| 00:00.03 | Godsey | and when agent hangs up, it crashes asterisk |
| 00:00.27 | file[laptop] | Godsey: I messaged you the fix... see if she works |
| 00:01.22 | firestrm | philz, just installed actos, it looks very usable.. i might even be able to understand it :) |
| 00:02.08 | *** join/#asterisk carlos-d-man (~carlos@201.128.216.12) |
| 00:02.10 | *** join/#asterisk wolfson (~hehe@65.174.122.198) |
| 00:04.28 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
| 00:04.31 | *** join/#asterisk firestrm (~vince@S010600047577bccd.gv.shawcable.net) |
| 00:04.59 | firestrm | phills, any idea what this might mean? *CLI> Jan 12 16:02:59 NOTICE[4033]: chan_iax2.c:5805 socket_read: Rejected connect attempt from 65.39.205.121, requested/capability 0x4/0x4 incompatible with our capability 0xf800. |
| 00:05.14 | Godsey | damn this place is too busy :) |
| 00:05.27 | Godsey | maybe someone will try to transfer so I can restart :) |
| 00:05.34 | devi0us | looks like a codec mismatch |
| 00:05.47 | file[laptop] | Godsey: hehe |
| 00:06.05 | wolfson | xoasisnetworks.com lists an asterisk box with 6 quad pri cards. I was under the impression anything over 2 cards was prone to not working |
| 00:06.10 | firestrm | devi0us, hmmm strange.. i followed the example exactly.. |
| 00:06.19 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 00:07.37 | Derkommissar | This is a long shot But here it goes,,,,,... I want to implement some queue's. and i want it to play a prompt when the agent awnsers the phone sayind this call is comming from bla, then ask press 1 to awnser.... is that dueable rigth now ? |
| 00:07.39 | Godsey | wolfson: maybe it's smp? |
| 00:07.44 | Godsey | more irqs? |
| 00:08.22 | Godsey | tho I don't see why anyone wouldn't use an AS5400HPX or APX8100 for that many ports :) |
| 00:08.41 | wolfson | or a lucent max tnt |
| 00:08.54 | firestrm | devi0us, if the incoming phone is using 729, could that be the problem? |
| 00:09.05 | Godsey | oh yes a max will work for only 144 ports |
| 00:09.21 | file[laptop] | mmm APX |
| 00:09.22 | wolfson | there is a ds3 version of the max tnt I thought |
| 00:09.48 | Godsey | ya but from what I've seen they don't do well when managing over 192 ports |
| 00:10.00 | Godsey | APX8000 chasis is better |
| 00:10.09 | file[laptop] | indeed |
| 00:10.12 | Godsey | I think you can use same cards, just much better backplane |
| 00:10.22 | file[laptop] | yup |
| 00:10.45 | file[laptop] | same DSP cards and stuff... rather nifty |
| 00:11.37 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 00:11.55 | wolfson | plus it handles all your compression, so no need to buy massive xeon machines |
| 00:12.11 | modulus_ | wb chuljin |
| 00:13.12 | Godsey | I was just looking at some voip hardware but can't remember their names |
| 00:13.19 | Godsey | they made RAS equipment too |
| 00:13.58 | modulus_ | row access strobe? |
| 00:14.00 | file[laptop] | my head hurts... and I need a drink |
| 00:16.07 | Godsey | geesh I feel dumb now :) |
| 00:17.12 | guugmember | what VoIP carrier do you recommend me if I am in LatinAmerica? |
| 00:17.36 | firestrm | ok now im getting this error... any ideas? Channel 'IAX2/iaxfwd@fwd-gw/2' sent into invalid extension 's' in context 'incoming', but no invalid handler |
| 00:17.38 | *** join/#asterisk vt2k1 (~kmescher@pcp0011112494pcs.elkrdg01.md.comcast.net) |
| 00:17.50 | Godsey | http://www.patton.com/ |
| 00:17.52 | Godsey | there it is :) |
| 00:17.55 | AgiNamu | What does make clean do? |
| 00:18.07 | file[laptop] | firestrm: read it |
| 00:18.09 | AgiNamu | does it just get rid of the 1000 .o and .so in the /usr/src/asterisk dirs? |
| 00:18.15 | Godsey | I wish they had a RAS+VoIP system |
| 00:18.18 | *** join/#asterisk xlyz (~xl@213-156-52-112.fastres.net) |
| 00:18.24 | wolfson | was the iax codec order stuff merged into cvs? |
| 00:18.25 | *** join/#asterisk frank_sbr (~frank_sbr@MTL-ppp-155094.qc.sympatico.ca) |
| 00:18.41 | firestrm | file[laptop], it doesnt make any sense... |
| 00:18.46 | file[laptop] | firestrm: yes it does, if you read it |
| 00:18.47 | frank_sbr | hello all |
| 00:19.13 | AgiNamu | hot shit! :D yey! Visual Studio 2005: Create new project from existing source code. /usr/src/asterisk... working.. done! now i get full CodeSense all over the project. |
| 00:19.15 | file[laptop] | the call was sent to the 's' extension in the 'incoming' context, but there was no s context so it failed |
| 00:19.30 | AgiNamu | I just need a compatible debugger and ill be in heaven |
| 00:20.05 | Godsey | I only have visual studio .net nfr :) |
| 00:20.17 | vt2k1 | Got a question about FXS modules: Can multiple telephones be connected to them? It it an issue of power requirements? |
| 00:20.26 | firestrm | file[laptop], arrrgh , i see where i went wrong... i put it into the default context.... sorry newby error.. |
| 00:20.28 | AgiNamu | Get VC++ 2005 Express |
| 00:20.29 | AgiNamu | it's free |
| 00:20.29 | guugmember | who has installed asterisk on ubuntu? |
| 00:20.43 | AgiNamu | as in free, as in how most of the world thinks of free when they think of products. |
| 00:20.46 | file[laptop] | firestrm: yes, so read errors - they're there for a reason |
| 00:20.51 | AgiNamu | not free as in I'm a crackhead. |
| 00:20.56 | file[laptop] | AgiNamu: but you are. |
| 00:21.03 | AgiNamu | that's besides the point. |
| 00:21.06 | file[laptop] | darn |
| 00:21.09 | AgiNamu | I didn't make VS2005 :) |
| 00:21.17 | file[laptop] | are you sure? |
| 00:21.21 | AgiNamu | Yea. well, i helped. |
| 00:21.26 | AgiNamu | and i talked to a lot of people who did. |
| 00:21.26 | file[laptop] | AHA! |
| 00:21.30 | file[laptop] | the truth comes out |
| 00:21.37 | AgiNamu | i was in some usability studies |
| 00:21.57 | AgiNamu | which was pretty cool |
| 00:22.07 | AgiNamu | and i reported a cool security holes |
| 00:22.11 | AgiNamu | and some other stuff |
| 00:22.14 | AgiNamu | nothing aamzing |
| 00:22.22 | AgiNamu | nothing like Anders |
| 00:22.27 | wolfson | aginamu: http://colo.aginet.com/lxr/http/source similar to codesense |
| 00:22.31 | AgiNamu | Heilsjberg or something :S |
| 00:22.46 | firestrm | file[laptop], getting closer, now i just have to figure out where the audio files are stored for playback |
| 00:22.54 | xlyz | hi! i'm trying to set up asterisk following the quick start guide on wiki. I'm stuck trying to add a sip softphone (linphone). not able to let him find asterisk. both are on the same pc. any hint? |
| 00:22.57 | tzafrir_home | guugmember: Do you have ubuntu? I'm interested to know how the asterisk packages from universe work there |
| 00:23.05 | AgiNamu | wolfson, you mean CodeSense as in Intellisense and everything VS does? |
| 00:23.13 | wolfson | yah |
| 00:23.21 | wolfson | not as powerful, but lxr will do crossreference |
| 00:23.27 | AgiNamu | thats nice |
| 00:23.29 | wolfson | but not edit time completion and stuff |
| 00:23.33 | AgiNamu | right |
| 00:23.38 | wolfson | which is really nice |
| 00:23.49 | file[laptop] | firestrm: /var/lib/asterisk/sounds |
| 00:23.53 | wolfson | not having to look up args for obscure functions is great |
| 00:23.54 | AgiNamu | I like VS's Code definition. when you type something, say I use ast_some_function, it'll lookup the header, and display it in a pane at the botom |
| 00:23.59 | tzafrir_home | heck, even vim has cross-references (using ctags) |
| 00:24.06 | file[laptop] | OMG BECKY |
| 00:24.40 | tzafrir_home | you should check out vim's simple ctrl-p and ctrl-n |
| 00:24.42 | firestrm | file[laptop], thanks, i figured it was somehting like that.. |
| 00:24.43 | Godsey | I have not really steped beyond php and perl |
| 00:24.46 | AgiNamu | is $350 for a quarter rack a good price for colo? includes all the bandwidth i need. |
| 00:24.52 | tzafrir_home | (yes, they do search in include files) |
| 00:25.03 | Godsey | AgiNamu: we generally pay $700 for full rack |
| 00:25.12 | Godsey | and $85/meg for bandwidth |
| 00:25.16 | bkw_ | ~seen TheLostOn |
| 00:25.18 | jbot | theloston <ZDEFAULT@i216-58-90-131.avalonworks.net> was last seen on IRC in channel #asterisk, 12d 4h 30m 54s ago, saying: 'ya but its a transport problem.... works ok on sip but not on iax'. |
| 00:25.42 | AgiNamu | Hmm |
| 00:25.50 | AgiNamu | how many U is a rack? |
| 00:25.59 | modulus_ | ours are 40u |
| 00:26.05 | modulus_ | err our cabinets are |
| 00:26.07 | AgiNamu | and can the 1U dell servers fit a digiumcard? |
| 00:26.12 | AgiNamu | (just for timing) |
| 00:26.29 | modulus_ | some 1u cases allow 1 pci card via reiser card |
| 00:26.33 | Godsey | I don't even know :) |
| 00:26.35 | modulus_ | depends on the design of the case |
| 00:26.36 | Godsey | 6' |
| 00:26.50 | Godsey | looks like 40U |
| 00:26.56 | Godsey | 1U=1.75" I think |
| 00:26.58 | modulus_ | is there an echo in here? |
| 00:27.16 | modulus_ | hehe |
| 00:27.36 | modulus_ | godsey what kind of bandwidth? |
| 00:27.40 | Godsey | eli.net |
| 00:28.05 | modulus_ | aginamu, aginamu, it depends on the bandwidth |
| 00:28.08 | Godsey | I can get it for $35/meg w/ wvfiber.com |
| 00:28.14 | modulus_ | aginamu, and i'm sure you'll have a limit |
| 00:28.26 | *** join/#asterisk Jammy (~jammy@24.244.183.75) |
| 00:28.54 | ta[i]nted | anyone know why my registration for gafachi would all the sudden be refused? |
| 00:29.02 | modulus_ | they hate you |
| 00:29.07 | Godsey | I am sysadmin @ an isp |
| 00:29.14 | Godsey | we have racks in 4 colo centers as backups |
| 00:30.11 | Godsey | one of the colo places charges us $500 for the rack and $1.35/gig transfered at commitment of 100gig/mo |
| 00:30.16 | Godsey | and $2.75/gig over |
| 00:30.20 | TrIpLeFFF | anybody use neteller or epassporte ? |
| 00:30.44 | firestrm | its so thrilling when the first light of knowlege breaks through the crack, and enlightenment begins... |
| 00:30.48 | modulus_ | godsey colo that charges total transfer usage? |
| 00:30.57 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 00:30.58 | Godsey | not total exactly but close |
| 00:31.06 | modulus_ | that's weird |
| 00:31.06 | Godsey | 5 minute windows, higher of in/out |
| 00:31.15 | modulus_ | usually it's 95th percentile |
| 00:31.28 | Godsey | ya only this 1 place bills that way |
| 00:31.38 | Godsey | 3 months left in that contract then we pull out :) |
| 00:31.41 | modulus_ | yeah |
| 00:31.45 | modulus_ | i bet it's someone's garage |
| 00:31.50 | AgiNamu | and i aways get confused: will a PCI-X slot work with normal PCI cards? |
| 00:31.58 | TrIpLeFFF | wvfiber.com / |
| 00:31.59 | TrIpLeFFF | ? |
| 00:32.03 | modulus_ | only hosting companies that over-subscribe charge per gig amount of transfer |
| 00:32.13 | TrIpLeFFF | doesn treposnd |
| 00:32.35 | AgiNamu | I want a rack mounted server that uses IDE drives |
| 00:32.39 | AgiNamu | im not gonna be nailing the hard disk |
| 00:32.42 | AgiNamu | and the price is silly |
| 00:32.48 | ta[i]nted | <PROTECTED> |
| 00:32.57 | ta[i]nted | oops |
| 00:33.29 | AgiNamu | $249 for a f*ing 36GB drive |
| 00:33.30 | AgiNamu | wtf |
| 00:33.39 | Godsey | na it's a nice datacenter |
| 00:33.39 | Godsey | it's just a way for them to be "different" |
| 00:33.39 | Godsey | eli is on 95th but we lock the port at 30meg |
| 00:33.43 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 00:33.44 | BoRiS | 10000 rpms? |
| 00:33.49 | ariel_ | hello everyone |
| 00:33.50 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 00:34.02 | AgiNamu | yea |
| 00:34.04 | AgiNamu | 10K |
| 00:34.06 | AgiNamu | they aren't selling less |
| 00:34.13 | AgiNamu | I dont know if Dell sells servers with SATA drives |
| 00:34.14 | AgiNamu | or IDe |
| 00:34.26 | ta[i]nted | AgiNamu: they do |
| 00:34.32 | AgiNamu | rack mounted? |
| 00:34.45 | ariel_ | anyone here configured sipura 2000 that is located via Sat Internet connection to an Asterisk server? |
| 00:35.52 | harryvv | arial, you mean its remote to your location |
| 00:35.52 | Jammy | anyone know if iaxtel is working or not? |
| 00:35.52 | TrIpLeFFF | wvfiber.com doenst reolve |
| 00:35.52 | TrIpLeFFF | Godsey: ? |
| 00:35.52 | ariel_ | harryvv, yes a remote user that is using a sat for internet connation. |
| 00:35.52 | AgiNamu | tainted, got a link? |
| 00:35.52 | AgiNamu | I want a dual Xeon, IDE drives |
| 00:35.55 | AgiNamu | cause paying $500 for disks is fucking retarded. |
| 00:35.58 | harryvv | Agi, go Opteron |
| 00:36.11 | AgiNamu | i dont like amd |
| 00:36.11 | bjohnson | Jammy: I was registering today but didn't try to connect |
| 00:36.45 | firestrm | AgiNamu, i can get dell precision 620's used for cheap |
| 00:36.51 | greg_work | ariel_: that sounds like you're begging for problems ;) |
| 00:36.54 | *** part/#asterisk guugmember (~nachoramo@200.49.163.49) |
| 00:36.59 | TrIpLeFFF | hey can anyone send me 5$ epassporte o neteller ? im missing 5 |
| 00:37.01 | harryvv | Cray computers used them to make a 6 teraflop cluster. 10k opterons. thay are cheaper and 20% faster then a Xeon. |
| 00:37.05 | TrIpLeFFF | just pm me |
| 00:37.05 | firestrm | dual xeon, scsi ultra |
| 00:37.06 | TrIpLeFFF | please |
| 00:37.14 | AgiNamu | yea, ok either way |
| 00:37.19 | AgiNamu | Dell doesnt sell them do they? |
| 00:37.21 | AgiNamu | I can't find em |
| 00:37.21 | bjohnson | any Canucks here need usb 1.1 4 port hubs? |
| 00:37.23 | firestrm | makes screaming asterisk boxes |
| 00:37.33 | AgiNamu | Does Itanium work well for Asterisk? |
| 00:37.40 | bjohnson | bjohnson: bunch on ebay you can get 4 for $20 CDN |
| 00:37.41 | ariel_ | greg_work, yes your right. But in any case I need to get them up and running. |
| 00:38.06 | harryvv | ohh hi tripleeff :) |
| 00:38.13 | TrIpLeFFF | hey hi harry |
| 00:38.16 | firestrm | AgiNamu, they are leasebacks, 2-3 years old.. |
| 00:38.19 | TrIpLeFFF | still in LEVIS ? |
| 00:38.21 | ariel_ | AgiNamu, most of dells dual servers are now either scsi or sata driver only. |
| 00:38.27 | greg_work | ariel_: well, its definately going to have lag issues. but .. theres no reason it won't "work" |
| 00:38.35 | TrIpLeFFF | so can anyone help with my 5$ problem ? |
| 00:38.48 | AgiNamu | ariel_ I havent seen any sata ones, for rack mounting |
| 00:38.51 | ariel_ | greg_work, my problem is not the lag I can make calls to there sipura without problem. |
| 00:39.16 | ariel_ | what I am having is problem of them making calls. the system shows them registered but when the dial it says not registered. |
| 00:39.51 | ariel_ | AgiNamu, I have but now dells I have setup some nice servers from Supermicro. |
| 00:40.06 | ariel_ | ./now/not |
| 00:40.12 | Godsey | ok testing fix for missing pointer :) |
| 00:40.50 | *** join/#asterisk PlainWhiteTrash (~matt@ps-ast-router.papersoft.com) |
| 00:41.05 | Godsey | www.wvfiber.com? |
| 00:41.08 | *** part/#asterisk xlyz (~xl@213-156-52-112.fastres.net) |
| 00:41.26 | Godsey | they purchased all of enrons ip hardware for $0.03/$1 |
| 00:41.31 | Godsey | er flip that :) |
| 00:41.46 | harryvv | whats a way to remove arts? its interfearing with my dial test on the cli |
| 00:41.58 | PlainWhiteTrash | Anyone have any ideas how I might debug a PRI issue I'm having with CVS head (older code works fine.) I'm getting a "Facility Not Subscribed" error back from the telco on current CVS head. |
| 00:42.12 | Damin | Bwahahahahahahaha.... |
| 00:42.14 | Damin | 12/21/2004 12:44 by jfenner: |
| 00:42.15 | Damin | Called Jim Doran. He just verbalized many obscenities and hung up. Returnin ticket to operations for further processing. |
| 00:42.38 | tzanger | Damin: hahaha |
| 00:42.55 | tzanger | PlainWhiteTrash: pri debug span 1 |
| 00:43.06 | PlainWhiteTrash | LOL, it's span 3 and I've already done that. |
| 00:43.08 | tzanger | and sort through the verbose output to try and see what it is specifically not liking |
| 00:43.18 | PlainWhiteTrash | I get the error message back... I'm not sure exactly what I'm sending that it doesn't want. |
| 00:43.19 | bkw_ | ~seen TheLostOn |
| 00:43.20 | jbot | theloston <ZDEFAULT@i216-58-90-131.avalonworks.net> was last seen on IRC in channel #asterisk, 12d 4h 48m 56s ago, saying: 'ya but its a transport problem.... works ok on sip but not on iax'. |
| 00:43.21 | Damin | tzanger: Did you get my message about the TE405P? |
| 00:43.34 | Damin | tzanger: Never mind then.. :) |
| 00:43.44 | Damin | Anyone else got one for sale? TE405P that is.. |
| 00:45.15 | file[laptop] | Godsey: did it work? |
| 00:45.15 | *** join/#asterisk ZX81 (matt@222-152-148-2.jetstream.xtra.co.nz) |
| 00:46.04 | bkw_ | do pigs fly? |
| 00:46.19 | file[laptop] | bkw_: only if you go near them |
| 00:46.31 | bkw_ | haha |
| 00:46.34 | harryvv | Godsey=bkw ? |
| 00:46.37 | bkw_ | no |
| 00:46.42 | bkw_ | i was being a ninny |
| 00:46.43 | harryvv | :) |
| 00:46.43 | ariel_ | is the search for user list on asterisk.org broken. when I use it to search the mailing list it just sends me to setup an email. |
| 00:46.51 | *** join/#asterisk Zaw (squid@synapse.subneural.net) |
| 00:47.02 | drumkilla | ariel_: just use google |
| 00:47.03 | harryvv | "Dont be such a niny" |
| 00:47.09 | harryvv | :) |
| 00:47.12 | PlainWhiteTrash | Anyone else? PRI "Facility not subscribed" ideas? |
| 00:47.15 | file[laptop] | bkw_ shut off your server because you're a ninny |
| 00:47.28 | ariel_ | drumkilla, yes I did. But it's was easyer to use the one at asterisk.org |
| 00:47.37 | file[laptop] | d rum killa! |
| 00:47.43 | drumkilla | yo file[laptop] !!! |
| 00:47.46 | file[laptop] | hey-o |
| 00:49.05 | drumkilla | how you doin' ? |
| 00:49.18 | file[laptop] | not too bad, u? |
| 00:49.26 | drumkilla | alright ... school started today |
| 00:49.39 | Godsey | file: testing now |
| 00:49.40 | file[laptop] | eek school |
| 00:49.46 | drumkilla | yep. |
| 00:49.58 | drumkilla | 17 hours of classes |
| 00:50.04 | AgiNamu | Add a second Xeon processor: 2.8Ghz: $399... 3.2GHz: $899 |
| 00:50.05 | file[laptop] | evil |
| 00:50.05 | AgiNamu | WTF? |
| 00:50.23 | Godsey | ya it worked! |
| 00:50.25 | Godsey | <PROTECTED> |
| 00:50.26 | drumkilla | file[laptop]: ooh! I have a good story from day 1 of class |
| 00:50.28 | Godsey | it says that now |
| 00:50.39 | file[laptop] | haha fun |
| 00:51.03 | AgiNamu | Oh, John Todd says that the sip channel uses a "SIP_CODEC" chanvar to override the preferred codec! |
| 00:51.10 | bkw_ | yes it does |
| 00:51.13 | bkw_ | or "should" |
| 00:51.17 | bkw_ | on a per call basis |
| 00:51.22 | AgiNamu | right! I was about to code that |
| 00:51.23 | bkw_ | IAX_CODEC is all we need now |
| 00:51.23 | AgiNamu | for IAX2 |
| 00:51.24 | drumkilla | So, in my operating systems class, my teacher says something like this while comparing C and java ... "You can take java code and run it through a C compiler. You may have to fix some syntax errors, but it will pretty much work in most cases." |
| 00:51.25 | bkw_ | who's up for it |
| 00:51.31 | AgiNamu | me |
| 00:51.34 | AgiNamu | I need it |
| 00:51.36 | AgiNamu | so ill do it |
| 00:51.37 | file[laptop] | drumkilla: O.O |
| 00:51.49 | bkw_ | drumkilla your teacher is on crack |
| 00:51.55 | drumkilla | I almost cried |
| 00:51.57 | bkw_ | haha |
| 00:52.03 | AgiNamu | well, it depends on what "some syntax errors" means :) |
| 00:52.14 | bkw_ | ALL OF THEM |
| 00:52.18 | bkw_ | every line just about |
| 00:52.20 | drumkilla | ha ... like ... rewriting the whole thing? |
| 00:52.22 | drumkilla | I mean, wtf |
| 00:52.24 | harryvv | Godsey, hows it going in oil refinery town :) |
| 00:52.31 | bkw_ | drumkilla did you tell him? |
| 00:52.33 | drumkilla | except for ... int i = 0; |
| 00:52.36 | bkw_ | I would have stood up and smacked him |
| 00:52.36 | Godsey | snow if you can believe it |
| 00:52.37 | bkw_ | and walked about |
| 00:52.39 | AgiNamu | well, if (someint == 0) {} -- that should work :D |
| 00:52.45 | bkw_ | "YOU MEAN I PAY FOR THIS CLASS and you're a dumb ass" |
| 00:52.49 | drumkilla | bkw_: I didn't want to get on his bad side on day 1 |
| 00:52.50 | harryvv | Godsey, we got some up here in vancouver |
| 00:52.56 | modulus_ | youuuuuuuucan't see meeeeeeeeeeee |
| 00:53.00 | AgiNamu | he has a bad side? |
| 00:53.03 | AgiNamu | How much worse can you get? |
| 00:53.04 | drumkilla | I already started a list of things |
| 00:53.08 | drumkilla | AgiNamu: haha |
| 00:53.11 | Godsey | harryvv: global freezing, it's the next epidemic |
| 00:53.26 | Damin | Called Jim Doran. He just verbalized many obscenities and hung up. Returnin ticket to operations for further processing. |
| 00:53.27 | drumkilla | I may go to the head of the department if he is as bad as he sounded today |
| 00:53.29 | drumkilla | he should be fired |
| 00:53.33 | AgiNamu | next he'll be saying "Visual Basic is just like Asterisk." |
| 00:53.38 | Godsey | now to figure out my IP300 phones having one way audio after a while |
| 00:53.51 | bkw_ | haha |
| 00:53.52 | bkw_ | no |
| 00:54.03 | bkw_ | drumkilla lord help us and he teachese? |
| 00:54.05 | bkw_ | er teaches |
| 00:54.11 | drumkilla | bkw_: yes. |
| 00:54.13 | bkw_ | oh lord |
| 00:54.14 | drumkilla | a college professor. |
| 00:54.15 | bkw_ | evne I know better |
| 00:54.24 | anthm | next thing ya know there will be javascript on asterisk too. |
| 00:54.26 | drumkilla | apparently he is new ... |
| 00:55.01 | AgiNamu | hey, I'd *like* a Javascript extensions file. |
| 00:55.04 | Godsey | drumkilla: you sure he didn't just make his ten year? |
| 00:55.06 | Godsey | :) |
| 00:55.14 | AgiNamu | I am seriously considering doing something like that, after I get ast_mono done. |
| 00:55.20 | drumkilla | Godsey: ha, he said this was his second semester ... |
| 00:55.24 | hermie | Godsey: I think you're thinking of 'tenure' :-) |
| 00:55.27 | harryvv | godsey, :) you have digium hardware installed or going to be a reseller? |
| 00:55.32 | Godsey | oh yes |
| 00:55.40 | hermie | drumilla: that's the kind of thing you just casually mention to another prof |
| 00:55.43 | harryvv | err reseller of bandwith |
| 00:55.48 | SplasPood | Godsey?! |
| 00:55.50 | file[laptop] | I like potatoes. |
| 00:55.58 | Godsey | linuxos to you too! |
| 00:56.01 | SplasPood | haha |
| 00:56.01 | Godsey | er hello |
| 00:56.05 | hermie | drumilla: "by the way, Prof. Dumbass lives up to his name..." |
| 00:56.06 | SplasPood | damn, we're all here... |
| 00:56.18 | SplasPood | Godsey: kidMagic comes around here too |
| 00:56.18 | Godsey | I'm a double fall out |
| 00:56.19 | bkw_ | haha |
| 00:56.24 | Godsey | I followed bkw from freebsdhelp |
| 00:56.26 | SplasPood | godsey: and coax used to |
| 00:56.30 | bkw_ | Godsey OMG YOU DID |
| 00:56.33 | bkw_ | haha |
| 00:56.45 | bkw_ | Godsey ArkyLady is here too |
| 00:56.51 | bkw_ | just not right now |
| 00:57.13 | Godsey | I haven't msg'd her for a very long time |
| 00:57.18 | Godsey | last I heard she sold voltage |
| 00:57.40 | bkw_ | yep |
| 00:57.43 | Godsey | you follow the exim mailing list? |
| 00:57.50 | bkw_ | I did at one time |
| 00:57.59 | bkw_ | but not anymore |
| 00:58.01 | bkw_ | that list is too active |
| 00:58.01 | Godsey | some guy asking about avoiding government easedropping |
| 00:58.06 | bkw_ | hahahahaha |
| 00:58.08 | bkw_ | don't talk |
| 00:58.10 | bkw_ | thats the only way |
| 00:58.10 | Godsey | I follow it and radiator |
| 00:58.18 | Godsey | it's funny |
| 00:58.25 | AgiNamu | well, not if you have an encrypting phone |
| 00:58.32 | AgiNamu | that'll avoid the easiest ways for them to listen |
| 00:58.35 | Godsey | he sets it up that he has friends in south america who's governments are corrupt and friends are in activist groups |
| 00:58.49 | Godsey | the funny bit is after looking, the guy hosts a company selling marijuana seeds |
| 00:58.50 | AgiNamu | Oh, he works with Fidel? |
| 00:58.54 | Godsey | I know what the REAL story is |
| 00:59.07 | AgiNamu | all governments are corrupt... so? |
| 00:59.20 | Godsey | I sent him a message: NEVER WRITE DOWN ANYTHING YOU DONT WANT THE WORLD TO KNOW |
| 00:59.25 | Godsey | and that's the truth |
| 00:59.32 | Godsey | let alone email :) |
| 00:59.37 | Jammy | heh |
| 00:59.43 | Godsey | only 1 person can keep a secret :) |
| 00:59.44 | AgiNamu | Well, if IAX2 phones start supporting encryption |
| 00:59.53 | AgiNamu | Then you could have end-2-end encrypted calls |
| 01:00.24 | Godsey | couldn't you make it? |
| 01:00.32 | AgiNamu | ? |
| 01:00.33 | Godsey | the same way firefly does video in txt packets? |
| 01:00.48 | AgiNamu | oh softphones yea |
| 01:01.10 | Godsey | I had to give up and stay out of grey area business ventures :) |
| 01:01.21 | Godsey | so I no longer really care about crypto :P |
| 01:01.44 | AgiNamu | .... even legit businesses should care |
| 01:01.55 | AgiNamu | Cause there's "grey and black" area guys who do |
| 01:02.07 | TrIpLeFFF | blah guuys ? |
| 01:02.25 | TrIpLeFFF | lol tought you said black guys.. grey and white is more michael jackson |
| 01:02.33 | Godsey | my business model is such that I don't store any sensitive information |
| 01:03.10 | modulus_ | jbot g-g-g-g? |
| 01:03.11 | jbot | G-UNIIT!! |
| 01:03.33 | Godsey | that reminds me |
| 01:03.33 | fafnir | Godsey: what about the sarbanesoxley ruling? |
| 01:03.34 | Godsey | JOHBOT ME |
| 01:03.35 | modulus_ | tear the roof on this motherfucka |
| 01:03.38 | modulus_ | light the roof on fire |
| 01:03.42 | modulus_ | jbot Godsey? |
| 01:03.47 | crash3m | burn that mofucka down |
| 01:04.01 | Godsey | fafnir: what about it? |
| 01:04.25 | Godsey | I will never be a publically traded company |
| 01:04.27 | Godsey | I'm a control freak |
| 01:04.48 | crash3m | you make more money as a privatly held anyway |
| 01:04.58 | Godsey | yes |
| 01:05.02 | crash3m | tough getting investors though |
| 01:05.20 | modulus_ | it's not tough |
| 01:05.28 | modulus_ | the money's just dirtier |
| 01:05.55 | crash3m | crack whores arent very good venture capitalists... |
| 01:05.57 | crash3m | :P |
| 01:06.05 | modulus_ | i keep getitng spam with the subject line: better than napster |
| 01:06.22 | KleinJonp | Hi people, what does "Found no matching peer or user for x.x.x.x" ? please, anyone ? |
| 01:06.30 | AgiNamu | pretty much anything's better than napster |
| 01:07.01 | Godsey | I'm self funded |
| 01:07.01 | Godsey | :) |
| 01:07.01 | Godsey | by not storing any sensitive info, I mean banking info |
| 01:07.01 | Godsey | all info is useful I supose |
| 01:07.01 | Godsey | I use ssh for remote server managment so it's not as if I don't benefit and use crypto |
| 01:07.05 | Godsey | I just don't obsess about it :) |
| 01:07.12 | Godsey | I don't have floppy pgp key swaping parties or anything :P |
| 01:07.20 | modulus_ | those people scare me |
| 01:07.30 | modulus_ | with their usb drives and their keys |
| 01:07.34 | Moc[dCAP] | . |
| 01:07.34 | AgiNamu | the PGP stuff is wierd |
| 01:07.46 | AgiNamu | in the sense of their trust |
| 01:07.56 | Godsey | how so? |
| 01:08.03 | TrIpLeFFF | got retina scanning device on your toilet ? |
| 01:08.05 | AgiNamu | how they do that circle jerk of trust or whatever its called |
| 01:08.15 | AgiNamu | might as well go and trust verisign. |
| 01:08.18 | TrIpLeFFF | and im not talking about that eye scanning |
| 01:08.18 | Godsey | I wrote a script for irc to use pgp keys :) |
| 01:08.48 | Godsey | only sort of irc |
| 01:08.58 | Godsey | mix of irc and perl cgi |
| 01:09.12 | AgiNamu | irc is a language? |
| 01:09.13 | *** join/#asterisk david (~dcoulson@muffin.davidcoulson.net) |
| 01:09.15 | david | hello |
| 01:09.23 | AgiNamu | Hi David, welcome to #asterisk! |
| 01:09.25 | Godsey | na the only thing irc was used for was to get a list of people |
| 01:09.42 | Godsey | <PROTECTED> |
| 01:09.48 | Godsey | would respond w/ your keyid |
| 01:10.04 | *** part/#asterisk charlesIII (~charlesII@65.171.196.23) |
| 01:10.06 | Godsey | then I'd type my msg and it'd crypt it to each of the keys and post to the cgi |
| 01:10.24 | Godsey | then privmsg $C :msgid 2998922 |
| 01:10.31 | Godsey | ugh |
| 01:10.42 | Godsey | msgid 2998922 keyid keyid keyid keyid |
| 01:10.53 | Godsey | for everyone I signed the message to |
| 01:11.03 | Godsey | and their script would grab it from the server |
| 01:11.10 | tzanger | Godsey: what's this for |
| 01:11.12 | Godsey | convoluded :) |
| 01:11.48 | Godsey | tzanger: mix of irc and web chat that used pgp |
| 01:11.54 | tzanger | ahh |
| 01:12.17 | Godsey | if someone in the group was busted, you revoke their key |
| 01:12.31 | Godsey | then when the fuzz sieze other machines none of the keys work |
| 01:12.40 | Godsey | all fun stuff in 95 :) |
| 01:12.41 | *** mode/#asterisk [-r] by bkw_ |
| 01:12.43 | *** join/#asterisk confbot (~root@66.250.68.194) |
| 01:12.44 | *** join/#asterisk confbot (~root@66.250.68.194) |
| 01:12.50 | *** join/#asterisk ArkyLady (nat32@93-95.hspg-ubr2-blk1.cablelynx.com) |
| 01:13.02 | *** mode/#asterisk [+r] by anthm |
| 01:14.30 | fafnir | Godsey: what do you do for work now? |
| 01:14.36 | Godsey | sysadmin of isp |
| 01:14.39 | AgiNamu | Wait, I don't think SIP_CODEC does what I want.... |
| 01:14.40 | Godsey | same as 95 |
| 01:14.50 | AgiNamu | It looks as if that only applies to when you receive a call or something? |
| 01:15.04 | AgiNamu | its in sip_answer... does sip_answer get called each time you dial? |
| 01:17.55 | firestrm | what do i use to convert sound file to/from gsm format? |
| 01:18.02 | tzanger | firestrm: sox |
| 01:18.16 | firestrm | thanks... |
| 01:18.46 | Godsey | soxmix |
| 01:18.55 | Godsey | right? |
| 01:18.58 | tzanger | soxmix is for mixing audio streams |
| 01:19.01 | Godsey | or is that for joining? |
| 01:19.04 | tzanger | sox file.gsm file.wav will work just fine |
| 01:23.04 | *** join/#asterisk josiah_ (~josiah@207.40.83.92) |
| 01:23.20 | josiah_ | hey all - good evening from indiana, usa |
| 01:24.31 | josiah_ | anybody here use Soyo SIP phones with Asterisk at all |
| 01:27.45 | Faithful | Why should I have so much lag on incoming ISDN calls VoIP out calls are fine??? |
| 01:28.11 | modulus_ | ISDN = it still does nothing |
| 01:28.50 | Faithful | modulus_: we _all_ know that, but I got it :p |
| 01:28.50 | *** join/#asterisk mikegrb (~michael@thegrebs.com) |
| 01:29.51 | bkw_ | asdf |
| 01:29.57 | mikegrb | ghjk |
| 01:30.06 | mikegrb | bkw_: how did you know my root password? |
| 01:30.10 | mikegrb | bkw_: did you hax0r me? |
| 01:30.19 | bkw_ | haha |
| 01:30.28 | mikegrb | Nugget: pingie |
| 01:31.00 | *** join/#asterisk JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 01:32.14 | TrIpLeFFF | anyone use paypal and epassporte or neteller / ? need to echange paypal for other |
| 01:32.30 | firestrm | is there an app that will play gsm files directly without conversion back to wav.. etc |
| 01:32.50 | Godsey | bkw: off topic but have you ever tried using Xen? |
| 01:32.58 | *** mode/#asterisk [-r] by anthm |
| 01:33.03 | *** join/#asterisk confbot (~root@66.250.68.194) |
| 01:33.03 | *** join/#asterisk juice (~juice@mo-65-40-189-111.dyn.sprint-hsd.net) |
| 01:33.06 | *** mode/#asterisk [+r] by anthm |
| 01:33.36 | *** join/#asterisk qwerp (~abc@219.93.57.62) |
| 01:33.50 | crash3m | why do you do +r anyway? |
| 01:33.59 | qwerp | when a call comes in, |
| 01:34.00 | bkw_ | spam bots |
| 01:34.03 | qwerp | it will ring a group. |
| 01:34.24 | crash3m | spam bots are handled pretty well on the server |
| 01:34.29 | crash3m | s/the server/this network |
| 01:34.51 | bkw_ | ya but since #asterisk was first on the list.. we would get 100+ join and parts per hour |
| 01:34.53 | qwerp | i am wondering, when someone is answering a call.. |
| 01:34.55 | bkw_ | quite anoying |
| 01:34.56 | Godsey | I hated +r till I figured out how to get irssi to auto register :) |
| 01:34.57 | qwerp | then a calls come in, |
| 01:35.17 | qwerp | anybody to hear my problem out? |
| 01:35.38 | Godsey | qwerp: I haven't seen your problem yet |
| 01:35.40 | bkw_ | not when you're a rude ass and dont say Hi |
| 01:35.50 | qwerp | errmm... |
| 01:35.57 | Godsey | I lurked here for 5 months before I started bugging bkw about asterisk bugs |
| 01:35.57 | qwerp | coz i did come a couple of times.. |
| 01:36.03 | Godsey | not that he really helps |
| 01:36.11 | Godsey | but I think he has my photo in his wallet |
| 01:36.13 | qwerp | n someone told me just ask, dun ask to ask |
| 01:36.26 | Godsey | qwerp: continue w/ your problem |
| 01:36.31 | bkw_ | +shaddap |
| 01:36.35 | confbot | ok bkw_ command 'shaddap' complete. |
| 01:36.42 | qwerp | calls comes in ---> a group of numbers. |
| 01:36.54 | Godsey | qwerp: that part is easy |
| 01:37.02 | qwerp | one ppl is answering a call, so his ext will not ring.. |
| 01:37.22 | qwerp | assume he is the only person left in the group.. |
| 01:37.40 | Godsey | you want it to ring many extensions at once |
| 01:37.43 | Godsey | not a queue group right? |
| 01:37.43 | qwerp | another call comes in, while he is on the phone, the other extension will ring.. |
| 01:37.51 | qwerp | not a queue.. |
| 01:38.12 | Godsey | ok :) |
| 01:38.23 | Godsey | when you say only one left in group I was thinking agents |
| 01:38.30 | qwerp | so the problem is, so if the person wans to ansewr the call, logically, if he hang up the first call, then second incoming call should ring on his extension as well |
| 01:39.12 | qwerp | its something like Dial(Zap/1-1,Zap/2-1,Zap/3-1) |
| 01:41.02 | Godsey | only if his line is available when the call starts |
| 01:41.02 | Godsey | it won't start ringing in the middle |
| 01:41.02 | Godsey | there is a way around that |
| 01:41.02 | Godsey | you can setup a loop w/ a counter |
| 01:41.02 | Godsey | and sleep |
| 01:41.03 | Godsey | poor man queue group |
| 01:41.06 | Godsey | if all extensions are busy/unavailable sleep 2 and loop again |
| 01:41.08 | Godsey | and have a counter to break the loop if they go over 10 times or whatever |
| 01:41.10 | Godsey | then it will work as you want |
| 01:41.12 | Godsey | right |
| 01:41.15 | Godsey | put a ring time limit on it |
| 01:41.18 | Godsey | of like 5 seconds |
| 01:41.20 | Godsey | or 10? |
| 01:42.20 | Brixius | hello |
| 01:42.42 | qwerp | yes |
| 01:42.52 | qwerp | a loop at dialplan or? |
| 01:43.11 | Brixius | I'm happy, dundi is working. :) :) :) |
| 01:43.52 | bkw_ | its & |
| 01:44.08 | bkw_ | Dial(Zap/1-1&Zap/2-1&Zap/3-1) |
| 01:44.52 | Slainte | I have an * box at work and one at home. I want to do two things. 1. Ring a work extension at home. 2. Pass all my Long Distance through work. Should I register as a SIP peer or IAX? |
| 01:45.36 | anthm | chan_rtfm |
| 01:45.41 | bkw_ | haha |
| 01:45.45 | bkw_ | so so true |
| 01:45.52 | Brixius | that sound's a little like theft of service |
| 01:46.06 | bkw_ | not if he owns the company |
| 01:46.11 | Brixius | this is true |
| 01:46.16 | *** join/#asterisk _Vile (~vile@90.b160.bendtel.net) |
| 01:46.27 | _Vile | asdf |
| 01:46.30 | Slainte | I am just wondering what the best way to do it is |
| 01:46.57 | _Vile | bkw what does this mean? Signalling requested on channel 72 is PRI Signalling but line is in Unknown signalling 896 signalling |
| 01:47.24 | qwerp | Godsey: loop in dial plan? |
| 01:47.34 | Godsey | that's how i did it for a while |
| 01:47.39 | Godsey | but just switched to agents and queues |
| 01:47.41 | Godsey | works better :) |
| 01:48.07 | qwerp | humm.. |
| 01:48.22 | qwerp | but queues when u answer, u need to press # key |
| 01:48.25 | *** join/#asterisk Huezo (~huezo@201.128.214.198) |
| 01:48.36 | tzanger | qwerp: you do? |
| 01:48.39 | tzanger | (I haven't used queues) |
| 01:48.48 | tzanger | I figured the queue rang your phone and it saw you answered |
| 01:49.42 | modulus_ | hm |
| 01:49.50 | modulus_ | sip does need call answer notification |
| 01:49.56 | qwerp | tzanger: i did tried, yes i do |
| 01:49.57 | modulus_ | like the zap/3c |
| 01:49.58 | modulus_ | thingy |
| 01:50.08 | tzanger | qwerp: interesting |
| 01:50.43 | qwerp | btw, how many ways a agent cal login? |
| 01:50.50 | qwerp | 1 is callbacklogin.. |
| 01:51.06 | Godsey | I use agentcallbacklogin but it has some issues |
| 01:51.08 | qwerp | i wonder can i just register a few extension as a agents? |
| 01:51.23 | Godsey | I do that also |
| 01:51.31 | Godsey | I switch between agents and dynamic queue members |
| 01:51.48 | samueltc | hey, what is the 'Predictive Dialer' feature in asterisk? |
| 01:52.06 | qwerp | Godsey: do u need to press # key if u use agentcallbacklogin? |
| 01:53.34 | bkw_ | moc |
| 01:53.38 | bkw_ | wake up |
| 01:54.30 | Godsey | you don't have to |
| 01:54.34 | Godsey | but you can set it up to do that |
| 01:55.50 | file[laptop] | I'm hungryyyyyyyyyyy |
| 01:56.01 | fafnir | me too |
| 01:56.03 | wolfson | does fxotune not get built with a normal make in zaptel? |
| 01:56.04 | fafnir | take me out to dinner |
| 01:56.10 | fafnir | i promise you sex afterwards |
| 01:56.22 | *** join/#asterisk lilneon (~tj_r3@cuscon4937.tstt.net.tt) |
| 01:56.23 | file[laptop] | ...nah |
| 01:56.27 | lilneon | hi everyone |
| 01:56.29 | tzanger | fafnir: damn I can guarantee you're not female... hahaha |
| 01:56.56 | fafnir | tzafrir; noone said it would be straight sex |
| 01:57.06 | Nivex | tzanger: and fafnir promised sex but did not say with whom |
| 01:57.10 | Nivex | tzanger: or what |
| 01:57.21 | sung | bkw. |
| 01:57.22 | tzanger | Nivex: no but he promised sex after dinner |
| 01:57.24 | tzanger | women don't promise that |
| 01:58.13 | _Vile | bkw -> Signalling requested on channel 72 is PRI Signalling but line is in Unknown signalling 896 signalling -- Source code says "Unable to find locate channel/trunk group" |
| 01:58.32 | fafnir | tzafrir: really hungry women do |
| 01:58.40 | tzanger | fafnir: hahaha |
| 02:01.05 | *** join/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx) |
| 02:03.58 | *** join/#asterisk roche (~roche@216.194.173.2) |
| 02:06.02 | lilneon | has anyone gotten a softphone on another lan to connect to an asterisk box on a different lan thru the internet? |
| 02:06.23 | lilneon | i have tried every phone from iaxcomm to sjphone and nothing.. please help |
| 02:07.05 | *** join/#asterisk apsaras (~antonis@serial03-01.que01.acl.hol.gr) |
| 02:07.19 | lilneon | hello? |
| 02:07.22 | apsaras | hi |
| 02:07.28 | apsaras | first time in here |
| 02:07.38 | wolfson | lilneon: double nat? and iax or sip? |
| 02:07.49 | tzanger | lilneon: yup I made a call on firefly on my laptop through dialup to my asterisk box on my DSL line, back out to nufone |
| 02:07.52 | tzanger | no nat through |
| 02:08.13 | tzanger | honestly though with nat and iax it isn't an issue at all |
| 02:09.02 | apsaras | hi all. I have a problem configuring a TE405P can anyone help? |
| 02:09.19 | tzanger | apsaras: state the problem with as much detail as you can manage |
| 02:09.41 | lilneon | tzanger:could u guide me thru |
| 02:09.49 | tzanger | but do not paste configs or anything, use pastebin.ca for that |
| 02:09.54 | tzanger | lilneon: there wasn't anything to it |
| 02:10.04 | tzanger | firefly config I had my * box set up as the network |
| 02:10.10 | tzanger | that was it |
| 02:10.16 | tzanger | ilbc codec since i was on dialup |
| 02:10.30 | tzanger | it was incredible call quality so long as I did not try to do anything else besides the VOIP call on the dialup :-) |
| 02:10.56 | lilneon | tzanger: well i have firefly |
| 02:11.17 | *** join/#asterisk trazillion (~traz@65.96.21.145) |
| 02:11.25 | apsaras | when I load the module for zaptel driver I get an error "ZT_CHANCONFIG failed on channel 97". I configured the card as a E1 but the driver still see it as a T1 |
| 02:11.30 | lilneon | i see it register on the asterisk box but ??.. |
| 02:11.49 | trazillion | What is the typical problem that causes the error message: span=1,1,0,esf,b8zs |
| 02:11.49 | trazillion | bchan=1-23 |
| 02:11.49 | trazillion | dchan=24 |
| 02:11.49 | trazillion | ? |
| 02:11.54 | trazillion | err |
| 02:12.02 | trazillion | Unable to create channel of type 'Zap' |
| 02:12.17 | tzanger | lilneon: ok |
| 02:12.25 | *** mode/#asterisk [-r] by bkw_ |
| 02:12.28 | apsaras | my conf file is like that |
| 02:12.34 | tzanger | set up the network such that you are using iax2 and you have a valid user on the * box |
| 02:12.41 | tzanger | really it is *no* different than them being on the same network |
| 02:12.41 | modulus_ | bleh |
| 02:13.02 | lilneon | tzanger: ok |
| 02:13.09 | apsaras | span=1,2,0,ccs,hdb3,crc4 |
| 02:13.09 | apsaras | bchan=1-15 |
| 02:13.09 | apsaras | dchan=16 |
| 02:13.09 | apsaras | bchan=17-31 |
| 02:13.21 | tzanger | apsaras: go to www.pastebin.ca and post your zaptel.conf |
| 02:13.34 | tzanger | apsaras: first thigns first -- did you change the 4 jumpers on the TE405 to "E1" from "T1" |
| 02:13.37 | tzanger | ? |
| 02:14.00 | *** join/#asterisk riksta (~rick@81-178-234-208.dsl.pipex.com) |
| 02:14.01 | modulus_ | oh man |
| 02:14.05 | apsaras | No |
| 02:14.07 | modulus_ | wouldn't that be something? |
| 02:14.12 | apsaras | I didn't know that |
| 02:14.41 | lilneon | tzanger: so once i get it working on my local lan it shoulds work fine over the internet? |
| 02:14.42 | apsaras | where can I find the jumpers' correct conf |
| 02:15.22 | tzanger | lilneon: so long as the two sides can see each other yes |
| 02:15.31 | tzanger | apsaras: it's right on the card |
| 02:15.39 | *** join/#asterisk Matt-E- (~matto@66-224-125-137.atgi.net) |
| 02:15.40 | tzanger | there are four jumpers right above the red "dial" selector switch |
| 02:15.46 | tzanger | it says "T1 = closed, E1 = open" |
| 02:15.50 | tzanger | all four need to be open |
| 02:15.57 | apsaras | ok |
| 02:16.02 | apsaras | I ll check |
| 02:16.05 | apsaras | thnx |
| 02:16.08 | lilneon | tzanger: i believe that's the problem right there.. |
| 02:16.15 | lilneon | my dsl gets a dynamic IP |
| 02:16.22 | lilneon | and so does my computer @ work |
| 02:16.55 | modulus_ | that sure does suck |
| 02:16.58 | trazillion | Nobody has seen this message? (Unable to create channel of type 'Zap') |
| 02:18.08 | Matt-E- | does any one know of cheaper equivilants to the TDM400P? |
| 02:18.19 | tzanger | lilneon: well yeah |
| 02:18.27 | PatrickDK | what is it about people and cheap |
| 02:18.30 | PatrickDK | cheap don't work |
| 02:18.41 | tzanger | asterisk isn't black voodoo, the packets need to be able to get from Point A to Point B just as if it were a game server or pr0n site |
| 02:18.48 | PatrickDK | cheap and equiv should never be in the same page |
| 02:18.49 | tzanger | PatrickDK: amen! |
| 02:18.57 | tzanger | Matt-E-: there is none. |
| 02:19.08 | Matt-E- | good to konw |
| 02:19.09 | Matt-E- | know |
| 02:20.21 | ta[i]nted | does asterisk pick up callerid from international calls? |
| 02:20.54 | tzanger | ta[i]nted: unplug asterisk and plug in a callerid phone -- does the phone pick up the international callerid? |
| 02:21.16 | harryvv | lil, the one at work is it connected directly to the dsl? |
| 02:21.19 | harryvv | err |
| 02:21.22 | tzanger | ta[i]nted: as I said to Matt-E- -- asteirsk is not black voodoo, it gets the information from the same places your telephone gets it |
| 02:21.25 | harryvv | yea |
| 02:21.25 | Nivex | Initial reviews of the SPA-841 are in on the voxilla forums. Looks like I stick with adapters for the moment. |
| 02:21.49 | *** join/#asterisk sudoer2 (~toy@h000d56c772cb.ne.client2.attbi.com) |
| 02:21.55 | harryvv | is that the handheld wifi unit? |
| 02:22.34 | Nivex | harryvv: no, that is Sipura's new $90 hardphone |
| 02:23.08 | devi0us | the 2100 is a pain in the ass |
| 02:24.44 | *** join/#asterisk DevilFish (DevilFish@216.163.40.2) |
| 02:25.03 | DevilFish | anyone home? |
| 02:27.51 | DevilFish | Anyone here cause I got what I think is a hell of a question |
| 02:28.30 | modulus_ | i'm not falling for that one |
| 02:28.41 | wolfson | i would just ask the question, then if someone cares to answer they will |
| 02:28.41 | DevilFish | ok then :) |
| 02:28.50 | DevilFish | I see |
| 02:28.57 | DevilFish | wasn't sure if this was that kind of chan |
| 02:30.06 | *** join/#asterisk rsb__ (~rsb@adsl-69-107-37-126.dsl.pltn13.pacbell.net) |
| 02:30.28 | rsb__ | Hi guys. |
| 02:31.24 | rsb__ | Just got a te410P installed and I'm surprised *not* to see any interrupts being generated. |
| 02:31.29 | *** join/#asterisk enzo123 (~enzo123@rdu26-62-161.nc.rr.com) |
| 02:31.56 | rsb__ | Does anyone know if this card generates interrupts without a span plugged in? |
| 02:32.00 | *** join/#asterisk MEMEyou (~memeyou@hnllhi1-ar7-4-7-215-027.hnllhi1.dsl-verizon.net) |
| 02:32.02 | enzo123 | just called garmin gps tech supprt .. did a enum lookup before the call and it found their number and made the call via IAX |
| 02:32.06 | rsb__ | I assumed that all the digium cards do |
| 02:32.22 | wolfson | is the new hardware echo cancel stuff in wctdm not ready? |
| 02:32.33 | enzo123 | guess garmin is using * |
| 02:32.39 | trazillion | Has anyone seen this message: Unable to create channel of type 'Zap' ? |
| 02:32.44 | MEMEyou | where do i go if i want to learn all about VoIP/PBX, Asterisk and Phones? |
| 02:33.01 | MEMEyou | we currently use NBX |
| 02:33.05 | PatrickDK | voip-info.org :) |
| 02:33.15 | MEMEyou | ++) |
| 02:33.33 | DevilFish | I have Asterisk 1.0.3 on RHFC2, SIP channel registered to a metaswitch, Polycom IP600 phones... any calls to the PSTN result in a disconnections after 3-20 mins. Debug is showin "Jan 12 16:13:08 DEBUG[2252]: Failed to grab lock, trying again..." and then somthing like "Jan 12 16:13:08 DEBUG[2252]: Bridge stops because we're zombie or need a soft hangup: c0=SIP/302-cb02, c1=SIP/sip_proxy-out-618f, flags: No,No,No,Yes" anyone here ever see this zombie stu |
| 02:33.55 | enzo123 | wow.. holly pastebin |
| 02:34.10 | modulus_ | holy word wrap devilfishman! |
| 02:34.15 | DevilFish | sorry |
| 02:34.28 | DevilFish | trying to cram in the info I guess |
| 02:34.40 | Cresl1n | wolfson: What do you mean? |
| 02:34.42 | DevilFish | anyways I've been at this one for like 3 days now |
| 02:35.24 | enzo123 | anyone here front ending * with SER ? |
| 02:35.35 | drumkilla | Cresl1n! |
| 02:35.42 | Cresl1n | drumkilla!!! |
| 02:35.42 | Cresl1n | :-) |
| 02:36.17 | _Vile | Cresl1n, drumkilla!!!!!!!!!!!!!!!!!! |
| 02:36.31 | Cresl1n | _Vile!!!! |
| 02:36.31 | Cresl1n | :-) |
| 02:37.07 | Slainte | Devil, what SIP ver you using on the 600? |
| 02:37.45 | DevilFish | lemme check |
| 02:39.51 | enzo123 | anyone here using iax.cc or sixtel ? |
| 02:40.55 | nestAr | i do |
| 02:40.58 | *** join/#asterisk Yellow_Fuzzy (Yellow_Fuz@c211-31-41-9.wavrl1.nsw.optusnet.com.au) |
| 02:41.01 | Yellow_Fuzzy | hi |
| 02:41.20 | Faithful | Trying to patch asterisk for broadvoice and it tells me that the patch only contains garbage... ? |
| 02:41.47 | *** join/#asterisk dallaschumly (dallaschum@c-24-0-130-236.client.comcast.net) |
| 02:42.25 | *** join/#asterisk MrEntropy (~entropy@ppp44-247.lns1.adl1.internode.on.net) |
| 02:42.26 | MrEntropy | yo |
| 02:42.32 | Yellow_Fuzzy | im looking at implementing asterisk within an organisation with about 40 - 50 users positioned all around the world with each being provided an extension & voicemail plus conferencing services, for something this size is it worth using mysql db for configs over the flatfile configs? |
| 02:42.56 | DevilFish | Slainte: is there anyway to tell the ver from the web interface or otherwise cause the phone is not here me. (dang it) |
| 02:43.16 | ManxPower | Yellow_Fuzzy: For your R&D systems flat file is fine. For production, you should consider a database. |
| 02:43.52 | DevilFish | wait maybe boot log can tell me |
| 02:43.52 | file[laptop] | okay everyone, commence self destruct sequence |
| 02:43.57 | ManxPower | BTW, if you don't have a R&D system, don't expect to have a working production system. There's just too much to learn. |
| 02:44.06 | firestrm | hi |
| 02:44.45 | firestrm | what would cause the first few seconds of an incoming call over iax to become garbled? |
| 02:44.56 | dallaschumly | I have a very broad and sweeping question |
| 02:45.07 | enzo123 | we have a broom.. |
| 02:45.14 | Slainte | Devil, the boot log will tell you. Are you botting off FTP/TFTP and if so do you have a new sip.ld in your ftp dir? |
| 02:45.14 | Yellow_Fuzzy | ManxPower: yes i do have an R&D system and a few production servers atm |
| 02:45.22 | ManxPower | firestrm: A slow system or a crappy router. |
| 02:45.28 | dallaschumly | LOL, maybe better get a pan for this one |
| 02:45.37 | dallaschumly | Kind of basic, but... |
| 02:45.43 | Yellow_Fuzzy | ManxPower: is there any particular type of DB that works best with asterisk? |
| 02:45.54 | dallaschumly | How do I connect Asterisk to the PSTN? |
| 02:45.56 | mtqh | Who hosts the hell room? |
| 02:46.16 | ManxPower | Yellow_Fuzzy: I'm partial to MySQL. It's easy, it's fairly lightweight, and for that number of users should be just fine. |
| 02:46.31 | dallaschumly | One way, order 20 phone lines, and buy 20 digium cards. Not reality |
| 02:46.34 | ManxPower | dallaschumly: A Digium card or a VoIP provider. |
| 02:46.47 | Yellow_Fuzzy | also ive been reading through the voip wiki but it has really confused me as to which i actually need to follow get it working in realtime |
| 02:46.56 | enzo123 | or a t1 card and a chan bank |
| 02:47.01 | DevilFish | Slainte, I'm booting via FTP but have no sip.ld but boot log is showing somthing like "Version=2.5.0.0006" in it |
| 02:47.10 | ManxPower | dallaschumly: Uh, the T-1/E-1 cards support up to 30x4 lines per card. |
| 02:47.12 | firestrm | ManxPower, system is a dual xeon 1ghz, router.. maybe... netgear |
| 02:47.19 | dallaschumly | Shouldn't I be able to connect to PSTN through a ISDN or T1? |
| 02:47.20 | Slainte | Devil, that is the BootRom not the SIP version |
| 02:47.20 | enzo123 | devilfish a have a 7960 running sip. |
| 02:47.35 | ManxPower | firestrm: Is there a bathroom nearby? Accidently drop the nethear in the toilet, then buy a new router. |
| 02:47.44 | DevilFish | Thats what I thought |
| 02:47.45 | firestrm | lol |
| 02:48.02 | ManxPower | firestrm: ACTUALLY, I don't personally know about netgear and VoIP. |
| 02:48.08 | firestrm | that bad huh |
| 02:48.13 | dallaschumly | Ah, so if I just buy a T-1 or E-1 card, it will connect? No fee to the telco? |
| 02:48.18 | Slainte | Devil, it might be worth getting this file http://www.freedomphones.net/polycom/files/SoundPointIP_SIP_1_3_1.zip |
| 02:48.26 | Moc_ | polycom rock ;) |
| 02:48.27 | ManxPower | I just know every netgear box I've tried failed randomly and miserably. |
| 02:48.48 | ManxPower | dallaschumly: If you want free PSTN access then just take lots of drugs. |
| 02:48.49 | firestrm | maybe i will try direct connection ie. no routr |
| 02:48.57 | dallaschumly | Is the T-1 fee the right to connect to a telco? |
| 02:49.15 | *** join/#asterisk BigCanOfTuna (~chatzilla@S0106000c41b313f3.cg.shawcable.net) |
| 02:49.22 | ManxPower | dallaschumly: Nothing is ever free with the telco. Expect $600/month - $2,400/month for 24 lines. |
| 02:49.23 | Slainte | Devil, Unzip it, and make sure your 000MAC00.cfg has a pointer to the new sip.ld |
| 02:49.38 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 02:49.50 | DevilFish | yeah my phones are using the sip.ld that they came with |
| 02:49.55 | enzo123 | more like telco = bend over |
| 02:50.00 | DevilFish | could that be hosing me up? |
| 02:50.00 | ManxPower | The ACTUAL T-1 fee is included in the monthly service. |
| 02:50.17 | DevilFish | I can see the boot log "|Loaded application sip.ld succesfully," but no sip.ld version |
| 02:50.17 | enzo123 | or PRI 8> |
| 02:50.33 | dallaschumly | The $2,400/month is for the T-1 or ISDN? Does this give me the right to connect to the local telco? |
| 02:50.38 | Slainte | Is there a sip.ld in your ftp dir? |
| 02:50.43 | ManxPower | enzo123: No sane person would get a T-1 without PRI. |
| 02:50.53 | DevilFish | no sip.ld in my ftp dir |
| 02:50.55 | Slainte | drop the new one in, and have it load it up |
| 02:50.58 | tzanger | ManxPower: I use CAS T1s all the time with my channel banks :-) |
| 02:51.06 | DevilFish | I thought you only had to have one if you wanted to update the phone |
| 02:51.16 | ManxPower | dallaschumly: For the whole package. PRI served over T-1, and the numbers I said would include 24 voice channels |
| 02:51.22 | enzo123 | manxpower .. never know.. |
| 02:51.33 | Slainte | Devil, the FTP/TFTP does three things. |
| 02:51.34 | ManxPower | tzanger: To the telco? |
| 02:51.41 | tzanger | ManxPower: no :-) |
| 02:51.50 | tzanger | T100P <--T1--> Channel Bank |
| 02:51.52 | Slainte | first, it has the config files for your phone based on the login FTP credentials i.e sip details |
| 02:52.00 | Slainte | second, it provides a place to dump the logs |
| 02:52.08 | ManxPower | enzo123: No sane person would get a T-1 without PRI for connection to the telco. |
| 02:52.09 | DevilFish | and updates :) |
| 02:52.10 | Slainte | third, it provide a place for updating both firm and sip ware |
| 02:52.23 | Slainte | wait for it chico :) |
| 02:52.28 | DevilFish | gotcha, could that be my wierd call problem? |
| 02:52.47 | Slainte | re booting a phone with the new sip.ld in the ftp dir is not going to break it |
| 02:52.52 | ManxPower | DevilFish: We had pretty poor luck with firmware previous to 1.3.1 |
| 02:52.53 | Slainte | def worth the first try |
| 02:52.55 | DevilFish | I hear ya there |
| 02:53.07 | enzo123 | manxpower ive seen mci sell some crazy shit.. 8> |
| 02:53.12 | ManxPower | I leave you all with a macro. |
| 02:53.14 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 02:53.27 | ManxPower | enzo123: I was referring to customers. We all know the telcos are not sane. |
| 02:53.29 | DevilFish | yeah, I'll snag the updates and then test again tomorrow morning |
| 02:53.39 | Slainte | Devil, |
| 02:53.43 | DevilFish | yes? |
| 02:53.52 | Slainte | I have a few pots lines at work I will try to replicate your problem |
| 02:54.07 | Slainte | drop in sometime tomorrow and put me on notify. I will let you know if I can replicate |
| 02:54.38 | DevilFish | thx, I'm gonna be all over it, I got promises to keep by tuesday :) |
| 02:54.41 | Slainte | I need to know if it is an issue as I will soon have a ton of the P600's in the field |
| 02:54.57 | DevilFish | p600 and p500s are great |
| 02:55.31 | dallaschumly | Hmm, I'm kind of slow in math, BUT.... |
| 02:55.34 | dallaschumly | $2,400/month, and I can terminate for $0.001/minute? |
| 02:55.35 | DevilFish | I love em, except this is happening, I'm worried that it migh have somthing to do too with the metaswitch and that is really worrying me |
| 02:56.04 | DevilFish | gonna try and squeeze some logs out of it tomorrow as well |
| 02:56.12 | dallaschumly | That's a lot of minutes! |
| 02:56.16 | Slainte | Make sure you set your loggin as high as possible |
| 02:56.16 | *** join/#asterisk Funbags (~Funbags@ool-182e52ab.dyn.optonline.net) |
| 02:57.04 | DevilFish | on the meta? or asterisk ? BTW what is max verbose for asterisk? |
| 02:57.22 | *** join/#asterisk fleeeb (~basin@pc-24-151-28-122.newm2.ct.charter.com) |
| 02:57.25 | Yellow_Fuzzy | ManxPower: so if i want to set it up with mysql with real time changes for sip,voicemail & extensions how would i go about doing it? |
| 02:57.28 | *** join/#asterisk ZX81 (matt@222-152-148-2.jetstream.xtra.co.nz) |
| 02:57.41 | ZX81 | is this for real: Purchase Digium Hardware this month and help the Tsunami relief effort! |
| 02:57.49 | ZX81 | ? |
| 02:57.54 | ZX81 | before I post it as news |
| 02:57.56 | *** join/#asterisk Funbags (~Funbags@ool-182e52ab.dyn.optonline.net) |
| 02:58.20 | *** join/#asterisk rtcg (~rtcg@bdsl.66.15.181.96.gte.net) |
| 02:58.27 | Slainte | Devil, never needed anything more then 5 verbose |
| 02:58.29 | fleeeb | hello all has anyone installed meetme2.c? for apps on asterisk |
| 02:58.36 | *** join/#asterisk s2angel (~s2angel@CPE00119508a661-CM000039529869.cpe.net.cable.rogers.com) |
| 02:58.57 | ZX81 | ~ping pong pang |
| 02:58.59 | jbot | pong pong pang |
| 02:59.02 | fleeeb | im just wondering if I needed pgres header files to compile it |
| 02:59.06 | *** join/#asterisk Funbags (~Funbags@ool-182e52ab.dyn.optonline.net) |
| 03:00.14 | DevilFish | Slainte: I'll try and catch you in here tomorrow then, I want to thankyou, the sip.ld had totally slipped my mind and I'll bet that mine is not current. |
| 03:00.20 | DevilFish | bootrom too |
| 03:00.34 | Slainte | no prob. glade I could be a little help |
| 03:00.39 | Slainte | chat tomorrow |
| 03:00.42 | Moc_ | ZX81 dono |
| 03:00.47 | DevilFish | cool, seeya |
| 03:01.49 | ZX81 | hmmmk |
| 03:01.49 | Yellow_Fuzzy | fleeeb: yes ive used meetme2.c before |
| 03:02.18 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 03:02.31 | fleeeb | Yellow_Fuzzy: i've patched the meetme2.c for the lastest cvs head but I keep getting htis error http://pastebin.com/228259 |
| 03:02.37 | Funbags | u sure it was rugby |
| 03:03.03 | ManxPower | Digium should document the interface between the modules and the card for their TDM400P card. Then other companies can make modules that Digium isn't interested in, like a Euroland ISDN module. |
| 03:04.02 | Moc_ | ManxPower, there is doc about it |
| 03:04.11 | Yellow_Fuzzy | fleeeb: from looking at that i would guess that you need the pgres head files |
| 03:04.22 | Moc_ | they actually just used a prototype of the people who made the chip |
| 03:04.42 | Moc_ | find the chip maker, of the TDM400 and you will find a 1 Port TDM400 like card |
| 03:05.19 | fleeeb | Yellow_Fuzzy: did you have them installed? |
| 03:05.20 | ZX81 | anyone know CDT time in USA? |
| 03:05.36 | fleeeb | Yellow_Fuzzy: because I really dont feel like installing them - I really just use mysql |
| 03:05.39 | Moc_ | must be 21h05 Central |
| 03:05.39 | ManxPower | ZX81: 905pm |
| 03:05.43 | ZX81 | ok |
| 03:05.45 | _Vile | yup |
| 03:05.46 | Moonwick | it's GMT-05 when it's in effect. |
| 03:05.49 | Moonwick | which it is not |
| 03:05.52 | ZX81 | :-) |
| 03:06.01 | ManxPower | LOL! |
| 03:06.04 | ManxPower | Well, yes. |
| 03:06.33 | ManxPower | It's 9:05pm CDT, which would be 10:05pm in CDT |
| 03:06.59 | *** join/#asterisk fofoz (~chiardon@200.24.76.91) |
| 03:07.03 | ManxPower | *sigh* It's been a long day. Just ignore me. |
| 03:07.09 | apsaras | hi. does anyone knows what is this message "TE410P: Double/missed interrupt detected" |
| 03:07.13 | fofoz | Hi |
| 03:07.20 | tzanger | apsaras: yeah -- your hardware's iffy :-) |
| 03:07.29 | ZX81 | ~seen twisted[work] |
| 03:07.30 | jbot | twisted[work] is currently on #asterisk (1d 9h 28m 44s). Has said a total of 72 messages. Is idling for 1d 2h 52m 52s |
| 03:07.35 | apsaras | iffy? |
| 03:07.54 | _Vile | ~seen _Vile |
| 03:07.55 | jbot | _vile is currently on #asterisk (1h 21m 39s). Has said a total of 40 messages. Is idling for 1s |
| 03:08.39 | tzanger | apsaras: yes, as in the system's either super-loaded down (CPU-bound), yrou PCI subsystem is unable to handle the interrupt rate the TE410 requires, you're sharing IRQs... problems like that |
| 03:08.44 | fofoz | hello |
| 03:08.54 | tzanger | apsaras: what system are you running this on, and what else is running on the system |
| 03:08.57 | *** join/#asterisk Kumbang (~ecvs@167.205.22.54) |
| 03:09.38 | tzanger | in particular are you running X11 or a distributed computing client or maybe a busy webserver? Are you sharing an IRQ with another card (check /proc/interrupts) |
| 03:10.08 | *** part/#asterisk lilneon (~tj_r3@cuscon4937.tstt.net.tt) |
| 03:10.29 | samueltc | hey, anyone tried the SPA 841? |
| 03:10.34 | samueltc | from sipura |
| 03:11.12 | fofoz | What can i do to connect the ANI fron an incoming call....match it whit an scheduling database an put on the screen the form related with this register??? |
| 03:11.13 | *** join/#asterisk josiah_ (~josiah@207.40.84.121) |
| 03:11.21 | josiah_ | hello all (again) |
| 03:11.34 | s2angel | anyone running asterisks from debian? |
| 03:11.48 | tzafrir_home | me |
| 03:11.55 | fofoz | all abou related with something like a call center function |
| 03:11.55 | ZX81 | bkw_: your 666 changed? |
| 03:12.01 | *** part/#asterisk Matt-E- (~matto@66-224-125-137.atgi.net) |
| 03:12.14 | fofoz | hi bkw!!! |
| 03:12.15 | MrEntropy | anyone here have some experience with freeradius? I'd like to ask some generic questions |
| 03:12.27 | s2angel | I just wanna get mine to run lol that would be a big step for me , trying to get mysql to work with it |
| 03:12.28 | bjohnson | anyone done voice recognition with asterisk? |
| 03:12.38 | ZX81 | sphinx |
| 03:12.40 | ZX81 | but not yet |
| 03:12.48 | ZX81 | bjohnson: |
| 03:12.52 | tzafrir_home | s2angel: what debian? |
| 03:13.26 | Slainte | s2angel. the Rapid Live CD is debian based |
| 03:13.31 | s2angel | I get the same error.. ugh see I am using some config scripts from a frind of mine |
| 03:14.01 | fofoz | how i can take the ani ....match it with DB and getting the form from it??? |
| 03:14.15 | josiah_ | does anyone know if asterisk can do generic SIP to PBX bridging (assuming dedicated FXO card to PBX analog port)? basically just bridge the analog PBX port to remote SIP or AIX client, both for dialing calls and receiving calls |
| 03:14.18 | ManxPower | The form? |
| 03:14.26 | *** join/#asterisk mikegrb (~michael@thegrebs.com) |
| 03:14.43 | fofoz | the information related with the matching register(ani and DB) |
| 03:14.47 | tzafrir_home | s2angel: what debian: stable? sarge? sid? |
| 03:14.56 | PTG123 | ~seen czero |
| 03:14.57 | jbot | czero <~me@CPE0090f800c5b0-CM001225704b6e.cpe.net.cable.rogers.com> was last seen on IRC in channel #asterisk, 10h 12m 25s ago, saying: 'bjohnson correct from canada'. |
| 03:14.57 | ManxPower | fofoz: It sounds like what you need to do is complex enough that you might get a better response if you post your question AND details to the mailing list. |
| 03:15.16 | modulus_ | ~seen jbot |
| 03:15.17 | jbot | jbot is currently on #ipaq (19d 5h 56m) #how (19d 5h 56m) #bzleague (19d 5h 56m) #newircd (19d 5h 56m) #storm (19d 5h 56m) #orkut (19d 5h 56m) #uphpu (19d 5h 56m) #va (19d 5h 56m) #asterisk (19d 5h 56m) #nslu2-linux (19d 5h 56m) #magnia (19d 5h 56m) #aegis (19d 5h 56m) #ol (19d 5h 56m) ... |
| 03:15.21 | bjohnson | ZX81: was just reading a page about X10 hardware controllers that seemed to suggest that voice recognition for hardware control had been done with combined FESTIVAL and IBM Viavoice SDK |
| 03:15.39 | ZX81 | Festival -> synthesis |
| 03:15.43 | ZX81 | not recognition |
| 03:15.48 | s2angel | sec not familuar with xchat sec |
| 03:15.48 | ZX81 | ViaVoice ah |
| 03:15.53 | fofoz | but it coud be something like CDR reading?? |
| 03:15.53 | ZX81 | for the recognition |
| 03:15.57 | ZX81 | haven't used it |
| 03:16.06 | bjohnson | josiah_: yes .. but it is a fxs card to the pbx and a fxo to the telco |
| 03:16.07 | ZX81 | I've heard of people getting sphinx working |
| 03:16.08 | *** join/#asterisk kludgebox (~bob@ppp-70-243-202-249.dsl.rcsntx.swbell.net) |
| 03:16.52 | bjohnson | ZX81: I assume for voice you usually want 2 way so voice generation with FESTIVAL and recognition with Viavoice |
| 03:16.58 | ZX81 | :-) |
| 03:16.59 | ZX81 | yeah |
| 03:16.59 | Brixius | ok, I have a small delima in my dialplan and I'm a little confused as to how to fix it, with regards to e164 phone #'s and looking them up, I have a dialplan that requires a 9 to get an outside line, but to lookup e164 phone#'s the 9 can't be there. |
| 03:17.01 | ZX81 | or sphinx |
| 03:17.03 | s2angel | I am runing knoppix harddrive installed from like a year ago been doing apt-get upgrades |
| 03:17.07 | fofoz | thanks manxpower but whats the reason that makeyou think that it is so hard to get?? |
| 03:17.13 | bjohnson | ZX81: http://w3.misterhouse.net:81/mh.html#usage_options |
| 03:17.17 | JonR800 | I thought IBM ditched viavoice for linux? |
| 03:17.28 | josiah_ | bjohnson: fxs? .... the asterisk server is just bridging pbx<->(sip|aix), no direct connect to telco...so fxs is appros for connect to analog pbx (specifically, BizFon 680)? |
| 03:17.50 | *** join/#asterisk ^Slash (slash@220-245-175-162-sa-pppoe.tpgi.com.au) |
| 03:17.57 | ManxPower | fofoz: It isn't, if you have everything configured. the database, the schema, the the AGI or other interface from Asterisk into the database. |
| 03:18.09 | bjohnson | josiah_: a fxs provides dial tone .. I use a fxs to connect to our PBX |
| 03:18.18 | Hymie | is there an option, which will cause asterisk to re-register via the iax register command? |
| 03:18.26 | Hymie | as in, re-register every 4 hours or so.. or some such |
| 03:18.29 | fofoz | but in your experience wich could be the best way?? |
| 03:18.35 | ManxPower | bjohnson: How is that working. Weren't you the one thwarted by the Nortel ATA? |
| 03:18.42 | s2angel | I am missing a file I comiled the one from the website .. where do I get the mysql file |
| 03:18.43 | ^Slash | when i change the tx/rxgain and i run ztmonitor should i see noticeable change when im talking ? |
| 03:19.13 | s2angel | I guess sarge parts and sid |
| 03:19.26 | Hymie | ^Slash: depends how much you change, I guess |
| 03:19.31 | *** join/#asterisk humberto (~hav@201.128.177.84) |
| 03:19.42 | josiah_ | bjohnson: hmm...dunno...i dont really need a dialtone _to_ the pbx - id like to bridge 6 dedicated ports on the pbx to six dedicated fx* cards in the asterisk server - so the cards would just need to receive ring voltage for incomming, and pickup hook and emit dtmf for outgoing, right? so would that still be fxs or fxo? |
| 03:19.43 | s2angel | tzafrir_home: its a mix.. not sure anymore |
| 03:19.46 | humberto | hello to all ,,, :) |
| 03:19.50 | *** join/#asterisk syslod (~sysglod@65.114.0.198) |
| 03:19.54 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-34-60.d4.club-internet.fr) |
| 03:19.56 | josiah_ | hello humberto |
| 03:20.02 | fofoz | whre i can look for interfaces from asterisk to the DB??? |
| 03:20.26 | ManxPower | fofoz: search the mailing list for MYSQL FRIENDS |
| 03:20.34 | *** join/#asterisk cblackbu (~cblackbu@c-24-23-43-130.client.comcast.net) |
| 03:20.39 | s2angel | tzafrir_home: I have been adding repositories and upgrading it for like a year now |
| 03:20.49 | fofoz | nothing in postgres that you know??? |
| 03:20.53 | *** join/#asterisk Funbags (~Funbags@ool-182e52ab.dyn.optonline.net) |
| 03:20.55 | ^Slash | i cant seem to get rid of this echo when dialing out over the fxo, i have changed the zconfig.h to inclead the aggressive_echo_can i have tried diff echo_can i have tried differnet rx/tx gains, nothing seems to change it, am i missing something ?? |
| 03:21.04 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 03:21.08 | ^Slash | im using a tdm400 card... |
| 03:21.12 | tzafrir_home | There are asterisk packages in backports.org |
| 03:21.20 | *** join/#asterisk wepy (~wepy@pcp09870567pcs.ewndsr01.nj.comcast.net) |
| 03:21.22 | ManxPower | fofoz: I'm sure there's postgress support too. Not much documentation. Search the mailing list archives. |
| 03:21.25 | ManxPower | To search the Asterisk mailing list archive go to www.google.com and put site:lists.digium.com in addition to your other query terms. |
| 03:21.25 | wepy | hello :) |
| 03:21.31 | Hymie | ^Slash: did you try turning echo cancellation on? |
| 03:21.34 | s2angel | can anyone point me to something that explains the mysql or something on databasing the calling and stuff.. need to read stuff |
| 03:21.37 | ^Slash | yeah its on |
| 03:21.43 | ^Slash | ive tried the echo training |
| 03:21.49 | ^Slash | all of it :-/ SIP calls are fine |
| 03:22.05 | ^Slash | its when i try to dial over the pstn it turns to echo central :( |
| 03:22.15 | *** join/#asterisk snewpy_ (~markl@203-206-235-209.dyn.iinet.net.au) |
| 03:22.17 | wepy | my company is thinking of purchasing a really really expensive VoIP system.. i was wondering if there are any docs available taht might tell me how to set up something nice with asterisk? |
| 03:22.21 | tzafrir_home | s2angel: database for what exactly? |
| 03:22.21 | fofoz | thanks manx power you are so cool!!! |
| 03:22.27 | Hymie | well... do you get the echo when you bridge to the pstn from sip? |
| 03:22.32 | *** join/#asterisk offroadgeek (~offroadge@dsl027-191-249.sfo1.dsl.speakeasy.net) |
| 03:22.54 | s2angel | tzafrir_home: holding the Calling logs |
| 03:23.13 | ^Slash | yep, i pick up the sip phone press extn 9 which is set to dial(zap/4) i get the normal dialtone i dial and when i speak i can hear my self |
| 03:23.54 | ^Slash | i have ztmonitor up on the screen and regardless of the changes i do, it shows the same results |
| 03:23.56 | Hymie | well.. if all the echo cancel / etc is on, and you just can get rid of it, I'd say it's time to check your wiring |
| 03:23.58 | *** join/#asterisk rumba1207 (~ropawa@cs68201148-205.sw.rr.com) |
| 03:23.58 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 03:24.12 | ^Slash | when i plug a normal phone into the line thats going to the fxo |
| 03:24.15 | ^Slash | its perfect |
| 03:24.20 | ^Slash | no crackle, no echo, nothing |
| 03:24.22 | Hymie | that's not relevant |
| 03:24.28 | ^Slash | oh ? |
| 03:24.31 | Hymie | it could still be wired wrong |
| 03:24.38 | ManxPower | ^Slash: You have to run ztconfig if you mess with /etc/zaptel.conf and stop/start Asterisk if yo mess with /etc/asterisk/zapata.conf |
| 03:24.39 | Hymie | there's a few good pages on echo cancelation |
| 03:24.40 | s2angel | tzafrir_home: I first have to get the file it requires first |
| 03:24.51 | Hymie | in the wiki |
| 03:24.59 | ^Slash | read heaps :-/ |
| 03:25.00 | s2angel | cdr_mysql.so |
| 03:25.03 | ManxPower | Also remember this is DB. You might want to increse by increments of 3 or 5 to get a noticable effect. |
| 03:25.10 | ^Slash | im not messing with /etc/zaptel |
| 03:25.30 | ^Slash | only /etc/asterisk/zapata.conf |
| 03:25.32 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 03:25.49 | Hymie | ^Slash: you do need to restart asterisk for changes to take effect in that file |
| 03:25.52 | ^Slash | yeah |
| 03:25.55 | ^Slash | i understand that |
| 03:25.59 | ^Slash | to reload the config |
| 03:26.08 | Hymie | right, a reload doesn't do it |
| 03:26.14 | Hymie | only actual restart does |
| 03:26.18 | ^Slash | yep, which i do |
| 03:26.21 | Hymie | volume isn't the issue though |
| 03:26.22 | ^Slash | i terminate the prog |
| 03:26.26 | Hymie | shit! sec! |
| 03:26.59 | ^Slash | but the fact that rxgainand txgain show no diff when i change that under ztmonitor to me means i must have something miss configd |
| 03:27.33 | Hymie | I would think so |
| 03:27.48 | Hymie | but.. you'll see a change going from an RX of 0 to 10, not 0 to 0.5 |
| 03:27.49 | *** join/#asterisk Funbags (~Funbags@ool-182e52ab.dyn.optonline.net) |
| 03:27.49 | ^Slash | like i said i even tried diff can options in the zconfig.h file for zaptel driver |
| 03:28.03 | humberto | Guys I wonder if someone can help me ,, I am getting some troubles with my main termination service provider I wonder if there is a Methond to do a "Fake ringback" and if congestion go to service provider 2 ,, (newbie) |
| 03:28.04 | ^Slash | yeah i have it set to 10 and -4.5 |
| 03:28.13 | ^Slash | 10 for rx and -4.5 to tx |
| 03:28.15 | ^Slash | no diff |
| 03:28.21 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 03:28.25 | s2angel | http://www.voip-info.org/wiki-Asterisk+cdr+mysql Never mind found a link talking about what I want to learn about.. thank you for your time |
| 03:28.34 | ^Slash | would that assume then the zaptel driver is broken ? |
| 03:29.28 | Hymie | well, it isn't here |
| 03:29.32 | Hymie | and for a lot of others |
| 03:29.35 | Hymie | are you using cvs or something? |
| 03:29.42 | ^Slash | surely if i was to set txgain and rxgain to something stupid like 50 if i wisperd it would jmp off the scale on the monitor ? |
| 03:29.43 | ^Slash | yep cvs |
| 03:29.52 | ^Slash | stable tho |
| 03:29.54 | ^Slash | not the dev |
| 03:29.59 | Hymie | ok... hmm |
| 03:30.09 | ^Slash | checked that a few times heh |
| 03:30.15 | Hymie | I think 50 would destroy a third world country ;) |
| 03:30.18 | bjohnson | ManxPower: only thwarted by not being able to use the ATA port to connect to another fxo to allow users to dial that extension to talk to asterisk |
| 03:30.42 | ^Slash | heh yeah, but i just set that terminated asterisk and loaded it up again and still ztmonitor has changed |
| 03:30.51 | ^Slash | :-/ |
| 03:30.53 | Hymie | don't know about that dude |
| 03:30.57 | ^Slash | hmm |
| 03:31.00 | Hymie | sonuds quite odd tohugh |
| 03:31.05 | ^Slash | heh it does |
| 03:31.06 | bjohnson | ManxPower: I already have 3 fxo/fxs pairs connecting the telco lines to the analogue lines in to the Nortel and another fxs providing a line 4 into the Nortel system |
| 03:31.20 | ManxPower | *nod* |
| 03:31.26 | *** join/#asterisk apsaras (~antonis@serial03-01.que01.acl.hol.gr) |
| 03:31.28 | Hymie | does ztmontior show you when you talk? do the levels change during conversation? |
| 03:31.49 | ^Slash | yeah when i talk it shows the levels moving |
| 03:31.53 | Hymie | well |
| 03:31.56 | Hymie | you're just weird |
| 03:31.56 | ^Slash | tx more so then rx |
| 03:32.13 | bjohnson | I was trying to use the Nortel ATA port as a 5th line to the * server (the Nortel hardware currently only supports 4 lines) |
| 03:32.46 | humberto | Manxpower,, do you know how do I do to have a backup route , if termination by service provider 1 fails , try service provider 2 (while I am in Ringback or waiting) |
| 03:32.47 | ManxPower | bjohnson: Have you considered eBay as a source for a used PRI or T-1 card for the system? |
| 03:32.57 | *** join/#asterisk ACiDV (~joel@122-68-181.dr.cgocable.ca) |
| 03:33.01 | ^Slash | what card are you using Hymie ? |
| 03:33.10 | Hymie | humberto: using Linux, are you? |
| 03:33.17 | humberto | yes |
| 03:33.32 | fofoz | manx power do you have another guide to helpme to find the way to connect the caller id with its information in a DB in real time??? |
| 03:33.38 | Hymie | ^Slash: the 4 port jigsaw puzzle card, one FXO, one FSO |
| 03:33.48 | Hymie | humberto: look at the LARTC howto |
| 03:34.08 | Hymie | humberto: some patches discussed there allow for automatic route failure detection and failover |
| 03:34.21 | bkw_ | you mean FXS |
| 03:34.27 | bkw_ | and FXO |
| 03:34.29 | bjohnson | ManxPower: yes .. but we've been happily using 3 lines and this gives us 4 lines .. plus if needed I have an extra analogue handset in an extra fxs port that I had .. so we don't really need the extra lines enough to spend $500 to $1000 |
| 03:34.30 | Hymie | bkw_: sure, that sounds about right ;) |
| 03:34.47 | humberto | Hymie: I am talikning about a congestion call not Ip routing |
| 03:34.50 | apsaras | does anyone know how to solve this proble? "TE410P: Double/missed interrupt detected" |
| 03:34.54 | Hymie | humberto: ah |
| 03:35.04 | ^Slash | would changing the fxols to the other fxo types make a diff ? |
| 03:35.04 | ManxPower | humberto: When a dial ends ${DIALSTATUS} is set. You should also look at the "g" option for Dial. |
| 03:35.24 | humberto | if a call was rejected if my service provider has no capacity or wrong route of a call |
| 03:35.29 | wepy | is there any Asterisk for Newbies kinda site? |
| 03:35.38 | ManxPower | also see qualify=yes in iax.conf or sip.conf |
| 03:35.55 | ACiDV | Hi, when I load zaptel driver for TDM400P, they show : Module x: AUTO FXO or AUTO FXS, etc... using printk (wctdm.c), so It "autodetect" modules present on the card. Does it's possible go add these information to /proc/zaptel/x ? |
| 03:35.59 | humberto | thanks Manxpower let me wiki a while on that |
| 03:36.04 | Hymie | humberto: look on the wiki for the dialplan snippet that allows for detection of busy, etc... and use it to try again, incrementing a variable with a wait |
| 03:36.12 | ^Slash | wepy voip-info.org |
| 03:36.18 | wepy | thanks |
| 03:36.32 | Hymie | ^Slash: sorry man, your weird problem is... weirding me out ;) |
| 03:36.36 | redder86 | yeah, like voip-info.org is even close to newbie level |
| 03:36.39 | Hymie | ^Slash: not sure where you should go from this point |
| 03:36.46 | ^Slash | :-/ |
| 03:36.53 | ^Slash | yeah its rather odd |
| 03:37.15 | humberto | thanks Hymie ,,, Wiki-ing |
| 03:37.32 | Hymie | ^Slash: except, obviously the levels shoudl change.. have you reloaded the modules since the problem started? done a cold boot (not use the soft power off button, but actually powered off the machine by the back switch, and left it off for 10+ seconds?) |
| 03:37.35 | redder86 | understanding voip-info.org at least requires a bit of introduction to know what to look for and also some discretion to know to not trust everything you see there |
| 03:37.54 | *** join/#asterisk jsharp (~jsharp@c-24-131-56-53.mw.client2.attbi.com) |
| 03:38.00 | redder86 | handing a newbie the VoIP encyclopedia is not always what the newbie wants |
| 03:38.01 | trazillion | Has anyone seen this message: Unable to create channel of type 'Zap' ? |
| 03:38.05 | bkw_ | yes |
| 03:38.09 | ^Slash | Hymie, tried it all heh |
| 03:38.10 | redder86 | trazillion: who hasn't seen it? |
| 03:38.12 | bkw_ | what else does it say |
| 03:38.14 | ^Slash | this is a fresh install tooo |
| 03:38.14 | Hymie | redder86: take that back! everything on voip-info.org is the plain, unvarnished truth! I'm going there now, to write a warning article about you! |
| 03:38.16 | jsharp | What's the status on app_chanspy. Does it work? Work well? |
| 03:38.19 | bkw_ | it says more than that |
| 03:38.25 | redder86 | Hymie: :-P |
| 03:38.26 | trazillion | Nothing |
| 03:38.37 | bkw_ | it should have a cause number |
| 03:38.55 | trazillion | no |
| 03:38.56 | ZX81 | :-( bkw changed 666 to get rid of me :-( |
| 03:39.01 | bkw_ | no |
| 03:39.03 | trazillion | <PROTECTED> |
| 03:39.03 | bkw_ | its still there |
| 03:39.10 | ZX81 | :-) |
| 03:39.13 | trazillion | <PROTECTED> |
| 03:39.16 | bkw_ | IAX2/guest@66.250.68.194/886 |
| 03:39.19 | bkw_ | er 996 |
| 03:39.22 | ZX81 | yah |
| 03:39.37 | trazillion | You can see http://asterisk.xvoip.com/viewtopic.php?p=2782#2782 for a full description |
| 03:39.47 | bkw_ | trazillion did you reinstall zaptel? |
| 03:39.48 | bkw_ | and all that |
| 03:39.51 | bkw_ | whats ztcfg -vvv |
| 03:39.52 | bkw_ | say? |
| 03:40.08 | Hymie | bkw_ |
| 03:40.09 | Moc_ | anyone use my dialconfirm app here ? |
| 03:40.11 | Hymie | talks like |
| 03:40.12 | trazillion | Channel 12: FXS Loopstart (Default) (Slaves: 12) |
| 03:40.12 | trazillion | Channel 21: FXO Loopstart (Default) (Slaves: 21) |
| 03:40.15 | Hymie | william shatner ;) |
| 03:40.19 | bkw_ | why 12-21 |
| 03:40.21 | bkw_ | wtf |
| 03:40.23 | trazillion | (first line for 1-12, second for 21-24) |
| 03:40.27 | bkw_ | ah |
| 03:40.29 | trazillion | (just didn't want to paste them all) |
| 03:40.32 | bkw_ | this is an E1? |
| 03:40.42 | bkw_ | nm |
| 03:40.43 | bkw_ | no |
| 03:40.47 | trazillion | T1 from TA750 |
| 03:40.51 | bkw_ | ok 1-12 are what kind of ports? |
| 03:40.58 | bkw_ | and 13-24 are? |
| 03:41.02 | trazillion | FXO on TA750 |
| 03:41.05 | bkw_ | no |
| 03:41.07 | redder86 | What is a TA750 ? |
| 03:41.11 | bkw_ | tell me physically what 1-12 is |
| 03:41.25 | trazillion | err, not 12-24 |
| 03:41.27 | trazillion | bah |
| 03:41.37 | bkw_ | what ever |
| 03:41.37 | trazillion | not 13-24, but 21-24 are fxs |
| 03:41.41 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 03:41.42 | bkw_ | ok |
| 03:41.50 | qwerp | harlo bkw_ |
| 03:41.57 | lilo | okay, so I'm finally in a position to get asterisk running 8) |
| 03:41.59 | bkw_ | trazillion I can only guess |
| 03:42.02 | qwerp | can i ask u a question? |
| 03:42.02 | Faithful | Is there place to get 1.0.3 deb package? |
| 03:42.15 | trazillion | ta750 is http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-30413019648.htm |
| 03:42.28 | bkw_ | zttool says? |
| 03:42.30 | lilo | what do I need to configure with zaptel? |
| 03:42.40 | trazillion | and we have it configured with the three fxo modules and one fxs module |
| 03:42.42 | redder86 | trazillion: does Zaptel work with that? |
| 03:42.45 | Faithful | I downloaded 1.0.3 but the compile failed at zaptel section |
| 03:42.53 | trazillion | yes, it looks like it's configuring it ok. |
| 03:43.01 | bkw_ | show me how you dial |
| 03:43.09 | bkw_ | post your zapata.conf and /etc/zaptel.conf to pastebin.ca |
| 03:43.26 | lilo | is there someplace I can look for an explanation of how to configure zaptel? |
| 03:43.27 | trazillion | I already had the URL it's posted at: http://asterisk.xvoip.com/viewtopic.php?p=2782#2782 |
| 03:44.02 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 03:44.30 | bkw_ | trazillion you can pay someone to look at this one |
| 03:44.50 | *** join/#asterisk mhnoyes_ (~mhnoyes@user-2ivfk3u.dialup.mindspring.com) |
| 03:44.53 | trazillion | I'm sure, but I was looking for a free solution. ;) |
| 03:45.19 | bkw_ | you're gonna have ot have someone that knows what they are doing look at it hands on.. such as me or many of the other people |
| 03:45.56 | bkw_ | it could be 100 diffrent things |
| 03:46.03 | bkw_ | why are you using ls? |
| 03:46.15 | bkw_ | that thing not support anything else? |
| 03:46.20 | trazillion | because the ta750 only supports ls and gs |
| 03:46.27 | bkw_ | tried gs? |
| 03:46.52 | trazillion | No, because we were guessing that ls is the standard setup for the telco |
| 03:47.12 | bkw_ | telco and standard in the same sentence |
| 03:47.13 | bkw_ | funny |
| 03:47.33 | trazillion | (well, the tech for the ta750 suggested that would be the case) |
| 03:48.24 | trazillion | Is there a way to tell other than changing the setup? |
| 03:48.35 | bkw_ | thats all I would do if I were there |
| 03:48.38 | bkw_ | is change it all |
| 03:48.45 | bkw_ | flip it on its head |
| 03:49.02 | Hymie | exit |
| 03:49.28 | josiah_ | aight guys...well im headn to bed...night all |
| 03:51.04 | trazillion | ok, we just changed everything to groundstart, but same result |
| 03:51.22 | trazillion | From the configuration we described, do the zaptel.conf and zapata.conf files look right? |
| 03:51.36 | Bentley | was DUNDi written primarily for ISPs, or can any * user use it to make free LD calls? |
| 03:51.45 | trazillion | Should we need to adjust any of the other parameters, such as gain? And if so, how do we know what values? |
| 03:52.12 | implicit | anyone here currently using ValetParking |
| 03:52.18 | implicit | and cvs-head |
| 03:52.39 | syslod | implicit: I have it but haven't tested . |
| 03:52.44 | implicit | spent a couple hours trying to port it over i am having weird coredumps now though |
| 03:52.51 | redder86 | Bentley: I have a hard time understanding why DUNDi was written. It appears to simply be a different way to produce a network of VoIP numbers. |
| 03:52.54 | harryvv | hi implicit :) |
| 03:52.54 | *** join/#asterisk _gorman (~lehmann@sulaco.franken.de) |
| 03:53.02 | wolfson | bently: anyone can use dundi |
| 03:53.07 | implicit | syslod: you changed it over to work with current HEAD? |
| 03:54.06 | syslod | We'll I've been merging changes but havn't specificly had to do thing for valet. |
| 03:54.15 | Bentley | wolfson: as i understand it, to get started, I need to execute the GPA with another peer. Where do I find a peer in the global 'trust group'? |
| 03:54.24 | implicit | syslod: there are LOTS of changes in how all that works |
| 03:54.43 | Bentley | redder86: yes, kinda like gnutella i think |
| 03:54.47 | Brixius | Dundi when peered with a significant sized network allows you to discover if there is an ip route to a remote #, if not you can route via PSTN if you want. |
| 03:54.57 | redder86 | Bentley: like you'd think that if there were another system out there with extensions that wanted you to dial to them that they would provide you access information independent from DUNDi. |
| 03:55.10 | implicit | syslod: res_features.c is a lot diff |
| 03:55.12 | syslod | U mean compile or issues trying to get it to work? |
| 03:55.42 | redder86 | Bentley: if my friend "Joe" wants me to call him on his VoIP, then he would give me his VoIP access information... that doesn't require DUNDi. |
| 03:55.42 | implicit | well a lot of the asterisk internals have changed |
| 03:55.48 | implicit | so i've been changing the code |
| 03:55.53 | syslod | Lemme park a call. |
| 03:55.54 | implicit | to try to get it to work properly |
| 03:56.00 | implicit | whic hversion of HEAD do you have? |
| 03:56.27 | syslod | version of HEAD? |
| 03:56.28 | trazillion | Well, we are signing off.... bkw_ if you have any more thoughts, please post them on that forum page for us. Thanks! |
| 03:56.37 | implicit | syslod: what date |
| 03:56.47 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 03:56.49 | implicit | it shouldn't compile for you if you have recent head |
| 03:56.50 | Bentley | It's confusing to me. It appears that "the DUNDi E164 web of trust" would allow any * user to make free calls ... but how to execute that initial GPA? |
| 03:56.51 | ^Slash | would i be right in thinking that ztmonitor should show balanced rx and tx levels ?? |
| 03:56.52 | blitzrage | wow, funny that I know almost all the people listed in my google ads |
| 03:57.02 | harryvv | I dont see config info on one such softphone clinet. that info goes into iax and sip.conf and if so, are thay configured with the ip adress of those clients? |
| 03:57.09 | blitzrage | astricon.net, atacomm, asterlink and signate |
| 03:57.29 | redder86 | Bentley: why would you want any * users making free calls on your system? |
| 03:58.02 | syslod | implicit: cvs up -d -A |
| 03:58.02 | Bentley | redder86, same reason I let people download my mp3s over gnutella i guess ....? |
| 03:58.38 | Brixius | redder86: a better example would be, I have ast sys1 peered with sys2 who is peered with sys3, now from sys1 I dial a sys3 #, dundi will ask sys2 do you know how I get to xyz phone #, sys 2 asks sys3 do you know, sys 3 say's yes, here's the info I have that #, then sys2 tells sys1 how to get there. |
| 03:58.41 | implicit | syslod: you know that valetparking doesn't come with asterisk by default right? |
| 03:58.42 | Brixius | if you can follow that |
| 03:58.46 | Bentley | redder86, but good question ;-) I'm trying to understand it .. |
| 03:58.49 | syslod | Right. |
| 03:58.49 | redder86 | Bentley: are they really your MP3s? ;-) |
| 03:59.07 | Bentley | it doesn't matter up hear in Canada ;-) |
| 03:59.11 | implicit | syslod: when did you do your cvs update? |
| 03:59.17 | implicit | and do you have valetparking compiled? |
| 03:59.34 | harryvv | 13 lines back |
| 03:59.37 | syslod | ValetParking: Valet Parking is in applications |
| 03:59.38 | redder86 | Brixius: well, I understand what DUNDi does and I kind of get why it is there, only I don't understand why anyone would use it. |
| 03:59.55 | syslod | did cvs up -d -A a few minutes ago and compiled. |
| 04:00.09 | file[laptop] | okay, scenario people: You have multiple asterisk spread out at offices for your company, they each have their own extensions |
| 04:00.20 | file[laptop] | how are you going to interconnect your systems and have all the extensions accessible everywhere? |
| 04:00.23 | Moc_ | syslod, I got a nice upcoming patch for dialconfirm |
| 04:00.25 | implicit | syslod: yeah but did valetparking compile for you |
| 04:00.36 | Moc_ | I just got 1 thing to change and it done |
| 04:00.46 | implicit | syslod: asterisk compiles fine, but valetparking doesnt thats why i have been patching it |
| 04:00.47 | redder86 | file: I do that without DUNDi quite easily. |
| 04:00.50 | Bentley | file[laptop], iax.conf and extensions.conf? |
| 04:00.54 | syslod | implicit: Yea but current I can't keep a SIP phone registered long enough to park a call at 7XX |
| 04:01.02 | file[laptop] | redder86: for large installations it makes more sense and stuff |
| 04:01.06 | syslod | Moc: Thats cool is it in the bugs db? |
| 04:01.18 | implicit | syslod: is this CVS head or CVS stable? |
| 04:01.28 | Moc_ | syslod, not yet |
| 04:01.28 | file[laptop] | to add an extension all you do is add it on one box, publish the extension and tada... all the rest can suddenly get to it... |
| 04:01.30 | syslod | Should be HEAD. |
| 04:01.34 | redder86 | file: okay, I get that, but why would they peer up with "any old asterisk user" ? |
| 04:01.39 | file[laptop] | you don't have to |
| 04:01.44 | Bentley | file[laptop], so it's more for enterprise use, and not for small * users to bypass VSPs, etc? |
| 04:01.50 | implicit | syslod: mind showing me a copy of your app_valetparking.c |
| 04:01.50 | Mavvie | the asterisk dev kit PCI card.... does it come with some kind of card documentation so people can write drivers for it? |
| 04:01.52 | file[laptop] | dundi-test is a test environment :) |
| 04:01.53 | redder86 | file: but that's what it's being sold as |
| 04:01.54 | file[laptop] | thus the name |
| 04:02.30 | file[laptop] | DUNDi can be used within an enterprise to create a fully-federated PBX with no central point of failure, and the ability to arbitrarily add new extensions, gateways and other resources to a trusted web of communication servers, where any adds, moves, changes, failures or new routes are automatically absorbed within the cloud with no additional configuration. |
| 04:02.58 | file[laptop] | there is a DUNDi group yes, but you don't have to join it |
| 04:03.02 | file[laptop] | you can use DUNDi internally... |
| 04:03.37 | file[laptop] | ugh, my neck hurts now |
| 04:03.39 | *** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg) |
| 04:03.44 | syslod | implicit: I installed valet when I first started testing * HEAD weeks ago. Is it possible to have it in show applications but not compiled in? Or are there multiple valetparking apps? |
| 04:03.58 | Brixius | redder86: you wouldn't really want to publish your trunk's to "any old asterisk user" you can publish your phone #, and lookup other's phone #'s so as to not use the pstn to get there. |
| 04:04.09 | bkw_ | IBM alone was granted 3,248 patents last year. |
| 04:04.19 | bkw_ | HOLY SHIT MAN |
| 04:04.23 | bkw_ | thats almost 10 a day |
| 04:04.23 | file[laptop] | Brixius: indeed |
| 04:04.34 | implicit | syslod: yeah you compiled it before when it hadn't changed yet |
| 04:04.39 | implicit | bkw_: mind pming me for a sec ;) |
| 04:04.40 | Mavvie | there are only 250 days in a year. |
| 04:04.47 | Mavvie | you can't file them on saturday or sunday. |
| 04:04.54 | bkw_ | true |
| 04:05.00 | syslod | implicit: That must be it cause it doesn't seem to work. |
| 04:05.01 | bkw_ | implicit in a fwe |
| 04:05.22 | redder86 | Mavvie: plus there's at least two-weeks' worth of gov't holidays |
| 04:05.35 | redder86 | Mavvie: well, maybe only one |
| 04:05.57 | *** part/#asterisk jsharp (~jsharp@c-24-131-56-53.mw.client2.attbi.com) |
| 04:06.32 | *** join/#asterisk BigCanOfTuna (~chatzilla@S0106000c41b313f3.cg.shawcable.net) |
| 04:06.43 | BigCanOfTuna | Hello everyone. |
| 04:07.23 | *** part/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 04:07.31 | Brixius | If I put part of my extensions.conf up on pastebin does anyone have time to take a look at it to see if there's an easy to change it so e164 lookup's work with it? if nobody has time I'm not going to post it. |
| 04:07.48 | *** join/#asterisk aokaze (aokaze@modem170.supermusca.bigsky.net) |
| 04:08.08 | Brixius | I'm a little confused right now. |
| 04:09.12 | *** join/#asterisk redder86 (~lee@gateway.howardsilvan.com) |
| 04:10.18 | BigCanOfTuna | How do I get the last number that was entered? I want to prompt the user for the conference number, when he enters it, I want it to use it for a parameter in the next command. |
| 04:12.08 | ^Slash | Hymie i fixed my problem, apparently if i specify in the /etc/asterisk/zapata.conf at the end of the config the signaling and channels it all works, i still have echo but the tx/rx is now working |
| 04:12.11 | harryvv | in sip.conf defaultip-ipofclient like a softphone? |
| 04:12.18 | harryvv | = |
| 04:12.27 | Moc_ | syslod, try http://bugs.digium.com/bug_view_page.php?bug_id=0002714 plz |
| 04:12.39 | syslod | Moc_: K |
| 04:12.39 | Moc_ | the only apply the app_dial.c-confirm_rev1.txt |
| 04:12.47 | Bentley | hey all, one more Q about DUNDi: What is the purpose of the "DUNDi E164 web of trust"? (@ http://www.voip-info.org/wiki-Asterisk+DUNDi+Call+Routing) |
| 04:12.50 | harryvv | moc that is right? |
| 04:12.51 | bjohnson | BigCanOfTuna: hmm .. I saw that in an example earlier today on the wiki .. LASTNUM I htink |
| 04:12.52 | Moc_ | no need of app_dialconfirm anymore |
| 04:13.03 | BigCanOfTuna | bjohnson: thanks! |
| 04:13.07 | lilo | is it possible to use more than one X100P in a single machine? |
| 04:13.16 | harryvv | yes |
| 04:13.31 | lilo | I just specify the configuration in /etc/zaptel.conf? |
| 04:13.40 | harryvv | yes |
| 04:13.43 | syslod | Moc: Nice |
| 04:13.49 | nestAr | whee.. i have polycom ip300 on my home desk now.. |
| 04:14.03 | Moc_ | nestAr, coool ;) |
| 04:14.09 | bjohnson | whee .. I have 2 SPA 3000 and 2 SPA 2000 coming |
| 04:14.11 | Moc_ | you like the options ? ;) |
| 04:14.15 | s2angel | snom phones are pretty cool |
| 04:14.16 | syslod | Poly is doing a backlit very soon. |
| 04:14.19 | Moc_ | syslod, hey, I hope it work .. |
| 04:14.30 | Moc_ | nice .. hope they keyboard is backlit too ;) |
| 04:14.35 | syslod | I'll know in a minute. |
| 04:14.43 | s2angel | got 10 190 snom.. they got Linux in side .. |
| 04:14.44 | syslod | No keyboard :( |
| 04:14.48 | BigCanOfTuna | bjohnson: no =( |
| 04:14.55 | Moc_ | syslod, it a new model ? |
| 04:15.06 | harryvv | Can somone confirm to me if this is right. I am going to use a softphone client for now untill I get somephones. Is the defaultip= in sip.conf the ip of the subject host machine that has the softclient on it? |
| 04:15.11 | syslod | I'm getting a Pana IP beta which is cool but they will list at $700 |
| 04:15.21 | s2angel | snom phones you can like mod them to use your own ring samples .. kool |
| 04:15.28 | syslod | Its not out. Our dist send a cheat sheet. |
| 04:15.47 | Moc_ | send me, Im the big polycom fan here ;) |
| 04:16.00 | nestAr | now i need to get a better DSL filter and hook my SPA-3000 to my pots line.. |
| 04:16.11 | s2angel | Get snom german IP phone.. the company cares |
| 04:16.11 | syslod | The pana TDA phones are gonna go IP and will support SIP. They are nice and have backlit indiglo. |
| 04:16.30 | s2angel | you use your phones n the dark? |
| 04:16.49 | s2angel | our office is well lit |
| 04:16.51 | syslod | Moc_: It'll be based on the 600 and will have a backlit display and a 2.5mm jack option. |
| 04:17.13 | Moc_ | ha that it :( |
| 04:17.16 | syslod | I have both backlit and poly phone and the backlit makes a huge diff. |
| 04:17.21 | syslod | Thats it. |
| 04:17.33 | syslod | Moc_: Take a look at the TDA phones they are really nice. |
| 04:17.43 | syslod | If you have ppl willing to pay the $700 |
| 04:17.44 | s2angel | can the poly do intercom? |
| 04:17.46 | Moc_ | well 700 $!!!! Im not that crazy... not anymore ;) |
| 04:17.48 | Moc_ | ofcourse |
| 04:18.01 | s2angel | snom not that expensive |
| 04:18.07 | syslod | 35% markup is better on a 700 phone :) |
| 04:18.09 | s2angel | frig it super cheep at 10 phones |
| 04:18.19 | syslod | cool = $$$ |
| 04:18.30 | s2angel | aprox $200 a phone if you order like 10 |
| 04:18.31 | Moc_ | polycom are cheap and are the best phone outthere |
| 04:18.32 | syslod | poly can do intercom. |
| 04:18.41 | syslod | Yea I aggree thats what we are pushing. |
| 04:18.48 | s2angel | does it run linux |
| 04:18.57 | s2angel | snom runs linux |
| 04:20.07 | s2angel | I think this snom 190 I got suports IAX even .. wow |
| 04:20.09 | ACiDV | Someone here have test Aastra 480i with latest firmware ? (1.0.0.78) |
| 04:20.14 | syslod | Moc_: I'm happy I got logos and microbrow working. |
| 04:20.35 | syslod | Now if the buttons just matched up it would be great. |
| 04:21.04 | Moc_ | buttons matched up ? |
| 04:21.27 | s2angel | they must be diffrent colours |
| 04:21.30 | syslod | YEa. Instructions don't match 500 buttons and the buttons functions aren't right. |
| 04:21.44 | s2angel | bad documentation? |
| 04:21.54 | syslod | Bad docs or bad phones? |
| 04:22.16 | syslod | They are programmable but I've been lazy. I just though it should be right outta the box. |
| 04:22.25 | syslod | Moc: Ur buttons right? |
| 04:22.51 | *** join/#asterisk JimVanM (~JimVanM@HSE-MTL-ppp77210.qc.sympatico.ca) |
| 04:23.02 | s2angel | why spend then have to pay a licence and not have access to the source code.. when you can support a company that cares about the hardwdware and keeping things open.. |
| 04:23.33 | syslod | s2angel: I've never seen a snom phone. |
| 04:23.50 | s2angel | snom for future support and no restrictions.. documention seems pkentiful.. |
| 04:24.17 | Moc_ | syslod, yes |
| 04:24.33 | Moc_ | btw you can reconfigure the location of the button on the phone via the xml config |
| 04:24.39 | s2angel | they are rying the american market.. but I think not many know that they are there.. yet.. or that they are so open.. |
| 04:24.58 | s2angel | I am german so I knoticed them at a show |
| 04:25.19 | Moc_ | I didnt got impress by snom phone, EXCEPT, they have a console phone, it the only thing that could interest me |
| 04:25.43 | syslod | Yea one of the guys developing our desktop controler setup his buttons. |
| 04:25.45 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 04:25.57 | s2angel | when a manufactor decieds that htis product is end of line.. your crap out of luck.. with open system.. it just seems to be supported longer.. |
| 04:26.14 | syslod | Yea that console issue has a lot of customers going cisco or Key. |
| 04:26.39 | s2angel | say a new codecs just tottaly changes vopip and makes a ton of phones outta date fast.. humm.. It was a big factor in our big expense.. |
| 04:28.03 | s2angel | the fact that it does the job .. allow to make phone calls.. its openn.. and its less then a 7960 c has the quality |
| 04:28.08 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 04:28.40 | s2angel | no back lite that I see on this snom 190.. but Its software is changing so dam fast |
| 04:29.07 | syslod | software is not gonna get a backlit display. |
| 04:29.28 | s2angel | not happy withthe software and a bit of a linux guru then compile your own kernel I think mine has 16 megs ram |
| 04:29.46 | *** join/#asterisk NTJOCK (~brian@txshirts.com) |
| 04:29.56 | Moc_ | polycom keep adding stuff, it crazy... xml file is getting big now ;) |
| 04:30.06 | s2angel | true.. I did'nt say its perfect.. but if it seems to be a feature .. I am sure they will offer a model.. |
| 04:30.08 | Moc_ | they will need to be bzip soon |
| 04:30.11 | *** join/#asterisk kludgebox (~bob@adsl-70-243-202-249.dsl.rcsntx.swbell.net) |
| 04:30.16 | s2angel | I think they do offer a model with a backlite.. |
| 04:30.17 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
| 04:30.24 | s2angel | 200 I think |
| 04:30.27 | s2angel | snom.com |
| 04:30.48 | ACiDV | about polycom, where I can get XML config "specification" ? for provisioning |
| 04:30.59 | NTJOCK | Question: When configuring Zaptel.conf and you have more then one card, what is the syntax to reference each card in the config fileS? Or where is this documented? |
| 04:31.00 | Moc_ | ACiDV, check voip-info |
| 04:31.10 | ^Slash | with the echo cancel, is it ment to start off realy bad echo and then clear up ? |
| 04:31.15 | Moc_ | there is link to documentation, xml example, and 1.3.1 firmware |
| 04:31.42 | s2angel | does the poly have a web interface ? |
| 04:31.47 | syslod | Yep |
| 04:31.51 | Moc_ | yes, but dont use it ;) |
| 04:31.54 | syslod | It sux. |
| 04:31.56 | Moc_ | use the ftp server |
| 04:31.58 | s2angel | it was our second choice |
| 04:32.09 | Moc_ | polycom had too much config option |
| 04:32.11 | syslod | Wish they had https. |
| 04:32.13 | s2angel | the snom has a rather nice one |
| 04:32.17 | ACiDV | Moc_ thanks, I've found |
| 04:32.27 | Moc_ | wish it used a sftp, I guess I should ask |
| 04:33.09 | Moc_ | s2angel, check polycom xml config PDF doc, and you will see why web interface for it is useless hehe |
| 04:33.51 | s2angel | cost made a diffrence.. and the fact alot of development on intercom and asterisk support s something that turned our heads toward it.. .. we could of went to the silly buget tones.. |
| 04:33.51 | Moc_ | you can get the phone working via the web, but there is just so much you can do via xml |
| 04:33.51 | s2angel | having 7 lines was nice too |
| 04:34.18 | s2angel | moc_ is there a link to the licenced poly xml pdf .. i'd like to see this.. maybe I will get a poly for a test phone |
| 04:35.05 | harryvv | seems arts interfears with the cli but its needed to make a custom greeting. |
| 04:35.14 | NTJOCK | anyone? zatpel.conf how to reference more then one digium card? |
| 04:35.14 | s2angel | I got a linksys pap2-na for a cordless phone at home with 2 lines on the cordless phone.. thous little sipura knock off is pretty decent |
| 04:35.15 | harryvv | artsd if im not mistaken |
| 04:37.45 | Mavvie | NTJOCK: just keep counting. |
| 04:37.57 | NTJOCK | so card 1 is 1-4 |
| 04:38.02 | NTJOCK | card 2 is 5-8? |
| 04:38.05 | Mavvie | yes |
| 04:38.10 | NTJOCK | ok cool |
| 04:38.12 | NTJOCK | just checking |
| 04:38.19 | NTJOCK | and how do I determine card 1 and 2? |
| 04:38.41 | harryvv | 1-4?? |
| 04:38.52 | lilo | hmmmm, is this my X100P? 0000:00:0d.0 Communication controller: Unknown device e059:0001 |
| 04:38.57 | Mavvie | NTJOCK: it's in order of PCI numbering. |
| 04:39.05 | harryvv | ntkock i sent you the info in msg format. |
| 04:39.13 | NTJOCK | oh ok |
| 04:39.43 | harryvv | mavie, you mean that the pci slots are in ordered number? |
| 04:39.43 | NTJOCK | well we'll see if I get it right... if I don't I'll get errors.. :) |
| 04:40.47 | Mavvie | harryvv: they have a certain order yes. |
| 04:40.54 | *** join/#asterisk hmmhesays (~noway@24-116-232-138.cpe.cableone.net) |
| 04:41.10 | harryvv | I see |
| 04:41.17 | Mavvie | if you do lspci you will see them: |
| 04:41.21 | NTJOCK | well that did work |
| 04:41.21 | NTJOCK | :) |
| 04:41.25 | harryvv | sure |
| 04:41.26 | NTJOCK | now on to zapata.conf |
| 04:41.28 | ^Slash | whats a good codec for sip phones to be using on the local lan ? |
| 04:41.28 | Mavvie | 00:00.0 Host bridge: Intel Corp. 82875P Memory Controller Hub (rev 02) |
| 04:41.28 | Mavvie | 00:03.0 PCI bridge: Intel Corp. 82875P Processor to PCI to CSA Bridge (rev 02) |
| 04:41.29 | Mavvie | 00:1c.0 PCI bridge: Intel Corp. 6300ESB 64-bit PCI-X Bridge (rev 02) |
| 04:41.33 | Mavvie | that's all internal on the mother board. |
| 04:41.39 | Mavvie | 02:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) |
| 04:41.39 | Mavvie | 02:06.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) |
| 04:41.45 | NTJOCK | Digium suggested the GSM was a good codec for general purpose |
| 04:41.56 | *** part/#asterisk JimVanM (~JimVanM@HSE-MTL-ppp77210.qc.sympatico.ca) |
| 04:41.58 | Mavvie | that is most likely the digium card. |
| 04:42.01 | harryvv | mavie you have a zap or two? |
| 04:42.07 | ^Slash | im using digium tdm400 |
| 04:42.08 | Mavvie | so on PCI ID 02:03.0 is the first card. |
| 04:42.10 | ^Slash | so i should use gsm ? |
| 04:42.15 | NTJOCK | said most people wouldn't be able to tell the difference between GSM and Ulaw/Alaw |
| 04:42.16 | Mavvie | and on PCI ID 02:06.0 is the second card. |
| 04:42.30 | NTJOCK | I've heard Alaw is better |
| 04:42.39 | NTJOCK | but GSM is what your cell phone uses (if it's new) |
| 04:42.42 | NTJOCK | and it does work |
| 04:42.55 | Mavvie | gsm is good enough, but *law is better. |
| 04:43.05 | NTJOCK | the direct answer is it doesn't really matter what you use on the local lan |
| 04:43.17 | NTJOCK | unless you are sending calls out by VOIP in which case use what your VOIP path will use |
| 04:43.23 | Mavvie | on the local LAN I would just use alaw/ulaw. |
| 04:43.25 | NTJOCK | that way you avoid transcoding the call from one codec to the other |
| 04:43.38 | NTJOCK | that's the gist of what I have picked up so far |
| 04:43.47 | ^Slash | yeah im testing em out and i can slightly tell the diff between them |
| 04:43.59 | ^Slash | alaw seems to be the clearest of em all |
| 04:44.10 | ^Slash | no doubt the most bw too heh :) |
| 04:44.32 | Mavvie | it's a trade off. |
| 04:44.34 | ^Slash | and as for echo its normal to have some fir a few seconds and then for it to clear up ? going over the pstn that is |
| 04:44.45 | ^Slash | fir=for |
| 04:45.01 | ^Slash | sip -> sip seems fine, just over the pstn is a tad dodge :-/ |
| 04:45.05 | *** join/#asterisk daver76 (~daver@ns1.tmok.com) |
| 04:46.03 | *** join/#asterisk xbmodder (~xbmodder@adsl-67-119-13-192.dsl.snfc21.pacbell.net) |
| 04:46.04 | xbmodder | hey |
| 04:46.16 | xbmodder | can asterisk call out? |
| 04:46.41 | *** join/#asterisk _gorman (~lehmann@sulaco.franken.de) |
| 04:46.42 | ^Slash | yeah with a zap device |
| 04:47.03 | xbmodder | zap device? |
| 04:47.12 | xbmodder | a normal modem will work... right? |
| 04:47.13 | BigCanOfTuna | Is it possible to conditionally include contexts? For example, if the user is calling from an internal SIP phone, allow include the [local.pstn] context. |
| 04:47.16 | ^Slash | www.digium.com |
| 04:47.39 | xbmodder | grrrrr |
| 04:47.59 | xbmodder | any programs you know that will dial out and play a pre-recorded message |
| 04:47.59 | Moc_ | xbmodder, get one, or use a voip provider |
| 04:48.00 | ^Slash | not sure |
| 04:48.14 | xbmodder | VOIP provider |
| 04:48.18 | xbmodder | like vonage? |
| 04:48.21 | ^Slash | im using a digium.com card |
| 04:48.27 | xbmodder | can i use vonage softphone? |
| 04:48.28 | ^Slash | ;-/ |
| 04:48.29 | Moc_ | no, like nufone, or voiceconduits |
| 04:48.42 | Moc_ | you can use x-ten softphone with * |
| 04:48.56 | Moc_ | nufone is about 2cent/min anywhere in US/Can |
| 04:49.14 | Moc_ | you can get a 1800# too, it 2cent/min also |
| 04:49.33 | harryvv | That would be nice to have a long list of phone numbers and * would dial out say a message of some importants to tenents of a complex or something. |
| 04:50.28 | harryvv | moc, nufone was talking to me the other day about consulting fees :) i said possibly in the future. I did not relize thay were also in that line of biz. |
| 04:50.45 | daver76 | I am using VoicePulse for my VoIP line |
| 04:50.46 | daver76 | works fine... |
| 04:51.25 | harryvv | I may have to sign up as a simple account to dial into my box. testing with a cell phone is getting a little pricy. Finding people willing to do it is not going to work. |
| 04:51.51 | Moc_ | I got my firefox today atless |
| 04:52.14 | Moc_ | hope to get my polycom tomorow |
| 04:52.14 | lilo | hmmmm, it doesn't seem to be finding the X100P card |
| 04:52.58 | NTJOCK | I'm getting an error "Line 140: Unable to open master device '/dev/zap/ctl'" |
| 04:53.01 | Moc_ | and I hope I can get this polycom soundstation 3000 to work with * |
| 04:53.02 | NTJOCK | how do I fix this? |
| 04:53.19 | NTJOCK | Using a debian package install |
| 04:53.21 | NTJOCK | new system |
| 04:53.30 | harryvv | nt, ivr seen that and fixed that prlblem. That is probebly on a google search. |
| 04:53.36 | *** join/#asterisk [TripleF] (kvirc@67.70.100.136) |
| 04:53.37 | NTJOCK | I've tried creating the object |
| 04:53.44 | harryvv | NTJOCK that is on google. |
| 04:53.55 | *** join/#asterisk IronHelixz (~irc@ool-182c8f9f.dyn.optonline.net) |
| 04:54.12 | harryvv | nt, in fact go and verify that /dev/zap exist |
| 04:54.17 | NTJOCK | it does |
| 04:54.21 | NTJOCK | been there done that |
| 04:54.26 | harryvv | okay..in my case it did not. |
| 04:54.27 | NTJOCK | tried creating the object by hand |
| 04:54.35 | NTJOCK | the package installer ran into some sort of problem and died |
| 04:54.38 | harryvv | googled it? |
| 04:54.45 | zigman | ztcfg -vv |
| 04:54.47 | NTJOCK | so I opened another SSH shell and created it by hand per the directions |
| 04:54.53 | NTJOCK | no I haven't googled it yet |
| 04:54.53 | ^Slash | i had the same probs nt |
| 04:54.56 | *** join/#asterisk m-00kie (3704558@pcp0010383411pcs.arlngt01.va.comcast.net) |
| 04:54.58 | ^Slash | so i installed it all with cvs |
| 04:55.05 | NTJOCK | I did not need to see any porn ... but I will in a minute |
| 04:55.18 | NTJOCK | google has quality issues. |
| 04:55.26 | NTJOCK | how did you fix it Slash? |
| 04:55.56 | NTJOCK | the package has issues anyhow.... it defaults to grabbing the Zap source instead of the binaries. |
| 04:56.08 | NTJOCK | which is ok, but not exactly how a package should work. |
| 04:56.25 | daver76 | NTJOCK, you need the source because drivers are generally dependent on kernel versions |
| 04:56.29 | NTJOCK | I'm not sure if it is a l or 1 |
| 04:56.39 | NTJOCK | ah ok, thanks daver |
| 04:56.39 | daver76 | it makes sense if you think about it... |
| 04:56.42 | NTJOCK | yea it does |
| 04:56.49 | NTJOCK | and DSP is a low level task when done right |
| 04:56.54 | NTJOCK | let me try a make clean |
| 04:57.35 | ^Slash | i fixed it by grabbing the cvs |
| 04:57.39 | ^Slash | and manually installing |
| 04:57.43 | devi0us | anyone have any idea why festival sounds fine on sjphone, but sounds like crap on a sipura ata? |
| 04:58.13 | *** join/#asterisk jjg (~clh@63.227.141.77) |
| 04:58.17 | jjg | hi |
| 04:59.31 | devi0us | both being on the same segment as the ast server, so no latency or packet loss |
| 05:00.26 | jjg | i've almost got voicemail going but i'm getting the following error : No application 'VoiceMail' for extension |
| 05:00.28 | jjg | ? |
| 05:00.33 | harryvv | what should i use to create a custom greeting? |
| 05:01.20 | jjg | audacity and sox is what i just used for the first time |
| 05:01.29 | jjg | super easy |
| 05:01.37 | daver76 | I just set up an extension that records to a file... |
| 05:01.43 | jjg | :\ |
| 05:01.45 | jjg | heh |
| 05:01.49 | jjg | how? |
| 05:01.54 | jjg | rtfm |
| 05:01.55 | jjg | heh |
| 05:02.15 | daver76 | exten => 101,1,Record(welcome:gsm) |
| 05:02.18 | daver76 | something like that |
| 05:02.42 | jjg | my * isn't recognizing VoiceMail |
| 05:02.51 | jjg | i have noload for vmail2 commented out |
| 05:03.03 | jjg | and i'm using VoiceMail(u5000@default) |
| 05:03.13 | jjg | but it says No Application |
| 05:03.27 | devi0us | is 5000 set up in voicemail.conf? |
| 05:03.52 | jjg | yeah |
| 05:04.22 | devi0us | do any other applications work? like MusicOnHold? |
| 05:05.36 | *** join/#asterisk _TripleF_ (kvirc@Toronto-HSE-ppp3880909.sympatico.ca) |
| 05:05.40 | harryvv | jjg yea, what was the command to record? |
| 05:05.59 | *** join/#asterisk FryGuy- (fryguy@c-24-2-50-122.client.comcast.net) |
| 05:06.34 | jjg | Background and Playback do |
| 05:06.34 | *** join/#asterisk brian23_ (~konversat@209.114.219.98) |
| 05:06.38 | jjg | does the vmail context need tobe the same as the context fromwhich it is called in extensions.conf? |
| 05:06.44 | *** join/#asterisk brc_ (~brian@ip24-251-178-25.ph.ph.cox.net) |
| 05:07.35 | *** join/#asterisk m-00kie (3704558@pcp0010383411pcs.arlngt01.va.comcast.net) |
| 05:10.29 | PTG123 | ~seen czero |
| 05:10.30 | jbot | czero <~me@CPE0090f800c5b0-CM001225704b6e.cpe.net.cable.rogers.com> was last seen on IRC in channel #asterisk, 12h 7m 58s ago, saying: 'bjohnson correct from canada'. |
| 05:11.16 | jjg | harryvv : exten => 101,1,Record(welcome:gsm) |
| 05:11.41 | ^Slash | when somone calls and asterisk answers if they hang up it doesnt detect that the other end has gone, im using fxo_gs which i thought detected that for you ? |
| 05:12.06 | ^Slash | or is there a trick to this ? :) |
| 05:12.24 | blankman | Hey, so anyone want to talk about the ins and outs of making * use IMAP instead of the file system for voicemail storage? |
| 05:13.01 | jjg | should this work : exten => s,3,VoiceMail(u5000@default) ? |
| 05:13.32 | blankman | jjg, the voicemail context needs to match the context that you put the extension in. Are you use the db or the voicemal.conf? |
| 05:13.40 | jjg | .conf |
| 05:13.50 | blankman | Did you make any additional contexts? |
| 05:14.03 | blankman | [context-name] |
| 05:14.07 | jjg | no |
| 05:14.24 | blankman | Then yes, that would work, but you could also not have a context, as the default context is default. |
| 05:14.28 | jjg | i have 5000 => 5000 in voicemail.conf |
| 05:14.49 | blankman | ummm .. that doesn't look right have you looked that wiki? |
| 05:15.06 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 05:15.07 | jjg | well, 5000 => 5000,username,email |
| 05:15.20 | blankman | oh ... okey. |
| 05:15.21 | *** join/#asterisk greendisease (~jack@greendisease.fedora) |
| 05:16.02 | jjg | but I don't have to explicity goto 5000 in the dialplan, right? it should just go right to the vmbox with VoiceMail(u5000@default) ? |
| 05:16.06 | blankman | so .. yes, your orig. question should work. |
| 05:17.31 | *** join/#asterisk DavidSpears (~spears_da@65.163.100.254) |
| 05:17.31 | blankman | Well yes, but you need something like this exten => {some_ext},{some_priotity},Voicemail([u|b]5000) |
| 05:17.51 | blankman | Otherwise there is no way to get to it ... did that make sense to you? |
| 05:18.02 | jjg | i have that |
| 05:18.05 | jjg | yes |
| 05:18.09 | jjg | thanks |
| 05:18.11 | ST-3 | slash: do you really have gs lines? |
| 05:19.04 | DavidSpears | Excuse me, does anyone know if there is a special DTMF setting for the Cisco 7960 phones to work with Asterisk using a 4 port FXO Digium Card? |
| 05:19.21 | mishehu | DavidSpears: I've never heard of any such thing. |
| 05:19.25 | ST-3 | david: need to make sure that the dtmf setting on the phone matchines * |
| 05:20.04 | mishehu | if it works with an x100/101p, it should work with a tdm04b |
| 05:21.03 | DavidSpears | This has been checked... When dialing out, its as though the numbers entered on the phone dont go out to the telco correctly and the operator comes in asking you to dial the number again, its about to drive me up the wall. |
| 05:21.19 | ST-3 | OH |
| 05:21.28 | ST-3 | thats not a dtmf thing |
| 05:21.42 | ST-3 | sounds like your dial string out the zap isnt dialing properly |
| 05:21.49 | ST-3 | whats your dial string look like? |
| 05:22.13 | *** join/#asterisk kludgebox (~bob@adsl-70-243-202-249.dsl.rcsntx.swbell.net) |
| 05:22.35 | mishehu | dtmf_inband: 1, dtmf_outofband: avt |
| 05:22.42 | mishehu | that's in the SIPDefault.cnf for me, |
| 05:22.52 | mishehu | and in sip.conf, I use rfc2833 for dtmfmode |
| 05:23.01 | mishehu | my 7960 works fine. |
| 05:23.05 | DavidSpears | Well I basicly copied the one from Digium's site. from http://www.digium.com/index.php?menu=configuration#TDM0XB |
| 05:23.09 | ST-3 | mishehu: doesn't it sound like when * is dialing out it isnt dailing the #? |
| 05:23.16 | jjg | do i need to have voicemail and voicemail2 commented out in modules.conf? |
| 05:23.26 | ST-3 | like he did Dial(Zap/g1) isntead of Zap/g1/${EXTEN} |
| 05:24.15 | DavidSpears | I used the Dial(Zap/g1/${EXTEN} |
| 05:24.38 | DavidSpears | with the \20t |
| 05:24.54 | ST-3 | umm |
| 05:25.15 | DavidSpears | If I set it for 9 to dial/zap/g1 it grabs an outside dial tone and I can call fine. |
| 05:25.54 | DavidSpears | If I use the {EXTEN} to pass the digits through, It will sometimes go out and then other times it wont. |
| 05:26.00 | ST-3 | hrmm |
| 05:26.04 | DavidSpears | I have 4 lines |
| 05:26.09 | DavidSpears | I have swapped lines. |
| 05:26.14 | mishehu | DavidSpears: your gains set properly? |
| 05:26.29 | DavidSpears | let me see what they are set to. |
| 05:26.38 | ST-3 | I have 0 experience with * and analog lines so I can't comment.. but I do have 7960's feeding a PRI |
| 05:26.50 | mishehu | I have actually found a txgain/rxgain combination that makes dialing completely cease functioning. |
| 05:27.05 | DavidSpears | Tx and Rx are at 0.0 |
| 05:27.14 | mishehu | shoudl be fine. |
| 05:27.21 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 05:27.24 | mishehu | what firmware is the cisco running? |
| 05:27.46 | AgiNamu | damn... refactor anyone? |
| 05:27.56 | DavidSpears | 6-3-00 |
| 05:28.05 | jjg | should i see app_voicemail.so from "show modules" at the CLI ? |
| 05:28.09 | AgiNamu | Would Marc accept patches if they were massive refactorings? |
| 05:28.27 | mishehu | DavidSpears: have you tried different versions of zaptel? |
| 05:28.31 | mishehu | and asterisk? |
| 05:28.31 | AgiNamu | err, Mark even |
| 05:28.33 | ST-3 | david: you maight want to NoOp(${EXTEN}) before your dial and make sure its not messed up |
| 05:29.15 | mishehu | I've never seen these types of problems like he is having. |
| 05:29.43 | ST-3 | mishehu: same here |
| 05:29.58 | ST-3 | david can you paste/email/watever your dialplan? |
| 05:30.16 | DavidSpears | Sure where to? |
| 05:30.22 | ST-3 | umm email it to me, ser@tch.org |
| 05:30.46 | ST-3 | ok.. i normally hate Starbucks, but there are some HOT grrrls in here |
| 05:30.46 | mishehu | pastebin |
| 05:30.56 | mishehu | I'd use pastebin instead |
| 05:30.59 | jjg | ok, so i had to manually load app_voicemail |
| 05:31.03 | ST-3 | that works too |
| 05:31.04 | jjg | that normal? |
| 05:31.04 | mishehu | jbot pastebin |
| 05:31.05 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try pastebin.ca |
| 05:31.28 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 05:31.28 | mishehu | jjg: I normally let the applications autoload |
| 05:32.00 | DavidSpears | Its on the wire. |
| 05:32.22 | DavidSpears | Its my outbound dial plan, I am compiling the entire dial plan now. |
| 05:32.31 | *** join/#asterisk tenz (tenz@134.115.159.169) |
| 05:33.04 | ST-3 | \20t ? |
| 05:33.10 | ST-3 | shouldnt that be ,t |
| 05:33.17 | tenz | hi guys where should a newbie who has just installed asterisk be directed to? :) |
| 05:33.25 | AgiNamu | www.voip-info.org |
| 05:33.28 | Silik0n | yeah |
| 05:33.31 | ST-3 | yah voip-info |
| 05:33.33 | tenz | ty |
| 05:33.33 | jjg | mishehu: how? |
| 05:33.45 | jjg | i have it uncommented in modules.conf |
| 05:33.46 | tenz | im reading the handbook atm :) |
| 05:33.54 | jjg | heh |
| 05:34.00 | jjg | just got vmail emailed ot myself |
| 05:34.05 | jjg | trying to get xmms to openit |
| 05:34.22 | jjg | yahoo plugin alerted me via gaim |
| 05:34.26 | jjg | nice |
| 05:34.34 | ST-3 | davidspears: I dunno what this \20t is |
| 05:35.03 | AgiNamu | Hmm, if I go into the channels directory, and do "make chan_iax2.so" -- it doesn't work |
| 05:35.09 | AgiNamu | but doing a make from the main dir does.. what give?S |
| 05:35.24 | *** join/#asterisk shaZwaz (~luckyali@203.81.196.167) |
| 05:35.46 | shaZwaz | morning ppl |
| 05:35.57 | jjg | morning |
| 05:36.54 | DavidSpears | I just emailed you my entire dial plan, and then I emailed you the information from Digium's site as they say to set it up. Does it look like I set it wrong? |
| 05:37.26 | ST-3 | david: I am not familar with this \20 thing |
| 05:38.18 | DavidSpears | how would you do it? |
| 05:38.31 | DavidSpears | I was not sure about it either but I was taking their word for it. |
| 05:39.39 | ST-3 | should be ,20,t |
| 05:39.52 | shaZwaz | \20 ? |
| 05:40.07 | shaZwaz | is it |20 |
| 05:40.10 | ST-3 | <PROTECTED> |
| 05:40.14 | ST-3 | shaz: yah | will work too |
| 05:40.20 | DavidSpears | did you get the last email where they had it on the website? think it was a typo? |
| 05:40.26 | ST-3 | yah i see that |
| 05:40.38 | AgiNamu | yey! I think i found the place where the format is decided to send |
| 05:40.45 | ST-3 | david: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Dial |
| 05:40.57 | AgiNamu | nested *8* levels deep |
| 05:41.37 | kram | whois file |
| 05:41.45 | jjg | so what is my voicemail passwordfor vmail.cgi? i don't see anything about setting the password |
| 05:42.01 | AgiNamu | Because obviously a 600 line switch is easy to read and maintain :) :) :) |
| 05:42.52 | ST-3 | david: I just reread that example, its |20 not \20 |
| 05:43.05 | AgiNamu | We could implement a shared "preferred codec", if we add ast_best_codec2, and have it accept a channel, to grab channelvars. |
| 05:43.16 | AgiNamu | Would that be better than having "IAX2_CODEC"? |
| 05:43.20 | mishehu | o/~ love is in the air o/~ |
| 05:43.22 | DavidSpears | ST-3? I just tried the .t as you suggested.. I have called 7 local numbers and all 7 connected. |
| 05:43.32 | ST-3 | ,t (comma t) |
| 05:43.36 | ST-3 | or |t |
| 05:43.53 | mishehu | mrT |
| 05:44.02 | *** join/#asterisk blankman (~blankman@h000d88a1570c.ne.client2.attbi.com) |
| 05:44.02 | mishehu | (I pity tha foo!) |
| 05:44.12 | DavidSpears | exten => _NXXXXXX,1,Dial(Zap/g1/${EXTEN}|t) Correct? |
| 05:44.15 | ST-3 | I PITY THE FOO WHO INVESTED MONEY IN LUCENT |
| 05:44.49 | ST-3 | david: right, unless you want the 20 sec timeout in which case its Dial(Zap/g1/${EXTEN},20,t) |
| 05:45.02 | ST-3 | , and | seem to be interchangeable |
| 05:45.17 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-83-43.sentechsa.net) |
| 05:45.25 | ST-3 | (someone correct me if I am wrong) |
| 05:45.56 | AgiNamu | in IAX2, "format" referrs to preferred codec, and "capabilities" to all possible codecs? |
| 05:47.26 | mishehu | ST-3: no, you are correct. |
| 05:47.29 | mishehu | no need to correct you |
| 05:47.43 | shaZwaz | ! false |
| 05:47.54 | shaZwaz | which is true |
| 05:48.19 | ST-3 | mishehu: I havn't seen that documented anywhere, but over time I have used every combination of | , and using and not using ( ) 's :) |
| 05:48.23 | jjg | anyone know where to set the webvmail password? |
| 05:48.37 | ST-3 | isnt the web vmail pass the same as the pass in voicemail.conf? |
| 05:49.47 | jjg | i think but it isn't working |
| 05:50.16 | ST-3 | is vmail.cgi using the wrong context? |
| 05:50.26 | ST-3 | does vmail.cgi get confused by extra spaces in the .conf? |
| 05:51.05 | jjg | ST-3 |
| 05:51.06 | jjg | on |
| 05:51.07 | jjg | no |
| 05:52.23 | *** join/#asterisk mrsquiggle (~galec@adsl-35-88.swiftdsl.com.au) |
| 05:53.28 | AgiNamu | Is there an app or var that has the current codec in use? |
| 05:53.56 | jjg | ok, i got it |
| 05:54.54 | bkw_ | AgiNamu no |
| 05:56.00 | AgiNamu | bkw_, is it as simple as mychannel->pvt->rawreadformat ? |
| 05:56.07 | Kumbang | anyone experienced with e&m with E1 |
| 06:02.00 | DavidSpears | ST-3, Do you see a problem with me leaving it as exten => _NXXXXXX,1,Dial(Zap/g1/${EXTEN}) |
| 06:02.20 | DavidSpears | I did that and I have connected with every local and LD call placed from the phones. |
| 06:02.28 | DavidSpears | No timeouts specified. |
| 06:02.56 | ST-3 | I dont see why that would be a problem |
| 06:03.42 | ST-3 | bbl |
| 06:03.46 | DavidSpears | Then thats what I will do, I appreciate all your help, have a good remainder of the night. |
| 06:03.51 | ST-3 | no prob |
| 06:06.36 | niZon | hmm, my music on hold always seems to suddenly stop working |
| 06:06.44 | niZon | and after a few restarts it works again |
| 06:06.48 | Wi_Fi | yo |
| 06:08.00 | shaZwaz | hi Wi_Fi |
| 06:10.38 | Wi_Fi | how doo |
| 06:11.11 | AgiNamu | i mean, really, is there no reason not to refactor a lot? |
| 06:11.20 | AgiNamu | the code would be a shitload easier to follow.. |
| 06:11.37 | Wi_Fi | anyone usin fwd able to receive calls from fwd? |
| 06:16.05 | shaZwaz | I am Wi_Fi |
| 06:16.07 | *** join/#asterisk reni (matt@pcp0011024458pcs.huntsv01.al.comcast.net) |
| 06:18.20 | *** join/#asterisk qwenky45 (~cooked@c-67-170-199-46.client.comcast.net) |
| 06:18.26 | qwenky45 | Hello |
| 06:24.03 | *** join/#asterisk Guest^DJ (~some@218.208.234.20) |
| 06:26.14 | *** join/#asterisk jontow (~jontow@ws.woflsys.net) |
| 06:26.51 | AgiNamu | in iax2_call(), will the current pvt's owner have the channelvars from the dialplan? |
| 06:26.56 | jontow | hmmm |
| 06:26.58 | jontow | hi all :) |
| 06:27.00 | AgiNamu | hi |
| 06:27.37 | libpcp | any implemented 4xE1 ISDN in asterisk located in singapore? |
| 06:27.39 | *** join/#asterisk ta[i]nted (~tainted@65-60-70-242-cust.telepacific.net) |
| 06:27.46 | Guest^DJ | hi all , IAX vs SIP , which is better to call over broadband ? |
| 06:27.55 | libpcp | where can i find a 4 port ISDN card and its driver? |
| 06:28.40 | wolfson | guest^dj, the wiki has a large comparison |
| 06:28.43 | wolfson | each has strengths |
| 06:29.12 | Guest^DJ | wolfson: hmmmm ok |
| 06:29.44 | qwenky45 | I have a NMS Systems AG4000 board, will that work with Astrex? |
| 06:29.51 | qwenky45 | *Asterisk |
| 06:29.57 | *** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com) |
| 06:30.15 | zigman | has anyone tried to chroot asterisk ? |
| 06:32.12 | *** join/#asterisk CybreWulf (cybre@killcybre.org) |
| 06:33.03 | ^Slash | anyway to make the hold music not as loud ? it distorts when i press hold |
| 06:33.23 | ^Slash | like it plays but at 300% more volume then it should |
| 06:33.41 | tzanger | http://slashdot.org/comments.pl?sid=135863&cid=11345225 |
| 06:33.43 | tzanger | HAHAHAHHAHA |
| 06:33.46 | tzanger | the child comments too |
| 06:34.21 | *** join/#asterisk hmodes (hmodes@dsl092-231-153.phl1.dsl.speakeasy.net) |
| 06:34.59 | shaZwaz | basically it's a toothbrush-cum-vibrator |
| 06:35.20 | Wi_Fi | shaZwaz hey |
| 06:35.31 | *** join/#asterisk sgnome` (~s_gnome65@ip70-176-146-245.ph.ph.cox.net) |
| 06:36.08 | Wi_Fi | shaZwaz how is your setup for incoming calls from fwd |
| 06:36.39 | shaZwaz | its working |
| 06:36.39 | Wi_Fi | i mean are you able to call yourself thru fwd exten? |
| 06:37.00 | shaZwaz | I think so |
| 06:37.01 | sgnome` | does anyone have time to help someone with their first asterisk configuration? |
| 06:37.30 | Wi_Fi | shaZwaz call 545296 |
| 06:37.42 | ta[i]nted | lol |
| 06:37.57 | ta[i]nted | i heard that's the same reason by those "back" massagers are so popular |
| 06:38.28 | shaZwaz | Wi_Fi: went dead |
| 06:38.28 | sgnome` | i've been working on this for like 2 hours and read everything on the voip-info.org wiki .. and still cannot get my softphone to register .. or my FWD to work |
| 06:38.50 | Wi_Fi | thats my problem |
| 06:39.06 | Wi_Fi | i can call out but nothing can get in |
| 06:39.16 | shaZwaz | Wi_Fi: 411 is working |
| 06:39.17 | Kumbang | can anyone translate this cisco3260 configs to zaptel configs |
| 06:39.22 | Kumbang | controller E1 0/0 |
| 06:39.22 | Kumbang | <PROTECTED> |
| 06:39.22 | Kumbang | <PROTECTED> |
| 06:39.22 | Kumbang | <PROTECTED> |
| 06:39.22 | Kumbang | <PROTECTED> |
| 06:39.26 | Guest^DJ | sgnome`: u need to configure on your softphone and sip.conf |
| 06:39.35 | Guest^DJ | and extensions.conf too |
| 06:39.39 | xbmodder | i got a SIP phone company |
| 06:39.41 | AgiNamu | meh, when iax2_call() is called, how can I get the channelvars? |
| 06:39.42 | sgnome` | i edited sip.conf and extensions.conf .. and still no luck |
| 06:39.47 | AgiNamu | What's the best way to dump all channelvars on a channel? |
| 06:39.50 | AgiNamu | (in C)? |
| 06:39.56 | xbmodder | i set it up in kphone |
| 06:40.05 | Guest^DJ | your softphone configured? |
| 06:40.05 | xbmodder | how do i do so in asterisk |
| 06:40.17 | xbmodder | Guest^DJ, i set it up in kphone |
| 06:40.25 | Wi_Fi | shaZwaz yes 411 works awsome |
| 06:40.27 | xbmodder | no asterisk support yes... |
| 06:41.05 | sgnome` | guest^dj .. i am not quite sure how to configure my softphone to use the asterisk box |
| 06:41.30 | Guest^DJ | sgnome`: what softphone you using? |
| 06:41.37 | sgnome` | x-lite or iaxcomm |
| 06:41.56 | shaZwaz | Wi_Fi: most of the time I get user offline for some numbers |
| 06:42.06 | *** join/#asterisk StilexIP (~none@pc-24-151-59-224.newt1.ct.charter.com) |
| 06:42.12 | JerJer | 7960! no less |
| 06:42.15 | shaZwaz | tho still on xlite it works |
| 06:42.16 | JerJer | accept no less |
| 06:42.22 | StilexIP | anyone get meetme2 working on the lastest cvs head? |
| 06:42.24 | AgiNamu | Anyone here familiar with chan_iax2.c? |
| 06:42.36 | xbmodder | register => 1<yournumber>:<secret>@atlas-east.mmfff44.net:5060 #where do i put this? |
| 06:43.24 | sgnome` | guest^dj .. mind if i msg you? |
| 06:43.52 | AgiNamu | xb in the sip.conf, general section |
| 06:45.06 | xbmodder | ok |
| 06:45.20 | xbmodder | grrr... |
| 06:45.22 | xbmodder | not working |
| 06:47.59 | xbmodder | Jan 12 22:47:45 WARNING[16384]: chan_sip.c:1881 sip_register: Host 'host.vopr.vonage.net' not found at line 32 |
| 06:48.45 | JerJer | it won't |
| 06:48.58 | ta[i]nted | will vonage work with *? |
| 06:49.07 | xbmodder | it should |
| 06:49.14 | xbmodder | its a SIP provider |
| 06:49.16 | JerJer | only if your dumb enough to pay for the soft client account |
| 06:49.27 | xbmodder | uhh well i am that dumb |
| 06:49.33 | xbmodder | i use the soft client alot |
| 06:49.34 | ST-3 | i hear vonage is pretty unfriendly to customer supplied equipment |
| 06:49.36 | ta[i]nted | why would it be considered dumb |
| 06:49.43 | ST-3 | (asterisk in this case) |
| 06:49.44 | xbmodder | i hate there equipment |
| 06:49.58 | xbmodder | well i know all the setting |
| 06:50.07 | xbmodder | i wanna get angry at asterisk |
| 06:51.23 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:52.11 | StilexIP | anyone get meetme2 app working with cvshead? |
| 06:52.40 | AgiNamu | why would anyone pay for vonage, and pay $10 a month extra to use your own stuff |
| 06:53.21 | xbmodder | i like vonage |
| 06:53.22 | ST-3 | why would anyone pay for vonage for use with *? Seems to me like there are lots of other providers that are cheaper then are * friendly |
| 06:53.38 | xbmodder | well how do i configure it |
| 06:53.59 | JerJer | ST-3: ppl are sheep. they really believe its unlimited |
| 06:54.12 | WilliamK | heya JerJer |
| 06:54.19 | xbmodder | Jan 12 22:51:20 WARNING[16384]: chan_sip.c:1881 sip_register: Host 'host.vopr.vonage.net' not found at line 32 ##<< I really need to fix this... |
| 06:54.19 | JerJer | moo |
| 06:54.39 | JerJer | um perhaps that hostname does not exist |
| 06:54.50 | ST-3 | jerjer: yah, but look at like broadbvoice and iconnecthere, dont they have unlimited plans too? |
| 06:55.13 | JerJer | ** server can't find host.vopr.vonage.net: NXDOMAIN |
| 06:55.17 | WilliamK | Jer, do you know if greg got a chance to look at the pri debug that I sent him? |
| 06:55.20 | *** join/#asterisk HjemmeRoyK (~roy@110.80-203-29.nextgentel.com) |
| 06:55.36 | JerJer | nobody has a "without bounds, unrestricted" plan |
| 06:55.40 | xbmodder | does that mean that kphone is superior? |
| 06:56.01 | HjemmeRoyK | <PROTECTED> |
| 06:56.13 | brc_ | hi jerjer |
| 06:56.16 | JerJer | moo |
| 06:56.16 | brc_ | how's it going |
| 06:56.28 | brc_ | that well eh? |
| 06:56.31 | brc_ | going to von? |
| 06:56.51 | brc_ | g'nite |
| 06:56.57 | xbmodder | yay fixed... |
| 06:56.59 | xbmodder | or not |
| 06:57.04 | xbmodder | it does not pickup |
| 06:57.08 | xbmodder | i called into the number |
| 06:57.10 | xbmodder | nothjing |
| 06:57.14 | xbmodder | gr |
| 06:57.21 | ST-3 | jerjer: thats crazy though.. are these people really using that many minutes? |
| 06:57.26 | ST-3 | doesn't seem worth it |
| 06:57.49 | JerJer | vonage (et al) will cut you off at some point |
| 06:57.55 | JerJer | thus its not unlimited |
| 06:57.55 | ST-3 | of course they will |
| 06:57.57 | xbmodder | i want it to call out and give the user a message |
| 06:58.15 | xbmodder | like call me at 3 PM and remind me to go to john's house |
| 06:58.46 | JerJer | then do that |
| 06:59.06 | *** join/#asterisk aceton (~aceton@wsip-24-248-81-121.ph.ph.cox.net) |
| 06:59.11 | xbmodder | how |
| 06:59.18 | aceton | morning |
| 06:59.43 | JerJer | xbmodder: cron and a call file |
| 06:59.45 | xbmodder | i want asterisk to call me and say 'hello sargun, this is a reminder that you have to go to john's house' |
| 06:59.46 | JerJer | cake |
| 06:59.55 | xbmodder | call file eh? |
| 07:00.10 | JerJer | do i need to drop the RTFM bomb? |
| 07:00.16 | xbmodder | AT |
| 07:00.26 | xbmodder | you agree at is better |
| 07:01.21 | aceton | i was wondering if anyone could suggest some good hardware |
| 07:01.27 | aceton | i just ordered the Wildcard TE110P T1 card and the TDM40B |
| 07:01.35 | aceton | need a good motherboard for these little guys |
| 07:01.37 | JerJer | Dell 2650 |
| 07:02.01 | aceton | really? |
| 07:02.03 | devi0us | ok. theoretical question. i have an application that i need to record all traffic on a nufone iax2 DID. what's the best way of recording it? |
| 07:02.13 | xbmodder | haha |
| 07:02.13 | JerJer | app_monitor |
| 07:02.17 | *** join/#asterisk kludgebox (~bob@adsl-70-243-202-249.dsl.rcsntx.swbell.net) |
| 07:02.53 | aceton | is the 2650 rackmount? |
| 07:03.01 | JerJer | yes |
| 07:03.04 | AgiNamu | so no one here can help me figure out some channel code? |
| 07:03.17 | _Vile | 1+1=2 |
| 07:03.32 | JerJer | 1+1 = 3 |
| 07:03.39 | JerJer | (in 9 months) |
| 07:03.43 | _Vile | the answer to life and everything is 42 |
| 07:03.57 | aceton | ok, i am not finding one, the 2850 replaced it right? |
| 07:04.08 | devi0us | what's a 2650 going for these days? i'm used to HP gear. the 2650 is a dl380 equiv |
| 07:04.18 | _Vile | cheap |
| 07:04.24 | _Vile | bbl |
| 07:04.25 | zigman | Oh yes, there is 10 types of people in the |
| 07:04.25 | zigman | world, one who can read binary and one who cannot |
| 07:05.20 | aceton | ok, so the 2850 should be cool then? how many pci slots does it have? |
| 07:06.07 | devi0us | is app_monitor included anymore? |
| 07:06.42 | xbmodder | Jan 12 23:06:23 NOTICE[65541]: pbx.c:1214 pbx_extension_helper: Cannot find extension context 'incoming' |
| 07:06.50 | StilexIP | hey can anyone help me - I've installed app_meetme2.c and everything has compiled correctly - but its not writing to my sql database or even attempting to - I was wondering if someone can point me in a direction so I can fix this |
| 07:06.51 | xbmodder | now thats what i get |
| 07:07.06 | devi0us | n/m. i'm a dumbass |
| 07:07.26 | zigman | devi0us right ;) |
| 07:11.01 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 07:11.17 | aceton | anyone else have any luck with and outer mb/server implementations |
| 07:14.32 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 07:16.38 | AgiNamu | Well, I got my patch for "IAX_CODEC" ("PREFERRED_CODEC") working |
| 07:17.08 | AgiNamu | But I had to patch app_dial ,a nd that sucks. so im looking for a ncier way, specifically, how do I get the channel with the variables from iax2_call()? |
| 07:17.08 | xbmodder | god |
| 07:18.54 | *** part/#asterisk aceton (~aceton@wsip-24-248-81-121.ph.ph.cox.net) |
| 07:20.14 | JerJer | um why not simply choose one codec per type=peer or type=user ? |
| 07:21.15 | jontow | hmm.. i was crawling through our server room hardware racks and found a strange dual-RJ11-port dialogic card of some sort |
| 07:21.22 | jontow | i may have to google some part numbers off that one :) |
| 07:24.27 | *** join/#asterisk JamesDotCom (~james@sweep.bur.st) |
| 07:25.51 | *** join/#asterisk KeRneL (~VeriaBLes@85.96.82.28) |
| 07:27.05 | *** join/#asterisk CpuID (~nathan@dsl-202-173-176-86.qld.westnet.com.au) |
| 07:27.08 | CpuID | afternoon ppls |
| 07:31.14 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 07:36.04 | *** join/#asterisk johngalt (1000@dsl081-088-238.lax1.dsl.speakeasy.net) |
| 07:38.50 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 07:40.23 | johngalt | I'm trying to understand call flow in *. are channels and extentions the same thing? |
| 07:40.53 | shaZwaz | how can I send web 2sms in germany vodafone |
| 07:41.02 | johngalt | at least mostly having a 1 to 1 corospondance |
| 07:42.52 | jontow | johngalt; no. |
| 07:43.12 | jontow | extensions are completely virtual; channels are tied to something (usually a device like a SIP phone) .. then there are the "Local" channels |
| 07:43.19 | jontow | check the wiki.. good explanations of all :) |
| 07:43.23 | jontow | http://www.voip-info.org/ |
| 07:44.17 | *** join/#asterisk alakdan (~dax@saidi-gwy01.meridiantelekoms.net) |
| 07:51.23 | CpuID | hrm, anyone know if i can select not to play vm-intro when going to a specific voicemail box? |
| 07:52.07 | JerJer | show application Voicemail |
| 07:52.30 | *** join/#asterisk Inv_arp (junya@adsl-8-230-104.mia.bellsouth.net) |
| 07:52.45 | CpuID | sec |
| 07:53.02 | CpuID | aha so thats what the prefix s does :) |
| 07:53.10 | CpuID | was wanting to know that earlier, thx |
| 07:53.30 | *** join/#asterisk phaze (~phaze@pcbcu420a.unil.ch) |
| 07:58.42 | *** join/#asterisk Duckbizkit (~jcunningh@24-240-243-142.charter.com) |
| 07:59.45 | Duckbizkit | nice, awesome to see * get ./ed |
| 08:01.02 | Inv_arp | hm that spipura 841 looks neat nice price too |
| 08:01.04 | *** join/#asterisk cdegroot (~Cees@tai42.xs4all.nl) |
| 08:04.27 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 08:13.09 | johngalt | jontow: does each channel have an exctention? I'm trying to figure out how the mapping works. I have a fxo zap channel and an fxs zap channel and it would seem that they have the context of default in zapada.conf I'm trying to figure how these map into the extentions.conf I'm writing |
| 08:13.15 | *** join/#asterisk lele (~lele@rivendell.windmill.it) |
| 08:14.02 | johngalt | when a call comes in on the fxo port how does * know what to do with it? |
| 08:15.14 | *** part/#asterisk Duckbizkit (~jcunningh@24-240-243-142.charter.com) |
| 08:16.19 | johngalt | conversely when the fxs port dials how does it know how to match the call? is matching limited to that context? |
| 08:17.01 | *** part/#asterisk alakdan (~dax@saidi-gwy01.meridiantelekoms.net) |
| 08:17.16 | Mavvie | unless there is something like calledid (not the be confused with callerid), I think you can have only one (1) number on an fso. |
| 08:17.18 | Mavvie | I could be wrong. |
| 08:18.54 | *** join/#asterisk Legend (~legend@adsl-69-107-115-138.dsl.pltn13.pacbell.net) |
| 08:20.59 | *** join/#asterisk pranav (pranav@203.115.90.246) |
| 08:21.18 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 08:21.34 | pranav | hello |
| 08:21.45 | datareactor | hello |
| 08:21.59 | pranav | need help to start gui for asterisk |
| 08:22.03 | pranav | can any one help |
| 08:22.22 | datareactor | what gui too you are using ? |
| 08:22.39 | pranav | astguicliant |
| 08:23.13 | pranav | i am totaly fresh for asterisk as well a linux |
| 08:23.20 | pranav | we managed to install asterisk |
| 08:23.25 | pranav | but for gui got stuck |
| 08:23.27 | jontow | good luck, and follow the SCRATCH_install document on the wiki WORD FOR WORD |
| 08:23.32 | jontow | its not easy |
| 08:23.47 | jontow | i'm well seasoned in unix/linux and pretty decent with asterisk.. and i had trouble ;) |
| 08:23.50 | datareactor | i have only experiecne with ACTOS http://www.ifrance.com/belikewater/code/actos.html |
| 08:24.18 | pranav | thanks |
| 08:24.29 | datareactor | ACTOS is quite easy to use |
| 08:24.33 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:24.43 | pranav | is it opensource? |
| 08:24.48 | jontow | yes |
| 08:24.58 | jontow | rather, astguiclient is.. don't know anything about ACTOS |
| 08:24.58 | pranav | ok |
| 08:25.04 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 08:25.21 | *** join/#asterisk Tiny_D (~Sascha@83.169.145.25) |
| 08:28.00 | pranav | thankx guy |
| 08:28.09 | pranav | will ready the link u gave me data |
| 08:28.41 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:29.43 | *** join/#asterisk Zeeek (~icechat5@84.4.11.61) |
| 08:29.53 | Guest^DJ | i set up a menu in extensions.conf and if caller did not key in anything. how do i get * to hangup ? |
| 08:30.03 | Guest^DJ | now it seems to loop forever |
| 08:30.48 | *** join/#asterisk Burglar (~ema@grog.scumm.it) |
| 08:31.39 | jontow | timeout? |
| 08:32.01 | pranav | hello Burglar |
| 08:32.37 | *** join/#asterisk helios4 (~utente1@host10-52.pool8175.interbusiness.it) |
| 08:33.36 | Guest^DJ | jontow: i wan * to repeat the menu again, if no respond, then * hangup |
| 08:34.10 | Zeeek | ("Hello World") |
| 08:34.18 | Guest^DJ | hi Zeeek |
| 08:35.34 | jontow | you need to learn variables then :) |
| 08:35.41 | jontow | and GotoIf() :) |
| 08:35.51 | jontow | SetVar() and GotoIf() are your keys.. |
| 08:35.53 | Guest^DJ | grrr |
| 08:35.58 | jontow | grrr? |
| 08:36.24 | Guest^DJ | jontow: ok will look into SetVar and Gotoif, tks for the tips |
| 08:36.48 | *** join/#asterisk Kumbang (~ecvs@167.205.22.54) |
| 08:37.24 | *** join/#asterisk KahiN (~DeatLeas@212.253.56.73) |
| 08:38.44 | *** join/#asterisk ptblank (~MURDER1@67.21.12.21) |
| 08:39.07 | Guest^DJ | is the * management portal software any good ? any one tried it yet ? |
| 08:39.14 | pranav | is there any place i can get more infor for gui setup? |
| 08:39.34 | RoyK | pranav: there's hardly anything available that works |
| 08:39.37 | jontow | exten => s,1,SetVar(LOOPME=0) |
| 08:39.38 | RoyK | pranav: why? |
| 08:39.42 | jontow | exten => s,2,playmenu |
| 08:40.16 | datareactor | pranav http://www.voip-info.org/tiki-index.php?page=Asterisk%20GUI |
| 08:40.48 | Guest^DJ | jontow: 2 lines can problem slove? |
| 08:41.40 | pranav | thankx |
| 08:41.45 | jontow | exten => s,3,GotoIf($[${LOOPME} = 1]?5) |
| 08:41.55 | jontow | exten => s,4,SetVar(LOOPME=1) |
| 08:42.00 | jontow | exten => s,5,Hangup |
| 08:42.00 | RoyK | ~pastebin |
| 08:42.02 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try pastebin.ca |
| 08:42.05 | RoyK | ~lart jontow |
| 08:42.09 | jontow | man |
| 08:42.11 | jontow | that wasn't a paste |
| 08:42.15 | jontow | thats all just typing |
| 08:42.16 | pranav | we need to setup gui for one clian with asterisk and we are totaly fresh with linux |
| 08:42.33 | Guest^DJ | jontow: looks compilcated but will try |
| 08:42.35 | RoyK | pranav: there isn't any |
| 08:42.36 | jontow | im not gonna go dump it into pastebin for 5 hand-typed lines :( |
| 08:42.51 | RoyK | jontow: it makes reading a lot easier if you do |
| 08:43.16 | Guest^DJ | pranav: i am only 5 days old with linux and *, more reading and it will really help |
| 08:43.16 | jontow | its an untested example.. in fact i'd be highly surprised if it works.. its just to give him the general idea of the logic behind it |
| 08:43.26 | jontow | i'd prefer he not copy/paste it anywhere ;) |
| 08:43.42 | Guest^DJ | jontow: i DID copy and paste in my extensions.conf and was abt to try |
| 08:43.47 | jontow | :P |
| 08:43.48 | jontow | damnit |
| 08:43.49 | jontow | heheh |
| 08:43.51 | Guest^DJ | hehhe |
| 08:44.03 | jontow | hey, if it works.. im glad.. i've never used SetVar or GotoIf ;) |
| 08:44.17 | jontow | thats just from a quick glance at the wiki (which you should be taking more than a quick glance at) |
| 08:44.53 | RoyK | pranav: please do a manual setup. you'll prolly get into it quite easily if you try hard enough :P |
| 08:45.19 | _TripleF_ | hehehe |
| 08:45.30 | _TripleF_ | this is whats it is being drunk |
| 08:45.37 | Guest^DJ | jontow: will keep as reference, tks |
| 08:45.52 | jontow | pranav; you're far better off knowing WHAT you have to do to setup an asterisk system than having something to do it for you.. it'll probably generate something braindead or overcomplicated and really push you away |
| 08:46.14 | jontow | the system itself is as simple as you wish to make it.. but if you want to do complicated things.. you'll see the results of those wishes quickly come to terms :) |
| 08:46.52 | pranav | thankx guys |
| 08:46.54 | Guest^DJ | pranav: i bought a book on * from ..... and i still think wiki have more info than the book |
| 08:47.02 | pranav | will be back with more question after i finish reading wiki |
| 08:47.03 | jontow | Signate Publishing? |
| 08:47.12 | Guest^DJ | jontow: yeah |
| 08:47.31 | jontow | not a bad book.. needs typos corrected and elaboration upona lot of things.. |
| 08:47.41 | jontow | but definitely consolidates a lot of ideas i couldn't hit on my own :) |
| 08:48.08 | pranav | can some one courier that book to me in india? |
| 08:48.11 | _TripleF_ | jontow: hey if you nedc ommercial quality u can look anywehere |
| 08:48.27 | _TripleF_ | * is GPL and open source |
| 08:48.32 | pranav | as its not available here |
| 08:48.39 | jontow | triplef; i'm just giving an opinion on the book.. * is top notch :) |
| 08:48.41 | _TripleF_ | that measn help as you can |
| 08:48.50 | _TripleF_ | hey i think too |
| 08:49.02 | jontow | i help wherever i can and whenever i can :) |
| 08:49.07 | _TripleF_ | just that i think quality is not enough to be production |
| 08:49.13 | Guest^DJ | buy ebook like i did, download it |
| 08:49.32 | jontow | i got a paper copy.. and its pencilled in all over the place where i took notes :) |
| 08:50.03 | _TripleF_ | u cant expklain to 10,000 user why teh 1800 DID what not connecting etc. and telco had strict rules |
| 08:50.17 | jontow | ;) you are drunk sir |
| 08:50.29 | _TripleF_ | who me ? |
| 08:50.29 | jontow | and i TOTALLY don't blame you |
| 08:50.57 | jontow | (the other guy with the crazy typing and drunken grammar ;)) |
| 08:51.16 | Guest^DJ | after reading materials from wiki, now the books is beginning to make sense to me |
| 08:51.21 | Guest^DJ | i can see the light now |
| 08:51.22 | Guest^DJ | lol |
| 08:51.25 | pranav | lol |
| 08:51.32 | _TripleF_ | ouch |
| 08:51.35 | _TripleF_ | i hear you |
| 08:51.35 | pranav | dj from where can i get that book |
| 08:51.40 | _TripleF_ | but i can't feal you |
| 08:51.53 | Guest^DJ | i got mine from www.osoft.com |
| 08:52.15 | pranav | thankx |
| 08:52.23 | RoyK | pranav: asteriskdocs.org has some good reading |
| 08:52.54 | pranav | ok |
| 08:52.57 | jontow | i daresay.. no one resource has it all |
| 08:53.00 | pranav | roy where u from |
| 08:53.08 | jontow | the whole point of the collective mind that * brings together.. |
| 08:53.10 | _TripleF_ | heeheh dialconfirm moc you there ? |
| 08:53.19 | _TripleF_ | just got tons of emails on your patch |
| 08:57.13 | pranav | roy where u from |
| 08:58.11 | *** join/#asterisk stuart_ (~schneis@70.57.124.226) |
| 08:58.22 | _TripleF_ | ??? |
| 08:58.28 | _TripleF_ | marc ?| |
| 08:58.29 | *** join/#asterisk RoyKa (~roy@80.239.107.80) |
| 08:58.57 | *** join/#asterisk TrisTan (Colgate@81.212.12.110) |
| 08:59.08 | _TripleF_ | anyone up or am i silent ? |
| 08:59.26 | stuart_ | I can read you TripleF |
| 08:59.27 | modulus_ | you're silent |
| 09:00.10 | stuart_ | Well, hi everyone - I'm a bit new to IRC, but have what's hopefully a quick * question that I haven't been able to find an answer for.... |
| 09:00.41 | _TripleF_ | hey hi all its 4am here |
| 09:00.43 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:00.46 | jontow | here too :/ |
| 09:00.55 | Guest^DJ | is 5pm here |
| 09:00.58 | _TripleF_ | stuart_: whats your q? |
| 09:00.59 | stuart_ | Only 1am here... ;-) |
| 09:01.18 | jontow | i wish it was 7am.. i'd be in bed. |
| 09:01.35 | stuart_ | I'm trying to setup my * system so that calls coming in from FWD automatically rings an extension on my SPA2000 |
| 09:01.55 | _TripleF_ | hehehe |
| 09:01.58 | *** join/#asterisk o7o (~Administr@S0106000f664eec84.vs.shawcable.net) |
| 09:01.58 | _TripleF_ | sap2000 ? |
| 09:02.04 | _TripleF_ | the fuck is that |
| 09:02.07 | _TripleF_ | a fxs ? |
| 09:02.10 | stuart_ | I've got that part working, but I'd like to find some way that I can abort the dial if the caller presses a number so that they could transfer to a specific extension |
| 09:02.20 | stuart_ | ...sorry SPA2000 from Sipura |
| 09:02.22 | _TripleF_ | sorry but im a noobies with knoledge |
| 09:02.28 | Zeeek | tih are you among us ? |
| 09:02.28 | stuart_ | Guess I can't type tonight |
| 09:02.38 | _TripleF_ | yesh change your batteries |
| 09:03.18 | stuart_ | The best I've found so far is to to be able to press * to cancel the dial |
| 09:03.36 | _TripleF_ | whatu wanto do ? |
| 09:03.40 | stuart_ | I was hoping for some way to be able to catch any DTMF similar to how Background() works |
| 09:03.53 | _TripleF_ | ok |
| 09:04.03 | _TripleF_ | u got waht as a card stuart_ |
| 09:04.14 | o7o | Hey everybody. o7o the noob here... looking for the development / API docs for asterisk.. is there a single uber source for this stuff? |
| 09:04.18 | jontow | "early dialing" is that called? |
| 09:04.26 | jontow | o7o; cvs.digium.com :) |
| 09:04.31 | stuart_ | Might be - I was one reference to that |
| 09:04.35 | o7o | thanks |
| 09:04.41 | RoyK | jontow: early B2? |
| 09:05.05 | jontow | rather, CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot ; cvs co asterisk zaptel libpri asterisk-addons asterisk-sounds |
| 09:05.15 | stuart_ | I've got it setup so that a call coming in rings my extension, but I can't abort the Dial() on a DTMF |
| 09:05.37 | jontow | that oughtta do the trick, o7o .. grab all sources from CVS HEAD branch.. the other end-all of API docs that I know of is the wiki.. update as you find more ;) www.voip-info.org |
| 09:05.41 | _TripleF_ | stuart pm me |
| 09:05.50 | Delvar | PING |
| 09:05.52 | WifiFred | PONG |
| 09:05.55 | RoyK | PANG |
| 09:05.58 | Delvar | lol |
| 09:06.21 | Guest^DJ | lol |
| 09:07.01 | Mavvie | I must be confused. I thought I saw 1.0.3 announced somewhere. |
| 09:07.20 | [Sim] | 1.0.3 is out there, yes |
| 09:07.31 | Mavvie | it is? then I must be really confused :-) |
| 09:07.51 | tzafrir | good morning (noon, whatever) |
| 09:07.51 | jontow | mavvie; i haven't seen a tarball.. but you can get it from the same place that i just referred o7o to.. |
| 09:08.02 | Zeeek | tzafrir hello |
| 09:08.13 | jontow | just use "-r v1-0" to get the 1-0 stable branch |
| 09:08.17 | Mavvie | there isn't a V1-0-3 tag in cvs.... |
| 09:08.21 | Mavvie | aha. |
| 09:08.36 | RoyK | Mavvie: don't use CVS if you need a stable server |
| 09:08.44 | RoyK | Mavvie: use 1.0.3 |
| 09:08.52 | RoyK | Mavvie: beleive me on that |
| 09:08.54 | Mavvie | RoyK: I'm not going for HEAD, I was looking for V1-0-3 in. |
| 09:08.57 | Mavvie | s/in.$ |
| 09:08.59 | jontow | royk; i was under the impression v1-0 was stable? |
| 09:08.59 | Zeeek | tzafrir do you do WIndows at all? (development question) |
| 09:09.05 | jontow | i am WELL aware that HEAD isn't stable ;) |
| 09:09.12 | jontow | but i've had no issues with -r v1-0 |
| 09:09.17 | tzafrir | Zeeek, not really. me personnaly: no |
| 09:09.22 | jontow | well.. save self-inflicted issues, of course ;) |
| 09:09.27 | RoyK | jontow: it's "stable", but all "stable" patches are put into CVS v1-0 and tested there before they are released |
| 09:09.45 | RoyK | jontow: beleive me. I had a server crash because of this the other day |
| 09:10.14 | tzafrir | anything you need from latest stable CVS? |
| 09:10.24 | Zeeek | tzafrir I asked because what you have done with asterisk (tho it is of no interest to me personally) would be great to do with a certain project I'm doing no on a WIn) |
| 09:10.31 | RoyK | tzafrir: prolly not |
| 09:10.55 | tzafrir | Zeeek, feel free to msg me |
| 09:11.02 | jontow | aha. |
| 09:11.38 | Zeeek | tzafrir nah, if you're not currently doing this kinda thing it prolly won't be of interest |
| 09:12.25 | Zeeek | just so you know it has to do with using Access databases in a PHP Intranet using the kind of thing I think you do, generating the configs etc. |
| 09:12.48 | Zeeek | I've never liked WIndows PHP devel but it is nec for some customers |
| 09:13.09 | *** part/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 09:13.10 | *** join/#asterisk ^HeLL^ (~admin@217.11.115.168) |
| 09:13.19 | ^HeLL^ | hello all |
| 09:13.23 | tzafrir | hrrmmm, not for me/us, I guess |
| 09:14.01 | tzafrir | which was not refering to the hello above |
| 09:15.47 | Mavvie | Somebody with a music server here? |
| 09:21.08 | ^HeLL^ | anybody knows, why, when I plug an Digium TDM400P card, to use with asterisk, when switch on the computer run fine, but if I do a software reboot it don't run... anybody happens something like that? |
| 09:21.31 | Moonwick | I think that's a known issue |
| 09:22.08 | ^HeLL^ | i was seeing a asterisk users list but i didnt find the solution |
| 09:23.24 | Moonwick | heh. the solution is to just flip the power switch. :) |
| 09:23.40 | *** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es) |
| 09:24.04 | _TripleF_ | hey |
| 09:24.22 | *** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62) |
| 09:26.14 | RoyK | ^HeLL^: reseat the PCI card, perhaps clean it. prolly that |
| 09:26.16 | Mavvie | hmmm... v1-0 doesn't support "next" in the dialplan. |
| 09:26.40 | *** join/#asterisk Bloody (~MiLitan@81.212.12.167) |
| 09:27.36 | Zeeek | HeLL by software rebott you mean stopping and starting asterisk ? |
| 09:29.07 | ^HeLL^ | Zeeek : software reboot stop and starting every apps |
| 09:29.20 | ^HeLL^ | doing 'reboot' on linux console |
| 09:30.20 | Zeeek | I've not had that problem |
| 09:30.27 | Zeeek | running 1.0.3 STABLE |
| 09:30.54 | RoyK | Zeeek: not from cvs, rait..... |
| 09:31.04 | Guest^DJ | guys, the SetVar in windows , is it the same as * command SetVar ? |
| 09:31.14 | *** join/#asterisk xbmodder (~xbmodder@adsl-67-119-13-192.dsl.snfc21.pacbell.net) |
| 09:31.19 | RoyK | Guest^DJ: What is windows? |
| 09:31.38 | ^HeLL^ | windows is an space on the wall |
| 09:31.38 | Guest^DJ | microsoft, hehehe |
| 09:31.41 | Guest^DJ | hahahha |
| 09:32.27 | Guest^DJ | i assume is different |
| 09:32.28 | RoyK | Window from old nordic language, Vindauge, meaning Vind == Wind and Auge == Eye, and eye towards the wind |
| 09:33.41 | *** join/#asterisk thefallen (PolarBear@thefallen.user) |
| 09:42.02 | RoyK | <PROTECTED> |
| 09:44.56 | *** join/#asterisk jas_williams (~jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 09:45.07 | jas_williams | morning |
| 09:45.29 | RoyK | <PROTECTED> |
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| 09:50.13 | *** mode/#asterisk [+o denon] by ChanServ |
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| 09:59.38 | _TripleF_ | ??? |
| 10:00.26 | _TripleF_ | .me see ny pillow sayinG FRANK COME GET ME |
| 10:05.36 | Zeeek | NEXT! |
| 10:06.34 | datareactor | anyone i have voip telephony and asterisk book soft copy |
| 10:07.17 | johngalt | could someone place a test call tome on iaxtel? |
| 10:07.25 | Zeeek | Number please? |
| 10:07.36 | clive- | john calll yourself |
| 10:07.45 | Zeeek | that's no fun |
| 10:08.15 | clive- | lol |
| 10:08.16 | johngalt | zeeek: 17002167766 |
| 10:08.31 | johngalt | ext 20 if u need it... |
| 10:08.32 | *** join/#asterisk phaze (~phaze@wl1-40.unil.ch) |
| 10:08.35 | Zeeek | is there an auto-answer? |
| 10:09.25 | shaZwaz | hi Zeeek , RoyK |
| 10:09.33 | johngalt | dont know just trying to get * setup for iaxtel and their instructions are somewhat vague |
| 10:09.58 | Zeeek | ringing? |
| 10:10.58 | Zeeek | I really need to get a way to click on a number and call it |
| 10:11.06 | fleeeb | anyone know of something besides festival that sounds more realistic |
| 10:11.19 | Zeeek | unfortunately, for complex technical reasons no one will undertand, that is very hard to do |
| 10:11.22 | RoyK | fleeeb: recorded speach |
| 10:11.27 | RoyK | s/each/eech/ |
| 10:11.35 | Zeeek | fleeeb - must it be free or will you pay? |
| 10:11.49 | fleeeb | well looking for dynamic |
| 10:11.50 | Zeeek | Free Speach is a famous idiot spammer ! |
| 10:11.54 | shaZwaz | Allison |
| 10:12.13 | Zeeek | fleeeb go listen to the demo of rhetoris |
| 10:12.18 | Zeeek | rhetoric |
| 10:12.24 | Zeeek | it's excellent |
| 10:12.53 | fleeeb | well im not looking for anything static - I can do that on my own |
| 10:13.15 | fleeeb | just looking for a dynamic solution that isnt robotic sounding if at all possible |
| 10:13.22 | Zeeek | feeeb I use another system but the voice costs $30 |
| 10:13.43 | fleeeb | rhetoric? |
| 10:13.57 | Zeeek | no that would be too expecsive |
| 10:14.16 | fleeeb | I wouldnt mind paying for a license if its decent |
| 10:14.24 | Zeeek | I can't remember the name ! |
| 10:14.28 | Zeeek | let me look |
| 10:14.30 | fleeeb | hehe ok |
| 10:15.27 | Zeeek | ah the bill.... Cepstral |
| 10:15.32 | viper-net | . |
| 10:15.38 | Zeeek | they have several voices avail |
| 10:15.52 | Zeeek | it works well but it a little mechanical sounding |
| 10:16.14 | Zeeek | not sure if better than festival but I heard so many negative remarks about festival I never installed it |
| 10:16.21 | *** join/#asterisk ebell_ (eric@CPE-65-30-220-56.wi.rr.com) |
| 10:16.34 | Zeeek | you can hear the demo on the web - also comes with sound effects |
| 10:16.57 | Zeeek | http://www.cepstral.com/cgi-bin/demos/general |
| 10:17.54 | Zeeek | while talk is quiet here, for the record I have had several mails from support at Grandstream about my DNS problem. At least they answer ! |
| 10:18.01 | fleeeb | hmm |
| 10:18.04 | fleeeb | yeah I've seen them before |
| 10:18.19 | fleeeb | I guess thats the best that you can get aside from recording it yourself |
| 10:18.26 | Zeeek | fleeeb I use "Dave" I thionk it's decent for the price but far from perfect |
| 10:20.54 | *** join/#asterisk Command (NoTice@81.212.173.50) |
| 10:22.44 | *** part/#asterisk datareactor (datareacto@203.81.192.33) |
| 10:22.59 | *** join/#asterisk johngalt (1000@dsl081-088-238.lax1.dsl.speakeasy.net) |
| 10:23.16 | johngalt | damn hardware! |
| 10:23.58 | johngalt | how do I specify where incomming iaxtel calls go? |
| 10:24.31 | Zeeek | context ? |
| 10:24.34 | Zeeek | Starter tutorial: |
| 10:24.34 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 10:24.34 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 10:24.34 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 10:24.34 | Zeeek | http://www.fnords.org/~eric/asterisk/ |
| 10:25.58 | johngalt | been there but how does an incomming call find the context to begin in? specificly iax? |
| 10:26.30 | Zeeek | how can you have read those docs and not know the answer which is clearly stated in all of the above? |
| 10:26.34 | johngalt | I seem to be missing how a given call finds it's context |
| 10:26.50 | Zeeek | which is why you should look at the first article |
| 10:27.03 | Zeeek | it's purpose is to give that info and that is what it does |
| 10:27.20 | johngalt | I'll look again |
| 10:28.39 | shaZwaz | I am getting some interference (like on radio) on my analog exts, any idea Zeeek |
| 10:30.39 | Zeeek | describe further please? |
| 10:31.43 | shaZwaz | like what ? |
| 10:32.11 | Zeeek | doctor, I am sick |
| 10:32.27 | Zeeek | what's wrong? It's my arm. |
| 10:32.34 | Zeeek | what about your arm |
| 10:32.38 | shaZwaz | its a analog extension connected to channel bank (Adit 600) |
| 10:32.39 | Zeeek | it hurts when I do this. |
| 10:32.45 | Zeeek | "So don't do this" |
| 10:33.21 | Zeeek | I'm not famil at all w channel banks but maybe some kind soul will read this |
| 10:34.30 | shaZwaz | don't know whats causing this, may be the rx /tx again need adjustment or may be its the electrical interference |
| 10:34.31 | Zeeek | tih ? |
| 10:34.44 | Zeeek | got a noise suppressor ? |
| 10:34.53 | shaZwaz | yes |
| 10:35.16 | shaZwaz | I have already enabled the aggressive supressor |
| 10:35.22 | Zeeek | any way to stop nearby stuff to check if that's the cause? |
| 10:35.46 | Zeeek | sometimes video cards cause noise - you could try changing or moving the video |
| 10:35.58 | Zeeek | obviously IRQ sharing has been checked |
| 10:36.16 | Zeeek | power supplies sometimes also generate noise on the buss |
| 10:36.23 | shaZwaz | the video card is built in |
| 10:36.41 | Zeeek | that's about what I remember from my audio workstation days :) |
| 10:36.56 | Zeeek | stupid idea here but try turning off the monitor |
| 10:36.57 | *** join/#asterisk Miker (~mike@201.129.119.248) |
| 10:37.04 | sxpert_work | the EPIA 800 is known for having bus noise issues. |
| 10:37.14 | sxpert_work | the M series doesn't have those anymore |
| 10:37.24 | shaZwaz | I will start with lowering the rx/txgains |
| 10:37.30 | Slimey | Anyone aware of a linux compatible DECT module? (USB preferably) |
| 10:37.31 | johngalt | Zeeek: what I seem to be missing is that the article covers sip and I have zap channels and iax...no mention is made of getting iax working and I do not see where I'd specify which context iax calls are routed to |
| 10:37.51 | shaZwaz | Zeeek: no monitor attached |
| 10:38.30 | Zeeek | johngalt pretend it's SIP. Further, there are smaple files in the asterisk source/config directory that have all the possible params |
| 10:38.55 | Zeeek | for x sake, context is the most basic concept to master - you won't get far without understanding it, believe me |
| 10:39.28 | Zeeek | shaZ - absolutely no idea then.. except... have you a cellphone nearby ? |
| 10:40.11 | *** join/#asterisk Manipura_ (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 10:40.31 | Zeeek | http://asteriskdocs.org : if you read this through completely and English is your first language and you don't understand context and dialplan, you will need a consultant to use asterisk |
| 10:40.48 | *** join/#asterisk macmx (~mike@201.129.119.248) |
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| 10:41.29 | johngalt | I think I get context - it's the mapping I don't seem to get...isn't that setup in the iax.conf file - but I don't see it there? |
| 10:41.49 | *** join/#asterisk [1]JohnJacob (~JohnJacob@pcp619824pcs.mainf01.in.comcast.net) |
| 10:42.15 | Zeeek | there are many iax.conf examples out there, a good one is provided to you with iaxtel already configured |
| 10:42.27 | Zeeek | but context is used in all channels |
| 10:42.39 | Zeeek | it isn't tied to SIP |
| 10:43.11 | Zeeek | a call comes in in a context. There is a context by default and a context for every peer and freind entry (or it will use default) |
| 10:43.36 | Zeeek | this is exactly what the 4000 web pages about asterisk say in the first few paragraphs of getting started |
| 10:44.02 | *** join/#asterisk _Poincare (~jefffnode@dD5779B07.access.telenet.be) |
| 10:44.55 | Zeeek | anyone remember my question about the 10 minutes of frenzyied iax errors every day? |
| 10:45.48 | *** join/#asterisk shaZwaz_ (~luckyali@203.81.196.167) |
| 10:46.32 | Zeeek | hey, this is interesting;: http://www.voip-info.org/wiki-Asterisk+config+iax.conf |
| 10:47.35 | *** join/#asterisk nextime (~nextime@danex.i-m-c.it) |
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| 10:47.35 | Zeeek | . |
| 10:47.35 | *** join/#asterisk kevinl (~klindsay@staff-nat.netnation.com) |
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| 10:53.36 | johngalt | is the context that iaxtel calls are branched to defined in iax.conf? yes I know in the sample files that are provided branch to the default context...but is this hard coded or what? I'm not seeing quite where it's specified |
| 10:53.42 | *** join/#asterisk finnjet (~finnjet@port-212-202-29-12.dynamic.qsc.de) |
| 10:53.59 | johngalt | (incomming iaxtelcalls) |
| 10:54.11 | Zeeek | it doesn't seem possible... |
| 10:56.53 | Zeeek | Last attempt: johngalt : http://www.loligo.com/asterisk/current/iax.conf |
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| 11:10.44 | ^Slash | hey guus, is there anyway to control the level of volume for the hold music ?? mine sounds like somone wants to blow some shit up... |
| 11:10.52 | ^Slash | about 500% way too loud |
| 11:10.59 | ^Slash | yet the mp3 is normal |
| 11:11.22 | Zeeek | take a look in musiconhold.conf |
| 11:11.31 | ^Slash | i have |
| 11:11.35 | ^Slash | that variable quietmp3 |
| 11:11.40 | Zeeek | isn't there something about "loud" |
| 11:11.41 | ^Slash | isnt realy a volume control |
| 11:11.51 | ^Slash | yeah i dont want it loud |
| 11:11.54 | Zeeek | what are you listenting on? |
| 11:12.12 | ^Slash | what do you mean ? |
| 11:12.19 | Zeeek | ZAP? SIP ? |
| 11:12.30 | ^Slash | somone calls me i put them on hold and they have to take the earpiece away coz its 500x way oo loud and distorted |
| 11:12.33 | ^Slash | both |
| 11:12.35 | ^Slash | ZAP and SIP |
| 11:12.55 | ^Slash | these are the stock mp3s that it came with |
| 11:12.58 | ^Slash | which i tested and sound fine |
| 11:13.08 | Zeeek | it's lous on both? I think the music file is too maximized (to make it sound loud) |
| 11:13.22 | ^Slash | plays fine on my px |
| 11:13.23 | ^Slash | er pc |
| 11:13.27 | ^Slash | normal mp3 |
| 11:13.28 | Zeeek | ah the mp3s it came with - I'm not sure why that would be |
| 11:13.36 | ^Slash | yep... |
| 11:13.45 | Zeeek | there is something in the mailing list about native player ? |
| 11:13.47 | ^Slash | im using mpg123 |
| 11:14.00 | ^Slash | im using the recomended player |
| 11:14.01 | Zeeek | search the list something just went by about this |
| 11:14.04 | Zeeek | ok |
| 11:14.06 | ^Slash | oh ? |
| 11:14.10 | Zeeek | the native thingie? |
| 11:14.17 | ^Slash | just thought it was odd why i couldnt change the volume |
| 11:15.20 | Zeeek | I know little about these things on linux - you'd think there would be a command line param to *lower* volume of a flie for the player |
| 11:15.36 | Zeeek | a lot of people say it's not loud enuf |
| 11:15.47 | Zeeek | s/flie/file/ |
| 11:16.51 | ^Slash | not loud enuf ? lol |
| 11:17.29 | Zeeek | well, with a name like Slash - I thought you'd want it ear-bleeding loud ? |
| 11:17.32 | ^Slash | i should setup a sip client for you and you dial my place youll be crying in pain from the volume |
| 11:17.38 | ^Slash | hahah |
| 11:17.40 | ^Slash | no |
| 11:17.49 | Zeeek | gimmie a number to call - you have iaxtel? |
| 11:17.59 | ^Slash | nope |
| 11:18.04 | ^Slash | just set the server up |
| 11:18.09 | Zeeek | well, that's what it's for |
| 11:18.12 | ^Slash | messing with it for a business |
| 11:18.15 | Zeeek | or maybe not... |
| 11:18.25 | Zeeek | muhahahaha business |
| 11:18.31 | Zeeek | Bizness ??? |
| 11:18.51 | Zeeek | "asterisk means business" |
| 11:19.17 | PoWeRKiLL | Hi |
| 11:19.24 | ^Slash | eh |
| 11:19.25 | ^Slash | ? |
| 11:19.26 | Zeeek | shit I'm starved and no food here. I'm gonna have to... hav to go... oh... out into meat space |
| 11:19.45 | PoWeRKiLL | is it possible to connect a avaya argentphone to asterisk ? |
| 11:19.58 | Zeeek | Slash while I'm out, set up an iaxtel number and I'll call |
| 11:20.14 | Zeeek | or even a guest SIP or IAX |
| 11:20.14 | *** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk) |
| 11:20.35 | ^Slash | yeah guest sip will do |
| 11:20.45 | ^Slash | let me pfw on the router |
| 11:20.53 | DrPete | YOu see i have one phone line, and i want to recieve voice and faxes on that one line. I was hoping that my linux box could answer, reiceve a fax if it was a fax, or allow me to answer if it was voice |
| 11:21.07 | DrPete | is this possible? |
| 11:21.27 | DrPete | Sorry, if it reads funny, I pasted it from another channel |
| 11:22.35 | ^Slash | hmm |
| 11:22.41 | ^Slash | im sure it can be done |
| 11:22.54 | ^Slash | im new to the asterisk game so cant help :-/ |
| 11:23.05 | ^Slash | im sure a macro has been done for it tho google is ya friend :) |
| 11:23.14 | Zeeek | but like a dog walking on its hind legs, it isn't done well |
| 11:23.20 | *** join/#asterisk qwerp (~abc@219.93.57.62) |
| 11:23.25 | qwerp | harlo... |
| 11:23.26 | Zeeek | [haven't got out the door yet] |
| 11:23.41 | ^Slash | what port is needed for sip ? |
| 11:23.42 | Zeeek | DrPete look up spandsp |
| 11:23.46 | Zeeek | 5060 |
| 11:23.52 | qwerp | is there anyway that i can play MOH music as a background music? |
| 11:23.53 | DrPete | Zeeek: ooh oki |
| 11:24.12 | Zeeek | qzerp you mean like all the time? |
| 11:25.20 | qwerp | no.. |
| 11:25.33 | qwerp | is like i play a recording like.. |
| 11:25.46 | qwerp | XXXXXXX if u know the person's ext, please dial now. |
| 11:26.02 | qwerp | then it starts to play moh music while waiting for caller to dial ext |
| 11:26.13 | DrPete | Zeeek: does it work well? Can windows XP cleints send and rec faxes though a the server easliy? |
| 11:26.54 | *** join/#asterisk BuzzBud (~adoroar@90.62-97-254.bkkb.no) |
| 11:28.02 | qwerp | its likw replacing Background(XXX) with MOH but still allow caller to dial an ext |
| 11:32.36 | Zeeek | who had the caramel chicken ? |
| 11:32.47 | ^Slash | guest sip |
| 11:32.50 | ^Slash | setup |
| 11:33.15 | Zeeek | greasy fingers from samoussa can't dial |
| 11:34.22 | Zeeek | qwerp that's just a question of taking a moh file and mixing in the announcement |
| 11:34.25 | ^Slash | user guest |
| 11:34.28 | ^Slash | ip |
| 11:34.40 | ^Slash | 220.245.163.107 |
| 11:34.43 | ^Slash | no pass |
| 11:34.44 | ^Slash | no nothing |
| 11:34.49 | ^Slash | dial 6789 |
| 11:34.54 | ^Slash | and enjoy the BOOOM |
| 11:35.13 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 11:35.13 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:35.37 | ^Slash | hopefully its going to work via the nat |
| 11:36.29 | Zeeek | Not found slash |
| 11:36.36 | ^Slash | yeah ? |
| 11:36.40 | ^Slash | 5060 UDP ? |
| 11:36.42 | ^Slash | thats the port |
| 11:37.13 | Zeeek | yes |
| 11:37.19 | Zeeek | maybe tcp too |
| 11:37.26 | ^Slash | hmmm |
| 11:37.33 | ^Slash | netstat dont show tcp |
| 11:37.34 | ^Slash | only udp |
| 11:38.00 | ^Slash | listening on 0.0.0.0 |
| 11:38.21 | Zeeek | I called guest@ that ip |
| 11:38.29 | ^Slash | called ? |
| 11:38.34 | ^Slash | i thought you were going to register |
| 11:38.41 | ^Slash | to the sip serv |
| 11:38.48 | ^Slash | then dial ext 6789 |
| 11:38.53 | Zeeek | no you need to set up a default context for guests |
| 11:39.06 | ^Slash | oh ? |
| 11:39.11 | ^Slash | why cant you just reg |
| 11:39.12 | ^Slash | heh |
| 11:39.14 | Zeeek | why would I register to call you? YOu screwing Jennifer Anston or something? |
| 11:39.45 | ^Slash | how hard is it to put in username guest, ip and pick up your phone and dial 6789 ?: |
| 11:39.51 | ^Slash | ok so what should i do then |
| 11:40.01 | Zeeek | when you're eating chinses food at the same time, HARD |
| 11:40.07 | ^Slash | lol |
| 11:40.33 | Zeeek | first call Jennifer Anston and see if she's interested |
| 11:40.39 | *** join/#asterisk phaze (~phaze@pcbcu420a.unil.ch) |
| 11:41.43 | Zeeek | it's prolly a good idea though to set something up in a default context, then comment it out after tests are over |
| 11:42.08 | ^Slash | i made a guest context for that sip user |
| 11:42.31 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 11:42.40 | Zeeek | Executing Dial("Zap/4-1", "SIP/guest@220.245.163.107|15") in new stack |
| 11:42.40 | Zeeek | <PROTECTED> |
| 11:42.40 | Zeeek | <PROTECTED> |
| 11:42.56 | ^Slash | call grandtsream1 |
| 11:43.02 | Zeeek | what do y u see on CLI? |
| 11:43.13 | ^Slash | i see nothing come up |
| 11:43.14 | Zeeek | exit |
| 11:43.21 | ^Slash | only when i pick up the phone |
| 11:43.23 | Zeeek | heh |
| 11:44.05 | ^Slash | so what should i have in my sip.conf |
| 11:44.18 | Zeeek | your guest entry isn't working but worse, if you don't see the call on CLI it ain't getting there |
| 11:44.44 | Zeeek | you have a "default" context in the general section ? |
| 11:44.51 | ^Slash | you got a 404 from my serv |
| 11:44.57 | PoWeRKiLL | is it possible to connect a avaya argentphone to asterisk ? |
| 11:45.00 | ^Slash | which file we looking at |
| 11:45.14 | Zeeek | sip.conf |
| 11:45.17 | ^Slash | k |
| 11:45.20 | Mavvie | jbot: stun? |
| 11:45.22 | jbot | hmm... stun is that feeling you get when you realise your SIP call actually got through! |
| 11:45.32 | Mavvie | hahaha. |
| 11:45.33 | ^Slash | yep context=default |
| 11:45.53 | Mavvie | jbot: STUN? |
| 11:45.54 | jbot | it has been said that stun is that feeling you get when you realise your SIP call actually got through! |
| 11:45.57 | Mavvie | *growl* |
| 11:45.59 | Zeeek | I'm trying to eat greasy chinese food on the desk of the girl with nice legs who will be here soon |
| 11:46.23 | Zeeek | I spilled some on my crotch and have a big grease stain there now |
| 11:47.09 | ^Slash | Zeeek... |
| 11:47.16 | Zeeek | so then if SIP calls are supposed to "default" in extensions.conf, look in extensions.conf... |
| 11:47.44 | ^Slash | yep |
| 11:47.44 | ^Slash | done |
| 11:47.44 | Zeeek | I could maybe put the candied ginger on my crotch and say "work on this" |
| 11:47.56 | ^Slash | ive included the context guest |
| 11:47.59 | ^Slash | [guest] |
| 11:48.01 | ^Slash | which has |
| 11:48.09 | Zeeek | shit what if she's on IRC here now watching (looks at list) |
| 11:48.17 | ^Slash | exten => 6789,1,MsuicOnHold(default_ |
| 11:48.20 | Zeeek | WAIT WAIT |
| 11:48.25 | ^Slash | ) |
| 11:48.32 | Zeeek | STOP |
| 11:48.36 | ^Slash | . |
| 11:48.49 | Zeeek | besides the fact that I doubt MsuicOnHold will do a lot... |
| 11:48.54 | ^Slash | hah |
| 11:48.56 | ^Slash | no typo |
| 11:49.10 | ^Slash | im typing 1 hand while on phone |
| 11:49.20 | ^Slash | MusicOnHold(default) |
| 11:49.21 | Zeeek | look again at sip.conf and consider that you have a line in general that say context= |
| 11:49.35 | ^Slash | yeah all makes sense |
| 11:49.36 | Zeeek | and a the right of the equal sign is a word |
| 11:49.45 | ^Slash | default |
| 11:49.47 | Zeeek | and that word is..... default |
| 11:49.50 | Zeeek | YES |
| 11:50.01 | ^Slash | and in my default in extensions.conf i have include guest |
| 11:50.03 | ^Slash | and under guest |
| 11:50.03 | Zeeek | default by default - logical and easy to remember/understand |
| 11:50.05 | shaZwaz_ | ~ping |
| 11:50.08 | jbot | pong |
| 11:50.08 | ^Slash | i have the line above |
| 11:50.18 | ^Slash | so you should call 6789@my_ip |
| 11:50.21 | ^Slash | correct ? |
| 11:50.32 | *** join/#asterisk Nuttah (~andrew@amber.interdart.co.uk) |
| 11:51.05 | Zeeek | I have to go clean my crotch or face serious sexual harassment charges |
| 11:51.22 | Zeeek | sine I am now consuming all the ginger |
| 11:51.27 | ^Slash | :-/ |
| 11:52.00 | *** join/#asterisk NeroLab1 (~Andrew@c-24-19-145-211.client.comcast.net) |
| 11:52.10 | *** join/#asterisk Specky[W] (~sspecken-@p508ED9FC.dip0.t-ipconnect.de) |
| 11:52.26 | *** part/#asterisk Specky[W] (~sspecken-@p508ED9FC.dip0.t-ipconnect.de) |
| 11:52.43 | Zeeek | FUCK THAT IS LOUD! |
| 11:52.47 | ^Slash | says started music on hold :) |
| 11:52.49 | DrPete | whats a zap channel? |
| 11:52.49 | *** part/#asterisk NeroLab1 (~Andrew@c-24-19-145-211.client.comcast.net) |
| 11:52.52 | ^Slash | haha thats default hahah |
| 11:52.56 | Zeeek | DrPete? |
| 11:53.00 | Zeeek | Starter tutorial: |
| 11:53.00 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 11:53.00 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 11:53.01 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 11:53.01 | Zeeek | http://www.fnords.org/~eric/asterisk/ |
| 11:53.12 | Zeeek | ok Slash, you are not crazy |
| 11:53.23 | ^Slash | heh |
| 11:53.25 | ^Slash | so no answers ? |
| 11:53.33 | Zeeek | is there any way you can play the files? You have a sound card and all? |
| 11:53.46 | ^Slash | yeah i played em on the pc |
| 11:53.48 | ^Slash | they play fine |
| 11:54.01 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 11:54.12 | Zeeek | brb |
| 11:55.04 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 11:56.55 | *** join/#asterisk NeroLab1 (~Andrew@c-24-19-145-211.client.comcast.net) |
| 11:58.37 | DrPete | Zeeek: man, all this pbx stuff is complicated. I have found a site that states asterisk can be used as a fax/voice switch. however it talks about it can send the fax to a fax entention, if its a zap channel. I just wondered if i can do this will a modem and a linux box. If its using extensions I imagine I am gonna need more hardware :( |
| 11:58.39 | *** join/#asterisk Logan (~logan@planetmath.cc.vt.edu) |
| 11:58.59 | Zeeek | You are also gonna need to do a lot of reading |
| 11:59.24 | Zeeek | but in a word, my own experience with fax is that it won't be consistent enouf for porfessional use |
| 11:59.49 | Zeeek | and you can't just use a modem with asterisk |
| 11:59.57 | Zeeek | so yes, you'll need some hardware |
| 12:00.22 | Zeeek | try to read that first article, it's a very good intro to asterisk and voIP |
| 12:00.33 | Zeeek | the second one tells how to hook to the phone line |
| 12:01.29 | fafnir | say i have a modem on my computer, that in windows i could make a phone call through it, and now i have linux install, think itll wokr with asterisk and voip so i can make outgoing cals through voip? just at a guess, i dont need anything definitive |
| 12:01.43 | Zeeek | No. |
| 12:01.57 | DrPete | Zeeek: what am I looking at minimum hardware cost would you think? where can I buy hardware? can you surgest anything cheap, heh |
| 12:02.12 | Zeeek | for pro use or hobby? |
| 12:03.16 | Zeeek | fafnir look up asterisk modem on google. You will see a lot of info about your question |
| 12:03.41 | Zeeek | DrPete if you are willing to forget fax, there are a lot of solutions |
| 12:03.47 | DrPete | Zeeek: its for a small business, i send and rec a few faxes and I have one phoneline and just a single phone, so its nothing crazy for now |
| 12:03.58 | *** part/#asterisk qwerp (~abc@219.93.57.62) |
| 12:04.14 | Zeeek | DrPete my honest free advice? Buy a cheap fax or better yet PC software |
| 12:04.47 | shaZwaz_ | ~ping |
| 12:04.48 | jbot | pong |
| 12:04.49 | Zeeek | for the phone, there are official Digium cards and there are clones starting at $10 |
| 12:06.23 | *** part/#asterisk NeroLab1 (~Andrew@c-24-19-145-211.client.comcast.net) |
| 12:06.53 | ^Slash | so zeeek any ideas ? |
| 12:06.55 | DrPete | Zeeek: I use PC software at the moment, and it workes well. I however when I rec a call, i pick up the phone and if its a fax I have to be there at the PC to click answer. I wanted a solution that would run on one of my linux servers that could detect if it was a fax and recive it. If not the phone would ring and I could pick it up if I was there. Does this make sense? |
| 12:07.08 | *** join/#asterisk Astinus_ (~abba@213.167.111.138) |
| 12:07.18 | ^Slash | not possible |
| 12:07.23 | ^Slash | unless it answers it first |
| 12:07.30 | ^Slash | detects the beep beep from the other fax |
| 12:08.08 | Zeeek | DrPete - I've been wanting to do the same forever. The fax stuff is called spandsp - it works great with some fax machines but not others. In a business context, that won't do at all |
| 12:08.09 | ^Slash | then if no beep it will transfer so the guy on the other end will pay for the call even if you aint home |
| 12:08.22 | fafnir | theres a service that does two numbers with different rings for one phone line |
| 12:08.36 | Zeeek | You can't tell an important client, "my fax software didn't get it, will you try again?" |
| 12:08.43 | ^Slash | lol |
| 12:08.46 | ^Slash | get 2 lines |
| 12:08.53 | Zeeek | DrPete - there are fax to email services that work PERFECTLY every time |
| 12:09.02 | DrPete | Zeeek: thats a shame, could I use asterisk as the switch and use say halyfax to get the faxes? |
| 12:09.12 | Astinus_ | hmm, does modems have anything to do with asteriks, i mean can a modem be used to receive calls from outside world? |
| 12:09.12 | Zeeek | They would use a different umber though |
| 12:09.31 | Zeeek | DrPete - ask that question on the mailing list, several people are doing just that |
| 12:09.48 | *** part/#asterisk Kumbang (~ecvs@167.205.22.54) |
| 12:09.56 | shaZwaz_ | DrPete: Zeeek is right |
| 12:09.57 | DrPete | Zeeek: Hmm, fax to email that would work, can asterisk do that? |
| 12:10.00 | Zeeek | Astinus sure a modem can dial and answer the pohone |
| 12:10.10 | ^Slash | im over this music |
| 12:10.11 | ^Slash | lol |
| 12:10.12 | Zeeek | DrPete yes but the faxes don't always work |
| 12:10.17 | ^Slash | sounds like death metal |
| 12:10.24 | shaZwaz_ | I have succeded with receiving faxes but the quality is not good |
| 12:10.31 | DrPete | Zeeek: oh right I see. |
| 12:10.43 | Zeeek | Slash the only concrete solution I know is to find and use softer music |
| 12:10.51 | DrPete | There must be a work around for fax and voice on the same line :( |
| 12:10.54 | ^Slash | yeah thats what i figured |
| 12:11.05 | Zeeek | the most important point is, the fax is not always ever received readable |
| 12:11.07 | ^Slash | but fore the default stuff not to work is kinda sad :-/ |
| 12:11.20 | Zeeek | DrPete there are fax machines for $50 that will do this |
| 12:12.36 | shaZwaz_ | ~seen n3o |
| 12:12.38 | jbot | n3o <~N3O@202.142.189.86> was last seen on IRC in channel #asterisk, 26d 20h 58m 31s ago, saying: 'use pager email for 2nd option docelmo'. |
| 12:12.52 | maruz | there someone present now with asterisk realtime mysql installed and running? my * hungs during authen |
| 12:12.54 | DrPete | Zeeek: heh, if thats the case, linux should do it for free, heh. I hate the idea of wasting desk space and paper on a fax machine, working with the PC is much better. also I want to make up some fax mailing lists and send a loads of flyers out etc |
| 12:14.39 | mikegrb | DrPete: the unreliable bit is software emulation of a fax modem, you can put a real fax modem in and hook it to an fxs and send faxes to it's extention |
| 12:15.04 | mikegrb | the zap cards will even autodetect fax tones and tell asterisk so asterisk can forward to the right extention automatically |
| 12:15.25 | Astinus_ | hmm not many books on amazon for asterisk |
| 12:16.39 | DrPete | mikegrb: hmm, oki. I have a fax modem in the server, I was planning to use that on the phone line. Wont that work with asterisk? |
| 12:17.23 | mikegrb | dunno |
| 12:18.53 | Zeeek | Astinus there are many free ones though |
| 12:18.55 | Zeeek | Starter tutorial: |
| 12:18.55 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 12:18.55 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 12:18.55 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 12:18.55 | Zeeek | http://www.fnords.org/~eric/asterisk/ |
| 12:19.02 | Zeeek | http://asteriskdocs.org |
| 12:19.19 | Zeeek | mikegrb excellent point although it will be a little pricey |
| 12:20.01 | DrPete | mikegrb: to have different extesions wont I need hardware to plug it all into? |
| 12:20.35 | Astinus_ | Zeeek: i'm having trouble getting graps of consepts like pots, ptsn, iax, sip, all the codecs t1/e1 fxo fso .. |
| 12:21.04 | Zeeek | Astinus which is why you need to read through some of that material |
| 12:21.45 | Zeeek | ~seen tih |
| 12:21.46 | jbot | tih is currently on #asterisk (3d 4h 17m 59s). Has said a total of 57 messages. Is idling for 22h 32m 34s |
| 12:26.17 | DrPete | I have a hayes optima external business modem, it says it supports class 1 and class 2 fax, I must be able to use this somehow :( |
| 12:26.37 | Zeeek | you can but not with asterisk |
| 12:26.49 | shaZwaz_ | passthru |
| 12:27.28 | BoRiS | G711u with *very* low latency and no packetloss |
| 12:27.54 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 12:28.09 | DrPete | Zeeek: Oh i see what you are saying now, to use asterisk you need a special modem, and that is not a fax modem, and therefore the problem, ahh |
| 12:28.10 | bowman | question. ChanIsAvail(SIP/phone3) is correct regarding syntax, right? |
| 12:28.28 | Zeeek | now I see how you made it thru med school, Dr Pete |
| 12:31.13 | DrPete | Zeeek: heh, yeah. Its all new to me. Well I have found some of them cards on ebay, they are only 10 pounds GBP, so thats good. Ok, what about this. I have both on the same server, then when a call is answered asterix detects the call and puts in though to the faxmodem that answers the call with fax software, sound good? My only question is what hardware do I need to have different exensions |
| 12:31.27 | Zeeek | I don't think that's gonna work |
| 12:31.47 | Zeeek | for the hardware question you need a TDM400P |
| 12:31.58 | Zeeek | look on http://digium.com |
| 12:32.39 | Zeeek | oops requires the www |
| 12:32.40 | DrPete | aww how come it wont work, lol |
| 12:32.47 | Zeeek | here's the TDM400P: http://www.digium.com/index.php?menu=wildcard_tdm400p2 |
| 12:33.22 | Zeeek | you'd need one FXO for the phone line and as many FXS as you want extensions |
| 12:33.40 | Zeeek | you're talkin a couple hundred pounds here |
| 12:33.49 | DrPete | lol what, man |
| 12:34.15 | Zeeek | $313 for 3FXS and one FXO |
| 12:34.34 | Zeeek | and it helps pay for the free software |
| 12:35.09 | Zeeek | plus shipping, taxes and possibly customs |
| 12:35.16 | Zeeek | batteries not included |
| 12:35.25 | Zeeek | not recommended for kids over 15 |
| 12:35.25 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 12:35.59 | DrPete | heh, yah I see the one, anyone want to donate one to me :) |
| 12:36.40 | Zeeek | If you have need of asterisk I again recommend you do some reading about it |
| 12:36.52 | Zeeek | it can do almost anything |
| 12:37.12 | Zeeek | but requires some understanding to accomplish what you want (or the help of a consultant) |
| 12:39.02 | *** join/#asterisk CybreWulf (cybre@killcybre.org) |
| 12:39.05 | DrPete | Zeek I think i will have to read more. Yeah, I wanted to get an idea if what I wanted to do was possible, then take it from there. I dont suppose anyone knows how many rings a standard fax machine will wait for and answer? |
| 12:39.10 | *** join/#asterisk Thumann (Bush@is.a.retard.dk) |
| 12:39.21 | Zeeek | it's usually possible to configure them |
| 12:39.28 | Zeeek | minimum is often 2 rings |
| 12:40.03 | Zeeek | I'm sure someone make a fax/answeringmachine + cordless phone |
| 12:40.11 | Zeeek | that'd do it all |
| 12:40.41 | Zeeek | you gotta love the names: |
| 12:40.42 | Zeeek | From: "Jungfrau K. Overweening" <antoney@eyesonff.com> (Rr one 1 message from kally awaits you aback) |
| 12:43.40 | DrPete | Zeeek: yeah, I know what you mean. I am sure you can buy a small bit of hardware that plugs in to the phone line and acts as a fax switch, u basically plug your phone and fax into it. That would work, buts its a shame linux dont make one |
| 12:44.24 | Zeeek | it's a shame hundres of thousands of people were killed too |
| 12:44.44 | *** join/#asterisk doughecka1 (~Doug@adsl-18-116-90.sdf.bellsouth.net) |
| 12:44.55 | Zeeek | you've spent more time here than the lousy $200 it'd cost to buy what you need |
| 12:46.11 | DrPete | Zeeek: Hmm, you could say that |
| 12:46.20 | tzanger | he just did say that |
| 12:46.28 | Zeeek | gosh what kind of business could it be - dentist ? |
| 12:46.48 | tzanger | hell I use app_rxfax to receive beteween 30 and 50 faxes a day with no problem |
| 12:47.14 | *** part/#asterisk Burglar (~ema@grog.scumm.it) |
| 12:47.21 | *** join/#asterisk Logan (~logan@planetmath.cc.vt.edu) |
| 12:47.25 | tzanger | TDM430P to connect two fax machines and a stamp machine |
| 12:47.40 | Zeeek | you use app fax as the switch? |
| 12:47.49 | tzanger | no but I could |
| 12:47.59 | Zeeek | actually isn't it asterisk that detects? or rather zaptel? |
| 12:48.00 | tzanger | I use app_fax to pull in faxes from our fax DID |
| 12:48.19 | Zeeek | I can tell you I never got it to work 100% reliably |
| 12:48.30 | tzanger | if you tell it to |
| 12:48.31 | tzanger | I don't |
| 12:49.08 | DrPete | tzanger: So do you recevice fax and voice one line? and it works ok? |
| 12:49.22 | tzanger | DrPete: that is not what I said I did |
| 12:49.26 | tzanger | but it would work easily |
| 12:50.01 | DrPete | tzanger: sorry, i must have miss read. How could I do that, with the minimum of cost? |
| 12:50.28 | tzanger | DrPete: get yourself a TDM110P, 1 FXS, 1 FXO |
| 12:50.31 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 12:50.51 | tzanger | or get a SPA3k or something and try doing it over VOIP |
| 12:50.54 | tzanger | on a lan it should work |
| 12:51.11 | tzanger | unless you're willing to invest about $3-500 don't bother |
| 12:51.24 | tzanger | you're better off with POTS for the frustration it'll cause you |
| 12:51.37 | tzanger | VOIP and cheap don't really mix well |
| 12:51.40 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@ALe-Mans-201-2-1-65.w193-251.abo.wanadoo.fr) |
| 12:51.46 | tzanger | the TDM series of cards is pretty damned inexpensive |
| 12:51.48 | DrPete | tzanger: I dont mind spending money, if I get what I want |
| 12:52.13 | tzanger | to be perfectly honest I am having issues with the FXS modules and sending a fax through it to a fax machine |
| 12:52.24 | tzanger | which is why I use app_rxfax and it's actually working out better in the end |
| 12:52.39 | tzanger | I can send faxes from the fax machine, through the TDM430P and out just fine |
| 12:52.52 | tzanger | but the other way 'round I get very poor fax success rate |
| 12:53.03 | DrPete | tzanger: I see |
| 12:53.13 | tzanger | I didn't have this problem at all with the T100P+Channel Bank so I imagine it's something with my mobo or the TDM card itself |
| 12:53.20 | tzanger | it's not system load or anything |
| 12:53.25 | JonR800 | i've been getting 100% reliablity with a x100p, spa-2000, fax modem, and hylafax |
| 12:53.40 | tzanger | JonR800: you miss the point entirely |
| 12:53.42 | JonR800 | it's all about tuning your rxgain and txgain. |
| 12:53.47 | JonR800 | sorry |
| 12:53.55 | tzanger | JonR800: I am talking about a real, physical fax machine on my end |
| 12:53.56 | RoyK | soooooooooooooooriiiiiiiiiiii? |
| 12:54.13 | JonR800 | shouldn't be much different tzanger. |
| 12:54.25 | tzanger | I am haivng trouble getting faxes through the TDM card to either one of them (both are G3 compatible too) |
| 12:54.29 | datareactor | HI ALL |
| 12:54.51 | tzanger | JonR800: well considering I am saying I am getting 100% success with app_rxfax I think there is a significant difference |
| 12:54.52 | JonR800 | tzanger: that's probably because the TDM card is slipping.. |
| 12:54.54 | tzanger | :-) |
| 12:55.01 | tzanger | JonR800: there's no reason for it to be slipping |
| 12:55.04 | datareactor | i want to connect 20 analog phone to my asterisk box which card should i use for that |
| 12:55.15 | tzanger | plenty of CPU, the same motherboard is handling the T100P just fine, etc., etc. |
| 12:55.40 | tzanger | datareactor: T100P and the cheapest FXS channel bank you can get your hands on |
| 12:55.50 | tzanger | I am partial to the Access Bank I for FXS-only situations, and the Adit600 for FXS+FXO situations |
| 12:55.51 | JonR800 | tzanger: well sorry for jumping in :) but yes, cheapest is spandsp. |
| 12:56.26 | tzanger | JonR800: yeah it works... I think there's something with the FXS module or FXS driver causing issue but the kicker is that I can use the same fax machine and same TDM card and same data path to send faxes just fine, I just can't receive them |
| 12:56.27 | *** join/#asterisk dan2 (dan@dan2.student.supporter.pdpc) |
| 12:56.35 | DrPete | tzanger: Basically, I have a fax modem on a linux server. and a single phone line. I want to reive a call, if its a fax recive it, if its voice allow me to answer it. I know I could buy fax machine, but I would like to use a PC. whats the best way I can do that? |
| 12:56.37 | tzanger | I can receive them on the same machine to app_rxfax so that is why I believe it's a TDM thing |
| 12:56.43 | markit | tzafrir: I've 2 tdm400p, and I've a fax machine also that I would like to connect. Since I've echo with voice calls also, could it be a echo problem? do you have echo with phones on fxs ports? |
| 12:56.48 | markit | sorry |
| 12:56.50 | markit | tzanger: |
| 12:57.03 | datareactor | tzanger T100P for pstn connectivety ? |
| 12:57.08 | tzanger | DrPete: is there a reason that you need to use the fax modem? i.e. if you used app_rxfax and app_txfax directly on the asterisk box would htat be acceptable? |
| 12:57.40 | tzanger | markit: no it's not an echo issue, like I said I can send faxes with the same fax+tdm+data path, I just can't receive to that fax with the TDM :-) |
| 12:58.11 | markit | tzanger: but did you have echo issues and how did you resolve it? |
| 12:58.27 | DrPete | tzanger: as long as it works yeah. I am new to all this, I have never exploered telephony on linux. I dont mind learning what I need, but i want to know it will work before I start |
| 12:58.31 | tzanger | datareactor: if you're using hte T100P to connect to PSTN then you're doing it over CAS T1 or PRI, and FXS channel banks don't come into the picture |
| 12:58.38 | markit | if I call with budgetone, I've no echo, while if I use a phone connected with tdm400 I've some |
| 12:59.04 | tzanger | markit: that's all on the wiki -- first off set your gains to 0.0 and turn on the =echo canceller and maybe echotraining=800 too if you still can't get it |
| 12:59.09 | tzanger | then and only then do you start playing with gain |
| 12:59.28 | tzanger | DrPete: I have excellent success with the app_rxfax and app_txfax |
| 12:59.37 | tzanger | DrPete: as I said I receive between 30 and 50 faxes a day without issue |
| 12:59.57 | tzanger | we're using a real fax machine to send (think old office politics rather than tech issues) but soon. :-) |
| 13:00.10 | tzanger | DrPete: my faxes come in, I convert them to PDF and dump them on a samba share |
| 13:00.14 | DrPete | tzanger: so what hardware would I require? |
| 13:00.28 | tzanger | we're starting ot route the fax to the appropriate person via the claler id and received CSR |
| 13:00.43 | tzanger | an FXO port to connect asterisk to the phone network |
| 13:00.48 | DrPete | tzanger: thats exactly what I need. I use a winbox just now to send faxes, mailing list etc, and it works great, I dont want a machine |
| 13:00.56 | tzanger | hell I receive many faxes at once even, it works great :-) |
| 13:01.05 | datareactor | tzanger FXS channel band you recommend ? |
| 13:01.18 | datareactor | bank |
| 13:01.52 | tzanger | DrPete: ok -- just be aware that Digium FXO ports do not "share the line" well -- i.e. if you have an FXO port and a phone sharing a phone line asterisk doesn't reliably deal with "oh the line is in use" issues |
| 13:02.01 | tzanger | datareactor: for FXS only: Carrier Access Access Bank I |
| 13:02.23 | tzanger | they're cheap (US$250ish) and work fine for FXS. They have FXO issues with far-end-disconnect supervision |
| 13:02.44 | DrPete | tzanger: I see, I assume that I will need a card that has an FXS? do that I can use a phone for voice calls? |
| 13:02.50 | bowman | on what condition does ChanIsAvail decide if the requested channel is available? If I speak on phone1 and do a chanisavail(sip/phone1), I get the response that a channel to phone1 is available. eh?? |
| 13:02.56 | mquin | tzanger: yeah - my experiments with an x100p clone on a shared line resulted in me blasting DTMF into an active conversation |
| 13:03.13 | tzanger | mquin: exactly |
| 13:03.28 | tzanger | DrPete: no |
| 13:03.36 | tzanger | FXS port = plug phones/equipment into it |
| 13:03.40 | tzanger | FXO port = plug a phone LINE into it |
| 13:03.50 | tzanger | oh |
| 13:04.00 | DrPete | tzanger: wont i need one of each? |
| 13:04.14 | tzanger | sorry yes if you intend on having asterisk handle voice calls you will need at least one FXS port (or VOIP phones) |
| 13:04.31 | tzanger | I need to get going, gotta triple-s and get the kids ready, I'll be on around 9am (1 hour from now) |
| 13:04.53 | DrPete | tzanger: thanks, laters. |
| 13:05.11 | DrPete | Man, i need to eat. Thanks for all the help everyone |
| 13:05.18 | datareactor | Thanks tzanger for the help |
| 13:07.37 | Zeeek | another happy customer - kerchink ! |
| 13:07.43 | Zeeek | kerchunk ! |
| 13:08.26 | markit | anyone using the new "native asterisk transfer"? bug/patch 3241 |
| 13:08.36 | *** join/#asterisk Fabe (~spamhere@pD95B2A09.dip0.t-ipconnect.de) |
| 13:10.58 | *** join/#asterisk nextime (~nextime@danex.i-m-c.it) |
| 13:11.06 | markit | wb nextime |
| 13:12.14 | *** join/#asterisk nextime (~nextime@ns0.nexlab.net) |
| 13:13.26 | datareactor | i want that when i dial 100 asterisk answer the call playback mysong |
| 13:13.54 | datareactor | 1,Answer() |
| 13:14.12 | *** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es) |
| 13:14.38 | *** join/#asterisk ToyMan (~stuq@204.8.82.238) |
| 13:15.00 | *** join/#asterisk Mike9 (~sturdee@sun.mikesweb.com) |
| 13:15.08 | *** join/#asterisk PakiPenguin (~info@203.130.21.42) |
| 13:15.33 | datareactor | exten => 100,1,Answer() exten => 100,2,Playback(mysong) is this OK |
| 13:15.43 | *** part/#asterisk Sound (~Sound@adsl-115-159.37-151.net24.it) |
| 13:18.14 | JonR800 | what format is the song? |
| 13:18.35 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:19.15 | Faithful | does anyone know if the patch for broadvoice is in the latest CVS |
| 13:19.42 | modulus_ | i have latest cvs |
| 13:19.46 | modulus_ | i never had to use the patch |
| 13:19.50 | modulus_ | i quit cold-turkey |
| 13:20.06 | JonR800 | it was in the latest |
| 13:20.16 | modulus_ | broadvoice kinda sucks |
| 13:20.18 | *** join/#asterisk tdonahue (~tdonahue@haynesmail.haynes-group.com) |
| 13:21.19 | JonR800 | actually it's in stable too. |
| 13:23.41 | Faithful | modulus_: why so? |
| 13:23.49 | modulus_ | huh? |
| 13:23.53 | modulus_ | why so what? |
| 13:24.05 | Faithful | modulus_ broadvoice kinda sucks |
| 13:24.13 | modulus_ | all voip providers kinda suck |
| 13:24.21 | modulus_ | i swear they're run by 17 year olds in basements |
| 13:24.28 | modulus_ | if you notice, none of them have real bandwidth |
| 13:24.37 | datareactor | JonR800 its mp3 |
| 13:25.02 | Faithful | modulus_: I imagine that they run a tight budget |
| 13:25.08 | *** join/#asterisk pranav (pranav@203.115.90.235) |
| 13:25.26 | Zeeek | what's wrong with 17 year olds in basements? I was once one |
| 13:25.49 | Zeeek | many moons ago in a galaxy far, far away |
| 13:25.59 | *** join/#asterisk Miker (~mike@201.129.119.248) |
| 13:25.59 | PakiPenguin | Zeeek: i was a 17 yr, old too , a decade ago though :p |
| 13:26.12 | Zeeek | ah, but were you geeking around in a basement |
| 13:26.58 | PakiPenguin | Zeeek: pretty much :) |
| 13:27.46 | Zeeek | I was building a TV camera and playing TV star with my friends |
| 13:27.53 | Zeeek | well that was actually more like 13 |
| 13:28.07 | Zeeek | at 17 I was pretending to tbe the Beatles |
| 13:29.03 | *** join/#asterisk pranav (pranav@203.115.90.235) |
| 13:29.08 | *** part/#asterisk pranav (pranav@203.115.90.235) |
| 13:30.05 | bjohnson | at 17 I was trying to get laid |
| 13:30.13 | Zeeek | me too! |
| 13:30.21 | Zeeek | unsuccessful for a long time too |
| 13:30.22 | bjohnson | everything else was secondary |
| 13:30.35 | shaZwaz_ | later guys |
| 13:30.36 | Zeeek | then I got in a rock band and things moved faster |
| 13:30.39 | bjohnson | Faithful: any luck connecting to bv? |
| 13:30.41 | Zeeek | bye shaZ |
| 13:30.48 | bjohnson | hehe .. me too .. drummer |
| 13:31.02 | Zeeek | someone has to pack the shit in the truck :) |
| 13:31.25 | bjohnson | modulus_: so in your opinion, which voip provider "sucks" least? |
| 13:31.47 | Zeeek | I have had good luck with: voicepulse, nufone, iconnecthere for different reasons |
| 13:32.00 | Zeeek | I am not _modulus though |
| 13:32.21 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 13:32.30 | Faithful | No I have been running 1.0.2-2 in debian and it isn't patched for bv so now I am installing another box with current |
| 13:32.50 | Zeeek | what is the specific problem everyone is having with BV ? |
| 13:33.25 | Faithful | It doesn't authenticate properly (even tho it says it does) |
| 13:33.55 | Faithful | and you need a patched chan_sip.c to do it.... current has it in??? |
| 13:34.34 | datareactor | how can i convert mp3 files to gsm ? |
| 13:34.41 | Faithful | sox |
| 13:34.57 | Faithful | datareactor: it is in the wiki |
| 13:35.04 | datareactor | Faithful Thanks |
| 13:35.05 | Faithful | quite easy |
| 13:37.06 | *** part/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 13:37.06 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 13:37.44 | RoyK | what was that 'simple url' or 'short url' site? |
| 13:38.20 | Zeeek | you have a great but long porn site you wanna share, RoyK ? |
| 13:38.28 | pointer-gaim | [slightly OT] anyone know of a forum package (with a good security track record) that integrates well with mailman? ....for the broadvoice-users forum/ml |
| 13:39.10 | RoyK | Zeeek: not with you, sorry |
| 13:39.19 | *** join/#asterisk HiTech69 (~hitech@155-105.202-68.tampabay.rr.com) |
| 13:39.24 | Zeeek | bad joke |
| 13:39.26 | pointer-gaim | phpBB has one on sf.net...but phpbb doesn't have what I'd consider a good security track record...it almost seems like the sendmail/bind of the forum world :-\ |
| 13:39.59 | Zeeek | that's a bad insult to sendmail ! |
| 13:40.08 | ^Slash | hey Zeeek how do i make it so anon ppl cant just call my sip server |
| 13:40.21 | ^Slash | also what software did you use to connect to me ? |
| 13:40.31 | Zeeek | Slash just comment that thing out until youneed it |
| 13:40.56 | ^Slash | insip.conf ? the context=default |
| 13:41.05 | bowman | mh, I can't build the current CVS code - is it known to be broken? |
| 13:41.06 | ^Slash | what software did you use ? whats recomended ? |
| 13:41.45 | Zeeek | Slash - leave default context in sip. Just don't put anything useable in [default] in extensions.conf |
| 13:42.00 | *** join/#asterisk Caede (~chatzilla@204.94.248.81) |
| 13:43.10 | bjohnson | ^Slash: make a refuse context in extensions.conf that just has congestion tone and point your sip guest account at that |
| 13:43.33 | ^Slash | ahh ok |
| 13:43.40 | blankman | Hey guys, simple question. When configuring the 410p, does one have to 'skip' the 24 when configuring the channels in zapata.conf? |
| 13:43.46 | bjohnson | ^Slash: and stop using ^ at the start of your nick .. if f*cking annoying for tab completion |
| 13:44.15 | blankman | Meaning the 24th channel of each T1? |
| 13:44.26 | slash^ | ^ <-- from the days of irc 7 years ago when someone stole my nick on efnet heh |
| 13:44.50 | Zeeek | slashdot |
| 13:44.52 | slash^ | Zeeek what software did you use ? |
| 13:45.04 | Zeeek | for what? i don need no stinking software |
| 13:45.16 | bjohnson | blackdog: try again later .. looks like a lot of the PRI users aren't here yet |
| 13:45.17 | slash^ | to connect to me |
| 13:45.28 | bjohnson | blankman: that was really for you |
| 13:45.37 | Zeeek | there's this program called asterisk - dunno if you've heard of it? |
| 13:45.48 | *** join/#asterisk rb|den (~redbaron@r.u.gg) |
| 13:45.49 | slash^ | oh |
| 13:45.51 | slash^ | you used that |
| 13:45.58 | slash^ | not a proggie from the pc |
| 13:45.59 | slash^ | haha ok |
| 13:46.01 | PakiPenguin | :) lol |
| 13:46.04 | bjohnson | Faithful: what's the bv sip issue? I ws thinking of signing up but don't feel like screwing around with patches, etc |
| 13:46.05 | Zeeek | I added a little guest out thingie that makes it easier to do that sort of test call |
| 13:46.14 | slash^ | fair enuf :) |
| 13:46.32 | slash^ | anyone else want to hear the incredibly loud hold music ? heh |
| 13:46.37 | slash^ | Zeeek loved it |
| 13:46.38 | Zeeek | now I just need to put the guest URL in a db variable and dial the same number |
| 13:47.08 | slash^ | hmmm |
| 13:47.12 | Zeeek | dbput Family guestout SIP/1234@diomain.tld |
| 13:47.40 | Zeeek | then I can call 99 all day to test. When someone else asks, I just enter the new guest out URL |
| 13:47.43 | PakiPenguin | slash^: i do |
| 13:47.58 | Zeeek | which can be done using the web interface if I'm not logged in |
| 13:48.46 | blankman | bjohnson, thanks ... I will try again later tonight. |
| 13:49.21 | pointer-gaim | blankman: I believe you don't skip it |
| 13:49.32 | pointer-gaim | I have a t1 card and 1 analog port |
| 13:49.39 | pointer-gaim | the analog port is zap/25 |
| 13:49.53 | pointer-gaim | that would lead me to believe 24 is used |
| 13:49.56 | slash^ | eh PakiPenguin ? |
| 13:50.00 | pointer-gaim | let me check zaptel/zapata |
| 13:50.03 | slash^ | 6789@my_ip |
| 13:50.07 | slash^ | have a listen |
| 13:50.10 | slash^ | you will love t |
| 13:50.11 | slash^ | it |
| 13:50.18 | slash^ | and then tell me how to make it not so loud |
| 13:50.33 | rb|den | anyone got any idea why my meetme confs gradually get a longer and longer delay on them? 5s delay after 10mins conf |
| 13:51.04 | pointer-gaim | blankman: zaptel.conf: bchan=1-23 (newline) dchan=24 |
| 13:51.56 | pointer-gaim | blankman: in zapata.conf, I have 2 groups... 1-4 and 5-23...then the analog port on 25 |
| 13:52.06 | pointer-gaim | trunk |
| 13:52.09 | pointer-gaim | groups that is |
| 13:52.45 | datareactor | i have error when using sox "sox down.mp3 down.gsm" ox: Failed reading down.mp3: Do not understand format type: mp3 |
| 13:53.19 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 13:53.31 | fishboy1669 | hi |
| 13:56.06 | tzanger | ... large trout? |
| 13:56.09 | *** join/#asterisk Hymie (hymie@L8R.NET) |
| 13:57.30 | RoyK | tzanger: no. small reddish fish from Lake Tanganyika in mid Africa |
| 14:00.29 | tzanger | ok I'm back |
| 14:00.33 | tzanger | anyone still have fxo/fxs questions |
| 14:01.00 | Zeeek | I thought bkw was a small reddish fish from Lake Tanganyika in mid Africa |
| 14:01.35 | fishboy1669 | hey zeek thanks for your help demo with customer went ok today |
| 14:01.38 | fishboy1669 | i think there intereseted |
| 14:01.39 | tzanger | Zeeek: close but he's actually a minor variant of that fish: asteriskhelpdeskius gaymanicus |
| 14:01.47 | Zeeek | hah |
| 14:01.56 | Zeeek | fichboy - you OWE ME ! |
| 14:03.40 | fishboy1669 | and a few others zx81 erncic and who ever else has helped |
| 14:03.59 | fishboy1669 | of course the asterisk bitch where ever he has gone |
| 14:04.02 | fishboy1669 | ;-) |
| 14:04.07 | *** join/#asterisk gabb0 (~gabe@indo1.indosoft.unb.ca) |
| 14:04.19 | gabb0 | good day |
| 14:09.25 | fishboy1669 | http://www.zapatatelephony.org |
| 14:10.06 | fishboy1669 | hey anyone know what http://www.govarion.com/ are like for supplies? are they ethical? or is digium the place to go? |
| 14:10.52 | *** join/#asterisk _TripleF_ (kvirc@Toronto-HSE-ppp3884328.sympatico.ca) |
| 14:11.01 | Nuttah | actually i do have a fxo zaptel question tz if ya interested :) |
| 14:12.20 | Nuttah | any way of of zap answering the call straight away instead of waiting 4 rings before answering? |
| 14:12.35 | Nuttah | through a wildcard x100p |
| 14:12.45 | fishboy1669 | there is a setting i think |
| 14:13.03 | fishboy1669 | in phone.conf |
| 14:13.12 | fishboy1669 | mode=immediate |
| 14:13.14 | fishboy1669 | i think |
| 14:13.33 | gabb0 | immediate=yes in zapata.conf I believe |
| 14:13.50 | Nuttah | cheers.. |
| 14:16.07 | Nuttah | hmm that doesnt seem to do it |
| 14:16.33 | Nuttah | still does chan_zap.c:5097 ss_thread: Got event 2 (Ring/Answered)... |
| 14:16.40 | Nuttah | 4 times |
| 14:16.49 | rb|den | waiting to grab callerid? |
| 14:16.52 | *** join/#asterisk firstsword (~root@host6614613596.biz.tor.fcibroadband.com) |
| 14:16.52 | fishboy1669 | the one in zapata.conf decided wether to wait till u enter full number in phone befor dialing or to accept the number as u dial it |
| 14:17.02 | bjohnson | Nuttah: I don't think the x100p uses phone.conf |
| 14:17.04 | rb|den | try turning callerid off and see if it's faster |
| 14:17.10 | bjohnson | Nuttah: check zapata.conf |
| 14:17.21 | fishboy1669 | http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
| 14:17.25 | Nuttah | i have immediate=yes in zapata.conf |
| 14:17.32 | Nuttah | doesnt seem to help |
| 14:17.36 | bjohnson | Nuttah: also make sure the context in extensions.conf doesn't use a high wait() setting |
| 14:18.35 | Nuttah | i dont |
| 14:18.40 | bjohnson | I don't think you want immediate=yes in zapata.conf but I'm not sure .. it depends if you want to answer ALL calls or do some callerid checking first |
| 14:18.44 | Nuttah | i have a wait, 1 which capi takes |
| 14:18.47 | Nuttah | but zap ignores |
| 14:19.01 | Nuttah | at the moment all calls |
| 14:19.19 | Nuttah | cli I do not need right now |
| 14:19.42 | bjohnson | Nuttah: my x100p answers after the first ring and I have immediate=yes in zapata.conf |
| 14:19.51 | Nuttah | hmm what else? |
| 14:19.51 | PoWeRKiLL | why do I have span that restart all time ? |
| 14:20.09 | Nuttah | mebbe i've mmispelled something |
| 14:20.29 | bjohnson | Nuttah: I think my zapata.conf is the default with signalling and channel added at the end |
| 14:20.57 | Nuttah | i have 2 x100p's so i have all my settings then the 2 channels |
| 14:21.10 | Nuttah | only have echocancel and immediate i believe |
| 14:21.35 | bjohnson | I also have rxwink=300 |
| 14:21.41 | *** join/#asterisk Samoied (~samoied@popeye.opens.com.br) |
| 14:22.00 | Samoied | Hello! |
| 14:22.23 | Nuttah | hmm |
| 14:22.29 | *** join/#asterisk PaulOnly (Paul@host213-120-111-169.in-addr.btopenworld.com) |
| 14:22.32 | fishboy1669 | hi |
| 14:22.37 | PaulOnly | Hello |
| 14:22.40 | fishboy1669 | hi |
| 14:22.46 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 14:22.47 | Nuttah | one sec.. checking its reload correctly |
| 14:23.08 | Samoied | What Analog Telefone adapter is better? |
| 14:23.20 | Samoied | I preffer with GSM or ilbc codecs |
| 14:23.55 | bjohnson | and then some callforwarding and callerid stuff that I can't imagine would have an impact. My extensions.conf context that is used is just a wait(2) and then a dial to ring sip extensions |
| 14:24.38 | Nuttah | i have wait also |
| 14:24.45 | Nuttah | i dont think its an extension issue |
| 14:24.52 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-246-241.buff.east.verizon.net) |
| 14:24.57 | bjohnson | Samoied: fxo or fxs? Are you providing two model to say which is better? |
| 14:25.58 | SuPrSluG | <PROTECTED> |
| 14:26.08 | SuPrSluG | soory |
| 14:26.14 | SuPrSluG | hello |
| 14:27.24 | bowman | do I need a timing source for commands like musiconhold, mp3player etc.? |
| 14:27.34 | Delvar | nope |
| 14:27.42 | Delvar | only timing for meetme etc.. |
| 14:27.48 | bowman | ok |
| 14:29.58 | *** join/#asterisk kludgebox (~bob@12.171.178.194) |
| 14:30.14 | *** join/#asterisk Moc_ (~mochouina@modemcable212.49-80-70.mc.videotron.ca) |
| 14:31.06 | *** join/#asterisk smurfix (~smurf@run.smurf.noris.de) |
| 14:31.47 | *** join/#asterisk Laura (nisse16@200.226.102.123) |
| 14:31.56 | bjohnson | what does the broadvoice patch actually do to sip.conf? |
| 14:32.01 | Nuttah | bjohnson: not having much luck here |
| 14:32.18 | tzanger | oh yay |
| 14:32.27 | tzanger | canada plans on joining the missle defence bullshit |
| 14:33.27 | Nuttah | have only immediate=yes and echocancel=yes at teh beginning of zapata.conf, yet it still rings 4 times before zap answers |
| 14:33.35 | Zeeek | mistles? I'm leaving for my shelter ! |
| 14:34.14 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:34.18 | bjohnson | tzanger: plans to join? What happend to Norad? |
| 14:34.45 | RoyK | http://bugs.digium.com/bug_view_page.php?bug_id=0003331 |
| 14:34.48 | RoyK | stupid bug |
| 14:35.29 | *** join/#asterisk KahiN (Watts@212.253.56.43) |
| 14:35.45 | bjohnson | Nuttah: I don't know .. maybe try watching the logs with verbose increased to see if it doesn't even sense the ring until after a few have passed? |
| 14:35.54 | Nuttah | no |
| 14:35.58 | tzanger | if canada doesn't it will go ahead without norad control |
| 14:36.00 | Nuttah | already done that |
| 14:36.10 | Nuttah | frist ring is picked up |
| 14:36.13 | Nuttah | first even |
| 14:36.57 | Nuttah | tis a wee bit strange |
| 14:38.20 | *** join/#asterisk scrub1 (~scrubb@OCI-19-41.OneCall.Net) |
| 14:39.05 | *** join/#asterisk simong (~simong@h166n2fls35o884.telia.com) |
| 14:39.24 | simong | <PROTECTED> |
| 14:39.54 | Nivex | simong is now rapidly changing his password :) |
| 14:39.54 | bowman | wee, thanks. |
| 14:39.56 | bowman | :D |
| 14:39.59 | simong | Yep |
| 14:40.19 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 14:40.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:40.43 | simong | Copy paste is dangerous |
| 14:40.56 | eKo1 | you need to learn how to do it |
| 14:40.59 | eKo1 | properly |
| 14:43.43 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 14:44.09 | suma | hi |
| 14:44.41 | suma | on what basis the x100p card is recognized as x100p card ? |
| 14:47.35 | fishboy1669 | no wildcard or |
| 14:47.51 | fishboy1669 | something else i cant remember at moment |
| 14:48.15 | Nuttah | x101p |
| 14:48.33 | suma | yes |
| 14:48.38 | fishboy1669 | there is another name as well |
| 14:48.41 | suma | but how it is a modem right ? |
| 14:48.49 | Nuttah | yeah mean at bootup fish? |
| 14:49.04 | suma | why not other modems are not recognized as x101p or wildcard or x100p ? |
| 14:49.17 | suma | i mean with modprobe wcfxo ! |
| 14:49.18 | nestAr | because they are not the same card |
| 14:49.26 | Nuttah | i dont follow suma? |
| 14:49.34 | suma | it says intel 537 chipset right ? |
| 14:49.37 | nestAr | why isn't your ethernet card picked up as a x100p |
| 14:49.45 | nestAr | not all 537 chipsets are the same |
| 14:49.55 | nestAr | i have 2 different 537's at home\ |
| 14:49.56 | suma | mm |
| 14:50.10 | nestAr | one works as a x100p, one won't even post in my * box |
| 14:50.17 | suma | which intel chipsets are recognised as Wildcard ? |
| 14:50.25 | fishboy1669 | nuttah isnt it called cheta or something |
| 14:50.38 | suma | what key factor recognizes that the card is x100p ? |
| 14:50.39 | SuPrSluG | suma: it must have md3200 chipset |
| 14:51.02 | bjohnson | the true x100p is a tigerjet |
| 14:51.14 | fishboy1669 | thats the one i knew it was some sort of cat |
| 14:51.14 | nestAr | mine shows up as a tigerjet in a lspci |
| 14:51.15 | SuPrSluG | suma: the EPxxxxxxx don't work |
| 14:51.20 | fishboy1669 | cheers bjohnson |
| 14:51.53 | suma | same me too |
| 14:51.54 | suma | 00:09.0 Communication controller: Tiger Jet Network Inc. Intel 537 |
| 14:51.54 | *** join/#asterisk sarumont (~sarumont@pcp04563795pcs.dallas01.ga.comcast.net) |
| 14:52.04 | fishboy1669 | suma thats the one |
| 14:52.07 | bjohnson | suma: tiger jet is a company |
| 14:52.25 | fishboy1669 | sumawhats your problem with it |
| 14:52.28 | fishboy1669 | suma |
| 14:52.30 | suma | but i bought this as a intel modem from my local computer store |
| 14:52.38 | *** join/#asterisk josiah_ (~josiah@207.40.85.50) |
| 14:52.44 | fishboy1669 | does it work |
| 14:52.51 | suma | i want to know whether i can bring any other 537 chipsets under this |
| 14:52.56 | fishboy1669 | what happens when u modprobe wcfxo? |
| 14:53.00 | suma | i want one more x100p |
| 14:53.11 | suma | it was out of stock in the near by store |
| 14:53.32 | suma | he has intel 537 chipsets that is why i asked |
| 14:53.40 | fishboy1669 | pass i have been wondering how many x100p's i could put in a box without it breaking |
| 14:53.46 | suma | will any other chipsets will work other than md3200 ? |
| 14:54.01 | fishboy1669 | u need to check the wiki see compatable hardware |
| 14:54.10 | SuPrSluG | not that i know of |
| 14:54.22 | suma | i checked in the wiki, i could not find anything out there |
| 14:54.28 | fishboy1669 | poo |
| 14:54.33 | fishboy1669 | fisypoo |
| 14:54.38 | fishboy1669 | fishpoo even |
| 14:54.49 | nestAr | suma: i would seriously consider forking out the cash for a real x100p |
| 14:55.03 | nestAr | i'm still trying to get caller id to work on my modem |
| 14:55.15 | nestAr | not to mention that it seems to introduce noise into my pots line |
| 14:55.20 | fishboy1669 | nestAr where u from? |
| 14:55.26 | fishboy1669 | im in uk and have problem with it |
| 14:55.31 | nestAr | US |
| 14:55.36 | fishboy1669 | something to do with bt and how they give out clid |
| 14:55.37 | suma | my x100p works fine |
| 14:55.44 | bjohnson | I love my SPA 3000 |
| 14:55.56 | fishboy1669 | spa 3000 what is that? |
| 14:55.56 | cripito | :) and i my SPA 1001 |
| 14:56.03 | bjohnson | and I have a real x100p for timing of meetme calls |
| 14:56.03 | fishboy1669 | what are they? |
| 14:56.07 | SuPrSluG | anyone doing dundi. my problem = has become TOO LAGGED (3416 ms) . but pings are low. |
| 14:56.40 | fishboy1669 | how many x100p's can u put in one box with out confilict? |
| 14:56.50 | bjohnson | sipura SPA 3000 .. one fxs and one fxo in a separate, small box that connects to LAN and each port is a SIP client |
| 14:56.56 | suma | i have three in mine |
| 14:56.59 | suma | it is working fine |
| 14:57.07 | fishboy1669 | propper x100p's? |
| 14:57.08 | fishboy1669 | cool |
| 14:57.08 | bjohnson | fishboy1669: as many as pci slots you have |
| 14:57.18 | fishboy1669 | wupee i can have fun then |
| 14:57.35 | bjohnson | I love my $100 USD SPA 3000 |
| 14:57.43 | fishboy1669 | sept im having problems with horendus echo on it |
| 14:57.44 | cripito | :) |
| 14:57.47 | bjohnson | (soon to have 3 of them) |
| 14:58.05 | cripito | bj i don't change my 1001 |
| 14:58.12 | cripito | i fxs port 2 lines ;) |
| 14:58.16 | cripito | 1 fxs |
| 14:58.20 | cripito | sorry happy fingers |
| 14:58.21 | suma | all i have is the modem from local computer store |
| 14:58.39 | cripito | :D suma what brand? |
| 14:58.42 | fishboy1669 | suma are they propper x100p's |
| 14:58.52 | bjohnson | fishboy1669: hell no |
| 14:59.28 | suma | the software detects as x100p clones |
| 14:59.33 | fishboy1669 | cool |
| 14:59.41 | fishboy1669 | as long as it works |
| 15:00.46 | suma | yes |
| 15:00.59 | *** join/#asterisk DevilFish (~me@staff211.qtm.net) |
| 15:01.04 | suma | i could not get those cards anymore |
| 15:01.19 | suma | that is why i asked is there is any alternative chipsets which can perform the same |
| 15:05.02 | bjohnson | sometimes people on ebay are stupid. Why would you pay the same (within $5) for a used SPA 3000 as a new one that includes a month of voip service? |
| 15:05.21 | *** join/#asterisk mogorman (~mogorman@216.207.244.182) |
| 15:05.28 | suma | mm |
| 15:05.36 | bjohnson | btw, a SPA 2000 on there for $46 right now |
| 15:05.45 | Delvar | anyone got a link to http download of grandstreams 1.0.5.16 ? |
| 15:05.46 | suma | i have seen auction in ebay for x100p for $0.99 |
| 15:06.02 | Nuttah | same reason as ppl seem to buy printers and the like in online auctions for more than a brand new printer |
| 15:06.42 | Nuttah | delvar: grandstreams download not working? |
| 15:07.15 | bjohnson | suma: likely not a real one. And likely shipping is for $20 more than real cost |
| 15:07.41 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 15:07.41 | suma | $12 is the shipping cost to local US |
| 15:07.51 | suma | is it not cheap ? |
| 15:08.01 | suma | total of $13 ! |
| 15:08.12 | Nuttah | its not real by the sounds of things |
| 15:08.22 | suma | means ? |
| 15:08.41 | suma | they cheat by getting money ? |
| 15:09.22 | fishboy1669 | nuttah i got one from there |
| 15:09.23 | Nuttah | its probably an wildcard x100p box or something like that |
| 15:09.25 | fishboy1669 | real thing |
| 15:09.32 | Nuttah | for .99 cent? |
| 15:09.41 | fishboy1669 | cost me £18 |
| 15:09.46 | fishboy1669 | inc postage |
| 15:09.47 | fishboy1669 | uk |
| 15:09.47 | *** join/#asterisk srosebush (~steve@84-104-133-239.cable.quicknet.nl) |
| 15:09.55 | Nuttah | impressive |
| 15:09.56 | fishboy1669 | no support though |
| 15:10.03 | bjohnson | then it's not a real one |
| 15:10.05 | fishboy1669 | dont tell everyone though |
| 15:10.12 | fishboy1669 | as they arnt made anymore |
| 15:10.14 | *** join/#asterisk Sn0p (~aggrw@cm-84.118.197.101.chello.no) |
| 15:10.17 | suma | i have seen listing in UK, they sell it for 9.99 plus shipping |
| 15:10.17 | fishboy1669 | so everyone will go buying them |
| 15:10.18 | fishboy1669 | up |
| 15:10.29 | fishboy1669 | ye they went down after i bough one |
| 15:10.30 | fishboy1669 | :( |
| 15:10.34 | suma | i bought one of them, and they provided excellent support |
| 15:10.36 | Nuttah | not a wildcard then |
| 15:10.45 | Nuttah | as in digium |
| 15:10.48 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 15:10.51 | suma | what you mean support ? |
| 15:11.00 | _Brian | has anyone been able to get the Dial-Macro code to compile cleanly on the cvs-stable release? ( http://bugs.digium.com/bug_view_page.php?bug_id=0002905) |
| 15:11.11 | _Brian | or should I even say patch cleanly.. |
| 15:11.14 | fishboy1669 | suma u bought one of ebay? |
| 15:11.17 | fishboy1669 | and got support? |
| 15:11.19 | fishboy1669 | how |
| 15:11.23 | suma | yes fishyboy1669 |
| 15:11.34 | blitzrage | morning all |
| 15:11.37 | fishboy1669 | ? oh who supported it digium? |
| 15:11.38 | suma | actually x100p does not need much support |
| 15:11.39 | Nuttah | still cant get this fecker to work bj... i dont think my cards like me |
| 15:11.39 | cjk | hi, are there any prefixes which are meant for internal routing roules, so i can use them in * withouht interferring with the international prefixes etc... |
| 15:11.53 | Nuttah | yes |
| 15:11.53 | srosebush | hello -- just got an FXO clone card for my home based PBX, and was wondering, i still have extensions throughout the house that are still connected directly to the PSTN, and not thru * yet.. i cant seem to get asterisk to stop ringing my SIP devices if someone has already answered an incoming call from the PSTN? |
| 15:11.53 | bjohnson | * isn't fun until you have BOTH fxo AND fxs interfaces |
| 15:11.55 | *** join/#asterisk __andy___ (~andy@wsip-64-147-11-54.oc.oc.cox.net) |
| 15:11.56 | fishboy1669 | better register mine then lol |
| 15:12.10 | fishboy1669 | hey anyone any experience with a BT Speedway ISDN PCI Card |
| 15:12.16 | fishboy1669 | using it as a bri |
| 15:12.17 | fishboy1669 | ? |
| 15:12.29 | PaulOnly | Hello there, sorry if this question has been asked before, but can asterisk be used to interface into a GSM card to allow it to send txts or route calls to a mobile? |
| 15:12.40 | fishboy1669 | suma did u get support from digium? |
| 15:12.41 | suma | fishboy1669 ! what you mean supported it digium ? |
| 15:12.44 | suma | nope |
| 15:12.45 | bjohnson | PatrickDK: I believe so |
| 15:12.51 | suma | i got the support from the seller |
| 15:12.56 | fishboy1669 | suma aha cool |
| 15:12.57 | bjohnson | PaulOnly: check the wiki and the mailing list |
| 15:13.07 | fishboy1669 | suma did u need support from seller? |
| 15:13.11 | fishboy1669 | plug and play really |
| 15:13.14 | suma | actually no |
| 15:13.18 | fishboy1669 | lol |
| 15:13.21 | fishboy1669 | lmfao |
| 15:13.22 | suma | yes, it is plug and play |
| 15:13.33 | PaulOnly | Yeah I am reading all that sort of stuff now bj, just wondered if anyone knew off the top of their head :) Thanks anyway. |
| 15:13.34 | suma | suppport means it is real bull shit for x100p card |
| 15:13.59 | fishboy1669 | hi |
| 15:14.00 | DevilFish | Anyone here have trouble getting the newst bootrom and sip release for the Polycom IP 600s and 500s ? |
| 15:14.00 | fishboy1669 | hi |
| 15:14.01 | fishboy1669 | hi |
| 15:14.14 | josiah_ | hi fishboy1669 |
| 15:14.16 | fishboy1669 | anyone any idea about using a BT Speedway ISDN PCI Card as a bri???? |
| 15:14.19 | Nuttah | na i used a eicon diva card for my ISDN bri card |
| 15:14.39 | fishboy1669 | ye i got this bt one for free but dont know if it will work |
| 15:14.51 | bjohnson | PaulOnly: there is also this type of thing that could be paired with a fxs port: http://store.voxilla.com/customer/product.php?productid=16136&cat=0&page= |
| 15:14.54 | fishboy1669 | hey anyone from uk here or am i only uk asterisker? |
| 15:14.55 | Nuttah | prolly need isdn4linux |
| 15:15.05 | Nuttah | er no theres a few of us m8 :) |
| 15:15.14 | fishboy1669 | cool |
| 15:15.18 | fishboy1669 | was getting lonely |
| 15:15.19 | fishboy1669 | lol |
| 15:15.28 | fishboy1669 | nuttah which county im cheshire |
| 15:15.33 | Nuttah | herts |
| 15:15.35 | fishboy1669 | cool |
| 15:15.55 | bjohnson | spoke to fellow who was helping me here from Sommerset a few days ago |
| 15:16.01 | fishboy1669 | hey is the eicon the bt one rebranded? |
| 15:16.09 | Nuttah | dont think so |
| 15:16.22 | Nuttah | i recommand eicon cards tho |
| 15:16.27 | Nuttah | -a |
| 15:16.27 | Nuttah | +E |
| 15:16.54 | suma | what chipset is used for X100P and what chipset is for X101P ? |
| 15:17.26 | Nuttah | bj, can ya think of anything else matey... ive been scratching ma head for an hour of this |
| 15:17.35 | bjohnson | PaulOnly: http://voxilla.com/forum-viewtopic-t-1924.html |
| 15:18.19 | fishboy1669 | hi |
| 15:18.20 | fishboy1669 | The BT Speedway ISDN PCI card is an AVM Fritz! card - cheap on EBay. |
| 15:18.23 | fishboy1669 | cool |
| 15:18.26 | *** join/#asterisk qfkrbb (~el@p5082E709.dip0.t-ipconnect.de) |
| 15:18.28 | fishboy1669 | sounds prommising |
| 15:18.30 | Nuttah | then ya need isdn4linux |
| 15:18.31 | bjohnson | Nuttah: no .. but like I said .. I'm no expert .. I have really played with even the default settings at all |
| 15:18.53 | Nuttah | ah well.. back to prodding it with a stick |
| 15:20.03 | srosebush | hello -- just got an FXO clone card for my home based PBX, and was wondering, i still have extensions throughout the house that are still connected directly to the PSTN, and not thru * yet.. i cant seem to get asterisk to stop ringing my SIP devices if someone has already answered an incoming call from one of those extensions still connected direct to the PSTN |
| 15:20.10 | *** join/#asterisk Zeeek (~Zeeek@80.125.80.38) |
| 15:20.54 | Nuttah | probably because it doesnt get the pickup response from the pstn lines |
| 15:21.05 | Nuttah | i dont think you can get around that |
| 15:21.15 | srosebush | i think it does get it |
| 15:21.31 | srosebush | im not sure just a sec id have to call myself again |
| 15:22.12 | Nuttah | can you ask yourself if you know an zaptel expert? |
| 15:22.19 | Nuttah | :) |
| 15:22.29 | srosebush | lol? |
| 15:22.46 | Nuttah | sorry, i'm in a funny mood today |
| 15:22.56 | srosebush | :P |
| 15:23.30 | *** join/#asterisk h4mm3r` (~h4mm3r@81-208-60-202.fastres.net) |
| 15:23.57 | *** join/#asterisk zwi (~zwi@216.88.131.43) |
| 15:24.19 | srosebush | ok it seemed to work this time lol -- Jan 13 16:24:47 NOTICE[8377]: chan_zap.c:5363 ss_thread: Got event (Ring/Answered)... |
| 15:24.54 | srosebush | weee !! tks cya |
| 15:24.56 | *** part/#asterisk srosebush (~steve@84-104-133-239.cable.quicknet.nl) |
| 15:24.57 | *** join/#asterisk helvetico (~bs@80-218-190-36.dclient.hispeed.ch) |
| 15:26.38 | gabb0 | anyone here running * on a dell poweredge 1800? |
| 15:28.31 | *** join/#asterisk darkskiez (~mbryars@usergc137.dsl.pipex.com) |
| 15:28.34 | Samoied | What ATA yours recommends? |
| 15:28.40 | cjk | grandstream |
| 15:28.44 | Samoied | I preffer with ilbc or GSm codecs |
| 15:28.45 | cjk | i like grandstream |
| 15:28.48 | cjk | the best ones |
| 15:28.54 | Samoied | cjk: 286 or 486? |
| 15:28.55 | Nivex | the handytone has ilbc? |
| 15:29.01 | cjk | yes |
| 15:29.02 | Samoied | 486 has |
| 15:29.03 | cjk | and ilbc rocks |
| 15:29.05 | Nivex | schweet! |
| 15:29.09 | Samoied | 286 not |
| 15:29.12 | Nivex | crap |
| 15:29.24 | cjk | both have |
| 15:29.28 | cjk | just do the firmware upgrade |
| 15:29.31 | Nuttah | i quite like both 486 and sipura |
| 15:29.34 | cjk | all gs products suppoprt ilbs |
| 15:29.35 | cjk | all gs products suppoprt ilbc |
| 15:30.06 | Samoied | cjk: the spec of 286 not say ilbc |
| 15:30.18 | Samoied | http://www.grandstream.com/HandyTone_Spec.pdf |
| 15:30.25 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 15:30.36 | Samoied | cjk: You have one 286 ? |
| 15:30.37 | cjk | Samoied, im quite sure they support it |
| 15:31.06 | Samoied | cjk: Is possible to transfer the call with 286? |
| 15:31.13 | cjk | Samoied, yeah |
| 15:31.24 | Samoied | cjk: How? |
| 15:31.30 | cjk | dont know |
| 15:31.33 | cjk | did never test it |
| 15:31.34 | Nivex | The codec faq (http://www.grandstream.com/FAQ-Codec.pdf) says all support ilbc |
| 15:31.39 | cjk | but its in the manuar |
| 15:31.42 | cjk | but its in the manual |
| 15:31.47 | Samoied | cjk: thanks |
| 15:31.52 | cjk | all grandstreams have the same firmware |
| 15:31.54 | Samoied | Nivex: thanks |
| 15:32.08 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmt8.dialup.mindspring.com) |
| 15:32.15 | Samoied | cjk: You use ilbc ? |
| 15:32.36 | *** join/#asterisk Casper_UA (~casper@ragu.bestnet.kharkov.ua) |
| 15:33.01 | *** join/#asterisk pi-cubic (~vallenti@woodpecker.net.informatik.tu-muenchen.de) |
| 15:33.16 | cjk | Samoied, of course |
| 15:33.19 | cjk | its best |
| 15:33.27 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 15:33.31 | Samoied | cjk: I has wanted to use GSM |
| 15:33.31 | cjk | anyone here got a tftp server for gs firmware |
| 15:33.33 | *** part/#asterisk christo (~chris@office.enovi.com) |
| 15:33.50 | cjk | i prefer gsm |
| 15:33.51 | Samoied | but not have ATAs with this codec |
| 15:34.11 | cjk | grandstream have that codec with the new firmware |
| 15:34.17 | *** part/#asterisk pi-cubic (~vallenti@woodpecker.net.informatik.tu-muenchen.de) |
| 15:34.20 | Samoied | cjk: Yes? |
| 15:34.24 | Samoied | with GSM? |
| 15:34.58 | *** join/#asterisk datareactor (datareacto@203.81.215.23) |
| 15:35.34 | Nivex | cjk: you can update directly from GS according to their page |
| 15:36.24 | datareactor | hi i want to setup call centre using asterisk from 12 seats inbound only |
| 15:36.28 | Zeeek | cjk which new firmwatre has gsm ? |
| 15:36.59 | datareactor | what hardware cards i have to buy ? |
| 15:37.04 | cjk | sorry all i did i typo. i meant I PREFER ILBX |
| 15:37.06 | cjk | sorry all i did i typo. i meant I PREFER ILBC |
| 15:37.11 | tzanger | datareactor: PRI |
| 15:37.16 | cjk | sorry zeek |
| 15:37.23 | Samoied | cjk: thans |
| 15:37.34 | Samoied | ow |
| 15:37.51 | Samoied | have new firmware for add ilbc in 286 |
| 15:37.52 | datareactor | tzanger my calls will come from internet |
| 15:37.57 | Samoied | cjk: is that? |
| 15:38.16 | *** join/#asterisk PakiPenguin (~umair@202.147.174.97) |
| 15:38.39 | datareactor | and i want use ATA OR IP Phone ? |
| 15:39.05 | Samoied | And Boxes with FXO ports? |
| 15:39.19 | Samoied | What yours recommends? |
| 15:39.22 | bjohnson | datareactor: easiest is with 6 Sipura SPA 2000 ..but won't do conference calls well .. for that get a four port fxo from Digium and 4 SPA 2000 |
| 15:39.30 | fishboy1669 | can some one clarify zaptel is just basically the code to use a isdn bri as an interface for a voice call |
| 15:39.55 | PoWeRKiLL | any french guy here ? |
| 15:40.07 | jero | me |
| 15:40.15 | Logan | I don't think I like these Sipura phones. |
| 15:40.21 | bjohnson | datareactor: voxilla has the 2000 for $75 ea, includes some free voip service, free US shipping, and there is a $10 *users coupon |
| 15:40.22 | Zeeek | 1/2 french |
| 15:40.26 | jero | PoWeRKiLL |
| 15:40.26 | Fabe | fishboy1669: zaptel isn't doing isdn bri... use bristuff e.g. for that |
| 15:40.44 | *** join/#asterisk reni (~nobody@dhcp-157.digium.com) |
| 15:40.45 | datareactor | 1/2 french :) |
| 15:40.56 | Zeeek | oh? so there is ONE full french person here |
| 15:41.00 | jero | 1/2 = 1 |
| 15:41.01 | Zeeek | 1/2 + 1/2 |
| 15:41.22 | Zeeek | tu veux quoi ? |
| 15:41.26 | jero | french citizenship is transmitted either by birth or by one of the parents |
| 15:41.34 | Zeeek | BUZZZZ wrong |
| 15:41.37 | fearnor | fatal sexually transmitted disease? |
| 15:41.43 | jero | lol |
| 15:42.12 | bjohnson | where I'm from, you're not french if you speak english (at all) |
| 15:42.26 | fearnor | bj: quebec? |
| 15:42.38 | Zeeek | where I'm from it's -30°F in the winter |
| 15:42.46 | *** join/#asterisk dragonius (~fr@pD95F2857.dip.t-dialin.net) |
| 15:42.48 | Zeeek | but no one speaks french |
| 15:42.50 | dragonius | good morning |
| 15:42.55 | jero | hi |
| 15:43.02 | Zeeek | dragonius, good morning to you |
| 15:43.04 | dragonius | how can i call my asterisk server with a normal phone? |
| 15:43.18 | dragonius | hi Zeeek, hi Jero |
| 15:43.24 | Zeeek | by dialing into a speacial number that calls it using SIP or IAX? |
| 15:43.43 | Zeeek | or buy purchasing the hardware to convert the phonbe to SIP or IAX |
| 15:43.43 | dragonius | hm, where can i find one of those service numbers? i live in germany |
| 15:43.43 | jero | use a card to connect it to a phone line, or connect it to a provider that will provide you with a real number |
| 15:43.51 | Zeeek | siphone.de |
| 15:43.56 | dragonius | ok, thank you :) |
| 15:44.00 | jero | good luck |
| 15:44.05 | Zeeek | there is also nikotel i think? |
| 15:44.17 | Zeeek | sorry - sipgate.de |
| 15:44.26 | gabb0 | anyone else here having issues with no audio playback using apps such as background, playback, voicemailmain, etc |
| 15:44.31 | Zeeek | I have had an account for months and never use it |
| 15:44.42 | Zeeek | dragonius, sipgate.de |
| 15:44.53 | datareactor | bjohnson can you explain what is voxilla used for ? |
| 15:45.08 | Zeeek | http://www.sipgate.de |
| 15:45.14 | fishboy1669 | fabe whats zaptel do then |
| 15:45.24 | *** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net) |
| 15:45.32 | datareactor | gabbO what error did you get on asterisk console |
| 15:45.37 | Zeeek | gabb0 check your codecs |
| 15:45.40 | Fabe | fishboy1669: zaptel is a channel driver for digium cards. (fxo, fxs, pri .... ) |
| 15:45.49 | Derkommissar | roundrobin means that it rings in the agents always in the same order.... |
| 15:46.37 | bjohnson | datareactor: I think it's basically a store with support forums (there is a lot of good info there) |
| 15:46.40 | dragonius | is there a way to use a microphone connected to my pc to 'phone' my asterisk server? |
| 15:46.59 | bjohnson | dragonius: tes |
| 15:47.12 | bjohnson | dragonius: check the wiki for a list of softphone |
| 15:47.13 | gabb0 | none. not a codec issue. Here's the deal, when zaptel drivers are not loaded and using sip it works fine. WIth the drivers loaded and using sip all it does is show it's playing on the console but nothing is played. NO errors anywhere (console, logs, dmesg, etc) |
| 15:47.22 | dragonius | ok |
| 15:47.23 | dragonius | thx |
| 15:47.41 | *** join/#asterisk Samoied (~samoied@popeye.opens.com.br) |
| 15:47.41 | gabb0 | I'm using dell poweredge 1800, fc3 2.6 kernel. |
| 15:47.46 | bjohnson | Someone should post the wiki address in the topic |
| 15:48.02 | Zeeek | the codec mismatch ? |
| 15:48.02 | datareactor | bjohson what card should i used for incomming calls from internet |
| 15:48.16 | bjohnson | datareactor: nothing |
| 15:48.38 | Zeeek | here's a new one http://voxilla.com/modules.php?op=modload&name=News&file=article&sid=39 |
| 15:48.42 | bjohnson | datareactor: you will need a fxs port if you want to answer it with a regular phone .. or you will need a voip phone |
| 15:49.06 | bjohnson | datareactor: you could always answer it with a softphone too (but sound quality on softphones usually suck) |
| 15:49.08 | datareactor | bjohnson i want to ip phones i will connect all phones and asterisk in same switch thats it |
| 15:49.19 | gabb0 | TO note, I've also tried cvs head, stable 1.0.3 as well as stable from 2004. |
| 15:49.55 | bjohnson | datareactor: so you will be using voip phones? then you don't need any other hardware to make and recieve calls over the internet |
| 15:50.41 | datareactor | bjohnson what if i want to use analog phones |
| 15:51.09 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 15:51.16 | bprice20 | wow no auth |
| 15:51.16 | bprice20 | nice |
| 15:51.21 | datareactor | bjohnson do i need ATA for each analog phone ? or any other solution |
| 15:51.32 | bjohnson | does anyone know of another voip provider that has a similar $20 unlimited plan to broadvoice? I don't want to screw around with sip patches in order to use bv |
| 15:51.35 | bprice20 | datareactor yes you do |
| 15:51.49 | bprice20 | or get ip phone |
| 15:51.52 | bprice20 | phones |
| 15:52.00 | pointer-gaim | bjohnson: not that I know of... |
| 15:52.11 | bjohnson | datareactor: you need a fxs port for each analogue phone you want to connect to a voip system. some ATA have one port, some have 2 |
| 15:52.17 | pointer-gaim | bjohnson: the patch is easy to apply....what's the complication? you're compiling anyway, right? |
| 15:52.27 | heison | ~seen jerjer |
| 15:52.29 | jbot | jerjer is currently on #asterisk (6d 10h 39m 54s). Has said a total of 141 messages. Is idling for 8h 22s |
| 15:52.34 | bjohnson | pointer-gaim: I'm a fat, lazy, prick |
| 15:52.36 | datareactor | thanks bjohnson . bprice20 |
| 15:53.18 | *** join/#asterisk psycodad_ (~obiwan@2001:4060:4419:b1:0:0:0:2) |
| 15:53.21 | bjohnson | any channel ops here? |
| 15:53.27 | pointer-gaim | bjohnson: if you find another provider with similar service plans and pricing, please let me know...I haven't seen any...esp that give me unlimited calling to .de, .pl, and .it |
| 15:53.29 | Casper_UA | heison, smth h323 related? |
| 15:53.44 | nestAr | so far, the only thing i don't like about the polycom ip300 and 500 phones is the lack of backlighting |
| 15:53.44 | heison | Casper_UA: not really... |
| 15:53.49 | bjohnson | pointer-gaim: I found some for US/CAN but not overseas |
| 15:54.18 | datareactor | bjohnson which fxs card do u recommend ? |
| 15:54.29 | Casper_UA | heison: ok |
| 15:54.38 | *** join/#asterisk powD (~atob100@83.146.53.34) |
| 15:55.10 | *** join/#asterisk SkyTel_DK (~skytel@83.91.243.210) |
| 15:56.14 | bjohnson | (public service announcement) I have Sipura devices that I've used for a grand total of 3 weeks. I love them. I don't think there is a cheaper or better per port device than the SPA 2000 and the SPA 3000. The cheapest I found for both was at voxilla.com and they even offer a *users $10 off coupon |
| 15:56.36 | Logan | bjohnson: Do you haev a copy of the spc utility? |
| 15:56.37 | powD | Hello, |
| 15:56.43 | SkyTel_DK | hi all |
| 15:56.44 | cjk | hi, anyone here who does firmware upgrades over http on the grandstream |
| 15:56.49 | bjohnson | Logan: no |
| 15:56.58 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
| 15:57.01 | powD | Can anyone tell me if there is a patch to enable callerID on the X100P for UK users? |
| 15:57.22 | Logan | bjohnson: Apparently we went with the Sipura phones instead of the Polycom IP-300s because of a $40 price difference, but I think we're losing more than $40 in quality. |
| 15:57.37 | Logan | (I'ev been working with an IP-300 and IP-500 the past week or so) |
| 15:57.48 | bjohnson | Logan: I haven't any experience with voip phones .. these are fxo and fxs devices |
| 15:58.05 | Logan | Oh, I'm referring to the SPA-841. |
| 15:58.16 | fishboy1669 | bye guys |
| 15:58.32 | SkyTel_DK | is it possible to send a command that dials a number and when answered sends a voice promt from the command prompt on the box |
| 15:58.38 | josiah_ | anyine know if the SPA-2100's dual fxs lines can be used concurantly - e.g. both lines talking at the same time? |
| 15:58.48 | bjohnson | (caveat) separate small port devices may not be the best for everyone. Especially if you have numerous lines or hansets or want to do conference calls) |
| 15:58.59 | cjk | where can i get the latest grandstream firmware |
| 15:59.15 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 15:59.20 | Godsey | Logan: have you had the IP-300 phones begin having one way audio after 30 minutes of taking calls? |
| 15:59.40 | Godsey | I have the seperate user and peer entries for them under sip.conf |
| 15:59.51 | bjohnson | SkyTel_DK: voice promt from the command prompt on the box? |
| 15:59.51 | *** join/#asterisk Darwin35 (~Darwin@c-24-3-241-22.client.comcast.net) |
| 16:00.07 | SkyTel_DK | sorry |
| 16:00.16 | powD | cjk: For firmware 1.0.5.16 TFTP to 168.75.215.189 |
| 16:00.17 | Godsey | drop an outbound call into the dialplan? |
| 16:00.26 | SkyTel_DK | from the linux cmd prompt |
| 16:00.32 | SkyTel_DK | dial |
| 16:00.37 | Godsey | I beleive you can using call files |
| 16:00.42 | powD | cjk: See http://www.grandstream.com/y-downloads.htm |
| 16:00.43 | SkyTel_DK | say some words then hang up |
| 16:00.49 | josiah_ | anyone here used the Sipura SPA-2100 with asterisk? |
| 16:00.58 | Nivex | josiah_: I do. |
| 16:01.00 | SkyTel_DK | asterisk is running in background |
| 16:01.05 | Nivex | josiah_: oh wait.. I have the 2000 sorry |
| 16:01.11 | bjohnson | SkyTel_DK: I agree with Godsey .. check the wiki for call spool references |
| 16:01.21 | SkyTel_DK | thanx |
| 16:01.29 | josiah_ | Nivex, both look similar - two fxs lines right? dual analog jacks? |
| 16:01.30 | bjohnson | 2100 should be the same but also acts as router |
| 16:01.43 | bjohnson | 2100 has 2 nic ports |
| 16:01.50 | bjohnson | (plus 2 fxs) |
| 16:01.50 | josiah_ | bjohnson, ahh k... |
| 16:02.22 | josiah_ | bjohnson / Nivex - can both analog ports be used simultaneously on seperate sessions / calls? |
| 16:02.25 | *** join/#asterisk illek (mike@ip68-13-238-168.ok.ok.cox.net) |
| 16:02.27 | Logan | Godsey: I can't say I've had many conversations over them. :P |
| 16:02.36 | bjohnson | josiah_: lots of people use the SPA 2000 .. not that many the 2100 because not that many people want the router function |
| 16:02.51 | odie_flocon | I'm planning on ordering a bunch of polycom ip500's are they any good? |
| 16:02.55 | Nivex | josiah_: yes, both ports are independent of each other |
| 16:03.09 | bjohnson | josiah_: the 2100 is more suited to a remote connection into an asterisk box |
| 16:03.14 | josiah_ | Nivex, thanx...exactly what i needed to hear |
| 16:03.36 | bjohnson | josiah_: can both analog ports be used simultaneously on seperate sessions / calls? definitely .. I love it |
| 16:03.41 | josiah_ | bjohnson, aight...brb |
| 16:04.06 | Nivex | heck both ports don't even need to be tied to the same service |
| 16:04.06 | odie_flocon | Hey Mitel say's that the Mitel 5220 IP phone is sip compatable? can anybody confirm this? |
| 16:04.24 | Nugget | why do you disbelieve them? |
| 16:04.28 | *** join/#asterisk Brixius[work] (~Brixius@2.nexusinfosys.com) |
| 16:04.51 | odie_flocon | I would just like to use them in a sip configuration. |
| 16:05.13 | odie_flocon | and I can't find any settings that would let me do that... |
| 16:05.23 | bjohnson | odie_flocon: don't you think you'd be wiser to ask if sound quality is good .. or if there are stability problems? |
| 16:05.57 | odie_flocon | I know the phones are stable. |
| 16:06.08 | odie_flocon | I don't know if the SIP Stack on them is stable. |
| 16:06.17 | odie_flocon | I know the sound quality is good. |
| 16:06.29 | bjohnson | odie_flocon: http://www.voip-info.org/wiki-Asterisk+phone+Mitel+5220 .. google is my friend .. google can be your friend too |
| 16:06.39 | odie_flocon | I want to take them off a NEW Mitel PBX. |
| 16:06.44 | Nugget | bjohnson is feeling lucky. |
| 16:06.59 | datareactor | :) |
| 16:07.13 | dragonius | is there a tutorial out there for plain network phone with softphones? |
| 16:07.34 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 16:07.43 | bjohnson | dragonius: explain more |
| 16:07.55 | odie_flocon | isn't it great, I can build an entire * PBX, for 10,000 less then what it would cost to upgrade this Mitel pbx |
| 16:07.59 | dragonius | i just want to phone using my computer and a microphone |
| 16:08.04 | dragonius | i dont want to buy hardware |
| 16:08.12 | dragonius | of course i just want to phone another asterisk server |
| 16:08.14 | Zeeek | http://xten.net |
| 16:08.15 | Nugget | you don't need asterisk to do that. |
| 16:08.27 | bjohnson | odie_flocon: 9000 less .. don't forget your required donation to bkw_ |
| 16:08.29 | dragonius | but i want to learn how it works |
| 16:08.30 | Nugget | just get a softphone and a fwd account |
| 16:08.32 | odie_flocon | and if that's the case you can use *'s console to do that. |
| 16:08.38 | odie_flocon | hehehe |
| 16:08.48 | bjohnson | dragonius: cheeck the wiki for a list of about 50 softphones |
| 16:08.49 | dragonius | i just want to test it in my lan ;) |
| 16:08.51 | dragonius | ok |
| 16:08.53 | dragonius | thx |
| 16:08.53 | odie_flocon | hey that would still make it 2,000 cheaper. :D |
| 16:08.55 | Zeeek | dragonius: 1 download X-Lite from the link. 2) go to FWD and talk there |
| 16:09.02 | ManxPower | A customer of mine wants to put 40 cordless phones in the same office. |
| 16:09.04 | dragonius | ok |
| 16:09.04 | Samoied | Anyone use USB phone adapter under linux? |
| 16:09.07 | dragonius | thank you Zeeek |
| 16:09.10 | odie_flocon | I like to use DIAX |
| 16:09.11 | Zeeek | np |
| 16:09.29 | bjohnson | dragonius: using the * console for calls is for advanced users |
| 16:09.29 | odie_flocon | Diax is freee. |
| 16:09.40 | datareactor | dragonius if you can get voip fundamentals by ciscopress good book |
| 16:09.42 | Zeeek | dragonius : http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi#general_cat |
| 16:10.04 | bjohnson | Samoied: eeks .. don't buy one |
| 16:10.07 | Zeeek | FWD is a good way to get started. Lots of asterisk users over there too |
| 16:10.20 | odie_flocon | cordless as in 802.11 sip phones? |
| 16:10.23 | bjohnson | dragonius: I like iaxcomm |
| 16:10.26 | dragonius | what is FWD? |
| 16:10.29 | dragonius | forward? |
| 16:10.33 | Nugget | http://fwd.pulver.com |
| 16:10.36 | *** join/#asterisk wolfson` (~hehe@65.174.122.198) |
| 16:10.37 | dragonius | ok |
| 16:10.47 | ManxPower | odie_flocon: She doesn't know the difference. But you CANNOT put 40 standard 2.4ghz phones in a small area. |
| 16:10.58 | odie_flocon | Hitachi WIP-5000 IP Phone |
| 16:10.58 | odie_flocon | 349.95 |
| 16:11.00 | josiah_ | bjohnson / anyone: any idea where is good to buy SPA-2000s? |
| 16:11.00 | Zeeek | sorry I misread thought you mentioned fwd, it was someone else |
| 16:11.03 | Nugget | FWD is nice because it allows you to learn that softphones suck without all the complication and frustration of installing a local asterisk server. :) |
| 16:11.11 | ManxPower | Also the WiFi SIP phones all SUCK and would be twice the cost of the current proposal |
| 16:11.20 | Zeeek | no softphones are very useful for testing |
| 16:11.20 | datareactor | Free DialUP world asterisk peering network |
| 16:11.21 | josiah_ | (im lookn at http://www.telephonyware.com/telephonyware/scan/fi=products/st=db/co=yes/sf=category/se=PhonesAdapters/op=eq/tf=category,description.html and the 2100 are cheaper than the 2000..hmm) |
| 16:11.23 | bjohnson | josiah_: were you late for class? |
| 16:11.35 | bjohnson | (public service announcement) I have Sipura devices that I've used for a grand total of 3 weeks. I love them. I don't think there is a cheaper or better per port device than the SPA 2000 and the SPA 3000. The cheapest I found for both was at voxilla.com and they even offer a *users $10 off coupon |
| 16:11.41 | Zeeek | WiFi SIP phone do suck according to every post I've ever seen on the net |
| 16:12.04 | josiah_ | bjohnson: heh...no, boss came over to ask about asterisk stuff and how soon i can have it online |
| 16:12.06 | bjohnson | Zeeek: and security issues .. and too expensive to upgrade often |
| 16:12.10 | ManxPower | Zeeek: I suggested to the MIS manager that he get this office manager into therapy for her wire phobia. |
| 16:12.12 | Nugget | the pulver/zyxel ones suck for certain. I haven't used or met anyone who has used either the hitachi one or that other one. |
| 16:12.26 | datareactor | one if friend got ordinary PBX which support ISDN how can i intregate asterisk with it |
| 16:12.50 | Zeeek | personally I'm not that happy with every single device int he world being wireless |
| 16:12.52 | Samoied | FXO gateways? |
| 16:12.53 | ManxPower | datareactor: ISDN PRI or ISDN BRI and what country are you in? |
| 16:12.56 | Samoied | preferences? |
| 16:13.12 | Zeeek | it's getting hard to find decent corded phones these days tho |
| 16:13.16 | bjohnson | ManxPower: I got a $30 2.4G cordless phone with callerid and MWI that works great with my fxs |
| 16:13.20 | *** join/#asterisk ToyMan_ (~stuq@204.8.82.238) |
| 16:13.29 | ManxPower | bjohnson: So do I. But not FORTY of them. |
| 16:13.34 | Zeeek | bjohnson what brand? |
| 16:13.47 | bjohnson | ManxPower: just make sure they have enough channels |
| 16:14.01 | ManxPower | bjohnson: I think 204Ghz has like 6 channels. |
| 16:14.07 | datareactor | ManXpower PRI we follow eurpoen standard |
| 16:14.10 | bjohnson | Zeeek: some cheap noname thing I found at the local discount store |
| 16:14.26 | ManxPower | datareactor: Then any of the E* or TE* cards from Digium will work. |
| 16:14.37 | Zeeek | you may find the buttons or display wearing out early but for everyday use, it sounds good |
| 16:14.43 | *** join/#asterisk Slainte (~Slainte@66.55.113.205.ppp.northrock.bm) |
| 16:14.48 | bjohnson | ManxPower: wifi has 11 channels .. my cordless phone advertises something like 20 channels |
| 16:15.01 | ManxPower | datareactor: The nice think about standards is that there are do many of them. What COUNTRY is the line in? |
| 16:15.04 | Slainte | bjohnson what type of phone is it? |
| 16:15.10 | dragonius | how can i phone an ip with xlite? |
| 16:15.27 | bjohnson | Zeeek: for $30 CDN I can afford 10 of them for the same price as a wifi sip phone |
| 16:15.28 | Zeeek | the way to go would be digital encoding in the wireless phone itself and then connect it to anything else, phone line, asterisk |
| 16:15.32 | tzanger | datareactor: sorry about the delay (long phone call, and yes using asterisk, heh) |
| 16:15.35 | ManxPower | bjohnson: Yes, but that's still a lot less than 40. And 40 is the number of people they have right now, they expect close to 100 people within 24 months |
| 16:15.36 | *** join/#asterisk rowter (~Drake@201.133.209.110) |
| 16:15.39 | bjohnson | Zeeek: and they have built in security |
| 16:15.48 | tzanger | datareactor: if all yoru calls are coming in off the internet you need to decide how ot terminate the calls |
| 16:15.49 | *** part/#asterisk illek (mike@ip68-13-238-168.ok.ok.cox.net) |
| 16:15.49 | Zeeek | dragionius you are getting into the issues for which there is only one good answer |
| 16:15.59 | ManxPower | I told the person: It is impossible using current tech to do what you want. |
| 16:16.01 | tzanger | plain phones (use a T1 card and FXS channel bank), VOIP phones, what |
| 16:16.07 | dragonius | ;) |
| 16:16.10 | dragonius | "buy a book" |
| 16:16.13 | dragonius | "get a brain"? |
| 16:16.14 | dragonius | ;) |
| 16:16.22 | Zeeek | read the ${MACRO_EXPLETIVE} X-Lite documentation |
| 16:16.30 | dragonius | lol |
| 16:16.31 | dragonius | ok ;) |
| 16:16.39 | Zeeek | they went to the trouble of making up big PDF files |
| 16:16.39 | dragonius | thought there might be a quick answer :) |
| 16:16.43 | rowter | this an asterisk variable to get the billing time, I need to do a card time recharger, I been doing it with agi, but I want to know If I could do it with dialpan |
| 16:16.56 | rowter | get billing time when it hang up.. |
| 16:17.03 | bjohnson | Slainte: some cheap ass thing .. I think I should buy a few more |
| 16:17.11 | Zeeek | the quick answer is configur ip dialing= yes and enter the ip |
| 16:17.21 | dragonius | kk |
| 16:17.22 | dragonius | thx sir |
| 16:17.31 | Slainte | bjohnson, if it has 20 channels, I wonder if it is using OFDM |
| 16:17.33 | bjohnson | Slainte: I don't have it here .. I'll check at lunch. Stay tuned to this channel for more info this aft |
| 16:17.53 | bjohnson | Slainte: I'm guessing at that 20 channels .. I think it was something like that |
| 16:18.08 | Slainte | If it is using OFDM, you could even use each channel twice, in ideal conditions |
| 16:18.10 | *** join/#asterisk dfallen (~steve@84-104-133-239.cable.quicknet.nl) |
| 16:18.16 | Zeeek | ManxPower "it is impossible to do what you want within the current tech..." is what we tell our customers our competitors will tell them |
| 16:18.22 | Slainte | but I have never seen ideal conditions :) |
| 16:18.26 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 16:18.27 | *** part/#asterisk HiTech69 (~hitech@155-105.202-68.tampabay.rr.com) |
| 16:18.32 | Zeeek | then we tell them "that is stupid fucking idea!" |
| 16:18.45 | dfallen | hey there :D I got a question about voicemail.conf.. is there a way to specify multiple e-mail addresses to send e-mail notification to, I want to set about 3?? |
| 16:18.45 | ctooley | Anyone else having problems with Polycom SIP phones and CVS HEAD lately? |
| 16:18.47 | ManxPower | Zeeek: Our first Asterisk deployment used analog phones. It was a user interface disaster. |
| 16:19.04 | Zeeek | how so Manx? |
| 16:19.10 | ManxPower | Zeeek: Granted, with smart users it would not be a problem. |
| 16:19.20 | bjohnson | datareactor: true integration with any PBX totally depends on the PBX abilities. Most people stick asterisk in front of the PBX and provide connection to the telco + provide connection to the PBX. Then use * to route calls over voip as desired |
| 16:19.21 | ManxPower | Zeeek: They have problems with call waiting, transfers, three-way calling, etc. |
| 16:19.33 | Slainte | I need to get "comfort" noise working. |
| 16:19.44 | Zeeek | Manx those are feature that require a $10000 pbx, remember? :) |
| 16:19.44 | ManxPower | Slainte: Not if you are using asterisk. |
| 16:20.03 | ManxPower | Zeeek: Yes, but with PBXs the way you access the features is more familiar to users. |
| 16:20.05 | Slainte | Manx, Asterisk has no Comfort noise? |
| 16:20.13 | ManxPower | Slainte: Correct. No VAD either. |
| 16:20.33 | Zeeek | dfallen set up an alias in the sendmail file |
| 16:20.43 | Slainte | Why is it that hard to send? |
| 16:20.49 | dfallen | ooh right. thanks |
| 16:20.50 | *** part/#asterisk dfallen (~steve@84-104-133-239.cable.quicknet.nl) |
| 16:20.52 | bjohnson | ManxPower: you could maybe do a mixture of wifi phones, fxs + 900Mhz cordless, fxs + 2.4GHz cordless, fxs + 5Ghz, and wired phones with bluetooth headsets |
| 16:21.02 | Zeeek | you can send two emails using the pager email |
| 16:21.19 | Slainte | Or you can setup an email list, and send it to the one addy on the list |
| 16:21.27 | Slainte | probably better management anyway |
| 16:21.37 | ManxPower | bjohnson: I would rather remove myself from consideration for the project. |
| 16:21.45 | Nugget | wifi and 2.4ghz cordless will shit all over each other. |
| 16:21.55 | Nugget | but that idea is fundamentally ridiculous. |
| 16:21.59 | ManxPower | They also wanted a WiFi network for that office. |
| 16:22.01 | dragonius | Zeeek, i cant find the menu entry for "ip dialing"... |
| 16:22.01 | dragonius | ;| |
| 16:22.03 | Inv_arp | how can i tell which version of * this box is running? |
| 16:22.18 | Nugget | it would be a political and training nightmare to have a 40 person office with five different types of phone. |
| 16:22.20 | ManxPower | As I said, the office manager seems to have a phobia of wires. |
| 16:22.22 | Slainte | especially if they are VTECH phones. Some of them put out at a watt, and will take your put your signal to noise ratio in the toilet |
| 16:22.29 | Zeeek | ManxPower I really wish it was possible to imitate an old four line phone with the lights and hold button mechanism and all. |
| 16:22.43 | ManxPower | Zeeek: Good SIP phones seem to do that. |
| 16:22.52 | dragonius | anybody knows where the ip-dialing configuration is hiding in xlite? |
| 16:22.52 | mquin | Inv_arp: show version from the * console |
| 16:22.57 | ManxPower | We will use Polycom IP 300's unless the SIPura phone works out. |
| 16:22.58 | Zeeek | they have leds lit for lines in use? |
| 16:23.15 | bjohnson | ManxPower: does anybody actually get a desk? |
| 16:23.18 | Zeeek | dragonius stop being lazy and look it up in the doc |
| 16:23.24 | ManxPower | Zeeek: In theory, but I've never tried it. Zeeek see the "hint" priority in Asterisk |
| 16:23.35 | ManxPower | dragonius: The FNORDS site has links to two X-Lite config docs |
| 16:23.36 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 16:23.37 | ctooley | ManxPower, The bitch with 300's is the mono speakerphone |
| 16:23.42 | Zeeek | I don't have any worth SIP phones to try it with |
| 16:23.47 | bjohnson | ManxPower: what about regular phones and the users dials and extension to tell the system which handset their at? |
| 16:23.55 | ManxPower | ctooley: They pack the people in so tight that lack of speakerphone is a good thing. |
| 16:24.10 | ManxPower | bjohnson: I idea. |
| 16:24.16 | Zeeek | what use are links if no one clicks on them? What use are balls if no one squeezes them? |
| 16:24.27 | Zeeek | oh shit a cust call.... damn it! |
| 16:25.14 | ctooley | Well, that and that crappy display make me recommend the 500's |
| 16:25.23 | Zeeek | I hate when customers interrupt my useless comments |
| 16:26.27 | Zeeek | that customer was having problems sending an 8 meg file. I sent her to http://sendit.com |
| 16:26.38 | Zeeek | great tool and free |
| 16:26.57 | tzanger | whoa |
| 16:27.05 | tzanger | I just got a floating point exception with app_rxfax |
| 16:27.09 | Zeeek | bjohnson - I have those findme extension in asterisk |
| 16:27.24 | Zeeek | you of the "I use spandsp all the time" fame ? |
| 16:28.00 | *** join/#asterisk jtodd (~jtodd@66.165.175.229) |
| 16:28.20 | tzanger | uh |
| 16:28.26 | tzanger | is it normal for a bt to have over 7000 entries? |
| 16:28.38 | bjohnson | Zeeek: huh? |
| 16:29.18 | Zeeek | didn't you say phones that you dial to say where ayou are? |
| 16:29.46 | Zeeek | when I go to the office I dial a three digit code and it says my extension |
| 16:29.47 | bjohnson | oh .. slightly different idea |
| 16:29.56 | Zeeek | anyone calling the last two digits will ring that phone |
| 16:30.07 | Zeeek | it's been done a lot on asterisk - simple to do |
| 16:30.16 | Slainte | I must say I am well impressed with these Poly600's |
| 16:30.59 | bjohnson | Zeeek: I was suggesting to ManxPower that the cordless office consider using corded phones and create an asterisk app to allow an employee to dial in, input a PIN, and then input the handset extension number of the phone they'd like their calls to be forwarded to |
| 16:31.57 | Zeeek | bjohnson a better idea is to buy a GSM gateway and let them use the cells |
| 16:31.57 | ManxPower | bjohnson: We may do that anyway. |
| 16:31.57 | Zeeek | 40 cordless phones is really not a good way to go |
| 16:31.57 | bjohnson | that would likely handle most of the needs |
| 16:32.31 | Nugget | use rfid and automate it. :) |
| 16:32.39 | bjohnson | Zeeek: I thought a Gsm gateway really just meant you could call a cellular network provider and all the hadsets then needed individual cell. service package |
| 16:32.40 | bprice20 | bkw_ need smore help |
| 16:32.45 | bprice20 | bkw_ |
| 16:32.58 | bprice20 | bkw_ how the fuc am i going to do call forwarding |
| 16:33.03 | Nugget | smore help? easy. first you toast the marshmallows. |
| 16:33.12 | bprice20 | nice :) |
| 16:33.21 | bjohnson | Nugget: yes. or callerid if someone is calling in from outside |
| 16:33.23 | Zeeek | I think the gsm gateway can look like a private netwrok |
| 16:33.30 | Nugget | bjohnson: I think you misunderstood my suggestion. |
| 16:33.35 | bprice20 | ayone mind helping me implement call forwarding |
| 16:33.39 | tzanger | where's coppice when you need him :-) |
| 16:33.46 | twisted[work] | yee haw. these storms are fun |
| 16:33.55 | jlewis | anyone know whats involved in moving a g729 license key from one server to another? |
| 16:34.12 | bjohnson | bprice20: just Dial a different number |
| 16:34.22 | ManxPower | jlewis: It should work if you also move the network card. |
| 16:34.40 | ManxPower | If you can't move the network card, then you'll have to contact Digium support. |
| 16:34.56 | Casper_UA | ~seen rkarlsba |
| 16:34.58 | jbot | i haven't seen 'rkarlsba', Casper_UA |
| 16:34.59 | bjohnson | Nugget: ahh .. rfid chips on the employees? And GIS to determine which phone is the closest? |
| 16:35.04 | Nugget | right |
| 16:35.10 | bprice20 | bjohnson yeah I know I can so that but I mean how will I add it to a dialplan |
| 16:35.11 | eKo1 | wtf!? why do i have 1092636494 as the dipostion in my db. |
| 16:35.18 | jlewis | ManxPower: can't do that...we have extra licenses (2 keys) on serverA, and we want to move one key to serverB because we want licenses now and would rather not wait the 24h digium usually takes |
| 16:35.25 | bprice20 | so that it will ring both or 1 then the other |
| 16:35.31 | bprice20 | waith i think i got it |
| 16:35.34 | tzanger | jlewis: you're pretty much screwed then |
| 16:35.34 | Nugget | personally, I think a solution that requires people to notify asterisk of their location is horrible. but I suppose you can coerce people to do it. |
| 16:35.47 | bjohnson | bprice20: use & in the Dial command to ring more than one |
| 16:35.50 | ManxPower | jlewis: It sucks to be you. Moving licenses is not something you can do without help from Digium. |
| 16:35.54 | Casper_UA | eKo1: what's wrong with dispositions? |
| 16:35.59 | Nugget | I would expect that in reality such a system would just end up being a mess of misdirected calls |
| 16:36.04 | ManxPower | Nugget: I agree with that. |
| 16:36.11 | mquin | 'if you move the card' - I guess it's locked to the MAC address, presuably if you've got a card that's reprogramable you can just switch the MAC addresses? |
| 16:36.22 | maruz | Nugget: are u thinking to phones far miles from employees? |
| 16:36.22 | jlewis | tzanger: whys that? AFAIK, you can reregister a key once or twice without digiums help...but I assume to do this properly, I need to somehow remove the key from serverA |
| 16:36.23 | bprice20 | bjohnson, what I', really asking is how to implement it into my dial plan |
| 16:36.33 | bprice20 | bjohnson care to help me there |
| 16:36.36 | ManxPower | mquin: The Digium code binary registers and locks to the MAC address |
| 16:36.44 | bjohnson | Dial(phone1 & phone2) |
| 16:36.52 | bjohnson | rings 2 phones at once |
| 16:37.02 | ManxPower | jlewis: Sre you trying to MOVE the license or COMBINE the license? |
| 16:37.08 | tzanger | jlewis: well unless you can get a hold of someone I imagine it's not going to happen |
| 16:37.34 | Nugget | maruz: I have no idea what you just tried to ask me. |
| 16:37.36 | tzanger | was not having a few spare licenses unbound and unregistered worth the $20 you saved? |
| 16:37.42 | bjohnson | Dial(phone1 & phone2) .. not sure what the caller gets if one phone is busy |
| 16:37.43 | Zeeek | email digium - I had to resolve something similar once and it's not that big a deal. The hardest part is getting an answer |
| 16:37.45 | jlewis | serverA has 2 separate 23 channel license keys, I just want to steal one of those away for serverB |
| 16:37.47 | bprice20 | bjohnson http://pastebin.ca/4143 |
| 16:38.04 | ManxPower | I was not aware that you could have two licenses on the same server. |
| 16:38.18 | Zeeek | jlewis - that's less easy I think |
| 16:38.19 | *** join/#asterisk fantomax1 (~fanto@81.208.114.250) |
| 16:38.23 | fantomax1 | hi all |
| 16:38.27 | ManxPower | jlewis: Does show g729 <tab> show all 46 licenses? |
| 16:38.29 | fantomax1 | I have a question |
| 16:38.44 | jlewis | yeah |
| 16:38.44 | Casper_UA | fantomax1: about spam calls again? |
| 16:39.00 | tzanger | jlewis: perhaps I'm being a little nasty but I really dislike when people say things like what you just described and then try to say it's somehow Digium's fault or a failing of Digium that you can't do $WEIRD_ADMIN_TASK immediately when you didn't provide for such a task yourself |
| 16:39.07 | fantomax1 | spam ? |
| 16:39.08 | fantomax1 | no |
| 16:39.14 | fantomax1 | Asterisk and Proxy |
| 16:39.30 | Casper_UA | fantomax1: what proxy? |
| 16:39.31 | jlewis | not saying anythings digiums fault...just trying to deal with management saying we need a key in 2 hours |
| 16:39.49 | jlewis | I'll call them and ask about moving a key |
| 16:40.00 | fantomax1 | I need to configure a proxy pointing to asterisk |
| 16:40.07 | bjohnson | spectravoice has an unlimited plan that may be suitable for CDN users .. but doesn't include world calls .. just NA |
| 16:40.13 | fantomax1 | do I have to set something on asterisk ? |
| 16:40.29 | Casper_UA | fantomax1: what protocol? |
| 16:40.37 | fantomax1 | sip |
| 16:41.13 | bjohnson | bprice20: what is exten => _+NXXNXXXXXX,1,Dial(IAX2/${EXTEN:1},7,r)? Go to the wiki and read about the dial command .. or use show application dial in the * cli |
| 16:41.38 | bjohnson | bprice20: you need to specify at least one phone |
| 16:42.47 | dragonius | SJPhone is way easier to use :) |
| 16:43.39 | PoWeRKiLL | what parameter do I have to use for HDSL in zaptel.conf please ? |
| 16:44.48 | bjohnson | bprice20: what does the + do? I haven't seen that one before and can't find info on it |
| 16:45.09 | dragonius | how can i make * forward a call to a softphone in my LAN |
| 16:46.19 | bjohnson | dragonius: make an extenion that gets the incoming call and use the Dial command to ring the softphone |
| 16:46.20 | bprice20 | bjohnson |
| 16:46.27 | bprice20 | thats basically for level 3 |
| 16:46.41 | *** join/#asterisk ast_freak (~yircme@hades-out.universalsystems.net) |
| 16:46.42 | bprice20 | our carrier needs the + added before all numbers |
| 16:47.06 | *** join/#asterisk jcims (~rrich@cpe-69-135-121-57.columbus.rr.com) |
| 16:47.09 | bjohnson | and incoming calls come in that way? |
| 16:47.47 | bkw_ | ok riddle me this |
| 16:48.03 | bkw_ | has anyone ever had one way audio on a pots line? |
| 16:48.16 | bkw_ | not running thru asterisk |
| 16:48.18 | tzanger | bkw_: only if the mic or earpiece were bad |
| 16:48.23 | tzanger | or the mute button is stuck |
| 16:49.05 | bkw_ | thats not it |
| 16:49.15 | bjohnson | bprice20: here is a multi phone dial .. exten => s,3,Dial(${PHONE1}&${PHONE2},15,Ttm) where PHONE1=SIP/2201 |
| 16:49.35 | bkw_ | now the pots line wont take DTMF |
| 16:49.39 | bjohnson | bprice20: I haven't tested what happens if one of the two phones is in use |
| 16:49.41 | Nuttah | bj managed to find the issue |
| 16:49.49 | dragonius | reboot |
| 16:49.49 | dragonius | brb |
| 16:49.52 | Nuttah | seemed to be an extension problem |
| 16:49.57 | *** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net) |
| 16:50.24 | bjohnson | Nuttah: glad to hear .. call didn't go where you thought? |
| 16:50.25 | Nuttah | s,1,wait,1 line caused it |
| 16:50.33 | *** join/#asterisk PTG123 (~PTG123@ip68-106-19-249.ph.ph.cox.net) |
| 16:50.59 | Nuttah | no it did, it just got effected by something I didnt think was causing it |
| 16:51.13 | Nuttah | to be honest I cant see why it caused such and issue |
| 16:51.25 | Nuttah | whats you wait line ? |
| 16:51.28 | Nuttah | your |
| 16:51.34 | PTG123 | ~seen czero |
| 16:51.36 | jbot | czero <~me@CPE0090f800c5b0-CM001225704b6e.cpe.net.cable.rogers.com> was last seen on IRC in channel #asterisk, 23h 49m 4s ago, saying: 'bjohnson correct from canada'. |
| 16:51.43 | *** join/#asterisk florz_ (nobody@Iac55.i.pppool.de) |
| 16:51.46 | bprice20 | bjohnson thx that helps |
| 16:52.23 | jcims | does asterisk have the capability of literally 'switching' a PSTN call...so the two lines are physically connected rather than being codec'ed? |
| 16:52.44 | jcims | (obviously assuming pstn in and out) |
| 16:52.47 | bkw_ | no |
| 16:52.54 | bjohnson | Nuttah: same but Wait(1) |
| 16:53.01 | Nuttah | k lemme try that |
| 16:53.28 | bjohnson | jcims: depends on what hardware is neing used |
| 16:53.38 | *** join/#asterisk Yoda-BZH (~yoda-bzh@80.125.209.17) |
| 16:53.46 | fearnor | jcims: sort of. see dacs functionality in zaptel |
| 16:54.26 | jcims | does zaptel do ISDN or just pots? |
| 16:54.41 | bjohnson | jcims: * can let 2 SIP devices talk directly to each other once the call has been connected for instance. If you had a multi fxo port sip device that supported it .. you could achieve what you ask |
| 16:54.48 | jcims | (not sure if analog == pots, meant analog) |
| 16:55.05 | Yoda-BZH | Le bonjour je vous souhaite / Hi ppl |
| 16:55.39 | Nuttah | i think I was a tad premature |
| 16:55.40 | Zeeek | I think you coders should build commands in like they do at Mozilla so you can type "sip show peerz" and see some wacky message |
| 16:55.41 | jcims | i'm actually thinking about colocating a box where i can get free PSTN long distance for pretty cheap, then use a call back setup to bridge inbound/outbound calls from my (and my fam's) home phones |
| 16:55.44 | bjohnson | jcims: you can get hardware that will work with asterisk for pots/analog lines, ISDN BRI, and PRI |
| 16:56.14 | jcims | i'm attempting it now using VOIP but the call quality is suffering |
| 16:56.22 | bjohnson | free PSTN long distance for pretty cheap? |
| 16:56.32 | jcims | it's $50 a month per line. |
| 16:56.52 | jcims | so $100 per month split 4 ways or so |
| 16:56.53 | bjohnson | jcims: you're missing the concept of voip I think |
| 16:57.19 | jcims | no, i get it, but i can't get my brother in law doing voip when he can't get much more than a 23k connection with his dialup |
| 16:57.23 | *** join/#asterisk IOscanner (~IOscanner@c-24-0-186-72.client.comcast.net) |
| 16:57.24 | jcims | we're all in the b00nies |
| 16:57.28 | bjohnson | jcims: look at Broadvoice that has unlimited world plan for as little as $20/mo |
| 16:57.30 | Faithful | I have no messaging sound what's the most common cause? |
| 16:57.46 | jcims | actually i'm not in the boonies, but everyone else is |
| 16:58.00 | jcims | but i have sprint, who can't seem to figure out how to setup a reasonble long distance plan |
| 16:58.05 | Zeeek | Faithful CLI messages ? |
| 16:58.09 | bjohnson | jcims: just get a voip DID that is local to them .. or a 800 number |
| 16:58.17 | jcims | there are none |
| 16:58.21 | jlewis | ok...just talked to digium...I can just re-register the old key on the new server, and they apparently contact the old server and tell it "you just lost this key" |
| 16:58.29 | Faithful | Zeeek: none |
| 16:58.33 | jcims | 800 number might work...not sure |
| 16:58.47 | bjohnson | Faithful: I don't understand the question |
| 16:58.48 | Zeeek | try nufone |
| 16:59.02 | bjohnson | jcims: ^^ |
| 16:59.02 | jcims | i'm just thinking put the * number on speed dial, * gets the call and calls the user back in 10 seconds or so, with a dialtone |
| 16:59.08 | Zeeek | ICH used to have $10 800 # but now it's more |
| 16:59.10 | jcims | i have nufone now... |
| 16:59.17 | jlewis | you get to do that once, after which you have to have them reset your license attempts |
| 16:59.21 | Zeeek | nufone does 888 toll free DID |
| 16:59.24 | jlewis | if you want to do it again |
| 16:59.37 | tzanger | Zeeek: u.s. only -- I need to get a good canadian 800# terminator |
| 16:59.43 | jcims | hmmm |
| 16:59.43 | jcims | didn's ee that |
| 16:59.46 | jlewis | so I can just move the key and then buy a new one for the server I stole it from |
| 16:59.57 | Zeeek | jcims you want Canada? |
| 17:00.00 | bjohnson | jcims: get a 800 and forget the callback .. just do auth from callerid info |
| 17:00.06 | Zeeek | tzanger ya seen calgarytelecom? |
| 17:00.11 | tzanger | no |
| 17:00.14 | *** join/#asterisk Mike9 (~sturdee@sun.mikesweb.com) |
| 17:00.16 | tzanger | what have they got for rates |
| 17:00.17 | Zeeek | I have a test acct |
| 17:00.25 | Zeeek | I dunno I'm not in Canada |
| 17:00.31 | jcims | don't i pay more for the 800 number tho? |
| 17:00.32 | Makenshi | Can you signup to Broadvoice in UK? |
| 17:00.38 | tzanger | Makenshi: why would you? |
| 17:00.38 | Zeeek | they are asterisk friendly tho |
| 17:00.39 | bjohnson | jcims: user calls 800 number .. it connects to * .. if callerid is ok .. it answers and gives dialtone for call out |
| 17:00.47 | tzanger | Zeeek: haha okay |
| 17:00.56 | tzanger | bjohnson: I'm doing the exact same thing |
| 17:01.04 | jcims | yeah, but there's usage on the 800 number |
| 17:01.05 | bjohnson | jcims: you going to pay about $0.02 x 2 / minute per call anyway |
| 17:01.05 | tzanger | if callerid is empty or unrecognized, asks for an accountcode |
| 17:01.06 | Makenshi | tzanger, because it's a good plan |
| 17:01.14 | tzanger | Makenshi: but the service blows hairy goatass |
| 17:01.20 | Zeeek | tzanger http://calgarytelecom.com/ |
| 17:01.28 | Makenshi | oh, well ill just ignore it then :) |
| 17:01.29 | tzanger | Zeeek: danke |
| 17:01.35 | jcims | if i use a usage based account... |
| 17:01.48 | Makenshi | i have a payg account |
| 17:01.50 | Faithful | bjohnson: I have no sound coming from asterisk to the phones |
| 17:01.50 | Zeeek | but I see no DID yet : http://calgarytelecom.com/?sMenu=main&spdrButtons=information.service_plans |
| 17:01.53 | jcims | i can get pstn with free long distance to the 48 states for $50 flat a month |
| 17:01.57 | Makenshi | 1p/min to most countries, it does me |
| 17:02.01 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 17:02.03 | Makenshi | with babble |
| 17:02.11 | tzanger | yeah I don't see 'em either |
| 17:02.14 | *** join/#asterisk xelpud (irc@CPE0080c6efe7fc-CM0012254495cc.cpe.net.cable.rogers.com) |
| 17:02.26 | bjohnson | Makenshi: Faithful (from AU) signed up to bv yesterday |
| 17:02.30 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 17:02.35 | Zeeek | are there no North American tolld free # ? |
| 17:02.52 | Makenshi | thanks tzanger, bjohnson |
| 17:02.57 | *** join/#asterisk wizhipp1 (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 17:03.06 | bjohnson | Faithful: from what I read .. that is often a codec issue |
| 17:03.20 | jcims | anyone know how to get a shell from bitchx? :P |
| 17:03.24 | tzanger | hahaha calgarytelecom sits on efnet #calgarytelecom |
| 17:03.37 | Zeeek | tzanger http://webstore.sprint.ca/webstore/jsp/display.do?jsp=/jsp/ld_14a.jsp |
| 17:03.38 | bjohnson | jcims: so you would need 2 of those |
| 17:03.47 | Zeeek | tell em I sent ya ! |
| 17:03.55 | Zeeek | calgary not sprint :) |
| 17:04.06 | jcims | bjohnson: yep. it's probably an illconceived plan, but yeah, 2x$50 a month for free long distance for anyone i know |
| 17:04.16 | jcims | bjohnson: one at a time, of course :) |
| 17:04.39 | bjohnson | jcims: you could get a $20 unlimited for outgoing and a 800 at $0.02 /minute for incoming .. let's see .. ($100 - $20)/0.02 = 4000 minutes |
| 17:04.58 | JerJer | no such thing as unlimited |
| 17:05.05 | jcims | i know.. ) |
| 17:05.18 | Zeeek | JerJer did you get my mail about the 10 minute weirdness ? |
| 17:05.21 | bjohnson | jcims: of course that would also include one DID that could be used by some people to avoid the $0.02/minute |
| 17:05.47 | bjohnson | JerJer: I can never find the "real" limits in the web sites |
| 17:05.58 | Zeeek | the mimit is when you get their letter |
| 17:06.05 | jcims | ^^exactly |
| 17:06.09 | bjohnson | jcims: the voip option could be multiple concurrent calls |
| 17:06.12 | Zeeek | saying you are "abusing" |
| 17:06.34 | Faithful | does asterisk actually use the sound card to process sound to the phones? |
| 17:06.36 | jcims | something about two lines being used at 70% duty cycle for a month from a residential account :) |
| 17:06.40 | Zeeek | Faithful no |
| 17:06.50 | Faithful | hmmm. |
| 17:07.11 | tzanger | thanks Zeeek |
| 17:07.15 | Zeeek | so don't load the modules if you don't have a sound card |
| 17:07.33 | Zeeek | tzanger you may wanna talk to calgary - they seem nice enuf |
| 17:07.34 | *** join/#asterisk VerMan (~none@201.130.4.100) |
| 17:07.39 | VerMan | hello |
| 17:08.03 | VerMan | we're using FireFly soft phone with asterisk |
| 17:08.05 | redder86 | Anyone here have access to current new Polycom firmware? |
| 17:08.11 | Zeeek | "customer care" is such a crock ! |
| 17:08.30 | VerMan | how can we configure asterisk in order to make calls from one firefly to another (let's say we have 50 softphones) |
| 17:08.32 | *** join/#asterisk telme_ (~trillian@sta-207-174-139-178.rockynet.com) |
| 17:08.39 | Zeeek | "customer <--- we don't give a shit" is much more accurate |
| 17:08.59 | jcims | VerMan, is it hammering your cpu? |
| 17:09.11 | VerMan | i don't think so |
| 17:09.27 | jcims | i'm trying to use FireFly and a) it sets itself to abovenormal priority and b) sits there and runs at 99% |
| 17:09.41 | jcims | so i start it and within 10 seconds my system is hammered. |
| 17:09.48 | jcims | anyway, you had a question? :) |
| 17:10.07 | VerMan | so Firefly is crap? |
| 17:10.23 | jcims | Zeeek: the problem with 'customer care' is that it costs $$$$, but no customers want to pay the premium |
| 17:10.32 | ast_freak | VerMan, get a copy of X-lite |
| 17:10.43 | Zeeek | Yeah but when you are paying, it sucks |
| 17:10.45 | jcims | VerMan: i think it's got a pretty slick little interface, it's just got some questionable behavior on my system |
| 17:11.07 | VerMan | my question is how to configure asterisk to run n-softphones (be firefly or x-lite= |
| 17:11.26 | VerMan | how to configure their extensions, and with the same user/pass |
| 17:11.27 | ast_freak | VerMan, have you looked at the Wiki? |
| 17:11.31 | jcims | Zeeek, well, _thats_ because the customer service _used_ to be good, but then some PHB decided to lay off half of the service staff to 'keep costs down', but didn't drop prices :) |
| 17:11.36 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 17:11.41 | VerMan | I'm at the wiki but it's dam huge |
| 17:12.06 | ast_freak | VerMan, points of interest are: |
| 17:12.15 | VerMan | sip.conf? |
| 17:12.19 | Zeeek | VerMan - the nice thing is while you're looking to solve one problem, you see a lot of info you'll need to know about other stuff |
| 17:12.26 | jcims | anyone using the AMP (AMD?) web interface to manage *? |
| 17:12.32 | Inv_arp | how can i tell which version of * this box is running? |
| 17:12.38 | VerMan | that's the prob |
| 17:12.40 | ast_freak | http://voip-info.org/wiki-Asterisk+phone+firefly |
| 17:13.00 | jcims | VerMan, i'm still having that prob...info overload |
| 17:13.02 | ast_freak | http://voip-info.org/wiki-Asterisk+phone+xten+xlite |
| 17:13.34 | ManxPower | and/or read the two X-Lite config docs linked on the fnords site. |
| 17:13.42 | puzzled | anyone know how much that Sipura SPA-841 costs? |
| 17:13.47 | VerMan | can you change GUI or look&feel in x-lite? |
| 17:13.52 | ManxPower | puzzled: under $100 |
| 17:13.56 | puzzled | nice |
| 17:14.14 | ast_freak | VerMan, I don't think so. |
| 17:14.19 | *** join/#asterisk D1ng0 (~dingo@130.205.8.67.cfl.rr.com) |
| 17:14.29 | VerMan | well, that's another prob |
| 17:14.57 | ManxPower | Verliba: All softphones suck. X-Lite seems to suck less than others, but they all still stuck. |
| 17:15.01 | ManxPower | and suck too |
| 17:15.10 | puzzled | firefly sucks too? |
| 17:15.12 | *** join/#asterisk HitTop (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 17:15.14 | Faithful | correction no sound out over ISDN |
| 17:15.15 | Zeeek | do not! |
| 17:15.18 | ManxPower | puzzled: ALL of them do. |
| 17:15.21 | Zeeek | do not! |
| 17:15.22 | puzzled | heheh |
| 17:15.26 | Zeeek | do not! not not not! |
| 17:15.34 | VerMan | at least firefly is GUI configurable |
| 17:15.34 | D1ng0 | do to to to to to |
| 17:15.36 | D1ng0 | :) |
| 17:15.40 | Zeeek | notnotnotnotnotnotnotnoto |
| 17:15.41 | VerMan | and that's what we need |
| 17:15.44 | ManxPower | For X-Lite and Firefly the suckage actually seems to come from Windows and PC hardware, rather than a moronic programmer. |
| 17:16.06 | puzzled | surely the softphone extension for emacs doesn't suck |
| 17:16.09 | Zeeek | X-Lite is excellent and comes with a detailed document which most people have never looked at |
| 17:16.20 | ast_freak | ManxPower, touchette. |
| 17:16.23 | *** join/#asterisk jemfinch (~jfincher@dhcp065-024-075-056.columbus.rr.com) |
| 17:16.42 | Zeeek | although... there are secret handshakes you need to know to get X(-Lite to work |
| 17:16.54 | HitTop | Hi~ inside /usr/lib/asterisk/modules i have a file called app_intercom.so... |
| 17:16.54 | HitTop | but under asterisk console when i typed show applications, its not there.. how do I load intercom.so when i run asterisk? |
| 17:17.04 | Zeeek | and all ManxPowers servers are hard coded not to work in X-Lite |
| 17:17.12 | ast_freak | Zeeek, what are you talking about... secret handshakes. |
| 17:17.28 | Zeeek | worldwide masonry ? |
| 17:17.39 | ManxPower | HitTop: app_intercom is schedule to be removed I think. Replaced by chan_alsa and chan_oss w/auto answer config |
| 17:17.41 | Zeeek | you gotta know the secret shit |
| 17:18.16 | *** part/#asterisk wizhipp1 (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca) |
| 17:18.35 | tzanger | Zeeek: yeah I'm emailing them as we speak |
| 17:19.09 | Zeeek | are you near calgary? |
| 17:19.29 | tzanger | no, I'm in Ontario, two provinces east |
| 17:19.32 | Zeeek | Mike ROberts was who I met here - they do IAX |
| 17:19.40 | HitTop | but is it chan_alsa and chan_oss only available in newer version of asterisk? Im using v1-0. Or anyone know of any intercom paging solution using /dev/dsp? |
| 17:20.04 | Zeeek | accessible from Canada and the USA." |
| 17:20.06 | ManxPower | HitTop: No, those channels are available in 1.0.x too. |
| 17:20.22 | ManxPower | HitTop: We use chan_alsa for overhead paging in 1.0.x |
| 17:20.50 | *** part/#asterisk jemfinch (~jfincher@dhcp065-024-075-056.columbus.rr.com) |
| 17:20.59 | Zeeek | tzanger they have an 877 number that is good for US/Canada |
| 17:21.11 | HitTop | thx.. I will find more info for chan_alsa.. thank you manxpower! |
| 17:21.12 | Zeeek | not sure what it's ofr |
| 17:21.18 | tzanger | Zeeek: yeah email is easier for me |
| 17:22.07 | Zeeek | if you don't get an answer, try mike -at- the domain .com - he'll likely let you set up an IAX test acct |
| 17:22.28 | VerMan | how can you assign different extensions to different installations of firefly? |
| 17:22.59 | *** join/#asterisk dragonius (~fr@pD95F2857.dip.t-dialin.net) |
| 17:23.05 | dragonius | how do i disable the CAPI module? |
| 17:23.45 | Zeeek | VerMan |
| 17:23.46 | Zeeek | The answers to the questions you've been asking are probably here: |
| 17:23.46 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 17:23.46 | Zeeek | http://iheavy.com/modules.php?op=modload&name=News&file=article&sid=35&mode=thread&order=0&thold=0 |
| 17:23.46 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 17:23.46 | Zeeek | http://automated.it/guidetoasterisk.htm |
| 17:23.47 | Zeeek | http://www.asteriskdocs.org |
| 17:23.49 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/book1.html |
| 17:23.56 | *** part/#asterisk Casper_UA (~casper@ragu.bestnet.kharkov.ua) |
| 17:24.49 | Nuttah | ok now i have it sussed |
| 17:24.51 | *** join/#asterisk ethzer0 (~ethzer0@d141-233-214.home.cgocable.net) |
| 17:25.36 | VerMan | thanks man |
| 17:26.03 | Nuttah | bj, twas callerid being on that caused it |
| 17:26.19 | Nuttah | and more specifically my BT line not supporting that function |
| 17:26.51 | Zeeek | i hate when that happens |
| 17:27.20 | Nuttah | I've decided just to hate BT |
| 17:27.46 | Zeeek | they do drive on the left, no question :) |
| 17:28.04 | Zeeek | offense |
| 17:28.17 | Zeeek | tyres |
| 17:28.21 | Nuttah | }:P |
| 17:28.28 | Zeeek | i need a strong beer |
| 17:28.35 | Nuttah | not an american one then |
| 17:29.46 | Nuttah | heh |
| 17:29.57 | Zeeek | 500ml to boot :)e nearby fridge |
| 17:30.35 | Nuttah | so nearly a pint.. but not quite :) |
| 17:30.48 | Zeeek | how many lires in a pint? |
| 17:30.53 | Zeeek | litres |
| 17:30.59 | Zeeek | a pint is how many |
| 17:31.06 | Zeeek | 0.55241324 |
| 17:31.08 | Zeeek | ? |
| 17:31.17 | Nuttah | er.. .568 summink i think |
| 17:31.44 | Zeeek | well if I feel the 0,5L is insufficient, Ihave several more in the fridge |
| 17:31.54 | Nuttah | i'd get a jug |
| 17:32.00 | Nuttah | have a try german tankard |
| 17:32.10 | Nuttah | have to try a |
| 17:32.15 | Zeeek | I am winning a domain transfer battle little by little |
| 17:32.22 | Zeeek | deserves a toast |
| 17:32.43 | xelpud | |
| 17:32.44 | Nuttah | toast you need more than one person surely |
| 17:32.47 | Zeeek | I wonder if asterisk consultants are abusing people like domain outfiots do? |
| 17:32.54 | *** join/#asterisk sektor195 (~please@216.86.45.98) |
| 17:33.03 | Zeeek | charging $100/year for domains |
| 17:33.08 | Nuttah | ouch |
| 17:33.10 | fearnor | sure. |
| 17:33.10 | DevilFish | Anyone here work with Polycom 600s ? |
| 17:33.13 | Nuttah | .coms no? |
| 17:33.15 | Godsey | is it possible to read the Polycom IP-300 "MyStatus" from the phones over sip? |
| 17:33.19 | fearnor | why wouldn't we? |
| 17:33.26 | Zeeek | so I told 'em , we'll do it for you for $20 ! |
| 17:33.26 | Godsey | so that it would show up in sip show peer 1234 |
| 17:33.30 | fearnor | its all about whether client feels they are getting a fair value from me or not. |
| 17:33.32 | ArkyLady | Zeeek are people still trying to charge that much? gah |
| 17:33.43 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:33.43 | fearnor | i have people paying 500$/mbit |
| 17:33.49 | fearnor | and people paying 15$/mbit |
| 17:33.54 | fearnor | it is called business. |
| 17:33.56 | DevilFish | Random cutoffs dialing out to the PSTN after like 20-60 mins |
| 17:33.58 | Zeeek | ArkyLady - they can only get away with this until someone honest comes along like we did |
| 17:34.24 | fearnor | "honest"? |
| 17:34.30 | fearnor | zeek: i beg to differ |
| 17:34.37 | ArkyLady | the thing people did with domains that irritated me the most was registering them in the hosting companies name so the customer couldn't move them to a new host |
| 17:34.40 | Zeeek | well to be fair, they offer DNS "free" with the $100 :) hah |
| 17:34.47 | Nuttah | dns is free |
| 17:34.50 | Nuttah | pff :P |
| 17:34.53 | fearnor | you go to an upscale restaurant, and pay 100$ for a bottle of wine and a dinner |
| 17:35.01 | sektor195 | I'm trying to implement agentcallbacklogin without using the pound key to make it as easy as possible, with a trick I found on the wik, which looks like this This removes the need for the user to press the # key on his/her phone |
| 17:35.01 | sektor195 | <PROTECTED> |
| 17:35.01 | sektor195 | <PROTECTED> |
| 17:35.02 | sektor195 | <PROTECTED> |
| 17:35.06 | fearnor | same bottle of wine in a wine store costs 20$ |
| 17:35.13 | sektor195 | but I keep getting invalid login |
| 17:35.14 | Zeeek | that's different |
| 17:35.17 | fearnor | is the restaurant dishonest for selling it to you at 100? |
| 17:35.17 | ManxPower | fearnor: But everyone still pays the same amount for the same service that go to that place. |
| 17:35.24 | sektor195 | brb |
| 17:35.34 | fearnor | that's all a load of crap. |
| 17:35.36 | Zeeek | it is a high markup but (I am in the wine business) it's a lot more complicated than that |
| 17:35.40 | Nuttah | just for freek # is hash not a pound! |
| 17:35.45 | fearnor | like i said, i have people paying 500$/mbit and 15$/mbit |
| 17:35.47 | ManxPower | sektor195: That will not work. Dial does not exit until the call ends. |
| 17:35.48 | PTG123 | anyone know of any good hash table libraries? |
| 17:35.56 | fearnor | different commits, different people, different services. |
| 17:36.04 | Nuttah | freek? zeek even.. gosh darnit |
| 17:36.15 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-139-202.dsl.scarlet.be) |
| 17:36.26 | Zeeek | I could tell you more but in principle, yes wine costs about 2-3 times what you would pay in the store and they don't pay that much |
| 17:36.27 | fearnor | i sell adsi phones for 50$ to some customers and for 200$ to others |
| 17:36.30 | ManxPower | Hmmm...I'm not jittery enough. Time for more coffee. |
| 17:36.38 | Zeeek | but it isn't really on topic here |
| 17:36.44 | Zeeek | hah Manx |
| 17:36.52 | ArkyLady | mmm coffee |
| 17:37.00 | ManxPower | I generally charge the same to all customers giving the same set of variables. |
| 17:37.01 | Zeeek | fearnot and some rest will let me bring my own wine |
| 17:37.16 | Zeeek | no coffee here thx, it's almost 7PM |
| 17:37.17 | fearnor | manx: means you aren't making all money you could be. |
| 17:37.26 | mquin | Zeeek: if you think the restaurant is reaming you on wine, you don't want to knwo the markup they are making on the food... |
| 17:37.29 | *** join/#asterisk rowters (~Drake@201.133.209.110) |
| 17:37.38 | fearnor | frankly, the customers who pay me 500$/mbit would *not* be my customers if i told them it costs 50. |
| 17:37.41 | fearnor | they *want to pay more* |
| 17:37.43 | ManxPower | fearnor: I'm sure that's the case, but I could not sleep well at night if I deliberatly screwed csutomers. |
| 17:37.49 | Nuttah | fearnor: you mean manx is exploiting the most out of his customers? |
| 17:37.55 | Zeeek | mquin in the US that may be true but if the food is good and the service is up to snuff they actuall only make their money on wine |
| 17:38.01 | Nuttah | is=isnt |
| 17:38.14 | fearnor | 'exploiting' is a loaded word |
| 17:38.33 | Zeeek | I know a lot more about this than I do about asterisk but this isn't the time or place |
| 17:38.40 | fearnor | manx: there you go |
| 17:38.48 | fearnor | idiot fees, handholding fees, etc. |
| 17:38.49 | Zeeek | we all charge what the market will bear to some extent |
| 17:38.56 | fearnor | customers who need more handholding pay more |
| 17:39.04 | Nuttah | i'm not getting into this... |
| 17:39.07 | Zeeek | I try to keep it relative to the time I REALLY have to spend |
| 17:39.12 | fearnor | when i feel the customer is very price-sensitive, i try to give competitive rate. |
| 17:39.15 | Nuttah | aye i agree |
| 17:39.24 | Nuttah | but service price stays the same |
| 17:39.28 | Zeeek | in fcat I'd really like to just bill hours like, cough, lawyers |
| 17:39.51 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 17:39.53 | Zeeek | we all give time away to cust we like too, no? |
| 17:39.57 | fearnor | when i know i'm coming in heavily favored and customer will buy no matter what is the price, obviously, i don't have pressure of trying to be competitive |
| 17:40.02 | fearnor | and give 'rack rates' |
| 17:40.13 | Zeeek | I spend hours helping them configure MS Outlook os some shit I never have even used :) |
| 17:40.32 | fearnor | zeeek: when you stop configuring outlook and start doing real business, you'll understand what i mean. |
| 17:40.42 | Nuttah | unless you exagerrating your "rack rates" are 500% to 1000% more expensive |
| 17:40.44 | Zeeek | you know nothing about what I do |
| 17:40.54 | fearnor | i used to hate businesses who don't have upfront prices on website |
| 17:40.59 | Zeeek | that was an example of helping a good cust |
| 17:41.01 | fearnor | now i know why ;) |
| 17:41.09 | Zeeek | heh |
| 17:41.17 | ManxPower | I charge $120/hr, %20 discount for invoices that are over $100 AND are paid within 15 calendar days from the date of the invoice. |
| 17:41.35 | *** join/#asterisk inetdcrsh (~inetdcrsh@res-24-196-244-071.spa.sc.charter.com) |
| 17:41.39 | fearnor | manx: reasonable rates, if you could be busy 30hrs/week :) |
| 17:41.44 | Nuttah | ok laters guys.. this looks like breaking into a "willy waving" competition |
| 17:41.49 | Nuttah | have fun |
| 17:41.51 | inetdcrsh | i'm considering implementing asterisk in a new office my company is opening |
| 17:41.59 | ManxPower | fearnor: I'm busy more like 10 hrs/week and I like it like that. |
| 17:42.13 | inetdcrsh | i'm thinking of using the asterisk pbx software with the dialogic hardware and cisco 7960 phones |
| 17:42.23 | *** join/#asterisk dolson_ (~dana@Sudbury-HSE-ppp3984249.sympatico.ca) |
| 17:42.28 | inetdcrsh | my other option is the 3com nbx system |
| 17:42.31 | ManxPower | I live simply, have money left over for fun stuff, and am not working my ass off. |
| 17:42.40 | Zeeek | what fun stuff? |
| 17:42.45 | inetdcrsh | can any of you offer me any feedback as to the merits of the asterisk setup versus the 3com? |
| 17:42.47 | fearnor | inet: you don't get 'good value' from using dialogic with asstorisk |
| 17:42.53 | Nuttah | prossies booze drugs.. you know :P |
| 17:42.59 | ManxPower | Zeeek: toys of various stuff, some limited travel, spending time with my lovers, etc. |
| 17:42.59 | inetdcrsh | care to elaborate |
| 17:43.00 | Zeeek | the usual. ok. |
| 17:43.01 | fearnor | inet: expandability and feeling of control. ;) |
| 17:43.11 | sektor195 | ManxPower: Is there a solution that will work that will allow me to do the same thing? |
| 17:43.17 | fearnor | inet: if you want shit to "just work" - use 3com nbx. |
| 17:43.18 | Zeeek | Manx time is the main thing that they ain't making any more of |
| 17:43.21 | inetdcrsh | i've been a o/s zealot for years... my biggest issue is that phones are a mission critial system |
| 17:43.21 | ManxPower | sektor195: What do you want to do? |
| 17:43.32 | inetdcrsh | and this is a new employer.. |
| 17:43.45 | fearnor | inet: there you go. is this your first asterisk install? does it have to absolutely positively work x days from now? |
| 17:43.45 | inetdcrsh | i don't want to install something that's going to piss people off and have to be replaced with the NBX in a year |
| 17:43.55 | fearnor | inet: if yes, don't try *. |
| 17:43.59 | inetdcrsh | yes, it's my first asterisk install |
| 17:44.06 | inetdcrsh | but, i've got a rather flexible schedule |
| 17:44.14 | fearnor | experiment on a smaller install until you get comfortable with features |
| 17:44.15 | Nuttah | double it |
| 17:44.21 | inetdcrsh | their existing office has a crappy norstar system |
| 17:44.27 | fearnor | deploy in a small[er] environment, let users beat on it etc |
| 17:44.31 | inetdcrsh | they're building a new office and i basically have free reign |
| 17:44.35 | sektor195 | Basically allow agents to login without having to memorize numbers, because the department that we want to implement this feature gets very busy and are not to tech savy |
| 17:44.35 | telme_ | anyone using freebsd? |
| 17:44.43 | sektor195 | I need to make it as simple as possible |
| 17:44.49 | inetdcrsh | really... my biggest question |
| 17:44.53 | ManxPower | sektor195: I have no idea. |
| 17:44.59 | Zeeek | telme_ have used it with * but no hardware |
| 17:45.17 | telme_ | Zeek: hrm, that's what my question is concerning |
| 17:45.19 | inetdcrsh | using the dialogic gear and cisco phones, 1. is it robust enough for production use 2. is it cool enough to impress the PHB's |
| 17:45.20 | sektor195 | is there a way to get that above function to work? |
| 17:45.31 | Zeeek | telme_ isn't it always ? |
| 17:45.37 | telme_ | heh |
| 17:45.41 | Zeeek | look at the mailing list too |
| 17:45.54 | ManxPower | inetdcrsh: Dialogic is poorly supported in Astersik and the drivers are not free. |
| 17:45.54 | inetdcrsh | because.... they're already impressed with the idea on the costs savings basis... if i can deliver a slick solution, then the flexibility to use o/s for future projects will remain |
| 17:46.00 | telme_ | Zeeek: yeah, been throuth that |
| 17:46.01 | inetdcrsh | ok.... what would you use? |
| 17:46.03 | fearnor | inet: dialogic does not give you a good value |
| 17:46.08 | fearnor | use digium cards. |
| 17:46.09 | ManxPower | inetdcrsh: Digium cards |
| 17:46.12 | fearnor | and channel banks |
| 17:46.15 | fearnor | wait wait |
| 17:46.19 | Zeeek | telme_ I don't know who write the FreeBSD drivers |
| 17:46.22 | fearnor | you using cisco phones? why do you need dialogic at all? |
| 17:46.27 | fearnor | cisco phones = ip |
| 17:46.29 | inetdcrsh | i need a way to get lines into the system |
| 17:46.31 | fearnor | dialogic = tdm/analog |
| 17:46.39 | inetdcrsh | and i need to hook fax machines up |
| 17:46.42 | fearnor | if you have a PRI, buy digium cards |
| 17:46.45 | inetdcrsh | no pri |
| 17:46.47 | inetdcrsh | analog lines |
| 17:46.48 | fearnor | chan-t1? |
| 17:46.51 | fearnor | ok |
| 17:46.52 | *** join/#asterisk kryptops (~dsp@eor.nuqe.net) |
| 17:46.54 | ManxPower | fearnor: No. Dialogic has both analog and digital cards |
| 17:46.57 | fearnor | buy digium card and a channel bank :) |
| 17:46.59 | inetdcrsh | not that big of an installation |
| 17:47.05 | inetdcrsh | 8 lines and 14 phones |
| 17:47.07 | kryptops | 'lo all. |
| 17:47.32 | ManxPower | inetdcrsh: If you have analog lines then get 2xTDM400P w/FXO and use SIP phones or get a channel bank and a Digium T-1 card. |
| 17:47.38 | Zeeek | telme_ What hardware ? |
| 17:47.44 | fearnor | inet: adit 600, one fxo card, two fxs cards, 600$, done/done |
| 17:47.45 | inetdcrsh | what sip phones would you recommend? |
| 17:47.51 | sektor195 | ManxPower: So the only way I can get that to work is if I have them login with their password and extension |
| 17:47.54 | ManxPower | inetdcrsh: Polycom |
| 17:47.57 | inetdcrsh | i just looked at the cisco because they're fancy looking and will make the PHB feel cool |
| 17:48.04 | ManxPower | sektor195: I don't know. I DON'T USE QUEUES. |
| 17:48.09 | sektor195 | oh ok |
| 17:48.18 | ManxPower | But I use Dial and Dial does not continue thru the dial plan until it exits. |
| 17:48.44 | sektor195 | anyone here that uses queues that can offer some input. |
| 17:48.48 | kryptops | I've set up asterisk with an IAX connection to FWD, which works fine. I've also got sipgate SIP connections that have PSTN gateways. DTMF isn't being recognised at all, even when I set dtmftype=inband and make sure I'm using either ulaw or alaw codecs. Any ideas as to why? |
| 17:49.02 | ManxPower | inetdcrsh: Cisco 7905G, 7940G and 7960G will work, but does not include power supply or SIP firmware. Both of which are extra cost with the Cisco. |
| 17:49.24 | kryptops | Er, DTMF isn't being recognised on the *PSTN* lines I should say, SIP clients with other dtmf types work fine. |
| 17:49.29 | ManxPower | KryoStoffer: Maybe the other side is set for a different DTMF mode. |
| 17:49.31 | inetdcrsh | my other thing on the cisco, i'm a ccie and have a full access cisco account |
| 17:49.32 | *** join/#asterisk starman (user@user-161.lightlink-ds1-825e-highlanddr-lehi.fiber.net) |
| 17:49.35 | inetdcrsh | i can get the firmware for free |
| 17:49.42 | inetdcrsh | and i get good prices on reman phones + power supply |
| 17:49.49 | ManxPower | inetdcrsh: Yes. However, it's still not licensed. |
| 17:49.54 | inetdcrsh | true |
| 17:49.58 | redder86 | Polycom's are extremely difficult to configure, in comparison to most other phones I've tried. |
| 17:50.04 | fearnor | gotte juarez and BSA bustin' yo doors?! |
| 17:50.04 | inetdcrsh | for testing purposes though, it's fine |
| 17:50.06 | telme_ | Zeeek: seems the t100 has some drivers for freebsd but not sure whether to install libpri from ports or cvs |
| 17:50.19 | inetdcrsh | what polycom phone do you prefer? |
| 17:50.25 | redder86 | Plus, the Polycom firmware isn't downloadable just anywhere. Only a reseller can get it. |
| 17:50.26 | ManxPower | inetdcrsh: IP500 |
| 17:50.37 | Zeeek | telme_ you've obviously seen http://www.voip-info.org/wiki-FreeBSD+zaptel |
| 17:50.52 | inetdcrsh | hey.. that's a slick phone |
| 17:51.03 | inetdcrsh | and cheaper then the cisco |
| 17:51.16 | redder86 | inetdcrsh: and from what I hear, better than Ciscos |
| 17:51.20 | ManxPower | I think Cisco and Polycom are the best phones for Asterisk. |
| 17:51.24 | redder86 | although I've never used a Cisco |
| 17:51.37 | inetdcrsh | ok.. |
| 17:51.48 | inetdcrsh | but, generally speaking... you guys think that it's a workable solution |
| 17:51.52 | redder86 | ManxPower: got a firmware that's newer than what's on freedomphones' page? |
| 17:51.57 | inetdcrsh | using the digium cards and the polycom phones |
| 17:52.30 | ManxPower | redder86: Yes, but it doesn't seem to fix any problems |
| 17:52.34 | redder86 | inetdcrsh: why wouldn't it work |
| 17:52.39 | telme_ | Zeeek: yeah, and i installed zaptel via svn, which seem to work well, but libpri won't install from cvs, so i wondering if it's okay from ports |
| 17:52.44 | redder86 | ManxPower: what problems are you having? |
| 17:52.45 | ManxPower | inetdcrsh: I would get a Polycom and a Cisco and see how they work. |
| 17:52.57 | Zeeek | talme_ can't say since I didn't need ti |
| 17:52.58 | inetdcrsh | i've just never done it before and i don't want to spend the money just to find out that the result sucks |
| 17:52.59 | inetdcrsh | :-p |
| 17:53.00 | ManxPower | redder86: None, but the release notes do not indicate any significant bugs fixed. |
| 17:53.03 | HitTop | Hi, for chan_alsa module.. I wonder if it would show there in show modules, coz its not for me even though i set load => chan_alsa.so in modules.conf |
| 17:53.15 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
| 17:53.18 | ManxPower | inetdcrsh: If you do not do a lot of R&D and testing your project will fail. |
| 17:53.22 | inetdcrsh | agreed |
| 17:53.23 | Zeeek | heh the beer is kicking in |
| 17:53.27 | inetdcrsh | this is the beginning of R&D |
| 17:53.34 | ManxPower | HitTop: can you use alsa apps outside of Asterisk |
| 17:53.35 | _Vile | mmm beer |
| 17:53.44 | redder86 | ManxPower: I've got a problem where the call rings the phone but there's no way to answer the call. The person picks up the receiver and nothing, presses buttons, nothing. The phone keeps on ringing. |
| 17:53.51 | inetdcrsh | ask everybody for as much info as i can glean before i start buying or compiling anything |
| 17:53.56 | ManxPower | redder86: I've never had that problem. |
| 17:54.17 | redder86 | ManxPower: at the same time, the person can't place calls out, either. They pick up the receiver and get dialtone for a second or two, and then a busy signal. |
| 17:54.20 | Zeeek | redder86 what phone? |
| 17:54.23 | *** part/#asterisk kryptops (~dsp@eor.nuqe.net) |
| 17:54.31 | redder86 | Zeeek: I've had it happen on the IP600 and the IP500. |
| 17:54.36 | HitTop | wats alsa app? oh.. i just notice that i don even have chan_alsa.so module.. and its also not in the /usr/src/asterisk/channels |
| 17:54.41 | redder86 | ManxPower: will you share your firmware? |
| 17:55.17 | Zeeek | wow it goes from 'the best phones for asterisk" to "I can't get it working" in five minutes |
| 17:55.18 | ManxPower | redder86: I have had no problems with the Polycoms running 1.3.1 and Asterisk CVS-v1-0-12/08/04-11:02:07 |
| 17:55.23 | redder86 | The Polycom is a nicer phone, by far, than the Grandstreams, but I've had so little problems with the Grandstreams in comparison to any other phone. |
| 17:55.29 | inetdcrsh | also, have any of you tried using ip phones over 802.11 using an ethernet bridge? |
| 17:55.35 | ManxPower | redder86: No. I got it from the guy that said in a message on the mailing list that he had 1.3.4 |
| 17:56.15 | redder86 | ManxPower: so he will share it? |
| 17:56.20 | redder86 | apparently? |
| 17:56.38 | ManxPower | asterisk@markovprocesses.com |
| 17:56.45 | redder86 | thanks |
| 17:56.49 | DevilFish | I got 1.3.4 |
| 17:57.04 | redder86 | DevilFish: will you share it? |
| 17:57.13 | DevilFish | yes |
| 17:57.13 | ManxPower | We put 1.3.4 on one phone and since it didn't do anything different we didn't upgrade the other phones. |
| 17:57.32 | DevilFish | I'm fighting my IP600s now actually |
| 17:57.35 | redder86 | Well, I'd like to see if it fixes these problems we're having. |
| 17:57.46 | DevilFish | where you want it? |
| 17:57.49 | redder86 | DevilFish: what problems are you having? |
| 17:58.04 | DevilFish | calls magically terminate after 20 -60 mins |
| 17:58.27 | DevilFish | looks like this |
| 17:58.28 | DevilFish | Jan 13 11:02:31 DEBUG[2252]: Ignoring too old packet packet 721599706 (expecting >= 721599707) |
| 17:58.35 | ManxPower | MY polycom problems were always solved by upgrading ASTERISK. |
| 17:58.38 | DevilFish | and then.. |
| 17:58.40 | DevilFish | this |
| 17:58.41 | DevilFish | Jan 13 11:02:31 DEBUG[2252]: Didn't get a frame from channel: SIP/sip_proxy-out-bffd |
| 17:58.54 | redder86 | DevilFish: looks like the polycom isn't updating the timestamps on the packets. |
| 17:58.56 | DevilFish | think I shold go to latest CVS? |
| 17:59.22 | redder86 | I'm using a very recent version of CVS. |
| 17:59.27 | redder86 | stable, of course |
| 17:59.44 | inetdcrsh | thanks for all the help guys... i'll let ya know how it goes |
| 17:59.49 | Zeeek | Manx how do you "upgrade" asterisk? Are their Platinum modules ? |
| 17:59.57 | ManxPower | I only use 1.0.x STABLE |
| 18:00.05 | ManxPower | Zeeek: Huh? |
| 18:00.10 | Zeeek | asterisk Gold... now more than ever |
| 18:00.45 | Zeeek | "<ManxPower> MY polycom problems were always solved by upgrading ASTERISK." |
| 18:00.47 | redder86 | Zeeek: okay, maybe update is a better word |
| 18:00.52 | ManxPower | I download the lastest v1-0 CVS, install it on my home server, if that works, install it on the lover's server, if that works install it on the voIP server, if that works install it on the Ormond server, if that works..... |
| 18:00.56 | Zeeek | I want to UPGRADE!!!! |
| 18:00.56 | DevilFish | I'm testing right now with xlite to see if the same cutoffs happen |
| 18:01.21 | Zeeek | I want ot get rid of the nag screens in CLI :) |
| 18:01.25 | ManxPower | Zeeek: You can't "upgrade" you can only install a different version on top of what you are running now. |
| 18:01.30 | redder86 | DevilFish: it will be curious if they do |
| 18:01.42 | Zeeek | but you said... sniff |
| 18:01.54 | DevilFish | well the xlite is at 39mins and holding so far |
| 18:02.16 | DevilFish | the Polycom 600 made it 1hr and 6mins this morning after upgrade to 1.3.4 |
| 18:02.33 | redder86 | DevilFish: where did you get that firmware? |
| 18:02.38 | DevilFish | but... I think I migh go to latest CVS of asterisk as well later to day and see if that yields anything |
| 18:02.46 | DevilFish | Polycom |
| 18:02.56 | redder86 | DevilFish: are you a reseller? |
| 18:03.02 | DevilFish | :) |
| 18:03.06 | DevilFish | social engineer |
| 18:03.17 | redder86 | hehe |
| 18:03.21 | DevilFish | ;) |
| 18:03.25 | redder86 | I'm using CVS 1.0 from 01/05/05 |
| 18:03.52 | DevilFish | right now I'm using asterisk 1.0.3 |
| 18:04.03 | DevilFish | not sure of the date |
| 18:04.03 | redder86 | yeah, so was I up until the 5th |
| 18:04.12 | ManxPower | try CVS v1-0 or 1.0.2 |
| 18:04.48 | redder86 | I was having problems with stuff that ended up not being Asterisk's fault, and I updated Asterisk in the course of troubleshooting |
| 18:04.49 | DevilFish | can I just follow the CVS instructions here?? http://asterisk.org/index.php?menu=download |
| 18:04.55 | DevilFish | I know squat about CVS |
| 18:05.06 | redder86 | DevilFish: yes, those instructions work |
| 18:05.11 | viper-net | hello |
| 18:05.16 | DevilFish | will it get me the version I want? |
| 18:05.34 | *** part/#asterisk helios4 (~utente1@host10-52.pool8175.interbusiness.it) |
| 18:05.38 | redder86 | make sure that you follow the instructions that include "-r v1_0" |
| 18:05.52 | DevilFish | check looks like |
| 18:05.52 | DevilFish | cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 18:06.04 | DevilFish | isn't that just version 1.0 ? |
| 18:06.05 | redder86 | yes, but you probably don't need all of those |
| 18:06.12 | redder86 | zaptel libpri asterisk |
| 18:06.36 | redder86 | I don't know if you use all of them, just itemize the ones you need. Make sure you update all of the ones that you use together. |
| 18:06.36 | DevilFish | I think I just need asterisk and asterisk-addons |
| 18:06.56 | DevilFish | we're using straight sip channel to a metaswitch |
| 18:07.01 | redder86 | DevilFish: -r v1_0 refers to the "stable" branch of CVS. |
| 18:07.06 | DevilFish | Ahhh I get it |
| 18:07.19 | DevilFish | so "latest stable" then |
| 18:07.25 | redder86 | yeah |
| 18:07.38 | HitTop | ManxPower: I wonder if u need libalsa2-devel in order to use chan_alsa? |
| 18:07.44 | redder86 | to get a "tagged" version you use -t instead of -r, I think. |
| 18:07.54 | DevilFish | tagged? |
| 18:07.58 | ManxPower | HitTop: Of course you do. How else would Asterisk build the channel driver? |
| 18:08.29 | ManxPower | DevilFish: You don't want a tagged version. Those versions never get updated. |
| 18:08.40 | DevilFish | whats -r then? |
| 18:08.44 | redder86 | DevilFish: "tagged" as "1.0.3" or "1.0" or "1.0.1" ,etc. |
| 18:08.49 | DevilFish | I see |
| 18:08.57 | redder86 | DevilFish: -r is "repository", I think. |
| 18:09.05 | ManxPower | DevilFish: -r is a BRANCH and branches drow with bugfixes and updates. |
| 18:09.11 | redder86 | DevilFish: it refers to "branches" of CVS |
| 18:09.14 | *** join/#asterisk eGnarF (mrbk@eris.hb.lu.se) |
| 18:09.30 | ManxPower | The correct command is normally in the /topic of #asterisk-stable |
| 18:09.35 | DevilFish | gotcha |
| 18:10.11 | redder86 | yeah, here we talk so much about DUNDi... :-P |
| 18:10.17 | redder86 | and slashdot |
| 18:11.36 | eGnarF | Hi! Is there any way to enable/disable the zaptel echocanceller from the dialplan? |
| 18:12.08 | Godsey | my ip-300 phones end up w/ one way audio after a while |
| 18:12.26 | redder86 | Godsey: what firmware rev? |
| 18:13.24 | Godsey | 1.3.0.0711 |
| 18:13.37 | DeviFish | share away redder86 ;) |
| 18:13.38 | *** join/#asterisk burton27_ (mimx@w201.ljudmila.org) |
| 18:13.48 | Godsey | I don't know how to upgrade the firmware either :) |
| 18:13.58 | Godsey | just stick it in the ftp dir? |
| 18:15.04 | redder86 | Godsey: pretty much, yes |
| 18:15.06 | DeviFish | yup |
| 18:15.40 | Godsey | do I have to worry about having a IP-300 and finding firmware for IP-600? |
| 18:15.40 | *** join/#asterisk terner (LKL@202.92.53.5) |
| 18:15.49 | redder86 | Godsey: I don't think so. |
| 18:16.05 | DeviFish | I think they all use same bootrom and sip.ld |
| 18:16.13 | DeviFish | 300,500, and 600 |
| 18:16.14 | redder86 | Godsey: but where are you going to get firmware? Does the place you got your phone support you with new firmware? |
| 18:16.31 | Godsey | I shall use google power |
| 18:16.32 | Godsey | :) |
| 18:16.51 | Godsey | http://www.freedomphones.net/polycom/files/ has slightly newer |
| 18:16.53 | redder86 | Godsey: go ahead, and let us know how well that time was spent |
| 18:16.57 | ManxPower | With Cisco anyone can get the firmware...if you pay enough money. With Polycom anyone can get the firmware..if they can find it. |
| 18:17.19 | DeviFish | well thats the 1.3.1 stuff |
| 18:17.45 | redder86 | Godsey: that has 1.3.1 DeviFish gave me 1.3.4 and I'd be happy to share the power |
| 18:17.57 | redder86 | unless you still want to google. |
| 18:18.18 | Godsey | I'll take any help I can get :) |
| 18:19.04 | redder86 | msg me your e-mail |
| 18:19.44 | Godsey | thanks :) |
| 18:20.28 | DeviFish | Umm... has anyone read the security advisory at http://www.mpg123.de/ about mpg123 ??? |
| 18:20.51 | redder86 | how recent is the advisory? |
| 18:21.00 | bjohnson | for those who want info about the cheap cordless phone I referred to this morning .. it is an XACT XG2200 (www.getxact.com) 2.4GHz cordless phone with a working MWI that I use with *. According to the manual, it scans 50 channels to find the clearest one when you hit the chan button |
| 18:21.04 | *** join/#asterisk Samoied (~samoied@popeye.opens.com.br) |
| 18:21.08 | DeviFish | yesterday |
| 18:21.12 | *** join/#asterisk burton27_ (mimx@w201.ljudmila.org) |
| 18:21.20 | Samoied | Grandstream phones support ADSI? |
| 18:21.31 | redder86 | bjohnson: is it a SIP phone or do you use it with an ATA? |
| 18:21.33 | Godsey | the website only has 1.3.1 geesh |
| 18:21.41 | Godsey | redder86, thanks |
| 18:21.47 | redder86 | got it ? |
| 18:22.03 | redder86 | Godsey: thank DeviFish |
| 18:22.05 | bjohnson | redder86: SIpura SPA 3000 fxs port |
| 18:22.06 | fearnor | samo: no. |
| 18:22.06 | DeviFish | np |
| 18:22.15 | fearnor | adsi = analog, grandstream = ip |
| 18:22.33 | Godsey | thanks DeviFish :) |
| 18:22.40 | DeviFish | No problmo |
| 18:22.44 | DeviFish | share away |
| 18:23.08 | Godsey | do I extract the files or leave them as .zip in ftproot? |
| 18:23.15 | redder86 | I don't understand why Polycom doesn't let everyone download firmware. |
| 18:23.24 | *** join/#asterisk Mike9 (~sturdee@sun.mikesweb.com) |
| 18:23.25 | redder86 | Godsey: extract them |
| 18:23.35 | Godsey | k |
| 18:23.38 | redder86 | Godsey: are you already configuring the phones via ftp? |
| 18:23.43 | Godsey | yes |
| 18:23.47 | Samoied | fearnor: sorry |
| 18:23.49 | redder86 | okay, then you're good to go |
| 18:23.49 | Godsey | dhcp+ftp |
| 18:23.59 | redder86 | getting the ftp configuration going is a bit of a trick at first |
| 18:24.12 | redder86 | setting up the ftp user |
| 18:24.13 | Godsey | voip-info held my hand pretty good :) |
| 18:24.35 | redder86 | voip-info does a fair job on the Polycom stuff. |
| 18:24.37 | Godsey | ya I had to change ftp software to allow for case insensitive |
| 18:25.16 | redder86 | I didn't run into a case sensitivity problem, but I did need to alter /etc/passwd by hand. |
| 18:25.24 | redder86 | I think that was it. |
| 18:26.09 | Godsey | for some silly reason my system doesn't like mixed case |
| 18:27.55 | Bentley | question for those using multipline phones: If line#1 of the phone is in-use, and a 2nd call comes in, does your phone automatically route the 2nd call to line#2 (or do you need to config your dialplan to do this)? |
| 18:28.32 | *** join/#asterisk RevK-Laptop (~RevK@irc.asterisk-plus.ltd.uk) |
| 18:28.33 | redder86 | Bentley: depends how those phones are configured. What kind of phones are you talking about? Usually you have to do that with dialplan. |
| 18:28.35 | viper-net | asterisk is very cool, i solved nearly all problems now, only 1 left |
| 18:28.39 | viper-net | :) |
| 18:29.05 | redder86 | viper-net: 1 problem: you can't fax over ULAW reliably? |
| 18:29.19 | Bentley | redder86, I'm not using one - just want to config system to be flexible for multiline phones (AMP) |
| 18:29.40 | DeviFish | Well I'm past the 1hour 6min mark with the X-Pro soft phone |
| 18:29.44 | viper-net | no, i use an active isdn card with chan_capi and when someone phones me the last digit gets cut |
| 18:29.52 | DeviFish | looks like my problem might actually be the Polycom 600 |
| 18:29.56 | redder86 | Bentley: generally you'd have to configure a queue of sorts to do what you mention, it's a frequent Asterisk configuration. |
| 18:30.04 | DeviFish | or a combo of it and asterisk 1.0.3 |
| 18:30.06 | redder86 | DeviFish: yup |
| 18:30.11 | DeviFish | crap |
| 18:30.30 | DeviFish | I can't ditch the Polycom and I'm up to latest realeases on them |
| 18:30.33 | Godsey | oh these example phone1.cfg etc files have lots more options than the one I started with :) |
| 18:30.34 | DeviFish | so now what |
| 18:30.38 | viper-net | i mean an external user phones me, and when he first press dial and then presses the digits asterisk accepts the call to early and doesn't take the last digit |
| 18:30.41 | Bentley | redder86, thx, I'll look at building exten 'failover' then |
| 18:30.47 | viper-net | so i don't get the call |
| 18:30.48 | DeviFish | goto latest CVS I can find? |
| 18:30.49 | redder86 | DeviFish: does Polycom have some direct support channels? |
| 18:31.03 | DeviFish | they don't want to mess with asterisk |
| 18:31.14 | redder86 | DeviFish: yeah, I know. |
| 18:31.15 | viper-net | but if he presses redial it works without any problem, because then he transmits the whole number |
| 18:31.19 | viper-net | it's strange |
| 18:31.26 | viper-net | maybe i can set a delay ? |
| 18:31.42 | redder86 | viper-net: I don't know anything about CAPI |
| 18:31.49 | eGnarF | viper-net: put a wait(1) in your dialplan before you answer. |
| 18:32.15 | eGnarF | viper-net: It might help. |
| 18:32.50 | IAX | Cool! It's available |
| 18:33.44 | bjohnson | ? |
| 18:33.51 | IAX | the nick "IAX" |
| 18:34.14 | IAX2 | hehe |
| 18:34.41 | IAX2 | much better 8-) |
| 18:34.46 | IAX2 | oh! |
| 18:34.56 | IAX3 | I am from the future! |
| 18:35.04 | DeviFish | hehe |
| 18:35.23 | gambolputty | IAX3 will have to be written just to enable IPv6 in *. |
| 18:35.37 | IAX3 | But then I'll be so fat! |
| 18:36.11 | DeviFish | yes OC3 to my house will be common :) |
| 18:36.43 | Makenshi | DeviFish, i think ethernet will be more popular than sonet for ftth due to the commodity nature of ethernet hardware |
| 18:37.03 | DeviFish | yes |
| 18:39.29 | Godsey | rebooting phone to see if it picks up the files :) |
| 18:39.44 | IAX3 | Do you think this protocol makes me look fat? |
| 18:40.03 | Godsey | updating bootrom sexy |
| 18:40.16 | eGnarF | IAX3: Not fat. Just built =) |
| 18:40.23 | *** join/#asterisk Gul-Dukat (2400@203-59-52-190.dyn.iinet.net.au) |
| 18:40.26 | Godsey | yikes formatting file system :) |
| 18:40.31 | DeviFish | oh oh |
| 18:40.36 | Godsey | my mom's phone doesn't do this! |
| 18:40.36 | Godsey | :P |
| 18:40.36 | DeviFish | just kidding :) |
| 18:42.25 | redder86 | DeviFish: that seems to have resolved my problems. |
| 18:42.39 | redder86 | DeviFish: well, either that or any of the other myriad of things I tried overnight. |
| 18:42.45 | dragonius | Jan 13 19:40:57 NOTICE[1088912304]: chan_sip.c:7571 handle_request: Registration from '<sip:fsteiger@192.168.0.3>' failed for '192.168.0.3' |
| 18:42.57 | dragonius | who can help me with registration? |
| 18:43.00 | dragonius | i dont know what's wrong |
| 18:43.21 | redder86 | dragonius: your Asterisk server is at 192.168.0.3 ? |
| 18:43.25 | IAX3 | dragonius: You don't have a [fsteiger] in sip.conf |
| 18:43.43 | dragonius | ok, IAX3, ill change that :) |
| 18:44.34 | dragonius | lol |
| 18:44.36 | dragonius | no, i got it |
| 18:44.40 | dragonius | thx for the quick help |
| 18:44.53 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
| 18:45.00 | DeviFish | redder86: I forgot ... what were you having? |
| 18:46.00 | redder86 | DeviFish: the phone would ring and ring with no way to answer it |
| 18:46.18 | redder86 | DeviFish: and the user couldn't call out on it |
| 18:46.31 | DeviFish | wierd |
| 18:46.36 | DeviFish | weird |
| 18:46.48 | redder86 | I don't know how a SIP application update would fix that, but either that or the configuration changes I made did. |
| 18:47.18 | redder86 | fortunately, they don't have hour-long conversations frequently, if ever |
| 18:47.18 | DeviFish | well its best to be up to snuff anyways |
| 18:47.26 | Godsey | humm |
| 18:47.29 | *** join/#asterisk randu (~randy@207.144.31.103) |
| 18:47.35 | Godsey | I put my .wav files into the ipmid file |
| 18:47.42 | Godsey | but I don't think the phone downloaded it :) |
| 18:47.52 | redder86 | Godsey: I've never done that |
| 18:48.03 | DeviFish | gota make sure its the right format |
| 18:48.09 | Godsey | oh ya |
| 18:48.14 | DeviFish | 16Khz mono |
| 18:48.15 | Godsey | mono 8000 |
| 18:48.48 | randu | Hello. We have 4 vonage lines and we want to use asterisk as an auto-attendant. We do I start? |
| 18:48.54 | DeviFish | I used Audcity |
| 18:49.05 | randu | I have debian installed with asterisk 1.0.2 |
| 18:50.04 | redder86 | isn't it Audacity? |
| 18:50.09 | bjohnson | randu: scrap vonage and use a good voip provider? |
| 18:50.53 | randu | which do you recommend? |
| 18:51.07 | bjohnson | randu: nufone is popular |
| 18:51.09 | redder86 | Vonage charges for numbers of simultaneous calls ("lines") ? |
| 18:51.46 | randu | they have unlimited calling now or do you mean something different? |
| 18:51.56 | bjohnson | randu: define unlimited |
| 18:51.58 | redder86 | well, you said that you have 4 lines |
| 18:52.07 | Godsey | vonage works w/ their own ata hardware |
| 18:52.13 | redder86 | does that mean that only 4 people can call you at the same time? |
| 18:52.15 | Godsey | using vonage softphones is a pain too |
| 18:52.36 | bjohnson | randu: is it unlikely you will be able to use any of the hardware provided by vonage |
| 18:52.42 | Godsey | get 1 line w/ someone like broadvoice or voicepulse and have all the inbound calls you can handle |
| 18:52.45 | randu | yea 4 lines business unlimted calling, no that means 4 different lines |
| 18:52.59 | bjohnson | randu: google for asterisk and vonage to get config info |
| 18:53.10 | Godsey | you'll need to get some X100P cards |
| 18:53.14 | redder86 | I don't understand the term "lines", though, as it applies to VoIP. |
| 18:53.16 | Godsey | if you want to make it simple |
| 18:53.22 | Delvar | nn all u blokes! |
| 18:53.32 | Godsey | for all intents and purposes vonage hides the fact it's voip |
| 18:53.34 | Delvar | and any bloke'et! |
| 18:53.49 | bjohnson | randu: do you have 4 DIDs in the same area? |
| 18:53.55 | redder86 | Godsey: so it's an ATA with 4 FXS ports? |
| 18:54.04 | randu | not sure what you mean by DID's |
| 18:54.11 | bjohnson | randu: phone numbers |
| 18:54.16 | modulus_ | morning |
| 18:54.18 | redder86 | randu: DIDs are your phone numbers |
| 18:54.34 | randu | oh yea we have not bought the other 3 lines yet, but yes they will all have the same are code and exchange |
| 18:54.58 | redder86 | ah, maybe "lines" means "DIDs" to Vonage |
| 18:55.14 | bjohnson | redder86: typically it's a 2 port fxs by Linksys with the admin password already set .. so the hardware talks directly to vonage and can't be used for anything else |
| 18:55.30 | bjohnson | randu: so there is some cost savings there |
| 18:55.47 | sektor195 | how can I configure an extension for Agentcallback login so that it knows the information without entering it and pressing # |
| 18:55.50 | Godsey | no |
| 18:55.51 | Godsey | they probably have 4 ata devices |
| 18:55.51 | Godsey | real waste of money |
| 18:56.09 | bjohnson | randu: in voip, you can get a DID that permits multile concurrent incoming and outgoing calls |
| 18:56.32 | bjohnson | Godsey: and with * they won't even be able to use them for phones |
| 18:56.43 | sektor195 | I have a line right now where it just asks for your password and new extension |
| 18:56.45 | redder86 | sounds like an amazingly silly waste of VoIP technology - limiting the user to any number of simultaneous calls or devices. |
| 18:56.45 | randu | interesting so you would only need 1 phone number then for incoming calls? |
| 18:56.46 | *** join/#asterisk file[tech] (~jcolp@66.199.241.90) |
| 18:56.52 | file[tech] | kaboom into existence I am |
| 18:56.54 | bjohnson | randu: yes |
| 18:56.57 | sektor195 | if it just asks for password then that is fine |
| 18:57.42 | DeviFish | Anyone mess with recording phone conversations in * ?? |
| 18:57.46 | randu | ok so I should go read broadvoice or voicepulse websites to learn more? |
| 18:57.53 | modulus_ | no |
| 18:57.56 | modulus_ | you shouldn't |
| 18:57.57 | bjohnson | randu: there is a huge list of voip providers on the wiki .. find one that you like .. some have unlimited incoming plans and some have unlimited outgoing. Some have per minute pricing .. some have 800 numbers .. find on that best suits you |
| 18:58.00 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 18:58.05 | Godsey | sox is making Microsoft PCM not ITU G.711 mu-law wav files |
| 18:58.23 | DeviFish | I need an example of how to record a conversation that I'm on |
| 18:58.24 | viper-net | i have a network with sip phone in conjunction with isdn lines, what codec would you choose ? |
| 18:58.26 | bjohnson | randu: then go buy some hardware (no vonage style discounts that they charge you for when you cancel service) |
| 18:58.27 | modulus_ | sox doesn't encode to ulaw? |
| 18:58.39 | Godsey | trying to figure out how to make sound files the phones like |
| 18:58.42 | bjohnson | randu: then you won't believe what you can really do with voip |
| 18:58.48 | modulus_ | godsey, rh works |
| 18:58.48 | bprice20 | is there any way I can change the bitrate of the wav's asterisk creates for voicemail |
| 18:58.52 | modulus_ | using sox |
| 18:58.52 | file[tech] | DeviFish : use the Monitor application before you Dial to the other individual in your dialplan, use show application monitor to see the options available for it |
| 18:59.23 | redder86 | VoIP is so much more powerful than what it appears Vonage is letting its users have. |
| 18:59.28 | tzanger | hmm |
| 18:59.30 | bjohnson | sektor195: I'd like to see the dialplan for that |
| 18:59.39 | tzanger | spandsp looks like it was compiled with -g but I have no line number info |
| 18:59.48 | bprice20 | bkw_ is there any way I can change the bitrate of the wav's asterisk creates for voicemail |
| 18:59.55 | bjohnson | sektor195: I've considered also using callerid info to authorize some people to use outgoing lines |
| 18:59.59 | redder86 | tzanger: was that you with the spandsp problem on -users? |
| 19:00.09 | randu | ok. I will look them over. and then I will be back :-) |
| 19:00.21 | tzanger | redder86: on -dev |
| 19:00.27 | redder86 | oh, it was -dev |
| 19:00.53 | DeviFish | What about if I want to record somone who has called me? |
| 19:01.26 | Godsey | ok needed to use -U for conversion :) |
| 19:01.32 | dragonius | asterisk says it is playing stuff like "the person is not available" but i cant hear anything |
| 19:01.38 | file[tech] | if the call has already started, it's not exactly possible |
| 19:01.51 | bjohnson | redder86: yes .. vonage takes the MS approach.. simplify for general users by removing features |
| 19:01.54 | file[tech] | just put a monitor before your phone is called using the dial application, and it'll record it |
| 19:01.58 | *** join/#asterisk _chad (~Chad@c-24-6-142-55.client.comcast.net) |
| 19:02.01 | *** join/#asterisk devi0us (devi0us@west.philly.ghetto.org) |
| 19:02.04 | _chad | anyone else having trouble w/ nufone? |
| 19:02.24 | modulus_ | i've always had trouble with nufone |
| 19:02.30 | _chad | ugh |
| 19:02.30 | DeviFish | I'd better go read up on monitor then... thankyou |
| 19:02.34 | modulus_ | in fact, i've always had trouble with _any_ voip provider i've used |
| 19:02.35 | redder86 | I've rarely had trouble with nufone. |
| 19:02.54 | _chad | my iax registration is now being rejected randomly |
| 19:02.56 | dragonius | asterisk says it is playing stuff like "the person is not available" but i cant hear anything |
| 19:03.27 | tzanger | redder86: I've *never* had trouble with them that was their fault |
| 19:03.28 | bjohnson | randu: for example .. you could get one DID (different providers can get different DID areas) that allows 4 concurrent incoming/outgoing calls for like $40 / mo and includes unlimited LD to NA |
| 19:03.57 | sektor195 | bjohnson: which dialplan do you want to see where it just asks for password and extension, but knows agent id? |
| 19:04.21 | redder86 | well, I can't seem to place calls through NuFone right now, though. |
| 19:04.26 | redder86 | I'm registered fine. |
| 19:04.39 | bjohnson | randu: but really evaluate what incoming and outgoing minutes you will really use .. don't just run to an unlimited plan cause it sounds nice |
| 19:04.43 | modulus_ | ERROR 2006 (HY000): MySQL server has gone away |
| 19:04.43 | modulus_ | No connection. Trying to reconnect... |
| 19:04.43 | modulus_ | Connection id: 257932 |
| 19:04.43 | modulus_ | Current database: CUSTOMERS |
| 19:04.43 | modulus_ | Query OK, 1 row affected (0.12 sec) |
| 19:04.43 | modulus_ | mysql> |
| 19:04.45 | modulus_ | haha |
| 19:04.47 | modulus_ | mysql is funny! |
| 19:04.48 | redder86 | hrmm... oh well. I only use them for overflow. |
| 19:05.11 | modulus_ | error 2006 "MySQL server has gone away" |
| 19:05.23 | bjohnson | sektor195: I thought you had one for an incoming call that asks for PIN and then allows use of outgoing lines |
| 19:05.46 | Godsey | -rw-r--r-- 1 plcmspip users 972 Jan 13 11:03 000000000000-directory.xml |
| 19:05.46 | sektor195 | no, I'm talking about login for queues |
| 19:05.53 | Moc | IM TIRED OF THE WORLD .... PEOPLE CAN'T DO THINGS RIGHT.... |
| 19:05.56 | Godsey | do I rename this to 000000000001-directory to re-push a new ver? |
| 19:06.04 | bjohnson | sektor195: sorry .. different topic then |
| 19:06.19 | sektor195 | its ok |
| 19:06.26 | *** join/#asterisk RafaeL (iskendeR@81.212.13.58) |
| 19:06.53 | ManxPower | Godsey: The polycom phone uses a mixed case userid and password. |
| 19:07.04 | Moc | UPS can't read a Postal Code... LG, Whirlpool, can't get good appliance out of their factore, The Brick can't get good people to deliver them.... LG doesnt have good repair men, and Whirlpool doesnt call back when you ask for repair ... |
| 19:07.16 | bjohnson | I'm not a big user but aleph has been a good voip provider for me |
| 19:07.27 | Godsey | ManxPower: I mean the filename to push a new directory to phone |
| 19:07.29 | bjohnson | and they support the astcc app |
| 19:07.31 | *** join/#asterisk roche (~roche@216.194.173.2) |
| 19:07.35 | modulus_ | aleph? |
| 19:07.39 | modulus_ | jewish voip provider? |
| 19:07.50 | fearnor | mod: who *isn't* a jewish provider |
| 19:07.52 | tzanger | hahaha |
| 19:07.54 | bjohnson | modulus_: don't know, don't care |
| 19:07.55 | fearnor | vonage = jeff citron, JEW |
| 19:08.01 | modulus_ | lol |
| 19:08.03 | fearnor | broadvoice = nathan stratton, JEW |
| 19:08.07 | modulus_ | HAHAHA |
| 19:08.17 | modulus_ | my gawd i didn't realize the folly of my comment |
| 19:08.21 | tzanger | I don't know why but having my voip provider's IVR say "GOYM" would just get me tittering |
| 19:08.29 | fearnor | i dunno if jerjer is a JEW but could well be ;) |
| 19:08.38 | modulus_ | and which of the * developers *isn't* homosexual? |
| 19:08.43 | modulus_ | HAHAHHAA |
| 19:08.51 | tzanger | HAHAHAH |
| 19:08.55 | tzanger | too fucking funny |
| 19:09.19 | fearnor | bkw is not a jew. but he looks to me like he ate one! |
| 19:09.21 | fearnor | :) |
| 19:10.03 | modulus_ | jerjer looks like he at two. |
| 19:10.08 | modulus_ | s/at/ate |
| 19:10.14 | fearnor | heh |
| 19:10.46 | bjohnson | Moc: I bought a hot tub from Home Depot last spring. They 1. delivered it to the wrong address on a date that they didn't coordinate with me (and they were coming in from another city) 2. lost it in the store on the date they were supposed to deliver it 3. delivered it all banged up and the delivery guy suggested we refuse it 4. lost it again 5. tried to deliver it to the wrong address again 6. 6th times a charm I guess |
| 19:11.21 | _chad | hmmmm |
| 19:11.23 | _chad | Call accepted by 66.225.202.72 (format ULAW) |
| 19:11.23 | _chad | <PROTECTED> |
| 19:11.23 | _chad | <PROTECTED> |
| 19:11.23 | _chad | <PROTECTED> |
| 19:11.55 | bjohnson | _chad: comiserate with modulus_ |
| 19:12.11 | _chad | so thats on nufone then? |
| 19:12.26 | bjohnson | sounds like it from others saying same thing |
| 19:12.42 | _chad | man what kind of ghetto shit is this |
| 19:12.42 | bjohnson | _chad: you have a backup voip provider right? |
| 19:12.59 | _chad | i've been waiting a month for my did from them |
| 19:13.07 | devi0us | email them |
| 19:13.12 | devi0us | it took them 2 weeks for mine |
| 19:13.15 | modulus_ | _chad: you have a backup of a backup of a backup of a backup voip provder right? |
| 19:13.21 | bjohnson | ahh .. incoming is one thing .. always have a backup for outgoing |
| 19:13.36 | _chad | lol |
| 19:15.07 | *** join/#asterisk ckruetze (~ckruetze@i3ED61609.versanet.de) |
| 19:16.00 | ckruetze | Hi |
| 19:17.16 | _chad | to add a redundant backup do i simply add another outbound extention after my primary? |
| 19:18.46 | _chad | hmm |
| 19:18.55 | _chad | nothin's simple lol |
| 19:19.37 | Godsey | trying "reset settings" on phone menu :) |
| 19:19.55 | Godsey | wish the phone would just auto download speed dial list every reboot |
| 19:20.02 | Godsey | I must be missing something simple |
| 19:20.05 | _chad | well i figured out the problem, i just want to add another outbound termination if the 1st one fails |
| 19:20.22 | tzanger | _chad: no |
| 19:20.27 | tzanger | look at DIALSTATUS and HANGUPCAUSE |
| 19:20.32 | tzanger | I just posted this to -users |
| 19:20.40 | redder86 | modulus_: Mark is homosexual? |
| 19:20.42 | _chad | on the wiki? |
| 19:20.55 | tzanger | no, -users |
| 19:21.12 | tzanger | redder86: no, we were fighting over who gets first dibs on Alison if her engagement breaks off |
| 19:21.28 | *** join/#asterisk Tili (~Tili@202-133-67-65-dialup.sat.net.pk) |
| 19:21.29 | _chad | mailinglist? sorry i'm not familiar w/ '-users' :) |
| 19:21.40 | redder86 | yeah, that's why modulus's comment struck me as odd |
| 19:22.02 | modulus_ | it was a contextual joke |
| 19:22.08 | tzanger | _chad: look for my post to "Multiple gateways for same dial pattern" in asterisk-users |
| 19:22.16 | tzanger | I posted it yesterday at 10:05:24pm |
| 19:22.22 | _chad | k, thanks |
| 19:22.28 | redder86 | I guess he could be bi, though. |
| 19:22.33 | tzanger | _chad: I use this already and it works very well |
| 19:23.30 | *** join/#asterisk freestyle_networ (~freestyle@S0106000f6630d841.ed.shawcable.net) |
| 19:24.19 | redder86 | tzanger: I've never met Allison. The voice is nice, but it would be weird to talk to someone who sounds like "my phone system". |
| 19:24.50 | tzanger | voice is soooooooooo sexy |
| 19:24.58 | ArkyLady | Allison is hot :) |
| 19:25.00 | tzanger | she's got that hint of laughter in her voice that makes Shania Twain's voice so hot too |
| 19:25.14 | Moc | Anyone recommend me a BRI * card that work for Canada/USA ? |
| 19:25.21 | DeviFish | anyone actually seen Allison? |
| 19:25.24 | modulus_ | allison looks like she could give me southern hospitality |
| 19:25.27 | redder86 | Shania Twain's voice is sexier than Allisons. |
| 19:25.30 | modulus_ | err s/give/show |
| 19:25.55 | ArkyLady | redder but Allison is way cool and might actually talk to ya hehe |
| 19:25.59 | tzanger | redder86: they're both very similar and it's that quality I described -- they can intonate very easily and it adds so much "depth" to their voice |
| 19:26.07 | tzanger | I've talked to Shania Twain before |
| 19:26.20 | *** join/#asterisk firestrm (~vince@S010600047577bccd.gv.shawcable.net) |
| 19:26.25 | ArkyLady | was she saying "Get your hands off my boobies!!" ? lol |
| 19:26.26 | redder86 | did she talk back to you? |
| 19:26.29 | firestrm | hello |
| 19:26.30 | tzanger | ArkyLady: hahahaha |
| 19:26.57 | tzanger | redder86: yes, we were talking about the weather actually (it's a canadian thing) |
| 19:27.37 | firestrm | anyone ever seen this error?[root@localhost ~]# modprobe wcfxo |
| 19:27.37 | firestrm | Notice: Configuration file is /etc/zaptel.conf |
| 19:27.37 | firestrm | line 0: Unable to open master device '/dev/zap/ctl' |
| 19:27.46 | redder86 | I don't think that a sexy lady would refer to her breasts as "boobies". |
| 19:27.57 | DeviFish | is this her ?? http://www.allisonsmith.org/ |
| 19:28.03 | redder86 | DeviFish: yes |
| 19:28.11 | DeviFish | no joke? |
| 19:28.14 | DeviFish | Dang |
| 19:29.11 | redder86 | I don't have a very good voice, but quite often I end up having to record voice prompts and menus for my customers because they never record them when I ask them to. |
| 19:29.36 | redder86 | I *hate* hearing recordings of myself. |
| 19:30.40 | sektor195 | How can I modify this dial plan exten => 999,1,AgentCallbackLogin(${CALLERIDNUM}|'#') so it doesn't ask for a new extension, the password is fine |
| 19:30.43 | Godsey | redder86: do you touch them up w/ audio editor? :) |
| 19:30.56 | mogorman | I dont think that is allison |
| 19:31.04 | Godsey | sektor195: to auto log out? |
| 19:31.13 | sektor195 | no login |
| 19:31.31 | redder86 | Godsey: no, I don't put much time into the recordings because "any day now" my customer will have one of the office girls do the recordings. |
| 19:31.45 | redder86 | mogorman: why not? |
| 19:32.05 | Godsey | exten => _*XXXX,2,AgentCallBackLogin(${MYAGENT}|${MYAGENT}@sipextbusy) |
| 19:32.08 | Godsey | I do it that way |
| 19:32.50 | sektor195 | godsey how come you have _* instead of an extension ? |
| 19:33.26 | sektor195 | I mean extension # that is |
| 19:33.37 | Godsey | because that's how I know what agent to login |
| 19:33.40 | *** join/#asterisk AgiNamu (~AgiNamu@200.12.43.74) |
| 19:33.54 | AgiNamu | Does anyone know of plans for the next asterisk stable? |
| 19:33.57 | sektor195 | oh ok |
| 19:33.58 | AgiNamu | as far as timelines? |
| 19:34.32 | sektor195 | but I can substitute _*XXXXX with an extension and it will still work right? |
| 19:34.59 | sektor195 | and all they would have to do is dial the same extension to log out right |
| 19:36.05 | Godsey | I have not used that for a bit, memory isn't fresh |
| 19:36.08 | AgiNamu | hey, the channelvars starting with _ -- that was added after stable 1.0.3? |
| 19:36.11 | Godsey | I use dynamic queue memebers now |
| 19:36.36 | sektor195 | oh |
| 19:36.52 | sektor195 | yeah all the queue members of this que are going to be static |
| 19:37.10 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 19:37.44 | *** join/#asterisk OhMyAchinLap (~chris@acs-24-144-154-158.zoominternet.net) |
| 19:38.19 | randu | I have looked at several of the voips listed at http://www.voip-info.org/wiki-VOIP+Service+Providers+Business and do not see of them that say anything about simultaneous incoming calls on one voip line. Does someone have an example? |
| 19:38.47 | *** join/#asterisk jbresler (~jbresler@worrior.cs.uiuc.edu) |
| 19:39.56 | Samoied | Is possible to use Conexant Softmodem with asterisk? |
| 19:40.22 | bjohnson | randu: yeah .. they don't usually advertise it since it confuses vonage type subscribers (just joking) .. email them to confirm |
| 19:40.36 | bjohnson | Samoied: only if you write a driver |
| 19:40.48 | sektor195 | Godsey: That line doesn't work |
| 19:40.59 | sektor195 | it just gives me a busy signal |
| 19:41.01 | Samoied | bjohnson: and what chan_modem_bestdata.so is for? |
| 19:41.34 | bjohnson | mogorman: that is alison .. the "voice" |
| 19:41.50 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 19:41.56 | sektor195 | Godsey: Originally I was trying to this setup exten => 905,1,Dial(Local/999/n,,D(#)) |
| 19:41.56 | sektor195 | <PROTECTED> |
| 19:41.56 | sektor195 | <PROTECTED> |
| 19:42.12 | bjohnson | sambal: I don't know |
| 19:42.13 | sektor195 | but it keeps giving me invalid login |
| 19:42.36 | AgiNamu | btw, wtf is up with naming in the code? |
| 19:42.39 | *** join/#asterisk jake (~jake@207.224.48.130) |
| 19:42.42 | AgiNamu | "ast_ha" ?? |
| 19:42.47 | bjohnson | AgiNamu: channel extensions starting with _ denote a match pattern |
| 19:42.57 | AgiNamu | bjohnson, no I'm talking about channelvars |
| 19:43.05 | bjohnson | AgiNamu: priority variables? |
| 19:43.07 | AgiNamu | supposedly "_" starting vars will get copied over to a channel that's dialed. |
| 19:43.11 | sektor195 | and someone else said that it couldn't be done with the Dial app |
| 19:43.27 | bjohnson | AgiNamu: oh. Did you have any luck with setup? |
| 19:43.27 | *** join/#asterisk firestrm (~vince@S010600047577bccd.gv.shawcable.net) |
| 19:43.30 | ManxPower | sektor195: Well what does ${CALLERIDNUM} expand to when you see the AgentCallbacklogin on the CLI? |
| 19:43.45 | sektor195 | hold on |
| 19:43.46 | AgiNamu | bjohnson, well, I think it's in head, not 1.0.3. so igotta go |
| 19:43.50 | AgiNamu | get that and patch it |
| 19:44.10 | AgiNamu | I take it that no one really cares about the maintainability or readability of the code |
| 19:44.15 | AgiNamu | it's more like 'what can I patch on quickly' |
| 19:44.26 | AgiNamu | and "how big a hotshot can i be" |
| 19:45.11 | firestrm | kernel upgrade (to hopefully fix new security holes) and zaptel is broken |
| 19:45.29 | AgiNamu | the code metrics for the project are eriously screwed up |
| 19:45.33 | bjohnson | for SetCIDName(${ARG7}) within a macro I can't seem to get the format for the variable right .. but I can get SetCIDNum(${ARG8}) working. Should ${ARG7} be sent within double quotes, single quotes, or nothing? |
| 19:46.14 | *** join/#asterisk telme_ (~trillian@sta-207-174-139-178.rockynet.com) |
| 19:46.22 | sektor195 | Maxnpower: It looks like this AgentCallbackLogin("SIP/202-5aeb", "202|#") |
| 19:46.40 | *** join/#asterisk telme__ (~trillian@sta-207-174-139-178.rockynet.com) |
| 19:46.56 | *** join/#asterisk telme__ (~trillian@sta-207-174-139-178.rockynet.com) |
| 19:47.01 | firestrm | anyone know where i should look to fix this? |
| 19:47.03 | firestrm | Notice: Configuration file is /etc/zaptel.conf |
| 19:47.03 | firestrm | line 0: Unable to open master device '/dev/zap/ctl' |
| 19:47.23 | bjohnson | can someone post a SetCIDName example? |
| 19:47.54 | Godsey | exten => s,1,SetCIDName(Tech ${CALLERIDNUM}) |
| 19:47.54 | Godsey | exten => s,1,SetCIDName(Blng ${CALLERIDNUM}) |
| 19:47.54 | Godsey | exten => s,1,SetCIDName(Mngr ${CALLERIDNUM}) |
| 19:48.10 | Godsey | this is how I let phone folks know what queue is calling them |
| 19:48.30 | modulus_ | how noble of you |
| 19:48.40 | modulus_ | i just let them see sip userid |
| 19:48.42 | modulus_ | haha |
| 19:50.03 | Godsey | for inbound calls? |
| 19:50.36 | bjohnson | hey I didn't realize Alison was one of the daughters on Kate and Allie !! |
| 19:51.19 | tzanger | what?? |
| 19:51.40 | *** join/#asterisk Handsome-Cexy (~sshah@202.125.141.6) |
| 19:52.41 | bjohnson | Godsey: no for outgoing but should be the same .. I think my voip provider is doing something with CIDName .. I can set the number but not the name |
| 19:52.55 | AgiNamu | bjohnson, to the PSTN? |
| 19:53.00 | AgiNamu | you can't set CIDName |
| 19:53.05 | tzanger | bjohnson: you're full of shit... kate and allie weren't their real names :-) |
| 19:53.09 | AgiNamu | that's done by a DB lookup |
| 19:53.15 | tzanger | ahh alison smith is there I see now |
| 19:53.30 | bjohnson | tzanger: on http://www.allisonsmith.org/ |
| 19:53.47 | *** join/#asterisk ScubaSteve (~steve@office6.neonova.net) |
| 19:53.47 | tzanger | she was married? |
| 19:53.49 | firestrm | how could a simple kernel ugrade break zaptel? |
| 19:53.51 | bjohnson | AgiNamu: voip provider to our incoming pstn line |
| 19:53.54 | tzanger | Married Randall Grimmett on June 23, 2001 |
| 19:54.06 | Handsome-Cexy | hi |
| 19:54.40 | Dr-Linux | any body can help me to configure voicemail.conf on local LAN ? |
| 19:55.04 | ManxPower | Dr-Linux: Um, if you can't configure voicemail.conf you have a lot more problems than Asterisk |
| 19:55.18 | tzanger | I think libtool is stripping something out |
| 19:55.23 | sgnome` | i installed asterisk last night and it stopped working for no apparent reason .. so i reinstalled linux and asterisk .. and now when i make install for asterisk i get an error |
| 19:55.23 | bjohnson | Dr-Linux: voicemail.conf |
| 19:55.46 | Dr-Linux | ManxPower: sir, everyone starts from more problems |
| 19:55.53 | ChArLeS___ | does anybody knows a softphone with G723 ? |
| 19:55.54 | bjohnson | Dr-Linux: then lookup voicemailmain on the voip wiki .. it's easy |
| 19:56.16 | ManxPower | ChArLeS___: See the following macro |
| 19:56.17 | Dr-Linux | i need just commands on my requirements |
| 19:56.18 | Moc | any voiceconduits people in here ? |
| 19:56.20 | ManxPower | Here is the licensing priceing info for G723.1 direct from the patent holder's web site: http://www.dspg.com/technology/LicensePricing.html |
| 19:56.24 | Dr-Linux | i have already done that .. |
| 19:56.36 | ManxPower | Dr-Linux: Most people just start with the voicemail.conf.sample in the Asterisk source. |
| 19:56.44 | Dr-Linux | but little doubt, |
| 19:56.58 | Dr-Linux | ManxPower: yae sir |
| 19:57.00 | bjohnson | Dr-Linux: so make an extension like exten => 88,1,Voicemailmain |
| 19:57.08 | ManxPower | ChArLeS___: Since Asterisk does not support G723.1 transcoding..... |
| 19:57.29 | Dr-Linux | bjohnson: wait sir let me tell me .. |
| 19:57.33 | Samoied | ManxPower: You know about chan_modem_bestdata.so? |
| 19:57.33 | mishehu | speex up, I can't hear you |
| 19:57.55 | ManxPower | Samoied: I know it doesn't work unless you hack it. |
| 19:57.56 | viper-net | eGnarF: are you still here ? |
| 19:58.16 | Dr-Linux | bjohnson: can i paste my voicemail.conf entries in your pvt? |
| 19:58.21 | ManxPower | Samoied: And I do not know of anyone that has ever gotten it working. |
| 19:58.31 | *** join/#asterisk OnuR (666@212.77.218.204) |
| 19:59.01 | bjohnson | Dr-Linux: voicemail.conf just sets up some options and accounts .. it doesn't enable it or anything |
| 19:59.15 | bjohnson | Dr-Linux: extensions.conf directs calls to voicemail |
| 19:59.16 | iCEBrkr | Damin: damnit! |
| 19:59.35 | bjohnson | Damint? |
| 19:59.50 | bjohnson | Dachocolate? |
| 19:59.56 | Dr-Linux | bjohnson: okey sir let me show you my extensions.conf and voicemail.conf entries |
| 20:00.20 | redder86 | I'm not all too impressed about distribution participation in vendor-sec. Seems like only Debian and Gentoo take it seriously. Maybe SuSE. |
| 20:01.17 | firestrm | what would cause /dev/zap/ctl to not be created when installing zaptel? |
| 20:01.28 | firestrm | where did i go wrong? |
| 20:03.14 | *** part/#asterisk starman (user@user-161.lightlink-ds1-825e-highlanddr-lehi.fiber.net) |
| 20:03.27 | firestrm | anyone |
| 20:03.33 | *** join/#asterisk lattice (~lattice@S0106000f669498f3.vc.shawcable.net) |
| 20:03.39 | Samoied | ManxPower: The voip-infoorg say: |
| 20:03.42 | Samoied | MODEM_BESTDATA Description: BestData (Conexant V.90 Chipset) VoiceModem Driver |
| 20:03.50 | firestrm | a hint, a wild guess, anything |
| 20:03.52 | Samoied | I have a Conexant modem |
| 20:04.01 | Samoied | V.90 Voice |
| 20:04.11 | ManxPower | Samoied: And I do not know of anyone that has ever gotten it working. |
| 20:04.22 | Samoied | ManxPower: thanks |
| 20:04.51 | firestrm | can anyone even see my messages, is this thing broken too? |
| 20:06.00 | *** join/#asterisk ennuyeux7 (~ennuyeux7@83.146.53.34) |
| 20:06.10 | *** join/#asterisk fleeeb (~basin@pc-24-151-28-122.newm2.ct.charter.com) |
| 20:06.12 | Himeko | firestrm yes it's broken |
| 20:06.32 | firestrm | Himeko, i was beginning to wonder.. |
| 20:06.54 | Godsey | firestrm are you using devfs? |
| 20:07.01 | modulus_ | tear the ROOF on this muthaFUCKA |
| 20:07.05 | modulus_ | light da ROOF on FIRE |
| 20:07.23 | firestrm | Godsey, duuhh... i dunno..... how do i tell? |
| 20:07.33 | redder86 | ManxPower: I was able to get chan_modem working back in June of last year, but chan_modem only supports half duplex, and really, what's the point of installing Asterisk to merely turn it into an answering machine... something easily done with vgetty. |
| 20:07.56 | redder86 | ManxPower: I did have to hack on the source code for chan_modem to get it working, though. |
| 20:08.12 | firestrm | Godsey, everything was working perfectly until up2date upgraded my kernel for me.. |
| 20:08.24 | modulus_ | redhat UGH |
| 20:08.25 | firestrm | now everything broken |
| 20:08.30 | redder86 | ManxPower: chan_modem doesn't properly support IS-101. |
| 20:08.46 | redder86 | firestrm: recompile zaptel |
| 20:08.59 | redder86 | firestrm: and next time don't let up2date update your kernel. |
| 20:09.03 | HitTop | Manxpower: Could you pls tell me what I need to write on extension for chan_alsa? |
| 20:09.10 | ManxPower | anytime the kernel changes you have to rebuild zaptel |
| 20:09.12 | firestrm | redder86, i did.. make clean;make linux26;make install |
| 20:09.31 | redder86 | firestrm: and did you reload the modules? |
| 20:09.34 | ManxPower | exten => 3311,1,Playback(beep) |
| 20:09.35 | ManxPower | exten => 3311,2,Dial(Console/1) |
| 20:09.46 | firestrm | yes, but it dies on wcfxo |
| 20:09.59 | redder86 | firestrm: what's the error? |
| 20:10.07 | firestrm | line 0: Unable to open master device '/dev/zap/ctl' |
| 20:10.12 | ManxPower | firestrm: You have an X100P? |
| 20:10.15 | firestrm | yes |
| 20:10.19 | viper-net | can i use whatever codec i want or are the technical restrictions ? |
| 20:10.21 | ManxPower | firestrm: Did you read the README.udev too? |
| 20:10.27 | viper-net | the phone ? sip ? |
| 20:10.48 | viper-net | i think about using speex or g.729 |
| 20:10.49 | firestrm | ManxPower, ummm no.. as i said, it was working fine untill i was upgraded |
| 20:11.12 | bjohnson | viper-net: yes some devices/voip providers/software only support certain codecs |
| 20:11.12 | viper-net | with cisco 7905 phones with sip firmware |
| 20:11.42 | viper-net | in conjunction with isdn lines, so i don't have another provider |
| 20:12.14 | ManxPower | firestrm: Read it. |
| 20:12.22 | viper-net | hmm ok, so it depends on the phone if it supports the codec ? but there are no restrictions in sip in general ? |
| 20:12.38 | firestrm | ManxPower, reading it now... i can see hw that could have changed with a kernel upgrade.. but i will try it.. |
| 20:12.46 | modulus_ | which codec does * support the best |
| 20:12.47 | modulus_ | ? |
| 20:12.48 | ManxPower | viper-net: The codecs that Cisco and Asterisk have in common are ulaw, alaw, G726 (depending on the phone), and G729 |
| 20:12.57 | modulus_ | all ulaw seems to work really well for me |
| 20:13.00 | ManxPower | modulus_: All of the free ones. |
| 20:13.05 | *** join/#asterisk |Vulture| (~Vulture@247.131.vbnet.net) |
| 20:14.43 | redder86 | I have an office full of women who apparently have never had speakerphones before, so now they all use them religiously rather than picking up the darn receiver and being polite. Being on speakerphone is so annoying. |
| 20:14.48 | viper-net | hmm would be interesting if the cisco phones only support all these codecs with the original firmware |
| 20:14.55 | viper-net | or also with sip firmware |
| 20:15.11 | firestrm | ManxPower, do i have to rebuild zaptel again after? |
| 20:17.54 | firestrm | ManxPower, or do i have to restart udevd? |
| 20:18.30 | ManxPower | firestrm: I know nothing about 2.6 and udev except what I've seen on this channel and on the mailing lists. I cannot help you further. |
| 20:19.20 | firestrm | ManxPower, ok thanks... that is so wierd, i was running 2.6.9, worked fine.. upgraded to 2.6.10, broken |
| 20:19.23 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 20:19.50 | sgnome` | anyone have a problem with asterisk seg faulting on ultrasparc hardware? |
| 20:20.48 | *** join/#asterisk Shido6 (~shido@d57-87-253.home.cgocable.net) |
| 20:20.53 | modulus_ | sgnome`, what OS? |
| 20:20.56 | *** join/#asterisk DiveFox (~DiveFox@dhcp024-209-065-230.woh.rr.com) |
| 20:20.59 | sgnome` | debian |
| 20:21.01 | Shido6 | yep |
| 20:21.14 | modulus_ | OMG |
| 20:21.14 | sgnome` | it also seg faults on all my FreeBSD x86 boxes |
| 20:21.18 | modulus_ | debian on sparc hardware |
| 20:21.23 | modulus_ | kill me now |
| 20:21.29 | |Vulture| | Anyone know if the "buddy" function or "Status" settings can be used by *? |
| 20:21.35 | |Vulture| | urg in the Polycom IP500 |
| 20:21.43 | DiveFox | Hey, anyone know what would make Asterisk echo at random when bridging SIP to Zap? |
| 20:21.50 | firestrm | modulus_, actually it works very well |
| 20:21.53 | modulus_ | divefox, probably the fxo |
| 20:22.01 | Shido6 | not using a barbietone on speakerphone are we? |
| 20:22.06 | firestrm | modulus_, im running my mailserver in that config |
| 20:22.09 | sgnome` | i don't get it .. it just seg faults every damn time i try to run it |
| 20:22.12 | sgnome` | it installed fine |
| 20:22.15 | DiveFox | It makes a loud ping sound and then echos for 10-15 secs |
| 20:22.34 | tclark | DiveFox: piss poor echo can in * :) |
| 20:22.48 | DiveFox | Well, the customer's getting pissed. |
| 20:22.51 | tclark | yup |
| 20:22.58 | DiveFox | Should I just put crack in the water or something? |
| 20:22.59 | *** join/#asterisk Yoda-BZH (~yoda-bzh@80.125.209.17) |
| 20:23.14 | tclark | get your self an in line echo can if you on t1 to channel bank |
| 20:23.28 | firestrm | arrgh.. still broken.. |
| 20:23.30 | Yoda-BZH | re |
| 20:23.51 | tclark | if your on a tdm pos welcome the echo with open arms |
| 20:23.56 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 20:23.56 | sgnome` | ========================================================================= |
| 20:23.56 | sgnome` | [ Booting...........WARNING[1024]: File chan_iax.c, Line 4289 (set_config): Cannot allow unknown format 'uall' |
| 20:23.56 | sgnome` | .WARNING[1024]: File chan_oss.c, Line 410 (soundcard_init): Unable to open /dev/dsp: No such device |
| 20:23.56 | sgnome` | ...................Segmentation fault |
| 20:24.05 | Dr-Linux | :( |
| 20:24.06 | modulus_ | heh |
| 20:24.07 | modulus_ | uall? |
| 20:24.13 | modulus_ | never heard of that codec |
| 20:24.18 | sgnome` | i just installed it .. never touched anything |
| 20:24.20 | modulus_ | i'd segfault too |
| 20:24.26 | modulus_ | and dump core on your chest |
| 20:24.28 | modulus_ | oh baby |
| 20:24.30 | sgnome` | so what file needs to be modified |
| 20:25.04 | modulus_ | just cvsup HEAD |
| 20:25.11 | sgnome` | ? |
| 20:25.14 | modulus_ | i get HEAD every day |
| 20:25.19 | tzanger | modulus_: uall is the codec used in the Southern U.S. a lot |
| 20:25.19 | modulus_ | and it's wonderful |
| 20:25.29 | modulus_ | tzanger, really? never heard of it. |
| 20:25.36 | bjohnson | hehe |
| 20:25.38 | sgnome` | so you're saying cvsup instead of using the .deb |
| 20:25.38 | modulus_ | of course you could be pulling my leg |
| 20:25.39 | tzanger | u'aul is used extensively for data transfer |
| 20:25.39 | sgnome` | aght |
| 20:25.44 | bjohnson | sgnome`: ulaw? |
| 20:25.58 | sgnome` | it's a new install .. i didn't change shit |
| 20:26.24 | DiveFox | Odd, it seems only the local SIP interfaces hear it |
| 20:26.42 | bjohnson | sgnome`: any chance you looked in iax.conf for a line that says uall |
| 20:27.53 | tzanger | modulus_: actually I am pulling your leg.. uall is a protocol used down there, not a codec... as in "uall have a good day now, yahear?" |
| 20:27.53 | tzanger | as you can see, it's a framing protocol |
| 20:28.19 | sgnome` | no |
| 20:28.32 | sgnome` | i am cvsuppin asterisk .. i guess the deb is just messed up |
| 20:29.22 | tclark | DiveFox: you have tdn card with fxs and doing a fxo in o out bound no echo ? |
| 20:29.56 | DiveFox | What? I have SIP phones and FXO cards. Internal phones are on SIP. |
| 20:30.31 | tclark | i mean do you have fxs devices on that tdm card ? |
| 20:30.46 | sgnome` | bah .. the cvs version of asterisk i got won't compile |
| 20:30.54 | DiveFox | No |
| 20:31.05 | bjohnson | hmm 355ms to iaxtel |
| 20:31.18 | Nugget | sgnome`: head or stable? |
| 20:31.32 | sgnome` | dunno |
| 20:31.34 | tclark | ah ok so you can point a smoking gun at the fxo card per say it might be some issue with the sip devices |
| 20:31.37 | sgnome` | i followed the instructions on the site |
| 20:31.38 | Nugget | well, that's kind of significant. |
| 20:31.54 | sgnome` | how can i tell? |
| 20:31.55 | tclark | err can't |
| 20:32.03 | *** join/#asterisk edeehobb1 (~ehobbs@rrcs-24-242-177-226.sw.biz.rr.com) |
| 20:32.06 | Nugget | what did you do to check it out? |
| 20:32.20 | *** part/#asterisk roche (~roche@216.194.173.2) |
| 20:32.47 | tclark | DiveFox: have tried to minimze the echo via tx/rx gain & tried some of the alternate echo cans ? |
| 20:32.51 | ManxPower | sgnome`: PASTE the result of "show version" |
| 20:33.01 | Nugget | ManxPower: he can't get it to compile. |
| 20:33.23 | DiveFox | It could be the RX gain being cranked way too high at customer's request? |
| 20:33.26 | *** join/#asterisk cdegroot (~Cees@tai42.xs4all.nl) |
| 20:33.27 | sgnome` | cd /usr/src ; export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot ;cvs login ; cvs checkout libpri asterisk |
| 20:33.39 | Nugget | the digium download page is poorly organized. I wish it didn't show the cvsup first on top. |
| 20:33.40 | DiveFox | RX gain is like... 10.0 |
| 20:33.44 | ManxPower | sgnome`: That is the dev/unstable asterisk |
| 20:33.48 | DiveFox | TX gain is 0.0 |
| 20:33.49 | tclark | yah the will bite ya |
| 20:33.55 | Nugget | the stable line is way at the bottom and hard to miss. |
| 20:34.17 | tclark | you can trt yhe monitor app in zaptel to tune the gains |
| 20:34.19 | wolfson` | anything special to enable the new transfer stuff in cvs. I added the lines to features.conf, but *X seems to not do anything |
| 20:34.49 | bjohnson | sgnome`: don't stop at the line that says sudo cd /; rm -rf |
| 20:35.00 | bjohnson | sgnome`: keep going |
| 20:35.05 | sgnome` | ? |
| 20:35.17 | bjohnson | sgnome`: read me friend |
| 20:35.20 | Nugget | sgnome`: # cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds |
| 20:35.22 | bjohnson | sgnome`: read my friend |
| 20:35.23 | Nugget | that's what you want. |
| 20:35.43 | Nugget | you want to check out the "v1-0" branch |
| 20:36.11 | *** part/#asterisk edeehobb1 (~ehobbs@rrcs-24-242-177-226.sw.biz.rr.com) |
| 20:36.16 | ManxPower | wolfson`: Make sure your phone is passing those digits to Asterisk |
| 20:36.35 | Nugget | bjohnson: is just trying to make a joke, but it's not funny and wouldn't even work if you fell for it. |
| 20:37.04 | bjohnson | yeah .. I keep screwing up my jokes |
| 20:37.14 | *** join/#asterisk zugzwang (~ehobbs@rrcs-24-242-177-226.sw.biz.rr.com) |
| 20:37.14 | tclark | DiveFox: try this for dtls on tuning tx/rx http://www.voip-info.org/tiki-index.php?page=Asterisk%20x100p%20echotraining#comments |
| 20:37.19 | Nugget | the rm command is malformed, lacking a target, and wouldn't be run within sudo anyway since the ";" terminates the sudo command. |
| 20:38.00 | wolfson` | manxpower, ok, will try again. i see it mapping when I reload res_features. I assume a t or T has to be in the Dial command as well |
| 20:38.10 | sgnome` | that version won't compile either : |
| 20:38.12 | sgnome` | :/ |
| 20:38.21 | Moc | Anyone got DID trunk with analog line with * ? |
| 20:38.21 | ManxPower | wolfson`: ONLY if you are trying to use # transfers. |
| 20:38.21 | Nugget | sgnome`: well, then the problem probably isn't asterisk's fault. |
| 20:38.48 | ManxPower | There HAVE been some reports of today's CVS (stable and head) are not compileing. |
| 20:38.52 | wolfson` | manxpower: so atxfer type should work without a t or T? just want to make sure |
| 20:39.01 | bjohnson | Moc: ? |
| 20:39.17 | ManxPower | wolfson`: ANY transfer will work with t and T except #-based transfers. |
| 20:39.21 | ManxPower | ..er.. |
| 20:39.21 | Nugget | ManxPower: interesting, but since he also couldn't get the debian package to install (which led him to cvs) I still suspect local problems. |
| 20:39.26 | ManxPower | wolfson`: ANY transfer will work WITHOUT t and T except #-based transfers. |
| 20:39.35 | wolfson` | manxpower: ok, thanks |
| 20:39.35 | ManxPower | Nugget: Ah. Yes. |
| 20:39.48 | *** part/#asterisk zugzwang (~ehobbs@rrcs-24-242-177-226.sw.biz.rr.com) |
| 20:40.03 | ManxPower | t/T are miserable excuses of a hack and should not even be documented. |
| 20:40.11 | sgnome` | well .. the reason i am trying this machine is because the other box i had it on .. and got it working fine on .. had everything configured .. just stopped workgin out of nowhere |
| 20:40.19 | wolfson` | ought to be a seperate: don't native transfer flag |
| 20:40.50 | *** join/#asterisk ToyMan (~stuq@user-12lcqq2.cable.mindspring.com) |
| 20:40.57 | ManxPower | or maybe things like T and t and r should be only shown with the command "show application dial hacks" |
| 20:41.51 | *** join/#asterisk JakBeatZ (~JakBeatZ@216.7.194.254) |
| 20:42.07 | wolfson` | Remapping feature Attended Transfer (atxfer) to sequence '*2' -- yet *2 from a sip phone does nothing. same phone that a # transfer has worked from |
| 20:42.13 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 20:42.19 | ctooley | Hello again. |
| 20:42.20 | JakBeatZ | Hey, anyone know where I can get app_conference? wiki says sourceforge cvs tarball but it doesn't seem to include it from what I can see :/ |
| 20:42.32 | ManxPower | wolfson`: Perhaps the phone is eating the code. Phone do that. Often. |
| 20:42.52 | wolfson` | i can hear the dtmf on the other leg of the call |
| 20:42.58 | ManxPower | The SIPura devices have LOTS of local calling features accessed by disling numbers starting with * |
| 20:43.13 | wolfson` | its a budgettone, also tried with a cisco 7960 |
| 20:43.27 | nestAr | gah. using somoene elses ipmid.cfg for my polycom's.. can't get his lastcallreturn to go away |
| 20:43.40 | ManxPower | wolfson`: What does show application dial show for attended transfers? |
| 20:44.58 | wolfson` | manxpower: not sure what you mean, there is not an attended transfer flag for app_dial with the new cvs code |
| 20:45.47 | *** join/#asterisk lo7k (~charly@pD9E006EC.dip.t-dialin.net) |
| 20:45.59 | Hymie | hmm |
| 20:46.04 | Hymie | distinctive ring doesn't seem to work here |
| 20:46.22 | *** join/#asterisk drachenfels (~ba@sm1ple2.gotadsl.co.uk) |
| 20:46.28 | ctooley | nestAr, pastebin it if you're needing help |
| 20:46.28 | drachenfels | lo all. |
| 20:46.35 | Hymie | that is, Dial(Zap/1r2) doesn't do a distictive ring?? |
| 20:47.03 | drachenfels | Got a problem with our digium card, think it's hardware related. |
| 20:47.21 | ctooley | I know this may sound dumb but I wanted to check for sure, I will never get anything except 0-9 into SetCIDNum, correct? |
| 20:47.23 | drachenfels | it's a tdm400 |
| 20:47.23 | DiveFox | Now, how do I do transfer from sip phone to sip phone? |
| 20:47.44 | ctooley | DiveFox, depends on how you set up your extensions. |
| 20:47.51 | drachenfels | I can call sip -> pstn fine on ports 1,3, and 4 |
| 20:47.57 | lo7k | current -rv1.0 zaptel cvs: battdebounce compile errors. zaptel/wcfxs.c:2190: Fehler: »battdebounce« nicht deklariert (erste Benutzung in dieser Funktion) |
| 20:47.58 | drachenfels | just port 2 doesn't work :( |
| 20:48.01 | ctooley | DiveFox, and which phones you have |
| 20:48.02 | lo7k | does russel know that` |
| 20:48.10 | DiveFox | 2 xlite and 1 barbietone |
| 20:48.17 | blitzrage | I'm having a problem where I'm connecting to a couple of providers. They are registered fine, and username and secret are defined (IAX2). However, when I do a Dial(IAX2/provider/${EXTEN},23), the dial executes fine, but the Asterisk output says Ringing then Hungup pretty quickly after another, and the call just goes to fast busy. What obvious thing have I overlooked? |
| 20:48.18 | drachenfels | if I call into the pbx I get an engaged tone !? |
| 20:48.30 | drachenfels | and I can't call out... |
| 20:48.34 | drachenfels | any ideas ne1? |
| 20:49.13 | rsb__ | anyone have a good source for cheap polycoms? |
| 20:49.14 | ctooley | blitzrage, if you replace "provider" with "username:secret@hostname" does it work? |
| 20:49.23 | bjohnson | tzanger: I modified your call macro to include a count setting and a CID setting to limit how many calls go out on a group |
| 20:49.29 | ctooley | rsb__, depends how cheap. |
| 20:49.42 | ctooley | rsb__, and which phones |
| 20:50.03 | rsb__ | ctooley, well, cheaper than atacomm. I need 50 phones, a mix of 300s, 500s, 600s |
| 20:50.06 | drachenfels | also, does anyone know what this means: application asterisk uses obsolete OSS audio interface |
| 20:50.15 | drachenfels | I get it when typing dmesg |
| 20:50.47 | ctooley | rsb__, our current prices are 140, 190, and 280 |
| 20:51.47 | bjohnson | does iaxtel have multiple proxies? |
| 20:52.08 | *** part/#asterisk allyour80211b (~allyour80@208.178.154.99) |
| 20:52.08 | ctooley | drachenfels, are you using ALSA on the machine? |
| 20:52.18 | blitzrage | ctooley: I was trying to avoid that, but you're right, I should try that next to make sure the calls are at least able to authenticate that way. |
| 20:52.37 | ctooley | blitzrage, if that works, then I'd be happy to look at your configs to help. |
| 20:53.54 | wolfson` | manxpower: atxfer would not work till t was added to the dial command |
| 20:54.01 | *** part/#asterisk DiveFox (~DiveFox@dhcp024-209-065-230.woh.rr.com) |
| 20:54.16 | *** join/#asterisk FryGuy (fryguy@c-24-2-50-122.client.comcast.net) |
| 20:54.30 | ManxPower | wolfson`: The GS phone only supports blindxfer, so you would need t for that, what about the other phone you have? |
| 20:55.01 | *** join/#asterisk JochenA (~jochen@12-216-247-62.client.mchsi.com) |
| 20:55.29 | bjohnson | if I want to count concurrent voip calls I can do it for outgoing .. but how do I do it for incoming? just setgroup accordingly for each incoming call before answer? |
| 20:55.45 | wolfson` | manxpower: 6970 which has attended native, but thats why I asked if you needed t for the new dtmf based atxfer code |
| 20:55.55 | wolfson` | manxpower: it works fine now though |
| 20:56.25 | ManxPower | wolfson`: You only need t or T for Asterisk handled DTMF transfer. |
| 20:57.01 | wolfson` | manxpower: thats what I thought. then you said only #, and I was confused. All is well. |
| 20:57.10 | ScubaSteve | Anyone in here use BroadVoice BYOD? |
| 20:57.49 | ManxPower | wolfson`: All DTMF transfers are just # transfers. If you want to change the key trigger, that's just cosmetic. |
| 20:57.51 | ctooley | ScubaSteve, I do |
| 20:58.24 | nestAr | ctooley: thanks, i sat down and read the manual. i was grepping for "lastcall" and i wanted "last-call" |
| 20:58.27 | ScubaSteve | ctooley: Cool. Do they allow you to setcallerid on your outbound calls? I plan on using them for outbound only, and need the number to show as my inbound. |
| 20:58.31 | nestAr | i just needed to vent. :) |
| 20:58.42 | JakBeatZ | Hey, anyone know where I can get app_conference? I've downloaded the cvs snapshot as per the wiki but app_Conference doesn't seem to be included |
| 20:58.45 | ctooley | ScubaSteve, you can set it all you want, they overwrite mine though. |
| 20:58.47 | JakBeatZ | IT's not in freebsd ports is it? |
| 20:58.59 | *** join/#asterisk berrayahkamel (~berrayahk@82.101.128.6) |
| 20:59.00 | ScubaSteve | ctooley: That's what I figured. Darn. |
| 20:59.19 | ScubaSteve | ctooley: Thanks for answering, saved me the hassle of signing up and cancelling it. |
| 20:59.30 | ManxPower | JakBeatZ: A better questions would be "Does ANYONE here use app_conference?" I suspect the answer will be "no" |
| 20:59.37 | ctooley | ScubaSteve, usually if your plan is too good to be true, it's not true. I tried broadvoice for outbound voicepulse for inbound too, it doesn't work out so well. :) |
| 21:00.14 | Hymie | ScubaSteve: livevoip allows you to see callerid on outgoing calls.. but ... well, I haven't been entirely happy with their service yet |
| 21:00.39 | ScubaSteve | ctooley: Yeah.. I hear ya... I'm looking for an unlimited, at least unlimited local outbound plan. |
| 21:00.55 | ScubaSteve | ManxPower .. hug away dude. |
| 21:01.03 | file[laptop] | ooh muffin tastic |
| 21:01.04 | ScubaSteve | File! Quit it. |
| 21:01.13 | file[laptop] | what? I needed to butter my muffin |
| 21:01.52 | nestAr | ScubaSteve: i feel your pain |
| 21:01.56 | ScubaSteve | file.. you have a muffin? |
| 21:01.56 | nestAr | i've got a couple of those |
| 21:02.05 | Hymie | ScubaSteve: ? |
| 21:02.06 | file[laptop] | why yes, I do |
| 21:02.09 | ScubaSteve | hee hee |
| 21:02.49 | mquin | mmm... muffins |
| 21:02.58 | Hymie | ScubaSteve: are you blind, or rude? |
| 21:03.06 | ScubaSteve | Hymie.. huh? |
| 21:03.19 | Hymie | ScubaSteve: scroll back, I've been typing at you |
| 21:03.33 | ScubaSteve | Hymie.. sorry let me look. |
| 21:04.03 | ScubaSteve | Hymie: Not blind, not rude.. just stupid :) |
| 21:04.19 | bjohnson | ScubaSteve: voicepulse? |
| 21:04.35 | ScubaSteve | bjohnson: Connect IAX service doesn't do unlimited anything. |
| 21:04.37 | Hymie | ScubaSteve: livevoip has had some downtime on me.. but the quality of service has been great when they are working |
| 21:04.59 | ScubaSteve | bjohnson: Their support sucked rocks. I had a problem when I was a customer (was) and they didn't even know what IAX was. |
| 21:05.18 | ScubaSteve | bjohnson: The only way to get things done was to /IM Charlie (VoicePulse) in here and beg.. |
| 21:05.29 | bjohnson | good to know |
| 21:05.41 | ScubaSteve | Hymie: I recall looking at LV's website some time back and it had very little info about their voip product... |
| 21:05.55 | ScubaSteve | Hymie: Am poking around now... also downtime.. happens :( |
| 21:07.02 | ScubaSteve | Hymie: Their "Asterisk Plan" doesn't seem like a great deal at all. |
| 21:07.53 | file[laptop] | 1.29 cents/minute, billed 6/6 isn't good? |
| 21:07.56 | Hymie | ? |
| 21:08.06 | bjohnson | ScubaSteve: Connect IAX = iax.cc? |
| 21:08.09 | ScubaSteve | Well... They have a $30/month minimum! |
| 21:08.13 | Hymie | no |
| 21:08.17 | Hymie | not per month |
| 21:08.19 | Hymie | per purchase |
| 21:08.22 | ScubaSteve | bjohnson: No.. iax.cc = SixTel ... I use them for inbound. |
| 21:08.25 | Hymie | the minutes are good for a year or so |
| 21:08.26 | ScubaSteve | Hymie: ah. |
| 21:08.31 | Hymie | 30 per month is insane ;) |
| 21:08.35 | file[laptop] | silly people |
| 21:08.39 | ScubaSteve | Hymie: Ok... Rates are about the same as VoipJet then (usa). |
| 21:08.54 | JakBeatZ | ManxPower: So everyone uses MeetMe? Hmm.. |
| 21:09.06 | Hymie | ScubaSteve: their 800 number is a good rate.. $0.99 per month + minutes used.. but it seems down more (incoming) than their regular service |
| 21:09.16 | *** join/#asterisk RDF (~leonardo@S010600055d210201.vs.shawcable.net) |
| 21:09.20 | ScubaSteve | bjohnson: I think VoicePulse and SixTel both backend to Level3. Almost had the same DID when I cancelled VP and went with 6. |
| 21:09.52 | ScubaSteve | Hymie: Seems like you've had a lot of downtime with LV. |
| 21:10.00 | odie_flocon | RDF are you from Vancouver??? |
| 21:10.08 | RDF | odie yes i am. |
| 21:10.16 | odie_flocon | RDF Coool.... |
| 21:10.45 | Hymie | ScubaSteve: if you think of the 24 hours in the day, then think of the # of times you use the phone.. THEN think of the amount of downtime.. it makes me wonder what downtime I'm _not_ seeing |
| 21:10.48 | odie_flocon | I'm in Freezing Calgary |
| 21:10.55 | RDF | I guess...im not to happy right now actually a little pissed had a nice dial plan made out and lost power here. It was only 1 second and that was it. |
| 21:11.05 | RDF | lost it. |
| 21:11.19 | Darwin35 | we should do what asterisk is best for trunking and sharing connections |
| 21:11.19 | odie_flocon | that's what save is for.... |
| 21:11.21 | RDF | Time to ask for a ups |
| 21:11.37 | Hymie | what is your editor?! vi or emacs should have a backup saved |
| 21:11.38 | Darwin35 | everyone trunk to everyone and then no need to worry |
| 21:11.38 | odie_flocon | yeah UPS is good. |
| 21:11.44 | RDF | odie, ive only been on for a hour when it happened. yea, i save before i close it but was in the middle of it. |
| 21:11.44 | bjohnson | ScubaSteve: broadband.spectravoice.com it's CDN |
| 21:11.54 | *** join/#asterisk blankman (~blankman@h000d88a1570c.ne.client2.attbi.com) |
| 21:11.55 | ScubaSteve | bjohnson: Let's take a peek. |
| 21:12.01 | odie_flocon | hmmmmmmmmmmmm. |
| 21:12.05 | odie_flocon | that sux. |
| 21:12.08 | bjohnson | ScubaSteve: I have no idea what service is like |
| 21:12.11 | odie_flocon | hey I gotta get back to work now. |
| 21:12.17 | blankman | Hey guys. |
| 21:12.26 | RDF | Hymie, well for some reason vi is not working. I was editing in vi |
| 21:12.28 | RDF | err |
| 21:12.30 | RDF | vim |
| 21:12.30 | Hymie | bjohnson: any plan that has a minimum fee is evil |
| 21:12.38 | RDF | hymie, i was in vim |
| 21:12.51 | RDF | vim drives me nuts. need to get vi reinstalled. :) |
| 21:12.55 | *** join/#asterisk greg_work (~greg@d221-73-198.commercial.cgocable.net) |
| 21:12.57 | Hymie | RDF: I THINK YOU ARE A LIAR! DO YOU HEAR ME, RDF??!! I THINK YOU LIE! |
| 21:12.58 | bjohnson | Hymie: he was looking for unlimited outbound |
| 21:13.00 | Hymie | heheh |
| 21:13.03 | Hymie | RDF: just kidding ;) |
| 21:13.09 | blankman | So, I have two questions. Is there anyone on that is using th 410 with more than one t1 connected? |
| 21:13.11 | Hymie | bjohnson: really? |
| 21:13.18 | Hymie | bjohnson: he's insane then ;) |
| 21:13.19 | RDF | RDF never lies it always finds the source |
| 21:13.20 | bjohnson | Hymie: I _think_ so |
| 21:13.46 | ScubaSteve | Hear what? |
| 21:13.48 | bjohnson | Hymie: I think that's why he was with Broadvoice |
| 21:13.48 | Hymie | bjohnson: at the cheapo rates of soem voip dudes though, unlimited isn't as attactive for me, I guess |
| 21:14.00 | RDF | brb |
| 21:14.03 | bjohnson | Hymie: totally depends on volume |
| 21:14.20 | blankman | I have some config questions about the channel numbering. |
| 21:14.21 | ScubaSteve | bjohnson: Hm, try signing up on that site.. no matter what area code/city you pick, it says "no, can't do that, don't have any DID's there"... |
| 21:14.26 | ScubaSteve | bjohnson: Pretty amusing. |
| 21:14.30 | nestAr | is there anyway to make the polycom's download the main directory without reseting them to defaults? |
| 21:15.03 | bjohnson | ScubaSteve: oh well |
| 21:15.03 | ScubaSteve | bjhonson: I also forgot to mention service has to be SIP/IAX. |
| 21:15.03 | ctooley | nestAr, should just touch the timestamp on the directory |
| 21:15.17 | blankman | Second question. I am trying to get the spool to work, and I have tried to ge the "document" that is on the wiki to work, but asterisk says it doesn't know about: Action ... any idea's? |
| 21:15.17 | ScubaSteve | bjohnson: Also PHP/mysql errors on the page.. Warning: mysql_num_rows(): supplied argument is not a valid MySQL result resource in /usr/local/apache2/broadband/signup_phone1.php on line 15 |
| 21:15.19 | bjohnson | ScubaSteve: I think there are SIP .. the hardware they provide if you buy it is Mediatrix 2102 or something .. a 2 port SIP fxs |
| 21:15.42 | bjohnson | ScubaSteve: so from your quicl review .. what is still in the running? |
| 21:16.17 | ScubaSteve | bjohnson: Well, I really need to be able to set callerid for outbound calls if the provider doesn't have 919 area codes. |
| 21:16.41 | ScubaSteve | bjohnson: Would have been thrilled with BV but absolutely need a 919 DID (which they don't have) or the ability to set the DID to a 919 number I already have for incoming. |
| 21:17.11 | blankman | ScubaSteve: ... did you look at NuFone? |
| 21:17.25 | ScubaSteve | Hymie I still want 'unlimited' to some degree though. |
| 21:17.34 | blankman | I know I am late to this line ... but they will let you do that ... as will gafachi. |
| 21:18.11 | RDF | im looking at a local voip carrier and dial back into my box to test my connection. Does anyone know of a voip carrier to setup a end user account to do such? |
| 21:18.21 | ScubaSteve | blankman: 1000 minutes for $22/mo isn't bad... In theory shouldn't break that ever.. but would be OK with paying $50 a month and not having to worry about it... |
| 21:18.48 | ScubaSteve | RDF VoipJet will give you 25 cents worth of outbound calls by IAX ,,, |
| 21:19.14 | Hymie | ScubaSteve: well, 1.3 cents per minute is the same as about 2000 minutes for $26 per month |
| 21:20.00 | sgnome` | ok .. i got asterisk compiled and running fine .. but now my damn softphone keeps timing out during login |
| 21:20.05 | sgnome` | what could be causing that? |
| 21:20.12 | RDF | scuba, what client voip software should i work with that? |
| 21:20.15 | ScubaSteve | Hymie: Hadn't actually sat down and done the math. As I said, I'd gladly pay $50/mo to not have to deal with an occasional high bill. |
| 21:20.38 | ScubaSteve | Hymie: If I could use Asterisk with Packet8 and not have to use the dorky ATA with an X100P I'd use that. |
| 21:20.38 | Hymie | ScubaSteve: er... but at the above, rate, that's 33 hours of calls for $26 per month... howeve,r if you use LESS than that, you pay less! |
| 21:20.51 | ScubaSteve | Packet8 with the ATA under * stinks, but otherwise the Packet8 service is fantastic.. |
| 21:21.06 | ScubaSteve | Hymie: I understand how that works :) |
| 21:21.14 | sgnome` | i don't even see the softphone attempting to register in *CLI |
| 21:21.20 | Hymie | ScubaSteve: I'm just saying it's going to be hard for you to break 33 hours in a month |
| 21:21.21 | telme_ | have you looked at teliax? |
| 21:21.28 | *** part/#asterisk mogorman (~mogorman@216.207.244.182) |
| 21:21.36 | ScubaSteve | Hymie: True. |
| 21:21.52 | blankman | ScubaSteve, you just want insurance that you wont have to pay more than 50 on any given month right? |
| 21:22.05 | ScubaSteve | blankman: Pretty much. |
| 21:22.11 | Hymie | that's 60+ hours of calls though with livevoip ;) |
| 21:22.21 | ScubaSteve | Hymie: Nice! |
| 21:22.22 | blankman | So if some came to you and give you an agreement like we do for bandwidth you would be cool with that yes? |
| 21:22.45 | ScubaSteve | blankman: Do you provide IAX termination service? |
| 21:22.49 | *** join/#asterisk santiago (~santiago@63.245.86.97) |
| 21:22.57 | telme_ | ScubaSteve: give teliax a try |
| 21:22.58 | Hymie | ScubaSteve: heheh, I was wonderign that myself ;) |
| 21:23.20 | RDF | scuba, what client voip software should i work with that? |
| 21:23.25 | blankman | yes ... but only iax. |
| 21:23.37 | ScubaSteve | blankman: /msg me! |
| 21:23.38 | Silik0n | firefly == great IAX2 client |
| 21:23.52 | ScubaSteve | RDF: No idea. I don't use softphones. |
| 21:23.59 | sgnome` | why would x-lite time out during login? |
| 21:24.14 | ScubaSteve | BRB smokie treat |
| 21:24.16 | Hymie | ScubaSteve: the best thing about setting your callerID, is that you can scare stupid people. I called a friend of mine, and set my caller ID to his number! When he answered, I told him I was in his basement! He was freaked!!! |
| 21:24.16 | sgnome` | and not even show up as trying to register in *CLI? |
| 21:24.23 | Silik0n | login info is wrong or you just added it to sip.conf and didnt do sip reload |
| 21:24.50 | Silik0n | make sure all the settings in the SIP Proxy configs are correct |
| 21:24.51 | RDF | Silkon mmm i have voipet and thay give me alot of information. more then goes in firefly. |
| 21:24.57 | sgnome` | login info is correct and i did reload |
| 21:25.03 | Silik0n | and if you have to, set verbose 5 |
| 21:25.34 | RDF | silkon you have a account with them? |
| 21:25.41 | Silik0n | with who? |
| 21:25.49 | RDF | voipjet? |
| 21:26.00 | sgnome` | Silik0n .. even in verbose 5 i still see nothing |
| 21:26.09 | sgnome` | and the login timed out again |
| 21:26.28 | HitTop | hi, when I load chan_alsa.so, I got the error: No sound card detected -- console channel will be unavailable. Can anyone tell me wat I am missing? |
| 21:26.29 | Silik0n | nope no voipjet account... they guy thgat runs that used to pop in here on occation as voipjet |
| 21:26.55 | RDF | ohh |
| 21:27.06 | Silik0n | sgnome` : double check the username and password and the IP address of the sip proxy, and make sure register is set to always (or default) in x-lite |
| 21:27.34 | blankman | Soooo ... anyone help me with the pbx_spooler application not lik'n my Action: command? |
| 21:27.48 | *** join/#asterisk Pantanero (~Pantanero@a213-22-82-157.netcabo.pt) |
| 21:27.51 | Silik0n | sgnome` and I have noticed at times that if the context in sip.conf for the x-lite client doesnt match the user name, it doesnt want to work sometimes, not sure why tho |
| 21:28.14 | RDF | yea, thay are a whosale account. give the rudimentry config info and basicly you are on your own :) not something for the public to understand unless you have a understanding of it. I did exactly as it was stated in the instructions and tested it out via cli early on and got no responce. |
| 21:28.20 | sgnome` | the context does match |
| 21:28.26 | HitTop | or even for chan_oss.so, I got the same msg and couldn't get sound to work in console |
| 21:28.28 | bjohnson | telme_: you use teliax? |
| 21:28.41 | *** join/#asterisk Samoied (~samoied@popeye.opens.com.br) |
| 21:28.42 | *** join/#asterisk JerJer (~JerJer@d11-164.rt-bras.che.centurytel.net) |
| 21:28.59 | Samoied | anyone use a FXO channel bank? |
| 21:29.08 | *** join/#asterisk scythelx (~fuckoiff@near.outpost.com) |
| 21:29.26 | blankman | Hey JerJer ... did you ever get you cluster working? |
| 21:29.30 | bjohnson | Samoied: you're going from trying to use a Conexant modem to a channel bank? That's a huge jump |
| 21:29.49 | sgnome` | Silik0n it still times out |
| 21:29.51 | Samoied | bjohnson: no, no |
| 21:30.01 | scythelx | for a large scale asterisk rollout with 10000+ extensions, is it possible for different companies/accounts to share the same extension? or would each extension have to be unique throughout the whole dialplan |
| 21:30.06 | sgnome` | and when i try using iaxcomm .. it also does not show up as attempting to register in *CLI |
| 21:30.17 | *** part/#asterisk ethzer0 (~ethzer0@d141-233-214.home.cgocable.net) |
| 21:30.18 | Samoied | bjohnson: I want to use a real channel bank. |
| 21:30.25 | RDF | Hymie does vim backup text into a file if the power is lost? |
| 21:30.35 | sgnome` | hrmm |
| 21:30.36 | Samoied | bjohnson: you recommend one? |
| 21:30.40 | Silik0n | sgnome`: does your asterisk box have multiple ethernet interfaces? |
| 21:30.47 | sgnome` | no |
| 21:30.51 | Hymie | RDF: it backs up files as soon as you edit them |
| 21:30.57 | sgnome` | lmfao .. i think i found the problem |
| 21:30.58 | bjohnson | Samoied: I don't have one but Adtran 750's have been popular |
| 21:31.04 | Hymie | RDF: you weren't working in /tmp, were you? |
| 21:31.09 | Silik0n | is eth0 actuallyup and running? (can you ping the * box from the windows client) |
| 21:31.14 | sgnome` | there's nothing regarding sip when i do help |
| 21:31.19 | sgnome` | i am logged into it via ssh atm .. yeah |
| 21:31.20 | RDF | hymie, well this was a new file. and how do i know i was working in tmp? |
| 21:31.28 | Silik0n | firewall? |
| 21:31.32 | Hymie | RDF: you didn't save it once? |
| 21:31.40 | Hymie | RDF: if not, you're fubarred |
| 21:31.47 | sgnome` | the .deb i installed had nothing dealing with sip .. i am gonna instal the cvs version |
| 21:31.50 | Hymie | RDF: next time give a scratch name, even, to the file |
| 21:31.52 | Hymie | <PROTECTED> |
| 21:31.52 | Moc | fucking voip DID providers... |
| 21:31.58 | Silik0n | if IAX and SIP arent connecting it sounds like you have a default denny policy firewall |
| 21:32.00 | Samoied | bjohnson: but it have a ethernet connectuon? |
| 21:32.05 | Samoied | bjohnson: or T1 only? |
| 21:32.05 | RDF | Hymie a scratch name? |
| 21:32.08 | Silik0n | moc quit your bitching |
| 21:32.24 | sgnome` | i have rules setup for the sip already |
| 21:32.29 | bjohnson | Samoied: can be picked up used on ebay loaded for about $700 |
| 21:32.37 | nestAr | hrmm.. i touched the 000000000000-directory.xml file and rebooted the phone.. phone didn't pick up the changes |
| 21:32.41 | RDF | Hymie i made it as extensions.conf and it opened as a new file. |
| 21:32.44 | bjohnson | Samoied: usually T1 |
| 21:32.59 | Samoied | bjohnson: I want one with Ethernet |
| 21:33.16 | Samoied | For rack with a asterisk server |
| 21:33.20 | Hymie | RDF: so you were editing the file "extensions.conf", not just editing a nameless file? |
| 21:33.24 | bjohnson | how many ports? |
| 21:33.38 | Samoied | bjohnson: 12-24 |
| 21:33.41 | Hymie | RDF: do a ls -la in the directory you were editing in, then, if it has a real name.. if not, it's gone |
| 21:33.42 | RDF | no, i created it as vi /etc/asterisk/extensions.conf |
| 21:33.53 | bjohnson | scythelx: however you wanted to set them up |
| 21:33.54 | RDF | k |
| 21:33.55 | Hymie | so, do that again.. and it should prompt you to recover |
| 21:34.06 | Silik0n | sgnome` well first disable the firewall allow any to any and test it |
| 21:34.23 | bjohnson | Samoied: look at mediatrix or quitum units .. you will be looking at $3000 and up |
| 21:35.13 | RDF | Hymie, I found the file but it has the old file info in it before i saved the new info. |
| 21:35.19 | *** join/#asterisk epoch (epoch@octane.breakbeats.org) |
| 21:35.21 | Moc | Anyone here in MTL who want DIDs ? |
| 21:36.00 | Silik0n | hah |
| 21:36.10 | Silik0n | www.moc-voip.com |
| 21:36.17 | Moc | no www.moctel.com ;) |
| 21:36.18 | Samoied | bjohnson: But if I use a channel bank with a T1 interface, I have to buy a T1 card |
| 21:36.23 | Silik0n | moc: i'll take a couple of MTL DIDs if they are free |
| 21:36.31 | Moc | you crazy |
| 21:36.38 | Samoied | bjohnson: How cost a T1 card? $1000 ? |
| 21:36.41 | Silik0n | <PROTECTED> |
| 21:36.44 | Moc | Im not gonna spend $$$ to give free DID!! |
| 21:36.56 | Silik0n | heh |
| 21:37.02 | Moc | Samoied, depend check digium site |
| 21:37.29 | Moc | Im probably going to go with analog DID trunk |
| 21:37.40 | Samoied | No |
| 21:37.43 | Silik0n | a single port T1 card is only about 495 |
| 21:37.44 | Samoied | $500 |
| 21:37.55 | Samoied | $500 + $700 |
| 21:38.05 | Samoied | $1200 for 24 FXO ports |
| 21:38.24 | Moc | 700$ ? what it that 700$ ? |
| 21:38.48 | Samoied | Moc: for Channel bank |
| 21:38.54 | RDF | lease fee |
| 21:38.57 | Moc | ha |
| 21:39.10 | Silik0n | moc what about trunking on bri? |
| 21:39.18 | *** join/#asterisk DavidSpears (~spears_da@65.163.100.254) |
| 21:39.24 | RDF | I think pri here is 1,200 in bc canada |
| 21:39.33 | Samoied | TDM400P with 4 FXO is $337 |
| 21:40.08 | Samoied | $337 x 6 = 2022 |
| 21:40.47 | Samoied | Channel bank with T1 is better choice :) |
| 21:40.47 | scythelx | <PROTECTED> |
| 21:42.08 | Silik0n | what you mean multiple companies? |
| 21:42.26 | Silik0n | as in multiple stations show as the same extension? |
| 21:42.39 | bjohnson | scythelx: however you wanted to set them up |
| 21:43.02 | markit | anyone here using the new "native" transfer of asterisk cvs head? patch 3241 |
| 21:43.35 | scythelx | well, company a has extensions 1000-2000 , company b also has extensions 1000-2000. would that conflict with eachother? would i have to make company b only use extensions 2001-3000? |
| 21:43.39 | bjohnson | eg each company could have extension 1111 .. but to transfer from Company A to Company B the Company A user would append digits like 3331111 |
| 21:43.53 | bjohnson | the 333 would be stripped off before sending to Company B |
| 21:44.05 | scythelx | deploying a large scale asterisk server to multiple companies, but i dont want to have them have like extension 394234 |
| 21:44.06 | scythelx | heh |
| 21:44.13 | RDF | scy, i think for simplicity deviding them into two number groups would be easier. |
| 21:44.15 | _TripleF_ | apaceh defaul;t |
| 21:44.17 | bjohnson | Samoied: 5743278757 on ebay.ca |
| 21:44.29 | _TripleF_ | moc maybe |
| 21:44.41 | scythelx | ohh ok that makes sense |
| 21:44.45 | scythelx | pattern matching |
| 21:45.28 | bjohnson | Samoied: that happens to be 24 fxs .. you just have to keep an eye out to try to get one close to what you want |
| 21:45.45 | Samoied | bjohnson: I want 24 FXO |
| 21:46.06 | bjohnson | Samoied: that happens to be 24 fxs .. you just have to keep an eye out to try to get one close to what you want |
| 21:46.23 | Samoied | bjohnson: thanks |
| 21:46.52 | *** join/#asterisk Monitor (~ezgiiii@pD9E4990D.dip.t-dialin.net) |
| 21:46.53 | *** join/#asterisk [TripleF] (kvirc@HSE-QuebecCity-ppp3496808.sympatico.ca) |
| 21:47.24 | bjohnson | Samoied: for 24 incoming .. why don't you just get a T1 PRI plus one Digium T1 card and a single fxo for an extra analog line |
| 21:47.41 | [TripleF] | quick question anyone know a program to monitor windowz process to see if a program is rurnning and email if not / |
| 21:47.44 | Samoied | bjohnson: T1 in Brazil is a big problem |
| 21:47.54 | Samoied | bjohnson: We have E1 |
| 21:47.59 | bjohnson | Samoied: but you can get 24 analog? |
| 21:48.04 | Samoied | bjohnson: but is much expensive |
| 21:48.11 | Samoied | bjohnson: Yes :) |
| 21:48.51 | bjohnson | up here a PRI approx = cost of 23 analog lines |
| 21:49.08 | *** join/#asterisk BurnedOutGeek (~BurnedOut@216.215.202.4.nw.nuvox.net) |
| 21:49.30 | *** join/#asterisk jdemajo (~jdemajo@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 21:50.23 | blankman | Samoied, where in BZ are you? |
| 21:50.35 | blankman | North Fortaleza or South Rio? |
| 21:50.51 | blankman | Or West Brazilia? |
| 21:50.59 | Samoied | blankman: In Santa Catarina |
| 21:51.35 | blankman | So you have along pull back to a major POP huh ... |
| 21:52.05 | Samoied | blankman: You are in Brazil? |
| 21:52.14 | blankman | Nope ... just used to live there ;-) |
| 21:52.16 | blankman | Rio. |
| 21:52.31 | blankman | Thinking of "going home" some day soon :-) |
| 21:52.49 | Brixius[work] | hello |
| 21:52.51 | Samoied | blankman: You speak Portuguese? |
| 21:53.08 | RDF | btw, whats this about voip-info providing a link to a company that sells gsm/voip gateway boxes for cell phones? |
| 21:54.23 | RDF | mmm gone quiet in here. |
| 21:54.51 | blankman | Speak yes ... write not so good ... but then again write in english not so good either :p |
| 21:55.28 | Samoied | blankman: My english is very poor |
| 21:55.40 | Samoied | blankman: but I try... |
| 21:55.52 | bjohnson | RDF: I don't know |
| 21:55.55 | blankman | Samoied, :-) |
| 21:56.01 | *** join/#asterisk jdemajo (~jdemajo@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 21:56.22 | Brixius[work] | Question: If I have the following in a macro in my dialplan, exten => s,1,StripMSD(1), when the macro is done, will it continue on with the origional exten or will it now be missing a digit? |
| 21:56.56 | Samoied | Anyone recommend a standalone box with more than 4 FXO ports? |
| 21:57.09 | Samoied | like Sipura 3000 but with more than 4 FXo ports |
| 21:57.14 | Samoied | Anyone know? |
| 21:57.20 | *** join/#asterisk jdemajo (~jdemajo@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 21:57.40 | blankman | Samoied, buils a * server for that ... it's cheaper and easier to use. |
| 21:58.05 | Samoied | blankman: Yes but is much expensive for Brazilian guys :) |
| 21:58.07 | redder86 | Samoied: an Asterisk box with a TDM card |
| 21:58.15 | Samoied | blaI was using a Antek gateway |
| 21:58.31 | blankman | Samoied, why? the cards or the box? |
| 21:58.35 | Samoied | blankman: I was using Antek gateways |
| 21:58.43 | Samoied | blankman: the cards |
| 21:59.14 | Samoied | They say: "Is this? A computer with cards?" |
| 21:59.31 | Samoied | They preffer a box |
| 21:59.36 | Samoied | with ports in the front |
| 21:59.37 | slePP | why does faxing have to be such a pain with asterisk? :P |
| 21:59.52 | slePP | well, voip in general, anyway |
| 21:59.53 | RDF | slepp, its not a reliable transport? |
| 22:00.02 | *** join/#asterisk SexyKen (~sexyken@64-121-7-74.c3-0.snmt-ubr2.sfrn-snmt.ca.cable.rcn.com) |
| 22:00.04 | slePP | no, the transport is fine |
| 22:00.10 | slePP | no loss, 0.2ms jitter or so |
| 22:00.12 | slePP | it's the codecs :P |
| 22:00.25 | RDF | so the fax image looks like crap? |
| 22:00.36 | SexyKen | Hey guys, anyone know of a good toll free #asterisk provider? I signed up with NuFone over a month ago and my vanity #asterisk still isn't active. |
| 22:00.47 | ManxPower | RDF: VoIP/RTP is not "reliable transport" |
| 22:01.01 | ManxPower | SexyKen: NuFone is one of the best out there. |
| 22:01.22 | RDF | Manx, you mean the connectablity can fail right? |
| 22:01.30 | RDF | or the quality ;) |
| 22:01.36 | slePP | it means you can lose packets |
| 22:01.41 | ManxPower | RDF: No, I mean if a packet is lost it's not retransmitted. |
| 22:01.41 | mquin | you'd have though fax would have been less hassle to send over IP than voice |
| 22:01.42 | RDF | yea udp |
| 22:01.55 | blankman | Ummm ... so your are saying that if I sell you an embedded box, that it will be better than a box? |
| 22:01.56 | slePP | fax w/ t.38 is nice |
| 22:01.58 | mquin | seeing as it's basically data |
| 22:02.01 | slePP | but, of course, asterisk has nothing for t.38 |
| 22:02.05 | blankman | sorry that was for Samoied |
| 22:02.17 | ManxPower | mquin: No. Fax is basically REALLY weird voice. |
| 22:02.24 | slePP | fax is modulated audio bitstreams |
| 22:02.29 | Samoied | blankman: yes |
| 22:02.41 | slePP | which needs a nice, and fairly solid, audio tone |
| 22:02.50 | ManxPower | slePP: That's what I just said. 8-) |
| 22:02.52 | Samoied | blankman: a embedded asterisk with FX? ports in the front is more beautifull |
| 22:02.54 | SexyKen | •ManxPower• When you say best how do you mean? Surely their customer service isnt the best. And surely it isn't their turnaround time. I wouldn't know about their network as they haven't enabled my #asterisk yet. |
| 22:02.55 | slePP | yup :> |
| 22:03.04 | [TripleF] | http://www.activexperts.com/activmonitor/ that seems usefull |
| 22:03.19 | ManxPower | SexyKen: I'm saying that most VoIP providers are not any better. |
| 22:03.37 | ManxPower | SexyKen: Have you e-mailed support@nufone.net? |
| 22:05.33 | [TripleF] | ? |
| 22:05.33 | mquin | it's a modulated audio yes, but what's been moduated is image data. Speaking from a pureley theoritical point of view (I've no idea what's available in practice) it would make sense to demodulate it as soon as it leaves the analogue domain, shunt it over the network as data then remodulate when it needs to go back over analogue lines |
| 22:06.46 | ManxPower | mquin: Yes, it would make sense. |
| 22:07.20 | SexyKen | •ManxPower• I've been talking to Greg for over 3 weeks now...trying to get it taken care of. |
| 22:07.25 | RDF | manx, seen another example online again eheh how many of those do you have floating about? |
| 22:07.50 | ManxPower | RDF: Huh? |
| 22:08.01 | RDF | manx, dialplan examples |
| 22:08.08 | nestAr | feh |
| 22:08.22 | nestAr | damn you polycom phone, update my directory! |
| 22:08.37 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 22:09.27 | ManxPower | RDF: Where did you see it? |
| 22:09.38 | RDF | manx, doing google searches :) |
| 22:09.53 | blankman | Hey ManxPower, could you help me with why I can't get the spooler to work? |
| 22:11.11 | ManxPower | blankman: The help is already here. You just have to open your eyes. A good place to help you open your eyes is http://www.fnords.org/~eric/asterisk/downloads/abandoned-stuff/ Pay attention to the callback and vmoutcall scripts. |
| 22:11.26 | slePP | anyone know how to tell winfax to send at 9600 or 4800 instead of 14.4? |
| 22:11.31 | ManxPower | blankman: But you'd better be able to read perl |
| 22:11.47 | mishehu | slePP: its in the settings for the device |
| 22:11.50 | mishehu | in winfax |
| 22:12.48 | slePP | it gave me two options, 19200 and 2400 |
| 22:12.53 | slePP | set it to 2400, no difference, still trains to 14.4 |
| 22:14.13 | redder86 | slePP: why do you want to do it? |
| 22:14.34 | slePP | a client wants to keep using winfax over a pap2 |
| 22:15.00 | sgnome` | what needs to be done to allow for incoming and outgoing calls via FWD? |
| 22:15.38 | redder86 | slePP: hrmm... well, I could tell you how to do it in HylaFAX but I've had maybe 2 hours of experience with Winfax ... as I uninstalled it. |
| 22:16.17 | redder86 | slePP: I really wonder if going to 9600 would improve things for them over 14400, though. |
| 22:17.11 | Dr-Linux | whts diffence between voicemail and voicemailmain ? |
| 22:17.23 | Dr-Linux | difference ? |
| 22:17.33 | slePP | redder86: we can usually get it to go at 9600 or 4800, but very rarely at 14400 with the pap2 |
| 22:17.35 | ManxPower | Dr-Linux: "show application voicemail" "show application voicemailmain" |
| 22:17.39 | slePP | winfax is icky |
| 22:17.42 | redder86 | slePP: generally the problem with faxing over VoIP isn't so much an issue of faxing speeds as it is an issue of latency, packetloss, jitters, etc. between VoIP endpoints. |
| 22:17.54 | slePP | yeh |
| 22:18.05 | slePP | it's not jittering much, latency is 20ms, and packetloss is nil |
| 22:18.13 | redder86 | slePP: I use the SPA-2000 which is supposedly very similar to the PAP2. I fax using ULAW at 14.4 all the time. |
| 22:18.31 | redder86 | slePP: of course, I use HylaFAX, though. |
| 22:18.39 | slePP | Jan 13 16:48:31.96: [29835]: RECV: TCF 1893 bytes, 43% non-zero, 940 zero-run |
| 22:18.39 | slePP | Jan 13 16:48:31.96: [29835]: RECV: reject TCF (too many non-zero, max 10%) |
| 22:18.39 | slePP | Jan 13 16:48:31.96: [29835]: RECV: reject TCF (zero run too short, min 1200) |
| 22:18.44 | redder86 | slePP: and MultiTech MT5634ZBA-V92 modems |
| 22:19.13 | slePP | some of the people have no problems at all with 14.4 and a pap2, and some do. i don't get the difference between them that would cause issues... |
| 22:19.13 | redder86 | slePP: that's very bad |
| 22:19.39 | redder86 | TCF is supposed to be 100% zeros. 1.5 seconds of it. |
| 22:19.40 | slePP | however, on the last call, it was Jan 13 16:48:19.99: [29835]: RECV: TCF 1893 bytes, 3% non-zero, 1832 zero-run |
| 22:19.58 | redder86 | slePP: looks like a modulator incompatibility between sender and receiver. |
| 22:19.58 | slePP | and the call before that, 0% non-zero |
| 22:20.15 | redder86 | or the sender's line is just plain noisy |
| 22:20.25 | slePP | there's no noise. the sender is the pap2 end |
| 22:20.35 | slePP | 18" cable to the fax machine, to the pap2, into voip, out the PRI downtown |
| 22:20.42 | redder86 | ah, well VoIP generates it own set of noise |
| 22:20.52 | ManxPower | slePP: ulaw codec, I assume. |
| 22:20.59 | slePP | yeh. i wonder if alaw would be better than ulaw.. they do a bit of different compression in the end.. |
| 22:21.34 | RDF | slepp did he/she has fax problems before this? with a traditional phone line |
| 22:21.40 | slePP | no |
| 22:21.59 | redder86 | then the VoIP must be introducing all of the problems. |
| 22:22.17 | redder86 | all hail T.38 support |
| 22:22.23 | RDF | slepp, what is the fax machine pluged into a ata? |
| 22:22.27 | slePP | yeh, it is the voip. i know that :P |
| 22:22.28 | AgiNamu | Anyone interested in IAX2 FXS systems? G729,723,G711 and GSM. |
| 22:22.33 | slePP | RDF: pap2-na |
| 22:22.41 | RDF | pap2? |
| 22:22.50 | slePP | yeh |
| 22:22.51 | RDF | what is it physically pluged into |
| 22:22.56 | slePP | winfax |
| 22:23.04 | RDF | winfax modem? |
| 22:23.05 | redder86 | modem |
| 22:23.10 | RDF | k |
| 22:23.18 | ManxPower | RDF: With me showing up on Google so much you'd think someone would offer me a high paying job, huh? |
| 22:23.48 | RDF | manx, why not go out and introduce your self to companies and do consulting work. |
| 22:23.54 | redder86 | when did employers start looking for employees on google? |
| 22:24.01 | ManxPower | RDF: I currently do consulting work. |
| 22:24.23 | *** join/#asterisk SkyTel_DK_Down (~skytel@83.91.243.210) |
| 22:24.27 | RDF | manx, IT jobs in canada are overly saturated. Sucks as it can be but its true. Yea push your self and bend over backwards for them :) |
| 22:24.31 | Nugget | I always google for potential hires. |
| 22:24.34 | ManxPower | I get requests for info like once a week, but they are either cheap or want programming. |
| 22:24.40 | Nugget | before google I'd altavista for them. |
| 22:24.52 | sgnome` | anyone here use FWD? |
| 22:25.13 | slePP | t.38 is so nice to play with, but, alas, asterisk doesn't do pass through on it for some reason |
| 22:25.15 | redder86 | "IT" is a big field. General IT is saturated, but there are niche markets where, if you can find a demand, there will be nobody to compete with you. |
| 22:25.30 | RDF | Manx, goto the city hall and find out who is starting a new bizz. If its a call center mabey thay have not decided on a pbx yet? |
| 22:25.37 | ManxPower | redder86: I want someone in the Toronto area. |
| 22:25.58 | ManxPower | RDF: You're joking right? Startups are cheap, unstable, and poorly managed. |
| 22:26.26 | Nivex | Why, oh why, did I have to get stuck in a saturated field? |
| 22:26.30 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
| 22:26.33 | RDF | yea thats probebly true. |
| 22:26.44 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:26.57 | ta[i]nted | go for new office branches then |
| 22:27.04 | ManxPower | I participated in the start up of 3 ISPs. Never again. |
| 22:27.04 | RDF | Niven, because alot of .com had unproven ideas that bet on luck :) |
| 22:27.08 | ta[i]nted | Nivex: what field is that |
| 22:27.20 | Brixius[work] | I'd get the $$$ up front for any investment I had to make in equipment, so as to minimize the out of pocket when dealing with a startup. |
| 22:27.34 | *** join/#asterisk _TripleF_ (kvirc@Toronto-HSE-ppp3881368.sympatico.ca) |
| 22:27.39 | AgiNamu | redder: google is huge for potential hires. |
| 22:27.48 | ManxPower | I have a good consulting gig, but I want to move to Canda. |
| 22:28.00 | ta[i]nted | ManxPower: why |
| 22:28.08 | RDF | ohh manx, I thought you were back east? |
| 22:28.16 | ManxPower | ta[i]nted: You want the short or long answer to that? |
| 22:28.20 | ta[i]nted | long |
| 22:28.23 | ManxPower | RDF: I live near New Orleans |
| 22:28.38 | ta[i]nted | seafood gumbo!!! |
| 22:28.51 | ManxPower | ta[i]nted: Since the 1980's and the War on (Some) Drugs, the USA has been enacting law after law to infringe on civil liberties. |
| 22:29.07 | ta[i]nted | like ... |
| 22:29.08 | ManxPower | The short answer is: I don't want to live in a country that would elect George Bush for a SECOND term. |
| 22:29.17 | Brixius[work] | You don't want to move somewhere cold, it's supposed to be 22 below up here tomorrow(in mn that is)and that's without windchill. |
| 22:29.22 | AgiNamu | haha |
| 22:29.49 | Nivex | ta[i]nted: "IT" |
| 22:29.50 | ta[i]nted | let me give u a short reply to that: people will not switch presidents during wartime |
| 22:30.00 | ta[i]nted | it just doesn't happen |
| 22:30.15 | RDF | Dont come to BC for a job or consulting its saturated here. Also alot of people like me who immigrate here looking to work in the same hightech field are out of luck. Very highly qualified people immigrate here thinking it wont be a huge issue looking for simular work only to work in the low pay service sector. I met a east indian who had a chemistry degree with experaince selling vacumes. a cabbie driver who was a 727 pilot a arctitect |
| 22:30.18 | Nivex | ta[i]nted: Right now I am a "Desktop Support Specialist" (read Hell Desk), but I have aspirations to be a sysadmin, netadmin, perhaps even some VoIP work. |
| 22:30.20 | RDF | jobs are in the states. |
| 22:30.49 | ManxPower | ta[i]nted: The RICO laws, the post 9/11 laws. |
| 22:30.49 | Moc | it good to live in canada... |
| 22:30.49 | ManxPower | The "we can take your property without due process", the "we can hold you without being charged with a crime for as long as we want" laws, etc. |
| 22:30.49 | sgnome` | can someone help me configure FWD on my * box pls? |
| 22:30.53 | Moc | like those people being held in prison in Cuba by the US |
| 22:31.17 | fearnor | startups are tough |
| 22:31.18 | RDF | moc, its nice to live here but the death of taxes in bc and lower pay and lack of really good paying jobs make me wonder. |
| 22:31.22 | ManxPower | Moc: Canada isn't perfect, but I feel they are at least going in the right direction. |
| 22:31.22 | ManxPower | Moc: Yes, that too. |
| 22:31.25 | ta[i]nted | RDF: true that.. I used to live in Seattle and visited Vancouver often... there is no money for high tech there.. |
| 22:31.30 | *** part/#asterisk Samoied (~samoied@popeye.opens.com.br) |
| 22:31.37 | redder86 | those people in Cuba have US civil liberties because ... ? |
| 22:31.48 | Moc | ManxPower, ofcourse it aint perfect, but when I look at friend south... Im so happy to be here.. |
| 22:31.50 | ManxPower | I'd rather live in Benelux, but that's too big of a change for me. |
| 22:32.03 | AgiNamu | redder86, the people in cuba are still covered by the geneva convention |
| 22:32.07 | fearnor | red: because they are held by US authorities etc etc. |
| 22:32.11 | ta[i]nted | nothing beats Southern Hospitality (tm) |
| 22:32.11 | fearnor | gitmo. |
| 22:32.13 | ManxPower | The low crime rate in .ca is nice too. |
| 22:32.19 | AgiNamu | if you're referring to that u.s. base. |
| 22:32.28 | redder86 | AgiNamu: geneva liberties and US civil liberties are different |
| 22:32.31 | RDF | I feel that to strive in IT one has to move. move to another country if you have to. One woman in the states exasted all her interviews and excepted a job in ireland. So what the heck :) |
| 22:32.36 | ManxPower | AgiNamu: The US says they are NOT covered under Geneva Convention |
| 22:32.36 | AgiNamu | redder, sure. but usa respects neither. |
| 22:32.44 | Moc | I like my canada liberty ;) |
| 22:32.44 | AgiNamu | I know, the US is in such violation |
| 22:32.50 | RDF | manx, true. low crime rate...depends where you live ;) |
| 22:32.55 | mquin | the US government has ducked the geneva convention as they didn't actually declare war on afganistan |
| 22:32.59 | AgiNamu | it's a shame there's not an international body who will stand up to them... like the Federation. |
| 22:33.02 | ta[i]nted | there are bad areas in Canada |
| 22:33.05 | *** join/#asterisk dextro (~dextro@209.163.224.58) |
| 22:33.08 | ManxPower | RDF: Uh, pretty much any place in .ca has a low crime rate compated to the USA. |
| 22:33.10 | ta[i]nted | where they still club baby seals |
| 22:33.11 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 22:33.11 | AgiNamu | tainted, yea, where moose bite you. |
| 22:33.23 | mquin | so the people they captured aren't POWs |
| 22:33.27 | fearnor | if canada was in caribbean, i'd move there. |
| 22:33.30 | AgiNamu | and US civil liberties are being eroded too , with crap like the PATRIOT act |
| 22:33.32 | fearnor | heh |
| 22:33.46 | redder86 | AgiNamu: I'm not saying what the US does or doesn't do, but people in Cuba aren't afforded US civil liberties because they're not US citizens and because they're not on US soil. |
| 22:33.58 | AgiNamu | redder, I thought the base was us soil? |
| 22:34.00 | RDF | manx, canada per 1000 pop is low. Saw this on a web site that tracks capital murder in all countries. Can you figure out which country had the highest per capical murder rate per 1000 ? |
| 22:34.06 | AgiNamu | if not, why doesnt castro go kick em off :) |
| 22:34.09 | redder86 | AgiNamu: nope, it's still Cuba |
| 22:34.17 | ta[i]nted | fearnor: I have a question I've been dying to ask you.. which do you believe is the bigger scum, porn peddlers or spammers? |
| 22:34.19 | redder86 | AgiNamu: bay of pigs thing |
| 22:34.23 | AgiNamu | still , dodging the Geneva convention |
| 22:34.24 | fearnor | it is technically under cuban souvereignity but practically, it is US. |
| 22:34.29 | ManxPower | RDF: Per city/metro area is a much more accurate measurement. |
| 22:34.31 | dextro | anyone have experience with the VoiceMail call back macro? I am having an issue with the Hangup(h) section not being jumped to when a user hangs up after a VoiceMail is recorded |
| 22:34.32 | fearnor | ta: uhh..."porn peddlers"? |
| 22:34.38 | *** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com) |
| 22:34.48 | ta[i]nted | really? |
| 22:34.49 | fearnor | 80% of my customer base and 99% of bandwidth are used by "pr0n peddlers", so to speak. |
| 22:34.56 | RDF | columbia had the highest per capita murder rate :) must be all that cocaine atributing to it ;) |
| 22:35.02 | redder86 | AgiNamu: I'm not in a position to say whether or not they follow the Geneva convention or not. But Manx was talking about "US civil liberties" for "those people in Cuba". |
| 22:35.16 | fearnor | ta: i love them. i love pr0n too. |
| 22:35.22 | fearnor | when you get married, you'll know why! |
| 22:35.26 | AgiNamu | oh, well, why doesnt Manx first talk about US Civil liberties for us citizens in the usa first :) |
| 22:35.37 | RDF | :) |
| 22:35.49 | ManxPower | RDF: Some years New Orleans 1 in 1000 people are murdered. |
| 22:35.50 | redder86 | A lot also depends upon the state that you happen to live in. |
| 22:35.50 | ta[i]nted | why love porn when u could just have affairs |
| 22:36.04 | ta[i]nted | ;) |
| 22:36.04 | AgiNamu | haha |
| 22:36.05 | RDF | tainted, because i love my familly? |
| 22:36.12 | ta[i]nted | i was talking to fearnor |
| 22:36.17 | RDF | k |
| 22:36.29 | dextro | does the h extension ever get run out of a macro? |
| 22:36.35 | ManxPower | dextro: Well you can't Hangup(h) |
| 22:37.10 | Dr-Linux | hello |
| 22:37.14 | Dr-Linux | exten => 9999,1,VoicemailMain(${CALLERID}) |
| 22:37.20 | ManxPower | The USA was founded by Puritans. Enough said. |
| 22:37.28 | ManxPower | Dr-Linux: That will NEVER work. |
| 22:37.31 | *** part/#asterisk lele (~lele@rivendell.windmill.it) |
| 22:37.35 | ta[i]nted | ManxPower: what are you? |
| 22:37.38 | ManxPower | Try exten => 9999,1,VoicemailMain(${CALLERIDNAME}) |
| 22:37.40 | Dr-Linux | {CALLERID} << here i'll put user/number or it will be as it is ? |
| 22:37.49 | ManxPower | ta[i]nted: An ultra liberal |
| 22:38.03 | ta[i]nted | move to amsterdam |
| 22:38.06 | ManxPower | Dr-Linux: see README.variables in the Asterisk source. |
| 22:38.18 | dextro | ManxPower, ?, it works fine when the user is nice and pushes # to exit, but nobody is going to do that =] I thought h would catch the hangup but it doesnt happen.... |
| 22:38.24 | ManxPower | ta[i]nted: I would get on a plane tomorrow for Amsterdam if I had a job waiting and immigration taken care of. |
| 22:38.34 | Dr-Linux | exten => 9999,1,VoicemailMain(${1115}) << is it okey ? |
| 22:38.41 | ManxPower | Dr-Linux: no. |
| 22:38.48 | ManxPower | exten => 9999,1,VoicemailMain(1115) |
| 22:38.56 | Dr-Linux | oo okey |
| 22:39.08 | ta[i]nted | maybe you just need to get out of New Orleans |
| 22:39.10 | Dr-Linux | okey |
| 22:39.10 | Nugget | for a doctor you sure don't seem to have done much studying. |
| 22:39.10 | ManxPower | Dr-Linux: What if you want to check your voicemail from an extension that is not your extension? |
| 22:39.11 | RDF | manx, my wife is experessing the idea of living in arizona for a few months out of the year..she has cronic back problems that is affected by the weathr up here. |
| 22:39.24 | ta[i]nted | try going to LA or NY or SF |
| 22:39.33 | ta[i]nted | plenty of radical liberals |
| 22:39.34 | Dr-Linux | ManxPower: what is 9999 here in this ? |
| 22:39.35 | ManxPower | ta[i]nted: I lived in LA for 5 years. |
| 22:39.45 | ta[i]nted | how about SF |
| 22:39.53 | ManxPower | Dr-Linux: in your exmaple 9999 is what you dial to access voicemailmain |
| 22:39.53 | ta[i]nted | lots of digerati like yourself there |
| 22:39.59 | ManxPower | ta[i]nted: Tempting, but too cold in the summer. |
| 22:40.11 | RDF | ohh I hear the philipines is very cheap to live and some americans are going there to work. |
| 22:40.13 | ManxPower | I want a nice multicultural city with good public transic. |
| 22:40.20 | ta[i]nted | "too cold in the summer." <--- and your considering Canada? |
| 22:40.22 | ManxPower | And I like Toronto a lot. |
| 22:40.25 | Nugget | now we're back to amsterdam, then. :) |
| 22:40.31 | ManxPower | ta[i]nted: Canada isn't cold in the summer. |
| 22:40.41 | RDF | Bc is very very nice in the summer |
| 22:40.50 | ta[i]nted | maybe you'd like Seattle |
| 22:40.54 | ta[i]nted | a lot of Toronto |
| 22:41.03 | ManxPower | My family is in Michigan. |
| 22:41.10 | RDF | I need to get out of here ;) see ya ;) |
| 22:41.15 | hermie | ManxPower: where in MI? |
| 22:41.16 | ta[i]nted | alot like toronto |
| 22:41.19 | redder86 | Unless you like city you proably wouldn't want to live *in* Seattle. More like somewhere "around" Seattle. |
| 22:41.19 | ManxPower | If I can't live in Toronto or Benelux it's not worth moving out of the USA. |
| 22:41.26 | ManxPower | ta[i]nted: Holland Michigan. |
| 22:41.40 | Yoda-BZH`ZzZ | une bonne nuit je vous souhaite ! / A good night I wish you |
| 22:41.43 | hermie | ManxPower: cool I don't like to far from there... |
| 22:41.44 | ta[i]nted | redder86: yea but who knows Issaquah or Redmon |
| 22:41.51 | redder86 | tainted: I do |
| 22:41.56 | ta[i]nted | redder86: just giving him a reference point |
| 22:42.04 | ManxPower | With enough painkillers Toronto winter should not be too bad. |
| 22:42.10 | redder86 | most people here should know "Redmond" anyway, from that big giant that lives there. |
| 22:42.14 | redder86 | :-) |
| 22:42.28 | ta[i]nted | you mean Redmond Town Center? |
| 22:42.30 | ta[i]nted | lol |
| 22:42.49 | redder86 | yes |
| 22:42.54 | ta[i]nted | ok let's ask ManxPower |
| 22:42.58 | redder86 | okay |
| 22:43.06 | ta[i]nted | ManxPower: do you know Redmond, WA? |
| 22:43.31 | ManxPower | YES! My Sipura phone shipped! |
| 22:43.42 | niZon | anyone know why music on hold would just suddenly stop working? |
| 22:43.42 | ManxPower | ta[i]nted: I am not interested in moving back to the west coast. |
| 22:43.42 | redder86 | apparently where Sipuras come from |
| 22:43.57 | ta[i]nted | apparently |
| 22:43.59 | redder86 | ManxPower: it was a trivia question, not a suggestion |
| 22:44.04 | ManxPower | ta[i]nted: Redmond WA is the Seat of Evil in the workd. |
| 22:44.08 | ta[i]nted | doh |
| 22:44.12 | redder86 | see , see |
| 22:44.21 | ta[i]nted | Redmond Town Center isn't that evil |
| 22:44.29 | *** join/#asterisk cdcopier (~sdcsdc@rhhe10-250.2wcm.comporium.net) |
| 22:44.46 | ManxPower | I WOULD like to visit there sometime, but only to piss on the Microsoft sign. |
| 22:44.50 | _TripleF_ | wow |
| 22:44.51 | ta[i]nted | redder86: are you from Seattle? |
| 22:45.02 | redder86 | tainted: I was in Kent for a year, but now I'm in Shelton. |
| 22:45.05 | ta[i]nted | so much anger |
| 22:45.23 | redder86 | tainted: you? |
| 22:45.26 | ta[i]nted | Shelton.. isn't that where you strap on your bindings to go skiing? lol |
| 22:45.41 | ta[i]nted | s/skiing/snowboarding |
| 22:45.51 | ManxPower | ta[i]nted: much less so since I stopped supporting MS products. |
| 22:45.55 | AgiNamu | Why's Redmon the seat of evil? |
| 22:45.57 | redder86 | hehe, no, mostly an older logging community, now turning into an outpost for Olympia workers |
| 22:46.15 | redder86 | AgiNamu: google for Redmond, WA |
| 22:46.16 | ManxPower | AgiNamu: MS HQ is thee. |
| 22:46.32 | ta[i]nted | ManxPower: i've never bought a single seat of Microsoft software |
| 22:46.46 | ta[i]nted | although i did win Microsoft Bob at one point in my life |
| 22:46.58 | redder86 | I don't think there are any ski resorts in the Olympic mountains. |
| 22:47.00 | AgiNamu | Manx... so? I've been there |
| 22:47.04 | AgiNamu | a few times. even interviewed there. |
| 22:47.13 | AgiNamu | it's an amazing place. |
| 22:47.13 | redder86 | they're like 30 minutes from here |
| 22:47.24 | AgiNamu | almost as cool as something from a Studio Ghibli film :) |
| 22:47.25 | ta[i]nted | redder86: that's pretty damn close |
| 22:47.28 | redder86 | yeah |
| 22:47.40 | ta[i]nted | AgiNamu: are u talking about MS campus? |
| 22:47.50 | AgiNamu | yea |
| 22:48.11 | ta[i]nted | half of my high school friends work at MS |
| 22:48.17 | redder86 | we moved here because of the location. We liked being ~60 minutes from the Pacific Ocean, 5 minutes from the Sound, and 30 minutes from one of the best National Parks on the continent |
| 22:48.19 | ta[i]nted | it was 'the place to work' |
| 22:48.28 | samueltc | hi |
| 22:48.43 | AgiNamu | tainted, when did they join? |
| 22:48.59 | redder86 | AgiNamu: if you live up there and want a job. |
| 22:49.04 | redder86 | tainted: where are you at? |
| 22:49.17 | ta[i]nted | AgiNamu: at least 5 years ago now |
| 22:49.33 | ta[i]nted | they do really well |
| 22:49.45 | ta[i]nted | redder86: i love olympic national park |
| 22:49.47 | redder86 | MS generally is known to treat its employees very well. |
| 22:49.54 | ta[i]nted | ever go to the sulfur springs? |
| 22:49.59 | redder86 | no, not yet |
| 22:50.03 | ta[i]nted | amazing hike |
| 22:50.06 | AgiNamu | tainted, wow... your highschool had some top people |
| 22:50.19 | ManxPower | redder86: The class action lawsuit by contractors doesn't count, I assume. |
| 22:50.20 | cdcopier | I'm new to VoIP and need some help if someone has some time to spare. I would like to have my office (7-8 lines) switched over to VoIP and use asterisk. I have been told a few different thing. One of suggestion was to get a voip adapter (Sipura SPA-2000) and signup with boardvoice. Will I be able to use asterisk with this setup? |
| 22:50.20 | samueltc | I want ~8 fxo, what should I buy? channel bank? I want something that fit ina 2U case |
| 22:50.25 | ta[i]nted | also helps that we had a lot parents who work for MS |
| 22:50.42 | ta[i]nted | cdcopier: yes |
| 22:50.50 | ManxPower | cdcopier: There are a zillion ways you can do it. |
| 22:50.55 | *** join/#asterisk KleinJonp (~chatzilla@dsl-213-023-227-242.arcor-ip.net) |
| 22:50.57 | ta[i]nted | redder86: I'm in Los Angeles now |
| 22:51.09 | ManxPower | Remember that any VoIP outside of your local office depends on the internet. |
| 22:51.28 | cdcopier | what about incoming calls? |
| 22:51.35 | cdcopier | same? |
| 22:51.38 | ta[i]nted | yes |
| 22:52.16 | ManxPower | cdcopier: You would need a provider with local numbers, but you are still depending on your internet connection. |
| 22:52.35 | ManxPower | cdcopier: *I* use local phones lines going directly into Asterisk, then analog or SIP phones on the desks. |
| 22:52.38 | ta[i]nted | redder86: the thing I miss most about Puget Sound.. are the crabs |
| 22:52.48 | *** join/#asterisk Voicelynx (VoiceLynx@user-0cdv656.cable.mindspring.com) |
| 22:52.50 | redder86 | ManxPower: they treat contractors very poorly |
| 22:53.02 | redder86 | contractor != employee |
| 22:53.08 | ta[i]nted | I must have the cholestrol level of a 70 year old considering how much crab I've eaten |
| 22:53.13 | ManxPower | I simply could never work for Microsoft. |
| 22:53.24 | redder86 | me neither, but I'm just saying |
| 22:54.01 | ta[i]nted | ManxPower: what exactly does a radical liberal believe in? Anarchy? |
| 22:54.13 | cdcopier | ManxPower: what hardware do you use for that setup? I don't want to have pay for the phone lines tho. Why do you use them and not Inet? |
| 22:54.29 | Dr-Linux | ManxPower: how i can all extentions status with command ? |
| 22:54.46 | ManxPower | ta[i]nted: Where to start? I think all laws must expire after 10 years unless they are passed again. |
| 22:54.48 | redder86 | tainted: my wife grew up in Kent. She loves crab and I go out to catch it every week during the summer. |
| 22:55.08 | ManxPower | ta[i]nted: I think most victimless crimes should be abolished. |
| 22:55.23 | ta[i]nted | redder86: I envy you. |
| 22:55.27 | ManxPower | ta[i]nted: I think there should be constituional right to privacy. |
| 22:55.31 | ta[i]nted | All laws must expire after 10 years! wow |
| 22:55.48 | ManxPower | ta[i]nted: If people still want the law it can be passed again. |
| 22:56.00 | Dr-Linux | ManxPower: how i can check all extentions status with command ? |
| 22:56.27 | ManxPower | <PROTECTED> |
| 22:56.32 | redder86 | ManxPower: re-passing a bunch of laws to renew them every 10 years would require a lot of government overhead. |
| 22:56.42 | ManxPower | Dr-Linux: I don't have any idea or I would have answered you. |
| 22:56.56 | ManxPower | redder86: *nod* Prolly easier to just let most of them expire. |
| 22:57.21 | cdcopier | ManxPower: what hardware do you use for that setup? I don't want to have pay for the phone lines tho. Why do you use them and not Inet? |
| 22:57.24 | ta[i]nted | would you eliminate precedence too then? |
| 22:57.36 | ManxPower | cdcopier: You'll ALWAYS have to pay for phone lines. |
| 22:57.40 | redder86 | ManxPower: I think that you'd find, though, that such an arrangement would just make big government moreso than less laws. Sure some of the crusty laws would disappear, but I think in the end it would just make for bigger government. |
| 22:57.56 | ManxPower | cdcopier: I don't rely on VoIP because I don't trust it to be reliable enough for me. |
| 22:58.10 | ta[i]nted | cdcopier: because he probably does least cost routing and wants redundancy in case VoIP poops out |
| 22:58.17 | ManxPower | redder86: MAYBE. |
| 22:58.31 | ManxPower | I also think that most drugs should be legalized and taxed. |
| 22:58.42 | ManxPower | I also believe that states do not have enough rights. |
| 22:58.56 | ta[i]nted | wow that is pretty radical |
| 22:58.59 | ta[i]nted | do you have a blog |
| 22:59.35 | ManxPower | ta[i]nted: Yeah. I also think 1/2 of the tax money collected from drugs should be allocated for free addiction treatment. |
| 22:59.49 | ManxPower | ta[i]nted: No, I just rant on IRC occasionally. *grin* But you DID ask. 8-) |
| 22:59.56 | ta[i]nted | how about the resulting increase in crime? |
| 23:00.09 | ManxPower | ta[i]nted: you mean decrease in drime. |
| 23:00.32 | ta[i]nted | yea you should move to Canada |
| 23:00.37 | ManxPower | LOL! |
| 23:00.41 | ta[i]nted | that stuff ain't gonna fly here |
| 23:00.50 | ta[i]nted | US is too religious |
| 23:01.25 | *** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk) |
| 23:01.40 | DrPete | tzanger: You there? |
| 23:01.51 | *** join/#asterisk wolfson (~hehe@bcp-68-187-182-173.man.nc.charter.com) |
| 23:01.59 | redder86 | too religious? |
| 23:02.13 | Nugget | the US is a big place. Almost too big to be "too" anything. |
| 23:02.14 | redder86 | everyone has a philosophy |
| 23:02.42 | ManxPower | ta[i]nted: The whole banning of gay marriage pisses me off too. |
| 23:02.44 | redder86 | If anything we should be governed by our own philosophy. Call that religion, I guess. |
| 23:03.02 | cdcopier | ManxPower: So, can I can get away with only using Sipura SPA-2000, broadvoice, and asterisk (Only use phonelines if this is not a critical situation)? |
| 23:03.04 | ManxPower | ta[i]nted: I also think that all govt involvement in "marriage" should be abolished as well. |
| 23:03.08 | derrick_ | i want the goverment to supply my weed |
| 23:03.19 | derrick_ | and i want nice sticky stuff, not damn mx brick |
| 23:03.26 | ta[i]nted | we should just abolish marriage |
| 23:03.31 | ManxPower | derrick_: Why? If it was legal you could just drow it in your garden. |
| 23:03.39 | ManxPower | GROW, that is. |
| 23:03.40 | AgiNamu | marriage is not a government institution, so i dont see why people care. |
| 23:03.40 | derrick_ | manx, i'm lazy :) |
| 23:03.58 | AgiNamu | I refused to get legally married for quite a while |
| 23:04.02 | ManxPower | AgiNamu: marriage SHOULD not be a government institution, but it is. |
| 23:04.07 | AgiNamu | went to the church, that's it. |
| 23:04.10 | Dr-Linux | 1115 can call others id, but if someone dial 1115 they get error Call failed: 404 Not Found ? |
| 23:04.10 | AgiNamu | fuck what the government says. |
| 23:04.13 | ManxPower | I was married once at one time. |
| 23:04.18 | Dr-Linux | where is problem |
| 23:04.18 | redder86 | goverment is involved in marriage for tax reasons |
| 23:04.27 | AgiNamu | oh that's it you see -- i dont pay tax either. |
| 23:04.42 | ManxPower | redder86: It could just as easily be done by adapting current corporate law. |
| 23:04.43 | ta[i]nted | redder86: not really.. they earn more in taxes from single people than married couples |
| 23:05.04 | redder86 | ManxPower: yeah, if you like what corporate law has done for the US |
| 23:05.14 | ManxPower | cdcopier: Yes, but *I* would be VERY unhappy with that. |
| 23:05.14 | derrick_ | and as a single person i have to pay taxes for all the breeders and their critters |
| 23:05.17 | ManxPower | redder86: Have you ever looked into LLPs? |
| 23:05.27 | ta[i]nted | LOL |
| 23:05.33 | ta[i]nted | ManxPower: you want marriage == LLP;? |
| 23:05.36 | ta[i]nted | hahaha |
| 23:05.36 | Dr-Linux | ManxPower: |
| 23:05.39 | Dr-Linux | 1115 can call others id, but if someone dial 1115 they get error Call failed: 404 Not Found ? |
| 23:05.42 | tzanger | derrick_: and it's us breeders whose offspring will be paying for your retirement |
| 23:05.44 | Dr-Linux | where is problem ? |
| 23:05.45 | redder86 | tainted: yes, they give tax breaks to married people because more voters are married and shifted the tax burden to others |
| 23:05.46 | ManxPower | cdcopier: If you do not build and test a prototype system you will fail.. |
| 23:05.50 | Nugget | yeah, I pay a fortune in taxes so the local high school can have a kickass football team. such a waste. |
| 23:05.55 | ManxPower | ta[i]nted: Why not? Or at least an LLP type thing. |
| 23:05.56 | AgiNamu | derrick_ , governemtns should pay people with high iqs to reproduce, and pay to sterilize low IQ people. |
| 23:05.57 | derrick_ | tzanger , doubt that |
| 23:06.03 | Nugget | doesn't do me a bit of good. |
| 23:06.17 | Nugget | well, aside from keeping idle teenagers from vandalizing my house, I guess. |
| 23:06.24 | redder86 | LLP's suck - anything with "limited liability" in it sucks. People should be accountable for their mistakes. |
| 23:06.27 | ta[i]nted | Nugget: there you go |
| 23:06.40 | Nugget | there are cheaper ways to do that! :) |
| 23:06.45 | tzanger | redder86: if the lawsuit industry didn't exist, I'd agree with you |
| 23:06.45 | ta[i]nted | redder86: Personally accountable for their business decisions? |
| 23:07.00 | redder86 | AgiNamu: in WA if you're low-income or otherwise "poverty" stricken you can be sterilized by free by the state |
| 23:07.02 | ManxPower | I'm not a corporate lawyer, but as I understand it LLPs do not get taxed, only the members of the LLP get taxed. |
| 23:07.03 | cdcopier | ManxPower: I am...I am trying to figure out what I need for a test and then figure out where I would like to go from there. Can you give me a recommendation on hardware/setup? You can PM me if you want. |
| 23:07.14 | AgiNamu | redder, but there's no *incentive* |
| 23:07.21 | cdcopier | ManxPower, ta[i]nted: thanks 4 help |
| 23:07.27 | Dr-Linux | ManxPower: u are not helping :( |
| 23:07.27 | ManxPower | cdcopier: Yes, what you said will work. |
| 23:07.28 | redder86 | tainted: yup, if people embezzle they shouldn't be able close the LLP and be done with it |
| 23:07.34 | AgiNamu | likewise, high IQ people should have to pay a fine for getting sterilization |
| 23:07.42 | tzanger | redder86: that's different. I'm all for the CDP |
| 23:07.50 | redder86 | AgiNamu: the incentive is no need for birth control |
| 23:07.53 | ta[i]nted | cdcopier: what costs are you trying to save. VoIP is not the answer to everything |
| 23:08.01 | AgiNamu | redder, meh |
| 23:08.06 | Nugget | eep. I'll need to get mine done before AgiNamu gets elected king. |
| 23:08.17 | cdcopier | longdistance, (7-8) phone lines |
| 23:08.20 | AgiNamu | No! Stand up! Save the world! |
| 23:08.20 | fearnor | corporations are engines of progress. |
| 23:08.24 | cdcopier | and 800 # |
| 23:08.26 | ta[i]nted | redder86: ok I agree with that.. too many fly-by-night operations |
| 23:08.35 | ta[i]nted | cdcopier: long distance to where? |
| 23:08.36 | AgiNamu | the world is being overrun with fucking retards cause they keep knocking kids out |
| 23:08.44 | ta[i]nted | cdcopier: US, Europe, Asia? |
| 23:08.47 | cdcopier | in and out of state...no INT. |
| 23:08.49 | AgiNamu | Whereas smarter people choose "a differnet lifestyle" and have maybe one kid |
| 23:09.03 | tzanger | AgiNamu: I have three, does that mean I'm stupid? |
| 23:09.05 | ManxPower | AgiNamu: or none |
| 23:09.14 | AgiNamu | tzanger, 3 is nothing |
| 23:09.21 | ManxPower | tzanger: Yes. You didn't stop after 1. *tease* |
| 23:09.23 | AgiNamu | if you have an IQ above 120, go have 8 kids |
| 23:09.30 | tzanger | I'd love to have a dozen kids but I can't financially support it |
| 23:09.34 | fearnor | thing is |
| 23:09.37 | fearnor | smart people don't have many kids. |
| 23:09.38 | AgiNamu | Thats why the government should help |
| 23:09.53 | ta[i]nted | cdcopier: do you currently have a PBX or just regular phones |
| 23:09.59 | AgiNamu | the guy from fairchild semiconductor came up with this research a long time ago |
| 23:10.02 | cdcopier | reg. phones |
| 23:10.04 | DrPete | tzanger: I dont suppose you have any info on the method you use to answer faxes you mentioned earlier? |
| 23:10.04 | AgiNamu | shockley |
| 23:10.08 | cdcopier | just getting into VoIP |
| 23:10.11 | tzanger | DrPete: of course I do |
| 23:10.17 | fearnor | how about instead of letting smart people have more kids |
| 23:10.19 | *** join/#asterisk DevilFish (DevilFish@216.163.40.2) |
| 23:10.21 | fearnor | government-sponsored pussy |
| 23:10.53 | ta[i]nted | cdcopier: just setup an asterisk server with broadvoice and purchase a few FXS cards for your local lines |
| 23:11.09 | cdcopier | so keep the lines? |
| 23:11.12 | tzanger | <PROTECTED> |
| 23:11.12 | tzanger | After being reported missing a day or two later, rangers found two bears, |
| 23:11.12 | tzanger | one a male, one a female, looking suspiciously overstuffed. They killed |
| 23:11.12 | tzanger | the female, autopsied her, and sure enough, found the German and the Pole. |
| 23:11.12 | tzanger | <PROTECTED> |
| 23:11.15 | tzanger | <PROTECTED> |
| 23:11.17 | tzanger | that's terrible. hahaha |
| 23:11.18 | ta[i]nted | i mean for your local phones |
| 23:11.32 | ta[i]nted | cancel your local lines if you want |
| 23:11.35 | ta[i]nted | or keep one/two |
| 23:11.38 | lters | only children can be taken to heaven |
| 23:12.04 | ManxPower | "Every time a child cries god kills a puppy" |
| 23:12.07 | ta[i]nted | cdcopier: it's not a difficult setup.. you can do it in a week |
| 23:12.41 | lters | children are an heritage |
| 23:12.45 | DrPete | tzanger: I was thinking of using VOIP as well, so I can use the single line for faxes and voice with faxes being filtered out. Can you point me in the right direction? I assume I need one of these http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItem&category=40995&item=5743029165&rd=1 |
| 23:12.56 | fearnor | manx: nah |
| 23:13.05 | AgiNamu | yea, that "everytime you masterbate, God kills a kitten" always bothered me. |
| 23:13.18 | ManxPower | I like children as long as they stay at least 300 meters from me. |
| 23:13.18 | fearnor | agi: "every time a woman masturbates, god creates a puppy" |
| 23:13.22 | lters | Happy is the man that hath his quiver full of them |
| 23:13.26 | fearnor | "so please, think of the puppies" |
| 23:13.31 | *** part/#asterisk stuart_ (~schneis@70.57.124.226) |
| 23:13.39 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 23:13.51 | redder86 | Do Polycom IP500s not have an HTTP interface? |
| 23:14.14 | pointer-gaim | they do |
| 23:14.16 | ta[i]nted | #asterisk - raw and unadulterated |
| 23:14.28 | tzanger | yes you'll need one of those, but don't get the X100P/X101P if you're in the UK, get a TDM401 or 411P, they work with UK impedances IIRC |
| 23:14.44 | tzanger | basically an asterisk dev kit lite will get you what you want |
| 23:14.53 | tzanger | fax over IP doesn't work very well, just be forewarned |
| 23:14.59 | redder86 | pointer-gaim: hrmm. this one doesn't respond |
| 23:15.01 | DevilFish | redder86: you gotta turn on the web interface n the cfg files |
| 23:15.12 | tzanger | it's impossible ot guarantee the strict timing you need for data transmission when you're spewing out UDP packets in a hostile environment |
| 23:15.21 | ManxPower | redder86: It takes like 3 min for the web interface to come up after the phone boots |
| 23:15.29 | DevilFish | yup that too |
| 23:15.29 | mquin | tzanger: theres an impedance problem with the x100 on uk lines? |
| 23:15.30 | redder86 | I'll wait. |
| 23:15.34 | DevilFish | they suck that way |
| 23:15.49 | tzanger | personally I think Digium should be suing the people who are marketing their knockoff cards as X100P and not X100P-compatible |
| 23:15.50 | ta[i]nted | tzanger: fax works great if you're willing to retry and retry and retry... and retry |
| 23:15.59 | *** join/#asterisk sgnome` (~s_gnome65@ip70-176-146-245.ph.ph.cox.net) |
| 23:16.32 | tzanger | mquin: IIRC the X100P's hybrid works for north american line impedance. UK is different so you have a much harder time eliminating echo |
| 23:16.35 | DrPete | Ahh, I see oki. |
| 23:16.45 | redder86 | DevilFish: how do I turn it on? |
| 23:17.16 | DrPete | Anyone know of a UK suppiler of cards and voip phones etc? |
| 23:17.22 | mquin | tzanger: ah |
| 23:17.31 | sgnome` | anyone know how to configure * for outgoing FWD calls? |
| 23:17.35 | sgnome` | i have incoming FWD working |
| 23:17.47 | DevilFish | can't remember but I had to do it in one of the cfg files ipmid.cfg I think |
| 23:18.05 | DevilFish | had to down load the admin guidw from Polycom and read the whole thing |
| 23:18.12 | DevilFish | guide even |
| 23:19.08 | redder86 | well it is in ipmid.cfg and it is enabled there, only it's not doing anything. |
| 23:19.15 | *** join/#asterisk tainted- (~tainted@65-60-70-242-cust.telepacific.net) |
| 23:20.11 | mquin | sgnome`: take a look at the advanced->iax page on the FWD website - it details how to set up asterisk to route calls to and from FWD |
| 23:20.21 | sgnome` | k |
| 23:20.25 | DevilFish | ip of phone in webbrowser yields nothing? |
| 23:20.40 | redder86 | DevilFish: nope |
| 23:20.42 | *** join/#asterisk sudoer2 (~toy@h000d56c772cb.ne.client2.attbi.com) |
| 23:20.47 | tzanger | DrPete: http://www.mixdown.ca/~andrew/dump/rxfax-example.txt |
| 23:20.48 | sgnome` | oh .. that's the page i used to setup incoming .. but following their directions didn't make outgoing work |
| 23:21.19 | sgnome` | does the 393 extension stuff need to be added to my from-sip context or to another one? |
| 23:22.04 | redder86 | it doesn't even create an app.log on boot like the IP600 does |
| 23:22.08 | DevilFish | redder86: Jeeze I'm not sure then |
| 23:23.13 | redder86 | oh wait, it just came up |
| 23:23.24 | redder86 | that took forever an a half |
| 23:23.32 | DevilFish | yeah they suck that way :) |
| 23:23.54 | ctooley | Anyone in the Dallas Area do general Linux consulting? |
| 23:24.05 | DevilFish | web gui on polycoms suck anyhow since they are going to get options at boot time from ipmid.cfg |
| 23:24.08 | DrPete | tzanger: I assume that is a script to rec the faxes |
| 23:24.15 | DevilFish | and other such cfg files |
| 23:24.23 | tzanger | DrPete: that is the dialplan and the script to do so, yes |
| 23:24.30 | tzanger | if by rec you mean "receive" |
| 23:24.47 | ctooley | DevilFish, The web ui isn't able to set user level settings anyway |
| 23:24.55 | DrPete | tzanger: yeah, hehe I see. When you say fax of IP is not that great what do you mean |
| 23:24.55 | redder86 | DevilFish: yes, but at least I can go in and see that it loaded the settings properly from the config files |
| 23:25.11 | DevilFish | I suppose so |
| 23:28.28 | ctooley | Not only that, but you can get a phone to register without having to go thorugh the ftp/tftp process for testing just one phone. |
| 23:29.20 | redder86 | in this case it was helpful in showing me where I screwed up on the cfg files |
| 23:29.27 | tzanger | DrPete: are you intending on receiving faxes that come in through the FXO (i.e. a local phone line) or are you going to try and receive faxes from a DID you're leasing from a VOIP provider? |
| 23:30.13 | *** join/#asterisk hengwee (~chinhw@cm232.gamma209.maxonline.com.sg) |
| 23:30.33 | DrPete | tzanger: I was thinking of just though the phone line |
| 23:30.37 | redder86 | yay! it's registering now |
| 23:30.58 | redder86 | and it took the firmware update |
| 23:30.59 | ctooley | redder86, which firmware are you running? |
| 23:31.02 | redder86 | 1.3.4 |
| 23:31.07 | redder86 | thanks to DevilFish |
| 23:31.11 | ctooley | ah, and which Asterisk checkout? |
| 23:31.33 | redder86 | CVS stable from the 5th of this month |
| 23:31.39 | ctooley | ok |
| 23:31.42 | tzanger | DrPete: that works perfectly |
| 23:31.47 | DrPete | Basically, I have one phone line. I want to use it for reciving faxes when they arrive, but letting voice though. I want to use a linux server and I like the idea of VOIP |
| 23:31.49 | tzanger | well, as perfect as fax is at any rate :-) |
| 23:31.55 | redder86 | funny that it registered but still no HTTP server |
| 23:32.14 | DrPete | tzanger: well that sounds good to me |
| 23:32.16 | ctooley | redder86, did you turn off the HTTP server in the config? |
| 23:32.35 | tzanger | DrPete: if you want to dump all your callers into an IVR you can use zapata's automatic fax detection |
| 23:33.00 | tzanger | basically while you're saying "welcome to my IVR" if * hears a fax tone it will jump to the fax extension and there you will call the receive-fax macro |
| 23:33.02 | redder86 | ctooley: no, it just takes a long time to come up |
| 23:33.06 | tzanger | you'll never hear a fax tone on your line again |
| 23:33.07 | redder86 | ctooley: it's up now |
| 23:33.30 | ctooley | redder86, cool |
| 23:33.49 | ctooley | well, I had problems with Polycom's and CVS HEAD yesterday, so be forewarned |
| 23:34.25 | DrPete | tzanger: I like the sound of that, and this is all though asterisk, it seems to be very impressive |
| 23:34.46 | tzanger | DrPete: asterisk is very impressive |
| 23:35.50 | redder86 | ctooley: well, I don't use HEAD, I'm not that masochistic |
| 23:35.57 | DrPete | tzanger: when you say it will jump to the fax extension, do you mean it can answer and say drop the fax in a samba share or mail it to me or something, as i dont want to use a machine |
| 23:36.36 | tzanger | DrPete: yes, app_rxfax receives a fax electronically, no fax machine needed. Similarly app_txfax sends a .tiff file to a fax machine without you needing a fax machine yourself |
| 23:37.15 | DrPete | tzanger: ahh, so I could use a windows client to send via the server? |
| 23:38.09 | redder86 | DrPete: you'd have to construct one |
| 23:38.16 | tzanger | yes basically something that dumps to pdf or tiff and then some script magic to look for new tiffs and figure out where to send them |
| 23:39.02 | bjohnson | tzanger: while on the topic, is receiving via a voip provider different? |
| 23:39.22 | redder86 | bjohnson: yes |
| 23:39.26 | tzanger | bjohnson: yes much |
| 23:39.27 | DrPete | tzanger: man, u guys need to get on and code that :) |
| 23:39.49 | tzanger | DrPete: my paypal account is akohlsmith-pp@mixdown.ca, feel free to pay the US$200 bounty |
| 23:40.04 | redder86 | bjohnson: there will be a lot more packet loss, latency, jitters, etc with VoIP |
| 23:40.30 | tzanger | it has everything to do with jitter -- latency can more or less be accounted for and most VOIP connections don't drop packets |
| 23:41.07 | DrPete | tzanger: well said :) |
| 23:41.42 | DrPete | if VOIP quality is anything like skype, i will be impressed |
| 23:41.55 | tzanger | skype *is* VOIP |
| 23:42.20 | *** join/#asterisk RoyKa (~roy@110.80-203-29.nextgentel.com) |
| 23:42.20 | *** join/#asterisk Miker (~mike@201.129.119.248) |
| 23:42.29 | DrPete | tzanger: hehe, i what i mean, is if I setup asterisk and its as good I will be happy :) |
| 23:42.54 | tzanger | ahh |
| 23:42.55 | *** join/#asterisk macmx (~mike@201.129.119.248) |
| 23:43.23 | RoyKa | bkw_, here? |
| 23:43.35 | DrPete | tzanger: If if its a pain to use the server to send faxes, could i have the winbox act as a voip phone, and just send it from the box, if you see what I mean |
| 23:43.37 | RoyKa | kram, ? |
| 23:43.53 | tzanger | it's more work than skype and there are several key differences but with a GOOD VOIP provider (I use nufone exclusively) and a good network connection, it's excellent, I consider it indistinguishable from POTS lines for quality |
| 23:44.39 | tzanger | DrPete: you can't fax over a voip phone |
| 23:44.48 | *** join/#asterisk jcims (~chatzilla@cpe-69-135-121-57.columbus.rr.com) |
| 23:44.52 | *** join/#asterisk planetWayne (~wayne@cpc1-lich2-5-0-cust62.brhm.cable.ntl.com) |
| 23:44.53 | tzanger | you can, though, use hylafax and a faxmodem to achive very good results |
| 23:45.00 | planetWayne | hello all. |
| 23:45.34 | jcims | hello |
| 23:45.54 | tzanger | ok gone to wrestle wit the kids for a bit, be back in about an hour |
| 23:45.59 | DrPete | tzanger: oooh, i like that, i have a very nice haynes modem, i would like to use |
| 23:47.04 | DrPete | tzanger: could i just plug that in the windows box and have that connected to the same phone line? Could it conflict with asterisk? |
| 23:47.14 | RoyKa | FUCK. someone gave me Karma -2 for reporting a fecking bug: |
| 23:47.32 | RoyKa | http://bugs.digium.com/bug_view_page.php?bug_id=0003331 |
| 23:47.46 | planetWayne | if you DIAL() multiple SIP channels on an outbound call - how does * handle this? does it dial all of the avaliable lines or will it progressivly go through each untill it finds a free line? (hope that makes sence) |
| 23:49.45 | jero | if it's a group, it will pickup the first available line |
| 23:49.56 | jero | sorry |
| 23:51.21 | bkw_ | yes |
| 23:51.24 | bkw_ | RoyKa i'm here |
| 23:51.32 | bkw_ | you do realize what you reported isn't a bug |
| 23:51.41 | derrick_ | palnetwayne, if you dial(sip/bob&iax2/henry) it'll dial both at the same time, whoever answers gets the call |
| 23:51.59 | slePP | is there a way to set restrictcid on a per-call basis? |
| 23:51.59 | bkw_ | you start any process that prints to stdout and close the terminal it will slowly eat itself because the fd it was writing to is blocking |
| 23:52.00 | *** join/#asterisk hardocp (~hardocp@ool-44c1f7b4.dyn.optonline.net) |
| 23:52.11 | *** join/#asterisk ccfiel (~chatzilla@210.213.141.130) |
| 23:52.19 | hardocp | can someone please help me setup call groups with iax trunks? |
| 23:52.25 | sung | bkw! |
| 23:52.49 | ctooley | G.729a is the codec that requires a license from Digium, correct? |
| 23:53.29 | RoyKa | bkw_, Mark's giving me a -2 for fecking reporting a bug |
| 23:53.34 | bkw_ | yes |
| 23:53.36 | bkw_ | because its not a bug |
| 23:53.39 | RoyKa | bkw_, and that's a BAD THING |
| 23:53.40 | bkw_ | in any sense of the word |
| 23:53.44 | RoyKa | bkw_, it is |
| 23:53.48 | bkw_ | no you made the decision to start asterisk in that manner |
| 23:53.58 | bkw_ | try starting apache int he same way with it outputing stuff to the console |
| 23:54.01 | bkw_ | then close the thing |
| 23:54.03 | RoyKa | bkw_, WHATEVER |
| 23:54.05 | bkw_ | and watch it do the exact same thing |
| 23:54.12 | bkw_ | it is not a bug |
| 23:54.17 | RoyKa | bkw_, it is |
| 23:54.21 | bkw_ | the FD gets closed that we write to because you DO NOT START ASTERISK LIKE THAT |
| 23:54.52 | planetWayne | derrick_ - ah yeah - sorry - my bad I just thought that was the same as how I am doing inbound... I ment to say Zap channels - eg. 4 analogue lines (zap/1 -> zap/4) if you do a dial(zap/1/{exten}&zap/2/{exten}& {etc}) will * attempt to pick up all the lines and dial the number? |
| 23:54.56 | RoyKa | bkw_, and whatever, the other bugs I've filed have been pretty fscking well documented, but do I get a + for that? |
| 23:55.00 | jcims | sounds like more of a 'gotcha' to me :P |
| 23:55.50 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 23:56.11 | AgiNamu | royka, what's up with the "whatever"? |
| 23:56.25 | redder86 | yet another reason people are afraid to submit bug reports: karma |
| 23:56.32 | RoyKa | bkw_, btw, I've been trying to fax Mark some stuff, but he's prolly using some asterisk HEAD, I only end up with an error |
| 23:56.38 | AgiNamu | when faced that you are wrong, then just admit it and apologize. a bit of humility goes a long way. |
| 23:56.42 | RoyKa | AgiNamu, read the bug |
| 23:57.10 | redder86 | if you make so many hurdles for a user/administrator to report something that they find is behaving incorrectly then you lose opportunity |
| 23:57.31 | redder86 | because you scare them away with stuff like "negative karma" |
| 23:57.46 | redder86 | such an ethereal and misty thing as that |
| 23:57.54 | *** join/#asterisk roger__ (~roger@smack.isp.net.au) |
| 23:58.02 | redder86 | *karma* of all the things to scare people with, *karma* |
| 23:58.22 | AgiNamu | RoyKa, I read the damn bug. when you act like a prick with mark, you should get even more |
| 23:58.28 | redder86 | it's like throwing air at someone |
| 23:58.33 | AgiNamu | the box might crash if you yank you the TDM card in the middle of a call. |
| 23:58.33 | roger__ | hi guys. i'm running cvs head and parking a call seems to segfault * |
| 23:58.55 | AgiNamu | redder86, i dont think it's supposed to be a threat :) |
| 23:59.10 | redder86 | AgiNamu: then there should be no such thing as "negative" karma |
| 23:59.26 | AgiNamu | sure there should. it's not a threat, its for others to know how serious to take you |
| 23:59.38 | AgiNamu | if some guy with a +10000 karma comes on and says he found something, you tend to know he knows what;'s going on |
| 23:59.39 | redder86 | "negative" karma implies "hey we're not going to pay attention to you in the future" |
| 23:59.56 | AgiNamu | companies deal with this by having employees versus "some random dong running our sofware" :) |