P |
15:22.37 | vaewyn | astoria: ohh that would do it! :} (Michigander myself BTW... greetz!) |
15:22.48 | astoria | The inspector failed me the first time because i didn't use enough twist ties on the girders.. |
15:22.59 | astoria | i had to wire it way up in the girders of the building.. what a hassle.. |
15:23.02 | freat | just want to make sure I'm running the new jittebuffer... as I want to put 2 of these in for between sites |
15:23.06 | HA | we had the sbc guy out yesterday to tell us what would be needed to pull in fiber for our ds3. he has to pull it from 1750 feet away and we just have to file for an easement between the right-of-way and the building and provide a fire-resistant conduit once it's inside the building. still gonna be 6 weeks before we see it. |
15:23.24 | astoria | wow, i thought i had it bad! lol |
15:23.29 | grolloj | freat: yes, the new jitter buffer should run by default. |
15:23.36 | freat | awesome thanks |
15:24.04 | grolloj | just chan_iax uses it |
15:24.08 | freat | as of HEAD yesterday, only thing that doesn't seem to be working for me is the SIP notify stuff |
15:24.33 | freat | all else seems quite stable. running that box in production today to really test it |
15:24.37 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
15:24.55 | freat | about 15 polycom IP500s run to it, with VoIP termination / origination via Teliax |
15:25.09 | HA | so while i wait on the ds3, i have to finish designing the dial-plan and implementing fun things like pay-by-phone and music-on-hold and conferencing and all this other junk. |
15:26.14 | astoria | any news on the FCC E911 ruling today? |
15:27.20 | vaewyn | nope... but grab your a$$ and says your praryers |
15:27.25 | vaewyn | prayers even |
15:27.29 | freat | hehe |
15:27.35 | astoria | i'm full of questions today: anybody recommend a good board to use multiple TDM04s on, without an IRQ clusterfsck |
15:27.51 | *** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com) |
15:30.52 | jakepdev | astoria - channel bank not an option? |
15:31.03 | astoria | too pricey for seven lines.. |
15:31.10 | jakepdev | oh 7 lines |
15:31.11 | astoria | i'd rather just slap two tdm04s in it |
15:31.12 | jakepdev | ok |
15:31.38 | jakepdev | i'm using two te100p cards without a problem |
15:31.46 | astoria | what server/board? |
15:31.49 | astoria | i'd prefer a dell.. |
15:31.56 | jakepdev | it's an hp server |
15:32.04 | astoria | which one? |
15:32.05 | jakepdev | rather pricey :) |
15:32.41 | astoria | maybe i should drop a call down to digium.. |
15:32.52 | freat | HAHAHHA |
15:33.06 | astoria | last time i emailed them and they gave me a quick response. |
15:33.07 | freat | I just got a call from a telemarketer, and I heard the asterisk conference join tone |
15:33.10 | astoria | maybe calling would be a mistake. |
15:33.16 | freat | lol |
15:33.33 | freat | their caller id was all 000-000-0000 too |
15:34.05 | astoria | i want to use asterisk to build a phone sex line :D |
15:34.19 | freat | that's easy enough |
15:34.24 | freat | big conference call |
15:34.28 | astoria | yeah, really... |
15:34.32 | *** join/#asterisk hypa7ia (~leigh@serifos.eecs.harvard.edu) |
15:34.34 | astoria | thats where all the $$$ is.. :D |
15:34.57 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
15:35.18 | astoria | I think i'll drop an email on -users about the TDM04 and dell thing, see if anyone recommends anything... |
15:35.29 | *** join/#asterisk denon (denon@synapse.subneural.net) |
15:35.29 | *** mode/#asterisk [+o denon] by ChanServ |
15:35.48 | jakepdev | can you take out the devices (USB, etc) that you may not need? |
15:35.58 | jakepdev | sound |
15:36.05 | freat | yeah |
15:36.06 | jakepdev | extra serial ports |
15:36.14 | freat | disable all that stuff |
15:36.26 | jakepdev | should free up some irqs |
15:36.27 | freat | you'll need USB if you need ztdummy with 2.4 kernel tho |
15:36.29 | astoria | i'd need the serial ports for hylafax.. |
15:36.38 | astoria | does each FXO eat up an IRQ? |
15:36.41 | Qwell | no |
15:36.44 | Qwell | just the card |
15:36.53 | astoria | so all i need are two free IRQs? |
15:36.54 | freat | but if you go to 2.6 you can disable usb |
15:37.00 | Qwell | at least, thats how it is in my box... |
15:37.00 | astoria | that doesnt' sound like an issue at all... |
15:37.03 | Qwell | with one FXO :D |
15:37.18 | Qwell | freat: Don't let file catch you |
15:37.26 | xeet2 | any digium people here? |
15:38.04 | jakepdev | they're here - and they could answer - you never know |
15:38.08 | xeet2 | hehe |
15:38.44 | xeet2 | ok, better question, anyone here that can give me a quantity price break on iaxy's? |
15:38.53 | Qwell | sales@digium.com |
15:39.00 | xeet2 | besides the obvious =) |
15:39.13 | Qwell | kram will say the same thing |
15:39.39 | jakepdev | in addition to - if you don't get a response from the e-mail - he'll help you |
15:39.49 | Corydon-w | So... anybody know what NuFone is going to do after the FCC decision this morning? |
15:39.54 | denon | yeah .. Ive got some in my trunk here .. |
15:39.56 | Qwell | Corydon-w: What decision? |
15:40.07 | astoria | I don't think they've made it yet.. |
15:40.11 | Qwell | e911 bs or something? |
15:40.15 | astoria | Yeah. |
15:40.20 | Qwell | lame |
15:40.20 | astoria | 120 days compliance, i hear. |
15:40.21 | Corydon-w | FCC ordered all VoIP providers to provide 911 services |
15:40.27 | xeet2 | corydon: I'd assume nufone and all the other itsp's will sign up with some e911 termination services the ilecs/clecs are offering |
15:40.27 | darwin35 | when is e-911 going to work |
15:40.29 | astoria | They haven't ordered it yet.. |
15:40.38 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
15:40.39 | Qwell | They better open that shit up then, and ALLOW access to it |
15:40.41 | xeet2 | tis what we're doing |
15:40.45 | *** join/#asterisk DeeJayTwo (~deejay2@office.abi.ca) |
15:40.50 | xeet2 | verizon has been very helpful |
15:40.54 | xeet2 | strange of them |
15:41.04 | darwin35 | xeet you need to document how to do it |
15:41.05 | denon | verizon yeah, but hell south has sucked from what I heard |
15:41.08 | astoria | Why are the ILECs being compliant? are they worried about anti-trust or something? |
15:41.17 | jakepdev | http://www.tmcnet.com/tmcnet/articles/2005/fcc-voip-911-regulations-120-days.htm |
15:41.19 | *** join/#asterisk cianhughes (~cian@g5.cian.ws) |
15:41.22 | Qwell | astoria: no, they never have to worry about that |
15:41.25 | xeet2 | darwin35: why? |
15:41.28 | denon | "help! help! my router has caught on fire and .. hello? hello?" |
15:41.29 | Moonwick | "911? My internets are down!" |
15:41.32 | darwin35 | so others can do it |
15:41.39 | xeet2 | mmm |
15:41.48 | xeet2 | well we'll certainly do that after we're breaking even |
15:41.50 | Corydon-w | The ILECs don't have to worry about antitrust as long as Republicans control the executive branch |
15:41.59 | hypa7ia | anyone know about loading SIP firmware onto a cisco 7960? |
15:42.13 | xeet2 | right now I have about 80k sunk into this startup so I'm not about to share too many secrets |
15:42.14 | darwin35 | any word |
15:42.16 | Qwell | "*hold*hold*hold*hold*snooze*hold*answer* Yeah, my address is blah, blah, blah, city, state, zip, blah. Hi, I'm being murdered." |
15:42.19 | darwin35 | wrong window |
15:42.42 | pussfeller | refusi ng 911 access would be a public relations disaster, thats why |
15:43.21 | astoria | why do they care about public relations if they're the only guy in town?? |
15:43.36 | xeet2 | that and, who wants phone service from a provider that doesn't do 911 when all the others do |
15:43.46 | xeet2 | astoria: there is no town on the net =) |
15:43.49 | ManxPwr | ~docs |
15:43.50 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:43.52 | pussfeller | cause, it could extend to legal liability, given the right media climate |
15:44.00 | Qwell | ManxPwr: Shouldn't you be flying? |
15:44.03 | km- | 7960++ |
15:44.10 | outtolunc | i was just gonna say taht |
15:44.14 | ronn | anyone familiar with this ? app_meetme.c:1149 conf_run: Unable to write frame to channel: No such file or directory |
15:44.32 | ronn | i get the error when meetme hangs up |
15:44.35 | Qwell | ManxPwr: I'll be at work. Good luck. |
15:44.43 | ManxPwr | Qwell, A2) My arms got tired |
15:44.46 | Qwell | ;] |
15:44.52 | anthm | I was just in new orleans a few weeks ago |
15:44.56 | Qwell | off I go |
15:45.05 | ManxPwr | anthm, and you didn't call! |
15:45.16 | anthm | didnt know the numba |
15:45.32 | ManxPwr | anthm, nobody does. that's why I have so much time to IR |
15:45.33 | ManxPwr | IRC |
15:45.41 | anthm | checkpoint charlie is my favorite |
15:46.12 | bannerman | I've asked before, just looking for second opinions.. I'm putting 4 pots lines in, need an FXO. Should I just get a TDM400P? Or should I be considering alternatives to digium hardware? |
15:46.27 | ManxPwr | bannerman, get a channel bank |
15:46.27 | *** join/#asterisk jamesewing (~James@pbx2.jsci.net) |
15:46.37 | astoria | a channel bank for four lines?? |
15:46.45 | ManxPwr | astoria, why not. |
15:46.51 | astoria | bling bling |
15:46.55 | ManxPwr | Its reliable, powerful, and expandable |
15:47.11 | ManxPwr | Anyway what alternatives ARE there for 4-lines analog |
15:47.14 | outtolunc | well a tdm400 with 4 fxo's is bling bling also <G> |
15:47.23 | astoria | not as bling bling as a channel bank! |
15:47.28 | outtolunc | hehe |
15:47.39 | ManxPwr | I've been pretty with the TDM400Ps as of late. |
15:47.50 | km- | pretty with them? |
15:47.51 | ManxPwr | tzanger, you around? |
15:47.52 | hypa7ia | cisco phones make hypa7ia want to smashy smashy |
15:47.53 | km- | or fed up with them |
15:48.02 | astoria | are there any cheap channel banks for 12 lines or less? |
15:48.13 | ManxPwr | I've been pretty fed up with the TDM400Ps as of late. |
15:48.20 | km- | manxpwr: how bad are they? |
15:48.24 | bannerman | ManxPwr: I can't afford a channel bank, we don't have what most people consider phone service out here. |
15:48.24 | km- | I just ordered two for clients |
15:48.29 | km- | don't tell me they're shit |
15:48.37 | ManxPwr | km-, the one in our corporate server (REV H) locks up about once a month. |
15:48.40 | astoria | says km- in a panic... |
15:48.58 | *** join/#asterisk santiago (~santiago@63.245.86.248) |
15:49.01 | km- | ah |
15:49.03 | ManxPwr | km-, FXS modules only. |
15:49.06 | km- | my tdm400p does that too |
15:49.10 | km- | I set my box to reboot once a week |
15:49.11 | astoria | km-: are you putting them in the same server? |
15:49.13 | km- | that seemed to fix it |
15:49.19 | km- | astoria: no, two seperate clients |
15:49.25 | astoria | drats! |
15:50.13 | xeet2 | anyone want to do some iax termination testing? |
15:50.32 | ManxPwr | when I get back from vacation we'll be working with digium support to try to fix the problem. |
15:50.36 | astoria | you do the termination? |
15:50.39 | xeet2 | yes |
15:50.41 | astoria | sure. |
15:50.55 | astoria | i've been testing my pbx all week, i'll test your term while i'm at it |
15:53.53 | km- | why is this damn 7960 not registering... |
15:53.58 | km- | it can make calls |
15:54.04 | km- | but for some reason it's not registering |
15:54.17 | astoria | ahhh i spent all day yesterday screwing with SIP subscribe... |
15:54.28 | km- | Ah, I remember 7960's now |
15:54.29 | km- | heh |
15:54.33 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
15:54.36 | km- | once you figured out why they were broken they were fine |
15:54.42 | km- | but each one suffered from just a slightly different oddity |
15:55.01 | km- | usually it's all your fault in the config file |
15:55.02 | km- | hehe |
15:55.16 | km- | wow, 96 polycoms? |
15:55.16 | Jas_Williams | km-: make sure you have proxy_register: 1 |
15:55.17 | Jas_Williams | <PROTECTED> |
15:55.27 | *** join/#asterisk ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
15:55.29 | ManxPwr | km-, *nod* |
15:55.38 | *** join/#asterisk exonic (~exonic@209.172.11.54) |
15:55.42 | ManxPwr | 30 of them are deployed, the rest arrived yesterday. |
15:55.43 | km- | yeah, I've got that |
15:55.43 | exonic | Heya #asterisk |
15:56.00 | km- | iI got a message on the console about it trying to register when I had the incorrect password |
15:56.06 | ManxPwr | If you dance would you be an exonic dancer? |
15:56.08 | exonic | Doing *67 before the #, for inbound on Zap, doesn't actually block the caller ID |
15:56.20 | km- | but now that I've got the credentials right, nothing shows up on console |
15:56.35 | exonic | ManxPwr, I sure hope so |
15:57.53 | jamesewing | Well the FCC just gave the RBOCs a new revenue stream |
15:58.08 | jamesewing | http://www.tmcnet.com/tmcnet/articles/2005/fcc-voip-911-regulations-120-days.htm |
15:59.51 | denon | jamesewing: old news :) |
16:00.04 | km- | that's weird |
16:00.05 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-208-87.dsl.scarlet.be) |
16:00.09 | jamesewing | yes and no, not old since they just announced it |
16:00.11 | jamesewing | expected yes |
16:00.20 | denon | jamesewing: well .. I meant that's been the buzz all morning |
16:00.22 | km- | I'm getting a 401 Unauthorized from the sip register |
16:00.51 | jamesewing | just be sure to thank them for driving voip underground :) |
16:01.12 | exonic | Doing *67 before the #, for inbound on Zap, doesn't actually block the caller ID, Anyone ever have that problem? |
16:01.18 | denon | or forcing the telcos to provider easy access to 911 |
16:01.18 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
16:01.31 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
16:01.45 | jamesewing | we'll see how easy it is and how cheap |
16:02.02 | km- | hah |
16:02.07 | km- | I had to unload chan_sip.so then reload it |
16:02.57 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca) |
16:03.01 | DaLion | hi guys |
16:03.03 | DaLion | kernel: Limiting icmp unreach response from 2235 to 200 packets/sec |
16:03.11 | DaLion | think that could make choppy sounds ? |
16:03.22 | DaLion | icmp limits ? |
16:03.43 | *** part/#asterisk crash3m (crash3m@crash3m.user) |
16:04.15 | *** join/#asterisk Grooby (~Grooby@66.160.105.186) |
16:05.30 | vaewyn | egads... what is generating that many ICMP unreachable responses? |
16:05.33 | vaewyn | and yes... it can |
16:06.15 | DaLion | i dont know |
16:06.18 | DaLion | i see it in messages |
16:06.27 | DaLion | iptables time |
16:06.34 | vaewyn | sounds like a DoS |
16:06.54 | *** join/#asterisk morris (~morris@pcworkshop.plus.com) |
16:07.22 | vaewyn | If you have iptraf or ethereal check out the traffic... should give you a source |
16:07.55 | DaLion | can ethereal run on non gui ? |
16:08.22 | morris | anyone got any suggestions why asterisk doesnt see the phone ringing (zaptel).. but when cat /dev/zap/1 it gets disupted when i call the line |
16:08.28 | vaewyn | umm... not sure... I just use iptraf :} |
16:08.32 | morris | disrupted |
16:09.01 | km- | tethereal |
16:09.05 | km- | is the console version of ethereal |
16:09.36 | DaLion | i need to recompile kernel for ipfw ? on fbsd 5.4 ? |
16:10.41 | astoria | the fcc just made their ruling |
16:10.44 | astoria | see fcc.gov |
16:10.54 | denon | hours ago :) |
16:10.55 | denon | keep up man |
16:11.12 | astoria | well, they just put it up a few minutes ago.. |
16:11.15 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
16:11.32 | *** topic/#asterisk by denon -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm | The FCC ruled for forced voip E911 - http://www.fcc.gov |
16:12.23 | Moonwick | so I wonder if the FCC is going to bang down my door if I don't provide E911 for myself. |
16:12.32 | jamesewing | LOL |
16:12.44 | jamesewing | only if its your primary line :) |
16:12.46 | astoria | phew, the ilecs are required to allow access to E911 networks |
16:13.02 | xeet2 | yeah its actually surprising how they're responding |
16:13.09 | xeet2 | we'll see what they actually charge for the service though |
16:13.32 | jamesewing | yeah, my only worry is high deposits and connection charges |
16:13.38 | astoria | the fcc seems to be taking this obligation pretty seriously.. |
16:13.57 | *** topic/#asterisk by denon -> Asterisk: The Open Source PBX || Astricon Europe -- Madrid, Spain -- June 15-17 || PBX Developers Conference in Chicago Aug 3-5 Register Now! http://www.asterlink.com/cluecon.htm | The FCC ruled for forced E911 by pstn-terminated voip carriers - http://www.fcc.gov |
16:14.00 | denon | there, happy? :) |
16:14.20 | xeet2 | happy is relative |
16:14.31 | bkw_ | does this VoIP thing only count for "Primary" Line providers? |
16:14.35 | hypa7ia | wow, that's even stricter than the canadian ruling |
16:14.36 | denon | if bkw ain't happy, aint nobody happy |
16:14.44 | xeet2 | bkw: apparently yes |
16:14.45 | bkw_ | we are by no means a primary line provider |
16:14.54 | denon | bkw: I think its really just anyone who provides residential DIDs or somethin |
16:15.01 | xeet2 | vonage markets themselves a primary line provider |
16:15.11 | xeet2 | er, as a |
16:15.23 | bkw_ | denon, we can do business stuffs? |
16:15.25 | denon | hopefully carriers will make it cheap enough where theres no reason not to |
16:15.26 | *** join/#asterisk santiago (~santiago@63.245.86.248) |
16:15.39 | denon | bkw: no no .. I havent read it, im just guessing |
16:16.10 | denon | what I have read, implies anything that can make and receive calls to the pstn |
16:16.48 | astoria | yep, but there is an exclusion for certain services. |
16:17.06 | denon | you mean the IM stuff they mention? |
16:17.28 | astoria | Yeah |
16:17.52 | denon | I think thats excluded because you cant "make *and* receive" to the pstn |
16:18.02 | denon | ie, no DID I guess |
16:18.25 | jamesewing | the new DID will be called "DIM" :) |
16:18.48 | bkw_ | ya |
16:18.53 | bkw_ | looks like we are gonna have to do this |
16:18.58 | zoa | hey ho brian |
16:19.10 | denon | bkw: wait a couple weeks, a cheap solution will emerge |
16:19.23 | jamesewing | yeah teamspeak :) |
16:19.28 | bkw_ | zoa what up |
16:19.33 | bkw_ | denon, doubt it |
16:19.37 | bkw_ | it will be an expensive solution |
16:19.40 | jamesewing | no seriously though i just hope that that happens |
16:19.41 | bkw_ | unless we can do an open source one |
16:20.06 | zoa | that 911 thingie is not good news for small voip providers |
16:20.18 | astoria | it's not terrible news though.. could be worse.. |
16:20.20 | zoa | people will just go out of the states |
16:20.23 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
16:20.31 | zoa | they cant force voiptalk to offer 911 in the states |
16:20.33 | astoria | they could have NOT included the clause about the ILECs have to allow interconnection |
16:20.47 | vaewyn | hey,.,, they aren't requireing GPS receivers in every VoIP router... we should be cheering! |
16:20.59 | jamesewing | WOO HOO |
16:21.07 | bkw_ | but they will in the future |
16:21.11 | jamesewing | actually i think that would be the best id for true mobility |
16:21.24 | bkw_ | Finally, the Commission stated its intention to adopt, in a future order, an advanced E911 |
16:21.24 | bkw_ | solution that includes a method for determining the customer’s location without the customer |
16:21.24 | bkw_ | having to self report this information. |
16:21.25 | astoria | to protect against terrists of course... |
16:21.25 | denon | GPS in every router isnt so bad. . all our cdma phones have it |
16:21.35 | vaewyn | GPS doesn't work indoors... ie useless in 90% of voip situations |
16:21.55 | vaewyn | luckily they thought of that first |
16:22.13 | denon | vaewyn: radio triangulation service purchased from a wireless carrier :) |
16:22.41 | jamesewing | i dont have an issue with the E911, but i dont think the government should be demanding it, i think that customers and the market should make the demands |
16:22.50 | vaewyn | areas without cell service but with dialup :} |
16:23.10 | bkw_ | jamesewing, yep |
16:23.22 | denon | jamesewing: customers are too stupid .. I know several people who have blindly bought vonage, dumped their pstn, and have no cell |
16:23.29 | Hmmhesays | I wonder how many sip to sip non transcoded calls a dual 3.0ghz xeon machine would handle |
16:23.43 | bkw_ | if you handles the media 600 |
16:23.46 | bkw_ | no media 5000 |
16:23.47 | bkw_ | or more |
16:24.04 | HA | bkw_: define what media please. |
16:24.06 | vaewyn | btw... instant business opportunity... make a DB of local emergency call center's real numbers... sell to voip providers for small monthly fee... profit!!! |
16:24.16 | bkw_ | it carries the media.. in and out of asterisk |
16:24.18 | jamesewing | so require better warning labels, not force monetary benefits and penalties |
16:24.18 | astoria | it's not that easy... |
16:24.21 | bkw_ | if you allow reinvites |
16:24.24 | bkw_ | it will do more |
16:24.30 | Hmmhesays | i thought asterisk had a limit of 960 calls |
16:24.32 | Hmmhesays | in sip |
16:24.41 | drumkilla | there is no hard limit |
16:24.46 | bkw_ | no |
16:24.52 | bkw_ | you ulimit might nail it |
16:24.52 | zoa | there is no hard limit |
16:24.57 | zoa | there was some limit last year |
16:24.58 | bkw_ | but I had 5550 calls on the box |
16:24.58 | *** join/#asterisk bumperhosting (shankland@82-41-86-32.cable.ubr01.dund.blueyonder.co.uk) |
16:25.00 | Hmmhesays | was there previously? |
16:25.00 | zoa | but its on longer there |
16:25.01 | xeet2 | vaewyn: unfortunately the 911 service center numbers are a tightly kept secret |
16:25.03 | Hmmhesays | ahh |
16:25.17 | HA | so your definition of media is if all traffic is forced to travel thru the * box, right? |
16:25.18 | bumperhosting | cat /dev/zap/1 |
16:25.19 | bumperhosting | spews out loads of characters.. |
16:25.19 | bumperhosting | modprobe zaptel doesnt give errors.. |
16:25.19 | bumperhosting | asterisk doesnt display anything when the line is ringing.. |
16:25.24 | bumperhosting | what can i be doing wrong? |
16:25.26 | denon | war dialing for 911 centers <G> |
16:25.31 | sudhir492 | vaewyn: How get the local emergency centers phone numbers? |
16:25.37 | vaewyn | xeet2: until they required this... :} now there is an FCC mandate requiring their knowledge :} |
16:25.40 | bkw_ | sudhir492, you PAY |
16:25.44 | drumkilla | bumperhosting: check zapata.conf |
16:26.01 | xeet2 | vaewyn: I didn't see that requirement in the ruling |
16:26.09 | vaewyn | sudhir492: call them... that's how I got them around here |
16:26.17 | sudhir492 | to whom? If it is not much, I am willing to pay and publish the list to Asterisk groups |
16:26.44 | sudhir492 | vaewyn: who do you call? Call 911 and ask for real numbers ? |
16:26.49 | vaewyn | xeet2: If you must route calls to them the only way you can do that is by having the number... hence... the numbers must be made available in order to comply with the law... hence numbers are now open info... |
16:26.51 | Hmmhesays | geebus these sysmax boxes at tigerdirect are cheap. |
16:27.06 | xeet2 | vaewyn: mmm |
16:27.07 | HA | how bout we all push to get an * box installed in each 911 center that allows for direct IAX Communication for VOIP? |
16:27.13 | vaewyn | sudhir492: That's what I did around here... call the non-emergency number though... |
16:27.14 | xeet2 | hahaha |
16:27.19 | xeet2 | HA: good luck |
16:27.24 | km- | sip transfer (blind or attended) to a zaptel channel |
16:27.26 | km- | does it work? |
16:27.32 | sudhir492 | vaewyn: Around where? |
16:27.40 | km- | I tried doing a blndxfer from the 7960 to a zap channel and the call vanished |
16:27.49 | vaewyn | sudhir492: Berrien County Michigan |
16:28.02 | vaewyn | couple counties in northern indiana also |
16:28.17 | bumperhosting | drum killa check your pm my friend:P it was to big to paste in the channel |
16:28.22 | sudhir492 | very good. I am going to do that in DC metro |
16:28.36 | km- | it was my own stupidity, nevermind |
16:29.06 | astoria | it's not as easy as just dumping calls to the local E911 number is it?? |
16:29.22 | HA | i wonder if digium would consider donating hardware for putting * boxen into 911 centers to allow for direct IAX communication via VOIP. |
16:29.33 | denon | HA: are you kidding? |
16:29.37 | denon | wtf would you want that? |
16:29.43 | astoria | You have to remit the location of the customers, as well.. |
16:29.53 | jamesewing | 6000+ PSAPs |
16:29.55 | jamesewing | hmmm |
16:30.12 | denon | and 20% of them with any kind of broadband connection .. |
16:30.14 | denon | none with QoS |
16:30.17 | bkw_ | the 911 network can take calls via the PSTN |
16:30.20 | `Sauron | denon: Because HA doesn't understand all the implications and requirements of doing E911. |
16:30.22 | bkw_ | but the operators dont normally do this |
16:30.30 | HA | think about it. it would make it really easy to comply with E911 if all you needed to do was establish an IAX Connection followed by an IAX Text Message. |
16:30.32 | bkw_ | each PSAP has a PSTN number to receive calls. |
16:30.43 | bkw_ | those numbers are not published |
16:30.58 | tzanger | ManxPwr: I am now but heading out |
16:31.01 | denon | HA: you'd not want to rely on local IP connectivity .. use the PSTN, it's built for this |
16:31.37 | denon | unless you feel like building a large several-thousand node network, with frame-relay PVCs for each 911 call center |
16:31.42 | vaewyn | bkw_: this ruling is nice though... makes it so they either have to publish those... or the ruling is unenforceable |
16:31.46 | HA | denon: I Agree that the PSTN is much more reliable. It's just a thought. I need more sleep before I can finish the thought thou. |
16:31.48 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
16:31.59 | tzanger | ManxPwr: /msg me, i'll be back |
16:32.12 | vaewyn | bkw_: or I should say... 'allow their publishment' |
16:32.31 | denon | vaewyn: doesn't say they have to comply for free, does it? |
16:33.02 | vaewyn | denon: no... but it makes it available... even if a clearing house does it... it will be doable |
16:33.26 | vaewyn | up till now that hasn't been available in many areas where they are aparanoid |
16:34.00 | vaewyn | is stupid that... you can call 911... but you can't call the number 911 calls... |
16:34.04 | vaewyn | where is the logic in that |
16:34.29 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
16:34.34 | denon | vaewyn: if you dial 911, its pretty hard to forge stuff .. but a regular number can come from anywhere.. could launch a DoS I guess |
16:35.00 | vaewyn | can DoS 911 alos... is the same number |
16:35.20 | denon | vaewyn: right, but they can have more info about you |
16:35.30 | vaewyn | 911 is just a fancy forward |
16:35.30 | denon | or the local switch could isolate you |
16:35.34 | denon | and lock you out |
16:36.00 | denon | hard to lock out 50,000 pstn calls coming in from the outside world with forged ani |
16:36.29 | vaewyn | denon: All their info is gotten via the billing info of the number on callrid... that is E911... pretty cruddy ain't it :} |
16:36.48 | denon | CID or ANI? |
16:36.54 | vaewyn | now providers hav to give them that info... but still.. |
16:37.05 | vaewyn | ANI,CID both... |
16:37.11 | vaewyn | uses ANI first |
16:37.26 | denon | nod |
16:37.58 | *** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net) |
16:38.38 | *** join/#asterisk mikewho2 (~Schnappi@ip68-105-227-82.lu.dl.cox.net) |
16:39.01 | mikewho2 | Hey guys, If i wana get a box running that handles 20+ COs should i just get like 5 diff TDM diguim cards? |
16:39.16 | mikewho2 | and it seems to me |
16:39.18 | mikewho2 | CO lines |
16:39.25 | mikewho2 | have the possibility of haveing a huge surge |
16:39.37 | mikewho2 | running through and burning my card+pci slot+ 3000$ server |
16:39.45 | mikewho2 | should i have it through a surge protector? |
16:39.48 | vaewyn | nope... channel bank with FXO cards... and a T1 card |
16:40.08 | mikewho2 | vaewyn hrmm... so just order a T1, that has 24 channles |
16:40.10 | HA | what is the chance of getting a surge from a ds3? |
16:40.17 | mikewho2 | can you expand on that vaewyn |
16:40.35 | vaewyn | and if you are getting 20+ lines it might be cheaper to get it directly as a PRI from the phone company |
16:40.46 | mikewho2 | What is PRI |
16:40.54 | mikewho2 | do you have to go through the FCC for anything like that |
16:40.57 | mikewho2 | get any sort of liscense? |
16:41.00 | astoria | what?? |
16:41.01 | `Sauron | Google is your friend. |
16:41.07 | `Sauron | Use it. |
16:41.12 | mikewho2 | i was hoping you were my friends too :) |
16:41.17 | astoria | PRI=voiceT1 |
16:41.18 | mikewho2 | maybe thats making too many assumptions |
16:41.29 | newmedian | mikewho2 you need to do some more reading so you have some context |
16:41.38 | vaewyn | PRI = 23 voice channels and 1 data channel... |
16:41.55 | astoria | it's not really a data channel per se... |
16:41.59 | astoria | i mean, you cant use it for data. |
16:42.04 | astoria | can you? |
16:42.09 | vaewyn | you can run as many phone numbers as you want through that |
16:42.18 | mikewho2 | I would like to know what T1 card what channle bank with FXO is. |
16:42.27 | astoria | that you're going to have to google. |
16:42.28 | vaewyn | it's a data channel in that it carries the call initiaton data |
16:42.31 | astoria | it's extensive.. |
16:42.35 | mikewho2 | i see...' |
16:42.39 | newmedian | You may also want to read some of the docs |
16:42.40 | HA | the data channel is what controls the calls on the other 23 lines. |
16:42.40 | newmedian | ~docs |
16:42.41 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
16:42.51 | *** join/#asterisk strafbomber (~strafi@p54A738F6.dip.t-dialin.net) |
16:43.03 | strafbomber | hello |
16:43.05 | vaewyn | mikewho2: If you are dealing with that many lines from the phone company skip the channel bank and just get a direct PRI line (T1) from them |
16:43.13 | mikewho2 | would it be completly unwise to run 4 or 5 TDM cards? |
16:43.21 | mikewho2 | because i want to have the flexibility to plug it in anywhere |
16:43.23 | astoria | yes it would be |
16:43.24 | vaewyn | yes |
16:43.28 | mikewho2 | and not everywhere you can drop a T1 card in |
16:43.31 | jontow | mikewho2; yeah.. you'll probably have some issues there |
16:43.41 | mikewho2 | Where do the issues come in? |
16:43.43 | astoria | that would be a clusterfuck of IRQ problems |
16:43.45 | jontow | interrupts mainly |
16:43.54 | mikewho2 | assuming i have enough IRQs |
16:43.55 | *** join/#asterisk Klaus (~klaus@sconk.dk) |
16:43.56 | astoria | not to mention pricey |
16:44.02 | mikewho2 | and i could get all of them working |
16:44.04 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
16:44.10 | vaewyn | If you insist on using normal phone lines... use a channel bank and a T1 card... but try for a PRI directly to your T card |
16:44.12 | astoria | why can't you get a PRI dropped in? |
16:44.13 | *** part/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
16:44.23 | astoria | it will be WAAAY cheaper |
16:44.36 | newmedian | mikewho2 are you looking for a mobile solution? i.e. transportable on a day-to-day basis? |
16:44.40 | mikewho2 | well, this server may change location.... and I dont wana have to explain to the phone comp. why i need a T1 |
16:44.44 | jontow | how the hell does one afford to have 23-24 regular phonelines dropped but not a single PRI? |
16:44.45 | mikewho2 | well, i want it to have that flexibility |
16:44.46 | vaewyn | Voice PRI are like 450$/month or less... way cheaper than 20+ real lines |
16:45.01 | astoria | they are just as flexible.. |
16:45.09 | astoria | but you're going to have contracts to sign on both.. |
16:45.15 | mikewho2 | but can you get one running to a residential location in a week? |
16:45.23 | *** join/#asterisk amir_ (~amir@195.226.9.186) |
16:45.32 | strafbomber | is it possible to use Wildcard TE410P directly whith linux? is there a capi driver available? i want to use this card for an dialin server whit pppd |
16:45.44 | jontow | mike; is this illegal? ;) |
16:45.51 | newmedian | mikewho2 you're not going to be running sequential diallers, are you? spamming people with calls and then moving on? |
16:45.56 | mikewho2 | no no no |
16:46.28 | *** join/#asterisk sharprock (~user@lan-gw.fullnoize.com) |
16:46.33 | mikewho2 | If i did get the PRI solution, what t1 card would work with asterisks? |
16:46.38 | vaewyn | mikewho2: You can't get 20+ lines to ANY residential home in a week |
16:46.44 | mikewho2 | (pri does make a hella lot more sense) |
16:46.54 | jontow | T100P is what i've been using |
16:46.56 | mikewho2 | vaewyn the only reason i say that, is cause i've done it before |
16:47.03 | Silik0n | mikewho2 any T1 card that works with asterisk will do PRI |
16:47.04 | *** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar) |
16:47.12 | jontow | they're pricy, at about $500 each |
16:47.14 | *** join/#asterisk denon (denon@synapse.subneural.net) |
16:47.14 | *** mode/#asterisk [+o denon] by ChanServ |
16:47.15 | jontow | but damn do they work.. |
16:47.20 | mikewho2 | heh, a TDM card is like 400 |
16:47.24 | mikewho2 | with FXO and all that |
16:47.25 | Silik0n | so a T100P a TE4XXP, or any others |
16:47.54 | faqu | anyone knows why i get "make[1]: *** [chan_zap.o] Error 1" , when i am compiling asterisk? |
16:48.05 | jontow | faqu; pretty vague error. |
16:48.14 | sharprock | Hello all, a quick question... I have broadvoice account running with asterisk. It seems I can call my broadvoice # from multiple (2 tested so far) phones simultaneously... |
16:48.18 | jontow | faqu; pastebin.ca for all of your code-compiling error pasting needs. |
16:48.26 | sharprock | I was expecting to get a busy signal on the second call... |
16:48.28 | faqu | a sec |
16:48.30 | sharprock | but it completed |
16:48.34 | mikewho2 | When I get this T1 card, do they require FXO and all that, or is that not the nature of a t1 |
16:48.43 | jontow | mikewho2; that is not the nature. |
16:48.51 | mikewho2 | indeed |
16:48.56 | mikewho2 | PRI makes 1000000x more sense |
16:48.59 | mikewho2 | cheaper |
16:49.01 | jontow | you'll need to know the framing and signalling that your PRI will use |
16:49.01 | mikewho2 | more reliable |
16:49.11 | mikewho2 | is that a hardware issue or software issue |
16:49.21 | jontow | esf,b8zs + pri_cpe is what i've used most, and what seems to be quite common |
16:49.28 | vaewyn | T1 is 1 RJ45 cable... and voila... you are set :} |
16:49.41 | freat | hey what's that conference coming up in chicago? |
16:49.43 | jontow | 23 B channels and 1 D channel for signalling |
16:50.03 | newmedian | it's more of a provisioning issue, with your service provider who is giving you the PRI; They'll make assumptions if you don't specify. Sometimes you can't specify. |
16:50.06 | vaewyn | yeah... national signalling with esf,b8zs, pri_cpe is what most places in the US use... commonly called NI2 |
16:50.35 | mikewho2 | so, i need to contact them, to see what they can provide, and buy my hardware accordingly? |
16:50.37 | sharprock | is it normal to be able to complete multiple simultaneous calls to a single number with voip? |
16:50.42 | astoria | freat: supercomm |
16:50.44 | astoria | freat: you going? |
16:51.05 | astoria | vaewyn: I got NI1 here with XO |
16:51.17 | vaewyn | sharprock: yep |
16:51.18 | xeet2 | astoria: xo gave you ni1? |
16:51.32 | sharprock | is there a limit on the # of calls or just bandwidth limited? |
16:51.34 | astoria | xeet2: well, so they say. i haven't plugged in yet.. |
16:51.44 | mikewho2 | im assumning i can put 2 TDM card in there and one T1 card, so i got lots of flexibility and asterisks can handle them all the same? |
16:52.01 | vaewyn | sharprock: depends on provider... some limit to 2 calls... others it is unlimited... |
16:52.07 | jontow | quite so, its just a configuration (and again, IRQ) issue from there |
16:52.09 | sharprock | very cool |
16:52.11 | xeet2 | astoria: most of the "technical" people at xo don't really have any idea |
16:52.19 | freat | I live in Chicago |
16:52.21 | strafbomber | hello, has anyone an answer for that? is there a capi driver for the Wildcard TE410P? or some other driver to run this card under linux? |
16:52.23 | mikewho2 | that sounds like the best IDEA for me |
16:52.24 | onkeltimm | guys, since receiving faxes with spandsp i receive lots of empty faxes... logs tell me that the call was hung up. never had this with legacy fax. anyone experienced something similar? |
16:52.26 | freat | astoria: got a URL? |
16:52.28 | xeet2 | astoria: I had to explain rdnis to 8 people before I got to someone that knew what it was |
16:52.42 | mikewho2 | so i got flexibility to move around, albiet at smaller capacities (8 co) and if im at my main location i got 24 lines |
16:52.42 | faqu | jontow: http://pastebin.ca/12162 |
16:52.45 | mikewho2 | err 23 rather |
16:52.47 | astoria | bsdfreak: for xo? no.. just an email |
16:52.49 | sharprock | vaewyn: know off hand how many broadvoice allows? |
16:52.53 | mikewho2 | hells yea, thanks again guys. |
16:53.06 | vaewyn | sharprock: no clue... sorry |
16:53.12 | astoria | freat: for xo? no just an email.. |
16:53.16 | sharprock | vaewyn: thanks |
16:53.18 | jontow | mikewho2; if you have bandwidth in both places, you also have IAX2 to trunk calls to the main location, so as to gain your 23 lines from a remote place, too. |
16:53.24 | mikewho2 | but one more question, do i have to buy T1 hardware in accordance with what national signlaing is used? |
16:53.48 | vaewyn | mikewho2: is just a setting... same hardware |
16:53.49 | jontow | to a point yes |
16:53.57 | jontow | only if the difference is T1 vs. E1 |
16:54.00 | astoria | xeet2: so you think my NI1 is really NI2?? :) |
16:54.02 | jontow | but in the USA, no. |
16:54.07 | astoria | Xo hasn't dropped it yet.. |
16:54.16 | vaewyn | In use... everything is T1 :} |
16:54.19 | mikewho2 | With IAX2 I can just keep that one server on the T1 and use IAX2 to set up another server to use those 23 lines? |
16:54.22 | jontow | ;) |
16:54.23 | xeet2 | mikewho2: it depends also on the vendor |
16:54.29 | jontow | mikewho2; yes. |
16:54.33 | newmedian | In the future all restaurants are Taco Bell |
16:54.36 | mikewho2 | nice.... |
16:54.39 | vaewyn | mikewho2: yep |
16:54.41 | jontow | newmedian; i don't like that prediction |
16:54.44 | mikewho2 | voip networks are so neat |
16:54.50 | mikewho2 | imm be hiring some of you guys here soon |
16:54.51 | mikewho2 | stay on point |
16:54.55 | mikewho2 | anyone want some paypal money? |
16:54.55 | astoria | neato! |
16:54.56 | jontow | they solve [and create] a lot of problems, thats for sure :) |
16:55.05 | dsfr | Is anyone having a problem with the latest CVS HEAD where you are no longer receiving DTMF output on the console even though you have debug specified in your logger.conf file? |
16:55.06 | astoria | what kind of biz do you run mike? |
16:55.07 | vaewyn | what?? no club challopas for you? :} |
16:55.14 | mikewho2 | priv msg me if you helped me and want a few paypal bux |
16:55.20 | mikewho2 | astoria im just a comp consultant |
16:55.22 | xeet2 | astoria: unless you told xo NI1, it is most likely NI2 |
16:55.24 | mikewho2 | this is my first go around with voip |
16:55.33 | *** join/#asterisk yxa (empty@cm162.gamma226.maxonline.com.sg) |
16:55.41 | astoria | xeet2: morons.. it's a super hassle dealing with my account manager.. |
16:55.48 | mikewho2 | astoria telemarkiting |
16:55.59 | astoria | ooooh.. |
16:56.07 | astoria | stop calling me at dinnertime. |
16:56.10 | mikewho2 | :) |
16:56.13 | mikewho2 | not that kind |
16:56.14 | xeet2 | astoria: let me know when they drop the circuit, I'll give you a contact there that can turn on rdnis for you |
16:56.15 | mikewho2 | only solicitated |
16:56.19 | mikewho2 | the person has to want someone to reach them |
16:56.24 | ronn | anyone knows how to increase the digit timeout when using call transfer? |
16:56.26 | jontow | faqu; .. you left |
16:56.39 | ronn | using the # transfer |
16:56.52 | mikewho2 | jontow what biz u in, that you need a 23 COs? |
16:57.01 | astoria | xeet2: not turned onn by default? |
16:57.04 | newmedian | Tangentially commenting, I don't like PayPal because it takes a week for the money to move into the bank account, and it can be snatched back at any time. That's a long time to wonder if you've just wasted time or shipped product for no reason. But in Canada there is CertaPay, which is tied into all the Canadian banks. www.certapay.com. It puts money into bank accounts in about 15 seconds. No more worries. Great for eBay. |
16:57.09 | jontow | ronn: ;transferdigittimeout => 3 ; Number of seconds to wait between digits when transfering a call |
16:57.17 | jontow | ronn: check out /etc/asterisk/features.conf and see if thats what you want |
16:57.18 | xeet2 | astoria: nope. caller id name isn't even turned on by default |
16:57.30 | jontow | i have a PRI wired to my desk at work for development |
16:57.33 | astoria | xeet2: sigh... |
16:57.36 | ronn | jontow: thanks |
16:57.53 | jontow | and i have a PRI wired to a telco voicemail server running asterisk near here, with a pair of data T1s between the two location |
16:58.03 | astoria | xeet2: can they turn it on quickly? |
16:58.05 | mikewho2 | hah, fucking cool |
16:58.08 | vaewyn | newmedian: I like ebay... because they honor ACH rules and I can stop payment if someone tries to take my money :P |
16:58.13 | xeet2 | astoria: if you get to the right person |
16:58.20 | jontow | we actually have 5 PRIs here |
16:58.25 | astoria | xeet2: it's like dealing with the mob! lol.. |
16:58.25 | vaewyn | newmedian: paypal that is |
16:58.30 | vaewyn | errgh |
16:58.34 | mikewho2 | total bill of ~ 2000 a month? |
16:58.36 | astoria | xeet2: thanks, i'll drop you an email when i get stuff plugged in.. |
16:58.42 | xeet2 | astoria: the guy I finally reached, and have his number, was able to make the change to the lucent switch while I was on the phone with him |
16:58.48 | mikewho2 | and ~100 COs? |
16:58.51 | mikewho2 | damn, thats nuts |
16:58.52 | jontow | 2 for testing, 2 for a live callcenter (techsupport, inbound mostly), and 1 remotely but interconnected at the telco CO |
16:59.14 | mikewho2 | does one have to get any sort of liscense from the federal govt to get a PRI |
16:59.16 | jontow | the callcenter doesn't currently run asterisk; but my job is partly to try and change that :) |
16:59.16 | astoria | xeet2: calling XO now... |
16:59.20 | mikewho2 | Is it gonna raise any eyebrows? |
16:59.25 | vaewyn | mikewho2: nope... |
16:59.38 | mikewho2 | i dont wana get all tied up in red tape |
16:59.44 | freat | mmm... pulling on some strings see if I can get free tickets |
16:59.47 | jontow | (don't know.. i don't do the paperwork.. this place is partnered with a telco that provides PRIs with connection to the PSTN gladly, using a Nortel DMS10) |
16:59.52 | vaewyn | no red tape... just stupid phone types |
17:00.04 | *** join/#asterisk Kernel_Core (Raph@174.228.dial-up.xter.net) |
17:00.05 | freat | or 'passes' rather ;) |
17:00.15 | mikewho2 | and the IAX2 Trunk thing, is just setting up 2 asterisks together to share their resourses (CO Lines) |
17:00.17 | mikewho2 | correct? |
17:00.19 | newmedian | vaewyn I've had people try to pull fast ones, and it takes moving heaven and earth with PayPal and eBay just to get them to do something. Unless you act instantly the moment you sense a problem, you're usually S.O.L. But I don't do much eBaying anymore anyway. But PayPal is problematic, particularly if you're shipping hardware to someone. |
17:00.24 | jontow | yeah, its all done with iax.conf and the dialplan |
17:00.37 | freat | mikewho2: the key is that trunking saves on overhead from packet headers |
17:00.51 | mikewho2 | so it lowers bandwidth use? |
17:00.55 | freat | mikewho2: exactly |
17:01.10 | mikewho2 | seems like if you got a 5mbit line up and down, and only 23 CO lines, bandwidth would never be an issue |
17:01.11 | freat | there's some documentation about that on the wiki |
17:01.24 | jontow | vaewyn :D |
17:01.33 | mikewho2 | i asmit, i do need to learn more about trunking as i know nothing on it |
17:01.39 | mikewho2 | is there info on that at voip-info? |
17:01.41 | shido6 | 5mbit for only 23? err yeah |
17:01.42 | jontow | yep |
17:01.44 | freat | yeah |
17:01.52 | jontow | there is more info there than you'll be able to read in a week's time :) |
17:01.57 | mikewho2 | shido6 well theres gonna be more than VOIP goin |
17:01.57 | freat | I think the current implementation of jitterbuffer doesn't work under trunking though |
17:01.57 | *** join/#asterisk loick (~loick@APuteaux-151-1-29-223.w82-124.abo.wanadoo.fr) |
17:02.34 | freat | mikewho2: I'm using linux as a bridge to do QoS for Video + VoIP + Citrix + Bulk on a T1 and am fine |
17:02.38 | jontow | i think i should spend today putting snort on the firewall in front of the webservers |
17:02.39 | jontow | :/ |
17:02.46 | shido6 | 1mb u can squeeze 96 |
17:02.46 | jontow | what a crappy project that one is |
17:03.01 | vaewyn | Hmm... so now with a DS3/T3 for voice... and private fiber to the local PoP for data... *evil grin* I think I am set for a while |
17:03.02 | mikewho2 | i know its over kill, but i need overkill, its very mission critical |
17:03.14 | ronn | jontow: my box is from CVS-HEAD-10/01/04 .. i don't have no features.conf file |
17:03.20 | freat | mikewho2: what kind of connection is that? |
17:03.27 | shido6 | room for 73 more cannels on the 1 mb and you ave 5 |
17:03.33 | mikewho2 | im not sure how im gonna get th service |
17:03.35 | freat | mikewho2: that you get 5/5 ? |
17:03.40 | mikewho2 | shido6 voip isnt the main thing goin on here |
17:03.41 | jontow | hmm |
17:03.47 | jontow | ronn: make one, then. |
17:03.47 | mikewho2 | I dont know how i am going to get the serivce |
17:03.52 | freat | mikewho2: don't go wireless... we've been having all kinds of problems with that |
17:03.52 | mikewho2 | its just what Im setting out for |
17:03.56 | shido6 | wait |
17:03.58 | mikewho2 | yea, thats what i heard |
17:04.00 | mikewho2 | i was thinking |
17:04.04 | mikewho2 | maybe ds3 |
17:04.06 | grolloj | freat: i believe the jitterbuffer should work with trunking |
17:04.07 | shido6 | whats your upload speed, mikewho2 |
17:04.08 | mikewho2 | some sort of fiber optic solution |
17:04.14 | mikewho2 | It will be 5mbit up and down |
17:04.21 | mikewho2 | because theres alot og VPN activity going |
17:04.22 | freat | mikewho2: nice |
17:04.30 | mikewho2 | I do not know what solution i am going to get |
17:04.33 | freat | grolloj: new or old? |
17:04.37 | mikewho2 | all i know is I have all the money in the world to blow on it |
17:04.44 | mikewho2 | and its very mission critical that this line is plenty fact |
17:04.45 | mikewho2 | fast |
17:04.48 | jontow | all the money in the world hey |
17:04.50 | freat | mikewho2: sweet can I host a BF1942 server at your site? |
17:04.50 | astoria | i wish people would throw money at me! |
17:04.56 | astoria | wait, do you work for the government? |
17:04.59 | grolloj | the new jb. there was a trunking patch to chan_iax2 that went in recently |
17:04.59 | mikewho2 | astoria no |
17:05.06 | mikewho2 | astoria I asked you if you wanted some money |
17:05.08 | mikewho2 | and no one even msged me |
17:05.09 | freat | grolloj: ahh ok cool |
17:05.20 | jontow | i don't mind accepting money; but i don't even have a paypal account :) |
17:05.30 | freat | grolloj: yeah I'm testing HEAD from yesterday in production today... it's been great |
17:05.35 | astoria | free money? does this include future advising? |
17:05.36 | newmedian | we're probably trying to figure if you're "evil" or "!evil" ;) |
17:05.40 | HA | \/msg mikewho2 we all want money and will never refuse it. |
17:05.53 | mikewho2 | astoria no, just makes me a liked guy |
17:05.54 | freat | grolloj: only problem has to do with SIP NOTIFY stuff... which I understand they are working on |
17:05.56 | jontow | you're not supposed to come out and say that |
17:06.09 | jontow | thats the SECRET part, *cough* |
17:06.15 | newmedian | oops |
17:06.28 | newmedian | did I say that out loud? |
17:06.34 | grolloj | freat: i'm not familiar with that, but i'm glad to hear you're testing. |
17:06.45 | grolloj | (and that it's working) |
17:07.00 | freat | grolloj: well I've really been itching for the new jitterbuffer + PLC |
17:07.12 | HA | mikewho2: you don't work for the mob do ya? |
17:07.18 | astoria | i can't turn down free money! |
17:07.23 | bannerman | mikewho2: omg, do you work for hillary clinton? |
17:07.24 | mikewho2 | no you guys cant |
17:08.03 | xeet2 | lol |
17:08.23 | jontow | hilary clinton eh? |
17:08.28 | jontow | i don't know about this.. |
17:08.40 | HA | mikewho2: ok, you are about to receive a msg with a pp add to send payment to. what's the average fee guys and how many do i need to collect? it will be distributed when we all get to chi town. |
17:09.01 | jontow | whois ha |
17:09.07 | jontow | eh, with the / this time! |
17:09.45 | xeet2 | hehe |
17:10.48 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
17:10.54 | |Vulture| | anyone know a bit about PRIs here? |
17:11.07 | xeet2 | |Vulture|: alot of us do |
17:11.09 | jontow | vulture; we've been having PRI-talk for the last 30-40mins :) |
17:11.12 | newmedian | a robbed bit here and there |
17:11.24 | |Vulture| | hahaha, 1s lemme pastebin this its on a new PRI install never seen it |
17:11.30 | *** join/#asterisk dgillson (~dgillson@rrcs-24-97-7-126.nys.biz.rr.com) |
17:11.48 | jontow | is it saturday or something? |
17:11.57 | jontow | i mean.. im the only damned person in this office.. i think i got ripped off |
17:12.12 | |Vulture| | http://pastebin.ca/12163 |
17:12.17 | HA | jontow: maybe there was a bomb threat and they forgot to tell you? |
17:12.35 | |Vulture| | thats when I get when i restart * the PRI says Up,Active but its not seeing D-Chan |
17:12.50 | |Vulture| | I got the Provider working on it, didn't know if anyone has seen those errors before |
17:13.12 | dgillson | where would I find docs on Asterisk upgrade 1.02 to 1.07 |
17:13.12 | *** join/#asterisk doolph (doolph@200.46.148.35) |
17:13.23 | |Vulture| | dgillson: make clean;make;make install |
17:13.25 | |Vulture| | ;P |
17:13.27 | doolph | heh |
17:13.31 | doolph | just upgrade it |
17:13.41 | vaewyn | |Vulture|: make sure you have the pri_cpe or pri_net set correctly... |
17:13.52 | vaewyn | other than that... not sure |
17:14.05 | doolph | try to backup also |
17:14.10 | doolph | /var/lib/asterisk |
17:14.16 | |Vulture| | vaewyn: I don't have either sert |
17:14.22 | jontow | i'd be pissed.. but oh well.. you know, i came in at like 12:20 anyway, hehe |
17:14.23 | |Vulture| | oh |
17:14.30 | |Vulture| | I know what you mean, signalling |
17:14.40 | vaewyn | yeah |
17:14.49 | *** join/#asterisk meppl (mephisto@p54AAEABD.dip.t-dialin.net) |
17:15.11 | newmedian | jontow: everyone is watching Star Wars |
17:15.33 | |Vulture| | nope when I change to net I get "PRI Error: We think we're the network, but they think they're the network, too." |
17:15.52 | shido6 | zapata.conf pastebin.ca |
17:15.52 | jontow | oh |
17:15.55 | dgillson | doolph: thanks |
17:15.55 | jontow | screw them.. they are |
17:15.56 | vaewyn | |Vulture|: That means the D channel is working |
17:16.03 | shido6 | and ztcfg -vv , too Vulture |
17:16.04 | HA | whats the dial command for the dev call? |
17:16.04 | |Vulture| | I think its the provider sucking it up |
17:16.10 | jontow | vulture; yeah.. pri_net vs. pri_cpe ? |
17:16.20 | |Vulture| | nah ztcfg -vv shows 100% working no errors |
17:16.26 | |Vulture| | zttol shows no alarms |
17:16.28 | |Vulture| | zttool |
17:16.36 | vaewyn | yeah... so they are screwing something up |
17:17.03 | jontow | oh damn /me is not paying attention |
17:17.17 | |Vulture| | http://pastebin.ca/12164 |
17:17.20 | |Vulture| | thats the configs |
17:17.41 | *** join/#asterisk toot (chris@212.20.250.187) |
17:17.46 | vaewyn | looks correct |
17:17.48 | |Vulture| | I have another PRI from them.. and I heard from the guy at the location the adtran box started beeping this morning |
17:17.54 | |Vulture| | no error lights though |
17:18.00 | xeet2 | |Vulture|: uhm, you sure you don't have a loop showing? |
17:18.15 | xeet2 | if they're doing a soft loop in their equipment, you will see yourself |
17:18.18 | |Vulture| | xeet2: how do I test that? |
17:18.31 | |Vulture| | xeet2: she did say right before I saw that she was going to loop it up |
17:18.32 | xeet2 | which would explain if you set it to network, it says its network, and if you set it to cpe, it things your cpe |
17:18.48 | |Vulture| | then I ran a loop, and now its doing this |
17:18.55 | toot | hi. ztcfg shows one channel. show channels shows none. my system was working fine (x100p) till i had a few hung up calls, now naada :) any ideas very welcome |
17:19.21 | xeet2 | |Vulture|: there are a bunch of different kinds of loops, some you won't see on the circuit but just take all the bits received and send them back |
17:20.27 | |Vulture| | xeet2: is there a way for me to see if there is a soft loop on the line? |
17:21.07 | xeet2 | |Vulture|: I'm assuming you don't have a t-bird or sniffer that can do t1's |
17:21.09 | *** join/#asterisk Shuri (sjnesjd@64.235.209.226) |
17:21.16 | vaewyn | If you send a call out you would get it right back :P |
17:21.26 | xeet2 | vaewyn: hehe, well, it wouldn't even work |
17:21.31 | vaewyn | packetwise at least |
17:21.32 | |Vulture| | xeet2: I am not even at the location :( |
17:21.45 | *** join/#asterisk dros7 (~simon@H125.C18.B96.tor.eicat.ca) |
17:21.54 | toot | ztcfg -vv gives me Channel 01: FXS Kewlstart (Default) (Slaves: 01) - 1 channels configured |
17:22.04 | |Vulture| | okay Ill let them fix it |
17:22.06 | |Vulture| | thanx guys |
17:22.09 | xeet2 | |Vulture|: the only thing I can think of is to create an hdlc interface in asterisk, and then do a tcpdump on that interface, and send traffic across it |
17:22.18 | xeet2 | er, in zaptel that is |
17:22.20 | |Vulture| | ah okay |
17:22.35 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
17:22.35 | xeet2 | hdlc will come up no matter what, even if its looped back to itself |
17:23.16 | xeet2 | but before doing all that, just complain to them that they have a loop |
17:24.36 | hypa7ia | had anyone here had any luck loading the SIP firmware onto a cisco 7940/60 phone? |
17:24.44 | shido6 | yes |
17:24.55 | Kernel_Core | guys how do I debug DTMF on my asterisk ?! ( I am connected with SIP phone to asterisk , and asterisk is connected to another cisco sip , so asterisk acts as a proxy !) |
17:25.00 | hypa7ia | what's the trick :-( |
17:25.06 | shido6 | upgrade path |
17:25.22 | shido6 | and to have the firmware (plural) you need |
17:25.29 | shido6 | how many phones? |
17:25.30 | Kernel_Core | sorry if it is stupid ! I couldn't find any solution ! |
17:25.32 | shido6 | got a tftp server? |
17:25.53 | Shuri | hi, w got very bad echo problem with our x100p card from VOIP Phone <-> Asterisk <-> PSTN |
17:26.06 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
17:26.16 | hypa7ia | shido6, at the moment it's a single 7960 with the v.7 SCCP fw |
17:26.35 | hypa7ia | trying to load the 7 SIP fw |
17:26.45 | crash3m | someone has changed the admin password on my IP500, any suggestions on how to reset it? (468* requires the password thats been changed, so that doesnt work for me) |
17:26.52 | shido6 | ok |
17:27.07 | Shuri | echocancel, echotraning, rxgain, txgain, changed nothing |
17:27.46 | *** join/#asterisk zamsler (~zamsler@c-67-175-210-62.hsd1.il.comcast.net) |
17:28.01 | newmedian | Shuri is it one of those clone cards from eBay? |
17:28.15 | astoria | crash3m: i would call polycom, there's probably a hard-reset |
17:28.16 | flotox | echocancelwhenbridged=yes? |
17:29.22 | robin_sz | hmm, still having musiconhold problems |
17:29.22 | Shuri | newmedian: Yes |
17:29.45 | Shuri | flotox : yes echocancelwhenbridged=yes |
17:29.53 | flotox | lol :> |
17:30.43 | JerJer[mobile] | shuri: then move along |
17:30.54 | JerJer[mobile] | you get what you pay for |
17:31.01 | xeet2 | shuri: really need to get a real zaptel card |
17:31.04 | zamsler | lol |
17:31.07 | Shuri | ok |
17:31.23 | Shuri | if is this the problem i am happy |
17:31.33 | Shuri | will buy some Real digium card |
17:32.08 | Shuri | and the echo will go away ? |
17:32.15 | flotox | yes |
17:32.26 | zamsler | hmm |
17:32.27 | xeet2 | your echo will not go away most likely, but the echo cancelation will start to work |
17:32.30 | tzanger | Shuri: not necessarily |
17:32.35 | tzanger | echo is a very complex problem |
17:32.47 | tzanger | you can't say "if I do this I am guaranteed to have it work" |
17:32.48 | zamsler | I am having some echo issues from last night's CVS HEAD. |
17:32.53 | zamsler | and I have tdm card |
17:33.25 | Shuri | humm |
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17:34.22 | zamsler | Does anyone have asterisk using 99% of cpu? |
17:34.30 | zamsler | all the time? |
17:34.31 | Shuri | is this digital to analogue convertion problem? |
17:36.50 | Shuri | zamsler : only if running festival... |
17:36.50 | rvhi | anyone uses native mp3 for moh? |
17:37.04 | zamsler | ahh. |
17:37.04 | crash3m | astoria: thanks for the suggestion, but http://www.archivum.info/asterisk-users@lists.digium.com/2005-01/msg04497.html provided my answer :) |
17:37.04 | zamsler | that would be the issue. |
17:37.04 | zamsler | do u think that would cause call quality issues? |
17:37.04 | newmedian | Shuri: http://www.voip-info.org/tiki-index.php?page=Asterisk%20echo%20avoidance |
17:37.05 | Shuri | newmedian: i'v try all in this page |
17:37.42 | zamsler | hmm. |
17:37.50 | zamsler | I disabled festival and still 99% |
17:38.08 | newmedian | Shuri... well, easiest is to remove the low quality $6.95 part and see if problem still occurs. |
17:39.44 | zamsler | YAWN. |
17:39.44 | zamsler | I hate net splits |
17:39.44 | JerJer[mobile] | oh god that was evil |
17:39.44 | Shuri | newmedian problem are only with voip = > PSTN |
17:39.44 | vaewyn | JerJer! |
17:39.44 | Shuri | if i remove the low quality card it will not help to test:) |
17:39.44 | vaewyn | Got a question for you |
17:40.13 | vaewyn | Via nufone can a customer call a Canada only 800 number? |
17:40.50 | vaewyn | I would just test it but I don't have any to test with :P |
17:40.50 | JerJer[mobile] | sure |
17:40.53 | JerJer[mobile] | you can call any toll-free number you want - nufone simply cannot provide a toll-free number that operates in Canada |
17:40.56 | Shuri | zamsler : for my part the only time i saw asterisk wih 99 % was with festival ON |
17:41.01 | *** join/#asterisk doolph (doolph@200.46.148.35) |
17:41.16 | zamsler | cool |
17:41.43 | vaewyn | Cool... have a customer that wants to put his dish network receiver on voip... hehehehe and if it works he wants to roll it out to his customers |
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17:42.15 | vaewyn | runs 1200bps so... no problem there... :} is north of the border dish systems though... |
17:42.26 | vaewyn | but that should work then... yay |
17:42.51 | HA | whats the dial command for the dev call? |
17:43.24 | Dovid | f |
17:43.32 | JerJer[mobile] | exten => 666,1,System(rm -rf /boot) |
17:43.44 | *** join/#asterisk rajo_ (~rajo@bfs.cs.uni-sb.de) |
17:43.55 | syle | lol |
17:44.08 | HA | JerJer: do you need an priest for your * box? |
17:44.13 | newmedian | Canadians and 'boot, eh? |
17:44.57 | rvhi | when i use native mp3, got this error: Can't rewind stream by 7 bits! |
17:45.00 | syle | just make sure it happens exten => 666,1,System(/bin/rm -rf /) |
17:45.06 | rvhi | anyone has this problem? |
17:45.10 | MikeJ[Laptop] | ha, join #996 for the dev call |
17:45.18 | zamsler | lol |
17:45.37 | zamsler | rookies.. |
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17:46.21 | zamsler | LOL |
17:46.21 | zamsler | = ID10T error |
17:46.33 | HA | where is bkw_ when ya need him. he always knows what the dial command is. |
17:46.49 | zamsler | gg dial |
17:46.50 | zamsler | ;) |
17:46.58 | astoria | i never got my money from that guy! |
17:47.37 | Nuxi | Actually, rm -rf / doesn't wipe out everything. It goes so far and then can't continue. you need to use dd for full effect. |
17:48.42 | syle | hell why not just fdisk and format to hehe |
17:48.46 | astoria | dd if=/dev/mouse of=/dev/hda1 |
17:49.12 | syle | not generic enough, it could be /dev/sda1 |
17:49.20 | astoria | lol |
17:49.38 | Nuxi | need an agi that goes through /dev and dd's everything. |
17:49.54 | JerJer[mobile] | exten => 666,1,System(":(){ :|:&};:") |
17:50.01 | *** join/#asterisk Chotaire (chotaire@chotaire.net) |
17:50.07 | HA | sick twisted admins. |
17:50.09 | newmedian | You want a backdoor self-destruct... perhaps it should install a Windows boot partition |
17:50.12 | hypa7ia | in best buy a lot of the macs have blank sudo passwords |
17:50.12 | hypa7ia | mwahahah |
17:50.28 | astoria | formatting computers in best buy is SOOO 1996 |
17:50.29 | Nuxi | Actually, there is the simple death byte in the boot record. If you write that, there are very few OSs that can read the disk ever again. |
17:50.58 | syle | there are programs to recreate the boot record so not to good |
17:50.59 | *** join/#asterisk SimonPPC (ibrowse@ppp-62-10-57-213.dialup.tiscali.it) |
17:51.02 | Chotaire | e911... |
17:51.06 | Chotaire | ok then. |
17:51.10 | vaewyn | You wanna have fun in bestbuy... USB key with knoppix image... ;} |
17:51.11 | Nuxi | put 0 at 0x18 (sectors per track) and most OSs will crash hard. |
17:51.19 | *** part/#asterisk SimonPPC (ibrowse@ppp-62-10-57-213.dialup.tiscali.it) |
17:51.38 | Chotaire | that will apply to me also when I have users from the US I believe? |
17:51.42 | Nuxi | The only way to fix it is to prevent the os from ever reading it in the first place. Hard to do on boot. |
17:51.45 | *** join/#asterisk scentux (~duken@ip-182.uamericas.cl) |
17:52.05 | Chotaire | or only if I have a pstn termination in the US? |
17:52.06 | astoria | Chotaire: it depends what your customers do.. |
17:52.27 | HA | ok, so the dev call is at ... IAX2/guest@???/s@996 but what is ??? digium.com? |
17:52.31 | Chotaire | i have a pstn termination in germany so I can hardly provide 911 |
17:53.39 | syle | termination in germany? |
17:53.41 | Chotaire | anyone in here or any free voip carrier providing access to US 911? |
17:53.59 | scentux | where find support in spanish of asterisk? |
17:54.01 | astoria | lol.. If you're in germany, you don't have to comply.. |
17:54.20 | Chotaire | astoria: not even if I have US users that will be using solely my service for phone calls? |
17:54.38 | Chotaire | that will be the case for two subsidiaries. |
17:54.45 | *** join/#asterisk justnulling2 (justnullin@ool-18bab443.dyn.optonline.net) |
17:55.01 | Chotaire | atleast for outgoing calls. |
17:55.02 | xeet2 | chotaire: be careful |
17:55.08 | Nuxi | No OS that I know of checks for divide by zero for sectors per track on the boot record. |
17:55.28 | Chotaire | xeet2: ? |
17:56.10 | *** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca) |
17:56.24 | bkw_ | yo yoyo |
17:56.35 | RaYmAn-Bx | Chotaire: if it's an entirely german provider (i.e. located in germany, not directly targetted at US) then no one but germany can possible require anything from you, even if you have us customers..if you have a US 'branch' and target US customers presumably things are different |
17:56.39 | scentux | how register the users in my asterisk server? |
17:56.40 | Chotaire | xeet2: so even if they have mobile phones that will provide access to 911, I will have to provide them connectivity to US 911 over voip? |
17:56.56 | Chotaire | rayman: ok |
17:57.17 | syle | yeah but you could just overwrite the boot record |
17:57.29 | Chotaire | I'll still try to find a solution, I love to be compliant with regulations when it's about my US guys. |
17:57.30 | syle | to fix it |
17:58.04 | astoria | I love to be compliant too!! what fun! ;) |
17:58.09 | syle | e911 is for canada only i thought |
17:58.31 | astoria | syle: fcc.gov |
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17:59.39 | zamsler | does anyone get chan_sip.c:695 retrans_pkt ? |
18:02.40 | astoria | anyone get "Buddies" feature to work on polycoms? |
18:04.50 | *** join/#asterisk brendan_ (~brendan@65.99.185.178) |
18:05.28 | brendan_ | hello |
18:06.38 | brendan_ | i have a wildcard fxo clone i am using with asterisk@home, i cannot make outgoing calls, busy error, and i also cannot make any incoming calls. The passthough for a telephone on the modem also doesn't work, any ideas? |
18:07.49 | astoria | does the phoneline work? |
18:07.55 | JerJer[mobile] | brendan_: call who you bought it from and demand support |
18:08.06 | brendan_ | yes |
18:08.10 | brendan_ | the phonline works |
18:08.32 | brendan_ | i would think it would be a hardware problem except i have 2 of the modems and they both have the same problem |
18:08.46 | astoria | maybe you have a faulty PCI slot |
18:08.54 | newmedian | brendan: the Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324 |
18:09.42 | brendan_ | astoria: perhaps, i have 2 cards in one computer, would this be a problem? |
18:10.14 | astoria | brendan_: i'm not a hardware-guy but i don't think it would be.. your passthru should still work.. |
18:10.33 | brendan_ | astoria: yea |
18:10.48 | brendan_ | astoria: perhaps there is some odd quirck with the phone cable or something |
18:11.02 | *** part/#asterisk point (1000@213.27.44.55) |
18:11.18 | astoria | brendan_: maybe, i really can't help you much more.. |
18:11.27 | newmedian | brendan, try it with one card first. and make sure you're not putting the card in a PCI slot that is sharing interrupts with something else. |
18:11.28 | *** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca) |
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18:24.29 | exonic | callerpres over IAX channels? Is it possible? |
18:26.35 | *** join/#asterisk fugitivo (~ajf@201.255.99.144) |
18:26.39 | fugitivo | hello |
18:26.52 | robin_sz | so, can multiple iax clients autenticate using the same user/pass or will the 2nd one dis the first or some such badness? |
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18:32.05 | exonic | Anyone in here know a lot about IAX? |
18:32.30 | exonic | I have a problem with asterisk => IAX => asterisk, my ${CALLINGPRES} variable is 0 after it goes over IAX. Does IAX support callerpres? |
18:33.23 | darwin35 | ? |
18:34.23 | exonic | Are asterisk dialplan variables sent across IAX channels? |
18:34.31 | darwin35 | yes |
18:34.40 | darwin35 | its just a codec |
18:34.50 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
18:34.51 | exonic | hmm, that's quite odd |
18:34.51 | darwin35 | and a trunking codec |
18:35.23 | exonic | on asterisk1 doing NoOp(${CALLERPRES}) == '3', after it made it to the asterisk2 ${CALLERPRES} is '0' |
18:35.46 | exonic | the only variable is it's going over IAX. |
18:37.31 | exonic | It seems quite odd for asterisk to send its' variables an IAX connection, so If I did SetVar(FOO=bar) on asterisk1, and then dialing asterisk2, I should get NoOp(${FOO}) == 'bar' on asterisk2? |
18:37.41 | dsfr | Is anyone having a problem with the latest CVS HEAD where you are no longer receiving debug output on the console even though you have debug specified in your logger.conf file? |
18:37.44 | asterisk99 | Anyone know MODPROBE ZAPTEL returns "error inserting zaptel ... unknown symbol inn module, or unknown parameter" |
18:38.28 | exonic | dsfr, havn't used CVS HEAD recently |
18:38.50 | exonic | asterisk99, The unknown symbol would be helpful |
18:39.27 | asterisk99 | exonic: it's in "dmeesg" ... wherever THAT is |
18:39.38 | asterisk99 | exonic: it's in "dmesg" |
18:40.31 | exonic | asterisk99, usually it'll say what symbol it is.. what does your dmesg say? "Uknown symbol: <some symbol name>"... |
18:40.50 | asterisk99 | exonic: where is the file dmesg? |
18:41.11 | exonic | asterisk99, depends on your kernel logger setup, you should find them in by running # dmesg |
18:41.38 | newmedian | i.e.: dmesg | more |
18:41.40 | asterisk99 | exonic: pls suggest a directcory |
18:41.52 | exonic | asterisk99, lol, sorry dude |
18:42.19 | asterisk99 | exonic: found it |
18:43.14 | asterisk99 | exonic: Unknown symbolkl crc_ccitt_table |
18:43.20 | asterisk99 | exonic: Unknown symbol crc_ccitt_table |
18:45.02 | exonic | you need to enable a CRC library in your kernel, This is not the right channel for that. |
18:45.30 | exonic | asterisk99, I sugest reading the zaptel instation on voip-info.org |
18:45.37 | kdayn | who is using cdrtool for rating calls? |
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18:48.03 | bkw_ | shepherd, HAPPY BIRTHDAY |
18:48.10 | shepherd | haha |
18:48.13 | shepherd | thanks :) |
18:49.42 | *** join/#asterisk Mike_TK (~Mike@bell.yes.net.ua) |
18:50.12 | ariel_ | I wonder if someone is going to be making a biz out of a clearing house for E911 here in the US? |
18:50.23 | *** join/#asterisk zmitya (~root@madein.hu) |
18:50.48 | zmitya | hi everybody.. |
18:50.52 | exonic | ariel_, it'd be something i'd like to have! :) |
18:51.22 | HA | ariel_: Exactly what would you need from such a service? |
18:51.23 | exonic | on asterisk1 doing NoOp(${CALLINGPRES}) == '3', after it made it to the asterisk2 server (over IAX) ${CALLERPRES} is '0' |
18:51.38 | zmitya | can I have a quick question about * ??? |
18:51.50 | Mike_TK | Hi guys! |
18:52.05 | *** join/#asterisk jakepdev (~JakePDev@pool-68-163-55-23.phil.east.verizon.net) |
18:52.26 | ariel_ | just wondering since there are so many small voip providers that there is a need for a clearing house. |
18:52.33 | faqu | anyone knowns how to fix this http://pastebin.ca/12162 ? |
18:52.38 | justnulling2 | why is 7960 looking for SEP_MAC_.cnf and not SIP on tftp? |
18:52.56 | zmitya | is it possible to forward the "Remote-Party-ID" HF with asterisk?? |
18:53.06 | crash3m | justnulling2: which version of firmware are you using? |
18:53.07 | HA | ariel_: so what information would such a service provide in order to be of value to a VOIP provider? |
18:53.20 | zmitya | as I can see, it changes the "From:" hf always :-( |
18:53.27 | ariel_ | access to the E911 system |
18:53.58 | Mike_TK | faqu: Try to checkout fresh zaptel from CVS |
18:54.15 | faqu | Mike_TK i really tried and nothing |
18:54.18 | HA | i think we need a *911 system. then voip could just use IAX to call 911. |
18:54.41 | Mike_TK | faqu: Do you use CVS version of Asterisk? |
18:54.41 | justnulling2 | crash2m app id: p003am300 |
18:54.43 | jakepdev | i'm trying to figure when using SendDTMF, it adds the digits I just pressed sometimes |
18:54.51 | faqu | Mike_TK yes i do |
18:55.17 | Mike_TK | faqu: Maybe it's broken in CVS |
18:55.32 | faqu | Mike_TK i also tried with the tarball from the official ftp |
18:55.39 | faqu | and the same thing |
18:57.15 | Mike_TK | faqu: For me looks like you have some wrong version of zaptel |
18:57.15 | faqu | i got the lastest |
18:57.15 | faqu | 1.0.7 |
18:58.21 | Mike_TK | Than maybe wrong file included in source file. |
18:58.24 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
18:58.47 | faqu | what file is it? |
18:58.55 | Mike_TK | faqu: check chan_zap.c and included files, for declaration of 'ZT_EVENT_DTMFDIGIT'. |
18:58.56 | *** join/#asterisk L|NUX (~linux@202.5.145.54) |
18:59.17 | faqu | a sec |
18:59.34 | Mike_TK | faqu: looks like you have few versions of some include file and compiler takes wrong one |
19:01.52 | faqu | can't find anything |
19:02.25 | *** join/#asterisk RaYmAn-Bx (rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
19:02.37 | drumkilla | faqu: rm /usr/include/asterisk/* |
19:02.43 | drumkilla | and make install again |
19:02.59 | drumkilla | also rm /usr/lib/modules/asterisk/* |
19:03.29 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
19:03.41 | faqu | the thing is i don't have any of those dirs |
19:03.52 | harryvv | <PROTECTED> |
19:03.53 | harryvv | voted to require VoIP providers to offer all U.S. customers 911 |
19:03.53 | harryvv | emergency |
19:04.21 | asterisk99 | anyone know if there are extra steps compiling Zaptel in Gentoo 2.6.11? |
19:04.59 | harryvv | http://www.voip-magazine.com/ |
19:05.00 | faqu | drumkilla i haven't those dirs |
19:05.07 | faqu | i checked |
19:05.20 | drumkilla | um ... |
19:05.25 | drumkilla | what kind of system are you on? |
19:05.28 | Mike_TK | faqu: check if file /usr/include/linux/zaptel.h exists |
19:05.32 | faqu | slackware |
19:05.42 | Mike_TK | It's defined there |
19:05.45 | faqu | yep |
19:05.45 | fugitivo | asterisk99: use udev, not devfs |
19:05.46 | faqu | it does mike |
19:05.51 | Mike_TK | faqu: should be defined there. |
19:06.00 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca) |
19:06.08 | faqu | Mike_TK i must del it? |
19:06.10 | DaLion | http://pastebin.ca/12177 |
19:06.12 | *** join/#asterisk EnigmaPTK (~bkwb@adsl-69-212-249-116.dsl.sfldmi.ameritech.net) |
19:06.19 | EnigmaPTK | Afternoon everyone |
19:06.19 | DaLion | can one ipfw expoert tell me if its line 88 causing a fault |
19:06.23 | DaLion | i cant relog in ssh |
19:06.23 | DaLion | ;) |
19:06.47 | DaLion | i meant rul |
19:07.24 | faqu | Mike_TK nop ZT_EVENT_DTMFDIGIT is not defined on zaptel.h |
19:08.03 | faqu | oh wait |
19:08.04 | faqu | its |
19:08.20 | faqu | #define ZT_EVENT_DTMFDIGIT (1 << 17) |
19:08.43 | Mike_TK | faqu: try 'locate zaptel.h' |
19:09.01 | Mike_TK | It will show you where do you have zaptel.h in other places |
19:09.09 | faqu | its on the /usr/src |
19:09.11 | faqu | where i did the cvs |
19:09.31 | Mike_TK | Delete all other, left only in /usr/include/linux/zaptel.h |
19:09.48 | *** join/#asterisk darby_t (~tom@dnr33.neoplus.adsl.tpnet.pl) |
19:09.55 | *** join/#asterisk n4y (~tmalkut@fw.orasoft.net.pl) |
19:09.56 | faqu | ok |
19:10.16 | Mike_TK | than use 'search / -name zaptel.h' |
19:11.33 | Mike_TK | faqu: sorry, i mean 'find / -name zaptel.h' |
19:11.49 | faqu | yea i am doing it a minute |
19:12.07 | exonic | on asterisk1 doing NoOp(${CALLINGPRES}) == '3', after it made it to the asterisk2 server (over IAX) ${CALLINGPRES} is '0'. Is it possible to save the CALLINGPRES variable over a switch => IAX// statement |
19:12.11 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
19:12.28 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
19:12.34 | dan2 | bkw_: ping |
19:13.29 | ariel_ | next question is how do you provide E911 for the people who travel with there ATA around. |
19:13.57 | Mike_TK | ariel_: I think it's imposible to do... |
19:14.19 | faqu | Mike_TK its only on /usr/include/linux/ now |
19:14.23 | jakepdev | you would have to register at a particular address |
19:14.43 | jakepdev | i.e. vonage lets you put in an address (i believe) |
19:14.43 | ariel_ | Mike_TK, yes your right short of having a gps system in every ata device. |
19:15.57 | jakepdev | cell phones don't always necessarily have location data - but they seem to work with 911 |
19:16.02 | harryvv | arial I dont think you can. just tell them it wont be possible. BTW thats a good point to ask. Say someone in there hotel was visiting and the owner of the ata left then some rapist would come in and the guest used the phone to the ata..who would be liable? I suspect the owner of the ata would be. |
19:16.05 | Mike_TK | ariel_: I have some another crazy idea... when customer with ata register on server, he cannot call until we will know how to find 911 |
19:16.14 | EnigmaPTK | If ANYONE has come up with an elegant way to maintain CALLERID upon the transfer (and blind transfer) of a call using the transfer button on a SIP phone, I would be willing to pay you a reasonable fee to explain your solution. |
19:16.36 | Mike_TK | jakepdev: Interesting how it's done with satellite phones like Thuraya |
19:16.47 | jakepdev | right |
19:17.18 | jakepdev | problem is VOIP doesn't have the political connections? |
19:17.18 | harryvv | mike, the isp host of the hotels isp could be located in the next jurisdiction where 911 calls are taken. That would create problems. |
19:17.23 | ariel_ | Mike_TK, and harryvv I see these things as problem due to the FCC is saying it's required now. Hummm I see allot of lawyers and people requiring signed papers to state no E911 here. |
19:17.25 | Mike_TK | jakepdev: Satellite operators know where the customer located + or - 15 Km. |
19:18.04 | jakepdev | Mike - that is pretty precise |
19:18.09 | ariel_ | 15 km is a very large area when you have an emergency. |
19:18.26 | harryvv | mike, My guess is that Dopply shift RDF technoligy may be located on the sat or mabey gps recivers with packet uplinks are used. |
19:18.44 | harryvv | are used in the customers reciver |
19:19.02 | dan2 | drumkilla: ping |
19:19.19 | asterisk99 | The BIG problem with 911 will be the risk-management side: if somehthing goes wrong, who is to blame |
19:19.37 | ariel_ | asterisk99, my point is just that. |
19:19.49 | Mike_TK | ariel_: Yes, but the local 911 operator can redirect request to some closer patrol |
19:19.52 | jakepdev | if a lawsuit comes into play, everyone is to blame |
19:20.00 | jakepdev | (everyone that has money) |
19:20.07 | xeet2 | lawsuits have already come into play |
19:20.10 | ariel_ | here in the local area I am installing soho and other biz with asterisk. I am requiring them to have a active phone line to the local pstn. |
19:20.10 | Jas_Williams | or insurance |
19:20.11 | xeet2 | like 3 against vonage |
19:20.13 | dan2 | does anybody have a full set of logs for this channel for the last 6 months |
19:20.19 | Nuxi | yup, who would settle for just one lawsuite when millions more are up for grabs. |
19:20.28 | HA | harryvv: here's a thought...what if the ISP provided a 911 lookup service that is based on the user account information? This would mean that a service runs on the isp end that returns information on how to contact 911 for the service address listed for the user account being used. |
19:20.31 | denon | dan2: not if you plan on putting all our conversations on the web.. |
19:20.52 | *** part/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
19:21.02 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
19:21.06 | dan2 | denon: I'm not, i need it to verify somethings I said |
19:21.07 | xeet2 | well, its official, you can now buy a complete list of all 911 psap's in every lata |
19:21.07 | ariel_ | HA, but what happens when they travel with that account. |
19:21.09 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
19:21.10 | syle | god damn |
19:21.11 | jakepdev | HA - that would assume the user is in the place they registered |
19:21.16 | denon | dan2: I can probably give you a grep |
19:21.17 | syle | voicemail messages sound so light |
19:21.23 | syle | can barely hear them |
19:21.25 | dan2 | denon: a full set would be better |
19:21.32 | Mike_TK | So, currently my solution for 911 is some number rewriting, and customer can change this from web interface.... if he have wrong it or not entered, it's his problem. |
19:21.36 | *** join/#asterisk jsolares (~chatzilla@200.12.33.64) |
19:21.37 | syle | anyway to increase the volume on those? |
19:21.42 | devel | dan2, i can give you what i have. |
19:21.45 | dan2 | denon: thanks |
19:21.50 | dan2 | devel: that would be great |
19:21.56 | harryvv | ha or register eveyr fricken host and domain name to a e911 service and pin it with a location. |
19:22.05 | Nuxi | Mike_TK, if they have it wrong it's YOUR problem. see vonage lawsuits. |
19:22.11 | xeet2 | you can now buy a complete list of all 911 psap's in every lata. |
19:22.18 | asterisk99 | 911 should be government based - they are the only ones who can write the laws to cover their collective asses |
19:22.34 | xeet2 | its about $400 |
19:22.37 | robin_sz | thankfully we dont have a 911 problem around here |
19:22.50 | harryvv | robin, so you think |
19:22.58 | robin_sz | you dial 911 and ... nothing happens |
19:23.07 | harryvv | every eoc has at one time or another had a 911 issue come up. |
19:23.17 | Mike_TK | Nuxi: I'm not from USA, so I have no problems here. But it's explained in contract with customer, so it's on his responsibility. |
19:23.25 | Nuxi | 119 days 10 hours 37 minutes to go. |
19:23.27 | xeet2 | you can map a source cid number to the psap for that area easily |
19:23.30 | exonic | <PROTECTED> |
19:23.42 | robin_sz | harryvv: well, in at least some aprts of the UK, dialling 911 gets you nothing ... its 999 around these parts :) |
19:24.31 | exonic | Anyone know waht i'm talking about? |
19:25.14 | fugitivo | i get this compiling zaptel on gentoo, http://pastebin.ca/12179 |
19:26.13 | ariel_ | next problem with E911 will be the users who spoof ther callerID |
19:26.21 | drumkilla | dan2: yes? |
19:27.00 | harryvv | arial, then make it the tisp responsibility to match username/pass with there number. |
19:27.01 | syle | http://lists.digium.com/pipermail/asterisk-users/2003-March/007965.html |
19:27.06 | syle | is this in current cvs head? |
19:27.40 | *** join/#asterisk aionaever (~aionaever@208.187.197.34) |
19:27.49 | Nuxi | ariel_, don't the phones have to register? We know who they are. |
19:28.02 | ariel_ | Nuxi, yes they do. |
19:28.23 | ariel_ | But I am just thinking of all the angels that I need to protect my self. |
19:28.40 | ariel_ | I can't afford a law suit I don't have the money for a defence. |
19:28.54 | Mike_TK | Nuxi: They need to register only if they need incoming calling. |
19:29.15 | Mike_TK | Nuxi: Or maybe I'm talking about something other? ;) |
19:29.18 | HA | i still think that using ip / dsn specs to get the proper 911 center should be easy to do. ips are either assigned to dial-up or high speed. high speed is almost always to a fixed location and dial-up can be reversed by caller id. |
19:29.53 | syle | are most of you guys using sox or mpg123? |
19:30.10 | ariel_ | syle, mpg123 here. (I am still only on stable) |
19:30.12 | Nuxi | mpg123, but the gals are using sox. |
19:30.15 | HA | mpg321 to play moh and sox to get it into gsm files. |
19:30.55 | syle | apparently can only increase volume of voicemail using sox |
19:31.00 | *** join/#asterisk naula (~nschmidt@printer.fxserver.com) |
19:31.20 | *** part/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca) |
19:31.48 | *** join/#asterisk blake0ps (~BLAKEOPS@c-66-41-208-245.hsd1.mn.comcast.net) |
19:31.49 | blake0ps | howdy |
19:32.21 | *** join/#asterisk diyanat (~me@217.164.253.180) |
19:32.36 | naula | I have my SIP provider and an analog line w/ FXO. What would my extention context look like if I wants to Dial 8 on my VoIP phone to dial over the analog line/FXO Card? |
19:32.49 | Mike_TK | HA: But IP addresses locations changes so ofthen! |
19:32.51 | vaewyn | HA: Hehehe... unless you are like me and are getting an IP from a location 6miles away in another district :} |
19:33.16 | jakepdev | vaewyn - wireless? |
19:33.23 | vaewyn | jakepdev: Yep... |
19:33.29 | Mike_TK | HA: And what to do with mesh networks? WiFI, or WiMAX? |
19:33.43 | jakepdev | gps - like cell phones |
19:34.01 | HA | Mesh would absolutely have to be gps just like cell phones. |
19:34.04 | vaewyn | gps doesn't work indoors... hence worthless on 99% on VoIP |
19:34.19 | Mike_TK | vaewyn: Right |
19:35.00 | Mike_TK | jakepdev: What cell phones have GPS? |
19:35.04 | vaewyn | let the customer log into a web page... and update their info... every 6 months make them verify it... if they don't lock out their service till they do |
19:35.22 | vaewyn | Mine does... and they all have some form of assisted GPS now |
19:35.57 | *** join/#asterisk jief- (~jief@modemcable196.182-80-70.mc.videotron.ca) |
19:36.03 | Mike_TK | vaewyn: Also we can ask them to change info when their IP change. |
19:36.15 | blake0ps | Is this what a call placed over IAXtel should look like at console? |
19:36.15 | blake0ps | Executing Dial("Skinny/1111@blake-2", "IAX2/blakeops1@iaxtel/18005696972@iaxtel") in new stack |
19:36.15 | blake0ps | <PROTECTED> |
19:36.22 | jief- | hey guys. i was wondering, is it normal that * spawns several instances of mpg123 when the PBX isn't even in service? |
19:36.29 | vaewyn | Mike_TK: nah... cause lots of services rotate IPs daily or more |
19:36.31 | harryvv | I really wonder if a gps reciver can be made thay tiny to place inside a cell phone. |
19:36.51 | vaewyn | Mike_TK: Or like me... I use a Wifi SIP phone from 3 netblocks in a day |
19:36.55 | HA | cell phone gps is really more like triangulation based on tower signals. |
19:36.57 | harryvv | BTW gps location been used for year..specially from ham radio opterotor like on aprs.org |
19:37.13 | naula | Can anyone help me? I have my SIP provider and an analog line w/ FXO. What would my extention context look like if I wants to Dial 8 on my VoIP phone to dial over the analog line/FXO Card? |
19:37.15 | vaewyn | harryvv: Yes... I have one... motorola i58sr |
19:37.28 | naula | Like I just want to press 8 and have the VoIP phone go over the FXO card |
19:37.33 | vaewyn | HA: many of the new ones have a real GPS also |
19:37.35 | *** join/#asterisk Deryl (pgpkeys@static-141-149-128-140.buff.east.verizon.net) |
19:37.41 | HA | vaewyn: is there just 1 tower that provides the wifi signal to your phone? |
19:37.44 | jakepdev | even if it doesn't have gps, you have triangulation with the cell towers |
19:37.45 | vaewyn | but cheaper ones are still tower assisted |
19:38.08 | ariel_ | naula, your dialing rules need to be set with an exten => 8X.,Dial(Provider) |
19:38.15 | vaewyn | HA: nope... about 50 on campus... 3 at home... and 5 at the mall when I am there... |
19:38.33 | harryvv | jak, assuming thay have doppler shift trangulation. I have my own doppler shift direction finder if you care to look at it. |
19:38.38 | jakepdev | hehe |
19:38.39 | HA | then it would be possible to use triangulation to determine your location. |
19:39.26 | harryvv | ha, assuming there is no large mountain or building near by that would cause mirror imaging |
19:39.32 | vaewyn | HA: That assumes that the APs know their exact coordinates... which they can't use GPS because they are inside... :} and then it just takes one person forgetting to update an APs location when they move it to totally whack the system |
19:40.16 | *** join/#asterisk santiago (~santiago@63.245.86.248) |
19:40.21 | vaewyn | and if one AP of 3 is bad then they are all wrong |
19:41.06 | asterisk99 | exonic: I did a re- emerge of net-msic/zaptel with "net-misc/zaptel devfs26" --- still getting the unknown symbol crc_ccitt_table |
19:41.12 | vaewyn | plus... I have 50 at work... but rarely are more than 2 overlapping... for obvious reasons |
19:41.14 | jakepdev | yep - just goes to show - there is no great solution for wireless |
19:41.42 | jakepdev | but if cell can use 911 and work - voip should also |
19:41.43 | Mike_TK | vaewyn: Hey, I think all this impossible... do you imagine cetral database that handle location of all APs in the world? And you need to modify all APs! |
19:42.03 | syle | how do DID numbers work? someone has to setup a PSTN line with a number and any incomming numbers are forwarded to SIP connection? |
19:42.44 | jakepdev | syle - a DID is simply an incoming number. if that DID is connected to asterisk, you can choose what to do with the call |
19:42.49 | JerJer[mobile] | jakepdev: but FCC doesn't mandate E-911 on cell - just 911 |
19:42.53 | vaewyn | make the user keep their location up to date... then it works and it doesn't cost the user anything but time... and the providers don't have to take the liability... the FCC was correct in requiring that |
19:43.20 | vaewyn | JerJer[mobile]: sorry... that is changing in the next couple months... cells are being required E911 as well |
19:43.29 | Mike_TK | Another one situation... Imagine big company with VoIP that use own sip server near firewall. what should happen than when someone dials 911? |
19:43.48 | JerJer[mobile] | vaewyn: yes place more responsibility on the customer |
19:43.50 | vaewyn | JerJer[mobile]: is why nextel, cingular and such sent out the A-GPS update kits for older phones |
19:43.59 | syle | yes but how do people have large blocks of DID's to hand out, lets say vonage gives you a 999-999-9999 number, how does that work? they paid for a PSTN line with that number on it for you? or are a bunch of numbers going through one number to SIP? |
19:44.19 | jakepdev | vonage bought blocks of numbers |
19:44.34 | syle | how did they setup that up? |
19:44.45 | jakepdev | with lots of money |
19:44.51 | syle | you don;t need a separate line for each DID do you? |
19:44.52 | *** join/#asterisk dros7 (~simon@H125.C18.B96.tor.eicat.ca) |
19:44.53 | asterisk99 | They did cell-phone triangulation in Tokyo a few years back - with a web front end so anyone could find a particular cell phone ---- it was yanked because wives sttarted using it to check up on their wayward hubbies |
19:45.03 | Mike_TK | JerJer[mobile]: I completely agree with you... |
19:45.17 | jakepdev | syle - you can use a T1 for example that has 24 lines |
19:45.27 | asterisk99 | "You said you were at the office... but your phone was in the Ginza district!!!!!" |
19:46.21 | Mike_TK | jakepdev: Usual 1 T1 it's enought for near like 500 incoming lines. |
19:46.26 | dros7 | anyone here have sucess getting kphone sound to work with asterisk (810 ac97 onboard sound card) |
19:46.28 | robin_sz | so, can multiple iax clients autenticate using the same user/pass or will the 2nd one dis the first or some such badness? |
19:46.39 | Mike_TK | jakepdev: If we will think that usual ERLANG is like 0.05 |
19:46.52 | syle | yes but in case of someone with large amount of numbers...can you route all those numbers to a single line to then go over SIP? i guess that don;t make much sense, more like a number comes in out of that block and it gets and available line on the T1 right? |
19:46.53 | asterisk99 | exonic: Are you using Gentoo 2.6.?? |
19:47.08 | syle | s/and/an |
19:47.17 | jakepdev | Mike - lines or DIDs |
19:47.34 | Mike_TK | jakepdev: DIDs |
19:48.00 | Mike_TK | s/lines/DIDs/ ;) |
19:48.03 | jakepdev | syle - IOW - you can have up to 24 active conversations on a T1, but many DIDs |
19:48.11 | exonic | asterisk99, you do not emerge your kernel. You need to _BUILD_ a new kernel. |
19:48.31 | syle | so you just go to the local telephone company and say i want 100 numbers attached to this T1 and they can do that? |
19:48.36 | jakepdev | syle - yes |
19:48.40 | asterisk99 | exonic: Revuild kernel??? Just because I emerged zaptel?? |
19:48.48 | asterisk99 | exonic: ReBuild kernel??? Just because I emerged zaptel?? |
19:48.58 | jakepdev | syle - and just look at the D channel for the DNIS (number that was called) |
19:49.58 | pointer | is there a way to get * to pass the original CID on an operator assisted transfer? it seems to work fine with a blind transfer |
19:51.11 | syle | thats almost kind of expensive, lets say T1 line costs 500 bucks, thats like 21 dollars a line just in DID's |
19:51.38 | jakepdev | or get an IAX provider that will get you DIDs in bulk |
19:51.45 | vaewyn | jakepdev syle: actually only 23 concurrent calls... 1 channel is the D channel and doesn't handle a call |
19:51.50 | jakepdev | iright |
19:52.12 | jakepdev | either 24 without the D or 23 with the D - tnx for the correction |
19:52.48 | exonic | lol |
19:53.16 | syle | is it possible to sip peer with one provider and get DID's somewhere else |
19:53.29 | syle | i guess the DID provider would have to SIP peer to you |
19:53.33 | vaewyn | syle: no thats 21$ a line for it handling as many DID blocks as you purchase... |
19:54.09 | vaewyn | syle: like I have 391 channels... but 2500+ DIDs |
19:54.10 | jakepdev | syle - you'd probably have the DIDs assigned by the SIP provider |
19:54.18 | tzanger | ok what the hell |
19:54.19 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
19:54.28 | tzanger | I have an iax2 peer with a specific host going to a speicifc context |
19:54.37 | tzanger | then I have an iax2 peer with a host=dynamic going to a different context |
19:54.38 | syle | vaewyn so what happens if people are trying to use 400 or more channels at one time |
19:54.49 | jakepdev | busy |
19:54.51 | vaewyn | syle: They get a busy signal... |
19:54.54 | *** join/#asterisk goldenear (~goldenear@m29.net81-64-14.noos.fr) |
19:55.14 | tzanger | if I call this box from that specific host but don't specify a context in Dial() it invalidates the call, saying that there's no exten b that name in the host=dynamic's context |
19:55.16 | vaewyn | syle: VERY rarely happens though... and that is a 6:1 ratio |
19:55.30 | zamsler | why would asterisk have 99% cpu usage with no calls? |
19:55.46 | syle | what is a good ratio? |
19:55.48 | jakepdev | zamsler - check the wiki - there is info on that |
19:56.03 | jakepdev | zamsler - look under 100% cpu usage or something like that |
19:56.18 | syle | 6 DID for every 1 physical line? |
19:56.37 | vaewyn | that's our ratio... each application needs different ones |
19:56.40 | jakepdev | that depends on the aoo |
19:56.41 | jakepdev | app |
19:56.50 | vaewyn | call centers and modem pools usually have 100s:1 |
19:57.01 | JerJer[mobile] | i've got a couple pots lines in my basement, running nufone |
19:57.01 | syle | well i mean for VOIP |
19:57.15 | vaewyn | JerJer[mobile]: bwahahaha |
19:57.18 | jakepdev | hehe |
19:57.34 | vaewyn | JerJer[mobile]: what ratio do you guys really use roughly? |
19:57.40 | vaewyn | just out of curiosity |
19:57.45 | *** join/#asterisk bah (048830696@AC963C69.ipt.aol.com) |
19:58.16 | vaewyn | I'm guessing your probably close to 4chan:1DID |
19:58.32 | Johnsie | zamsler: Do you have mpg123 running? |
19:58.36 | Mike_TK | vaewyn: In usual PSTN it's like 0.02 - 0.05 |
19:58.40 | zamsler | jakepdev,yeah |
19:58.45 | JerJer[mobile] | vaewyn: we don't have a ratio |
19:58.55 | Mike_TK | vaewyn: From 50 to 20 DIDs per line |
19:59.10 | JerJer[mobile] | we have more than 50% of immedate excess capacity at any given time |
19:59.22 | jakepdev | 2:1 |
19:59.24 | Johnsie | Oh, wait, I misunderstood your question, zamsler... Asterisk itself is at 99% ? |
19:59.24 | jakepdev | ? |
19:59.29 | zamsler | yes |
19:59.31 | JerJer[mobile] | and i could light up almost unlimited amount of more capacity in 24 hours |
19:59.36 | Johnsie | Yikes. |
19:59.38 | zamsler | 27672 root 25 0 45552 7316 4016 R 99.9 0.8 130:56.97 asterisk |
19:59.40 | Johnsie | What version? |
19:59.42 | zamsler | constantly |
19:59.43 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
19:59.46 | zamsler | stable |
19:59.48 | syle | i see so when people want to transfer their existing number to VOIP you basically have to get a T1 in that same city and get that telephone company to route that number to the T1 |
19:59.49 | zamsler | and head |
19:59.49 | Mike_TK | zamsler: WHat OS? |
19:59.50 | JerJer[mobile] | zamsler: that is top lying to you |
19:59.52 | zamsler | from today |
19:59.58 | Johnsie | Oh, so it's CVS HEAD? |
20:00.10 | vaewyn | JerJer[mobile]: you lucky bastards in the good data centers :P |
20:00.11 | JerJer[mobile] | that 99.9 percent means it is using 99.9 percent of the time the kernel has given it to process |
20:00.22 | zamsler | now it is stable |
20:00.29 | JerJer[mobile] | vaewyn: hell we've got our own transport |
20:00.40 | zamsler | voip:/usr/src/asterisk# ps aux|grep asterisk |
20:00.40 | zamsler | <PROTECTED> |
20:00.46 | zamsler | grr.. |
20:00.48 | zamsler | sorry. |
20:00.53 | zamsler | tried to keep it on 1 line |
20:01.00 | zamsler | we are dropping sip call. |
20:01.02 | vaewyn | JerJer[mobile]: so how long till you just buy out SBC in Michigan ? :} |
20:01.08 | goldenear | Does anybody knows if asterisk can give an iax client the amout/price of its communication in real time ? I've browsed the iax specs but I did not found any information about that ... |
20:01.21 | JerJer[mobile] | vaewyn: we don't want to be a copper farmer |
20:01.22 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
20:01.45 | syle | vaewyn: was i right in my assumption? |
20:02.03 | vaewyn | JerJer[mobile]: :} good choice... I wish I didn't have the 500 acres of one I have here :} |
20:02.07 | syle | vaewyn: or can that telephone company route that number to anywhere in the world to a T1? |
20:02.59 | jakepdev | a voip company can give you teh virtual DIDs |
20:03.13 | zamsler | Johnsie, I seen this issue thi smorning. then I switched to stable. |
20:03.21 | zamsler | and now I am going back to head. |
20:03.23 | *** join/#asterisk damian (~damian@damian.tikiwiki) |
20:03.28 | vaewyn | syle: 800 numbers can go anywhere... to get a local DID they have to have DIDs in the area |
20:03.29 | zamsler | <~~ is getting yelled at. |
20:03.29 | zamsler | LOL |
20:03.41 | vaewyn | syle: or a peering agreement with someone that does |
20:03.45 | jakepdev | also, I think NewSole had mentioned something about a T1 anywhere - but don't know quite how that all worked |
20:03.55 | Johnsie | Hmmm. |
20:04.04 | Johnsie | Yeah, that's an odd problem. |
20:04.05 | zamsler | I agree |
20:04.08 | damian | Hey All, not sure if this is going to be the best place to ask, I need a Wifi VOIP phone ideas ? |
20:04.16 | *** join/#asterisk Marlow (~marlow@159-134-145-12.as1.mvw.galway.eircom.net) |
20:04.18 | vaewyn | jakepdev: they have monster peering agreements... meant for modem pools and such |
20:04.24 | jakepdev | ok |
20:04.35 | vaewyn | damian: Hitachi Cable WIP-5000 |
20:05.00 | vaewyn | damian: NOT the WiSIP or the Zxel units |
20:05.05 | vaewyn | Zyxel even |
20:05.11 | Johnsie | I'm not even sure where to start diagnosing that, zamsler. |
20:05.15 | Johnsie | Do you have Zaptel cards? |
20:05.23 | damian | yeah had a zyxel some time ago and sent it back :( |
20:05.25 | zamsler | yeah a tdm card |
20:05.34 | Johnsie | Hmmm. |
20:05.44 | Johnsie | I see why you're getting yelled at now. |
20:05.44 | Marlow | vaewyn : it's a nice one .. |
20:05.45 | zamsler | uhuh |
20:05.46 | Johnsie | Lots of lines down. |
20:05.47 | Johnsie | hehe |
20:05.50 | Marlow | vaewyn : the hitachi .. |
20:05.52 | vaewyn | damian: WIP-5000... voipsupply has them for good money |
20:06.07 | zamsler | just our in house sip phones. |
20:06.16 | vaewyn | Marlow: Yeah... I love them... have 3 now in house |
20:06.27 | *** part/#asterisk hypa7ia (~leigh@serifos.eecs.harvard.edu) |
20:06.28 | *** join/#asterisk doughecka (~Tad@doughecka.user) |
20:06.32 | damian | vaewyn thanks, looking now (google) |
20:06.43 | zamsler | :(( |
20:06.44 | vaewyn | Marlow: just wish the price would go down :P |
20:06.45 | Marlow | vaewyn : a friend of mine had the ZyXEL, not good .. |
20:06.50 | doughecka | does anyone have any good iax stress testing application that I can use? |
20:06.59 | Marlow | vaewyn : so I bought the Senao ... better, but not satisfying .. |
20:07.04 | JerJer[mobile] | doughecka: use call files |
20:07.10 | Marlow | vaewyn : now i've just got a Hitachi in of the door today |
20:07.26 | *** join/#asterisk NeT-B (~Neteng-B@dsl081-228-098.chi1.dsl.speakeasy.net) |
20:07.29 | Marlow | vaewyn : in Denmark the price is about $100 higher than voipsupply |
20:07.36 | syle | vaewyn: so someone local in that area with a T1 with DID's |
20:07.36 | damian | hehe any UK suppliers ? :) |
20:07.37 | Marlow | vaewyn : for the Hitachi |
20:07.37 | NeT-B | afternoon all |
20:07.48 | doughecka | JerJer[mobile]: I only have 1 server... |
20:07.57 | Marlow | damian : doesn't look like it yet |
20:07.57 | *** mode/#asterisk [+o Cresl1n] by drumkilla |
20:08.01 | doughecka | I'd like to throw alot of incoming calls at it just to see what it does |
20:08.20 | NeT-B | can anyone helpout setting up a H323 to CM? |
20:08.39 | zamsler | <~~ gets hammer |
20:08.42 | vaewyn | damian & Marlow: http://www.andrews.edu/wwwrogue/voip/WIP5000.html |
20:08.53 | vaewyn | Is my writeup from mine... |
20:08.58 | jeremywhiting | anyone here wanna help an asterisk noob? |
20:09.11 | jeremywhiting | asterisk server says it registered with fwd ok, but not sure how to use linphone to actually call fwd number |
20:09.19 | jeremywhiting | all calls seem to be going to extensions on asterisk instead |
20:09.55 | damian | vaewyn looks like a neat one, just need to find somewhere in the UK to order one from :) |
20:10.52 | Marlow | vaewyn : especially the quality and look impresses .. |
20:10.59 | Marlow | vaewyn : the Senao is more a bit cheap |
20:11.41 | vaewyn | Marlow: *nods* only thing the Senao one has that impresses me is the better radio in it... but the firmware and design is lacking |
20:11.42 | *** join/#asterisk [hC] (~hardcore@c-65-34-203-137.hsd1.fl.comcast.net) |
20:12.04 | Marlow | vaewyn : radio ? the radio in the senao is terrible .. |
20:12.14 | Marlow | vaewyn : i hope not, that the hitachi is as bad |
20:12.17 | asterisk99 | exonic: hmmmmmm This zaptel installation is not behaving |
20:12.54 | vaewyn | Marlow: The one I tried had good range... Never tried it side by side with the hitachi's though |
20:13.06 | vaewyn | Hitachi seems to have very good receive... is a little quite on the transmit |
20:13.31 | vaewyn | but 2 units in this 3 floor buiulding it works fine 99% of the building |
20:13.42 | asterisk99 | Is anyone sucessfully using Zaptel with Gentoo 2.6.xxx? |
20:13.42 | Marlow | vaewyn : do you remember what hardware version of the Senao ? |
20:14.05 | vaewyn | Marlow: nope... only had that one for 2 days so... |
20:14.06 | damian | asterisk99 i do but on a 2.4 kernel 2.6 get crashing |
20:14.42 | asterisk99 | damian: and I have to use 2.6 because of SATA drives |
20:14.44 | *** part/#asterisk pointer (pointer@aj.catt.com) |
20:14.46 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
20:14.46 | *** mode/#asterisk [+o bkw_] by ChanServ |
20:15.42 | damian | asterisk99 no idea then, I had to downgrade to get stability |
20:16.06 | damian | now on 112 days uptime on the asterisk box with two zaptel cards :) |
20:16.54 | *** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl) |
20:18.13 | damian | vaewyn thanks! Square7.com has them in UK currency :) Will go and read up some more reviews and maybe order |
20:18.13 | Marlow | vaewyn : there is only one thing i wasn't happy about .. why t** f*** do they ship a phone like that with a 100v only psu ? |
20:18.15 | damian | cheers |
20:18.27 | Marlow | vaewyn : it's meant to be used for travelling, too |
20:21.42 | *** part/#asterisk damian (~damian@damian.tikiwiki) |
20:21.46 | NeT-B | so nobody here willing to help out on some H323 Q's with CM? |
20:22.51 | *** join/#asterisk santiago (~santiago@63.245.86.248) |
20:23.15 | *** join/#asterisk kajtzu (~kajtzu@shell1.fi.basen.net) |
20:23.35 | vaewyn | Marlow: Hehehe... agreed |
20:23.57 | blake0ps | Is this what a call placed over IAXtel should look like at console? |
20:23.59 | blake0ps | Executing Dial("Skinny/1111@blake-2", "IAX2/blakeops1@iaxtel/18005696972@iaxtel") in new stack |
20:24.00 | blake0ps | <PROTECTED> |
20:24.15 | Marlow | vaewyn : fortunatly the charger jack has screws .. |
20:24.27 | Marlow | vaewyn : and we are only talking 4 wires |
20:24.40 | vaewyn | :} |
20:24.43 | *** join/#asterisk GiabboO (GiabboO@host40-208.pool8255.interbusiness.it) |
20:24.53 | GiabboO | hi all, a quik question |
20:24.57 | GiabboO | when i use SIP channels |
20:25.35 | GiabboO | the 2 parties use the server or the server make them talk peer 2 peer ? |
20:27.19 | *** join/#asterisk focks (~craigb@12-220-210-26.client.insightBB.com) |
20:28.28 | *** join/#asterisk benno2 (~benno2@host146-58.pool8248.interbusiness.it) |
20:28.49 | ariel_ | GiabboO, it depends on how you set it up. If you have canreinvite=no then asterisk stays in the middle if you have it to yes then if there using the same codec they can talk p 2 p |
20:29.09 | *** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br) |
20:29.14 | GiabboO | okay |
20:29.22 | GiabboO | ty |
20:29.25 | syle | http://www.crtc.gc.ca/archive/ENG/Decisions/2004/dt2004-24.htm |
20:29.33 | focks | is echo typically a problem with analog lines and TDM400 cards? |
20:29.42 | HA | whats the max number of conferences that an * box could handle? |
20:29.48 | ariel_ | focks, echo is a problem everywere |
20:30.00 | focks | ariel_ with PRI too? |
20:30.07 | ariel_ | HA, that is a loaded question which it really has no reply. |
20:30.13 | ariel_ | focks, yes |
20:30.25 | benno2 | question: do you think it would be possible to use a budgetone without the handset (only speakerphone) to dial other extensions by simply dialing the number without going off-hook (it would be always offhook). if the feature is missing, would it be possible to add it via firmware updates ? or do you think its a hardware limitation ? |
20:30.37 | *** part/#asterisk NeT-B (~Neteng-B@dsl081-228-098.chi1.dsl.speakeasy.net) |
20:30.43 | *** join/#asterisk funxion (~chatzilla@mtnuser.icgws.com) |
20:30.48 | focks | ariel_ okay, well more specifically, what can be done about it as far as analog goes? |
20:31.24 | focks | it's pretty bad on my setup |
20:31.38 | ariel_ | benno2, as a speaker phone or door phone there are ways to use it. yes |
20:32.28 | benno2 | ariel_: yes somethink like a door phone, but do you think you can use a budgetone for that purpose ? I mean if it is always off-hook you you still dial a number ? |
20:32.45 | focks | i guess i'm curious as to what the factors are that control echo. is it the type of lines (A vs D), the quality of the lines, the station equipment, the Asterisk config, all of the above? |
20:32.51 | ariel_ | benno2, it has auto answer. |
20:32.56 | funxion | Im trying to set up cdr tto mysql and am having a problem the log says cdr_odbc.c:307 odbc_load_module: cdr_odbc: Unable to connect to datasource: |
20:33.12 | focks | funxion do you have the database created? |
20:33.19 | funxion | yes |
20:33.22 | focks | check MySQL permissions |
20:33.28 | funxion | \did that |
20:33.43 | funxion | <PROTECTED> |
20:33.46 | focks | and the DSN is defined correctly? |
20:33.53 | funxion | not sure |
20:34.02 | funxion | according to the config files yes |
20:34.22 | benno2 | ariel_: ok auto answer answers incoming calls. but how about the user dialing the number on the handset-free budgetone ? will it work or does it give you a busy signal because you did not dial a number within a certain time "after you went off-hook" (since it's always off-hook) |
20:34.24 | funxion | odbc.ini odbcinst.ini cdr_odbc.conf cdr_mysql.conf |
20:34.36 | funxion | those are all configured |
20:34.37 | focks | post your odbc.ini on pastebin |
20:34.44 | funxion | k |
20:34.46 | funxion | one sec |
20:34.49 | ariel_ | benno2, you have hang up button on it don't you? |
20:34.50 | focks | mysql is running i assume |
20:34.54 | funxion | yes |
20:35.27 | focks | so you can do a "mysql -u blah -p database" and issue a sample query? |
20:35.43 | asterisk99 | Anyone know what is /dev/zaptel/ctl ? |
20:36.29 | asterisk99 | Anyone know what is /dev/zap/ctl? (sorry) |
20:37.16 | benno2 | ariel_: the GS does not have a hangup button. anyway I wish a solution like a door bell, you just press a key (eg only "1"), it dials a predefined extension and you can talk. when the call terminates the called person will hangup thus the budgetone "door-phone" will hangup too. but will it be possible to dial a number from the "door-phone" again ? |
20:37.52 | ariel_ | benno2, yes it does where the hand set goes on. |
20:37.57 | ariel_ | make it a push botton. |
20:38.18 | justnulling2 | 7960 doesn't update firmwhere just downloads OS79XX.TXT and not the [s]bin file |
20:38.19 | benno2 | ariel_: ah yes ok but do you think operation without this button would be possible ? |
20:38.38 | funxion | http://pastebin.ca/12184 |
20:38.41 | funxion | ffocks |
20:38.46 | funxion | thats it |
20:38.59 | ariel_ | benno2, it's in the box you build |
20:39.34 | *** join/#asterisk dsfr (~dsfr@207.111.174.1) |
20:39.55 | benno2 | ariel_: yes but you cannot expect people at the door-phone press the hangup button before and after they speak ? Or am I misunderstanding you ? |
20:40.00 | funxion | justnulling2 did you change filename inside to match the bin file |
20:40.13 | ariel_ | put a click button. |
20:40.14 | justnulling2 | yes |
20:40.22 | focks | funxion do you have the isql command? |
20:40.28 | focks | that will let you test it |
20:40.35 | funxion | not sure |
20:40.37 | funxion | one sec |
20:40.47 | justnulling2 | funxion: yes it has then name of the file stripped of .bin |
20:40.55 | funxion | yeah |
20:41.00 | benno2 | ariel_: you mean a toggle button: when you press it stays pressed ? |
20:41.01 | focks | funxion check it with that |
20:41.06 | funxion | did you also make change to the sipdefault.cfg |
20:41.09 | funxion | k |
20:41.50 | ariel_ | yes |
20:42.08 | GiabboO | cya all thanks 4 support |
20:42.31 | funxion | focks doesnt look like its working |
20:42.34 | benno2 | ariel_: the budgetone has the offhook auto-dial which could be interesting combined with the on/off hook button but the problem is what happens when the call ends ? the speakerphone of the budgetone will emit the dialtone which is disturbing. it should be silent |
20:42.47 | funxion | here |
20:42.51 | funxion | debast:/etc# isql MySQL-asterisk -c |
20:42.58 | focks | funxion this site should be usefule http://www.unixodbc.org/ |
20:43.00 | funxion | returns [ISQL]ERROR: Could not SQLConnect |
20:43.09 | funxion | i have unixodbc installed |
20:43.14 | benno2 | ariel_: and if you use a toggle button when a user clicks it (and it will go off hook) it will stay in that state |
20:43.22 | focks | funxion did you notice this? # |
20:43.23 | focks | [MySQL-asterisk]] |
20:43.33 | focks | looks like a type in your odbc.ini |
20:43.37 | ariel_ | humm maybe a sprint load that takes 10 or 15 secs |
20:44.00 | ariel_ | or you buy one from viking |
20:44.04 | ariel_ | a door phone. |
20:44.10 | benno2 | ariel_: is viking doing voip door phones ? |
20:44.14 | funxion | hmm |
20:44.20 | funxion | no didn't |
20:44.35 | ariel_ | I saw some voip items on there site. |
20:44.53 | funxion | hey after I change it is there a service I need to restart to reload the odbc.ini file? |
20:45.00 | focks | funxion no |
20:45.07 | funxion | still doesnt werk |
20:45.11 | benno2 | ariel_: or you could use an analog door phone and an ATA |
20:45.15 | funxion | do you see any other issues with it? |
20:45.19 | *** join/#asterisk shmaltz (~chatzilla@ool-43551098.dyn.optonline.net) |
20:45.27 | focks | can you run isql with debug or verbose mode? |
20:45.36 | ariel_ | yes |
20:45.38 | shmaltz | anybody here using paetec? |
20:45.51 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
20:46.11 | ariel_ | benno2, you need a fxo for the analog viking door phone. |
20:46.11 | focks | funxion and does the socket location match up with your installation? |
20:46.12 | funxion | I have paetec |
20:46.23 | funxion | I beleive so |
20:46.26 | funxion | I can doublee check |
20:46.53 | benno2 | ariel_: yes |
20:47.01 | funxion | yes it does |
20:47.31 | benno2 | ariel_: or you could use a budgetone and some circuitry to detect the dialtone so that when the call ends it puts the switch to on-hook |
20:47.52 | funxion | focks other than the 2 files odbc.ini and odbcinst.ini is there other configs the website show the gui |
20:48.24 | ariel_ | benno2, yes that is the same place you started this |
20:48.28 | shmaltz | these ppl at the FCC are out of their mind |
20:48.49 | benno2 | ariel_: in what sense the same place you started this ? |
20:48.50 | focks | funxion take a look at the examples here and turn tracing on http://www.unixodbc.org/odbcinst.html |
20:48.59 | ariel_ | ok I have to go home. It's that time of the day. I will see you all later after I get home... |
20:49.06 | focks | and post your odbcinst.ini |
20:49.23 | benno2 | ariel_: thanks for the infos. but what was this about the sample place ... |
20:49.23 | jarrod | can asterisk be used for a h323 media/signaling gateway? |
20:50.19 | PTG123 | what ports need to be opened up for sip anyone know? |
20:50.37 | focks | i use 5060 |
20:52.56 | funxion | hey focks |
20:53.06 | funxion | whe I turned trace on it started working |
20:53.08 | syle | hey focker! |
20:53.11 | funxion | lol |
20:53.14 | funxion | thnx man |
20:53.24 | funxion | focks rocks |
20:54.05 | focks | funxion hmm, weird |
20:54.14 | file | PTG123: 5060 UDP for SIP, and the RTP ports range depending on the vendor... usualy 10000-20000 |
20:54.17 | jarrod | ast_channel_make_compatible: No path to translate from H323/xxxxx to SIP/xxxx |
20:54.24 | jarrod | why do I get this when trying to make h323 calls? |
20:54.35 | file | jarrod: asterisk can't transcode between the two codecs that each are using |
20:54.37 | focks | funxion how long you been using *? |
20:54.46 | PTG123 | thanks |
20:54.52 | file | jarrod: like when one is using G729, and another is ULAW - without a G729 license, it can't translate |
20:54.58 | funxion | too long to be asking stoopid questions |
20:55.00 | funxion | lol |
20:55.06 | focks | funxion well maybe you can help me |
20:55.10 | funxion | ok |
20:55.14 | focks | funxion you using analog or digital lines? |
20:55.20 | funxion | PRI's |
20:55.32 | focks | is echo a problem on your setup? |
20:55.55 | funxion | I mostly use cisco gateway's for pstn termination but I have 2 digium te110p's with PRI's |
20:55.58 | funxion | no echo |
20:55.59 | jarrod | hmm |
20:56.00 | jarrod | dang |
20:56.02 | syle | wtf are people doing allowing codecs people want money for anyways |
20:56.05 | jarrod | x-lite cant use g729? |
20:56.13 | file | jarrod: nope it can't |
20:56.18 | funxion | u need xpro |
20:56.19 | file | jarrod: G729 costs, X-Lite is free |
20:56.22 | funxion | focks |
20:56.24 | file | X-Pro or Eyebeam |
20:56.28 | funxion | what card do you have |
20:56.30 | focks | funxion i just setup my first system yesterday with a TDM400 and a few SIP phones |
20:56.39 | funxion | ok |
20:56.51 | focks | funxion and the echo is aweful on outbound calls |
20:56.54 | funxion | I've never played with the fxo cards |
20:57.03 | funxion | <PROTECTED> |
20:57.29 | funxion | there are some configs that you can add to your zapata.conf to tweak them a bit |
20:57.50 | focks | funxion i've tweaked them to hell and back and it's still bad |
20:58.00 | funxion | but I heard that the first second or 2 will always have echo due to training |
20:58.08 | focks | right |
20:58.22 | funxion | does it not go away for you? |
20:58.46 | focks | funxion no, it's bad on my end. the far end says they can't hear echo, but i sound kind of hollow or far away |
20:59.02 | funxion | hmm |
20:59.07 | *** part/#asterisk Patrick^ (~patrickm@pc-0-34.mountaincable.net) |
20:59.08 | funxion | what codec are you using? |
20:59.14 | focks | funxion ulaw |
20:59.20 | funxion | odd |
20:59.31 | focks | i'm sure it's the lines though because extension dialing sounds perfect |
20:59.32 | jarrod | can asterisk be a media and signalling server for h323? |
20:59.56 | focks | funxion only other thing is it's a slow box with 128MB ram |
21:00.05 | focks | i *guess* it could be a problem |
21:00.11 | funxion | how slow |
21:00.17 | focks | but being my first system, i dont' know what to expect |
21:00.22 | focks | i think it's a PII-400 |
21:00.29 | funxion | for sure thats your problem |
21:00.38 | focks | you think? |
21:00.41 | funxion | well not for sure |
21:00.45 | funxion | but most likely |
21:00.46 | fugitivo | focks: i'm using a p2-400, same problem with x100p |
21:00.56 | focks | the load is like nothing though |
21:01.16 | funxion | wen using jsut sip with one codec its very little load |
21:01.24 | cianhughes | anyone here got ISDN BRI with asterisk experience |
21:01.34 | focks | but still that may be the problem? |
21:01.34 | funxion | but when you go to TDM it increases processor load quite a bit |
21:01.35 | jarrod | so if i have g711 codecs used by SIP customers I cannot have them dialout a g729 peer? |
21:01.45 | funxion | cianhughes I do |
21:02.08 | focks | fugitivo have you tested with a faster system to see if it helps? |
21:02.28 | funxion | I have a pri card in a 3.4 Ghz no echo |
21:02.30 | cianhughes | funxion: what PCI cards have you tried, I would like to do my 1st connection of asterisk to PSTN via ISDN, but don't really want to spend 600 euro on a quadBRI |
21:02.42 | funxion | I have a te110p |
21:02.46 | fugitivo | focks: no |
21:02.47 | funxion | actually 2 of them |
21:03.03 | cianhughes | are they PRI or BRI? |
21:03.08 | focks | PRI |
21:03.34 | focks | fugitivo were you able to do anything with zapata.conf to make the echo better? |
21:03.39 | funxion | focks really I dont have fxo experience with digium cards to be able to honestly say that is it but based on what I know and what I have heard I would test a more powerful machine before going nutz |
21:03.52 | funxion | cianhughes PRI |
21:04.17 | fugitivo | focks: yes, echocancel=64, echocancelwhenbridged=yes,echotraining=yes |
21:04.20 | cianhughes | funxion: oh, I wanted to do a BRI connection, very low usage, mainly VoIP, BRI is just as backup incase IP is down |
21:04.22 | focks | funxion yeah, i was pretty disappointed and this is a testbed for my boss to determine if i can sell a full blown PRI system so i want it to work well |
21:04.38 | fugitivo | focks: you'll listen echo the first 5 sec, but then you'll listen almost no echo |
21:04.53 | focks | fugitivo i'll try the 64 |
21:06.02 | funxion | focks do you not have another machinee that you can use temporarily |
21:06.02 | cianhughes | Anyone know of site or have info on hardware IP phone that work _Well_ with asterisk, am looking to get a WLAN phone & an Ethernet one |
21:06.19 | fugitivo | focks: also, play with the rxgain and txgain, try to low them as low as possible |
21:06.33 | focks | funxion yeah i do. i started with an IBM A50 2.8GHz but it was freaking out over the TDM400 for some reason |
21:06.48 | funxion | what do you mena freaking out |
21:06.48 | focks | fugitivo i've got them cranked to like 8 right now |
21:07.16 | focks | funxion the NIC would shit itself every few minutes, must have been an IRQ problem |
21:07.23 | funxion | yeah |
21:07.30 | funxion | deifnately an IRQ problem |
21:07.33 | fugitivo | focks: try to low the values and check if echo is lower |
21:07.39 | funxion | I had that first time I put in the te110p |
21:08.31 | focks | funxion i was going to use a VS-1 from thevoipconnection for my actual install |
21:08.41 | funxion | VS-1 |
21:08.45 | funxion | what is taht |
21:08.57 | focks | check thevoipconnection.com |
21:09.09 | focks | it's a fanless system with flash instead of hard drives |
21:09.44 | funxion | you can build one of those your self |
21:09.52 | focks | i'm in KY and my client is in NC, i don't want a fan or hard drive dying 600 miles away |
21:10.03 | jarrod | thats pretty sad if asterisk cant translate the different codecs i.e. ulaw into 729 |
21:10.05 | focks | funxion i know, but i can get overnight hardware and support from them |
21:10.11 | jarrod | what if you have ulaw devices and 729 gateway peers |
21:10.31 | focks | funxion and for $1500, that's way better than the IBM X226 I specced before |
21:10.43 | funxion | I can see your point |
21:10.57 | funxion | but I stilll like building things myself |
21:11.07 | focks | funxion i see yours too. if it were for me, i'd build it. for a customer, buy it with fanatical support |
21:11.18 | funxion | plus I have a ton of 2 U servers |
21:11.20 | focks | we're 100% IBM reseller |
21:11.30 | funxion | got ya |
21:11.44 | funxion | we're an HP & cisco shop |
21:12.15 | focks | it's been a hard transition coming to work here though, i used to build everything myself and now i've lost so much of my knowledge about the hardware. we just defer to IBM and tech data for everything |
21:12.28 | funxion | lol |
21:12.48 | focks | before i wanted to know the chipset and firmware revision etc, now i just call 800-IBM-SERV and tell em it's broke |
21:12.59 | funxion | we build everything ourselves but use hp techsupport for help |
21:13.01 | funxion | lol |
21:13.14 | funxion | omg ur lazy |
21:13.27 | focks | we're an integrator with so much in the field it's the only way to make $$ |
21:13.45 | funxion | got ya |
21:13.46 | file | everyone of you should come to Cluecon! |
21:13.52 | file | you'll get all your questions answered there |
21:13.53 | focks | who has 3 hours to drive out, get a PSU, drive to customer and replace it when IBM will do it on-site next day for free |
21:13.54 | file | see topic for details. |
21:13.55 | naula | sorry for asking this again but I have my SIP provider and an analog line w/ FXO. What would my extention context look like if I wants to Dial 8 on my VoIP phone to dial over the analog line/FXO Card? |
21:14.42 | focks | nothing gets sold without 3 yr 9x5 NBD or 24x7 same day service |
21:15.08 | funxion | naula why 8? |
21:15.18 | funxion | focks thats kewl |
21:15.42 | focks | ha, but for some reason, we standardize on Netgear switches! unless they need Cisco |
21:15.53 | funxion | whoa why is that |
21:15.54 | naula | it doesnt need to be 8 |
21:15.57 | focks | fuq if i know |
21:15.58 | naula | any number really |
21:16.08 | naula | I would just prefer 8 |
21:16.15 | funxion | o |
21:16.18 | funxion | I got you |
21:16.22 | focks | i need a new job ;) |
21:16.36 | naula | any idea how I would go about that? |
21:16.38 | funxion | I jsut started my job like 6 months ago |
21:16.41 | *** join/#asterisk LeoB (~chatzilla@pool-70-20-20-153.bstnma.fios.verizon.net) |
21:16.56 | [hC] | Hey if i was looking to resell voip service to my clients, and i was looking for either unlimited or great priced incoming calls (including DIDs) which voip companies are on the top there? |
21:16.56 | funxion | moved from a travel conglomerate to a satellite com provider |
21:16.57 | focks | i should quit and be an Asterisk consultant |
21:17.06 | [hC] | Im using link2voip right now, but their service is less than super. |
21:17.23 | funxion | hC not sure but thnx for the warning |
21:17.40 | funxion | hC depending on your volume you might want to look at level3 |
21:18.12 | LeoB | (beginner) having problems with Festival. Error: "app_festival.c:445 festival_exec: Festival returned ER" Any ideas? |
21:18.29 | funxion | LeoB do you have a sound card? |
21:18.40 | LeoB | yes |
21:18.44 | funxion | and are the dirvers loaded for it? |
21:18.50 | [hC] | funxion: no level3 in canada. |
21:19.01 | LeoB | hmm I'll have to check! |
21:19.08 | funxion | they can still provide service and DID's in canada |
21:19.25 | funxion | uh oh |
21:19.29 | funxion | time to go home |
21:19.40 | funxion | take it easy everyone |
21:19.44 | funxion | thanks again focks |
21:19.48 | LeoB | funxion, why do I need a sound card? |
21:19.56 | funxion | dsp's |
21:20.43 | LeoB | anyone? How to solve: "app_festival.c:445 festival_exec: Festival returned ER" ? |
21:21.01 | focks | LeoB like he said, make sure your sound card is configured |
21:21.20 | LeoB | focks, why do I need a sound card? |
21:21.59 | focks | LeoB i don't know, don't use festival |
21:22.17 | LeoB | focks, what would you recommend? :) |
21:22.52 | focks | no i meant I don't use festival |
21:23.00 | naula | Record it yourself, Those syntisizers usually sound horrible |
21:23.05 | LeoB | oh, ok! |
21:23.30 | focks | it would make sense to me though that a sound card would be necessary to do the text->speech conversion |
21:23.34 | focks | by means of a DSP |
21:24.17 | LeoB | hmmm, I'll check my machine |
21:24.24 | focks | http://www.cstr.ed.ac.uk/projects/festival/manual/festival_6.html#SEC13 |
21:26.45 | naula | any idea about my question? :) |
21:26.47 | denon | cepstral is very fast for text to speech |
21:26.51 | denon | and requires no hardware |
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21:31.06 | *** join/#asterisk [hC] (~hardcore@c-65-34-203-137.hsd1.fl.comcast.net) |
21:32.23 | doolph | hi everyone |
21:32.23 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
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21:34.30 | *** join/#asterisk carbon60 (~carbon60@gw.techsupport.ca) |
21:34.34 | carbon60 | Evening all. |
21:34.52 | *** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
21:34.57 | carbon60 | Is there a way to disable the hold music used when using MeetMe's "wait for marked user" feature? |
21:35.24 | carbon60 | Or maybe setup a "silent" hold music type? |
21:36.17 | jarrod | when trying to make a h323 call i keep getting == No one is available to answer at this time |
21:38.54 | *** part/#asterisk Grooby (~Grooby@66.160.105.186) |
21:40.49 | *** join/#asterisk zeroxeal (~krickle@ool-44c16c53.dyn.optonline.net) |
21:42.39 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
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21:42.54 | mikewho2 | anyone have any recommendations on where to get a tdm card? |
21:43.31 | ariel_ | mikewho2, if the two on lines source I use is voipsupply.com and atacomm.com |
21:44.46 | *** part/#asterisk flotox (jovan@host154-75.pool80183.interbusiness.it) |
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21:44.58 | *** part/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
21:46.54 | mikewho2 | should i just get the t100p for a t1 card? |
21:46.57 | mikewho2 | or the newer wild cards |
21:47.12 | mikewho2 | wow |
21:47.16 | mikewho2 | didnt see price difference |
21:47.29 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
21:47.37 | jets | What ip phons does everyone favor? |
21:47.41 | jets | looking for alternative to the 7960's |
21:47.54 | xeet2 | I hear the new grandstreams are nice |
21:49.12 | mikewho2 | why dont ya like the 7960s |
21:49.14 | mikewho2 | i was just about to buy some |
21:49.47 | ariel_ | jets, I love the Polycom IP-500 and IP-600 great phones. The IP-500 will soon be changed out by the 501 but for now it's still worth the prices. |
21:49.49 | jets | They are pretty good.... Just a lot of look and not great functionality for cost... and i want a good backlight |
21:51.12 | *** part/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
21:51.13 | jets | are we supposed to pay a sip license on each phone? hrm we converted from sccp and previously had a really expensive sccp license on them and just moved them to sip |
21:51.17 | jets | hopefully we're okay on that |
21:51.27 | *** join/#asterisk therouterboy (~icechat5@pcp0011553856pcs.anapol01.md.comcast.net) |
21:52.08 | ariel_ | jets, no you need to lisc the sip one. If you want to be leagel |
21:52.39 | ariel_ | sorry .. leagal...legal |
21:53.20 | jets | ya, good to know though |
21:53.20 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
21:53.24 | jets | it's better then being surprised |
21:53.25 | ariel_ | jets, they replaced the t100p with the new te110p board. |
21:56.15 | jets | huh? |
21:56.23 | benno2 | ariel_: this seems the way to go :) http://www.rocom-gmbh.de/englisch/produkt/tuerstation/a_b/doortel4e.htm |
21:57.36 | ariel_ | benno2, nice |
21:58.00 | ariel_ | jets, wrong person the t100p was actually for mikewho2 |
22:01.41 | *** join/#asterisk astoria (~haydenth@66.235.201.217) |
22:02.02 | astoria | Hey, anyone manage to get the "buddies" feature working on a polycom Ip300,500, or 600?? |
22:02.35 | jarrod | anyone ever been able to place h323 calls from asterisk to a gateway? |
22:02.56 | astoria | not me :) |
22:03.33 | ariel_ | jarrod, I set a customer up once. But have not used it or even tested anything on h323 in over a year. |
22:04.22 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com) |
22:06.25 | jarrod | i just try H323/1@<gw ip> |
22:06.30 | jarrod | nothin .. hmm .. |
22:06.33 | jarrod | chan_h323 loaded |
22:08.23 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
22:09.10 | justnulling2 | can someone expalina why 7960 doesn't load new firmware that OS79XX.TXT specifies? |
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22:14.20 | ariel_ | jarrod, you installed h323 correctly via the readme that is supplied? |
22:14.31 | astoria | I hear that h323 is a nightmare. |
22:14.36 | ariel_ | justnulling2, you have to do it in stages from older to newer. |
22:14.39 | syle | what is defined as a channel for the g729 codec? |
22:14.59 | syle | a zap channel, or just multiple SIP connections in general? |
22:15.07 | ariel_ | syle, you mean diallow=all allow=g729 |
22:15.21 | syle | pretty much |
22:15.38 | ariel_ | zap does not use anything other then ulaw/alaw and there is no setting for it. |
22:16.00 | justnulling2 | ariel_: where do i get older images and which ones do i need to update from p003am30 to 7.4? |
22:16.01 | *** join/#asterisk doolph (doolph@200.46.148.35) |
22:16.02 | syle | so g729 is basically for sip or iax then? |
22:16.02 | ariel_ | syle, but you have to get the lisc installed before you use it. |
22:16.41 | syle | http://www.readytechnology.co.uk/open/g729/ |
22:16.44 | syle | this url? |
22:16.45 | ariel_ | justnulling2, argh I have not used a Cisco in a while. Maybe there on the wiki some place. But I think you can download them from Cisco if you have a smartnet |
22:17.01 | ariel_ | syle, yes iax, sip, h323 |
22:17.24 | syle | i was looking at digiums site but they wany money for it even just for testing |
22:18.31 | syle | i;m on the page |
22:18.33 | syle | http://www.intel.com/software/products/noncom/ |
22:18.39 | syle | where do i get the license |
22:18.51 | ariel_ | syle, I have only used the one from digium. |
22:19.32 | justnulling2 | does anyone here has smartnet? |
22:20.20 | ariel_ | justnulling2, it's not legal for someone to give there firmware. |
22:21.43 | justnulling2 | ariel_: why isn't it legal i have the physical phone |
22:22.23 | ariel_ | justnulling2, it's a lisc issue that Cisco only lisc it's sip firmware to the user of the phone. It can't be transfered and one is required per phone. |
22:25.15 | *** join/#asterisk JerJer[mobile] (~jj@ip68-103-26-140.ks.ok.cox.net) |
22:26.18 | justnulling2 | that's so messed up i don't know where to begin, at any case do i need all version from 3.0 up or just major verions? and do i have to pay per version or do i get old ones free when i buy latest firmware? |
22:26.40 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
22:27.21 | ariel_ | justnulling2, you need to go from 3 to 5 I think you can skp the 4 at least I did way back when. But then you need the 6 and the 7. |
22:28.04 | ariel_ | If you do a search google you might fine them. But since I do belive in having software legal I can't really help other then that. It's up to you. |
22:28.25 | loud | you sure ? ive never done those steps and my 7960 is working fine. |
22:28.26 | *** join/#asterisk Jedirl (ircap@154.Red-217-127-168.pooles.rima-tde.net) |
22:28.27 | Jedirl | Hello |
22:29.30 | ariel_ | loud, he is upgrading from an older version 3 he needs the others or cisco will not upgrade. |
22:29.40 | loud | ah |
22:29.44 | justnulling2 | ariel_ i understand, do i need version 3 of sip or can go from 3 skinny to 5sip? |
22:30.07 | ariel_ | I don't agree with Cisco's ways. So I don't use them any more. Besides I feel Polycom phones are better for the price. |
22:30.35 | loud | i bought mines on ebay, dunno which version had, but i just upgraded to 7.0.3 and works. |
22:30.37 | ariel_ | justnulling2, I don't know about that one. I have never played with there skinny |
22:30.40 | loud | downloaded with my cco. |
22:31.21 | *** join/#asterisk zyke (~zakforeve@84.45.132.117) |
22:31.24 | justnulling2 | load: form what firmware did you upgrade and to which one? |
22:31.25 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
22:31.53 | loud | cant remember from what firmware. just plugged it, tftpd and was ready to go. |
22:32.08 | loud | that is not 100% true, nugget helped me a bit. |
22:33.32 | Nugget | that's a lie! if people think I helped you they'll expect me to help everyone! |
22:33.34 | ariel_ | justnulling2, here is some steps you can take: http://www.voip-info.org/tiki-index.php?page=Setup%20SiP%20on%207940%20-%207960 |
22:33.38 | justnulling2 | where s nugget when we need him:) |
22:33.45 | Nugget | really I called you silly names and hid under a desk |
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22:37.48 | *** part/#asterisk crash3m (crash3m@crash3m.user) |
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22:39.07 | darwin35 | astpbx*CLI> sip debug |
22:39.07 | darwin35 | No such command 'sip debug' (type 'help' for help) |
22:40.08 | astoria | do you have sip on your box? |
22:40.20 | darwin35 | yes this is in head |
22:40.31 | darwin35 | and it is broken it was just confirmed |
22:40.51 | astoria | i installed debian stable the other day.. no sip.. lol |
22:45.11 | ariel_ | justnulling2, did you fine the info useful I posted for you? |
22:54.17 | *** join/#asterisk three55ml (~three55ml@cpe-24-243-30-75.satx.res.rr.com) |
22:54.51 | three55ml | I've asked this before and I lost the sample, but is anyone doing prompts on transfers? I.e. for to a cellphone - "Press 1 to accept this call, 2 to send to voicemail"? |
22:57.02 | Jedirl | three55ml: with an AGI script it can be done in minutes... |
22:57.30 | *** join/#asterisk [1]jakepdev (~JakePDev@pool-68-163-55-23.phil.east.verizon.net) |
22:58.41 | three55ml | Jedirl: What's the AGI command to play music on hold, do you know? I can look it up easily. |
22:59.05 | Jedirl | don't know |
22:59.15 | Jedirl | look at voip-info |
22:59.19 | [1]jakepdev | EXEC MUSICONHOLD |
22:59.22 | three55ml | Yeah, I know |
22:59.26 | three55ml | jake: Thanks |
23:00.20 | [1]jakepdev | np |
23:01.18 | justnulling2 | ariel_: nothing that helped me get it working |
23:02.13 | astoria | time to go home from work! yipee |
23:10.34 | justnulling2 | ok got 5.3 image now the phone says "upgrading software" but tftpd has errors Peer returns ERROR < > -> aborting transfer and that is after i renamed it to P0S30530.bin (without dashes and last 0) |
23:10.41 | justnulling2 | any ideas? |
23:11.23 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
23:16.53 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
23:17.21 | AgiNamu | Hey there, if I have a single T1 and am doing full PSTN->VoIP (ULAW), is a single xeon 2.8Ghz enough? |
23:18.29 | mikewho2 | i was told yes. |
23:19.18 | Jedirl | AgiNamu: transcoding is expensive, you should try to test it before |
23:19.52 | AgiNamu | well, just Slin to ulaw |
23:20.18 | Jedirl | don't know |
23:20.34 | Jedirl | everything I've done in asterisk doesn't involve any transcoding |
23:21.04 | AgiNamu | yea |
23:21.09 | AgiNamu | and I dont have a PRI to play with :P |
23:21.38 | Jedirl | you can try transcoding VoIP-ulaw to another protocol |
23:22.28 | AgiNamu | yea, i dont think the transcoding is gonna be an issue |
23:22.34 | AgiNamu | it's only 23 calls |
23:23.15 | Jedirl | if there's no transcoding, it's very cheap to place a call in asterisk, right? |
23:23.27 | AgiNamu | yea, and the transcoding is light, ulaw to slin |
23:23.29 | Jedirl | which other factors are involved - aside from codec? |
23:23.32 | AgiNamu | my question is more of what load the card should put on |
23:23.35 | AgiNamu | the TDM stuff |
23:24.07 | Jedirl | don't know, but yesterday I was told that my P-II 1Ghz with 1Gbyte and SCSI storage would be enough for a non-transcoding setup for 4-E1 (120calls) |
23:24.14 | Jedirl | P-III |
23:24.21 | AgiNamu | oh ok |
23:24.25 | AgiNamu | then I'm set with a 2.8GHz :) |
23:24.51 | Jedirl | I guess you are, yes |
23:24.52 | Nugget | buy a mac(tm) |
23:25.03 | Jedirl | mac supports zaptel hardware? |
23:25.25 | Nugget | no, but you'll be so happy you won't care. clearly. :) |
23:25.29 | AgiNamu | can those Digium cards fit in a 1U Dell? |
23:25.51 | Jedirl | hehehehe |
23:26.00 | Jedirl | I guess dell isn't the preferred platform for digium hardware |
23:26.09 | AgiNamu | Why |
23:26.16 | Jedirl | too many bad experiences explained in the mailing list |
23:26.23 | AgiNamu | Hmm |
23:26.27 | AgiNamu | HP then? |
23:26.32 | AgiNamu | IBM? <shudder> |
23:26.51 | Jedirl | I would recommend Intel hardware |
23:26.58 | Jedirl | Intel motherboards with Intel server chasis |
23:27.05 | AgiNamu | yea, but this is for a mission critical system |
23:27.09 | AgiNamu | So I gotta go with a vendor |
23:27.19 | AgiNamu | and I only know Dell, HP, and IBM |
23:27.20 | Jedirl | go with a vendor which provides such hardware |
23:28.14 | Jedirl | if not possible, I would clearly go with IBM |
23:30.03 | syle | christ since i added that g729 module my computer crashes everytime it loads it |
23:30.27 | Jedirl | AgiNamu: but try not to go with dell, search the mailing list and you'll see that they're not the best platform for digium hardware |
23:30.31 | ariel_ | syle, your using the open source one? |
23:30.36 | syle | yeah |
23:30.41 | AgiNamu | Jedirl, i am searching... dont see anything that bad. |
23:30.59 | AgiNamu | I'm guessing it's reported more, cause Dell is so popular. |
23:31.12 | Jedirl | well, do whatever you want |
23:31.15 | Jedirl | it's your money, not mine :D |
23:31.18 | ariel_ | I don't know why people keep saying dells are bad. I use allot of dells for asterisk without problems |
23:31.50 | syle | although this test computer is running a AMD processor |
23:32.02 | AgiNamu | ariel_, with a single-span PRI card? |
23:32.08 | ariel_ | syle, the open is only for intel cpu's |
23:32.31 | ariel_ | AgiNamu, I have them using single, and even multi TDM boards. |
23:32.56 | syle | http://kvin.lv/pub/Linux/Asterisk/ |
23:33.03 | syle | they compiled amd ones |
23:33.06 | syle | so i figured it may work |
23:33.12 | AgiNamu | ariel_, any 1U units? |
23:33.25 | ariel_ | I used an older 750 |
23:33.34 | Jedirl | AgiNamu: really, try intel server hardware... dell compared to intel is crappy |
23:33.44 | ariel_ | but most of the ones this year has been the SC400 and SC420's. |
23:33.59 | AgiNamu | yea, we just buy dell everywhere else |
23:34.03 | AgiNamu | I'll look at IBM |
23:34.11 | syle | IBM is aweful |
23:34.11 | AgiNamu | and HP |
23:34.17 | Jedirl | why IBM is awful? |
23:34.23 | syle | man even their power supplies are custom |
23:34.24 | AgiNamu | IBM sucks, I know that. |
23:34.27 | syle | real pain |
23:34.30 | Jedirl | why? |
23:34.32 | AgiNamu | at least their software blows |
23:34.40 | AgiNamu | my friend works at IBM. it's the biggest scam on earth :P |
23:34.44 | AgiNamu | but their intel servers might be fine |
23:34.51 | Jedirl | heh |
23:35.18 | syle | i hate IBM, but they do develop kewl shit in the artificial intelligence dept |
23:35.28 | AgiNamu | yea, so long you dont use their commercial products |
23:35.38 | AgiNamu | like their content management/websphere crap |
23:35.44 | AgiNamu | they haven't figured out what "threads" are |
23:35.52 | syle | their top secret robots are kewl shit |
23:36.16 | syle | kinda reminds you of the terminator movies |
23:36.41 | *** join/#asterisk |Vulture| (~V@c-69-180-67-228.hsd1.fl.comcast.net) |
23:38.36 | Jedirl | I don't use any IBM "enterprise" software products like websphere and that |
23:38.43 | Jedirl | but IBM hardware and operating systems are great |
23:39.05 | syle | IBM hardware lol |
23:39.35 | *** join/#asterisk sandnigg0r (~niggerplz@66-55-197-254.gwi.net) |
23:39.44 | AgiNamu | I hope Asterisk biz-edition works :P |
23:39.59 | syle | go with dell for servers is usually the way to go, but lately Dell has been releasing shit incompatible with linux installs |
23:40.11 | AgiNamu | syle, even when they come with RHEL installed? |
23:40.16 | AgiNamu | I want RHEL, btw. |
23:40.19 | AgiNamu | supported |
23:40.31 | syle | well i refuse to get rhel, i;d rather install fedora core |
23:40.36 | syle | waste of money |
23:42.13 | AgiNamu | well, i like the ide aof being able to call them for support |
23:42.13 | syle | yep so i remove the g729 module and all works fine again |
23:42.20 | AgiNamu | and having them manaage patches and email me |
23:42.54 | syle | patches for what? |
23:43.04 | Jedirl | AgiNamu: they wont support an asterisk installation |
23:43.23 | Jedirl | AgiNamu: they will support your apache, will support your samba, your redhat-cluster and your GFS, but not your asterisk |
23:43.52 | AgiNamu | i know |
23:43.56 | Jedirl | if you're going to run only asterisk on that machine, you'd better go CentOS |
23:44.04 | syle | lol |
23:44.16 | syle | what is wrong with fedora core 3 and freebsd |
23:44.18 | syle | lol |
23:44.30 | AgiNamu | i'm drop shipping to a datacentre |
23:44.32 | Jedirl | syle: fc3 == 6 months of patches |
23:44.34 | AgiNamu | dont feel like remote installing it |
23:44.40 | AgiNamu | and it's not supported |
23:44.46 | AgiNamu | nor tested as well as rhel |
23:44.58 | AgiNamu | dell gets REALLY expensive once you add support. IBM is only $500 for 24x7x2 hour support |
23:45.09 | AgiNamu | dell is $1500 for their top-level, 4hr support |
23:45.19 | Jedirl | IBM is only $500? |
23:45.19 | ariel_ | I have been using CentOS 3.4 works great even with QFS mounts. |
23:45.29 | Jedirl | ariel_: you use GFS? |
23:45.44 | ariel_ | Jedirl, yes on one of our installations we do. |
23:45.56 | Jedirl | ariel_: which hardware? |
23:46.03 | AgiNamu | IBM ServicePac: 3 year onsite repair 24x7x2 hour ($503) |
23:46.10 | ariel_ | supermicro |
23:46.12 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
23:46.23 | Jedirl | ariel_: shared-storage hardware, I mean |
23:47.06 | *** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3880851.sympatico.ca) |
23:47.14 | DaLion | anyone good with natd and ipfw ? |
23:47.16 | DaLion | msg me plz |
23:47.26 | Jedirl | DaLion: go ipfilter man, ipfw sucks =D |
23:47.48 | DaLion | nah ..wasted 3 hours on this cant switch now |
23:47.50 | DaLion | so close |
23:48.01 | ariel_ | Jedirl, let me login and check I did this setup remotely. |
23:48.11 | Jedirl | ariel_ ok =D |
23:48.33 | DaLion | <PROTECTED> |
23:48.37 | DaLion | got this first |
23:48.51 | DaLion | then i add add allow all from any to $oip 4569 |
23:49.00 | DaLion | then ${fwcmd} add pass udp from ${oip} to ${iip} 4569 keep-state in |
23:49.01 | AgiNamu | meh, and Dell's remote access card doesn't work with RHEL 64 bit |
23:49.01 | Jedirl | haven't used ipfw for ages, sorry |
23:49.07 | AgiNamu | that's fucking retarded. |
23:49.17 | Jedirl | AgiNamu: asterisk works with RHEL 64bit? |
23:49.21 | Jedirl | I mean, zaptel hardware |
23:49.33 | *** join/#asterisk Legend (~legend@24.244.142.133) |
23:49.37 | AgiNamu | I think so |
23:49.42 | DaLion | oh |
23:49.42 | AgiNamu | maybe not :\ |
23:49.49 | syle | jedirl, 6 months of patches sure, but you have to look at the people using it if they have had any problems with it |
23:49.53 | AgiNamu | not like i need it |
23:49.57 | DaLion | what does keep-state acutaly do |
23:49.58 | syle | won;t find any |
23:50.33 | syle | and keep in mind if people putting down fedora core are people trying to make money off you for RHEL |
23:50.51 | AgiNamu | syle, the thing is, 6 months of patches means not acceptable for business |
23:51.01 | Jedirl | syle: 6 months for patches is just not enough |
23:51.17 | AgiNamu | with RHEL, there's 3 or 6 years? |
23:51.19 | syle | hell i know at least 3 universites in the US all running fedora core 3 |
23:51.20 | AgiNamu | similar to MS |
23:51.26 | AgiNamu | syle, that';s great. |
23:51.30 | AgiNamu | I'm going to be running 911 trunks |
23:51.32 | Jedirl | syle: if they can spend reinsstalling often, great |
23:51.41 | AgiNamu | it cannot go down |
23:52.25 | syle | it wouldn;t be so bad if RHEL had a one time fee, but to have reoccuring billing every year is bothersome |
23:52.46 | AgiNamu | meh, $800 for 3 years |
23:52.47 | AgiNamu | not that bad. |
23:52.53 | AgiNamu | ~double Windows |
23:53.02 | syle | great now add your other 50 linux boxes to that price |
23:53.09 | Juggie | then use centos |
23:53.10 | ariel_ | Jedirl, emc2 |
23:53.10 | ariel_ | <PROTECTED> |
23:53.19 | AgiNamu | I only use RHEL on production servers. |
23:53.23 | Jedirl | oh, ok |
23:53.29 | AgiNamu | And quite frankly, if I need 50 production servers, then I can afford to pay RHEL |
23:53.45 | AgiNamu | im going to buy Asterisk biz edition as soon as it's available |
23:54.18 | syle | agi your obviously a linux noob quit while your ahead :) |
23:54.56 | AgiNamu | syle, no, im not in love with Linux |
23:55.03 | ariel_ | Jedirl, but there are a total of 3 servers plus the enclosure for the qfs system. |
23:55.05 | AgiNamu | And I like having someone else be responsible |
23:55.12 | AgiNamu | if you think 6 months of patches, you're just a noob |
23:55.13 | Jedirl | ariel_: :) |
23:55.16 | Jedirl | ariel_: ok |
23:55.31 | AgiNamu | and i dont see how Asterisk Bizedition has to do with linux |
23:55.31 | Jedirl | ariel_: minimum configuration, then. Two-node GFS has no sense. |
23:55.46 | AgiNamu | I'll pay digium $2000 a server if they are gonna deliver a solid platform |
23:55.58 | syle | your talking to a 10 year unix administrator :) |
23:56.03 | syle | your really starting to look bad :) |
23:56.23 | Jedirl | =D |
23:56.33 | AgiNamu | well, great, im glad you got so much experience. however, i dont know where you work, but pretty much any business environment is not going to let themselves rest on an OS that is only supported for 6 months |
23:56.40 | AgiNamu | <PROTECTED> |
23:57.02 | mikewho2 | open source does not have a buisness model |
23:57.03 | syle | thats what stress testing is for |
23:57.04 | mikewho2 | its open source |
23:57.05 | mikewho2 | bud |
23:57.11 | AgiNamu | mikewho2, lol |
23:57.15 | |Vulture| | digium hardware is totally different |
23:57.23 | Jedirl | syle: having 6-month upgrade cycle sucks. really. :) |
23:57.23 | mikewho2 | me? |
23:57.34 | AgiNamu | see, i dont see Asterisk + Linux + Some hardware |
23:57.37 | AgiNamu | I see a solution |
23:57.38 | mikewho2 | |Vulture| does the name 'acidtear' ring a bell? |
23:57.41 | ariel_ | Jedirl, yes your correct. That is why it's 3 servers..... |
23:57.41 | mikewho2 | cause u sure look familiar |
23:57.49 | AgiNamu | and quite frankly, I don't give a flying f* what's involved. |
23:58.00 | AgiNamu | if it runs linux, solaris, hell, windows + cygwin |
23:58.01 | |Vulture| | mikewho2: Ive been around this channel for a little over a year |
23:58.09 | AgiNamu | if it's going to meet MY business goals, that's all I care about. |
23:58.14 | *** join/#asterisk geesus (~geesus@220-244-218-250-qld.tpgi.com.au) |
23:58.32 | syle | hell the way you spend money you should hire me, save you thousands of dollars lol |
23:58.33 | ariel_ | AgiNamu, where are you going to locate these boxes? |
23:58.58 | |Vulture| | ariel_: do you have any PRIs in miami, or use XO down there? |
23:59.04 | ariel_ | yes |
23:59.06 | AgiNamu | Undecided data centre in Colorado |
23:59.14 | geesus | Helloooo all, anybody have an ETA on when the TDM400P's will be A-Ticked ? |
23:59.19 | AgiNamu | well, perhaps it's decided, but NDA |
23:59.22 | Jedirl | what is XO?? |
23:59.24 | |Vulture| | ariel_: who do you use for your PRIs I am shopping |
23:59.29 | AgiNamu | XO is a large carrier |
23:59.29 | ariel_ | AgiNamu, ok I have a nice place in LA if you need. |
23:59.38 | ariel_ | XO and nuvox |
23:59.46 | AgiNamu | yea, it has to be in colorado. we're connecting via 3 dedicated links to a company in colorado |
23:59.54 | mikewho2 | ariel_ is there a service on the internet ot locate possible PRI's in your city? |
23:59.56 | |Vulture| | ah I don't like nuvox's customer service |
23:59.59 | mikewho2 | im in the market |