| 00:01.40 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:01.49 | *** part/#asterisk sfChrisJacob (n=cjacob@70-32-0-48.frdrmd.adelphia.net) |
| 00:02.02 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:05.22 | Hymie | do you guys know if there are any voip phones with a simple hub or switch in them? I know it's silly, but running an extra drop of network cable to each desk, just for a phone, is a PITA |
| 00:05.33 | Qwell[] | Hymie: Most phones do |
| 00:05.43 | MRH2 | most hardphones |
| 00:05.45 | Hymie | Qwell[]: really? that's convienent |
| 00:06.01 | Hymie | do you guys know if the cisco 79xx series does, or the polycoms do? |
| 00:06.07 | Qwell[] | 79xx should |
| 00:06.09 | MRH2 | yes |
| 00:06.13 | MRH2 | both |
| 00:06.14 | Qwell[] | I don't know about the polycoms |
| 00:06.16 | DarthClue | ip501 does, and the 79xx should |
| 00:06.17 | Hymie | cool |
| 00:06.32 | Hymie | I guess the low end units may not have such things |
| 00:06.34 | Hymie | to cut down on cost |
| 00:06.41 | Hymie | from no-name brand companies and such |
| 00:06.57 | MstlyHrmls | all the polycoms do, except the IP 4000 conference phone |
| 00:07.24 | Hymie | thanks ;) |
| 00:07.37 | rajiv | my crappy low end ip phone with no hold button even has 2 eth jacks |
| 00:07.49 | Hymie | hehe |
| 00:08.27 | rajiv | though watch out for the spa-942. i hear that the 2nd jack is only 10mb |
| 00:08.46 | *** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 00:09.03 | Hymie | hmm, thanks |
| 00:09.20 | troyb1 | hey guys when i receive calls through my DID it terminates at my phone instead of [mainmenu] can you help me out? |
| 00:10.33 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 00:11.08 | *** join/#asterisk Soul (n=Soul@87-196-46-44.net.novis.pt) |
| 00:13.34 | rayvd | TEE HEE. |
| 00:14.32 | rayvd | k cut that out. |
| 00:14.41 | troyb1 | ;_; |
| 00:16.24 | FLeiXiuS | Is there any way to map outside calls to not have to dial 9 before dialing out. |
| 00:16.52 | FLeiXiuS | perhaps when asterisk recieves a 10 digit number, it'll dial outbound rather then locally. |
| 00:19.16 | MRH2 | use 0 to indicate an external call |
| 00:19.38 | FLeiXiuS | 0 + NUMBER? |
| 00:20.05 | Qwell[] | that would be pointless |
| 00:20.49 | FLeiXiuS | Qwell[]: I'm sorry, what? |
| 00:21.16 | Qwell[] | using 0 instead of 9 |
| 00:21.19 | Qwell[] | it defeats the purpose |
| 00:21.43 | FLeiXiuS | Oh yeah, absolutely...that'd be an indefinite solutiont to what already have |
| 00:21.46 | MRH2 | depends where ur... in the uk all numbres start with a zero 0800xxyyz |
| 00:22.59 | Qwell[] | it's possible, but all internal extensions would require you to wait |
| 00:23.07 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 00:23.14 | Qwell[] | that's why people use 9 |
| 00:24.12 | delmar | FLeiXiuS, i do exactly what u are asking in my dialplan... |
| 00:24.45 | delmar | FLeiXiuS, here normal dialing is 0xxxxxxxx for national.. and 00etc for international. |
| 00:25.03 | delmar | FLeiXiuS, so not only does 9 or 1 work... but plain dialing works too. |
| 00:25.11 | MRH2 | in the US do u just dial "3053248811" for example? |
| 00:25.54 | rayvd | you generally prefix a long distance number with a 1 |
| 00:26.36 | rayvd | in some cities though, like phoenix, az, there are two area codes, 602 and 480. if you're a 480 number you don't need to dial 1-602-xxx-xxxx, just 602-xxx-xxxx :) |
| 00:27.07 | MRH2 | ah different strokes |
| 00:27.12 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:28.46 | delmar | FLeiXiuS, you got it figured or anna see the couple of lines I have in my dialplan? |
| 00:29.06 | Mavvie | nothing is more annoying than being on hold on an international line |
| 00:29.25 | *** part/#asterisk Utah_Dave (n=boucha@0-3pool128-60.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 00:30.10 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
| 00:30.23 | Psykick | justinu: are you here? |
| 00:30.54 | FLeiXiuS | delmar: yes if you could please post it |
| 00:31.06 | FLeiXiuS | I've been trying to figure it out without using _9 |
| 00:31.30 | Qwell[] | FLeiXiuS: A non-US dialplan won't help you much |
| 00:31.33 | delmar | FLeiXiuS, leave _1 or _9 there anyway.. for plebs that try to dial 9 for outside line... |
| 00:31.48 | delmar | FLeiXiuS, but my plain dialing catch lines.. look something like.... |
| 00:31.52 | delmar | National |
| 00:31.52 | delmar | exten => _0N.,1,Dial(IAX2/user@provider/64${EXTEN:1},,rT) |
| 00:31.52 | delmar | International |
| 00:31.52 | delmar | exten => _00.,1,Dial(IAX2/user@provider/${EXTEN:2},,rT) |
| 00:32.05 | Qwell[] | like I said...doesn't help much :p |
| 00:32.19 | FLeiXiuS | Yeah, international..bleh..thanks anyways! |
| 00:32.23 | delmar | yeah a non-US thingie like mine .. might now. |
| 00:32.34 | delmar | might not* |
| 00:32.48 | delmar | perhaps we have it better here :P |
| 00:32.58 | Qwell[] | FLeiXiuS: You either need a dialout prefix (such as 9), or every internal extension will have to wait after dialing 4-5 digits |
| 00:33.13 | delmar | _0N. matches quite well N=2-9. there is no 1 in our area codes. |
| 00:33.15 | Qwell[] | so...wait 5 seconds, or press 9 |
| 00:34.07 | Qwell[] | or hardcode every areacode in the US/Canada, and change your internal extensions every time a new one is added |
| 00:34.12 | Qwell[] | same with prefixes in your areacode |
| 00:34.12 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 00:34.47 | Qwell[] | "sorry John, we have to take 6267 away from you, because 626 is a new areacode" |
| 00:34.58 | MRH2 | isn't there some area code used in the movies that isn't real? (just out of interest) |
| 00:35.00 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-10-165.cybersurf.com) |
| 00:35.11 | Qwell[] | 555 |
| 00:35.18 | Qwell[] | but that leaves...10 internal extensions |
| 00:35.22 | Qwell[] | if you use 4 digits |
| 00:36.47 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 00:38.11 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
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| 00:38.36 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 00:38.56 | delmar | is a TDM400 PCI hard hot-swap cabable? :P |
| 00:39.59 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 00:40.00 | delmar | heh. ive had network cards pop out (because someone didnt screw it it) while the system was up and running... and put it back in and continued running... so its an interesting thought :P |
| 00:40.37 | delmar | i better not try i guess. better shutdown my box. (sigh). |
| 00:40.42 | rayvd | test it out. i'm sure digium will replace it under warranty if you tell them the reason it died ;) |
| 00:41.01 | delmar | rayvd, that was my main thought :P |
| 00:41.20 | Qwell[] | :D |
| 00:41.21 | delmar | as it is I think its faulty for some unknown reason.... |
| 00:41.44 | delmar | ill run it past u all in case someone new here didnt see my fault description 6+hrs ago. ... |
| 00:42.31 | delmar | during calls... the TDM400 card seems to be making little chrip/beep noises.. quite randomly both in occurance and volume... sounds like u are in a supermarket checkout line on a quiet day.. sorta thing. |
| 00:42.59 | delmar | I didn't notice this happening a few months ago... so it's something new. |
| 00:43.13 | delmar | im about to try the card in a new PC to rule out the PC hardware. |
| 00:43.15 | rayvd | have you ruled out being in a supermarket checkout line? |
| 00:43.37 | delmar | rayvd, yep. no high-school gurls wearing supermarket uniform on my line.. im sad to say. |
| 00:43.58 | rayvd | :D that sounds like a good thing to try. |
| 00:44.15 | MRH2 | supermarket uniform? sexy? |
| 00:44.41 | delmar | rayvd, yep ill try another PC.. then ill try another phone line somewhere else... then I will submit all that and request a replacement card. |
| 00:45.10 | delmar | MRH2, depends on the dress code :P |
| 00:45.27 | Psykick | does anyone know if I should be worried if I'm seeing something like this (IAX Packet 131864 from circuit ids 228->18conflicts with earlier call with circuit ids 191->18) |
| 00:45.47 | MRH2 | <--clearly has 2 learn to shop at a better class of supermarket ;) |
| 00:46.08 | rayvd | hahaha |
| 00:46.18 | rayvd | .. or a worse class |
| 00:50.45 | *** join/#asterisk ComputerWarm (n=workingg@HS196-230-97.nt.net) |
| 00:54.06 | jesster | can anyone tell me if any Soekris box can handle 50 concurrent G711 calls |
| 00:54.14 | jesster | over SIP |
| 00:55.20 | delmar | jesster, whats the specs on the box? |
| 00:55.52 | jesster | 266mhz 586class cpu |
| 00:56.11 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 00:56.11 | delmar | jesster, Soekris means little to me. might as well be a toaster. |
| 00:56.14 | jesster | Soekris net4801 |
| 00:56.21 | jesster | ah, popular appliance |
| 00:56.25 | delmar | 266mhz what kind? intel/amd/cyrix/risc thing? |
| 00:56.49 | delmar | im thinking 50 calls might be pushing it there dude. |
| 00:57.28 | jesster | im sure the answer is puretly mathematical, just not sure and don't have resources to try it ;) |
| 00:58.09 | delmar | when more powerful hardware is quite cheap I wouldn't be paying for one of those to put into a commercial situation expecting 50 concurrent calls. |
| 00:58.41 | delmar | jesster, will there be any transcoding at all? |
| 00:58.49 | jesster | delmar: no |
| 00:58.50 | Psykick | I think for at least 50 concurrent calls you'll need something at least 800Mhz - 1Ghz |
| 00:58.58 | delmar | jesster, not in the slightest? |
| 00:59.14 | Psykick | actually ... a 500Mhz should be ok |
| 00:59.16 | MRH2 | SIP->SIP only? or using Zap |
| 00:59.25 | jesster | delmar: nope |
| 00:59.27 | Psykick | since there will be no transcoding |
| 00:59.33 | jesster | 100% sip |
| 00:59.39 | delmar | Psykick, probably not that high. 20x g729 calls working fine on an Intel 550Mhz box... and ulaw is much less cpu intensive. |
| 00:59.58 | MRH2 | might do it with reinvites |
| 01:00.20 | Psykick | I prefer having that little bit of extra power especially if your running scheduled tasks |
| 01:00.34 | delmar | jesster, im thinking 50 ulaw calls wouldnt be bad on a 550Mhz or better. but if it were me i wouldnt want to push 50ulaw of calls on much less than that. |
| 01:01.08 | delmar | Psykick has a godo point there jesster. u would need to be careful about other system tasks and when they run. |
| 01:01.34 | jesster | delmar: it runs on astlinux, made for embedded appliances |
| 01:01.38 | delmar | jesster, u should be able to pickup an intel 1gig deal cheap enuf. |
| 01:01.56 | jesster | delmar: a soekris mobo w/ case is about 300 total (tops) |
| 01:01.57 | Psykick | could always go for an EPIA board |
| 01:02.02 | *** join/#asterisk [hC] (n=hardcore@209.153.195.139) |
| 01:02.25 | Psykick | can mount the thing in a old video player |
| 01:02.35 | delmar | heh. lol |
| 01:02.35 | Psykick | only uses 12v |
| 01:02.47 | delmar | now that.. is shoe string budget. :P |
| 01:02.51 | Psykick | so don't need to worry too much about over heating |
| 01:02.52 | Psykick | ;) |
| 01:02.58 | delmar | bbs. hardware changes. |
| 01:05.13 | MRH2 | anyone know how to do a svn checkout of a certain date like with cvs? |
| 01:05.47 | Psykick | jester: I'm lookin at using these soon (http://www.via.com.tw/en/products/mainboards/mini_itx/epia_tc/index.jsp) |
| 01:06.54 | Psykick | and I'll probably get the CF card adapter and load asterisk onto that |
| 01:07.06 | jesster | ok |
| 01:07.38 | Psykick | nice ... quite, small and overall pretty cost effective |
| 01:08.29 | [hC] | hey i just got one of those epia's |
| 01:08.31 | [hC] | gotta play with it |
| 01:08.39 | MRH2 | I think i need to role back to yesterday to get a working copy |
| 01:09.35 | Psykick | [hC]: I love em ... |
| 01:10.00 | MRH2 | ah found it finally |
| 01:16.00 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 01:16.34 | *** join/#asterisk delmar (n=Delmar@203-114-178-231.inspire.net.nz) |
| 01:21.34 | [hC] | So anyone here using a cisco 7970? I have a couple operational questions.. |
| 01:23.29 | *** join/#asterisk doushanes (n=dwright@71.194.48.241) |
| 01:25.11 | delmar | with all the government departments and tv shows using cisco phones on their desks.. u would think they would start to pop up on ebay quite inexpensive by now. |
| 01:25.39 | delmar | if love to get a cisco phone to play with and see if they really are worth the extra $. |
| 01:25.54 | *** join/#asterisk _deg_ (n=deg@201.22.31.111.adsl.gvt.net.br) |
| 01:26.20 | doushanes | delmar: 24 comes to mind when you say that :) |
| 01:26.20 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 01:27.18 | delmar | doughecka, yeah they thrash cisco hardware there for sure.. but its not too far fetched either... everywhere I look government have cisco phones. |
| 01:27.35 | delmar | doughecka, there are plenty of other shows where u see them too. |
| 01:27.47 | delmar | doughecka, altho they are probably the same phone props from the 24 set :P |
| 01:27.55 | doughecka | ? |
| 01:28.13 | mzo | um, speaking as someone who actually works in film. |
| 01:28.13 | delmar | doughecka, my bad. :P nick completion error :P |
| 01:28.16 | doughecka | woot |
| 01:28.22 | mzo | the cisco phones we get for prop use are fakes. |
| 01:28.26 | mzo | dummies with no insides. |
| 01:28.30 | delmar | mzo, oh really? damn |
| 01:28.33 | mzo | yeah. |
| 01:28.37 | mzo | they are like $18 bucks each |
| 01:28.39 | delmar | mzo, the 24 ones are fake? |
| 01:28.39 | doughecka | the island had a future cisco video phone :D |
| 01:28.49 | mzo | usually they wire them up with fake lights inside if they need them for on-screen use |
| 01:28.50 | delmar | mzo, the 24 ones have nice lcd, make noise.. the works... |
| 01:28.53 | mzo | Delmar, probably. |
| 01:29.01 | mzo | is it the same LCD as what the real phones show? |
| 01:29.08 | delmar | mso, heh. ok but that asside.. government is a huge user. |
| 01:29.12 | delmar | mzo* |
| 01:29.14 | delmar | but then... |
| 01:29.20 | mzo | i only know about the ones i've seen on film. |
| 01:29.25 | mzo | most of the stuff is prop use. |
| 01:29.31 | mzo | there's a division of cisco who deals in prop sales. |
| 01:29.34 | delmar | the us government DoD devision has allocated to them... almost 200 Million IP addresses. |
| 01:29.46 | mzo | they call it 'branded media relations' and they get all the fake ones, |
| 01:29.56 | mzo | they also sell empty cisco cases for 'data center' shots |
| 01:30.01 | doushanes | delmar: there's you cheap cisco phone you wanted |
| 01:30.03 | mzo | like it's just a fake case with power lights |
| 01:30.18 | delmar | doushanes, yep. lol. look good ont he desk too huh :P |
| 01:30.28 | doushanes | I'll take 20 |
| 01:30.56 | delmar | mzo, what do u do .. when u say u work in film? |
| 01:31.10 | doushanes | customers roll in and see the nice phones. Good marketing |
| 01:31.25 | mzo | delmar, i'm a 'grip' |
| 01:31.32 | mzo | i'm still in film school. |
| 01:31.32 | doushanes | ....hype |
| 01:31.39 | delmar | doushanes, yeah but its harder to get a fake Jag parked in the CEO carpark :P |
| 01:31.48 | doushanes | doh |
| 01:31.50 | mzo | but i did set dressing, it's kinda boring shit. I'm naturally geeky and i thought they used real shit, too. |
| 01:32.00 | mzo | but almost every big company has a division that specializing in media stuff |
| 01:32.14 | delmar | mzo, heh. funny.. i already guessed before u said it.. the words " key grip" came into mind before u said that. :P |
| 01:32.18 | mzo | and they pay for on-screen stuff, so it's kinda promotional. |
| 01:32.18 | doushanes | you could always do the chyrsler labaron kit conversion to a mercedes!!!!! |
| 01:32.21 | mzo | i'm not a key-grip, i wish |
| 01:32.33 | mzo | i need another year of work before i get my union card and i could be a key grip |
| 01:32.42 | delmar | mzo, but yeah.. figured a grip of some sort. |
| 01:32.50 | mzo | heh, the shitty jobs :P |
| 01:33.04 | delmar | mzo, not really. |
| 01:33.18 | delmar | mzo, all kindsa fun. |
| 01:33.45 | delmar | mzo, so u are enjoying film school? |
| 01:33.48 | mzo | i'm at work at 4am. |
| 01:33.51 | mzo | yeah, it's fun. |
| 01:33.58 | mzo | some shoots last until 8 or 9, and then up at 4am again. |
| 01:33.59 | mzo | bleh |
| 01:34.01 | delmar | mzo, are u getting any film crew jobs as well? |
| 01:34.05 | mzo | yeah, |
| 01:34.07 | mzo | i take anything i can get |
| 01:34.08 | delmar | nice. |
| 01:34.15 | mzo | they offer bullshit crap at school for student productions. |
| 01:34.21 | mzo | desperate to get anything. :P |
| 01:34.24 | mzo | at least yo uget food |
| 01:34.35 | *** join/#asterisk themolester (i=themoles@30-54.34-65.tampabay.res.rr.com) |
| 01:34.39 | delmar | mzo, theres work here in NZ off and on all the time.. lots of productions over here.. especially with jackson barely stopping for a glass of wine before he gets into something else. |
| 01:35.22 | mzo | there's work everywhere :P |
| 01:35.23 | mzo | i mean in most countries anyway |
| 01:35.23 | mzo | and big cities ahve some kind of it |
| 01:35.23 | delmar | mzo, yep. |
| 01:35.23 | mzo | bleh, someone make my fwd dialup work! |
| 01:35.29 | mzo | i'm rejected going on four days now *Sigh* |
| 01:35.41 | delmar | mzo, sip or iax? |
| 01:35.54 | mzo | iax |
| 01:36.10 | mzo | says code 29 as the reason why |
| 01:36.21 | delmar | and its rejecting? wierd. try a password reset ? |
| 01:36.34 | mzo | um, where? on their site? |
| 01:36.43 | themolester | before i go read 100 docs, and such... i just wanted a yes no answer to whether asterisk will do what i need... what i need is... a 4 line telephone pool provided by viop (broadvoice?) all will ring to phone 1 which can be voip, or standard. once answered, the receptionist must be able to forward the call to one of at least 3 extentions, which will be analog phone cables |
| 01:36.55 | themolester | dunno if that all went through, but is that possible? |
| 01:37.06 | *** join/#asterisk ManxPower (i=ewieling@224.sub-70-197-246.myvzw.com) |
| 01:37.24 | mzo | i checked my password and it's teh same one that lets me log onto fwd so i'm not sure. |
| 01:37.27 | delmar | mzo, yeah i guess. never had to do one but i assume u can login and change the password on their end via the website? |
| 01:37.35 | mzo | i did that i think. |
| 01:37.37 | delmar | mzo, ok fair enuf. |
| 01:37.54 | ManxPower | Are you talking about IAXtel? |
| 01:37.56 | lunaphyte | themolester: sure |
| 01:38.20 | mzo | i mean fwddialup. |
| 01:38.27 | litage | is it possible to set the packetisation rate (eg: 20ms) in Asterisk, or is that an end-user-device setting/ |
| 01:38.31 | mzo | i didn't try iaxtel, they were supposed to send me a password but it's never arrived |
| 01:38.39 | themolester | ok, and one more... any one of those phones can be picked up, and after dialing 9, pick up one of the available pool lines? |
| 01:40.23 | lunaphyte | i'm no expert, but that sounds pretty basic. |
| 01:40.40 | ManxPower | litage, Asterisk requires 20ms of audio per packet. |
| 01:40.56 | ManxPower | Specifically MeetMe will sound horrible if it's not 20ms |
| 01:41.04 | themolester | lunaphyte ya, that one i figured was basic... the others diddn't know |
| 01:41.16 | ManxPower | themolester, what specific phone?: |
| 01:41.36 | themolester | was thinking a cisco phone for receptionist |
| 01:41.41 | themolester | and for one of the offices |
| 01:41.43 | lunaphyte | what do i need to do to match a sip peer? |
| 01:41.46 | litage | ManxPower: is it possible to change that to a different value though? say, 10ms? |
| 01:41.50 | themolester | the rest will be analog |
| 01:42.03 | ManxPower | litage, no. |
| 01:42.17 | ManxPower | Asterisk assumes 20ms packets EVERYWHERE. |
| 01:42.24 | themolester | i cant find ANY sip phones that are wireless (without using wifi ptewwie) |
| 01:42.43 | litage | ManxPower: so Asterisk forces that upon all codecs? |
| 01:42.43 | ManxPower | themolester, a SIPura + a cordless phone |
| 01:42.54 | ManxPower | litage, I believe so |
| 01:42.56 | lunaphyte | ptewwie? |
| 01:43.00 | litage | thanks ManxPower |
| 01:43.11 | themolester | lunaphyte spitting sound |
| 01:43.18 | themolester | i hate wifi |
| 01:43.21 | lunaphyte | oh |
| 01:43.26 | lunaphyte | why? |
| 01:43.41 | themolester | crap distance, laughable security |
| 01:43.44 | ManxPower | Wow! someone else that thinks Wifi not the solution |
| 01:43.56 | lunaphyte | it serves it purpose |
| 01:44.01 | themolester | not for phones |
| 01:44.07 | themolester | imo |
| 01:47.04 | mzo | heh, i have a wifi phone with wep. it's the biggest crap ever |
| 01:47.47 | themolester | assuming this question isn't too retarded... is there anything that broadvoice (or another byod provider) offers that could allow transferring extentions beyond your office... say you have two branch offices, and want to transfer calls between them... without having to buy a t1 or some outrageous internet line to rebroadcast the call |
| 01:48.06 | *** part/#asterisk Psykick (n=anon@203.167.226.250) |
| 01:48.23 | lunaphyte | just do it on your own |
| 01:48.45 | themolester | wouldn't that require doubling the bandwidth (and increasing latency) |
| 01:48.59 | themolester | or is it in the sip featureset somewhere |
| 01:49.12 | themolester | im still getting into voip |
| 01:49.19 | lunaphyte | the traffic has to go from point a to point b one way or another. |
| 01:50.21 | themolester | well... |
| 01:50.38 | themolester | on packet 8... they have the pbx hosted on their end... and you can do such a thing |
| 01:50.55 | themolester | but their boxes are not as advanced and i diddn't see any byod options on their site |
| 01:53.04 | DarthClue | themolester, check out asterlink, i'm pretty sure they can do what you want. |
| 01:54.04 | themolester | if i did run it over the internet a second time, do you think it would break up with using a business class roadrunner (cable) account with 7mb/1mb ? |
| 01:54.22 | lunaphyte | depends |
| 01:54.31 | lunaphyte | if there was no other traffic, no. |
| 01:54.42 | themolester | ? |
| 01:54.56 | *** join/#asterisk Himeko (i=himeko@S01060040ca128fc3.ed.shawcable.net) |
| 01:55.13 | themolester | do you mean if there was traffic, no |
| 01:55.33 | themolester | cause that doesn't sound so much like a depends |
| 01:55.34 | lunaphyte | no |
| 01:55.34 | themolester | heh |
| 01:55.48 | themolester | oh, my bad |
| 01:56.00 | themolester | im tired hehe |
| 01:56.12 | themolester | no it wont break up, not no it wont work |
| 01:56.27 | themolester | *sorrow |
| 01:56.53 | lunaphyte | with a connection like that, if you can't support 2 ip calls when there is no other traffic, i'd call your provider and complain |
| 01:57.10 | themolester | umm... i did not say 2 calls |
| 01:57.34 | lunaphyte | you didn't say any. ;) |
| 01:57.43 | themolester | 4 lines, plus 1 fax in each location |
| 01:57.46 | lunaphyte | unless you mean all 4? |
| 01:57.48 | lunaphyte | ah |
| 01:57.55 | *** join/#asterisk pengyong (n=lala@222.188.136.120) |
| 01:58.02 | lunaphyte | so i guess i'm tired :) |
| 01:58.30 | De_Mon | If I want to send and receive calls over a POTS line, would kinda card do I need? FXO/FXS are only one way it seems |
| 01:59.23 | *** join/#asterisk tuxinator_linux (n=tuxinato@m190e36d0.tmodns.net) |
| 01:59.49 | lunaphyte | themolester: iiuc, ip calls can use as little as ~ 8k. make it 10 for easy math, and you end up with 100k of voice traffic. |
| 01:59.53 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 02:00.22 | themolester | is that good enough quality? i thought the major providers used 64k streams? |
| 02:00.24 | De_Mon | wait a sec... I just need an FXO card, yes? |
| 02:01.20 | lunaphyte | themolester: not sure what they use. |
| 02:02.27 | themolester | what does 8k sound like? |
| 02:02.56 | themolester | also, whats the minimum needed for teh fax lines |
| 02:03.04 | Abydos313 | 8k sounds like you're going to be paying for a codec |
| 02:04.45 | mzo | paying for a codec? :P |
| 02:05.01 | mzo | what's the deal with that payin anyway? i'm curious how that works? |
| 02:05.29 | litage | when an ATA or IP phone's web gui asks for a value for the RTP packet size, what unit is it referring to? kilobytes? milliseconds? |
| 02:08.42 | De_Mon | wtf? I can't find any single or dual line cards on digiums site |
| 02:10.24 | troyb1 | hey guys, when i receive a call through my DID it rings directly to my extension not [mainmenu] can anyone help me? |
| 02:11.26 | litage | De_Mon: not sure what you mean. they're all listed on Digium's hardware page: Pattern: [-A-Za-z0-9 '.,]+ |
| 02:12.11 | litage | troyb1: check out what Asterisk's debug messages say about the path the call takes when coming in |
| 02:13.54 | troyb1 | litage its coming through [default] |
| 02:14.13 | troyb1 | however for it to get there, the context = mainmenu, thus the call has to traverse -> mainmenu -> submenu -> default |
| 02:14.48 | litage | troyb1: i'm afraid i'm not an expert in this area, so i can't be of any additional help =/ |
| 02:15.05 | troyb1 | litage its okay :) |
| 02:17.04 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 02:20.35 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.254) |
| 02:21.00 | De_Mon | litage the TDM400 is the lowest end card they haven't discontinued according the wiki |
| 02:21.19 | De_Mon | sorry, don't wana shell our $250 just yet... ^_^ |
| 02:21.32 | litage | De_Mon: $250 + modules =P |
| 02:21.35 | troyb1 | De_Mon could always buy a clone. |
| 02:21.50 | troyb1 | i paid less then that for my cisco 7940 :P |
| 02:21.50 | De_Mon | litage 250 includes 1 fxo and 1 fxs module ;) |
| 02:22.00 | troyb1 | hh |
| 02:22.02 | ThaZZa_Work | Can anyone help me with asterisk startup issue.. |
| 02:22.12 | De_Mon | ThaZZa_Work not unless you state what they are |
| 02:22.18 | ThaZZa_Work | i am doing strace asterisk and getting the following error. |
| 02:22.22 | ThaZZa_Work | open("/var/run/asterisk/asterisk.pid", O_WRONLY|O_CREAT|O_TRUNC, 0666) = 3 |
| 02:22.56 | litage | De_Mon: really! that's not bad at all |
| 02:22.57 | De_Mon | litage probably just bits |
| 02:23.36 | litage | ThaZZa_Work: can asterisk write to /var/run/asterisk/ ? |
| 02:24.05 | ThaZZa_Work | litage: this is full command pastebin.. http://pastebin.com/569583 |
| 02:25.35 | litage | ThaZZa_Work: your error is on line 2 of that paste |
| 02:26.00 | litage | ThaZZa_Work: when reading debug output, always look for the very first error, because later errors can be a result of earlier errors |
| 02:31.15 | *** join/#asterisk VeNoMouS_ (n=venom@60-234-209-199.bitstream.orcon.net.nz) |
| 02:31.53 | VeNoMouS_ | hay has anyone come aross a problem where you can call asterisk from a landline and ave an exten where it plays voicemial, but it dont work from cell phone providers? |
| 02:32.04 | ThaZZa_Work | litage: You mean "etc/ld.so.preload", O_RDONLY) = -1 ENOENT" |
| 02:32.22 | litage | ThaZZa_Work: yes, that's an error |
| 02:32.27 | VeNoMouS_ | on vodafone it just disconnects right away, on telecom it just sits there but u cant hear anything or say anything |
| 02:32.50 | ThaZZa_Work | litage: Yeah yet i get that same error at home. and asterisk works fine. |
| 02:32.54 | VeNoMouS_ | ie |
| 02:32.55 | VeNoMouS_ | <PROTECTED> |
| 02:32.55 | VeNoMouS_ | <PROTECTED> |
| 02:32.55 | VeNoMouS_ | <PROTECTED> |
| 02:33.26 | VeNoMouS_ | ^^ thats vodafone |
| 02:34.08 | litage | VeNoMouS_: use a pastebin |
| 02:34.09 | DeadZen | moo |
| 02:34.29 | mzo | ~pb |
| 02:34.31 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 02:34.56 | litage | ThaZZa_Work: why do you think line 21 of the paste is an error that's preventing asterisk from running? |
| 02:34.57 | VeNoMouS_ | yea i know what a pastebin is |
| 02:35.01 | iCEBrkr | werd |
| 02:35.24 | litage | thanks mzo, i can never remember the macro for each channel i'm in =p |
| 02:35.47 | *** part/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 02:36.52 | ThaZZa_Work | litage: I lied. it is not.. it is a safe_asterisk issue i can see now. |
| 02:37.20 | ThaZZa_Work | litage: cause if i run this from prompt it works fine. 'asterisk -vvvc -U asterisk" |
| 02:38.35 | ThaZZa_Work | lilo: I seem to find out. it is not working.. it is the user path that is cauing issues. |
| 02:39.05 | litage | ThaZZa_Work: =) glad you found the problem |
| 02:40.15 | mzo | i always forget, |
| 02:40.26 | ThaZZa_Work | litage: Not yet check out pastebin.. (and yes i can't upgrade yet) http://pastebin.com/569603 |
| 02:41.40 | litage | ThaZZa_Work: strace it again, i guess |
| 02:42.33 | ThaZZa_Work | litage: very good.. found a bug. stat64("/etc/asterisk/phone.conf", 0xbfffcd3c) = -1 EACCES (Permission denied) |
| 02:42.45 | SplasPood | Anyone played with a WIP300 yet? |
| 02:43.00 | SplasPood | Linksys wifi SIP phone.. |
| 02:43.56 | ThaZZa_Work | litage: Yet it is owned by root, and asterisk has read permission. |
| 02:45.16 | [hC] | to make a Zap channel not answer the phone, it should be as simple as taking the context out, right? |
| 02:45.45 | litage | is it possible to set the packetisation rate (eg: 20ms) in Asterisk, or is that an end-user-device setting? |
| 02:47.01 | ThaZZa_Work | litage: the permissions are: onwer - root : group - asterisk and 644 is the perm's |
| 02:49.11 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 02:49.25 | litage | ThaZZa_Work: very strange |
| 02:49.47 | litage | ThaZZa_Work: actually, it might be that when you run asterisk, it runs as user 'asterisk' but doesn't run as group 'asterisk' |
| 02:50.11 | ThaZZa_Work | litage: Just worked that one out. thanks mate.. |
| 02:50.17 | litage | no worries |
| 02:50.24 | ThaZZa_Work | litage: now i need to fix my safe_asterisk script. |
| 02:52.14 | litage | ThaZZa_Work: you're an aussie! |
| 02:52.18 | litage | =P |
| 02:52.57 | ThaZZa_Work | litage: you saying something is bad about being an aussie? |
| 02:53.22 | litage | ThaZZa_Work: haha on the contrary, i'm in the gold coast |
| 02:54.24 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:56.02 | ThaZZa_Work | litage: Well i am not far away then.. Do you know where the default safe_asterisk script is stored, before you compile? |
| 02:56.18 | litage | ThaZZa_Work: 'fraid not |
| 02:57.15 | litage | ThaZZa_Work: http://rafb.net/paste/results/J0IgrL75.html |
| 02:59.43 | [hC] | what the junk here... |
| 02:59.58 | [hC] | Ive removed context= from my zapata.conf, yet the modem still answers the line |
| 03:01.45 | ThaZZa_Work | litage: Found where the bug is.. looks like the person who setup /etc/init.d/asterisk didn't install start-stop control. |
| 03:02.10 | ThaZZa_Work | litage: and also the permissions on /etc/asterisk should be asterisk:asterisk |
| 03:02.19 | litage | ThaZZa_Work: annoying. you might want to submit your fix(es) |
| 03:06.37 | *** join/#asterisk dVoka (n=dVoka@CPE-65-29-147-86.wi.res.rr.com) |
| 03:08.21 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 03:10.08 | *** part/#asterisk dVoka (n=dVoka@CPE-65-29-147-86.wi.res.rr.com) |
| 03:10.36 | themolester | what kind of device would i use to connect a analog phone to an asterisk |
| 03:10.45 | Qwell | themolester: an ata, or an analog Digium card |
| 03:11.17 | themolester | also, are there any good companies to buy a single box unit with asterisk? |
| 03:11.42 | themolester | like has all the equipment built together in a compact form but with all the features of asterisk |
| 03:11.43 | Qwell | single box unit? |
| 03:11.55 | Abydos313 | hire Qwell to build and set it up ;) |
| 03:11.58 | themolester | the analog jacks, the ethernet, a web interface, etc |
| 03:11.59 | Qwell | indeed |
| 03:12.20 | Abydos313 | don't buy a box , buy an end solution |
| 03:12.24 | themolester | so that you dont have to buy a seprate pc, install an os etc |
| 03:13.20 | themolester | Abydos313 sounds like what i want... any places i should look? |
| 03:14.15 | Abydos313 | no idea |
| 03:15.50 | [hC] | Hmm. for some reason 5 g729 channels took 60kbit/channel... |
| 03:16.12 | [hC] | is g729 8 kB or kb/s ? must be kB |
| 03:17.39 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 03:18.10 | ThaZZa_Work | litage: Don't think i will. They are probally already fixed in 1.2.4, thats why it works fine at home.. This is works pabx. and it is running (Yucky) 1.2.0 |
| 03:18.38 | ThaZZa_Work | litage: You haven't played with the Sipura SPA-3000 and asterisk have you? |
| 03:19.13 | litage | ThaZZa_Work: is the SPA-3000 the same as the Linksys PAP2? if so, yes |
| 03:24.02 | *** join/#asterisk krischnoff (n=chatzill@209.77.205.12) |
| 03:26.03 | *** join/#asterisk PaulHuynh (n=paulhuyn@c-68-44-237-105.hsd1.de.comcast.net) |
| 03:26.14 | PaulHuynh | hello |
| 03:26.14 | PaulHuynh | <PaulHuynh> i got my pap2 to talk to asterkis via g711u |
| 03:26.14 | PaulHuynh | <PaulHuynh> but i need to cut back on bandwidth so i need to be able to use g729 but when i set allow=g729 i got no audio |
| 03:26.50 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 03:27.29 | SwK | most likely you rae trying to transcode then |
| 03:27.53 | SwK | if thats the case, you'll need some g729 lic's from digium... they are only $10 each so order a few and be done with it |
| 03:29.03 | *** join/#asterisk opsys (n=opsys@68-235-141-52.miamfl.adelphia.net) |
| 03:29.34 | WasPhantom | http://www.trademe.co.nz/Home-living/Outdoor-garden-conservatory/Other/auction-48293162.htm - heh |
| 03:36.41 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 03:36.46 | PakiPenguin | hello everyone |
| 03:36.52 | PakiPenguin | anyone of you use astguiclient? |
| 03:37.52 | *** join/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 03:38.10 | propagandhi | what does the qualify parameter on an IAX trunk do |
| 03:39.12 | DeadZen | http://www.dice.com/beingit/ |
| 03:40.08 | *** join/#asterisk jero (n=jero@modemcable062.46-82-70.mc.videotron.ca) |
| 03:40.48 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
| 03:44.05 | *** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca) |
| 03:48.54 | *** join/#asterisk bmg505 (n=leon@dsl-146-50-124.telkomadsl.co.za) |
| 03:49.00 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 03:51.33 | *** join/#asterisk normast (n=NormAst@toronto-HSE-ppp3975028.sympatico.ca) |
| 03:52.48 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 03:53.53 | lunaphyte | how do i make a call ring on multiple extensions? |
| 03:54.37 | Qwell | SIP/1&SIP/2 |
| 03:54.50 | propagandhi | lunaphyte: or create a queue |
| 03:56.29 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:56.58 | lunaphyte | hmm. reading about queues, ringall sounds promising. |
| 03:57.01 | propagandhi | lunaphyte: and u can set a ringall strategy |
| 03:57.05 | propagandhi | yes thats it |
| 03:57.10 | lunaphyte | :) |
| 03:59.17 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 03:59.35 | iaxy | phillip, you here? |
| 04:00.29 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 04:00.38 | ComputerWarm | Hello all question anyone know of a program that will show ASR ACD for asterisk |
| 04:01.59 | *** join/#asterisk Gir19 (i=Gir@67.189.110.174) |
| 04:02.45 | iaxy | Hi all. |
| 04:02.49 | websae | hey |
| 04:03.03 | Gir19 | hey |
| 04:03.34 | Gir19 | any of you hear about the IRC/norton PF bug? |
| 04:04.12 | Abydos313 | anyone ever use this device? http://cgi.ebay.com/IAX-Native-FXS-for-Digium-Asterisk-VoIP-PBX-Beats-IAXy_W0QQitemZ5870915404QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 04:04.29 | Gir19 | not that it has anything to do with asterisk, just a side thing I found out. |
| 04:04.43 | Abydos313 | Gir19 i heard of it |
| 04:05.57 | Gir19 | Adydos313: ever try it to see if it actually worked? |
| 04:05.58 | iaxy | I knew a girl19 once.....course, that was when I was 18 |
| 04:06.45 | Abydos313 | Gir19 that is what i'm asking? i need to buy two devices..one for the asterisk server and other for remote client extension and wanted to use iax..so i was looking at that device |
| 04:07.00 | iaxy | Anyone played with a sip handset gateway? specifically citel box? |
| 04:08.05 | iaxy | look at x100p.com for their new iax ata |
| 04:08.15 | Gir19 | Adydos313: I have not used that one, but I have used others and they do work, but not for fax machines. |
| 04:08.24 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 04:08.54 | Abydos313 | hmm. why not fax? |
| 04:09.18 | Gir19 | iaxy: I've used the Sipura SPA-2100 and it works like all other ATA's but I cannot use them as I need to be able to use fax machines. |
| 04:09.25 | sevard | I'm still looking for a WiMax SIP phone :( make one god! |
| 04:09.30 | Abydos313 | iaxy that is the device i pasted. it was a link of theirs from ebay |
| 04:09.41 | Abydos313 | have you used the fxs 2.0 |
| 04:10.10 | Abydos313 | Gir19 so what do you use and what protocol |
| 04:10.13 | ThaZZa_Work | litage: Found the bug.. It was so simple. :-( |
| 04:10.19 | iaxy | oh...sorry |
| 04:10.39 | Gir19 | Adydos313: Fax to Foip does not work well, if at all because of network bursts and what the fax machines see as line noise will cause problems. |
| 04:11.30 | Abydos313 | ok, faxing is really not an issue at this time, hope there is a future working solution |
| 04:12.15 | Abydos313 | iaxy have you ever used the fxs for asterisk? |
| 04:12.51 | *** join/#asterisk livinded (n=livinded@cpe-24-24-190-252.socal.res.rr.com) |
| 04:13.05 | Gir19 | as of right I only know of installing a mini asterisk server into the location I need to do faxing at and let it use a tdm20b |
| 04:13.18 | iaxy | If you mean fax, yes...but I go into x100p and asterisk creates pdf and emails it |
| 04:13.20 | livinded | why do only extensions under the demo header work, all the other ones say unavailable |
| 04:14.02 | Gir19 | I don't use the fax to email thing cause alot of people cannot accept emails with there fax machines. |
| 04:15.01 | russellb | who has a fax machine that accepts email? |
| 04:15.09 | russellb | I'd like to see it! that would be cool |
| 04:15.20 | file | russellb: go away! |
| 04:15.20 | websae | hehehe |
| 04:15.25 | websae | file: how are you? |
| 04:15.26 | gaupe | wouldn't be to hard to make |
| 04:15.34 | file | I exist. |
| 04:15.35 | file | you? |
| 04:15.40 | livinded | oh, i need to include them in default |
| 04:15.48 | websae | file: i just wrote you an email about 800, how much it is and does it include US 48 and Canada |
| 04:15.53 | russellb | file: shush you |
| 04:15.56 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 04:16.19 | file | $1.95/mth, 2 cents per minute inbound from US48, 10 cents per minute inbound from Canada |
| 04:16.33 | websae | wow, expensive Canadian rates, heh? |
| 04:16.34 | websae | lol |
| 04:16.35 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-96-61.qld.bigpond.net.au) |
| 04:16.37 | file | yes |
| 04:16.42 | websae | why? |
| 04:16.49 | file | because that's what Qwest charges us |
| 04:16.51 | websae | i think i found it for 3.5cents/min |
| 04:16.56 | file | then use them |
| 04:17.00 | websae | might have to |
| 04:17.01 | file | I can't change our rates from Qwest :) |
| 04:17.13 | russellb | file: Qwest is for n00bs |
| 04:17.20 | websae | i don't like Qwest too much, they charge a lot for T1s eeeks |
| 04:17.31 | file | russellb: no no, Qwest ARE n00bs |
| 04:17.37 | niZon | what's the least expensive device that will give me 1FXO and 1FXS (simultaneously) for asterisk? |
| 04:18.19 | [TK]D-Fender | niZon : SPA-3000 |
| 04:18.24 | niZon | thought so |
| 04:18.25 | iaxy | sipura has one |
| 04:18.35 | iaxy | thats it |
| 04:18.46 | niZon | anyone used the grandstream one... the HT488 i think? |
| 04:18.49 | russellb | iaxy: so, why is your name iaxy? |
| 04:18.56 | Abydos313 | iaxy that same device is 30 bucks cheaper than a spa3k |
| 04:19.39 | [TK]D-Fender | iaxy does not have an FXO port, and frankly is pretty flakey... |
| 04:19.53 | iaxy | russellb:because I like it better than Grandstream |
| 04:20.00 | russellb | cool :) |
| 04:20.07 | *** join/#asterisk ircritesh (n=riteshir@c-24-6-80-84.hsd1.ca.comcast.net) |
| 04:20.08 | mzo | i have a grandstream 286 thing, it's kinda crappy :P |
| 04:20.28 | ircritesh | any idea/blog on how to connect asterisk to gtalk? |
| 04:20.38 | websae | grandstream ATAs are crappy? |
| 04:20.40 | russellb | ircritesh: you can't ... yet |
| 04:20.45 | ircritesh | i am looking for a lead so that I can start thinking of writing the interface? |
| 04:20.47 | evilbuny | iaxy would be compared to the spa1001, not the 3000 |
| 04:20.48 | ircritesh | why not yet? |
| 04:20.48 | Abydos313 | what are good ata's? |
| 04:20.51 | mzo | mine is, but i don't know, i have the old POS one ;) |
| 04:20.52 | iaxy | I don't know what guys think of the gxp2000, I just got one the other day and it is a world better than the budget tone |
| 04:20.55 | evilbuny | spa1001 = ~$50 |
| 04:21.00 | ircritesh | russellb? |
| 04:21.01 | websae | i have the gxp2000 |
| 04:21.03 | russellb | ircritesh: because the support isn't written. However, someone is working on it. |
| 04:21.14 | russellb | ircritesh: it's in a public branch in svn if you're interested. |
| 04:21.15 | websae | i like it a lot |
| 04:21.22 | ircritesh | yes, I am interested.... |
| 04:21.32 | ircritesh | could you point me to the branch please? |
| 04:21.37 | ircritesh | very interested |
| 04:21.50 | russellb | ircritesh: http://svn.digium.com/svn/asterisk/team/mogorman/asterisk-xmpp |
| 04:22.06 | ircritesh | thanks a bunch russelb |
| 04:22.08 | russellb | It's not functional for what you want, yet |
| 04:22.13 | russellb | but that is where it is being worked on |
| 04:22.20 | russellb | you're welcome |
| 04:22.32 | ircritesh | you have any idea how far is it right now? |
| 04:22.45 | ircritesh | i want to start contributing so wanted to know in general |
| 04:22.50 | iaxy | Has anyone worked on a Citel handset gateway? |
| 04:23.00 | russellb | No. You'll have to ask the author. He's on here as some variation of "mog" |
| 04:23.43 | russellb | ircritesh: or you could probably email him at mogorman@digium.com |
| 04:24.00 | ircritesh | great!! you are good... |
| 04:24.53 | websae | file: are you still there? |
| 04:25.04 | websae | file: are there 800 numbers that mexico can dial? |
| 04:25.10 | websae | if so what's the rate |
| 04:25.26 | iaxy | I need a Citel tech.... this thing is pissing me off!!!! |
| 04:26.06 | file[laptop] | I don't know how those are handled really... I think I saw them in the rateplan... probably billed at US48 rate |
| 04:26.20 | websae | could you find out? |
| 04:26.22 | websae | and let me know |
| 04:26.31 | websae | by tomorrow |
| 04:26.46 | file[laptop] | well, it's already tomorrow here... and I'm in bed... but I can look |
| 04:26.53 | websae | you can check right now? |
| 04:26.59 | file[laptop] | once I login, yes |
| 04:27.27 | De_Mon | file[laptop] in bed with a laptop? how.. single |
| 04:27.37 | file[laptop] | I'm single and available. |
| 04:27.38 | websae | no he has a king size bed |
| 04:27.40 | iaxy | I wanna to develop a 24 port iax gateway... |
| 04:27.47 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-96-61.qld.bigpond.net.au) |
| 04:28.10 | file[laptop] | websae: Mexico is on the rate plan for Extreme Asterlink, so calls would be accepted inbound from there |
| 04:28.15 | file[laptop] | as long as the callerid was a Mexico number |
| 04:28.26 | websae | what is extreme asterlink? |
| 04:28.45 | file[laptop] | it's one of the two plans... just has you configure how inbound calls are handled |
| 04:28.46 | websae | is that 2cents/min as well on that 800 number for mexico users dialing it? |
| 04:28.50 | file[laptop] | but it uses a different rateplan |
| 04:28.51 | file[laptop] | yes |
| 04:28.58 | file[laptop] | it's treated as an international call for them |
| 04:28.59 | websae | uses a different rate plan? |
| 04:29.08 | websae | ohh, it cost them? |
| 04:29.10 | file[laptop] | yes |
| 04:29.24 | websae | so 800 number is how much to setup? |
| 04:29.39 | file[laptop] | an 800 number from the pool costs nothing to setup, and $1.95/mth |
| 04:29.43 | websae | 800 number--- Canada: 10cents/min US: 2cents/min Mexico 2cents/min |
| 04:29.54 | websae | correct? |
| 04:29.57 | file[laptop] | yes |
| 04:30.07 | websae | so no mexico 800 numbers that don't cost them |
| 04:30.32 | file[laptop] | we don't do international toll-free number stuff |
| 04:30.39 | iaxy | Anyone tried the ST-302 iax phone? |
| 04:31.22 | file[laptop] | Mexico is international... so dialing a toll-free number would be considered an international call for them |
| 04:31.31 | ircritesh | hmm...wonder how jabphone guys got ahead...must be some smarts....i loved the jabphone interface for asterisk? |
| 04:31.38 | ircritesh | and gtalk |
| 04:32.48 | mzo | what's the deal with vonage? |
| 04:32.54 | mzo | can you actually call vonage through fwd? |
| 04:34.17 | *** join/#asterisk iaxy (n=iaxy@modemcable236.55-131-66.mc.videotron.ca) |
| 04:35.40 | sevard | So i'm trying to build an * system from the ground up as I've begun to hate amp, i'm adding configuration files as I go and I just ran into "no such command 'sip'" which disturbs me |
| 04:36.21 | Abydos313 | run asterisk -rvvvv |
| 04:36.27 | Abydos313 | and then sip commands work |
| 04:36.29 | sevard | That's what i'm running |
| 04:36.39 | Abydos313 | type help |
| 04:36.44 | sevard | help shows all commands |
| 04:36.57 | sevard | but sip <anything> says no such command sip |
| 04:37.00 | Abydos313 | and those commands dont' work |
| 04:37.06 | Abydos313 | sip show peers |
| 04:37.10 | sevard | Everything works except sip anything |
| 04:37.16 | sevard | which gives me no such command |
| 04:37.23 | Abydos313 | no idea |
| 04:37.34 | sevard | same here, i would have thought taking out conf files wouldn't break the cli |
| 04:37.53 | mzo | no one knows, vonagae through fwd? :) |
| 04:37.59 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
| 04:38.25 | russellb | ircritesh: there he is now :) |
| 04:39.44 | iaxy | sevard: type sip and hit the tab key. Does it show you the options? |
| 04:40.25 | troyb1 | sip error.. shitty protocol *kidding* |
| 04:40.26 | ircritesh | mogorman, just sent you an email. Interested in Asterisk XMPP. Wanna provide some updates... |
| 04:40.39 | ircritesh | I want to contribute |
| 04:40.51 | ircritesh | actually, going through your code right now.. |
| 04:42.45 | *** join/#asterisk livinded (n=livinded@cpe-24-24-190-252.socal.res.rr.com) |
| 04:42.58 | sevard | iaxy: No, it doesn't. Is SIP in a module somewher that i'm not enabling? |
| 04:43.03 | livinded | do i only need libpri is i am using a pri and an fxo card? |
| 04:43.16 | sevard | I |
| 04:43.23 | sevard | I'm building all of the conf files by hand. |
| 04:44.40 | sevard | chan_sip.so perhaps... |
| 04:44.54 | troyb1 | sevard i dont suggest building the conf by hand.. |
| 04:45.04 | sevard | troyb1: why's that? |
| 04:45.31 | troyb1 | make a copy of the default sip.conf / iax.conf / extensions.conf , read through *everything* asterisk configuration isnt straightforward, you can miss something really easily |
| 04:45.49 | troyb1 | i missed one option and it took me nearly 2 hours to unwind |
| 04:46.53 | troyb1 | by the same token i have been playing with asterisk for 5 days and i have a nearly complete phone system, there is a learning curve, (you have to visualize everything) |
| 04:47.11 | troyb1 | its just like programming, if you've learned java |
| 04:47.31 | livinded | do i need libpri is i'm not using a pri and onyl am using ztdummy from zaptel? |
| 04:48.14 | troyb1 | livinded if i were you i'd compile zaptel / libpri / asterisk in the order specified, it will take you maybe an extra 5 minutes but its probably worthwhile |
| 04:48.18 | troyb1 | there is no downside to compiling them |
| 04:48.41 | livinded | i didn't compile it though, is there any reason to go back uninstall astrisk and the install libpri? |
| 04:49.16 | troyb1 | i dont believe so.. i mean if the asterisk CLI isnt erroring and your switching packets, all the power to you |
| 04:49.45 | livinded | as of right now i'm just having codec issues |
| 04:49.56 | livinded | but i doubt that has nything to do with libpri |
| 04:49.58 | troyb1 | livinded the easiest thing to do, is allow=all |
| 04:50.04 | troyb1 | disable all the other codec options |
| 04:50.19 | *** join/#asterisk Huynh (n=chatzill@adsl-71-131-12-154.dsl.sntc01.pacbell.net) |
| 04:50.19 | livinded | i had that |
| 04:50.23 | troyb1 | they are just going to piss you off as you gamble one option at a time :P |
| 04:50.37 | troyb1 | did you comment the other codec opt's? |
| 04:50.42 | livinded | when i try to dial it says its not able to negotiate a codec |
| 04:50.49 | troyb1 | did you comment the other codec opt's? |
| 04:50.54 | troyb1 | using a ; |
| 04:50.55 | *** part/#asterisk krischnoff (n=chatzill@209.77.205.12) |
| 04:50.56 | livinded | i need to look |
| 04:51.00 | troyb1 | :P |
| 04:51.06 | troyb1 | thats something to look into :) |
| 04:51.12 | livinded | i just shutdown the box and don't plan on playign around with it anymore tonight |
| 04:51.22 | troyb1 | livinded, you shutdown an asterisk box? |
| 04:51.29 | livinded | ya |
| 04:51.36 | troyb1 | are you unable to pay for power? |
| 04:51.49 | livinded | no but my room is loud with it running, its hard to sleep |
| 04:52.10 | troyb1 | you will get used to it.. i have up to 6 machines running in here |
| 04:52.21 | troyb1 | i dont even need lights.. |
| 04:52.23 | livinded | i turn off all my machiens at night |
| 04:52.29 | troyb1 | :o your modem too? |
| 04:52.34 | livinded | no |
| 04:52.41 | livinded | just the machines in my room |
| 04:52.44 | troyb1 | your router? |
| 04:52.49 | livinded | no |
| 04:52.49 | ThaZZa_Work | Anyone got a good startup script for /etc/rcS.d/ to start ztdummy and /etc/init.d/asterisk |
| 04:52.52 | troyb1 | :) |
| 04:53.07 | livinded | i do need to buy a new switch though |
| 04:53.15 | livinded | i'm out of room on my 5port workgroup |
| 04:53.18 | troyb1 | livinded linux wasnt meant to shutdown :P i have seen phone switches with uptimes longer then guys i know |
| 04:53.27 | livinded | ya i know its not |
| 04:53.32 | livinded | but i don't need it up |
| 04:53.38 | troyb1 | hehe |
| 04:53.55 | Huynh | can someone help me with t1 pri? |
| 04:53.57 | livinded | it doesn't do anything so i might as well save my parent some money and make my room quieter |
| 04:54.18 | troyb1 | ahh, well then.. you should be commended for your efforts :) if your lucky you saved them a shiny loonie this month |
| 04:54.59 | troyb1 | okay im done :P |
| 04:55.02 | livinded | troyb1, i would have saved a loonie if i was in canada but i'm not so i saved a dollar or so |
| 04:55.08 | mzo | bleh, why doesn't fwd work |
| 04:55.22 | troyb1 | we call currency dollars too :P we call one dollar currency 'loonies' |
| 04:55.50 | mzo | toronto girls are hot |
| 04:55.55 | mzo | oops, wrong channel |
| 04:56.02 | troyb1 | mzo well then you havnt been to montreal my friend. |
| 04:56.28 | troyb1 | you wont get toronto girls on the first night ;) *or for the most part* |
| 04:57.20 | mzo | i've been to montreal |
| 04:57.26 | troyb1 | ohh well then :) |
| 04:57.28 | mzo | i've been all over canuckland ;P |
| 04:57.32 | mzo | newfies are hot too :P |
| 04:58.38 | mzo | bleh, i wish i knew why FWD wouldn't work for me |
| 04:58.38 | troyb1 | thats 220 by the way, so you go out and get a shiny transformer *rufflez* |
| 04:58.45 | troyb1 | mzo it works for me :o |
| 04:58.58 | livinded | my saved power is just a byproduct of my increasing my ability to sleep well by turning off the boxes not the other way around |
| 04:59.01 | mzo | haha |
| 04:59.03 | mzo | app_saycountpl.so] => (Say polish counting words) |
| 04:59.03 | mzo | ? |
| 04:59.06 | mzo | wtf is that? |
| 04:59.17 | troyb1 | thats just crazy, crazy like a cat, (i havnt seen that error), |
| 04:59.17 | mzo | troy, help? It won't register on iax |
| 04:59.21 | troyb1 | are you using the default config? |
| 04:59.24 | mzo | yah |
| 04:59.28 | troyb1 | :o |
| 04:59.28 | mzo | i just noticed that during a restart |
| 04:59.39 | troyb1 | you did your register => in iax.conf ? |
| 04:59.43 | mzo | yes |
| 04:59.47 | mzo | i did it through aah |
| 04:59.52 | troyb1 | ? |
| 04:59.56 | mzo | let me check |
| 04:59.57 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 05:00.04 | mzo | they have slightly different directions |
| 05:00.06 | mzo | but it still says refused |
| 05:00.27 | troyb1 | are you follow'in the dir's off voip-info? |
| 05:01.09 | FuriousGeorge | what does asterisk display for you guys when the CID is unknown? |
| 05:01.10 | troyb1 | i promise you if you follow the ones off the FWD site *it will* work |
| 05:01.17 | mzo | i'm following this |
| 05:01.20 | troyb1 | FuriousGeorge anonymous |
| 05:01.28 | troyb1 | or so my cisco 7940 says *pos* |
| 05:01.30 | mzo | http://asteriskathome.sourceforge.net/handbook/index.html |
| 05:01.36 | livinded | i miss the easyness that amp gave me |
| 05:01.41 | troyb1 | ohh asterisk@home theres your problem.. i know nothing about it. |
| 05:01.43 | livinded | almost makes me want to go back |
| 05:01.45 | FuriousGeorge | troyb1: mine pulls a random extensions CID name and number |
| 05:01.47 | mzo | the ones on the FWD site fucked up my configuration so bad that i had to reinstall |
| 05:02.12 | troyb1 | FuriousGeorge you must be furious :> (you can set cid in extensions.conf (outgoing)) |
| 05:02.23 | livinded | mzo, just rm * in /etc/asterisk and run make samples again |
| 05:02.25 | FuriousGeorge | troyb1: im sure i can set it incoming |
| 05:02.26 | troyb1 | i dont know why it'd be random |
| 05:02.32 | FuriousGeorge | its not really random |
| 05:02.39 | FuriousGeorge | its idiosyncratic tro every server |
| 05:02.41 | troyb1 | livinded no need to rm * the script auto rm -rf * |
| 05:02.49 | livinded | oh |
| 05:02.50 | FuriousGeorge | one does Maria 103 another does Asterisk 0 |
| 05:02.55 | troyb1 | FuriousGeorge im setting a 2 syllable rule.. say again? |
| 05:03.21 | Qwell | troyb1: id-io-syn-cra-tic |
| 05:03.26 | FuriousGeorge | lol, every server displays a different extension with unknown CIDs and no one can really tell me why |
| 05:03.43 | troyb1 | Qwell you may as well hit me a cluebat cause i still dont get it :( |
| 05:03.50 | Qwell | ~thwap troyb1 |
| 05:03.51 | jbot | ACTION thwaps troyb1 on the nose with a 2 by 4 |
| 05:04.01 | troyb1 | Qwell got wood? *grinz* |
| 05:04.13 | FuriousGeorge | im gonna just put it in my dialplan logic for inbound calls (as much as i dont want to) that if the CID = Maria 103, set it to UNKNOWN 0000000000 |
| 05:04.16 | troyb1 | FuriousGeorge well the foreign server is responsible for the cli incoming |
| 05:04.28 | troyb1 | your responsible outgoing |
| 05:04.30 | FuriousGeorge | its the telco |
| 05:04.41 | troyb1 | incoming server = telco |
| 05:04.42 | FuriousGeorge | the just leave it blank and my server picks an extension |
| 05:04.54 | troyb1 | wait is that right erm |
| 05:04.59 | troyb1 | foreign server = telco |
| 05:05.00 | FuriousGeorge | the telco does, i meant to say |
| 05:05.10 | FuriousGeorge | yeah, the central office leaves the CID blank |
| 05:05.14 | troyb1 | yeah, well i personally havnt set CLI *i just dont care* |
| 05:05.34 | FuriousGeorge | and my local * does something stupid with it and i seem to be the only one |
| 05:05.37 | troyb1 | FuriousGeorge you should walk into that co like kevin poulsen *stares at Qwell* |
| 05:06.15 | troyb1 | FuriousGeorge i wouldnt know what your telco does specifically, my brain doesnt have the capacity to remember every clec in the phon book |
| 05:06.26 | *** join/#asterisk natmlt (n=Moriah@ip24-251-50-94.ph.ph.cox.net) |
| 05:06.49 | *** join/#asterisk username_taken (n=stinky@203.202.184.90) |
| 05:06.51 | SkramX | gambolputty? where is he!!!! |
| 05:07.09 | troyb1 | SkramX have you checked the slut machines. |
| 05:07.13 | troyb1 | err *slot machines |
| 05:07.47 | livinded | ooooo that would be so cool to run linux on |
| 05:07.55 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 05:08.14 | troyb1 | anyways guys, my work requires my focus; enjoy |
| 05:08.26 | username_taken | Hey guys i just broke my asterisk box trying to get a te110p connected. I'm running A@H 2.0 and after i ran genzaptelconf i can't hear any messages |
| 05:08.37 | FuriousGeorge | Qwell: take a look at this and explain yourself http://pastebin.ca/42978 |
| 05:08.53 | FuriousGeorge | since i blame you for all my problems smart guy ;) |
| 05:09.11 | username_taken | phonecalls work fine, but say if I dial message bank I don't hear anything, but the log shows it playing sounds to me |
| 05:09.41 | FuriousGeorge | seriously though, someone explain that to me. two calls, the top one has a valid CID and is no problem, the second call has no CID so you see * set it to some dumb shit |
| 05:10.20 | FuriousGeorge | did we integrate SIMPLE into * yet? |
| 05:11.13 | tuxinator_linux | nope, * is still pretty complex |
| 05:11.25 | tuxinator_linux | (someone had to say it) |
| 05:11.32 | lovinit | * = asterisk mmm |
| 05:12.04 | FuriousGeorge | tuxinator_linux: simple as in sip messaging |
| 05:12.12 | FuriousGeorge | not the quality of simplicity :) |
| 05:12.18 | tuxinator_linux | FuriousGeorge: ahh, that would be handing |
| 05:12.27 | tuxinator_linux | s/handing/handy |
| 05:12.31 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 05:12.57 | FuriousGeorge | i heard file did it but it only worked wtih SIP so it didnt get integrated into the main branch |
| 05:13.28 | FuriousGeorge | or something |
| 05:13.36 | tuxinator_linux | have to ask him in the morning, he's sleeping |
| 05:13.53 | FuriousGeorge | personally im rooting for mog's astjab |
| 05:13.56 | FuriousGeorge | go mogorman |
| 05:15.09 | *** join/#asterisk natmlt (n=Moriah@ip24-251-50-94.ph.ph.cox.net) |
| 05:15.39 | *** join/#asterisk mphill (n=datix@CPE-70-92-245-5.wi.res.rr.com) |
| 05:16.30 | FuriousGeorge | i should go to law school or somethiing soon |
| 05:17.28 | FuriousGeorge | be damned if im gonna be struggling with understanding how CID is(n't) working in 5 years :) |
| 05:18.10 | tuxinator_linux | law school == CID, how? |
| 05:18.16 | mphill | what is the most cost effective trunk implimentation? |
| 05:18.16 | *** join/#asterisk flot (n=flot@user241.hovrino.net) |
| 05:18.50 | FuriousGeorge | mphill: get it transplanted from a dead elephant |
| 05:18.59 | FuriousGeorge | j/k iax does trunking |
| 05:19.43 | FuriousGeorge | tuxinator_linux: just thinking that this whole world of linux and asterisk probably isnt for me. but im making some money and its fun for now, except when its not |
| 05:20.08 | FuriousGeorge | mphill: i heard Xorcom came out with a USB channel bank |
| 05:20.27 | tuxinator_linux | <PROTECTED> |
| 05:20.34 | tuxinator_linux | I get bored really easily |
| 05:20.38 | FuriousGeorge | i guess it better than bartending |
| 05:20.43 | flot | åÓÔØ ËÔÏ ÐÏ ÒÕÓÓËÉ ÐÏÎÉÍÁÅÔ ? :) |
| 05:20.49 | tuxinator_linux | but I'd also be good at law |
| 05:21.02 | tuxinator_linux | flot: in english this time |
| 05:21.24 | flot | ok :( |
| 05:22.06 | *** part/#asterisk Navman (n=icechat5@62.108.206.82) |
| 05:23.54 | FuriousGeorge | tuxinator_linux: you in the US? |
| 05:24.15 | tuxinator_linux | FuriousGeorge: yep |
| 05:24.20 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 05:24.32 | FuriousGeorge | what do you get on * when there is no CID on an incoming call? |
| 05:25.07 | tuxinator_linux | I which I had enough time to have played with * enough to know that |
| 05:25.39 | FuriousGeorge | ya, i got 4 servers in production they all do dsomething different. pick a different local cid name and number |
| 05:25.48 | iaxy | you get nothing. At least nothing shows up in cdr |
| 05:26.14 | tuxinator_linux | I imagine it should be find if no CID is received |
| 05:26.45 | tuxinator_linux | s/find/fine |
| 05:27.24 | tuxinator_linux | flot: Do you speak English? |
| 05:27.36 | FuriousGeorge | iaxy: i know i get nothing |
| 05:27.36 | mphill | FuriousGeorge: i guess, its just I use vonage now for 3 lines, i feel like 80 bucks is high considering its VoIP. I have no contract, if I wanted cheap unlimited VoIP, but PSTN access, what would you suggest? |
| 05:28.17 | FuriousGeorge | i dont know why * decides to insert "Maria" 103, or how to stop it from doing it w/o hacking it into my dialplan which i feel i shouldnt have to do |
| 05:28.26 | FuriousGeorge | mphill: i agree 80 is high |
| 05:29.18 | mogorman | whats that |
| 05:29.20 | FuriousGeorge | mphill: you want a local number or not? |
| 05:29.22 | mogorman | people saying my name |
| 05:29.48 | FuriousGeorge | mogorman: i was talking to tuxinator_linux about simple, and i told him i was waiting on astjab personally |
| 05:29.51 | FuriousGeorge | then i cheered for you |
| 05:29.55 | mogorman | yay |
| 05:29.57 | mphill | FuriousGeorge: yes |
| 05:30.08 | FuriousGeorge | mphill: where do you live? |
| 05:30.14 | mphill | wisconsin |
| 05:30.33 | tuxinator_linux | people really live in Wisconsin/ |
| 05:30.37 | tuxinator_linux | ? |
| 05:30.38 | mphill | omg |
| 05:30.44 | mphill | where are you from? |
| 05:30.55 | FuriousGeorge | thats really a state? :) j/k |
| 05:30.55 | tuxinator_linux | SoCal |
| 05:31.13 | tuxinator_linux | OC of SoCal |
| 05:31.16 | FuriousGeorge | mphill: i dont know where to get dids in wisc |
| 05:31.37 | tuxinator_linux | let's see, is it snowing in Wisconsin? |
| 05:31.37 | FuriousGeorge | ubt the going rate if you do find one is usually lke 1 or 2 bucks a month and 1 or 2 cents a minute incoming |
| 05:32.11 | FuriousGeorge | check out sipphone.org, they got a lot of dids for i believe like 5 bucks a month with only one incoming call allowed at a time |
| 05:32.23 | FuriousGeorge | mphill: the above was to you |
| 05:32.41 | FuriousGeorge | as for outgoing you should not pay more than 2Cents a min with 0/month |
| 05:32.59 | FuriousGeorge | so add that all up, it should be less than 80 |
| 05:33.13 | FuriousGeorge | btw, that was sipphone.com |
| 05:33.17 | tuxinator_linux | mphill: I used to live in Arizona, people live there also, thou I don't know why, a bit hot |
| 05:33.30 | FuriousGeorge | tuxinator_linux: my roomate is going there |
| 05:33.36 | FuriousGeorge | moving there i should say |
| 05:33.39 | flot | I connect * to mobile from bluetooth. How to me to adjust an asterisk on reception of phone calls? |
| 05:33.40 | FuriousGeorge | i dunno why |
| 05:33.44 | *** join/#asterisk yxa (n=diablo@58.185.90.98) |
| 05:33.54 | yxa | what's C7 ISUP and does * support that? |
| 05:34.12 | FuriousGeorge | ~C7ISUP |
| 05:34.19 | FuriousGeorge | i guess not |
| 05:34.42 | tuxinator_linux | flot: I am not familiar with mobile or bluetooth. Good luck. |
| 05:35.02 | yxa | FuriousGeorge what's that BTW? |
| 05:35.26 | FuriousGeorge | yxa: i dunno, ask google.i was hoping jbot would tell me. watch: |
| 05:35.27 | tuxinator_linux | FuriousGeorge: the nice part about Arizona is the girls don't like the hot weather, so they wear less clothes |
| 05:35.30 | FuriousGeorge | ~furiousgeorge |
| 05:35.31 | jbot | somebody said furiousgeorge was a knife-fighting monkey last seen with The Man with the Yellow Bat |
| 05:36.15 | FuriousGeorge | i dont wanna live anywhere w/o an ocean and good ehnic food, so there goes a lot of the contry |
| 05:36.21 | FuriousGeorge | *ethnic |
| 05:36.35 | tuxinator_linux | FuriousGeorge: where did you say you are? |
| 05:36.39 | FuriousGeorge | nj |
| 05:37.07 | tuxinator_linux | FuriousGeorge: you talk funny then |
| 05:37.23 | FuriousGeorge | yeah, but wadayougonnado |
| 05:37.31 | tuxinator_linux | ;-) |
| 05:38.19 | flot | I have connected an asterisk to a mobile phone through chan_bloetooth. Somebody did similar? |
| 05:38.22 | *** join/#asterisk wrench (n=signal@68-118-224-178.dhcp.oxfr.ma.charter.com) |
| 05:38.47 | FuriousGeorge | not me flot |
| 05:40.25 | FuriousGeorge | thats it, unless anyone can tell me why, im gonna have to hack mydialplan for when i dont get the cid info, to prevent * from displaying Maria 103 |
| 05:40.37 | tuxinator_linux | Watching the Olympics? |
| 05:41.10 | wrench | hey all, im looking for a simple lightweight solution to answer high volume of brief calls to verify DTMF input against an SQL database or flatfile and act accordingly - would asterisk be a viable solution? |
| 05:41.14 | FuriousGeorge | no i heard who won on mike and the mad dog already (local sports radio show) so what i mean is, unless someone can tell me why this happens, im gonna do something ugly |
| 05:41.17 | pcm | flot: do you have to peer with the phone before you can use it ? |
| 05:41.24 | tuxinator_linux | FuriousGeorge: I think I remember a setting somewhere to set the 'default' CID or something, that was a while ago when someone else had that problem |
| 05:41.26 | FuriousGeorge | wrench: yes |
| 05:41.33 | pcm | flot: or can it work as a regular phone that registers automatically ? |
| 05:42.01 | FuriousGeorge | tuxinator_linux: hmmmm |
| 05:42.05 | tuxinator_linux | wrench: high volume to a flat file, eww |
| 05:42.16 | wrench | ok definitely DB :) |
| 05:42.35 | wrench | ive been reading/learning about dialplans to build IVR, though i cant see how to make use of external data like a DB |
| 05:43.00 | tuxinator_linux | the only part of the Olympics I like is the girls ice skating, Sasha Cohen is cute |
| 05:43.12 | pcm | wrench: suppose you have ordering tickets system |
| 05:43.21 | pcm | wrench: you need to let the db know you have one less |
| 05:43.22 | pcm | heh |
| 05:43.26 | wrench | yep |
| 05:43.29 | tuxinator_linux | wrench: asterisk_addons, if I'm not mistaken |
| 05:43.58 | flot | pcm: With registration all is normal. Similar it is necessary to register channels in zapatel? I not install zapatel |
| 05:44.00 | wrench | hmm pardon my newbness, addons for db interaction! excellent |
| 05:44.15 | wrench | ill check them out now |
| 05:44.27 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 05:44.32 | tuxinator_linux | wrench: good luck, and get the wiki |
| 05:44.33 | FuriousGeorge | wrench exten= s,1,answer, exten=> s,2,bacground(input_dtmf,30) exten => s,3,db(users/${exten}=1) or something |
| 05:44.39 | tuxinator_linux | ~wiki |
| 05:45.03 | sevard | Why would the CLI say a peer is online when it's not? |
| 05:45.09 | wrench | tux/Furious >> thanks! |
| 05:46.25 | flot | pcm: http://www.thetechguide.com/howto/asterisk/chanbluetooth.html --- I have not understood about zapatel |
| 05:47.02 | tuxinator_linux | flot: are you on debian or redhat based? |
| 05:47.23 | tuxinator_linux | flot: or something else |
| 05:48.01 | flot | tuxinator_linux: Gentoo linux |
| 05:48.47 | tuxinator_linux | flot: not familar with gentoo, but I was asking because.... each distro has it's special things you need to do |
| 05:49.48 | tuxinator_linux | it looks like you need to install zaptel |
| 05:49.59 | tuxinator_linux | you have not installed zaptel? |
| 05:51.20 | FuriousGeorge | mogorman: you know anything about where to set a default CID, or better yet why my * servers seem to each select a random extension to display every time when there is no CID? |
| 05:51.27 | tuxinator_linux | flot: the directions on the web page you are looking at is very specific to asterisk@home |
| 05:51.41 | mogorman | That is odd... |
| 05:51.47 | tuxinator_linux | flot: you will have to adapt accordingly |
| 05:51.56 | flot | tuxinator_linux: I compile all from initial texts. OK. Has gone to put zapatel |
| 05:52.20 | *** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
| 05:52.23 | FuriousGeorge | sometimes i think asterisk has a personality, and it can be best described as "this guy knows nothing.... i'm gonna screw with him" |
| 05:52.44 | tuxinator_linux | FuriousGeorge: I bet you have that problem with a bunch of programs ;-) |
| 05:52.52 | FuriousGeorge | no just * |
| 05:53.23 | tuxinator_linux | Back in a few minutes |
| 05:53.54 | FuriousGeorge | one time it decided to start playing /var/spool/voicemail/0/default/TEMP instead of ..../busy or .../iunavail eventually i just renamed unavail to temp and did without a busy |
| 05:54.12 | FuriousGeorge | that server still does it |
| 05:54.25 | FuriousGeorge | on that one mailbox |
| 05:55.31 | mogorman | lol |
| 05:55.35 | FuriousGeorge | but this CID thing has been happening since 1.0.9 for me so i thought it was everyone |
| 05:55.37 | mogorman | sorry i am trying to fix some stuff |
| 05:55.40 | mogorman | i will be back in a while |
| 05:55.43 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 05:55.49 | *** join/#asterisk aaml (n=aml@c-67-166-40-6.hsd1.co.comcast.net) |
| 05:55.54 | FuriousGeorge | np, i was bitching to tuxinator_linux |
| 05:56.57 | *** part/#asterisk aaml (n=aml@c-67-166-40-6.hsd1.co.comcast.net) |
| 05:58.38 | *** join/#asterisk aaml (n=aml@c-67-166-40-6.hsd1.co.comcast.net) |
| 05:58.48 | aaml | . |
| 05:58.55 | *** part/#asterisk aaml (n=aml@c-67-166-40-6.hsd1.co.comcast.net) |
| 06:01.15 | luke-jr_ | How can I create an extension that handles all calls for the context? |
| 06:01.24 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 06:02.50 | tuxinator_linux | My wife was trying on swimsuits, wow she's still a hottie |
| 06:03.16 | Qwell | FuriousGeorge: temp is always played if it exists |
| 06:03.26 | Qwell | if it exists, busy and unavail aren't played, iirc |
| 06:04.01 | tuxinator_linux | FuriousGeorge: did you check to change logs to see if the CID issue is resolved |
| 06:04.14 | brockj49464 | How do I hangup on an incomming call, wait 30 seconds, then start a new outbound call? |
| 06:04.39 | *** join/#asterisk a2m (n=aml@c-67-166-40-6.hsd1.co.comcast.net) |
| 06:04.42 | tuxinator_linux | flot: How are you doing? What are you having trouble with? |
| 06:06.16 | tuxinator_linux | brockj49464: you may want to explain a little better on what you are trying to do. |
| 06:06.43 | flot | tuxinator_linux: I do not know as to adjust * on reception of the call acting on a mobile phone. |
| 06:07.25 | brockj49464 | I am trying to get it so when I call with my cellphone it will NOT answer but set my outbound callerID to X and call my cellphone back on my VOIP line and give me VM. |
| 06:08.29 | username_taken | what exactly does a "RED" alarm on a T1 mean ? ( te110p ) |
| 06:08.37 | tuxinator_linux | brockj49464: okay, but not sure why you would want to do that. |
| 06:08.51 | tuxinator_linux | username_taken: it's borked |
| 06:09.02 | tuxinator_linux | username_taken: just kidding.... google it |
| 06:09.09 | brockj49464 | Provider is stuid with how they determine what is mobile-mobile... |
| 06:09.17 | lovinit | whee my menus' work *does a lil dance* |
| 06:10.04 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:10.20 | Qwell | brockj49464: try Sprint... |
| 06:10.29 | tuxinator_linux | lovinit: good for you |
| 06:10.31 | Qwell | brockj49464: voicemail ISN'T mobile-mobile! |
| 06:10.50 | lovinit | Qwell :) |
| 06:11.14 | tuxinator_linux | brockj49464: do you have to count your minutes? |
| 06:11.35 | troyb1 | username_taken red light on a csu/dsu means a nuclear bomb is coming |
| 06:12.28 | brockj49464 | Yes I have to count my cellphone->non provider minutes but not mobile-mobile minutes. |
| 06:12.43 | username_taken | gee thanks for that troy, I'll get down to my bombshelter straight away |
| 06:12.48 | *** join/#asterisk MrBelvedr (n=tt@ip70-187-237-193.dc.dc.cox.net) |
| 06:13.07 | troyb1 | username_taken haha, it probably means there is a line issue, *depending on the purpose of that LED* |
| 06:13.11 | troyb1 | really it could mean many things |
| 06:13.27 | tuxinator_linux | username_taken: what's the user manual say |
| 06:13.28 | username_taken | zttool shows RED in the status too |
| 06:13.29 | troyb1 | usually Yellow is safe, Red is usually not a good thing. |
| 06:13.37 | FuriousGeorge | Qwell: is that it? why isnt it deleting temp when my user sets the CID? i'm gonna try that in a second |
| 06:13.45 | Qwell | huh? |
| 06:14.08 | FuriousGeorge | tuxinator_linux: i checked master.csv and as far as it knows Maria 103 is the one who called everytime its unknown |
| 06:14.24 | troyb1 | FuriousGeorge your name is Martia? |
| 06:14.26 | Qwell | what does voicemail have to do with callerid? |
| 06:14.32 | FuriousGeorge | Qwell: i meant to say "sets the voicemail" |
| 06:14.53 | Qwell | temp won't be deleted until the user removes the temp message |
| 06:14.56 | troyb1 | Qwell ironically i know a lot of places exten => mailbox (no Dial) |
| 06:15.00 | tuxinator_linux | FuriousGeorge: hmm, Maria is popular |
| 06:15.07 | FuriousGeorge | my name isnt maria |
| 06:15.19 | FuriousGeorge | thats the name * gives unknown CIDs |
| 06:15.36 | tuxinator_linux | find it and change it? |
| 06:15.40 | troyb1 | Qwell i must say im very impressed with asterisk, but the learning curve sucks |
| 06:16.03 | troyb1 | username_taken do you have PRI terminated over the T1? |
| 06:16.14 | FuriousGeorge | Qwell: thats what i mean, my user sets the voicemail for u0 and b0 (general mail box), * creates unavail and busy accordingly, but plays temp |
| 06:16.40 | *** join/#asterisk lucasjb (n=lucas@mail.stabat.com) |
| 06:16.41 | Qwell | FuriousGeorge: yes...if temp is set, the other two won't be played, regardless of settings |
| 06:17.03 | username_taken | troyb1: yes i do |
| 06:17.08 | FuriousGeorge | Qwell: so all i ever had to do was delete temp, huh, lemme see |
| 06:17.19 | Qwell | FuriousGeorge: it can be done from within VM |
| 06:17.23 | Qwell | in fact... |
| 06:17.31 | troyb1 | username_taken so you have it split into voice/data? |
| 06:17.39 | Qwell | there is a patch on the bug tracker, that will remind you that you have temp set, when you login to VM |
| 06:17.41 | username_taken | it's all voice |
| 06:17.46 | Qwell | and allow you the option of removing it |
| 06:17.55 | troyb1 | ah okay, well if you have 23 channels up i'd suggest calling the telco, better safe then sorry |
| 06:18.10 | username_taken | yeah. am waiting on hold for the telco :) |
| 06:18.11 | *** join/#asterisk salviadud (n=mofo@201.133.209.101) |
| 06:18.19 | username_taken | could be a while |
| 06:18.23 | troyb1 | username_taken who are you with? |
| 06:18.31 | username_taken | optus ( in Australia ) |
| 06:18.51 | troyb1 | australia is a monopoly :) |
| 06:19.00 | flot | tuxinator_linux: I call from * and i have error: Unable to create channel of type 'BLT' (cause 0 - Unknown) |
| 06:19.03 | username_taken | not quite, but not far off |
| 06:19.14 | lucasjb | Hiyas, question: I have a Polycom IP 300 connected to my Asterisk system with NAT in between. I can call the Polycom handset with X-Lite connected to the same Asterisk switch, or on my regular phone through my upstream provider. I can also call from the Polycom to X-Lite. BUT I can't call from Polycom through upstream provider to a regular phone. The phone rings, but when answered, there's no audio. Where do I start looking to resolve this problem? |
| 06:19.15 | troyb1 | yeah i was being a little conservative there |
| 06:19.28 | salviadud | is this a proper way to answer an incoming call from a pstn line on a sipura 3000? |
| 06:19.31 | salviadud | http://pastebin.ca/42982 |
| 06:19.34 | luke-jr_ | How can I create an extension that handles all the calls for the context? |
| 06:19.43 | username_taken | anyways am gonna pull the te100p out of the production box and chuck it in a box i can play with without upsetting users |
| 06:19.58 | brockj49464 | So is there a way to start another call and hangup on the incoming one? |
| 06:20.01 | troyb1 | username_taken i know here in Canada you can get VPRI's around 300 but thats virtual, (no physical connection), data T1's go for 400-800 |
| 06:20.03 | tuxinator_linux | flot: I wish I could help you, let me see what I can found out about 'unable to create channel of type...' |
| 06:20.35 | tuxinator_linux | what's a virtual PRI? how is it different from a PRI? |
| 06:20.43 | salviadud | the sipura should answer it and send it to that context immediateley |
| 06:20.48 | troyb1 | tuxinator_linux there is no physical line termination, its routed over the net |
| 06:20.54 | troyb1 | virtual, (no physical connection), data T1's go for 400-800 |
| 06:20.54 | troyb1 | [01:20] <tuxinator_linux> flot: I wish I could help you, let me see what I can found out about 'unable to create channel of type...' |
| 06:20.54 | troyb1 | [01:20] <tuxinator_linux> what's a virtual PRI? how is it different from a PRI? |
| 06:20.54 | troyb1 | [01:20] <salviadud> the sipura should answer it and send it to that context immediateley |
| 06:20.55 | troyb1 | [01:20] <troyb1> tuxinator_linux there is no physical line termination, its routed over the net |
| 06:20.55 | salviadud | ?? |
| 06:20.56 | troyb1 | fux |
| 06:20.58 | troyb1 | im so sorry :( |
| 06:21.02 | FuriousGeorge | Qwell: sure enough, that worked. you have no idea how many times i asked why that was happening |
| 06:21.10 | Qwell | FuriousGeorge: none? |
| 06:21.10 | Qwell | ;] |
| 06:21.33 | troyb1 | username_taken so technically you could get a PRI + Data T1 for like 700 :P |
| 06:22.05 | flot | I write to exten Dial(BLT/number@K750i) - error |
| 06:22.21 | lucasjb | luke-jr_, Are you asking how to say "all calls arriving in this context should be sent to that extension"? |
| 06:22.30 | troyb1 | flot bacon lettuce and tomatoe? BLT |
| 06:22.36 | FuriousGeorge | you know, i've helped a few people set their voicemail, and i dont remember having to delete the temp message before |
| 06:22.45 | flot | I write to exten Dial(BLT/K750i/number) - not error , but not call |
| 06:22.49 | Qwell | FuriousGeorge: temp is something the user records |
| 06:23.02 | luke-jr_ | lucasjb: yes |
| 06:23.07 | troyb1 | username_taken i'd be concerned come peek call hour *grin* |
| 06:23.13 | FuriousGeorge | Qwell: for me it was allison saying "Please leave a message after the tone" |
| 06:23.39 | yxa | whats the difference between PRI and E1? E1 for data cost much more but they are can both be terminated using a E1 card? |
| 06:23.40 | Qwell | temp can be 0s |
| 06:23.54 | Qwell | PRI runs on top of E1 |
| 06:23.59 | troyb1 | yxa two totally different things :) |
| 06:24.08 | Qwell | troyb1: not different at all |
| 06:24.10 | troyb1 | Qwell remember e1 = 2.0Mbps T1 = 1.544, standard depends on country |
| 06:24.26 | troyb1 | shit i misunderstood :| |
| 06:24.44 | tuxinator_linux | any downside to using a VPRI? |
| 06:24.45 | yxa | so can i run a E1 for both voice and data? |
| 06:24.57 | flot | troyb1: I have not understood you |
| 06:25.00 | troyb1 | if your telco wants to sell you a voice / data E1 |
| 06:25.13 | troyb1 | flot i was just making a joke :) |
| 06:25.22 | lucasjb | luke-jr_, OK well it depends on what kind of extension you want to send the call to, but for SIP I think you could probably do it with s,1,Dial(SIP/${EXTEN}) |
| 06:25.41 | luke-jr_ | lucasjb: that will only work if the 's' extension is dialed in the context |
| 06:26.19 | troyb1 | ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer |
| 06:26.35 | flot | troyb1: Clearly, I badly understand English :))) |
| 06:27.02 | troyb1 | flot my bad :( maybe i should retake english ;_; |
| 06:27.13 | yxa | so i can have 30 voice lines and yet have have a 2Mbps link? |
| 06:27.31 | tuxinator_linux | flot: I'm curious, what is your primary language? |
| 06:27.34 | troyb1 | yxa a PRI is 23+1 |
| 06:27.44 | yxa | troyb1 E1 |
| 06:27.46 | flot | tuxinator_linux: russian |
| 06:27.54 | luke-jr_ | lucasjb: I want _., but without catching fake extensions like 'h' |
| 06:27.58 | troyb1 | if you want more then 24 then you need to ask your telco for a larger solution |
| 06:28.03 | [av]bani | \o/ |
| 06:28.06 | Qwell | mog_work: we need a translator! :) |
| 06:28.18 | troyb1 | Qwell i fail? |
| 06:28.55 | tuxinator_linux | flot: there are a few people on here that speak russion, but not many. |
| 06:28.56 | lucasjb | luke-jr_, Well in that case, sounds like you understand contexts better than me, but I'd be surprised if you couldn't have a rule that matched anything. |
| 06:29.18 | troyb1 | yxa E1 = speed of link A PRI is always 23 + 1 right Qwell? |
| 06:29.19 | luke-jr_ | lucasjb: the problem is having it *not* match non-dialed extensions |
| 06:29.42 | Qwell | troyb1: no, I'm pretty sure PRI can run on E1 also |
| 06:29.48 | yxa | troyb1 i think we call them ISDN30 here |
| 06:30.05 | flot | tuxinator_linux: russian channel * - #asteriskru :))) |
| 06:30.26 | troyb1 | Qwell yeah.. im just trying to say a PRI is always 23+1 regardless of the capacity they are routed over? |
| 06:30.40 | Qwell | troyb1: if PRI runs over E1...no |
| 06:30.42 | *** join/#asterisk Eggplant (i=No@dsl-438.cascadeaccess.com) |
| 06:30.53 | tuxinator_linux | flot: look at that, a channel for Russians |
| 06:30.53 | troyb1 | so a PRI isnt always 23+1? |
| 06:32.51 | Qwell | troyb1: if PRI runs over E1...no |
| 06:32.58 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 06:33.08 | troyb1 | Qwell but.. a PRI can run over anything? |
| 06:37.29 | [av]bani | a PRI can be 24 :)) |
| 06:38.28 | username_taken | an E1 can be 29+1 afaik |
| 06:38.34 | [av]bani | or 30 :)) |
| 06:39.03 | troyb1 | so thats what an extra 500k gets you :P |
| 06:40.54 | username_taken | yeah, 30+1 i meant |
| 06:40.59 | username_taken | wasn't thinking straight |
| 06:41.07 | blitzrage | there's no such thing as ANI over SIP, right? |
| 06:41.07 | username_taken | atm my e1 gives me 0+0 :( |
| 06:42.05 | *** part/#asterisk propagandhi (n=opera@CPE-144-131-132-107.nsw.bigpond.net.au) |
| 06:42.15 | troyb1 | username_taken haha :) |
| 06:44.23 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 06:48.59 | FuriousGeorge | im having trouble with this newfangled n priority and its possible labels |
| 06:49.02 | FuriousGeorge | http://pastebin.ca/42984 |
| 06:50.48 | FuriousGeorge | can anyone see what im doing wrong there |
| 06:51.49 | [av]bani | you cant goto n |
| 06:52.53 | [av]bani | n just prevents your from having to manually number your priorities, it doesnt eliminate them |
| 06:53.04 | [av]bani | goto still requires a number |
| 06:55.04 | evilbuny | [av]bani: you don't need numbers, you can use labels |
| 06:55.56 | evilbuny | FuriousGeorge: goto(dial) |
| 06:56.02 | FuriousGeorge | evilbuny: right |
| 06:56.02 | evilbuny | not goto(n(dial)) |
| 06:56.11 | FuriousGeorge | i just found the wiki page |
| 06:56.12 | FuriousGeorge | thanks |
| 06:56.25 | evilbuny | :) |
| 06:56.41 | [av]bani | yay |
| 06:56.57 | [av]bani | now do it in ael |
| 06:57.02 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 06:57.34 | evilbuny | FuriousGeorge: you can also drop the s, bit to |
| 06:57.39 | evilbuny | it's not needed |
| 06:57.48 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.adnc.com) |
| 06:58.49 | evilbuny | gotoif($[${CALLERIDNAME}=Maria]?open,s-fixcid,1|open,s,n(dial)) => gotoif($[${CALLERIDNAME}=Maria]?open,s-fixcid,1|dial) |
| 06:58.51 | evilbuny | etc |
| 06:59.07 | [av]bani | FuriousGeorge: this is some ex-girlfriend script? :)) |
| 07:00.20 | *** join/#asterisk mkl1525 (n=daniel@212.80.239.24) |
| 07:00.35 | FuriousGeorge | [av]bani: its not. by default thats the cid * displays when it gets none from the telco |
| 07:00.38 | luke-jr_ | What defines the default voicemail context? |
| 07:01.20 | FuriousGeorge | evilbuny: i know, since it wasnt working before i went as verbose as possible |
| 07:01.21 | evilbuny | [default] |
| 07:01.28 | evilbuny | FuriousGeorge: ahh ok :) |
| 07:02.36 | *** join/#asterisk brookshire[home] (n=matt@68.62.235.16) |
| 07:10.46 | mkl1525 | Hi, trying to integrate * into an existing pbx. The team function of this pbx rings all phones in a team if a call comes in. Problem is that * takes the call immediately and so the pbx won't ring all the other phones anymore. Is there a way to ring the sip phone on * without really taking the call? Have tried with wait but this doesn't work - any suggestions? |
| 07:11.49 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 07:16.46 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
| 07:17.22 | Psykick | hey guys |
| 07:21.12 | *** join/#asterisk X-Gen (n=x-gen@dsl-146-122-162.telkomadsl.co.za) |
| 07:21.22 | X-Gen | hey freaks |
| 07:21.22 | *** part/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 07:21.48 | blitzrage | heh cool, I just built a neat little AGI application to check with my SS to make sure E911 is active on a particular line |
| 07:22.21 | Qwell | SS? |
| 07:22.57 | blitzrage | softswitch |
| 07:23.01 | Qwell | right |
| 07:23.03 | X-Gen | from "show translation" which shows how many ms a codec adds to a conversion, how does one figure out max no of active channels that a CPU can handle ? |
| 07:23.17 | blitzrage | SIP doesn't have the concept of ANI does it? |
| 07:23.29 | blitzrage | I know callerID is sent via Remote-Party-ID field |
| 07:24.17 | X-Gen | max allowable lag (300ms) div by codec lag (30ms) = max of 10 calls |
| 07:29.39 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:29.52 | *** part/#asterisk username_taken (n=stinky@203.202.184.90) |
| 07:33.26 | *** join/#asterisk oej (n=oej@fuckoff012231-1-3.clients.easynet.fr) |
| 07:33.32 | DeadZen | ? |
| 07:34.11 | FuriousGeorge | errr, why is allison saying "...is unavailable." after my voicemail message |
| 07:34.21 | Qwell | FuriousGeorge: u |
| 07:34.30 | FuriousGeorge | Qwell: well obviously |
| 07:34.43 | FuriousGeorge | Qwell: im looking for a more precise cause |
| 07:34.48 | Qwell | so...whats the problem? |
| 07:35.01 | blitzrage | oej: !!! are you there? I have a question :) |
| 07:35.09 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 07:35.15 | FuriousGeorge | :) everyother mailbox she goes right into "Please leave a message." |
| 07:35.18 | DeadZen | http://www.ncbi.nlm.nih.gov/Structure/CN3D/cn3dinstall.shtml |
| 07:35.45 | FuriousGeorge | the one we just fixed by deleting temp she goes into "...is unavailable." then does the "Please leave a message" |
| 07:36.11 | FuriousGeorge | oej is here? tell him i say the sineapps article and i promise to test metermaid |
| 07:36.12 | Qwell | so stop using option u to voicemail() |
| 07:36.15 | FuriousGeorge | not that he knows me |
| 07:36.48 | oej | FuriousGeorge: Cool |
| 07:36.53 | FuriousGeorge | Qwell: your not getting it, cuz im not explaining right |
| 07:37.08 | FuriousGeorge | oej: my pleasure |
| 07:38.18 | FuriousGeorge | Qwell: every other mailbox, when i dial u${MAILBOXNUM}, * plays the unavailable message followed by allison saying "Please leave a message... blah blah" |
| 07:38.42 | Qwell | If one exists |
| 07:38.46 | Qwell | otherwise, you'll get Allison |
| 07:39.10 | FuriousGeorge | on this particular mailbox, it plays the message, then says "... is unavailable. Please leave a message... blah blah" |
| 07:39.27 | FuriousGeorge | like it was expecting my name not my greeting |
| 07:39.47 | Qwell | yeah...intended behavior |
| 07:39.51 | FuriousGeorge | maybe before when i replaced temp with alison i remobed unavail.... brb |
| 07:40.10 | FuriousGeorge | i mean when i replaced temp with unavail cuz i couldnt figure out why it was playing temp |
| 07:40.56 | [av]bani | FuriousGeorge: got that agi working? keysystem.agi :) |
| 07:41.07 | FuriousGeorge | yup, thats what i did, my bad |
| 07:41.32 | FuriousGeorge | [av]bani: X-Rob talked me out of it, but im sure someone else could do it easily |
| 07:41.38 | [av]bani | bah |
| 07:41.46 | FuriousGeorge | someone with a better understanding of the manager interface |
| 07:41.52 | FuriousGeorge | somone like [av]bani |
| 07:41.55 | [av]bani | would you pay for one :)) |
| 07:42.26 | FuriousGeorge | [av]bani: instead i promised oej i would test metermaid, so it will make it into 1.4 and i wont have to |
| 07:42.36 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 07:43.08 | [Airwolf] | Can anyone tell me how I can redirect a extension in my extensions.conf to another extension ? |
| 07:43.13 | [Airwolf] | Of isn't that possible ? |
| 07:43.15 | [Airwolf] | or |
| 07:43.20 | Qwell | Goto() |
| 07:43.57 | [Airwolf] | Qwell, thank you |
| 07:45.14 | [av]bani | FuriousGeorge: would you pay for a key system emulator :)) |
| 07:45.50 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 07:46.13 | FuriousGeorge | [av]bani: no, i just thought that would be snazzy for the barge and the reverse x-fer functionality |
| 07:46.14 | franck | Hi |
| 07:46.21 | FuriousGeorge | i'd much rather not use meetme |
| 07:46.42 | franck | I have a lot polarity reversed on my zaptel card.. What is happening? |
| 07:47.17 | [av]bani | FuriousGeorge: well, one could just wait for bridged appearances... |
| 07:47.24 | [av]bani | in 6mo or so :/ |
| 07:47.38 | FuriousGeorge | [av]bani: plus it wouldnt emulate it right. how are you gonna be able to kill the room seemlessly, or get a caller out of the conference once hes in there |
| 07:47.50 | FuriousGeorge | as for the second question you definately need the api for that |
| 07:48.01 | FuriousGeorge | and if i could solve those problems, it would take 6 months |
| 07:48.29 | [av]bani | FuriousGeorge: you can do anything from agi |
| 07:48.46 | [av]bani | forcibly transferring out of the conference. .. shrug |
| 07:48.47 | FuriousGeorge | can you assign device states to parking spots? |
| 07:48.58 | FuriousGeorge | if you can tell X-Rob and me about it |
| 07:49.19 | [av]bani | no idea, maybe hacking the asterisk code would do it |
| 07:50.12 | FuriousGeorge | yeah throw in a "NoOP" ${PARKINGSPOTNUMBER} in there :) |
| 07:51.20 | FuriousGeorge | im gonna watch nova and hit the sack |
| 07:51.52 | FuriousGeorge | [av]bani: you know what ive come to conclude, if this could have been done someone as bright as myself with more * experience would have :) |
| 07:52.22 | FuriousGeorge | ask oej if he appears again in here. like x-rob said to me yesterday "application hints are not easy" |
| 07:52.44 | [av]bani | not yet, but 1.4 is supposed to fix that |
| 07:52.49 | [av]bani | in 6mo :/ |
| 07:52.58 | Qwell | application hints aren't easy, but they sure would be nice |
| 07:53.10 | Qwell | like, say...meetme |
| 07:53.32 | Qwell | you could hack around it, but... |
| 07:53.39 | FuriousGeorge | i know, im hoping for that and mogs xmpp branch, even though im not sure how its gonna work |
| 07:54.10 | FuriousGeorge | Qwell: x-rob already yelled at me for thinking about it with meetme :) |
| 07:55.07 | FuriousGeorge | Qwell: but it would be cool cuz you have built in barge and reverse transfer by led |
| 07:55.08 | [av]bani | Qwell: you could hack individual apps to provide hints, but... ick |
| 07:55.13 | Qwell | [av]bani: yeah |
| 07:55.23 | [av]bani | hopefully a unified API will fix that in 1.4 |
| 07:55.27 | [av]bani | then any app can provide hints |
| 07:55.35 | [av]bani | that will r0x0r |
| 07:55.55 | [av]bani | of course, wont fix the fundamental problems with the polycom - 7 buddy limit |
| 07:56.29 | [av]bani | youd want a sort of 'virtual extension' i suppose |
| 07:56.41 | [av]bani | with a bridged appearance |
| 07:57.38 | FuriousGeorge | ooohhh, nova is about the big bang again. go string theory |
| 07:58.36 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 08:08.03 | Qwell | bed |
| 08:09.21 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:11.16 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 08:11.45 | sl16 | Failed to authenticate on INVITE ? |
| 08:12.19 | sl16 | i am trying outbound call to some number with hardware sip phone via * |
| 08:12.29 | sl16 | inbounds are ok |
| 08:14.01 | sl16 | when try to set authusername and pass on the phone and the authusername and pass the phone don't what to register in * |
| 08:14.06 | sl16 | what to do ? :( |
| 08:16.18 | sl16 | is it possible to dial some number via my sip VoIP provider with any of the internal phones |
| 08:17.22 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-204.claranet.co.uk) |
| 08:18.19 | *** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net) |
| 08:20.09 | franck | When I dial I get RING/NO RING and sometimes a NO BAT |
| 08:20.23 | franck | Asterisk does not pick up the line, why? |
| 08:22.24 | franck | I have also a couple of lines that have a lot of polarity switch.... |
| 08:23.42 | franck | How can I tell asterisk to pick up the phone after one ring instead of 3? |
| 08:35.34 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 08:35.51 | *** join/#asterisk Himeko (i=himeko@S01060040ca128fc3.ed.shawcable.net) |
| 08:39.05 | flot | Who used chan_bluetooth? |
| 08:40.49 | *** join/#asterisk Psykick (n=anon@60-234-220-253.bitstream.orcon.net.nz) |
| 08:41.01 | Psykick | hi guys |
| 08:41.15 | Psykick | does asterisk listen on a default interface or does it listen on all interfaces? |
| 08:41.18 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:42.34 | X-Gen | netstat -lap | more will tell you Psykick |
| 08:43.18 | Psykick | hmmm |
| 08:43.20 | X-Gen | if its listening on *:port that means all afaik |
| 08:43.52 | Psykick | ok |
| 08:43.53 | franck | why do I get some nobat when there is a call in? |
| 08:44.03 | Psykick | where can I specify in asterisk for it to listen on all interfaces? |
| 08:44.55 | Psykick | hmm |
| 08:44.58 | Psykick | never mind |
| 08:45.07 | Psykick | I see in iax.conf its bound to 0.0.0.0 |
| 08:45.36 | Psykick | strange thing is I'm not seeing any traffic coming in on the interface I'm expecting to see data coming in on |
| 08:48.47 | PakiPenguin | anyone here uses astguiclient? |
| 08:49.07 | Psykick | this is really weird |
| 08:50.33 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 08:51.32 | Psykick | only think I can think of is that asterisk may need to be restarted |
| 08:55.22 | Psykick | does anyone have any ideas why I'm not seeing any connections coming in on public IP address |
| 08:57.01 | trixter | cause no one wants to connect to your box |
| 08:57.16 | trixter | either that or your internet connection to that box is not configured |
| 08:58.30 | Psykick | I'm trying to connect to it from home |
| 08:58.56 | Psykick | I also have the default route pointing to the gateway |
| 08:59.03 | trixter | connect how? |
| 08:59.11 | Psykick | using a softphone |
| 08:59.23 | trixter | you asked a very general question then followed up with generic info |
| 08:59.41 | trixter | ok, so a softphone wont connect, do you know that it isnt connecting? or is something else going on like a NAT issue? |
| 09:00.01 | Psykick | I'm not natting the connections through |
| 09:00.09 | Psykick | the asterisk box has a public IP address |
| 09:00.17 | Psykick | that address is part of my dmz |
| 09:00.32 | Psykick | I'm configured the firewall to allow UDP 4569 in and out of the dmz |
| 09:00.50 | Psykick | I meant I've configured the firewall to allow 4569 to the dmz |
| 09:01.18 | Psykick | in both directions |
| 09:01.39 | Psykick | from the dmz to the net and from the net to the dmz |
| 09:04.11 | X-Gen | Psykick: use something like netcat to test your connectivity. it can listen on any udp/tcp port. make sure that remote connections to it are working. |
| 09:05.06 | Psykick | netcat? |
| 09:05.48 | X-Gen | or use something like tcpdump -I interface to see what traffic is comming in on that interface |
| 09:06.51 | Psykick | been doing that |
| 09:06.55 | Psykick | and I don't see anything |
| 09:07.55 | Psykick | eg I'm not seeing any UDP packets coming in |
| 09:08.06 | Psykick | and I don't see anything hitting the firewall either |
| 09:08.09 | X-Gen | your tcpdump setup for udp traffic ? |
| 09:08.39 | Psykick | no I'm using tethereal |
| 09:09.16 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:12.27 | *** join/#asterisk real-dev (n=dev@master.netzflocken.de) |
| 09:14.40 | *** join/#asterisk ChrisUK (n=chris@82.108.126.170) |
| 09:14.42 | real-dev | hi, I have a question regarding setting calleridname for an incoming call. Can this be done via AMI setvar or only via AGI set caller(name)? |
| 09:15.41 | *** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net) |
| 09:16.14 | *** join/#asterisk jebba (n=jebba@70.90.114.101) |
| 09:19.12 | dezent | i got a problem with echo and meetme app.. im using ztdummy and i wonder if there is a way to use echocancel and echotraining anyway ? from what ive seen if using ztdummy i dont use zapata.conf.. is that correct ? |
| 09:21.29 | trixter | ztdummy doesnt really do anything but provide a timing interface. zapata.conf contains the configuration for hardware |
| 09:21.43 | trixter | that hardware is specifically limited to what chan_zap is responsible for |
| 09:21.54 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 09:22.21 | trixter | as such what channel would you do echo cancelation on? it certainly wont be a zap channel so zapata.conf wont apply |
| 09:23.05 | dezent | trixter: no its soft-phone -> asterisk <- softphone, and calls from my pstn gateway provider.. |
| 09:23.09 | *** join/#asterisk t0ke (n=kaka@125.Red-83-53-40.dynamicIP.rima-tde.net) |
| 09:23.30 | *** join/#asterisk astar` (n=astar@ANantes-154-1-93-196.w86-210.abo.wanadoo.fr) |
| 09:23.34 | t0ke | morning |
| 09:23.35 | trixter | so why do you think that echo cacncelation on a zap channel would have any effect on that call setup? |
| 09:23.56 | t0ke | anyone have rx and tx gain value for Telefonica Spain lines? |
| 09:24.10 | trixter | t0ke: what type of line? |
| 09:24.26 | t0ke | trixter, I am configuring FXO ports |
| 09:25.00 | dezent | trixter: i dont.. but im trying to find a solution and my question was if its possible to do that anyway |
| 09:25.15 | trixter | to a point that would depend on how far you are from the CO, some lines may not need as much gain as others, you may want to play with ztmonitor and see what works well for you where you are with all the cross connects that are between you and the phone company |
| 09:25.43 | *** part/#asterisk Z_God (n=Z_God@jabber.xs4all.nl) |
| 09:25.53 | trixter | dezent: well even if it was, I am trying to get you to think about what channels are actually used and how configuration on an unused channel type wouldnt affect what is actually being used |
| 09:25.53 | t0ke | trixter, I thinked would to exist any default values by companies |
| 09:26.06 | trixter | its more of a way of thinking about it so you can try to figure out the problem |
| 09:26.15 | trixter | typically with softphones the echo is at the end point not in the middle |
| 09:26.30 | trixter | someone has a mic and speakers and the mic picks up the speakers (VERY common) - solution get a headset |
| 09:26.41 | trixter | other problems are bad sound cards that are mixing the audio (less common) |
| 09:27.12 | trixter | t0ke: the defaults are 0dB for both, but you may need to adjust that depending on the physical properties of the wire that goes between you and the phone company |
| 09:27.23 | trixter | t0ke: are you having a specific problem that you need to adjust the gain? |
| 09:27.53 | dezent | trixter: that sounds very reasonable, thank you, i have a try |
| 09:28.45 | t0ke | trixter, no..I was configuring one FXO ports and I thinked I had to set that value. I will triy with 0db. tnks |
| 09:30.16 | trixter | t0ke: often you dont have to do anything but you should at least try it there first to see... also if you have echo you may want to switch from the default echo canceler, there are a couple, I think steve has one that is pretty good.. The default one many people have had problems with and as a result have to turn the rx gain down so low you cant really hear what is being said :( |
| 09:33.28 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:33.56 | X-Gen | speex is quite cpu intensive codec |
| 09:34.51 | trixter | yup |
| 09:34.58 | trixter | over 2x g.729 |
| 09:35.11 | trixter | if that is true then it appears that it takes about 60MHz on a modern cpu to do it which sucks |
| 09:35.27 | trixter | 60MHz per channel jsut for that |
| 09:35.41 | trixter | echo cancelation and g.729 each take 30MHz on a modern cpu according to industry experts |
| 09:35.53 | trixter | and speex takes more bandwidth than g.729 |
| 09:36.10 | trixter | but its the only low bandwidth codec supported on my ipaq softphone so I use it :) |
| 09:36.18 | X-Gen | from "show translation" which shows how many ms a codec adds to a conversion, how does one figure out max no of active channels that a CPU can handle ? |
| 09:36.24 | X-Gen | max allowable lag (300ms) div by codec lag (30ms) = max of 10 calls |
| 09:36.33 | X-Gen | is that correct ? |
| 09:36.37 | trixter | not totally |
| 09:37.01 | trixter | becuase if you are running other applications it takes load and such |
| 09:37.21 | trixter | I do 100 channels of g.729 with little lag on a single cpu box |
| 09:37.27 | X-Gen | so how do u work out the max channels ? |
| 09:38.01 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 09:38.12 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 09:38.20 | trixter | a rule of thumb would be 30MHz for g.729 per channel, plus if you run an application what that application takes, plus if you have horrible hardware that generates a lot of interrupts how much ever it needs, plus if you do anything else like echo cancelation etc add that |
| 09:38.20 | evilbuny | X-Gen: you also have bandwidth issues |
| 09:38.53 | X-Gen | evilbuny: on 100Mbit ? |
| 09:38.57 | evilbuny | k |
| 09:39.06 | X-Gen | :P |
| 09:39.08 | evilbuny | trixter: pass through :) |
| 09:39.13 | evilbuny | someone elses problem the :) |
| 09:39.22 | trixter | in theory it would seem to make sense that if it takes 30ms to translate that 10 channels means a 300ms lag but that doesnt seem to work right for some reason |
| 09:39.28 | trixter | passthrough != translation |
| 09:39.37 | evilbuny | exactly :) |
| 09:40.15 | trixter | but stuff like zap based cards (the driver some cards like sangoma can actually save you a TON of interrupts) require more IRQs generated which requires more context switches which increase cpu load |
| 09:40.38 | trixter | sangoma without zap can be t old to 1. use the hardware based g.729 that some of em have, and 2. to give you 20ms frames for example instead of a constant stream |
| 09:40.45 | trixter | which saves you a ton of cpu power right there |
| 09:41.09 | trixter | hardware echo cancelers leave more cpu available for other things |
| 09:41.27 | X-Gen | so there is no way to workout the max no. of channels u can run with show translation ? |
| 09:41.38 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 09:41.45 | heroine | hi |
| 09:42.28 | trixter | this box says 8 in show translation, however it can do 100 channels without a problem and there is not 800ms lag |
| 09:42.35 | trixter | so I really dont know |
| 09:43.01 | evilbuny | that makes no sense |
| 09:43.10 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:43.21 | *** join/#asterisk brettnem (n=brettnem@nemeroff.com) |
| 09:43.22 | evilbuny | wouldn't it be CPU (eg 2Ghz) 2000/30 |
| 09:43.52 | *** join/#asterisk power1 (i=daemon@dsl-146-46-84.telkomadsl.co.za) |
| 09:43.52 | trixter | I think show translation lies :P |
| 09:43.56 | evilbuny | no |
| 09:44.01 | evilbuny | it gives you the time to translate |
| 09:44.09 | evilbuny | which isn't directly related to channels per cpu |
| 09:44.21 | X-Gen | time 2 translate 1 second of sound |
| 09:44.48 | trixter | if it takes 8ms on an idle cpu why doesnt 100 channels cause 800ms lag? since all 100 are concurrent so the other 99 channels would interfere with that one being processed |
| 09:44.54 | X-Gen | it is evilbuny, faster your cpu more channels u can translate |
| 09:45.01 | power1 | Hey all, I am going to go out now and buy a board and cpu for an asterisk box.... gonna use 10 sip phones and one sip trunk. I want it to be super stable on compatible with a tdm400p card....any suggestions on what board / cpu I should go for? |
| 09:45.04 | trixter | oh evilbunny btw thanks for my tollfree listing in e164 :) |
| 09:45.05 | evilbuny | X-Gen: you're thinking linearly |
| 09:45.09 | evilbuny | CPUs have pipes |
| 09:45.24 | trixter | the cpu isnt doing only one thing though |
| 09:45.33 | evilbuny | exactly my point |
| 09:45.36 | trixter | there are several tasks running on this box, so those are always getting flushed |
| 09:45.40 | evilbuny | they can process multiple things per hertz |
| 09:46.16 | trixter | I think you are looking at it different, encoding g.729 takes about 30MHz on a modern cpu by in large (decoding is easier than encoding) |
| 09:46.26 | trixter | and that is what he is talking about |
| 09:46.35 | trixter | intel was the one that came up with the 30MHz figure as well |
| 09:46.36 | X-Gen | trixter: where did u get that 30Mhz value from ? |
| 09:46.43 | trixter | but the exact g.729 implementation matters there |
| 09:47.03 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 09:47.21 | trixter | the 30MHz for software echo cancelaqtion came from a sangoma guy that was talking about hardware vs software echo cancelers |
| 09:47.55 | RoyK | 30MHz what? |
| 09:48.10 | power1 | I am going to go out now and buy a board and cpu for an asterisk box.... gonna use 10 sip phones and one sip trunk. I want it to be super stable on compatible with a tdm400p card....any suggestions on what board / cpu I should go for? |
| 09:48.17 | trixter | a frequency where HF ends and VHF starts |
| 09:48.18 | trixter | :P |
| 09:48.26 | X-Gen | RoyK: these people are telling me lies |
| 09:48.28 | X-Gen | :P |
| 09:48.38 | evilbuny | power1: there are other options |
| 09:48.53 | RoyK | sangoma hardware is cool but their drivers could be better |
| 09:49.16 | trixter | well their drivers in asterisk are limited by the zapata interface |
| 09:49.22 | trixter | many of the features on the card cant be supported |
| 09:49.23 | RoyK | sangoma generally blames echocan for cpu usage, but their wanpipe and related drivers uses far too much cpu on high load |
| 09:49.36 | X-Gen | RoyK: is no way to workout the max no. of channels u can run with show translation info ? |
| 09:49.39 | trixter | the newest drivers is what freeswitch is doing and you generate like 50 IRqs/sec instead of 1000 under zap |
| 09:49.42 | trixter | WAY better |
| 09:49.45 | RoyK | i ran some profiling on that lately.... |
| 09:50.02 | trixter | context switches are the single most expensive operation on a x86 cpu. each irq cauess at least 2 context switches |
| 09:50.29 | RoyK | but then |
| 09:50.33 | X-Gen | page faults are worse |
| 09:50.36 | RoyK | when is freeswitch going to be stable? |
| 09:50.42 | trixter | it started last november |
| 09:50.52 | RoyK | iirc page faults causes context switching. no? |
| 09:51.00 | trixter | the fact that it does sip, h.323, iax2, sangoma pri, and a few other things all from scratch isnt that bad |
| 09:51.11 | trixter | going form user to kernel is a context switch under protected mode |
| 09:51.12 | RoyK | does it _work_ yet? |
| 09:51.18 | X-Gen | RoyK: any comment on my translation question ? |
| 09:51.24 | trixter | you dont really have em in real mode but you dont have the protection that is desired in real mode so ... |
| 09:51.27 | trixter | yes it works |
| 09:51.36 | RoyK | X-Gen: translation? |
| 09:51.49 | RoyK | eh. |
| 09:51.59 | RoyK | no |
| 09:52.02 | trixter | its just not finished as much.. and you are also not really thinking about what it is (or so it seems based on your comments).. freeswitch is a switching core, not a all inclusive pbx that tries to do everything |
| 09:52.18 | X-Gen | no ? no it cant be done or no - no comment |
| 09:52.20 | trixter | that core is used by applications (ivr, voicemail, etc) and is not the applications directly |
| 09:52.30 | RoyK | i maxed out a dual xeon64 3.2 with 8 PRIs on two A104s |
| 09:52.39 | RoyK | each on a sole pci bus |
| 09:52.41 | trixter | freeswitch can also deal with wideband codecs something asterisk currently doesnt do well |
| 09:52.44 | trixter | :) |
| 09:52.56 | RoyK | wideband codecs is cool |
| 09:53.01 | power1 | evilbuny, care to elaborate on other options? |
| 09:53.08 | trixter | they can make it sound better, especially people with hearing loss |
| 09:53.27 | RoyK | in comparison, we run te410p on a single xeon32 3.0 which is doing well below 50% on full load |
| 09:53.35 | X-Gen | RoyK: how did u know it was maxed out ? extra lag in the speech ? |
| 09:54.03 | RoyK | vmstat |
| 09:54.03 | trixter | 93 gsm channels less than 50% cpu usage |
| 09:54.04 | RoyK | sar |
| 09:54.06 | RoyK | top |
| 09:54.08 | trixter | on this box |
| 09:54.39 | trixter | sar wow havent heard anyone talk about that in like a decade |
| 09:54.59 | mkl1525 | Hi, when I change the ring strategy of a call queue is a reload of the configuration eough to change the strategy? (cause it didn't work doing it that way) |
| 09:55.34 | trixter | if it didnt work my guess is that no its not enough |
| 09:56.56 | RoyK | X-Gen: also, asterisk is quite bad if just bridging calls. i maxed out a single xeon 3.2 with 500 concurrent calls with no transcoding at all |
| 09:57.08 | MGSsancho | >.> |
| 09:57.17 | remiss | hmm.. when i use the call-files and it rings back at the specified context, it doesn't give what number has been called to the telephone.. any ideas on how to set it? |
| 09:57.32 | *** join/#asterisk oej (n=oej@swissco012231-3-3.clients.easynet.fr) |
| 09:57.52 | RoyK | oej: morgon |
| 09:58.14 | remiss | god morgen |
| 09:59.12 | *** join/#asterisk Guest^DJ (i=me@211.24.146.12) |
| 09:59.20 | RoyK | remiss: #asterisk-no :) |
| 09:59.55 | X-Gen | RoyK: how did u simulate the calls ? |
| 10:00.18 | oej | Morgon, RoyK |
| 10:00.46 | RoyK | just dialed into box a which dialed box b on extension 10 which dialed box a on extension 11 etc until ext 510 where it was sent to app_echo |
| 10:00.59 | X-Gen | i take it those where just sip calls that u setup right ? |
| 10:01.41 | X-Gen | * metrics still seams like a black undocumented art :P |
| 10:01.43 | RoyK | sip, yes, with rtp |
| 10:02.27 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:02.53 | *** join/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 10:03.21 | trixter | the metrics are only meaningful if you have an identical system because its really hard to quantify all the cpu suckers |
| 10:03.44 | trixter | your ethernet card for example can be a good one that doesnt cause as much cpu load or a really horrible one that requires 2x the cpu to process |
| 10:04.16 | MGSsancho | like shity PCI based ones |
| 10:04.18 | trixter | something that people may not even consider, and there may be other non-asterisk processes running that arent considered, there may be just bad hardware that spew garbage to the bus and you never see that bad data but it causes slowdowns |
| 10:04.34 | trixter | my moms laptop had a bad CDROM, replacing it caused a *noticable* speed boost becuase of that very issue |
| 10:04.36 | dezent | trixter: we have done some brief testing now and i think i get a better picture over the problem.. thanks again |
| 10:04.49 | RoyK | trixter: the testing I did was with well tuned, good e1000 cards, and what was spending time was kernel calls, not the e1000 driver |
| 10:04.51 | trixter | dezent: that is good |
| 10:04.58 | dezent | yes =) |
| 10:05.01 | RoyK | trixter: please do that testing yourself with oprofile :) |
| 10:05.09 | trixter | yeah I am just saying though that your results dont translate well to say my system |
| 10:05.11 | trixter | becuase of those factors |
| 10:05.17 | sergeus | ~seen jwu |
| 10:05.23 | jbot | jwu <n=jwu@221.221.11.37> was last seen on IRC in channel #asterisk, 1d 22h 36m 11s ago, saying: 'idunno, can you help?'. |
| 10:05.24 | flot | Who used chan_bluetooth? |
| 10:05.24 | X-Gen | * run on solaris ? they seam to have THE profiler atm |
| 10:05.36 | *** join/#asterisk Modcuts (n=bob@proporta.gotadsl.co.uk) |
| 10:05.48 | trixter | so while it may have worked for you at 500 calls I might be able to only get 300 or perhaps 700 either way and the difference is motherboard, ram, ethernet card, etc. even if the cpus are identical |
| 10:06.09 | RoyK | trixter: have you tried setting up two boxes and ping/pong-dialed between them with canreinvite=no? |
| 10:06.15 | trixter | so its a potential problem to say that if you have X cpu you can do Y channels of type Z becuase that may be a general rule but not true all the time |
| 10:06.18 | MGSsancho | could be like database machines, its all about the quality of the NICs |
| 10:06.30 | X-Gen | but surely there should be some test u can run on your system and it would say..well give the crap hardware u have...u can run x channels |
| 10:06.38 | RoyK | trixter: i have tried both on an HP box and on a couple of different IBM boxes |
| 10:06.58 | RoyK | it seems they max out at about 4-600 calls per cpu (2.4-3.0GHz) |
| 10:07.03 | RoyK | so it's pretty stable |
| 10:07.05 | trixter | RoyK: regardless of how many you have tested that doesnt mean that someone else can use those numbers as absolutes,that is all I am saying |
| 10:07.13 | RoyK | sure |
| 10:07.20 | RoyK | i'm not talking about absolutes |
| 10:07.35 | trixter | I am not questioning your ability to do this, only that the numbers dont mean a whole lot outside of your particular set up for a variety of factors, and that was in response to someone saying that metrics were a black undocumented art |
| 10:07.40 | RoyK | just saying there's stuff in asterisk that should be tuned better to perform this task |
| 10:07.46 | trixter | giving a reason why it is and to a point why it should be generally |
| 10:07.58 | trixter | oh yeah there is a lot that should be |
| 10:08.04 | trixter | but first it has to work :P |
| 10:08.32 | MGSsancho | roy means that hp and ibm are emaning less unless you mention a model number so we can see waht it has |
| 10:08.36 | X-Gen | but i want my little test program that says..look u can have 100 g711 calls and 10speex calls and 1 alaw call :P |
| 10:08.54 | MGSsancho | like saying my pentium4 and ati card get 60fps in quake 4 |
| 10:09.03 | RoyK | X-Gen: g.711 == alaw |
| 10:09.13 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 10:09.15 | MGSsancho | well there were 20 diffrent P4 and like 40 diffrent ati cards |
| 10:09.19 | RoyK | X-Gen: also, you can't get that. just do some testing.... |
| 10:09.24 | MGSsancho | was it agp or pcie? |
| 10:09.59 | MGSsancho | has to do with benchmark |
| 10:10.03 | MGSsancho | roy want spefics |
| 10:10.07 | X-Gen | RoyK: there are ways to bench your display card, how do u bench * ? |
| 10:10.07 | trixter | I think he was speaking generically saying that without very specific hardware details its meaning less to quote metrics |
| 10:10.13 | MGSsancho | like was it 100mb lan? or 1gb? |
| 10:10.20 | MGSsancho | might might a diffrence <_< |
| 10:10.38 | astar` | i have a dialplan that does _02. to call via voip but I want to force per example by pressing 9 and my phone number to call by zap |
| 10:10.40 | RoyK | X-Gen: script it |
| 10:10.42 | X-Gen | i thought show translations was a bench, but it appears to be meaning less |
| 10:11.02 | RoyK | it's highly theoretical |
| 10:11.17 | trixter | well it also means only that when the codec is loaded, becuase that is when its calculated, what if you do it at boot when a bunch of stuff is starting, perhaps in the background |
| 10:11.22 | RoyK | it only shows an index of the transcoding part |
| 10:11.25 | trixter | and the system is generally more busy than it normally is |
| 10:11.28 | MGSsancho | most of asterisk is I/O |
| 10:11.42 | RoyK | but as i said, we maxed out a box with ~400 sip/sip calls with no transcoding at all |
| 10:11.48 | RoyK | with a good server indeed |
| 10:12.11 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 10:12.17 | [av]bani | RoyK: use reinvite :/ |
| 10:12.49 | MGSsancho | im guessing the cpu was basically handeling the psi bus |
| 10:12.56 | MGSsancho | pci is bad in that aspect |
| 10:13.03 | MGSsancho | cpu as to do a lot of the work |
| 10:13.15 | [av]bani | no, its all mem bandwidth issues |
| 10:13.30 | MGSsancho | cpu to ram? |
| 10:13.36 | RoyK | [av]bani: no it is not mem bandwidth |
| 10:13.38 | MGSsancho | fsb? |
| 10:13.39 | X-Gen | then amd opteron >> all |
| 10:13.39 | [av]bani | cpu to pci |
| 10:13.52 | MGSsancho | yeah i know of the pit fall of that |
| 10:13.57 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:14.00 | MGSsancho | that aspect pcie ovens |
| 10:14.05 | MGSsancho | *owns |
| 10:14.11 | X-Gen | network card -> cpu -> mem and back |
| 10:14.12 | RoyK | [av]bani: this is a high-end computer and it can handle quite a lot more, but it seems * doesn't use much like zerocopy stuff |
| 10:14.26 | RoyK | X-Gen: that is done by dma in these days |
| 10:14.31 | [av]bani | RoyK: you cant use zerocopy if you're mangling data, like asterisk does |
| 10:14.48 | [av]bani | RoyK: and your app has to be specifically written to do zerocopy |
| 10:14.49 | RoyK | no, but why does it mangle rtp data if bridging? |
| 10:14.56 | RoyK | i'm aware of that :) |
| 10:15.00 | RoyK | but still it should be done |
| 10:15.01 | [av]bani | RoyK: headers |
| 10:15.19 | [av]bani | udp is problematic in that respect, some hardware has tcp assist |
| 10:15.30 | [av]bani | but udp... not simple |
| 10:15.32 | MGSsancho | if your box does say 500 calls, see how you can slim it down. can you use sata hdds instead of scsi and get same results? can you put in 2gb ram instead of 4 gb and get same results? |
| 10:15.41 | trixter | RoyK: how many people have a realtek 8139 that doesnt do dma well? :P |
| 10:16.06 | trixter | although I really would prefer device polling to be more standard than it is with hardware access, but that is just me |
| 10:16.14 | MGSsancho | now once you have a box running with max calls at say 500 calls per box, then you know the ratio. so if you need 10k calls, then you get more boxes |
| 10:16.27 | [av]bani | RoyK: also, afaik zerocopy doenst work properly on connectionless sockets anyway.. and zerocopy in linux doesnt work the way you expect usually... there are some annoying limitations in the way it works |
| 10:16.28 | MGSsancho | lol |
| 10:16.46 | trixter | RoyK: nah but I am saying a lot of people do use them and they arent that good, and cause the cpu to work more, yet another reason why random numbers of call limits and other stuff is not quite portable across systems |
| 10:17.01 | [av]bani | RoyK: last time i used zerocopy, it was only good for tcp sockets, and then only if you didnt even touch the data -- just shuffling from fd to fd |
| 10:17.28 | RoyK | trixter: sure, but these tests were done with e1000 tuned up with huge buffers so they generated < 10interrupts/s |
| 10:17.54 | [av]bani | RoyK: buffers dont matter, its all about interrupt mitigation |
| 10:18.01 | RoyK | mitiwhat? |
| 10:18.21 | MGSsancho | lol |
| 10:18.23 | RoyK | :) |
| 10:18.34 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 10:18.37 | puzzled | morning |
| 10:18.42 | RoyK | morning, puzzled |
| 10:18.46 | MGSsancho | morning |
| 10:18.47 | X-Gen | g'day |
| 10:18.50 | X-Gen | freak |
| 10:19.18 | MGSsancho | i just run asterisk on my linksys router |
| 10:19.28 | MGSsancho | <-- student |
| 10:19.41 | MGSsancho | maybe 2 calls at most |
| 10:19.54 | MGSsancho | lol |
| 10:19.56 | puzzled | actually that's 20Mb |
| 10:20.11 | MGSsancho | so i dont need a google farm to have a zillion connections lol |
| 10:20.21 | MGSsancho | you dont live in the US do you? |
| 10:20.29 | puzzled | nope |
| 10:20.36 | [av]bani | RoyK: one might have better luck writing a kernel module to handle zerocopy rtp |
| 10:20.48 | puzzled | X-Gen: feed your bits more coffee |
| 10:21.06 | MGSsancho | over clock? |
| 10:21.27 | trixter | puzzled: ahh today you are only 2 hours late :P |
| 10:21.42 | trixter | need to set your alarm earlier :D |
| 10:21.42 | MGSsancho | lol |
| 10:21.54 | MGSsancho | <--2:21 am |
| 10:22.26 | puzzled | trixter: hehe. but yesterday I was still online at 03:00ish. and now I have a major headache |
| 10:22.34 | trixter | heh |
| 10:23.25 | MGSsancho | heh |
| 10:23.36 | MGSsancho | then i assume your no 20 years old :) |
| 10:23.39 | MGSsancho | *not |
| 10:23.48 | puzzled | I left that station years ago |
| 10:24.08 | MGSsancho | lol |
| 10:24.40 | MGSsancho | i just wish school was serious |
| 10:24.51 | MGSsancho | lame stuff you alredy know |
| 10:24.56 | MGSsancho | cant really skipp classes |
| 10:25.24 | MGSsancho | not till you get to like 4th year college do you learn usefull stuf |
| 10:25.32 | MGSsancho | like bash scripting |
| 10:25.48 | MGSsancho | needs to be a test |
| 10:26.18 | MGSsancho | if you can compile ur own kernal and modules in 24 hours then you pass 3 years of bs intro to comps classes |
| 10:28.26 | astar` | i have a dialplan that does _02. to call via voip but I want to force per example by pressing 9 and my phone number to call by zap |
| 10:29.08 | MGSsancho | i dunno >.< |
| 10:31.01 | *** join/#asterisk basta (n=basta@194.150.162.129) |
| 10:32.30 | RoyK | [av]bani: doing that in kernelspace is ugly |
| 10:32.39 | Curus | Zerocopy would be a bit silly on 100 byte packets |
| 10:32.53 | basta | ever happened that you can't call only some numbers ? you get busy even if the called phone isn't really busy, seems the problem is with the end-provider |
| 10:32.56 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 10:33.10 | RoyK | [av]bani: i've started working on another approach, a standalone userspace thingie combined with iptables QUEUE |
| 10:33.39 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:33.44 | RoyK | the QUEUE target app will just handle sip signalling and set DNAT rules for RTP |
| 10:33.57 | RoyK | methinks that will work well :) |
| 10:39.10 | MGSsancho | ok ive go to goto sleep now @_@ |
| 10:39.28 | MGSsancho | good talking to ur guys(girls? you never know online) |
| 10:39.30 | MGSsancho | ga'night |
| 10:49.17 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 10:52.34 | trixter | Concerned about VoIPE911? Here's a list of all PSAPs in the US http://blogs.zdnet.com/ip-telephony/index.php?blogthis=1&p=930 |
| 10:52.40 | trixter | woah that is gonna be totally abused!! |
| 10:53.44 | RoyK | hm |
| 10:53.52 | RoyK | how do you call a function from agi? |
| 10:53.57 | RoyK | perl agi |
| 10:54.06 | RoyK | setlanguage is gone, so i need to use LANGUAGE |
| 10:54.14 | mrtwister|mobile | hi. question. canteinvite=no sholud exclude media proxy on the asterisk? or other setting? |
| 10:54.25 | RoyK | it should |
| 10:55.09 | RoyK | eh |
| 10:55.22 | RoyK | canreinvite=no will make asterisk bridge the media |
| 10:57.18 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 11:00.02 | *** join/#asterisk X-Rob_ (n=me@CPE-138-130-232-234.qld.bigpond.net.au) |
| 11:00.44 | *** join/#asterisk fulgas (n=fulgas@82.102.2.254) |
| 11:13.25 | *** join/#asterisk guzzler (n=chatzill@213.63.190.66) |
| 11:13.47 | *** join/#asterisk apardo (n=apardo@87.218.44.213) |
| 11:20.14 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 11:20.17 | wiseguy_ | hello |
| 11:20.24 | puzzled | hi |
| 11:20.27 | wiseguy_ | anybody using asterisk with vizufon? |
| 11:20.33 | puzzled | nope |
| 11:23.18 | *** join/#asterisk QuAtRo[NL] (n=dennis@dsl-083-247-051-039.solcon.nl) |
| 11:25.02 | QuAtRo[NL] | 1. Currently my agents have to login before they are able to answer calls, how can i edit the configuration that they don't have to login anymore? 2. They can not choose whether they want to accept a call or not, how can they choose? |
| 11:25.41 | trixter | lets play the myth game.. fact or myth if there is about 30ms lag (60ms RTT) will faxing work? |
| 11:25.54 | puzzled | ask coppice :) |
| 11:26.07 | trixter | I am gonna vote for myth |
| 11:26.40 | trixter | and here is why, edinburgh scotland to los angeles california is 5161 miles as the crow flies. fiber/copper rarely goes as the crow flies, but that distance it takes light about 27.5 ms to go that far |
| 11:27.18 | trixter | add in telco latency the fact that the circuit is likely to not be a totally straight line, etc 30ms is fair to say for that.. yet faxing works |
| 11:27.32 | trixter | I just recall people saying that if its > 20ms it will never work properly |
| 11:27.54 | QuAtRo[NL] | Does someone know the answer on one of my questions above? |
| 11:28.11 | trixter | dynamic agents have to log in, however agents dont have to be dynamic |
| 11:28.24 | evilbuny | that psap list doesn't have call numbers |
| 11:28.27 | trixter | http://www.orderlyq.com/asteriskqueues.html |
| 11:28.42 | trixter | evilbuny: yeah I misread it has the 'psap number' not the phone number |
| 11:28.53 | trixter | but it still makes it a lot easier to know who covers what region since they dont overlap |
| 11:29.05 | evilbuny | i wish it did have numbers |
| 11:29.07 | X-Rob_ | Heya evilbuny |
| 11:29.07 | evilbuny | phone |
| 11:29.10 | evilbuny | lo x-rob |
| 11:29.13 | X-Rob_ | we're in t'mba at the mo |
| 11:29.13 | trixter | you know the city and state and entity name so google or something should help you find them |
| 11:29.14 | evilbuny | how's melb |
| 11:29.48 | evilbuny | o |
| 11:29.48 | evilbuny | trixter: mmmm |
| 11:29.48 | trixter | if it had the emergency number that would be spread as a '911 backdoor list' and 911 services abused much more than they are |
| 11:29.48 | evilbuny | lol |
| 11:33.03 | *** join/#asterisk belgarioz (n=belgario@212.17.211.2) |
| 11:33.06 | belgarioz | hi |
| 11:33.20 | belgarioz | can anyone help me with a BRI ISDN? |
| 11:33.27 | belgarioz | i really don't know how to make it work :( |
| 11:33.42 | belgarioz | mISDN documentation is quite confusing |
| 11:37.02 | evilbuny | trixter: the reason I'm after numbers is so I can link it to 911 on asterisk boxes |
| 11:37.54 | mrtwister|mobile | <PROTECTED> |
| 11:37.59 | mrtwister|mobile | what is it/ |
| 11:38.08 | trixter | well you generally have to have a contract to do that anyway |
| 11:38.22 | evilbuny | this is for personal stuff |
| 11:38.36 | evilbuny | ie friends |
| 11:38.59 | trixter | if you send a psap a lot of calls to their real did then they wont get address info etc.. so they might have issues with that |
| 11:39.12 | trixter | there is a voip standard that was approved a couple months ago so that may start popping up places |
| 11:44.02 | *** join/#asterisk vinsik (n=vinsik@gw-ff.verkkokauppa.com) |
| 11:45.48 | vinsik | Hei guys!!! I have a little problem understanding fax operativity.. the * is saying unknown codec 100 received. I have 1.2.4 version with steveu's T38 support for it, but still it doesent work. anybody knows what to do? |
| 11:46.40 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 11:50.52 | *** join/#asterisk Majestik (n=Majestik@S0106000024c058cc.ed.shawcable.net) |
| 12:03.25 | *** join/#asterisk d-tech (n=dtc@h-72-245-233-107.sfldmidn.covad.net) |
| 12:07.14 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 12:11.57 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:19.32 | *** part/#asterisk X-Gen (n=x-gen@dsl-146-122-162.telkomadsl.co.za) |
| 12:21.47 | astar` | i have a dialplan that does _02. to call via voip but I want to force per example by pressing 9 and my phone number to call by zap |
| 12:29.20 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:30.10 | *** join/#asterisk my007ms (n=my@196.202.70.99) |
| 12:30.33 | my007ms | Hello all |
| 12:31.51 | my007ms | hello |
| 12:32.48 | my007ms | i have new card DTM2424E |
| 12:33.07 | my007ms | i must see one i do lpci |
| 12:33.13 | my007ms | ? |
| 12:34.21 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 12:34.40 | *** join/#asterisk mut (n=animenod@65.111.201.79) |
| 12:35.48 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 12:41.05 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 12:41.59 | fourcheeze | my007ms: I'm having difficulty understanding your question |
| 12:50.28 | astar` | ok i found if someone has same prob ask me :) |
| 12:52.40 | heroine | has somebodyt there used spandsp and app_rxfax/app_txfax successfully ? |
| 12:55.39 | my007ms | pleas all i need help |
| 12:56.15 | my007ms | what i must see when i do lspci when i install TDM2424E card |
| 12:56.46 | my007ms | in all server i install b4 i was see Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 12:56.57 | my007ms | no i see unknow divece |
| 12:58.19 | my007ms | fourcheeze do u have any idea |
| 12:58.39 | RoyK | http://photos.photosig.com/photos/88/38/1703888-b8bae0792e073dcc.jpg |
| 12:59.27 | Skid | lol |
| 12:59.52 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 13:00.38 | *** join/#asterisk newl (n=newlook@203-59-100-129.dyn.iinet.net.au) |
| 13:06.26 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:06.33 | *** join/#asterisk oej (n=oej@swissco012231-4-3.clients.easynet.fr) |
| 13:06.37 | fourcheeze | my007ms: that's the one |
| 13:07.04 | fourcheeze | my007ms: the Tiger Jet Network thing |
| 13:07.09 | fourcheeze | unless you have another ISDN card |
| 13:09.32 | fourcheeze | in voicemail.conf should the format configuration affect the outgoing message format? |
| 13:09.41 | fourcheeze | if not, what should |
| 13:10.05 | mut | woot |
| 13:10.12 | mut | we finally got some racks put in |
| 13:10.21 | mut | get to move everything over tonite |
| 13:10.35 | mut | take down internet and phones all over! |
| 13:14.27 | jbalcomb | RoyK: that's nice. they seem so emotionless though. kinda kills the mood. |
| 13:15.31 | my007ms | so what is wrong with my new card i don't see this |
| 13:15.55 | my007ms | i don't have irq confilact |
| 13:16.50 | RoyK | ~lart jbalcomb |
| 13:18.51 | my007ms | fourcheeze |
| 13:18.55 | fourcheeze | my007ms: have you already accounted for the Tiger card? |
| 13:19.01 | my007ms | yes |
| 13:19.22 | my007ms | but i see unknow thing in PCI |
| 13:19.36 | my007ms | all othre card was show Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 13:19.55 | jbalcomb | VAXen? |
| 13:19.55 | my007ms | is TDM2424E need something othere don't |
| 13:19.56 | my007ms | ?? |
| 13:20.25 | my007ms | i mean by othere othere TDM card |
| 13:20.37 | my007ms | and sorry for my english |
| 13:20.57 | jbalcomb | ~english |
| 13:21.01 | jbot | i guess english is more geographically spread than any other language |
| 13:21.44 | jbalcomb | nice work jbot |
| 13:22.06 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
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| 13:39.53 | vinsik | Hei guys!!! I have a little problem understanding fax operativity.. the * is saying unknown codec 100 received. I have 1.2.4 version with steveu's T38 support for it, but still it doesent work. anybody knows what to do? |
| 13:42.25 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:44.10 | bronze | ~jbot |
| 13:44.15 | jbot | hmm... jbot is only marginally useful at best, or a silly little bugger |
| 13:44.23 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 13:45.01 | asteriskmonkey | can anyone tell me what the difference is bettween 1.2 and 1.2-netsec on the branches svn |
| 13:46.03 | heroine | vinsik: you have the T38 support in your asterisk ? |
| 13:47.44 | asteriskmonkey | so no one know if there is why there is a 1.2-netsec veriosn? |
| 13:48.32 | heroine | asteriskmonkey: from what i have read from the Changelog, it disables some features which seems buggy, corrects some memory leak |
| 13:48.34 | puzzled | asteriskmonkey: it is a version of asterisk that supports firewall technology that automagically opens/closes ports as needed |
| 13:49.04 | puzzled | heroine: I just compiled it in |
| 13:49.36 | heroine | puzzled: asterisk-netsec or T38 support ? :) |
| 13:49.43 | asteriskmonkey | is there a list of features it disables? |
| 13:50.18 | puzzled | heroine: T38 passthrough |
| 13:50.23 | *** part/#asterisk mphill (n=datix@CPE-70-92-245-5.wi.res.rr.com) |
| 13:50.48 | puzzled | asteriskmonkey: unless you have a firewall that supports this netsec stuff you don't need it |
| 13:50.57 | asteriskmonkey | ok cool |
| 13:51.11 | heroine | you used steveu t38-bits patchs ? |
| 13:51.16 | puzzled | yup |
| 13:51.20 | asteriskmonkey | why not make it an add on module rather than an entire build though :P |
| 13:51.28 | puzzled | ask the developers |
| 13:51.53 | asteriskmonkey | ill ask mark in a couple week at von |
| 13:51.56 | heroine | damn .. i can't manage to patch chan_sip.c on a* 1.2.4 :/ i get some compiling errors |
| 13:53.45 | heroine | did you get the t38 patch on bugs.digium.com or on steveu homepage ? |
| 13:55.08 | puzzled | bugs |
| 13:55.22 | heroine | hmm k thanx :) |
| 13:55.42 | Skid | hmm, is there a 'GD" channel on freenode? GD the graphics package |
| 13:56.58 | puzzled | Skid: try /join #gd |
| 13:57.07 | *** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it) |
| 13:57.08 | Skid | yeah, I have |
| 13:57.16 | Skid | it's +f #gamedev :/ |
| 13:57.22 | Skid | just thought someone may know :) |
| 13:59.15 | m-i-a- | Anyone in here at the TAUG event the other day? |
| 13:59.17 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 14:06.51 | wiseguy_ | anybody using asterisk with vizufon? |
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| 14:14.10 | gaupe | :q |
| 14:14.13 | gaupe | oops |
| 14:14.21 | fafnir | OOOPS |
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| 14:19.55 | iCEBrkr | yo yo yo |
| 14:20.12 | Aughey | anyone use the snom 320 phones? |
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| 14:22.22 | Aughey | it's quiet this morning |
| 14:22.31 | b0ef | ehlo |
| 14:22.46 | heroine | snom 360 here |
| 14:23.10 | Aughey | heroine: what's the trick to get it to auto answer for intercom? |
| 14:23.23 | heroine | to auto answer ? |
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| 14:23.33 | b0ef | is foo@bar.baz a valid uri with the iax protocol? |
| 14:23.46 | b0ef | wouldn't that be an account? |
| 14:24.27 | heroine | Aughey: don't know |
| 14:24.33 | Aughey | on grandstream phones, I add SIPAddHeader(Call-Info: answer-after=0) and then run Page(SIP/extension) |
| 14:24.52 | Aughey | paging snom phones that way will just ring the phone, they don't answer |
| 14:25.07 | Aughey | heroine: do you use dialplans on the phone so they don't have to press the Ok key? |
| 14:25.22 | heroine | hmm no we don't use dialplan on the phone |
| 14:25.42 | Aughey | well, I'm 0 for 2 this morning :) |
| 14:25.43 | iCEBrkr | Aughey: Page() is just a function to get an 'all page' type system working, it's not going to make the phones auto-answer |
| 14:26.50 | Aughey | iCEBrkr: for Grandstream, I have to add the SIPAddHeader to get it to answer right away. For "all page" I believe the solutio nis to use that allpage.agi script |
| 14:27.44 | iCEBrkr | I just know Page() is a clever hack to get 'page' a group of phones |
| 14:28.23 | iCEBrkr | Now if you can issue SIP headers to instruct the phone to auto-answer, that's a differnet story :) |
| 14:29.01 | Aughey | I mostly just want someone to confirm that the snom 320 can do intercom and a dialplan without pressing OK. I have one of the phones, but it's not accessable to me right now and I need to purchase 4 more. |
| 14:29.12 | Aughey | I want to make sure those two features are truely available before ordering them |
| 14:29.29 | Aughey | the wiki seems to say they do, but when I tried it didn't work right away so I'm hesitant |
| 14:30.44 | Aughey | my customer likes the hands free audio quality of the snom phone, but the price and look of the Grandstream. So we're going to use both phones in different areas |
| 14:34.49 | fugitivo | anyone is using asterisk as a recording addon for a pbx? |
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| 14:39.32 | *** join/#asterisk ChrisUK (n=chris@82.108.126.170) |
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| 14:45.10 | fourcheeze | Aughey: I think the intercom functions on snoms is a bit of a hack in * |
| 14:45.18 | fourcheeze | but I did see it somewhere |
| 14:46.18 | real-dev | how can I set the outgoing callerid conditionally? |
| 14:46.32 | Aughey | fourcheeze: I see it in the asterisk-users list. I'll investigate it further. Thanks |
| 14:47.51 | real-dev | I have a hw pbx with two phones attached to *, one phone is '1', the second '9' |
| 14:48.14 | real-dev | Accepting overlap voice call from '9' to '012345' on channel 0/1, span 1 |
| 14:48.43 | real-dev | this I get and I want to set a different cid for '9' than for '1' |
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| 15:03.18 | mkl1525 | Hi, I've configured, compiled and installed the cdr_addon_mysql.so module, restartet *. show modules shows cdr_addon_mysql.so (with a use count of 0) but when a call comes in it isn't logged to mysql - so is there a setting I've missed? |
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| 15:06.41 | heroine | mkl1525: which version of asterisk ? |
| 15:09.26 | mkl1525 | 1.0.10 |
| 15:13.45 | heroine | hmm .. we are on 1.2.x here :/ |
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| 15:19.30 | *** join/#asterisk Seyr (n=seyr@cpe-67-10-139-141.houston.res.rr.com) |
| 15:19.54 | Seyr | When you call into to someones voicemail, while the greeting is being played, you should be able to press 0 to go to an o extension right? |
| 15:20.03 | Seyr | or is that only while your recording a message? |
| 15:20.39 | heroine | Seyr: i didn't read anything about that .. |
| 15:21.31 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 15:21.37 | [TK]D-Fender | Seyr : Thats what the "operator=yes" flag is in voicemail.conf |
| 15:21.43 | m-i-a- | I am trying to duplicate the way my company I work for uses their nortel system into a * system. Basicly they have 1 room with a bunch of phone operators and as calls come in they pickup each line individually and if really busy they put current call on hold and pickup another line and put that call on hold. Here's the kicker, everyone else in the other rooms will pickup a call if the lines go crazy but that only |
| 15:21.49 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
| 15:21.56 | m-i-a- | How are you [TK]D-Fender? |
| 15:22.17 | [TK]D-Fender | m-i-a- : "Still breathing" |
| 15:22.43 | mkl1525 | heroine, is there something like a diff page with the differences of * 1.0 and 1.2? |
| 15:22.48 | m-i-a- | [TK]D-Fender I got the go ahead to put in * for July this year |
| 15:22.51 | [TK]D-Fender | So your operators are bouncing itno each other's calls? |
| 15:23.07 | [TK]D-Fender | m-i-a- : Congrats... not you need to start planning the outlay for it. |
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| 15:23.48 | heroine | mkl1525: what i know between * 1.0 and 1.2 is that mysql support was reworked in what is called realtime to support several way to fetch data (pgsql/mysql/ldap) .. but that's the only difference i know about |
| 15:24.13 | heroine | it seems to also have fixed a bug in the way sdp packets are built when * is bound on several interfaces |
| 15:25.01 | mkl1525 | heroine, thanks for your help |
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| 15:25.08 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:26.16 | m-i-a- | [TK]D-Fender, I think what I need is a way so if there are more than 2 or 3 people on hold in a queue that it then rings everyone phone in the whole office to try and accept the calls |
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| 15:27.08 | Seyr | [TK]D-Fender: it doesnt work. I set operator=yes, but pressing 0 only does something when it starts to record |
| 15:27.29 | m-i-a- | They want to keep the personal touch, so the clients dont feel like they are talking to a machine |
| 15:27.31 | heroine | i'm having a strange behavior .. havnig a DeadAGI() calling a non existent script and asterisk log repors that the AGI completed returning 0 where i expect a message more like "error, agi not found" or something similar .. |
| 15:28.25 | heroine | (i removed the script has it seems it wasn't even called) |
| 15:28.26 | [TK]D-Fender | Seyr : Pastebin all the related parts so I can see how you did it. |
| 15:28.53 | Seyr | [TK]D-Fender: What other parts are there besides setting operator=yes? |
| 15:29.12 | [TK]D-Fender | m-i-a- : You can have it so that before you send them to an operator queue it looks at the back-log and then routes to "everyone" in case its overflowing. |
| 15:29.39 | [TK]D-Fender | Seyr : pastebin your voicemail.conf and the extensions.conf sections where you are implementing it. and I mean the entire context. |
| 15:30.45 | Seyr | [TK]D-Fender: i'll have to re-add them. took it all out :-( lemme get it back in |
| 15:31.12 | m-i-a- | [TK]D-Fender they everyone queue could be setup so there is no agent login needed correct? |
| 15:31.43 | *** join/#asterisk EnErGY_CSDX (n=energy@193.242.114.14) |
| 15:33.20 | [TK]D-Fender | m-i-a- : You could do that, or based on how many people there are you could just a DialAll line. |
| 15:34.02 | [TK]D-Fender | m-i-a- : How big an office are you in? |
| 15:34.37 | m-i-a- | but if there was two overflow calls happening at the same time and two dialall commands happening would the second dial all not get a busy because everyones phones are ringing? |
| 15:34.49 | m-i-a- | 15 ppl 23 channel |
| 15:34.50 | m-i-a- | s |
| 15:35.39 | *** join/#asterisk duckz (n=duckz@193.192.47.26) |
| 15:35.56 | m-i-a- | I got to do a chit load more reading on dialplan logic |
| 15:37.08 | m-i-a- | assuming that all the sip hard phones only had a single line appearance |
| 15:38.08 | [TK]D-Fender | m-i-a- : Depends. Multi-line phones can have as many rining channels as they've got free... |
| 15:38.20 | [TK]D-Fender | m-i-a- : not in MY world... |
| 15:38.58 | [TK]D-Fender | Polycom's can have 24 calls per line key * 6 keys on an IP 601. Thats a LOT BTW..... |
| 15:39.13 | m-i-a- | =) |
| 15:39.19 | m-i-a- | very true |
| 15:39.27 | [TK]D-Fender | At a minimum people here have 3 line keys that can handle 1 call each. |
| 15:39.49 | [TK]D-Fender | If you need more, you're corporate planning is flawed :) |
| 15:40.02 | m-i-a- | lol |
| 15:40.09 | m-i-a- | its a very whacked business |
| 15:40.51 | [TK]D-Fender | *b00m* ! |
| 15:42.09 | m-i-a- | The sangoma 4 port T1 card is the only one with echo can? |
| 15:42.22 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 15:42.31 | m-i-a- | I did not see a single port with echo can |
| 15:42.47 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 15:43.38 | m-i-a- | Did you see rhino is giving away there t1 card free with every channel bank purchase until the end of march |
| 15:43.48 | *** join/#asterisk EnErGY_CSDX (n=energy@193.242.114.14) |
| 15:44.13 | *** part/#asterisk EnErGY_CSDX (n=energy@193.242.114.14) |
| 15:44.24 | *** join/#asterisk EnErGY_CSDX (n=energy@193.242.114.14) |
| 15:45.26 | *** join/#asterisk flynux (i=8x2fw5m@gs10-2.openweb.be) [NETSPLIT VICTIM] |
| 15:46.13 | heroine | yo flynux |
| 15:46.34 | [TK]D-Fender | m-i-a- : I wouldn't touch Rhino's card until a great review comes out. As it is it has no E1 capability. As for Echo Cancel cards, Digium's TE411P and TE406P has on-board EC as well. |
| 15:48.05 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 15:48.31 | m-i-a- | But if it's free why not try it out? :) |
| 15:48.50 | [TK]D-Fender | Yes, FREE might be overpriced for it :) |
| 15:48.56 | m-i-a- | lol |
| 15:49.13 | [TK]D-Fender | But frankly I have a disdain for CB's.... |
| 15:49.25 | m-i-a- | I have a feeling Rhino is going to release a pci TDM card this year |
| 15:50.41 | [TK]D-Fender | m-i-a- : I think the market is going to get very volatile now that there are multiple options and the integration, loyalty, and "spirit" factors are likely to suffer at the same time as mass interest picks up speed. |
| 15:51.28 | m-i-a- | I think you are right |
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| 15:53.40 | *** join/#asterisk essaredee (n=srd@cpc1-lich2-0-0-cust257.brhm.cable.ntl.com) |
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| 15:54.47 | [TK]D-Fender | I usually is when it comes to bad news. I'm like some sort of "harbinger" I swear.... its *&^#$ depressing... |
| 15:55.15 | essaredee | I've got a few iax incoming accounts, which in extensions.conf they all point to an extension via goto, the extension picks up plays a file then rings the phone |
| 15:55.45 | essaredee | I've set indications=uk, and all incoming iax accounts but one are doing uk ring style, but one isn't |
| 15:55.58 | essaredee | the one that isn't sounds like a us telephone ring |
| 15:56.20 | essaredee | might there be a reason for this? I can't seem to figure out why |
| 15:57.58 | *** part/#asterisk EnErGY_CSDX (n=energy@193.242.114.14) |
| 15:59.16 | m-i-a- | [tk] what I am confused about is if a PRI connection is digital why would there be a need for echo can on the t1 card? isent the echo generated on the digital to analog conversion? so the telce would need the echo can not me? |
| 15:59.26 | *** join/#asterisk EnErGY_CSDX (n=energy@193.242.114.14) |
| 16:00.10 | MikeJ[Laptop] | m-i-a-, you can still have issues when talking to analog circuits at the other end.. |
| 16:00.35 | m-i-a- | hmmmm... |
| 16:00.52 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 16:04.06 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 16:04.10 | [TK]D-Fender | m-i-a- : Where do you think calls TO your PRI are coming from? the ANALOG world. |
| 16:04.25 | Hmmhesays | hmm voipjet and sixtel are both failing calls to russia |
| 16:04.44 | m-i-a- | but i thought the telco would handle the echo can at that side |
| 16:04.47 | [TK]D-Fender | m-i-a- : And believe me, without good EC you will find yourself hairless in short order... |
| 16:04.50 | m-i-a- | but i guess not |
| 16:05.07 | [TK]D-Fender | m-i-a- : Only on things they terminate. they leave digital alone. |
| 16:06.02 | Hmmhesays | i could use a good provider for +74 |
| 16:07.39 | MikeJ[Laptop] | where is +74? |
| 16:08.31 | EnErGY_CSDX | MikeJ[Laptop] it`s Russia |
| 16:08.51 | MikeJ[Laptop] | hmmm |
| 16:09.02 | MikeJ[Laptop] | we might?? |
| 16:09.14 | MikeJ[Laptop] | try |
| 16:10.19 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 16:10.21 | Hmmhesays | voipjet and sixtel won't terminate like 75% of the calls we are trying to send to them |
| 16:10.35 | essaredee | If I've set country=uk in indications.conf, made an extension in extensions.conf to answer and then ring a sip phone.. why does the ringing on the callers side sound like they were calling a us phone and not a uk one? |
| 16:10.35 | MikeJ[Laptop] | Hmmhesays, send them to us then! |
| 16:10.37 | MikeJ[Laptop] | :P |
| 16:10.42 | xachen | first of all, your lucky sixtel can terminate calls :P |
| 16:10.56 | xachen | hey sean |
| 16:11.01 | essaredee | hey |
| 16:11.11 | Hmmhesays | MikeJ[Laptop] give me some rates |
| 16:11.18 | brad_mssw | MikeJ[Laptop]: who is 'us' ? |
| 16:11.24 | MikeJ[Laptop] | now your asking me all the hard questions... |
| 16:11.27 | MikeJ[Laptop] | ask file? |
| 16:11.33 | MikeJ[Laptop] | asterlink. |
| 16:11.40 | MikeJ[Laptop] | I don't really do that stuff.. |
| 16:11.41 | *** join/#asterisk coppice (n=chatzill@250.194.17.210.dyn.pacific.net.hk) |
| 16:11.45 | brad_mssw | oh, yeah, that's right |
| 16:12.05 | *** join/#asterisk flynux (i=pmn8tyf@cl-8.bru-01.be.sixxs.net) |
| 16:12.08 | MikeJ[Laptop] | it's mr. underwood!!! |
| 16:12.21 | brad_mssw | steveu's the man |
| 16:12.36 | MikeJ[Laptop] | well.. |
| 16:12.39 | MikeJ[Laptop] | a man |
| 16:12.51 | MikeJ[Laptop] | he is THE dsp man! |
| 16:12.56 | brad_mssw | right |
| 16:12.57 | Hmmhesays | I'll ask him when he's around |
| 16:13.07 | synthetiq | do star codes work on zap channel phones? |
| 16:13.15 | MikeJ[Laptop] | he is now |
| 16:13.22 | MikeJ[Laptop] | star codes? |
| 16:13.33 | synthetiq | like *72 |
| 16:13.46 | MikeJ[Laptop] | ummm |
| 16:13.55 | Hmmhesays | Oh file |
| 16:13.57 | [TK]D-Fender | synthetiq : If you put it in your dial-plan, SURE. |
| 16:13.59 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 16:14.03 | MikeJ[Laptop] | that would depend on if you do somthing with them in dialplan |
| 16:14.07 | synthetiq | ok |
| 16:14.09 | MikeJ[Laptop] | or if the line handles it |
| 16:14.14 | MikeJ[Laptop] | and you send that to the line |
| 16:14.46 | [TK]D-Fender | Thats one reason I prefer gateways to channel banks. Dealing with CLASS codes and Zaptel..... |
| 16:14.51 | file | eh? |
| 16:14.55 | [TK]D-Fender | BEtter to off-load it. |
| 16:15.13 | Hmmhesays | I'm looking for a provider to do +74 terminations |
| 16:15.31 | EnErGY_CSDX | Hmmhesays sipnet.ru |
| 16:15.38 | Katty | Hmmhesays: how about a hug instead? |
| 16:15.44 | MikeJ[Laptop] | awww |
| 16:16.02 | tmccrary | Hi, I have a trunk setup between two asterisk pbx's and when I'm having an issue with voicemail / dial plan execution. When I call through from trunk A to trunk B, the voicemail entry in the dial plan never gets called (it says 487 Request Terminated), however when i call directly on the same pbx, it goes to voicemail with no problem. |
| 16:16.51 | tmccrary | anyone have an issue like that? |
| 16:16.56 | [TK]D-Fender | Katty: mew. |
| 16:16.59 | Katty | mister fender (= |
| 16:17.13 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:17.31 | [TK]D-Fender | Ms. Katty =) |
| 16:18.25 | Hmmhesays | how you feeling Katty? |
| 16:19.40 | *** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it) |
| 16:19.44 | Micetto | hi :) |
| 16:19.59 | Micetto | I have a question |
| 16:19.59 | *** join/#asterisk fu3 (n=kka@234-200-29-134.hcc.mnscu.edu) |
| 16:20.03 | fu3 | hey lads |
| 16:20.18 | Micetto | Can I connect a ISDN phone to a quadBRI ISDN card? |
| 16:20.22 | fu3 | Ijust got off the phone with qwest. They are telling me that I could be getting only a 300ms battery drop to indicate a hangup with loopstart. |
| 16:20.29 | fu3 | I understand that it is supposed to be 500ms |
| 16:20.34 | fu3 | is there any way I can change that? |
| 16:20.39 | fu3 | in the zap driver, or somewhere? |
| 16:20.44 | AndyCap | Micetto: if the driver and card supports NT mode, sure |
| 16:20.59 | tzanger | fu3: I don't think that you're going to be able to convince them to bump it up :-) |
| 16:21.11 | fu3 | I know |
| 16:21.14 | tzanger | fu3: I imagine it'd be in the wctdm driver |
| 16:21.15 | Micetto | AndyCap, the driver that I use is mISDN and I think that support NT mode |
| 16:21.16 | fu3 | so i need to change it within zap or asterisk. |
| 16:21.34 | fu3 | tzanger.. cool.. i'd have to do that in the source eh? |
| 16:21.47 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
| 16:21.58 | Micetto | AndyCap: but if i connect telephone to card the led remaning red |
| 16:22.00 | tzanger | yep I do believe so |
| 16:22.07 | fu3 | cool.. I'll check that out. |
| 16:22.08 | fu3 | thanks |
| 16:22.19 | Micetto | AndyCap: it's possible that I should set the jumper on card ? |
| 16:22.28 | AndyCap | Micetto: yes, should work, but you have to configure the port to NT mode. haven't used misdn much so I dunno |
| 16:23.00 | *** part/#asterisk Aughey (n=jha@ns1.washucsc.org) |
| 16:23.02 | gaupe | Micetto: and the you need to provide power to the bus |
| 16:23.03 | AndyCap | Micetto: btw. is the phone powered? |
| 16:23.15 | Micetto | no |
| 16:23.23 | Micetto | :( |
| 16:23.39 | AndyCap | Micetto: then you need to hack a power supply to the bus |
| 16:23.41 | Micetto | gaupe: I provide power to the bus |
| 16:23.52 | gaupe | Micetto: with what? |
| 16:24.03 | Micetto | with an external alimentator connected to the card |
| 16:24.27 | AndyCap | Micetto: what card is this btw? |
| 16:24.27 | Micetto | sorry....alimentator=power supply |
| 16:24.52 | Micetto | HFC-4S |
| 16:25.04 | Micetto | Cologne Chip |
| 16:25.23 | Micetto | 4 port ISDN |
| 16:25.35 | coppice | alimentator sounds like it means something rather different than power supply |
| 16:25.45 | AndyCap | eliminator.. |
| 16:26.22 | coppice | the chinese for power supply literally means fire cow. never worked that one out :-\ |
| 16:26.25 | Micetto | eheh :P sorry...I'm italian and I write in English I think in Italian :P |
| 16:26.37 | MikeJ[Laptop] | fire cow... |
| 16:26.46 | MikeJ[Laptop] | nice |
| 16:26.52 | Micetto | ops...sorry... |
| 16:27.02 | AndyCap | coppice: probably some nasty farm accident with an early model |
| 16:27.58 | Micetto | AndyCap: it's possible that I should set the jumper on card for NT and TE mode? |
| 16:27.59 | AndyCap | Micetto: but the card is (should be) able to power the bus on its own? |
| 16:28.13 | Micetto | AndyCap: yes |
| 16:28.17 | AndyCap | Micetto: wouldn't think so no. possibly to set termination |
| 16:28.29 | AndyCap | Micetto: what's the exact model and make of the card? |
| 16:28.34 | AndyCap | might want one myself. :P |
| 16:28.42 | Micetto | quadBRI.... |
| 16:29.06 | Micetto | I don't know the exact model...but on the chip |
| 16:29.19 | Micetto | there is "IOB4ST" HFC-4S |
| 16:29.33 | AndyCap | junghanns.net card? |
| 16:29.51 | AndyCap | optional S0 bus power feeding module available |
| 16:30.12 | AndyCap | Micetto: yup, it has jumpers you need to set |
| 16:30.19 | Micetto | ah1 |
| 16:30.20 | AndyCap | http://www.junghanns.net/downloads/quadbrijumpersnew.pdf |
| 16:30.23 | Micetto | ah! right! :D |
| 16:30.38 | AndyCap | as well as termination jumpers |
| 16:31.01 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 16:31.11 | jbalcomb | exten => _X.,1,Dial(Zap/g2/${EXTEN}) |
| 16:31.11 | jbalcomb | exten => _X.,2,Hangup |
| 16:31.11 | jbalcomb | exten => t,1,Hangup |
| 16:31.41 | jbalcomb | whats the _X about? (sorry about the pasting error) |
| 16:31.46 | Micetto | I'm going to set jumper |
| 16:31.51 | Micetto | thank for help... |
| 16:32.00 | fu3 | why would the phone company program my channel banks to operate at lower voltages? |
| 16:32.04 | fu3 | any conceivable reason? |
| 16:32.13 | fu3 | i only get ~38vdc on a line |
| 16:32.23 | jbalcomb | fu3 energy star complient? |
| 16:32.25 | fu3 | they claim that the CB's can be coded to operate at lower voltages. |
| 16:32.28 | [TK]D-Fender | jbalcomb : Thats FUGLY.... it sends WHATEVER you dial to the PSTN as-is.... |
| 16:32.28 | fu3 | hahahahahah not a chance :) |
| 16:32.32 | jbalcomb | fu3 saving some money? |
| 16:32.34 | fu3 | this gear is from way back |
| 16:32.39 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 16:32.41 | SpaceBass | hey folks |
| 16:32.51 | SpaceBass | anyone know if Broadvoice prevents calling mobile number in france? |
| 16:33.37 | *** join/#asterisk dpolitech (n=Owner@207.224.48.130) |
| 16:33.50 | jbalcomb | SpaceBass dunno. someone was in here earlier compaining about some other company blocking calls to Russia |
| 16:33.53 | *** join/#asterisk mrgoby (n=mrgoby@aa.linuxbox.com) |
| 16:34.09 | freat | Hello all... I just got approved Dual Opterons for an Asterisk box. Should I be OK using the x86 64 version of CentOS? (this is my first time w/ 64 bit linux) |
| 16:34.35 | SpaceBass | ok...its actually on their site...and it looks like they permit calls to 6xxxxxxxx numbers... |
| 16:34.42 | SpaceBass | but I get a error about invalid URI when I try it |
| 16:34.43 | heroine | puzzled: are you using app_rxfax/app_txfax ? or just doing T38 passthrought ? |
| 16:35.03 | jbalcomb | [TK]D-Fender its for the incoming context. does that mean all incoming calls will match? |
| 16:35.54 | *** join/#asterisk Champi (i=Champi@damn.e-leet.be) |
| 16:35.55 | puzzled | heroine: only T.38 but I'm planning to add rxfax/txfax to my rpm |
| 16:36.32 | [TK]D-Fender | jbalcomb : Basically if that's an incoming context it looks like it takes everything in and spits it right out a PRI lings from what it appears. |
| 16:36.58 | jbalcomb | puzzled: are you having much success with that? we've been trying to use rxfax and some other hack job but we are seeing about 30% failure rate on faxes. very inhibiting. |
| 16:37.16 | coppice | someone has T.38 termination working now :-) |
| 16:37.21 | heroine | puzzled: good luck .. after having patch asterisk with the latest patch on bugs.digium , i added app_rxfax/app_txfax and i got new compiling errors :/ |
| 16:37.43 | jbalcomb | [TK]D-Fender ok, thats the line I have for forwarding our fax calls over to our analog PBX via PRI/T1-Xover |
| 16:37.49 | heroine | jbalcomb: you succeeded in getting the whole stuff compiling ? |
| 16:38.10 | *** join/#asterisk dVoka (n=dVoka@CPE-65-29-147-86.wi.res.rr.com) |
| 16:38.13 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 16:38.52 | jbalcomb | heroine: i wasn't involved in the setup of those things. I'm just taking over an existing system. I'm actually in the process of setting up an analog pbx to handle faxing |
| 16:39.07 | heroine | hmm ok |
| 16:39.11 | jbalcomb | heroine I was looking here: http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+fax&diff2=32 |
| 16:39.24 | jbalcomb | heroine: and ended up here: http://www.hylafax.org/archive/2004-07/msg00096.html |
| 16:39.49 | dVoka | Has anyone here tied in Zimbra with Asterisk? |
| 16:40.26 | jbalcomb | heroine: I'd like to keep on with rxfax though. we do a lot with client faxes so it has to work 100% but I'd like to have internal faxing for everyone. |
| 16:40.42 | jbalcomb | dVoka: Not yet, its on the project board for March. |
| 16:41.10 | heroine | hmm .. i see .. |
| 16:41.12 | coppice | jbalcomb: well rxfax works just fine for thousands of people, so you could try fixing your system |
| 16:41.21 | heroine | i'll take a look at that |
| 16:41.37 | tmccrary | In asterisk is a a rule that is the absolute default "if all else fails" rule? the t rule isn't working |
| 16:42.18 | jbalcomb | coppice: its in the works. just got my PO approved today for our new server. Once I get everything working as /normal/ I'm using our current server for testing fun features |
| 16:42.50 | jbalcomb | tmccrary: you looking for a 'exten => ?' that catches the call if nothing else works? |
| 16:44.08 | tmccrary | hi jbalcom, let me repost my current problem. Thank you for the help |
| 16:44.17 | tmccrary | <PROTECTED> |
| 16:44.41 | tmccrary | At that point, between trunks, I get a forbidden on the phone. However, if I call the same extension locally, the voicemail works okay |
| 16:45.04 | tmccrary | but yes, I am looking for an exten rule that is the abolute catch all |
| 16:45.50 | tzanger | tmccrary: use _X. -- if you use _. you will catch all the special stuff too (oshiat) and you don't want that |
| 16:46.05 | *** join/#asterisk natmlt (n=Moriah@216.138.71.250) |
| 16:46.36 | tzanger | tmccrary: I use _X. a lot when I'm trying to figure out what someone's sending me |
| 16:46.38 | tmccrary | yes, I am using _XXXX to match the extension that is called, however, after that rule times out it doesn't go to the next rule priority, it just gives a 487 Forbidden |
| 16:46.46 | tmccrary | exten => _25XX,1,Dial(SIP/${EXTEN},12,Ttr) |
| 16:46.46 | tmccrary | exten => _25XX,2,Voicemail(${EXTEN}) |
| 16:46.47 | tzanger | something like exten => _X.,1,NoOp(I got "${EXTEN}") |
| 16:47.08 | tzanger | if it times out it should go to 't' or 'T' extensions depending on the timeout |
| 16:47.27 | tmccrary | there, locally that works okay. But if I call from a trunk another asterisk, it never gets to prio 2 and gives a 487 |
| 16:47.37 | jbalcomb | what about exten => _s. ? |
| 16:48.10 | tzanger | _s.? |
| 16:48.33 | tmccrary | I don't think the issue is about a starting rule |
| 16:51.04 | jbalcomb | ok, other than that I don't see any reason why 't' shouldn't work. |
| 16:51.19 | AlexCTI | Hi, Someone can hel me with this message: process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 65.240.225.5 |
| 16:51.19 | jbalcomb | tmccrary you check asterisk log to see what its doing with the call instead? |
| 16:51.49 | jbalcomb | AlexCTI: turn off silence suppression, CNG, & VAD on your phones perhaps? |
| 16:52.12 | AlexCTI | but that is comes from my VoIP provider |
| 16:53.28 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 16:53.34 | tmccrary | i'm an idiot! I figured out the problem :) |
| 16:53.40 | jbalcomb | AlexCTI your VoIP providor connects to your Asterisk server? perhaps they can turn it off if you ask them. seems like you could probably ignore it too. I'm sure it will be implemented soon enough. |
| 16:53.50 | jbalcomb | tmccrary bravo! |
| 16:53.54 | eKo1 | anyone here use fonality's pbxtra? |
| 16:54.00 | jbalcomb | tmccrary what was it? |
| 16:54.16 | tmccrary | Basically, my trunk's timeout was the same as the other trunk, so the one that originated the call was terminating a little bit too soon, thus not allowing voicemail to be reached |
| 16:54.27 | jbalcomb | eKo1: no. you thinking about checking it out or having trouble with one? |
| 16:54.39 | tmccrary | so i increased my timeout at Trunk A to be longer than Trunk B, so it would ensure the call would live long enough |
| 16:54.43 | jbalcomb | tmccrary: ah, you found that in the logs? |
| 16:54.57 | AlexCTI | jbalcomb: I'll request them that |
| 16:54.59 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
| 16:55.01 | tmccrary | no,It just kind of popped into my head that that could be the problem |
| 16:55.05 | gambolputty | hi |
| 16:55.17 | jbalcomb | tmccrary good enough |
| 16:55.26 | tmccrary | but looking at the logs is definately something I should have done. I went through the cdr log but it didn't have the info I needed |
| 16:55.31 | tmccrary | thank you for the help jbalcomb |
| 16:55.50 | jbalcomb | iiyo |
| 16:56.06 | tmccrary | is there a way to turn on verbose logging or something similar? I know about the asterisk -vvvvv stuff, but that obviously isn't too useful for production systems |
| 16:56.12 | *** part/#asterisk jebba (n=jebba@70.90.114.101) |
| 16:56.21 | *** join/#asterisk Rhizome (n=rhizome@tor/session/x-0da23a28c5f4b414) |
| 16:56.35 | gambolputty | Is there a way on an analog phone line for * to kill a channel immediately if a caller hangs up prematurely before the timeout value expires in the Dial command? |
| 16:56.44 | jbalcomb | tmccrary: you want verbose in the messages.asterisk but not on the console? |
| 16:57.52 | tmccrary | well asterisk -vvvv is just to standard out. More verbose output in the console would be good as well as something in /var/log |
| 16:58.26 | tmccrary | is there a verbosity command in the console? I've never tried it or is it like asterisk -rvvvvvv |
| 16:58.55 | jbalcomb | tmccrary isn't that what the logger.conf is about? |
| 16:59.10 | jbalcomb | tmccrary set verbose X |
| 16:59.10 | tmccrary | ah, i never noticed that file, lol |
| 16:59.17 | tmccrary | awesome, sounds easy enough |
| 16:59.19 | tmccrary | thank you |
| 16:59.20 | tmccrary | ! |
| 16:59.22 | eKo1 | jbalcomb: i'm thinking about getting one |
| 16:59.38 | eKo1 | but it would be nice to have some opinions about it before i get one |
| 16:59.59 | *** join/#asterisk SibRw0rk (n=SibRw0rk@66.234.235.84) |
| 17:00.39 | jbalcomb | eKo1 it seems like a good price but I wish the basic had the uploadable voice prompts |
| 17:01.36 | jbalcomb | eKo1 depending on the server though, seems like for 995 if it sucks you can just scratch it and use the box for a custom install |
| 17:02.12 | jbalcomb | eKo1 I think I'm going to purchase one for a client in a couple of weeks. |
| 17:03.10 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:04.00 | eKo1 | yeah, i'm thinking the same thing |
| 17:07.09 | guzzler | Hi all! I'm having problems with app_rxfax. It gives me the following error: Unable to find a codec translation path from unknown to unknown |
| 17:07.30 | guzzler | and then, Unable to restore write format on 'Zap/1-1' |
| 17:07.48 | coppice | app_rxfax doesn't give you that error. something in * does |
| 17:08.12 | Damin | Anyone alive? |
| 17:08.13 | guzzler | Anyone has any idea of what might be wrong? |
| 17:08.48 | MikeJ[Laptop] | Damin, nope.. all dead... |
| 17:08.54 | MikeJ[Laptop] | wassup |
| 17:09.05 | guzzler | I'm here |
| 17:09.28 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 17:09.38 | guzzler | coppice: channel.c |
| 17:10.08 | Damin | MikeJ[Laptop]: Want do do me a favor? I need someone to do an SMTP relay test on a box from off our netblock and all the boxes i have access to are behind filters that block outbound port 25! :( |
| 17:10.45 | Damin | MikeJ[Laptop]: We're testing a box that seems to have mucked up relay permissions.. |
| 17:11.04 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 17:12.46 | guzzler | coppice: channel.c gives the first error, and app_rxfax gives the second |
| 17:12.54 | asterisk99 | Security question: I created a user-id (foobar), created a directory (foodir) plus a few files; then did a chown -R foobar:foobar foodir; I want to edit these files under my main login ID; I can't; I tried making my ID a member of foobar; still no-go.... What am I missing???? (Other than the ability to understand the manual) |
| 17:13.19 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 17:13.22 | iDunno | erm - you're missing group write permissions on the files |
| 17:13.28 | iDunno | (probably) |
| 17:13.41 | iDunno | (this is a guess, and off the top of my head) |
| 17:14.23 | iDunno | I'd guess that the permissions on the files are -rw-r--r-- and you want -rw-rw-r-- |
| 17:14.34 | iDunno | (difference between 644 and 664) |
| 17:14.53 | asterisk99 | IDunno: OK... (reason for this is I want to edit the extensions.conf file which I moved to foodir ... otherwise I have term into root and do it with nano (uggh)) |
| 17:15.13 | jesster | anyone having compile issues with asterisk-1.2 release and spandsp 0.2pre25? Im getting problems starting with app_rxfax.c and undeclared items |
| 17:16.27 | *** join/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 17:17.55 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:18.01 | generalhan | whats going on everyone ? |
| 17:18.37 | *** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it) |
| 17:18.41 | Micetto | :( |
| 17:19.32 | Micetto | now i try to change driver |
| 17:19.36 | generalhan | i remember reading something a long time regarding a GotoIf statement if ${CALLERID(number)} is null, does anyone have any idea what im talking about ? LOL |
| 17:19.52 | Micetto | and i use bristuff driver |
| 17:19.58 | *** join/#asterisk RoyK (n=roy@ti211310a080-11530.bb.online.no) |
| 17:20.19 | Micetto | but if i load qozap driver |
| 17:20.27 | Micetto | with ports=1 |
| 17:20.58 | Micetto | the PC crash! O_O |
| 17:21.14 | *** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br) |
| 17:21.56 | [TK]D-Fender | generalhan : if one side of a comparison is null in a $[] evaluation it will bomb out. You need to put a safeguard in to ensure that both side will have something. |
| 17:22.21 | generalhan | [TK]D-Fender: thats it exactly thats what i need to figure out how to do |
| 17:22.37 | sivana | is there a place to download ulaw music on hold on the net? |
| 17:22.49 | generalhan | some people are passing a null CIDNUM and its messing EVERYTHING up so i need a test when the call come in that if its null i assign it SOMETHING to make everything work again |
| 17:24.18 | generalhan | [TK]D-fender: "exten => s,2,GotoIf($["${CALLERIDNUM}" = ""]?s|1000)" this is what i just found ... will this suffice for what i need ? |
| 17:24.58 | generalhan | then i could use a s,1000,SetCALLERIDNUM= whatever |
| 17:25.20 | [TK]D-Fender | generalhan : yup. that would be "" ="" in a null scenario |
| 17:25.25 | *** join/#asterisk SplasPood (n=jwb@host-64-246-146-171.ubr0.alb1.inoc.net) |
| 17:25.33 | [TK]D-Fender | generalhan : which is just fine |
| 17:26.01 | generalhan | [TK]D-Fender: perfect !!! thanks a lot im gonna try that out !! |
| 17:26.41 | generalhan | [TK]D-Fender: can i make CALLERID(number) alpha chars ?? or will it only hold numbers / |
| 17:26.56 | [TK]D-Fender | Actually you don't need the extra white-space or the "s|" part either. |
| 17:27.09 | generalhan | really ? |
| 17:27.12 | [TK]D-Fender | generalhan : I *think* it can hold alpha.... not sure... |
| 17:27.45 | [TK]D-Fender | exten => s,2,GotoIf($["${CALLERID(number)}"=""]?1000) |
| 17:27.50 | [TK]D-Fender | that is the proper 1.2 way |
| 17:28.09 | generalhan | YES !! do you have a link for that so i can reference this later if i forget ?! |
| 17:28.34 | justinu | _Sam--: hello? |
| 17:28.34 | [TK]D-Fender | ~docs |
| 17:28.37 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:28.41 | [TK]D-Fender | WIKI |
| 17:28.58 | generalhan | thanks ... im gonna go test this out !! |
| 17:29.38 | justinu | d-fender: heh... i setup an ip301 for the customer who wanted to use GXP2000s last night... they were blown away |
| 17:31.16 | *** join/#asterisk NassoR (n=root@step.funcitec.rct-sc.br) |
| 17:31.26 | [TK]D-Fender | justinu : as in they liked it? |
| 17:31.28 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 17:31.35 | justinu | yeah... couldn't believe how good it sounded |
| 17:31.44 | *** part/#asterisk MasterYoda (n=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 17:31.46 | bweschke | that's a fairly common reaction with polys |
| 17:31.58 | justinu | yep, which I kept telling them |
| 17:32.03 | [TK]D-Fender | justinu : I'm not sure anything about them counts towards "blown away" so much as good value phone for those not needing speakerphone. |
| 17:32.03 | justinu | but they didn't believe it until they heard it |
| 17:32.12 | austinnichols101 | I was thinking of ordering a couple for the office... |
| 17:32.19 | [TK]D-Fender | justinu : They got to compare side by side right? |
| 17:32.23 | justinu | yes |
| 17:32.30 | justinu | that's where the "blown away" came from |
| 17:32.48 | [TK]D-Fender | justinu : Yeah the audio quality is all there.... the GXP looks more impressive per dollor, but it IS cheap. |
| 17:32.56 | [TK]D-Fender | (negatively speaking) |
| 17:33.07 | justinu | interesting enough... some of the GXP's sound better than others |
| 17:33.12 | justinu | i verified that myself |
| 17:33.23 | [TK]D-Fender | I want one of each of the IP X01 family for when I start my company this year |
| 17:33.23 | justinu | same firmware rev on all of them too |
| 17:33.44 | [TK]D-Fender | justinu : Sounds like shit MFG constraints or they are lying about something... |
| 17:33.54 | *** join/#asterisk ToTo (n=ToTo@host4-163.pool872.interbusiness.it) |
| 17:34.01 | justinu | lying? |
| 17:34.24 | *** join/#asterisk exonic (n=exonic@209.172.11.54) |
| 17:34.28 | [av]bani | [TK]D-Fender: you have a gxp? |
| 17:34.46 | [TK]D-Fender | [av]bani : nope. |
| 17:34.55 | exonic | Hey folks, i've successfully detected faxes with my digum hardware, but a problem i'm hyaving is keeping track of the current ${EXTEN}, it seems to be 'fax' at this point. |
| 17:35.05 | [av]bani | [TK]D-Fender: then how do you know about the construction? |
| 17:35.06 | exonic | Is there a env variable or do I have to set it ? |
| 17:35.13 | [av]bani | [TK]D-Fender: or the audio... |
| 17:35.35 | [TK]D-Fender | [av]bani : It was an abstract comment on how justinu could have 2 otherwise identical phones that differ so much. |
| 17:36.05 | [av]bani | justinu: what mac addrs? i am sure there are different hw rev's |
| 17:36.13 | justinu | one of the gxp's sounded excessively hollow |
| 17:36.31 | [TK]D-Fender | [12:33] <justinu> same firmware rev on all of them too |
| 17:36.33 | [av]bani | did you check the codecs it was using? |
| 17:36.40 | [av]bani | [TK]D-Fender: fw rev != hw rev |
| 17:36.55 | justinu | all running g71 |
| 17:36.57 | justinu | g711 |
| 17:37.17 | *** part/#asterisk NassoR (n=root@step.funcitec.rct-sc.br) |
| 17:37.28 | [av]bani | did you have any of the problematic 03:xx:xx macs? |
| 17:37.36 | justinu | nope |
| 17:37.37 | *** join/#asterisk natmlt (n=Moriah@216.138.71.250) |
| 17:37.43 | [av]bani | :< |
| 17:38.10 | [av]bani | [TK]D-Fender: speakerphone works, it is just a bit quiet. you can bump it up all the way and its usable |
| 17:38.12 | justinu | all my mac's are 05:xx:xx |
| 17:38.44 | [TK]D-Fender | [av]bani : Speakerphone? huh? what? |
| 17:38.57 | [av]bani | [TK]D-Fender: justinu : I'm not sure anything about them counts towards "blown away" so much as good value phone for those not needing speakerphone. |
| 17:39.24 | justinu | they're ok with not having a speakerphone |
| 17:39.40 | [TK]D-Fender | [av]bani : I was talking about the IP 301, not the GXP. |
| 17:40.06 | [TK]D-Fender | Blown away would entail getting a 601 class phone at lower pricing. |
| 17:40.25 | justinu | i'm strictly referring to audio quality |
| 17:40.31 | [av]bani | justinu: the snom 300 looks like it is aimed to kill the ip301 |
| 17:40.38 | [TK]D-Fender | justinu : Yeah, they DSP it up real good.... |
| 17:40.39 | justinu | these jagoffs couldn't care less about the features, etc. |
| 17:40.56 | [TK]D-Fender | [av]bani : link? |
| 17:40.59 | [av]bani | [TK]D-Fender: from what i can tell, polycom cheats... they bass boost the daylights out of the phone |
| 17:41.09 | justinu | that sounds accurate, but it works |
| 17:41.21 | [av]bani | yeah, its a cheat though :) |
| 17:41.27 | justinu | customer comment: "it sounds like the person is sitting right next to me" |
| 17:41.32 | [av]bani | like vendors do with headphones |
| 17:41.45 | justinu | nice guys finish last :P |
| 17:42.07 | [av]bani | http://www.snom.com/snom300_voip_phone.html?&L=1 |
| 17:42.25 | [av]bani | 6 lines vs 2, and FD speakerphone |
| 17:42.45 | justinu | is that new? |
| 17:42.49 | [av]bani | yes |
| 17:42.55 | justinu | cool... i'll have to buy one |
| 17:42.57 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 17:43.02 | diclophis | hello everyone |
| 17:43.03 | justinu | anyone selling them? |
| 17:43.13 | *** join/#asterisk high_ (n=mtryfoss@80.239.93.22) |
| 17:43.16 | [av]bani | not yet |
| 17:43.18 | diclophis | does anyone know of an alternative to the app_voicemail system? |
| 17:43.47 | [av]bani | diclophis: yea, hire a receptionist |
| 17:44.01 | diclophis | ha, nice |
| 17:44.06 | [av]bani | :)) |
| 17:44.28 | diclophis | so i guess it is app_diy.c then |
| 17:44.47 | [av]bani | pretty much |
| 17:44.51 | high_ | how many iax2->sip calls will a "normal" server handle (P4 Xeon, 3 ghz), when no transcoding is used..? (alaw) |
| 17:44.57 | [av]bani | feel free to hack app_voicemail.c |
| 17:45.11 | brad_mssw | any preference for alaw vs ulaw ? |
| 17:45.23 | [av]bani | high_: hmm, 400? |
| 17:45.27 | diclophis | yea.. i all ready have dived into it, but i ami am trying to find some other options |
| 17:46.02 | high_ | okay, thanks |
| 17:47.27 | [TK]D-Fender | [av]bani : And on the downside for the Snom 300 : no PoE, 2 line display w/o soft-keys, only 1 call/line key and if you want to expand you'll have to sacrifice actual call features, and there is no PRICE listed so far. |
| 17:48.15 | [TK]D-Fender | justinu : Is the IP 301 wall-mountable? |
| 17:48.56 | kFuQ | http://krugle.com/product/ <--- a search engine for code |
| 17:50.36 | jesster | [TK]D-Fender: I believe all SoundPoint IP phones are wall mountable, it just depends if the wall mount comes with the phone or not.. |
| 17:50.45 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp55-adsl-212.ath.forthnet.gr) |
| 17:51.57 | *** part/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:52.06 | [TK]D-Fender | jesster : I've never seen the part advertised and its a funny think for me to try to imagine. |
| 17:52.45 | *** join/#asterisk natmlt (n=Moriah@216.138.71.250) |
| 17:52.51 | Abydos313 | good morning everyone |
| 17:54.15 | jesster | [TK]D-Fender: if you need part numbers let me know, usually in stock at ingram micro.. |
| 17:55.40 | [TK]D-Fender | jesster : Yeah I could sue them actually... I have some crap phones I'l like to replace with IP 301's |
| 17:55.51 | [TK]D-Fender | if they'd fit the bill |
| 17:56.49 | *** part/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 17:59.47 | *** join/#asterisk rustyb (n=rustyb@68-235-135-252.atlsfl.adelphia.net) |
| 18:00.44 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc156.dialup.mindspring.com) |
| 18:01.00 | *** join/#asterisk |omni| (i=rob@net98.limelyte.net) |
| 18:01.46 | *** join/#asterisk RoyK (n=roy@ti211310a080-11530.bb.online.no) |
| 18:05.40 | *** join/#asterisk CleanerX (n=nix@p54A3B177.dip0.t-ipconnect.de) |
| 18:05.44 | RoyK | wtf is xpp? |
| 18:06.40 | mog_work | its the xorcom usb channel bank |
| 18:07.42 | justinu | yep, all the polycom phones are wall mountable |
| 18:08.03 | mog_work | its not a new feautre |
| 18:08.06 | mog_work | err feature |
| 18:08.10 | mog_work | it is new hardware |
| 18:08.14 | mog_work | no change in api |
| 18:08.22 | [TK]D-Fender | justinu : Link to the mounting plat and hopfully finished mounted smple? |
| 18:08.27 | RoyK | still new functionality |
| 18:08.29 | *** join/#asterisk katakefalos (i=katakefa@194.214.77.65.in-addr.arpa.ethernext.com) |
| 18:08.35 | mog_work | how so? |
| 18:08.47 | RoyK | the sip jitterbuffer doesn't change the api either |
| 18:08.52 | justinu | fender: it's the same plate the phone comes with... you reverse it |
| 18:08.53 | mog_work | its 0 risk to anyone except xorcom channel bank people |
| 18:08.56 | RoyK | nor does quite a lot of other things |
| 18:08.57 | mog_work | yes it does roy |
| 18:08.58 | justinu | the instructions that come with the phone show you what to do |
| 18:09.16 | *** join/#asterisk trelane_ (n=trelane@209.43.90.13) |
| 18:09.16 | RoyK | mog_work: how? the api? the structure is all the same |
| 18:09.45 | mog_work | oh roy, did you see where i fixed your voicemail thing by the way |
| 18:09.49 | mog_work | it was a bug in list_macro |
| 18:09.54 | mog_work | fixed in head and 1.2 |
| 18:09.58 | mog_work | but anyways |
| 18:10.08 | mog_work | do you really need me to go find the changes? |
| 18:10.23 | RoyK | no |
| 18:10.25 | RoyK | not really |
| 18:10.40 | RoyK | i just pondered about what 'feature freeze' meant |
| 18:10.42 | RoyK | never mind |
| 18:10.49 | RoyK | thanks, anyway |
| 18:11.12 | mog_work | it means feature freeze.... |
| 18:11.18 | mog_work | if digium made a new card |
| 18:11.30 | mog_work | we would add the driver into 1.2 much like we did with 1.0 |
| 18:11.34 | mog_work | if it didnt change anything |
| 18:11.36 | RoyK | just a curious question, should variables and stuff change names in 1.2? |
| 18:11.37 | mog_work | but added said new driver |
| 18:11.41 | RoyK | ok |
| 18:11.55 | iCEBrkr | RoyK: Change names in 1.2? |
| 18:12.17 | RoyK | between revisions |
| 18:12.36 | iCEBrkr | RoyK: I'm still not quite sure I understand what you mean. |
| 18:12.50 | iCEBrkr | like ${DNIS}? |
| 18:13.17 | RoyK | nah |
| 18:13.21 | RoyK | just internal variables |
| 18:13.31 | iCEBrkr | in the code? |
| 18:13.48 | mog_work | we dont do that in an even branch |
| 18:13.54 | mog_work | only in trunk |
| 18:13.58 | RoyK | yes |
| 18:14.28 | mog_work | i think it happened a few times cerca 1.0 but not on purpose |
| 18:14.32 | RoyK | like the 'toms' variable around line 3218 in 1.2.0 that was renamed to 'to' in 1.2.2 |
| 18:14.32 | iCEBrkr | Changing variable names from version to version for whatever reason is high-risk for problems or at least creating a lot more work that's needed. |
| 18:14.44 | RoyK | it's not :( |
| 18:14.46 | RoyK | :) |
| 18:14.50 | RoyK | i was just curious |
| 18:14.55 | mog_work | okies |
| 18:14.56 | RoyK | because it makes patching annoying |
| 18:14.58 | mog_work | happy with your answer? |
| 18:15.01 | RoyK | sure |
| 18:15.11 | mog_work | cool |
| 18:15.31 | iCEBrkr | I know I hate it when I have to rename a variable in my code, it's a bunch of search and replace and hope it didn't stomp on some other variable name or whatever.. |
| 18:15.35 | iCEBrkr | It's a pain in the ass |
| 18:15.46 | RoyK | this was only a local |
| 18:16.04 | RoyK | who = ast_waitfor_n(cs, 2, &toms); |
| 18:17.23 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:19.04 | [av]bani | [TK]D-Fender: ip301 has poe and softkeys? |
| 18:19.54 | justinu | no poe |
| 18:20.02 | justinu | you need an adapter |
| 18:20.07 | [av]bani | so his criticism of the snom 300 applies to the ip301 as well |
| 18:20.26 | justinu | the only polycom that has native 802.3af PoE is the 601 |
| 18:20.29 | justinu | afiak |
| 18:20.35 | [av]bani | look that way |
| 18:20.51 | [av]bani | snom claims poe would add $25 to the mfg cost of the phone |
| 18:20.53 | *** join/#asterisk heroine (n=heroine@m37.net81-66-58.noos.fr) |
| 18:20.54 | [av]bani | i find that suspect |
| 18:20.56 | heroine | back |
| 18:21.34 | *** part/#asterisk flot (n=flot@user241.hovrino.net) |
| 18:21.46 | coppice | PoE should add about $3 right now, which is a heck of a lot |
| 18:23.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:23.40 | Nivex | I'm trying to set up MeetMe so participants need no code, but a moderator can log in with a code and mute the participants |
| 18:23.49 | justinu | so how can grandstream sell a poe phone for so cheap? |
| 18:23.59 | Nivex | I'm reading up on MeetMe in the wiki, but it's not jiveing so far |
| 18:24.32 | RoyK | http://photos.photosig.com/photos/88/38/1703888-b8bae0792e073dcc.jpg |
| 18:25.18 | mog_work | i wil never brush my teeth again RoyK .... |
| 18:25.33 | RoyK | lol |
| 18:25.37 | FuriousGeorge | file[laptop]: are you around |
| 18:26.56 | [TK]D-Fender | RoyK : That is the most curious picture I've seen in a LONG time. its benign at the same time as atristically provocative. |
| 18:27.14 | *** join/#asterisk jasonKr (n=jason@67-41-47-125.bois.qwest.net) |
| 18:27.29 | dVoka | for those of you setting up asterisk pbx installs of 15 - 25 extensions ... what service are you recommending for dedicated bandwidth to ensure the possability of all 25 lines being active with 100 kbps upload bandwidth (I.e. dual t1s, dual 1.5 cable, etc.)? Thanks. |
| 18:27.54 | katakefalos | what would be the best billing solution for commercial use of asterisk any suggestions? |
| 18:29.22 | jasonKr | Anyone know anything about DTMF detection? I'm having loads of problems with my TDM400P detecting DTMF. Some digits are missed others are duplicated |
| 18:29.37 | katakefalos | dVoka> http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 18:29.55 | [TK]D-Fender | dVoka : its not a question of service but of codec. I'd aim for IAX termination so you can trunk it and save on the overhead and then choose a decent codec. |
| 18:30.15 | hensema | any tips on how to prevent robot-like sound? |
| 18:30.29 | hensema | large jitter buffers? |
| 18:31.27 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:31.37 | FuriousGeorge | mog_work: have you heard something about xten releasing a jabber/sip client ala eyebeam |
| 18:31.52 | mog_work | nope |
| 18:31.56 | mog_work | that would be spiffy |
| 18:32.14 | FuriousGeorge | you and me both |
| 18:32.23 | FuriousGeorge | for now ill settle for this snom 360 :) |
| 18:32.28 | FuriousGeorge | and gaim |
| 18:32.54 | [TK]D-Fender | hensema : Use a quality phone with good jitter control and choose your codecs carefully. |
| 18:32.59 | [av]bani | if snom will onlyt fix the (*!@&#)(*&! snom 360 firmware bugs... |
| 18:33.02 | coppice | DTMF detection? nah. nobody here knows anything about that :-) |
| 18:33.14 | jasonKr | I guess not |
| 18:33.15 | justinu | heh |
| 18:33.16 | [av]bani | [TK]D-Fender: the jitter buffers in snom are first-rate, even better than polycom afaict |
| 18:33.19 | jasonKr | My boss is furious |
| 18:33.28 | jasonKr | he wants me to buy a "real" pbx |
| 18:33.30 | mog_work | man i would love to hack on a snom360 |
| 18:33.34 | mog_work | make my own firmware |
| 18:33.37 | mog_work | do iax and jabber |
| 18:33.38 | justinu | coppice: trivia question... why alaw vs ulaw? |
| 18:33.42 | mog_work | but i have no time |
| 18:33.49 | justinu | why two different companding algorithms? |
| 18:33.49 | [av]bani | mog_work: http://www.openhardphone.org/ |
| 18:34.08 | *** join/#asterisk mphill (n=datix@CPE-70-92-245-5.wi.res.rr.com) |
| 18:34.12 | jasonKr | wish I could fix it but I've treid even messing with the defines in dsp.c |
| 18:34.24 | jasonKr | to no avail |
| 18:34.46 | coppice | justinu: ulaw had limitations. the europeans used it for a bit, and before it got too entrenched decided to re-tweak it, producing a-law. |
| 18:34.47 | FuriousGeorge | jasonKr: echo? |
| 18:34.56 | mphill | what command would get asterisk to reload all the config files, and effectivley restart wihthout rebooting the box. I change an extension and its still thinking the extension exists |
| 18:35.05 | justinu | coppice: what kind of limitations? |
| 18:35.16 | mog_work | cool site av |
| 18:35.17 | jasonKr | I've tried with both echocancel on and off doesn't seem to make s diff |
| 18:35.20 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 18:35.25 | jasonKr | altough most problems do come from cell phones |
| 18:35.28 | heroine | mphill: killall asterisk ; asterisk ? :) |
| 18:35.28 | justinu | jasonKr: try runing your rx/txgain values |
| 18:35.34 | justinu | s/runing/tuning/ |
| 18:35.43 | jasonKr | mphill> reload |
| 18:35.57 | jasonKr | up or down? |
| 18:36.03 | FuriousGeorge | jasonKr: for zap definately |
| 18:36.04 | jasonKr | I've tried going up |
| 18:36.05 | FuriousGeorge | down |
| 18:36.06 | justinu | probably down if you're echoing |
| 18:36.07 | mphill | /etc/init.d/asterisk reload? |
| 18:36.09 | FuriousGeorge | tone it down scott |
| 18:36.10 | jasonKr | k |
| 18:36.21 | jasonKr | I'll give that a try |
| 18:36.26 | coppice | justinu: a-law compression is better optimised |
| 18:36.32 | FuriousGeorge | i think you gotta reload zap service |
| 18:36.39 | justinu | coppice: so does it "sound" any different? |
| 18:36.45 | FuriousGeorge | jasonKr: also, what kind of handsets are you using |
| 18:36.59 | jasonKr | all the problems are from the outside lines |
| 18:37.02 | jasonKr | inside work fine |
| 18:37.24 | FuriousGeorge | outside people hear themselves echo? maybe im c@rzy but ive noticed when that is the case its usually cuz your phone stinks |
| 18:37.27 | Abydos313 | it would be nice if a manufacturer put an irc client in the phones :)) |
| 18:37.35 | coppice | justimu: not much. the difference is minor. had u-law become more entrenched before the europeans started playing with this, they would never have bothered |
| 18:37.42 | [av]bani | justinu: http://en.wikipedia.org/wiki/Mu-law_algorithm http://en.wikipedia.org/wiki/A-law_algorithm |
| 18:38.02 | justinu | i've read those |
| 18:38.05 | jasonKr | lol yeah definately worse on cell phones and voip phones calling into the POTS lines |
| 18:38.06 | FuriousGeorge | jasonKr: and the wiring of course |
| 18:38.27 | FuriousGeorge | right, because there is a greater delay |
| 18:38.29 | jasonKr | yeah but the *old* pbx worked on the same lines for years... they seem pretty clear |
| 18:38.34 | justinu | it's the gains |
| 18:38.35 | [av]bani | justinu: alaw has slightly greater dynamic range, but at the cost of small deltas having larger error |
| 18:38.40 | FuriousGeorge | the old pbx mechanically switched? |
| 18:38.44 | [av]bani | justinu: its apples/oranges |
| 18:38.46 | jasonKr | cool I'll definately play with the gains |
| 18:39.07 | jasonKr | No I don't think so. I didn't install... |
| 18:39.14 | FuriousGeorge | jasonKr: and if you insist on using chan_zap get new phones and maybe even clean up your wiring |
| 18:39.23 | [av]bani | justinu: alaw certainly doesnt compress any "better" than ulaw. its just 'different', like PAL/NTSC :/ |
| 18:39.25 | FuriousGeorge | or you will always have crappy sound |
| 18:39.29 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:39.52 | FuriousGeorge | jasonk, then go to snom.com and look at the 360 and tell your boss to buy REAL phones |
| 18:39.57 | jasonKr | Wiring is pretty clean, but then why does digium even sell the TDM400P? |
| 18:40.03 | jasonKr | if it's crap |
| 18:40.11 | FuriousGeorge | its not crap |
| 18:40.12 | coppice | PAL is clearly superior to NTSC. a-law is just a minor tweak |
| 18:40.16 | jasonKr | ?? |
| 18:40.16 | FuriousGeorge | but sip is so much cooler |
| 18:40.21 | jasonKr | LOL |
| 18:40.45 | jasonKr | I hear u |
| 18:40.49 | FuriousGeorge | i got an old pbx sitting on top of asterisk in one place |
| 18:40.54 | FuriousGeorge | and this one handset sounds aweful |
| 18:41.18 | fu3 | does anyone know specifically where in the freebsd zap driver, I can change the loopstart hangup battery drop time? Aparently its set to 500ms, but the phone company says mine is at 300ms. |
| 18:41.31 | FuriousGeorge | meanwhile with sip, i got a remote snom 360, and no echo at all when i take a call here. people think im in the building |
| 18:41.43 | jasonKr | hmm |
| 18:41.59 | jasonKr | we'll I was trying to take the asterisk conversion one step at a time |
| 18:42.08 | jasonKr | replacing the old pbx |
| 18:42.12 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 18:42.13 | jasonKr | then added sip |
| 18:42.14 | justinu | coppice: bah... SECAM-DK forever!! |
| 18:42.25 | jasonKr | maybe I'll have to byte the bullet |
| 18:42.27 | fu3 | wow.. ok i just got "justinu" |
| 18:42.34 | justinu | ? |
| 18:42.36 | FuriousGeorge | jasonKr: i hear ya, maybe it woulda been better to keep the oldpbx on chan zap like i did then slowly add sip phones |
| 18:42.37 | fu3 | just in u |
| 18:42.39 | fu3 | ? |
| 18:42.39 | justinu | lol |
| 18:42.44 | justinu | it's my first name, and my last initial |
| 18:42.46 | fu3 | oh :) |
| 18:42.48 | fu3 | well it rocks |
| 18:42.51 | justinu | lol thx |
| 18:42.53 | coppice | SECAM is great for huge sprawling countries, but makes less sense in smaller ones |
| 18:43.05 | jasonKr | yeah the higher ups are a bit nervous about * now |
| 18:43.11 | jasonKr | grrr |
| 18:43.12 | justinu | coppice: why's that? |
| 18:43.30 | jasonKr | sucks cause my test asterisk setup worked great in my house ;-) |
| 18:43.39 | jasonKr | *still does* |
| 18:43.47 | coppice | interestingly in 1954 the NTSC commitee was in such a hurry to get going, they were presented with the PAL idea and said they didn't have time to get it into the spec :-) |
| 18:43.52 | [av]bani | coppice: yeah, which is why france uses secam ... |
| 18:44.12 | *** join/#asterisk qermit (n=m@chello212186129075.chello.pl) |
| 18:44.16 | justinu | former soviet union also |
| 18:44.37 | coppice | france uses secam cos a french guy invented it. russia and some middle east countries use it for engineering reasons |
| 18:44.38 | FuriousGeorge | jasonKr: you should try the txgain -5 then if that doesnt help -5 the rxgain (i believe its on a 100 -100 scale but i heard you shouldnt go beyond +/- 15 |
| 18:44.42 | fu3 | better ideas are crushed every day simply because of politics |
| 18:44.45 | fu3 | it sucks, but it happens. |
| 18:44.50 | [av]bani | if you look at the secam users, its mainly piddly little countries and communist bloc |
| 18:45.02 | justinu | coppice: could you elaborate on those reasons? |
| 18:45.03 | jasonKr | yeah trying it now |
| 18:45.05 | coppice | secam degrades more gracefully in fringe areas |
| 18:45.06 | [av]bani | and of course ex french colonies |
| 18:45.07 | justinu | i'm just curious |
| 18:45.07 | jasonKr | -5 |
| 18:45.15 | justinu | i see |
| 18:46.21 | [av]bani | the problem with pal and ntsc -- theres 1 ntsc standard, 3 pal and 7 secam (go frenchies!) |
| 18:47.22 | AndyCap | and now htey repeat it with ATSC? |
| 18:47.35 | [av]bani | or DVB :) |
| 18:47.40 | coppice | the 3 pals are due to people trying to block easy imports |
| 18:47.55 | [av]bani | coppice: pal-m is 60hz ... |
| 18:47.57 | AndyCap | http://en.wikipedia.org/wiki/NTSC nice map |
| 18:47.58 | FuriousGeorge | jasonKr: does the outside hear you echo, you hear yourself echo, or both ways? |
| 18:48.00 | [av]bani | nothing to do with imports there... |
| 18:48.33 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:48.34 | FuriousGeorge | jasonKr: and you do realize what you are doing is essentially lowering the incoming and outgoing volume so there is less to echo |
| 18:48.48 | PakiPenguin | evening |
| 18:49.11 | shido6 | erf? |
| 18:49.26 | coppice | that map makes the TV world look quite evenly split :-) |
| 18:49.31 | jasonKr | FuriousGeorge: Yeah -5 does help definately |
| 18:49.47 | jasonKr | FuriousGeorge: still a little off, volume seems good though |
| 18:49.48 | AndyCap | coppice: make it a little RISK game |
| 18:50.10 | AndyCap | what? no tv standard in antarctica? |
| 18:50.17 | [av]bani | coppice: would be more interesting to rank it by population |
| 18:50.22 | heroine | nice week end ppl |
| 18:50.26 | jasonKr | FuriousGeorge: cell phone works now, packt8 phone still doesn't |
| 18:50.34 | coppice | of course, china and india tip PAL massively as the most popular |
| 18:50.39 | [av]bani | yes |
| 18:51.00 | *** join/#asterisk steveaj (n=steve@62.55.147.53) |
| 18:51.04 | [av]bani | but if you rank it by population who actually have a tv ... ;) |
| 18:51.27 | [av]bani | japan and NA have highest tv penetration, india and china the lowest |
| 18:51.30 | coppice | same for china. by viewers probably thre same for india |
| 18:51.39 | FuriousGeorge | packt8 phone echoes when you call analog? |
| 18:52.00 | *** part/#asterisk mphill (n=datix@CPE-70-92-245-5.wi.res.rr.com) |
| 18:52.08 | FuriousGeorge | or analog hears themselves when they call it |
| 18:52.08 | [av]bani | some X billiion population, but less than 1 in 50 have a tv, or something like that |
| 18:52.13 | justinu | technical question, what's the most wattage you can provide over 802.3af PoE? |
| 18:52.23 | coppice | china's TV ownership is huge. in rural india is more a communal viewing thing, i think |
| 18:53.26 | [av]bani | coppice: and yet, china has only ~105m phones |
| 18:53.41 | *** join/#asterisk arctic_import (n=jasonj@mail.uui-alaska.com) |
| 18:53.46 | [av]bani | hm, 300m TVs in mainland china according to stats |
| 18:53.48 | coppice | china has about 250M land lines and 400M GSM phones |
| 18:54.11 | arctic_import | Is there a good book on asterisk? I'm having trouble and needs some reading material |
| 18:54.19 | justinu | ~thebook |
| 18:54.22 | jbot | rumour has it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 18:54.24 | AndyCap | justinu: 12.95W |
| 18:54.32 | justinu | AndyCap: is that a hard limit? |
| 18:54.45 | AndyCap | http://standards.ieee.org/getieee802/download/802.3af-2003.pdf |
| 18:55.27 | AndyCap | page 39 and on |
| 18:55.28 | jasonKr | FuriousGeorge: srry was messing with the config. Packet8 calling into analog line on * |
| 18:55.48 | justinu | andycap: thanx |
| 18:55.48 | [av]bani | ok, china telephones is 1 in 5 per population |
| 18:56.02 | AndyCap | justinu: and no, it seems to support more. :) |
| 18:56.03 | jasonKr | FuriousGeorge: also, my earlier reports were greatly exagrerated... cell phone still not working |
| 18:56.13 | jasonKr | I'm messing with the echotraining now |
| 18:56.16 | jasonKr | and rxgain |
| 18:56.31 | [av]bani | justinu: probably a guide, so you dont melt RJ45 |
| 18:56.56 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
| 18:56.59 | jasonKr | when changing the zapata.conf is a reload all that is necessary or do I need to restart *? |
| 18:57.36 | AndyCap | :P |
| 18:57.59 | justinu | i'm wondering if we can have PoE computers |
| 18:58.07 | justinu | or charge my laptop over PoE |
| 18:58.07 | AndyCap | justinu: jackpc |
| 18:58.40 | AndyCap | justinu: http://www.chippc.com/products/jackpc/ runs windows ce though. :( |
| 18:59.05 | jasonKr | dunno do you really want to have cisco power your equip? |
| 18:59.08 | jasonKr | LOL |
| 18:59.12 | justinu | not disco |
| 18:59.16 | justinu | er cisco |
| 18:59.17 | AndyCap | jasonKr: who said anything about cisco |
| 18:59.22 | justinu | i use dlink 802.3af poe switches |
| 18:59.32 | jasonKr | POE where's the *POWER* comming from |
| 18:59.39 | [av]bani | justinu: i imagine one could use two POE for > watts |
| 18:59.51 | AndyCap | jasonKr: yes, and there are other suppliers of switches than cisco |
| 18:59.52 | jasonKr | ok do you want <insert switch manufacturer here> powering your equip? |
| 19:00.01 | jasonKr | :-) |
| 19:00.14 | [av]bani | heh, the pc-in-a-wall-socket |
| 19:00.14 | AndyCap | jasonKr: for all the equipment that's useless without network, sure why not |
| 19:00.15 | [av]bani | ick |
| 19:00.28 | jasonKr | yeah guess so good point |
| 19:00.34 | justinu | jasonKr: yeah - it's better than cheap chinese wallwarts |
| 19:00.42 | AndyCap | f-f-fire |
| 19:00.48 | justinu | fire fire! |
| 19:00.50 | jasonKr | LOL that's for sure |
| 19:01.12 | [av]bani | amd risc, ew |
| 19:01.20 | jasonKr | of course that's what will be powering ur switch too ;-) |
| 19:02.40 | AndyCap | of course the way this is going you can't run your switches on stock 10A iec connectors anymore. :P |
| 19:02.54 | justinu | yeah |
| 19:03.20 | jasonKr | that's all I need a 20A connector on my switch |
| 19:03.30 | justinu | 30a twist lock! |
| 19:03.47 | FuriousGeorge | jasonKr: you got echocancelwhenbridged=yes? |
| 19:03.51 | [av]bani | of course for a 24 port poe switch, thats still a lot of watts |
| 19:04.04 | justinu | my dlink has a total power budget of 150w I think |
| 19:04.06 | jasonKr | FuriousGeorge: no. I'm down to -15 on rxgain... |
| 19:04.08 | justinu | not all that impressive |
| 19:04.11 | jasonKr | FuriousGeorgetrying that now |
| 19:04.13 | justinu | jasonKr: can you still hear ok? |
| 19:04.17 | justinu | with rxgain set to -15? |
| 19:04.20 | AndyCap | seems you have to supply 15 or so and max draw is 13 |
| 19:04.24 | jasonKr | Dunno just set it |
| 19:04.28 | jasonKr | calling in now |
| 19:05.24 | x86 | whats a cheap IAX2-capable PSTN provider? |
| 19:05.37 | FuriousGeorge | jasonKr: oh heres a big one: build asterisk with MMX support |
| 19:05.44 | FuriousGeorge | you gotta uncomment a line in the make file |
| 19:05.52 | jasonKr | ok |
| 19:06.03 | FuriousGeorge | jasonKr: you gott restart asterisk and zap service you know |
| 19:06.09 | FuriousGeorge | after changing txgain |
| 19:06.09 | jasonKr | sure |
| 19:06.13 | jasonKr | oh |
| 19:06.16 | jasonKr | opps |
| 19:06.19 | jasonKr | LOL |
| 19:06.32 | FuriousGeorge | you may find it real quiet at -15 now |
| 19:06.32 | jasonKr | if I change zapata.conf reload doesn't change it? |
| 19:06.48 | justinu | no, restart * |
| 19:06.52 | justinu | completely |
| 19:06.53 | jasonKr | gotcha |
| 19:06.57 | jasonKr | well then |
| 19:06.59 | FuriousGeorge | i could be wrong but i think you gotta restart both services to be safe |
| 19:07.13 | justinu | zaptel doesn't need to be restarted |
| 19:07.18 | justinu | just asterisk |
| 19:07.19 | jasonKr | k |
| 19:07.21 | AndyCap | <PROTECTED> |
| 19:07.49 | shido6 | wow |
| 19:08.01 | AndyCap | Hmm, that makes 333*13 = 4.3kW. |
| 19:08.32 | AndyCap | in addition to running the monster itself, and cooling it all. |
| 19:08.50 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 19:08.58 | justinu | that's only like 40 amps |
| 19:09.50 | AndyCap | 20 in the civilized (i.e. non-ntsc) parts of the world. ;) |
| 19:10.12 | justinu | heh |
| 19:10.26 | generalhan | lol |
| 19:10.53 | AndyCap | you should fear the -48V |
| 19:11.00 | justinu | no, i deal with that |
| 19:11.13 | brad_mssw | x86: been testing sellvoip.net recently, they're about the cheapest |
| 19:11.26 | x86 | brad_mssw: are they any good? |
| 19:11.27 | brad_mssw | x86: next would be teliax ... |
| 19:11.49 | fu3 | I wish I could get -48V |
| 19:11.50 | jasonKr | FuriousGeorge/Justinu: Restarted * with -5, seems to have helped. LOL before I wasn't doing anything by doing just reloads... |
| 19:11.56 | fu3 | my GD lines are at -38 |
| 19:11.56 | brad_mssw | x86: dunno, haven't run much for testing yet on sellvoip, they have a few servers around the US though (East and West coast) |
| 19:12.09 | brad_mssw | x86: teliax's bad part is they're primarily west coast |
| 19:12.28 | brad_mssw | x86: teliax had been fine for us until latency got bad ... |
| 19:12.42 | x86 | ah |
| 19:12.59 | justinu | jasonKr: common mistake |
| 19:13.20 | brad_mssw | x86: 39ms to sellvoip ny, 16ms to sellvoip miami, 68ms to teliax :/ |
| 19:13.23 | x86 | brad_mssw: does sellvoip allow incoming fax? also, do they let you do LNP? |
| 19:13.48 | jasonKr | <- * newb |
| 19:13.49 | brad_mssw | x86: yes and yes ... though incoming fax is hit and miss obviously, latency is the killer there ... you really don't want over 45ms or so |
| 19:14.17 | FuriousGeorge | jasonKr: im glad it helped |
| 19:14.18 | x86 | i have ~26ms to google... rather consistantly too... |
| 19:14.19 | brad_mssw | x86: sellvoip's web interface is the worst I've seen though, they say they're fixing it though |
| 19:14.23 | jasonKr | Much better thanks |
| 19:14.28 | FuriousGeorge | jasonKr: now test a bunch and see if you have to tweak more |
| 19:14.30 | jasonKr | still a little fussy on the pack8 |
| 19:14.31 | justinu | i was under the impression that fax wasn't affected by latency so much as packet loss |
| 19:14.32 | brad_mssw | x86: where are you located? (state) |
| 19:14.37 | jasonKr | yeah definately |
| 19:14.40 | justinu | coppice: can you comment on that? |
| 19:14.41 | x86 | brad_mssw: illinois |
| 19:14.48 | FuriousGeorge | jasonKr: call cell phones and voip extensions where there is more likely to be delay that would cause echo |
| 19:15.07 | FuriousGeorge | remote voip extensions |
| 19:15.11 | brad_mssw | x86: try 70.42.34.200 (sellvoip's NY server) and voip-co3.teliax.com |
| 19:15.22 | [av]bani | FuriousGeorge: LD calls |
| 19:15.23 | FuriousGeorge | ohter asterisk servers over iax with zap channels |
| 19:15.37 | FuriousGeorge | jasonKr: yeah make a few LD calls too |
| 19:15.47 | [av]bani | fax isnt affected by latency at all. its one way transmission, so no echo |
| 19:15.51 | brad_mssw | x86: for pinging/tracerouting that is |
| 19:15.53 | FuriousGeorge | call france and pretend to be Jerry Lewis. I hear that |
| 19:15.54 | coppice | justinu: for fax packet loss is a killer. latency isn't usually too critical, but huge latency causes timeouts |
| 19:15.56 | [av]bani | or rather, EC is totally disabled |
| 19:15.57 | FuriousGeorge | gets you places |
| 19:16.01 | x86 | brad_mssw: ~36ms average to sellvoip |
| 19:16.06 | justinu | coppice: huge meaning like 500ms or more? |
| 19:16.14 | FuriousGeorge | gotta run |
| 19:16.22 | [av]bani | justinu: i dont think huge latency matters at all for fax. its one way transmission. |
| 19:16.27 | x86 | brad_mssw: voip-co3.teliax.com doesnt even ping ;) |
| 19:16.31 | katakefalos | what would be the best billing solution for commercial use of asterisk any suggestions? |
| 19:16.40 | [av]bani | justinu: only somethin glike 15000ms would matter |
| 19:16.45 | justinu | k |
| 19:16.52 | brad_mssw | x86: odd, you're right, i guess they turned of icmp |
| 19:17.31 | x86 | brad_mssw: i get about to denver... and that's already more latency than sellvoip ;) |
| 19:17.32 | AndyCap | umm, you shouldn't measure rtp latency with icmp echo |
| 19:17.37 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 19:17.37 | brad_mssw | x86: I'd look at the avg time for a traceroute to the last visible hop, which should be something at rockynet.com |
| 19:17.57 | x86 | yeah |
| 19:18.03 | x86 | that's what i was talking about |
| 19:18.07 | brad_mssw | x86: yeah, sellvoip is pretty good |
| 19:18.11 | brad_mssw | x86: latency wise |
| 19:18.25 | brad_mssw | x86: they don't yet offer callerid name incoming, but they said some time in march .... so .... we'll see |
| 19:18.38 | [av]bani | justinu: but any packet loss will kill fax totally |
| 19:18.43 | Netgeeks | kata: kind of hard to give any recommendation, unless you add some detail to the term commercial... what to you want to be able to bill for? calling cards, pre-paid phone service, virtual hosted pbx, etc? |
| 19:19.02 | [av]bani | justinu: t38 is the bestest solution of course... |
| 19:19.06 | justinu | i understand that clever PLC can alleviate some problems with packet loss |
| 19:19.48 | brad_mssw | coppice: really? I thought latency was a killer because > 20ms of silence would cause major issues, or something ... my experience has been that > 60ms you can almost never fax, but I didn't really look at packet loss in that scenerio either |
| 19:19.58 | [av]bani | is ahrimanes around? |
| 19:20.04 | AndyCap | so people don |
| 19:20.40 | Netgeeks | how in the world are you measuring packet loss in milliseconds? |
| 19:21.10 | fu3 | packet transmission is timed in ms. |
| 19:21.11 | brad_mssw | [av]bani: of course, we're all waiting on t38 support :) wonder how much $$ coppice needs to stay motivated ;) Need a donate to steveu button somewhere |
| 19:21.35 | brad_mssw | Netgeeks: who said anything about measuring packet loss in milliseconds ? |
| 19:22.27 | Netgeeks | brad: I misread your statement, apparently, the packet loss in that scenerio wasn't particularly tied to the packet transit time |
| 19:22.32 | [av]bani | brad_mssw: t38 passthru would be sufficient even.... but then all your hw needs to speak t38 |
| 19:22.50 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 19:23.07 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 19:23.08 | brad_mssw | [av]bani: true, actually, got the t38 passthru patch in place already, just no provider that supports it :/ |
| 19:23.21 | [av]bani | brad_mssw: so beat them up till they do |
| 19:24.03 | brad_mssw | [av]bani: heh, most of them seem to use * to terminate their PRIs ... so ... ;) |
| 19:24.14 | [av]bani | the good ones dont |
| 19:24.19 | brad_mssw | this is probably true |
| 19:24.29 | brad_mssw | any recommendations? |
| 19:24.31 | [av]bani | one does not use a pbx instead of a switch |
| 19:24.41 | [av]bani | just because it can does not mean it should :)) |
| 19:25.20 | brad_mssw | had a hell of a time finding a voip provider that supports t38 and business services |
| 19:26.15 | [av]bani | one can use * to band-aid retarded vendor hardware, eg gateway h323 or sip to idiot vendor PRI-only stuff |
| 19:26.24 | [av]bani | but one probably should not run an ITSP off * boxes |
| 19:26.59 | [av]bani | does voicepulse do t38? |
| 19:27.09 | brad_mssw | not that I'm aware of |
| 19:27.22 | x86 | brad_mssw: hmm... sellvoip.net you have to pre-pay... i like getting billed... |
| 19:27.30 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 19:27.40 | brad_mssw | x86: ehh, most IAX providers are pre-pay |
| 19:28.00 | x86 | brad_mssw: and sellvoip.net has $0.01 per minute in the states... i want unlimited US calling |
| 19:28.15 | Egonis | My sangoma card is echoing a fair amount -- will this affect ppl trying to dial extensions incoming? |
| 19:28.18 | x86 | brad_mssw: know of any flat-rate US calling providers doing IAX2? |
| 19:28.19 | Egonis | b/c that's what's happening |
| 19:28.26 | justinu | egonis: tune the gains |
| 19:28.47 | justinu | x86: there's no such thing as "unlimited" |
| 19:28.49 | brad_mssw | x86: doubt you'll find true unlimited calling |
| 19:29.04 | brad_mssw | x86: they all have caps ... some call 2500 minutes unlimited and charge you $.02 after that |
| 19:29.17 | x86 | well that's plenty... |
| 19:30.01 | justinu | then you don't need unlimited |
| 19:30.01 | Netgeeks | marketing folks use different dictionaries that the rest of us |
| 19:30.01 | x86 | hmm |
| 19:30.01 | justinu | you want a 2500 minute package |
| 19:30.01 | x86 | what about vonage? |
| 19:30.01 | brad_mssw | x86: exactly, you don't need unlimited then ... do the math, you won't find a so-called unlimited one for cheaper |
| 19:30.07 | brad_mssw | x86: you can't use asterisk with vonage |
| 19:30.07 | x86 | vonage offers true unlimited, but they dont do IAX :( |
| 19:30.15 | x86 | brad_mssw: what about SIP trunking? |
| 19:30.17 | [av]bani | http://www.youtube.com/watch?v=UBYPaNc57Ik |
| 19:30.19 | Egonis | justinu: I've tried that, is there an easy way about it? should it always be negative gain? |
| 19:30.29 | justinu | i heard that vonage would open up the SIP credentials on residential accounts |
| 19:30.43 | [av]bani | Egonis: what sangoma model? |
| 19:30.46 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 19:30.47 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 19:30.52 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 19:30.54 | justinu | egonis: it depends... you want it to be as low as possible, without people having a hard time hearing. |
| 19:30.55 | brad_mssw | justinu: eh ? their business plus supposedly allows it, but when we were with vonage, they told us we weren't allowed to use asterisk |
| 19:31.19 | Egonis | justinu: So rxgain = -1.0 does about the same as -3.0.. should I go into the positive? |
| 19:31.33 | justinu | which side hears echo? |
| 19:31.37 | justinu | your side, or the far side? |
| 19:31.46 | Egonis | justinu: Internal, the outside caller hears perfect clarity |
| 19:31.53 | justinu | ok, then adjust your txgain down. |
| 19:32.01 | justinu | try increments of 4 unites |
| 19:32.02 | Egonis | justinu: k, ty |
| 19:32.03 | justinu | units |
| 19:32.11 | Egonis | justinu: i.e. -0.40? |
| 19:32.12 | justinu | make the change, restart * completely not just reload |
| 19:32.19 | Egonis | yes |
| 19:32.21 | justinu | no, +/- 4.0 |
| 19:32.29 | Egonis | justinu: Jebus! Okay then |
| 19:32.31 | justinu | so if you're at 0/0, try going -4 txgain |
| 19:32.38 | justinu | make test calls |
| 19:32.43 | justinu | make sure people can hear you speak ok |
| 19:32.51 | justinu | if they can still hear you ok, and you still have echo, go another -4 |
| 19:33.19 | justinu | if you've gone past the point where they start to not be able to hear ok (your voice is too quiet) you'll need to go back up |
| 19:33.24 | [av]bani | http://www.youtube.com/watch?v=UBYPaNc57Ik |
| 19:33.30 | AndyCap | "can you hear me now?" |
| 19:33.31 | justinu | if you still have echo at the very bottom, the problem isn't txgain |
| 19:33.54 | cpm | Why does my asterisk installation suck? |
| 19:34.25 | justinu | did you install it? |
| 19:34.36 | brad_mssw | coppice: btw, are you also working on V150 (Modem over IP) support? Just curious, please don't flame me ;) |
| 19:34.37 | [av]bani | Egonis: what sangoma card you having echo issues with? hopefully not a 104d |
| 19:34.50 | Egonis | [av]bani: A200 |
| 19:34.58 | justinu | egonis: i'm using A101u |
| 19:35.04 | fu3 | how many conversations go on here every day revolving around tx and rxgain? |
| 19:35.08 | fu3 | it seems like its very common |
| 19:35.12 | justinu | that and what phone to use |
| 19:35.15 | fu3 | perhaps it needs to be better documented? |
| 19:35.17 | Egonis | justinu, [av]bani: I adjusted the txgain down to -4.0, and rxgain is -0.0... -4.0 to -8.0 is no different.. but 0.0 to -4.0 is better |
| 19:35.34 | [av]bani | Egonis: hmm.. a200 shouldnt have any echo.. you should call sangoma |
| 19:35.37 | justinu | egonis: -8 still gives echo? |
| 19:35.44 | Egonis | justinu: yes |
| 19:35.51 | justinu | a200 is the analog board, right? |
| 19:35.56 | [av]bani | yes |
| 19:36.00 | [av]bani | with hw ec |
| 19:36.05 | justinu | my advice is generally only useful for digital stuff |
| 19:36.13 | justinu | since I try and avoid analog as much as possible |
| 19:36.13 | Egonis | justinu: Yes |
| 19:36.15 | [av]bani | i would say call sangoma, its their responsiblity to support the card |
| 19:36.23 | Egonis | fudge |
| 19:36.30 | [av]bani | and they should know their own product |
| 19:36.33 | gambolputty | Is there a way on an analog phone line for * to kill a channel immediately if a caller hangs up prematurely before the timeout value expires in the Dial command? |
| 19:36.38 | coppice | brad_mssw: I considered it, but there doesn't seem to be a lot of demand. POS is possibly the main potential user |
| 19:36.52 | justinu | POS? |
| 19:37.00 | [av]bani | point of sale |
| 19:37.05 | [av]bani | eg, credit card verification units |
| 19:37.26 | Egonis | [av]bani, justinu: They tell me to play with gains too |
| 19:37.26 | justinu | ah |
| 19:37.26 | FuriousGeorge | [av]bani: i take it you like the sangoma better than the tdm400/2400 |
| 19:37.26 | x86 | w00t |
| 19:37.29 | AndyCap | they should get with the program and move to ip. :> |
| 19:37.30 | x86 | found a free DID provider :) |
| 19:37.30 | [av]bani | FuriousGeorge: i dont have any of them actually :)) |
| 19:37.35 | [TK]D-Fender | [av]bani : Not all combinations of the A200 have EC.... its an OPTIONAL board... |
| 19:37.36 | justinu | egonis: do you have echo when you call internet * extensions? |
| 19:37.38 | FuriousGeorge | [av]bani: if you had to pick? |
| 19:37.39 | justinu | like voicemail |
| 19:37.40 | [av]bani | [TK]D-Fender: :< |
| 19:37.44 | Egonis | justinu: Not at all |
| 19:37.50 | [av]bani | FuriousGeorge: a200 ec |
| 19:37.51 | brad_mssw | [av]bani: ironically, I write payment processing software for a living ... one connectivity method is ... modem |
| 19:37.55 | justinu | egonis: so what happens w/ txgain -12? |
| 19:38.00 | justinu | can the far end hear you? |
| 19:38.15 | Egonis | justinu: Good question |
| 19:38.15 | FuriousGeorge | [av]bani: yeah and its about the same price too |
| 19:38.20 | [av]bani | brad_mssw: in a past life i wrote software to interface with the little ccv boxes |
| 19:38.35 | brad_mssw | coppice: yeah, we'd be interested in V150 :) how different is it really from T.38? I would assume they have a lot in common |
| 19:38.52 | [av]bani | brad_mssw: and the cc software in the ofis uses 1200 baud modem (whee!) because the banks are stupid |
| 19:39.12 | [av]bani | FuriousGeorge: but i would avoid pstn totally if possible |
| 19:39.14 | brad_mssw | [av]bani: yep, all our modem stuff is still at 1200 baud (for credit/debit/gift) ... |
| 19:39.29 | [av]bani | brad_mssw: i love how financial institutions are still living in the 1980s |
| 19:39.31 | coppice | [av]bani: they actually use 2400, to make it as fast as possible |
| 19:39.36 | justinu | egonis: btw, i think you're the first in channel to have this card |
| 19:39.40 | [av]bani | v32? omg! |
| 19:39.44 | justinu | one of the first, for sure |
| 19:39.51 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 19:40.03 | brad_mssw | coppice: well, that's not necessarily true ... most still say 1200 baud in their specs ... some do allow 2400 buad though, that's true |
| 19:40.04 | [TK]D-Fender | justinu : Amongst, but I've spoken with others... |
| 19:40.23 | [av]bani | coppice: what? shannon says its impossible to >2400 over pots! my engineers say so! blabla |
| 19:40.30 | justinu | heh |
| 19:40.45 | [av]bani | brad_mssw: nobody i ever dialed would allow 2400.. tis all 1200 |
| 19:40.49 | brad_mssw | [av]bani: yeah, well, you think, negotiation time for 56k can be like 45s, where 1200baud is more like 10s ... if the packets are small enough, it doesn't make sense to connect at too high a speed because of that negotaition time |
| 19:40.50 | coppice | [av]bani: try thinking, instead of just trying to be clever |
| 19:40.52 | justinu | maybe ulaw isn't giving enough dynamic range :P |
| 19:41.05 | FuriousGeorge | [av]bani: i'd avoid it too, but i havent done any work that ever required a T1. those are like 500/mo here |
| 19:41.29 | AndyCap | [av]bani: 4800 baud is still the max for pots iirc. |
| 19:41.32 | justinu | i remember playing with modem latency back in the day |
| 19:41.32 | fourcheeze | anyone got func_odbc.so to return a value? |
| 19:41.36 | justinu | 4800bps was the best |
| 19:41.45 | *** join/#asterisk gunsch (n=linux@pD95416EA.dip0.t-ipconnect.de) |
| 19:41.54 | fourcheeze | I'm executing queries fine with func_odbc but no values are returned |
| 19:42.01 | fourcheeze | any ieas? |
| 19:42.05 | fourcheeze | ideas even |
| 19:42.13 | justinu | not until you send us a 4 cheeze pizza |
| 19:42.19 | [av]bani | brad_mssw: 1200baud is usually 5s.. 9600 is a bit more.. and 14.4 and above is ages :) |
| 19:42.41 | AndyCap | baud != bps |
| 19:42.46 | brad_mssw | [av]bani: well, I was including initialization and dial time in there :) |
| 19:43.01 | FuriousGeorge | i heard 4800Bps (big B) is the limitation of copper wiring |
| 19:43.03 | justinu | we know |
| 19:43.06 | justinu | baud is symbol rate |
| 19:43.08 | [av]bani | brad_mssw: S40=10 :) |
| 19:43.10 | *** join/#asterisk unmanaged (n=kvirc@64.89.118.139) |
| 19:43.11 | FuriousGeorge | or rather its the limit |
| 19:43.27 | brad_mssw | [av]bani: not familiar with S40 ... S55=25 :) |
| 19:43.32 | [av]bani | FuriousGeorge: its the limit of 8bits and 8khz sampling |
| 19:43.37 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 19:43.49 | FuriousGeorge | [av]bani: now that you mention it, of course :) |
| 19:43.51 | [av]bani | FuriousGeorge: since DSL would disagree with you |
| 19:44.05 | brad_mssw | [av]bani: oop, S11=25 |
| 19:44.09 | FuriousGeorge | right, i meant with the screeching noise |
| 19:44.16 | [av]bani | brad_mssw: been about 15+ years, that sounds about right :) |
| 19:44.42 | justinu | DSL is not modulated AF, iirc |
| 19:44.53 | AndyCap | you kids and all your new fangled non hayes registers |
| 19:45.02 | [av]bani | +++ |
| 19:45.03 | [av]bani | ATH |
| 19:45.55 | AndyCap | [av]bani: no, you're still here. :P |
| 19:46.07 | [av]bani | curses, foiled again |
| 19:46.07 | brad_mssw | AndyCap: pretty sure S11 is standard hayes ... meaning milliseconds between dialing digits |
| 19:46.15 | AndyCap | S40 and 55 isn't |
| 19:46.49 | brad_mssw | AndyCap: correct, S55 was a typo, meant S11 ... dunno what S40 is |
| 19:46.54 | [av]bani | i was actually reminescing last night w/friends online about old suprafaxmodem |
| 19:47.01 | [av]bani | with the uber LED display |
| 19:47.05 | nestar | s40 is a dope lookin volvo |
| 19:47.06 | nestar | ;) |
| 19:47.20 | [av]bani | how we <3 those modems back in the day |
| 19:47.27 | justinu | i had one of those supra modems |
| 19:47.31 | justinu | it ran pretty hot |
| 19:47.38 | [av]bani | yeah, but ti was a metal case |
| 19:47.44 | [av]bani | intead of cheap plastic |
| 19:47.46 | [av]bani | hmm |
| 19:47.48 | brad_mssw | [av]bani: heh, we just got a couple supramax modems in for testing ... win-modems only ... what a shame |
| 19:47.50 | nestar | http://www.rsportscars.com/foto/05/s40evolve04_02.jpg <-- s40 |
| 19:47.50 | [av]bani | i want a metal hardphone :) |
| 19:47.52 | justinu | i enjoyed my USR Courier Dual Standard more |
| 19:48.02 | [av]bani | if supra made hardphones |
| 19:48.03 | [av]bani | ... |
| 19:48.12 | AndyCap | when will someone make a voip buttphone? |
| 19:48.16 | mikefoo | anyone suggest a decent did inbound provider for production use? |
| 19:48.16 | justinu | supra got bought up by creative, iirc |
| 19:48.24 | [av]bani | yeah, pity |
| 19:48.29 | mikefoo | in the US |
| 19:48.59 | [av]bani | the reason supra went poof is coz their main modem codec guy died |
| 19:49.04 | [av]bani | and then ... thats all she wrote |
| 19:49.05 | justinu | ack |
| 19:49.13 | brad_mssw | mikefoo: teliax, sellvoip, junctionnetworks |
| 19:49.17 | [av]bani | so they sold off to creative... and creative ran it into the ground |
| 19:49.23 | [av]bani | end of an era... pity |
| 19:49.48 | brad_mssw | [av]bani: yeah, the 3com merger almost killed US Robotics too |
| 19:50.04 | brad_mssw | [av]bani: now Zoom bought Hayes ... it's odd how the market has changed |
| 19:50.06 | fourcheeze | first modem I had was 300baud |
| 19:50.19 | fourcheeze | can't remember the name of it now |
| 19:50.23 | AndyCap | mm, read some portrait article of Hayes a while ago |
| 19:50.33 | justinu | i remember the day when everyone had USR sportsters |
| 19:50.38 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 19:50.43 | fourcheeze | still got one of those somewhere |
| 19:50.43 | AndyCap | eew. sportster |
| 19:50.45 | [av]bani | http://modem.dp.ua/PICS/sup288e.jpg ... sniff sniff |
| 19:50.45 | justinu | they must have sold millions of those things |
| 19:50.51 | [av]bani | how we <3 you suprafaxmodem |
| 19:50.55 | AndyCap | now Courier. that's a proper modem |
| 19:50.56 | fourcheeze | my dad still uses a sportster |
| 19:51.09 | [av]bani | ick.. sportsters |
| 19:51.09 | Nugget | My first modem ever was an MPP (Supra, before they were called Supra) |
| 19:51.12 | justinu | i had a courier |
| 19:51.23 | brad_mssw | [av]bani: heh, I've got a few that are just like that |
| 19:51.24 | [av]bani | the ISP i worked for had a rack of 300 sportsters ... scary |
| 19:51.39 | justinu | 1337 dual 16550 UART card |
| 19:51.41 | Nugget | Courier HST is the only "real" modem, naturally. |
| 19:51.42 | AndyCap | [av]bani: they did what? eeew |
| 19:51.49 | justinu | courier dual standard |
| 19:51.50 | Nugget | 16550 or 16550A? ;) |
| 19:51.57 | justinu | HST sucked... it was like ADSL |
| 19:52.00 | [av]bani | http://www.markus-debus.de/museum/img/modem-dataphons21d-full.jpg yay modem! |
| 19:52.07 | justinu | but the direction was switchable |
| 19:52.20 | AndyCap | why didn't they implement that in adsl? |
| 19:52.22 | [av]bani | http://www.cs.unc.edu/Outreach/vr/computer_museum/acousticcoupler.JPG <3 |
| 19:52.25 | AndyCap | ping-pong |
| 19:52.28 | justinu | andycap: good question |
| 19:52.42 | AndyCap | stupid content providers |
| 19:52.46 | Nugget | http://slacker.com/photos/computers/IMG_0864 <-- my third courier |
| 19:53.05 | [av]bani | justinu: HST? no .. you are thinking of PEP |
| 19:53.11 | justinu | no, it was HST |
| 19:53.13 | [av]bani | PEP was half duplex |
| 19:53.19 | [av]bani | and like ADSL... many bands |
| 19:53.20 | AndyCap | [av]bani: hst had 450bps back channel |
| 19:53.22 | Nugget | HST was asymmetric too. |
| 19:53.27 | [av]bani | AndyCap: so did PEP ... |
| 19:53.30 | [av]bani | well |
| 19:53.40 | [av]bani | it had like 30bps backchannel for signaling, not data |
| 19:53.43 | coppice | PEP was ahead of its time |
| 19:53.44 | justinu | the courier kicked ass at busy number detection |
| 19:53.51 | justinu | it was the fastest |
| 19:54.01 | [av]bani | PEP was (is still) good for shit lines |
| 19:54.09 | justinu | i never heard of PEP |
| 19:54.13 | [av]bani | widely used in russia on their crappy lines for FIDONET links |
| 19:54.18 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 19:54.18 | Nugget | PEP was Hayes' competition to HST. |
| 19:54.24 | justinu | ah |
| 19:54.26 | coppice | PEP is quite similar to the way ADSL works |
| 19:54.30 | [av]bani | yep |
| 19:54.31 | AndyCap | Nugget: umm, no Telebit |
| 19:54.34 | Nugget | but since Hayes was late to the game with a sysop program, *all* the boards used HST. |
| 19:54.39 | justinu | telebit... i do remember them |
| 19:54.47 | [av]bani | the $1000 modems as large as a PC |
| 19:54.56 | justinu | telebit trailblazer |
| 19:55.06 | AndyCap | motorola bought them? |
| 19:55.14 | x86 | is ther another outbound PSTN provider like FWD i can use for only toll-free calls? |
| 19:55.18 | austinnichols101 | I remember when we got our first Hayes 1200 full-card. Everyone in the office stood around and marveled at the speed (4 x our 300 baud modem) |
| 19:55.25 | [av]bani | dunno what happened to telebit |
| 19:55.37 | [av]bani | other than... v32 and family killed them |
| 19:55.54 | jasonKr | FuriousGeorge/Justinu: Thanks a million, probably saved my * setup (I didn't want to go buy another pbx!) DTMF is much improved rxgain=-2.1 THANK! |
| 19:55.59 | jasonKr | THANKS! |
| 19:56.03 | justinu | jasonKr: w00t! |
| 19:56.10 | [av]bani | ah.. cisco bought telebit in 1996 for $200m (!!) |
| 19:56.14 | freat | anyone here using 64 bit linux for their asterisk install? I just got a dually opteron box approved... wondering if anyone has trouble w/ asterisk on 64 bit or not... |
| 19:56.15 | justinu | wow |
| 19:56.24 | [av]bani | freat: works fine |
| 19:56.25 | AndyCap | acha. cisco bought them |
| 19:56.36 | AndyCap | http://www.brouhaha.com/~eric/personal/telebit.html |
| 19:56.36 | freat | [av]bani: cool thanks |
| 19:56.51 | justinu | http://en.wikipedia.org/wiki/Telebit |
| 19:56.55 | mikefoo | anyone in US and using * for commercial use? No what inbound provider are you using!! |
| 19:57.02 | justinu | L3 |
| 19:58.53 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 20:03.52 | *** join/#asterisk festr_ (n=festr@ns.regnet.cz) |
| 20:06.02 | *** part/#asterisk jasonKr (n=jason@67-41-47-125.bois.qwest.net) |
| 20:06.59 | *** join/#asterisk pussfeller (n=todd@166-82-183-46.quickclick.ctc.net) |
| 20:07.07 | brodiem | Could someone offer a bit of advice in implementing agent logins in a call center environment? I'm just not sure which method to go about it, such as using AddQueueMember with the SIP CID (SIP/XXX) or with the agent ID (Agent/XXXX), and whether it's better to dynamically add to queue or specify members in queue config, etc |
| 20:07.39 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 20:08.06 | *** join/#asterisk ravenpi (n=chatzill@host-64-65-199-187.man.choiceone.net) |
| 20:08.49 | ravenpi | Dammit. I had some darn SIP soft phone for my notebook's XP partition, but now it's bluescreening. What's a good SIP soft phone for 'doze? |
| 20:09.10 | ravenpi | (Preferably OSS, though inexpensive and/or freeware works, too. ;-) |
| 20:09.33 | arctic_import | YES! I finally got a sip soft phone working. |
| 20:09.39 | brodiem | ravenpi, http://xten.com |
| 20:09.54 | ravenpi | brodiem: Thanks! I'll check it out ASAP. |
| 20:10.21 | brodiem | didn't use the win version, but I'm assuming its the same as the linux ver |
| 20:11.01 | *** join/#asterisk mphill (n=datix@CPE-70-92-245-5.wi.res.rr.com) |
| 20:11.10 | arctic_import | justinu, L3 uses * are you sure? |
| 20:11.28 | mphill | can we ask asterisk@home questions, namely AMP, namely the digital receptionist? |
| 20:11.35 | justinu | no, but I'm using them for SIP origination/termination |
| 20:11.37 | justinu | and I use * |
| 20:12.02 | arctic_import | justinu, Ah. okay. Does it work pretty well? |
| 20:12.14 | justinu | fan-fucking-tastic |
| 20:12.44 | [TK]D-Fender | mphill .... |
| 20:12.45 | [TK]D-Fender | ~amp |
| 20:12.46 | jbot | amp is probably NOT supported here! people using it should join #amportal |
| 20:13.03 | pussfeller | u think a 100mz computer work for an extremely low volume asterisk box |
| 20:13.08 | pussfeller | heh |
| 20:13.09 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 20:13.17 | brodiem | [TK]D-Fender, you run a small call center don't you? That's what the guy at scopserv said anyway |
| 20:13.26 | [TK]D-Fender | yup, 4 agents |
| 20:13.39 | [TK]D-Fender | brodiem : What do you expect of it? |
| 20:13.47 | brodiem | herew as my question: |
| 20:13.49 | ravenpi | pussfeller: I wouldn't try it. I know that my old 90 MHz system had trouble with real-time playing of MP3s; I have to wonder what issues a 90 MHz system would run into. |
| 20:13.51 | brodiem | Could someone offer a bit of advice in implementing agent logins in a call center environment? I'm just not sure which method to go about it, such as using AddQueueMember with the SIP CID (SIP/XXX) or with the agent ID (Agent/XXXX), and whether it's better to dynamically add to queue or specify members in queue config, etc |
| 20:14.03 | arctic_import | someone needs to write an ncurses interface to asterisk administration. All these web based things are nice but they all require alot of software, apache, perl, etc.. A simple perl/python ncurses interface would be awesome. |
| 20:14.07 | mphill | [TK]D-Fender: what would you suggest for that sort of application? |
| 20:14.35 | *** part/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 20:14.48 | pussfeller | ravenpi: i could just use wavs for the moh i guess, but i bet the real hang up would be the conversions between codecs |
| 20:15.09 | [TK]D-Fender | brodiem : thats a cut & paste of your last question. Often I find static agents works best, other times agentcallbackloing. I've seen "psycho" call centers using AgentLogin directly.... thats where they must watch their agents like HAWKS... |
| 20:15.15 | ravenpi | pussfeller: Exactly. That, and, God help you, voicemail. "Just say no." |
| 20:16.05 | brodiem | [TK]D-Fender, in my scenario, agents do not have the same desk/phone every day, and we need to be able to get metrics per agent login |
| 20:16.27 | brodiem | [TK]D-Fender I think agentcallbacklogin is what we'll use for logins, but it's getting those agents to the queue |
| 20:16.56 | brodiem | I noticed that when doing an AddQueueMember based on the SIP channel, the "autologoff" doesn't work so phones just keep ringing if there's nobody at a desk |
| 20:17.22 | [TK]D-Fender | brodiem : AgentCallbackLogin seems to be what you're looking for then... You should also be using the pause feature to let them walk away later... |
| 20:17.49 | justinu | i just found a dialogic D/40B card in my closet |
| 20:17.50 | justinu | go ISA! |
| 20:18.05 | Juggie | justinu, i have a JCT-120LS somewhere. |
| 20:18.11 | brodiem | [TK]D-Fender, would having the agent dial an ext to pause themselves be any better then just using a DND function on the phone so that it makes their extension ring busy? |
| 20:18.31 | justinu | juggie: I used to use the DID-120 cards |
| 20:18.34 | justinu | groundstart |
| 20:18.36 | x86 | what's the cheapest ATA one could buy? |
| 20:18.48 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 20:18.52 | [TK]D-Fender | brodiem : yes it would for sure. |
| 20:19.03 | Egonis | How can I force a callerid for an incoming call? |
| 20:19.19 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:19.36 | Nugget | set the callerid on the incoming call. |
| 20:19.38 | Egonis | i.e. I call a number, zaptel answers, hands it to SIP/100 and says 'BOO!' for CallerID |
| 20:19.44 | Egonis | Nugget: Is it dead simple? |
| 20:20.00 | Nugget | yes |
| 20:20.06 | brodiem | [TK]D-Fender what way is it better? for reporting maybe? and just remove the DND button from the phones |
| 20:20.08 | Egonis | So RTFM, right? |
| 20:20.12 | file | Nugget: what's today's specialty sauce? |
| 20:20.12 | Nugget | precisely :) |
| 20:20.13 | justinu | egonis: how's that gain tuning? |
| 20:20.34 | Egonis | justinu: Much MUCH better w/ -12.0 but now it pops and crackles when they speak (inside the network) |
| 20:20.36 | Nugget | Sweet and Sour! |
| 20:20.42 | file | yay! |
| 20:20.49 | katakefalos | what would be the best billing solution for commercial use of asterisk any suggestions? |
| 20:20.59 | justinu | egonis: pops and crackles... odd. |
| 20:21.01 | file | katakefalos: your own |
| 20:21.05 | Katty | i'mmmmmmmmm the very model of a modern major general |
| 20:21.08 | real-dev | is there a way to set the callerid for an incoming call with AIM? |
| 20:21.10 | justinu | egonis: inside the network? |
| 20:21.13 | Egonis | justinu: yeah, waiting for sangoma to call back |
| 20:21.13 | Katty | i've information animal vegetable and mineral |
| 20:21.15 | justinu | katty: feeling better today, eh? |
| 20:21.18 | Katty | i know the kings of england and i quote.. |
| 20:21.20 | Katty | justinu: yes :> |
| 20:21.22 | justinu | w00t |
| 20:21.22 | Katty | the facts historical! |
| 20:21.24 | real-dev | or is agi the only way? |
| 20:21.29 | Katty | from marathon to waterloo in order categorical! |
| 20:21.46 | katakefalos | file> with acc? |
| 20:22.03 | file | katakefalos: what? |
| 20:22.10 | [TK]D-Fender | brodiem : Well....the "best" way to guarantee things is to use the AgenLogin method which will literally kick them out, but it FORCES them to answer. |
| 20:22.27 | Katty | file: do you still love me? even though my wisdom teeth are gone and i lack wisdom now? |
| 20:22.30 | katakefalos | file> i mean with the asterisk calling card add on? |
| 20:22.45 | justinu | katty: it builds character |
| 20:22.48 | file | katakefalos: I didn't say that, and you didn't say that, the simple answer is the best billing solution for commercial use is your own |
| 20:22.49 | [TK]D-Fender | Katty: And lots about the square of the hypontenuse? ;) |
| 20:22.57 | file | because no matter what you're going to want it to work some way that nobody else has done |
| 20:22.58 | Katty | [TK]D-Fender: yes, many cheerful facts. |
| 20:23.04 | file | Katty: of course! |
| 20:23.08 | [TK]D-Fender | ARRRGGGH!!! |
| 20:23.09 | Katty | file: :> |
| 20:23.10 | brodiem | [TK]D-Fender yeah I definitely don't like that one since they need to keep the phones on-hook |
| 20:23.19 | katakefalos | file> so there is nothing good available |
| 20:23.31 | Katty | mister anthm! |
| 20:23.33 | file | katakefalos: did I say that? |
| 20:23.44 | file | katakefalos: there's many solutions available for many different things, find one that fits what you want - or write your own |
| 20:23.50 | Katty | actually, file you'd probably know. |
| 20:23.55 | file | what do I know? |
| 20:23.55 | Katty | file: when the next cluecon is scheduled. |
| 20:23.59 | Katty | file: or if it's been scheduled yet |
| 20:24.05 | file | I don't think it has, bkw does that stuff |
| 20:24.13 | Katty | bkw_: this is your reminder dear! |
| 20:24.14 | justinu | i heard cairns australia |
| 20:24.15 | file | I just go where the plane ticket tells me to |
| 20:24.41 | katakefalos | ok thanks file thats why am here to ask if anyone can suggest one that he is using...maybe i have to find another channel that has to do with asterisk... |
| 20:24.49 | Katty | file: to the angeliverse. |
| 20:24.59 | file | katakefalos: well, your best bet is to be more specific - what exactly do you want to do? and Google. |
| 20:25.05 | file | Katty: ooooooooh |
| 20:25.22 | katakefalos | k thanks |
| 20:25.30 | brodiem | [TK]D-Fender, I just don't like the fact that if for example an agent logs in and is dynamically added to the queue, and then that person leaves (and doesn't log off the queue), then somebody else coming in on a shift automaticallys starts getting calls but under a different agent ID |
| 20:25.31 | Katty | JunK-Y: which reminds me, i still need to talk to you. |
| 20:25.33 | *** join/#asterisk backblue (n=moo@87-196-34-196.net.novis.pt) |
| 20:25.53 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 20:25.56 | Katty | twisted[asteria]: and i definately need to talk to you. |
| 20:26.04 | Katty | twisted[asteria]: mister workaholic. |
| 20:26.06 | [TK]D-Fender | brodiem : I'm not sure how you'd deal with that unfortunately.... |
| 20:26.11 | brodiem | [TK]D-Fender, I just wish there was a way there could be some visuals on the phone of some sort |
| 20:26.45 | [TK]D-Fender | brodiem : On a phone with XML/XHTML you could keep a screen refreshed with login status, etc.... |
| 20:26.50 | iCEBrkr | brodiem: Sounds like a custom app to me :P |
| 20:27.35 | iCEBrkr | brodiem: Or people just need to be trained to make sure they log out |
| 20:27.44 | iCEBrkr | and a manger needs to keep an eye on it. |
| 20:27.52 | brodiem | [TK]D-Fender it's an aastra 480i so it supports XML, but I don't know of a way for it to just display a banner... you can "push" XML to the phone, but in such a way that displays over top of the entire display until you push "done" |
| 20:27.58 | iCEBrkr | ...at least that's how our call-center software worked. |
| 20:28.18 | brodiem | yeah I suppose so |
| 20:28.20 | file | food consumption in progress |
| 20:28.45 | iCEBrkr | brodiem: Our guys had to login to the desktop software and set their In/Out status |
| 20:29.56 | mikefoo | justinu: you had to go thru testing phases with L3 before you got an account with them? |
| 20:30.03 | justinu | mikefoo: yes |
| 20:30.06 | brodiem | iCEBrkr, weird |
| 20:30.22 | mikefoo | yah the testing phase took us 3 months to complete. |
| 20:30.28 | justinu | took me about a month |
| 20:30.46 | mikefoo | using them for inbound or outbound? |
| 20:30.51 | iCEBrkr | brodiem: well it was an integrated phone system for call centers. |
| 20:30.58 | iCEBrkr | brodiem: so, it makes sense. |
| 20:31.45 | justinu | mikefoo: both |
| 20:32.24 | mikefoo | justinu: cool, how about pricing as such, you need to commit to million a month or something? |
| 20:32.31 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
| 20:32.37 | mikefoo | million minutes that is.. |
| 20:32.44 | djMax | so... I'm hoping I've caught chan_sip red handed. |
| 20:33.01 | justinu | mikefoo: i think it's 25,000USD/mo |
| 20:33.10 | justinu | but i didn't negotiate the contract, i just made it work |
| 20:33.18 | djMax | Here's what I knows. I turned sip debug on, watched a registration from a device with expires=3600, saw it succeed, and yet sip show peers didn't have it. |
| 20:33.25 | mikefoo | justinu: truu.. |
| 20:35.32 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 20:37.05 | djMax | no takers? Unfortunately the amount of changes in chan_sip in 1.2 are daunting in terms of isolating the cause |
| 20:37.47 | Egonis | djMax: That's messed |
| 20:38.27 | djMax | tell me about it... Everybody is trying to get to the * box to kick it out of the building. And I've got no reason to stop them really, it's definitely totally busted. |
| 20:39.09 | justinu | djMax: still having reg problems with your IP500s? |
| 20:39.18 | justinu | what software do you run on the phones? |
| 20:39.32 | *** join/#asterisk dce (n=dce@schools.dce.k12.wi.us) |
| 20:39.32 | djMax | yeah, vicious. But I'm starting to really think it's * |
| 20:39.32 | justinu | I run 501's w/ 1.6.3 |
| 20:39.34 | djMax | * sends back an "OK" to the reg, and STILL drops it. |
| 20:39.35 | justinu | and asterisk 1.2 and don't have this problems |
| 20:39.43 | djMax | I'm not on 1.6.3, that's for sure. |
| 20:39.44 | justinu | drops it right away? |
| 20:39.54 | djMax | hard to define right away, but way before the expire. |
| 20:40.07 | djMax | The expire in the OK ACK was 3600, I checked in like 20 seconds and it was gone from peers. |
| 20:40.07 | justinu | what's your default/min/maxexpiry set at? |
| 20:40.11 | justinu | ok |
| 20:40.11 | *** join/#asterisk steveaj (n=steve@62.55.147.53) |
| 20:40.21 | djMax | default = 3700, min = 600, max = 5000 |
| 20:40.23 | Egonis | djMax: is it doing this to all peers? |
| 20:40.30 | justinu | djMax: try enabling full debug in logger.conf |
| 20:40.30 | Egonis | djMax: wait... do you have pedantic=yes? |
| 20:40.33 | *** join/#asterisk X-Rob_ (n=me@CPE-138-130-232-234.qld.bigpond.net.au) |
| 20:40.35 | dce | Does anyone know if asterisk works as well in 64bit (opteron) or should it be run at 32 bit? |
| 20:40.51 | djMax | hard to know. I see most peers dropping and coming back, but there are 9, so not easy to definitively answer. |
| 20:40.56 | Egonis | dce: I have an Athlon64, but use x86 |
| 20:41.14 | Egonis | djMax: Check pedantic=yes, it should be NO! |
| 20:41.19 | justinu | djMax: well, definitely check to pendantic |
| 20:41.24 | Egonis | djMax: I set pedantic=yes once, and broke the whole setup |
| 20:41.26 | djMax | ok, will turn on logging. pedantic not set at all. |
| 20:41.26 | AlexCTI | <PROTECTED> |
| 20:41.29 | justinu | djMax: and enable the full log, and enable sip debug |
| 20:41.40 | Egonis | djMax: What changed since it worked? |
| 20:41.42 | justinu | djmax: then pastebin that log, and lets take a look. |
| 20:41.51 | natmlt | I one single opteron setup with debian amd64 and one dual opteron setup. The work well. |
| 20:41.55 | dce | egonis: Ok. I'm just trying to figure out if it's worth going with real server hardware or just desktop level stuff |
| 20:42.09 | djMax | I went from 1.2.(something) to 1.2.4. |
| 20:42.27 | Egonis | dce: in terms of zaptel compatability, I use x86 |
| 20:42.30 | justinu | djMax: I'm running 1.2.0 and 1.2.1 |
| 20:42.35 | Egonis | dce: otherwise I'd use x64 |
| 20:42.45 | dce | egonis: I'll need zaptel as well |
| 20:42.47 | djMax | SVN-trunk-r10766 for me |
| 20:42.54 | justinu | i wouldn't run trunk |
| 20:43.01 | natmlt | We use zaptel fine in our Opteron system. |
| 20:43.07 | djMax | is there an easy way to get back to the one you have? |
| 20:43.16 | justinu | yeah - download 1.2.1 tarball |
| 20:43.36 | djMax | and put it in /usr/src where the built one was? |
| 20:43.46 | djMax | perhaps I can find some SVN command to get to that ver |
| 20:43.57 | justinu | djMax: you can download it from the asterisk.org site |
| 20:44.07 | justinu | extract it anywhere, run make install |
| 20:44.23 | justinu | might be a good idea to move all old modules out of /usr/lib/asterisk/modules first. |
| 20:44.30 | djMax | just gives me the shakes to have two independent vers on the same box, but yeah, you're right. |
| 20:45.06 | fu3 | hey.. what are those green panels with the fuses and 66 blocks called? |
| 20:45.30 | Katty | file: conjunction junction, what's your function! |
| 20:45.55 | iCEBrkr | Katty: What's your disfunction? |
| 20:46.22 | file | Katty: I have no function |
| 20:46.24 | file | I'm functionless |
| 20:46.32 | Katty | file: you're a variable without a function? |
| 20:46.38 | iCEBrkr | Undefined Function file() on line 12 |
| 20:46.54 | file | EXACTLY! |
| 20:47.06 | *** join/#asterisk mzo (n=moz@ool-435193b3.dyn.optonline.net) |
| 20:47.21 | mzo | someone fix my fwd? *cry* |
| 20:47.31 | iCEBrkr | file: Dude. HTML invoices without a mime-type or multipart SUCKS, BTW |
| 20:47.45 | file | iCEBrkr: ok |
| 20:47.45 | *** join/#asterisk SplasPood (n=jwb@host-64-246-146-171.ubr0.alb1.inoc.net) |
| 20:47.50 | justinu | file.c:432: warning: unused variable 'file' |
| 20:47.54 | iCEBrkr | file: I'm just saying |
| 20:48.27 | file | iCEBrkr: alrighty |
| 20:50.28 | mzo | anyone have luck why my fwd would still be broken after five days? It still says registration refused over iax? |
| 20:51.35 | mzo | argh brb |
| 20:51.50 | iCEBrkr | How man toll-free NPAs are there.. 4? 800, 866,877,888? |
| 20:52.10 | fu3 | Have you guys heard of NEC ISC-303 channel banks? |
| 20:52.15 | fu3 | Thats what I have here |
| 20:52.42 | fu3 | the voltage going in is -53VDC, and coming out is -35VDC :| |
| 20:53.04 | fu3 | also, does anyone know what "Facility Loss of Signal" errors indicate? |
| 20:53.07 | file | iCEBrkr: for right now yes |
| 20:53.09 | fu3 | a lot of my cards have that error up. |
| 20:53.42 | iCEBrkr | file: I need to filter on those numbers.. NANPA doesn't have a list :( |
| 20:55.46 | *** join/#asterisk rob0 (i=1002@cardinal.lizella.net) |
| 20:57.01 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 20:57.05 | MattB2 | hi all |
| 20:57.20 | mishehu | gah. I got a plantronics headset with rj11, and this cisco 7960 doesn't seem to like it. I get no volume no matter what I do... |
| 20:57.30 | *** part/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 20:57.49 | outtolunc | http://docs.nanpa.com/cgi-bin/npa_reports/nanpa?function=list_npa_non_geo |
| 20:57.50 | MattB2 | got a question about queues - i have a roundrobin queue, 2 members, timeout=10, retry=0. When I execite the queue command I give it a timeout of 30, but it quits after 20 - like it's ringing each of the members but not going back to the first again when both are unavailable. Any ideas? |
| 20:58.47 | iCEBrkr | mishehu: smash it |
| 20:59.04 | mishehu | iCEBrkr: with an icebreaker? |
| 21:00.19 | iCEBrkr | mishehu: That'll do it! |
| 21:00.51 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 21:02.36 | Netgeeks | NPA 500 Personal Communication Services - whats this? is this NANPA's version RFC1918 address space? |
| 21:04.08 | outtolunc | http://www.nanpa.com/number_resource_info/500_codes.html |
| 21:04.43 | rob0 | I was just looking in on a * server running music on hold, and noticed that the mpg123 is running as root. Is there a way to make that run as a non-priv user? |
| 21:05.39 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 21:06.33 | x86 | i found a Sipura SPA-2002 for $59... is that about the cheapest 2-port ATA i'm going to find? |
| 21:07.06 | x86 | rob0: make asterisk run as a regular user, then it wont be able to spawn processes as root ;) |
| 21:07.31 | justinu | x86: yeah, that's a good price. |
| 21:07.40 | justinu | you might find a handytone for less, but beware |
| 21:07.52 | rob0 | hmmm I suppose so. |
| 21:08.03 | rob0 | I guess I assumed * had to run as root. |
| 21:08.58 | rob0 | x86: where was the sipura? |
| 21:09.25 | x86 | justinu: beware? |
| 21:09.42 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 21:10.12 | justinu | x86: handytone isn't as good as sipura, imo |
| 21:10.35 | x86 | would it be best to get a cisco ata-186 or a sipura spa-2002 ? |
| 21:10.39 | mikefoo | beware of doggie |
| 21:11.03 | MattB2 | handytone is awful |
| 21:11.04 | justinu | i'd go with sipura |
| 21:11.44 | dce | Interested in your opinion.. since I have to have a system at each of 11 locations (for 911 calls) wouldn't it make sense to size them so all transcoding is done at each site and a 12th server deals with our T1 lines? |
| 21:12.26 | *** join/#asterisk Himeko (i=himeko@S01060040ca128fc3.ed.shawcable.net) |
| 21:12.43 | justinu | eh? |
| 21:16.16 | _Paulo_ | I'm having 6ms blanks (Unicall/MFCR2 signalling). It's imperceptible in voice calls, but this are ruining my fax transmissions. |
| 21:17.22 | _Paulo_ | coppice told me that its may be due to buffer underuns |
| 21:17.32 | jbalcomb | iCEBrkr http://astguiclient.sourceforge.net/ |
| 21:17.42 | fu3 | how did you detect something at 6ms? |
| 21:17.48 | jbalcomb | iCEBrkr http://astguiclient.sourceforge.net/vicidial.html |
| 21:17.57 | fu3 | Im trying to find something at 500ms and my voltmeter doesnt have that kind of resolution |
| 21:18.10 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc156.dialup.mindspring.com) |
| 21:18.10 | _Paulo_ | recording app_txfax transmissions and decoding them after. |
| 21:18.14 | fu3 | cool |
| 21:18.22 | jbalcomb | fu3 ms or ma? |
| 21:18.25 | fu3 | ms |
| 21:18.31 | fu3 | well |
| 21:18.36 | fu3 | im trying to find a 500ms battery drop. |
| 21:18.36 | jbalcomb | what is ms? |
| 21:18.44 | fu3 | miliseconds |
| 21:19.22 | jbalcomb | didn't realize that was part of the electricity world |
| 21:19.24 | _Paulo_ | may the slips be due to clock sync problems with the telco? |
| 21:19.32 | fu3 | isnt it? |
| 21:19.34 | iCEBrkr | jbalcomb: Vividial is 20mins from me in Clearwater.. Matt joined the Tampa/St. Pete Asterisk Users Group |
| 21:19.42 | fu3 | I should see the line go to 0VDC for those 500ms |
| 21:19.48 | iCEBrkr | err Vicidial |
| 21:19.53 | jbalcomb | fu3 too bad i still can help you. :/ |
| 21:20.01 | fu3 | thanks just the same |
| 21:20.08 | fu3 | its nice to just talk about the issues |
| 21:20.09 | jbalcomb | iCEBrkr right on. isn't that inline with what your trying to do? |
| 21:20.19 | iCEBrkr | jbalcomb: Kinda sure. |
| 21:20.45 | jbalcomb | iCEBrkr are you working with the project or you still having trouble being a team players? ;) |
| 21:21.03 | iCEBrkr | jbalcomb: F you RON |
| 21:21.24 | jbalcomb | haha |
| 21:21.33 | jbalcomb | qtracker rulez! |
| 21:21.46 | iCEBrkr | :-S |
| 21:21.48 | djMax | getting the "super latest" may have fixed it (SIP registration), but still testing. |
| 21:24.04 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 21:24.40 | dce | Anyone have experience with Intel Zeon vs AMD Opteron processor systems? |
| 21:28.41 | *** join/#asterisk tuppa (n=tuppa@eclipse.tuppa.org) |
| 21:28.44 | rob0 | Aha, as usual the wiki to the rescue: http://www.voip-info.org/wiki/view/Asterisk+non-root |
| 21:28.58 | rob0 | but even that page says that mpg123 needs to be SUID |
| 21:29.11 | fu3 | I like my opterons |
| 21:30.29 | AlexCTI | ICEbrkr, Do you know how to Asterisk make perfect sync working as CO agains a PBX attached to it? |
| 21:30.48 | iCEBrkr | ?? |
| 21:31.02 | Netgeeks | Intel Xeon, I suspect you mean, dce |
| 21:31.36 | Katty | Netgeeks :> |
| 21:32.58 | justinu | smoove |
| 21:33.42 | *** join/#asterisk bprice20 (n=brandon@Dynamic-216.120.224.160.hrnoc.net) |
| 21:34.32 | bprice20 | Does anyone have the envelope option for voicemail.conf working with realtime? |
| 21:35.57 | [av]bani | rob0: i prefer using native moh and format_mp3 |
| 21:36.32 | iCEBrkr | HIPPIES! |
| 21:36.53 | Katty | make config, not war. |
| 21:37.10 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:37.25 | iDunno | (quite often) |
| 21:37.36 | iDunno | well, actually, it's ant war, but same thing ;) |
| 21:37.51 | bprice20 | Does anyone have the envelope option for voicemail.conf working with realtime? |
| 21:38.27 | Netgeeks | bprice20: I wasn't going to anwer because I didn't specifically try to do anything with it, but yes, the envelope works with realtime |
| 21:38.34 | bprice20 | oops sorry |
| 21:38.36 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 21:38.41 | Netgeeks | I should clarify... |
| 21:38.42 | bprice20 | my irc client did that |
| 21:38.51 | iDunno | mmhmm ;) |
| 21:38.55 | justinu | heh |
| 21:38.59 | Netgeeks | my dog told to me to say it twice! |
| 21:39.00 | bprice20 | really, it did |
| 21:39.08 | bprice20 | the devil made me do it |
| 21:39.10 | Netgeeks | my cat just looks at me like I'm crazy..... |
| 21:39.18 | rob0 | [av]bani: thanks, I guess personally I don't care much about MOH, but some of the people I work with do. Just seems odd that mpg123 has to be root ... and the Wiki doesn't have much to say on that issue. |
| 21:39.18 | Katty | you are crazy, dear. |
| 21:39.22 | iDunno | heh |
| 21:39.50 | Netgeeks | Says alot about the cat and much less about the dog, doesn't it, Katty? |
| 21:40.05 | cpm | Okay, CAC AB1, (which sucks I know) *still* can't get a ring on my fxs ports. I've changed cards, I changed the card frame, I've changed out the backplane, Anyone have clues? |
| 21:40.24 | iDunno | rob0: I don't, it always worried me ;) |
| 21:40.24 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 21:40.28 | bprice20 | I have serious case of keyboard face from falling asleep trying to get openser to do what i wanted it to do |
| 21:40.32 | bprice20 | it hurts |
| 21:40.33 | Netgeeks | bprice: to clarify, I'm using dynamic realtime voicemail and envelopes work. iDunno about static realtime however |
| 21:40.56 | Katty | Netgeeks: clearly. |
| 21:40.58 | rob0 | True, MOH does tell you that the other side hasn't hung up yet. |
| 21:41.10 | bprice20 | thx netgeeks, did you create a field called envelope and put yes in there, or use the options table |
| 21:41.25 | fu3 | yay.. I just connected my first VOIP phone via SIP. |
| 21:41.28 | fu3 | im so happy! |
| 21:41.32 | Netgeeks | bprice: let me look at the db, and I'll let you know |
| 21:41.38 | iDunno | go fu3 |
| 21:41.39 | *** join/#asterisk kpettit (n=keith@user-0cet2g6.cable.mindspring.com) |
| 21:41.42 | fu3 | :) |
| 21:41.44 | bprice20 | netgeeks: thx |
| 21:41.50 | bprice20 | w00t fu3 |
| 21:41.59 | fu3 | that was easier than I suspected :) |
| 21:42.02 | Katty | fu3: playing with the dialplan is sheer joy :> |
| 21:42.02 | fu3 | how enjoyable. |
| 21:42.06 | fu3 | it sure is! |
| 21:42.07 | iDunno | it's very very easy :) |
| 21:42.17 | fu3 | Although I do get a serious echo on this Snom 190. |
| 21:42.26 | fu3 | do you think that is a local phone issue, or a config issue? |
| 21:42.39 | Katty | is it just that snom? |
| 21:42.42 | Netgeeks | bprice: options |
| 21:42.46 | fu3 | its the only voip phone I Have. |
| 21:42.50 | Katty | hmm. |
| 21:42.51 | fu3 | i could use a soft phone i suppose eh |
| 21:42.57 | Katty | is it analog lines? |
| 21:43.00 | Katty | or from a t1 |
| 21:43.09 | bprice20 | netgeeks, thx |
| 21:43.10 | fu3 | analog to a t1 to a ds3 to the co. |
| 21:43.12 | Katty | or pure voip |
| 21:43.32 | Katty | so you have a channel bank which takes your t1 and turns it into analog lines? |
| 21:43.33 | fu3 | i get the echo when I connect over pure VOIP though |
| 21:43.36 | cpm | morn'n kpettit |
| 21:43.47 | Katty | fu3: hmm, that sounds like an asterisk config issue. |
| 21:43.51 | Katty | fu3: tx and rx values |
| 21:43.53 | fu3 | from the voip phone, to the demo asterisk setup. |
| 21:44.02 | fu3 | ohhh NO!!! not the rx tx stuff! |
| 21:44.02 | fu3 | :) |
| 21:44.07 | Netgeeks | snom 190's are pretty yucky phones |
| 21:44.10 | fu3 | i hear about that ALL THE TIME. But I look forward to playing with it. |
| 21:44.19 | Netgeeks | they are like 2 generations back |
| 21:44.20 | fu3 | Netgeeks.. agreed, so far. It came with the TDM400P. |
| 21:44.24 | Katty | they are kinda icky |
| 21:44.29 | [av]bani | fu3: sounds like the local echo canceler is disabled somehow |
| 21:44.30 | Katty | we have one here, and it's sitting in the warehouse |
| 21:44.30 | Netgeeks | I have one here, it's a cats toy |
| 21:44.35 | Katty | the polycom500s are much dreamier. |
| 21:44.39 | fu3 | but $$ |
| 21:44.57 | fu3 | [av]bani.. agreed. I will check into that. |
| 21:44.58 | kpettit | lo cpm what's up |
| 21:45.19 | kpettit | I bought a couple of those new iax fxs devices. they kick ass |
| 21:45.25 | [av]bani | fu3: for sip-sip you should never get any echo. only time you should possibly get echo is sip->pstn |
| 21:45.27 | kpettit | I'll never buy a Digium IAXY again |
| 21:45.40 | Katty | or when your volume is all screwed up |
| 21:45.42 | Netgeeks | fu3: you aren't using a speakerphone on either end of that call are you? |
| 21:45.49 | Katty | like when Netgeeks calls and yells at me |
| 21:45.52 | Katty | and my phone goes nuts |
| 21:45.57 | Netgeeks | !! |
| 21:45.57 | fu3 | [av]bani.. i'll verify. hang on. |
| 21:46.06 | fu3 | Netgeeks.. no speakerphone. |
| 21:46.49 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.8) |
| 21:46.57 | bprice20 | Netgeeks: you have envelope=yes in options to get that working right |
| 21:47.24 | kpettit | Anybody know how I can have voicemail to email without it storing the voicemail? |
| 21:47.38 | kpettit | I just want the voicemail sent to email, but not stored after it's been sent |
| 21:47.41 | Netgeeks | attach=yes,delete=yes |
| 21:47.46 | Netgeeks | previous for kpettit |
| 21:47.49 | kpettit | sweet!!! Thanks Netgeeks |
| 21:48.03 | Netgeeks | checking again bprice, forgot already will get you exact format |
| 21:48.14 | bprice20 | Netgeeks: why would that play the envelope? |
| 21:48.22 | Netgeeks | that was for kpettit |
| 21:48.30 | fu3 | yeah |
| 21:48.32 | bprice20 | Netgeeks: thx |
| 21:48.35 | kpettit | Netgeeks, is that seperated by comma or | ? |
| 21:48.37 | fu3 | that echo only happens when I bring a POTS phone into the picture. |
| 21:48.48 | kpettit | Netgeeks, |tz=central|attach=yes,delete=yes|saycid=yes|review=yes|envelelope=yes is what I have now |
| 21:48.48 | fu3 | so, tx and rxgain then eh |
| 21:48.54 | [av]bani | fu3: then your fxo is at fault |
| 21:49.00 | Netgeeks | kpettit, check the same voicemail.conf, should be an exact sample like showing that |
| 21:49.20 | fu3 | I can hear the echo on the VOIP phone |
| 21:49.22 | fu3 | but not on the POTS phone. |
| 21:49.45 | [av]bani | your pots has echo most likely, but the latency is so low you dont notice it |
| 21:49.52 | [av]bani | what fxo you using? |
| 21:50.06 | fu3 | I have two FXO ports on a TDM400P card. |
| 21:50.12 | fu3 | and two FXS. |
| 21:50.29 | Katty | someone remind me what the difference between fxo and fxs is |
| 21:50.31 | [av]bani | do you get echo on the fxs? |
| 21:50.37 | Katty | one is for an analog line, and the other is for a phone, right? |
| 21:50.43 | Netgeeks | envelope=no is what I have now (I didn't want envelope anymore, I have envelope=yes when I had it working) |
| 21:50.51 | [av]bani | i bet you will |
| 21:51.04 | fu3 | well.. when I call into the PBX from a cell phone, and connect to the VOIP phone, I get the echo on the voip phone and not the cell. If that answers that? |
| 21:51.07 | Netgeeks | kpettit use the | as the separator, they are two options, just like all the other options |
| 21:51.17 | [av]bani | you mean 'no echo on the pots phone' as in, phone->pstn right ? not phone->fxs->fxo->pstn |
| 21:51.19 | bprice20 | Netgeeks,hmm not for me |
| 21:51.27 | bprice20 | well I know if that syntax worked for you |
| 21:51.31 | bprice20 | it does work |
| 21:51.31 | Netgeeks | bprice: what asterisk verison? |
| 21:51.35 | bprice20 | 1.2.4 |
| 21:51.41 | fu3 | yeah i meant the latter. |
| 21:51.50 | fu3 | the pots phone I have in here is connected to the FXO |
| 21:51.53 | Netgeeks | hrm, might wanna dig through code. I'm using CVS HEAD from 2005-11-28 |
| 21:51.56 | fu3 | but I can do the same thing with a cell phone |
| 21:52.03 | [av]bani | fu3: local echo is caused by your local side interface to the pstn |
| 21:52.09 | fu3 | ok |
| 21:52.12 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-234-37-160.dsl.tul2ok.sbcglobal.net) |
| 21:52.12 | Netgeeks | you could try adding a envelope column and see if it works |
| 21:52.27 | Netgeeks | won't hurt anything and should be a 5 minute test |
| 21:52.29 | [av]bani | fu3: the echo can in the tdm400p is a problem for many people. try the mg2 canceller |
| 21:52.41 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-73.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 21:52.44 | fu3 | I dont know what that is, but I will begin my research. |
| 21:52.47 | [av]bani | fu3: also, some hardware is incompatible with the tdm |
| 21:52.48 | fu3 | Thanks for the tip! |
| 21:53.00 | fu3 | any particular hardware come to mind? |
| 21:53.02 | bprice20 | Netgeeks: yep done that |
| 21:53.12 | [av]bani | fu3: but i would guess your echo canceller in tdm400 driver is misconfigured |
| 21:53.18 | fu3 | ok |
| 21:53.25 | fu3 | I will start digging :) |
| 21:53.37 | Netgeeks | fxs connects to a office line, and fxo connects to a phone... I think |
| 21:53.47 | bprice20 | Netgeeks I tried adding envelope and putting yes in there |
| 21:53.49 | bprice20 | no avail |
| 21:53.56 | fu3 | Netgeeks.. correct. |
| 21:54.17 | [av]bani | fu3: http://www.voip-info.org/wiki-Asterisk+echo+cancellation |
| 21:54.24 | fu3 | thank you |
| 21:55.07 | delmar | fu3, i found that the tdm400 I have was much better for echo compared to the X100 i tried previously. |
| 21:55.26 | [av]bani | heh, x100 is cheaps |
| 21:55.35 | fu3 | I hear the same things.. |
| 21:56.03 | delmar | fu3, u can only tweak the txgain so much before u have comprimised the audio to the extent that the other end can't hear you well.... once you get it as low as is bareable for the other party.. then work on echo |
| 21:56.16 | Netgeeks | I try to remember it by the s in fxs stands for station, meaning the device uses station signalling and thus needs an office line to connect to... whereas the o in fxo means office, and it only talks to things that look like phones! |
| 21:56.21 | delmar | peopl ehere say mg2... i found the kb1 as standard in the Asterisk release.. works great. |
| 21:56.35 | Netgeeks | but sometimes I still get confused, because thats too much to remember for a crazy person |
| 21:56.37 | [av]bani | fu3: http://www.voip-info.org/wiki/view/TDM400P |
| 21:57.16 | [av]bani | echo is the #1 problem with voip installs... |
| 21:57.21 | delmar | yup |
| 21:57.24 | delmar | pain in the bloody ass |
| 21:57.32 | [av]bani | and you cant skimp on it... |
| 21:57.32 | oogle | ew |
| 21:57.52 | oogle | mein minds eyes! |
| 21:57.54 | [av]bani | echo can in the PC is bad craziness |
| 21:58.02 | fu3 | thanks for the URL [av]bani |
| 21:58.25 | [av]bani | fu3: i will send a bill later |
| 21:58.26 | fu3 | delmar.. I dont really want to mess with tx and rxgain if I can avoid it. I heard its best to leave them alone unless there are other issues. |
| 21:58.38 | fu3 | [av]bani send it to Katty |
| 21:59.45 | delmar | fu3, mine are both 0.0 on my tdm400 and things work fine. |
| 22:00.00 | Katty | Katty is not accepting bills at this time. |
| 22:00.01 | Katty | kthxbi. |
| 22:00.07 | delmar | fu3, just re-tested the x100 too.. and thats now 0.0.. workin pretty good. not quite as good as the tdm400 but almost |
| 22:00.19 | joe | anyone running asterisk on centos that has decent packages? |
| 22:00.40 | RoyK | joe: there's source on asterisk.org :) |
| 22:00.43 | fu3 | delmar, thats good to know. |
| 22:00.55 | Netgeeks | Oh joy... I've been trying to log into a clients system via ssh -l root... they use dynamic dns for name resolution, but i've been getting ssh auth rejects.... so after about 10 attempt and some small nmap testing, I get an email from the client.... 'due to recent brute force ssh attempts on our server the password has been changed, and the dynamic dns name now points at www.fbi.gov, please use the ip address.....' |
| 22:01.07 | joe | RoyK: indeed, but no spec files that I saw unless I'm blind to make decent packages |
| 22:01.17 | delmar | NOTE: Common artififacts of various motherboard incompatibilities by the Digium TDM400P are snap/crackle/pop on PSTN calls, and even worse intermittent echo. |
| 22:01.18 | joe | RoyK: :P |
| 22:01.19 | delmar | fuck. |
| 22:01.22 | delmar | this could be my problem... |
| 22:01.29 | delmar | wish i had seen this earlier... |
| 22:01.50 | delmar | accept my audio sounds like .. more like little beeps at different volumes... |
| 22:01.56 | delmar | like supermarket checkout line :P |
| 22:02.50 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 22:03.20 | Netgeeks | what motherboard do you have? |
| 22:03.49 | RoyK | strange..... |
| 22:04.08 | dVoka | IAX or SIP - either of you prefer one over the other (recently setup an asterisk box & configured all inner network tests; now I'm ordering a DID)? I read a few comparisons online and came away from the review thinking that SIP is more "universal" but that theri were some benefits to IAX; curious if either of you have an opinion based on usage. |
| 22:04.34 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 22:04.36 | delmar | the main benifit to IAX is its nat traversal... |
| 22:04.44 | delmar | it simply slides in/out of nat without issue. |
| 22:05.02 | delmar | if you are linking asterisk boxes.. IAX is the one to use. |
| 22:05.03 | russellb | and bandwidth usage, and simplicity of configuration, and ... |
| 22:05.26 | dVoka | really ... would have expected that of SIP too, but read that to be one of the benefits of IAX. Thanks for the feedback! |
| 22:05.29 | delmar | russellb is right.. iax is simple and can save bandwidth when u start trunking and such |
| 22:05.56 | delmar | dVoka, for a non-Nat phone to asterisk solution.. SIP is going to be great. |
| 22:06.14 | russellb | Hey file! |
| 22:06.25 | RoyK | hmmmmmmmm |
| 22:06.41 | Netgeeks | However.... If I follow the discussion correctly IAX has a problem with high volumes. There is a single handler for the port and this can cause issues under high load.... but, there are others on this channel who know about this far more than I |
| 22:06.42 | dVoka | So IAX is better on bandwidth, and config simplicity? am going to have to try both - thanks Delmar and Russelb! |
| 22:07.02 | file | russellb: hi russellb! |
| 22:07.19 | russellb | Netgeeks: that's true. However, we know the solution for it ... it just hasn't been done. :) |
| 22:07.20 | RoyK | connecting a client to server A running 1.0.7 generates audio both ways. connecting the same client to another box with indentical configuration, except that box is running 1.2.4, creates one-way audio with one client, not all, though |
| 22:07.31 | delmar | dVoka, only saves u bandwidth if you are trunking multiple calls |
| 22:07.41 | Netgeeks | russelb: Good to hear! |
| 22:08.13 | delmar | dVoka, if you are simply trying to figure out what to use to terminate to an upstream supplier... just go IAX. |
| 22:08.29 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:08.34 | Netgeeks | russelb: btw: got screenies of the system I've been working on for a while at http://www.netgeeks.net/ you should take a gander some time |
| 22:08.44 | dVoka | delmar: thanks much - at order screen now ... think I'll choose IAX and see how it goes. |
| 22:09.14 | delmar | dVoka, if your Asterisk box is behind NAT ie a router.. ie your Asterisk box's IP is of the 192.168, 10.x.x.x or such.. then you really do need to use IAX. |
| 22:10.25 | delmar | dVoka, my main upstream supplier uses SIP on an as5300, so I had to setup my Asterisk box to actually BE the router.. with the public internet IP on it's eth1, and run an iptables firewall from there. it was just too much of a pain in the ass trying to get SIP to work on the Asterisk box when it was behind the router/firewall |
| 22:10.47 | dVoka | delmar: Cool ... yep, that's how my asterisk box is setup & others I plan to setup would be the same. |
| 22:11.32 | *** join/#asterisk doug (i=doug@zaxxon.telerama.com) |
| 22:12.14 | dVoka | delmar: thanks for the feedback - I surely want to keep * behind router & firewall. Looks like IAX is the way to go. |
| 22:13.04 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 22:13.25 | robin_sz | ooh, the new GXP2000 software is different ... |
| 22:13.41 | robin_sz | the previous version used to make the screen go upside-down |
| 22:13.52 | robin_sz | this version just makes it go blank. neat. |
| 22:14.02 | Netgeeks | lol |
| 22:14.02 | fu3 | in soviet russia, screen make you go upside-down. |
| 22:14.07 | fu3 | :| |
| 22:14.24 | doug | how's one to goto based on the password entered? |
| 22:14.44 | doug | like, bob uses 1234 and gets his app, alice uses 4321 and gets something different... |
| 22:15.08 | fu3 | ? |
| 22:15.10 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 22:15.34 | doug | pseudocode might be something like: |
| 22:15.35 | doug | exten => auth,2,Authenticate(1234,Goto(bob,menu,1)) |
| 22:15.39 | stack_ | I'm looking at picking up Linksys Sipura 941's for my Asterisk setup... can anyone suggest a good headset to go with these? |
| 22:15.47 | doug | exten => auth,2,Authenticate(4321,Goto(alice,menu,1)) |
| 22:15.48 | doug | for example |
| 22:15.55 | [av]bani | robin_sz: progress! |
| 22:16.07 | doug | tho i guess i could treat the passwords as extensions... |
| 22:16.08 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:16.12 | fu3 | haha |
| 22:16.13 | fu3 | just in u |
| 22:16.18 | fu3 | I will think that every time now |
| 22:16.28 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 22:17.27 | fu3 | now you hate me :| |
| 22:20.36 | Netgeeks | doug, I would either use a db lookup and the read function to do that, or you could use read and pattern matching with a context full of 'pins' used to get the user to thier app |
| 22:22.20 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 22:23.00 | Netgeeks | good afternoon, Qwell |
| 22:23.11 | doug | ah |
| 22:23.30 | Qwell[] | Netgeeks: How's the testing going? |
| 22:23.50 | stack_ | can anyone recommend a good, cheap headset? |
| 22:24.08 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:24.10 | Qwell[] | good...and cheap? |
| 22:24.48 | stack_ | yeah I know, quite the paradox |
| 22:24.51 | Netgeeks | Qwell: I've run into a bit of a granite wall |
| 22:24.57 | Qwell[] | Netgeeks: Fun |
| 22:24.59 | doug | i use the plantronics and logitech |
| 22:25.02 | stack_ | but my boss is cheap |
| 22:25.10 | robin_sz | sigh .. blank GXP2000 .. again .. less than 5 minutes that time |
| 22:25.10 | doug | dual-earphone biggies |
| 22:25.25 | stack_ | k I'll take a look at those, thanks |
| 22:25.36 | Netgeeks | seems on both PC hardware as well as the sun hardware, the true CPU load in non-transcoded traffic is NIC interrupt handling |
| 22:26.54 | Netgeeks | since even in a multi-processor environment the physical interrupt from a nic card can only be handled by a single cpu, we limit ourselves to the capacity of that CPU to handle NIC interrupts |
| 22:27.12 | Netgeeks | The sun being a bunch of low-powered procs is even more crippled by this |
| 22:28.00 | Netgeeks | however, even a dual opteron 275 hits interrupt handling long before it's ability to handle sip to sip calls with media |
| 22:28.26 | Netgeeks | so I've been looking at options to reduce the interrupts generated per packet recieved |
| 22:28.48 | Netgeeks | there are some decent looking solutions |
| 22:28.52 | Abydos313 | anyone using iax only.. no sip clients..or combo of iax/sip? curious what firewall issues you have |
| 22:29.39 | Netgeeks | some linux kernel based, some hardware based. Ive ordered some hardware for testing (higher end nic cards that can do IRQ limiting) |
| 22:30.24 | Netgeeks | But this also means that we might be able to see much higher loads on cheap pc hardware than thought possible by the general asterisk users community |
| 22:30.48 | russellb | Netgeeks: that's very interesting. I'd be interested to hear how that goes. |
| 22:31.33 | *** join/#asterisk lesouvage (n=lesouvag@82.74.19.41) |
| 22:31.44 | justinu | netgeeks: i've been doing similar benchmarking |
| 22:32.00 | justinu | netgeeks: and my findings agree with yours |
| 22:32.22 | Qwell[] | hmm |
| 22:32.24 | Netgeeks | The linux kernel has some code referred to as NAPI (New API) which will dynamically change the number of packets per interrupt if the NIC card supports it (Intel E1000/pro and Broadcom BCM57XX cards) |
| 22:32.33 | Netgeeks | This New API looks promissing |
| 22:32.45 | justinu | i love it when programmers call something "new" |
| 22:32.50 | justinu | like java nio (new io) |
| 22:32.56 | robin_sz | hah |
| 22:32.57 | justinu | what happens when a newer io package comes out? |
| 22:33.02 | robin_sz | shows what you know |
| 22:33.10 | robin_sz | thats not new io |
| 22:33.17 | robin_sz | that nonblocking |
| 22:33.39 | Netgeeks | if it doesn't work some NICs support IC (Interupt Coalescence) allowing you to set the max interrupt rate you want to see, after that the card sends more than one frame per interrupt |
| 22:33.47 | robin_sz | camee out in java 1.4 iirc |
| 22:33.53 | doug | this is timeout: exten => t,1,Goto(tauth,1) |
| 22:33.56 | justinu | robin_sz:http://java.sun.com/j2se/1.4.2/docs/guide/nio/ |
| 22:33.58 | doug | what is it when the "t" is an "i" instead? |
| 22:34.04 | justinu | "The new I/O (NIO) APIs introduced in v 1.4 provide new features and improved performance in the areas of buffer management, scalable network and file I/O, character-set support, and regular-expression matching. The NIO APIs supplement the I/O facilities in the java.io package." |
| 22:34.11 | justinu | shows what you know |
| 22:34.23 | robin_sz | ok |
| 22:34.29 | robin_sz | then shows what I know ;) |
| 22:34.32 | robin_sz | bugger all :) |
| 22:34.48 | Netgeeks | so I've got a couple E1000/PRO and Broadcom 5704 cards on the way, when they get here, testing recommences |
| 22:35.15 | Qwell[] | Netgeeks: So, the QFEs only used one CPU? |
| 22:35.20 | Qwell[] | each, or? |
| 22:35.27 | justinu | netgeeks how many CS/sec can you get out of your machines before you hit the wall? |
| 22:35.39 | justinu | (i get about 20k/sec out of xeon machines) |
| 22:35.42 | Netgeeks | Yes, the QFE's got tied to the SBUS port, and each SBUS port was locked to a physical IRQ |
| 22:35.46 | doug | i? anyone? |
| 22:35.53 | austinnichols101 | justinu: new++ |
| 22:36.00 | justinu | austinnichols101: i was thinking newer |
| 22:36.05 | justinu | then maybe newest |
| 22:36.07 | Netgeeks | Call setups, Justin? |
| 22:36.14 | justinu | Netgeeks: context switches |
| 22:36.15 | Qwell[] | I could see how a full QFE on a 450mhz CPU could hurt... |
| 22:36.29 | austinnichols101 | justinu: or new enhanced ultra pro |
| 22:36.42 | Netgeeks | ah, I'm not sure, I'd have to actually go back and look, that wasn't a parameter I was tracking at all |
| 22:37.09 | justinu | netgeeks: vmstat will show you CS counts |
| 22:37.46 | Netgeeks | Qwell, yep, on the sun, using 20ms rtp packets, a single proc would cap when it's associated NIC was doing right around 50Mbps |
| 22:38.00 | justinu | 50mbps is way better than I got |
| 22:38.23 | Qwell[] | I imagine it was around 13mbps per port |
| 22:38.25 | *** join/#asterisk jyukes (n=jyukes@4.78.144.84) |
| 22:38.30 | Qwell[] | which may have helped with the ints some? |
| 22:38.36 | jyukes | yo can BackgroundDetect be used from AGI? |
| 22:39.22 | doug | ah, i => "invalid" |
| 22:39.28 | Netgeeks | I did some bonding across three nics in three different SBUS slots (on three different IRQ's) and was able to hit 140Mbps before it fell apart |
| 22:39.37 | [av]bani | Netgeeks: the e1000 are somewhat better than the broadcoms |
| 22:39.37 | justinu | 140mbps is pretty good |
| 22:39.55 | Qwell[] | Netgeeks: And what was the opteron pushing? |
| 22:40.35 | Netgeeks | Qwell, I didn't get that far.. Only tested the AMD with one gig NIC (a broadcomm onboard nic) |
| 22:41.04 | [av]bani | heh.. broadcom seem favored for onboard nic |
| 22:41.09 | Netgeeks | it's cheaper |
| 22:41.22 | Qwell[] | I wonder if a 10gbit nic would lower the ints any? |
| 22:41.26 | [av]bani | no |
| 22:42.16 | Netgeeks | the E1k/pros have hardware IC capability. if you have one you could look up how to set it and run a quick test to see how much of an effect it has |
| 22:42.22 | jyukes | you can use interrup moderation |
| 22:42.40 | [av]bani | Netgeeks: just about every gbe chip has interrupt mitigation, hw checksum, and all the fixings |
| 22:42.58 | [av]bani | even without interrupt mitigation, most gbe drivers let you switch to polling |
| 22:43.30 | Qwell[] | So...can a single box not actually hangle a gbit NIC fully? |
| 22:43.37 | Qwell[] | or is it just because of the way the data is set? |
| 22:43.45 | [av]bani | Qwell[]: a single gbe nic can max out pci/33 |
| 22:44.08 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 22:44.27 | kuku5 | Anyone know pri's well, Xo emailed me that * is sending out a reset command for the b-channel |
| 22:44.41 | kuku5 | using a q931 trace |
| 22:44.50 | freat | [av]bani: I've been able to get 33MHz PCI up to about 129MBps (megabytes)... but that was bursting from my hdd's so I dunno if that affects it |
| 22:45.16 | [av]bani | Qwell[]: single boxes can saturate gbe, but not with a pci/33 card.. pci/66 and pci-x can do it |
| 22:45.44 | freat | [av]bani: once I moved my RAID array to the on-board (Intel) controller, I jumped up to almost 200MBps (dual wd raptors raid-0) |
| 22:45.47 | Qwell[] | well, they're saying 50mbps was about the limit...why? |
| 22:45.49 | jbalcomb | AlexCTI: if your T1 is going into asterisk then you should just be able to define the span for the Asterisk<->PBX link as not being a timing source. the pbx should take its timing from asterisk. |
| 22:45.59 | [av]bani | freat: thats because its not using the pci bus.. its using the southbridge |
| 22:46.06 | [av]bani | Qwell[]: 50mbps for what? |
| 22:46.08 | freat | [av]bani: yes exactly |
| 22:46.24 | freat | [av]bani: tis why I got the new mobo ;) |
| 22:46.48 | Qwell[] | [av]bani: ints were the problem - not the bus |
| 22:47.19 | Netgeeks | the 50Mbps was the limit i had on the sun using a single nic Fast Ethernet interface running full duplex. It was at this level of traffic (using 20ms rtp packets for the traffic) that the CPU servicing the interrupt from that NIC hit 100% cpu load |
| 22:47.34 | Qwell[] | oh |
| 22:47.40 | Qwell[] | so the other CPUs were "okay"? |
| 22:47.44 | Netgeeks | so the issue is interrupts generated by the nic due to the pps level |
| 22:47.57 | Netgeeks | yes, all the other CPUs were under 5% load |
| 22:47.58 | freat | Netgeeks: ahh... cause the packets are so small |
| 22:48.07 | freat | Netgeeks: a lot of overhead there |
| 22:48.12 | Netgeeks | freat: right, the packet per second rate was huge |
| 22:48.35 | freat | Netgeeks: yeah I think uLaw IAX2 is about 204 Bytes per packet |
| 22:48.43 | justinu | netgeeks you can see that reflected in the CS/sec rate also |
| 22:48.44 | freat | including header etc at 20ms sample rate |
| 22:48.51 | wunderkin | kuku5, i dont know a whole lot about it but yes, asterisk does reset all unused b channels every 60 minutes, whats the problem with that.. or rather what is your real problem? |
| 22:48.53 | justinu | since each interrupt causes a switch to kernel space to run the ISR |
| 22:48.58 | justinu | which is at least one CS |
| 22:49.04 | freat | CS ? |
| 22:49.06 | Netgeeks | anyone doing networking back in the days that the cisco 7500 routers just came out will remember that cisco had GRAND bandwidth numbers for the RSPs, but it turned out only if you used HUGE packet sizes could you ever get close |
| 22:49.10 | justinu | context switch |
| 22:49.10 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 22:49.14 | freat | thx |
| 22:49.20 | [av]bani | Qwell[]: maybe sbus has such limitations... |
| 22:49.28 | [av]bani | Qwell[]: but 50mbps is _nothing_ even for a desktop pc |
| 22:49.53 | Netgeeks | sbus in the E4500 runs at 32 bits wide by 33MHz |
| 22:50.03 | freat | I thought sbus was pretty fast... I mean, they make fibre channel controllers for it |
| 22:50.12 | freat | guess I was wrong |
| 22:50.15 | [av]bani | Netgeeks: sounds like interrupts are pretty expensive to service on the e4500 |
| 22:50.18 | justinu | it is the small packet size that's part of the issue |
| 22:50.29 | justinu | it generates the same amount of interrupts for less bandwidth |
| 22:50.31 | Netgeeks | but... each sbus port is it's own bus |
| 22:50.34 | freat | is TDMoE a possiblity in your environment? |
| 22:50.52 | Netgeeks | so you don't share bandwidth on the bus with anything else |
| 22:51.02 | [av]bani | i had no problem with 100k interrupts/sec on my amd64 with gbe |
| 22:51.05 | *** join/#asterisk SibRw0rk (n=SibRw0rk@66.234.235.84) |
| 22:51.11 | freat | Netgeeks: kind of like pxi-e |
| 22:51.18 | Netgeeks | the sbus host cards plug into the E4500's backplace which is a 2.8 GBps backplane |
| 22:51.29 | freat | pci-e |
| 22:52.00 | Netgeeks | av[bani]: the e4500 procs are years old... 400Mhz Ultrasparc 2's (v9) |
| 22:52.06 | justinu | but they have 8 meg of L2 |
| 22:52.11 | freat | wow |
| 22:52.11 | justinu | pretty impressive |
| 22:52.14 | Netgeeks | yep |
| 22:52.17 | freat | that's nuts expensive |
| 22:52.24 | [av]bani | Netgeeks: probably slow to service interrupts then, and likely the NICs have no hw assist |
| 22:52.27 | justinu | E4500 was mad bucks in the day |
| 22:52.28 | freat | the die must be huge |
| 22:52.38 | justinu | 150k machine for a 4 cpu, iirc |
| 22:52.41 | freat | the silicon wafer or whatever |
| 22:52.46 | [av]bani | Netgeeks: since jsut about ant $15 gbe nic you get these days has total hw assist on every level |
| 22:52.51 | [av]bani | any |
| 22:53.00 | [av]bani | even the cheap-ass chinese POS nics |
| 22:53.04 | [av]bani | (rtl...) |
| 22:53.07 | darkskiez | so are digium making pci-express cards then soon? |
| 22:53.40 | Netgeeks | [av]bani: yep all the sbus nics have no interrupt mitigation |
| 22:53.43 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 22:53.49 | *** join/#asterisk tmccrary (n=tmccrary@68.78.185.254) |
| 22:53.55 | [av]bani | Netgeeks: sux :) |
| 22:54.03 | [av]bani | Netgeeks: likely no hw csum either |
| 22:54.14 | Netgeeks | the sbus gbe nics do, however, they only come in 1000SX format, and I don't have any switches that support that |
| 22:54.18 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 22:54.23 | freat | speaking of old computers... hehe so what was your first? Mine was a wonderful Apple IIe+ (80 column card) |
| 22:54.25 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 22:54.26 | tmccrary | if I get a TE110P, can I use all 24-channels to voice or does it have to be split 50/50 for voice/data? |
| 22:54.30 | justinu | heh, nothing like obsolete standards |
| 22:54.32 | Netgeeks | soooo, I bought a PCI host card for the 4500, and will be testing pci gbe cards |
| 22:54.51 | [av]bani | Netgeeks: my guess, a $15 gbe pci nic will work better than a $2500 sbus nic |
| 22:55.12 | Netgeeks | I will have to agree for the specific scenerio we are talking about |
| 22:55.32 | justinu | i tested on commodity hardware and got less results than he did |
| 22:55.50 | Netgeeks | I had a sinclair thingy first off, then went to an apple //, and then to a //c :) |
| 22:55.54 | austinnichols101 | tmccrary: I have a TE110P with 7 channels voice + 1 D channel (PRI) |
| 22:55.59 | [av]bani | justinu: commodity PCs these days can push as much packets as a $1m cisco did not so long ago |
| 22:56.01 | justinu | ti-99/4a here |
| 22:56.04 | austinnichols101 | tmccrary: the rest is data |
| 22:56.06 | freat | yeah a buddy of mine has been buying up compaq proliants for his house... I keep telling him he would be better off with 1 good server with VMWare or XEN |
| 22:56.11 | justinu | [av]bani: apparently not with asterisk :P |
| 22:56.20 | austinnichols101 | tmccrary: but I also have an adtran in the middle to strip off the data |
| 22:56.22 | [av]bani | justinu: thats the fault of asterisk. i hear SER is good |
| 22:56.28 | justinu | ser doesn't proxy RTP |
| 22:56.32 | [av]bani | justinu: and $1m cisco does not pbx, it routes only |
| 22:56.58 | freat | austinnichols101: I was just talking about that w/ a buddy of mine today. You've got it working good with the adtran channel bank in the middle? |
| 22:56.58 | [av]bani | justinu: SIPX |
| 22:56.59 | austinnichols101 | justinu: siproxd does.. |
| 22:57.12 | justinu | siproxd isn't the same as SER |
| 22:57.39 | tmccrary | austinnichols101: So that is configurable then? thanks |
| 22:57.53 | freat | austinnichols101: so the adtran splits the data out, then you run the partial PRI to the Digium card right? |
| 22:57.54 | austinnichols101 | tmccrary: not by the te110p itself |
| 22:58.04 | Netgeeks | what I really want to compare is this E4500 with the opteron in doing database service... with the 10G of RAM and the 10 procs I bet the sun would make a killer database server.... not that that has anything to do with asterisk |
| 22:58.11 | justinu | bah, you can do that with a single sangoma card, no DSU required |
| 22:58.43 | [av]bani | Netgeeks: compared to... 4-way opteron for 1/10 the cost :)) |
| 22:58.56 | austinnichols101 | freat: yes |
| 22:59.03 | freat | justinu: in our case I want to run the PRI into a redfone device and have that failing over TDMoE between two Asterisk boxes |
| 22:59.15 | justinu | cool |
| 22:59.16 | freat | austinnichols101: cool... good to hear that works well |
| 22:59.19 | Netgeeks | oh, want to hear something else that was very interesting in the sun test? |
| 22:59.21 | tmccrary | why only 7 channels, I though you're supposed to be able to use 12 channels (at least) |
| 22:59.22 | justinu | what if the redphone fails? ;) |
| 22:59.30 | tmccrary | austinnichols |
| 22:59.41 | austinnichols101 | tmccrary: I only wanted 7 voice channels. |
| 22:59.51 | Qwell[] | Netgeeks: Sure |
| 22:59.53 | freat | justinu: hehe... then we call the phone company, and have them forward the number to a VoIP number and handle the calls that way heh |
| 22:59.54 | Netgeeks | I built two sytem drives, one with linux (Aurora beta2) and one with Solaris 10... |
| 22:59.56 | doug | how do i pause or "sleep" for 10 seconds? |
| 22:59.59 | doug | in extensions.conf ... |
| 23:00.00 | justinu | freat: heh |
| 23:00.03 | Qwell[] | aurora did better? |
| 23:00.06 | Netgeeks | Asterisk on solaris 10 was more reliable |
| 23:00.19 | freat | justinu: so you wear a tin-foil hat ;) |
| 23:00.28 | freat | jk |
| 23:00.30 | tmccrary | austinnichols101: I see, thanks |
| 23:00.35 | justinu | no, i'm just used to looking for single points of failure ;) |
| 23:00.41 | Netgeeks | at the extreme end of load, I got maybe 10% better performance.. and the failure modes were much cleaner on solaris versus linux |
| 23:00.52 | Netgeeks | I could reproduce these differences without issue |
| 23:00.58 | [av]bani | Netgeeks: what failure modes? |
| 23:01.00 | freat | justinu: yeah my suspicion is the redfone device is asterisk with a digium card and a NIC |
| 23:01.02 | Qwell[] | 10% better performance with solaris? |
| 23:01.11 | justinu | freat: hmm, i hope not |
| 23:01.16 | doug | sleep? pause? what's the command, anyone know offhand? |
| 23:01.20 | justinu | wait? |
| 23:01.20 | doug | wait? |
| 23:01.24 | justinu | ding ding |
| 23:01.29 | justinu | we have a weener |
| 23:01.30 | Netgeeks | for examply under linux when the nic hit the limit due to cpu interrupt handling, if I kept pushing more calls, then stopped and let all the active calls die |
| 23:01.37 | doug | ah |
| 23:01.42 | doug | you guys are so helpful!! |
| 23:01.51 | justinu | any time, pal |
| 23:01.56 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 23:01.57 | freat | well.. shortly after digium came out w/ quad-span cards, they added quad T-1 / E-1 capability |
| 23:01.58 | freat | http://www.red-fone.com/ |
| 23:02.06 | Netgeeks | on linux there would always be a couple hundred left over calls hanging in limbo, and asterisk would not take a stop now, restart now, etc. command from CLI |
| 23:02.20 | Netgeeks | under solaris, there were no hanging calls and asterisk didn't 'stick' |
| 23:02.22 | [av]bani | Netgeeks: interrupt handling limit? what were you pushing? |
| 23:02.23 | Qwell[] | Netgeeks: Fun... |
| 23:02.28 | justinu | freat: i hope you're wrong .... |
| 23:02.28 | justinu | heh |
| 23:02.39 | tmccrary | Netgeeks: It's freaking solaris what did you expect? :) |
| 23:02.59 | freat | later everyone... good weekend |
| 23:03.02 | *** part/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
| 23:03.12 | Netgeeks | [av]bani: Not sure I know what you are asking here: Netgeeks: interrupt handling limit? what were you pushing? |
| 23:03.20 | [av]bani | Netgeeks: how many packets/sec |
| 23:03.23 | doug | i luf u guyzzzzzz |
| 23:03.34 | Netgeeks | ah, around 50Mbps of 20ms rtp packets |
| 23:03.41 | Netgeeks | didn't look at the actual pps rate |
| 23:04.15 | [av]bani | Netgeeks: i ask because i have no problems pushing 100kpps and no interrupt handling issues... even on low end hardware |
| 23:04.39 | *** join/#asterisk NeonLevel (n=NeonLeve@201.145.227.218) |
| 23:04.52 | NeonLevel | good day everyone |
| 23:04.53 | [av]bani | 50m.. |
| 23:05.07 | Netgeeks | at 20ms packets, that would be what... um 50 pps per call? |
| 23:05.21 | [av]bani | checking |
| 23:05.22 | NeonLevel | is there some doc on how to use the *8 "pickup function" ? |
| 23:05.30 | websae | i am curious---1000gb of bandwidth a month---would that be enough to host 50 clients? |
| 23:05.43 | [av]bani | 172 bytes... |
| 23:05.46 | justinu | 50pps |
| 23:05.48 | justinu | is right |
| 23:06.16 | [av]bani | that was around 40kpps by my calculation |
| 23:06.30 | [av]bani | assuming g711u |
| 23:06.46 | doug | anyhone know where i can snag a pleasant "bong" sound? |
| 23:06.50 | Netgeeks | so about 68.8kbps payload, or about 80kbps payload and 50pps... so given those numbers.... 50Mbps would be |
| 23:06.55 | doug | (and i don't mean huff+bubbles) |
| 23:07.04 | Qwell[] | doug: in that case...no |
| 23:07.06 | doug | a GSM bong sound... |
| 23:07.06 | Netgeeks | 8kbps overall I mean |
| 23:07.10 | Netgeeks | 80 |
| 23:07.11 | doug | 8k |
| 23:07.23 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 23:07.28 | [av]bani | Netgeeks: sounds more like a scheduling issue with linux kernel, solaris' scheduler is different |
| 23:07.36 | [av]bani | Netgeeks: 40kpps / 50mbps is nothing ... |
| 23:07.56 | Qwell[] | redder86: good luck |
| 23:07.56 | Netgeeks | 32k pps or so should be what I was seeing |
| 23:08.14 | redder86 | Qwell: no doubt |
| 23:08.14 | Netgeeks | I'll re-run the test in a couple minutes and see what I come up with |
| 23:08.28 | redder86 | the tech support agent is having trouble identifying the product as theirs |
| 23:08.35 | Netgeeks | just need to reconfigure and boot (hard power-on for the 4500 is a 15 minute ordeal) |
| 23:08.37 | [av]bani | Netgeeks: if you have high sys %cpu then something is dreadfully wrong with your setup |
| 23:08.39 | Qwell[] | redder86: because it isn't |
| 23:08.55 | Qwell[] | 15 minutes? |
| 23:09.02 | redder86 | Cisco bought Sipura and put it under Linksys |
| 23:09.06 | [av]bani | Netgeeks: 50mbps is only ~6mbyte/sec |
| 23:09.10 | Qwell[] | no, linksys bought sipura |
| 23:09.19 | Qwell[] | doesn't mean linksys provides sipura support already though |
| 23:09.29 | Netgeeks | Qwell: yep, 15 minutes |
| 23:09.32 | justinu | the e4500 does take quite some time to boot |
| 23:09.33 | Qwell[] | Netgeeks: Why so? |
| 23:09.34 | [av]bani | did they ever? afaict sipura support was axed when linksys bought them |
| 23:09.34 | redder86 | The Sipura phone numbers get me to Linksys |
| 23:09.37 | Netgeeks | 15 minutes of POST |
| 23:09.40 | justinu | it goes thru some insane hardware POST |
| 23:09.49 | Qwell[] | disable it :p |
| 23:09.54 | websae | question for everyone---what's the ratio for end users and available channels.....for 100 end users how many channels should i have availabe for calling out/recieving calls ? |
| 23:10.02 | Netgeeks | it runs full checks on all the CPUs and runs full mem tests on the 10GB of mem.. lol |
| 23:10.02 | Qwell[] | OBP is flexible, heh |
| 23:10.03 | [av]bani | there havent been any firmware updates to spa3k in a while now |
| 23:10.09 | [av]bani | and the bugs pile up |
| 23:10.09 | justinu | you can't... you can't even get into OpenBoot until it's done, iirc |
| 23:10.17 | Qwell[] | Netgeeks: put in a 128mb stick...then hotswap to 10gb |
| 23:10.23 | [av]bani | Netgeeks: sounds like HP9k... |
| 23:10.27 | Qwell[] | geez, you call yourself a computer guy :P |
| 23:10.27 | websae | any response greatly appreciated :) |
| 23:10.35 | [av]bani | Netgeeks: HP9k is like 15 min boot |
| 23:10.56 | Netgeeks | They really didn't think peiople would turn these big systems off and on often... soft reset is only a 2 minute ordeal |
| 23:11.03 | Qwell[] | I suppose that's true |
| 23:11.15 | Qwell[] | so...why do you? |
| 23:11.16 | [av]bani | well when you turn them on and off the lights dim |
| 23:11.33 | [av]bani | so you try to avoid that :)) |
| 23:11.39 | websae | is it 1 channel/6 end users? is that the ratio? |
| 23:11.40 | justinu | i never had an HP9k that big |
| 23:11.45 | justinu | only the crappy desktop models |
| 23:11.46 | Netgeeks | it generates a good bit of heat and noise, and I have it here in my office |
| 23:11.58 | Netgeeks | so I turn it off when I'm not running tests |
| 23:12.08 | [av]bani | justinu: no 19" washing machine HP9ks? pity. |
| 23:12.15 | justinu | nope |
| 23:12.37 | [av]bani | nothing beats booting a hp9k/712 off a 9-track tape |
| 23:12.44 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 23:12.50 | justinu | IBM AS/400s also take a long time to IPL (boot) |
| 23:12.50 | Netgeeks | the E4500 is cool, it has a sequencer to limit intial power draw... |
| 23:13.01 | Zodiacal | any ideas why i don't have a command in CLI: "zap show channels" zap isn't found :/ |
| 23:13.18 | Zodiacal | im pretty sure they are loaded and configured properly |
| 23:13.22 | [av]bani | Netgeeks: sequencer... engage! polyphasic coils charging.... prepare the photonic torpedo launchers! |
| 23:13.34 | Netgeeks | stepping away a few minutes... client call... lol [av]bani |
| 23:13.55 | [av]bani | launch then e4500 warheads |
| 23:14.19 | *** part/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 23:14.20 | justinu | bah... clients |
| 23:15.39 | dpolitech | YES |
| 23:15.44 | dpolitech | STOOPID CLIENTS |
| 23:15.48 | dpolitech | heh |
| 23:16.04 | justinu | lusers |
| 23:16.15 | lesouvage | websae: This depends completely on the ind of business you are running. Using voip connections incoming lines are not the same as outgoing lines. I think with 1/6 you are save. |
| 23:16.28 | x86 | whats a good, cheap PTSN provider? |
| 23:16.30 | websae | does anyone know what the ratio is for available channels to end users??? is is for every 6 users have 1 available channel (like on a PRI)??? |
| 23:16.51 | justinu | ~erlang |
| 23:16.52 | jbot | Full-featured programming language developed at the Ericsson CS Laboratory. URL: http://www.erlang.org/ |
| 23:17.00 | [av]bani | ~lua |
| 23:17.02 | jbot | somebody said lua was a powerful light-weight programming language designed for extending applications. It is often used as a configuration language. See http://www.lua.org/. |
| 23:17.04 | justinu | ~erlang-b |
| 23:17.31 | x86 | no one? |
| 23:17.37 | [av]bani | x86: your local ilec |
| 23:17.49 | x86 | IAX2-trunkable |
| 23:17.54 | justinu | except they're expensive, and they suck |
| 23:18.18 | *** join/#asterisk AJay-MN (i=AJay@63.231.252.9) |
| 23:18.33 | AJay-MN | Has anyone gotten Distintive ring to work with Broadvoice? |
| 23:20.51 | lesouvage | websae: if you pick a proper ip telephone provider it will be very easy to scale capacity to your needs. |
| 23:21.09 | kuku5 | wunderkin: XO emailed me asking why are we sending it |
| 23:23.08 | wunderkin | kuku5, so you don't have any problems, they are just bugging you that you are resetting the b chans? ummm k.. |
| 23:23.42 | doug | i'm having my PBX dial numbers out for me (i'm using it instead of a calling card) |
| 23:23.46 | wunderkin | thats not a standard thing to do? |
| 23:23.53 | doug | anyone know how i can store the last number dialed and make it available for redial? |
| 23:24.06 | Qwell[] | doug: astdb? |
| 23:24.43 | doug | i'd like to find an example extensions.conf that implemented it... |
| 23:24.46 | doug | any ideal, qwell? |
| 23:24.49 | doug | idea...not ideal... |
| 23:24.51 | *** join/#asterisk flashnet (i=flashnet@Darkstar.AceShells.com) |
| 23:26.10 | kuku5 | its dropping calls sometimes |
| 23:26.44 | *** join/#asterisk Grizzy (i=Generic@adsl-68-127-29-121.dsl.pltn13.pacbell.net) |
| 23:26.45 | wunderkin | and when you drop calls is it when it does the resets? this sounds like a different problem |
| 23:27.00 | kuku5 | thats what we are trying to figure out |
| 23:27.06 | kuku5 | b/c then it would mean it would reset every call |
| 23:27.08 | kuku5 | but it doenst |
| 23:27.34 | wunderkin | it should only be resetting channels that arent in use, i was asking if you looked in the logs when this has happened to know if that is when you loose the calls |
| 23:27.50 | wunderkin | i have heard of people having the problem but im not sure what causes it |
| 23:30.11 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:30.11 | *** join/#asterisk JuRaDoZ (n=Jurado@200.157.129.80) |
| 23:30.31 | doug | how do you store a PBX dialed number in astdb? |
| 23:30.51 | Netgeeks | Okay, results from test run on Solaris.... max traffic hit was just shy of 50Mbps, 1200 active SIP channels |
| 23:31.16 | Netgeeks | context switchs according to vmstat 1 peaked at 26985 |
| 23:31.37 | [av]bani | context switches != interrupts :) |
| 23:31.50 | [av]bani | well maybe on solaris it does |
| 23:31.50 | Netgeeks | justinu wanted to know the cs number |
| 23:32.13 | justinu | interrupts cause context switches, don't they? |
| 23:32.35 | wunderkin | kuku5, i would probably contact digium support, but first find out when you are dropping calls and look at the log |
| 23:32.35 | [av]bani | they can, syscalls do too |
| 23:32.45 | justinu | apparently threads yeilding cause CS |
| 23:32.47 | Netgeeks | the cpu handling the nic card reporded 91% cpu interrupt handling 8% sys, 1% user |
| 23:32.55 | [av]bani | wow... thats a lot |
| 23:33.14 | Netgeeks | of the 9 other cpus, the highest used one was at 95% idle |
| 23:33.18 | [av]bani | no bottleneck in asterisk :) |
| 23:33.39 | Netgeeks | most usage was in user |
| 23:33.43 | Netgeeks | for the idle cpus |
| 23:33.48 | [av]bani | so if you get hw assist... |
| 23:33.52 | Netgeeks | a couple had 1% is sys |
| 23:34.54 | Grizzy | someone willing to help me get started, trying to connect x-ten to a running asterisk, have a small extensions.conf, sip.conf. ? |
| 23:35.12 | Netgeeks | this test was run in solaris. |
| 23:35.27 | Netgeeks | I'll repeat in linux, if I remember I saw alot more cpu in sys under linux |
| 23:35.59 | [av]bani | what linux hw? |
| 23:36.09 | Netgeeks | justinu: how does tha CS number look compared to what you saw? |
| 23:36.20 | justinu | i got around 19,000 |
| 23:36.23 | Netgeeks | same hardware, just rebooted into linux 2.6.13 |
| 23:36.23 | justinu | that was my wall |
| 23:36.34 | doug | ah, gonne crib from a2billing... |
| 23:36.36 | [av]bani | oh, sbus/sparc? |
| 23:36.42 | doug | anyone ever use a2billing? |
| 23:36.42 | Netgeeks | yep |
| 23:37.02 | [av]bani | heh. dunno if linux is really good for sparc. but a cheapy x86 would spank that |
| 23:37.32 | [av]bani | i would expect solaris to be best for e4500... |
| 23:37.58 | *** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 23:37.58 | Netgeeks | given what I've seen, I would agree |
| 23:38.35 | *** part/#asterisk JuRaDoZ (n=Jurado@200.157.129.80) |
| 23:38.41 | [av]bani | an opteron would spank that however :)) |
| 23:38.47 | [av]bani | a sun opteron, even |
| 23:39.13 | Netgeeks | *nod* but a sun opteron wouldn't cost $950 |
| 23:39.20 | [av]bani | actually, they do |
| 23:39.30 | [av]bani | i think it was $750? |
| 23:40.00 | [av]bani | lemme dig it up |
| 23:40.19 | [av]bani | sun had some promo on 1u opterons |
| 23:40.25 | [av]bani | yea... 745 |
| 23:40.37 | Qwell[] | plus support? |
| 23:41.04 | [av]bani | http://www.sun.com/servers/entry/x2100/ |
| 23:41.04 | *** join/#asterisk bprice20 (n=brandon@cpe-24-194-238-98.nycap.res.rr.com) |
| 23:41.07 | justinu | i thought htey'd give you the system if you bought the 3year contract |
| 23:41.08 | sandos | what, is sun opterons different from AMD opterons? ;) |
| 23:41.18 | Qwell[] | sandos: Of course |
| 23:42.07 | [av]bani | 745 includes 3 year warranty |
| 23:42.13 | [av]bani | no OS |
| 23:42.16 | Qwell[] | the $745 is for a single code single cpu... |
| 23:42.21 | Qwell[] | single core* |
| 23:42.43 | Qwell[] | an opteron 146 |
| 23:43.20 | Qwell[] | a single opteron isn't gonna come anywhere near the e4500... |
| 23:43.36 | Netgeeks | if we can get around the network limit in the 4500 |
| 23:43.41 | Netgeeks | if we can't it will |
| 23:43.58 | Netgeeks | i should say the nic/interrupt issue |
| 23:44.01 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 23:44.11 | [av]bani | Qwell[]: 400mhz sparcs are not exactly screamers. the 146 is 2ghz and has a faster fsb |
| 23:44.33 | Netgeeks | someone else brought up a neat use for the 4500 if the network issue is resolvable, a high volume call recorder |
| 23:44.50 | [av]bani | Netgeeks: how many cpus you got in the 4500? |
| 23:45.03 | Netgeeks | the oen i have has 10, but it can be configured up to 14 |
| 23:45.05 | [av]bani | usparc2... |
| 23:45.40 | Netgeeks | I'll bring it up to 14 cpus when I am done testing the 10 cpu config |
| 23:46.08 | [av]bani | a 900mhz usparc3 is about equivalent to an athlon tbird 750... |
| 23:46.12 | [av]bani | integer |
| 23:47.02 | [av]bani | hmm i have some usparc2 numbers... 360mhz usparcII is about equialent to a p3/500 |
| 23:47.34 | [av]bani | and a sun 3/80 is the slowest sun i have ever used :)) |
| 23:48.41 | *** part/#asterisk doug (i=doug@zaxxon.telerama.com) |
| 23:48.44 | Qwell[] | speaking of the slowest sun ever... |
| 23:48.48 | Netgeeks | however, as we all know basic cpu performance measurements i.e. MIPS/FLOPS don't tell you much about how a cpu runs a particular app |
| 23:49.08 | [av]bani | Netgeeks: would be interesting to benchmark the 10 cpu e4500 vs a 1 cpu sun fire x2100 |
| 23:49.24 | [av]bani | Netgeeks: its a rough estimate of integer + cache performance |
| 23:49.47 | [av]bani | unless your application is mostly fp, its a good rule of thumb |
| 23:49.54 | Netgeeks | I don't have a x2100, but I will bench it against a dual opteron 275 |
| 23:49.58 | *** part/#asterisk steveaj (n=steve@62.55.147.53) |
| 23:50.00 | Netgeeks | which I do have |
| 23:50.08 | [av]bani | i woudl expect the opteron to smack it down |
| 23:50.24 | Netgeeks | I'll also compare it against a dual Xeon 3.2GHz |
| 23:50.44 | [av]bani | xeon should do well also... xeon has deep caches |
| 23:50.57 | Netgeeks | i'll even be using the same nic cards |
| 23:51.04 | [av]bani | what nics |
| 23:51.44 | Netgeeks | 1 E1k/Pro and 1 Broadcom 5704 |
| 23:52.48 | Netgeeks | hehehe, back to the slowest sun ever.... |
| 23:53.29 | Qwell[] | Netgeeks: I'm currently ssh'd through, and serial console'd to, my slow sun |
| 23:53.56 | Netgeeks | Linux ns1.xxxxxxxx.xxx 2.1.130 #1 Tue Mar 23 23:39:23 EST 1999 sparc unknown |
| 23:54.02 | Qwell[] | serial console really sucks when you've got two connections to it... |
| 23:54.18 | Netgeeks | BogoMips : 49.86 |
| 23:54.38 | Netgeeks | cpu : Texas Instruments, Inc. - MicroSparc |
| 23:54.49 | Qwell[] | and it locked up...fun |
| 23:55.34 | trixter | telepacket is having more issues |
| 23:55.37 | Netgeeks | <PROTECTED> |
| 23:55.54 | trixter | its like every week or two their mysql server goes down or something and no one can auth |
| 23:56.28 | Qwell[] | Netgeeks: umm... |
| 23:56.34 | Qwell[] | wtf? |
| 23:56.39 | Netgeeks | it's a dns server |
| 23:56.42 | Netgeeks | does nothing else |
| 23:56.54 | Netgeeks | it's one of those little shoebox suns |
| 23:57.11 | Netgeeks | I installed it in my rack in a colo in Laurel MD in 1999 and it's been going ever since |
| 23:57.31 | trixter | that would still be 6.5 years without any kernel upgrades, I bet there are remote vulns |
| 23:57.35 | Netgeeks | it's got a sister sitting next to it, but it's been rebooted a couple times, had a scsi drive go out on it |
| 23:57.53 | Netgeeks | I'm sure there are some holes |
| 23:58.12 | Netgeeks | but the only thing running is ssh and named which are both up to date |
| 23:58.21 | trixter | and the kernel |
| 23:58.23 | Qwell[] | solaris? |
| 23:58.34 | trixter | there have been remoately exploitable kernel bugs |
| 23:58.36 | trixter | remotely |
| 23:58.40 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:58.41 | Netgeeks | you should see ssh compile on that system.... it's a very long task |
| 23:58.43 | trixter | malformed packets and all |
| 23:58.53 | Netgeeks | linux 2.1 |
| 23:58.54 | [av]bani | trixter: "iptables" :)) |
| 23:59.00 | Qwell[] | Netgeeks: I've compiled bx on my ss5...yeah... |
| 23:59.20 | [av]bani | Qwell[]: try a sun 3/80 |
| 23:59.23 | Qwell[] | takes a good...30 minutes, to generate keys? |
| 23:59.25 | [av]bani | 20mhz 68030... |
| 23:59.28 | trixter | its linux from 6 years ago? yeah its got remote kernel vulns |
| 23:59.41 | tronix | heh I once compiled 'acm' (flight sim) on a 3/50. it took so long I did it as a cron job; it finished in just under 24 hours. |
| 23:59.44 | [av]bani | Qwell[]: a kernel compile takes... several days |
| 23:59.45 | tronix | that was fun :) |
| 23:59.49 | Netgeeks | if it gets hacked, oh well, it's lasted alot longer than I ever thought |
| 23:59.55 | trixter | [av]bani: I had a 3/60 - same cpu though iirc |
