00:00.15 | SpaceBass | not i |
00:02.00 | Hmmhesays | looks alright |
00:02.05 | Hmmhesays | and AMP'ish |
00:02.06 | lowridese | spacebass: do you have that screen name to send? |
00:02.22 | lowridese | no one is answering there with them right now |
00:03.57 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3634.nb.aliant.net) |
00:04.26 | SpaceBass | back |
00:04.39 | SpaceBass | lowridese: no I sure don't |
00:05.04 | lowridese | but they had someone that helped you in the end? |
00:05.36 | SpaceBass | yeah, he didnt work for BV, but he helped |
00:05.54 | SpaceBass | and then I scrapped * when I needed help again Ariel_ was a big help |
00:07.36 | SpaceBass | ok, evoloution on os x... slow |
00:11.38 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
00:11.45 | infinity1 | can someone spend a few cycles with me on this CHANUNAVAIL problem i'm having with my dialplan? |
00:11.47 | shido6 | what are u runnin os x on? |
00:12.07 | SpaceBass | powerbook 12: |
00:12.09 | SpaceBass | 12" |
00:12.13 | SpaceBass | 1 gb ram |
00:13.25 | SpaceBass | not that impressed with evolution... uses OWA to talk to exchange |
00:13.50 | Hmmhesays | infinity1 might help if you post some cli output on pastebin |
00:14.55 | infinity1 | Hmmhesays: k. i'll do that. |
00:15.24 | Hmmhesays | then pat your head, rub your tummy and whistle |
00:15.47 | tzafrir | does: asterisk -rx 'restart now' work (restarts, doesn't crash) for anybody here that runs asterisk as non-root? |
00:15.50 | *** join/#asterisk yoink (~yoink@MTL-HSE-ppp165512.qc.sympatico.ca) |
00:16.13 | tzafrir | Seems broken in 1.0 . I wonder if it got fixed in HEAD |
00:17.20 | jmann | Question for all-- is there a way that I can have a dns name in the externip= option in the sip.conf file |
00:17.40 | jmann | this is due to having DHCP from my provider |
00:17.41 | SpaceBass | jmann: sure |
00:18.12 | jmann | what is the syntax |
00:18.34 | Hmmhesays | I would try externip=www.myrclean.com |
00:18.46 | Hmmhesays | heh, once again a good joke foiled by bad typing |
00:18.58 | infinity1 | Hmmhesays: http://pastebin.ca/16893 |
00:18.59 | jmann | I am getting an Invalid address to externip |
00:19.34 | Hmmhesays | ok infinity set verbose to something high make a call and paste the cli output in there |
00:20.48 | infinity1 | Hmmhesays: http://pastebin.ca/16895 |
00:21.23 | infinity1 | i'm trying to get it to dialout on nufone if the zap interface is busy. |
00:23.16 | Hmmhesays | ${DIALSTATUS} = CHANUNAVAIL looks like that is returning false |
00:23.19 | infinity1 | Hmmhesays: it seems to me that dialstatus ins't being set to unavail. |
00:24.02 | infinity1 | Hmmhesays: any idea what my problem is? i think this is suppose to work |
00:28.42 | Hmmhesays | well as i'm sure you noticed the call is not making it to s,22 |
00:29.35 | infinity1 | yea. its jumping to 122 |
00:30.11 | infinity1 | hm. on second though. that should be a 123 i think. |
00:30.31 | Hmmhesays | doesn't dial return n+101 on failure? |
00:30.38 | infinity1 | yea |
00:30.55 | infinity1 | oh. wait. |
00:30.59 | infinity1 | 122 is right :) |
00:31.20 | Hmmhesays | If all the called channels are busy, Dial will exit with a return code of 0 and will continue in the current context at priority n+101, if it exists, where n is the priority of the Dial command. |
00:31.46 | infinity1 | http://lists.digium.com/pipermail/asterisk-users/2005-February/089017.html |
00:31.48 | infinity1 | same problem |
00:32.33 | Hmmhesays | comment out 122 and reload and try your call |
00:32.44 | infinity1 | Hmmhesays: i tried that. it works. |
00:32.53 | Hmmhesays | did you see what I said above? |
00:33.31 | infinity1 | so that way it is working is correct? |
00:33.33 | SarahEmm | hihi |
00:33.35 | SarahEmm | backies :o) |
00:33.38 | Hmmhesays | infinity1: yes |
00:33.42 | infinity1 | it retruns busy instead of CHANUNAVAIL |
00:33.50 | infinity1 | argh. |
00:33.59 | infinity1 | so how do i use CHANUNAVAIL? :) |
00:34.05 | Hmmhesays | leave that busy commented out |
00:34.12 | Hmmhesays | or... |
00:34.33 | Hmmhesays | set s,122 to goto s,22 |
00:35.14 | *** join/#asterisk iq (~IQ@70-59-162-85.omah.qwest.net) |
00:37.06 | SarahEmm | shido6: should i resend my request to the main email address, or is it okay as-is? |
00:37.10 | infinity1 | so whoever i stole the code from is an idiot ...and i'm prolly a bigger one . hah |
00:37.49 | infinity1 | Hmmhesays: thanks for the help. i'll play with it some more. |
00:38.41 | file[laptop] | stealing?!? |
00:39.04 | Hmmhesays | heh, stole? that's an amp generated config i'm sure |
00:39.04 | infinity1 | file[laptop]: borrowed :) |
00:39.16 | infinity1 | Hmmhesays: whats amp? |
00:39.28 | Hmmhesays | where you borrowed that dialplan from |
00:39.35 | Hmmhesays | ~amp |
00:39.35 | jbot | it has been said that amp is an Audio MPEG Player. [non-free], or http://amp.coalescentsystems.ca/ |
00:39.43 | litage | how exactly does asterisk work with e164.org? |
00:39.48 | Hmmhesays | heh, that's not quite right |
00:40.19 | infinity1 | do most people use something like amp to create dialplans? or do they do it by hand? |
00:40.27 | Hmmhesays | amp is a gui for asterisk |
00:41.05 | file[laptop] | by hand. |
00:41.24 | Hmmhesays | yeah, i'd agree with that |
00:41.51 | infinity1 | aite. i guess i'll keep banging my head learning this |
00:41.52 | infinity1 | heh |
00:42.22 | Hmmhesays | helps if you start off with your own dialplan... you'll learn more banging your head that way |
00:42.53 | infinity1 | so it works like you said. |
00:42.54 | infinity1 | <PROTECTED> |
00:43.13 | Hmmhesays | i suppose it does |
00:45.30 | Hmmhesays | now if your iax connection returns busy it should still play the busy prompt |
00:47.16 | SarahEmm | hihi litage |
00:48.28 | *** join/#asterisk NetSkier (~ns@ca-redbch-cuda1-c9b-a-152.stmnca.adelphia.net) |
00:49.53 | litage | howdy there SarahEmm |
00:49.53 | litage | what's cookin |
00:50.15 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net) |
01:01.28 | SarahEmm | not too much |
01:01.28 | SarahEmm | brain is full |
01:01.28 | SarahEmm | relaxing |
01:01.28 | syle | looking for area code 204 DID's |
01:01.28 | SarahEmm | syle: have you talked to blitzrage? |
01:01.28 | syle | yeah he told me about a site, i emailed them and never got a response back |
01:01.28 | syle | so still looking |
01:01.28 | SarahEmm | which site? |
01:01.28 | syle | http://www.mixnetworks.com/ |
01:01.28 | SarahEmm | ahh |
01:01.28 | SarahEmm | err |
01:01.28 | SarahEmm | blitzrage works with them |
01:01.28 | SarahEmm | he can set you up himself afaik |
01:01.28 | Hmmhesays | how many incoming simultaneous calls are you looking for to one did? |
01:01.28 | SarahEmm | Hmmhesays: me? one |
01:01.28 | SarahEmm | oh, duh |
01:01.28 | SarahEmm | nevermind, not me :P |
01:01.28 | syle | right now just 1 hmmhesays, but if i ever get a company contract more |
01:01.28 | SarahEmm | thanks Hmm-home |
01:01.28 | SarahEmm | Hmmhesays |
01:01.28 | Hmmhesays | vonage has a business plan you can use with asterisk |
01:01.29 | Hmmhesays | omg I have a clone |
01:01.29 | wunderkin | damn why doesnt anything ever work right for me ;/ |
01:01.29 | Hmmhesays | vatsamatta |
01:03.09 | wunderkin | i have ztdummy installed, when i try to make a conference, it says invalid conference number |
01:03.22 | SarahEmm | that's not a ztdummy issue anyway... |
01:03.24 | Hmmhesays | lsmod |
01:03.54 | wunderkin | yeah it shows |
01:04.11 | Hmmhesays | SarahEmm rmmod ztdummy and see what your asterisk says |
01:04.33 | SarahEmm | Hmmhesays: *blinks* you're serious? 'no timing source' shows 'invalid conference number'??? |
01:04.48 | Hmmhesays | *that is not a valid conference number please try again* |
01:04.58 | SarahEmm | *blinks* |
01:04.58 | SarahEmm | gah! |
01:05.01 | SarahEmm | that's intuitive :P |
01:05.12 | SarahEmm | do you at least get something on the console? |
01:05.32 | Hmmhesays | good question, I don't remember |
01:05.48 | PatrickDK | sounds like meetme.conf isn't setup right |
01:06.34 | infinity1 | i don't suppose there is a way to check if a zap interface has dialtone before using it |
01:06.42 | wunderkin | you cant do.. like meetme(1|d) and it will create #1? |
01:07.03 | SarahEmm | infinity1: well, x100p's throw a RED alarm if there isn't a line hooked up.. does that help you? |
01:07.14 | PatrickDK | hmm, dunno, I haven't done dynamic conferences |
01:07.18 | PatrickDK | never had a reson to |
01:07.40 | *** join/#asterisk PBXtech (~nik@001-740-536.area1.spcsdns.net) |
01:07.44 | infinity1 | SarahEmm: whats the command to see this alarm? |
01:07.50 | wunderkin | well i dont really want to, but just trying to figure out a workaround for another problem |
01:08.27 | PBXtech | why am i getting permission errors on call files when moved to the outbound dir? directory and file are chmod777 and user is asterisk.. |
01:08.45 | litage | how exactly does asterisk work with e164.org? can you tell your viop device to dial john@smith.com and asterisk will query e164.org for the corresponding voip/phone #? |
01:08.51 | *** join/#asterisk redG ([U2FsdGVkX@67.51.185.15) |
01:08.51 | redG | <PROTECTED> |
01:09.09 | SarahEmm | infinity1: zttool will show it. asterisk will throw a msg on the console when it goes into RED. i'm not sure how to get it from code |
01:09.22 | PatrickDK | litage, no, you query the phonenumber at e164, and it tells you if there is a match |
01:09.44 | PatrickDK | read up on e164 |
01:09.57 | litage | PatrickDK: i have been, but they're a bit fuzzy on details |
01:10.08 | PatrickDK | fuzzy? |
01:10.14 | litage | unspecific |
01:10.19 | PatrickDK | it very simple, does phone number xxx-xxx-xxxx exist |
01:10.23 | infinity1 | SarahEmm: hm., i was trying to see if there is a way to not use a zap interface if the phone line is broke for dialing out. guess it requires some serious scripting |
01:10.28 | PatrickDK | yes it does, it's at sip/user@ip |
01:10.49 | *** join/#asterisk rnovotny22 (~rnovonty2@c-66-41-170-125.hsd1.mn.comcast.net) |
01:10.55 | litage | PatrickDK: yes but e164.org says that you can call someone via their email address |
01:10.57 | PatrickDK | infinity, I never found a way for asterisk to do that, my cisco box's do though |
01:11.10 | PatrickDK | litage, that must be a custom extention |
01:11.14 | SarahEmm | infinity1: well, you should be able to |
01:11.24 | SarahEmm | infinity1: you'd need to detect if there's an alarm on the channel before dialing out |
01:11.40 | SarahEmm | infinity1: i'm going to be Super Busy until saturday, i'll try to investigate then and if there's no way to do it now, make one |
01:11.45 | SarahEmm | there prolly is tho |
01:12.32 | litage | PatrickDK: so the user has to query e164.org manually, find a match, and put that phone # into their voip device? |
01:13.01 | PatrickDK | na, normally the voip device does that |
01:13.14 | syle | anyone tried the iaxy's? |
01:13.25 | infinity1 | i see. zttool changes the active flag. |
01:13.32 | DarthClue | litage: or you write some code to have * do it via res_js or agi or something of that nature. |
01:13.34 | infinity1 | too bad that info isn't availabe as a dialplan command |
01:13.46 | litage | thanks DarthClue, i'll look into res_js and agi |
01:13.54 | PatrickDK | darthclue, asterisk already does it |
01:14.01 | SarahEmm | infinity1: it's not in any variable? |
01:14.59 | DarthClue | PatrickDK: using what? |
01:15.37 | PatrickDK | e164 |
01:15.37 | infinity1 | SarahEmm: i'm still looking . i haven't found one |
01:15.37 | PatrickDK | enum.conf |
01:15.37 | SarahEmm | infinity1: 'kay. i'll look into it on saturday and look at fixing it if i don't find a way to do it. |
01:15.37 | SarahEmm | infinity1: what id you try dialing out on an alarmed channel? it doesn't just jump to priority+100 or somesuch? |
01:15.37 | DarthClue | litage: there ya go. |
01:15.44 | syle | i got a question: when i am dialing out my analog phone provider line on my fxo port , *67 in asterisk doesn;t execute *67 on my phone providers line, how can i fix this? |
01:16.05 | PatrickDK | syle, make asterisk support *67 |
01:16.09 | PatrickDK | that is alot of fun :) |
01:16.13 | infinity1 | SarahEmm: well, the problem is that you cant jump to 101 because it also means that the chanels are ful |
01:17.20 | SarahEmm | Patrick^: make it support? huh? |
01:17.20 | infinity1 | SarahEmm: so if you use zap + iax to dialout, 101 wouldnt' work. according to the docs +101 is for busy/no avail chanels |
01:17.20 | SarahEmm | Patrick^: isn't it just normal DTMF? |
01:17.20 | syle | well we are just talking about digit pass-through on analog lines |
01:17.20 | PatrickDK | sarahemm, yes, his dialplan doesn't support it though |
01:17.20 | litage | DarthClue: what do you mean? |
01:17.20 | SarahEmm | Patrick^: oh. well that's a dialplan issue then :) |
01:17.20 | infinity1 | is zttool only ncurses? its really annoying. |
01:17.38 | syle | can i setup a *67 extension in dialplan to override asterisk's default *67? |
01:17.41 | SarahEmm | infinity1: okay, but if the channel is in alarm, it's unavaialble.. do you need to seperate the two? |
01:17.41 | PatrickDK | syle, it's all in extentions.conf |
01:17.45 | DarthClue | litage: take a look at http://www.voip-info.org/tiki-index.php?page=Asterisk+E164+Call+Routing and http://www.voip-info.org/tiki-index.php?page=Asterisk+config+enum.conf |
01:17.55 | PatrickDK | syle, asterisk doesn't have a default *67 |
01:18.34 | syle | sure it does |
01:18.35 | PatrickDK | unless your using zaptel fxs ports |
01:18.35 | infinity1 | SarahEmm: yea. because unavail also means it is in use |
01:18.35 | syle | yeah i was using zaptel fxs ports for testing |
01:18.35 | infinity1 | SarahEmm: wait. |
01:18.35 | PatrickDK | ah, I only use sip phones |
01:18.41 | infinity1 | SarahEmm: hmm |
01:18.48 | syle | it don;t work on sip? |
01:19.25 | syle | hmmm, so you do *67, and in that extension do like setCID caller unknown kind of thing? |
01:19.47 | infinity1 | SarahEmm: i think it would work if the dial returned control to the dial plan and returned CHANUNAVAIL |
01:20.10 | SarahEmm | infinity1: it doesn't? what happens? |
01:20.23 | infinity1 | SarahEmm: it just sits there |
01:20.37 | syle | patrick what ip phone do you use? |
01:20.52 | SarahEmm | infinity1: and you've verified the channel is showing RED right now? |
01:21.50 | infinity1 | SarahEmm: when i removed the conection to the zap interface, it still tries to use it and doesn't fail fast enough |
01:21.50 | PatrickDK | syle, yuou don't unset cid, unless your using a sip/iax voip provider |
01:21.50 | file | DarthClue: I'm awake, or atleast I think I am! |
01:21.50 | PatrickDK | if you using fxo out, you have to dial *67 before your number |
01:21.50 | infinity1 | SarahEmm: i've never seen this RED thing. i loaded zttool and i only see active chaning |
01:21.50 | infinity1 | er s/chaning/changing/ |
01:21.54 | DarthClue | file: it is taking you way too long to respond, you are absolutely asleep. |
01:22.08 | file | am not |
01:22.10 | file | I'm working |
01:22.19 | infinity1 | SarahEmm: active changes when i unplug and plug the zap interface in. |
01:22.52 | DarthClue | file: everything you are doing is just a dream, therefore you must be asleep. |
01:23.49 | syle | patrick your just horrid to talk to lol |
01:23.49 | syle | bbiab |
01:23.49 | infinity1 | SarahEmm: i'm using a tdm400p. maybe becuase it only has one interface it doesn't go red? |
01:23.49 | file | DarthClue: wow - your logic is great! |
01:23.49 | infinity1 | SarahEmm: er i mean because it has multiple interfaces |
01:23.49 | PatrickDK | syle, I will help yuou, I will not do it for you |
01:23.49 | SarahEmm | infinity1: ahh, i don't have a tdm400 so i have no idea =-/ |
01:23.49 | SarahEmm | x100p here |
01:24.23 | infinity1 | SarahEmm: when you unplug the pstn from the x100p it goes red? mine just changes activity to 0 instead of 1 |
01:24.23 | wunderkin | valetparking doesnt work right for me either ;/ |
01:24.39 | DarthClue | wunderkin : which valet parking are you using? |
01:24.59 | wunderkin | DarthClue, umm.. app_valetparking from bkw i think |
01:25.09 | SarahEmm | infinity1: mine goes red in zttool |
01:25.19 | infinity1 | SarahEmm: interesting. |
01:25.21 | DarthClue | wunderkin: verify that and then tell us what it is doing. |
01:25.23 | SarahEmm | using CVS HEAD zap stuff |
01:25.37 | infinity1 | SarahEmm: i'm using 1.0.9 debian unstable |
01:25.40 | SarahEmm | ahh |
01:25.43 | SarahEmm | maybe it's a HEAD thing |
01:25.48 | wunderkin | DarthClue, well.. maybe not.. in the header it says anthm? |
01:26.07 | DarthClue | wunderkin: that's the right one. what is it doing? |
01:27.55 | wunderkin | DarthClue, i have exten => 7,1,ValetParkCall(1|mylot|600|1|2|dial) / exten => 8,1,ValetUnParkCall(1|mylot); i park it ok and it keeps saying the lot number.. when i pick it up it works but keeps saying the lot number..then when i hangup it crashes asterisk.. or i think it did something else weird the time before that |
01:28.22 | DarthClue | wunderkin: stable or head? |
01:28.28 | wunderkin | head from 6/30 |
01:29.50 | DarthClue | wunderkin: i don't think anthm or bkw_ are here at the moment. they would be the ones to provide the best assistance. drop them an email or wait till morning (12 hours) and they will be around. |
01:30.19 | file | grab a backtrace and see where it's crashing |
01:30.26 | wunderkin | ok.. are there other versions i can try? |
01:30.27 | DarthClue | unless file wants to volunteer. |
01:30.42 | file | well there's super valet parking too |
01:30.45 | wunderkin | yeah |
01:31.05 | DarthClue | valet_parking supercedes super_valet_paerking. |
01:31.08 | file | but uh backtrace! |
01:31.12 | DarthClue | er...parking even. |
01:32.33 | wunderkin | ill try |
01:33.03 | *** join/#asterisk likwid-- (~likwid@nc-65-41-255-12.dyn.sprint-hsd.net) |
01:33.11 | wunderkin | there it went with the weirdness this time.. didnt crash |
01:34.31 | wunderkin | ok well theres no core so i guess i have to attach to it |
01:35.05 | wunderkin | nice gdb isnt installed |
01:39.45 | tzafrir | maybe you have strace? |
01:40.02 | tzafrir | Sometimes it is good enough |
01:40.19 | wunderkin | well i just installed gdb .. i havent used it that often so.. ill try :D |
01:40.31 | SarahEmm | strace rocks |
01:41.41 | wunderkin | ok i don tknow what im doing |
01:43.12 | tzafrir | wunderkin, do you have a core dump? |
01:43.20 | wunderkin | not for this issue |
01:43.29 | tzafrir | (did you run asterisk with -g to get one?) |
01:43.42 | wunderkin | it looks like asterisk gets killed |
01:43.45 | wunderkin | yes |
01:44.00 | tzafrir | you can attach gdb/strace to a runnig asterisk process |
01:44.31 | wunderkin | well, its attached to something but i dont know where to go from there |
01:44.52 | wunderkin | i did gdb program then the pid of asterisk.. at least one of them |
01:44.53 | tzafrir | wunderkin, what is the issue you try to isolate? |
01:45.19 | wunderkin | a problem with valetparking.. it is doing weird things and asterisk ends up getting killed |
01:45.33 | SarahEmm | what makes it 'valet' call parking? |
01:45.49 | SarahEmm | does a little asterisk process come and move your call to the parking lot instead of manually having to drive it there yourself? |
01:46.18 | wunderkin | file wanted me to backtrace it |
01:48.03 | wunderkin | ok, well i did an strace on it |
01:48.11 | wunderkin | file, is that good enough? |
01:48.48 | DarthClue | SarahEmm: yeah, something like that. |
01:52.14 | wunderkin | i dont know how to use gdb on a running program |
01:52.50 | wunderkin | strace says its getting a sigkill |
01:56.13 | *** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net) |
01:56.13 | milkyflava | hello |
01:56.13 | milkyflava | I have finally got my TDM400P and I am going through the setup at asteriskdocs.org |
01:56.13 | milkyflava | Do I need to use the IAX or SIP config files or just the zaptel.conf file? Do I setup the zaptel.conf then use either IAX or SIP? |
01:56.17 | milkyflava | Is zaptel for setting up the card then the IAX or SIP is for making calls? |
01:56.40 | milkyflava | Am I even making sense? |
01:57.28 | wunderkin | well.. you use zap to access the card... if you have other phones off of asterisk that arent connected to your card, then you would use sip or iax |
01:58.12 | milkyflava | would thise be like soft phones and IP phones? |
01:58.18 | wunderkin | yes |
01:58.18 | milkyflava | thise=those |
01:59.50 | milkyflava | Thanks, this stuff is confusing! |
01:59.50 | wunderkin | if you have a phone on a fxs port, it would use zap |
01:59.50 | milkyflava | But I can use them together also, correct? |
01:59.50 | infinity1 | milkyflava: hah. you haven't even touched the surface. |
01:59.50 | milkyflava | lol, I know |
01:59.50 | wunderkin | to dial out the fxo port it would use zap too |
01:59.50 | wunderkin | well you could have a sip phone dial out on your card |
01:59.50 | milkyflava | I have a dev card with 1 fxo and 1 fxs port on it the 11B I think it was called |
02:01.54 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-HSE-ppp168389.qc.sympatico.ca) |
02:01.54 | SarahEmm | hihi |
02:01.54 | SarahEmm | again |
02:01.54 | SarahEmm | WAP crashed :P |
02:01.54 | milkyflava | wunderkin: Thank you. |
02:01.54 | DarthClue | wunderkin: are you using super valet parking or valet parking? |
02:01.54 | wunderkin | DarthClue, just regular right now.. |
02:02.08 | wunderkin | i was just trying to figure out what the difference was with super |
02:02.12 | DarthClue | ok, super is old so i just wanted to make sure. |
02:02.18 | wunderkin | oh.. |
02:02.28 | DarthClue | super was an alpha that anthm did as a whim. valet is what has been officially released. |
02:02.36 | wunderkin | ok |
02:03.12 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
02:04.15 | wunderkin | this is really just trying to hack around attended transfer.. which doesnt work right the way im doing things |
02:04.16 | wunderkin | unless it just doesnt work the way im wanting it to |
02:06.01 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
02:06.01 | bjohnson | totally unrelated to asterisk but I can't seem to bring up a web site. does somebody know how to use a public http proxy? |
02:07.42 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
02:09.02 | *** join/#asterisk chonlada (chonlada@202.29.6.19) |
02:09.39 | chonlada | Hi all, can i find NEC Neax 7400 ICS manual ? |
02:10.35 | Pete_Largo | try google? |
02:10.54 | chonlada | yes, i am try. but cannot found. |
02:11.18 | Pete_Largo | http://www.sundance-communications.com/cgi-bin/forumdisplay.cgi?action=topics&forum=NEC&number=14&DaysPrune=1000&LastLogin= |
02:14.35 | Pete_Largo | someone there may be able to help |
02:14.36 | SarahEmm | nini all |
02:19.24 | Pete_Largo | night Sarah |
02:19.24 | wunderkin | WOOOOOHOOOO |
02:19.24 | wunderkin | (sorta) |
02:19.24 | Pete_Largo | good news wunderkin? |
02:19.24 | wunderkin | yeah i finally figured out why atxfer wasnt working right for me :D |
02:19.24 | wunderkin | but now i gotta figure out a caller id problem.. i think that one is a source problem |
02:19.24 | Pete_Largo | good for you :) |
02:19.24 | chonlada | Pete_Largo thank you. |
02:19.24 | *** join/#asterisk CpuID (~nathan@dsl-202-173-176-82.qld.westnet.com.au) |
02:19.24 | wunderkin | i was using # for atxfer which i guess # is reserved for the regular transfer.. i was wondering why it was doing blind :) |
02:19.24 | CpuID | ok ppls, question, is asterisk HEAD these days kinda like a 1.1 branch or something? |
02:19.24 | JunK-Y | 1.2 |
02:19.25 | DarthClue | 1.2 |
02:19.37 | CpuID | ah k |
02:19.59 | CpuID | so will 1.0.x lead up to 1.1, and then HEAD as you said will stay 1.2 right? |
02:24.46 | CpuID | just confirming, so i know whats what when reading through src :) |
02:24.46 | CpuID | or will 1.1 get skipped altogether? |
02:24.46 | rabelais | do I need "insecure = very" to get incoming calls from fwd? |
02:24.46 | Pete_Largo | fwd has a great setup guide on their web site |
02:24.46 | loud | that parameter should be changed or something, people think that they can get owned with that. |
02:24.46 | rabelais | I can't get calls otherwise, so is it not a security hole to allow that? |
02:24.48 | loud | nope |
02:25.31 | loud | its to allow reged hosts to call without re doing auth. |
02:26.39 | rabelais | ok, I read that on the wiki, I was a little confused, thanks for clarifying it :) |
02:36.46 | CpuID | i think its good having insecure=very as is :) |
02:37.05 | CpuID | gives hosts a good reason to change their shit, so people wont think their gonna get owned all the time lol :) |
02:37.14 | CpuID | the name suits nicely |
02:39.59 | loud | me too, unless it brings a security flaw, i dont care, but people think its like a doyouwantogethacked = yes |
02:39.59 | CpuID | lol |
02:39.59 | *** join/#asterisk darylp (~daryl@63-208-162-60.digitalrealm.net) |
02:43.35 | Ariel_ | Hello everyone |
02:48.18 | litage | if my voip provider only uses H323 and my network uses NAT, how can i setup asterisk to allow multiple simulataneous incoming/outgoing calls? |
02:51.37 | *** join/#asterisk spoot_nick (~julio@CPE-147-10-171-214.nsw.bigpond.net.au) |
02:51.37 | Ariel_ | h323 is not very friendly to nat. |
02:51.37 | spoot_nick | does anybody know why asterisk (using the samples conf files) would get calls, but not output any sound at all? |
02:51.37 | spoot_nick | error msg: WARNING[15566]: file.c:550 ast_readaudio_callback: Failed to write frame |
02:51.37 | litage | Ariel_: yeah that's what i've read, so i'm curious for any solutions that might exist |
02:51.59 | Ariel_ | litage, everyone I know that uses h323 is on an static IP address. |
02:52.21 | Ariel_ | spoot_nick, is this sip your not getting sound on? |
02:52.27 | spoot_nick | yes |
02:52.48 | Ariel_ | spoot_nick, is the server on the same network as the sip device? |
02:52.58 | spoot_nick | Ariel_: yep |
02:53.06 | spoot_nick | no NAT between them, local area network |
02:53.11 | spoot_nick | fresh install according to the "10 minute guide to asterisk", http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart |
02:53.18 | spoot_nick | slackware 10.1 |
02:53.55 | spoot_nick | i'm using a sjphone to test it. i dialed "1000" for the congrats msg, and no sound gets out of it. same with the echo service |
02:54.24 | litage | Ariel_: can those people you know using h323 do multiple, simultaneous in/out calls? |
02:54.27 | Ariel_ | spoot_nick, is the sip phone registered? |
02:54.35 | Ariel_ | litage, yes |
02:54.53 | spoot_nick | Ariel_: yes it is. example console output: Registered SIP 'julio' at 192.168.0.102 port 5060 expires 120 |
02:54.59 | litage | Ariel_: how did they accomplish that? |
02:55.50 | Ariel_ | litage, I have not used h323 in over a year. It's a bear to get configured actually to get it to install on the asterisk box. But it does work once you get pass that. |
02:56.13 | spoot_nick | weird thing is i don't get any error msgs. except for that one 'ast_readaudio_callback: Failed to write frame' |
02:56.27 | spoot_nick | which makes sense, since it's just audio that seems to be failing me |
02:56.46 | Ariel_ | spoot_nick, I don't know the 10 minute setup let me read what it does. But first I would check your settings as it not configured correctly. |
02:56.59 | litage | Ariel_: all the info and docs needed to get h323 configured on asterisk is available on voip-info.org? |
02:57.26 | Ariel_ | spoot_nick, for your sip.conf account do this disallow=all allow=ulaw canreinvite=no |
02:57.49 | Ariel_ | litage, most of it. yes and there is readme's as well. |
02:58.12 | litage | Ariel_: as in Asterisk's readmes, right? |
02:58.49 | Ariel_ | litage, there is an h323 channel drivers that comes with asterisk there is a readme there. |
03:02.17 | litage | Ariel_: what exactly is a channel? |
03:02.17 | spoot_nick | Ariel_: my current conf files, http://202.92.94.93/~julio/ast_confs.tar.gz |
03:02.17 | tzanger | litage: a channel is something that audio comes in and out of |
03:02.17 | tzanger | a resource is something that can manipulate a call (might provide audio, mangle audio, detect audio, etc.) |
03:02.17 | Ariel_ | spoot_nick, use pastebin.ca I don't have any way to unzip them on my remote system. |
03:02.17 | Ariel_ | hello tzanger |
03:02.17 | litage | ah i see |
03:02.17 | tzanger | hello Ariel_ |
03:02.57 | spoot_nick | well, http://202.92.94.93/~julio/ast_confs/sip.conf, then iax.conf and extensions.conf in the same folder |
03:03.00 | spoot_nick | tks |
03:03.46 | *** join/#asterisk brookshire (~matt@esbrooks3.traveller.com) |
03:05.05 | *** join/#asterisk sroddy (~sroddy@republic.ligo-la.caltech.edu) |
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03:05.19 | Ariel_ | spoot_nick, ok now do what I asked you to do. put disallow=all allow=ulaw and add canreinvite=no |
03:05.31 | lowridese | anyone able to help out with broadvoice incoming and voipjet outgoing setup? been trying for a week now |
03:05.58 | Ariel_ | lowridese, wow inbound via bv and outbound via voipjet. |
03:05.58 | spoot_nick | Ariel_: did it |
03:06.16 | Ariel_ | spoot_nick, ok and? |
03:06.30 | Ariel_ | lowridese, what is the problem you have? |
03:06.30 | spoot_nick | Ariel_: no sound. commited the changes and restarted the server |
03:07.05 | lowridese | every problem in the book, if you can think of a better provider to setup with im game. i cant get BV to register with them and only got VOIPJET to work once |
03:07.17 | loud | both are good ones. |
03:07.25 | lowridese | been through so many different configurations and im sure its easy to get up and running |
03:07.26 | spoot_nick | Playing 'demo-congrats' (language 'en') |
03:07.28 | Ariel_ | spoot_nick, when you have no sound 90% of the time is due to a natted network. |
03:07.29 | brookshire | voicepulse rocks |
03:07.30 | brookshire | :) |
03:07.37 | loud | vp is great too. |
03:07.41 | file[laptop] | Mattttttttt |
03:07.43 | lowridese | isnt it expensive? |
03:07.46 | brookshire | hey file! |
03:07.49 | brookshire | not really |
03:07.50 | file[laptop] | hi hi |
03:07.56 | brookshire | i think i pay like $13 a month |
03:07.56 | spoot_nick | Ariel_: gotcha. but that's weird. we're in a LAN here, 192.168.0.0/24 |
03:08.06 | brookshire | for both incoming and outgoing |
03:08.12 | lowridese | unlimited? |
03:08.14 | loud | lowridese, pastebin your stuff. |
03:08.18 | brookshire | no.. not unlimited |
03:08.25 | brookshire | unlimited in |
03:08.29 | JunK-Y | yo brookshire. |
03:08.31 | lowridese | ok unlimited in works |
03:08.41 | Ariel_ | spoot_nick, you said your on the same network. Your not? |
03:08.46 | brookshire | lowrie; plus, they support iax :) |
03:08.55 | brookshire | but! |
03:09.01 | Ariel_ | voicepulse is great I have had them for over a year without any issues. |
03:09.03 | spoot_nick | i am. the server running asterisk and the desktop computer running the sjphone |
03:09.03 | brookshire | you also don't have to use voicepulse for out |
03:09.11 | loud | why not |
03:09.14 | brookshire | if you want to use voipjet or something still |
03:09.37 | spoot_nick | Ariel_: no hops between both computers |
03:11.21 | lowridese | ok will do |
03:11.31 | Ariel_ | spoot_nick, have you reloaded since your setting up the config file? |
03:11.55 | lowridese | http://pastebin.ca/16900 |
03:12.12 | spoot_nick | yes. actually stopped asterisk and ran it again |
03:12.18 | lowridese | if anyone can make sense of those that is great. i also edited my iax.conf file and sip.conf file and extensions.conf file |
03:12.49 | loud | what about the dialplan where you put the outbound |
03:13.48 | lowridese | voicepulse has a big startup fee it looks like |
03:13.51 | spoot_nick | Ariel_: does it matter to have a default gateway? wouldn't make much sense to me, since it's not NAT |
03:14.08 | spoot_nick | Ariel_: in the same network class, of course |
03:14.14 | loud | but are up most of the time |
03:14.32 | lowridese | do they have a byod plan? |
03:14.43 | loud | of course, connect.voicepulse.com |
03:14.57 | Ariel_ | spoot_nick, you keep saying that. But what device is inbetween them two systems. |
03:15.58 | spoot_nick | Ariel_: a switch, and a windows 2k as a gateway |
03:16.06 | spoot_nick | Ariel_: no firewalls |
03:16.10 | lowridese | if i go with voicepulse is there a good place for setup on it? |
03:16.11 | Ariel_ | ahh windows 2k |
03:16.20 | lowridese | anyone check out the pastebin i put up? |
03:16.50 | loud | do you get to register with bv ? |
03:17.02 | Ariel_ | lowridese, give me a minute or two I will look at it then. Trying to get my self in a better spot with this wireless tablet. |
03:17.23 | lowridese | my connecion they say they can see but i am having authentication issues 401 error. and does not register |
03:17.24 | lowridese | ok thanks' |
03:17.55 | brookshire | lowridese: they give you example asterisk setup confs |
03:18.24 | spoot_nick | Ariel_: technically it's a router. all other type of connection i use goes through fine. |
03:20.25 | lowridese | who is they? |
03:20.39 | loud | lowridese, you are missing authuser on bv |
03:20.54 | Ariel_ | spoot_nick, windows 2k will not forward udp ports 10,000 to 20,000 |
03:21.32 | spoot_nick | i thought the port in question was 5060 udp |
03:21.45 | lowridese | so i would add authuser=(phone #)? |
03:22.05 | Ariel_ | spoot_nick, only for registration sound is via rtp which is on ports 10,000 to 20,000 |
03:22.45 | spoot_nick | Ariel_: well, thanks a lot =) i'll remove the default gateway and do a straight connection between these 2 pcs |
03:22.52 | spoot_nick | should workd |
03:22.59 | *** join/#asterisk |Vulture| (~V@user-0c6tr11.cable.mindspring.com) |
03:23.14 | loud | yes lowridese |
03:23.23 | loud | let me pastebin for you |
03:23.34 | loud | http://pastebin.ca/16902 |
03:23.54 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net) |
03:24.33 | |Vulture| | Anyone know what this notice refers to "NOTICE[15159]: cdr.c:1157 do_reload: CDR simple logging enabled."? I looked in cdr.conf and couldn't find it |
03:24.43 | *** join/#asterisk vuvie (~S@bb219-74-45-31.singnet.com.sg) |
03:24.51 | lowridese | loud:that is a simple setup right that should work? :) |
03:25.23 | loud | directly from my working * |
03:25.47 | lowridese | so you are using broadvoice, what do you use for outgoing? BV? |
03:26.22 | lowridese | does my registration string look correct? |
03:26.24 | loud | PRI, and voicepulse, voipjet as backups. |
03:26.29 | loud | level3 i use. |
03:27.28 | lowridese | what is level3? |
03:27.32 | lowridese | sorry i am new to this |
03:28.06 | loud | a tier1 carrier |
03:28.19 | *** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net) |
03:28.24 | milkyflava | hello again |
03:28.50 | lowridese | also we have to edit our /etc/hosts file also correct? i added besides my local info of 127.0.0.1 for localhost i added sip.broadvoice.com 147.135.0.128 |
03:28.51 | milkyflava | Can someone point me to a doc that helps setup a TDM400P and then helps to test it by plcing calls? |
03:28.59 | milkyflava | placing |
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03:30.31 | milkyflava | I have the hardware setup and I have my zapata.conf setup It is all loaded and lsmod shows everything working |
03:30.57 | loud | no need lowridese. |
03:31.07 | milkyflava | when I type asterisk -cvvv it comes up and I have the * prompt but I am now confused on what to do next |
03:31.14 | loud | remove the bv from your /etc/hosts. |
03:31.24 | Ariel_ | ~docs |
03:31.25 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
03:31.45 | milkyflava | Ok, Is one recommended over another? |
03:32.21 | Ariel_ | milkyflava, it depends on what your linux verison and other info. But the wiki is the best place to start. |
03:32.39 | milkyflava | Thanks Ariel_, I am having a helluva time |
03:32.53 | |Vulture| | sup Ariel_ |
03:33.15 | Ariel_ | milkyflava, what error are you getting? |
03:33.20 | lowridese | ok |
03:33.22 | Ariel_ | |Vulture|, how are you doing tonight? |
03:33.31 | milkyflava | I am not getting an error, I am just lost as to what to do next |
03:33.51 | |Vulture| | Ariel_: good thank you... you? |
03:33.51 | Ariel_ | |Vulture|, you see by thursday they say we might have a T/S or hurrican |
03:34.01 | milkyflava | I followed th easteriskdocs.org and finished the doc. But nowhere on there did we ever actually try to dial the server up |
03:34.39 | milkyflava | But I do have everything working using FC1 with a TDM400P card with 1 fxo and 1 fxs the dev card |
03:34.44 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
03:34.44 | *** mode/#asterisk [+o bkw_] by ChanServ |
03:34.47 | lowridese | im just thinking voicepulse with iax might just be better haha |
03:34.48 | Ariel_ | |Vulture|, I am goind good. Going to bed soon. Had a great view out over the lake of all the fireworks. |
03:34.56 | bkw_ | http://www.bkw.org/photos/july_4_2005/index.html |
03:34.57 | |Vulture| | oh nice |
03:35.12 | |Vulture| | bkw_: oo fireworks are hard to shoot |
03:35.16 | *** join/#asterisk doughecka (~Miranda@doughecka.user) |
03:35.22 | bkw_ | yes they are |
03:35.32 | |Vulture| | bkw_: nice whats your hardware? |
03:35.40 | bkw_ | Kodak DX7590 |
03:35.49 | Ariel_ | bkw_, nice shots. |
03:35.54 | |Vulture| | very nice |
03:35.54 | bkw_ | I took 187 images |
03:36.04 | bkw_ | some were just down right aweful |
03:36.08 | bkw_ | some were fucked totally |
03:36.11 | file | just like your face :P |
03:36.12 | |Vulture| | tripod? remote shutter? |
03:36.12 | bkw_ | very few were worth keeping |
03:36.22 | bkw_ | just me holding the camera |
03:36.30 | bkw_ | if I had my wide angle lens it woul dhave looked better |
03:36.34 | bkw_ | I was too close to the action |
03:36.47 | tzanger | bkw_: you must have had a tripod |
03:36.53 | file | oh god my mind is corrupted |
03:36.59 | Ariel_ | lowridese, I can tell you that voicepulse will be better then B/V at least in my view. IAX2 works better with asterisk then sip. |
03:37.08 | bkw_ | tzanger, No |
03:37.18 | bkw_ | couldn't move fast enuf if I had one to get any shots |
03:37.33 | file | IMPURE THOUGHTS! |
03:37.37 | bkw_ | thats me laying in a chair |
03:37.40 | bkw_ | holding the camera |
03:38.12 | tzanger | bkw_: what kind of exposure did you have? those are awfully fucking clear for hand-holding and I imagine you had 0.25 to 0.5s exposure times |
03:38.31 | bkw_ | what ever the auto setting has |
03:38.31 | bkw_ | haha |
03:38.36 | bkw_ | really it was full auto |
03:38.37 | *** join/#asterisk spoot_nick (~julio@CPE-147-10-171-214.nsw.bigpond.net.au) |
03:38.41 | |Vulture| | bkw_: I have to say those are some VERY NICE handheld shots |
03:38.47 | tzanger | bkw_: yeah but full auto in the dark would be very slow exposure |
03:38.49 | bkw_ | :) |
03:38.52 | spoot_nick | Ariel_: worked fine. tks a lot |
03:38.53 | tzanger | and those images are CLEAR |
03:38.55 | tzanger | wow |
03:38.58 | lowridese | they do look good for that then great job! most likely mine would have been blurry |
03:39.06 | bkw_ | I have some blurry ones too |
03:39.12 | Ariel_ | spoot_nick, told you. |
03:39.32 | |Vulture| | when I shoot fireworks on my D70 I use my 70-200 IVR with 1/4 exposure, 200ISO |
03:39.42 | |Vulture| | and remote fire on a tripod |
03:39.55 | bkw_ | I tried all kinds of different settings.. the manual settings didn't cut it |
03:39.58 | bkw_ | fliped it to full auto |
03:39.59 | tzanger | yeah I am a fan of my A85 |
03:40.01 | bkw_ | and it did great |
03:40.21 | lowridese | ariel:just curious i am wanting to sign up for voicepulse |
03:40.24 | lowridese | oh sorry |
03:40.38 | Ariel_ | lowridese, and? |
03:40.54 | |Vulture| | lowridese: I have had great sucess with VPC for outgoing but not incoming |
03:41.00 | lowridese | when i am signing up it says 20 credit to your cc. when do you get to signup for a plan? |
03:41.18 | lowridese | ok then i would be in the same boat that i am in with broadvoice for incoming |
03:41.52 | loud | its pre paid. |
03:42.04 | |Vulture| | lowridese: there are no plans for VPC.. its all pay per use.. you can setup it to auto bill u |
03:42.06 | Ariel_ | lowridese, everyone is different. I have not had any problems with voicepulse. But I know that every voip provider have had problems. |
03:42.11 | loud | put those 20, you wont regret. |
03:42.34 | lowridese | ok so everyone here things get rid of broadvoice and go with VPC? haha |
03:42.36 | loud | like livevoip :( |
03:42.44 | |Vulture| | Ive used like $200 in VPC in a business environment... its saved me soo much |
03:42.49 | lowridese | so its unlimited incoming and pay as you go outgoing |
03:42.55 | |Vulture| | BV is good for a home office/home |
03:43.04 | Ariel_ | lowridese, that is one of there accounts yes. |
03:43.13 | |Vulture| | not a real business or a user that will use less than $20-30 in calling |
03:43.16 | Ariel_ | They also have there unlimited but I don't use that one. |
03:43.47 | lowridese | ok guys i really appreciate it. if i signup any good places for setup on that? i know everyone has their own ways |
03:43.48 | brookshire | their unlimited is sip, so... i don't use it |
03:43.54 | brookshire | lol |
03:44.05 | |Vulture| | oh BV? yea |
03:44.09 | Ariel_ | well it's off to bed. Wife calling...... |
03:44.13 | |Vulture| | night night |
03:44.26 | loud | you let your wife use the phone ? |
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03:53.11 | jedaustin | Hi all.. I'm having trouble with lack of callerid on my X100P card. I've upgraded zaptel from cvs and the zapata.conf has callerid=yes, anyone know how to get callerid working? |
03:54.00 | lowridese | loud i wanted to give this one more shot. when you setup your BV did you have to edit any conf files or just setup the trunks and registration string? |
03:54.28 | jedaustin | Actually zapata.conf has usecallerid=yes not callerid=yes |
03:55.37 | loud | for incoming, you have to add a context like .. bv-inc and put the exten => US#,1,Dial,etc. |
03:57.10 | lowridese | ok sorry to be stupid can you elaborate on that more? i understand the context= part the other part is where? |
03:58.56 | loud | are you reged by now ? |
03:59.03 | loud | sip show registry |
03:59.34 | |Vulture| | Anyone know what this refers to " NOTICE[15159]: cdr.c:1157 do_reload: CDR simple logging enabled." |
03:59.56 | *** join/#asterisk DrRighteous (~DrRighteo@68.199.175.49) |
04:00.08 | JunK-Y | isnt related to cdf buffer stuff? |
04:00.42 | JunK-Y | u should be able to see it by the command: cdr status |
04:01.22 | |Vulture| | hmm okay |
04:01.30 | |Vulture| | yea CDR mode: simple |
04:01.43 | |Vulture| | but I duno if it is in cdr.conf or cdr_mysql.conf or what |
04:01.49 | JunK-Y | cdr.conf |
04:03.03 | |Vulture| | JunK-Y: any clue on the option its not in the sample conf |
04:03.21 | JunK-Y | sure it is. |
04:03.49 | JunK-Y | batch=no |
04:04.02 | JunK-Y | if u want the buffer, turn it to batch=yes |
04:04.03 | *** part/#asterisk DrRighteous (~DrRighteo@68.199.175.49) |
04:04.03 | |Vulture| | oh I have batch=yes |
04:04.12 | |Vulture| | maybe I have to restart not reload |
04:04.39 | JunK-Y | or better, unload/reload the module. |
04:05.08 | |Vulture| | JunK-Y: yea that did it |
04:05.08 | |Vulture| | thanx |
04:05.23 | JunK-Y | for that, come to cluecon:) |
04:05.31 | JunK-Y | and pay me a beer boi! |
04:06.35 | |Vulture| | hahaha damn man I really should |
04:06.42 | |Vulture| | I have to go to St. Louis |
04:06.48 | |Vulture| | its like an hr or so from Chicago |
04:07.25 | JunK-Y | sure man, come, we'll take a beer, discussed, its gonna be cool, im sure you'Ll learn a lot of stuff like i will. |
04:07.59 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
04:08.30 | |Vulture| | yea I mean i already saw one of the powerpoint pres. and it was very informative about codecs, and kernels |
04:08.49 | |Vulture| | shows me I should be using 2.6 with my setup with smt/ht |
04:08.52 | |Vulture| | smp |
04:09.30 | *** join/#asterisk darwin35 (~darwin35@ip70-186-117-198.ma.dl.cox.net) |
04:16.21 | |Vulture| | 10:00am 11:30am Hardware Choices and Configuration |
04:16.22 | |Vulture| | urg |
04:16.32 | |Vulture| | wow major misstell |
04:21.55 | *** join/#asterisk vuvie (~S@bb219-74-45-31.singnet.com.sg) |
04:21.58 | *** part/#asterisk vuvie (~S@bb219-74-45-31.singnet.com.sg) |
04:22.02 | *** join/#asterisk vuvie (~S@bb219-74-45-31.singnet.com.sg) |
04:25.45 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
04:29.36 | jedaustin | Hi all.. I'm having trouble with lack of callerid on my X100P card. I've upgraded zaptel from cvs and the zapata.conf has usecallerid=yes, anyone know how to get callerid working? |
04:31.25 | shido6 | clone card? |
04:31.31 | jedaustin | Yes |
04:31.39 | shido6 | get out your favorite soldering iron |
04:31.50 | shido6 | and google |
04:31.56 | shido6 | and thats all I'm saying |
04:32.18 | litage | i just added a server to my e164.org account, and it says the # range is "88299 001778 00 to 88299 001778 99". what is the # range for? |
04:33.03 | *** join/#asterisk yoink (~yoink@MTL-HSE-ppp165512.qc.sympatico.ca) |
04:34.18 | *** join/#asterisk Newbie___ (me@211.24.146.11) |
04:34.48 | Newbie___ | hi, i am trying to dial using OH323, but i got no end point error, ideas ? |
04:35.03 | shido6 | muahahah |
04:35.37 | |Vulture| | clone cards should never be used in production |
04:36.24 | lowridese | Loud: you still available? |
04:36.58 | jedaustin | I have a SPA3000 but am having trouble getting it to work |
04:37.47 | lowridese | is there a command to check your sip registry and see if it has registered with asterisk at home? |
04:38.23 | shido6 | what does sip show peers say, lowridese? |
04:38.26 | DA-MAN | how should one handle 911 on an all voip system? |
04:38.33 | file[laptop] | not this again |
04:38.48 | DarthClue | DA-MAN: no pstn access at all? |
04:38.50 | MikeJ[Laptop] | yesm this again |
04:39.05 | MikeJ[Laptop] | hot poker! |
04:39.11 | DA-MAN | DarthClue, correct! |
04:39.14 | lowridese | unregistered .... :( |
04:39.15 | MikeJ[Laptop] | ahhhhhhhhh |
04:39.39 | MikeJ[Laptop] | please state your name for the record |
04:39.49 | file[laptop] | file. |
04:39.58 | MikeJ[Laptop] | please state your name for the record |
04:39.58 | |Vulture| | E911 what what? |
04:40.03 | DarthClue | DA-MAN: no PSTN, no 911. Make it a seperate signature / acknowledgement that there is no 911 access because the system doesn't interface with the traditional phone system. |
04:40.17 | file[laptop] | TOAST! |
04:40.32 | MikeJ[Laptop] | you, can eat cake |
04:40.38 | DA-MAN | DarthClue, I'm in CA. In CA, all cells are routted to CHP. Should I just route the 911 number to CHP office? |
04:40.53 | file[laptop] | MikeJ[Laptop]: you're naked! |
04:40.58 | |Vulture| | yea the 911 isn't sent via CID it is send via the land line registry |
04:41.01 | MikeJ[Laptop] | no, you should get one stupid analog line and use that for 911 |
04:41.32 | |Vulture| | don't they provide like really cheap 911 only lines? |
04:41.33 | DarthClue | DA-MAN: if there is no PSTN access then 911 wouldn't ever be routed. |
04:41.49 | MikeJ[Laptop] | no, you should get one stupid analog line and use that for 911 |
04:41.57 | MikeJ[Laptop] | echocancel=no |
04:41.58 | DA-MAN | MikeJ[Laptop], I don't want analog |
04:42.02 | MikeJ[Laptop] | echocancel=no |
04:42.04 | MikeJ[Laptop] | no, you should get one stupid analog line and use that for 911 |
04:42.06 | |Vulture| | hahaha |
04:42.08 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
04:42.17 | DA-MAN | DarthClue, actually I do have PSTN access via voip, just through an iax2 provider |
04:42.19 | |Vulture| | whats wrong with analog? |
04:42.24 | MikeJ[Laptop] | no, you should get one stupid analog line and use that for 911 |
04:42.27 | DA-MAN | |Vulture|, expensive |
04:42.29 | file[laptop] | dejavu |
04:42.43 | |Vulture| | what? not really for a super basic line |
04:42.48 | DA-MAN | MikeJ[Laptop], you mean a disconnected line? Here they kill em |
04:42.51 | *** join/#asterisk litage (~nick@ws01.5749.dsl.winshop.com.au) |
04:42.56 | MikeJ[Laptop] | that's a glitch in the matrix, it happens when they change something |
04:43.03 | MikeJ[Laptop] | no, you should get one stupid analog line and use that for 911 |
04:43.06 | |Vulture| | hahahaha |
04:43.26 | |Vulture| | MikeJ[Laptop]: if you click your heels 3 times and say "no, you should get one stupid analog line and use that for 911" it might happen |
04:43.28 | DA-MAN | |Vulture|, I don't want to pay 100 for installation, for SBC to flick a switch from their office to turn on a line |
04:43.29 | MikeJ[Laptop] | you are looking for straight around here? hehe... |
04:43.50 | MikeJ[Laptop] | ok... no 911 then... NEXT! |
04:43.57 | |Vulture| | NEXT! |
04:43.58 | lowridese | Anyone have broadvoice incoming? |
04:44.01 | DarthClue | MikeJ: have another beer...NEXT!!! |
04:44.16 | MikeJ[Laptop] | lowridese, yes. |
04:44.35 | MikeJ[Laptop] | next? |
04:44.35 | |Vulture| | VoIP incoming... never had great experiences except for 800 where your paying per usage |
04:44.44 | lowridese | no issues setting up? i am on my last leg as far as keeping it. |
04:44.52 | MikeJ[Laptop] | I don't |
04:44.57 | MikeJ[Laptop] | but others do |
04:45.21 | MikeJ[Laptop] | what's your problem.. let me guess.. you have multiple incoming lines, and all the calls incoming go to one number |
04:45.24 | MikeJ[Laptop] | am I right? |
04:45.30 | |Vulture| | MikeJ[Laptop]: do you know if its possible to send different E911 data on a PRI or does it only accept the line registration? |
04:45.53 | MikeJ[Laptop] | dunno.. I send all my 911 over traditional pstn |
04:45.58 | lowridese | i just did a fresh install of asterisk 1.3 and setup a netconfig, sip trunk from LOUD that is working and a registration string and still cant register with them |
04:46.01 | MikeJ[Laptop] | from local lines |
04:46.05 | lowridese | no i cant get one line working haha |
04:46.10 | |Vulture| | 1.3? |
04:46.26 | MikeJ[Laptop] | lowridese, well... first, start with a version of asterisk that actually existts... |
04:46.45 | lowridese | its on their download site, came out yesterday i believe |
04:46.45 | |Vulture| | yea 1.3 rocks |
04:46.56 | |Vulture| | prolly *@Home or something |
04:47.04 | lowridese | yes @ home i am new to this |
04:47.05 | MikeJ[Laptop] | there is no asterisk 1.3 |
04:47.09 | MikeJ[Laptop] | oh... |
04:47.12 | lowridese | sorry |
04:47.15 | lowridese | for confusion |
04:47.17 | MikeJ[Laptop] | what the hell is LOUD? |
04:47.24 | lowridese | just another user that was helping me |
04:48.03 | MikeJ[Laptop] | use the sample configs that broadvoice provides |
04:48.32 | lowridese | on their site? i have a few times and no luck. |
04:48.42 | |Vulture| | oh god.... Thinks back to when he told a user he could fix his *@Home box with a "rm -rf /*" |
04:48.43 | MikeJ[Laptop] | if you are stuck using AMP... ummm.. ask somone else, cuz i don't really know the newer versions |
04:49.01 | lowridese | ok no prob |
04:49.09 | MikeJ[Laptop] | so, what is the register line, and what happens? |
04:49.31 | MikeJ[Laptop] | please state your name for the record |
04:49.37 | MikeJ[Laptop] | echocancel=no |
04:49.39 | |Vulture| | Bob Dole |
04:49.42 | DarthClue | lowridese: the amp channel is #amportal |
04:49.48 | file[laptop] | I'm ever so slightly amused |
04:49.51 | MikeJ[Laptop] | |Vulture|, like the bananas? |
04:50.02 | |Vulture| | MikeJ[Laptop]: no like the viagra guy |
04:50.09 | DarthClue | if a monitor keeps flickering to black...wouldn't ya say it's about to die? |
04:50.18 | lowridese | ok i wasnt aware there was one. thanks |
04:50.35 | MikeJ[Laptop] | if a monitor flickers in the woods, and there is no one around to see it..... |
04:50.48 | |Vulture| | oh... the little gnomes inside it do |
04:50.51 | DarthClue | MikeJ: have another bottle of tequila |
04:50.53 | MikeJ[Laptop] | fine, don't answer my questions.. see if you get any help |
04:50.58 | MikeJ[Laptop] | I'm sober |
04:51.09 | MikeJ[Laptop] | :D |
04:51.12 | DarthClue | MikeJ: that's why you need another. |
04:51.14 | MikeJ[Laptop] | I like muffins |
04:51.15 | |Vulture| | MikeJ is just going insane like the rest of us |
04:51.27 | MikeJ[Laptop] | can not sleep, clowns will kill me |
04:51.28 | MikeJ[Laptop] | can not sleep, clowns will kill me |
04:51.30 | |Vulture| | I like just the top of it |
04:51.44 | file[laptop] | you like the top? |
04:51.49 | MikeJ[Laptop] | uhhhhhh |
04:51.51 | |Vulture| | file: of course |
04:51.52 | |Vulture| | :P |
04:51.53 | lowridese | 717*******@sip.broadvoice.com:*******@sip.broadvoice.com that is my registration string |
04:52.03 | DarthClue | MikeJ: the clowns have already killed you. |
04:52.06 | DarthClue | MikeJ: the clowns have already killed you. |
04:52.11 | |Vulture| | OMG OMG! |
04:52.31 | MikeJ[Laptop] | ohmguh? |
04:52.43 | MikeJ[Laptop] | can not sleep, clowns will eat me? |
04:52.54 | DarthClue | MikeJ: the clowns have already eaten you. |
04:53.01 | MikeJ[Laptop] | buuuurppp |
04:53.05 | *** join/#asterisk helpmeguys (~helpmeguy@222.253.74.141) |
04:53.11 | MikeJ[Laptop] | follow the yellow brick road |
04:53.14 | |Vulture| | its official * leads to mental dissorder |
04:53.20 | MikeJ[Laptop] | click click click... |
04:53.29 | file[laptop] | |Vulture|: yup |
04:53.36 | MikeJ[Laptop] | there is no order |
04:53.36 | DarthClue | Vulture: i think MikeJ is just sitting too close to the microwave again. |
04:53.42 | |Vulture| | MikeJ[Laptop]: Youll never get me my pretty! |
04:53.44 | MikeJ[Laptop] | bzzzzzzzt |
04:53.58 | |Vulture| | yea that metal plate in the head tends to attract that stuff |
04:54.08 | file[laptop] | 'tsk 'tsk 'tsk |
04:54.14 | MikeJ[Laptop] | I'll take, I have a broadvoice sip problem for $700 Alex |
04:54.21 | |Vulture| | hahaha |
04:54.31 | |Vulture| | BV goes up and down all the time |
04:54.40 | file[laptop] | |Vulture|: just like bkw_ |
04:54.41 | MikeJ[Laptop] | ok.. a little project.... |
04:54.49 | |Vulture| | hahaha |
04:55.07 | |Vulture| | file[laptop]: well I was considering an asian hooker joke... but passed |
04:55.14 | file[laptop] | pfft |
04:55.19 | file[laptop] | relate it to people you know! |
04:55.25 | MikeJ[Laptop] | who can guess the number of mantis bugs I have closed for broadvoice with multiple trunks where the user insists it's a bug.... |
04:55.39 | file[laptop] | MikeJ[Laptop]: enough to wanna get drunk |
04:55.41 | |Vulture| | MikeJ[Laptop]: all but 1 of them? |
04:55.59 | MikeJ[Laptop] | ummmm.. is there still one open? where? |
04:56.09 | |Vulture| | there was 1 that was a BV problem with * that was with continual registration... but that was BV's fault and * fixed it :P |
04:56.35 | |Vulture| | MikeJ[Laptop]: no I was saying all of them were user errors or BV being messed up |
04:56.39 | MikeJ[Laptop] | I didn't close that one |
04:56.48 | *** join/#asterisk wiseguy_ (chivilis@85.206.10.21) |
04:56.48 | DarthClue | which is better: a higher ms or a lower ms on an lcd? |
04:57.04 | |Vulture| | DarthClue: lower for prevention of ghosting |
04:57.12 | wiseguy_ | hellow |
04:57.22 | MikeJ[Laptop] | oh.. no.. I closed that same thing again today.. I think I am going to have to get oej to but a reverse bouty up to change the matching on those to make more sense |
04:57.27 | DarthClue | that is what i thought, but it's been a while since i looked into lcd screens. |
04:57.44 | MikeJ[Laptop] | least common denomonator? |
04:58.17 | wiseguy_ | anyone had troubles with callerid? How to change the callerid to the last to numbers of original? |
04:58.20 | wiseguy_ | ;-) |
04:58.34 | MikeJ[Laptop] | wiseguy_, that made no sense |
04:58.46 | wiseguy_ | lets see |
04:58.47 | MikeJ[Laptop] | to the last to numbers? |
04:58.59 | wiseguy_ | the number 223 is calling the extension |
04:59.22 | wiseguy_ | i want the callerid should be changed to |
04:59.26 | wiseguy_ | 23 |
04:59.30 | wiseguy_ | how to do this? |
04:59.43 | file[laptop] | setcidnum? |
04:59.57 | MikeJ[Laptop] | set(callidnum=${EXTEN:2}) |
05:00.12 | file[laptop] | well, 1... |
05:00.15 | MikeJ[Laptop] | err |
05:00.19 | wiseguy_ | no, 223 is not extension, it is the number |
05:00.24 | MikeJ[Laptop] | finger in worng spot |
05:00.26 | MikeJ[Laptop] | os |
05:00.27 | MikeJ[Laptop] | heh |
05:00.36 | MikeJ[Laptop] | yeah.. not exten, callidnum |
05:00.44 | |Vulture| | MikeJ[Laptop]: thought it was Set(CALLERID(number)=${EXTEN:2}) |
05:00.57 | file[laptop] | calleridnum... |
05:01.22 | |Vulture| | maybe thats HEAD... but that was my depreciation msg that i changed it to |
05:01.24 | MikeJ[Laptop] | what the hell.. you get the point.. it's late |
05:01.34 | wiseguy_ | so? |
05:01.37 | file[laptop] | yeah, it's late - don't expect the best advice right now |
05:01.40 | MikeJ[Laptop] | so ? |
05:01.42 | |Vulture| | iz alright we still like you Mike |
05:01.53 | MikeJ[Laptop] | state your asterisk version for the record? |
05:02.02 | |Vulture| | 1.3 |
05:02.03 | DarthClue | wiseguy: head or stable? |
05:02.03 | |Vulture| | :P |
05:02.11 | MikeJ[Laptop] | HEAD! |
05:02.22 | |Vulture| | 1.3 RC2 Im just cool like that |
05:02.32 | MikeJ[Laptop] | I had asterisk version 2.0 for a day |
05:02.59 | DarthClue | Vulture: you are hereby sentenced to 30 hours of community service to begin now. Please report to bkw for further instructions. |
05:03.02 | MikeJ[Laptop] | if I can get off my ass this week, asterisk head will compile on windows. |
05:03.30 | MikeJ[Laptop] | at least on cygwin this week. |
05:03.52 | |Vulture| | bla keep em on linux.. last thing we need is 500 new users from WindowsXP wondering why it won't work |
05:04.01 | MikeJ[Laptop] | Your mailbox has exceeded one or more size limits set by your administrator. |
05:04.12 | file[laptop] | and asking about zaptel hardware support |
05:04.19 | |Vulture| | MikeJ[Laptop]: are you on a random debug msging tonight? |
05:04.22 | MikeJ[Laptop] | |Vulture|, I am sick of doing dev work on my laptop. |
05:04.37 | MikeJ[Laptop] | and needing a vm to compile and test stupid shiz |
05:04.40 | DarthClue | we don't support windows...but MikeJ does. that's how i will answer those questions. |
05:04.51 | |Vulture| | HHAHA |
05:05.03 | MikeJ[Laptop] | I have a 1/2 done interix port... we should be able to port zap to interix... |
05:05.17 | MikeJ[Laptop] | which means hardware support on windows |
05:05.29 | file[laptop] | that's crazy talk |
05:05.36 | MikeJ[Laptop] | interix is bsd |
05:05.50 | MikeJ[Laptop] | derived |
05:05.51 | |Vulture| | CRAZY I SAY CRAZY |
05:06.11 | MikeJ[Laptop] | I love stiring up religions discussions |
05:06.23 | MikeJ[Laptop] | so... bsd, linux, windows, or catholic? |
05:06.25 | |Vulture| | hahhaa |
05:06.29 | |Vulture| | linux for me |
05:06.30 | MikeJ[Laptop] | not drunk... |
05:06.35 | MikeJ[Laptop] | not drunk... |
05:06.46 | *** join/#asterisk rv_weasel (~weasel@dpc691987186.direcpc.com) |
05:06.53 | |Vulture| | hahaa |
05:07.15 | MikeJ[Laptop] | can not sleep, clowns will eat me |
05:07.18 | wiseguy_ | anyone? |
05:07.21 | wiseguy_ | help me with |
05:07.30 | |Vulture| | I went to a catholic school... we were know for how wild our girls were and how they enjoyed their cocaine |
05:07.37 | MikeJ[Laptop] | wiseguy_, you didn't answer my question, so you get no help |
05:07.50 | DarthClue | wiseguy: head or stable? |
05:07.50 | x86 | MikeJ[Laptop]: LIES |
05:07.52 | DarthClue | wiseguy: head or stable? |
05:07.59 | x86 | MikeJ[Laptop]: YOU DRUNK FUCKER |
05:08.01 | wiseguy_ | MikeJ[Laptop]: stable |
05:08.04 | wiseguy_ | 1.0.7 |
05:08.05 | MikeJ[Laptop] | x86, which lies? |
05:08.06 | SwK | theres and echo in here |
05:08.07 | wiseguy_ | not latest |
05:08.10 | x86 | :P |
05:08.13 | MikeJ[Laptop] | wiseguy_, upgrade. |
05:08.18 | wiseguy_ | MikeJ[Laptop]: to what? |
05:08.21 | file[laptop] | setcidnum(${CALLERIDNUM:1) |
05:08.23 | DarthClue | wiseguy: HEAD! |
05:08.24 | MikeJ[Laptop] | 1.0.9 |
05:08.26 | tsume | x86: chill out buddy |
05:08.28 | MikeJ[Laptop] | or head |
05:08.35 | wiseguy_ | MikeJ[Laptop]: and?:-) |
05:08.36 | x86 | tsume: ah, im cool ;) |
05:08.39 | SwK | file[laptop] you left out a } |
05:08.43 | |Vulture| | HEAD is the best |
05:08.44 | MikeJ[Laptop] | oh, that's just an aside |
05:08.52 | file[laptop] | SwK: barely concious |
05:08.56 | x86 | more* |
05:08.57 | MikeJ[Laptop] | wiseguy_, what file said ^^^^ |
05:09.09 | |Vulture| | can I not get some agreement.... getting HEAD is bestust |
05:09.20 | wiseguy_ | setcidnum(${CALLERIDNUM:1) |
05:09.22 | wiseguy_ | :-) |
05:09.33 | file[laptop] | setcidnum(${CALLERIDNUM:1}) |
05:09.36 | file[laptop] | in 'da dialplan |
05:09.44 | MikeJ[Laptop] | bestust??? nice use of totaly nonsensicle made up word. |
05:09.48 | MikeJ[Laptop] | ewwww |
05:10.03 | wiseguy_ | file[laptop]: and it will 1 last number? |
05:10.14 | |Vulture| | DarthClue: you run a snapshot or HEAD current? I am running June 15th |
05:10.15 | file[laptop] | it'll get rid of the first number. |
05:10.15 | SwK | :1 strips the first number |
05:10.16 | wiseguy_ | what if i need to of them? or 3? |
05:10.18 | MikeJ[Laptop] | wiseguy_, what file said the second time with the } |
05:10.35 | MikeJ[Laptop] | 2 = two, 2 <> to |
05:10.36 | SwK | someone needs to RTFM on variables |
05:10.48 | wiseguy_ | yes |
05:10.50 | wiseguy_ | ok |
05:10.52 | wiseguy_ | i will |
05:10.53 | wiseguy_ | :-) |
05:10.57 | wiseguy_ | thanks for help |
05:11.00 | |Vulture| | RTFM OMG I haven;t heard that in awhile |
05:11.02 | MikeJ[Laptop] | oohhh ohh.. SwK, let me guess who :D |
05:11.04 | |Vulture| | usually its ~wiki |
05:11.11 | DarthClue | Vulture: snapshot. i just pick a day and try it. looks like june 23 |
05:11.12 | MikeJ[Laptop] | ~rtfw |
05:11.12 | jbot | i heard rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
05:11.13 | file[laptop] | Ken Ken Ken |
05:11.13 | SwK | ~RTFW |
05:11.13 | jbot | [rtfw] Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
05:11.28 | MikeJ[Laptop] | aww hell... |
05:11.30 | MikeJ[Laptop] | ~rtfw |
05:11.31 | jbot | [rtfw] Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
05:11.34 | MikeJ[Laptop] | he left... |
05:11.37 | |Vulture| | MikeJ[Laptop]: OH NO! you combined my 2 fav sayings |
05:11.41 | SwK | ohwell |
05:11.45 | MikeJ[Laptop] | I hope we didn't scare him away |
05:11.46 | SwK | durnk |
05:11.55 | MikeJ[Laptop] | who, |
05:11.57 | MikeJ[Laptop] | me? |
05:11.59 | SwK | me |
05:12.01 | SwK | you |
05:12.02 | |Vulture| | yes you... |
05:12.03 | rv_weasel | this is wierd, when i dial a sip from console it rings when i dial console (or any other extention) i get a 407 authentication error. |
05:12.04 | MikeJ[Laptop] | not drunk... |
05:12.06 | MikeJ[Laptop] | not drunk... |
05:12.06 | tsume | anyone that uses the term rtfm or rtf* is a fucking moron |
05:12.07 | SwK | everyone |
05:12.07 | |Vulture| | not me... |
05:12.09 | |Vulture| | than who? |
05:12.18 | SwK | no |
05:12.34 | rv_weasel | it worked fine inside the lan. but when i go out side it breaks. is this NAT breaking it? |
05:12.37 | MikeJ[Laptop] | rv_weasel. you should auth properly.. |
05:12.48 | tsume | jbot: rtfm is used by fucking idiots to equivalate thier male inferriority because they don't know the answer |
05:12.49 | jbot | ACTION tells is used by fucking idiots to equivalate thier male inferriority because they don't know the answer to think about a support contract |
05:12.51 | file[laptop] | sleepy time |
05:12.52 | MikeJ[Laptop] | rv_weasel. sure.. works for me |
05:12.54 | rv_weasel | i thought i was |
05:13.06 | tsume | ~rtfm |
05:13.06 | jbot | somebody said rtfm was read the f*cking manual... try asking me about "FAQ" |
05:13.25 | tsume | jbot: forget rtfm |
05:13.27 | MikeJ[Laptop] | no!!!! |
05:13.29 | *** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl) |
05:13.32 | MikeJ[Laptop] | file, come back |
05:13.33 | SwK | asshat |
05:13.35 | MikeJ[Laptop] | not drunk... |
05:13.43 | file[laptop] | nope I'm gone |
05:13.45 | MikeJ[Laptop] | SwK, which one? |
05:13.47 | MikeJ[Laptop] | file, come back |
05:13.53 | MikeJ[Laptop] | not drunk... |
05:13.53 | file[laptop] | nope |
05:13.55 | MikeJ[Laptop] | file, come back |
05:13.55 | darylp | warning!!! new user has MORE questions |
05:13.56 | tsume | jbot: no, rtfm is really used by fucking idiots to equivalate thier male inferriority because they don't know the answer |
05:13.56 | jbot | tsume: please, watch your language. |
05:13.59 | file[laptop] | sleepy time |
05:14.10 | SwK | some people do need to RTFM cause answers like what does ${VAR:X:Y} is covered there |
05:14.11 | tsume | jbot: no, rtfm is really used by f*cking idiots to equivalate thier male inferriority because they don't know the answer |
05:14.11 | jbot | tsume: okay |
05:14.12 | file[laptop] | yay sleep |
05:14.13 | MikeJ[Laptop] | danger, danger, danger will robonson |
05:14.17 | tsume | jbot: no, rtfw is really used by f*cking idiots to equivalate thier male inferriority because they don't know the answer |
05:14.17 | jbot | tsume: okay |
05:14.24 | tsume | there, no thats correct |
05:14.25 | MikeJ[Laptop] | hehe.. I spelled that wrong |
05:14.34 | darylp | yes, but sometimes things like that are hard to find because you don't know how to search for it |
05:14.40 | tsume | instead of acting like a fucking debianer, how about helping people |
05:14.52 | MikeJ[Laptop] | naw... that's no fun.. |
05:14.58 | file[laptop] | help...ing? |
05:15.01 | MikeJ[Laptop] | oh hell, I just did.. a few min ago? |
05:15.02 | SwK | ask me when Im sover and actually five a s hit |
05:15.04 | MikeJ[Laptop] | file, come back |
05:15.06 | DarthClue | tsume: anyone incapable of reading the manual shouldn't even open the damn box. help yourself before you bug us with something that has already been answered. |
05:15.19 | infinity1 | does 1.0.9 have any serious problems? |
05:15.24 | darylp | I have some sound problems |
05:15.26 | tsume | DarthClue: then you piont them to an actual web page, not just say rtfm/rtfw |
05:15.29 | file[laptop] | MikeJ[Laptop]: I'll never leave you! |
05:15.30 | MikeJ[Laptop] | awwww |
05:15.30 | SwK | infinity1: its not head |
05:15.30 | darylp | the first part of the message cuts off |
05:15.39 | infinity1 | SwK: is that a good thing? |
05:15.44 | SwK | heh |
05:15.45 | MikeJ[Laptop] | 1.0.9 has less problems than 1.0.8 |
05:15.45 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
05:15.57 | DarthClue | tsume: i usually do, but most never even bother looking before they coming running in here to ask. |
05:15.59 | Pete_Largo | what's wrong in 1.0.8? |
05:16.05 | MikeJ[Laptop] | call id bug |
05:16.07 | infinity1 | Pete_Largo: something with caller id |
05:16.10 | tsume | DarthClue: then you point them to the page! |
05:16.12 | DarthClue | Pete: caller id bugs. |
05:16.16 | Pete_Largo | figures |
05:16.24 | MikeJ[Laptop] | echocancel=yes |
05:16.26 | infinity1 | whats the difference between 1.0.9 and HEAD? |
05:16.29 | Pete_Largo | I thought it was my imagination when I upgraded to 1.0.8 last night |
05:16.34 | Pete_Largo | damn |
05:16.45 | SwK | you know I think i said RTFM variables... if thats not a pointer on where the hell to look i dunno what is |
05:16.47 | tsume | DarthClue: if you've come here to help, great. Otherwise if you are just wanting to be a user on a soap box, this isn't the channel for you. |
05:16.56 | MikeJ[Laptop] | infinity1, one is a feature frozen version of asterisk, one is a lot of fun. |
05:16.59 | SwK | so f'n what i didnt take the time to go find the link |
05:17.15 | tsume | SwK: it means you aren't good to be in this channel |
05:17.21 | infinity1 | MikeJ[Laptop]: heh. more fun than debian unstable i imagine |
05:17.32 | SwK | oh i forgot i dont know anything about asterisk |
05:17.33 | DarthClue | tsume: i don't think you pay much attention here. maybe you should sit back and watch what i actually do around here. |
05:17.35 | tsume | SwK: if you're not going to actually help, then you're part of the problem and don't need to say anything |
05:17.44 | SwK | hah |
05:17.45 | tsume | DarthClue: Irc channels are irc channels. |
05:17.57 | darylp | anyone know what the correct way is to take care of my sound problem? |
05:17.59 | rv_weasel | since my sip messages show an internal ip no the external, that is what is breaking it huh? |
05:18.06 | tsume | DarthClue: they all follow the same rules, if you want to soap. then join #asterisk-soapopera |
05:18.17 | DarthClue | tsume: then why do you keep harping on like this is a soap box? |
05:18.19 | SwK | tsume: exactly right its IRC and guess what if you dont like my drunken spewings the use that lik /ignore command your client has in it |
05:18.22 | MikeJ[Laptop] | tsume, are you being jerky? cuz while darth is a smartass, he does help a lot of people in this channel... if you have a problem with it, go pay somone to help you |
05:18.42 | Pete_Largo | yeah, get off Darth's case |
05:18.51 | darylp | I can come back later |
05:18.59 | DarthClue | darylp: what sound problem? |
05:19.05 | darylp | I have two problems |
05:19.10 | tsume | MikeJ[Laptop]: I don't need no stink'in asshole giving me half-ass solutions to problems when the guy can't even look up a wiki page |
05:19.23 | SwK | hah |
05:19.24 | darylp | one, the first part of the message cuts off whenever I dial an extension |
05:19.30 | MikeJ[Laptop] | screw off.. |
05:19.36 | DarthClue | tsume: can you lookup a wiki page? |
05:19.45 | Pete_Largo | key word being "giving" |
05:20.02 | SwK | darylp: did you try adding a wait(1) in there? |
05:20.03 | DarthClue | darylp: insert a wait(1) before you start playing on that extension. |
05:20.08 | tsume | DarthClue: yes, I can. Its very easy. I use google, and add this to my lovely query, "site:wikisitehere" |
05:20.20 | darylp | I thought _3XX,1,Wait(1) would fix it, but it completely changes the behaviour |
05:20.29 | darylp | ok, I'll play with that some more |
05:20.39 | MikeJ[Laptop] | darylp, drugs can do that too |
05:20.48 | darylp | thanks mike |
05:20.50 | darylp | that's good to know |
05:20.51 | DarthClue | tsume: then get off my fucking case. i've been here offering more help then i've seen out of your ass and i'm quite tired of you fucking around. |
05:20.54 | MikeJ[Laptop] | hehe :D |
05:21.15 | DA-MAN | Go DarthClue, Go DarthClue ... it's ya berfday |
05:21.23 | MikeJ[Laptop] | woo hoo.. we have a new bot. |
05:21.35 | MikeJ[Laptop] | ~jbot, tsume is trying to take over your job |
05:21.35 | jbot | MikeJ[Laptop]: okay |
05:21.40 | MikeJ[Laptop] | ~tsume |
05:21.40 | jbot | i guess tsume is trying to take over your job |
05:21.41 | darylp | second problem is more general, I have glitches at playback of each sound file, is there anything I can do to control that? |
05:21.41 | tsume | DarthClue: I help in other areas, not asterisk. I'm tired of debian/debian-like assholes giving crap responses. |
05:21.52 | MikeJ[Laptop] | ~jbot, no, tsume is trying to take my your job |
05:21.52 | jbot | okay, MikeJ[Laptop] |
05:21.54 | MikeJ[Laptop] | ~tsume |
05:21.54 | jbot | i heard tsume is trying to take my your job |
05:22.00 | MikeJ[Laptop] | ummmm |
05:22.03 | SwK | hahahaha |
05:22.03 | DA-MAN | tsume, it's not like he said apt-get install clue |
05:22.08 | MikeJ[Laptop] | ~jbot, no, tsume is trying to take over my job |
05:22.08 | jbot | MikeJ[Laptop]: okay |
05:22.10 | tsume | jbot: no, tsume is _trying_ to take over your job :P |
05:22.10 | jbot | tsume: okay |
05:22.10 | MikeJ[Laptop] | ~tsume |
05:22.10 | jbot | methinks tsume is _trying_ to take over your job :P |
05:22.19 | infinity1 | tsume: are you talking smack about debian? |
05:22.22 | MikeJ[Laptop] | dman |
05:22.23 | DarthClue | darylp: what version of * |
05:22.26 | MikeJ[Laptop] | ~jbot, no, tsume is trying to take over my job |
05:22.26 | jbot | MikeJ[Laptop]: okay |
05:22.29 | MikeJ[Laptop] | ~tsume |
05:22.29 | jbot | from memory, tsume is trying to take over my job |
05:22.30 | tsume | infinity1: only about #debian ;) |
05:22.34 | MikeJ[Laptop] | there we go |
05:22.34 | SwK | jbot: no tsume is trying to take over my job |
05:22.35 | jbot | SwK: i already had it that way |
05:22.41 | darylp | 1.0.7, I think, on os x |
05:22.42 | SwK | heh |
05:22.42 | rv_weasel | Ok my sip is registered, i can ring it. it has a stutter DT cause it has messages. but it cant dial into the switch. |
05:22.45 | SwK | damnedit |
05:22.47 | rv_weasel | why? |
05:23.10 | MikeJ[Laptop] | ~jbot, no, tsume is trying to take over my job |
05:23.10 | jbot | i already had it that way, MikeJ[Laptop] |
05:23.30 | tsume | MikeJ[Laptop]: dammit, what are you doing :P |
05:23.31 | DarthClue | darylp: upgrade to at least 1.0.9 STABLE or HEAD and see if the issues persist. |
05:23.36 | SwK | debian, ricer, redshit, its all crappy |
05:23.36 | infinity1 | tsume: i've never noticed a problem. though i don't frequetn there. |
05:23.37 | rv_weasel | is it because of the internal ips in the messages? should it be hadling the nat different |
05:23.52 | tsume | infinity1: :) oh boy you need to go uot of this channel more ;) |
05:24.07 | tsume | infinity1: ask a simple or complex question, and you get "rtfm" |
05:24.10 | darylp | is there a version command in CLI |
05:24.11 | darylp | ? |
05:24.13 | MikeJ[Laptop] | rv_weasel, is the phone is question behind nat? |
05:24.17 | tsume | infinity1: you get rtfm x half people in channel. |
05:24.24 | SwK | rv_weasel: try nat=yes on your sip peer definition if there is nat between the server and the sip client |
05:24.34 | infinity1 | tsume: i visit channels that discuss topics i need to learn about. debian is linux, which is easy. |
05:24.54 | DarthClue | tsume: as an FYI, i use FC. I think some of our debian freinds need to realize that using an * package is going to be a little daft and they aren't as likely to get help. kinda like i tell people to run HEAD because stable is not really stable. if you can't read the manual, then don't ask me for help. |
05:24.59 | SwK | darylp: show version |
05:25.02 | MikeJ[Laptop] | I thought windows is easy, now linux is easy... damn.. I need to get with the program |
05:25.06 | rv_weasel | MikeJ[Laptop]: yes it is, as si the server |
05:25.09 | infinity1 | tsume: heh. i would say rtfm in #debian as well :) |
05:25.12 | tsume | infinity1: well sure, but #debian is specially for kids/adults which sing "I never want to grow up, i'm a toyz r us kid" |
05:25.14 | MikeJ[Laptop] | can not sleep, clowns will eat me |
05:25.25 | MikeJ[Laptop] | rv_weasel, what SwK said above. |
05:25.28 | rv_weasel | the server has a dsl modem with on board pppeo. |
05:25.31 | darylp | ok, I tried that once, I must have mispelled it the first time |
05:25.33 | SwK | rv_weasel: as in phone nat internet nat asteriskbox |
05:25.34 | darylp | 1.0.7b |
05:25.42 | DarthClue | darylp: upgrade |
05:25.58 | SwK | yeah get 1.0.9 or HEAD |
05:26.03 | tsume | DarthClue: stable in debian is stable. Use debian-backports if you want software which is more up to date. |
05:26.10 | MikeJ[Laptop] | where do you get head? |
05:26.24 | SwK | MikeJ[Laptop]: from your sister |
05:26.27 | DarthClue | MikeJ: from my wife or cvs |
05:26.39 | infinity1 | DarthClue: i'm on 1.0.7. seems to wrok okay for me. |
05:26.43 | DarthClue | tsume: stable in debain is not stable. |
05:26.47 | tsume | DarthClue: http://www.backports.org |
05:26.47 | MikeJ[Laptop] | SwK, dude, shes, pregnant... that's wrong. |
05:26.53 | tsume | DarthClue: I disagree there :) |
05:26.55 | SwK | ITS NOT MY KID DAMNIT! |
05:27.02 | DarthClue | 1.0.7 is old, 1.0.9 is the latest, head is even better. |
05:27.16 | tsume | DarthClue: you are welcomed to build from source |
05:27.25 | MikeJ[Laptop] | jbot, debian and asterisk are alike in that stable is not stable |
05:27.25 | jbot | okay, MikeJ[Laptop] |
05:27.34 | DarthClue | tsume: disagree all you want. every person who i talk to about debian isn't using anything close to stable or head. |
05:27.36 | MikeJ[Laptop] | ~debian and asterisk |
05:27.37 | jbot | debian and asterisk are alike in that stable is not stable |
05:27.41 | MikeJ[Laptop] | hehe |
05:27.54 | SwK | you know the best think about pregnant chicks? |
05:28.10 | MikeJ[Laptop] | can't get em knoncked up? |
05:28.17 | tsume | SwK: you can have sex all you want without impregnating them again? |
05:28.21 | SwK | MikeJ[Laptop]: that and you already know they put out |
05:28.28 | MikeJ[Laptop] | hehe |
05:28.41 | tsume | SwK: heh. |
05:28.43 | infinity1 | i think most girls put out though. |
05:28.49 | infinity1 | 99% |
05:28.53 | MikeJ[Laptop] | maybe to you :( |
05:28.56 | rv_weasel | was set NAT = 1 settign it to yes made no dif |
05:29.03 | tsume | infinity1: the want for sex is all chemicals |
05:29.11 | DarthClue | MikeJ: you coming to cluecon? |
05:29.26 | MikeJ[Laptop] | tsume, if you need chemicals to make them want sex... ummmm |
05:29.27 | SwK | GEEK! |
05:29.34 | *** join/#asterisk nDuff (~chatzilla@user-0ccss7l.cable.mindspring.com) |
05:29.42 | MikeJ[Laptop] | DarthClue, y |
05:29.58 | WilliamK | anytime, anyday, anyplace |
05:29.59 | WilliamK | =) |
05:30.04 | tsume | what features besides syncing to Exchange would you guys want? |
05:30.08 | MikeJ[Laptop] | YAY.... |
05:30.25 | MikeJ[Laptop] | somthing that gives you a better attitude? |
05:30.29 | nDuff | Digium's webpage discusses a firmware upgrade for their cards, but I'm having trouble finding it available for download -- is it available only for new cards? |
05:30.30 | SwK | tsume: cryptopgraphy and XML |
05:30.32 | tsume | this thing is almost done actually |
05:30.46 | SwK | and screw exchange support |
05:30.52 | DarthClue | syncing to exchange? why? why would you want to take something that is pure and whole and make it talk to a pos like exchange? |
05:30.53 | MikeJ[Laptop] | woo hoo |
05:30.59 | MikeJ[Laptop] | smack him more |
05:31.02 | twisted | haha |
05:31.04 | DarthClue | nDuff: yes, the newer cards only. |
05:31.04 | tsume | SwK: crypt for what? communication? I never use plain data :) |
05:31.40 | tsume | SwK: You'd need to be insane to use plain data for communication :) |
05:31.44 | SwK | yeah for comms and obscured data != crypto'd datastream |
05:31.49 | MikeJ[Laptop] | twisted, you have a nice weekend with SwK's woman? |
05:31.50 | SwK | not really |
05:31.55 | twisted | MikeJ[Laptop], lol |
05:32.06 | SwK | IRC's plain text and you're using it |
05:32.07 | nDuff | DarthClue, oh. Heckuvadeal. (How new? We just bought a second TE110P a few weeks ago). |
05:32.45 | MikeJ[Laptop] | well hell.. you think he dosn't notice that she get's home from her "girls weekend" at the same time you log in and say you just got back.... |
05:32.53 | DarthClue | nDuff: probably newer than that. call digium support and they can tell you. you might have to send it in to get a replacement so that you can have the upgradeable firmware. |
05:33.04 | MikeJ[Laptop] | can not sleep, clowns will eat me |
05:33.10 | twisted | MikeJ[Laptop], SHHHHHHH |
05:33.22 | nDuff | hmm. Okay; thanks! |
05:33.23 | MikeJ[Laptop] | ohhh.. sorry |
05:33.27 | DarthClue | MikeJ: the clowns have already eaten your ass, now shutup and digest already. |
05:34.11 | MikeJ[Laptop] | I should sleep. |
05:34.19 | MikeJ[Laptop] | I should sleep. |
05:34.25 | MikeJ[Laptop] | echocancel=yes |
05:34.51 | MikeJ[Laptop] | sorry about that.. forgot to turn it back on |
05:35.25 | MikeJ[Laptop] | you sank my jitterbuffer... |
05:35.42 | MikeJ[Laptop] | G27 |
05:35.45 | MikeJ[Laptop] | BINGO! |
05:36.13 | MikeJ[Laptop] | follow follow follow follow follow the yellow brick road. |
05:36.53 | MikeJ[Laptop] | http://bugs.digium.com/view.php?id=4511 |
05:37.02 | MikeJ[Laptop] | hey... somone test it... |
05:37.04 | MikeJ[Laptop] | NOW! |
05:37.07 | darylp | ok, it's non trivial to build asterisk for os x, does anyone know where I can download prebuilt binaries for os x? |
05:37.14 | *** join/#asterisk Assid (~assid@203.115.64.61) |
05:37.25 | MikeJ[Laptop] | darylp, it should be trivial\ |
05:37.41 | MikeJ[Laptop] | head, not stable. |
05:37.48 | darylp | well, bison is complaining profusely |
05:38.09 | MikeJ[Laptop] | ummm... oh osx boys, where did you go? |
05:38.15 | darylp | I downloaded asterisk-1.0.9.tgz |
05:38.20 | MikeJ[Laptop] | use head. |
05:38.26 | DarthClue | darylp: use head |
05:38.27 | MikeJ[Laptop] | from cvs. |
05:38.44 | MikeJ[Laptop] | don't think 1.0.x will work |
05:39.10 | MikeJ[Laptop] | if todays does not... 6-17-05 should... but no reason today's shouldn't |
05:39.29 | MikeJ[Laptop] | darylp, old bison? |
05:39.33 | darylp | so I follow these instructions :To get the current snapshot from the release branch of CVS, issue the following command: |
05:39.38 | darylp | default os x bison |
05:39.40 | Assid | anyone spoken to shido6 recently? |
05:39.48 | MikeJ[Laptop] | he was on earlier |
05:39.55 | *** join/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET) |
05:40.01 | DarthClue | darylp: the wiki has instructions for cvs...one sec. |
05:40.03 | Assid | bought a nufone account |
05:40.04 | MikeJ[Laptop] | ~seen shido6 |
05:40.04 | jbot | shido6 is currently on #asterisk (12h 36m 31s). Has said a total of 74 messages. Is idling for 1h 1m 41s |
05:40.06 | Assid | cant register |
05:40.19 | MikeJ[Laptop] | call nufone support... |
05:40.36 | DarthClue | http://www.voip-info.org/wiki-Asterisk+Download |
05:40.48 | MikeJ[Laptop] | wiki wiki |
05:40.53 | MikeJ[Laptop] | it's like pizza pizza |
05:41.03 | Assid | will do that |
05:41.39 | MikeJ[Laptop] | Assid, and use the configs they send you.. they work |
05:41.50 | darylp | If you want to be at the bleeding edge, use cvs to checkout the latest version: |
05:41.55 | MikeJ[Laptop] | at least mine did when I had an acocunt from them |
05:42.01 | darylp | so I use these instructions : If you want to be at the bleeding edge, use cvs to checkout the latest version: ? |
05:42.12 | MikeJ[Laptop] | ummmm |
05:42.24 | DarthClue | darylp: see where it says development version on that link i posted? that is cvs-head. |
05:42.27 | DarthClue | ~cvs-head |
05:42.27 | jbot | i guess cvs-head is the latest and greatest version of *. directions for download are at http://www.voip-info.org/wiki-Asterisk+Download |
05:42.36 | darylp | ok, that's what I thought |
05:42.42 | MikeJ[Laptop] | yeah |
05:42.48 | MikeJ[Laptop] | the 4 lines below that... |
05:43.04 | MikeJ[Laptop] | dl, |
05:43.12 | MikeJ[Laptop] | if you are upgrading from stable |
05:43.22 | MikeJ[Laptop] | make sure to checkout to a new dir |
05:43.30 | MikeJ[Laptop] | and remove the old asterisk modules. |
05:43.48 | MikeJ[Laptop] | and make your bed |
05:44.07 | MikeJ[Laptop] | tap tap tap.. is this thing on? |
05:44.29 | MikeJ[Laptop] | http://bugs.digium.com/view.php?id=4511 |
05:44.34 | darylp | ...downloading |
05:44.37 | WilliamK | hey Mike, had an issue the last time I did a recompile, it keeps thinking a few mods are older than the cvs, etc.. |
05:44.55 | WilliamK | not standard to delete the var/lib/modules/asterisk is it? |
05:44.57 | darylp | ...still downloading |
05:45.18 | MikeJ[Laptop] | WilliamK, no, it just bitches at you about them.. cuz they could be out of tree modules |
05:45.58 | *** join/#asterisk clive- (~pirch@rndf-146-5-08.telkomadsl.co.za) |
05:46.00 | MikeJ[Laptop] | wait, didn't we do a make superduperstuperclean or somthing that did that? |
05:46.11 | MikeJ[Laptop] | I don't recall.. |
05:46.26 | MikeJ[Laptop] | ok.. sleep |
05:46.32 | shido6 | who need NuFone support? |
05:46.35 | shido6 | +s |
05:46.50 | MikeJ[Laptop] | assid can't register |
05:47.28 | MikeJ[Laptop] | really this time |
05:48.25 | *** join/#asterisk Tili (~Tili@202-133-65-175-dialup.sat.net.pk) |
05:48.41 | rv_weasel | instead of just getting a notice that sipreg failed. can i get some info on why? |
05:48.56 | darylp | hmm, I must not understand something |
05:48.59 | DarthClue | rv_weasel: how high is your verbose level set? |
05:49.14 | rv_weasel | like 8 |
05:49.18 | DarthClue | darylp: what's that? |
05:49.41 | darylp | I have xxx,1,Dial(yyy) xxx,2,Wait(1), xxx, 3,Voicemail(yyy) |
05:49.51 | DarthClue | rv_weasel: that is probably enough, but you could set it to about 30 and see what you get. |
05:49.53 | darylp | and it doesn't seem to wait, I still have the cutoff |
05:50.02 | litage | i added a server to e164.org, and it gave me a # range (88299 001778 00 to 88299 001778 99). does that mean that anyone who calls a # in that range is sent to my server? |
05:50.20 | milkyflava | I am trying to start * with asterisk -vvvgc but it never gets to the console, where does it log why it is not working? |
05:50.38 | DarthClue | darylp: custom recorded greetings for vm or defaults? |
05:50.42 | clive- | how would one know if there is a memory leak in cvs head ? |
05:50.51 | milkyflava | I have looked in /var/log/messages /var/log/asterisk/ and have found no reason why it isn't working |
05:51.12 | DarthClue | milkyflava: it should tell you why when it is loading. |
05:51.19 | darylp | defaults |
05:51.32 | DarthClue | try increasing it to wait(2) |
05:51.48 | darylp | yeah, I went all the way to 5, the behavior doesn't change |
05:52.01 | DarthClue | 1.0.9? or head? |
05:52.18 | milkyflava | DarthClue, I see some conf files that it can't find but it worked before a shutdown and now it doesn't |
05:52.37 | milkyflava | everything loads fine but asterisk will not get to a cli |
05:52.43 | DarthClue | milkyflava: pastebin the output. |
05:55.49 | milkyflava | http://pastebin.ca/16907 |
05:58.58 | DarthClue | milkyflava: try asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvgc |
05:59.17 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
05:59.35 | tsume | http://news.bbc.co.uk/2/hi/europe/4649987.stm <-- what a joke. |
06:00.11 | tsume | the person sueing is obviously an anti-american |
06:00.44 | milkyflava | DarthClue, paste that also or should that be painfully obvious? |
06:01.06 | `Sauron | And what's wrong with being anti-american? |
06:01.44 | Assid | shido6?????? |
06:01.56 | DarthClue | milkyflava: if you don't see something with that command, there is something really, really wrong. |
06:02.18 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
06:02.22 | DarthClue | tsume: no, they are just plain stupid and are trying to make a quick buck. |
06:02.42 | milkyflava | :) I am really, really new. I just got my card and am just trying to do my first install. |
06:03.20 | DarthClue | milkyflava: not a problem, what results do you see with that new command? |
06:04.39 | milkyflava | It looks the same |
06:04.58 | DarthClue | milkyflava: what kind of card? |
06:05.16 | milkyflava | TDM11B the dev card with 1 fxo and 1 fxs |
06:05.17 | clive- | ayone else using CVS head as of yesterday? |
06:05.48 | DarthClue | milkyflava: did you setup your zaptel.conf and zapata.conf |
06:05.57 | DarthClue | clive: why? |
06:06.15 | milkyflava | DarkClue, yes I have them both set up |
06:06.24 | darylp | if anyone cares os x bison must be upgraded before you can build asterisk, the version in darwin ports looks like it's working (it didn't error out on bison) |
06:06.33 | clive- | Darth my free memory is plummetting, and looks very worrying to me |
06:07.38 | DarthClue | clive: don't know. mine is from the 23rd. if no one else responds, you might check back later and see if anyone else is using it or just back up a few days. |
06:07.41 | darylp | if I make install will it override my conf files? |
06:08.12 | DarthClue | darylp: no, but make samples will and it is a good idea to back up the conf files and then copy / paste what you need into the new ones. |
06:08.27 | darylp | thanks |
06:08.39 | DarthClue | milkyflava: did you modprobe zaptel and wctdm? |
06:09.07 | milkyflava | DarkClue, yes, they both loaded without a problem |
06:09.57 | DarthClue | milkyflava: the only thing i can see is that it looks like it isn't happy that so many conf files are missing. maybe you need to re-install. |
06:10.00 | darylp | do I need to make zaptel? |
06:10.05 | darylp | cause it's erroring like a sob |
06:10.30 | milkyflava | DarkClue, Ok, I don't mind installing 6-7 times I will get better with each one |
06:10.50 | milkyflava | but is there an install using FC* with this card that you could recommend? |
06:10.55 | DarthClue | darylp: zaptel is required for ztdummy. if you want meetme or anything that requires timing then you don't need zaptel |
06:11.36 | DarthClue | milkyflava: i use fc3 with that card and it works just fine. did you download from cvs? |
06:12.20 | darylp | ok |
06:12.25 | darylp | I dont' think I have a choice |
06:12.26 | milkyflava | DarkClue, I d/led it tonight from CVS |
06:12.33 | darylp | zaptel seems very linux specific |
06:12.40 | milkyflava | using FC1 and the asteriskdocs website |
06:12.57 | darylp | libpri won't build either |
06:13.59 | helpmeguys | I need help: http://bugs.digium.com/view.php?id=2266 how do I get the 24 bytes key to send reboot to Grandstream phone? |
06:14.08 | loud | i had no problems, fc4_64+t410p+2.6.12 |
06:15.09 | loud | helpmeguys |
06:15.13 | darylp | well, it built, and installed |
06:15.14 | DarthClue | milkyflava: i use FC3 so some things may be different in FC1. but i use the docs on voip-info and it works flawlessly. get rid of the existing files / directories in etc/asterisk, var/lib/asterisk and /usr/src that pertain to asterisk, then go to the site i am about to post the link for and follow the directions for cvs-head and then scroll down to where it says upgrading asterisk and follow the directions for compile and |
06:15.22 | darylp | the sound is improved, but I still have glitches |
06:15.25 | loud | curl -c cookies.txt -d"P2=PASSWORD&Login=Login&gnkey=0b82" http://*.*.*.*/dologin.htm |
06:15.29 | loud | curl -b cookies.txt http://*.*.*.*/rs.htm |
06:16.37 | helpmeguys | loud: thanks for the tip, but I want to be able to send SIP NOTIFY message, since the phone might be behind NAT and stuffs |
06:16.43 | DarthClue | darylp: not sure, could be an osx issue. bkw might be able to help you, but i don't recall if he runs it on osx or not. you'll have to wait a few hours for the rest of the U.S. to wake up. |
06:17.47 | darylp | ok, well, I can deal with it for now, it's not like I don't have a hundred other issues to work out |
06:18.19 | milkyflava | DarthClue, It was my undrstanding that using this card I wouldn't need SIP or IAX is that correct? |
06:18.34 | milkyflava | I could just use my POTS line |
06:19.37 | DarthClue | milkyflava: if you want just standard pots communication and don't want any voip then yes. |
06:20.38 | milkyflava | DarthClue, but I could still get the PBX stuff working and use IP phones inside my network? |
06:21.04 | DarthClue | milkyflava: you would then need to use SIP or IAX in your network for the IP Phones. |
06:21.34 | milkyflava | DarthClue, Ok, thanks, I have been going at this for about 8 hours and am getting more confused as I go. |
06:21.58 | *** join/#asterisk tessier_ (~treed@222.253.74.141) |
06:22.06 | DarthClue | milkyflava: just take it slow. it will all be clear in the end. |
06:23.16 | milkyflava | DarthClue, I hope so, it's hard just finding one good doc that will walk a newb through from beginning to end |
06:23.49 | DarthClue | milkyflava: the wiki is pretty good, if you can navigate it. otherwise, my only answer is that it is being worked on. |
06:24.19 | DarthClue | ~cluecon |
06:24.19 | jbot | cluecon is probably http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
06:24.19 | tessier_ | Where in the realtime config do I tell it what db to connect to? ip, user, etc? |
06:24.31 | milkyflava | DarthClue, The wiki is pretty hard to navigate through, I will keep at it and eventually get it...I hope. |
06:24.43 | tessier_ | The extconfig.conf just tells it mysql, db name, table name |
06:25.29 | DarthClue | tessier: i don't understand your question? |
06:25.59 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
06:26.09 | tessier_ | DarthClue: I am setting up realtime for asterisk. I want to set up a voicemail system. I want the voicemail configs to be in my mysql db on the same host. |
06:26.25 | blitzrage | DarthClue: lol - no more of those :) |
06:26.30 | milkyflava | lol |
06:26.31 | tessier_ | I added this to extconfig.conf: voicemail => mysql,asterisk,voicemail_users |
06:26.42 | blitzrage | DarthClue: theres a really cute chick who waitresses at vball. She's hot. |
06:26.52 | tessier_ | That is all the wiki says I have to do. |
06:27.07 | DarthClue | ok, and what happens? |
06:27.10 | tessier_ | I am wondering where I can tell it what my db ip is, what username/pass etc? |
06:27.22 | tessier_ | I don't see how it can possibly connect with just that line. |
06:28.29 | DarthClue | tessier: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime ... there is a section that says 'How to configure Realtime - MySQL Method' ... i think that's what you want. |
06:31.32 | helpmeguys | I need help: http://bugs.digium.com/view.php?id=2266 how do I get the 24 bytes key to send NOTIFY reboot to Grandstream phone? I don't want to use curl because the phone might be behind NAT |
06:31.37 | *** join/#asterisk iceyp (~icepick@firewall.unix.co.nz) |
06:32.01 | milkyflava | DarthClue, Can you post that link to the wiki doc? |
06:32.16 | blitzrage | DarthClue: don't think I'm going to make it to cluecon |
06:32.16 | iceyp | hey guys, anyone upgraded a 7940 phone here? I can get P0S3-05-3-00 onto the phone, but it wont accept anything newer |
06:32.21 | DarthClue | hmg: are you trying to find out if the bug was applied or what? |
06:32.27 | iceyp | I'm guessing i need to upgrade the boot loader somehow? |
06:33.03 | DarthClue | milkyflava: which one? |
06:33.20 | iceyp | the phone keeps saying "Booting DSP" when i go into menu's etc |
06:33.43 | milkyflava | DarthClue, you said you would post one but you never did. :) |
06:34.00 | DarthClue | i did? |
06:34.05 | milkyflava | lol, yeah |
06:34.29 | helpmeguys | DarthClue: yes, was it applied? If it was not, I can apply it for my case, but I will need the grandstream key anyway :( |
06:35.20 | milkyflava | DarthClue, you said "go to the site I am about to post the link..." but you never did |
06:36.08 | DarthClue | doh...http://www.voip-info.org/wiki-Asterisk+Download |
06:36.18 | DarthClue | sorry, it's late. |
06:37.01 | *** join/#asterisk af_ (~af@ip-131-22.sn2.eutelia.it) |
06:38.25 | milkyflava | DarthClue, Thanks, don't worry, I am the one asking for help. :) |
06:39.56 | DarthClue | hmg: i would say it wasn't applied. let me on something though. |
06:41.13 | DarthClue | hmg: there is a sip notify command...not sure if it can do what you want or not. |
06:41.53 | iceyp | can one upgrade to POS3-07-4-00 when using boot load ID PC030301 |
06:42.00 | iceyp | on a cisco 7940 phone |
06:42.23 | DarthClue | iceyp: i don't think any of the cisco guys are here right now. you might have to wait a few hours for them to wake up. |
06:42.31 | iceyp | sweet |
06:42.33 | helpmeguys | DarthClue, the bug says "You also need grandstreams administration toolkit to produce the small key file that we attach to the NOTIFY" ... that is what I'm trying to find |
06:43.34 | helpmeguys | DarthClue, I already tried "sip notify" and it worked for CISCO phones, however Grandstreams returns with 415 response .. I need an additional key for grandstream |
06:45.04 | Pete_Largo | question: does the 's' extension work for calls that have a 'DID' number? |
06:45.24 | Pete_Largo | or rather, is it supposed to? |
06:46.14 | DarthClue | Pete: i believe it depends on the digits being sent. usually, it doesn't go to s, but to the matching digits as sent. eg, if the number is 555-1212 and 1212 gets sent, then 1212 is the extension that gets matched. |
06:47.19 | Pete_Largo | well, let's say that I want all incoming calls from a certain context to go to a certain phone, regardless of the DID number. can I use 's' or do I need to list out each DID (or match it with X's)? |
06:49.06 | DarthClue | Pete: i use _X. to match all incoming |
06:49.19 | Pete_Largo | instead of s ? |
06:49.21 | Pete_Largo | why is that? |
06:49.29 | Pete_Largo | just out of curiosity |
06:50.10 | Pete_Largo | I mean, really, what would the difference be between 's' and _X. ? |
06:50.17 | DarthClue | Pete: i also have s. not everything goes to s, so _X. picks up what doesn't. it may not be the right way to do it, this box is just a test. |
06:50.28 | DarthClue | _X. will match everything if i remember right. |
06:50.33 | Pete_Largo | same here Darth, just testing, and trying to understand... |
06:50.49 | Pete_Largo | OK, so if _X. will match EVERYTHING, then why have 's'? |
06:51.23 | Pete_Largo | ok, not true that _X. will match _EVERYTHING_, it requires at least one digit |
06:51.29 | Pete_Largo | hmm |
06:51.56 | Pete_Largo | so is 's' for calls that come in with ZERO digits but won't match calls with ONE or MORE digits? |
06:53.33 | DarthClue | Pete: the wiki page on s...http://www.voip-info.org/tiki-index.php?page=Asterisk+s+extension |
06:54.00 | Pete_Largo | yeah, I've made it to here... http://www.voip-info.org/tiki-index.php?page=Asterisk+standard+extensions |
06:54.11 | Pete_Largo | I guess I'm just bored |
06:54.56 | Pete_Largo | thanks (yet again) DarthClue |
06:55.17 | DarthClue | Pete: not a problem. |
07:01.05 | iceyp | hmmm, for some reason my cisco phone is looking for a non existent file off the tftp called P0S3-07-4-00.sbn , one i dont have, anyone got any ideas? |
07:02.45 | *** join/#asterisk wiseguy_ (chivilis@85.206.10.21) |
07:03.04 | wiseguy_ | help me again, how to get last two numbers of callerid, in extensions.conf? |
07:03.05 | wiseguy_ | :_) |
07:03.17 | wiseguy_ | ${CALLERID}:1, do not work |
07:03.31 | Assid | how the hell do i stop this lady from saying "welcome"!!! |
07:03.31 | wiseguy_ | anyone |
07:03.39 | wiseguy_ | :-)) |
07:04.09 | DarthClue | wiseguy: what version of *? |
07:04.26 | wiseguy_ | DarthClue: 1.0.9 |
07:04.50 | DarthClue | what is the exact line you have in your dialplan? |
07:05.06 | litage | i added a server to e164.org, and it gave me a # range (88299 001778 00 to 88299 001778 99). does that mean that anyone who calls a # in that range is sent to my server? |
07:05.14 | wiseguy_ | exten => _86XXXXXXX,1,SetCallerID(${CALLERID}:1); |
07:05.14 | wiseguy_ | exten => _86XXXXXXX,2,Dial,Zap/g1/${EXTEN} |
07:05.42 | DarthClue | litage: we don't know. maybe you should ask e164.org |
07:06.03 | DarthClue | wiseguy: change it to CALLERIDNUM |
07:06.44 | wiseguy_ | DarthClue: :1 it means what? |
07:08.15 | Assid | hrmm |
07:08.24 | DarthClue | wiseguy: go here http://www.voip-info.org/tiki-index.php?page=Asterisk+variables |
07:08.28 | Assid | how do i know whether or not my card supports callerid? |
07:08.30 | DarthClue | Assid: what lady? |
07:08.40 | Assid | ? |
07:08.40 | DarthClue | Assid: is it a clone? |
07:09.15 | Assid | yes |
07:09.29 | Assid | 0000:00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
07:09.32 | DarthClue | i don't think the clones support callerid. |
07:09.39 | Assid | how do i find out? |
07:09.45 | twisted | try it? |
07:09.46 | twisted | maybe? |
07:10.41 | Assid | <PROTECTED> |
07:10.42 | Assid | <PROTECTED> |
07:10.42 | Assid | ? |
07:11.53 | DarthClue | Assid: try CALLERIDNUM |
07:12.03 | DarthClue | or CALLERID |
07:12.14 | DarthClue | i am pretty sure they are case-sensitive |
07:12.27 | darylp | ugh |
07:12.54 | darylp | on debian (ubuntu-hoary) the build processing is complaining I don't have termcap support |
07:13.23 | nDuff | If I don't have an action defined for when fax detection kicks in, will having faxdetect=incoming have any effect? (I want fax detection to take effect in a particular menu where I have "exten => fax,1" defined, but not alter behaviour elsewhere; is this general behaviour?). |
07:13.58 | DarthClue | nDuff: it will look for a fax extension everywhere i think. |
07:14.05 | DarthClue | darylp: i don't know. |
07:14.15 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
07:14.30 | nDuff | DarthClue: erm. And if it doesn't find one, does it terminate the call, or pretend it didn't happen? |
07:14.40 | *** join/#asterisk epablo (~epablo@WLL-24-pppoe197.t-net.net.ve) |
07:14.57 | epablo | Hi people.. How's it going? |
07:14.57 | Assid | Jul 5 12:44:30 WARNING[3954]: pbx.c:5754 pbx_builtin_setvar_old: SetVar is deprecated, please use Set instead. |
07:14.58 | DarthClue | it will terminate the call based on invalid-extension |
07:15.34 | Zeeek | nDuff you are testing spandsp? |
07:15.34 | nDuff | Oh. That's what I was afraid of. |
07:15.41 | DarthClue | epablo: run! |
07:15.53 | darylp | one needs the libcurses dev files |
07:15.58 | nDuff | Zeeek: no, I just want to route fax calls to the fax machine even if they're made to the primary number. |
07:15.59 | DarthClue | Assid: yes, it is deprecated, but it won't kill you just yet. |
07:16.10 | Zeeek | nDuff makes sense |
07:16.14 | DarthClue | darylp: yeah, i think so. |
07:16.33 | epablo | I'm using 1.0.7. Earlier today I got chan_h323 to leave me cdrs on my cdr_odbc. But it broke with no reason. Can anyone helpme out? |
07:16.44 | Zeeek | nDuff so you detect the fax and send it to an extension? |
07:16.55 | nDuff | Zeeek: I've played with spandsp a bit, but app_rxfax's library dependencies made Asterisk die an awful, awful death, and I'm not going to go back there until getting the more important (redirect to the *real* fax machine) functionality working first. |
07:17.02 | Assid | nope |
07:17.04 | DarthClue | epablo: step one, upgrade to at least 1.0.9 or HEAD. |
07:17.09 | Assid | dont think callerid works on this clone |
07:17.10 | nDuff | Zeeek: yup. Only trick is that we have several extensions with fax machines, and I don't want to redirect any of them. |
07:17.14 | *** join/#asterisk Myshenka (~gunde@217.9.101.85) |
07:17.17 | epablo | sorry chan_oh323. I hace the amaFlags=billing |
07:17.21 | nDuff | erm, redirect calls initially intended for any of them. |
07:17.28 | Zeeek | nDuff just yesterday another fax machine kept trying and didn't make it through |
07:17.28 | epablo | ok.. I'll try that |
07:18.03 | epablo | DarthClue: where there any bugs found in 1.0.7 related to cdrs? |
07:18.05 | DarthClue | epablo: that's step one, once you've done that, try again, and then come back if it still doesn't work. |
07:18.29 | nDuff | and the other problem is not having a second fax (modem|machine) handy and consequently being unable to test. That's probably a big enough issue that I'm going to put things back and call it a night shortly. |
07:18.30 | DarthClue | epablo: i doubt it, but 1.0.9 is the latest stable and most of us prefer HEAD |
07:19.09 | nDuff | DarthClue: is HEAD actually sanely usable in a production environment? |
07:19.24 | DarthClue | nDuff: yes |
07:19.26 | epablo | DarthClue: I used to use HEAD. But since I'm on a production enviroment.. |
07:19.50 | DarthClue | HEAD is used in all sorts of production environments |
07:20.21 | epablo | DarthClue: We'll I'll give 1.0.9 a try.. be back with you in a while to see how it went |
07:21.15 | Zeeek | nDuff you need a test call? |
07:22.11 | nDuff | Zeeek: I'd need a bunch of them -- I don't anticipate this working out-of-the-box. The offer's appreciated, but I'll just hold off 'till I'm able to handle it on my own. |
07:22.13 | Zeeek | CALLING EVERYONE IN PARIS: lunch with Mark Spencer (who?) this Friday; be there or be there |
07:22.36 | Zeeek | nDuff ok, I have jfax for now so I could easily help you out at some point |
07:23.43 | Assid | hrmm.. this thing keeps saying that welcome |
07:23.44 | Assid | and i dont want it |
07:23.51 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
07:24.01 | Zeeek | you don't want to be welcomed? |
07:24.06 | Zeeek | Get Out! |
07:24.07 | Assid | eeks |
07:24.08 | Assid | sorry.. mine.. |
07:24.15 | Zeeek | (that's how it feels not to be welocmed) |
07:24.55 | DarthClue | Assid: what thing keeps saying welcome? |
07:25.05 | Zeeek | WELCOME |
07:26.01 | DarthClue | DarthClue knows that format c: is possible when using /u |
07:26.09 | Assid | DarthClue: my bad.. i added a background of welcome |
07:26.15 | Assid | to play the "welcome" from the operator |
07:26.30 | Assid | and what happens on pstn.. people dont hear that.. so it sounds like CUM |
07:26.38 | Assid | which is kind of embarassing |
07:26.48 | Zeeek | yes, that would be in many cases |
07:26.48 | Assid | coz my aunt just missed the pbx |
07:26.59 | Zeeek | Assid add a wait(1) |
07:27.11 | Assid | nah.. just knocked it off |
07:27.58 | Assid | but when i dial a known extension.. shouldnt it automatically go to the DialStatus ? |
07:28.05 | Assid | i mean if nonswer and busy etc |
07:28.08 | Assid | coz its not doing it |
07:28.19 | Assid | or do i have to define it every context? |
07:30.45 | Assid | hrmm.. time for another shower |
07:30.49 | Assid | too damn hot |
07:32.38 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
07:36.03 | Zeeek | actually cooled down a lot here and floods in the north |
07:36.12 | Zeeek | it's a nice 18°C now |
07:36.25 | *** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net) |
07:37.23 | nDuff | Zeeek: actually, if you wouldn't mind sending a fax to 512 874 7601 and seeing if it gets redirected to a fax machine, I'd rather appreciate that. |
07:38.29 | DarthClue | down to 71F here...too bad the humidity is still sitting at 87%! |
07:38.44 | Zeeek | nDuff sure just a sec (need to login and remember password etc) |
07:38.45 | Myshenka | Are there any known issues with the asterisk users mailing list? With addresses from a certain domain I cannot subscribe at all, and being subscribed with a different address my mails dont seem to get published when I write to the list. |
07:39.21 | Myshenka | And yes, concerning the first issue I did write to the mailing list owners, but didnt get a reply |
07:40.01 | Pete_Largo | is anything broken in 1.0.9? |
07:41.09 | DarthClue | Pete: shouldn't be, why? |
07:41.56 | wiseguy_ | am, i have 12 e1 channels, howto describe to asterisk, that he should use one of available channels from these 12? |
07:41.59 | wiseguy_ | :-) |
07:42.15 | DarthClue | wiseguy? |
07:42.35 | wiseguy_ | ammm |
07:42.39 | Zeeek | nDuff I'm having a small problem... but it'll happen RSN |
07:42.42 | wiseguy_ | Dial,Zap/g1 |
07:42.49 | wiseguy_ | it will do it? |
07:44.15 | Pete_Largo | no reason |
07:44.35 | Zeeek | nDuff ok fax sent |
07:45.38 | nDuff | Zeeek: excellent, it detected the fax and redirected to the appropriate line. Do you have any feedback on whether your side thinks it was correctly sent? (I'm not in-office to grab the physical piece of paper). |
07:45.50 | Zeeek | I'll see |
07:46.23 | Zeeek | yep they report sucessful transmission |
07:46.28 | nDuff | sweet! |
07:46.29 | nDuff | many thanks. |
07:46.37 | Zeeek | np |
07:47.13 | nDuff | (there's a nonprofit we're going to be giving some building space and access to our phone system to -- this means I'll be able to put them behind just one extension, rather than needing a second one for their phone line as well). |
07:47.36 | epablo | I just upgraded to 1.0.9 and now I'm getting this: usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: ast_smoother_feed |
07:47.42 | epablo | Any ideas? |
07:48.09 | nDuff | s/for their phone line/for their fax machine/ |
07:48.12 | nDuff | gah. |
07:48.36 | Myshenka | epablo, before you upgraded, did you delete the contents of the asterisk modules directory? |
07:49.16 | epablo | Myshenka: No. But I built my own rpm and installed it with -Uvh |
07:50.42 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
07:51.07 | epablo | Myshenka: all my modules are from today.. |
07:51.57 | Myshenka | epablo: I am a newbie here, just the error message looked familiar (-: |
07:52.29 | *** join/#asterisk jansaell (~jan@192.71.194.38) |
07:52.54 | epablo | Myshenka: Ok.. thanks.. all ideas are well recived ;) |
07:55.15 | *** join/#asterisk truescot (~ts@84.119.222.91) |
07:56.47 | truescot | can anyone help me please, i have installed free world dialup and can make outgoing calls and everythign else works fine except i get no incoming calls, it doest even show anything in the asterisk console when an incoming call is started |
07:57.10 | truescot | i am guessing i have not forwarded all correct ports to the machine on my router, has anyone got any info? |
08:00.19 | Zeeek | truescot with sip debug ? |
08:06.09 | Assid | shido6? |
08:06.32 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
08:08.42 | infinity1 | i have a weird problem. i can dial my sip phone's extension from my zap interface but i can't dial the zap from the sip phone. |
08:09.06 | Assid | did u set up a extension? |
08:09.07 | infinity1 | is there somewhere else to define an extension for a phone other than extensions.conf |
08:09.12 | Assid | to handle through zap |
08:09.26 | Assid | show me your dialplan for calling through zap |
08:09.27 | infinity1 | exten => 110,1,Macro(stdexten,110,${PHONE3}) |
08:09.48 | infinity1 | i have the sip fone right above it |
08:10.58 | wiseguy_ | help me |
08:11.06 | wiseguy_ | with channels |
08:11.14 | wiseguy_ | i have e1 card with 12 channels |
08:11.15 | Assid | i dont see a Dial(ZAP/1... |
08:11.36 | Assid | therefore you arent making any call through zap |
08:11.51 | Assid | all your dial() would be SIP/IAX2 if you look at it |
08:12.38 | Assid | wiseguy_: sorry.. dunno how.. just started using 1 extension last night on x100p clone |
08:13.05 | infinity1 | Assid: how weird. |
08:13.30 | infinity1 | Assid: for some reason you don't have to have sip phone extensions listed in extensions.conf. i didn't have the context included. |
08:13.34 | *** join/#asterisk truescot (~ts@84.119.222.91) |
08:13.41 | infinity1 | Assid: but the sip phones still worked. weird. |
08:13.47 | Assid | infinity1: right.. |
08:13.53 | Assid | exten => _1.,1,Dial(Zap/1/${EXTEN}) |
08:13.56 | Assid | try that |
08:14.01 | Assid | and call any number starting with 1 |
08:14.10 | Assid | it should dial through your zap interface |
08:14.53 | infinity1 | yea. that works. for some reason you don't need it for sip. only zap. |
08:15.14 | Assid | as i said.. you need to tell zap when to be called |
08:15.29 | infinity1 | Assid: you don't for sip? |
08:15.36 | Assid | also.. your dial plan must include a ZAP interface calling.. hence Dial(Zap/1.. |
08:15.52 | Assid | sip as well |
08:15.58 | Assid | hence why i said starting with 1 |
08:16.07 | infinity1 | Assid: what i'm saying is you don't need it for sip. it works without it |
08:16.09 | Assid | what you could do is have * for pstn.. and 1 for sip |
08:16.23 | Assid | yes.. cause you are using Dial(SIP |
08:16.29 | Assid | not Dial(Zap |
08:16.37 | Assid | it doesnt know where to route the calls |
08:16.40 | infinity1 | i didn't have dial anything |
08:17.27 | Assid | you have to tell * which interface to use to make the call.. |
08:17.32 | Assid | do you want zap/sip protocol |
08:18.02 | infinity1 | Assid: thats what i though. i removed the lines and it works for sip |
08:18.24 | Assid | if you read the stdexten context.. you will notice that all of them dial through sip |
08:18.55 | infinity1 | Assid: mine doesn't |
08:19.05 | Assid | eh? |
08:19.14 | Assid | dont you have a Dial() ? |
08:19.19 | infinity1 | no |
08:19.22 | infinity1 | not for sip. |
08:19.29 | infinity1 | it must be using the context name in sip.conf |
08:19.52 | Assid | prolly.. i have it. but then mine isnt the best way to go |
08:20.56 | infinity1 | very strange. well, i learned something strange/new. |
08:21.30 | Assid | i like having to tell * what to do.. lets me send things down IAX when i want to |
08:21.40 | Assid | and Zap and SIP when it suits me |
08:22.46 | Assid | shido6 you there? |
08:24.32 | *** join/#asterisk Vlasis (user@194.219.121.194) |
08:25.55 | Vlasis | hello all, does anyone else have problems with chan_capi 0.3.5 after upgrading to * 1.0.9? |
08:26.03 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
08:27.16 | Vlasis | I keep on getting :"error sending DATA_B3_REQ" |
08:27.44 | Assid | does anyone have a nufone account> |
08:28.32 | CpuID | Assid, ya |
08:28.37 | Assid | sup CpuID |
08:28.38 | CpuID | access to 3 actually |
08:28.40 | CpuID | nada u |
08:28.42 | tzafrir | Vlasis, what 1.0.9 exactly? |
08:28.48 | Assid | goin on |
08:28.50 | Assid | u on dal? |
08:28.55 | tzafrir | from debian unstable? |
08:28.57 | CpuID | nah not on dal |
08:29.01 | CpuID | havent been on dal much recently |
08:29.05 | Assid | hehe |
08:29.10 | CpuID | cbf connecting to multiple networks all the time |
08:29.34 | Assid | i cant seem to get my nufone connected |
08:29.50 | Assid | didnt know u were into * man |
08:30.00 | CpuID | lol yea |
08:30.00 | *** join/#asterisk Newbie___ (me@211.24.146.11) |
08:30.04 | skrusty | morning |
08:30.04 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
08:30.09 | CpuID | ive got an * box at work, and one at home and one at my colleagues house |
08:30.16 | Assid | aaah |
08:30.24 | Assid | havent seen u on for quite some time |
08:30.31 | CpuID | 5 cisco 7960s at work, a yuxin YWH100 iax handset at home, and an IAXy for my gf's place shortly |
08:30.33 | CpuID | just gotta configure it |
08:30.42 | Assid | hehe |
08:30.46 | CpuID | heh just cos i aint in a channel dont mean i aint on ;) |
08:30.50 | Vlasis | tzafrir: I upgraded to Asterisk version 1.0.9 yesterday, and it seems that there are problems with chan_capi |
08:30.54 | CpuID | usually always on FN here |
08:30.58 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
08:31.03 | CpuID | so whats up with your nufone? |
08:31.07 | Assid | yeah... ive barely logged in myself |
08:31.07 | CpuID | are you doing 1way or 2way? |
08:31.08 | Assid | hrmm |
08:31.10 | Assid | cant get it to work |
08:31.25 | Assid | keeps saying no authority found |
08:31.49 | Vlasis | tzafrir: i used chan_capi 0.3.5 up to yesterday with Asterisk 1.0-RC2 and all worked fine, but after upgrading to Asterisk 1.0.9 the isdn channels give "error sending DATA_B3_REQ" |
08:31.49 | CpuID | ok, this for outgoing calls? |
08:31.50 | *** join/#asterisk Jas_Williams (~Jason@host86-130-10-146.range86-130.btcentralplus.com) |
08:32.03 | Assid | yep |
08:32.14 | Assid | put in $10 for testing.. |
08:32.50 | Assid | since last night i cant get it up |
08:33.29 | CpuID | sec gonna pn you |
08:33.34 | CpuID | pm* |
08:33.38 | Jas_Williams | Assid: bad choice of words |
08:33.41 | skrusty | yeah ;) |
08:34.31 | CpuID | lol Jas_Williams |
08:34.32 | CpuID | :) |
08:34.44 | CpuID | that almost deserves to be quoted somewhere ;) |
08:35.38 | skrusty | like the topic? ;) |
08:35.52 | CpuID | er...yes sounds good :) |
08:37.26 | CpuID | damn i love nufone CID |
08:37.31 | CpuID | so fun to prank people :) |
08:37.35 | skrusty | hehe |
08:37.36 | truescot | btw i managed to get iax to work on FWD, i just forgot to put a did in :) |
08:41.07 | CpuID | Assid, get my msgs? |
08:42.03 | *** part/#asterisk Vlasis (user@194.219.121.194) |
08:43.18 | *** join/#asterisk loick (~loick@81.255.80.161) |
08:48.13 | *** join/#asterisk vuvie (~S@bb219-74-45-31.singnet.com.sg) |
08:56.29 | tzafrir | CpuID, so now you can blackmail Assid with a bash.org submission? |
08:56.44 | *** part/#asterisk Myshenka (~gunde@217.9.101.85) |
08:57.01 | CpuID | lol :) |
09:00.42 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
09:02.03 | *** join/#asterisk Assid (~assid@203.115.64.61) |
09:03.21 | Assid | CpuID: u there? |
09:03.25 | CpuID | ya |
09:03.27 | CpuID | get my msgs? |
09:03.33 | Assid | nope |
09:03.34 | Assid | i got cut |
09:03.35 | CpuID | arr |
09:03.40 | CpuID | damnit i closed all my shit |
09:03.43 | Assid | stupid isp's antennae died |
09:12.29 | *** join/#asterisk jeroko (~root@220.Red-83-38-229.pooles.rima-tde.net) |
09:13.58 | CpuID | who was it here who runs nufone again? |
09:14.03 | CpuID | i know the guy used to hang around here |
09:14.10 | Assid | chido |
09:14.34 | CpuID | ah |
09:15.03 | DarthClue | JerJer runs nufone i believe |
09:15.33 | CpuID | hmm, yea that nick is more familiar to me |
09:15.43 | Assid | he aint around either |
09:16.16 | DarthClue | it's too early in the U.S. for anybody to be around. I have to get up in less than 2 hours and I have not yet even gone to bed. |
09:16.31 | Assid | dont sleep |
09:16.36 | Assid | it will give you a headache |
09:16.50 | Assid | hrmm i need to start increasing th number of hours i sleep |
09:17.02 | Assid | 5-6 is starting to make me feel really odd |
09:17.33 | DarthClue | i usually get maybe 2 a night during the week then crash on the weekends for 12 to 16 hours at a time. |
09:17.44 | Assid | thats the thing |
09:17.48 | Assid | i cant get more than 6 |
09:17.51 | Assid | no matter what |
09:17.57 | Assid | sunday.. maybe 8. if im lucky |
09:18.09 | Assid | friday night.. 4 |
09:18.12 | Assid | even 3 |
09:18.13 | DarthClue | go a week without and it will be easy to get hours and hours |
09:18.21 | Assid | hrmm |
09:18.24 | Assid | already 3 months |
09:20.23 | jeroko | this channel is for making questions about asterisk? |
09:20.55 | Assid | nah.. its for answering them |
09:20.58 | Assid | hehe |
09:21.27 | jeroko | then, here is one.. |
09:21.36 | DarthClue | jeroko: no one is awake right now, you'll have to make your question in 4 or 5 hours when the rest of the world wakes up. |
09:22.01 | Assid | We are unavailable right now, please leave a message after the tone.. BEEEEEEEEEEP |
09:23.33 | jeroko | can asterisk creat CDR's in realtime (with live calls)? |
09:23.58 | jeroko | feel free to answer those who are not aslept |
09:25.16 | Jas_Williams | jeroko: Yes when a call finishes cdr is written |
09:28.32 | jeroko | yes, i have that already working, but i need a crd at the beginning of the call and then to fill it with complete information after the call has ended |
09:28.47 | jeroko | just to keep track of current calls |
09:29.05 | DarthClue | jeroko: to keep track of current calls? |
09:32.10 | jeroko | my english is a little poor, so i don't know if i am been well understood |
09:34.26 | tzafrir | jeroko, asterisk does create CDR in real time by default. |
09:34.43 | tzafrir | tail -f /var/log/asterisk/cdr-csv/Master.csv |
09:35.07 | tzafrir | oh, that kind of real-time... |
09:35.11 | tzafrir | no |
09:35.24 | Zeeek | I think he means looking at it during the call |
09:35.35 | Zeeek | there is no CDR record until the call completes AFAIK |
09:35.37 | jeroko | yes, that is |
09:35.57 | Zeeek | jeroko you need to use the manager interface to get the info you want |
09:36.20 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
09:36.21 | Zeeek | if it is just to look at, Flash Operator Panel does a good job |
09:36.28 | tzafrir | Some applications try to do the math on their own, assuming they know about all the calls the user makes |
09:37.29 | jeroko | i've been reading about the manager interface, but i haven't find much documentation |
09:39.36 | jeroko | there has to be some already-built aplication for manager interface |
09:47.02 | Jas_Williams | jeroko: what type of application are you trying to write ? |
09:47.38 | jeroko | what i am looking for is a way to obtain running calls information( source, destiny, current time...) from a DB that is wroten everytime asterisk begin or ends a call |
09:48.39 | *** join/#asterisk key2 (~tree@gob75-2-81-56-64-17.fbx.proxad.net) |
09:49.44 | jeroko | the idea is to make a web where someone can watch what calls are going on |
09:50.23 | Jas_Williams | jeroko: Then the manager interface is what you require http://www.voip-info.org/wiki-Asterisk+manager+API checkout the perl modules |
09:50.26 | key2 | jeroko: yeah but u don't need the manager for that |
09:50.31 | key2 | just connect to asterisk and do a show |
09:50.37 | key2 | it will give you all the details |
09:53.19 | Jas_Williams | jeroko: you could also do asterisk -rx "show channels" |
09:53.32 | Jas_Williams | from your application and parse the output |
09:54.52 | jeroko | but there isn't enough information, after that, i should make a "show channel <channel name>" for each one |
09:55.15 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
09:55.26 | shido | ZzzZz |
09:55.39 | shido | turned my box off , now it wont turn back on because its too hot |
09:56.47 | jeroko | do u know if there is any script doing that?, if not i'm affraid of learning perl from scratch |
09:56.49 | Assid | which box? |
09:57.01 | Assid | shido: i had CpuID test my account.. he has the same issue |
09:58.10 | shido | ok, its back on, brb |
09:58.17 | CpuID | ya |
09:58.33 | CpuID | dang, gf had a little prang in her car |
09:58.38 | CpuID | nothing major, just car damage |
09:58.51 | CpuID | will soon find out how bad lol |
09:59.12 | Assid | oh boy |
09:59.24 | Assid | was it "parked"? |
09:59.25 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
09:59.32 | Assid | shido6:!! |
09:59.36 | shido6 | sup? |
09:59.39 | Assid | dude.. its def. the account |
09:59.46 | shido6 | really? |
09:59.46 | Assid | i had CpuID check it as well |
09:59.51 | Assid | yep |
10:00.00 | Assid | and he already has an account |
10:00.01 | Assid | his works |
10:00.02 | Assid | mine doesnt |
10:00.13 | Assid | you dont like me do you? |
10:00.29 | CpuID | yep |
10:00.36 | CpuID | i tested my configs, just changed the user/secret |
10:00.39 | CpuID | no authority found |
10:00.47 | CpuID | switched straight back to my own, worked perfect |
10:01.15 | *** join/#asterisk The_Duke (~the_duke@80.92.64.103) |
10:01.46 | Assid | shido6? |
10:02.23 | shido6 | I need a moment to check the switch again |
10:02.29 | Assid | okie |
10:02.33 | shido6 | thumbscans at 6 am are confusing |
10:02.56 | shido6 | make a phone call, Assid |
10:03.20 | Assid | nope |
10:03.42 | Assid | btw: there is no facility to change password? |
10:05.37 | *** join/#asterisk ZeeLax (~zeelax@217.22.162.234) |
10:05.46 | *** part/#asterisk jeroko (~root@220.Red-83-38-229.pooles.rima-tde.net) |
10:05.59 | shido6 | you're right |
10:06.05 | shido6 | assid takes a dump |
10:06.07 | shido6 | no authority found |
10:06.08 | shido6 | odd |
10:06.36 | Assid | see |
10:06.41 | Assid | so why dont you delete my account |
10:06.44 | Assid | and recreate it |
10:06.50 | Assid | would be easier to debug |
10:07.05 | Assid | make it manually |
10:07.06 | *** join/#asterisk meppl (mephisto@p54AAC178.dip.t-dialin.net) |
10:08.17 | CpuID | manually...err, lol |
10:09.25 | *** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
10:09.25 | Assid | bah |
10:09.42 | CpuID | pfft |
10:09.51 | shido6 | make a call |
10:10.12 | Assid | nope |
10:10.14 | Assid | didnt work |
10:10.24 | Assid | i have no authority |
10:10.43 | *** join/#asterisk helpmeguys (~helpmeguy@203.210.213.52) |
10:10.46 | *** join/#asterisk meppl (mephisto@p54AABE80.dip.t-dialin.net) |
10:12.21 | CpuID | stfu lol |
10:13.02 | Assid | im planning to add thumb print scanning to my pc.. |
10:13.07 | Assid | to "protect" my files |
10:13.08 | Assid | hehe |
10:13.16 | Assid | passwords dont cut it |
10:13.25 | *** part/#asterisk toresbe (tsb@developer.skolelinux.no) |
10:13.56 | Assid | only certificates and thumb scan |
10:14.05 | Assid | maybe do both together |
10:14.19 | Assid | hell.. add a retina scan in there |
10:15.40 | truescot | u know the new ibm laptops come with a fingerprint scanner |
10:16.01 | Assid | yeah |
10:16.02 | Assid | i know |
10:16.08 | Assid | i aint buying a new laptop coz of it |
10:16.22 | Assid | just the fingerprint scanner.. for the hell of it |
10:16.23 | Assid | hehehe |
10:16.25 | truescot | we have a few at work and a pile of fingerprint scanners for accessing bloomberg |
10:16.29 | Assid | i wonder how well it works on linux |
10:16.39 | truescot | i put it on for logon as well but i got bored of it |
10:16.42 | Assid | fingerprint scanners to access the news? |
10:17.09 | *** part/#asterisk vuvie (~S@bb219-74-45-31.singnet.com.sg) |
10:17.38 | truescot | blomberg is more than news, and costs a fortune to run and bloomberg is damn tight on the licenses and insist everyone logs in with fingerprints |
10:17.59 | Assid | oh.. you work @bloomberg? |
10:18.03 | truescot | nope |
10:18.12 | truescot | i work at a private bank with bloomberg terminals |
10:19.06 | *** join/#asterisk florinm (~florin@212.240.100.208) |
10:19.21 | florinm | hi boys and girls ;) |
10:19.29 | florinm | a small problem with a TE110 :( |
10:19.36 | shido6 | whats wrong now, florinm ? |
10:19.36 | truescot | go for it |
10:19.42 | florinm | i can receive ok calls via all the channes |
10:19.48 | florinm | but i can't dial out :( |
10:20.07 | truescot | are u using asterisk or asterisk@home? |
10:20.08 | florinm | -- Executing Dial("SIP/200-f089", "ZAP/g0/907951000003") in new stack |
10:20.08 | florinm | <PROTECTED> |
10:20.12 | florinm | asterisk |
10:20.39 | Assid | shido6: any luck? |
10:21.00 | shido6 | no, I have not received your password change request yet |
10:21.11 | Assid | no no..to get it working |
10:21.17 | truescot | i had this problem when i had not set the outgoing trunk properly |
10:21.21 | Assid | can change the password later.. |
10:21.37 | bublbobl | Hi all ! What is the 1st sound file played when VoiceMailMain() is called ? |
10:21.55 | florinm | and how u fixed it ? |
10:22.08 | shido6 | yes, change the password |
10:23.01 | Assid | okay fine |
10:23.04 | Assid | email sent |
10:23.12 | Assid | but i really just care to get it working first.. |
10:25.25 | truescot | i had to sort out the correct channels in group 0 in zaptel.conf |
10:25.54 | florinm | what u mean ? |
10:26.03 | florinm | i have the TE110 card |
10:26.14 | florinm | E1 with 30 voice chanels |
10:26.19 | truescot | sorry i ment zapata.conf |
10:26.26 | truescot | same as me |
10:28.45 | florinm | can u show me ur conf ? |
10:28.57 | truescot | i am not at work atm so i don't have it |
10:29.03 | florinm | k |
10:29.10 | truescot | give me a sec and i will see if i can get it |
10:29.18 | florinm | cheers ;) |
10:29.25 | *** join/#asterisk postel (~el@53f944c513317627.session.tor) |
10:34.17 | Jas_Williams | florinm: post you zapata.conf to pastebin :) |
10:34.42 | florinm | 2 sec |
10:35.30 | Jas_Williams | florinm: also you are sending the 9 out over the PSTN change the {EXTEN} in the dial command to to {EXTEN:1} |
10:35.31 | florinm | http://pastebin.ca/16922 |
10:35.39 | *** join/#asterisk jonathh (~asd@host81-136-131-42.in-addr.btopenworld.com) |
10:35.40 | florinm | yes is ok :P |
10:35.48 | florinm | it need a 9 in front! |
10:36.09 | florinm | the setup is a bit dodgy here |
10:36.21 | bublbobl | Sorry all, i don't have my asterisk nearby, does anyone know the sound file played when you invoke VoiceMailMain() ? :-$ |
10:36.26 | jonathh | hello, anyone here setup or maintain asteriisk in a production environment? Are there any tried and tested ways to harden the install? read-only partitions UPS etc? etc? |
10:36.28 | florinm | isdn <-> asphire<->asterisk |
10:36.35 | shido6 | "Comedian Mail" |
10:36.39 | shido6 | "Mailbox?" |
10:36.55 | bublbobl | shido6> yes, comdeian mail |
10:37.32 | bublbobl | when you invoke VoiceMail(), I think it is vm-intro, but I forgot for VoiceMailMain() |
10:37.41 | shido6 | <PROTECTED> |
10:37.49 | bublbobl | shido6> |
10:37.54 | bublbobl | tahnk you |
10:37.57 | shido6 | :) |
10:38.01 | florinm | Jas_Williams : any idea ? |
10:39.33 | Jas_Williams | florinm: looks fine try more debug 'debug span 1' in the cli when you make the call attempt |
10:40.12 | florinm | k |
10:40.25 | florinm | linux*CLI> debug span 1 |
10:40.26 | florinm | No such command 'debug span' (type 'help' for help) |
10:40.43 | Jas_Williams | oh |
10:40.55 | Jas_Williams | zap debug span 1 |
10:41.04 | florinm | linux*CLI> zap debug span 1 |
10:41.05 | florinm | No such command 'zap debug' (type 'help' for help) |
10:41.34 | Pete_Largo | help zap debug |
10:41.35 | Jas_Williams | I should really remember these things :) |
10:41.36 | florinm | debug channel Enable debugging on a channel |
10:41.36 | florinm | <PROTECTED> |
10:41.41 | Jas_Williams | pri debug span 1 |
10:42.06 | *** part/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
10:42.10 | florinm | k :P |
10:42.27 | Jas_Williams | You will now get a lot of output paste the results to pastebin |
10:42.37 | florinm | no extra info!!!! |
10:43.44 | florinm | seems that debug didn't done any good? as i didn't got any extra info ?? |
10:44.04 | Jas_Williams | Then the call is not going out PRI |
10:44.17 | florinm | <PROTECTED> |
10:44.17 | florinm | <PROTECTED> |
10:44.34 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
10:44.37 | Jas_Williams | There are no availiable channels in PRI |
10:44.47 | Jas_Williams | show channels |
10:44.58 | florinm | Channel (Context Extension Pri ) State Appl. Data |
10:44.58 | florinm | 0 active channel(s) |
10:44.58 | florinm | 0 active call(s) |
10:45.20 | Jas_Williams | zap show channels |
10:45.31 | PoWeRKiLL | hello ! |
10:45.50 | florinm | http://pastebin.ca/16924 |
10:47.49 | Jas_Williams | florinm: try selecting an individual channel in your dial eg daial(ZAP/30/${EXTEN}) |
10:48.10 | florinm | k |
10:48.59 | bublbobl | bye all |
10:49.07 | *** part/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr) |
10:49.42 | florinm | <PROTECTED> |
10:49.42 | florinm | <PROTECTED> |
10:50.38 | Jas_Williams | florinm: you say incomming calls work |
10:50.42 | florinm | yes |
10:50.56 | florinm | i can dial in no problem |
10:51.00 | *** join/#asterisk darylp (~darylp@63-208-162-60.digitalrealm.net) |
10:51.15 | Jas_Williams | This is a TE110P single port card ? |
10:51.17 | *** join/#asterisk moonaddict (~moonaddic@213.129.253.62) |
10:51.33 | moonaddict | hey all. any idea where I could by a digium card in austria? |
10:51.48 | florinm | yes |
10:52.43 | Jas_Williams | moonaddict: Australian Technology Partners in Melbourne |
10:52.55 | moonaddict | no sorry austria/europe |
10:53.01 | Jas_Williams | opps |
10:53.04 | Delvar | lol |
10:54.04 | moonaddict | :) |
11:00.43 | darylp | any osx experts? |
11:03.59 | *** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca) |
11:05.54 | moonaddict | have I understood this correctly? to connect a bunch of analog extension (i.e. phones, faxes) I need e.g. A DIGIUM TDM400P + FXS modules |
11:06.15 | moonaddict | and then another BRI (zap) interface that can connect to my telco's NTBA |
11:06.17 | moonaddict | right? |
11:06.21 | Assid | hrmm i am using DISA.. however.. when i put off the phone from the remote location.. it didnt hangup |
11:06.28 | moonaddict | my mouse is hovering above the "buy now" button :) |
11:06.41 | Assid | the call was yet on between the asterisk box .. and sip provider |
11:06.47 | Assid | anyone know how to stop this? |
11:14.01 | *** join/#asterisk kakazz (~Krab@203.81.232.176) |
11:14.31 | *** join/#asterisk cpm (~Chip@cpm.sustaining.supporter.pdpc) |
11:14.57 | kakazz | hey |
11:15.49 | kakazz | need little help on PBX |
11:16.44 | kakazz | hello |
11:16.47 | kakazz | any one out there |
11:17.25 | Assid | just ask your question |
11:17.31 | Assid | if anyone can help you they will |
11:17.55 | *** join/#asterisk The_Duke2 (~the_duke@80.92.64.103) |
11:18.03 | The_Duke2 | hello |
11:18.07 | Assid | hrmm.. how much bandwith does GSM take? |
11:18.09 | The_Duke2 | can messages like -- parse_srv: SRV mapped to host sipgate.de, port 5060 |
11:18.14 | The_Duke2 | sorry... |
11:18.26 | Assid | i mean the bitrate is 13.2 .. what about the bandwith? |
11:18.46 | The_Duke2 | are messages like "Got SIP Response 404 Not Found...." stored in a variable??? |
11:19.02 | Assid | The_Duke: not found extension? |
11:19.05 | PatrickDK | bandwidth for gsm is like 27kbps |
11:19.10 | The_Duke2 | the DIALSTATUS variable does have NOT FOUND? |
11:19.27 | The_Duke2 | Assid: not found extension, never heard of that..... |
11:19.48 | Assid | well |
11:20.00 | Assid | the number/extension your trying to reach is not there? |
11:20.47 | Assid | kakazz: did you try to message me? it just auto banned you |
11:21.06 | Assid | err.. ignore |
11:21.23 | kakazz | wt eve |
11:21.32 | kakazz | <PROTECTED> |
11:21.44 | kakazz | for 30 people |
11:24.14 | The_Duke2 | Assid: OK what I need to do is tell my * to something special when i get's a message 404 not found back via SIP... |
11:24.22 | kakazz | any know what hardware card should i use for interconnecting 30 people |
11:24.38 | Assid | The_Duke2: there is a good chance its not even reaching your provider |
11:24.40 | Assid | so.. |
11:24.42 | The_Duke2 | Assid: this could be a voice message, the number you dialled has not been allocated... etc etc etc.... |
11:24.43 | Assid | check for that first |
11:25.08 | Assid | The_Duke2: dont know how to do that.. google? |
11:25.20 | Assid | kakazz: ip phones? |
11:25.34 | kakazz | softphone |
11:25.38 | Assid | well |
11:25.43 | Assid | then you dont need a hardware card |
11:25.48 | The_Duke2 | I tried. but everybody talks about ${DIALSTATUS} which cannot have a value of NOTFOUND (or anything similar).... |
11:25.59 | kakazz | and if i be using a IP phones |
11:26.04 | Assid | hardware cards are when you want hardware phones --analog/ip |
11:26.17 | Assid | hrmm |
11:26.19 | Assid | yeah |
11:26.26 | Assid | i guess dialstatus may do the trick |
11:26.50 | kakazz | if i m using Outbound calls on Soft phone ,, and internally i m using Analog phone wht will be your advice |
11:27.13 | *** part/#asterisk moonaddict (~moonaddic@213.129.253.62) |
11:27.25 | Assid | FXO cards |
11:27.41 | kakazz | thanks assid |
11:27.42 | kswail | moonaddict - yes you need FXS modules for your internal phones |
11:27.42 | Assid | depending on the number of extensions |
11:27.59 | Assid | actually |
11:28.02 | Assid | FXS cards |
11:28.04 | kakazz | how much i can expand in that matter |
11:28.07 | Assid | not FXO |
11:28.11 | Assid | well |
11:28.18 | Assid | you could buy a T1 card |
11:28.27 | Assid | that comes iwth 24/32 lines |
11:28.35 | kakazz | thanks |
11:29.03 | kakazz | do you ref this Digital CTI Cards : Digium TE110P |
11:29.16 | Assid | never used it.. cant recommend it |
11:29.21 | Assid | but digium makes good hardware |
11:29.23 | kakazz | ok |
11:29.33 | Assid | specially since its asterisk compliant |
11:29.43 | kakazz | i m just new to it .. |
11:29.55 | Assid | hey till last night.. i didnt do a FXO interface |
11:30.08 | Assid | today i am already rerouting PSTN->iax calls |
11:30.21 | The_Duke2 | assid: I would really use DIALSTATUS, but it returns BUSY for calls which return a 404 Not Found.... which is incorrect... |
11:30.22 | Assid | the guys in this channel ROCK |
11:30.38 | Assid | you may wanna look for NOTFOUND |
11:30.53 | Assid | instead of busy |
11:32.25 | The_Duke2 | Assid: I would like to, but DIALSTATUS can only have the following states: |
11:32.33 | The_Duke2 | CHANUNAVAIL | CONGESTION | BUSY | NOANSWER | ANSWER | CANCEL | HANGUP |
11:45.12 | Zeeek | The_Duke_of_Oil? |
11:49.24 | *** join/#asterisk Thumann (~Thumann@217.157.30.66) |
11:52.32 | *** join/#asterisk Bhaal (bhaal@bhaal.staff.freenode) |
11:52.53 | Bhaal | Hey guys, just got a quick question.. |
11:53.17 | Zeeek | go go go |
11:53.25 | *** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo) |
11:53.48 | Bhaal | Im using asterisk to allow multiple sip fones use 1 account with a sip provider... Just wondering if the callerID number will translate through asterisk to the handsets? |
11:54.16 | Zeeek | should |
11:54.41 | Bhaal | For an incoming call I mean, forgot to clarify that... |
11:54.57 | Zeeek | I think it shoud work |
11:55.29 | Bhaal | Cool, not able to test yet, provider isnt able to supply me with a pstn number as they dont have any allocated for my area... |
11:56.00 | Thumann | hey peeps! ;> |
11:56.36 | Zeeek | Bhaal interestingly FWD allows multiple clients on th same account |
11:57.03 | djin | Did anyone have any luck setting up modem dial-up (Banking) throught SIP? |
11:57.25 | Bhaal | Im in australia .. Im not sure if the provider Im using allows the same thing, although I dont really want it setup that way... If the line is being used, its being used... Like a normal fone... |
11:57.46 | djin | Trying Draytek Router icw. G.711a and no luck so far. |
11:58.22 | Zeeek | oh it still would be busy if busy. Anyway the multiple clients would all ring at once where with asterisk you could ring different ones at different time of day, etc |
11:59.10 | Bhaal | Zeeek: Yeah... I actually wanna setup a 'who do you want to call' type stuff as there are 4 housemates living here... |
11:59.47 | Zeeek | "to speak to the pimply teenager on the second floor, please press 1" |
11:59.56 | Thumann | lol |
12:00.05 | Zeeek | "for information on coke pricing..." |
12:00.26 | Thumann | "if you are an ignorant phonesalesman who deserves to die press 2" |
12:00.42 | Zeeek | "to hear the most recent insult about the person calling from: three three one four two three one... please..." |
12:00.59 | *** join/#asterisk Samoied (~Samoied@popeye.opens.com.br) |
12:01.22 | Zeeek | "please do not touch any buttons on your phone while we adjust the voltage" |
12:01.35 | Zeeek | "ok, now, do you feel lucky today?" |
12:02.05 | Wonka | "please do not touch that button again" |
12:02.23 | Thumann | hah! |
12:02.32 | Zeeek | "I SAID PRESS THREE! " |
12:02.35 | Thumann | :) i nned that.. |
12:02.47 | Zeeek | "oh oh oh!" |
12:02.53 | Thumann | but i dont think 48v is enough to actually boil someone.. :-/ |
12:03.12 | Zeeek | but maybe a loud tone could be a powerful message |
12:03.17 | robl^ | 48V DC can kill |
12:03.30 | Thumann | true.. with enough amps... |
12:03.43 | Zeeek | How about a general filter: "please type 123 to continue" |
12:04.03 | Zeeek | that kills any auto call machines |
12:04.11 | robl^ | there are many cases where people have used a phone while in a bathrtub and then droped it in the water.. it was enuff to stop their heart |
12:04.25 | Zeeek | that was ground differential thoug |
12:04.36 | Thumann | what about a dice roll kinda thing.. # to start stop roll.. 1-9 is hangup.. 0 is try again.. :D |
12:05.16 | *** join/#asterisk doughecka_ (~Miranda@doughecka.user) |
12:06.03 | Zeeek | still a few seats left for lunch with kram this friday - c'mon, we'll pay for your ticket to Paris |
12:06.14 | robl^ | Zeeek, those "feneral filters" also filter out mst granmothers. Mine can barely dial a phone number correctly.. let alone use an IVR. |
12:06.26 | Zeeek | lunch will cost just $2,299 |
12:06.42 | *** join/#asterisk mountie (~mountie@trb229.travel-net.com) |
12:06.51 | Zeeek | yeah I know, IVR suck anyway, but they're fun to play with for a while |
12:07.06 | Zeeek | I hate getting them as a caller though |
12:12.06 | ManxPower | a ring is about 90V in the USA. |
12:12.30 | Zeeek | ManxPower you are soooobehind |
12:15.29 | *** join/#asterisk virterm (~virterm@207.107.229.2) |
12:17.32 | ManxPower | Zeeek: I'm still 1/2 asleep. |
12:17.41 | ManxPower | Anyway, I'm about 7 hours behind 8-) |
12:17.41 | Zeeek | get some cafeine in you man! |
12:18.07 | ManxPower | working on it. |
12:18.12 | Zeeek | I'm actually sleepy after being online for 7 hours |
12:20.23 | cochi | I'm actually sleepy after being asleep for 7 hours ;) |
12:21.26 | Assid | is there a way to increase the number of rings before it picks up pots? |
12:22.03 | Zeeek | Assid who? |
12:26.01 | Assid | pstn |
12:27.49 | ManxPower | i use CLID matching to automatically forward calls from my grandmother to my phone. |
12:31.00 | *** join/#asterisk salvini_fs (~felipesal@200164050004.user.veloxzone.com.br) |
12:31.27 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
12:32.59 | *** join/#asterisk jhava (~icechat5@200.58.26.21) |
12:33.18 | Jas_Williams | Assid: yes add a wait before an onward dial in extensions.conf |
12:33.35 | Assid | yeah.. just added a wait(5) |
12:37.18 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
12:45.07 | *** join/#asterisk DarthClue (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
12:47.29 | *** join/#asterisk evangelion (~manzy_zet@ip-185-118.sn1.eutelia.it) |
12:47.43 | evangelion | hi all |
12:52.15 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
12:54.19 | *** join/#asterisk lodeon (~not4u@as1-6-4.ld.bonet.se) |
12:54.38 | lodeon | corydon: there? |
12:55.01 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
12:59.41 | *** join/#asterisk rv_weasel (~no@adsl-68-91-83-3.dsl.ksc2mo.swbell.net) |
13:01.47 | rv_weasel | i had everything working. i needed to change some network configuration, so i reboot. now it wont come back up. look at my pastebin http://pastebin.com/307775 |
13:02.52 | *** join/#asterisk DarthClue (HeadachesA@wsip-68-99-73-32.tu.ok.cox.net) |
13:04.19 | rv_weasel | wait, i have to run ztcfg at startup? |
13:04.53 | jontow | yes |
13:05.04 | *** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk) |
13:06.30 | rv_weasel | i think i found the problem. i needed to install wcfxo and run ztcfg a second time in my modprobe.conf |
13:06.54 | rv_weasel | i guess i shoule reboot to be sure huh? |
13:07.00 | ManxPower | if /etc/modules.conf is set up correctly, then ztcfg will be called after every driver load. |
13:07.26 | Assid | hrmm... how do you get the logging and ocnfiguration of * on pgsql? |
13:07.31 | ManxPower | That's done by the post-install line in modules.conf. |
13:07.46 | evangelion | have you ever run ser on a mosix cluster? |
13:07.53 | ManxPower | If you want zaptel to start automagically, do a make config in the zaptel dir, that will install the init script. |
13:09.09 | tzafrir | on debian: simply add it to /etc/modules |
13:09.10 | rv_weasel | run twice in modprobe.preload: install wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg |
13:09.12 | rv_weasel | right? |
13:09.26 | rv_weasel | two cards |
13:09.37 | ManxPower | rv_weasel: I have 8 asterisk servers and never needed to do that. |
13:10.10 | tzafrir | There is basically no reason it should be run in the post-install there if you can guarantee it would be run later (e.g: init.d script) |
13:10.21 | ManxPower | I just do a "chkconfig zaptel on". |
13:10.36 | rv_weasel | ok |
13:10.40 | rv_weasel | i follow ya |
13:10.59 | ManxPower | tzafrir: not everyone uses the init scripts. |
13:11.30 | tzafrir | ManxPower, I said that an init.d is *one way* to verify that. |
13:11.42 | tzafrir | running the ztcfg there automatically is configusing |
13:11.50 | ManxPower | *nod* |
13:12.09 | tzafrir | configusing is a confusing way to write confusing |
13:12.37 | *** part/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM0012c999e6a0.cpe.net.cable.rogers.com) |
13:12.51 | darylp | has anyone used galaxyvoice? |
13:13.10 | Jas_Williams | 2 |
13:13.26 | *** join/#asterisk mithro (~tim@212.129.242.85) |
13:13.36 | MRH2 | anyone know why I don't get echo on a PSTN->voip provider->LAN but do get an echo over the LAN? is it likely echo cancellation with the provider? |
13:14.11 | tzanger | MRH2: you don't get it over the pstn but you do over the lAN... how in the hell would the provider be responsible for that? |
13:14.29 | rv_weasel | ok, all set rebooting to be sure |
13:15.01 | MRH2 | wondering why I don't get it with the provider (although this is a good thing) |
13:16.03 | tzanger | MRH2: is your earpiece or mic volume cranked? |
13:16.12 | Jas_Williams | MRH2: what are you using as phones to get echo on the LAN ? |
13:17.04 | MRH2 | yep cranked to the max - this is why i get the echo on the LAN (Poly IP500) |
13:17.51 | MRH2 | but how is the voip provider getting rid of this echo? |
13:17.59 | MRH2 | do u think? |
13:18.38 | crash3m | the echo cancellation patch? |
13:19.35 | MRH2 | do all PSTN gateways normally use some sort of echo cancellation |
13:20.49 | ManxPower | If they do PSTN/VoIP, yes. |
13:21.02 | darylp | nobody is using galaxy? |
13:21.39 | MRH2 | cool that explains it |
13:23.09 | *** join/#asterisk \usr\sbin (~squirrelv@202.57.81.146) |
13:24.47 | *** join/#asterisk af_ (~af@ip-131-22.sn2.eutelia.it) |
13:26.14 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
13:26.49 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
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13:28.43 | *** join/#asterisk indra (~indra_wat@microinfo.rain.fr) |
13:28.59 | indra | anyone up? |
13:29.10 | darylp | I am |
13:29.35 | indra | good, u might be my saviour :) |
13:29.51 | darylp | not likely, I'm here cause I'm stuck |
13:30.01 | indra | ah :( |
13:30.24 | darylp | *sigh* and it's not like EVERY day I get a chance to be a savior |
13:30.39 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
13:30.42 | indra | u might still try to give me a clue no? |
13:30.58 | darylp | I'm really new, but others might answer ifyou ask your question |
13:31.22 | darylp | savior=saviour |
13:31.37 | indra | thanks daryl :) |
13:31.37 | SplasPood | holy crap.. first time ENUM has actually worked in a real world scenario for me! |
13:32.06 | indra | guess living in France these past 5 years have completely dulled my english :( |
13:32.29 | darylp | no, you were right, I was correcting my mistake |
13:32.30 | darylp | haha |
13:32.40 | indra | I have never done any telephony stuffs in my life (am in critical software stuffs) |
13:33.10 | indra | bouhahahahah :D well GLAD to know my english is not THAT bad yet :D |
13:33.45 | indra | so, i am doing my summer internship and my boss has decided to put me on a cheap project using Asterisk |
13:34.23 | darylp | sounds like a match made in heaven |
13:34.30 | darylp | or perhaps some other place |
13:35.03 | indra | well i won't bother everyone here with the story of my life, just want to know if there is any way to treat call waiting with Asterisk |
13:35.27 | indra | the features mention that it is possible, but i haven't found any line in the doc about it |
13:35.35 | ManxPower | indra: What SPECIFIC problem are you having with Call Waiting? |
13:35.45 | indra | well, how to configure the call waiting? |
13:36.01 | ManxPower | With Zap, SIP, MGCP, SCCP, or H323? |
13:36.04 | indra | i mean to swith back and forth between several calls on a terminal |
13:36.06 | indra | SIP |
13:36.16 | ManxPower | indra: that is TOTALLY controlled by your SIP device. |
13:36.29 | ManxPower | Most things in SIP are totally controlled by your SIP device. |
13:36.40 | indra | well, that is bad news for me but thanks manxpower |
13:37.27 | ManxPower | indra: Asterisk will always send a second (or more) call to a SIP device. What happens at that point is up to the SIP device. |
13:37.50 | indra | but i thought asterisk does the job of the call queueing? |
13:38.06 | ManxPower | indra: "queueing" is the wrong term. |
13:38.40 | ManxPower | Asterisk will handle call waiting internally on ZAP devices, as well as (I think) MGCP and SCCP devices. |
13:38.40 | indra | ah? |
13:39.17 | ManxPower | But with SIP, asterisk will just send the call. The device will usually send back a BUSY HERE message to Asterisk if the device can't handle (or is configured to not handle) another call. |
13:39.39 | ManxPower | SIP expects most of the smarts to be in the phone, not in the PBX. |
13:39.52 | indra | ManxPower : okay, i got the image |
13:40.31 | indra | ManxPower : any clue for the internal document for the SIP device - Asterisk communication? |
13:42.30 | ManxPower | indra: Huh? The SIP RFC, but that's not going to help you. You need to look at the docs for the SIP device you are using. |
13:42.35 | *** join/#asterisk af_ (~af@ip-131-22.sn2.eutelia.it) |
13:43.23 | ManxPower | What SIP device ARE you using, indra? |
13:43.46 | indra | ManxPower : great, just what I need. This company of mine has made their own modem and I am supposed to be making an application bridging their modem and asterisk application |
13:45.22 | ManxPower | A SIP device is NOT a modem. |
13:45.43 | mishehu | perhaps he means ATA ? |
13:46.08 | indra | mishehu : they call this modem as RETA but it is something they build on their own |
13:46.24 | indra | mishehu : a specific solution they offer to particular clients |
13:46.52 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
13:46.55 | mishehu | indra: stop talking in marketspeak, and explain what *exactly* you mean. |
13:47.23 | indra | mishehu : i have an asterisk application in one end |
13:47.32 | mishehu | you sound like you're reading an advertising brochure. |
13:48.23 | indra | mishehu : worse than a brochure, i only have a vague description!! :( |
13:48.35 | darylp | my problem is easier, I'm sure of it |
13:48.44 | mishehu | then you can't do anything with a vague description, can you? |
13:48.52 | *** join/#asterisk rittwage[work] (~peter@intsys.net) |
13:49.12 | darylp | what does RETA mean? |
13:49.20 | ManxPower | indra: you can't do your job if you don't have the technical info. |
13:49.38 | rittwage[work] | Hi guys- I know this has been asked 1,000 times, but is there an easy to use softphone-based operator console? FOP does funny things for me and isn't a one-click solution. |
13:49.40 | cypromis | reverse engineer it |
13:49.41 | cypromis | hehe |
13:49.44 | mishehu | darylp: my guess is that RETA is market speak |
13:49.51 | indra | mishehu : well i need to gather whatever information i can get on asterisk to make this bridge (an application) between asterisk and the modem |
13:50.20 | mishehu | indra: you need to know what the hell it is you are trying to get to work with asterisk before you start messing with asterisk. |
13:50.21 | indra | darylp : mishehu was right, this RETA stuff is just a name invented by the sales dept |
13:50.26 | darylp | what exactly must the bridge do, surely you have that information |
13:50.39 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
13:50.44 | mishehu | it's like trying to find a cure when you don't even know the disease. |
13:51.01 | darylp | the modem answers a line and then.... |
13:51.11 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
13:51.12 | darylp | a fax comes in and.... |
13:51.19 | *** join/#asterisk drumkilla (~russell@drumkilla.developer.and.stable.maintainer.asterisk) |
13:51.20 | *** mode/#asterisk [+o drumkilla] by ChanServ |
13:51.22 | ManxPower | A modem takes serial data and converts it to a modulated audio tone. |
13:51.28 | darylp | well yes |
13:51.33 | ManxPower | That has nothing to do with VoIP or SIP. |
13:51.39 | indra | darylp : simply for calls, no fax included |
13:52.17 | darylp | ok, so the modem is a voice/data product? |
13:52.54 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
13:52.56 | indra | yes |
13:52.57 | fantomax1 | hi all |
13:53.09 | darylp | I guess modem is the wrong term when you aren't really doing any modulating or demodulating |
13:53.13 | darylp | nevertheless |
13:53.20 | fantomax1 | can anyone suggest me a tool to control the performances of an * server ? |
13:53.29 | darylp | money |
13:53.39 | fantomax1 | in term of resources .. etc |
13:53.45 | SplasPood | more money |
13:53.57 | ManxPower | fantomax1: It's called "ulimit". |
13:53.59 | SplasPood | and a whip |
13:54.02 | fantomax1 | i know |
13:54.10 | fantomax1 | ulimit is a command |
13:54.16 | ManxPower | And? |
13:54.25 | fantomax1 | can i use it as testing? |
13:54.31 | ManxPower | It's not like you can control Asterisk's resource usage in any other way. |
13:54.34 | darylp | since there's so much explaining going on right now perhaps someone would like to help with my simple problem |
13:55.05 | Hmmhesays | heh... always love questions that start off like that |
13:55.07 | florinm | any unlimited VOIP provider inclusive mobile (uk) out there so i can use with asterisk ?? |
13:55.29 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
13:55.34 | darylp | xlite->galaxyvoice, behind nat, works fine, can make and receive calls |
13:55.36 | Nugget | unlimited voip == punch the monkey for the free ipod |
13:55.47 | florinm | ??? |
13:55.52 | rittwage[work] | nobody is ever asked for an operator function that FOP doesn't handle or did I just miss a particular phone model or softphone that does it... |
13:55.54 | indra | manxpower: if u could give me a hint as where to look for the documentation explaining the interaction between asterisk and SIP devices ? |
13:55.57 | rittwage[work] | ? |
13:56.03 | darylp | asterisk-> galaxy voice, behind same nat, sends request but never gets a reply, can't receive calls |
13:57.59 | Hmmhesays | probably because your invite contains the internal address of the asterisk box, if I had to guess |
13:58.32 | darylp | so how do I fix that? |
13:58.49 | skeffling | Is there a script around that imports a .csv file in to the cidname database? (no point re-inventing the wheel!) |
13:58.56 | florinm | externalip=xxx.xxx.xxx.xx something like so |
13:59.02 | darylp | both xlite and asterisk are on the same internal network |
13:59.07 | florinm | in sip.conf |
13:59.07 | darylp | ok, I will try taht |
13:59.10 | *** join/#asterisk nylon1234 (~tonton123@203.131.137.76) |
13:59.54 | florinm | and in the galaxy do u have something like "Use NAT IP " ? |
14:00.03 | Hmmhesays | http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+externip |
14:00.12 | florinm | i had same problem with a grandstream 486 |
14:00.16 | nylon1234 | hi guys......anyone had used vicidial here....i have a hardtime making the auto dial work...any help pls? |
14:00.18 | darylp | nat=yes |
14:00.35 | Katty | alive, but terribly sleepy |
14:00.47 | florinm | and i had to put there the enternal ip of the internet where the grandsteam was on |
14:01.08 | florinm | :D |
14:02.25 | Hmmhesays | it has indeed been a party weekend |
14:02.28 | Katty | DarthClue: i have an infection in the uretha |
14:02.42 | Katty | DarthClue: on antibodics....but still alive ;) |
14:02.43 | Hmmhesays | painful |
14:03.00 | ManxPower | I wonder who my IRC client automagically blocked when they tried to /msg me. |
14:03.07 | Katty | Hmmhesays: quite. |
14:03.08 | darylp | hmm, it has no effect |
14:03.16 | darylp | affect? |
14:03.20 | ManxPower | I have it set up that way because so many assholes try to get private support for free. |
14:03.27 | Hmmhesays | hmmm, try setting externip |
14:03.34 | Hmmhesays | instead of nat=yes |
14:03.37 | florinm | and ?? |
14:03.40 | Hmmhesays | :D |
14:03.47 | Katty | yes. i'm SICK |
14:03.49 | florinm | no, need to have nat=yes too ;) |
14:03.49 | Katty | sniffle. |
14:04.15 | mutilator | o_O |
14:04.15 | Hmmhesays | last time I tried that I got a pen in the gut |
14:04.27 | Hmmhesays | ;( |
14:04.35 | tzanger | Hmmhesays: you got a pen in the gut for what |
14:04.35 | Katty | ha. |
14:04.59 | Hmmhesays | hugging Katty, lol |
14:05.03 | tzanger | hahaha |
14:05.06 | Hmmhesays | watch out for the sunburn |
14:05.56 | Katty | you goofball |
14:05.57 | `Sauron | I think I got smacked last time I tried to do that to Katty |
14:06.02 | Katty | didn't you wear sunblock? |
14:06.11 | tzanger | sunblock's for the weak |
14:06.12 | `Sauron | True to her name, she is. :p |
14:06.26 | mishehu | `Sauron: like yoda, speak you. |
14:06.29 | Katty | tzanger: and for those who don't want to burn |
14:06.35 | Hmmhesays | sunblock? not soon enough... 5 hours on the river is a long time |
14:06.38 | tzanger | my goal is to be mistaken for a puerto rican (spelled wrong I'm sure) by the end of hte summer |
14:06.46 | tzanger | Hmmhesays: I did that may 24 weekend |
14:06.50 | tzanger | longer actually I think |
14:06.51 | Katty | Hmmhesays: :<<< |
14:06.53 | tzanger | the big float :-) |
14:07.14 | Hmmhesays | we have a river near here, it's just a giant party, you pay 5 bucks and float down it |
14:09.39 | indra | manxpower: got to go, but thanks for the info on SIP devices |
14:09.39 | Hmmhesays | i'm suprised people don't drown actually |
14:10.34 | `Sauron | drunk people float by default. |
14:10.45 | Hmmhesays | but do they float right side up |
14:10.57 | indra | bye all, have a nice swim for those who decide to jump into the river :) |
14:11.13 | Hmmhesays | <chuckle> |
14:12.15 | tzanger | Hmmhesays: sounds like what we did |
14:12.39 | *** join/#asterisk jhiver (~jhiver@AStDenis-101-2-4-33.w193-253.abo.wanadoo.fr) |
14:12.52 | jhiver | Anybody from NuFone about? |
14:12.58 | tzanger | we went up to paisley, ontario (about 1.5hrs from where I live) and partied on a private campground, then the owners of the campground threw us in a truck, drove us 10mi upriver and dropped us in a bunch of canoes |
14:13.35 | *** join/#asterisk daryl_just_daryl (~daryl_jus@63-208-162-62.digitalrealm.net) |
14:14.08 | Hmmhesays | tzanger: sounds like a blast, we didn't have canoes though, just tubes |
14:14.21 | tzanger | Hmmhesays: tubes? good lord man where do you keep the booze? |
14:14.52 | Hmmhesays | tzanger: in coolers tied to the inside of the tubes |
14:15.02 | tzanger | Hmmhesays: sounds like a lot of work :-) |
14:15.36 | Damin_ | Floating coolers, with really good latches! :) |
14:15.43 | Hmmhesays | takes like five minutes to tie up a cooler tube, it's one larger tube with a smaller tube tied in the middle, then you set the cooler in there and strap it down |
14:15.50 | Damin_ | Specially designed for high performance drunken tubing! |
14:17.01 | Hmmhesays | one group of people had one of those huge 70 somethine quart coolers, in a big huge tractor tube |
14:17.11 | tzanger | Hmmhesays: sounds like an easy way to crush your nuts if you hit a rapid or a shallow spot |
14:17.21 | tzanger | still sounds like fun though |
14:17.39 | Hmmhesays | haha, no.. you aren't in the same tube.. you tie it off to the tube you ride in |
14:18.03 | Damin_ | tzanger: Tubing is a blast.. You hang your ass in the tube, sit back and go for a ride.. When you get hot, you just jump into the water.. |
14:18.25 | Hmmhesays | or fall off trying to surf the cooler |
14:18.25 | Damin_ | tzanger: You don't have to even let your ass touch the water if you don't want.. |
14:18.46 | Damin_ | That too... |
14:18.55 | tzanger | Damin_: hehe yes I've been tubing many times |
14:19.15 | Damin_ | I used to be a Summer Camp Counselor in my teen years, and one of the activities was tubing.. I'd do it twice a week for like 10 weeks in a row.. :) |
14:19.21 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
14:19.23 | SpaceBass | morning |
14:19.35 | Damin_ | tzanger: I got to be an expert at it. ;) |
14:19.45 | Hmmhesays | passing out beer to the teens? |
14:19.46 | Hmmhesays | lol |
14:19.52 | Hmmhesays | Morning SpaceBass |
14:19.54 | tzanger | Damin_: :-) |
14:19.56 | Damin_ | tzanger: But the best trips where when you headed about 30 miles up river and made a day out of it. :) |
14:20.19 | tzanger | Damin_: yup, I think we only went about 10 miles upriver to canoe but nobody was paddling so it took all day :-) |
14:20.36 | Hmmhesays | some guys had a party barge this time around... a bunch of tubes with a piece of plywood and a picnic table on top.. with tunes |
14:20.42 | tzanger | http://www.mixdown.ca/~andrew/photos/PaisleyMay24-2005/ |
14:20.43 | Damin_ | tzanger: And when they were paddling, they were pointed the wrong way.. |
14:20.51 | tzanger | Damin_: that too |
14:21.03 | tzanger | the canoe trip starts about 1/2way down there |
14:21.19 | Hmmhesays | which one is you? |
14:21.30 | tzanger | Hmmhesays: hahaha I ain't telling |
14:21.52 | Hmmhesays | if I had to guess |
14:21.54 | Hmmhesays | http://www.mixdown.ca/~andrew/photos/PaisleyMay24-2005/tn/img_3040.jpg.html |
14:22.05 | tzanger | haha |
14:22.08 | tzanger | no that is not me |
14:22.20 | tzanger | there are no pics of me in teh canoe |
14:22.29 | tzanger | I wouldn't give the camera to anyone else on the water for fear of them dropping it in the river |
14:22.52 | Hmmhesays | there's one guy in there that looks like this guy named 'goose' i know |
14:22.59 | tzanger | who's that |
14:23.04 | Damin | You guys didn't go canoeing.. You went drinking, and brought canoes along to sit in... |
14:23.13 | tzanger | Damin: well yes |
14:23.17 | Hmmhesays | http://www.mixdown.ca/~andrew/photos/PaisleyMay24-2005/tn/img_3048.jpg.html on the right |
14:23.18 | daryl_just_daryl | great I've left, I can get my nick back |
14:23.28 | clive- | any jitter buffer experts about |
14:23.30 | tzanger | haha yeah that's mike |
14:23.33 | tzanger | clive-: what's the question |
14:23.34 | Hmmhesays | darylp: ghost |
14:23.34 | Damin | Look at image 3040.. |
14:23.43 | darylp | connection issues |
14:23.45 | *** join/#asterisk thal (~thalunil@walledcity.de) |
14:23.49 | thal | salut |
14:23.52 | darylp | seems to be my story today |
14:23.54 | Damin | Got his feet kicked up, beer about to be chugged.. |
14:23.59 | Hmmhesays | darylp you can ghost your nick any time you want |
14:24.02 | tzanger | Damin: yup |
14:24.08 | Damin | Alright.. I need to go to work.. |
14:24.11 | tzanger | we did some pretty intense paddling a couple of times... but that's about it |
14:24.17 | clive- | tzanger for some strange reason the RTT value goes very high very quick, causing teh jitter buffer to go wacky on me |
14:24.25 | darylp | I don't know how to do that |
14:24.30 | tzanger | clive-: what version |
14:24.37 | clive- | cvs-head |
14:24.57 | tzanger | clive-: cvs head shouldn't have that problem it should chop off the top 5% of jitter values |
14:25.00 | ManxPower | clive-: What DATE of CVS-HEAD? |
14:25.10 | thal | i have problems getting chan-capi-0.35 getting compiled. |
14:25.11 | tzanger | ManxPower: it should be fine from like april even I think |
14:25.16 | clive- | yesterdays date |
14:25.19 | thal | what can /usr/lib/gcc-lib/i486-linux/3.3.5/include/stddef.h:213: error: syntax error befo |
14:25.19 | thal | re "typedef" |
14:25.19 | thal | <PROTECTED> |
14:25.35 | ManxPower | clive-: both sides are the same CVS-HEAD date? |
14:25.41 | clive- | tzanger the wierd ting is that my RTT is never over 1000 |
14:26.09 | tzanger | clive-: define "goes wacky" |
14:26.14 | clive- | Manx the other side.....I really dunno which version they are running...is theer a big version skew issue? |
14:26.24 | tzanger | clive-: who are you terminating to |
14:27.07 | clive- | tzanger, well the RTT jumps from the normal 350ms value up to 1200ms..stays there for a bit and eventualy goes back down, in the meanwhle the conversation is basicallly a mess |
14:27.10 | clive- | magrathea |
14:27.19 | tzanger | CVS HEAD before about may (I think) had an issue where the jitter buffer'd take a big shit if you sent DTMF |
14:27.30 | tzanger | it's actually a zaptel issue and hte far end needs to fix it |
14:27.41 | jhiver | ~seen shido6 |
14:27.41 | jbot | shido6 is currently on #asterisk (4h 28m 16s). Has said a total of 17 messages. Is idling for 3h 49m 44s |
14:27.56 | clive- | zaptel?.... |
14:27.59 | Hmmhesays | I need to read a perl book |
14:28.28 | tzanger | clive-: I'm not sure what the new command is but you might want to try jb debug and see what it looks like when that happens |
14:28.33 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
14:28.33 | *** mode/#asterisk [+o anthm] by ChanServ |
14:28.36 | tzanger | you shoudl normally see vvvvvvvvvvvvvvvvvvvvvvvvvv |
14:28.41 | clive- | tzanger thanks ill try that |
14:30.04 | tzanger | if the far end stops sending packets you'll see LLLLLL (loss) which would happen with a jump in lag |
14:30.19 | *** join/#asterisk mbranca (~matteo@81.208.92.210) |
14:30.29 | tzanger | you'll see GGGG too which is "grow" (jitter buffer growth) -- followed by some 'l' (late) |
14:30.31 | *** join/#asterisk wek (leroy@yoyoyo.net) |
14:30.32 | tzanger | I have to go over this again |
14:31.05 | clive- | thousands of these vclamping target from 206 to 180 |
14:31.06 | clive- | vclamping target from 206 to 180 |
14:31.11 | clive- | vclamping target from 184 to 180 |
14:31.11 | clive- | vclamping target from 181 to 180 |
14:31.15 | clive- | vclamping target from 181 to 180 |
14:31.17 | clive- | oops,,,sorry |
14:31.23 | tzanger | clive-: that's a very very small jitter buffer |
14:31.27 | tzanger | what are your jitter buffer settings? |
14:31.34 | *** join/#asterisk jeffik (~Jeff@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
14:31.36 | faa | anthm Hi |
14:31.52 | clive- | tzanger 180 is small? |
14:31.57 | tzanger | 180 is tiny |
14:32.06 | tzanger | you're clamping to 180 which is useless |
14:32.09 | tzanger | may as well not have one at all |
14:32.11 | clive- | tzanger what do you recommend |
14:32.12 | clive- | ? |
14:32.16 | tzanger | clive-: what are your settings? |
14:32.27 | clive- | 180 ms |
14:32.32 | clive- | for jitter buffering |
14:32.34 | tzanger | clive-: WHAT ARE YOUR SETTINGS |
14:32.42 | tzanger | as in pastebin.ca your jitter buffer settings |
14:32.47 | tzanger | 180 tells me nothing |
14:33.08 | clive- | in iax.conf? |
14:33.18 | tzanger | yes of course |
14:33.50 | clive- | jitterbuffer=yes |
14:33.51 | clive- | qos=lowdelay |
14:33.52 | clive- | ;dropcount=1 |
14:33.53 | clive- | maxjitterbuffer=180 |
14:33.54 | clive- | ;maxexcessbuffer=250 |
14:33.59 | tzanger | erase that maxjitterbuffer line |
14:34.01 | tzanger | let it default |
14:34.21 | MikeJ[Laptop] | wow... |
14:34.24 | MikeJ[Laptop] | ~pastebin |
14:34.24 | jbot | [pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca |
14:34.29 | MikeJ[Laptop] | clive-, use it |
14:34.29 | clive- | with the newjb I commented out teh otehr stuff |
14:34.34 | *** join/#asterisk jmacz (~jmacz@63.245.86.244) |
14:34.34 | *** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca) |
14:34.36 | clive- | sorry MIke |
14:34.39 | MikeJ[Laptop] | :P |
14:34.59 | clive- | tzanger will it find its own jiter buffer settings? |
14:35.22 | tzanger | yes that's the whole point of hte new jitter buffer |
14:35.29 | tzanger | I have jitterbuffer=yes and forcejitterbuffer=no |
14:35.30 | tzanger | that's it |
14:36.36 | [TK]D-Fender | I can't seem to remember the Polycom web-admin default passwords, anyone care to remind me? (doesn't appear to be "polycom" / "456") |
14:36.47 | clive- | forcejitterbuffer?...tas a new one |
14:36.59 | *** join/#asterisk inNEEDofHELP (~jordan@rrba-146-118-79.telkomadsl.co.za) |
14:37.06 | clive- | let em wait for this person to get off the call so I can restart asterisk |
14:37.19 | *** join/#asterisk virterm (~virterm@shiva.kanatek.com) |
14:37.35 | clive- | tzanger what value do you get on your line? |
14:38.33 | tzanger | clive-: it's dynamic |
14:38.36 | tzanger | that's the whole point |
14:39.15 | inNEEDofHELP | hello,i just need to know if my panasonic pbx d500 e1 connection can pass calls to0 asterisk server? any help would be greatly appreciated. |
14:41.00 | clive- | inneedofhelp, howzit,,, |
14:41.04 | thal | inNeedofHelp: ist it able to speak SIP? |
14:41.36 | clive- | tzanger, ok, I have removed the maxjitterbuffer,,,we will see if thatworks,,still confused why the RTT jumps from 350 to 1200 |
14:41.48 | inNEEDofHELP | thal,no i dont think so :) |
14:42.04 | inNEEDofHELP | clive-howzit..any ideas? |
14:42.23 | *** join/#asterisk brookshire (~matt@207.111.174.1) |
14:43.21 | vaewyn | BOOYAHH! errr... morning all |
14:44.30 | tzanger | clive-: tha'ts your network, it has nothing to do with the jb, the jb is trying to compensate |
14:44.50 | eKo1 | inNEEDofHELP: does it speak ISDN? |
14:44.55 | vaewyn | anyone played with TCP SMDI and * yet? Or other methods of doing MWI from/to Norhell switches? |
14:46.02 | SpaceBass | is there an easy way to disable music on hold on one specific extension? |
14:46.24 | *** join/#asterisk hubbaba (~me@68.114.5.160) |
14:46.31 | ChkDigit | SpaceBass: Put that extension into another context. |
14:46.41 | clive- | thanks for your help |
14:46.56 | SpaceBass | ChkDigit thats what I was trying to avoid |
14:47.38 | *** join/#asterisk da_monumental_1 (~da_monume@rrcs-24-172-102-198.midsouth.biz.rr.com) |
14:48.53 | SpaceBass | how about for the outbound routing, can I disable it at that point? |
14:49.22 | *** part/#asterisk clive- (~pirch@rndf-146-5-08.telkomadsl.co.za) |
14:50.12 | *** join/#asterisk Myshenka (~gunde@217.9.101.85) |
14:50.23 | hubbaba | Anyone had any luck setting up an AdTran TSU 600 with 24 FXS ports. I have a TE410P which has span 1 configured and working with a PRI. Span 2 goes to the CSU and is using a T1 crossover cable. Both ports light up green on the card when modprobed and ztcfg shows the channels properly. My problem is that when I plug in an analog phone, I don't get a dial tone. |
14:51.22 | ChkDigit | SpaceBass: I suppose you could add a SetMusicOnHold to any outbound extension, and make it go to dead air. |
14:52.09 | SpaceBass | dead air is great |
14:52.11 | Myshenka | Can anybody recommend a page where I can read more about placing the contents of extensions.conf in a mysql table when I want to use Realtime? I know how to deal with the static bits and with exten=>... but what to do with include=> and switch=> ? |
14:52.54 | SpaceBass | ChkDigit have two trunks one personal one business... dont want bruce springsteen or dave mathews playing when I put work calls on hold :) |
14:53.30 | ChkDigit | Gotcha. You should have a context for each trunk, selecting the right music on hold. |
14:53.53 | SpaceBass | ChkDigit that will work |
14:55.00 | Hmmhesays | openvpn just rocks some days |
14:55.11 | ChkDigit | openvpn rocks everyday. |
14:57.35 | jhiver | I can't get openvpn to always give machines the same private IP address, no matter how hard I try |
14:57.42 | jhiver | other than that, it works great |
14:58.20 | Hmmhesays | assign them statically |
14:58.40 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
14:59.25 | Nuxi | Definitely Asteirsk related => http://nytimes.com/2005/07/05/health/05sex.html?hp&ex=1120622400&en=98912eace83692a8&ei=5094&partner=homepage |
14:59.43 | Nuxi | lol |
14:59.55 | tzafrir | Nuxi, and for those of us unwilling to subscribe? |
15:00.43 | Nuxi | make up a name, etc. I use a new one every time the cookie expires. |
15:01.39 | *** part/#asterisk hubbaba (~me@68.114.5.160) |
15:02.25 | *** join/#asterisk santiago (~santiago@63.245.86.165) |
15:03.36 | Nuxi | tzafrir, found a "mirror" http://www.contracostatimes.com/mld/cctimes/living/science/12056017.htm |
15:05.34 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
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15:17.08 | eKo1 | Hmm... i have 'exten => 399,1,voicemailmain(@myvoicemailcontext)' and it still asks me for the mailbox. What am I doing wrong? |
15:20.37 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
15:21.28 | shido6 | yess |
15:21.36 | shido6 | it doesnt know what your voicemail mailbox is |
15:21.40 | shido6 | you're not passing that to it |
15:22.47 | vaewyn | ${EXTEN}@myvoicemailcontext :} |
15:24.47 | wek | trafrir- bugmenot.com |
15:28.15 | eKo1 | hmm...but it says in the documentation that the mailbox is an optional argument. |
15:28.37 | Jas_Williams | eKo1: it is without it it prompts you to enter you mailbox :) |
15:28.43 | file | yeah, so if it doesn't get it... it has to ask you |
15:28.46 | file | it's not psychic |
15:28.53 | *** join/#asterisk doolph (doolph@201.226.146.178) |
15:29.20 | eKo1 | Oh, I see. I thought it 'figured it out' automagically. |
15:30.08 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc0ml.dialup.mindspring.com) |
15:30.31 | Jas_Williams | eKo1: it will do if you put ${CALLERIDNUM}@myvoicemailcontext |
15:30.57 | eKo1 | yes, i was about to do that |
15:31.06 | Jas_Williams | if the callerID is not a voicemial box then it will prompt for one |
15:32.52 | eKo1 | worked like a charm. thanks |
15:34.33 | wunderkin | what are some codecs that can survive packet loss better than ulaw.. other than g729 |
15:34.35 | *** join/#asterisk Assid (~assid@203.115.64.61) |
15:34.37 | Assid | heya |
15:35.02 | wunderkin | out of 123 packets, i have 6% loss now |
15:35.16 | vaewyn | just NEVER use ${CALLERID} and the 's' extension unless you control ALL callerid sources and can prove it :P |
15:35.28 | vaewyn | s/extension/option |
15:35.41 | vaewyn | wunderkin: ilbc |
15:35.50 | vaewyn | handles packet loss the 'best' audibly |
15:36.10 | *** join/#asterisk funxion (~funxion@63.214.236.140) |
15:36.12 | wunderkin | ok |
15:36.35 | wunderkin | thanks, my cable modem used to be perfect before the holiday.. now its starting to suck.. :( |
15:36.45 | vaewyn | if it is packet loss due to bandwidth restrictions then try GSM also if you feel like it |
15:36.59 | wunderkin | na i have plenty of bandwidth |
15:39.00 | *** join/#asterisk Silik0n (~krice@newrso.suspicious.org) |
15:40.57 | Assid | hell yeah |
15:46.03 | tzanger | wunderkin: ulaw probably "survives" packet loss better than most others since hte least amount of voice information is sent per packet |
15:46.12 | tzanger | throw a good PLC jitterbuffer on it (asterisk has one) and it works well |
15:46.35 | tzanger | ilbc and g729 and perhaps speex too all have packet loss concealment algorithms to try and improve lost packet sound |
15:47.09 | wunderkin | so try turning on the jitter buffer? ilbc still is a little shaky |
15:47.53 | *** join/#asterisk |Vulture| (~V@user-0c6tr11.cable.mindspring.com) |
15:47.53 | wunderkin | i dont need it for all connections, only the one to my house.. which right now goes to a sip phone |
15:48.53 | tzanger | wunderkin: ilbc sounds like ass I've found, and I can't tell why, it's supposed to have excellent MOS |
15:49.00 | wunderkin | the box is in a data center, the problem is just the connection to my house.. |
15:49.02 | wunderkin | yeah |
15:49.07 | tzanger | I use gsm or g729, that's it. (ulaw for faxes) |
15:49.15 | *** part/#asterisk evangelion (~manzy_zet@ip-185-118.sn1.eutelia.it) |
15:49.27 | Assid | i thought gsm is "as clear" as ilbc |
15:49.28 | |Vulture| | tzanger: you don't like ilbc? |
15:49.33 | doolph | cant g729 support fax? |
15:49.39 | tzanger | |Vulture|: I want to like it (open source, open protocol) but it sounds like ass |
15:49.40 | |Vulture| | no ulaw for fax ONLY |
15:49.40 | eKo1 | no |
15:49.45 | tzanger | doolph: no compressed voice codec can do fax |
15:49.45 | Nivex | the few times I've used ilbc, it's been nice, but I tend to stick with ulaw and gsm for compat purposes |
15:49.57 | doolph | mmm |
15:49.58 | |Vulture| | tzanger: I thought it sounds better than gsm |
15:50.02 | tzanger | every time I turn on ilbc I get complaints about quality |
15:50.04 | doolph | dan |
15:50.06 | tzanger | use gsm and nobody complains |
15:50.12 | Damin | Oh shit.. |
15:50.12 | *** join/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca) |
15:50.14 | Optic | hey yo |
15:50.14 | ManxPower | I use Speex mostly. |
15:50.16 | |Vulture| | strange |
15:50.45 | |Vulture| | Ill have to try GSM again |
15:50.47 | Assid | isnt that supposed to be robotic? |
15:50.52 | Nivex | for some reason I had Speex enable itself on a link this weekend and the other end said I was really choppy |
15:51.08 | Nivex | Assid: gsm codec doesn't sound any worse than a cellphone |
15:51.15 | ManxPower | Nivex: That would be when you don't use disallow=all and then allow=thecodecyouwant |
15:51.17 | Assid | same thing? |
15:51.17 | |Vulture| | cell phones sound bad |
15:51.18 | doolph | for me its same |
15:51.19 | Optic | i've got a little system with 10 spa-841's and 5 polycom ip500's... when a spa-841 calls a polycom, the spa user's voice is very quiet to the polycom user... polycom to polycom or spa to spa calls are fine |
15:51.22 | Assid | hrmm |
15:51.27 | Assid | thats weird |
15:51.33 | Assid | cellphones sound good here |
15:51.35 | Nivex | ManxPower: yeah. I went back and got rather specific with it. |
15:51.38 | doolph | because polycom is better |
15:51.46 | tzanger | Assid: lpc10 is robotic (and very funny) |
15:51.46 | [TK]D-Fender | Hey, anyone know where I can get documentation on Polycom's Micro-Browser XML format? Or at least some samples... |
15:51.53 | |Vulture| | I notice a big diff on gsm and ulaw |
15:51.58 | wunderkin | is gsm known as something else? in a number form? i dont see it on the list on my gs101 |
15:52.02 | tzanger | |Vulture|: well of course you will. :-) |
15:52.10 | Optic | the polycom phones are nice :) |
15:52.16 | vaewyn | domo ari gato mr roboto |
15:52.27 | Nivex | What I really couldn't figure out is why my friend's asterisk box would allow a ulaw call but then couldn't hear the audio. I switched to gsm and it worked just fine. |
15:52.29 | [TK]D-Fender | wunderkin : GS101 doesn't support GSM AFAIK |
15:52.30 | |Vulture| | Optic: agreed just about to get some new 501s |
15:52.35 | wunderkin | damn |
15:52.43 | Optic | it's a bit of a problem in our system though |
15:52.51 | Optic | they don't play well with the sipuras |
15:53.18 | Jas_Williams | 2 |
15:59.12 | *** join/#asterisk santiago (~santiago@63.245.86.244) |
15:59.41 | *** join/#asterisk vmlinuz (~nabudocon@red-corp-200.76.231.29.telnor.net) |
15:59.57 | |Vulture| | g729 sounds good.. but too expensive to use other than on external phones linked to * for me |
16:00.02 | *** join/#asterisk blazint (~blazin@cm99.epsilon201.maxonline.com.sg) |
16:00.18 | |Vulture| | isn't there some GPL version of g729 in the works? |
16:00.26 | tzanger | |Vulture|: $10 is too expensive? |
16:00.32 | loud | the intel one, but don't be cheap |
16:00.34 | vaewyn | still not legel in the US even if it is GPL |
16:00.34 | tzanger | |Vulture|: no, g729 cannot be GPLd |
16:00.42 | loud | oh |
16:00.51 | |Vulture| | tzanger: ah okay yea I have 4 licenses |
16:01.12 | doolph | what's the difference between the intel one |
16:01.14 | |Vulture| | tzanger: I am using ilbc between offices and for LD calls, then g729 to phones outside the LAN |
16:01.39 | tzanger | |Vulture|: you might want to try gsm -- smaller bandwidth footprint and (in my testing with a staff of 40) far better quality |
16:01.58 | |Vulture| | tzanger: I will try it tomorrow morning |
16:02.02 | vaewyn | I was real tempted to go beat the #@!$@#$ outta the patent holders when I found they were at VON :} |
16:02.10 | *** join/#asterisk HDG-laws|sl33p (~yo@adsl-69-208-237-54.dsl.sfldmi.ameritech.net) |
16:02.14 | |Vulture| | lol |
16:02.42 | *** part/#asterisk Myshenka (~gunde@217.9.101.85) |
16:02.44 | wunderkin | well.. should i try the sip jitterbuffer |
16:02.57 | ManxPower | wunderkin: That stuff is only in CVS-HEAD |
16:03.03 | vaewyn | do... or do not... there is no try ;P |
16:03.06 | HDG-laws|sl33p | Anyone around who can help answer a few questions about Asterisk for use with Snom 360 + PSTN failover configuration? |
16:03.14 | wunderkin | yeah im using head |
16:05.00 | *** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk) |
16:05.09 | tzanger | vaewyn: what's wrong with the g729 patent? |
16:05.11 | tzanger | seriously? |
16:05.39 | tzanger | it's a novel, innovative invention. While I dont' think that software patents should last for more than 5 years, it's a valid patent in my eyes |
16:06.07 | |Vulture| | its a good codec |
16:06.29 | vaewyn | software patents in general I have issue with... but not even making it 'free for non-commercial use' is just being greedy |
16:06.31 | |Vulture| | I just wish I could buy like 50 licenses and have them be used dynamically on all my servers |
16:06.58 | doolph | why do you need 50 licenses? |
16:07.07 | tzanger | I have no issues with software patents, just as I have no issues with hardware patents. it takes a lot of work and expense to do this kind of work and there should be protections in place to let me recoup my investment before the floodgates open |
16:07.08 | doolph | will it runs within internet or just lan? |
16:07.17 | tzanger | now 17 years is a bit long, even for hardware |
16:07.22 | vaewyn | I just have 2 servers with g729a landing points... rest of them don't talk it |
16:07.27 | ManxPower | I don't have much of a problem with the G729 patent holders. They will at least license it in small qtys. |
16:07.35 | tzanger | and the whole idea of frivilous and stupid, broad, non-novel patents sucks |
16:07.36 | |Vulture| | I have 9 * servers each making ~10 calls at a time |
16:07.36 | cpm | But software patents DO last for more than 5 years, a lot more in fact. This whole software patent thing is bogus. It's ALL derived, everything software, and most "things" in general are inspired by something else. It's all bogus. |
16:07.46 | tzanger | cpm: bullshit |
16:07.47 | *** join/#asterisk trickyrick77 (~rsegrest@207.111.174.1) |
16:08.02 | vaewyn | ManxPower: umm... your definition of 'small quantities' must be different then mine |
16:08.16 | ManxPower | vaewyn: Um, 1 license at a time? |
16:08.19 | |Vulture| | lol |
16:08.20 | tzanger | cpm: software patents are no more evil than hardware patents. And yes things are based off other things but that doesn't mean that the approach or resolution of the technical issues was easy or obvious. |
16:08.34 | ManxPower | vaewyn: Maybe you are thinking of the greedy G723.1 bastards? |
16:08.39 | tzanger | cpm: as I said, I feel that 17 years for software is insane. But patenting software isn't insane. |
16:08.52 | tzanger | cpm: just as I said that I feel that 17 years for hardware's insane. |
16:08.59 | vaewyn | ManxPower: go ahead... call them up and try and buy one license... they will refer you to a 'reseller' that has purchased a 10k block from them |
16:09.13 | ManxPower | vaewyn: I just call up digium. |
16:09.26 | ManxPower | Digium issues me a license. All done. |
16:09.30 | vaewyn | exactly... hence... the patent holder is not allowing small scale sales |
16:09.35 | vaewyn | it's just digium being nice |
16:09.47 | vaewyn | they had to buy a block of them to start |
16:09.54 | cpm | What's the point of a patent? |
16:09.57 | ManxPower | vaewyn: Hmm? Until about a year ago, digium got each license, individually from the patent holders. |
16:10.00 | tzanger | vaewyn: the patent holder is allowing it. how is it bad? |
16:10.07 | *** join/#asterisk alephant (~cmd@c-24-3-52-93.hsd1.pa.comcast.net) |
16:10.11 | alephant | Hi all... |
16:10.22 | tzanger | you can't buy individual MAC addresses either, you have to buy a big block... how is this a problem? |
16:10.48 | Nuxi | you can make up a mac address with out a license. |
16:10.56 | vaewyn | exactly :} |
16:11.01 | vaewyn | good call Nuxi |
16:11.09 | tzanger | Nuxi: actually that's not supposed to be allowed |
16:11.14 | alephant | ...if I connect * to an analog phone line, and I want to bounce incoming calls to my cell phone, how many phone lines and FXS's do I need? |
16:11.21 | ManxPower | <PROTECTED> |
16:11.32 | tzanger | same with PCI IDs... you can hijack someone else's (digium did this even) or make one up |
16:11.42 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
16:11.51 | Nuxi | oops was that out loud. |
16:13.26 | vaewyn | I think... 1-5 years... let them patent something they can PROVE within reasonable doubt is unique (and not just because it is done on a computer) and rape the crud outta it... but after that let it be free |
16:13.54 | cpm | tzanger, dont' get me wrong, I don't blame digium in the least, that's the way of the world, but I do stand by the concept that "ideas" should be patented is bullshit. |
16:14.39 | mutilator | whats being patented? |
16:14.53 | vaewyn | heck... even real 'hardware' patents are getting out of control these days... I mean.. they let them patent a swing for @@#$@ sake |
16:14.53 | shido6 | ideas shouldnt be patented? |
16:15.09 | shido6 | wow |
16:15.54 | vaewyn | I love it that I can't patent 1 + 1 = 2... but I can patent my $tot = 1 + 1; print $tot; |
16:16.06 | shido6 | you ever had an idea taken from you and someone else made millions because of it? |
16:16.18 | mutilator | doesn't everyone |
16:16.22 | Nuxi | actually you can patent 1 + 1 = 2. It's a method. |
16:16.34 | mutilator | w00t |
16:16.36 | shido6 | no |
16:16.42 | mutilator | charge elementary schools to use it |
16:16.54 | Nuxi | It's a method of constructing 2. |
16:16.56 | tzanger | cpm: ideas shoudl not be patented, and indeed they can't be |
16:17.00 | tzanger | PROCESSES are patented |
16:17.01 | *** join/#asterisk mkrufky (~mk@68.160.103.77) |
16:17.07 | Assid | right |
16:17.14 | Assid | hyperthreading.. hypertransport |
16:17.16 | Nuxi | ideas can easily be wrapped in process. |
16:17.23 | mutilator | i love it when you talk dirty Assid |
16:17.26 | Assid | bottom line.. nothing gets done |
16:17.45 | Assid | mutilator: now unless your a girl.. you aint getting none |
16:18.28 | Assid | burn! |
16:18.29 | Assid | okay |
16:18.30 | *** join/#asterisk cinix (~ax@24-52-166-190.lndnnh.adelphia.net) |
16:18.37 | Assid | i desperately need to get more sleep in |
16:19.06 | eKo1 | ditto |
16:19.07 | DarthClue | Assid: sleep is waste of time. I should know. |
16:19.24 | cpm | Sure they can. How many software patents would like to review that are nothing more than ideas? |
16:19.56 | *** join/#asterisk Lee__ (~lee@ool-44c26fa3.dyn.optonline.net) |
16:20.10 | *** join/#asterisk damajor (~damajor@tuxmania.org) |
16:20.12 | cpm | some history, http://www.vrijschrift.org/swpat/030508_1/ |
16:20.20 | Nuxi | It used to be you had to have an implementation to patent. now you can patent a "gee wouldn't it be nice if ..." |
16:20.35 | DarthClue | Come to cluecon. The only voip conference to feature nearly every major VOIP developer. |
16:20.47 | Assid | DarthClue: tell me about it |
16:20.55 | *** join/#asterisk rob (~rob@2001:4c20:0:2:211:43ff:fecd:8834) |
16:21.05 | Lee__ | if there are any app_conference users here, I get no sound when connecting two SIP phones to the same room. Has someone else found this? |
16:21.08 | Assid | 4-5/day.. everyday ..since months |
16:21.12 | Assid | starting to really get to me |
16:21.48 | vaewyn | eep! prior art! |
16:21.57 | Assid | Nuxi: you are in violation |
16:22.04 | rob | Hi, has anyone got Asterisk running properly on OS X 10.4.1? |
16:22.05 | Assid | DarthClue is already patented by me |
16:22.14 | Assid | you now owe me 100 million USD |
16:22.18 | *** join/#asterisk vmlinuz (~nabudocon@red-corp-200.76.231.14.telnor.net) |
16:22.21 | vaewyn | Nuxi: is that argv[0] or argv[1] ? ;P |
16:22.37 | Lee__ | rob, no. |
16:22.54 | Nuxi | Assid, no, I patented a diferent way of using DarthClue. not the closet use you used. |
16:22.55 | Assid | i wanna patent the process of patenting |
16:22.55 | rob | There's a package floating round which is a 2003-10 CVS, which seems to be broken with dtmf on the 7960. |
16:23.12 | Lee__ | have you contacted the package author? |
16:23.12 | Assid | Nuxi: i patented DarthClue itself.. |
16:23.25 | Assid | therefore you are in violation |
16:23.29 | Nuxi | Assid, I'll see you in court. |
16:23.32 | Assid | fine |
16:23.39 | Nuxi | fine. |
16:23.46 | rob | Lee__: The developer's site on versiontracker takes you to Asterisk.org. |
16:23.47 | Assid | you gotta travel 7000 miles to come and meet me now |
16:24.15 | Assid | DarthClue: that was already patented earlier.. |
16:24.20 | Lee__ | rob, see if you can track down an email address. there's also lists.digium.org. |
16:24.22 | Assid | yuo can add it to your tab |
16:24.48 | Assid | but since you helped me earlier.. i give you 1 month free usage.. license free |
16:25.29 | mkrufky | this patent stuff is all a big joke, right? (just entered room) |
16:25.43 | vaewyn | yes and no :P |
16:25.47 | mkrufky | heh |
16:25.49 | vaewyn | SWEET!!!! |
16:26.06 | Nuxi | mkrufky, some people are grumpy about codec patents. |
16:26.29 | mkrufky | i am grumpy about it too... |
16:26.32 | mkrufky | :-( |
16:26.35 | rob | Lee__: I will do, but is there any other way that I can deal with the problem other than having to chase someone down? |
16:26.38 | vaewyn | the other half shall be solved when you open the patents on killing politicians :P |
16:27.12 | mutilator | those are called hits ;P |
16:27.30 | Lee__ | rob. have you tried setting dtmf=rfc2833 for the Cisco phone in sip.conf? |
16:27.36 | mkrufky | nuxi: you AGREE with these patents?!? |
16:27.46 | vaewyn | but I get to use openoffice for free ;P |
16:27.46 | |Vulture| | are we still on this? |
16:27.50 | mkrufky | lol |
16:27.51 | vaewyn | yep |
16:28.09 | mkrufky | gotcha |
16:28.20 | Lee__ | couldn't resist |
16:28.23 | rob | Lee__: Yes, I have, unfortunately it seems to be doing nothing else. |
16:28.34 | tzanger | Lee__: :-) |
16:28.43 | Lee__ | rob, you may not like my other suggestion. |
16:28.44 | tzanger | Nuxi: you can't patent ideas |
16:28.47 | tzanger | you patent processes |
16:28.57 | mkrufky | i think that some patents are okay, but not the stupid little patents where m$ will patent the use of scrollbars or something basic like that |
16:29.01 | Nuxi | any idea can be wrapped loosely in a process. |
16:29.04 | Lee__ | rob, use Linux. |
16:29.12 | vaewyn | tzanger: go look at the recent patent filings... find a line of code in them... trust me... they ain't there |
16:29.26 | tzanger | mkrufky: those are obvious and trivial patents and should never be allowed. that's a failure of the patent system being greedy |
16:29.33 | mkrufky | exactly |
16:29.39 | tzanger | vaewyn: see my last comment to mkrufky |
16:30.02 | mkrufky | tzanger: it's like a new word for technological monopoly |
16:30.10 | Lee__ | it would definitely benefit Asterisk if all the codecs used internally were non-patented. |
16:30.32 | vaewyn | heck... we might as well move to... anyone can patent anything that hasn't been patented before... but you only get it for 1 year... you can't get a hold inthe market in that long... screw you |
16:30.43 | Nuxi | It benefits Asterisk to be able to use all codecs. |
16:30.50 | Nuxi | patented or not. |
16:31.16 | mkrufky | i havent looked into that... how does asterisk use a patented codec? |
16:31.32 | mkrufky | is there a binary version of the codec that must be linked or something? |
16:31.39 | Lee__ | mp3 for moh, g729 for SIP/IAX channels. |
16:31.48 | Lee__ | yeah, binary. |
16:32.02 | mkrufky | what a PITA |
16:32.02 | vaewyn | or... even better make the patents only patentable by a single person that is responsible for it... no more companies holding patents and such |
16:32.04 | Lee__ | or compiled outside of Asterisk in the case of mp3 support. |
16:32.16 | shido6 | whew |
16:32.39 | Nuxi | make mpg123 |
16:32.42 | Lee__ | mkrufky: tip your glass to the companies that patented the codec for that one. |
16:32.53 | mkrufky | lol |
16:33.08 | mkrufky | the companies are taking this stuff too far if u ask me |
16:33.29 | mkrufky | i cant believe the hurdles we have to go through just to get a datasheet to write a linux driver |
16:33.45 | mkrufky | <-- video4linux dev |
16:33.56 | *** join/#asterisk shaonss (~shaonss@61.68.15.40) |
16:34.21 | vaewyn | actually... in the long run the best thing would be if all standards bodies required free-to-use patents grants for anything they ratified... that would rock right there... and drive restrictive patent holders out of business |
16:34.26 | rob | Lee__: Ah, I found the developer's site and have now upgraded. |
16:34.32 | rob | However, I still have the problem. |
16:34.50 | |Vulture| | Anyone know if it is possible to subtract or add to a ${TIMESTAMP}, I need to add 3 seconds to it |
16:35.04 | mutilator | i think they do that with medicine now don't they vaewyn? |
16:35.14 | Lee__ | mkrufky: this month's LJ says the V4L driver is currently unmaintained. Is that true? |
16:35.28 | vaewyn | mutilator: in some cases yes... I think more are running that way |
16:35.53 | shaonss | calling card application need help!!!!!!! |
16:36.04 | mkrufky | Mauro Carvalho Chehab has just stepped up as the new maintainer |
16:36.06 | mutilator | i just heard something about that in the news a bit ago, they don't require royalties on exprimental use of stuff |
16:36.18 | mkrufky | vaewyn: Mauro Carvalho Chehab has just stepped up as the new maintainer |
16:36.39 | mkrufky | vaewyn: I have been helping him out a lot lately with patching to -mm |
16:36.56 | shaonss | which calling card application is the best for asterisk? |
16:37.05 | mkrufky | vaewyn: and also helping to mediate between v4l and dvb trees |
16:37.10 | Lee__ | cool. I don't use video much but it's good to know it's not abandoned in the kernel. |
16:37.38 | Lee__ | I just compiled 2.6.12 and noticed a load of DVB additions since 2.6.8 |
16:37.48 | mkrufky | Lee__: in fact, there has been a TON of activity in v4l lately |
16:38.05 | mkrufky | Lee__: DVB is a separate project, although there is some overlap with v4l |
16:38.08 | |Vulture| | Anyone know how to add seconds to a ${TIMESTAMP}? |
16:38.13 | vaewyn | speaking of which... I need my trusty friend to finish the DVB drivers so they will work with analog alos attached to the pcHDTV2000/3000 |
16:38.20 | Lee__ | Delta Song uses Linux for each seat's video display. I saw the FB penguin boot up :) |
16:38.28 | mkrufky | vaewyn: do u have a link to that article in lj? |
16:38.31 | *** join/#asterisk Cheetah (~Snak@62.217.48.111) |
16:38.49 | mkrufky | vaewyn: i believe that card has full analog support |
16:38.55 | Cheetah | hi folks |
16:39.07 | Cheetah | is there a way to redirect an incoming call in extensions.conf to the voicemail box after a certain amount of rings? |
16:39.09 | shaonss | can anyone please help for calling card application? |
16:39.12 | rob | Lee__: How will Linux help me on one simple dtmf problem, surely the configuration issue could exist on anything? |
16:39.20 | mkrufky | vaewyn: If the analog part needs fixing, let me know and I can probably take care of that for you |
16:39.20 | Cheetah | like, 10 seconds/rings and nobody picked up |
16:39.30 | vaewyn | mkrufky: was Lee__ that saw that |
16:39.39 | Lee__ | rob, because I am not using OS X and I don't have said DTMF problem and we have the same phones. |
16:39.50 | mkrufky | oops |
16:40.03 | mkrufky | Lee__: do you have a link to that article in lj ? |
16:40.08 | Lee__ | mkrufky: it wasn't an article, it was a one liner in the diff -u kernel notes section. |
16:40.19 | mkrufky | oh, i gotcha |
16:40.33 | mkrufky | u will notice that the MAINTAINER file has been updated in 2.6.13-rc1 |
16:40.37 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.190.115.68.195.rev.coltfrance.com) |
16:40.43 | mkrufky | :-) |
16:40.43 | vaewyn | mkrufky: problem right now is that I can't seem to get it to run in analog and DVB... I have to toast the modules and reload the other set to switch between DVB/HDTV and analog |
16:40.53 | Lee__ | I'm not that cutting edge with kernels. I stick to "releases" if there are such things. |
16:40.55 | rob | Lee__: What are your settings for SIPDefault: dtmf_inband:0 , dtmf_outofband: avt_always, dtmf_db_leve: 3, dtmf_avt_payload: 101? |
16:40.56 | vaewyn | mkrufky: can't do that from in mythtv so :P |
16:41.28 | *** join/#asterisk NightHawke (~NightHawk@66-190-111-175.static.sprn.tx.charter.com) |
16:41.33 | NightHawke | hi'lo |
16:41.40 | *** join/#asterisk yoink (~yoink@MTL-HSE-ppp165512.qc.sympatico.ca) |
16:41.42 | Lee__ | rob, I'll msg you. is that ok? |
16:41.48 | *** join/#asterisk The_LightSide (~dial@tpr-165-247-230.telkomadsl.co.za) |
16:41.57 | rob | Lee__: That's fine thanks. |
16:42.12 | mkrufky | vaewyn: interesting...... it is cx88 board #22, correct? |
16:43.49 | NightHawke | i got a set of TDM 400P FXO/FXS cards. what i want to do is hook up a set of analog phones into the FXO card. |
16:43.59 | NightHawke | how do i register each extension in AMP? |
16:44.12 | |Vulture| | FXS for phones;FXO for lines |
16:44.19 | NightHawke | FXS thanks |
16:44.30 | NightHawke | i got the ZAP extensions in there but when i try to get a outside line i get a busy |
16:44.50 | |Vulture| | NightHawke: I am not sure for amp, but what you want to do is edit zaptel.conf and zapata.conf to see the TDM cards |
16:44.51 | mkrufky | vaewyn: its very busy in here..... meet me in #v4l and I can probably help you with your board |
16:44.54 | mutilator | fricking rye chips |
16:45.01 | mutilator | thats what i hate about chex mix |
16:45.07 | mutilator | the damn rye chips |
16:45.08 | NightHawke | AMP has direct links to see the conf files |
16:45.08 | Lee__ | so...anyone using app_conference successfully? When I connect to a conference I get no audio. |
16:45.33 | |Vulture| | NightHawke: then you will just say Dial(Zap/(Zap FXS Line),20(time in seconds to dial),r) |
16:45.50 | vaewyn | mkrufky: :} yeah... it gets that way.. not sure which chip... not in front of me at the moment... I have the dmesg | grep bttv output for it |
16:46.17 | |Vulture| | mutilator: I don't like all the pretzles |
16:46.18 | NightHawke | |Vulture|, i see the extensions in zaptel.conf now |
16:46.58 | vaewyn | mkrufky: hahahahaha I think it know's you... it just worked without reloading hahahaha |
16:47.16 | mkrufky | vaewyn: problem solved already??? |
16:47.42 | *** join/#asterisk LeoB (~chatzilla@wireless-107.media.mit.edu) |
16:47.57 | vaewyn | mkrufky: I think you scared it into submission... that is wierd... oh well... if it happens again I'll get some diag info on it |
16:48.00 | mkrufky | vaewyn: i just noticed that pchdtv3000 had two different boards, one is bttv-based and the other cx88-based |
16:48.04 | mkrufky | haha |
16:48.08 | shido6 | oh lord |
16:48.12 | shido6 | dont get me STARTED |
16:48.14 | mkrufky | well i'll be happy to help you |
16:48.14 | vaewyn | mine is the 2000 board so... |
16:48.17 | shido6 | MSI TV@anywhere |
16:48.17 | LeoB | hello there, how can I prevent log messages from eating my entire hard drive? |
16:48.19 | shido6 | BS |
16:48.36 | NightHawke | ok vulture, i'm lost on this... that and the effing phone is ringing |
16:48.36 | shido6 | stinkin ati and linux hate each other, two tv cards |
16:48.44 | NightHawke | i'll ahve to deal with it later on |
16:48.47 | shido6 | i cant get sound to work on my TV@nywhere |
16:48.50 | shido6 | conexant based |
16:49.00 | mkrufky | shido6: meet me in #v4l |
16:49.03 | LeoB | they've eaten 13GB already... |
16:49.07 | Lee__ | LeoB: logger.conf |
16:49.09 | vaewyn | shido6: hehehe... I am fairly sure I will never buy an ATI device for under linux ever agian :P |
16:49.28 | mkrufky | vaewyn: maybe pchdtv3000 and pchdtv2000 use same chips? i dont see direct support for 2000 board anywhere |
16:50.35 | vaewyn | mkrufky: I know they use different firmware for one chip at least 'or51211' versus like 'or51221' or such |
16:50.44 | vaewyn | not sure which chip that is though |
16:52.00 | Lee__ | nobody with app_conference? too specialized? a hack? |
16:52.15 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
16:53.38 | NightHawke | Dial(Zap/(Zap FXS Line),20(time in seconds to dial),r) i assume that 20 is the extension # assigned to the zap extension |
16:54.21 | |Vulture| | NightHawke: I doubt its 20 since you only have 2 TDM cards so 1-4,5-8 |
16:54.25 | vaewyn | umm... no... it is the time in seconds to ring the zap interface before giving up and going on in the dialplan |
16:54.36 | HDG-laws|sl33p | Are any digium cards required for a pure SIP / * setup ? |
16:54.47 | |Vulture| | no |
16:54.50 | shido6 | vaewyn NEVER |
16:54.50 | vaewyn | HDG-laws|sl33p: nope |
16:54.55 | HDG-laws|sl33p | If I want failover to PTSN, then I will |
16:54.57 | shido6 | I love the AIW in xp but |
16:54.57 | HDG-laws|sl33p | Correct? |
16:54.58 | shido6 | not |
16:55.00 | shido6 | in linux |
16:55.38 | mkrufky | vaewyn: i dont know... i havent seen any bttv/dvb boards yet.. so im not sure how they fall in v4l/dvb trees |
16:56.00 | NightHawke | Dial(Zap/(Zap 1-3),20(time in seconds to dial),r) |
16:56.35 | vaewyn | shido6: if I'm gonna have that much $$ and hassle though I am going to have my hdtv ;P so linux with the pchdtv or air2pc cards is the way to go :P |
16:56.56 | spyroux | does the version 1.0.9 of asterisk include Realtime configuration ? |
16:57.42 | vaewyn | I think realtime is cvs head only |
16:58.17 | shido6 | ooh |
16:58.18 | shido6 | http://www2.ati.com/drivers/linux/linux_8.14.13-inst.html |
16:58.24 | shido6 | but Ive done this manually |
16:58.50 | ManxPower | Realtime is only in CVS-HEAD |
17:00.13 | HDG-laws|sl33p | With a 100% SIP system + *, what would I need in order to have a POTS failover ? |
17:00.17 | ManxPower | Through the generosity of our Premier sponsor [for Cluecon], Sangoma Technologies |
17:00.21 | ManxPower | Ick. |
17:00.25 | spyroux | ManxPower: thanks |
17:02.30 | |Vulture| | ManxPower: any clue how I would add 3 seconds to a TIMESTAMP? |
17:02.48 | mkrufky | vaewyn: as of yesterday, there is now support for fusionhdtv3 gold (t/q) in video4linux cvs for both dvb and analog |
17:04.20 | vaewyn | |Vulture|: I think you have to use a temp variable... ${TIMESTAMP} is not writeable |
17:05.01 | Lee__ | I'm going to call it on the wiki. app_conference is broken with SIP phones and CVS HEAD. |
17:05.09 | |Vulture| | vaewyn: yes but right now I am just using TEMP, I have to isolate the timestamp's second side, then subtract then readd it together |
17:05.15 | |Vulture| | file: any ideas? |
17:05.48 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net) |
17:06.05 | |Vulture| | I have: Math(TEST,${TIMESTAMP}+3) but returns "TEST|20050705-130136+3" = 20050708.000000 |
17:06.41 | outtolunc | Variables marked with a * are builtin functions and can't be set, |
17:06.41 | outtolunc | only read in the dialplan. Writes to such variables are silently |
17:06.41 | outtolunc | ignored. |
17:06.49 | outtolunc | ${TIMESTAMP} * Current date time in the format: YYYYMMDD-HHMMSS |
17:07.13 | |Vulture| | yea... so I need to isolate HHMMSS and then subtract 3, then recombine it with the YYYYMMDD |
17:07.50 | |Vulture| | but how do I isolate it? |
17:07.53 | vaewyn | |Vulture|: SetVar(TEMP=$[${TIMESTAMP}+2]) |
17:08.00 | vaewyn | err... or 3... not 2 :} |
17:08.11 | |Vulture| | vaewyn: Ill try that |
17:08.48 | *** join/#asterisk brenda (~b@c-67-182-205-227.hsd1.ut.comcast.net) |
17:09.36 | *** join/#asterisk DrRighteous (~DrRighteo@68.199.175.49) |
17:09.44 | ManxPower | put a space around the + |
17:09.49 | |Vulture| | kk |
17:10.04 | DrRighteous | anyone have experience with mutlple instances of asterisk on a box? |
17:10.06 | vaewyn | ergh... that's right also |
17:10.17 | ManxPower | I use this: SetVar(LOOP=$[${LOOP} + 1]) |
17:10.46 | ManxPower | DrRighteous: Only when the version of "ps" on the box shows each thread as a seperate instance of Astersik |
17:10.53 | vaewyn | someone needs to program a 'foreach' application :} |
17:10.53 | |Vulture| | ManxPower: when I put the space it set it to 3 |
17:11.10 | ManxPower | |Vulture|: then try my method |
17:11.25 | ManxPower | vaewyn: there might be in AEL? |
17:11.32 | bkw_ | ManxPower, so why the Ick? |
17:11.38 | DrRighteous | ManxPower: my problem is that Im using the -C to allow for multiple asterisk directories for resources/pids... but I still getAsterisk already running on /var/run/asterisk/asterisk.ctl |
17:11.53 | ManxPower | bkw_: Having a primary competitor sponsering an "asterisk developers conference" |
17:12.08 | file | Cluecon isn't an asterisk developers conference |
17:12.12 | ManxPower | Has Sangnoma even contributed any code to the Asterisk CVS. |
17:12.18 | bkw_ | yes |
17:12.24 | ManxPower | file: then someone should change the /topic |
17:12.30 | |Vulture| | nah that returned: WARNING[15159]: ast_expr.y:666 op_plus: non-numeric argument |
17:12.36 | ManxPower | Ah, PBX, not Asterisk. |
17:12.38 | file | the topic is perfectly fine :P |
17:12.39 | bkw_ | its about 33% asterisk related |
17:12.39 | ManxPower | That's not so bad then. |
17:13.00 | ender | anybody know what the default username/password is on Sipura phones? |
17:13.03 | ender | IP301 ? |
17:13.05 | ManxPower | |Vulture|: maybe you need a newer version of bison. All I can say is that The sample I pasted is verbatum from my extensions.conf |
17:13.20 | vaewyn | ummm then if it is only 33% asterisk is it taking 50% of the /topic? ;P |
17:13.23 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
17:13.31 | bkw_ | ManxPower, and since when is illegal to be a competitor in an open market? |
17:13.40 | |Vulture| | ManxPower: bison (GNU Bison) 1.875c what are you running? |
17:13.52 | tsume | *0.00000000001kb |
17:14.07 | bkw_ | tsume, what software? |
17:14.20 | tsume | bkw_: its medical building software :) |
17:14.24 | bkw_ | ah |
17:14.26 | vaewyn | ender: IP301 is polycom... not sipura |
17:14.38 | ender | oh shit. |
17:14.38 | tsume | bkw_: I didn't feel like charging 700 extra dollars to all customers |
17:14.41 | ender | I'm a dork. |
17:14.43 | file | bkw_: I want to KILL cube1 |
17:14.44 | vaewyn | :} |
17:14.49 | ender | I just replaced my sipura w/ a polycom. |
17:14.53 | Optic | mooo |
17:14.56 | ender | anybody know what the ploycom default password is? (: |
17:14.59 | Optic | ender: which polycom? |
17:15.02 | Optic | ender: 456 |
17:15.06 | vaewyn | polycom/456 |
17:15.10 | Optic | ender: for the web it's Polycom/456 |
17:15.14 | tsume | bkw_: they kb'ed me fast out the room like a car on nitros |
17:15.15 | Optic | you need the cap P :) |
17:15.36 | tsume | bkw_: Its like channel of deliquent trolls :P |
17:15.38 | ender | Optic: 301 thakns. |
17:15.43 | Optic | polycom's ftp provisioning is a bit shady |
17:16.10 | ender | hrm, 'polycom' user and '456' password is not working. |
17:16.32 | Optic | you need the cap P :) |
17:16.35 | Optic | Polycom/456 |
17:16.50 | ender | ah |
17:17.35 | Optic | 301 eh? We have a bunch of 500's here |
17:17.42 | ender | yeah, 301. |
17:17.48 | Optic | sipura callers sound very quiet on them |
17:17.53 | Optic | what's the difference with the 1's? |
17:17.56 | Optic | 301, 501, etc? |
17:18.02 | ender | seems like a decent value bump from the Sipura phones |
17:18.08 | Optic | yes |
17:18.10 | ender | more memory and something else I thought. |
17:18.13 | vaewyn | more mem |
17:19.01 | *** part/#asterisk loick (~loick@81.255.80.161) |
17:20.32 | vaewyn | I wish they would enable the microbrowser on the 500s as well... |
17:20.42 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
17:21.41 | HDG-laws|sl33p | Anyone using Snom 360, I'm preparing to buy 15 of them |
17:24.05 | HDG-laws|sl33p | :( I killed the channel |
17:24.34 | vaewyn | sorry... snom's are technically very nice phones... but their user interface and feel are attrocious |
17:24.46 | PBXtech | what is the cisco directory that was included in the older Ast@home? |
17:25.09 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
17:25.10 | HDG-laws|sl33p | Any other suggestions then for 6 CO line capable SIP? |
17:25.14 | SpaceBass | PBXtech what is it? xml |
17:25.17 | PBXtech | ya |
17:25.53 | vaewyn | Polycom 600s |
17:26.08 | SpaceBass | PBXtech its just an xml file that you have to manually edit |
17:26.20 | SpaceBass | for 6 lines, the 7960 isnt bad |
17:26.21 | PBXtech | they had a web interface to it |
17:26.33 | SpaceBass | configuring 15 is just as easy as 1 |
17:27.12 | MikeJ[Laptop] | ~rtfw |
17:27.12 | jbot | hmm... rtfw is really used by f*cking idiots to equivalate thier male inferriority because they don't know the answer |
17:27.23 | bkw_ | jbot forget rtfw |
17:27.36 | MikeJ[Laptop] | jbot, remeber the old rtfw :) |
17:27.38 | crash3m | read the fuckin web? |
17:27.39 | bkw_ | jbot no rtfw |
17:27.50 | file | jbot: forget rtfw |
17:27.54 | SpaceBass | always thought it was stfw |
17:27.57 | crash3m | me to |
17:28.00 | vaewyn | bkw_ forget jbot |
17:28.01 | SpaceBass | and pretty much a rude thing to say, to boot |
17:28.02 | vaewyn | :P |
17:28.04 | file | jbot: no rtfw |
17:28.08 | *** join/#asterisk Assid (~assid@203.115.64.61) |
17:28.18 | vaewyn | ~rtfm |
17:28.18 | jbot | i heard rtfm is really used by f*cking idiots to equivalate thier male inferriority because they don't know the answer |
17:28.27 | vaewyn | hehehe |
17:28.36 | SpaceBass | ~stfw |
17:28.37 | jbot | methinks stfw is Search The F*cking Web. See http://justf*ckinggoogleit.com/ |
17:28.43 | SpaceBass | lol |
17:28.55 | *** join/#asterisk mrgoby (~mrgoby@64.208.211.80) |
17:28.59 | *** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc) |
17:28.59 | MikeJ[Laptop] | jbot, forget rtfm |
17:29.05 | *** join/#asterisk gtigene (~chatzilla@70.89.216.41) |
17:29.27 | DarthClue | ~rtfw |
17:29.27 | jbot | from memory, rtfw is http://www.voip-info.org, the only place to get any real answers about * and it's many, many uses. |
17:29.37 | MikeJ[Laptop] | jbot, no rtfm |
17:29.38 | jbot | somebody said no rtfm was Read The Fucking Manual (TM) |
17:29.52 | gtigene | My asterisk uses SIP phones on a local area network but sometimes it drops calls. Another time I could hear my caller but he could not hear me. I am using CVS Head. What could be the cause, or where should I look.. |
17:30.11 | PBXtech | i found it http://sf.net/projects/webaddressbook |
17:30.13 | Optic | sounds like it's not just a lan |
17:30.18 | Optic | i've never had sip fail me on a regular basic lan |
17:31.05 | gtigene | Optic, was that a response to my question? |
17:31.10 | Optic | yes |
17:31.19 | gtigene | Optic: thank you |
17:31.28 | HDG-laws|sl33p | Sounds like NAT. |
17:31.29 | Optic | are you sure you don't have any routers, or nat, or anything like that between your asterisk and your phones? |
17:31.39 | vaewyn | ~rtfm |
17:31.39 | jbot | from memory, rtfm is Read the Fine Manual (or in other words we are too busy for your question and the answer exists elsewhere) |
17:31.40 | vaewyn | :} |
17:31.53 | gtigene | Optic, there is only an Ethernet switch |
17:32.07 | Optic | gti: that is a very strange thing then |
17:32.16 | HDG-laws|sl33p | Only a switch? What is the switch connected to? |
17:32.16 | Optic | i can't help you because I have never had that problem :) |
17:32.27 | gtigene | Optic, OK |
17:32.32 | Assid | hrmm.. if an ip phone is on NAT but the asterisk server is on the net.. will the devices connect? |
17:33.10 | gtigene | HDG-laws|sl33p: Switch is connected directly to phones and directly to Asterisk host. |
17:33.58 | HDG-laws|sl33p | Where is the internet coming into play? |
17:34.00 | ender | is it still true that Polycom phones don't like type=friend ? |
17:34.13 | ManxPower | ender: that has NEVER been true |
17:34.39 | vaewyn | type=friend is just semi evil... and the people that first ran into that were using polycoms... |
17:34.48 | ender | hrm, ok. |
17:34.51 | ender | wiki says it is. |
17:34.54 | vaewyn | bugs by association... not really a problem |
17:34.58 | gtigene | HDG: There is one phone, hardly ever used, that is not on the LAN. They are using it to test something called a GRE tunnel. It talks to us through a router. |
17:35.10 | ManxPower | good thing they are making peers and users going away soon and everyone will use friend |
17:35.33 | vaewyn | ManxPower: really? is there a thread on that? |
17:35.41 | ManxPower | vaewyn: I'm sure there is. 8-) |
17:35.46 | vaewyn | I hope they have fixed the friend issues :} |
17:36.04 | ManxPower | Fortunatly, type=friend isn't REALLY evil. It just appears that way to people that don't understand their =friends |
17:36.31 | *** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com) |
17:36.34 | vaewyn | there are things you can do (correctly) with the user/peer that friend barfs on still |
17:36.42 | ManxPower | type=friend assumes incoming and outgoing authentication username/secret are the SAME. This is usually true for phones, but not always true with ITSPs or Gateways |
17:36.44 | Assid | HDG-laws|sl33p: just curious if that question was for me |
17:36.59 | vaewyn | ManxPower: exactly |
17:36.59 | ManxPower | vaewyn: I think all that stuff is/will be fixed. |
17:37.37 | HDG-laws|sl33p | Assis No, I don't believe so |
17:37.38 | ManxPower | they SHOULD have had type=incoming|outgoing|bidirectional |
17:37.40 | HDG-laws|sl33p | Assid* |
17:37.47 | vaewyn | hehehe... now instead of seperate entries we'll have 'peersecret=blah usersecret=blah2' :P |
17:38.07 | Assid | i wanna link 2 * boxes!!! |
17:38.11 | Assid | dunno who with |
17:38.12 | Assid | hehe |
17:38.21 | Assid | hrmm |
17:38.27 | vaewyn | ManxPower: peer/user is much better for defining than 'incoming/outgoing' cause those terms are relative |
17:38.30 | Assid | what about voip gateways? |
17:39.31 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
17:39.39 | Assid | shido6 my man |
17:39.54 | vaewyn | Will be nice to not have to dupe some info in both entries though... friend is a good idea... just needs to be 'fixed' |
17:40.09 | bkw_ | shido6, yo |
17:40.12 | shido6 | ? |
17:40.14 | shido6 | bkw |
17:40.18 | bkw_ | you gonna still speak? |
17:40.20 | shido6 | you talked jeremy into using your sys |
17:40.22 | shido6 | yeah |
17:40.25 | shido6 | Fraud |
17:40.32 | Assid | fraud? |
17:40.39 | shido6 | CC fraud |
17:40.43 | Assid | who did it? |
17:40.45 | shido6 | Rates Fraud |
17:40.48 | bkw_ | can you go register and select the speaker option.. then I can get you into the head count? |
17:40.59 | shido6 | Assid, its what took live voip out in addition to other things |
17:41.05 | shido6 | ok |
17:41.11 | Assid | hrmm |
17:41.11 | bkw_ | no livevoip was setup to be a fraud from day one |
17:41.17 | bkw_ | this isn't the first time they have done this |
17:41.35 | bkw_ | shido6, I have found some interesting info |
17:41.42 | *** join/#asterisk point (1000@213.27.44.55) |
17:42.05 | tzanger | oh? |
17:43.13 | shido6 | greg@cluecon, bkw |
17:43.49 | *** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt) |
17:44.33 | darwin35 | and the day drags on |
17:46.12 | ManxPower | If my damn customer hurries up and gets to the remote site, I MIGHT be able to see a movie this afternoon |
17:46.52 | *** join/#asterisk cpm (~Chip@cpm.sustaining.supporter.pdpc) |
17:48.14 | *** join/#asterisk jfonsecausa (~jfonseca@12.42.141.34) |
17:48.21 | eKo1 | what movie? |
17:50.22 | Bhaal | Hey guys, with Asterisk how easy is it to setup an automated operator so people can choose the extension they want to call once asterisk has answered the call? and voice mail if the extension isnt answered? |
17:50.46 | Assid | Bhaal: pretty easy |
17:50.58 | Assid | comparatively |
17:51.05 | vaewyn | Bhaal: not too hard... hey... just out of curiosity... do I remember you from #fnr? |
17:51.19 | Bhaal | vaewyn: yes, you do |
17:51.33 | vaewyn | Bhaal: ;} cool... long time :} |
17:51.36 | Assid | CpuID u up? |
17:51.39 | Bhaal | Assid: So its reasonably well explained in the docs? |
17:51.45 | mishehu | bah. |
17:51.54 | Bhaal | vaewyn: hehe indeed... Hows things? |
17:51.58 | mishehu | Bhaal: good source of info is at http://www.voip-info.org |
17:52.09 | Assid | Bhaal: yeah.. you can find some info on t |
17:52.10 | Bhaal | mishehu: Ahhh thanks |
17:52.40 | vaewyn | Bhaal: it's not spelled out... but everything is there... and once you get the hang of it... these are easy apps... :} and we are always here to help with problems |
17:52.49 | vaewyn | Bhaal: Things are good :P |
17:52.53 | *** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt) |
17:53.29 | Assid | Bhaal: just record your greetings and stuff and keep them ready |
17:53.32 | *** part/#asterisk point (1000@213.27.44.55) |
17:53.38 | Assid | once you start working on it.. things move rather fast |
17:54.03 | SpaceBass | Assid you ever get up and running? |
17:54.13 | Bhaal | sweet... I saw a festival conf file, how easy is it to setup with that? |
17:54.28 | Assid | SpaceBass? |
17:54.30 | vaewyn | Bhaal: You know... the #fnr guys seem to get into asterisk a lot :} I've seen SynRG and Mysticone and now you :P |
17:54.41 | Bhaal | haha |
17:54.43 | SpaceBass | festival works fine, but it sounds like crap... think 1982 robot voice |
17:54.45 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
17:55.09 | vaewyn | Bhaal: festival is a bitfun... but have someone with a good voice just record it... way easier and cheaper on the CPU... and sounds better :P |
17:56.00 | *** join/#asterisk darylp (~daryl_jus@63-208-162-62.digitalrealm.net) |
17:56.08 | Assid | SpaceBass: acutally.. i got it to "work" but 2/4 times.. it doesnt really do what i want and hangs up my zap |
17:56.09 | Assid | brb |
17:56.12 | Assid | phone call |
17:56.16 | Bhaal | Ive bought a sip <-> regular fone adapter .. and have a sip provider (no incoming number yet coz they havent got any allocated) .. Ive got asterisk working as the proxy etc ... just waiting for the incoming number and I will set the rest up... |
17:56.27 | *** join/#asterisk tris (~tristan@camel.ethereal.net) |
17:56.57 | vaewyn | wow... #fnr is gone... 1 person... :{ |
17:57.24 | Bhaal | vaewyn: Its #freematrix or something like that now |
17:57.25 | SpaceBass | what was #fnr? |
17:57.35 | Bhaal | SpaceBass: Freenode Radio |
17:57.39 | SpaceBass | ahhh |
17:57.42 | vaewyn | SpaceBass: freenode radio back in the day :P |
17:57.44 | vaewyn | it rocked |
17:58.02 | vaewyn | or in otherwords... geeks with mics :P |
17:58.08 | Bhaal | hahaha indeed |
17:58.43 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
17:58.44 | *** part/#asterisk Optic (dfraser@H31.C18.B96.tor.eicat.ca) |
17:58.51 | MattH | Hi.. does anyone know what's up (or down) with digium's 877 number? |
17:58.53 | MattH | I get a fast busy signal |
17:58.56 | Bhaal | vaewyn: I have a radio show on Pulseradio (www.pulseradiogroup.com) its called Preparation Zero ... We have a different DJ each week etc... |
17:59.13 | Bhaal | vaewyn: Not restricted to free music or anything, its all dance related music |
17:59.21 | vaewyn | Bhaal: cool... I'll check it out |
17:59.50 | vaewyn | Bhaal: monkey still glowing? ;P |
18:00.04 | Bhaal | That wasnt my show hahaha, that was Barbicane |
18:00.11 | vaewyn | ohh that's right |
18:00.27 | Bhaal | I cant even remember what my show on FNR was called... |
18:00.32 | vaewyn | getting my nicks messed up |
18:00.45 | Bhaal | hehe |
18:01.07 | Bhaal | Anyway Im off to bed, going to get a 3-4hrs sleep before heading into work... |
18:01.18 | vaewyn | Bhaal: have fun... nice seeing you again! |
18:01.20 | HDG-laws|sl33p | vaewyn So Polycom 600 over Snom 360? |
18:01.26 | bjohnson | woo hooo. Intellicad is released for linux workstations |
18:01.33 | Bhaal | yup, I will be hanging around for a while.. chat to ya later |
18:01.55 | vaewyn | HDG-laws|sl33p: For our application it made sense... especially with the microbrowser to play with |
18:02.41 | HDG-laws|sl33p | vaewyn This is for an insurance agency, I just hope I'm not stepping into quicksand doing this instead of having a PBX installed. |
18:02.46 | HDG-laws|sl33p | ;) |
18:03.30 | greg_work | HDG-laws|sl33p: just watch those voip providers.. so many sketchy ones |
18:03.43 | HDG-laws|sl33p | I was looking at Broadvoice I believe it was |
18:03.45 | HDG-laws|sl33p | They have BYOD |
18:03.57 | Lee__ | only good for wholesale though. |
18:04.32 | InfraRed | rnk |
18:04.39 | InfraRed | url ? |
18:04.40 | HDG-laws|sl33p | How much of a pain is it going to be to setup a POTS failover with the SIP (whichever phone I end up with) and * |
18:04.56 | greg_work | wholesale is better than those packaged "unlimited" deal things |
18:05.05 | vaewyn | HDG-laws|sl33p: Broadvoice isn't bad... I recommend Nufone primarily though cause I have had less connection issues to them (caveat: I have a short path to them so... it may just be that) |
18:05.07 | InfraRed | and where are they based? |
18:05.23 | greg_work | ie, you usually just pay a flat rate of somwhere between 1 and 2 cents/min .. if you don't use it, you don't pay anything |
18:05.44 | HDG-laws|sl33p | Would be nice if the IP Dialtone providers mentioned their connectivity so you would know if you had local peering or not |
18:06.17 | greg_work | if something happens and you get high latency to one provider, you switch to another. they also allow multiple concurrent calls, while those broadvoice-type services often don't |
18:06.41 | wunderkin | well it looks like my audio problem was caused by my provider.. or their isp.. suckage.. :/ |
18:07.17 | greg_work | you're also not paying extra to get "features" like call waiting, caller id, voice mail etc.. |
18:07.28 | greg_work | if you have *, it can do all that stuff. all you need is a raw pipe coming in |
18:08.05 | Lee__ | rnk is $0.007/minute but you have to deposit $5000 up front. |
18:08.26 | InfraRed | 5k |
18:08.27 | InfraRed | ouch |
18:08.37 | Lee__ | HDG-laws|sl33p: there is no dialtone with IP phones. It's a recording in the phone. |
18:08.38 | jhiver | what's 'rnk'? |
18:08.48 | ender | is there a page out there for using the web config of these Polycom phones for * use? |
18:09.20 | mishehu | ender: it's pretty straight forward, as long as you're using the sip firmware... |
18:09.28 | greg_work | i use sixtel for some long distance.. i think i had to put down $10 and get 1.5c/min or so |
18:09.29 | Lee__ | RNK will let you test their system for a whole day if you call them. it worked well here and had < 20ms latency. |
18:10.26 | ender | mishehu: you'd think so, however there are some things which seem very ambiguous. |
18:10.30 | greg_work | i have a DID with another provider for $5 cdn /mo (and that's only because the number is not in a big city, where it would be $2.50/mo), and i pay 1.1c/min within their calling area |
18:10.34 | mishehu | ender: such as? |
18:10.40 | ender | mishehu: like sip configuration vs registration |
18:11.02 | ender | mishehu: and Identification with a server address box as well as Server 1 w/ an address box |
18:11.30 | greg_work | to cmopare to broadvoice, i'd have to use over 2700 minutes of LD just to make it break even (more, and broadvoice is cheaper) |
18:11.31 | ender | mishehu: and 'outbound proxy' with a server box and 'Server 1' with a server box under sip config |
18:11.59 | greg_work | of course, read the fine print with those 'unlimited' services ... they often consider "unlimited" usage to be ie, 3000 mins.. |
18:12.20 | mishehu | ender: if something is repeated, it is because you can set different line appearances to use a different config, but there is also a "central" config that will be used as default. |
18:12.23 | greg_work | and that's not even taking into account that they may not even support >1 concurrent call |
18:12.42 | vaewyn | umm... 5000$/.007 is 714285 minutes... or 496 days... egads that's a lot to prepay |
18:13.20 | greg_work | yeah, that seems more like actual wholesale targetted at a LD provider or something |
18:13.28 | vaewyn | *nods* |
18:13.32 | greg_work | whereas the "wholesale" service I get is more like ... no-frills voip |
18:13.33 | ender | mishehu: yeah, I'm having a hard time figuring out what is the 'central' one, I haven't got my phone to register correctly yet. |
18:13.38 | vaewyn | would work great for here at the univeristy :} |
18:13.58 | ender | mishehu: is 'outbound proxy' the 'core' SIP server I should use (IE * ) |
18:14.19 | *** join/#asterisk fugitivo (~ajf@168.226.246.44) |
18:14.19 | fugitivo | hello |
18:14.26 | jhiver | hi |
18:14.54 | ender | mishehu: also there are a lot of radio buttons that have no label for each option. |
18:14.59 | jhiver | I agree with greg_work. 'Pay as you go' is as good as it gets with VoIP |
18:15.15 | ender | mishehu: there is a Type: line w/ a radio button choice of 2, no label on what each of those two are, so how the heck do I know what I'm selecting? |
18:15.39 | ManxPower | ender: that's a well known bug in the polycoms. No idea how to fix it, since I use config files for them. |
18:16.30 | The_LightSide | does anyone know where i can get a DSS console for * ? |
18:16.35 | ender | ManxPower: ah ok. |
18:16.38 | ender | ManxPower: I foudn it in the source. |
18:16.42 | greg_work | jhiver: well, there's just effectively two different classes of voip providers... the end-user targetting ones, and the *-user targetting ones.. services targetted at end-users who want to plug a sip phone into their router are just not the best fit with asterisk |
18:16.45 | mishehu | ender: and I don't have a phone here that I can view at the moment... |
18:16.49 | mishehu | so I can't help really. |
18:16.55 | mishehu | got my hands full at teh moment. |
18:17.04 | jhiver | yep |
18:17.40 | jhiver | altough if there was a kick ass IAX ATA this could change as you could start a end-user IAX VoIP provider service |
18:18.15 | tclark | jhava: pa168V |
18:18.17 | jhiver | but I think it would mean 'forget about unlimited plans' which isn't a bad thing anyways |
18:18.20 | [TK]D-Fender | Hey, anyone know where I can get documentation on Polycom's Micro-Browser XML format? Or at least some samples for em to play with? |
18:18.45 | Lee__ | [TK]D-Fender: IIRC it's valid XHTML |
18:18.46 | HDG-laws|sl33p | greg_work So you are saying using * for SIP isn't a good fit? |
18:19.02 | greg_work | HDG-laws|sl33p: no, not at all |
18:19.07 | jhiver | I guess the good thing about 'unlimited' plans when you're a provider is that it's pretty easy to bill |
18:19.36 | jhiver | but then you have to keep an eye on your expenses anyways |
18:19.38 | jhiver | sooo.... |
18:19.53 | greg_work | HDG-laws|sl33p: but there's no point in getting a service that provides you with call waiting , voicemail, etc etc etc, when you're connecting it to *, which can provide voicemail, calll waiting, etc etc but on YOUR terms |
18:20.12 | The_LightSide | does anyone know where i can get a DSS console for asterisk@home?? |
18:20.14 | greg_work | ie-.. say you use a BV line for an incoming business line. if you use BV voicemail, your whole business gets one mailbox |
18:20.22 | wunderkin | well.. the only reason you would want your provider to have voicemail is if you arent registered |
18:20.34 | HDG-laws|sl33p | greg_work: Hmm, the more and more I read the more and more nervous I get ;O |
18:20.43 | greg_work | if you use *, instead of using BV you can have an autoattendant, and have individual mailboxes for every person |
18:21.24 | greg_work | wunderkin: true. but if not being registered is an issue (ie, you don't have a dedicated internet connection of some sort that has an SLA) then you shouldn't be relying on it for your primary phone line |
18:21.43 | jhiver | HDG-laws|sl33p: greg_work is telling you the right thing: you should probably worry about finding a solid, basic VoIP connection and leave all the fancy 'added value services' to asterisk |
18:21.53 | jhiver | so that you keep the 'added value' for yourself ;) |
18:22.08 | HDG-laws|sl33p | I'm not concerned about "value added services", I'm more concerned with providing my boss a "maximum cost per month" |
18:22.16 | HDG-laws|sl33p | Going on a per usage leaves that "open-ended" |
18:22.45 | jhiver | garbage |
18:22.57 | jhiver | just look at your phone invoices for the last X month |
18:22.59 | greg_work | HDG-laws|sl33p: i work at a small business, and here's what we do: we have 3 incoming POTS lines on a hunt group, that go into *. I also have a DID number that comes in over VoIP.. that DID is also in the hunt group (well, it will be shortly, i'm waiting for them to change it now) |
18:23.40 | [TK]D-Fender | Lee__ : I didn't really give it a full XHTML header so I guess that could be it. Do you have a sample file by any chance? |
18:24.05 | greg_work | HDG-laws|sl33p: outgoing LOCAL calls are made using two of our POTS lines (leaving one free for incoming), or if they're both busy, a VoIP provider. outgoing LD calls are done using a Voip provider first, and if that's busy, or our internet is down, it fails over to POTS lines |
18:24.19 | *** join/#asterisk coldfeet (~cold@dsl-80-46-109-145.access.as9105.com) |
18:24.22 | HDG-laws|sl33p | jhiver: Charges are never the same, we have annual renewals so one month you will be talking to X client and X client's insurance company in X state and some months you won't. I could look at the most ever used and use that for a basis, but it is easier to tell them "look, $30/mo per line" |
18:24.27 | Lee__ | [TK]D-Fender: nope, but I used google's XHTML page and it loaded. |
18:24.44 | coldfeet | hi guys, anyone here using MYSQL in extensions, I have the following query, but it fails, cant see why |
18:24.46 | Lee__ | hee hee. tiny http://www.google.com/xhtml |
18:24.47 | coldfeet | select\ a.reseller_id\ from\ subscriber\ s\,\ subscriber_add\ a\ where\ s.phplib_id\ =\ a.phplib_id\ and\ s.username\ =\ \'040600\' |
18:24.50 | HDG-laws|sl33p | greg_work you sound like the guy I need to speak with, you are doing what I want to do |
18:24.53 | Lee__ | ah!!!! |
18:24.54 | greg_work | effectively, it means if our internet connection goes down, our call capacity just decreases to 3 conncurrent calls (on POTS).. we're not crippled, but people may get congested tones when trying to place calls |
18:24.59 | HDG-laws|sl33p | 6 CO lines with POTS failover |
18:25.19 | jhiver | Yeah but I don't see any business right in their minds doing 'all you can eat' for businesses |
18:25.34 | jhiver | although some of them do 'softcaps', I remember having seen one doing that |
18:25.42 | greg_work | imho, VoIP over the internet is just not reliable enough to depend on, ESPESSIALLY if you don't have a dedicated T1 or Fiber or better connection, with a tight SLA |
18:26.00 | jhiver | they would look at the last average 3 month consumption and as long as it's below a certain level not overcharge |
18:26.06 | Lee__ | yeah, get an SLA if your client can sue you. |
18:26.26 | *** join/#asterisk WeezeyD (~ohno@206.210.111.115) |
18:26.30 | greg_work | Lee__: well, i'm thinking more for customers.. if you can't make/get phone calls, it's pretty crippling to a business |
18:26.42 | HDG-laws|sl33p | We aren't at capacity of requiring a T1, but we are given a five 9's SLA from Comcast I believe. |
18:26.54 | HDG-laws|sl33p | But I wanted to have at least a 3 line POTS failover |
18:27.00 | HDG-laws|sl33p | In the event of internet downtime |
18:27.08 | HDG-laws|sl33p | So we could AT LEAST answer calls. |
18:27.10 | *** join/#asterisk toot (~chris@212.20.250.186) |
18:27.15 | greg_work | VoIP for inter-office communications (between branch offices) is probably ok.. just make sure there's a backup way to communicate |
18:27.42 | greg_work | HDG-laws|sl33p: how many concurrent calls are you planning for? |
18:27.46 | toot | hey anyone using festival ? is the quality good or any other recommendaions? |
18:27.53 | WeezeyD | I set up TDMoE, but callerid isn't being passed, does caller ID not work with TDMoE? |
18:27.54 | [TK]D-Fender | Lee__ : I just saved Google's page over the one I was testing with and get another error |
18:28.02 | HDG-laws|sl33p | 3-4 concurrent, we've maxed out the 6 lines quite often |
18:28.06 | [TK]D-Fender | Lee__ : Could you pastebin your sample |
18:28.16 | Lee__ | that /is/ my sample. |
18:28.24 | Lee__ | Ploycom 600 |
18:28.34 | *** join/#asterisk Corydon76-home (three@Corydon76-home.sustaining.supporter.pdpc) |
18:28.38 | [TK]D-Fender | eek... I did google.ca, I'll try google.com |
18:28.58 | jhiver | hahaha, that rnkvoip 'phone for life' offer sounds just so bogus |
18:29.00 | *** part/#asterisk vaewyn (freeman@mail.parrishmachine.com) |
18:29.07 | greg_work | toot: i was mildly satisfied by one of the voices from festvox.org .. also tried cepstral, thought it sounded worse than the default festival voice |
18:29.19 | jhiver | 'give us loads of money upfont, and you'll have phone for life until we're bust' |
18:29.37 | greg_work | HDG-laws|sl33p: in that case, i'd say use your POTS as your primary lines, and use VoIP for extra capacity |
18:29.38 | mrgoby | toot: festival is the best tts that is open source, imo... quality is okay... i had best results with mbrola voices |
18:29.40 | *** join/#asterisk salvini_fs (~felipesal@200164050004.user.veloxzone.com.br) |
18:30.01 | [TK]D-Fender | Lee__ : Dammit, all failures |
18:30.02 | mrgoby | the festival app though is unreliable in asterisk |
18:30.09 | mrgoby | best to use AGI workarounds |
18:30.16 | HDG-laws|sl33p | Pots for both in/out or just for in? |
18:30.26 | greg_work | mrgoby: tried festvox.org? i use voice_cmu_us_bdl_arctic_hts .. didn't try or hear mbrola though |
18:30.31 | ender | hrm, I have a weird issue w/ my polycom. |
18:30.40 | ender | We are using 4 digit extensions here, 100X for now. |
18:30.47 | greg_work | toot: and I second the AGI thing.. i have a php script that does TTS for me, and it caches |
18:30.53 | mrgoby | greg_work, not sure if i tried that... will check though |
18:31.05 | ender | on my polycom, when I press '10' it immediately tries to do some sort of call, it fails, then I get my prompt back half a second later and I can continue putting digits in. |
18:31.15 | ender | something is making the phone send something after 10 is hit. |
18:31.19 | mrgoby | the approache in php_agi is a good one |
18:31.34 | HDG-laws|sl33p | greg_work: I don't follow |
18:31.40 | mrgoby | in regards to festival caching |
18:31.47 | *** join/#asterisk vaewyn (freeman@mail.parrishmachine.com) |
18:32.07 | greg_work | HDG-laws|sl33p: for both .. but really, i've been fairly happy with my setup. it's a simle LCR (least cost routing) config for outbound |
18:32.24 | vaewyn | Bhaal: Southern Exposure was yours :} thank goodness for irc logs :P |
18:32.28 | greg_work | LD goes over VoIP (where it's cheaper permin) while local calls go over POTS first (since it's 0/min) |
18:33.09 | greg_work | but if all POTS are used up, it uses VoIP, if VoIP is down (for LD) it uses POTS, and if all POTS are used up and voip is down, they just get a busy signal |
18:33.13 | HDG-laws|sl33p | greg_work How do I work the SIP in with the POTS, I've gotten confused ;o |
18:34.20 | greg_work | HDG-laws|sl33p: well if you use a config like AMPortal, then there's a whole section for configuring how calls are routed and over which trunks (disclaimer: i wrote the outbound routing stuff in amp ;) ) |
18:34.26 | *** join/#asterisk loick (~loick@ATuileries-151-1-19-219.w82-123.abo.wanadoo.fr) |
18:35.04 | HDG-laws|sl33p | greg_work But there would be no issues having all phones in the new office be SIP ? |
18:35.19 | greg_work | But basically you make a dial macro .. like [macro-dial] 1,Dial(Zap/g0/${ARG1}) 102,Dial(SIP/provider/${ARG1}) |
18:35.38 | HDG-laws|sl33p | Just a second, let me get out my Greek dictionary |
18:35.41 | greg_work | .. if the first dial fails it moves to the next one .. |
18:35.42 | greg_work | heh |
18:35.45 | *** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net) |
18:35.49 | milkyflava | hello |
18:35.50 | HDG-laws|sl33p | I understand the concept of the failover |
18:36.08 | milkyflava | Can someone recommend a good softphone that uses IAX? |
18:36.10 | HDG-laws|sl33p | I'm just afraid I don't have the entire hardware list or know exaclty what I need to make this come together |
18:36.13 | greg_work | AMP (a web gui / config for *) does all that stuff.. but if you really want to do it by hand then you need to read up on dialplan stuff |
18:36.19 | Lee__ | milkyflava: no |
18:36.23 | milkyflava | lol |
18:36.41 | greg_work | HDG-laws|sl33p: well, as far as in the office, it depends on what you want to do, and how your network is configured |
18:37.22 | milkyflava | Thanks Lee__, I'll get SIP through NAT then. |
18:37.41 | The_LightSide | possibly iaxcomm... very basix though |
18:38.05 | greg_work | HDG-laws|sl33p: here, we use all SIP phones. I have at least two cat5's running to every desk, and the voip phones are on a physically different switch than the computers |
18:38.07 | vaewyn | iaxcomm works well |
18:38.19 | HDG-laws|sl33p | So you don't use the internal switching |
18:38.35 | greg_work | (though they are connected .. i don't really know if theres a huge benefit to that or not.. but at least the majority of the traffic is seperated.. i've never had an issue with not enough bandwith on the local network) |
18:38.50 | greg_work | no, my phones don't even have switches |
18:38.55 | shido6 | poe switches on ups? |
18:39.09 | HDG-laws|sl33p | Ah, the phones I'm looking at have internal switches to connect PC or networkable printer to |
18:39.16 | HDG-laws|sl33p | So I can avoid extra runs of cat6 |
18:39.21 | HDG-laws|sl33p | or 5e |
18:39.23 | Katty | mew |
18:39.29 | greg_work | we just moved in here 7 months ago, so i had the benefit of wiring it. did it all with multiple cat5e's |
18:39.29 | JerJer | mooo |
18:39.32 | *** join/#asterisk inspired (mikael@213.197.167.61) |
18:39.41 | Katty | suddenly, i'm all inspired. |
18:39.45 | greg_work | when you wire a building, cat5e becomes dirt cheap |
18:39.58 | HDG-laws|sl33p | I can wire it all up, I just don't know what I'm doing as per running the wiring thru the ceilings and down the walls ;o |
18:39.59 | greg_work | probably cheaper than the extra money for voip phones that have switches in them ;) |
18:40.08 | shido6 | Lord |
18:40.17 | HDG-laws|sl33p | The voip phones that have switches in them also seem to be the same phone with 6 lines ;) |
18:40.20 | vaewyn | heck yeah... 'good' cat5e is 70$/1000ft |
18:40.24 | shido6 | I did Detroit Public Schools , wiring buildings sucks |
18:40.25 | vaewyn | dirt cheap |
18:40.31 | shido6 | esp OLD buildings |
18:40.32 | robl^ | cat5e!?!? what?!?! no one still uses thinnet 10Base2 RG59 coax for networking? |
18:40.37 | HDG-laws|sl33p | shido6: Where are you? I am in Hamtramck! |
18:40.46 | shido6 | Im in Windsor right now |
18:40.47 | Lee__ | HDG-laws|sl33p: Snom and Cisco phones with switches are good. |
18:40.48 | vaewyn | ~lart robl^ |
18:40.54 | greg_work | shido6: it's ready for PoE, but i didn't use it because i have spa-841's and they removed the poe feature (grr) |
18:40.57 | shido6 | HDG-laws|sl33p, NuFone is a michigan based company you know? |
18:41.03 | The_LightSide | robl^: dont u mean RG58? |
18:41.07 | HDG-laws|sl33p | Lee___ I was prepared to buy Snom 360's today and now I'm backing off that and looking to Poly 600's |
18:41.10 | mutilator | isn't everything |
18:41.15 | HDG-laws|sl33p | shido6 I did not know that, interesting |
18:41.16 | greg_work | I do have an ATA in the server room on UPS though, that powers two analog phones (one cordless) |
18:41.29 | The_LightSide | my bad! |
18:41.29 | milkyflava | DarthClue, You awake? |
18:41.32 | robl^ | The_LighterSide, whats a few RG's between geeks? :) |
18:41.37 | The_LightSide | lol |
18:42.18 | greg_work | isn't rg59 the really cheap crappy coax cable they used to use for tv? |
18:42.24 | greg_work | sadly people probably HAVE used that for thinnet |
18:42.25 | SpaceBass | pretty much |
18:42.36 | The_LightSide | cheers for now! catch u all later! |
18:42.36 | SpaceBass | rg6 is the only way to go for digital cable or satelite |
18:42.55 | SpaceBass | hey... 14mb tolken ring was much better than 10baseT |
18:42.58 | vaewyn | *nods* rg6 or rg11 on really long runs |
18:43.13 | greg_work | HDG-laws|sl33p: so are you wiring from scratch then? |
18:43.23 | SpaceBass | but then again, I really liked working on as400s too :) |
18:43.24 | robl^ | Tolkien Ring? the ONE RING ethernet :) |
18:43.28 | HDG-laws|sl33p | greg_work THere is wiring in place that is going BYE BYE. |
18:43.38 | HDG-laws|sl33p | OLD cat5 |
18:43.56 | SpaceBass | whats the typical rate for hireing someone for structured wireing these days? |
18:44.20 | jhiver | hey, what's asterisk packetization time when using SIP? |
18:44.28 | greg_work | hm. not a clue.. it's really not that hard though, probably not worth the money :p |
18:44.33 | greg_work | i'd imagine $50-70/hr |
18:44.42 | greg_work | that's a lot to pay somenoe to pull cable |
18:44.48 | SpaceBass | yeah it is |
18:44.53 | HDG-laws|sl33p | I have someone willing to run the cable for $85/run |
18:44.58 | greg_work | hell, even pull the cable yourself and then get someone to come in and terminate it all, if you don't know how to do that |
18:45.01 | Lee__ | get an intern. |
18:45.15 | SpaceBass | thinking about homes really... if you don't do it yourself its hard to find someone to wire cat5 in a house |
18:45.25 | jhiver | ok I have my answer: 20ms |
18:45.41 | greg_work | but really.. invest in a decent crimp tool ($50+), a bag of rj45's (you'll use 3-4 per cable when you first start) .. and its pretty simple |
18:46.00 | greg_work | HDG-laws|sl33p: i also highly recommend a patch panel |
18:46.13 | greg_work | HDG-laws|sl33p: this is mine: http://office.mwater.ca/files/office/patchfront.jpg |
18:46.25 | toot | will give it a good - thanks for the comments |
18:46.48 | SpaceBass | I really needed a patch pannel... I ran everything stright to the switch |
18:46.56 | HDG-laws|sl33p | greg_work Patch panel is a guarantee |
18:47.33 | HDG-laws|sl33p | greg_work I know how to do all that stuff, it is the running of the wires down walls that I'm "uncertain of" |
18:47.37 | greg_work | with RJ45, you can plug RJ11 in (ie, analog phone lines) .. since I have cat5e running everywhere, it's very simple to add an analog phone line (ie, for a fax/interac machine, or from my SIP ATA adapter to the analog cordless phone) anywhere |
18:47.48 | HDG-laws|sl33p | greg_work As well as exactly what hardware I need to do this cross between VoIP and POTS |
18:47.48 | SpaceBass | HDG-laws|sl33p what are you wiring? |
18:48.17 | mutilator | A BOMB! |
18:48.22 | SpaceBass | RUN! |
18:48.31 | tsume | they set us up the bomb |
18:48.35 | greg_work | HDG-laws|sl33p: drop ceiling? most places are constructed with the tops of walls open .. ie, there will be studs that go ALL the way to the top of the roof (usually 1-4' above the drop ceiling) |
18:48.40 | tsume | all your base are belong to us |
18:48.58 | greg_work | and the drywall will just go a few inches above the drop ceiling and stop, leaving an open gap .. easy to drop down wires |
18:49.04 | Lee__ | has someone compiled CVS HEAD today? |
18:49.14 | jhiver | Hey guys |
18:49.16 | SpaceBass | greg_work too bad houses aren't built that way! wiring b/t floors is a pain! |
18:49.18 | Lee__ | cause my update seems to be missing a directory. |
18:49.22 | greg_work | SpaceBass: yes :) |
18:49.29 | jhiver | Any ideas what UDP range is asterisk using when using SIP? |
18:49.51 | Lee__ | outtolunc: yes, it is missing a directory? |
18:49.51 | SpaceBass | jhiver for rtp its configured in rtp.conf |
18:50.20 | outtolunc | iirc there were a couple for gnophone and others |
18:50.26 | greg_work | HDG-laws|sl33p: to get POTS lines into your * box you need something with fxo ports on it .. like a TDM400 from digium (... avoid, imo) or an ATA ("analog telephony adapter" ... i think) |
18:50.42 | jhiver | Cheers |
18:50.54 | SpaceBass | and ATA is an fxs not an fxo :) |
18:50.57 | Lee__ | make: build_tools/mkdep: Command not found |
18:50.58 | SpaceBass | clear as mud? |
18:50.58 | HDG-laws|sl33p | greg_work So I'd either have to run a digium card out of the * or use ATA's? |
18:51.14 | greg_work | SpaceBass: is it? i didnt know that |
18:51.22 | *** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
18:51.27 | Lee__ | the Sipura has an FXS and FXO |
18:51.36 | greg_work | wellgate makes a 4-port fxo device that talks SIP |
18:51.37 | Lee__ | as to some big media gateways |
18:51.41 | greg_work | yeah, i think it is an ATA still |
18:51.44 | Lee__ | *do |
18:52.07 | Damin | Wiiwiiwii... |
18:52.09 | greg_work | HDG-laws|sl33p: i have a TDM400P with 4 fxo ports, and personally, i wouldn't buy one again |
18:52.18 | HDG-laws|sl33p | greg_work eek! |
18:52.21 | greg_work | it's crashed on me about 5 times now (in 6 months) |
18:52.21 | obsidian-studios | greetings, anyone got any suggestions for keeping CID/ANI info off a caller, that say calls, does not leave a message, and does not reach a person via an extension? Ideally a email report or something? |
18:52.56 | outtolunc | lee: didn't see that, and my 5000 line buffer only has from preprocess to the end |
18:53.01 | ManxPower | greg_work: Did you report the problem to Digium? |
18:53.07 | outtolunc | (i've been busy since then) |
18:53.15 | greg_work | ManxPower: no, because i wasn't running the lastest driver at the time |
18:53.32 | ManxPower | obsidian-studios: Perhaps if you reworded your question, you might get an answer. |
18:53.35 | SpaceBass | obsidian-studios you could use AMP and the reports it generates... export to PDF... im sure there is a way to automate it and use sendmail |
18:53.38 | obsidian-studios | greg_work: FYI, I have a TDM400P in a * deployment where a hurricane took out the old pbx system, and so far the TDM400 card has worked perfectly |
18:53.53 | obsidian-studios | SpaceBass: cool |
18:54.09 | greg_work | it just suddenly scrolls "TDM PCI master abort" over and over on the local console, and otherwise totally freezes. cold reboot is the only fix |
18:54.17 | *** join/#asterisk Skarmeth (~Skarmeth@201009024136.user.veloxzone.com.br) |
18:54.30 | obsidian-studios | I seem to have people calling in and hanging up, trying to get info on them, any suggestions? |
18:54.38 | obsidian-studios | ManxPower: is that worded better :) |
18:54.39 | greg_work | i think it's the drivers and not the hardware.. but the bottom line is i don't totally trust it |
18:54.44 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
18:54.52 | greg_work | obsidian-studios: caller id |
18:55.35 | obsidian-studios | greg_work: it's the caller id info I need in an accessible place, as I am talking about someone who calls, listens say to the main menu and hangs up |
18:55.54 | obsidian-studios | I guess I could keep a regular phone plugged in as well and look at it from time to time |
18:56.13 | obsidian-studios | was inquiring on a more elegant solution, but I do not really need all the info in a cdr |
18:56.14 | greg_work | obsidian-studios: oh. well, use CDR |
18:56.23 | SpaceBass | obsidian-studios I still suggest amp and its reports |
18:56.24 | Lee__ | I guess I just needed to checkout the CVS again. It got the missing directory. |
18:56.29 | obsidian-studios | greg_work: thought a cdr would have to much info |
18:56.43 | obsidian-studios | SpaceBass: cool, I will go research and check it out now |
18:56.47 | greg_work | obsidian-studios: or put Noop(caller id is ${CALLERIDNUM}) in the dialplan .. it will get logged (if you have logging on) |
18:56.59 | greg_work | and you can grep "caller id is" /var/log/asterisk/full |
18:57.07 | obsidian-studios | greg_work: yeah I know I can log it, but then I have to look at logs etc |
18:57.26 | obsidian-studios | would be totally cool if I knew someone was in the system before they enter my extension or etc |
18:57.28 | *** join/#asterisk pa (~Paolo@pa.user) |
18:57.32 | pa | Hello! |
18:57.33 | pa | wow |
18:57.38 | greg_work | crontab -e 59 23 * * * grep "caller id is" /var/log/asterisk/full |
18:57.41 | pa | an IRC channel for Asterisk! |
18:57.49 | greg_work | then you'll get an email every night (assuming you rotate that file nightly) |
18:57.51 | obsidian-studios | greg_work: :) nice |
18:58.02 | obsidian-studios | greg_work: any ideas for real time |
18:58.21 | greg_work | use FOP |
18:59.00 | greg_work | or use CDR and write a two-second web app that does SELECT callerid, date FROM cdr ORDER BY date DESC LIMIT 10 |
18:59.07 | pa | I have a simple question (then i go to RTFM): Can i call someone by phone with Asterisk not being at home on the machine with ISDN card, but from remote over TCP? |
19:00.02 | *** join/#asterisk meshuga (meshuga@c-24-20-154-158.hsd1.wa.comcast.net) |
19:00.03 | pa | that is: ME -- TCP --> HOME + Asterisk + TA ISDN -- PHONE NETWORK --> Called |
19:00.35 | greg_work | pa: yes, you just have to use a SIP or IAX phone and open the approate ports on the firewall to let it connect to asterisk |
19:01.04 | meshuga | so, i recently switched from ztdummy to a digium wildcard as a timing source, now i cant hear anything in VM. is there configuration changes I need to make to tell asterisk to use the new timing source? |
19:01.06 | obsidian-studios | greg_work: why do you not like the TDM400 cards? |
19:01.19 | meshuga | the console acts like its playing |
19:01.24 | meshuga | but you cant hear anything thru my sip clients |
19:01.33 | pa | greg_work: i think they are VoIP phones.. there are linux applications that act like a such voip phone using microphone and speakers or earphones? |
19:01.33 | Skarmeth | hi all, I've a simple ask about Asterisk and X100P cards... I'm searching for it about two days without sucess... I've a extension from my analog PBX connected to it and I want to make calls throung it using my SIP/AIX2 soft phones. |
19:01.58 | Skarmeth | How I can allow the soft phone dial the destination number? |
19:02.04 | SpaceBass | pa tons... iaxcomm and i think one for gnome are peopular |
19:02.05 | meshuga | pa: iaxcomm woulda |
19:02.06 | greg_work | obsidian-studios: getting a call on my cell phone early in the morning "greg, the phones are not working! what do we do!!" and finding out that it's scrolling "TDM PCI master abort" on the console, not responding on the network, and requires a reboot |
19:02.07 | meshuga | works |
19:02.13 | meshuga | gnomephone doesnt work at all |
19:02.24 | greg_work | those calls - espessially when they wake me up - put me in a bad mood |
19:02.25 | pa | thanks a lot! going to RTFM :-) |
19:02.29 | SpaceBass | Skarmeth did you set up the cards on the box using the ztcfg tool? |
19:02.35 | Skarmeth | using a fixed extension, I can complete the call, but not allowing the shoft phone tell it |
19:02.36 | pa | but first add #Asterisk to list :-) |
19:02.39 | greg_work | and i've had about 3 of them since i started with *. not impressed at all |
19:02.41 | obsidian-studios | greg_work: what kernel and etc? I see really good uptime and etc with the TDM400 at a clients, with allot of abuse? |
19:02.54 | obsidian-studios | greg_work: wow |
19:02.54 | greg_work | 2.6 kernel, * stable |
19:03.02 | greg_work | 2.6.8 to be exact |
19:03.04 | obsidian-studios | greg_work: distro? |
19:03.12 | greg_work | debian sarge |
19:03.19 | obsidian-studios | hmm, might be env? |
19:03.33 | obsidian-studios | I run Gentoo, and bet I am using newer version of everything than you? |
19:03.34 | meshuga | greg_work: i havent had good luck with tdm400s either |
19:03.38 | Skarmeth | SpaceBass, it's working, I can get call coming from it and transfer to SIP/IAX2 clients |
19:03.45 | meshuga | greg_work : sangomas work better, IMO |
19:03.47 | Lee__ | newer != better |
19:03.59 | obsidian-studios | greg_work: running 2.6.11 or 2.6.12 kernels ATM |
19:04.00 | meshuga | but i'd just use a media gateway really |
19:04.19 | meshuga | once i get to multiplexing in 4 t1s, i don't like trusting something that if it fails the whole pbx goes down to change. |
19:04.23 | greg_work | obsidian-studios: my ztcfg binary is dated 2005-06-09 |
19:04.26 | obsidian-studios | greg_work: could be what you are seeing is weirdness, incompatibility, or instability due to other things? |
19:04.32 | Skarmeth | my /etc/zaptel.conf looks like: fxsks=1 newline defaultzone=us newline loadzone=us |
19:04.41 | mutilator | it possible to start recording calls from the cli? |
19:04.50 | meshuga | mutilator : no |
19:04.56 | greg_work | i WAS using a slightly older (december 2004) one until then though |
19:05.08 | greg_work | i'm actually not sure if it has crashed since then |
19:05.09 | obsidian-studios | greg_work: yes, but the gcc it's using, glibc, kernel calls, kernel headers, etc might be older? So it's more what your current stuff is compiled against |
19:05.12 | mutilator | just manager interface & extensions then |
19:05.22 | meshuga | exactly |
19:05.30 | obsidian-studios | greg_work: only time I have seen * stop working or fail, is when I have done something |
19:05.32 | greg_work | at the same time though, i had to resort to using a cron job to unload and reload the wctdm driver every night |
19:05.36 | Nuxi | manager scripts can be called from the cli. |
19:05.38 | meshuga | well, or you can ethereal and pick up the RTP stream ;) |
19:05.47 | greg_work | obsidian-studios: the kernel header is upto date |
19:05.56 | greg_work | gcc 3.3.5-1 |
19:06.04 | obsidian-studios | greg_work: something is not right, but I am not sure i would blame * or the drivers, or hardware without looking at all vars |
19:06.18 | meshuga | those tdm400p's are very static prone. |
19:06.19 | *** join/#asterisk meppl (mephisto@p54AABE80.dip.t-dialin.net) |
19:06.25 | greg_work | ii libc6 2.3.2.ds1-20 GNU C Library: Shared libraries and Timezone data |
19:06.26 | meshuga | i've rma'd 2 or 3 |
19:06.40 | meshuga | i just stick with a maximum of 2pt sangomas these days |
19:06.47 | meshuga | and media gateways for anything higher |
19:06.50 | meshuga | uh |
19:06.57 | meshuga | Nuxi: none of that stuff has 'cvs heads' |
19:07.15 | Skarmeth | SpaceBass, can you send me a example extension to make calls throung X100P? |
19:07.16 | Nuxi | oh, well, the latest and greatest of each. |
19:07.18 | meshuga | and plus, that'd be not smart to use development gcc and glib and whatnot. |
19:07.25 | meshuga | Nuxi: incorrect sir |
19:07.26 | greg_work | obsidian-studios: well, the actual system it's running on is the same as a couple other servers i have |
19:07.28 | SpaceBass | Skarmeth think there is one on the wiki |
19:07.31 | SpaceBass | ~wiki |
19:07.35 | obsidian-studios | well I am glibc 2.3.4.20041102-r1 and gcc 3.4.2-r2 on the * box |
19:07.51 | meshuga | Skarmeth : there is, look for 'development kit lite' instructions, it used sample configs for the x100p |
19:07.58 | Nuxi | wow, thanks meshuga, I've been knighted. |
19:08.02 | greg_work | well, i dunno |
19:08.03 | obsidian-studios | greg_work: I would start looking to the * developers and the env they use |
19:08.22 | greg_work | i would rather just spend an extra $200 or whatever to get a media gateway |
19:08.23 | *** join/#asterisk MRH2 (~Mr_happy@fcirc-adsl.demon.co.uk) |
19:08.28 | meshuga | Nuxi: the libc5 and glibc2 issues come to mind, plus of course gcc 2.95 and gcc 3.* as well. |
19:08.40 | meshuga | greg_work : not even that much more these days |
19:08.59 | obsidian-studios | greg_work: are you pulling * from CVS? Or using Debian binaries? |
19:09.06 | greg_work | cvs |
19:09.17 | obsidian-studios | greg_work: when I left RH I attempted to go Debian but had way to many problem on stable |
19:09.18 | meshuga | i've been using octtel media gateways |
19:09.20 | meshuga | to decent sucess |
19:09.29 | meshuga | debian stable == 3 years old |
19:09.33 | obsidian-studios | greg_work: kernel stuff on machines that ran the same kernel config and etc under RH for years |
19:09.38 | greg_work | meshuga: well, not anymore ;) |
19:09.39 | obsidian-studios | greg_work: horrible |
19:09.43 | meshuga | true, as of last week? |
19:09.44 | greg_work | i've only ever used sarge though |
19:09.50 | greg_work | meshuga: nah, 3 or 4 weeks now |
19:09.55 | MRH2 | anyone experince backgound noise fading in and out (when someone speaks) when you call noisy environments |
19:09.55 | meshuga | ah, i was off abit |
19:09.57 | obsidian-studios | greg_work: you are working from current cvs I bet coming off developers machines using much more current stuff than you |
19:10.03 | meshuga | i'm pretty done with debian |
19:10.22 | brookshire | asterisk compiles just fine on debian |
19:10.23 | obsidian-studios | greg_work: to pull from cvs, you need to make sure you are compiling against what the developers are, or you will get unexpected, unreliable, unstable results |
19:10.26 | mishehu | MRH2: that probably has to do with your phone or software. |
19:10.26 | brookshire | did you use the wiki? |
19:10.41 | greg_work | meshuga: its just a mind thing. debian stable (with woody, anyway) really was STABLE.. but it got old |
19:10.48 | meshuga | brookshire : we didn't say it didnt, hes having problems with tdm400p and debian. |
19:10.53 | greg_work | great for servers, where you get it working and don't want to screw with it (besides security updates) |
19:10.55 | obsidian-studios | brookshire: the problem is greg_work: is seeing problems with * and unreliability I have not see |
19:11.02 | meshuga | greg_work: slack is still better ;) |
19:11.03 | greg_work | but effecively, you can use testing and it's quite safe |
19:11.11 | brookshire | meshuga: did you call digium support? or email them? |
19:11.18 | greg_work | obsidian-studios: it's not *.. it's zaptel drivers |
19:11.25 | MRH2 | I have tried different combos with the same result though |
19:11.26 | meshuga | brookshire : its not my problem. why would i do that? |
19:11.28 | *** join/#asterisk djin_ib (~djin_ib@gridfox.xs4all.nl) |
19:11.30 | obsidian-studios | greg_work: in my case it was all servers I had problems with Debian, BIND servers would kernel panic |
19:11.43 | obsidian-studios | greg_work: it's all related |
19:11.48 | meshuga | hahaha bind would kernel panic? |
19:11.51 | obsidian-studios | greg_work: * the drivers, etc |
19:11.52 | meshuga | you got to be kidding. |
19:11.54 | greg_work | i have a lot of debian servers out there, and i trust it 100% |
19:12.05 | rob | Does anyone use OS X as their primary Asterisk server platform? |
19:12.12 | greg_work | i don't think i've had any of them kernel panic in production |
19:12.16 | MRH2 | and am not using noise suppression, echo cancellation or neting like that |
19:12.19 | vaewyn | Ohh.. I hope not rob... |
19:12.21 | obsidian-studios | meshuga: it's like this, I took a kernel config from RH, ran make oldconfig etc under debian, compiled the kernel, ran it, and after a few days I got a kernel panic |
19:12.27 | meshuga | rob : my benchmarks indicated not being able to push out more then 50 calls |
19:12.30 | meshuga | on a g5 |
19:12.33 | meshuga | concurrently |
19:12.34 | obsidian-studios | so I tossed Debian, and went Gentoo, will never look back |
19:12.40 | vaewyn | :} within an order of magnitude |
19:12.47 | meshuga | rob: plus, it lagged in weird places. i wouldn't use it as a primary. |
19:12.48 | rob | I'm just looking at it running locally for something for my home single phone. |
19:12.56 | meshuga | rob: use a wrt54g? |
19:12.57 | brookshire | i've never had a problem with debian and zaptel either, oh well :/ |
19:13.02 | rob | I'm just having some crazy issues with getting my phone to work with it properly. |
19:13.03 | greg_work | oh yeah.. thats the other big thing. for the extra money a media gateway costs compared to a TDM400p, not having to screw around with echo canncelatoin settings would easily make up for it |
19:13.24 | greg_work | mine STILL echos from time to time, and i've spent hours now messing with it |
19:13.26 | meshuga | greg_work: plus, the ability to swap in a spare in <20 seconds |
19:13.28 | meshuga | is ideal |
19:13.30 | greg_work | yes |
19:13.30 | rob | meshuga: Well, currently, I have an Asterisk server in a datacentre, which is using an IAX tunnel to get to me. |
19:13.42 | obsidian-studios | greg_work: allot of things in your env does not sound right |
19:13.44 | meshuga | rob: wrt54g will be iax tunnels. |
19:13.46 | *** join/#asterisk maik (~maik@bfs.cs.uni-sb.de) |
19:13.52 | *** join/#asterisk rajo_ (~rajo@bfs.cs.uni-sb.de) |
19:13.53 | meshuga | er will do. |
19:13.55 | rob | meshuga: And then a local asterisk, which has my 7960 connected. |
19:13.55 | greg_work | obsidian-studios: specifics? |
19:14.00 | obsidian-studios | greg_work: I would recommend not using CVS unless you are running Debian unstable |
19:14.13 | rob | meshuga: Ah, I see. I'm just wondering whether I've found a specific bug in the OS X version with some phone models, but I'm unsure. |
19:14.16 | greg_work | obsidian-studios: i'm pulling from cvs, but i'm using *-STABLE.. not head |
19:14.26 | brookshire | :) |
19:14.33 | JerJer | obsidian-studios: there is no need for anyone to run STABLE |
19:14.37 | obsidian-studios | anything in CVS is going to be compiled against newer stuff I would bet |
19:14.41 | meshuga | rob: some phone models? very doubtful. the sip implementation is solid. is your 7960 running latest sip? |
19:14.42 | JerJer | HEAD is far superior code |
19:14.52 | brookshire | yeah.. head is where it's at |
19:15.02 | obsidian-studios | greg_work: well it's like this, your problems are weird, and I would look else where |
19:15.05 | greg_work | JerJer: except its too much work to follow to be comfortable to use |
19:15.11 | rob | meshuga: Yep. it's just that at the moment, can't get audio source d at the 7960 to go anywhere else. |
19:15.16 | JerJer | greg_work: ? |
19:15.18 | greg_work | JerJer: ie, how do i know if i pull HEAD today that it will be suitable for production use? |
19:15.19 | obsidian-studios | greg_work: before you bash *, the drivers, or hardware you would have to rule it all out |
19:15.19 | brookshire | greg_work: how so? |
19:15.24 | brookshire | it's one command |
19:15.25 | JerJer | yeah how so |
19:15.26 | brookshire | make install |
19:15.32 | obsidian-studios | greg_work: I bet if it was Gentoo, most if not all would go away |
19:15.38 | meshuga | hahha |
19:15.39 | obsidian-studios | or if you were using newer stuff in your env |
19:15.47 | JerJer | obsidian-studios: asterisk doesn't give a shit about the distro you run |
19:15.53 | meshuga | obsidian is hilarious |
19:15.59 | rob | meshuga: I've had a friend around and we had it fine running on a desktop machine, however, it's a little broken using OS X. |
19:16.06 | obsidian-studios | greg_work: FYI, the * stuff in Gentoo is not as up to date as * in general, because the Gentoo guys like the test, get feedback, fix stuff etc |
19:16.18 | greg_work | are there any plans to release a 1.1 anytime soon anyways? |
19:16.21 | JerJer | don't ever run a distro version of asterisk |
19:16.25 | meshuga | rob: yes, its quite broken. the fact that apple keeps breaking backwards compatibility with every point release has alot to do with it |
19:16.26 | jfonsecausa | easy to updte from debian -asteriskk 1.0.5 to debian Asterisk 1.0.9? Which best practice CVS-HEAD or Stable from Digium? |
19:16.28 | JerJer | check it out of the cvs yourself |
19:16.35 | brookshire | 1.2 is the next planned release |
19:16.36 | obsidian-studios | JerJer: aything you compile cares about the version of gcc, glibc, kernel, kernel headers, give me a break, go program a gi |
19:16.37 | meshuga | obsidian-studios : the founder also is a microsoft employee. |
19:16.53 | JerJer | obsidian-studios: all of which has nothing to do with the specific distro you run |
19:16.56 | meshuga | obsidian-studios : actually, i disagree. |
19:16.57 | brookshire | use CVS-HEAD for now |
19:17.00 | brookshire | until 1.2 |
19:17.06 | brookshire | then after that, who knows :) |
19:17.13 | meshuga | obsidian-studios : i havent had a gcc problem since 3.* became mainstream |
19:17.13 | obsidian-studios | I know I have had apps break across version of gcc, glibc and etc |
19:17.16 | greg_work | well see, is HEAD working today? |
19:17.18 | rob | meshuga: Ah, so generally it's not too reliable? I figure I might try and route some IP space to my phone so that it's not behind NAT or anything then, and abandon the IAX tunnel. |
19:17.20 | JerJer | there is nothing stopping you from upgrading gcc in a really old version of debian |
19:17.21 | meshuga | same with glibc2 hitting mainstream |
19:17.25 | JerJer | or the kernel |
19:17.25 | brookshire | HEAD usually works |
19:17.29 | obsidian-studios | there are differences in gcc 3.3, 3.4 and 3.5 |
19:17.31 | meshuga | rob: use a wrt54g over a os x box. |
19:17.34 | greg_work | brookshire: so how do you know? |
19:17.41 | obsidian-studios | such that when using things like distcc things will majorly break bad |
19:17.42 | brookshire | but then again.. let's not talk about 1.0.8 stable not working |
19:17.45 | JerJer | obsidian-studios: and what does this have to do with what distro you run? |
19:17.48 | rob | meshuga: I'll have to source one, thanks for your help. |
19:17.52 | meshuga | obsidian-studios : and i havent personally witnessed one package that had a problem |
19:17.53 | greg_work | i mean ,say today there's some obscure bug? |
19:17.54 | obsidian-studios | JerJer: it;s all about your env |
19:17.58 | meshuga | rob: they are <$70 on froogle |
19:18.09 | obsidian-studios | JerJer: we are talking an open source app built on other open source dependencies |
19:18.10 | JerJer | obsidian-studios: ok so why in hell do you think gentoo would run any different than debian? |
19:18.12 | obsidian-studios | you got to resolve all |
19:18.16 | JerJer | ok and ? |
19:18.16 | brookshire | greg_work: i just do ;) |
19:18.27 | obsidian-studios | JerJer: using much newer software, much more current with all project release |
19:18.27 | JerJer | this has nothing to do with what distro you run |
19:18.32 | JerJer | asterisk does not care |
19:18.48 | greg_work | brookshire: because you're a dev, or follow development/cvs updates? i would love to be able to, but i simply don't have time to do it |
19:18.49 | obsidian-studios | programs are developed and compiled and tested in certain env |
19:19.05 | obsidian-studios | if you are not using at least the same versions of everything or newer you will have problems |
19:19.08 | obsidian-studios | and weird ones |
19:19.17 | obsidian-studios | we are talking about differences on the asm levels at time |
19:19.22 | greg_work | obsidian-studios: not necessarily |
19:19.26 | JerJer | again this has nothing to do with a specific distro |
19:19.27 | greg_work | and in fact, not likely |
19:19.43 | obsidian-studios | greg_work: well I have never ever had the type of unreliability and etc you speak of |
19:19.48 | obsidian-studios | greg_work: and I am in FL |
19:19.56 | greg_work | yes no one has, afaik |
19:19.59 | obsidian-studios | greg_work: it rains daily, we have all kinds of lighting, surges etc |
19:20.04 | greg_work | thuogh i don't know what location has to do with it |
19:20.06 | obsidian-studios | greg_work: so listen |
19:20.17 | obsidian-studios | greg_work: weidness like that will come from your env |
19:20.33 | obsidian-studios | when I use distcc and one machine is out of sync with gcc and glibc my binaries turn to crap |
19:20.40 | JerJer | don't use distcc |
19:20.57 | obsidian-studios | it's very very important when working with C and C++ stuff you use the same env or similar stuff, or newer |
19:21.03 | obsidian-studios | headers, libraries, etc |
19:21.08 | JerJer | no |
19:21.20 | obsidian-studios | I am a programmer and deal with this stuff daily |
19:21.35 | JerJer | i've ran asterisk on Linux based every environment i can find it it simply works |
19:21.37 | obsidian-studios | ask other * programmers, I am not a * programmer but I bet it's no different for them |
19:21.48 | pa | excuse me, where can i find source for iaxcomm? i have linuxppc and there isn't binaries for linuxppc |
19:21.58 | obsidian-studios | JerJer: ok so you explain why greg_work: has problems no one else has? Or a resolution or etc? |
19:22.04 | JerJer | obsidian-studios: i am an asterisk contributor |
19:22.16 | obsidian-studios | JerJer: not contributor, developer |
19:22.28 | obsidian-studios | talk to drumkilla and ask |
19:22.30 | JerJer | um i contribute code, dumbass |
19:22.32 | greg_work | because there's a difference? |
19:22.58 | obsidian-studios | JerJer: contribute, test, are responsible for, paid to develop etc is not all the same |
19:23.09 | JerJer | what? |
19:23.13 | JerJer | go anony someone els e |
19:23.13 | pa | ok found :) |
19:23.18 | obsidian-studios | I have contributed to many projects, but do not know them like the orginal developers |
19:23.34 | obsidian-studios | or people maintaining or responsible for tracking down bugs, fixing them etc |
19:23.49 | shido6 | clueless |
19:23.50 | greg_work | out of curiousity, JerJer what's your opinion on using a TDM400 vs external media gateway? |
19:23.55 | obsidian-studios | JerJer: bottom line I am trying to help greg_work: |
19:23.56 | JerJer | obsidian-studios: ok and your point is? |
19:24.03 | JerJer | TDM400 any day |
19:24.08 | obsidian-studios | ***Nuxi JGenerator and others I did not request to be on the project |
19:24.23 | obsidian-studios | <PROTECTED> |
19:24.31 | JerJer | obsidian-studios: find a clue |
19:24.38 | The_LighterSide | anyone know anything about a software DSS panel? |
19:24.41 | obsidian-studios | JerJer: ask drumkilla |
19:24.45 | file[laptop] | break it up. |
19:24.50 | greg_work | The_LighterSide: FOP (asternic.org i think) |
19:24.51 | obsidian-studios | JerJer: drumkilla maintains the 1.* branch and etc |
19:24.58 | The_LighterSide | thanks :) |
19:25.00 | MikeJ[Laptop] | find a clue? |
19:25.32 | file[laptop] | at Cluecon! |
19:25.34 | file[laptop] | ~cluecon |
19:25.34 | jbot | rumour has it, cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
19:25.48 | JerJer | obsidian-studios: i don't run STABLE for obvious reasons |
19:26.08 | obsidian-studios | bottom line I do not consider the TDM400P cards to be crap, and I chimed in because greg_work: was mentioning it, and problems I have never ever ever heard of |
19:26.36 | obsidian-studios | so I was trying to help out, and before I called Digium or said their cards, drivers or etc was junk, I would fully research, test and etc |
19:27.25 | MRH2 | is there any recommendation enabling / disabling /or no real difference hyperthreading on a machine running asterisk. |
19:27.33 | greg_work | obsidian-studios: i never said they were junk, i just said based on my experience i don't trust them, and wouldn't recommend them |
19:27.38 | crash3m | fragmentation of a SIP register packet would cause issues right? |
19:27.49 | obsidian-studios | greg_work: yes I know but you have to look at why or find a reason |
19:28.06 | obsidian-studios | greg_work: I would bet $ that if you say switched to Gentoo that problem would go away, |
19:28.08 | crash3m | MRH2: I thought I heard multithreading was found very insecure recently |
19:28.25 | greg_work | obsidian-studios: well, thats out of the question, and frankly, a rediculus fix |
19:28.27 | obsidian-studios | greg_work: I would love to be wrong and the problem exist regardless of env |
19:28.46 | Assid | hrmm.. do you guys suggest using CVS for production? |
19:28.47 | obsidian-studios | greg_work: well then go find out what the min gcc, glibc and etc the * developers recommend |
19:28.49 | brookshire | always multithread |
19:28.57 | greg_work | 1, i don't know gentoo.. 2, this is a production server that runs a critical system |
19:28.57 | JerJer | i have countless TDM400P based systems and have never had a single problem with FXS |
19:29.03 | *** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
19:29.05 | obsidian-studios | greg_work: as I bet there is something in your env specifically causing the weirdness |
19:29.11 | greg_work | i'm not switching OSs |
19:29.19 | greg_work | JerJer: this is 4-port fxo |
19:29.31 | obsidian-studios | greg_work: I run production critical * deployments on Gentoo with no problems |
19:29.35 | coldfeet | does nayone know howto use variables in a mYSQL query in asterisk... |
19:29.48 | coldfeet | ...if ${name} =paul |
19:30.02 | coldfeet | how would you incorporate this in a where statement |
19:30.04 | obsidian-studios | greg_work: I have had a TDM400P in use for close to a year now without any problems, doing all kinds off stuff |
19:30.17 | greg_work | JerJer: i have intermittent echo issues, but the biggest issue is that i ocasionally get a message scrolling "TDM PCI master abort" and the machine effectively locks up |
19:30.19 | brookshire | where? |
19:30.31 | coldfeet | where fname=${name} or where fname=\${name} |
19:30.34 | obsidian-studios | greg_work: you do not have to switch, but research to see what the mins you need to get for your env are, and work on getting them |
19:30.34 | MRH2 | hyperthreading as 1 cpu lookis like 2 |
19:30.38 | brookshire | no second where |
19:30.41 | coldfeet | tried all types of escpaes but no luck |
19:30.45 | JerJer | greg_work: I have setup like 3 or 4 FXO systems and they are doing fine |
19:30.54 | greg_work | obsidian-studios: it would take me abuot 12 seconds to upgrade to newer gcc or whatever |
19:30.55 | *** join/#asterisk darkskiez (~mhb@host-84-9-95-155.bulldogdsl.com) |
19:31.06 | brookshire | don't forget the '' |
19:31.07 | obsidian-studios | greg_work: at min you should use a newer kernel, since I bet the zaptel, and etc drivers out of cvs are meant for a newer kernel? |
19:31.10 | brookshire | learn mysql :) |
19:31.12 | crash3m | mrgoby: CORRECT |
19:31.21 | funxion | coldfeet: maybe System(mysql -e select blah blah blah) |
19:31.22 | JerJer | greg_work: are you running a known supported motherboard? |
19:31.28 | JerJer | funxion: oh god no |
19:31.29 | *** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
19:31.33 | obsidian-studios | greg_work: so mess around with it, I have a very strong feeling your weirdness is due to something in your env |
19:31.49 | greg_work | JerJer: well, i don't know what the issue is. it's hard for me, because it's production. i can't reliably reproduce it, it just happens after somewhere between 2 and 30 days |
19:32.15 | mrgoby | crash3m: about what ? |
19:32.19 | greg_work | i reload the wctdm drivers nightly, and it seems to help (it crashed this morning, but someone else rebooted, so i don't know if that was the issue .. probably though) |
19:32.43 | Mw3 | hi, i have a tdm400p with 4 fxs in our office. sometimes the secretary is annoying me with this: when she made a mistake during dialing and she press down the hook button for a 1sec there will be no dial tone |
19:32.47 | obsidian-studios | greg_work: you can point and blame *, zaptel, and the drivers all you want, but they are not getting complied by them selves or in a env or world all to themselves. It is pulling stuff from your env, and depending on what it gets, might vary your output |
19:33.00 | JerJer | obsidian-studios: bullshit |
19:33.11 | greg_work | JerJer: i'm not sure, where's the list of supported boards? |
19:33.17 | mishehu | I don't think I'll get any more of the tdm400 cards... |
19:33.38 | JerJer | http://www.digium.com/index.php?menu=compatibility |
19:33.39 | obsidian-studios | greg_work: or look at it like this, you bought a Digium card that comes with Digium support so call Digium |
19:33.46 | crash3m | mrgoby: misfire, sorry |
19:33.52 | obsidian-studios | <PROTECTED> |
19:33.54 | crash3m | MRH2: correct |
19:34.18 | JerJer | obsidian-studios: everything came from cvs at some point |
19:34.19 | JerJer | so yes |
19:34.25 | obsidian-studios | <PROTECTED> |
19:34.35 | JerJer | and you are talking pure bullshit |
19:34.44 | obsidian-studios | JerJer: you get me an min * requirement sheet or compatibility for gcc, glibc , etc |
19:34.51 | JerJer | it doesn't care |
19:35.20 | *** part/#asterisk cpm (~Chip@cpm.sustaining.supporter.pdpc) |
19:35.21 | JerJer | i've ran asterisk on gcc 2.95 to 3.4 without a single issue |
19:35.25 | brookshire | obsidian-studios: at least not for gcc and glibc |
19:35.33 | JerJer | glibc and uclibc - same |
19:35.37 | meshuga | hahah obsidian is still going off on how much gentoo is better |
19:35.39 | brookshire | now there are a few necessary libraries |
19:35.40 | JerJer | it simply doesn't matter |
19:36.14 | obsidian-studios | meshuga: I am not talking about gentoo, If I develop an app with a version of glibc and compile it with gcc, and you do it with older versions you are going to have issues |
19:36.16 | brookshire | any distro that you have to compile 'nslookup' first in order to use it, is crap |
19:36.19 | mutilator | whats a good prog to merge in and out monitoring? |
19:36.21 | brookshire | :) |
19:36.23 | mutilator | wavs |
19:36.54 | JerJer | soxmix |
19:37.07 | obsidian-studios | Gentoo just happens to be a source distro so you are compiling and running the latest version of software from each of the open source projects |
19:37.23 | JerJer | obsidian-studios: then you are a shitty developer |
19:37.27 | mutilator | =\ |
19:37.29 | mutilator | no apt source |
19:37.44 | obsidian-studios | JerJer: great I am glad you feel that way, as you know what about me |
19:37.55 | obsidian-studios | JerJer: did you get drumkilla's input? |
19:38.08 | funxion | mutilator there is apt source for sox |
19:38.10 | obsidian-studios | JerJer: or anyone at Digium to back up your statements? |
19:38.20 | greg_work | JerJer: nothing on that page says anything about my hardware. |
19:38.30 | MRH2 | crash3m: k hyperthreading enabled it is |
19:38.35 | JerJer | i'm speaking about my personal experiences with asterisk |
19:38.37 | mutilator | ;P |
19:38.39 | obsidian-studios | greg_work: contact Digium you get paid support when you bought the card |
19:38.51 | obsidian-studios | greg_work: that's the horses mouth, so hear it there |
19:39.04 | JerJer | i have done thouands of installs in various environments |
19:39.22 | greg_work | yeah, the only thing is i don't have anything to tell them |
19:39.23 | obsidian-studios | JerJer: great, so explain and fix greg_work problem |
19:39.29 | JerJer | don't run stable |
19:39.33 | greg_work | "my tdm400p sometimes dies" |
19:39.51 | greg_work | HEAD scares me |
19:39.58 | obsidian-studios | greg_work: they will work with you and not leave you hanging, I bet they will start asking about the env and etc |
19:40.04 | Assid | head scares you? |
19:40.07 | Assid | why? |
19:40.09 | MRH2 | lol |
19:40.11 | greg_work | if i had time to follow the -dev list and cvs updates, i would run it |
19:40.14 | obsidian-studios | Assid: teeth |
19:40.16 | JerJer | i love head - i wish i could get some more of it |
19:40.17 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
19:40.22 | Assid | hehe |
19:40.48 | JerJer | greg_work: who says you have to follow anything? |
19:40.56 | greg_work | as is, i don't have any way of knowing that it's suitable for production if i check it out right now |
19:40.57 | brookshire | just update every month or so |
19:40.59 | brookshire | lol |
19:41.05 | JerJer | greg_work: very simple |
19:41.10 | JerJer | cd /usr/src |
19:41.11 | greg_work | other than if * crashes tomorrow, then i know it wasn't ;) |
19:41.13 | JerJer | mkdir production |
19:41.15 | JerJer | mkdir development |
19:41.18 | JerJer | cd development |
19:41.23 | JerJer | cvs co asterisk zaptel |
19:41.31 | JerJer | then compile those |
19:41.33 | Assid | JerJer: what u suggest? head/production? |
19:41.44 | JerJer | then make install everything |
19:41.46 | brookshire | HEAD |
19:41.48 | brookshire | lol |
19:41.50 | CoaxD | anyone have any idea how to unfuck a locked e-brake (drum!) on a 1997 plymouth breeze? :) |
19:41.51 | JerJer | then if you find a problem in the development directory |
19:41.52 | brookshire | HEAD HEAD HEAD |
19:41.56 | brookshire | USE HEAD |
19:41.57 | brookshire | :) |
19:41.58 | Assid | bah.. dev. / production |
19:41.59 | JerJer | simply cd /usr/src/production |
19:42.01 | CoaxD | the cable lets down.. but the damn brake wont unlock! *blah* |
19:42.07 | JerJer | and make install those and reload the kernel module |
19:42.11 | JerJer | back to your old sysem |
19:42.14 | JerJer | system |
19:42.19 | greg_work | JerJer: oh yeah, fair enough |
19:42.20 | JerJer | 29 seconds |
19:42.21 | tzanger | werd to the jer jer herd |
19:42.24 | CoaxD | Got the tire off. need to get the drum off, but the brakes are locked. blah. |
19:42.24 | greg_work | except "find a problem" |
19:42.37 | JerJer | come in a 2am and make those changes |
19:42.40 | JerJer | then make a dozen calls |
19:42.41 | greg_work | that means.. the phone system crashes? what if i'm not in the office? what if it's 3am? |
19:42.48 | brookshire | CoaxD: #1997plymouth :) |
19:43.00 | brookshire | i doubt anyone will know that answer in here |
19:43.01 | brookshire | lol |
19:43.11 | CoaxD | brookshire: HEY! bah. i thought.. why not just spew my pain in an entirely off-topic manner, here in a group with a lot of people, and see what happens.. :) |
19:43.11 | greg_work | what if it's a bug (like whatever the problem is with my tdm400p) that will show up at some random point after days of running smoothly? |
19:43.12 | funxion | lol |
19:43.16 | MRH2 | yay compile finished bbl |
19:43.19 | ManxPower | OR you can just wait a few days after a 1.0.x release and then not have to do as much pre-production testing. |
19:43.27 | SpaceBass | wow... just noticed you can get allison to record voice prompts through digium's site |
19:43.34 | CoaxD | brookshire: Hey, at least my asterisk has a 3 month uptime, and doesnt require fixing! *lol* |
19:43.37 | brookshire | lol |
19:43.41 | JerJer | greg_work: then you know the problem still exists |
19:43.44 | greg_work | ManxPower: like exactly what i do now. except everyone is telling me i shuoldn't run 1.0.x because HEAD is better :) |
19:43.47 | obsidian-studios | SpaceBass: really for free or $? |
19:43.59 | JerJer | to which you can say "this problem exists in cvs head as of such and such date" |
19:44.02 | greg_work | JerJer: and i'd be totally willing to test that on a testing system |
19:44.06 | ManxPower | greg_work: They are blinded by all the cool new features. |
19:44.13 | greg_work | but i don't feel as adventerous with my production systems |
19:44.16 | JerJer | so then buy another TDM400P with one FXO and test |
19:44.17 | tzanger | ManxPower: oh please. :-) |
19:44.32 | greg_work | JerJer: if digium wants to give me a tdm400p and a system, i'd be enitrely willing to test |
19:44.40 | greg_work | i doubt that will happen though |
19:44.44 | JerJer | buy one |
19:44.44 | ManxPower | They are prolly the same people that try to get their Longhorn and then deploy it in production |
19:44.46 | JerJer | they are cheap |
19:45.02 | obsidian-studios | greg_work: have you spoke to Digium at all or any of their developer/employees |
19:45.02 | robl^ | SpaceBass, you can also contact allison directly. Digium's site is good for small volume recording (3-4 prompts), but Allison will work by the hour and it will be cheaper if you want LOTS of prompts |
19:45.03 | *** join/#asterisk AlexCeli (~Alex@200.37.85.90) |
19:45.12 | greg_work | if i'm going to buy anything, it will likely be a media gateway ;) |
19:45.17 | salvini_fs | can someone help me setting up asterisk with mywebcalls? |
19:45.24 | greg_work | obsidian-studios: no |
19:45.36 | obsidian-studios | greg_work: give it a shot, you already paid for it when you purchased the card |
19:45.37 | greg_work | maybe i'll just email them now |
19:45.39 | JerJer | greg_work: then you will find problems in chan_sip |
19:45.42 | brookshire | robl^: sshhh! don't tell everyone our secert :) |
19:45.45 | obsidian-studios | greg_work: should get you somewhere faster than here |
19:45.49 | greg_work | JerJer: hehe, its possible ;) |
19:45.59 | obsidian-studios | greg_work: I would call, not email, but either way |
19:46.04 | SpaceBass | robl^ somehow I cannot justify her for my home * box IVR |
19:46.06 | robl^ | brookshire, ooops! sorry! you'll have to punish me! |
19:46.15 | SpaceBass | robl^ but it would be fun |
19:46.19 | obsidian-studios | greg_work: I would be prepared to grant them access to the machine if possible |
19:47.06 | *** join/#asterisk anti (russ@anti.developer.gentoo) |
19:47.13 | SpaceBass | LOL |
19:47.27 | SpaceBass | I've never been whipped with cat5... but imagine it might hurt... fiber would too |
19:47.41 | brookshire | not cat5, cata5! |
19:47.44 | brookshire | :) |
19:49.18 | pa | what is "predictive dialer"? |
19:49.29 | ManxPower | Wonka: Great idea! I'll have to try making one of those, as well as a fiber version |
19:49.34 | *** join/#asterisk johnm (~johnm@johnm.developer.gentoo) |
19:49.43 | ender | is there a page on the wiki that details the RFC for digitplan syntaxes? |
19:49.50 | JerJer | greg_work: you check the standard shit? interrupt sharing? |
19:49.52 | MikeJ[Laptop] | NERDS! |
19:50.12 | eKo1 | cat5 is nothing, try rg-49 coaxial.... |
19:50.44 | greg_work | JerJer: as much as i could, i don't really know enough about that anymore (ie, it said irq 207 or something) |
19:50.55 | funxion | pa: google it |
19:50.58 | Wonka | one with two markings, to hang it on the wall with a vampire clamp |
19:50.59 | SpaceBass | I'm a functional nerd... quite similar to a functional alcholic... I keep it well hidden... I am an normal member of societ... I never wear my "got root" shirt out of the house |
19:51.00 | JerJer | ?! 207 |
19:51.05 | JerJer | cat /proc/interrupts |
19:51.10 | SpaceBass | of course I'm also a functioning alcholic... but thats another room |
19:51.12 | funxion | lol |
19:51.14 | *** join/#asterisk [Jedi] (iRRRRrrm@154.Red-217-127-168.pooles.rima-tde.net) |
19:51.16 | [Jedi] | Hello |
19:51.17 | Wonka | and one with some good handle, to use as a LART |
19:51.19 | greg_work | 201: 27529176 IO-APIC-level wctdm |
19:51.30 | JerJer | wholy WTF batman |
19:51.43 | greg_work | thats what I thought ;) |
19:51.55 | greg_work | sec, i'll show you the whole thing |
19:52.05 | JerJer | use pastebin |
19:52.13 | greg_work | http://pastebin.ca/16968 |
19:52.34 | ManxPower | greg_work: if you use APIC you can get things on interrupts 2 - 240 or so. |
19:53.22 | greg_work | ManxPower: APIC is new to me.. i stopped messing with hardware a long time ago, that was the first time i've seen an irq > 15 :p |
19:53.37 | JerJer | yeah i don't know much about APIC |
19:53.55 | JerJer | can you try to turn that off in bios? |
19:54.04 | greg_work | someone suggested changing the PCI slot, i haven't had a chance to do that yet |
19:54.05 | JerJer | and turn off the parallel port and usb (if you are not using them) |
19:54.07 | greg_work | hm, not sure |
19:54.12 | greg_work | yeah, thats a good idea |
19:54.19 | JerJer | and eth1 |
19:54.23 | obsidian-studios | you can turn off APIC at the kernel level on boot apic=no |
19:54.24 | JerJer | again if not in use |
19:54.26 | greg_work | i use eth1 |
19:54.28 | JerJer | ok |
19:54.38 | ManxPower | greg_work: APIC gives you LOTS of interrupts. |
19:54.44 | obsidian-studios | whoops apic=off |
19:54.50 | ManxPower | IF your motherboard uses it well. |
19:54.55 | JerJer | but does it also give you LOTS of problems? |
19:55.10 | greg_work | ManxPower: which frankly is probably a good thing :) 15 irqs always used to be the biggest PITA... :) |
19:55.12 | obsidian-studios | greg_work: make another entry in grub or what ever boot loader and append apic=off to turn off and see |
19:55.49 | obsidian-studios | greg_work: even if present in bios, using that command will disable in kernel, you can also not build APIC support in the kernel to go even further |
19:56.02 | greg_work | ManxPower: well, being the only person who seems to know about APIC here :) .. do you think theres a benefit to turning it off? |
19:56.11 | obsidian-studios | greg_work: APIC not really a great idea for servers and etc |
19:56.27 | obsidian-studios | greg_work: unless you have a need you are aware of |
19:56.58 | obsidian-studios | greg_work: you will not hurt anything by turning off APIC |
19:57.55 | greg_work | (btw, i'm using an Asus A7N8X mobo, with an athlon XP 2000+ cpu) |
19:58.45 | obsidian-studios | greg_work: hardly a specific server board? Kill APIC and see if you problems go away |
19:59.52 | greg_work | alright, well, i'll try some stuff in about 1.5 hours, (5:30 EST) if you guys are still around |
20:00.10 | greg_work | people don't take kindly to taking the phone system offline in the middle of the day ;P |
20:01.51 | ManxPower | greg_work: Not really. In theory APIC might increase your interrupt latency a little bit, but that's the only issue I know of. |
20:02.24 | [Jedi] | Last week I bought a new TE405P |
20:02.30 | ManxPower | greg_work: if you disable APIC, then you'll have to do all the work to confirm the card is not sharing interrupts, of course. |
20:02.31 | [Jedi] | and it says "Found a Wildcard: Wildcard TE410P/TE405P (1st Gen)" |
20:02.35 | obsidian-studios | ok I was totally off base with what I was looking for with regard to * call notification. Now I am leaning toward notification via some IM protocol. Anyone got info if one is better, more reliable than another. I currently can use IRC, or GAIM/AIM. I see ways for Jabber, but do not have a Jabber account etc? |
20:02.37 | [Jedi] | what does (1st Gen) mean? |
20:02.47 | [Jedi] | is my card an old model? |
20:02.48 | file[laptop] | first generation |
20:02.53 | ManxPower | [Jedi]: That means it's not a second generation card. |
20:03.06 | file[laptop] | imagine that! |
20:03.06 | [Jedi] | and what's a second generation card? what's the difference? |
20:03.28 | file[laptop] | http://www.digium.com/index.php?menu=press/pr_2gen_firm |
20:03.33 | mutilator | when it has babies |
20:03.42 | ManxPower | thank you, file |
20:03.47 | [Jedi] | thankx file |
20:03.48 | file[laptop] | go look at that URL |
20:04.00 | file[laptop] | then talk to Digium, cause that's the extent of our knowledge really in this area |
20:04.37 | [Jedi] | why have they sold me an old card if there're newer models available? |
20:04.59 | [Jedi] | well not they but their 'reseller' in spain... :// |
20:05.02 | funxion | obsidian-studios: what type of call notification are you doing? |
20:05.03 | darkskiez | as thats _just_ been released |
20:05.09 | file[laptop] | maybe the reseller didn't have it yet |
20:05.25 | [Jedi] | I think they should have told me |
20:05.39 | [Jedi] | it was my fault, in fact |
20:05.43 | [Jedi] | but they should have told me |
20:05.45 | file[laptop] | so talk to them... |
20:05.48 | file[laptop] | nothing we can do about it |
20:05.55 | obsidian-studios | <PROTECTED> |
20:05.58 | ManxPower | [Jedi]: for %90 of people there is no functional difference. |
20:06.02 | darkskiez | [Jedi]: you can post the card back and have it updated |
20:06.04 | *** join/#asterisk JohnJacob (~JohnJacob@pcp0011543387pcs.mainf01.in.comcast.net) |
20:06.18 | [Jedi] | it's just a firmware updatE? |
20:06.23 | darkskiez | yes |
20:06.28 | funxion | AHH |
20:06.31 | [Jedi] | uhm and it can't be done by software? |
20:06.38 | darkskiez | one of the features of the new firmware is - Field upgradable firmware for future updates. |
20:06.43 | [Jedi] | hehehe |
20:07.21 | ender | which dtmf mode should I use for Polycom phones? |
20:07.51 | [Jedi] | well I'll keep my old "1st gen" card... and I'll buy next cards directly from digium |
20:08.07 | darkskiez | [Jedi]: whened you buy the card? |
20:08.22 | [Jedi] | when? last week |
20:08.26 | Assid | okay heres something weird.. |
20:08.40 | Assid | i have to connect asterisk as a front end to a voip gateway |
20:08.43 | darkskiez | [Jedi]: dont think 2nd gen existed last week |
20:08.48 | [Jedi] | wow |
20:08.54 | [Jedi] | are they that new? |
20:09.04 | Assid | when a person presses an extension.. how do i send that to the gateways extension |
20:09.06 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
20:09.07 | [Jedi] | then... bad luck for me :) |
20:09.19 | JerJer | Assid: by dialing |
20:09.31 | Assid | JerJer: a bit more complicated.. |
20:09.35 | JerJer | no |
20:09.37 | JerJer | not at all |
20:10.03 | darkskiez | I concur |
20:11.48 | [Jedi] | which is the top number of TDM400 cards with FXO in a system? |
20:11.58 | [Jedi] | are these cards CPU-intensive like PRI cards? |
20:12.03 | JerJer | i wouldn't go any more than 1 maybe 2 |
20:12.12 | *** join/#asterisk wasim (~wasim@wasim.active.supporter.pdpc) |
20:12.17 | JerJer | because a T1 card+channel bank will be more effective |
20:12.32 | [Jedi] | but if it PRI wasn't in any way available |
20:12.40 | [Jedi] | and I really needed as many FXO's as possible |
20:12.40 | brookshire | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
20:12.43 | *** join/#asterisk heka (~heka@82.114.68.124) |
20:12.50 | [Jedi] | how much FXO's could work in a single system? |
20:12.59 | InfraRed | one million |
20:13.00 | Assid | JerJer: i dont knowhow to accept authorization from this device |
20:13.02 | JerJer | [Jedi]: so then get a T100P and an Adtran TA750 with FXO cards |
20:13.11 | JerJer | Assid: a type=user |
20:13.17 | ManxPower | [Jedi]: With a 4-port card and a channel bank you can put 4x24 ports |
20:13.20 | Assid | hrmm |
20:13.24 | ManxPower | well, 4 channel banks |
20:13.32 | JerJer | yp |
20:13.33 | JerJer | yep |
20:13.34 | Assid | lemme do this.. lemme set up the softphones.. and tell them to test on this.. |
20:13.41 | [Jedi] | that would be for PRI T1/E1 cards |
20:13.48 | [Jedi] | what I need are standard phone-line interfaces |
20:13.51 | JerJer | no |
20:13.57 | JerJer | a channel bank gives you FXO ports |
20:13.59 | brookshire | you could use e1 and get more channels ;) |
20:14.00 | [Jedi] | uhm |
20:14.05 | JerJer | and aggregates them into a T-1 |
20:14.11 | [Jedi] | oh I see |
20:14.26 | [Jedi] | and how much does such a thing cost? |
20:14.33 | ManxPower | [Jedi]: A channel bank converts each channel in a T-1/E-1 into an analog port. (FXO or FXS, depending on the cards in the channel bank) |
20:14.34 | JerJer | depends |
20:14.42 | ManxPower | [Jedi]: I just bought a channel bank off eBay for $180 |
20:14.44 | brookshire | adtran is expensive, but high quality |
20:14.59 | [Jedi] | ManxPower: 180$ for how many E1s? |
20:14.59 | JerJer | and a T-1 card is like 500 $ |
20:15.05 | [Jedi] | ando how many FXOs? |
20:15.06 | ManxPower | Adtran Total Access 750 (Adtran TA is the only channel bank I'll buy) |
20:15.08 | JerJer | you don't want E-1s |
20:15.22 | brookshire | e1 equip is more expensive |
20:15.37 | ManxPower | [Jedi]: 12 ports, these were FXS. You'll have trouble finding FXOs. Why do you want FXOs anyway? |
20:15.46 | [Jedi] | ManxPower: long history |
20:15.59 | ManxPower | [Jedi]: You will be VERY unhappy with analog FXO. |
20:15.59 | brookshire | cheaper than a t1? |
20:16.16 | [Jedi] | cheaper prices for fxo termination |
20:16.18 | [Jedi] | for some destinations |
20:16.24 | brookshire | make since :) |
20:16.35 | ManxPower | We paid $300 for a 24 port FXS Adtran TA 750 a few months ago |
20:16.37 | brookshire | digium is coming out with a 24port fxo card soon, maybe 2 months |
20:16.44 | [Jedi] | uhmmmm interesting |
20:16.52 | ManxPower | brookshire: We've been hearing that for the past 18 months. |
20:17.01 | brookshire | yeah.. prototype is in |
20:17.01 | ManxPower | I'll believe it when it SHIPS. |
20:17.03 | [Jedi] | less interesting then |
20:17.08 | [Jedi] | :) |
20:17.25 | [Jedi] | so i could use an E1/T1 port of my TE405P and a channel bank |
20:17.35 | brookshire | yeah |
20:17.44 | [Jedi] | and the quality is that bad, brookshire? |
20:17.49 | darkskiez | 24port FXO? that must be a beasty cable that attaches. |
20:17.56 | ManxPower | [Jedi]: Yes. But remember since FXOs are much less common in channel banks, you'll have a harder time finding them on the used market. |
20:17.56 | brookshire | no.. what do you mean? |
20:18.01 | *** join/#asterisk shido (~greg@d57-87-253.home.cgocable.net) |
20:18.03 | ManxPower | darkskiez: Amphenol, I'm sure. |
20:18.12 | ManxPower | Just like channel banks use |
20:18.20 | shido | yep |
20:18.24 | brookshire | amphenol connector |
20:18.29 | brookshire | yes! |
20:18.43 | mutilator | J1 = ? |
20:18.54 | brookshire | j1 = japanese t1 |
20:18.59 | darkskiez | would one of them fit on a pci? |
20:19.04 | mutilator | ..o |
20:19.16 | Silik0n | when are the tXXXp's getting J1 part cert? |
20:19.20 | brookshire | yeah |
20:19.33 | brookshire | Silik0n: probably never |
20:19.36 | brookshire | :) |
20:19.49 | Silik0n | I know how to get that done ;) |
20:19.55 | brookshire | really now |
20:19.58 | Silik0n | yeah ;) |
20:20.02 | brookshire | because that's impossible in the us |
20:20.04 | brookshire | LOL |
20:20.08 | Silik0n | nope |
20:20.10 | Silik0n | not really |
20:20.22 | ManxPower | You bribe the certification authority, of course. |
20:20.26 | brookshire | talk with malcolm please :) |
20:20.46 | darkskiez | Silik0n: they'll never know you've connected an uncertified card :) |
20:20.49 | brookshire | hit them with the cata5 pair |
20:21.23 | Silik0n | darkskiez but most japanese purchasers of such equipment will not even consider it if its not cert'd |
20:21.38 | darkskiez | Silik0n: they wont even know there is certification. |
20:21.53 | Silik0n | darkskiez are you japanese? |
20:22.05 | *** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net) |
20:22.08 | milkyflava | hello |
20:22.08 | darkskiez | Dont have a mirror handy |
20:22.37 | Silik0n | or should I say do you really understand how rigid their society is... especially their business society... |
20:22.40 | milkyflava | I have a TDM11B dev card and I have * setup and my analog phone connected to it |
20:22.42 | eKo1 | i though japan uses the US system... |
20:22.49 | darkskiez | Silik0n: are the Sangoma cards certified there? |
20:22.49 | Silik0n | hell they dont even fart without making sure their asses are covered |
20:22.55 | brookshire | no |
20:22.55 | Silik0n | nope |
20:23.09 | Silik0n | sangomas are not cert'd at this time |
20:23.10 | eKo1 | US carrier system that is |
20:23.18 | darkskiez | Silik0n: get a business sponser and get them certified yourself |
20:23.18 | milkyflava | I was wondering if a digital phone in the 2.4ghz range would not work plugged directly into an * box |
20:23.25 | ManxPower | eKo1: no. Japan uses their own varient of T-1 |
20:23.43 | Silik0n | eKo1 actually the japanese telecom system is sorta like hte US system only bastardized to lock out non-japanese equipment and such |
20:23.44 | brookshire | ?? |
20:23.46 | darkskiez | a DECT FXS base card would be nice. |
20:23.53 | brookshire | 2.4ghz phone? |
20:23.57 | brookshire | why wouldn't it? |
20:24.03 | milkyflava | yes, from uniden |
20:24.08 | ManxPower | all mine do. |
20:24.10 | milkyflava | excellent |
20:24.11 | eKo1 | Silik0n: ah, that said it all |
20:24.28 | *** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1) |
20:24.32 | milkyflava | I am getting a dial tone but as soon as I hit a number it hangs up so I just wanted to make sure |
20:24.36 | Silik0n | eKo1 they dont even have GSM ... they has some whacky implementation of CDMA |
20:24.46 | milkyflava | thanks brookshire |
20:24.49 | eKo1 | cdma2000? |
20:24.58 | eKo1 | wait, that's qualcom... |
20:25.03 | Silik0n | i forget what it is, but its not "standard" cdma |
20:25.18 | Silik0n | its the Japanese version... heh cant use a japanese cell phone anywhere but in japan |
20:25.30 | darkskiez | Silik0n: could you get a certified convertor ? |
20:25.42 | Silik0n | there are people working on such beasts |
20:25.43 | Himeko | unless it's a vodafone phone |
20:25.54 | Himeko | vodafone is running w-cdma 2100 |
20:27.03 | Himeko | the other providers are using whacky stuff |
20:27.14 | darkskiez | Silik0n: http://www.voip-info.org/tiki-index.php?page=Sunrise+J1+PRI+solutions |
20:27.24 | darkskiez | <PROTECTED> |
20:27.37 | Himeko | ouch |
20:27.39 | Silik0n | darkskiez yeah |
20:27.42 | [Jedi] | wow |
20:27.44 | [Jedi] | that's a lot |
20:27.46 | Silik0n | thats only $500 USD |
20:27.51 | Himeko | liek is liek $4000 |
20:27.51 | [Jedi] | 500 $? |
20:27.55 | Silik0n | or is it $5000 USD |
20:27.58 | [Jedi] | this is like 3500EUR |
20:28.11 | Himeko | er that is like $4000 |
20:28.18 | darkskiez | <PROTECTED> |
20:28.23 | Silik0n | I think its 105 jpy to 1USD |
20:28.37 | Silik0n | or something like that anyway |
20:28.39 | shido | errr |
20:28.44 | [Jedi] | 498 JPY or 498000 JPY? |
20:28.45 | shido | why not use the quad t1 |
20:28.47 | shido | e1 in |
20:28.48 | shido | t1 out |
20:28.48 | [Jedi] | hehe |
20:28.49 | shido | done |
20:28.59 | darkskiez | 1 USD = 111 JPY |
20:29.09 | Silik0n | 498.000 is japnese for 498,000 for you north americans |
20:29.17 | Himeko | .=, as a delimiter in manyplaces of the wor;d |
20:29.33 | *** join/#asterisk astor (~chatzilla@135.80-203-85.nextgentel.com) |
20:29.35 | [Jedi] | almost everywhere excepting english, . is the delimiter |
20:29.44 | milkyflava | When I hit a number button on my phone in the * cli it says Hung up 'Zap/1-1' why is that happening? |
20:30.07 | darkskiez | so, 4500 USD, nice. |
20:30.44 | astor | are there any regulatory issues with using a Uniden UIP1868 wireless handset which operates on 5.8GHz in europe? |
20:31.33 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
20:32.00 | darkskiez | Not sure what this is, but would you use two to do a J1/T1 convertor thing http://www.pcuniverse.com/464991.htm ? |
20:32.35 | mogorman | digium card will work on a j1 line |
20:32.50 | Assid | is incoming free on michigan did's for nufone? |
20:33.10 | darkskiez | mogorman: he needs a certified thing tho. |
20:33.19 | mogorman | ouch |
20:33.27 | mogorman | i dont know of anyone who has it cert. |
20:33.45 | brookshire | mogormon: you can use j1 internally in a company though |
20:34.06 | milkyflava | I think it has to do with my zapata.conf file but am not sure |
20:34.59 | *** part/#asterisk [Jedi] (iRRRRrrm@154.Red-217-127-168.pooles.rima-tde.net) |
20:35.14 | *** join/#asterisk The_LightSide (~dt@wbs-196-2-121-52.wbs.co.za) |
20:35.37 | *** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-94.dsl.spfdmo.swbell.net) |
20:35.56 | *** join/#asterisk loick (~loick@ATuileries-151-1-13-190.w82-123.abo.wanadoo.fr) |
20:36.18 | *** join/#asterisk bankrupt (~bank@pcp08695394pcs.500ash01.tn.comcast.net) |
20:36.46 | Ayano | what's new in the asterisk world? |
20:37.02 | InfraRed | nothing |
20:37.13 | InfraRed | i'l just having problems with 1.0.10 |
20:37.17 | InfraRed | i'm |
20:37.24 | Ayano | what is it doing? |
20:37.46 | InfraRed | nothing i want it to do |
20:37.55 | Ayano | I've been there befor |
20:38.13 | darkskiez | Silik0n: http://lists.digium.com/pipermail/asterisk-dev/2004-September/006090.html |
20:38.51 | *** join/#asterisk nosc (~nosc@200.121.129.178) |
20:39.38 | milkyflava | does [trunkgroups] have to be in the zapata.conf? |
20:40.08 | Cresl1n | you can use it publicly too |
20:40.12 | Cresl1n | just don't make a big deal about it :-) |
20:40.39 | brookshire | you just can't use it to connect to the japanese telco |
20:40.59 | Ayano | or india's pots lines |
20:41.50 | Assid | 1,0.10 out ? |
20:42.21 | InfraRed | :) |
20:42.23 | InfraRed | finally |
20:42.30 | Assid | its out??? |
20:42.40 | InfraRed | someone who noticed my subtle lame joke |
20:42.59 | Ayano | OOPPS |
20:43.16 | InfraRed | well done Assid |
20:43.22 | InfraRed | you win a cup of tea |
20:43.29 | InfraRed | bit cold |
20:43.33 | InfraRed | but it's nice |
20:43.34 | InfraRed | :) |
20:44.30 | *** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net) |
20:45.40 | *** join/#asterisk [Jedi] (iRRRRrrm@154.Red-217-127-168.pooles.rima-tde.net) |
20:45.59 | Assid | brb |
20:46.15 | milkyflava | Can someone look at this http://pastebin.ca/16972 and tell me where to look. |
20:46.47 | *** join/#asterisk The_LightSide (~dt@wbs-196-2-123-224.wbs.co.za) |
20:46.55 | milkyflava | I think it has to do with my zapata.conf file that I am probably missing something |
20:47.36 | milkyflava | I have a TDM11B on FC1 trying to get my analog phone to work, I get a dial tone but when I try to call it from my cell I am getting that in the CLI of * |
20:47.56 | milkyflava | When I try to dial a number it immediatley hangs up and I get a busy signal |
20:49.18 | *** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca) |
20:50.00 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-HSE-ppp168389.qc.sympatico.ca) |
20:50.01 | SarahEmm | hihi |
20:50.05 | Beirdo | heya |
20:50.12 | SarahEmm | hihi Beirdo :o) |
20:50.12 | newmedian | waves to SarahEmm |
20:50.13 | SarahEmm | yarrr. |
20:50.17 | SarahEmm | shido: *pokies* |
20:50.18 | Beirdo | yarr! |
20:50.24 | *** join/#asterisk Assid (~assid@203.115.64.61) |
20:51.25 | *** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-94.dsl.spfdmo.swbell.net) |
20:51.42 | Beirdo | SarahEmm: that game is sooo evil :) |
20:52.44 | funxion | milkyflava: do you have anything entered in the default context of your extensions.conf? |
20:53.07 | SarahEmm | Beirdo: yepyep ;) |
20:53.11 | Beirdo | heh |
20:53.14 | Assid | wheres shido??? |
20:53.20 | Assid | i wanna buy a did man |
20:53.34 | milkyflava | Funxion, I was just looking and no I don't even have [default] just [incoming] I have been trying multiple things to figure this out and must have deleted it |
20:53.38 | Assid | SarahEmm: which service? |
20:53.45 | Beirdo | mine works great. |
20:53.47 | SarahEmm | Assid: trying to sign up with xetricom :) |
20:53.47 | *** join/#asterisk _mwoodj_ (~mwoodj@hyper-eye.digium.sponsor.pdpc) |
20:54.26 | funxion | well from what I can tell your zapata drops you into the default context which is missing |
20:54.45 | funxion | you need to create [default] and add some exten's |
20:55.04 | Assid | where the hell is JerJer anyways |
20:55.11 | Assid | i want a DID |
20:55.15 | milkyflava | funxion, you are exactly correct. I will do that and try again. Thanks so much, I am getting completely lost. :) |
20:55.15 | CoaxD | michigan |
20:55.19 | Assid | is incoming free on michigan did's ? |
20:55.26 | funxion | np |
20:55.27 | CoaxD | assid: Ya |
20:55.28 | file | yes |
20:55.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
20:55.31 | Assid | okay |
20:55.37 | Beirdo | I can attest to that |
20:55.40 | CoaxD | assid: If yer a voip telco and not offering "free" incoming, you' |
20:55.42 | loud | nufone does did's ? didnt know that. |
20:55.44 | CoaxD | you're not gonna last long |
20:55.59 | CoaxD | loud: Hell yea man. They're also the only voip telco that offers free 800 DIDs |
20:56.06 | Assid | they do.. michigan/toll free only |
20:56.07 | Assid | now.. |
20:56.09 | CoaxD | loud: (You just pay for the incoming minutes you use) |
20:56.19 | Assid | so thats not free incoming |
20:56.20 | loud | how cool, do you know if they have cali dids ? |
20:56.28 | CoaxD | assid: Of course not |
20:56.32 | Assid | free incoming -- unlimited incoming |
20:56.32 | Beirdo | Assid: the MI ones are $0/min |
20:56.43 | Assid | okay.. |
20:56.50 | Assid | i wanna purchase a number man.. |
20:56.54 | SarahEmm | well, a few |
20:56.58 | SarahEmm | s'easier said than done ;) |
20:57.01 | Assid | i just wanna do some testing |
20:57.08 | CoaxD | assid: So get one! |
20:57.09 | Assid | wheres JerJer/chido anyways |
20:57.09 | Assid | ? |
20:57.19 | Beirdo | just order it online, no? |
20:57.22 | CoaxD | assid: Your whole damn setup with any of these telcos, the most its ognna cost you is the first months' fee on a DID |
20:57.30 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
20:57.41 | CoaxD | assid: hell, JerJer doesnt even bill me for my MI did. I just use it for checking my voicemail, anyway |
20:57.54 | Beirdo | shh |
20:57.57 | Beirdo | heh |
20:57.58 | SarahEmm | Beirdo: depends which company |
20:58.03 | SarahEmm | Beirdo: not for anyone that offers 416 DIDs |
20:58.04 | Assid | actualy |
20:58.09 | CoaxD | (Then again, i do business with him on other areas, so..) |
20:58.12 | Beirdo | SarahEmm: yeah, sucky |
20:58.12 | SarahEmm | they're all 'email person X' |
20:58.15 | SarahEmm | yeah... |
21:00.19 | Beirdo | but the only VoIP provider I've found in Puerto Rico so far is Verizon or one the other big ones... not really asterisk-friendly. |
21:05.38 | brookshire | got to love how the big telecos don't get it |
21:06.08 | SarahEmm | yeah... |
21:06.25 | Beirdo | yeah, sucketh |
21:06.30 | opus__ | i'm getting call disconnects during calls, like 3 4 minutes into a call |
21:06.38 | opus__ | i'm using alaw |
21:06.44 | opus__ | should I try g729a instead? |
21:07.48 | *** join/#asterisk alerios (~c81e471a@208.195.214.137) |
21:10.52 | alerios | Hi!,.. I've a strange problem (like all of them): most peers can make calls throug a Zap channel, but a couple of peers can't, and they are in the rigth context, and the channel is free. ¿any ideas? |
21:15.15 | *** join/#asterisk harryvv (~noyb@S010600a0c93f6f7e.vs.shawcable.net) |
21:16.31 | astor | can anyone recommend some wireless ip phones? I need to be able to have 2-3 lines operational at the same time. does this preclude the cheap non-wifi systems? |
21:16.31 | *** part/#asterisk Moc (~mochouina@h66-201-214-109.gtconnect.net) |
21:17.23 | SarahEmm | ast_freak: don't get a wifi system. :) |
21:17.45 | SarahEmm | i've heard of few that are good, and i don't know of any wifi ones... you can't use POTS ones with FXS? |
21:17.46 | SarahEmm | +ports |
21:18.43 | astor | SarahEmm: I can, but I thought it could be easier with wifi + I'd save having to buy the FXS stuff. |
21:21.16 | alerios | astor: I have a Zyxel p2000w and is not good |
21:21.52 | alerios | is not much configurable |
21:22.48 | astor | alerios: what kind of configuration do you need? |
21:23.57 | alerios | for example, i couln't make it to send dtmf with rfc2833, icould only sent it inband |
21:24.09 | astor | I worry that some of these phones don't support anything else than WEP security. But maybe that's no worse than what DECT or other digital wireless systems offer. Anyone knows? |
21:25.50 | LoRez | DECT probably doesn't have ready-made tools to crack it easily available via the net with an off the shelf wireless card |
21:25.57 | SarahEmm | astor: you'd think (be easier and save having to buy FXS) but most of the wifi phones Suck :P |
21:26.09 | LoRez | that's kinda run-onish, but I'm sure you get my meaning |
21:28.09 | anti | Anyone have any idea when I try to place calls to my asterisk at home via iax2, I get the message "No authority found" on the calling end, and the called end I get a connection attempt rejected. |
21:28.48 | astor | SarahEmm: so what's the current view on UIP1868 and similar proprietary wireless systems with voip gateways? |
21:29.03 | SarahEmm | astor: i dunno :) |
21:29.11 | SarahEmm | astor: i just relay what i hear here and reading review. |
21:29.19 | SarahEmm | i've never passed voice through * actually heh |
21:29.20 | SarahEmm | just text |
21:29.24 | astor | SarahEmm: that's what I'm looking for :-) |
21:29.42 | astor | SarahEmm: gossip.. :-) |
21:30.33 | astor | the UIP1868 could save me having to buy FXS-stuff, but I'm not sure if I can install two such systems in the same area. |
21:30.34 | SarahEmm | lol |
21:30.38 | SarahEmm | i've not heard any gossip on that yet |
21:31.20 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
21:32.15 | *** join/#asterisk maruk (~maruk@i-194-106-46-242.freedom2surf.net) |
21:32.23 | ender | when using a polycom IP-30[01], and I want both line buttons to be forthe same extension, is it best to just register 2 lines to the same number or is there a better way? |
21:33.00 | astor | alerios: so rfc2833 should be on my list of required features.. |
21:35.47 | alerios | astor: for me, it is |
21:36.02 | *** join/#asterisk Prion (~swschulz@cpe-024-211-202-206.nc.res.rr.com) |
21:36.14 | astor | alerios: I see that the F1000 is supposed to support rfc2833 |
21:43.07 | *** join/#asterisk znoG (~gs@200.115.216.109) |
21:43.19 | pjz | ender: I've got a 500 and I just register the first button and leave the others blank |
21:44.07 | ender | pjz: yeah, I just tried that and that seems better. |
21:44.16 | alerios | astor: gotta go. good luck |
21:44.39 | pjz | ender: I use the other stuff for group-ring stuff so people can tell where they're getting the call from |
21:44.55 | ender | pjz: have you gotten sntp working? My GMT offset isn't being used really. |
21:47.13 | Assid | hrmm how do i roll over a call to athe next free extension? |
21:47.56 | *** join/#asterisk AlexCeli (~Alex@200.37.85.97) |
21:48.20 | Assid | suppose i have 101, 102, 103 extensions.. and if i dial to 101.. it should roll to 102 if 101 is busy.. or 103 .. if 102 and 101 is busy |
21:48.37 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
21:48.51 | Assid | or even cycle like 103 if 102 is busy.. then at 103.. go back to 101.. if 103 is busy |
21:49.01 | *** join/#asterisk cgcorea (~cgcorea@63.245.14.194) |
21:49.24 | SarahEmm | that's just dialplan logic |
21:49.34 | SarahEmm | you've rtfw? |
21:49.43 | harryvv | checking out the *@home example seems that not all nick driver libraries are installed |
21:49.59 | harryvv | Hello SarahEmm |
21:50.06 | SarahEmm | hihi |
21:50.15 | harryvv | :) |
21:50.33 | harryvv | ender is port 123 open |
21:51.00 | Assid | ChanIsAvail(SIP/2001&SIP/3001) |
21:51.02 | Assid | ok |
21:52.22 | abatista | hello everyone |
21:52.29 | SarahEmm | hihi |
21:52.54 | ender | harryvv: open on what? |
21:53.06 | ender | harryvv: it seemed to get the right date, just the wrong time. |
21:53.24 | ender | although I don't recall if it had the right date before. and I can't find a way to manually set the time. |
21:55.05 | *** join/#asterisk |Vulture| (~V@user-0c6tr11.cable.mindspring.com) |
21:55.13 | |Vulture| | Anyone using PoE on the IP500/501? |
21:55.42 | *** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net) |
21:55.48 | *** join/#asterisk DJ-Pyro (~DJ-Pyro@207.250.58.17) |
21:55.57 | harryvv | right now in the process of getting @home setup. |
21:56.00 | pjz | ender: yeah, it looks to me like the sntp implementation won't traverse a gateway, though, so I just set up a server on my pbx |
21:56.10 | harryvv | Vulture, not right now but need a switch |
21:56.10 | ariel_ | quick question. I need to setup asterisk on a via-c3 the wiki does not give any real help on getting it working other then saying it needs a kernel-source select "CyrixIII/VIA-C3" |
21:56.18 | pjz | ender: which is also my dhcp/ftp server |
21:56.41 | ariel_ | is this on basic kernel build option or what. I don't seem to find it on CentOS 3.4/3.5 |
21:56.54 | |Vulture| | harryvv: do they require an different cable? or just a regular RJ45 cable with a 802.3af switch? |
21:56.54 | pjz | |Vulture|: afaict, all IP500s use POE :) |
21:57.10 | |Vulture| | pjz: yea I didn't know if they could accept Netgear POE |
21:57.20 | ender | pjz: ah, that would be why. Hrm, ok. |
21:57.22 | pjz | |Vulture|: oh, I think POE is pretty standard |
21:57.31 | pjz | ender: I ran into the same thing :) |
21:57.38 | |Vulture| | I heard 79XX can't use Netgear's |
21:57.44 | pjz | oh, really? |
21:57.45 | |Vulture| | okay then looks like Ill be getting one |
21:57.51 | harryvv | vulture, I dont think thay use the same cable. |
21:57.52 | pjz | |Vulture|: hrm, well, don't take my word for it |
21:58.21 | pjz | |Vulture|: get a salesrep somewhere to promise that it works |
21:58.23 | |Vulture| | okay |
21:58.29 | |Vulture| | good idea |
21:58.30 | DJ-Pyro | so we're not getting any audio on our ds3 from global crossing, they had a tech note that went out last week specifically about asterisk and no audio being passed, I just upgraded to cvshead and we're still having this issue, anyone know anything about this or why it won't work? |
21:58.36 | harryvv | its pretty easy to seperate DC from the TCP/IP pulses from ether net cable so yes it should be the same. |
21:58.39 | pjz | |Vulture|: then if it breaks you can give it back to them and say 'you promised' |
21:58.57 | |Vulture| | lol |
21:59.30 | SarahEmm | DJ-Pyro: what are you using to interface a DS-3 with *? |
21:59.55 | *** part/#asterisk jfonsecausa (~jfonseca@12.42.141.34) |
22:00.48 | |Vulture| | Sangoma makes a DS3 card |
22:01.16 | mogorman | they have asterisk drivers? |
22:01.21 | |Vulture| | yes |
22:01.24 | DJ-Pyro | SarahEmm: DS3 splits off through an adtran mx2800 into 7 servers, each with a 4 port card in it |
22:01.33 | DJ-Pyro | the card is fine, channels don't alarm, dchannel is up |
22:01.42 | ariel_ | |Vulture|, no it does not use the normal PoE polycom IP-500 |
22:01.48 | mogorman | how many channels can you actually drive in a machine? |
22:01.49 | |Vulture| | ariel_: thank you |
22:01.51 | DJ-Pyro | we see the calls come in, play tt-monkey, get 2 digits, and read back 4 |
22:02.03 | DJ-Pyro | don't hear any audio and don't see the dtmf tones |
22:02.08 | SarahEmm | DJ-Pyro: nice |
22:02.19 | SarahEmm | (well except for that) |
22:02.45 | DJ-Pyro | I know, GC put on a tap in chicago and didn't see any traffic |
22:03.09 | DJ-Pyro | it's also something to note that globalcrossing knows about asterisk, they put out an internal tech notice about issues customers have been having with it and no audio |
22:03.11 | mogorman | i mean vulture, whats the point of ds3 card if your machine can only push 200 or so channels |
22:04.11 | *** join/#asterisk JonUK (~me@81-178-16-14.dsl.pipex.com) |
22:04.19 | |Vulture| | mogorman: yea I know thats why DJ-Pyro's setup makes more sense |
22:04.37 | JonUK | hi everyone, anyone care to offer any advice on what ISDN BRI card works best with * |
22:05.19 | JonUK | I have been looking at the AVM and Eicon cards, but cannot see much difference between them |
22:06.01 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
22:06.53 | JonUK | no one?? |
22:06.55 | SarahEmm | blitzrage: you here? |
22:06.59 | SarahEmm | JonUK: i have no idea, sorry |
22:07.04 | SarahEmm | shido: mew? |
22:07.33 | milkyflava | I just got my first phone call to work into my * box to my analog phone! |
22:07.50 | JonUK | well done milkyflava |
22:08.07 | milkyflava | Thanks JonUK! |
22:08.55 | shmaltz | anybody here knows what happens if A receives a call from B and B transfers A to C using the transfer button on their phone (attd xfer), what shows up in the CDR? under channel will it show B or A? |
22:09.10 | shmaltz | I know I can test it but if someone else did the test already why should I bother |
22:09.32 | JonUK | shmaltz, cant answer, sorry |
22:10.10 | DJ-Pyro | anyone have any insight? |
22:11.28 | JonUK | shmaltz, I would think that nothing would be shown in CDR, as the phone is handling the transfer to the Destination C |
22:12.09 | shmaltz | JonUK, well what you think and what * thinks are 2 different thigs |
22:12.18 | shmaltz | thigs = things |
22:12.27 | JonUK | ok, was just a guess ! |
22:12.31 | DJ-Pyro | http://pastebin.ca/16979 |
22:12.38 | DJ-Pyro | that's what the console says |
22:12.43 | DJ-Pyro | I don't see what we're doing wrong |
22:12.59 | Inv_arp | ariel_: should work fine since its x86 |
22:15.42 | ariel_ | Inv_arp, it's not working fine. I can't get asterisk to run on it. |
22:15.47 | shmaltz | JonUK, were u under the impression that my guessing talent is not as good as yours? |
22:16.37 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
22:17.32 | ariel_ | Inv_arp, I found some emails that said you need to make sure you compile it for Cyrii/via-c3 But I can't seem to find the option. |
22:20.10 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
22:20.49 | puzzled | ariel_: i think someone on the list mentioned compiling for i586 will work |
22:21.47 | ariel_ | puzzled, what is the option for that? I have had one of my worst days today and I am not thinking correctly. |
22:22.19 | pjz | ariel_: compile it for pentium architecture |
22:22.23 | puzzled | ariel_: you are talking about compiling * right? |
22:22.33 | *** join/#asterisk Dan_K (dan@c-24-8-35-8.hsd1.co.comcast.net) |
22:22.56 | ariel_ | puzzled, yes or do I need to set the kernel up first differently? |
22:23.24 | puzzled | ariel_: if the kernel runs fine on the box I guess there is no need to change the kernel |
22:23.36 | puzzled | ariel_: try editing PROC in the asterisk Makefile |
22:23.41 | ariel_ | puzzled, I keep getting asterisk unable to load. |
22:24.41 | ariel_ | puzzled, I will do that. My head is just not in it. I am going back to the hospital.. see you all. |
22:24.48 | Dan_K | Hello, I am a new asterisk user. I signed up with broadvoice and their minimalist byod plan so I could play with a real voip connection. However when trying to dial out from asterisk, it seems to ignore my fromuser and fromdomain directives in the sip.conf for broadvoice |
22:25.10 | Dan_K | Is there anything I need to look at specifically? |
22:25.16 | puzzled | ariel_: in my asterisk Makefile on line 81 it says: #PROC=i586. enable that and try again |
22:25.20 | *** join/#asterisk gbdrbob (drbob@alltalk.demon.co.uk) |
22:25.36 | *** part/#asterisk gbdrbob (drbob@alltalk.demon.co.uk) |
22:29.14 | Dan_K | ah, nevermind, I figured it out. helps if I name the section correctly in sip.conf and extensions.conf |
22:30.09 | SarahEmm | shido? blitzrage? |
22:31.11 | blitzrage | SarahEmm: yes? |
22:33.12 | pjz | how do I prepend something to the CID string in stable? |
22:33.28 | pjz | what's the function call in asterisk? |
22:33.34 | blitzrage | ${CALLERID}this_is_the_string I believe |
22:33.34 | pjz | er, in extensions.conf |
22:33.55 | pjz | but how do I set it? |
22:34.07 | pjz | something like SetCallerID() or something? |
22:34.17 | blitzrage | yes |
22:34.42 | pjz | so SetCallerID(FOO: ${CALLERID}) ? |
22:35.37 | *** join/#asterisk SarahEmm (~sarahemm_@MTL-HSE-ppp168389.qc.sympatico.ca) |
22:35.40 | SarahEmm | wheee wifi-- |
22:36.41 | blitzrage | pjz: believe so - give it shot |
22:37.26 | tzanger | mmmmmmmmmm stale nonce... my favourite |
22:37.50 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
22:37.51 | blitzrage | tasty |
22:42.00 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
22:42.18 | dca[laptop] | hey anthm, you around? |
22:42.49 | anthm | yes |
22:43.18 | blitzrage | SarahEmm: were you looking for me? |
22:43.31 | tzanger | my blackfin books came today :-) |
22:43.53 | dca[laptop] | anthm: has any else reported problems with ChanSpy cuasing seg faults? |
22:44.00 | Assid | shido ??? |
22:44.02 | Assid | did? |
22:44.17 | tzanger | Assid: does the channel really need to know about your issues? :-/ |
22:44.38 | Assid | hehe |
22:44.40 | Assid | sorry |
22:44.52 | anthm | not that I know of |
22:45.47 | SarahEmm | blitzrage: err.... yes, but now i can't remember why :) Sowwy |
22:45.58 | SarahEmm | Assid: shido seems to not be around.. i'm looking for him too heh |
22:46.01 | SarahEmm | Assid: wondering what's up with my DID :) |
22:46.33 | SarahEmm | ooh right |
22:46.37 | SarahEmm | blitzrage: ETA on just a 416 DID? |
22:48.15 | Assid | you bought one? |
22:48.30 | SarahEmm | Assid: from shido? i'm trying to |
22:48.50 | SarahEmm | Assid: i have no idea where things are tho =-/ |
22:49.07 | SarahEmm | he said 416 DIDs should be quick, but i'm not finding that heh |
22:49.10 | blitzrage | SarahEmm: ummm... like 1-2 weeks I think to order |
22:49.27 | Assid | how the hell can i get a did of only 9 digits? |
22:49.34 | SarahEmm | Assid: huh? |
22:49.34 | Assid | are there 9 digit numbers? |
22:49.36 | SarahEmm | 9?? |
22:49.39 | Assid | yeah |
22:49.42 | Assid | thats what i got |
22:49.43 | blitzrage | seems wrong |
22:49.45 | SarahEmm | Assid: oh. where in the world are you? |
22:49.51 | SarahEmm | blitzrage: uhh... 1-2weeks for a 416? |
22:50.00 | SarahEmm | that seems long |
22:50.08 | SarahEmm | Assid: if you're in NA that's wrong... |
22:50.27 | dca[laptop] | anthm: did you catch that last question? |
22:50.40 | blitzrage | SarahEmm: yah.. I think thats what I've been told, although I've found it seems to be closer to 4-5 days |
22:51.43 | SarahEmm | hrm. |
22:51.45 | SarahEmm | okay.... |
22:52.13 | opus__ | cool i got g729 to work |
22:52.53 | Assid | hrmm time to start packing up.. |
22:52.56 | Assid | 4.30 am |
22:53.23 | pjz | No application 'SetCallerID(TOC: ${CALLERID}' |
22:53.33 | opus__ | missing ) |
22:54.04 | Assid | gnight folks |
22:56.28 | pjz | okay, so why didn't the CID change work? |
22:57.57 | anthm | yah my answer was, not that I know of |
22:58.44 | *** part/#asterisk mkrufky (~mk@68.160.103.77) |
22:58.54 | blitzrage | pjz: missing ) |
22:59.02 | pjz | I see the SetCallerID() execute in the stack |
22:59.05 | pjz | blitzrage: fixed that |
22:59.17 | pjz | but my phone doesn't show the modified CID |
23:02.14 | dca[laptop] | anthm, sorry didn't see it, tks guess something is up with my box then |
23:03.04 | *** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt) |
23:08.30 | *** join/#asterisk WeezeyD (~ohno@206.210.109.226) |
23:08.42 | WeezeyD | anyone have a TDMoE set up between two asterisk boxes? |
23:08.50 | key2 | when someone calls me on my zap, is {exten} the callerid ? |
23:08.52 | *** part/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
23:09.12 | *** join/#asterisk psywar (psywar@rasterburn.org) |
23:09.26 | psywar | anyone have problems with a SPA-2000 throwing the occasional bad UDP checksum? |
23:09.42 | pjz | ohhh.. is there a max length to CID? |
23:09.43 | psywar | it terminates my calls, really annoying |
23:10.15 | WeezeyD | my SPA-3000s always gives that checksum thing |
23:10.51 | psywar | does it terminate your calls and annoy you greatly? |
23:10.59 | WeezeyD | no |
23:11.03 | psywar | hrm |
23:11.08 | WeezeyD | it just gives that error |
23:11.19 | psywar | I guess I need to do more correlation. |
23:11.28 | psywar | maybe the two events aren't correlated. |
23:11.30 | *** join/#asterisk pingywon (~mike@pcp0010034410pcs.reding01.pa.comcast.net) |
23:12.35 | psywar | I'm only doing VoIP over about 1m of cat5 cable, dunno why I'm getting so many abrupt terminations |
23:12.35 | psywar | seems to happen in bursts, and only with certain callers |
23:12.35 | infinity1 | i see a volume voicemail patch. is that available in the main code? |
23:12.35 | WeezeyD | maybe a bad switch or cable? |
23:12.36 | psywar | could be cable, since I make them myself |
23:12.42 | psywar | I doubt it though |
23:14.18 | *** join/#asterisk AgiNamu (~bob@200.6.217.202) |
23:14.28 | AgiNamu | Hey, does SIP md5secret for users work with non-Asterisk users? |
23:19.49 | *** join/#asterisk Chotaire (chotaire@chotaire.net) |
23:19.56 | Chotaire | good morning vietnam. |
23:20.37 | SarahEmm | hi Chotaire |
23:21.20 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
23:22.30 | terrapen | anybody use Cingular here? |
23:22.42 | terrapen | gosh, i know nothing about mobile phones these days |
23:23.01 | greg_work | JerJer: still around? disabled everything in bios, now my interrupts look like this: http://pastebin.ca/16992 |
23:24.10 | greg_work | weird thing though.. i had been using only ports 1, 3 and 4 of my TDM400P (all fxo).. forget exactly why, but I was having a problem with port 2 not answering or something.. anyway, just tried to go back to using 1, 2 and 3, and port 2 seems to be permenantly offhook |
23:25.43 | terrapen | cingular has this EDGE thing |
23:25.49 | terrapen | supposedly quite fast |
23:26.04 | terrapen | dunno anybody who uses it |
23:26.09 | WeezeyD | g'damn. CallerID doesn't seem to be working over TDMoE |
23:28.22 | *** join/#asterisk josephcool (~asterisk@adsl-67-121-209-208.dsl.sndg02.pacbell.net) |
23:30.03 | josephcool | I have a problem. When I do a Playback(custom/aa_1) I don't get playback, but if I do Playback(/var/lib/asterisk/sounds/custom/aa_1) it will work. How do I setup my default directory for this? |
23:30.05 | niZon | has anyone used a handytone 488? |
23:30.45 | josephcool | I haven't but I love this Sipura SPA-841... |
23:30.52 | niZon | hmm |
23:30.59 | *** join/#asterisk _DAW (~bob@68-114-110-210.dhcp.slid.la.charter.com) |
23:31.18 | josephcool | I don't know how to set the directory correct so asterisk will see the sound files. |
23:31.31 | niZon | i want a cisco 7940/60 I have some uses for the XML features... |
23:31.46 | niZon | but right now I want to know some stuff about the ht488 |
23:32.10 | josephcool | Do you know anything about how asterisk directories works? |
23:33.08 | niZon | my directory is /usr/share/asterisk/sounds/ |
23:33.16 | niZon | i dunno how to change it |
23:33.38 | josephcool | hmmm |
23:33.40 | *** join/#asterisk JunK-Y (~junky@69.156.216.243) |
23:33.40 | infinity1 | ~voicemailvolume |
23:34.18 | josephcool | I BET That is the problem... |
23:34.20 | pjz | josephcool: afaik that's hardcoded somewhere |
23:34.22 | josephcool | How do you change it? |
23:34.33 | josephcool | oh... hardcoded? guh. |
23:34.36 | josephcool | ugh even. |
23:34.39 | niZon | do a "locate sounds" or "find /*|grep sounds" |
23:34.41 | niZon | *shrug* |
23:34.54 | _DAW | josephcool - you could move it with sim link |
23:35.35 | josephcool | no, I see that... holy smoke, I get it now... the problem is I installed it apt-get... that must make the /usr/share sounds directory, but AMP creates and tries to use /var/lib/asterisk/sounds... |
23:36.02 | pjz | josephcool: I'd suggest a symlink |
23:36.09 | josephcool | instead of a recompile? |
23:36.12 | pjz | josephcool: yeah |
23:36.13 | *** join/#asterisk cfrank_ (~cfrank@bi01p1.co.us.ibm.com) |
23:36.14 | WeezeyD | ~tdmoe |
23:36.14 | jbot | hmm... tdmoe is for a situation where you need TDM reliability without traditional TDM hardware. TDMoE is useful because it allows the above familiarity, flexibility, and reliability of TDM but over inexpensive Ethernet instead of T1s or E1s. |
23:36.17 | josephcool | True... |
23:36.18 | infinity1 | anyone know anything about the voicemail volume problems? is this some kind of sick joke? |
23:36.22 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
23:36.30 | josephcool | ok, how do I create a simlink? |
23:36.34 | SarahEmm | what problem infinity1? |
23:36.38 | WeezeyD | ln -s |
23:36.41 | SarahEmm | (i'm just curious, i have no idea what the issue is or a fix btw |
23:36.44 | infinity1 | SarahEmm: http://lists.digium.com/pipermail/asterisk-users/2005-June/113872.html |
23:36.44 | SarahEmm | shido: mewmewmew? |
23:37.10 | file[laptop] | SarahEmm: KITRICH! |
23:37.10 | infinity1 | SarahEmm: from what i'm reading voicemail/zap is not possible. |
23:37.12 | WeezeyD | ~callerid |
23:37.24 | *** join/#asterisk tinch0 (morey@84.77.99.140) |
23:37.39 | WeezeyD | infinity1: sending Zap to voicemail? works fine. |
23:37.41 | SarahEmm | hihi file[laptop] |
23:37.51 | SarahEmm | infinity1: i don't beleive that... |
23:37.53 | tinch0 | hi |
23:37.54 | SarahEmm | infinity1: lots of people do that. |
23:38.01 | infinity1 | SarahEmm: i feel betrayed. i spent many days on this * stuff and its broke. |
23:38.51 | infinity1 | SarahEmm: ...and it seems no one cares. i don't get it. there must be more to this. |
23:38.58 | tinch0 | how is the right syntax for Dial/SIP if I want to use a certein account at a proxy server? |
23:38.58 | SarahEmm | infinity1: well, i'm sure zap and voicemail is possible. a ton of people do it |
23:39.05 | SarahEmm | you're getting low volume, infinity1? just in voicemail, no other * stuff? |
23:39.07 | niZon | I want * to ring certain voip extensions when a call comes in via PSTN on the HT488, but I want the HT-488 to keep the PSTN line on hook until I pick up a voip extension, anyone tried this? |
23:39.20 | ManxPower | infinity1: that voicemail volume problem only happens with one of the formats |
23:39.45 | infinity1 | ManxPower: what format should voicemail be using? |
23:39.52 | ManxPower | infinity1: Try gsm |
23:40.01 | infinity1 | ManxPower: i read the bug report on it. and has a LOT of comments |
23:40.06 | tinch0 | similar to IAX syntax I think... anybody? |
23:40.09 | ManxPower | infinity1: Yup. |
23:40.40 | WeezeyD | infinity1: mine works fine, I use wav49 and gsm, wav49 for email attachments. |
23:41.01 | infinity1 | WeezeyD: using zap as a receiving interface? |
23:41.06 | WeezeyD | yep |
23:41.21 | infinity1 | WeezeyD: i just got a VM in my email. i couldn't hear shit. |
23:41.27 | infinity1 | WeezeyD: i'll try some other formats. |
23:43.06 | josephcool | So how do I create a symlink that will re-point the /usr/share/asterisk directory to /var/lib/asterisk like amp wants. and will that take everything recursive? |
23:43.59 | ManxPower | infinity1: What format ARE you using? WAV has known problems |
23:44.13 | pcm | ttvdy is a special protocol that does nothing |
23:44.18 | pcm | ~ttvdy |
23:44.47 | pcm | hmmm ... though the bot was learning from the channel ? |
23:45.09 | SarahEmm | ttvdy? what's that? |
23:45.18 | SarahEmm | pcm: it is, but you have to prefix it with jbot: |
23:45.19 | ManxPower | jbot, ttvdy is a special protocol that does nothing |
23:45.19 | jbot | ManxPower: okay |
23:45.33 | infinity1 | ManxPower: mine was set to format=wav49|gsm|wav |
23:45.40 | ManxPower | jbot, ttvdy is a special protocol invented by the CIA. Now I have to kill you. |
23:45.40 | jbot | ...but ttvdy is already something else... |
23:45.45 | ManxPower | jbot, no, ttvdy is a special protocol invented by the CIA. Now I have to kill you. |
23:45.45 | jbot | okay, ManxPower |
23:45.51 | infinity1 | ManxPower: i changed it to wav, from reading the bug report, i thought that was best. |
23:45.58 | ManxPower | infinity1: And what do you use each of those formats for? |
23:46.04 | pcm | ~ttvdy |
23:46.04 | jbot | rumour has it, ttvdy is a special protocol invented by the CIA. Now I have to kill you. |
23:46.05 | ManxPower | There should be a REASON for each format. |
23:46.09 | infinity1 | ManxPower: that was the default |
23:46.28 | ManxPower | We all know Asterisk's defaults suck. |
23:46.36 | *** join/#asterisk outtolunc (outtolunc@adsl-69-110-52-142.dsl.pltn13.pacbell.net) |
23:46.46 | infinity1 | ManxPower: i just started using this. i just got a call over one zap, and it bridged to my cordless phone (zap). there was echo on my end. weird. |
23:47.05 | *** join/#asterisk rv_weasel (~no@adsl-68-93-10-206.dsl.ksc2mo.swbell.net) |
23:47.06 | infinity1 | ManxPower: so you recommend gsm for the format? |
23:47.19 | SarahEmm | pcm: what *is* ttvdy? |
23:47.33 | infinity1 | ManxPower: any idea whats up with the echo? |
23:47.46 | rv_weasel | ok, i need a good sip client for linux. one with console features would rock too |
23:47.54 | pcm | sarahemm: it's nothing :) |
23:48.04 | ManxPower | infinity1: Yes. |
23:48.15 | rv_weasel | I got my system all setup though!!! everything is workign great |
23:48.25 | *** join/#asterisk eng1neer (1000@adsl-68-94-42-172.dsl.rcsntx.swbell.net) |
23:48.26 | ManxPower | Echo is caused by using only 2 wires for tx and rx. |
23:48.28 | *** join/#asterisk jpablo (~jpablo@201.138.154.114) |
23:48.40 | infinity1 | ManxPower: so i need a special cable? |
23:48.41 | ManxPower | infinity1: Don't try to run before you learn to walk, grasshopper. |
23:48.42 | jpablo | hi, anyone here has an tdm in production ? |
23:48.44 | jpablo | with 4 fxo modules. |
23:48.45 | rv_weasel | routing all my calls out through the net at 1.1 cent/min |
23:48.57 | ManxPower | infinity1: no, you need to balance your volume levels and turn on echo can |
23:48.59 | infinity1 | ManxPower: what do you mean? i just got a working dial plan |
23:49.03 | josephcool | So I can't see where I would change the directories from /usr/share/asterisk to /var/lib/asterisk Anyone know how to change this? |
23:49.20 | josephcool | I understand I can symlink it... but how does it normally setup? |
23:49.23 | *** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com) |
23:49.25 | ManxPower | infinity1: Exactly. You just got a working dialplan and now you are asking about echo, one of the hardest things in telecom to get rid of. |
23:49.41 | infinity1 | ManxPower: ouch. |
23:49.42 | jpablo | I'm having LOTS of problems, with it, the fxo lines just hang or something, it says they are Onhook, when they are idle and they don't work for anything. |
23:49.48 | *** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net) |
23:50.01 | infinity1 | ManxPower: i just dialed someone through the pbx (zap/zap) and there was no echo. weird. |
23:50.23 | ManxPower | That's because the audio path delay in zap<->zap is not enough for you to HEAR the echo. |
23:50.31 | *** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-7-173.w81-248.abo.wanadoo.fr) |
23:51.39 | josephcool | Anyone know how I switch directories for asterisk? |
23:51.46 | ManxPower | all this stuff is covered over and over and over and over and over and over and over and over and over again in the mailing list archives. |
23:51.53 | rv_weasel | i was wondering about echo cancelation. it seems the digital ends always have echo. i need to balance the analog ends rx and tx gain to get rid of that dont i? |
23:51.57 | ManxPower | josephcool: "cd directory" |
23:51.57 | infinity1 | ManxPower: err ..when someone called me zap/zap, i heard the echo. is that different then when i initiate the call |
23:52.01 | josephcool | lol manx... |
23:52.06 | infinity1 | ManxPower: k |
23:52.18 | infinity1 | ManxPower: i somehow got on the topic. i didn't do any research yet. |
23:52.21 | josephcool | I mean to change from /usr/share/asterisk to /var/lib/asterisk like AMP wants. |
23:52.24 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp09940120pcs.roylok01.mi.comcast.net) |
23:52.33 | infinity1 | ManxPower: so i'll just leave the VM to gsm, see how it goes, and google on the echo thing. |
23:52.35 | ManxPower | josephcool: I'm sorry, I cannot help you with that. |
23:52.39 | infinity1 | ManxPower: thanks :) |
23:53.00 | MikeJ[Laptop] | googleit |
23:54.03 | rv_weasel | now here is a question. if i got a DID account with IAX i could forward my single landline to that and make a cheap trunk |
23:59.33 | jpablo | is they anyway to tell a tdm to use other irq ? |