| 00:00.04 | justinu | probably should be foreign exchange, but that's just a guess |
| 00:00.45 | ManxPower | kink0, define "plan" |
| 00:01.13 | kink0 | guest@switch-3.asterlink.com/996 sounds very fine |
| 00:01.32 | justinu | heh |
| 00:01.34 | kink0 | plan=dialplan |
| 00:02.00 | kink0 | extension I have defined, guest@switch-3.asterlink.com/996 sound perfect |
| 00:02.13 | marcus2 | i'm really curious why i get just lots of "?" and "(null)"s when i have intense debugging turned on |
| 00:02.18 | ManxPower | kink0, that would be "some providers sound perfect." |
| 00:02.43 | kink0 | but if I dial IAX2/216.207.245.8:4569-1 sound distortioned |
| 00:02.47 | ManxPower | Also, remember when you dial by hostname or IP you do not get any of the options in iax.conf or sip.conf (you MIGHT get the [general] options, you might not) |
| 00:03.10 | marcus2 | hm ok |
| 00:03.14 | marcus2 | now to try "switch type" |
| 00:03.16 | *** join/#asterisk coppice (n=chatzill@199.192.17.210.dyn.pacific.net.hk) |
| 00:03.16 | justinu | marcus: probably hdlc errors? |
| 00:03.45 | kink0 | ManxPower, yes, I am ussing the same [general] for both calls. |
| 00:04.30 | marcus2 | the magix gives me a few options... 4ess, 5ess, dms-250, dms-100, dex600e |
| 00:04.40 | justinu | whatever your asterisk box is pretending to be |
| 00:04.48 | marcus2 | yeah, both are set to 5ess right now |
| 00:04.53 | marcus2 | i'm wondering if something else is preferrable |
| 00:04.58 | kink0 | how will be able to test an incoming call since nobody calls me ? |
| 00:05.11 | justinu | probably not, since it doesn't have NI-2 support? |
| 00:05.18 | marcus2 | nope, no ni-w support, heh |
| 00:05.19 | justinu | kink0: what's your number? |
| 00:05.21 | kink0 | there any "call back" or so ? |
| 00:05.22 | marcus2 | er, ni-2 |
| 00:05.39 | kink0 | justinu, AIX2/guest@asterisk.interec.org/1 |
| 00:05.51 | justinu | oh, no pstn to that |
| 00:05.55 | kink0 | I hope that runs, I never got a call |
| 00:06.18 | kink0 | no.. just compiled and testing, my asterisk is not yet connected to PSTN or GSM |
| 00:07.14 | *** join/#asterisk Suckysucky (i=Borgon@70-100-55-174.dsl1.tbr.ga.frontiernet.net) |
| 00:07.33 | Suckysucky | Nov 13 18:46:32 WARNING[235]: chan_sip.c:1900 create_addr: No such host: gw5.voicepulse.com/18004444444 ..... Can anyone tell me what causes this error? |
| 00:07.48 | bugz | wow i totally hacked my wifi driver setup |
| 00:07.50 | bugz | i hate think pads |
| 00:08.20 | justinu | hippie |
| 00:08.27 | bugz | i have a t43 with a fingerprint reader |
| 00:08.37 | bugz | im bout to get this thing integrated into xlock |
| 00:09.03 | Nugget | I have a laptop that works without having to spend hours hacking drivers. :) |
| 00:09.04 | bugz | Nugget: yeah yeah |
| 00:09.06 | bugz | haha |
| 00:09.11 | justinu | i have one too |
| 00:09.15 | bugz | not me dude |
| 00:09.22 | bugz | i have two laptops that both suck |
| 00:09.32 | bugz | one of which i have spent years perfecting a fbsd install on |
| 00:09.41 | bugz | even the dvd player and IR work on it |
| 00:09.48 | justinu | lol |
| 00:09.51 | justinu | "years" |
| 00:10.00 | bugz | imagine that, stuff works on it |
| 00:10.04 | bugz | its a sony kds |
| 00:10.07 | bugz | nasty little bugger |
| 00:10.17 | justinu | you are obviously a masochist |
| 00:10.18 | marcus2 | oh well, i think the next step is to replace asterisk@home with gentoo |
| 00:10.30 | asterboy | or lfs |
| 00:10.30 | bugz | so ive been told |
| 00:10.42 | Suckysucky | Anyone have recommendations on how to fix Nov 13 18:46:32 WARNING[235]: chan_sip.c:1900 create_addr: No such host: gw5.voicepulse.com/18004444444 |
| 00:11.15 | Nugget | can you resolve that hostname normally? |
| 00:11.30 | file[laptop] | what's your dial line look like? |
| 00:11.30 | asterboy | I'd check ny hosts file after I find out what ip its talkin about. |
| 00:11.45 | bugz | speaking of masochism, im working with some friends to put gentoo and * on a via embedded system |
| 00:11.53 | marcus2 | eh, screw lfs |
| 00:12.07 | asterboy | ya thats what I say about gentoo. |
| 00:12.20 | marcus2 | under openwrt, tho, not gentoo |
| 00:12.29 | bugz | marcus2: what hardware platform? |
| 00:12.32 | h3x | is that openwrt+asterisk stuff stable now |
| 00:12.38 | marcus2 | linksys wrt54gs |
| 00:12.44 | marcus2 | stable enough for home use |
| 00:12.44 | bugz | oh lol |
| 00:12.45 | Suckysucky | exten => _1NXXNXXXXXX,2,Dial(SIP/hkM33nQI:vSn72dad@gw5.voicepulse.com/${EXTEN}) |
| 00:12.55 | bugz | how do you fit a PRI card in that |
| 00:12.59 | file[laptop] | yeah that's not right for SIP |
| 00:13.00 | marcus2 | sweet, now i can make all those international calls i've been needing to make |
| 00:13.01 | Nugget | I hope you changed that password. |
| 00:13.02 | justinu | hey thanks for the voicepulse account! |
| 00:13.08 | marcus2 | heheh |
| 00:13.10 | Suckysucky | file[laptop]: how would it look like |
| 00:13.11 | file[laptop] | and hahaha |
| 00:13.15 | Suckysucky | justinu: haha nope =) |
| 00:13.21 | Suckysucky | i removed some of the password haha |
| 00:13.22 | marcus2 | bugz; the linksys is just for home, the PRI is at the office |
| 00:13.24 | file[laptop] | you would setup an entry in sip.conf as a peer with that host, username, and password |
| 00:13.30 | file[laptop] | and then do SIP/${EXTEN}@voicepulse |
| 00:13.36 | file[laptop] | where voicepulse is the name of the peer entry |
| 00:13.45 | file[laptop] | probably also need to set fromuser to your username |
| 00:13.49 | bugz | what you dont have 24 T1's in your house? get outta here.. |
| 00:13.52 | justinu | warning: voicepulse's sip support SUCKS |
| 00:13.56 | Suckysucky | k thanks, ill try it |
| 00:14.09 | marcus2 | if i had T1s at my house, i'd be using more than just a linksys for asterisk :) |
| 00:14.13 | asterboy | what does voicepulse cost? |
| 00:14.14 | Suckysucky | justinu: lol yes.. i got iax workign fine but i see nothing in their docs about sip |
| 00:14.19 | *** join/#asterisk jbl (n=jbl@ool-18bf1e0a.dyn.optonline.net) |
| 00:14.22 | h3x | i used to have a PRI at home |
| 00:14.22 | h3x | heh |
| 00:14.27 | justinu | sucky: it works, but not correctly. |
| 00:19.03 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 00:20.56 | asterboy | http://www.voice-plus.com/onenumber.htm |
| 00:21.04 | asterboy | Is that the voiceplus your talkin about? |
| 00:21.38 | br00ksh1r3 | it's voicepulse |
| 00:21.48 | asterboy | doh.. |
| 00:22.02 | br00ksh1r3 | Suckysucky: is your dns working? |
| 00:22.12 | br00ksh1r3 | sounds like a dns problem the more i look at it |
| 00:23.17 | kink0 | is normal I get if I set SIP instead IAX2 this message: chan_sip.c:1947 create_addr: No such host: switch-3.asterlink.com/996 |
| 00:23.19 | kink0 | ? |
| 00:23.46 | kink0 | or means my SIP is not working ? |
| 00:24.03 | *** part/#asterisk bugz (n=bugz@cpe-24-27-67-66.houston.res.rr.com) |
| 00:25.48 | file[laptop] | br00ksh1r3: it's not... |
| 00:25.59 | file[laptop] | chan_sip doesn't parse like that... |
| 00:26.06 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 00:26.13 | file[laptop] | so it ends up trying to do a DNS lookup on the full string including /extension |
| 00:26.53 | Suckysucky | W00 TW00T |
| 00:26.54 | Suckysucky | W00T W00T |
| 00:26.56 | Suckysucky | IT WORKS IT WORKS |
| 00:27.06 | Suckysucky | thanks everyone |
| 00:27.46 | kink0 | this is what I get: |
| 00:27.47 | kink0 | Nov 14 01:19:15 WARNING[17826]: chan_sip.c:1947 create_addr: No such host: switch-1.ofllc.com/7070 |
| 00:27.48 | kink0 | Nov 14 01:19:15 NOTICE[17826]: app_dial.c:975 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 00:27.56 | file[laptop] | I just explained why |
| 00:28.00 | file[laptop] | I'm not repeating myself |
| 00:28.18 | br00ksh1r3 | :/ |
| 00:28.24 | Druken | read the damn error.. |
| 00:28.26 | coppice | file: but you will. everyone does here :-) |
| 00:28.27 | Druken | no such host... |
| 00:28.33 | br00ksh1r3 | anyone going to vegas? |
| 00:28.43 | file[laptop] | Mattttttttttt |
| 00:28.55 | br00ksh1r3 | hey file! |
| 00:29.03 | marcus2 | hrm |
| 00:29.04 | file[laptop] | br00ksh1r3: what madness are you up to today? |
| 00:29.11 | marcus2 | 464gb usable on my * server |
| 00:29.15 | marcus2 | wonder if that will be enough |
| 00:29.21 | kink0 | but runs fine if I use AIX2 instead SIP |
| 00:29.33 | br00ksh1r3 | me and twisted have to be up at 6am tomorrow |
| 00:29.40 | br00ksh1r3 | so we're not going to sleep tonight |
| 00:29.48 | br00ksh1r3 | going out drinking |
| 00:29.49 | br00ksh1r3 | :D |
| 00:30.05 | twisted | word |
| 00:30.08 | twisted | drunken flying++ |
| 00:30.14 | br00ksh1r3 | he wanted to rent a car tonight |
| 00:30.15 | mog_home | what you say brooks |
| 00:30.16 | file[laptop] | br00ksh1r3: ooh |
| 00:30.20 | twisted | no, i said i thought about it |
| 00:30.23 | twisted | i didn't want to |
| 00:30.31 | br00ksh1r3 | because it would have been cheaper to rent one than leave one there |
| 00:30.31 | br00ksh1r3 | lol |
| 00:30.52 | br00ksh1r3 | at the airport |
| 00:30.59 | Katty | :< |
| 00:31.01 | twisted | aww |
| 00:31.05 | twisted | poke-fu is fun |
| 00:31.08 | Katty | k |
| 00:31.17 | br00ksh1r3 | f-u |
| 00:31.37 | twisted | brookshire, look at what i was telling you in the other chan |
| 00:32.01 | br00ksh1r3 | they have wifi at bumpers? |
| 00:32.24 | twisted | *nods* |
| 00:32.28 | br00ksh1r3 | lame |
| 00:32.35 | br00ksh1r3 | i would get distracted |
| 00:32.39 | br00ksh1r3 | i need to finish this tonight |
| 00:32.46 | twisted | no you wouldn't |
| 00:32.50 | twisted | you'd have beer and a laptop |
| 00:32.55 | twisted | and a comfy sofa |
| 00:33.15 | twisted | and it's early, so i doubt you'd get distracted until at least 10:30-11ish |
| 00:33.44 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 00:35.11 | br00ksh1r3 | why don't you just go then.. |
| 00:35.15 | br00ksh1r3 | and i'll meet you up there |
| 00:35.22 | br00ksh1r3 | at 10:30ish |
| 00:35.23 | br00ksh1r3 | :D |
| 00:35.25 | twisted | hahaha |
| 00:38.17 | file | ooh warm |
| 00:38.40 | twisted | yeah, it's a heated blanket apparently |
| 00:38.44 | twisted | natural heat. |
| 00:40.26 | tzanger | nice |
| 00:40.40 | tzanger | I eat beans |
| 00:40.43 | tzanger | that provides natural heat |
| 00:41.02 | twisted | no |
| 00:41.06 | twisted | that provides natural gas |
| 00:41.08 | twisted | there's a difference |
| 00:41.27 | tzanger | a rather big blob |
| 00:42.54 | twisted | damnit |
| 00:42.56 | twisted | you broke my trigger |
| 00:43.19 | file | ugh, that was horrible |
| 00:43.26 | tzanger | yikes |
| 00:43.33 | tzanger | I didn't touch your trigger |
| 00:43.50 | twisted | SOMEONE did. |
| 00:44.02 | file | wasn't me! |
| 00:44.07 | file | Queue Shaggy song. |
| 00:44.17 | IronHelix | but she cought me in the shower! |
| 00:44.24 | file | wasn't me. |
| 00:44.24 | Druken | uhmm... i am assuming you three live close or something? |
| 00:44.32 | IronHelix | saw me bangin on the sofa |
| 00:44.46 | twisted | file lives in nova scotia or somewhere weird |
| 00:44.47 | file | I'd be scared if I did |
| 00:44.51 | file | New Brunswick |
| 00:44.53 | twisted | don't know where tzanger lives |
| 00:45.12 | mog_home | jerjer around? |
| 00:45.19 | Katty | over the river and through the woods, twisted |
| 00:45.20 | Druken | wuts wrong with nb? |
| 00:45.42 | twisted | Katty, he's grandma!? |
| 00:46.22 | Qwell | nope, daddy...he just still lives at home with mommy. :( |
| 00:46.23 | tzanger | I live in ontario |
| 00:46.29 | twisted | ahh |
| 00:46.33 | Katty | twisted: obviously. |
| 00:46.48 | twisted | Katty, ooh... perhaps he's the wolf in grandma's clothing. |
| 00:46.55 | Druken | tzanger: aren't you the chap who lives west of me? |
| 00:47.10 | tzanger | perhaps |
| 00:47.19 | twisted | i always loved that story. crossdressing wolves trying to eat jailbait. |
| 00:47.20 | tzanger | I'm just outside of a small town called Listowel |
| 00:47.24 | tzanger | twisted: hahaha |
| 00:47.40 | Druken | that's right |
| 00:47.56 | Druken | i remember now... we were talking about wireless shit... |
| 00:48.11 | tzanger | perhaps yes |
| 00:48.18 | *** join/#asterisk hhoffman (i=tor@c212-151-198-78.swipnet.se) |
| 00:48.53 | Katty | twisted: possibly |
| 00:49.00 | Katty | twisted: oh there's a thought |
| 00:49.06 | Katty | twisted: mister wolf is a crossdresser |
| 00:49.07 | hhoffman | hi, I'm just starting out and have a FXS card so I'm dialing into my PSTN line and asterisk is answering but everything is very choppy and staticy... |
| 00:49.14 | twisted | indeed |
| 00:49.19 | hhoffman | is there something I can do to stop this? |
| 00:49.19 | Katty | twisted: or a transvestite! |
| 00:49.25 | Katty | twisted: an /executive/ transvestite (= |
| 00:50.00 | twisted | muaahahahaha |
| 00:50.10 | kink0 | runs with SIP !! was I am nated !! |
| 00:50.27 | twisted | don't run with SIP, you could fall and poke your eyebeam out |
| 00:50.37 | Qwell | twisted: That was bad. Even for you. :p |
| 00:50.42 | twisted | Qwell, yea yeah |
| 00:50.57 | Dr_Ray | hhoffman , check your IRQs? |
| 00:51.01 | Druken | hehe |
| 00:51.15 | twisted | oh gnoes! |
| 00:51.51 | Katty | you've clearly all insaned. |
| 00:51.55 | twisted | my hair! it like, disappeared 1/8" of an inch from my scalp |
| 00:52.09 | Katty | next you shall core dump and kernel panic |
| 00:52.12 | Katty | with optional deadlocking |
| 00:52.42 | Katty | twisted: i make up for your lack of hair. |
| 00:52.55 | twisted | yes yes |
| 00:54.25 | hhoffman | Dr_Ray: how do i do that? |
| 00:54.34 | SwK | killall -9 twisted |
| 00:54.45 | file | SwK: Permission denied |
| 00:55.04 | hhoffman | <PROTECTED> |
| 00:55.04 | hhoffman | <PROTECTED> |
| 00:55.05 | hhoffman | <PROTECTED> |
| 00:55.05 | hhoffman | <PROTECTED> |
| 00:55.05 | hhoffman | <PROTECTED> |
| 00:55.07 | hhoffman | <PROTECTED> |
| 00:55.09 | hhoffman | <PROTECTED> |
| 00:55.10 | SwK | cut twisted && kill twisted && rm -rf twisted |
| 00:55.11 | hhoffman | <PROTECTED> |
| 00:55.14 | hhoffman | <PROTECTED> |
| 00:55.15 | hhoffman | <PROTECTED> |
| 00:55.18 | hhoffman | <PROTECTED> |
| 00:55.19 | *** kick/#asterisk [hhoffman!n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted] by twisted (flood detected) |
| 00:55.25 | file | hehe haha |
| 00:55.33 | *** join/#asterisk hhoffman (i=tor@c212-151-198-78.swipnet.se) |
| 00:55.46 | hhoffman | doh, sorry... guess pasting isn't allowed? |
| 00:55.46 | SwK | thanks hhoffman we just love getting flooded in irc |
| 00:55.51 | hhoffman | sorry |
| 00:55.52 | SwK | you think>? |
| 00:55.53 | hhoffman | :-( |
| 00:55.59 | twisted | pastebin.ca |
| 00:56.04 | SwK | pastebin.ca live it love it use it |
| 00:56.12 | hhoffman | k, will do from now on... sorry all |
| 00:56.31 | *** join/#asterisk implicit (n=implicit@ip68-7-154-222.sd.sd.cox.net) |
| 00:56.33 | twisted | that's how. |
| 00:56.44 | file | noooooooo |
| 00:57.42 | Qwell | I bet he would fit in bind fairly easily |
| 00:57.43 | hhoffman | anyone help with the static sounds? |
| 00:57.44 | twisted | SwK knows what i'm talking about |
| 00:58.59 | twisted | PING |
| 00:59.06 | rajiv | interesting that hhoffman is using PIC and not APIC |
| 00:59.34 | Druken | who wants to buy me a smartboard for my payphone? |
| 00:59.44 | file | Druken: I don't. |
| 00:59.47 | Dr_Ray | druken - I might have one for you |
| 01:00.01 | Dr_Ray | you have to buy your own relay and hopper for it |
| 01:00.08 | hhoffman | it's a older PII system |
| 01:00.12 | SwK | heh |
| 01:00.20 | Druken | Dr_Ray: got the relay and hopper... |
| 01:00.32 | Dr_Ray | it has to be matching voltage for the board |
| 01:00.57 | Druken | really.... hmm... |
| 01:00.58 | Qwell | hmm |
| 01:01.07 | Druken | how do i know what volage the relay is? |
| 01:01.08 | Evanrude | heya... i have a quick opinion question |
| 01:01.09 | Qwell | If you play a sound, and you get static... |
| 01:01.15 | Qwell | will it sound exactly the same next time you play it? |
| 01:01.17 | konfuzed | hhhhmmmm id like to check in on the real need for ram |
| 01:01.24 | Qwell | If not, I propose we start calling it dynamic. |
| 01:01.45 | Evanrude | which, if any, are the best operator consoles (for monitoring extensions, transferring calls, etc).. |
| 01:02.02 | konfuzed | will 256 - 133 at 7.5 be sufficient for a debian box running asterisk no X or gui |
| 01:02.06 | Druken | i use Flash Operator Panel |
| 01:02.13 | Dr_Ray | I use asterisk -r |
| 01:02.20 | konfuzed | is there a real need to make it 512 mb ram |
| 01:02.21 | Dr_Ray | fop stinks |
| 01:02.24 | Evanrude | Druken: is that the one that comes with AMP |
| 01:02.26 | Druken | Dr_Ray: i use that too... |
| 01:02.42 | Druken | Evanrude: probably |
| 01:02.46 | konfuzed | is there a page on RAM needs some where |
| 01:03.05 | IronHelix | asterisk+dimensioning |
| 01:03.07 | IronHelix | on voip info |
| 01:03.09 | IronHelix | as i recall |
| 01:03.27 | twisted | ram is cheap |
| 01:03.32 | Evanrude | Druken: I'll check it out... last time I think I used it it wasn't exactly user friendly |
| 01:03.34 | Druken | Dr_Ray: how do i know the voltage of the relay ? |
| 01:03.44 | twisted | just maximize your capacity |
| 01:03.50 | Nugget | stick your tongue on it |
| 01:03.59 | twisted | Druken, cross ref the part number |
| 01:04.07 | Dr_Ray | do a google search for ernest relay payphone and then use a voltmeter to test yours |
| 01:04.18 | *** join/#asterisk wmandra (n=wmandra@bgp504084bgs.verona01.nj.comcast.net) |
| 01:04.54 | wmandra | whats goin on all..... has anyone tried updating asterisk via cvs today?? |
| 01:04.55 | Druken | hmm.... so in otherwords... who knows.. ok :) |
| 01:05.10 | mog_home | there were things going on with mirrors |
| 01:05.12 | mog_home | go again |
| 01:06.39 | Qwell | omg, its the mog |
| 01:06.49 | Qwell | hmm |
| 01:06.57 | Qwell | same letters |
| 01:07.08 | twisted | hah |
| 01:07.23 | *** join/#asterisk SwK (n=SwK@dpc6745230018.direcpc.com) |
| 01:07.36 | Qwell | So whats up in Vegas exactly? |
| 01:07.37 | Nugget | what flight? |
| 01:07.49 | mog_home | yeah why do you think im mog |
| 01:10.02 | konfuzed | mog_home: <-------- starts with mog |
| 01:10.12 | konfuzed | 8^) |
| 01:10.23 | file | he's an Ogorman |
| 01:10.28 | mog_home | the ogorman |
| 01:10.34 | konfuzed | what makes you think im konfuzed anyway |
| 01:10.40 | tzanger | mike, montgomery... |
| 01:10.41 | mog_home | from the land of cork |
| 01:10.46 | IronHelix | <-- isnt ironhelix!! |
| 01:10.49 | konfuzed | :^) |
| 01:12.25 | mog_home | erm? |
| 01:13.04 | konfuzed | darn this AMD 1300 i have only has 256 mb ram |
| 01:13.26 | konfuzed | but its lesser quality ram than the 256 in my amd 600 |
| 01:15.32 | konfuzed | damn these other two ram stick dont say how much ram but the chips indicate 10ns |
| 01:16.34 | konfuzed | alright I'll take it as a blessing that this 256 mb 7.5ns happens to match and make this box a 512 mb athlon 600 |
| 01:17.49 | *** join/#asterisk kn0x (n=nunya@tor/session/x-bd9201179476f930) |
| 01:19.07 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 01:20.31 | konfuzed | ok so next |
| 01:21.18 | konfuzed | voipinfo says a lot about dialup but I did not see any thing about using a dial up modem in potential telephony vs dialup to internet connection |
| 01:21.29 | konfuzed | fax modem |
| 01:21.36 | konfuzed | any use I should I pull it out |
| 01:21.53 | konfuzed | any use to keep it, or should I pull it out |
| 01:22.13 | vira | are there any linux softphones w/ esd support? |
| 01:24.25 | Druken | has anyone actually used adsi with asterisk ? |
| 01:24.29 | tzanger | yes |
| 01:24.34 | tzanger | simply |
| 01:24.36 | tzanger | but yes |
| 01:24.56 | mog_home | yes |
| 01:25.00 | twisted | oh fuck a duck |
| 01:25.16 | Druken | tzanger: does it work over ata's or just FXS channels connected directly? |
| 01:25.26 | twisted | i have to check-in in person because they have overbooked the flights. |
| 01:25.27 | tzanger | it'll work over any voice channel |
| 01:25.38 | tzanger | A guy walks into a doctor's office with a duck on his head |
| 01:25.42 | tzanger | they doc says "Can I help you?" |
| 01:25.50 | tzanger | the duck says "yeah... get this guy off my ass." |
| 01:26.13 | Druken | what a quack |
| 01:26.20 | twisted | omg |
| 01:26.30 | twisted | that was bad. |
| 01:26.37 | tzanger | it's one of my favourite jokes |
| 01:26.39 | twisted | made me physically groan |
| 01:27.35 | marcus2 | thats pretty bad |
| 01:27.55 | twisted | you know what's worse? |
| 01:28.10 | marcus2 | i'm not sure i want to |
| 01:28.28 | twisted | the duck had avion flu |
| 01:28.59 | tzanger | I'm not sure if I get that joke |
| 01:29.06 | tzanger | if 'avion' is misspelled on purpose or not |
| 01:29.09 | konfuzed | the flu took a plane to get to the duck? |
| 01:29.22 | twisted | tzanger, no, it's not on purpose. |
| 01:29.30 | twisted | no |
| 01:29.56 | twisted | you can only get avian flu if you are exposed phsically to bird poop |
| 01:30.03 | twisted | *physically |
| 01:30.05 | konfuzed | image the flu flying on a french plane to chase down a duck |
| 01:30.10 | tzanger | twisted: you sure about that? |
| 01:30.26 | tzanger | H5N1 is airborne amongst birds |
| 01:30.27 | twisted | tzanger, according to the CDC/WHO, yea |
| 01:30.31 | bsdfreak | ahahahh |
| 01:30.41 | tzanger | heh that *is* bad isn't it |
| 01:31.02 | tzanger | I'll make myself sit in the corner for that |
| 01:32.09 | konfuzed | according to the head of the CDC he was told in may 2001 by a us general that a terrosit attack was going to take down the towers |
| 01:32.23 | konfuzed | my only point being to not believe everything the WHO tells you |
| 01:32.58 | tzanger | but they had such great hits |
| 01:33.36 | twisted | uhh |
| 01:33.47 | konfuzed | he also said the Avian Flu was GMOd at us military bioweapons labs |
| 01:33.52 | twisted | CDC/WHO has nothing to do with the terrorist attacks.. that would be the FBI/CIA/NSA |
| 01:34.09 | konfuzed | us General told the head of the CDC that it would happen |
| 01:34.21 | tzanger | GMO'd? |
| 01:34.24 | konfuzed | s/us/U.S./ |
| 01:34.26 | tzanger | I'm drowning in TLA soup here |
| 01:34.35 | konfuzed | Genetically Modified Organism |
| 01:34.48 | konfuzed | custom military organism |
| 01:34.58 | rajiv | i have a zap channel and a sip phone on my desk. the pots line has call waiting from my CLEC. how so i send a hook flash on the sip phone so that i can answer the call waiting ? |
| 01:35.00 | twisted | well yeah, it's a highly effective killer |
| 01:35.18 | tzanger | we have equipment in the Canadian equivalent of the CDC's highest biohazard building |
| 01:35.22 | tzanger | really cool actually |
| 01:35.27 | tzanger | how they seal everything up |
| 01:35.37 | tzanger | the buildings themselves aren't special |
| 01:35.44 | tzanger | just the actual level 'x' rooms |
| 01:35.55 | tzanger | concrete block rooms |
| 01:35.56 | justinu | rajiv, i don't think it works like that |
| 01:36.06 | tzanger | network cable, everything is "unwoven" and run through conduit |
| 01:36.08 | konfuzed | Head of CDC says its not if but when the break out really hits that a world wide pendemic will be recognized |
| 01:36.15 | tzanger | then the conduit is filled with epoxy |
| 01:36.27 | rajiv | justinu: i guess it doesn't make sense to connect a pots line with call waiting to * |
| 01:36.33 | tzanger | the room is painted with a kind of epoxy paint to seal everything |
| 01:36.37 | justinu | rajiv: no, i think you can do that |
| 01:36.43 | Druken | if we ever get wifi phones working right, can we still charge like 10-40 cents a minute?? :) |
| 01:37.09 | justinu | charge whatever people will pay |
| 01:37.28 | file[laptop] | twisted: http://www.midsouthmarketplace.com/~krice/gallery/view_photo.php?set_albumName=album04&id=IMG_4619 |
| 01:37.31 | tzanger | and of course negative pressure on everything to force all air through the scrubbers |
| 01:37.41 | twisted | file[laptop], ? |
| 01:38.02 | justinu | butter against violence? |
| 01:38.06 | tzanger | wtf is it with you people and muffins? |
| 01:38.13 | konfuzed | tzanger: it was probably the secret testing on open territories to rate the human impact that led to the accidental release into the public bird population |
| 01:39.00 | konfuzed | certain not those Canadian clean rooms |
| 01:39.01 | tzanger | I dunno |
| 01:39.05 | konfuzed | ;^) |
| 01:39.11 | tzanger | that's awfully conspiracy theorist |
| 01:39.23 | konfuzed | not if you talk to the head of the CDC |
| 01:39.30 | konfuzed | thats the scary part |
| 01:39.47 | tzanger | konfuzed: got a press release with that? |
| 01:39.49 | konfuzed | any way somebody else brought it up but i wont say any more on that one |
| 01:39.51 | tzanger | or is it FOAF type stuff |
| 01:40.12 | *** join/#asterisk lo2 (n=lo2@ti112210a080-2163.bb.online.no) |
| 01:40.23 | konfuzed | tzanger: I can get you an audio recoding of an interview with the head of the CDC |
| 01:40.28 | tzanger | nice |
| 01:40.51 | konfuzed | but I said I wont say any more so may be some other time |
| 01:40.51 | tzanger | I'd actually like to hear that (not to mistrust you but I am geniuinely interested in hearing what else he said) |
| 01:46.01 | *** join/#asterisk tainted_ (n=identd@adsl-71-129-43-66.dsl.irvnca.pacbell.net) |
| 01:46.10 | tainted_ | ~seen |nix |
| 01:46.14 | jbot | |nix <n=inix@cm240.gamma120.maxonline.com.sg> was last seen on IRC in channel #asterisk, 94d 10h 49m 46s ago, saying: 'any ideas?'. |
| 01:49.17 | konfuzed | ~eeks |
| 01:49.19 | jbot | hmm... eeks is the Eeks eeks run for the hills IAX2 is here to stay |
| 01:49.34 | konfuzed | ;^) |
| 01:50.47 | konfuzed | so is there actually any use for a PCI fax/modem or should it be pulled out just to be on the safe side |
| 01:50.53 | konfuzed | ;^) |
| 01:52.06 | *** join/#asterisk lenne_dk (n=mirc@83.72.129.7.ip.tele2adsl.dk) |
| 01:52.11 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 01:53.37 | lenne_dk | Hello, group :-) I need to know why an incoming call begins at a certain entry in the dialplan. |
| 01:53.48 | lenne_dk | Anyone here? |
| 01:54.25 | lunk | if i have 100 calls to make and n outbound connections, what's the best algorithm to keep as many outbound lines as busy as possible |
| 01:54.41 | justinu | lol |
| 01:54.50 | lunk | this is for reminders for appointments |
| 01:55.29 | konfuzed | lenne_dk: at best some one is virtually here |
| 01:55.56 | lenne_dk | Is that you, Eliza? |
| 01:56.07 | justinu | lol |
| 01:56.29 | konfuzed | lenne_dk: no but perhaps the entry in the dial plan has an 'a' after the number |
| 01:56.49 | lenne_dk | And what might that mean? |
| 01:57.01 | justinu | Nov 13 17:58:28 WARNING[29835]: interface.c:215 decodeMP3: Junk at the beginning of frame 00000000 |
| 01:57.03 | justinu | ?? |
| 01:57.12 | konfuzed | lenne_dk: sorry Im only virtually helpless |
| 01:57.34 | lenne_dk | And I'm clueless :-) |
| 01:57.46 | konfuzed | well formed statement |
| 01:57.50 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 01:58.03 | deezed | justinu why not open it up in foobar2000 and either fix the mp3 header, or renencode it |
| 01:58.15 | konfuzed | lenne_dk: you dont present enough info for a valid question |
| 01:58.33 | justinu | deezed: does that mean the mp3's that came with asterisk are teh suck? |
| 01:58.50 | lenne_dk | I have 4 numbers registered. two goes directly to extensions, one goes to menu, and the last is supposed to go to menu too, but goes to extension instead. |
| 01:58.55 | konfuzed | wheres that - asking good questions - link when ya need it |
| 01:59.07 | deezed | odd.. heh i doubt it. never used them though |
| 01:59.33 | lenne_dk | I'm not finished... |
| 02:00.29 | brettnem | hey all |
| 02:00.42 | lenne_dk | I really just want to turn on some debugging, so I can see "incoming call from ... to sip:123@my.host" |
| 02:00.58 | brettnem | hey anyone know if app_devstate will work with Asterisk 1.2?? |
| 02:01.04 | lenne_dk | And content/extension/priority |
| 02:01.07 | marcus2 | did you try 'sip debug' ? |
| 02:01.41 | deezed | whats the most calls has anyone taken with nufone iax at the same time? |
| 02:02.31 | lenne_dk | Much too verbose, and as I have 4 numbers registered, I have much irrelevant data. And it doesn't show the context/extension/priority |
| 02:03.50 | brettnem | anyone using devstate? |
| 02:04.42 | lenne_dk | I might try adding some debug to the source, but I'm not sure in which part I'll have to start. chan_sip.c? Or in a common module for all channels? |
| 02:04.56 | brettnem | hmm guess now |
| 02:04.58 | brettnem | er not |
| 02:07.32 | *** join/#asterisk kris2k (n=kris@lnx001.nat.hst.tmcsolutions.net) |
| 02:07.52 | lenne_dk | Is building HEAD messed up, or is it me? For the last hour or so make has just looped, making version.h.tmp over and over |
| 02:07.52 | *** join/#asterisk PBXtech (i=nik@164.sub-70-213-172.myvzw.com) |
| 02:08.16 | marcus2 | hahah theres a chan_bluetooth |
| 02:09.29 | Druken | Dr_Ray: you still around? |
| 02:11.12 | *** join/#asterisk asteriskgeeks (n=SIPdawg@64.5.53.45) |
| 02:11.13 | asteriskgeeks | <PROTECTED> |
| 02:11.19 | justinu | marcus2: you get your pbx up? |
| 02:15.27 | MikeJ[Laptop] | <PROTECTED> |
| 02:16.03 | marcus2 | justin; no :/ |
| 02:16.09 | marcus2 | i'm reinstalling the asterisk server tho |
| 02:16.14 | marcus2 | replacing a@h with gentoo |
| 02:16.24 | MikeJ[Laptop] | mog_home, are you a bot? |
| 02:16.28 | justinu | use 1.2-rc2 |
| 02:16.31 | marcus2 | since i needed to do that anyway, and asterisk was being wierd on a@h |
| 02:16.36 | marcus2 | the pri debugging was junk |
| 02:16.46 | mog_home | mog_home is a tx1000 with a neural net proccessor |
| 02:16.48 | mog_home | a learning computer |
| 02:16.59 | MikeJ[Laptop] | :D |
| 02:17.08 | Dr_Ray | Druken - yes |
| 02:17.09 | justinu | thinking machines cm-5 |
| 02:17.49 | deezed | sorry this problem is making me go insane.. how do i check the debug ---> res_config_mysql.c:495 reload: MySQL RealTime: Couldn't establish connection. Check debug. |
| 02:17.57 | MikeJ[Laptop] | does matt riddle really think it is likely that osip and exosip would be disclaimed? |
| 02:20.05 | marcus2 | yeah justin, i'm definately going to put 1.2 on it |
| 02:20.32 | MikeJ[Laptop] | deezed, set debug to somthing... and make sure in logger.conf that debug is going to console or file or somewhere you can see it.. |
| 02:20.37 | marcus2 | i need to find someone that knows how to program the stupid magix pbx, tho |
| 02:20.42 | marcus2 | i'm even willing to pay |
| 02:20.51 | MikeJ[Laptop] | stupid magix ? |
| 02:20.54 | MikeJ[Laptop] | heh |
| 02:20.58 | deezed | thanks mike |
| 02:21.04 | MikeJ[Laptop] | np |
| 02:21.45 | asterboy | Can I have 4 fxo ports route to 2 fxs ports AND give me distinctive ring on for the missing 2 fxs ports? |
| 02:22.14 | justinu | going crazy with this on hold music noise |
| 02:22.31 | asterboy | From what I've read so far it's just a matter of creating the dial plan with the right options. |
| 02:22.53 | asterboy | noice is usually a bad contact, no? |
| 02:23.03 | justinu | lol |
| 02:23.25 | asterboy | Just want to confirm distinctive ring is easily setup in my case. |
| 02:24.21 | marcus2 | fucking emerge |
| 02:24.26 | asterboy | or 2 fxo, (pstn lines) ----> 1 fxs, (telephone) using distinctive ring to distinguish. |
| 02:24.38 | asterboy | gentoo...lol, thats why I went to lfs. |
| 02:24.43 | lenne_dk | http://www.marko.net/asterisk/archives/0205/0140.html |
| 02:25.01 | lenne_dk | exten => 1,1,Dial,Zap/28 ; Ring Zap/28 normally |
| 02:25.01 | lenne_dk | exten => 2,1,Dial,Zap/28r1 ; Ring Zap/28 with ring #1 |
| 02:25.01 | lenne_dk | exten => 3,1,Dial,Zap/28r2 ; Ring Zap/28 with ring #2 |
| 02:26.12 | lenne_dk | SIP doesn't support distinctive ring, I believe |
| 02:26.53 | justinu | it does |
| 02:27.22 | marcus2 | yeah, my sip devices have configuration settings for distinctive ring |
| 02:27.23 | *** join/#asterisk gpx1000 (n=gpx1000@12.196.68.26) |
| 02:27.26 | *** join/#asterisk PBXtech (i=nik@164.sub-70-213-172.myvzw.com) |
| 02:27.33 | gpx1000 | <PROTECTED> |
| 02:27.45 | gpx1000 | when it tries to link chan_alsa.o |
| 02:29.14 | Lostfrog | _ALERT_INFO= ? |
| 02:31.19 | gpx1000 | Anyone know what I'm missing? |
| 02:32.37 | ptiggerdine | have you got all the alsa-devel pkgs |
| 02:32.49 | gpx1000 | ptiggerdine: yes... |
| 02:33.32 | ptiggerdine | then you might need to look at mantis |
| 02:33.38 | lenne_dk | How do you dial that? WARNING[614]: chan_sip.c:1947 create_addr: No such host: 36949608r1 |
| 02:34.03 | justinu | what is that? |
| 02:34.21 | lenne_dk | Tried exten => 661,1,dial,SIP/36949608r1 |
| 02:34.35 | justinu | what kind of number is that? |
| 02:35.18 | lenne_dk | I can call my desktop phone at dial,SIP/36949608 |
| 02:35.30 | konfuzed | heck it out - this amd 600 i got seems to have an fm tuner on the motherboard |
| 02:35.48 | konfuzed | play the radio to the music on hold |
| 02:35.51 | konfuzed | ;^) |
| 02:35.55 | justinu | heh |
| 02:36.07 | konfuzed | gotta try that one |
| 02:36.13 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 02:36.14 | lenne_dk | So How do I do distinctive ring on SIP? |
| 02:36.15 | asterboy | lenne_dk: thanks for the distinctive ring config!!! |
| 02:36.41 | justinu | lenne_dk: it varies from device to device |
| 02:37.44 | lenne_dk | So that what "There is no standard for distinctive ring on SIP" means. I've got two grandstreams and a Fritz!FON box |
| 02:37.54 | justinu | yeah |
| 02:38.02 | justinu | grandstream can do it based on callerid |
| 02:38.07 | justinu | and by registration |
| 02:38.12 | justinu | and possibly a few other ways |
| 02:38.17 | justinu | you gotta look at the manual |
| 02:38.44 | asterboy | Can DID numbers come from the Internet? |
| 02:38.59 | justinu | yes |
| 02:39.07 | lenne_dk | What do you mean by distinctive ring by registration? |
| 02:39.12 | asterboy | Is it a cheaper way to get a POTS line replacement? |
| 02:39.17 | justinu | yes |
| 02:39.37 | asterboy | Who is selling those in Canada? |
| 02:39.38 | Lostfrog | There could be 911 issues, ast_freak. |
| 02:39.42 | Lostfrog | asterboy, even. |
| 02:39.51 | justinu | lenne_dk: the phone registers one 4 "line appearances", and depending on which one is called, it rings differently. |
| 02:39.54 | tzanger | asterboy: lots of people providing Canadian DIDs |
| 02:40.00 | justinu | s/one/on |
| 02:40.38 | asterboy | So my High Speed internet plugs to my *box and ouput via IAX or FXS? |
| 02:40.39 | Lostfrog | justinu: I am reluctant to have multiple registrations per phone.. that could get messy quickly. |
| 02:40.49 | lenne_dk | That must be the new "Office" grandstream, not the BT101 or HT286 |
| 02:40.56 | tzanger | asterboy: or SIP or h323 or OSS/Alsa, yes |
| 02:41.02 | justinu | Lostfrog: yeah, there are other better ways of doing it |
| 02:41.07 | asterboy | excellent...that has to be the best way to do it. |
| 02:41.17 | Lostfrog | I haven't found it for the snom320/360 |
| 02:41.21 | asterboy | instead of getting vonage etc. |
| 02:41.23 | Lostfrog | I can't even get a wav to play. |
| 02:41.37 | justinu | Lostfrog: i have a snom360, i'll help you get it working |
| 02:41.41 | gpx1000 | yep, alsa was broken... it's making fine now... |
| 02:41.45 | asterboy | so "Even" is one of the providers? |
| 02:42.18 | tzanger | who's Even? |
| 02:42.19 | lenne_dk | Somebody build HEAD recently? My build loops, it didn't use to. |
| 02:42.32 | asterboy | Lostfrog · asterboy, even. |
| 02:42.41 | asterboy | missread that no doubt. |
| 02:43.16 | Lostfrog | No.. I meant the message I typed to ast_freak to be to you, asterboy. |
| 02:43.17 | justinu | Lostfrog: the snom 360 is pretty slick actually |
| 02:44.00 | asterboy | oh, lol...ya we need another "aster" something in the list :P |
| 02:45.03 | asterboy | hmmm...link2voip.com sells them for $4.50usd/month |
| 02:45.13 | asterboy | anyone know the best price? |
| 02:45.18 | justinu | that's not bad |
| 02:45.28 | justinu | is that for usa too? |
| 02:45.59 | asterboy | don't think so. |
| 02:47.11 | *** join/#asterisk devonst17 (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net) |
| 02:47.29 | gpx1000 | lenne_dk: I'm building it now |
| 02:47.57 | kn0x | can someone take a look at my sip.conf |
| 02:48.01 | kn0x | fwd wont register |
| 02:48.03 | jebba | is there anyone here that has experience with asterisk and a sipura 2002? I can't get it to register no matter how much i kick it. |
| 02:48.05 | kn0x | http://pastebin.ca/28623 |
| 02:48.57 | MacRohard | anyone know a sip provider that offers service in area code 902 (nova scotia) ? |
| 02:49.05 | MacRohard | inbound did i mean. |
| 02:49.59 | Katty | time for random question |
| 02:50.05 | Katty | what is your /favorite/ holiday cookie? |
| 02:50.29 | file | I like shortbread cookies |
| 02:50.45 | Katty | file: does your mother make them? |
| 02:50.50 | lenne_dk | Please be serious and on topic |
| 02:50.50 | file | no |
| 02:50.58 | Katty | lenne_dk: i'm quite serious, thank you |
| 02:51.03 | Katty | file: are they store bought? |
| 02:51.09 | file | Katty: yes |
| 02:51.13 | Katty | file: :< |
| 02:51.16 | lenne_dk | Then you are just in the wrong conference |
| 02:51.20 | Katty | haha. |
| 02:51.25 | file | 1. This is an IRC channel |
| 02:51.29 | Katty | twisted: what is favorite christmas cookie? |
| 02:51.40 | file | 2. I've been here for a long time :P |
| 02:51.40 | lenne_dk | Wrong channel then |
| 02:51.48 | Katty | lenne_dk: lies. |
| 02:51.54 | file | and random stuff like that isn't too bad |
| 02:51.57 | Katty | lenne_dk: you're clearly missinformed. |
| 02:52.07 | lenne_dk | My asterisk doesn't make cookies |
| 02:52.14 | Katty | lenne_dk: then you're missing modules. |
| 02:52.42 | kn0x | could someone look at http://pastebin.ca/28623 asterisk isnt registering to fwd properly |
| 02:52.57 | twisted | lenne_dk, get over it. |
| 02:53.02 | twisted | Katty, gingerbread ;) |
| 02:53.06 | Katty | file: what brand of shortbread cookies do you eat? |
| 02:53.11 | Katty | twisted: hrmm. |
| 02:53.15 | Katty | twisted: do you have recipe? |
| 02:53.22 | file | Katty: something from the local grocery |
| 02:53.29 | Katty | figures. |
| 02:53.30 | twisted | Katty, no :( |
| 02:53.36 | Katty | file: attention to details next time! |
| 02:53.41 | Katty | file: kthx |
| 02:53.45 | Katty | twisted: does your mother have it? |
| 02:53.50 | twisted | Katty, possibly |
| 02:53.55 | twisted | i'm sure my grandmother does |
| 02:53.57 | Katty | twisted: bribe her. |
| 02:53.58 | Katty | twisted: i want recipe. |
| 02:55.40 | twisted | haha |
| 02:55.42 | lenne_dk | Wjy don't yoi make cookies in #asterisk-devel instead? Nobody's there |
| 02:55.47 | twisted | you're enjoying it, aren't you |
| 02:55.55 | Katty | i think lenne_dk has insaned. |
| 02:55.56 | twisted | lenne_dk, why don't you stop complaining and ask a question if you have one. |
| 02:56.30 | lenne_dk | Somebody build HEAD recently? My build loops, it didn't use to. |
| 02:56.38 | asterboy | Are there other products that do the same thing as digium's fxs and fxo? |
| 02:56.44 | twisted | if your build loops, make clean, or move the tree |
| 02:56.50 | Katty | asterboy: sangoma (= |
| 02:56.55 | twisted | i built the tree just today and it built fine |
| 02:57.08 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
| 02:57.09 | Katty | ..! |
| 02:57.11 | Katty | that's it! |
| 02:57.17 | Katty | i shall make a sangoma cookie recipe! |
| 02:57.25 | Katty | file: find joey. |
| 02:57.48 | file | Katty: otay |
| 02:57.53 | Katty | i must figure out what defines all things sangoma and dreamy |
| 02:58.00 | file | he's not online sadly enough |
| 02:58.04 | Katty | :< |
| 02:58.10 | file | he'll be online tomorrow when he's @ work |
| 02:58.11 | file | :p |
| 02:58.19 | Katty | k, that's soon enough |
| 02:59.15 | asterboy | I'm guessing DID via HighSpeed is cheaper than ISDN correct? |
| 02:59.26 | lenne_dk | Make clean didn't help. A new tree didn't loop. At least not yet |
| 02:59.39 | asterboy | T1/E1 equipment is expensive, plus you need to pay for those line costs. |
| 02:59.51 | asterboy | But probably better for big ops? |
| 03:00.40 | asterboy | Could not fint the Sangoma cards...any part numbers? |
| 03:01.15 | Katty | asterboy: i think voip-supply.com has sangoma cards |
| 03:01.32 | asterboy | ya, just found them...looking. |
| 03:01.48 | Corydon76-home | There's fewer things that you can find to annoy Mark than to discuss Sangoma in here |
| 03:02.05 | Katty | he's a big boy |
| 03:02.26 | *** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 03:02.29 | Katty | i'm sure he can handle it (= |
| 03:02.33 | asterboy | sangoma has only t1/e1 stuff. |
| 03:02.38 | Corydon76-home | Keep in mind that the less Digium and the more Sangoma, the less Asterisk development can get done |
| 03:02.47 | wmandra | hey guys anyone have any tips for getting * cvshead to hear dtmf tones from a cisco 7960?? |
| 03:02.58 | Katty | asterboy: the analog cards are still in the making, i do believe. |
| 03:03.03 | Corydon76-home | The purpose of Digium is to help finance Asterisk development |
| 03:03.21 | asterboy | In Canada it seems they want a lot for the Digium cards...over $100 per port. |
| 03:03.28 | mog_home | and keep mog_home off the street |
| 03:03.30 | kn0x | can someone help me with my sip.conf? |
| 03:03.34 | kn0x | http://pastebin.ca/28623 |
| 03:03.39 | asterboy | Which is cheap considering what it use to cost for propietary stuff. |
| 03:03.40 | kn0x | fwd wont register |
| 03:03.44 | lenne_dk | Seems ok to me, kn0x |
| 03:04.11 | kn0x | well heres the situation, asterisk is behind nat with an ip of 192.168.0.10 |
| 03:04.15 | asterboy | what is the lowest price found for a digium card? say the tdm40b? |
| 03:04.18 | kn0x | i have dyndns running |
| 03:04.26 | asterboy | I have found $296 |
| 03:04.29 | Lostfrog | justinu: I like the snoms so far. I would have responded sooner, but, Adelphia cable sucks. |
| 03:04.31 | kn0x | its being updated on the router because i have a dyn ip |
| 03:04.52 | mog_home | lowest is tdm10b |
| 03:04.56 | kn0x | i have ports 5060, and 10000-20000 forwareded to the box |
| 03:05.08 | Lostfrog | UDP, kn0x? |
| 03:05.12 | asterboy | yes lower, but I mean to compare apples to apples. |
| 03:05.34 | asterboy | Say a loaded TDM FXS or FXO |
| 03:05.52 | asterboy | Think FXS is cheaper to produce. |
| 03:06.03 | kn0x | lostfrog- yes |
| 03:06.08 | kn0x | bosth tcp/udp |
| 03:06.11 | kn0x | http://pastebin.ca/28624 |
| 03:06.17 | kn0x | theres the sip headers from the debug |
| 03:07.38 | asterboy | At about $100 per port for the FXS, you might as well put that money towards an IAX/SIP capable phone. |
| 03:07.41 | kn0x | any ideas? |
| 03:07.48 | asterboy | Anyone know the best phones for the price??? |
| 03:07.54 | kn0x | Nov 13 15:07:47 WARNING[2398]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x819e9b8 (len 397) to 69.90.155.70:-1 returned 5060: Bad file descriptor |
| 03:08.03 | kn0x | i keep getting this on the cli... |
| 03:08.08 | kn0x | whats that about |
| 03:08.40 | kn0x | im running the latest CVS HEAD as of a week ago |
| 03:08.47 | asterboy | grandstream still the best bang for buck? |
| 03:09.02 | justinu | just do yourself a favor and get a snom 360 or something |
| 03:09.08 | lenne_dk | Have you connected to FWD with a real phone instead of asterisk sometime? |
| 03:09.25 | wmandra | hey guys * is not hearing dtmf tones from my 7960, anyone have any ideas? |
| 03:09.54 | *** part/#asterisk WillySilly (n=WillySil@c-24-23-145-194.hsd1.ca.comcast.net) |
| 03:10.09 | Qwell | wmandra: dtmfmode |
| 03:10.10 | justinu | wmandra: set dtmfmode=rfc2833 |
| 03:10.32 | *** part/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 03:10.35 | asterboy | ooooo....http://www.snom.com/661.html |
| 03:11.11 | asterboy | snom 360 $250 usd |
| 03:11.29 | asterboy | what is the cheapest 2 port RJ45 phone??? |
| 03:11.44 | Qwell | 2 port, as in with a builtin switch? |
| 03:12.01 | lenne_dk | Like BT102, I presume |
| 03:12.11 | *** join/#asterisk M-I-A (n=chaguy42@209.161.199.142) |
| 03:12.35 | asterboy | bt102 is grandstream I think. |
| 03:12.41 | lenne_dk | Yes |
| 03:12.55 | wmandra | justinu, didn't work..... |
| 03:13.52 | M-I-A | wow there are a lot of people in here... I did not realize Asterisk had this kind of following |
| 03:13.58 | *** join/#asterisk JunK-Y (n=junky@69.156.217.51) |
| 03:14.35 | Lostfrog | Or a snom 320. |
| 03:14.56 | Lostfrog | Just like a 360, except smaller screen and fewer simultaneous calls. |
| 03:15.49 | IronHelix | m-i-a alot of the users here idle and ask a question every now and then |
| 03:15.53 | asterboy | can the bt102 do call conferencing...guess its an asterisk function so it should???? |
| 03:16.01 | IronHelix | but asterisk certainly has a following, it is saving real people real money |
| 03:16.45 | Lostfrog | asterboy: meetme conferencing is *.. 3-way calling is on the phone. |
| 03:16.50 | asterboy | what is really saving the money is switch pots to voip...otherwise the equipment is much the same price. |
| 03:17.19 | Lostfrog | asterboy: and flexibility and scalabality |
| 03:17.37 | asterboy | yes for sure...thats a harder sell I'm finding. |
| 03:18.02 | justinu | wmandra: check your phone config then |
| 03:18.04 | JunK-Y | me hugs Katty too. |
| 03:18.11 | asterboy | ok so the bt102 most likely will not support 3-way calling...I thought * could do this on the server side and supply it to the FXS |
| 03:18.13 | twisted | JUNK-Y! |
| 03:18.24 | JunK-Y | hey twisted ! |
| 03:18.27 | Katty | JunK-Y++ |
| 03:18.29 | twisted | what's going on? |
| 03:18.31 | file | oh noes it's JunK-Y |
| 03:18.32 | Lostfrog | asterboy: it may or may not.. read it's manual. |
| 03:18.40 | JunK-Y | how do u like my new t-shirt btw? |
| 03:18.46 | Lostfrog | I don't think it can handle multiple calls at the same time. |
| 03:18.50 | twisted | so JunK-Y, out with the details... did you cuddle with file? :P |
| 03:18.53 | wmandra | justinu, i've checked it and rechecked it, not working...... |
| 03:18.53 | JunK-Y | sqrt(junky) = katty |
| 03:19.01 | file | LOL |
| 03:19.07 | Katty | that does not parse. |
| 03:19.12 | file | I cuddled with his girlfriend when he was gone! OH NO!!! |
| 03:19.13 | twisted | Katty, apparently you're his square root |
| 03:19.20 | JunK-Y | u know twisted, me and file are in terrible love. |
| 03:19.27 | twisted | JunK-Y, i thought so ;) |
| 03:19.31 | file | it's true |
| 03:19.32 | Katty | twisted: yes, but that does not parse. |
| 03:19.34 | Lostfrog | You need to include math.h and use -lmath when linking, Katty. |
| 03:19.51 | Lostfrog | excuse me.. -lm |
| 03:19.57 | twisted | hah |
| 03:19.59 | asterboy | Support 3-way conferencing (Model 102D), |
| 03:20.00 | Katty | JunK-Y: so..i'm squared at all times around you? |
| 03:20.12 | busdriver202 | file: its 2.50!!! |
| 03:20.24 | twisted | whoa, Katty^2 would make my brain rupture... |
| 03:20.25 | IronHelix | 102d is cancelled |
| 03:20.29 | twisted | (not in a bad way) |
| 03:20.29 | IronHelix | but as i recall |
| 03:20.32 | busdriver202 | after the diamonds, lets move to 202! |
| 03:20.38 | IronHelix | new beta firmware for bt1xx series has 3way |
| 03:20.48 | Katty | twisted: you couldn't handle two of me. |
| 03:20.56 | twisted | Katty, you sure? |
| 03:21.21 | busdriver202 | twisted, katty: if u wanna take a look at our last week, http://www.midsouthmarketplace.com/~krice/gallery/view_album.php?set_albumName=album04 |
| 03:21.22 | IronHelix | asterboy http://www.grandstream.com/BETATEST/BT100_HT286_HT486/ |
| 03:21.22 | asterboy | wow thanks for that IronHelix. |
| 03:21.27 | twisted | uh oh... my feline companions have realized that I have a packed suitcase in the living room |
| 03:21.29 | file | lol |
| 03:21.34 | IronHelix | the 1.0.7.11 firmware supports conference |
| 03:21.40 | twisted | busdriver202, i saw it already :) |
| 03:21.41 | Lostfrog | Ok.. Pineapple upside-down cake is good in small amounts. |
| 03:21.46 | justinu | twisted: you're in trouble now |
| 03:22.04 | twisted | justinu, i am? |
| 03:22.10 | kn0x | what does this mean: Nov 13 15:21:36 WARNING[2398]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x819e9b0 (len 397) to 69.90.155.70:-1 returned 5060: Bad file descriptor |
| 03:22.12 | justinu | cats know you're leaving |
| 03:22.13 | kn0x | ? |
| 03:22.15 | twisted | oh yea |
| 03:22.20 | twisted | but they also know i'll be back in a few days |
| 03:22.26 | asterboy | So it looks like the cheapest way for me to build a system is to offer DID at $4.50 and issue it via Grandstream BT102. |
| 03:22.43 | busdriver202 | twisted: wanna know the daily tips? never put red bull in ur eyes! |
| 03:22.43 | asterboy | Won't need the Digium hardware then. |
| 03:22.50 | twisted | busdriver202, NO KIDDING |
| 03:22.53 | busdriver202 | hheheh |
| 03:22.57 | justinu | asterboy: sounds like a plan |
| 03:23.06 | IronHelix | asterboy what are you trying to build? ITSP? office PBX? |
| 03:23.07 | asterboy | interesting. |
| 03:23.13 | asterboy | office PBX |
| 03:23.23 | IronHelix | hosted? |
| 03:23.29 | asterboy | no |
| 03:23.36 | twisted | busdriver202, was that your gf in some of those pics? |
| 03:23.41 | asterboy | turnkey installed server. |
| 03:23.45 | busdriver202 | twisted: yes |
| 03:23.52 | twisted | busdriver202, she's cute |
| 03:24.00 | busdriver202 | NO KIDDING! :P |
| 03:24.05 | IronHelix | thats good. a suggestion, bt102/bt101 (same thing really) feel like cheap plastic toys. doestn give biz users warm fuzzies unless they are really pressed for costs |
| 03:24.13 | busdriver202 | thats why shes my gf :P |
| 03:24.18 | justinu | IronHelix: agreed |
| 03:24.25 | busdriver202 | taht girl is great, smart, cute, special. |
| 03:24.27 | twisted | busdriver202, that's the oly reason? |
| 03:24.29 | twisted | oh |
| 03:24.30 | IronHelix | they are fun to play around with but they look absurd on somebodys desk |
| 03:24.32 | twisted | okay good ;) |
| 03:24.39 | busdriver202 | plus, shes learning * too. |
| 03:24.45 | *** join/#asterisk fafnir (n=hello@tdds-gw.Moscow.gldn.net) |
| 03:25.02 | IronHelix | for a bit more $$ check out the grandstream GXP2000 ($100 or so USD) |
| 03:25.05 | justinu | rule #1: never date a girl who likes computers |
| 03:25.09 | IronHelix | at least it doestn LOOK like a toy |
| 03:25.14 | IronHelix | also sipura 841 is nice |
| 03:25.24 | justinu | 841 is nice except for the crap keypad |
| 03:25.28 | IronHelix | yeah |
| 03:25.32 | IronHelix | thats true |
| 03:25.35 | asterboy | ok, so go with the grandstream GXP-2000?? |
| 03:25.36 | twisted | busdriver202, uh oh |
| 03:25.50 | hhoffman | anyone here using teliax? |
| 03:26.01 | IronHelix | if you can afford 150/user check out the sipura/linksys spa941 or anything by SNOM |
| 03:26.08 | IronHelix | AAstra is also good i've heard |
| 03:26.22 | Lostfrog | justinu: why shouldn't you date a femaled geek? |
| 03:26.26 | justinu | i have an aastra 480i on the way |
| 03:26.37 | justinu | Lostfrog: i dunno, i like to keep my interests seperate |
| 03:26.57 | Lostfrog | At least she won't be pissed off about you not spending time with her. |
| 03:26.59 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 03:26.59 | asterboy | missed your post IronHelix...ya, thats a great suggestion. |
| 03:27.03 | IronHelix | asterboy whatever you get, buy one or two of them first for yourself and learn the quirks of it before you commit client $$ |
| 03:27.12 | lenne_dk | A femaled geek? Doesn't that mean a geek who had a feemale? |
| 03:27.13 | justinu | that's good advice |
| 03:27.15 | Katty | JunK-Y: neato. |
| 03:27.25 | asterboy | The BT101 and BT102 are a major difference if you don't want the extra cost of Hubbing that RJ45 connection. |
| 03:27.35 | IronHelix | true, BUT |
| 03:27.35 | Lostfrog | No.. a geek with a sex changes, lenne_dk. |
| 03:27.37 | Lostfrog | :) |
| 03:27.37 | asterboy | good suggestion to try out first. |
| 03:27.43 | justinu | the gpx2000 has 2 ports |
| 03:27.45 | JunK-Y | katty: u will have to come one day too. |
| 03:27.46 | IronHelix | keep in mind the bt102 only has 10mbit/sec ethernet ports |
| 03:27.50 | Katty | nothing wrong with sex changes. |
| 03:27.52 | asterboy | so does the BT102 |
| 03:27.59 | Katty | JunK-Y: kthen |
| 03:28.01 | asterboy | hence the btxx2 |
| 03:28.02 | IronHelix | the gxp as i recall has 100mbit |
| 03:28.07 | justinu | yeah |
| 03:28.18 | Lostfrog | As does the snoms |
| 03:28.20 | IronHelix | so if you're gonna use dualport phones to save cabling, make sure you have enough ports |
| 03:28.21 | Lostfrog | do |
| 03:28.23 | JunK-Y | how do u like my evil pig? |
| 03:28.27 | IronHelix | eh make sure you have the speed |
| 03:28.28 | justinu | asteriskboy: you'll find out that business will go ga-ga over the polycom 601 |
| 03:28.30 | IronHelix | or that your users wont miss it |
| 03:28.34 | JunK-Y | blitz loved him SO MUCH! |
| 03:28.44 | twisted | evil pig? |
| 03:28.47 | twisted | do explain |
| 03:28.51 | JunK-Y | http://www.midsouthmarketplace.com/~krice/gallery/view_photo.php?set_albumName=album04&id=evil_pig |
| 03:29.01 | JunK-Y | its my bank for my change :) |
| 03:29.04 | twisted | that pig is not evil |
| 03:29.06 | twisted | that pig is happy |
| 03:29.16 | IronHelix | polycoms have orgasmic speakerphones (one of polycoms original business was conference room speakerphones) but you have to deal with VARs to get firmware |
| 03:29.19 | Katty | JunK-Y: it's kinda cute. |
| 03:29.27 | file | the pig is h-o-t |
| 03:29.37 | JunK-Y | like bb? |
| 03:29.42 | file | quite |
| 03:29.48 | asterboy | wow, that polycom rocks! |
| 03:29.55 | asterboy | $300usd |
| 03:29.59 | JunK-Y | hey bb you're hurting me. |
| 03:30.13 | Lostfrog | I may have to get a polycom for the conference room.. |
| 03:30.17 | file | [TK]D-Fender: nooooooooo |
| 03:30.19 | JunK-Y | we're saying that shit since last week. |
| 03:30.24 | [TK]D-Fender | hehhe |
| 03:30.27 | Lostfrog | Why don't they make an IP phone that has a FXS port on it? |
| 03:30.31 | JunK-Y | [TK]D-Fender: send me my butt's pic now. |
| 03:30.33 | file | [TK]D-Fender: see MSN for a present. |
| 03:30.43 | Katty | wish i could have gone, but company didn't want to fork out the money :< |
| 03:30.45 | [TK]D-Fender | I did, thx :) |
| 03:31.00 | IronHelix | lostfrog- the only one i've seen is the uniden uip1688 |
| 03:31.11 | IronHelix | its a sip based 5.8ghz cordless phone base |
| 03:31.31 | Lostfrog | I would love to be able to plug our polycom into an IP phone. |
| 03:31.37 | IronHelix | but it doesnt look like a biz phone |
| 03:31.43 | justinu | IronHelix: http://www.freedomphones.net/polycom/files/ |
| 03:31.43 | *** join/#asterisk Corydon76-home (i=beige@pdpc/supporter/sustaining/Corydon76-home) |
| 03:31.45 | IronHelix | well you can always use an ATA :( |
| 03:31.53 | Lostfrog | True. |
| 03:32.05 | IronHelix | jus- useful |
| 03:32.11 | IronHelix | is that kept up to date? |
| 03:32.16 | justinu | yeah, it has the latest |
| 03:32.25 | Lostfrog | But that would mean two devices in the same room.. because people aren't going to give up the IP phone. |
| 03:32.26 | JunK-Y | file |
| 03:32.32 | JunK-Y | ive a great pic to let u see. |
| 03:32.37 | IronHelix | either way- polycom is still better than cisco |
| 03:32.40 | Lostfrog | Maybe.. I have some time to play. |
| 03:32.41 | Katty | file's going to be a pretty boy when he grows up |
| 03:32.57 | Katty | i mean pretty boy(tm) |
| 03:32.59 | IronHelix | you have to pay them $$$ for a support contract to be able to legally get the firmware at all |
| 03:33.02 | file | ut roh |
| 03:33.04 | file | what pic? |
| 03:33.22 | twisted | haha |
| 03:33.56 | [TK]D-Fender | file : its one of mine... PRICELESS, trust me ... muahahahhaha *cough* |
| 03:33.59 | Lostfrog | ut roh, roy. |
| 03:34.01 | file | oh god |
| 03:34.03 | file | this can't go well |
| 03:35.02 | IronHelix | also asterboy |
| 03:35.02 | JunK-Y | http://junky.homelinux.org/nice_yeah.jpg |
| 03:35.07 | IronHelix | check out the 3com nj100 |
| 03:35.13 | [TK]D-Fender | Lostfrog : what kind of polycom are you trying to plug into an IP phone (not sure I even follow you, but I'll bite...) |
| 03:35.13 | Lostfrog | justinu: May I PM you? |
| 03:35.23 | Lostfrog | Soundstation 100. |
| 03:35.59 | [TK]D-Fender | Lostfrog : thats analog isn't it? |
| 03:36.03 | IronHelix | its an in wall ethernet switch... add a 802.3af injector on the other side and for $130 you've turned one port into 4. It also has one passthru 802.3af port so you can power a phone off it too |
| 03:36.15 | Lostfrog | Yes.. hence why I spoke about an FXS port on an IP Phone. |
| 03:36.23 | twisted | JunK-Y, what the... |
| 03:36.24 | Katty | JunK-Y: they look girly |
| 03:36.44 | Lostfrog | I should be investing in PoE. |
| 03:36.54 | lenne_dk | I'd hate to meet boys looking like that |
| 03:36.56 | IronHelix | good way to expand a site capability without running cable |
| 03:37.01 | JunK-Y | twisted: nice pic huh? :) |
| 03:37.15 | JunK-Y | lenne_dk: i some some guys which look like that too :P |
| 03:37.16 | Katty | JunK-Y: no goth girl :< |
| 03:37.30 | IronHelix | i wish a network vendor would wake up one day and deploy a good cheap line of powered gigabit switches |
| 03:37.36 | IronHelix | i'd buy tons of those |
| 03:37.38 | Katty | JunK-Y: that makes me all sad inside. |
| 03:37.48 | [TK]D-Fender | Lostfrog : Just use an ATA. When I got my Ploycom Wireless conferencing module I made sure to make it analog an not digital (I had a Norstar setup at the time). |
| 03:37.56 | JunK-Y | katty: then take ur video cam and go make me a pic :P |
| 03:37.58 | [TK]D-Fender | Lostfrog : ATA works just great. |
| 03:38.01 | asterboy | ok, you still run 1 rj45 cable..just that it splits like a multiplex. |
| 03:38.05 | Katty | JunK-Y: pfft. |
| 03:38.08 | IronHelix | sadly, gigabit switches arent cheap, powered switches are even less cheap, and powered gigabit if its even sold yet is probably very expensive |
| 03:38.15 | Katty | JunK-Y: kats do not get on camera. |
| 03:38.17 | IronHelix | asterboy- exactly, its a network switch built into the jack |
| 03:38.20 | fafnir | JunK-Y: only 3 of those girls belong |
| 03:38.29 | Lostfrog | [TK]D-Fender: I know. |
| 03:38.30 | IronHelix | same as a desk hub or switch, only you cant walk away with it |
| 03:38.39 | Lostfrog | But, I would still need an IP phone. |
| 03:38.51 | JunK-Y | fafnir: which ones? |
| 03:39.05 | IronHelix | also makes one PoE injector do double duty (the switch and the phone) |
| 03:39.15 | [TK]D-Fender | Powered gigabit isn't quite real yet. 803.11af uses 2 pairs for power and doesn't put data over it so those ports are 10/100. I have heard of 1 model or so taht offers you 1000 *IF* there is no PoE |
| 03:39.22 | IronHelix | so that removes two power bricks from your overall setup |
| 03:39.31 | asterboy | Another wall jack that might interest some...http://www.windowsfordevices.com/news/NS3139003780.html |
| 03:39.33 | [TK]D-Fender | PoE injectors only do 10/100 AIRC |
| 03:39.48 | IronHelix | as i understand it, and i could be wrong, gigabit uses differential signalling so it would be compatible with PoE |
| 03:40.03 | Lostfrog | Grrr.. put spaces before your URLs. :) |
| 03:40.20 | IronHelix | http://www.windowsfordevices.com/news/NS3139003780.html clickable |
| 03:40.23 | asterboy | wonder how those skype phones compare. |
| 03:40.24 | IronHelix | damn, thats cool |
| 03:40.45 | fafnir | JunK-Y: bottom left, top right, actually 2 dont belong |
| 03:40.49 | fafnir | other four are fine |
| 03:41.16 | IronHelix | i think the main problem with gigabit poe is theres little/no demand for it |
| 03:41.22 | kn0x | can someone help me with registering asterisk to free world dialup from inside nat |
| 03:41.28 | kn0x | i have port forwarding |
| 03:41.37 | Lostfrog | That would be cool.. |
| 03:41.38 | IronHelix | knox- port forward. in sip.conf set externip= and localnet= |
| 03:41.41 | kn0x | 5060, and 10000-20000 |
| 03:41.46 | Lostfrog | Put two (or more) computers in each room.. |
| 03:41.47 | [TK]D-Fender | Ok, I'm off for the night, later peeps.... JunK-Y : I'll have your pics ready for tomorrow night. |
| 03:41.57 | IronHelix | then set qualify=yes canreinvite=no under the fwd section |
| 03:41.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 03:42.01 | IronHelix | 90% of the demand for ports is standard 10/100 unpowered ports |
| 03:42.01 | Lostfrog | Run terminal services on a dual/quad xeon. :) |
| 03:42.09 | kn0x | http://pastebin.ca/28623 is my sip.conf |
| 03:42.19 | kn0x | ironhelix- im also getting this message on the cli |
| 03:42.21 | IronHelix | gigabit is gaining acceptance, but is still at a premium |
| 03:42.23 | fafnir | JunK-Y: thanks now i want to go have sex with some 18 year old girl, thanks. |
| 03:42.30 | kn0x | Nov 13 15:21:40 WARNING[2398]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x819e9b0 (len 397) to 69.90.155.70:-1 returned 5060: Bad file descriptor |
| 03:42.45 | Lostfrog | fafnir: send me one. :) |
| 03:43.02 | fafnir | 200 a pop |
| 03:43.06 | IronHelix | kn0x- i dont think * supports a domain name in externip= |
| 03:43.07 | JunK-Y | fafnir: theres all over 21. |
| 03:43.09 | IronHelix | i could be wrong tho |
| 03:43.17 | Lostfrog | fafnir: night or hour? |
| 03:43.22 | Lostfrog | I'm joking people. |
| 03:43.28 | fafnir | night |
| 03:43.31 | fafnir | but you buy drinks |
| 03:43.52 | fafnir | JunK-Y: doesnt stop me from wanting an 18 year old girl, preferably still in HS |
| 03:44.31 | IronHelix | scandalous! |
| 03:44.32 | JunK-Y | sorry, i dont have to get girls drunk to ... them :P |
| 03:44.43 | Lostfrog | JunK-Y: It does help, though. :) |
| 03:44.50 | fafnir | no no, its not getting them drun |
| 03:44.56 | fafnir | you already paid for their services |
| 03:45.00 | fafnir | they just want the liqour |
| 03:45.02 | JunK-Y | file: HELP ME, tell him how sweet is bb |
| 03:45.03 | IronHelix | ouch! |
| 03:45.03 | fafnir | makes it easier for them |
| 03:45.09 | lenne_dk | I know what .... means, but what is ... ? |
| 03:45.19 | IronHelix | show them his phone system |
| 03:45.22 | IronHelix | :) |
| 03:45.54 | JunK-Y | mouhaha |
| 03:46.00 | kn0x | ironhelix- im not using externip.. im using externhosy |
| 03:46.02 | kn0x | *host |
| 03:46.03 | JunK-Y | hey, wanna see my PBX baby? |
| 03:46.10 | Lostfrog | lol |
| 03:46.11 | fafnir | youre all a bunch of nerds |
| 03:46.13 | fafnir | NERDZ |
| 03:46.19 | lenne_dk | I've got 4 phones for 2 people on my system. Is that a good pickup line? |
| 03:46.20 | kn0x | i know |
| 03:46.21 | IronHelix | the bad file descriptor could be a larger problem tho |
| 03:46.22 | Lostfrog | fafnir: and *proud* of it. |
| 03:46.27 | IronHelix | maybe its having trouble with your network interface |
| 03:46.44 | fafnir | im one too :( |
| 03:46.48 | JunK-Y | fafnir: http://www.midsouthmarketplace.com/~krice/gallery/view_photo.php?set_albumName=album04&id=IMG_4699 |
| 03:46.48 | lenne_dk | And I can call my fridge to hear the temperature |
| 03:46.55 | IronHelix | hey baby, wanna provision a SIP channel? we can qualify back and forth all night long... |
| 03:46.56 | Lostfrog | Any Zap, is a good pickup line, lenne_dk. |
| 03:47.12 | fafnir | pfft |
| 03:47.18 | fafnir | i whip out my gumstix |
| 03:47.25 | lenne_dk | No ZAP, only SIP and ATA |
| 03:47.25 | fafnir | and their like 'whooooa' |
| 03:47.37 | fafnir | "Omg thats a computer hehe' |
| 03:47.37 | asterboy | Can't wait to put my asterisk install on this: http://web.archive.org/web/20041127094503/http://www-ccs.cs.umass.edu/~shri/iPic.html |
| 03:47.43 | cybertank | Anyone else using Asterisk@Home? |
| 03:48.00 | Lostfrog | Not the way it was intended, cybertank. |
| 03:48.01 | fafnir | yeah |
| 03:48.06 | fafnir | okay |
| 03:48.07 | cybertank | Kik |
| 03:48.12 | fafnir | not gonna happen :P |
| 03:48.13 | Qwell | I use *@h |
| 03:48.24 | Qwell | I burned a crapload of them, to stop my table from wobbling |
| 03:48.24 | twisted | Qwell, yes, and you must suffer for it |
| 03:48.29 | cybertank | If I update the kernel... it will die right? |
| 03:48.34 | Qwell | it's perfectly sturdy now |
| 03:48.36 | twisted | oh |
| 03:48.37 | twisted | hahah |
| 03:48.37 | twisted | ok |
| 03:48.43 | Qwell | I would have used Windows CDs, but... |
| 03:48.48 | Qwell | we all know how stable /that/ is |
| 03:48.55 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 03:48.58 | fafnir | Ha |
| 03:49.00 | twisted | yeah, it'd suck if your table bluescreened |
| 03:49.08 | Qwell | it'd be worse if it crashed |
| 03:49.14 | Qwell | especially during dinner or something |
| 03:49.19 | JonR800 | is it bad joke night in #asterisk? :) |
| 03:49.26 | Qwell | JonR800: What, are you new? |
| 03:49.32 | devonst17 | lol |
| 03:49.33 | JonR800 | im a lurker |
| 03:49.35 | Qwell | That's practically the topic |
| 03:49.44 | devonst17 | Then i think I will hang out here more! |
| 03:49.46 | devonst17 | :D! |
| 03:49.57 | twisted | i'll get you my pretty |
| 03:49.58 | cybertank | Has anyone brough Centos up to date and kept Asterisk stable?? |
| 03:50.02 | twisted | and you're little box, too |
| 03:50.05 | Qwell | twisted: You think I'm pretty? :) |
| 03:50.11 | twisted | s/you're/your |
| 03:50.20 | asterboy | Digg.com had this great link: http://linuxgazette.net/120/smith.html |
| 03:50.24 | twisted | i'm watching wizard of oz on tv |
| 03:50.32 | Qwell | oh :( |
| 03:50.36 | devonst17 | Lol |
| 03:50.41 | twisted | FUCK ME WITH A TABLESAW |
| 03:50.45 | twisted | i forot to put out the trash |
| 03:50.46 | devonst17 | twisted: You got Qwell's hopes up |
| 03:50.57 | devonst17 | oh fuck, so did I |
| 03:51.11 | cybertank | Damn... we must be in the same timezone! Same here |
| 03:51.17 | fafnir | lately i've been finding myself attracted to girls that arent that... attractive |
| 03:51.25 | twisted | fafnir, you're getting older |
| 03:51.27 | asterboy | lol, I'm watching that too...I love the quote, "I don't think were in Kansas anymore Dorthy!" |
| 03:51.29 | cybertank | fafnir... that sucks |
| 03:51.34 | JunK-Y | twisted: saw? :P u have to see that movie (saw 2) |
| 03:51.40 | JunK-Y | i went with file, its crazy! |
| 03:51.41 | fafnir | well, i did just turn 21 |
| 03:51.42 | twisted | JunK-Y, i haven't seen the second one yet |
| 03:51.44 | fafnir | but thats not that old |
| 03:51.46 | *** join/#asterisk bmg505 (n=leon@rndf-146-47-98.telkomadsl.co.za) |
| 03:51.54 | asterboy | Is saw2 available for download yet? |
| 03:51.55 | twisted | fafnir, oh, then it's the alcohol |
| 03:51.59 | wasim | you'd think not being in kansas anymore would be a good thing [tm] |
| 03:52.05 | fafnir | i dont drink |
| 03:52.06 | asterboy | lol |
| 03:52.11 | Qwell | JunK-Y: re shirts; you'd like mine |
| 03:52.14 | twisted | fafnir, or lack thereof |
| 03:52.17 | devonst17 | fafnir: I have the same problem, except its not girls that arent attractive... its girls with huge racks... |
| 03:52.17 | asterboy | "I'll get you my pretty!" |
| 03:52.20 | Lostfrog | wow.. asterboy: that is an awesome little computer. |
| 03:52.22 | Corydon76-home | asterboy: uh, just how much do you like Wizard of Oz? |
| 03:52.24 | Qwell | JunK-Y: "I'm an asshole. What's your excuse?" |
| 03:52.25 | JunK-Y | which one? |
| 03:52.26 | fafnir | well |
| 03:52.39 | fafnir | its like 'huh i could see myself having sex with her in an hour' |
| 03:52.39 | asterboy | Depends if I'm high or not. :P |
| 03:52.42 | Corydon76-home | asterboy: friend of Dorothy? |
| 03:52.43 | twisted | Qwell, you're not an asshold |
| 03:52.43 | JunK-Y | hhehee |
| 03:52.46 | twisted | er hole |
| 03:52.47 | twisted | hahaha |
| 03:52.49 | devonst17 | LOL |
| 03:52.53 | Qwell | man... |
| 03:52.58 | asterboy | Kinda like watching Alice in Wonderland. |
| 03:53.12 | Qwell | twisted: I can be... |
| 03:53.28 | Qwell | You cross me...and I'LL CUT YOU! |
| 03:53.34 | twisted | okay, SwK |
| 03:53.39 | Corydon76-home | Qwell: no, that's "bitch" |
| 03:53.59 | asterboy | or how about old Spiderman with those Acid Wash backgrounds...real trippy. |
| 03:54.47 | lenne_dk | Nice. Asterisk HEAD just finished compiling on FreeBSD, without patching. |
| 03:54.52 | cybertank | Damn... I thought it was the drugs... you saw the acid wash backgrounds too?? |
| 03:54.57 | asterboy | Lostfrog: ya that iPIC chip is my next project for a web page. |
| 03:55.09 | asterboy | lol |
| 03:55.42 | asterboy | I got real screwed up when they mixed Rocket Robinhood with Spiderman and Dimensure V |
| 03:55.44 | *** join/#asterisk chandi (n=burni13@modemcable248.1-201-24.mc.videotron.ca) |
| 03:55.48 | Corydon76-home | Enjoying that, devon? |
| 03:55.56 | devonst17 | Not really |
| 03:55.58 | devonst17 | lol |
| 03:56.50 | chandi | hey guys, little question. If there's voicemail messages but asterisk doesn't send a sip notify with a MWI flag. where should I have a look ? ;) |
| 03:56.53 | cybertank | So... *@h... anyone try updating the kernel? Good Bad Ugly? |
| 03:57.03 | Qwell | chandi: app_voicemail |
| 03:57.15 | IronHelix | very few of use *@h |
| 03:57.24 | IronHelix | i dont see any reason why it would hugely fail |
| 03:57.46 | IronHelix | chandi- sip.conf, is mailbox set to user@context? |
| 03:57.49 | chandi | qwell : which settings ? |
| 03:57.58 | mog_home | indeed |
| 03:57.58 | Qwell | or what he said |
| 03:58.12 | chandi | IronHelix : it's set. @context = context of voicemail.conf, right ? |
| 03:58.17 | Qwell | mog_home: I tried that jabber thing...umm...yeah |
| 03:58.20 | Qwell | java POS, heh |
| 03:58.21 | mog_home | lol |
| 03:58.25 | mog_home | whats up? |
| 03:58.27 | IronHelix | yup |
| 03:58.33 | Qwell | I couldn't figure out how to run the damn thing. :P |
| 03:58.35 | mog_home | lol |
| 03:58.38 | Qwell | Is it like...a client...or what? |
| 03:58.41 | mog_home | mine or the other one |
| 03:58.42 | chandi | IronHelix and user= extension number in voicemail.conf ? like 120@default |
| 03:58.45 | mog_home | it can be either |
| 03:58.45 | Qwell | the java one |
| 03:58.52 | mog_home | oh the java one |
| 03:58.54 | mog_home | is not my stuff |
| 03:58.56 | Qwell | yeah, I couldn't figure out for the life of me how to run the client, heh |
| 03:58.57 | IronHelix | doesnt matter what user= is set to |
| 03:58.58 | mog_home | but it uses manager |
| 03:59.01 | Qwell | yeah, I know. you just recommended it |
| 03:59.09 | mog_home | i do |
| 03:59.15 | mog_home | because my stuff isnt done |
| 03:59.16 | IronHelix | extensions.conf will (if set up right) send the call to voicemail(mailbox) |
| 03:59.21 | asterboy | wow, saw 2 is available. |
| 03:59.30 | IronHelix | the mailbox si what shoudl be in sip.conf in mailbox= |
| 03:59.36 | Qwell | somebody should have gone to vegas then :p |
| 03:59.39 | Lostfrog | I thought saw 2 was still in theaters. |
| 03:59.43 | mog_home | lol |
| 03:59.49 | mog_home | i cant miss any more classes |
| 03:59.52 | chandi | IronHelix it does send it to the right voicemail but it doesn't send a MWI flag :( |
| 03:59.54 | Lostfrog | You wouldn't do anything illegal, would you, asterboy? |
| 03:59.59 | mog_home | ive missed like 3 weeks |
| 04:00.06 | asterboy | *cough* no..no way! |
| 04:00.09 | IronHelix | two separate things |
| 04:00.16 | IronHelix | extensions.conf send the call to voicemail |
| 04:00.16 | chandi | IronHelix and asterisk says that that user has 4 messages |
| 04:00.28 | IronHelix | are you sure the phone is picking it up right? |
| 04:01.02 | chandi | IronHelix I've been sniffing the traffic between the sipura ATA and asterisk but asterisk doesn't send any NOTIFY RMI thing |
| 04:02.03 | chandi | IronHelix : so it doesn't seem to be something about the mime-type cuz I see nothing between the client and server about MWI |
| 04:02.09 | IronHelix | hmmm |
| 04:03.05 | IronHelix | any cli output? |
| 04:03.10 | chandi | IronHelix: weird, hey ? |
| 04:03.15 | IronHelix | yo |
| 04:03.18 | *** join/#asterisk DCGrendel (n=DCGrende@ip68-1-157-197.mc.at.cox.net) |
| 04:03.19 | justinu | chandi: did you set mailbox= in sip.conf? |
| 04:03.48 | chandi | IronHelix no.. well.. when verbose is high enough I just see that a new message arrived in the mailbox |
| 04:03.59 | DCGrendel | is there a way to get more detailed output than just "CAUSE : No such context/extension" when debugging an incoming iax call that fails? |
| 04:04.00 | chandi | justinu : yes, mailbox=120@default |
| 04:04.10 | DCGrendel | like the specific extension lookup that failed... |
| 04:04.13 | chandi | justinu does that seem right ? |
| 04:04.16 | justinu | yeah |
| 04:04.35 | chandi | justinu and voicemail.conf has : 120 => 2,Samuel Dion,chandi@estriesansfil.org |
| 04:04.38 | IronHelix | and thats in the entry for the sipura box, not in general |
| 04:04.39 | chandi | in @default |
| 04:04.49 | chandi | IronHelix yup |
| 04:04.55 | justinu | hmm, you did a sip reload? |
| 04:04.56 | IronHelix | the context is @default or default? |
| 04:05.53 | chandi | IronHelix just default |
| 04:06.10 | chandi | justinu well.. stopped asterisk many times since then |
| 04:06.16 | justinu | should work |
| 04:06.46 | chandi | justinu I know :i |
| 04:07.02 | *** join/#asterisk BleedingMe (n=Bleeding@ppp-71-137-216-107.dsl.scrm01.pacbell.net) |
| 04:07.26 | BleedingMe | is there a new script in 1.2 to add a mailbox? |
| 04:07.31 | justinu | chandi: phone type? |
| 04:07.37 | chandi | justinu and Iron : is * supposed to send a notify mwi upon registration ? |
| 04:07.45 | BleedingMe | addmailbox seems to be gone |
| 04:07.48 | justinu | it sends it periodically |
| 04:07.49 | chandi | justinu sipura 2001 |
| 04:07.54 | chandi | justinu ok |
| 04:08.05 | chandi | justinu it never did :I |
| 04:09.08 | justinu | i don't use the @default thing |
| 04:09.13 | justinu | just looked over my config |
| 04:09.18 | justinu | i just put mailbox=4512 |
| 04:09.23 | chandi | justinu ok, I'll try that |
| 04:09.59 | justinu | checkmwi=10 |
| 04:10.13 | justinu | i think that means it sends once every 10 minutes |
| 04:10.16 | chandi | justinu is that 10 minutes or seconds ? |
| 04:10.17 | chandi | ok |
| 04:10.22 | chandi | justin : mine is set to 1 |
| 04:10.29 | justinu | strange |
| 04:10.42 | justinu | everything is working except the MWI? |
| 04:10.56 | chandi | justin exactly |
| 04:11.06 | justinu | version of asterisk? |
| 04:11.23 | chandi | justinu HEAD |
| 04:11.26 | Qwell | maybe they don't have VM? :D |
| 04:12.02 | justinu | i think it sends the notify anyways |
| 04:12.02 | chandi | Qwell : Oh ;) the power was off on the sipura ;) just kidding |
| 04:12.14 | Lostfrog | lol |
| 04:12.17 | *** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net) |
| 04:12.30 | justinu | chandi: sounds like you've done everything right |
| 04:12.50 | chandi | what's vmexten= in sip.conf ? |
| 04:13.19 | justinu | that's the "callback" that gets sent to the phone in the notify |
| 04:13.24 | justinu | tells it where to dial back for it's vm |
| 04:13.27 | Lostfrog | chandi: it's used for phones that can't figure out how to retrieve voicemail. |
| 04:13.47 | Lostfrog | like snoms. :) |
| 04:13.50 | justinu | snom needs that, |
| 04:13.51 | justinu | heh |
| 04:14.04 | DCGrendel | Nov 13 23:12:31 NOTICE[2576]: Rejected connect attempt from (scrubbed), request '(scrubbed)@fromiaxfwd' does not exist |
| 04:14.10 | Lostfrog | Snoms need a sledge hammer to make vm work. |
| 04:14.18 | chandi | justinu ok... grrr |
| 04:14.23 | justinu | lostfrog: took me a while, but I got it finally |
| 04:14.30 | Lostfrog | Took me 1 hour. |
| 04:14.31 | chandi | Lostfrog haha |
| 04:14.34 | Lostfrog | That is a long time for me. |
| 04:14.37 | file[laptop] | DCGrendel: the extension, in the context, doesn't exist |
| 04:14.45 | justinu | next task: get BLA's working |
| 04:14.45 | Lostfrog | That is with the help from the wiki. |
| 04:14.52 | DCGrendel | file[laptop]: but i can see it in extensions.conf |
| 04:15.09 | file[laptop] | DCGrendel: it's going to the fromiaxfwd context, and the extension (which is given to you), doesn't exist |
| 04:15.11 | Lostfrog | In the right context, DCGrendel? |
| 04:15.17 | DCGrendel | yes |
| 04:15.34 | file[laptop] | if it says it doesn't exist, chances are - it doesn't exist |
| 04:15.34 | DCGrendel | [fromiaxfwd] |
| 04:15.34 | DCGrendel | exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r) |
| 04:15.38 | chandi | justinu: does the sipurabox has to ask the server if it has mail or not ? |
| 04:15.53 | justinu | chandi: i think it's a server push |
| 04:15.58 | file[laptop] | and FWDNUMBER is set as a global variable of your FWD number? |
| 04:16.05 | DCGrendel | yea |
| 04:16.20 | DCGrendel | can you have multiple [global] sections? |
| 04:16.22 | justinu | chandi: in my experience, when mailbox= is set in sip.conf, the server pushes the notify to the UA |
| 04:16.27 | file[laptop] | DCGrendel: no... |
| 04:16.28 | chandi | justinu ok.... |
| 04:16.33 | justinu | chandi: like instantly |
| 04:16.37 | DCGrendel | file[laptop]: not even in included files? |
| 04:16.42 | file[laptop] | weird stuff might happen... |
| 04:16.50 | file[laptop] | like, it won't work |
| 04:16.55 | chandi | justinu : Hmmm ok. weird! |
| 04:17.02 | justinu | well, pizza time |
| 04:17.05 | justinu | good luck! |
| 04:17.33 | chandi | take care justinu, thanks |
| 04:18.20 | Lostfrog | justinu: you coming back on later? |
| 04:18.46 | BleedingMe | does anyone know if there is a replacement for the addmailbox script in beta 2? |
| 04:18.57 | DCGrendel | ok, working now :) |
| 04:19.03 | DCGrendel | mthx |
| 04:19.26 | file[laptop] | BleedingMe: app_voicemail automatically does that stuff, so addmailbox has been deprecated for a LONG time |
| 04:19.43 | BleedingMe | ah.. i kinda had a feeling... thanks for the info |
| 04:21.00 | Qwell | file[laptop]: you can add new users from app_voicemail? |
| 04:21.25 | file[laptop] | addmailbox was a script that made all the scripts and stuff |
| 04:21.28 | file[laptop] | er |
| 04:21.31 | file[laptop] | made all the folders |
| 04:21.39 | Qwell | oh...cheesy |
| 04:21.49 | mog_home | cheese! |
| 04:22.00 | file[laptop] | very very cheesy |
| 04:22.06 | Qwell | mmm |
| 04:22.48 | *** join/#asterisk cjk_ (n=cjk@11.121.9.213.dsl.getacom.de) |
| 04:23.05 | asterboy | ok that web server is back...here is the live link: http://www-ccs.cs.umass.edu/~shri/iPic.html |
| 04:23.16 | asterboy | my new asterisk box :P |
| 04:25.01 | IronHelix | i wonder how many channels of ILBC that can encode... |
| 04:25.02 | IronHelix | :) |
| 04:27.52 | *** part/#asterisk Corndawg_ (i=whoisit@c-66-176-66-83.hsd1.fl.comcast.net) |
| 04:28.26 | chandi | IronHelix I've found the MWI problem :I and I'm kind of shy |
| 04:28.48 | chandi | IronHelix so next time I come here I'll have a different nickname ;) |
| 04:28.57 | IronHelix | lol |
| 04:29.00 | IronHelix | whats the problem? |
| 04:29.03 | IronHelix | and dont be shy |
| 04:29.05 | IronHelix | or change your nick |
| 04:29.07 | IronHelix | thats dumb |
| 04:29.09 | chandi | hahah |
| 04:29.13 | IronHelix | we've all been n00bs at one point |
| 04:29.22 | IronHelix | im sure i've asked my share of dumb questions |
| 04:30.00 | chandi | IronHelix I was just kidding. I did define my box twice in sip.conf, once with mailbox=120 and the second without any maibox config |
| 04:30.11 | chandi | IronHelix so the last one was taken into account |
| 04:30.18 | IronHelix | :) good job |
| 04:30.36 | chandi | IronHelix I guess I did cut and paste it twice from my 1.0.9 config file |
| 04:30.50 | chandi | IronHelix thanks a lot for your time :) |
| 04:31.06 | IronHelix | np |
| 04:31.10 | IronHelix | glad it works :) |
| 04:31.53 | chandi | IronHelix any idea where to look to set my phone to ring with a distinctive ringer ? |
| 04:32.21 | IronHelix | hmmm |
| 04:32.25 | asterboy | thats exentensions.conf no? |
| 04:32.44 | chandi | asterboy what kind of config? |
| 04:32.57 | chandi | I mean what apps ? |
| 04:33.21 | asterboy | I was asking something like this earlier...2 fxo --- 1 fxs |
| 04:34.09 | chandi | ahh |
| 04:34.14 | asterboy | · http://www.marko.net/asterisk/archives/0205/0140.html |
| 04:34.14 | asterboy | (19:24) · lenne_dk · exten => 1,1,Dial,Zap/28 ; Ring Zap/28 normally |
| 04:34.14 | asterboy | (19:24) · lenne_dk · exten => 2,1,Dial,Zap/28r1 ; Ring Zap/28 with ring #1 |
| 04:34.14 | asterboy | (19:24) · lenne_dk · exten => 3,1,Dial,Zap/28r2 ; Ring Zap/28 with ring #2 |
| 04:34.14 | asterboy | (19:26) · lenne_dk · SIP doesn't support distinctive ring, I believe |
| 04:34.29 | IronHelix | chandi http://www.voip-info.org//tiki-index.php?page=Sipura+SPA-2000 set ALERT_INFO to Bellcore-rx (replace x with 1-8) |
| 04:34.41 | chandi | great!! |
| 04:34.44 | chandi | thanks |
| 04:34.57 | asterboy | your doing it with your phone...I'm doing it on the fxs channel. |
| 04:35.41 | chandi | oh gosh..It's late ;) time to go to bed |
| 04:35.43 | chandi | thanks guys! |
| 04:36.01 | IronHelix | np |
| 04:36.02 | IronHelix | nite |
| 04:36.04 | stbain | another satisfied customer... off to dream about all things asterisk |
| 04:36.12 | chandi | stbain hahaha |
| 04:36.24 | IronHelix | customer? he left without paying! |
| 04:36.37 | stbain | he paid us by gracing us with his presence |
| 04:36.38 | DCGrendel | under what conditions would app_meetme.so not be built during install? |
| 04:36.51 | stbain | (or something like that) |
| 04:36.53 | *** join/#asterisk nords2 (n=nords@S0106001217abcbc3.no.shawcable.net) |
| 04:37.12 | IronHelix | damn :( i was hoping for dollars |
| 04:37.44 | nords2 | I am getting a core dump when I send an originate command through the Asterisk Manager Interface with a variable |
| 04:37.57 | nords2 | works fine without any variables |
| 04:38.27 | asterboy | Another good phone? http://cgi.ebay.ca/SPA-841-Sipura-Residential-Phone-Basic-SIP-Phone-Re_W0QQitemZ5826438969QQcategoryZ61840QQssPageNameZWD1VQQrdZ1QQcmdZViewItem |
| 04:38.35 | nords2 | my backtrace: 0x080a8e7c in astman_get_variables (m=0x0) at manager.c:328 |
| 04:38.35 | nords2 | #1 0x00000000 in ?? () |
| 04:39.21 | DCGrendel | ah, lacking kernel source + zaptel driver :P |
| 04:39.29 | devonst17 | Why are SIP phones so damn expensive/ |
| 04:39.40 | DCGrendel | denon: $50 is expensive? |
| 04:39.53 | IronHelix | http://www.voipsupply.com/product_info.php?products_id=322 same thing asterboy |
| 04:39.53 | DCGrendel | er devonst17 |
| 04:40.04 | devonst17 | lol |
| 04:40.23 | DCGrendel | $55 for a Budgetone 101 |
| 04:40.30 | IronHelix | cheaper from voipsupply and it doesnt say on the auction if thats a 4 line or 2 line version |
| 04:40.34 | asterboy | ah thats better. |
| 04:40.34 | DCGrendel | at that store IronHelix just said |
| 04:40.45 | asterboy | true |
| 04:40.46 | nords2 | seems to have been introduced in the last week, because i working versions from early Nov |
| 04:40.51 | asterboy | and the url is damn long! |
| 04:41.02 | Sedorox | I don't recommened a budgetone unless you really want only the basics.... |
| 04:41.08 | DCGrendel | heh |
| 04:41.09 | asterboy | how do I make those clickable? |
| 04:41.12 | *** part/#asterisk wmandra (n=wmandra@bgp504084bgs.verona01.nj.comcast.net) |
| 04:41.13 | Sedorox | I've had it less then a year and have grown out of it :p |
| 04:41.18 | asterboy | "/" something. |
| 04:41.24 | Sedorox | but unfortinatly I don't have the money for anything better at the moment |
| 04:41.43 | IronHelix | asterboy the 841 is a nice phone, the only problem is the keypad is annoying on some copies of the phone (rubberized button gets stuck below the plastic faceplate) and it has no LCD backlight |
| 04:41.59 | asterboy | yikes.. |
| 04:41.59 | IronHelix | asterboy- its a client feature, just type the url or click it. Make sure the URL has a space before and after it |
| 04:42.09 | IronHelix | its not that huge a problem from what i hear |
| 04:42.12 | asterboy | ah |
| 04:42.18 | IronHelix | but enough to be worthy of mention |
| 04:42.24 | *** join/#asterisk rikstah (n=rick@62.6.163.90) |
| 04:42.34 | IronHelix | also, the 841 is VERY configurable |
| 04:43.00 | IronHelix | with config files on a tftp server you can customize down to blink patterns of the LED lights |
| 04:43.40 | asterboy | nice |
| 04:44.24 | DCGrendel | IronHelix: backlights are easy to add :) |
| 04:44.41 | IronHelix | hehe i debated doing just that |
| 04:44.48 | IronHelix | ended up going with a gxp2000 (grandstream) tho |
| 04:44.59 | IronHelix | also asterboy- the spa841 is SMALL |
| 04:45.00 | asterboy | how do you like the gxp? |
| 04:45.02 | DCGrendel | they want $30 to install the firmware update for you |
| 04:45.09 | asterboy | yuk! |
| 04:45.19 | IronHelix | its not the firmware update, its enabling the other 2 lines |
| 04:45.29 | IronHelix | you can update the firmware, it will just stay a 2 line phone |
| 04:45.33 | DCGrendel | same thing isnt it? |
| 04:45.35 | DCGrendel | wth? |
| 04:45.36 | IronHelix | no |
| 04:45.53 | IronHelix | firmware upgrade = 1.1.2 to 1.1.5. phone config stays the same |
| 04:46.09 | DCGrendel | ok, well can't you edit the phone config w/o paying them to do it for you? |
| 04:46.12 | IronHelix | the $30 upgrade = fimrware stays at v.whatever, only now you get 4 sip registrations instead of 2 |
| 04:46.16 | IronHelix | sure you can |
| 04:46.23 | DCGrendel | so why pay them $30 |
| 04:46.24 | IronHelix | you can edit and flash whatever you want |
| 04:46.31 | IronHelix | to turn the 2 line model into the 4 line model |
| 04:47.30 | iCEBrkr | Thank you Fedora Core 4, for making me switch to a different flavor of distro.. YOU FUCKING PIECE OF SHIT |
| 04:47.35 | IronHelix | firmware upgrades you can do as much as you want for free. The $30 is to unlock the extra 2 lines |
| 04:47.39 | DCGrendel | iCEBrkr: HEH. |
| 04:47.52 | DCGrendel | iCEBrkr: i just installed CentOS 4.2 and A@H ontop of it. |
| 04:47.53 | iCEBrkr | apt-get is all jacked up |
| 04:47.53 | IronHelix | whats wrong with FC4? |
| 04:47.56 | iCEBrkr | yum doesn't have shit |
| 04:48.03 | IronHelix | lol |
| 04:48.05 | iCEBrkr | Can't even RPM install Pine |
| 04:48.19 | DCGrendel | iCEBrkr: why would you want that old wooden email reader? |
| 04:48.28 | iCEBrkr | DCGrendel: Cuz I'm old Skewl, yo. |
| 04:48.38 | DCGrendel | not old school enough |
| 04:48.40 | Qwell | pine is "non free" |
| 04:48.43 | Qwell | or something |
| 04:48.59 | iCEBrkr | DCGrendel: Plus, I'm always SSH'd in, I don't use those crazy GUI based email readers :) |
| 04:49.16 | iCEBrkr | Qwell: hehe yea, the re-distrib. license is wonky |
| 04:49.22 | DCGrendel | nah, use the old one, mail or whatever it was |
| 04:49.28 | IronHelix | but yeah asterboy- the gxp is a nice phone, with firmware 1.0.1.12 the speakerphone doesnt suck ass anymore (finally implemented AEC) and as of 1.0.1.13 it supports real intercom (with alert info) and BLF (asterisk hint LED) |
| 04:49.29 | iCEBrkr | ew |
| 04:50.14 | DCGrendel | i've not used pine in ages, but mail i practially memorised how to erase the logs i'm always getting filling my mailbox :) |
| 04:50.31 | Druken | d * |
| 04:50.32 | Druken | :) |
| 04:50.36 | DCGrendel | yeps |
| 04:50.42 | iCEBrkr | DCGrendel: I used 'mail' back when I was at 1200 baud on our local FreeNet. |
| 04:50.45 | iCEBrkr | No thanks. :) |
| 04:50.46 | *** join/#asterisk SwK (n=SwK@dpc6745230018.direcpc.com) |
| 04:50.48 | DCGrendel | but h first to make sure theres nothing important there |
| 04:51.00 | asterboy | For the money it seems choosing the polycom over the gxp is a better choice...although the ploycom 301 is only a 2 line phone. |
| 04:51.06 | iCEBrkr | I've tried to use Mutt.. I just don't like it |
| 04:51.14 | DCGrendel | iCEBrkr: i use mail on my 1200 baud serial console to my vax sitting 8ft from me. |
| 04:51.25 | iCEBrkr | and I can't force myself to use SquirrelMail, tho I'm always sure to have it installed and configured :-/ |
| 04:51.53 | DCGrendel | nothing better than a 200w 1984 space heater made by DIGITAL |
| 04:51.57 | asterboy | I wish the number of lines was a function of the fxs port. |
| 04:52.02 | iCEBrkr | haha |
| 04:52.20 | DCGrendel | with 32MB of ram! |
| 04:52.30 | iCEBrkr | LOOK OUT! |
| 04:53.01 | iCEBrkr | That's funny, I just rebuilt this laptop from scraps and it's got a 128m of ram and a whopping 32meg video card ( shared, of course ) |
| 04:53.05 | IronHelix | yeah, * needs better support for SLA (shared line appearances), so it can emulate a key system for people with primarily analog systems |
| 04:53.22 | stbain | I remember when I bumped my Pentium 133 from 16MB to 32MB of RAM. Quake ran so much better. |
| 04:54.04 | IronHelix | speaking of which... does anybody know of a method or script that when called will figure out who's using a ZAP channel, pull them and the zap channel into a meetme conference and then connect you to it? |
| 04:54.44 | devonst17 | Anyone know how to configure Asterisk on MacOSx? |
| 04:54.45 | IronHelix | so someone can yell BOB PICK UP LINE 3 and not have to xfer or park the call |
| 04:54.58 | devonst17 | :) |
| 04:55.14 | Qwell | key systems are kinda useless... |
| 04:55.33 | Sedorox | devonst17: there are a few in here who use * on OS X.... but I'm not one of them |
| 04:55.42 | Sedorox | I dunno if nay are here right now.. but it seems to work fine |
| 04:55.47 | IronHelix | true, but being able to forcably enter a conversation has uses in a smaller office |
| 04:55.49 | Sedorox | 'cept maybe FXO... *shrugs* |
| 04:56.01 | nords2 | anyone what to help me debug a core dump? |
| 04:56.25 | devonst17 | aite... its too late... I will ask again later, thanks Sedorox |
| 04:56.49 | stbain | IronHelix: can you park the call in a special parking lot and then have that parking spot's pre-designated Meetme conference pulled from a database? |
| 04:57.08 | Sedorox | hehe |
| 04:57.16 | IronHelix | yeah but that requires the person that answered the call to do something |
| 04:57.57 | IronHelix | as i see it- call comes in on zap and is answered by (whatever). call is connected. At this point if (another guy) wants to enter the convo, he can't. He can zapbarge and listen, but he cant talk |
| 04:58.20 | IronHelix | brb |
| 04:59.25 | Qwell | meh, just send all calls to meetme, heh |
| 04:59.56 | Qwell | "dial 6024" "Bob is currently on a call. Press 1 to join him." |
| 05:01.05 | stbain | IronHelix: http://www.asterlink.com/svp/00README |
| 05:01.30 | stbain | that help? |
| 05:01.35 | IronHelix | checking |
| 05:01.45 | IronHelix | qwell- i'd be happy to but then how to call actual people? |
| 05:02.08 | IronHelix | i guess you'd need to generate a call file somehow that would dial from the conference to (extensiosn, voicemail, whtaever) |
| 05:02.29 | joelsolanki | Does the transcoding from g711 to g729 takes huge bandwidth ? i have linksys pap2 ..it only support g729 at a time so the other call made is g711 and transcoded to g729 .does it take huge bandwidth ? |
| 05:02.59 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 05:03.11 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 05:03.44 | justinu | transcoding takes CPU, not bandwidth |
| 05:03.57 | *** join/#asterisk grimse (n=grimse@p5481C97F.dip.t-dialin.net) |
| 05:04.15 | DCGrendel | justinu: it takes bus bandwidth... |
| 05:04.35 | DCGrendel | not that it applies |
| 05:04.43 | justinu | ok, try and make him right... :P |
| 05:04.50 | DCGrendel | not gonna |
| 05:04.55 | joelsolanki | oh ok. i was not getting clarity while using transcoding. i will look on this particular ..ip how much it consumes the bandwidht |
| 05:05.11 | IronHelix | stbain- i will look more into that tomorrow |
| 05:05.13 | justinu | g729 is like 10kb/sec? |
| 05:05.13 | IronHelix | might be what i need |
| 05:05.23 | IronHelix | little under 10 |
| 05:06.04 | justinu | anyone run g729 over dialup? |
| 05:06.13 | justinu | does it work? |
| 05:06.21 | IronHelix | joel- which codec you choose affects how much bandwidth you use. g.711 ulaw (ulaw, pcmu, 711u, etc) uses the most, about 64kbit/sec. G.729 uses the least at about 8kbit/sec. |
| 05:06.44 | IronHelix | joel- converting one codec to another uses cpu power. however which codec ends up being used will determind bandwidth |
| 05:07.18 | IronHelix | i assume you mean the pap2 can only do one 729 call at a time, so the other one goes to your next codec selection |
| 05:07.23 | justinu | ironhelix: when do you start the training course for asterisk n00bs? |
| 05:07.25 | IronHelix | and 711u does use huge bandwidth |
| 05:07.33 | IronHelix | ? |
| 05:07.53 | justinu | you seem like a natural teacher |
| 05:07.58 | IronHelix | hahahaha |
| 05:08.05 | stbain | justinu: problem running over dialup w/ g729 isn't so much the bandwidth (8kb < 33.6kb upstream on a 56k connection), but latency |
| 05:08.12 | IronHelix | i dunno |
| 05:08.16 | justinu | stbain: yeah, i figured |
| 05:08.34 | asterboy | I'm going to get the Poloycom SP-601...looks like a great demo to take around and showcase to potential clients. |
| 05:08.46 | justinu | asterboy: it's awesome |
| 05:08.47 | asterboy | Plus it will work well in my office of 4 lines. |
| 05:08.59 | IronHelix | asterboy- you'll have fun with that. keep in mind tho, your clients may not want $300 phones for everybody |
| 05:09.03 | asterboy | I'm tired of people telling me I sound like I'm talking in a toilet. |
| 05:09.07 | justinu | lol |
| 05:09.11 | justinu | so get a ip301 |
| 05:09.20 | IronHelix | my advice, if you have some free cash, is get a few different models |
| 05:09.31 | IronHelix | learn their quirks and you'll know which ones to pitch to which clients |
| 05:09.39 | asterboy | yes, but I want the wow factor, and then let them decide to downgrade |
| 05:09.59 | asterboy | The expansion module is fantastic. |
| 05:10.03 | IronHelix | if you want real wow factor get one of those $800 cisco phones that can surf the web and run word files |
| 05:10.06 | asterboy | simple and clean looking. |
| 05:10.09 | stbain | yes... always show the Caddilac before you take them to the other side of the lot and offer to sell them the Pinto |
| 05:10.19 | asterboy | lol |
| 05:10.23 | IronHelix | lol |
| 05:10.25 | justinu | wow, ztdummy is not working for me |
| 05:10.30 | justinu | meetme sound is terrible |
| 05:10.50 | bumble | so asterisk doesn't work with mpg321? |
| 05:11.01 | stbain | bumble: not from everything I've read |
| 05:11.18 | stbain | hence the handy "make mpg123" included w/ the asterisk MakeFile |
| 05:11.24 | asterboy | IronHelix: You mean Cisco CP-7970G Color TouchScreen |
| 05:11.29 | IronHelix | yea |
| 05:11.37 | asterboy | that is sweet! |
| 05:11.38 | bumble | ahhhhhhhhhhhhhhh... |
| 05:11.54 | IronHelix | only problem is there you DO have to pay for firmware |
| 05:12.16 | asterboy | extra? |
| 05:12.16 | IronHelix | cisco makes good products, but i stay away from them for that reason |
| 05:12.19 | IronHelix | yup |
| 05:12.21 | IronHelix | support contract |
| 05:12.24 | asterboy | how much? |
| 05:12.33 | asterboy | support contract??? yikes! |
| 05:12.53 | IronHelix | i dont know exactly, i've heard everywhere from $7 to $100 |
| 05:12.57 | asterboy | I'll make my own touch screen interface for that price. |
| 05:13.07 | IronHelix | and i've also heard the sign up process is NOT pleasant or automated in any way |
| 05:13.20 | asterboy | forget that nonesense. |
| 05:13.34 | stbain | IronHelix: you mean you have to buy Smartnet? |
| 05:13.36 | bumble | but what about the buffer overflow in mpg123? |
| 05:16.16 | asterboy | is the 601 web interfaced? |
| 05:16.16 | IronHelix | maybe thats it |
| 05:16.22 | IronHelix | i just hate the concept of paying for firmware |
| 05:16.36 | *** part/#asterisk nords2 (n=nords@S0106001217abcbc3.no.shawcable.net) |
| 05:16.37 | Qwell | smartnet for a cisco phone is about $13 |
| 05:17.34 | Qwell | Does anybody happen to know if it's legal to use a cisco phone (sccp or sip) on asterisk, without paying the $98 for the license? |
| 05:17.42 | Qwell | I assumed the license was for ccm/ccme |
| 05:18.03 | IronHelix | afaik yeah the license is for CCM |
| 05:18.17 | IronHelix | they wouldnt sell the phone itself it you werent licensed to use it |
| 05:18.22 | DCGrendel | heh |
| 05:18.26 | DCGrendel | sure they would! |
| 05:18.55 | DCGrendel | these are the same folks who require you sell them your soul to download drivers for your wifi card. |
| 05:19.10 | IronHelix | lol |
| 05:21.30 | *** join/#asterisk tengulre (n=tengulre@222.90.66.156) |
| 05:23.27 | tengulre | hi,all |
| 05:23.52 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 05:23.54 | devonst17 | Night all |
| 05:26.37 | tengulre | does the asterisk support OEM X100P - FXO PCI Card |
| 05:26.45 | tengulre | ? |
| 05:26.49 | IronHelix | yes |
| 05:27.12 | ManxPower | tengulre, No. Asterisk support |
| 05:27.27 | ManxPower | 's Digium's X100P cards. Other cards are compatable with the Digium card. |
| 05:27.40 | IronHelix | ah yes good point |
| 05:27.54 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 05:28.00 | IronHelix | if your question is will it work, the answer is yes. however official x100's are out of production, so this is a clone card |
| 05:28.09 | IronHelix | it will work fine, but digium wont support it |
| 05:28.09 | tengulre | ManxPower, I want to buy a this card, but I m at china! |
| 05:28.29 | tengulre | :( |
| 05:28.37 | justinu | where in china? |
| 05:28.56 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 05:29.04 | tengulre | justinu: Me! |
| 05:29.18 | IronHelix | meh |
| 05:29.19 | justinu | what city |
| 05:29.19 | IronHelix | im out |
| 05:29.19 | IronHelix | nite |
| 05:29.22 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 05:29.26 | justinu | bye |
| 05:29.30 | mog_home | stupid sloppy focus |
| 05:30.26 | tengulre | justinu: Xi'an! |
| 05:31.17 | justinu | neat |
| 05:31.35 | justinu | you know of any ITSP's selling DIDs in taiwan? |
| 05:33.27 | konfuzed | any debian users in tonight |
| 05:34.05 | konfuzed | installing this box on sarge and want to know if I should go with testing or stick with stable |
| 05:35.28 | konfuzed | never used deb before and there could be somethin that I need out of testing for happy use with asterisk or FM or something |
| 05:39.39 | asterboy | IronHelix: nite, thanks for your help! |
| 05:43.24 | *** join/#asterisk clive- (n=pirch@ndn-165-132-91.telkomadsl.co.za) |
| 05:46.39 | DJ-Pyro | konfuzed: is this machine in production or just for testing? |
| 05:48.46 | *** part/#asterisk DCGrendel (n=DCGrende@ip68-1-157-197.mc.at.cox.net) |
| 05:49.20 | DJ-Pyro | if you're in production stick with stable |
| 05:49.27 | DJ-Pyro | otherwise you can play in testing land |
| 05:50.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 05:56.45 | tengulre | hi,all |
| 05:57.02 | tengulre | what's the FXO and FXS means? |
| 05:57.13 | Qwell | ~fxsfxo |
| 05:57.17 | jbot | [fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. An FXS port expects to provide dialtone and provide ring voltage. |
| 05:57.20 | mog_home | wow |
| 05:57.27 | mog_home | i didnt think that would work qwell |
| 05:57.41 | Qwell | ManxPower added both |
| 05:58.25 | tengulre | jbot: thanks! |
| 05:58.25 | jbot | tengulre: no worries |
| 05:58.31 | asterboy | Interesting there are problems with the Polycom 601 and the expansion module. |
| 05:58.32 | asterboy | http://voxilla.com/PNphpBB2-viewtopic-t-6350.html |
| 06:04.17 | *** join/#asterisk svenna_ (n=svenna@p548D28D0.dip0.t-ipconnect.de) |
| 06:05.03 | Qwell | mog_home: I talked to my boss about contribing changes I make at work back...he said it was cool |
| 06:05.22 | Qwell | So, essentially, I'll be getting paid to write open source code. :D |
| 06:08.01 | asterboy | nice |
| 06:08.44 | mog_home | YAY! |
| 06:08.51 | mog_home | you rock qwell |
| 06:09.01 | Qwell | if I don't quit...heh |
| 06:09.02 | mog_home | what you been working on? |
| 06:09.15 | Qwell | just chan_sccp lately |
| 06:09.48 | mog_home | the real one or the sccp one sourceforge |
| 06:09.55 | Qwell | the one from berlios.de |
| 06:10.56 | Qwell | which one is the "real" one? Besides chan_skinny |
| 06:11.24 | mog_home | skinny |
| 06:11.38 | mog_home | i didnt realize |
| 06:22.37 | *** join/#asterisk Igbothom (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 06:23.57 | rajiv | i have a phone with a 'function' key. is there a standard way to configure phone keys via SIP ? |
| 06:28.04 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 06:28.28 | YoMama | wcusb and the S100U are as crappy as a baby's diaper |
| 06:29.24 | Qwell | s100u? |
| 06:29.38 | YoMama | Qwell: the USB FXS thing that Digium used ot make |
| 06:29.45 | YoMama | "supported" by the wcusb driver |
| 06:29.57 | Qwell | oh |
| 06:30.23 | wasim | has anybody worked with SIPI interconnect? is it a standard? |
| 06:31.35 | *** join/#asterisk Callum (n=callummc@ns2.ains.net.au) |
| 06:31.53 | asterboy | Anyone tried the Aastra 480i CT IP SIP Phone??? |
| 06:32.18 | *** join/#asterisk zobia (n=laura_sh@218.6.242.212) |
| 06:32.25 | zobia | Hello everyone |
| 06:32.36 | asterboy | yo |
| 06:33.36 | zobia | is there any way to reload the asterisk 's conf files automatically , like by addons etc? |
| 06:34.00 | zobia | I want to reload these config files or asterisk by remote way |
| 06:34.43 | asterboy | asterisk -r 'extensions reload' |
| 06:34.52 | asterboy | pwd |
| 06:34.55 | asterboy | oops |
| 06:35.39 | Callum | Hi all....Having some (lots) of problems with my * machine today.... anyone care to help me with some troubleshooting ? |
| 06:36.07 | YoMama | Callum: i'm about to fall asleep but i'll do what i can |
| 06:36.56 | Callum | thx |
| 06:37.08 | Callum | Basically... we have our lines maxing out |
| 06:37.30 | Callum | We have an ISDN30 (PRI) line.... lots of call queues etc |
| 06:38.29 | Callum | getting lots of messages like; |
| 06:38.30 | Callum | chan_zap.c:8311 pri_dchannel: Ring requested on channel 0/17 already in use on span 1. Hanging up owner. |
| 06:38.38 | tzafrir_laptop | zobia, what do you mean "automatically"? when? |
| 06:38.39 | Callum | when the box is melting down |
| 06:38.49 | Callum | any thoughts ? |
| 06:40.13 | tzafrir_laptop | Callum, sanity check: you do have free channels, right? |
| 06:40.20 | Callum | yes... |
| 06:40.23 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 06:40.27 | Callum | we rarely use the full 30 |
| 06:40.30 | zobia | hello asterboy |
| 06:40.36 | wasim | Callum: are you specifying Dial(Zap/17) ? |
| 06:40.37 | zobia | thank you for answer |
| 06:40.38 | Callum | I have some debug output to post to anyone interested |
| 06:40.44 | asterboy | hey zobia! |
| 06:41.12 | Callum | (I'm new to the whole IRC thing - so if someone can tell me how to do that without spamming everyone, that would be great !) |
| 06:41.25 | zobia | i want to reload the extensions.conf remotely like from web , do u know how to resolve this? |
| 06:41.26 | wasim | pastebin.ca |
| 06:41.43 | asterboy | cgi |
| 06:41.50 | asterboy | perl or php |
| 06:42.11 | zobia | but the web server is not on asterisk server |
| 06:42.34 | litage | how does scopserv (www.scopserv.com) compare with what asterisk 1.2.0 natively has? |
| 06:42.39 | asterboy | ok, so you have to communicate to the remote somehow |
| 06:42.40 | Callum | wasim, I'm not sure what you mean when you say specifying Dial(Zap17).... |
| 06:42.49 | Callum | why would you dial a Zap channel ? |
| 06:42.49 | asterboy | say ftp, ssh, rlogin |
| 06:42.57 | asterboy | email |
| 06:43.00 | Callum | (please excuse my ignorance) |
| 06:43.18 | wasim | Callum: just trying to figure out if you're hardcoding chans in the dialplan |
| 06:43.24 | zobia | okay. i will try telnet to 5038 first then do it |
| 06:43.26 | asterboy | cause that's just the command to "bridge" |
| 06:43.38 | zobia | thank you. |
| 06:43.39 | asterboy | don't focus too much on the "dial" part |
| 06:44.12 | asterboy | telnet should work, but its fun trying to command line that. |
| 06:44.12 | Callum | no we are not |
| 06:44.44 | Callum | we are using SIP Phones (mix of Polycom 300's, 600's & Grandstream 2000's) |
| 06:45.44 | tzafrir_laptop | ~pastebin |
| 06:45.45 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca/ |
| 06:46.13 | Callum | ok here it is; http://pastebin.ca/28636 |
| 06:46.16 | YoMama | sheesh |
| 06:46.24 | Callum | this is the output taken from some time ago |
| 06:46.31 | Callum | but the same thing is happening today |
| 06:46.38 | Callum | over and over again |
| 06:47.32 | wasim | hmm ... its showing all your channels are in use, so the err is right |
| 06:47.38 | Callum | ok |
| 06:47.53 | Callum | but the strange thing is that the channels are not in use... |
| 06:48.12 | Callum | when people get through to our queues, they do not hear music on hold |
| 06:48.22 | Callum | and consequently hang up |
| 06:48.37 | Callum | but the call seems to be stuck in the channels list |
| 06:48.44 | Callum | calls never get through to agents |
| 06:48.55 | Callum | but music on hold and outbound calls all still work |
| 06:49.00 | wasim | Callum: post your zapata.conf and your extensions.conf |
| 06:49.06 | Callum | (until Asterisk has a fit and shuts down) |
| 06:51.02 | Callum | ok - zapta is http://pastebin.ca/28638 |
| 06:52.06 | Callum | Ext.conf - http://pastebin.ca/28639 |
| 06:53.00 | infinity1 | tzafrir_laptop: thanks for the work on making deb packages! |
| 06:53.58 | *** part/#asterisk Maksim (n=max@213.142.207.20) |
| 06:56.14 | *** join/#asterisk lehel (n=lehel@82.79.20.17) |
| 06:57.22 | wasim | Callum: ains.conf too please |
| 06:59.20 | wasim | and please egrep -v ";" prior to pasting |
| 06:59.23 | Callum | ains.conf http://pastebin.ca/28640 |
| 06:59.46 | Callum | oh - sorry, missed that one - will do on next post |
| 07:00.17 | asterboy | You know, after reviewing all the different phones...they are so primative...should be much farther ahead. |
| 07:00.49 | asterboy | color touch displays with totally cusomisable html displays. |
| 07:01.00 | Qwell | like the 7970? |
| 07:01.10 | asterboy | yes, but they want a kings ranson. |
| 07:01.13 | asterboy | ransom |
| 07:01.16 | Qwell | $500? |
| 07:01.19 | Qwell | That isn't a lot |
| 07:01.22 | asterboy | and charge you a support fee. |
| 07:01.28 | Qwell | $15? |
| 07:01.40 | asterboy | is that 1 time? |
| 07:01.47 | Qwell | is support ever 1 time? |
| 07:01.58 | asterboy | lol...so yearly? |
| 07:02.02 | Qwell | yeah |
| 07:02.13 | asterboy | hmmm |
| 07:03.43 | joelsolanki | hello all: i m using sip.conf for connecting to my provider. I have buyed and 6 license of g729 and installed in asterisk. when i call and monitor the bandwidth usage it is taking 27 to 30 kbps. i m using linksys pap2 as an endpoint. my setup consist of ser+asteisk |
| 07:03.55 | joelsolanki | any hints what could be wrong on my side. |
| 07:04.10 | asterboy | how customisable is the display of the 7970? |
| 07:04.25 | asterboy | how about a company logo or picture pop up on a caller id? |
| 07:04.54 | tzafrir_laptop | infinity1, hi |
| 07:05.04 | asterboy | wireless headset? |
| 07:05.33 | wasim | Callum: i got lost, perhaps what you need to do, is build a test dialplan entry which just sends it to queues and see whats happening |
| 07:05.43 | Qwell | asterboy: you could probably write something. And they sell third party headsets that are wireless |
| 07:06.13 | Qwell | like plantronics |
| 07:06.45 | asterboy | How about something that interfaces with the computer to allow a web page to do all the functions of transfer hold,etc. |
| 07:07.22 | asterboy | The astra looks good, however, it sounds like it is still in transition. |
| 07:07.28 | *** join/#asterisk kumamoto (n=eryco@68-116-142-128.dhcp.ftwo.tx.charter.com) |
| 07:07.36 | joelsolanki | as per docs g729 take 8 kbps bandwidth but in my case it take 3 times more :( |
| 07:07.54 | wasim | joelsolanki: show channel and see what format |
| 07:09.47 | joelsolanki | wasim: let me call and check. |
| 07:10.52 | *** join/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
| 07:11.25 | Qwell | joelsolanki: http://www.digium.com/index.php?menu=faq#Codecs_0 |
| 07:12.19 | zobia | hello asterboy |
| 07:13.47 | joelsolanki | Qwell: i m looking for it. |
| 07:14.03 | Qwell | that link will take you right to it |
| 07:14.16 | Qwell | replace gsm with g729 |
| 07:15.16 | joelsolanki | ok |
| 07:16.47 | zobia | hello. doesn anyone know how to add a extension context dynamically to asterisk ? |
| 07:17.28 | Callum | wasim, the problem is that this issue takes a lot of time in production before it actually appears |
| 07:17.41 | Callum | so a test dial plan will not cut it, unfortunately |
| 07:18.08 | Callum | the problem has been cropping up nearly once every two weeks |
| 07:18.22 | Callum | but today it has happened about 8-9 times already |
| 07:18.27 | Callum | (nothing has changed) |
| 07:20.23 | wasim | hmm |
| 07:20.56 | Callum | yes... :P I've been "hmmmmm"ing all day :D |
| 07:21.15 | *** part/#asterisk FaithX (n=FaithX@vg28.vodafone.com.au) |
| 07:21.28 | wasim | the only thing i can make out is that after the queue, the channels isn't being hungup, per se |
| 07:21.44 | Callum | yeah - that seems to be about it |
| 07:21.53 | *** join/#asterisk dasuberdavid (n=egg@pcp01534754pcs.huntsv01.al.comcast.net) |
| 07:22.16 | Callum | I've had a bug lodged for it |
| 07:22.17 | Callum | http://bugs.digium.com/view.php?id=5487&nbn=12 |
| 07:22.51 | zobia | can any one suggest a sharp guy to solve me ARA problem? i am looking for the answer for a longtime. but still no luck. |
| 07:23.00 | Qwell | zobia: ara? |
| 07:23.02 | Callum | and I have purchased Digium Support (or will do tomorrow when their back), but today it's just started going balistic, so I thought I'd see if anyone else can help |
| 07:23.37 | Qwell | ~ara |
| 07:23.45 | Qwell | asterisk realtime architecture, or something? |
| 07:23.51 | zobia | yes. The Asterisk Realtime Interface |
| 07:24.03 | Qwell | zobia: Whats the problem? |
| 07:24.36 | zobia | Qwell. thanks , my problem is can not add a context dynamically to my dialplan. |
| 07:24.50 | Qwell | Why not? |
| 07:25.24 | zobia | i created 2 context in the database . but if i jump from 1 to another. the second one could not be recognize. |
| 07:25.51 | zobia | but if i add switch=> context2@... to the dialplan it could recognize |
| 07:26.19 | zobia | but the point is i should not edit the extensions.conf manually. i need something could do automaticlly |
| 07:27.17 | Qwell | Is that what realtime switch is for? |
| 07:27.18 | wasim | hack ... make a #include workaround.conf ... have your script write to that, then reload |
| 07:28.34 | zobia | yes. |
| 07:28.42 | mog_home | <PROTECTED> |
| 07:28.51 | *** part/#asterisk bumble (n=b@69-160-145-156.ontrca.adelphia.net) |
| 07:28.53 | Qwell | mog_home: huh? What'd he tell you? |
| 07:29.15 | Qwell | Whatever he said, he lied |
| 07:29.51 | mog_home | lol |
| 07:30.03 | zobia | Qwell. then is there any way i could just update the database not touch the extensions.conf to create goto operation between two realtime context? |
| 07:30.30 | Qwell | zobia: I kinda thought that was the whole point of realtime, was to not have to touch extensions.conf. Adding contexts should be trivial. |
| 07:30.38 | Qwell | I'd have to RTFM, but... |
| 07:31.28 | zobia | Qwell i thought the same like u thought , but the truth is it could not do that. i already tested it |
| 07:32.25 | *** part/#asterisk opus_ (n=opus@dahphish.org) |
| 07:34.46 | zobia | Qwell: i paste it http://pastebin.ca/28642 |
| 07:35.21 | *** join/#asterisk Bullseye (n=bharatsa@210.211.246.47) |
| 07:35.23 | Qwell | zobia: Why are you adding that stuff in the dialplan? I thought you were doing realtime? |
| 07:35.40 | Bullseye | hello Qwek |
| 07:35.42 | Bullseye | Qwell |
| 07:36.00 | zobia | yes. i want to do extension realtime. |
| 07:36.12 | Bullseye | i am trying to configure my voicemail as realtime |
| 07:36.14 | zobia | no , this what i added in to my database |
| 07:36.14 | litage | "Qwek"...i like that =P |
| 07:36.37 | Qwell | Bullseye: fairly easy to do...check the wiki |
| 07:36.39 | Qwell | ~wikis |
| 07:36.40 | jbot | [wikis] http://www.voip-info.org |
| 07:36.46 | zobia | Qwell. i just tranlate what i added to my database records to the corresponding dialplan. |
| 07:36.59 | Bullseye | but when I have made the relevant changes i am gettingthe error on the CLI as "No entry in voicemail config file for '555" |
| 07:37.42 | Qwell | zobia: are you using Switch => ? |
| 07:37.46 | Qwell | erm, lowercase s |
| 07:38.38 | zobia | yes. |
| 07:38.45 | Qwell | Bullseye: did you setup extconfig.conf? |
| 07:38.51 | Qwell | zobia: What does that like look like? |
| 07:39.38 | zobia | Qwell: [context1] |
| 07:39.39 | zobia | switch => Realtime/context1@realtime_ext |
| 07:39.55 | Qwell | zobia: try this |
| 07:39.56 | zobia | [context2] |
| 07:39.57 | zobia | switch => Realtime/context2@realtime_ext |
| 07:40.04 | Qwell | switch => Realtime/@realtime_ext |
| 07:41.02 | zobia | oh. the manual said if we leave the context empty, it will find the current context . but anyway let me try |
| 07:41.23 | Qwell | dunno, I'm barely RTFMing |
| 07:41.28 | Bullseye | Qwell: i have added the line in the extconfig "voicemail => mysql,localpbx,voicemai" |
| 07:41.38 | Bullseye | where mysql is the driver |
| 07:41.53 | Bullseye | and voicemail is the table name |
| 07:43.06 | *** join/#asterisk Msalim5 (n=msalim_5@210.211.246.47) |
| 07:43.11 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 07:43.15 | Bullseye | Qwell, what role does the family name in the extcongfig.conf ? |
| 07:43.27 | Bullseye | thats the voicemail in my case |
| 07:43.49 | Qwell | Bullseye: it's driver,database[,table] |
| 07:45.08 | Bullseye | ya thats fine |
| 07:45.56 | *** join/#asterisk Msalim6 (n=msalim_5@210.211.246.47) |
| 07:46.01 | Qwell | Bullseye: Does it say anything when verbose or debug are up? |
| 07:46.10 | Qwell | something like not being able to connect for whatever reason, perhaps? |
| 07:47.36 | Bullseye | nop |
| 07:47.59 | Qwell | You have of course reloaded since you made the change? |
| 07:48.12 | Bullseye | yup |
| 07:48.15 | Bullseye | i have |
| 07:48.35 | Qwell | You should be getting messages when it tries to connect to the db, I'd think |
| 07:49.06 | zobia | Qwell: sorry i added switch => Realtime/@realtime_ext but could not recognize any of them |
| 07:49.09 | Bullseye | the voicemail.conf is gonna hold only the db login details ? dont they? |
| 07:49.40 | Qwell | Bullseye: I don't think it uses voicemail.conf at all when using realtime |
| 07:49.49 | Qwell | though, maybe |
| 07:50.02 | zobia | yes. bullseye, it use extconfig to config voicemail db connection |
| 07:50.08 | Qwell | I set it up at work, but...yeah |
| 07:50.09 | zobia | not voicemail.conf |
| 07:50.26 | Bullseye | so what zobia? |
| 07:50.45 | zobia | i config realtime of voicemail for mysql. it config it it in extconfig.conf |
| 07:51.00 | Qwell | Bullseye: unixodbc is installed and setup, right? |
| 07:51.42 | zobia | hello Qwell. any idea of my problem? |
| 07:51.53 | *** join/#asterisk otaku42 (i=otaku@madwifi/developer/otaku42) |
| 07:51.55 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 07:52.07 | otaku42 | hi all. |
| 07:53.03 | Bullseye | But Qwell, as I am using Mysql, does unixobdc installation might be one of the reason of my problem? |
| 07:53.13 | otaku42 | question: anyone knows an howto that explains how to use asterisk as "softphone"? in theory that should be possible, since it supports alsa for sound input/output... |
| 07:53.52 | *** join/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 07:54.07 | zobia | Bullseye. read http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail |
| 07:54.10 | wasim | otaku42: its simple |
| 07:54.31 | wasim | otaku42: you can use a manager api based app to create a visual interface or just use the console to Dial |
| 07:54.36 | wasim | otaku42: and answer, etc |
| 07:54.48 | nick125 | ok, quick question about meetme and agi: if i used agi meetme, would i have to redo the admin menu for it? |
| 07:55.00 | asterboy | $250 for a Polycom IP601 phone...anyone find a better price? |
| 07:55.12 | *** join/#asterisk Frawg (n=Frawg@unaffiliated/frawg) |
| 07:55.58 | wasim | nick125: wtf is an agi meetme? meetme called from an agi will operate the same way as called from the dialplan, or should, rather ... |
| 07:56.06 | otaku42 | wasim: yes, but how do i make use of alsa support? never did that so far, just read that it is there. |
| 07:56.15 | zobia | Qwell, are u there? |
| 07:56.36 | nick125 | wasim: well, im talking like when you add the b flag to the meetme() command |
| 07:56.51 | Qwell | otaku42: modules.conf load => chan_alsa.so |
| 07:57.23 | zobia | Qwell, can u give me a hint of my question? |
| 07:57.28 | Qwell | zobia: if I knew, sure |
| 07:57.29 | litage | how does scopserv (www.scopserv.com) compare with what asterisk 1.2.0 natively has? |
| 07:58.08 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 07:58.14 | zobia | Qwell, you also don't know why it could not jump between two context? |
| 07:58.29 | wasim | litage: seems to be basically * with config/management front ends |
| 07:58.32 | otaku42 | Qwell: and what about configuration? do you know if voip-info.org has something on that? |
| 07:58.50 | litage | wasim: does asterisk 1.2 come with any config/mgmt frontends? |
| 07:58.53 | Qwell | otaku42: there isn't much config |
| 07:58.58 | Qwell | litage: no |
| 07:59.32 | wasim | litage: its also based on 1.0.7 |
| 07:59.52 | otaku42 | Qwell: there should be at least a way to tell which alsa interface should be used for input and output, no? |
| 07:59.58 | wasim | otaku42: alsa.conf |
| 08:00.30 | otaku42 | wasim: ah... |
| 08:00.35 | litage | wasim: the pre-reqs say "Asterisk 1.0.7 or above" |
| 08:00.40 | otaku42 | wasim, Qwell: looking at it, thanks for your help. |
| 08:00.58 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:01.23 | zobia | Qwell , do u know any person could know this problem. i sent email to the author of this realtime code. but he didnot reply. i really need to resolve this badly. thank you inadvanced. |
| 08:04.16 | nick125 | is there a way to go and tell when a specific users joins and leaves a meetme? |
| 08:05.00 | wasim | nick125: put it in conf-background.agi |
| 08:05.18 | wasim | or use the |w flag |
| 08:05.35 | nick125 | but, in the voip-info page, it says that only works with ZAP channels |
| 08:06.20 | nick125 | does the conf-background.agi file already exist somewhere or do i have to create it from scratch? |
| 08:07.56 | nick125 | hrm...the MEETMESECS field looks interesting.. |
| 08:07.59 | Bullseye | Is there any command to check whether Asterisk is able to connect to the MySQL db for which it is configured? |
| 08:09.18 | nick125 | MEETMESECS is defined after the user disconnects? |
| 08:10.25 | Frawg | anyone running asterisk on fbsd? |
| 08:10.33 | nick125 | i am |
| 08:11.02 | Frawg | also running it on linux |
| 08:11.02 | Frawg | just wanna know if there is any difference in performance |
| 08:11.07 | nick125 | not that ive really noticed..least in my opinion |
| 08:11.13 | Frawg | ahhk |
| 08:13.16 | lehel | if i change the gain levels in zapata. conf do i need to restart *? |
| 08:13.23 | wasim | yep |
| 08:13.29 | lehel | ahh |
| 08:14.41 | lehel | wasim, while i'm doing ztmonitor.. on TX is always higher.. i mean i have echo on a zap chan |
| 08:15.13 | lehel | i have echocancel, echocanwhe..,echotraining=yes |
| 08:15.37 | wasim | echopraygoaway=yes |
| 08:15.39 | lehel | with rx/txgain can i adjust to normal? |
| 08:15.55 | wasim | lehel: sometimes reducing txgain can help |
| 08:16.14 | lehel | wasim, wht do you mean with praygoaway? |
| 08:16.49 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 08:17.41 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@d14-69-8-30.try.wideopenwest.com) |
| 08:17.48 | zobia | so desperate , no one can resolve. |
| 08:19.20 | nick125 | btw, for, meetme, do i have to enable any modules or such? |
| 08:20.37 | Qwell | zobia: Digium paid support is always an option |
| 08:21.30 | Qwell | nick125: app_meetme.so |
| 08:21.34 | zobia | thanks Qwell. |
| 08:21.37 | Qwell | and probably chan_zap |
| 08:21.57 | nick125 | Qwell: hrm, in my modules folder, i dont see app_meetme, is it in the asterisk-addons? |
| 08:22.19 | Qwell | no. You need to compile/install zaptel before you compile asterisk, or it won't be there |
| 08:22.27 | nick125 | aah |
| 08:22.29 | Qwell | afaik anyhow |
| 08:22.43 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:22.58 | nick125 | i hope compiling zaptel on my freebsd box isnt too hard :/ |
| 08:23.13 | Frawg | what release/version? |
| 08:23.15 | lehel | Qwell, any idea why do i have echo en zap when in call with iax/sip virtual, and none when in call with pstn? |
| 08:23.17 | nick125 | 6 |
| 08:23.21 | Frawg | ahh |
| 08:23.33 | Qwell | lehel: no, I don't do the whole echo thing |
| 08:23.37 | Frawg | i took my devel box to 6 today |
| 08:23.40 | Frawg | (from 5.4) |
| 08:23.47 | Frawg | only thing that broke was freeradius |
| 08:23.49 | mog_home | yay for cs homework |
| 08:23.55 | lehel | ok;\ |
| 08:23.57 | mog_home | i get to write a calculator |
| 08:23.59 | Frawg | recompiing with ULE instead of 4BSD |
| 08:24.00 | mog_home | !!! |
| 08:24.01 | Frawg | heh |
| 08:24.06 | Qwell | Who's the echo guy from oz? |
| 08:24.28 | lehel | "oz"?;) |
| 08:24.38 | Qwell | oz-tray-lea |
| 08:25.13 | nick125 | wow, i jsut relized that i had 42 firefox tabs open |
| 08:25.16 | mog_home | with not one but 4 functions!!! |
| 08:25.17 | c0w | Callum, i'm having the same issue only not using it as callcentre, |
| 08:25.33 | Qwell | eh...I forget his name |
| 08:25.41 | Qwell | he's like the king of echo though |
| 08:26.12 | lehel | Qwell, please tell me if u remember |
| 08:26.13 | nick125 | wow |
| 08:26.15 | c0w | Callum, i have the box in this state at the moment, and mark said he will look when he has time. |
| 08:26.28 | nick125 | this zaptel module which isnt supposed to work with 6.0 just worked with 6.0 |
| 08:26.41 | nick125 | 32 7 0xc7ddf000 2e000 zaptel.ko |
| 08:27.14 | Frawg | ;) |
| 08:27.25 | Frawg | heh |
| 08:27.32 | Frawg | swapped to ULE |
| 08:27.32 | Qwell | lehel: <X-Rob> we have _awful_ lines out here. I'm the fucking GOD on echo. |
| 08:27.39 | Qwell | Self proclaimed god of echo. :p |
| 08:27.49 | lehel | lool |
| 08:27.55 | lehel | thanks;) |
| 08:28.01 | nick125 | lol |
| 08:28.04 | nick125 | Doesn't work on 6.x yet. Expected to be complient soon. < lol |
| 08:33.38 | marcus2 | so will there be a new zaptel with 1.2? |
| 08:33.45 | Qwell | yes |
| 08:34.29 | marcus2 | oh, i see it now |
| 08:36.58 | *** join/#asterisk konfuzed (n=KonfuzeD@H129.C72.B0.tor.eicat.ca) |
| 08:38.10 | Qwell | off to bed |
| 08:38.35 | mog_home | <PROTECTED> |
| 08:39.07 | Qwell | hmm |
| 08:39.19 | Qwell | is it bad when the "surge" light on my power strip is on constant? |
| 08:39.29 | mog_home | i just ignore things like that |
| 08:39.39 | mog_home | i mean if i am gonna get f-ed |
| 08:39.39 | Qwell | good plan |
| 08:39.41 | mog_home | probablly |
| 08:39.43 | mog_home | but meh |
| 08:39.49 | mog_home | thats why i have backups |
| 08:39.59 | Qwell | heh |
| 08:40.11 | Qwell | okay, off to bed - to not worry about the surge light |
| 08:40.34 | Qwell | I'll fix it in the morning, like I did with the collision light on my switch |
| 08:40.45 | Qwell | electrical tape :D |
| 08:40.59 | wasim | erp ... 144 run deficit |
| 08:41.18 | *** join/#asterisk genmud (n=genmud@ip68-98-82-206.ph.ph.cox.net) |
| 08:42.01 | mmmToop | hey...wasim...haven't seen you for agest...where have you been hiding? |
| 08:42.37 | wasim | mmmToop: out and about |
| 08:45.05 | mmmToop | ...well good to c u again anyway ; ) |
| 08:45.24 | wasim | ; ) |
| 08:46.47 | *** join/#asterisk johnrage (n=jabetong@212.93.201.89) |
| 08:47.15 | johnrage | hello |
| 08:47.35 | johnrage | I am newbie in asterisk |
| 08:47.49 | johnrage | anybody can help me setup from scratch? |
| 08:48.03 | skyen | asterisk is set up from scratch |
| 08:48.12 | skyen | just run it with your example configuration |
| 08:48.43 | johnrage | I am planning to run DID |
| 08:48.47 | *** join/#asterisk nroej (n=joern@134.147.62.143) |
| 08:48.49 | nroej | hi |
| 08:49.14 | wasim | johnrage: wiki |
| 08:49.22 | wasim | ~docs |
| 08:49.24 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 08:49.24 | vira | johnrage, there is lots of into at http://www.voip-info.org/wiki-Asterisk |
| 08:50.06 | johnrage | yes I been through to the follwing URL and read all the info however am a newbie on this thing |
| 08:50.40 | johnrage | we run our own calling card platform using cisco but never been with asterisk |
| 08:51.27 | *** join/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net) |
| 08:54.03 | *** join/#asterisk Kerath (n=Kerath_R@41-209.35-65.tampabay.res.rr.com) |
| 08:54.50 | *** part/#asterisk Kerath (n=Kerath_R@41-209.35-65.tampabay.res.rr.com) |
| 08:56.01 | *** join/#asterisk steff (n=steff@80.125.254.220) |
| 08:56.29 | nick125 | anyone here have the asterisk book? if so, what do you think of it? |
| 08:57.30 | *** join/#asterisk christo (n=chris@brezhnev.spiration.co.uk) |
| 08:57.39 | christo | morning all |
| 08:57.46 | nick125 | morning |
| 08:58.17 | johnrage | anyone here who have a knowledge to setup asterisk server for DID?? Let me know |
| 08:58.52 | wasim | johnrage: most everybody would |
| 08:59.09 | wasim | johnrage: so would you, if you have read the wiki |
| 08:59.19 | nick125 | lol |
| 08:59.20 | nick125 | Paperback: 404 pages |
| 09:01.10 | wasim | good paperweight eh? |
| 09:01.24 | wasim | too light for a door stop |
| 09:01.55 | johnrage | yes guys..it takes a lot of painful steps to do it |
| 09:02.01 | nick125 | "Become a PowerPoint Superhero!" /me pukes |
| 09:02.09 | johnrage | I want somebody who have knowledge to guide us |
| 09:02.15 | johnrage | WE are willing to pay |
| 09:02.42 | johnrage | privte message to me |
| 09:03.37 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.132) |
| 09:04.31 | lehel | nick125: why did u asked about the book? |
| 09:04.48 | *** part/#asterisk newmember (i=user@S010600036d1139fb.cg.shawcable.net) |
| 09:04.51 | nick125 | just wondering if its a good read or not |
| 09:05.01 | MuppetMaster | nick125: Which book? |
| 09:05.08 | lehel | tfot book |
| 09:05.11 | lehel | probably |
| 09:05.31 | nick125 | MuppetMaster: the asterisk book put out from oreilly |
| 09:05.49 | MuppetMaster | nick125: A great read overall. Also, you may download the PDFs and have a look before you buy. |
| 09:05.51 | MuppetMaster | I recommend it. |
| 09:06.09 | lehel | i'm reading it, and .. it's great; i mean good |
| 09:06.56 | lehel | could someone tell me what is the meaning of this?: Nov 14 11:02:13 WARNING[22364]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 6d5d97f45256b1880b8326ae5ad19b5a@172.24.2.2 for seqno 102 (Non-critical Request) |
| 09:08.06 | lehel | the sip virtual is just registered.. no call .. nothing |
| 09:08.52 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 09:16.10 | Delvar | lehel: sounds like a nat issue |
| 09:17.26 | lehel | yap, could be possible Delvar, but i'm not that good yet with NATs ;\ |
| 09:17.51 | Delvar | lehel: in sip.conf add teh lines nat=yes and canreinvite=no to teh sip entity |
| 09:18.17 | Delvar | lehel: will let asterisk work out the nat problems in most cases |
| 09:19.03 | lehel | thanks Delvar |
| 09:20.26 | *** join/#asterisk c0w (n=c0w@staff-ns50-3.as25178.net) |
| 09:20.34 | lehel | Delvar, i put qualify=yes and tells me: NOTICE[22364]: chan_sip.c:10014 sip_poke_noanswer: Peer '291' is now UNREACHABLE! Last qualify: 0 |
| 09:20.56 | lehel | however i have it logined |
| 09:21.20 | Delvar | lehel: i usualy dont use qualify, i set it to no |
| 09:21.35 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 09:21.37 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.132) |
| 09:21.39 | Delvar | lehel: try qualify=8000 to give it 8 seconds to respond |
| 09:26.06 | vira | i'm getting that same error from voicepulse's servers right now :/ |
| 09:31.41 | *** part/#asterisk kumamoto (n=eryco@68-116-142-128.dhcp.ftwo.tx.charter.com) |
| 09:31.45 | _m_ | Does ael support #include? |
| 09:54.49 | *** join/#asterisk hohum (i=corbe@snoop.burghcom.com) |
| 09:54.52 | *** join/#asterisk Danett (n=cyrieldo@tbnb-165-211-26.telkomadsl.co.za) |
| 09:54.54 | Danett | heya. |
| 09:55.30 | Danett | When i call my access number (wich bridges to my mobile phone) the line isn't hangup when the command is issued |
| 09:55.59 | Danett | it gives: -- Hungup 'IAX2/4518@4518-1' |
| 09:56.06 | Danett | But it's still ringing |
| 09:56.19 | hohum | anyone ever play with the Polycom IP phones? |
| 09:58.35 | *** join/#asterisk pr0m_ (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 10:01.45 | jebba | I have a sipura->asterisk set up. If you want extention one, you just hit "1#" to get the extension. How can I make it so you just hit "1" and don't need the "#". In other words, just a dozen possible extensions, and less "keystrokes" ? :) |
| 10:03.01 | jebba | or even better, when the receiver goes Off Hook, it immediately connects. E.g. OffHook->Background(some-song) |
| 10:10.02 | *** join/#asterisk stevie20 (n=stevie@mini.fdknet.de) |
| 10:10.09 | stevie20 | hello |
| 10:10.51 | *** join/#asterisk demetrio (n=demetrio@host199-134.pool872.interbusiness.it) |
| 10:11.09 | demetrio | how do I find out a phone's user agent? |
| 10:11.26 | stevie20 | i have put in my sip.conf in the section generall the following lines: |
| 10:11.48 | stevie20 | disallow=all ; First disallow all codecs |
| 10:11.48 | stevie20 | allow=ulaw ; Allow codecs in order of preference |
| 10:11.48 | stevie20 | ;allow=alaw |
| 10:11.48 | stevie20 | ;allow=gsm |
| 10:11.48 | stevie20 | ;allow=ilbc ; Note: codec order is respected only in [general] |
| 10:12.01 | stevie20 | but why is asterisk using alaw !? |
| 10:12.34 | stevie20 | any ideas? |
| 10:14.00 | stevie20 | demetrio: |
| 10:14.13 | stevie20 | sip show peer <peername> |
| 10:14.27 | stevie20 | and in this list, you have the field useragent |
| 10:14.47 | Danett | I do you "catch" an return code returned by the Dial command? |
| 10:15.30 | Danett | s/I/How |
| 10:15.46 | *** join/#asterisk Jkx (n=soif@80.125.251.112) |
| 10:15.51 | Jkx | Hi all ;) |
| 10:16.34 | Jkx | something, I'm wondering, can we use the Handy Tone 486, as a asterisk gateway ? |
| 10:16.39 | lehel | stevie20, look at the trunks |
| 10:16.45 | Jkx | I mean the fxo |
| 10:16.55 | christo | I'm using the manager API to start a meetme conference, but the call recipients get 'this is not a valid conference number. Please try again'. Why could this be? |
| 10:20.28 | mover | anyone here experienced wit t38 stuff in HEAD? |
| 10:20.53 | mover | i cant compile the stuff after apply the patches |
| 10:20.59 | mover | chan_sip.c:2121: error: `p' undeclared (first use in this function) |
| 10:21.19 | mover | there is something wrong i guess |
| 10:21.34 | stevie20 | lehel, sorry, where should i look? |
| 10:22.59 | joelsolanki | hi all..how much sip g729 codec uses bandwidth ? any rough idea |
| 10:23.01 | joelsolanki | :) |
| 10:23.05 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:23.45 | lehel | stevie20, this alaw connection is between..? |
| 10:23.58 | *** join/#asterisk MuppetMaster (n=MuppetMa@62.37.170.132) |
| 10:24.07 | joelsolanki | I tested with linksys pap2 it is taking RX 20 / TX 20 kbps. but when i use softphone it only uses 20 to 27 kbps |
| 10:24.09 | MuppetMaster | Anyone getting continual crashes with 1.2RC2? *** glibc detected *** double free or corruption (out): 0x081a5b50 *** Aborted |
| 10:24.10 | joelsolanki | any ideas |
| 10:24.17 | mrtwister | joelsolanki, at voip-info.org, i was seen info about all codecs |
| 10:24.41 | mrtwister | joelsolanki, and maybe www.voiponline.com, not sure, there maybe you can find bandwidth calculator |
| 10:24.53 | stevie20 | lehel between Asterisk and a SIP Gateway... the connection to the Zap Channel is in ulaw... |
| 10:25.06 | joelsolanki | mrtwister: yes i have also seen that but dont know my linksys is not working or something wrong. it is consuming around 40 kbps total |
| 10:26.00 | mrtwister | joelsolanki, huh. it is wrong. |
| 10:26.09 | mrtwister | joelsolanki, not more than 25k |
| 10:26.20 | stevie20 | lehel may it be, that the openser, which is our SIP Gateway has got restrictions to the codec? |
| 10:27.02 | *** join/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 10:27.16 | joelsolanki | mrtwister: yes it should not max then 25kbps. but it is consuming more i m getting mad :) i have setup of asterisk+ser and i have 6 g729 licenses for testing. |
| 10:27.57 | joelsolanki | mrtwister: any ideas ? |
| 10:28.25 | mrtwister | joelsolanki, www.translate.ru to www.asterisk-support.ru, push files, download g729 and g723 from there and test |
| 10:28.39 | lehel | stevie20, [not sure], so you don't use trunks? |
| 10:28.50 | mrtwister | joelsolanki, it is compiled ipp libs/codecs |
| 10:29.49 | stevie20 | i'm not sure... i am sure, that i am using 2 trunkgroups with a zaptel device in each group... |
| 10:30.15 | stevie20 | but i dont know, wether i am using trunks for sip calls or not... |
| 10:30.16 | demetrio | stevie20, thanks |
| 10:30.52 | lehel | stevie20, in your trunkgroups there isn't defined any codec? |
| 10:31.36 | *** part/#asterisk MuppetMaster (n=MuppetMa@62.37.170.132) |
| 10:31.53 | stevie20 | lehel the trunkgroups defined by zapata.conf ? |
| 10:32.18 | lehel | oh, not there |
| 10:32.54 | stevie20 | hmm.. where should i define the trunkgroups? |
| 10:33.00 | demetrio | after a while (few hours) from startup, asterisk will unregister from an account and further attempts will timeout. the only way I found to make it register again is disconnecting & reconnecting to the internet (changing IP). |
| 10:34.24 | demetrio | I thought that maybe the provider is banning certain IPs when they see the asterisk user agent, so no I'm trying to make it look like the pone the provider gave me. this doesn't make much sense, however, because if I try to register directly with the phone it works even if asterisk doesn't, so IP ban is not the case. |
| 10:34.27 | stevie20 | in the extensions.conf ? |
| 10:34.59 | lehel | stevie20, trunkgroups ok there.. i thought about smthg else |
| 10:35.04 | demetrio | and even user agent ban isn't, given that with a fresh IP even asterisk will work |
| 10:37.04 | stevie20 | ok lehel... other ideas, where to define the preferred or the must be used codec? |
| 10:38.34 | lehel | stevie20, i'm using sip/iax2.. and for me works the codec-changing in sip or iax.conf |
| 10:40.30 | stevie20 | hmmm.... which version do you use? |
| 10:41.57 | *** join/#asterisk Rawplayer (n=kevin@ipc31055d2.oom-killer.org) |
| 10:42.14 | jebba | I have a sipura->asterisk set up. If you want extention one, you just hit "1#" to get the extension. How can I make it so you just hit "1" and don't need the "#"? |
| 10:42.23 | lehel | stevie20, 1.0.9 |
| 10:42.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:43.13 | lehel | which is ur ver. stevie20 ? |
| 10:43.25 | RoyK | ~seen zoa |
| 10:43.27 | jbot | zoa <n=zoa@pirus.securax.be> was last seen on IRC in channel #asterisk, 12h 5m 59s ago, saying: 'he's just crazy'. |
| 10:43.36 | stevie20 | 1.0.8 |
| 10:44.00 | stevie20 | hhmmm.... should i upgrade? |
| 10:44.25 | lehel | i don' think it could be a ver. issue, but u could try an upgrade, yes |
| 10:49.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:49.21 | christo | I'm using manager API to start a dynamic meetme conference, but the recipients hear 'this is not a valid conference number. Please try again'. |
| 10:49.26 | lme | jebba: I think it should be corrected in your sipura's dialplan, not on the asterisk side... But I don't know anything about sipura ! Sorry |
| 10:50.33 | jebba | lme, ok. thx. /me looking |
| 10:53.42 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 10:54.23 | jebba | lme, yep. that was it. Thanks. I wasn't even thinking about looking at that side. Got off-track. ;) thx |
| 10:54.25 | stevie20 | lehel i gonna try to completly restart the asterisk, maybe the reload is not enough... |
| 10:54.45 | lme | jebba : cool ! u'r welcome |
| 10:56.41 | *** join/#asterisk langals (n=icechat5@196.7.14.183) |
| 10:57.20 | langals | Hi there....I am using an Iax softphone dialling into meetme conferences. Sometimes calls are randomly dropped, and the following error message is given: |
| 10:57.31 | stevie20 | hmm... 2 concurrent calls all the time.... |
| 10:58.50 | langals | WARNING [9490]: cahn_iax2.c:1480 attempt_transmit: Max retries exeeded to host 195.2.4.56 on IAX2/bob@195.2.4.56 (type = 6, subclass =11, ts = 120019, seqno =37) |
| 11:00.13 | lme | langals: it's sounds like a network problem to me |
| 11:00.31 | langals | what do you think it could be? |
| 11:01.06 | langals | because it basically works, but erratically will drop a call - normally the first time you try and connect |
| 11:01.18 | langals | it will throw the error |
| 11:01.20 | lme | langals: r u experiment loss or something like that on your network ? |
| 11:01.44 | lme | langals: you're showing a public ip address, is it thru the Internet ? |
| 11:01.47 | langals | lme: no, not really |
| 11:02.06 | *** join/#asterisk ful|work (n=fulgas@209.8.233.170) |
| 11:02.11 | langals | lme: asterisk is on a public IP, clients are behind a NAT |
| 11:02.11 | ful|work | hey |
| 11:02.31 | langals | lme - normally once you are connected then there isn't a problem |
| 11:03.11 | lme | langals: until you lost your nat, or too many packets |
| 11:03.36 | langals | lme - just sometimes the first time you try and connect drops the call and now sometimes when a user is in the middle of a call |
| 11:03.55 | langals | lme - do you think the NAT may reject packets when too many |
| 11:04.12 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:04.48 | lme | langals: no, (of course depending on the nat device) but it could lost entry, due to a stack overflow or something like that |
| 11:05.21 | lme | langals: maybe you try to tcpdump a session to see if everything is okay |
| 11:05.42 | langals | lme: do I do that on the NAT |
| 11:06.02 | lme | langals: on the asterisk box |
| 11:06.07 | langals | ok |
| 11:06.42 | lme | langals: you should take a look on your nat device... and look at the nat entries when you lost your session |
| 11:07.59 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 11:09.19 | langals | lme - thanks - I will try that |
| 11:14.21 | christo | Should it be possible to build a conference using meetme on a * box, with the calls routed over IAX to a media gateway? ie the conferences are built and controlled on a different machine to the mgw. I'm having problems with this setup |
| 11:16.07 | christo | don't worry |
| 11:16.15 | christo | I fixed it - I had a typo in my dial plan :S |
| 11:23.02 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 11:24.02 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 11:29.27 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 11:39.04 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:39.28 | puzzled | hi |
| 11:39.50 | *** join/#asterisk InfraRed (n=bigboss@master.subhi.com) |
| 11:40.03 | lehel | hello, puzzled |
| 11:40.04 | InfraRed | hi all |
| 11:41.07 | RoyK | hi |
| 11:41.41 | InfraRed | i have this setup atm: ATA GW -> Asterisk (local also NAT gw) -> remote asterisk -> termination server. In sip debug i keep getting from/to unknown@ip between the 2 asterisk servers, any ideas. what sip.conf option should i set |
| 11:42.58 | wasim | use iax between * servers and save yourself some headaches |
| 11:43.26 | wasim | or atleast change the position of the pain :( |
| 11:45.23 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 11:46.09 | InfraRed | hmmm |
| 11:46.24 | InfraRed | the reason of using the 2nd * server is for billingh |
| 11:46.53 | InfraRed | but i suppose the billing should be ok since it'll be sip -> * -> IAX -> * -> SIP |
| 11:49.29 | *** join/#asterisk folsson (n=filip@h147n1fls32o985.telia.com) |
| 11:51.31 | *** join/#asterisk jeffik (n=Jeff@CPE0011505c92d3-CM014350000760.cpe.net.cable.rogers.com) |
| 11:54.42 | *** join/#asterisk Speeder (n=psilva@pal-213-228-158-41.netvisao.pt) |
| 11:56.50 | wasim | morning astmaster |
| 11:58.06 | RoyK | góðanhelvÃtisdaginn |
| 11:58.19 | RoyK | kram: you up this early? |
| 11:58.21 | syle | wtf is assmaster |
| 11:58.45 | wasim | syle: gurus on #asstricks |
| 11:59.05 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:59.11 | RoyK | syle: other word for gay? |
| 11:59.49 | *** part/#asterisk Jkx (n=soif@80.125.251.112) |
| 11:59.55 | syle | fag |
| 12:00.35 | syle | sexually challenged, sexually steriotyped ? |
| 12:00.45 | drumkilla | ... |
| 12:01.01 | drumkilla | please move on. |
| 12:01.23 | syle | drumkilla you;ve said nothing and thats the first thing you say? |
| 12:01.43 | kram | i'm on my way to vegas this a.m. |
| 12:01.51 | drumkilla | i looked over for a second and saw that, so yeah. that's all I had to say. |
| 12:01.58 | mut | and ya didn't invite me? |
| 12:02.13 | wasim | oooh, put $10 on red-14 on the roulette table for us |
| 12:02.39 | wasim | and remember dealer has to hit on 16 |
| 12:02.42 | mut | i got $20 on #17 |
| 12:04.41 | mut | what ast_log level is verbose 1? |
| 12:04.58 | mut | NOTICE and WARNING are verbose off and still show.. |
| 12:06.11 | syle | how are they verbose off then? |
| 12:06.25 | mut | they show if verbose is off |
| 12:06.29 | mut | .. |
| 12:06.33 | mut | don't ask me why |
| 12:06.36 | mut | cause i dunno |
| 12:06.47 | syle | what shows it off? |
| 12:06.56 | mut | um |
| 12:07.12 | mut | Verbosity is now OFF |
| 12:07.14 | mut | that |
| 12:07.15 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 12:08.31 | mut | must be anything that is sent to ast_log goes to cli... |
| 12:09.16 | syle | i would imagine anything under "console" in logger.conf myself |
| 12:09.37 | *** join/#asterisk zotz (n=zotz@24.231.47.168) |
| 12:10.00 | mut | ya |
| 12:10.58 | _m_ | Is there any support for #include in AEL? |
| 12:11.06 | RoyK | _m_: iirc yes |
| 12:11.08 | syle | i thinnk verbosity is more a debug level thing |
| 12:11.37 | RoyK | syle: not really. very, very, very useful to have logs when things go wrong |
| 12:11.55 | mut | well |
| 12:11.56 | syle | not really what? |
| 12:12.09 | _m_ | RoyK: the syntax seems to differ from extensions.conf's #include. I'm getting pbx_ael.c:1115 handle_root_token: Unknown root token '#include' |
| 12:12.14 | mut | i'm just string to get rid of my sip registration failure cli output |
| 12:12.24 | mut | w/o getting rid of all warnings |
| 12:12.24 | RoyK | mut: rtfs :) |
| 12:12.27 | mut | or notices |
| 12:12.36 | mut | RoyK? |
| 12:12.39 | *** join/#asterisk coppice (n=chatzill@7.206.17.210.dyn.pacific.net.hk) |
| 12:12.59 | RoyK | mut: use the source, luke |
| 12:13.04 | mut | i am.. |
| 12:13.15 | mut | just figuring out ast_log |
| 12:14.48 | mut | like LOG_DEBUG |
| 12:15.13 | mut | i'll just use that |
| 12:15.20 | mut | since i'm not outputtin it to console |
| 12:15.33 | mut | dirty |
| 12:17.17 | syle | what are you doing ? |
| 12:17.38 | mut | just making it so my sip registration failures aren't output to console |
| 12:17.59 | syle | owww |
| 12:18.21 | syle | then just grep the source and change from whatever log level to log_debug |
| 12:18.28 | mut | thats what i did |
| 12:18.46 | mut | working just spiffy |
| 12:19.24 | mut | man the winds here were nuts the past few days |
| 12:19.30 | mut | we've got towers without power all over |
| 12:19.40 | mut | customers with their poles bent right over |
| 12:19.46 | syle | hehe |
| 12:20.00 | syle | yeah we had once of those this summer to |
| 12:20.05 | coppice | i'd hate to have my pole bent right over |
| 12:20.15 | syle | as long as its not your pole! |
| 12:20.16 | syle | haha |
| 12:20.17 | mut | i can't believed mine wasn't |
| 12:20.22 | mut | guyed wires held it decent |
| 12:20.28 | pooh_ | g'day everyone |
| 12:20.30 | mut | the top 10ft i was amazed didn't fall over tho |
| 12:20.44 | mut | have that heavy router on top of the pole |
| 12:20.51 | mut | swaying like mad in the wind |
| 12:21.19 | syle | notice when there is a big storm, there is ALWAYS a car accident somewhere |
| 12:21.28 | syle | blows me away |
| 12:21.59 | demetrio | cars are obsolete |
| 12:22.19 | syle | you got a transporter i can use? |
| 12:22.20 | syle | hehe |
| 12:22.28 | demetrio | you got it, too |
| 12:22.34 | mut | heh yesterday i was watching the powerlines do jumprope with themselves |
| 12:22.35 | demetrio | they're called feet |
| 12:22.47 | mut | the bottom wire was wrapping itelf around the one above it |
| 12:22.53 | syle | yeah well that isn;t going to help you get 100 miles out in the country |
| 12:22.54 | mut | then unwrapping and doin it the other way |
| 12:23.03 | demetrio | syle: impatience is bad |
| 12:23.04 | demetrio | :D |
| 12:23.12 | joelsolanki | hello all: my linksys pap2 is consuming very much bandwidth 40 kbps ..does any body know anything to setup in that ? payload ? |
| 12:23.28 | syle | yeah well some would call that life is to short, to each his own i guess |
| 12:24.02 | demetrio | I assure you, after you experiencing a couple times your car swinging on ice on a steep mountain road those 5 miles really don't seem too much to walk anymore |
| 12:24.07 | clive- | joel sounds about right to me |
| 12:24.27 | syle | hehe |
| 12:24.33 | syle | you must be from the states |
| 12:24.36 | syle | come to canada and say that |
| 12:24.37 | fourcheeze | can anyone point me to a url for a sample asterisk/ser config? |
| 12:24.39 | demetrio | nope |
| 12:24.44 | joelsolanki | clive: means it is ok ?/ |
| 12:24.53 | mut | my headphones should be here this week |
| 12:24.54 | demetrio | I'm from Italy, the mountains are the dolomites |
| 12:24.55 | mut | YAYZ |
| 12:24.55 | syle | well whatever works in your country i guess |
| 12:24.58 | fourcheeze | I just want ser to proxy for asterisk |
| 12:25.05 | syle | italy yeah i;d probably walk |
| 12:25.13 | mut | they had better be teh awesome like everyone says |
| 12:25.24 | clive- | joel yup...on g729 you will end with like 35 odd kbps so its about right |
| 12:25.25 | syle | make sure i;m drunk first though |
| 12:25.28 | syle | :) |
| 12:26.13 | syle | actually being from italy, do you drink alot of wine? |
| 12:26.36 | syle | wondering what the percentage of beer vs wine drank is there |
| 12:26.49 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 12:27.04 | demetrio | well, every transport method needs some combustible |
| 12:27.26 | syle | not so, there are some electric transport means |
| 12:27.33 | fourcheeze | demetrio: not all |
| 12:27.35 | joelsolanki | clive: but only 1 single call take this much bandwidth ? |
| 12:27.57 | fourcheeze | syle: depends how you got the electricity in the first place |
| 12:28.18 | fourcheeze | but I find traveling from the top to bottom of a hill to not need combustibles |
| 12:28.29 | syle | well for the cars its quite inefficent right now, 6 car batteries linked together |
| 12:28.32 | clive- | joel yup... |
| 12:28.37 | syle | doesn;t get you very far |
| 12:28.40 | demetrio | anyway, I'm from an area in Italy that's famous for its alcoholics |
| 12:28.50 | fourcheeze | so we should urge town planners to make sure all roads go downhill - it's straightforward |
| 12:29.10 | syle | demetrio, which area is that? i got to visit sometime :) |
| 12:29.14 | demetrio | it's not about beer or wine, here you drink mainly grappa (I'm not sure if there's a translation for it) |
| 12:29.20 | demetrio | veneto |
| 12:29.28 | demetrio | north of venice :) |
| 12:29.29 | syle | kewl |
| 12:29.35 | joelsolanki | clive: dont u think that only 1 single call takes 35 kbps its much ? |
| 12:29.54 | syle | i hear housing is overpriced in italy though |
| 12:29.54 | joelsolanki | clive: is that only with linksys pap2 ? or with all boxes ? |
| 12:30.03 | syle | like a million dollars for a house |
| 12:30.33 | demetrio | syle: I don't know, would you call overpriced a bed in a double room for $500/month if you're lucky? :) |
| 12:30.49 | syle | lol |
| 12:31.03 | syle | get your own apt for that here :) |
| 12:31.16 | joelsolanki | clive: ??? |
| 12:31.28 | fourcheeze | demetrio: you get your own bed for that?? |
| 12:31.35 | fourcheeze | you don't even have to share? |
| 12:31.41 | demetrio | nope, I'm lucky enough to have relatives that can host me |
| 12:32.11 | syle | no wonder italy families are so tight, they can;t afford to get out on their own because of those real estate prices |
| 12:32.17 | clive- | joel which codec are you using...g711 will use over 90kbps |
| 12:32.35 | demetrio | well, to tell the truth, not everywhere it's like that |
| 12:32.46 | fourcheeze | syle: I don't think italy is that much different to the rest of Europe in the main |
| 12:32.47 | demetrio | I'm talking about a loan that a student has to afford in milan |
| 12:32.50 | joelsolanki | clive: i m using linksys pap2 with g729 codec. |
| 12:32.59 | clive- | joel sounds right |
| 12:33.00 | shido6 | i love my pap |
| 12:33.13 | syle | hate mine |
| 12:33.24 | demetrio | but still, if you want an apt you must get ready to pay at least €300k in a minor city |
| 12:33.25 | syle | can i ask you what your settings are shido? |
| 12:33.55 | joelsolanki | clive: but when i m using both lines of linksys pap2 it is using 100 kbps :( |
| 12:34.15 | syle | seems every hour on the dot, my pap resets or something |
| 12:34.24 | joelsolanki | clive: linksys pap2 doesnt have 2 g729 support. it can use g729 one at a time. |
| 12:34.27 | *** join/#asterisk Laerte (n=io@82.112.222.200) |
| 12:34.29 | Laerte | hy all |
| 12:34.30 | syle | not hardware resets, asterisk |
| 12:34.44 | joelsolanki | clive: so 100 kbps is ok for both lines on linksys pap2 ? |
| 12:35.00 | syle | no it can;t only use 1 uncompressed codec at a time |
| 12:35.12 | clive- | yup,,,but it cant do 2x g729 |
| 12:35.15 | syle | compressed |
| 12:35.25 | joelsolanki | yes |
| 12:35.40 | syle | yeah sucks what can you do |
| 12:35.44 | syle | buy sipura hehe |
| 12:35.54 | syle | or just g729 and ulaw |
| 12:35.56 | clive- | sipura 2000 also cant do 2x g729 |
| 12:36.13 | joelsolanki | i heard sipura 2100 has 2* g729 |
| 12:36.18 | syle | yeah me to |
| 12:36.23 | clive- | yup, you need the 2100 model |
| 12:36.41 | Delvar | pap is a sipura?? |
| 12:36.44 | syle | well thats what everyone gets hehe |
| 12:36.47 | syle | no linksys |
| 12:36.58 | joelsolanki | so sipura 2100 using both both channels. how much bandwidth it will consume ? |
| 12:37.08 | Delvar | yes but it is a sipura jsut slightly differrent front end |
| 12:37.44 | syle | well assuming sip and both lines calling at same time.... |
| 12:38.00 | *** join/#asterisk br00ksh1r3 (n=matt@206.166.206.34) |
| 12:38.01 | stevie20 | demetrio italy, nice... ive made holidays with my parents in the dolomites... near of Bozen (Bolzano) |
| 12:38.03 | syle | 94.5 Kbps |
| 12:38.31 | joelsolanki | ahhh 94 kbps is very much i guess :) ? |
| 12:39.03 | joelsolanki | i tested the softphone with g729 it consumes only 20 to 25 kbps per call. |
| 12:39.07 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 12:39.07 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 12:39.10 | twisted[mobile] | hsv airport #2 |
| 12:39.25 | br00ksh1r3 | yeah.. i agree |
| 12:39.55 | twisted[mobile] | we're so rox0ring |
| 12:40.04 | br00ksh1r3 | yeaaaaaaaaaaaaaaah |
| 12:40.07 | twisted[mobile] | and stuff |
| 12:40.08 | twisted[mobile] | damn i'm tired. |
| 12:40.09 | Laerte | anyone had problem with group and attendend transfer ? |
| 12:40.15 | demetrio | stevie20, I live about 80 miles from bozen, and usually when I tell strangers that I don't even know how to ski they stare at me dubious |
| 12:41.09 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 12:41.59 | *** join/#asterisk Tili (n=Tili@213.132.60.182) |
| 12:42.12 | stevie20 | demetrio i was there only in the summer ;-) |
| 12:42.39 | *** join/#asterisk aaronz (n=aaronz@pdpc/supporter/student/aaronz) |
| 12:43.24 | stevie20 | this is an nice area there... |
| 12:43.58 | Druken | hey ppl, exactly how hard would it be to have asterisk monitor the rtp and destroy a call when the data reaches 0 ? |
| 12:44.31 | syle | well if you don't ski in italy you probably don't travel alot |
| 12:44.33 | Druken | i'm finding i'm getting lost calls all the time, at least 1 a day |
| 12:44.47 | demetrio | I bet you were in merano |
| 12:44.48 | syle | i take a plane to go ski'ing wherever |
| 12:44.58 | twisted[mobile] | time goes by |
| 12:45.00 | twisted[mobile] | so slowly |
| 12:45.10 | br00ksh1r3 | z0mg |
| 12:45.15 | mut | no it doesnt |
| 12:45.18 | twisted[mobile] | yes it does |
| 12:45.22 | twisted[mobile] | we've been up all night |
| 12:45.22 | mut | i don't even know where last week went |
| 12:45.24 | br00ksh1r3 | mattf wants twisted's powerbook |
| 12:45.25 | stevie20 | demetrio in merano for day trips.... |
| 12:45.25 | tzanger | that's relativity |
| 12:45.26 | twisted[mobile] | tiredness == slow time |
| 12:45.27 | syle | last time i tried snowboarding, man did i ever land on my ass a million times |
| 12:45.39 | mut | before i knew it i was sleeping on my desk on friday |
| 12:45.46 | tzanger | put your hand on a hot stove element for a minute and it feels like an hour... sit with a pretty girl for an hour and it feels like a minute |
| 12:45.55 | demetrio | and I swear at ski related tourism that ruin my mountains instead |
| 12:46.14 | twisted[mobile] | aiyeeee |
| 12:46.26 | tzanger | time to get my ass to work |
| 12:46.48 | twisted[mobile] | i agree |
| 12:46.52 | twisted[mobile] | oh wait |
| 12:46.56 | twisted[mobile] | i'm flying soon |
| 12:49.14 | stevie20 | we had apartments round about "Voels am Schlern", Fie allo Sciliar, |
| 12:49.29 | mut | fookin as |
| 12:49.29 | mut | p |
| 12:49.30 | *** join/#asterisk SERGEUS|WORK (n=SERGEUS@ippe-245.ippe.ru) |
| 12:49.32 | mut | argh@ |
| 12:49.42 | mut | Operation not Allowed |
| 12:49.42 | mut | /Default.asp, line 0 |
| 12:49.47 | demetrio | don't know that |
| 12:49.50 | mut | the hells that supposed to mean! |
| 12:49.51 | SERGEUS|WORK | hi! can anybody help me with SIP headers? |
| 12:50.05 | stevie20 | or "Rosengarten" demetrio .. sorry, i cant remeber the italy names ;-) |
| 12:50.10 | lme | demetrio: where r u from ? |
| 12:50.12 | SERGEUS|WORK | i wonder which SIP header is used for MWI |
| 12:50.17 | demetrio | well, those are actually german names :) |
| 12:50.25 | demetrio | lme: belluno, veneto, italy |
| 12:51.03 | stevie20 | yes demetrio, you are right ;-) |
| 12:51.09 | Laerte | anyone had problem with group and attended transfer ? |
| 12:51.18 | SERGEUS|WORK | is it possible to set MWI signal manualy? |
| 12:51.30 | Delvar | use a stick and a bit of gum |
| 12:51.38 | demetrio | Laerte, I recall there's a bug with attended transfer & groups in 1.0.x |
| 12:51.46 | lme | ouch at the opposite from me... |
| 12:52.03 | demetrio | lme, you're from... ? |
| 12:52.04 | Delvar | SERGEUS|WORK: yes using sipsac its easy |
| 12:52.12 | Laerte | yes demetri, but i have tried *-HEAD and i have the same problem |
| 12:52.21 | lme | demetrio: Bourg-St-Maurice/ Les Arcs |
| 12:52.34 | lme | lme: France |
| 12:52.57 | lme | damned |
| 12:52.59 | br00ksh1r3 | lates |
| 12:53.03 | lme | demetrio, not lme |
| 12:53.04 | Laerte | cmq grazie demetrio |
| 12:53.21 | stevie20 | oh, france... one political question.. what do you think about the riots in france? |
| 12:53.30 | lme | oh my god |
| 12:53.33 | lme | please |
| 12:53.33 | demetrio | haha |
| 12:53.44 | demetrio | good luck lme :) |
| 12:53.52 | mut | http://www.bash.org/?575675 |
| 12:54.06 | lme | don't speak about riots, i will not about political situation in de :) |
| 12:54.14 | *** join/#asterisk Danett (n=cyrieldo@tbnb-165-211-26.telkomadsl.co.za) |
| 12:54.17 | Danett | heya. |
| 12:54.51 | Danett | When i make a call trough my sip provider, there is no sound. The asteriskbox from where i originate the call is behind NAT, how can i solve this problem? |
| 12:54.51 | stevie20 | lme ok ;-) |
| 12:55.04 | SERGEUS|WORK | Delvar, what is sipsac? |
| 12:55.17 | demetrio | Danett, port forwarding |
| 12:55.23 | Delvar | ~google sipsac |
| 12:55.35 | Delvar | its a tool to send SIP messages |
| 12:55.41 | Delvar | used mainly for debug etc |
| 12:55.53 | Delvar | but can be used to send MWI manualy |
| 12:55.56 | Delvar | very usfull |
| 12:55.59 | Delvar | gogoel fo rit |
| 12:56.05 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:56.28 | SERGEUS|WORK | checked voip-info with google :) and there were nothing - will check other internet |
| 12:56.40 | Danett | demetrio: i forwarded 5060 |
| 12:56.50 | SERGEUS|WORK | is there any standard features in * which can be used for it? |
| 12:57.39 | stevie20 | danett maybe you should forward the RTP ports too... |
| 12:57.45 | *** join/#asterisk feenikz (n=dave@tbnb-165-216-08.telkomadsl.co.za) |
| 12:57.59 | stevie20 | just to get right, signalling works, or not Danett ? |
| 12:58.01 | demetrio | Danett, forward 10000 to 20000 too |
| 12:58.03 | Laerte | demetrio, you hare italian ? |
| 12:58.10 | demetrio | yes I am |
| 12:58.32 | Ahrimanes | anyone here using a zyxel prestige 2002 ata? it's registering and calling fine, but sound recieved is really bad, sound sent is fine.. |
| 12:58.41 | Laerte | i like to find some italian to work with asterisk ( i'm also italian demetrio) |
| 12:58.48 | RoyK | demetrio: you'll love asterisk then. the code is very italian-inspired |
| 12:58.56 | RoyK | demetrio: in terms of spaghetti |
| 12:58.56 | stevie20 | Ahrimanes, this is like our problems... |
| 12:59.07 | Ahrimanes | stevie20: with what equiptment ? |
| 12:59.32 | stevie20 | i am using an digium TE205P for connecting to the PBX and on the other side a SIP gateway... |
| 13:00.00 | stevie20 | outgoing sound is finde, incoming has got silence supression... |
| 13:00.03 | demetrio | RoyK, as long as it works I'm not complaining :) |
| 13:00.04 | *** join/#asterisk RedDane (n=bharatsa@210.211.246.47) |
| 13:00.10 | RedDane | hello |
| 13:00.12 | RedDane | there |
| 13:00.23 | stevie20 | and this is very, very uncomfortable... |
| 13:00.26 | Ahrimanes | stevie20: hmm.. i disabled such features on this one |
| 13:00.28 | lme | stevie20 : on sip or on TE205 side ? |
| 13:00.34 | RedDane | I am configuring the Asterisk queues using realtime |
| 13:00.44 | stevie20 | lme sound coming from the SIP side.... |
| 13:00.50 | lme | !! |
| 13:00.55 | lme | same issue here |
| 13:01.11 | RedDane | after making changes inthe extconfig.conf are there any changes to be made inthe queues.conf file?? |
| 13:01.24 | stevie20 | i'm going to test, if this is an issue with decoding alaw und encoding to ulaw... |
| 13:01.24 | Ahrimanes | stevie20: * -> zyxel 2002 -> analogue phone here.. |
| 13:01.33 | RedDane | hello |
| 13:01.37 | RedDane | plesae answer me |
| 13:01.54 | stevie20 | but i can test ist finally round about 15:00 GMT |
| 13:02.07 | stevie20 | .. test it... |
| 13:02.38 | *** part/#asterisk feenikz (n=dave@tbnb-165-216-08.telkomadsl.co.za) |
| 13:02.44 | RedDane | hello |
| 13:02.47 | Ahrimanes | hm this friggin ata has german language config interface, but guess i should try changing stuff |
| 13:03.13 | RedDane | does anybody know the answer to my question on the queues? |
| 13:03.13 | Druken | RedDane: either have patients, or you'll never get an answer :) i don't see a cheque in my hands yet... |
| 13:03.22 | Druken | i only take shit from people who pay me |
| 13:03.23 | stevie20 | for this test i gonna remove all other codes from the module directory... just to get sure, the is only the ulaw codec... |
| 13:03.30 | fugitivo | hello |
| 13:04.16 | fugitivo | i´m trying to register to a remote sip gateway, and i get this message -- Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx |
| 13:06.33 | lme | RedDane: not me sorry ! |
| 13:06.53 | fugitivo | any idea about that interval too brief? |
| 13:07.34 | fugitivo | is a problem from * side or the gateway side? |
| 13:08.02 | lme | fugitivo: in the sip header you should see who speak to who |
| 13:08.19 | lme | fugitivo: but at my sense, this is the other side which reject you |
| 13:08.39 | lme | fugitivo: too many bad tries or interval between re authentication too short |
| 13:08.58 | stevie20 | lme has you tried anything, to get the silencesuprresion out of the sound? |
| 13:09.23 | fugitivo | yes, i know that the other side is rejecting me, but i don't get authentification rejects |
| 13:09.50 | *** join/#asterisk bweschke (i=bweschke@243.sub-70-209-36.myvzw.com) |
| 13:10.44 | lme | stevie20: except praying no, I don't get the hand on the other side.. I'm rejecting all but alaw on my sip gateway. Inside i'm with cisco's phone and chan_sccp |
| 13:11.17 | *** join/#asterisk cica (i=Lamercz@81.30.249.241) |
| 13:11.32 | lme | one day, i will get some english grammatical lessons |
| 13:14.53 | *** join/#asterisk Gourou_fou (n=x@ACaen-151-1-27-155.w86-195.abo.wanadoo.fr) |
| 13:15.01 | Gourou_fou | beuaaaah vive la révolution |
| 13:15.01 | stevie20 | hmm... ok lme.. when my test works, i gonna update you... |
| 13:16.35 | *** join/#asterisk yxa (i=empty@cm121.gamma228.maxonline.com.sg) |
| 13:16.41 | *** part/#asterisk SERGEUS|WORK (n=SERGEUS@ippe-245.ippe.ru) |
| 13:16.54 | lme | stevie20: ok... 'cause you mentioned it, I'm seriously thinking about bring my * box in out streets to study about heat effects on silent suppression.... |
| 13:17.31 | lme | I hate silent suppression, Silent suppression is evil ! |
| 13:18.04 | stevie20 | *lol* |
| 13:18.08 | coppice | VAD is OK, though |
| 13:18.19 | demetrio | I think it's called silence suppression, "silent suppression" sounds like a good name for an action b-movie to me :) |
| 13:18.29 | lme | yes |
| 13:18.32 | lme | but i'm french |
| 13:18.38 | Ahrimanes | oh my.. |
| 13:19.07 | lme | and we do speak english as bad as spanish's cows do for french... |
| 13:19.25 | Ahrimanes | lol |
| 13:19.40 | cica | hi! can anybody help me setup Sangoma card A101 and R2/MFC signalig? |
| 13:20.11 | coppice | cica: do you have the sangoma card set up yet? |
| 13:20.31 | cica | yes but with E1 ISDN |
| 13:20.44 | stevie20 | coppice we dont like silence here... we need comfort noice generating, if someone suppress silence... |
| 13:21.49 | coppice | I think the sangoma config file needs changing to put the card in CAS mode. then the zaptel.conf file needs similar changes. I'm not too familiar with the sangoma config, though. I know people are using sangoma with my R2 software |
| 13:21.55 | lme | it's an health problematic... If my boss doesn't here noise while he's phoning, he smash me... |
| 13:22.13 | coppice | stevie20: silence suppression is always horrible. you need proper VAD |
| 13:22.16 | lme | hear |
| 13:22.51 | stevie20 | coppice, whats the difference? where to configure VAD? ;-) |
| 13:23.13 | coppice | detecting voice != simply detecting something |
| 13:23.49 | stevie20 | ok, but for which purpose do you want to detect voice? |
| 13:23.52 | demetrio | is it possible to use one of those PCMCIA umts modem cards as call gateways? |
| 13:24.10 | lme | maybe with oss ? |
| 13:24.15 | coppice | to insert comfort noise when there is no voice |
| 13:24.52 | stevie20 | ok... is asterisk capable of comfort noise generating? |
| 13:25.00 | coppice | no |
| 13:25.13 | coppice | it doesn't do VAD either. |
| 13:26.11 | stevie20 | damn... so i got the same problem as before... i just saw a light on the horizon, but that was firework... i need to get the silence suppression out here... *grrr* |
| 13:26.27 | cica | coppice: I saw some cofiguration on the internet but it din't work :-( |
| 13:26.29 | Ahrimanes | submit patch :) |
| 13:26.50 | lme | stevie20: have you finished your test ? |
| 13:27.51 | coppice | cica: config for what? |
| 13:28.02 | cica | for sangoma card |
| 13:28.17 | stevie20 | no, i cant start these at the moment... we have about 5 concurrent calls now and i am not allowed to disconnect these calls... :-( |
| 13:28.40 | clive- | steve just give them a "soft hangup" |
| 13:28.43 | lme | stevie20: i've got no silence suppression problem with calls over my zap cards (fxo wildcard, te110P, quadbri junghaans) |
| 13:29.05 | coppice | the config file generator for the sangomas is hopeless. I had to do things by hand. so far I only set up a sangoma for PRI use, though. |
| 13:29.20 | stevie20 | lme dito.. i dont have silence compression issues, when i just forward the calls to the PMX... |
| 13:29.43 | stevie20 | ....silence suppression...... |
| 13:29.53 | lme | soft hangup sounds like "squeeze me gently"... Sounds good, but occurs bad |
| 13:30.20 | stevie20 | i think, this could be an issue of decoding alaw and encoding this to ulaw... |
| 13:31.21 | lme | stevie20: if so, I wonder why my disallow all have no effect... |
| 13:31.44 | stevie20 | lme i have got the same problem with disallow all... |
| 13:31.53 | lme | gee ! |
| 13:31.56 | stevie20 | i said, disallow all und only allow ulaw |
| 13:32.01 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:32.05 | stevie20 | but this doesnt matter |
| 13:32.06 | cica | coppice: thank's for info. I try contact sangoma for help |
| 13:32.14 | stevie20 | every call to sip is initiated with alaw |
| 13:32.55 | coppice | cica: once you can get the sangoma config sorted out you should be able to follow my instructions for the rest of the installation. it should be exactly the same as using a Digium card from there on |
| 13:34.22 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 13:34.50 | cica | coppice: I allready setup once you software with digum card just for test it if it's work :-) |
| 13:35.38 | coppice | cica I did some bug fixing this week, so make sure you get the latest libmfcr2 |
| 13:36.42 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:37.16 | cica | coppice: I will, thanks |
| 13:41.33 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 13:42.27 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 13:46.45 | stevie20 | hhmm.... |
| 13:47.39 | stevie20 | i moved /usr/lib/asterisk/modules/codec_* to another directory and put only codec_ulaw in the modules directory... |
| 13:48.01 | stevie20 | why can asterisk talk in alaw !? |
| 13:48.27 | stevie20 | the codec is not there... from where does asterisk get its codecs? |
| 13:48.36 | skyen | i believe that codecs are only used when converting between clients |
| 13:48.50 | skyen | if you've got two sipclients talking alaw, the asterisk won't have to modify the rtp-stream |
| 13:48.56 | skyen | hince, codecs aren't used |
| 13:49.14 | stevie20 | skyen asterisk must code this way, because the one side is an E1 interface and the other a sip client |
| 13:49.23 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
| 13:49.26 | skyen | ah, nm then ;) |
| 13:49.38 | stevie20 | ;-) |
| 13:49.49 | stevie20 | but where does he ever know about alaw !? |
| 13:51.52 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-242-4.buckeyecom.net) |
| 13:54.07 | lme | stevie20: you only got ulaw codec ??!! |
| 13:54.34 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 13:55.24 | stevie20 | lme in the modules directory is only the codec_ulaw.so |
| 13:55.36 | stevie20 | but, asterisk is using anything else than ulaw... |
| 13:55.38 | lme | okay |
| 13:55.40 | lme | so |
| 13:55.43 | stevie20 | and i dont knwo why |
| 13:56.02 | lme | I gonna try to work as a farmer.... Safest |
| 13:56.06 | Gourou_fou | how can i can create .gsm sounds ? |
| 13:56.20 | stevie20 | yes lme, this seems so ;-) |
| 13:56.25 | gambolputty | record them from a phone into * |
| 13:56.33 | Gourou_fou | !!! |
| 13:56.50 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 13:56.58 | *** join/#asterisk deezed (i=r00t@Stay.away.He.is.krazy.us) |
| 13:57.08 | Gourou_fou | and if i want convert a wav, mp3, ogg ... sound |
| 13:57.18 | Gourou_fou | other sound than voice :) |
| 13:57.48 | gambolputty | sox |
| 13:58.08 | lme | Gourou_fou: sox |
| 13:58.12 | Gourou_fou | mmh |
| 13:58.18 | lme | damn |
| 13:58.24 | stevie20 | Nov 14 14:57:38 DEBUG[3303]: Set channel Zap/62-1 to read format alaw |
| 13:58.24 | stevie20 | Nov 14 14:57:38 DEBUG[3303]: Set channel SIP/194.97.4.21-3d71 to write format alaw |
| 13:58.24 | stevie20 | Nov 14 14:57:38 DEBUG[3303]: Set channel Zap/62-1 to write format alaw |
| 13:58.24 | stevie20 | Nov 14 14:57:38 DEBUG[3303]: Set channel SIP/194.97.4.21-3d71 to read format alaw |
| 13:58.34 | Gourou_fou | http://www.voip-info.org/wiki/view/sox |
| 13:58.36 | stevie20 | grml |
| 13:58.36 | Gourou_fou | ok :) |
| 13:58.40 | Gourou_fou | thanks !! |
| 13:58.46 | lme | ah ah ah !!!! |
| 13:59.03 | lme | * the undead |
| 13:59.20 | lme | try to rm -rf / just to see |
| 14:00.36 | Meaty | .. |
| 14:00.46 | Gourou_fou | :p |
| 14:00.51 | stevie20 | *lol* ;-) |
| 14:01.14 | Gourou_fou | try /Quit |
| 14:01.25 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 14:01.49 | yxa | does anyone has an example on how to use the Authenticate command? |
| 14:03.26 | deezed | Authenticate([insertpassword]) |
| 14:03.29 | stevie20 | ok... another one.. has anybody the option disallow=all in sip.conf working ? |
| 14:03.54 | deezed | like exten => s,1,Authenticate(1111) |
| 14:03.57 | stevie20 | version 1.0.9 |
| 14:04.04 | stevie20 | ? |
| 14:04.13 | Meaty | yes stevie20 |
| 14:04.14 | yxa | deezed i want the passwd to read from a file... |
| 14:04.49 | stevie20 | Meaty, really? did you test with a sip client, if you can use another codec the the ones, specified to be allowed? |
| 14:05.01 | yxa | deezed that matches the number to be called. is that possible? |
| 14:05.07 | stevie20 | meaty i placed |
| 14:05.11 | stevie20 | disallow=all |
| 14:05.12 | deezed | like exten => s,1,Authenticate(/passwordfile) |
| 14:05.16 | stevie20 | allow=ulaw |
| 14:05.21 | Meaty | ok |
| 14:05.23 | stevie20 | in sip.conf [general] |
| 14:05.28 | Meaty | and ? |
| 14:05.30 | stevie20 | but sip is using alaw... |
| 14:05.35 | deezed | yxa what I did was had mysql pull it from a table |
| 14:05.35 | Meaty | :o |
| 14:05.48 | stevie20 | i dont have alaw set to be allowed |
| 14:06.30 | Meaty | ok stevie20 |
| 14:06.32 | deezed | exten => s,2,MYSQL(Query resultid ${connid} SELECT\ `password`\ FROM\ `voicemail_users`\ WHERE\ `customer_id`\ =${EXTEN}) |
| 14:06.32 | deezed | exten => s,3,MYSQL(Fetch fetchid ${resultid} password) |
| 14:06.32 | deezed | exten => s,4,Authenticate(${password}) |
| 14:06.33 | Meaty | hmm |
| 14:07.25 | stevie20 | can you crosscheck this, Meaty ? |
| 14:07.46 | yxa | deezed hmm.. so when the auth succeeds, what happens? |
| 14:08.00 | deezed | goes to s,5 |
| 14:08.08 | deezed | or whatever the next priority is |
| 14:08.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:08.12 | Meaty | stevie20 I see for it, one moment. |
| 14:08.19 | stevie20 | thx meaty |
| 14:08.37 | yxa | deezed i see..i'm gonna try and digest that |
| 14:08.43 | Delvar | deezed: whay not do the lot in an AGI script? never liked the idea od SQL in the dialplan... |
| 14:08.49 | yxa | deezed does * support PGSQL natively too? |
| 14:09.08 | deezed | cuz i pheer agi... and this is easy |
| 14:09.21 | Delvar | oh :) |
| 14:09.32 | Katty | morning |
| 14:09.36 | docelmo | sup sup.. |
| 14:09.38 | *** join/#asterisk jmjones (n=jmjones@adsl-223-72-14.aep.bellsouth.net) |
| 14:09.40 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 14:09.40 | yxa | Delvar you have any alternatives to authentication w/o using sql? :) |
| 14:09.53 | deezed | yxa i believe so... you have to make install the asterisk addons for mysql to work |
| 14:10.18 | Delvar | yxa: no but id use an aqi scrip tto do it :) |
| 14:10.30 | Delvar | agi* |
| 14:10.36 | deezed | yxa you can do it from a text file with every password in it |
| 14:10.48 | deezed | you can't select it buy EXTEN though |
| 14:10.56 | Meaty | stevie20 ! I realy dont have this problem, try with another codec. |
| 14:11.04 | Meaty | Only for test |
| 14:11.21 | synthetiq | what would be a reason for phones to contantly lose registraion on the same lan all the time when others are fine....? |
| 14:11.28 | stevie20 | ok meaty, thank you... |
| 14:11.34 | yxa | deezed how is the syntax of the passwd file like? just a passwd every new line? |
| 14:11.47 | deezed | yes |
| 14:12.00 | yxa | deezed seems like a security hole to me |
| 14:12.02 | stevie20 | lme, we two are the only ones, which has these problem, thas disallow=all wont work... |
| 14:12.51 | lme | *************** |
| 14:13.33 | jmjones | i've got my asterisk server running behind my router on my lan at home. i've got "nat=route" in the [general] section of my sip.conf file and have ports 5060-5070 and 8766-35000 forwarded in to my asterisk server |
| 14:13.57 | jmjones | but when i call it from the outside, i get a connection, but can't hear anything... |
| 14:14.15 | InfraRed | do you get anything on the CLI |
| 14:14.15 | deezed | yep. for my dialplan i just want the users to be able to use their realtime voicemail password so they can change it via the website or voicemail app, but i didn't want to use the VMAuthenticate app |
| 14:15.10 | jmjones | InfraRed - doh, lemme check the logs. this was from yesterday. i got a voicemail message and no error on the client, so i'm assuming it must not have had too big of a problem... |
| 14:19.14 | *** join/#asterisk graphyx (n=mike@67.50.46.118) |
| 14:19.26 | graphyx | Can I put a line in the dial plan that logs a CDR record? |
| 14:19.56 | Gourou_fou | http://www.ges.fr/voip/product_info.php?products_id=221&osCsid=bae74b8452aab08eab8bdb466749c140 |
| 14:20.07 | Gourou_fou | 65 euros for iax tel |
| 14:20.48 | InfraRed | graphyx: look at AGI's in voip-info.org |
| 14:21.03 | InfraRed | graphyx: and cdr enteries |
| 14:22.46 | *** join/#asterisk Inkubot (n=inkubot@200.75.4.7) |
| 14:23.01 | Inkubot | how can i use the flash key for transfer ? |
| 14:23.16 | Meaty | graphyx : You can use userfield in the cdr to addinfo |
| 14:23.18 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 14:23.49 | Meaty | graphyx : with setuserfield or appenduserfield |
| 14:25.15 | *** join/#asterisk Gh0sty (n=ghosty@ip-81-11-227-234.dsl.scarlet.be) |
| 14:25.43 | kippi | on asterisk, can you have a group, and add a group into that group? |
| 14:25.43 | Gh0sty | hello |
| 14:25.46 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 14:25.52 | Gh0sty | bit of a crowd here :) |
| 14:26.25 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 14:26.26 | Gh0sty | is there a way to make the time it takes to pick up the analog line shorter? |
| 14:26.36 | Meaty | kippi : group of channels? |
| 14:26.43 | kippi | yeah |
| 14:27.38 | kippi | also is there away you can ring a group from a handset? |
| 14:27.44 | Gh0sty | seems it rings twice before asterisk picks up, then it takes another 5 seconds until you are connected with the extension (which is 5 seconds of silence on the caller ...) |
| 14:27.49 | *** join/#asterisk zagaya971 (n=almeli@APointe-a-Pitre-102-1-11-9.w81-248.abo.wanadoo.fr) |
| 14:28.16 | Meaty | you can create group with Zap channel whith group=1 and to Dial(Zap/g1) (group 1) |
| 14:28.59 | Katty | kippi: yes. |
| 14:29.20 | Katty | kippi: you can also call sip/2&sip/1&sip/2000005 etc |
| 14:29.55 | Katty | kippi: like extension 17 is a blast group for multiple sip peoples. |
| 14:30.27 | yxa | when i place a call file to do authenticate on say Zap/1/1234567, it starts executing when Zap/1-1 was answered. That's not what supposed to happen. its supposed to answer when the land line connected. how do i correct this? |
| 14:30.53 | *** join/#asterisk robtro (i=rob@unaffiliated/robtro) |
| 14:31.20 | robtro | i have a SIP client on a wireless connection (802.11x internally) that doesnt recieve INBOUND calls, where should i start looking? |
| 14:31.26 | robtro | does sip qualify have anything to do with that |
| 14:31.40 | Katty | robtro: hi |
| 14:31.47 | robtro | Katty: hi |
| 14:31.52 | Katty | robtro: how're you today? |
| 14:32.12 | robtro | wonderful how are you |
| 14:32.18 | lme | robtro: outgoing calls are ok ? sip and rtp streams ? |
| 14:32.26 | Katty | robtro: do you usually get version information? |
| 14:32.31 | Katty | robtro: do you like my client? heh |
| 14:32.35 | robtro | is this a bot |
| 14:33.35 | [TK]D-Fender | robtro : have you tried it with that PC wired? |
| 14:33.49 | InfraRed | anyone here using IAX -> IAX -> SIP setup ? |
| 14:33.51 | robtro | its a laptop - network card doesn't work. |
| 14:34.06 | robtro | everything else is fine. all other machines and phones are working allright. |
| 14:34.12 | robtro | just this one, it's the only wireless one. |
| 14:34.13 | [TK]D-Fender | robtro : IS it your own personal lan? AAny NAT involved? |
| 14:34.26 | robtro | not a personal lan, but yes - MY lan |
| 14:34.40 | [TK]D-Fender | Ok, so the * box is on the same subnet then? |
| 14:34.44 | robtro | yep. |
| 14:34.53 | lme | laptop ? so this is a softphone ? |
| 14:34.57 | robtro | lme: yes. |
| 14:35.01 | [TK]D-Fender | which? |
| 14:35.11 | ManxPower | I have seen Linksys wireless boxes prevent devices from communitcating between the LAN ports of the box |
| 14:35.13 | lme | robtro: just to be sure... no firewall problems ? xp sp2 ?! |
| 14:35.32 | robtro | firewall `problems' ? |
| 14:35.40 | robtro | this is a cisco aironet BRIDGE |
| 14:35.44 | robtro | plugged directly into the switch. |
| 14:35.45 | Katty | sp2 :<<<< |
| 14:35.59 | [TK]D-Fender | robtro : just make sure UDP is allowed through. So which softphone are you trying? |
| 14:36.00 | ManxPower | robtro, at least you are using high end equipment 8-) |
| 14:36.01 | robtro | yeah it's SP2 |
| 14:36.11 | robtro | xten, as well as sjphone |
| 14:36.13 | robtro | same issue. |
| 14:36.23 | Katty | robtro: make sure your windows firewall thingy is off |
| 14:36.31 | Katty | robtro: and if you have norton internet security, kill it |
| 14:36.36 | lme | robtro: imho, your windows's firewall is blocking SIP headers. |
| 14:36.40 | robtro | that's all off. |
| 14:36.49 | robtro | windows firewall is always off. |
| 14:36.51 | robtro | it's the devil. |
| 14:36.55 | Katty | hmm. |
| 14:37.07 | ManxPower | doing a tcpdump/ethereal on the Asterisk box would give you additional information |
| 14:37.21 | Laerte | anyone has problem with group ad attended transfer in *-HEAD ? |
| 14:37.35 | robtro | when i call the laptop, * says the users is unvaialable, doesnt even ring. |
| 14:37.46 | robtro | its ringing now |
| 14:37.46 | robtro | weird |
| 14:37.48 | lme | robtro: is this an intel centrino device ? |
| 14:37.59 | robtro | no |
| 14:38.01 | robtro | lucent card. |
| 14:38.22 | ManxPower | robtro, "sip show peers" should show the peer's SIP lag. |
| 14:38.31 | lme | maybe you should try to windump your 802.11b/g/a link |
| 14:38.46 | *** join/#asterisk rocket (n=rocket@gentoo/developer/rocket) |
| 14:40.11 | *** part/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 14:40.36 | robtro | no leg |
| 14:40.40 | ManxPower | yxa, I have an answer for you. |
| 14:40.45 | robtro | STATUS Unmonitored |
| 14:41.17 | yxa | ManxPower i'm listening |
| 14:41.46 | ManxPower | yxa, ANALOG FXO ports are considered ANSWERED as soon as dialing is finished. |
| 14:41.47 | *** join/#asterisk gres (n=serg@62.152.85.54) |
| 14:42.02 | yxa | ManxPower yeah i figured. any workarounds? |
| 14:42.26 | ManxPower | yxa, nothing good. My fix is to always use a PRI where I need that functionality |
| 14:43.27 | ManxPower | yxa, Traditional PBXs have the same issue with analog FXO ports. |
| 14:43.28 | yxa | ManxPower pretty out the question... |
| 14:43.45 | ManxPower | yxa, HINT: Practically all ITSPs use PRIs |
| 14:44.35 | gres | Hi all. Does anybody have succes with accessing cisco ip phone (7960) from console cable? |
| 14:45.47 | *** join/#asterisk pattieja (n=pattieja@adsl-69-153-174-41.dsl.stlsmo.swbell.net) |
| 14:46.14 | lme | robtro: if you want it to be monitored, you have to add a qualify directive |
| 14:46.31 | yxa | ManxPower how abt: can i loop some music until a key is pressed? |
| 14:47.15 | robtro | lme: qualify= what |
| 14:47.19 | ManxPower | yxa, I don't know. |
| 14:47.24 | ManxPower | robtro, qualify=yes |
| 14:47.30 | yxa | ManxPower then do the authenticate or whatever |
| 14:47.39 | ManxPower | yxa, I use a .call file to play a message over and over to the callee |
| 14:48.04 | lme | robtro: qualify=xxx xxx=time in ms to be considered as up |
| 14:48.09 | robtro | 18ms |
| 14:48.21 | ManxPower | "yes" is 2000 |
| 14:48.29 | yxa | ManxPower how do i loop a background? |
| 14:48.36 | ManxPower | robtro, now, look at that same number when you can't call the phone |
| 14:49.11 | ManxPower | yxa, use Local/whatever in the Channel: field of the .call file where the "whatever" is an extension in your dialplan that plays a message over and over until someone presses a key |
| 14:50.01 | ManxPower | Or put the Zap channels in the Channel: field of the .call file and the specify the whatever extension on the Extension: and Priorty: fields of the .call file. |
| 14:50.27 | ManxPower | yxa, Search the mailing list archives, you are not the first person with this issue. |
| 14:50.37 | InfraRed | I have the following setup : SIP phone -> *(1) -> IAX -> *(2) -> SIP termination company. I keep getting Call rejected by *(2): No such context/extension i have the extention dial and context set in the dialplan on the *(2).... any ideas? |
| 14:51.17 | *** join/#asterisk toddf (n=toddf@ns0.fries.net) |
| 14:51.28 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 14:52.11 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 14:52.15 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 14:52.17 | lme | <PROTECTED> |
| 14:52.25 | ManxPower | infinity1, do you have the context set in the iax.conf section of the [whatever] section of sip.conf? |
| 14:52.27 | yxa | ManxPower trying my best... |
| 14:52.28 | *** join/#asterisk Abbas (n=Abbas@203.81.194.242) |
| 14:52.28 | zoa | hey ho |
| 14:53.09 | InfraRed | lme: fancy having a look at my conf ? |
| 14:53.13 | InfraRed | could be something silly |
| 14:54.22 | fourcheeze | InfraRed: did you misspell "context" in your sip.conf? |
| 14:55.08 | InfraRed | it should be iax.con |
| 14:55.10 | InfraRed | right ? |
| 14:55.13 | InfraRed | hmm |
| 14:55.14 | InfraRed | i get you |
| 14:55.16 | InfraRed | worth checking |
| 14:55.54 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-b262d7444c057a9f) |
| 14:56.04 | *** join/#asterisk jmacz (n=neuroman@200.24.113.153) |
| 14:58.30 | InfraRed | fourcheeze: no :/ |
| 14:58.37 | InfraRed | it's all right in there |
| 14:59.15 | hhoffman | hi, I'm using a Tiger3XX card, X100P clone I believe. Are they known to cause alot of crackling? |
| 15:07.01 | *** join/#asterisk paryl (n=paryl@216-201-177-82.res.logixcom.net) |
| 15:07.11 | paryl | g'morning all |
| 15:08.29 | *** join/#asterisk mkrufky (n=mk@68.160.103.77) |
| 15:09.30 | ManxPower | hhoffman, no. But it can happen if there's a problem, like the card shareing interrupts |
| 15:10.27 | paryl | when using a TE205P to connect to a T1, is echo ever an issue? i ask because i just finished messing with TDM400P's, and the echo issues drove me crazy |
| 15:10.37 | kippi | I would like to try and get a meeting with the develp. of asterisk, how easy do u think this will be? |
| 15:11.56 | zoa | kippi, there is not just 1 developper of asterisk |
| 15:11.58 | ManxPower | paryl, Echo has NOTHING to do with YOUR interface to the PSTN, just the interface of the OTHER END. |
| 15:11.59 | stbain | kippi: depends... you have time to drive and/or fly to Huntsville, Alabama? |
| 15:12.16 | zoa | the best thing is to attend a conference |
| 15:12.25 | ManxPower | paryl, Granted, if you have an analog interface to the PSTN you could also get echo with your connection to the internet. |
| 15:12.26 | zoa | why do you want a meeting ? |
| 15:12.34 | zoa | i missed the first part of your talks |
| 15:12.45 | ManxPower | zoa, He works for Nortel and wants to take out all the Asterisk developers at once. |
| 15:13.14 | zoa | haha |
| 15:13.15 | stbain | get 'em all in the same room and "poof!" no more competition |
| 15:13.16 | [TK]D-Fender | paryl : Echo is a problem for any digital device. I had it pretty bad with my TE405P's, my home TDM400 as well. |
| 15:13.35 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 15:13.48 | hhoffman | ManxPower: cat /proc/interrupts shows wcfxo using int 11, I don't see anything else using that |
| 15:13.54 | ManxPower | [TK]D-Fender, Yes, but you only HEAR echo when you have a high latency connection |
| 15:14.13 | ManxPower | ANY VoIP would be "high latency? |
| 15:14.52 | kippi | stbain: is there no one in the uk that we can talk to ? |
| 15:15.05 | zoa | depends on what you need kippi |
| 15:15.08 | zoa | see private msg |
| 15:15.16 | ManxPower | Cell phone would also be a high latency connection, which why cell carriers have incredible amounts of EchoCan |
| 15:15.29 | stbain | I'm sure there's probably at least one devel in the UK. |
| 15:17.38 | paryl | manx, fender: thanks |
| 15:20.12 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 15:21.31 | steff | i all, i have an X100P who don't answering incoming call, ztmonitor show ring , but no action in asterisk (sorry for my poor english :-) |
| 15:21.33 | *** join/#asterisk johnrage (n=jabetong@212.93.201.89) |
| 15:22.08 | coppice | digital cellular really drove the development of good echo can |
| 15:22.35 | coppice | stbain: there's over 600 devils in parliament alone |
| 15:24.59 | *** join/#asterisk Speeder (n=psilva@pal-213-228-158-41.netvisao.pt) |
| 15:26.19 | johnrage | hello guys. |
| 15:27.03 | johnrage | anybody here can help me setup a VIRTUAL numbers or DID using an ATA device. PM me |
| 15:28.27 | _Sam-- | /quit |
| 15:28.33 | [TK]D-Fender | johnrage : What is the origin of the call? |
| 15:29.16 | *** join/#asterisk _Thor (i=CS@user-vc8fl7n.biz.mindspring.com) |
| 15:29.39 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
| 15:30.40 | _Thor | anyone with experience in installing mysql who can help me? |
| 15:31.56 | stbain | apt-get install mysql |
| 15:32.13 | stbain | errr.... what sort of assistance do you need, _Thor |
| 15:32.23 | _Thor | I wish, I have been trying to do the rpm install... maybe 6 times |
| 15:32.26 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 15:32.44 | _Thor | I will like to install mysql 5.0 using the rpm package |
| 15:32.49 | jvictorfc | is possible convert mp3 to gsm format? |
| 15:33.03 | _Thor | ...but I am getting some kind of key error |
| 15:33.18 | jvictorfc | for using with playback() and backgroud()? |
| 15:33.20 | _Thor | ...not only that, it doen't cretae /usr/bin/mysql |
| 15:34.00 | _Thor | I did install 4.1 fully, but I did not find /usr/bin/mysql, and I uninstalled it |
| 15:35.14 | stbain | RPM package off of the MySQL site, or RPM package from your distribution's site (e.g. from redhat.com)? |
| 15:35.33 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
| 15:36.11 | _Thor | specific question is: in the rpm version, anyone has had trouble with the key?, or else the other question is: why will it not do the /usr/bin/mysql? |
| 15:36.23 | _Thor | RPM from the mysql.com |
| 15:36.35 | asterboy | also make sure you check all possible places for mysql, somethimes it installs in other directories like /usr/local |
| 15:37.21 | asterboy | If 4.1 installed without errors, it was successfully put on your system...do a "find / -name mysql" |
| 15:37.29 | _Thor | all the instructions indicate to install in /usr/local |
| 15:37.39 | _Thor | ...which is where I installed |
| 15:38.23 | _Thor | I want to uninstall 5.0 |
| 15:38.33 | _Thor | and I erased the files |
| 15:39.04 | _Thor | but when I do a rpm -qa | grep MySQL it says it is installed |
| 15:39.09 | *** join/#asterisk rikstah (n=rick@80.229.114.105.plusnet.pte-ag2.dyn.plus.net) |
| 15:39.16 | _Thor | who do I erase it completely from linux? |
| 15:39.55 | lunk | rpm is the devil |
| 15:40.58 | _Thor | you are right |
| 15:41.45 | _Thor | do you know how to wipe it out? |
| 15:43.42 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 15:44.12 | *** join/#asterisk DYOGI_B (n=Jade@dsl-202-173-190-245.qld.westnet.com.au) |
| 15:44.35 | *** part/#asterisk lme (n=lme@gw-r-ifc.infoclip.fr) |
| 15:44.39 | DYOGI_B | hi anyone know why I can't connect to the call manager this is driving me razyyyyyyyyy |
| 15:45.00 | jvictorfc | is possible convert mp3 to gsm format for using playback() or backgroud()? |
| 15:46.02 | hhoffman | hmm, so if I play a file when calling in on my PSTN line that I don't get any static, but if I use Echo there is tons of noise :-? |
| 15:46.03 | DYOGI_B | dont thinks so |
| 15:46.30 | DYOGI_B | i mean sorry use wave pad that will help |
| 15:47.00 | DYOGI_B | and it is free |
| 15:47.09 | Lostfrog | Has anyone gotten call parking working with a function key on a snom phone? |
| 15:47.10 | DYOGI_B | mp3-->gsm |
| 15:47.27 | DYOGI_B | does anyone know manager.conf |
| 15:47.35 | *** join/#asterisk Voicelynx (n=rda@8.8.197.77) |
| 15:48.03 | asterboy | morning. |
| 15:48.20 | docelmo | YIPPIE! |
| 15:48.43 | asterboy | it no longer burns to pee? |
| 15:48.49 | asterboy | :P |
| 15:49.14 | tzanger | hahaha |
| 15:49.38 | _Thor | DYOGI_B: What about manager.conf |
| 15:49.40 | asterboy | the wonders of medication! |
| 15:50.39 | DYOGI_B | does anyone know manager.conf |
| 15:50.50 | DYOGI_B | well it makes me crazy |
| 15:51.07 | DYOGI_B | i can't connect anything to it |
| 15:51.07 | DYOGI_B | just won't work |
| 15:51.13 | DYOGI_B | is there any reason why this could be |
| 15:52.14 | ManxPower | ~mailinglist |
| 15:52.15 | jbot | it has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 15:52.20 | ManxPower | ~docs |
| 15:52.22 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 15:52.27 | _Thor | DYOGI_B: check username and pwd |
| 15:52.57 | *** part/#asterisk Bentley (n=Bentley@S0106000f3d016dd2.cg.shawcable.net) |
| 15:53.22 | _Thor | DYOGI_B: check port number, and... are you sure your sockets are talking? |
| 15:54.27 | tclark_ | anyone want a deal on the old 4 port t400 t1 cards ? |
| 15:55.09 | tclark_ | were talking blow out here $200-300 |
| 15:55.11 | Lostfrog | Nice.. parked calls go back to the device they came in on.. :( |
| 15:55.27 | Lostfrog | Parked SIP/192.168.11.95-08760848 on 71. Will timeout back to extension [incomingfxo] 401, 1 in 45 seconds |
| 15:55.31 | Lostfrog | Explain that to me. |
| 15:57.24 | ManxPower | Lostfrog, It's the way all other PBXs do call parking. |
| 15:57.59 | ManxPower | 1.2x has does the CORRECT thing. 1.0.x just timed out to exten s, which is not the way other PBXs do it. |
| 15:58.16 | Lostfrog | Umm.. send the call back to the fxo port?? |
| 15:58.21 | asterboy | scary, everything here is logged: http://www.asterisknerds.com/logs/irclogger_logs/asterisk I'm sure that's the case for all IRC |
| 15:58.29 | ManxPower | Lostfrog, Hmm? |
| 15:58.32 | asterboy | great to search through for support. |
| 15:58.36 | Lostfrog | Say the call came in on zap/1, should the call be return to zap/1? |
| 15:58.40 | ManxPower | No other PBXs send the call back to the phone that parked it. |
| 15:58.51 | Lostfrog | Correct, which I would like to happen. |
| 15:58.54 | ManxPower | and 1.2x now does that too. |
| 15:58.56 | *** join/#asterisk emakris2 (n=emakris@c-24-128-56-2.hsd1.ma.comcast.net) |
| 15:58.56 | *** part/#asterisk graphyx (n=mike@67.50.46.118) |
| 15:59.01 | Lostfrog | 401 is the extension is came in on. |
| 15:59.08 | Lostfrog | 401@anotherdevice. |
| 15:59.19 | asterboy | Where is the official place to purchase the Asterisk Book, "The Future of Telephony" |
| 15:59.19 | asterboy | ?? |
| 15:59.21 | ManxPower | so what's the extension that PARKED it? |
| 15:59.26 | Lostfrog | 1011 |
| 15:59.29 | ManxPower | asterboy, ora.com |
| 15:59.34 | ManxPower | Lostfrog, report it as a bug. |
| 15:59.37 | asterboy | thnk |
| 15:59.41 | ManxPower | and do it fast before 1.2 is released |
| 15:59.56 | Lostfrog | Let me do one more test. |
| 16:00.17 | asterboy | dam, I hate going to ora.com! |
| 16:00.41 | brettnem | hey anyone using DEVSTATE? |
| 16:00.53 | *** join/#asterisk ecto (n=ectospas@69.85.202.2) |
| 16:01.24 | ecto | The safe_asterisk man page mentions screendump, but I don't have that on my system. Does anyone know what package that's a part of? |
| 16:01.45 | ManxPower | ecto, "screen" or "screendump" |
| 16:02.08 | Lostfrog | Today is not my day.. |
| 16:02.08 | ecto | screen will not work for my purposes |
| 16:02.11 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:02.17 | Lostfrog | The hookswitch on one of my snoms is backwards. |
| 16:02.27 | asterboy | now I have to buy a reg_ex book and a perl book and a python book and a php book and a mysql book and a javascript book and a ... |
| 16:02.29 | ecto | screendump isn't on my system. I need to look at virtual terminal 9, remotely |
| 16:02.31 | Lostfrog | You take the phone off hook and the speaker phone comes on. :( |
| 16:03.25 | mut | k i gotta name a new server but i can't decide what to use |
| 16:03.29 | mut | VOTE! |
| 16:03.30 | mut | foghorn, pepe, or yosemite |
| 16:03.48 | ManxPower | ecto, Why do you need to look at vtty9? |
| 16:03.55 | tclark_ | and also have 1 t100p & zplex 24 fxs channel bank yo can have for $250 |
| 16:04.01 | Chuji | Is there any problem with type=friend in 1.2? I can get my asterisk A server (1.0) to talk to my asterisk B server (1.2) but not vice versa |
| 16:04.13 | ecto | Because the output on STDERR for my AGI script doesn't show up, and that's the last place I've got left to look |
| 16:04.39 | Chuji | That's over IAX2 by the way |
| 16:04.56 | ManxPower | ecto, start screen, then start asterisk as asterisk -cvvv that will put STDERR on your current console, "screen" will let you disconnect and reconnect without killing Asterisk |
| 16:05.06 | Lostfrog | tclark_: That is an awesome price. |
| 16:05.10 | Lostfrog | Wish I could use it. :) |
| 16:05.38 | ecto | But screen has its own scroll buffer, and I can't see this output. |
| 16:05.42 | tclark_ | yah just getting dust here, I switch over to sangoma gear |
| 16:06.33 | Lostfrog | ManxPower: What info do I need to find out to submit this? |
| 16:06.45 | Chuji | Call rejected by 10.50.4.9: No authority found |
| 16:06.47 | ManxPower | Lostfrog, not much as far as I can tell. |
| 16:06.56 | Chuji | Is that basically username/pass problems? |
| 16:07.05 | ManxPower | Chuji, that means the incoming call had a username/secret/context that didn't match your iax.conf |
| 16:07.33 | *** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com) |
| 16:08.08 | Chuji | ManxPower : That's what I would have thought, but everythign looks peachy in iax2 debug |
| 16:08.08 | mut | O_o |
| 16:08.08 | mut | no votes eh |
| 16:08.21 | brettnem | hey anyone using DEVSTATE with 1.2? |
| 16:08.24 | ManxPower | Just remember that for incoming calls it's match against [blahsection] in the destination iax.conf |
| 16:08.25 | asterboy | tclark_: I'm interested in the fxs stuff |
| 16:08.26 | mut | 1-0-1 so far |
| 16:09.23 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 16:09.55 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@mail.airlinksystems.com) |
| 16:10.46 | asterboy | tclark_: what is the web page for zplex? |
| 16:11.24 | Lostfrog | zhone.com? |
| 16:11.30 | Chuji | ManxPower : what does it match on though? |
| 16:12.02 | Lostfrog | oh wait.. zhone doesn't avow the existence of the zplex10. |
| 16:12.17 | asterboy | zphone.com not available. |
| 16:12.20 | Chuji | ManxPower : How does it know what the name of the incoming context? |
| 16:12.25 | asterboy | who makes it? |
| 16:12.49 | *** join/#asterisk rjuan (n=rjuan@233.Red-217-127-61.staticIP.rima-tde.net) |
| 16:15.26 | *** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au) |
| 16:15.42 | DYOGI_B | da |
| 16:15.45 | DYOGI_B | this sucks |
| 16:17.03 | tclark_ | asterboy: zhone.com |
| 16:17.16 | ManxPower | Chuji, it doesn't unless you specify it on the Dial line or the local config in iax.conf. I never specify a context on my dial and let the context= on the remote server route the call |
| 16:17.27 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 16:17.58 | Lostfrog | ManxPower: now it is working. :( |
| 16:18.08 | asterboy | http://www.wwworks-inc.com/asterisk/ <--- has a nice table of hardware. |
| 16:18.47 | Chuji | ManxPower : ok, I think I found the problem, seems buggy to me |
| 16:19.23 | Chuji | ManxPower : If I use context=default and default has include=>incoming it doesn't work |
| 16:19.34 | Chuji | ManxPower : If I use context=incoming, it works |
| 16:19.49 | Lostfrog | Now, if someone call tell me how to park calls on a snom without using #700. |
| 16:19.55 | ManxPower | Chuji, is the destination extension a non-number? |
| 16:19.57 | asterboy | zplex has some callerID issues. |
| 16:20.10 | Chuji | Nope, 4303 |
| 16:20.12 | asterboy | power supplies are reported problems |
| 16:20.25 | ecto | ManxPower: how do you scroll back in screen? The problem I have with it is that I can't see more than 24 lines at a time, and this * server is too busy for me to see my output, if it's there at all... |
| 16:20.27 | tclark_ | only on fxo ports |
| 16:20.38 | ManxPower | ecto, "man screen" |
| 16:20.41 | *** join/#asterisk Bicster_ (n=Bicster@pdpc/supporter/active/Bicster) |
| 16:20.43 | Lostfrog | My callerid on my zplex10a works fine. |
| 16:20.49 | Chuji | ecto ^a ^[ |
| 16:20.55 | tclark_ | and p/s was wade issue this one has run 24/7 since i bought it afew yesars ago |
| 16:20.57 | Chuji | puts you in scrollback |
| 16:20.58 | Bicster_ | does anyone know if SBC will put a rollover on a residential POTS line in TX? |
| 16:21.22 | ecto | thanks Chuji |
| 16:21.30 | ManxPower | Bicster_, The call it "Call Forward/NoAnswer and Call Forward/Busy" Some telcos call it the "voicemail companion package" |
| 16:21.46 | *** join/#asterisk azzie (n=az@azzie.net) |
| 16:21.52 | Bicster_ | ManxPower, they only seem to offer that call forward feature on business lines...and I don't want the "no answer" part of the equation |
| 16:22.03 | ManxPower | Bicster_, if you use the term "rollover" or "hunt group" they will want you to get a business line |
| 16:22.03 | asterboy | ok good to hear, sounds like a great product to connect a lot of analog phones. |
| 16:22.10 | tclark_ | yah wade pages on the Z-PLEX-10-24S/O |
| 16:22.20 | Chuji | ManxPower : So does that seem buggy to you? That's not the behavior in 1.0. It does honor includes |
| 16:22.22 | tclark_ | yah that works fine |
| 16:22.34 | ManxPower | Chuji, you have something else going on. |
| 16:22.42 | tclark_ | this a Z-PLEX-10-24S |
| 16:24.45 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 16:25.26 | *** part/#asterisk Bicster_ (n=Bicster@pdpc/supporter/active/Bicster) |
| 16:26.45 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 16:26.51 | asterboy | Anyone done an asterisk and hylafax install? |
| 16:28.24 | asterboy | I'd be amazed if the Zaptel can send/receive faxes...if I'm reading the docs correctly, it is possible, no? |
| 16:29.55 | sahafeez | ~fax |
| 16:29.56 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 16:31.48 | sahafeez | asterboy: http://www.google.com/custom?tk=9caca5c460c38f97a129&q=fax&sa=Google+Search&cof=S%3Ahttp%3A%2F%2Fwww.voip-info.org%3BGL%3A0%3BAH%3Aleft%3BBGC%3AE9ECEF%3BL%3Ahttp%3A%2F%2Fwww.voip-info.org%2Fimages%2FVOIP-info.jpg%3BAWFID%3A866fed4e998eaa65%3B&domains=www.voip-info.org&sitesearch=www.voip-info.org |
| 16:32.15 | *** part/#asterisk aaronz (n=aaronz@pdpc/supporter/student/aaronz) |
| 16:34.49 | asterboy | Interesting...looks like the most reliable method is to dedicate a fax line and let hylafax do its job using modems, instead of the X100P emulating a modem, no? |
| 16:34.55 | RoyK | does anyone know an easy way to use asterisk for outgoing faxes from a windows box? perhaps a windows 'printer'? or emeail? |
| 16:35.09 | tzanger | RoyK: hylafax has several "printers" that do just this |
| 16:35.37 | asterboy | RoyK: I use hylafax and samba to do that very thing. |
| 16:36.27 | asterboy | RoyK: Once you capture output from the windows box, email, print, fax all become easy. |
| 16:36.45 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 16:36.53 | christo | asterboy - spandsp and txfax/rxfax are good |
| 16:36.55 | RoyK | asterboy: using this with spandsp? |
| 16:37.06 | RoyK | i have spandsp setup for incoming faxes |
| 16:37.10 | asterboy | no...just hylafax. |
| 16:37.28 | asterboy | I want to try that spandsp though...sounds like the asterisk way to do things. |
| 16:37.37 | RoyK | hylafax doesn't speak with te410p cards.... |
| 16:37.43 | RoyK | nor PRIs |
| 16:38.10 | asterboy | RoyK: http://www2.stealthdigitalservice.com:8080/download/Print2File/print2file.html <--- this may be what your loking for. |
| 16:39.06 | asterboy | yes thats most likely the case. |
| 16:39.49 | asterboy | but I imagine you could detect the fax and transfer it to a modem, no? |
| 16:40.07 | RoyK | spandsp |
| 16:40.15 | RoyK | and app_rxfax |
| 16:40.17 | RoyK | and app_txfax |
| 16:40.48 | zoa | http://www.asteriskguru.com/tutorials/spandsp.html |
| 16:42.46 | asterboy | Can anyone speak for the asterisk to fax connection reliability?? |
| 16:43.13 | ManxPower | asterboy, Yes. It sucks. |
| 16:43.27 | asterboy | that is what I figured...it is after all emulating. |
| 16:44.19 | asterboy | It does this through the X100P? or in the case of DID numbers...hmmmm. |
| 16:44.42 | asterboy | sounds like it doesn't matter what the hardware is, just emulates. |
| 16:50.55 | asterboy | ok, so I have asterisk and hylafax on the same box...no matter...the modems for hylafax can do their magic while asterisk does its magic. |
| 16:52.33 | asterboy | DID numbers for Alberta are $3 |
| 16:52.43 | asterboy | Anyone know of better price? |
| 16:53.17 | ManxPower | asterboy, I pay $20/month per 100 DIDs |
| 16:53.27 | asterboy | sweet! |
| 16:53.42 | Uther_P | ManxPower: what area are the did's ? |
| 16:53.52 | ManxPower | Uther_P, I get them from my local CLEC |
| 16:53.56 | asterboy | Hence the DID exchange we site...you can sell some of those. |
| 16:54.02 | asterboy | s/we/web |
| 16:54.15 | Uther_P | did exchange site... dude, where? |
| 16:54.24 | johnrage | I am looking for PHILIPPINE DID |
| 16:54.41 | asterboy | http//:didx.org |
| 16:54.55 | Uther_P | cool, thans |
| 16:54.57 | Uther_P | k |
| 16:55.08 | ManxPower | Why would I want the hassle of selling the DIDs? |
| 16:55.19 | asterboy | I guess you just pay a small membership fee and post up what you want to sell or buy. |
| 16:55.25 | ManxPower | Especially since my servers are all behind a firewall |
| 16:55.56 | asterboy | ManxPower: In case your like me and want just a few but have to buy them wholesale bulk. |
| 16:56.04 | asterboy | ManxPower: to get a good price. |
| 16:56.24 | ManxPower | asterboy, We have a good relationship with our CLEC |
| 16:56.27 | asterboy | ManxPower: or maybe you want a grab bag of area codess. |
| 16:56.34 | asterboy | ManxPower: sounds like it! |
| 16:56.47 | ManxPower | For my PERSONAL/SOHO stuff I pay the VoIP DID rates. |
| 16:57.12 | ManxPower | asterboy, We are our CLEC's 3rd largest customer. So they pretty much do anything we tell them to do. 8-) |
| 16:57.19 | asterboy | ManxPower: lol |
| 16:57.41 | hhoffman | are DIDs portable? |
| 16:57.49 | asterboy | I'm a noob to DID, how exactly does it work and setup on asterisk? |
| 16:58.25 | asterboy | Sounds like its the cheapest way to get phone numbers cause you can bypass the digium equipment. |
| 16:58.34 | asterboy | Do it all via HighSpeed Internet. |
| 16:58.44 | *** join/#asterisk kink0 (n=k@62.37.205.161) |
| 16:58.46 | kink0 | hello !! |
| 16:58.47 | Qwell | "cheapest" is almost never "best" |
| 16:59.06 | LostFrog | You still have to pay a provider to provide termination. |
| 16:59.15 | kink0 | I have dialed from one PC to other PC, there anyway I play a mp3 or so to hear in the called PC ? |
| 16:59.20 | [hC] | cheap, quick, good.. pick two. |
| 16:59.44 | Qwell | [hC]: if that many, sometimes |
| 16:59.45 | ManxPower | asterboy, You mean route your phones calls across the "super reliable internet"? I want some of the drugs you are on. |
| 16:59.47 | asterboy | well, ignore the word cheap |
| 16:59.51 | asterboy | "efficient" |
| 16:59.59 | asterboy | s/cheap/efficient |
| 17:00.04 | *** join/#asterisk afrosheen (n=afro@c-67-187-0-137.hsd1.tx.comcast.net) |
| 17:00.23 | ManxPower | asterboy, I do everything I can to avoid routing calls over the internet. |
| 17:00.30 | asterboy | ok, so how do you terminate the DID at your end? With T1/E1 |
| 17:00.32 | asterboy | ? |
| 17:00.55 | ManxPower | asterboy, Yes. Except for the 2 - 3 personal low use numbers. |
| 17:01.14 | asterboy | ah...for those you route via internet? |
| 17:01.25 | ManxPower | We are slowly putting in place the ability for Asterisk to do incoming and outgoing calls when 1) the PRI is down or 2) when the PRI is full. |
| 17:01.27 | afrosheen | asterboy: did you get your fxo/fxs stuff working yesterday? |
| 17:01.36 | *** part/#asterisk otaku42 (i=otaku@madwifi/developer/otaku42) |
| 17:02.31 | asterboy | afrosheen: yes, it works perfectly...IronHelix caught the hardware going to Channel 5, I was assuming channel 1 |
| 17:02.47 | afrosheen | good eye :) |
| 17:03.02 | asterboy | yes, now I'm exploring all the possibilities with Asterisk |
| 17:03.25 | afrosheen | there are alot of them, I'd explore a little at a time if I were you |
| 17:03.44 | asterboy | My ADHD doesn't allow me to do that :( |
| 17:04.27 | afrosheen | when I can't stop fiddlin', I just takes me ritalin <--simpsons quote |
| 17:04.43 | asterboy | lol...that stuff is legal cocain. |
| 17:04.57 | asterboy | none for me thanks...I just hit the bong |
| 17:05.27 | afrosheen | did you say 'thank you jesuuuuuus' after you hit it? |
| 17:05.42 | afrosheen | tom licas reference..probably lost here :) |
| 17:05.47 | asterboy | on my knees...arms raised! |
| 17:05.53 | *** join/#asterisk hhrp (i=zloydyad@c-66-176-86-87.hsd1.fl.comcast.net) |
| 17:05.59 | hhrp | hi |
| 17:06.03 | asterboy | high |
| 17:06.07 | asterboy | :-> |
| 17:06.27 | asterboy | couldn't resist that. |
| 17:06.44 | hhrp | i did update my * from cvs, did compile it but it still shows an old version running |
| 17:06.49 | hhrp | what could be the prob? |
| 17:06.52 | syle | whats wrong with routing calls out the internet |
| 17:07.03 | afrosheen | security..latency...etc. |
| 17:07.13 | syle | never had a problem |
| 17:07.34 | asterboy | ya, I gotta say, so far I've not had a problem. |
| 17:07.49 | afrosheen | we've had problems before, our provider blows up switches monthly |
| 17:07.53 | asterboy | Is there a way to get DID over internet and terminate with SIP phones? |
| 17:08.03 | afrosheen | yeah definitely |
| 17:08.09 | syle | well yeah colocation is key |
| 17:08.11 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:08.14 | asterboy | That is what I'm looking for. |
| 17:08.32 | afrosheen | asterboy: our provider offers it cheap, www.txlink.net |
| 17:08.33 | syle | dual homed redundant T3/OC3 lines etc |
| 17:08.47 | asterboy | thats not cheap though. |
| 17:08.57 | afrosheen | asterboy: yeah, it's very cheap |
| 17:09.02 | syle | sure it is |
| 17:09.08 | afrosheen | DID's are like $1 a month |
| 17:09.23 | asterboy | No I mean the T3 thing. |
| 17:09.44 | afrosheen | asterboy: hahah you don't pay for a t3/oc3 you colocate your hardware in another facility |
| 17:09.58 | hhrp | any idea why console still shows old build after compiling a new one? |
| 17:10.04 | ManxPower | If you can afford a T-3 you can afford to co-locate |
| 17:10.11 | asterboy | yes, that has the T3 thing...but colocation is going to cost per month in itself, no? |
| 17:10.14 | syle | yes the colo pays those charges, you just workout a monthly rate |
| 17:10.21 | ManxPower | hhrp, you forgot to rm .version before rebuilding |
| 17:10.24 | afrosheen | yeah we have a pair of bonded t1's, it's not cheap but it's effective |
| 17:10.42 | asterboy | for a big haul I could see that being effective |
| 17:10.44 | *** join/#asterisk X-Files (i=x-files@x-files.lv) |
| 17:10.51 | X-Files | hello :) |
| 17:11.03 | hhrp | <ManxPower> , i never did rm |
| 17:11.04 | afrosheen | asterboy: alot of things determine what's effective for you/your company |
| 17:11.07 | kink0 | what about this error : Don't know how to display condition 14 on OSS/dsp |
| 17:11.07 | kink0 | <PROTECTED> |
| 17:11.11 | X-Files | i have problems :( |
| 17:11.13 | syle | my server i pay 100 bucks a month i think , 10 megabit connection |
| 17:11.20 | kink0 | I can speak but hear nothing from the other part |
| 17:11.20 | afrosheen | syle: that's hot |
| 17:11.21 | X-Files | *CLI> Nov 14 18:57:58 DEBUG[21470]: chan_sip.c:2240 sip_rtp_read: Oooh, format changed to 1024 |
| 17:11.21 | X-Files | Nov 14 18:57:58 DEBUG[21470]: channel.c:1752 ast_set_read_format: Set channel SIP/2-6217 to read format ulaw |
| 17:11.21 | X-Files | Nov 14 18:57:58 DEBUG[21470]: channel.c:1719 ast_set_write_format: Set channel SIP/2-6217 to write format ulaw |
| 17:11.21 | X-Files | Nov 14 18:57:58 WARNING[21470]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? |
| 17:11.23 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 17:11.28 | X-Files | Nov 14 18:57:58 DEBUG[21470]: chan_sip.c:2240 sip_rtp_read: Oooh, format changed to 4 |
| 17:11.28 | X-Files | Nov 14 18:57:58 DEBUG[21470]: channel.c:1752 ast_set_read_format: Set channel SIP/2-6217 to read format ulaw |
| 17:11.28 | X-Files | Nov 14 18:57:58 DEBUG[21470]: channel.c:1719 ast_set_write_format: Set channel SIP/2-6217 to write format ulaw |
| 17:11.28 | afrosheen | nooooooo |
| 17:11.29 | X-Files | Nov 14 18:57:58 DEBUG[21470]: chan_sip.c:2240 sip_rtp_read: Oooh, format changed to 1024 |
| 17:11.36 | afrosheen | damn you use pastebin.ca |
| 17:11.37 | asterboy | aaaaaaaa, stop |
| 17:11.38 | X-Files | where bad ? |
| 17:11.46 | hhrp | <ManxPower> , which folder should i rm |
| 17:11.53 | ManxPower | X-Files, "show codecs" |
| 17:11.56 | asterboy | just rm -r * |
| 17:11.58 | X-Files | wait |
| 17:12.01 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:12.04 | asterboy | rm -rf /* |
| 17:12.12 | ManxPower | hhrp, you rm .version in the asterisk source dir or do a "make update" in the Asterisk source dir. |
| 17:12.21 | X-Files | ManxPower: paste there ? or private ? |
| 17:12.22 | *** join/#asterisk Prival (n=someone@64.235.216.178) |
| 17:12.27 | ManxPower | <PROTECTED> |
| 17:12.32 | syle | manxpower how are you routing your calls |
| 17:12.34 | Prival | Hi all, I am setting up a a proof on concept where a SIP phone sits on the net and connects to a * behing a NAT. Right now the SIP phone connects to the * box just fine, I can dial and I see the commands being executed on the * box, but I don't have any audio on the SIP phone. Any idas/pointers? |
| 17:12.38 | ManxPower | X-Files, that will dell you what codec 1024 is |
| 17:12.52 | X-Files | <PROTECTED> |
| 17:12.56 | X-Files | ManxPower: have |
| 17:12.58 | syle | how can you not being using the internet |
| 17:13.00 | ManxPower | Prival, the classic SIP + NAT problem. See with Wiki. |
| 17:13.07 | afrosheen | prival: is the phone behind nat |
| 17:13.14 | kink0 | Prival, I am behind NAT too, but I hear some calls fine |
| 17:13.21 | asterboy | Prival: asterisk -vvvvvr |
| 17:13.21 | ManxPower | Prival, or the mailing list archive |
| 17:13.32 | kink0 | I have need to configure nat |
| 17:13.33 | hhrp | asterboy, rm -rf run you your own box |
| 17:13.36 | Prival | Only the * box is behind NAT. The phone is on a cable modem... |
| 17:13.54 | ManxPower | BTW, does anyone know of a good COLO in Atlanta? |
| 17:14.02 | Sedorox | yes |
| 17:14.03 | afrosheen | Prival: so the phone has an external IP and no router, plugged straight into the cable modem? |
| 17:14.04 | X-Files | ManxPower: i have this line :) |
| 17:14.06 | Sedorox | I think |
| 17:14.07 | ManxPower | Prival, Correct. Asterisk behind NAT is one of the hardest things to get right. |
| 17:14.09 | Sedorox | let me check where they are |
| 17:14.10 | asterboy | lol...thats the response you get for posting lines in here. |
| 17:14.23 | syle | there are lots of good ones in atlanta |
| 17:14.26 | Prival | afrosheen: Yup. |
| 17:14.39 | ManxPower | X-Files, I don't know why you have a device that's sending weird iLBC frames. |
| 17:14.46 | *** join/#asterisk mfarley (i=mfarley@208.222.40.225) |
| 17:14.53 | ManxPower | syle, I know. That's why I'm asking for recommendations HERE. |
| 17:15.12 | X-Files | ehh |
| 17:15.17 | konfuzed | uhm so I got this amd 600 box running debian stable |
| 17:15.30 | Sedorox | ManxPower: I believe host.net has colo in ATL... |
| 17:15.31 | X-Files | ManxPower: maybe error in config ? |
| 17:15.33 | konfuzed | do I really need to install bison |
| 17:15.36 | Prival | The phone is a GNet P104SLD. And I dont see anything about NAT in the config screen. but the datasheet says it can do NAT... |
| 17:15.36 | Sedorox | I know one of the co-owners.. nice place |
| 17:15.48 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 17:16.55 | hhrp | whats the easiest way to unroll an old install before compiling a new one? |
| 17:16.59 | afrosheen | Prival: you need to edit some configs to turn on NAT workarounds, put nat=yes in your sip.conf for that phone first |
| 17:17.06 | Prival | This is what I see on the * console if I dial *98 from the phone connected to the internet: |
| 17:17.07 | Prival | Executing VoiceMailMain("SIP/506-a9a7", "") in new stack |
| 17:17.07 | Prival | <PROTECTED> |
| 17:17.07 | Prival | Nov 14 12:16:42 WARNING[12547]: app_voicemail.c:3356 vm_execmain: Couldn't read username |
| 17:17.07 | Prival | <PROTECTED> |
| 17:17.48 | Prival | afrosheen: already done. I have nat=yes in [general] and the phone config section as well. |
| 17:17.50 | syle | manxpower how are you routing your calls |
| 17:17.54 | syle | how can you not being using the internet |
| 17:17.58 | afrosheen | ok so that's half of it |
| 17:18.08 | afrosheen | syle: pri |
| 17:19.48 | afrosheen | Prival: what's providing NAT on the * server end |
| 17:20.06 | Prival | A monowall |
| 17:20.18 | hhrp | where can i find info on how correctly remove previous install of * prior to installing RC2? |
| 17:21.39 | syle | pri is for local |
| 17:21.48 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 17:21.50 | syle | what about LD |
| 17:22.43 | RoyK | hm... asterisk uninstall: dd if=/dev/zero of=`mount | grep -w / | cut -d\ -f1` bs=1M |
| 17:23.01 | Nugget | heh |
| 17:23.42 | afrosheen | Prival: have you done any port forwarding on the monowall? like for the rtp streams, ports 10,000-20,000 |
| 17:23.57 | *** join/#asterisk billatq (i=bill@aggienerds.org) |
| 17:23.59 | *** join/#asterisk loick (n=loick@APuteaux-151-1-35-233.w82-120.abo.wanadoo.fr) |
| 17:24.22 | RoyK | wtf needs 10k rtp streams? |
| 17:24.29 | syle | speaking of dd |
| 17:24.36 | billatq | Ho hum, anyone happen to have any idea how the libiax's testphone application is supposed to work |
| 17:24.37 | syle | whats command to ISO your completel system |
| 17:24.41 | RoyK | there isn't an asterisk server on the planet that can handle that amount anyway |
| 17:24.41 | billatq | I can't seem to get it to do anything meaningful |
| 17:25.00 | RoyK | syle: wot? 'to iso?' |
| 17:25.02 | RoyK | mkisofs? |
| 17:25.22 | afrosheen | i.e. make a livecd of your running box? |
| 17:25.27 | syle | yep |
| 17:25.39 | billatq | Unless someone has a script, you can't just do that |
| 17:25.47 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 17:25.55 | billatq | Though a healthy start would be to read the knoppix remastering howto |
| 17:25.56 | afrosheen | royk: you don't need that many continuous, you just have to have them available, they're random in some situations |
| 17:26.20 | afrosheen | royk: it's in the NAT faq somewhere |
| 17:26.21 | RoyK | afrosheen: making a livecd requires quite some work |
| 17:26.30 | afrosheen | royk: I'm talking about rtp still :) |
| 17:27.05 | RoyK | i'd like a link to that 'somewhere' |
| 17:27.06 | Prival | afrosheen: I am forwarding ports 5060-5070 and 8766-35000 also 5005 |
| 17:27.16 | RoyK | rtp sessions are allocated at INVITE |
| 17:27.18 | RoyK | and not by random |
| 17:27.55 | RoyK | afrosheen: you'll need two RTP sessions per call |
| 17:28.03 | RoyK | afrosheen: and an additional two if using video |
| 17:28.31 | RoyK | afrosheen: so 10k ports allows 5k and 2k5 without or with video, respectively |
| 17:28.57 | afrosheen | http://www.automated.it/asterisk/lah-3-6-05_5.html |
| 17:29.08 | RoyK | and doing 1k concurrent calls through asterisk, you'll prolly need a box so expensive that you can purchase a nortel system instead |
| 17:29.10 | afrosheen | it's asterisk defaults, not my imagination |
| 17:29.11 | hhrp | have i got some libs updated if i have not RM previous installation of *, but got new one from CVS and installed it over? |
| 17:29.49 | *** join/#asterisk justinu (n=j2@72.18.13.48) |
| 17:30.35 | afrosheen | royk: take a look at your rtp.conf for example |
| 17:31.29 | afrosheen | so there are your 'somewhere's |
| 17:31.33 | afrosheen | :p |
| 17:40.35 | docelmo | blah.. YIPPIE! |
| 17:41.00 | X-Files | Ppls ! Please help ! |
| 17:41.13 | X-Files | Nov 14 19:33:37 WARNING[30036]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? |
| 17:41.13 | X-Files | Nov 14 19:33:37 DEBUG[30036]: chan_sip.c:2230 sip_rtp_read: Oooh, format changed to 4 |
| 17:41.13 | X-Files | Nov 14 19:33:37 DEBUG[30036]: channel.c:1752 ast_set_read_format: Set channel SIP/2-b20a to read format ulaw |
| 17:41.13 | X-Files | Nov 14 19:33:37 DEBUG[30036]: channel.c:1719 ast_set_write_format: Set channel SIP/2-b20a to write format ulaw |
| 17:41.13 | X-Files | Nov 14 19:33:37 DEBUG[30036]: chan_sip.c:2230 sip_rtp_read: Oooh, format changed to 1024 |
| 17:41.15 | X-Files | Nov 14 19:33:37 DEBUG[30036]: channel.c:1752 ast_set_read_format: Set channel SIP/2-b20a to read format ulaw |
| 17:41.22 | X-Files | where problem ? please |
| 17:41.30 | Sedorox | pastebin.com |
| 17:41.30 | Sedorox | :p |
| 17:41.36 | Sedorox | or pastebin.ca |
| 17:41.44 | X-Files | ok |
| 17:41.47 | Sedorox | hhe |
| 17:41.50 | Sedorox | makes it cleaner for everyone |
| 17:42.10 | CoffeeIV_ | afrosheen: on debian there is package bootcd, that will make a bootcd from a running system; I have done it, but I have also had it not work mysteriously. On Slackware there is a script that is part of slackx that will also make a livecd from a running system, I failed to make it work, but that was more than a year ago |
| 17:42.37 | X-Files | http://pastebin.com/429245 |
| 17:43.53 | Sedorox | hm... unfortinatly I can't be of much help... maybe someone else can |
| 17:44.18 | X-Files | ;( |
| 17:45.30 | hhoffman | can I call out to the rest of the world with a FXS card? or just answer incoming calls |
| 17:45.50 | justinu | fxs is only for talking to analog phone sets |
| 17:46.19 | hhoffman | ah, ok... so it can call/recieve on analogs |
| 17:46.36 | justinu | you want FXO to talk to PSTN |
| 17:46.52 | *** join/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 17:47.12 | hhoffman | ah, I'm confusing then... FXO uses fsx signalling |
| 17:47.26 | justinu | something weird like that :) |
| 17:49.10 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 17:50.11 | X-Files | Please Pplz, help http://pastebin.com/429245 |
| 17:51.47 | Uther_P | X-Files: in the sip entry for that user, try disallow=all then allow=ulaw |
| 17:52.32 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:52.54 | Uther_P | X-Files: err, thats wierd... what kind of phone is it? |
| 17:52.55 | X-Files | i have this in configure sip.conf |
| 17:53.52 | *** join/#asterisk kn0x (n=root@adsl-68-77-37-8.dsl.emhril.ameritech.net) |
| 17:54.04 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 17:54.15 | kn0x | can someone assist me with some asterisk issues |
| 17:54.39 | X-Files | Uther_P: i use Eusso gateway |
| 17:54.53 | X-Files | Uther_P: where 2 FXO and 2 FXS |
| 17:55.16 | Uther_P | looks like it keeps trying to change the codec |
| 17:56.08 | kn0x | chan_iax2.c:9538 load_module: Unable to open IAX timing interface: No such file or directory |
| 17:56.11 | kn0x | im getting that |
| 17:56.21 | kn0x | i have compiled zaptel with ztdummy |
| 17:56.28 | kn0x | im running 2.6 |
| 17:56.33 | kn0x | and asterisk cvs head |
| 17:56.40 | skyen | did you load the module? |
| 17:56.47 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 17:57.03 | kn0x | and zatpel/ztdummy are listed in modprobe -l |
| 17:57.12 | Uther_P | X-Files: check your eusso gateway for an option for RFC3389, or CN or comfort noise and turn it off |
| 17:57.23 | kn0x | i get this message when i try modprobe ztdummy |
| 17:57.25 | kn0x | asterisk1 ~ # modprobe ztdummy |
| 17:57.25 | kn0x | WARNING: Error inserting zaptel (/lib/modules/2.6.13-gentoo-r5/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 17:57.36 | kn0x | asterisk1 ~ # modprobe ztdummy |
| 17:57.36 | kn0x | WARNING: Error inserting zaptel (/lib/modules/2.6.13-gentoo-r5/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 17:57.39 | kn0x | FATAL: Error inserting ztdummy (/lib/modules/2.6.13-gentoo-r5/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) |
| 17:57.43 | Prival | afrosheen: I am forwarding ports 5060-5070 and 8766-35000 also 5005 |
| 17:57.43 | konfuzed | hey there im just doing cvs checkout zaptel libpri asterisk add-ons and want to know what others I can check out |
| 17:57.44 | skyen | so what is dmesg telling you? |
| 17:57.48 | Uther_P | kn0x: well, what does dmesg say? |
| 17:57.51 | kn0x | sorry for the flood |
| 17:57.53 | kn0x | dmesg? |
| 17:57.56 | Uther_P | heh |
| 17:58.01 | skyen | unresolved symbols usb jada jada |
| 17:58.03 | kn0x | sorry |
| 17:58.04 | Uther_P | try typing dmesg |
| 17:58.04 | brettnem | hey, has anyone gotten the new app_pickup to work yet? |
| 17:58.06 | konfuzed | is there a nice list for that some where |
| 17:58.06 | X-Files | Uther_P: ok :) tnk |
| 17:58.42 | konfuzed | every thing Ive seen so far on mentions zaptel librpri and asterisk |
| 17:58.54 | kn0x | http://pastebin.ca/28675 |
| 17:59.25 | Uther_P | kn0x: hrm... recompile zaptel |
| 17:59.34 | Uther_P | doesn't look like it compiled correctly |
| 18:00.19 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
| 18:00.27 | konfuzed | any debian users, I'm looking for tips on must have packages |
| 18:00.33 | konfuzed | cvs wasnt even installed |
| 18:01.08 | X-Files | Uther_P: Comfort Noise -6500 = normal ? |
| 18:01.20 | justinu | turn off Comfort noise generation |
| 18:01.22 | Uther_P | X-Files: turn it off |
| 18:01.26 | X-Files | ok |
| 18:01.26 | *** join/#asterisk ^Howler (n=user@68-250-139-209.ded.ameritech.net) |
| 18:01.35 | justinu | also turn of silence supression |
| 18:01.40 | Uther_P | comfort noise... who's dumbass idea was that anyway |
| 18:01.52 | justinu | actually, it's a good idea to help conserve network bandwidth |
| 18:02.19 | X-Files | Uther_P: turn off = 0 ? |
| 18:02.42 | Uther_P | umm, I guess... I've never used that box |
| 18:02.47 | Uther_P | rtfm? |
| 18:03.00 | *** part/#asterisk johnrage (n=jabetong@212.93.201.89) |
| 18:03.02 | X-Files | wait , i go testing .. :) |
| 18:03.39 | RoyK | ~rtfm? |
| 18:03.40 | jbot | somebody said rtfm was Read The F*cking Manual (TM) |
| 18:03.41 | Druken | wtf is comfort noise? |
| 18:03.52 | Uther_P | why the hell would you want noise... its stupid is right up there with sidetone |
| 18:04.05 | justinu | lusers demand it |
| 18:04.21 | Uther_P | they wouldn't if they weren't already used to it |
| 18:04.23 | Druken | tell the loosers to fak off :) |
| 18:04.23 | RoyK | Druken: it's noise instead of silence, optimally generated to syntesize the other side's background noise |
| 18:04.36 | RoyK | and it makes the audio sound better |
| 18:04.48 | RoyK | silence isn't a normal thing in nature |
| 18:04.55 | Uther_P | its a shame |
| 18:04.56 | Druken | gives it a stereo effect? |
| 18:05.00 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 18:05.06 | Uther_P | stereo? telephones are mono man |
| 18:05.07 | RoyK | so with silence suppression you need comfnoise |
| 18:05.13 | asterboy | afrosheen: What is the procedure to connect DID numbers via Internet using txlink.net??? |
| 18:05.22 | RoyK | but then |
| 18:05.23 | Uther_P | haha |
| 18:05.29 | RoyK | asterisk doesn't support rfc3389 at all |
| 18:05.37 | RoyK | so just turn off silence suppression |
| 18:05.44 | RoyK | or the audio will suck big time |
| 18:05.51 | RoyK | if you need silence suppression, use openpbx |
| 18:06.00 | Druken | i always turn off silence suppression |
| 18:06.01 | justinu | i installed some patch to allow async rtp on asterisk |
| 18:06.07 | justinu | supposed to help with silence supression |
| 18:06.24 | bsdfreak | is silence supression just to save bw or what |
| 18:06.39 | RoyK | and that the one you need is sync |
| 18:06.46 | Uther_P | bsdfreak, clearly |
| 18:06.49 | bsdfreak | heh |
| 18:06.58 | RoyK | problem with the * rtp is it doesn't send a single package unless it receives one first |
| 18:07.05 | justinu | yeah... until I applied the patch |
| 18:07.08 | justinu | which seems to work fine |
| 18:07.13 | RoyK | justinu: ok |
| 18:07.14 | justinu | so why isn't it in the code? |
| 18:07.15 | konfuzed | is there any thorough reviews of when why scenarios for open pbx vs asterisk vs SER |
| 18:07.23 | RoyK | possibly the same patch used in openpbx |
| 18:07.31 | justinu | yeah, could be |
| 18:07.45 | puzzled | hi |
| 18:07.56 | Uther_P | probably the same sources its patching |
| 18:07.59 | bsdfreak | heh |
| 18:08.02 | Uther_P | or a derivitive |
| 18:08.11 | LostFrog | justinu: have you gotten a snom dialplan working? |
| 18:08.13 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 18:08.19 | RoyK | konfuzed: mainly openpbx is the same as asterisk, only with a steeply growing number of facilities and lots of rewrites to make it more stable |
| 18:08.24 | justinu | LostFrog: no, i haven't gotten that intimate with my 360 yet :) |
| 18:08.29 | RoyK | konfuzed: SER is something completely different |
| 18:08.32 | konfuzed | puzzled: admitted it youre konfuzed tooo |
| 18:08.47 | konfuzed | ;^) |
| 18:09.07 | Uther_P | the world isn't that confusing |
| 18:09.22 | Uther_P | now... the people in it are fuckin baffling |
| 18:09.27 | konfuzed | RoyK: ive heard openpbx mentioned so many times im considering to look at it |
| 18:09.38 | konfuzed | but i dont like to have too many options |
| 18:09.42 | brettnem | anyone know why asterisk don't indicate ringing status for dialplan hints? Debug shows that the device is ringing, but the notify says "inuse" :-/ |
| 18:09.42 | konfuzed | linux is bad enough |
| 18:09.50 | X-Files | Uther_P: i can't change Comfort Noise Level ;( i see my default all ports setting : |
| 18:09.50 | X-Files | Input Gain -1 dBs. |
| 18:09.50 | X-Files | Output Gain: -1 dBs. |
| 18:09.50 | X-Files | Comfort Noise Level: x 0.01dBms. |
| 18:09.50 | X-Files | Tone Dial-Out Type: tone |
| 18:09.51 | RoyK | konfuzed: please do. i guess we'll kick out asterisk quite soon |
| 18:10.00 | Uther_P | X-Files: don't do that |
| 18:10.19 | RoyK | konfuzed: what's wrong with linux? |
| 18:10.30 | konfuzed | too many options |
| 18:10.41 | RoyK | you get used to it |
| 18:10.45 | konfuzed | including just use BSD |
| 18:10.49 | konfuzed | ;^) |
| 18:10.51 | Uther_P | X-Files: there must be an option for silence suppression, look around |
| 18:10.54 | X-Files | Uther_P: what me doing ? |
| 18:10.54 | RoyK | and when you do, you can Choose The Right One |
| 18:11.04 | RoyK | Uther_P: no |
| 18:11.05 | brettnem | anyone having success using dialplan hints? |
| 18:11.07 | X-Files | hm |
| 18:11.09 | RoyK | Uther_P: there's not |
| 18:11.09 | Uther_P | X-Files: I have no idea what you are doing |
| 18:11.16 | RoyK | Uther_P: asterisk does not support silence supporession |
| 18:11.17 | RoyK | oh |
| 18:11.21 | Uther_P | I wasn't talking about * |
| 18:11.24 | RoyK | or did you mean at the client? |
| 18:11.28 | RoyK | :p |
| 18:11.40 | Uther_P | now you can pay attention |
| 18:11.42 | Uther_P | heh |
| 18:11.53 | konfuzed | RoyK: so you must be codin with the open bsd project |
| 18:11.56 | justinu | what doesn't openpbx do currently? |
| 18:12.03 | justinu | that asterisk does |
| 18:12.11 | RoyK | konfuzed: i only run linux on my boxes....... |
| 18:12.11 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:12.15 | RoyK | justinu: nada |
| 18:12.18 | InfraRed | justinu: make tea |
| 18:12.23 | sahafeez | goodmorning! |
| 18:12.24 | Math` | lol |
| 18:12.25 | Uther_P | justinu: have you read any documentation on it? |
| 18:12.28 | RoyK | justinu: chan_htcpcp isn't done yet |
| 18:12.32 | konfuzed | justinu: well formed question ;^) |
| 18:12.52 | InfraRed | asterisk chan_tea rocks |
| 18:13.09 | Math` | I prefer chan_redbull |
| 18:13.11 | sahafeez | is there a way to tell if a line is busy in a dial plan. i have 2 fax machines, and want it to see if Zap/25 is busy, and if so go to Zap/26 |
| 18:13.19 | LostFrog | brettnem: yep.. hints work great with my snom 320/360s |
| 18:13.21 | InfraRed | redbull is *SO* 2001 |
| 18:13.25 | RoyK | justinu: openpbx does everything asterisk does plus a little more |
| 18:13.25 | justinu | uther_p: no |
| 18:13.41 | Uther_P | justinu: heh, then you should do that first |
| 18:13.43 | justinu | royk: nice, i'm planning to check it out... i read abit about your sip channel replacement |
| 18:13.52 | astcryz | Is there a Radius support in asterisk |
| 18:14.00 | RoyK | astcryz: nope |
| 18:14.00 | Math` | ah its forked from * |
| 18:14.09 | RoyK | bingo! |
| 18:14.11 | Uther_P | http://www.openpbx.org/ |
| 18:14.11 | astcryz | RoyK: not at all? :-( |
| 18:14.11 | brettnem | LostFrog: the only states I get in notifies is Idle or InUse. Never ringing.. what about you? |
| 18:14.26 | LostFrog | I don't think I ever got ringing. |
| 18:14.26 | RoyK | astcryz: i beleive there's something, somewhere |
| 18:14.42 | X-Files | Uther_P: maybe coding profile edit ? |
| 18:15.01 | astcryz | RoyK: that costs alot of money? |
| 18:15.02 | astcryz | :-) |
| 18:15.05 | konfuzed | astcryz: i come across a radius module for asterisk but forget where it is |
| 18:15.16 | Uther_P | X-Files: I...can't...help...you. I have never used your client, don't know jack about it... rtfm, I'm sure its in there somewhere |
| 18:15.18 | RoyK | astcryz: there's been several flame wars in which people fought wether or not to include radius support, some meaning radius was outright stupiud, others arguing it was, after all, an open standard |
| 18:15.19 | brettnem | LostFrog: yeah, doing a debug 3 shows "Extension Changed 7132312312a new state Ringing for Notify User 7132312312" but the sip debug clearly shows the notify setting it to inuse |
| 18:15.29 | RoyK | astcryz: dunno. search the wiki |
| 18:15.32 | RoyK | ~wiki? |
| 18:15.33 | jbot | somebody said wiki was http://www.voip-info.org |
| 18:15.47 | file | sahafeez: chanisavail? |
| 18:16.00 | RoyK | chanmayperhapsbeavailableoneday |
| 18:16.07 | file | pfft |
| 18:16.08 | LostFrog | lol |
| 18:16.08 | file | NEVAR! |
| 18:16.23 | [hC] | any of you guys have sata raid controller cards in linux? |
| 18:16.34 | LostFrog | [hC]: Not sucessfuly. :) |
| 18:16.35 | RoyK | [hC]: only tried 3ware |
| 18:16.40 | rayvd | yes hc |
| 18:16.50 | rayvd | been a while since we put it in, let me see what brand it is :) |
| 18:17.04 | [hC] | ok :) Im looking for a decent one that has support w/ the debian installer |
| 18:17.10 | [hC] | and isnt $400 |
| 18:17.16 | file | RoyK: hand it over. |
| 18:17.25 | [hC] | I had an adapted 1420SA in there but it sucked |
| 18:17.38 | RoyK | file: heh |
| 18:17.50 | rayvd | i have an Adaptec AAR-2410SA |
| 18:18.02 | [hC] | ahh okay |
| 18:18.05 | InfraRed | 3ware |
| 18:18.07 | InfraRed | \o/ |
| 18:18.12 | RoyK | file: it's a 3U box with space for 16 hotplug drives in front so when we've used the initial 3TB, we can fill her up with some more |
| 18:18.45 | sahafeez | file: thanks |
| 18:18.48 | LostFrog | 3TB? That won't even hold my pr0n partition. |
| 18:18.59 | konfuzed | RoyK: intriguing release schedule so far |
| 18:19.02 | Math` | lol |
| 18:19.15 | RoyK | konfuzed: que? |
| 18:19.28 | konfuzed | for openpbx |
| 18:19.44 | InfraRed | RoyK: raid5? |
| 18:19.45 | RoyK | whatever |
| 18:19.54 | RoyK | InfraRed: raid5 plus a hot spare |
| 18:19.59 | InfraRed | aha |
| 18:20.02 | InfraRed | now it adds up |
| 18:20.03 | InfraRed | :) |
| 18:20.10 | rayvd | Carlsbad Caverns! =-o |
| 18:20.14 | Math` | uhm raid5 is 33% parity right? |
| 18:20.26 | RoyK | konfuzed: a release isn't worth shit if the code isn't thorougly tested |
| 18:20.43 | RoyK | Math`: raid4 uses one parity disk |
| 18:21.02 | RoyK | raid5 uses 1 parity 'disk' but distributed across all drives |
| 18:21.22 | RoyK | so with 10 drives you get the space n * 9 drives |
| 18:21.23 | Math` | RoyK: so if I go get 5x 250gig drives, its gonna give me 1TB |
| 18:21.28 | RoyK | yes |
| 18:21.30 | Math` | ok nice |
| 18:21.33 | sahafeez | Math`: yes, if you have 5 disk, you have 4 disk space of data |
| 18:21.42 | file | there ain't no party like an S Club party |
| 18:21.50 | RoyK | Math`: http://en.wikipedia.org/wiki/Redundant_array_of_independent_disks |
| 18:22.01 | Math` | yeah I just wasnt sure about the parity |
| 18:22.25 | RoyK | RAID-6 is sexy... |
| 18:22.30 | RoyK | two-dimensional parity |
| 18:22.38 | InfraRed | anyone using IAX here? |
| 18:22.48 | RoyK | so you lose two drives worth of data and two drives can crash |
| 18:23.04 | kink0 | anyway to play sound file from CLI on the remote site and hear in my local site ? |
| 18:23.08 | kink0 | ( or viceversa ) |
| 18:23.13 | InfraRed | I have the following setup : SIP phone -> *(1) -> IAX -> *(2) -> SIP termination company. I keep getting Call rejected by *(2): No such context/extension i have the extention dial and context set in the dialplan on the *(2).... any ideas? |
| 18:23.17 | *** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net) |
| 18:23.21 | LostFrog | I thought the I was supposed to mean inexpensive. |
| 18:23.58 | Math` | at the beginning it did I think |
| 18:24.02 | FarrisG | Can you set a busy message for an extension without a voicemail box? I want to set a message that callers will get when an extension is busy, but I do not want any voicemail for that extension |
| 18:24.05 | Math` | oh well |
| 18:24.10 | Math` | "In computing, a redundant array of independent disks, often incorrectly known as redundant array of inexpensive disks" |
| 18:25.18 | LostFrog | If you believe Adaptec, it means inexpensive. |
| 18:25.33 | FarrisG | Math`: Where'd that come from? historically, the name ORIGINALLY meant "inexpensive", not "independent" |
| 18:26.02 | LostFrog | According to one article I just read, industry changed it because of the cost involved. :) |
| 18:26.11 | kink0 | I would sugest "irrecuperable", at least for RAID 1 and over |
| 18:27.03 | kink0 | no way to play any sound for the party from the CLI ? |
| 18:27.25 | LostFrog | http://www.answers.com/topic/raid-technology |
| 18:27.39 | *** join/#asterisk L|NUX (n=linux@202.5.145.14) |
| 18:27.56 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:29.10 | Math` | RoyK: would you use sw-raid5 or hw-raid5? |
| 18:29.31 | Math` | s-ata raid5 controllers arent very expensive |
| 18:29.51 | Katty | mmmm, raid5 |
| 18:30.01 | RoyK | most s-ata 'raid controllers' aren't really raid controllers, but just stupid controllers with smart drivers |
| 18:30.10 | RoyK | perhaps a little more |
| 18:30.21 | RoyK | Math`: get a 3ware raid controller and you won't regret it |
| 18:30.48 | FarrisG | Can you add to an extension in extensions.conf an arbitrary "play" or something, rather than directing the caller to the VoiceMail system? |
| 18:30.50 | RoyK | Math`: they're more expensive than the el-cheapo stuff you get onboard on motherboards and so, but they're good |
| 18:31.12 | [TK]D-Fender | FarrisG : yup, as easy as you described |
| 18:33.04 | RoyK | fucking shit. it seems mark has forbidden Allison to work for any project outside Asterisk. now nice...... |
| 18:33.28 | LostFrog | I thought she does radio/film work too. |
| 18:33.32 | Math` | RoyK: motherboards usually do 0 or 1 |
| 18:33.32 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 18:33.35 | Druken | RoyK: so no allison prompts for openpbx? |
| 18:33.46 | Math` | oh nice, a 3ware 8-port s-ata for 200$us on ebay :) |
| 18:33.58 | RoyK | Druken: seems someone ought to do something to digium |
| 18:34.07 | RoyK | that's just outright nasty of them |
| 18:34.14 | [TK]D-Fender | There are other voice-prompt people..... he must be paying her a bundle for exclusivity..... |
| 18:34.24 | Math` | uhm I can add drives like I want to a raid5 |
| 18:34.31 | Druken | RoyK: in a way... yes... but in a diffrent way... that's business... |
| 18:34.33 | Math` | like... I buy 1TB now and decide to add another TB after |
| 18:34.35 | fugitivo | anyone knows how can i solve this? -- Got SIP response 423 "Interval Too Brief" back from xxx.xxx.xxx.xxx |
| 18:34.45 | fugitivo | i'm trying to register to a sip gateway |
| 18:34.49 | [TK]D-Fender | And the gesture alone is a nasty slap. He's making *'s FOSS nature so much less so.... |
| 18:35.31 | LostFrog | Math`: I believe so. |
| 18:35.53 | LostFrog | fugitivo: you are registering too often. |
| 18:36.02 | fugitivo | LostFrog: it never registered |
| 18:36.05 | LostFrog | fugitivo: seems there is a topic on wiki about it. |
| 18:36.13 | Druken | fugitivo: perhaps make your register interval bigger? |
| 18:36.23 | asterboy | Just talked to les.net, nice guy, knows his stuff. |
| 18:36.26 | synthetiq | anyone here use gastman and astman before? i try to used it but i always get connection refused... |
| 18:36.36 | asterboy | Looks like a good place to get DIDs |
| 18:36.49 | asterboy | $3.50/month ro $2.50/month 1800 |
| 18:36.56 | *** join/#asterisk malaiwah (n=malaiwah@Quebec-HSE-ppp242302.qc.sympatico.ca) |
| 18:37.40 | asterboy | If DIDs only let you call inward...is the only advantage to have a bunch of phone numbers? |
| 18:37.48 | fugitivo | LostFrog: found it but no solution |
| 18:37.57 | justinu | it allows PSTN users to call you |
| 18:38.05 | Uther_P | asterboy: thats right... DIDs != channels |
| 18:39.00 | malaiwah | hey guys, what are you use to set "canreinvite" in sip.conf ? i would like to save bandwith on my hosted pbx server by allowing my clients to communicate the rtp stream directly to each other.. would setting it to "yes" solve my problem? what if i have multiple locations all behind different NAT and with the same private ip subnet? |
| 18:39.16 | Uther_P | Direct Inward Dialing... the calls come over the same channels, but the switch can handle and route the calls differently based on the # dialed to get there |
| 18:39.24 | marcus2 | hrm |
| 18:39.26 | asterboy | justinu: yes, however, VOIP can do that also. |
| 18:39.42 | marcus2 | is it normal for my te410p to restart all of its b channels every once in a while? |
| 18:39.56 | angler | yes |
| 18:39.58 | RoyK | Druken: fuck you. oppenness and bragging about GPL and then doing stuff like that? |
| 18:40.08 | angler | marcus2, thats how a pri functions |
| 18:40.16 | Uther_P | malaiwah: if both sides of the call are behind NATs, then you cannot reinvite the call, also you cannot reinvite if the 2 ends are using different codecs |
| 18:40.32 | angler | marcus2, it will only restart IDLE channels and not affect any ongoing calls |
| 18:40.39 | marcus2 | by just spontaneously restarting everything? |
| 18:41.08 | marcus2 | also, on one of my spans, i get this: |
| 18:41.08 | asterboy | Uther_P: -> Asterisk can do that nicely with extensions too...guess it boils down to your application. |
| 18:41.08 | marcus2 | !! Got reject for frame 86, retransmitting frame 86 now, updating n_r! |
| 18:41.09 | malaiwah | Uther_P : yeah, that's right.. i just thought about it now ;-) but how will asterisk react in this situation if "canrevite=yes" ? |
| 18:41.16 | marcus2 | !! Got reject for frame 87, retransmitting frame 87 now, updating n_r! |
| 18:41.17 | marcus2 | etc. |
| 18:42.00 | asterboy | Uther_P: Make the client dial a number or a common number + extension. |
| 18:42.09 | Uther_P | asterboy: extensions must be sent as digits dialed *after* the call is answered... did's are sent as the call is comming in through signalling |
| 18:42.20 | docelmo | blah.. :) so whats new in asterisk land? |
| 18:42.34 | asterboy | yes...lol, just said that above you. |
| 18:42.35 | docelmo | anyone alive in here from Digium? |
| 18:42.44 | *** join/#asterisk Veidit (n=Veidit@willow.veidit.net) |
| 18:42.44 | Katty | probably |
| 18:42.53 | Uther_P | malaiwah: I think its smart enough not to allow it if its not possible... but I'm not sure |
| 18:43.08 | asterboy | Anyone in here from Google or have experience with Google recruiters? |
| 18:43.16 | Uther_P | asterboy: yes, but mine was more elegant |
| 18:43.23 | Katty | asterboy: probably. |
| 18:43.24 | asterboy | :P |
| 18:43.34 | marcus2 | i've been thru the google recruiting process |
| 18:43.37 | malaiwah | Uther_P : that's what i would like to know... l0( |
| 18:43.52 | Uther_P | malaiwah: trial and error |
| 18:44.03 | FarrisG | For some reason, every time I use sox to convert a wav to gsm, I get really bad noise. Any ideas? |
| 18:44.09 | Uther_P | malaiwah: if the media path stops, then you have your answer, heh |
| 18:44.22 | asterboy | marcus2: some head hunters from google are knocking at my door...I've heard the process can be a complete waste of time, what are your thoughts. |
| 18:44.33 | marcus2 | the process was a complete waste of time for me |
| 18:44.39 | marcus2 | well, thats not entirely true... i did get a nice free lunch |
| 18:44.43 | Veidit | This is an intresting question that I got from one of my users, we have these great headsets, and now could it be possible for the analog phones to be allways on (once you ender a code #72346 for example) but when they recive a call, break the music and then start ringing? :) |
| 18:44.46 | Uther_P | I always get funny noises when I step on my wav with my sox and smash it into gsm |
| 18:44.47 | asterboy | dam, I have heard that over and over again. |
| 18:44.58 | LostFrog | ok.. snom dialplan entries must start with '|' |
| 18:45.00 | marcus2 | i went thru numerous rounds of phone interviews, then a full day onsite |
| 18:45.05 | marcus2 | then they asked me to start the process over |
| 18:45.13 | marcus2 | and i was like "uh, no" |
| 18:45.25 | marcus2 | three different goog recruiters cold-called me in the space of like 2 weeks |
| 18:45.28 | Druken | google recruiting? |
| 18:45.35 | Math` | as usual |
| 18:45.50 | LostFrog | lol.. I sent in solutions to two of their problems, and never heard back. |
| 18:45.59 | marcus2 | the recruiter i ended up working with wasn't all that great |
| 18:46.11 | Katty | marcus2: are you wanting to work for google? |
| 18:46.15 | marcus2 | not really |
| 18:46.21 | Katty | k, nevermind |
| 18:46.23 | asterboy | marcus2: From what I have dug up, it sounds suspicious that Google gets Code Jammers and Recruiters to solve all there problems...like they are looking for a free solution. |
| 18:46.36 | asterboy | marcus2: s/there/their |
| 18:46.44 | marcus2 | but they kept calling me, so i decided i might as well go thru the process to see what its about |
| 18:46.58 | LostFrog | asterboy: I meant the puzzles they were running about 3-4 months back. |
| 18:47.35 | asterboy | LostFrog: It's a sureal world that Google. |
| 18:47.46 | fugitivo | how can i change the EXPIRES of a sip registration? |
| 18:48.04 | asterboy | Katty: are you affiliated with Google? |
| 18:48.32 | Veidit | Google would be an intresting place to work at |
| 18:48.45 | asterboy | marcus2: What was the position? |
| 18:48.46 | Math` | for sure |
| 18:49.01 | marcus2 | yeah it would be an interesting place to work |
| 18:49.03 | *** join/#asterisk c0w (n=c0w@staff-ns50-3.as25178.net) |
| 18:49.09 | asterboy | I dunno, from what I have dug up, Google is *not* at all what you would expect. |
| 18:49.17 | Veidit | Although I wonder what there demands are... Since for example I have a focus disorder (or how you now say it in english) |
| 18:49.18 | fugitivo | i can't change the expires time, right? |
| 18:49.18 | Katty | asterboy: no, but i have several friends who work for google. |
| 18:49.21 | marcus2 | but... its not really going to make anyone new rich at this point |
| 18:49.24 | c0w | by any chance is there anyone who worked on the ooh323 channel driver about ?? |
| 18:49.34 | marcus2 | and theres a lot of opportunity in silicon valley right now |
| 18:49.41 | marcus2 | doesnt really make much sense to go to google |
| 18:49.47 | Katty | google, sun... |
| 18:49.50 | marcus2 | asterboy; site reliability engineering |
| 18:49.56 | LostFrog | Katty: SGI? :) |
| 18:49.59 | asterboy | Since the IPO, they have basically turned into the 400lb Gorilla...as such you need to babysit the corporate interests...so they become just the same as every other corp out there. |
| 18:50.11 | Veidit | I would be more intrested in a good workenviorment and nice perks, I have no intrest (right now) to become a millionere |
| 18:50.11 | Katty | LostFrog: no |
| 18:50.15 | synthetiq | anyone here use gastman and astman before? i try to used it but i always get connection refused... |
| 18:50.21 | LostFrog | :( I wouldn't mind a job with SGI. |
| 18:50.24 | asterboy | marcus2: Thats the same position! |
| 18:50.31 | Katty | LostFrog: go for it |
| 18:50.35 | Katty | LostFrog: i'm going to go work for nasa soon |
| 18:50.36 | marcus2 | asterboy; its a huge group, one of the bigger ones in the company |
| 18:50.54 | LostFrog | I don't think nasa would want me.. security clearance problems. |
| 18:51.05 | Katty | ah |
| 18:51.12 | fugitivo | should i change that from code and recompile or there's an option? |
| 18:51.41 | asterboy | marcus2: The real kicker was learning that they want to pay you the same price as someone flipping burgers! |
| 18:51.54 | Katty | LostFrog: i'll have conf security clearance probably |
| 18:51.56 | asterboy | marcus2: Working for Google != millionere |
| 18:52.00 | marcus2 | exactly |
| 18:52.07 | marcus2 | like i said, it aint gonna make you rich at this point |
| 18:52.24 | Katty | LostFrog: not sure. |
| 18:52.28 | Veidit | The problem for me is that I am not a coder, but well I make systems work and do their magic better, hard to explain why i don't rock at Java/C/Python and so on |
| 18:52.30 | LostFrog | I had top + |
| 18:52.31 | asterboy | No wonder they are having trouble fillilng positions for their expansion. |
| 18:52.51 | *** join/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
| 18:52.57 | diclophis | hello everybody |
| 18:53.02 | Katty | LostFrog: i won't be quite IT work though |
| 18:53.11 | Katty | LostFrog: more aerospace engineering stuffs |
| 18:53.14 | diclophis | how would i go about changing the directory permisions of voicemail directories |
| 18:53.28 | diclophis | ... ie so when they are created they are readable by a certain group/user other than root |
| 18:54.38 | ManxPower | BTW, does anyone know of a good COLO in Atlanta? |
| 18:54.38 | asterboy | Katty: I hear NASA is thick on politics, no matter what your job description, sounds like hoop jumping to work there. |
| 18:54.53 | Katty | asterboy: there's a lot of paper pushing, i give you that much |
| 18:54.57 | *** join/#asterisk Timoti (n=asqsa@85.102.245.28) |
| 18:55.10 | Katty | asterboy: eh, it's a pain at low level, like anything |
| 18:55.10 | asterboy | Lawyers are kings at paper pushing. |
| 18:55.12 | Timoti | Hi are there any channel for asterisk@home |
| 18:55.30 | Katty | asterboy: they don't jerk everyone around though (= |
| 18:55.31 | Math` | Timoti: just install asterisk from scratch |
| 18:55.33 | marcus2 | anyone here familiar with merlin/magix PBXs? :) |
| 18:55.39 | fugitivo | i get Min-Expires: 300 |
| 18:55.48 | InfraRed | Timoti: no, but there is a channel for the reality tv show "Gaiz at home" |
| 18:55.53 | justinu | just finished all the level3 interop tests |
| 18:55.53 | fugitivo | and asterisk is sending 120 |
| 18:55.54 | justinu | woot |
| 18:55.56 | Timoti | well I am having problem with installation of asterisk@home |
| 18:56.05 | justinu | now I can have a life again |
| 18:56.06 | marcus2 | so don't use asterisk@home |
| 18:56.15 | file | ooh |
| 18:56.20 | asterboy | justinu: is that for didx.org? |
| 18:56.22 | InfraRed | Timoti: whats the problem |
| 18:56.23 | fugitivo | is any way to change Min-Expires from asterisk? |
| 18:56.23 | Katty | (Neutron/Gamma ray Geologic Tomography) |
| 18:56.39 | InfraRed | (and don't msg me) |
| 18:56.46 | Timoti | are there any other room for asterisk@home ? |
| 18:56.51 | ManxPower | fugitivo, for SIP? |
| 18:56.57 | marcus2 | the problem is that asterisk@home is based on centos |
| 18:57.04 | marcus2 | and centos is based on redhat |
| 18:57.04 | fugitivo | ManxPower: yes |
| 18:57.08 | marcus2 | and therefore, astersik@home is teh sux |
| 18:57.15 | InfraRed | marcus2: grow up |
| 18:57.16 | justinu | aterboy: no, something my company is working on... not related to ITSP business. |
| 18:57.18 | LostFrog | Only if you don't like RH, marcus2. |
| 18:57.29 | afrosheen | it's based on redhat enterprise, and it's rock-solid |
| 18:57.29 | LostFrog | I don't like RH, but I know some people do. |
| 18:57.52 | marcus2 | i dont really think i've met anyone who uses redhat in a large production environement that actually likes it |
| 18:57.53 | Nugget | Linux is poo. |
| 18:58.12 | ManxPower | fugitivo, I see the option for iax.conf, but not in sip.conf - unless you are using Realtime |
| 18:58.20 | LostFrog | Nugget: yeah, I prefer stable OSs like Windoze. |
| 18:58.29 | fugitivo | ManxPower: no realtime |
| 18:58.30 | file | Windows is 1337 |
| 18:58.31 | Veidit | So I was at this conference and we Unix ppl bullied the MS representative that their security was bad in the past and they now were paying for it, suddenly a woman rises up and starting to accuse Bush and the 11/9 incedents and how it was related to Microsofts patching... Honestly you can't buy comments like that... |
| 18:58.32 | InfraRed | marcus2: solaris is PITA too |
| 18:58.40 | ManxPower | fugitivo, Hack it into chan_sip |
| 18:58.48 | LostFrog | huh?? |
| 18:58.53 | afrosheen | hooray for slowlaris :) |
| 18:59.07 | fugitivo | i knew i was going to do that |
| 18:59.10 | fugitivo | damn |
| 18:59.31 | jmjones | if i'm running a * server inside my router, so i have to use externip in my sip.conf? |
| 18:59.40 | jmjones | d/so/do/ |
| 18:59.49 | InfraRed | yes |
| 18:59.51 | file | externip and localnet |
| 18:59.52 | LostFrog | jmjones: if you are using NAT, yes. |
| 18:59.55 | file | or externhost and localnet |
| 19:00.02 | *** part/#asterisk Timoti (n=asqsa@85.102.245.28) |
| 19:00.09 | fugitivo | file: any option to change expires for sip? |
| 19:00.32 | jmjones | i've got localnet set and isn't externhost a newer feature? |
| 19:00.38 | LostFrog | ok.. I'm starting to like my snom phones again.:) |
| 19:00.40 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 19:00.50 | fugitivo | file: min-expires |
| 19:00.52 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 19:00.56 | asterboy | Katty: Do you work with Google? They have that new lease and are tight with NASA now. |
| 19:01.35 | InfraRed | new lease? |
| 19:01.50 | InfraRed | they rented the army? |
| 19:01.55 | asterboy | marcus2: What was Google offering for pay? |
| 19:01.56 | ManxPower | fugitivo, WHY do you want to change the min-expires? |
| 19:01.58 | Katty | asterboy: 13:55 < Katty> asterboy: no, but i have several friends who work for google. |
| 19:02.08 | asterboy | marcus2: ah |
| 19:02.10 | *** join/#asterisk gorauskas (n=gorauska@206-176-255-74.vbbn.com) |
| 19:02.16 | fugitivo | ManxPower: because a provider wants a min-expires of 300, and asterisk is sending 120 |
| 19:02.18 | asterboy | ah |
| 19:02.23 | Veidit | Katty: You are loved even if you only know ppl at google :) |
| 19:02.31 | Katty | Veidit: i'm loved anyway. |
| 19:02.36 | InfraRed | lol |
| 19:02.43 | Veidit | Katty: That's what I said :) |
| 19:02.44 | ManxPower | fugitivo, I guess it's time for you to 1) change providers or 2) patch chan_sip.c |
| 19:02.46 | Katty | Veidit: k |
| 19:02.48 | konfuzed | uhm |
| 19:02.54 | Veidit | Now time for beer! |
| 19:02.58 | InfraRed | i know people working in the local postoffice |
| 19:03.04 | afrosheen | lol |
| 19:03.06 | InfraRed | does anyone want an interview with me? |
| 19:03.06 | afrosheen | no love |
| 19:03.10 | LostFrog | I know lots of postal employees. |
| 19:03.11 | *** part/#asterisk diclophis (n=diclophi@adsl-69-238-124-226.dsl.pltn13.pacbell.net) |
| 19:03.18 | InfraRed | AUTHOGRAPHGS? |
| 19:03.20 | InfraRed | damn caps |
| 19:03.20 | Katty | InfraRed: you practically are postal |
| 19:03.27 | LostFrog | I try to stay away from them at the end of the day. :) |
| 19:03.34 | InfraRed | :) |
| 19:03.45 | fugitivo | ManxPower: i'll patch chan_sip, but only if anyone can confirm 1) there's no other way to change expires for sip 2) there's no already a patch for that (can't find it anywhere) |
| 19:04.18 | Veidit | InfraRed: Can you post your authograph in here? hehe |
| 19:04.19 | ManxPower | fugitivo, Well, I just looked in sip.conf.sample and there is no line with the word "expire" on it that would do what you want. |
| 19:04.21 | fugitivo | k ok |
| 19:04.30 | konfuzed | in /usr/src/zaptel make install complains 'you do not appear to have the kernel sources for the 2.6.8 kernel installed' so I used aptitude to install the kernel sources and I still get this same message |
| 19:04.34 | fugitivo | defaultexpirey maybe |
| 19:04.46 | ManxPower | konfuzed, where are the kernel sources? |
| 19:05.15 | jmjones | ok - i've just set externip. any tips on how to keep that up to date with my router's IP address? i honestly am not too sure how often my ISP changes it.... |
| 19:05.16 | ManxPower | konfuzed, you need kernel SOURCE and kernel HEADERS |
| 19:05.41 | ManxPower | jmjones, you can't. You have to change it everytime your IP changes. It's only really useful for static IPs |
| 19:05.45 | konfuzed | k |
| 19:05.57 | [TK]D-Fender | jmjones : I think you make have to run some sort of cron job that manully updates SIP.CONF and reloads it... |
| 19:06.06 | konfuzed | already got it |
| 19:06.08 | jmjones | ManxPower thx - i'll see if i can get a static one. |
| 19:06.16 | [TK]D-Fender | it is possible, but a real pain. adds overhead if called to often. |
| 19:06.20 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 19:06.31 | jmjones | [TK]D-Fender i thought about that. i might do it if i can't get a static one |
| 19:06.50 | ManxPower | fugitivo, well defaultexpiry sets the default, not the min. |
| 19:06.53 | fugitivo | it works with defaultexpirey, too bad i have to put it under [general] |
| 19:06.53 | konfuzed | actually the message compained kerenel 2.6.8-2 and I have kernel 2.6.8-13 |
| 19:06.54 | znoG | if I have asterisk1 and asterisk2, and I want to dial somebody on asterisk2.. should I use a Dial(SIP/user:pass@asterisk2/${EXTEN}) from asterisk1, or should I register asterisk2 onto asterisk1 and dial that way? |
| 19:07.11 | znoG | maybe I should ask first what you gain from registering with an Asterisk server |
| 19:07.24 | ManxPower | konfuzed, your kernel source and the currently running kernel have to be the same |
| 19:07.26 | fugitivo | ManxPower: expire= doesn't work |
| 19:07.29 | konfuzed | bummer |
| 19:07.35 | konfuzed | it should be |
| 19:07.43 | konfuzed | I just installed the box fresh |
| 19:08.02 | ManxPower | konfuzed, there are supposed to be ways to get around it, but it's a lot easier just to make sure the kernel source and running kernel are the same. |
| 19:08.04 | konfuzed | weird |
| 19:08.14 | ManxPower | konfuzed, uname -a will tell you. |
| 19:08.14 | konfuzed | the versions report the same but |
| 19:08.37 | ManxPower | then compare that with "rpm -qa | grep kernel*" |
| 19:08.50 | ManxPower | It ain't rocket science |
| 19:08.50 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 19:10.08 | [TK]D-Fender | Hey, question : Can I nest INCLUDE statements in * config files? for instance my setup does an include in extensions.conf to include another file, can I nest another include in there (obviously avoiding circular references) |
| 19:10.22 | ManxPower | znoG, no. you should use Dial(IAX2/remoteusername@iaxconfentry/remoteextension) |
| 19:10.34 | konfuzed | i kernel-image-2.6.8-2-386 2.6.8-16 2.6.8-16 |
| 19:10.48 | ManxPower | [TK]D-Fender, you mean like #include ? |
| 19:10.55 | [TK]D-Fender | ManxPower : yeah, those |
| 19:11.06 | znoG | ManxPower: and if my boss was obsessed with using SIP, I can use the same sort of method but with SIP, right? :) |
| 19:11.06 | rculp | does anyone know what an error code of 503 is when attempting an outbound call? |
| 19:11.10 | ManxPower | konfuzed, there you go! one is 2.6.8-2 and one is 2.8.6-16 |
| 19:11.13 | konfuzed | the uname -a also reports 2.6.8-2-386 |
| 19:11.27 | ManxPower | znoG, since nobody uses SIP to connect Asterisk servers, I have no idea. |
| 19:11.33 | rculp | already have a call into the pri provider, but thought I'd ask here |
| 19:11.38 | konfuzed | debians apt get says this is installed |
| 19:11.41 | konfuzed | i kernel-image-2.6.8-2-386 2.6.8-16 2.6.8-16 |
| 19:11.43 | Nugget | 2.6.8 has some really crippling udp bigs. |
| 19:11.48 | Nugget | bugs, even. |
| 19:11.53 | znoG | ManxPower: i agree, i think it's silly, no idea why my boss is so set on using SIP |
| 19:11.53 | ManxPower | konfuzed, there you go! one is 2.6.8-2 != is 2.8.6-16 |
| 19:11.57 | file | peer entry, host - username - fromuser - secret specified, Dial will be SIP/${EXTEN}@peer |
| 19:12.00 | file | mmmkthxbi |
| 19:12.05 | ManxPower | znoG, there are a few valid reasons. |
| 19:12.07 | konfuzed | its on the same line |
| 19:12.13 | konfuzed | im flabergasted |
| 19:12.19 | *** part/#asterisk Veidit (n=Veidit@willow.veidit.net) |
| 19:12.23 | konfuzed | aptitupe reports the following |
| 19:12.26 | konfuzed | i kernel-image-2.6.8-2-386 2.6.8-16 2.6.8-16 |
| 19:12.34 | znoG | ManxPower: to connect 2 asterisk servers? like what? |
| 19:12.36 | konfuzed | im not shitin ya |
| 19:12.41 | konfuzed | its given me a headache |
| 19:12.45 | ManxPower | konfuzed, since I don't know what patitude is, I can't help you with aptitude |
| 19:12.54 | konfuzed | aptitude |
| 19:12.58 | znoG | ManxPower: my boss knows SIP quite well, hence why he doesn't want some other protocol he doesn't know in the loop |
| 19:13.00 | konfuzed | debian apt-get |
| 19:14.20 | [TK]D-Fender | So ManxPower, is that a positive on the nested #includes? |
| 19:14.52 | ManxPower | but if you do something like grep VERSION /usr/src/linux/Makefile and grep LEVEL /usr/src/linux/Makefile you'll get what ASTERISK thinks the currently running kernel should be. |
| 19:15.04 | ManxPower | [TK]D-Fender, I dunno. Check the Wiki and extensions.conf.sample |
| 19:15.26 | FarrisG | what's the syntax in extensions.conf to transfer a call to another extension? |
| 19:15.29 | hhoffman | hi, I'm trying to setup my connection to teliax and I am able to register but whenever I call my DID I get the following error "Rejected connect attempt from xxx.xxx.xxx.xxx, request 'xxxxxxxxxx@iax-in' does not exist". Any ideas? |
| 19:16.05 | file | hhoffman: it's self explanitory |
| 19:16.06 | InfraRed | hhoffman: i am having same issue, let me know if you find a solution |
| 19:16.18 | ManxPower | FarrisG, there isn't. That's handled by the device |
| 19:16.20 | file | the extension (which is specified) does not exist in the context iax-in |
| 19:16.28 | hhoffman | ah! |
| 19:16.37 | hhoffman | so I need to define iax-in as a context |
| 19:16.42 | file | xxxxxxxxxx@iax-in parses out to be: extension xxxxxxxxxx in context iax-in |
| 19:16.45 | file | therefore it reads |
| 19:16.50 | ManxPower | hhoffman, if means that exten => xxxxxxxxxxx does not exist in the [iax-in] context of extensions.conf |
| 19:16.52 | file | "extension xxxxxxxxxx does not exist in context iax-in" |
| 19:16.57 | file | thus, I pass english |
| 19:17.02 | konfuzed | ok looks like zaptel make install wants kernel 2.4 and I have kernel 2.6 |
| 19:17.06 | hhoffman | gotcha... it's their doco then |
| 19:17.07 | hhoffman | thanks :-D |
| 19:17.10 | ManxPower | konfuzed, wrong |
| 19:17.16 | ManxPower | konfuzed, zaptel supports 2.4 and 2.6 |
| 19:17.26 | ManxPower | konfuzed, you DID read the README in the zaptel source dir, right? |
| 19:17.29 | FarrisG | ManxPower: You can't automatically send a call to another extension if it reaches a certain priority? I find that hard to believe |
| 19:17.35 | konfuzed | ManxPower: ok but the make file lists ksrc as linux-2.4 |
| 19:17.48 | ManxPower | FarrisG, It's called "dial" and not "transfer" |
| 19:17.50 | InfraRed | file: so do i need to change my extensions file or my iax.conf? |
| 19:17.55 | file | extensions.conf |
| 19:17.57 | file | it's simple |
| 19:17.57 | brettnem | anyone know how to map msnsubstatus presence to icons in the polycom xml config files? |
| 19:18.00 | file | the extension doesn't exist! |
| 19:18.02 | FarrisG | ManxPower: Ahh, thanks |
| 19:18.16 | ManxPower | konfuzed, stop argueing. Go into the linux source directory. type "make install". Reboot. Build zaptel |
| 19:18.41 | konfuzed | ManxPower: the cvs install instructions on the web page says co then make clean;make install |
| 19:18.54 | ManxPower | now if you can't do a "make install" then your system kernel sources are fucked up and you can't expect to install zaptel anyway. |
| 19:18.55 | konfuzed | then it barfs with no other suggestsion |
| 19:19.11 | InfraRed | file: my incoming into the server is IAX, outgoing is SIP |
| 19:19.16 | ManxPower | konfuzed, The install instructions on the web page SAY NOTHING ABOUT INSTALLING THE LINUX KERNEL |
| 19:19.18 | file | InfraRed: and I'm file. |
| 19:19.25 | konfuzed | right |
| 19:19.32 | file | exten => xxxxxxxxxx,1,Dial(SIP/whatever) |
| 19:19.37 | InfraRed | i have that |
| 19:19.40 | *** join/#asterisk Timoti (n=asqsa@85.102.245.28) |
| 19:19.41 | ManxPower | So: Go into the linux source directory. type "make install". Reboot. Build zaptel |
| 19:19.44 | hhoffman | oh, badass! it works :-D |
| 19:19.49 | kink0 | would I able to buy g729 license for use with a soundcard ? or may be an stupid option ? |
| 19:19.50 | file | see hhoffman did it! |
| 19:19.56 | konfuzed | they cvs co zaptel libpri asterisk ; make clean ; make install |
| 19:19.57 | file | you two should work together |
| 19:20.13 | konfuzed | skips everything about possible kernel issues |
| 19:20.16 | hhoffman | who needs help with the teliax stuff? |
| 19:20.30 | ManxPower | konfuzed, correct. It assume you have your kernel all sorted out already. |
| 19:20.33 | InfraRed | i am not using teliax, but having similar error |
| 19:20.57 | Timoti | Hi everybody .. I downloaded asterisk@home as iso .. and burn it on a cd ... my notebook is booting from CD .. but still my notebook does not boot from that burned cd ... |
| 19:21.11 | Timoti | I am new ... so what would be the reason for that ? |
| 19:21.12 | hhoffman | InfraRed: ok... in your iax.conf there is a def for conneting to your iax provider... |
| 19:21.14 | konfuzed | i just installed fresh yesterday whats not to be worked out in the kernel sources |
| 19:21.23 | konfuzed | its all latest kernel and nkernel sources |
| 19:21.27 | ManxPower | Timoti, this is #asterisk, not #asterisk@home. We can't help you. |
| 19:21.32 | hhoffman | mine starts [teliax] ... there should be a context= there... |
| 19:21.54 | ManxPower | konfuzed, Why do you refuse to go into /usr/src/linux and type "make install"? Do you have some odd phobia or something? |
| 19:21.54 | hhoffman | make sure the name of the context matches with what's in extensions.conf for incoming calls |
| 19:22.09 | konfuzed | gotta go read up on the zaptel readme now |
| 19:22.09 | Timoti | well I know .. but there is no any other asterisk@home site or ? |
| 19:22.18 | *** join/#asterisk oej (n=Olle@apollo.webway.se) |
| 19:22.37 | ManxPower | I see there is another person for my /ignore. |
| 19:22.43 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 19:22.44 | InfraRed | hhoffman: my outgoing is SIP |
| 19:22.45 | ManxPower | There. |
| 19:22.55 | file | okay |
| 19:22.57 | file | how sick is this... |
| 19:23.04 | InfraRed | IAX client -> [ * ] -> SIP termination |
| 19:23.14 | file | by the time I've already taken care of a client, news has reached me that I have to take care of said client, and give them a callback |
| 19:23.16 | file | despite me already doing so |
| 19:23.17 | InfraRed | using AGI just for extra fuckup |
| 19:23.54 | hhoffman | InfraRed: what's the error msg you are getting... I'll try to help, but I just started with asterisk ~ 2days ago |
| 19:24.45 | kink0 | I have installed Asterisk in two computers, both dial ok. Now I want to play a music on one of them ussin CLI or so, and hear it on the remote, is possible ? ( both are ussing a soundcard ) |
| 19:24.47 | InfraRed | Nov 14 16:22:21 NOTICE[173]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 195.147.223.253, request '441314433637@goat' does not exist |
| 19:25.10 | InfraRed | [goat] in extensions is defained to sIP dial the termination company |
| 19:25.19 | file | extension 441314433637 does not exist in context goat |
| 19:25.48 | *** join/#asterisk Lurr (n=pr0ph3t@pcp04927291pcs.wolfrd01.fl.comcast.net) |
| 19:26.06 | ManxPower | InfraRed, the context you specified for the device in iax.conf does not exist in extensions.conf |
| 19:26.15 | *** part/#asterisk Lurr (n=pr0ph3t@pcp04927291pcs.wolfrd01.fl.comcast.net) |
| 19:26.16 | hhoffman | InfraRed, in [goat] put exten => 441314433637,WhatEverYouWantToDO |
| 19:26.42 | brettnem | anyone in here really good with the polycom phones?? |
| 19:26.43 | InfraRed | hmm |
| 19:26.47 | InfraRed | good point |
| 19:26.52 | InfraRed | will try the static approach |
| 19:26.53 | Rawplayer | hallo, when you want to use a asterisk server should i use a particulair fs to have the best performance or can i just use reiserfs or ext3? |
| 19:27.08 | ManxPower | Rawplayer, use anything you want. Asterisk is not disk intensive |
| 19:27.23 | *** join/#asterisk ^Howler (n=wwolfe@12.33.170.149) |
| 19:27.26 | Rawplayer | k |
| 19:28.38 | rculp | is anyone familiar with error 503 when attempting to make outbound calls on a config that appears to be accurate? |
| 19:29.36 | [TK]D-Fender | brettnem : I run 26 IP 600's here, what do you want to know? |
| 19:30.18 | brettnem | [TK]D-Fender: I'm trying to make ringing indications work on the phone.. with hints and presence.. |
| 19:30.51 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 19:30.55 | hhoffman | ok, is this something I should worry about? Unknown option '-' in '1-1/5551212' ? It seems to be b/c my card is called Zap/1-1 |
| 19:31.05 | brettnem | [TK]D-Fender: Currently asterisk only sends out inuse or online... not ringing.. If I could just figure out what status name to send the phone.. and then how to make the phone associate that status name with an icon, I could show a remote line ringing.. |
| 19:31.08 | Prival | Hi all, trying to setup a SIP phone connecting to a * behing a NAT firewall (monowall). The phone can connect to the * box, I can dial the extension of a SIP phone which is on the same * box but inside the firewall, but I can not have any voice... Any hints/pointers? |
| 19:31.22 | Chuji | There shouldnt' be any difference in bandwidth between a Sip outbound call versus inbound call using ulaw is there? |
| 19:31.35 | [TK]D-Fender | Ah.... rining? not sure on that one. |
| 19:31.41 | ManxPower | hhoffman, There are no zap channels called Zap/1-1 There is only one called Zap/1 and the system adds a -1 or -2 or whatever for each call on the channel. |
| 19:31.43 | [TK]D-Fender | inuse I've seen, but not ringing.... |
| 19:32.00 | InfraRed | Chuji: no |
| 19:32.00 | Chuji | Seems like Asterisk - > Sip/phone = bad and Sip/phone -> Asterisk = good |
| 19:32.04 | [TK]D-Fender | I'm gearing up for it myself. |
| 19:32.07 | ManxPower | you NEVER put in the -1 -2 -3 or whatever |
| 19:32.10 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 19:32.11 | hhoffman | ManxPower: ah, ok... so it adds the -1... is that warning anything to worry about? |
| 19:32.12 | afrosheen | Chuji: not unless it's trunking somehow |
| 19:32.25 | hhoffman | oh! I think I know what you're saying |
| 19:32.34 | brettnem | [TK]D-Fender: well any idea on how to match a status name to an icon? I see all these states in the XML file.. under CTX_ |
| 19:32.37 | ManxPower | hhoffman, Yes it's something to worry about. You are specifying the -1 in the Dial command and it won't work |
| 19:32.51 | hhoffman | ManxPower: right, sorry... I'm a little slow ;-) |
| 19:33.08 | LostFrog | ok.. what is the web page for looking up NPA/NXX calling areas? |
| 19:33.11 | [TK]D-Fender | brettnem : Have to say I've never tried. My IP 601 w/ Attendant modules is on backorder.... |
| 19:33.15 | konfuzed | shite im using debian for the first time and debian doesnt use /usr/src/linux for kernels or kernel sources |
| 19:33.18 | ManxPower | LostFrog, there isn't. |
| 19:33.24 | ManxPower | What specific are are you interested in? |
| 19:33.26 | brettnem | [TK]D-Fender: well I've sent it different msnsubstatus and I can see it change.. in fact, I've made it display 3 different icons.. idle., ringing, and busy.. |
| 19:33.36 | afrosheen | konfuzed: symlinks? |
| 19:33.37 | [TK]D-Fender | brettnem : haven't gotten to actually TRYING it yet... |
| 19:33.44 | hhoffman | ManxPower: thanks, that was it |
| 19:33.46 | brettnem | [TK]D-Fender: however, ringing shows as "away".. I can't figure out how to change the icon |
| 19:33.58 | LostFrog | I saw one before where you put in your exchange and area code and it lists the exchanges that are local. |
| 19:34.04 | brettnem | [TK]D-Fender: yeah.. just starting on it myself really.. but the docs suck |
| 19:34.09 | *** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net) |
| 19:34.14 | [TK]D-Fender | brettnem : could be on the indication level that * reports back. Hint implementation is somewhat spotty right now IIRC |
| 19:34.20 | konfuzed | [14:39:55] <stew> konfuzed: you can feel free to make the link from /usr/src/linux to /usr/src/kernel-headers-2.x.x.x-x-x but if that fixes it you should file a bug with asterisk |
| 19:34.31 | [TK]D-Fender | brettnem : I know... have you tried calling Polycom on it directly? |
| 19:34.46 | brettnem | [TK]D-Fender: I have the functionality working just fine.. I just need to knwo what "text" to send to the phone to get it to display a ringing ion |
| 19:34.51 | konfuzed | is this a common thing with debian installs |
| 19:35.04 | brettnem | [TK]D-Fender: polycom won't talk to anyone who isn't a authorized polycom retailer. |
| 19:35.06 | LostFrog | Like this: http://members.dandy.net/~czg/lca_prefix.php :) |
| 19:36.02 | Math` | thats a damn useful site :) |
| 19:36.14 | [TK]D-Fender | brettnem : My retailer got me in direct touch with them... check with yours |
| 19:36.28 | brettnem | [TK]D-Fender: ok, I'll check that.. |
| 19:36.42 | brettnem | LostFrog: I don't think that site actually tells you what the local calling area is made of tho.. careful |
| 19:37.16 | ManxPower | Your local calling area is defined BY YOUR CARRIER. |
| 19:37.37 | ManxPower | MY local calling are is the ENTIRE state of Louisiana and Mississippi, for example. |
| 19:37.43 | brettnem | ManxPower: it's defined in the carrier's tariffs. it's a complicated system |
| 19:38.01 | brettnem | well, for ILECs at least.. RBOCs and such |
| 19:38.09 | ManxPower | brettnem, I would call that "It is defined by your carrier" since your carrier files for the tarrifs. |
| 19:38.15 | brettnem | sure |
| 19:38.21 | ManxPower | Yes, the ROBCS are somewhat limited in what they can offer. |
| 19:38.40 | brettnem | I'm a carrier and I define my basic local calling area to match the local provider |
| 19:38.45 | brettnem | (SBC) |
| 19:39.11 | Prival | Chuji: That's whet I'm thinking, but I believe I forward everything in all directions 5060, 10000-2000 41000-45000 (rtp.conf) |
| 19:39.56 | konfuzed | ManxPower: OK i got rtfm README.Linux26 |
| 19:41.42 | *** join/#asterisk }cytrak{ (n=kvirc@208.63.19.172) |
| 19:41.46 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-79.cpe.netcabo.pt) |
| 19:41.47 | }cytrak{ | does asterisk compile on a Dual core processor ? |
| 19:41.49 | fulgas | hey |
| 19:41.54 | }cytrak{ | Do I need something more than a single XEON for an IVR system of with 20 simultaneous calls |
| 19:42.06 | afrosheen | our box is dual xeon, it never hurts :) |
| 19:42.07 | Math` | }cytrak{: which codec |
| 19:42.20 | InfraRed | }cytrak{: check voip-info.org |
| 19:42.31 | Math` | }cytrak{: bah... I suggest you to buy a dual xeon system anyways... 40% performance more for a 10% cost increase, I think its worth it |
| 19:42.32 | InfraRed | article is called "dimensioning a server" |
| 19:42.36 | ManxPower | }cytrak{, We get dial SMP Xeon motherboards w/ 1 CPU. Then we can add a 2nd CPU later if we need it. |
| 19:42.37 | InfraRed | or soemthing like that |
| 19:42.42 | Math` | }cytrak{ and, your server will last longer |
| 19:42.44 | ManxPower | dial == dual |
| 19:43.10 | InfraRed | btw |
| 19:43.16 | InfraRed | thanks for the help guys |
| 19:43.24 | Math` | ManxPower: er, you got to gake the cpu from the same batch (aka same stepping) |
| 19:43.26 | afrosheen | cytrak asked a good question, how is dual core support in linux right now, treated like SMP or what? |
| 19:43.37 | InfraRed | the extension was the issue putting a static entry there worked |
| 19:44.03 | Math` | afrosheen: dual core are supported just like if it was 2 cpus |
| 19:44.22 | afrosheen | so yeah, SMP. that's cool then. |
| 19:44.41 | ManxPower | Math`, I guess that depends on the motherboard, but I'm pretty sure we will be able to get that exact same Xeon for a while. |
| 19:44.46 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
| 19:44.50 | zoa | helooooooooooooooo |
| 19:44.53 | zoa | doberden |
| 19:44.54 | Math` | so is hyperthreading, except a preemptive scheduler have been implementd |
| 19:45.19 | ManxPower | Math`, actually hyperthreading can cause problems with Asterisk |
| 19:45.23 | zoa | nah |
| 19:45.27 | ManxPower | see the mailinglist archives |
| 19:45.33 | zoa | hyperthreading gives no problem |
| 19:45.40 | Math` | ManxPower: and dual-processor doesnt? |
| 19:45.42 | zoa | but the preemptive thing gives a problem with misdn |
| 19:45.46 | ManxPower | zoa, we had some pretty significant issues with hyperthreading |
| 19:45.51 | zoa | i never had |
| 19:45.53 | *** join/#asterisk pussfeller (n=todd@12.150.129.170) |
| 19:45.55 | Prival | afrosheen: Any new hints for me? I now can dial from the SIP phone that is on the internet to a SIP phone that is connected to the local net behind the firewall, but still no voice... I believe I forward everything in all directions 5060, 10000-2000 and 41000-45000 (rtp.conf) |
| 19:45.56 | zoa | but you need to use 2.6 |
| 19:45.59 | ManxPower | Math`, dual processer / dial core processor is NOT the same as hyperthreading |
| 19:46.13 | ManxPower | zoa, that could be the case. |
| 19:46.23 | Math` | ManxPower: application-wise should be the same |
| 19:46.30 | afrosheen | Prival: well if you get no voice, it's still some kind of rtp problem |
| 19:46.30 | Math` | except if your talking about Zap drivers |
| 19:46.31 | ManxPower | Prival, and your externip and localnet= settings are correct? |
| 19:46.39 | zoa | my recommendation is: 2.6 -> ht, 2.4 -> definately no ht |
| 19:47.02 | ManxPower | Prival, and your NAT box is not the same as your Asterisk box? |
| 19:47.48 | ManxPower | afrosheen, I give up when people don't answer me or argue with me. |
| 19:47.50 | Prival | ManxPower: externip is set to my monowall (firewall) IP and internip is ser to my internal IP range (behind the firewall) |
| 19:48.19 | ManxPower | Prival, you mean your localnet= is set to your internal IP range? |
| 19:48.42 | ManxPower | since there is not an internip= option |
| 19:48.47 | konfuzed | shite I have a symlink from /usr/src/linux to /lib/modules/2.6.8-2-386/ and I still get the apears to be no sources error |
| 19:48.47 | Prival | I don't have a localnet directive in sip.conf... |
| 19:48.56 | konfuzed | when I do make linux26 |
| 19:48.58 | ManxPower | Prival, well that's why it's not working. |
| 19:49.05 | ManxPower | Or at least one of the reasons. |
| 19:49.08 | }cytrak{ | ok I checked some of the voip-info.org notes and I guess a Xeon w/ 2GB should be enough for what I'm plannign to do with asterisk |
| 19:49.32 | zoa | ask me and i will tell you for sure :) |
| 19:49.46 | konfuzed | there is no directory called build in the /lib/modules/2.6.8-2-386 directory |
| 19:49.53 | }cytrak{ | it's just an IVR system that uses a T100P and can only handle 24 calls concurrently |
| 19:50.00 | Prival | ManxPower: Ok, let me try that... |
| 19:50.01 | ManxPower | }cytrak{, That would depend on what you are doing with the system. For exmaple are you wanting to handle 24 calls using G729 codec? |
| 19:50.01 | *** join/#asterisk MattH (n=MattH@63.174.244.174) |
| 19:50.06 | zoa | go easy on the ram, you dont need that much |
| 19:50.11 | zoa | a xeon will do more than ok |
| 19:50.18 | }cytrak{ | really ? |
| 19:50.25 | }cytrak{ | 1-2GB |
| 19:50.27 | zoa | i'd use a p4 for it |
| 19:50.31 | zoa | 512 will be ok too |
| 19:50.34 | ManxPower | }cytrak{, And are you actually using a T100P or will you use a TE110P? |
| 19:50.43 | konfuzed | zoa: is there troubles with asterisk o nkernel 2.6 on a simple athlon processor |
| 19:50.48 | MattH | Hi, when creating a call file... is there some way that I can make asterisk adhear to my call plan to choose where to send the file? IE.. rather then doing Channel: Zap/g1/number is there some way I can just do "number" and have it figure out where to send it? |
| 19:50.50 | ManxPower | The TE series cards are supposed to do DMA and so, in theory, can handle more channels with less CPU |
| 19:51.16 | ManxPower | Math`, you mean like Channel: Local/915551212@outgoing-zap |
| 19:51.19 | Prival | ManxPower: It works!!! Thanks all for the great help! |
| 19:51.51 | }cytrak{ | ManxPower: TE110P |
| 19:52.13 | MattH | ManxPower, kinda.... but what's outgoing-zap? |
| 19:52.38 | ManxPower | Math`, whatever context the exten => 91NXXXXXX,1,Dial(blah line is in |
| 19:52.47 | MattH | ok |
| 19:52.56 | MattH | so I'd do something like Local/5705551212@from-inside |
| 19:53.06 | MattH | I didn't think about doing it that way |
| 19:53.07 | }cytrak{ | shouln't ulaw be enough |
| 19:53.07 | MattH | :) |
| 19:54.02 | ManxPower | }cytrak{, if you are doing only PSTN/ULAW to/from IVR you should be able to handle at least 96 calls with that setup |
| 19:55.20 | }cytrak{ | ManxPower: I'm sorry for my stupid questions but I'm new to the telephone stuff ... but since the T1 can only give me 24 channels shouldn't I be only able to get 24 cocurrent calls ? |
| 19:55.37 | ManxPower | }cytrak{, not if you have a 4 port card |
| 19:55.51 | ManxPower | Add another CPU and you should be able to put in a 2nd 4-port card. |
| 19:56.16 | }cytrak{ | by the way the TE110P (single span) would be connected to a siemens PBX not the pstn |
| 19:57.04 | }cytrak{ | I can configure the PBX to use a signiling protocol such as g.711 I think |
| 19:57.29 | }cytrak{ | I'm getting my TE110P soon and then i will know for sure |
| 19:57.31 | }cytrak{ | :-) |
| 19:57.32 | ManxPower | }cytrak{, All calls on voice T-1s in the USA are ulaw |
| 19:57.45 | ManxPower | ulaw is a CODEC, not a PROTOCOL. The PROTOCOL you would want is PRI |
| 19:57.49 | }cytrak{ | cool thanks for letting me know that |
| 19:58.10 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:58.33 | MattH | hrmm |
| 19:58.48 | MattH | well local/ seems to work except when I go out over my LD provider.. it seems to connect but never goes and connects to the internal extension |
| 19:58.50 | InfraRed | T1s suck |
| 19:58.55 | InfraRed | E1s > T1s |
| 19:58.56 | InfraRed | :) |
| 20:00.03 | }cytrak{ | ManxPower: so I don't have to worry about the codec on the PBX part , I just got make sure the PBX and the asterisk server w/ TE110P use the same signalling protocol ? |
| 20:00.26 | ManxPower | }cytrak{, correct |
| 20:00.42 | }cytrak{ | thanks |
| 20:01.47 | brettnem | what's the bandwidth of a DS0 on an E1.. anyone? |
| 20:02.06 | }cytrak{ | hehe I'm checking the voip-info.org now and the asterisk protocols are only SIP/AIX(2)/MGCP/H323 |
| 20:02.10 | }cytrak{ | no PRI |
| 20:02.53 | brettnem | come on.. anyone.. 56K or 64K? |
| 20:03.14 | MattH | I'm thinking 64K brettnem |
| 20:05.07 | kink0 | another newbie question: when I dial 8500 I am asked about login and password, what must I do ? ( default sample files ) |
| 20:05.15 | afrosheen | cytrak: I believe asterisk interfaces with PRI via zaptel stuff |
| 20:05.28 | ManxPower | brettnem, 64K |
| 20:05.42 | brettnem | ahh. thx |
| 20:05.49 | ManxPower | }cytrak{, then voip-info sucks. Try Digium's page for the card you will use. |
| 20:06.56 | brettnem | }cytrak{ try again: http://www.voip-info.org/wiki/view/Asterisk |
| 20:08.21 | *** part/#asterisk gorauskas (n=gorauska@206-176-255-74.vbbn.com) |
| 20:12.57 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:13.31 | [TK]D-Fender | blitzrage : I just want...bang!bang!bang! |
| 20:13.34 | [TK]D-Fender | ;) |
| 20:13.36 | blitzrage | LOL |
| 20:13.54 | blitzrage | good times... :) |
| 20:14.08 | [TK]D-Fender | Did you guys go to Aria? |
| 20:14.38 | blitzrage | yah... me and JunK-Y did |
| 20:15.01 | *** part/#asterisk Uther_P (n=uther_p@66.180.120.82) |
| 20:15.18 | *** join/#asterisk cripito (n=ncripito@82.153.157.113) |
| 20:15.24 | cripito | hi |
| 20:16.07 | [TK]D-Fender | How late/early? |
| 20:16.29 | blitzrage | we got back at 5:30am when Julie got up to go to work :) |
| 20:17.24 | Math` | lol |
| 20:17.25 | brettnem | hey all.. anyone using app_page? |
| 20:17.35 | brettnem | the new one, that is.. |
| 20:18.27 | *** join/#asterisk gorauskas (n=root@206-176-255-74.vbbn.com) |
| 20:19.47 | brettnem | anyone? I am speaking to 312 people, right? ;) |
| 20:19.52 | [TK]D-Fender | blitzrage : Were'nt out for long then... dead? |
| 20:20.08 | [TK]D-Fender | brettnem : *crickets* |
| 20:20.13 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 20:20.15 | brettnem | no kidding |
| 20:20.16 | [TK]D-Fender | I've never used PAGE |
| 20:20.19 | blitzrage | [TK]D-Fender: yah... out for a couple of hours at the club. Then I crashed :) |
| 20:20.27 | [TK]D-Fender | How was it? |
| 20:20.37 | brettnem | it's a neat app.. but it's not setting Alert_info right |
| 20:21.33 | *** join/#asterisk sigwerk (n=sigwerk@athena.rootednetworks.com) |
| 20:24.07 | mfarley | brettnem: I have given it a try (haven't gotten it to quite work, though). |
| 20:24.23 | *** part/#asterisk gorauskas (n=root@206-176-255-74.vbbn.com) |
| 20:25.06 | mfarley | brettnem: I think it may have to do with the SNOM phones I am trying to page. |
| 20:26.28 | docelmo | brett what are you trying to figure out? |
| 20:26.42 | docelmo | I am using GXP2000's and I dont know if its working to be quite honest.. :) |
| 20:27.13 | konfuzed | so i got through zaptel make install and lib pri make install but /usr/src/asterisk/make is comlaining of termcap support not found yet apt-get install termcap-compat says it is already the altest version |
| 20:27.29 | konfuzed | is this the wrong termcap or something |
| 20:28.06 | *** join/#asterisk bumblefsck (n=bumblefs@69-160-145-156.ontrca.adelphia.net) |
| 20:29.04 | konfuzed | checking for tgetent in -ltermcap.... no in -ltinfo... no in -lcurses... no in -lncurses... no |
| 20:30.00 | rayvd | those are sad times! :( :( |
| 20:31.09 | brettnem | docelmo: I need to set _ALERT_INFO to make my phones do a ring-answer |
| 20:31.16 | *** join/#asterisk freespace-in (n=special@ppp-70-225-137-82.dsl.ipltin.ameritech.net) |
| 20:31.27 | brettnem | just doing a setvar isn't passing it to the outbound channels.. ideas? |
| 20:31.27 | docelmo | ohh mine are automatic.. |
| 20:31.32 | freespace-in | has anyone used mgcp to dial outbound? |
| 20:31.37 | docelmo | anyone know how to remove this from the sip messages |
| 20:31.38 | docelmo | "rport" tag in the VIA line |
| 20:31.50 | freespace-in | it didn't seem to like exten => _NXXXXXX,1,Dial(MGCP/aaln/S1/4@c3810/${EXTEN}) |
| 20:32.07 | rayvd | docelmo: edit the source or use SER? :) |
| 20:32.08 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:32.21 | docelmo | Its being produced by SER that was the problem |
| 20:32.36 | brettnem | mfarley: are you setting ALERT_INFO? |
| 20:32.38 | *** join/#asterisk gorauskas (n=gorauska@206-176-255-74.vbbn.com) |
| 20:32.53 | freespace-in | i get Nov 14 15:32:29 NOTICE[27785]: chan_mgcp.c:1639 find_subchannel_and_lock: Gateway 'c3810/6527650' (and thus its endpoint 'aaln/S1/4') does not exist |
| 20:32.53 | freespace-in | Nov 14 15:32:29 WARNING[27785]: chan_mgcp.c:3509 mgcp_request: Unable to find MGCP endpoint 'aaln/S1/4@c3810/6527650' |
| 20:33.17 | hhoffman | does the user asterisk need to own everything in /var/spool/asterisk/voicemail? |
| 20:33.39 | hhoffman | I'm getting "Unable to create lock file '/var/spool/asterisk/voicemail/default/1000/INBOX': No such file or directory" |
| 20:33.43 | brettnem | all your base |
| 20:33.58 | X-Files | hello pippls ! I use gateway EUSSO UTG7104 Protocol SIP , i have problem :( please check : http://pastebin.com/429483 |
| 20:35.20 | mfarley | brettnem: Yeah, I have tried various things with ALERT_INFO, but haven't gotten the phones to do a page yet. |
| 20:36.14 | brettnem | mfarley: I'm not even sure where in the source to put it.. I'm trying to insert a pbx_builtin_setvar_helper(chan, "_ALERT_INFO", "Intercom"); somewhere useful in the source.. |
| 20:36.47 | *** join/#asterisk darby_t (i=darby_t@dla115.neoplus.adsl.tpnet.pl) |
| 20:37.41 | mfarley | brettnem: Yeah, I've been a bit puzzled by all that myself. I can successfully specify a ringtone on an unpatched system using ALERT_INFO, however. |
| 20:38.05 | mfarley | brettnem: I was hoping the 'intercom' thing would be that easy as well, but no such luck. |
| 20:38.43 | lunk | word, upgrading from 4.5MB Data only to 6.0MB and 16 Sip channels for only $37.50 more / month |
| 20:39.38 | brettnem | mfarley: seems like an obvious bug.. I can think of ways to do it with local channels, butthat's ugly |
| 20:39.44 | brettnem | actually, I'm going to try that now |
| 20:40.21 | *** join/#asterisk delphiuk (n=Richard@host86-128-157-3.range86-128.btcentralplus.com) |
| 20:41.34 | konfuzed | ok so now /usr/src/asterisk/ make compplains about -lssl |
| 20:41.39 | konfuzed | openssl is installed |
| 20:42.04 | konfuzed | openssl-dev is not an available package |
| 20:42.32 | konfuzed | anyone know what cause this make error |
| 20:43.10 | brettnem | mfarley: also channels with an active call will get added late to it.. |
| 20:45.32 | X-Files | Please need help !!! I use gateway EUSSO UTG7104 Protocol SIP , i have problem :( please check : http://pastebin.com/429483 |
| 20:45.42 | brettnem | looks like app_page was an attempt to win some public option, but it really kinda sucks |
| 20:46.23 | brettnem | mfarley: btw, adding paging local channels works.. just point to a context that does nothing but set the variable then goto the regular context that dials the phone.. |
| 20:48.10 | *** join/#asterisk copantl (n=galel@205.240.205.192) |
| 20:48.29 | copantl | i need support from digium!! |
| 20:48.47 | file | so call them |
| 20:48.49 | *** join/#asterisk argos73 (n=mike@adsl-70-228-109-5.dsl.akrnoh.ameritech.net) |
| 20:48.52 | *** part/#asterisk delphiuk (n=Richard@host86-128-157-3.range86-128.btcentralplus.com) |
| 20:48.55 | *** join/#asterisk c0w (i=c0w@cpc1-staf1-3-0-cust86.brhm.cable.ntl.com) |
| 20:49.08 | copantl | very smart fike!!! :) |
| 20:49.18 | copantl | file.. sorry |
| 20:49.53 | mfarley | brettnem: What syntax are you using for that variable-setting? |
| 20:50.25 | copantl | i have a wte110p in my asterisk , and is connected to a lucent switch |
| 20:50.46 | copantl | via PRI |
| 20:52.04 | copantl | and, i have a particular behavior, when i place a call i got answer suppervision |
| 20:52.12 | brettnem | mfarley: I'm old school.. setvar(_ALERT_INFO=Intercom) |
| 20:52.35 | ManxPower | brettnem, Naw, OLD school is setvar(ALERT_INFO=Intercom) |
| 20:52.49 | brettnem | ManxPower: that's just plain different.. ;) |
| 20:53.00 | ManxPower | brettnem, from the times before _ and __ variables |
| 20:53.14 | copantl | the call is ring and the asterisk give me a onhook before the other party answer |
| 20:53.42 | copantl | any body know? |
| 20:54.29 | ManxPower | copantl, what country are you in? Where is the line coming from? Telco? PBX? |
| 20:54.40 | ManxPower | copantl, What digium cards do you have? |
| 20:55.00 | freespace-in | i answered my own question |
| 20:55.06 | freespace-in | it should look like this |
| 20:55.06 | freespace-in | exten => _NXXXXXX,1,Dial(MGCP/aaln/S1/4@c3810|${EXTEN}) |
| 20:55.39 | copantl | wait |
| 20:55.49 | copantl | im in Honduras |
| 20:56.11 | copantl | the DID comes from a Lucent switch |
| 20:56.15 | hhoffman | when I call comes in on the console CLI> is there a way to see the caller-id? |
| 20:56.31 | ManxPower | copantl, Digium analog cards do not suport DIDs |
| 20:56.34 | copantl | my card is TE110P |
| 20:56.43 | copantl | is not a analog card |
| 20:56.53 | copantl | is a single E1 card |
| 20:56.57 | ManxPower | hhoffman, Yes, put Noop(CALLERID=${CALLERID}) in a priority where the call is coming in on |
| 20:57.12 | ManxPower | copantl, you need to make sure the signaling is the same on the Lucent side and on the Asterisk side. |
| 20:57.21 | hhoffman | ManxPower: thanks! |
| 20:57.32 | konfuzed | well thats one big pile of code |
| 20:58.10 | copantl | yes is the same but some... the courious thing is that some calls dont have FAS |
| 20:58.21 | copantl | Answer supervision |
| 20:58.22 | konfuzed | perhaps I should have apt-get install asterisk to grab all the dependencies and the uninstall just asteterisk and then do the cvs co asterisk |
| 20:58.35 | konfuzed | maybe next time |
| 20:59.11 | hhoffman | ManxPower: so, exten => s,1,Noop(CALLERID=${CALLERID}) |
| 20:59.15 | brettnem | HAHA |
| 20:59.27 | copantl | only for 233-xxxx dont have the anwer supervision |
| 21:00.04 | copantl | any idea? |
| 21:00.07 | brettnem | mfarley: ok, so I did a page using the local channel driver.. didn't answer the page.. rolls into voicemail... Now my app_voicemail is in app_meetme.. haha and my presence indicator is showing constantly busy |
| 21:00.32 | jontow | did mysql_vm_routines.h disappear from CVS HEAD? |
| 21:00.36 | copantl | my country have several different kinds of switches, like OKI, SIEMENS and Lucent |
| 21:01.35 | jontow | nm.. yes it did :) |
| 21:01.38 | copantl | ManxPower: any idea? |
| 21:02.13 | hhoffman | so, for a IAX DID is exten => always going to start out with the DID number? |
| 21:03.05 | ManxPower | hhoffman, only if the incoming calls hit exten => s (like a call coming in on an ANALOG FXO port) |
| 21:03.44 | ManxPower | copantl, You need to find out if the signalling is PRI, E&M, Wink, or whatever. then set that in Asterisk to match what the Lucent is expecting |
| 21:03.47 | *** join/#asterisk darby_d (i=darby_t@dkv194.neoplus.adsl.tpnet.pl) |
| 21:03.59 | copantl | is PRI S5 |
| 21:04.04 | copantl | for Lucent |
| 21:04.10 | ManxPower | Sounds like a classic Asterisk is set for Loopstart or Kewlstart and the telco switch is set for E&M/Wink |
| 21:04.20 | hhoffman | ManxPower: ok... then I'm a little confused :-( I've got a DID from teliax, and I've setup a context called [incoming-iax]... can I just use regular vmail extension within that? like exten => 500,1,VoiceMailMain() ? |
| 21:04.28 | ManxPower | copantl, then asterisk has to be set to pri_net or PRI_CPE |
| 21:04.38 | konfuzed | wow i got to make progdocs and now it keeps scrolling problems running dot. check your isntallation sh: line 1: |
| 21:04.44 | konfuzed | wow too fast for me to read |
| 21:04.45 | copantl | is PRI_CPE |
| 21:04.53 | *** join/#asterisk Paolo1000 (n=Paolo@ppp-62-10-136-167.dialup.tiscali.it) |
| 21:04.56 | ManxPower | hhoffman, if you have a DID from Teliax Asterisk will route the call to the exten => line that matches your DID |
| 21:04.57 | copantl | i recieve clock |
| 21:04.58 | Paolo1000 | hi |
| 21:05.06 | konfuzed | yeah it finally stopped |
| 21:05.17 | konfuzed | crazy thousands of lines like that |
| 21:05.18 | ManxPower | copantl, then you need to look thru the mailing list archives to try to find a solution |
| 21:05.20 | Paolo1000 | i'm going to 1 istallation and i have a pro.. |
| 21:05.21 | konfuzed | no shittin |
| 21:05.22 | ManxPower | ~mailinglist |
| 21:05.25 | jbot | it has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 21:05.35 | copantl | ok thanx any way |
| 21:05.52 | Paolo1000 | in asterisk fedora core 3 from voip-info.org |
| 21:05.55 | Paolo1000 | ... |
| 21:06.15 | hhoffman | ManxPower: then how can I offer the voicemail menu when I call in thru the Teliax DID number? |
| 21:06.35 | Paolo1000 | ln -s /lib/modules/2.6.9-1.667/build/linux-2.6 |
| 21:06.40 | hhoffman | sorry to be asking so many questions... I'm currently reading the book, and googleing to try and figure all of this out |
| 21:06.53 | Paolo1000 | i didnt this file |
| 21:07.32 | Paolo1000 | i have an linux-2.6.x in etc/... |
| 21:08.17 | Paolo1000 | what should i do? |
| 21:08.21 | ManxPower | hhoffman, How about this: http://pastebin.ca/28701 |
| 21:08.24 | konfuzed | hm |
| 21:08.35 | ManxPower | BTW, [incoming] is where all calls from untrusted sources go into. |
| 21:09.21 | brettnem | ?!? why does app_voicemail have a temporary greeting recording, but no way to use it?! wtf |
| 21:09.33 | ManxPower | brettnem, file a bug |
| 21:09.41 | brettnem | fabulous |
| 21:09.59 | ManxPower | brettnem, Hurry, we are already on RC2 |
| 21:10.12 | hhoffman | ManxPower: thanks, I think I understand what's going on there |
| 21:10.22 | brettnem | it's been around for quite a while acutally.. I just tested it on rc1 and on cvs like from march |
| 21:10.25 | ManxPower | hhoffman, that is from my own dialplan |
| 21:10.57 | *** join/#asterisk oelewapperke (i=oelewapp@alf.ulyssis.student.kuleuven.be) |
| 21:11.10 | oelewapperke | any hope for asterisk's sip-over-tcp these days ? |
| 21:11.23 | hhoffman | ManxPower: ah, ok. cool. Does it make sense to allow ppl calling on via Teliax to get to the VoiceMailMain() ? |
| 21:11.33 | docelmo | From what I heard its in the works oel.. |
| 21:11.48 | brettnem | ManxPower: am I missing something? Is there some hidden way to get the temporary greeting to work? |
| 21:11.54 | ManxPower | hhoffman, Depends on your local policy, but usually yes. |
| 21:12.04 | oelewapperke | docelmo: any status pages / mail/news posts on it ? |
| 21:12.06 | Paolo1000 | someone can say me what i should do? |
| 21:12.06 | ManxPower | brettnem, No idea. I've never tried to use it. |
| 21:12.17 | SwK[Work] | why do companies insist on using asterisk@home? |
| 21:12.18 | docelmo | Check bugs.digium.com |
| 21:12.25 | X-Files | Please need help !!! I use gateway EUSSO UTG7104 Protocol SIP , i have problem :( please check : http://pastebin.com/429483 |
| 21:12.28 | SwK[Work] | done they see the "@HOME" part of the name? |
| 21:12.31 | docelmo | I just know they said they were gonna start on it. |
| 21:12.36 | docelmo | Cause Companies are stupid |
| 21:12.45 | hhoffman | ManxPower: ok, great. This is so much fun... I haven't had something to learn this complex in quite a long time :-) |
| 21:13.00 | docelmo | and whoever made AMP / A@H is gonna burn in hell |
| 21:13.50 | oelewapperke | ah nice thx |
| 21:14.02 | CoffeeIV_ | no, call it asterisk@Work |
| 21:14.19 | docelmo | Ya and it will be something that actually works |
| 21:14.39 | CoffeeIV_ | then I can use it at home |
| 21:14.52 | mog_work | lol |
| 21:14.54 | mog_work | genius |
| 21:14.57 | ManxPower | hhoffman, Basically I have three types of incoming calls into Asterisk. Calls from the PSTN, VoIP Providers and stuff like that, those all go into the [incoming] context and can dial extensions. I have calls that come from trusted sources and they go into the [toll-access] and [local-access] contexts that are allowed to dial out of the system |
| 21:15.26 | hhoffman | ManxPower: yeah, that's the exact setup I'm trying to get to |
| 21:15.56 | ManxPower | hhoffman, all my [incoming] context does it to route calls to the place I want them (USUALLY a menu) |
| 21:16.33 | Druken | ManxPower: sounds very simular :) |
| 21:16.38 | hhoffman | ManxPower: it's possible then to have a incoming call on either line dial out through the other line, right? |
| 21:17.02 | hhoffman | so, if it's local to the PSTN dial out there if coming on via IAX |
| 21:18.51 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:19.12 | infinity1 | i'm using voipjet and i'm getting |
| 21:19.13 | infinity1 | <PROTECTED> |
| 21:19.13 | infinity1 | <PROTECTED> |
| 21:19.28 | infinity1 | it was working this morning. something is intermittent |
| 21:19.44 | Druken | would be a problem on their side... |
| 21:19.54 | InfraRed | http://www.honest.demon.nl/stuff/capfaggo.jpg |
| 21:20.18 | infinity1 | they seem to have this intermittent problem often. this is the first time i've tried tracking it down. |
| 21:20.47 | hhoffman | ManxPower: does you extensions file just have all of the local vmail box extensions then? |
| 21:20.55 | Druken | i've heard a few people bitch about voipjet... |
| 21:21.02 | infinity1 | Druken: how might i narrow the problem down further. |
| 21:21.18 | infinity1 | Druken: i can now see why :) ...lucky i use someone else to receive calls. |
| 21:21.38 | Druken | :) |
| 21:22.09 | infinity1 | Druken: is there someone else that works good (er when voipjet works, it seems fine) and has descent rates? |
| 21:22.15 | infinity1 | .011 for US calls ain't bad. |
| 21:22.46 | Druken | 1.1 cents is pretty good.. yeah.. i know i couldn't touch that... :) |
| 21:23.26 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 21:23.56 | infinity1 | i guess i'll have to setup some logic to try something else if they don't work |
| 21:24.02 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:24.12 | infinity1 | i can do .015 :) |
| 21:24.13 | infinity1 | heh |
| 21:24.21 | Druken | infinity1: hehe |
| 21:24.28 | asterboy | Any good rates for companies doing VOIP termination? |
| 21:24.33 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:24.39 | infinity1 | thats the highest i'll pay. final offer! |
| 21:24.53 | Druken | no deal :) |
| 21:25.24 | Paolo1000 | during installation of * i need to install an x100p |
| 21:25.32 | LostFrog | Free for US calls is good here. |
| 21:25.34 | LostFrog | <-- in US. |
| 21:25.43 | Paolo1000 | is right next steps? |
| 21:25.54 | Paolo1000 | Installation of Fedora 3 |
| 21:25.58 | Druken | LostFrog: i'll take free us calls... sounds good :) |
| 21:26.06 | Paolo1000 | installation of required rpm |
| 21:26.14 | InfraRed | US sucks for phones |
| 21:26.22 | Paolo1000 | yum for updates |
| 21:26.29 | InfraRed | why would anyone pay for incoming calls! |
| 21:26.33 | infinity1 | thats why everyone needs voip already. then it would all be free! |
| 21:26.37 | Paolo1000 | <PROTECTED> |
| 21:26.48 | asterboy | I bought one of those OEM Asterisk X100P to try...anyone report? |
| 21:26.49 | Druken | InfraRed: your talking cells... that's diffrent |
| 21:26.59 | Paolo1000 | %define buildsource 1 |
| 21:27.25 | Druken | asterboy: it's good to get your feet wet with... |
| 21:27.42 | Paolo1000 | now what i should change? |
| 21:27.44 | InfraRed | traceroute to skype.com (198.63.210.250), 64 hops max, 44 byte packets |
| 21:27.44 | InfraRed | <PROTECTED> |
| 21:27.44 | asterboy | sure is a good price at $10 |
| 21:27.45 | InfraRed | :D |
| 21:28.14 | asterboy | I would really prefer to terminate with VOIP connections though. |
| 21:28.27 | asterboy | Not sure who is a good company to go with here in Canada.?? |
| 21:28.37 | Druken | asterboy: so? bring in the calls by the x100, term by voip... i do that all the time |
| 21:28.51 | Druken | asterboy: depends on where in canada you are |
| 21:28.52 | asterboy | That is what I'm soing. |
| 21:28.52 | Paolo1000 | !? |
| 21:29.10 | asterboy | Just like to skip that step and go VOIP right into Asterisk |
| 21:29.32 | hhoffman | so I did a make samples but the O'Reilly book suggests not using the samples... I've created a fresh iax.conf, extensions.conf, and voicemail.conf... Should I just throw away the other conf files? |
| 21:29.35 | asterboy | There must be somebody providing that service. |
| 21:29.46 | Druken | asterboy: where in the west are ya? |
| 21:29.55 | asterboy | hhoffman: no |
| 21:30.09 | hhoffman | ok |
| 21:30.13 | asterboy | hhoffman: you'll see if you start asterisk that it will complain. |
| 21:30.19 | Paolo1000 | someone want help me? |
| 21:30.20 | hhoffman | gotcha |
| 21:30.27 | ender | file: file[laptop]: ping |
| 21:30.43 | hhoffman | Paolo1000: yep, go ahead and install that card if you are going to dial out on your POTS line |
| 21:31.15 | asterboy | Paolo1000: you need to define your question a little clearer. |
| 21:31.39 | asterboy | Paolo1000: I think hhoffman has you pegged. |
| 21:31.58 | asterboy | Druken: Alberta bound! |
| 21:32.14 | Paolo1000 | what should i do... |
| 21:32.14 | asterboy | Druken: I'm a hick, red-neck to the bone. |
| 21:32.17 | Paolo1000 | now.. |
| 21:32.18 | Druken | asterboy: 780 or 604 ? |
| 21:32.22 | asterboy | Druken: 780 |
| 21:32.31 | Druken | er.. 403 |
| 21:32.33 | Druken | damnit! |
| 21:32.33 | Paolo1000 | in guide that i have i have example of isdn |
| 21:32.37 | Paolo1000 | and 400 |
| 21:32.50 | Druken | asterboy: 7806691305 :) |
| 21:32.50 | Paolo1000 | and not x100 |
| 21:33.07 | asterboy | Druken: ya that is always a limiting factor...Calgary or Edmonton |
| 21:33.13 | asterboy | Druken: 403 or 780 |
| 21:33.24 | asterboy | Druken: yes |
| 21:33.42 | asterboy | Druken: the 669 exchange is very popular right now. |
| 21:33.50 | asterboy | Druken: what number is that? |
| 21:33.50 | Druken | yup |
| 21:34.29 | Druken | asterboy: mine... |
| 21:34.40 | morale | asterboy, there is a 669 exchange? i could only find 668 in calgary for my place |
| 21:35.15 | asterboy | Druken: intersting...so calgary is assign 668 and edmonton 669 |
| 21:36.13 | docelmo | Anyone know if I can pull the billable seconds from a variable in the dialplan? |
| 21:36.17 | Druken | asterboy: diffrent areacodes |
| 21:36.28 | docelmo | And if so.. What that variable is? |
| 21:37.11 | docelmo | asterboy what are you looking for? |
| 21:37.13 | docelmo | I do termination |
| 21:37.50 | docelmo | How many minutes? |
| 21:37.56 | Druken | asterboy: if you want.. gimie a call... 780-669-1305 x 101 and i'll get ya setup with someone for edmonton :) |
| 21:37.57 | asterboy | Just want to skip the FXO/FXS part of my asterisk box...go straight to the source via Internet. |
| 21:38.07 | Druken | i'm heading out, but you'll get my cell... |
| 21:38.15 | asterboy | ok calling... |
| 21:39.16 | X-Files | Please need help !!! I use gateway EUSSO UTG7104 Protocol SIP , i have problem :( please check : http://pastebin.com/429483 |
| 21:39.35 | Druken | asterboy: phone rang.. sorry :) |
| 21:39.44 | asterboy | left message. |
| 21:39.46 | Druken | k |
| 21:40.23 | asterboy | is that going to an asterisk box, 101 ==> cell etc.. |
| 21:40.30 | Druken | yeah |
| 21:40.36 | asterboy | Its great to see this stuff in action. |
| 21:40.50 | asterboy | voice is clear. |
| 21:40.50 | docelmo | So does anyone know the VAR for pulling billing seconds from MySQL? |
| 21:40.56 | docelmo | err asterisk? |
| 21:40.59 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 21:41.40 | asterboy | Not sure myself, but when I'm looking for that kind of thing I list all the tables and try to get a clue from the naming conventions. |
| 21:41.48 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 21:42.09 | asterboy | It could be a calc based on call start and end. |
| 21:43.09 | *** join/#asterisk Igbothom_III (n=HiltonT@static-84.217.240.220.dsl.comindico.com.au) |
| 21:43.20 | docelmo | That gives total duration.. but not when it answered till end.. |
| 21:43.22 | shmaltz | anybody seen this before: |
| 21:43.23 | *** join/#asterisk R3DB0x (i=nobody@66.142.28.36) |
| 21:43.24 | shmaltz | Cisco 7960 gets acall from zap/1, hits conf to call out on zap/2, then hits join, after a while cisco hangsup, at which point zap/1 and zap/2 can still talk, shouldn't asterisk hangup on all three? |
| 21:43.55 | docelmo | yes codec issue |
| 21:45.29 | asterboy | Anyone have some S100M modules to sell? |
| 21:46.07 | asterboy | Should start a telco used equipment classified |
| 21:47.00 | shmaltz | asterboy, asterisk-biz list is made just for that |
| 21:47.25 | asterboy | ah! I though there was something like that around. |
| 21:47.43 | asterboy | shoulda just asked :^) |
| 21:48.44 | hhoffman | thanks everyone for the help, esp. you ManxPower. Have a good night all |
| 21:50.29 | wunderkin | would anyone here have any idea why my provider wouldn't be able to loop my te410p even when it is in otherwise normal operation? |
| 21:51.26 | brettnem | wunderkin: not sure if those cards support that kind of remote loopup |
| 21:51.40 | wunderkin | kpfleming told me yes and support did too *shrug* |
| 21:52.02 | brettnem | well, there are about 3 or 4 different codes.. need to be sure the one they are sending is supported. |
| 21:52.11 | wunderkin | ah |
| 21:52.26 | wunderkin | well when i try to run zttool and press loop they say they can't see anything either |
| 21:52.47 | wunderkin | and when they are telling me to try to loop it i am seeing a lot of hdlc aborts, i run zttool and press loop nothin happens |
| 21:53.15 | rculp | if anyone is available to answer, does anyone know how to disable the auto-hangup after 20000 ms without answer? doesn't give me enough time to leave voicemail when calling someone |
| 21:53.36 | brettnem | and of course, if you can't get the t1 to sync, chances are they can't send you the code either |
| 21:53.46 | wunderkin | brettnem, the line works fine otherwise |
| 21:53.52 | brettnem | line? |
| 21:53.57 | wunderkin | yeah the t1 |
| 21:54.12 | brettnem | what makes you think that? |
| 21:54.14 | shmaltz | anybody seen this before: |
| 21:54.14 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 21:54.15 | shmaltz | Cisco 7960 gets acall from zap/1, hits conf to call out on zap/2, then hits join, after a while cisco hangsup, at which point zap/1 and zap/2 can still talk, shouldn't asterisk hangup on all three? |
| 21:54.26 | Math` | rculp: thats a dial() option |
| 21:54.27 | wunderkin | other than the intermittant red alarms :P yes the timing is fine i already checked that.. |
| 21:54.38 | brettnem | shmaltz: you just did an attendend transfer.. congrats |
| 21:54.46 | brettnem | you get intermitant reds? |
| 21:54.52 | wunderkin | yeah |
| 21:54.55 | brettnem | do you have timing set right? |
| 21:54.57 | shmaltz | brettnem, you wrong read my post again |
| 21:55.06 | wunderkin | yup i know thats the first thing i checked |
| 21:55.06 | brettnem | what's your span line look like? |
| 21:55.29 | brettnem | shmaltz: still sounds like an attended trasfer to me. ;) |
| 21:55.32 | wunderkin | span=1,1,0,esf,b8zs |
| 21:55.36 | Math` | shmaltz: he isnt wrong |
| 21:55.36 | wunderkin | span=2,2,0,esf,b8zs |
| 21:55.37 | brettnem | hmm |
| 21:55.47 | shmaltz | brettnem, you speak english? |
| 21:55.48 | wunderkin | i have problems on both, usually the 2nd |
| 21:55.55 | brettnem | shmaltz: sorry no.. I don't |
| 21:55.57 | *** join/#asterisk kn0x (i=tor@c212-151-198-78.swipnet.se) |
| 21:56.01 | wunderkin | heh |
| 21:56.18 | brettnem | wunderkin: have you asked them to kill the t1 and can you verify your ckt goes red constant? |
| 21:56.27 | brettnem | make sure you are looking at the same ckt |
| 21:56.27 | shmaltz | Math, no I'm not we are talking about a local confference issue and not an xfer |
| 21:56.52 | brettnem | shmaltz: depends on how those capabilities are implemented.. local mixing, server.. |
| 21:56.55 | wunderkin | brettnem: no.. but they did testing and they say its clean.. last time they did testing on 1 of them i saw things happen on the cli |
| 21:57.02 | InfraRed | gopher://sdf.lonestar.org/ |
| 21:57.19 | Math` | shmaltz: some phone works like that... when you conf 2 people and then hangup, the conf'd persons are transfered together |
| 21:57.21 | brettnem | wunderkin: if they didn't do a loopback test with your T1 card in loop, they CAN'T see errors from them to YOU |
| 21:57.38 | brettnem | shmaltz: actually, I think it's a configurable option |
| 21:57.45 | shmaltz | brettnem, thank you, i know its an implemetation issue (vs aconfig or user issue, like you suggestted it being a user issue), but am I the first one complaining? |
| 21:57.56 | wunderkin | brettnem: well i put a loopback plug on my side before the equipment too and they tested both that way and it was clean |
| 21:58.33 | wunderkin | its in a colo it goes niu -> 66 block -> cabinet - patch panel - me |
| 21:58.36 | shmaltz | Math, not traditional ones, maybe IP phones, but then one should have an option to hang them up first (maybe like the polycomd split) |
| 21:58.40 | brettnem | shmaltz: I didn't suggest it was a user error.. I suggested what happened was an attended transfer.. I didn't suggest you did it knowingly it's just kinda what happened |
| 21:59.10 | brettnem | wunderkin: and you are sure they tested to the plug and not just the nid? sounds like a cable issue.. |
| 21:59.13 | juanjoc | Hi, has anyone used the RxFAX and TxFAX apps from spandsp to talk to each other? |
| 21:59.18 | brettnem | wunderkin: have you checked for irc conflicts? |
| 21:59.22 | brettnem | er.. haha irq |
| 21:59.36 | shmaltz | brettnem, well maybe thats what you meant but this is what you worte: |
| 22:00.04 | brettnem | shmaltz: I'm not here to help you with symantics.. if you don't like what I have to say, just ignore me |
| 22:00.04 | shmaltz | brettnemshmaltz: you just did an attendend transfer.. congrats |
| 22:00.16 | asterboy | shmaltz: you talking about this link for asterisk-biz??? http://lists.digium.com/mailman/listinfo/asterisk-biz |
| 22:00.19 | brettnem | ~gwypf |
| 22:00.20 | jbot | from memory, gwypf is Get What You Pay For - this channel is full of volunteers who are here to help you. However, we can't hold your hand. If you need a specific problem solved immediately, there is a list of for-hire consultants located at: http://www.voip-info.org/tiki-index.php?page=Asterisk+Consultants |
| 22:00.24 | shmaltz | brettnem, that is not the case |
| 22:00.28 | *** join/#asterisk Flauto (n=zhao@c-24-14-197-214.hsd1.il.comcast.net) |
| 22:00.38 | Flauto | hi all |
| 22:00.42 | wunderkin | brettnem: hehe ouch.. well i told them how its connected and they said that it would have tested to me.. thats what i would think too, it would have to be the cable from the panel in my cabinet to the card though.. it cant be a problem with the card.. i have 2 other t1s crossed to another box and they are always fine.. i tested with all 4 to the other machine when i was doing testing at home |
| 22:00.42 | shmaltz | asterboy, yep |
| 22:00.52 | asterboy | shmaltz: ok thanks. |
| 22:01.49 | docelmo | ACK! RPORT SUCKS! |
| 22:01.54 | wunderkin | brettnem: i did the windows trick on it and i havent had any problems so far on it.. about 24 hours now.. theres only been one whole day lately that it didnt have problems before so im just letting it ride a bit |
| 22:01.57 | docelmo | What the hell is it used for anyhow? |
| 22:02.08 | shmaltz | brettnem,if this is not fixed in 1.2bx or HEAD then it should be fixed, since otherwise local conf will have to be disabled on these phones, polycom has a much better way of handling these |
| 22:02.43 | brettnem | wunderkin.. you have a RJ-45 loopback plug? |
| 22:02.49 | *** join/#asterisk zeedo (n=zeedo@80.68.92.188) |
| 22:02.49 | brettnem | what kind of loopback are you doing? |
| 22:03.12 | wunderkin | brettnem: yeah.. i had a hard loopback .. rj45 loopback plug in the patch panel in my cabinet |
| 22:03.33 | brettnem | and to hook up your server.. you remove the plug and attach a simple cable to the server. |
| 22:03.41 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:04.15 | wunderkin | yeah im just using a regular ethernet cable to my machine, my voip provider does the same and they havent ever had problems with cat5 cable.. im colo'd with them |
| 22:04.40 | wunderkin | i know you are supposted to have extra shielding :P |
| 22:05.02 | Flauto | is there anyone using vbuzzer? |
| 22:05.13 | Flauto | i mean using vbuzzer with asterisk |
| 22:05.20 | wunderkin | thats one thing i was wondering about but i dunno, normally happens in the wee hours of the morning, sometimes off and on all day.. seems to me like interference |
| 22:05.24 | marcus2 | i just tried making a modem connection over a linksys pap2 |
| 22:05.28 | marcus2 | oddly enough, it didnt work |
| 22:05.43 | Druken | ok.... |
| 22:06.28 | Math` | how are modem-over-ata stuff handled? |
| 22:06.50 | Math` | is there a special codec/protocol for data? |
| 22:07.30 | marcus2 | i'm wondering if i will have any better luck doing it over a channel bank connected directly to my * server |
| 22:07.38 | wunderkin | brettnem: they were saying something about a female loopback? if i was unable to loop my equipment.. whats that? |
| 22:08.02 | Flauto | vbuzzer seems registered but i can not receive any call and when i make calls it hear ringing tone and then, quiet |
| 22:08.20 | Flauto | i opened the rtp ports already |
| 22:09.00 | wunderkin | brettnem: im just trying to figure out what that would be used for in this case, how would it be different than using an rj45 loopback plug |
| 22:09.39 | justinu | wunderkin: there's no problem using a cat5 for t1 |
| 22:09.52 | justinu | unless it's like a reeally really long run |
| 22:09.53 | wunderkin | brettnem: am i supposted to go from the patch panel to the t1 card and then have something stick off of that to force it to loop? |
| 22:10.50 | wunderkin | justinu, yeah well im pretty sure thats probably what the colo is using |
| 22:11.35 | wunderkin | i think maybe i get the picture |
| 22:11.59 | *** part/#asterisk rculp (n=rculp@66.173.240.20) |
| 22:12.18 | *** join/#asterisk br00ksh1r3 (n=matt@wsip-24-120-36-162.lv.lv.cox.net) |
| 22:12.30 | kn0x | ztdummy isnt working... heres my dmesg output, it looks strange.... http://pastebin.ca/28717 |
| 22:12.51 | wunderkin | justinu, do you have any idea about this female loopback? |
| 22:13.04 | justinu | never heard of it |
| 22:13.10 | wunderkin | its just forcing it to loop right? |
| 22:13.17 | justinu | sounds like a hard loop plug, or something |
| 22:13.23 | nick125 | kn0x: what kernel? |
| 22:13.23 | X-Files | Please need help !!! I use gateway EUSSO UTG7104 Protocol SIP , i have problem :( please check : http://pastebin.com/429483 |
| 22:13.42 | wunderkin | i think something like that, im wondering can i just switch the pins on the cable from the patch panel to the card? |
| 22:13.53 | kn0x | sorry, fixed |
| 22:13.54 | kn0x | http://bugs.digium.com/view.php?id=5236&nbn=14 |
| 22:14.00 | wunderkin | all it does is switch tx and rx right? |
| 22:14.05 | kn0x | nick125- 2.6.13 |
| 22:14.05 | wunderkin | or is there something more |
| 22:14.08 | kn0x | my bad |
| 22:14.18 | nick125 | aah |
| 22:14.23 | justinu | wunderkin: yeah |
| 22:14.45 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 22:14.45 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 22:15.06 | wunderkin | so if i make a cable with a normal end that plugs into the patch panel and then switch rx and tx pairs going to the card, that should work? |
| 22:15.11 | wunderkin | i mean, for a loop |
| 22:15.23 | *** join/#asterisk br00ksh1r3 (n=matt@wsip-24-120-36-162.lv.lv.cox.net) |
| 22:15.28 | wunderkin | but then again its not using the original cable but at least it gets me to the card |
| 22:15.31 | twisted[mobile] | wheee |
| 22:15.34 | twisted[mobile] | vegas is HOT |
| 22:15.38 | br00ksh1r3 | heck yeah it is |
| 22:15.40 | twisted[mobile] | and I don't mean HOT like paris hilton hot |
| 22:15.42 | twisted[mobile] | i mean HOT |
| 22:15.47 | br00ksh1r3 | it's like 100 |
| 22:16.10 | Igbothom_III | Paris Hilton is not hot |
| 22:16.14 | Igbothom_III | slutty, maybe |
| 22:16.15 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 22:16.18 | Igbothom_III | but NOT hot |
| 22:16.34 | brettnem | wunderkin: I was just suggesting that 1: the loopback plug should be placed as close to the equipment as possible and 2: testing for errors is only something you can do with a RECIEVER.. unless the far end is sending PM or in Loop |
| 22:17.07 | wunderkin | brettnem: umm well yeah they are doing something on their side and telling me to loop my equipment |
| 22:17.39 | justinu | there are such things as loopup/down codes |
| 22:18.10 | wunderkin | yeah, well i must not be sending the right ones.. it says its looping it up but they dont see anything |
| 22:18.20 | justinu | i've rarely gotten them to work myself |
| 22:18.30 | wunderkin | shitty |
| 22:18.56 | justinu | what you need, is a sunset t1 |
| 22:19.02 | wunderkin | yeah, well |
| 22:19.14 | wunderkin | i was hoping qwest would stay long enough for me to see if they could test it to my side |
| 22:19.24 | wunderkin | they were there when i got there but not by the time i could get to find them :P |
| 22:19.31 | justinu | yeah, how uncool of them to bail |
| 22:19.43 | wunderkin | they have tested before from the niu to me and said it was ok |
| 22:20.09 | wunderkin | so well that would test the 1 pair at least |
| 22:20.38 | wunderkin | too bad kpfleming doesn't have one :( |
| 22:20.43 | justinu | i had a bitch of a problem with a T1 once that ended up being a bad wirewrap on the carriers DSX panel |
| 22:20.57 | justinu | took months to sort that out |
| 22:21.15 | wunderkin | yeah i imagine since its on their side |
| 22:21.17 | justinu | they never found it because when they went to test, they plugged in AT the DSX panel, which isolated that wirewrap |
| 22:21.28 | justinu | so all their testing was always clean |
| 22:21.32 | wunderkin | yup |
| 22:21.45 | justinu | you may even have a similar situation |
| 22:21.58 | wunderkin | i didn't think about that |
| 22:22.05 | wunderkin | i'm not sure where they are testing it from on the other end |
| 22:22.13 | justinu | if they can loop from their switch framer to you |
| 22:22.30 | wunderkin | i know they said they were using test equipment but i dont know where |
| 22:22.32 | justinu | and then loop from their dsx to you |
| 22:22.36 | justinu | you could isolate it maybe |
| 22:23.29 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:23.30 | wunderkin | i probably should get the notes of what all has been done, i almost was at that point before and they finally figured it out |
| 22:23.35 | justinu | yep |
| 22:23.52 | *** join/#asterisk shmaltz (n=chatzill@69.28.255.210) |
| 22:24.17 | X-Files | People, are an opportunity that Internet Telephony Gatewey it was possible authorization of number of ports and devices that others could not will connect without the password or something such like.. Protocol H323 ? |
| 22:24.43 | shmaltz | brettnem, according to cisco it's configurable, so I was able to solve it, in SIPDefault.cnf one has to put in: |
| 22:24.44 | shmaltz | # Allow for the bridge on a 3way call to join remaining parties upon hangup |
| 22:24.46 | shmaltz | cnf_join_enable: 1; 0-Disabled, 1-Enabled (default) |
| 22:25.08 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 22:25.48 | kn0x | do i need GRE tunnels over IP enabled for asterisk? |
| 22:25.59 | shmaltz | kn0x, to do what? |
| 22:25.59 | Math` | no |
| 22:26.13 | Math` | except if you want to tunnel to the LAN using asterisk using GRE tunnels |
| 22:26.33 | wunderkin | justinu, i'm not sure if you said.. what about my cabling idea, will that work? |
| 22:26.42 | justinu | i dunno if i saw that |
| 22:27.17 | wunderkin | so if i make a cable with a normal end that plugs into the patch panel and then switch rx and tx pairs going to the card, that should work for a loop <-- |
| 22:27.32 | X-Files | grrr :( |
| 22:27.41 | shmaltz | wunderkin, why not? |
| 22:27.51 | wunderkin | i guess, just never tried it |
| 22:28.15 | shmaltz | wunderkin, in fact you should only do it on *one* single point in the run like this: |
| 22:28.33 | wunderkin | yeah thats what i mean |
| 22:28.41 | X-Files | People, are an opportunity that Internet Telephony Gatewey it was possible authorization of number of ports and devices that others could not will connect without the password or something such like.. Protocol H323 ? It is possible? |
| 22:28.49 | shmaltz | ___________/------------------------------------ |
| 22:28.51 | shmaltz | -----------------/_______________________ |
| 22:28.57 | shmaltz | and not like this: |
| 22:29.32 | shmaltz | _________________________/--------------------------\_________________________ |
| 22:29.33 | shmaltz | --------------------------------------\_________________/--------------------------- |
| 22:29.48 | shmaltz | because then you end up with the same pairs at the card |
| 22:30.19 | Math` | thats some ascii-art talent |
| 22:30.21 | shmaltz | and from a wiring point of view, you nvever do it on the patch panel you always do it on the patch cable |
| 22:30.27 | shmaltz | Math lol |
| 22:30.29 | wunderkin | the cable yes |
| 22:31.06 | wunderkin | i mean i make the cable, normal end and then on the other end of the cable reversed |
| 22:31.18 | shmaltz | yeah |
| 22:31.27 | Math` | what kind of wiring does a T1 line use? |
| 22:31.35 | *** join/#asterisk xphreakster (n=xphreaks@ns1.zrlocal.net) |
| 22:31.37 | wunderkin | that should be the same as using the loopback plug but its going into the card |
| 22:32.12 | shmaltz | Math rj45 |
| 22:32.27 | shmaltz | but tq uses pair 1 and 3 |
| 22:32.37 | shmaltz | while ethernet uses pair 2 and 3 |
| 22:32.45 | Druken | mmmmm, pizza |
| 22:32.46 | shmaltz | tq=t1 |
| 22:32.57 | justinu | wunderkin: yeah |
| 22:33.09 | xphreakster | HI everybody |
| 22:33.09 | xphreakster | I have installed ztdummy for kernel 2.6, and the module is loaded and also the zaptel module |
| 22:33.09 | xphreakster | When I try to use the MeetMe application in the dialplan I get the following error: |
| 22:33.09 | xphreakster | No application 'MeetMe' for extension |
| 22:33.09 | xphreakster | What am I doing WRONG ???? |
| 22:33.39 | ManxPower | xphreakster, zaptel must be installed or Asterisk won't build the MeetMe Application when you compile Asterisk |
| 22:33.42 | shmaltz | xphreakster, first no flooding, second looks like a timer issue |
| 22:33.42 | justinu | math: t1 uses 1,2 + 4,5 |
| 22:33.44 | wunderkin | justinu, so that should be a good test too just that its not using the original cable |
| 22:33.54 | ManxPower | Since you must have zaptel timer in order to use meetme |
| 22:34.13 | justinu | wunderkin: so you're gonna make a your own custom LB plug by somehow crossing the tx and rx pairs? |
| 22:34.32 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:34.37 | fugitivo | xphreakster: did you edit /etc/asterisk/meetme.conf? |
| 22:34.56 | xphreakster | fugitivo: yep |
| 22:35.13 | xphreakster | I see so first I have to install the zaptel and then to install asterisk ?? |
| 22:35.18 | fugitivo | yes |
| 22:35.29 | xphreakster | I had done it vice versa :( |
| 22:36.00 | xphreakster | is it possible to recompile asterisk again, and reinstall it again over the current version ???? |
| 22:36.11 | shmaltz | Math, the wiki got some good links for the wiring thingy |
| 22:36.14 | Math` | cat5e is ok too? |
| 22:36.14 | xphreakster | will it then compile the MeetMe to be usable ? |
| 22:36.21 | wunderkin | justinu, yeah i wanted to test it all the way to the card so i wanted to make a loop cable to it instead.. use a normal end from the patch panel and then on the other end that goes into the card - reverse the pairs |
| 22:36.34 | Math` | Katty: its 1.0.9 |
| 22:36.38 | Katty | eww |
| 22:36.40 | shmaltz | xphreakster, yeah just run make clean then make then make install |
| 22:36.50 | shmaltz | katty, nope |
| 22:36.57 | Katty | pity |
| 22:37.23 | xphreakster | thanks guys for your help, you are lifesavers :) |
| 22:37.23 | shmaltz | math even cat3 is |
| 22:37.28 | Math` | ok |
| 22:37.30 | *** join/#asterisk enemy^x (i=lkqw@212.62.250.98) |
| 22:37.31 | shmaltz | cat3 was meant for t1 |
| 22:37.51 | shmaltz | xphreakster, tell that my wife :) |
| 22:38.06 | justinu | i always preferred the rum flavored ones |
| 22:38.17 | Katty | hmm, rum |
| 22:38.19 | InfraRed | rum flavored T1ws? |
| 22:38.21 | xphreakster | :) |
| 22:38.23 | InfraRed | T1s |
| 22:38.30 | Katty | hmmmmmmmmmmmmmmmmmmmmm, rum. |
| 22:38.41 | justinu | wunderkin: if you can make that work, it should be a valid test |
| 22:38.47 | shmaltz | InfraRed, yeah you need some color coding system in the wiring closet |
| 22:38.53 | shmaltz | why not use a flavor |
| 22:38.53 | wunderkin | justinu, well thats why im asking if it would work ;) |
| 22:39.14 | justinu | yeah, it all comes down to your ability to construct a solid LB plug :) |
| 22:39.26 | wunderkin | justinu, well i can make cables :D |
| 22:39.31 | Druken | shmaltz: would you stick the wires in the closet in your mouth ?? |
| 22:39.33 | Druken | i know i wouldn't |
| 22:39.50 | shmaltz | Druken after a bottle of rum, maybe |
| 22:39.58 | Druken | :) |
| 22:40.14 | shmaltz | justinu, I believe you can buy couplers that do that for you |
| 22:40.31 | justinu | yeah, i have a few lying around |
| 22:40.58 | justinu | sunset t1, lb plugs, bantam cables, etc, etc, etc. |
| 22:41.11 | justinu | hehe |
| 22:41.31 | justinu | i have a sunset t3 also |
| 22:41.34 | justinu | does t1 and t3 testing |
| 22:41.44 | wunderkin | i would need to borrow you too since i wouldn't know what to do with it all :P |
| 22:41.50 | justinu | the t1 is cooler because it has 2 t1 interfaces |
| 22:41.55 | justinu | you can do drop and insert |
| 22:41.57 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:42.43 | shmaltz | gtg guys |
| 22:42.46 | shmaltz | c ya |
| 22:42.47 | wunderkin | cya |
| 22:42.47 | justinu | wunderkin: you are starting to see why i'm just sick of dealing with TDM |
| 22:42.48 | shmaltz | bye |
| 22:43.04 | wunderkin | justinu, so you don't need that equipment anymore? :D |
| 22:43.22 | justinu | hehe |
| 22:43.30 | justinu | belongs to the company I work for anyways :) |
| 22:43.33 | wunderkin | ah |
| 22:44.22 | *** join/#asterisk mashedpotats (n=potats@pool-151-203-73-60.bos.east.verizon.net) |
| 22:44.22 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:44.22 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 22:44.28 | justinu | i'd beg qwest to come out again and force them to stay until they test the CPE cabling |
| 22:44.54 | wunderkin | right... keep dreaming |
| 22:45.06 | justinu | why not? |
| 22:45.18 | Druken | hold them hostage |
| 22:45.20 | justinu | most of those hi-cap repair techs are nice guys |
| 22:45.24 | justinu | they've always helped me out |
| 22:45.27 | asterboy | money |
| 22:45.40 | Flauto | i agree with igb |
| 22:45.41 | asterboy | why else? |
| 22:45.50 | *** join/#asterisk Math[laptop] (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 22:45.57 | asterboy | lol |
| 22:46.16 | asterboy | I need a dime...payphones are .35c |
| 22:46.23 | *** join/#asterisk bkw_ (n=bkw_@adsl-69-155-20-23.dsl.tulsok.swbell.net) |
| 22:46.24 | *** join/#asterisk hhoffman (n=hhoffman@tor/session/x-2271902f93fe75f2) |
| 22:46.25 | Math[laptop] | ast_expr2.c:1454: undefined reference to `__builtin_stpcpy' <-- anyone had that with head? |
| 22:46.28 | Druken | cheap bastards |
| 22:46.33 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 22:46.43 | *** part/#asterisk bkw_ (n=bkw_@adsl-69-155-20-23.dsl.tulsok.swbell.net) |
| 22:46.47 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:46.50 | *** join/#asterisk nagl (n=nagl@213.235.241.6) |
| 22:46.53 | Katty | yay, Ariel_ |
| 22:46.56 | hhoffman | hi, anyone know how to keep asterisk from answering the PSTN line before X rings? |
| 22:47.02 | IronHelix | sure |
| 22:47.09 | Ariel_ | Katty, hugs hope your doing well. |
| 22:47.10 | IronHelix | in extensions.conf dont have answer() as your first priority |
| 22:47.15 | Katty | Ariel_: yes, thanks (= |
| 22:47.16 | IronHelix | or anything else that will cause it to answer |
| 22:47.22 | Druken | hhoffman: can't be done |
| 22:47.27 | IronHelix | use Wait(x) with x being the number of seconds to wait |
| 22:47.34 | Ariel_ | hhoffman, wait(10) |
| 22:47.36 | hhoffman | IronHelix: thx :-) |
| 22:47.39 | IronHelix | np |
| 22:49.30 | Druken | asterboy: get my email ? |
| 22:49.43 | Math[laptop] | asterboy, payphones @ 0.35$? where ? |
| 22:50.04 | asterboy | in hicks ville, Alberta |
| 22:50.30 | kink0 | why I hear nothing while Executing MusicOnHold("OSS/dsp", "Santana") in new stack ? |
| 22:51.19 | Math[laptop] | asterboy, telus? |
| 22:51.23 | asterboy | yes |
| 22:51.31 | Math[laptop] | rippers lol |
| 22:51.42 | Druken | but i do like their cell service... :) |
| 22:51.43 | asterboy | unless they have raised the rate...its been a while since I looked. |
| 22:51.59 | Math[laptop] | payphones always were 0.25 iirc |
| 22:52.01 | asterboy | Yes, Telus here are thieves! |
| 22:52.11 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 22:52.11 | ManxPower | kink0, you have a MoH class called "Santana" (in that case)? |
| 22:52.25 | *** join/#asterisk dalabera (n=dalabera@146.82.190.164) |
| 22:52.25 | asterboy | I have a client with $1000 phone bill . |
| 22:52.41 | ManxPower | asterboy, Write more secure dialplans 8-) |
| 22:52.41 | Math[laptop] | asterboy, I have a client with a 10 000$ bill for bandwidth quota exceeding |
| 22:52.42 | asterboy | They try to get away with everything. |
| 22:52.47 | Math[laptop] | with bell business internet |
| 22:52.49 | Druken | i really hope that's not for a month |
| 22:52.55 | Katty | Math[laptop]: i keep mistaking you for MikeJ[Laptop] |
| 22:52.58 | asterboy | yes for 1 month! |
| 22:53.00 | Math[laptop] | 20$/gig, no limit! |
| 22:53.04 | Math[laptop] | Katty, lol |
| 22:53.14 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 22:53.17 | asterboy | That's exactly the bull Telus will pull. |
| 22:53.17 | Katty | lolzkthxbi |
| 22:53.18 | Math[laptop] | Katty, file was file[laptop] too before coming back from mtl :P |
| 22:53.24 | asterboy | They stole $5000 from me. |
| 22:53.26 | Katty | yay, `Sauron |
| 22:53.28 | ManxPower | Gads, even when I was using a hosted unified messaging service, my phone bills were never more than $400 in a month |
| 22:53.30 | Druken | Math[laptop]: wholly shit.... |
| 22:53.36 | `Sauron | Katty: wot? |
| 22:53.37 | Math[laptop] | asterboy, I hope you're not using them anymore :P |
| 22:53.41 | kink0 | ManxPower, yes !! Santana => quietmp3:/var/lib/asterisk/moh/Santana |
| 22:53.43 | Katty | `Sauron: allo. |
| 22:53.51 | asterboy | Ripped the wires from my house! |
| 22:53.57 | `Sauron | Ah, allo |
| 22:54.00 | kink0 | and I have one mp3 under Santana folder |
| 22:54.00 | ManxPower | kink0, do other MOH classes work? |
| 22:54.03 | asterboy | Now I undercut them everywhere I go. |
| 22:54.15 | asterboy | I'm slashing and cutting until they bleed. |
| 22:54.18 | Math[laptop] | hehe |
| 22:54.34 | kink0 | ManxPower, no, no one. I am playing changing fews lines at musiconhold.conf and extensions.conf |
| 22:54.40 | Druken | touchy... :) |
| 22:54.42 | asterboy | I pulled an ad in the yellow pages for the $5000 they owe me...lets see them try to get paid! |
| 22:54.54 | Math[laptop] | lol |
| 22:55.37 | ManxPower | kink0, and your "mpg123 -v" shows 0.59r |
| 22:56.05 | kink0 | Version 0.59q (2002/03/23 |
| 22:56.09 | kink0 | q instead r |
| 22:56.23 | ManxPower | kink0, it needs to be r |
| 22:56.43 | kink0 | ManxPower, ahhh ok, just r or may be newest ? |
| 22:56.48 | ManxPower | kink0, not newest, r |
| 22:57.08 | ManxPower | In fact, it's SO important that you can do a "make mpg123" in the asterisk source dir and it will download the right version and compile it |
| 22:57.09 | Math[laptop] | kink0, make mpg123 will get the right one |
| 22:57.36 | kink0 | ok, I will search now for 59r version |
| 22:57.47 | Math[laptop] | don't search for it... just make mpg123 |
| 22:59.34 | asterboy | ya business in Alberta is cut throat...if you can make it here...you can make it anywhere. |
| 23:00.03 | ManxPower | I prefer Louisiana - everyone is so tech.stupid you don't have any real competition |
| 23:00.29 | asterboy | lol |
| 23:00.42 | *** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net) |
| 23:01.26 | asterboy | It does amaze me how many businesses here are still paying $10,000s of dollars in phone bills when we can do it for $100s |
| 23:01.48 | Druken | well, lets not get too carried away... :) |
| 23:01.55 | Druken | say 1000's.... |
| 23:02.01 | hypa7ia | asterboy: you've never worked on a cisco install have you :( |
| 23:02.01 | asterboy | no 10,000 |
| 23:02.04 | *** part/#asterisk mkrufky (n=mk@68.160.103.77) |
| 23:02.12 | asterboy | you don't know Telus very well. |
| 23:02.30 | Druken | i'm not saying their cost... i'm, saying what you could cut it down to... |
| 23:02.39 | asterboy | nope, no cisco...too expensive for what a linux box can do better. |
| 23:02.41 | hypa7ia | asterboy: where in AB are you? |
| 23:03.00 | asterboy | Druken: yes true. |
| 23:03.03 | hypa7ia | i put in a big callmanager in calgary |
| 23:03.10 | hypa7ia | this spring |
| 23:03.22 | asterboy | Red Deer |
| 23:03.49 | ManxPower | The halmark of a young newbie is that they think Linux makes a good router for corporate use. |
| 23:04.10 | ManxPower | Cisco is not all that expensive if you do it right. |
| 23:04.39 | hypa7ia | ManxPower: i should have beenmore clear, wasn't talking cisco routing and switching |
| 23:04.44 | asterboy | I've been using Linux for corporate use for the longest time...no probs. |
| 23:05.01 | hypa7ia | ManxPower: their IP tel stuff is expensive no matter how you slice it |
| 23:05.27 | asterboy | I'm not talkin...Ubuntu here. |
| 23:05.30 | Druken | asterboy: when i say corp i'm talking about 1000+ clients, and servers behind... with more than a c block to it's network... |
| 23:05.34 | ManxPower | hypa7ia, We install Cisco 550x switches CHEAP. We use Cisco 2621 routers, usually from eBay if it's not an emergency. |
| 23:05.45 | ManxPower | hypa7ia, Yes, Cisco's telephony suff is expensive |
| 23:05.54 | hypa7ia | ManxPower: callmanager software alone if like 10K CAD |
| 23:06.01 | hypa7ia | s/if/is |
| 23:06.13 | hypa7ia | it's insane :( |
| 23:06.15 | myke420247 | no way, you can route all that on an old 486/120 with 3 isa nics |
| 23:06.21 | myke420247 | it works fine at my house |
| 23:06.24 | asterboy | lol |
| 23:06.26 | myke420247 | therefore it will work anywhere |
| 23:06.43 | *** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com) |
| 23:06.49 | asterboy | there is something to be said for the "appliance" |
| 23:07.14 | kink0 | ManxPower, just compiled and install version 59r but still not sounds !! |
| 23:07.33 | asterboy | I'd love to take on 1000+ clients with a linux router to see how it compares. |
| 23:07.37 | myke420247 | kink0, make sure you either have working zaptel hw or the ztdummy driver |
| 23:07.42 | hypa7ia | asterboy: callmanager is so far from an appliance it's not even funny |
| 23:07.48 | myke420247 | it's not documented but moh won't work w/o a timing source |
| 23:07.57 | kink0 | myke420247, no, I am just ussing my soundcard |
| 23:07.58 | hypa7ia | it runs on win2k and everything |
| 23:08.07 | kink0 | mpg123 file.mp3 runs fine |
| 23:08.07 | asterboy | interesting |
| 23:08.13 | myke420247 | kink0, then you need ztdummy properly installed |
| 23:08.37 | Druken | asterboy: just make sure that's a fast linuxbox and has a few gig of ram :) |
| 23:09.14 | Druken | myke420247: it'll work... but it'll sound like shit.... |
| 23:09.16 | myke420247 | i had the same problem then i redid my * setup following along the asterisk book which has you install a bunch of stuff like that, then moh worked |
| 23:09.30 | myke420247 | it sounds fine on my setup |
| 23:09.32 | asterboy | multi-processor |
| 23:09.39 | ManxPower | myke420247, Yes, MoH will work without a timing source. |
| 23:09.49 | asterboy | Think about the processing power you can buy for cheap! |
| 23:09.51 | myke420247 | before i had ztdummy it kept halting |
| 23:09.51 | ManxPower | kink0, Asterisk can fail killing MoH |
| 23:10.28 | ManxPower | kink0, stop asterisk, kill all instances of mpg123 (ps -ax), make sure the old version of mpg123 is not on the system |
| 23:10.35 | kink0 | ManxPower, but I kill asterisk and restart it, then I try with "dial 2" where I have set the musiconhold |
| 23:10.40 | ManxPower | myke420247, that can be an issue in underpowered systems. |
| 23:11.10 | kink0 | ManxPower, yes, I have also deleted the prior version of mpg123, and new mpg123 is in the path |
| 23:11.47 | ManxPower | kink0, dial an extension that runs MusicOnHold(yourmohclass) |
| 23:13.00 | kink0 | yes, exten => 2,1,Answer() |
| 23:13.00 | kink0 | exten => 2,2,MusicOnHold(Santana) |
| 23:13.21 | kink0 | and I got the message as was playing, but not sounds |
| 23:13.22 | ManxPower | kink0, what's exten =>2,1 |
| 23:13.28 | kink0 | xecuting MusicOnHold("OSS/dsp", "Santana") in new stack |
| 23:13.29 | ManxPower | Ah. |
| 23:13.46 | ManxPower | kink0, and you confirmed there were NO stray instances of mpg123 running when you stop asterisk? |
| 23:14.16 | myke420247 | manx, same system, it's a dell sc420 with no other calls so that wasn't it |
| 23:14.54 | asterboy | Gotta like this page: http://www.freesco.org/ |
| 23:15.01 | myke420247 | kink0, try with the default moh sounds first |
| 23:15.05 | asterboy | although no match for the high end stuff. |
| 23:15.41 | justinu | asterboy: you should run some arcnet |
| 23:15.45 | justinu | very l337 stuff |
| 23:16.10 | asterboy | arcnet? Man I have not heard that for a long time. |
| 23:16.11 | myke420247 | i had a whole cluster working on tokenring a while back |
| 23:16.21 | justinu | that freesco.org page mentions arcnet |
| 23:16.23 | myke420247 | great to keep randoms from plugging their laptops into your lan |
| 23:16.26 | justinu | made me chuckle |
| 23:16.32 | asterboy | true |
| 23:16.39 | Druken | ok... who called me from cooksville? |
| 23:16.51 | asterboy | that protocol had its virtues. |
| 23:17.01 | justinu | myke420247: was that the 4mbps or 16mbps variant of TR? :) |
| 23:17.05 | myke420247 | 16 |
| 23:17.11 | myke420247 | i'm not *that* retro |
| 23:17.13 | justinu | lol |
| 23:17.17 | tzanger | haha |
| 23:17.26 | myke420247 | this was for a public cluster at a "hacker" con, so requirements were a bit different from your typical home or corporate setup |
| 23:17.31 | tzanger | you've got a PCMCIA token ring card? |
| 23:17.33 | justinu | that's kinda fun |
| 23:17.34 | myke420247 | i did |
| 23:17.35 | asterboy | more : http://www.imagestream.com/Cisco_Comparison.html |
| 23:17.37 | myke420247 | they're cheap on ebay |
| 23:17.41 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:17.41 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:17.45 | myke420247 | i got rid of all that crap at the end of the con tho |
| 23:17.52 | myke420247 | it's so cheap you can give it away and not care |
| 23:17.54 | kink0 | no, there no any other mpg123 running |
| 23:17.57 | kn0x | i cant get any sip peers to register |
| 23:18.00 | kn0x | chan_sip.c:1045 __sip_xmit: sip_xmit of 0x812fb28 (len 418) to 204.147.183.18:-1 returned 5060: Bad file descriptor Nov 14 11:17:30 NOTICE[7156]: chan_sip.c:5247 sip_reg_timeout: -- Registration for '498897@fwd.pulver.com' timed out, trying again (Attempt #2) |
| 23:18.00 | myke420247 | sure didn't want to ship it back to my house |
| 23:18.03 | justinu | TR was pretty cool when they used those bug ass IBM modular plugs |
| 23:18.07 | justinu | s/bug/big |
| 23:18.13 | kn0x | whats that all about |
| 23:18.17 | myke420247 | yeah ibm structured cabling |
| 23:18.25 | myke420247 | nice and expensive |
| 23:18.27 | tzanger | yeah |
| 23:18.44 | myke420247 | i had an application server, and a bunch of 486/25's as graphical dumb terminals |
| 23:18.47 | myke420247 | "thin clients" |
| 23:19.00 | myke420247 | worked good, and no jerks arp spoofing the router or anything |
| 23:19.04 | justinu | heh |
| 23:19.14 | tzanger | token ring was nice because you could actually hit its limits |
| 23:19.29 | justinu | ATM is next on the list of dead networking protocols |
| 23:19.35 | tzanger | doubtful |
| 23:19.36 | myke420247 | hmm yeah\ |
| 23:19.41 | tecnico | very out of topic.. but I'm desperately looking for an Actel ProAsic FPGA programmer box. Anyone here (and in the U.S.) that has one? |
| 23:19.45 | *** join/#asterisk Anthro (n=keljsrh@pdpc/supporter/active/Anthro) |
| 23:19.45 | tzanger | it runs pretty much every single ADSL rollout in north america |
| 23:19.49 | myke420247 | maybe for the next con i'll pick up a bunch of desktop atm adapters |
| 23:19.56 | LostFrog | Hmm.. no way to look at globalvars from the CLI? |
| 23:19.59 | justinu | yeah, I know that, but can you think of any corporate user that runs ATM seriously anymore? |
| 23:20.02 | myke420247 | tho i don't think i can get those for $2 a pop like i can tokenring nics |
| 23:20.18 | tzanger | oh on a LAN? no |
| 23:20.27 | justinu | talk about a good idea, but ruined by complexity |
| 23:20.29 | justinu | that's ATM |
| 23:20.35 | tzanger | no |
| 23:20.38 | tzanger | ATM sucked from the start |
| 23:20.42 | tzanger | faaaaaaaaaaaaar too much overhead |
| 23:20.48 | Anthro | Nifty, though. |
| 23:21.01 | *** part/#asterisk xphreakster (n=xphreaks@ns1.zrlocal.net) |
| 23:21.23 | tzanger | heh |
| 23:21.26 | myke420247 | atm didn't exist when i was in college |
| 23:21.27 | tzanger | and then over your DSL connection |
| 23:21.30 | kink0 | I see I have |
| 23:21.31 | tzanger | ATM over UDP over ATM |
| 23:21.32 | kink0 | root 21412 0.0 0.0 3884 604 pts/9 S+ 00:18 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 Carlos_Santana-Jam_With_Santana-14-Europa_(FT).mp3 |
| 23:21.35 | myke420247 | tho FORE was founded from grads from my college |
| 23:21.36 | kink0 | but not sound !! |
| 23:21.37 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 23:21.39 | *** join/#asterisk warthog (n=nvadekar@216.249.38.137.ppp.northrock.bm) |
| 23:21.59 | kink0 | and asterisk reports at CLI Executing MusicOnHold("OSS/dsp", "") in new stack |
| 23:22.08 | warthog | anyone know how to check the version of zaptel on a running machine? |
| 23:22.20 | LostFrog | Hmm.. there is an alisson sound for 'day', but not 'night'. |
| 23:22.24 | asterboy | Cisco does not guarantee any of its specifications...these guys do for 31 days. |
| 23:22.31 | tzanger | LostFrog: so just have her say "not day" :-) |
| 23:22.32 | justinu | i'm getting annoying noise in my hold music running format_mp3 |
| 23:22.37 | justinu | should I not use that? |
| 23:22.43 | LostFrog | tzanger: My users would flip. :) |
| 23:23.03 | tzanger | ehehe |
| 23:23.05 | asterboy | http://www.imagestream.com/Cisco_Comparison.html#4 |
| 23:23.20 | Anthro | I am just getting started with Asterisk. I am debating whether it is important enough to use a recent version or not. The reason this comes up is that I was planning on using Debian stable (rather than testing or unstable) if I could get away with it. Any thoughts? |
| 23:23.24 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:24.18 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 23:24.18 | tzanger | Anthro: use 1.2rc then |
| 23:24.23 | LostFrog | No simple 'not' sound either. |
| 23:24.39 | LostFrog | Debian 3.1 works fine for me. |
| 23:24.59 | Druken | anyone know why voicemail has a limit of 100 messages? |
| 23:25.00 | tzanger | I really don't recommend 1.0.x |
| 23:25.11 | Anthro | LostFrog: You are using 1.0.7? |
| 23:25.12 | kn0x | chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81a3c28 (len 418) to 204.147.183.18:-1 returned 5060: Bad file descriptor |
| 23:25.15 | justinu | Druken: that's a hard coded limit |
| 23:25.19 | LostFrog | 1.2RC2 |
| 23:25.22 | LostFrog | I like to bleed. |
| 23:25.26 | kn0x | none of my sup peers will register |
| 23:25.29 | Druken | justinu: ok, but why? :) |
| 23:25.32 | Anthro | tzanger: What are the advantages of 1.2 over 1.0? |
| 23:25.50 | justinu | Druken: dunno, but just increase the constant :) |
| 23:25.57 | tzanger | Anthro: it's current code |
| 23:26.04 | tzanger | TONS of featurs and fixes over 1.0.x |
| 23:26.26 | Anthro | I don't suppose there's a readable (i.e. comprehensible) changelog somewhere? |
| 23:26.29 | Druken | justinu: not that i'd have 100 voicemail unless something fucks up... :) |
| 23:26.32 | *** join/#asterisk SplasPood (n=sp@paravolve.net) |
| 23:26.34 | tzanger | Anthro: yes, download 1.2 |
| 23:26.40 | LostFrog | Grr.. I just want it to say whether it's in night or day mode. |
| 23:26.50 | denon | Druken: that limit is set in voicemail.conf now |
| 23:26.52 | justinu | Druken: some lusers actually leave that much crap in their mailbox |
| 23:26.56 | denon | look at the sample |
| 23:27.09 | denon | I hit that limit all the time, but I believe its now configurable |
| 23:27.18 | Druken | justinu: my wife would.... i swear she doesn't know the the number 7 is... |
| 23:27.27 | justinu | i knew a guy who saved all his vm messages |
| 23:27.29 | justinu | weirdo |
| 23:27.37 | LostFrog | I save most of my e-mail. |
| 23:27.37 | Druken | agreed |
| 23:27.43 | justinu | email is a bit different |
| 23:27.45 | ManxPower | Anthro, read the UPGRADE.txt and Changelog of 1.2 to see the changes |
| 23:27.46 | LostFrog | I call it CYA. |
| 23:27.56 | Anthro | tzanger: Any guess on when 1.2 will be released instead of merely a release candidate? |
| 23:28.00 | tzanger | Anthro: soon |
| 23:28.12 | justinu | this guy had stacks of cassette tapes with answering machine messages on them |
| 23:28.16 | myke420247 | 1.2rc2 works fine for me |
| 23:28.23 | Anthro | tzanger: Hrm. How different is the configuration? |
| 23:28.25 | ManxPower | Anthro, at this point there will only be bug fixes before release. I'm running RC2 on 1 of my servers and it seems to work pretty good. |
| 23:28.28 | Anthro | ManxPower: URL? |
| 23:28.35 | LostFrog | Dang.. there is an 'on' but no 'off' |
| 23:28.45 | ManxPower | Anthro, download the source code for 1.2, then read those files |
| 23:28.46 | tzanger | Anthro: you want 1.2rc2. just trust us on this |
| 23:28.53 | Druken | justinu: that is just strange.... i know i have an option to monitor incoming calls... but that is for a purpose |
| 23:28.54 | warthog | anyone know how to determine the version of zaptel on a running machine? |
| 23:29.05 | ManxPower | warthog, not that I am aware of |
| 23:29.06 | tzanger | Anthro: I know you're a little nervous but we're all here to help |
| 23:29.49 | LostFrog | You would think modinfo zaptel would tell you. :) |
| 23:29.50 | Anthro | tzanger: Actually, it's more about laziness. 1.2rc2 isn't packaged for Debian AFAIK. I like having someone else package my software. |
| 23:30.02 | justinu | that's the wrong attitude to have around here |
| 23:30.03 | tzanger | you do NOT want to run Debian's packages |
| 23:30.06 | tzanger | they are junk |
| 23:30.09 | ManxPower | Anthro, Asterisk is the ONLY package I don't use packages for. |
| 23:30.14 | LostFrog | Anthro: just compile *. it's simple. |
| 23:30.30 | LostFrog | As a matter of fact, my gf could do it, Anthro. |
| 23:30.47 | Math[laptop] | mine too |
| 23:30.58 | justinu | not mine |
| 23:31.03 | Math[laptop] | (well if I dictate her what to type) |
| 23:31.10 | Anthro | Oh, I can compile. Hell, I compiled ircii back in the day, and xpilot, and innumerable other pre-web entertainments. I just... got bored with it. |
| 23:31.15 | Callum | c0w, are you still about ? |
| 23:31.20 | Math[laptop] | good |
| 23:31.42 | Anthro | All right, I'm convinced. I'll grab the code and install in /usr/local. |
| 23:31.59 | LostFrog | There is no version string in zaptel that I can find. |
| 23:32.10 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 23:32.10 | ManxPower | Anthro, I understand system admin lazyness. I have an extreme case of it, but I still download and install Asterisk by hand |
| 23:32.37 | LostFrog | I love apt-get. |
| 23:32.53 | Anthro | ManxPower: Okay. Well, here goes. Not that it will be speedy. VIA chip and a 5400 RPM disk. |
| 23:33.28 | warthog | perhaps someone who know someone the halls of power can suggest that zaptel version info be placed in modinfo!! |
| 23:33.34 | Anthro | LostFrog: Likewise |
| 23:33.46 | ManxPower | Anthro, the people working on Asterisk are brilliant programmers, but they suck when it comes to portable code and portable Makefiles and relocation of Asterisk's config and binary files. |
| 23:34.23 | *** join/#asterisk in-side (n=lowgitek@es-217-129-27-34.netvisao.pt) |
| 23:34.29 | Anthro | ManxPower: Uhhhh... that doesn't bode well. Where does it want to install itself? |
| 23:34.35 | in-side | Hi |
| 23:34.40 | *** join/#asterisk fafnir (n=hello@tdds-gw.Moscow.gldn.net) |
| 23:34.44 | in-side | anybody here uses freebsd? |
| 23:34.55 | in-side | I have a weirdo problem with my fbsd 6.0 |
| 23:34.55 | ManxPower | Anthro, traditional places /etc/asterisk /usr/bin /usr/lib/asterisk /var/spool/asterisk, etc |
| 23:34.58 | tzanger | /usr/sbin/asterisk and a couple other apps htere, and then /usr/lib/asterisk |
| 23:35.07 | tzanger | oh and /etc/asterisk and /var/spool asterisk, as ManxPower says |
| 23:35.11 | warthog | can anyone tell me if 1.2 of zaptel is better at echo management than 1.09.2? |
| 23:35.21 | ManxPower | Anthro, 1.2 is better about reloacting it |
| 23:35.27 | ManxPower | warthog, Yes. |
| 23:35.32 | ManxPower | But it's still not perfect. |
| 23:35.34 | LostFrog | ./usr/include/asterisk |
| 23:36.01 | in-side | I got Undefined symbol "ast_pthread_create" |
| 23:36.03 | warthog | manxpower, how much do the echo cancel cards help, is it worth it? |
| 23:36.05 | in-side | by everywhere |
| 23:36.09 | in-side | it never happened to me |
| 23:36.15 | ManxPower | warthog, the echo daughter cards are not cheap. |
| 23:36.17 | in-side | does anybody has a clue |
| 23:36.21 | in-side | what is happenning ? |
| 23:36.28 | in-side | I'm using 1.0.9 |
| 23:36.38 | in-side | I have the same setup working perfect at 5.4 |
| 23:36.45 | Anthro | Hrmph. I do want to keep it in /usr/local. If I must build my own software, I really want it to avoid a pissing contest with my packaging system. |
| 23:36.51 | ManxPower | warthog, we can normally get rid of echo even with 1.0.x, but it can be a long process. We are slowly working on using Tellabs external echocan |
| 23:37.07 | warthog | manxpower, they seem to be cheap with the 2400 series cards as it only cost me I think 200 more for a 2431 card |
| 23:37.21 | in-side | hey guys |
| 23:37.26 | warthog | with echocancel chip onboard that is |
| 23:37.27 | in-side | adoes anyone can help me? |
| 23:37.42 | Math[laptop] | in-side, do you have libpthread installed? |
| 23:38.05 | in-side | yah I think so I have reinstalled the kernel |
| 23:38.07 | sahafeez | in-side: if you do not get an answer here post to the mailing list. someone will answer |
| 23:38.14 | in-side | but maybe somethin missing :( |
| 23:38.16 | ManxPower | or SEARCH the mailing lists |
| 23:38.18 | ManxPower | ~mailinglist |
| 23:38.21 | jbot | extra, extra, read all about it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 23:38.23 | in-side | sahafeez: I did it |
| 23:38.28 | sahafeez | ~god |
| 23:38.30 | jbot | well, god is a llama, or real unless declared integer |
| 23:38.34 | RoyK | kram: ping |
| 23:38.41 | in-side | but it seems my isp server is doing a bad job again |
| 23:38.46 | RoyK | mark is full of shit |
| 23:38.48 | ManxPower | Results 1 - 10 of about 146 from lists.digium.com for ast_pthread_create. (0.16 seconds) |
| 23:39.01 | in-side | well I will try to recomplie my bvsd kernel again |
| 23:39.09 | in-side | there something broke here for shure |
| 23:39.18 | fulgas | hey in-side :P |
| 23:39.27 | in-side | hey fulgas |
| 23:39.39 | in-side | are you fine rapaz |
| 23:39.41 | in-side | ;) |
| 23:39.52 | Anthro | Do I need the addons and sounds as well? Do I want them? |
| 23:39.54 | fulgas | yah sure :) |
| 23:39.57 | RoyK | see the -users ml |
| 23:40.01 | fulgas | long time no read :) |
| 23:40.02 | fulgas | heh |
| 23:40.12 | Math[laptop] | in-side, this isnt a kernel problem, your lacking a library |
| 23:40.13 | in-side | ya.. kind busy you |
| 23:40.30 | RoyK | fucking mark has turned Allison away from all other projects than asterisk |
| 23:40.44 | in-side | Math[laptop]: I gonna check eeverything again |
| 23:41.33 | *** join/#asterisk aaronz (n=yoyo@pdpc/supporter/student/aaronz) |
| 23:42.29 | RoyK | fuck digium |
| 23:43.11 | aaronz | I'm trying to write an app which simply pipes the audio stream (read&write) over tcp to another local computer. What is the best way to implement this? a loop in exec that does ast_waitfor(chan, 2), reads it in, sends it over the network, then reads from network, and writes in 1 frame? |
| 23:43.48 | aaronz | while the remote server sends audio in 2msec payloads? |
| 23:43.57 | RoyK | prolly using something apart from asterisk |
| 23:43.59 | RoyK | :P |
| 23:44.42 | aaronz | also, can i dynamically load the apps or do i have to compile it w/the server? (i would assume dynamic, but i havent tried to compile/load yet) |
| 23:45.12 | Druken | RoyK: you still bitching about that ? |
| 23:45.22 | mog_work | royk is a troll |
| 23:45.27 | mog_work | hes just looking for food |
| 23:45.37 | wunderkin | mm food |
| 23:46.07 | mog_work | you make your whole side just look worse |
| 23:46.10 | mog_work | you realize that right |
| 23:46.23 | mog_work | you also have no idea what happened in said situation |
| 23:46.26 | mog_work | so you can f off |
| 23:46.34 | justinu | lol |
| 23:46.35 | *** join/#asterisk docelm0 (n=docelmo@static-71-251-95-2.tampfl.fios.verizon.net) |
| 23:46.52 | kn0x | heres my sip.conf http://pastebin.ca/28725 |
| 23:47.00 | kn0x | no sip peers will register |
| 23:47.01 | docelm0 | Does anyone know how I can cut a CDR when the dial command is finished so I can update the information in the CDR table? |
| 23:47.17 | wunderkin | <3 |
| 23:47.17 | kn0x | and im getting a "chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8155500 (len 398) to 69.90.155.70:-1 returned 5060: Bad file descriptor " on the cli |
| 23:47.17 | RoyK | mog_work: have you taken your medicin? |
| 23:47.18 | *** join/#asterisk SwK (n=SwK@dpc6745230018.direcpc.com) |
| 23:47.21 | docelm0 | right now I cant seem to update the CDR cause the row doesnt exist. |
| 23:47.29 | mog_work | roy, you know im right |
| 23:47.35 | RoyK | mog_work: no |
| 23:47.37 | RoyK | mog_work: n |
| 23:47.37 | mog_work | i dont know why you bother comming in here |
| 23:47.45 | RoyK | mog_work: you're not right |
| 23:47.46 | mog_work | you are so polarly apposed to this project |
| 23:47.54 | mog_work | just go sqwaller in your channel |
| 23:48.03 | mog_work | if you cant just enjoy both sides |
| 23:48.03 | RoyK | apposed... what word is that? |
| 23:48.12 | LostFrog | Geez, kids. |
| 23:48.24 | mog_work | sorry i failed 6th grade spelling |
| 23:48.30 | kn0x | RoyK... what is your problem with asterisk? |
| 23:48.31 | mog_work | and i dont have gtkspell on this box |
| 23:48.31 | *** part/#asterisk warthog (n=nvadekar@216.249.38.137.ppp.northrock.bm) |
| 23:48.40 | kn0x | RoyK it is a great project |
| 23:48.40 | RoyK | kn0x: quite a lot |
| 23:48.42 | justinu | hooked on monkey fonics! |
| 23:48.45 | LostFrog | ok.. Can the catfight end? |
| 23:48.59 | mog_work | im happy for it to end |
| 23:48.59 | mog_work | just stop trolling |
| 23:49.05 | SwK | CATFIGHT CAT FIGHT! |
| 23:49.10 | mog_work | lol swk |
| 23:49.16 | SwK | what up mog |
| 23:49.22 | mog_work | not much |
| 23:49.25 | mog_work | chillin |
| 23:49.26 | mog_work | you |
| 23:49.34 | justinu | tear off her bra! |
| 23:49.42 | SwK | was going to get some beer and pretzles so i could watch a fight but I guess i missed it |
| 23:49.56 | *** join/#asterisk bweschke (n=bweschke@wsip-24-120-60-190.lv.lv.cox.net) |
| 23:50.02 | in-side | damn pthread |
| 23:50.05 | in-side | and bsd |
| 23:50.05 | in-side | :S |
| 23:50.15 | mog_work | heh |
| 23:50.20 | mog_work | scroll up and then scroll down |
| 23:50.23 | mog_work | reall slow... |
| 23:50.40 | mog_work | hehe |
| 23:50.41 | LostFrog | The problem with testing a PBX, is you need calls.. |
| 23:50.43 | SwK | hah |
| 23:50.44 | LostFrog | :( |
| 23:50.55 | justinu | call yourself |
| 23:51.07 | kn0x | heres my sip.conf http://pastebin.ca/28725 no sip peers will register and im getting a "chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8155500 (len 398) to 69.90.155.70:-1 returned 5060: Bad file descriptor " on the cli |
| 23:51.14 | LostFrog | I can't handle two calls together.. |
| 23:51.18 | LostFrog | Three, even. |
| 23:51.22 | Druken | LostFrog: which is why all our cell bills went for shit when we started.... |
| 23:51.23 | Druken | gegege |
| 23:51.26 | Druken | hehehe |
| 23:51.35 | justinu | use a sip call generator |
| 23:51.53 | mog_work | go write your number on a bathroom wall |
| 23:51.55 | ManxPower | kn0x, does the problem go away when you restart Asterisk? |
| 23:51.55 | mog_work | youll get calls |
| 23:51.57 | justinu | hah |
| 23:52.09 | SwK | mog knows all about his number on the bathroom stalls |
| 23:52.14 | SwK | he writes it there all the time |
| 23:52.17 | mog_work | heh |
| 23:52.21 | justinu | glory hole! |
| 23:52.26 | mog_work | that actually happened to me one time |
| 23:52.27 | SwK | mog arent you supposed to be in vegas? |
| 23:52.28 | *** join/#asterisk twisted[mobile] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 23:52.28 | *** mode/#asterisk [+o twisted[mobile]] by ChanServ |
| 23:52.29 | LostFrog | umm.. yeah.. I'm going to write my business number in public.. lol |
| 23:52.30 | mog_work | when i was in hs |
| 23:52.32 | mog_work | no im in hsv |
| 23:52.42 | SwK | hah |
| 23:52.44 | SwK | twisted |
| 23:52.47 | LostFrog | Hey.. It's a lawyers office.. I could put in bar restrooms. :) |
| 23:52.50 | kn0x | manx- umm, no |
| 23:52.53 | LostFrog | Probably get some business. |
| 23:52.56 | twisted[mobile] | swk, eat me. |
| 23:52.58 | Druken | LostFrog: well.... i write my business number in public all the time... just not bathroom walls.... |
| 23:53.00 | mog_work | when i was in highschool there was a guy who stalked a bunch of the kids |
| 23:53.03 | SwK | whats up y0 |
| 23:53.06 | mog_work | called us at 4 in the morning |
| 23:53.07 | mog_work | etc |
| 23:53.10 | mog_work | loads of fun |
| 23:53.12 | kn0x | any ideas? |
| 23:53.23 | LostFrog | Druken: you know what i mean. |
| 23:53.29 | Druken | :) |
| 23:53.39 | LostFrog | lol.. I just noticed that 'Druken' doesn't gave a n in the middle. |
| 23:54.08 | wunderkin | LostFrog: he's so drunk he forgot the n? |
| 23:54.21 | LostFrog | I'm so daft, I thought it said drunken, wunderkin. |
| 23:55.08 | Druken | ya daft punk :) |
| 23:56.42 | Druken | fucken weather network... tellin me were going to start getting that god damn lake effect snow.... |
| 23:56.55 | justinu | snow sucks |
| 23:56.57 | *** join/#asterisk Frawg (n=Frawg@unaffiliated/frawg) |
| 23:57.18 | Druken | justinu: ya think ? |
| 23:57.19 | ManxPower | We never get lake effect snow here, because we don't get snow! hahahahahha! |
| 23:57.26 | justinu | no snow here either |
| 23:58.03 | ManxPower | Druken, move south. |
| 23:59.09 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 23:59.10 | Druken | nah.... i wouldn't thank ya to be american... |
