00:01.42 | Ariel_ | sivana, it's been up and down all day |
00:02.00 | SarahEmm | for days... |
00:04.03 | doughecka | drink water |
00:04.07 | doughecka | its better than coffee |
00:04.17 | fugitivo | try decafeinated |
00:04.31 | doughecka | try unleaded |
00:04.47 | fugitivo | drink tea |
00:04.57 | doughecka | drink ocean water |
00:04.59 | Robot_ | i agree with tea |
00:05.00 | fugitivo | or mate |
00:05.02 | Ariel_ | not the same coffee is well coffee.. yummmm |
00:05.16 | fugitivo | did you try mate? |
00:05.20 | hardwire | http://www.redcoat.net/pics/ponchomo.jpg |
00:05.42 | doughecka | hahah |
00:05.52 | doughecka | was that taken at the astricon? |
00:05.55 | Ariel_ | ponch.... |
00:08.08 | *** join/#asterisk Nukemizer (~Nuke@67.137.28.165) |
00:15.15 | fugitivo | http://www.thinkgeek.com/caffeine/drinks/6147/ |
00:15.20 | *** join/#asterisk MagicFab (~chatzilla@modemcable094.36-203-24.mc.videotron.ca) |
00:15.32 | MagicFab | Hello |
00:15.42 | fugitivo | much better than coffee |
00:15.55 | MagicFab | Wondering if there a french / quebec asterisk user group(s) ? |
00:17.09 | NewSole | damd french... they always want to be seperest.... |
00:17.37 | NewSole | they want to seperate from canada and now they want thier own groups |
00:17.57 | fugitivo | lol |
00:18.21 | MagicFab | Any other smart people around ? :P |
00:18.38 | NewSole | asterisk is a US Product from califonia.... you want to seperate from that... microsoft offers Live server |
00:18.46 | fugitivo | not me, sorry :) |
00:20.05 | NewSole | sorry... I FELL NO PITTY for Quebeckers.... The Demand you speek french and english in Ontario.... but they make it illegal to put an english sign up in quebec |
00:20.34 | MagicFab | Of course everyone should speak only english and be as ignorant as NS - get a life |
00:20.40 | *** part/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca) |
00:20.47 | MagicFab | the law changed long ago BTW |
00:20.52 | MagicFab | :D :D |
00:20.53 | NewSole | no I belive fair is fair... |
00:21.19 | NewSole | no... it changed that you have to have french first then english... |
00:21.19 | sivana | NS = nova scotia? :) |
00:21.29 | fugitivo | i agree with quebec goverment, some languages are dying (ex: french) |
00:21.38 | sivana | and english can only be 1px font size |
00:22.07 | fugitivo | english is going to die soon |
00:22.28 | NewSole | english will die and everyone will speek geek |
00:22.32 | Ariel_ | english is not going to die. |
00:22.45 | colinm_ | pig latin for the masses! |
00:22.48 | Ariel_ | english is going to morph |
00:23.42 | fugitivo | we'll all speak speranto |
00:23.51 | MagicFab | Trolling is so '90s |
00:24.04 | MagicFab | a well.. back to the n00b forums |
00:24.18 | MagicFab | tx for nothing |
00:26.29 | NewSole | sorry folks... I live near Ottawa and I think its bull crap... people from hull look at people like they are skum cause they dont talk to them in french |
00:26.49 | sivana | quoi? |
00:27.00 | NewSole | and when ever there is something good out they demand a french version of it. |
00:28.12 | sivana | Asteriske |
00:29.09 | NewSole | my friend worked in public works in hull... |
00:29.29 | NewSole | and they fired him because all his C code was not French |
00:30.25 | sivana | you mean the comments? |
00:30.34 | NewSole | no the code |
00:30.54 | NewSole | his comments were in english and french |
00:31.04 | sivana | how can C code be french? |
00:31.38 | NewSole | thats why it was stupid... |
00:31.46 | NewSole | C code is english based |
00:31.55 | SpaceBass | alors qu'est-ce-cest le problem? |
00:32.09 | Ariel_ | SpaceBass, says what's the problem |
00:32.20 | SpaceBass | very poorly at that |
00:32.22 | NewSole | I know |
00:32.24 | JerJer | the functions he created could be named using the frechy lingo |
00:32.36 | sivana | ah ya |
00:32.39 | NewSole | ya but not print |
00:32.46 | NewSole | or while |
00:32.49 | *** join/#asterisk popooya (~popooya@07738e59f333d31b.session.tor) |
00:32.49 | NewSole | or for |
00:32.58 | NewSole | etc... |
00:34.01 | b0lt | Hey |
00:34.06 | JerJer | hoe |
00:34.11 | b0lt | I just installed asterisk, and want to use it with broadvoice |
00:34.18 | JerJer | good luck |
00:34.40 | b0lt | is there any way I could plug a phone into a modem on the asterisk server to get a dial tone, etc? |
00:34.44 | SpaceBass | b0lt did you check out the article from geek gazette on it |
00:34.51 | b0lt | SpaceBass, yeah |
00:35.04 | b0lt | except it uses official digitel hardware |
00:35.05 | b0lt | 00:03.0 Communication controller: Rockwell International HCF 56k Data/Fax Modem (rev 01) |
00:35.11 | b0lt | i have a conexant softmodem =) |
00:35.22 | b0lt | (using linuxant drivers) |
00:35.34 | b0lt | the modem operates through /dev/modem just like any other |
00:35.39 | SpaceBass | b0lt typically regular modems lack the digital audio converters for voice |
00:36.06 | b0lt | this one said it was a voice/fax modem on the box |
00:36.25 | sivana | SpaceBass: where's the article? |
00:36.41 | SpaceBass | b0lt my understanding is that it could work in theroy if you had drivers... but I've never heard of winmodem drivers existing |
00:36.52 | SpaceBass | you can get an x100p card on ebay for $10.00 |
00:36.56 | SpaceBass | thats your best bet |
00:36.57 | b0lt | http://www.linuxant.com/company/ |
00:37.10 | *** join/#asterisk darwin_35 (user@cas-11.ftsm-noc.mtlnk-129.valuelinx.net) |
00:37.45 | SpaceBass | but those aren't * drivers |
00:38.09 | b0lt | yeah, they are |
00:38.25 | b0lt | http://www.linuxant.com/drivers/ |
00:38.50 | SpaceBass | hummm not seeing any |
00:39.31 | SpaceBass | sivana I'm looking for the article |
00:39.42 | sivana | SpaceBass: about Broadvoice? |
00:39.44 | niZon | yay |
00:39.46 | b0lt | SpaceBass, http://www.linuxant.com/drivers/hcf/downloads-license.php |
00:39.49 | SpaceBass | sivana yep |
00:39.50 | niZon | link2voip gave me a refund |
00:39.58 | b0lt | http://www.google.com/url?sa=U&start=3&q=http%3A//geekgazette.com/index.php%3Foption%3Dcom_content%26task%3Dview%26id%3D20%26Itemid%3D26&ei=W7O4QsGuAqCUsAHyuayWCw&sig2=p3gbWcoHyQQRo4F7EOBq4w |
00:41.14 | SpaceBass | sivana thats the one http://geekgazette.com/index.php?option=com_content&task=view&id=2&Itemid=26 |
00:41.46 | SpaceBass | b0lt that linuxant.com link was for a license |
00:41.46 | meppl | gute nacht - good night |
00:41.59 | harryvv | summer has not reached vanouver bc yet |
00:42.00 | harryvv | :) |
00:43.00 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) [NETSPLIT VICTIM] |
00:44.41 | *** join/#asterisk drbrown (~chatzilla@user-0cdv208.cable.mindspring.com) |
00:44.56 | Pete_Largo | anyone using voip 'did' numbers for incoming faxes? |
00:45.38 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
00:45.42 | SpaceBass | Pete_Largo not yet... plan on trying |
00:45.57 | Pete_Largo | how are you thinking of doing it spacebass? |
00:46.30 | SpaceBass | Pete_Largo in what sense? |
00:46.38 | SpaceBass | thinking of using a BV account |
00:46.52 | Pete_Largo | point the did to a fax machine/ata or some sort of fax to mail? |
00:47.02 | SpaceBass | using amp? |
00:47.10 | Pete_Largo | amp? |
00:47.13 | Pete_Largo | ~amp |
00:47.14 | jbot | it has been said that amp is an Audio MPEG Player. [non-free], or http://amp.coalescentsystems.ca/ |
00:47.21 | Pete_Largo | that amp? |
00:47.30 | SpaceBass | * management portal - web front end |
00:47.43 | SpaceBass | comes with *@home |
00:47.51 | Pete_Largo | well, ok, but what about the back end? |
00:48.08 | SpaceBass | leme look at my fax setup |
00:49.10 | *** join/#asterisk hassler (~hassler@cpe-65-31-36-179.woh.res.rr.com) |
00:49.18 | SpaceBass | Pete_Largo well, I have a ext-fax context, so I have the did just do a s,1,goto(ext-fax,in_fax,1) |
00:49.53 | SpaceBass | but that ext-fax is from amp/asterisk@home I think |
00:50.10 | hassler | did I just jump in the middle of a fax discussion? |
00:50.13 | Pete_Largo | what is in_fax ? |
00:50.21 | Pete_Largo | yeah, trying to figure out if * can do it |
00:50.40 | SpaceBass | it works on my pstn line |
00:50.49 | harryvv | now is asterisk at home a iso and what OS does it use? |
00:50.49 | hassler | mine are always garbled -- someone said timing problems in zap are causing it, but haven't been able to verify |
00:50.58 | SpaceBass | I'll patebin my ext-fax |
00:51.25 | Pete_Largo | thanks |
00:53.55 | *** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
00:54.20 | SpaceBass | http://pastebin.ca/15449 |
00:55.37 | khemir | harryvv: Centos 4 |
00:55.41 | colinm_ | harryvv: it won't run from the cd, but it is distributed as an iso. based on centos, which is based on rhel. |
00:56.02 | bkw_ | oh guys |
00:56.16 | bkw_ | go check pbxfreeware.org out.. I just posted a more extensive res_js example |
00:56.23 | bkw_ | that interfaces with an external website |
00:57.42 | bkw_ | lalalla |
00:57.44 | bkw_ | NEXT!!!!! |
00:57.49 | WilliamK | hey brian |
00:57.51 | doughecka | hail bkw_ |
00:57.57 | bkw_ | WilliamK, yes? |
00:58.28 | bkw_ | wasabi |
00:58.51 | WilliamK | lots |
00:58.52 | WilliamK | =) |
00:59.03 | WilliamK | libpri was broken a bit ago after that one major update of ANI |
00:59.05 | harryvv | khemir: is centos like fedora? |
00:59.13 | bkw_ | you coming to cluecon? |
00:59.14 | doughecka | wasabi |
00:59.16 | doughecka | huh |
00:59.19 | WilliamK | -- Extension 'AD D' in context 'incoming-trunk01' from '9038834303' does not exist. Rejecting call on channel 0/1, span 1 |
00:59.32 | WilliamK | no idea how it came up with AD D |
00:59.57 | doughecka | bkw_ make it closer and I will go |
00:59.58 | SpaceBass | Pete_Largo - does that help? I use it for distenctive ring right now |
00:59.59 | cluecon | harryvv: centos is based on rhel. |
01:00.12 | Pete_Largo | can you pastebin the macro? |
01:00.19 | SpaceBass | which'en? |
01:00.27 | SpaceBass | faxrecieve? |
01:01.02 | *** join/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca) |
01:01.10 | doughecka | :) |
01:01.18 | SarahEmm | heya ;) |
01:01.24 | SpaceBass | [macro-faxreceive] |
01:01.24 | SpaceBass | exten => s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif) |
01:01.24 | SpaceBass | exten => s,2,SetVar(EMAILADDR=${FAX_RX_EMAIL}) |
01:01.24 | SpaceBass | exten => s,3,rxfax(${FAXFILE}) |
01:01.24 | SpaceBass | exten => s,103,SetVar(EMAILADDR=${FAX_RX_EMAIL}) |
01:01.24 | SpaceBass | exten => s,104,Goto(3) |
01:01.28 | SpaceBass | oops... sorry |
01:01.28 | harryvv | i see |
01:01.33 | SpaceBass | itchy trigger finger |
01:01.36 | bkw_ | SpaceBass, DUDE |
01:01.39 | bkw_ | he said pastebin |
01:01.41 | bkw_ | NOT HERE boi |
01:01.43 | bkw_ | *SMACK* |
01:01.54 | hassler | are you folks receiving faxes correctly? |
01:02.07 | bkw_ | yes |
01:02.09 | bkw_ | why? |
01:02.13 | bkw_ | you have squished ones? |
01:02.16 | hassler | mine are always garbled |
01:02.21 | bkw_ | on a PRI? |
01:02.22 | doughecka | bkw_ how reliable are your faxes? |
01:02.24 | hassler | squished may be a good description |
01:02.24 | bkw_ | check your clocking |
01:02.31 | bkw_ | your clock src is wrong |
01:02.34 | bkw_ | and you have frame slips |
01:02.39 | bkw_ | you won't hear it usually on voice calls |
01:02.42 | bkw_ | but its a clicking noice |
01:02.44 | SarahEmm | i'm not.. but i don't have it set up |
01:02.45 | bkw_ | er noise |
01:03.06 | hassler | how? using digium wildcard tdm400p card |
01:03.14 | doughecka | hahaha |
01:03.43 | bkw_ | hassler, no go |
01:03.49 | bkw_ | you can't use a tdm400p for faxing |
01:03.51 | bkw_ | it just won't work |
01:03.53 | bkw_ | give up now |
01:03.58 | bkw_ | you'll never have it work right |
01:04.14 | hassler | that's what I've been hearing, someone else said it was a timing issue broken in zap code |
01:04.26 | doughecka | bkw_: would you rely on ALL faxes coming in through asterisk? |
01:04.29 | bkw_ | no its not the code |
01:04.30 | harryvv | bkw,what do you recomend? |
01:04.33 | *** join/#asterisk remmo (~rem@smack.isp.net.au) |
01:04.35 | bkw_ | dougheckawe do right now |
01:04.35 | bkw_ | why? |
01:04.45 | bkw_ | well 1. make digium fix it |
01:04.51 | bkw_ | 2 bitch and moan till they do. |
01:04.54 | bkw_ | or |
01:04.55 | doughecka | I want to know if I can rely on it for ALL my faxes... |
01:04.56 | bkw_ | 3 |
01:04.59 | bkw_ | get some X101p's |
01:05.04 | doughecka | everyonce in awhile I will get a messed up page |
01:05.10 | harryvv | the 101 is a replacment for the 100? |
01:05.13 | bkw_ | doughecka yes you will get some messed up pages from time to time |
01:05.19 | doughecka | bkw_: :I could never get x101 to recieve faxes ok |
01:05.22 | bkw_ | but usually its the sending fax machine's fault |
01:05.28 | bkw_ | I got mien to do it 100% |
01:05.33 | SpaceBass | my x100p recieves faxes |
01:05.35 | SarahEmm | what's the difference between x100p and x101p? |
01:05.35 | bkw_ | I setup our system to fax hammer it one night |
01:05.40 | bkw_ | for 8 hours straight |
01:05.41 | bkw_ | it worked |
01:05.44 | bkw_ | 100% |
01:05.47 | harryvv | bkw, in what way those faxes dont follow fax protocol standards? |
01:05.53 | doughecka | bkw_: cool |
01:05.57 | bkw_ | have you read the fax standards? |
01:06.01 | harryvv | nope |
01:06.02 | bkw_ | its worse than SIP |
01:06.09 | hassler | x100p? |
01:06.13 | doughecka | haha |
01:06.22 | bkw_ | hassler, yes same thing |
01:06.26 | *** join/#asterisk eric`` (~e@adsl-9-111-66.mia.bellsouth.net) |
01:06.35 | bkw_ | some fax machiens just boldly break the standard |
01:06.38 | hassler | ahhh, the one port FXO card. |
01:06.40 | bkw_ | some just don't follow it much if any |
01:06.43 | harryvv | bkw, so what card did you use for the fax machine or was it machines ? |
01:06.46 | doughecka | go where no fax machine has gone before! |
01:07.00 | bkw_ | harryvv, our hylafax system sending faxes to my x101p in my box here |
01:07.03 | bkw_ | for 8 hours straight |
01:07.08 | bkw_ | one fax every 3 min |
01:07.12 | bkw_ | 2 pages long |
01:07.14 | hassler | I'd love to have the fax line detect fax or voice |
01:07.15 | harryvv | yes, i have read about hylafax |
01:07.56 | hassler | does the x100p work the same otherwise as the tdm400p? |
01:08.05 | harryvv | no |
01:08.36 | harryvv | i need to get my ip500 completed. fun fun. |
01:09.18 | bkw_ | its just an FXO |
01:09.18 | *** join/#asterisk [illuminatus] (~illuminat@cpe-65-185-103-95.woh.res.rr.com) |
01:09.47 | [illuminatus] | any idea on why i can make outgoing calls but I can't receive incoming calls? |
01:09.52 | hassler | but would work the same as a tdm400 with a single fxo daughter? |
01:09.55 | SpaceBass | ha! |
01:10.06 | SpaceBass | sshed into my * box... saw my wife call my cell before it rang |
01:10.30 | Pete_Largo | I would like to receive incoming faxes over a voip 'did' number... |
01:11.19 | harryvv | [illuminatus]: check your context |
01:11.27 | doughecka | SpaceBass: didja kill the connection before it went through? |
01:11.43 | SpaceBass | lol |
01:11.44 | SpaceBass | should have |
01:11.46 | Chuji | [illuminatus] : We're going to need a little more info |
01:11.49 | *** join/#asterisk eric`` (~e@adsl-9-111-66.mia.bellsouth.net) |
01:12.01 | Chuji | SpaceBass : One day I printed out a usage report for my wife out of the CDR tables and she wasn't amused |
01:12.11 | SpaceBass | she doesnt like those eithers |
01:12.13 | Chuji | I told her, you talk on the phone too much, and here is proof |
01:12.19 | SpaceBass | and she claims not all sip channels ring |
01:12.22 | *** join/#asterisk jerlique (jerlique@lnk2.adl.adsl.esc.net.au) |
01:12.29 | SpaceBass | when i call in |
01:13.01 | [illuminatus] | Chuji: such as? |
01:13.01 | wunderkin | SpaceBass, grandstream phones? |
01:13.23 | SpaceBass | one hitachi, one cisco, one siemens, one pos ata |
01:13.26 | wunderkin | oh |
01:13.27 | Chuji | [illuminatus] : Did it ever work? SIP? IAX? tdm? |
01:14.12 | *** join/#asterisk fugitivo (~ajf@168.226.244.221) |
01:14.39 | [illuminatus] | incoming calls never worked. I just got outgoing calls to properly work. It's SIP |
01:14.43 | [illuminatus] | there's no NAT in betwee |
01:14.47 | [illuminatus] | between* |
01:15.18 | Chuji | [illuminatus] : Does the CLI give an error? |
01:15.47 | [illuminatus] | let me SSH in and look |
01:15.56 | [illuminatus] | http://pastebin.ca/15454 for config |
01:17.30 | Chuji | [illuminatus] : I'll need to see the Dial line |
01:19.01 | [illuminatus] | ok console is spitting out rap whe ppl call |
01:19.21 | *** join/#asterisk dalaila (Santon@adsl-200-136.tricom.net) |
01:19.42 | dalaila | I have problem with Asteriskhome |
01:19.45 | *** join/#asterisk LapTop006 (~laptop006@sparc006.chriskaine.com.au) |
01:19.56 | [illuminatus] | http://pastebin.ca/15455 |
01:20.52 | Chuji | ~asterisk@home |
01:20.53 | jbot | i guess asterisk@home is http://asteriskathome.sourceforge.net/, or http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home |
01:21.03 | *** join/#asterisk cluecon (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
01:21.34 | [illuminatus] | Chuji: what do you mean by the dial line? |
01:21.46 | *** join/#asterisk dos000 (~dos000@66.11.173.123) |
01:23.12 | *** join/#asterisk bjohnson (~bjohnson@ip172-172.dsl.istop.com) |
01:23.59 | Chuji | [illuminatus] : That all looks good. Doesn't look like your provider is accepting the call |
01:24.39 | [illuminatus] | but it seems to be getting to our * box... |
01:25.03 | [illuminatus] | why do you think the provider isn't accepting the calls? (j/w) |
01:25.32 | Chuji | [illuminatus] : Take a look at this |
01:25.33 | Chuji | Found peer 'pstn'\ |
01:25.33 | Chuji | it's using that peer for incoming |
01:25.38 | Chuji | but in that peer you have not defined a context |
01:26.23 | [illuminatus] | ok. so what should I do? |
01:26.31 | Chuji | put a context = incoming (or wherever your incoming is) |
01:26.39 | Chuji | in that sip peer entry |
01:27.46 | Chuji | Have you defined an incoming context in your dialplan? |
01:27.46 | [illuminatus] | let me look at it |
01:27.46 | dalaila | My problem is with the user that asterisk create, this user doesn't autenticated by web |
01:27.46 | SpaceBass | dalaila web what? voicemail? |
01:28.43 | [illuminatus] | sorry i'm a little bit of a nub at this |
01:28.51 | [illuminatus] | i can't even find the dial plan now :( |
01:29.10 | SpaceBass | [illuminatus] is this a sip or pstn ? |
01:29.13 | Chuji | [illuminatus] : extensions.conf |
01:29.19 | Chuji | [illuminatus] : or |
01:29.22 | [illuminatus] | it's a SIP trunk to our provider |
01:29.28 | Chuji | 'show dialplan' from the CLI |
01:29.32 | SpaceBass | [illuminatus] bv? |
01:29.43 | [illuminatus] | CentricVoice |
01:29.47 | dalaila | SpaceBass The user that Asterisk gave to me, Does authenticated |
01:30.26 | dalaila | SpaceBass the user that I put is admin |
01:30.35 | SpaceBass | dalaila don't follow... what are you trying to authenticate to? |
01:31.12 | dalaila | Asterisk Management Portal |
01:31.13 | [illuminatus] | pstn is not defined in extentions |
01:31.18 | [illuminatus] | only from-pstn is |
01:31.31 | SpaceBass | [illuminatus] using amp? |
01:31.35 | [illuminatus] | unless i just said something stupid |
01:31.37 | [illuminatus] | AMP is installed yes |
01:31.45 | SpaceBass | dalaila and what user do you want to authenticate? |
01:31.51 | dalaila | It give me for default Admin |
01:32.08 | SpaceBass | dalaila and you want to create another user? |
01:32.09 | dalaila | the password for default is password |
01:32.15 | Chuji | [illuminatus] : as a termporary fix, you can put 'context = from-pstn' in sip.conf under [pstn] |
01:32.31 | Chuji | see if that gets you going |
01:32.53 | SpaceBass | [illuminatus] you need to look in /etc/asterisk/extensions_additional.conf or in amp for the context |
01:33.10 | SpaceBass | [illuminatus] ignore me and follow chuji thats a good idea |
01:33.37 | SpaceBass | Chuji its easiest with *@h to treat sip trunks as pstn for inbound |
01:33.50 | SpaceBass | dalaila you can change that password if thats what you want |
01:33.59 | dalaila | SpaceBass I tried with another user but it came up with the same error |
01:34.03 | Chuji | more and more people are using it around here. I guess I should install it once to see what it looks like |
01:34.10 | dalaila | SpaceBass I change the password |
01:34.11 | Chuji | Seems unnecessary to me |
01:34.24 | SpaceBass | dalaila a OS user or an extension you created in asterisk? |
01:34.35 | jontow | ugh.. installed a SCO OpenServer machine tonight.. *shudder* |
01:34.38 | jontow | what a nasty thing that is :) |
01:34.45 | Chuji | ~sco |
01:34.46 | jbot | it has been said that sco is the new name for Caldera, and they're making right fools of themselves, or the antichrist, or supposed to sue linux users today, or a bunch of jerks |
01:34.55 | SpaceBass | dalaila you can create another unix user like maint, but you cannot use an extension you have created to log into AMP |
01:35.02 | jontow | you got it, jbot. |
01:35.25 | dalaila | SpaceBass `How can I do this? |
01:35.27 | SpaceBass | Chuji its slick for what it is, but it creates newbiews like me and [illuminatus]... but thats probabbly a good thing in the end |
01:35.39 | SpaceBass | dalaila create a linux user? |
01:35.44 | SpaceBass | #linuxhelp :) |
01:36.02 | [illuminatus] | <PROTECTED> |
01:36.04 | dalaila | SpaceBass I create one name dalila |
01:36.08 | Chuji | ~man adduser |
01:36.15 | dalaila | SpaceBass longtime ago |
01:36.43 | SpaceBass | Pete_Largo I counldnt get anything until 1.0 to work correctly |
01:36.53 | SpaceBass | *@H 1.0 that is |
01:37.08 | Chuji | [illuminatus] : 'show dialplan from-pstn' | pastebin.ca |
01:37.10 | Pete_Largo | I'd rather use the CLI anyway |
01:37.27 | SpaceBass | can use cli with *@h |
01:37.41 | SpaceBass | *@H enabled me to learn enough to be able to manually create dial plans, etc |
01:37.44 | Pete_Largo | should have read 'CLI version' |
01:38.40 | Qwell | omg...my friend just explained the phone/network situation at his work... |
01:38.50 | SpaceBass | i read the handbook several times and was getting the hang of it.. I think I *could* have done it... what thew me was all the agi stuff |
01:38.52 | Qwell | They're sitting on about 20 phone lines (no, they don't have a T1), and using 56k dialup |
01:39.11 | Damin | Morning everyone.. |
01:39.17 | SpaceBass | cluecon if that existed I'd be a happy man! |
01:39.22 | dalaila | SpaceBass my problem is not to create a user, my problem is thAt i can not enter with the user and the password that I asigned |
01:39.40 | cluecon | Qwell: is this a company located in the US? |
01:39.48 | Qwell | cluecon: yes! |
01:39.54 | Qwell | makes me sad |
01:40.09 | Qwell | I'm gonna go there and regulate...smack the IT guy around a bit |
01:40.32 | harryvv | Qwell, no t-1? thats dumb with that many lines. |
01:40.33 | SpaceBass | dalaila I'm not sure how amp is set up... may need to create the user in /var/www/html/pannel |
01:40.38 | Sedorox | what does the place do? |
01:40.40 | Qwell | harryvv: exactly my point |
01:40.40 | SpaceBass | i haven't looked closely |
01:40.46 | Qwell | Sedorox: motorcycle dealership |
01:40.52 | SpaceBass | dalaila check #amportal those folks could tell you quickly |
01:40.59 | harryvv | I wonder how much thay are being charged with those 20 lines |
01:41.00 | Qwell | two locations, actually |
01:41.06 | Sedorox | I could see the lines... but I dun think they would really need very high speed internet... |
01:41.07 | [illuminatus] | http://pastebin.ca/15461 |
01:41.09 | Qwell | harryvv: alot more then a T1 would cost, thats for sure |
01:41.16 | harryvv | well, at least thay could get a fractional t-1 |
01:41.18 | [illuminatus] | the other two includes don't eit |
01:41.20 | Qwell | Sedorox: they have a second location... |
01:41.27 | Sedorox | ah |
01:41.27 | Qwell | data transferred back and forth all day |
01:41.32 | Sedorox | dammn |
01:41.35 | cluecon | Qwell: If i'm doing the math right...a standard phone line is at least $20 bucks. That is at least $400 a month. Your average T1 is gonna run about that. |
01:41.37 | dalaila | \j #amportal |
01:41.39 | Qwell | over a single 56k...heh |
01:41.49 | Qwell | cluecon: business lines are alot more, I'm sure |
01:41.50 | harryvv | Qwell, whats the location there and how much for a t-1 can that be combined voice/data? |
01:41.53 | SpaceBass | [illuminatus] there is no s in from-pstn |
01:41.55 | Sedorox | upgrade time! |
01:41.56 | Sedorox | :p |
01:42.02 | SpaceBass | i think thats your problem |
01:42.06 | Qwell | harryvv: not sure about pricing out there, Fort Worth, IN |
01:42.18 | *** join/#asterisk milkyflava (~milkyflav@240-7-237-24.gci.net) |
01:42.19 | Pete_Largo | IN = Indiana? |
01:42.22 | Chuji | [illuminatus] : Well, that is your problem. You need to define what happens when an incoming call hits your dialplan |
01:42.23 | Qwell | yeah |
01:42.25 | milkyflava | hello |
01:42.28 | Chuji | [illuminatus] : What should it do? |
01:42.38 | harryvv | cluecon: our t-1 here in bc cost anywhere from $750-1,100 CDN here |
01:42.42 | [illuminatus] | it is defined though. it goes to extention 502... |
01:42.44 | SpaceBass | Chuji there is a context he can goto with amp/aah |
01:42.46 | SpaceBass | leme look |
01:42.47 | [illuminatus] | it works when i dial 7777 |
01:43.04 | Pete_Largo | Off the top of my head I can't think of any CLECs that service IN |
01:43.10 | *** part/#asterisk SarahEmm (~sarahemm_@Toronto-HSE-ppp3681993.sympatico.ca) |
01:43.17 | milkyflava | stupid question - can I set up asterisk on my local network and make calls to computers in my local network without having to sign up for a service? |
01:43.20 | harryvv | Qwll, thats a pretty good price :) |
01:43.29 | Qwell | harryvv: for what? |
01:43.36 | harryvv | t1/pri |
01:43.39 | Sedorox | I'm helping the one IT guy... the new school they just built here is having all cisco equipment put in, including VoIP phones in every room... |
01:43.43 | Qwell | whats a good price? |
01:43.44 | Sedorox | the 7960's are pretty nice... |
01:43.59 | harryvv | I wonder how vonage does it by terminating there pstn in vaginia. |
01:44.14 | SpaceBass | Chuji and [illuminatus] i understand the problem and fix, but dont know enough to articulate it... when you hit 777 its going to from-pstn which includes time-check and thus from-pstn-reghours-nofax which then has the dial/sip502,FOO,BAR |
01:44.28 | jontow | milkyflava; yes. |
01:44.37 | milkyflava | thanks |
01:44.56 | Pete_Largo | vonage is probably using carrier ld termination |
01:45.01 | SpaceBass | [illuminatus] instead of goto(from-pstn, s,1)... |
01:45.19 | milkyflava | but if I want to make calls going outside to PSTN phones I would need to sign up, correct? |
01:45.24 | SpaceBass | [illuminatus] make it goto(from-pstn-reghours-nofax,s,1) |
01:45.37 | SpaceBass | mike-ff signup for what? |
01:45.50 | [illuminatus] | SpaceBass: i would if I knew how :( |
01:46.02 | cluecon | harryvv: we have a local T1 that is costing us about $600 USD. We are about to get some additional T1s put in that are just a per minute charge and will soon also have a DS3 circuit which has a cost of about 135 USD per T1 if we have the whole thing active. |
01:46.03 | *** join/#asterisk dyl0n (~mhappe@p548B28C8.dip0.t-ipconnect.de) |
01:46.15 | SpaceBass | [illuminatus] are you comfortable editiing the config files? |
01:46.38 | [illuminatus] | sure |
01:46.44 | harryvv | cluecon: what services do you sell |
01:46.48 | milkyflava | like with voipjet or broadvoic services |
01:46.57 | SpaceBass | [illuminatus] that pastebin you pasted earlier ... which .conf was it from? |
01:47.14 | Chuji | 135 per T1? in US? you should be able to do better than that w/ a DS3 |
01:47.27 | [illuminatus] | extentions.conf and then i included what it included from extentions_additional.conf |
01:47.29 | cluecon | harryvv: we don't sell, we use. it's a collections company and we run thru quite a few calls. |
01:47.37 | harryvv | ahhhh |
01:47.43 | SpaceBass | milkyflava you can sign up with one to make calls from 'the net' to pstn phones or you can use a FXO which allows you to plug a PSTN into * directly (and many other options including t1, etc) |
01:47.48 | harryvv | now thats a company i can hit on for biz :) |
01:47.50 | Pete_Largo | Chuji, the loop cost is probably raising the price |
01:48.10 | SpaceBass | [illuminatus] you have a line that is exten => 12143294838,1,Goto(from-pstn,s,1) |
01:48.22 | harryvv | cluecon who implemented the entire project? |
01:48.32 | SpaceBass | change it to goto(from-pstn-reghours-nofax, s,1) |
01:48.35 | cluecon | harryvv: define implement. |
01:48.39 | milkyflava | thats what was confusing me, so all I really need is a tmd400p 11b card from digium then I dont need to sign up with anyone but use my existing PSTN line, correct? |
01:48.45 | SpaceBass | and remove the include = > ext-did-custom |
01:48.48 | SpaceBass | dont think you need that |
01:48.52 | harryvv | who started and installed and configured all of it. |
01:49.08 | SpaceBass | milkyflava yep, or a x100p from ebay for $10.00usd |
01:49.18 | Chuji | ~x100p |
01:49.19 | jbot | i heard x100p is an obsolete card. you don't want to bother trying to make it (or any of the "digium compatible" clones work. Get a TDM01P, you will save your sanity. |
01:49.25 | cluecon | harryvv: i am. we are about to replace our current dialer with * boxes. |
01:49.40 | SpaceBass | milkyflava if you are just playing around you can get a $5/month account from BV and cancel anytime... some people have reported issues with them though |
01:50.07 | harryvv | cluecon how many channels is it going to handle? Im asuming the load is distributed among several boxes. |
01:50.20 | milkyflava | SpaceBass thanks, thats exactly what I am doing. |
01:50.30 | SpaceBass | whats the general idea b/h loadbalancing with * |
01:50.36 | SpaceBass | how do you do it... put some channels on each box? |
01:50.46 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
01:50.56 | Chuji | ~dundi |
01:50.57 | jbot | rumour has it, dundi is http://www.dundi.com |
01:51.00 | SpaceBass | milkyflava enjoy... and of note: BV allows more than one incoming call... but they dont tell you that :) |
01:51.07 | cluecon | harryvv: we are limiting it to 1 quad t card per box (max 96 zap channels and i think 30 zip per box). |
01:51.23 | harryvv | big call center |
01:51.24 | harryvv | :) |
01:51.44 | [illuminatus] | OMGWTFBBQ? |
01:51.52 | [illuminatus] | I can't test it from my PSTN line |
01:51.56 | harryvv | before your company did this, were thay given quotes by other commerical voip sla providers? |
01:51.59 | SpaceBass | this cheesy hotel room has a f'ing hot tub in the center of it... HA! |
01:52.02 | milkyflava | SpaceBass, excellent, thanks for the tip |
01:52.08 | [illuminatus] | because it when I dial the country code it says I don't need and and when I don't dial the country code it says I do need it |
01:52.28 | SpaceBass | [illuminatus] sounds like a dial pattern problem |
01:52.31 | cluecon | initial plan was to start out with 8 ts active but we just heard that sbc actually got its approval to come onto the property so i am trying to talk the boss into starting it up with 16 active. |
01:52.43 | [illuminatus] | SpaceBass: on my regular phone line? |
01:52.58 | harryvv | sbc come onto the property and do what? |
01:53.00 | harryvv | :) |
01:53.02 | SpaceBass | milkyflava one last one... google geek gazette asterisk@home |
01:53.02 | cluecon | harryvv: we use pure pri at the moment. it is better for us (due to the cost of data bandwidth) to use pri. |
01:53.07 | milkyflava | one more question - Should I install using source or *@home |
01:53.14 | cluecon | harryvv: run fiber to support the ds3. |
01:53.19 | SpaceBass | milkyflava good tutorial for setting up BV with * (if you can find the article, i never can) |
01:53.33 | milkyflava | SpaceBass I think that answered it. :) |
01:53.38 | Pete_Largo | it's on the broadvoice site in the support section |
01:53.42 | cluecon | milkyflava: use CVS-HEAD. |
01:53.54 | SpaceBass | Pete_Largo that doesnt work... at least i've never gotten it to successfully |
01:53.58 | cluecon | *@H is not what you want. |
01:54.02 | harryvv | cluecon yea thats nifty. Any ideas how vonage can sell service here in bc canada when there "I suspect" pstn termination is in vaginia? |
01:54.07 | Pete_Largo | http://www.broadvoice.com/support_install_asterisk.html |
01:54.12 | SpaceBass | ha ha ha! I'm spreading the *@H plague :) |
01:54.13 | Pete_Largo | it's what I used, and it works fine |
01:54.17 | milkyflava | ah, ok, thanks cluecon |
01:54.20 | Sedorox | *@H sucks..... |
01:54.27 | Sedorox | :p |
01:54.39 | Sedorox | lol |
01:54.44 | SpaceBass | <-- *@h lepper |
01:54.56 | Sedorox | :p |
01:55.04 | Pete_Largo | lol lepper |
01:55.23 | Sedorox | I have it.. but only to use parts of the config from (like setting it up to recieve a fax |
01:55.35 | *** join/#asterisk darwin_35 (user@cas-11.ftsm-noc.mtlnk-129.valuelinx.net) |
01:55.41 | cluecon | harryvv: simple. pstn termination is just required to get back to the traditional land lines. if you are big enough (vonage is) you can get great deals with the land line owners (sbc, qwest, sprint, etc.) that allow you to make a profit off of the voip service. |
01:56.07 | SpaceBass | Sedorox thats my take... use it for some stuff like fax and time check |
01:56.14 | SpaceBass | and wake up call and weather |
01:56.16 | Sedorox | yea |
01:56.27 | Sedorox | stuff you don't feel like making thats already done :p |
01:56.33 | SpaceBass | exactly |
01:57.08 | SpaceBass | bbl |
01:57.10 | Sedorox | actually... I believe I run release |
01:57.10 | Sedorox | lol |
01:57.30 | Sedorox | lol |
01:57.44 | Sedorox | nobody was insisting on it |
01:57.47 | Sedorox | ut oh |
01:58.43 | milkyflava | So is there a one doc that is preferred over others that is specific to on distro? |
01:59.06 | DarthClue | voip-info.org has some pretty straightforward docs. |
01:59.08 | milkyflava | or just follow the docs at the * site |
01:59.29 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
02:00.21 | milkyflava | ok, that one I have bookmarked, how about a distro or just whatever I'm used too or does it even matter |
02:00.25 | DarthClue | voip-info.org is slightly better. |
02:00.55 | DarthClue | I use FC because it is what I am familiar with. Some distros have better luck than others. What is your preferred distro? |
02:01.03 | milkyflava | FC |
02:01.11 | SpaceBass|AWAY | fc |
02:01.20 | fugitivo | i don't have a preferred distro, i use the best distro, gentoo |
02:01.24 | DarthClue | then use FC. You will get alot of flac about it, but use it if you like it. |
02:01.26 | Sedorox | Gentoo |
02:01.28 | DarthClue | GENTOO SUCKS!!!! |
02:01.35 | fugitivo | DarthClue-- |
02:01.38 | milkyflava | lol |
02:01.42 | Sedorox | lol |
02:01.49 | Pete_Largo | I'm using FC3 and it works fine |
02:01.49 | darwin_35 | Slack or Debian |
02:01.55 | Pete_Largo | for me |
02:01.57 | Sedorox | each to his own |
02:02.08 | darwin_35 | Feudoria sucks |
02:02.11 | Pete_Largo | just remember to compile zaptel with make linux26 |
02:02.14 | harryvv | slack has almost been free of crashes...on its own. |
02:02.15 | fugitivo | fc is for l00sers |
02:02.19 | harryvv | err i mean fc3 |
02:02.21 | fugitivo | lol |
02:02.35 | milkyflava | windows 3.1 is rock solid |
02:02.38 | Pete_Largo | harryvv, go take a long walk off a short pier |
02:02.43 | [illuminatus] | grrr |
02:02.47 | jontow | why .. i mean, honestly.. why |
02:02.48 | harryvv | fug, fc3 runs my phones just phone |
02:02.51 | [illuminatus] | still not working |
02:02.51 | [illuminatus] | http://pastebin.ca/15467 |
02:02.52 | fugitivo | harryvv: crashes? only if you use windows |
02:02.55 | darwin_35 | Slack10 rocks 10.1 you have to replace gcc |
02:02.59 | sivana | does linux have a .zip unzip command? |
02:03.01 | jontow | "WINDOWS XP SUCKS, WINDOWS 2000 ALL THE WAY" <-- seriously, do you guys see what you're saying? |
02:03.10 | jontow | shut up and use what works best for you |
02:03.14 | Sedorox | [22:02] <Pete_Largo> just remember to compile zaptel with make linux26 |
02:03.15 | Sedorox | I haven't had to |
02:03.19 | darwin_35 | FreeBSD |
02:03.21 | fugitivo | harryvv: i'm just kidding, i like linux in general |
02:03.27 | DarthClue | But windows XP does suck. |
02:03.31 | milkyflava | love the BSD |
02:03.32 | jontow | ;) |
02:03.48 | darwin_35 | Windows is Great as a end user os but not a server |
02:03.49 | fugitivo | yeah, openbsd is the winner |
02:04.08 | harryvv | linux is okay..i made the mistake of downloading 64 bit for fc3 and then use it as a work station and server. besides, who needs more memory then what 32 bit can provide. |
02:04.09 | *** join/#asterisk |Ladybug| (~ladybug@219.95.78.33) |
02:04.13 | milkyflava | I read what the openBSD developer had to say about linux |
02:04.14 | fugitivo | darwin_35: windows as end user is a pain in the ass |
02:04.16 | |Ladybug| | hi all |
02:04.40 | fugitivo | harryvv: i use gentoo 64bit in my laptop |
02:04.47 | jontow | personally, my first 2 choices are always FreeBSD and NetBSD .. but with asterisk; i tend to not think that.. they still have some serious issues to work through (most specifically with random zaptel hardware); so i use gentoo, and i've gladly used debian though i just wasn't comfortable with it.. i now am also using beehive linux on a very weak machine and have a lot more call volume to push through the machine |
02:04.50 | darwin_35 | weI agree but was trying to be nice |
02:05.04 | harryvv | fug, why? |
02:05.07 | fugitivo | jontow++ |
02:05.09 | [illuminatus] | anyone know why i can't receive calls? |
02:05.10 | harryvv | why why |
02:05.15 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
02:05.15 | [illuminatus] | everything seems to be setup correctly |
02:05.23 | darwin_35 | Freebsd and Zaptel are stable |
02:05.28 | [illuminatus] | http://pastebin.ca/15467 <-- what happens when i try to make a call |
02:05.39 | |Ladybug| | i need help with asterisk... ne1 can help? |
02:05.39 | jontow | if i had the time/energy/knowledge, and/or someone was paying me to learn.. i'd gladly fix all of the bugs i could find with freebsd and netbsd using zaptel hardware. |
02:05.40 | fugitivo | harryvv: why what? |
02:05.53 | darwin_35 | I hope to have Festival 1.95 working soon |
02:05.54 | jontow | however.. none of the above is true, so i'm content with gentoo until most of those conditions can be met |
02:06.08 | jontow | darwin_35; using a T100P under FreeBSD? |
02:06.22 | |Ladybug| | i wrote a program using vc++ to interact with asterisk |
02:06.29 | fugitivo | gentoo is the easiest distribution to mantain |
02:06.31 | |Ladybug| | i am using a softphone called sjphone |
02:06.35 | Sedorox | I've had ok luck with zaptel and fbsd |
02:06.35 | jontow | agreed.. |
02:06.39 | milkyflava | Thanks all for the answers to my questions, I'm gonna get some CVS-HEAD and start learning |
02:06.43 | Sedorox | but then it started locking the kernel :( |
02:07.08 | jontow | "emerge --sync ; emerge --deep update world" rocks my --deep world.. |
02:07.08 | |Ladybug| | but everytime i transfer call... my program crashed |
02:07.08 | jontow | ;) |
02:07.09 | darwin_35 | not yet . but I have a tdm400 |
02:07.09 | |Ladybug| | can ne1 help? |
02:07.13 | fugitivo | jontow: don't forget emerge search, it's great |
02:07.18 | jontow | :) |
02:07.23 | darwin_35 | man I hate dialup |
02:07.23 | jontow | i use locate for a lot of my searching |
02:07.23 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
02:07.34 | jontow | but im not too familiar with portage yet.. still learning :) |
02:07.37 | darwin_35 | I want my highspeed back |
02:07.45 | jontow | darwin_35; me too. mine burnt. |
02:07.54 | fugitivo | jontow: just emerge search whateveryouwant |
02:08.04 | darwin_35 | I only use the ports for base install but build * from cvs head |
02:08.06 | jontow | now i have 15 machines on a 26.4kbps dialup |
02:08.12 | |Ladybug| | ello??? |
02:08.25 | DarthClue | Ladybug: what is the question? |
02:08.47 | fugitivo | |Ladybug|: softphones for linux are not what I call "stable" |
02:08.56 | |Ladybug| | oh |
02:09.02 | |Ladybug| | i wrote a program using vc++ to interact with asterisk |
02:09.04 | darwin_35 | there is one from Novel |
02:09.06 | |Ladybug| | i am using a softphone called sjphone |
02:09.06 | jontow | ladybug; part of the issue here is that you asked about VC++ and asterisk + program crashes when calls are transferred; sounds like a debugging issue.. and .... under windows? |
02:09.09 | darwin_35 | I forget the name |
02:09.16 | |Ladybug| | yes... |
02:09.18 | darwin_35 | SFLPHone |
02:09.21 | |Ladybug| | it is running under windows |
02:09.29 | fugitivo | darwin_35: novell? |
02:09.32 | jontow | ok, so you're already outside of the comfort zone for a lot of people here :) |
02:09.42 | darwin_35 | yes |
02:09.46 | |Ladybug| | i know... i never like microsoft |
02:09.49 | darwin_35 | Novell |
02:09.51 | fugitivo | darwin_35: for linux? |
02:09.54 | |Ladybug| | but my co's licience is using microsoft |
02:09.55 | |Ladybug| | =( |
02:10.09 | |Ladybug| | but the asterisk server is running under linux |
02:10.20 | jontow | i understand the need for it.. in a lot of ways, as i'm stuck using a lot of it here (and maintaining it.. ugh. frontpage+coldfusion+win2k+IIS5+ms-sql7 == UGGGGGGHHHH) |
02:10.37 | fugitivo | jontow: man, i'm sorry |
02:10.39 | jontow | touch any one of the 5 and the rest just come crashing down |
02:10.57 | jontow | i've happily deemed it "house of cards syndrome", and laugh whenever possible |
02:11.17 | jontow | the problem is .. i rarely get time to update/maintain my unix machines because.. honestly, they never need attention |
02:11.19 | fugitivo | lol |
02:11.43 | jontow | the win2k machines always need the attention.. whining crybabies they are.. "someone poked me with a portscan, IM TELLING MOM" etc. |
02:11.52 | jontow | stupid crap. |
02:11.55 | darwin_35 | brb |
02:11.57 | jontow | ;) |
02:11.59 | |Ladybug| | just asking... when i transfer/redirect the calls... what shud be the response of the asterisk? |
02:12.08 | jontow | darthclue; exactly. |
02:12.38 | fugitivo | |Ladybug|: what kind of application did you develop with vc++? |
02:12.42 | jontow | ladybug; depends.. how're you transferring? # or client-initiated or manager-api initiated, etc? |
02:12.53 | |Ladybug| | manager api |
02:13.03 | |Ladybug| | i am developing a dll... then another MFC program to run it |
02:13.05 | jontow | have you successfully transferred one by hand via the manager api? |
02:13.10 | DarthClue | jontow: i put a linux box in as a web / email / database server when i started this job last september. i haven't had to do anything to it since then. i have a windows server that i have to 'fix' everyday. |
02:13.13 | jontow | (and watched the response?) |
02:13.17 | |Ladybug| | yes |
02:13.24 | |Ladybug| | the originate button r perfectly ok... |
02:13.31 | |Ladybug| | just transfering is a prob |
02:13.37 | |Ladybug| | i have 2 ipphone here |
02:13.41 | fugitivo | |Ladybug|: did you try with another softphone? |
02:13.46 | |Ladybug| | using a softphone called sjphone |
02:14.00 | jontow | try diax2, x-lite, or something else and see if you can replicate the situation |
02:14.02 | fugitivo | |Ladybug|: try with another softphone, xlite is free |
02:14.06 | |Ladybug| | used anoter phone called sumthing like Bellphone |
02:14.08 | jontow | you may be ahead to find out that its a bug in sjphone.. |
02:14.11 | |Ladybug| | oh... i m downloading now |
02:14.49 | |Ladybug| | try diax2... |
02:14.57 | |Ladybug| | it doesnt seem to like my pc at all |
02:14.59 | darwin_35 | man this exten.conf I am working on is a killer |
02:15.06 | jontow | i'd so like to quit this job and get something where i can spend a little more time troubleshooting/developing/maintaining/supporting asterisk and related applications |
02:15.18 | darwin_35 | has anyone here mapped all th *XX nmbrs |
02:15.23 | fugitivo | |Ladybug|: xlite seems stable |
02:15.34 | DarthClue | jontow: come to cluecon. |
02:15.37 | jontow | as is i get slightly into a project with * and get pulled to fix a windows server thats on fire |
02:15.43 | Sedorox | exten => *XX.,1,YayItWorks :p |
02:15.47 | Sedorox | ops |
02:15.50 | jontow | darthclue; and who'll throw my ticket? |
02:15.51 | Sedorox | exten => _*XX.,1,YayItWorks :p |
02:15.51 | |Ladybug| | is the x-lite using the SIP too? |
02:15.52 | Sedorox | there.. done |
02:15.53 | Sedorox | :p |
02:15.53 | jontow | :/ |
02:16.05 | DarthClue | jontow: plane ticket? where you located at? |
02:16.06 | fugitivo | |Ladybug|: yes |
02:16.07 | jontow | (and my gf's .. since we never leave home without eachother.. long story) |
02:16.10 | jontow | upstate NY |
02:16.11 | darwin_35 | no I melike *10 and so on |
02:16.24 | fugitivo | i want a ticket too, i live in argentina |
02:16.29 | *** part/#asterisk brian13 (~user@c-24-98-71-208.hsd1.ga.comcast.net) |
02:16.32 | jontow | ;) |
02:16.38 | darwin_35 | I have a few of them but there are alot more |
02:16.40 | Sedorox | darwin_35: ouch.. your seperating them? |
02:16.47 | jontow | "yeah.. while you're handing out tickets.. can i have 4?" |
02:16.49 | jontow | :D |
02:16.54 | Nugget | I miss all the travel I used to do, but I don't miss it too. it's nice to be able to make plans for more than a week or two in advance. |
02:17.22 | Nugget | but I miss globetrotting. |
02:17.32 | |Ladybug| | ok.. thanks |
02:17.39 | harryvv | nugget where and for what reason did you tracel |
02:17.43 | harryvv | travel |
02:18.08 | Nugget | I used to fly to our customers and teach them how to use our software. but I left the company and now I work for one of the customers now |
02:18.25 | harryvv | I remember being in a airforce base on leave taking millitary hops for free back home and saw a flight out to hawaii for 10 dollars. that was tempting. |
02:18.28 | fugitivo | i don't miss travel at all, i had a bad experience with a plane, and now i have trauma to flying |
02:18.36 | Nugget | UK, france, switzerland, australia, and japan. I flew about 100,000 miles a year for three years. |
02:18.46 | |Ladybug| | internet is slow here/... =( |
02:18.50 | Nugget | now I don't leave my house. it was quite a culture shock |
02:18.51 | darwin_35 | http://pastebin.ca/15468 this is what I mean |
02:19.09 | harryvv | nugget yea I bet :) |
02:19.14 | jontow | some of our customers pay way better than this company |
02:19.18 | harryvv | lady, you f? |
02:19.26 | |Ladybug| | yes... |
02:19.26 | jontow | kinda makes me sad because many of them are far less skilled and yet still are over double what i get paid |
02:19.36 | |Ladybug| | how can a male named themselves as Lady? |
02:19.47 | harryvv | anyone can mask anyone on irc :) |
02:19.49 | fugitivo | jontow: it's always like that |
02:20.03 | Nugget | I miss getting to hang out in paris or tokyo on someone else's dime. but I'm enjoying being able to rediscover what "weekend" means and being more involved locally |
02:20.05 | jontow | ladybug; san francisco? |
02:20.31 | harryvv | nugget yea :) how about doing the occational voip install out of state? |
02:20.46 | darwin_35 | I found the list on the net of the mapped *XX exten I am just mappping them in * |
02:20.49 | jontow | or for that matter any major metropolis? :) i hear there are lots of guys calling themselves lady under the right circumstances.. |
02:21.02 | |Ladybug| | nope... i m from malaysia |
02:21.25 | harryvv | lady was Salam? |
02:21.49 | |Ladybug| | Salam? |
02:21.54 | |Ladybug| | i am a chinese lady |
02:22.05 | harryvv | mandarin or cantonese? |
02:22.05 | Nugget | flying to cleveland just doesn't have the same appeal as flying to sydney. :) |
02:22.14 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
02:22.24 | harryvv | Nugget, what did you do for that company? |
02:22.31 | darwin_35 | my pet project is to map them all in * |
02:22.35 | Nugget | professional services. |
02:22.39 | darwin_35 | for all users |
02:22.40 | |Ladybug| | mandarin |
02:22.52 | Nugget | I trained people to use the product and I did app development. |
02:22.55 | Nugget | (for ud.com) |
02:22.55 | harryvv | Lady, we have a huge chinese population here in vancouver bc. |
02:22.57 | fugitivo | i want to learn mandarin |
02:23.13 | harryvv | fug, move to vancouver and learn it :) |
02:23.54 | SpaceBass|AWAY | in the year of darkness 2029? |
02:24.05 | fugitivo | i know it's impossible to learn to speak it correctly |
02:24.16 | harryvv | or 2007 |
02:24.36 | jontow | darthclue; dude.. you don't want to know that.. :) |
02:24.39 | DarthClue | no. The |
02:24.52 | harryvv | I think both languages have the most difficulty because the way the mouth and tounge muscles work. |
02:24.55 | darwin_35 | if anyone can help with this please do |
02:24.57 | DarthClue | jontow: too late, i already know. |
02:25.02 | |Ladybug| | just installing xlite |
02:25.03 | jontow | thats too bad :/ |
02:25.07 | |Ladybug| | how to setup? =( |
02:25.10 | DarthClue | darwin: help with what? |
02:25.23 | Nugget | I find japanese easy, but french is totally obtuse to me. french has sounds that my ear just can't hear. |
02:25.24 | harryvv | lady, first of all do you have a firewall? |
02:25.30 | Nugget | german is practically english |
02:25.34 | |Ladybug| | nope |
02:25.35 | fugitivo | mandarin is difficult because the same sound, with another tone, means two different things |
02:25.37 | jontow | ladybug; play with it for a bit.. it happens easily after a few minutes of changing settings :) |
02:25.44 | |Ladybug| | our co dun have firewall |
02:25.45 | SpaceBass|AWAY | yeah the intonations are crazy |
02:25.53 | |Ladybug| | ok... i am going to the website n read a bit |
02:26.07 | |Ladybug| | will disturb u guys again if i dun understand |
02:26.13 | jontow | :) |
02:26.20 | fugitivo | japanese has easy sounds |
02:26.24 | Nugget | yeah |
02:26.24 | |Ladybug| | sorry... just started to work.. so a lot of thing i dun understand |
02:26.45 | Nugget | the hardest thing with japanese is fighting the tendancy to inflect. |
02:26.53 | harryvv | I say to my wife "ya lublu teba darinaga" |
02:27.23 | harryvv | Which in russian means "I love you much precios" |
02:27.31 | harryvv | precious |
02:27.41 | SpaceBass|AWAY | i want to learn russian and japanese next... |
02:27.43 | SpaceBass|AWAY | on my list |
02:27.49 | harryvv | rusisan is on my list |
02:28.05 | jontow | that ain't no frodo |
02:28.05 | harryvv | As soon as I get my hamradio station up...then i can start learning other languages. |
02:28.10 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
02:28.24 | darwin_35 | night |
02:28.26 | *** part/#asterisk darwin_35 (user@cas-11.ftsm-noc.mtlnk-129.valuelinx.net) |
02:28.29 | jontow | but anyway --- to do the background research via IRC once again.. anyone played with SMDI and asterisk? |
02:28.40 | harryvv | btw, voip took off on hamradio a few years ago. www.irlp.org |
02:28.42 | jontow | specifically.. the Sun Netra-based Coppercom soft-switches? |
02:28.43 | DarthClue | jontow: define SMDI. |
02:29.02 | jontow | ~SMDI |
02:29.03 | jbot | SMDI serial and TCP session protocol as APE library. URL: http://www.voxilla.org/projects/projsmdi.html |
02:29.03 | jontow | (?) |
02:29.13 | DarthClue | nope. |
02:29.16 | jontow | now .. admittedly, APE libraries mean absolutely nothing to me.. |
02:29.31 | jontow | but ... i did do a quick rewrite in C vs. C++ |
02:29.58 | jontow | and it still didn't help, i guess the real issue is.. the serial<-->ethernet converter that the coppercom softswitches use to pass SMDI .. just wanted to know if anyone had seen one and/or made sense of it |
02:30.10 | jontow | i suspect not.. pretty new stuff as far as telephony goes [on the market] |
02:31.33 | jontow | alternately.. anyone know of a cheap-yet-effective PCI-X board that gives one regular RS232 serial port[s] ? |
02:31.41 | jontow | (that works under linux...) |
02:32.18 | *** join/#asterisk Derkommissar (~total@66.64.215.6.nw.nuvox.net) |
02:32.34 | Derkommissar | anyone has any cool asterisk wallpapers :-) like a 3d asterisk ? |
02:33.17 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
02:33.25 | fugitivo | it's time to do asterisk hats and tshirts |
02:33.40 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
02:34.08 | Mavvie | somebody here with experience of running asterisk on a sparc with linux on it? |
02:34.23 | shido | it has been done |
02:34.26 | shido | it can be done |
02:34.32 | Mavvie | that's a good start! |
02:34.34 | jontow | http://www.softio.com/ic0607kb.htm |
02:34.35 | Mavvie | thanks. |
02:34.37 | jontow | guess that fits my needs |
02:34.37 | jontow | :) |
02:35.13 | jontow | (weeee) |
02:35.23 | jontow | i do'nt even know if that damned thing boots anymore [once again a fire victim] |
02:35.44 | MikeJ[Laptop] | shido, did you get your unicall stuff working? |
02:35.53 | jontow | i wonder if linux even supports sparc32 anymore |
02:36.11 | jontow | if not; netbsd works great on that box.. haven't tried 2.0.2 though |
02:36.20 | jontow | i think i should find a new HDD for it and bring it to work to install upon |
02:40.10 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
02:40.57 | fugitivo | anyone is using broadvoice? |
02:42.35 | Sedorox | fugitivo: lots |
02:42.43 | fugitivo | is it good? |
02:42.55 | Sedorox | I don't, I know others that do |
02:42.58 | Derkommissar | anyone have a cool asterisk wallpaper ? like a 3d asterisk or something ? |
02:47.51 | SwK | does david of res_php hang out in here? |
02:48.07 | SwK | or has anyone gotten it to compile? |
02:48.09 | jontow | swk; AgiNamu, i believe, wrote an alternative, i believe. |
02:48.14 | jontow | lemme see if i can dig it up :) |
02:48.40 | SwK | I just saw that on sineapps |
02:48.43 | jontow | http://jontow.hijacked.us/~jontow/asterisk_php.tgz |
02:48.57 | jontow | he pointed the channel at it for feedbakc |
02:49.05 | SwK | werd |
02:49.06 | jontow | if you have a good use for it, i urge you to do so; it'll only help |
02:49.06 | jontow | :) |
02:49.18 | jontow | thats the copy i grabbed from his site; which i fail to remember now |
02:49.47 | jontow | David Eder .. that his name? |
02:50.12 | SwK | yeah |
02:50.18 | SwK | I have a copy from his website |
02:50.33 | SwK | I'm having compile issues |
02:51.04 | jontow | aha |
02:54.11 | fugitivo | unlimited calls to argentina |
02:54.16 | fugitivo | only for 25 dollars |
02:54.43 | fugitivo | i pay more for my pots line, and it's not unlimited |
02:55.46 | wunderkin | so what would a res_php accomplish exactly? |
02:55.49 | *** join/#asterisk _0_0_ ([U2FsdGVkX@96837297623a2575.session.tor) |
02:57.57 | *** join/#asterisk Ahmuck (~chatzilla@24.225.23.102) |
02:58.14 | NewSole | so where is twisted sister |
02:58.56 | SwK | wunderkin: res_php is to asterisk as mod_php is to apache |
02:58.58 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
02:59.09 | tengulre | does asterisk support FreeBSD? |
02:59.25 | wunderkin | yeah, i dont know how that works either |
02:59.29 | wunderkin | tengulre, yeah |
02:59.49 | tengulre | wunderkin,thanks :), I'm beginner! |
02:59.52 | SwK | tengulre: works fine on FBSD but dont bet on hardware support thats worth a damn |
03:00.16 | tengulre | SwK,thank u! |
03:00.58 | wunderkin | i guess somehow it can do the php within asterisk instead of having to spawn a php process for each thing.. or something |
03:01.21 | tengulre | I agree u! |
03:01.42 | *** join/#asterisk Junk-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
03:02.43 | tengulre | I like asterisk, but the asterisk can't support more hardware! |
03:04.44 | fugitivo | anyone using broadvoice? |
03:06.01 | tengulre | fugitivo, yes im |
03:06.09 | fugitivo | tengulre: what plan? |
03:07.30 | tengulre | what kind of broad are u use? |
03:07.52 | fugitivo | i don't use it, i want to try it |
03:08.31 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
03:08.50 | jontow | there an easy way to hangup a channel? softhangup that is? ie. not destroying the channel or anything like that? |
03:08.58 | jontow | ah.. stupid question |
03:09.16 | jontow | :) |
03:09.23 | tengulre | plaint!!:) |
03:09.25 | *** join/#asterisk Qorky (~Pooa@dip-202-72-131-243.wa.westnet.com.au) |
03:09.49 | Qorky | can anyone here help me with isdn and asterisk. I cant get chan_capi to load |
03:09.50 | Qorky | please |
03:11.15 | shido | 4th quarter |
03:15.37 | jontow | greeaaaat; upstream dialplan at the telco switch is failing so i can't call from one of my PBXes to the other |
03:15.53 | *** join/#asterisk freat (~freat@h-69-3-229-184.chcgilgm.covad.net) |
03:16.49 | *** join/#asterisk da_monumental_1 (~da_monume@rrcs-24-172-102-197.midsouth.biz.rr.com) |
03:18.26 | jontow | wtf |
03:18.33 | jontow | Zap/23-1 answered IAX2/intnic@intnic/1 |
03:18.40 | jontow | (am i missing something here or is this just messed up?) |
03:19.01 | jontow | using a zaptel channel for a 100% voip call? |
03:19.45 | Nuxi | Hmmm... looks like people been talking about me when I was gone... |
03:19.47 | jontow | actually .. 2 Zap/N channels for that call |
03:19.50 | jontow | thats messssssed up |
03:20.36 | Nuxi | SwK, jontow, compile problems with res_php? |
03:20.45 | jontow | aha !5!#^* it wasn't AgiNamu at all |
03:20.53 | jontow | i knew i'd confused someone :( [and figured it was myself] |
03:21.22 | jontow | SwK was having the issues |
03:21.37 | jontow | i was just trying to offer the absolutely vague insight that i had |
03:21.38 | jontow | :) |
03:22.02 | Nuxi | current code is at http://eder.us/projects/asterisk_php/ needs a few tweaks to work with cvs-head. |
03:24.17 | *** join/#asterisk Ahmuck (~chatzilla@24.225.23.102) |
03:24.33 | Nuxi | jontow, there's also res_bf at http://pastebin.ca/15414 for your amusement. (brainfu**) |
03:24.42 | jontow | ahahahahahah |
03:24.53 | jontow | brilliant; yet so damned twisted :) |
03:24.59 | Hogie | my head hurts, I was at a sausage fest all day |
03:25.29 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
03:35.08 | *** part/#asterisk Cresl1n (~matt@216.207.245.23) |
03:35.43 | SwK | ok... NOTE TO SELF: **DO NOT** walk of the front door with a blade in your hand when cops are rolling up on you |
03:36.01 | Sedorox | lol |
03:36.38 | jontow | good call |
03:36.56 | Nuxi | SwK, did you ever get res_php to compile? |
03:37.09 | Corydon76-home | ~res_brainfuck |
03:37.11 | jbot | methinks res_brainfuck is at http://www.tubgirl.com/ |
03:37.19 | SwK | Nuxi: nopw... but I'm trying against head |
03:37.35 | SwK | Sedorox: that shit just happened to me at my house |
03:37.50 | Sedorox | :( |
03:38.20 | Sedorox | why were the cops there.. and why did you happen to have a knife... wait.. I dun wanna know |
03:38.28 | jontow | http://pastebin.ca/15414 <-- corydon76; go there instead. |
03:38.40 | SwK | so a kid about 15yrs old comes beating on my door... doods holding my female cousin nextdoor and is possibly raping her.... |
03:38.50 | SwK | hilarity ensues |
03:38.58 | Nuxi | stuff moves around in head. http://pastebin.ca/15473 has the correct include paths. |
03:39.23 | Sedorox | :/ |
03:39.57 | Corydon76-home | jontow: what, no res_ook? ;-) |
03:39.59 | *** join/#asterisk ptiggerdine (~ptiggerdi@c210-49-98-194.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM] |
03:39.59 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) [NETSPLIT VICTIM] |
03:39.59 | *** join/#asterisk newl (~newlook@203-59-112-225.dyn.iinet.net.au) [NETSPLIT VICTIM] |
03:39.59 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM] |
03:41.34 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
03:41.55 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
03:53.40 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
03:53.40 | *** topic/#asterisk is Asterisk: The Open Source PBX || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com |
03:53.43 | NewSole | setting up peerin and want to see if rings phone |
03:53.48 | *** join/#asterisk roywish (~roy_wish@202.155.41.20) |
03:53.59 | NewSole | if I try call inside net wit wont go out |
03:54.38 | DarthClue | jontow: wikipedia may have more content but it isn't asterisk related. I am working on a creating a copy of voip-info.org that would be more scalable and less likely to die. |
03:54.38 | roywish | hi all |
03:55.07 | postel | DarthClue: a mirror or from scratch? |
03:55.07 | roywish | anyone have succesfull install oh323 |
03:55.13 | roywish | please help me |
03:55.33 | DarthClue | postel: combination. |
03:55.35 | jontow | cool. |
03:55.57 | sivana | exit |
03:56.00 | NewSole | DarthClue... can you try it for me |
03:56.01 | sivana | ack |
03:56.07 | DarthClue | we need docs that are more specific that we can point newbies to so that we don't keep answering the same questions over and over again. |
03:56.34 | roywish | anyone have succesfull install oh323 |
03:56.35 | roywish | please help me |
03:56.55 | postel | DarthClue: why dont you work a bit on the asterisk handbook? its been left outdated for some time now |
03:57.16 | DarthClue | NewSole: not sure that i can. my box isn't exactly in a position to be tweaked at the moment. |
03:57.49 | NewSole | ok... |
03:57.58 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
03:58.23 | roywish | anyone have succesfull install oh323 |
03:58.24 | roywish | please help me |
03:58.55 | DarthClue | postel: asterisk handbook does need work. I have looked at that and passed it off as being out of date when compared to the wiki. |
04:00.50 | roywish | hi all |
04:00.53 | NewSole | Anyone fell helpfull to make a test call for me... |
04:01.16 | postel | NewSole: people asked you before, scroll up, WHAT kind of test call? |
04:01.44 | NewSole | only one did DarthClue |
04:01.52 | *** part/#asterisk roywish (~roy_wish@202.155.41.20) |
04:01.55 | postel | how many you wanted? a test pool? |
04:02.21 | NewSole | I just need someone to call a number and see if it rings phone |
04:02.36 | NewSole | see if guy has peer hooked up right |
04:02.38 | postel | give the damn number then, and someone might call |
04:02.55 | freat | mmm... phones |
04:02.57 | NewSole | 5194884245 |
04:02.57 | postel | you want a SLA signed before that? |
04:03.35 | NewSole | thnx |
04:03.40 | freat | np |
04:05.03 | Sedorox | night |
04:05.27 | twisted | html documentation? why? can't read .txt files? |
04:06.00 | DarthClue | just to make you ask why. No other reason than to make twisted ask why. |
04:06.05 | Nuxi | The problem is that documentation has to be written after a release because trying to keep up with head is chaos. |
04:06.34 | twisted | DarthClue, nice attitude. that'll get ya somewhere |
04:08.21 | DarthClue | twisted, i'm joking. It would be nice to have html docs (even lynx could read them) that work similar to the wiki included with the CVS-HEAD download that would make it easier for newbies to get going. |
04:08.29 | *** join/#asterisk SleepyCow (SleepyCow@wnpgmb09dc1-79-72.dynamic.mts.net) |
04:08.39 | twisted | i see |
04:08.55 | DarthClue | it's just an idea since the wiki is down so much. |
04:09.29 | fugitivo | damn broadvoice |
04:09.33 | fugitivo | i can't make any call |
04:09.38 | SleepyCow | Hello all; I have some questions regarding the business end of providing VOIP services. Does anyone have a few miniutes or some links to resources? |
04:11.15 | SleepyCow | anyone? |
04:14.05 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
04:14.14 | SleepyCow | hello (echo) ? |
04:14.33 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
04:14.51 | Nuxi | hmmm. echo cancelation must not be working. ;) |
04:15.01 | SleepyCow | har. |
04:15.18 | *** part/#asterisk hassler (~hassler@cpe-65-31-36-179.woh.res.rr.com) |
04:15.20 | SleepyCow | Seriously, I just need a few questions answered that noone seems to be abel to answer.... |
04:15.31 | SleepyCow | Basicly , on the provider side: how do you get into the PSTN |
04:15.35 | SleepyCow | and how much does it cost? |
04:15.48 | `Sauron | You get a smart trunk from the *LEC |
04:16.24 | SleepyCow | now those have a fixed number of 'lines' , eg connections to the pstn simultaneously? |
04:16.26 | SleepyCow | right? |
04:16.58 | `Sauron | Yes. |
04:17.02 | `Sauron | 23, to be exact |
04:17.03 | `Sauron | per trunk |
04:17.34 | SleepyCow | so, with one trunk , i can have at most 23 incomming + outgoing calls to/from the pstn |
04:17.52 | *** join/#asterisk mxmasster (~maxc@216.152.251.69) |
04:17.53 | mxmasster | hi all |
04:17.57 | SleepyCow | greetings |
04:17.58 | `Sauron | Correct. |
04:18.14 | *** join/#asterisk nitram (foo@superblob.com) [NETSPLIT VICTIM] |
04:18.27 | SleepyCow | now that doesnt mean i can have only up to 23 phone numbers: i could have say 35 lines, since most wont always be in use |
04:18.31 | SleepyCow | right? |
04:18.37 | fugitivo | broadvoice doesnt work |
04:19.03 | mxmasster | HA |
04:19.14 | SleepyCow | wrong? |
04:19.15 | SleepyCow | mabee? |
04:19.23 | `Sauron | Correct. |
04:19.37 | SleepyCow | okay; now how do i find out how much a trunk is from the ilec |
04:19.41 | `Sauron | At work, we have 65 numbers allocated to a single T1/trunk |
04:19.43 | `Sauron | Call them up. |
04:19.59 | SleepyCow | sso at work you hcan have 20odd maximum simultaneous calls |
04:20.07 | `Sauron | yeah |
04:20.10 | SleepyCow | can someone give me a general price range in whcih to look |
04:20.16 | `Sauron | 23 |
04:20.22 | `Sauron | Nope. |
04:20.25 | `Sauron | It varies from lec to lec |
04:20.27 | Ariel_ | we have a pri with 100 did's asigned to it. just love it. we have only maxed it's 23 channels once. |
04:20.39 | SleepyCow | Okay, sounds good |
04:20.47 | SleepyCow | I have a supplier willing to give me did's at $5 / each |
04:20.49 | `Sauron | And there's bulk discounts, grandfather discounts, competetive discounts, cash cow discounts, etc... |
04:20.51 | SleepyCow | which seems cheap |
04:21.06 | Ariel_ | x/o here provides a pri with 23 channels for around $ 549.00 per month plus $ 3.20 for 20 did's. |
04:21.06 | SleepyCow | how much would it costs to do it myself is what im trying to find out |
04:21.45 | SleepyCow | so roughly $25 per 'real line' |
04:21.50 | Ariel_ | 100 did's are prices at 3.00 per group of 20. |
04:22.02 | *** join/#asterisk cmk (~cmk_@p54A3E583.dip.t-dialin.net) |
04:22.06 | `Sauron | se, now our prices are different. :) |
04:22.08 | SleepyCow | how much does it costs for the actual calls over the line? what is free? what is local? how do you get ld? |
04:22.12 | `Sauron | see above discount list |
04:22.21 | `Sauron | $700-some for the circuit |
04:22.30 | `Sauron | but I think we got the numbers for free |
04:22.35 | Ariel_ | all inbound for us are free and all local calls are free. |
04:22.35 | SleepyCow | ? |
04:22.39 | `Sauron | 'course, we already have an entire NXX from them |
04:22.52 | `Sauron | 324-xxxx is all ours |
04:23.00 | wunderkin | wow |
04:23.08 | SleepyCow | okay well we aremuch smaller than that |
04:23.14 | wunderkin | you whore :D |
04:23.28 | `Sauron | hum, not really.. we have ~10k employees |
04:23.32 | wunderkin | ah |
04:23.36 | `Sauron | 7-8k, actually |
04:23.50 | SleepyCow | so how do i get ld? |
04:23.56 | `Sauron | call them up |
04:23.58 | SleepyCow | how does that work? |
04:23.59 | `Sauron | once again |
04:24.05 | Ariel_ | 7 to 8 thousand employee's large co for me. |
04:24.07 | SleepyCow | you get ld from the ilec? |
04:24.14 | `Sauron | Depends |
04:24.18 | `Sauron | who do you want LD service from |
04:24.24 | `Sauron | I think we get it from MCI here |
04:24.25 | SleepyCow | whomever is cheap ;) |
04:24.27 | `Sauron | or at least used to |
04:24.32 | `Sauron | whatever MCI is called now |
04:24.43 | Ariel_ | mci |
04:24.52 | `Sauron | Unless SBC managed to give us a better deal |
04:25.01 | `Sauron | Hell if I know, that's all telecom stuff, not my worry. :) |
04:25.12 | SleepyCow | thats what i need to leanr about :( |
04:25.23 | fugitivo | damn, this really pissed me off |
04:25.24 | `Sauron | well |
04:25.32 | `Sauron | start taking LEC sales drones out to lunch |
04:25.33 | Ariel_ | fugitivo, what does? |
04:25.38 | fugitivo | Ariel_: broadvoice |
04:25.39 | `Sauron | or, rather.. have them take you out to lunch |
04:25.46 | fugitivo | Ariel_: i can't make any call |
04:25.48 | `Sauron | BV having problems tonight? |
04:25.51 | Ariel_ | ahh the up and down service |
04:25.51 | `Sauron | Hum di dum |
04:25.59 | fugitivo | i don't know, i just signed in |
04:26.07 | fugitivo | and i only can call the technical service |
04:26.12 | SleepyCow | how do i get the list of xLEC's in an area? |
04:26.13 | Ariel_ | sip behind a nat? |
04:26.17 | fugitivo | no |
04:26.34 | wunderkin | yeah xo wanted to take me to lunch.. im like.. dude its just a quote hehe |
04:26.35 | Ariel_ | did you put insecure=very |
04:26.44 | fugitivo | i call a number, and the voice says "we're sorry, we can't establish the call right now" or something like that |
04:27.10 | Ariel_ | all circuits are busy please try your call later. |
04:27.21 | `Sauron | hum |
04:27.24 | `Sauron | di dum |
04:27.25 | fugitivo | "we're sorry, you're call cannot be completed at this time, please hung up and bla bla bal" |
04:27.32 | `Sauron | Sorry |
04:27.33 | `Sauron | works fine here |
04:27.34 | *** part/#asterisk tengulre (~tengulre@61.185.238.166) |
04:27.42 | `Sauron | just called my cell from BV |
04:27.42 | wunderkin | think im going with sprint for ld pri t1 here |
04:27.54 | fugitivo | then should i wait or should i complain? |
04:28.06 | `Sauron | complain |
04:28.17 | fugitivo | great, the dtmf doesnt work |
04:28.21 | fugitivo | :/ |
04:28.24 | *** join/#asterisk sbingner (~thanotos@adsl-699.flex.com) |
04:28.55 | sbingner | hey I got more freezing up with my zaptel card, had been very stable until recently |
04:29.15 | sbingner | not worth opening a bug report, I dont have enough info :b |
04:29.41 | `Sauron | hum |
04:29.41 | Ariel_ | wunderkin, mistake |
04:29.46 | fugitivo | oh, only inband |
04:29.50 | `Sauron | I get to multihome-nat my machines at home |
04:29.55 | `Sauron | how fun does that sound |
04:30.18 | wunderkin | Ariel_, what kind of problems have you had with them |
04:30.21 | SleepyCow | so how do i find a list of all the lec for an area? |
04:30.33 | `Sauron | hum |
04:30.38 | `Sauron | wonder if nether.net has a lec list |
04:30.41 | `Sauron | puck.nether.net |
04:30.43 | Ariel_ | wunderkin, from billing problems, poor data lines and just plain stupid support people nothing. |
04:30.44 | `Sauron | go look |
04:30.57 | wunderkin | Ariel_, was that you that replied on the list a day or two ago about sprint? |
04:31.11 | Ariel_ | I did reply once. |
04:31.39 | wunderkin | ok |
04:31.41 | fugitivo | grrrrrrrrr |
04:32.10 | Ariel_ | fugitivo, in the morning I just got an email that a customer wants me to setup bv for him. Argh. |
04:32.40 | fugitivo | Ariel_: lol, i'm calling support number and dtmf doesnt work |
04:32.42 | wunderkin | Ariel_, did you just have data with them? |
04:33.02 | SleepyCow | sauron, the NOC list? |
04:33.27 | `Sauron | Umm |
04:33.27 | `Sauron | no |
04:34.10 | Ariel_ | wunderkin, no just t1 phone service. |
04:34.20 | Ariel_ | they way to expensive for data |
04:34.34 | SleepyCow | where then? |
04:34.35 | wunderkin | oh |
04:35.08 | wunderkin | Ariel_, what kind of billing problems? dont have the email anymore |
04:35.10 | Ariel_ | we have a support department that needed to call via modem to take over systems. They were not able to via sprint. |
04:35.26 | *** join/#asterisk techie (gus@antibala.com) |
04:35.33 | Ariel_ | we finally switch ld to WorldCom for the support department. |
04:36.20 | Qwell | Offtopic...would anyone happen to know how the intel 915g chipset and the ADD2 cards works? I'm trying to figure out how it works, but there is very little info available on google... |
04:36.45 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
04:38.52 | jdv79 | how can make samples fail? haha - this is starting to get hilarious. |
04:39.42 | drumkilla | because you're not teh root! |
04:40.14 | jdv79 | i am |
04:40.17 | jdv79 | make: *** No rule to make target `/usr/include/asterisk/version.h', needed by `cli.o'. Stop.make: *** No rule to make target `/usr/include/asterisk/version.h', needed by `cli.o'. Stop. |
04:40.53 | Qwell | jdv79: it isn't a cvs upgrade, is it? |
04:41.12 | drumkilla | make clean! |
04:41.15 | jdv79 | no |
04:41.19 | jdv79 | its a clean checkout from head |
04:41.23 | Qwell | drumkilla: I was getting to that. :p |
04:41.26 | `Sauron | I love how there was make clean |
04:41.31 | `Sauron | then there came make distclean |
04:41.37 | `Sauron | then there's make spotless |
04:41.39 | Qwell | mrproper? |
04:41.43 | jdv79 | i'm just thinking today is cursed |
04:41.45 | `Sauron | I'm waiting for make spitshine |
04:41.54 | Qwell | `Sauron: give me 5 minutes |
04:41.56 | jdv79 | maybe if i go to sleep everything will work tomorrow |
04:42.00 | jdv79 | :) |
04:42.20 | Qwell | `Sauron: rm -rf . && cvs co asterisk |
04:42.26 | `Sauron | hehe |
04:42.38 | `Sauron | make antibacterial? |
04:42.48 | `Sauron | <PROTECTED> |
04:42.52 | `Sauron | hehn |
04:43.31 | fugitivo | oh well |
04:43.38 | fugitivo | who is using broadvoice with asterisk? |
04:43.48 | fugitivo | dmtf is not working for me |
04:44.24 | jdv79 | the CTO of broadvoice used to be my boss |
04:44.30 | jdv79 | for a very short period, thank god |
04:44.57 | fugitivo | i signed up today, i'm going to cancel the service tomorrow |
04:45.04 | jdv79 | i don't doubt it |
04:45.18 | fugitivo | maybe i'll not wait until tomorrow |
04:45.50 | jdv79 | what's the deal? |
04:46.32 | fugitivo | i signed up, i can't make any call (when i make a call, it says "we're sorry, we can't complete your call at this moment, please hangup and ...") |
04:47.02 | fugitivo | and dtmf doesnt work, but maybe that's my fault |
04:51.11 | jdv79 | msg me your # |
04:51.22 | jdv79 | maybe i can get something done about it |
04:53.00 | `Sauron | I'm using it |
04:53.08 | `Sauron | fugitivo: you need ot use inline |
04:53.11 | jdv79 | my contact has no attention span by the way |
04:53.34 | fugitivo | i'm using inline |
04:54.14 | jdv79 | is that your BV #? |
04:54.22 | fugitivo | yes |
04:57.29 | jdv79 | best way to deal with it is through BV support |
04:57.46 | fugitivo | yes, buy i can't access support because dtmf isnt working for me, lol |
04:58.02 | jdv79 | can't you get to another phone? |
04:58.18 | fugitivo | it seems it works only to another BV # |
04:58.18 | jdv79 | good god man you think VOIP is reliable?! |
04:58.51 | fugitivo | `Sauron: could i try to call your BV #? |
04:59.46 | `Sauron | Umm, sure |
04:59.47 | `Sauron | 'sec |
05:01.19 | *** join/#asterisk kisu (~daniel@3ffe:831f:cbe9:9a0a:0:fbcc:25d9:5f26) |
05:01.34 | `Sauron | check /msg |
05:03.21 | `Sauron | try forcing it to alaw/ulaw |
05:03.43 | fugitivo | what about no? |
05:03.45 | fugitivo | now |
05:03.49 | fugitivo | no error, right? |
05:03.49 | `Sauron | no errors |
05:03.52 | fugitivo | ok |
05:03.57 | `Sauron | yup |
05:04.02 | fugitivo | i'll try the support line now :) |
05:05.11 | `Sauron | try other numbers |
05:05.15 | `Sauron | maybe that's what was causing problems |
05:05.24 | fugitivo | no, still not working |
05:05.32 | fugitivo | and dtmf isnt working neither |
05:05.38 | `Sauron | hum |
05:05.49 | Qorky | can anyone here help me with isdn and asterisk. I cant get chan_capi to load |
05:05.51 | Qorky | please |
05:06.30 | X-Rob | qorky, you're usually better asking in the morning |
05:06.32 | X-Rob | the yanks are awake then |
05:06.36 | X-Rob | but what's your problem? |
05:06.44 | jdv79 | the yanks not suffering from insomnia that is |
05:07.00 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
05:07.15 | *** part/#asterisk tengulre (~tengulre@61.185.238.166) |
05:07.17 | X-Rob | Well yeah. __Except__ the yanks suffering from insomnia |
05:09.40 | Qorky | well X-Rob. im just a bit lost. cant seem to load everything right. |
05:09.50 | fugitivo | its working |
05:09.56 | Qorky | i've tried different drivers and still cant seem to get anywhere. |
05:10.13 | Qorky | im using a fritz pci card. |
05:10.32 | *** join/#asterisk Ayano (~erik_leee@dsl093-034-050.snd1.dsl.speakeasy.net) |
05:10.54 | Ayano | has anyone authenticated to primus network before? |
05:11.08 | fugitivo | well, i can call to any number now, but dtmf is not working yet |
05:11.22 | X-Rob | OK. So Quorky, you can't load the module, asterisk crashes when you try? |
05:11.30 | X-Rob | or what exactly is happening |
05:11.48 | Qorky | well i've tried lots of ways, so its hard to explain. |
05:11.54 | wunderkin | fugitivo, are you using ulaw and inband dtmf? |
05:11.56 | Qorky | im currently trying on a 2.6.12 kernel |
05:12.18 | Qorky | i have the following loaded |
05:12.18 | Qorky | root@asterisk:/tars# lsmod |
05:12.18 | Qorky | Module Size Used by |
05:12.18 | Qorky | hisax_fcpcipnp 12416 0 |
05:12.18 | Qorky | hisax_isac 10388 1 hisax_fcpcipnp |
05:12.19 | Qorky | hisax 115104 2 hisax_fcpcipnp,hisax_isac |
05:12.21 | Qorky | isdn 89120 1 hisax |
05:12.33 | X-Rob | looks good |
05:12.38 | Qorky | so the kernel pcpcipnp driver is loaded. as far as i know thats what i want. |
05:12.58 | Qorky | so yeah.. i have all the /dev/capi/blah stuff. that all looks good. |
05:13.16 | Qwell | what actually happens? |
05:13.26 | Qorky | when i start asterisk it has a cry. |
05:13.39 | Qorky | let me see |
05:13.52 | Qorky | == Parsing '/etc/asterisk/capi.conf': Found |
05:13.52 | Qorky | Jun 22 13:04:47 NOTICE[2557]: chan_capi.c:2636 load_module: CAPI not installed! |
05:14.17 | Qwell | Did you install capi? |
05:14.19 | Qorky | and i've made and installed chan_capi-0.3.5 |
05:14.48 | Qorky | and added the lines in the modules.conf file |
05:14.49 | X-Rob | lsmod | grep -i capi |
05:15.26 | Qorky | lsmod | grep -i capi = blank |
05:15.33 | Qorky | but. |
05:15.34 | Qorky | CAPI Subsystem Rev 1.1.2.8 |
05:15.34 | Qorky | capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) |
05:15.34 | Qorky | capifs: Rev 1.1.2.3 |
05:15.34 | X-Rob | you haven't loaded the capi module |
05:15.37 | X-Rob | Ah |
05:15.38 | Qorky | tis from dmesg |
05:15.42 | X-Rob | you compiled it in, rather than as a module? |
05:16.00 | Qorky | yar. if i dont. and i load the modules i never get the /dev/capi stuff. |
05:16.07 | Qorky | but the modules load fine. |
05:16.32 | Qorky | you reccommend using the module eh ? |
05:16.45 | X-Rob | Doesn't make a difference |
05:17.09 | X-Rob | Um |
05:17.10 | Qorky | hmm ok. |
05:17.11 | Qorky | <PROTECTED> |
05:17.11 | Qorky | Jun 22 13:07:59 NOTICE[2825]: chan_capi.c:2636 load_module: CAPI not installed! |
05:17.11 | Qorky | Jun 22 13:07:59 WARNING[2825]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returni ng -1 |
05:17.21 | Qorky | the load_module failed sounds like my prob. |
05:17.26 | Qorky | but dunno what to do. |
05:17.36 | X-Rob | Hrm. |
05:17.53 | Qorky | also. is there another way i can see if the drivers are ok? like capiinfo etc ? |
05:18.02 | X-Rob | yeah |
05:18.06 | Qorky | because i cant get capiinfo and capiinit to work. i do it manually |
05:18.41 | Qorky | also, i dont need a special version of asterisk or anything do i? because im just running the 1.0.7 version. |
05:18.50 | X-Rob | I don't have a fritz card, so I've never tried it |
05:18.57 | X-Rob | It's better to run HEAD |
05:19.09 | Qorky | but it should work no ? |
05:19.09 | X-Rob | Stable == not moving, it doesn't mean 'won't crash' |
05:19.30 | X-Rob | I'd start with head |
05:19.33 | X-Rob | what distro are you using? |
05:19.39 | Qorky | slackware 10.0 |
05:19.40 | jdv79 | it implies that its been looked at though and it has less of a chance to crash |
05:20.05 | Qorky | i can change distro if necisary. but I'm very used to slack. |
05:20.20 | X-Rob | That should be fine |
05:20.43 | jdv79 | isn't slack the fBSD lovers linux? |
05:20.50 | Qorky | it ships with 2.4.26 but i've upgraded and got further with 2.4.12. |
05:20.50 | X-Rob | nah |
05:21.14 | X-Rob | http://www.aussievoip.com.au/wiki-Asterisk-HEAD |
05:21.15 | drray | HEAD gets changed and broken from time to time, STABLE should work |
05:21.18 | X-Rob | Qorky - look at that. |
05:21.22 | drray | I run HEAD |
05:21.25 | X-Rob | head works at the moment |
05:21.40 | X-Rob | sip was borked over the weekend, but that's the first time it's been broken in ages |
05:22.38 | *** join/#asterisk DrRighteous (~DrRighteo@68.199.175.49) |
05:22.39 | X-Rob | jdv - anyone who runs slackware has a clue |
05:22.48 | X-Rob | as it's not a distribution for the weenies. |
05:24.41 | Qorky | i have a little clue. but not that much |
05:24.48 | Qorky | hecne why im stuck |
05:24.52 | Damin | Wow.. |
05:25.03 | Damin | Pulling up carpeting sucks ass... |
05:25.48 | drray | I use fedora core 3, so I can save my clue for * and not setting up X windows |
05:31.21 | Qorky | meh. i'll keep trying. |
05:31.28 | Qorky | tah for trying to help |
05:35.06 | X-Rob | qorky - got a free world dialup accoutn set up? |
05:37.12 | der[mat] | g00d morning * |
05:40.46 | twisted | wheee |
05:41.45 | *** join/#asterisk jerlique (jerlique@lnk2.adl.adsl.esc.net.au) |
05:43.04 | Ayano | has anyone used primus before? |
05:44.46 | twisted | Ayano, yeah, they were my lienholder for my car |
05:45.06 | DrRighteous | any user-mode linux guru's around? |
05:45.26 | X-Rob | ...you're trying to run asterisk under uml? |
05:45.45 | DrRighteous | X-Rob: yeah, a number of virtual * boxes |
05:46.11 | DrRighteous | X-Rob: all dual-xeon systems |
05:47.02 | X-Rob | what's not working? |
05:48.22 | DrRighteous | X-Rob: Debian box, UML is loading, using TUN/TAP, but UML networking can only ping/reach host machine, not public internet. |
05:48.28 | DrRighteous | X-Rob: driving me crazy |
05:49.31 | X-Rob | How many machines do you have on your LAN, and, is the UML machine the internet gateway? |
05:50.45 | DrRighteous | large number of machines, have a full class C, .1 is a Cisco 12000GSR and is gateway/ |
05:51.41 | X-Rob | http://user-mode-linux.sourceforge.net/networking.html |
05:51.43 | X-Rob | Have you read that? |
05:51.48 | DrRighteous | yes... |
05:51.55 | X-Rob | and you're using tun/tap? |
05:52.57 | DrRighteous | X-Rob: yes I am... this is my first attempt at tun/tap.. but once again the IP I assign the UML eth0 as a public IP can ping the hosts public IP, but can't ping any other box on the LAN |
05:53.11 | DrRighteous | no NAT/private IPs on LAN either |
05:53.12 | *** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com) |
05:53.13 | RoyK | anyone that knows how i can use soxmix to mix two audio files into one stereo file with each input file as left and right channel? |
05:53.58 | X-Rob | so what's your command line parameter for the uml kernel? |
05:55.15 | RoyK | asterisk under uml??? |
05:55.23 | DrRighteous | RoyK: yes |
05:55.25 | RoyK | under vmware |
05:55.36 | DrRighteous | RoyK: no vmware |
05:55.48 | RoyK | still |
05:56.12 | RoyK | you'll lose the realtime performance in linux by using a virtual machine |
05:56.17 | RoyK | which is bad for telephony |
05:56.44 | cochi | sounds like he's doing virtual server * hosting. then he has not many options |
05:56.47 | RoyK | asterisk on colinux on win2k3 on vmware on linux 2.2! |
05:57.00 | cochi | vmware, uml, virtuozzo, .. |
05:57.09 | DrRighteous | RoyK: ACK WINDOWS...!!!! |
05:57.47 | RoyK | perhaps just chrooting stuff could help |
05:58.36 | DrRighteous | RoyK: I need a degree of security between virtual machines. And I don't think something like VMWare with Windows would be better, the GUI overhead alone! |
05:58.57 | RoyK | i was joking about the windows stuff |
05:59.17 | RoyK | DrRighteous: but how many asterisk installations do you want to do per box? |
05:59.43 | RoyK | and what sort of hardware is this? |
05:59.49 | RoyK | can you isolate one vm per cpu? |
06:00.41 | RoyK | or could you do better with at stack of mini-itx via c3 cards? |
06:01.09 | *** join/#asterisk shido6 (~shido6@d57-87-253.home.cgocable.net) |
06:03.31 | X-Rob | the via boards don't have a very fast FPU |
06:03.39 | X-Rob | and they suck at transcoding |
06:03.45 | X-Rob | as long as you don't plan on doing any, they rock. |
06:04.29 | DrRighteous | RoyK: Dual-Xeon's, 4-8G of ram |
06:04.55 | DrRighteous | RoyK: have a ton of these and don't want a small * install to take over each box. |
06:08.08 | *** join/#asterisk kaldemar (~kaldemar@xdsl-204-1.nblnetworks.fi) |
06:10.35 | RoyK | DrRighteous: i wonder what sort of asterisk installation that will require > 256MB RAM ......... |
06:11.00 | RoyK | my largest one has allocated 5MB |
06:15.19 | |Ladybug| | i m back |
06:17.01 | *** join/#asterisk Kieran2 (~kieran@2.197.221.203.velocitynet.com.au) |
06:21.00 | Kieran2 | Are there any documents on machine hardware requirements for Asterisk servers? |
06:21.07 | Kieran2 | I havnt been able to find anything specific. |
06:23.14 | shido6 | what do you want to do Kieran2 ? |
06:24.20 | Kieran2 | Just a basic system to route maybe 10 voip lines, and 15 - 20 isdn lines. The machine will deal with 3 different incoming numbers, and will need about 10 different voicemail boxes. |
06:24.27 | Kieran2 | It'll need to handle stuff like call queues too.. |
06:24.28 | |Ladybug| | hi all |
06:24.44 | Kieran2 | So i'm thinking a 386 won't quite do it :) |
06:25.06 | Robot_ | has to support MMX |
06:25.47 | |Ladybug| | ne1 know wat kind of processing i need for the origintae n transfer command? |
06:25.49 | *** join/#asterisk gres (~gres@81.222.48.242) |
06:26.06 | |Ladybug| | i developed a vc++ program to interact with asterisk |
06:26.20 | |Ladybug| | for the oroginate command... it crash my prog sumtimes.. |
06:26.33 | |Ladybug| | n i cant even trasnfer |
06:26.39 | |Ladybug| | ne1 can give me some advice? |
06:26.44 | *** join/#asterisk Inv_arp (junya@adsl-3-251-225.mia.bellsouth.net) |
06:27.24 | SwK | |Ladybug|: sounds like an issue w/ the vc++ program |
06:27.49 | *** part/#asterisk shido6 (~shido6@d57-87-253.home.cgocable.net) |
06:28.24 | RoyK | |Ladybug|: doesn't really sound like an asterisk problem |
06:28.30 | SwK | anyone got that res_php module to compile right? |
06:28.53 | jdv79 | use perl instead, you'll thank me later |
06:28.54 | jdv79 | ;) |
06:29.09 | Qwell | real men use res_assembly |
06:29.22 | SwK | hah |
06:29.23 | jdv79 | i didn't see that one |
06:29.35 | RoyK | &drink(0xc0ffee) while ($tired--); |
06:29.36 | SwK | jdv79: i loath perl |
06:29.37 | Qwell | ~lart jdv79 |
06:29.40 | Qwell | there |
06:29.50 | jdv79 | haha |
06:30.08 | RoyK | ~lart SwK for saying bad things about perl |
06:30.22 | Qwell | RoyK: nice |
06:30.51 | SwK | yeah roy |
06:30.54 | SwK | it doesnt work |
06:31.01 | RoyK | it does indeed :) |
06:31.05 | jdv79 | that's an interesting point of view SwK cause PHP basically wants to be perl when and if it ever grows up. plus the irony thickens since perl gave birth to PHP inn a manner of speaking:) |
06:31.23 | SwK | maybe so |
06:31.28 | jdv79 | that's my rant for the night |
06:31.30 | SwK | but i still dont care for perl |
06:31.33 | RoyK | php5 is pretty fscking far from perl |
06:31.34 | jdv79 | ok |
06:32.06 | jdv79 | so is perl6 but lets not go there 'til it exists at least:) |
06:33.48 | RoyK | i remember talking about the soon-to-come perl6 in 1998 |
06:34.07 | jdv79 | yeah, but now we have a robot on the task and his name is autrijus |
06:34.21 | jdv79 | ;) |
06:34.33 | jdv79 | who knew AI existing already |
06:35.01 | JerJer | just hope Autrijus doesn't go crazy and bust into your operating room |
06:35.03 | |Ladybug| | royk... do u know what kind of response would asterisk give back when i click transfer buttib? |
06:35.16 | |Ladybug| | royk.. u have ne idea how to improve the processing so that i dont clash? |
06:37.59 | SwK | |Ladybug|: use non-blocking IO to the manager interface |
06:41.11 | |Ladybug| | oh.. ok |
06:41.41 | |Ladybug| | i m using asterisk manager api n a sjphone |
06:42.03 | *** join/#asterisk whmok (whmok@218.208.97.102) |
06:43.38 | *** join/#asterisk jjg (jjg@adsl-69-226-248-4.dsl.pltn13.pacbell.net) |
06:47.07 | *** join/#asterisk amir (~amir@195.226.9.186) |
06:54.42 | jjg | anyone doin any production VoIP over VSAT? |
06:55.45 | SwK | hah |
06:55.52 | SwK | it works |
07:02.58 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
07:04.11 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
07:04.28 | lehel | hello |
07:07.12 | *** join/#asterisk denon (denon@synapse.subneural.net) |
07:07.12 | *** mode/#asterisk [+o denon] by ChanServ |
07:17.57 | *** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net) |
07:17.58 | *** join/#asterisk bewest (~bewest@h48.204.255.206.cable.srcy.cablelynx.com) [NETSPLIT VICTIM] |
07:17.58 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) [NETSPLIT VICTIM] |
07:21.03 | *** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net) |
07:21.10 | *** join/#asterisk Broesmeli (~broesme@195.65.2.68) |
07:24.05 | *** join/#asterisk kks (~kks@203.115.208.140) |
07:24.42 | *** join/#asterisk stkn (~stkn@stkn-active-pdpc.developer.gentoo) |
07:24.54 | *** join/#asterisk kks (~kks@203.115.208.140) |
07:28.11 | *** join/#asterisk kks (~kks@203.115.208.140) |
07:30.32 | *** join/#asterisk rLg (~umairbari@202.142.189.86) |
07:30.32 | |Ladybug| | what is the mwaning of non-blocking IO? |
07:31.36 | *** join/#asterisk flyingmayo (~flyingmay@ip-66-218-239-201.cableaz.com) |
07:34.44 | jansaell | woip+info seems to have serious problems at the moment |
07:34.59 | jansaell | voip-info.com i meant |
07:35.14 | *** join/#asterisk rLg (~umairbari@202.142.189.86) |
07:35.17 | DA-MAN | hehe voip-info has been damn shitty the past few months |
07:35.21 | DA-MAN | wonder what the problem is |
07:35.28 | JerJer | cpu |
07:35.30 | drray | popularity |
07:35.52 | DA-MAN | JerJer, you run that box? |
07:35.57 | JerJer | fuck no |
07:36.01 | Qwell | DA-MAN: I have a feeling it'd be up if he did |
07:36.23 | DA-MAN | Qwell, wouldn't know . . . |
07:36.24 | DA-MAN | hehe |
07:36.33 | DA-MAN | JerJer, meant no offense by it |
07:36.45 | JerJer | this is irc |
07:36.59 | JerJer | if someone takes offense to anything said in irc they need their head examined |
07:37.09 | drray | I'm offended by that |
07:37.24 | JerJer | good |
07:37.27 | DA-MAN | JerJer, lol |
07:38.33 | DA-MAN | man, I wish I had picked a more reputable iax provider, I went with sixTel and those bastards still haven't activated my toll free #. It's been a month |
07:38.47 | JerJer | fun |
07:39.25 | DA-MAN | pretty good for non-toll free though, i use it for home service |
07:40.08 | jansaell | now vopi-info seems to work again |
07:40.20 | JerJer | someone must have kicked it |
07:40.30 | jansaell | yes i think so to |
07:41.00 | JerJer | i offered to provide collocation space and bandwidth |
07:41.21 | JerJer | but they never took me up on the offer |
07:41.36 | drray | what about just a mirror? |
07:41.39 | DA-MAN | JerJer, you work for a colo |
07:41.46 | drray | like linux from scratch does |
07:41.51 | DA-MAN | JerJer, who is "they" |
07:42.00 | JerJer | wiki's are not friendly to mirror |
07:42.33 | JerJer | someone else in here said they need to run media wiki |
07:43.22 | JerJer | which could get installed on a dual xeon box i just have running distributed.net stuff since it has nothing real to do |
07:43.53 | JerJer | but then data has to be either migrated or created from scratch |
07:43.59 | *** join/#asterisk rLg (~umairbari@202.142.189.86) |
07:44.03 | JerJer | which i ain't doin' |
07:44.05 | JerJer | :) |
07:44.19 | DA-MAN | JerJer, sounds like a bitchin machine, what kinda pipe would you put somethin like that on? |
07:44.36 | JerJer | 66.225.202.65 would be the router |
07:44.55 | JerJer | phat pipe - last i checked we had 21 gigabits per second aggregate |
07:44.58 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
07:45.15 | JerJer | 100BaseT switched to gig-e core |
07:45.18 | DA-MAN | wow, that's awesome |
07:45.38 | JerJer | downtown chicago |
07:46.41 | DA-MAN | 11 unknown.ord.scnet.net (66.225.202.65) 64.765 ms 64.963 ms 64.983 ms |
07:47.01 | JerJer | yeah reverse dns is broken, but decent latency |
07:47.18 | JerJer | and 11 hops is about average |
07:47.33 | JerJer | unfortunately |
07:47.51 | DA-MAN | not too bad |
07:47.53 | DA-MAN | seen worse |
07:47.57 | JerJer | oh yeah |
07:48.07 | JerJer | it gets better to though |
07:48.17 | JerJer | i'm like 8ms from there and i'm in michigan :P |
07:48.36 | DA-MAN | whats scnet do |
07:49.04 | JerJer | that 8ms is basically the pushing limits of the speed of light |
07:49.13 | JerJer | server central |
07:49.22 | Qwell | 38 unknown.ord.scnet.net (66.225.202.65) 662.173 ms 610.391 ms 598.640 ms |
07:49.27 | Qwell | nice |
07:49.30 | JerJer | they run the kick ass ip network |
07:49.34 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
07:49.45 | DA-MAN | sweet |
07:50.03 | JerJer | Jordan and company rock |
07:50.19 | DA-MAN | i do very little network stuff, mostly a system guy myself |
07:50.27 | mini | does asterisk in version 1-0-7 understands ptime in SDP for rtp? |
07:50.29 | JerJer | if you notice sixtel is also very close to that ip address (there is a reason for it :) |
07:50.55 | Qwell | off to bed |
07:51.09 | Qwell | JerJer: the above was fake, btw, in case you cared. :p |
07:52.13 | JerJer | i figured vsat or something |
07:52.19 | DA-MAN | holy crap, you're right. I never did traceroute sixtel |
07:52.31 | JerJer | or some crap in .pk or .iq |
07:53.02 | JerJer | damn!!! |
07:53.02 | JerJer | <PROTECTED> |
07:53.06 | JerJer | it was 8ms |
07:53.10 | JerJer | this is great |
07:53.46 | Qwell | ...my god |
07:53.47 | DA-MAN | <PROTECTED> |
07:53.51 | DA-MAN | this always trips me out!!! |
07:53.52 | Qwell | I get 10ms to my first hop |
07:53.58 | Qwell | SIX FREAKING INCHES AWAY |
07:54.12 | JerJer | atm? |
07:54.17 | Qwell | sadly, no |
07:54.23 | JerJer | odd |
07:54.26 | JerJer | bad cable? |
07:54.27 | Qwell | not sure whats going on with that... |
07:54.34 | Qwell | slow ass router |
07:54.39 | Qwell | downloading at 500k/s currently |
07:54.40 | JerJer | polish the fiber again :P |
07:54.50 | DA-MAN | qwell, wireless? |
07:55.03 | Qwell | <PROTECTED> |
07:55.10 | Qwell | DA-MAN: nope, ethernet |
07:55.16 | Qwell | 5' cable maybe |
07:55.25 | JerJer | cat-3 :) |
07:55.32 | Qwell | nah, its cat5 |
07:55.36 | JerJer | lol |
07:55.37 | DA-MAN | wow, to quote a common slashdot phrase, "that is teh sucks" |
07:56.53 | DA-MAN | so anyone else have the horror of trying to configure the 4port welltech fxo? |
07:56.54 | *** part/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net) |
07:56.59 | *** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net) |
07:57.01 | DA-MAN | oops |
07:58.32 | Qwell | anyhow, bed |
07:58.53 | DA-MAN | night |
08:00.20 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
08:06.09 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
08:10.51 | GuERo | I what to send a variable to emialbody of voicemail, but, this variable need coming from perl script (agi). ¿ This is posible ? |
08:11.50 | *** join/#asterisk shido (~greg@d57-87-253.home.cgocable.net) |
08:13.52 | *** join/#asterisk pooh_ (~pooh_@a213-84-220-3.adsl.xs4all.nl) |
08:17.40 | *** join/#asterisk olivier__ (~olivier@sud35-3-82-240-204-182.fbx.proxad.net) |
08:24.38 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
08:38.28 | key2 | yop |
08:38.33 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
08:39.17 | key2 | anyone knows how one asterisk could tell an otherone to transfer a call to a specific SIP phone ? |
08:40.07 | infi | semaphore? |
08:40.32 | key2 | infi: ? |
08:40.37 | infi | no no, rfc1149 IP over Avian Carrier |
08:40.56 | key2 | what u mean ? |
08:41.16 | infi | passenger pigeons, man |
08:41.51 | Zeeek | Warning: Heathcliff is UNREACHABLE, lag=1241342314231233324132413322 milliseconds |
08:46.08 | *** join/#asterisk jackthe (~jesse@d594f03e.ftth.concepts.nl) |
08:47.49 | *** part/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net) |
08:53.11 | Romik_ | somebody uses voipjet? it down now? |
08:54.15 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
08:54.38 | Zeeek | what is down? |
08:54.43 | Zeeek | they appear to be up |
08:56.31 | Romik_ | http://pastebin.ca/15516 |
08:56.51 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
08:57.53 | Romik_ | zeeek: to any number i dialed i getting : == No one is available to answer at this time |
08:59.28 | Zeeek | what is that 800 numer in pastebin? Something I can try to call? |
09:00.11 | Zeeek | no that's yours I guess... let me look around |
09:01.20 | Zeeek | Romik_ which server do you use? I've heard there were problems with the west coast one |
09:01.37 | Romik_ | zeeek: which one you use? |
09:01.45 | Zeeek | east, I'm in Eu it's closer |
09:02.11 | Romik_ | i use new york one |
09:02.28 | Romik_ | i make test on this one : 18882175648 |
09:03.28 | Zeeek | works for me |
09:03.38 | Zeeek | you owe me $0.014 |
09:03.48 | Romik_ | do you accept paypalk? |
09:04.00 | Zeeek | sorry, $0.013 |
09:04.07 | Zeeek | no karma willbe fine |
09:04.17 | Zeeek | so that number works for me |
09:04.24 | Romik_ | 1.4 per min...? increment 6 sec? |
09:04.36 | Romik_ | which ip do you use? |
09:04.37 | Romik_ | <PROTECTED> |
09:04.53 | Zeeek | sorry: total cost of call: 0.00130 |
09:05.03 | Romik_ | ok |
09:05.07 | Romik_ | tell me which server? |
09:05.31 | Romik_ | which protocl? ulaw or ilbc? |
09:05.36 | Zeeek | that's the one, New York primary - so the problem is with you |
09:05.41 | Zeeek | ulaw |
09:06.22 | Romik_ | <PROTECTED> |
09:06.23 | Romik_ | <PROTECTED> |
09:06.23 | Romik_ | <PROTECTED> |
09:06.24 | Romik_ | strange... |
09:07.12 | Zeeek | check your callerid - mine is 8663241324 - no '1' |
09:07.43 | Zeeek | that's the format I mean not the exact callerid I use |
09:07.48 | Romik_ | zeeek: callerid i can anything... |
09:08.04 | Zeeek | some 800 nums will not answer calls with certain prefixes, especially 800 ones |
09:09.11 | Romik_ | like problem with their server |
09:11.46 | Romik_ | since 20-jun - there no calls via voipjet |
09:12.08 | Zeeek | for me it works |
09:16.28 | *** join/#asterisk bjohnson (~bjohnson@ip172-172.dsl.istop.com) |
09:21.24 | *** join/#asterisk [Ladybug] (~ladybug@219.95.78.33) |
09:21.34 | [Ladybug] | hi all |
09:24.30 | drray | lo |
09:24.38 | Zeeek | yo |
09:25.24 | [Ladybug] | need help with the asterisk n little with my programing |
09:25.33 | Ahrimanes | mornin' |
09:25.34 | [Ladybug] | just need to know more behavior abt asterisk |
09:25.38 | [Ladybug] | morning/evening |
09:27.02 | [Ladybug] | i wrote a vc++ to interact with asterisk |
09:27.25 | [Ladybug] | but once i idle more than 5 mintues... n i call out using the softphone.. it crashes |
09:27.42 | [Ladybug] | can i know if there is any internal activities inside asterisk itself... |
09:27.51 | [Ladybug] | what;s it format pattern? |
09:28.09 | Zeeek | NAT probs? |
09:28.36 | [Ladybug] | what does NAT means? |
09:28.42 | [Ladybug] | sorry.. i am very new to * |
09:29.09 | Zeeek | what is your c++ app? A softphone? |
09:30.16 | jansaell | NAT - Netrowk Address Translation |
09:30.18 | [Ladybug] | my c++ application is a program which got the manager api button |
09:30.22 | [Ladybug] | thanks jans |
09:30.30 | jansaell | np |
09:30.50 | [Ladybug] | for ex.. when i click Login... the asterisk will response back the response to me.. telling me login successful n etc |
09:30.59 | [Ladybug] | the softphone i am using is sjphone |
09:31.12 | Zeeek | I never got thatone to work |
09:31.17 | [Ladybug] | which 1? |
09:31.21 | [Ladybug] | u means sjphone? |
09:31.50 | Zeeek | yes, sjp^hone - so I can't help you with that |
09:32.11 | Zeeek | I thought you were writing the phone... which is why I asked about NAT |
09:32.25 | [Ladybug] | oh... |
09:32.30 | [Ladybug] | i m not writing the phone |
09:32.33 | [Ladybug] | i am using that |
09:32.39 | [Ladybug] | i have 2 usbphone |
09:32.52 | [Ladybug] | n i use my sjphone to dial to the usbphone |
09:33.11 | [Ladybug] | my c++ program will detect the response n send the response back to my vc++ |
09:33.36 | *** join/#asterisk Qorky (~Pooa@dip-202-72-131-243.wa.westnet.com.au) |
09:34.09 | der[mat] | anyone knows a d-link cpe device with mgcp support? |
09:40.17 | Qorky | ne1 know why im getting this: No ISA tormenta card found at d0000 ? |
09:45.59 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
09:48.13 | *** join/#asterisk smeevil (~smeevil@gremesh1.demon.nl) |
09:48.34 | smeevil | good localtime :) |
09:50.06 | smeevil | i was wondering if any of you could tell me if its possible to generate some static on calls that are in progress.. a lot of people think the line is dead when one keeps silent since they are used to hear some static... i tried to search on google for it and only thing i ran into is that this phenomenon is called comfort noise ? |
09:51.06 | drray | a solution in search of a problem |
09:51.11 | drray | comfort noise |
09:51.50 | drray | give everyone bothered by it a wrapper from a pack of smokes and let them crinkle that over the phone |
09:52.03 | smeevil | LOL true enough :) |
09:53.36 | smeevil | i guess a _real_ solution would be to create a small gsm file that contains low volume static and play that looping in the background ? |
09:55.27 | jackthe | most SIP-phones have some advanced settings for CN |
09:56.08 | jackthe | less load for your *-box |
09:56.29 | jackthe | adding CN to a call is eating CPU... |
09:57.47 | Ahrimanes | app_meetme2.c:646: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) <- hmmm |
09:57.51 | smeevil | using cisco 7905....go figure |
09:58.34 | drray | real solution is to tell them that like cave painting going away, static free phone calls are a good thing |
09:59.04 | jackthe | it has CNG |
09:59.07 | jackthe | Voice-activity-detection (VAD) and comfort-noise-generation (CNG) support |
09:59.20 | jackthe | here is the source http://www.cisilion.com/7905.htm |
09:59.42 | jackthe | don't know how to set the CNG on, but Cisco does ;) |
10:00.05 | AndyCap | drray: static free is good, dead silence is not. :-) |
10:00.26 | drray | I fail it, I guess |
10:00.47 | drray | ideas like CNG boggle my brain |
10:00.54 | smeevil | ty jackthe |
10:01.00 | smeevil | will take a look in that |
10:01.00 | drray | :) |
10:01.45 | jackthe | good luck with Cisco getting the manual to change your CNG-settings |
10:01.57 | drray | :) |
10:11.13 | *** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt) |
10:24.58 | *** join/#asterisk Mavvie (edwin@dsl-35.56.240.220.dsl.comindico.com.au) |
10:27.57 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
10:29.35 | *** join/#asterisk lehel (~lehel@82.79.20.17) |
10:29.44 | lehel | hello |
10:31.19 | *** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
10:32.18 | *** join/#asterisk meppl (mephisto@p54AAD58D.dip.t-dialin.net) |
10:33.57 | *** join/#asterisk zotz (~zotz@208.196.247.140) |
10:34.35 | *** join/#asterisk speakman (~speak@t2o30p7.telia.com) |
10:34.45 | speakman | hi folks |
10:35.51 | speakman | none awake? |
10:36.00 | Ahrimanes | kinda |
10:36.07 | jackthe | sleepwalking |
10:36.08 | souls | hehe |
10:36.10 | speakman | so many people... so many idlers... :P |
10:36.11 | souls | lurking mode :P |
10:36.22 | speakman | it might be night in the US? :) |
10:36.35 | souls | i am not in the US ... |
10:36.37 | Ahrimanes | people should be waking up in an hour or so |
10:36.38 | souls | so no worries |
10:36.41 | jackthe | me nether |
10:36.46 | zotz | @ 6:30 am on the east coast |
10:36.46 | speakman | hehe ok |
10:36.48 | speakman | ok |
10:37.04 | speakman | I do agree it's too early yet then.. ;) |
10:37.33 | speakman | Anyone of you using ZapRAS in any application? |
10:37.40 | jackthe | it's lunchtime here ;) |
10:37.51 | speakman | jackthe: CET? |
10:38.05 | speakman | or GMT+1 |
10:38.15 | jackthe | GMT+1 |
10:38.29 | *** join/#asterisk FreezeS (~gido_b@83.103.170.130) |
10:38.30 | speakman | or +2 depending on light saving times |
10:38.37 | speakman | or something.. hehe |
10:38.40 | jackthe | netherlands |
10:38.48 | souls | jackthe: hehe nice |
10:38.49 | jackthe | easier |
10:38.50 | FreezeS | hello |
10:38.51 | souls | .nl r0x |
10:38.52 | speakman | oh, netherlands.. sweden here.. |
10:38.56 | jackthe | ah |
10:38.58 | souls | germany here :P |
10:39.00 | FreezeS | I have a problem with an ISDN card |
10:39.01 | souls | hehe |
10:39.04 | Ahrimanes | .dk 0wnz |
10:39.12 | FreezeS | can anybody help me ? |
10:39.23 | souls | FreezeS: depends on some more details |
10:39.45 | FreezeS | I'm using zaphfc |
10:39.52 | FreezeS | and aparently it's all ok |
10:40.01 | FreezeS | except that I can't make any calls |
10:40.06 | speakman | I have problems with ZapRAS but no one seems to be using that.. |
10:40.12 | speakman | tried the mailing list for months.. :P |
10:40.26 | souls | speakman: sorry, never used that one |
10:40.39 | jackthe | use nether of them |
10:40.43 | souls | hehehe |
10:40.54 | speakman | who? what? |
10:40.59 | souls | FreezeS: how do you take it it is all ok? |
10:41.00 | jackthe | I'm more the research type |
10:41.14 | speakman | ?! |
10:41.30 | FreezeS | can I paste 4 lines here ? |
10:41.53 | jackthe | try it, if it doesn't work try 2 lines twice |
10:42.01 | FreezeS | I mean, is it agains policy ? |
10:42.02 | FreezeS | ok |
10:42.04 | *** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net) |
10:42.04 | FreezeS | Chan Extension Context Language MusicOnHold |
10:42.04 | FreezeS | <PROTECTED> |
10:42.04 | FreezeS | <PROTECTED> |
10:42.04 | FreezeS | <PROTECTED> |
10:42.18 | FreezeS | that shows up when I type |
10:42.22 | FreezeS | zap show channels |
10:42.33 | key2 | FreezeS: pastbin |
10:42.56 | FreezeS | key2: what's that ? |
10:43.04 | key2 | for not pasting like that in the chan |
10:43.07 | key2 | before getting KB |
10:43.13 | FreezeS | ok, that's why I asked |
10:43.32 | FreezeS | I knew that on some channels they use things like that |
10:43.52 | key2 | FreezeS: http://pastebin.ca |
10:44.05 | FreezeS | thanks |
10:45.07 | *** join/#asterisk psywar (psywar@rasterburn.org) |
10:46.18 | psywar | Hey can anyone tell me what digits I hit on a Sipura SPA-2000 to get the features listed at the end of the manual? For some reason it has blanks where the two digits should be. |
10:46.59 | FreezeS | http://pastebin.ca/15534 |
10:47.02 | speakman | no one using pppd at all within asterisk? :P |
10:49.18 | FreezeS | souls: did you look at pastebin ? |
10:49.36 | souls | about to |
10:50.02 | FreezeS | there is another problem... I've seen it just now... |
10:50.11 | FreezeS | the signalling is PRI |
10:50.21 | souls | hmmmm |
10:50.26 | souls | good question, never had that error |
10:50.31 | FreezeS | but in zapata.conf I have signalling = bri_cpe |
10:50.45 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
10:50.53 | FreezeS | and zaphfc is loaded normally (not NT) |
10:50.59 | souls | hmmm |
10:51.10 | souls | as far as i know NT mode is what you would want |
10:51.22 | FreezeS | hmm |
10:51.28 | FreezeS | my setup is as follows: |
10:51.34 | souls | at least from my experience with ISDN |
10:51.47 | souls | even though i havent used that in years literally |
10:51.56 | FreezeS | one computer, with debian, with one isdn card, and a lot of SJPhones connected via network |
10:52.17 | FreezeS | ok, so I'll try NT |
10:52.23 | FreezeS | I'm a noob with telephony |
10:53.16 | souls | everyone here has been at some point in his life |
10:53.23 | souls | and being "pro" is very relative |
10:55.01 | FreezeS | ok, tried with NT |
10:55.03 | FreezeS | same result |
10:55.41 | smeevil | no luck yet with cng on the cisco's :/ |
10:55.50 | FreezeS | the weird thing is that although I have signalling = bri_net, it shows Signalling Type: PRI Signalling |
10:56.57 | jackthe | smeevil: try the Cisco helpdesk, maybe they can help. |
10:56.58 | *** part/#asterisk popooya (~popooya@07738e59f333d31b.session.tor) |
10:57.48 | smeevil | jackthe: yeah, will do that |
10:58.30 | jackthe | otherwise always have some music on in your office, then there is enough Comfort Noise (try to find a nice jazzy radiostation to fall asleep during worktime ;)) |
10:58.51 | smeevil | friggin cisco phones really, every logical thing is in the wrong place, to transfer a call you need to go to a sub menu before you can actually transfer, default only has one ringtone, etc etc etc |
11:00.00 | FreezeS | so, nobody else has any experience with zaphfc ? |
11:03.07 | postel | smeevil: it doesnt haev a single one, it has Chirp 1 and 2, resample new sounds , edit RINGLIST.DAT and reboot to get more sounds, to tranfer its in the same bottom menu with everything else (hit more once) on CCM you can even remove items or change the order, do you have a question or just randomly bitching? |
11:06.02 | smeevil | postel: well i know all that, using atftpd to see what happends, the phone only asks for the lddefault files, even though i have all the files in the directory of tftpd, default it only has chirp1 (cisco 7905) , but i still need to try the latest firmware. that might solve a lot prolly |
11:09.08 | postel | you got right permissions for /tftpboot? got the files from CCO or emule? |
11:10.31 | *** join/#asterisk emboss (emboss@caffeine.blacknight.ie) |
11:10.52 | emboss | Hi, I have 3 numbers from my telco, i'm looking to setup a pbx for about 20 users |
11:11.08 | emboss | can anyone recomend a card to suit? |
11:17.50 | FreezeS | I've made some advancements.... now, if I use signalling = bri_net_ptmp, it doesn't show the error |
11:18.13 | FreezeS | but the call isn't coming through eighter.... |
11:19.11 | *** join/#asterisk gres (~gres@81.222.48.242) |
11:19.44 | gres | hi all. Can anybody help me? |
11:27.40 | FreezeS | does anybody know how can I dial from asterisk command line ? |
11:29.34 | *** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net) |
11:31.16 | Zeeek | show applications |
11:34.28 | lehel | i had an extension "211" .. i deleted it, but still finds somewhere |
11:34.38 | lehel | i can't find anywhere.. |
11:37.41 | *** join/#asterisk pooh_ (~pooh_@a213-84-220-3.adsl.xs4all.nl) |
11:38.19 | Zeeek | lehel how can it be? |
11:39.04 | Ahrimanes | deleted it in the conf file but didnt reload? |
11:39.05 | drray | have you reloaded the server? |
11:39.06 | Ahrimanes | hey Zeeek :D |
11:39.13 | Zeeek | beer? |
11:39.17 | Ahrimanes | where? |
11:39.40 | jansaell | FreezeS: you have to have the console chanel loaded to be able to dial from the command line i think |
11:46.15 | Qorky | . |
11:47.09 | *** join/#asterisk Astermigraine (~psolomon@69-165-217-96.atlsfl.adelphia.net) |
11:47.52 | *** join/#asterisk pawalls (~pawalls@pawalls.teamgleim.com) |
11:48.24 | pawalls | Can someone PLEASE help me diagnose a problem with a partial T1 PRI I'm having? |
11:48.59 | pawalls | I'm using a TE110P, our incoming line from the phone company is 18 channels (1-18), plus the D-channel on channel 24 |
11:49.35 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
11:50.32 | pawalls | I've configured zaptel using every single combination of fx(s|o)(k|l|g)s and none of them work. All I get when I dial in to the system is a busy signal. There are no alarm states on the card, all of the LEDs on the DS1 look exactly like they do when it's in the current phone server. /proc/zaptel/1 shows all of the channels correctly and no alarm states. |
11:50.42 | pawalls | zttool shows "OK" as the status of the span. |
11:50.47 | FreezeS | aparently bristuff hates 2.4.x kernel |
11:51.02 | FreezeS | I'm trying now with the latest 2.6. |
11:51.26 | lehel | Zeeek: it's ok |
11:51.39 | pawalls | But for some forsaken reason, it is giving me nothing but a busy signal when I call. I've tried every possible combination of options I can think of. I know the connection is esf/b8zs so it can't be that... |
11:51.40 | lehel | the extension number had autoforward to itself.. |
11:51.44 | pawalls | Can ANYONE think of what would cause this? |
11:52.56 | pawalls | I've followed the instructions step by step from the Digium website for configuring a system with a PRI trunk. |
11:53.22 | pawalls | But I'm running into this same wall, and there is not a single bit of useful debugging information to tell me what is wrong. |
11:53.52 | pawalls | dmesg says nothing, all proc files show that it is working properly. I'm definitely at a loss at this point, any remote amount of help would be greatly appreciated at this point. |
11:54.07 | Ahrimanes | http://pastebin.ca/15543 anyone know what could cause this kind of error? |
11:56.22 | Delvar | ferits |
11:56.32 | Delvar | ferits with evil mind rays |
11:58.38 | drray | is the light green on the PRI? |
12:00.59 | Zeeek | Ahrimanes a stab in the dark, you sure all the version of various stuff are up to date? |
12:01.41 | Ahrimanes | Zeeek: yeah, same source code that i am running from now, just added this module |
12:01.53 | Ahrimanes | ah but could be that the module requires cvs and not 1.0.7 |
12:02.01 | Ahrimanes | where's areski when i need him, hehe |
12:02.14 | Zeeek | that's the kind of thing I was talking about |
12:05.58 | MikeJ[Laptop] | everyone checked out the cool stuff at pbxfreeware.org? |
12:08.23 | InfraRed | MikeJ[Laptop]: any good porn ? |
12:09.04 | MikeJ[Laptop] | sure |
12:09.15 | MikeJ[Laptop] | <sigh> |
12:10.31 | *** join/#asterisk mrproper_ (~psynode@CPE-60-225-51-188.nsw.bigpond.net.au) |
12:10.46 | mrproper_ | what do i need to connect normal analogue lines to asterisk? |
12:11.21 | Zeeek | in PC FXO card or modules or FXO devices |
12:11.26 | psywar | if you don't need support, an IA92 winmodem |
12:11.30 | Zeeek | like SIpura, Grandstream ATA etc |
12:11.51 | psywar | yeah the winmodem talks FXO (to the telco) |
12:12.16 | psywar | for FXS (handset), either a softphone or an ATA like the SPA-2000, about $75 |
12:12.26 | psywar | the winmodem is like $7 |
12:14.39 | mrproper_ | basically i have 4 normal analogue lines going out to the telco, i want to feed them into asterisk and have the sip phones use them for in and out dial |
12:15.45 | *** part/#asterisk lehel (~lehel@82.79.20.17) |
12:16.51 | psywar | ah you may want something higher-density than the IA92 |
12:16.58 | MikeJ[Laptop] | mrproper_, I don't recomend the $7 modems.. |
12:17.00 | psywar | check out digium's FXO cards |
12:17.03 | jansaell | mrproper: You can use the digium tdm400p for that setup |
12:17.04 | MikeJ[Laptop] | especially for that many |
12:17.11 | MikeJ[Laptop] | what they said ^^^ |
12:18.21 | mrproper_ | thanks guys |
12:18.26 | jansaell | np |
12:24.56 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
12:25.08 | cjk | hi, anyone here who uses loadbalancers and asterisk |
12:26.09 | *** join/#asterisk jerlique (~jerlique@lnk254.adl0.adsl.esc.net.au) |
12:26.53 | Qorky | anyone here setup a fritz card and capi ? im stuck! please |
12:27.57 | *** join/#asterisk heka (~heka@82.114.68.126) |
12:35.12 | *** join/#asterisk Romik_ (~romik@212.143.5.146) |
12:38.28 | *** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com) |
12:38.51 | *** join/#asterisk Tili (~Tili@202-133-67-212-dialup.sat.net.pk) |
12:45.16 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
12:45.23 | *** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
12:45.43 | Ariel_ | Hello everyone |
12:47.00 | *** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo) |
12:48.23 | [TK]D-Fender | Good morning |
12:50.18 | jansaell | well aftrenoon |
12:53.31 | *** join/#asterisk [flop] (~flop@socks.epitech.net) |
12:53.48 | Ariel_ | jansaell, if you say so then it's afternoon at your location. |
12:55.28 | *** join/#asterisk Morex (~blah@host81-157-226-241.range81-157.btcentralplus.com) |
12:55.32 | Morex | Hello all |
12:55.36 | [flop] | I'm using asterisk 1.07 on 5.4 FreeBSD station and i have a problem : it seems that mpg123 is going into an infinite loop |
12:55.51 | *** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com) |
12:55.54 | Tili | hello |
12:56.01 | gambolputty | what do you mean by infinite loop? |
12:56.08 | gambolputty | consuming memory like crazy? |
12:56.23 | Tili | is there someone here from Pakistan and has to throw some dialogic hardware. cuz I want some stuff |
12:56.32 | [TK]D-Fender | Dear God... Anthm has been busy with res_EVERYTHING....... |
12:56.35 | [flop] | i'm lauching asterisk with -vvvgc then when trying to stop now, it loops |
12:56.52 | gambolputty | kill the process then |
12:57.14 | [flop] | with a ps aux, it seems that mpg123 is the only process related to aterisk wich is not stoped! |
12:57.30 | [flop] | gambolputty indeed but it should work by itself |
12:57.44 | Ariel_ | Tili, your not wanting to use dialogic boards with asterisk are you? |
12:57.51 | gambolputty | maybe don't use mpg123 then |
12:57.58 | gambolputty | convert mp3 files to gsm |
12:58.04 | gambolputty | something like that |
12:58.11 | [flop] | my main problem is that i don't have any sound on my sip phones and i tought it was related to mpg123 |
12:58.29 | [flop] | gambolputty is the demo using an mp3? |
12:58.38 | gambolputty | what demo? |
12:58.46 | [flop] | the demo context |
12:58.53 | Ariel_ | [flop], no sound on sip phones is a problem with nat settings most of the time. |
12:58.54 | gambolputty | I think so |
12:59.00 | [flop] | you know, when it says welcome to asterisk... :) |
12:59.32 | [flop] | Ariel_ we are on the same subnet |
12:59.45 | [flop] | (the phone and the server) nothing more classical :) |
13:00.38 | drray | nat/udp |
13:00.59 | [flop] | drray not sure to understand.. :\ |
13:01.24 | Ariel_ | [flop], do you have a firewall on the fbsd where asterisk is running? |
13:01.37 | [flop] | no |
13:01.42 | *** join/#asterisk dsfr (~dsfr@207.111.174.1) |
13:02.03 | [flop] | let me explain : |
13:02.06 | drray | is the IP phone and server on the same subnet? |
13:02.13 | [flop] | yes |
13:02.27 | [flop] | no connexion problem between both |
13:02.45 | drray | when did you build asterisk? and was it head or stable? |
13:02.52 | [flop] | the ip phone can even identify itself to the asterisk |
13:03.12 | [flop] | drray from ports (supposed stable) |
13:03.16 | drray | sip show peers |
13:03.20 | drray | ? |
13:03.45 | [flop] | when typing 2 on the ip phone, the asterisk got it, then launch the demo, but no sounds on the phone |
13:04.02 | [flop] | and i tought that mpg123 going on infinite loop |
13:04.18 | [flop] | drray yes it shows |
13:04.27 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
13:04.50 | *** join/#asterisk inspired (mikael@host-81-191-123-197.bluecom.no) |
13:05.33 | [flop] | let me know, a sound card isn't necessary to make asterisk works (i'm in doubt) |
13:05.45 | Ahrimanes | it isnt |
13:06.07 | drray | you do need a timing device |
13:06.15 | drray | but sip should work for a while |
13:06.37 | [flop] | timing device? |
13:06.46 | [flop] | wich means? |
13:06.52 | drray | I'm actually outside my area of expertise |
13:07.23 | drray | but the mailing list was talking about how some sip calls went out of sync without a zaptel timing device |
13:08.15 | *** join/#asterisk jr99 (~jr99@73.230.165.24.cfl.res.rr.com) |
13:09.01 | jr99 | anyone have experience with the sipura device which provides an FXO interface? Can you use the FXO interface with asterisk or is it just for failover/routing within the sipura? |
13:09.21 | drray | flop - have you specified in the sip.conf what protocol you are using? |
13:09.57 | [TK]D-Fender | JR99 : you can use it as a full incoming line to * and it can act as a failover all by istelf as well |
13:10.02 | *** part/#asterisk smeevil (~smeevil@gremesh1.demon.nl) |
13:10.37 | *** join/#asterisk gres (~gres@81.222.48.242) |
13:11.38 | jr99 | [TK]D-Fender: ok. So in theory these could replace my TDM400P with one FXO and one FXS port? |
13:11.55 | MikeJ_ | [TK]D-Fender, you like the toys out on pbxfreeware.org? |
13:12.00 | *** join/#asterisk makhtar (~ageller@mail.bulletinnews.com) |
13:14.21 | gres | I have question. I have active channel IAX2/incoming-from-x. In op_buttons.cfg, i wrote [IAX2/incoming-from-x] ... - but on FOP. I don't see anything. Can anybody help me? |
13:15.10 | gres | Maybe i mast write in op_buttons.cfg [_IAX2/incoming.*]? |
13:15.38 | gres | But it dosn't work too. |
13:16.36 | [TK]D-Fender | jr99 : exactly, and at a cheaper price-point. However you would lose your Zaptel timing device and have to rely on ZTDUMMY |
13:17.58 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
13:18.04 | jr99 | [TK]D-Fender: ahh.. thats right.. forgot about the timing.. My problem is that my TDM400P has always been flakey.. |
13:18.32 | jr99 | [TK]D-Fender: does ztdummy work well? |
13:18.59 | [TK]D-Fender | jr99 : mANY HAVE NO COMPLAINS. mY tmd400 IS FLAKEY AS WELL... WHAT QUIRKS DOES YOURS HAVE? |
13:19.07 | [TK]D-Fender | oops, caps... |
13:19.37 | jr99 | heh.. well sometimes it will fail to initalize.. and you must reload the zaptel drivers a dozen times until it works.. |
13:20.50 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
13:20.54 | [TK]D-Fender | Had that happen to me for a while (I modprobe it in rc.local now) and mine "spontaneously" drops calls. |
13:21.19 | MikeJ[Laptop] | call digium support |
13:21.25 | [TK]D-Fender | MikeJ[Laptop] : Yeah the Res_*Stuff* looks neat, but complicated to set up. I look forward to ResPHP running here soon |
13:21.32 | MikeJ[Laptop] | that's what you pay for when you buy there cards |
13:21.34 | jr99 | My thoughts were that having the sipura would keep eveything SIP based...And would be $32/FXS.. |
13:21.47 | jr99 | Yea.. I've called them.. never got anywhere.. maybe I should try again. |
13:21.47 | MikeJ[Laptop] | [TK]D-Fender, should not be that complicated to set up... |
13:22.50 | MikeJ[Laptop] | Will touch base with bkw and anthm today to see if we can get a better out of tree module installer andded to contrib. there is already one that works for single app modules |
13:27.01 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
13:27.01 | *** mode/#asterisk [+o twisted] by ChanServ |
13:27.19 | Qorky | anyone here setup a fritz card and capi ? im stuck! please |
13:30.31 | mini | Qoeky: ? |
13:32.30 | *** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com) |
13:35.11 | *** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com) |
13:38.43 | *** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com) |
13:39.29 | gambolputty | how can gsm be used for music on hold? |
13:40.18 | *** join/#asterisk Stephnie (dfsdf@203.215.180.254) |
13:40.19 | Delvar | hmm there was a mod to do that... basicly a perl sccript that did a cat of the file and piped to asterisk |
13:40.37 | Delvar | no idea what its called tho :) |
13:40.47 | Stephnie | I just compiled and installed asterisk ..... first time it run fine. |
13:40.52 | Stephnie | I rebooted my system |
13:40.53 | Stephnie | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
13:40.57 | Stephnie | now getting this msg |
13:42.09 | MikeJ[Laptop] | Delvar, rawplayer |
13:42.33 | MikeJ[Laptop] | you can also use native MOH in head |
13:42.33 | Delvar | ah thats it! |
13:42.45 | Stephnie | how to run asterisk as Daemon? |
13:42.46 | MikeJ[Laptop] | it's in contrib\utils |
13:42.56 | Delvar | cool |
13:42.56 | MikeJ[Laptop] | Stephnie, type asterisk |
13:43.14 | Stephnie | ok |
13:43.24 | Stephnie | yes, now asterisk is fine |
13:47.13 | [TK]D-Fender | MikeJ[Laptop] : Just biding my time until these new tools are more easily merged and for 1.0.8 to be released. I am quite close to * acceptance here at work now. |
13:47.31 | der[mat] | is it possible to run asterisk as server and simultaneous as a sip-client? |
13:47.45 | cochi | o.O |
13:47.51 | cochi | you mean to register at sip providers? |
13:48.43 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
13:52.47 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:52.47 | *** mode/#asterisk [+o anthm] by ChanServ |
13:53.59 | der[mat] | cochi: jebb |
13:54.24 | der[mat] | afk |
13:54.24 | der[mat] | pardon me |
13:54.25 | *** part/#asterisk der[mat] (~mat@gate-nue0.bintec.de) |
13:55.09 | *** join/#asterisk Katty (~angela@68.112.15.110) |
13:58.29 | *** join/#asterisk orospakr (~orospakr@ip-168.82.126.206.dsl-cust.ca.inter.net) |
13:58.36 | *** join/#asterisk VoIpMaster (VoIp@194.105.96.243.static.cablesurf.de) |
13:58.47 | VoIpMaster | hi to all |
13:59.24 | VoIpMaster | the first question is does anybody know a german asterisk channel? I'm here in Germany and English is only a native language for me ... |
13:59.28 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
14:00.30 | InfraRed | you can always start #asterisk-de |
14:00.52 | *** join/#asterisk olivier_ (~olivier@sud35-3-82-240-204-182.fbx.proxad.net) |
14:01.33 | orospakr | hi! I am trying to build asterisk on an ubuntu x86_64 box, and I'm seeing: cc1: error: bad value (k8) for -march= switch. |
14:01.49 | VoIpMaster | thx InfraRed |
14:02.44 | VoIpMaster | but there is only a bot into the channel :) so i think stay here :) |
14:03.32 | Stephnie | when I reboot my machine....asterisk doesnt run automatically... |
14:03.48 | Stephnie | how to make it run automatically when computer resets... |
14:04.05 | *** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net) |
14:04.23 | InfraRed | Stephnie: what distro |
14:04.28 | Stephnie | RH9 |
14:04.50 | InfraRed | put A STARTUP SCRIPT IN /ETC/INIT.D |
14:04.57 | InfraRed | damn capa lock |
14:05.26 | Stephnie | how ? |
14:05.45 | InfraRed | caps |
14:05.52 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
14:06.20 | [flop] | i'm using te405p on a freebsd 5.4 with zaptel drivers and i have no sound on my sip phones, but when i unload the kernel module of the card i do have sound and everything's working. do someone experienced such problem? |
14:07.31 | VoIpMaster | hi flop, is it possible that your soundcard and your te card uses the same irq's? |
14:07.34 | Stephnie | I dont know how to do that |
14:07.41 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
14:07.46 | Stephnie | to put a startup script in /etc/init.d |
14:08.13 | Corydon76-home | Then hire someone who knows what that means and can do it |
14:08.16 | Stephnie | I think I can do it thru command...to make asterisk run automatically with the OS |
14:08.25 | Stephnie | I hired myself :p |
14:09.04 | Stephnie | ok ok . gonna do that . |
14:09.06 | [flop] | VoIpMaster don't thnik so, but what's the relationship beetwen the sound card and the te |
14:10.22 | VoIpMaster | i don't know flop |
14:11.10 | VoIpMaster | actual i'm fighting on a clean install of RH9 and the i get no network ... that's very cracy... |
14:11.13 | orospakr | ah, my problem was ubuntu' |
14:11.20 | orospakr | s choice of gcc 3.3, not 3.4 |
14:11.42 | Hmmhesays | ~seen bkw_ |
14:11.42 | jbot | bkw_ is currently on #asterisk |
14:12.13 | *** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com) |
14:12.43 | VoIpMaster | I'm away for the next 60 minutes, if important leave a PM |
14:13.28 | Hmmhesays | anyone in here using dell poweredge's? |
14:15.16 | [TK]D-Fender | I keep getting a "file not found" type error looking for "/etc/init.d" ;) |
14:16.09 | Hmmhesays | haha RoyK |
14:18.23 | *** join/#asterisk brookshire (~matt@207.111.174.1) |
14:18.37 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
14:19.18 | Hmmhesays | that girl on the digium site is kind of cute |
14:20.11 | Wonka | hmm, he says... |
14:20.34 | Wonka | yeah, definitely considerable |
14:20.35 | Hmmhesays | indeed |
14:20.41 | Katty | Hmmhesays: mew |
14:20.53 | Hmmhesays | Katttaaaaay |
14:21.28 | brimstone | pamples? |
14:21.33 | Katty | yes |
14:21.36 | Katty | :< |
14:22.00 | Hmmhesays | how goes it this morning? |
14:22.24 | Katty | am sleepy |
14:23.11 | *** join/#asterisk stormfr (~StorM@alf94-3-82-66-251-138.fbx.proxad.net) |
14:23.58 | stormfr | hello, did anybody already see a iax trunk not anymore working ? (cvs head 2005-06-09) |
14:24.31 | orospakr | hi! I'm getting chan_zap not found whenver I start asterisk. I didn't compile in zapata, and I don't need to use the cards, so what do I need to do to tell asterisk to not load the plugin? |
14:25.27 | InfraRed | look in the conf files asterisk.conf or something like modules.conf |
14:25.28 | InfraRed | cant remmber the name |
14:25.34 | InfraRed | noload => module |
14:25.35 | *** join/#asterisk dalabera (~Dalabera@pmr.pmrtechnologies.com) |
14:26.11 | brimstone | /etc/asterisk/modules.conf |
14:26.36 | brimstone | nload => chan_zap.so |
14:26.43 | `Sauron | SNORK |
14:26.56 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
14:28.06 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM] |
14:28.06 | *** join/#asterisk newl (~newlook@203-59-112-225.dyn.iinet.net.au) [NETSPLIT VICTIM] |
14:28.06 | *** join/#asterisk ptiggerdine (~ptiggerdi@c210-49-98-194.rochd1.qld.optusnet.com.au) [NETSPLIT VICTIM] |
14:28.31 | brimstone | i prefer meaty smoothies |
14:29.04 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) [NETSPLIT VICTIM] |
14:29.11 | Hmmhesays | and suddenly I felt very alone on asimov |
14:29.14 | Hmmhesays | lol |
14:30.27 | orospakr | ah, I found my problem. it was actually a larger config problem. |
14:36.38 | *** join/#asterisk mjman (~mike@205.158.42.66.ptr.us.xo.net) |
14:37.46 | mjman | Hi, I am having some issues with a SNOM 190 SIP phone, and the page on the wiki does not have what I need. Call waiting simply does not work on these phones. When I test it out, I get a fast busy on the second call. Does anyone have experience with this or a similar phone?? |
14:38.58 | *** join/#asterisk Evanrude (~david@wsip-68-15-251-34.dl.dl.cox.net) |
14:40.49 | *** join/#asterisk file[laptop] (~file[lapt@mctn1-3366.nb.aliant.net) |
14:41.01 | file[laptop] | #42 7*6 |
14:41.26 | file[laptop] | I'm still lacking in sleep |
14:41.43 | mjman | file[laptop]: what? |
14:42.04 | file[laptop] | my prom was lastnight, and safegrad |
14:42.04 | *** join/#asterisk zeedo (~zeedo@www.reboot-robot.net) |
14:42.11 | drumkilla | file[laptop]: !!!!!!!!!!!!!!! |
14:42.15 | drumkilla | how was it?!?!?!?!?! |
14:42.22 | file[laptop] | hi drumkilla |
14:42.27 | file[laptop] | it was good |
14:42.31 | drumkilla | yay!!!!! |
14:42.37 | drumkilla | that's just so cute ... |
14:42.40 | file[laptop] | in the end I said screw it to everything safegrad and went to the loft and slept |
14:42.47 | *** join/#asterisk eric`` (~e@adsl-9-111-66.mia.bellsouth.net) |
14:42.55 | file[laptop] | during which I won a cake in my sleep |
14:42.58 | bkw_ | OMG you use punctuation like a gay man!!!!!!!!!!!! |
14:43.00 | drumkilla | pics? |
14:43.18 | file[laptop] | pics? meh |
14:43.39 | brookshire | me too |
14:43.43 | food | now eat me |
14:43.46 | *** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net) |
14:43.47 | brookshire | i'm hungry and it's like 9:43 |
14:43.56 | DrRighteous | hi bkw & file! morning |
14:43.59 | file[laptop] | it's 11:43AM, and I haven't eaten since... well, I don't remember |
14:44.03 | file[laptop] | hi |
14:44.04 | brookshire | hey file |
14:44.08 | brookshire | :P |
14:44.18 | file[laptop] | brookshire: Matttttt |
14:44.31 | mjman | Hi, I am having some issues with a SNOM 190 SIP phone, and the page on the wiki does not have what I need. Call waiting simply does not work on these phones. When I test it out, I get a fast busy on the second call. Does anyone have experience with this or a similar phone?? |
14:44.47 | drumkilla | I'm drinking red bull |
14:44.47 | brookshire | OH GNO! |
14:44.47 | file[laptop] | drumkilla: when aren't you drinking red bull? |
14:44.47 | brookshire | NOT THE REDBULL |
14:44.54 | drumkilla | yes. the RED. BULL. |
14:44.59 | brookshire | Red.bull. |
14:45.17 | file[laptop] | yay - stumbles! |
14:45.46 | brookshire | REDBULL.ORG.UK is free |
14:45.47 | brookshire | hehe |
14:46.04 | orospakr | Can't locate Asterisk/AGI.pm in @INC. wjat |
14:46.12 | orospakr | s/wjat/what's up with this?/ |
14:46.16 | DrRighteous | trademark warming!!! whoo! whoo! |
14:46.30 | *** join/#asterisk Cramnoselo (fwuser@host-22.216-16-72.iw.net) |
14:46.33 | brookshire | RedBull :) |
14:46.42 | brookshire | RedBull!! |
14:47.35 | file[laptop] | orospakr: you don't have the Asterisk AGI perl stuff installed |
14:47.35 | *** join/#asterisk tagore (~ircap8b@r200-125-24-169-dialup.adsl.anteldata.net.uy) |
14:47.41 | Cramnoselo | Anyone using TDM04B and Asterisk @ home succesfully??? |
14:48.02 | Cramnoselo | I am having issues with the card not recognizing after running yum to update |
14:48.03 | *** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com) |
14:48.19 | orospakr | file[desk], is that included in one of the official asterisk packages or is that completely 3rd party? |
14:48.28 | orospakr | er |
14:48.32 | orospakr | file[laptop]. whoops. |
14:48.34 | doolph | Cramnoselo follow the instructions |
14:48.50 | Cramnoselo | which instructions??? |
14:48.55 | doolph | at their page |
14:49.02 | doolph | there's a how to to install TDM |
14:49.07 | doolph | cards |
14:49.07 | file[laptop] | orospakr: Google it |
14:49.15 | shmaltz | what does this mean: |
14:49.17 | Cramnoselo | I run the genzaptelconf -s -d and it does not work... |
14:49.17 | shmaltz | <PROTECTED> |
14:49.20 | Hmmhesays | bkw_: were you the poweredge fan? |
14:49.27 | file[laptop] | orospakr: type in Asterisk::AGI - first result that comes up |
14:49.54 | *** join/#asterisk Betu| (~betul@62.244.193.101) |
14:50.44 | shmaltz | why am I getting this: |
14:50.46 | shmaltz | <PROTECTED> |
14:51.10 | brookshire | too much lag, or the server isn't responding |
14:51.39 | brimstone | does Asterisk @ Home have a support channel? |
14:52.02 | drumkilla | brimstone: #amportal |
14:52.19 | brookshire | lol |
14:53.26 | orospakr | ugh. now this: |
14:53.28 | Hmmhesays | lemme guess, amp screwed up your config |
14:53.32 | orospakr | Use of uninitialized value in pattern match (m//) at /var/lib/asterisk/agi-bin/dialparties.agi line 123, <STDIN> line 39. |
14:53.33 | Hmmhesays | and you can't fix it |
14:53.36 | Cramnoselo | I run the genzaptelconf -s -d and it does not work... |
14:53.46 | file[laptop] | what am I, an AGI debugger? |
14:54.37 | orospakr | well, it *would* be rather odd for a regular user like me to run into a whole new compile-time bug, yes? |
14:54.39 | Ahrimanes | yeah |
14:54.51 | *** part/#asterisk kaldemar (~kaldemar@xdsl-204-1.nblnetworks.fi) |
14:54.53 | Betu| | hi, i have DIGIUM TDM11B and want to all module for pstn,Do i need to by fxo modules for TDM11B,dont i |
14:55.29 | brookshire | yes |
14:55.39 | brookshire | x100m |
14:55.56 | file[laptop] | orospakr: compile-time bug? |
14:55.58 | file[laptop] | orospakr: dialparties.agi isn't part of asterisk that I know of |
14:56.12 | orospakr | hmmm |
14:56.18 | Hmmhesays | i believe that is part of amp |
14:56.19 | orospakr | perhaps amportal is munging something |
14:56.20 | file[laptop] | it's from AMP |
14:56.55 | Betu| | Is x100m another type of card or the name of fxo module ? |
14:57.06 | Hmmhesays | you don't have to use amp with * @ home... in fact the use of custom-context is a good way to go |
14:57.13 | dsfr | Beirdo: it's the FXO module. |
14:57.20 | Beirdo | ? |
14:57.21 | bkw_ | it tells you the problem |
14:57.34 | dsfr | oops, went that for Betu| |
14:57.37 | Hmmhesays | that way you can leave the amp configs alone... and just use your own.... it provides a good interface for editing the *.conf files |
14:58.10 | orospakr | yeah, rerunning AMP solved it. |
14:58.13 | orospakr | d'oh! |
14:58.15 | Hmmhesays | bkw_: were you the dell poweredge fan? |
14:58.22 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
14:58.27 | bkw_ | Hmmhesays, yes |
14:58.30 | BoRiS | bkw!!!!!!!! |
14:58.44 | orospakr | anyway, it seems that I am back at the same problem as I had before: asterisk segfaults on SIP phone hangup. |
14:58.57 | Cramnoselo | Anyone using TDM04B and Asterisk @ home succesfully??? |
14:59.20 | Betu| | thanks |
14:59.23 | *** part/#asterisk Betu| (~betul@62.244.193.101) |
14:59.28 | Hmmhesays | curious I'm poking through the dell website... what OS do you like to run on those... dell gives you windows or redhat from the factory |
14:59.33 | Cramnoselo | I am having issues with the card not recognizing after running yum to update |
15:00.28 | JerJer | use the source |
15:01.16 | file[laptop] | yay people's court |
15:01.36 | *** join/#asterisk ilan (ilan@69.60.110.251) |
15:01.39 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
15:01.49 | tagore | Cramnoselo, i'm using TDM22B and is running fine |
15:02.04 | IPmonger | hey folks |
15:02.08 | Hmmhesays | it's not people's court without judge wapner |
15:02.20 | IPmonger | quick question: is there a open-source SIP call generator? |
15:02.36 | Cramnoselo | tagore, when I run the genzaptelconf -s -d, it tells me there is no zaptel... |
15:02.39 | Hmmhesays | ~google |
15:02.39 | jbot | it has been said that google is a search engine found at http://www.google.com/ |
15:02.39 | Cramnoselo | Any ideas? |
15:02.41 | JerJer | IPmonger: asterisk |
15:03.07 | tagore | change the pci slot and try again |
15:03.10 | tagore | :O |
15:03.35 | Cramnoselo | Do you think I should reinstall the whole system or just move the card? |
15:04.12 | tagore | you can try move the card, if this not work , reinstall, and not ryn yum again :P |
15:04.15 | tagore | run* |
15:04.16 | *** part/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
15:05.04 | tagore | the card is working before you run yum? |
15:06.05 | Cramnoselo | it seems to be the LEDs light before yum, but not after... |
15:06.24 | tagore | crap !! |
15:06.49 | tagore | try reinstall, and not run yum :S |
15:07.21 | Cramnoselo | will it run stable without running the updates? |
15:08.21 | bkw_ | crontab -e |
15:08.24 | bkw_ | doh |
15:08.41 | tagore | i'm running the stable without updates |
15:08.50 | tagore | and is runnign fine |
15:08.58 | tagore | and is in lan ;) |
15:09.13 | tagore | low risk |
15:09.45 | Nuxi | res_php works with cvs-head (it even compiles) http://eder.us/projects/asterisk_php/asterisk_php.0.1a.tgz |
15:10.02 | DarkSpectre | hey, where are the voicemail recordings actually stored at |
15:10.35 | Nuxi | <PROTECTED> |
15:10.54 | DarkSpectre | cool. |
15:11.12 | Cramnoselo | will do tagore, thanks for the help!!! |
15:11.23 | tagore | u'welcome |
15:12.32 | dalabera | IPMONGER take a look onto this? http://www.inaccessnetworks.com/projects/asterisk-oh323/utils |
15:12.42 | *** join/#asterisk jansaell (~jan@c80-216-185-161.cm-upc.chello.se) |
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15:17.36 | *** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net) |
15:18.43 | Hmmhesays | anyone using a dell poweredge 1800? |
15:19.15 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
15:20.08 | file[laptop] | hi Mike |
15:21.16 | *** join/#asterisk Corydon-w (grey@vcchgate.vcch01.springfield.tn.us.vcch.net) |
15:23.09 | *** join/#asterisk grolloj (~grolloj@slim-eth0.horizonlive.net) |
15:34.18 | rephorm | has anyone ever had zap channels get stuck in ast_waitfor_nandfds? (like, for ~120 hours after the call was hung up). this is on a zap to zap call using a channel bank and a wc t100p |
15:36.19 | *** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net) |
15:36.27 | shido6 | ouch' |
15:36.47 | [TK]D-Fender | I'm trying to pick a server to use for * for my company and am not liking the $ figure Dell is giving me. Any suggestions? |
15:37.42 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
15:37.49 | doolph | you can build your own [flop] |
15:37.52 | doolph | i mean |
15:37.52 | cluecon | TKD: how soon do you need an answer and what price range are you targetting? |
15:37.56 | Ariel_ | [TK]D-Fender, look into a supermicro |
15:37.58 | doolph | [TK]D-Fender |
15:38.47 | [TK]D-Fender | Well I quoted 2000$CAD (rough) for a quicky server that is a local store white box with known * compatible parts (just the server w/o cards) |
15:39.12 | [TK]D-Fender | And told them I may be willing to look a little higher, but a raided Dell 2850 is running over 5K |
15:39.45 | [TK]D-Fender | but is much better supported. I like the low-end idea for the ability to use ATA drives which can be swapped at the drop of a hat though and cloned as easily. |
15:40.06 | *** join/#asterisk mjman (~mikem@205.158.42.66.ptr.us.xo.net) |
15:40.07 | *** part/#asterisk mjman (~mikem@205.158.42.66.ptr.us.xo.net) |
15:40.14 | [TK]D-Fender | and of course the cost factor. ATA raid has me a little nervous because of how often the OS sees both drives independantly (Slack 10) |
15:40.20 | [TK]D-Fender | So I'm a little conflicted |
15:40.29 | cluecon | TKD: I am building an AMD64 3000+ with 4 Gigs of RAM and 1 TB of storage for around $1500 USD |
15:41.28 | [TK]D-Fender | cluecon : I know I can do that white-boxing, but its a question of corporate approval as well... |
15:42.13 | [TK]D-Fender | And I've heard here that there may be interrupts issues on AMD based solutions (although I swear by them for PC's) |
15:42.15 | jontow | hey wtf is this "X-Lite v2.0 Build 1105d for Linux" ;) |
15:47.34 | orospakr | http://pastebin.ca/15558 <-- log of segmentation fault. on hangup. |
15:47.38 | orospakr | it's consistent. |
15:48.28 | cluecon | orospakr: what version of * are you using? |
15:48.37 | orospakr | 1.0.7 on x86_64. |
15:48.42 | orospakr | compiled with gcc 3.4. |
15:48.57 | orospakr | the original ubuntu package for 1.0.6 on x86_64 did the same. |
15:51.40 | *** join/#asterisk dennis (dennis@dennis.mod.unixboard) |
15:51.52 | cluecon | orospakr: have you tried CVS-HEAD? |
15:52.05 | Silik0n | orospakr or did you get backtrace? |
15:52.25 | orospakr | Silik0n, I can't seem to run gdb on asterisk (either my build or ubuntu's) |
15:52.31 | orospakr | it just crashes right away in some nptl stuff |
15:52.49 | orospakr | hmm, perhaps I could try attaching... |
15:53.13 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
15:54.25 | orospakr | ah HAH |
15:55.28 | orospakr | http://pastebin.ca/15559 <-- backtrace. |
15:56.46 | *** join/#asterisk JerJer (~JerJer@jerjer.bronze.supporter.pdpc) |
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15:57.35 | brettnem | good morning |
15:57.45 | [TK]D-Fender | What is the easiest way to follow the changes being made in CVS-Head? Scanning the mailing list archives looks a little messy right now. Is there a better way (with an eye on following mod's and documentation on their use) |
15:58.07 | brettnem | eyow! |
15:59.54 | *** join/#asterisk kFuQ (~somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
16:01.15 | *** join/#asterisk vooduhal (~christoph@67.19.25.178) |
16:01.41 | *** join/#asterisk Stephnie (dfsdf@203.215.180.254) |
16:02.09 | jontow | x-lite on linux runs .. extremely well |
16:02.17 | harryvv | it does? |
16:02.24 | harryvv | I have some errors from it. |
16:02.29 | harryvv | missing some kind of library |
16:02.47 | vooduhal | Quick question guys. We're using one of the 4 port modular cards with 4 FXO modules and when dialing on a Zap channel, it takes ~4-5 seconds before the call starts ringing. Does anyone know why this is and a way to get shorten that period? |
16:03.10 | *** join/#asterisk bonez41 (~aint@drjones.dsl.xmission.com) |
16:03.10 | harryvv | I can see xlite as a temperary solution to ip phones but thats it. |
16:03.10 | Silik0n | vooduhal: out pulse time to send digits to the CO |
16:03.39 | orospakr | kphone works well now. |
16:03.44 | vooduhal | Silik0n: I'm not sure I understand. |
16:03.45 | bkw_ | cvs-stable needs to die |
16:03.46 | Ariel_ | vooduhal, that is due to it's trying to wait for caller ID info |
16:03.47 | rephorm | vooduhal: dialing out? |
16:03.52 | harryvv | jontow what distro are you using? |
16:03.53 | vooduhal | Yes, going out. |
16:03.54 | bkw_ | I think some people would agree with me |
16:04.02 | bonez41 | is it possibl with asterisk, to have a specific incoming call routed to another number? i.e., if party X calls and I want them to just call party Y...can asterisk do that? |
16:04.02 | vooduhal | We've resolved the inbound time. |
16:04.07 | orospakr | anyway, no one has any ideas about that mysql_real_escape_string problem? |
16:04.08 | bonez41 | possibl=possible |
16:04.16 | Ariel_ | bkw_, no I like stable |
16:04.18 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
16:04.31 | jlewis | bkw_: whats wrong with cvs-stable? |
16:04.31 | file[laptop] | vooduhal: it takes time to dial the digits... |
16:04.39 | Ariel_ | vooduhal, are you dialing from a sip phones? |
16:05.07 | vooduhal | Ariel_: Yes, but the call from the phone to asterisk is pretty much instantious (sp). |
16:05.09 | jlewis | I've got several servers running various -stable snapshots |
16:05.17 | rephorm | vooduhal: what device are you calling (i.e. what is ringing 4-5 seconds after you dial?) |
16:05.39 | Ariel_ | jlewis, I have lots of asterisk boxes running stable. |
16:06.03 | vooduhal | rephorm: Nothing. That's the problem. It takes 4-5 seconds before the ringing on the line starts. |
16:06.05 | *** join/#asterisk unabonger (BhongwanSh@stilyagihall.themoon.org) |
16:06.25 | jontow | harryvv; Fedora Core 3 on the machine i tested with |
16:06.42 | *** join/#asterisk cmk (~cmk_@p54A3EBD2.dip.t-dialin.net) |
16:07.22 | *** join/#asterisk jtodd (~jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
16:07.36 | harryvv | is that the 32 bit version? |
16:07.40 | jontow | yes |
16:07.53 | harryvv | okay had problems with it in 64 bit version. |
16:08.17 | jontow | no idea about 64bit :/ |
16:08.17 | jontow | never owned one of them :) |
16:08.46 | harryvv | I might just download burn and install 32 bit version of fc3. just to much is missing to make this a effective combined ws/ser |
16:08.58 | jontow | :/ |
16:09.31 | harryvv | or fc4 seems that it has alot of the fixes thats a part of fc3 |
16:09.58 | unabonger | any FreeBSD Asterisk gurus? Got a st()()pid question about a performance problem after changing from Shrike (RH9.0) to FreeBSD for my Asterisk server. |
16:10.08 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
16:10.18 | rephorm | vooduhal: seems like that's just the way it is. takes a few seconds for the card to answer the call, then dial the numbers, and then for the fxs at the other end to send back a ring tone |
16:12.40 | vooduhal | rephorm: No, this is from an FXO going out. |
16:13.17 | vooduhal | Ie, SIP phone calls asterisk, asterisk routes the call out via the FXO on a CO line to our DMS. |
16:13.38 | vooduhal | One thing that we've noticed that is really odd, is that it when it dials the card pauses for 1 second before the last digit. |
16:13.50 | vooduhal | So its like 555121---2 |
16:15.27 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
16:15.41 | unabonger | can anybody tell me how to check disk cache performance and read/write speeds on FreeBSD? I think my disk might be munging my performance during certain calls. Either that or Asterisk Server to NuFone latency probs. |
16:16.12 | tagore | hdparm -Tt ? |
16:16.32 | Stephnie | exten => _7.,1,Dial(SIP/${EXTEN:1}@outgoing) <--- I dont want the prefix 7 ....I want to number to be dialed as it is...what should I use? |
16:17.02 | unabonger | tagore ,thanks I couldn't remember. |
16:17.06 | tagore | :D |
16:17.07 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
16:17.07 | *** mode/#asterisk [+o bkw_] by ChanServ |
16:17.31 | unabonger | dangit |
16:17.42 | tagore | exten => _.,1,Dial(SIP/${EXTEN:1}@outgoing) |
16:17.45 | unabonger | that's how I would do it on RedHat, but that doesn't seem to work on FreeBSD |
16:18.13 | Stephnie | tagore: asterisk told me what to use:) it says ..use _X |
16:18.27 | tagore | ok |
16:18.39 | Stephnie | not working :S |
16:20.07 | tagore | and _. ? |
16:20.32 | Silik0n | _. catches everything include s, i, and h |
16:20.44 | tagore | is true |
16:20.47 | Katty | Hmmhesays: mrow? |
16:20.51 | Silik0n | _X. is what you really want |
16:21.13 | Silik0n | exten => _X.,1,Dial(SIP/${EXTEN:1}@outgoing) or something |
16:21.26 | Hmmhesays | Hey Katty, I got split before |
16:21.36 | tagore | unabonger http://lists.freebsd.org/pipermail/freebsd-questions/2004-October/061533.html |
16:21.40 | Hmmhesays | suddenly I was on asimov with like 3 people |
16:22.52 | Katty | asterisk does faxes :> |
16:22.53 | Katty | i'm all excited and bouncy now :> |
16:22.53 | Hmmhesays | I was rambling on about my weekend too.. to myself |
16:23.01 | Katty | heh |
16:23.22 | Hmmhesays | it sure does |
16:24.00 | Stephnie | Silik0n: _X. is not working |
16:24.08 | Hmmhesays | in fact if you have to asterisk endpoints you can store and forward |
16:25.17 | Beirdo | bouncy? |
16:25.21 | Katty | yes |
16:25.27 | Stephnie | _7. works but _X. doesnt work. |
16:25.36 | Beirdo | hehe, you really shouldn't put such thoughts in the minds of geeks :) |
16:25.38 | Hmmhesays | Katty: i should send you an episode of bonkers |
16:25.43 | Silik0n | Stephnie what kind of phone are you using? |
16:25.49 | Hmmhesays | i'm reminded of that every time you say "bouncy" |
16:26.35 | Silik0n | katty: A/S/L/ |
16:26.46 | Silik0n | Katty: oh sorry... forgot this was a tech chan heh |
16:27.02 | Stephnie | Silik0n : X-Pro |
16:27.56 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
16:28.00 | Silik0n | Stephnie can you paste that portion of your dial plan to pastebin.ca with a verbose capture from the console you you try to dial out? |
16:28.37 | Stephnie | ok |
16:29.44 | Stephnie | Silik0n: _X. means any prefix ...but there should be a prefix...for example: to dial 5551212 ...it could be X5551212 |
16:29.47 | Stephnie | I just checked it out. |
16:30.09 | Silik0n | _X. mean anything that starts with a digit 0 - 9 |
16:30.16 | Stephnie | yes.... |
16:30.19 | Silik0n | so 0123123123 or 0 would match |
16:30.27 | Hmmhesays | single digits don't match that |
16:30.31 | Stephnie | yes...correct but I need to dial number direct.. |
16:30.31 | Silik0n | ture |
16:30.35 | Stephnie | 5551212 .... |
16:30.39 | Silik0n | i meant 00 |
16:30.53 | Silik0n | _X would match single digits |
16:31.07 | Hmmhesays | ja |
16:31.23 | Hmmhesays | doncha know |
16:31.23 | Stephnie | yes. but I need to dial 5551212 without 0, 7 , 8 ... |
16:31.31 | Stephnie | without any prefix.. |
16:31.36 | Hmmhesays | pastebin.ca |
16:32.00 | Silik0n | once its on pastebin.ca let us see it |
16:32.29 | jansaell | do you men that you whant to dial wothout a prefix? |
16:32.36 | Silik0n | you might be better doing something like _9. for the pattern then you would dial 95551212 and ${EXTEN:1} would give you 5551212 |
16:32.38 | jansaell | without... |
16:32.48 | Stephnie | _X. means X5551212 ..... I need to dial 5551212 |
16:33.48 | jansaell | the ${EXTEN:1} drops the first character in the current extension |
16:33.54 | Stephnie | Silik0n: I dont want to send any extra digits ( _X. ) with number |
16:34.08 | jansaell | so if you dial 95551234 ${EXTEN:1} return 5551234 |
16:34.38 | lesouvage | I run festival --language cepstral_diane --server after running the cepsral script and copying to the festival dir. This is the errormessage: "Unsupported language, using English" SIOD ERROR: unbound variable : voice_rab_diphone. Any clue? |
16:34.52 | Stephnie | I think u guys dont understand what I mean.. |
16:34.53 | Stephnie | :) |
16:35.14 | Stephnie | I dont want the number (5551212) to be followed by any DIGIT..... |
16:35.42 | Stephnie | If I need to dial 5551212 then I should dial it 5551212 from my x-pro...not 75551212 or 95551212 |
16:35.45 | Stephnie | got it ? |
16:35.55 | Nugget | yes, we get it. |
16:36.00 | Nugget | we got it before |
16:36.01 | Nugget | we get it now |
16:36.16 | Stephnie | then what were you doing?? sleeping ? |
16:36.17 | Stephnie | :p |
16:36.19 | *** join/#asterisk DarkSpectre (Jesus2@adsl-69-155-22-158.dsl.tulsok.swbell.net) |
16:36.23 | DarkSpectre | in asterisk, can you have the option of when a user calls in, it can login to the system via a "username" and "password"? |
16:36.46 | Nugget | getting angry with the people who are trying to help you is not a path for success. |
16:37.05 | Stephnie | if is it for me ? then I am not angry :) |
16:37.25 | Stephnie | there is difference between anger and joke ;) |
16:37.32 | Stephnie | _X.,1,Dial(SIP/${EXTEN:1}@outgoing) <----- so ? |
16:37.41 | jansaell | ok Stepanie: exten => _XXXXXXX,1,Dial(SIP/${EXTEN}@outgoing) |
16:37.53 | Nugget | better to use NXXXXXX |
16:37.53 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
16:37.55 | jansaell | that match 7 chars |
16:37.56 | Seyr | heyas people |
16:39.10 | lesouvage | Stephnie: make an extension with the external numer you want to call and pass it to the zap channel exten => 1234567,1,Dial(Zap/1/91234567). |
16:39.43 | lesouvage | sorry i tried to cancel my suggestion but I pressed Enter by accident. |
16:39.45 | Nugget | What Stephnie is missing is the function of the ":1" in the variable. |
16:40.34 | Nugget | :1 makes it strip off the first digit, which I guess has led Stephnie to assume that a prefix is mandatory, since asterisk was always dropping that first digit. |
16:40.35 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.168.115.68.195.rev.coltfrance.com) |
16:41.02 | Nugget | but asterisk was just doing what the dialplan said. Drop the :1 and the while exten will be dialed |
16:41.06 | Nugget | s/while/whole/ |
16:41.52 | *** join/#asterisk DarkSpectre (Jesus2@adsl-69-155-22-158.dsl.tulsok.swbell.net) |
16:41.59 | DarkSpectre | dargh |
16:42.00 | Nugget | this whole ordeal is rapidly turning into "just tell me what to type" and it sounds like you could be well-served by spending some more time with the documentation. |
16:42.14 | orospakr | how would I go about disabling the mysql CDR module? just another noload? |
16:42.32 | Nugget | how variables are expressed in the dialplan, including using :1 to strip off a prefix, is fundamental asterisk knowledge that you should have before you dive too deep |
16:43.53 | Stephnie | exten => _XXXXXXX,1,Dial(SIP/${EXTEN:1}@outgoing) |
16:43.57 | Stephnie | not working |
16:44.04 | Stephnie | oops |
16:44.05 | *** join/#asterisk Kernel_core (Raph@9.229.dial-up.xter.net) |
16:44.07 | Nugget | of course not. |
16:44.10 | *** join/#asterisk Romik_ (~romik@212.143.5.146) |
16:44.14 | Stephnie | I got it |
16:44.14 | jansaell | no drop the :1 |
16:44.15 | Stephnie | I got it |
16:44.16 | Stephnie | :) |
16:44.18 | jontow | well.. the good news is.. x-lite works on netbsd to a point |
16:44.19 | Stephnie | :D yes |
16:44.25 | jontow | i still haven't been able to get it to register to my * though |
16:46.00 | Kernel_core | hi all , I am looking for equvalinet command ( output attenuation , input gain) in cisco for asterisk! For SIP and IAX channel !does asterisk have such command ? |
16:46.01 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
16:47.06 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
16:47.22 | MikeJ[Laptop] | anybody have a good source for nortel referbished gear? |
16:48.04 | Stephnie | Thank you guys.....:) working now.... |
16:48.05 | Stephnie | :) |
16:48.33 | Stephnie | Nugget: thanks....... |
16:48.36 | Stephnie | jansaell: thanks |
16:48.37 | Stephnie | bye |
16:49.56 | *** join/#asterisk ennuyeux72 (~ennuyeux7@83.146.53.34) |
16:53.22 | *** join/#asterisk cluecon (cluecon@wsip-68-99-73-32.tu.ok.cox.net) |
16:53.28 | DarkSpectre | quick question |
16:53.47 | DarkSpectre | i'm gonna have a box running to accept at least 6 lines. which card would be best suited for this type of project? |
16:54.18 | MikeJ[Laptop] | you want 6 lines on 1 card, then you are using a t1 with a channel bank |
16:54.30 | MikeJ[Laptop] | or you could use a couple tdm cards |
16:55.07 | DarkSpectre | so, a Wildcard TE405P would work then |
16:55.42 | Ariel_ | DarkSpectre, yes if your going to pluging 4 channel banks or 4 t1/pri lines |
16:56.07 | DarkSpectre | working on a project for a radio station. I'm not sure exactly what their incoming lines would be |
16:56.30 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
16:56.57 | PBXtech | its not normal when a phone re-registers to pick a new port each time is it? |
16:57.10 | Seyr | If I have SER in front of Asterisk and I am just passing outbound calls to *, what config changes do I need to make to the * box? |
16:58.04 | Seyr | so far all I have found is to set "autocreatepeer=yes" in sip.conf |
16:58.41 | dalabera | correct me if I wrong please: if I do a cvs checkout zaptel,libpri I'm getting latest dev code. Can I mix it with asterisk stable code ?? |
16:59.37 | jontow | well i had x-lite under netbsd registered and making a call; now im stuck at "Discovering Firewall..." when i start it |
16:59.40 | jontow | that sucks :) |
17:00.14 | cluecon | dalabera: it is not recommended to mix dev and stable. the results are unreliable at best. |
17:01.08 | harryvv | jontow, is that x-lite on the same network as the * box and you dialing out to another x-lite or how is your x-lite setup? |
17:05.34 | jansaell | me |
17:05.55 | *** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com) |
17:06.07 | Seyr | Anyone familiar with using SER in front of Asterisk? |
17:06.46 | *** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net) |
17:06.46 | harryvv | seyr, have you bothered to read the documentation? |
17:07.30 | Seyr | harryvv: whats a documentation? |
17:07.43 | unabonger | Wiki-Tiki-Tavi, baby! |
17:08.12 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
17:08.28 | MikeJ[Laptop] | Seyr, you have this really bad echo.. the same thing keeps coming out of you every hour or so for days now... |
17:08.37 | MikeJ[Laptop] | do you need help with SER and asterisk? |
17:08.55 | jontow | same network as * (192.168.2.0/24) |
17:09.11 | jontow | dialing to an internal extension on that particular *, not outbound at all |
17:09.14 | Seyr | MikeJ[Laptop]: You have a bad memory :-) yesterday I asked about difference between SER and OpenSER, today I am asking about config :-) |
17:09.37 | MikeJ[Laptop] | if you need help, I know some people who can help... |
17:09.54 | MikeJ[Laptop] | but not free |
17:10.00 | Seyr | MikeJ[Laptop]: I have SER up and running, just having problems with outbound calls going through * |
17:10.21 | MikeJ[Laptop] | well, if you are willing to pay, let me know and I will have them contact you |
17:10.27 | dalabera | cluecon thanks for advice |
17:11.12 | cluecon | dalabera: no problem. |
17:13.32 | *** join/#asterisk allanon (allanon@c-24-18-190-208.hsd1.wa.comcast.net) |
17:14.08 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
17:14.53 | *** join/#asterisk Nukemizer (~Nuke@67.137.28.165) |
17:16.19 | PBXtech | its not normal when a phone re-registers to pick a new port each time is it? |
17:17.50 | *** part/#asterisk Cramnoselo (fwuser@host-22.216-16-72.iw.net) |
17:22.58 | *** join/#asterisk fugitivo (~ajf@168.226.244.27) |
17:25.48 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
17:25.53 | yaaar | word |
17:25.54 | fugitivo | hello |
17:25.59 | cluecon | PBXTech: what kind of phone and define picka a new port. |
17:26.03 | cluecon | sentence. |
17:26.16 | PBXtech | 7960 out a nat |
17:26.27 | PBXtech | define what? |
17:26.45 | cluecon | what do you mean by it picks a new port? Is it using sip protocol? |
17:27.02 | PBXtech | every time it re-registers is has a new nat port |
17:27.04 | PBXtech | sip yup |
17:27.45 | PBXtech | it seems to loose the ability to dial to the phone |
17:27.54 | cluecon | sip uses 5060 to register and RTP 10000:20000 to carry voice packets. |
17:28.05 | cluecon | is it using a port outside that group? |
17:28.27 | PBXtech | 3819/3819 66.219.xx.xx D N 255.255.255.255 1125 Unmonitored |
17:28.57 | PBXtech | so * is calling the phone on 1125 |
17:29.18 | *** join/#asterisk ArkyLady (ArkyLady@adsl-66-142-125-19.dsl.ltrkar.swbell.net) |
17:29.40 | *** join/#asterisk hholzer (~hholzer@ip130.may.co.at) |
17:30.13 | PBXtech | and it will register after 3600 and pick a new port |
17:30.13 | PBXtech | doesnt seem normal |
17:30.46 | cluecon | i'm not familiar with the 7960 but does the phone have a setting that allows it to pick a random port? i know the grandstream i'm presently testing with does. if so, that needs to be disabled so it will use the standard port. |
17:31.10 | PBXtech | not aware of that setting |
17:31.46 | orospakr | \ |
17:31.51 | orospakr | oops! |
17:32.34 | *** join/#asterisk afrosheen (~afro@c-24-0-141-232.hsd1.tx.comcast.net) |
17:32.52 | file[laptop] | PBXtech: it's the NAT that is doing it... |
17:33.01 | file[laptop] | PBXtech: my friend's D-Link does it... |
17:33.08 | afrosheen | file[laptop]: what, 1 way audio? |
17:33.08 | file[laptop] | PBXtech: just enable qualify to keep the UDP mapping open |
17:33.16 | file[laptop] | no, port madness |
17:35.18 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
17:36.29 | hholzer | hi, can someone help me at a problem with a Digium TDM20B Card ? |
17:36.39 | Cresl1n | hholzer: what's up man? |
17:36.40 | hholzer | the wcfxs module didnt want to load and i got "ProSLIC 3210 version 2 is too old" in the message log |
17:36.47 | *** join/#asterisk CoffeeIV (~rristroph@mail.airlinksystems.com) |
17:36.53 | Cresl1n | hholzer: update to latest CVS |
17:36.59 | Cresl1n | (your zaptel) |
17:37.03 | afrosheen | proslic? |
17:37.07 | Cresl1n | either CVS-HEAD or CVS-STABLE |
17:37.12 | Cresl1n | I think both should work |
17:37.23 | hholzer | i am using zaptel-1.0.7 |
17:37.24 | Cresl1n | or check to see if there's a newer version of cvs-stable |
17:37.29 | hholzer | is this too old ? |
17:37.35 | afrosheen | what is proslic anyway |
17:37.38 | Cresl1n | hholzer: apparently :-) |
17:37.47 | Cresl1n | afrosheen: it's the line interface part on the fxs module |
17:38.42 | CoffeeIV | I am recording my calls through *. I am using the example from voip-info which does stereo mp3 files. I would also like to email the recording to the email address for that extension; is there an easy way to do that ? |
17:39.42 | CoffeeIV | I am about to write a perl script to parse voicemail.conf and do it "by hand", but I don't want to if there is another way |
17:41.23 | cluecon | CoffeeIV: use res_js |
17:41.57 | cluecon | http://www.pbxfreeware.com |
17:42.58 | Nuxi | or res_php with php's builtin ini parser. |
17:43.18 | *** join/#asterisk truz24 (~raydogg@12-220-103-82.client.insightBB.com) |
17:44.22 | Nuxi | http://us2.php.net/manual/en/function.parse-ini-file.php |
17:45.42 | *** join/#asterisk Maxxed (~max@cpe-70-114-238-9.houston.res.rr.com) |
17:45.51 | Maxxed | beh |
17:46.18 | Maxxed | hey, i have a issue with my cisco ip phones, the time in the upper part of the lcd disapears after a short while |
17:46.23 | Maxxed | i reboot the phones |
17:46.25 | Maxxed | and its back again |
17:46.34 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
17:46.52 | Maxxed | iv spokem with another asterisk user the otherday in regards to the issue, and they have the exact same problem |
17:47.05 | truz24 | I just installed an analog card and asterisk, what is a quick way to test to make sure it works? |
17:47.18 | truz24 | asterisk is running. |
17:47.19 | CoffeeIV | cluecon: thanks, i'm glad to know about those links -- I think I am going to write my own perl script though, it seems simpler than learning a whole new interface |
17:47.20 | Maxxed | the solution to bring the clock back that they found was to set the time to 24hr format and switch it back |
17:47.29 | Maxxed | and it will come back with out rebooting the phones |
17:47.35 | Maxxed | but it will even disapear after a while |
17:47.43 | wunderkin | maybe its a feature |
17:48.12 | Maxxed | anybody know about this issue? its gota be somthing cisco related |
17:48.18 | Maxxed | firmware issue or something? |
17:48.29 | cluecon | CoffeeIV: res_js is pretty simple to use and will make it really easy to do the record and email that you want to do. |
17:49.17 | Maxxed | im running sip 7-4-00 and bootloader pc03a300 |
17:50.36 | loud | Maxxed, you use any ntp server ? |
17:50.51 | cluecon | ~cluecon |
17:50.51 | jbot | it has been said that cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
17:51.52 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
17:52.15 | Maxxed | loud: nope, at least i dont think i do |
17:52.28 | *** join/#asterisk sangee (~rkuru@207.188.77.86) |
17:53.50 | Maxxed | ah, well, i do have the phones set to get there time from another ntp server |
17:53.56 | Maxxed | not directly from the asterisk |
17:54.16 | Maxxed | i have a ntp server on my gateway that the ipphones use |
17:54.26 | sangee | I want to pass through fax, how do i setup Asterisk for that |
17:54.43 | loud | reboot the phone, * 6 + settings without the ntp |
17:54.56 | loud | youll see the time. |
17:57.00 | *** join/#asterisk riksta (~rick@212.85.228.176) |
17:57.50 | Maxxed | loud: the ntp is what freaks the time out? |
17:58.13 | Maxxed | loud: there could be an issue with my firewall that isnt permiting the phones to "talk" to the ntp server |
17:58.50 | lesouvage | Can somebody please tells me what file hides the standard voice for festival? I run festival --server |
17:59.08 | lesouvage | I mean the setting of the standard voice. |
17:59.12 | cluecon | lesouvage: are you wanting to change the voice? |
17:59.23 | loud | waht i did works, to me. don't know about the firewall issue, my 7960 has a public address. |
17:59.46 | *** join/#asterisk ctooley (~ctooley@rrcs-24-227-212-181.sw.biz.rr.com) |
17:59.57 | *** join/#asterisk Trakk (~Trakk@adsl-8-244-88.mia.bellsouth.net) |
18:00.01 | lesouvage | cluecon: yes everything is in place except the change of the default voice. |
18:00.07 | ctooley | Does asterisk's agi "STREAM FILE" have the ability to stream from a file socket? |
18:00.36 | Nuxi | nope |
18:00.41 | ctooley | cat file.gsm > /tmp/SOCKET |
18:00.46 | loud | you unpolite people never say hi when joining this room. |
18:00.48 | loud | :) |
18:00.50 | Maxxed | loud: il check it out, thanks :) |
18:00.51 | ctooley | and have asterisk read from that? |
18:00.58 | Trakk | loud: sorry |
18:00.59 | Trakk | hi |
18:01.06 | Trakk | my client is about to quit |
18:01.07 | Maxxed | hi Trakk |
18:01.09 | Maxxed | :) |
18:01.16 | Trakk | IRC client that is |
18:01.39 | Trakk | be back on later then. |
18:02.24 | lesouvage | cluecon: I expected a .conf file where I had to change a line. The point is that I can't find out what file to change. |
18:02.41 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net) |
18:03.10 | pooh_ | Is there anybody who has a few minutes to check something about AGI with me pls ? |
18:03.27 | cluecon | lesouvage: one sec, I'm looking for mine. it is part of the festival conf not * |
18:03.35 | Nuxi | pooh_, whatcha got? |
18:03.59 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
18:04.56 | cluecon | lesouvage: you need to edit the file siteinit.scm. Mine is located under festival/lib. My voice line is as follows: |
18:04.56 | P-Chan | What would cause Asterisk to give a "Zap/2-1 is busy" when trying to call out? I keep getting "All circuits busy"? We have more than enough circuits available. |
18:04.56 | cluecon | (set! voice_default 'voice_cstr_us_jmk_arctic_multisyn) |
18:05.14 | ctooley | Nuxi, was your "nope" directed at me? |
18:05.24 | pooh_ | Nuxi: need to change the context for a call with 'set context' |
18:05.54 | Nuxi | yes, but I was thinking network sockets. don't know about unix sockets. |
18:06.14 | pooh_ | Nuxi: based upon a dbget value the context has to be set to continue dialing |
18:06.37 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
18:06.56 | pooh_ | Hi ManxPower, Eindhoven was fun! :-) |
18:06.58 | Nuxi | pooh_, so what is the problem SET CONTEXT <context> |
18:07.04 | truz24 | How do i install the zaptel module in ubuntu ? |
18:07.10 | truz24 | I got the x100p digium card |
18:07.20 | lesouvage | cluecon: thanks |
18:07.23 | pooh_ | Nuxi: The problem is that I do not have anything, just this wish |
18:07.29 | ManxPower | truz24, Um, Digium no longer sells that card |
18:08.19 | cluecon | lesouvage: your welcome. |
18:08.27 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-5-125.d4.club-internet.fr) |
18:08.40 | truz24 | ManxPower, so? |
18:08.54 | *** join/#asterisk frogy (~edmund@cm218-255-137-80.hkcable.com.hk) |
18:09.37 | pooh_ | ManxPower: got my idea *almost* working (HotDesk, but need to be able to set the correct context before commencing the dialplan withing the correct context |
18:09.40 | frogy | anyone has problem use the Meetme conference here? |
18:10.22 | pooh_ | NuxI: I guess it would be a 1 or 2 line agi before continue a call in the given context |
18:10.57 | frogy | My conference is having huge delay, and makes it less than usable. |
18:11.04 | [TK]D-Fender | Extensions.conf should have an "include" facility to link in other .conf files so as not to create 1 monstrous file..... |
18:11.05 | *** join/#asterisk clive- (~pirch@rrba-146-88-153.telkomadsl.co.za) |
18:11.32 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
18:12.14 | sivana | [TK]D-Fender: you can |
18:12.25 | [TK]D-Fender | how? |
18:12.28 | *** join/#asterisk bofh42 (~bofh42@p5482B957.dip0.t-ipconnect.de) |
18:12.36 | [TK]D-Fender | I must have missed that somewhere... |
18:12.44 | sivana | ; Macros File |
18:12.44 | sivana | #include macros.inc |
18:12.56 | sivana | the file.ext can be anything |
18:13.11 | cluecon | TKD: #include "my-ext.cluecon" |
18:13.26 | sivana | I dont' have quotes |
18:13.43 | frogy | can anyone give me some advise on how to solve the Meetme delay problem? |
18:14.05 | JerJer | meetme delay problem? |
18:14.15 | cluecon | quotes are optional i think. |
18:14.25 | pooh_ | Nuxi: ping ? |
18:14.27 | cluecon | they are shown in the docs, but i don't use them either. |
18:14.44 | Nuxi | pooh_: pong |
18:15.04 | Nuxi | what language are you friendly with? |
18:15.06 | cluecon | I have heard that the meetme delay problem may be addressed at cluecon. |
18:15.07 | bkw_ | frogy, hop to cluecon |
18:15.48 | pooh_ | Nuxi: for now the few agi scripts I have (stolen/use) are in perls and I have the perl stuff installed |
18:15.56 | heison | does anyone know if Asterisk can act as a SIP register proxy? i.e. when User-Agent A register with Asterisk via sip.conf, can Asterisk take the registration and register with SER for User-Agent A, such that incoming calls received by SER will be directed to Asterisk |
18:15.57 | [TK]D-Fender | Ok, I see it now |
18:16.01 | [TK]D-Fender | <- blind |
18:16.06 | Hmmhesays | I hear I still need to find money to go to cluecon |
18:16.08 | DrRighteous | cluecon? |
18:16.12 | bkw_ | asterisk is a B2B UA |
18:16.24 | heison | that is incoming calls to user agent A received by SER |
18:16.30 | *** join/#asterisk Lethargicclown (~chatzilla@ool-44c28c91.dyn.optonline.net) |
18:16.34 | cluecon | ~cluecon |
18:16.34 | jbot | i heard cluecon is http://www.cluecon.com - Cluecon is an Open Source Telephony Expo and Developer's Conference geared towards open source Telephony enthusiasts and developers around the world. There will be a full schedule of expert speakers as well as many presentations and booths from various telephony related businesses. |
18:17.03 | Nuxi | pooh_ I don't know anything about the perl stuff. |
18:17.04 | JerJer | heison: why not have ppl register to SER? |
18:17.13 | bkw_ | ya what JerJer said |
18:17.14 | bkw_ | hehe |
18:17.19 | pooh_ | Nuxi: a little background, I need to be able to direct a call to the correct context based upon a dbget related to a user, NOT a device (e.g. sip.conf) |
18:17.21 | *** join/#asterisk pjz (~pj@place.org) |
18:17.36 | *** part/#asterisk DrRighteous (~DrRighteo@68.199.175.49) |
18:17.37 | heison | because SER is a proxy but I need to control features provided by Asterisk |
18:17.45 | frogy | yeah, I use SIP phones to connect to the Meetme. At first, it works ok fine. But after 30 seconds, it starts come the delay. And it can be as much as 4 seconds for someone to hear each others. |
18:17.55 | heison | SER is currently running on hardware I have no control over |
18:18.04 | bkw_ | frogy, we told everyone how to fix it |
18:18.08 | bkw_ | and even tried to give back changes |
18:18.08 | pooh_ | Nuxi: so when a user places a call, a dbget value is checked, and accordignly the correct context is parsed instead of the context given in sip.conf |
18:18.18 | bkw_ | but they were rejected! |
18:18.21 | heison | I could do static routing on the SER, but like to avoid if possible |
18:18.27 | pooh_ | Nuxi: fine by me |
18:18.44 | anthm | pooh, are you using *choke* stable |
18:18.46 | pooh_ | Nuxi: php >= 4.3.x is ok |
18:18.49 | frogy | I read the post in the Mantis too. |
18:19.20 | heison | JerJer: what we currently have is SUA -> SER -> Cisco, and we want to implement SUA -> Asterisk -> SER -> Cisco |
18:19.26 | frogy | And applied the patch, but it doesn't help. I'm running 1.0.7. |
18:20.18 | pooh_ | OR.... I need to be able to change the cached dialplan and cached sip.conf values.... |
18:20.42 | JerJer | heison: why? |
18:20.45 | frogy | bkw_ where can I find the solution? |
18:20.58 | Nuxi | pooh_ you should be able to do that without agi. I think anthm can help. |
18:21.01 | cluecon | frogy: come to cluecon. |
18:21.21 | heison | JerJer: i want to provide call features with Asterisk |
18:21.22 | bkw_ | frogy, well the problem comes up because mark is chincy with threads in meetme |
18:21.22 | Nuxi | [[anthm is much smarter than Nuxi]] |
18:21.47 | bkw_ | if you dedicate a thread to playing sounds and other conf tasks it works great |
18:22.02 | anthm | aww, go on, Nuxi made res_php! |
18:22.11 | pooh_ | anthm: Greetings stranger, the force led me to you with my search for a solution |
18:22.13 | bkw_ | we have fought with mark over this.. he thinks it would be too resource intensive to fire up a thread |
18:22.53 | anthm | well using stable is your first obsticle, I'm pretty sure there is a secret bristuff for head |
18:23.04 | bkw_ | considering you can fire off 1000's of threads per second on a nice box .. |
18:23.05 | bkw_ | and do millions of malloc' |
18:23.05 | JerJer | another new thread for each new call? |
18:23.05 | bkw_ | no |
18:23.05 | bkw_ | one total thread for the entire conf |
18:23.06 | bkw_ | its job is to do things like play files |
18:23.06 | Hmmhesays | is there a patch floating around that might do that? |
18:23.06 | truz24 | Does anyone run asterisk in ubuntu? |
18:23.20 | anthm | if you do want stable and you want to use agi just rember the golde rule |
18:23.27 | bkw_ | JerJer, maybe mark was thinking two threads per caller. |
18:23.40 | anthm | in and out, asap |
18:23.56 | pooh_ | anthm: I guess my wish has nothing to do with whatever patch/version, but rather to change the given context entry within sip/iax.conf |
18:24.08 | anthm | agi should do minimal task and exit back to the dialplan never prolong an agi session |
18:24.42 | wunderkin | is that to get bfs if you use asterisk? what about gfs? |
18:24.43 | bkw_ | Nuxi, what was that link again |
18:24.45 | frogy | bkw_ so is there any way to get it to work in 1.0.7 ? |
18:25.09 | anthm | well, um it has to do with it a little since you could do it in a 1 line callflow in head =p |
18:25.17 | Nuxi | res_bf.c => http://pastebin.ca/15414 |
18:25.51 | bkw_ | where was the page about bf? |
18:26.15 | Nuxi | ~google bf |
18:26.23 | pooh_ | anthm: so where do I go from here and with 1.0.7, to change the context given in sip/iax.conf preceding processing a call by a user |
18:26.37 | Nuxi | http://www.muppetlabs.com/~breadbox/bf/ |
18:27.35 | *** join/#asterisk sudhir492 (~sudhir@pool-71-114-77-25.washdc.dsl-w.verizon.net) |
18:27.51 | heison | JerJer: I have a work around... although not what I wanted... I can have register lines in sip.conf for all the numbers I want to take from the SER over to Asterisk, if an incoming call from PSTN terminates on Asterisk and the SUA is not available, I can play Congestion |
18:27.59 | anthm | you want to alter the default context of the peer triggered by a call? |
18:28.24 | pooh_ | anthm: yes, exactly |
18:28.27 | Hmmhesays | why not just send them to that context |
18:28.30 | Hmmhesays | ? |
18:28.39 | Hmmhesays | based upon their callerid |
18:29.12 | pooh_ | Hmmhesays: not possible, give it a try and you will see that the cached dialplan will play you tricks based upong sip/iax.conf |
18:29.15 | frogy | bkw_ can u please tell me more about how to fix the meetme? |
18:29.39 | anthm | send them all to the same context and use dynamc switching logic to direct them to the desierd context. |
18:29.49 | Hmmhesays | heh, i hope that wasn't for me anthm |
18:30.30 | anthm | boing!! boing!! no its for pooh woo hoo hoo |
18:30.44 | pooh_ | so simply put, I need to set the context=.... value within sip/iax.conf from the dialplan to overrule the given values within sip/iax.conf, at the time of calling |
18:31.04 | anthm | no you are over-complicating yourself into a corner |
18:31.08 | Hmmhesays | pooh_ cmd gotoif would be one way to do it |
18:31.20 | anthm | take a step back and let the abstraction aroma sink in |
18:31.25 | *** join/#asterisk Romik_ (~romik@1.fix.netvision.net.il) |
18:31.26 | JerJer | heison: i don't follow the problem |
18:31.51 | pooh_ | Hmmhesays: yeah, RoyK advised me that, using accountcode and gotoif from a default context, but that does not work for now |
18:31.51 | Hmmhesays | anthm: i'm stealing that |
18:31.57 | frogy | can anyone tell me more about how to fix the meetme delay problem? |
18:32.05 | JerJer | what is this delay problem? |
18:32.20 | JerJer | i don't experience any delay on meetme |
18:32.25 | kajtzu | hmm. what delay problem? |
18:32.30 | Hmmhesays | pooh_ : then you are doing something wrong |
18:33.04 | bkw_ | its a problem when playing the enter exit tones in meetme |
18:33.05 | bkw_ | and or using the menu |
18:33.06 | pooh_ | yes |
18:33.06 | pooh_ | I know |
18:33.06 | pooh_ | :-) |
18:33.06 | bkw_ | I think its more voip related than TDM JerJer |
18:33.14 | frogy | JerJer: When using SIP phones to connect to the Meetme. it can has as much as 4 seconds delay for everyone to hear each others. |
18:33.30 | pooh_ | ok... let me explain in a bit more detail |
18:33.38 | pooh_ | * is very device oriented |
18:33.39 | *** join/#asterisk MikeJ_ (~ircatjerr@mi.origenfinancial.com) |
18:33.47 | pooh_ | I want * to be user oriented |
18:33.53 | pooh_ | so...... |
18:34.00 | *** join/#asterisk AgiNamu (~Michael@200.6.215.184) |
18:34.04 | AgiNamu | hello |
18:34.39 | frogy | I have used the q option, but delay is still there. |
18:34.40 | pooh_ | walk to a device, login and from that moment that device is your phone and ALL calls in/out will be controlled by your account, not the device |
18:35.03 | pooh_ | e.g. not controlled by sip/iax.conf |
18:35.07 | anthm | how many ways you want to solve that? |
18:35.12 | anthm | 2, 4? |
18:35.14 | AgiNamu | Hey, any downsides to storing a north american 10-digit number as a 32-bit int? |
18:35.15 | pooh_ | and I mean everything related to your account, vm, calls in/out |
18:35.16 | sudhir492 | I see the following error on the server when trying to send a fax using PAP2-NA: ast_rtp_read: Unknown RTP codec 100 received |
18:35.20 | file[laptop] | anthm: 42 |
18:35.53 | pooh_ | I am very close, just need to be able to set the correct context, and I am done |
18:36.02 | sudhir492 | AgiNamu: why would you want to do that? |
18:36.07 | DannyF | res_js is neat *grin* |
18:36.12 | Hmmhesays | sounds like a pain in the @ss pooh_ |
18:36.28 | file[laptop] | anthm: such insanity! |
18:36.35 | AgiNamu | sudhir, well, it is WAY less to store an int32 than a string. |
18:36.46 | AgiNamu | I'm guessing 28 bytes less |
18:36.51 | pooh_ | Hmmhesays: I know, I have been trying for 2 days now.... no solution yet |
18:36.57 | pooh_ | Hmmhesays: but I will not give up :-) |
18:37.05 | AgiNamu | and indexing, etc. will be smaller, faster, etc. |
18:37.13 | DannyF | AgiNamu, unless the number never starts with a 0 all is good ,) |
18:37.20 | kajtzu | AgiNamu: you're assuming north american numbers? there's the rest of the world too ;-) |
18:37.28 | AgiNamu | DannyF, yea, a north american 10 digit |
18:37.36 | Hmmhesays | pooh_ are you using realtime? |
18:37.36 | bkw_ | DannyF, did you see the orderstatus example I posted |
18:37.40 | DannyF | should be fine |
18:37.41 | AgiNamu | kajtzu, that is correct. only 10 digit numbers. this thing can't touch other numbers anyways |
18:37.50 | DannyF | bkw_, yeah, playing with it ;) |
18:37.52 | pooh_ | Hmmhesays: nope, all from the dialplan or a AGI script |
18:37.53 | AgiNamu | i thought it'd be fine, but i just wanted to see if i was missing anything hue. |
18:38.00 | bkw_ | did you see the format of the return data from the remote side? |
18:38.08 | anthm | here's 1 way. store all that info in a mysql or any db, and make frontends to all of the functions you need to offer that are agi that look up the info in the db based on the callerid when you log in and out you install your caller id into the db |
18:38.09 | Lethargicclown | does the wildcard tdm400p make the asterisk box work like an ATA? |
18:38.11 | file[laptop] | bkw_: OMG BECKY!!! |
18:38.13 | DannyF | myhash? |
18:38.27 | sudhir492 | AgiNamu: faster indexing etc. seems a valid reason. Not the storage. If you are sticking to US and Canada numbers only, then I guess no problem in storing as int |
18:38.28 | bkw_ | its asterisk config file like... [blah]\nname=blah\n |
18:38.36 | bkw_ | but the remote CGI can output that on request |
18:38.42 | bkw_ | and you can suck it up into myhash |
18:38.42 | DannyF | ahvalue pairs eh |
18:39.07 | DannyF | or whatchacallem ini blocks? |
18:39.15 | Hmmhesays | another way you could do it is use an agi to modify your sip.conf and reload sip |
18:39.18 | fugitivo | Lethargicclown: not exactly, ATA means analogic telephone adapter, asterisk with an fxo card is more complex than that |
18:39.24 | bkw_ | DannyF, yep |
18:39.30 | AgiNamu | sudhir492, storage in memory == faster |
18:39.35 | DannyF | very usefull |
18:39.39 | file[laptop] | memory is always faster... |
18:39.44 | AgiNamu | no, |
18:39.54 | AgiNamu | i mean, less memory , i.e., less storage, results in faster code |
18:39.57 | Hmmhesays | unless you've been drinking ;) <chuckle> |
18:40.08 | AgiNamu | keep as much stuff in cache as possible |
18:40.16 | Lethargicclown | Fugitivo: More complex how? |
18:40.21 | pooh_ | the problem is that * always starts with the context given in sip.conf/iax.conf thus device oriented, while I look at communiction as users using a stupid device, whatever device |
18:40.31 | AgiNamu | thanks everyone |
18:40.33 | DannyF | bkw_, sure makes some nice things a helluwa lot easier to make in a few minutes |
18:40.33 | anthm | you make them all use the same context |
18:40.55 | pooh_ | anthm: Tried that, not working because of the cahed dialplan |
18:41.11 | pooh_ | or I need to be able to change the cached dialplan |
18:41.12 | anthm | tried what |
18:41.33 | pooh_ | but I do not know the cached variable names if at all accessable |
18:41.33 | file[laptop] | Incense and Peppermints! |
18:41.43 | anthm | you have 1 context the sip device tries to call somewhere and hit's _X.1 to an agi that sets the proper goto context exten pri |
18:42.03 | anthm | _X.,2 goto's the contents of ${GOTO} that the agi set |
18:42.43 | pooh_ | anthm: exactly what RoyK adviced, and I treid, but within the 'end' context there is no real 's' exten because the context consists of includes only |
18:43.20 | pooh_ | AND I rather change the context from the 'start'... |
18:43.37 | anthm | one more time |
18:43.48 | *** join/#asterisk dwiltjer (~dwiltjer@63.227.22.62) |
18:44.09 | file[laptop] | it's great, I got it from bkw |
18:44.13 | anthm | you install the context in the db associated with the user where you surely are storing it anyway if you want to bother having users |
18:44.31 | anthm | you dial 1234 |
18:44.34 | DannyF | bkw_, did u try run festival stuff (or similar) though it? |
18:44.51 | dwiltjer | i have an error 'Ring/Off-hook in strange state 6 on channel X' and i've researched it a bit, it told me to turn off a few settings in my zap.conf file, but that dind't fix it |
18:44.54 | anthm | the catchall looks up your callerid in the db and sets the context variable |
18:44.57 | anthm | then |
18:45.18 | anthm | Goto(${MYCONTEXT},${EXTEN},1) in the next line |
18:45.24 | pooh_ | anthm: that is the idea, but how to make it work. I tried a 1Mio ways |
18:45.24 | anthm | tada dynamic contexts |
18:45.39 | dwiltjer | does anyone have any suggestions? its on a T1 with a TDMX100P card |
18:45.41 | anthm | i just practicly showed you working code |
18:46.05 | file[laptop] | anthm: you should know better then to help in here, it just makes you angry and raises your blood pressure |
18:46.13 | pooh_ | (20:41:50) pooh_: anthm: exactly what RoyK adviced, and I treid, but within the 'end' context there is no real 's' exten because the context consists of includes only |
18:46.37 | anthm | when you dial the login ext you go to an agi that does auth if it's a success you put the remote device in the db as a relation for the dynamic outbound |
18:46.57 | anthm | i read that |
18:47.08 | anthm | and said to myself, wtf does that mean and moved on |
18:47.23 | pooh_ | anthm: It sounds like the way to do it, but since the real trageted (outgoing) exten is included, it can not be found within the 'Goto' context |
18:47.37 | *** join/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
18:47.38 | dwiltjer | does anyone know anything about the 'ring/off hook in strange state' error? |
18:48.16 | yaaar | can anybody take a quick look at these 4 lines? http://pastebin.com/302599 ....they're error messages I get when trying to make an outbound call across my x100p.... |
18:48.44 | pooh_ | anthm: please explain a bit more about : when you dial the login ext you go to an agi that does auth if it's a success you put the remote device in the db as a relation for the dynamic outbound |
18:48.50 | anthm | if you say Goto(mycon,1000,1) and 1000 is in mycon by way of an include it will still find it |
18:49.34 | pooh_ | anthm: I have all the includes in mycon, but it will not find the exten, instead it will loop forever |
18:49.59 | MikeJ_ | file[laptop]!!!! |
18:50.03 | pooh_ | but then again, I am a real nOOb |
18:50.06 | anthm | you dial login ext, agi answers, it knows your remote sip id so it inserts it into the db in the same table as your profile so you know ext 1234 is SIP/3.4.5.6 |
18:50.06 | cluecon | file: report to room 996 for your meeting with the principal. |
18:50.33 | pooh_ | anthm: I have all of that working |
18:50.56 | pooh_ | anthm: even technology is set |
18:51.02 | pooh_ | sip/iax2 |
18:51.23 | pooh_ | cause the dial command NEEDS the technology of a device |
18:52.08 | anthm | a sip call reaction to the context= is not at all different from how the goto would have done it |
18:52.09 | pooh_ | anth: I am trying to seperate sip/iax.conf from what a user can do on the system |
18:52.33 | pooh_ | Goto requires a real exten or priority |
18:52.36 | sivana | when a file has * next to it, what does that mean? |
18:52.50 | pooh_ | while my contexts has includes and no real extens |
18:53.06 | anthm | does goto yell at you saying it doesnt exist ? |
18:53.39 | pooh_ | Nope it will loop. It will go to the right context, can not find it and loops, cause the fallback context says try again |
18:53.48 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-206-168-218-206.rockynet.com) |
18:53.56 | file[laptop] | nice dialplan logic. |
18:53.59 | *** join/#asterisk santiago (~santiago@63.245.86.198) |
18:54.09 | pooh_ | the dial command says to look for context,exten,prio |
18:54.16 | pooh_ | while context only has includes |
18:54.26 | pooh_ | no go |
18:54.26 | bkw_ | oh file dear |
18:54.35 | file[laptop] | bkw_: oh go jump off a bridge |
18:54.36 | bkw_ | can I see the pictures.. please please please |
18:54.41 | *** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com) |
18:54.52 | bkw_ | I take it you had a bad night? |
18:54.52 | bkw_ | NOT DIGITAL? |
18:54.56 | file[laptop] | no, not digital |
18:55.03 | fugitivo | why i get "sorry, your call can't be completed at this time.... " using broadvoice? any idea? |
18:55.05 | file[laptop] | I haven't slept in over a day |
18:55.09 | MikeJ_ | what kind of geek are you? |
18:55.15 | MikeJ_ | sad... so sad. |
18:55.17 | anthm | this is 1.07 ? |
18:55.26 | pooh_ | ye |
18:55.28 | pooh_ | syes |
18:55.32 | file[laptop] | okay people, answer me this... I'm in the prom stuff, how was I supposed to take pics? :P |
18:55.32 | pooh_ | damn, yes |
18:55.50 | file[laptop] | clone myself perhaps... |
18:56.15 | MikeJ[Laptop] | yes |
18:56.17 | MikeJ[Laptop] | ummm |
18:56.23 | file[laptop] | they did take pictures |
18:56.26 | MikeJ[Laptop] | yes |
18:56.33 | MikeJ[Laptop] | well, we want to see them |
18:56.36 | file[laptop] | my mother doesn't like the digital camera because she likes real prints, and they're supposedly expensive |
18:56.41 | Ariel_ | file[laptop], did you get lucky? |
18:56.42 | file[laptop] | so shutup before I hunt you down and kill you |
18:56.47 | file[laptop] | I'm not in the mood to bitch about pictures |
18:56.49 | MikeJ[Laptop] | :P |
18:56.56 | MikeJ[Laptop] | ok |
18:57.00 | MikeJ[Laptop] | your still loved |
18:57.03 | yaaar | does anybody have any idea what would cause * to say "could not create channel of type Zap'" when I try to execute something like 'exten => s,1,Dial(Zap/1/${EXTEN})' ? |
18:57.03 | *** join/#asterisk Derkommissar (Derkommiss@241.sub-166-139-71.myvzw.com) |
18:57.08 | Derkommissar | Hello |
18:57.20 | *** join/#asterisk Poincare (~jefffnode@dD5779BD2.access.telenet.be) |
18:57.20 | cluecon | yaaar: do you have a Zap channel 1 defined? |
18:57.29 | MikeJ[Laptop] | yaaar, it's inability to create a zap Channel |
18:57.53 | yaaar | MikeJ[Laptop]: gee, that's brilliant |
18:57.58 | MikeJ[Laptop] | ~lart cluecon |
18:58.02 | MikeJ[Laptop] | thanks |
18:58.57 | MikeJ[Laptop] | ~lart cluecon |
18:59.09 | pooh_ | anthm: so, goto a context that has all kinds of includes will not recognise the overall available extensions from the includes |
19:00.46 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
19:00.51 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
19:03.05 | Derkommissar | im having trouble compiling oh323 |
19:03.12 | Derkommissar | is there anyone here that can help me ? |
19:04.20 | *** join/#asterisk Darien (~darien@office.mvcard.com) |
19:04.25 | Darien | hi all, some questions on IAX trunking |
19:04.37 | Darien | I'm reading the IAX2 bandwidth page on voip-info.org and reading about codecs |
19:04.50 | Hmmhesays | yeah Derkommissar: use debian install the packages |
19:05.10 | Darien | now, the setup we're going to have is phones -> local PBX -> remote call router |
19:05.41 | Darien | it looks like ILBC is the most efficient in terms of bandwidth, but I'm wondering what drawbacks it has |
19:05.43 | anthm | are you sure, there is no code in goto to enforse the existance of the destination |
19:06.22 | Darien | wait, I misread that |
19:06.31 | Darien | Hmmhesays: was that to me? |
19:06.38 | Hmmhesays | Darien: yeah |
19:06.40 | Darien | yeah |
19:06.51 | Darien | we have one location in India |
19:06.54 | Darien | but most of our clients will be here in Montreal |
19:07.06 | Darien | I know |
19:07.12 | *** join/#asterisk jhiver (~jhiver@AStDenis-101-2-4-33.w193-253.abo.wanadoo.fr) |
19:07.22 | Hmmhesays | how many calls are you talking about? |
19:07.22 | jhiver | hi all |
19:07.23 | Hmmhesays | simultaneous |
19:07.34 | Darien | twenty at the most |
19:07.40 | Darien | at least, for the time being |
19:07.51 | Hmmhesays | and you are conerned enough about bandwidth to use iax trunking? |
19:08.24 | Darien | not really bandwidth |
19:08.26 | Darien | but I don't know what the state of broadband is in India |
19:08.26 | harryvv | there should be a channel on realestate thats where alot of the money is being made at :) |
19:08.28 | Hmmhesays | buy 20 g.729 licenses and call it good |
19:08.40 | harryvv | darien, bandwith is not a issue in some parts of india. |
19:08.45 | Darien | ok |
19:09.01 | Darien | it's probably safe to assume that his call centre is located in a not-an-issue part |
19:09.27 | Hmmhesays | you putting asterisk boxes at each location? |
19:09.28 | Darien | Hmmhesays: is g.729 worth paying for? |
19:09.31 | Darien | Hmmhesays: yes |
19:09.36 | Darien | and each will trunk to us |
19:09.39 | harryvv | ask him if there is alot of high tech buildings where he is at. if not and its poor area then no chance for good bandwith. |
19:09.45 | Hmmhesays | Darien: it's not worth it if you don't need it |
19:09.58 | Darien | well, I can just ask him what he has for an internet connection |
19:09.58 | Darien | Hmmhesays: that's what I thought, and it doesn't look like I need it |
19:10.02 | jhiver | g.729 give you good quality for little bandwith |
19:10.10 | Hmmhesays | if you can get away with ilbc then use it |
19:10.10 | Darien | GSM should be fine though for what we're doing |
19:10.17 | Hmmhesays | gsm does't sound quite as good |
19:10.19 | Darien | mm, it says it's robotic-sounding |
19:10.56 | Hmmhesays | IMHO ilbc sounds better |
19:10.57 | Darien | but I'll use whatever gives toll-quality sound for less than a megabit per 20 callers |
19:10.58 | harryvv | what do cell phones use mostly? |
19:10.58 | jhiver | if you can't use g.711 then really g.729 is second best... |
19:11.10 | Hmmhesays | i dunno, there are some intermediate codecs that sound pretty damn good |
19:11.12 | jhiver | plus it only uses 24 kbits/s (using SIP) |
19:11.25 | *** join/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl) |
19:11.32 | Darien | the thing is, the sites are going to be using vicidial/astguiclient |
19:11.34 | pooh_ | back, connection problem |
19:11.41 | Darien | which I guess can do 30 callers per 3 ghz machine |
19:11.43 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
19:11.54 | pooh_ | anthm: send me a message in the last 5 minutes? |
19:12.11 | *** join/#asterisk loick (~loick@APuteaux-151-1-27-74.w82-124.abo.wanadoo.fr) |
19:12.40 | Hmmhesays | ever listen to g.726? |
19:12.49 | jhiver | I didn't no |
19:13.03 | Darien | I've never heard any of it (or at least, I don't know what I've heard) |
19:13.04 | jhiver | I tried g.711, g.729, GSM, iLBC |
19:13.11 | Darien | the way we're doing billing is to basically modify astcc to do it, and use the trunk username as the account number |
19:13.21 | Darien | so we're using trunking regardless |
19:13.22 | Hmmhesays | try it out sometime |
19:13.25 | anthm | are you sure, there is no code in goto to enforse the existance of the destination |
19:13.28 | Hmmhesays | Darien: why are you using trunking? |
19:13.40 | jhiver | g.711 is excellent but you need so much bandwith! |
19:13.53 | harryvv | any of you using the ip500 |
19:13.59 | Darien | Hmmhesays: bandwidth efficiency and to have a single username/connection to watch for |
19:14.03 | Darien | which I guess would be the case anyway |
19:14.07 | jhiver | g.729 sounds almost as good (except for some 'f' that sound like 's' with certain voices) |
19:14.12 | Hmmhesays | Darien: indeed |
19:14.22 | Darien | anyway, bandwidth is enough reason for me |
19:14.27 | Darien | we do have to pay for it after all |
19:14.32 | Hmmhesays | I thought you said bandwidth wasn't an issue |
19:14.36 | Darien | well |
19:14.42 | Darien | it's not a technical issue |
19:14.49 | Darien | it's a financial issue |
19:14.55 | Hmmhesays | I didn't ask if it was a technical issue |
19:14.59 | Darien | and we're going to have a machine at their location anyway |
19:15.13 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
19:15.14 | Darien | so unless there are downsides to trunking....? |
19:17.13 | ManxPower | in 1.0.x trunking breaks the iax2 jitterbuffer |
19:17.23 | Hmmhesays | last I read about it was quite some time ago, and then it had jitter problems |
19:17.47 | clive- | manx, i belive cvs head that is fixed |
19:18.03 | ManxPower | clive-, *nod* that's why I said "in 1.0.x" 8-) |
19:18.56 | clive- | manx, how stable is CVS head at the moment >? |
19:19.38 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com) |
19:20.20 | Hmmhesays | what is head at now? |
19:20.56 | clive- | I am waiting for 1.2 but its taking its time appearing |
19:21.01 | Darien | hmm |
19:21.11 | Darien | do you guys think GSM would be sufficient for a call centre environment? |
19:21.21 | Darien | 100% clarity is not a requirement, nor likely an expectation |
19:21.36 | clive- | Iuse GSM, and its fine |
19:21.43 | Darien | good enough for me |
19:21.43 | *** join/#asterisk torment0r (~torment0r@md-frdrck-cmts3a-b-222.shphwv.adelphia.net) |
19:21.48 | clive- | I use,.,,, |
19:21.56 | Darien | I have another question about PRIs, which maybe someone can answer |
19:22.09 | Darien | or rather, long-distance in general |
19:22.17 | Darien | I'm looking for the best deal I can get on North American long-distance |
19:22.22 | Darien | American mostly |
19:22.36 | Darien | right now we have 0.9 cents/min, but does anyone know any cheaper/better rates than that? |
19:22.53 | torment0r | I have an X100P card, and for some reason at bootup the status is in RED Alarm |
19:23.07 | torment0r | what could cause that? |
19:23.12 | rephorm | torment0r: have you configured it with ztcfg? |
19:23.19 | torment0r | yep |
19:23.33 | rephorm | and you have the t1 plugged into it? |
19:23.34 | ender | Darien: CHeck out timewarner. |
19:23.41 | Darien | ender: ? |
19:23.44 | torment0r | it's just a single card, i have fxsks=1 |
19:23.53 | torment0r | in the /etc/zaptel.conf |
19:23.56 | torment0r | then i run ztcfg -vv |
19:23.57 | Darien | ender: for commercial dial-out ? |
19:24.11 | ender | Darien: we got a really good deal on a T1+PRI line (bonded, PRI with higher priority) and really good long distance. |
19:24.15 | rephorm | oh x100p. sorry |
19:24.26 | Darien | ender: what kind of price, if you don't mind my asking? |
19:24.49 | Darien | I'm just looking for a ballpark |
19:24.50 | ender | Darien: TW has a nifty little box that as you add T1 lines, you immediately get more bandwidth and you can choose how much of that bandwidth to allow for callout. It's all balanced and shared on the fly |
19:24.51 | rephorm | torment0r: do you have a line plugged into it? |
19:25.00 | ender | Darien: for the line cost or just the long distance? |
19:25.09 | torment0r | no, does there need to be for it not to be in red alarm? |
19:25.17 | Darien | ender: total |
19:25.18 | rephorm | torment0r: i believe so. |
19:25.22 | ender | Darien: and local long distance or long long distance? |
19:25.24 | torment0r | lemme give this a try |
19:25.34 | pooh_ | back, damn ISP |
19:25.35 | Darien | ender: lower 48 states |
19:25.37 | ender | Darien: actually nwo that I think about it, we used AT&T for our long long distance calling, TW for local long distance. |
19:25.42 | Darien | ender: very rarely alaska/hawaii |
19:25.52 | Darien | ender: it would just be for long-distance though, we don't have any physical presence in the US |
19:26.06 | Darien | so we'd be colocating in probably New York or Los Angeles |
19:26.20 | ender | Darien: for a 2T1 set and the PRI stuff, it was 1000 a month I do believe. Tack on the AT&T on top of that. |
19:26.43 | ender | Darien: and that was with 200 Direct Dial numbers. |
19:26.52 | torment0r | interesting.. now it's working.. awesome :) |
19:27.08 | torment0r | i have a vonage ATA box connected to an x100p box |
19:27.17 | torment0r | and it's working.. awesome |
19:27.18 | Darien | ender: so that's 47 lines and local long distance for $1k? |
19:27.36 | torment0r | now to test it with my cell phone |
19:27.37 | Darien | how local is the local long distance? |
19:27.53 | ender | Darien: there wasnt' really 'lines' to speak of, just bandwidth. |
19:27.57 | Darien | hrm |
19:28.20 | ender | Darien: time warner took the phone stuff and turned it into voip to go from our office to their office, and then it went back to voice. |
19:28.23 | Darien | ender: was this a digital telephony/VoIP solution? |
19:28.24 | Darien | aaah |
19:28.26 | Darien | I see |
19:28.30 | *** join/#asterisk brettnem (~brettnem@207.90.232.34) |
19:28.38 | ender | Darien: but I think it was only 23 lines because we used the second T1 for internet bandwidth only. |
19:28.40 | Darien | well how local was the local LD then? |
19:28.46 | ender | Darien: we could do any split we wanted with teh channels. |
19:28.49 | Darien | lower 48, or just in-state? |
19:28.53 | ender | Darien: I thin kit was just instate. |
19:28.56 | Darien | damn |
19:29.02 | Pete_Largo | Darien, how many channels do you need and how many minutes of use per month? |
19:29.11 | ender | Darien: TW said they didnt' have good long long distance, the big boys like AT&T were better with that. |
19:29.21 | Darien | ender: noted |
19:29.46 | Darien | Pete_Largo: 20 channels (at the moment; needs to be quickly expandable) and our first client is looking about about 480,000 minutes/month |
19:29.49 | *** join/#asterisk wozto1s (~wozto1s@host81-155-20-99.range81-155.btcentralplus.com) |
19:30.16 | Darien | well, he'll be starting at 144,000 but he's going to move up to 480,000 |
19:30.31 | Pete_Largo | ~ (24*60)*30 |
19:30.31 | jbot | 43200 |
19:30.31 | *** join/#asterisk Meaty (~cp_simbul@office.abi.ca) |
19:30.46 | Darien | ? |
19:30.47 | Pete_Largo | ~43200*20 |
19:30.47 | jbot | 864000 |
19:31.05 | Pete_Largo | just using jbot for some math :) |
19:31.07 | Darien | nod |
19:31.31 | Pete_Largo | Darien, are you going to colocate or will it be your location? |
19:31.48 | Darien | Pete_Largo: we have a colocation for the central call routing server |
19:31.53 | Darien | all the calls will be going through that |
19:32.09 | Pete_Largo | where is your colocation located (what state/city)? |
19:32.13 | *** join/#asterisk flotox (~jovan@host52-75.pool80183.interbusiness.it) |
19:32.31 | Robot_ | ~2^2 |
19:32.31 | jbot | 2^2 is, like, a bad question. You want 2**2. |
19:32.31 | Darien | Montreal,QC |
19:32.43 | Robot_ | ~2**2 |
19:32.43 | jbot | 4 |
19:32.48 | Robot_ | ~2**10 |
19:32.49 | jbot | 1024 |
19:32.51 | Darien | however, if we can save money, we will open up other call centres and put colocations there (e.g. New York, Los Angeles) |
19:32.53 | Robot_ | ~2**32 |
19:32.54 | jbot | 4294967296 |
19:32.58 | Robot_ | ~2**1000 |
19:32.58 | jbot | a number with quite a few digits... |
19:33.05 | Robot_ | ~2**100 |
19:33.05 | jbot | a number with quite a few digits... |
19:33.12 | Robot_ | ~2**64 |
19:33.12 | jbot | 18446744073709551616 |
19:33.19 | Darien | :/ |
19:33.34 | Darien | Pete_Largo: why do you ask? |
19:33.40 | h3x0r | you cant fit 480k minutes in 20 channels |
19:33.41 | Pete_Largo | I found a LA colocation for approx 550 month, in which we can get a single full PRI for approx 250 month |
19:33.48 | Pete_Largo | or a DS3 of PRI for 1500 |
19:33.51 | Pete_Largo | month |
19:34.04 | h3x0r | http://www.carrierone.net |
19:34.08 | brettnem | 480k is a few t1s if I recall. I think 140k is about 1 |
19:34.12 | Darien | h3x0r: 60 minutes per hour times 20 hours per day times 20 days per month times 20 channels |
19:34.13 | Pete_Largo | what kind of per minute rate do you want? |
19:34.17 | brettnem | depends on your application of course. |
19:34.24 | wozto1s | evening....don't suppose anybody has experience of using a TDM card with BT in the UK do they? I'm NOT a newbie btw and my setup IS working, but the TDM card isn't quite doing what it should! |
19:34.27 | Darien | Pete_Largo: less than $0.009/min USD |
19:34.40 | h3x0r | Darien: well, that is including non billable time i guess |
19:34.59 | Darien | h3x0r: yeah, it's 20 stations, 20 days, 20 hrs/day |
19:35.03 | Darien | which is a lot |
19:35.04 | harryvv | Darian, from where to where for that price? |
19:35.24 | h3x0r | what, are you having people call in or something? |
19:35.48 | Darien | harryvv: from Montreal, QC to the US (lower 48 is fine), or if the savings are significant enough, we can set up a colocation in the US (NY/LA) to handle it as well |
19:35.50 | Pete_Largo | Darien, is it all outbound or inbound or mixed? |
19:35.51 | brettnem | 20 hrs/day.. you got migrant workers or something? |
19:36.08 | Darien | brettnem: 20 hrs/day per station, probably 6 hour shifts or something |
19:36.14 | Darien | Pete_Largo: my impression is outgoing only |
19:36.14 | brettnem | ah |
19:36.29 | Pete_Largo | let me make a quick phone call... |
19:36.30 | Darien | or 10 hour shifts, I dunno what normal working days are in India |
19:36.32 | Darien | ok |
19:36.53 | h3x0r | anyway like brettnem said, you cant really bill more than 150k MMOU per T1 |
19:36.56 | h3x0r | if you arent using it 24/7 |
19:37.02 | h3x0r | if you were using it 24/7 then like 250K MMOU |
19:37.06 | h3x0r | er MOU |
19:37.10 | Darien | MOU? |
19:37.16 | h3x0r | minutes of usage |
19:37.28 | Darien | what do you mean, 'can't really bill more than' ? |
19:37.31 | brookshire | http://www.digium.com/index.php?menu=press/pr_2gen_firm |
19:37.33 | brettnem | yeah.. consider average call time and peak calling times |
19:37.34 | Derkommissar | can anyone here help me with chan_oh323 |
19:37.48 | brettnem | you simply can't put more than 250k minutes onto the T1.. |
19:37.51 | h3x0r | because, calls spend a lot of time ringing, busy, etc |
19:37.54 | brettnem | or somewhere around then |
19:38.12 | brettnem | or channels are just idle.. like a 4am |
19:38.13 | Pete_Largo | you could put approx 480k minutes per month on a T1 |
19:38.16 | Pete_Largo | 24/7 |
19:38.24 | Darien | well, this is an approximate |
19:38.25 | wozto1s | hope you havn't better luck than me Derkommissar...I didn't even get a hello :( |
19:38.37 | h3x0r | yeah if you left every call off hook |
19:38.39 | Pete_Largo | ~ ((24*60)*30)*23 |
19:38.39 | jbot | 993600 |
19:38.39 | brettnem | assuming a perfectly uniform calling pattern 24/7 |
19:38.41 | h3x0r | but how many uses are there for that |
19:38.51 | Ariel_ | wozto1s, hello |
19:39.02 | Darien | we won't know what the REAL numbers are until we start doing them |
19:39.05 | brettnem | you got to engineer it for your peak calling volume |
19:39.07 | Ariel_ | Derkommissar, hello how was your trip? And what's the problem? |
19:39.16 | h3x0r | i know what the real numbers are |
19:39.16 | h3x0r | ive sold like hundreds of t1s |
19:39.20 | Derkommissar | It was good :) |
19:39.23 | h3x0r | ive never seen anybody do more than 250k a t1 before ever |
19:39.26 | Derkommissar | h323 hates me:( |
19:39.29 | Pete_Largo | like hundreds? |
19:39.35 | brettnem | yeah, there is pretty good historical data for T1 usage |
19:39.39 | Derkommissar | but the trip was good |
19:39.39 | *** join/#asterisk _-Jon-_ (jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com) |
19:39.45 | _-Jon-_ | Hey |
19:39.47 | brettnem | Derkommissar: don't feel bad, h323 hates everyone. |
19:39.48 | Ariel_ | Derkommissar, did you not allready have this working before? |
19:39.51 | wozto1s | o right...don't ask about my problem then! :) |
19:39.55 | _-Jon-_ | Anyone else here having problems registering with BroadVoice? |
19:39.57 | brettnem | Derkommissar: you know, just yesterday, I found out that chan_sccp hates me |
19:40.48 | Darien | so what you guys are saying is that thanks to busy signals, etc. we probably won't hit 480k minutes of usage, right? |
19:40.53 | brettnem | wozto1s: you'll get a better response with a hello, then a question, and some active participation |
19:40.56 | Derkommissar | sccp ? |
19:41.22 | brettnem | Darien: it's more like typical usage.. you'll peak out durning the day, taper off at night.. |
19:41.23 | Derkommissar | here is the errors |
19:41.24 | Derkommissar | http://pastebin.com/302627 |
19:41.30 | Pete_Largo | Broadvoice is working ok for me right now... |
19:41.32 | Darien | brettnem: probably, yeah |
19:41.33 | wozto1s | i did say evening...which is short for 'Good Evening' to my knowledge...and I don't know anything about what your talking about so how take I take part????? |
19:41.36 | brettnem | Darien: lots of ringing and call failures.. no anwers.. |
19:41.47 | Darien | nod |
19:42.18 | Darien | like I said, we won't know real numbers until we start routing his calls |
19:42.20 | brettnem | it's unlikely that you'll use more than 250k per T1 unless you are doing something weird.. like not talking to PEOPLE but machines. |
19:42.26 | h3x0r | only in vegas... |
19:42.53 | brettnem | h3x0r: how about being 100% humidity and not raining.. only in texas... |
19:42.59 | h3x0r | hahah |
19:43.15 | brettnem | what is 100% humiditiy anyway.. seems like you'd be underwater |
19:43.29 | h3x0r | it means the air has saturated as much water as it possibly can |
19:43.43 | h3x0r | and i guess thats relative to berometric pressure |
19:43.50 | brettnem | and tempreture |
19:43.55 | h3x0r | well yeah |
19:43.59 | _-Jon-_ | Hmm, I wonder why my registration is failing then |
19:44.11 | h3x0r | so its 100 degrees, 17% and raining go figure |
19:44.21 | brettnem | Derkommissar: sccp is Cisco Call Manager emulation |
19:44.30 | Ariel_ | Derkommissar, your using asterisk head? |
19:44.32 | *** join/#asterisk pifiu (~myassisbi@208.205.181.170) |
19:44.41 | Derkommissar | yes |
19:44.57 | brettnem | ok.. back to work |
19:45.20 | *** join/#asterisk wasim (~wasim@203.81.220.100) |
19:46.48 | Ariel_ | Derkommissar, you downloaded all the PWlib's |
19:48.17 | Ariel_ | also your not on FreeBSD are you? |
19:48.30 | Darien | hmm |
19:50.05 | Derkommissar | yes |
19:51.05 | Ariel_ | Derkommissar, yes to which one? |
19:52.12 | Derkommissar | i downloaded all the pwlibs |
19:52.19 | Derkommissar | its all compiled as the readme says |
19:52.53 | torment0r | w00t |
19:53.00 | torment0r | i got it working like a ninja |
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19:54.44 | *** join/#asterisk cluecon (cluecon@wsip-68-99-73-32.tu.ok.cox.net) |
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19:55.47 | Darien | does anyone know what kind of a talk time ratio vicidial can get? |
19:55.55 | Darien | like 40 minutes out of the hour, 50, etc? |
19:56.43 | *** join/#asterisk MrbBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
19:57.03 | MrbBelvedr | i have asterisk set to start at bootup, and it works great |
19:57.14 | *** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net) |
19:57.17 | shido | ZzZZ |
19:57.20 | shido | I wanna be |
19:57.20 | MrbBelvedr | now, I want to make sure it never starts at bootup |
19:57.21 | shido | I wanna |
19:57.25 | shido | i wanna be , I wanna |
19:57.30 | shido | I wanna , I wanna |
19:57.34 | shido | I wanna be a hilton |
19:57.49 | cluecon | what is the best way to resolve the issue of "this driver does not support 1st gen modules" with a te405p card? |
19:58.23 | MrbBelvedr | how do i set it up to never start at boot? |
19:58.47 | Darien | remove the entry in rc.d or init.d |
19:59.08 | *** join/#asterisk bjohnson (~bjohnson@ip172-172.dsl.istop.com) |
20:00.03 | *** join/#asterisk darwin_35 (user@cas-11.ftsm-noc.mtlnk-129.valuelinx.net) |
20:01.25 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
20:02.26 | *** part/#asterisk wozto1s (~wozto1s@host81-155-20-99.range81-155.btcentralplus.com) |
20:04.45 | cluecon | has bug 4155 been resolved? |
20:05.44 | jdv79 | is there a test suite for asterisk? |
20:06.47 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
20:07.39 | *** join/#asterisk jmacz (~jmacz@63.245.86.234) |
20:08.49 | *** join/#asterisk proti (~seb@calypso.frankengul.org) |
20:09.34 | harryvv | has there been any bad issue between the ip500 and tftp or ftp server? |
20:11.02 | Ariel_ | harryvv, I use the IP-500 via ftp. Works great |
20:11.17 | lesouvage | cluecon: I finmally found out what it was. With a debian sustem the settings are in /etc/festival and not in siteinit.scm I have it up and running now although it sounds a little bit loud. |
20:11.38 | lesouvage | and distorted |
20:11.50 | *** join/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET) |
20:15.49 | *** join/#asterisk Umaro (~umaro@209.140.74.64) |
20:15.56 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
20:16.00 | Umaro | Hey guys, I've got a pri.. how do I set the callerid to private? |
20:16.20 | eKo1 | callerid=private? |
20:17.15 | MrbBelvedr | so if i remove the file called "asterisk" from the init.d directory, it will not start at boot? |
20:19.08 | *** join/#asterisk Tili (~Tili@202-133-67-239-dialup.sat.net.pk) |
20:19.39 | *** join/#asterisk Netview (~Netview@p54AF8279.dip0.t-ipconnect.de) |
20:20.06 | rephorm | MrbBelvedr: what os are you on? |
20:20.24 | MrbBelvedr | rechat - centos |
20:20.34 | MrbBelvedr | redhay |
20:20.37 | MrbBelvedr | fucj |
20:20.44 | rephorm | MrbBelvedr: man chkconfig |
20:20.49 | *** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
20:21.03 | Umaro | eKo1: so like SetCallerID(private) would work? |
20:21.25 | rephorm | MrbBelvedr: 'chkconfig asterisk off' will stop it from loading on boot |
20:21.49 | Nethab | Set(CALLERID(name)=private) |
20:22.15 | jhiver | ~seen p0lar |
20:22.15 | jbot | p0lar <~p0lar@64.254.225.62> was last seen on IRC in channel #asterisk, 95d 23h 45m 21s ago, saying: 'time to order..hehe'. |
20:22.34 | jhiver | ~seen Zeeek |
20:22.34 | jbot | zeeek <~Zeeek@Zeeek.sustaining.supporter.pdpc> was last seen on IRC in channel #asterisk, 15d 3h 23m 20s ago, saying: 'good luck'. |
20:23.04 | MrbBelvedr | i did chkconfig --del asterisk |
20:23.28 | MrbBelvedr | so that will completely remove the startup script right? |
20:25.10 | wizhippo | just the links in the runlevels not the init script |
20:26.48 | Umaro | Nethab: huh? |
20:27.03 | Nethab | the new func_callerid.so stuff |
20:34.20 | *** join/#asterisk DarthClue (~Headaches@adsl-70-244-228-14.dsl.tulsok.swbell.net) |
20:35.52 | Silik0n | anyone remember about how long ago ast_request_and_dial() changed? |
20:39.21 | *** join/#asterisk DarkSpectre (Jesus2@adsl-69-155-22-158.dsl.tulsok.swbell.net) |
20:39.37 | torment0r | in the extension.conf, how do you get the caller id information |
20:41.08 | rephorm | ${CALLERID} |
20:41.20 | torment0r | that's it? |
20:41.34 | torment0r | exten => s,1,${CALLERID} |
20:41.40 | torment0r | or whatever |
20:41.44 | MikeJ[Laptop] | torment0r, no, the rest are in readme.variables |
20:42.04 | rephorm | well, if you just want to see it in the cli, do a Noop: exten => s,1,Noop(${CALLERID}) |
20:42.32 | torment0r | i'm actually trying to use it from an AGI script, but i think that will work |
20:42.38 | torment0r | you r0x0r, thx :) |
20:44.04 | Cresl1n | r0x0r is pretty cool |
20:44.10 | rephorm | torment0r: from agi, you can do a GET VARIABLE to get CALLERIDNUM and CALLERIDNAME |
20:44.16 | Cresl1n | d0 u sp3ak in 1337? |
20:44.16 | *** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net) |
20:44.19 | Cresl1n | :-p |
20:44.31 | Silik0n | nac i fi tahw |
20:44.51 | *** join/#asterisk greg_work (~greg@d221-73-237.commercial.cgocable.net) |
20:44.52 | Silik0n | tsetseb eht si kaeps 73313 |
20:44.53 | *** part/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET) |
20:45.00 | Silik0n | 31337 sP[-@k !S +h[- |3E$T[-S~|~ |
20:45.05 | torment0r | lol |
20:45.44 | Silik0n | eVeN If !+ DOEs dr1ve 5()|\/|3 PeOPLe tO 1N$TA|\|!tY'S |)0OR S~|~Ep |
20:45.52 | Cresl1n | yeah ;-) |
20:45.55 | Cresl1n | that's the problems |
20:46.02 | Nivex | SSdsbCBzZWUgeW91ciBsMzN0IGFuZCByYWlzZSB5b3Ugc29tZSBiYXNlNjQu |
20:46.03 | Silik0n | heh |
20:46.20 | jontow | $foo |
20:46.23 | brimstone | hey, that's an idea |
20:46.29 | brimstone | let's md5 encode everything we say |
20:46.46 | `Sauron | That looked like base64 |
20:46.48 | Silik0n | ok so does anyone know what bug/patch went in for changing the ast_request_and_dial() call? |
20:47.12 | brimstone | Silik0n: a little cvs on diff action will tell ya |
20:47.12 | pifiu | hey guys wasup |
20:47.23 | pifiu | can anyone recommend a good stable motherboard for an asterisk pbx? |
20:47.34 | Nugget | a powermac. :) |
20:47.53 | brimstone | asus? |
20:48.18 | pifiu | those are stable for a pbx? |
20:48.24 | pifiu | just a standard one? |
20:48.38 | pifiu | or would you guys recommend like an entry level intel server board? |
20:48.52 | devel | pifiu, we're running an intel board, regular desktop. rock solid. |
20:49.00 | pifiu | which model? |
20:49.25 | Nugget | if you plan to do any zaptel hardware at all, you'll want to make sure the bios lets you diddle with interrupts. |
20:49.41 | `Sauron | I finally beat my close card into submission |
20:49.43 | Nugget | if no zaptel is planned it doesn't matter at all. just buy something stable |
20:49.50 | `Sauron | yeah, it's lame to use a clone card, but it works |
20:49.52 | devel | pifiu, i'd have to open the box back up to tell you. |
20:49.54 | pifiu | yeah for this new setup I wouldnt need zaptel anymore |
20:50.04 | Nugget | then get a powermac or an xserve :) |
20:50.06 | pifiu | lol |
20:50.09 | `Sauron | I didn't diddle interrupts, though. Linux did it for me. |
20:50.32 | pifiu | and what is asterisk heavy on when designing a new system? |
20:50.33 | Nugget | I had to turn off some of the onboard crap on my home machine because the bios kept wanting to share the interrupts |
20:50.40 | pifiu | would need to be handling like 5-10 calls at a time |
20:50.50 | pifiu | actually correction. 5 calls each server |
20:50.56 | pifiu | memory? cpu? cache? |
20:51.28 | *** join/#asterisk dos000 (~dos000@66.11.173.123) |
20:51.54 | `Sauron | Nugget, what's dB's preferred addressed name online.. dB or his irc nic? |
20:51.56 | `Sauron | nick |
20:55.01 | *** join/#asterisk `Sauron (sauron@h-69-3-12-50.hstqtx02.covad.net) |
20:55.06 | `Sauron | hum, that didn't work |
20:55.23 | *** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net) |
20:55.26 | Nugget | decibel or thecougar. |
20:55.31 | Nugget | I know he's bitched that decibel is taken here. |
20:55.43 | Nugget | I can't keep track of all his fscking nicks. it's confusin |
20:55.48 | Nugget | :) |
20:56.05 | Nugget | I wrote an irssi script that sprays privmsgs to all of them, that way I don't have to keep track |
20:56.19 | Nugget | http://slacker.com/~nugget/stuff/irssi/scripts/decimessage.pl |
20:58.12 | pifiu | ok update now |
20:58.25 | pifiu | i will need ONE zaptel card to be able to dial 911 in case of emergencys |
20:58.36 | `Sauron | rofl |
21:00.27 | `Sauron | so irssi is a commandline client, huh? |
21:00.30 | `Sauron | IIiiiinteresting. |
21:01.42 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
21:01.53 | cochi | irssi is like ircii or bitchx but cooler ;) |
21:02.09 | *** join/#asterisk bitslave (~psolomon@69-165-217-96.atlsfl.adelphia.net) |
21:02.13 | `Sauron | I hate bX |
21:02.19 | `Sauron | I <3 ircII |
21:02.26 | `Sauron | I might have to try out irssi |
21:02.27 | cochi | irssi is cool.just give it a tr |
21:02.27 | *** join/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
21:02.30 | cochi | y |
21:02.42 | cochi | used to chat with the developer for like 1.5 years back then. nice guy |
21:02.51 | Romik_ | anybody can advice about IAX2 server <-> server encryption? |
21:02.51 | `Sauron | Is there a way to convince it to run ircII scripts? :) |
21:02.59 | `Sauron | I might have to check out new versions of epic too |
21:03.26 | Nugget | irssi is the best. |
21:03.43 | Nugget | it does ssl natively, scripting in perl or tcl, solid multi-server support |
21:03.44 | Nugget | it rocks |
21:04.10 | `Sauron | we'll see |
21:04.34 | cochi | i wrote a patch for converting the timestamp to swatch beats. dunno if timo included it somewhen |
21:04.35 | file[laptop] | Romik_: it doesn't work, it'll crash your box |
21:04.39 | `Sauron | I'll have to figure out how to hack in all my mods to epic before I'll be able to use it fulltime |
21:05.35 | Nuxi | irssi aslo has php support. |
21:05.41 | Romik_ | file[laptop]: any advice what work? |
21:05.51 | cochi | does it have ruby, too? |
21:05.56 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
21:06.41 | Nugget | sure, but using php for that is beyond stupid. |
21:06.55 | cochi | using php is generally quite.. well.. anyway ;) |
21:07.25 | Nuxi | as opposed to perl? |
21:07.35 | Nugget | people who insist on using php for anything other than page-generation are falling victim to the "one tool in my toolbox" problem. |
21:08.06 | cochi | perl.. cough.. erm unqualified to give out any statements there ;) |
21:08.27 | cochi | php is just ... ugly. a badly designed language |
21:08.40 | cochi | with frequent unannounced api changes. thanks guys, had a lot of fun with these |
21:08.43 | Hmmhesays | php isn't that bad |
21:08.52 | `Sauron | it's not bad |
21:08.52 | PatrickDK | cochi, it may be, but it's still quick to code in |
21:08.58 | DannyF | Nugget, each to his own... |
21:08.59 | `Sauron | <PROTECTED> |
21:08.59 | Nugget | sure, but it was never intended to be a general purpose language and it shows. |
21:09.01 | jontow | yes; but the reason for that as i see is the quickly ballooning ability of the language |
21:09.03 | PatrickDK | but that does make it pretty ugly |
21:09.06 | cochi | and very error-prone. and not really straight forward |
21:09.12 | `Sauron | I like the combination of php for web, perl for non-web - in applications. |
21:09.13 | cochi | that's the reason why i love Ruby |
21:09.31 | PatrickDK | I don't like ruby |
21:09.39 | PatrickDK | heh, why yet another language |
21:09.55 | Nuxi | Why does everyone always insist that every application has a single language and every language has a single application? |
21:09.56 | cochi | it's not easy to get used to it. but the oop stuff in there is so clean <3 |
21:10.00 | Nugget | nobody's said anything religious. |
21:10.06 | cochi | yet |
21:10.07 | cochi | ;) |
21:10.09 | DannyF | yet :) |
21:10.12 | DannyF | lol |
21:10.13 | cochi | *g* |
21:10.19 | cochi | that just -had- to happen ;) |
21:10.27 | DannyF | yupp |
21:10.27 | PatrickDK | I just find it annoying to have 25 different compilers on the system is all :) |
21:10.28 | X-Rob | uh. Praise Larry! |
21:10.38 | Hmmhesays | coming from the author of res_php, lol |
21:10.58 | Nuxi | javascript is only for web browsers. |
21:10.58 | DannyF | PHP grew on me ;) I like bkw's res_js it's cute as hell |
21:11.02 | jontow | as i went about smackin' people down yesterday about the distribution wars.. each language is useful for something or other.. use what you're comfortable with and don't argue :) [im being gentle today 'cause i got too much to do] :D |
21:11.21 | Nugget | it's not a matter of comfort, it's a matter of appropriate use. |
21:11.35 | jontow | when i program something; if it takes 3 languages to get a task done efficiently and effectively; then i will use 3 languages |
21:11.37 | cochi | well don't get me wrong. i used to be a pro php developer since 1997 now. working with it every day, writing huge amounts of code. but that just showed my its deficies ;) |
21:11.38 | Nuxi | perl is the only language! |
21:11.42 | Nugget | letting comfort dictate language choice is what leads to people writing scripts in php. :) |
21:11.57 | tclark | Nuxi: http://pbxfreeware.com/res_js.tgz |
21:11.59 | Nuxi | * should have been written in perl! |
21:12.01 | Nugget | (or trying to write full apps in perl) |
21:12.02 | X-Rob | perl is the one true language |
21:12.03 | DannyF | Nugget, although it's quite convenient if you have stuff that dont require 1000 calls / sec to re use PHP code u have laying about |
21:12.05 | jontow | maybe so, but if that is what they're good with.. that is what they're good with |
21:12.08 | `Sauron | Just listen to nugget, people. |
21:12.13 | X-Rob | anyone who says difference is a heretic. |
21:12.15 | `Sauron | he knows that he talks about ;) |
21:12.17 | cochi | wasnt freepbx written in perl, nuxi?;) |
21:12.30 | dos000 | is there an /etc/init.d/asterisk for autostarting asterisk .. i am googling and i see somewhere it mentions is part of the source but i could not find it. |
21:12.39 | DannyF | dos000, safe_asterisk |
21:12.44 | Nuxi | linux should have been written in perl to run on windows! |
21:12.56 | dos000 | DannyF, was that ? is it in the source ? |
21:13.01 | X-Rob | nuxi, linux should have been written in vbScript. |
21:13.04 | X-Rob | Doh. |
21:13.08 | DannyF | created when u do make install |
21:13.09 | jontow | yeah, with activeX |
21:13.15 | Nuxi | dos000 look in the redhat folder for an init script. |
21:13.17 | cochi | linux shouldnt have been written.. but been -designed- ;) |
21:13.43 | Nugget | heh |
21:13.56 | DannyF | not much on the net today technology wise that didnt suck initially ;) |
21:14.03 | Nuxi | we should write laws against writting in anything but perl. |
21:14.06 | `Sauron | you're seeing the movie, right? |
21:14.08 | Nuxi | perl should be written in perl. |
21:14.08 | dos000 | Nuxi, i am on debian ... DannyF where does it put it. |
21:14.11 | cochi | danny really not? read the Geforce7800 reviews already? |
21:14.18 | `Sauron | Ack, ignore that here. :) |
21:14.19 | cochi | wasn't pascal 7 written in pascal 7 ? *mh* |
21:14.29 | cochi | dos on debian? /etc/init.d |
21:14.34 | Nuxi | dos000 dl the source and it is in <*src>/redhat |
21:14.41 | cochi | gotta enable startup in the /etc/defaults anyway |
21:14.42 | DannyF | dos /usr/sbin/safe_asterisk usually |
21:15.10 | *** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
21:17.32 | `Sauron | alright |
21:17.48 | dos000 | Nuxi, hopefully this is ok in debian based systems ? |
21:18.54 | DannyF | Nuxi, same thing diff name |
21:21.11 | Nuxi | I know, I know I should be running it on windows with http://www.asteriskwin32.com/ lol |
21:27.30 | pifiu | anyone running an intel entry level server board for asterisk? |
21:27.34 | pifiu | a 775 basd one? |
21:30.52 | harryvv | mmm my ip500 is failing to register against the asterisk box. |
21:31.08 | *** join/#asterisk Topslack (~topslakr@24-53-13-169.lndnnh.adelphia.net) |
21:31.37 | fugitivo | damn |
21:31.58 | fugitivo | all day, trying to figure out why broadvoice wasn't working, and it's because videosupport=yes |
21:32.02 | harryvv | what fug |
21:32.11 | harryvv | really |
21:32.21 | harryvv | that should not matter |
21:32.28 | fugitivo | it does! |
21:32.40 | rephorm | harryvv: how are you configuring it? (xml files or web interface?) |
21:33.50 | fugitivo | i'm going to put that on the wiki |
21:33.55 | FuRR_ | anyone know where i can find an emulator for cisco ip phones |
21:34.08 | FuRR_ | or does their SoftPhone support their XML applications |
21:34.34 | harryvv | well, the initial ip and what was on the phone has been done. astrisk sees the ip address trying to register against it but its failing. I have gone into the web interface in register and only provided the basics on the first block of information. |
21:34.45 | Nuxi | woohoo documentation going on to the wiki! |
21:36.11 | harryvv | Furr, I have theidentification block filled out but is the Server 1 really needed? |
21:36.50 | fugitivo | is this a bug? broadvoice not working when videosupport=yes ? |
21:36.52 | harryvv | also I am assuming the address in the identification is asking for the asterisk server ip? |
21:37.17 | rephorm | harryvv: set 'address' to the name in sip.conf |
21:37.36 | rephorm | (identification address) |
21:37.41 | harryvv | ohh |
21:38.15 | Nethab | broadvoice works fine and i have videosupport=yes |
21:38.22 | Nethab | but i don't have a video phone |
21:38.32 | harryvv | name as in regexten or username ? |
21:38.39 | fugitivo | Nethab: me neither, but i comment the line videosupport=yes, and it works |
21:38.39 | harryvv | or username? |
21:38.44 | fugitivo | Nethab: what asterisk version? |
21:38.54 | *** part/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
21:39.03 | Nethab | i use todays HEAD |
21:39.10 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
21:39.11 | fugitivo | maybe it's fixeds |
21:39.15 | fugitivo | it doesn't work on 1.0.7 |
21:39.21 | Nethab | but it's always worked with videosupport since january |
21:39.38 | Nethab | i just leave it on cause i thought it was cool |
21:40.19 | fugitivo | i don't really need videosupport, so i'll leave it commented until i upgrade to head |
21:40.51 | rephorm | harryvv: username |
21:40.57 | harryvv | k |
21:41.20 | harryvv | seems a little confusing why it would ask for a address in the registration |
21:41.23 | harryvv | of the ip500 |
21:41.44 | rephorm | harryvv: yeah. the polycom wording is different from asterisk wording |
21:42.13 | rephorm | its a sip thing. the full address of the phone is SIP:username@serverip |
21:42.23 | harryvv | i see |
21:42.29 | rephorm | or, in polycom terms, SIP:address@serveraddress |
21:42.37 | harryvv | and the point of having server 1 ? |
21:42.50 | rephorm | server1 is the asterisk box |
21:43.42 | harryvv | fantastic its working |
21:44.05 | harryvv | Thanks rephorm :) |
21:44.56 | rephorm | np |
21:45.02 | *** join/#asterisk jdv79 (~jdv79@u1057064.ul.warwick.net) |
21:45.15 | jdv79 | call failed to go through, reason 8 - is that common? |
21:46.01 | jdv79 | no! |
21:47.19 | pjz | how do I check to see if my zaptel cards are working correctly? |
21:47.43 | pjz | ah, hrm, nm |
21:47.47 | pjz | it looks like they're not configred right |
21:47.53 | harryvv | what do you have |
21:47.53 | pjz | I added a tdm400 to my existing te110p |
21:48.02 | pjz | and now neither is setting up correctly |
21:48.08 | pjz | it looks like maybe the tdm is being found first? |
21:48.22 | pjz | what should my zaptel.conf look like? |
21:48.32 | pjz | I've got teh stuff in it for the te110p, what do I need for the tdm400 ? |
21:49.34 | pjz | and how do I tell which one is providing which channels? |
21:50.03 | *** part/#asterisk wrmem (~wrmem@monnin-win.cso.uiuc.edu) |
21:51.47 | pigpen | anyone have anything good or bad to say about voipjet.com? |
21:52.33 | file[laptop] | yes. |
21:52.37 | VoIpMaster | does anybody knows a good instruction (step by step) for isntalling Asterisk REALTIME ?? |
21:52.41 | pigpen | ok..which one.? |
21:52.51 | file[laptop] | yes. |
21:52.56 | pjz | anyone? |
21:52.59 | pjz | anyoen able to help me? |
21:53.28 | riksta | i wish people would get the fuck off the -users list writing useless shit |
21:53.28 | pigpen | file[laptop], hmm...good or bad? |
21:53.36 | pigpen | I am hoping good. |
21:53.37 | file[laptop] | pigpen: see mailing list. |
21:53.44 | pigpen | pjz, not me. |
21:53.53 | pigpen | hmm..ok.. |
21:53.57 | *** join/#asterisk jaike (~a@203.131.137.76) |
21:54.11 | pigpen | a hint? |
21:54.26 | file[laptop] | nope, no hint |
21:54.34 | file[laptop] | let all the posts be your help! |
21:54.36 | VoIpMaster | does nobody habe experience in working with asterisk REALTIME? |
21:54.39 | file[laptop] | thus you can make your own decision |
21:54.51 | pjz | anyone know how to set up both a te110p and a tdm400p in the same box? |
21:55.28 | jaike | hi. newbie here. if i wanted to enable blindxfer and atndxfer, i enable it on features.conf and add Tt to Dial? |
21:55.53 | harryvv | what codec is ulaw anyway |
21:55.55 | jaike | exten => 312,1,Dial(SIP/312,20) |
21:56.05 | jaike | exten => 312,1,Dial(SIP/312,20,Tt) ? |
21:56.29 | file[laptop] | jaike: using # yes |
21:56.35 | pigpen | file[laptop], ok..I see some concerns...but nufone had issues too...are they doing better? |
21:56.41 | file[laptop] | er well using the internal asterisk mumbo jumbo transfer thing |
21:56.47 | file[laptop] | pigpen: not as many |
21:56.47 | VoIpMaster | okay, another way, does anybody here have experience with asterisk and MySQL .. ? or know a easy to use how to ? |
21:57.01 | file[laptop] | VoIpMaster: there's a guide on voip-info.org I do believe |
21:57.23 | jdv79 | hmm, 404 not found for a sip call |
21:57.44 | pigpen | file[laptop], well, I guess I will give nufone another chance. I think they have a better reputation... |
21:57.55 | file[laptop] | pigpen: or try http://www.asterlink.com/ |
21:57.57 | VoIpMaster | really ? file[laptop] ... i can't find it today when i'm searching for it on the website ... |
21:57.58 | jdv79 | Channel: SIP/bla@10.10.10.10 - does that looks right? |
21:58.13 | file[laptop] | jdv79: if you're not using a peer entry for authentication, yes |
21:58.26 | file[laptop] | VoIpMaster: are you gonna make me look? |
21:58.46 | jdv79 | why would i get a 404 then? |
21:58.53 | jdv79 | bla exists in sip.conf on 10.10.10.10 |
21:59.10 | file[laptop] | VoIpMaster: type realtime config in for a search string :p |
21:59.22 | jdv79 | even so, shouldn't it not go 404 and instead go right to the context as defined in the general section of sip.conf? |
21:59.28 | file[laptop] | jdv79: you don't dial a peer entry, you dial an extension in extensions.conf |
21:59.32 | twisted[work] | WORD |
21:59.32 | twisted[work] | i have the most awesomest warning message evar |
21:59.40 | twisted[work] | it reminds me of BKW at astricon atlanta last year |
21:59.42 | file[laptop] | twisted[work]: ... |
21:59.42 | twisted[work] | Jun 22 16:43:05 WARNING[13326]: chan_sip.c:1026 __sip_pretend_ack: Have a packet that doesn't want to give up! |
21:59.47 | file[laptop] | lol |
21:59.51 | file[laptop] | that's a new one |
21:59.51 | pigpen | file[laptop], hmm..well, I already have a pri, just looking for good reliable cheap long distance |
21:59.57 | twisted[work] | file[laptop], that's what I said |
22:00.04 | VoIpMaster | okay, i will try that tomorrow ... thx the first .. i will came back to you when i can't find a good howto :) |
22:00.21 | jdv79 | oops |
22:01.54 | *** join/#asterisk jief- (~jief@digitalized.ca) |
22:01.56 | file[laptop] | darn I helped, what was I thinking |
22:02.02 | jdv79 | can i not do a SIP call without finding a valid extention? |
22:02.02 | harryvv | pigpen what did your pri cost you |
22:02.10 | *** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net) |
22:02.14 | file[laptop] | jdv79: correct |
22:02.17 | VoIpMaster | okay guys i wish all here a nice night .. open your brains to bring the nicest PBX up !!! |
22:02.27 | VoIpMaster | cu ... we see us tomorrow! |
22:02.28 | jdv79 | then what is the context entry in sip.conf for? |
22:02.33 | jief- | hello, im going to deploy a call center soon with *. I was wondering, is it possible for agents to have not-available codes? like 611 bathroom, 612 on break, etc ? |
22:02.39 | file[laptop] | jdv79: actually your question confused me, rephrase it |
22:02.43 | file[laptop] | I'm not awake too much |
22:02.52 | pigpen | harryvv, $175/mo |
22:02.55 | harryvv | ji, as in a meeting or on break ? |
22:03.03 | jdv79 | i want all sip session to terminate in a context - that's all |
22:03.05 | jdv79 | just for testing |
22:03.13 | file[laptop] | jdv79: you can send them to a default context, sure |
22:03.16 | harryvv | t-1 pri 24 channels for 175 a month? |
22:03.22 | jdv79 | i tried but i get a 404 |
22:03.26 | jief- | harryvv: yes like what you can get with Symposium for example |
22:03.30 | file[laptop] | 404 means the extension doesn't exist |
22:03.33 | pigpen | yep...and for $20 I got 100 did's |
22:03.44 | shido | ok |
22:03.45 | pigpen | and yes..it is inbound/outbound. |
22:03.46 | harryvv | ji, polycom ip500 has those features in the phone. |
22:03.48 | shido | its FOOOOOOOOOOd time |
22:03.48 | shido | brb |
22:03.49 | file[laptop] | if you do a sip debug on the other box, it'll say where it's looking for the extension (which context) if you wanna make sure |
22:04.02 | jdv79 | ok |
22:04.05 | harryvv | pigpen who gave you that deal? |
22:04.13 | jief- | harryvv: ok, so its a phone related feature and not relative to the pbx? |
22:04.15 | file[laptop] | but anyway, your issue is completely configuration issue |
22:04.28 | harryvv | ji, dont know about that. |
22:04.33 | pigpen | My box is located in a telco hotel. I have over 600MB of bandwith with a telco...they like me. |
22:04.34 | jief- | harryvv: ok |
22:04.53 | jdv79 | got it |
22:04.56 | harryvv | pigpen, obviosly! |
22:04.58 | jdv79 | thanks file[laptop] |
22:05.07 | file[laptop] | jdv79: you're welcome |
22:05.09 | harryvv | pigpen, in a hotel? doing what |
22:06.02 | wunderkin | pigpen, what city? |
22:06.18 | file[laptop] | no one loves oily Homer! |
22:06.26 | harryvv | I have never heard of those rates for a t-1 |
22:07.03 | wunderkin | my ld t1 pri quotes were much less |
22:07.08 | pjz | if I have multiple zap devices, how do I tell which are which channels in zaptel.conf? |
22:08.17 | harryvv | wunderking, Only T1 rates I have seen here is 650-1,100 per month in BC canada. |
22:08.51 | wunderkin | well this is also in a telco hotel |
22:11.02 | InfraRed | http://www.ananova.com/news/story/sm_1436341.html |
22:12.27 | *** join/#asterisk meppl (mephisto@p54AAD58D.dip.t-dialin.net) |
22:12.33 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l02v-5-125.d4.club-internet.fr) |
22:13.13 | pjz | argh |
22:13.30 | pjz | does no one know how to deal with both a pri and a coupel fxo ports in teh same box? |
22:13.42 | pjz | they're both zaptel, but I don't know how to write my zaptel.conf |
22:14.53 | pjz | since it needs both span= and fxoks= |
22:19.40 | harryvv | I am impressed with the sound quality and lack of any echo on the polycom soundpoint ip500. |
22:19.48 | jief- | harryvv: Sprint has the best T1 prices in Canada |
22:19.49 | harryvv | intercom worked great. |
22:19.59 | harryvv | jief-: ohh really |
22:20.07 | jief- | harryvv: as far as i know yes |
22:20.16 | jief- | harryvv: at least around montreal |
22:21.41 | wunderkin | sprint has had the best pricing for me here too, im getting quotes through another reseller, he usually gets a little better deals |
22:21.46 | *** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net) |
22:22.04 | jief- | yeah we are a Sprint partner, might be why we get a better deal though |
22:24.15 | jief- | i was attending a Cisco conference on VoIP today. they have neat products |
22:25.07 | pawalls | A "custom Lucent 5E" switch would be the "5ess" switchtype, right? |
22:25.16 | pawalls | jief-, Cisco has insane stuff. |
22:25.22 | jief- | which reminds me, does * have support for encrypted communications? |
22:25.33 | jief- | pawalls: CallManager + IPCC is kinda cool |
22:25.41 | pawalls | When I toured a Cisco facility in 1999 (9th grade), we got to use several of their different VoIP phones. |
22:25.54 | pawalls | They have IP based cameras, TVs, a 10Gbit network, etc... |
22:25.55 | TokyoJimu | What's a good way to debug sip registrations? I have several "register" lines in sip.conf but "sip show registry" shows nothing. |
22:26.03 | pawalls | Everything was being controlled from a wireless PDA. |
22:26.06 | TokyoJimu | I have sip debug on. |
22:26.20 | pawalls | He was changing channels, repositioning the camera, starting a video conference.. wicked stuff. |
22:26.21 | jief- | pawalls: Cisco has mobile labs you can borough and use for tests |
22:26.25 | Romik_ | anybody can advice about IAX2 server <-> server encryption? |
22:26.50 | harryvv | pawalls, i think that was the year cisco started to put out voip products. |
22:27.08 | jief- | harryvv: they started in '96 |
22:27.13 | pawalls | harryvv, Yeah. I was taking CCNA courses. So we got to see the bleeding edge. |
22:27.27 | harryvv | ohh as far back as then. I took the CCNA course in 1998 |
22:27.44 | *** part/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net) |
22:28.15 | pawalls | Cisco is a pretty cool company.. somewhat evil at times, but they've got wicked technologies. |
22:28.22 | pawalls | And great product line. |
22:28.33 | jief- | yes and their conferences are cool |
22:28.39 | jief- | in their state of the art labs ;) |
22:29.08 | pawalls | Anyway, so my phone company says the switchtype on our incoming circuit is a "custom Lucent 5E", anyone know if that is the same as "5ess" switchtype ? |
22:29.23 | TokyoJimu | pawalls: yes it is. |
22:29.33 | pawalls | TokyoJimu, Great, thanks :) |
22:29.43 | pawalls | Finally replacing a 15 year old Comdial POS. |
22:29.51 | TokyoJimu | Comdial. Yuck. |
22:29.53 | pawalls | That machine is a nightmare... |
22:30.02 | pawalls | Sadly all of our phones are proprietary Comdial Impacts. |
22:30.08 | TokyoJimu | American technology at its worst. |
22:30.12 | pawalls | They were 300$ a piece... |
22:30.19 | pawalls | And I think all ~200 of them are going to go to waste. |
22:30.35 | TokyoJimu | What are you replacing them with? |
22:30.36 | jief- | at the office, it came up cheaper to buy all new SIP phones, and new Catalysts than upgrading our Lucent system |
22:30.58 | pawalls | We're using softphones for most of our personnel now. |
22:31.08 | pawalls | However, I can't for the life of me find a softphone with configurable buttons! |
22:31.18 | pawalls | It seems like this would be an obvious feature. |
22:31.19 | TokyoJimu | Are they happy with that? I don't think I'd want to use a softphone all day. |
22:31.30 | pawalls | Our sales people use headsets anyway. |
22:31.42 | pawalls | So it's no difference to them really. |
22:31.50 | jief- | ok, Cisco CallManager let you encrypt communication between phones using IPsec, is it possible with *? |
22:32.33 | pawalls | If only I could find a free softphone with programmable buttons... like if you click Button FOO it dials "*510" or something. |
22:33.47 | harryvv | what bit length encryption do credit cards use |
22:33.50 | pawalls | Anyone know of one by chance? |
22:33.54 | pawalls | jief-, Sorry, no clue :-/ |
22:33.55 | *** part/#asterisk pooh_ (user78@cust.15.241.adsl.cistron.nl) |
22:34.46 | jief- | that's one of the features that Cisco came up with that's really nice. and some other stuff they coded for the US DOD |
22:34.49 | pawalls | jief-, http://www.voip-info.org/wiki-Asterisk+Bounty+SIP+encryption |
22:34.58 | jief- | pawalls: yeah just saw that :( |
22:35.43 | harryvv | so what kind of encryption does it have |
22:35.51 | jief- | none |
22:35.56 | harryvv | ohh nice |
22:36.03 | pawalls | None. |
22:36.08 | pawalls | But you could use a VPN tunnel I guess.. |
22:36.16 | harryvv | so much for selling it to the federal goverment. |
22:36.29 | jief- | MD5 only for IAX |
22:36.55 | *** join/#asterisk Chrif (~fritsch@p54BF376B.dip.t-dialin.net) |
22:36.57 | jief- | pawalls: yes, but with Cisco, the communication between any Cisco phones can be encrypted, no need for VPN. it uses 3DES and AES128 |
22:36.59 | Chrif | Hi |
22:38.13 | pawalls | Hmm.. |
22:38.27 | pawalls | I wonder what kind of need there is for an xml configurable softphone. |
22:38.33 | Chrif | some time ago, someone told me a bug-number for asterisk with two IPs (Lan and WAN) - i cant find this entry in the bugtracingsystem anymore - anybody a hint? |
22:38.34 | pawalls | Where you can add custom buttons and such. |
22:38.55 | pawalls | Basically interface would be defined as an XML document or something.. |
22:38.56 | jief- | pawalls: call center app on your phone maybe |
22:39.04 | *** join/#asterisk khemir (user119@201.133.129.105) |
22:39.06 | harryvv | so the phones them self can do 3des and aes128 and asterisk will pass it. |
22:39.36 | jief- | harryvv: i doubt it, its configurable in CallManager, not on the phone as far as i remember |
22:39.43 | harryvv | i see |
22:40.19 | terrapen | http://www.bordergatewayprotocol.net/jon/humor/images/lic_plate.jpg |
22:40.56 | ender | terrapen: heh. |
22:41.12 | ender | terrapen: I"d be surprised if that is a real plate and the state let them get away w/ it. |
22:41.26 | terrapen | im not sure |
22:42.32 | khemir | hi |
22:42.52 | khemir | some have expirience in Ovislink whit h323? |
22:43.48 | *** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net) |
22:44.27 | jdv79 | are there best practices for tuning an install or is the answer immersion again:) |
22:44.39 | jdv79 | i just hit my proc limit |
22:44.48 | jdv79 | without even doing anything cool |
22:48.08 | jdv79 | is call setup and teardown intensive? |
22:48.26 | jdv79 | i saw an initial surge of load and then it tapered rapidly |
22:52.45 | *** join/#asterisk iq (~iq@204-26-74-11.omah.qwest.net) |
22:53.01 | iq | hi all |
22:55.40 | pawalls | Hmmm... |
22:55.55 | pawalls | In our current phone system, we have call parking (which I set up in Asterisk also) |
22:56.31 | pawalls | However, with our current system you can see which one currently have people waiting on them by looking at the phone. There is an indicator LED for parked call slot. |
22:56.46 | pawalls | Is there any way to do with with Asterisk and the hardware IP phones? |
22:56.59 | ender | pawalls: how would you define the slots? |
22:57.09 | ender | calls can be parked anywhere from 701 to 799 |
22:57.20 | znoG | is there an easy to setup stun server for linux? |
22:57.31 | TokyoJimu | How can I see current IAX registry? |
22:57.37 | znoG | im not sure i even need it. I have a SIP client behind NAT, however, the SIP server is NOT behind NAT. |
22:57.47 | TokyoJimu | There is a "sip show registry" but not an "iax show registry" |
22:58.01 | robl^ | pawalls, nt easily. unless you use a phone like Cisco 79xx using SCCP |
22:58.32 | Micc | TokyoJimu, iax2 show registry |
22:58.39 | TokyoJimu | Ahh. Thanks! |
23:03.07 | *** join/#asterisk Defraz (~t0tal@tim.ibccom.net) |
23:03.08 | *** join/#asterisk Nemesis760 (~nemesis@71.36.28.33) |
23:03.09 | TokyoJimu | Our voicepulse stopped working. We are registered but: chan_iax2.c: Host 66.234.228.170 failed to authent |
23:03.09 | TokyoJimu | icate as voicepulse-in-01 |
23:03.15 | znoG | so anyone know much about SIP, NAT, STUN? |
23:04.15 | fugitivo | it sucks |
23:04.58 | Nemesis760 | I'm placing an order for a channelized DS3 (PRI) to be terminated via M13 MUX -> Sangoma A104s/Asterisk and need some assistance with various line options... can someone here help. Will pay (reasonable) consulting fee if nec. |
23:05.19 | *** join/#asterisk outtolunc (outtolunc@adsl-69-110-5-162.dsl.pltn13.pacbell.net) |
23:07.24 | *** part/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
23:07.25 | Nemesis760 | Over 200 peeps here, and no takers?! |
23:07.36 | *** join/#asterisk Exstatica (exstatica@65.119.22.200) |
23:08.03 | *** join/#asterisk harryvv (~leonardo@S010600a0c93f6f7e.vs.shawcable.net) |
23:08.19 | jdv79 | national bz8 - oh i forget that shit |
23:08.25 | ilan | Nemesis760: have you tried the asterisk-biz list? |
23:08.59 | Nemesis760 | Not yet... was hoping for instant gratification. ;) |
23:09.27 | *** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv) |
23:10.19 | brookshire | yeah.. good luck with that |
23:10.19 | pawalls | robl^, That phone does support it? |
23:10.26 | pawalls | is SCCP the name of the protocol that does that sort of thing? |
23:11.02 | pawalls | And are there any other phones that have similar functionality.. |
23:11.05 | Nemesis760 | I got the basics... NI1 B8ZS ESF, etc. But no clue on "D-Channel Backup" or Wink, or # of outpulsed digits. |
23:13.13 | pawalls | Hmm... also.. |
23:13.20 | pawalls | What phone support automatic answer? :-/ |
23:13.21 | alt | pawalls: SCCP is Cisco's proprietary IP Phone protocol. |
23:13.52 | pawalls | With the current phones, if you dial an internal extension, it automatically picks up the phone. |
23:13.56 | pawalls | (on speaker) |
23:14.14 | harryvv | anyone have experaince with the cdr_addon_mysql ? |
23:14.56 | *** join/#asterisk imcdona (~imcdonald@imail.speakeasy.net) |
23:15.08 | harryvv | paw mm you mean pickup good question. |
23:15.23 | pawalls | harryvv, Yeah |
23:15.31 | robl^ | pawalls, sccp is tghe protocol. With the right tricks you can make it display any text message on the phone display or you can control lights / status icons next to LCDs. you can can create the function you want |
23:15.52 | harryvv | pawalls, I know there are some heard of them but dont recall. |
23:16.01 | pawalls | robl^, Interesting. |
23:16.09 | pawalls | robl^, Does asterisk speak sccp ? |
23:16.17 | Sedorox | yay for skinny |
23:16.37 | Sedorox | (aka sccp) |
23:16.45 | pawalls | Sedorox, Ah. |
23:16.49 | robl^ | pawalls, alost. its not as mauture as SIP or IAX2 support.. but there is chan_sccp and chan_skinny that both talk sccp to some degree |
23:16.50 | Nemesis760 | How 'bout another topic... Anyone here familiar with the recent CVS Commit dealing with X86_64 processors? It seems to me that the patch assumes that a X86_64 processor is going to be -march=k8. Doesn't this discount the existance of EM64T procs? |
23:17.27 | Sedorox | bah |
23:17.34 | Sedorox | there is no such thing as a EM64T |
23:17.34 | Sedorox | :p |
23:17.48 | Sedorox | whistles* |
23:17.54 | *** join/#asterisk ozus (~level3@h-67-101-176-189.lsanca54.covad.net) |
23:17.54 | robl^ | what about XY64Q? :) |
23:18.32 | ender | Nemesis760: the difference between k8 opts and em64t ops is virtually nonexistant. |
23:18.33 | Nemesis760 | Funny... I just bought 6 Dell Machines w/ 'em. And RHES for EM64T. |
23:19.00 | ender | Nemesis760: for most/all compiler cares, EM64T is just x86_64 |
23:19.12 | Nemesis760 | Ok... so I guess it'll be up to me to prove that.? Hopefully I prove you right. |
23:19.27 | key2 | does Freebsd support SMP ? (for dual xeon) or is it more recommended to set up a linux ? |
23:19.29 | jdv79 | is a two way g711 call about 70K or 150K of traff? |
23:19.49 | ender | Nemesis760: feel free. There was a large discussion about this on the fedora-devel-list. |
23:19.57 | brookshire | yes.. freebsd supports smp, but linux is better for asterisk anyways |
23:20.18 | xkev | jdv79, about 80k each way |
23:20.21 | xkev | 64k payload + headers |
23:20.55 | robl^ | but if you use "trunking" you and reduce the headers :) |
23:20.56 | Nemesis760 | Considering I'm supposed to go live in 2 weeks with a pretty large project, I'll be hammering the hell outta these boxes... If I still have a job after drop dead date, then I guess it works. ;) |
23:21.12 | xkev | and about 30pps, depending on how many voice frames per packet you configure (not sure if * even has an option for frames per packet) |
23:21.34 | jdv79 | so for 2way audio we're talkin' ~ 160K roughtly? |
23:21.45 | key2 | brookshire " can u explain why ? cause i think freebsd has a better kernel and better stability ? |
23:22.00 | Sedorox | key2: zaptel isn't as developed on freebsd |
23:22.01 | brookshire | cuz asterisk works better with linux.. |
23:22.10 | Sedorox | and asterisk has some bugs (granted I haven't ran into any) |
23:22.15 | brookshire | and it was designed for linux |
23:22.22 | key2 | oki |
23:22.27 | Nemesis760 | I remember reading that there are a couple 64bit instructions that are not implemented in k8 that are in EM64T... So is the consensus that if it's built for k8 it'll work on both, but not nec. the other way around? |
23:22.29 | Sedorox | the way it was written was around linux.. which has a totally different kernel design then freebsad |
23:22.30 | brookshire | and is supported only on linux |
23:22.56 | Nuxi | let me take this opportunity to remind everyone that asterisk runs on windows, but better on freebsd. |
23:23.13 | wunderkin | i hate linux, i did an emerge --unmerge ssmtp and now libstdc++.so.5 is missing |
23:23.15 | key2 | :0 |
23:23.27 | brookshire | oh.. you hate gentoo |
23:23.30 | brookshire | me too |
23:23.31 | brookshire | :D |
23:23.37 | wunderkin | no i hate all distros |
23:23.37 | Nemesis760 | ouch. |
23:23.56 | harryvv | run away from 64 bit if you are to use it as a work station |
23:24.28 | ilan | hmm, anyone using asterisk on the IBM openpower servers? |
23:24.36 | ender | Nemesis760: I think thats about right. adn those few differences really make no difference to applicaiton performance. |
23:24.40 | Nemesis760 | It is to be used solely as a server... in a pretty heavy environment. 8-T1's / box. |
23:25.22 | Nemesis760 | Need every performance advantage I can get. =) Can't wait for Sangoma's DSP cards to be released. 70 days and counting. |
23:25.34 | brookshire | digium cards are out now |
23:25.40 | brookshire | with echo can |
23:25.42 | ender | if I recall, the differences were for perhaps some desktop or graphical apps. |
23:25.48 | brookshire | and better firmware |
23:26.01 | ender | brookshire: including the E1/T1 cards? |
23:26.04 | brookshire | yes |
23:26.07 | ender | awesome |
23:26.08 | Nemesis760 | Yeah? Echo can + hardware channelization? |
23:26.18 | brookshire | yup |
23:26.29 | ender | are they just newer revisions of existing cards, or will we have to order something specific? |
23:26.29 | brookshire | and out perform the current sangoma cards |
23:26.30 | pawalls | wunderkin, I hate windows. I installed new soundcard drivers, and then Windows XP refused to boot (even into safe mode) after that :-P |
23:26.30 | Nemesis760 | Good to know. |
23:26.39 | pawalls | Required a complete reinstall. |
23:26.51 | brookshire | http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE411P&tab=details |
23:27.04 | brookshire | and they are certified |
23:27.14 | pawalls | We all have our preferences I suppose. Your problem is an odd one. |
23:27.25 | bkw_ | brookshire, where is your data to back that up... I just wanna see numbers |
23:27.44 | brookshire | bkw_: ask malcolm |
23:27.59 | bkw_ | and you should trust that opinion.. I want hard facts and figures here |
23:28.19 | brookshire | bkw_: why are you hating? |
23:28.25 | brookshire | :) |
23:28.32 | bkw_ | i'm not promoting either card.. but if you're going to say its better you better have numbers to back it up. |
23:28.35 | bkw_ | don't be like signate |
23:28.36 | bkw_ | :P |
23:28.41 | brookshire | they are there.. we did tests |
23:29.10 | brookshire | sangoma couldn't do a full e1 + g729 |
23:29.13 | brookshire | but digium could |
23:29.40 | brookshire | on a A dual-processor, 3-GHz 800FSB Intel XEON server with 1MB L2 cache |
23:29.52 | brookshire | this is without echocan |
23:29.57 | brookshire | card |
23:30.27 | bkw_ | and I still don't see hard performance numbers |
23:30.36 | brookshire | ehh.. oh well.. |
23:30.55 | brookshire | because i don't know them off the top of my head |
23:30.56 | bkw_ | you buy a porsche because you know its gonna go fast |
23:31.00 | bkw_ | it has that history of going fast |
23:31.14 | brookshire | if you want them.. ask malcolm |
23:31.19 | bkw_ | and I know mark did alot of work on those cards to improve performace |
23:31.29 | brookshire | http://www.digium.com/index.php?menu=press/pr_2gen_firm |
23:32.15 | Nemesis760 | Maybe a stupid question... but what the heck is "J1" |
23:32.16 | Nemesis760 | ? |
23:32.19 | bkw_ | japan |
23:32.23 | Nemesis760 | ah. |
23:32.24 | brookshire | Japanese T1 basically |
23:32.28 | brookshire | different protocol |
23:32.28 | bkw_ | and I like how they say they support J1 when infact know it doesn't work |
23:32.39 | brookshire | nope.. j1 support is in |
23:32.45 | bkw_ | no it doesn't work right |
23:32.50 | bkw_ | just ask benjk |
23:32.50 | brookshire | and tested on digium hardware |
23:32.57 | brookshire | won't work with sangoma |
23:33.01 | bkw_ | what won't? |
23:33.02 | bkw_ | J1? |
23:33.05 | brookshire | yup.. won't work |
23:33.07 | bkw_ | riight |
23:33.09 | bkw_ | ok |
23:33.12 | brookshire | until they update their driver |
23:33.25 | brookshire | using digium's code of course |
23:33.33 | bkw_ | and you work for digium? |
23:33.43 | brookshire | yeah |
23:34.03 | bkw_ | J1 does work on sangoma |
23:34.10 | brookshire | crestl1n finished the testing on J1 two weeks ago |
23:34.11 | bkw_ | benjk and I had a talk about it last week infact |
23:34.18 | bkw_ | it might work on this new board |
23:34.31 | MikeJ[Laptop] | ahhhhhhhhhh |
23:34.47 | MikeJ[Laptop] | PURPLE MONKEYS!!! |
23:34.51 | brookshire | all digium cards support j1 now |
23:34.57 | brookshire | even the old ones |
23:35.11 | *** part/#asterisk pussfeller (~todd@t1-rtc-woodlawn.rtcol.com) |
23:35.19 | bkw_ | well if you're going to be saying these thigns you better post hard facts and figures to back it up |
23:35.26 | bkw_ | the exact configs |
23:35.32 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
23:35.34 | bkw_ | if you wish to compare them |
23:36.37 | bkw_ | and for the record I don't hate |
23:36.50 | bkw_ | I have never personally bought any digium or sangoma hardware. |
23:36.58 | bkw_ | if you wish to hate go check out the home page of voipsupply |
23:37.07 | bkw_ | and look at the sangoma board with digiums name below it on the front page :P |
23:38.00 | bkw_ | I think they fixed it now |
23:38.16 | shido | . |
23:38.47 | bkw_ | the company I work for on the other hand has a boat load of 410's |
23:38.48 | bkw_ | haha |
23:39.03 | *** join/#asterisk torment0r (~torment0r@md-frdrck-cmts3a-b-222.shphwv.adelphia.net) |
23:39.07 | torment0r | i'm back |
23:39.11 | bkw_ | OH NO |
23:39.15 | bkw_ | :P |
23:39.18 | ender | *chuckle* |
23:39.19 | lesouvage | I have festival running with Diane from Cepstral. The last problem s that she is kind of shouting. Does anybody know how I can pump down the volume too a sophisticated level? |
23:39.54 | torment0r | this agi script i'm working is pretty 0wn4ge.. i'm using asterisk to deturn an ex-girlfriend :) |
23:40.05 | torment0r | good stuff |
23:40.59 | Nemesis760 | Does anyone here know about "D Channel Backup"? |
23:41.12 | MikeJ[Laptop] | torment0r, learn to type english.. cuz all that cr4p is l4me |
23:41.40 | MikeJ[Laptop] | Nemesis760, yes |
23:41.45 | torment0r | pfft.. don't be scared :) |
23:42.04 | MikeJ[Laptop] | torment0r, you are such a l337 hax0r... |
23:42.05 | Nemesis760 | I think I get what it is, just don't know how / if it's implemented, and weather or not I want it. |
23:42.18 | torment0r | i never said all that now |
23:42.39 | ender | MikeJ[Laptop]: it's a maturity measurement device. |
23:42.40 | Nemesis760 | MikeJ: Care to share some knowledge? |
23:42.47 | MikeJ[Laptop] | Nemesis760, do you have a question, or are you just speaking (err, typing) out loud? |
23:42.54 | MikeJ[Laptop] | nfas |
23:43.30 | MikeJ[Laptop] | lets you have 1 or more d channels running a bunch of pri's (usually want to keep it around 20 at most by convention) |
23:43.34 | Nemesis760 | I'm ordering a channelized DS3 (28 PRIs) and want to know if this is something I can / should take advantage of. |
23:43.48 | MikeJ[Laptop] | Nemesis760, that is where it is useful, yes |
23:43.57 | torment0r | lets you have?? what kind of english is that? |
23:44.00 | torment0r | lol |
23:44.07 | *** join/#asterisk Weezey (WeezeyD@206.210.109.233) |
23:44.08 | MikeJ[Laptop] | my english |
23:44.10 | MikeJ[Laptop] | :D |
23:44.25 | torment0r | exactly, 0wn3d |
23:44.46 | MikeJ[Laptop] | Nemesis760, the big advantage in that situation is you can use fewer chan's for D chan. |
23:44.54 | Nemesis760 | So another part to this question... does this imply that not every PRI has to utilize a D-Channel and that I can gain extra voice channels? |
23:45.05 | Nemesis760 | Preempted me. |
23:45.10 | MikeJ[Laptop] | so you can get another 20+ voice channels out of your ds3 |
23:45.12 | *** part/#asterisk jaike (~a@203.131.137.76) |
23:45.28 | Weezey | When I go from SIP into asterisk and out IAX, I get: un 22 19:40:37 WARNING[22181]: channel.c:1607 ast_indicate: Unable to handle indication 3 for 'SIP/2527-0a15' and the call doesn't ring. Any clues? |
23:45.36 | ender | Nemesis760: I think that is exactly what it means. |
23:45.41 | MikeJ[Laptop] | well, all of the pri's "use" a D channel, just not one on the same t1 |
23:45.59 | Nemesis760 | And this can be implemented this way in asterisk? Sorry if I'm being redundant... just want to be sure. |
23:46.17 | MikeJ[Laptop] | now if you are talking to a quad t1 card, you are going to need to have at least one (perferabally 2, primary and backup) per card |
23:46.41 | MikeJ[Laptop] | err.. maybe per server now, I am not sure with the recent 2nd gen enhancements. |
23:46.50 | Nemesis760 | Ok, and if 2 Quad spans/server... could I still get away with just 2 D Channels? |
23:47.07 | MikeJ[Laptop] | I am not sure.. what cards are you planning on using |
23:47.08 | Nemesis760 | Damn your good... 1 step ahead of my ???s. =) |
23:47.13 | sivana | quit |
23:47.16 | sivana | ack |
23:47.16 | Nemesis760 | Sangoma A104s. |
23:47.23 | MikeJ[Laptop] | give them a ring. |
23:47.29 | stormfr | i have rawplayer using 100% cpu. Load still ok anyway but is there a problem in handle thread with rawplayer/asterisk ? |
23:47.40 | Nemesis760 | Will do... thanks for the skinny. |
23:47.42 | MikeJ[Laptop] | stormfr, stable or head |
23:47.43 | MikeJ[Laptop] | np |
23:47.56 | sivana | hehe |
23:48.15 | stormfr | cvs head 2005-06-09 |
23:48.26 | stormfr | (currently upgrade to current) |
23:48.53 | bkw_ | doubt it |
23:48.58 | bkw_ | you must be doing something very wrong |
23:49.03 | *** join/#asterisk SwK (~SwK@12-219-156-206.client.mchsi.com) |
23:49.26 | Nemesis760 | MikeJ: Are you available for consulting? |
23:49.35 | Nemesis760 | Paid of course. |
23:49.46 | shido | go MikeJ[Laptop] |
23:49.57 | shido | before someone else snatches him up |
23:49.59 | *** join/#asterisk eric`` (~e@adsl-9-111-66.mia.bellsouth.net) |
23:50.04 | shido | :) |
23:50.27 | MikeJ[Laptop] | what do you need? |
23:50.31 | MikeJ[Laptop] | and where are you at? |
23:51.31 | Nemesis760 | I just want to have someone at my disposal to bounce questions off as I go into final ordering phase for this project... Want to make sure I get no surprises.... |
23:51.49 | Nemesis760 | I'm in Boise, Id. My Company is based out of San Diego, CA. |
23:52.15 | Nemesis760 | You can be anywhere.... pmt via PayPal, or whatever's clever. |
23:52.21 | *** part/#asterisk bonez41 (~aint@drjones.dsl.xmission.com) |
23:52.22 | brookshire | just call sangoma.. damn |
23:52.40 | Nemesis760 | brookshire: Have other questions. |
23:52.40 | brookshire | i mean.. you did buy the cards from them.. ask them |
23:52.59 | bkw_ | brookshire, this is #asterisk not #digium |
23:53.24 | brookshire | digium did start #asterisk, remember |
23:53.31 | Nemesis760 | bkw_: THANK YOU. :P |
23:53.35 | bkw_ | yes but without the asterisk community it wouldn't be here. |
23:53.46 | bkw_ | so alot of digiums success is here in the community. |
23:53.47 | brookshire | without digium, asterisk wouldn't be here |
23:53.54 | brookshire | they go hand in hand |
23:53.57 | bkw_ | no something else would be in its place |
23:54.02 | Weezey | chicken + egg = dinner |
23:54.14 | torment0r | yummie |
23:54.17 | Nemesis760 | Chicken / Egg... Ok Digium was the egg, and it did come first... But the community helped it hatch. |
23:54.58 | bkw_ | brookshire, without the work of Jim Dixon digium wouldn't exist. |
23:54.59 | brookshire | true.. but nemesis, really.. maybe 7 employees of digium are here in this chatroom.. and would help you.. but we're not going to support you if you do not support us |
23:55.04 | bkw_ | so do please give credit where credit is due. |
23:55.28 | bkw_ | brookshire, digium employees very rarely help anyone |
23:55.35 | brookshire | bkw_: without mark + jim, the tormenta would still be isa |
23:55.36 | bkw_ | I have yet to see much help from the digium staff in here. |
23:55.42 | brookshire | and not pci |
23:55.50 | bkw_ | thats neither here nor there. |
23:55.55 | brookshire | that's the truth |
23:56.04 | bkw_ | and the tor2 is GPL hardware |
23:56.07 | bkw_ | moving on |
23:56.13 | bkw_ | Nemesis760, you had some questions? |
23:56.15 | Nemesis760 | I'm happy to support Digium... would pay them for consulting... But I need a certified platform, for me that means Dell, and I'd heard of problems with Digium cards on Dell 2850 s. |
23:56.15 | torment0r | i talked with mark on the phone once, that was pretty awesome |
23:56.16 | brookshire | and was just as much designed by mark as it was jim |
23:56.18 | brookshire | joint project |
23:56.22 | torment0r | i was like.. i'm not worthy |
23:57.14 | X-Rob | well, I buy TDM400 cards rather than SPA2000 and 3000's, so, pfft. |
23:57.42 | bkw_ | I just use 10 dollar modems |
23:57.46 | Nemesis760 | bkw_: Wink / immediate outpulse... what's preferred and why? |
23:57.55 | bkw_ | you have a CT1 |
23:58.01 | bkw_ | why did you not order PRI? |
23:58.04 | X-Rob | bkw, you're in the US, you can. Our PSTN impedence is all screwed up compared to yours. |
23:58.12 | bkw_ | X-Rob, true |
23:58.28 | Nemesis760 | I am ordering PRIs... Does that mean the Wink question is irrelevent? |
23:58.28 | bkw_ | Nemesis760, Every CT1 I have ever used is Wink start |
23:58.33 | bkw_ | yes |
23:58.36 | bkw_ | you don't wink a PRI |
23:58.40 | Nemesis760 | Ok... that was easy. |
23:58.49 | brookshire | digium supports dell 2850 |
23:59.02 | brookshire | and has been throughly tested with Asterisk Business Edition |
23:59.04 | brookshire | for all cards |
23:59.10 | bkw_ | brookshire, does digium have a good list of systems that have passed to publish? |
23:59.15 | torment0r | asterisk business edition??.. this is news to me |
23:59.20 | brookshire | i have three.. i'll get them |
23:59.23 | bkw_ | its a closed src version of asterisk |
23:59.23 | Nemesis760 | Number of outpulsed digits, is this a user preference, or dictated by *? |
23:59.25 | SwK | Nemesis760: just remember on a 2850 TDM400s w/ FXS wont work (no power inside for the SLICs and most of those clones of the x100p kill the PCI buss and the box wont boot |
23:59.45 | bkw_ | Nemesis760, that shoudln't matter its PRI |
23:59.51 | torment0r | that's pretty interesting, when did this branch come about? |
23:59.59 | bkw_ | torment0r, go check their website |