00:00.12 | [hC] | i love the thin line that people operate between |
00:00.29 | trixter | and spreading falsehoods about guis. the gui in astlinux is basically a web based text editor but oops its a gui so it must be evil and not let you do everything and ... |
00:00.31 | [hC] | most people that say serious admins dont use a gui, are the people that have their 3-4 unix boxes that are tuned out the ass |
00:00.39 | [hC] | i would not call that serious admin |
00:00.54 | [hC] | anyone who manages 100, 500, 1000, 10000 machines will know that you need an interface to manage all of that |
00:01.12 | [hC] | doesnt mean it has to be a stupid dinky interface, but you need *something* |
00:01.31 | trixter | saying that the free ones overwrite manual changes, when amp itself doesnt even do that ... and saying that you cant be serious if you use either tool just for some ego related issue.. |
00:01.34 | Qwell | vi + rsync |
00:01.35 | Qwell | heh |
00:01.42 | mog_work | amen brother qwell |
00:01.50 | [hC] | :P |
00:01.52 | mog_work | but guis arent bad in general |
00:01.53 | Qwell | trixter: amp most certainly does overwrite manual changes |
00:01.55 | trixter | yes amen lets restrict cohice! |
00:02.00 | [hC] | good guis are good, and bad guis are bad |
00:02.00 | mog_work | but just no good ones yet |
00:02.05 | [hC] | look at windows and osx for example. |
00:02.09 | Qwell | My guis are good |
00:02.10 | mog_work | exactly |
00:02.15 | trixter | qwel: no it doesnt. becuase it doesnt overwrite sip.,conf for example it writes sip_additional.conf |
00:02.17 | trixter | which is included |
00:02.20 | mog_work | when i see a good gui for asterisk i will praise it |
00:02.22 | Qwell | I have a way to manage queues...and it's done RIGHT. |
00:02.25 | mog_work | but i havent see one yet |
00:02.27 | Qwell | but...it won't be released. :( |
00:02.37 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
00:02.37 | [hC] | amp is getting *better* but its still young. |
00:02.45 | mog_work | code tease |
00:02.53 | Qwell | until AMP stops writing to it's own database...it'll forever be junk in my eyes |
00:03.00 | mog_work | i think complete realtime would make a gui doable |
00:03.01 | [hC] | and is still geared towards the simple everything-on-one-box user. |
00:03.06 | Qwell | if it wants to write to a database...great, use realtime |
00:03.06 | mog_work | but editing flat files is dumb |
00:03.08 | trixter | my understanding is that the mac asterisk group has some nice guis, although I havent seen them just heard about them |
00:03.13 | Qwell | mog_work: I use realtime in my guis |
00:03.16 | Qwell | extensively |
00:03.20 | Qwell | You should see my dialplan |
00:03.20 | mog_work | its only way to do it right |
00:03.29 | mog_work | lol denon |
00:03.33 | mog_work | i dont quite believe that one |
00:03.46 | denon | oh, and cpu is at 2% util |
00:03.50 | Qwell | mog_work: Did I show you my EVIL func_odbc call? |
00:03.52 | trixter | [hC]: lets face it that is how most asterisk installs are done by volume (of installs not necissarily calls). |
00:04.03 | trixter | even the 'real admins' tend to run everything on one box :P |
00:04.29 | razu_ | hi ... anyone using realtime ? |
00:04.32 | [hC] | im not sure 'real' is the word i'd use :) |
00:04.39 | Qwell | razu_: get with the program :P |
00:04.44 | trixter | [hC]: heh.. why it was in quotes :P |
00:04.54 | razu_ | :) |
00:05.03 | [hC] | asterisk wise, alot of stuff makes sense to go on the same box.. sometimes. |
00:05.08 | [hC] | depending on what you're going for. |
00:05.24 | theblue | Do 'user's take calls or make them? |
00:05.47 | fugitivo | _Sam--: 6 servers? and you consider yourself a real admin? bah |
00:05.50 | razu_ | Qwell : so i have this kind of problem ... i have phones behind different nat-s and all i see is the damn public address ... not the nat address ... :( |
00:05.55 | OloBola | in the following example, how would I get "yada" from within my php/agi script? exten => 1,2,agi,test.py|yada |
00:06.02 | Qwell | mog_work: see msg for very EVIL fund_odbc call |
00:06.07 | Qwell | func_odbc rather |
00:06.07 | fugitivo | i want to see that guy administering 100 servers with a GUI |
00:06.23 | Qwell | fugitivo: never gonna happen |
00:06.46 | [hC] | again, it depends on the gui |
00:06.47 | razu_ | Qwell : why is that ? |
00:06.53 | [hC] | if you have a gui for each server |
00:06.55 | [hC] | that is insanity |
00:06.59 | fugitivo | [hC]: any gui |
00:07.02 | [hC] | if you have a gui that represents each server inside one interface. |
00:07.05 | [hC] | thats different.y |
00:07.06 | *** join/#asterisk Chiardon (n=yo@200.71.58.39) |
00:07.15 | fugitivo | [hC]: it's faster vi, scp and rsync |
00:07.25 | [hC] | It depends what we're talking about here |
00:07.29 | [hC] | Im talking about servers in general |
00:07.31 | fugitivo | in general |
00:07.32 | [hC] | not necessarily asterisk ones. |
00:07.35 | Qwell | mog_work: I had to expand the default buffer of 512 chars |
00:07.43 | mog_work | heh |
00:07.49 | fugitivo | i have 10 years of linux admin experience |
00:07.51 | fugitivo | never used a gui |
00:08.02 | trixter | and that is your choice |
00:08.08 | [hC] | I have about the same, and I use a mix. the gui's are usually ones ive written myself |
00:08.11 | trixter | but to mock people who want to give others the choice doesnt seem like a good idea |
00:08.21 | [hC] | having to push the same config changes out to 100 boxes simultaneously |
00:08.38 | fugitivo | trixter: people can use whatever they want, i just don't like when they use a gui and come here for help |
00:08.41 | [hC] | it all depends on what you're trying to do. |
00:08.43 | trixter | all I am saying is that people should be given the choice and falsehoods about guis and mocking of people that would write them, or use them doesnt seem wise |
00:08.53 | Qwell | fugitivo: I think you just summed up the argument... |
00:09.00 | trixter | aparently people cant use whatever they want without ridicule however |
00:09.01 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
00:09.04 | Qwell | It makes it difficult for US, when YOU use a GUI. :) |
00:09.13 | Qwell | because we prefer not to use them |
00:09.18 | trixter | people that use a gui know no more about asteirks than someone who just installed it |
00:09.20 | fugitivo | Qwell: that's what i wanted to say |
00:09.26 | [hC] | The problem with people using asterisk guis, is that the people in here do NOT use them, therefore do not know how they work under the hood, making it hard to help. |
00:09.31 | [hC] | whereas if you do it by hand, you know. |
00:09.32 | OloBola | damn gui queers |
00:09.43 | trixter | to think that people that are predisposed to use a gui somehow are born with less knowledge than someone who isnt predisposed is silly |
00:09.49 | trixter | everyone starts with 0 knowledge |
00:10.05 | xheliox | OloBola: wtf? |
00:10.09 | Qwell | I knew how to config asterisk, when I was born |
00:10.17 | Qwell | it just...you know...wasn't invented for 20 more years |
00:10.20 | trixter | that explains a lot about your comments here |
00:10.21 | fugitivo | trixter: people using a gui, come here asking question related the gui, using terminology used by the gui, they think we know what they're talking about |
00:10.27 | [hC] | i agree with both sides. sometimes its much more convenient to NOT use a gui, particularly when you have no CHOICE, and you have to use a gui rather than do it by hand, when you know it would be easier to do by hand |
00:10.41 | trixter | ahh so you dont like people talking about something you dont know about |
00:10.43 | theblue | How do I change the context for a register statement? |
00:10.45 | [hC] | and in other cases, doing it by hand are a time waste, and having a custom gui you've written to manage something can save tremendous amounts of time. |
00:10.46 | trixter | rather than them not knowing something you do |
00:10.49 | trixter | I start to understand |
00:11.20 | trixter | [hC]: the mac asterisk gui has a templating engine to do most common tasks for people very quickly |
00:11.21 | OloBola | I was suddenly reminded of the end of the movie brokeback mountain, sorry |
00:11.30 | trixter | or so they said I dont use a mac so I dont know |
00:11.40 | *** part/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net) |
00:11.40 | *** join/#asterisk Tiger (n=navajodu@adsl-69-106-247-196.dsl.pltn13.pacbell.net) |
00:11.45 | trixter | infact I dont think I have even seen a screenshot of it, just heard its some cocoa app |
00:11.58 | fugitivo | trixter: this is #asterisk, asterisk has plain text config files, i like people that come here asking for those files, not a GUI |
00:12.18 | trixter | ok so we cant talk about asterisk addons either |
00:12.21 | trixter | becuase that isnt plain asterisk |
00:12.28 | fugitivo | ok |
00:12.31 | Qwell | but many people here use it |
00:12.31 | fugitivo | enough, i have to go |
00:12.34 | trixter | or any questions that relate to connecting asterisk to any non asterisk device |
00:12.38 | trixter | that makes this channel quite useless |
00:12.50 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
00:13.13 | fugitivo | trixter: look, if you like your web interface, go ahead, do it for regular users, i'm not telling you not to do it, just make sure you create a support irc channel for your gui |
00:13.17 | [hC] | trixter: now you're getting it. |
00:13.18 | [hC] | :) |
00:13.27 | Chiardon | hey, anybody knows how do I find out which R2 variant do I have? |
00:14.05 | trixter | [hC]: exactly that and people that are opposed appear to have said clearly they dont like people asking questions about stuff they dont know |
00:14.15 | trixter | becuase they want to be able to answer every question rather than admit they dont know something |
00:14.24 | trixter | so rather than learn they want to attack people who do something they dont know about |
00:14.31 | alephcom | That's like the "outtolunch" guy the other night. |
00:14.31 | trixter | I think I finally understand the state of things |
00:14.38 | [hC] | trixter: its pretty comon. |
00:14.47 | [hC] | trixter: in life in general. |
00:15.08 | trixter | yeah, why it takes people like me to speak up for choice and freedom rather than lack of choice and lack of freedom |
00:15.23 | trixter | but then I am not all about removing choice from people or making them feel worthless for doing something I dont know about |
00:15.30 | trixter | I like to learn, maybe that is the difference |
00:15.44 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
00:15.56 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
00:16.16 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
00:16.58 | *** part/#asterisk Tiger (n=navajodu@adsl-69-106-247-196.dsl.pltn13.pacbell.net) |
00:17.27 | fugitivo | trixter: NOBODY is removing choice from people, if they choice to use AMP, then they should ask questions in #amportal, not here, get it? |
00:17.30 | [hC] | trixter: yeah, same for me, ive tried amp. it doesnt suit me, but i tried it out to see if it would. |
00:18.26 | trixter | you remove choice by attcking people for *writing* guis, you remove choice by attakcing people that use them |
00:18.31 | blitzrage | GUIs are not meant to give you total functionality of the system. The point is to make the commons tasks easier to do. You still need an administrator. Just like you still need a Windows admin in a networked server environment |
00:18.36 | trixter | get it? |
00:19.03 | trixter | blitzrage: I tried to bring up the 80% thing and got attacked for suggesting that |
00:19.05 | blitzrage | I wrote my own GUI to control several Asterisk boxen from a central portal |
00:19.08 | fugitivo | trixter: ok, i'm a bad person |
00:19.10 | trixter | someone else was attacked for saying they were writing a gui |
00:19.23 | fugitivo | "attacked" |
00:19.28 | blitzrage | trixter: you can't take things too personally in here |
00:19.33 | trixter | yeah told they arent real admins for writing software |
00:19.34 | fugitivo | you should be in politics for the way you talk |
00:19.40 | trixter | told that they are somehow inferior |
00:19.42 | trixter | that is attacking |
00:19.47 | fugitivo | ok |
00:19.51 | fugitivo | i'm a bad person |
00:19.55 | fugitivo | and i attacked you |
00:19.57 | fugitivo | bah |
00:22.13 | *** join/#asterisk jbot (i=ibot@rikers.org) |
00:22.13 | *** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org |
00:22.15 | blitzrage | trixter: I've been on IRC for years, I know where you're coming from. My statement still stands. |
00:22.23 | Qwell | blitzrage: You just HAD to start the os/distro war, didn't you? :) |
00:22.26 | trixter | even though iuts inaccurate? |
00:22.30 | trixter | you dont know where I am coming from |
00:22.34 | [hC] | Qwell: blasphemy! osx rules too! you suck for not liking it! |
00:22.36 | blitzrage | Qwell: just changing the flame war to something else, then I'll leave :D |
00:22.43 | Qwell | blitzrage: you're a pro |
00:22.45 | mog_work | qwell is infaliable |
00:22.48 | mog_work | like the pope |
00:22.53 | Qwell | which one? |
00:22.56 | trixter | I am trying to gain understanding of the mindset that inhibits choice, that actually takes a bit of backbone |
00:28.40 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
00:28.40 | *** topic/#asterisk is Asterisk 1.2.1 has been released! -//- http://www.asterisk.org |
00:28.44 | mog_work | im not just closing em |
00:28.54 | JohnnyC | Anyone works with phpagi ? |
00:29.18 | wunderkin | hes putting little smilie faces on them too |
00:29.18 | trixter | I have |
00:29.32 | trixter | ahh well I would rather have a smiley face than working code |
00:29.51 | trixter | why I have 5M of patches to asterisk that I have to install before delviering a system |
00:29.53 | mog_work | hey qwell your a marshall... you looked at http://bugs.digium.com/view.php?id=5443 |
00:30.13 | *** join/#asterisk imran (n=codentes@cpe-68-206-53-16.houston.res.rr.com) |
00:30.18 | mog_work | 5M of patches? |
00:30.24 | trixter | yeah for various things |
00:30.25 | mog_work | thats like half size of asterisk proper |
00:30.29 | mog_work | what are you doing |
00:30.38 | mog_work | are they 0 padded? |
00:30.39 | trixter | fiuxing a lot of broken stuff |
00:31.01 | Qwell | mog_work: I don't even see files on there |
00:31.06 | mog_work | yeah i know |
00:31.09 | Qwell | erm, nm, heh |
00:31.13 | Qwell | I didn't read the lines before that |
00:31.13 | mog_work | but if i assigned it to you |
00:31.17 | mog_work | would you deal with it |
00:31.18 | trixter | the CDR problem where a call is marked failed when its really completed is one fine example that needs to be fixed before distributing a system |
00:31.24 | Qwell | mog_work: I don't use that skinny |
00:31.26 | mog_work | or just assign it to yourself |
00:31.30 | Qwell | I use chan_sccp from berlios.de |
00:31.30 | mog_work | and... ^_ |
00:31.31 | mog_work | ^ |
00:31.47 | [hC] | is skinny even worth keeping in *? why not pull in chan_sccp? |
00:31.51 | mog_work | and you havent gone and fixed it because? |
00:32.14 | Qwell | [hC]: I've been trying, heh |
00:32.15 | mog_work | it is as the one from berlios is in legal contest i thought |
00:32.43 | mcquaid | how do i install chan_sip2? |
00:33.02 | [hC] | urg. legal contest as in how he got it to work, or licensing conflicts with *? |
00:33.09 | mog_work | no not on our end |
00:33.19 | mog_work | i wasnt sure if the code was all legal or licensed correctly |
00:33.21 | mog_work | on their end |
00:33.26 | [hC] | ah |
00:33.26 | mog_work | i rember reading that months ago |
00:33.51 | *** join/#asterisk anonymouz666 (n=lynx@200.218.193.6) |
00:34.14 | Qwell | well, I asked the guy about it...I forget what he said |
00:34.24 | mog_work | oh? |
00:34.36 | mog_work | i imagine the fargo would care about it qwell |
00:35.00 | Qwell | I think it originally came from asterisk proper |
00:35.09 | Qwell | then it turned into something else, then into what he's got |
00:35.18 | mog_work | thats what i heard too |
00:35.21 | Qwell | lemme peek at the copyright notices |
00:35.28 | [hC] | man. i just need a reliable ssh tunnel that will restart itself if it gets killed. this is not behaving. |
00:36.00 | OloBola | should I just parse agi_request to get variables passed from dialplan to agi? exten => 1,2,agi,test.php|I_NEED_THIS |
00:36.06 | Qwell | * The original chan_sccp driver that was made by Zozo which itself was derived from the chan_skinny driver. |
00:36.07 | Qwell | <PROTECTED> |
00:36.32 | ManxPowe | OloBola, That is not a variable, that is an arguement |
00:36.44 | OloBola | argument |
00:37.12 | ManxPowe | so look up how to access "command line arguements" in PHP. |
00:37.21 | Qwell | does chan_skinny even work? |
00:37.24 | Qwell | like...well? |
00:37.58 | mog_work | yes |
00:38.07 | mog_work | but not with features is my understanding |
00:38.24 | Qwell | chan_sccp has a bunch of nice features |
00:38.27 | mog_work | do you have a 7920? |
00:38.31 | Qwell | I wish |
00:38.35 | Qwell | I'd LOVE to test one of those |
00:38.39 | mog_work | damn i think we have one |
00:38.45 | mog_work | im gonna see if i can replicate bug |
00:39.57 | brockj49464 | Got a question...I need to use dtmfmode=rfc2833 before my register string but when I use AMP it forgets about that. Any other place I can put it? |
00:40.18 | *** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it) |
00:40.42 | [hC] | I have a 7960, 7914, and 7970 running on chan_sccp right now |
00:40.43 | Ariel_ | brockj49464, in the sip.conf file at the top part you can put dtmfmode=rfc2833 |
00:41.13 | Qwell | [hC]: cool |
00:41.20 | Qwell | [hC]: the berlios one? |
00:41.26 | Qwell | Try my realtime patch. :) |
00:41.28 | [hC] | Qwell: yup. |
00:41.31 | tzafrir_laptop | brockj49464, detete that peer and add it in sip_custom.conf :-( |
00:41.34 | [hC] | I dont use realtime yet. |
00:41.49 | [hC] | I keep getting ready to try it, and people go 'eeeeaaghh realtime' |
00:41.52 | [hC] | so i put it off |
00:42.21 | Qwell | I love realtime |
00:42.32 | [hC] | I'll have to pick your brain about it sometime soon |
00:42.45 | [hC] | Im gonna be redoing our infrastructure a bit soon |
00:42.48 | [hC] | to use realtime |
00:42.56 | [hC] | and be very db driven for config and provisioning |
00:43.07 | [hC] | tie it into our client database, etc |
00:43.28 | Chiardon | hewllo |
00:43.37 | trixter | is there a list of all the things that are different with realtime, especially things that no longer work quite the same |
00:43.40 | Chiardon | hey, anybody knows how do I find out which R2 variant do I have? |
00:43.41 | trixter | or at all |
00:43.56 | [hC] | What the hell is an R2 |
00:43.56 | trixter | I recall people talking about that on the mailing lists but didnt know if anyone compiled a list in one place |
00:44.01 | [hC] | I'd tell you if i knew what it was |
00:44.02 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
00:44.06 | trixter | [hC]: prolly ccitt5 r2 signalling |
00:44.27 | trixter | as opposed to R1 which is basically what a bluebox did |
00:45.22 | Chiardon | some kind of legacy protocol |
00:45.46 | wunderkin | nvbackgrounddetect isnt working with sip, it does on a pri though :( |
00:45.53 | troyb | how do i dial an @ sign? |
00:46.17 | marcus2 | how 'stable' is AEL? |
00:46.41 | Qwell | marcus2: it 'works' |
00:46.48 | Qwell | try the new AEL2 parser |
00:47.01 | *** join/#asterisk bn-7bc (n=a@pppoecl82052.minlos.no) |
00:47.02 | Qwell | http://bugs.digium.com/view.php?id=6021 |
00:47.10 | mog_work | it works just fine, it converts it to dial plan |
00:47.18 | mog_work | there is no 'stableness' to it |
00:47.25 | mog_work | however not everything is implemented |
00:47.28 | marcus2 | ah ok |
00:47.34 | mog_work | like im not sure if hints made it in yet |
00:47.47 | mog_work | but you can still do all of the things that are not done in extensions.conf |
00:47.48 | Qwell | I think AEL2 does hints... |
00:47.50 | mog_work | as it reads both |
00:47.52 | Qwell | I'm not sure |
00:48.37 | marcus2 | maybe i will just stick with old style dialplans |
00:48.54 | marcus2 | i'm finally getting around to upgrading my personal asterisk box from 1.0.7 |
00:49.52 | mog_work | ahh |
00:49.56 | trixter | ael2 has the ability to include other files and even has a cli parser so you can check syntax before laoding into asterisk, and claims to have a better syntax checking, although I havent tried it yet the aditional stuff looks nice |
00:49.58 | mog_work | ael is cool as crap |
00:50.20 | marcus2 | maybe i will migrate my work install to ael sometime tho |
00:50.57 | bn-7bc | to any developers in here: thank you for a great peace of software, I installed it just as a test, and it works like a dream, this is IMHO a vary good axample tha FOSS works |
00:51.08 | mog_work | but as my dial plan here is 4 lines |
00:51.12 | mog_work | i dont have much need for it |
00:51.16 | mog_work | yay! bn-7bc |
00:51.59 | tzanger | mog_work: hahaha |
00:52.06 | tzanger | what's your dialplan, set callerid and dial? |
00:52.09 | mog_work | i am debating realtime |
00:52.19 | tzanger | don't do it man |
00:52.25 | marcus2 | i've been pondering realtime for work too |
00:52.29 | tzanger | it's evil... EEEEEEvil |
00:52.33 | mog_work | send caller id to jabber, see if i want to take call, take call or send it to vm |
00:52.33 | Qwell | realtime <3 |
00:52.33 | marcus2 | but really i'm more interested in some good ldap integration |
00:52.45 | tzanger | but yeah the recent changes you have made to the realtime interface eliminate pretty much all my arguments against it |
00:52.52 | tzanger | marcus2: yes that too |
00:53.00 | tzanger | only really for callerID lookup for me though |
00:53.05 | tzanger | I wouldn't store a dialplan or IVR in LDAP |
00:53.06 | Qwell | tzanger: seen func_odbc? It's wonderful |
00:53.11 | tzanger | no |
00:53.49 | marcus2 | yeah, callerid lookup is a primary use, but also sip/iax2 auth and some aspects of the dialplan |
00:54.11 | tzanger | yes that too, I had forgotten about that which is actually troubling :-) |
00:54.36 | Qwell | Ariel_: missing out |
00:54.44 | Ariel_ | Qwell, not really |
00:54.50 | marcus2 | almost all othre aspects of our IT infrastructure "just work" for users once they are added to lda |
00:55.11 | marcus2 | it would be nice if they could use a sip phone and have their extension routed to it as soon as they wre in ldap |
00:55.11 | tzanger | I have recently put my CDRs in postgres |
00:55.34 | marcus2 | alos, voicemail configs in ldap could be nice |
00:56.03 | tzanger | I need to write a patch for cdr_pgsql that will queue 'em up in batch mode if for some reason it couldn't write to the db |
00:56.15 | *** join/#asterisk sneak (n=sneak@datavibe.net) |
00:56.27 | *** join/#asterisk iq (n=iq@71-38-75-128.omah.qwest.net) |
00:57.00 | Ariel_ | it's getting too cold tonight. argh can you guys up north of here shut the door on the frig.. please |
00:57.14 | troyb | hey guy's for some reason im having trouble calling SIP url's |
00:57.24 | tzanger | Ariel_: :-) |
00:57.26 | troyb | it's a cisco 7940 |
00:58.02 | *** join/#asterisk locid (n=locid@206-248-133-11.dsl.teksavvy.com) |
00:58.16 | Ariel_ | ~weather ktmb |
00:58.38 | alephcom | Ariel: We're trying to help you get rid of the bugs |
00:58.49 | Ariel_ | 52 is cold for me.. it's suppose to get donw to 40... |
00:59.07 | Ariel_ | alephcom, I don't have any heaters in my house.. argh |
00:59.12 | tzanger | it's -1C here right now |
00:59.25 | Qwell | 68F...gotta love CA |
00:59.34 | tzanger | yeah I love .ca :-) |
01:00.28 | Ariel_ | Qwell, nice... but it's only for 2 days that it gets this cold. |
01:00.46 | alephcom | Ariel: I can understand you then. :-) |
01:01.27 | Ariel_ | alephcom, I like cold weather if your homes are able to take it. But most down here are not really made for cold weather... |
01:01.49 | troyb | Ariel_ more like if you can afford the heating bill *grin* |
01:01.50 | Qwell | Ariel_: You're in what, FL? |
01:02.45 | Ariel_ | Qwell, South Florida. actually Homestead.... |
01:03.44 | mcquaid | is it still possible to compile chan_sip2? or necessary? |
01:04.26 | Ariel_ | why??? |
01:04.45 | Qwell | I need JD, and quick |
01:04.57 | tzanger | I'm exhausted |
01:05.05 | tzanger | I was out in the snow with the kids |
01:05.11 | Ariel_ | it's 2nd game start time..... |
01:05.29 | tzanger | I found it is actually easier to weed in the winter than the summer |
01:05.46 | tzanger | pulling out bigass burr bushes |
01:06.00 | tzanger | heh |
01:06.18 | *** join/#asterisk zhao__ (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
01:07.11 | *** join/#asterisk sneak (n=sneak@datavibe.net) |
01:08.04 | slappingt | wW had a foot of snow in December here in Missouri. Today it was in the 60's. |
01:09.05 | Ariel_ | slappingt, our high was 65 that is the lowest it's been for a high in a very long time here. |
01:11.44 | slappingt | cold in the 60's there and people here broke out the shorts and short sleeves. |
01:13.07 | Ariel_ | burr |
01:13.32 | *** join/#asterisk _Sam-- (n=sam@phone2.kneedraggers.com) |
01:17.35 | *** part/#asterisk slappingt (n=randygre@pcp03933849pcs.sthind01.mo.comcast.net) |
01:23.51 | harry8 | has anyone had problems installing zaptel on FC4? |
01:23.59 | Qwell | harry8: none |
01:24.04 | Qwell | What problems are you seeing? |
01:24.04 | harry8 | really? |
01:24.20 | harry8 | I tried compiling from source but the problem has to do with the kernel sources |
01:24.26 | harry8 | and then when I tried an RPM |
01:24.28 | Qwell | so install the kernel sources |
01:24.32 | harry8 | i did |
01:24.38 | harry8 | but i keep getting this error |
01:24.48 | harry8 | Module zaptel does not exist in /proc/modules |
01:25.00 | harry8 | missing /dev/zap! |
01:25.02 | Qwell | on an rmmod |
01:25.11 | Qwell | harry8: udev? read README.udev |
01:25.28 | harry8 | what's udev? |
01:25.35 | Qwell | like devfs, but not |
01:25.44 | Qwell | (any new distro likely uses it) |
01:25.44 | mcquaid | is there an up to date listing of all cmds for * besides at the wiki |
01:26.04 | Qwell | mcquaid: type help at the CLI, or press tab, or ?, and show applications |
01:26.07 | Qwell | and show functions |
01:26.08 | Qwell | etc |
01:26.27 | ManxPowe | mcquaid, you mean like "show applications" in the Asterisk CLI? |
01:26.59 | mcquaid | sorry i mean all variables, that you could use say in sip.conf |
01:27.16 | Qwell | sip.conf.sample |
01:27.52 | harry8 | thanks taking a loook at README.udev |
01:29.03 | mcquaid | Qwell, outboundproxy isn't documented in sip.conf.sample |
01:29.21 | harry8 | Qwell: did you compile the zaptel devices or use an RPM? |
01:30.12 | Qwell | harry8: compile |
01:30.23 | Qwell | weekly, heh |
01:30.41 | Qwell | several times a day for * |
01:30.50 | harry8 | so do you have to do a full linux compile of the kernel before you use zaptel? |
01:31.05 | harry8 | fc4 doesn't come with the kernel-sources rpm |
01:31.28 | harry8 | so I find that to even compile zaptel i have to go through the entire kernel compilation process |
01:31.34 | harry8 | FC1 was flawless |
01:32.00 | Qwell | kernel-headers |
01:33.41 | *** join/#asterisk redd (n=redd@81-178-129-92.dsl.pipex.com) |
01:33.46 | *** part/#asterisk redd (n=redd@81-178-129-92.dsl.pipex.com) |
01:34.21 | harry8 | hmm |
01:34.23 | mcquaid | i'm trying to ensure outboundproxy is set, i should see that when i start * no? |
01:34.34 | harry8 | i have installed glibc-kernheaders-2.4-9.1.94.i386.rpm |
01:35.16 | harry8 | but the linux/version.h file only gets made when I do a make |
01:36.30 | Qwell | harry8: Does FC4 still have a kernel-headers package? |
01:37.01 | harry8 | when I did a search for kernel-headers on rpm.phone.net |
01:37.04 | ManxPowe | mcquaid, Asterisk does not support "outbound proxy" in the traditional sense of SIP. |
01:37.18 | Qwell | harry8: no, use yum |
01:37.26 | harry8 | ? |
01:37.29 | harry8 | oh |
01:37.43 | mcquaid | ManxPowe, isn't that what this is?: http://www.voip-info.org/wiki-Asterisk+SIP+chan_sip2 |
01:37.56 | mcquaid | near bottom |
01:38.09 | ManxPowe | mcquaid, I believe chan_sip2 is still in development. |
01:38.21 | ManxPowe | you can check sip.conf.sample in the asterisk source |
01:38.35 | mcquaid | yes but i read that most of chan_sip2 is ported to chan_sip |
01:38.40 | mcquaid | i checked sample, no mention |
01:38.41 | harry8 | i always wondered what yum was for heheh |
01:38.43 | file | chan_sip2 doesn't exist anymore |
01:38.52 | mcquaid | but outboundproxy and port is in chan_sip.c |
01:39.24 | mcquaid | i realize that chan_sip2 no longer exists, trying to confirm if this variable made it over |
01:39.37 | mcquaid | it seems like it did as it's in the c file |
01:39.54 | mcquaid | but can't find it documented anywhere else |
01:40.27 | mcquaid | i need it for my voip provider |
01:40.46 | ManxPowe | mcquaid, then you would be one of the few. |
01:41.02 | Qwell | outboundproxy is in chan_sip, yes |
01:41.07 | mcquaid | hmm, maybe i'm on the wrong path |
01:41.18 | mcquaid | right now, i can call my * box with my sip client fine |
01:41.28 | mcquaid | i can call out to my voip to a pstn number |
01:41.41 | Qwell | "You can't have a dynamic outbound proxy, you big silly head" |
01:41.42 | mcquaid | but if i call in from pstn i see asterisk answer it but hear nothing either way |
01:41.45 | Qwell | heh |
01:42.03 | ManxPowe | mcquaid, What is your provider? |
01:42.05 | Qwell | it's a general option |
01:42.13 | mcquaid | vbuzzer.com |
01:42.31 | Qwell | [general] \n outboundproxy=ip \n outboundproxyport=1234 |
01:42.51 | ManxPowe | in sip.conf, of course |
01:42.53 | mcquaid | it definitely was a general option for chan_sip2 but according to wiki it's a peer option now |
01:42.58 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
01:43.03 | Qwell | mcquaid: according to the code, it's a general option |
01:43.10 | mcquaid | ok i'll try there again |
01:43.19 | mcquaid | how can i verify it's being set? |
01:43.31 | Qwell | <PROTECTED> |
01:43.31 | Qwell | <PROTECTED> |
01:43.34 | Qwell | dunno |
01:43.52 | *** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
01:44.10 | mcquaid | damn, i see asterisk answering it, but no sound |
01:44.15 | Qwell | okay, it can be a peer option too |
01:44.23 | Qwell | but it's kinda funky, and very similar to host |
01:44.29 | ManxPowe | mcquaid, no sound is usually a codec mismatch or a NAT problem |
01:45.10 | mcquaid | well, in the two clients i've got working with this voip provider (linphone, twinklephone) it only worked once i plugged in the outboundproxy |
01:45.17 | mcquaid | i didn't need to enable nat nor stun |
01:45.30 | mcquaid | trying to get the same going with asterisk |
01:45.49 | mcquaid | i'm new to this so i might be on the wrong path, but i can't think of any other reason why it's not working |
01:45.59 | mcquaid | when i debug rtp i see packets going back and forth |
01:46.08 | harry8 | kernel-smp-devel-2.6.11-1.1369_FC4.i686.rpm fixed the problem thanks |
01:46.14 | mcquaid | and as i said, when i call out i can hear/speak fine |
01:46.17 | harry8 | now i'll try to figure out how to use YUM |
01:46.23 | Qwell | harry8: kernel-devel...silly Redhat |
01:46.29 | ManxPowe | mcquaid, Is the IP info inside the packets the internal or external address of your network? |
01:46.30 | Qwell | never would have said to install that |
01:47.02 | marcus2 | wtf |
01:47.06 | marcus2 | i have an 888 DID from nufone |
01:47.22 | marcus2 | when calls come in on it now, the callerid is set to some number i've nevr seen before |
01:47.26 | marcus2 | always the same, it seems |
01:47.46 | mcquaid | checking |
01:49.03 | mcquaid | ManxPowe, internal |
01:49.19 | mcquaid | but i was trying to even just have asterisk to answer and play a file |
01:49.30 | ManxPowe | mcn, there's your problem |
01:49.33 | mcquaid | for those it said my voips ip obviously |
01:49.35 | ManxPowe | mcquaid, that s |
01:49.53 | mcquaid | any suggestions how to fix? i've already tried setting the externip and local |
01:50.29 | Ariel_ | nat=yes |
01:50.44 | mcquaid | tried that |
01:51.47 | *** join/#asterisk fgravato (n=frankie@mail.serversforless.com) |
01:52.26 | *** part/#asterisk fgravato (n=frankie@mail.serversforless.com) |
01:52.30 | *** join/#asterisk fgravato (n=frankie@mail.serversforless.com) |
01:52.33 | fgravato | hello |
01:52.45 | fgravato | anyone know if you can do more then one bindaddr in iax.conf? |
01:52.57 | fgravato | for both internal & external ips |
01:53.56 | Qwell | fgravato: 0.0.0.0 |
01:54.01 | mcquaid | or could siproxd maybe help here? |
01:54.10 | Qwell | then let iptables take care of blocking |
01:54.47 | mcquaid | i thought for sure it was outboundproxy, but that doesn't seem to help |
01:54.55 | Qwell | mcquaid: pastebin an rtp debug |
01:55.01 | mcquaid | ok |
01:55.02 | Qwell | You just get no audio, right? Everything else works? |
01:55.06 | mcquaid | yes |
01:55.14 | mcquaid | no audio on incoming from outside |
01:55.25 | Ariel_ | ports blocking |
01:55.27 | Qwell | and you're positive you've got the rtp ports open? |
01:55.32 | Qwell | AND forwarded? |
01:55.34 | mcquaid | audio on outgoing calls to pstn, audio if i call from sip phone to * server |
01:56.32 | Ariel_ | incoming from outside voip provider |
01:57.20 | mcquaid | Qwell, http://pastebin.ca/36146 |
01:57.32 | Qwell | answer the other question, then I'll look at it :p |
01:57.41 | mcquaid | and i am behind a firewall, i have no access to it. |
01:57.50 | Qwell | Then you're toast |
01:57.56 | mcquaid | i thought i can get other sip clients working fully, why not * |
01:57.57 | X-Rob | heh |
01:58.15 | Ariel_ | X-Rob, long time no see. how are you doing? |
01:58.16 | Qwell | What is 10.1.0.12? |
01:58.30 | Qwell | and did you change your RTP ports? |
01:58.30 | mcquaid | my ip behind lin server |
01:58.34 | X-Rob | Ariel_, not bad. Been busy over xmas/ny |
01:58.34 | Ariel_ | canreinvite=no |
01:58.39 | file[laptop] | ...hi |
01:58.44 | Qwell | file[laptop]: http://pastebin.ca/36146 fix |
01:58.45 | Qwell | :P |
01:59.01 | file[laptop] | externip/externhost AND localnet set? RTP ports forwarded? |
01:59.05 | file[laptop] | if not - ha too bad |
01:59.08 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
01:59.25 | Qwell | file[laptop]: No ":(" |
01:59.33 | Ariel_ | file[laptop], he has no access to his fw |
01:59.40 | mcquaid | i've tried setting externip, and local |
01:59.50 | file[laptop] | then your firewall is probably blocking the RTP stream |
01:59.52 | mcquaid | i've done nothing with rtp ports |
02:00.01 | file[laptop] | no audio foo you! |
02:00.12 | file[laptop] | Hackers is on |
02:00.12 | mcquaid | doubtful, i can call out. and as i said why do sip clients work fine |
02:00.47 | file[laptop] | doubtful? there's no RTP packets coming in to you from the provider :P |
02:00.51 | Ariel_ | mcquaid, outbound opens the rtp port for outbound calls but it's not opening for incoming |
02:01.11 | mcquaid | in the sip clients i've tried, i dont enable nat or stun, just put in the outbound proxy and they both worked |
02:01.17 | Ariel_ | firewalls most of the time will allow outbound first then inbound. But not inbound first |
02:01.42 | file[laptop] | oh well, I've said what I've said... |
02:01.43 | mcquaid | then i wouldn't be able to use sip clients directly to the voip no? |
02:01.59 | mcquaid | if it was a firewall issue |
02:02.07 | file[laptop] | do this |
02:02.16 | file[laptop] | make sure localnet and externip/externhost is set |
02:02.19 | file[laptop] | do a sip debug |
02:02.27 | perlmonkee | after placing a call, and having it connect, I press *1 - the asterisk console tells me that I have started recording. |
02:02.30 | file[laptop] | and see what Asterisk says to use for the IP and port in the SDP content |
02:02.43 | perlmonkee | The .wav files appear in the proper place under /var/spool |
02:02.47 | mcquaid | ok |
02:02.50 | perlmonkee | when I stop recording, the files are merged. |
02:02.52 | file[laptop] | if it's your public IP address, and an rtp debug still says no RTP coming from your provider - then chances are it's getting blocked at your firewall |
02:02.57 | perlmonkee | (automagically by asterisk it would seem) |
02:03.02 | perlmonkee | I now try to play this resulting wav file |
02:03.07 | perlmonkee | and it contains several minutes of total silence. |
02:03.17 | mcquaid | i was setting externip but not externhost before |
02:03.28 | file[laptop] | externhost is if you use a dynamic DNS host |
02:03.37 | file[laptop] | externip is for IP address |
02:03.42 | file[laptop] | you just need one |
02:03.44 | mcquaid | no fixed ip |
02:03.48 | file[laptop] | but you have to use it in connection with localnet |
02:03.51 | file[laptop] | or else it's useless |
02:03.57 | perlmonkee | can anyone help me with automon and call recording? |
02:04.13 | file[laptop] | Qwell: wanna help me hax0r teh g1b50n? |
02:05.49 | Qwell | file[laptop]: k |
02:06.14 | mcquaid | file[laptop], ok i did what you said |
02:06.28 | file[laptop] | pastebin, pastebin, it's so cool |
02:06.39 | mcquaid | one sec |
02:07.30 | Qwell | perlmonkee: What are you trying to play it with? |
02:08.13 | mcquaid | file[laptop], http://pastebin.ca/36147 |
02:08.35 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
02:10.29 | file[laptop] | you are indeed sending public IP in SDP |
02:10.53 | mcquaid | thats good right? |
02:11.08 | jahani | how can i use asterisk to send fax by email? |
02:11.23 | file[laptop] | yes, it means they are sending audio to you on that IP |
02:11.33 | Qwell | but the firewall is eating it! |
02:11.43 | mcquaid | weird, cause i hear nothing either way |
02:11.57 | file[laptop] | jahani: well, first you research... as there is documentation, and scripts and stuff to do... exactly that! |
02:12.46 | mcquaid | why isn't the firewall eating it, with a basic sip client connected directly? |
02:13.16 | mcquaid | there must be some way to mimic what these sip clients are doing differently |
02:13.57 | jahani | file[laptop] where i can download the script? |
02:14.07 | file[laptop] | jahani: I don't have links here, why don't you Google? |
02:14.24 | Ariel_ | argh it's too early to be this cold outside tonight...5.0°C (41°F) |
02:14.26 | file[laptop] | for example I just typed "asterisk fax e-mail" into Google, and the first result is: |
02:14.33 | file[laptop] | http://www.voip-info.org/wiki-Asterisk+Fax+to+email |
02:14.46 | file[laptop] | which has a macro, and a script |
02:14.51 | file[laptop] | to do exactly what you want :) |
02:15.34 | jahani | ok thank you very much |
02:16.37 | mcquaid | file[laptop], my sip.conf if you could look and see if something doesn't look right |
02:16.40 | mcquaid | http://pastebin.ca/36148 |
02:18.10 | file[laptop] | never used an outbound proxy myself |
02:19.21 | Qwell | file[laptop]: watch with me! |
02:19.33 | file[laptop] | Qwell: you just want to escape your family |
02:19.37 | Qwell | I do |
02:19.40 | file[laptop] | toooo bad! |
02:19.45 | Qwell | pwned |
02:21.03 | mcquaid | how well does * work with siproxd? i'm thinking on trying that |
02:21.20 | Qwell | mcquaid: Does it follow standards? |
02:21.27 | mcquaid | not sure if i can run siproxd behind the firewall |
02:21.33 | mcquaid | i believe so |
02:21.38 | Qwell | mcquaid: You'll probably have the same problem with that |
02:21.59 | mcquaid | in it's documents it mentions * not supporting outboundproxy but they're probably outdated |
02:22.14 | mcquaid | damn |
02:22.30 | mcquaid | but yet vbuzzers client works fine, so does twinkle, and linphone |
02:22.37 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
02:22.38 | Qwell | For incoming calls? |
02:22.41 | mcquaid | how are they getting around nat/firewall issues |
02:22.43 | mcquaid | yes |
02:22.45 | Qwell | It's a whole different deal with incoming vs outgoing |
02:22.49 | mcquaid | they work for both in/out fine |
02:23.42 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
02:24.06 | harry8 | hey Quell, what is a good development tool to use for developing asterisk? |
02:24.21 | harry8 | i am new to the linux dev arean |
02:24.25 | harry8 | arena |
02:24.49 | harry8 | is Eclipse a good editor for c? |
02:26.30 | sivana | <PROTECTED> |
02:28.47 | Qwell | harry8: I just use nano, heh |
02:29.35 | harry8 | where is a good place to get started just by looking at the source? |
02:29.45 | Qwell | harry8: doxygen docs |
02:29.51 | tzanger | harry8: somewhere where you have an interest |
02:29.55 | tzanger | harry8: I started in zaptel |
02:30.02 | tzanger | then pulled in chan_zap |
02:30.04 | harry8 | I am interested especially in the part i mentioned earlier about the soft video client + hard phone |
02:30.06 | tzanger | jumped into chan_iax2 |
02:30.07 | harry8 | integration |
02:30.15 | tzanger | dropped down to channel.c |
02:30.18 | harry8 | similar to the cisco VT Advantage thing |
02:30.23 | tzanger | (or rose to channel.c, I guess it's higher level) |
02:30.29 | tzanger | did some pbx.c stuff |
02:30.31 | Qwell | lower level |
02:30.40 | tzanger | but you need to start somewhere where you have interest |
02:30.41 | tzanger | Qwell: ?? |
02:30.45 | tzanger | channel.c is hte highlevel channel code |
02:30.52 | tzanger | then you drop down as you get closer to the wire |
02:31.01 | tzanger | whether that wire be cat5 or cat3. :-) |
02:31.04 | Qwell | okay, long day :p |
02:31.04 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-94-42.msy.bellsouth.net) |
02:31.04 | harry8 | yes that's the part i would like to figure out not sure how difficult it is |
02:31.16 | harry8 | or if it's even possible to do with asterisk.. |
02:31.22 | tzanger | harry8: asterisk is not easy to understand but in its own sick and twisted way it does make sense |
02:31.32 | mcquaid | Qwell, any other suggestions in the light that other clients work? |
02:31.54 | Qwell | mcquaid: I saw it was sending rtp to 5062...is that what asterisk expects them on? |
02:31.59 | harry8 | tzanger: how would you approach this scenario: I have a hard phone + a soft video client |
02:32.10 | tzanger | eek |
02:32.11 | Qwell | or maybe that's just the source port |
02:32.12 | tzanger | not exactly sure |
02:32.15 | mcquaid | i noticed that too, i don't know why it's doing that |
02:32.15 | harry8 | I want to have the application so that when I receive a video call on my hard phone |
02:32.21 | mcquaid | i bound my local ip to port 6000 |
02:32.23 | harry8 | I get a video pop up on my screen |
02:32.25 | tzanger | it'd definitely be SIP and chan_sip.c eats babies, this is a documented fact. |
02:32.25 | Qwell | it surely sees them |
02:32.29 | harry8 | and vice versa |
02:33.04 | *** part/#asterisk JohnnyC (n=JoaoCorr@195-23-115-68.net.novis.pt) |
02:33.08 | tzanger | anyway I am totally wiped, I'm heading to bed |
02:33.11 | harry8 | from a high level i would guess that there would be some way to configure your SIP phone to send a message to your computer to launch a video screen |
02:33.18 | harry8 | that would be a cool feature :0 |
02:33.23 | Qwell | harry8: You have canreinvite=no, right? |
02:33.34 | tzanger | harry8: nah, just before you Dial() the sip phone you do something (System?) to notify the PC |
02:33.36 | Qwell | erm |
02:33.38 | Qwell | mcquaid: |
02:33.46 | mcquaid | yes |
02:34.16 | Qwell | mcquaid: What happens when you get an incoming call? |
02:34.21 | Qwell | in your dialplan |
02:34.30 | Qwell | Do you Dial() a SIP phone or something? |
02:34.46 | harry8 | qwell: hmm not sure let me check |
02:34.50 | harry8 | eg: |
02:34.52 | mcquaid | exten => _4.,1,Dial(SIP/home) |
02:34.54 | Qwell | Try an Answer() then an Echo() |
02:34.57 | harry8 | i have a 7960G phone |
02:35.03 | harry8 | if I pick that up |
02:35.10 | mcquaid | but before i was trying to just have * answer and play monkeys |
02:35.12 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
02:35.22 | mcquaid | it says it's playing it but no audio |
02:35.34 | Qwell | I thought it was incoming audio you didn't get? |
02:35.35 | harry8 | then a software client (like xten eyebeam?) |
02:35.44 | Qwell | harry8: wrong person, sorry |
02:35.48 | harry8 | oh |
02:35.51 | harry8 | heheh duh! |
02:35.52 | perlmonkee | can anyone help me with automon and call recording? I end up with wav files containing silence. |
02:35.56 | *** join/#asterisk jyukes (n=jameshot@pool-138-89-211-251.atc.east.verizon.net) |
02:36.20 | mcquaid | on incoming calls from outside in, i get no audio either way. if i call out to pstn i get full audio both ways |
02:36.21 | Qwell | perlmonkee: How are you playing the files? |
02:36.26 | xheliox | perlmonkee: Maybe no one is talking. ;) |
02:36.36 | Qwell | mcquaid: okay, I see |
02:36.48 | perlmonkee | Qwell: I tried with `play`, `mplayer`, and `xmms` |
02:36.56 | Qwell | perlmonkee: try to play them from asterisk |
02:37.02 | perlmonkee | xheliox: There was plenty of talking =/ |
02:37.03 | jyukes | when i do Ringing in dialplan it gives SIP msg 180 with no audio.... how do u get 183 Ringing with 1 way audio? |
02:37.25 | xheliox | perlmonkee: Maybe it was telepathy and you just don't realize it. |
02:37.41 | harry8 | tzanger: so technically, could you overload the Dial() functionaly to add dialing to video? or create a new function called DialWithVideo() |
02:37.45 | perlmonkee | xheliox: amusing but not helpful. |
02:37.46 | harry8 | or something like that :p |
02:37.58 | Qwell | harry8: * technically supports video |
02:38.06 | xheliox | perlmonkee: that's my purpose in life. |
02:38.20 | jyukes | does anyone know how to do Verizon-style ringback music? So when i call an extension instead of ringing, it owuld play a mp3? |
02:38.32 | jyukes | this would need to be a SIP 183 message with 1way audio |
02:39.06 | harry8 | i've seen that it supports video, i was just wondering if there are any plans to do the hard phone + soft video client integration |
02:39.19 | harry8 | that would really bring good desktop video conferencing i think |
02:39.34 | harry8 | so you don't have to have a phone that has a video camera |
02:39.38 | Qwell | harry8: * would have to split the audio and video packets, and send them to different devices |
02:39.55 | Qwell | and good luck syncing outgoing audio/video |
02:40.05 | harry8 | Couldn't it be setup |
02:40.20 | perlmonkee | harry8: What he's saying is "Nice idea, but more trouble than its worth for obvious technical reasons." |
02:40.29 | Qwell | no, not even that |
02:40.33 | harry8 | hmm |
02:40.40 | Qwell | nice idea, doable, may look like crap |
02:40.46 | harry8 | Cisco has done it with their VT Advantage product |
02:40.58 | harry8 | works very well |
02:41.15 | Qwell | harry8: the ip phone probably calls the PC (in fact, I know it does), which then calls the PBX |
02:41.31 | Qwell | So, all traffic would be done through the PC, including audio |
02:41.39 | harry8 | imagine every hard phone you have could technically become a video phone :) |
02:41.40 | Qwell | the client splits the audio from the video, and sends it to the phone |
02:41.51 | Qwell | however, the model you want, is different |
02:42.01 | harry8 | yes i know the cisco VT has the phone call the PC |
02:42.01 | Qwell | the PBX would need to split the audio and video, and also combine them |
02:42.27 | Qwell | and also...if done at a purely RTP level... |
02:42.34 | Qwell | you would almost instantly support sccp and mgcp |
02:42.42 | harry8 | hmm |
02:42.48 | Qwell | (big bonus there) |
02:42.57 | harry8 | sounds hard :) |
02:43.15 | Qwell | harry8: I'd ask on the asterisk-dev mailing list. Lay out your plans, ask if they have any advice/input |
02:43.35 | harry8 | good idea you think? |
02:43.50 | Qwell | yep, I think so |
02:44.00 | harry8 | well great that's one other person heheh :0 |
02:44.01 | Qwell | I mean...in reality, all softphones should just support it |
02:44.22 | harry8 | i just think that a feature like that |
02:44.26 | Qwell | the phone gets rtp (which contains video, and perhaps empty audio), and proceeds happily |
02:44.34 | harry8 | would really bring video and desktop conferencing to a reality |
02:44.55 | Qwell | but, like I said...good luck syncing the two |
02:45.01 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
02:45.22 | harry8 | the problem with the cisco VT advantage is that you have to have the pc connected to the IP Phone |
02:45.26 | Qwell | I don't know much about SIP or RTP, but, you may have issues if one client lags a little, or... |
02:45.41 | harry8 | agreed, but it would be a good start though heheh |
02:45.56 | harry8 | so post this on the asterisk-dev mailing list? |
02:45.58 | Qwell | harry8: The easiest way, would obviously be to do it how cisco is doing it, and write it as a seperate app, but... |
02:46.11 | Qwell | it's definitely not as cool that way |
02:46.17 | perlmonkee | Qwell: that would only handle outgoing calls. |
02:46.25 | Qwell | incoming too |
02:46.35 | perlmonkee | Unless you have the incoming calls routed to the PC, and then split the audio to the phone. |
02:46.37 | Qwell | instead of sending calls to the phone, you send them to the soft client |
02:46.39 | harry8 | hmm, maybe i can pay someone to do it |
02:46.41 | perlmonkee | right |
02:46.43 | harry8 | Ii don't think i could write this |
02:46.43 | Qwell | perlmonkee: yes, that's exactly what cisco does |
02:46.48 | *** join/#asterisk coppice (n=chatzill@166.168.17.210.dyn.pacific.net.hk) |
02:47.07 | Qwell | but, that way is cheesy |
02:47.10 | *** join/#asterisk techie (n=gus@antibala.com) |
02:47.15 | harry8 | that is a good idea though |
02:47.25 | harry8 | you definately have more control with the PC |
02:47.31 | perlmonkee | Harry8: I don't see how your plan supports outgoing Soft video client + hard phone situations |
02:47.31 | harry8 | than with an IP phone |
02:47.48 | Qwell | perlmonkee: the PBX would do the work that the cisco client does |
02:47.53 | harry8 | the pbx would still handle the setup |
02:48.02 | perlmonkee | Qwell: I understand how it could do that for incoming |
02:48.04 | perlmonkee | but not for outgoing. |
02:48.29 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
02:48.32 | Qwell | perlmonkee: [bob] \n videopartner=10.1.1.15 |
02:48.36 | perlmonkee | You place an outgoing call from your hardphone and...? * rings your softphone to setup the video channel? |
02:48.44 | Qwell | "Ahh, okay bob, I see you have a video partner...I'll send the video to him." |
02:48.50 | harry8 | yup |
02:48.52 | perlmonkee | not SEND |
02:48.55 | harry8 | that's what I was thinking |
02:49.05 | perlmonkee | Qwell: I'm talking about *you* setting up an *outgoing* video stream. |
02:49.07 | Qwell | perlmonkee: and it would also know to receive rtp packets from that |
02:49.26 | Qwell | perlmonkee: okay, it's like this |
02:49.28 | perlmonkee | if your initiating the call from the hard phone, the soft phone would have no idea. |
02:49.39 | Qwell | [bob] and [bobvideo] |
02:49.40 | harry8 | the pbx |
02:49.46 | harry8 | would tell the softphone |
02:49.49 | Qwell | bob calls * |
02:49.54 | perlmonkee | unless * rings the soft phone and says "start giving me video" |
02:50.01 | Qwell | * says "hi bob, oh, okay, hold on, let me dial bobvideo |
02:50.07 | harry8 | or you can configure the pbx |
02:50.14 | Qwell | bobvideo rings. answer bobvideo, and now you have video |
02:50.19 | harry8 | so that it already knows ahead of time that the phone and client support video |
02:50.28 | perlmonkee | this is all wonderful, but I still get silence when using automon =( |
02:50.34 | harry8 | Qwell: could you write this :) |
02:50.44 | Qwell | perlmonkee: for the third time, try playing the file from asterisk |
02:50.47 | harry8 | and add the video recording funcationality too heheh |
02:50.47 | Qwell | harry8: it'll cost you a bit. :p |
02:50.48 | Qwell | but, no |
02:50.55 | harry8 | how much ? |
02:51.04 | Qwell | I think there was work done recently to record, to do video better, or something |
02:51.14 | Qwell | harry8: I don't think I could write that |
02:51.16 | harry8 | i guess I'll email the guys at Digium |
02:51.20 | perlmonkee | Qwell: I'm sorry, I never saw you say that before. |
02:51.27 | perlmonkee | How would I play a file from inside asterisk? |
02:51.35 | Qwell | Play() from the dialplan |
02:51.40 | perlmonkee | hurrr |
02:51.54 | harry8 | qwell: thanks for the help with FC4 btw got it working |
02:51.56 | perlmonkee | Play(/full/path/to/file.wav) ? |
02:52.08 | harry8 | qwell: where's the link for asterisk-dev list? |
02:52.36 | Qwell | harry8: lists.diguim.com |
02:52.45 | Qwell | perlmonkee: relativepath/to/file |
02:52.47 | Qwell | no extension |
02:53.07 | Qwell | and steals his laptop! |
02:54.02 | file[laptop] | nooooooooo |
02:54.18 | harry8 | thanks guys |
02:54.23 | harry8 | for all the help |
02:54.36 | perlmonkee | Qwell: relative? starting from where? |
02:54.39 | Qwell | I bet file could write all that! :D |
02:54.49 | Qwell | perlmonkee: /var/lib/asterisk/sounds, or something |
02:55.06 | file[laptop] | what am I writing now? |
02:55.19 | Qwell | file[laptop]: something like Cisco's video stuff |
02:55.27 | Qwell | where it has a seperate video client, from the IP phone |
02:55.34 | file[laptop] | interesting |
02:55.38 | Qwell | very |
02:56.06 | Qwell | but, cisco's way is kinda hackish...it has to be a seperate client. Would be VERY cool to have * do it |
02:56.24 | *** join/#asterisk morale (i=russell@S010600111155e117.cg.shawcable.net) |
02:56.47 | morale | anyone know why whenever i call my home phone number, the status code it returns after the call is over is FAILED? |
02:56.50 | Qwell | not only would you have near instant support for sip, sccp, and mgcp, but also for just about any client, AND...you could like... |
02:57.01 | Qwell | you could totally have an SCCP hardphone, and a SIP video client |
02:57.40 | Qwell | which is exactly what * is good at doing, is interprotocol communication |
02:57.43 | rob0 | morale: Asterisk doesn't think the call went well. Try calling for pizza delivery. ;) |
02:58.01 | Qwell | morale: there is a bug on the tracker. |
02:58.03 | rob0 | seriously, I wondered about that too. |
02:58.04 | morale | can i make Zaptel return a nicer return code? |
02:58.07 | morale | ah ok |
02:58.17 | Qwell | I think all of them are relating to zaptel too...odd |
02:58.28 | Qwell | morale: could you read the notes, and post your findings? |
02:58.49 | Qwell | I'm starting to think it's directly related to zaptel. I think everybody has said it only happens with incoming zap chans... |
02:59.06 | Qwell | zaptel/chan_zap rather |
02:59.10 | morale | Qwell: yeah, i will have a look |
02:59.54 | perlmonkee | Qwell: I'm now playing it from inside asterisk |
02:59.55 | perlmonkee | I get silence. |
03:00.05 | perlmonkee | (also it was Playback(), not Play() ) |
03:00.13 | Qwell | morale: 5918 |
03:00.20 | Qwell | perlmonkee: right, yeah |
03:00.45 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
03:01.08 | harry8 | there i've sent it off |
03:01.09 | *** join/#asterisk iq (n=iq@71-38-75-128.omah.qwest.net) |
03:01.10 | harry8 | :0 |
03:01.34 | perlmonkee | so... I get huge valid .wav files with nothing but silence. |
03:01.37 | perlmonkee | any more ideas? |
03:02.04 | Qwell | perlmonkee: nope... |
03:02.14 | perlmonkee | =( |
03:02.17 | Qwell | automon works great for me... |
03:02.24 | perlmonkee | Then I just hate you. |
03:02.24 | [hC] | me too |
03:02.26 | Qwell | perlmonkee: does it leave the old files around? |
03:02.27 | perlmonkee | And you. |
03:02.32 | perlmonkee | the split channels? |
03:02.33 | perlmonkee | no. |
03:02.33 | [hC] | try upgrading to an i686 compiled kernel |
03:02.34 | Qwell | yes |
03:02.35 | [hC] | :) |
03:02.37 | Qwell | heh |
03:02.52 | perlmonkee | ? |
03:03.31 | morale | is there a function to end the dialplan? like.. s-ANSWERED,2,Exit ? |
03:04.15 | [hC] | Hangup ends the dialplan |
03:05.58 | *** join/#asterisk jef_ (i=fischer@p54847C7F.dip.t-dialin.net) |
03:08.00 | perlmonkee | Okay... |
03:08.13 | perlmonkee | I just started a new recording and copied the wav files out before they could be merged |
03:08.18 | perlmonkee | they contain the proper audio. |
03:08.47 | perlmonkee | so when they are merged they become no good. |
03:10.11 | morale | Qwell: i've added onto that ticket anything i can think is relevant. |
03:11.17 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
03:12.02 | harry8 | have you guys had any problems with choppy audio? |
03:12.07 | harry8 | i just finished my install |
03:12.17 | harry8 | when i access voicemail, the prompt is choppy |
03:12.22 | harry8 | but the phone conversations are fine |
03:12.35 | morale | slow machine? |
03:12.40 | dorphalsig | hey, I finally installed R2 support for *, but when I try to run ... |
03:12.41 | harry8 | uh no |
03:12.43 | dorphalsig | I get this |
03:12.56 | dorphalsig | Jan 7 21:49:57 WARNING[5896] loader.c: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging |
03:12.56 | dorphalsig | Jan 7 21:49:57 WARNING[5896] loader.c: Loading module chan_unicall.so failed! |
03:12.59 | harry8 | i am running it in vmware on a 2x dual core operton |
03:13.14 | harry8 | with 16 gig of ECC ram |
03:13.15 | harry8 | :p |
03:13.25 | SkramX | harry8: shiiit |
03:13.26 | harry8 | it's not a speed issue |
03:13.31 | SkramX | why must it run vmware? |
03:13.33 | harry8 | I didn't have this problem with FC1 though |
03:13.37 | marcus2 | yeah the ecc makes it *fast* |
03:13.46 | Qwell | in vmware? eww |
03:13.50 | dorphalsig | 16 gig? |
03:13.52 | harry8 | i am running the pbx in VMWare |
03:13.57 | harry8 | yes 16 gig |
03:14.00 | dorphalsig | shit |
03:14.09 | harry8 | yeah it's one of our vm hosts |
03:14.12 | dorphalsig | what you running, hotmail? :P |
03:14.14 | coppice | dorphalsig: chan_unicall cannot find the unicall library. If you installed it in /usr/local, is /usr/local in your /etc/ld.so.conf file? |
03:14.26 | SkramX | lol |
03:14.29 | harry8 | hehhe |
03:14.37 | perlmonkee | harry8: You sound like a UnixShell# user |
03:14.52 | perlmonkee | except they don't use vmware because they don't SUCK as much as you do apparently. |
03:14.52 | harry8 | pretty cool actully with vmware and virtual server |
03:15.04 | harry8 | unixshell? |
03:15.12 | perlmonkee | harry8: Look into XEN - it will make all the difference. |
03:15.27 | harry8 | XEN? |
03:15.41 | perlmonkee | http://www.cl.cam.ac.uk/Research/SRG/netos/xen/ |
03:15.45 | SkramX | LINUX-VSERVER.ORG |
03:15.45 | harry8 | why do I suck? what's your problem? |
03:15.47 | perlmonkee | many many times faster than VMWare |
03:15.52 | *** join/#asterisk GD_ (n=GD@ppp20-adsl-165.ath.forthnet.gr) |
03:15.56 | perlmonkee | Harry8: becuase you use VMWare instead of XEN |
03:16.21 | harry8 | thanks, how old are you? 10? |
03:16.49 | dorphalsig | coppice --> yes, just added. tried again and the same mistake |
03:17.04 | Qwell | harry8: posted a reply, with more detail |
03:17.08 | X-Rob | ooh, there's coppice. Heya steve. |
03:17.12 | coppice | well, it cannot find the library for some reason |
03:17.16 | perlmonkee | harry8: Youre just wasting so much of your hardware by using vmware. |
03:17.23 | coppice | X-Rob: you've been quiet lately |
03:17.26 | X-Rob | dorphalsig, try running 'ldconfig' |
03:17.35 | perlmonkee | the overhead to run VMWare is absurd and disgusting. |
03:17.45 | Qwell | no it isn't |
03:17.45 | dorphalsig | coppice --> may I priv ya? |
03:17.53 | X-Rob | coppice, I've been running around australia.. Heading off to Darwin on tuesday morning, then back, then up to cairns, then down to melbourne. |
03:18.03 | harry8 | fine then just give a recommendation don't direct your personal remarks at me I don't know you |
03:18.05 | Qwell | you aren't using vmware workstation, are you? |
03:18.11 | harry8 | no GSX Server |
03:18.20 | Qwell | yeah, with a machine like that, I'd hope to |
03:18.21 | coppice | dorphalsig: you'll only get the same answer in private :-) |
03:18.23 | harry8 | but i don't think it's a problem with the VM |
03:18.23 | Qwell | so* |
03:18.24 | SkramX | harry8: what is the base-os? |
03:18.26 | SkramX | Windows? |
03:18.29 | harry8 | Yes :) |
03:18.35 | SkramX | harry8: LINUX |
03:18.36 | dorphalsig | coppice --> yeah man, but ldconfig returns nothing |
03:18.37 | Qwell | okay, NOW it is a problem :P |
03:18.43 | harry8 | I have a confession, I am a Microsoft Junkie :0 |
03:18.46 | X-Rob | dorphalsig, it's mean to return nothing. |
03:18.46 | SkramX | Qwell: Indeed! |
03:18.50 | X-Rob | that means 'it worked' |
03:18.51 | harry8 | but i do use linux :) |
03:18.53 | coppice | X-Rob: I'm off to Delhi in a could of hours. travelling sucks |
03:19.02 | X-Rob | Oooh! Delhi is fun! |
03:19.04 | SkramX | harry8: come over to my place.. we will break that windows-pussy in :) |
03:19.06 | X-Rob | Lucky bugger! |
03:19.21 | coppice | X-Rob: yeah. right. |
03:19.25 | X-Rob | Next time I'm there, I'm _so_ going to make time to go to agra. |
03:19.26 | harry8 | yes I am 80% windows 20% Linux :p |
03:19.27 | X-Rob | Well |
03:19.30 | X-Rob | fun if you don't have to work |
03:19.34 | harry8 | but i do like Linux |
03:19.55 | [hC] | Im such a nerd, i get a kick out of reading new pages and new page modifications on voip-info. :( If their RSS feeds worked better, i'd subscribe to the page mods :S |
03:20.41 | X-Rob | [hC], keep that up and you'll become uberclued at asterisk. |
03:20.52 | perlmonkee | harry8: You run windows on a dual core dual processor opteron system? |
03:20.53 | dorphalsig | X-Rob --> Yet, I still get the same darn message |
03:21.03 | GD_ | hello.. could you tell me whether all HFC-S PCI A 2BDS0 cards are expected to work in NT mode? |
03:21.09 | perlmonkee | harry8: you do suck =/ |
03:21.20 | SkramX | :| |
03:21.20 | [hC] | X-Rob: yeah, i'm well on my way. |
03:21.21 | coppice | X-Rob: I'm a kind of anti-tourist. |
03:21.22 | harry8 | perlmonkee: yes one of our servers is a 2 x 2 dual core AMD |
03:21.40 | X-Rob | hurm. |
03:21.41 | SkramX | for what? a fucking asterisk pbx |
03:21.45 | SkramX | thats way overkill, bro |
03:21.47 | harry8 | no |
03:21.51 | harry8 | it's not only for the pbx |
03:21.53 | perlmonkee | no, its a VServer box. |
03:21.58 | harry8 | it is a virtual machine host |
03:21.59 | SkramX | Okay. |
03:22.01 | harry8 | :) |
03:22.02 | perlmonkee | a windows VServer box. |
03:22.08 | SkramX | Yeah, perlmonkee we sell Vservers. |
03:22.16 | X-Rob | coppice, dorphalsig -- maybe he's botched the compile somehow? Howabout LD_PRELOAD=/usr/wherever/libunicall.so |
03:22.22 | perlmonkee | thats nice, SkramX. |
03:22.40 | SkramX | My company uses http;//vserver-linux.org, it has much less overhead than virtuozzo and even less than Xen |
03:23.18 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
03:23.30 | dorphalsig | X-Rob coppice --> I did a find / -name libunicall.so |
03:23.42 | dorphalsig | and it returned /usr/lib/asterisk/modules/chan_unicall.so |
03:24.00 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
03:25.03 | perlmonkee | SkramX: I don't understand the joke. |
03:25.10 | SkramX | nothing; |
03:25.16 | coppice | dorphalsig: that does make sense. you search for libunicall.so, and get chan_unicall.so back? |
03:25.16 | SkramX | Tootles, 'Will be back later |
03:25.38 | dorphalsig | ooops |
03:25.42 | perlmonkee | SkramxX: the URL you provided, as I assume you know, forwards to microsoft.com |
03:25.45 | dorphalsig | searched for chan_unicall |
03:25.47 | dorphalsig | :$:$:$:$:$:$:$ |
03:25.47 | X-Rob | coppice, he's obviously using find --with-esp |
03:26.17 | coppice | dorphalsig "whereis libunicall.so" would be more interesting |
03:26.23 | GD_ | the bristuff packages is referred to in the asterisk's gentoo ebuild changelog. Does this mean that if I emerge asterisk with the bri use flag turned on, bristuff will be compiled as well? |
03:28.36 | SkramX | perlmonkee: oops. linux-vserver.org lol |
03:28.56 | SkramX | perlmonkee: it doesnt do that for me. |
03:29.23 | SkramX | perlmonkee: http://www.linux-vserver.org/ <-- thats what I /meant/ |
03:29.24 | perlmonkee | SxramX: yeah, I realized after I said what I did that I directly followed your link - except not - it was improper (http; <- semicolon) and thus FireFox annoyingly went to google, got the first hit for http (microsoft.com) and sent me on my way. |
03:29.40 | perlmonkee | so... we're both idiots =/ |
03:29.42 | SkramX | perlmonkee: oh, oops.. sorrry. |
03:29.44 | SkramX | :) |
03:30.18 | perlmonkee | interesting to know that microsoft.com is the first hit for http |
03:31.07 | SkramX | that is weird; indeed |
03:32.25 | dorphalsig | X-Rob coppice --> I did a find / -name libunicall.so --> just added /usr/local/lib to my ld.so.conf |
03:32.34 | dorphalsig | yet I still have the problem :s |
03:32.42 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
03:32.58 | kuku5 | What do I need to do to have 1 port PoE for a 7940g ? |
03:33.38 | X-Rob | dorphalsig, did you run ldconfig after you added it? |
03:34.40 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
03:39.25 | _Sam-- | how would i configure cdr_mysql.conf to log to more than sql database? |
03:39.33 | _Sam-- | or more than host |
03:39.36 | _Sam-- | more than 1 host |
03:39.47 | _Sam-- | sorry i cant speak english tonite |
03:39.53 | _Sam-- | log to more than 1 host or database |
03:40.26 | _Sam-- | just add another context for the second one? |
03:40.59 | X-Rob | don't think you can. |
03:41.09 | *** join/#asterisk justinu (n=justinu@207.181.0.86) |
03:42.18 | _Sam-- | i guess i could do the second one via odbc |
03:44.20 | mcquaid | anyone have experience with siproxd? |
03:46.51 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
03:48.57 | *** join/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
03:50.05 | _Vile | HI |
03:51.57 | *** join/#asterisk bmg505_ (n=leon@dsl-146-28-210.telkomadsl.co.za) |
03:56.42 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
04:01.47 | *** join/#asterisk Defraz (n=t0tal@72.24.220.144) |
04:03.41 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
04:08.56 | *** join/#asterisk anon-troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
04:09.10 | [hC] | can i use particular expressions in my dial plan like |
04:09.19 | [hC] | 10[23] to represent 102 or 103? |
04:09.23 | [hC] | like in bash? if so, how? |
04:09.44 | mog_work | exactly like that hc |
04:09.45 | [hC] | I guess thats technically a valid regular expression |
04:09.49 | [hC] | hm. |
04:09.54 | *** join/#asterisk Defraz_ (n=t0tal@72.24.220.144) |
04:10.01 | mog_work | you can do ranges |
04:10.05 | mog_work | as well |
04:10.10 | anon-troyb | i have setup my cisco 7940 with the SIP image however i want to be able to dial URL's |
04:10.24 | [hC] | I wonder why this isnt working then. |
04:10.28 | [hC] | exten => _1NXXNXXXXXX,/_10[12],Macro(voip-out,${EXTEN}) |
04:10.33 | [hC] | oops |
04:10.36 | [hC] | friggin comma |
04:10.42 | mog_work | heh |
04:11.34 | [hC] | odd. still didnt like it. |
04:12.23 | mog_work | i can double check |
04:12.28 | mog_work | but that is valid regex |
04:12.32 | [hC] | yeah.. |
04:13.15 | mog_work | yeah thats right |
04:13.18 | mog_work | looking at example |
04:13.20 | mog_work | config |
04:13.23 | [hC] | im stupid |
04:13.24 | mog_work | but thats for cid |
04:13.33 | [hC] | i removed the comma |
04:13.34 | mog_work | because its after / |
04:13.36 | [hC] | notice anything else missing? |
04:13.38 | [hC] | the ,1, |
04:13.56 | [hC] | butterfingers in vi tonight. |
04:14.32 | _Vile | vi can do that to you... |
04:14.34 | [hC] | works now, thanks for humoring me :) |
04:15.33 | mog_work | no problem we are here to help |
04:16.20 | coppice | I never knew that |
04:16.39 | justinu | heh |
04:17.19 | _Vile | shit, I thought this was an idling channel? |
04:17.25 | troyb | anyone here experienced with cisco ip phone setp? |
04:17.27 | troyb | *setup |
04:17.39 | justinu | most people are idling here |
04:17.46 | justinu | a small fraction are active |
04:17.51 | troyb | yeah makes sense |
04:17.57 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
04:18.14 | troyb | work on a saturday night.. thats depressing eh? mog_work |
04:18.18 | justinu | bigtime |
04:18.23 | mog_work | nope |
04:18.27 | mog_work | its actually pretty nice |
04:18.32 | mog_work | i get to do real work |
04:18.38 | znoG | guys, any of you sell X100P cards in .au? |
04:18.50 | _Vile | noone to bother you when noone's around.. |
04:18.50 | troyb | mog_work as long as you enjoy what you do :) |
04:18.55 | *** join/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
04:18.58 | mog_work | you probably want a real hw in australlia as the lines suck |
04:19.22 | znoG | mog_work: actually i need to buy it in AU but it's for another country |
04:19.38 | mog_work | i see |
04:19.41 | znoG | argentina, where the lines would probably suck a lot more :) |
04:19.44 | mog_work | which country? |
04:19.53 | mog_work | nah argentina isnt as bad... |
04:20.03 | troyb | anyone know how to setup URL SIP dialing? |
04:20.38 | znoG | mog_work: distinctive ring detection doesn't work with argentinian style of distinctive ring... which doesn't make the lines bad, but anyway.. :) |
04:21.06 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
04:21.10 | mog_work | if you are doing that troyb why go through asteirsk |
04:21.15 | kuku5 | What do I need to do to have 1 port PoE for a 7940g ? |
04:21.16 | justinu | what is the argentinian style? |
04:21.40 | znoG | justinu: no idea, but asterisk doesn't detect it. |
04:21.59 | znoG | justinu: if i ring number 1, or number 2, both come in with cadence 0,0,0 |
04:22.08 | justinu | hmm |
04:22.40 | troyb | mog_work im not :) but this is one of the few channels where there are people who are knowledgable |
04:23.06 | mog_work | asterisk is not meant to be a sip proxy |
04:23.11 | justinu | you can uri dial thru any sip proxy, check out ser |
04:23.19 | mog_work | but you could probably allow asterisk to route stuff like that for you |
04:23.23 | mog_work | its just pointeless |
04:23.32 | justinu | or i think you can uri dial off of fwd's proxies |
04:23.36 | troyb | mog_work yeah i just dont have a production asterisk server at my disposal |
04:23.37 | morale | <PROTECTED> |
04:23.38 | morale | ls |
04:23.48 | mog_work | you dont have a linux box? |
04:23.49 | Qwell | You can dial URIs right from you phone, can't you? |
04:24.10 | mog_work | yes but most people need proxy to get out of lan qwell |
04:24.45 | Qwell | oh, right |
04:24.58 | mog_work | that whole nat issue.... |
04:24.58 | troyb | the LAN is mine if i need to port forward or anything thats fine |
04:25.11 | troyb | mog_work there is no NAT on this network :) |
04:25.28 | mog_work | oh well than just tell your phone to allow uri dialing |
04:25.31 | mog_work | it wont touch asterisk |
04:25.32 | justinu | can you do reinvites with iax? |
04:25.33 | mog_work | and it will work |
04:25.38 | mog_work | yes justinu |
04:25.43 | justinu | cool, didn't know that |
04:25.44 | troyb | mog_work i dont mean to sound like an idiot but how do i do that :P |
04:25.45 | mog_work | well its an iax native transfer |
04:25.50 | mog_work | the path leaves the middle point |
04:25.53 | justinu | yeah |
04:25.57 | mog_work | but you also lose signalling |
04:25.59 | justinu | oh |
04:26.01 | mog_work | as iax is all together |
04:26.02 | justinu | not so good |
04:26.07 | mog_work | so it wouldnt work if you were charging |
04:26.10 | justinu | yeah |
04:26.13 | mog_work | but if its internal |
04:26.15 | mog_work | its great |
04:26.18 | mog_work | like inter office |
04:26.29 | Qwell | I wonder if sccp can reinvite...probably not |
04:26.29 | _Vile | znoG, you should have logs |
04:26.30 | justinu | any plans to change that, or would you just say use SIP? |
04:26.34 | _Vile | about not being able to set cadence |
04:26.39 | _Vile | on distincitive ring |
04:26.49 | justinu | qwell: sccp uses rtp, right? so it should |
04:27.02 | Qwell | sccp is stupid |
04:27.32 | mog_work | it has been discussed justinu |
04:27.32 | _Vile | it's set as a LOG_WARNING, so if you're logging warnings, look for "Unable to set distinctive ring cadence" |
04:27.36 | justinu | i like sccp |
04:27.38 | mog_work | like cloaning signalling |
04:27.42 | Qwell | I love sccp |
04:27.50 | troyb | mog_work how do i allow uri dialing? |
04:27.52 | Qwell | I want it to have my children |
04:27.56 | mog_work | its a phone issue |
04:27.56 | justinu | but I like SIP too, they're just two completely opposite ideas |
04:28.09 | _Vile | though, maybe it's deeper than that -- and it's not even getting to that ast_log. hm |
04:28.27 | mog_work | sip is good if there is no nat |
04:28.27 | znoG | _Vile: i start asterisk with -vvvvvv and where it should wait a couple of seconds to detect the type of ring, it just goes straight in with cadence 0,0,0 |
04:28.32 | mog_work | otherwise pttttttttttth on sip |
04:28.42 | justinu | there's solutions for that |
04:28.46 | justinu | session border controllers |
04:28.57 | Qwell | I need an sccp proxy |
04:28.58 | mog_work | i prefer iax instead |
04:28.59 | znoG | _Vile: asterisk should wait a couple of seconds when it gets a ring, AFAIK, and it's not. It just goes straight in, and usedistinctiveringdetection is definately on. |
04:29.28 | Qwell | If sccp, sip, and iax were in a cage match...you know which one would win? |
04:29.37 | justinu | sip would sit on sccp and iax, and win |
04:29.50 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
04:29.51 | trixter | sounds about right |
04:29.52 | justinu | 10,000 pages of RFCs and drafts |
04:29.56 | Qwell | pfft, sip has to stick to the RFC. No dirty fighting |
04:30.01 | troyb | mog_work how do i allow uri dialing? |
04:30.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
04:30.08 | mog_work | its a feature of the phone |
04:30.16 | mog_work | it would be different for each phone |
04:30.19 | troyb | i see though i cant find it in the system menu |
04:30.22 | wunderkin | mog_work are we there yet? |
04:30.28 | troyb | (cisco 7940) |
04:30.28 | _Vile | interesting |
04:30.51 | mog_work | are we where yet |
04:30.51 | justinu | wherever you go, there you are |
04:31.02 | mog_work | unless its the event horizon |
04:31.23 | Qwell | The matter has left the event horizon |
04:31.35 | Qwell | That doesn't work, does it? |
04:34.08 | _Vile | 7940? |
04:34.27 | _Vile | you dial, and switch to alpha I think |
04:35.02 | justinu | in the cisco world, how do you setup voip trunks? what protocol? |
04:36.22 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
04:36.37 | _Vile | mgcp or sip |
04:37.03 | justinu | ic |
04:37.10 | _Vile | oh ur kidding |
04:37.15 | justinu | no |
04:37.42 | _Vile | ok, what gateway |
04:38.00 | justinu | i dunno anything about cisco gateways, so i was curious |
04:38.16 | justinu | i need to find something that speaks SIP-T |
04:38.26 | justinu | something cheap |
04:38.29 | mog_work | you can use asterisk |
04:38.29 | justinu | or free |
04:38.33 | mog_work | go grab that patch |
04:38.36 | justinu | i coudln't get * to work with sip-t |
04:38.39 | mog_work | you mean sip tcp |
04:38.41 | mog_work | right |
04:38.42 | justinu | nope |
04:38.49 | mog_work | whats sip-t |
04:39.29 | _Vile | no |
04:39.30 | _Vile | he means: |
04:39.38 | _Vile | http://www.voip-info.org/wiki/view/RFC3372 |
04:39.41 | justinu | SIP-T is like SS7 over sip, SIGTRAN |
04:39.44 | *** join/#asterisk Shakhruz (i=Shakhruz@83.221.169.216) |
04:40.01 | _Vile | that would be isup over sip right? |
04:40.04 | justinu | yes |
04:40.08 | justinu | i need it |
04:40.10 | _Vile | that's done.... |
04:40.15 | _Vile | tmk |
04:40.30 | _Vile | shit, let me pull my notes |
04:40.32 | justinu | tmk? |
04:40.37 | _Vile | to my knowledge |
04:40.42 | justinu | ah |
04:40.42 | _Vile | there is ISUP code over SIP |
04:40.47 | _Vile | already in CVS |
04:40.49 | justinu | sweet |
04:40.53 | _Vile | or it's an add-on |
04:40.55 | _Vile | or something |
04:40.55 | _Vile | sec |
04:41.24 | justinu | level3 wants to give me SIP-T |
04:41.28 | justinu | which seems nice |
04:41.47 | _Vile | well |
04:41.49 | _Vile | its new code |
04:41.56 | _Vile | less than a few months |
04:41.58 | justinu | it's ok, all their shit is beta too :) |
04:42.01 | _Vile | ok |
04:42.05 | justinu | i'll be doing interop tests with it |
04:42.12 | justinu | so I'll probably work on the code myself |
04:44.39 | _Vile | I could be wrong here, I may be thinking of chan_ss7 ---- give me a min |
04:44.49 | _Vile | I remember isup over sip being in * |
04:45.01 | _Vile | something to do w/ verisign's sip-7 service |
04:45.07 | justinu | i tried it, and * chocked on the multi-part mime type that l3 was sending me |
04:45.08 | sivana | *sigh* |
04:48.11 | znoG | does anyone actually use distinctive ring detection in Asterisk? i would *love* to see a working zapata.conf |
04:49.26 | mog_work | i use the ring codes |
04:50.06 | troyb | damn it now my phone lines went missing off the display lol |
04:50.19 | JunK-Y | yo guys |
04:50.25 | mog_work | yo JunK-Y |
04:50.28 | JunK-Y | whats up? |
04:50.35 | file | yo yo you crazy french person :P |
04:50.37 | Qwell | JunK-Y: y0 |
04:50.39 | *** join/#asterisk sim5im (n=root@71.196.10.2) |
04:50.50 | JunK-Y | hey hey bus driver 202! |
04:50.59 | JunK-Y | lo Qwell |
04:54.02 | Qwell | JunK-Y: If you hear anything more about ETel, you should msg me, so we can work out details if we need to. Cool? |
04:54.16 | JunK-Y | all right houston. |
04:54.25 | JunK-Y | same thing for ya. |
04:54.28 | Qwell | yep |
04:54.58 | Qwell | and this time... |
04:55.04 | Qwell | I'm gonna practice playing pool... |
04:55.09 | Qwell | and I'm gonna kick yer ass! :P |
04:55.19 | JunK-Y | ya ya keep dreaming. |
04:55.19 | JunK-Y | :) |
04:55.21 | Qwell | heh |
04:55.23 | sivana | heh |
04:55.34 | JunK-Y | sivana: are ya comin' too? |
04:55.38 | Qwell | It was the lighting in that bar, I tell ya... |
04:55.45 | sivana | where we going? |
04:55.49 | Qwell | sivana: ETel |
04:55.50 | JunK-Y | etel! |
04:55.56 | sivana | in SF? |
04:55.58 | Qwell | yeah |
04:56.02 | sivana | I wish... no can do :) |
04:56.10 | sivana | just got back from 12 days in AZ/CA :) |
04:56.25 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
04:56.34 | troyb | any reason a phone would all of a sudden stop recognizing the lines and say loads error? |
04:56.46 | harryvv | nope |
04:56.53 | troyb | ironically it's still loading my .cnf because it says my custom headername.. yet my lines disappeared |
04:58.29 | file | mmm melted M&Ms |
04:58.48 | JunK-Y | time for read. see you tomorrow guys. |
04:59.38 | troyb | damn it says no load specified |
05:02.04 | troyb | also will unplugging the phone too much lead to problems? |
05:02.29 | brockj49464 | How do I debug if I am getting CIDNAME from my voip provider? |
05:02.42 | *** part/#asterisk Defraz (n=t0tal@72.24.220.144) |
05:03.49 | _Vile | justin, I don't think SIP-T has been implemented. |
05:04.06 | justinu | ok |
05:04.18 | _Vile | but |
05:04.19 | JunK-Y | dumpchan() |
05:04.19 | _Vile | check out: |
05:04.27 | _Vile | http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+faq |
05:04.35 | _Vile | http://www.voip-info.org/wiki-Asterisk+SS7 |
05:04.44 | _Vile | and http://www.voip-info.org/wiki/index.php?page=Asterisk+ss7+commands |
05:05.01 | _Vile | there is code, just needs to be worked |
05:05.56 | dorphalsig | Hello, I'm trying (without much success) to install R2 Support for *, I dont know if anybody can give me a hand |
05:06.13 | _Vile | also, search around a bit about asterisk and the Verisign SIP-7 service |
05:06.38 | dorphalsig | So far, I downloaded and compiled all the libs the wiki instructed |
05:06.46 | dorphalsig | and patched the channel makefile accordingly |
05:07.23 | justinu | vile: thanks for the pointers |
05:07.23 | dorphalsig | I'm 90% thru unicall.conf |
05:07.39 | justinu | talk to coppice about that, i think |
05:07.56 | dorphalsig | Coppice ... you there? |
05:08.21 | coppice | <PROTECTED> |
05:08.27 | dorphalsig | ok can ya help me out? |
05:08.53 | coppice | ok |
05:09.08 | dorphalsig | I am unsure which group should I put on my unicall.conf |
05:09.22 | troyb | is there a way to restart a cisco phone gracefully as opposed to pulling the cord? |
05:09.28 | _Vile | yes |
05:09.34 | _Vile | *,6,settings |
05:09.39 | _Vile | at the same time |
05:10.00 | troyb | thanks :) |
05:10.05 | dorphalsig | I am really unsure what to put there, to be quite sincere |
05:10.08 | troyb | i just get the feeling unplugging = bad |
05:10.12 | coppice | dorphalsig: groups are not a unicall issue. they are a general * issue |
05:10.27 | dorphalsig | I have some groups defined in my zapata.conf |
05:10.35 | _Vile | you can also telnet into it |
05:10.36 | _Vile | and reboot |
05:10.45 | _Vile | unplugging is ok, but *shrug* |
05:10.58 | coppice | dorphalsig: do you understand what a group is? |
05:11.02 | _Vile | I use the * command or reboot from the telnet prompt |
05:11.04 | troyb | for some reason _Vile the phone lost it's calling functions |
05:11.16 | _Vile | just one phone? |
05:11.21 | _Vile | or multiple phones? |
05:11.23 | troyb | yeah i only have one |
05:11.26 | _Vile | ok |
05:11.34 | _Vile | interesting, describe |
05:11.36 | troyb | i didnt make any changes which is weird |
05:11.48 | dorphalsig | not quite:s |
05:11.55 | _Vile | describe what you did prior to it losing its call function |
05:11.55 | troyb | it is now grabbing my load file from TFTP after a reboot (it never used to do that) |
05:12.01 | troyb | oh, |
05:12.03 | dorphalsig | I think its groups of channels with the same priviledges |
05:12.04 | dorphalsig | right? |
05:12.11 | troyb | i made changes to the proxy settings because im not behind nat |
05:12.14 | dorphalsig | or rather destined to the same thing |
05:12.18 | dorphalsig | Ie. outbound calls |
05:12.18 | _Vile | on the phone? |
05:12.21 | troyb | yup |
05:12.32 | troyb | because at that point settings wern't updating from the config file to the phone |
05:12.36 | troyb | so i made the changes directly |
05:12.43 | _Vile | yep |
05:12.48 | _Vile | now you have to make the changes in the config |
05:12.57 | _Vile | and let it tftp grab those changes |
05:13.12 | troyb | so im having these problems because the tftp config file isnt the same as the one on the phone? |
05:13.31 | _Vile | no, the phone will align itself to the config that you're letting it grab |
05:13.34 | coppice | dorphalsig: a group is a bunch of trunks which are interchangable. all the channels in a n E1 typically for a group. however, you might have an E1 where the telco has dedicated some channels to be incoming and some to be outgoing. |
05:13.42 | _Vile | you have to reconfigure the tftp config file |
05:13.44 | _Vile | for what you want |
05:13.48 | troyb | oh i agree |
05:13.49 | _Vile | and let the phone grab it |
05:13.59 | troyb | the file is sitting in the TFTP root nothing is stopping it :) |
05:14.05 | troyb | and the file is being grabbed without hesitation |
05:14.08 | _Vile | ok |
05:14.14 | _Vile | what does the phone screen show |
05:14.18 | troyb | but its now saying phone unprovisioned |
05:14.20 | _Vile | 7940 right? |
05:14.21 | _Vile | ok |
05:14.22 | troyb | yup |
05:14.27 | _Vile | so, your config file needs to be fixed. |
05:14.33 | _Vile | unprovisioned in all caps? |
05:14.35 | troyb | i wanted a 7960 but it was a little out of my reach :) |
05:14.38 | troyb | nope |
05:14.41 | _Vile | ok |
05:14.42 | troyb | Phone Unprovisioned |
05:14.45 | _Vile | can you view the tftp |
05:14.49 | _Vile | err |
05:14.50 | troyb | yes i can :) |
05:14.50 | _Vile | config file |
05:14.54 | troyb | yup i can |
05:15.01 | troyb | do you want me to do a pastebin? |
05:15.05 | _Vile | can you pastebin it, removing the password crap |
05:15.06 | _Vile | yep |
05:15.06 | _Vile | thx |
05:15.11 | troyb | no thank you :) |
05:15.13 | troyb | err |
05:15.16 | troyb | thank you |
05:15.29 | troyb | expressions dont come out right on irc :P |
05:15.43 | _Vile | and do you have TFTP logs? |
05:15.46 | _Vile | and pastebin those |
05:15.50 | _Vile | might be looking for dialplan etc |
05:15.58 | troyb | sure |
05:16.02 | _Vile | thx |
05:16.04 | _Vile | brb in 2 min |
05:16.53 | troyb | http://pastebin.com/495996 |
05:18.35 | _Vile | ok |
05:18.40 | _Vile | sec |
05:19.02 | *** part/#asterisk joat (n=joat@ip70-160-150-20.hr.hr.cox.net) |
05:19.12 | troyb | np |
05:19.17 | _Vile | where part of the screen does it say "Phone Unprovisioned"? |
05:19.23 | troyb | bottom center |
05:20.00 | _Vile | ok |
05:20.17 | _Vile | do you have TFTP url? |
05:20.27 | _Vile | pastebin'd TFTP url? |
05:20.34 | troyb | i dont understand |
05:20.45 | troyb | i have a tftp server yes.. it's internal |
05:20.48 | _Vile | you should have TFTP logs in your /var/log directory |
05:20.51 | _Vile | I need to see those |
05:20.52 | _Vile | sec |
05:21.04 | troyb | ohh i can turn logging on and restart the phone sec |
05:21.18 | _Vile | in your messages file |
05:21.30 | _Vile | ok thx |
05:21.47 | troyb | *6settings is a lot easier then reaching for the power brick |
05:21.52 | _Vile | yep ;) |
05:22.00 | Qwell | psh, SIP |
05:22.06 | Qwell | real men need to do **#** |
05:22.19 | troyb | _Vile it also worries me that if i screw that brick up another one is an arm + leg |
05:22.34 | _Vile | no worries |
05:22.40 | _Vile | relatively cheap on ebay |
05:22.56 | troyb | i hate ebay :) i have only bought one thing on it and its under dispute right now bastards |
05:22.58 | _Vile | I think we got 10 for $200 or $300 |
05:23.29 | troyb | heh |
05:23.29 | troyb | http://pastebin.com/496000 |
05:23.33 | _Vile | you can use PoE too, but more expensive |
05:23.34 | _Vile | ok brb |
05:23.37 | troyb | yeah :) |
05:30.18 | brockj49464 | _Vile: Any idea on figuring out if VOIP provider is sending CIDNAME in sip? |
05:30.33 | *** join/#asterisk inv_Arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
05:33.46 | _Vile | Do a NoOp,${CALLERIDNAME} right after your Answer |
05:33.56 | _Vile | if empty, no. |
05:34.11 | troyb | _Vile i replied to your pm |
05:34.45 | _Vile | didnt get it |
05:35.03 | *** join/#asterisk Cyon (n=cyon@cyons.net) |
05:35.03 | troyb | ohh |
05:35.06 | *** part/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
05:35.33 | *** join/#asterisk ManxPowe (i=ewieling@24.sub-70-210-168.myvzw.com) |
05:36.47 | brockj49464 | I have done a dumpvar when it sets the CallerID and CallerIDName is not set. Would it set it AFTER answer? |
05:37.10 | _Vile | it gets set after answer |
05:37.58 | *** join/#asterisk sim5im (n=root@71.196.10.2) |
05:38.01 | _Vile | if you're using an x100p, it gets set after the first ring, if it's pri or sip it's available after answer |
05:38.26 | brockj49464 | Let me check... |
05:38.33 | _Vile | or iax.. I'm leaving that out because a real man's protocol is sip ;) |
05:38.48 | _Vile | actually, I like iax |
05:39.10 | _Vile | troyb, still no messages? |
05:39.59 | Cyon | lol |
05:40.18 | troyb | i sent you a pm again |
05:40.22 | _Vile | nothing |
05:40.22 | Cyon | what if I were a woman, which is the appropriate feminine protocol to use? :-P |
05:40.26 | _Vile | troy |
05:40.26 | troyb | oh my.. |
05:40.28 | _Vile | in here is fine |
05:40.31 | troyb | okay :) |
05:40.35 | _Vile | just not multi-line |
05:40.42 | troyb | <troyb> but it says at the top Troy's IP Phone SIP |
05:40.42 | troyb | <troyb> it would of had to read the config file to get that |
05:40.46 | troyb | ohh :P |
05:40.52 | _Vile | two lines is ok |
05:40.55 | _Vile | sec |
05:40.56 | Cyon | lol |
05:40.59 | *** join/#asterisk GD_ (n=GD@ppp68-adsl-215.ath.forthnet.gr) |
05:41.10 | troyb | i read the error status of the phone and it said: |
05:41.16 | troyb | <troyb> <troyb> W351 unprovisioned proxy emergency |
05:41.17 | troyb | <troyb> <troyb> W350 unprovisioned proxy_ backup |
05:41.17 | _Vile | troy |
05:41.23 | troyb | yup |
05:41.24 | _Vile | did it say Troy's IP Phone before? |
05:41.40 | troyb | yes but i edited and rebooted just to see if it would change and it did |
05:41.50 | GD_ | hello.. has anybody come up with a solution to the unresolved symbols error which occurs after modprobing zaptel and zaphfc? |
05:42.03 | _Vile | interesting |
05:42.52 | _Vile | it reset to "Troy's IP Phone" |
05:42.56 | _Vile | or "Troy's IP Phone SIP" |
05:43.21 | Cyon | GD_: as in where if you have asterisk running, then taking either of those actions, asterisk essentially freezes? |
05:43.22 | troyb | well it says SIP in the right hand corner regardless |
05:43.30 | troyb | right now it says "Troy's IP Phone" |
05:43.31 | _Vile | oh, yes |
05:43.45 | brockj49464 | I have sip debug ip for my provider and I still don't see anything after being answered. WHat would I be looking for? Are you sure sip sends it after answer? |
05:43.48 | Cyon | GD_: That's all I know of, and it's still happening in 1.2.1 for me. |
05:43.53 | _Vile | on the right top, I expected that not to be included ok |
05:44.03 | troyb | ;) |
05:44.13 | _Vile | change it in the config, to say "blah" |
05:44.18 | _Vile | and *6settings it |
05:44.21 | troyb | okay |
05:44.23 | GD_ | no i can't get it to run.. I tried asterisk -vvvvgc and I get fatal errors.. so I thought that it might be due to my not having any drivers loaded... but loading modprobing the modules fails... |
05:45.07 | GD_ | oh well... I found somewhere of the net that some of these symbols have been deprecated as of kernel 2.6.13... do you think that might be the reason? I'm using 2.6.15 at the moment.. |
05:45.17 | troyb | its rebooting |
05:45.27 | troyb | universal application loader.. sounds cool :P |
05:45.39 | _Vile | heh |
05:45.42 | _Vile | I may have found it |
05:45.46 | Cyon | GD_: well since you have a kernel newer than the deprecation, no. ;-) Unless that was a typo |
05:45.55 | _Vile | http://pastebin.com/495996 |
05:46.02 | _Vile | # |
05:46.02 | _Vile | # Outbound Proxy info |
05:46.02 | _Vile | # |
05:46.02 | _Vile | outbound_proxy: "" |
05:46.02 | _Vile | # |
05:46.03 | _Vile | outbound_proxy_port: "5060" |
05:46.07 | _Vile | well, shit |
05:46.15 | _Vile | ok anyway, outbound_proxy needs to be set |
05:46.21 | GD_ | no I'm using 2.6.15... according to that I need an OLDER kernel... |
05:46.21 | troyb | oh really? |
05:46.24 | Cyon | GD_: And asterisk should run; test it without zaptel.conf; see if it will work when not getting any zap channels |
05:46.37 | GD_ | Jan 8 08:46:56 WARNING[19556]: chan_zap.c:924 zt_open: Unable to open '/dev/zap/channel': No such file or directory |
05:46.38 | GD_ | Jan 8 08:46:57 ERROR[19556]: chan_zap.c:6473 mkintf: Unable to open channel 1: No such file or directory |
05:46.38 | troyb | okay it says Blah on the screen |
05:46.40 | GD_ | is this normal? |
05:46.54 | Qwell | GD_: you using udev? |
05:46.59 | Qwell | GD_: README.udev |
05:47.04 | GD_ | yeap.. gentoo does by default |
05:47.06 | sim5im | outbound proxy dont need to be set unless you're using it |
05:47.07 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
05:47.09 | Qwell | (assuming the modules are loaded successfully) |
05:47.20 | troyb | _Vile what should i set the address too? my router? |
05:47.21 | _Vile | troy, set outbound_proxy |
05:47.26 | troyb | sure ^^ |
05:47.31 | GD_ | what modules? zaptel and zaphfc? well they dont!!! |
05:47.34 | _Vile | no, set it to 10.0.69.100 |
05:47.37 | Qwell | GD_: well, why not? |
05:48.00 | GD_ | because I get an unresolved symbols error after modprobing them... and they won't load |
05:48.03 | sim5im | troyb: also you need a port |
05:48.04 | _Vile | which should be your * box, or voip provider or whatever |
05:48.20 | _Vile | he's got one |
05:48.22 | _Vile | 5060 |
05:48.23 | _Vile | outbound_proxy_port: "5060" |
05:48.29 | troyb | :_ |
05:48.31 | troyb | * :) |
05:48.31 | sim5im | proxy1_port |
05:48.42 | troyb | yup thats already set to 10.0.69.100 |
05:48.43 | _Vile | bouncing sec |
05:48.44 | sim5im | or was it in the proxy1_address ip:port |
05:48.52 | sim5im | i just setup a 7960 on friday |
05:48.59 | troyb | coolness |
05:49.07 | _Vile | don't need a port on proxy1_address |
05:49.10 | _Vile | just need outbound |
05:49.15 | sim5im | but i dont have axs to the tftp at the moment , to look at the conf |
05:49.23 | sim5im | I never use outbound proxy |
05:49.35 | _Vile | I actually have proxy_emergency, proxy_backup, and outbound_proxy used |
05:49.49 | sim5im | yeah redundancy |
05:49.55 | troyb | _Vile when i look at phone status it says error proxy emergency and proxy backup not set |
05:49.56 | _Vile | but none the less |
05:50.02 | troyb | and it still says Phone Unprovisioned |
05:50.03 | _Vile | troyb |
05:50.04 | _Vile | ok |
05:50.11 | sim5im | im on a VMware emulated box right now, instaling openH323 |
05:50.11 | _Vile | now, sec |
05:50.20 | _Vile | see ur msg's |
05:50.36 | sim5im | and gonna try run asterisk on this vmware virtual linux box |
05:50.49 | troyb | _Vile got it.. does it matter that i dont have a box with that ip assigned? |
05:50.56 | _Vile | yes |
05:50.59 | _Vile | see the next msg |
05:51.01 | troyb | sure |
05:51.07 | troyb | got it |
05:51.16 | troyb | i dont have a registered nickname which is why you dont get my pm's |
05:51.36 | _Vile | ahhhhhh |
05:51.40 | _Vile | you should register |
05:51.41 | _Vile | ;) |
05:51.42 | troyb | sure :) |
05:51.51 | troyb | it seems troyb is taken. jerk, its actually my name :P |
05:52.15 | _Vile | and vile was my original nick on EF |
05:52.28 | _Vile | *shrug* |
05:52.37 | troyb1 | alright im regsiered :) |
05:52.41 | Cyon | lol |
05:52.49 | Cyon | I got Cyon. ;-) :-P |
05:53.08 | _Vile | I figured noone would ever use vile, go figure... do a whois |
05:53.09 | _Vile | ;) |
05:55.09 | Cyon | Yeah...I've had mine over 10 years...now it's a cell phone brand.. :-( http://en.wikipedia.org/wiki/Cyon |
05:56.15 | brockj49464 | vile: So should I be able to see it in a "sip debug ip (provider ip)" or not? |
06:00.44 | znoG | wikipedia is like hotmail, every dictionary word is taken |
06:01.20 | justinu | SIP-T isn't taken |
06:01.35 | justinu | i guess that's not in any dictionary but newton's tho |
06:02.57 | trixter | even all the words from the pr0n dictionaries? |
06:03.13 | trixter | stuff like snowball, dirty sanchez, hot karl |
06:03.23 | trixter | I am sure there are dictionaries you cna still add to wikipedia with |
06:03.24 | justinu | some of those cyon phones look pretty cool |
06:03.47 | Cyon | justinu: lol That makes me feel so much better :-P |
06:04.13 | justinu | better than the junk we get shoved down our throats here |
06:04.25 | Cyon | lol |
06:04.44 | Cyon | Very true; we think the US is so advanced; but we get hand-me-down technology in everything |
06:05.05 | justinu | the only reason people think we're advanced is the've never been anywhere really advanced |
06:05.16 | Cyon | Exactly |
06:05.51 | Cyon | Hell, I didn't know much better, even while reading all sorts of tech news..I just assumed I'd never bothered looking for it here. When I was in Japan, omfg. lol |
06:06.01 | *** join/#asterisk PJMattF (n=matt@cpc2-fare3-3-0-cust79.cos2.cable.ntl.com) |
06:06.07 | justinu | yeah, i'm headed there in a few months |
06:06.11 | Cyon | Sweet :) |
06:06.17 | Cyon | Speak the language? |
06:06.21 | justinu | nope |
06:06.37 | justinu | wish I could |
06:06.41 | coppice | I loved the US ads someone did for GSM roaming phones, before the US had much GSM - "works anywhere outside the western hemisphere" weird :-) |
06:06.52 | Cyon | It isn't needed at all; but just picking up some basic words helps a lot |
06:07.12 | justinu | i carry a quad band gsm phone |
06:07.21 | justinu | so wherever I go, I can just buy prepaid cards |
06:07.39 | coppice | nobody seems to have a 5 band phone yet :-) |
06:07.49 | justinu | are there 5 gsm bands yet? |
06:07.56 | Cyon | Hmmm, be careful; almost no phones are compatible with Japan's network, even if the technolgy could be, they just don't let it work. |
06:08.01 | justinu | oh really? |
06:08.04 | justinu | no GSM there? |
06:08.13 | trixter | justinu: beer and coffee are almost the same so at least you can get breakfast and afternoon beverages without too much of a problem :) |
06:08.16 | justinu | i gotta ask benjk about that, he'll know |
06:08.24 | trixter | justinu: there is gsm400 that is coming out |
06:08.24 | justinu | trixter: lol |
06:08.43 | trixter | so yeah there would be 4+ that are techniclaly available although I dont know if there are any phones that do all 4 |
06:08.43 | PJMattF | I'm having problems installing * 1.2.1 onto my new Ubuntu Breezy install - I've got it apparently building OK, but can't get the ztdummy module to work ("FATAL: Module ztdummy not found" when trying to modprobe it) |
06:09.00 | Cyon | "Note however, that due to different technologies used, mobile phones from your home country, including GSM phones, are likely not to work in Japan." |
06:09.05 | Cyon | http://www.japan-guide.com/e/e2223.html |
06:09.07 | coppice | our local CDMA rand UMTS phones roam to Japan |
06:09.09 | Nugget | "not likely" heh. |
06:09.27 | justinu | i don't want to roam, i want to be onnet to whatever is there |
06:09.29 | *** join/#asterisk [hC] (i=turnerd@66.199.130.40) |
06:09.40 | justinu | my phone issn't subsidy locked either |
06:09.54 | [hC] | anyone have a regex to match NXXNXXXXXX OR 1NXXNXXXXXX without having to explicitly specify both? |
06:09.58 | Nugget | just call your provider and rent a phone from them for while you're there. |
06:10.03 | [hC] | maybe do a [1|] or something? |
06:10.08 | Nugget | or rent a phone from your hotel and forward your number to it. |
06:10.11 | justinu | i hate my provider, they can kiss my ass |
06:10.16 | coppice | only a few countries do the loocking thing with GSM phones |
06:10.18 | Corydon76-home | [hC]: no such regex |
06:10.20 | Nugget | it'll be about $100 a day and $2 a minute. |
06:10.24 | Cyon | I rented...got it in the airport, and dropped it off when leaving |
06:10.34 | justinu | they must have prepaid service in japan |
06:10.42 | Nugget | not for gaijin. |
06:10.45 | justinu | seriously? |
06:10.52 | Cyon | Nugget: Policies have changed...now you can |
06:10.55 | [hC] | Corydon76-home: suck. :( |
06:11.03 | Corydon76-home | Or just do 1NXXNXXXXXX,1,Goto(${EXTEN:1},1) |
06:11.18 | Nugget | could be, I haven't been to japan in about 10 months. |
06:11.20 | [hC] | That would work, depending... i suppose. |
06:11.29 | trixter | I want the 7.2Mbps data that tmobile is doing in germany |
06:11.34 | [hC] | Ive got to maintain a list of every NPA in our area codes |
06:11.38 | trixter | that much bandwidth portable and wireless ... Mmmm |
06:11.50 | justinu | you know what they charge for that service? |
06:11.52 | Corydon76-home | Gee, wow, gratitude for fixing your problem. Thanks. |
06:11.54 | [hC] | because my telco plays the 'you have dialed a long distance number' instead of simply rejecting the call |
06:12.01 | Cyon | Nugget: I couldn't say when, but I walked into narita, asked for the cell phone rental at the information booth, anmd they pointed around the corner, 5 minutes later I had my cell |
06:12.02 | Nugget | I always just rent a phone from american express' vendor, which ain't cheap but is painless and hassle-free. |
06:12.03 | justinu | none of my german friends can afford wireless data |
06:12.16 | Cyon | justinu: I had it 8 days, and it was < $100 total; one second |
06:12.24 | Nugget | not bad at all. |
06:12.36 | Nugget | I recommend that, then. :) |
06:12.42 | justinu | are there wifi hotspots in japan? |
06:12.59 | Cyon | http://www.narita-airport.jp/en/guide/service/list/svc_19.html |
06:13.36 | Cyon | It seemed to me that all their rates were close enough I just didn't care who it was. |
06:13.36 | trixter | justinu: nope dont know, saw the PR release not any thing else |
06:13.36 | justinu | http://www.telecomsquare.co.jp/en/index.html |
06:13.53 | justinu | about 7 bucks a day |
06:13.58 | justinu | i guess i can deal with that |
06:14.59 | justinu | 4.2 yen per second? |
06:15.02 | Cyon | Yeah, sounds about right |
06:15.07 | justinu | isn't that like 4 cents a second? |
06:15.18 | justinu | they must mean minute |
06:15.42 | Nugget | I wouldn't be too sure of that. |
06:15.46 | Cyon | lol |
06:15.48 | [hC] | yen != cent, is it? |
06:15.52 | Nugget | japan (well, tokyo) is ridiculously expensive. |
06:15.53 | justinu | that's fucking crazy |
06:15.59 | Cyon | International is extremely expensive |
06:16.01 | Nugget | it's a little less than a cent, but it's easiest to just pretend it is. |
06:16.17 | [hC] | Im worried to see what my cell bill is going to be after roaming and placing calls in costa rica for the last 12 days |
06:16.17 | Nugget | the place I stay for NTT Data is 28,000 yen a night, and it's nothing special. |
06:16.22 | justinu | i'll be using my ip phone as much as I can |
06:16.39 | Cyon | 1 USD = 114.665 JPY |
06:17.02 | Cyon | 1 JPY = 0.00872106 USD |
06:17.07 | [hC] | exten => s,n,GotoIf($["${CALLERIDNUM}" = ""]?SetUnknownCID) |
06:17.08 | [hC] | exten => s,n(SetUnknownCID),Set(CALLERID(num)=Unknown) |
06:17.13 | [hC] | Eaaargh |
06:17.16 | [hC] | I hate windows |
06:17.18 | [hC] | Sorry. |
06:17.33 | Cyon | lol We all do....I hope *peers around* |
06:17.48 | [hC] | Im puttying in a remote office right now and hit the right mouse button by accident. |
06:17.48 | Nugget | I hate windows. I hate Linux too. |
06:17.55 | [hC] | Amen nugget. |
06:18.03 | Cyon | Nugget: Hmmm, it's possible to be cheap, just takes work. |
06:18.05 | justinu | all computer operating systems suck |
06:18.11 | Nugget | Cyon: yeah |
06:18.22 | Corydon76-home | [hC]: Ugh, don't use ${CALLERIDNUM} |
06:18.31 | Nugget | it's hard if you have work to do and barely have conversational japanese skills, though. :) |
06:18.31 | Corydon76-home | Use ${CALLERID(num)} |
06:18.44 | justinu | yeah, it's 2 bucks a minute |
06:18.46 | [hC] | Corydon76-home: i was actually replacing that.. |
06:18.50 | justinu | incoming is free |
06:18.50 | Cyon | Nugget: I found the absolute best prices were when I went to a travel stand thingy and got a room for the night on the best deal they had...they always had super deals to use if it was the same day |
06:19.02 | justinu | i'll just use my * box as an intl callback |
06:19.03 | justinu | fuck it |
06:19.15 | Cyon | Nugget: True; but I have no japanese skills at all...Maybe they pitied me. ;-) |
06:19.17 | Corydon76-home | So, do you find it difficult to conceptualize setting a function to a value? |
06:19.21 | Nugget | I can't tell you how many times I've walked into a restaurant and started to order, gotten 30 seconds into the conversation at which point they realize I barely speak japanese, and then had my menu switched out for the gaijin menu with pictures and doubled prices. :) |
06:19.23 | Cyon | justinu: lol |
06:19.26 | justinu | they're charging in 1second increments |
06:19.34 | justinu | i can make it dial the digits pretty quick |
06:19.45 | justinu | make the phone dial, that is |
06:19.53 | Cyon | Nugget: lol My favorite places were where I get a ticket for food. :-x |
06:19.57 | Nugget | heh |
06:20.23 | Cyon | Nugget: The food was good, it was cheap, and didn't need to say much at all. lol |
06:20.33 | Cyon | Nugget: omg yes. Yuki...*sigh* |
06:20.37 | justinu | how about the prostitutes, are they as expensive as everything else? |
06:20.49 | Corydon76-home | Hmmm, I wonder if anybody has attempted to patent international callback yet... that would be a tidy little racket... |
06:21.15 | Cyon | justinu: I don't know...:-P |
06:21.27 | [hC] | Corydon76-home: Yeah, the variable-functions are... not the most intuitive at first.. |
06:21.55 | justinu | corydon: i could swear i remember hearing that someone did |
06:21.58 | Corydon76-home | Good, because I'm the one who got that functionality added |
06:22.13 | [hC] | Nugget: Dell is selling the 30" Cinema HD display for $1999 now in their case. You probably paid almost double that for yours, huh? :x |
06:22.36 | [hC] | Corydon76-home: why, just to be unique? :) |
06:22.44 | dorphalsig | hey, can anybody help me with this error msg? |
06:22.45 | dorphalsig | [chan_unicall.so]Jan 8 01:10:52 WARNING[6376]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging |
06:22.58 | Cyon | justinu: But just go to here http://www.japan-guide.com/e/e3011.html and you'll be all set ;-) |
06:23.15 | Corydon76-home | No, to be an asshole. I wanted to make sure that it was quite possibly the most difficult system to conceptualize, so people would continue to pay me the big bucks |
06:23.28 | Cyon | CoaxD |
06:23.30 | Cyon | Bah |
06:23.56 | justinu | lol |
06:23.58 | [hC] | Corydon76-home: You must work at microsoft? |
06:24.04 | [hC] | :) |
06:25.04 | Corydon76-home | Actually, I hate seeing code that polls for a change in a variable. I'd much rather have code that is triggered upon a value change |
06:25.06 | dorphalsig | Anybody? |
06:25.18 | PJMattF | I'm having problems getting (yes you guessed it) ztdummy going, on my new install of ubuntu breezy - it *looks* like it's building properly (I've overcome the obvious "not got ..." issues) but it won't modprobe. can anyone help? |
06:25.23 | justinu | polling is usually never a good idea |
06:25.40 | Corydon76-home | which is exactly what setting a function does... act as a trigger... |
06:25.43 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
06:25.51 | justinu | you mean a callback? |
06:26.00 | [hC] | PJMattF: i presume you did 'make install' and 'depmod -a' before trying a modprobe? |
06:26.03 | Corydon76-home | No, a trigger |
06:26.09 | [hC] | Corydo76: Makes sense. |
06:26.12 | [hC] | doh. |
06:26.14 | justinu | sounds like a callback to me |
06:26.21 | [hC] | exten => s,n,GotoIf($["${CALLERIDNUM}" = ""]?SetUnknownCID) |
06:26.21 | [hC] | exten => s,n(SetUnknownCID),Set(CALLERID(num)=Unknown) |
06:26.24 | [hC] | son of a. |
06:26.28 | Cyon | rofl |
06:26.41 | [hC] | there. whitespace in my copy buffer now. |
06:26.45 | Corydon76-home | It's not a callback, because you aren't passing around a pointer to the function |
06:26.50 | justinu | ok |
06:27.02 | PJMattF | [hC]: yep, still gives me "module ztdummy not found" and the matching "error running install command for ztdummy" failures. |
06:27.11 | Qwell | [hC]: umm... |
06:27.19 | Qwell | Why are you setting cidnum if cidname is empty? |
06:27.21 | Qwell | :) |
06:27.32 | Qwell | You probably meant to set cidname... |
06:27.47 | PJMattF | I picked up Asterisk fresh from svn earlier today, it appears to have done the UDEV changes (I recall doing that manually before - I rebuilt my machine today with different distro) |
06:28.04 | dorphalsig | Hi, I'm trying to get R2 to work with *, I already added /usr/local/lib to ld.conf.so, but I keep getting this when trying to start *. [chan_unicall.so]Jan 8 01:10:52 WARNING[6376]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging |
06:28.08 | [hC] | Qwell: er? That checks callerid num, and sets caller id num.. to unknown. |
06:28.16 | dorphalsig | can anybody help me? |
06:29.11 | Cyon | Qwell: Yeah...It looks logically right to me, what am I missing? |
06:29.18 | [hC] | Qwell is high. |
06:29.22 | Cyon | dorphalsig: Sorry, not a clue |
06:29.31 | [hC] | Dont mind him. |
06:29.37 | justinu | lol |
06:29.45 | Cyon | haha |
06:29.57 | [hC] | have to get incoming caller id turned on on these lines yet. Was sick of seeing 'asterisk' |
06:30.19 | justinu | the idea that you have to pay extra for caller ID is a fucking joke |
06:30.48 | Corydon76-home | Wow, you're using a module not included in Asterisk, yet asking the Asterisk folks to help. |
06:31.23 | Corydon76-home | Excuse me, #debian, but could you help me out with a problem I've been having on my RedHat 9 system? |
06:32.00 | PJMattF | and the little light in my microwave is broken |
06:32.26 | justinu | then openss7.org website sucks |
06:32.29 | justinu | none of the links work |
06:33.08 | [hC] | Anyone have any ideas when the Linksys SPA-942 will be released? |
06:33.13 | [hC] | I really want dual ethernet and PoE |
06:33.25 | [hC] | that would really make this 941 much better. |
06:33.51 | Cyon | dual ethernet definitely PoE; don't overly care |
06:33.51 | [hC] | Heh, found a funny bug with it though, if you set a user password on the web interface, trying to use the 'redial' function on the phone asks you for the user password. |
06:34.14 | Cyon | [hC]: Hmmm, only 941, not the 841s? |
06:34.20 | [hC] | Cyon: not the end of the world no, but it would be nice. its about the only thing its missing. |
06:34.30 | [hC] | Cyon: not sure, havent played with an 841. |
06:35.01 | Qwell | [hC]: hmm, okay |
06:35.10 | Cyon | [hC]: Yeah, I can see nice |
06:36.28 | Qwell | [hC]: well...be consistant with your variable/function use! :P |
06:36.44 | Qwell | I read that like 4 times too, kept reading it as name |
06:36.48 | [hC] | Qwell: haha. I've already switched it. |
06:37.12 | [hC] | I wonder what linksys is using to power their spa-9000's pbx functions.. I doubt they used asterisk |
06:37.22 | jbroome | linksysone! |
06:38.57 | SkramX | I have asterisk questions galore! |
06:39.20 | SkramX | later. |
06:39.21 | SkramX | lol |
06:39.21 | rajiv | i wonder when we will see pics of the new linksys spa phones |
06:40.05 | SkramX | rajiv: which model? |
06:40.49 | rajiv | spa901 spa921 spa922 spa942 |
06:40.55 | rajiv | http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_News_C2%26cid%3D1136499819516&pagename=Linksys%2FCommon%2FVisitorWrapper |
06:41.04 | [hC] | Hmm. I guess they did use asterisk. |
06:41.06 | [hC] | Linksys has made a special version of Asterisk that integrates with the SPA-9000. The changes Linksys has made to Asterisk are getting integrated into future Asterisk releases. |
06:41.17 | Qwell | [hC]: oh? |
06:41.24 | Qwell | What changes? |
06:41.41 | [hC] | Not sure, thats all the article says. |
06:42.00 | [hC] | Thats on the SPA9000 PBX/Router/ATA/SIP server thing they released |
06:42.10 | [hC] | looks alright for a very very basic ~5 phone small office install |
06:42.18 | [hC] | very cheap, 399 |
06:42.19 | marcus2 | the iax.conf that comes with 1.2.1 mentions some compile-time jitterbuffer options |
06:42.26 | rajiv | but only 1 pots port |
06:42.32 | marcus2 | but i cant figure out where they would be controlled.. the makefile doesnt have anything obvious |
06:42.39 | _Vile | hc, what features? |
06:43.05 | _Vile | only 1 pots out/inbound? |
06:43.59 | [hC] | _Vile check out the article on voxilla.com |
06:44.04 | SkramX | hey... anyone ever used asterisk to like keep track of a call, and make sure the call doesnt go over -X- ammount of minutes |
06:44.11 | _Vile | thx |
06:45.17 | SkramX | ?? |
06:45.29 | [hC] | SkramX: yes, lots ... One option that comes to mind is AbsoluteTimeout |
06:45.36 | _Vile | skram, yes |
06:45.42 | _Vile | skram /|\ |
06:46.01 | SkramX | well, i want to announce right before it hangs up |
06:46.03 | *** join/#asterisk dancing_ninja (n=chatzill@61.246.9.140) |
06:46.13 | dancing_ninja | hello friends |
06:46.31 | _Vile | skram, is it AGI? |
06:46.36 | [hC] | SkramX: im sure if you search for absoultetimeout on voip-info.org you will find some relevant examples. |
06:46.40 | SkramX | I dont have anything set up yet |
06:46.47 | dancing_ninja | i was wondering if we can use asterisk in buig call center setting? |
06:47.02 | _Vile | this is for a pre-paid type app? |
06:47.14 | SkramX | _Vile: yeah |
06:48.20 | _Vile | http://www.voip-info.org/wiki-Asterisk+Prepaid+Applications |
06:48.28 | SkramX | _Vile: I need to code my own |
06:48.28 | _Vile | there is a lot of code there |
06:48.30 | SkramX | any agi scripts? |
06:48.31 | _Vile | ahh |
06:48.33 | _Vile | yes |
06:48.43 | SkramX | PHP AGI? |
06:48.48 | troyb1 | _Vile im surprised you are not ready to hibernate after you finished with me? |
06:49.21 | _Vile | skram, not that I know of |
06:49.32 | _Vile | but the syntax should be easy to move from perl to php |
06:49.40 | _Vile | troy, ;) |
06:50.31 | troyb1 | are you EST? |
06:50.35 | _Vile | pst |
06:50.49 | _Vile | though, I'll be up 'til 1 or 2 tonight |
06:50.51 | justinu | pst? where you at? |
06:50.51 | _Vile | pst :) |
06:50.57 | _Vile | oregon |
06:51.00 | _Vile | you're in cali |
06:51.00 | justinu | ah |
06:51.02 | justinu | yes |
06:51.08 | troyb1 | only 7 am for you then? |
06:51.08 | _Vile | :) |
06:51.16 | _Vile | 10:51pm |
06:51.17 | _Vile | here |
06:51.24 | troyb1 | so much for that idea |
06:51.26 | _Vile | haha |
06:51.27 | troyb1 | naw im in Toronto |
06:51.34 | _Vile | I have a friend in toronto |
06:51.39 | troyb1 | your Northern neighbor :) |
06:51.42 | _Vile | former business partner |
06:51.52 | _Vile | yeah, you guys quietly come down and steal our cheese.. |
06:52.05 | troyb1 | haha yeah.. i just came back from Florida actually |
06:52.11 | _Vile | what part? |
06:52.14 | troyb1 | ironically i did buy cheese today |
06:52.16 | troyb1 | Naples |
06:52.21 | _Vile | nice place |
06:52.32 | troyb1 | i call it god's waiting gate |
06:52.40 | _Vile | hahaha |
06:52.46 | _Vile | I call it, quiet |
06:53.02 | troyb1 | yeah it's slow paced.. damn good miniature golf course there ;) |
06:53.46 | dancing_ninja | i was wondering if we can use asterisk in big call center setting? |
06:53.56 | troyb1 | i went to Miami for a few days but i mean it costs a lot more to stay there then naples for sure |
06:54.06 | rajiv | [hC]: i was mistaken. 2 pots ports not 1 |
06:54.42 | trixter | I love my 1 litre insulated coffee mug |
06:54.48 | trixter | er my 2 litre |
06:54.52 | justinu | lol |
06:55.03 | justinu | that's a huge mug! |
06:55.05 | dancing_ninja | am i invisible |
06:55.19 | justinu | ninja's are pretty stealthy |
06:56.20 | rajiv | dancing_ninja: you can use asterisk where ever you like |
06:56.22 | troyb1 | damn it would suck working for time.gov if you were even 10 minutes late there would be no arguing |
06:56.22 | trixter | justinu: yeah.. I have a couple cups of something a day |
06:56.50 | dancing_ninja | yeah they sure are ......but can some body pl guide oe provide a link where I can find links to setup asterisk in a big call ceter (1000 seater) |
06:57.01 | trixter | beer is nice when I am out and stuff |
06:57.21 | justinu | i doubt anyone here has info on a call center that big |
06:57.27 | dancing_ninja | i need to get ademo up and convince my ceo otherwise we buy avaya :-( |
06:57.30 | justinu | that's serious stuff, and serious money |
06:57.38 | SkramX | is there like a way to make an announcement 10 minutes into a call? |
06:57.44 | troyb1 | dancing_ninja hire a professional team to do that for oyu |
06:57.46 | troyb1 | *you |
06:57.46 | SkramX | well, after a Dial() command |
06:57.50 | justinu | skramx: show application dial |
06:57.55 | trixter | SkramX: technically yes, there is app_bridge |
06:57.56 | justinu | it does all that shit |
06:57.59 | dancing_ninja | from where ? |
06:58.02 | troyb1 | i dont know about you but i wouldnt want to unpack 1000 IP Phones |
06:58.08 | trixter | you can bridge to something that plays an anouncement if you really really wanted |
06:58.08 | justinu | troyb1: lol |
06:58.18 | trixter | or you can whisper into it |
06:58.20 | trixter | or something |
06:58.28 | troyb1 | dancing_ninja are you in the states? |
06:58.34 | dancing_ninja | i need to get a demo up ...some kind of feasibility study before that |
06:58.50 | troyb1 | most corporations can provide that data to you |
06:58.52 | _Vile | dancing, 1000 wouldn't be too difficult |
06:59.01 | justinu | skramx: http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
06:59.02 | troyb1 | _Vile i have seen bigger :) |
06:59.02 | dancing_ninja | no i am in India and mlaysia |
06:59.07 | _Vile | me too |
06:59.23 | _Vile | India, call center for mainly us traffic? |
06:59.29 | justinu | skramx: look at open L() |
06:59.33 | justinu | s/open/option/ |
06:59.38 | dancing_ninja | yup US and UK |
06:59.54 | _Vile | what do you have in place now? |
07:00.06 | trixter | I am doing a US call center for a US company.. same state even but only 10M minutes a month pending the 600k minute test goes well |
07:00.19 | dancing_ninja | we got cisco and avaya stuff |
07:00.21 | trixter | why export jobs when its so cheap to do it here? |
07:00.33 | SkramX | justinu: i am looking |
07:00.39 | justinu | everyone hates telemarketers |
07:00.42 | _Vile | trixter, expanding the wealth, is what I've come to the conclusion of |
07:00.42 | troyb1 | trixter the US economy cant support it is my guess |
07:00.47 | SkramX | so I would do Dial(who,L(x,y,z))? |
07:00.59 | justinu | yeah |
07:01.00 | dancing_ninja | weneed to open a new center so looking for alternatives |
07:01.15 | troyb1 | if your really going to try and setup a thousand cisco IP Phones then call up a few companies have meetings with them |
07:01.30 | troyb1 | i have a feeling none of them would think twice about inviting you out to lunch :) |
07:01.40 | _Vile | dancing, what software pops up on the screen -- or is it a simple call agent pool? |
07:01.50 | _Vile | yeah |
07:02.09 | trixter | SkramX: Hmm.. just thought of another ghetto way too. With dial you can goto a macro on connect, now I havent tried this but I would guess you can idle in that macro for 10 minutes then play some tone or something |
07:02.10 | _Vile | and dinner, a few brunches... |
07:02.22 | troyb1 | _Vile no questions asked. |
07:03.06 | SkramX | macro == agi? |
07:03.19 | Cyon | SkramX: No, macro == macro |
07:03.30 | SkramX | alright |
07:03.33 | SkramX | never written a macro |
07:03.34 | troyb1 | it would definetely have to be a 6 figure budget just for the phones alone |
07:03.35 | SkramX | hmm |
07:03.35 | dancing_ninja | does asterisk support skype ? |
07:03.57 | justinu | troyb1: i bet he wants to use softphones :P |
07:03.59 | dancing_ninja | or only sip based stuff like vonage |
07:05.47 | SkramX | how do i know how long a call has been in progress? |
07:06.04 | SkramX | dancing_ninja: NO |
07:06.50 | dancing_ninja | what abt a sip based stuff ? |
07:07.12 | rajiv | sip is no problem |
07:08.55 | dancing_ninja | grt ...so i can use any sip based network like vonnage or wengo and and use softphones ....? |
07:09.17 | trixter | SkramX: macros are kinda like contexts in the dialplan although they are slightly different ... they are defined within the dialplan though |
07:09.18 | Cyon | dancing_ninja: Sure |
07:09.28 | SkramX | meh |
07:09.44 | Cyon | SkramX: They are really simple once you see an easy example |
07:10.04 | trixter | they arent that hard.. it would be like [macro-waitfor10] exten => s,1,wait(600) exten => s,2,playback(beep) |
07:10.08 | SkramX | any examples :? |
07:10.35 | trixter | for what you want anyway.. then call it like Macro(waitfor10) or something.. the M option in dial is what you need to look at for your use |
07:10.43 | trixter | its totally ghetto but it should work |
07:10.52 | SkramX | well it would need to be a variable it waits for. |
07:11.02 | OloBola | Hello. When trying to view my arguments passed from dialplan to php: $agi->verbose($argv[1]);, PHP hangs, never prints anything. I can get the first argument fine ($agi->verbose($argv[0]);) so I know my code is ok |
07:11.56 | trixter | ok.. [macro-waittone] exten => s,1,wait(${arg1}) exten => s,2,playback(beep) |
07:12.21 | trixter | then Dial(SIP/somewhere/something,M(waittone,600)) |
07:12.26 | trixter | or something |
07:12.40 | trixter | no error checking but you get the idea |
07:12.45 | trixter | or should |
07:13.16 | trixter | that should work with a lot of versions of asterisk regardless of extra addons |
07:15.34 | dancing_ninja | is there java based interface or api with asterisks ? |
07:18.56 | rajiv | you can call java programs from asterisk if you want. there is the AGI api which works with anything that can read from STDIN and write to STDOUT |
07:19.54 | trixter | dancing_ninja: how exactly do you mean? |
07:20.02 | trixter | there are programs that do call management and other functions in java |
07:20.20 | trixter | there are other java based programs that interface with java directly... just dont know what the end goal is |
07:21.22 | dancing_ninja | i mean to say ......if there is java wrapper to asterisk api so that i call in using java program rather than jni |
07:21.37 | trixter | what specifically are you going toi call though |
07:21.58 | trixter | generaelly when people make reference to asterisk they mean the program not the libraries |
07:22.11 | trixter | your comments make me think you want libraries basically |
07:23.24 | dancing_ninja | yeah ...i play to integrate asterisk/voip intoour crm and groupware |
07:23.57 | dancing_ninja | and make a custom control panel and quatility dashboard |
07:24.37 | dancing_ninja | which hopefully will be released under lgpl eventually |
07:27.17 | trixter | there are a few that already exist |
07:27.33 | trixter | bet if you just wanted to mod one you could use their base class files to get the functionality |
07:27.41 | trixter | btw you appear to want the "manager api" |
07:27.57 | trixter | not "asterisk api" -- so you have better terms to look for it |
07:28.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
07:29.08 | dancing_ninja | ok ... |
07:29.42 | dancing_ninja | trixter: can u pl provide some links |
07:30.47 | trixter | www.google.com |
07:30.57 | justinu | ~docs |
07:30.58 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
07:31.05 | trixter | offhand I dont know where a java class would be for the asterisk manager api |
07:31.17 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
07:31.23 | trixter | justinu: he doesnt want docs he wants a java class for the manager api |
07:31.31 | Qwell | http://www.voip-info.org/wiki/view/JAVA-AGI |
07:31.37 | trixter | not for agi |
07:31.38 | trixter | manager api |
07:31.46 | Qwell | oh |
07:31.49 | OloBola | is this correct? exten => s,2,agi(test.php|'testing 1 2 3') trying to pass 'testing 123' |
07:40.48 | *** join/#asterisk Defraz (n=t0tal@72.24.220.144) |
07:41.13 | *** part/#asterisk justinu (n=justinu@207.181.0.86) |
07:44.05 | trixter | I feel like a monkey, eating trail mix, nuts, raisins and M&Ms |
07:44.23 | trixter | "Red M&M, blue M&M, they all are the same color in the end" - homer simpson |
07:46.14 | trixter | is there any digium t1 card that does atm? |
07:46.25 | trixter | or do I have to use sangnoma? |
07:46.32 | trixter | cause I know they do it |
07:50.54 | _Vile | trixter, what are you looking to do? |
07:51.11 | trixter | T1 in atm mode |
07:51.13 | trixter | not channelized |
07:51.18 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
07:51.24 | _Vile | interesting |
07:51.47 | _Vile | tmk, the digium code doesn't support this, but gimme a sec to confirm |
07:51.52 | trixter | a lot of people have done it a lot of times in the past |
07:54.19 | _Vile | looks like sangoma |
07:54.24 | _Vile | though |
07:54.32 | _Vile | *asterisk developers* |
07:54.38 | _Vile | this topic was brought up in late 2003! |
07:55.09 | _Vile | http://lists.digium.com/pipermail/asterisk-users/2003-September/021893.html |
07:55.32 | _Vile | lemme check code |
07:56.10 | _Vile | nuffin' |
07:57.25 | _Vile | trixter, you a coder? |
07:57.58 | trixter | once or twice |
07:58.48 | _Vile | well, if you decide to take the hard road and figure it out on asterisk, throw it back, seems useful |
07:59.40 | _Vile | s/asterisk/digium |
07:59.41 | trixter | asterisk is gpl that makes it really hard for me |
07:59.47 | trixter | its against my religion to contribute to gpl products |
08:00.20 | *** join/#asterisk Defraz_ (n=t0tal@72.24.220.144) |
08:00.56 | _Vile | im about out of steam, going to bed in a min |
08:01.24 | _Vile | quiet night |
08:01.49 | SkramX | oh |
08:05.53 | _Vile | ok later |
08:07.50 | *** join/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
08:14.47 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-118.claranet.co.uk) |
08:18.47 | *** join/#asterisk MatsK2 (i=enforcer@c83-253-29-22.bredband.comhem.se) |
08:24.09 | trixter | Hmm.. got confirmation today that with asterisk modules at least (big explanation of how they interface with the system) the modules themselves dont have to be gpl just gpl compatible which means that I might actually od something with regards to asterisk development becuase I can use a BSD lciense for my modules |
08:24.20 | trixter | which alieviates a lot of my problems with development for asterisk |
08:27.00 | jahani | hi |
08:27.04 | jahani | what mean this error ? |
08:27.04 | jahani | ERROR[2354]: chan_sip.c:10856 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete. |
08:27.35 | *** join/#asterisk lorinc (n=ang@caracas-0315.adsl.interware.hu) |
08:29.54 | mogorman | no trixter |
08:29.59 | mogorman | one simple thing |
08:30.06 | mogorman | you have to load a module with gpl key |
08:30.16 | mogorman | as well as libraries that are gpl |
08:30.29 | mogorman | you can no use gpl libs in non gpl code |
08:30.32 | mogorman | perio |
08:30.32 | mogorman | d |
08:30.42 | trixter | according to the FSF I can do that |
08:30.45 | trixter | I will take their word over yours |
08:30.49 | mogorman | i know |
08:30.55 | mogorman | i talked with benjk for an hour |
08:30.56 | mogorman | about it |
08:31.01 | mogorman | it will be resolved this week |
08:31.03 | mogorman | i hope |
08:31.09 | benjk | it has already been resolved |
08:31.14 | mogorman | and i imagine there will be a statement of some kind |
08:31.28 | trixter | well regardless licensing@fsf.org staes that I can, so I will if you as a digium spokesperson want to make this a legal issue go ahead I can give you my lawyers name and address |
08:31.45 | trixter | I am doing what the FSF states I can do under section 7 of the gpl you cant impose extra terms on me |
08:31.50 | mogorman | i am taking it up with fsf |
08:31.57 | mogorman | i know exactly what you are implying |
08:32.00 | benjk | asking a second time, the FSF insisted that LGPL is a proper license for Asterisk modules |
08:32.04 | trixter | so if you want to violate the gpl to tell me I cant go ahead I will do it anyway, the only way to stop me is to file a lawsuit, is that your intention? |
08:32.06 | mogorman | but it is a simple fact |
08:32.12 | mogorman | gpl is a 1 way street |
08:32.17 | trixter | its a fact that I have documentation from the FSF on my side |
08:32.48 | mogorman | ugh im tired of having argument again |
08:33.01 | trixter | so if you want to file a lawsuit lemme know I will forward you the name and contact information of my lawyer |
08:33.04 | mogorman | like i told benjk |
08:33.11 | mogorman | i am going to discuss this with fsf |
08:33.17 | trixter | othwrise I will side with the FSF on the issues relating to the gpl in this matter |
08:33.19 | mogorman | and disclose any contact i have |
08:33.19 | benjk | mog: why do you think legal issues are black and white and simplistic if it takes hard work and study to go through law school and take the bar exam |
08:33.23 | mogorman | if this issue is resolved |
08:33.28 | mogorman | in any way |
08:33.42 | mogorman | i will make steps ness. to have lgpl and bsd keys added |
08:33.49 | mogorman | agreed |
08:33.53 | mogorman | my brother is a lawyer |
08:33.58 | mogorman | as are both my uncles |
08:34.10 | trixter | well anyway, I am going to release something tonight just to do it, if you want to attempt to file a restraining order to prevent me go ahead, if you want to file against me for licensing infringement go ahead I will give you the name of my lawyer |
08:34.25 | benjk | and they specialise in intellectual property with regards to software and publishing? |
08:34.26 | mogorman | as im not the co holder |
08:34.52 | mogorman | actually yes |
08:34.53 | mogorman | one is uber geek |
08:34.57 | trixter | well if its so clear cut why does the general counsel to the open source initiative agree with me on this issue |
08:35.01 | trixter | per a 2003 linuxworld article |
08:35.06 | trixter | or was it linux journal? |
08:35.08 | mogorman | i know trixter |
08:35.10 | trixter | one of those two |
08:35.13 | tzafrir_laptop | basically you can not use gpl-conflicting modules: res_crypto (openssl) and chan_[o]h323 |
08:35.14 | mogorman | im not going to make desc. |
08:35.17 | mogorman | i will let fsf |
08:35.17 | benjk | and they told you that Asterisk modules cannot be LGPL licensed? |
08:35.18 | mogorman | do it |
08:35.22 | trixter | however the FSF is on my side in this so feel free |
08:35.29 | tzafrir_laptop | Oh, and codec_g729a |
08:35.29 | mogorman | but i want a formal statement |
08:35.32 | mogorman | not you telling me |
08:35.40 | benjk | so how do you explain that the FSF licensing department states otherwise |
08:35.47 | mogorman | that too tzafrir_lapto |
08:35.51 | trixter | I will follow the FSF directions in one other matter |
08:35.53 | tzafrir_laptop | Anybody wants to rewrite res_crypto with gnutls? |
08:35.56 | mogorman | i am going to fsf |
08:35.59 | trixter | Yes, taking into account all the information you have provided, |
08:35.59 | trixter | the LGPL is an acceptable license for the modules. Please refer the |
08:35.59 | trixter | Digium licensing team to: |
08:35.59 | trixter | <PROTECTED> |
08:36.01 | mogorman | like i said |
08:36.03 | benjk | the licensing specialists of the organisation that released the GPL and LPGL licenses |
08:36.05 | trixter | unless of course you arent part of hte licensing team |
08:36.13 | mogorman | i know |
08:36.14 | trixter | that is from zak, licensing@fsf.org |
08:36.19 | mogorman | i realize |
08:36.31 | tzafrir_laptop | what's the FSF here for? |
08:36.33 | mogorman | you guys obviously want it cleared up |
08:36.39 | mogorman | thats what i want to have happen |
08:36.52 | mogorman | but right now its obviously a grey area with us in a dispute |
08:36.53 | trixter | email zak, its quite simple |
08:36.56 | trixter | anyone can actually do that |
08:36.58 | mogorman | can you please let me get a formal answer |
08:36.59 | trixter | and get the same information |
08:37.02 | mogorman | and make the changes ness. |
08:37.05 | benjk | as far as I am concerned the FSF has cleared it up for us |
08:37.15 | trixter | I will be releasing something, not quite sure what yet, but something tonight |
08:37.17 | mogorman | then why dont i have said email |
08:37.18 | benjk | but you did insist that we were doing something wrong |
08:37.22 | mogorman | why isnt it public anywhere |
08:37.33 | mogorman | i want to do this in public arena |
08:37.37 | trixter | if you want my lawyer info to serve with a restrianing order to prevent me from licensing with a compatible but not gpl license lemme know |
08:38.14 | tzafrir_laptop | mogorman, again, is there anything that isn't clear? you can't use real-time mysql together with dundi, as things stand now |
08:38.14 | trixter | that email is from tonight btw, so as of about 1 hour or so ago zak was answering emails |
08:38.17 | trixter | he may still be |
08:38.33 | mogorman | thats true tzanger |
08:38.37 | mogorman | err tzafrir_laptop |
08:38.43 | mogorman | unless they issue us an exception |
08:39.06 | tzafrir_laptop | they won't. Unless mysql decide to change their license |
08:39.08 | benjk | whether MySQL's policies are in line with the GPL is an entirely different question |
08:39.22 | tzafrir_laptop | Or unless somebody bothers rewirting res_crypto |
08:39.25 | mogorman | they can issue an exception |
08:39.28 | benjk | but since that question doesn't concern me, I am happy not to spend my time on it |
08:39.30 | mogorman | without changing license |
08:39.47 | mogorman | as co holder you can do anything you want |
08:40.17 | benjk | no you can't |
08:40.24 | mogorman | actually you can |
08:40.33 | mogorman | if you are copy right holder of a work |
08:40.35 | tzafrir_laptop | As for me, I'd be hapier to have something that is stricktly GPL, so I won't have any problem using other GPL libs |
08:40.40 | mogorman | you can do anything you want with said work |
08:40.53 | benjk | for example, if you release code under the GPL, you cannot tell people that they cannot use or redistribute that GPL licensed code in ways that the GPL allows |
08:40.53 | trixter | so digium owns the copyright on the netbsd code in asterisk? |
08:41.00 | trixter | what about the libc code that is part of it too? |
08:41.22 | trixter | benjk: per section 7 of the gpl no extra restrictions :) |
08:41.27 | benjk | what you license under the GPL grants recipients the rights the GPL grants |
08:41.33 | trixter | which includes removing the ability to use gpl compatible licenses |
08:41.40 | benjk | you cannot take those rights away with EULAs etc |
08:41.54 | Qwell | exception != restriction |
08:41.57 | tzafrir_laptop | benjk, well? |
08:42.06 | mogorman | its like religious debate at this point |
08:42.09 | benjk | and MySQL AB are quite likely overstepping this line |
08:42.17 | tzafrir_laptop | benjk, mysqlab sell you mysql under a different license |
08:42.30 | tzafrir_laptop | that is, not you specifically, I figure |
08:42.38 | benjk | but as I said, I don't care to spend my time on that issue because I don't use MySQL |
08:42.40 | trixter | mogorman: I dont see it as religious, the FSF has spoken digium is infringing on the gpl rights of the people that get the gpl version |
08:42.48 | trixter | its quite simple, or at least a digium representative is trying to |
08:42.54 | mogorman | ? |
08:43.03 | mogorman | ? |
08:43.11 | benjk | tzafrir: I am talking about GPL MySQL |
08:43.16 | mogorman | one i dont rep. digium in this argument |
08:43.36 | trixter | oh so saying I cant is a moot point then |
08:43.37 | mogorman | and two i never in my time at digium have seen any debate between us and fsf if anything we are good friends |
08:43.40 | trixter | at least that is cleared up |
08:43.46 | tzafrir_laptop | trixter, where has the FSF spoken so, could you please give me a link? |
08:43.47 | mogorman | i disagree |
08:44.04 | Qwell | trixter: I too would like to see the communication back and forth between them regarding this |
08:44.05 | benjk | according to MySQL AB's terms they claim several restrictions on what you can do with GPL MySQL which go beyond what the GPL says |
08:44.08 | mogorman | which is why when i speak with fsf i will disclose any info i recieve |
08:44.09 | trixter | tzafrir_laptop: I copied in the email, and even gave zak's email address (licensing@fsf.org) if anyone wanted independant confirmation |
08:44.17 | *** join/#asterisk ManxPower (i=ewieling@24.sub-70-210-168.myvzw.com) |
08:44.20 | trixter | they suggested that digium licensing team read their faq and pointed to a specific entry |
08:44.41 | benjk | and the GPL says that for code you release under GPL terms you must not impose any such additional restrictions |
08:44.43 | Qwell | benjk: With MySQL, using it with a NON GPL product, WOULD VIOLATE the GPL. MySQL is doing "The Right Thing" by selling it under a diff license |
08:44.50 | trixter | zak was there 1.5 hours ago, prolly still is (I just checked the timestamp) |
08:45.10 | benjk | consequently, I consider MySQL AB in violation of the GPL in respect of their GPL-MySQL release |
08:45.20 | trixter | using gpl code with a commercial product isnt necessarily violating the license, becuase of how mysql defines that |
08:45.23 | Qwell | benjk: If you're at ETel, I'll GLADLY talk to my contact at MySQL with you |
08:45.37 | trixter | mysql states that if you store data in their database with a commercial app or was handled by a commercial app you have to get a commercial license |
08:45.51 | Qwell | we can let him clear it up. You *KNOW* they have lawyers, and you *KNOW* they've talked with the FSF on multiple occasions. |
08:45.53 | trixter | they FURTHER state that if you use mysql commercially you have to get a commercial license, even if the software itself isnt commercial |
08:46.00 | benjk | especially the claus that says that you cannot make your customer to install MySQL with an installer for example |
08:46.01 | Qwell | trixter: no, they don't. :) |
08:46.13 | tzafrir_laptop | trixter, is this in reference to plugins from asterisk-addons? |
08:46.31 | trixter | now this is over mysql |
08:46.35 | trixter | getting a link hold on |
08:47.20 | OloBola | how do I go about sending arguments to an agi script? I've seen four different ways so far, can't get any of them to work. exten => s,2,agi(test.php,10001) Trying to send 10001 to test.php |
08:47.25 | benjk | no it is a general assessment of how MySQL AB claim restrictions for the GPL release of their software which go beyond the restrictions of the GPL itself |
08:47.44 | Qwell | benjk: come to ETel |
08:47.55 | Qwell | benjk: I'll let you talk to a MySQL rep in person. I'll even introduce you. |
08:48.25 | tzafrir_laptop | That said, practically any file in the source tree calims to be GPL. And actually don't refer to a modified license or any excptions. |
08:48.31 | benjk | I don't have the resources to fight this through, but I do have the right to state that there are such additional restrictions |
08:48.42 | benjk | consequently I don't use MySQL |
08:48.48 | X-Rob | trixter, the important bit is 'if the program you're using it with isn't GPL'ed' |
08:48.56 | Qwell | X-Rob: exactly |
08:49.02 | Qwell | which I've stated repeatedly |
08:49.03 | benjk | so I don't risk to release an installer which MySQL then take issue with |
08:49.09 | X-Rob | if the program you're using it with _IS_ GPL'ed then it's fine and no-one cares. |
08:49.38 | X-Rob | wups |
08:49.38 | mogorman | X-Rob! |
08:49.42 | X-Rob | *eek* |
08:49.44 | mogorman | i havent seen you in ages |
08:49.49 | X-Rob | I've been busy. |
08:49.51 | X-Rob | 8-\ |
08:49.56 | mogorman | heh |
08:49.59 | mogorman | how you been |
08:50.05 | benjk | those companies have the right to try an be as restrictive as they can or as they can get away with |
08:50.18 | benjk | and I have the right to avoid their products as a result |
08:50.34 | benjk | and to recommend to all my customers never to touch those products |
08:51.10 | X-Rob | http://aussievoip.com.au/salute1.wmv |
08:51.26 | mogorman | ? |
08:51.30 | X-Rob | 8-( |
08:51.31 | benjk | PostgreSQL is available as an alternative, I can use that without even having to worry about what may or what may not upset MySQL AB's legal folks |
08:51.53 | X-Rob | Funer. |
08:52.32 | X-Rob | Fus all hell coz my PC is day. |
08:52.36 | X-Rob | Gah. I'm laggy as all hell coz my PC is rendering |
08:52.41 | X-Rob | yay it's finished. |
08:52.49 | mogorman | X-Rob, hows the amp side of force? |
08:53.05 | X-Rob | mogorman, I've been busy as crap recently. |
08:53.21 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
08:56.50 | *** join/#asterisk elcuco (n=diego@local.xorcom.com) |
08:57.03 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
08:58.17 | X-Rob | Hurm. Anyone know anything about quicktime, and the best codec to use whilst encoding? |
08:58.27 | X-Rob | (I realise this is a totally inappropriate channel for this questions) |
08:58.36 | mogorman | for what kinda video |
08:58.37 | benjk | Quicktime uses a variety of codecs |
08:58.42 | mogorman | with quicktime you can use h264 |
08:58.47 | mogorman | proccessor intensive |
08:58.49 | mogorman | but tiny as hell |
08:58.55 | benjk | the latest version even includes H.264 now |
08:59.11 | benjk | but that runs only on G5 IIRC |
08:59.12 | X-Rob | mogorman, to re-render that link I posted above. It's crappy for a 5mb file. |
08:59.30 | mogorman | hrm? |
08:59.52 | elcuco | "when you are looking out the window for rain, put your glases on... otherwise you will not see the rain." |
09:00.28 | mogorman | i see |
09:01.33 | OloBola | how do I go about sending arguments to an agi script? I've seen four different ways so far, can't get any of them to work. exten => s,2,agi(test.php,10001) Trying to send 10001 to test.php |
09:02.12 | OloBola | I tried this to: exten => s,2,agi,test.php|10001|10001|10001 |
09:02.19 | OloBola | too |
09:02.41 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
09:02.52 | OloBola | I can get the first argument, not the second. |
09:03.04 | OloBola | using agiphp |
09:05.51 | OloBola | from php: $agi->verbose($argv[1]); |
09:06.03 | OloBola | it just hangs |
09:06.52 | *** join/#asterisk ptblank (n=MURDER1@68-169-161-61.lmdaca.adelphia.net) |
09:07.10 | *** join/#asterisk zhao__ (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
09:08.55 | *** join/#asterisk jebba (n=jebba@ip-216-17-203-198.rev.frii.com) |
09:09.08 | jebba | anyone using a utstarcom f1000? :) |
09:11.17 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
09:26.47 | *** join/#asterisk zhao_ (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
09:31.04 | *** join/#asterisk bmg505 (n=leon@dsl-146-16-132.telkomadsl.co.za) |
09:34.40 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
09:41.25 | *** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net) |
09:43.15 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
09:46.43 | Qwell | bed time...I think |
09:48.19 | *** join/#asterisk Kidaz (n=kidaz@thedowds.demon.co.uk) |
09:52.29 | *** join/#asterisk mkl1525 (n=daniel@84.19.199.22) |
09:58.00 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
10:00.10 | *** join/#asterisk jebba (n=jebba@ip-216-17-154-58.rev.frii.com) |
10:01.14 | *** part/#asterisk mogorman (i=root@user-24-236-84-48.knology.net) |
10:17.18 | *** join/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
10:18.25 | Gordo | Hi Guys... I need Help!!! Big Time.... I have been at this for days.... |
10:18.44 | Gordo | I can make calls to test extensions @ the * server, but not between phones |
10:18.51 | Gordo | Its killing me!! |
10:23.52 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
10:28.23 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
10:42.55 | *** join/#asterisk eXeCuT (n=root@81.213.241.86) |
10:43.02 | *** part/#asterisk eXeCuT (n=root@81.213.241.86) |
10:46.51 | *** join/#asterisk zhao__ (n=zhao@c-24-13-6-136.hsd1.il.comcast.net) |
10:47.02 | trixter | my bsd licensed asterisk app that does really nothing at all.. http://www.trxtel.com/app_bsdapp.c -- basically written becuase I was told I couldnt |
10:47.40 | *** join/#asterisk MT`AwAy (n=admin@bzq-218-195-128.red.bezeqint.net) |
10:47.53 | MT`AwAy | hello~ |
10:48.39 | MT`AwAy | got a little problem : when someone uses a service where I use WaitExten to get the code they type, I get the last key they entered in the CDR records as DST field |
10:49.42 | MT`AwAy | anyone has an idea ? |
10:50.05 | *** join/#asterisk svenna_ (n=svenna@p548D1EE2.dip0.t-ipconnect.de) |
10:51.47 | *** part/#asterisk svenna_ (n=svenna@p548D1EE2.dip0.t-ipconnect.de) |
10:59.03 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
11:09.51 | Gordo | Any chance of some help from the * guru's? |
11:14.40 | *** join/#asterisk aksis (i=aksis@universalkingdomofgod.net) |
11:15.11 | *** join/#asterisk _4d4m_ (n=adam@44-8-101-159.adsl.legend.co.uk) |
11:24.16 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
11:29.18 | *** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it) |
11:41.00 | *** join/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
11:41.39 | tzafrir_laptop | Gordo, try detailing your problem. Maybe one of the non-gurus lurking here could help |
11:42.59 | tzafrir_laptop | Please explain exactly what does work, and what doesn't. Also state version of Asterisk |
11:44.49 | Gordo | I have a fresh installation of * on a gentoo box (sparc), I have an Aastra 480i sip phone and a couple of soft phones... All my phones are registered, and they can call test extensions with recorded messages |
11:44.57 | Gordo | but hte phones cannot call each other |
11:45.13 | iDunno | have you got extensions set up for the phones? |
11:45.18 | Gordo | yeah |
11:45.32 | iDunno | what happens when you dial the extension? |
11:45.52 | Gordo | I can see that the call is setup in *, but then it puches me straight to voicemail |
11:46.04 | Gordo | retrans_pkt: Maximum retries exceeded on call |
11:46.22 | Gordo | No one is available to answer at this time |
11:46.35 | iDunno | phones are SIP? |
11:46.38 | Gordo | yep |
11:46.46 | iDunno | asterisk can talk out on port 5060 to the phones? |
11:47.09 | iDunno | phones on the same lan segment as the asterisk box? |
11:47.19 | iDunno | firewall anywhere in the middle? |
11:47.34 | Gordo | phones are on a different subnet, but only a routehr in hte middle |
11:47.40 | Gordo | no firewall or ACL's |
11:48.05 | iDunno | try putting one of them on the same subnet. |
11:48.14 | iDunno | see if that then works. |
11:48.20 | Gordo | ok, wont take long... |
11:53.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
11:53.52 | aksis | if I have a static ip, do I need to use FWD iax setup? |
11:54.17 | aksis | im having a little trouble grasping the big picture of voip |
11:54.19 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
11:54.45 | *** join/#asterisk rabelais (n=blank@ca-stmnca-cuda4-gen2q-213.vnnyca.adelphia.net) |
11:55.18 | Gordo | iDunno: Ok, changed ip of my * box and configured sip phones... but exactely the same problem |
11:55.39 | Gordo | What gets me is that I can call a test extension and hear a recorded message fine |
11:58.02 | iDunno | suggests that the phones can talk to * but * can't talk to the phones to me, so a networking or router problem. |
11:58.06 | tzafrir_laptop | Gordo, seems like it's time for you to pastebin extensions.conf , sip.conf and a trace from the CLI in verbose mode |
11:58.33 | Gordo | ok.. all phones are on the same VLAN/ subnet now |
11:58.36 | Gordo | pasting now |
11:58.47 | tzafrir_laptop | ~pastebin |
11:58.49 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
12:00.54 | Gordo | ok, the two conf files @ http://pastebin.com/496229 |
12:01.45 | Gordo | cli trace @ http://pastebin.com/496230 |
12:02.17 | rabelais | do I have to compile in MeetMe support? |
12:05.51 | *** join/#asterisk Gr1NcHeuX (n=devine@AStDenis-105-1-22-116.w81-248.abo.wanadoo.fr) |
12:06.06 | Gr1NcHeuX | salut ici et bo nene a tous ^^ |
12:06.11 | Gr1NcHeuX | une kestion |
12:06.50 | Gordo | any luck guys? |
12:07.28 | Gr1NcHeuX | kkun sait quoi ca correspond ? |
12:07.31 | Gr1NcHeuX | -rw-r--r-- 1 root root 163 2006-01-02 11:23 mamie-a-la-plage.avi |
12:07.46 | Gr1NcHeuX | -rw-r--r-- 2 root root 163 2006-01-02 11:23 mamie-au-resto.avi |
12:08.02 | Gr1NcHeuX | la diff entre le 1 et le 2 ? |
12:09.09 | *** join/#asterisk Oryn (i=oryn@falcore.fsck.tv) |
12:09.24 | *** join/#asterisk tsetane (n=tsetane@pppoecl69000.minlos.no) |
12:09.55 | rabelais | Gr1NcHeuX: I'm not sure what you're asking, but is seems like you're asking if there is a difference between the two files |
12:10.15 | Oryn | anyone know why my music-on-hold seems to be playing very very slowly? (sounds like a special effect from a star trek movie) |
12:10.38 | rabelais | Gr1NcHeuX: there doesn't seem to be one, but you can compare using diff, enter: diff mamie-a-la-plage.avi mamie-au-resto.avi if there is any output, then they're not the same |
12:11.03 | Oryn | bye the way it does work great for scaring away sales reps :P |
12:11.14 | Gr1NcHeuX | no it between 1 and 2 |
12:11.22 | Gr1NcHeuX | what's the differcne ? |
12:11.40 | Gr1NcHeuX | erf sorry for my por english i'm a french guy :) |
12:12.48 | Mother | greetings |
12:13.12 | *** join/#asterisk MatsK (i=enforcer@c83-253-29-22.bredband.comhem.se) |
12:13.13 | Mother | I have a couple of boxes running HEAD from march 2005, and want to roll them up to 1.2.1, any particular caveats? |
12:13.29 | Mother | I remember reading someplace that the extensions format has been changed a lot |
12:13.41 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
12:16.13 | rabelais | Gr1NcHeuX: I told you, there doesn't look like there's a difference, if you want to be sure, you can use the "diff" program to check the file content and it will report any differences between the two files |
12:16.46 | rabelais | but when I say there doesn't "appear" to be any differences, all that means is that the files have the same size and were modified at the same time |
12:16.56 | rabelais | that's no guarantee that the files have the some content |
12:16.59 | Gr1NcHeuX | ok rabelais thx |
12:17.06 | Gordo | iDunno: What would prevent * from talking to the phones if hte phones can talk to * |
12:17.53 | iDunno | firewall, blocked UDP ports, something like that. |
12:18.24 | iDunno | Gordo: I'd put * and a phone on the same switch. |
12:18.43 | Gordo | They are |
12:18.50 | Gordo | Same Subnet, same VLAN, same switch |
12:19.50 | MT`AwAy | Gr1NcHeuX> tu peux aussi utiliser md5sun |
12:19.51 | MT`AwAy | erf |
12:19.53 | MT`AwAy | md5sum |
12:20.07 | *** join/#asterisk Dovid (i=dovi5988@53.sub-70-193-162.myvzw.com) |
12:20.29 | Gr1NcHeuX | deja fais MT`AwAy le probleme c ke les fichiers sont differents en taille |
12:20.40 | Gr1NcHeuX | donc le md5sum differe |
12:21.08 | MT`AwAy | bah donc les fichiers sont differents |
12:21.21 | Gr1NcHeuX | 1 root root |
12:21.25 | Gr1NcHeuX | 2 root root |
12:21.29 | *** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au) |
12:21.44 | Gr1NcHeuX | en dehors de leur taille c le chiffre ki me pose un soucis |
12:21.51 | *** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
12:22.07 | MT`AwAy | y'aurais pas un repertoire dans le lot ? |
12:22.17 | Gr1NcHeuX | nop c deux videos |
12:22.52 | MT`AwAy | aussi loin que je me souvienne le numero juste apres les permissions correspond au nombre de fichiers visibles dans le repertoire |
12:23.08 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
12:23.25 | MT`AwAy | ou bien |
12:23.59 | mrtwister | hello. i currently use oh323, but there i cannot set fast start and h245 options per gateway (peer). cna i do it in other channels (h323 and ooh323 )? |
12:24.02 | Gordo | tzarfir: Any luck with those configs? |
12:26.40 | Gr1NcHeuX | MT`AwAy, c aussi c que g savais mais y a un soucis dans le sens ou ce sont des videos donc un seul fichier logiquement |
12:27.22 | MT`AwAy | bah oui |
12:27.41 | MT`AwAy | sinon il me semble que pour les fichiers ca peut egalement representer le nombre de 'hard link' |
12:28.05 | MT`AwAy | -rw-r--r-- 1 magicaltux magicaltux 0 2006-01-08 14:27 test |
12:28.11 | MT`AwAy | et apres un hardlink : |
12:28.12 | MT`AwAy | -rw-r--r-- 2 magicaltux magicaltux 0 2006-01-08 14:27 test |
12:28.13 | MT`AwAy | -rw-r--r-- 2 magicaltux magicaltux 0 2006-01-08 14:27 test2 |
12:28.36 | MT`AwAy | en gros t'as une autre mami au resto sur ton disque |
12:28.43 | elcuco | ... i think my irc client is broken... |
12:28.46 | Gr1NcHeuX | ok cool |
12:29.02 | Gr1NcHeuX | g'avais pige merci pour ca :) |
12:29.13 | MT`AwAy | pas de soucis~ |
12:29.19 | MT`AwAy | sinon je vois pas trop le rapport avec asterisk |
12:29.22 | Gordo | BRB guys |
12:29.23 | Gr1NcHeuX | c sympa 2 mamie au resto +tot k'une :p |
12:29.36 | Gr1NcHeuX | MT`AwAy, g me suis juste plante de chan pour demander |
12:29.44 | MT`AwAy | 'kay |
12:29.52 | Gr1NcHeuX | et g un asterisk chez moi c pour ca ke chui la de tps zen tps |
12:30.01 | MT`AwAy | ok |
12:30.11 | MT`AwAy | j'utilise aussi asterisk, mais pour le public ;) |
12:30.18 | MT`AwAy | (petit serv public en france, pour des francais) |
12:31.34 | Gr1NcHeuX | ta vue ke chez un fournisseur de telephonie ils ont une offre a 0.01e la min de portable en france ? |
12:32.39 | Gr1NcHeuX | cf budget-telecom |
12:32.56 | Gr1NcHeuX | stu mets ca sur ton serv tu vas gagner des sous ;) |
12:33.33 | *** join/#asterisk p0g0__ (n=p0g0@mrtc-mm-590160.mis.net) |
12:33.58 | *** join/#asterisk Gordo (n=bs@203-56-245-33.cpe.vic-1.comcen.com.au) |
12:34.20 | Gordo | Idunno: you have me thinking that the windows firewall is blocking my softphone from working actually... |
12:34.38 | Gordo | I have one desktopn sip phon and two softphones for testing atm |
12:34.39 | trixter | damn firewalls |
12:34.42 | *** join/#asterisk MatsK_away (i=enforcer@c83-253-29-22.bredband.comhem.se) |
12:34.45 | iDunno | oh, that's always possible. the windows firewall is a pain |
12:34.47 | trixter | the world would be an easier place without em |
12:34.53 | Gordo | hrm yeah... and as if it would be as simple as turning it off |
12:35.05 | Gordo | noooo, group policy prohibits it |
12:35.31 | trixter | doesnt the windows firewall like to let programs open up ports? like fairly easily? I know that microsoft messenger uses upnp to disable firewall settings on any MS inspired network firewall (like their access points) |
12:35.51 | MT`AwAy | Gr1NcHeuX> je pense pas qu'ils autorisent la revente de leur offre |
12:35.55 | Gordo | hah, really |
12:35.59 | Gordo | that would be right |
12:36.24 | Gr1NcHeuX | ben t'es pas oblige de la revendre hein |
12:36.38 | trixter | we have one here and there is no way to disable that via any of the standard tools (obviously you can rewrite the firmware but ...) |
12:37.10 | trixter | so users can trump the admin, and its basically soap or soap style at least data so its easy to manipulate to open ports and whatnot.. very bad mojo |
12:37.29 | p0g0__ | trixter: use un pnp from here to nail the messenger http://www.grc.com/freepopular.htm |
12:37.41 | MT`AwAy | Gr1NcHeuX> dans ce cas je vois pas trop comment je vais gagner des sous~ |
12:37.56 | p0g0__ | gibson has been around for 30 years & is very competant |
12:37.57 | Gr1NcHeuX | MT`AwAy, sur l'offre globale |
12:38.05 | trixter | p0g0__: its not a big problem, just one that would have been nice if it were docuented |
12:38.10 | *** join/#asterisk backblue (n=moo@87-196-6-89.net.novis.pt) |
12:38.27 | trixter | and right now nothing is even connected to that AP its just for testing |
12:39.39 | MT`AwAy | Gr1NcHeuX> sur mon offre globale j'apelle deja plus de 30 pays gratuitement (pour les fixes, et les mobiles pour 3 de ces pays) |
12:40.33 | Gr1NcHeuX | gratuitement ? |
12:40.38 | Gr1NcHeuX | sans abonnement ? |
12:41.09 | MT`AwAy | Gr1NcHeuX> juste un petit abonnement de 25 euro/mois |
12:41.30 | Gr1NcHeuX | ok abo valable pour des comm entreprises ? |
12:41.40 | MT`AwAy | vi |
12:41.47 | Gr1NcHeuX | cool chez ki ? |
12:41.50 | MT`AwAy | enfin pour les entreprises c'est 30 euro au lieu de 25 |
12:41.54 | MT`AwAy | par contre pas de revente |
12:42.04 | Gr1NcHeuX | ouep du direct |
12:42.10 | Gr1NcHeuX | c chez ki ? |
12:42.29 | MT`AwAy | broadvoice |
12:42.52 | MT`AwAy | je teste le service actuellement~ ca marche pas mal |
12:43.16 | *** join/#asterisk svenna_ (n=svenna@p548D1EE2.dip0.t-ipconnect.de) |
12:43.43 | Gr1NcHeuX | g deja essaye chez eux mais la ou g suis g trop de ping avec eux |
12:43.57 | MT`AwAy | ok |
12:44.05 | Gordo | Besides 5060 what ports should I be allowing on hte windows firewall |
12:44.14 | trixter | your rtp ports |
12:44.24 | tzafrir_laptop | basically, all of the RTP ports, see rtp.conf |
12:44.25 | trixter | rtp.conf - default is 10000-20000 |
12:44.44 | trixter | udp |
12:44.55 | Gordo | there is no way to open a port range on XP firewall is there...? |
12:45.03 | trixter | dont use it dont know |
12:45.43 | *** join/#asterisk welles (n=welles@222.90.155.122) |
12:46.44 | Gordo | Thanks for all your help guys, but I need sleep.... |
12:47.24 | Gr1NcHeuX | Gordo, the better solution is to put a debian on ure machine ;) |
12:47.43 | trixter | I say no firewall no service packs |
12:47.46 | trixter | that will fix it right up! |
12:48.38 | *** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it) |
12:49.28 | *** join/#asterisk wellng (n=welles@222.90.155.122) |
12:49.48 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
12:51.30 | *** part/#asterisk wellng (n=welles@222.90.155.122) |
13:08.13 | *** join/#asterisk elcuco (n=diego@local.xorcom.com) |
13:20.11 | OloBola | trixter: should I be able to grab these as well: exten => s,2,agi(dtmf.php,10001|11569|friday|1|22|2006) $argv[2], $argv[3] etc? |
13:20.54 | *** join/#asterisk Guest3456 (n=root@200-233-134-246.xd-dynamic.ctbcnetsuper.com.br) |
13:23.45 | *** join/#asterisk Skymarshal (n=Skymarsc@p54AF67BB.dip.t-dialin.net) |
13:24.06 | OloBola | I got it. |
13:24.21 | Skymarshal | Hi, |
13:24.23 | Skymarshal | does anybody know what the difference between VM_CALLERID and VM_CIDNUM is? |
13:25.08 | blitzrage | Skymarshal: CALLERID has both the name and number, CIDNUM is just the number |
13:25.24 | Skymarshal | aaaaah!!!! Thanks! :-) |
13:25.35 | blitzrage | try using them in a NoOp() statement next time to see the actual output of what they are |
13:26.01 | blitzrage | or check your asterisk source code under the ./doc/ directory in README.variables <----- your friend!!! |
13:26.02 | zoa | hey ho blitzrage |
13:26.08 | blitzrage | zoa: ho hey! |
13:26.14 | Guest3456 | i found a bug in res_config_mysql.c, when the sip user unregister the file res_config_mysql.c put on mysql a lot of garbage on the tables sip_buddies in the fields port and ipaddr...anyone know how to fix it? |
13:26.40 | blitzrage | zoa: do you know of that application that lets you share the screen between multiple computers like a KVM kinda thing, but its software for Linux/Windows? Its on sourceforge, but I can't seem to remember the name of it |
13:26.49 | zoa | vnc |
13:26.55 | blitzrage | zoa: nope, thats not it :) |
13:27.10 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-24-214.38-151.net24.it) |
13:27.12 | zoa | you mean like one big screen ? |
13:27.12 | blitzrage | zoa: it lets you use a single KB/mouse combo and move across screens |
13:27.26 | blitzrage | zoa: yah, kinda. Its like a KVM, but you run the software on all the computers |
13:27.27 | tzanger | blitzrage: x2vnc |
13:27.29 | tzanger | x2x |
13:27.34 | blitzrage | tzanger: don't think thats it either |
13:27.39 | p0g0__ | blitzrage: tightvnc, realvnc , etc |
13:27.40 | tzanger | blitzrage: that's what I use at work |
13:27.44 | blitzrage | its not a VNC :) |
13:27.52 | tzanger | tightvnc or RDP if I need to actually see a remote screen on my laptop |
13:28.06 | Skymarshal | bye guys |
13:28.09 | tzanger | but at work I have a 20" monitor beside the laptop so I use x2vnc so I can just use my laptop mouse/kb on the PC |
13:28.10 | blitzrage | when I find it I'll tell you what it is because its awesome :) |
13:28.11 | zoa | hmm, blitzrage: i didnt know that existed, so if you find it, let me know |
13:28.12 | zoa | :) |
13:28.22 | blitzrage | zoa: yah, its vicked c00 |
13:29.46 | dpryo | blitzrage: synergy lets you share keyboard and mouse, but not display |
13:29.56 | blitzrage | dpryo: thats the one I wanted :) |
13:30.04 | blitzrage | dpryo: it lets you move the mouse/KB across multiple screens |
13:30.12 | dpryo | :) |
13:30.29 | tzanger | dpryo: yeah that's what I use x2vnc and x2x for |
13:30.31 | dpryo | I'm using it between my linux, mac and windows |
13:30.49 | blitzrage | dpryo: yah, I have 3 comps here at the school and odn't want to move my hands to another KB/mouse |
13:31.00 | tzanger | yeah that's exactly what these programs do |
13:31.03 | blitzrage | and I'd used that software before and it works great |
13:31.23 | blitzrage | vnc is a bitch since you have to control the remote computer on the same screen |
13:31.29 | Guest3456 | i found a bug in res_config_mysql.c, when the sip user unregister the file res_config_mysql.c put on mysql a lot of garbage on the tables sip_buddies in the fields port and ipaddr...somebody know how to fix it? |
13:31.31 | blitzrage | I want the same kb/mouse across multiple screens |
13:31.32 | tzanger | blitzrage: huh? |
13:31.38 | blitzrage | Guest3456: we heard you the first time |
13:31.52 | blitzrage | Guest3456: and we still don't know |
13:31.54 | zoa | blitzrage: tell me when you found it |
13:32.04 | tzanger | blitzrage: my laptop runs linux. I run x2vnc -east compiler.ca.benshaw.com:0. |
13:32.16 | tzanger | then I move my linux mouse over to the right of hte screen and now the kb/mouse runs the windows machine beside it |
13:32.17 | blitzrage | zoa: http://synergy2.sourceforge.net/ |
13:32.42 | tzanger | you could run a second one -west or -north or whatever and get a second/third/fourth system in the fray |
13:32.59 | blitzrage | tzanger: using all 3 monitors? |
13:33.11 | tzanger | blitzrage: YES |
13:33.12 | blitzrage | tzanger: I still picture VNC as using remote computers on the same screen |
13:33.15 | blitzrage | tzanger: I don't believe you |
13:33.20 | blitzrage | :D |
13:33.23 | tzanger | blitzrage: that is what it typically does, but x2vnc does not |
13:33.28 | blitzrage | oh I see |
13:33.34 | blitzrage | you must be lieing though |
13:34.05 | tzanger | synergy does the exact same thing, but why run two daemons? I already have VNC on every machine so I can hit them remotely... x2vnc does it without placing the screen |
13:34.23 | tzanger | x2x does the same thing but connects to another X server, not a VNC one |
13:34.32 | tzanger | there's likely a win2vnc too IIRC |
13:34.36 | tzanger | I know there is in fact |
13:35.40 | dpryo | tzanger: Can one move xclients from one server to another? |
13:35.58 | dpryo | That would rock. |
13:36.11 | dpryo | xinerama between computers :) |
13:36.14 | tzanger | dpryo: not that I'm aware of. There's no way to "split" the ownership of an app across two X servers |
13:36.26 | dpryo | x.org-people are working on it though. |
13:36.27 | tzanger | you'd need a third x server and have it actually handle it :-) |
13:36.35 | dpryo | Yeah, some proxy. |
13:37.03 | tzanger | that has caught me a few times.. >I'm so used to x2vnc I sometimes take an app and try to move it ot the monitor to get it off the laptop and hten I wonder for a moment as to why it didn't owrk :-) |
13:37.10 | dpryo | hehe |
13:39.38 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
13:40.23 | tzanger | heh |
13:40.35 | tzanger | synergy can't handle the three-finger salute but it owrks great on x2vnc :-) |
13:40.54 | tzanger | it does have some nice features, scrolllock locking would be handy |
13:42.35 | fugitivo | what's synergy? |
13:42.37 | fugitivo | (morning) |
13:42.57 | tzanger | synergy is the coming together of disparate ideas or concepts in unrelated fields |
13:43.15 | blitzrage | smart ass |
13:43.25 | tzanger | no, I have a sweet ass |
13:43.29 | fugitivo | not at this time |
13:43.38 | blitzrage | I guess I'll try this x2vnc then I guess since tzanger thinks its so great and he is all knowing and such... |
13:44.01 | tzanger | blitzrage: your apology and beer will be accepted at the next torastricon meeting |
13:44.08 | blitzrage | tzanger: :D |
13:44.29 | tzanger | I need to get cracking on this telebridge thing, jimvanm will be upset if I don't have something wild to show him |
13:44.41 | *** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
13:44.44 | blitzrage | tzanger: yah... I'm working on a redundency solution with DUNDi today |
13:44.51 | tzanger | :-) |
13:44.58 | blitzrage | tzanger: I don't know anything about your telebridge app :) |
13:45.00 | fugitivo | did you try the "Desktop sharing" feature of kde? |
13:45.02 | fugitivo | it's great |
13:45.05 | tzanger | blitzrage: I know you don't |
13:45.09 | tzanger | it's top secret shit |
13:45.18 | blitzrage | tzanger: good thing we're talking about it in this public forum :D |
13:45.22 | tzanger | yep |
13:45.28 | tzanger | gotta create a buzz |
13:45.38 | blitzrage | fugitivo: I prefer gnome now (I used to use KDE -- its sloooooow) |
13:45.44 | *** join/#asterisk _4d4m_ (n=adam@44-8-101-159.adsl.legend.co.uk) |
13:45.46 | fugitivo | slow? |
13:45.48 | tzanger | blitzrage: you're on crack |
13:45.52 | fugitivo | with todays cpu? nahh |
13:45.53 | blitzrage | its slower than gnome I've found |
13:45.53 | tzanger | or using Debian |
13:46.03 | fugitivo | lol |
13:46.11 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
13:46.13 | fugitivo | my sister uses gnome |
13:46.14 | blitzrage | ...and don't even get my started on debian |
13:46.22 | blitzrage | fugitivo: smart girl. She hot? |
13:46.28 | tzanger | hahahahha |
13:46.34 | fugitivo | she's not smart |
13:46.43 | fugitivo | i don't know if she's hot, it's my sister |
13:46.44 | blitzrage | smarter than you in regards to Xwindows :D |
13:46.55 | blitzrage | fugitivo: oh come on... I'm sure you know whether she's attractive or not |
13:47.03 | fugitivo | no, i don't |
13:47.14 | tzanger | kde's integration is what keeps me there. Gnome is messy and "big" and don't even get me started on the build environment and dependency hell there |
13:47.23 | fugitivo | i second that |
13:47.36 | fugitivo | if you install kde you don't need anything else |
13:47.37 | iDunno | you're all broken, ion3 is the true way forwards, it stays out of the way, it's light, and it shows you just how many apps are completely fucking broken ;) |
13:47.50 | fugitivo | i don't care if it's light |
13:47.54 | fugitivo | i have a lot of cpu and ram |
13:48.02 | tzanger | nah, I used to be a big windowmaker fan |
13:48.07 | tzanger | still am, just can't do without the integration |
13:48.13 | blitzrage | tzanger: know anyone who needs 2x1GB Registered ECC CL2.5 RAM? |
13:48.14 | fugitivo | i have 64bits and 1gb ram |
13:48.17 | tzanger | dcop and ioslaves rock your world |
13:48.24 | tzanger | blitzrage: I have some of that myself |
13:48.31 | fugitivo | why should i use a light desktop? :) |
13:48.36 | blitzrage | tzanger: I have some that I'm trying to sell <-- got it for $100. |
13:48.43 | tzanger | heh |
13:48.50 | blitzrage | tzanger: doesn't work in my AMD machine :) |
13:48.59 | tzanger | first rule of sales: never reveal your margins |
13:49.07 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
13:49.10 | blitzrage | tzanger: sales suck |
13:49.20 | fugitivo | sales keep us alive |
13:49.20 | blitzrage | tzanger: I think I might just eBay it |
13:49.37 | blitzrage | I just let all the crooks deal with sales |
13:52.09 | *** join/#asterisk Xandor (n=astar@ANantes-154-1-4-15.w81-53.abo.wanadoo.fr) |
13:52.14 | *** join/#asterisk MatsK (i=enforcer@c83-253-29-22.bredband.comhem.se) |
13:52.16 | *** join/#asterisk xheliox (n=jeff@user-0c6s3h2.cable.mindspring.com) |
13:52.56 | Xandor | hello |
13:53.47 | Xandor | i want to do a shortcut ex : on my soft phone i type 5555 & it call the number 0243435458 |
13:54.18 | *** part/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
13:54.41 | blitzrage | Xandor: exten => 5555,1,DoSomething() |
13:54.58 | fugitivo | i love postgresql |
13:55.02 | blitzrage | fugitivo: me too |
13:55.09 | tzanger | postgres rox yer sox |
13:55.22 | blitzrage | and pgAdmin makes it bareable to use :) |
13:55.27 | fugitivo | yes |
13:55.29 | tzanger | hahah |
13:55.31 | tzanger | yes pgadmin is nice |
13:55.40 | tzanger | anyway gotta get the kids to their mom and spend the day with alina. :_0 |
13:55.40 | benjk | yeah, psgql is the way to go |
13:55.41 | tzanger | er :-) |
13:55.42 | tzanger | later |
13:55.45 | Xandor | exten => 5555,1,Dial(0243435458) i tried this but it doesnt work |
13:55.48 | blitzrage | tzanger: peas |
13:55.58 | blitzrage | Xandor: you have no technology |
13:55.59 | fugitivo | so those mysql lovers can kiss my ass when they say that postgresql is difficult to use |
13:56.08 | blitzrage | Xandor: how does Asterisk know how to connect the call? |
13:56.09 | benjk | try Dial(ZAP/1/number,... |
13:56.11 | tzanger | fugitivo: amen; I always found mysql more difficult to install and use |
13:56.16 | Xandor | exten => 5555,1,Dial(SIP/0243435458) i tried this but it doesnt work |
13:56.17 | fugitivo | i agree |
13:56.17 | Xandor | idem |
13:56.18 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
13:56.23 | benjk | or Dial(SIP/number@yourprovider, ... |
13:56.33 | blitzrage | Xandor: you still don't have the right format |
13:56.35 | fugitivo | Xandor: you want to use SIP or ZAP? |
13:56.41 | Xandor | SIP |
13:56.51 | benjk | SIP/number@provider |
13:56.57 | fugitivo | yeah, read what benjk said |
13:56.58 | blitzrage | Dial(SIP/name_in_square_brackets_in_sip.conf/number_to_dial) |
13:57.07 | MT`AwAy | oh btw anyone knows if there's a way to use meetme without loading a module ? Currently my kernel does not have support for loadable modules |
13:57.10 | blitzrage | or number @ provider :) |
13:57.26 | fugitivo | MT`AwAy: you need zaptel or ztdummy |
13:57.26 | blitzrage | MT`AwAy: asterisk loads the app_meetme.so module, not the kernel. |
13:57.33 | blitzrage | and that :) |
13:57.38 | MT`AwAy | yup |
13:57.41 | MT`AwAy | meetme requires a timer |
13:57.51 | MT`AwAy | but there's only timers as kernel modules |
13:57.56 | blitzrage | yep |
13:57.56 | fugitivo | yes |
13:57.57 | benjk | app_conference works without timers |
13:58.01 | MT`AwAy | and I can't load a module on my current kernel |
13:58.07 | fugitivo | then recompile |
13:58.08 | MT`AwAy | app_conference ? |
13:58.14 | benjk | yep |
13:58.19 | MT`AwAy | fugitivo> I can't reboot this server at all |
13:58.33 | blitzrage | benjk: didn't know that didn't need a timer. How does it work then? (just curious) |
13:58.49 | tzafrir_laptop | you don't need to boot to install zaptel |
13:58.51 | blitzrage | MT`AwAy: sounds like a server that shouldn't be running Asterisk then :) |
13:58.55 | MT`AwAy | benjk> I'll look that on google ;) |
13:58.59 | _4d4m_ | hi all. i have an * server on my home lan, connectivity provided via adsl. my router forwards all packets from my public IP to my * box. i connect to FWD via iax. internal phones are connected to * using SIP. I have one way audio when recieving calls from FWD numbers (phone rings and i hear them, they dont hear me). i've tried all the nat options and been through the docs but nothing seems to work. can anyone help? |
13:59.33 | benjk | app_conference has less features than app_meetme |
13:59.34 | blitzrage | alright, I'm disconnecting from screen now so I can get some work done. Lates! |
13:59.34 | tzafrir_laptop | MT`AwAy, why can't you load modules? |
13:59.44 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
13:59.53 | fugitivo | _4d4m_: too long for this day and hour |
14:00.26 | _4d4m_ | fugitivo: heh.. you and me both |
14:00.54 | _4d4m_ | been messing with it and searching docs for the last few hours |
14:01.10 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
14:01.14 | MT`AwAy | tzafrir_laptop> the kernel does not have support for loading modules, and I can't reboot on a new kernel for now |
14:01.37 | fugitivo | MT`AwAy: sometimes you need to sacrifice something to get another thing :) |
14:01.54 | tzafrir_laptop | anybody here with bristuff 0.3.0 installed? |
14:02.03 | MT`AwAy | fugitivo> yup, but I'll first try app_conference |
14:02.39 | Xandor | [14:54:50] <blitzrage> Dial(SIP/name_in_square_brackets_in_sip.conf/number_to_dial) |
14:02.46 | Xandor | thanks its works |
14:03.15 | Xandor | didn't find this on google |
14:03.41 | tzafrir_laptop | Looking at asterisk.patch there I notitced it seems to change the version of the manager interface from 1.0 to 1.2 . Is that right? |
14:04.13 | elcuco | tzafrir_laptop, yes, that right. you really found that. :) |
14:05.57 | tzafrir_laptop | I would be bothered if noone else had |
14:07.51 | *** join/#asterisk javar (n=javar@69.79.51.8) |
14:08.11 | *** join/#asterisk shawn (n=welles@219.145.57.45) |
14:09.46 | *** join/#asterisk johan_hammy (i=Hammy@dialup-4.159.56.172.Dial1.Chicago1.Level3.net) |
14:09.56 | johan_hammy | *poke* |
14:10.20 | *** join/#asterisk p0g0 (n=p0g0@mrtc-mm-590160.mis.net) |
14:10.44 | johan_hammy | Can anyone here assist me with a zaptel make install error? |
14:15.00 | tzafrir_laptop | johan_hammy, you posted to asterisk-users? |
14:15.50 | tzafrir_laptop | note the '[ -f ]' . That is a syntax error, I believe |
14:15.53 | *** join/#asterisk Skymarshal (n=Skymarsc@p54AF67BB.dip.t-dialin.net) |
14:15.54 | johan_hammy | *nods* |
14:16.05 | tzafrir_laptop | What version of asterisk? |
14:16.22 | johan_hammy | I'm in digest mode (I already get hundreds of e-mails a day), so I wouldn't see any replies for a few hours. |
14:16.25 | johan_hammy | 1.2.1 |
14:16.54 | Skymarshal | Hi, if I use |
14:16.56 | Skymarshal | mailbox = password,name,email,operator=yes |
14:16.57 | Skymarshal | how does the system know who is operator? Where is that defined? |
14:19.24 | johan_hammy | I do see a "/bin/sh: -c: line 0: `if [ -n "" ]; then if [ -f ]; then mv -f .bak ; fi;"... in the previous command executed. |
14:19.32 | Simon- | Skymarshal: -= Info about application 'VoiceMail' =- |
14:19.42 | Simon- | Skymarshal: If the caller presses '0' (zero) during the prompt, the call jumps to extension 'o' in the current context. |
14:19.58 | tzafrir_laptop | johan_hammy, MODCONF is not defined... |
14:20.10 | Skymarshal | Simon: Thanks! :-) |
14:20.45 | tzafrir_laptop | johan_hammy, what distro do you use? |
14:21.20 | johan_hammy | CentOS 4.2 is the host OS, but I'm running OpenVZ for some virtual servers. |
14:21.37 | tzafrir_laptop | what kernel version? |
14:21.55 | johan_hammy | 2.6.8-022stab050.1 |
14:23.00 | tzafrir_laptop | do you have ther /etc/modprobe.conf and/or /etc/modprobe.d ? |
14:23.14 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
14:24.30 | tzafrir_laptop | anyway, the workaround is to rem-out the line in the install: target that checks for $(MODCONF) . |
14:24.42 | tzafrir_laptop | You don't really need that file. |
14:24.49 | johan_hammy | When I locate modprobe, I get /sbin/modprobe. WHen I check the host OS, I see some other modprobe stuff. |
14:25.24 | johan_hammy | okay. THanks. Maybe what I'll do is talk to OpenVZ about modprobe, but if they're no help, I'll just comment it out. |
14:25.25 | tzafrir_laptop | Simply make sure you run ztcfgsometime before asterisk is started and after the modules are loaded |
14:26.05 | tzafrir_laptop | Anyway, please report a bug in the mantis. It's a buggy part in the makefile |
14:26.55 | johan_hammy | Thanks much |
14:27.44 | johan_hammy | I've only done a couple Asterisk systems. THis is my first 1.2.1 build and I decided to go at it on a Virtual Server. I've had quite a few bumps along the way. |
14:27.47 | Xandor | some one knwo disa? |
14:29.32 | Xandor | when i call ma asterisk & i recall another number if i stop the call , it continues ringing on the other side |
14:31.01 | *** join/#asterisk joesat (n=gianni@adsl-205-217.37-151.net24.it) |
14:32.26 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
14:37.04 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
14:38.16 | *** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it) |
14:46.37 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
14:47.20 | dorphalsig | Hi, I've recompiled asterisk with R2 Support, but when I try to load it, I get the following error [chan_unicall.so]Jan 8 09:34:24 WARNING[18213]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging |
14:47.20 | dorphalsig | Jan 8 09:34:24 WARNING[18213]: loader.c:554 load_modules: Loading module chan_unicall.so failed! |
14:49.29 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
14:50.27 | *** join/#asterisk bn-7bc (n=a@pppoecl82052.minlos.no) |
14:51.10 | dorphalsig | anybody= |
14:58.12 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
15:03.04 | Skymarshal | I am writing a documentation about the voicemailsystem of asterisk. Is there a plan of the voicemail menu? Where can I find it? |
15:03.47 | rob0 | ~docs |
15:03.50 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
15:04.12 | rob0 | particularly http://www.voip-info.org/wiki-Asterisk is best IMO |
15:04.32 | Skymarshal | But there is no plan of the menu. |
15:04.56 | rob0 | hmmm, Asterisk-cmd-Voicemail or something like that? |
15:05.54 | svenna_ | hi all |
15:06.08 | svenna_ | hey, voicemail is my problem at the moment :-) |
15:06.14 | svenna_ | im a bid stuck here |
15:06.23 | svenna_ | i got it working |
15:06.47 | ruud_org | Skymarshal: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain |
15:07.15 | Skymarshal | ruud_org: Thanks! :-) |
15:07.38 | svenna_ | but, when i add new busy message for a user and dial in, i get to here the users message AND the default one. |
15:07.54 | svenna_ | where can i configure to nit here the default one? |
15:13.07 | *** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
15:15.16 | *** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
15:16.31 | dorphalsig | Hey |
15:16.42 | dorphalsig | this is the 4th time I recompile libunicall |
15:16.58 | dorphalsig | and I still get WARNING[2183]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_unicall.so: undefined symbol: uc_set_logging |
15:16.59 | *** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
15:17.13 | dorphalsig | anybody knows anything about R2 support? |
15:27.39 | *** join/#asterisk davism (n=davism@dmcpherson.dsl.patriot.net) |
15:29.35 | *** join/#asterisk FastJack (i=fastjack@p5091FE06.dip.t-dialin.net) |
15:32.54 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
15:33.52 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
15:34.37 | ManxPower | dorphalsig, that sort of message usually means a mismatch between the version of the module and the version of Astersk |
15:35.02 | ManxPower | Are you SURE that Unicall has been ported to the version of Asterisk you are using? |
15:35.41 | *** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
15:39.12 | ManxPower | My internet connection sucks. I can either listen to streaming audio OR do something else that uses the internet, but not both. |
15:40.20 | *** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
15:40.29 | *** join/#asterisk jahani (n=k@adsl196-153-25-206-196.adsl196-1.iam.net.ma) |
15:47.46 | dorphalsig | ManxPower --> No, it hasnt, but ppl have installed it without problems ... |
15:48.33 | dorphalsig | I've had reports... of it working w/o problems... |
15:49.40 | svenna_ | how can i keep asterisk from playing: Playing 'vm-intro' (language 'en') and make it playing 'de'-files which are stored under /de? |
15:49.45 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfj4h.dialup.mindspring.com) |
15:49.58 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfj4h.dialup.mindspring.com) |
15:51.50 | ManxPower | svenna_, "show application setlanguage" |
15:52.20 | svenna_ | do i have to type that in asterisk console? |
15:52.28 | svenna_ | sorry, im a noob... |
15:53.32 | rob0 | svenna_: yes |
15:53.44 | svenna_ | ok i see |
15:53.46 | svenna_ | thx |
15:55.58 | svenna_ | but where do i SET the language? in extension.conf? |
15:56.57 | inv_Arp | svenna_: you got to start looking at www.voip-info.com |
15:57.26 | svenna_ | ok, i found it |
15:57.31 | svenna_ | sorry 4 that! |
15:58.05 | ManxPower | svenna_, you can set the language in several ways, as part of the extension in extensions.conf or for the device in zapata.conf, sip.conf, etc |
15:58.10 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-243-88-211.dsl.tulsok.swbell.net) |
15:58.16 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
15:58.35 | svenna_ | ah, ok thanks a lot! |
15:58.57 | svenna_ | i will look four it in capi.conf in my case |
15:59.25 | ManxPower | BTW, does anyone here know of 1u or 2u cases that can take a standard ATX motherboard and support at least 1 PCI card (with maybe a vert to horiz riser card)? |
16:00.17 | ManxPower | I'm looking at colocating at a local ISP. My current case is like 4U since until now it's always been in racks that are not charged by space used. |
16:01.42 | *** join/#asterisk kiwnix (n=egarcia@229.red-82-158-154.user.auna.net) |
16:02.21 | *** join/#asterisk puzzled (n=yeahrigh@53533C13.cable.casema.nl) |
16:05.39 | svenna_ | and it works :-) |
16:12.39 | _Sam-- | a 2u one shouldnt be hard to find |
16:13.05 | _Sam-- | the 1u will probably be hard to find a standard ATX compatible |
16:14.31 | Sparkz | I have a couple of http://www.chieftec.de/?page=products_big&id=95&k_id=3&language=uk |
16:14.41 | Sparkz | Works well |
16:16.15 | Mother | has anyone compiled chan_bluetooth for 1.2.1? |
16:16.46 | *** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
16:16.51 | Mother | I've been trying various fixes found googling around but nothing cures a bunch of undeclared's |
16:18.29 | *** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
16:21.30 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
16:23.07 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
16:23.36 | *** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
16:26.51 | *** join/#asterisk ToTo (n=ToTo@host242-83.pool8260.interbusiness.it) |
16:30.40 | *** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com) |
16:31.19 | ManxPower | Sparkz, thanks |
16:31.39 | ManxPower | Sparkz, I figured 2u would be the smallest I could get that would work with a standard motherboard |
16:32.46 | *** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
16:32.51 | Mother | ManxPower: do you know if chan_bluetooth can be compiled into 1.2.1? |
16:33.02 | ManxPower | Mother, I do not know. |
16:33.07 | Mother | OK thanks |
16:34.33 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
16:37.01 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
16:37.04 | PakiPenguin | hello everyone |
16:42.57 | *** join/#asterisk dsl1518 (n=dsl@p54A7EF32.dip.t-dialin.net) |
16:43.40 | *** join/#asterisk BriSch (n=BriSch@dslb-084-059-105-002.pools.arcor-ip.net) |
16:43.44 | *** join/#asterisk llirk (n=majestic@203-214-25-173.dyn.iinet.net.au) |
16:52.48 | *** part/#asterisk javar (n=javar@69.79.51.8) |
16:55.08 | *** join/#asterisk calennert (n=calenner@66-191-55-096.dhcp.gnvl.sc.charter.com) |
16:56.10 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
16:56.45 | *** part/#asterisk Naturalblue (n=Kay@195.26.12.229) |
17:07.39 | *** join/#asterisk cyberd0g (n=blah@ip70-177-41-173.br.br.cox.net) |
17:09.14 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
17:10.34 | *** join/#asterisk acehunky (n=chat_jok@59.184.5.167) |
17:10.45 | acehunky | hi ppl |
17:10.46 | mjmac | not too much in the way of SIP-based soft videophones, eh? |
17:12.04 | *** join/#asterisk iq (n=iq@71-38-75-128.omah.qwest.net) |
17:12.08 | acehunky | hi, i am looking for some document which explains what is context, dial plan, extension etc .. i have been surfing asteriskguru and voip-info .. but its kinda confusing |
17:12.58 | benjk | ~docs |
17:12.59 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
17:13.15 | *** join/#asterisk Assid (n=assid@203.115.64.5) |
17:13.33 | *** part/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
17:13.50 | ManxPower | ~mailinglist |
17:13.52 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
17:14.02 | Skymarshal | acehunky: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
17:16.44 | *** join/#asterisk Tili (i=Tili@202-133-65-187-dialup.sat.net.pk) |
17:17.32 | acehunky | umm lemme see the urls .. thanks jbot, Skymarshal |
17:18.33 | *** join/#asterisk roulduke_ (i=612j9l3n@p508D42CF.dip0.t-ipconnect.de) |
17:19.24 | acehunky | aah and one more thing that i wanted to ask ... how can i simulate a call via cli ? |
17:19.28 | acehunky | is there any way |
17:19.29 | acehunky | ? |
17:22.21 | benjk | you drop a call file into /var/spool/asterisk/outgoing |
17:22.36 | benjk | btw, 'jbot' is a bot |
17:23.40 | benjk | if you do ~docs or ~mailinglists etc etc, it will spit out some info it has been fed before |
17:23.53 | wunderkin | why when i do a svn checkout of asterisk-1.2 it says revision 7866 but show version says 7847M? |
17:24.09 | wunderkin | do they go off of different things? |
17:24.35 | Corydon76-home | 7847 was the latest change to that branch... 7866 is the latest revision to the tree |
17:25.01 | wunderkin | ok... |
17:25.16 | acehunky | oh thanks benjk ... good to know that .. do you know of any hacks with this bot ? |
17:26.31 | *** join/#asterisk lrizzo (n=luigi@81-174-21-10.f5.ngi.it) |
17:27.48 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
17:27.57 | wunderkin | so i probably gave the wrong revision on my bug.. oh well |
17:31.00 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
17:31.47 | benjk | hacks? |
17:32.34 | *** join/#asterisk chapeaurouge (n=chap@85.201.81.201) |
17:46.02 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
17:48.47 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
17:50.24 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
17:50.57 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
17:51.57 | lrizzo | ping... any autoconf guru around here ? |
17:52.50 | svenna_ | what does trunkmsg realy mean? or: "MSD digits ti strip" what are MSD digits? |
17:53.20 | tzafrir_laptop | acehunky, I use ! to run a script that generates a call file... |
17:54.21 | swm_ | !kill fags |
17:54.24 | swm_ | file dies.. |
17:54.35 | file | how rude. |
17:54.39 | bkw__ | swm_, you're rude |
17:54.55 | swm_ | LOL |
17:55.05 | Corydon76-home | Funny, he doesn't look like a cigarette |
17:55.18 | Corydon76-home | how do you kill a cigarette, anyway? |
17:56.07 | *** join/#asterisk theblue (n=theblue@pcp04402293pcs.nrockv01.md.comcast.net) |
17:56.09 | theblue | Hi all. |
17:56.28 | theblue | I'm trying to connect to FWD using asterisk, and I'm wondering how to use an outbound proxy, if its possible. |
17:57.40 | *** join/#asterisk Skymarshal (n=Skymarsc@p54AF67BB.dip.t-dialin.net) |
17:59.06 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
17:59.12 | swm_ | bkw_ is a child molestor |
18:01.19 | Corydon76-home | I'd strongly advise you to retract that |
18:02.39 | rob0 | Maybe swm_ is Fred Phelps? :) |
18:03.27 | Corydon76-home | It's still slander |
18:04.06 | rob0 | what brought all this on anyway? (/me scrolls up) |
18:04.23 | rob0 | hmmm |
18:04.30 | jbroome | slander is spoken |
18:04.35 | jbroome | libel is written |
18:04.37 | theblue | libel is written. |
18:04.38 | theblue | jinx. |
18:04.42 | jbroome | :) |
18:04.50 | dogtanian | nope |
18:04.53 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
18:04.55 | theblue | But does anyone know how I can use an outbound proxy with SIP? |
18:04.57 | dogtanian | liable is published |
18:05.05 | sivana | hehe |
18:05.12 | jbroome | theblue: on topic questions will not be tolerated |
18:05.24 | theblue | jbroome: OH MYE. |
18:05.29 | rob0 | file is a Fine Human Being (for a canuck kid, that is ;) ) |
18:05.31 | jbroome | :) |
18:05.39 | xachen | he is? |
18:05.46 | rob0 | definitely |
18:05.52 | rob0 | right file? |
18:05.53 | xachen | a Candian, eh? |
18:06.05 | xachen | Canadian* |
18:06.09 | xachen | gasp |
18:06.13 | xachen | New Brunswick |
18:06.17 | xachen | bloody NBers :P |
18:06.30 | rob0 | At least he's not a Newfie. |
18:06.32 | file | yup, all my fault |
18:06.36 | sivana | heh |
18:08.07 | xachen | Newfies make me laugh |
18:08.10 | xachen | << Albertan here |
18:08.17 | *** part/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
18:09.05 | svenna_ | is there a way to skipp vm-intro in voicemail? |
18:09.09 | *** join/#asterisk swm_ (n=admin@digitaldatabits.net) |
18:09.27 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
18:10.24 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
18:10.45 | xachen | actually New Brunswick people make me laugh |
18:10.51 | xachen | they all seem funny for some reason |
18:10.55 | xachen | goofy is the word actually :) |
18:10.56 | alexhopper | We are |
18:11.02 | alexhopper | Completely |
18:11.43 | alexhopper | Ask file about how I run band practices, he'll tell you the definition of goofy :p |
18:12.09 | file | or the definition of failure :P |
18:15.52 | swm_ | yawn |
18:19.41 | *** join/#asterisk cnet2 (n=jjohn@201.192.107.58) |
18:20.31 | *** join/#asterisk chat_jokey (n=chat_jok@59.184.13.225) |
18:24.27 | cnet2 | hi, i just bought a polycom 501, but i wanted to update the firmware. I'm not sure how it's done. I installed tftp server on my machine, and put the boot and sip files for the firmware on the root dir. How do I access the phone, or how do I make the phone go to my boot server? |
18:28.59 | theblue | cnet2: Usually there's a way to force a TFTP update on your phone. |
18:29.09 | cnet2 | got it.. jeje |
18:29.13 | theblue | cnet2: Do you have your tftp server running and ready to update at the moment? |
18:29.28 | *** join/#asterisk brookshire[home] (i=matt@69.247.184.46) |
18:29.37 | cnet2 | yes.. i the thing is i hadn't seen the tftp option on the phone.. lol :S, |
18:29.42 | theblue | Oh. |
18:36.15 | CoaxD | fuck. why do these stupid people crack a useful app, only to fucking put a virus in it when they release? |
18:37.05 | rob0 | for fun and profit? |
18:38.14 | CoaxD | rob0: I spose. But. c'mon. why not just release the crack? |
18:42.35 | *** join/#asterisk calennert (n=calenner@66-191-55-096.dhcp.gnvl.sc.charter.com) |
18:45.05 | tzafrir_laptop | Why do other _____ people use software from an untrusted source? |
18:45.54 | dogtanian | to make the rest of us look clever |
18:48.12 | tzafrir_laptop | As I wrote to -users: now that 1.2.2 has been released, I can safely announce my 1.2.1 debs :-( |
18:49.42 | tzafrir_laptop | hmm, why did I think 1.2.2 was released? |
18:49.54 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
18:50.27 | theblue | Can anyone tell me how to make it so that after I dial "393" and then any string of numbers, it only dials that string of numbers (of any length) to SIP/fwd? |
18:51.24 | wunderkin | from the dev topic |
18:51.48 | benjk | dial(SIP/${EXTEN:3}@fwd ... |
18:52.05 | theblue | But what extension do I do? |
18:52.15 | theblue | _393.? |
18:52.19 | benjk | _393X. |
18:52.25 | fndude | Hi all. I just installed asterisk. I am trying to test it, but currently I have no voip provider. I would like to connect via my local network/xp client. From what I gather I should have port 4569 open. Is this a tcp/ip port? |
18:52.32 | theblue | But that first number could also be a *. |
18:52.55 | benjk | _393. will do as well |
18:52.56 | theblue | fndude: No, UDP. |
18:53.10 | theblue | benjk: Ok, then why in hell does X-Lite keep giving me a 404 not found error? |
18:53.33 | theblue | (Actually, I'm doing this in IAX2) |
18:54.18 | fndude | theblue: ok very good. Thank you. |
18:54.23 | theblue | fndude: No prob. |
18:56.46 | *** join/#asterisk SwK_ (n=SwK@12-219-151-128.client.mchsi.com) |
18:57.32 | *** join/#asterisk matt_ (n=matt@cpe-66-108-191-40.nyc.res.rr.com) |
18:59.34 | benjk | sip debug is your friend |
19:00.32 | theblue | D'oh. |
19:01.12 | theblue | What I'm doing could be entirely done in SIP, you think SER would be easier to use for this/ |
19:02.54 | theblue | No, it wouldn't, apparently. |
19:03.03 | *** join/#asterisk jannic (n=jan@potemkin.hitnet.RWTH-Aachen.DE) |
19:06.55 | *** join/#asterisk ast_freak|Laptop (n=jesse@12.28.106.2) |
19:08.56 | dorphalsig | hey |
19:09.19 | dorphalsig | can anybody help me build unicall? I've been trying since this morning |
19:09.30 | dorphalsig | and all I get is a warning telling me spandsp.h is unusable |
19:10.51 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
19:12.57 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
19:15.09 | cnet2 | anyone had any problems with the digium TDM2400?, my xorcom asterisk installation doesn't seem to recognize it. any special drivers i should install? |
19:16.20 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
19:19.13 | *** join/#asterisk T42X (n=T42X@193.219.62.88) |
19:25.02 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
19:25.11 | *** join/#asterisk CleanerX (n=nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
19:27.52 | brookshire[home] | cnet2: what version of zaptel are you running? |
19:27.54 | brookshire[home] | do you know? |
19:27.55 | T42X | my asterisk is behind a nat, forwarded to outside. everyone can connect and call, but there are some problems though. if someone from outside calls someone who's inside the network - he only hears his voice, but the one who's inside cannot hear the callers voice. and when both callers are from the inside network - none of them can hear each other |
19:28.00 | cnet2 | nop.. |
19:28.04 | cnet2 | let me find out |
19:28.56 | *** join/#asterisk kenrstone (n=krstone@ool-4573f3dc.dyn.optonline.net) |
19:37.48 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
19:38.20 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
19:40.34 | fndude | Is the correct way to build your architecture under the extensions.conf file? |
19:41.11 | T42X | architecture? |
19:41.24 | fndude | The way your calls are recived, extensions, etc. |
19:42.30 | rob0 | dialplan is probably the term you want |
19:45.37 | mjmac | anyone else noticing a problem with VPC origination today? |
19:46.04 | file | brookshire[home]: !!!!!!!!!!!! |
19:49.56 | brookshire[home] | josh!!!!!! |
19:50.06 | file | omg omg omg Mattttttttttt |
19:50.09 | *** join/#asterisk O3OBBITHA (n=O3OBBITH@213.150.185.18) |
19:50.20 | file | brookshire[home]: fyi, you're H-O-T |
19:50.28 | brookshire[home] | oh? |
19:50.39 | file | a pixie told me so |
19:50.42 | file | so it MUST be true |
19:51.00 | brookshire[home] | heh.. |
19:51.11 | *** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
19:51.34 | *** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
19:51.52 | file | brookshire[home]: how are you feeling? |
19:52.03 | brookshire[home] | much better! |
19:52.08 | brookshire[home] | my mom is sick now :/ |
19:52.12 | file | :( |
19:52.22 | brookshire[home] | everyone around me is getting sick |
19:52.39 | brookshire[home] | i called it black death |
19:52.40 | brookshire[home] | hah |
19:53.26 | *** join/#asterisk xtr (n=01928375@S0106000c41ed11e1.vf.shawcable.net) |
19:53.46 | file | atleast I'm not getting sick |
19:53.53 | file | if so I would be very unhappy with you |
19:54.48 | brookshire[home] | i would be very unhappy with me too |
19:55.12 | brookshire[home] | i have a memory leak in java |
19:55.19 | brookshire[home] | i didn't think that was possible |
19:55.31 | brookshire[home] | but apparently it is! |
19:55.48 | file | that's because you're silly |
19:56.05 | brookshire[home] | i have no idea where to find it either |
19:56.06 | brookshire[home] | lol |
19:56.13 | *** join/#asterisk brif8 (n=The_Bear@lazyjtrainingcenter.com) |
19:56.19 | file | turn around three times, and say "memory leak be gone!" |
19:56.24 | brookshire[home] | i feel like doing some simple loop tests to see if it is just java |
19:56.37 | brookshire[home] | no clicking the heals? |
19:57.03 | file | nope |
19:57.15 | file | the system has been upgraded. |
19:58.02 | brif8 | curious question monitor() records the conversation to a sound file to be played back later. But is it possible to have phone 1 making a converation and phones 2&3 listening in. NO not NDA spying. But for training purposes to help train new help desk staff. We handle a medium helpdesk and constantly have a staff change over. I would like to use actual phone calls as part of the training ex. |
19:58.45 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
19:59.58 | *** join/#asterisk LoRez_ (i=lorez@freenode/staff/lorez) |
20:00.38 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
20:03.52 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
20:06.12 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
20:07.35 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.159) |
20:08.08 | *** join/#asterisk razu_ (n=razu@217-159-227-104-dsl.prn.estpak.ee) |
20:10.43 | *** join/#asterisk Kernel_co (n=I@73.230.dial-up.xter.net) |
20:17.34 | *** join/#asterisk file[laptop] (n=jcolp@mctnnbsa24w-142167048221.nb.aliant.net) |
20:18.25 | ruud_org | is there any way to pass voice mail status (mwi) from one asterisk box to another (let's say you have sip phones registered to box A, but their voicemail lives on box B) |
20:19.21 | rob0 | hey brif8 that reminds me, I never figured out my MixMonitor answer. What do you have to specify for sample rate, size and encoding for the resulting .raw file? |
20:19.43 | *** join/#asterisk john867530 (n=john8675@12-218-49-92.client.mchsi.com) |
20:20.48 | john867530 | I am trying to get zapras to work I installed and patch pppd and when I try to connect I get "LCP: timeout sending Config-Requests" also when I dial the number direct it does not make modem tones |
20:31.33 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
20:42.25 | *** join/#asterisk Defraz_ (n=t0tal@72.24.220.144) |
20:42.49 | _4d4m_ | hi all.. asterisk is refusing to accept calls from a user (over iax), claiming there is no context available to handle the call. the error message states it is looking for a context in the format s@<my-context-label> |
20:42.59 | *** join/#asterisk anon-troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
20:43.15 | _4d4m_ | and whatever i change the context label to, the error message reflects it.. any ideas? |
20:51.53 | *** join/#asterisk Defraz_ (n=t0tal@72.24.220.144) |
20:56.24 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
20:57.30 | De_Mon | _4d4m_ the context needs to exist in extensions.conf |
20:58.17 | *** join/#asterisk dorphalsig (n=dorphals@200.106.223.5) |
20:58.20 | dorphalsig | Hi |
20:58.31 | De_Mon | dorphalsig sup |
20:58.52 | dorphalsig | Anybody has any experience building spandsp? |
20:59.03 | dorphalsig | I build spandsp |
20:59.10 | dorphalsig | and then I try to build libsupertone |
20:59.12 | _4d4m_ | De_Mon: the context does.. it is listed in iax.conf and extensions.conf as [contextname], but whatever it is, the call is refused saying s@[contextname] does not exist |
20:59.45 | dorphalsig | and I get a funky message saying "spandsp.h is there but can not be compiled, make sure you have no missing headers bla bla bla" |
21:00.06 | dorphalsig | but I get no warnings or anything building spandsp |
21:00.11 | dorphalsig | and I need that to add r2 support |
21:00.56 | De_Mon | hrm |
21:01.41 | dorphalsig | checking spandsp.h usability... no |
21:01.41 | dorphalsig | checking spandsp.h presence... yes |
21:01.54 | dorphalsig | configure: WARNING: spandsp.h: present but cannot be compiled |
21:01.54 | dorphalsig | configure: WARNING: spandsp.h: check for missing prerequisite headers? |
21:02.32 | *** join/#asterisk mistral (i=mistral@jstevenson.plus.com) |
21:02.53 | dorphalsig | ideas? |
21:02.57 | dorphalsig | suggestions? |
21:03.03 | dorphalsig | miraculous solutions? |
21:04.31 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
21:05.29 | dorphalsig | ok I'll settle for a not so miraculous solutions |
21:05.33 | dorphalsig | as long as it works :P |
21:05.54 | QbY | I've got a queue, with agents that aren't actually on the phone system.. (They are in another office we connect via pstn) the Dial command has the t option so that they can transfer back in.. However, today for some reason, its saying -- Playing 'pbx-transfer' (language 'en') |
21:05.54 | QbY | <PROTECTED> |
21:05.54 | QbY | <PROTECTED> |
21:06.42 | QbY | its like it doesn't even listen to the entire extension (ie. 303) |
21:16.54 | *** join/#asterisk _mistral (i=mistral@jstevenson.plus.com) |
21:23.06 | *** join/#asterisk shacha (n=shacha@32.179.8.67.cfl.res.rr.com) |
21:24.01 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
21:27.05 | *** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
21:27.27 | mrtwister | testing sip under DMZ.. can someone call me? |
21:28.03 | dogtanian | whoch country? |
21:28.10 | dogtanian | *i |
21:28.55 | mrtwister | dogtanian: over sip |
21:29.02 | mrtwister | dogtanian: not intl call |
21:29.17 | dogtanian | ah |
21:29.26 | dogtanian | i only do iax to external :) |
21:29.40 | mrtwister | i see |
21:29.41 | dogtanian | fiddles about with sip for ages but it ended up pissing me off |
21:40.33 | Blankman | dogtanian: what are you using for your ata's? |
21:41.00 | Blankman | dogtanian: I do the same ... SIP is only for internal networks that are wired :-) |
21:41.44 | Blankman | Qwell, you on? |
21:41.48 | Qwell | Blankman: hey, I am |
21:41.52 | Qwell | never got an email from you |
21:42.02 | Blankman | Qwell: sorry I forgot to send you the stuff :) |
21:42.14 | Qwell | np, was wondering what happened |
21:42.34 | Blankman | I will get it to you later ... you around in a few hours? |
21:42.37 | *** join/#asterisk aksis (i=aksis@republicofarizona.net) |
21:42.41 | Qwell | Blankman: probably, yeah |
21:42.45 | Blankman | I have some ? to ask you about festival if your game? |
21:43.06 | SERGEUS | How can i get UNIQUEID inside my app? |
21:43.07 | Qwell | I've done festival twice...it only went well once |
21:43.12 | Qwell | SERGEUS: cdr? |
21:43.19 | Qwell | ${CDR(uniqueid)} |
21:43.38 | SERGEUS | char *uniqueid; |
21:43.38 | SERGEUS | <PROTECTED> |
21:43.39 | SERGEUS | ? |
21:44.14 | Qwell | no |
21:44.19 | Qwell | I don't think that'll work |
21:44.29 | Qwell | You'll want to pull it from the CDR |
21:44.35 | SERGEUS | yes :) that doesn't work - why? |
21:44.48 | Qwell | because uniqueid isn't a variable |
21:45.11 | SERGEUS | i thought it is ... http://www.voip-info.org/wiki-Asterisk+variables |
21:45.42 | *** join/#asterisk iq (n=iq@71-38-75-128.omah.qwest.net) |
21:47.21 | Qwell | okay, so it is |
21:48.22 | SERGEUS | i'll try ast_cdr_getvar(chan->cdr, "uniqueid",&tmp, workspace,sizeof(workspace), 0); now |
21:50.50 | *** join/#asterisk TaSo (i=licucude@ool-44c784a0.dyn.optonline.net) |
21:54.31 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
21:55.36 | SERGEUS | Qwell, thank you very much |
21:56.28 | *** join/#asterisk miguellinux (n=miguel@64.76.202.18) |
21:58.40 | SERGEUS | BTW, simple dump showing that there are only 4 variables defined for my SIP channel (SIPCALLID, SIPUSERAGENT, SIPDOMAIN, SIPURI) - and what about all other Variables described in WIKI? I'm doing something wrong, or eiki is outdated? |
22:00.11 | Qwell | SERGEUS: for the most updated variables, look in doc/README.variables |
22:00.31 | *** part/#asterisk shacha (n=shacha@32.179.8.67.cfl.res.rr.com) |
22:00.45 | blitzrage | Qwell: ! |
22:00.49 | Qwell | blitzrage: ! |
22:00.51 | SERGEUS | Qwell, thanks :) |
22:00.57 | blitzrage | Qwell: how goes? |
22:00.59 | Qwell | blitzrage: Shame you aren't going to ETel... |
22:01.03 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
22:01.08 | blitzrage | Qwell: I might actually |
22:01.12 | Qwell | oh yeah? |
22:01.17 | blitzrage | Qwell: its still a possibility |
22:01.22 | Qwell | What about the one in FL? |
22:01.29 | blitzrage | Qwell: hey, you can program in C right? |
22:01.33 | Qwell | yeah |
22:01.36 | blitzrage | Qwell: yah, don't think that's happening anymore |
22:01.49 | blitzrage | file: there you are :) |
22:01.51 | Qwell | ahh, sucks |
22:02.04 | Qwell | blitzrage: well, if you're at ETel...umm...you can buy the beer this time! :D |
22:02.28 | blitzrage | anyways... I'm looking for someone who could maybe make it so that when you have multiple regexten's for a peer, it adds them all, and not just the last one |
22:02.32 | blitzrage | Qwell: no problem |
22:02.41 | file | blitzrage: oh... that wouldn't be too hard |
22:03.00 | blitzrage | file: yah, I figured as much. Should simply be a loop checking for all regexten's |
22:03.02 | Qwell | yeah, could probably just copy the code from codecs or contexts |
22:03.05 | blitzrage | like... a while loop :) |
22:03.17 | file | codecs are different... |
22:03.21 | blitzrage | I need to go and look at C code one of these days and be able to implement this small stuff myself :) |
22:03.44 | Qwell | file: well, the adding to the flag part, yeah, but it still accepts multiple "allow=" lines |
22:03.46 | blitzrage | anyways. With it I will be able to dynamically find a peer and the DID associated with that peer (a peer may have multiple DIDs) |
22:04.08 | blitzrage | was surprised that wasn't how it worked anyways :) |
22:04.21 | *** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au) |
22:04.31 | file | ah crap I don't have my certificate on this box |
22:04.47 | blitzrage | Qwell: you going to Etel? |
22:05.15 | Qwell | blitzrage: very very likely |
22:05.16 | blitzrage | Qwell: Damin and KK and I are trying to hook up a hotel room so we can go |
22:05.29 | blitzrage | Qwell: I'll be in K.C. the week of Jan 16th anyways |
22:05.30 | Qwell | gonna share a room with JunK-Y if he goes |
22:05.34 | Qwell | KC? |
22:05.39 | blitzrage | Qwell: Kansas City |
22:05.43 | Qwell | right |
22:05.46 | blitzrage | Qwell: nice! he's cool |
22:05.50 | Qwell | What's KC have to do with SF? |
22:06.05 | blitzrage | Qwell: Astricon training the week of Jan. 16th |
22:06.09 | Qwell | ahh |
22:06.14 | Qwell | So, not in Canadialand |
22:06.18 | blitzrage | no sir |
22:06.22 | Qwell | I see |
22:06.37 | Qwell | If I can get off work, I might go to Spring VON too |
22:06.44 | blitzrage | VON... ewwww ;) |
22:06.51 | Qwell | file is making me |
22:06.52 | Qwell | heh |
22:06.54 | blitzrage | lol |
22:07.22 | file | I'm cheating |
22:07.39 | file | I'm flying out in March for a job interview, and cheating and doing a cheap US flight to San Jose :P |
22:07.50 | Qwell | s/cheating/commiting a felony/ |
22:07.51 | Qwell | :D |
22:07.55 | blitzrage | nice |
22:08.13 | Qwell | Can anybody hit battle.net? ;/ |
22:08.15 | *** join/#asterisk ys76 (i=ys76@linoa.etherkiller.de) |
22:08.15 | file | to quote the company, "we don't care when you leave here, as long as your destination is your home" |
22:08.18 | Qwell | Was wondering why wine wouldn't let me update, heh |
22:08.29 | Qwell | file: VON = home |
22:08.36 | Qwell | home is where your friends are |
22:08.36 | Qwell | heh |
22:08.36 | blitzrage | anyways... I'm outta here for a bit. I'll look up someone here if I can't figure out my regexten stuff myself and maybe throw some cash around |
22:08.40 | file | Qwell: haha |
22:08.44 | file | blitzrage: kk |
22:08.55 | *** part/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
22:08.59 | blitzrage | file / Qwell: peas in the middle east! |
22:09.09 | file | peace! |
22:12.39 | *** join/#asterisk Jestre (n=ack@dargo.trilug.org) |
22:12.45 | *** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net) |
22:12.59 | Jestre | Anyone having trouble with voicepulse connect today? |
22:13.31 | mogorman | hey file |
22:13.35 | mogorman | hows it going |
22:13.40 | file | peachy, you? |
22:13.43 | mogorman | lots of bugs on the bug tracker.... |
22:20.07 | *** join/#asterisk harryk (n=me@harry.org.ua) |
22:20.46 | *** join/#asterisk zebedee1 (n=zebedee1@allthebrooks.plus.com) |
22:25.14 | fugitivo | why i need to discuss search engine optimization if I can have google? |
22:25.34 | *** join/#asterisk SugarGuest142 (n=SugarGue@BSN-95-217-164.dial-up.dsl.siol.net) |
22:26.34 | *** join/#asterisk aksis (i=aksis@idea-anvil.net) |
22:28.06 | SugarGuest142 | hi! i'm new to asterisk |
22:28.16 | fugitivo | congratulations |
22:28.19 | SugarGuest142 | :) |
22:28.31 | *** join/#asterisk BladeRunner05 (n=gianni@adsl-146-214.38-151.net24.it) |
22:28.49 | SugarGuest142 | and I am looking for a install guide - aka For dummies - does it exist? |
22:29.00 | fugitivo | ~docs |
22:29.01 | jbot | well, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
22:29.05 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
22:29.08 | fugitivo | check the handbook |
22:29.12 | *** join/#asterisk a1fa|64 (n=a1fa@24.144.51.82) |
22:29.14 | a1fa|64 | i am so 31337 |
22:29.21 | a1fa|64 | teach me how to haXXX :p |
22:29.31 | a1fa|64 | god man! some kids tried to prank call me |
22:29.34 | SugarGuest142 | could you tell me where to start? |
22:29.42 | a1fa|64 | i had a blast sanding them with my asterix pbx :P |
22:29.44 | fugitivo | SugarGuest142: http://www.digium.com/handbook-draft.pdf |
22:30.18 | warthawg | is there a handy guide to explain how to config my bt-101 so i can use it with asterisk to talk to my iax provider? |
22:30.55 | dorphalsig | SugarGuest142 --> You speak spanish¿ |
22:31.01 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
22:31.30 | mogorman | asterix?!?! asterisk..... |
22:31.39 | mogorman | the best place is |
22:31.42 | a1fa|64 | hahaha :P |
22:31.43 | mogorman | ~thebook |
22:31.45 | jbot | rumour has it, thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
22:31.46 | a1fa|64 | i need to setup a trap for them |
22:31.55 | a1fa|64 | so they get lost in the world of IVR menus |
22:32.13 | warthawg | yo no soy marinero |
22:32.15 | fugitivo | alfa|64: there's something on the wiki for telemarketers |
22:32.17 | aksis | hi, i have asterisk setup and running with an iax connect to fwd. I am failing to grasp how to setup the exts and the sip so that I can be reached at sip:aksis@idea-anvil.net and call out with kphone. Here is my config http://www.idea-anvil.net/~aksis/ |
22:32.25 | a1fa|64 | fugitivo: awesome |
22:32.25 | a1fa|64 | hehee |
22:33.01 | Qwell | mogorman: umm...I need help |
22:33.08 | Qwell | ...I forgot what I was going to emerge. :( |
22:33.16 | mogorman | emerge -e world |
22:33.23 | mogorman | will keep you busy for a bit |
22:33.25 | Qwell | k |
22:34.06 | mogorman | qwell go kill some bugs |
22:34.22 | aksis | I have no problem dialing from the asterisk cli to 500 and interacting with FWD... |
22:34.39 | Qwell | wow that's a big list of packages |
22:34.53 | aksis | I am not quite clear on the relation between my pbx and FWD... do I even need FWD? |
22:35.03 | mogorman | no |
22:35.17 | aksis | Mother: is that no to me? |
22:35.29 | aksis | errr mogor: is that no to me? |
22:35.29 | warthawg | there are several basic things i am not clear on, this telelphony business, doodle erlang erlang |
22:36.04 | a1fa|64 | omfg |
22:36.06 | a1fa|64 | this is awesome |
22:36.07 | a1fa|64 | :p |
22:36.19 | a1fa|64 | talk about prank calling somebody |
22:36.19 | a1fa|64 | dial-out |
22:36.20 | a1fa|64 | and force them to use the menu |
22:36.22 | a1fa|64 | this is awesome |
22:36.29 | mogorman | alfa?! |
22:37.09 | rob0 | haha |
22:37.48 | a1fa|64 | this sucks |
22:37.52 | a1fa|64 | i cant play the .gsm in winampo |
22:38.32 | mogorman | install sox |
22:38.39 | a1fa|64 | sox? |
22:38.53 | a1fa|64 | sox@sourceforge? |
22:39.10 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
22:39.17 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
22:39.48 | dorphalsig | there is a winamp gsm plugin |
22:39.50 | mogorman | i dont know where sox is stored |
22:40.32 | warthawg | in the top drawer, your shorts go in the drawer below |
22:41.45 | a1fa|64 | omg |
22:41.48 | a1fa|64 | this guy i sannoying |
22:41.59 | a1fa|64 | http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture |
22:43.03 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
22:43.49 | a1fa|64 | this is great |
22:43.54 | a1fa|64 | i am making those people very mad |
22:44.15 | mogorman | bye |
22:44.31 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
22:44.31 | [TK]D-Fender | a1fa|64 : So hows you're system working now? |
22:44.41 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
22:44.49 | a1fa|64 | its working great |
22:45.10 | rob0 | bye mogorman |
22:45.37 | [TK]D-Fender | Good to hear. |
22:46.13 | a1fa|64 | yeah man |
22:46.18 | a1fa|64 | i had some people prank call my cellphone |
22:46.27 | a1fa|64 | so i strike back : http://www.voip-info.org/wiki-Asterisk+Telemarketer+Torture |
22:50.44 | a1fa|64 | .aif is much better quality |
22:50.50 | a1fa|64 | i am guessing, astierisk supports aif |
22:53.50 | a1fa|64 | anybody? |
22:54.07 | a1fa|64 | are there any voice filters for asterisk? |
22:55.19 | Qwell | a1fa|64: There is a thirdparty voicechanger app |
22:55.55 | Qwell | http://www.lobstertech.com/voicechanger/ |
22:56.17 | a1fa|64 | i am there |
22:56.21 | Qwell | JunK-Y's patch is very cool. I'm not sure if they've integrated it yet though |
22:56.21 | a1fa|64 | have you tried this patch>? |
22:56.37 | Qwell | yep, works nicely |
22:56.59 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
22:57.23 | a1fa|64 | mind trying it out? |
22:57.31 | a1fa|64 | record a demo |
22:57.33 | a1fa|64 | or something |
22:58.23 | a1fa|64 | http://sky.prohosting.com/oparviai/soundtouch/soundstretch.html#examples |
22:58.36 | a1fa|64 | 404 |
23:01.14 | Mother | I managed to get chan_bluetooth working! |
23:01.23 | *** join/#asterisk litage (n=nick@203.220.55.70) |
23:01.37 | *** join/#asterisk dflow (i=pch@yennefer.sisco.pl) |
23:01.40 | Mother | pairing it with my BT headset and works like a charm |
23:02.25 | *** part/#asterisk dflow (i=pch@yennefer.sisco.pl) |
23:02.52 | [TK]D-Fender | Mother : How do you dial with it? |
23:05.09 | benjk | [TK]D-Fender: you don't |
23:05.28 | benjk | you use a desktop dialer |
23:06.27 | Mother | [TK]D-Fender: well - this headset (SouthWing SH305) can either initiate voice-dial with +BVRA=1 or dial a pre-stored number |
23:06.48 | Mother | it can store the last received call's CallerID and dial it with a long press of the Push4 button |
23:07.53 | Mother | other than that as benjk says, or using an exten to dial using the headset |
23:11.29 | zebedee1 | Hi all. I'm trying to get my new asterisk box to play nice behind a masquerading firewall, and am having a few problems with Asterisk reaching the outside world. |
23:12.22 | [TK]D-Fender | zebedee1 : You need to set either "EXTERNHOST" or "EXTERNIP". Than you need to define "LOCALNET". This is all in the [general] section of SIP.conf |
23:13.24 | a1fa|64 | they need to merge voicechanger with 1.2.2 tree |
23:13.25 | a1fa|64 | :p |
23:13.48 | zebedee1 | At present, I've set 'externalip' to 84.92.168.173 (my external ip), and 'localnet' to 192.168.1.0/255.255.255.0 |
23:13.57 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
23:14.37 | zebedee1 | ...but no joy. Registration to 3rdP voip services keep timing out |
23:14.39 | RaYmAn-Bx | zebedee1: "externalip" or "externip"? |
23:14.44 | [TK]D-Fender | zebedee1 : And you'll need to forward 5060,10000-20000 all UDP to your * box |
23:15.03 | [TK]D-Fender | Thats supposed to be EXTERNIP. |
23:15.59 | zebedee1 | Ah! currently externALip. I'll change it over, and give it a go. |
23:16.06 | [TK]D-Fender | :P |
23:18.28 | file | [TK]D-Fender: !!! |
23:18.30 | *** join/#asterisk gnosys (n=gnosys@ip68-9-201-250.ri.ri.cox.net) |
23:18.39 | [TK]D-Fender | file: ~~~ |
23:18.43 | [TK]D-Fender | er.... |
23:18.46 | [TK]D-Fender | file: !!! |
23:18.47 | file | yeekz, almost stained my nice new white shirt |
23:19.00 | zebedee1 | Nice one - Asterisk is dialing out! Thanks a lot. |
23:19.07 | [TK]D-Fender | np |
23:20.15 | *** join/#asterisk sigmounte_ (n=sigmount@www.sighq.net) |
23:20.28 | *** join/#asterisk mrtwister (n=andrius@cable-10-68.cgates.lt) |
23:21.39 | mrtwister | hello... loose all evening... i setup DMZ, asterisk under nat, but incoming SIP calls are one-way audio... i map 5060 and ports 10000 to 25000 too, and calls have to go :) how to set asterisk for NAT? |
23:24.57 | [TK]D-Fender | mrt : You need to set either "EXTERNHOST" or "EXTERNIP". Than you need to define "LOCALNET". This is all in the [general] section of SIP.conf |
23:26.02 | wunderkin | echo.. |
23:26.20 | aksis | when someone calles into my box sip:aksis@idea-anvil.net its going to come in on 5060 right? So I have that forwared to the asterisk server. From there how is asterisk treating it? is "aksis" going to be the ext? how do I map aksis@idea-anvil.net to kphone? can kphone and asterisk be on the same box? |
23:26.21 | [TK]D-Fender | I prefer the term "cut&paste" ;) |
23:26.47 | *** join/#asterisk Defraz (n=t0tal@103-16.69-92-cpe.cableone.net) |
23:27.14 | infinity1 | realistically, how are you suppose to dial an email address from a phone? |
23:27.19 | *** join/#asterisk chiardon (n=dorphals@200.106.223.5) |
23:27.37 | [TK]D-Fender | aksis : You'll need to have "exten => aksis,1,dosomethingoranotherhere" in your incoming context in defined in sip.conf's [general] section |
23:27.45 | aksis | Im setting this up more for ip phones. |
23:28.13 | [TK]D-Fender | infinity1 : well you wouldn't dial an *E-MAIL* address... |
23:28.59 | [TK]D-Fender | aksis : what does that imply? |
23:29.08 | infinity1 | err ...i guess dialing an email address is for dialing with software? |
23:29.17 | dorphalsig | hey |
23:29.20 | Qwell | You don't dial email addresses |
23:29.20 | SkramX | j #XORG |
23:29.22 | SkramX | oops |
23:29.26 | Qwell | You dial SIP addresses, or something |
23:29.32 | dorphalsig | is there much difference between asterisk 1.1 and 1.2? |
23:29.42 | Qwell | dorphalsig: yes |
23:29.44 | Qwell | 1.2 actually exists |
23:29.45 | aksis | I am under the impression that you can dial an ip phone with a domain name and a prefix? |
23:29.56 | infinity1 | whats the difference between an email address and a sip address? aren't they generally the same? |
23:30.01 | aksis | errr prefix=ext |
23:30.22 | iDunno | infinity1: erm, they 'look' the same, they're handled differently. |
23:30.28 | aksis | infinity1: the domain translates to an ip and the user translates to an ext... |
23:30.36 | [TK]D-Fender | aksis : Sort of. Only if the hostname IS the IP in question. |
23:30.39 | iDunno | infinity1: email tends to use an MX record... |
23:30.52 | iDunno | there's a different chunk of evil magic DNS for SIP |
23:30.58 | iDunno | (which I can't remember :) |
23:31.02 | infinity1 | iDunno: and sip uses SRV |
23:31.10 | iDunno | infinity1: that's the one :) |
23:31.26 | infinity1 | but its still looks like an email address. which i can't type in using a hardphone |
23:31.38 | iDunno | you've got a crap hardphone, then. |
23:31.42 | Qwell | infinity1: Get a good one |
23:31.51 | [TK]D-Fender | infinity1 : Depends on the hardphone. I can from mine (SPA-941), and many others suppotr it as well. |
23:31.52 | iDunno | certainly you can type then on the Snom190 |
23:31.53 | aksis | hmmm so aslong as the A == MX then its all good? |
23:32.18 | infinity1 | i have a polycom. it doesn't have a qwerty keyboard. i guess i could type it in somehow, but it doesn't seem practical |
23:32.20 | aksis | inf I am maping aksis to 1001 |
23:32.28 | [TK]D-Fender | aksis : if you're already using *, all SIP calls hit it be default anyways. |
23:32.35 | iDunno | infinity1: it's got characters on the numbers, surely... |
23:32.36 | aksis | so either way the extention can be dialed... |
23:32.43 | [TK]D-Fender | if your want it to dial your phone right away, jsut do that in the dial-plan |
23:32.44 | iDunno | infinity1: you've used a mobile, right? ;) |
23:33.00 | aksis | hehe... this is where I am getting lost... |
23:33.08 | infinity1 | iDunno: my point is, it takes much more effort dial dail characters than #s |
23:33.28 | iDunno | infinity1: so set up extensions in your * setup to dial sip addresses ;) |
23:33.31 | infinity1 | does anyone even dial a sip address? |
23:33.44 | [TK]D-Fender | aksis : just add "exten => asksis,1,Dial(SIP/myphone,20)" or whatever into the context you dfeined in sip.conf |
23:34.07 | aksis | exten => aksis,1,Goto(999) ; alias aksis to 999 |
23:34.08 | aksis | exten => 999,2,Dial(${CONSOLE}) ; this will dial ${AKSIS} |
23:34.10 | benjk | infinity: most modern phones support some form of phonebook |
23:34.26 | aksis | thats what I was going to use, but CONSOLE isn't correct is it? |
23:34.34 | benjk | you put the sip address in the phone book, then dial by selecting from the phone book |
23:34.35 | infinity1 | benjk: what does that mean for dialing a sip addR? |
23:34.38 | [TK]D-Fender | aksis : that won't work. The goto tries to jump to PRIORITY 999. |
23:34.50 | infinity1 | benjk: so its like speed dial. |
23:34.52 | infinity1 | ? |
23:34.56 | [TK]D-Fender | and your exten 999 doesn't have a "1" priority |
23:35.00 | benjk | like on a mobile phone |
23:35.24 | benjk | where you have a menu of some kind and you can scroll through all the entries you entered |
23:35.36 | infinity1 | does anyone here ever dial sip addresses? |
23:35.44 | benjk | then just press the dial button when you found the desired entry |
23:35.54 | benjk | not manually |
23:35.57 | [TK]D-Fender | infinity1 : I have. A pin-in-the-ass, but can be done. |
23:36.03 | [TK]D-Fender | pain* |
23:36.11 | benjk | ouch |
23:36.13 | infinity1 | there must be a better way |
23:36.24 | benjk | pin in the a$$ - sounds painful indeed |
23:36.33 | [TK]D-Fender | infinity1 : there is. Put it in your dial-plan |
23:36.49 | infinity1 | [TK]D-Fender: what do you mean? |
23:37.30 | [TK]D-Fender | infinity1 : exten => 666,1,Dial(SIP/satan@hell.com,20) |
23:37.38 | infinity1 | ahh ..i see. |
23:37.52 | infinity1 | thats not a bad idea. but it still requires more work. |
23:38.22 | mrtwister | what is it? asterisk under nat |
23:38.24 | mrtwister | --- |
23:38.24 | mrtwister | set_destination: Parsing <sip:202.59.90.178:1000> for address/port to send to |
23:38.24 | mrtwister | set_destination: set destination to 202.59.90.178, port 1000 |
23:38.24 | mrtwister | Reliably Transmitting (NAT) to 202.59.90.178:1000: |
23:38.24 | mrtwister | BYE sip:202.59.90.178:1000 SIP/2.0 |
23:38.43 | infinity1 | is there some kind of directory * can subscribe to which looks up a phone # and dials it via sip instead of using pots if possible? |
23:38.55 | benjk | yes, enum |
23:38.56 | *** join/#asterisk r0d3nt|m (i=nobody@wsip-24-234-241-145.lv.lv.cox.net) |
23:39.11 | infinity1 | benjk: are any companies using it? |
23:39.15 | benjk | E164.org, E164.arpa end a bunch of others |
23:39.29 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
23:39.45 | infinity1 | benjk: is there some kind of standard? |
23:39.54 | benjk | there are already about 70-80 million numbers in there or so I seem to remember |
23:40.02 | mrtwister | infinity1: abou enum - ipkall.com use 100% |
23:40.03 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
23:40.30 | infinity1 | mrtwister: does vonage support dialing their customers using SIP? |
23:40.35 | justinu | no |
23:40.36 | benjk | infinity: we have a howto for enum |
23:40.54 | mrtwister | infinity1: to enum? think, not |
23:40.55 | benjk | http://www.astmasters.net/howtos.html |
23:41.11 | benjk | scroll down a bit to the Enum article |
23:41.30 | infinity1 | benjk: cool. |
23:41.41 | infinity1 | looks like something i should probably already know :) |
23:45.29 | *** join/#asterisk Dovid (i=dovi5988@17.sub-70-193-186.myvzw.com) |
23:47.52 | infinity1 | has anyone implemented enumlookup with their standard dialing with success? or is it better to have the user manually do an enum dial? |
23:48.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
23:49.16 | Dovid | hello there shmaltz |
23:49.41 | shmaltz | hi Dovid |
23:49.57 | SkramX | oy veu! |
23:50.00 | SkramX | *Vey! |
23:50.02 | SkramX | lol |
23:50.08 | shmaltz | lol |
23:52.29 | *** join/#asterisk rappu_noxep (n=me@harry.org.ua) |
23:53.25 | aksis | [TK]D-Fender: thanks, I am able to ring my phone now... |
23:54.51 | troyb1 | hey guys i have a cisco 7940 SIP 7.3 and im having trouble dialing sip email addresses |
23:55.59 | Qwell | They aren't email addresses |
23:56.14 | troyb1 | sorry for my mis-statement :) |
23:56.22 | Qwell | we just went over this :P |
23:56.35 | troyb1 | is it in my best interest to scroll up? |
23:56.36 | *** join/#asterisk PJMattF (n=matt@cpc2-fare3-3-0-cust79.cos2.cable.ntl.com) |
23:56.52 | *** join/#asterisk Vijay (i=Vijay@203.122.28.109) |
23:56.54 | mrtwister | someone, please call me at 500@80.240.10.68 over sip, i tesing NAT... |
23:57.01 | *** join/#asterisk DrukenHME (n=druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com) |
23:57.12 | troyb1 | mrtwister i'd volunteer but i cant even get mine working *grin* |
23:57.27 | DrukenHME | volunteer for what ? |
23:57.29 | Vijay | hello mrtwister |
23:57.38 | mrtwister | hi |
23:57.42 | troyb1 | calling his sip address |
23:57.54 | PJMattF | does anyone have any suggestions for a good (i.e. fairly easy-to-use) linux distro for experimenting with Asterisk - I'm sick of the amount of obvious stuff that's absent from Ubuntu. |
23:57.58 | Vijay | i need to configure my asterisk run to run behind the nat |
23:57.59 | DrukenHME | my system will do it |
23:58.08 | troyb1 | PJMattF ubuntu |
23:58.09 | mrtwister | PJMattF: ubuntu is very good |
23:58.14 | Vijay | gentoo is the best i would say |
23:58.15 | troyb1 | mrtwister i beat ya too it |
23:58.17 | DrukenHME | goto www.abss.ca and enter in your sip URL into the callme box |
23:58.19 | PJMattF | it's missing tons of damn packages though |
23:58.33 | rob0 | what is missing from Ubuntu? |
23:58.33 | PJMattF | and i have no idea which ones are stopping infernal ztdummy from building |
23:58.34 | mrtwister | PJMattF: @home and rapid |
23:59.05 | Dovid | lol |
23:59.06 | PJMattF | well it may not be "missing" but there's assloads which isn't in the default ubuntu install |
23:59.09 | mrtwister | PJMattF: but for live system ubuntu server, debian server is very good.... gentoo too, but i not like to wast time to compile whole distro |
23:59.12 | rob0 | most of the "easy" distros do not install what you need to compile things |
23:59.14 | Dovid | I like CENT OS works great for me |
23:59.16 | DrukenHME | troyb1: who needs a SIP url called? |
23:59.33 | mrtwister | someone, please call me at 500@80.240.10.68 over sip, i tesing NAT... |
23:59.34 | mrtwister | me |
23:59.36 | mrtwister | :) |
23:59.51 | Qwell | mrtwister: <DrukenHME> goto www.abss.ca and enter in your sip URL into the callme box |