| 00:00.34 | Moc[Toronto] | yes |
| 00:00.39 | blitzrage | too bad I'm in Seattle |
| 00:00.48 | blitzrage | (area*) |
| 00:01.15 | Druken | area? |
| 00:01.18 | blitzrage | Oakville |
| 00:01.22 | Druken | ahh cool |
| 00:01.26 | Moc[Toronto] | ha |
| 00:01.28 | blitzrage | aka tOakville |
| 00:02.07 | Moc[Toronto] | if mark told me when he would arrived... |
| 00:02.17 | *** join/#asterisk zione (~zione@62-101-126-208.fastres.net) |
| 00:03.57 | zione | hi!i've just got into this!I'd like to use it for in-house training of a call-centre staff... any suggestion/pointer/addon? shall i ask on a different chunnel? |
| 00:04.31 | blitzrage | zione: read, read read |
| 00:04.35 | blitzrage | zione: oh, and read too. |
| 00:04.48 | j_vianna | quit |
| 00:04.49 | Druken | btw, read :) |
| 00:04.51 | *** join/#asterisk mithro (~tim@150.203.247.9) |
| 00:04.56 | Druken | ~voipinfo |
| 00:04.58 | zione | blitzrage, that's what all those firefox tabs are there for... |
| 00:05.01 | bugbot | voipinfo is assigned nothing and reported nothing. |
| 00:05.11 | blitzrage | zione: ;) |
| 00:05.22 | blitzrage | ~voip-info |
| 00:05.23 | jbot | rumour has it, voip-info is the Voice Over IP wiki. It is a community resource which will answer all of your questions, from Asterisk to ZTDummy. You can find it over at http://www.voip-info.org - well worth bookmarking |
| 00:05.23 | bugbot | voip-info is assigned nothing and reported nothing. |
| 00:05.46 | zione | blitzrage, ok, thanks have not to start from that low, |
| 00:06.16 | zione | i wonder if astguiclint would be enough for in-house training without external lines... |
| 00:07.29 | zione | i can't find really much info on * running with no real lines (and if I can fake some) |
| 00:07.52 | Nugget | you're not looking very hard, then. |
| 00:08.10 | zione | so you confirm, i'd better start deploying |
| 00:08.35 | blitzrage | file[laptop]: !!! yo |
| 00:08.53 | blitzrage | brb |
| 00:13.57 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 00:17.31 | *** join/#asterisk MatsK (~matsk@107.80-202-57.nextgentel.com) |
| 00:24.01 | *** join/#asterisk ross_cav (~ross_cav@60-240-47-244.tpgi.com.au) |
| 00:25.08 | *** join/#asterisk mindamp (~unknown@AC8CF84A.ipt.aol.com) |
| 00:25.33 | mindamp | this PBX is awesome |
| 00:25.42 | L|NUX | :) |
| 00:26.13 | mindamp | i don't have to pay rent on an office anymore |
| 00:26.13 | mindamp | :-) |
| 00:26.18 | kam_ | is the astrisk config all comand line btw? |
| 00:26.54 | nestAr | what's a good phone that i can disable call waiting on? |
| 00:27.07 | nestAr | Obviously, these polycom's suck at that |
| 00:29.07 | L|NUX | nestAr : man |
| 00:29.20 | L|NUX | nestAr : can you help me in installing * on non-root |
| 00:29.29 | L|NUX | i read at voip-info.org |
| 00:29.35 | nestAr | what's the problem? |
| 00:29.37 | L|NUX | but it giving me error |
| 00:29.42 | nestAr | error? |
| 00:29.48 | L|NUX | yeah |
| 00:29.51 | nestAr | what error |
| 00:29.51 | L|NUX | permission error |
| 00:29.55 | nestAr | from? |
| 00:30.02 | L|NUX | now i can't change permission of all folders |
| 00:30.02 | L|NUX | like |
| 00:30.17 | L|NUX | well /usr/lib permission |
| 00:30.50 | nestAr | ls -lad /usr/lib |
| 00:31.06 | L|NUX | k |
| 00:31.43 | L|NUX | [asterisk@www asterisk]$ ls -lad /usr/lib |
| 00:31.44 | L|NUX | drwxr-xr-x 29 root root 12288 Apr 19 16:48 /usr/lib |
| 00:33.12 | L|NUX | so |
| 00:33.57 | Nugget | so install someplace else if you are not root. |
| 00:34.13 | L|NUX | hmm |
| 00:34.15 | Nugget | /usr/l|b/ or something stupid like that. |
| 00:34.16 | L|NUX | ok |
| 00:34.17 | nestAr | even the non-root instructions assume that you have root |
| 00:34.33 | nestAr | but you want to run the program as a user other than root |
| 00:34.39 | L|NUX | yeah |
| 00:34.46 | L|NUX | you means |
| 00:34.47 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:34.51 | L|NUX | i can install it as root |
| 00:34.59 | L|NUX | and then change owner ship |
| 00:35.06 | L|NUX | and run it as non root ? |
| 00:35.22 | nestAr | yes |
| 00:35.27 | Nugget | yes, just like practically every other unix service. |
| 00:35.31 | nestAr | that's what that document is geared to |
| 00:35.33 | L|NUX | ;) |
| 00:35.36 | L|NUX | ok got it |
| 00:39.38 | *** join/#asterisk ToyMan (~konversat@user-12lcqur.cable.mindspring.com) |
| 00:40.20 | Elwell | ok - Quite like this asterisk thingy ;-p so going to play at work with it. will using an avm fritz (BT speedway) isdn card work with ISDN2e? |
| 00:41.20 | Veto | Is there any difference between a cisco 7960g and a 7960g-ch1 for *? |
| 00:41.47 | Nugget | I'd never even heard of the -ch1 until you mentioned it earlier. I have no clue, sorry. |
| 00:41.53 | blitzrage | me either |
| 00:42.03 | blitzrage | whats the difference? |
| 00:42.18 | Veto | neither had I until I found one in pricegrabber! |
| 00:44.13 | Veto | I find mention of the ch1 being with Internaltionl icons and something about manager license |
| 00:44.27 | Nugget | the "g" is what brings the international icons. |
| 00:44.43 | Veto | but since this is my first 7960, i'm not sure. |
| 00:45.25 | Nugget | the 7960 has words on it and the 7960g (global) has the silly little pictures. |
| 00:45.49 | Nugget | ch1 could be a vendor notation as to whether or not it comes with the license you need to get a support contract on it, I suppose. |
| 00:45.54 | Nugget | (or the lack thereof) |
| 00:46.58 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 00:47.33 | nestAr | bitchx on aol? |
| 00:47.34 | nestAr | lol |
| 00:47.39 | nestAr | windows version, i guess |
| 00:49.13 | *** join/#asterisk mindamp (~mindamp@AC8CF84A.ipt.aol.com) |
| 00:49.20 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 00:49.45 | Veto | here is a search that shows both, maybe ch1 seems to mean manager set, no license...i think. |
| 00:49.52 | Veto | http://www.pricegrabber.com/search.php?topcat_search=1&form_keyword=cisco+7960&topcat_id=1&Search=Search |
| 00:54.49 | drumkilla | why has irc been so quiet lately? |
| 00:54.55 | file[laptop] | bah |
| 00:54.55 | drumkilla | almost ... too quiet. |
| 00:55.03 | drumkilla | creeps me out |
| 00:55.15 | *** mode/#asterisk [+o file[laptop]] by drumkilla |
| 00:55.20 | *** mode/#asterisk [-o file[laptop]] by drumkilla |
| 00:55.29 | drumkilla | just kidding, muahaha |
| 00:55.35 | tainted- | lol |
| 00:55.40 | file[laptop] | drumkilla: awww how sad |
| 00:55.41 | tainted- | did i join efnet/ |
| 00:55.47 | drumkilla | :) |
| 00:56.07 | tainted- | drumkilla why are there so many idlers on #asterisk |
| 00:56.15 | tainted- | i've always wondered |
| 00:56.16 | file[laptop] | Mental note: Bring a hat saying "+o" when I next see drumkilla |
| 00:56.23 | file[laptop] | :p |
| 00:56.30 | tainted- | is there a secret file offerbot somewhere i don't know about? |
| 00:56.34 | mindamp | well im idling to gain useful information |
| 00:56.45 | drumkilla | that's why most people do |
| 00:56.47 | mindamp | unfortunately.. |
| 00:56.48 | file[laptop] | they watch us... observe... blackmail... |
| 00:56.52 | drumkilla | they watch while theyarelearning |
| 00:56.55 | mindamp | it seem slike everyone else is idling |
| 00:56.55 | CoaxD | alright. I am the official weed dealer of tonight's online weed smoking activities. Stand in line please. |
| 00:56.57 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 00:56.58 | mindamp | ;-/ |
| 00:56.58 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:56.59 | drumkilla | my spacebaris hating life right now |
| 00:57.08 | tainted- | mmkay |
| 00:57.13 | tainted- | but wouldn't u learn more by talking? |
| 00:57.17 | file[laptop] | CoaxD: there is no line I'm afraid, only in your mind d00d |
| 00:57.23 | CoaxD | file: :/ |
| 00:57.23 | mindamp | send all the weed u have to 3291 springer ridge rd. |
| 00:57.27 | mindamp | carbondale, il |
| 00:57.29 | mindamp | 62901 |
| 00:57.36 | drumkilla | I deserve it more! |
| 00:57.38 | mindamp | ill send u a check once i get the goods |
| 00:57.41 | CoaxD | mindamp: alright dude |
| 00:57.53 | drumkilla | ~drumkilla |
| 00:57.54 | jbot | rumour has it, drumkilla is the Asterisk v1.0-stable maintainer. ph33r him. |
| 00:57.55 | CoaxD | mindamp: I'll include a return address in case it doesn't go through mmkhx? |
| 00:58.12 | file[laptop] | drumkilla is a *great* guy |
| 00:58.19 | mindamp | hah |
| 00:58.22 | file[laptop] | and hilarious when he freaks out |
| 00:58.23 | *** join/#asterisk itnomad (~jackal@rrcs-24-153-206-123.sw.biz.rr.com) |
| 00:58.26 | mindamp | my nuts are blue |
| 00:58.36 | tengulre | Hi,all! |
| 00:58.42 | drumkilla | hey tengulre ! |
| 00:58.52 | CoaxD | mindamp: I *HATE* it when that happens |
| 00:58.57 | file[laptop] | drumkilla: I need more suggestions for this quasi-app turned game |
| 00:58.58 | tainted- | tengulre we've missed u! |
| 00:59.09 | tengulre | ? |
| 00:59.21 | firestrm | mindamp, quit bathing in blue coolaid :) |
| 00:59.26 | CoaxD | <PROTECTED> |
| 00:59.28 | tengulre | Is the asterisk support IVR? |
| 00:59.50 | tainted- | yes the asterisk support the IVR |
| 00:59.58 | file[laptop] | tengulre: yes, it's on the long list at http://www.asterisk.org/ you should go read it |
| 01:00.01 | *** part/#asterisk itnomad (~jackal@rrcs-24-153-206-123.sw.biz.rr.com) |
| 01:00.11 | tengulre | tks |
| 01:00.11 | file[laptop] | under "Features" |
| 01:00.15 | tengulre | i see |
| 01:00.30 | Sedorox | does asterisk support VoIP? |
| 01:00.31 | Sedorox | O:-) |
| 01:00.35 | CoaxD | <PROTECTED> |
| 01:00.37 | tainted- | no |
| 01:00.44 | tainted- | the asterisk does not support the voip |
| 01:00.44 | file[laptop] | Sedorox: no, it does not - only VoP, Voice over Pidgeons |
| 01:00.46 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 01:00.47 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 01:00.50 | Sedorox | ahaha |
| 01:00.52 | Sedorox | ncie one file[laptop] |
| 01:00.59 | tzanger | Sedorox: heh |
| 01:01.01 | tengulre | I want use C/S struct? Server= Linux, Client=Win32 |
| 01:01.04 | firestrm | Sedorox, what is voip? very old important person? |
| 01:01.06 | drumkilla | actually,voiceUnderIP |
| 01:01.08 | firestrm | :P |
| 01:01.19 | tainted- | lol |
| 01:01.23 | tainted- | VuIP |
| 01:01.29 | Sedorox | lol |
| 01:01.32 | mindamp|away | i need some voice over pidgeons |
| 01:01.37 | mindamp|away | how exactly does that work? |
| 01:01.45 | mindamp|away | is my cell phone supported? |
| 01:01.46 | Sedorox | there is a RFC on it.. |
| 01:01.47 | Sedorox | .well |
| 01:01.48 | drumkilla | WeScreamThePackets |
| 01:01.51 | Sedorox | PoP |
| 01:01.55 | tengulre | I want use C/S struct? Server= Linux, Client=Win32 |
| 01:01.57 | Sedorox | packets over pigdeons |
| 01:01.58 | drumkilla | IHaveTotallyBrokenMySpaceBar,Folks |
| 01:02.04 | CoaxD | DUDE! IPoCP is a GREAT protocol! |
| 01:02.11 | file[laptop] | tengulre: yes. |
| 01:02.14 | Veto | Here is a better example of the -ch1 and plain 7960g, one is $270 and the -ch1 is $315: http://www.voltdepot.com/advanced_search_result.php?keywords=7960 |
| 01:02.22 | firestrm | army ant ip protocol.. white painted ant is a 1 bit a black painted ant is a 0 bit |
| 01:02.35 | tainted- | does voice over pidgeons replace voice under pidgeons? VuP was a constant mess... |
| 01:02.43 | mindamp|away | but what about the smaller ants |
| 01:02.46 | mindamp|away | and the queen ant |
| 01:02.54 | Sedorox | firestrm: does it include error correction? |
| 01:03.02 | file[laptop] | or the dreaded... voice over satellite over modem over walkie talkies |
| 01:03.04 | firestrm | queen ant is used as the header |
| 01:03.14 | tainted- | lol |
| 01:03.21 | mindamp|away | ah |
| 01:03.27 | mindamp|away | and the little ants are the libs? |
| 01:03.28 | firestrm | drones = bit stuffing |
| 01:03.32 | Sedorox | where { modem=28.8; }; |
| 01:03.36 | mindamp|away | ah |
| 01:03.44 | drumkilla | firestrm: how's the latency? |
| 01:04.03 | firestrm | bout the same as satelite |
| 01:04.04 | mindamp|away | a hair past no connectivity |
| 01:04.18 | mindamp|away | can Vup use satellite? |
| 01:04.25 | mindamp|away | or does the atmosphere get in the way.. |
| 01:04.30 | drumkilla | this is much more entertaining than studying. |
| 01:04.33 | mindamp|away | and how do i configure my pidgeon for asterisk? |
| 01:04.37 | Sedorox | lol Druken |
| 01:04.39 | Sedorox | drumkilla: |
| 01:04.46 | file[laptop] | mindamp|away: you need to reflash it's firmware |
| 01:04.46 | drumkilla | Sedorox: |
| 01:04.50 | mindamp|away | do i have to terminal in? |
| 01:04.54 | Sedorox | it was a correction |
| 01:04.55 | Sedorox | sorry |
| 01:05.02 | Sedorox | tab completetion... |
| 01:05.03 | Sedorox | :/ |
| 01:05.04 | mindamp | stick a serial cable up its ass? |
| 01:05.07 | mindamp | whats the deal here |
| 01:05.09 | drumkilla | :) |
| 01:05.12 | mindamp | theres no reset button either |
| 01:05.21 | mindamp | oh wait |
| 01:05.23 | mindamp | there it is |
| 01:05.23 | firestrm | i wonder what would be faster.. pigion or army ant protocol.. pigion, faster than ant but packets come in bursts, ant, slower but constant data stream' |
| 01:05.28 | mindamp | it just shit out a bunch of logs |
| 01:05.34 | drumkilla | firestrm: yeah, bad jitter |
| 01:05.40 | firestrm | lol |
| 01:05.42 | Sedorox | ant would be better for voice |
| 01:05.54 | drumkilla | could we implement a jitter buffer? |
| 01:05.55 | ariel_ | ~seen Mauricio |
| 01:05.57 | jbot | mauricio <thrill@adsl-69-227-38-177.dsl.pltn13.pacbell.net> was last seen on IRC in channel #asterisk, 2h 22m 2s ago, saying: 'anyone have AMP running on slackware?'. |
| 01:05.58 | tainted- | lol |
| 01:06.05 | Sedorox | drumkilla: an ant-eater? |
| 01:06.16 | firestrm | rotfl |
| 01:06.19 | tainted- | this is on the brink of lame |
| 01:06.23 | Sedorox | ahaha |
| 01:06.24 | tainted- | but not there yet |
| 01:06.24 | drumkilla | maybe set some food out |
| 01:06.33 | drumkilla | it was better than when it was totally quiet earlier |
| 01:06.34 | ariel_ | Hello everyone |
| 01:06.37 | firestrm | food = routing protocol |
| 01:06.45 | drumkilla | firestrm: brilliant |
| 01:06.49 | drumkilla | is someone writing this down? |
| 01:06.52 | Sedorox | oh.. so the ant-eater just gets the ants that are over their TTT, Time To Travel? |
| 01:06.53 | drumkilla | we may forget. |
| 01:07.06 | firestrm | rotfllmao |
| 01:07.14 | tainted- | TTL gets a whole new meaning as well |
| 01:07.19 | *** join/#asterisk ALapo (~hal9000@216-164-58-166.c3-0.129-ubr2.lnh-129.md.cable.rcn.com) |
| 01:07.22 | ALapo | howdy |
| 01:07.27 | tainted- | ALapo! |
| 01:07.30 | Sedorox | ahah |
| 01:07.32 | drumkilla | what would clasify as a hop? |
| 01:07.36 | ALapo | hi tainted- |
| 01:07.47 | drumkilla | not that, file[laptop] !!!!! |
| 01:07.50 | firestrm | hmm perhaps stream crossing? |
| 01:07.52 | ALapo | anyone have any tips for me and getting free world dialup working |
| 01:08.10 | ALapo | i set it up the wiki way, and i get "all circuits busy" i set it up the FWD way and it doesnt work |
| 01:08.12 | file[laptop] | yay |
| 01:08.29 | ALapo | im not experienced with asterisk yet, only on askterisk at home getting the groundwork in learning before i move to compile it myself |
| 01:08.51 | ALapo | so i was just wondering if anyone had tips for me |
| 01:09.17 | tainted- | it is hard to provide support for asterisk newbs |
| 01:09.23 | tainted- | even if u have the desire to |
| 01:09.25 | ALapo | yeah im sure |
| 01:09.26 | drumkilla | ALapo: dive in head first! |
| 01:09.28 | ALapo | it's complex as heck |
| 01:09.34 | tainted- | there are so many complicated concepts to grasp |
| 01:09.37 | drumkilla | but it's so much fun! |
| 01:09.39 | ALapo | yeah |
| 01:09.41 | ALapo | im enjoying it |
| 01:09.44 | drumkilla | :D |
| 01:09.46 | drumkilla | I'm addicted. |
| 01:09.47 | ALapo | any place to get those knock off cards cheap? |
| 01:09.51 | drumkilla | AH! |
| 01:09.53 | file[laptop] | you're sooooo nice! |
| 01:09.53 | ALapo | on ebay i found em for $8... but $8 shipping |
| 01:09.57 | file[laptop] | like a big teddy bear |
| 01:10.03 | mindamp | any hardware set modem |
| 01:10.04 | file[laptop] | except not. |
| 01:10.05 | ALapo | not that i wont spend $16 |
| 01:10.06 | tainted- | and it is becoming more and more polarize between asterisk vets and asterisk@home types |
| 01:10.13 | mindamp | with voice |
| 01:10.21 | ALapo | i own no voicemodems at the moment afaik |
| 01:10.22 | ALapo | :( |
| 01:10.28 | ALapo | i just want something that 100% will work |
| 01:10.32 | file[laptop] | you can't just use any voice modem with asterisk pfft |
| 01:10.34 | drumkilla | tainted-: it is good to have the asterisk@home stuff out there, though |
| 01:10.35 | mindamp | i had to dig through like 60 comps to fine 2 |
| 01:10.35 | ariel_ | ALapo, there is a site here call #amportal that might be able to help more with the FWD setup. |
| 01:10.35 | ALapo | if i got to spend $16 to avoid headache so be it |
| 01:10.37 | drumkilla | it helps gain adoption |
| 01:10.50 | ALapo | trust me, i will be skilled at asterick one day |
| 01:10.55 | ALapo | it's too valueable a skill to not have |
| 01:11.07 | MikeJ[Laptop] | ~kpflemming |
| 01:11.08 | file[laptop] | how cute! |
| 01:11.15 | tainted- | drumkilla but that is also like saying AOL increasing internet adoption |
| 01:11.17 | ALapo | like shit you can setup an almost free voip network... people pay thousands for those things a month to be maintained |
| 01:11.18 | ariel_ | That is knew skilled at asterisk.... |
| 01:11.19 | MikeJ[Laptop] | ~seen kpflemming |
| 01:11.20 | jbot | MikeJ[Laptop]: i haven't seen 'kpflemming' |
| 01:11.24 | tainted- | drumkilla it comes with big time drawbacks |
| 01:11.25 | drumkilla | it's quite possibly the most commonly mispronounced app name, ever :) |
| 01:11.26 | ALapo | thanks ariel_ i'll check it out |
| 01:11.26 | MikeJ[Laptop] | ~seen kpfleming |
| 01:11.27 | jbot | kpfleming <~kpfleming@207.111.174.1> was last seen on IRC in channel #asterisk, 19d 19h 15m 14s ago, saying: 'no, there is no specific plan at this time'. |
| 01:11.38 | file[laptop] | Aster-isk! |
| 01:11.43 | file[laptop] | IAX! |
| 01:11.47 | tainted- | asstricks |
| 01:11.48 | file[laptop] | not, Eye Ax! |
| 01:11.57 | ALapo | i h ate people who pronounce acronyms |
| 01:11.58 | ALapo | like GUI |
| 01:12.01 | ALapo | it is G U I |
| 01:12.04 | ALapo | not gooey |
| 01:12.07 | ALapo | lol |
| 01:12.09 | drumkilla | it's gooey! |
| 01:12.12 | ALapo | never!! |
| 01:12.13 | ALapo | :P |
| 01:12.16 | file[laptop] | then you don't wanna say IAX :p |
| 01:12.16 | mindamp | ill gooey u |
| 01:12.18 | tainted- | i cannot stand hearing grown men say "voeeeep" |
| 01:12.21 | ALapo | yeah |
| 01:12.23 | mindamp | er ill GUI u |
| 01:12.25 | mindamp | ;-) |
| 01:12.31 | tainted- | voeeeep |
| 01:12.34 | drumkilla | IAX is eeks! |
| 01:12.36 | ALapo | or like MySQL |
| 01:12.39 | ALapo | it isnt my seequll |
| 01:12.43 | ALapo | it's my S Q L |
| 01:12.43 | ALapo | lol |
| 01:12.44 | file[laptop] | drumkilla: right you are! |
| 01:12.44 | tainted- | drumkilla is it? |
| 01:12.49 | drumkilla | tainted-: yep |
| 01:12.55 | ALapo | anyways |
| 01:12.56 | tainted- | eeks!? |
| 01:13.00 | file[laptop] | yes, eeks |
| 01:13.01 | ALapo | any guides for me diving into asterisk |
| 01:13.02 | drumkilla | tainted-: yeah :) |
| 01:13.06 | ariel_ | it's even better with SCSI or Suzzzy |
| 01:13.08 | ALapo | im fairly good at linux |
| 01:13.12 | tainted- | i'll go with I-A-X |
| 01:13.16 | drumkilla | tainted-: comes from mark |
| 01:13.23 | ALapo | still learning some, but i have a lot of knowledge |
| 01:13.32 | drumkilla | ALapo: if you're good with linux, asterisk config shouldn't be too hard for you |
| 01:13.37 | ariel_ | ALapo, on the asterisk@home site there is a document that can help you get started. |
| 01:13.38 | ALapo | they dont sem hard |
| 01:13.40 | tainted- | drumkilla not true |
| 01:13.40 | drumkilla | ALapo: it's more the concepts to learn |
| 01:13.41 | ALapo | other than weird syntax |
| 01:13.41 | ariel_ | ~docs |
| 01:13.42 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:13.44 | file[laptop] | gah dang it I keep forgetting to e-mail Angela |
| 01:13.48 | ariel_ | also the above will help. |
| 01:13.58 | ALapo | slackware is my distro of choice |
| 01:13.59 | drumkilla | tainted-: you're killing me tonight :) |
| 01:14.02 | ALapo | but im moving to centos |
| 01:14.07 | ALapo | lack of PAM packages |
| 01:14.14 | tainted- | haha sorry drumkilla |
| 01:14.20 | ALapo | dont want to compile and integrate pam all by myself... not that advanced at linux yet |
| 01:14.21 | ALapo | : |
| 01:14.22 | ALapo | :( |
| 01:14.24 | Nugget | Linux is poo. |
| 01:14.31 | ALapo | PAM = pain in the ass |
| 01:14.32 | Sedorox | pam sucks... |
| 01:14.35 | drumkilla | tainted-: no problem ... I really can't remember what it was like to learn Asterisk ... I have been doing it every day for a while |
| 01:14.36 | Sedorox | but thats my opinion |
| 01:14.41 | ALapo | i need it for samba |
| 01:14.49 | ALapo | to pass through auth requests to a domain controller |
| 01:14.53 | Sedorox | you don't _need_ pam for samba |
| 01:14.53 | ALapo | err |
| 01:14.57 | ALapo | create the local accounts |
| 01:14.59 | Sedorox | lol |
| 01:15.02 | ariel_ | ALapo, just yum install samba |
| 01:15.06 | Sedorox | *shrugs* |
| 01:15.11 | tainted- | i can't think of another popular linux project that is harder to grasp |
| 01:15.12 | ALapo | ive researched this |
| 01:15.29 | ALapo | to pass through to domain controller, unless you want to manually create accounts or write your own script, requires PAM |
| 01:15.47 | drumkilla | tainted-: the technology, you mean? |
| 01:16.01 | ALapo | if you know otherwise let me know :) |
| 01:16.05 | tainted- | mmm.. no |
| 01:16.12 | drumkilla | tainted-: how to configure it? |
| 01:16.16 | tainted- | yea |
| 01:16.24 | tainted- | if u don't have a background in telephony |
| 01:16.26 | drumkilla | tainted-: well ... comes with the power of the app, I suppose |
| 01:16.35 | Sedorox | but then again. I didn't add users/machines through it.. so... |
| 01:16.40 | ALapo | i understand the concepts in asterisk |
| 01:16.44 | ALapo | just not the syntax |
| 01:16.46 | tainted- | dialplans seemed so archaic at first |
| 01:16.48 | *** join/#asterisk sniffe (~sniffe@ti211110a080-5150.bb.online.no) |
| 01:16.49 | drumkilla | tainted-: mark's background was not telephony ... nor is mine :) |
| 01:17.11 | ALapo | having a skill like this is too valuable |
| 01:17.13 | tainted- | actually mythtv is another one with steep learning curve |
| 01:17.14 | file[laptop] | I love asterisk! |
| 01:17.16 | ALapo | once im proficient in it |
| 01:17.19 | tainted- | dunno about now |
| 01:17.21 | sniffe | Anyone tried Zaptel with kernel 2.6.11.7 ? |
| 01:17.25 | ALapo | i could go around selling consulting services |
| 01:17.31 | drumkilla | tainted-: mythtv kills me ... I just want to watch tv!!! :) |
| 01:17.31 | sniffe | I'm havind major problems.. |
| 01:17.32 | ALapo | installing asterisk PBX |
| 01:17.33 | ALapo | >:) |
| 01:17.41 | vaxen | ALapo: alot of ppl are doing it now |
| 01:17.45 | ALapo | yeah im sure |
| 01:17.46 | tainted- | drumkilla yea talking about getting your ass handed to you |
| 01:17.48 | Veto | ~7960 |
| 01:17.50 | vaxen | there is a good market for it |
| 01:17.52 | ALapo | thats why i want to get on the wagon too :P |
| 01:17.53 | mindamp | mythtv is harder to get working that asterisk |
| 01:17.59 | ALapo | im sure |
| 01:18.00 | tainted- | mindamp no way |
| 01:18.01 | mindamp | i speak from experience |
| 01:18.02 | ALapo | ive tried mythtv |
| 01:18.03 | mindamp | :-) |
| 01:18.03 | drumkilla | tainted-: I move every 4 months, and reconfiguring my location is a pain ... |
| 01:18.05 | ALapo | it seems to be a pain |
| 01:18.15 | tainted- | asterisk is tougher |
| 01:18.18 | mindamp | see point and fact |
| 01:18.20 | ALapo | i just want to get FWD working and i'll be content |
| 01:18.25 | mindamp | alapo couldn't install myth |
| 01:18.31 | mindamp | yet he installed asterisk |
| 01:18.34 | mindamp | go figure |
| 01:18.37 | tainted- | umm |
| 01:18.37 | ALapo | i installed asterisk at home |
| 01:18.40 | tainted- | he can't get fwd working |
| 01:18.42 | ALapo | so i did nothing really |
| 01:18.46 | tainted- | and he is using *@home |
| 01:18.51 | drumkilla | FWD has asterisk config on their web page ... |
| 01:18.52 | vaxen | you guys manually configure or use AMP? |
| 01:18.57 | tainted- | that is like knoppmyth |
| 01:19.02 | ALapo | drumkilla: i dont understand the syntax tho |
| 01:19.02 | ariel_ | ALapo, there is a setup for FWD on AMP site I think. |
| 01:19.03 | drumkilla | manual all the way!!! |
| 01:19.07 | file[laptop] | I _need_ fooooooood |
| 01:19.08 | ALapo | i made the changes |
| 01:19.11 | ALapo | but they didnt work |
| 01:19.16 | drumkilla | ah. |
| 01:19.28 | file[laptop] | drumkilla: fooooooooooood |
| 01:19.29 | ariel_ | Food, did I hear food? |
| 01:19.33 | drumkilla | FOOD. |
| 01:19.33 | ALapo | i dial 393 and it wont connect |
| 01:19.40 | mindamp | food! |
| 01:19.43 | ALapo | with the wiki setup i would get "all circuits are busy" |
| 01:19.46 | tainted- | lol |
| 01:19.52 | ALapo | is that a FWD message or asterisk? |
| 01:20.00 | ariel_ | ALapo, you need to set it up the AMP way not off the wiki. |
| 01:20.06 | sniffe | Anyone had problems with "Badness in local_bh_enable at kernel/softirq.c:140" and ztdynamic (TDMoE) and Asterisk ? |
| 01:20.09 | ALapo | i did the AMP way |
| 01:20.17 | ALapo | through the wiki guide |
| 01:20.27 | ALapo | and i would get "all circuits are busy" |
| 01:21.08 | tainted- | i'm stalling to avoid rewriting some code |
| 01:21.19 | drumkilla | tainted-: what code? |
| 01:21.27 | Sedorox | i'm stalling doing final projects |
| 01:21.28 | Sedorox | :/ |
| 01:21.33 | ariel_ | ALapo, did you set the trunk |
| 01:21.42 | drumkilla | I'm stalling learning new material for a test ... that is tomorrow |
| 01:21.49 | ALapo | yep |
| 01:21.53 | tainted- | a billing statement dll |
| 01:21.55 | ALapo | im behind NAT btw |
| 01:22.05 | ALapo | but i opened 5060 and 10000-20000 TCP and UDP to the box |
| 01:22.15 | ariel_ | ALapo, then did you set up the DID route? |
| 01:22.21 | drumkilla | ALapo: you should use iax with fwd |
| 01:22.22 | ALapo | i think i did everything |
| 01:22.23 | *** join/#asterisk da-manFL (~claude_cu@adsl-065-006-172-248.sip.mia.bellsouth.net) |
| 01:22.26 | drumkilla | would make your life easier :) |
| 01:22.29 | ALapo | drumkilla: the FWD setup? |
| 01:22.33 | ALapo | i tried that too |
| 01:22.35 | ariel_ | Also have you added externip= and localnet= on your sip.conf file? |
| 01:22.39 | ALapo | i think i did it right, but i dont understand |
| 01:22.41 | ALapo | yep ariel_ |
| 01:22.56 | ALapo | dont understand all the syntax, so i dont know exactly what i was doing |
| 01:23.04 | tainted- | lol |
| 01:23.05 | tainted- | classic |
| 01:23.07 | ALapo | but i did the AMP config correct afaik |
| 01:23.18 | ALapo | tainted-: it's called playing around... good way of learning |
| 01:23.20 | ALapo | :) |
| 01:23.21 | PTG1234 | hey whats the stuff called, they are replacing price tags with that can sense the device wireless in stores |
| 01:23.26 | darwin35 | ~amp |
| 01:23.27 | jbot | rumour has it, amp is an Audio MPEG Player. [non-free], or http://amp.coalescentsystems.ca/ |
| 01:23.27 | elric | i want to punch Harry Potter :/ |
| 01:23.27 | ALapo | RFID |
| 01:23.28 | ariel_ | ALapo, ok now what does the cli tell you when you dial to fwd? |
| 01:23.39 | tainted- | PTG1234 RFID |
| 01:23.40 | ALapo | it dials through, but i get all circuits busy |
| 01:23.43 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
| 01:23.45 | drumkilla | tainted-: you beat me |
| 01:23.47 | TUplink | <PROTECTED> |
| 01:23.48 | TUplink | Apr 19 21:18:06 WARNING[480]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 31A5806A-4C77-43C1-B4DB-0FAFBF5EBA46@172.138.22.186 for seqno 12030 (Non-critical Response) |
| 01:23.50 | PTG1234 | thanks |
| 01:23.51 | TUplink | whats that mean |
| 01:23.52 | tainted- | ALapo beat us both |
| 01:23.54 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 01:23.58 | ALapo | well it did, i undid the changes temporarialy |
| 01:23.58 | drumkilla | dang! |
| 01:24.01 | ALapo | tainted-? |
| 01:24.11 | drumkilla | ALapo: to saying RFID |
| 01:24.17 | tainted- | ALapo u beat us to the RFID trivia |
| 01:24.22 | ariel_ | TUplink, you have canreinvite=yes ? |
| 01:24.29 | ALapo | huh |
| 01:24.31 | tainted- | what is the capital of washington? |
| 01:24.34 | tainted- | olympia |
| 01:24.35 | TUplink | wheres that go sip.conf |
| 01:24.37 | ALapo | i live in DC |
| 01:24.39 | drumkilla | tainted-: cheater! |
| 01:24.40 | tainted- | hmm.. looks like i got that one |
| 01:24.56 | tainted- | ok here's one for u drum |
| 01:24.58 | drumkilla | what is the capital of South Carolina? |
| 01:25.01 | tainted- | what rhymes with silver? |
| 01:25.01 | drumkilla | Columbia! |
| 01:25.02 | ALapo | 3.3.1 Free World Dialup (FWD) |
| 01:25.06 | ALapo | i did that |
| 01:25.11 | tainted- | omg i totally had that one |
| 01:25.15 | TUplink | should canreinvite be yes or no? |
| 01:25.22 | Qwell | Whats the capital of Chicago? |
| 01:25.29 | darwin35 | ~wiki |
| 01:25.30 | file[laptop] | Toast! |
| 01:25.33 | ariel_ | TUplink, in most cases no |
| 01:25.37 | Qwell | see, nobody knows... |
| 01:25.39 | tainted- | Qwell trick question! |
| 01:25.40 | ALapo | ariel_: http://asteriskathome.sourceforge.net/handbook/#Section_3.3.1 |
| 01:25.43 | ALapo | thats what i did |
| 01:25.47 | ALapo | and i get "all circuits busy" |
| 01:25.48 | *** join/#asterisk Smilk (~Ling@c-069-063-192-006.sd2.redwire.net) |
| 01:25.53 | ALapo | enabled IAX at FWD |
| 01:25.54 | tainted- | Smilk! |
| 01:25.56 | newl | chicago hahahahah |
| 01:25.57 | drumkilla | tainted-: Asterilver |
| 01:26.05 | Qwell | drumkilla: That a new convention? |
| 01:26.06 | tainted- | that's right ! |
| 01:26.09 | ALapo | is all circuits busy a FWD message or an asterisk, it could work but they could be having issues, im not sure |
| 01:26.15 | Smilk | anyone here in San Diego? hey there tainted |
| 01:26.28 | tainted- | Smilk any luck with your DID? |
| 01:26.29 | tainted- | lol |
| 01:26.33 | drumkilla | Qwell: yes. it's my convention ... held in my apartment. |
| 01:26.38 | Smilk | not yet.. trying still |
| 01:26.44 | ALapo | so 16 bux is the cheapest i can get that knock off card for? |
| 01:26.44 | TUplink | canreinvite yes or no? |
| 01:26.44 | Smilk | there must be a cheaper way initially |
| 01:26.49 | Smilk | that is still able to scale |
| 01:26.50 | tainted- | tell everyone your DID.. it's really amusing |
| 01:26.57 | Smilk | tainted-, lol |
| 01:26.59 | drumkilla | 2569223052 |
| 01:27.02 | ALapo | whats DID? |
| 01:27.12 | tainted- | what is that drum |
| 01:27.13 | Smilk | ok, ok, I DID my uncle... |
| 01:27.13 | file[laptop] | 800-555-1212 |
| 01:27.21 | drumkilla | tainted-: one of my dids! |
| 01:27.22 | ALapo | direct dial interface? |
| 01:27.23 | Sedorox | ~DID |
| 01:27.24 | jbot | from memory, did is Direct Inward Dialing |
| 01:27.29 | ALapo | ah |
| 01:27.30 | PTG1234 | hey anyone got any cool ideas for what you would want to see for the home of tommorow :) in your house? |
| 01:27.35 | tainted- | lol |
| 01:27.39 | tainted- | PTG1234 google for dilbert home |
| 01:27.40 | ALapo | ariel_: any tips? |
| 01:27.41 | Qwell | PTG1234: a microwave in every bedroom |
| 01:27.50 | Sedorox | sorry PTG1234 they are copyrighted... |
| 01:27.51 | PTG1234 | hah |
| 01:28.08 | file[laptop] | PTG1234: bah |
| 01:28.33 | ariel_ | ALapo, just a sec brb |
| 01:28.35 | drumkilla | you should be able to control everything with Asterisk |
| 01:28.38 | drumkilla | including make breakfast. |
| 01:28.45 | Qwell | drumkilla: and the restroom? |
| 01:28.50 | ALapo | thanks ariel_ |
| 01:28.53 | Qwell | "sink - activate" |
| 01:28.54 | file[laptop] | make muffins |
| 01:28.57 | file[laptop] | yay |
| 01:29.01 | PTG1234 | heh |
| 01:29.07 | ALapo | im re-applying the settings now |
| 01:29.08 | file[laptop] | make muffins TYPE=blueberry |
| 01:29.12 | drumkilla | Qwell: yes, dial 7 to flush |
| 01:29.14 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 01:29.19 | Qwell | drumkilla: batphones? |
| 01:29.21 | tainted- | drumkilla what's up with blueberry muffins anyway |
| 01:29.28 | drumkilla | tainted-: ask file[laptop] |
| 01:29.31 | tainted- | was reading about that someplace |
| 01:29.34 | drumkilla | that's his sillyness |
| 01:29.39 | tainted- | oh |
| 01:29.46 | file[laptop] | yes yes, it's all me |
| 01:29.50 | tainted- | i think on sineapps or something |
| 01:30.07 | tainted- | file[laptop] what's up with blueberry muffins |
| 01:30.14 | file[laptop] | what about them? |
| 01:30.17 | file[laptop] | they're goooooood |
| 01:30.28 | tainted- | peach is good too |
| 01:30.34 | tainted- | if they are fresh |
| 01:30.35 | Qwell | in a muffin? please |
| 01:30.50 | tainted- | Qwell u just don't know a real muffin |
| 01:30.52 | drumkilla | file[laptop]'s muffin gets around |
| 01:31.00 | tainted- | not your chain store variety |
| 01:31.36 | tainted- | ok that's a different muffin as well |
| 01:31.46 | file[laptop] | Mark is my muffin man |
| 01:32.01 | *** join/#asterisk dwmw2_gone (dwmw2@baythorne.infradead.org) |
| 01:32.13 | *** join/#asterisk PyroSteve (~steve@wsip-70-183-114-254.no.no.cox.net) |
| 01:32.21 | PyroSteve | hey guys |
| 01:32.26 | tainted- | PyroSteve! |
| 01:32.30 | drumkilla | PyroSteve: !!!!!!! |
| 01:32.35 | PyroSteve | A long time ago I got Voicepulse to work |
| 01:32.37 | ALapo | blah |
| 01:32.39 | ALapo | still doesnt work |
| 01:32.39 | ALapo | :( |
| 01:32.42 | PyroSteve | and has been working ever since |
| 01:33.20 | tainted- | i wish we had a forum someplace |
| 01:33.29 | PyroSteve | but I also remember being able to use the incoming caller id in place of the extension |
| 01:33.30 | tainted- | i would love to take a poll: why are u in #asterisk |
| 01:33.31 | file[laptop] | we do ... |
| 01:33.43 | PyroSteve | but that doesn't work with voicepulse |
| 01:33.48 | ALapo | i get 404 not found if i dial 393612 |
| 01:33.49 | file[laptop] | I SEE GAY PEOPLE! |
| 01:33.55 | ALapo | and if i just dial 393 i get all circuits busy |
| 01:34.09 | PyroSteve | Voicepulse seems to want the extension number on my own voicepilse DID ? |
| 01:34.11 | *** join/#asterisk ToyMan (~konversat@user-12lcqur.cable.mindspring.com) |
| 01:34.11 | pointer | anyone around that could look at my zaptel.conf and tell me what I'm typoing that's causing our second PRI not to work? (cfg is 1 port T1; single FXO; quad t1) |
| 01:34.21 | drumkilla | ALapo: is 612 a valid fwd number? |
| 01:34.31 | ALapo | yeah |
| 01:34.33 | ALapo | it should read me the time |
| 01:34.52 | ALapo | i only get all circuits busy if i add outbound routing rules for 393 |
| 01:34.53 | pointer | I'm getting hte ever descriptive "Invalid argument (22)" error |
| 01:35.06 | drumkilla | ALapo: does it strip the 393 from the number correctly? |
| 01:35.09 | pointer | s/hte/the/ |
| 01:35.15 | ALapo | my phone client?> |
| 01:35.21 | drumkilla | ALapo: asterisk |
| 01:35.28 | ALapo | i just get 404 not found |
| 01:35.28 | tainted- | drumkilla what are u studying |
| 01:35.31 | ALapo | w/o an outbound rule |
| 01:35.39 | drumkilla | tainted-: computer engineering |
| 01:35.45 | drumkilla | ALapo: want me to log in to your box? |
| 01:35.47 | *** join/#asterisk JMcA (~jmcadams@67.141.1.51) |
| 01:35.48 | ALapo | with an inbound rule, 393612 will get 404 not found and 393 will get all circuits busy |
| 01:35.54 | file[laptop] | drumkilla is all knowing in the computer engineering field |
| 01:36.06 | *** join/#asterisk |Vulture| (~V@64.234.204.68.cfl.res.rr.com) |
| 01:36.07 | drumkilla | yeah, right |
| 01:36.08 | ALapo | sure, but port 80 is blocked so it would have to be via shell if ya got the time |
| 01:36.17 | drumkilla | ALapo: shell is all I wanted |
| 01:36.23 | drumkilla | I just want to see the * CLI output |
| 01:36.24 | ALapo | does AMP get fubared if you modify by hand? |
| 01:36.32 | drumkilla | ALapo: probably :) |
| 01:36.36 | ALapo | ;/ |
| 01:36.36 | |Vulture| | oh got AMP... |
| 01:36.46 | |Vulture| | *god |
| 01:36.54 | ALapo | sorry i just started using asterisk today |
| 01:36.57 | tainted- | ALapo why don't u try to learn dialplan |
| 01:36.59 | ALapo | would rather use it and get a feel for it before compiling :) |
| 01:37.03 | tainted- | it's quite sexy |
| 01:37.04 | ALapo | dialplan? |
| 01:37.16 | drumkilla | yeah, the "manual" way |
| 01:37.25 | tainted- | the "man" way |
| 01:37.31 | |Vulture| | I think #asterisk should be reserved for people who actually use * and not AMP :P |
| 01:37.32 | tainted- | puts hair on your chest |
| 01:37.37 | r0d3nt | heh |
| 01:37.37 | ALapo | tainted-: i do plan to venture to it |
| 01:37.40 | tainted- | the hair that u pull out of your head |
| 01:37.48 | ALapo | just not yet |
| 01:37.55 | ALapo | want to get my feet wet, not cannonball into the pool |
| 01:37.56 | ALapo | :P |
| 01:37.58 | tainted- | venture forth! |
| 01:38.11 | ALapo | drumkilla: what outputs from shell did you want to see? |
| 01:38.16 | ALapo | anything that would help me diagnoe? |
| 01:38.20 | file[laptop] | where's my watch... |
| 01:38.21 | |Vulture| | ALapo: you will be more confused by doing AMP first |
| 01:38.32 | ALapo | probably |
| 01:38.39 | file[laptop] | who stole my watch?!? |
| 01:38.47 | drumkilla | ALapo: the line that says something about "Dial" :) |
| 01:38.47 | |Vulture| | file[laptop]: what kinda watch? |
| 01:38.50 | Qwell | hahaha! |
| 01:38.51 | ALapo | im going to use amp for a few days and then compile from scratch this/next weekend once my voice card comes |
| 01:38.53 | tainted- | if u don't learn fundamentals, u will be a howto whore forever |
| 01:38.53 | ALapo | in what conf file |
| 01:38.59 | *** part/#asterisk pointer (pointer@aj.catt.com) |
| 01:39.02 | tainted- | and we've all met the howto whores |
| 01:39.03 | ALapo | im waiting for my card |
| 01:39.12 | drumkilla | ALapo: asterisk -r |
| 01:39.15 | ALapo | i dont want to waste my time installing it w/o the card first |
| 01:39.16 | drumkilla | ALapo: then make the call |
| 01:39.23 | drumkilla | ALapo: then copy the line that says "Dial" |
| 01:39.59 | ALapo | is it supposed to display info? or is there a command i need to run to see the debug output for when i dial |
| 01:40.08 | ALapo | i dialed on the softphone, but i see nothing on the commandline spawned |
| 01:40.26 | drumkilla | it should display stuff |
| 01:40.32 | ALapo | it's not :S |
| 01:40.35 | drumkilla | bah ... I don't know if I can handle debugging AMP :) |
| 01:40.42 | ALapo | ah |
| 01:40.46 | Qwell | is there a difference between * and *@home? I do mean the * code itself, not the crap surrounding it |
| 01:40.46 | ALapo | when i dial other #'s it scrolls info |
| 01:41.02 | ALapo | i think @ home is just asterisk pre-configged |
| 01:41.03 | drumkilla | my guess is that it isn't stripping the 393 |
| 01:41.09 | PyroSteve | Both Nufone and VP Connect requires my DID as an extension instead of the 's' extension.... I want to use the incoming called id as the extensions instead .. how do I do this ? |
| 01:41.15 | ALapo | should i add an outbound call rule and check for info? |
| 01:41.20 | drumkilla | sure |
| 01:41.21 | drumkilla | heh |
| 01:41.42 | ALapo | nothing |
| 01:41.42 | drumkilla | I vote for saying screw it, and doing it with the conf files :) |
| 01:41.44 | drumkilla | then I can help |
| 01:41.48 | ALapo | but i get all circuits are busy now |
| 01:41.53 | ALapo | vs. 404 not available |
| 01:41.59 | ALapo | yeah thanks for the help |
| 01:42.08 | tainted- | ALapo let me ssh into your box |
| 01:42.08 | drumkilla | try the amp channel |
| 01:42.14 | ALapo | you'll probably see a lot of me once im compiling/configging from scratch |
| 01:42.14 | ALapo | :P |
| 01:42.16 | tainted- | i'll fix it in a jiffy |
| 01:42.26 | ALapo | i want to learn tho |
| 01:42.31 | ALapo | where do you sugest checking?> |
| 01:42.34 | tainted- | just watch the screen |
| 01:42.40 | ALapo | i dono if screen is installed |
| 01:42.46 | tainted- | what distro |
| 01:42.50 | tainted- | oh asterisk@home right |
| 01:42.51 | ALapo | it's @home |
| 01:42.53 | ALapo | centos3 i guess |
| 01:43.03 | drumkilla | cya folks |
| 01:43.05 | ALapo | thanks anyways drumkilla |
| 01:43.06 | tainted- | bye drum |
| 01:43.08 | ALapo | i'll be back ;) |
| 01:43.13 | ALapo | but hmm |
| 01:43.22 | ALapo | it doesnt look to me like asterisk is making config file changes |
| 01:43.24 | Qwell | PyroSteve: nufone comes in on my DID exten. |
| 01:43.25 | ALapo | AMP i mean |
| 01:43.26 | drumkilla | cya tainted- ALapo file[laptop] <everyone else> |
| 01:43.34 | file[laptop] | bye drumkilla |
| 01:43.37 | *** join/#asterisk three55ml (~three55ml@cpe-66-68-98-68.austin.res.rr.com) |
| 01:43.45 | at561 | pyrosteve that's extremely introductory level stuff. if you define an extension as exten => 6/8005551212,1,Playback(welcometoyou) it will create an extension "6" which only triggers if 8005551212 is the callerid |
| 01:43.49 | Qwell | PyroSteve: exten => 8003911234,1,something, works fine for me |
| 01:44.00 | ALapo | [outrt-002-FWD] |
| 01:44.00 | ALapo | include => outrt-002-FWD-custom |
| 01:44.00 | ALapo | exten => 393,1,Macro(dialout-trunk,2,${EXTEN}) |
| 01:44.00 | ALapo | exten => 393,2,Macro(outisbusy); No available circuits |
| 01:44.02 | ALapo | hmmm |
| 01:44.06 | PyroSteve | opps |
| 01:44.07 | ALapo | i see no available circuits |
| 01:44.07 | PyroSteve | never mind |
| 01:44.08 | PyroSteve | sorry |
| 01:44.09 | ALapo | that perks my ears |
| 01:44.11 | PyroSteve | i was doing |
| 01:44.18 | three55ml | Other than VoicePulse, are there any IAX originators with instant provisioning? Or at least decent turnaround? I'm tired of providers not taking 3-4 weeks with no replies. |
| 01:44.19 | ALapo | do those lines look correct? |
| 01:44.28 | PyroSteve | it wrong |
| 01:44.33 | three55ml | Taking 3-4 weeks rather |
| 01:44.42 | ALapo | ? |
| 01:44.51 | ALapo | oh not talkin to me PyroSteve |
| 01:45.46 | at561 | my front door now says "hello?", monitors for any reply, then buzzes whoever it is in anyway with instructions to leave the package at the door |
| 01:46.03 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 01:46.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:48.07 | ALapo | hey ariel_ back? |
| 01:48.39 | *** join/#asterisk tugalone (~tugalone@pcp0010303951pcs.avenel01.nj.comcast.net) |
| 01:48.53 | Qwell | three55ml: nufone? |
| 01:49.23 | three55ml | Qwell: They only offer Michigan numbers and aren't taking new customers. |
| 01:49.28 | Qwell | tollfree |
| 01:49.41 | Qwell | and, they can if you talk to them |
| 01:49.46 | three55ml | I don't have any problem getting toll-free numbers quickly. |
| 01:49.57 | three55ml | I don't like NuFone's owner's attitude the few times I've talked to him :) |
| 01:50.10 | tainted- | three55ml join the club ;) |
| 01:50.24 | three55ml | tainted-: Hehe, that was putting it nicely. |
| 01:50.32 | at561 | since asterisk apparently supports changing codec midstream it would be cool if there were an app interface to that. so for instance callerid could trigger a higher quality call |
| 01:50.39 | sivana | three55ml: you need toll-frees? |
| 01:50.42 | ALapo | ;/ |
| 01:50.51 | three55ml | sivana: Depending on the rate, more providers never hurts. |
| 01:51.26 | ALapo | can you fake caller ID with asterisk? |
| 01:51.28 | ALapo | outbound |
| 01:51.38 | at561 | why would you want to do that |
| 01:51.47 | ALapo | trick friends for no reason |
| 01:51.52 | Qwell | at561: I have a perfectly valid reason to fake my CID |
| 01:51.59 | ALapo | is it possible? |
| 01:52.01 | Qwell | I can't call some 8xx numbers unless I do |
| 01:52.02 | tainted- | ALapo u can if ur provider allows u to |
| 01:52.03 | Qwell | ALapo: sure |
| 01:52.06 | ALapo | ah |
| 01:52.07 | ALapo | cool |
| 01:52.19 | ALapo | in any event |
| 01:52.21 | denon | ALapo: it has nothing to do with Asterisk, you can do it with any PRI gear |
| 01:52.23 | Veto | so is the 7960g > 7960 except for the icons? |
| 01:52.33 | denon | Veto: yes, but thats about the only difference |
| 01:52.33 | Veto | err, ^>^= |
| 01:52.43 | ALapo | why do i get all circuits busy :( |
| 01:52.44 | Moc[Toronto] | 7960g support normal PoE I beleive |
| 01:52.49 | denon | nod |
| 01:52.53 | denon | though some people argue it doesnt |
| 01:52.55 | *** join/#asterisk tessier (~treed@203.210.209.79) |
| 01:53.04 | PyroSteve | at561: im sorry ... i was doing it wrong .. ive did it before and it wasn't working with VP Connect and i thought it was the way that my system had to be setup for VP COnnect |
| 01:53.12 | tainted- | PoE on a sip phone? ABOUT GOD DAMN TIME |
| 01:53.18 | at561 | now it works? |
| 01:53.27 | denon | tainted-: huh? everything support PoE .. has for years |
| 01:53.29 | Qwell | They need a PoE wifi phone. |
| 01:53.29 | Veto | okay, seems that the 7960g-ch1 = 7960g |
| 01:53.32 | PyroSteve | at561: and you're right .. that is introductory level configuration |
| 01:53.40 | Moc[Toronto] | Qwell funny |
| 01:53.43 | tainted- | denon if u hack your own break out box |
| 01:53.57 | Qwell | Moc[Toronto]: Was wondering if anybody would catch it :p |
| 01:54.02 | denon | tainted-: cisco's supported their own poe before the standard was ratified .. |
| 01:54.15 | denon | tainted-: and pretty much every other decent vendor supports either the standard, or both |
| 01:54.33 | ALapo | i got info |
| 01:54.36 | ALapo | from dialing now |
| 01:54.42 | tainted- | hmm |
| 01:54.44 | tainted- | didn't know that |
| 01:54.48 | Veto | denon: thanks info, think I go with just plain 7060 for ~$270 for first one...maybe G-CH1 later. |
| 01:54.57 | denon | 7060? |
| 01:55.04 | Veto | err, 7960 |
| 01:55.15 | ALapo | -- Call accepted by 65.39.205.121 (format ulaw) |
| 01:55.15 | ALapo | -- Format for call is ulaw |
| 01:55.25 | ALapo | Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX Subclass: ACK |
| 01:55.25 | ALapo | Timestamp: 02383ms SCall: 00001 DCall: 01169 [65.39.205.121:4569] |
| 01:55.25 | ALapo | -- Hungup 'IAX2/FWD/1' |
| 01:55.25 | ALapo | == No one is available to answer at this time |
| 01:55.26 | tainted- | ALapo what did u fix |
| 01:55.33 | ALapo | i got debug info to show |
| 01:55.43 | ALapo | so it seems to try to connect to them |
| 01:55.49 | file[laptop] | what's your dial line like? |
| 01:56.11 | tainted- | he is using AMP file |
| 01:56.26 | file[laptop] | yes I know but if he can see that then he is looking at the CLI |
| 01:56.31 | file[laptop] | and can see what the dial line looks like there |
| 01:56.41 | ALapo | dial line? |
| 01:56.58 | ALapo | -- Executing Macro("SIP/200-48f4", "dialout-trunk|2|393612") in new stack |
| 01:57.10 | ALapo | -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix |
| 01:57.10 | ALapo | fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf |
| 01:57.19 | file[laptop] | pastebin.ca |
| 01:57.19 | at561 | ALapo try calling my FWD number 638271 |
| 01:57.23 | file[laptop] | go, learn to use it :) |
| 01:57.24 | TUplink | how do i get a list of codecs to enable? |
| 01:57.27 | kam_ | hmm make clean;make install fails on zaptel |
| 01:57.34 | ALapo | CALLED NUMBER : 393393612 |
| 01:57.34 | ALapo | CALLING NUMBER : 645311 |
| 01:57.45 | ALapo | pastebin.ca? |
| 01:57.51 | at561 | but paste any results to a private message window or pastebin.ca |
| 01:57.52 | ALapo | where is a helpfile for it |
| 01:57.55 | file[laptop] | ALapo: I was talking to you. and btw, called number is wrong... the 393 should be gone |
| 01:58.03 | Qwell | its there twice even, heh |
| 01:58.04 | ALapo | hmm |
| 01:58.06 | file[laptop] | ALapo: you should only be calling 612 |
| 01:58.08 | at561 | it's just like a free forum where you paste crap and it generates a url for the crap you pasted |
| 01:58.09 | ALapo | what should my outbound rule be? |
| 01:58.27 | ALapo | at561: they asked for some info so i can ge some help, sorry |
| 01:58.45 | ALapo | dial pattern |
| 01:58.51 | at561 | something like exten => _393.,1,Dial(IAX2/yourfwdid:yourpass@iax2.fwdnet.net/${EXTEN:3},60,r) |
| 01:58.58 | ALapo | in what file? |
| 01:59.07 | at561 | extensions.conf, under a context you're dialing in |
| 01:59.14 | file[laptop] | oh oh oh |
| 01:59.34 | file[laptop] | eh eh eh |
| 02:00.04 | ALapo | exten => 393,1,Macro(dialout-trunk,2,${EXTEN}) |
| 02:00.04 | ALapo | exten => 393,2,Macro(outisbusy); No available circuits |
| 02:00.11 | ALapo | i have that under my extensions_additional.conf |
| 02:00.23 | file[laptop] | I looked at AMP for someone once, then I proceeded to stab my eyes |
| 02:00.47 | ALapo | dont worry, im going to learn asterisk once my card comes |
| 02:00.53 | ALapo | but not yet |
| 02:00.57 | ALapo | today is my first day with asterisk |
| 02:01.15 | at561 | it looks like you have an unnecessarily complex extensions.conf |
| 02:01.19 | TUplink | ALapo dont you need exten => _393.,1,Macro(dialout-trunk,2,${EXTEN}:3) |
| 02:01.24 | Qwell | at561: welcome to AMP |
| 02:01.24 | at561 | and i have to finish my own junk so afk |
| 02:01.37 | ALapo | it was setup by default TUplink when i made the changes via AMP |
| 02:01.59 | TUplink | ewwwww just use the text editor |
| 02:02.18 | ALapo | i am |
| 02:02.25 | ALapo | but i dont know whati m doing yet |
| 02:02.31 | ALapo | i will in a few days when i know more about asterisk |
| 02:02.37 | at561 | paste your entire conf file and its included crap into pastebin.ca |
| 02:02.37 | TUplink | well sorry but i have to go. |
| 02:02.45 | ALapo | which conf file? |
| 02:02.50 | at561 | extensions |
| 02:02.54 | file[laptop] | at561: you'll regret asking that |
| 02:02.55 | tainted- | the prettiest one |
| 02:02.57 | file[laptop] | cause it's U-G-L-Y |
| 02:03.06 | ALapo | ok |
| 02:03.07 | file[laptop] | and will make you blink and go "uh... I won't help with THAT" |
| 02:03.08 | at561 | okay nevermind then |
| 02:03.14 | ALapo | probably |
| 02:03.15 | ALapo | it's huge |
| 02:03.18 | at561 | just install regular asterisk because it's easier |
| 02:03.21 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 02:03.26 | TUplink | the way i see it is your extention is way wrong |
| 02:03.46 | TUplink | need _393. |
| 02:03.51 | ALapo | at561: i plan to in a few days once my new voicecard comes |
| 02:03.58 | TUplink | the FWD site tells you how to do it |
| 02:03.59 | ALapo | it's pasting to pastebin.ca |
| 02:04.07 | ALapo | i dont understand what they ask me to do |
| 02:04.10 | *** join/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net) |
| 02:04.25 | ALapo | for example |
| 02:04.26 | ALapo | Don't forget to remove FWD's sip register command in sip.conf. |
| 02:04.28 | at561 | if your conf file takes this long to upload something's profoundly wrong |
| 02:04.30 | ALapo | what do they mean by that |
| 02:04.40 | ALapo | pastebin.ca is being flakey |
| 02:05.10 | ALapo | wont load for me any more |
| 02:05.21 | tainted- | try pastebin.com |
| 02:05.26 | tainted- | it's up for me |
| 02:05.31 | at561 | start reading all of voip-info.org if it's up |
| 02:05.43 | ALapo | http://pastebin.com/273479 |
| 02:05.48 | ALapo | yeah i was doing that |
| 02:05.55 | ALapo | thats where i got the AMP info for FWD |
| 02:06.07 | tainted- | lol omg |
| 02:06.30 | *** join/#asterisk DannyF (~dannyf@h78n3c1o848.bredband.skanova.com) |
| 02:06.33 | ALapo | so i take it a "normal" extensions.conf file isnt this huge? |
| 02:06.37 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 02:06.46 | tainted- | normal ones are bigger |
| 02:06.48 | tainted- | and wider |
| 02:06.51 | ALapo | lol |
| 02:06.54 | ALapo | :S |
| 02:06.58 | tainted- | jk |
| 02:07.05 | tainted- | normal ones are maybe 1/10 that size |
| 02:07.07 | ALapo | ah |
| 02:07.10 | ALapo | theres another one too |
| 02:07.23 | ALapo | http://pastebin.com/273480 |
| 02:07.30 | ALapo | thats extensions_additional.conf |
| 02:07.38 | at561 | normal ones are also not a php script with #includes |
| 02:08.01 | tainted- | ALapo u are not going to get much support if u keep using AMP |
| 02:08.09 | ALapo | i just want to get FWD working |
| 02:08.12 | ALapo | with @ home |
| 02:08.19 | ALapo | i plan to venture into normal asterisk soon |
| 02:08.21 | ALapo | and then bug you all more :P |
| 02:09.03 | ALapo | but any reason you see why FWD would only report all circuits busy |
| 02:09.03 | tainted- | i have found the cause of the phrase 'rtfm' |
| 02:09.05 | at561 | you of course enabled iax2 access on freeworlddialup.com first, right? |
| 02:09.10 | ALapo | yep |
| 02:09.18 | ALapo | about 2 hours ago |
| 02:11.20 | ALapo | -- Called FWD/393393638271 |
| 02:11.25 | ALapo | seems like it's adding extra #'s? |
| 02:11.51 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 02:12.51 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 02:13.06 | ALapo | now if i just dial 393 i get # incomplete |
| 02:13.14 | ALapo | so it seems i got the extension configged correct nw |
| 02:13.48 | ALapo | blah |
| 02:14.20 | tainted- | lol |
| 02:14.20 | Qwell | suck it up, and use a real install of * |
| 02:14.35 | ALapo | i plan to |
| 02:14.42 | ALapo | once my FXS card comes |
| 02:14.46 | tainted- | i think ALapo just popped his cherry |
| 02:14.47 | ALapo | i was just playing around |
| 02:14.51 | Qwell | Why waste time screwing it up now? |
| 02:15.02 | ALapo | because i want to get my feet wet before jumping in the pool |
| 02:15.21 | Qwell | So install *, and setup a FWD account or something |
| 02:15.24 | tainted- | well u got it to say 'all circuits are busy now' |
| 02:15.27 | Qwell | You aren't learning shit... |
| 02:15.27 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 02:15.28 | Veto | Always dive in without testing the water...it could be cold ALapo. |
| 02:15.31 | tainted- | that's getting your feet wet |
| 02:16.05 | Qwell | its like hes taking a pH test, before judging the temperature of the water, so he can stick a toe in |
| 02:16.23 | ALapo | tainted-: been saying that since i first tried |
| 02:16.26 | Veto | lol |
| 02:16.26 | ALapo | so ive made no progress |
| 02:16.30 | ALapo | the recording is made locally |
| 02:16.33 | ALapo | not from FWD |
| 02:16.34 | ALapo | brb |
| 02:16.45 | tainted- | can't take the ph Test until the ph Test arrives in the mail |
| 02:16.59 | *** join/#asterisk iq (~iq@70-59-161-65.omah.qwest.net) |
| 02:17.10 | Qwell | tainted-: So, we still have time to whiz in it?.. |
| 02:17.31 | Veto | no shit, jump in before someone pisses in the pool! |
| 02:17.53 | tainted- | Qwell did u self test your prostate and bladder? |
| 02:18.07 | Qwell | tainted-: nope, I'm just gonna do it |
| 02:18.17 | Qwell | no fun waiting |
| 02:18.18 | tainted- | Veto and then get pissed on? no thanks, i'm gonna wait for the ph test |
| 02:18.51 | Veto | speaking of which...modifying [stdexten] to be [fwdexten] where pass primary and cell number, ring primary 15-20secs then transfer to cell |
| 02:19.08 | Veto | cell is via IAX2/<blah>/<cell number> |
| 02:19.46 | Veto | often seem to get 'all circuits are busy' in log when doing transfer to cell (SIP rings fine) |
| 02:20.03 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 02:20.10 | tainted- | which provider |
| 02:20.11 | Mazda-MX5 | Hi~~ All ,,,, |
| 02:20.20 | tainted- | Mazda-MX5! |
| 02:20.39 | Mazda-MX5 | ya , I am here |
| 02:21.00 | Veto | voipjet, direct call works fine. using sNOANSWER and adding a 1, playback(transfer), 2,Dial(<stuff) |
| 02:21.02 | tainted- | we've been waiting |
| 02:21.39 | tainted- | hmm |
| 02:22.12 | *** join/#asterisk cpatry (~JunK-Y@modemcable174.107-81-70.mc.videotron.ca) |
| 02:22.36 | Veto | just added one arg to stdexten so ${ARG3} is an external IAX2 via voipjet. |
| 02:22.49 | Mazda-MX5 | I have trouble , How can I enable "call waiting" function in SIP channel ? |
| 02:23.31 | Veto | Mazda-MX5, my wife still has her 90 miata, still a great car. |
| 02:23.54 | Mazda-MX5 | Oh, great ~~ |
| 02:24.18 | Mazda-MX5 | is 6MT ? or 4AT ? |
| 02:24.43 | tainted- | Veto just started happening? |
| 02:25.01 | *** join/#asterisk kanzure (~bbishop@cpe-66-68-141-11.austin.res.rr.com) |
| 02:25.21 | Veto | No, just trying to get the macro to work at all, worked a couple of times but I -think- i still hade $ARG3 hardcoded in (was in a rush) |
| 02:25.55 | Veto | Maz, don't know the difference...power package 2nd year, red. |
| 02:26.54 | Mazda-MX5 | oh , but I love MT very much , |
| 02:27.02 | Veto | Here is all I have changed from stdexten: |
| 02:27.13 | Veto | exten => s-NOANSWER,1,Playback(transfer) |
| 02:27.14 | Veto | exten => s-NOANSWER,2,Dial(${ARG3}) |
| 02:27.49 | Mazda-MX5 | "call waiting" is the standard function ? Why My asterisk ca not run |
| 02:28.05 | tainted- | mazda is either love it or hate it |
| 02:28.12 | Mazda-MX5 | I always become call forward |
| 02:28.25 | Veto | Where call looks like: s,1,Macro(fwdexten,1111,SIP/rsnow,IAX2/XYZ@voipjet/1NnnNnnNnnn) |
| 02:28.29 | at561 | ALapo, it looks like your asterisk is dialing FWD with the 393 prepended to the actual number. you should cut off the prefix when dialing by taking the variable name and putting a colon after it followed by the length of the prefix you want to cut. for instance Dial(IAX2/666:secret@iax2.fwdnet.net/${EXTEN:3}) will chop off 393 from ${EXTEN} |
| 02:28.35 | Mazda-MX5 | I always become "call forward if busy" |
| 02:28.55 | Veto | 268k miles on hers, still in GREAT shape..hardtop, rag top does need replacing. |
| 02:29.20 | nestAr | i had a miata |
| 02:29.21 | nestAr | miss it |
| 02:29.23 | nestAr | :) |
| 02:29.32 | tainted- | Veto no sure |
| 02:29.47 | tainted- | nestAr they haven't changed the styling on the miata in years |
| 02:29.55 | tainted- | Veto might want to look at what $ARG3 is |
| 02:29.59 | nestAr | well, 3rd gen is coming out soon |
| 02:30.09 | nestAr | first real restyling |
| 02:30.18 | Veto | tainted: a couple months ago someone told me I was going to run into some issue with this due to (maybe?) bug/issue with Dial/Dial. |
| 02:30.46 | Veto | tainted: ${ARG3} would be: IAX2/XYZ@voipjet/1NnnNnnNnnn |
| 02:30.58 | tainted- | is it completing the call? |
| 02:31.04 | tainted- | or just rejecting with all circuits busy |
| 02:31.04 | Veto | no |
| 02:31.25 | tainted- | are u sure direct dial works? |
| 02:31.31 | Veto | rejecting, looks like a handoff issue...voipjet doesn't seem to connect |
| 02:31.48 | Veto | log is: trying....all busy or the like. |
| 02:32.08 | tainted- | try with voip phone behind asterisk |
| 02:32.11 | *** part/#asterisk cpatry (~JunK-Y@modemcable174.107-81-70.mc.videotron.ca) |
| 02:32.21 | tainted- | that dialstring |
| 02:32.22 | Veto | I'll frutz with it somemore, just wondering if you knew the 'issue' I had been told about with trying this type of forwarding. |
| 02:32.34 | tainted- | and see if its provider or dialstring |
| 02:32.46 | tainted- | no, never heard of fwding issue |
| 02:32.59 | Veto | let me go try that. |
| 02:33.29 | Mazda-MX5 | Have another free software G729 codec ? |
| 02:33.54 | tainted- | Mazda-MX5 what do u mean |
| 02:34.17 | remmo | there are no free g729 codecs |
| 02:34.22 | Mazda-MX5 | beacuse of G729 of Digium is close source , I must porting into MIPS .. |
| 02:34.46 | tainted- | remmo define 'free' |
| 02:34.48 | Mazda-MX5 | I must have source to cross-compile into mips |
| 02:34.59 | remmo | you still have to pay royalties |
| 02:35.16 | tainted- | remmo there are illegal g729 codecs |
| 02:35.21 | tainted- | with sources |
| 02:35.23 | scrubb | I have a zaptel question. If I run two instances of asterisk as a non root user, can they share access to the zaptel device for timing and channel use? |
| 02:35.23 | remmo | you might find a g729 that is active but you are still obligated to pay royalties |
| 02:35.39 | remmo | tainted: YES, it does exist |
| 02:35.39 | tainted- | ahh |
| 02:35.40 | Mazda-MX5 | digium can ge me source when I pay lisence cost?? |
| 02:35.54 | tainted- | Mazda-MX5 no |
| 02:35.59 | tainted- | u don't get source |
| 02:36.02 | remmo | MX5: dont know, never played with digium codec |
| 02:36.07 | tainted- | u get object |
| 02:37.03 | scrubb | anyone know about sharing a 4 port pri board between multiple non-root asterisk processes? |
| 02:37.19 | scrubb | I know I can't share the same channels, but what about the spans and timing? |
| 02:38.02 | Mazda-MX5 | where are illegal g729 codecs with source ? |
| 02:39.13 | scrubb | on the illegal sites? |
| 02:40.44 | at561 | illegal.com |
| 02:41.07 | Veto | tainted: who knows, I put 'IAX2/<blah>/<number>' around it and it works fine now (single quotes added). maybe it wasn't me to begin with. |
| 02:41.11 | scrubb | www.ImTooCheapToPayOthersForTheirWork.com? |
| 02:41.26 | Veto | both sip -> sip -> cell and cell -> iax -> sip -> cell |
| 02:41.52 | scrubb | so no takers on the zaptel sharing question. . .. To many voipers around ;-) |
| 02:42.12 | at561 | yeah. but you can send me hardware and i'll look into it |
| 02:42.19 | *** join/#asterisk dwmw2_gone (dwmw2@baythorne.infradead.org) |
| 02:42.21 | scrubb | :-) |
| 02:42.27 | tainted- | Veto yea voipjet sucks ass |
| 02:42.34 | *** join/#asterisk Strom_C (~strom@66.159.243.60) |
| 02:42.40 | scrubb | yeah, I can, its just I don't have enough hours in a freakin day. |
| 02:43.13 | Veto | tainted: nice prices, but questionable in some ways it seems. |
| 02:43.23 | tainted- | Mazda-MX5 http://www.voip-info.org/wiki-ITU+G.729 |
| 02:43.29 | Veto | tainted: how about sixtel / iax.cc? |
| 02:43.31 | Strom_C | woot, I got my old packet8 dta-310 working with my asterisk box :) |
| 02:43.44 | tainted- | Veto i've heard that sixtel is reasonable |
| 02:44.01 | tainted- | Veto nufone sucks |
| 02:44.12 | tainted- | Veto voicepulse is decent |
| 02:44.23 | Mazda-MX5 | Oh~ great! Thank you very much , tainted |
| 02:44.41 | tainted- | Veto gafachi took my money and ran |
| 02:44.42 | Veto | tainted: I know support can be rough, but I want good price and quality iax...I can deal with tech support that isn't right there (as long as it works!) |
| 02:44.54 | Veto | tainted: ack! |
| 02:45.01 | tainted- | Veto voicepulse then |
| 02:45.10 | tainted- | anyone disagree pipe up |
| 02:45.20 | tainted- | idling masses |
| 02:45.24 | Strom_C | asterlink |
| 02:45.51 | Veto | tainted: isn't voicepulse single sip channel? |
| 02:45.53 | tainted- | Strom_C is that yours |
| 02:45.57 | Strom_C | no |
| 02:46.00 | tainted- | Veto connect.voicepulse.com |
| 02:46.03 | tainted- | they are IAX |
| 02:46.18 | Veto | ahh! I always saw people talking about them and couldn't find IAX. |
| 02:46.45 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 02:46.49 | at561 | me too |
| 02:46.55 | tainted- | asterlink supports fax spammers and autodialers |
| 02:47.00 | at561 | maybe nufone will be better after they add capacity |
| 02:48.19 | tainted- | someone who providers good support with quality service will clean up this market |
| 02:48.27 | tainted- | the market dictates it |
| 02:48.56 | tainted- | and these horseshit fly-by-nights will go bye bye |
| 02:49.05 | at561 | hm. voicepulse doesn't charge for incoming calls? |
| 02:49.18 | tainted- | at561 $11/month flat |
| 02:49.25 | tainted- | 4 concurrent incoming/outgoing |
| 02:49.35 | at561 | that page said no monthly fees or minimums |
| 02:49.40 | tainted- | i've had some issues though.. minor stuff |
| 02:49.57 | tainted- | well, US termination is like 2.5 or 3 cents / min |
| 02:50.24 | tainted- | only 'unlimited' provider is BV, and their 'unlimited' is around 1000 min/month |
| 02:50.50 | at561 | so what kind of service do i get from voicepulse for absolutely no monthly fee besides the account deposit? |
| 02:51.10 | Qwell | at561: I quite like nufone |
| 02:51.23 | tainted- | at561 up to 4 concurrent incoming |
| 02:51.24 | at561 | i haven't tried them yet, myself |
| 02:51.25 | at561 | can't |
| 02:51.32 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 02:51.42 | Qwell | at561: talk to shido, I bet he can get you started |
| 02:51.45 | at561 | what's the $11/flat for? |
| 02:51.48 | Qwell | speaking of whom...has anybody seen him lately? |
| 02:52.20 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 02:52.28 | at561 | do they sell dids, qwell? |
| 02:52.29 | Veto | off topic: ndmpcopy is slow... |
| 02:52.37 | Mazda-MX5 | Hi, QWell ~ |
| 02:52.37 | Qwell | at561: MI and US48 toll-free currently |
| 02:52.56 | at561 | how much for toll free? |
| 02:53.03 | Qwell | 2c/min incoming |
| 02:53.07 | tainted- | if there is anyone offering US term (SIP or IAX) please msg me |
| 02:53.09 | at561 | tnx |
| 02:53.29 | Veto | same as iax.cc/sixtel for incoming us48 toll-free then? 2c/min. |
| 02:53.35 | *** join/#asterisk Kel (~kel@c-67-181-56-160.hsd1.ca.comcast.net) |
| 02:53.38 | *** part/#asterisk Kel (~kel@c-67-181-56-160.hsd1.ca.comcast.net) |
| 02:53.41 | Qwell | Veto: except nufone doesn't suck... |
| 02:54.08 | tainted- | Veto just don't make any or take any calls |
| 02:54.16 | Veto | lol |
| 02:55.06 | ALapo | wow |
| 02:55.08 | ALapo | that was a long brb |
| 02:55.11 | at561 | voipjet has worked ok for me but i don't send out huge numbers of calls |
| 02:55.21 | at561 | well scroll back ALapo because there was a reply |
| 02:56.20 | ALapo | thanks at561 |
| 02:56.23 | ALapo | i'll check that out |
| 02:57.11 | ALapo | was that the extended_additional? |
| 02:57.14 | ALapo | err |
| 02:57.16 | ALapo | extensions |
| 02:57.38 | at561 | wherever the actual Dial is being invoked from. no idea! |
| 02:58.04 | at561 | the donated did from a friendly local has performed beautifully |
| 02:58.23 | at561 | i'm probably helped by the fact that his server is sitting in my same neighborhood |
| 02:58.41 | ALapo | yeah im confused where the dial is being invoked from |
| 02:58.45 | ALapo | i cant find it in any of the config files |
| 02:58.53 | at561 | only cthulhu knows |
| 02:59.10 | at561 | also, maybe your final dial is being invoked by somebody else who should trim off the 393 |
| 02:59.24 | ALapo | possibly |
| 02:59.35 | ALapo | im just going to order this card, give up for now, and start from scratch when the card comes |
| 02:59.35 | ALapo | :) |
| 02:59.49 | at561 | you could start early |
| 03:00.14 | ALapo | eh whn i learn to config it i prolly wont need to learn how to do it over voip iax initially |
| 03:00.22 | ALapo | just with the PSTN |
| 03:00.30 | Qwell | its practically the same config |
| 03:00.32 | CTJ | I got a message from asterisk saying "Registration from '<sip:username@asterisk_ip:5060>' failed for 'client_ip' How do I get asterisk to print out why it failed? |
| 03:00.37 | ALapo | all the better |
| 03:00.37 | ALapo | :) |
| 03:01.13 | ALapo | once i get proficient at asterisk... i wonder what people are charging to deploy IP phone systems |
| 03:01.26 | ALapo | >:) |
| 03:02.59 | ALapo | almost happy 4/20 to all |
| 03:05.08 | ALapo | i found the register= line |
| 03:08.27 | *** join/#asterisk theHub_ (~karl@ool-182cba82.dyn.optonline.net) |
| 03:17.48 | *** join/#asterisk iq|laptop (~iq@70-59-161-65.omah.qwest.net) |
| 03:20.41 | *** join/#asterisk reph (~r@user-24-236-86-244.knology.net) |
| 03:25.19 | jakepdev | anyone know about iax2 debugging - how do I find out if it's network or the server? |
| 03:29.29 | `Sauron | Learn to use a network sniffer |
| 03:30.57 | *** join/#asterisk bonez41 (~aint@c-67-166-77-14.hsd1.ut.comcast.net) |
| 03:34.59 | shmaltz | anybody here knows if there are any plans for T38 support in asterisk? |
| 03:36.43 | ALapo | bought my FXS card |
| 03:36.44 | ALapo | :) |
| 03:37.37 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 03:41.01 | Mazda-MX5 | I have trouble , Why do I get "Unknown RTP codec 96 received" when I dial ? |
| 03:44.45 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 03:44.52 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 03:46.22 | *** join/#asterisk Moc[Toronto] (~mochouina@209.47.87.2) |
| 03:52.20 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 03:52.23 | *** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net) |
| 03:54.04 | *** join/#asterisk SimonR (~SimonR@Toronto-HSE-ppp3736980.sympatico.ca) |
| 03:55.21 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 03:55.32 | Mazda-MX5 | brc_ , hi ~ |
| 03:56.13 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 03:56.49 | Mazda-MX5 | Why do I get "Unknown RTP codec 96 received" when I dail ?? |
| 03:57.02 | tessier | How do people handle cdr's when they have lots of calls? Everything in one table or one table per client? |
| 03:57.15 | brc_ | 19" LCD, $260 http://snurl.com/dell19_260 20" LCD, $393 http://snipurl.com/dell20_393 |
| 03:57.22 | tessier | I am looking at millions of cdr's in one big table in mysql. Shouldn't be a problem I would think. |
| 03:57.27 | brc_ | hello Mazda-MX5 |
| 03:57.39 | Mazda-MX5 | hello , brc_ |
| 03:57.47 | brc_ | hola Mazda-MX5 |
| 03:58.27 | *** join/#asterisk mithro (~tim@150.203.247.9) |
| 03:58.48 | Mazda-MX5 | have no body know ...... |
| 03:59.12 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 03:59.12 | *** mode/#asterisk [+o bkw_] by ChanServ |
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| 04:03.44 | reph | sup mark |
| 04:04.19 | Strom_C | a company I had two addresses ago is suddenly getting lots of credit card offers at my current address |
| 04:04.26 | Strom_C | thank god for my paper shredder |
| 04:04.33 | Corydon76-home | Mazda-MX5: probably your SIP phone is sending a codec it did not negotiate |
| 04:05.06 | Corydon76-home | It's not supposed to do that, but many still do (notably Cisco phones) |
| 04:05.11 | *** join/#asterisk tugalone (~tugalone@pcp0010303951pcs.avenel01.nj.comcast.net) |
| 04:05.21 | reph | Strom_C do like my mom does and save junk mail for the winter, saves wood =) |
| 04:05.47 | Mazda-MX5 | I am not use CISCO phone |
| 04:05.51 | Corydon76-home | Cisco phones love sending silence suppression, even when it's clear from the negotiation that the server doesn't support it |
| 04:06.09 | Corydon76-home | What SIP phone, then? |
| 04:06.48 | Mazda-MX5 | corydon76, thank you , this message is happen when I test "call waiting" |
| 04:07.00 | Micc | would it be cool to be able to place config commands into sql for the asterisk config system? |
| 04:07.00 | Corydon76-home | Using what phone? |
| 04:07.27 | Mazda-MX5 | a my three party product |
| 04:07.28 | newl | Micc: with Realtime you can do just that. |
| 04:07.50 | Micc | newl: Where can I get it? |
| 04:07.58 | Corydon76-home | Mazda-MX5: does it have a name? |
| 04:08.14 | Corydon76-home | Micc: realtime is CVS-HEAD only and it's very poorly documented |
| 04:08.36 | Mazda-MX5 | A and B is connect and talked , when I press B's digit , message "Unknown RTP codec 96 received" put |
| 04:08.41 | Corydon76-home | In fact, it's not very clear by reading the source how to interface with Realtime |
| 04:08.57 | Mazda-MX5 | ip phone name is "LP-201" |
| 04:09.06 | newl | the docs on voip-info are fine. |
| 04:09.14 | Corydon76-home | Mazda-MX5: model name and manufacturer |
| 04:09.31 | Micc | Corydon76-home: is CVS-HEAD pretty stable right now? |
| 04:09.56 | Micc | Corydon76-home: I think I got the head last week sometime. |
| 04:10.01 | *** join/#asterisk Rick_Hunter (~rhunter@02-083.008.popsite.net) |
| 04:10.02 | Micc | would it be in there? |
| 04:10.10 | Corydon76-home | Micc: I can't in good conscience recommend running CVS-HEAD to anybody who is not already a developer |
| 04:10.20 | Corydon76-home | at least, not in production |
| 04:10.31 | newl | What a well thought out answer. hehe |
| 04:11.24 | Mazda-MX5 | it is "1FXS/IFXO" |
| 04:11.36 | PTG1234 | anyone have a good source for ebooks? :) |
| 04:11.41 | Micc | I'm a developer but I need that functionality for a production system next week. |
| 04:11.44 | Qwell | PTG1234: what kind of ebooks? |
| 04:11.52 | Mazda-MX5 | i do not know this its all name |
| 04:12.12 | Corydon76-home | Mazda-MX5: ask your phone vendor for support |
| 04:12.14 | Mazda-MX5 | corydon76 , is it the ip phone problem ? |
| 04:12.40 | Corydon76-home | Tell them you want a SIP-compliant phone |
| 04:12.51 | Corydon76-home | Not the garbage you have now |
| 04:13.09 | PTG1234 | qwell: any |
| 04:13.11 | Mazda-MX5 | OK, thank you very much , corydon76 |
| 04:13.16 | Qwell | PTG1234: project gutenburg |
| 04:13.24 | PTG1234 | whats that |
| 04:13.30 | Qwell | erm, erg |
| 04:13.35 | Qwell | http://www.gutenberg.org/ |
| 04:13.47 | Qwell | books that are out of copyright, converted to ebook form |
| 04:13.50 | Qwell | (mostly) |
| 04:13.56 | jakepdev | Hey `Sauron - If you're still on - I know how to use a network sniffer. What I was hoping for is some Linux commands to tell me if the processor is overburdened and that is causing the audio to sounds crappy |
| 04:14.34 | Corydon76-home | jakepdev: See uptime(1) |
| 04:14.47 | jakepdev | ok thanks Corydon |
| 04:14.57 | Corydon76-home | Or top(1) |
| 04:15.02 | PTG1234 | hmm |
| 04:15.03 | dmccollum | or sar |
| 04:15.17 | SimonR | jakepdev, try # top -d 1 |
| 04:15.25 | Corydon76-home | $ man sar |
| 04:15.25 | Corydon76-home | No manual entry for sar |
| 04:15.27 | SimonR | just watch the processor idle |
| 04:15.39 | SimonR | any other ideas, anyone? |
| 04:15.58 | jakepdev | i'll try all of them :) |
| 04:16.00 | SimonR | you can use hdparm to check if disk i/o wait is an issue |
| 04:17.25 | jakepdev | I just had no idea being new at Linux. Thanks Cory/dm/Simon |
| 04:17.58 | dmccollum | top is the easiest just to see how busy the CPU is. |
| 04:18.18 | jakepdev | ok - i'll try that |
| 04:19.13 | Corydon76-home | Or: watch cat /proc/loadavg |
| 04:20.24 | jakepdev | k |
| 04:21.06 | Corydon76-home | Or, to limit it some more: watch "cut -f1-3 -d' ' < /proc/loadavg" |
| 04:22.23 | SimonR | you could install cactai to watch the load... check out hte screenshots for cactai or rrdtool |
| 04:22.50 | SimonR | anyone know what makes the audio sound crappy? |
| 04:23.09 | SimonR | I know that pci interrupt problems will do it, inadequate disk i/o, etc. |
| 04:23.19 | SimonR | If you disk is set to a dma level above 3, it will cause this |
| 04:27.21 | nestAr | hrmmm.. |
| 04:27.36 | nestAr | why does setgroup and checkgroup not seem to work in CVS-HEAD |
| 04:32.27 | *** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net) |
| 04:32.50 | nestAr | well |
| 04:32.57 | nestAr | i guess i roll back to stable |
| 04:43.00 | *** join/#asterisk ToyMan (~konversat@user-12lcqur.cable.mindspring.com) |
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| 05:04.39 | *** join/#asterisk TheEmperor (~user@203.114.48.47) |
| 05:06.20 | SimonR | how'd you monitor your ethernet interface for dropped packets and its response time? |
| 05:06.28 | SimonR | I mean from your local loopback? |
| 05:06.31 | SimonR | ping localhost? |
| 05:06.56 | SimonR | or is there an interface monitoring utility you can run short of using rrdtool or mrtg or simialr? |
| 05:08.17 | Micc | if I do cvs checkout zaptel libpri asterisk will that give me the latest? the CVS-HEAD |
| 05:09.58 | drumkilla | yes |
| 05:13.56 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 05:14.06 | tengulre | Hi, all |
| 05:14.27 | odie_flocon | Heyup |
| 05:16.16 | tengulre | odie_flocon, does the asterisk support SHT voice card? |
| 05:16.55 | Qwell | shit voice cards? yeah, it supports the clone x100p |
| 05:16.59 | Qwell | Thanks, I'll be here all night. |
| 05:17.01 | drumkilla | I don't know what a "SHT" card is, but it sounds pretty low quality ... hehe |
| 05:17.11 | drumkilla | dang. you beat me to the punch :( |
| 05:17.14 | Qwell | drumkilla: :p |
| 05:19.02 | tengulre | Qwell, the asterisk support SHT? really? |
| 05:19.12 | *** part/#asterisk theHub_ (~karl@ool-182cba82.dyn.optonline.net) |
| 05:19.14 | Qwell | tengulre: I don't know. I was making a (bad) joke. |
| 05:19.34 | tengulre | Qwell,:) |
| 05:22.07 | tengulre | Qwell,I have some message about this card, can you help me? does the card possible support it? |
| 05:22.19 | Qwell | I don't know |
| 05:22.38 | tengulre | Qwell,I sent you? |
| 05:23.44 | tengulre | Qwell, OK? |
| 05:23.59 | tengulre | any1 help me? |
| 05:24.01 | tengulre | Please? |
| 05:24.08 | mindamp|away | fuck off |
| 05:24.46 | tengulre | mindamp|away, what your means? |
| 05:26.45 | tengulre | hi,all! |
| 05:30.18 | Micc | so I pass -r HEAD ot cvs checkout command to get the head, right? |
| 05:30.26 | Micc | It seems to be working so I'll assume I'm right. |
| 05:35.36 | JonR800 | no.. you shouldn't have to put -r |
| 05:36.49 | tengulre | up |
| 05:39.18 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com) |
| 05:42.08 | *** join/#asterisk mindamp (~mindamp@AC8CF84A.ipt.aol.com) |
| 05:43.50 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
| 05:43.54 | ellvis | hi people |
| 05:45.47 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 05:46.24 | tengulre | ellvis,hi |
| 05:46.39 | ellvis | tengulre, hola |
| 05:47.04 | ellvis | i am scared to ask anything here today:) |
| 05:47.17 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 05:49.01 | tld | Does Asterisk support media and signaling encryption? |
| 05:49.28 | shmaltz | tld, yes if you use a VPN |
| 05:49.35 | shmaltz | but no within asterisk |
| 05:50.05 | *** join/#asterisk sympad (~Misha@195.138.127.98) |
| 05:50.09 | shmaltz | tld, you know of any phone that supports it? |
| 05:50.46 | tld | shmaltz: That would have been my next question. ;) |
| 05:51.03 | shmaltz | tld, just like PSTN |
| 05:51.10 | tld | shmaltz: The Cisco 7960 has some options for it, but I'm not sure it'll let you use it without the CallManager. |
| 05:51.41 | shmaltz | the only way to get secure transmission is to have 2 endpoints that encrypt/decrypt and then use analog phones |
| 05:51.57 | shmaltz | in voip you would use VPN |
| 05:57.05 | *** join/#asterisk tld (~tld@80.203.70.227) |
| 05:57.22 | tld | Sorry, didn't mean to be rude. Hardware trouble. |
| 05:58.07 | tld | There's SRTP and similair standards, so there's no reason why VoIP phones, and PBXes, should not be able to offer more security than the PSTN. |
| 05:59.46 | Qwell | tld: from what I understand (which isn't much), heres the problem |
| 06:00.14 | Qwell | There isn't much point for PBXs to support things like SRTP, if hardware (or even other PBXs) isn't going to support it |
| 06:00.31 | Qwell | and if the PBXs (and other hardware) don't support it, why would the phones need to support it? |
| 06:01.16 | Qwell | Of course, I have no clue...just a thought |
| 06:01.26 | tld | Yeah, probably part of the problem. |
| 06:01.33 | tld | Just a bit sad really. |
| 06:01.43 | Qwell | its unfortunate, sure, but its how it goes |
| 06:01.57 | Qwell | could always use a tunnel like shmaltz suggested |
| 06:02.26 | ScythelX | isnt rtp random? |
| 06:02.38 | tld | ScythelX: ? |
| 06:02.46 | tld | Tunnels create a lot of problems. |
| 06:02.51 | ScythelX | ¿ |
| 06:03.12 | tld | One of the nice features of SRTP is that with Asterisk and phone support for it, you could just plug your VoIP phone into the RJ45 at your hotel room, and have a secure line. |
| 06:03.33 | tld | If you want a tunnel, you'll need some kind of VPN hardware in addition to the phone. |
| 06:03.37 | tld | A lot less convenient. |
| 06:05.21 | tengulre | any1 know html2tiff? |
| 06:06.44 | Micc | What protocol does vonage use? Because I don't like the audio quality. Is asterisk better audio quality? |
| 06:07.08 | Qwell | Micc: It uses SIP. Protocols don't help audio quality. You're thinking codecs |
| 06:07.10 | tengulre | all, how can i use fax function? |
| 06:09.24 | tengulre | ? |
| 06:09.41 | Micc | Qwell, oh, ok so could I use my vonage box with asterisk and get better quality? |
| 06:09.45 | Qwell | No |
| 06:09.58 | Qwell | doing so is against Vonage policy |
| 06:10.03 | Micc | oh. |
| 06:10.20 | tengulre | oh, |
| 06:10.33 | Micc | So is there some other box I can get that I would be able to configure for my asterisk server that I can plug a phone into. |
| 06:10.48 | tld | Qwell: btw: When it comes to hardware implementations, even the pretty basic plain Sipura phone supports Secure RTP. |
| 06:10.49 | Qwell | Micc: sure, but you'll need/want another service |
| 06:10.56 | Micc | And is asterisk setup by default with a good codec for audio quality? |
| 06:11.08 | Qwell | tld: Like I said, I didn't know what I was talking about...just guessing. |
| 06:11.16 | Micc | Qwell, I'm getting setup with nufone. |
| 06:11.23 | Qwell | Micc: There are a bunch you can use, if your provider allows them |
| 06:11.28 | Qwell | ahh, good choice :p |
| 06:11.50 | Micc | So does nufone allow a good codec? |
| 06:12.02 | Qwell | I don't know what they allow. I use ulaw, and its working fine |
| 06:12.44 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 06:15.20 | Qwell | Micc: So, I take it you were able to get ahold of shido? |
| 06:16.18 | Micc | Yes, but I still have to send them some paypal funds. |
| 06:16.33 | Micc | is ulaw a good codec or a better one than gsm? |
| 06:16.52 | Qwell | I didn't like gsm, I don't know |
| 06:17.17 | tld | Zultys 4x4 also supports encryption. |
| 06:19.03 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 06:19.32 | Qwell | Micc: Should I assume you left out the "Qwell sent me" part? ;] |
| 06:19.49 | Micc | no, I put that in there. |
| 06:19.55 | Qwell | surprising, heh |
| 06:20.46 | Micc | Well I got ahold of him on here and was talking to him then he went AFK. |
| 06:21.05 | Micc | Qwell, I also sent him an email. I don't think he correlated the two. |
| 06:21.10 | *** join/#asterisk riksta (~rick@81-178-248-131.dsl.pipex.com) |
| 06:21.22 | *** join/#asterisk NeonLevel (~NeonLevel@201.129.134.132) |
| 06:21.24 | Micc | Qwell, but I'm sure I put the, qwell sent me part in there too. |
| 06:22.32 | Micc | Qwell, I want the best audio quality, what provider/codec should I use? |
| 06:22.59 | Qwell | Micc: You'll want to look at the wiki to look at the different codecs. Nufone is great though |
| 06:23.46 | Micc | Qwell, ok, well I can't wait to try their service. How will they bill me, I take it I don't get an online statement. |
| 06:25.15 | tld | Seems to me like about half of the phones support SRTP. I'm a bit surprised that Asterisk doesn't. |
| 06:25.39 | Micc | Qwell, I heard you guys saying yesterday that its cheaper to go with a company like nufone than vonage for unlimited. I'm planning on getting 20 lines and utilizing most of them all the time. |
| 06:25.51 | tld | Also, SRTP should be in the media negotiation, so using Asterisk with two SIP phones would work, right? Or are the voice channels *always* pulled through Asterisk? |
| 06:26.56 | *** join/#asterisk NeonLevel (~NeonLevel@201.129.134.132) |
| 06:26.57 | Qwell | tld: reinvite |
| 06:27.01 | Qwell | erm, heh |
| 06:27.02 | NeonLevel | hi all |
| 06:27.05 | Qwell | tired |
| 06:27.19 | Qwell | it will go out of the loop with the right settings...don't ask me though, I need sleep |
| 06:27.49 | Qwell | Micc: yeah, 20 lines with nufone would be pretty expensive, and not "unlimited" |
| 06:27.59 | Qwell | erm |
| 06:28.03 | Qwell | with vonage |
| 06:28.18 | Qwell | maybe I'll go to sleep |
| 06:28.57 | NeonLevel | sometime ago i needed to set some timeout in order to jump to the t extension now, i can't find that command! any help? |
| 06:30.29 | NeonLevel | no-one? |
| 06:30.42 | Qwell | NeonLevel: http://www.voip-info.org/wiki-Asterisk+standard+extensions |
| 06:31.03 | NeonLevel | checking it |
| 06:31.13 | Qwell | ResponseTimeout? |
| 06:32.08 | NeonLevel | thank you very much! |
| 06:32.19 | NeonLevel | i guess the commands are changing really fast! |
| 06:32.21 | NeonLevel | ;-) |
| 06:32.26 | NeonLevel | thanks Qwell |
| 06:32.33 | Qwell | Just send money... |
| 06:32.37 | NeonLevel | ha haha |
| 06:32.40 | NeonLevel | account? |
| 06:32.55 | Qwell | That actually works in this channel? My god |
| 06:33.24 | NeonLevel | i just asked never mentioned i will do |
| 06:33.31 | Sedorox | only the n00bs Qwell.. only the n00bs... |
| 06:33.33 | Qwell | :p |
| 06:33.49 | NeonLevel | :p |
| 06:33.52 | Qwell | Sedorox: I've donated to somebody once. And I told drumkilla I owe him a few beers |
| 06:34.05 | Sedorox | hehe |
| 06:34.57 | Micc | Why am I getting a TDS error when I build the HEAD? |
| 06:35.01 | NeonLevel | where are you gauys from? |
| 06:35.17 | Micc | Oh, I remember. |
| 06:35.25 | Micc | I've got to remove the tds part. I'm using odbc. |
| 06:35.37 | *** join/#asterisk zapa (~zapa@201.135.137.236) |
| 06:35.51 | Sedorox | off to bed |
| 06:35.51 | Sedorox | night |
| 06:36.24 | tengulre | 14:37 now! |
| 06:36.28 | tengulre | 14:37:23 now! |
| 06:38.23 | zapa | hi all, is there any way to send tones |
| 06:38.38 | jakepdev | sedDTMF |
| 06:38.42 | jakepdev | SendDTMF |
| 06:39.08 | zapa | thanks jakepdev |
| 06:39.12 | jakepdev | np |
| 06:39.54 | zapa | where i need to configure ring back tone in asterisk in zapata.conf ? |
| 06:48.50 | brc_ | http://arsenal.media.mit.edu/memes/mahnamahna.mpeg |
| 06:49.47 | tengulre | what's up? |
| 06:50.28 | remmo | i have a cisco 7960 ip phone here and trying to upgrade the firmware via tftp and its not working. can anyone provide any tips or pointers pls |
| 06:51.52 | remmo | tftp barfs because of too many little packets |
| 06:54.48 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 06:58.43 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 06:59.46 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 07:05.20 | Primer | I like how the sipura can update over http |
| 07:05.23 | Primer | what an innovation! |
| 07:05.32 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 07:22.14 | tengulre | hi,all is the asterisk support rockwell modem? |
| 07:23.03 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 07:26.17 | tengulre | hi,all! |
| 07:26.22 | tengulre | no-one! |
| 07:26.50 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 07:34.24 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 07:34.42 | *** join/#asterisk Gunnar (~gunnar@213-187-176-147.dd.nextgentel.com) |
| 07:36.41 | *** join/#asterisk three55ml (~three55ml@cpe-66-68-98-68.austin.res.rr.com) |
| 07:38.54 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 07:39.13 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 07:40.16 | *** join/#asterisk jollyr (jollyr@210.16.20.27) |
| 07:41.04 | jollyr | any one here offering a billing software |
| 07:41.05 | jollyr | ? |
| 07:41.07 | *** join/#asterisk Skaag (unix@80.178.109.238.adsl.012.net.il) |
| 07:41.20 | *** join/#asterisk riksta (~rick@81-178-209-106.dsl.pipex.com) |
| 07:41.29 | *** join/#asterisk haujobb2000 (~joerg@p3EE313BA.dip0.t-ipconnect.de) |
| 07:42.06 | Skaag | can someone please help me configure 'wcfxo' cards? (Kernel Found a Wildcard FXO: Generic Clone) |
| 07:42.06 | Skaag | As it is, i'm dialing to that modem and it's not picking up |
| 07:44.57 | *** join/#asterisk fenlander (~neils@82.152.81.57) |
| 07:45.02 | *** join/#asterisk zapa (~zapa@201.135.137.236) |
| 07:45.55 | Zeeek | what are the |
| 07:46.10 | Zeeek | steps you have taken so far? |
| 07:51.41 | tengulre | Hi, Zeeek! |
| 07:52.22 | Zeeek | hello |
| 07:52.43 | Zeeek | Skaag answer my question: what have you done so far? |
| 07:54.04 | Delvar | i thnk he died |
| 07:54.23 | Delvar | poke him with a stick see if he twitches |
| 07:54.24 | Zeeek | well I didn't see the question soon enuf. |
| 07:55.08 | Delvar | only a few minuets after he asked... |
| 07:55.25 | Zeeek | and after all this isn't a paid hotline, is it? :) |
| 07:55.26 | Delvar | not like you get paid to help noobs... |
| 07:55.31 | Delvar | hehe |
| 07:55.34 | Zeeek | exaklymenté |
| 07:55.45 | Zeeek | I wish someone would help me |
| 07:55.53 | pif | I have to demo * to an advertising agency today, what "WOW!" features should I show? |
| 07:55.56 | Zeeek | I am losing what is left of my mind |
| 07:56.12 | Delvar | what you whining about? not like your losing much :) |
| 07:56.12 | Zeeek | pif do a demo of the pr0n callout center |
| 07:56.35 | Zeeek | where asterisk calls random numbers and trys to sell sexual services to people |
| 07:56.36 | pif | where should I look it up? |
| 07:56.38 | Makenshi | pif, least cost routing, ivr, vm |
| 07:56.52 | Skaag | Zeeek: sorry |
| 07:56.56 | Makenshi | you can call me if you like, sip:chills@ne-worcs.ac.uk :> |
| 07:57.09 | Zeeek | Skaag tells us more about your worries |
| 07:57.17 | pif | Makenshi: in ivr what features are the nicest? |
| 07:57.18 | Skaag | Zeeek: I've setup an extension and have it working with X-Lite, I have made a greeting for the receptionist |
| 07:57.27 | Zeeek | by the way I hope you are male with a name like that |
| 07:57.40 | Makenshi | pif, writing custom agi in perl, python, or even .net :> |
| 07:57.47 | Zeeek | eeeeeewwwwwwww |
| 07:58.02 | Skaag | Zeeek: I am male ;-) |
| 07:58.06 | pif | Makenshi: thanks |
| 07:58.11 | Zeeek | Skaag how did you install your fxo? |
| 07:58.23 | Zeeek | aside from plugging it in |
| 07:58.25 | Delvar | custm AGI is good, also Realtime CDR reports, managers love that... se who is calling who in realtime :) |
| 07:58.28 | Makenshi | Zeeek, eeeeeewwwwww? :> |
| 07:58.32 | Skaag | I just stuck my cards there and the module wcfxo was loaded automatically |
| 07:58.35 | Zeeek | .NET eeeewwwwww |
| 07:58.39 | Skaag | And then I followed the Wiki instructions |
| 07:58.47 | Zeeek | Skaag ztcfg ? |
| 07:58.50 | Skaag | (I setup /etc/zaptel.cfg) |
| 07:58.52 | Makenshi | .Net is not ew, .Net is great :) |
| 07:59.02 | Skaag | Zeeek: No I didn't use ztcfg |
| 07:59.24 | Zeeek | then you didn't follow instructions |
| 07:59.42 | jollyr | i need a windows based monitor for callin callout zapchannels and sip channels |
| 07:59.43 | jollyr | ? |
| 07:59.58 | jollyr | something like a manager |
| 07:59.59 | Skaag | Zeeek: ztcfg produced the same zaptel.conf that I made myself |
| 08:00.12 | Skaag | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 08:00.19 | Delvar | there is a flash operators pannel we are currenly playinmg with, that look sprety cool |
| 08:00.46 | Zeeek | Delvar which? FOP works great and looks good |
| 08:01.22 | Skaag | is there a utility which will convert 16khz wav to 8khz? |
| 08:01.27 | Zeeek | sox |
| 08:01.34 | Zeeek | as in man sox |
| 08:01.39 | Skaag | thanks |
| 08:01.47 | Zeeek | np |
| 08:01.48 | Delvar | i cant rember off the top of my head ill find out, its all big buttons, with drag n drop calling and realtime status of extensions, got it hooke dup to a touch screen TFT |
| 08:02.01 | Zeeek | that sounds cool |
| 08:02.27 | Delvar | will be when its configured for the local bpx |
| 08:02.40 | Delvar | at the moment only got a couple extensions and queus hooked up |
| 08:04.23 | Delvar | http://www.asternic.org/ |
| 08:06.06 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 08:06.47 | Zeeek | yeah that's the one I use |
| 08:06.51 | *** join/#asterisk mbaron (~mbaron@AVelizy-154-1-28-151.w82-124.abo.wanadoo.fr) |
| 08:06.57 | Zeeek | very useful at home to watch the office pbx |
| 08:07.08 | *** part/#asterisk mbaron (~mbaron@AVelizy-154-1-28-151.w82-124.abo.wanadoo.fr) |
| 08:09.41 | Delvar | yeah, we are considdering using it for our smaller PBX role outs |
| 08:09.56 | Delvar | im not sure how it would cope with a 100 use pbx tho :) |
| 08:10.01 | Delvar | use/user |
| 08:12.39 | Makenshi | i think the answer to reliability is to use something like dect, maybe wimax |
| 08:12.39 | Makenshi | then all the phones will be like mobile phones, with their own batteries |
| 08:12.39 | Makenshi | its impractical to provide battery backup for every handset, even with power injectors |
| 08:20.00 | Skaag | strange I can not find the "Routing" menu |
| 08:20.10 | emrah | The new sip protocol software with the new cvs version of Asterisk in very interesting |
| 08:20.16 | emrah | is* |
| 08:20.25 | Makenshi | how so? |
| 08:20.59 | emrah | Are you speaking with me Makenshi ? |
| 08:21.09 | emrah | talking* |
| 08:21.15 | emrah | sorry my English is terrible |
| 08:21.55 | Skaag | of course... definitely not talking to me... nobody talks to me |
| 08:21.55 | Zeeek | Skaagnot true :) |
| 08:21.55 | Skaag | :-) |
| 08:21.57 | Zeeek | I still don't see what your problem is, since youdodn't state it clearly |
| 08:22.03 | Skaag | I'm trying to follow the guides, Zeeek, but it's just not clear |
| 08:22.05 | Makenshi | emrah, yes! |
| 08:22.14 | Skaag | Zeeek: Well, I have the modems installed and identified |
| 08:22.15 | Zeeek | what is the output when you type ztcfg -vvvv |
| 08:22.22 | Zeeek | what modems? |
| 08:22.29 | Skaag | Channel 01: FXS Kewlstart (Default) (Slaves: 01) |
| 08:22.31 | Skaag | 1 channels configured. |
| 08:22.42 | *** join/#asterisk JerJer (~JerJer@12.173.204.122) |
| 08:22.45 | Skaag | It's a Tiger Jet Network modem, and the module wcfxo is loaded for it |
| 08:22.50 | JerJer | happy 420 all |
| 08:23.46 | Skaag | thanks JerJer |
| 08:24.13 | Skaag | I tried following this one: http://www.shritel.com/downloads/asterisk@home_v0.8_setup_guide.pdf |
| 08:24.15 | emrah | Makenshi: It more compatible to connect two Asterisks together, for example. And the transmited packed are more compatible with some devices. |
| 08:24.18 | Skaag | But it's very sketchy |
| 08:24.26 | JerJer | learn asterisk for real |
| 08:24.32 | JerJer | asterisk@home is a joke |
| 08:24.47 | emrah | I've tried to compile also and other chan_sip.so too, to see the difference. |
| 08:24.52 | Skaag | i'm just trying to find out if it actually can pick up the line and sound me the receptionist |
| 08:25.02 | emrah | another* |
| 08:25.19 | Skaag | I stuck 4 modems in there, one has a line connected to it |
| 08:26.08 | Makenshi | i'm all for sip |
| 08:26.16 | emrah | Anyone please knows a provider offering voip2pstn calls, with interesing rates, good routing, and payable by Paypal, please? |
| 08:26.20 | Makenshi | iax tries to solve a problem that has already been solved |
| 08:26.30 | Makenshi | and it doesn't address the issue of roaming between networks |
| 08:27.06 | emrah | IAX2 is bether than sip to connect two Asterisks together |
| 08:27.25 | *** join/#asterisk user_suse9 (~user_suse@62.153.57.92) |
| 08:27.35 | user_suse9 | hallo |
| 08:27.41 | emrah | but you have to be carreful for security problems |
| 08:27.59 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 08:28.02 | emrah | I like very much this protocole |
| 08:28.12 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 08:28.40 | Makenshi | yeah, like iax doesn't support s/mime |
| 08:28.47 | Makenshi | and it's only useful for audio |
| 08:28.55 | tainted- | Makenshi i don't think it was ever meant to |
| 08:29.16 | Makenshi | tainted-, exactly |
| 08:29.43 | Makenshi | with sip i can establish an encrypted session in a similar manner to encrypting my email |
| 08:29.58 | Makenshi | and verify the authenticity of the caller/callee |
| 08:30.04 | JerJer | iax supports aes media encryption |
| 08:32.35 | jakepdev | emrah- http://www.voip-info.org/wiki-VOIP+Service+Providers |
| 08:33.34 | emrah | jakepdev: I'm behind a proxy |
| 08:33.44 | emrah | and many websites doesn't work |
| 08:34.30 | jakepdev | ok |
| 08:35.03 | jakepdev | what does that have to do with the link I sent you, can you not access it? |
| 08:35.19 | emrah | No |
| 08:35.21 | emrah | It doesn't work |
| 08:35.48 | emrah | and it shit to browse with links or lyns |
| 08:35.50 | emrah | lynx |
| 08:35.51 | jakepdev | how are you going to check out VOIP providers if you can't even sign up? |
| 08:36.04 | emrah | I'm comming back in 15 minutes |
| 08:36.05 | jakepdev | most of the sign ups are web based? |
| 08:36.06 | emrah | thanks |
| 08:37.52 | jakepdev | and I don't get what the big fuss is over @home... It works fine. may need to blow away some things, but besides that I haven't seen any problems on the few installations I've done |
| 08:38.43 | Skaag | damn this is frustrating |
| 08:39.00 | Skaag | is there any way to check if my modem is working at all with asterisk, before I get into complex dial plans and routing? |
| 08:39.29 | jakepdev | modem? |
| 08:39.55 | jakepdev | are you trying to use a voice modem with *? |
| 08:40.01 | Zeeek | Skaag sorry I got a phone call. I see you use @home so I can't really help you |
| 08:40.08 | Skaag | yes |
| 08:40.13 | Zeeek | you still didn't say what "modem" you're using |
| 08:40.20 | Skaag | I'm trying to connect an old style PSTN line to * |
| 08:40.28 | jakepdev | what modem? |
| 08:40.29 | Zeeek | what hardware? |
| 08:40.32 | Skaag | I did... it's a Tiger Jet Network modem |
| 08:40.38 | jakepdev | X100P? |
| 08:40.42 | Skaag | yes |
| 08:40.49 | Skaag | I did as instructed |
| 08:41.03 | Skaag | 02:08.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 08:41.04 | user_suse9 | Hi, Ihave installed asterisk 1.0.0.20040814-1 with asterisk-capi 0.3.5-3 under suse 9.2, but asterisk dosen't start. Is it possible to get some help here? |
| 08:41.24 | Zeeek | and you type zap show channels and see what? |
| 08:41.36 | jakepdev | suse9 - it's possible - but you have to be more specific |
| 08:41.50 | user_suse9 | allright, little moment |
| 08:42.19 | Skaag | I see 1 channels |
| 08:42.35 | Skaag | *CLI> zap show channels |
| 08:42.35 | Skaag | <PROTECTED> |
| 08:42.35 | Skaag | <PROTECTED> |
| 08:42.35 | Skaag | <PROTECTED> |
| 08:42.41 | jakepdev | ~pastebin |
| 08:42.42 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 08:42.44 | Skaag | ok |
| 08:42.54 | Skaag | but it was just 4 lines, but ok |
| 08:43.25 | user_suse9 | I have 3 warnings if I try to start with "asterisk -c" WARNING[1076838528]: loader.c:423 load_modules: Loading module app_capiCD.so failed! |
| 08:43.27 | Zeeek | and what is the symptom? You can't dial on the line? |
| 08:43.37 | Zeeek | Skaag |
| 08:43.55 | jakepdev | suse9 - no errors? |
| 08:44.01 | jakepdev | ok |
| 08:44.04 | jakepdev | saw the error |
| 08:44.16 | user_suse9 | no erros, but asterisk dosen't start! |
| 08:44.33 | user_suse9 | Sorry Im a linux noob, I make my first steps |
| 08:45.21 | jakepdev | suse9 - just dd a search on that error - came up with: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg26094.html |
| 08:45.23 | Skaag | Zeeek: I don't even know how to do it |
| 08:45.34 | Skaag | Zeeek: I try to dial into that line but the state of the channel never changes |
| 08:45.39 | Skaag | it always remains: Actual Hookstate: Onhook |
| 08:45.47 | Skaag | (it doesn't pick up) |
| 08:46.34 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 08:46.38 | Zeeek | how do you "try to dial" - put it in http://pastebin.ca |
| 08:47.02 | Zeeek | although as JerJer said, prolly better to learn to set up asterisk without @hole |
| 08:47.05 | Skaag | from a regular phone, to the number of this line |
| 08:47.18 | Skaag | actually from my cellphone to that line number |
| 08:47.29 | Zeeek | what output on CLI screen? |
| 08:47.40 | Skaag | there's no output on the CLI when I dial the number |
| 08:47.46 | *** join/#asterisk rlg (~umairbari@202.142.189.86) |
| 08:48.10 | Zeeek | sounds like the "modem" isn't compatible or you are in a country that doesn't work |
| 08:48.17 | Zeeek | with zaptel |
| 08:48.20 | Zeeek | stuff |
| 08:48.24 | Skaag | :-( |
| 08:48.37 | Skaag | I specifically ordered this modem as an Asterisk compatible modem |
| 08:48.43 | Skaag | 4 of them... |
| 08:48.54 | jakepdev | didn't even know they made those anymore |
| 08:48.57 | Zeeek | 4 modems? |
| 08:49.07 | Skaag | 4 cards |
| 08:49.09 | Skaag | yes |
| 08:49.17 | jakepdev | shoulda got the TDM400P |
| 08:49.17 | Zeeek | for one box? |
| 08:49.25 | Skaag | yes in one box |
| 08:49.33 | Skaag | they were all identified |
| 08:50.12 | Zeeek | usually you shouldn't have more than 2 cards in one box |
| 08:50.20 | Zeeek | also look at the IRQ situation |
| 08:50.26 | Zeeek | cat /proc/interrupts |
| 08:50.31 | Skaag | http://pastebin.ca/9885 |
| 08:51.02 | Skaag | they received IRQs 100, 10, 3 and 9 |
| 08:51.07 | Zeeek | my advice: pull out three of them and start over |
| 08:51.08 | Skaag | 11, 10, 3, 9 |
| 08:51.15 | Skaag | ok will do that... brb |
| 08:51.23 | Zeeek | they are all shar"ed except the first one |
| 08:51.34 | Zeeek | you don't want shared IRQ with zaptel hw |
| 08:52.05 | Zeeek | looks like the vendor said "'compatible", not Digium |
| 08:52.46 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 08:52.56 | Skaag | yah |
| 08:53.05 | Skaag | bastard.. |
| 08:53.07 | Zeeek | they may have been lyiong |
| 08:53.20 | Zeeek | but first, pull three of them out, |
| 08:53.31 | Skaag | did that, rebooting right now |
| 08:53.36 | Zeeek | and make sure theone is on its own IRQ |
| 08:53.45 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 08:54.36 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 08:54.37 | *** join/#asterisk Cardoe (~chatzilla@Cardoe.developer.gentoo) |
| 08:54.47 | Cardoe | Anyne ever try livevoip.com or voipjet.com ? |
| 08:54.54 | *** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 08:55.02 | jakepdev | i tried voipjet |
| 08:55.06 | Zeeek | see the mailing lis for extensive comments on both |
| 08:55.20 | Zeeek | jakepdev didn't like it? |
| 08:55.22 | Skaag | http://pastebin.ca/9886 |
| 08:55.49 | Skaag | holy shit, it works |
| 08:55.55 | Skaag | but it just said goodbye |
| 08:55.57 | Skaag | how rude |
| 08:55.58 | Skaag | :-) |
| 08:56.16 | jakepdev | didn't work for me... I'm on iaxtel for now - we'll see how that works out |
| 08:56.16 | Zeeek | Skaag ok. here is what I have: Found a Wildcard FXO: Wildcard X101P |
| 08:56.29 | Cardoe | jakepdev: iaxtel didn't work for me |
| 08:56.43 | Zeeek | I've had good results with voipjet |
| 08:56.46 | jakepdev | should revise that - it did work, but audio was choppy |
| 08:56.50 | Zeeek | but I don't call daily |
| 08:56.57 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 08:56.57 | Zeeek | never had that |
| 08:57.06 | Zeeek | voipjet has a FREE test |
| 08:57.08 | jakepdev | it all depends on the network path |
| 08:57.11 | Cardoe | voipjet looks pretty simple. |
| 08:57.14 | Skaag | ok so the problem was a hardware problem here |
| 08:57.17 | jakepdev | they're all simple |
| 08:57.20 | Skaag | cuz I stuck 4 cards in one box |
| 08:57.24 | Zeeek | jakepdev I'm far from all providers except voiptalk.org |
| 08:57.32 | Skaag | is there a solution for that, in general? |
| 08:57.37 | Zeeek | TDM400 |
| 08:57.42 | Zeeek | with 4 fxo modules |
| 08:57.43 | jakepdev | i hehe |
| 08:57.47 | Zeeek | but it isn't cheap |
| 08:57.50 | Skaag | it's one card? |
| 08:57.52 | Skaag | how much? |
| 08:57.58 | Zeeek | ya with 4 daughters |
| 08:58.05 | Zeeek | go look at digium.com |
| 08:58.07 | Skaag | ok |
| 08:58.12 | Zeeek | $380 I think |
| 08:58.23 | Zeeek | rather than the $40 you spent |
| 08:59.56 | jakepdev | i found it for under $300 |
| 09:00.07 | Zeeek | with 4 FXO? That's good |
| 09:00.08 | jakepdev | do a search on froogle |
| 09:00.12 | jakepdev | yep |
| 09:00.28 | jakepdev | oops |
| 09:00.32 | jakepdev | 4 FXS |
| 09:00.34 | jakepdev | ok |
| 09:00.41 | Zeeek | FXO are $10 more IIRC |
| 09:01.13 | jakepdev | http://www.verilan.com/store/customer/product.php?productid=16217 |
| 09:02.01 | Zeeek | where are these people? |
| 09:02.53 | jakepdev | Portland, OR |
| 09:03.08 | Zeeek | I se that now |
| 09:03.15 | Zeeek | that looks good |
| 09:03.18 | Skaag | too bad there's no way to fix the simpler cards |
| 09:03.23 | Skaag | cost is a serious issue here |
| 09:04.09 | user_suse9 | jakepdev, thanks for the link, now I have modified modules.conf and now I get a message that the capi dosen't installed. I use a AVM C2 ISDN controller. I think it is installed, but Im not really sure. |
| 09:04.25 | Skaag | so how do I grab an outside line now with x-lite? |
| 09:04.34 | Skaag | I tried 9xxxxxxxxx but it says all extensions are busy |
| 09:04.40 | jakepdev | don't think you're ready yet |
| 09:04.42 | Zeeek | Starter tutorial: |
| 09:04.42 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 09:04.43 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 09:04.43 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 09:04.43 | Zeeek | THE reference of the moment: |
| 09:04.43 | Zeeek | http://www.asteriskdocs.org |
| 09:05.16 | jakepdev | you have to get capi working before you can even think about making a SIP call |
| 09:05.19 | *** join/#asterisk psiderov (~pi@83.228.0.212) |
| 09:05.21 | jakepdev | over CAPI |
| 09:05.24 | Skaag | <PROTECTED> |
| 09:05.51 | jakepdev | you don't need to start with macros |
| 09:05.53 | psiderov | hi guys, is there option in asterisk for activating / deactivating VAD ? |
| 09:06.07 | jakepdev | start with the Dial cmd |
| 09:06.34 | jakepdev | but - if you want to call out your CAPI, you need to get it installed |
| 09:06.51 | Skaag | I thought it was already installed |
| 09:07.07 | jakepdev | you just said "now I get a message that the capi dosen't installed" |
| 09:07.25 | Skaag | no that's not me |
| 09:07.34 | Skaag | i said I get a message that all circuits are busy |
| 09:07.47 | Zeeek | Skaag take the time to read a few docs |
| 09:07.57 | Skaag | I did but it's a big mess |
| 09:08.03 | jakepdev | blur vision |
| 09:08.07 | Skaag | it's because asterisk is so flexible |
| 09:08.19 | user_suse9 | NOTICE[1076838528]: chan_capi.c:2636 load_module: CAPI not installed! |
| 09:08.20 | jakepdev | skagg - start simple |
| 09:08.20 | Skaag | every guide walks you through some kind of scenario (which is not mine) |
| 09:08.20 | Zeeek | look at the links I posted above |
| 09:08.32 | Skaag | Zeeek: i'm looking through them one by one |
| 09:08.36 | Zeeek | if you don't know how to use dial you haven't read much |
| 09:08.37 | jakepdev | suse - http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI |
| 09:09.06 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 09:09.24 | *** join/#asterisk Jas_Williams (~Jason@host217-44-216-142.range217-44.btcentralplus.com) |
| 09:10.14 | psiderov | hi guys, is there option in asterisk for activating / deactivating VAD ? |
| 09:10.27 | jakepdev | skaag - i found it was easier to rename extensions.conf to something else and start simple |
| 09:10.29 | psycodad | what do I need to do to receive 2 or more concurrent incoming sip calls from different local sip users to a snom190 ? Dial jumps always to 101+X when calling from the 2nd sip phone while having already a conversation from the snom to another local sip phone... (should both parties have reinvite on ev. ?) |
| 09:10.42 | jakepdev | ~VAD |
| 09:10.43 | jbot | somebody said vad was voice activated delivery |
| 09:10.58 | jakepdev | who squashed the bugbot? |
| 09:11.13 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 09:12.24 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 09:12.46 | psiderov | silence suppression |
| 09:13.07 | jakepdev | psyco - sounds like a dial err if it's jumping to 101+X. any errors you can see with sip debug? |
| 09:13.20 | psiderov | I have a problem w/ ATA186 |
| 09:13.30 | Skaag | jakepdev: I edited the [default] section of the extensions.conf file... |
| 09:14.34 | jakepdev | skaag - @home has too much stuff loaded in the dialplan (extensions.conf). it's better to start simple |
| 09:14.52 | jakepdev | mv extensions.conf extensions.conf.amp |
| 09:15.05 | jakepdev | nano extensions.conf |
| 09:15.34 | jakepdev | skaag - really gotta learn that dialplan though |
| 09:15.44 | Skaag | yah... :-( |
| 09:15.47 | Skaag | how complex |
| 09:16.11 | jakepdev | it's really not that bad once you see how it works... |
| 09:17.47 | jakepdev | skaag - at least read this: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 09:17.59 | jakepdev | that's the bare minimum |
| 09:18.03 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 09:18.40 | Mimmus | hi, where can I buy Digium PRI cards in Europe (Italy)? |
| 09:19.32 | Skaag | ok thanks |
| 09:19.47 | psycodad | jakepdev: I see only twice an OPTIONS to the SNOM without reply and then a cancel...however i can make the same calls concurrently when initiated from the snom |
| 09:20.41 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 09:20.42 | *** join/#asterisk Riddles (~mark@mail.wormgoor.com) |
| 09:21.13 | jakepdev | usually you'll see reject or something like that... pastebin your sip debug test calls... maybe I can see something you might have missed |
| 09:21.56 | psycodad | apropos: is there a way to catch debugging output outside the CLI console ? I use screen and have limited scrollback... |
| 09:22.11 | Zeeek | logging.conf |
| 09:22.15 | Riddles | It's in the FAQ |
| 09:22.27 | jakepdev | or an SSH client |
| 09:22.35 | jakepdev | with logging capibilities |
| 09:23.23 | psycodad | thnx, sorry for the question...already added debug to messages => |
| 09:23.45 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 09:23.50 | clive- | does anyone know why thiserror message appears: Unable to create channel of type 'SIP' |
| 09:24.01 | jakepdev | psiderov - nobody will be able to help you with your ATA186 problem if you don't say what the problem is |
| 09:24.24 | jakepdev | clive - the obvoius answer would be... |
| 09:24.36 | psycodad | clive-: you're prolly dialing a SIP without /XXX ;-) |
| 09:26.07 | jakepdev | ithe sun is almost out here |
| 09:26.09 | clive- | psycodad..thanks, my dial string looks like::_9X.,1,Dial(SIP/9991234@9991234,100) |
| 09:27.21 | *** join/#asterisk christo (~chris@office.enovi.com) |
| 09:27.25 | christo | aye |
| 09:28.26 | jakepdev | clive - what is 9991234? |
| 09:28.33 | Riddles | Does anyone here know about DTMF Callerid? Especially on FXS lines (it works on my FXO inpu). |
| 09:28.35 | Riddles | +t |
| 09:28.55 | jakepdev | riddles - what are you trying to do? |
| 09:29.07 | jakepdev | what's not working? |
| 09:29.08 | JerJer | Dial,SIP/peer |
| 09:29.10 | Cardoe | What do you guys think? Sipura SPA-1001 or just getting the phone? |
| 09:29.23 | Riddles | I am getting a DTMF callerid on my phoneline and it shows up on the logs. |
| 09:29.35 | Riddles | The callerid does show up in a connected SIP phone. |
| 09:30.02 | Riddles | But it seems like asterisk doesn't send it out on an FXS lines and my analog phones don'tshow the callerid. |
| 09:30.03 | jakepdev | riddles - you can try SetCallerID before you Dial |
| 09:30.19 | jakepdev | oh |
| 09:30.29 | jakepdev | hmm |
| 09:30.52 | jakepdev | *does* show up for SIP |
| 09:30.59 | Riddles | Even when in debug log, it seems like Asterisk never even tries to send it. |
| 09:31.49 | jakepdev | riddles - what card? |
| 09:32.11 | Riddles | TDM400P |
| 09:32.34 | Riddles | 31b really |
| 09:33.29 | jakepdev | says it supports callerid |
| 09:33.58 | jakepdev | is it enabled in zapata.conf? |
| 09:34.09 | Riddles | Yes, incoming... but does it support sending callerid on fxs? |
| 09:34.33 | fenlander | Riddles: which country are you in? |
| 09:34.38 | Riddles | NL |
| 09:35.11 | Riddles | This is in zapata.conf (extract) |
| 09:35.12 | Riddles | usecallerid=yes |
| 09:35.12 | Riddles | cidsignalling=dtmf |
| 09:35.12 | Riddles | cidstart=polarity |
| 09:35.12 | Riddles | hidecallerid=no |
| 09:35.23 | Riddles | It's in both the fxs and the fxo section |
| 09:36.07 | jakepdev | useincomingcalleridonzaptransfer |
| 09:36.28 | jakepdev | callerid=asreceived |
| 09:37.10 | Riddles | callerid=asreceived is in the fxo section (fxs signalling) |
| 09:37.12 | Riddles | which works |
| 09:37.22 | Riddles | I have callerid lines for all channels in the fxs section |
| 09:37.35 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-010.man.nc.charter.com) |
| 09:37.41 | Riddles | I thought useincomingcalleridonzaptransfer was for transfering calls only |
| 09:37.49 | jakepdev | don't know |
| 09:38.38 | jakepdev | when it syas it's dialing zap, can you see if the callerid is being sent? |
| 09:39.06 | *** join/#asterisk Newbie___ (me@218.111.8.142) |
| 09:39.45 | Newbie___ | hi, does anyone has any experience in setting up inphonex.com with * ? |
| 09:40.30 | jakepdev | i'm out guys - birds are chirping |
| 09:40.34 | jakepdev | good night |
| 09:40.53 | Makenshi | anyone working on dkms support for zaptel/libpri? |
| 09:40.54 | Riddles | night |
| 09:44.42 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 09:45.02 | Skaag | ok, so I managed to get an incoming call routed to my voicemail |
| 09:45.08 | Skaag | hurrey |
| 09:45.57 | *** join/#asterisk lbow (~steve@et0.cr2.ctn.nha.co.za) |
| 09:47.33 | Skaag | i have a feeling it does not detect hangups fast enough |
| 09:52.22 | psycodad | anybody using snom phones ? |
| 09:55.10 | *** join/#asterisk mozzi (~stefaans@darkstar.posix.co.za) |
| 09:55.58 | mozzi | hallo all |
| 09:55.59 | langals | Hi there...I have got a really strange issue and my problem is not that it is not working, but why it is working... |
| 09:56.14 | langals | I have Asterisk set up on a public IP, but the clients are behind a Port Restricted NAT with no support for UPnP... |
| 09:56.14 | *** part/#asterisk jollyr (jollyr@210.16.20.27) |
| 09:56.20 | langals | when I don't specify nat=yes in the sip.conf file, then it works...but not sure why it works because I cannot find any reference to the IP of the NAt in the SIP messages... |
| 09:56.26 | langals | My problem is that I don't know if it will work on other networks..... |
| 09:57.42 | mozzi | anyone here using Flash Operator Panel ? |
| 09:57.51 | Zeeek | ya |
| 09:58.10 | Newbie___ | hi, does anyone has any experience in setting up inphonex.com with * ? |
| 09:58.15 | mozzi | does it work well? |
| 09:58.24 | Zeeek | I like it |
| 09:58.39 | Zeeek | I even donated a few bucks |
| 09:58.40 | mozzi | are there other options? |
| 09:58.50 | Zeeek | a few |
| 09:58.54 | Zeeek | one with .NET |
| 09:59.08 | mozzi | I'll skip :P |
| 09:59.24 | Zeeek | yeah that's what I thought. Esp since it needs beta 2. |
| 09:59.30 | Zeeek | wich I don't have |
| 09:59.35 | *** join/#asterisk sleepy_one (~chatzilla@cpe-24-166-32-45.neo.res.rr.com) |
| 09:59.42 | sleepy_one | hello everyone |
| 09:59.54 | mozzi | I am looking to set it up for someone else for the first time |
| 10:00.09 | mozzi | so I need to make them interact with it easier |
| 10:00.15 | sleepy_one | I just upgraded to * 1.0.7 and DTMF on my cisco 7960Gs using SIP stopped working :'( any idea what's going on? |
| 10:00.34 | mozzi | I saw Asterisk@Home |
| 10:00.55 | mozzi | but I will need to put other services on there as well |
| 10:01.06 | sleepy_one | dtmfmode=inband like it's always been, my snoms with dtmfmode=rfc2833 are still working |
| 10:01.39 | sleepy_one | so are my IAXys |
| 10:02.18 | Zeeek | mozzi if you want to understand asterisk, avoid @hole and other interfaces that just muddy the water |
| 10:02.54 | mozzi | I have a good understanding of it(kinda0 |
| 10:02.57 | mozzi | ) |
| 10:03.12 | mozzi | just need to make it easy for someone else |
| 10:03.17 | fenlander | sleep_one: there are some dtmf negotiation issues in 1.0.6 & 1.0.7. I have seen a phone try to send rfc2833 when * was expecting inband. |
| 10:04.30 | fenlander | sleepy_one: the 7960G should work fine with rfc2833 |
| 10:04.41 | sleepy_one | I just switched and it DID |
| 10:04.47 | sleepy_one | thanks :) |
| 10:07.16 | *** join/#asterisk lbow (~steve@wbs-146-129-17.telkomadsl.co.za) |
| 10:07.24 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 10:08.31 | soundguy | =is chan_modem.so needed for asterisk to run? if just using it with SIP and IAX? |
| 10:09.27 | sleepy_one | I could be wrong but I don't think so |
| 10:09.59 | soundguy | asterisk is chucking a spaz because it cant find it on a new install in slackware |
| 10:10.05 | darkskiez | soundguy, u may have to take out the bri support too if it complains. |
| 10:10.22 | darkskiez | soundguy, chan_modem_isdn etc |
| 10:10.41 | soundguy | it has now complained about chan_modem_bestdata.so chan_modem_aopen.so |
| 10:10.46 | darkskiez | yup |
| 10:10.47 | soundguy | does it take wildcards? |
| 10:10.50 | darkskiez | take them out |
| 10:11.02 | soundguy | so I could add like: noload => chan_modem*.so ? |
| 10:11.02 | darkskiez | dunno about the wildcards. |
| 10:11.06 | darkskiez | its only 3 lines |
| 10:11.16 | soundguy | thers more |
| 10:11.20 | soundguy | :) |
| 10:11.49 | sleepy_one | comment them out with ; |
| 10:12.01 | soundguy | they arnt in modules.conf, so I have to add the noload part |
| 10:12.05 | sleepy_one | or noload them |
| 10:13.03 | sleepy_one | it doesn't take wildcards that I know of |
| 10:13.03 | soundguy | this didnt do that to me on a gentoo install :( |
| 10:13.26 | sleepy_one | what distro r u using? |
| 10:13.30 | soundguy | slackware |
| 10:13.35 | sleepy_one | dunno |
| 10:13.52 | soundguy | i just did a make install of asterisk |
| 10:13.56 | sleepy_one | * works beautifully in RH x.x FC1 FC2 and FC3 |
| 10:14.06 | soundguy | it works nicely in gentoo too |
| 10:14.20 | sleepy_one | it was a pain on debian but it worked on it too |
| 10:14.31 | sleepy_one | haven't tried slackware so I dunno |
| 10:14.35 | soundguy | yeah, cant say I like Debian |
| 10:14.41 | Makenshi | i suppose you have to recompile zaptel and libpri every time you upgrade your kernel? |
| 10:14.55 | sleepy_one | yup |
| 10:15.12 | Makenshi | they ought to support dkms |
| 10:15.25 | Makenshi | then you wouldn't have to |
| 10:21.33 | darkskiez | whats dkms do when its at home |
| 10:21.49 | darkskiez | oh, i see |
| 10:21.50 | sleepy_one | good nite / good morning all, cya l8r |
| 10:21.51 | darkskiez | dell thing |
| 10:23.31 | Makenshi | when you change your kernel, it automatically compiles modules if necessary |
| 10:23.38 | Makenshi | very handy |
| 10:25.35 | darkskiez | make-kpkg modules ? |
| 10:28.28 | *** join/#asterisk mithro (~tim@150.203.247.9) |
| 10:30.58 | *** join/#asterisk roamer323 (~sing@HSE-MTL-ppp64962.qc.sympatico.ca) |
| 10:32.20 | darwin35 | good morning |
| 10:34.01 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 10:36.54 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 10:37.01 | ManxPower | Is it safe to come back? |
| 10:39.11 | darwin35 | never |
| 10:48.16 | christo | I've beent trying all morning to set up an IAX2 connection between 2 servers.. Here's my latest attempt http://pastebin.ca/9893 where am I going wrong please? |
| 10:50.20 | ManxPower | darwin35: The most frustrating thing is that the newbies don't even have any idea of what to say when reporting a problem |
| 10:50.46 | *** join/#asterisk facek_ (faceoff@devel.acdbddh.eu.org) |
| 10:50.49 | facek_ | anyone? |
| 10:51.26 | ManxPower | christo: First make sure the username=blah is the same as [blah] |
| 10:51.40 | ManxPower | i.e. use [jones] and username=jones or whatever |
| 10:51.40 | facek_ | can I use that in zapata.conf |
| 10:51.41 | facek_ | channel => 1-2 |
| 10:51.41 | facek_ | channel => 4-5 |
| 10:51.56 | facek_ | if i have two carda.. and i want that same settings for that cards |
| 10:52.08 | christo | ManxPower - so the username has to be the same as the context header thing? |
| 10:52.21 | ManxPower | christo: Also don't use permit/deny at this point. You can add it latter. |
| 10:52.37 | christo | ok |
| 10:52.44 | ManxPower | christo: Not really, but it makes things MUCH easier to get working |
| 10:53.40 | ManxPower | Also don't specify a Context: in your .call file. |
| 10:54.57 | facek_ | ManxPower Can you help me. My asterisk doesn't answer calls |
| 10:55.00 | Skaag | So I understand that basically Asterisk@Home = Shit, and I should install the real thing, instead? |
| 10:55.11 | ManxPower | well, try not specifying a Contect: in your call file. |
| 10:55.45 | ManxPower | Skaag: Asterisk@Home seems to do a decent job of being a mostly preconfigured system for very simple home use. |
| 10:56.05 | ManxPower | If you have needs that are more than "very simple home use" then don't use Asterisk@Home |
| 10:57.03 | *** join/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz) |
| 10:57.26 | pdracevich | I need help with IAX relatime, I have got working SIP relatime. |
| 10:57.34 | pdracevich | I have looked everywhere |
| 10:58.59 | ManxPower | Personally I have no interest in using a build of Asterisk that thinks it wants to be Windows. |
| 10:59.15 | pdracevich | Same with me |
| 10:59.31 | christo | ManxPower: I have checked all of the above, and updated the paste: http://pastebin.ca/9895. I'm still not able to get authority for the connection |
| 10:59.42 | christo | I've not managed once to get the IAX2 connection accepted |
| 10:59.51 | ManxPower | christo: This is the error you care about: request '442077519062@iax-incoming' does not exist |
| 11:00.08 | ManxPower | that about says it all |
| 11:00.32 | christo | the request does not exist? |
| 11:00.37 | *** join/#asterisk Mother_ (~m@53.Red-217-126-93.pooles.rima-tde.net) |
| 11:00.49 | Mother_ | greetings |
| 11:02.06 | ManxPower | You do not have a matching exten => line in the iax-incoming context on the remote system |
| 11:03.00 | christo | I'm looking at that now |
| 11:03.09 | Mother_ | ManxPower is no longer grumpy_ |
| 11:03.10 | Mother_ | ? |
| 11:03.34 | Skaag | ManxPower: but it doesn't seem to be really properly configured out of the box as it says it is |
| 11:03.40 | ManxPower | Mother_: Don't push your lucj. |
| 11:04.01 | Mother_ | hehehe |
| 11:04.06 | tzanger | manxpower's always grumpy :-) |
| 11:04.29 | ManxPower | tzanger: I'll be MUCH more grumpy friday morning. |
| 11:04.32 | Mother_ | yeah but yesterday he was *really* grumpy, judging by his nick :D |
| 11:04.39 | Mother_ | why's that? |
| 11:04.44 | ManxPower | That's when the 60 phone Asterisk system goes into production |
| 11:04.49 | Mother_ | argh |
| 11:04.52 | Mother_ | good luck |
| 11:05.20 | tzanger | heh |
| 11:05.24 | tzanger | it'll go well I am sure, ManxPower |
| 11:05.30 | ManxPower | Mother_: Oh, I'm not worried about the technical parts of the install. I'm worried about the evil SOBs that are going to be using it. |
| 11:05.58 | ManxPower | "I don't have time for you to install my new phone today" <-- actual words from a user. |
| 11:06.06 | Mother_ | ManxPower: indeed, that's usually the worst problem - I've had all sorts of niceties on this install |
| 11:06.22 | Mother_ | s/user/(L)user |
| 11:06.41 | ManxPower | It doesn't help that 1/2 the people don't even have a computer in their office. |
| 11:06.52 | ManxPower | Another 1/3 don't even have a phone in their office. |
| 11:06.57 | Mother_ | one complained his phone wasn't working, when I look at it, he'd unplugged the ethernet cable, "because the ringing noise was giving him a headache" |
| 11:07.12 | tzanger | Mother_: HAAAAAAAAAAAAAHAHAHAHHAHAHAHA |
| 11:07.42 | ManxPower | Mother_: Do you sometimes fantasize about being able to say "You're fired!" to such people? |
| 11:07.46 | Mother_ | so I showed him that the ringing volume *can* be lowered on the 7012s, even by someone so technically incompetent as him |
| 11:07.58 | Mother_ | ManxPower: I dream of that almost every night! |
| 11:08.54 | Skaag | ManxPower: we have no phones in our office |
| 11:09.00 | Skaag | ManxPower: just computers :-) |
| 11:09.18 | tzanger | Skaag: sounds like heaven |
| 11:09.40 | Skaag | yes some of my staff does beleive whole heartedly that they have reached heaven and died |
| 11:09.51 | Skaag | except they wake up around 17:30 and leave the office... |
| 11:10.03 | Skaag | (I tried chains but they manage to break the combination on the locks) |
| 11:10.15 | tzanger | Skaag: hah |
| 11:10.24 | Skaag | damn websites that explain how to break locks... |
| 11:10.54 | Skaag | so now i'm trying to make their lives miserable by actually giving them phones, even if it's just x-lite with a headset |
| 11:11.10 | ManxPower | I'm pretty convinced that when a cockroach dies it's soul is reincarnated as a Real Estate Agent. |
| 11:11.44 | Skaag | that's why I never kill them... I actually beleived the same but with lawyers instead of real estate agents |
| 11:12.04 | Mother_ | LOL |
| 11:12.29 | Wonka | poor cockroaches - most of them don't deserve _that_ |
| 11:12.48 | Mother_ | it would be a full cycle then |
| 11:13.22 | ManxPower | Commercial Agents are not so bad. It's the residential agents that are evil. |
| 11:13.25 | Wonka | i think, they'd reincarnate as chicken in a egg factory, or some other very lousy life... |
| 11:13.40 | Wonka | cockroaches is not miserable enough |
| 11:14.10 | Skaag | ManxPower: I agree with you 100% |
| 11:14.42 | Skaag | by the way what's the absolute cheapest phone that works with *? |
| 11:14.56 | ManxPower | Skaag: $9 phone from walmart |
| 11:15.09 | Skaag | really?! |
| 11:15.18 | Skaag | you mean with an ATA adapter...? |
| 11:15.22 | christo | ManxPower - okay. things are looking up. I added exten => _XXX.,1,Dial(zap/g2/${EXTEN:1}) .. still not quite there tho. I'll update the paste ;) |
| 11:15.27 | ManxPower | or with a Digium card |
| 11:17.58 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 11:20.18 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 11:25.34 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 11:30.48 | Mother_ | so, any suggestion as to a cheap/problem-free card to start playing with ISDN? |
| 11:31.12 | Mother_ | plus, from what I read, zaphfc is better than CAPI? |
| 11:31.49 | ManxPower | Um, there is no cheap/problem-free card for ISDN BRI |
| 11:32.26 | Mother_ | heh I knew you'd say that |
| 11:32.39 | Mother_ | s/problem-free/least-problem-prone |
| 11:33.02 | Mother_ | I can get a Fritz locally for some 60 euros |
| 11:33.18 | Jas_Williams | Mother_: Europe or USA |
| 11:35.18 | *** join/#asterisk bzzz (bzzz@84.217.0.126) |
| 11:35.23 | Zeeek | anyone know anything about FTTH ? |
| 11:35.45 | Jas_Williams | Nah What does it mean ? |
| 11:35.56 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 11:37.02 | ManxPower | Fiber To The Home |
| 11:37.06 | Zeeek | Fiber to the Home! 100M/100M Internet access |
| 11:37.19 | Zeeek | for $85/mo |
| 11:37.30 | ManxPower | FTTH is what the Bells dangle in front of regulators to get what they want. |
| 11:37.32 | Zeeek | they're starting tests now here |
| 11:38.16 | soundguy | nie, if only they were going to have FTTH in Australia |
| 11:38.16 | Zeeek | I want FTTH now |
| 11:38.16 | ManxPower | FTTH requires putting new cables into the ground. It's not going to be widely available for decades. |
| 11:38.32 | Zeeek | widely, shmidely, as long azs I have it :) |
| 11:38.50 | Zeeek | in big cities they pass it through the swers |
| 11:38.59 | Zeeek | sewers... so no trench digging |
| 11:39.19 | langals | ManPower - Hi there...I spoke to you yesterday about NAT issues |
| 11:39.40 | ManxPower | Zeeek: I KNEW I should have patented that idea 12 years ago. |
| 11:39.54 | ManxPower | hello, langals |
| 11:40.05 | Zeeek | running fiber in sweres? |
| 11:40.25 | langals | ManPower - I am behind a port restricted nat, with no support for UPnP...there is also no support for NAT in my client.... |
| 11:40.27 | Zeeek | I got me second, no third IAX hardphone today |
| 11:40.29 | ManxPower | Zeeek: Yeah. |
| 11:40.34 | Zeeek | works well |
| 11:40.47 | Mother_ | sorry, phone rang - Jas_Williams: Europe |
| 11:40.52 | ManxPower | my specific idea was water pipes, actually. |
| 11:40.59 | langals | ManPower - now, if I do not specify nat=yes in Asterisk (which is on a public IP), then it works, no problem.... |
| 11:41.00 | ManxPower | In Jakarta |
| 11:41.05 | Zeeek | ah. |
| 11:41.12 | langals | ManPower - this is really strange because it should not work |
| 11:41.31 | Zeeek | still a few left: http://ipphone.eezeephone.com/index_files/x100.htm |
| 11:41.34 | langals | ManPower - there is also no reference to the IP of the NAT in my SIP messages |
| 11:42.08 | langals | ManPower - this is really bothering me because I am not sure if it will work over other NATs, and I cannot undertstand why it is working |
| 11:46.09 | *** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk) |
| 11:47.03 | langals | ManPower - when I put in nat=yes into sip.conf, then Asterisk adds the NAT IP and port to the sip message, but it times out when there is no rtp traffic - this may be because ping is disabled on the network.. |
| 11:47.27 | langals | ManPower - by the way, clients are dialling into meetme... |
| 11:48.05 | langals | ManPower, so the it seems to work better when I have NO nat support, even when clients are behind a nat...wierd |
| 11:48.37 | Zeeek | langlais I think it all depends on what ip the client is sending and the type of NAT |
| 11:48.57 | ManxPower | it depends on MANY things. |
| 11:49.09 | Mother_ | not all NATs are created equal |
| 11:49.13 | newl | soundguy: If you lived in a particular suburb in Bris you could be trialing FTTP. :) |
| 11:49.30 | Zeeek | the preceeding list was contractually non-exhaustive :) |
| 11:49.31 | langals | It is a Port Restricted nat |
| 11:49.32 | soundguy | P? |
| 11:49.49 | ManxPower | define "Port Restricted nat" |
| 11:50.11 | soundguy | fibre to the premesis? |
| 11:50.43 | Zeeek | a NAT that can only drink port |
| 11:51.02 | ManxPower | Zeeek: At least it has good taste. |
| 11:51.16 | Zeeek | like NAT, there is port and there port |
| 11:51.19 | langals | ManPower, Zeeek - hang on, I am looking for the def |
| 11:51.40 | Zeeek | as long as it isn't an anti-symetic NAT |
| 11:51.50 | ManxPower | I suspect "Port Restricted nat" is what we call "fucking braind dead nat" here on #asterisk. |
| 11:52.12 | Zeeek | actually, X-Lite seems to be able to tell you what the NAT is if you let it use their STUN |
| 11:52.32 | langals | ManPower, Zeeek - http://en.wikipedia.org/wiki/Port_restricted_cone_NAT |
| 11:52.34 | Zeeek | one of the areas where Skype shines of course - it doesn't care |
| 11:52.36 | darkskiez | thats part of the stun spec |
| 11:52.46 | Zeeek | to determine NAT type? |
| 11:52.49 | Mother_ | indeed |
| 11:52.52 | Zeeek | interesting |
| 11:53.35 | langals | All this NAT stuff is starting to confuse me |
| 11:53.51 | Zeeek | just think how much it confuses aasterisk |
| 11:53.56 | *** join/#asterisk amir2 (~amir@195.226.9.186) |
| 11:54.00 | Mother_ | so, langals, what the problem is that if the internal client does not 'open' a connection outbound, and keep it open, the mapping times out and thus the router stops forwarding inbound packets |
| 11:54.11 | Mother_ | can you not play with the router? |
| 11:54.20 | ManxPower | I think if it in simple terms 1) Network Address Translation w/ Port Address Translation 2) all other brain dead stupid NATs |
| 11:54.42 | Mother_ | sounds close enough |
| 11:54.52 | ManxPower | 1) is what most NAT routers do. |
| 11:55.17 | Mother_ | I remember the 3Com 812 router, it would forward ALL external ports to your IP once you started a TCP session or sent a UDP packet to an external IP |
| 11:55.23 | langals | Mother_ - that is what it seems to be - not sure if I can do anything with the router |
| 11:55.30 | Mother_ | i.e. visit a web site, that web server's IP has access to ALL your ports |
| 11:55.31 | ManxPower | Mother_: That's 2) |
| 11:55.41 | Mother_ | ManxPower: a big yes |
| 11:56.09 | Mother_ | and it's enabled like that by default, and the only way to turn it off is via CLI |
| 11:56.25 | Mother_ | oddly, 3Com insists in calling this "intelligent PAT" |
| 11:56.50 | ManxPower | what happens when a second PC behind the same NAT goes to the same web server? |
| 11:56.51 | langals | Mother_ - so is that an anti-symmetric nat? |
| 11:57.07 | Mother_ | ergh... |
| 11:57.22 | Mother_ | ManxPower: I never stopped to investigate, I simply turned it off |
| 11:58.08 | Mother_ | ManxPower: what I know is that the mapping timed out after 3 minutes |
| 11:59.42 | Mother_ | http://seclists.org/lists/vulndiscuss/2003/Apr/0000.html |
| 12:02.26 | psycodad | I am still looking for the reason why my snom190 yields back "Got SIP response 486 "Busy Here" back from 10.1.1.236" when trying to call from a second sip line |
| 12:02.31 | langals | Mother_, ManPower, do you guys know where the reference to the ip that rtp traffic should be sent to can be found in the sip / sdp packets? |
| 12:03.00 | Mother_ | have you tried sip debug? |
| 12:03.14 | Mother_ | in the debug you'll see the RTP ports clearly |
| 12:03.15 | ManxPower | langals: look on google. It's a well known issue with all SIP (or RTP) and NAT. |
| 12:03.16 | *** join/#asterisk kanzure (~bbishop@cpe-66-68-141-11.austin.res.rr.com) |
| 12:03.42 | Mother_ | in * you have rtp.conf to set the port range, which you must then map appropriately in the routers |
| 12:04.21 | ManxPower | Mother_: only if Asterisk be behind NAT. |
| 12:04.26 | langals | Mother_, ManPower - thanks for the help guys |
| 12:06.56 | Zeeek | asterisk behind NAT sends a Via header with ext ip and port |
| 12:07.55 | *** join/#asterisk genuix (~genuix@sobek.7g0.net) |
| 12:08.13 | langals | Zeeek - asterisk is not behind nat - my clients are |
| 12:08.51 | *** join/#asterisk kioko (~rkioko@196.200.25.253) |
| 12:12.06 | Zeeek | but the asterisk I refer to is a client to someone else |
| 12:12.50 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 12:14.15 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 12:14.40 | Mother_ | langals: then you shouldn't have so many problems, when it gets hairy is when both are behind their NATs |
| 12:15.22 | pif | using Festival totally crashes my * server (1.0.7) |
| 12:15.22 | Mother_ | ManxPower: yes, indeed, I assumed his * was also behind NAT |
| 12:15.22 | pif | anyone noticed? |
| 12:18.02 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 12:18.02 | pointer | anyone know when the digium support peoples show up for work? |
| 12:18.02 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:18.25 | jontow | ... |
| 12:18.32 | cjk | hi, anyoe here who can tell me if 533 mhz and 256 mb ram is enough for transcoding 16 channels from bri to ilbc ? |
| 12:18.34 | *** join/#asterisk MikeJ[Laptop] (~icechat5@mi.origenfinancial.com) |
| 12:18.34 | jontow | i thought they'd be EST since they're in Alabama, yes? |
| 12:18.42 | langals | Mother_, ManPower - This is my setup - asterisk on public IP, clients behind NATs, and clients dialling into Meetme conference...so the calls are always between clients behind NAT and meetme on public IP |
| 12:19.10 | pointer | Alabama is CT |
| 12:19.31 | jontow | ah.. hm.. wasn't aware of that |
| 12:19.32 | jontow | :o |
| 12:19.43 | pointer | jontow: http://www.timetemperature.com/tzus/alabama_time_zone.shtml |
| 12:19.49 | langals | Mother_, ManPower - so is it understandable then that it works without specify nat support (either in asterisk / clients) |
| 12:20.05 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 12:24.09 | ManxPower | Digium is in USA Eastern Time. Currently UTC-5 I think. |
| 12:24.28 | ManxPower | It's 8:25am Digium Time |
| 12:24.40 | Mother_ | langals: the clients should inform * of their NAT IP so that * doesn't forward RTP to an internal IP |
| 12:24.57 | Mother_ | langals: also, it's good to specify the localnet in sip.conf |
| 12:25.22 | bjohnson | cjk: unlikely .. since a 533 is likely a piii at best |
| 12:25.47 | bjohnson | also the ram is kinda low |
| 12:26.25 | Mother_ | i.e. externip=212.x.x.x and localnet=192.168.0.0/255.255.2550 |
| 12:26.36 | Mother_ | 255.0 in the netmask sorry |
| 12:27.21 | ManxPower | Mother_: that's only needed if ASTERISK is behind NAT. |
| 12:27.25 | ManxPower | That's not langals's case. |
| 12:27.47 | bjohnson | that gets anoying unless the remote sip devices stay in the same remote subnet and maintain a static ip |
| 12:28.02 | Weezey | ManxPower: EDT is -0400 |
| 12:28.11 | ManxPower | Weezey: I sit corrected |
| 12:28.15 | *** join/#asterisk gonzo- (~gonzo@portacare.portaone.com) |
| 12:28.25 | reph | I hear living in EDT causes cancer. |
| 12:28.33 | Weezey | EST is -0500 so in the winter you're right. |
| 12:28.37 | bjohnson | I think EST IS -5 |
| 12:28.46 | bjohnson | ahh |
| 12:28.46 | Mavvie | anybody here played with a Cisco 2821? |
| 12:28.47 | bjohnson | DST |
| 12:28.54 | reph | what exactly is the indianan time zone |
| 12:29.01 | reph | used to mess me up bad when I lived there =\ |
| 12:29.07 | Weezey | IND +3000 ? |
| 12:29.08 | langals | ManPower, Mother_ - so would you say that it won't work if I don't specify nat=yes in sip.conf, because I am not at the moment and it is working - the clients don't seem to be specifying their nat ip |
| 12:29.19 | ManxPower | Mavvie: How is that router different from the 2621? |
| 12:29.23 | reph | I wasl ike 5 min from michigan...and michigan changed times for Daylight savings... |
| 12:29.34 | reph | sometimes time was the same, sometimes an hour ahead |
| 12:29.41 | ManxPower | langals: many sip clients have their own methods for making NAT work. |
| 12:30.18 | cypromis | wrong |
| 12:30.29 | cypromis | many sip clients have their own way for making NAT not work |
| 12:30.42 | ManxPower | cypromis: LOL! |
| 12:30.55 | Mother_ | ManxPower: agreed |
| 12:31.42 | Mother_ | ManxPower: I keep falling back to the double NAT, which has been my obsession :) |
| 12:31.50 | Mother_ | bbl |
| 12:32.00 | Mavvie | ManxPower: the 2600 series are LAN/WAN routers. The 2800 series are Integrated Services Routers. (i.e. voip friendly) |
| 12:32.07 | Zeeek | when my connection works, double NAT works |
| 12:32.11 | cypromis | hmm |
| 12:32.16 | cypromis | Cisco and voip friendly ? |
| 12:32.17 | cypromis | nah |
| 12:32.18 | jontow | on the topic of Cisco equipment.. |
| 12:32.19 | langals | ManPower, Mother_ - I am using RTC Client API, which is what windows messenger is built on - it supports UPnP nats by default, but not non-UPnP nats |
| 12:32.24 | cypromis | cisco is only support agreement charge friendly |
| 12:32.29 | jontow | anyone done anything with * and cisco 3640 Voice Gateways ? |
| 12:33.25 | Zeeek | how does upnp affect NATs? Mine has always been disabled |
| 12:33.31 | Mavvie | right now it does do everything fine: accepting sip packets, setting up connections to the pabx. |
| 12:33.39 | Mavvie | all except setting up a proper RTP stream :-) |
| 12:33.54 | *** join/#asterisk theHub (~karl@69.177.93.36) |
| 12:33.57 | *** join/#asterisk daork (~daork@jade.daork.net) |
| 12:33.59 | jontow | hmm |
| 12:34.13 | daork | so, anyone here familiar with the Asterisk::AGI perl module? |
| 12:34.26 | langals | Zeeek - it allows applications to easily query the nat for address mapping, or something like that - microsoft is pushing it |
| 12:34.41 | jontow | i laugh.. you know.. they've chosen a bunch of outdated MGCP equipment at work and i just don't know what to think about it |
| 12:34.52 | jontow | except "i ain't supportin' that" :) |
| 12:36.00 | jontow | need some 4-16 port IAX2-based ATAs on the market.. damnit :) |
| 12:36.47 | ManxPower | it's called a TE110P and a channel bank. |
| 12:36.48 | clive- | jontow, dont we all |
| 12:37.04 | jontow | manx.. true, true ;) |
| 12:37.16 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 12:37.17 | clive- | Manx, thats like using a sledgehammer to crack a peanut |
| 12:37.37 | ManxPower | clive-: for 16 ports? It's a hell of a lot cheaper than a VOIP device to do the same thing |
| 12:37.49 | Mimmus | hi, can I receive faxes using a PRI board? |
| 12:37.54 | *** join/#asterisk easimon (~easimon@balu.kawo2.RWTH-Aachen.DE) |
| 12:37.58 | jontow | yeah, depending on factors.. it'd be about $1k yes? |
| 12:37.58 | ManxPower | Mimmus: maybe. |
| 12:38.04 | jontow | assuming a decent deal on the channel bank? |
| 12:38.16 | Mimmus | ManxPower: why 'maybe'? |
| 12:38.17 | ManxPower | jontow: maybe less |
| 12:38.39 | ManxPower | Mimmus: It may or may not work. You have to build external software, you may need to patch stuff. |
| 12:38.48 | ManxPower | Or you can just plug an analog fax machine into an analog port. |
| 12:38.57 | clive- | Manx, agreed, for 16 is cool, but not for 4 or 6..:) |
| 12:39.18 | Mimmus | ManxPower: I already tried with an ISDN BRI and it worked (patching capi_channel) |
| 12:39.42 | bjohnson | ManxPower: not a hell of a lot cheaper .. about the same as a bunch of 2 port ATAs |
| 12:39.45 | jontow | yeah i'd definitely be down with a 4, 6, or 8 port for $250-500 |
| 12:39.49 | *** join/#asterisk jtar (~zic@81-178-54-38.dsl.pipex.com) |
| 12:39.49 | daork | anyone have any idea when I do a VERBOSE to * from an AGI (using Asterisk::AGI) a subsequent GET DATA doesnt get any data back? |
| 12:39.55 | ManxPower | bjohnson: but much easier to manage |
| 12:40.00 | daork | is the VERBOSE expecting data from * for some reason? |
| 12:40.00 | bjohnson | yes |
| 12:40.28 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) |
| 12:40.31 | bjohnson | ManxPower: but using that argument would also apply to channel bank vs voip phones |
| 12:40.57 | ManxPower | bjohnson: Yes. Except that VoIP phones (usually) have a better user interface for features than analog does. |
| 12:41.08 | Mimmus | ManxPower: what external applications are involved in fax receiving/sending? |
| 12:41.24 | ManxPower | Mimmus: spandsp and txfax / rxfax |
| 12:41.38 | ManxPower | ~google site:lists.digium.com spandsp txfax rxfax |
| 12:41.41 | Mimmus | ManxPower: ok, I will look for them on teh wiki |
| 12:42.22 | bjohnson | jontow: you can get a used channel bank for as low as $200 I'm told .. personally I've only ever seen them as low as $500. And the T1 card is about $500 .. but then you have 24 chnnels ready to use |
| 12:43.45 | ManxPower | I got an Adtran TA 750 for $400 from eBay the other day. Included T-1 card and 24 FXS |
| 12:44.20 | pointer | ManxPower: nice |
| 12:44.38 | ManxPower | I think port 24 is bad, but still.... |
| 12:47.15 | jontow | neat |
| 12:47.27 | jontow | i wish i had the kinda cash to play with this stuff at home |
| 12:47.44 | ManxPower | Don't use YOUR money! Use someone else's money! |
| 12:47.53 | jontow | hahah |
| 12:48.09 | jontow | my money is the only money i have access to at home |
| 12:48.23 | jontow | at work.. hell.. i've managed to defintiely use their money to play with the toys |
| 12:48.29 | *** join/#asterisk stefanocarlini (~stefanoca@host149-108.pool8019.interbusiness.it) |
| 12:49.18 | stefanocarlini | hello kram |
| 12:49.22 | jontow | got 3 T100Ps, a PRI, a cisco 7960, a polycom IP600, couple grandstreams, IAXy, Sipura SPA-2000, etc |
| 12:49.39 | jontow | but it all be a learning experience.. for them and me |
| 12:49.47 | *** join/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 12:50.22 | nestAr | i like how some versions of asterisk, the cli is color... other versions.. not so much |
| 12:51.24 | stefanocarlini | hello I need some info about the asterisk's project |
| 12:51.32 | stefanocarlini | anynone can help me? |
| 12:51.41 | ManxPower | ~docs |
| 12:51.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:51.45 | ManxPower | There you go. |
| 12:52.04 | jabbzy | ~docs |
| 12:52.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 12:52.05 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
| 12:52.13 | stefanocarlini | Ok. I already read it. |
| 12:52.13 | ManxPower | ~doc |
| 12:52.14 | jbot | from memory, doc is The command is "~docs", moron! |
| 12:52.26 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 12:52.31 | Juggie | word |
| 12:52.32 | *** join/#asterisk AngelGabriel (~angel@host81-133-186-111.in-addr.btopenworld.com) |
| 12:52.35 | stefanocarlini | I'm a student of a Master on F/OSS |
| 12:52.55 | stefanocarlini | and I'm writing a doc about the asterisk dev project |
| 12:53.30 | AngelGabriel | Can asterisk be used with the Mitel Smart-1 phone switch?? |
| 12:53.32 | stefanocarlini | I need to know some info like how many developers work on the code |
| 12:53.42 | genuix | a few ;-)) |
| 12:53.49 | Juggie | anyone at von canada? |
| 12:54.05 | genuix | you better come to astricon to better know this |
| 12:54.10 | ManxPower | stefanocarlini: 4 or 5 people commonly comit changes to CVS. MANY MANY more people feed them code. |
| 12:54.46 | stefanocarlini | ok. so I can get many info accessing the CVS site |
| 12:55.22 | Mimmus | sorry for the FAQ but what is DID? peraphs in Italy another word is used... |
| 12:55.39 | *** part/#asterisk kanzure (~bbishop@cpe-66-68-141-11.austin.res.rr.com) |
| 12:55.39 | clive- | stefano, just visit bugs.digium.com and view whats going / gone on there |
| 12:55.52 | ManxPower | Mimmus: DID is the same as DDI. DID = USA term, DDI = German/EU term |
| 12:55.56 | *** join/#asterisk vinsci (~vinsci@dsl-sjkgw2jb1.dial.inet.fi) |
| 12:56.17 | newl | And elsewhere on the planet, we call then DN's :) |
| 12:56.33 | stefanocarlini | I'm going.... |
| 12:56.33 | ManxPower | newl: That's more of an ISDN term |
| 12:56.43 | newl | Whereas DID means something entirely different. |
| 12:56.57 | ManxPower | But is technically correct for most any telephone number. |
| 12:57.04 | Mimmus | ehm, ehm.... an example? |
| 12:57.04 | *** part/#asterisk pointer (pointer@aj.catt.com) |
| 12:57.16 | stefanocarlini | Mimmus: are you from Italy? |
| 12:57.25 | Mimmus | stefanocarlini: yes (si) |
| 12:57.36 | ManxPower | Mimmus: Company has 100 people with phone. They all have a direct phone number that does not go thru the front desk / recptionist. |
| 12:57.49 | stefanocarlini | Mimmus: where are you from? |
| 12:57.54 | newl | ManxPower: Primarily differs by the choice of the switch vendor as to the terminology I believe. |
| 12:58.06 | Mimmus | ManxPower: ok, peraphs I understand now |
| 12:58.19 | Mimmus | stefanocarlini: I'm from Cosenza, Calabria, SOuth of Italy |
| 12:58.30 | newl | i.e. Slight differences between AXE, S12, DMS100, ESS, etc. |
| 12:58.36 | stefanocarlini | ok, I'm from Mantova |
| 12:58.48 | Mimmus | stefanocarlini: weel, beautiful town! |
| 12:59.10 | stefanocarlini | yes. i never been to Cosenza |
| 12:59.44 | Mimmus | stefanocarlini: it's a small town, with a semi-decent quality of life |
| 13:00.30 | stefanocarlini | why are you working with asterisk? |
| 13:00.46 | *** join/#asterisk ckruetze_ (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 13:01.15 | Mimmus | stefanocarlini: company where I'm emplyed is interested to gradually replace classic, very expensive, traditional, analog Alcatel PBX |
| 13:01.46 | Mimmus | stefanocarlini: and you |
| 13:02.17 | *** join/#asterisk saabluvr (master@keeper.nc-ks.de) |
| 13:02.23 | *** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 13:02.30 | *** join/#asterisk tsetane (~tsetane@212.4.33.58) |
| 13:02.45 | stefanocarlini | I'm using asterisk beacuse I'm a ICT and F/OSS freelance consultant |
| 13:03.09 | *** join/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 13:03.53 | Mimmus | stefanocarlini: what is F/OSS? |
| 13:04.12 | bjohnson | ~DID |
| 13:04.15 | jbot | rumour has it, did is Direct Inward Dialing |
| 13:04.18 | stefanocarlini | F/OSS = Free and Open Source Source Software |
| 13:04.21 | ManxPower | ~ddi |
| 13:04.22 | jbot | hmm... ddi is Direct Dialling Inward, URL: http://www.wilco-telephony.co.uk/did.html |
| 13:04.28 | bjohnson | never heard of it |
| 13:04.39 | Mimmus | stefanocarlini: ah, ok, great! |
| 13:04.53 | Mimmus | jbot: in italian is 'selezione passante', I think |
| 13:04.54 | jbot | Mimmus: okay |
| 13:05.53 | stefanocarlini | I'leaving the chat... bye |
| 13:05.58 | *** part/#asterisk stefanocarlini (~stefanoca@host149-108.pool8019.interbusiness.it) |
| 13:06.12 | *** join/#asterisk phpboy (~sj@tpr-165-231-125.telkomadsl.co.za) |
| 13:06.14 | phpboy | hey guys |
| 13:06.28 | phpboy | does anyone have a nice how-to on how to setup dialing permissions? |
| 13:06.46 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 13:06.52 | ManxPower | http://www.wilco-telephony.co.uk/glossary.html |
| 13:07.13 | ManxPower | phpboy: no. |
| 13:07.17 | phpboy | :/ |
| 13:07.33 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 13:07.38 | phpboy | cause I wanna allow certain extentions not to beable to dial certain numbers |
| 13:07.39 | phpboy | etc |
| 13:07.41 | phpboy | etx |
| 13:07.43 | phpboy | ect |
| 13:08.01 | ManxPower | So don't put the pattern in the context. |
| 13:08.10 | Mimmus | phpboy: it can be achieved with the exoterical art of extensions.conf! |
| 13:08.48 | phpboy | and u put a certain user in more than one context? |
| 13:08.52 | ManxPower | no. |
| 13:09.06 | phpboy | hmmm, then I'm not too sure how I will achieve this |
| 13:09.26 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-142-202.buff.east.verizon.net) |
| 13:09.52 | ManxPower | I have three main contexts for phones. You set up seperate contexts for verious things. toll calls, local calls, premium calls. Then just include => those contexts into the context the phone is in. |
| 13:09.56 | bjohnson | it is controlled by device .. not user |
| 13:10.01 | Mimmus | phpboy: I'm not an * guru but dialing permissions depend on how to 'include' contexts in extensions.conf |
| 13:10.07 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 13:10.29 | phpboy | Mimmus: Makes sense what ur saying |
| 13:10.42 | phpboy | so |
| 13:10.45 | bjohnson | with softphones the distinction may be minimal .. but it is controlled by the devices configurations in sip.conf, iax.conf, etc .. they dump incoming calls into whatever context you specify |
| 13:10.46 | phpboy | what would make sense to me |
| 13:11.00 | phpboy | is making like 3 contexts in my extentions.conf |
| 13:11.08 | phpboy | 1 for internal calls |
| 13:11.15 | phpboy | 1 for international calls |
| 13:11.23 | phpboy | 1 for local calls |
| 13:11.39 | phpboy | then I put those in various groups |
| 13:11.39 | phpboy | no? |
| 13:11.45 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 13:11.46 | bjohnson | kind of |
| 13:11.48 | ManxPower | that's pretty much what I do |
| 13:11.54 | Mimmus | an incoming call will be put in a context, according to channel configuration (sip.conf or zaptel.conf, ...) |
| 13:12.16 | Aze` | I need help, scenario: SIP/201 xlite 172.16.0.10 -> * 172.16.0.30 <- Router Wan 80.105.162.40 -> Sip/202 Xlite with public ip (dialup) 62.11.x.x, why when SIP/201 call SIP/202 or reverse.. SIP/202 listen SIP/201 but SIP/201 cant listen SIP/202, why pls ? (RTP Port 10000-20000 and 5060 redirect on *) |
| 13:12.29 | Mimmus | then, Asterisk will look for a matching in that context only (and in the contexts included...) |
| 13:12.46 | Mimmus | no match = no dialing permission |
| 13:13.06 | bjohnson | phpboy: you'll likely find you want more contexts than that .. and then include the ones you want. Start editing and testing .. you'll see what I mean |
| 13:13.24 | phpboy | yeah, of coz |
| 13:13.26 | phpboy | I'll play around |
| 13:14.37 | bjohnson | that's what it takes |
| 13:15.20 | bjohnson | it difficult to visualize all the possible call routings when you are first starting with * |
| 13:16.21 | *** join/#asterisk LeoB (~chatzilla@wireless-236.media.mit.edu) |
| 13:16.37 | ManxPower | I want a nap |
| 13:17.00 | tzanger | whee I get to install a server at 151 Front |
| 13:18.28 | newl | chmod a+nap ManxPower && chown nobody.bother ManxPower :0 |
| 13:20.27 | ManxPower | newl: I leave for New Orleans in less than an hour. |
| 13:20.36 | *** join/#asterisk _Brian (brian@unix01.voicenet.com) |
| 13:21.02 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:21.11 | newl | that's right, you're traveling :) |
| 13:21.28 | ManxPower | nust for a few days for an Asterisk install |
| 13:21.31 | ManxPower | just, even |
| 13:21.43 | newl | sweet |
| 13:22.07 | *** join/#asterisk wiz8291 (~dang@kay.arcbox.com) |
| 13:22.14 | ManxPower | Not really. These people make Catbert look like a nice guy. |
| 13:22.21 | wiz8291 | hi guys, can anyone help me with an echo problem? |
| 13:22.23 | newl | haha |
| 13:22.24 | *** join/#asterisk jjcool (~jjcool@208.178.154.99) |
| 13:22.29 | ManxPower | They make Ratbert look like a member of Mensa |
| 13:23.12 | wiz8291 | ManxPower: seems my problem yesterday was a crap phone... plugged in one with its own power supply and it works... but i get some SERIOUS echo! |
| 13:23.33 | newl | hmm..all the iax connections I've got configured here are outbound only. I didn't think it suffered the NAT issues that plagued SIP. I need to forward the port don't I? |
| 13:25.49 | *** part/#asterisk saabluvr (master@keeper.nc-ks.de) |
| 13:26.15 | ManxPower | Today is Stoner Day |
| 13:26.25 | _Brian | yup :) |
| 13:26.26 | Moonwick | someone else is listening to NPR, eh? :) |
| 13:26.39 | AngelGabriel | I have a copy of asterisk @ home *i mean the actually distro, not home as in where I live* and I was wondering, if I installed it on a machine with no digium type hardware, can I still test it using a softphone? |
| 13:26.42 | _Brian | http://www.420.com |
| 13:26.55 | Moonwick | they're talking about it on the radio right now |
| 13:26.59 | ManxPower | Moonwick: I'm, uh, familiar with the term 420, but didn't think April 20 was considered special |
| 13:27.26 | jabbzy | yes, asterisk@home doesn't need digium h/w |
| 13:28.01 | ManxPower | It's also Hitler's birthday, BTW. |
| 13:28.05 | *** join/#asterisk lancey (Shady@213.137.58.124) |
| 13:28.07 | lancey | hi guys |
| 13:28.11 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 13:28.12 | jabbzy | can't be - he's dead |
| 13:28.23 | lancey | since an hour, i get a bunch of these: |
| 13:28.24 | lancey | <PROTECTED> |
| 13:28.25 | ManxPower | Maybe if he had smoked more pot..... |
| 13:28.33 | lancey | and no sound at all, what might be the reason?? |
| 13:28.55 | *** join/#asterisk iq (~iq@70-59-160-50.omah.qwest.net) |
| 13:29.03 | DrWho17 | heh, www.iaxprovider.net isn't very informative |
| 13:29.22 | DrWho17 | is there some comprehensive list of iax providers out there, besides what is on the wiki |
| 13:29.30 | lancey | anyone knows some reasons for this? |
| 13:30.09 | olivier_ | <lancey> have u check u're codecs ? |
| 13:30.27 | lancey | olivier_ i haven't changed anything in my configuration |
| 13:30.35 | lancey | it just started to give me these errors |
| 13:30.38 | lancey | since an hour |
| 13:30.49 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 13:31.06 | ManxPower | ~google site:lists.digium.com "Received mini frame before first full voice frame" |
| 13:31.12 | wiz8291 | grrrrrrrrrrrr |
| 13:31.23 | wiz8291 | this is annoying... everything sounds distorted |
| 13:31.32 | ManxPower | wiz8291: echo is one of the hardest things in telco to fix. |
| 13:31.42 | lancey | ManxPower thank you, i've went over this |
| 13:31.43 | wiz8291 | ja, i'd gathered that! lol |
| 13:32.01 | *** join/#asterisk tessier (~treed@210.245.38.70) |
| 13:32.29 | wiz8291 | is there a decent document about? |
| 13:32.38 | wiz8291 | i've tried all the echocancel, etc options |
| 13:32.42 | wiz8291 | they make no difference |
| 13:32.57 | ManxPower | echocancel=yes echotraining=yes|600|800 |
| 13:33.09 | ManxPower | make sure they are BEFORE the channel => line |
| 13:33.15 | ManxPower | make sure to RESTART Asterisk. |
| 13:33.20 | wiz8291 | yup |
| 13:33.26 | cypromis | make sure to pray |
| 13:33.29 | ManxPower | play with rxgain and txgain to help reduce echo. |
| 13:33.32 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:33.33 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:33.47 | ManxPower | Play with the zaptel header (zconfig.h?) and enable different echo cancelers |
| 13:33.56 | lancey | ManxPower, is it normal that i receive this message no matter what codecs i choose? |
| 13:34.00 | ManxPower | add an external T-1 echo canceler device. |
| 13:34.07 | ManxPower | lancey: I don't know. |
| 13:34.36 | lancey | the mailing lists don't know either :> |
| 13:34.38 | wiz8291 | i had to play with the gains to get DTMF working |
| 13:35.04 | ManxPower | wiz8291: echo is caused by the FAR END analog loop. |
| 13:35.19 | wiz8291 | this is just on an extension |
| 13:35.26 | ManxPower | It's there without VoIP, you just can't hear it because it echos back so fast. |
| 13:35.26 | wiz8291 | dialling the voicemail or echo test app |
| 13:35.45 | ManxPower | wiz8291: then you have some OTHER problem |
| 13:35.49 | wiz8291 | so phone -> channel bank -> asterisk voicemail |
| 13:36.10 | ManxPower | wiz8291: you don't have an echo issue. You have some other issue that happens to cause echo. IRQ shareing? |
| 13:36.15 | wiz8291 | the echo on the voicemail isn't there when called from a voip extension |
| 13:36.51 | ManxPower | wiz8291:percieved echo is caused by LATENCY. Analog should have much less latency than voip. |
| 13:36.55 | DrWho17 | wiz8291: sure, echo comes from the analog loop |
| 13:37.03 | wiz8291 | yup |
| 13:37.07 | wiz8291 | ok |
| 13:37.16 | wiz8291 | so a problem with the channel bank settings perhaps? |
| 13:37.27 | ManxPower | I'll bet your T-1 card is shareing an IRW |
| 13:37.29 | ManxPower | IRQ |
| 13:37.39 | ManxPower | or your motherboard has excessive PCI latency |
| 13:37.41 | DrWho17 | wiz8291: sure, mess with the gains |
| 13:37.49 | wiz8291 | just installing pciutils to have a look |
| 13:37.58 | ManxPower | cat /proc/interrupts |
| 13:38.03 | DrWho17 | usually echo comes from a level mismatch |
| 13:39.32 | wiz8291 | <PROTECTED> |
| 13:39.43 | wiz8291 | there we go |
| 13:39.52 | ManxPower | There's one potential cause of echo |
| 13:40.39 | wiz8291 | its hard to describe the sound |
| 13:40.43 | wiz8291 | the phone is very loud |
| 13:40.54 | wiz8291 | and everything has a sort of whine over it and sounds distorted |
| 13:41.03 | ManxPower | wiz8291: Stop. Get the card on it's own interrupt. Don't do anythng else until you get that done. |
| 13:41.48 | newl | Incorrect TN/BN configuration between the exchange and subscriber, 3KHz/Voice settings on the remote end (read your end), impedence settings on the line equipment incorrect..there's plenty of reasons. :) |
| 13:42.12 | ManxPower | newl: No telco involved in his setup. |
| 13:42.18 | wiz8291 | doing that now |
| 13:42.53 | newl | The T1 comes out of thin air? :) |
| 13:43.01 | lancey | thank you all for the help |
| 13:43.01 | ManxPower | newl: no, channel bank |
| 13:43.03 | lancey | bye guys |
| 13:43.11 | ManxPower | newl: So, "yes" |
| 13:43.31 | Gand_DJ | How's a channel bank compared to a T1 for termination? |
| 13:43.35 | Gand_DJ | If I'm reading right |
| 13:43.49 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 13:44.09 | bjohnson | Gand_DJ: you're not |
| 13:44.23 | bjohnson | Gand_DJ: you have an ATA right? |
| 13:44.29 | Gand_DJ | Not yet. |
| 13:44.30 | Gand_DJ | :( |
| 13:44.55 | bjohnson | channel bank is like a big ATA |
| 13:45.03 | bjohnson | but instead of connecting directly to the LAN |
| 13:45.10 | bjohnson | it connects to a T1 card |
| 13:45.12 | SuPrSluG | bjohnson:I rebooted the sipura and now no phantom rings. THANX |
| 13:45.22 | bjohnson | SuPrSluG: just the SPA? |
| 13:45.31 | SuPrSluG | yeah |
| 13:45.32 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 13:45.40 | bjohnson | I could have sworn I had to do a full reboot of the server |
| 13:45.42 | Gand_DJ | ok... so the channel bank is like an ata with 10+ fxs ports? |
| 13:45.57 | bjohnson | up to 31 .. but usually 24 |
| 13:46.06 | Gand_DJ | kewl |
| 13:46.13 | Gand_DJ | must be expensive for those suckers |
| 13:46.13 | Hmmhesays | http://www.webopedia.com/TERM/C/channel_bank.html |
| 13:46.26 | bjohnson | and some models can use fxo instead of some of the fxs |
| 13:46.59 | bjohnson | Gand_DJ: if you need over about 10 fxo or fxs ports .. a used channel bank and digium T1 card is worth considering |
| 13:47.02 | SuPrSluG | bjohnson:well to be fair i kicked out the electricity to my thin client.and now no problems.Go figure? |
| 13:47.28 | bjohnson | thin client? |
| 13:47.35 | DrWho17 | Gand_DJ: well we use channel banks to connect DS0's to channels of a T1 |
| 13:47.46 | jabbzy | how do you mean - phantom rings? |
| 13:47.50 | bjohnson | SuPrSluG: does the thin client have anything to do with *? |
| 13:48.05 | bjohnson | jabbzy: one ring, every 60 minutes |
| 13:48.18 | bjohnson | exactly every 60 minutes |
| 13:48.22 | jabbzy | bjohnson : what was causing that? |
| 13:48.29 | bjohnson | don't know |
| 13:48.38 | bjohnson | I'm thinking it was a dhcp issue |
| 13:49.07 | jabbzy | possibly. i always get a ring on startup / power up.... |
| 13:49.13 | jabbzy | of the atat |
| 13:49.15 | SuPrSluG | must have been something like that. |
| 13:49.23 | bjohnson | the SPAs ring once when you turn them on .. maybe it is to actually show that they have a valid LAN connection including a valid IP |
| 13:50.04 | bjohnson | if you see it again, try just assigning the SPA a static ip and see if the problem goes away |
| 13:51.09 | bjohnson | DrWho17: that use is a little harder to explain to a newbie |
| 13:51.38 | SuPrSluG | anyone else get this virus? You have just received the "Alabama Virus" As we ain't got no programming experience, this virus works on the honor system. Please delete all the files from your hard drive and manually forward this virus to everyone on your mailing lists. > > > > Thanks for your cooperation, > > University of Alabama Computer Engineering Department. |
| 13:51.57 | bjohnson | HAA HAAA |
| 13:52.38 | *** join/#asterisk kmi (~km@office-gw.westend.com) |
| 13:52.48 | SuPrSluG | it came in my IM |
| 13:52.48 | DrWho17 | bjohnson: yes, but that is what a channel bank does |
| 13:53.08 | *** join/#asterisk yhonnkar (~mrtg@200.75.105.33) |
| 13:53.22 | *** join/#asterisk laotan (~jesse@H38.C18.B96.tor.eicat.ca) |
| 13:53.25 | yhonnkar | hi |
| 13:53.41 | *** join/#asterisk doctorCTI (~DoctorCTI@modemcable094.64-80-70.mc.videotron.ca) |
| 13:53.49 | *** join/#asterisk laotan (~jesse@H38.C18.B96.tor.eicat.ca) |
| 13:55.03 | nestAr | Anyone using HEAD and have problems with SetGroup/CheckGroup? |
| 13:55.18 | cypromis | nestAr: check the bugtracker |
| 13:56.31 | nestAr | i checked. nothing. |
| 13:57.06 | nestAr | i'm just wondering if the syntax between stable and head is different |
| 13:58.05 | *** join/#asterisk dosle (~dosle@silenceisdefeat.org) |
| 13:58.06 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
| 14:00.10 | *** join/#asterisk vaewyn (freeman@mail.parrishmachine.com) |
| 14:00.33 | vaewyn | #include appropriate_GMT_based_greeting.h |
| 14:01.37 | vaewyn | ok... so does anyone know... is there a reason NOT to set nat=yes for sip.conf? Does it slow it down... or create problems or...??? |
| 14:01.44 | *** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it) |
| 14:02.41 | vaewyn | seems from my testing that they should just make that the default and fly with it |
| 14:03.40 | ManxPower | vaewyn: It causes problems with a small number of phones. I don't recall exactly which ones (SNOM?) |
| 14:04.03 | jabbzy | what problem mr ManxPower |
| 14:04.17 | ManxPower | jabbzy: they stop working |
| 14:04.20 | vaewyn | ManxPower: ok... but if your phones are ok... then just do it in the default and no ill effects? |
| 14:04.34 | mAsH` | how can i see the state of my sip client? |
| 14:04.35 | ManxPower | vaewyn: I think so. |
| 14:04.40 | vaewyn | Cause all mine seem to work just fine with it so... :P |
| 14:04.43 | ManxPower | mAsH`: "sip show peers" |
| 14:05.20 | mAsH` | ManxPower: i would to see if a phone is ringing or if is just talking with another client |
| 14:05.46 | ManxPower | mAsH`: "sip show channels" will tell you about ACTIVE SIP CALLS. |
| 14:05.57 | ManxPower | "sip show peers" just tells you about configured SIP devices. |
| 14:05.59 | jabbzy | oh ok - nice and simple! any particular reason why they stop working? |
| 14:06.16 | mAsH` | well ManxPower |
| 14:06.37 | ManxPower | ~google site:lists.digium.com nat=yes SNOM problem |
| 14:06.39 | *** join/#asterisk oden (~oden@194-237-146-22.customer.telia.com) |
| 14:06.43 | ManxPower | ~google site:lists.digium.com nat=yes SNOM |
| 14:06.51 | ManxPower | ~google site:lists.digium.com nat=yes causes problems |
| 14:06.58 | ManxPower | ugh. |
| 14:06.59 | mAsH` | but i can't see if the phone is ringing or if is in DND or if it has another active call |
| 14:07.12 | ManxPower | jabbzy: it was over a year ago that I saw a discussion on the list abou tit. |
| 14:07.15 | jabbzy | on a slighlty different note, is there anyway to purge the "sip show peers" table - to force re-registration? |
| 14:07.18 | ManxPower | mAsH`: no. |
| 14:07.30 | jabbzy | thanks Manx-Power |
| 14:07.33 | ManxPower | jabbzy: no. qualify=yes might do what you want. |
| 14:07.49 | nestAr | ugh |
| 14:07.56 | nestAr | i think i'm just going to buy new phone |
| 14:07.57 | nestAr | s |
| 14:08.05 | nestAr | :| |
| 14:08.07 | ctooley | with SIP qualify helps with a lot of problems |
| 14:08.26 | vaewyn | ummm... where did I get this in my sip.conf 'remote=remove' ?!? anyone have idea what that does... google is clueless on it |
| 14:08.42 | ManxPower | vaewyn: invlid options do not generate an error. |
| 14:08.57 | jabbzy | qualify=5000 is what i do at present, seems silly for the consolse not to be able to purge certain tables. |
| 14:09.01 | ManxPower | vaewyn: perhaps you put it in during a drunken stupor? I've done that. |
| 14:09.06 | vaewyn | ManxPower: time to hack and purge then :} |
| 14:09.14 | ManxPower | jabbzy: why do you want to purge? |
| 14:09.16 | vaewyn | ManxPower: don't drink so... probably not :P |
| 14:09.34 | ManxPower | vaewyn: perhaps lack of sleep then. |
| 14:09.46 | ManxPower | looks like we are out of coffee. /me points at CoffeeIV has quit () |
| 14:09.58 | vaewyn | ManxPower: quite a bit more likely :} |
| 14:10.00 | jabbzy | i don't in normal times, but when i am testing new systems. it is nice to be able to throw everything off and see it re-register |
| 14:10.33 | vaewyn | stop now does that :} |
| 14:11.22 | jabbzy | thats not exactly what i mean mr vaewyn - when you restart astersik, it still has old data in. |
| 14:11.46 | Mimmus | I'm planning the best path to introduce Asterisk in my company, integrating with an analog Alcatel PBX |
| 14:12.19 | Mimmus | I'm thinking to put it in front of Alcatel and transparently foward all calls, as first step |
| 14:12.40 | Mimmus | then I will gradually introduce SIP phones connected to Asterisk |
| 14:12.46 | cjk | 533, 256 mb ram mhz should be enough to transcode 8 channels from isdn to ilbc? |
| 14:12.47 | Mimmus | what do you think? |
| 14:12.59 | vaewyn | btw... is funny making Norhell jockey downstairs jump every time I find a hole in his setup... and I am just accessing over a PRI ;P |
| 14:13.06 | jabbzy | its always nice if you are troubleshooting to tell the end user switch off your phone, then at the console you remove its entry, the user switches the phone back on and by checking sip show peers see if it shows back up |
| 14:13.29 | CTJ | Good Morning. I am trying to use SJphone to connect to my asterisk and get a single message played back. I've managed to get SJPhone to register. I have [sip] exten=>s,1,Answer() exten => s,2,Playback(tt-weasle), exten=>s,3,Hangup() in my exten.conf file. But I can't seem to see any channels being created and SJphone is reporting: End reason: Critical transaction failed: Client INVITE. |
| 14:13.39 | *** join/#asterisk Voicelynx (VoiceLynx@user-0cdv656.cable.mindspring.com) |
| 14:13.54 | *** join/#asterisk sonic74 (~Sven@pinguin.tdb.de) |
| 14:14.01 | vaewyn | jabbzy: then turn your qualify down and when they unplug it the peer will be shown down... when they plug it back in it will be shown up |
| 14:14.40 | *** join/#asterisk rlg (~umairbari@202.142.189.86) |
| 14:15.27 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:15.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:15.38 | vaewyn | morning anthm |
| 14:15.49 | anthm | hi |
| 14:15.54 | *** part/#asterisk sonic74 (~Sven@pinguin.tdb.de) |
| 14:16.13 | wiz8291 | ok, irq is seperate... its better |
| 14:16.17 | wiz8291 | but still not great! |
| 14:17.13 | vaewyn | ohh egads... idiot I am helping is running g.729 transcoding on a P2-500 |
| 14:17.19 | vaewyn | guess how well that works :P |
| 14:17.20 | jabbzy | thats an option, but surley that increases network traffic? |
| 14:17.46 | LeoB | [HELP NEEDED] Although I've configured different softphones (SIP/IAX) according to the instructions, none of them seem to be able to register with my asterisk server. Can anyone help? |
| 14:18.05 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:18.16 | Moonwick | [WITTY REMARK] Ethernet cable plugged into the * server? |
| 14:18.30 | vaewyn | jabbzy: that's why you need to tune your qualify to meet your needs without eating too much network :} |
| 14:19.03 | jabbzy | well a thanks to all for at least confirming i wasn't mad |
| 14:19.36 | Gand_DJ | How would one enable * to allow iLBC or Speex? |
| 14:19.51 | vaewyn | Gand_DJ: ilbc is built in... not sure on speex |
| 14:19.53 | LeoB | the thing is that my server seems to work fine with FWD and IAXTel... |
| 14:19.54 | Gand_DJ | It seems that * only allows Ulaw/Alaw and GSM for trying Firefly codecs |
| 14:20.25 | vaewyn | Gand_DJ: what are your allow=lines?> |
| 14:20.28 | DrWho17 | Gand_DJ: disallow=all; allow=ilbc |
| 14:20.32 | Gand_DJ | they are blank. |
| 14:20.43 | bjohnson | does anyone know much about wi-max? |
| 14:20.53 | jabbzy | a millionare style fastest finger first question :) put the codecs in order of bandwidth - lowest first |
| 14:20.56 | bjohnson | I'm reading that it might compete with cellular |
| 14:21.02 | jabbzy | hopefully |
| 14:21.10 | Gand_DJ | lol |
| 14:21.12 | jabbzy | it will compete with celluree |
| 14:21.30 | vaewyn | bjohnson: it's wifi on steriods... long range... good speed... no line of site issues |
| 14:21.31 | jabbzy | full disclosure : made redundent by a cellurar operator |
| 14:21.47 | Gand_DJ | Shouldn't having nothing entered for allow / disallow be the same as allow all? |
| 14:21.49 | bjohnson | what kind of bandwidth, distance examples are available? |
| 14:21.51 | newl | firefly (well, freshtel is their provider) uses gsm, ilbc, and speex |
| 14:22.00 | bjohnson | and any cost examples |
| 14:22.17 | Gand_DJ | newl, I'm using 3rd party version.. for my * server.... along with another softphone client |
| 14:22.22 | vaewyn | bjohnson: bandwidth above wifi... 30-50 mile tower range... not sure on rest |
| 14:22.23 | jabbzy | eh? it will always have line of site issues, its radio |
| 14:22.51 | vaewyn | jabbzy: not like wifi does... a tree or 2 and you kill wifi |
| 14:23.00 | CoaxD | they need to make a wifi standard that works reliably over 100 mile omnidirectional. *lol* |
| 14:23.20 | bjohnson | ie .. is it suitable for SOHO/home type use or is it a more of a deployed system |
| 14:23.21 | vaewyn | CoaxD: amen! :} UWB might hit that soon |
| 14:23.37 | jabbzy | no, but i wouldn't want anyone listening to think that its magic ;) |
| 14:23.49 | newl | CoaxD: That's called rtty. :) |
| 14:23.52 | CoaxD | vaewyn: We have the technology to make shit like this happen nowadays. we shouldnt be stuck in the dark ages anymore. |
| 14:23.54 | vaewyn | bjohnson: most of the industry are talking both... cell type towers but also in home repeaters and such |
| 14:24.00 | bjohnson | I have a 600 m jump through trees that I'm wondering if this might work |
| 14:24.15 | CoaxD | bjohnson: You just need to get yourself a pringles can or two |
| 14:24.24 | Gand_DJ | for allowing multiple codecs, can I just do allow=gsm,ilbm,ulaw etc |
| 14:24.25 | vaewyn | 600m? dig a trench :} |
| 14:24.27 | *** join/#asterisk boch (~as24@200.59.172.98) |
| 14:24.29 | bjohnson | haven't tried wifi with antennaes cause I think that the tress would still kill the signal |
| 14:24.32 | jabbzy | aye, but rtty has a minor issue with it baud rates :) |
| 14:24.38 | LeoB | how to make sure RedHat is not preventing my softphones from registering with my server? |
| 14:24.42 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:24.54 | bjohnson | vaewyn: 600 m across about 12 different properties |
| 14:24.58 | Gand_DJ | LeoB, disable any firewall? |
| 14:25.01 | newl | jabbzy: yeah, but the distance is great on 10m haha |
| 14:25.07 | vaewyn | bjohnson: horizontal drill ;P |
| 14:25.22 | bjohnson | under their basements? |
| 14:25.27 | LeoB | Gand_DJ, do you know how to do that? |
| 14:25.28 | vaewyn | bjohnson: ;P |
| 14:25.34 | jabbzy | sure is sir, although where you land - nobody knows :) oh and the power availible is a tad higer too! |
| 14:25.37 | Gand_DJ | nope. kinda new also |
| 14:25.56 | *** join/#asterisk moy (~kvirc@201.135.105.124) |
| 14:25.58 | jabbzy | a millionare style fastest finger first question :) put the codecs in order of bandwidth - lowest first - please! |
| 14:26.24 | newl | LeoB: service iptables stop, I'd also suggest another channel for *nix help. :) |
| 14:26.28 | vaewyn | lowest bandwidth.... morse code ;P |
| 14:26.34 | Nugget | heh |
| 14:26.35 | newl | -.-. --.- |
| 14:26.39 | CoaxD | ... --- ... ...---... |
| 14:26.40 | jabbzy | not true - psk31 is lower |
| 14:27.08 | bjohnson | depending on message being sent .. true/false would be lower |
| 14:27.09 | LeoB | thanks, I'll try that out! |
| 14:27.20 | vaewyn | semaphores! |
| 14:27.21 | vaewyn | hehehe |
| 14:27.51 | darkskiez | if anyone cares about the lost packets problem i was having, the switch had decided to dump our 500mbit multicast backbone on every port. |
| 14:28.03 | CoaxD | does anyone else remember "FrontDoor"? |
| 14:28.10 | vaewyn | darkskiez: that would hurt |
| 14:28.16 | lbow | fidonet. lbow remembers monster phone bills |
| 14:28.18 | jabbzy | we care mr darkskies. thats never good. |
| 14:28.27 | CoaxD | lbow: Haha. Those were the days, my friend |
| 14:28.38 | newl | CoaxD: I remember a long time before FD. B) |
| 14:28.40 | bjohnson | Sophmores !! |
| 14:28.43 | CoaxD | lbow: I was up to 6 echomail downstreams |
| 14:28.44 | bjohnson | oops |
| 14:28.50 | CoaxD | lbow: *gah* |
| 14:29.06 | CoaxD | lbow: On a single 14.4k modem |
| 14:29.21 | CoaxD | lbow: I ahve no idea how people ever called that damn BBS |
| 14:29.30 | lbow | try fidonet polls from south africa to europe. yow |
| 14:29.37 | vaewyn | modem? you didn't just whistle into the phone? |
| 14:29.49 | CoaxD | lbow: No |
| 14:30.20 | lbow | whistling? luxury (in a yorkshire accent). we used to grunt. and we were happy. |
| 14:30.38 | CoaxD | <LeoB> [HELP NEEDED] Although I've configured different softphones (SIP/IAX) |
| 14:30.40 | vaewyn | *unnn umpph.. unnn* |
| 14:30.43 | CoaxD | err |
| 14:30.51 | CoaxD | http://biseonline.com/r19/files/backbone.html |
| 14:31.31 | lbow | coaxd: it's still there? |
| 14:31.50 | CoaxD | lbow: FidoNet still exists, sure |
| 14:32.05 | CoaxD | http://www.rxn.com/~net282/ <-- a good graph of fidonet's statistics |
| 14:32.21 | LeoB | CoaxD, were you trying to talk to me? |
| 14:32.28 | lbow | lbow remembers porting binkleyterm to the acorn archimedes. even gave it a simple gui |
| 14:32.34 | CoaxD | LeoB: no, sorry. i mispasted |
| 14:32.45 | CoaxD | lbow: Hahahaha |
| 14:33.33 | lbow | all that code - and dead. I had someone ask for source code to something I wrote on the HP3000 in the 80s. (how old am I!?) - so maybe old code is never quite dead |
| 14:33.46 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 14:34.16 | lbow | or something like that |
| 14:34.25 | Nugget | I was the echomail link between .us and .nl for about a year, too. |
| 14:34.28 | CoaxD | lbow: Heh. :) Those were the days. :/ |
| 14:34.36 | CoaxD | Nugget: GAH! |
| 14:34.42 | CoaxD | nugget: *THAT* musta cost some $$! |
| 14:34.56 | lbow | nugget: I assumed everyone who did mass echomail links was sponging of their companies phone bills |
| 14:35.13 | Nugget | nope |
| 14:35.48 | Nugget | it was $200-$300 a month or so, iirc, and partly subsidized by node dues. |
| 14:36.02 | Nugget | still cost a bundle, though. I split the cost with the guy on the other end |
| 14:36.24 | CoaxD | i do however remember when our shared echomail costs dropped drastically because our NEC had it set up such that the new echomail packets could be ftp'ed instead of retrieved via modem |
| 14:36.37 | CoaxD | (you know. on this new 'internet' thing.) |
| 14:37.46 | lbow | "internet"? it'll never catch on. lets assimilate them with MSN rather ... |
| 14:37.49 | Gand_DJ | Ne1 know how to get eval license for G729 for testing while testing this *@home system for family use. I'm running *@home through Virtual Server 2005 so the Mac addresses are done via software (and will change on re-installs) |
| 14:38.18 | nestAr | don't reinstall? |
| 14:38.19 | nestAr | :) |
| 14:38.33 | CoaxD | pkzipped, the nodelist is 262k right now |
| 14:38.39 | CoaxD | (remember when it was like 1.5mb zipped?) |
| 14:38.56 | nestAr | pkzip! |
| 14:39.03 | lbow | can you do fido protocols over tcpip nowadays? |
| 14:39.23 | CoaxD | lbow: Yeah, it aint tough.. Just package up the stuff you're routing, and send it via tcp/ip instead |
| 14:39.31 | *** part/#asterisk cjk (~cjk@80.92.64.103) |
| 14:39.34 | CoaxD | lbow: FidoNet mostly runs over ipv4 nowadays |
| 14:39.44 | CoaxD | lbow: (of course, totally eliminating half the fun) |
| 14:40.18 | nestAr | damn, why aren't my colors working in the CLI |
| 14:40.18 | nestAr | :( |
| 14:40.32 | *** join/#asterisk Nivex (kjotte@user-0c8hq5r.cable.mindspring.com) |
| 14:42.17 | jabbzy | omg, fidonet - a big step up from bbs! |
| 14:42.44 | jabbzy | black and white monitor? |
| 14:43.09 | Gand_DJ | lol.. I still have a BBS system on my Amiga 1200 that I had setup to use a telnet device for incoming connections instead of the standard serial device (for dialup) |
| 14:43.28 | CoaxD | Gand: Heh. Does it have users? |
| 14:43.44 | Gand_DJ | It had.. haven't ran it in years. |
| 14:43.49 | CoaxD | Gand: Heh :) |
| 14:44.05 | CoaxD | Gand: I had a 4 line chat board here in Minnesota. I've had a few here and there over time |
| 14:44.18 | Gand_DJ | I could try to fire it back up, but my amiga has been iffy ever since I converted the system from a desktop into a Zorro-based tower |
| 14:44.18 | jabbzy | omg, a bbs! on a amiga 1200 - any more nostalga and ill end up on a bidding frenzy on ebay |
| 14:44.19 | CoaxD | Gand: but "The Corner BBS" will always live on in my heart, sir! |
| 14:44.39 | CoaxD | jabbzy: Hahahahaha |
| 14:44.51 | CoaxD | jabbzy: Thats okay. pay your $10 for a new amiga and go with it. *lol* |
| 14:44.52 | Gand_DJ | The BBS software I used was Max's BBS, and RPG-BBS |
| 14:45.19 | CoaxD | jabbzy: but be *sure* to get the 20mb hard disk option. you need file downloads! |
| 14:45.25 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 14:45.25 | Gand_DJ | I tried setting up the bbs via WinUAE, but it doesn't seem to install & run properly |
| 14:46.14 | CoaxD | hate. that. |
| 14:46.38 | *** join/#asterisk dalabera (~Dalabera@mail.pmrtechnologies.com) |
| 14:47.36 | Gand_DJ | currently trying to install webmin into my *@home |
| 14:48.05 | jabbzy | whats webim mr_dj? |
| 14:48.34 | Gand_DJ | Webmin is a web-based interface for system administration for Unix |
| 14:48.39 | Gand_DJ | http://www.webmin.com/intro.html |
| 14:48.50 | Gand_DJ | Webmin consists of a simple web server, and a number of CGI programs which directly update system files like /etc/inetd.conf and /etc/passwd. The web server and all CGI programs are written in Perl version 5, and use no non-standard Perl modules. |
| 14:49.21 | Gand_DJ | runs on port 10000 iir |
| 14:49.59 | nestAr | anyone got a second to look at this SetGroup/CheckGroup pastebin and tell me if there's a reason this shouldn't work with HEAD? |
| 14:50.02 | nestAr | http://pastebin.ca/9907 |
| 14:51.13 | *** part/#asterisk yhonnkar (~mrtg@200.75.105.33) |
| 14:51.39 | jabbzy | anything that makes unix easier for the masses is a good thing imho |
| 14:53.03 | Jas_Williams | nestAr: You do not have a priority 103 for the Checkgroup to fail to ? |
| 14:54.06 | Jas_Williams | change the 104 to 103 |
| 14:54.31 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 14:55.36 | nestAr | Jas_Williams: but assuming that's the only problem, the call should just drop dead in the water, not continue on with the Dial... right? |
| 14:56.03 | nestAr | i changed it, btw |
| 14:56.12 | nestAr | thanks for pointing that out.. |
| 14:56.33 | Jas_Williams | nestAr: No if a priority n+101 does not exist then Checkgroup exits with -1 and continues at the next priority I think |
| 14:56.46 | Jas_Williams | so the goes and dials your sip phone |
| 14:56.49 | nestAr | Ok. |
| 14:57.09 | Jas_Williams | Question is does it work now ? |
| 14:57.33 | nestAr | Well, I'm running stable. It was/is working as pasted with v1-0 |
| 14:57.45 | nestAr | i need a test server |
| 14:57.51 | nestAr | i think i'll build one right now |
| 14:58.47 | vaewyn | hey... anyone have calleridname working over a PRI to a Norhell? |
| 14:59.04 | AngelGabriel | I have an Asterisk install up and running, and I have linphone on a diffrent machine, is it possible for me to connect linphone to the asterisk box? |
| 14:59.44 | Nugget | thee is no technical reason why you can't. |
| 15:00.32 | *** join/#asterisk pbxjunkie (~stormtroo@213.5.44.113) |
| 15:00.34 | *** part/#asterisk pbxjunkie (~stormtroo@213.5.44.113) |
| 15:01.30 | jabbzy | I need to build a system, thats going to be running a quad PRI card in the near future, I don't have a real PRI to test it with - If i get two cards and build 2 machines, one card in each, can i theoretically set one up to pretend to be the telco? |
| 15:01.34 | *** join/#asterisk paddyboy (paddy@c-67-174-243-209.hsd1.ca.comcast.net) |
| 15:01.52 | Moonwick | yeah, in theory I think you can do that |
| 15:01.59 | *** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com) |
| 15:03.07 | moy | Hi everybody |
| 15:03.10 | Gand_DJ | I called my telco.. can't believe to use a PRI, the monthly cost is $750-$1000 CDN / month |
| 15:03.28 | Gand_DJ | At least the DID's are only $2.50 each.. and sold in packs of 10... per month |
| 15:04.23 | moy | i have a Wildcard TDM400P, with 4 FX0 ports, i need to know when some line is connected, and when not, so i can tell the user that no line is connected, any ideas? |
| 15:04.32 | moy | zttool, does not say so |
| 15:05.53 | jabbzy | any special settings i should be aware of? do they work on a similar system to fxo/fxs ? |
| 15:08.51 | *** join/#asterisk Lee__ (~lee@cpe-69-203-206-248.nyc.res.rr.com) |
| 15:08.57 | *** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net) |
| 15:09.18 | lbow | moy: does zap show channel "n" show "InAlarm: 1" when there is no line connected? |
| 15:09.56 | rephorm | i get a whole lot of messages like the following in my log: "chan_iax2.c: Immediately destroying 8, having received INVAL". anyone seen something like this before? |
| 15:11.43 | rephorm | by alot, i mean 1 or 2 per second |
| 15:11.52 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 15:12.34 | Aze` | Who use bristuff on 2.4 ? |
| 15:13.34 | mcnobody | Aze me |
| 15:13.38 | *** join/#asterisk devolish (devolish@devolish.no) |
| 15:15.10 | Juggie | kram, when are you leaving for toronto |
| 15:16.05 | *** join/#asterisk BlackCrow (BlackCrow@phoenix.arminco.com) |
| 15:16.53 | *** join/#asterisk afrosheen (~afro@c-67-166-172-141.hsd1.tx.comcast.net) |
| 15:23.37 | *** join/#asterisk phpboy (~sj@tpr-165-231-125.telkomadsl.co.za) |
| 15:23.39 | phpboy | guys |
| 15:23.51 | phpboy | how would I go about blocking certain extentions |
| 15:23.54 | phpboy | for example |
| 15:23.58 | phpboy | I have this |
| 15:24.22 | phpboy | exten => _0XXXXXXXXXXXXXX,1,Dial,Zap/g1/${EXTEN} |
| 15:24.31 | phpboy | but that covers basically everything |
| 15:24.39 | phpboy | so what I like to beable to add to that context is |
| 15:24.42 | phpboy | something like |
| 15:24.51 | phpboy | !exten => _0XXXX082XXXXXXX,1,Dial,Zap/g1/${EXTEN} |
| 15:25.00 | phpboy | which would block any number matching that |
| 15:25.03 | phpboy | any idea's? |
| 15:25.05 | eKo1 | eh, add the first extension you want to block before that one and instead of dial, to a hangup |
| 15:26.02 | eKo1 | Those are long ass numbers by the way. |
| 15:26.03 | phpboy | ok |
| 15:26.08 | phpboy | eKo1: it is |
| 15:26.23 | phpboy | I'm currently behind a telco PBX system |
| 15:26.30 | phpboy | eKo1 so ! would work |
| 15:26.30 | phpboy | ? |
| 15:26.49 | phpboy | prefix by a '!' |
| 15:27.03 | eKo1 | eh no. |
| 15:27.22 | eKo1 | Say i want to block exten 310 but let all others go through. I would have: |
| 15:27.33 | eKo1 | exten => 310,1,hangup |
| 15:27.44 | eKo1 | exten => 3XX,1,dial(...) |
| 15:27.52 | phpboy | ah |
| 15:27.53 | phpboy | I see |
| 15:27.56 | phpboy | or rather |
| 15:28.01 | phpboy | exten => 310,1,hangup |
| 15:28.08 | *** join/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 15:28.08 | phpboy | exten => 3XX,2,dial(...) |
| 15:28.09 | phpboy | :P |
| 15:28.13 | phpboy | thanks a mil :) |
| 15:28.13 | eKo1 | eh no. |
| 15:28.17 | phpboy | hmmm |
| 15:28.24 | phpboy | both first process? |
| 15:28.24 | eKo1 | It's supposed to be exactly how I have it. |
| 15:28.38 | Gand_DJ | ne1 here successfully using FWD for incoming calls to *. |
| 15:28.48 | Gand_DJ | I can call out, but when someone calls in, they get busy signal |
| 15:28.50 | phpboy | ah, ok |
| 15:28.51 | phpboy | soz :< |
| 15:28.52 | Gand_DJ | it doesn't hit my box at all |
| 15:28.57 | phpboy | eKo1: one question though |
| 15:29.06 | phpboy | is there a way I could, instead of make it hang up |
| 15:29.11 | phpboy | make it go to a voice |
| 15:29.15 | eKo1 | Gand_DJ: Do a 'sip debug' and check it. |
| 15:29.18 | phpboy | stating you can't dial that numbeR? |
| 15:29.32 | eKo1 | phpboy: you can do whatever you want with it. |
| 15:29.44 | *** join/#asterisk RoyK (~roy@ua-213-115-144-116.cust.bredbandsbolaget.se) |
| 15:29.47 | RoyK | http://filebox.vt.edu/users/xmayox/5720322434.gif |
| 15:30.03 | nestAr | :( my ansi color stopped working. |
| 15:30.24 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 15:30.24 | phpboy | thanks a mil eKo1 :) |
| 15:30.29 | *** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 15:30.30 | Gand_DJ | I have stuff randomly showing up in sip debug |
| 15:30.30 | ariel_ | Hello everyone |
| 15:30.34 | eKo1 | ansi color? |
| 15:30.39 | nestAr | in the cli |
| 15:30.45 | Gand_DJ | FWD is IAX2 though, so should I be doing sip debug? |
| 15:30.59 | eKo1 | that happens to me every now and then. |
| 15:31.10 | Pinhole | I find that piping the output to /dev/null gets rid of any undesired output. |
| 15:31.12 | eKo1 | Gand_DJ: oh, nevermind |
| 15:31.13 | newl | Gand_DJ: I get the same. Outgoing okay, incoming notta on iax2 debug beyond the poke, pong, ack. |
| 15:31.19 | phpboy | exten => _0XXXX082XXXXXXX,1,Dial,Zap/g1/${EXTEN} |
| 15:31.19 | phpboy | exten => _0XXXXXXXXXXXXXX,1,Dial,Zap/g1/${EXTEN} |
| 15:31.20 | phpboy | exten => _0XXXXXXXXXXXXXX,2,Congestion |
| 15:31.25 | phpboy | eKo1: does that seem correct to you? |
| 15:31.37 | phpboy | oh shit |
| 15:31.42 | phpboy | nvm |
| 15:31.45 | phpboy | I see my mistake |
| 15:31.46 | phpboy | soz :P |
| 15:31.46 | Wazb | hi all |
| 15:32.05 | newl | "Shit? See no shit." - Johnny 5 |
| 15:32.40 | Wazb | i am looking at wakeup-call.pl , i need to use it but i dont how ? any help? |
| 15:33.54 | eKo1 | Wazb: run it as root and see what happens. |
| 15:34.28 | nestAr | nice! |
| 15:34.28 | nestAr | make: Warning: File `pbx.c' has modification time 4.8e+02 s in the future |
| 15:34.32 | moy | Ibow: no, zap show channel <num> does not say that is on alarm |
| 15:35.39 | eKo1 | nestAr: Looks like that file when into the future. |
| 15:35.56 | nestAr | yep |
| 15:36.10 | nestAr | fixed now |
| 15:36.15 | mishehu | ugh. |
| 15:36.38 | mishehu | polycom ip500 downloads the xml settings files it's always used, but the settings never seem to take hold. |
| 15:36.59 | nestAr | does it have write access to the ftp server? |
| 15:37.06 | mishehu | nestAr: sure does. |
| 15:37.22 | mishehu | nestAr: it was fine since january, suddenly one of the 5 phones here just borked. |
| 15:37.41 | facek_ | Do anyone have a some application to show on computer.. who is calling me? |
| 15:37.42 | nestAr | polycoms are just really weird |
| 15:37.45 | nestAr | i have a lot of them |
| 15:37.46 | facek_ | if i am using hardphone |
| 15:37.54 | nestAr | and i'm less impressed by the day |
| 15:38.12 | mishehu | nestAr: how do I reset all settings to factor default? |
| 15:38.15 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-205-173.dsl.scarlet.be) |
| 15:38.33 | nestAr | i think you hold down 4, 6, 8, 0 |
| 15:38.40 | nestAr | it's in the admin pdf |
| 15:38.51 | nestAr | it should prompt you for the admin password |
| 15:38.55 | rephorm | 4,6,8,* |
| 15:39.03 | nestAr | there you go. |
| 15:39.04 | nestAr | i was close |
| 15:39.07 | rephorm | :) |
| 15:39.48 | nestAr | i just love these phones because sometimes reseting the defaults doesn't seem to do much of anything |
| 15:40.18 | rephorm | i think the 4,6,8,* thing just whipes out any settings you put in manually through the phone, or through the web interface |
| 15:41.02 | nestAr | i like my 300, when i try to save a contact to the directory, it give me a Busy error |
| 15:41.06 | nestAr | and never saves it |
| 15:42.02 | Pinhole | Anybody got technical specs (SNMP mibs, bootfile, extra magic) for the motorola vt1000 or vt1005? (Not the vonage version, although that might be useful too) |
| 15:42.33 | Jas_Williams | facek_: http://www.voip-info.org/wiki-Asterisk+call+notification |
| 15:42.40 | facek_ | Jas_Williams thankls |
| 15:43.00 | facek_ | Jas_Williams this is not good for me |
| 15:43.12 | Jas_Williams | Why ? |
| 15:46.11 | *** join/#asterisk BSDnewbie (hehe@202.179.26.144) |
| 15:46.16 | BSDnewbie | wow |
| 15:46.18 | BSDnewbie | many ppl |
| 15:47.00 | mishehu | damn. I forgot what password I set for ftp for the user I made for the phones... |
| 15:47.07 | BSDnewbie | Has anyone uses asterisk in large Ip network with ~500 clients? |
| 15:47.24 | mishehu | looks like I'll need to reset all the phones... |
| 15:47.48 | *** join/#asterisk ToyMan (~konversat@204-8-82-238.webjogger.net) |
| 15:48.02 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 15:48.17 | wiz8291 | anyone recommend a good adsi phone that works well with asterisk? |
| 15:48.20 | moy | just in case someone is interested. To se if the zap card has a line connected to it, the OnHook parameter say so when doing zap show channel <chann> |
| 15:48.46 | moy | :P i just discovered it heheh |
| 15:48.48 | *** join/#asterisk sabre (urfos@pcp03281999pcs.gillst01.ar.comcast.net) |
| 15:49.05 | *** join/#asterisk ast_freak (~ast_freak@hades-out.universalsystems.net) |
| 15:50.01 | ast_freak | Has anyone done anything with SMDI (Simplified Message Desk Interface)? Is that feature available? |
| 15:51.07 | facek_ | no |
| 15:52.22 | *** join/#asterisk ManxPower (~eric@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 15:54.45 | *** join/#asterisk PBXtech[mobile] (~upirc@wirelessdata-031-010.mycingular.net) |
| 15:55.13 | RoyK | BSDnewbie: yes, but not with BSD |
| 15:55.26 | *** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net) |
| 15:56.12 | wiz8291 | anyone used the sayson phones? |
| 15:56.14 | BSDnewbie | why ? any disadvantages asterisk with BSD? |
| 15:57.03 | Gand_DJ | Is there a way in allow = to prioritize which codec to try first? |
| 15:57.05 | Nugget | BSDnewbie: the zaptel drivers can be a pain in the ass to work with. BSD support (well, FreeBSD) lags the Linux world. |
| 15:57.15 | Nugget | but for pure ip telephony, it's just super. |
| 15:57.23 | Gand_DJ | my softphone defaults to ulaw, and I want to use gsm first |
| 15:57.33 | Gand_DJ | but if gsm isn't supported, then try ulaw |
| 15:57.39 | Nugget | I run asterisk in linux, freebsd, and mac os x. |
| 15:57.51 | BSDnewbie | aahan |
| 15:59.12 | phpboy | BSDnewbie: I attempted a FreeBSD install |
| 15:59.18 | phpboy | HA HA HA :< |
| 15:59.20 | BSDnewbie | then? |
| 15:59.32 | BSDnewbie | just installan of freebsd? |
| 15:59.33 | phpboy | zaptel drivers cause kernel panics |
| 15:59.43 | phpboy | I've had to move a linux :< |
| 16:00.11 | BSDnewbie | aahan, |
| 16:00.25 | BSDnewbie | which branch of linux seems be good for asterisk? |
| 16:00.55 | BSDnewbie | redhat suse debian mandrak ...? |
| 16:01.17 | phpboy | BSDnewbie: I tried mandrake |
| 16:01.25 | phpboy | I wouldn't waste my time there if I was you |
| 16:01.27 | Jas_Williams | all work I use slackware, others use redhat, gento |
| 16:01.38 | Jas_Williams | Debian |
| 16:01.39 | phpboy | I'm using a distro of asterisk |
| 16:01.45 | phpboy | called asterisk at home |
| 16:01.52 | phpboy | nice to learn on |
| 16:01.57 | phpboy | OS and everything included |
| 16:01.59 | Jas_Williams | If you have no preference or are new use asterisk at home |
| 16:02.18 | Nugget | asterisk doesn't care what linux you use. so use whichever one is least awful for your needs. |
| 16:02.33 | Nugget | I went with slackware, but not for any logical reason. |
| 16:02.49 | Nugget | mostly I like how slackware stays out of my way and doesn't keep reminding me that it is linux |
| 16:02.58 | phpboy | slackware I'd trust |
| 16:03.00 | phpboy | personally |
| 16:03.01 | BSDnewbie | okay tnx ppl for suggestions, Now need to try me. |
| 16:03.04 | BSDnewbie | :D |
| 16:03.08 | phpboy | not speaking from experience though |
| 16:03.34 | phpboy | BSDnewbie: I'd suggest asterisk at home |
| 16:03.37 | phpboy | it's VERY VERY nice |
| 16:03.43 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 16:03.46 | BSDnewbie | aahan, |
| 16:05.29 | ast_freak | I like slack too. |
| 16:05.51 | PBXtech[mobile] | has anyone made any alarm notification stuff? |
| 16:06.04 | ast_freak | ~seen kald|work |
| 16:06.16 | jbot | kald|work <~KalD@proxy.corp.telesym.com> was last seen on IRC in channel #asterisk, 6d 18h 31m 10s ago, saying: 'wow - so my company just decided to go .BOMB...'. |
| 16:06.16 | Gand_DJ | dumb question.. does *@home support conference calling by default? |
| 16:06.32 | PBXtech[mobile] | yes gand |
| 16:06.46 | PBXtech[mobile] | 8+extension |
| 16:06.50 | Qwell | phpboy: and very very useless overall |
| 16:07.05 | phpboy | Qwell: what u mean? |
| 16:07.12 | Gand_DJ | *@home isn't useless |
| 16:07.14 | Qwell | You won't get much support in here for it. |
| 16:07.35 | bannerman | I just can't track this down :( 5-10 minutes into a call, I can still hear the other side, but my voice drops out for about 20 seconds. |
| 16:07.37 | phpboy | Qwell: the * itself on the system is pretty much identical to normal asterisk |
| 16:07.55 | bannerman | I'm using IAX2 via Nufone, and SIP phones, using ulaw to the phones and g729 to Nufone |
| 16:08.05 | Qwell | phpboy: watch next time somebody tries to get help with *@home in here |
| 16:08.14 | ast_freak | Does * use serial ports for anything right now? |
| 16:08.44 | *** join/#asterisk jakepdev (~jakepdev@pool-68-236-58-19.phil.east.verizon.net) |
| 16:08.47 | Gand_DJ | Qwell, you know if there is a way to prioritize what codecs to have the softphones use? if codec 1 isn't allowed, try codec 2, etc |
| 16:08.50 | phpboy | Qwell: what u mean? |
| 16:09.02 | Qwell | phpboy: They'll get shoved aside, and told to use a real install |
| 16:09.17 | phpboy | uhm |
| 16:09.24 | Qwell | For many things, you'll need to edit the configs |
| 16:09.32 | Qwell | and, if you're going to edit the configs, WHY even use *@home? |
| 16:09.33 | phpboy | Qwell: if you didn't know it was an asterisk @ home install |
| 16:09.38 | phpboy | u woudn't know the diffs |
| 16:09.53 | phpboy | Qwell: it's the initial install |
| 16:09.57 | Qwell | phpboy: The configs from *@home are freaking ugly |
| 16:10.02 | phpboy | that is true |
| 16:10.09 | phpboy | but I've rewritting all the configs |
| 16:10.15 | phpboy | I dun use the web interface |
| 16:10.16 | Qwell | Then *WHY* use *@home? |
| 16:10.17 | jakepdev | qwell - i used it becase it made the install quick - just renamed the configs if I ever want to use them later |
| 16:10.23 | phpboy | I must agree |
| 16:10.31 | phpboy | the web interface does suck |
| 16:10.32 | PBXtech[mobile] | qwell its nice |
| 16:10.45 | phpboy | Qwell: I did the same as jakepdev |
| 16:10.46 | PBXtech[mobile] | its easy |
| 16:10.49 | phpboy | for quick install |
| 16:10.52 | phpboy | it's VERY nice |
| 16:11.04 | PBXtech[mobile] | simple |
| 16:11.09 | PBXtech[mobile] | :) |
| 16:11.13 | Qwell | I bet I could do an install much faster then you with *@home, and I use gentoo |
| 16:11.25 | Qwell | from scratch, CD in drive, booting the installation |
| 16:11.29 | jakepdev | but you'd spend more time getting it up and running |
| 16:11.40 | Qwell | not a chance |
| 16:11.43 | phpboy | Qwell: each one to his own |
| 16:11.44 | PBXtech[mobile] | not all the config scripting it has |
| 16:11.52 | phpboy | I'm clearly not NEAR as good as you at this |
| 16:11.54 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 16:11.57 | phpboy | I'm from a FreeBSD world |
| 16:11.57 | Qwell | phpboy: I suck at this |
| 16:12.02 | phpboy | still getting to know Linux |
| 16:12.09 | phpboy | when I'm comfy with it |
| 16:12.13 | phpboy | I'll install manually |
| 16:12.14 | mishehu | ah thank gawd for ethereal |
| 16:12.18 | Qwell | anyhow, off to work |
| 16:12.32 | *** part/#asterisk PBXtech[mobile] (~upirc@wirelessdata-031-010.mycingular.net) |
| 16:12.32 | phpboy | cheers d00d |
| 16:12.33 | phpboy | :) |
| 16:12.34 | *** join/#asterisk brettnem (~mive29@user-0ccsr10.cable.mindspring.com) |
| 16:12.46 | brettnem | yawn.. hello all |
| 16:13.02 | ast_freak | Does asterisk use serial ports, anyone? |
| 16:13.04 | jakepdev | it's the classic - don't knowk it till ya try it |
| 16:13.11 | brettnem | woah! wtf, only 58 people in here??? what happened? |
| 16:13.22 | ast_freak | I read 315. |
| 16:13.29 | jakepdev | serial ports?? for what? |
| 16:13.30 | RestLessGemini | me too.. 315 |
| 16:13.34 | techie | 315 |
| 16:13.42 | phpboy | I see 315 |
| 16:13.43 | phpboy | :/ |
| 16:13.44 | jakepdev | 315 |
| 16:14.04 | eKo1 | brettnem: stop smoking crack |
| 16:14.16 | RestLessGemini | brettnem: i guess you are in b/w netsplit |
| 16:14.16 | jakepdev | yep - there are better drugs |
| 16:15.11 | bannerman | I don't see why everyone dislikes asterisk@home |
| 16:15.11 | ast_freak | anything. I'm thinking about writing SMDI into asterisk. Doesn't anyone else have a need for MWI to work on legacy switches? |
| 16:15.11 | bannerman | It seems like a pretty decent setup to me, what's wrong with it? |
| 16:15.11 | brettnem | anyone here?? what is going on? |
| 16:15.15 | ast_freak | brettnem: stop smoking crack |
| 16:15.19 | mogorman | it breaks asterisk cvs |
| 16:15.27 | mogorman | and zaptel cvs |
| 16:15.32 | mogorman | it is not as clean |
| 16:15.36 | eKo1 | bannerman: define 'decent' |
| 16:15.40 | phpboy | there's nothing wrong with asterisk @ home for most peeps |
| 16:15.47 | RestLessGemini | ok guys. off to home. take care.. |
| 16:15.48 | RestLessGemini | bye |
| 16:15.49 | jakepdev | bannerman - i could give a hunch. it's probably because most people who use @home are beginners, and although the install is easy, the dialplan included is too large and can be confusing to a beginner |
| 16:16.42 | Jas_Williams | Also the main experianced people here do not use @home and so cannot support ti |
| 16:16.51 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:16.59 | bannerman | Decent as in it's super easy to configure to the point that it functions, and much easier to track down what you've got to do manually in the config to get what you want if there's already something there |
| 16:17.01 | jakepdev | breaks cvs? i installed from cvs after using @home just fine |
| 16:17.04 | bannerman | the sample files that come with asterisk are horrible |
| 16:17.29 | mogorman | lol |
| 16:17.43 | eKo1 | the sample files are only to be used as a reference. |
| 16:17.48 | *** join/#asterisk TEKjacob (~chris@c2.efb7d1.client.atlantech.net) |
| 16:17.55 | bannerman | yeah, but there isn't much in them to refer to |
| 16:17.56 | mogorman | you really dont know what you are talking about |
| 16:18.08 | bannerman | mogorman: I know exactly what I'm talking abut |
| 16:18.12 | mogorman | lol |
| 16:18.14 | bannerman | mogorman: I'm the newb that has used both. |
| 16:18.14 | mogorman | okay |
| 16:18.16 | eKo1 | but I do agree, the word 'sample' is badly used. |
| 16:18.17 | mogorman | if you say so |
| 16:18.22 | tessier | I think the sample files are pretty good |
| 16:18.26 | tessier | I learned a lot from them |
| 16:18.28 | bannerman | mogorman: If anyone in the channel has the right to speak about this, I do. |
| 16:18.43 | malcolmd | that's a pretty high horse |
| 16:18.48 | mogorman | well as an asterisk "pro" i would say asterisk@home is bad |
| 16:18.51 | bannerman | malcolmd: A low horse, you mean. |
| 16:18.55 | TEKjacob | drumkilla: Thanks for the help the other night... Turns out the provider had the T1 prov. but didn't have the PRI up.... |
| 16:19.18 | jakepdev | mogorman - you haven't given any real reason why it is bad, have you? |
| 16:19.32 | mogorman | i work in support for digium |
| 16:19.34 | bannerman | mogorman: If you're talking about what to use for your VoIP provider, sure dude. But if you're talking about a sysadmin who doesn't have prior experience with asterisk, this gives people a place to start from. |
| 16:19.38 | jakepdev | and? |
| 16:19.45 | mogorman | we get more calls on asterisk@home screwing up |
| 16:19.49 | mogorman | than anything else |
| 16:19.57 | jakepdev | name one? |
| 16:20.00 | *** join/#asterisk jeffik (jefik@69.158.7.118) |
| 16:20.18 | jakepdev | not a challenge :P |
| 16:20.18 | mogorman | the zaptel source is different |
| 16:20.24 | mogorman | than standard cvs |
| 16:20.25 | mogorman | do a diff |
| 16:20.37 | jakepdev | couldn't you just reinstall * from cvs? |
| 16:20.45 | mogorman | which is what we do |
| 16:20.49 | jakepdev | that's what I did immediately |
| 16:20.53 | mogorman | but what is the point of haveing it |
| 16:20.58 | mogorman | on there |
| 16:21.01 | jakepdev | well - kind of - but not really |
| 16:21.04 | mogorman | whats the point of using it |
| 16:21.07 | mogorman | if you are just reinstalling on top |
| 16:21.12 | jakepdev | because all the other dependencies |
| 16:21.20 | Essobi | you commercially deploy *@home? |
| 16:21.30 | mogorman | all the other deps. there are almost no deps for asterisk |
| 16:21.46 | jakepdev | i tried to do an install from debian - all kinds of dependencies |
| 16:21.51 | jakepdev | ncurses, etc |
| 16:22.01 | jakepdev | just gave up |
| 16:22.04 | Essobi | Hehe. |
| 16:22.12 | mogorman | lol |
| 16:22.15 | mogorman | there are like 5 deps |
| 16:22.18 | Essobi | Took me about 15 minutes to get it up on debian. |
| 16:22.18 | mogorman | if that |
| 16:22.25 | jakepdev | caue you knew |
| 16:22.29 | bannerman | Yeah. You guys are already familiar with it. |
| 16:22.30 | mogorman | and most come pre packed with deb to my knowledge |
| 16:22.30 | Essobi | Then I promptly rm -Rfed it. |
| 16:22.31 | mogorman | wiki |
| 16:22.37 | Essobi | bannerman Nope. 1st install ever. |
| 16:22.37 | mogorman | lol |
| 16:22.41 | jakepdev | i read the wiki = step by step |
| 16:22.43 | *** join/#asterisk fugitivo (~ajf@201.255.99.221) |
| 16:22.53 | bannerman | Essobi: You are god. |
| 16:22.55 | fugitivo | hey |
| 16:22.57 | jakepdev | specifically the debian install |
| 16:22.58 | Essobi | *SHRUG* |
| 16:23.02 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:23.09 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 16:23.23 | Essobi | bannerman I just make the geek handsign and flash my FreeBSD tattoo and the linux distros cower in fear of me. |
| 16:23.37 | eKo1 | hehe |
| 16:23.39 | mogorman | <PROTECTED> |
| 16:23.43 | mogorman | that is why |
| 16:23.43 | Seyr | i got DTMF working, but having some timeout issues I believe |
| 16:23.45 | Essobi | I did? |
| 16:23.49 | mogorman | it serves no purpos |
| 16:23.51 | mogorman | no not you |
| 16:23.54 | Essobi | I don't like it because it's a kludge. |
| 16:23.55 | Essobi | :) |
| 16:23.58 | fugitivo | Seyr: what was the problem? |
| 16:24.02 | jakepdev | if they took the asterisk install out of *at home and had the user install from CVS, there would be no issues |
| 16:24.12 | Essobi | The DB design looks like it was made by a monkey.. (no offense if anyone in here wrote it.) |
| 16:24.16 | bannerman | Essobi: must be nice, but I'm here with minimal linux and asterisk experience. when I try to build sometihng and it gives an odd error that doesn't have anything to do with the actual dependancy it takes me awhile to figure it out. |
| 16:24.18 | *** join/#asterisk mgamboa (~mariop@201.138.186.69) |
| 16:24.20 | eKo1 | Bottom line: * > *@home |
| 16:24.20 | mogorman | but then what would the point be |
| 16:24.21 | TEKjacob | While I'm not a big fan of *@home, I like AMP... but even then, you should no the ins and outs of asterisk before you start using it.... To hard to troublehsoot if you don't know the way it _should_ work... |
| 16:24.23 | mogorman | and the config files |
| 16:24.26 | *** part/#asterisk mgamboa (~mariop@201.138.186.69) |
| 16:24.27 | mogorman | ugh all htose .inc |
| 16:24.28 | *** join/#asterisk mgamboa (~mariop@201.138.186.69) |
| 16:24.32 | mogorman | so annoying for editing |
| 16:24.34 | Seyr | fugitivo: when i call in, sometimes it recognizes every digit, sometimes it only catches first 4 digits |
| 16:24.38 | mogorman | so much easier with just one file |
| 16:24.39 | Essobi | bannerman Yea.. We all put in our dues. I was doing that umm. |
| 16:24.44 | *** join/#asterisk FITA1 (~m_ahmed@202.5.145.50) |
| 16:24.46 | Essobi | bannerman 12 years ago.. |
| 16:24.48 | johnnyb | Essobi: you should be apologizing to those of us in here who are monkeys. |
| 16:24.49 | mgamboa | hi all |
| 16:24.54 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 16:24.55 | FITA1 | hi all |
| 16:25.01 | mgamboa | i have a little question about asterisk? |
| 16:25.08 | mogorman | ask away |
| 16:25.10 | bannerman | Essobi: So that's why you think it's bad that there's an easy installation? |
| 16:25.13 | *** join/#asterisk Moc[Toronto] (~mochouina@209.167.66.3) |
| 16:25.13 | fugitivo | Seyr: that problem is using any zap channel? |
| 16:25.15 | Pinhole | mgamboa, you have my personal permission to ask |
| 16:25.23 | Essobi | johnnyb OOHHH OHHH AHHH.. |
| 16:25.24 | Moc[Toronto] | Hi everyone |
| 16:25.25 | mgamboa | asterisk support MFC/R2 in the e1 interface |
| 16:25.30 | mogorman | yes |
| 16:25.37 | mgamboa | is a plugin |
| 16:25.44 | Seyr | fugitivo: no clue what zap is. i only have a couple channels defined |
| 16:25.46 | mgamboa | or is a configuration i need to do |
| 16:25.51 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 16:26.06 | Essobi | bannerman Easy? I just don't like the Web Interface, or the other shit in it.. |
| 16:26.08 | Seyr | actually, only have 3 extensions and 1 s defined in default |
| 16:26.10 | mogorman | go to soft-switch.org |
| 16:26.10 | FITA1 | any One who can tell me about trunking |
| 16:26.10 | johnnyb | Essobi: You know my language! |
| 16:26.17 | phpboy | is there a default recording that says something in the lines of "If you know you parties extention, please dial it now" ? |
| 16:26.18 | mgamboa | ok |
| 16:26.24 | mgamboa | thanks |
| 16:26.25 | johnnyb | Essobi: oh, to hear the native tongue |
| 16:26.28 | Essobi | I'm all about Mini-distros with *... there should be a CVS upgrade path on them thou.. |
| 16:26.30 | mogorman | go into asterisk-sounds |
| 16:26.32 | fugitivo | Seyr: then you're not using zap, because if you're using zap, maybe you have the gain too low, and dtmf will not work correctly |
| 16:26.37 | mogorman | and read souunds.txt |
| 16:26.46 | FITA1 | I want to use different codec for client to asterisk and asterisk to asterisk |
| 16:27.20 | FITA1 | that is client ---- > asterisk gsm and asterisk ----> asteisk ilbc |
| 16:27.35 | Essobi | FITA1 You get no RTP re-invites with transcoding. |
| 16:27.37 | bannerman | Essobi: I do. I'm as comfortable with editing text configs as the next person, but it's tough to build a dialplan and things from the documentation that comes with asterisk and what you find online. I'm speaking as the authority on learning new here; you should listen to me, experts. |
| 16:28.03 | Seyr | fugitivo: Asterisk has to pass DTMF off to the end point, is there any timeout settings that would interfere with that? |
| 16:28.03 | pgpkeys | gsm <-> IAX2 -- IAX2 <-> iLBC |
| 16:28.22 | Essobi | The docs sucks, we all know that. |
| 16:28.29 | TEKjacob | bannerman: The problem is... you will never be comfotable with building the dialplan if you just rely on *@home... |
| 16:28.33 | bannerman | Essobi: asterisk@home filled in a lot of gaps for me. |
| 16:28.37 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 16:28.58 | fugitivo | you'll never learn anything that runs under linux if you don't touch the text files |
| 16:29.05 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 16:29.06 | bannerman | I didn't say I didn't touch the text files |
| 16:29.06 | Pinhole | phpboy, look through sounds-extra.txt in asterisk-sounds package. |
| 16:29.14 | Essobi | Take the time to learn it yourself.. you'll be a lot happier, especially when *@home gets abandoned and you have no upgrade path anymore. |
| 16:29.19 | FITA1 | Essobi: plz descibe what u said, I didn't get you |
| 16:29.28 | Essobi | FITA1 What what? |
| 16:29.29 | outtolunc | ~seen jerjer |
| 16:29.30 | jbot | jerjer <~JerJer@12.173.204.122> was last seen on IRC in channel #asterisk, 7h 22s ago, saying: 'Dial,SIP/peer'. |
| 16:29.32 | fugitivo | Seyr: mmm |
| 16:29.37 | bannerman | but I'm sure happy to let asterisk@home do most of the work for me, so I can look and see how you guys do all these poorly documented things in the first place |
| 16:29.44 | jonathh | hey gents |
| 16:29.49 | Essobi | FITA1 Your phone won't talk directly to the 2nd * box. |
| 16:29.52 | *** join/#asterisk felipeao (~felipeao@berg.viaip.com.br) |
| 16:29.57 | jonathh | anyone got an help to offer regarding mpg123 |
| 16:30.04 | jonathh | to peeps use it? or is there an alternative? |
| 16:30.07 | mgamboa | the use of trunk in asterisk in this case iax is for comunicate pbx to pbx? |
| 16:30.07 | jonathh | mpg321? |
| 16:30.12 | Essobi | Fita If you change codecs anywhere.. it has to bridge them.. |
| 16:30.18 | mgamboa | asterisk of course |
| 16:30.36 | FITA1 | Essobi: and how can I do that |
| 16:30.36 | Essobi | fita Otherwise, RTP (your voice stream) goes from endpoint to endpoint and not through your * box. |
| 16:30.51 | FITA1 | ok |
| 16:30.53 | heison | mgamboa: yes, it's header compression that is only benefitial between 2 hosts |
| 16:31.04 | vaewyn | Dialplans are hard to document because every country... city state... business does it WAY different |
| 16:31.12 | Essobi | FITA1 Change the codec priority associated with the peer you're calling. |
| 16:31.15 | mgamboa | the h.323 codec is include with asterisk or need to download the openh323 |
| 16:31.24 | Essobi | FITA1 /etc/asterisk/whatever |
| 16:31.27 | FITA1 | so it is not possible to use different codecs for iax2 ---> asterisk and asterisk --->iax2 |
| 16:31.38 | FITA1 | communication |
| 16:31.38 | bannerman | vaewyn: sure, otherwise it would come preconfigured with a dialplan that worked for everyone |
| 16:31.41 | Seyr | fugitivo: think maybe its BroadVoice not passing DTMF correctly? I mean, I got it to work..... it just doesnt work 100%, so either its a timeout issue of some sort, or its BroadVoice |
| 16:31.46 | Essobi | Sure you can.. you have to specify a different codec for the peer |
| 16:32.02 | Essobi | mgamboa H323 support is currently very sketchy still. |
| 16:32.04 | vaewyn | FITA1: don't use 'friend' and it is very easy :} |
| 16:32.12 | Essobi | And it's not a codec. ;) |
| 16:32.12 | fugitivo | Seyr: you said that using xlite without *, worked fine |
| 16:32.20 | TEKjacob | Anyone have voicemail set up so that during the outbound message the user can press * and log into VM? |
| 16:32.22 | bannerman | vaewyn: The problem is, the features aren't that well documented. And many things require little tricks.. I don't have an example right off the hand, but there are a lot of things unique to asterisk that aren't documented that you only learn with a lot of fiddling. |
| 16:32.34 | Essobi | Umm. |
| 16:32.38 | Essobi | It's called RTFS. |
| 16:32.39 | Essobi | :) |
| 16:32.42 | Seyr | fugitivo: yeh, but I got X-lite to work with Asterisk as well |
| 16:32.43 | mogorman | lol |
| 16:32.46 | bannerman | Essobi: lol. |
| 16:32.49 | Essobi | I do it everyday. |
| 16:32.50 | mogorman | well we should be better at documenting |
| 16:32.54 | Seyr | one sec, let me double check |
| 16:32.54 | Essobi | Sadly, I'm serious. |
| 16:32.55 | bannerman | Essobi: Good for you. |
| 16:33.03 | vaewyn | doesn't even have to be rtfs... 'show application (blah)' |
| 16:33.08 | TEKjacob | seems it should be relativly simple.... |
| 16:33.34 | bannerman | Essobi: I'd love to learn C, but unfortunately I have a full time job that doesn't revolve solely around maintaining asterisk. |
| 16:33.50 | Seyr | fugitivo: yeh, x-lite works fine going through asterisk now as well |
| 16:33.57 | Seyr | hmm... |
| 16:34.00 | Essobi | No offense to blitzrage or anything, but we do need better docs, and more docs... a big problem is * is a moving target. |
| 16:34.05 | Seyr | its gotta be BroadVoice |
| 16:34.18 | phpboy | who's go a nice doc on setting up an auto attendant |
| 16:34.19 | phpboy | ? |
| 16:34.21 | vaewyn | Essobi: 'moving target' hehehe understatement of the year :} |
| 16:34.29 | FITA1 | <Essobi> : I m unable to get you |
| 16:34.31 | Essobi | Evading target? |
| 16:34.35 | fugitivo | what's the problem with the actual docs? :) |
| 16:34.37 | pgpkeys | Essobi: then put documentation into the source and use doxygen or one of the other doc creators to make docs fromt he source :) |
| 16:34.42 | Essobi | Mutating-polymorphic target? |
| 16:34.56 | vaewyn | pgpkeys: have stuff like that for the API... still sucks :} |
| 16:35.03 | pgpkeys | hehe |
| 16:35.09 | *** join/#asterisk Skarmeth (~Skarmeth@201009023158.user.veloxzone.com.br) |
| 16:35.11 | Skarmeth | hi all |
| 16:35.13 | Seyr | Anyone know any good VoIP providers that work well with Asterisk and has no problems with DTMF?? :-) |
| 16:35.22 | Essobi | pgpkeys PFFFT, who wants to see docs in their code.. that's fine for documenting a API, but user docs.. that's silly. |
| 16:35.33 | Pinhole | So, when you figure it out, simply add it to the wiki. Then it's documented. |
| 16:35.39 | pgpkeys | Essobi: it was a joke. i figured the /me ducks would clue you |
| 16:35.49 | pgpkeys | obviously not i see :) |
| 16:35.52 | Essobi | I was typing my rant before you ducked. |
| 16:35.53 | Essobi | :) |
| 16:35.54 | *** join/#asterisk darwin35 (~darwin35@24.3.226.147) |
| 16:35.56 | pgpkeys | hehe |
| 16:36.04 | pgpkeys | so that's what the splatter was |
| 16:36.11 | mgamboa | the question is because i can to build the amportal and only show me sip iax and zap the h323 doesn't appear i need to do something via shell or were i can configurate the protocol h323 |
| 16:36.13 | *** join/#asterisk binhex (~bob@216.31.167.125) |
| 16:36.16 | Essobi | That wiki sucks too.. it's a terribly wiki. |
| 16:36.21 | *** join/#asterisk PBXtech (~nik@001-711-202.area1.spcsdns.net) |
| 16:36.32 | Essobi | Not that the content sucks.. the wiki is just... |
| 16:36.33 | Pinhole | I agree, but its about all we got. |
| 16:36.36 | Essobi | sketchy.. |
| 16:36.51 | Essobi | I wish I could volunteer some bandwidth. |
| 16:36.52 | Seyr | honestly, being new and all... i think the wiki for Asterisk is a LOT better than some I have seen |
| 16:37.06 | Skarmeth | I'm trying to set up a Asterisk server for a home test lab and I'm in troubles with FXO interfaces (a X100P clone)... I'm searching for docs about setting zap channels but without sucess |
| 16:37.15 | Essobi | Seyr God you havn't tried seraching for the word, Dial have you? :) |
| 16:37.19 | pgpkeys | i like the handbook but it still needs work. i'm not exactly an idiot when it comes to learning new things and I'm having difficulty setting ip dialplans myself. |
| 16:37.27 | fugitivo | i don't understand people complaining about the docs |
| 16:37.30 | bannerman | parking.conf or features.conf? |
| 16:37.30 | Seyr | lol Essobi |
| 16:37.32 | pgpkeys | i can see the argument but i'm still too new to asterisk to say anything more than that |
| 16:37.50 | FITA1 | Essobi: let me define you my problem again, I am using iax2 client on both ends. and want to use gsm format for client1 to server1 and client2 to sever2 and ilbc for server1 to server2. is it possible?? |
| 16:37.55 | bkw_ | Cluecon |
| 16:37.58 | Essobi | phpboy I guess I got it easy.. the whole idea of dialplan partitions is native to me coming in with a VoIP softswitch background. |
| 16:38.08 | bkw_ | Everyone needs to come to cluecon Aug 3rd - 5th |
| 16:38.12 | fugitivo | there is a lot of documentation, and an IRC channel, what else do they want? |
| 16:38.13 | johnnyb | pgpkeys: Until I figured out that extensions were scripts and not devices, I had trouble with dialplans. |
| 16:38.22 | mgamboa | i have a real question asterisk can do some or all feature like avaya definity |
| 16:38.25 | Essobi | FITA1 Yea. |
| 16:38.26 | mgamboa | or is limit |
| 16:38.34 | binhex | dose ne1 know if there is a limit in sip.conf? I seem to be able to add only about 60 or so peers |
| 16:38.42 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || ClueCon Dev Conf Aug 3rd - 5th |
| 16:38.43 | FITA1 | <Essobi>: How |
| 16:38.51 | johnnyb | mgamboa: like what? |
| 16:38.57 | Essobi | Server1 need to have a IAX peer pointed to Server2.. "disallow=all" "allow=ilbc" on tyhat peer.. |
| 16:39.13 | Essobi | IAX does some kind of reinviting too.. you might have to explicitly turn that off. |
| 16:39.16 | Seyr | fugitivo: any voip providers you could suggest instead of BroadVoice? |
| 16:39.24 | mgamboa | i don't know like hospitality feauture |
| 16:39.24 | bkw_ | Seyr, asterlink.com |
| 16:39.29 | Essobi | Seyr Me in about a year. :) |
| 16:39.33 | Seyr | thanks |
| 16:39.40 | Seyr | Essobi: ill call ya in a year then ;) |
| 16:39.41 | Essobi | Hi Brian! |
| 16:39.55 | fugitivo | Seyr: no, sorry :( |
| 16:40.20 | Seyr | kk, im gonna go get this thing running.... been working on it for days ;( |
| 16:40.33 | Seyr | thanks for the help again |
| 16:40.45 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 16:40.52 | bannerman | fugitivo: As a for instance, one of the things I ran into was the fact that I had um.. bandwidth=high or something in my iax.conf, and it wasn't until I came here and overheard someone else's question that I realized it was keeping me from allowing only certain protocols. |
| 16:41.00 | PBXtech | Avaya/definity is a better call center application |
| 16:41.29 | bannerman | fugitivo: So many little things that are very difficult to track down unless you know exactly what you're looking for (in which case you probably know the answer as well) |
| 16:42.10 | pgpkeys | and relying on an irc channel for the information isn't exactly a good thing when your setting something like this up for a business. |
| 16:42.13 | PBXtech | :) |
| 16:42.24 | Essobi | RTFS! |
| 16:42.31 | pgpkeys | like anything else, docs are always lacking. it's the nature of the game. |
| 16:42.33 | Pinhole | Google is better than IRC because it never tells you to rtfm. Unfortunately it is not as smart as all of you. |
| 16:42.39 | Essobi | Better yet.. contribute to the code. |
| 16:42.40 | PBXtech | would be cool if * did look ahead call forwarding in call center |
| 16:42.48 | fugitivo | bannerman: i think that working for years with linux teachs you to work that way |
| 16:42.49 | pgpkeys | Essobi: not everyone is a coder. |
| 16:43.00 | Essobi | pgpkeys well the should be. |
| 16:43.01 | pgpkeys | i can tell you're a coder, that's a coder comment ;) |
| 16:43.04 | pgpkeys | hehehe |
| 16:43.08 | *** join/#asterisk brettnem (~mive29@user-0ccsr10.cable.mindspring.com) |
| 16:43.16 | Pinhole | what is look ahead call forwarding? |
| 16:43.29 | Essobi | I was just thinking the same thing. |
| 16:43.30 | *** join/#asterisk __a (user@193.140.215.2) |
| 16:43.38 | Essobi | Save, I had a WTF in there.. |
| 16:43.40 | PBXtech | look ahead to another * system for queue info |
| 16:43.41 | bannerman | fugitivo: probably so |
| 16:43.47 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 16:43.53 | Essobi | PBXtech What what? |
| 16:44.05 | Essobi | PBXtech You mean like Queue spillover? |
| 16:44.17 | pgpkeys | fugitivo: I've been using linux for roughly 10 years, and sorry to say, for admins, that's great, but it's still not a panacea to good documentation. |
| 16:44.20 | Essobi | into anotother queue/vm? |
| 16:44.29 | heison | ~seen moc[toronto] |
| 16:44.31 | jbot | moc[toronto] is currently on #asterisk (19m 18s). Has said a total of 1 messages. Is idling for 19m 7s |
| 16:44.31 | Pinhole | You should be able to write an agi that probes the queue on the second machine via the manager interface. |
| 16:44.36 | bannerman | fugitivo: However, in this case, there's a way around a whole lot of the problems.. there's a packagae that assists you in setting up a working dialplan, and reduces the amount of "oh, duh" questions by a lot. |
| 16:44.39 | PBXtech | if my skill if pegged i want to look for an alternative queue on another * box and reroute |
| 16:44.52 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 16:44.53 | pgpkeys | bannerman: PLEASE don't say amp |
| 16:45.01 | Essobi | God I hate AMP. |
| 16:45.11 | darwin35 | ~amp |
| 16:45.12 | jbot | it has been said that amp is an Audio MPEG Player. [non-free], or http://amp.coalescentsystems.ca/ |
| 16:45.15 | PBXtech | call center reporting isnt much in * |
| 16:45.21 | Essobi | Not that AMP.. the other AMP |
| 16:45.24 | pgpkeys | hehe |
| 16:45.25 | *** join/#asterisk switch (~switch@61.206.115.5.user.ad.il24.net) |
| 16:45.25 | *** join/#asterisk DannyF (~dannyf@h3n9c1o848.bredband.skanova.com) |
| 16:45.33 | bannerman | lol. So, tell me what's wrong with amp. |
| 16:45.33 | pgpkeys | Essobi: misfire! :) |
| 16:45.35 | binhex | dose ne1 know if there is a limit in sip.conf? I seem to be able to add only about 60 or so peers |
| 16:45.35 | fugitivo | pgpkeys: that's what i'm talking about, you learn to work with that little problem |
| 16:45.47 | Essobi | PBXtech You can do, what you're talking about. |
| 16:45.47 | Pinhole | AMP = apache mysql php? |
| 16:45.50 | bannerman | And also, tell me why my outbound audio fades halfway through every long call! |
| 16:45.52 | Essobi | No. |
| 16:45.55 | Pinhole | just not necessarily on linux |
| 16:45.57 | PBXtech | how |
| 16:45.57 | Essobi | Asterisk MAnagement Portal |
| 16:46.04 | Essobi | IT's a cruft kludge. |
| 16:46.07 | pgpkeys | basicly a webmin for asterisk |
| 16:46.12 | *** part/#asterisk genuix (~genuix@sobek.7g0.net) |
| 16:46.30 | Essobi | pgpkeys webmin is bvetter designed then AMP. |
| 16:46.32 | PBXtech | you talking about just forwarding the call? all i want to do is query a far end skill and make a determination |
| 16:46.35 | pgpkeys | true |
| 16:46.44 | Essobi | "skill"? |
| 16:46.48 | Essobi | WTF is he talking about? |
| 16:46.55 | pgpkeys | however it's a similar functionality so figured i'd use webmin to describe the 'type' of thing amp is |
| 16:47.00 | PBXtech | skill/hunt group/queue whatever |
| 16:47.04 | felipeao | quit |
| 16:47.18 | Essobi | Oh umm. Hmm.. |
| 16:47.23 | Grooby | has anyone here uses DUNDi? |
| 16:47.44 | Essobi | PBXtech Yea.. write an AGI, or if it's high volume write an app. |
| 16:48.04 | pgpkeys | Essobi: tech support call center queue. usually called a skill since the call escalation is based on a skill level for a call. |
| 16:48.09 | Essobi | Hit the management port, query the queue for whatever.. and route based on the decision. |
| 16:49.06 | TEKjacob | Essobi: The problem is usgin vendor specicfic terms confuses everyone... Skills, Look Ahead Routing, etc... |
| 16:49.34 | TEKjacob | I think you can do exactly the same thing w/ asterisk... you just don't have the slick marketing guy to sell it to you... |
| 16:49.36 | Essobi | Right.. It's called a queue. |
| 16:50.16 | brettnem | have you seen ICD? |
| 16:50.50 | bkw_ | ICD RULES!!!!!! |
| 16:50.56 | Essobi | Briefly. |
| 16:51.03 | bkw_ | ICD is done the right way... mkay |
| 16:51.04 | Essobi | We know you work on it. |
| 16:51.09 | cypromis | he does ? |
| 16:51.15 | bkw_ | since when? |
| 16:51.21 | PBXtech | what is ICD? |
| 16:51.26 | Essobi | Oh.. Who was it that workd on ICD? |
| 16:51.26 | brettnem | Asterisk ICD (Intelligent Call Distribution) is an advanced mechanism for handling queues, conferences and agents and provides a unified API for interfacing to external dialer systems. |
| 16:51.28 | Essobi | worked. |
| 16:51.32 | Essobi | cypromis ? |
| 16:51.33 | brettnem | http://www.voip-info.org/tiki-index.php?page=ICD |
| 16:51.49 | Essobi | It's a #asterisk* regular. |
| 16:52.07 | Essobi | Mehe. |
| 16:52.11 | bkw_ | brettnem, anthm, cypromis and tclark are the ones to thank for it |
| 16:52.22 | rephorm | anyone know if there's a way to assign a pci slot an irq through linux (the bios is crappy and won't let me) |
| 16:52.28 | Essobi | :) |
| 16:52.33 | brettnem | heh, I'm on that list? |
| 16:52.48 | bkw_ | no that was directed at you |
| 16:52.49 | bkw_ | :P |
| 16:52.52 | brettnem | rephorm: try an alternative manufacturer for your card |
| 16:52.57 | Essobi | bkw_ Read my PM.. I don't want to spill the beans in here. :) |
| 16:53.00 | brettnem | haha ah.. ok.. :) please PLEASE don't thank me |
| 16:53.30 | brettnem | bkw_: I got SER+SEMS working as a conference bridge.... |
| 16:53.41 | brettnem | no beans allowed in chat |
| 16:53.54 | Essobi | :) |
| 16:54.13 | brettnem | I'm finding joy in asterisk alternatives |
| 16:54.15 | Pinhole | Says on slashdot that M$ patented 911, ... er something like that... |
| 16:54.16 | brettnem | I hate to say.. |
| 16:54.21 | brettnem | what? |
| 16:54.28 | *** join/#asterisk mw` (~michael@omega.gc-schwartz.de) |
| 16:54.29 | mogorman | what brettnem? |
| 16:54.59 | brettnem | heh |
| 16:55.07 | brettnem | well asterisk just seems so bloated.. |
| 16:55.15 | jonathh | hey guys.. if i have a context for incoming sip calls... how do i route that call to an extention that plays a message.. say? |
| 16:55.16 | mogorman | what why? |
| 16:55.17 | brettnem | I'm starting to lose faith |
| 16:55.27 | Pinhole | rm -rf /root/src/asterisk* not bloated any more |
| 16:55.37 | mogorman | why is it bloated? |
| 16:55.44 | brettnem | Pinhole: don't forget /usr/bin/asterisk |
| 16:55.58 | brettnem | it's got so much crap built into it. |
| 16:56.00 | Pinhole | rm -rf /, there, got it all |
| 16:56.05 | brettnem | there ya go |
| 16:56.07 | *** part/#asterisk binhex (~bob@216.31.167.125) |
| 16:56.09 | mogorman | apps dont load |
| 16:56.13 | mogorman | they arent bloat |
| 16:56.19 | mogorman | asterisk as it self doesnt do much |
| 16:56.26 | brettnem | it crashes |
| 16:56.29 | brettnem | :) |
| 16:56.31 | PBXtech | [brettnem]: why use SEMS instead of * ? |
| 16:56.33 | mogorman | lol |
| 16:56.33 | Pinhole | dd -someparameters /dev/null /hda |
| 16:56.33 | mogorman | sure... |
| 16:56.39 | brettnem | and the digium cards take over the box |
| 16:56.45 | Pinhole | maybe /dev/rand |
| 16:56.46 | mogorman | ? |
| 16:56.55 | brettnem | PBXtech: just for some functions.. like a ISDN GW |
| 16:57.06 | Pinhole | killall -9 asterisk |
| 16:57.13 | brettnem | FORMAT C: |
| 16:57.18 | Pinhole | shutdown -h now |
| 16:57.26 | Pinhole | reboot |
| 16:57.32 | Pinhole | poweroff |
| 16:57.42 | *** join/#asterisk lohelle (~post@213.161.252.253) |
| 16:57.51 | pgpkeys | magnet to the head |
| 16:57.52 | Pinhole | Let's fix the bloat! rewrite it in java! |
| 16:57.59 | jonathh | LMAO |
| 16:58.00 | pgpkeys | Pinhole: that was just bad |
| 16:58.15 | brettnem | really.. I get too high a count of crashes, lockup, hung channels, ast_walk_deadlock_avoided. whatever.. it's really starting to annoy me |
| 16:58.20 | pgpkeys | you sick sick person you :) |
| 16:58.22 | Pinhole | Or, maybe we'll make the source more readable so people can rtfs. rewrite it in perl. |
| 16:58.34 | brettnem | how about cobol? |
| 16:58.40 | pgpkeys | fortran |
| 16:58.42 | cypromis | algol |
| 16:58.43 | TEKjacob | fortan |
| 16:58.44 | pgpkeys | my wife could help with that |
| 16:58.46 | Pinhole | Or, to be really compsci, lisp |
| 16:58.47 | TEKjacob | tran |
| 16:58.56 | brettnem | or haskell |
| 16:58.59 | pgpkeys | well ex wife |
| 16:59.05 | Pinhole | English! |
| 16:59.15 | lohelle | hello! I have a question.. How do I add asterisk-addons to bristuff (junghanns.net)..? I would like to test mysql support, but I'm not using "standard" asterisk 1.07 (using bristuffed version 1.07) |
| 16:59.16 | TEKjacob | dropped her after the y2k rush? |
| 16:59.16 | brettnem | english? what is that? some high level language I bet |
| 16:59.40 | brettnem | mogorman: really I like asterisk, but in production, it's caused too many problems I think.. |
| 16:59.41 | pgpkeys | TEKjacob: hehe |
| 16:59.47 | pgpkeys | I needed a newer model |
| 16:59.49 | Pinhole | Let's make it crash faster! 80386 asm |
| 17:00.24 | mogorman | stable? |
| 17:00.26 | mogorman | or cvs |
| 17:00.34 | brettnem | yeah, "stable" |
| 17:00.37 | pgpkeys | TEKjacob: funny things about Wife units after the 3rd year of marriage. They develop this bug called Nagging |
| 17:00.37 | jonathh | hey guys.. if i have a context for incoming sip calls... how do i route that call to an extention that plays a message.. say? |
| 17:00.45 | pgpkeys | it's caused many a Man system to crash |
| 17:00.51 | mogorman | what bloat does stable |
| 17:00.53 | jonathh | the extension is in a differnt context... do i goto? or dial |
| 17:00.57 | Nugget | goto. |
| 17:01.08 | brettnem | that whole dialplan thing |
| 17:01.10 | Pinhole | my wife hasn't developed that bug yet and it's been 5 years. I must have got a better model. |
| 17:01.15 | mogorman | wtf |
| 17:01.17 | jonathh | is it clean to goto a context from another incoming one? |
| 17:01.23 | Nugget | yes |
| 17:01.23 | mogorman | man brettnem you are making 0 sense |
| 17:01.28 | Pinhole | oh, gotta go, I need to mow the lawn... |
| 17:01.30 | pgpkeys | Pinhole: be glad. it's a common affliction |
| 17:01.37 | pgpkeys | Pinhole: hahaha |
| 17:01.42 | jonathh | can you mow mine too? BBQ on sunday |
| 17:01.47 | brettnem | mogorman: if I simply want to take PRI and convert to sip, it's a lot of overhead |
| 17:02.16 | *** part/#asterisk __a (user@193.140.215.2) |
| 17:02.17 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 17:02.53 | TEKjacob | brettnem: you think? My machines sit mostly idle while users chat away on sip phones... |
| 17:02.55 | mogorman | not really |
| 17:02.55 | brettnem | mogorman,bkw: twice I've had CDR modules totally disable my box. Yes even with cdr_odbc |
| 17:02.59 | pgpkeys | if i wasn't so sick today i'd bbq myself |
| 17:03.10 | brettnem | TEKjacob: I want to do it with LOTS of PRIs |
| 17:03.19 | mogorman | how many calls are you pushing, that you can cause that |
| 17:03.24 | *** join/#asterisk nd^ (~nd@66-188-116-53.mad.wi.charter.com) |
| 17:03.24 | *** join/#asterisk point (1000@213.27.44.55) |
| 17:03.29 | mogorman | im sure mark would be interested in this |
| 17:03.32 | brettnem | well right now, not more than 20-30 at a time |
| 17:03.44 | TEKjacob | brettnem: how many is LOTS |
| 17:03.54 | brettnem | TEKjacob: hundreds |
| 17:04.00 | brettnem | of course, distributed |
| 17:04.03 | brettnem | somehow |
| 17:04.13 | brettnem | or not if we can get DSPs onto the hardware |
| 17:04.25 | Gand_DJ | Is there a channel for amp? Curious how to setup prefered codec order |
| 17:04.25 | brettnem | CT bus.. something a bit fancier |
| 17:04.48 | TEKjacob | Gand_DJ: #Amportal |
| 17:04.57 | TEKjacob | or #amportal |
| 17:05.07 | brettnem | I'm working with the SS7 guys.. but I'm still worried about implementing asterisk in production in my network. |
| 17:05.54 | nd^ | i have an T1/E1 card that i wish to write a linux driver for, and would like it to be possible to use it with asterisk (with ISDN PRI probably) -- any advice on the best way to do this (a capi driver? try to emulate a zaptel device?) |
| 17:06.09 | Gand_DJ | What's the difference between ulaw and alaw for codec? |
| 17:06.19 | brettnem | couple of bits |
| 17:06.39 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 17:07.18 | eKo1 | actually, if you're going through the pstn, you need to use ulaw if you're in the americas and alaw if you're in europe. |
| 17:07.30 | eKo1 | or so the saying goes |
| 17:07.55 | blitzrage | ulaw is better |
| 17:07.57 | blitzrage | :D |
| 17:07.58 | Pinhole | Gand_DJ, I believe it is Atlantic disposition |
| 17:08.20 | eKo1 | ~ulaw |
| 17:08.21 | blitzrage | note: ulaw and alaw are basically the same |
| 17:08.43 | Gand_DJ | ok. If you don't assign an allow or disallow for an extension, does it default to using the allow list from the General portion of sip.conf |
| 17:10.35 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 17:10.38 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 17:10.55 | vaewyn | Gand_DJ: yes |
| 17:10.59 | durex | hello asterisks... how should i accept incoming calls from Free World DialUp (FWD)? |
| 17:11.57 | *** join/#asterisk UBiQUiTY (~mike@68.160.103.76) |
| 17:12.07 | durex | Now I can dial-out, but anyone knows how to accept incoming calls? |
| 17:12.19 | Nugget | the FWD website knows how. |
| 17:12.21 | Gand_DJ | Do you get a busy signal? |
| 17:12.37 | Gand_DJ | few others get a busy signal if someone calls their * box... like me |
| 17:12.42 | Gand_DJ | calling out is fine though |
| 17:12.47 | Gand_DJ | I think it's issue on fwd end |
| 17:12.47 | UBiQUiTY | DUREX: to accept incoming calls. take a look inside extensions.conf |
| 17:13.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 17:13.48 | durex | UBiQUiTY thank u... let me see... |
| 17:13.57 | durex | Nugget i didn't found anything in www.freeworlddialup.com |
| 17:14.28 | *** join/#asterisk Barmal (~1@m615e36d0.tmodns.net) |
| 17:14.36 | UBiQUiTY | DUREX: ok... another hint.... u might wanna skip down to the [default] section to get yourself started... of course u should really read all the comments before that as well |
| 17:14.46 | Nugget | http://www.google.com/search?q=free+world+dial+up+asterisk has quite a number of useful pages. the number one hit is exactly what you asked, in fact. |
| 17:15.24 | Barmal | hello guys |
| 17:15.32 | Pinhole | and gurls? |
| 17:16.33 | durex | Nugget no... what is the URL that appears for u? I'm in other part of the world than u... and google... do u know... |
| 17:16.50 | Nugget | I'm not here to teach you how to use a search engine. |
| 17:17.14 | UBiQUiTY | ok i have a wierd situation, maybe one of you may know.... i'm trying to run a sound quality test on my hardware... I'm using one of the 4-port zaptel boards... I would like to dial out using the zaptel span 2 and receive the call using span 3 ... I have span 2 set up as pri_net and span 3 set up as pri_cpe , but i am getting the red alarm on both channels (span 1 is okay, hooked into a real pri) ... am i supposed to use a special cable for this? |
| 17:17.24 | Nugget | I know you're in brazil, but that doesn't mean you don't have to expend a little energy to solve your own problem. |
| 17:17.54 | Nugget | we're not here to tell you what to type. google, try, if you have specific questions feel free to ask. |
| 17:17.55 | durex | Nugget I think google doesn't give me the same results that it give to u |
| 17:18.06 | Nugget | I think that you shouldn't be thwarted by that. |
| 17:18.25 | Nugget | ~docs |
| 17:18.26 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 17:18.31 | Nugget | ^ start there |
| 17:18.48 | jakepdev | UBiQUiTY - i would think you'd want pri_cpe for both spans if they're both connected to the telco |
| 17:18.56 | UBiQUiTY | no |
| 17:19.12 | UBiQUiTY | i have span 3 connected to span 2 via an rj-45 cable |
| 17:19.17 | jakepdev | why :) |
| 17:19.19 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 17:19.26 | UBiQUiTY | i am trying to dial from 3 directly into 2 |
| 17:19.50 | jakepdev | there surly must be another way to accomplish what you want besides buring 2 spans |
| 17:19.52 | UBiQUiTY | im trying to determine where i am losing sound quality, so i am trying to do this in order to rule out the zaptel board |
| 17:19.54 | jakepdev | burning |
| 17:20.00 | jakepdev | ok |
| 17:20.04 | UBiQUiTY | this is just for a temporary test |
| 17:20.05 | Pinhole | durex, google gives me: http://www.voip-info.org/wiki-Free+World+Dialup |
| 17:20.35 | jakepdev | you might need to xover |
| 17:20.38 | durex | Pinhole thank u |
| 17:20.47 | UBiQUiTY | i tried that and it didnt work |
| 17:20.51 | UBiQUiTY | i will try again |
| 17:20.57 | jakepdev | only two configs :) |
| 17:21.05 | jakepdev | straight or xover |
| 17:22.11 | UBiQUiTY | is a pri xover cable wired the same as an ethernet xover? |
| 17:22.18 | jakepdev | nope |
| 17:22.19 | durex | Pinhole http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD not shows how to do it with incoming calls right.... |
| 17:22.45 | Pinhole | Is there a syntax checker for * config files? I think it would greatly speed up my 1,000,000 monkeys in their quest. |
| 17:22.48 | UBiQUiTY | damn... |
| 17:22.49 | jakepdev | http://www.voip-info.org/wiki-crossover+T1+cable |
| 17:22.56 | UBiQUiTY | ok thank you |
| 17:22.59 | jakepdev | np |
| 17:24.19 | Pinhole | sorry, durex, I only know how to use google. |
| 17:28.26 | jakepdev | UBiQUiTY: I was given some ideas yesterday that might help you... "hdparm" - checks if disk i/o wait is an issue or use "watch cat /proc/" or "loadavg |
| 17:28.29 | jakepdev | or watch "cut -f1-3 -d' ' < /proc/loadavg" |
| 17:28.48 | facek_ | do anybody have a some caller notifier? |
| 17:28.56 | *** join/#asterisk Kal_Zakath (~Kal_Zakat@213.219.184.108.adslpower.by.edpnet.be) |
| 17:28.57 | jakepdev | what? |
| 17:28.58 | Kal_Zakath | hi |
| 17:29.13 | jakepdev | what do you mean by "caller notifier"? |
| 17:29.15 | facek_ | jakepdev i want to get a balloon in windows, when somebody ic calling me |
| 17:29.20 | jakepdev | oh |
| 17:29.28 | jakepdev | can you write in vb or c#? |
| 17:29.30 | Kal_Zakath | I look for a cool iax linux softphone, does anybody has an idea ? |
| 17:29.37 | DrWho17 | facek_: the wiki has several listed |
| 17:30.13 | facek_ | jakepdev in c# |
| 17:30.18 | facek_ | i am writing now.. but i have little problem |
| 17:30.23 | facek_ | use the manager port is the best way? |
| 17:30.24 | jakepdev | how little? |
| 17:30.41 | jakepdev | i'd suggest FastAGI - but manager might work for you |
| 17:30.51 | facek_ | FastAGI |
| 17:30.59 | jakepdev | ~FastAGI |
| 17:31.24 | Kal_Zakath | no idea for a cool iax linux softphone ? |
| 17:31.40 | UBiQUiTY | JAKEPDEV: thanks for those ideas above... i was away for a minute... |
| 17:31.43 | facek_ | jakepdev problem is.. |
| 17:31.45 | bjohnson | cool? |
| 17:31.50 | facek_ | i can get who is calling |
| 17:31.56 | jakepdev | jbot, FastAGI is Implements (agi://) over TCP socket using port #4573 by default. |
| 17:31.57 | jbot | okay, jakepdev |
| 17:31.57 | bjohnson | isn't a linux iax softphone the essence of cool? |
| 17:32.05 | facek_ | but i dont know how to connect two information who and to who |
| 17:32.05 | jakepdev | jbot, FastAGI implements (agi://) over TCP socket using port #4573 by default. |
| 17:32.27 | UBiQUiTY | JAKEPDEV: looks like im gonna have to crimp my own crossover cable, if thats what's needed. any idea how i can tell for sure if i need to use xover to do this? |
| 17:32.43 | jakepdev | UBiQUiTY: i did |
| 17:33.05 | jakepdev | facek_: yes you can get any * variable |
| 17:33.12 | UBiQUiTY | oh, so ur saying that I definitely need the xover? |
| 17:33.16 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 17:33.29 | jakepdev | i needed it to connect * to another switch |
| 17:33.36 | UBiQUiTY | hmmmmm |
| 17:33.42 | UBiQUiTY | ya i guess ill need it then |
| 17:33.58 | jakepdev | facek - we can talk on the channel |
| 17:33.59 | UBiQUiTY | oh well, i left the crimper at home today .... theres always tomorrow |
| 17:34.28 | facek_ | jakepdev i get form manager |
| 17:34.35 | facek_ | if Zap/(.*) is calling |
| 17:34.37 | facek_ | i get tgat |
| 17:34.43 | facek_ | and latter get calleridnum |
| 17:35.01 | facek_ | and latter i must send it to called human |
| 17:35.03 | facek_ | but how? |
| 17:35.09 | facek_ | he can call me or friend |
| 17:35.10 | jakepdev | facek - using FastAGI - you can get anything by using GET VARIABLE |
| 17:35.51 | Gand_DJ | hrm, does asterisk support call parking by default? Ijust tried a test call using Xten softphone and the Park option said the feature isn't avail |
| 17:37.19 | *** join/#asterisk felipeao (~felipeao@berg.viaip.com.br) |
| 17:38.16 | felipeao | hi every1... do u guys know any command-line based SIP client that I can use under kernel 2.4+, withouth having to have X installed? |
| 17:38.17 | durex | please somebody can test call FWD# 644754 ??? |
| 17:38.42 | Gand_DJ | i will durex |
| 17:39.03 | durex | Gand_DJ please :-) |
| 17:39.19 | durex | let me see if incoming calls are working... ;-) |
| 17:39.25 | Gand_DJ | sat there for 5 seconds... and then just disconnected |
| 17:39.38 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 17:39.44 | durex | can u try again... i reload configuration here... |
| 17:39.55 | *** join/#asterisk stoyan (~stoyan@ns.burdenis.com) |
| 17:39.57 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 17:40.05 | Gand_DJ | same thing |
| 17:40.44 | point | felipeao, asterisk ;) |
| 17:43.25 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 17:43.31 | *** join/#asterisk laotan (~jesse@H38.C18.B96.tor.eicat.ca) |
| 17:45.34 | CoaxD | Bwahahahahaha |
| 17:45.37 | CoaxD | He said he emailed you with proposal. I've taken this last month's payment. |
| 17:45.37 | CoaxD | Account |
| 17:45.39 | CoaxD | err |
| 17:45.47 | CoaxD | http://irc.evtek.fi/urllog/urllog/2005/04/15/vim.gif |
| 17:46.16 | CoaxD | Thats one of the funniest things i've seen all week |
| 17:46.48 | vaewyn | CoaxD: bwahaha |
| 17:47.55 | *** join/#asterisk Luhiwu (~marsosa@200.63.89.245) |
| 17:48.13 | mogorman | hehe that is funny |
| 17:48.23 | UBiQUiTY | JAKEPDEV: u suggested i run hdparm to see if i/o wait is having an impact on my sound quality loss.... that would only apply if i was using an IDE drive, correct? (i'm using scsi) |
| 17:50.56 | *** join/#asterisk JerJer (~JerJer@12.173.204.122) |
| 17:52.14 | *** part/#asterisk Kal_Zakath (~Kal_Zakat@213.219.184.108.adslpower.by.edpnet.be) |
| 17:52.53 | *** join/#asterisk RevK_ (~adrianken@81.187.165.154) |
| 17:54.35 | jakepdev | UBiQUiTY: don't know for sure. these were just given to me yesterday as a few options |
| 17:54.52 | *** join/#asterisk Rick_Hunter (~rhunter@04-043.008.popsite.net) |
| 17:54.55 | UBiQUiTY | ok cool |
| 17:55.06 | *** join/#asterisk ramtha (~tk@td9091901.pool.terralink.de) |
| 17:55.14 | ramtha | hi |
| 17:55.35 | felipeao | point, how can I use asterisk as a sip client? |
| 17:55.48 | felipeao | I want to interconnect a sip client using asterisk as the sip proxy, to another sip client |
| 17:56.00 | felipeao | just for testing |
| 17:56.29 | jakepdev | felipeao - just make the proper enteries in sip.conf and in your dialplan (extensions.conf) |
| 17:56.34 | jakepdev | entries |
| 17:57.00 | felipeao | sure, but how am I going to speak and listen to the other endpoint? |
| 17:57.00 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) |
| 17:57.18 | point | felipeao, chan_alsa chan_oss ... |
| 17:57.28 | ramtha | im using digium card quad 4. works fine, but: if i call a cellphone over pstn (euroisdn) and not pickup the call at my gsm phone, by pressing the busy key, asterisk rings on and on and did not recognize the "hangup" from the cell phone |
| 17:57.30 | jakepdev | if you have two endpints, you call in from one to * and * dials out to the other |
| 17:58.09 | ramtha | calls to analog/isdn line works |
| 17:59.53 | felipeao | thanks. I will try to sort the config for alsa |
| 18:00.36 | *** join/#asterisk Lee__ (~lee@cpe-69-203-206-248.nyc.res.rr.com) |
| 18:02.20 | *** join/#asterisk SplasPood (jwb@paravolve.net) |
| 18:04.19 | bjohnson | felipeao: why don't you just use a softclient |
| 18:04.48 | bjohnson | felipeao: ie. call asterisk, have asterisk dial the other phone, answer the other phone, speak |
| 18:05.06 | bjohnson | that is the normal method for handling calls in * |
| 18:07.12 | *** join/#asterisk km- (~pgrace@brdgw1.rttx.com) |
| 18:07.30 | felipeao | sure, thats what I asked the first time ehhehe |
| 18:07.31 | km- | anyone here use SJPhone over X-Lite? |
| 18:07.51 | jakepdev | ?? |
| 18:08.04 | jakepdev | they are two different soft clients - can you clarify? |
| 18:08.21 | felipeao | quoting myself -> ~felipeao: hi every1... do u guys know any command-line based SIP client that I can use under kernel 2.4+, withouth having to have X installed?~ |
| 18:08.34 | km- | jakepdev: forgive me, "over" is an american colloquial |
| 18:08.35 | bjohnson | linphone |
| 18:08.49 | jakepdev | km - are you asking about preference? |
| 18:08.50 | km- | jakepdev: I meant, which one do the people here prefer? |
| 18:08.52 | km- | yeah. |
| 18:08.53 | felipeao | linphone? ok I will give it a try, thx! |
| 18:08.54 | jakepdev | ok |
| 18:09.10 | bjohnson | there is a gui version of linphone too |
| 18:09.19 | jakepdev | I've used both and they seem fine |
| 18:09.55 | jakepdev | personally - SJPhone was easier to config - but others may have different experience |
| 18:10.07 | km- | well, I dont like X-Lite because of the lack of transfer |
| 18:10.11 | km- | hmm... |
| 18:10.27 | km- | do I need something like cantransfer=yes in my sip.conf? |
| 18:10.33 | km- | that could be the problem in both instances |
| 18:10.41 | kajtzu | linphone is a pain in the ass to compile and get working on rhel4 and fc3 |
| 18:10.47 | kajtzu | and seems really unstable as well |
| 18:10.54 | kajtzu | could be it's just the new gnome versions |
| 18:10.54 | kajtzu | :) |
| 18:11.09 | bjohnson | installed from the rpm for me with fc3 |
| 18:11.28 | bjohnson | but that's the gnome version which isn't what he wants |
| 18:11.50 | kajtzu | I use 7960s on the desk and roadwarriors xten |
| 18:11.51 | jakepdev | km - i don't believe so |
| 18:12.40 | jakepdev | transfer=yes |
| 18:12.45 | jakepdev | you can try that |
| 18:14.11 | *** join/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 18:14.24 | jabbzy | evening all. |
| 18:14.58 | *** join/#asterisk Inv_arp (junya@adsl-8-231-139.mia.bellsouth.net) |
| 18:15.58 | Inv_arp | omg xchat logged in here automatically |
| 18:17.33 | Nugget | omg lol rofl +-+-+ |
| 18:17.42 | jabbzy | xchat? mac or liunx? |
| 18:18.00 | Inv_arp | linux |
| 18:18.23 | jabbzy | in the context of "sip chanel" iax channel, zap channel, whats a local channel? |
| 18:18.31 | Inv_arp | usaully i have to type /join $blah for this chan all other chans join fine |
| 18:18.39 | Inv_arp | err #blah |
| 18:19.00 | mogorman | kpfleming or bkw around? |
| 18:19.23 | bjohnson | Inv_arp: I think you have to register first |
| 18:19.40 | Inv_arp | bjohnson: yea xchat has a register line |
| 18:20.08 | Inv_arp | but for some reason even after registering i have to /join manually most of the time |
| 18:20.34 | Inv_arp | today it actualy went in .... hmm weird *shrug* |
| 18:21.11 | Inv_arp | jabbzy: no idea of local chan |
| 18:22.07 | Inv_arp | heh... im starting to move off of EFNET completely and to FREENODE |
| 18:22.42 | Inv_arp | efnet is getting to ridiculous with rules these days in many chans |
| 18:23.45 | Inv_arp | can't even have a quit message anymore in some chans |
| 18:25.07 | bjohnson | probably depends on the text in the quit message |
| 18:25.19 | moy | hi, quick question: if i do SetVar(somevar=somevalue) and somevar is a variable that previously existed as Global variable, other channels will be affected? |
| 18:25.33 | *** join/#asterisk criten (~peterh@dsl-202-173-184-65.nsw.westnet.com.au) |
| 18:25.42 | bjohnson | ie .. "I'm going for beer, follow me" might be acceptable |
| 18:26.04 | criten | anyone here from nufone? |
| 18:26.15 | bjohnson | I'm from Canad |
| 18:26.16 | bjohnson | a |
| 18:26.18 | bjohnson | haha |
| 18:26.26 | Inv_arp | damnit VP raised there rates to $11 |
| 18:26.50 | Inv_arp | i need a replacement for my BV incoming... since they only support ulaw |
| 18:27.09 | Inv_arp | voicepulse incoming |
| 18:27.16 | criten | I hear that canadians have trouble calling nufone freecall numbers |
| 18:27.23 | *** part/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 18:27.26 | bjohnson | freecall numbers? |
| 18:27.39 | bjohnson | toll free? |
| 18:27.44 | criten | yeah |
| 18:28.15 | bjohnson | we have trouble calling US toll free lines that aren't activated in Canada |
| 18:28.34 | bjohnson | just like US residents have trouble calling Canadian toll free numbers that aren't activated in the US |
| 18:28.53 | criten | are you saying nufone numbers are only activated for the us? |
| 18:29.02 | bjohnson | but then, I can't call UK toll free numbers either |
| 18:29.11 | bjohnson | I'm saying .. ask them |
| 18:29.35 | Gand_DJ | nufone would have issues with canadian toll-free, since it's usa based |
| 18:29.35 | TEKjacob | or just like $country_a has trouble calling tollfree numbers in $country_b that aren't activated in $country_a |
| 18:29.43 | Inv_arp | criten: shido6 is from nufone hes pretty good too work with |
| 18:29.43 | bjohnson | (well with voip you can get around these barriers for outgoing) |
| 18:29.45 | JerJer | "US48 Toll-Free number" isn't that not obvious enough? |
| 18:29.51 | bjohnson | TEKjacob: exactly |
| 18:30.17 | criten | JerJer, is it possible to get toll free numbers that work in canada? |
| 18:30.30 | JerJer | not from us |
| 18:30.36 | bjohnson | err .. yes |
| 18:30.52 | Gand_DJ | JerJer, us? |
| 18:30.52 | bjohnson | some voip providers offer them but they are usually more costly |
| 18:30.54 | JerJer | absolutely not worth the costs we have to incur |
| 18:30.55 | Weezey | criten: 5 cents per minute form my guy. |
| 18:31.06 | bjohnson | oh .. not from us .. not .. not from US |
| 18:31.18 | criten | Weezey, want to message me a url? |
| 18:31.35 | Inv_arp | anyone have an iax.cc router i can try to see ping delay before i join? |
| 18:31.39 | Weezey | criten: I can give you his contact info. |
| 18:31.55 | bjohnson | was a few on the asterisk-box list a couple of weeks ago |
| 18:32.12 | bjohnson | looklieloo keeps saying he has them but damn if I can get him to reply to my emails |
| 18:32.24 | bjohnson | les.net is CA too I think |
| 18:32.47 | bjohnson | I guess that was really the asterisk-biz mail list |
| 18:32.48 | Gand_DJ | les.net....lol |
| 18:32.50 | Gand_DJ | :P |
| 18:32.58 | bjohnson | yeah .. not much info available |
| 18:33.16 | *** join/#asterisk jetx (~jetx@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 18:33.24 | Gand_DJ | he's local to me sorta. Does long distance for a couple small towns on the city border |
| 18:34.22 | bjohnson | Inv_arp: 205.234.133.203 |
| 18:34.39 | Gand_DJ | Has anyone tried the broadvoice toll-free number for their acct? |
| 18:34.45 | Inv_arp | bjohnson: u use them ... pretty good? |
| 18:34.48 | Gand_DJ | It's pretty cheap iirc |
| 18:35.00 | bjohnson | I have them for a couple of low use incoming lines |
| 18:35.08 | bjohnson | iax.cc ^^ |
| 18:35.23 | bjohnson | tzanger uses them for a DID or 2 as well |
| 18:35.35 | Inv_arp | Gand_DJ: i use BV local incoming DID, but htey only support ulaw/alaw kills my bandwidth |
| 18:35.51 | *** join/#asterisk stevej (~stevej@67.97.36.243) |
| 18:35.52 | Gand_DJ | ok. |
| 18:36.05 | Inv_arp | bjohnson: they support gsm/ilbc? |
| 18:36.11 | UBiQUiTY | question: i have my pri correctly functioning on span 1 bchan=1-23 dchan=24 , i want to unplug it from span 1 and plug it into span 2, which has the exact same setup, but bchan=25-47,dchan=48 ... it wont work... telling me there is no dchannel on 48 ... does this mean that i must reconfigure my span 2 to be 1-24??? |
| 18:36.45 | UBiQUiTY | it doesnt seem to allow me to configure BOTH span 1&2 to be bchan=1-23 and dchan=24 |
| 18:36.55 | UBiQUiTY | (not at the same time, at least) |
| 18:37.38 | Weezey | I gotta get myself some g729 licenses. |
| 18:38.18 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 18:38.22 | Wazb | hi |
| 18:38.50 | bjohnson | Inv_arp: I think so |
| 18:39.14 | Wazb | i am getting NOT ACCEPTABLE MEDIA error , when i trying to send call from * to CISCO box, any help? |
| 18:39.30 | *** join/#asterisk haujobb2000 (~joerg@p3EE31342.dip0.t-ipconnect.de) |
| 18:39.55 | *** join/#asterisk AmaDEE0_ (~foo@209.52.90.78) |
| 18:40.32 | AmaDEE0_ | When I update my cvs of Asterisk I still see the version 1.0.03. Why does that not change? To 1.0.7 or whatever the latest is? |
| 18:40.40 | AmaDEE0_ | It does from time to time check out new files. |
| 18:40.59 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 18:42.21 | Damin | AmaDEE0_: cd ../asterisk |
| 18:42.21 | Damin | rm -f .version |
| 18:42.22 | Damin | make clean && make install |
| 18:43.09 | *** join/#asterisk MarkS__ (~MarkS@cpe-70-112-81-84.austin.res.rr.com) |
| 18:47.03 | km- | maybe I'm just a sped, but I can't get transfer to work in sjphone! |
| 18:47.46 | km- | you hit transfer, it brings up the menu, you hit "enter number", I dial the number I'm transferring to, then hit what? If I click transfer again, or if I hit enter, sjphone hangs up |
| 18:48.00 | km- | I love technology! :) |
| 18:48.44 | Damin | ~say I love the core file! |
| 18:48.46 | jbot | I love the core file! |
| 18:51.42 | AmaDEE0_ | Damin: It went up to 1.0.04, but still not to 1.0.7. |
| 18:51.46 | Wazb | i am getting NOT ACCEPTABLE MEDIA error , when i trying to send call from * to CISCO box, any help? |
| 18:53.24 | AmaDEE0_ | Does anyone know how to update via cvs to 1.0.7? I seem to only be able to get 1.0.04 when I update. |
| 18:53.38 | riksta | using the right codec Wazb ? |
| 18:55.08 | bjohnson | ?msg jbot |
| 18:55.27 | riksta | hey bjohnson |
| 18:56.20 | bjohnson | hi |
| 18:56.27 | Nugget | AmaDEE0_: you are reading the version string incorrectly. |
| 18:56.44 | Nugget | that "04" means april. |
| 18:56.49 | Nugget | read the whole string |
| 18:56.50 | Damin | http://pastebin.ca/9922 |
| 18:58.25 | vaewyn | hahaha... Norhell guy can't figure out why his switch is appending '000' to the end of 4 digit extensions callerid when it comes over my PRI hehehe |
| 18:58.30 | vaewyn | I love stumping him |
| 18:58.33 | bjohnson | ~jbot bjohnson |
| 18:58.34 | jbot | extra, extra, read all about it, bjohnson is a hoopy frood who really knows where his towel is |
| 18:58.56 | vaewyn | ~vaewyn |
| 18:58.57 | jbot | somebody said vaewyn was really a much cooler frood than vaewyn |
| 18:59.08 | vaewyn | :} |
| 19:01.27 | bjohnson | http://www.imdb.com/title/tt0371724/ hhgttg 2005 was just recently released |
| 19:01.44 | *** join/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 19:02.11 | vaewyn | *nods* must see hhgttg! |
| 19:02.40 | Wazb | i am using H323 to make calls but i getting no audio on either side |
| 19:03.03 | vaewyn | Wazb: must be pretty boring calls *rimshot* |
| 19:03.35 | Wazb | yes they are , any help |
| 19:03.40 | km- | vaewyn: is he on your staff? :) |
| 19:03.43 | km- | vaewyn: the norhell guy |
| 19:03.51 | vaewyn | km-: yeah |
| 19:04.11 | km- | vaewyn: nice. Would you mind proxying a question to him for me? |
| 19:04.14 | vaewyn | He really knows his norhell... but anytime you expose it he looses his mind :P |
| 19:04.25 | vaewyn | km-: I can try... what ya need? |
| 19:05.10 | km- | vaewyn: XO has a DMS-500 in our comm closet.. Apparently, angler_ tells me that it's most likely misconfigured. I send calls out on E&M Wink,but I never receive an "off-hook" signal from the DMS-500 -- the call originates fine, but the call stays in "Ringing" state the whole time... |
| 19:05.21 | Luhiwu | Wazb: did you check the codecs? |
| 19:05.35 | UBiQUiTY | all this talk (above) about version #'s made me want to check my version.... i JUST checked out from vcs using "cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds" and asterisk still reports version "CVS-v1-0-03/14/05-12:41:13" |
| 19:05.38 | km- | vaewyn: just wondered if there was some magical "Not-Take-Off-Hook" setting or something :) |
| 19:05.56 | UBiQUiTY | why doesnt it say v1-0-7 ? |
| 19:05.59 | vaewyn | km-: I'll ask him... but down he has a clue... He's never played with wink |
| 19:06.09 | km- | vaewyn: ahhh |
| 19:06.17 | vaewyn | doubt even |
| 19:06.23 | vaewyn | will ask though |
| 19:07.24 | *** join/#asterisk tekjacob (~myself@c2.efb7d1.client.atlantech.net) |
| 19:07.33 | km- | worth a shot |
| 19:07.37 | vaewyn | yep |
| 19:07.42 | km- | he might know how to get ahold of a DMS-500 user manual in pdf format |
| 19:07.48 | km- | that might be very helpful for me too :) |
| 19:07.56 | vaewyn | hehehe |
| 19:08.14 | km- | meanwhile, to figure out how to get transfers working via sip... |
| 19:08.21 | *** join/#asterisk durex (~ironman@weber.anpa.org.br) |
| 19:08.39 | durex | may someone can call an FWD# number to make incoming call test? |
| 19:10.35 | riksta | yeah |
| 19:11.00 | riksta | durex: number? |
| 19:12.23 | Seyr | I'm using IAX and incoming calls do not seem to send DTMF properly. If I type 1-2-3-4-5 it might grab 1-3-4-5 or 1-2-5. I was using a SIP peer at first, then switched to IAX and I still have the same problem. If I call from a softphone to the Asterisk box, it works fine. If I call through a VoIP line, I get the DTMF problems. |
| 19:12.31 | bjohnson | riksta: you still trying to get durex's number? |
| 19:12.35 | durex | riksta 644754 |
| 19:12.37 | bjohnson | www.durex.com |
| 19:12.39 | riksta | :) |
| 19:12.44 | riksta | i was waiting for that |
| 19:12.46 | durex | bjohnson :-0 |
| 19:12.55 | riksta | calling |
| 19:13.10 | durex | well... nothing here... |
| 19:13.14 | riksta | it's ringing |
| 19:13.33 | riksta | Seyr: try typing slower |
| 19:13.35 | durex | yeap, but it's not ringing here... |
| 19:13.59 | durex | well well well... does anybody are accepting incoming calls from FWD network ? |
| 19:14.10 | km- | ah shit |
| 19:14.21 | km- | vaewyn: if he's got any info, or that manual, can you forward it to pgrace@rttx.com? |
| 19:14.29 | km- | vaewyn: TIA! :) |
| 19:14.34 | vaewyn | will do |
| 19:14.36 | vaewyn | np |
| 19:14.37 | km- | you da man :) |
| 19:15.03 | *** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net) |
| 19:15.32 | bjohnson | durex: it has in the past. I don't use it often |
| 19:18.06 | *** join/#asterisk jtodd (~jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 19:18.40 | Wazb | <Luhiwu> G.711-ulaw |
| 19:18.51 | durex | bjohnson please look pvt |
| 19:20.04 | Micc | Is there a way to lookup the device by extension? In sip.conf you can have it register the extension, but that doesn't allow one to dial that extension from that context and ring that device. Is there a way to do this? |
| 19:20.41 | Micc | I've tried a couple combinations of exten => _7XXX,1,Dial(${EXTEN}) |
| 19:20.50 | Luhiwu | Wazb, did you see which codec do the other side support? |
| 19:21.39 | Wazb | <Luhiwu> the other is Cisco and it accepts all |
| 19:22.08 | Wazb | <Luhiwu> can i change codec to something else ... |
| 19:22.15 | kajtzu | Micc: Dial(SIP/10${EXTEN}) |
| 19:22.20 | kajtzu | Micc: Dial(SIP/${EXTEN}) |
| 19:22.20 | kajtzu | :) |
| 19:23.20 | Gand_DJ | Does * allow you to park a call directly from the "park" button on Xten softphone? |
| 19:23.35 | Gand_DJ | For some reason my softphone says my network doesn't support it |
| 19:24.00 | Wazb | <Luhiwu> i am calling like this Dial(H323/xxxx@xxxxx,20,rt) |
| 19:24.58 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 19:25.19 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 19:25.39 | *** join/#asterisk Alexandre22 (~agbsn@200.206.137.244) |
| 19:26.29 | *** part/#asterisk point (1000@213.27.44.55) |
| 19:26.45 | *** join/#asterisk ixx (foobar@cpe-70-113-47-137.austin.res.rr.com) |
| 19:26.59 | *** join/#asterisk Moc[Toronto] (~mochouina@209.167.66.3) |
| 19:27.08 | Moc[Toronto] | Hail |
| 19:27.24 | Alexandre22 | hello guys, does anyone here users g723 in pass-thru mode? |
| 19:27.47 | JerJer | with very recent cvs, sure |
| 19:27.51 | JerJer | like less than a couple days ago |
| 19:28.10 | Moc[Toronto] | ;nick |
| 19:28.13 | Alexandre22 | [JerJer] g723? |
| 19:28.17 | JerJer | sure |
| 19:28.42 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 19:28.52 | Cardoe | JerJer: you run iaxtel no? |
| 19:29.01 | Alexandre22 | just put canreinvire=yes and caninvite=yes and thats all? |
| 19:29.03 | JerJer | no, digium does |
| 19:30.04 | Wazb | i am calling using Dial(H323/xxxx@xxxxx,20,rt) to Cisco IP box ,how can i change codec at my end |
| 19:30.16 | Cardoe | JerJer: don't you run one of the providors? |
| 19:31.26 | Micc | kajtzu, I suppose I should use the extension as the context in sip.conf instead of a name. |
| 19:32.19 | *** join/#asterisk Inv_arp (junya@adsl-8-231-139.mia.bellsouth.net) |
| 19:33.11 | Wazb | i am calling using Dial(H323/xxxx@xxxxx,20,rt) to Cisco IP box how can i change codec at my end for H323 |
| 19:35.16 | Seyr | <PROTECTED> |
| 19:35.48 | Seyr | anyone have a clue? |
| 19:39.40 | bjohnson | I thought the toronto von was in may |
| 19:40.08 | bjohnson | guess not |
| 19:40.34 | bjohnson | so any way to connect to the floor and shout obcenities to passers by? |
| 19:40.58 | Essobi | :) |
| 19:42.01 | Essobi | Wazb H323 isn't stable yet. |
| 19:42.11 | facek_ | do anyone have some good idea to receive from manager who is calling me |
| 19:42.15 | facek_ | i know only my sip login |
| 19:42.23 | *** join/#asterisk ionix (~ionix@209.71.254.135) |
| 19:42.24 | facek_ | and i want to get who is calling my.. his callerid |
| 19:42.57 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 19:43.09 | jakepdev | facek - you try FastAGI? |
| 19:43.20 | MuppetMaster | FastAGI for what? |
| 19:43.35 | facek_ | MuppetMaster for receive who is calling me |
| 19:43.41 | facek_ | jakepdev but i dont know how to use it |
| 19:43.49 | facek_ | jakepdev i have many machine |
| 19:43.50 | facek_ | no only one |
| 19:44.59 | jakepdev | you could AGI connect to the IP address of the computer you want to send info to and have it request the data. |
| 19:45.31 | facek_ | jakepdev FastAGI server may be that some computer, which is astersi, yes? |
| 19:45.31 | *** join/#asterisk PBXtech (~nik@207.59.218.4) |
| 19:46.48 | PBXtech | are voicetronix cards good? |
| 19:47.30 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 19:47.38 | PBXtech | thinking of getting a 12 port card |
| 19:47.38 | JerJer | PBXtech: no |
| 19:47.45 | PBXtech | oh why |
| 19:48.10 | L|NUX | how can i allow more then one incoming call on same sip ? |
| 19:48.13 | mountie | PBXtech: They are not digium |
| 19:48.44 | PBXtech | i fully support digium and have bought a lot of T1 cards BUT they dont sell a 12 port fxo card either |
| 19:48.57 | CoaxD | mountie: If asterisk were to NOT support these products, maybe he'd have a reaosn to go elsewhere |
| 19:49.33 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 19:49.35 | CoaxD | er. he'd have NO reason. blah. forgive me. tired. |
| 19:49.40 | PBXtech | i just need something more than 4 ports |
| 19:49.49 | JerJer | so then get an adtran ta 750 and a single span t-2 card |
| 19:49.52 | CoaxD | yupper. totally understood |
| 19:49.58 | JerJer | t-1 |
| 19:49.59 | JerJer | :) |
| 19:50.00 | jakepdev | facek - there are a bunch of ways you can go about this, but I'd suggest you read a little about FastAGI and I can almost guarantee you'll have a lot less questions... |
| 19:50.07 | CoaxD | jerjer: um. |
| 19:50.22 | PBXtech | 2 vs 1 ? |
| 19:50.22 | JerJer | phat phinger |
| 19:50.28 | CoaxD | jerjer: That would be what they call 'overpaying and overthinking' |
| 19:50.36 | PBXtech | T1/channel bank or a 12 port card |
| 19:50.37 | facek_ | jakepdev I read about it.. nad this is not good for me |
| 19:50.44 | facek_ | I want to get that info from manager channel |
| 19:50.57 | JerJer | T1/Channel bank will give you room to grow |
| 19:51.10 | DrWho17 | unless you don't need to grow |
| 19:51.39 | file[laptop] | GR303! *G* sorry but I had to |
| 19:51.41 | PBXtech | understand, think ide prefer 1 card |
| 19:52.18 | DrWho17 | right if you are a small business 6-12 lines might be fine forever, and you only need 1 card, not a digium card and channel bank |
| 19:53.02 | PBXtech | ive spent far enough money to digium that they can loose 1 sale to something that works better for this solution. just havnt used it so i was looking for non-polotical opinion :) |
| 19:53.33 | bjohnson | PBXtech: don't know <- my opinion |
| 19:53.39 | PBXtech | heh |
| 19:53.54 | PTG1234 | anyone have a sample .htaccess file for passwd auth.. mine seems to not be working and i can not figure out why |
| 19:54.03 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 19:54.23 | bjohnson | PBXtech: but digium or not .. a lot of benefits from using a channel bank |
| 19:54.30 | JerJer | ~google .htaccess |
| 19:54.46 | PBXtech | yea ive installed them |
| 19:54.56 | JerJer | only load 12 channels into the channel bank |
| 19:55.12 | bjohnson | what have you got for pstn? |
| 19:55.18 | JerJer | buy off ebay and you just might get it for about the same as a 12 port PCI card |
| 19:55.39 | PBXtech | this Zhone channel banks are cheep and work well |
| 19:55.43 | L|NUX | ~google limit call |
| 19:55.45 | JerJer | oh god no |
| 19:55.53 | JerJer | zhone are the WORST |
| 19:56.09 | PBXtech | why, using it for fxs works fine |
| 19:56.25 | JerJer | they crash |
| 19:56.37 | bjohnson | L|NUX: I think you might be looking for setgroup and cheackgroup like in the superdial macro |
| 19:56.37 | PBXtech | i havnt see that yet |
| 19:56.46 | PBXtech | havnt tried fxo |
| 19:57.11 | PBXtech | i have 2 of them working, fine. and a couple carrier access |
| 19:57.41 | PBXtech | quite the price diff too heh |
| 19:57.57 | Seyr | ` I'm using IAX and incoming calls do not seem to send DTMF properly. If I type 1-2-3-4-5 it might grab 1-3-4-5 or 1-2-5. I was using a SIP peer at first, then switched to IAX and I still have the same problem. If I call from a softphone to the Asterisk box, it works fine. If I call through a VoIP line, I get the DTMF problems. Anyone have a clue? |
| 19:58.20 | bjohnson | dtmfmode and ulaw |
| 19:58.35 | Seyr | dtmfmode is rfc2833 |
| 19:58.41 | JerJer | Seyr: running which version of asterisk? |
| 19:58.45 | JerJer | i smell -stable |
| 19:59.02 | PBXtech | HeHeHe stable worse than cvs? |
| 19:59.06 | Seyr | latest CVS as of yesterday morning |
| 19:59.11 | *** join/#asterisk girabraz (root@200.121.129.180) |
| 19:59.29 | JerJer | cvs up today |
| 19:59.47 | JerJer | Log Message: ensure that sequential DTMF sent via RTP has some 'dead air' between digits (bug #3675) |
| 19:59.51 | file[laptop] | JerJer: you're awfully talkative today, be quiet |
| 20:00.07 | JerJer | just beating the n00b's down |
| 20:00.24 | *** join/#asterisk jbragnar (~ragnar@c5100AE12.xdsl.catch.no) |
| 20:00.27 | Seyr | thanks JerJer |
| 20:00.34 | *** part/#asterisk ionix (~ionix@209.71.254.135) |
| 20:00.57 | PBXtech | wiki needs a mirror |
| 20:01.05 | JerJer | not just a mirror |
| 20:01.08 | JerJer | a new CPU |
| 20:01.22 | PBXtech | :) |
| 20:01.22 | JerJer | the database activity is just stupid on tiwiki |
| 20:01.22 | boch | lol |
| 20:02.28 | ixx | . |
| 20:02.28 | boch | JerJer age? |
| 20:02.39 | JerJer | i offered bandwidth, but he said bandwidth wasn't the problem...it is the massive amount of DB queries per page laod |
| 20:02.41 | JerJer | load |
| 20:02.54 | JerJer | boch: to perfection |
| 20:03.26 | boch | are the main developers of * in this room? |
| 20:03.34 | kajtzu | is it the amount of qps or just that the queries are doing full table scans and not using indexes? (just thinking) |
| 20:03.42 | file[laptop] | boch: maybe, maybe not |
| 20:04.13 | bjohnson | depends who's asking |
| 20:04.25 | bjohnson | and how much money you're carrying |
| 20:05.10 | boch | aha |
| 20:05.32 | JerJer | boch: define main |
| 20:05.43 | boch | founders? |
| 20:05.48 | JerJer | that would be kram |
| 20:06.11 | bjohnson | I thought he meant Maine |
| 20:06.25 | bjohnson | file[laptop]: you're close to that right? |
| 20:06.34 | file[laptop] | yes surprisingly |
| 20:06.37 | L|NUX | bjohnson : no there is a way from which i can do in sip |
| 20:06.43 | L|NUX | i think like limitcall=6 |
| 20:06.45 | L|NUX | :$ |
| 20:06.52 | bjohnson | google it |
| 20:07.13 | L|NUX | doing |
| 20:07.15 | bjohnson | according to what I read, setgroup/checkgroup was created to replace that since it didn't work |
| 20:07.41 | *** part/#asterisk Seyr (~Seyr@rrcs-24-227-133-226.sw.biz.rr.com) |
| 20:07.53 | bjohnson | watch yer step |
| 20:08.20 | bjohnson | guano |
| 20:08.47 | *** join/#asterisk vinsci (~vinsci@dsl-sjkgw2jb1.dial.inet.fi) |
| 20:09.54 | *** join/#asterisk Veryhot (~tho@adsl-69-109-159-239.dsl.sndg02.pacbell.net) |
| 20:10.07 | Veryhot | Anyone tried netlogic.net ? |
| 20:11.02 | bjohnson | kissmyasterisk is still available as a domain |
| 20:11.10 | PBXtech | YAVOIP |
| 20:12.02 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 20:12.20 | Veryhot | netlogic.net have out free locals, do they have good network? |
| 20:12.47 | PBXtech | to many to keep track of |
| 20:14.09 | bjohnson | just how hot? |
| 20:14.21 | PBXtech | very |
| 20:15.11 | bjohnson | free locals? |
| 20:15.21 | smiley- | hmm.. I am creating a menu with Background() playing soundfiles.. when I choose an option it takes "forever" beforce it transfer me to the choosed option |
| 20:15.38 | Veryhot | bjohnson: yeah, free outgoing locals |
| 20:15.42 | PBXtech | sounds like you got a . |
| 20:16.10 | Veryhot | bjohnson: and LD is .015 |
| 20:16.18 | bjohnson | looks like $21/mo with a 1 year committment |
| 20:16.47 | bjohnson | couple of others offer LD for same or less without a committment or monthly fee |
| 20:16.58 | bjohnson | so you gotta ask yourself .. |
| 20:17.06 | Veryhot | bjohnson: signup with the wholesale side |
| 20:17.16 | bjohnson | do make a lot of outgoing calls local to the DID? |
| 20:17.20 | bjohnson | huh? do ya? |
| 20:18.00 | Veryhot | bjohnson: wholesale.netlogic.net |
| 20:18.49 | notVeryhot | sounds like a self pitch to me |
| 20:19.19 | Veryhot | hey pbx :) |
| 20:19.29 | Weezey | bjohnson: you offer service? |
| 20:19.38 | *** join/#asterisk ropeguru_work (~ropeguru@141.152.37.26) |
| 20:20.30 | smiley- | ah.. I found the error..DigitTimeout(1) |
| 20:20.55 | ropeguru_work | Anyone noticing long durations between posting to the -user list and getting a copy back?? |
| 20:22.12 | Veryhot | nope |
| 20:22.32 | *** part/#asterisk theHub (~karl@69.177.93.36) |
| 20:22.50 | ropeguru_work | Hmmm.. Ok. It has also been over an hour since I have seen any posts from the -user list. Guess I will have to check my mail server when I get home. |
| 20:23.25 | nd^ | i have an T1/E1 card that i wish to write a linux driver for, and would like it to be possible to use it with asterisk (with ISDN PRI probably) -- any advice on the best way to do this (a capi driver? try to emulate a zaptel device?) |
| 20:23.32 | bjohnson | $14/mo for "unlimited" incoming and outgoing local calls seems ok .. if you use more than 1100 minutes per month |
| 20:23.43 | bjohnson | (on each DID subscribed) |
| 20:24.40 | Veryhot | bjohnson: compared to voicepulse $11/DID |
| 20:24.41 | ropeguru_work | I found the best solution for me was to pay for a cheap DID with incoming only, then subscribe to VoIPjet at 1.3 cents per minute. I still have to maintain a pots line for dsl so I use that for all my local calling. |
| 20:24.46 | ropeguru_work | Saves me a bundle |
| 20:25.10 | bjohnson | nd^: try email the developer mailing list .. not too many people with driver writing experience here |
| 20:25.16 | Veryhot | ropeguru: that's what I been doing. But lowest unlimited DID is like $9 |
| 20:25.19 | nd^ | bjohnson: ah ok, i wasn't sure :) |
| 20:25.29 | bjohnson | Veryhot: compared to livevoip at $1 / mo plus 0.012 / minute |
| 20:25.42 | bjohnson | ropeguru_work: me too |
| 20:25.49 | *** join/#asterisk Derkommissar (~alberto@66.64.215.7.nw.nuvox.net) |
| 20:25.54 | ropeguru_work | I went with myphonecompany.com. They have a $4.99 unlimited incoming, 60 minutes outgoing plan. Been working great for me. |
| 20:25.57 | Derkommissar | anyone know of a cheap ss7 solution ? |
| 20:27.09 | Veryhot | bjohnson: but there is a min per DID at livevoip |
| 20:27.12 | bjohnson | ropeguru_work: only US DIDs available so not usuable by many people here |
| 20:27.25 | bjohnson | Veryhot: I gave you the math answers already |
| 20:28.12 | Veryhot | roper: oh let me check them out. |
| 20:28.44 | bjohnson | ($14 - $1/mo ) / 0.012 = 1100 minutes of usage (approx). Use it less (averaged out) than that and the per minute plan is cheaper |
| 20:30.10 | bjohnson | 1100 is a fair bit for most people |
| 20:30.31 | Veryhot | bjohnson: but there is a $29 min per DID at livevoip. |
| 20:30.31 | bjohnson | and for business, multiple concurrent calls is usually more important than saving a couple of bucks |
| 20:30.38 | Veryhot | bjohnson: is taht $29 per year? |
| 20:30.44 | ropeguru_work | Veryhot: You willprobably need to go to their FAQ page to find the plan. You have to fax in a form to them. It is not readily advertised. They have a $4.99 with 60 outgoing minutes and a $9.99 plan with 500 outgoing minutes. |
| 20:30.49 | bjohnson | that is the minimum payment on account |
| 20:31.11 | bjohnson | that is pay $29 .. use it up .. pay another $29 .. use it up |
| 20:31.51 | Veryhot | bjohnson: oh ic not bad. I thought it was like 29/mon min. per month |
| 20:32.33 | bjohnson | nope .. every payment is min. $29. I think one year expiry to use up that $29 |
| 20:32.36 | Veryhot | bjohnson: they have good network? |
| 20:32.44 | bjohnson | don't know |
| 20:32.53 | bjohnson | I have a did from them but it is low usage |
| 20:33.43 | bjohnson | but I have the opinion that most package plans are not cheaper for most people |
| 20:34.01 | bjohnson | the ones ropeguru_work refer to sounds good .. but not available in my country |
| 20:34.04 | Veryhot | bjohnson: could they handle multiple cdrs per did? |
| 20:34.14 | bjohnson | well .. the DIDs are not useful to me |
| 20:34.46 | bjohnson | you will find that most prepaid plans are quite open to multiple calls |
| 20:34.56 | bjohnson | and most packaged plans discourage them |
| 20:35.07 | Veryhot | bjohnson: how many concurrent connections with livevoip? |
| 20:35.17 | bjohnson | you better ask them |
| 20:36.59 | bjohnson | http://www.livevoip.com/index.php?subject=1&content=asteriskServerPlans lloks like up to 12 .. but the more you want the more money on account they want |
| 20:37.11 | bjohnson | makes sense I guess since you would be using it up faster |
| 20:37.30 | ropeguru_work | I am personally not very fond of LiveVoIP. Mainly because of the way they change their minds on packages, pricing and providers on a whim. |
| 20:37.52 | jakepdev | but looks like you can lock in for $12 |
| 20:38.06 | ropeguru_work | They recently decided to change DID providers, and rather than notifying their customers, they just cut off all the DID's that were in use by customers and told them too bad. |
| 20:38.15 | bjohnson | I think that most prepaid plans will handle 4 to 6 at a time but are likely willing to provide more if asked |
| 20:38.47 | *** join/#asterisk jmacz (~jmacz@63.245.86.136) |
| 20:39.31 | Veryhot | rope: let me give myphonecompany.com a call |
| 20:40.38 | Veryhot | ropeguru: myphonecompany.com does iax and sip ? |
| 20:42.01 | L|NUX | i have problem |
| 20:42.24 | L|NUX | when i am trying to run * as non-root then MOH is not working :( |
| 20:42.42 | L|NUX | also i can't dial from eyebeam to X-PRO |
| 20:42.44 | L|NUX | :( |
| 20:43.09 | jakepdev | :( |
| 20:47.52 | ropeguru_work | Varyhot: Unfortunately they are SIP only. But I have had no problems with them. |
| 20:48.31 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 20:49.24 | *** join/#asterisk ManxPower (~eric@stirprop-S0-0-0-26.ndcr2.datasync.net) |
| 20:49.55 | eKo1 | L|NUX: sounds like permission problems to me. |
| 20:50.11 | *** join/#asterisk Apophis (~apophis@px-ls-2.ipdvb.ru) |
| 20:50.25 | eKo1 | Of course, they all sound like that when running * as non-root. |
| 20:50.32 | *** part/#asterisk Apophis (~apophis@px-ls-2.ipdvb.ru) |
| 20:53.26 | L|NUX | eKo1 : hmm |
| 20:53.41 | L|NUX | eKo1 : so it won't works ? |
| 20:53.52 | eKo1 | What won't work. |
| 20:54.01 | L|NUX | well MOH ? |
| 20:54.19 | eKo1 | check the permissions on the moh directory. |
| 20:54.24 | L|NUX | ok |
| 20:54.25 | L|NUX | wait |
| 20:54.29 | *** join/#asterisk bah (048830696@AC865145.ipt.aol.com) |
| 20:54.54 | L|NUX | drwxr-x--- 2 asterisk asterisk 4096 Apr 19 19:36 mohmp3 |
| 20:55.10 | eKo1 | and the files within? |
| 20:55.11 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 20:55.38 | L|NUX | same |
| 20:55.43 | L|NUX | owner ship to * |
| 20:55.44 | L|NUX | :) |
| 20:55.59 | ropeguru_work | L|NUX: Also make sure that whatever you are using to play the MOH, like mpg123, is running as the non-root user. |
| 20:56.14 | L|NUX | <PROTECTED> |
| 20:56.14 | L|NUX | <PROTECTED> |
| 20:56.23 | L|NUX | hmm |
| 20:56.24 | L|NUX | oh |
| 20:56.25 | L|NUX | wait |
| 20:56.30 | Veryhot | brb |
| 20:57.23 | L|NUX | how can i do this |
| 20:57.36 | L|NUX | changing owner shif ? |
| 20:57.38 | Nugget | for a guy with the nick "linux" you sure don't know much about it. |
| 20:58.11 | Damin | ~say Fuck! |
| 20:58.13 | jbot | Fuck! |
| 20:58.28 | *** join/#asterisk LoRez (lorez@lorez.staff.freenode) |
| 20:58.35 | daork | Nugget: linux fanboys rarely do.. |
| 20:58.41 | L|NUX | Nugget : i am doing man |
| 20:58.53 | daork | Nugget: thats it :) |
| 20:59.43 | L|NUX | asterisk 16205 2.6 0.9 6152 4876 ? S 15:54 0:01 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 |
| 20:59.43 | L|NUX | asterisk 16206 0.0 0.5 3984 2676 ? S 15:54 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 fpm-world-mix.mp3 |
| 20:59.44 | *** part/#asterisk Primer (~primus@sh.nu) |
| 21:01.08 | Essobi | ~seen Oprah's ass but I hear it's hot. |
| 21:01.11 | jbot | Essobi: i haven't seen 'oprah's ass but i hear it's hot.' |
| 21:01.23 | daork | oh man |
| 21:01.35 | daork | when did this channel become trash? |
| 21:01.51 | Essobi | Since the unwashed masses started populating. |
| 21:02.56 | Essobi | daork Is that anything like Bjork? |
| 21:03.30 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 21:03.41 | daork | Essobi: nope |
| 21:04.19 | Essobi | It's quitting time people. Go home, and feed your children. |
| 21:06.27 | *** part/#asterisk ropeguru_work (~ropeguru@141.152.37.26) |
| 21:09.56 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 21:17.26 | ariel_ | Essobi, I am home and I have already given my baby her food. |
| 21:17.35 | ariel_ | But his happens every 4 hours or so. |
| 21:21.38 | *** join/#asterisk tessier (~treed@210.245.38.106) |
| 21:22.25 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 21:22.46 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM0012c999e6a0.cpe.net.cable.rogers.com) |
| 21:23.16 | Pinhole | anybody made any progress with sphinx2? |
| 21:24.01 | Veryhot | anyone using www.eezeephone.com the x10? |
| 21:27.15 | eKo1 | Man, everyday I have problems with this shitty fxo equipment. Argh. |
| 21:27.41 | eKo1 | ok. back to linear algebra. |
| 21:29.40 | *** join/#asterisk bjohnson (~bjohnson@ip159-181.tor.istop.com) |
| 21:34.16 | *** join/#asterisk Dishwasha (~chatzilla@208.251.32.70) |
| 21:34.20 | Dishwasha | Howdy folks |
| 21:34.37 | Dishwasha | I am trying to test to extension 1234 to hear the default asterisk message and I get 499 Not Acceptable Here from X-lite, any ideas? |
| 21:34.52 | Dishwasha | I'm using AMPortal BTW |
| 21:36.02 | *** join/#asterisk SilentBob420 (~silentbob@2001:618:400:0:0:0:18a3:c452) |
| 21:38.48 | Dishwasha | Hmm, kinda dead in here |
| 21:39.06 | file[laptop] | the mailing list went dead, IRC went dead |
| 21:39.09 | file[laptop] | what's up with that?!? |
| 21:40.12 | Hmmhesays | heh |
| 21:40.20 | Hmmhesays | you suck at teh internet |
| 21:40.43 | Dishwasha | alright, thanks for making me feel dumb |
| 21:40.46 | Dishwasha | I'm sure you get that one a lot |
| 21:40.50 | Dishwasha | codec issue |
| 21:41.27 | Dishwasha | BUT, not to continue asking dumb questions, but I set up a SIP trunk and it says all circuits are busy whenever I dial 9 and my number |
| 21:44.31 | eKo1 | Advice: do NOT use AMP. |
| 21:45.38 | Dishwasha | well, I tried doing it manually via the config files and I get the similar results. Obviously I don't quite know what I'm doing. |
| 21:46.27 | Dishwasha | I'm pretty sure I've successfully registered with my SIP provider because I see a message saying something registered successfully in the logs |
| 21:46.29 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 21:46.34 | Dishwasha | before I register my client |
| 21:46.37 | JerJer | pretty sure? |
| 21:46.51 | JerJer | why don't you know for a fact? |
| 21:46.53 | Dishwasha | well, I didn't have my registration string correct before and I was getting registration rejected messages |
| 21:46.59 | file[laptop] | JerJer: shhhh if you keep talking you'll ruin your reputation |
| 21:47.00 | Dishwasha | that's why I believe I have it correct now |
| 21:50.41 | *** join/#asterisk MarkS__ (~MarkS@cpe-70-112-81-84.austin.res.rr.com) |
| 21:52.02 | MarkS__ | Hey yall.. |
| 21:53.08 | Dishwasha | I'm getting that the number I'm dialing is not a local user in the logs, then it tries the SIP trunk line and says Forbidden - wrong password on authentication for INVITE to |
| 21:59.18 | eKo1 | replace wrong password with right one. |
| 21:59.32 | *** join/#asterisk ropeguru (~ropeguru@fw.ropeguru.com) |
| 22:00.09 | Dishwasha | eKo1: Where specifically? I have the correct password in all sections related to passwords that I can see. |
| 22:01.13 | eKo1 | Hell if I know. How do you know they are 'correct' in the first place. |
| 22:02.25 | Dishwasha | Because I have all the passwords for all my lines from my provider |
| 22:03.57 | Dishwasha | I've successfully set up X-lite directly to my SIP provider, just having trouble getting Asterisk to |
| 22:04.22 | Micc | Dishwasha, I setup x-lite this morning to connect to my asterisk server. |
| 22:04.41 | Micc | Dishwasha, follow the instructions on the wiki |
| 22:05.03 | Dishwasha | Oh, I have x-lite set up to connect to my asterisk server just fine, just not from asterisk to my SIP provider, or at least not from x-lite to asterisk to SIP provider |
| 22:05.17 | Dishwasha | Micc: I've followed instructions from several sites and can't quite get it right |
| 22:06.00 | eKo1 | Well, keep fiddling. It should work at some point. |
| 22:06.07 | pigpen | I am using the Polycom ip phones...when I have a voicemail message ....the message light is not blinking...any ideas? |
| 22:06.21 | eKo1 | The light is dead? |
| 22:06.39 | pigpen | no...it works when the phone rings... |
| 22:06.56 | pigpen | ie: the light blinks. |
| 22:06.58 | ctooley | is there any way to send out voiemails as mp3 instead of wav? |
| 22:07.08 | ctooley | the wav files are only occasionally playable |
| 22:07.25 | Dishwasha | yeah, you have to compile it with lame support |
| 22:08.10 | outtolunc | ML down? |
| 22:08.29 | ctooley | and do you know how I would do that? |
| 22:08.57 | ropeguru | outtolunc: Yeah... Seems to be for a while now. |
| 22:10.00 | Dishwasha | ctooley: type ./configure --help and it should tell you |
| 22:11.07 | ctooley | Dishwasha, configure ? |
| 22:11.38 | Dishwasha | ctooley: Did you build from source or pull a package? |
| 22:11.51 | ctooley | cvs checkout |
| 22:12.10 | ctooley | there are no configure/autotools files in the dir.. |
| 22:12.24 | ctooley | and grep -r lame * gives nothing |
| 22:12.49 | eKo1 | download lame and install it then |
| 22:12.55 | *** join/#asterisk daxvon (~dax@host81-130-106-159.in-addr.btopenworld.com) |
| 22:13.08 | ctooley | installing lame isn't going to change my Asterisk source tree |
| 22:13.27 | eKo1 | why would it change your * source tree? |
| 22:14.28 | ctooley | so how would downloading lame and installing make "grep" return anything when using "grep -ri lame *" in the asterisk source tree? |
| 22:15.00 | *** join/#asterisk zapa (~zapa@201.135.137.236) |
| 22:16.10 | Micc | So in sip.conf the contexts are names of users or devices but how can I make it so Dial(SIP/${EXTEN}) works because ${EXTEN} equals the extension not the name of the sip device/context. |
| 22:16.26 | Micc | I don't want to have to hard code each user. |
| 22:16.31 | eKo1 | lame is a program that has nothing to do with *. |
| 22:16.46 | eKo1 | You you can convert those wav files to mp3 |
| 22:16.54 | ctooley | which points out that youd didn't read any of the rest of the thread |
| 22:17.21 | eKo1 | Micc: it's dial(sip/devincename/${exten} |
| 22:17.30 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 22:17.36 | eKo1 | ctooley: what thread? |
| 22:18.25 | Micc | eKol can I leave out the devicename because in sip.conf that contains user specific info too. |
| 22:19.00 | Romik | for "model name : VIA Samuel 2 |
| 22:19.00 | Romik | " which version of g729 need to use from digium i386 or Athlon? |
| 22:19.05 | eKo1 | What do you mean leave out? Do a 'show application dial' on the cli and start reading. |
| 22:19.50 | Micc | eKol, I know what I need to do. I can have the user specifics in the auth section. |
| 22:20.28 | eKo1 | If you know what you need to do, then why are you here? |
| 22:23.21 | Micc | eKol, I know what to try next anyways. |
| 22:23.21 | *** join/#asterisk alt_phil (~alt_phil@abgtr1.abgnetwork.net) |
| 22:24.26 | alt_phil | Hey all, I'm pulling my hair out here. My CALLERID and CALLERIDNUM varibles are getting set correct, at least a NoOp in extentions shows them, but when I dial, only the CID name is passed, not the number. |
| 22:24.54 | alt_phil | So the phone I'm calling says "From: alt_phil, unknown number" |
| 22:24.57 | Micc | eKol, I want my x-lite installations to have their extensions register automatically then you just put in the extension and dial and it'll connect to that extension. But that doesn't seem to work, to transfer or dial I need a name not an extension. |
| 22:25.19 | alt_phil | Any hints as to what I'm missing here? |
| 22:25.54 | zapa | hi all i have trouble with DID, i am reciving the call, but if i dont pick up the phone the telco company bill my customer what could be wrong? |
| 22:26.13 | Blissex | Micc: IIRC X-Lite does not do incoming calls... |
| 22:26.59 | zapa | i had a E1 encore r2 to PRI <-- asterisk E1 whit telmex telco inmexico |
| 22:27.07 | *** join/#asterisk GordonF (~somedude@rrba-146-118-245.telkomadsl.co.za) |
| 22:27.11 | GordonF | Hello's |
| 22:28.27 | Micc | Blissex, it works fine when I do this: Dial(SIP/mic) |
| 22:31.03 | GordonF | Quick Installation Question (Hopefully). unable to open master device '/dev/zap/ctl' anyone got any help here? |
| 22:31.09 | MarkS__ | is there a command that will hsow me both iax2/iax/sip channels all at once instead of doing all of them seperate. And is there a way to put a call on hold via ssh/the CLI) |
| 22:31.12 | eKo1 | If you're just calling another UA, the dial(sip/devicename) is fine. |
| 22:31.15 | Lee__ | GordonF: install zaptel |
| 22:31.47 | eKo1 | MarkS__: show channels |
| 22:31.50 | GordonF | Lee__: I have repeatedly |
| 22:31.53 | MarkS__ | lol |
| 22:31.53 | MarkS__ | ok |
| 22:32.02 | eKo1 | MarkS__: the for your second ?, no. |
| 22:32.08 | MarkS__ | damn im stupid |
| 22:32.09 | MarkS__ | OK |
| 22:32.29 | GordonF | Lee__: I have a zapctl file in the dev folder though |
| 22:32.34 | MarkS__ | ok |
| 22:32.46 | Micc | I think I'm needing to do Dial(Local/${EXTEN}) |
| 22:32.49 | MarkS__ | So there is no way to put a call on hold and put them to music via ssh? |
| 22:33.14 | MarkS__ | or /CLI |
| 22:38.20 | eKo1 | no |
| 22:39.00 | eKo1 | unless you find a way to sabotage the channel and inject the appropriate music |
| 22:39.35 | *** join/#asterisk P-Chan (~jpfingstm@68.142.66.200) |
| 22:39.41 | GordonF | Redid ZAPTEL again by 'make clean' 'make linux26' make install' 'modprobe zaptel' All ok till here then 'modprobe wcfxs' then ztcfg and get the same error |
| 22:40.57 | Lee__ | what's dmesg say? |
| 22:42.18 | GordonF | unable to open master device '/dev/zap/ctl' |
| 22:42.40 | GordonF | but that directory has'nt been created |
| 22:43.03 | *** join/#asterisk tempnick0 (~grurnpy@83.146.62.65) |
| 22:44.04 | Lee__ | nothing about loading the zaptel modules? |
| 22:44.13 | tempnick0 | Can I ask a question by posting to the channel? |
| 22:44.22 | GordonF | Lee__ nothing |
| 22:45.42 | Lee__ | what's lsmod | grep zaptel say? |
| 22:45.53 | JerJer | tempnick0: can you? |
| 22:46.25 | tempnick0 | i would like to, please |
| 22:46.54 | GordonF | Lee__ :zaptel 190980 1 wctdm and crc_ccitt 6081 1 zaptel |
| 22:47.03 | zapa | any experiencie with Direct inward dialing ? |
| 22:47.13 | eKo1 | tempnick0: no, you can't sorry |
| 22:47.57 | Lee__ | it sounds like your not loading the right driver for your card |
| 22:48.20 | tempnick0 | ah, so if i have a question about obtaining asterisk, then it will remain unasked? |
| 22:49.54 | eKo1 | yes, it will. |
| 22:50.36 | Moonwick | say, is anyone in here in the US and using * with ISDN? |
| 22:52.39 | eKo1 | of course |
| 22:53.50 | tempnick0 | i am using FreeBSD-5.3 and * doesnt seem to be in the ports, anyone know how I can obtain it with pkg_add? thankyou |
| 22:53.59 | Weezey | tempnick: net/asterisk |
| 22:54.30 | Weezey | tempnick: cd /usr/ports; make fetchindex; make search key="asterisk" |
| 22:54.35 | *** join/#asterisk JunK-Y (~JunK-Y@modemcable174.107-81-70.mc.videotron.ca) |
| 22:54.41 | *** join/#asterisk Moc[Toronto] (~mochouina@209.47.87.2) |
| 22:54.55 | Weezey | Hey Moc, how are the toronto hoes? |
| 22:55.55 | Moc[Toronto] | going well, I think im going to see a basketball game or something |
| 22:56.02 | Weezey | cool |
| 22:56.23 | Weezey | you gonna be down there tomorrow? |
| 22:56.42 | Weezey | I'm going to walk the show. |
| 22:56.54 | Moc[Toronto] | yep |
| 22:56.59 | Moc[Toronto] | to von or Linuxworld ? |
| 22:57.03 | Weezey | VON |
| 22:57.14 | Moc[Toronto] | you get free access to the the other also .. |
| 22:57.18 | *** join/#asterisk GordonF1 (~somedude@rrba-146-66-232.telkomadsl.co.za) |
| 22:57.20 | Weezey | nice, then both |
| 22:57.27 | GordonF1 | Sorry ISP problems |
| 22:57.28 | *** join/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 22:57.34 | Weezey | VON first, It's more important to me right now. |
| 22:57.44 | tempnick0 | Weezey, yes * exists but it is not on the ftp server |
| 22:57.57 | GordonF1 | Lee__ I'm back. Did you get my last post of the lsmod? |
| 22:58.27 | Weezey | tempnick: hrmm, have you updated your ports lately? |
| 22:58.55 | tempnick0 | its a new installation |
| 22:59.03 | syle | lol |
| 22:59.08 | Weezey | tempnick: so no. |
| 22:59.27 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 22:59.51 | tempnick0 | but I want to stay at 5.3 |
| 23:00.09 | syle | cvsup -g -L 2 -h cvsup5.freebsd.org /usr/share/examples/cvsup/ports-supfile |
| 23:00.14 | syle | or whatever it is on 5.3 |
| 23:00.47 | Weezey | tempnick: cd /usr; rm -rf ports; ftp ftp://ftp2.freebsd.org/pub/FreeBSD/ports/ports/ports.tar.gz; tar -zxvf ports.tar.gz |
| 23:01.09 | tempnick0 | i wanted to install using pkg_add, rather than via a compilation |
| 23:01.37 | tempnick0 | however, * is not on any ftp server: pkg_add -r asterisk fails |
| 23:01.40 | syle | how is 5.x, i still refuse to leave 4.x |
| 23:02.13 | Sedorox | tempnick0: I'm not sure how long it'll be when there are packaged for asterisk... |
| 23:02.26 | syle | your better off using linux anyways, fbsd don;t support SMP very well |
| 23:02.29 | Sedorox | I would suggest just compiling it.... |
| 23:02.33 | Sedorox | it doesn't?! |
| 23:03.22 | syle | that being said i run fbsd on all my single cpu machines and linux on my dual processor systems |
| 23:03.22 | tempnick0 | ok... various sites suggest it is packaged. i will try compiling. thankyou |
| 23:03.26 | Sedorox | tempnick0: also.. make sure that if you use Zaptel.. .to use 0.9 |
| 23:04.12 | Sedorox | I'm not sure.. I've never done asterisk via prebuilt package so... |
| 23:04.43 | Sedorox | syle: apparently smp is fixed in 5.x |
| 23:04.56 | tempnick0 | i cant find the package on any freebsd server |
| 23:06.57 | GordonF1 | sheesh took 10 minutes for my nick to die off |
| 23:07.13 | GordonF1 | Lee__ U still available? |
| 23:08.09 | *** join/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 23:08.34 | *** join/#asterisk SimonR (~SimonR@Toronto-HSE-ppp3736980.sympatico.ca) |
| 23:08.54 | SimonR | or is there an interface monitoring utility you can run short of using rrdtool or mrtg or simialr? |
| 23:08.56 | *** part/#asterisk Veryhot (~tho@adsl-69-109-159-239.dsl.sndg02.pacbell.net) |
| 23:09.05 | SimonR | how'd you monitor your ethernet interface for dropped packets and its response time? |
| 23:09.25 | eKo1 | mrtg |
| 23:09.27 | eKo1 | nagios |
| 23:10.17 | eKo1 | I also monitor * with mrtg. |
| 23:10.42 | JerJer | i use cacti |
| 23:10.47 | JerJer | and nagios |
| 23:10.53 | SimonR | cactai is nice |
| 23:11.24 | SimonR | so you use nagios to send off alarms, and cactai view the stats? |
| 23:11.27 | *** join/#asterisk ScythelX (Fleb@pc-24-181-176-181.sbi.ct.charter.com) |
| 23:11.30 | ScythelX | hello all |
| 23:11.37 | ScythelX | anyone using asterisk with freebsd |
| 23:11.48 | JerJer | SimonR: yes - very trivial |
| 23:11.54 | SimonR | can cactai graph anything, for example, if you were to write a snmp trap to say how many channels were in use, cactai would graph it? |
| 23:11.57 | Nugget | I use asterisk with freebsd. |
| 23:12.11 | ScythelX | did you install via ports or CVS |
| 23:12.19 | SimonR | jerjer, do you have any cool snmp traps written for *? |
| 23:12.22 | Nugget | I run cvs stable. |
| 23:12.24 | ScythelX | trying to figure out what maybe best to use |
| 23:12.28 | SimonR | I'm thinking of writing one to measure call setup times, for instance. |
| 23:12.33 | JerJer | don't use snmp |
| 23:12.44 | TechDawg | I gave up on Asterisk on FreeBSD and just loaded debian. |
| 23:12.55 | SimonR | jerjer: what, then? |
| 23:13.00 | ScythelX | ok I'll install from CVS |
| 23:13.10 | ScythelX | Nugget: where did you get your mpg123 from |
| 23:13.34 | Nugget | make mpg123 |
| 23:13.38 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 23:13.44 | Nugget | it's a tag in the asterisk makefile |
| 23:13.52 | SimonR | jerjer, cactai is all snmp based, no? |
| 23:14.16 | Nugget | as long as you don't need zaptel (and meetme), asterisk runs just spiffy on *bsd or osx. |
| 23:14.25 | Nugget | if you need zaptel or meetme, don't bother. |
| 23:14.27 | Wazb | anyone know how to change change codec in H323 |
| 23:14.31 | Nugget | just suck it up and tolerate linux |
| 23:14.39 | JerJer | SimonR: hell no |
| 23:14.58 | SimonR | so cactai just sends the stats directly into a database |
| 23:15.04 | JerJer | no |
| 23:15.10 | JerJer | well rra |
| 23:15.10 | SimonR | err rrdtool sends the stats and cactai views them |
| 23:15.19 | syle | nope |
| 23:15.19 | JerJer | so sorta db |
| 23:15.21 | ScythelX | Nugget: Zaptel was easy to install on BSD |
| 23:15.22 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
| 23:15.30 | syle | i got confirmation that SMP is still unstable in fbsd 5.x |
| 23:15.37 | Nugget | install isn't even a 10th of the issue. |
| 23:15.43 | Nugget | it's unreliable |
| 23:15.44 | ScythelX | Nugget: but I dont know how it performs |
| 23:16.00 | ScythelX | Nugget: Well im using ztdummy right now |
| 23:16.24 | Nugget | I hate linux as much as anyone, but seriously, if you need ztdummy for meetme just run linux. it's the only parctical solution. |
| 23:16.37 | TUplink | i want to exten => 8500,1,Voicemailmain(users number they are calling from) how do i do that |
| 23:16.38 | TUplink | :? |
| 23:16.41 | Nugget | I run linux on my asterisk box where I have zaptel in use. |
| 23:17.11 | syle | i run linux on dedicated dell 2850 box , runs sip only |
| 23:17.42 | Nugget | if you're looking at cacti, you might also want to consider cricket (cricket.sourceforge.net) as an alternative. it's more mrtg-like than cacti, but solves the same problems with mrtg. |
| 23:17.43 | L|NUX | hello |
| 23:17.57 | Nugget | I don't care for cacti's interface and inflexibility |
| 23:18.09 | L|NUX | i am trying to call from one * to anoter * but its saying the personal you are calling is not available please try again |
| 23:18.15 | L|NUX | i am using eyebeam |
| 23:18.30 | *** join/#asterisk chaoscon (~ph33r@chaoscon.user) |
| 23:18.34 | Nugget | L|NUX: what does the asterisk console say? |
| 23:18.46 | Nugget | there are so many places where that can be misconfigured. |
| 23:19.18 | Dishwasha | Question, if Asterisk is not a SIP proxy, why do I put the IP address of my Asterisk server in the SIP Proxy section of X-lite? |
| 23:19.52 | L|NUX | nothing |
| 23:19.53 | L|NUX | :( |
| 23:20.00 | Nugget | then increase verbosity. |
| 23:20.08 | L|NUX | wait |
| 23:20.40 | L|NUX | still nothing :( |
| 23:20.44 | Nugget | then increase verbosity. |
| 23:20.50 | Dishwasha | Is my SIP client supposed to authenticate to the Asterisk server or to my SIP provider? |
| 23:20.51 | L|NUX | i did |
| 23:20.52 | Nugget | or add some log lines. |
| 23:21.01 | L|NUX | like |
| 23:21.01 | Nugget | NoOps in the dialplan will help. |
| 23:21.45 | TUplink | what is noOps? |
| 23:21.48 | TUplink | i want to exten => 8500,1,Voicemailmain(users number they are calling from) how do i do that? |
| 23:21.53 | Nugget | read the docs, TUplink. |
| 23:21.54 | SimonR | has anyone used opennms? |
| 23:21.58 | Nugget | dialplan application noop |
| 23:22.01 | Dishwasha | Yes, it sucks SimonR |
| 23:22.06 | L|NUX | can some one able to dial 1002@e-maili.com |
| 23:22.08 | SimonR | Dish, how so? |
| 23:22.13 | L|NUX | sip:1002@e-maili.com |
| 23:22.22 | L|NUX | from their * |
| 23:22.29 | Dishwasha | It doesn't really give that much information, plus all the tomcat stuff is a pain to set up and very slow |
| 23:22.29 | SimonR | I'm looking for a package that can receive SNMP traps to send alarms, and that I can write my own monitoring scripts around. |
| 23:22.38 | SimonR | oh, it's in java |
| 23:22.44 | Dishwasha | yeah |
| 23:22.52 | SimonR | cactai is in php |
| 23:23.12 | TUplink | how do i get caller ID? |
| 23:23.17 | SimonR | Cricket is in perl |
| 23:23.21 | JunK-Y | TUplink: see wikis. |
| 23:23.24 | Dishwasha | How can I be sure that Asterisk is passing the proper authentication to my SIP provider when I make a call through my Asterisk server? |
| 23:23.44 | SimonR | dish, do the calls go through? |
| 23:23.45 | Nugget | cactai is spelled cacti. |
| 23:23.49 | Dishwasha | Simon |
| 23:23.53 | Dishwasha | SimonR: Ever look at Nagios? |
| 23:23.54 | Nugget | (for anyone who is having trouble googling for it) |
| 23:24.16 | Dishwasha | SimonR: No, I get a Forbidden - wrong password on authentication for INVITE problem |
| 23:24.23 | SimonR | Nagios is pretty popular... it's more of a trap handler I suppose. |
| 23:24.27 | Dishwasha | and an all circuit are busy problem |
| 23:24.46 | SimonR | Dish, have you looked at the invites using sip debug? |
| 23:25.00 | Dishwasha | no, is that from the CLI prompt? |
| 23:25.08 | SimonR | yes... sip show debug |
| 23:25.14 | SimonR | or 'sip no debug' |
| 23:25.28 | Dishwasha | dump question, if I have asterisk running in the background as a daemon, how can I get to the CLI prompt additionally? |
| 23:25.40 | Nugget | it's just "sip debug" not "sip show debug" |
| 23:25.44 | Dishwasha | no, asterisk -r |
| 23:26.38 | SimonR | do you guys have a good collection of snmp monitoring traps? |
| 23:27.15 | SimonR | here's a question: |
| 23:27.28 | SimonR | Which monitoring package is going to have the most modules written for it that I can borrow? |
| 23:28.04 | SimonR | I assume a SNMP trap written for one will work with another, short of the alarm scripting you put together for the actual monitoring system? |
| 23:28.49 | *** join/#asterisk zapa (~zapa@201.135.137.236) |
| 23:29.39 | Dishwasha | Is there a way to compile SIP proxy capability in to Asterisk? |
| 23:29.53 | Dishwasha | I'm getting SIP/2.0 403 Route Not Available |
| 23:30.34 | Wazb | anyone know how to change change codec in H323 |
| 23:35.45 | blitzrage | SimonR: you creating SNMP monitoring for Asterisk? |
| 23:35.45 | *** join/#asterisk PBXtech[mobile] (~upirc@wirelessdata-031-011.mycingular.net) |
| 23:36.32 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 23:37.07 | PBXtech[mobile] | dude |
| 23:38.35 | niZon | how does $200US sound for a cisco 7940? |
| 23:39.06 | PBXtech[mobile] | get a 7960 for 250 |
| 23:39.14 | niZon | hm |
| 23:39.31 | niZon | i want one with SIP firmware.. unless chan_sccp works well with the 7960 |
| 23:39.32 | *** join/#asterisk jmacz (~jmacz@63.245.86.140) |
| 23:39.57 | PBXtech[mobile] | 7960 is sip |
| 23:41.03 | PBXtech[mobile] | man I'm bored |
| 23:41.09 | *** join/#asterisk easimon (~easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 23:41.22 | *** join/#asterisk wdatkinson (~wdatkinso@pcp986542pcs.northw01.in.comcast.net) |
| 23:41.30 | niZon | both can run the sip firmware... i hear getting the firmware is difficult |
| 23:41.34 | niZon | (legally) |
| 23:42.09 | PBXtech[mobile] | I get it from cisco legally |
| 23:42.33 | Dishwasha | Okay, I think I've finally figured my problem out. Is SER a good SIP proxy? |
| 23:42.48 | PBXtech[mobile] | yup good |
| 23:44.46 | PBXtech[mobile] | man the martinis got me tonight |
| 23:44.46 | Dishwasha | how does it compare to siproxyd? I need to be able to take an internal SIP address and proxy it through an external SIP address |
| 23:45.06 | *** part/#asterisk PBXtech[mobile] (~upirc@wirelessdata-031-011.mycingular.net) |
| 23:45.19 | GordonF1 | Cya's |
| 23:46.35 | Dishwasha | I need to be able to map different asterisk extensions to seperate SIP addresses |
| 23:47.01 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 23:48.58 | JerJer | Dishwasha: ok what's the problem? |
| 23:49.44 | Dishwasha | JerJer: Asterisk simply forwards registrations, so my VoIP provider is seeing the call coming from SIP:100@10.5.1.57 rather than SIP:line6@myVoIPdomain.com |
| 23:49.57 | SimonR | blitzrage: yes, I'm serious about asterisk monitoring |
| 23:49.57 | Dishwasha | So it's rejecting the INVITE command |
| 23:50.03 | *** join/#asterisk DrJolo (~chatzilla@ts171.internetdsl.tpnet.pl) |
| 23:50.18 | JerJer | ok then run SER |
| 23:50.35 | Dishwasha | That's what I was asking, if SER can do this. |
| 23:50.37 | SimonR | I know there's a nagios snmp trap... does anyone know of other useful plugins? |
| 23:50.55 | JerJer | just write a perl script (or ten) |
| 23:50.58 | *** join/#asterisk Storhost (~rewt@adsl-210-7-169.mco.bellsouth.net) |
| 23:51.01 | blitzrage | SimonR: keep me posted, I'm quite interested too. I used SNMP to monitor Asterisk for my final project at school. I'd like to help (I can't help with coding though... much) |
| 23:51.14 | Dishwasha | because I want different extensions from asterisk to use different real SIP lines on my real VoIP |
| 23:51.16 | JerJer | snmp is no longer necessary |
| 23:51.18 | Storhost | hello. |
| 23:51.27 | SimonR | blitzrage, you are lief, right? |
| 23:51.29 | JerJer | so asterisk isn't real VoIP? |
| 23:51.42 | SimonR | jerjer, what do you use instead of snmp? what's wrong with snmp? |
| 23:51.45 | Dishwasha | public rather than private then, jerjer |
| 23:51.46 | blitzrage | SimonR: yes, you are Simon from asterisk-on, yes? |
| 23:51.50 | SimonR | yes |
| 23:51.51 | JerJer | perl script |
| 23:51.58 | blitzrage | SimonR: thought so :) |
| 23:51.59 | SimonR | that sends a phone call? |
| 23:51.59 | Storhost | can I maintain an astrisk server at my colocation, and have it dial out to multiple users, regadless of where they are located? |
| 23:52.05 | JerJer | nothing is wrong with snmp, its just not necessary in my world |
| 23:52.13 | JerJer | i use snmp to query switches, etc |
| 23:52.17 | SimonR | jerjer, why not make the perl scripts send traps? |
| 23:52.18 | JerJer | but not asterisk |
| 23:52.28 | *** join/#asterisk outtolunc (~chatzilla@adsl-69-110-26-49.dsl.pltn13.pacbell.net) |
| 23:52.38 | JerJer | cuz i use nagios |
| 23:52.40 | blitzrage | I used a perl script and snmptrapd on Linux to do all the stuff I needed, but I wasn't monitoring Asterisk, I was monitoring SNMP enabled devices with Asterisk |
| 23:52.45 | JerJer | dont' need the overhead |
| 23:53.01 | SimonR | so it's a performance issue mainly? |
| 23:53.30 | SimonR | blitz, you used asterisk to read out the error messages over the phone, right? |
| 23:53.36 | blitzrage | SimonR: hopefully I'll be at the next Asterisk Meetup in May |
| 23:53.41 | blitzrage | SimonR: yes |
| 23:53.44 | SimonR | cool, there's one tomorrow with kram |
| 23:53.54 | blitzrage | SimonR: yah.. I'm in Seattle this week unfortunatly... :( |
| 23:53.55 | Storhost | anyone availble to answer a question? |
| 23:54.04 | blitzrage | Storhost: just ask, if someone is, they'll answer |
| 23:54.07 | SimonR | blitz, there are two Simons in Toronto... Simon Rowland (me) and Simon Ditner, who is one of the organizers of asterisk-on |
| 23:54.11 | Storhost | Q: can I maintain an astrisk server at my colocation, and have it dial out to multiple users, regadless of where they are located? |
| 23:54.30 | SimonR | The other Simon is actually helping me with my product development... |
| 23:54.34 | blitzrage | SimonR: ooooh... you're on my MSN right? |
| 23:54.39 | SimonR | yes |
| 23:54.53 | blitzrage | SimonR: ok, the other Simon runs the asterisk-on list, got it now :) |
| 23:55.00 | SimonR | The other Simon is taller and dare-I-say a bit nerdier than me :) |
| 23:55.10 | blitzrage | SimonR: lol, I haven't met either :) |
| 23:55.18 | *** join/#asterisk apardo (~apardo@80-28-166-72.adsl.nuria.telefonica-data.net) |
| 23:56.30 | SimonR | so once you have a bunch of snmp trap generating scripts that sit on your machine and run tests, you can use any monitoring package, be it nagios or big sister, or whatever it is. |
| 23:56.39 | Storhost | so, those of you using Asterisk, are you using it in an office location with a physical phone system? |
| 23:56.55 | Storhost | or are you using it in, say, a hosted environment where you have remote access to the server |
| 23:56.57 | SimonR | storhost: I have my stuff at a data center |
| 23:57.00 | Storhost | ok |
| 23:57.01 | apardo | somebody knows gsm/xfs pci cards ? |
| 23:57.05 | blitzrage | Storhost: all the above :) |
| 23:57.17 | Storhost | so, it can then dial out using the trunk line from the data center |
| 23:57.25 | Storhost | to say, mulitple cell phones |
| 23:57.31 | SimonR | yes, I have 42 PRI :) |
| 23:57.46 | Storhost | can it call multiple phones @ one time? |
| 23:57.52 | Dishwasha | so am I going to have to make special rewrite scripts to do proxying in SER? |
| 23:57.53 | SimonR | many at once |
| 23:57.57 | ScythelX | anyone get this error on freebsd when trying to compile in res_odbc /usr/bin/ld: cannot find -lodbc |
| 23:57.59 | Storhost | and make the connection based on who answers 2st? |
| 23:58.04 | Storhost | err |
| 23:58.05 | Storhost | 1st |
| 23:58.28 | Storhost | ...what about conference calling, I know it says 3 way.. can it handle conferences? |
| 23:58.41 | *** join/#asterisk mog_home (~mog_home@146.229.176.225) |
| 23:58.55 | Storhost | Conference Bridging <- is that same as a conference call? |
| 23:59.04 | Storhost | forgive me, as I am not too familiar with the terminology. |
| 23:59.15 | Weezey | ScythelX: I googled for that when I got that, there's a fix. |
| 23:59.41 | Weezey | Storhost: google for Asterisk cmd MeetMe |
| 23:59.59 | *** join/#asterisk PBXtech (~nik@207.59.218.4) |
